From 573e746914f14e069d8a2dca96ed32640e5796f9 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Per=20=C3=85hgren?= <peah@webrtc.org>
Date: Mon, 9 Feb 2026 14:17:39 +0100
Subject: [PATCH 001/472] Correct the switching between the coarse and refined
 filters
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This CL corrects the switching between the coarse and refined filters to
be done smoothly for all channels, and to be decided based on the _
content of all channels.


Bug: chromium:464314991
Change-Id: I5eb90848fbe135ec2c2ef9a5339920c6b7f56bbf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/447060
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46854}
---
 modules/audio_processing/aec3/echo_remover.cc | 95 +++++++++++--------
 1 file changed, 54 insertions(+), 41 deletions(-)

diff --git a/modules/audio_processing/aec3/echo_remover.cc b/modules/audio_processing/aec3/echo_remover.cc
index 1f0773780f..c720982193 100644
--- a/modules/audio_processing/aec3/echo_remover.cc
+++ b/modules/audio_processing/aec3/echo_remover.cc
@@ -108,6 +108,45 @@ void WindowedPaddedFft(const Aec3Fft& fft,
   std::copy(v.begin(), v.end(), v_old.begin());
 }
 
+// Detects whether the output of the refined filter is more appropriate to use
+// than the output of the coarse filter, returning the result as a bool.
+bool UseRefinedOutput(const SubtractorOutput& subtractor_output) {
+  // As the output of the refined adaptive filter generally should be better
+  // than the coarse filter output, add a margin and threshold for when
+  // choosing the coarse filter output.
+  if (subtractor_output.e2_coarse < 0.9f * subtractor_output.e2_refined &&
+      subtractor_output.y2 > 30.f * 30.f * kBlockSize &&
+      (subtractor_output.s2_refined > 60.f * 60.f * kBlockSize ||
+       subtractor_output.s2_coarse > 60.f * 60.f * kBlockSize)) {
+    return false;
+  }
+
+  // If the refined filter is diverged, choose the filter output that has
+  // the lowest power.
+  if (subtractor_output.e2_coarse < subtractor_output.e2_refined &&
+      subtractor_output.y2 < subtractor_output.e2_refined) {
+    return false;
+  }
+  return true;
+}
+
+// Forms the linear filter output by smoothly transition between the refined and
+// coarse filter outputs according to which of the filters is to be used.
+void FormLinearFilterOutput(bool refined_filter_output_last_selected,
+                            bool use_refined_output,
+                            const SubtractorOutput& subtractor_output,
+                            ArrayView<float> output) {
+  RTC_DCHECK_EQ(subtractor_output.e_refined.size(), output.size());
+  RTC_DCHECK_EQ(subtractor_output.e_coarse.size(), output.size());
+
+  SignalTransition(refined_filter_output_last_selected
+                       ? subtractor_output.e_refined
+                       : subtractor_output.e_coarse,
+                   use_refined_output ? subtractor_output.e_refined
+                                      : subtractor_output.e_coarse,
+                   output);
+}
+
 // Class for removing the echo from the capture signal.
 class EchoRemoverImpl final : public EchoRemover {
  public:
@@ -144,12 +183,6 @@ class EchoRemoverImpl final : public EchoRemover {
   }
 
  private:
-  // Selects which of the coarse and refined linear filter outputs that is most
-  // appropriate to pass to the suppressor and forms the linear filter output by
-  // smoothly transition between those.
-  void FormLinearFilterOutput(const SubtractorOutput& subtractor_output,
-                              ArrayView<float> output);
-
   static std::atomic<int> instance_count_;
   const EchoCanceller3Config config_;
   const Aec3Fft fft_;
@@ -385,14 +418,28 @@ void EchoRemoverImpl::ProcessCapture(
                       subtractor_output);
 
   // Compute spectra.
+  bool use_refined_output;
+  if (use_coarse_filter_output_) {
+    use_refined_output = false;
+    for (size_t ch = 0; ch < num_capture_channels_; ++ch) {
+      if (UseRefinedOutput(subtractor_output[ch])) {
+        use_refined_output = true;
+        break;
+      }
+    }
+  } else {
+    use_refined_output = true;
+  }
   for (size_t ch = 0; ch < num_capture_channels_; ++ch) {
-    FormLinearFilterOutput(subtractor_output[ch], e[ch]);
+    FormLinearFilterOutput(refined_filter_output_last_selected_,
+                           use_refined_output, subtractor_output[ch], e[ch]);
     WindowedPaddedFft(fft_, y->View(/*band=*/0, ch), y_old_[ch], &Y[ch]);
     WindowedPaddedFft(fft_, e[ch], e_old_[ch], &E[ch]);
     LinearEchoPower(E[ch], Y[ch], &S2_linear[ch]);
     Y[ch].Spectrum(optimization_, Y2[ch]);
     E[ch].Spectrum(optimization_, E2[ch]);
   }
+  refined_filter_output_last_selected_ = use_refined_output;
   const auto& nearend_spectrum = aec_state_.UsableLinearEstimate() ? E2 : Y2;
   // `y_old_` and `e_old_` now point to the current block. Though their channel
   // layout is already suitable for residual echo estimation, an alias is
@@ -494,40 +541,6 @@ void EchoRemoverImpl::ProcessCapture(
                         aec_state_.SaturatedCapture() ? 1 : 0);
 }
 
-void EchoRemoverImpl::FormLinearFilterOutput(
-    const SubtractorOutput& subtractor_output,
-    ArrayView<float> output) {
-  RTC_DCHECK_EQ(subtractor_output.e_refined.size(), output.size());
-  RTC_DCHECK_EQ(subtractor_output.e_coarse.size(), output.size());
-  bool use_refined_output = true;
-  if (use_coarse_filter_output_) {
-    // As the output of the refined adaptive filter generally should be better
-    // than the coarse filter output, add a margin and threshold for when
-    // choosing the coarse filter output.
-    if (subtractor_output.e2_coarse < 0.9f * subtractor_output.e2_refined &&
-        subtractor_output.y2 > 30.f * 30.f * kBlockSize &&
-        (subtractor_output.s2_refined > 60.f * 60.f * kBlockSize ||
-         subtractor_output.s2_coarse > 60.f * 60.f * kBlockSize)) {
-      use_refined_output = false;
-    } else {
-      // If the refined filter is diverged, choose the filter output that has
-      // the lowest power.
-      if (subtractor_output.e2_coarse < subtractor_output.e2_refined &&
-          subtractor_output.y2 < subtractor_output.e2_refined) {
-        use_refined_output = false;
-      }
-    }
-  }
-
-  SignalTransition(refined_filter_output_last_selected_
-                       ? subtractor_output.e_refined
-                       : subtractor_output.e_coarse,
-                   use_refined_output ? subtractor_output.e_refined
-                                      : subtractor_output.e_coarse,
-                   output);
-  refined_filter_output_last_selected_ = use_refined_output;
-}
-
 }  // namespace
 
 std::unique_ptr<EchoRemover> EchoRemover::Create(

From ffe79876ecad93bdf0c14d69ef9e7bd087a8fc09 Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Mon, 9 Feb 2026 20:08:39 -0800
Subject: [PATCH 002/472] Update WebRTC code version (2026-02-10T04:08:07).

Bug: None
Change-Id: I202f8ab3448cea9478227b187446ec670382d2b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/447940
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#46855}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index 0776b93707..2d297f3f41 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-02-09T04:06:40";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-02-10T04:08:07";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From 7c388cbabb1ba422c626addc37e40765de200169 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Per=20=C3=85hgren?= <peah@webrtc.org>
Date: Tue, 10 Feb 2026 10:33:22 +0100
Subject: [PATCH 003/472] Changing default values of multi channel processing
 to be true
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This CL changes the default values of multi channel processing to be true. The CL will be followed by another CL that removes these parameters once their use in Chromium has been removed.

The reason for their removal is that their initial purpose, which was allowing a fallback to single-channel processing until the multi-channel processing functionality was verified is no longer needed.

The CL also includes an update of the unittests where the tests was including an expectation on a code-independent test threshold value. That change is needed as the threshold took effect as part of this code-change.
A follow-up CL will be made to refactor that test.


Bug: chromium:464314991
Change-Id: Ie4141cd5b13c6b748f1ff1552b526c15155ac279
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/429343
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46856}
---
 api/audio/audio_processing.h                              | 6 +++---
 resources/audio_processing/output_data_float.pb.sha1      | 2 +-
 resources/audio_processing/output_data_float_avx2.pb.sha1 | 2 +-
 3 files changed, 5 insertions(+), 5 deletions(-)

diff --git a/api/audio/audio_processing.h b/api/audio/audio_processing.h
index a83078ae9b..1e0ea0beee 100644
--- a/api/audio/audio_processing.h
+++ b/api/audio/audio_processing.h
@@ -147,10 +147,10 @@ class RTC_EXPORT AudioProcessing : public RefCountInterface {
       // 32000 or 48000 and any differing values will be treated as 48000.
       int maximum_internal_processing_rate = 32000;
       // Allow multi-channel processing of render audio.
-      bool multi_channel_render = false;
+      bool multi_channel_render = true;
       // Allow multi-channel processing of capture audio when AEC3 is active
-      // or a custom AEC is injected..
-      bool multi_channel_capture = false;
+      // or a custom AEC is injected.
+      bool multi_channel_capture = true;
       // Indicates how to downmix multi-channel capture audio to mono (when
       // needed).
       DownmixMethod capture_downmix_method = DownmixMethod::kAverageChannels;
diff --git a/resources/audio_processing/output_data_float.pb.sha1 b/resources/audio_processing/output_data_float.pb.sha1
index 7dd76325f3..7035368d68 100644
--- a/resources/audio_processing/output_data_float.pb.sha1
+++ b/resources/audio_processing/output_data_float.pb.sha1
@@ -1 +1 @@
-d80019464496fec3a149fd36809769e84a9ea5ce
\ No newline at end of file
+c5f9b0b9c75f89e34e8f58cfb6e37e8a46271072
\ No newline at end of file
diff --git a/resources/audio_processing/output_data_float_avx2.pb.sha1 b/resources/audio_processing/output_data_float_avx2.pb.sha1
index ddca9839e7..6859932988 100644
--- a/resources/audio_processing/output_data_float_avx2.pb.sha1
+++ b/resources/audio_processing/output_data_float_avx2.pb.sha1
@@ -1 +1 @@
-8751057d01ad4d58c98781cd7236a65ece699706
\ No newline at end of file
+1623a92acd619a7ad2e599d7859463227179c1d4
\ No newline at end of file

From d01ecdaa05f48ed6095ed80d07142c7c76303055 Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Mon, 9 Feb 2026 15:59:55 +0100
Subject: [PATCH 004/472] Replace ArrayView::subview with subspan in logging/
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Fix ExtendLoggedBatch function to produce expected resolut when
`!output.empty()`. In particular avoid calling subspan function and
discard result as that would be a compilation error with std::span.

Bug: webrtc:439801349
Change-Id: I4a8e372417c974c796ec4eb5e1d406b13b59fa4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/447061
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46857}
---
 .../rtc_event_log/dependency_descriptor_encoder_decoder.cc    | 4 ++--
 logging/rtc_event_log/events/rtc_event_field_encoding.cc      | 2 +-
 .../rtc_event_log/events/rtc_event_field_encoding_parser.h    | 4 ++--
 3 files changed, 5 insertions(+), 5 deletions(-)

diff --git a/logging/rtc_event_log/dependency_descriptor_encoder_decoder.cc b/logging/rtc_event_log/dependency_descriptor_encoder_decoder.cc
index eaa75331c8..ca947ddfb0 100644
--- a/logging/rtc_event_log/dependency_descriptor_encoder_decoder.cc
+++ b/logging/rtc_event_log/dependency_descriptor_encoder_decoder.cc
@@ -45,7 +45,7 @@ RtcEventLogDependencyDescriptorEncoderDecoder::Encode(
   rtclog2::DependencyDescriptorsWireInfo res;
   const ArrayView<const uint8_t>& base_dd = raw_dd_data[0];
   auto delta_dds =
-      MakeArrayView(raw_dd_data.data(), raw_dd_data.size()).subview(1);
+      MakeArrayView(raw_dd_data.data(), raw_dd_data.size()).subspan(1);
 
   // Start and end bit.
   {
@@ -117,7 +117,7 @@ RtcEventLogDependencyDescriptorEncoderDecoder::Encode(
     std::vector<std::optional<std::string>> values(raw_dd_data.size());
     for (size_t i = 0; i < raw_dd_data.size(); ++i) {
       if (raw_dd_data[i].size() > 3) {
-        auto extended_info = raw_dd_data[i].subview(3);
+        auto extended_info = raw_dd_data[i].subspan(3);
         values[i] = {reinterpret_cast<const char*>(extended_info.data()),
                      extended_info.size()};
       }
diff --git a/logging/rtc_event_log/events/rtc_event_field_encoding.cc b/logging/rtc_event_log/events/rtc_event_field_encoding.cc
index e5b931d7e1..e43526a411 100644
--- a/logging/rtc_event_log/events/rtc_event_field_encoding.cc
+++ b/logging/rtc_event_log/events/rtc_event_field_encoding.cc
@@ -214,7 +214,7 @@ void EventEncoder::EncodeField(const FieldParameters& params,
   // Compute delta parameters
   ArrayView<const uint64_t> all_values(values);
   uint64_t base = values[0];
-  ArrayView<const uint64_t> remaining_values(all_values.subview(1));
+  ArrayView<const uint64_t> remaining_values = all_values.subspan(1);
 
   FixedLengthEncodingParametersV3 delta_params =
       FixedLengthEncodingParametersV3::CalculateParameters(
diff --git a/logging/rtc_event_log/events/rtc_event_field_encoding_parser.h b/logging/rtc_event_log/events/rtc_event_field_encoding_parser.h
index d46c6ac5ac..2de51895a3 100644
--- a/logging/rtc_event_log/events/rtc_event_field_encoding_parser.h
+++ b/logging/rtc_event_log/events/rtc_event_field_encoding_parser.h
@@ -212,9 +212,9 @@ PopulateRtcEventTimestamp(const ArrayView<uint64_t>& values,
 template <typename E>
 ArrayView<E> ExtendLoggedBatch(std::vector<E>& output, size_t new_elements) {
   size_t old_size = output.size();
-  output.insert(output.end(), old_size + new_elements, E());
+  output.resize(old_size + new_elements);
   ArrayView<E> output_batch = output;
-  output_batch.subview(old_size);
+  output_batch = output_batch.subspan(old_size);
   RTC_DCHECK_EQ(output_batch.size(), new_elements);
   return output_batch;
 }

From a27d608d4b063553c4c7fe9e1a7a8810a6855d6c Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Tue, 10 Feb 2026 08:30:02 +0000
Subject: [PATCH 005/472] Spanify rtc_base/buffer

This makes it possible to compile rtc_base/buffer with the checks
for unsafe buffers.

Bug: webrtc:478086887
Change-Id: I53f1161164e0408f985bd8395a0fd516e47bb0cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/444780
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46858}
---
 .../source/rtp_format_vp8_test_helper.cc      |  15 +-
 rtc_base/BUILD.gn                             |   1 +
 rtc_base/buffer.h                             |  40 ++++--
 rtc_base/buffer_unittest.cc                   | 135 +++++++++---------
 4 files changed, 105 insertions(+), 86 deletions(-)

diff --git a/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.cc b/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.cc
index ef54c0abe4..30cc3b7c7a 100644
--- a/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.cc
+++ b/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.cc
@@ -12,6 +12,7 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <iterator>
 
 #include "api/array_view.h"
 #include "modules/rtp_rtcp/source/rtp_format_vp8.h"
@@ -70,7 +71,7 @@ void RtpFormatVp8TestHelper::GetAllPacketsAndCheck(
     RtpPacketizerVp8* packetizer,
     ArrayView<const size_t> expected_sizes) {
   EXPECT_EQ(packetizer->NumPackets(), expected_sizes.size());
-  const uint8_t* data_ptr = payload_.begin();
+  Buffer::const_iterator data_it = payload_.begin();
   RtpPacketToSend packet(kNoExtensions);
   for (size_t i = 0; i < expected_sizes.size(); ++i) {
     EXPECT_TRUE(packetizer->NextPacket(&packet));
@@ -79,13 +80,15 @@ void RtpFormatVp8TestHelper::GetAllPacketsAndCheck(
 
     int payload_offset = CheckHeader(rtp_payload, /*first=*/i == 0);
     // Verify that the payload (i.e., after the headers) of the packet is
-    // identical to the expected (as found in data_ptr).
+    // identical to the expected (as found in data_it).
     auto vp8_payload = rtp_payload.subspan(payload_offset);
-    ASSERT_GE(payload_.end() - data_ptr, static_cast<int>(vp8_payload.size()));
-    EXPECT_THAT(vp8_payload, ElementsAreArray(data_ptr, vp8_payload.size()));
-    data_ptr += vp8_payload.size();
+    ASSERT_GE(std::distance(data_it, payload_.cend()),
+              static_cast<ptrdiff_t>(vp8_payload.size()));
+    EXPECT_THAT(vp8_payload,
+                ElementsAreArray(data_it, data_it + vp8_payload.size()));
+    data_it += vp8_payload.size();
   }
-  EXPECT_EQ(payload_.end() - data_ptr, 0);
+  EXPECT_EQ(payload_.end(), data_it);
 }
 
 int RtpFormatVp8TestHelper::CheckHeader(ArrayView<const uint8_t> buffer,
diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
index 309bc31fe7..3a253296b2 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -81,6 +81,7 @@ rtc_source_set("buffer") {
     ":type_traits",
     ":zero_memory",
     "../api:array_view",
+    "//third_party/abseil-cpp/absl/algorithm:container",
     "//third_party/abseil-cpp/absl/base:core_headers",
     "//third_party/abseil-cpp/absl/strings:string_view",
   ]
diff --git a/rtc_base/buffer.h b/rtc_base/buffer.h
index 5d9d8caabd..322cf2536a 100644
--- a/rtc_base/buffer.h
+++ b/rtc_base/buffer.h
@@ -16,9 +16,11 @@
 #include <algorithm>
 #include <cstring>
 #include <memory>
+#include <span>
 #include <type_traits>
 #include <utility>
 
+#include "absl/algorithm/container.h"
 #include "absl/base/attributes.h"
 #include "absl/strings/string_view.h"
 #include "api/array_view.h"
@@ -66,7 +68,8 @@ class BufferT {
 
  public:
   using value_type = T;
-  using const_iterator = const T*;
+  using iterator = std::span<T>::iterator;
+  using const_iterator = std::span<const T>::iterator;
 
   // An empty BufferT.
   BufferT() : size_(0), capacity_(0), data_(nullptr) {
@@ -121,6 +124,12 @@ class BufferT {
                 internal::BufferCompat<T, U>::value>::type* = nullptr>
   BufferT(U (&array)[N]) : BufferT(array, N) {}
 
+  // Construct a buffer from any type with a data() and size() member.
+  template <typename W,
+            typename std::enable_if<
+                HasDataAndSize<const W, const T>::value>::type* = nullptr>
+  explicit BufferT(const W& w) : BufferT(w.data(), w.size()) {}
+
   ~BufferT() { MaybeZeroCompleteBuffer(); }
 
   // Implicit conversion to absl::string_view if T is compatible with char.
@@ -205,20 +214,20 @@ class BufferT {
 
   T& operator[](size_t index) {
     RTC_DCHECK_LT(index, size_);
-    return data()[index];
+    return ArrayView<T>(*this)[index];
   }
 
   T operator[](size_t index) const {
     RTC_DCHECK_LT(index, size_);
-    return data()[index];
+    return ArrayView<const T>(*this)[index];
   }
 
-  T* begin() { return data(); }
-  T* end() { return data() + size(); }
-  const T* begin() const { return data(); }
-  const T* end() const { return data() + size(); }
-  const T* cbegin() const { return data(); }
-  const T* cend() const { return data() + size(); }
+  iterator begin() { return std::span(*this).begin(); }
+  iterator end() { return std::span(*this).end(); }
+  const_iterator begin() const { return std::span(*this).begin(); }
+  const_iterator end() const { return std::span(*this).end(); }
+  const_iterator cbegin() const { return begin(); }
+  const_iterator cend() const { return end(); }
 
   // The SetData functions replace the contents of the buffer. They accept the
   // same input types as the constructors.
@@ -288,7 +297,10 @@ class BufferT {
     const size_t new_size = size_ + size;
     EnsureCapacityWithHeadroom(new_size, true);
     static_assert(sizeof(T) == sizeof(U), "");
-    std::memcpy(data_.get() + size_, data, size * sizeof(U));
+    ArrayView<const U> source(data, size);
+    ArrayView<T> destination =
+        ArrayView<T>(data_.get(), capacity_).subview(size_, size);
+    absl::c_copy(source, destination.begin());
     size_ = new_size;
     RTC_DCHECK(IsConsistent());
   }
@@ -332,8 +344,8 @@ class BufferT {
     RTC_DCHECK(IsConsistent());
     const size_t old_size = size_;
     SetSize(old_size + max_elements);
-    U* base_ptr = data<U>() + old_size;
-    size_t written_elements = setter(ArrayView<U>(base_ptr, max_elements));
+    size_t written_elements =
+        setter(ArrayView<U>(data<U>(), size()).subspan(old_size));
 
     RTC_CHECK_LE(written_elements, max_elements);
     size_ = old_size + written_elements;
@@ -410,7 +422,7 @@ class BufferT {
       // It would be sufficient to only zero "size_" elements, as all other
       // methods already ensure that the unused capacity contains no sensitive
       // data---but better safe than sorry.
-      ExplicitZeroMemory(data_.get(), capacity_ * sizeof(T));
+      ExplicitZeroMemory(ArrayView<T>(data_.get(), capacity_));
     }
   }
 
@@ -418,7 +430,7 @@ class BufferT {
   void ZeroTrailingData(size_t count) {
     RTC_DCHECK(IsConsistent());
     RTC_DCHECK_LE(count, capacity_ - size_);
-    ExplicitZeroMemory(data_.get() + size_, count * sizeof(T));
+    ExplicitZeroMemory(MakeArrayView(data(), capacity_).subview(size_));
   }
 
   // Precondition for all methods except Clear, operator= and the destructor.
diff --git a/rtc_base/buffer_unittest.cc b/rtc_base/buffer_unittest.cc
index 88ebd0dd6a..064320f96d 100644
--- a/rtc_base/buffer_unittest.cc
+++ b/rtc_base/buffer_unittest.cc
@@ -10,6 +10,7 @@
 
 #include "rtc_base/buffer.h"
 
+#include <array>
 #include <cstddef>
 #include <cstdint>
 #include <cstring>
@@ -25,11 +26,18 @@ namespace webrtc {
 
 namespace {
 
+using ::testing::Each;
 using ::testing::ElementsAre;
 using ::testing::ElementsAreArray;
+using ::testing::Eq;
 
-constexpr uint8_t kTestData[] = {0x0, 0x1, 0x2, 0x3, 0x4, 0x5, 0x6, 0x7,
-                                 0x8, 0x9, 0xa, 0xb, 0xc, 0xd, 0xe, 0xf};
+auto kTestData =
+    std::to_array<uint8_t>({0x0, 0x1, 0x2, 0x3, 0x4, 0x5, 0x6, 0x7, 0x8, 0x9,
+                            0xa, 0xb, 0xc, 0xd, 0xe, 0xf});
+
+uint8_t* TestDataAtIndex(int index) {
+  return ArrayView<uint8_t>(kTestData).subview(index).data();
+}
 
 void TestBuf(const Buffer& b1, size_t size, size_t capacity) {
   EXPECT_EQ(b1.size(), size);
@@ -47,24 +55,24 @@ TEST(BufferTest, TestConstructEmpty) {
   // that to be considered a null pointer, which makes the call ambiguous.
   TestBuf(Buffer(0 + 0, 10), 0, 10);
 
-  TestBuf(Buffer(kTestData, 0), 0, 0);
-  TestBuf(Buffer(kTestData, 0, 20), 0, 20);
+  TestBuf(Buffer(kTestData.data(), 0), 0, 0);
+  TestBuf(Buffer(kTestData.data(), 0, 20), 0, 20);
 }
 
 TEST(BufferTest, TestConstructData) {
-  Buffer buf(kTestData, 7);
+  Buffer buf(kTestData.data(), 7);
   EXPECT_EQ(buf.size(), 7u);
   EXPECT_EQ(buf.capacity(), 7u);
   EXPECT_FALSE(buf.empty());
-  EXPECT_EQ(0, memcmp(buf.data(), kTestData, 7));
+  EXPECT_EQ(0, memcmp(buf.data(), kTestData.data(), 7));
 }
 
 TEST(BufferTest, TestConstructDataWithCapacity) {
-  Buffer buf(kTestData, 7, 14);
+  Buffer buf(kTestData.data(), 7, 14);
   EXPECT_EQ(buf.size(), 7u);
   EXPECT_EQ(buf.capacity(), 14u);
   EXPECT_FALSE(buf.empty());
-  EXPECT_EQ(0, memcmp(buf.data(), kTestData, 7));
+  EXPECT_EQ(0, memcmp(buf.data(), kTestData.data(), 7));
 }
 
 TEST(BufferTest, TestConstructArray) {
@@ -72,33 +80,33 @@ TEST(BufferTest, TestConstructArray) {
   EXPECT_EQ(buf.size(), 16u);
   EXPECT_EQ(buf.capacity(), 16u);
   EXPECT_FALSE(buf.empty());
-  EXPECT_EQ(0, memcmp(buf.data(), kTestData, 16));
+  EXPECT_EQ(0, memcmp(buf.data(), kTestData.data(), 16));
 }
 
 TEST(BufferTest, TestStringViewConversion) {
   Buffer buf(kTestData);
   absl::string_view view = buf;
-  EXPECT_EQ(view,
-            absl::string_view(reinterpret_cast<const char*>(kTestData), 16u));
+  EXPECT_EQ(view, absl::string_view(
+                      reinterpret_cast<const char*>(kTestData.data()), 16u));
 }
 
 TEST(BufferTest, TestSetData) {
-  Buffer buf(kTestData + 4, 7);
-  buf.SetData(kTestData, 9);
+  Buffer buf(TestDataAtIndex(4), 7);
+  buf.SetData(kTestData.data(), 9);
   EXPECT_EQ(buf.size(), 9u);
   EXPECT_EQ(buf.capacity(), 7u * 3 / 2);
   EXPECT_FALSE(buf.empty());
-  EXPECT_EQ(0, memcmp(buf.data(), kTestData, 9));
+  EXPECT_EQ(0, memcmp(buf.data(), kTestData.data(), 9));
   Buffer buf2;
   buf2.SetData(buf);
   EXPECT_EQ(buf.size(), 9u);
   EXPECT_EQ(buf.capacity(), 7u * 3 / 2);
-  EXPECT_EQ(0, memcmp(buf.data(), kTestData, 9));
+  EXPECT_EQ(0, memcmp(buf.data(), kTestData.data(), 9));
 }
 
 TEST(BufferTest, TestAppendData) {
-  Buffer buf(kTestData + 4, 3);
-  buf.AppendData(kTestData + 10, 2);
+  Buffer buf(TestDataAtIndex(4), 3);
+  buf.AppendData(TestDataAtIndex(10), 2);
   const int8_t exp[] = {0x4, 0x5, 0x6, 0xa, 0xb};
   EXPECT_EQ(buf, Buffer(exp));
   Buffer buf2;
@@ -111,33 +119,34 @@ TEST(BufferTest, TestAppendData) {
 TEST(BufferTest, TestSetAndAppendWithUnknownArg) {
   struct TestDataContainer {
     size_t size() const { return 3; }
-    const uint8_t* data() const { return kTestData; }
+    const uint8_t* data() const { return kTestData.data(); }
   };
   Buffer buf;
   buf.SetData(TestDataContainer());
   EXPECT_EQ(3u, buf.size());
-  EXPECT_EQ(Buffer(kTestData, 3), buf);
+  EXPECT_EQ(Buffer(kTestData.data(), 3), buf);
   EXPECT_THAT(buf, ElementsAre(0, 1, 2));
   buf.AppendData(TestDataContainer());
   EXPECT_EQ(6u, buf.size());
-  EXPECT_EQ(0, memcmp(buf.data(), kTestData, 3));
-  EXPECT_EQ(0, memcmp(buf.data() + 3, kTestData, 3));
+  EXPECT_EQ(0, memcmp(buf.data(), kTestData.data(), 3));
+  EXPECT_EQ(0, memcmp(ArrayView<uint8_t>(buf).subview(3).data(),
+                      kTestData.data(), 3));
   EXPECT_THAT(buf, ElementsAre(0, 1, 2, 0, 1, 2));
 }
 
 TEST(BufferTest, TestSetSizeSmaller) {
   Buffer buf;
-  buf.SetData(kTestData, 15);
+  buf.SetData(kTestData.data(), 15);
   buf.SetSize(10);
   EXPECT_EQ(buf.size(), 10u);
   EXPECT_EQ(buf.capacity(), 15u);  // Hasn't shrunk.
   EXPECT_FALSE(buf.empty());
-  EXPECT_EQ(buf, Buffer(kTestData, 10));
+  EXPECT_EQ(buf, Buffer(kTestData.data(), 10));
 }
 
 TEST(BufferTest, TestSetSizeLarger) {
   Buffer buf;
-  buf.SetData(kTestData, 15);
+  buf.SetData(kTestData.data(), 15);
   EXPECT_EQ(buf.size(), 15u);
   EXPECT_EQ(buf.capacity(), 15u);
   EXPECT_FALSE(buf.empty());
@@ -145,7 +154,7 @@ TEST(BufferTest, TestSetSizeLarger) {
   EXPECT_EQ(buf.size(), 20u);
   EXPECT_EQ(buf.capacity(), 15u * 3 / 2);  // Has grown.
   EXPECT_FALSE(buf.empty());
-  EXPECT_EQ(0, memcmp(buf.data(), kTestData, 15));
+  EXPECT_EQ(0, memcmp(buf.data(), kTestData.data(), 15));
 }
 
 TEST(BufferTest, TestEnsureCapacitySmaller) {
@@ -159,18 +168,18 @@ TEST(BufferTest, TestEnsureCapacitySmaller) {
 }
 
 TEST(BufferTest, TestEnsureCapacityLarger) {
-  Buffer buf(kTestData, 5);
+  Buffer buf(kTestData.data(), 5);
   buf.EnsureCapacity(10);
   const int8_t* data = buf.data<int8_t>();
   EXPECT_EQ(buf.capacity(), 10u);
-  buf.AppendData(kTestData + 5, 5);
+  buf.AppendData(TestDataAtIndex(5), 5);
   EXPECT_EQ(buf.data<int8_t>(), data);  // No reallocation.
   EXPECT_FALSE(buf.empty());
-  EXPECT_EQ(buf, Buffer(kTestData, 10));
+  EXPECT_EQ(buf, Buffer(kTestData.data(), 10));
 }
 
 TEST(BufferTest, TestMoveConstruct) {
-  Buffer buf1(kTestData, 3, 40);
+  Buffer buf1(kTestData.data(), 3, 40);
   const uint8_t* data = buf1.data();
   Buffer buf2(std::move(buf1));
   EXPECT_EQ(buf2.size(), 3u);
@@ -185,7 +194,7 @@ TEST(BufferTest, TestMoveConstruct) {
 }
 
 TEST(BufferTest, TestMoveAssign) {
-  Buffer buf1(kTestData, 3, 40);
+  Buffer buf1(kTestData.data(), 3, 40);
   const uint8_t* data = buf1.data();
   Buffer buf2(kTestData);
   buf2 = std::move(buf1);
@@ -206,14 +215,14 @@ TEST(BufferTest, TestMoveAssignSelf) {
   // state could be caught by the DCHECKs and/or by the leak checker.) We need
   // to be sneaky when testing this; if we're doing a too-obvious
   // move-assign-to-self, clang's -Wself-move triggers at compile time.
-  Buffer buf(kTestData, 3, 40);
+  Buffer buf(kTestData.data(), 3, 40);
   Buffer* buf_ptr = &buf;
   buf = std::move(*buf_ptr);
 }
 
 TEST(BufferTest, TestSwap) {
-  Buffer buf1(kTestData, 3);
-  Buffer buf2(kTestData, 6, 40);
+  Buffer buf1(kTestData.data(), 3);
+  Buffer buf2(kTestData.data(), 6, 40);
   uint8_t* data1 = buf1.data();
   uint8_t* data2 = buf2.data();
   using std::swap;
@@ -230,7 +239,7 @@ TEST(BufferTest, TestSwap) {
 
 TEST(BufferTest, TestClear) {
   Buffer buf;
-  buf.SetData(kTestData, 15);
+  buf.SetData(kTestData.data(), 15);
   EXPECT_EQ(buf.size(), 15u);
   EXPECT_EQ(buf.capacity(), 15u);
   EXPECT_FALSE(buf.empty());
@@ -250,8 +259,8 @@ TEST(BufferTest, TestLambdaSetAppend) {
   };
 
   Buffer buf1;
-  buf1.SetData(kTestData, 15);
-  buf1.AppendData(kTestData, 15);
+  buf1.SetData(kTestData.data(), 15);
+  buf1.AppendData(kTestData.data(), 15);
 
   Buffer buf2;
   EXPECT_EQ(buf2.SetData(15, setter), 15u);
@@ -270,8 +279,8 @@ TEST(BufferTest, TestLambdaSetAppendSigned) {
   };
 
   Buffer buf1;
-  buf1.SetData(kTestData, 15);
-  buf1.AppendData(kTestData, 15);
+  buf1.SetData(kTestData.data(), 15);
+  buf1.AppendData(kTestData.data(), 15);
 
   Buffer buf2;
   EXPECT_EQ(buf2.SetData<int8_t>(15, setter), 15u);
@@ -290,7 +299,7 @@ TEST(BufferTest, TestLambdaAppendEmpty) {
   };
 
   Buffer buf1;
-  buf1.SetData(kTestData, 15);
+  buf1.SetData(kTestData.data(), 15);
 
   Buffer buf2;
   EXPECT_EQ(buf2.AppendData(15, setter), 15u);
@@ -336,12 +345,12 @@ TEST(BufferTest, TestMutableLambdaSetAppend) {
   EXPECT_FALSE(buf.empty());
 
   for (uint8_t i = 0; i != buf.size(); ++i) {
-    EXPECT_EQ(buf.data()[i], magic_number + i);
+    EXPECT_EQ(buf[i], magic_number + i);
   }
 }
 
 TEST(BufferTest, TestBracketRead) {
-  Buffer buf(kTestData, 7);
+  Buffer buf(kTestData.data(), 7);
   EXPECT_EQ(buf.size(), 7u);
   EXPECT_EQ(buf.capacity(), 7u);
   EXPECT_NE(buf.data(), nullptr);
@@ -353,7 +362,7 @@ TEST(BufferTest, TestBracketRead) {
 }
 
 TEST(BufferTest, TestBracketReadConst) {
-  Buffer buf(kTestData, 7);
+  Buffer buf(kTestData.data(), 7);
   EXPECT_EQ(buf.size(), 7u);
   EXPECT_EQ(buf.capacity(), 7u);
   EXPECT_NE(buf.data(), nullptr);
@@ -377,20 +386,21 @@ TEST(BufferTest, TestBracketWrite) {
     buf[i] = kTestData[i];
   }
 
-  EXPECT_THAT(buf, ElementsAreArray(kTestData, 7));
+  EXPECT_THAT(buf, ElementsAreArray(kTestData.data(), 7));
 }
 
 TEST(BufferTest, TestBeginEnd) {
   const Buffer cbuf(kTestData);
   Buffer buf(kTestData);
-  auto* b1 = cbuf.begin();
+  auto b1 = cbuf.begin();
   for (auto& x : buf) {
     EXPECT_EQ(*b1, x);
+    ASSERT_NE(b1, cbuf.end());
     ++b1;
     ++x;
   }
   EXPECT_EQ(cbuf.end(), b1);
-  auto* b2 = buf.begin();
+  auto b2 = buf.begin();
   for (auto& y : cbuf) {
     EXPECT_EQ(*b2, y + 1);
     ++b2;
@@ -471,12 +481,13 @@ TEST(BufferDeathTest, DieOnUseAfterMove) {
 }
 
 TEST(ZeroOnFreeBufferTest, TestZeroOnSetData) {
-  ZeroOnFreeBuffer<uint8_t> buf(kTestData, 7);
+  ZeroOnFreeBuffer<uint8_t> buf(kTestData.data(), 7);
   const uint8_t* old_data = buf.data();
   const size_t old_capacity = buf.capacity();
-  const size_t old_size = buf.size();
   constexpr size_t offset = 1;
-  buf.SetData(kTestData + offset, 2);
+  // Pointer to the last five bytes of the underlying buffer.
+  auto to_be_zeroed = ArrayView<const uint8_t>(buf).subspan(2);
+  buf.SetData(TestDataAtIndex(offset), 2);
   // Sanity checks to make sure the underlying heap memory was not reallocated.
   EXPECT_EQ(old_data, buf.data());
   EXPECT_EQ(old_capacity, buf.capacity());
@@ -484,9 +495,7 @@ TEST(ZeroOnFreeBufferTest, TestZeroOnSetData) {
   // been zeroed.
   EXPECT_EQ(kTestData[offset], buf[0]);
   EXPECT_EQ(kTestData[offset + 1], buf[1]);
-  for (size_t i = 2; i < old_size; i++) {
-    EXPECT_EQ(0, old_data[i]);
-  }
+  EXPECT_THAT(to_be_zeroed, Each(Eq(0)));
 }
 
 TEST(ZeroOnFreeBufferTest, TestZeroOnSetDataFromSetter) {
@@ -497,10 +506,10 @@ TEST(ZeroOnFreeBufferTest, TestZeroOnSetDataFromSetter) {
     return 2;
   };
 
-  ZeroOnFreeBuffer<uint8_t> buf(kTestData, 7);
+  ZeroOnFreeBuffer<uint8_t> buf(kTestData.data(), 7);
   const uint8_t* old_data = buf.data();
+  auto to_be_zeroed = ArrayView<const uint8_t>(buf).subspan(2);
   const size_t old_capacity = buf.capacity();
-  const size_t old_size = buf.size();
   buf.SetData(2, setter);
   // Sanity checks to make sure the underlying heap memory was not reallocated.
   EXPECT_EQ(old_data, buf.data());
@@ -509,16 +518,14 @@ TEST(ZeroOnFreeBufferTest, TestZeroOnSetDataFromSetter) {
   // been zeroed.
   EXPECT_EQ(kTestData[offset], buf[0]);
   EXPECT_EQ(kTestData[offset + 1], buf[1]);
-  for (size_t i = 2; i < old_size; i++) {
-    EXPECT_EQ(0, old_data[i]);
-  }
+  EXPECT_THAT(to_be_zeroed, Each(Eq(0)));
 }
 
 TEST(ZeroOnFreeBufferTest, TestZeroOnSetSize) {
-  ZeroOnFreeBuffer<uint8_t> buf(kTestData, 7);
+  ZeroOnFreeBuffer<uint8_t> buf(kTestData.data(), 7);
+  auto to_be_zeroed = ArrayView<const uint8_t>(buf).subspan(2);
   const uint8_t* old_data = buf.data();
   const size_t old_capacity = buf.capacity();
-  const size_t old_size = buf.size();
   buf.SetSize(2);
   // Sanity checks to make sure the underlying heap memory was not reallocated.
   EXPECT_EQ(old_data, buf.data());
@@ -527,24 +534,20 @@ TEST(ZeroOnFreeBufferTest, TestZeroOnSetSize) {
   // been zeroed.
   EXPECT_EQ(kTestData[0], buf[0]);
   EXPECT_EQ(kTestData[1], buf[1]);
-  for (size_t i = 2; i < old_size; i++) {
-    EXPECT_EQ(0, old_data[i]);
-  }
+  EXPECT_THAT(to_be_zeroed, Each(Eq(0)));
 }
 
 TEST(ZeroOnFreeBufferTest, TestZeroOnClear) {
-  ZeroOnFreeBuffer<uint8_t> buf(kTestData, 7);
+  ZeroOnFreeBuffer<uint8_t> buf(kTestData.data(), 7);
+  auto to_be_zeroed = ArrayView<const uint8_t>(buf);
   const uint8_t* old_data = buf.data();
   const size_t old_capacity = buf.capacity();
-  const size_t old_size = buf.size();
   buf.Clear();
   // Sanity checks to make sure the underlying heap memory was not reallocated.
   EXPECT_EQ(old_data, buf.data());
   EXPECT_EQ(old_capacity, buf.capacity());
   // The underlying memory was not released but cleared.
-  for (size_t i = 0; i < old_size; i++) {
-    EXPECT_EQ(0, old_data[i]);
-  }
+  EXPECT_THAT(to_be_zeroed, Each(Eq(0)));
 }
 
 }  // namespace webrtc

From dc0041e35ea3253f4606377511e4760a151901e5 Mon Sep 17 00:00:00 2001
From: AhmadDurrani579 <ahmaddurranitrg@gmail.com>
Date: Fri, 6 Feb 2026 18:19:09 +0000
Subject: [PATCH 006/472] Add RTC_CHECK to fwrite calls in video_file_writer
 The compiler option `warn_unused_result` causes build failures if the return
 value of `fwrite` is ignored. Currently, `video_file_writer.cc` ignores these
 return values, which is also unsafe if the disk is full.

This CL wraps `fwrite` calls in `RTC_CHECK_EQ` to ensure the correct number of bytes are written, crashing safely if the write fails.

Bug: None
Test: Compiled locally.
Change-Id: I6cb62b673af2d3cefc6afdb9064d86bd9f080cd3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/447020
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46859}
---
 rtc_tools/video_file_writer.cc | 12 +++++++-----
 1 file changed, 7 insertions(+), 5 deletions(-)

diff --git a/rtc_tools/video_file_writer.cc b/rtc_tools/video_file_writer.cc
index 2a1bc67cc6..abfeb90b6b 100644
--- a/rtc_tools/video_file_writer.cc
+++ b/rtc_tools/video_file_writer.cc
@@ -15,6 +15,7 @@
 #include <string>
 
 #include "absl/strings/match.h"
+#include "absl/strings/string_view.h"
 #include "api/scoped_refptr.h"
 #include "api/video/video_frame_buffer.h"
 #include "rtc_base/checks.h"
@@ -42,8 +43,9 @@ void WriteVideoToFile(const scoped_refptr<Video>& video,
   }
   for (size_t i = 0; i < video->number_of_frames(); ++i) {
     if (isY4m) {
-      std::string frame = "FRAME\n";
-      fwrite(frame.c_str(), 1, 6, output_file);
+      absl::string_view frame = "FRAME\n";
+      RTC_CHECK_EQ(fwrite(frame.data(), 1, frame.length(), output_file),
+                   frame.length());
     }
     scoped_refptr<I420BufferInterface> buffer = video->GetFrame(i);
     RTC_CHECK(buffer) << "Frame: " << i
@@ -51,17 +53,17 @@ void WriteVideoToFile(const scoped_refptr<Video>& video,
     const uint8_t* data_y = buffer->DataY();
     int stride = buffer->StrideY();
     for (int j = 0; j < video->height(); ++j) {
-      fwrite(data_y + j * stride, /*size=*/1, stride, output_file);
+      RTC_CHECK_EQ(fwrite(data_y + j * stride, 1, stride, output_file), stride);
     }
     const uint8_t* data_u = buffer->DataU();
     stride = buffer->StrideU();
     for (int j = 0; j < buffer->ChromaHeight(); ++j) {
-      fwrite(data_u + j * stride, /*size=*/1, stride, output_file);
+      RTC_CHECK_EQ(fwrite(data_u + j * stride, 1, stride, output_file), stride);
     }
     const uint8_t* data_v = buffer->DataV();
     stride = buffer->StrideV();
     for (int j = 0; j < buffer->ChromaHeight(); ++j) {
-      fwrite(data_v + j * stride, /*size=*/1, stride, output_file);
+      RTC_CHECK_EQ(fwrite(data_v + j * stride, 1, stride, output_file), stride);
     }
   }
   if (ferror(output_file) != 0) {

From 7fa36c54a3a11ab96af41dfd06b517e1d4972768 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Erik=20Spr=C3=A5ng?= <sprang@webrtc.org>
Date: Tue, 10 Feb 2026 16:44:54 +0100
Subject: [PATCH 007/472] Transition quality scaler to new DropReason type.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This CL moves the `QualityScalerResource` and related files from using
the old `EncodedImageCallback::DropReason` enum type to the more
appropriate `VideoStreamEncoderObserver::DropReason`.

The prepares for the eventual removal of the former type.

Bug: webrtc:467444018
Change-Id: Ib8c3da0c1af917575c9b916f770b919329699795
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/447001
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46860}
---
 video/adaptation/quality_scaler_resource.cc   |  17 ++-
 video/adaptation/quality_scaler_resource.h    |   3 +-
 .../video_stream_encoder_resource_manager.cc  |   2 +-
 .../video_stream_encoder_resource_manager.h   |   2 +-
 video/video_stream_encoder.cc                 |  12 +-
 video/video_stream_encoder_unittest.cc        | 103 ++++++++++++++++++
 6 files changed, 126 insertions(+), 13 deletions(-)

diff --git a/video/adaptation/quality_scaler_resource.cc b/video/adaptation/quality_scaler_resource.cc
index 6f6e93caed..a0a1a1a163 100644
--- a/video/adaptation/quality_scaler_resource.cc
+++ b/video/adaptation/quality_scaler_resource.cc
@@ -24,6 +24,7 @@
 #include "modules/video_coding/utility/quality_scaler.h"
 #include "rtc_base/checks.h"
 #include "video/adaptation/video_stream_encoder_resource.h"
+#include "video/video_stream_encoder_observer.h"
 
 namespace webrtc {
 
@@ -78,17 +79,15 @@ void QualityScalerResource::OnEncodeCompleted(const EncodedImage& encoded_image,
 }
 
 void QualityScalerResource::OnFrameDropped(
-    EncodedImageCallback::DropReason reason) {
+    VideoStreamEncoderObserver::DropReason reason) {
   RTC_DCHECK_RUN_ON(encoder_queue());
-  if (!quality_scaler_)
-    return;
-  switch (reason) {
-    case EncodedImageCallback::DropReason::kDroppedByMediaOptimizations:
-      quality_scaler_->ReportDroppedFrameByMediaOpt();
-      break;
-    case EncodedImageCallback::DropReason::kDroppedByEncoder:
+  if (quality_scaler_) {
+    if (reason == VideoStreamEncoderObserver::DropReason::kEncoder) {
       quality_scaler_->ReportDroppedFrameByEncoder();
-      break;
+    } else if (reason ==
+               VideoStreamEncoderObserver::DropReason::kMediaOptimization) {
+      quality_scaler_->ReportDroppedFrameByMediaOpt();
+    }
   }
 }
 
diff --git a/video/adaptation/quality_scaler_resource.h b/video/adaptation/quality_scaler_resource.h
index 2e916bb586..3d23ddc151 100644
--- a/video/adaptation/quality_scaler_resource.h
+++ b/video/adaptation/quality_scaler_resource.h
@@ -21,6 +21,7 @@
 #include "modules/video_coding/utility/quality_scaler.h"
 #include "rtc_base/thread_annotations.h"
 #include "video/adaptation/video_stream_encoder_resource.h"
+#include "video/video_stream_encoder_observer.h"
 
 namespace webrtc {
 
@@ -41,7 +42,7 @@ class QualityScalerResource : public VideoStreamEncoderResource,
   void SetQpThresholds(VideoEncoder::QpThresholds qp_thresholds);
   void OnEncodeCompleted(const EncodedImage& encoded_image,
                          int64_t time_sent_in_us);
-  void OnFrameDropped(EncodedImageCallback::DropReason reason);
+  void OnFrameDropped(VideoStreamEncoderObserver::DropReason reason);
 
   // QualityScalerQpUsageHandlerInterface implementation.
   void OnReportQpUsageHigh() override;
diff --git a/video/adaptation/video_stream_encoder_resource_manager.cc b/video/adaptation/video_stream_encoder_resource_manager.cc
index 399430ec3c..b79a186012 100644
--- a/video/adaptation/video_stream_encoder_resource_manager.cc
+++ b/video/adaptation/video_stream_encoder_resource_manager.cc
@@ -496,7 +496,7 @@ void VideoStreamEncoderResourceManager::OnEncodeCompleted(
 }
 
 void VideoStreamEncoderResourceManager::OnFrameDropped(
-    EncodedImageCallback::DropReason reason) {
+    VideoStreamEncoderObserver::DropReason reason) {
   RTC_DCHECK_RUN_ON(encoder_queue_);
   quality_scaler_resource_->OnFrameDropped(reason);
 }
diff --git a/video/adaptation/video_stream_encoder_resource_manager.h b/video/adaptation/video_stream_encoder_resource_manager.h
index 66c94581dd..3645b22825 100644
--- a/video/adaptation/video_stream_encoder_resource_manager.h
+++ b/video/adaptation/video_stream_encoder_resource_manager.h
@@ -121,7 +121,7 @@ class VideoStreamEncoderResourceManager
                          int64_t time_sent_in_us,
                          std::optional<int> encode_duration_us,
                          DataSize frame_size);
-  void OnFrameDropped(EncodedImageCallback::DropReason reason);
+  void OnFrameDropped(VideoStreamEncoderObserver::DropReason reason);
 
   // Resources need to be mapped to an AdaptReason (kCpu or kQuality) in order
   // to update legacy getStats().
diff --git a/video/video_stream_encoder.cc b/video/video_stream_encoder.cc
index eea90cad3b..ad12271019 100644
--- a/video/video_stream_encoder.cc
+++ b/video/video_stream_encoder.cc
@@ -2443,7 +2443,17 @@ void VideoStreamEncoder::OnDroppedFrame(DropReason reason) {
   sink_->OnDroppedFrame(reason);
   encoder_queue_->PostTask([this, reason] {
     RTC_DCHECK_RUN_ON(encoder_queue_.get());
-    stream_resource_manager_.OnFrameDropped(reason);
+    switch (reason) {
+      case webrtc::EncodedImageCallback::DropReason::kDroppedByEncoder:
+        stream_resource_manager_.OnFrameDropped(
+            VideoStreamEncoderObserver::DropReason::kEncoder);
+        break;
+      case webrtc::EncodedImageCallback::DropReason::
+          kDroppedByMediaOptimizations:
+        stream_resource_manager_.OnFrameDropped(
+            VideoStreamEncoderObserver::DropReason::kMediaOptimization);
+        break;
+    }
   });
 }
 
diff --git a/video/video_stream_encoder_unittest.cc b/video/video_stream_encoder_unittest.cc
index 154716d0b4..016d3c2f09 100644
--- a/video/video_stream_encoder_unittest.cc
+++ b/video/video_stream_encoder_unittest.cc
@@ -43,6 +43,8 @@
 #include "api/test/rtc_error_matchers.h"
 #include "api/test/time_controller.h"
 #include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/frequency.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
 #include "api/video/builtin_video_bitrate_allocator_factory.h"
@@ -1148,6 +1150,9 @@ class VideoStreamEncoderTest : public ::testing::Test {
       if (is_qp_trusted_.has_value()) {
         info.is_qp_trusted = is_qp_trusted_;
       }
+      if (has_trusted_rate_controller_.has_value()) {
+        info.has_trusted_rate_controller = *has_trusted_rate_controller_;
+      }
       return info;
     }
 
@@ -1248,6 +1253,11 @@ class VideoStreamEncoderTest : public ::testing::Test {
       encoded_image_data_ = encoded_image_data;
     }
 
+    void ForceDropFrames(bool drop) {
+      MutexLock lock(&local_mutex_);
+      drop_frames_ = drop;
+    }
+
     void ExpectNullFrame() {
       MutexLock lock(&local_mutex_);
       expect_null_frame_ = true;
@@ -1297,6 +1307,11 @@ class VideoStreamEncoderTest : public ::testing::Test {
       is_qp_trusted_ = trusted;
     }
 
+    void SetIsRateControlTrusted(bool trusted) {
+      MutexLock lock(&local_mutex_);
+      has_trusted_rate_controller_ = trusted;
+    }
+
     VideoCodecComplexity LastEncoderComplexity() {
       MutexLock lock(&local_mutex_);
       return last_encoder_complexity_;
@@ -1308,6 +1323,13 @@ class VideoStreamEncoderTest : public ::testing::Test {
       {
         MutexLock lock(&local_mutex_);
         num_encodes_++;
+        if (drop_frames_) {
+          if (encoded_image_callback_) {
+            encoded_image_callback_->OnDroppedFrame(
+                EncodedImageCallback::DropReason::kDroppedByEncoder);
+          }
+          return WEBRTC_VIDEO_CODEC_OK;
+        }
         if (expect_null_frame_) {
           EXPECT_EQ(input_image.rtp_timestamp(), 0u);
           EXPECT_EQ(input_image.width(), 1);
@@ -1433,6 +1455,7 @@ class VideoStreamEncoderTest : public ::testing::Test {
         local_mutex_) = {.offset_x = 0, .offset_y = 0, .width = 0, .height = 0};
     std::vector<VideoFrameType> last_frame_types_;
     bool expect_null_frame_ = false;
+    bool drop_frames_ RTC_GUARDED_BY(local_mutex_) = false;
     EncodedImageCallback* encoded_image_callback_ RTC_GUARDED_BY(local_mutex_) =
         nullptr;
     NiceMock<MockFecControllerOverride> fec_controller_override_;
@@ -1445,6 +1468,8 @@ class VideoStreamEncoderTest : public ::testing::Test {
     absl::InlinedVector<VideoFrameBuffer::Type, kMaxPreferredPixelFormats>
         preferred_pixel_formats_ RTC_GUARDED_BY(local_mutex_);
     std::optional<bool> is_qp_trusted_ RTC_GUARDED_BY(local_mutex_);
+    std::optional<bool> has_trusted_rate_controller_
+        RTC_GUARDED_BY(local_mutex_);
     VideoCodecComplexity last_encoder_complexity_ RTC_GUARDED_BY(local_mutex_){
         VideoCodecComplexity::kComplexityNormal};
   };
@@ -1970,6 +1995,84 @@ TEST_F(VideoStreamEncoderTest, DropsPendingFramesOnSlowEncode) {
   EXPECT_EQ(1, dropped_count);
 }
 
+TEST_F(VideoStreamEncoderTest, DropsFramesOnBitrateOvershoot) {
+  ConfigureEncoder(video_encoder_config_.Copy());
+  video_stream_encoder_->OnBitrateUpdatedAndWaitForManagedResources(
+      kTargetBitrate, kTargetBitrate, 0, 0, 0);
+
+  // Set encoder rate control as not trusted - so that the external frame
+  // dropper is activated.
+  fake_encoder_.SetIsRateControlTrusted(false);
+
+  DataSize ideal_size_per_frame =
+      kTargetBitrate / Frequency::Hertz(kDefaultFramerate);
+  // Create an image buffer that is consistently four times larger than
+  // expected.
+  scoped_refptr<EncodedImageBuffer> encoded_image_buffer =
+      EncodedImageBuffer::Create(ideal_size_per_frame.bytes() * 4);
+  std::fill(encoded_image_buffer->data(),
+            encoded_image_buffer->data() + encoded_image_buffer->size(), 0);
+
+  bool has_media_opt_frame_drop = false;
+  stats_proxy_->SetDroppedFrameCallback(
+      [&has_media_opt_frame_drop](
+          VideoStreamEncoderObserver::DropReason reason) {
+        if (reason ==
+            VideoStreamEncoderObserver::DropReason::kMediaOptimization) {
+          has_media_opt_frame_drop = true;
+        }
+      });
+
+  for (int i = 0; i < 30; ++i) {
+    int64_t timestamp = CurrentTimeMs();
+    fake_encoder_.SetEncodedImageData(encoded_image_buffer);
+    video_source_.IncomingCapturedFrame(
+        CreateFrame(timestamp, codec_width_, codec_height_));
+    AdvanceTime(TimeDelta::Millis(kFrameIntervalMs));
+    if (has_media_opt_frame_drop) {
+      break;
+    }
+  }
+
+  EXPECT_TRUE(has_media_opt_frame_drop);
+
+  video_stream_encoder_->Stop();
+}
+
+TEST_F(VideoStreamEncoderTest, QualityScalerTriggersOnFrameDropping) {
+  // Set resolution high so there is a lower resolution to adapt to.
+  codec_width_ = 1280;
+  codec_height_ = 720;
+  ConfigureEncoder(video_encoder_config_.Copy());
+  video_stream_encoder_->OnBitrateUpdatedAndWaitForManagedResources(
+      kTargetBitrate, kTargetBitrate, 0, 0, 0);
+
+  // Set a low QP so that the qualitu scaler won't trigger because of QP.
+  fake_encoder_.SetQp(kQpLow);
+  fake_encoder_.SetQualityScaling(true);
+  video_source_.set_adaptation_enabled(true);
+
+  // Drop all frames so that the quality scaler will trigger (at the time of
+  // writing frame drop percentage > 65% is used in the quality scaler
+  // implementation).
+  fake_encoder_.ForceDropFrames(true);
+
+  for (int i = 0; i < 100; ++i) {
+    int64_t timestamp = CurrentTimeMs();
+    video_source_.IncomingCapturedFrame(
+        CreateFrame(timestamp, codec_width_, codec_height_));
+    AdvanceTime(TimeDelta::Millis(kFrameIntervalMs));
+    // After enough dropped frames the quality scaler should trigger.
+    if (stats_proxy_->GetStats().bw_limited_resolution) {
+      break;
+    }
+  }
+
+  EXPECT_TRUE(stats_proxy_->GetStats().bw_limited_resolution);
+
+  video_stream_encoder_->Stop();
+}
+
 TEST_F(VideoStreamEncoderTest, NativeFrameWithoutI420SupportGetsDelivered) {
   video_stream_encoder_->OnBitrateUpdatedAndWaitForManagedResources(
       kTargetBitrate, kTargetBitrate, 0, 0, 0);

From e902e15d1ca767288927e722a5af32ad12a48752 Mon Sep 17 00:00:00 2001
From: Sameer Vijaykar <samvi@google.com>
Date: Tue, 10 Feb 2026 16:08:06 +0100
Subject: [PATCH 008/472] Add network slice to Network::ToString()

BUG=webrtc:466507512

Change-Id: Ida1b809a139dc9da56c27fbc11f48cab9da07eba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/448040
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Cr-Commit-Position: refs/heads/main@{#46861}
---
 rtc_base/network.cc         |  3 +++
 rtc_base/network_monitor.cc | 14 +++++++++++---
 rtc_base/network_monitor.h  |  3 ++-
 3 files changed, 16 insertions(+), 4 deletions(-)

diff --git a/rtc_base/network.cc b/rtc_base/network.cc
index 16bb0e2fd6..676a11273f 100644
--- a/rtc_base/network.cc
+++ b/rtc_base/network.cc
@@ -1324,6 +1324,9 @@ std::string Network::ToString() const {
   if (IsVpn()) {
     ss << "/" << AdapterTypeToString(underlying_type_for_vpn_);
   }
+  if (network_slice() != NetworkSlice::NO_SLICE) {
+    ss << "/" << NetworkSliceToString(network_slice());
+  }
   ss << ":id=" << id_ << "]";
   return ss.Release();
 }
diff --git a/rtc_base/network_monitor.cc b/rtc_base/network_monitor.cc
index 0733d8eea6..2595b68a91 100644
--- a/rtc_base/network_monitor.cc
+++ b/rtc_base/network_monitor.cc
@@ -10,18 +10,26 @@
 
 #include "rtc_base/network_monitor.h"
 
-#include "rtc_base/checks.h"
+#include "absl/strings/string_view.h"
 
 namespace webrtc {
 
-const char* NetworkPreferenceToString(NetworkPreference preference) {
+absl::string_view NetworkPreferenceToString(NetworkPreference preference) {
   switch (preference) {
     case NetworkPreference::NEUTRAL:
       return "NEUTRAL";
     case NetworkPreference::NOT_PREFERRED:
       return "NOT_PREFERRED";
   }
-  RTC_CHECK_NOTREACHED();
+}
+
+absl::string_view NetworkSliceToString(NetworkSlice network_slice) {
+  switch (network_slice) {
+    case NetworkSlice::NO_SLICE:
+      return "NO_SLICE";
+    case NetworkSlice::UNIFIED_COMMUNICATIONS:
+      return "UNIFIED_COMMUNICATIONS";
+  }
 }
 
 NetworkMonitorInterface::NetworkMonitorInterface() {}
diff --git a/rtc_base/network_monitor.h b/rtc_base/network_monitor.h
index 9d14070762..10889afdf7 100644
--- a/rtc_base/network_monitor.h
+++ b/rtc_base/network_monitor.h
@@ -46,7 +46,8 @@ enum class NetworkSlice {
   UNIFIED_COMMUNICATIONS = 1,
 };
 
-const char* NetworkPreferenceToString(NetworkPreference preference);
+absl::string_view NetworkPreferenceToString(NetworkPreference preference);
+absl::string_view NetworkSliceToString(NetworkSlice network_slice);
 
 // This interface is set onto a socket server,
 // where only the ip address is known at the time of binding.

From b3bdddbbc11c1baf5eb7d8145fe37118f378db50 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Tue, 10 Feb 2026 14:03:10 -0800
Subject: [PATCH 009/472] Roll chromium_revision dbeec2f05b..be8f64a423
 (1580859:1582802)

Change log: https://chromium.googlesource.com/chromium/src/+log/dbeec2f05b..be8f64a423
Full diff: https://chromium.googlesource.com/chromium/src/+/dbeec2f05b..be8f64a423

Changed dependencies
* siso_version: git_revision:73b1681189a8d8a4d922cd84246dcec2494142a5..git_revision:2599cc0037d82d9187beea5c13349ce05716817b
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/a37e61dc22..e82653b704
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/6ce374626d..d8de59743d
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/b887106bc2..62968b4611
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/e43b96b7a6..f4814d241e
* src/third_party/android_build_tools/aapt2/cipd: s6POXpUalcnuPehDsORiojCpgbNXT4LYq7DVUYgsfxEC..vUP1cs7krVsNHbUJpwSDR7rvWuYLGzINVa9xopPpt74C
* src/third_party/android_build_tools/manifest_merger/cipd: YgbGvXYO4RiQ7zH0QiMCxe_f9hE2k9JW8lUOKTkYC70C..xxaa38R0ED95Wxo8P29BC-mB7bjFPi2HKMpiIJn-UhAC
* src/third_party/android_deps/autorolled/cipd: LStkoCV2NKgZ1hqhnILaH4soxZtZa2-8RG67-JxAVUsC..RiKmV-EpMhg5ibpMbPpENQmfU4IPLpo8cxer4F3NSUgC
* src/third_party/androidx/cipd: dJ3xGQgP3gjT2n9aXf2sj4QdckdRZXrVhhrlsHQg8QQC..us_8XGxuKdmIoJEbqTnp_Ehw-VEaxJIVTzMy2h0fnhwC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/305bcfce00..73cd946964
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/39805a224b..a476f554f8
* src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/d606e955ee..8658704610
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/9fd48a305e..42786f6e46
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/afb4380534..23fcc1c6c9
* src/third_party/ffmpeg: https://chromium.googlesource.com/chromium/third_party/ffmpeg.git/+log/9e588ab02e..ae11d2ba5c
* src/third_party/fuzztest/src: https://chromium.googlesource.com/external/github.com/google/fuzztest.git/+log/54dfec04d5..54b2ccd592
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/f394c3f07b..75ff03fd6c
* src/third_party/kotlin_stdlib/cipd: eZzGGvckJ-abo5lKKVxignwQeuPMQcgDp6p5ITnnI-kC..FfVJUc4q9S6JB4uqVGPJQ3anrc1Y_W0O-oGK_2MJA5gC
* src/third_party/kotlinc/current: f-d0mg_APXr9lbudXU46Wlvt0hnZpuViqvhw3MGhTHwC..KZWh3uhuenRgsoGqNYgSo82FO3sMwZjvmuR9TBvanTcC
* src/third_party/libaom/source/libaom: https://aomedia.googlesource.com/aom.git/+log/557586fde2..4018d3b634
* src/third_party/libvpx/source/libvpx: https://chromium.googlesource.com/webm/libvpx.git/+log/e83e25f791..a0ac06f5d3
* src/third_party/llvm-build/Release+Asserts: Linux_x64/clang-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz,Linux_x64/clang-tidy-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz,Linux_x64/clangd-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz,Linux_x64/llvm-code-coverage-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz,Linux_x64/llvmobjdump-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz,Mac/clang-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz,Mac/clang-mac-runtime-library-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz,Mac/clang-tidy-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz,Mac/clangd-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz,Mac/llvm-code-coverage-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz,Mac/llvmobjdump-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz,Mac_arm64/clang-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz,Mac_arm64/clang-tidy-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz,Mac_arm64/clangd-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz,Mac_arm64/llvm-code-coverage-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz,Mac_arm64/llvmobjdump-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz,Win/clang-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz,Win/clang-tidy-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz,Win/clang-win-runtime-library-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz,Win/clangd-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz,Win/llvm-code-coverage-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz,Win/llvmobjdump-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz..Linux_x64/clang-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Linux_x64/clang-tidy-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Linux_x64/clangd-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Linux_x64/llvm-code-coverage-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Linux_x64/llvmobjdump-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Mac/clang-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Mac/clang-mac-runtime-library-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Mac/clang-tidy-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Mac/clangd-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Mac/llvm-code-coverage-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Mac/llvmobjdump-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Mac_arm64/clang-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Mac_arm64/clang-tidy-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Mac_arm64/clangd-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Mac_arm64/llvm-code-coverage-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Mac_arm64/llvmobjdump-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Win/clang-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Win/clang-tidy-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Win/clang-win-runtime-library-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Win/clangd-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Win/llvm-code-coverage-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Win/llvmobjdump-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz
* src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/d99c56d4b9..4f89a98f36
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/2074b65d22..4d0f63f082
* src/third_party/r8/cipd: kCLXGwoL10sgeyycuTLyDhesWICC0DSVfVf-a6U6_yUC..TozMGRBtGJuZp31wJjyY7VkLaQ6SI8lsRnLVsSlmmogC
* src/third_party/siso/cipd: git_revision:73b1681189a8d8a4d922cd84246dcec2494142a5..git_revision:2599cc0037d82d9187beea5c13349ce05716817b
* src/third_party/turbine/cipd: j49Y1F_PAMGZdDf53rGaEzVXdT6jXWDhEm7UMfGhjswC..BMHNhxMhr7uGz1rh_Od_JE4kAdP9K5MXr6GN2R9tQkAC
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/f8f65faf07..17573b6d7d
Removed dependency
* src/third_party/android_deps/cipd/libs/org_jetbrains_kotlinx_kotlinx_coroutines_guava
DEPS diff: https://chromium.googlesource.com/chromium/src/+/dbeec2f05b..be8f64a423/DEPS

BUG=None

Change-Id: I2a4982cd6c99e13d073270735560cdd9839febb3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/448025
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#46862}
---
 DEPS | 241 ++++++++++++++++++++++++++++-------------------------------
 1 file changed, 115 insertions(+), 126 deletions(-)

diff --git a/DEPS b/DEPS
index 5040c451fe..3aeb7155f3 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': 'dbeec2f05bf41e3ef61dbfeb1486aa2fbc4fd7f6',
+  'chromium_revision': 'be8f64a4234dbe7b2308a01dc57dddfea118fc1d',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -50,7 +50,7 @@ vars = {
   # reclient CIPD package version
   'reclient_version': 're_client_version:0.185.0.db415f21-gomaip',
   # siso CIPD package version.
-  'siso_version': 'git_revision:73b1681189a8d8a4d922cd84246dcec2494142a5',
+  'siso_version': 'git_revision:2599cc0037d82d9187beea5c13349ce05716817b',
 
   # ninja CIPD package.
   'ninja_package': 'infra/3pp/tools/ninja/',
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@a37e61dc22fd633ab32146e7000068a57a2864ff',
+    'https://chromium.googlesource.com/chromium/src/build@e82653b70411841da3662ed6edf96694a55f4068',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@6a18683f555b4ac8b05ac8395c29c84483ac9588',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@6ce374626d5f5e43356fa2cc337a88ced639ef34',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@d8de59743dc98dc9932b7db60f90cc3db0149c2f',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@b887106bc278bb2a49e21f3a0ab2019574e7e47a',
+    'https://chromium.googlesource.com/chromium/src/testing@62968b461188cb73855fcc23c540ee8472eb81c4',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@e43b96b7a65dd3f45f066983061e6f8b2f3a112d',
+    'https://chromium.googlesource.com/chromium/src/third_party@f4814d241e26dc251fabe5c66b0c3b997f28a8c8',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -136,157 +136,157 @@ deps = {
     'objects': [
       {
         # The Android libclang_rt.builtins libraries are currently only included in the Linux clang package.
-        'object_name': 'Linux_x64/clang-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz',
-        'sha256sum': 'd373cde5b6f1c0da245ebcad93e4883252323c1ff283df96f0eb2e9180f1a537',
-        'size_bytes': 57692132,
-        'generation': 1769798229467453,
+        'object_name': 'Linux_x64/clang-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
+        'sha256sum': 'bab27d04b9740f772fb6f50461ba4c2cf96c57231eb83943f785775c0bc274a6',
+        'size_bytes': 57583584,
+        'generation': 1770400171446681,
         'condition': '(host_os == "linux" or checkout_android) and non_git_source',
       },
       {
-        'object_name': 'Linux_x64/clang-tidy-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz',
-        'sha256sum': '334d3f2eaf1abf19a34539faff8d20674ca0654994b330be1e240df3d6f925e8',
-        'size_bytes': 14414652,
-        'generation': 1769798229616278,
+        'object_name': 'Linux_x64/clang-tidy-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
+        'sha256sum': '00633aec3bf29aba53206cb1c58256367954670e1eb4665bd08c401a4c9e538d',
+        'size_bytes': 14383508,
+        'generation': 1770400171915898,
         'condition': 'host_os == "linux" and checkout_clang_tidy and non_git_source',
       },
       {
-        'object_name': 'Linux_x64/clangd-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz',
-        'sha256sum': '2b439769a7072f488c9c792bbbd7527215d8c0f3791bcedb2233c604e2ac736f',
-        'size_bytes': 14647728,
-        'generation': 1769798229728240,
+        'object_name': 'Linux_x64/clangd-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
+        'sha256sum': 'a67ff56b3a3feb84a204952b6e517686d70cc9eb3744694f8a4ee4e62538efd8',
+        'size_bytes': 14612000,
+        'generation': 1770400171913400,
         'condition': 'host_os == "linux" and checkout_clangd and non_git_source',
       },
       {
-        'object_name': 'Linux_x64/llvm-code-coverage-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz',
-        'sha256sum': '62ecd25bd5d8820fa694c68687b262bd85cce5512e640341eac80648a182a8bf',
-        'size_bytes': 2325696,
-        'generation': 1769798229958559,
+        'object_name': 'Linux_x64/llvm-code-coverage-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
+        'sha256sum': '8920f2faeb5c1620e2df2ebfc17afb1a7405a9ce0681560cd95caa72dee9348d',
+        'size_bytes': 2312556,
+        'generation': 1770400172534077,
         'condition': 'host_os == "linux" and checkout_clang_coverage_tools and non_git_source',
       },
       {
-        'object_name': 'Linux_x64/llvmobjdump-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz',
-        'sha256sum': '28df84ef25bae64ab82a48295bd09aa6689e436449bd78cdc83f58ef05bfdd6a',
-        'size_bytes': 5818392,
-        'generation': 1769798229787736,
+        'object_name': 'Linux_x64/llvmobjdump-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
+        'sha256sum': '5fa2372512142aba2824aadd652d8a5416d75806df046876be344b53c1c7d983',
+        'size_bytes': 5785056,
+        'generation': 1770400172080602,
         'condition': '((checkout_linux or checkout_mac or checkout_android) and host_os == "linux") and non_git_source',
       },
       {
-        'object_name': 'Mac/clang-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz',
-        'sha256sum': '0d785b952cdbba91e5ca4c72c8e12d3f9a5cc44fb37d58be87a62bdb3a0b191e',
-        'size_bytes': 54770836,
-        'generation': 1769798231469873,
+        'object_name': 'Mac/clang-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
+        'sha256sum': '89bf5f88dd0e919f1906eb3d3d1c3c7268dc8163d924f492dac4d2627f2c8d94',
+        'size_bytes': 54752956,
+        'generation': 1770400174327435,
         'condition': 'host_os == "mac" and host_cpu == "x64"',
       },
       {
-        'object_name': 'Mac/clang-mac-runtime-library-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz',
-        'sha256sum': 'd6f35ab337540994ee942f0300469bc627041b54cd0e8f1942db94da795ab8de',
-        'size_bytes': 1010540,
-        'generation': 1769798240394528,
+        'object_name': 'Mac/clang-mac-runtime-library-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
+        'sha256sum': '2c5a55301498d63286a4d249531254e586017c889e16d5d91fdb9db7fdc61451',
+        'size_bytes': 1014688,
+        'generation': 1770400193401638,
         'condition': 'checkout_mac and not host_os == "mac"',
       },
       {
-        'object_name': 'Mac/clang-tidy-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz',
-        'sha256sum': '0a89c1cc843537ab4a81b69f03923b8e29778b89c156341a5d589b8742f0c44f',
-        'size_bytes': 14492308,
-        'generation': 1769798231635674,
+        'object_name': 'Mac/clang-tidy-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
+        'sha256sum': 'fb67062f7fb7edc2ac5a2a7a698a29446f93b8e8c091258b5194aa2b8c166ce1',
+        'size_bytes': 14495128,
+        'generation': 1770400174657643,
         'condition': 'host_os == "mac" and host_cpu == "x64" and checkout_clang_tidy',
       },
       {
-        'object_name': 'Mac/clangd-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz',
-        'sha256sum': '768e49e08e0e07d536fea63bcfb7e138409972d8da1493419f438ea39947b50c',
-        'size_bytes': 16432880,
-        'generation': 1769798231738122,
+        'object_name': 'Mac/clangd-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
+        'sha256sum': '7d61e0cbca5ef8e9e2f2295097cb996abda3df85e003f9d64c54715b28312b12',
+        'size_bytes': 16448964,
+        'generation': 1770400174847386,
         'condition': 'host_os == "mac" and host_cpu == "x64" and checkout_clangd',
       },
       {
-        'object_name': 'Mac/llvm-code-coverage-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz',
-        'sha256sum': 'ec463ac915dd97d60715622799bf67b99ad56675917d1d12940c90da0846079a',
-        'size_bytes': 2358616,
-        'generation': 1769798231966010,
+        'object_name': 'Mac/llvm-code-coverage-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
+        'sha256sum': '35003b14528499ad7f51a5f0388f39708bc779502d289d884532275c3161f1fa',
+        'size_bytes': 2356044,
+        'generation': 1770400175337551,
         'condition': 'host_os == "mac" and host_cpu == "x64" and checkout_clang_coverage_tools',
       },
       {
-        'object_name': 'Mac/llvmobjdump-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz',
-        'sha256sum': '163ff23c49a6980d96bc540bd68fcfe6f2574ff7488446290dea8e378d659392',
-        'size_bytes': 5693264,
-        'generation': 1769798231791100,
+        'object_name': 'Mac/llvmobjdump-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
+        'sha256sum': '72ed5ca9220026efb8a4fabea6b9b2ad39ded51b1bdac8e2290602497bbd74aa',
+        'size_bytes': 5695864,
+        'generation': 1770400175016502,
         'condition': 'host_os == "mac" and host_cpu == "x64"',
       },
       {
-        'object_name': 'Mac_arm64/clang-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz',
-        'sha256sum': '45c4c6fa65d92ca1b7dc7e6221589cf7e7f3ae2b22300277150761f670e1b1b0',
-        'size_bytes': 45722620,
-        'generation': 1769798241897042,
+        'object_name': 'Mac_arm64/clang-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
+        'sha256sum': 'eb6086a89ae63808cb85a3b274b5c3870b832385f81b607f966d28b09971191f',
+        'size_bytes': 45747156,
+        'generation': 1770400195193634,
         'condition': 'host_os == "mac" and host_cpu == "arm64"',
       },
       {
-        'object_name': 'Mac_arm64/clang-tidy-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz',
-        'sha256sum': '32439ff7d3f5c6f431f3d543c1a59d0ea3c5f2639c4e77203f51b4e5e5aa6200',
-        'size_bytes': 12463612,
-        'generation': 1769798242077766,
+        'object_name': 'Mac_arm64/clang-tidy-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
+        'sha256sum': '77b5cdea3c51876a87c08a826297c1d113dd5dc298f973af12e98fee3afebd5a',
+        'size_bytes': 12451868,
+        'generation': 1770400195530884,
         'condition': 'host_os == "mac" and host_cpu == "arm64" and checkout_clang_tidy',
       },
       {
-        'object_name': 'Mac_arm64/clangd-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz',
-        'sha256sum': '4692dabe1495a5055db075b42b1b81511fef14a3d85fd7c374e83d4376d5a31f',
-        'size_bytes': 12850940,
-        'generation': 1769798242197514,
+        'object_name': 'Mac_arm64/clangd-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
+        'sha256sum': '032f8b1cac0e03703890fca0d0438515b92f36b438fd7d125e2b7bdb05216b3e',
+        'size_bytes': 12855596,
+        'generation': 1770400195729193,
         'condition': 'host_os == "mac" and host_cpu == "arm64" and checkout_clangd',
       },
       {
-        'object_name': 'Mac_arm64/llvm-code-coverage-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz',
-        'sha256sum': 'c52db431571b4d55cb7b6452b3c8524f890632d6909a09af93dbd488e0594918',
-        'size_bytes': 1981792,
-        'generation': 1769798242410391,
+        'object_name': 'Mac_arm64/llvm-code-coverage-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
+        'sha256sum': '7d696cdde5bdacde2eafcb3aaf5f7b3123f30998a75b50f84ce99aae57c59977',
+        'size_bytes': 1982584,
+        'generation': 1770400196199062,
         'condition': 'host_os == "mac" and host_cpu == "arm64" and checkout_clang_coverage_tools',
       },
       {
-        'object_name': 'Mac_arm64/llvmobjdump-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz',
-        'sha256sum': '0295ceffc9d4208b998ff51b4a7a0a89827a87f83b8ccbcbf6db4f904f8caff0',
-        'size_bytes': 5432532,
-        'generation': 1769798242264216,
+        'object_name': 'Mac_arm64/llvmobjdump-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
+        'sha256sum': '2656c91e724e5efaf844fdde1b2e14aeaf31bcb0766a7e28aa58f59a09b2b6b2',
+        'size_bytes': 5431708,
+        'generation': 1770400195856360,
         'condition': 'host_os == "mac" and host_cpu == "arm64"',
       },
       {
-        'object_name': 'Win/clang-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz',
-        'sha256sum': '2751f2de8cf1da96916fe124e1d5f6823b3c7810cb0e1436d58f82565a9676d5',
-        'size_bytes': 48990064,
-        'generation': 1769798253319534,
+        'object_name': 'Win/clang-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
+        'sha256sum': '36d195fd31754af28bb5e9203ea582b67813b2f5a7772f86d5d567ec887067a0',
+        'size_bytes': 49100488,
+        'generation': 1770400217126755,
         'condition': 'host_os == "win"',
       },
       {
-        'object_name': 'Win/clang-tidy-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz',
-        'sha256sum': '2e55b255c435e666804f6d1967f24b6be85f29515f4a8a890631df0139e6d442',
-        'size_bytes': 14394780,
-        'generation': 1769798253490087,
+        'object_name': 'Win/clang-tidy-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
+        'sha256sum': '533a1233533bd97b3cf9b80bf06124b49f981bdde28a8d302b4fb49eb94ba889',
+        'size_bytes': 14441820,
+        'generation': 1770400217463907,
         'condition': 'host_os == "win" and checkout_clang_tidy',
       },
       {
-        'object_name': 'Win/clang-win-runtime-library-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz',
-        'sha256sum': 'f1d21f63b5839dce9e53d965cca7b7252b39b2a0f13c447f151df01cdc33f1f7',
-        'size_bytes': 2545304,
-        'generation': 1769798262175764,
+        'object_name': 'Win/clang-win-runtime-library-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
+        'sha256sum': 'bed1dee4a3ab4e7cbfad5d33aeb3f5c4caed676de5ba0eb22a8eea69bfc0c670',
+        'size_bytes': 2590736,
+        'generation': 1770400239673699,
         'condition': 'checkout_win and not host_os == "win"',
       },
       {
-        'object_name': 'Win/clangd-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz',
-        'sha256sum': '6568f82ef6d3b8f19fae1e00f8534135edf86f195b0296f150784cc70d2dac93',
-        'size_bytes': 14793848,
-        'generation': 1769798253604910,
+        'object_name': 'Win/clangd-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
+        'sha256sum': '8c25b517a03977dfaf09e2b5d3ddf8e4602165ce692e43413e3d728e4618ace6',
+        'size_bytes': 14853072,
+        'generation': 1770400217655297,
         'condition': 'host_os == "win" and checkout_clangd',
       },
       {
-        'object_name': 'Win/llvm-code-coverage-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz',
-        'sha256sum': 'c96bb29efd1b4efae7a33cdc81f1d2d6104f9ad0d971a4c1b340cec6a5821fde',
-        'size_bytes': 2421196,
-        'generation': 1769798253771930,
+        'object_name': 'Win/llvm-code-coverage-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
+        'sha256sum': '8a6317407a98e3d3f99a30255c8559ff146d605b2d44e0e22d314cbd0c7f0636',
+        'size_bytes': 2466000,
+        'generation': 1770400218121410,
         'condition': 'host_os == "win" and checkout_clang_coverage_tools',
       },
       {
-        'object_name': 'Win/llvmobjdump-llvmorg-23-init-2224-g5bd8dadb-1.tar.xz',
-        'sha256sum': '54d94b4e1560870f94dee78c16c7828f95c185f363321fe605aa286d88104bfd',
-        'size_bytes': 5798400,
-        'generation': 1769798253622689,
+        'object_name': 'Win/llvmobjdump-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
+        'sha256sum': '9a78b48d45ea671e1eca8a9c35a1e89015505e0cbd33ccf2c29f9740ce1c0957',
+        'size_bytes': 5845536,
+        'generation': 1770400217790811,
         'condition': '(checkout_linux or checkout_mac or checkout_android) and host_os == "win"',
       },
     ]
@@ -331,13 +331,13 @@ deps = {
   'src/third_party/clang-format/script':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@c2725e0622e1a86d55f14514f2177a39efea4a0e',
   'src/third_party/compiler-rt/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@d606e955eec3d4fb0bf831dea336c3b24cba2f27',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@86587046105639b66fe40059bf8fdb8a33522f97',
   'src/third_party/libc++/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@7ab65651aed6802d2599dcb7a73b1f82d5179d05',
   'src/third_party/libc++abi/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@8f11bb1d4438d0239d0dfc1bd9456a9f31629dda',
   'src/third_party/llvm-libc/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@d99c56d4b9f6663bff528c4fac5313bceb32e762',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@4f89a98f364a8b42b19890afb5ba3ffa526a34f0',
   'src/third_party/libunwind/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@ba19d93d6d4f467fba11ff20fe2fc7c056f79345',
 
@@ -402,7 +402,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/android_build_tools/aapt2',
-              'version': 's6POXpUalcnuPehDsORiojCpgbNXT4LYq7DVUYgsfxEC',
+              'version': 'vUP1cs7krVsNHbUJpwSDR7rvWuYLGzINVa9xopPpt74C',
           },
       ],
       'condition': 'checkout_android',
@@ -488,11 +488,11 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@305bcfce00b189f2297f53365b0454f96009927d',
+    'https://boringssl.googlesource.com/boringssl.git@73cd946964c555a39730592db4b540accbafac6c',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@79099fdf668ae097c9eca7052fd5c4c5de6c9098',
   'src/third_party/catapult':
-    'https://chromium.googlesource.com/catapult.git@39805a224bb6c6e80e403a4ebe9a150c7ca0b4d1',
+    'https://chromium.googlesource.com/catapult.git@a476f554f8865b7d162ec9f1ad8aae1eab38e048',
   'src/third_party/ced/src': {
     'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5',
   },
@@ -505,13 +505,13 @@ deps = {
   'src/third_party/crc32c/src':
     'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558',
   'src/third_party/depot_tools':
-    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@9fd48a305e18b9bbaf61734557ce2c46497192b3',
+    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@42786f6e46c25c30dd58f69283ab6fcd0c959f58',
   'src/third_party/ffmpeg':
-    'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@9e588ab02e16326026aa61cc3b6515da20520cec',
+    'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@ae11d2ba5c835b822a61d6a99eeb853ca30d41d8',
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@f394c3f07b4c685b9f6948b36d65ca10a629f4fa',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@75ff03fd6c1763cbb4ab047df6628e57562237de',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -573,7 +573,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/kotlin_stdlib',
-              'version': 'eZzGGvckJ-abo5lKKVxignwQeuPMQcgDp6p5ITnnI-kC',
+              'version': 'FfVJUc4q9S6JB4uqVGPJQ3anrc1Y_W0O-oGK_2MJA5gC',
           },
       ],
       'condition': 'checkout_android',
@@ -584,7 +584,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/kotlinc',
-              'version': 'f-d0mg_APXr9lbudXU46Wlvt0hnZpuViqvhw3MGhTHwC',
+              'version': 'KZWh3uhuenRgsoGqNYgSo82FO3sMwZjvmuR9TBvanTcC',
           },
       ],
       'condition': 'checkout_android',
@@ -593,7 +593,7 @@ deps = {
   'src/third_party/libFuzzer/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt/lib/fuzzer.git@bea408a6e01f0f7e6c82a43121fe3af4506c932e',
   'src/third_party/fuzztest/src':
-    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@54dfec04d5c9ad1f22b08002ab6a5e2d0de77671',
+    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@54b2ccd59262eaeab887c5798ae6b12b3bb7b04b',
   'src/third_party/libjpeg_turbo':
     'https://chromium.googlesource.com/chromium/deps/libjpeg_turbo.git@6bb85251a8382b5e07f635a981ac685cc5ab5053',
   'src/third_party/libsrtp':
@@ -601,7 +601,7 @@ deps = {
   'src/third_party/dav1d/libdav1d':
     'https://chromium.googlesource.com/external/github.com/videolan/dav1d.git@b546257f770768b2c88258c533da38b91a06f737',
   'src/third_party/libaom/source/libaom':
-    'https://aomedia.googlesource.com/aom.git@557586fde2fdc09dff9c3edf6943d6d54aa8914c',
+    'https://aomedia.googlesource.com/aom.git@4018d3b63456eb657475e66c352bfa86f321e0f5',
   'src/third_party/libgav1/src':
     Var('chromium_git') + '/codecs/libgav1.git' + '@' + '40f58ed32ff39071c3f2a51056dbc49a070af0dc',
   'src/third_party/libunwindstack': {
@@ -609,11 +609,11 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '2074b65d22dd04b65c2688647b9386a63338f0b9',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '4d0f63f0824b488797427c6d9840932886bef221',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
-    'https://chromium.googlesource.com/webm/libvpx.git@e83e25f791932202256479052f18bdd03a091147',
+    'https://chromium.googlesource.com/webm/libvpx.git@a0ac06f5d3c83f890cb502c80470ef0502343fec',
   'src/third_party/libyuv':
     'https://chromium.googlesource.com/libyuv/libyuv.git@917276084a49be726c90292ff0a6b0a3d571a6af',
   'src/third_party/lss': {
@@ -644,7 +644,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/r8',
-              'version': 'kCLXGwoL10sgeyycuTLyDhesWICC0DSVfVf-a6U6_yUC',
+              'version': 'TozMGRBtGJuZp31wJjyY7VkLaQ6SI8lsRnLVsSlmmogC',
           },
       ],
       'condition': 'checkout_android',
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@f8f65faf07b1132c9d01932ef95846e9a82b8d54',
+    'https://chromium.googlesource.com/chromium/src/tools@17573b6d7d84674cc1012af9b31ff30e7c5fff53',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'dJ3xGQgP3gjT2n9aXf2sj4QdckdRZXrVhhrlsHQg8QQC',
+          'version': 'us_8XGxuKdmIoJEbqTnp_Ehw-VEaxJIVTzMy2h0fnhwC',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -718,7 +718,7 @@ deps = {
       'packages': [
           {
                'package': 'chromium/third_party/android_build_tools/manifest_merger',
-               'version': 'YgbGvXYO4RiQ7zH0QiMCxe_f9hE2k9JW8lUOKTkYC70C',
+               'version': 'xxaa38R0ED95Wxo8P29BC-mB7bjFPi2HKMpiIJn-UhAC',
           },
       ],
       'condition': 'checkout_android',
@@ -792,7 +792,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/turbine',
-              'version': 'j49Y1F_PAMGZdDf53rGaEzVXdT6jXWDhEm7UMfGhjswC',
+              'version': 'BMHNhxMhr7uGz1rh_Od_JE4kAdP9K5MXr6GN2R9tQkAC',
           },
       ],
       'condition': 'checkout_android',
@@ -842,7 +842,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/android_deps/autorolled',
-              'version': 'LStkoCV2NKgZ1hqhnILaH4soxZtZa2-8RG67-JxAVUsC',
+              'version': 'RiKmV-EpMhg5ibpMbPpENQmfU4IPLpo8cxer4F3NSUgC',
           },
       ],
       'condition': 'checkout_android and non_git_source',
@@ -864,7 +864,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '84818a41e074779dbb00521a4731d3e14160ff15',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + 'afb43805349cf1cbec0083d94256bb8f72cbc53b',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '23fcc1c6c9726ef212701971103b18cd1936ae9b',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',
@@ -1014,17 +1014,6 @@ deps = {
       'dep_type': 'cipd',
   },
 
-  'src/third_party/android_deps/cipd/libs/org_jetbrains_kotlinx_kotlinx_coroutines_guava': {
-      'packages': [
-          {
-              'package': 'chromium/third_party/android_deps/libs/org_jetbrains_kotlinx_kotlinx_coroutines_guava',
-              'version': 'version:2@1.8.1.cr2',
-          },
-      ],
-      'condition': 'checkout_android and non_git_source',
-      'dep_type': 'cipd',
-  },
-
   'src/third_party/android_deps/cipd/libs/org_jsoup_jsoup': {
       'packages': [
           {

From b9336fda1a1e4e9e85dcac34baf847c6332a03c8 Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Tue, 10 Feb 2026 20:03:07 -0800
Subject: [PATCH 010/472] Update WebRTC code version (2026-02-11T04:02:53).

Bug: None
Change-Id: Ie21c0a01a85724acd85353609b4b8318b7873b81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/448200
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#46863}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index 2d297f3f41..2e5114a394 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-02-10T04:08:07";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-02-11T04:02:53";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From 43845d364dd59f943ccdf53f9fc0a1548eb93c18 Mon Sep 17 00:00:00 2001
From: Victor Boivie <boivie@webrtc.org>
Date: Tue, 10 Feb 2026 13:32:17 +0100
Subject: [PATCH 011/472] dcsctp: Correct Fast Recovery retransmission logic

This commit fixes a logic error in OutstandingData that prevented the
correct application of Fast Recovery retransmission rules when the
Cumulative TSN Ack point advanced.

Previously, HandleSack would call RemoveAcked before
NackBetweenAckBlocks. Because RemoveAcked updates the internal
last_cumulative_tsn_ack_ to match the incoming SACK, the subsequent
check (cumulative_tsn_ack > last_cumulative_tsn_ack_) in
NackBetweenAckBlocks would always be false. This resulted in miss
indications not being incremented for all missing TSNs as required
during Fast Recovery.

Per RFC 9260 Section 7.2.4: "If an endpoint is in Fast Recovery and
a SACK chunks arrives that advances the Cumulative TSN Ack Point, the
miss indications are incremented for all TSNs reported missing in the
SACK chunk."

The fix involves determining if the cumulative ACK point is advancing
before the internal state is modified in HandleSack. A new unit test
FastRecoveryIncrementsNackCountWhenCumulativeTsnAdvances verifies
that missing TSNs correctly receive miss indications in this scenario.

No-Try: True
Bug: None
Change-Id: Ic8ef7026bfb3617bd2094bae47de13b9d0133930
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/448080
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46864}
---
 net/dcsctp/tx/outstanding_data.cc      |  8 +++-
 net/dcsctp/tx/outstanding_data.h       |  1 +
 net/dcsctp/tx/outstanding_data_test.cc | 56 ++++++++++++++++++++++++++
 3 files changed, 63 insertions(+), 2 deletions(-)

diff --git a/net/dcsctp/tx/outstanding_data.cc b/net/dcsctp/tx/outstanding_data.cc
index 65b31b4cc2..37ba07b822 100644
--- a/net/dcsctp/tx/outstanding_data.cc
+++ b/net/dcsctp/tx/outstanding_data.cc
@@ -143,6 +143,9 @@ OutstandingData::AckInfo OutstandingData::HandleSack(
     UnwrappedTSN cumulative_tsn_ack,
     webrtc::ArrayView<const SackChunk::GapAckBlock> gap_ack_blocks,
     bool is_in_fast_recovery) {
+  bool cumulative_tsn_ack_advanced =
+      cumulative_tsn_ack > last_cumulative_tsn_ack_;
+
   OutstandingData::AckInfo ack_info(cumulative_tsn_ack);
   // Erase all items up to cumulative_tsn_ack.
   RemoveAcked(cumulative_tsn_ack, ack_info);
@@ -152,7 +155,7 @@ OutstandingData::AckInfo OutstandingData::HandleSack(
 
   // NACK and possibly mark for retransmit chunks that weren't acked.
   NackBetweenAckBlocks(cumulative_tsn_ack, gap_ack_blocks, is_in_fast_recovery,
-                       ack_info);
+                       cumulative_tsn_ack_advanced, ack_info);
 
   RTC_DCHECK(IsConsistent());
   return ack_info;
@@ -225,6 +228,7 @@ void OutstandingData::NackBetweenAckBlocks(
     UnwrappedTSN cumulative_tsn_ack,
     webrtc::ArrayView<const SackChunk::GapAckBlock> gap_ack_blocks,
     bool is_in_fast_recovery,
+    bool cumulative_tsn_acked_advanced,
     OutstandingData::AckInfo& ack_info) {
   // Mark everything between the blocks as NACKED/TO_BE_RETRANSMITTED.
   // https://tools.ietf.org/html/rfc4960#section-7.2.4
@@ -237,7 +241,7 @@ void OutstandingData::NackBetweenAckBlocks(
   // in-flight and between gaps should be nacked. This means that SCTP relies on
   // the T3-RTX-timer to re-send packets otherwise.
   UnwrappedTSN max_tsn_to_nack = ack_info.highest_tsn_acked;
-  if (is_in_fast_recovery && cumulative_tsn_ack > last_cumulative_tsn_ack_) {
+  if (is_in_fast_recovery && cumulative_tsn_acked_advanced) {
     // https://tools.ietf.org/html/rfc4960#section-7.2.4
     // "If an endpoint is in Fast Recovery and a SACK arrives that advances
     // the Cumulative TSN Ack Point, the miss indications are incremented for
diff --git a/net/dcsctp/tx/outstanding_data.h b/net/dcsctp/tx/outstanding_data.h
index d8bc926ea9..81fb4c27af 100644
--- a/net/dcsctp/tx/outstanding_data.h
+++ b/net/dcsctp/tx/outstanding_data.h
@@ -327,6 +327,7 @@ class OutstandingData {
       UnwrappedTSN cumulative_tsn_ack,
       webrtc::ArrayView<const SackChunk::GapAckBlock> gap_ack_blocks,
       bool is_in_fast_recovery,
+      bool cumulative_tsn_acked_advanced,
       OutstandingData::AckInfo& ack_info);
 
   // Process the acknowledgement of the chunk referenced by `iter` and updates
diff --git a/net/dcsctp/tx/outstanding_data_test.cc b/net/dcsctp/tx/outstanding_data_test.cc
index 9a3b352ead..f56f1d5499 100644
--- a/net/dcsctp/tx/outstanding_data_test.cc
+++ b/net/dcsctp/tx/outstanding_data_test.cc
@@ -676,5 +676,61 @@ TEST_F(OutstandingDataTest, TreatsUnackedPayloadBytesDifferentFromPacketBytes) {
   EXPECT_EQ(buf_.unacked_items(), 2u);
 }
 
+TEST_F(OutstandingDataTest,
+       FastRecoveryIncrementsNackCountWhenCumulativeTsnAdvances) {
+  // This test verifies that the Fast Recovery retransmission rules are
+  // correctly applied when the Cumulative TSN Ack point advances. RFC 9260
+  // Section 7.2.4: "If an endpoint is in Fast Recovery and a SACK arrives that
+  // advances the Cumulative TSN Ack Point, the miss indications are incremented
+  // for all TSNs reported missing in the SACK."
+
+  for (int i = 10; i <= 16; ++i) {
+    buf_.Insert(kMessageId, gen_.Ordered({1}, ""), kNow);
+  }
+
+  // SACK 1: Cumulative Ack = 10. Gap blocks for 12, 14, 16.
+  // Missing: 11, 13, 15.
+  // This marks 12, 14, 16 as Acked.
+  // TSNs 11, 13, 15 get their 1st miss indication each.
+  std::vector<SackChunk::GapAckBlock> gab1 = {
+      SackChunk::GapAckBlock(2, 2),  // TSN 12
+      SackChunk::GapAckBlock(4, 4),  // TSN 14
+      SackChunk::GapAckBlock(6, 6)   // TSN 16
+  };
+  buf_.HandleSack(unwrapper_.Unwrap(TSN(10)), gab1,
+                  /*is_in_fast_recovery=*/false);
+
+  // SACK 2: Cumulative Ack advances to 11. Same gap blocks (12, 14, 16).
+  // Endpoint is now in Fast Recovery (is_in_fast_recovery = true). Because the
+  // Cumulative TSN Ack Point advanced from 10 to 11, 13 and 15 should get their
+  // 2nd miss indication.
+  std::vector<SackChunk::GapAckBlock> gab2 = {
+      SackChunk::GapAckBlock(1, 1),  // TSN 12
+      SackChunk::GapAckBlock(3, 3),  // TSN 14
+      SackChunk::GapAckBlock(5, 5)   // TSN 16
+  };
+  buf_.HandleSack(unwrapper_.Unwrap(TSN(11)), gab2,
+                  /*is_in_fast_recovery=*/true);
+
+  // SACK 3: Cumulative Ack advances to 12.
+  // Note: TSN 12 was already acked via gap block, so this just advances the
+  // Cumulative Ack. 13 and 15 should get their 3rd miss indication and trigger
+  // retransmission.
+  std::vector<SackChunk::GapAckBlock> gab3 = {
+      SackChunk::GapAckBlock(2, 2),  // TSN 14
+      SackChunk::GapAckBlock(4, 4)   // TSN 16
+  };
+  buf_.HandleSack(unwrapper_.Unwrap(TSN(12)), gab3,
+                  /*is_in_fast_recovery=*/true);
+
+  // 13 and 15 should now be retransmitted.
+  EXPECT_THAT(buf_.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(12), State::kAcked),
+                          Pair(TSN(13), State::kToBeRetransmitted),
+                          Pair(TSN(14), State::kAcked),
+                          Pair(TSN(15), State::kToBeRetransmitted),
+                          Pair(TSN(16), State::kAcked)));
+}
+
 }  // namespace
 }  // namespace dcsctp

From 90c4b85e77641334829b582f89bfcc9aca21beae Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Tue, 10 Feb 2026 18:23:16 +0100
Subject: [PATCH 012/472] Batch SetupMediaChannel blocking calls in rtp
 receiver classes.

The current implementation of media channel setup triggers individual
blocking calls to the worker thread for every transceiver during state
updates. When applying remote descriptions with multiple transceivers,
this results in frequent thread context switching and increased
signaling thread latency.

Modify RtpReceiverInternal and its implementations to return a deferred
task using absl::AnyInvocable instead of executing the setup
immediately. This allows the SdpOfferAnswerHandler to collect setup
requirements across all transceivers and execute them in a single
batched blocking call on the worker thread.

Key modifications:
- Update SetupMediaChannel and SetupUnsignaledMediaChannel to return
  a callable task.
- Refactor AudioRtpReceiver and VideoRtpReceiver to package their
  worker-thread logic into returnable objects.
- Implement task collection and batch execution logic within the
  SdpOfferAnswerHandler.
- Update associated mocks and unit tests to accommodate the new
  asynchronous execution pattern.

(using No-try because of `webrtc_linux_chromium` timeout issues)

Bug: webrtc:42222804
No-try: true
Change-Id: I4b4d114e3f750fd41eaa4ed29e5c8ee6976eca39
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/447063
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46865}
---
 pc/BUILD.gn                          |  3 +++
 pc/audio_rtp_receiver.cc             | 25 ++++++++++--------
 pc/audio_rtp_receiver.h              | 22 ++++++++--------
 pc/audio_rtp_receiver_unittest.cc    | 13 ++++++----
 pc/rtp_receiver.h                    | 13 ++++++++--
 pc/rtp_sender_receiver_unittest.cc   | 14 +++++++---
 pc/rtp_transmission_manager.cc       | 13 +++++-----
 pc/sdp_offer_answer.cc               | 17 +++++++++++--
 pc/test/mock_rtp_receiver_internal.h | 11 ++++++--
 pc/video_rtp_receiver.cc             | 38 +++++++++++++++-------------
 pc/video_rtp_receiver.h              | 24 ++++++++++--------
 pc/video_rtp_receiver_unittest.cc    |  8 ++++--
 12 files changed, 131 insertions(+), 70 deletions(-)

diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 9c802f75bb..3b692fcdc3 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -1808,6 +1808,7 @@ rtc_library("rtp_receiver") {
     "../api:scoped_refptr",
     "../media:media_channel",
     "../rtc_base:threading",
+    "//third_party/abseil-cpp/absl/functional:any_invocable",
   ]
 }
 
@@ -1839,6 +1840,7 @@ rtc_library("audio_rtp_receiver") {
     "../rtc_base:macromagic",
     "../rtc_base:threading",
     "../rtc_base/system:no_unique_address",
+    "//third_party/abseil-cpp/absl/functional:any_invocable",
     "//third_party/abseil-cpp/absl/strings:string_view",
   ]
 }
@@ -1873,6 +1875,7 @@ rtc_library("video_rtp_receiver") {
     "../rtc_base:macromagic",
     "../rtc_base:threading",
     "../rtc_base/system:no_unique_address",
+    "//third_party/abseil-cpp/absl/functional:any_invocable",
     "//third_party/abseil-cpp/absl/strings:string_view",
   ]
 }
diff --git a/pc/audio_rtp_receiver.cc b/pc/audio_rtp_receiver.cc
index 2d7b03cf4f..e1f5acb032 100644
--- a/pc/audio_rtp_receiver.cc
+++ b/pc/audio_rtp_receiver.cc
@@ -17,6 +17,7 @@
 #include <utility>
 #include <vector>
 
+#include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
 #include "api/crypto/frame_decryptor_interface.h"
 #include "api/dtls_transport_interface.h"
@@ -200,18 +201,20 @@ void AudioRtpReceiver::Stop() {
   track_->internal()->set_ended();
 }
 
-void AudioRtpReceiver::RestartMediaChannel(std::optional<uint32_t> ssrc) {
+absl::AnyInvocable<void() &&>
+AudioRtpReceiver::GetRestartFunctionForMediaChannel(
+    std::optional<uint32_t> ssrc) {
   RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
   bool enabled = track_->internal()->enabled();
   MediaSourceInterface::SourceState state = source_->state();
-  worker_thread_->BlockingCall([&]() {
-    RTC_DCHECK_RUN_ON(worker_thread_);
-    RestartMediaChannel_w(std::move(ssrc), enabled, state);
-  });
   source_->SetState(MediaSourceInterface::kLive);
+  return [this, ssrc = std::move(ssrc), enabled, state]() mutable {
+    RTC_DCHECK_RUN_ON(worker_thread_);
+    GetRestartFunctionForMediaChannel_w(std::move(ssrc), enabled, state);
+  };
 }
 
-void AudioRtpReceiver::RestartMediaChannel_w(
+void AudioRtpReceiver::GetRestartFunctionForMediaChannel_w(
     std::optional<uint32_t> ssrc,
     bool track_enabled,
     MediaSourceInterface::SourceState state) {
@@ -240,14 +243,16 @@ void AudioRtpReceiver::RestartMediaChannel_w(
   Reconfigure(track_enabled);
 }
 
-void AudioRtpReceiver::SetupMediaChannel(uint32_t ssrc) {
+absl::AnyInvocable<void() &&> AudioRtpReceiver::GetSetupForMediaChannel(
+    uint32_t ssrc) {
   RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
-  RestartMediaChannel(ssrc);
+  return GetRestartFunctionForMediaChannel(ssrc);
 }
 
-void AudioRtpReceiver::SetupUnsignaledMediaChannel() {
+absl::AnyInvocable<void() &&>
+AudioRtpReceiver::GetSetupForUnsignaledMediaChannel() {
   RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
-  RestartMediaChannel(std::nullopt);
+  return GetRestartFunctionForMediaChannel(std::nullopt);
 }
 
 std::optional<uint32_t> AudioRtpReceiver::ssrc() const {
diff --git a/pc/audio_rtp_receiver.h b/pc/audio_rtp_receiver.h
index 1324a10930..70247478a4 100644
--- a/pc/audio_rtp_receiver.h
+++ b/pc/audio_rtp_receiver.h
@@ -17,6 +17,7 @@
 #include <string>
 #include <vector>
 
+#include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
 #include "api/crypto/frame_decryptor_interface.h"
 #include "api/dtls_transport_interface.h"
@@ -49,8 +50,9 @@ class AudioRtpReceiver : public ObserverInterface,
   // instance at construction time without having to call `SetMediaChannel()`
   // on the worker thread straight after construction.
   // However, when using that, the assumption is that right after construction,
-  // a call to either `SetupUnsignaledMediaChannel` or `SetupMediaChannel`
-  // will be made, which will internally start the source on the worker thread.
+  // a call to either `GetSetupForUnsignaledMediaChannel` or
+  // `GetSetupForMediaChannel` will be made, which will internally start the
+  // source on the worker thread.
   AudioRtpReceiver(Thread* worker_thread,
                    absl::string_view receiver_id,
                    std::vector<std::string> stream_ids,
@@ -106,8 +108,8 @@ class AudioRtpReceiver : public ObserverInterface,
 
   // RtpReceiverInternal implementation.
   void Stop() override;
-  void SetupMediaChannel(uint32_t ssrc) override;
-  void SetupUnsignaledMediaChannel() override;
+  absl::AnyInvocable<void() &&> GetSetupForMediaChannel(uint32_t ssrc) override;
+  absl::AnyInvocable<void() &&> GetSetupForUnsignaledMediaChannel() override;
   std::optional<uint32_t> ssrc() const override;
   void NotifyFirstPacketReceived() override;
   void NotifyFirstPacketReceivedAfterReceptiveChange() override;
@@ -136,12 +138,12 @@ class AudioRtpReceiver : public ObserverInterface,
       VoiceMediaReceiveChannelInterface* media_channel,
       RemoteAudioSource::OnAudioChannelGoneAction source_gone_action);
 
-  void RestartMediaChannel(std::optional<uint32_t> ssrc)
-      RTC_RUN_ON(&signaling_thread_checker_);
-  void RestartMediaChannel_w(std::optional<uint32_t> ssrc,
-                             bool track_enabled,
-                             MediaSourceInterface::SourceState state)
-      RTC_RUN_ON(worker_thread_);
+  absl::AnyInvocable<void() &&> GetRestartFunctionForMediaChannel(
+      std::optional<uint32_t> ssrc) RTC_RUN_ON(&signaling_thread_checker_);
+  void GetRestartFunctionForMediaChannel_w(
+      std::optional<uint32_t> ssrc,
+      bool track_enabled,
+      MediaSourceInterface::SourceState state) RTC_RUN_ON(worker_thread_);
   void Reconfigure(bool track_enabled) RTC_RUN_ON(worker_thread_);
   void SetOutputVolume_w(double volume) RTC_RUN_ON(worker_thread_);
 
diff --git a/pc/audio_rtp_receiver_unittest.cc b/pc/audio_rtp_receiver_unittest.cc
index 0471003fae..a1c5e37ce7 100644
--- a/pc/audio_rtp_receiver_unittest.cc
+++ b/pc/audio_rtp_receiver_unittest.cc
@@ -13,6 +13,7 @@
 #include <atomic>
 #include <cstdint>
 #include <string>
+#include <utility>
 #include <vector>
 
 #include "api/make_ref_counted.h"
@@ -73,7 +74,8 @@ TEST_F(AudioRtpReceiverTest, SetOutputVolumeIsCalled) {
   receiver_->track()->set_enabled(true);
   receiver_->SetMediaChannel(&receive_channel_);
   EXPECT_CALL(receive_channel_, SetDefaultRawAudioSink(_)).Times(0);
-  receiver_->SetupMediaChannel(kSsrc);
+  auto setup_task = receiver_->GetSetupForMediaChannel(kSsrc);
+  std::move(setup_task)();
 
   EXPECT_CALL(receive_channel_, SetOutputVolume(kSsrc, kVolume))
       .WillOnce(InvokeWithoutArgs([&] {
@@ -96,10 +98,11 @@ TEST_F(AudioRtpReceiverTest, VolumesSetBeforeStartingAreRespected) {
   receiver_->SetMediaChannel(&receive_channel_);
 
   // The previosly set initial volume should be propagated to the provided
-  // media_channel_ as soon as SetupMediaChannel is called.
+  // media_channel_ as soon as GetSetupForMediaChannel is called.
   EXPECT_CALL(receive_channel_, SetOutputVolume(kSsrc, kVolume));
 
-  receiver_->SetupMediaChannel(kSsrc);
+  auto setup_task = receiver_->GetSetupForMediaChannel(kSsrc);
+  std::move(setup_task)();
 }
 
 // Tests that OnChanged notifications are processed correctly on the worker
@@ -114,8 +117,8 @@ TEST(AudioRtpReceiver, OnChangedNotificationsAfterConstruction) {
 
   EXPECT_CALL(receive_channel, SetDefaultRawAudioSink(_)).Times(1);
   EXPECT_CALL(receive_channel, SetDefaultOutputVolume(kDefaultVolume)).Times(1);
-  receiver->SetupUnsignaledMediaChannel();
-  loop.Flush();
+  auto setup_task = receiver->GetSetupForUnsignaledMediaChannel();
+  std::move(setup_task)();
 
   // Mark the track as disabled.
   receiver->track()->set_enabled(false);
diff --git a/pc/rtp_receiver.h b/pc/rtp_receiver.h
index 6772016739..3ef0274811 100644
--- a/pc/rtp_receiver.h
+++ b/pc/rtp_receiver.h
@@ -21,6 +21,7 @@
 #include <string>
 #include <vector>
 
+#include "absl/functional/any_invocable.h"
 #include "api/dtls_transport_interface.h"
 #include "api/media_stream_interface.h"
 #include "api/rtp_receiver_interface.h"
@@ -48,11 +49,19 @@ class RtpReceiverInternal : public RtpReceiverInterface {
 
   // Configures the RtpReceiver with the underlying media channel, with the
   // given SSRC as the stream identifier.
-  virtual void SetupMediaChannel(uint32_t ssrc) = 0;
+  // This function returns a callable object that must be invoked on the worker
+  // thread by the caller to complete the operation. This is done to allow the
+  // caller to batch up tasks for the worker thread.
+  [[nodiscard]] virtual absl::AnyInvocable<void() &&> GetSetupForMediaChannel(
+      uint32_t ssrc) = 0;
 
   // Configures the RtpReceiver with the underlying media channel to receive an
   // unsignaled receive stream.
-  virtual void SetupUnsignaledMediaChannel() = 0;
+  // This function returns a callable object that must be invoked on the worker
+  // thread by the caller to complete the operation. This is done to allow the
+  // caller to batch up tasks for the worker thread.
+  [[nodiscard]] virtual absl::AnyInvocable<void() &&>
+  GetSetupForUnsignaledMediaChannel() = 0;
 
   virtual void set_transport(
       scoped_refptr<DtlsTransportInterface> dtls_transport) = 0;
diff --git a/pc/rtp_sender_receiver_unittest.cc b/pc/rtp_sender_receiver_unittest.cc
index 31a29a4e8e..f900787d8c 100644
--- a/pc/rtp_sender_receiver_unittest.cc
+++ b/pc/rtp_sender_receiver_unittest.cc
@@ -18,6 +18,7 @@
 #include <vector>
 
 #include "absl/algorithm/container.h"
+#include "absl/functional/any_invocable.h"
 #include "api/audio_options.h"
 #include "api/crypto/crypto_options.h"
 #include "api/crypto/frame_decryptor_interface.h"
@@ -315,7 +316,9 @@ class RtpSenderReceiverTest
     worker_thread_->BlockingCall([this] {
       audio_rtp_receiver_->SetMediaChannel(voice_media_receive_channel());
     });
-    audio_rtp_receiver_->SetupMediaChannel(kAudioSsrc);
+    auto setup_task = audio_rtp_receiver_->GetSetupForMediaChannel(kAudioSsrc);
+    worker_thread_->BlockingCall(
+        [task = std::move(setup_task)]() mutable { std::move(task)(); });
     audio_track_ = audio_rtp_receiver_->audio_track();
     VerifyVoiceChannelOutput();
   }
@@ -327,7 +330,9 @@ class RtpSenderReceiverTest
     worker_thread_->BlockingCall([this] {
       video_rtp_receiver_->SetMediaChannel(video_media_receive_channel());
     });
-    video_rtp_receiver_->SetupMediaChannel(kVideoSsrc);
+    auto setup_task = video_rtp_receiver_->GetSetupForMediaChannel(kVideoSsrc);
+    worker_thread_->BlockingCall(
+        [task = std::move(setup_task)]() mutable { std::move(task)(); });
     video_track_ = video_rtp_receiver_->video_track();
     VerifyVideoChannelOutput();
   }
@@ -348,7 +353,10 @@ class RtpSenderReceiverTest
     worker_thread_->BlockingCall([this] {
       video_rtp_receiver_->SetMediaChannel(video_media_receive_channel());
     });
-    video_rtp_receiver_->SetupMediaChannel(primary_ssrc);
+    auto setup_task =
+        video_rtp_receiver_->GetSetupForMediaChannel(primary_ssrc);
+    worker_thread_->BlockingCall(
+        [task = std::move(setup_task)]() mutable { std::move(task)(); });
     video_track_ = video_rtp_receiver_->video_track();
   }
 
diff --git a/pc/rtp_transmission_manager.cc b/pc/rtp_transmission_manager.cc
index 4cbf60910c..25f1360c34 100644
--- a/pc/rtp_transmission_manager.cc
+++ b/pc/rtp_transmission_manager.cc
@@ -471,11 +471,11 @@ PLAN_B_ONLY void RtpTransmissionManager::CreateAudioReceiverPlanB(
   auto audio_receiver = make_ref_counted<AudioRtpReceiver>(
       worker_thread(), remote_sender_info.sender_id, streams, false,
       voice_media_receive_channel());
-  if (remote_sender_info.sender_id == kDefaultAudioSenderId) {
-    audio_receiver->SetupUnsignaledMediaChannel();
-  } else {
-    audio_receiver->SetupMediaChannel(remote_sender_info.first_ssrc);
-  }
+  auto task = (remote_sender_info.sender_id == kDefaultAudioSenderId)
+                  ? audio_receiver->GetSetupForUnsignaledMediaChannel()
+                  : audio_receiver->GetSetupForMediaChannel(
+                        remote_sender_info.first_ssrc);
+  worker_thread()->BlockingCall([&]() mutable { std::move(task)(); });
 
   auto receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
       signaling_thread(), worker_thread(), std::move(audio_receiver));
@@ -497,11 +497,12 @@ PLAN_B_ONLY void RtpTransmissionManager::CreateVideoReceiverPlanB(
   auto video_receiver = make_ref_counted<VideoRtpReceiver>(
       worker_thread(), remote_sender_info.sender_id, streams);
 
-  video_receiver->SetupMediaChannel(
+  auto task = video_receiver->GetSetupForMediaChannel(
       remote_sender_info.sender_id == kDefaultVideoSenderId
           ? std::nullopt
           : std::optional<uint32_t>(remote_sender_info.first_ssrc),
       video_media_receive_channel());
+  worker_thread()->BlockingCall([&]() mutable { std::move(task)(); });
 
   auto receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
       signaling_thread(), worker_thread(), std::move(video_receiver));
diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc
index 2931698178..5242ec13d5 100644
--- a/pc/sdp_offer_answer.cc
+++ b/pc/sdp_offer_answer.cc
@@ -2349,6 +2349,7 @@ void SdpOfferAnswerHandler::ApplyRemoteDescriptionUpdateTransceiverState(
   std::vector<scoped_refptr<RtpTransceiverInterface>> remove_list;
   std::vector<scoped_refptr<MediaStreamInterface>> added_streams;
   std::vector<scoped_refptr<MediaStreamInterface>> removed_streams;
+  std::vector<absl::AnyInvocable<void() &&>> worker_tasks;
   flat_map<std::string, DtlsTransportAndName> dtls_transports_by_mid =
       GetDtlsTransports(*transceivers(), context_->network_thread(),
                         transport_controller_s());
@@ -2445,12 +2446,24 @@ void SdpOfferAnswerHandler::ApplyRemoteDescriptionUpdateTransceiverState(
       if (!media_desc->streams().empty() &&
           media_desc->streams()[0].has_ssrcs()) {
         uint32_t ssrc = media_desc->streams()[0].first_ssrc();
-        transceiver->receiver_internal()->SetupMediaChannel(ssrc);
+        worker_tasks.push_back(
+            transceiver->receiver_internal()->GetSetupForMediaChannel(ssrc));
       } else {
-        transceiver->receiver_internal()->SetupUnsignaledMediaChannel();
+        worker_tasks.push_back(transceiver->receiver_internal()
+                                   ->GetSetupForUnsignaledMediaChannel());
       }
     }
   }
+
+  if (!worker_tasks.empty()) {
+    context_->worker_thread()->BlockingCall([&worker_tasks] {
+      for (auto& task : worker_tasks) {
+        if (task) {
+          std::move(task)();
+        }
+      }
+    });
+  }
   // Once all processing has finished, fire off callbacks.
   pc_->RunWithObserver([&](auto observer) {
     for (const auto& transceiver : now_receiving_transceivers) {
diff --git a/pc/test/mock_rtp_receiver_internal.h b/pc/test/mock_rtp_receiver_internal.h
index baebd44361..8d3bc701dd 100644
--- a/pc/test/mock_rtp_receiver_internal.h
+++ b/pc/test/mock_rtp_receiver_internal.h
@@ -16,6 +16,7 @@
 #include <string>
 #include <vector>
 
+#include "absl/functional/any_invocable.h"
 #include "api/crypto/frame_decryptor_interface.h"
 #include "api/dtls_transport_interface.h"
 #include "api/media_stream_interface.h"
@@ -71,8 +72,14 @@ class MockRtpReceiverInternal : public RtpReceiverInternal {
               SetMediaChannel,
               (webrtc::MediaReceiveChannelInterface*),
               (override));
-  MOCK_METHOD(void, SetupMediaChannel, (uint32_t), (override));
-  MOCK_METHOD(void, SetupUnsignaledMediaChannel, (), (override));
+  MOCK_METHOD(absl::AnyInvocable<void() &&>,
+              GetSetupForMediaChannel,
+              (uint32_t),
+              (override));
+  MOCK_METHOD(absl::AnyInvocable<void() &&>,
+              GetSetupForUnsignaledMediaChannel,
+              (),
+              (override));
   MOCK_METHOD(std::optional<uint32_t>, ssrc, (), (const, override));
   MOCK_METHOD(void, NotifyFirstPacketReceived, (), (override));
   MOCK_METHOD(void,
diff --git a/pc/video_rtp_receiver.cc b/pc/video_rtp_receiver.cc
index 622efd006a..f3e4f9db7f 100644
--- a/pc/video_rtp_receiver.cc
+++ b/pc/video_rtp_receiver.cc
@@ -17,6 +17,7 @@
 #include <utility>
 #include <vector>
 
+#include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
 #include "api/crypto/frame_decryptor_interface.h"
 #include "api/dtls_transport_interface.h"
@@ -136,18 +137,19 @@ void VideoRtpReceiver::Stop() {
   track_->internal()->set_ended();
 }
 
-void VideoRtpReceiver::RestartMediaChannel(std::optional<uint32_t> ssrc) {
+absl::AnyInvocable<void() &&>
+VideoRtpReceiver::GetRestartFunctionForMediaChannel(
+    std::optional<uint32_t> ssrc) {
   RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
   MediaSourceInterface::SourceState state = source_->state();
-  // TODO(tommi): Can we restart the media channel without blocking?
-  worker_thread_->BlockingCall([&] {
-    RTC_DCHECK_RUN_ON(worker_thread_);
-    RestartMediaChannel_w(std::move(ssrc), state);
-  });
   source_->SetState(MediaSourceInterface::kLive);
+  return [this, ssrc = std::move(ssrc), state]() mutable {
+    RTC_DCHECK_RUN_ON(worker_thread_);
+    GetRestartFunctionForMediaChannel_w(std::move(ssrc), state);
+  };
 }
 
-void VideoRtpReceiver::RestartMediaChannel_w(
+void VideoRtpReceiver::GetRestartFunctionForMediaChannel_w(
     std::optional<uint32_t> ssrc,
     MediaSourceInterface::SourceState state) {
   RTC_DCHECK_RUN_ON(worker_thread_);
@@ -199,14 +201,16 @@ void VideoRtpReceiver::SetSink(VideoSinkInterface<VideoFrame>* sink) {
   }
 }
 
-void VideoRtpReceiver::SetupMediaChannel(uint32_t ssrc) {
+absl::AnyInvocable<void() &&> VideoRtpReceiver::GetSetupForMediaChannel(
+    uint32_t ssrc) {
   RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
-  RestartMediaChannel(ssrc);
+  return GetRestartFunctionForMediaChannel(ssrc);
 }
 
-void VideoRtpReceiver::SetupUnsignaledMediaChannel() {
+absl::AnyInvocable<void() &&>
+VideoRtpReceiver::GetSetupForUnsignaledMediaChannel() {
   RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
-  RestartMediaChannel(std::nullopt);
+  return GetRestartFunctionForMediaChannel(std::nullopt);
 }
 
 std::optional<uint32_t> VideoRtpReceiver::ssrc() const {
@@ -354,18 +358,18 @@ std::vector<RtpSource> VideoRtpReceiver::GetSources() const {
   return media_channel_->GetSources(current_ssrc.value());
 }
 
-void VideoRtpReceiver::SetupMediaChannel(
+absl::AnyInvocable<void() &&> VideoRtpReceiver::GetSetupForMediaChannel(
     std::optional<uint32_t> ssrc,
-    MediaReceiveChannelInterface* media_channel) {
+    VideoMediaReceiveChannelInterface* media_channel) {
   RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
   RTC_DCHECK(media_channel);
   MediaSourceInterface::SourceState state = source_->state();
-  worker_thread_->BlockingCall([&] {
+  source_->SetState(MediaSourceInterface::kLive);
+  return [this, ssrc = std::move(ssrc), media_channel, state]() mutable {
     RTC_DCHECK_RUN_ON(worker_thread_);
     SetMediaChannel_w(media_channel);
-    RestartMediaChannel_w(std::move(ssrc), state);
-  });
-  source_->SetState(MediaSourceInterface::kLive);
+    GetRestartFunctionForMediaChannel_w(std::move(ssrc), state);
+  };
 }
 
 void VideoRtpReceiver::OnGenerateKeyFrame() {
diff --git a/pc/video_rtp_receiver.h b/pc/video_rtp_receiver.h
index 444cf9ddfa..6025bed1ca 100644
--- a/pc/video_rtp_receiver.h
+++ b/pc/video_rtp_receiver.h
@@ -17,6 +17,7 @@
 #include <string>
 #include <vector>
 
+#include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
 #include "api/crypto/frame_decryptor_interface.h"
 #include "api/dtls_transport_interface.h"
@@ -87,8 +88,8 @@ class VideoRtpReceiver : public RtpReceiverInternal {
 
   // RtpReceiverInternal implementation.
   void Stop() override;
-  void SetupMediaChannel(uint32_t ssrc) override;
-  void SetupUnsignaledMediaChannel() override;
+  absl::AnyInvocable<void() &&> GetSetupForMediaChannel(uint32_t ssrc) override;
+  absl::AnyInvocable<void() &&> GetSetupForUnsignaledMediaChannel() override;
   std::optional<uint32_t> ssrc() const override;
   void NotifyFirstPacketReceived() override;
   void NotifyFirstPacketReceivedAfterReceptiveChange() override;
@@ -109,17 +110,18 @@ class VideoRtpReceiver : public RtpReceiverInternal {
 
   std::vector<RtpSource> GetSources() const override;
 
-  // Combines SetMediaChannel, SetupMediaChannel and
-  // SetupUnsignaledMediaChannel.
-  void SetupMediaChannel(std::optional<uint32_t> ssrc,
-                         MediaReceiveChannelInterface* media_channel);
+  // Combines SetMediaChannel, GetSetupForMediaChannel and
+  // GetSetupForUnsignaledMediaChannel.
+  absl::AnyInvocable<void() &&> GetSetupForMediaChannel(
+      std::optional<uint32_t> ssrc,
+      VideoMediaReceiveChannelInterface* media_channel);
 
  private:
-  void RestartMediaChannel(std::optional<uint32_t> ssrc)
-      RTC_RUN_ON(&signaling_thread_checker_);
-  void RestartMediaChannel_w(std::optional<uint32_t> ssrc,
-                             MediaSourceInterface::SourceState state)
-      RTC_RUN_ON(worker_thread_);
+  absl::AnyInvocable<void() &&> GetRestartFunctionForMediaChannel(
+      std::optional<uint32_t> ssrc) RTC_RUN_ON(&signaling_thread_checker_);
+  void GetRestartFunctionForMediaChannel_w(
+      std::optional<uint32_t> ssrc,
+      MediaSourceInterface::SourceState state) RTC_RUN_ON(worker_thread_);
   void SetSink(VideoSinkInterface<VideoFrame>* sink) RTC_RUN_ON(worker_thread_);
   void SetMediaChannel_w(MediaReceiveChannelInterface* media_channel)
       RTC_RUN_ON(worker_thread_);
diff --git a/pc/video_rtp_receiver_unittest.cc b/pc/video_rtp_receiver_unittest.cc
index 15b23a6d08..8756c10b76 100644
--- a/pc/video_rtp_receiver_unittest.cc
+++ b/pc/video_rtp_receiver_unittest.cc
@@ -14,6 +14,7 @@
 #include <functional>
 #include <memory>
 #include <string>
+#include <utility>
 #include <vector>
 
 #include "api/make_ref_counted.h"
@@ -205,10 +206,13 @@ TEST_F(VideoRtpReceiverTest, EnablesEncodedOutputOnChannelRestart) {
   MockVideoSink sink;
   Source()->AddEncodedSink(&sink);
   EXPECT_CALL(channel_, SetRecordableEncodedFrameCallback(4711, _));
-  receiver_->SetupMediaChannel(4711);
+  auto setup_media_channel = receiver_->GetSetupForMediaChannel(4711);
+  worker_thread_->BlockingCall(
+      [task = std::move(setup_media_channel)]() mutable { std::move(task)(); });
   EXPECT_CALL(channel_, ClearRecordableEncodedFrameCallback(4711));
   EXPECT_CALL(channel_, SetRecordableEncodedFrameCallback(0, _));
-  receiver_->SetupUnsignaledMediaChannel();
+  auto setup_task = receiver_->GetSetupForUnsignaledMediaChannel();
+  worker_thread_->BlockingCall([&]() mutable { std::move(setup_task)(); });
 }
 
 }  // namespace

From a5e2e7b8d4cf7ee88ba9ad76bf830e5fa0678d95 Mon Sep 17 00:00:00 2001
From: Nfrederiksen <nicholas.frederiksen@eyevinn.se>
Date: Fri, 30 Jan 2026 17:02:07 +0100
Subject: [PATCH 013/472] video_capture: move capture_checker check after
 StopCapture in V4L2 dtor

Before: The V4L2 capture destructor ran
`RTC_CHECK_RUNS_SERIALIZED(&capture_checker_)` before `StopCapture()`,
which can assert while the capture thread is still active.
After: The destructor now calls `StopCapture()` first, then validates
serialized access, with an explanatory comment about the capture
thread lifecycle.

Bug: webrtc:479850459
Change-Id: I147b55bf0f1592586290dcdeba6efc3f1025406a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/444800
Reviewed-by: Jan Grulich <grulja@gmail.com>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Cr-Commit-Position: refs/heads/main@{#46866}
---
 modules/video_capture/linux/video_capture_v4l2.cc | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

diff --git a/modules/video_capture/linux/video_capture_v4l2.cc b/modules/video_capture/linux/video_capture_v4l2.cc
index 1cb83f82d5..3f836efc95 100644
--- a/modules/video_capture/linux/video_capture_v4l2.cc
+++ b/modules/video_capture/linux/video_capture_v4l2.cc
@@ -108,9 +108,10 @@ int32_t VideoCaptureModuleV4L2::Init(const char* deviceUniqueIdUTF8) {
 
 VideoCaptureModuleV4L2::~VideoCaptureModuleV4L2() {
   RTC_DCHECK_RUN_ON(&api_checker_);
-  RTC_CHECK_RUNS_SERIALIZED(&capture_checker_);
 
   StopCapture();
+
+  RTC_CHECK_RUNS_SERIALIZED(&capture_checker_);
   if (_deviceFd != -1)
     close(_deviceFd);
 }

From 964afe58c8ef59fcf4ba0f5243709345305959c7 Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Tue, 10 Feb 2026 18:46:15 +0100
Subject: [PATCH 014/472] Replace ArrayView::subview with subspan in H26x
 parsers

Bug: webrtc:439801349
Change-Id: Iad6e02836cc344a6e8b16896b42adfb37fa63e91
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/448100
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46867}
---
 common_video/h264/h264_bitstream_parser.cc          |  6 +++---
 common_video/h264/pps_parser_unittest.cc            |  2 +-
 common_video/h264/sps_vui_rewriter.cc               |  6 +++---
 common_video/h265/h265_bitstream_parser.cc          | 10 +++++-----
 common_video/h265/h265_bitstream_parser_unittest.cc |  6 ++----
 modules/video_coding/h264_sps_pps_tracker.cc        |  8 ++++----
 modules/video_coding/h26x_packet_buffer.cc          |  4 ++--
 sdk/objc/components/video_codec/nalu_rewriter.cc    |  6 +++---
 8 files changed, 23 insertions(+), 25 deletions(-)

diff --git a/common_video/h264/h264_bitstream_parser.cc b/common_video/h264/h264_bitstream_parser.cc
index 7bc6ab40bf..1a66e5a88a 100644
--- a/common_video/h264/h264_bitstream_parser.cc
+++ b/common_video/h264/h264_bitstream_parser.cc
@@ -318,13 +318,13 @@ void H264BitstreamParser::ParseSlice(ArrayView<const uint8_t> slice) {
   H264::NaluType nalu_type = H264::ParseNaluType(slice[0]);
   switch (nalu_type) {
     case H264::NaluType::kSps: {
-      sps_ = SpsParser::ParseSps(slice.subview(H264::kNaluTypeSize));
+      sps_ = SpsParser::ParseSps(slice.subspan(H264::kNaluTypeSize));
       if (!sps_)
         RTC_DLOG(LS_WARNING) << "Unable to parse SPS from H264 bitstream.";
       break;
     }
     case H264::NaluType::kPps: {
-      pps_ = PpsParser::ParsePps(slice.subview(H264::kNaluTypeSize));
+      pps_ = PpsParser::ParsePps(slice.subspan(H264::kNaluTypeSize));
       if (!pps_)
         RTC_DLOG(LS_WARNING) << "Unable to parse PPS from H264 bitstream.";
       break;
@@ -347,7 +347,7 @@ void H264BitstreamParser::ParseBitstream(ArrayView<const uint8_t> bitstream) {
   std::vector<H264::NaluIndex> nalu_indices = H264::FindNaluIndices(bitstream);
   for (const H264::NaluIndex& index : nalu_indices)
     ParseSlice(
-        bitstream.subview(index.payload_start_offset, index.payload_size));
+        bitstream.subspan(index.payload_start_offset, index.payload_size));
 }
 
 std::optional<int> H264BitstreamParser::GetLastSliceQp() const {
diff --git a/common_video/h264/pps_parser_unittest.cc b/common_video/h264/pps_parser_unittest.cc
index e0c1480152..eb2891c8cb 100644
--- a/common_video/h264/pps_parser_unittest.cc
+++ b/common_video/h264/pps_parser_unittest.cc
@@ -232,7 +232,7 @@ TEST_F(PpsParserTest, ParseSliceHeader) {
     if (nalu_type == H264::NaluType::kIdr) {
       // Skip NAL type header and parse slice header.
       std::optional<PpsParser::SliceHeader> slice_header =
-          PpsParser::ParseSliceHeader(chunk.subview(
+          PpsParser::ParseSliceHeader(chunk.subspan(
               index.payload_start_offset + 1, index.payload_size - 1));
       ASSERT_TRUE(slice_header.has_value());
       EXPECT_EQ(slice_header->first_mb_in_slice, 0u);
diff --git a/common_video/h264/sps_vui_rewriter.cc b/common_video/h264/sps_vui_rewriter.cc
index beb32fa148..4d23071e9c 100644
--- a/common_video/h264/sps_vui_rewriter.cc
+++ b/common_video/h264/sps_vui_rewriter.cc
@@ -233,10 +233,10 @@ Buffer SpsVuiRewriter::ParseOutgoingBitstreamAndRewrite(
 
   for (const H264::NaluIndex& nalu_index : nalus) {
     // Copy NAL unit start code.
-    ArrayView<const uint8_t> start_code = buffer.subview(
+    ArrayView<const uint8_t> start_code = buffer.subspan(
         nalu_index.start_offset,
         nalu_index.payload_start_offset - nalu_index.start_offset);
-    ArrayView<const uint8_t> nalu = buffer.subview(
+    ArrayView<const uint8_t> nalu = buffer.subspan(
         nalu_index.payload_start_offset, nalu_index.payload_size);
     if (nalu.empty()) {
       continue;
@@ -261,7 +261,7 @@ Buffer SpsVuiRewriter::ParseOutgoingBitstreamAndRewrite(
       output_nalu.AppendData(nalu[0]);
 
       ParseResult result =
-          ParseAndRewriteSps(nalu.subview(H264::kNaluTypeSize), &sps,
+          ParseAndRewriteSps(nalu.subspan(H264::kNaluTypeSize), &sps,
                              color_space, &output_nalu, Direction::kOutgoing);
       if (result == ParseResult::kVuiRewritten) {
         output_buffer.AppendData(start_code);
diff --git a/common_video/h265/h265_bitstream_parser.cc b/common_video/h265/h265_bitstream_parser.cc
index 38ffe47db4..23fbbf2f46 100644
--- a/common_video/h265/h265_bitstream_parser.cc
+++ b/common_video/h265/h265_bitstream_parser.cc
@@ -531,7 +531,7 @@ void H265BitstreamParser::ParseSlice(ArrayView<const uint8_t> slice) {
       std::optional<H265VpsParser::VpsState> vps_state;
       if (slice.size() >= H265::kNaluHeaderSize) {
         vps_state =
-            H265VpsParser::ParseVps(slice.subview(H265::kNaluHeaderSize));
+            H265VpsParser::ParseVps(slice.subspan(H265::kNaluHeaderSize));
       }
 
       if (!vps_state) {
@@ -545,7 +545,7 @@ void H265BitstreamParser::ParseSlice(ArrayView<const uint8_t> slice) {
       std::optional<H265SpsParser::SpsState> sps_state;
       if (slice.size() >= H265::kNaluHeaderSize) {
         sps_state =
-            H265SpsParser::ParseSps(slice.subview(H265::kNaluHeaderSize));
+            H265SpsParser::ParseSps(slice.subspan(H265::kNaluHeaderSize));
       }
       if (!sps_state) {
         RTC_LOG(LS_WARNING) << "Unable to parse SPS from H265 bitstream.";
@@ -558,7 +558,7 @@ void H265BitstreamParser::ParseSlice(ArrayView<const uint8_t> slice) {
       std::optional<H265PpsParser::PpsState> pps_state;
       if (slice.size() >= H265::kNaluHeaderSize) {
         std::vector<uint8_t> unpacked_buffer =
-            H265::ParseRbsp(slice.subview(H265::kNaluHeaderSize));
+            H265::ParseRbsp(slice.subspan(H265::kNaluHeaderSize));
         BitstreamReader slice_reader(unpacked_buffer);
         // pic_parameter_set_id: ue(v)
         uint32_t pps_id = slice_reader.ReadExponentialGolomb();
@@ -568,7 +568,7 @@ void H265BitstreamParser::ParseSlice(ArrayView<const uint8_t> slice) {
         IN_RANGE_OR_RETURN_VOID(sps_id, 0, 15);
         const H265SpsParser::SpsState* sps = GetSPS(sps_id);
         pps_state =
-            H265PpsParser::ParsePps(slice.subview(H265::kNaluHeaderSize), sps);
+            H265PpsParser::ParsePps(slice.subspan(H265::kNaluHeaderSize), sps);
       }
       if (!pps_state) {
         RTC_LOG(LS_WARNING) << "Unable to parse PPS from H265 bitstream.";
@@ -639,7 +639,7 @@ void H265BitstreamParser::ParseBitstream(ArrayView<const uint8_t> bitstream) {
   std::vector<H265::NaluIndex> nalu_indices = H265::FindNaluIndices(bitstream);
   for (const H265::NaluIndex& index : nalu_indices)
     ParseSlice(
-        bitstream.subview(index.payload_start_offset, index.payload_size));
+        bitstream.subspan(index.payload_start_offset, index.payload_size));
 }
 
 std::optional<int> H265BitstreamParser::GetLastSliceQp() const {
diff --git a/common_video/h265/h265_bitstream_parser_unittest.cc b/common_video/h265/h265_bitstream_parser_unittest.cc
index 7cfa1eabdd..1045a47ebd 100644
--- a/common_video/h265/h265_bitstream_parser_unittest.cc
+++ b/common_video/h265/h265_bitstream_parser_unittest.cc
@@ -195,10 +195,8 @@ TEST(H265BitstreamParserTest, ReportsFirstSliceSegmentInPicFalse) {
 }
 
 TEST(H265BitstreamParserTest, ReportsFirstSliceSegmentInPicParseInvalidSlice) {
-  ArrayView<const uint8_t> slice_data(kH265SliceChunk);
-  EXPECT_THAT(
-      H265BitstreamParser::IsFirstSliceSegmentInPic(slice_data.subview(50)),
-      Eq(std::nullopt));
+  EXPECT_THAT(H265BitstreamParser::IsFirstSliceSegmentInPic({}),
+              Eq(std::nullopt));
 }
 
 }  // namespace webrtc
diff --git a/modules/video_coding/h264_sps_pps_tracker.cc b/modules/video_coding/h264_sps_pps_tracker.cc
index 94e6411ded..3f76077e3e 100644
--- a/modules/video_coding/h264_sps_pps_tracker.cc
+++ b/modules/video_coding/h264_sps_pps_tracker.cc
@@ -115,7 +115,7 @@ H264SpsPpsTracker::FixedBitstream H264SpsPpsTracker::CopyAndFixBitstream(
   }
 
   if (h264_header.packetization_type == kH264StapA) {
-    ByteBufferReader nalu(bitstream.subview(1));
+    ByteBufferReader nalu(bitstream.subspan(1));
     while (nalu.Length() > 0) {
       required_size += sizeof(start_code_h264);
 
@@ -159,7 +159,7 @@ H264SpsPpsTracker::FixedBitstream H264SpsPpsTracker::CopyAndFixBitstream(
 
   // Copy the rest of the bitstream and insert start codes.
   if (h264_header.packetization_type == kH264StapA) {
-    ByteBufferReader nalu(bitstream.subview(1));
+    ByteBufferReader nalu(bitstream.subspan(1));
     while (nalu.Length() > 0) {
       fixed.bitstream.AppendData(start_code_h264);
 
@@ -206,9 +206,9 @@ void H264SpsPpsTracker::InsertSpsPpsNalus(const std::vector<uint8_t>& sps,
     return;
   }
   std::optional<SpsParser::SpsState> parsed_sps = SpsParser::ParseSps(
-      ArrayView<const uint8_t>(sps).subview(kNaluHeaderOffset));
+      ArrayView<const uint8_t>(sps).subspan(kNaluHeaderOffset));
   std::optional<PpsParser::PpsState> parsed_pps = PpsParser::ParsePps(
-      ArrayView<const uint8_t>(pps).subview(kNaluHeaderOffset));
+      ArrayView<const uint8_t>(pps).subspan(kNaluHeaderOffset));
 
   if (!parsed_sps) {
     RTC_LOG(LS_WARNING) << "Failed to parse SPS.";
diff --git a/modules/video_coding/h26x_packet_buffer.cc b/modules/video_coding/h26x_packet_buffer.cc
index dc141e5a83..10c22f3dc9 100644
--- a/modules/video_coding/h26x_packet_buffer.cc
+++ b/modules/video_coding/h26x_packet_buffer.cc
@@ -361,9 +361,9 @@ void H26xPacketBuffer::InsertSpsPpsNalus(const std::vector<uint8_t>& sps,
     return;
   }
   std::optional<SpsParser::SpsState> parsed_sps = SpsParser::ParseSps(
-      ArrayView<const uint8_t>(sps).subview(kNaluHeaderOffset));
+      ArrayView<const uint8_t>(sps).subspan(kNaluHeaderOffset));
   std::optional<PpsParser::PpsState> parsed_pps = PpsParser::ParsePps(
-      ArrayView<const uint8_t>(pps).subview(kNaluHeaderOffset));
+      ArrayView<const uint8_t>(pps).subspan(kNaluHeaderOffset));
 
   if (!parsed_sps) {
     RTC_LOG(LS_WARNING) << "Failed to parse SPS.";
diff --git a/sdk/objc/components/video_codec/nalu_rewriter.cc b/sdk/objc/components/video_codec/nalu_rewriter.cc
index 236ef32d08..5d73376687 100644
--- a/sdk/objc/components/video_codec/nalu_rewriter.cc
+++ b/sdk/objc/components/video_codec/nalu_rewriter.cc
@@ -275,7 +275,7 @@ bool AnnexBBufferReader::ReadNalu(ArrayView<const uint8_t>& out_nalu) {
     return false;
   }
   out_nalu =
-      buffer_.subview(offset_->payload_start_offset, offset_->payload_size);
+      buffer_.subspan(offset_->payload_start_offset, offset_->payload_size);
   ++offset_;
   return true;
 }
@@ -312,10 +312,10 @@ bool AvccBufferWriter::WriteNalu(ArrayView<const uint8_t> data) {
   // Write length header, which needs to be big endian.
   uint32_t big_endian_length = CFSwapInt32HostToBig(data.size());
   memcpy(buffer_.data(), &big_endian_length, sizeof(big_endian_length));
-  buffer_ = buffer_.subview(sizeof(big_endian_length));
+  buffer_ = buffer_.subspan(sizeof(big_endian_length));
   // Write data.
   memcpy(buffer_.data(), data.data(), data.size());
-  buffer_ = buffer_.subview(data.size());
+  buffer_ = buffer_.subspan(data.size());
   return true;
 }
 

From cb94a77c5d9e4df73f1c4fdbd0a679b909b267f3 Mon Sep 17 00:00:00 2001
From: Rasmus Brandt <brandtr@webrtc.org>
Date: Wed, 11 Feb 2026 12:47:34 +0100
Subject: [PATCH 015/472] Video timing simulator: Improve per-frame margin
 metrics.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This change improves the per-frame margin metrics for the
`RenderingSimulator`:

1) `RenderMargin` is renamed to `AssembledMargin`, to better describe
   the purpose: difference between the target render timestamp and
   the assembly timestamp. This margin can be both positive, zero,
   and negative.
2) A new `RenderedMargin` is added, which is the differerence between
   the target render timestamp and the actual render timestamp. This
   margin should be ==0 most of the time, and <0 whenever an
   underrun happens.

In summary, the new `AssembledMargin` will describe dynamics _before_
the frame buffering, and the new `RenderedMargin` will describe dynamics
_after_ the frame buffering.

Other smaller changes: a) fix bug in render timestamp, b) add total
buffer duration helper, c) add "in time" bool helpers.

Bug: b/423646186
Change-Id: I78e0e2434b4771e35713520a5842f074d3b8c84c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/447840
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46868}
---
 video/timing/simulator/rendering_simulator.h  | 94 ++++++++++++++-----
 .../simulator/rendering_simulator_unittest.cc | 90 ++++++++++++++++++
 2 files changed, 161 insertions(+), 23 deletions(-)

diff --git a/video/timing/simulator/rendering_simulator.h b/video/timing/simulator/rendering_simulator.h
index eb0d05afa4..0624476e5f 100644
--- a/video/timing/simulator/rendering_simulator.h
+++ b/video/timing/simulator/rendering_simulator.h
@@ -26,6 +26,7 @@
 #include "api/units/timestamp.h"
 #include "logging/rtc_event_log/rtc_event_log_parser.h"
 #include "modules/video_coding/timing/timing.h"
+#include "rtc_base/checks.h"
 #include "video/timing/simulator/frame_base.h"
 #include "video/timing/simulator/results_base.h"
 #include "video/timing/simulator/stream_base.h"
@@ -105,6 +106,7 @@ class RenderingSimulator {
     TimeDelta PacketBufferDuration() const {
       return assembled_timestamp - first_packet_arrival_timestamp;
     }
+
     // Time spent waiting to be decoded, after assembly. (This includes a,
     // currently, zero decode duration.) Note that this is similar to
     // `jitter_buffer_delay`, except that the latter is 1) recorded by the
@@ -113,14 +115,25 @@ class RenderingSimulator {
     TimeDelta FrameBufferDuration() const {
       return decoded_timestamp - assembled_timestamp;
     }
+
     // Time spent waiting to be rendered, after decode.
     TimeDelta RenderBufferDuration() const {
       return rendered_timestamp - decoded_timestamp;
     }
-    // Margin between render timestamp (target) and
-    // assembled timestamp (actual):
+
+    // Total duration in all three buffers: from first packet to rendered.
+    TimeDelta TotalBufferDuration() const {
+      TimeDelta total_duration = PacketBufferDuration() +
+                                 FrameBufferDuration() + RenderBufferDuration();
+      RTC_DCHECK_EQ(total_duration,
+                    rendered_timestamp - first_packet_arrival_timestamp);
+      return total_duration;
+    }
+
+    // Pre-buffer margin between render timestamp (target) and
+    // assembled timestamp (actual arrival):
     //   * A frame that is assembled early w.r.t. the target (<=> arriving
-    //     on time from the network) has a positive margin.
+    //     in time from the network) has a positive margin.
     //   * A frame that is assembled on time w.r.t. the target (<=> arriving
     //     slightly late from the network) has zero margin.
     //   * A frame that is assembled late w.r.t. the target (<=> arriving
@@ -130,28 +143,52 @@ class RenderingSimulator {
     // delay. In terms of margin, that means low positive margin values, a
     // couple of frames with zero margin, and very few frames with
     // negative margin.
-    TimeDelta RenderMargin() const {
-      return render_timestamp - assembled_timestamp;
+    // TODO: b/423646186 - Change this to be `DecodabilityMargin`.
+    TimeDelta AssembledMargin() const {
+      // Subtract `kRenderDelay`, since that is what `VCMTiming` and
+      // `VideoRenderFrames` also do.
+      return (render_timestamp - kRenderDelay) - assembled_timestamp;
     }
-    // Split the margin along zero: "excess margin" for positive margins and
-    // "deficit margin" for negative margins.
-    // A jitter buffer would generally want to minimize the number of frames
-    // with a deficit margin (delayed frames/buffer underruns => video freezes),
-    // while also minimizing the stream-level min/p10 of the excess margin
-    // (early frames spending a long time in the buffer => high latency).
-    std::optional<TimeDelta> RenderExcessMargin() const {
-      TimeDelta margin = RenderMargin();
-      if (margin < TimeDelta::Zero()) {
-        return std::nullopt;
-      }
-      return margin;
+    bool AssembledInTime() const {
+      return AssembledMargin() > kInTimeMarginThreshold;
     }
-    std::optional<TimeDelta> RenderDeficitMargin() const {
-      TimeDelta margin = RenderMargin();
-      if (margin > TimeDelta::Zero()) {
-        return std::nullopt;
-      }
-      return margin;
+
+    // Split the assembled margin along zero: "excess margin" for positive
+    // margins and "deficit margin" for negative margins. A jitter buffer would
+    // generally want to minimize the number of frames with a deficit margin
+    // (delayed frames/buffer underruns => video freezes), while also minimizing
+    // the stream-level min/p10 of the excess margin (early frames spending a
+    // long time in the buffer => high latency).
+    std::optional<TimeDelta> AssembledMarginExcess() const {
+      return AssembledInTime() ? std::optional<TimeDelta>(AssembledMargin())
+                               : std::nullopt;
+    }
+    std::optional<TimeDelta> AssembledMarginDeficit() const {
+      return !AssembledInTime() ? std::optional<TimeDelta>(AssembledMargin())
+                                : std::nullopt;
+    }
+
+    // Post-buffer margin between render timestamp (target) and
+    // rendered timestamp (actual render):
+    //   * Frames should not be rendered early, if the timing works well.
+    //   * A frame that is rendered on time w.r.t. the target has zero margin.
+    //   * A frame that is rendered late w.r.t. the target has negative margin.
+    TimeDelta RenderedMargin() const {
+      return (render_timestamp - kRenderDelay) - rendered_timestamp;
+    }
+    bool RenderedInTime() const {
+      return RenderedMargin() > kInTimeMarginThreshold;
+    }
+
+    // A well-functioning jitter buffer would have very few (or non) frames with
+    // a render margin excess. Render margin deficits can happen though.
+    std::optional<TimeDelta> RenderedMarginExcess() const {
+      return RenderedInTime() ? std::optional<TimeDelta>(RenderedMargin())
+                              : std::nullopt;
+    }
+    std::optional<TimeDelta> RenderedMarginDeficit() const {
+      return !RenderedInTime() ? std::optional<TimeDelta>(RenderedMargin())
+                               : std::nullopt;
     }
   };
 
@@ -169,7 +206,18 @@ class RenderingSimulator {
   };
 
   // Static configuration.
+
+  // The "render delay" that is passed through the timing component and
+  // render buffer. It is added and subtracted through the pipeline, so it is
+  // important to have it set.
   static constexpr TimeDelta kRenderDelay = TimeDelta::Millis(10);
+  // The threshold used for "in time" calculations of assembled and rendered
+  // margins. A threshold of `TimeDelta::Zero()` wouldn't work well, because
+  // the recorded render timestamp of the frames have a loss of precision:
+  // the scheduling is done on a microsecond level, but the
+  // `EncodedFrame::RenderTimestamp()` returns values on a millisecond level.
+  // See https://g-issues.webrtc.org/issues/483303559.
+  static constexpr TimeDelta kInTimeMarginThreshold = TimeDelta::Micros(-500);
 
   explicit RenderingSimulator(Config config);
   ~RenderingSimulator();
diff --git a/video/timing/simulator/rendering_simulator_unittest.cc b/video/timing/simulator/rendering_simulator_unittest.cc
index 2dba5281be..debec90aee 100644
--- a/video/timing/simulator/rendering_simulator_unittest.cc
+++ b/video/timing/simulator/rendering_simulator_unittest.cc
@@ -36,6 +36,96 @@ using ::testing::SizeIs;
 using Frame = RenderingSimulator::Frame;
 using Stream = RenderingSimulator::Stream;
 
+TEST(RenderingSimulatorFrameTest, BufferDurationsAndMarginsForInTimeFrame) {
+  RenderingSimulator::Frame frame{
+      .first_packet_arrival_timestamp = Timestamp::Zero(),
+      .assembled_timestamp = Timestamp::Millis(1),
+      .render_timestamp =
+          Timestamp::Millis(10) + RenderingSimulator::kRenderDelay,
+      .decoded_timestamp = Timestamp::Millis(9),
+      .rendered_timestamp = Timestamp::Millis(10)};
+
+  // Buffer durations.
+  EXPECT_EQ(frame.PacketBufferDuration(), TimeDelta::Millis(1));
+  EXPECT_EQ(frame.FrameBufferDuration(), TimeDelta::Millis(8));
+  EXPECT_EQ(frame.RenderBufferDuration(), TimeDelta::Millis(1));
+  EXPECT_EQ(frame.TotalBufferDuration(), TimeDelta::Millis(10));
+
+  // Margins.
+  EXPECT_EQ(frame.AssembledMargin(), TimeDelta::Millis(9));
+  EXPECT_TRUE(frame.AssembledInTime());
+  EXPECT_TRUE(frame.AssembledMarginExcess().has_value());
+  EXPECT_FALSE(frame.AssembledMarginDeficit().has_value());
+  EXPECT_EQ(frame.RenderedMargin(), TimeDelta::Zero());
+  EXPECT_TRUE(frame.RenderedInTime());
+  EXPECT_TRUE(frame.RenderedMarginExcess().has_value());
+  EXPECT_FALSE(frame.RenderedMarginDeficit().has_value());
+}
+
+TEST(RenderingSimulatorFrameTest, BufferDurationsAndMarginsForDelayedFrame) {
+  RenderingSimulator::Frame frame{
+      .first_packet_arrival_timestamp = Timestamp::Millis(12),
+      .assembled_timestamp = Timestamp::Millis(12),
+      .render_timestamp =
+          Timestamp::Millis(10) + RenderingSimulator::kRenderDelay,
+      .decoded_timestamp = Timestamp::Millis(12),
+      .rendered_timestamp = Timestamp::Millis(12)};
+
+  // Buffer durations.
+  EXPECT_EQ(frame.PacketBufferDuration(), TimeDelta::Zero());
+  EXPECT_EQ(frame.FrameBufferDuration(), TimeDelta::Zero());
+  EXPECT_EQ(frame.RenderBufferDuration(), TimeDelta::Zero());
+  EXPECT_EQ(frame.TotalBufferDuration(), TimeDelta::Zero());
+
+  // Margins.
+  EXPECT_EQ(frame.AssembledMargin(), TimeDelta::Millis(-2));
+  EXPECT_FALSE(frame.AssembledInTime());
+  EXPECT_FALSE(frame.AssembledMarginExcess().has_value());
+  EXPECT_TRUE(frame.AssembledMarginDeficit().has_value());
+  EXPECT_EQ(frame.RenderedMargin(), TimeDelta::Millis(-2));
+  EXPECT_FALSE(frame.RenderedInTime());
+  EXPECT_FALSE(frame.RenderedMarginExcess().has_value());
+  EXPECT_TRUE(frame.RenderedMarginDeficit().has_value());
+}
+
+TEST(RenderingSimulatorFrameTest,
+     InTimeHelpersReturnTrueWhenRenderTimestampRoundedDownToMsPrecision) {
+  Timestamp processing_timestamp = Timestamp::Micros(10499);
+
+  RenderingSimulator::Frame frame{.assembled_timestamp = processing_timestamp,
+                                  .rendered_timestamp = processing_timestamp};
+  // Simulate the rounding down in `SetRenderTime(render_time.ms());`
+  frame.render_timestamp = Timestamp::Millis(processing_timestamp.ms()) +
+                           RenderingSimulator::kRenderDelay;
+  EXPECT_EQ(frame.render_timestamp,
+            Timestamp::Millis(10) + RenderingSimulator::kRenderDelay);
+
+  // Margins.
+  EXPECT_EQ(frame.AssembledMargin(), TimeDelta::Micros(-499));
+  EXPECT_TRUE(frame.AssembledInTime());
+  EXPECT_EQ(frame.RenderedMargin(), TimeDelta::Micros(-499));
+  EXPECT_TRUE(frame.RenderedInTime());
+}
+
+TEST(RenderingSimulatorFrameTest,
+     InTimeHelpersReturnTrueWhenRenderTimestampRoundedUpToMsPrecision) {
+  Timestamp processing_timestamp = Timestamp::Micros(10500);
+
+  RenderingSimulator::Frame frame{.assembled_timestamp = processing_timestamp,
+                                  .rendered_timestamp = processing_timestamp};
+  // Simulate the rounding up in `SetRenderTime(render_time.ms());`
+  frame.render_timestamp = Timestamp::Millis(processing_timestamp.ms()) +
+                           RenderingSimulator::kRenderDelay;
+  EXPECT_EQ(frame.render_timestamp,
+            Timestamp::Millis(11) + RenderingSimulator::kRenderDelay);
+
+  // Margins.
+  EXPECT_EQ(frame.AssembledMargin(), TimeDelta::Micros(500));
+  EXPECT_TRUE(frame.AssembledInTime());
+  EXPECT_EQ(frame.RenderedMargin(), TimeDelta::Micros(500));
+  EXPECT_TRUE(frame.RenderedInTime());
+}
+
 Matcher<const Frame&> EqualsFrame(const Frame& expected) {
   return AllOf(
       Field("num_packets", &Frame::num_packets, Eq(expected.num_packets)),

From 8485ffd5806c41dc4a038470c2644f08a2807d07 Mon Sep 17 00:00:00 2001
From: Jeremy Leconte <jleconte@webrtc.org>
Date: Wed, 11 Feb 2026 09:58:08 +0100
Subject: [PATCH 016/472] Add network queue to the schedulable network.

Also remove SimulatedNetworkNode::Build that takes a NetworkEmulationManager as argument. It was added 6 years ago but was never used.

Change-Id: Ieb02304f45e31feb321e92e068aee5ecd6e8e141
Bug: b/482318882
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/446880
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#46869}
---
 api/test/network_emulation/BUILD.gn           |  2 +-
 .../schedulable_network_node_builder.cc       | 12 +++++-
 .../schedulable_network_node_builder.h        |  3 ++
 api/test/network_emulation_manager.cc         | 38 +++++--------------
 api/test/network_emulation_manager.h          |  3 --
 test/network/BUILD.gn                         |  2 +
 test/network/schedulable_network_behavior.cc  | 11 ++++--
 test/network/schedulable_network_behavior.h   |  6 ++-
 .../schedulable_network_behavior_test.cc      | 34 +++++++++++------
 9 files changed, 62 insertions(+), 49 deletions(-)

diff --git a/api/test/network_emulation/BUILD.gn b/api/test/network_emulation/BUILD.gn
index 9d0ec6da4e..7e9e0ff8af 100644
--- a/api/test/network_emulation/BUILD.gn
+++ b/api/test/network_emulation/BUILD.gn
@@ -24,8 +24,8 @@ if (rtc_enable_protobuf) {
     ]
     deps = [
       ":network_config_schedule_proto",
+      ":network_queue",
       "../..:network_emulation_manager_api",
-      "../../../rtc_base:timeutils",
       "../../../test/network:schedulable_network_behavior",
       "../../units:timestamp",
       "//third_party/abseil-cpp/absl/functional:any_invocable",
diff --git a/api/test/network_emulation/schedulable_network_node_builder.cc b/api/test/network_emulation/schedulable_network_node_builder.cc
index 8040e2b903..d959338434 100644
--- a/api/test/network_emulation/schedulable_network_node_builder.cc
+++ b/api/test/network_emulation/schedulable_network_node_builder.cc
@@ -15,7 +15,9 @@
 #include <utility>
 
 #include "absl/functional/any_invocable.h"
+#include "api/test/network_emulation/leaky_bucket_network_queue.h"
 #include "api/test/network_emulation/network_config_schedule.pb.h"
+#include "api/test/network_emulation/network_queue.h"
 #include "api/test/network_emulation_manager.h"
 #include "api/units/timestamp.h"
 #include "test/network/schedulable_network_behavior.h"
@@ -34,6 +36,11 @@ void SchedulableNetworkNodeBuilder::set_start_condition(
   start_condition_ = std::move(start_condition);
 }
 
+void SchedulableNetworkNodeBuilder::set_queue_factory(
+    NetworkQueueFactory& queue_factory) {
+  queue_factory_ = &queue_factory;
+}
+
 EmulatedNetworkNode* SchedulableNetworkNodeBuilder::Build(
     std::optional<uint64_t> random_seed) {
   uint64_t seed =
@@ -41,8 +48,11 @@ EmulatedNetworkNode* SchedulableNetworkNodeBuilder::Build(
           ? *random_seed
           : static_cast<uint64_t>(
                 net_.time_controller()->GetClock()->CurrentTime().ns());
+  std::unique_ptr<NetworkQueue> network_queue =
+      queue_factory_ ? queue_factory_->CreateQueue()
+                     : std::make_unique<LeakyBucketNetworkQueue>();
   return net_.CreateEmulatedNode(std::make_unique<SchedulableNetworkBehavior>(
       std::move(schedule_), seed, *net_.time_controller()->GetClock(),
-      std::move(start_condition_)));
+      std::move(start_condition_), std::move(network_queue)));
 }
 }  // namespace webrtc
diff --git a/api/test/network_emulation/schedulable_network_node_builder.h b/api/test/network_emulation/schedulable_network_node_builder.h
index ee15215e40..307c7cabb9 100644
--- a/api/test/network_emulation/schedulable_network_node_builder.h
+++ b/api/test/network_emulation/schedulable_network_node_builder.h
@@ -15,6 +15,7 @@
 
 #include "absl/functional/any_invocable.h"
 #include "api/test/network_emulation/network_config_schedule.pb.h"
+#include "api/test/network_emulation/network_queue.h"
 #include "api/test/network_emulation_manager.h"
 #include "api/units/timestamp.h"
 
@@ -31,6 +32,7 @@ class SchedulableNetworkNodeBuilder {
   // NetworkConfigScheduleItem is used. There is no guarantee on which
   // thread/task queue that will be used.
   void set_start_condition(absl::AnyInvocable<bool(Timestamp)> start_condition);
+  void set_queue_factory(NetworkQueueFactory& queue_factory);
 
   // If no random seed is provided, one will be created.
   // The random seed is required for loss rate and to delay standard deviation.
@@ -41,6 +43,7 @@ class SchedulableNetworkNodeBuilder {
   NetworkEmulationManager& net_;
   network_behaviour::NetworkConfigSchedule schedule_;
   absl::AnyInvocable<bool(Timestamp)> start_condition_;
+  NetworkQueueFactory* queue_factory_ = nullptr;
 };
 
 }  // namespace webrtc
diff --git a/api/test/network_emulation_manager.cc b/api/test/network_emulation_manager.cc
index dc737f8ee7..e12caf5872 100644
--- a/api/test/network_emulation_manager.cc
+++ b/api/test/network_emulation_manager.cc
@@ -146,27 +146,14 @@ NetworkEmulationManager::SimulatedNetworkNode
 NetworkEmulationManager::SimulatedNetworkNode::Builder::Build(
     uint64_t random_seed) const {
   RTC_CHECK(net_);
-  return Build(net_, random_seed);
-}
-
-NetworkEmulationManager::SimulatedNetworkNode
-NetworkEmulationManager::SimulatedNetworkNode::Builder::Build(
-    NetworkEmulationManager* net,
-    uint64_t random_seed) const {
-  RTC_CHECK(net);
-  RTC_CHECK(net_ == nullptr || net_ == net);
-  std::unique_ptr<NetworkQueue> network_queue;
-  if (queue_factory_ != nullptr) {
-    network_queue = queue_factory_->CreateQueue();
-  } else {
-    network_queue = std::make_unique<LeakyBucketNetworkQueue>();
-  }
-  SimulatedNetworkNode res;
+  std::unique_ptr<NetworkQueue> network_queue =
+      queue_factory_ ? queue_factory_->CreateQueue()
+                     : std::make_unique<LeakyBucketNetworkQueue>();
   auto behavior = std::make_unique<SimulatedNetwork>(config_, random_seed,
                                                      std::move(network_queue));
-  res.simulation = behavior.get();
-  res.node = net->CreateEmulatedNode(std::move(behavior));
-  return res;
+  return SimulatedNetworkNode{
+      .simulation = behavior.get(),
+      .node = net_->CreateEmulatedNode(std::move(behavior))};
 }
 
 std::pair<EmulatedNetworkManagerInterface*, EmulatedNetworkManagerInterface*>
@@ -177,15 +164,10 @@ NetworkEmulationManager::CreateEndpointPairWithTwoWayRoutes(
   auto* alice_node = CreateEmulatedNode(config);
   auto* bob_node = CreateEmulatedNode(config);
 
-  std::vector<EmulatedEndpoint*> alice_endpoints;
-  for (int i = 0; i < alice_interface_count; i++) {
-    alice_endpoints.push_back(CreateEndpoint(EmulatedEndpointConfig()));
-  }
-
-  std::vector<EmulatedEndpoint*> bob_endpoints;
-  for (int i = 0; i < bob_interface_count; i++) {
-    bob_endpoints.push_back(CreateEndpoint(EmulatedEndpointConfig()));
-  }
+  std::vector<EmulatedEndpoint*> alice_endpoints(
+      alice_interface_count, CreateEndpoint(EmulatedEndpointConfig()));
+  std::vector<EmulatedEndpoint*> bob_endpoints(
+      bob_interface_count, CreateEndpoint(EmulatedEndpointConfig()));
 
   for (auto alice_endpoint : alice_endpoints) {
     for (auto bob_endpoint : bob_endpoints) {
diff --git a/api/test/network_emulation_manager.h b/api/test/network_emulation_manager.h
index 52adc336af..611a4b349f 100644
--- a/api/test/network_emulation_manager.h
+++ b/api/test/network_emulation_manager.h
@@ -186,7 +186,6 @@ class NetworkEmulationManager {
     class Builder {
      public:
       explicit Builder(NetworkEmulationManager* net) : net_(net) {}
-      Builder() : net_(nullptr) {}
       Builder(const Builder&) = default;
       // Sets the config state, note that this will replace any previously set
       // values.
@@ -204,8 +203,6 @@ class NetworkEmulationManager {
       Builder& avg_burst_loss_length(int avg_burst_loss_length);
       Builder& packet_overhead(int packet_overhead);
       SimulatedNetworkNode Build(uint64_t random_seed = 1) const;
-      SimulatedNetworkNode Build(NetworkEmulationManager* net,
-                                 uint64_t random_seed = 1) const;
 
      private:
       NetworkEmulationManager* const net_;
diff --git a/test/network/BUILD.gn b/test/network/BUILD.gn
index 9fce95253b..2e9818f673 100644
--- a/test/network/BUILD.gn
+++ b/test/network/BUILD.gn
@@ -346,6 +346,7 @@ if (rtc_enable_protobuf) {
       "../../api:simulated_network_api",
       "../../api/task_queue",
       "../../api/test/network_emulation:network_config_schedule_proto",
+      "../../api/test/network_emulation:network_queue",
       "../../api/units:data_rate",
       "../../api/units:time_delta",
       "../../api/units:timestamp",
@@ -368,6 +369,7 @@ if (rtc_enable_protobuf) {
         "../../api:network_emulation_manager_api",
         "../../api:simulated_network_api",
         "../../api/test/network_emulation:network_config_schedule_proto",
+        "../../api/test/network_emulation:network_queue",
         "../../api/units:time_delta",
         "../../api/units:timestamp",
         "../../system_wrappers",
diff --git a/test/network/schedulable_network_behavior.cc b/test/network/schedulable_network_behavior.cc
index f602eb39f6..494947218e 100644
--- a/test/network/schedulable_network_behavior.cc
+++ b/test/network/schedulable_network_behavior.cc
@@ -11,12 +11,14 @@
 
 #include <algorithm>
 #include <cstdint>
+#include <memory>
 #include <utility>
 
 #include "absl/functional/any_invocable.h"
 #include "api/sequence_checker.h"
 #include "api/task_queue/task_queue_base.h"
 #include "api/test/network_emulation/network_config_schedule.pb.h"
+#include "api/test/network_emulation/network_queue.h"
 #include "api/test/simulated_network.h"
 #include "api/units/data_rate.h"
 #include "api/units/time_delta.h"
@@ -77,10 +79,13 @@ SchedulableNetworkBehavior::SchedulableNetworkBehavior(
     network_behaviour::NetworkConfigSchedule schedule,
     uint64_t random_seed,
     Clock& clock,
-    absl::AnyInvocable<bool(Timestamp)> start_callback)
-    : SimulatedNetwork(GetInitialConfig(schedule), random_seed),
+    absl::AnyInvocable<bool(Timestamp)> start_condition,
+    std::unique_ptr<NetworkQueue> queue)
+    : SimulatedNetwork(GetInitialConfig(schedule),
+                       random_seed,
+                       std::move(queue)),
       schedule_(std::move(schedule)),
-      start_condition_(std::move(start_callback)),
+      start_condition_(std::move(start_condition)),
       clock_(clock),
       config_(GetInitialConfig(schedule_)) {
   if (schedule_.item().size() > 1) {
diff --git a/test/network/schedulable_network_behavior.h b/test/network/schedulable_network_behavior.h
index 53cf05de94..c38858dd8f 100644
--- a/test/network/schedulable_network_behavior.h
+++ b/test/network/schedulable_network_behavior.h
@@ -12,10 +12,12 @@
 #define TEST_NETWORK_SCHEDULABLE_NETWORK_BEHAVIOR_H_
 
 #include <cstdint>
+#include <memory>
 
 #include "absl/functional/any_invocable.h"
 #include "api/sequence_checker.h"
 #include "api/test/network_emulation/network_config_schedule.pb.h"
+#include "api/test/network_emulation/network_queue.h"
 #include "api/test/simulated_network.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
@@ -34,8 +36,8 @@ class SchedulableNetworkBehavior : public SimulatedNetwork {
       network_behaviour::NetworkConfigSchedule schedule,
       uint64_t random_seed,
       Clock& clock,
-      absl::AnyInvocable<bool(webrtc::Timestamp)> start_condition =
-          [](webrtc::Timestamp) { return true; });
+      absl::AnyInvocable<bool(webrtc::Timestamp)> start_condition,
+      std::unique_ptr<NetworkQueue> queue);
 
   bool EnqueuePacket(PacketInFlightInfo packet_info) override;
 
diff --git a/test/network/schedulable_network_behavior_test.cc b/test/network/schedulable_network_behavior_test.cc
index 2108e0caa9..62e4e065a8 100644
--- a/test/network/schedulable_network_behavior_test.cc
+++ b/test/network/schedulable_network_behavior_test.cc
@@ -14,6 +14,7 @@
 #include <vector>
 
 #include "api/test/create_network_emulation_manager.h"
+#include "api/test/network_emulation/leaky_bucket_network_queue.h"
 #include "api/test/network_emulation/network_config_schedule.pb.h"
 #include "api/test/network_emulation_manager.h"
 #include "api/test/simulated_network.h"
@@ -64,8 +65,10 @@ TEST(SchedulableNetworkBehaviorTest, NoSchedule) {
   SchedulableNetworkBehaviorTestFixture fixture;
 
   network_behaviour::NetworkConfigSchedule schedule;
-  SchedulableNetworkBehavior network_behaviour(schedule, kRandomSeed,
-                                               fixture.clock());
+  SchedulableNetworkBehavior network_behaviour(
+      schedule, kRandomSeed, fixture.clock(),
+      [](webrtc::Timestamp) { return true; },
+      std::make_unique<LeakyBucketNetworkQueue>());
   webrtc::Timestamp send_time = fixture.TimeNow();
   EXPECT_TRUE(network_behaviour.EnqueuePacket({/*size=*/1000 / 8,
                                                /*send_time_us=*/send_time.us(),
@@ -85,8 +88,10 @@ TEST(SchedulableNetworkBehaviorTest, ScheduleWithoutUpdates) {
   initial_config->set_link_capacity_kbps(10);
   initial_config->set_queue_delay_ms(70);
 
-  SchedulableNetworkBehavior network_behaviour(schedule, kRandomSeed,
-                                               fixture.clock());
+  SchedulableNetworkBehavior network_behaviour(
+      schedule, kRandomSeed, fixture.clock(),
+      [](webrtc::Timestamp) { return true; },
+      std::make_unique<LeakyBucketNetworkQueue>());
   webrtc::Timestamp send_time = fixture.TimeNow();
   EXPECT_TRUE(network_behaviour.EnqueuePacket({/*size=*/1000 / 8,
                                                /*send_time_us=*/send_time.us(),
@@ -124,8 +129,10 @@ TEST(SchedulableNetworkBehaviorTest,
   // 55ms.
   updated_capacity->set_link_capacity_kbps(100);
 
-  SchedulableNetworkBehavior network_behaviour(schedule, kRandomSeed,
-                                               fixture.clock());
+  SchedulableNetworkBehavior network_behaviour(
+      schedule, kRandomSeed, fixture.clock(),
+      [](webrtc::Timestamp) { return true; },
+      std::make_unique<LeakyBucketNetworkQueue>());
   MockFunction<void()> delivery_time_changed_callback;
   network_behaviour.RegisterDeliveryTimeChangedCallback(
       delivery_time_changed_callback.AsStdFunction());
@@ -177,7 +184,8 @@ TEST(SchedulableNetworkBehaviorTest, ScheduleStartedWhenStartConditionTrue) {
       .InSequence(s)
       .WillOnce(Return(true));
   SchedulableNetworkBehavior network_behaviour(
-      schedule, kRandomSeed, fixture.clock(), start_condition.AsStdFunction());
+      schedule, kRandomSeed, fixture.clock(), start_condition.AsStdFunction(),
+      std::make_unique<LeakyBucketNetworkQueue>());
 
   EXPECT_TRUE(network_behaviour.EnqueuePacket(
       {/*size=*/1000 / 8,
@@ -213,8 +221,10 @@ TEST(SchedulableNetworkBehaviorTest, ScheduleWithRepeat) {
   // config should again take 100ms to send.
   schedule.set_repeat_schedule_after_last_ms(200);
 
-  SchedulableNetworkBehavior network_behaviour(schedule, kRandomSeed,
-                                               fixture.clock());
+  SchedulableNetworkBehavior network_behaviour(
+      schedule, kRandomSeed, fixture.clock(),
+      [](webrtc::Timestamp) { return true; },
+      std::make_unique<LeakyBucketNetworkQueue>());
 
   webrtc::Timestamp first_packet_send_time = fixture.TimeNow();
   EXPECT_TRUE(network_behaviour.EnqueuePacket(
@@ -254,8 +264,10 @@ TEST(SchedulableNetworkBehaviorTest, ScheduleWithoutRepeat) {
   // A packet of size 1000 bits should take 10ms to send.
   updated_capacity->set_link_capacity_kbps(100);
 
-  SchedulableNetworkBehavior network_behaviour(schedule, kRandomSeed,
-                                               fixture.clock());
+  SchedulableNetworkBehavior network_behaviour(
+      schedule, kRandomSeed, fixture.clock(),
+      [](webrtc::Timestamp) { return true; },
+      std::make_unique<LeakyBucketNetworkQueue>());
 
   webrtc::Timestamp first_packet_send_time = fixture.TimeNow();
   EXPECT_TRUE(network_behaviour.EnqueuePacket(

From 7ce221014c9ce6403604bb6a1126aec04404d3f4 Mon Sep 17 00:00:00 2001
From: Evan Shrubsole <eshr@webrtc.org>
Date: Wed, 11 Feb 2026 12:00:57 +0000
Subject: [PATCH 017/472] Use clock in rtc_event_log_unittest_helper
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Bug: webrtc:42223992
Change-Id: I1436f1045a1c0dd55243388e99eb29466a6a6964
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/448300
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46870}
---
 logging/rtc_event_log/rtc_event_log_unittest_helper.cc | 8 ++++----
 1 file changed, 4 insertions(+), 4 deletions(-)

diff --git a/logging/rtc_event_log/rtc_event_log_unittest_helper.cc b/logging/rtc_event_log/rtc_event_log_unittest_helper.cc
index 719ec53d75..db573e7913 100644
--- a/logging/rtc_event_log/rtc_event_log_unittest_helper.cc
+++ b/logging/rtc_event_log/rtc_event_log_unittest_helper.cc
@@ -87,7 +87,6 @@
 #include "rtc_base/buffer.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/random.h"
-#include "rtc_base/time_utils.h"
 #include "system_wrappers/include/ntp_time.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
@@ -238,13 +237,14 @@ std::unique_ptr<RtcEventFrameDecoded> EventGenerator::NewFrameDecodedEvent(
   constexpr VideoCodecType kCodecList[kNumCodecTypes] = {
       kVideoCodecGeneric, kVideoCodecVP8,  kVideoCodecVP9,
       kVideoCodecAV1,     kVideoCodecH264, kVideoCodecH265};
-  const int64_t render_time_ms =
-      TimeMillis() + prng_.Rand(kMinRenderDelayMs, kMaxRenderDelayMs);
+  const Timestamp render_time =
+      clock_.CurrentTime() +
+      TimeDelta::Millis(prng_.Rand(kMinRenderDelayMs, kMaxRenderDelayMs));
   const int width = prng_.Rand(kMinWidth, kMaxWidth);
   const int height = prng_.Rand(kMinHeight, kMaxHeight);
   const VideoCodecType codec = kCodecList[prng_.Rand(0, kNumCodecTypes - 1)];
   const uint8_t qp = prng_.Rand<uint8_t>();
-  return Create<RtcEventFrameDecoded>(render_time_ms, ssrc, width, height,
+  return Create<RtcEventFrameDecoded>(render_time.ms(), ssrc, width, height,
                                       codec, qp);
 }
 

From 5738f0130a8a81b017b539325844b1a73e9a88ee Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Wed, 11 Feb 2026 04:03:16 -0800
Subject: [PATCH 018/472] Roll chromium_revision be8f64a423..974282c604
 (1582802:1583127)

Change log: https://chromium.googlesource.com/chromium/src/+log/be8f64a423..974282c604
Full diff: https://chromium.googlesource.com/chromium/src/+/be8f64a423..974282c604

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/e82653b704..d0790d1596
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/d8de59743d..66d45f56c2
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/62968b4611..4b9107b4d6
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/f4814d241e..0cee9b0c86
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/73cd946964..ec1c593db0
* src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/8658704610..7a0968b56b
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/42786f6e46..6c6a6572eb
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/75ff03fd6c..3151eefe26
* src/third_party/libaom/source/libaom: https://aomedia.googlesource.com/aom.git/+log/4018d3b634..37b2db4334
* src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/4f89a98f36..2f7df07563
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/4d0f63f082..ab1d624455
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/17573b6d7d..b503bfc821
DEPS diff: https://chromium.googlesource.com/chromium/src/+/be8f64a423..974282c604/DEPS

BUG=None

Change-Id: Ica4c3a8fdbd57ace5619fb950c544fd61191139a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/448320
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#46871}
---
 DEPS | 26 +++++++++++++-------------
 1 file changed, 13 insertions(+), 13 deletions(-)

diff --git a/DEPS b/DEPS
index 3aeb7155f3..e573ffd0f3 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': 'be8f64a4234dbe7b2308a01dc57dddfea118fc1d',
+  'chromium_revision': '974282c604825b4b9a3bcff508fa433e1b84dc5b',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@e82653b70411841da3662ed6edf96694a55f4068',
+    'https://chromium.googlesource.com/chromium/src/build@d0790d159683f591bcbea765f8f72f9c8f62f866',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@6a18683f555b4ac8b05ac8395c29c84483ac9588',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@d8de59743dc98dc9932b7db60f90cc3db0149c2f',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@66d45f56c2145b9f0bbe28b26aae967c546bb94b',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@62968b461188cb73855fcc23c540ee8472eb81c4',
+    'https://chromium.googlesource.com/chromium/src/testing@4b9107b4d692153ec821742de16cf049435318c7',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@f4814d241e26dc251fabe5c66b0c3b997f28a8c8',
+    'https://chromium.googlesource.com/chromium/src/third_party@0cee9b0c86b98cc0ce990de4d7bd8bf036e8f2df',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -331,13 +331,13 @@ deps = {
   'src/third_party/clang-format/script':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@c2725e0622e1a86d55f14514f2177a39efea4a0e',
   'src/third_party/compiler-rt/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@86587046105639b66fe40059bf8fdb8a33522f97',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@7a0968b56bfaafaf09775f0b2f06a79fc47c83b2',
   'src/third_party/libc++/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@7ab65651aed6802d2599dcb7a73b1f82d5179d05',
   'src/third_party/libc++abi/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@8f11bb1d4438d0239d0dfc1bd9456a9f31629dda',
   'src/third_party/llvm-libc/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@4f89a98f364a8b42b19890afb5ba3ffa526a34f0',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@2f7df07563d93f442b05a711769e2613715ad279',
   'src/third_party/libunwind/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@ba19d93d6d4f467fba11ff20fe2fc7c056f79345',
 
@@ -488,7 +488,7 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@73cd946964c555a39730592db4b540accbafac6c',
+    'https://boringssl.googlesource.com/boringssl.git@ec1c593db0ef36ed47634f9795022e102ec17f71',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@79099fdf668ae097c9eca7052fd5c4c5de6c9098',
   'src/third_party/catapult':
@@ -505,13 +505,13 @@ deps = {
   'src/third_party/crc32c/src':
     'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558',
   'src/third_party/depot_tools':
-    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@42786f6e46c25c30dd58f69283ab6fcd0c959f58',
+    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@6c6a6572eb0e27efd41f8724b4631ac2aa9df460',
   'src/third_party/ffmpeg':
     'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@ae11d2ba5c835b822a61d6a99eeb853ca30d41d8',
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@75ff03fd6c1763cbb4ab047df6628e57562237de',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@3151eefe26976c60f2ddd9bf74513dd0152136ea',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -601,7 +601,7 @@ deps = {
   'src/third_party/dav1d/libdav1d':
     'https://chromium.googlesource.com/external/github.com/videolan/dav1d.git@b546257f770768b2c88258c533da38b91a06f737',
   'src/third_party/libaom/source/libaom':
-    'https://aomedia.googlesource.com/aom.git@4018d3b63456eb657475e66c352bfa86f321e0f5',
+    'https://aomedia.googlesource.com/aom.git@37b2db4334d3b90c3b3b6e188659a2214a411907',
   'src/third_party/libgav1/src':
     Var('chromium_git') + '/codecs/libgav1.git' + '@' + '40f58ed32ff39071c3f2a51056dbc49a070af0dc',
   'src/third_party/libunwindstack': {
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '4d0f63f0824b488797427c6d9840932886bef221',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'ab1d624455b26a0094f946b88f5e772a0baa1d3a',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@17573b6d7d84674cc1012af9b31ff30e7c5fff53',
+    'https://chromium.googlesource.com/chromium/src/tools@b503bfc8210edd598e2006ceae2f9287869b533d',
 
   'src/third_party/espresso': {
       'packages': [

From 27eddf019f4282999afd957721d2ed8f98dd826d Mon Sep 17 00:00:00 2001
From: Evan Shrubsole <eshr@webrtc.org>
Date: Wed, 11 Feb 2026 12:15:53 +0000
Subject: [PATCH 019/472] Use clock in Logging Perf test

Bug: webrtc:42223992
Change-Id: If5e4dd0a2111b63498b0fe1bcdf9e2566a6a6964
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/448282
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46872}
---
 rtc_base/logging_unittest.cc | 15 ++++++++-------
 1 file changed, 8 insertions(+), 7 deletions(-)

diff --git a/rtc_base/logging_unittest.cc b/rtc_base/logging_unittest.cc
index 47c2deea70..217fed9bdf 100644
--- a/rtc_base/logging_unittest.cc
+++ b/rtc_base/logging_unittest.cc
@@ -17,9 +17,10 @@
 #include <string>
 
 #include "absl/strings/string_view.h"
+#include "api/units/timestamp.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/platform_thread.h"
-#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/clock.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
 
@@ -309,6 +310,8 @@ TEST(LogTest, CheckTagAddedToStringInDefaultOnLogMessageAndroid) {
 
 // Test the time required to write 1000 80-character logs to a string.
 TEST(LogTest, Perf) {
+  // This test should probably be turned into a benchmark.
+  Clock& clock = *Clock::GetRealTimeClock();
   std::string str;
   LogSinkImpl stream(&str);
   LogMessage::AddLogToStream(&stream, LS_VERBOSE);
@@ -325,19 +328,17 @@ TEST(LogTest, Perf) {
   str.reserve(120000);
   static const int kRepetitions = 1000;
 
-  int64_t start = TimeMillis(), finish;
+  Timestamp start = clock.CurrentTime();
   for (int i = 0; i < kRepetitions; ++i) {
     LogMessageForTesting(__FILE__, __LINE__, LS_VERBOSE).stream() << message;
   }
-  finish = TimeMillis();
+  Timestamp finish = clock.CurrentTime();
 
   LogMessage::RemoveLogToStream(&stream);
 
   EXPECT_EQ(str.size(), (message.size() + logging_overhead) * kRepetitions);
-  RTC_LOG(LS_INFO) << "Total log time: " << TimeDiff(finish, start)
-                   << " ms "
-                      " total bytes logged: "
-                   << str.size();
+  RTC_LOG(LS_INFO) << "Total log time: " << (finish - start)
+                   << " total bytes logged: " << str.size();
 }
 
 TEST(LogTest, EnumsAreSupported) {

From d0aec8fb0c2c4c628b7678689865fb0dfc616e1e Mon Sep 17 00:00:00 2001
From: Rasmus Brandt <brandtr@webrtc.org>
Date: Wed, 11 Feb 2026 12:44:00 +0100
Subject: [PATCH 020/472] Video timing simulator: Add SSRC filtering to
 `RtcEventLogDriver`
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This will be useful when only analyzing a single / few streams in a log.

Bug: b/384950328
Change-Id: Ic304f88f70bcbee5cd24a1e377dce232bbf2ae43
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/448280
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46873}
---
 .../simulator/decodability_simulator.cc       |  4 ++-
 .../timing/simulator/decodability_simulator.h |  5 ++++
 video/timing/simulator/rendering_simulator.cc |  5 ++--
 video/timing/simulator/rendering_simulator.h  |  5 ++++
 .../timing/simulator/rtc_event_log_driver.cc  | 14 +++++++++-
 video/timing/simulator/rtc_event_log_driver.h |  9 +++++++
 .../rtc_event_log_driver_unittest.cc          | 27 +++++++++++++++++++
 7 files changed, 65 insertions(+), 4 deletions(-)

diff --git a/video/timing/simulator/decodability_simulator.cc b/video/timing/simulator/decodability_simulator.cc
index 21c0c4b735..856bf78b9c 100644
--- a/video/timing/simulator/decodability_simulator.cc
+++ b/video/timing/simulator/decodability_simulator.cc
@@ -186,7 +186,9 @@ DecodabilitySimulator::Results DecodabilitySimulator::Simulate(
   // Decodability should not be a function of any field trials, so we pass the
   // empty string here.
   RtcEventLogDriver rtc_event_log_simulator(
-      {.reuse_streams = config_.reuse_streams}, &parsed_log,
+      {.reuse_streams = config_.reuse_streams,
+       .ssrc_filter = config_.ssrc_filter},
+      &parsed_log,
       /*field_trials_string=*/"", std::move(stream_factory));
   rtc_event_log_simulator.Simulate();
 
diff --git a/video/timing/simulator/decodability_simulator.h b/video/timing/simulator/decodability_simulator.h
index 347f18d021..5fc56f365e 100644
--- a/video/timing/simulator/decodability_simulator.h
+++ b/video/timing/simulator/decodability_simulator.h
@@ -12,6 +12,7 @@
 #define VIDEO_TIMING_SIMULATOR_DECODABILITY_SIMULATOR_H_
 
 #include <cstdint>
+#include <set>
 #include <vector>
 
 #include "absl/algorithm/container.h"
@@ -35,6 +36,10 @@ class DecodabilitySimulator {
     // same SSRC. This can be useful for simulation, but likely not for data
     // analysis.
     bool reuse_streams = false;
+
+    // If non-empty, will only simulate video streams whose main SSRCs is
+    // contained in the set.
+    std::set<uint32_t> ssrc_filter = {};
   };
 
   // Metadata about a single decodable frame.
diff --git a/video/timing/simulator/rendering_simulator.cc b/video/timing/simulator/rendering_simulator.cc
index ea846ee25d..e2101bd9f3 100644
--- a/video/timing/simulator/rendering_simulator.cc
+++ b/video/timing/simulator/rendering_simulator.cc
@@ -240,8 +240,9 @@ RenderingSimulator::Results RenderingSimulator::Simulate(
                                                       rtx_ssrc, &results);
   };
   RtcEventLogDriver rtc_event_log_simulator(
-      {.reuse_streams = config_.reuse_streams}, &parsed_log,
-      config_.field_trials_string, std::move(stream_factory));
+      {.reuse_streams = config_.reuse_streams,
+       .ssrc_filter = config_.ssrc_filter},
+      &parsed_log, config_.field_trials_string, std::move(stream_factory));
   rtc_event_log_simulator.Simulate();
 
   // Return.
diff --git a/video/timing/simulator/rendering_simulator.h b/video/timing/simulator/rendering_simulator.h
index 0624476e5f..83f3b46a36 100644
--- a/video/timing/simulator/rendering_simulator.h
+++ b/video/timing/simulator/rendering_simulator.h
@@ -15,6 +15,7 @@
 #include <functional>
 #include <memory>
 #include <optional>
+#include <set>
 #include <string>
 #include <vector>
 
@@ -51,6 +52,10 @@ class RenderingSimulator {
     // Whether or not to reset the stream state on newly logged streams with the
     // same SSRC.
     bool reuse_streams = false;
+
+    // If non-empty, will only simulate video streams whose main SSRCs is
+    // contained in the set.
+    std::set<uint32_t> ssrc_filter = {};
   };
 
   // Metadata about a single rendered frame.
diff --git a/video/timing/simulator/rtc_event_log_driver.cc b/video/timing/simulator/rtc_event_log_driver.cc
index 729a08e791..e46a6edc00 100644
--- a/video/timing/simulator/rtc_event_log_driver.cc
+++ b/video/timing/simulator/rtc_event_log_driver.cc
@@ -140,6 +140,18 @@ void RtcEventLogDriver::OnLoggedVideoRecvConfig(
   uint32_t rtx_ssrc = config.config.rtx_ssrc;
   HandleEvent(config.log_time(), [this, ssrc, rtx_ssrc]() {
     RTC_DCHECK_RUN_ON(simulator_queue_.get());
+
+    all_known_ssrcs_.insert(ssrc);
+
+    // Skip setting up stream if not included in the filter.
+    if (!config_.ssrc_filter.empty() && !config_.ssrc_filter.contains(ssrc)) {
+      RTC_LOG(LS_INFO) << "OnLoggedVideoRecvConfig being skipped for "
+                       << "ssrc=" << ssrc << ", rtx_ssrc=" << rtx_ssrc
+                       << " (simulated_ts=" << env_.clock().CurrentTime()
+                       << ")";
+      return;
+    }
+
     RTC_LOG(LS_INFO) << "OnLoggedVideoRecvConfig for "
                      << "ssrc=" << ssrc << ", rtx_ssrc=" << rtx_ssrc
                      << " (simulated_ts=" << env_.clock().CurrentTime() << ")";
@@ -183,7 +195,7 @@ void RtcEventLogDriver::OnLoggedRtpPacketIncoming(
                     packet.log_time());
       RTC_DCHECK_EQ(env_.clock().CurrentTime(), packet.log_time());
       it->second->InsertSimulatedPacket(simulated_packet);
-    } else {
+    } else if (!all_known_ssrcs_.contains(ssrc)) {
       RTC_LOG(LS_WARNING) << "Received packet for unknown ssrc=" << ssrc
                           << " (simulated_ts=" << env_.clock().CurrentTime()
                           << ")";
diff --git a/video/timing/simulator/rtc_event_log_driver.h b/video/timing/simulator/rtc_event_log_driver.h
index c4b3386429..72e7a71c08 100644
--- a/video/timing/simulator/rtc_event_log_driver.h
+++ b/video/timing/simulator/rtc_event_log_driver.h
@@ -14,9 +14,11 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <set>
 
 #include "absl/base/nullability.h"
 #include "absl/container/flat_hash_map.h"
+#include "absl/container/flat_hash_set.h"
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
 #include "api/environment/environment.h"
@@ -57,6 +59,10 @@ class RtcEventLogDriver {
     // `LoggedVideoRecvConfig` was logged for the same stream. (This might
     // happen around `SetRemoteDescription`s.)
     bool reuse_streams = false;
+
+    // If non-empty, will only simulate video streams whose main SSRCs is
+    // contained in the set.
+    std::set<uint32_t> ssrc_filter = {};
   };
 
   // A stream that is driven by simulated RTP packets coming from the log.
@@ -131,6 +137,9 @@ class RtcEventLogDriver {
   // Streams for reception. Keyed by both `ssrc` and `rtx_ssrc`.
   absl::flat_hash_map<uint32_t, StreamInterface*> receiving_streams_
       RTC_GUARDED_BY(simulator_queue_);
+  // Keep track of all logged `ssrc`s for text logging purposes.
+  absl::flat_hash_set<uint32_t> all_known_ssrcs_
+      RTC_GUARDED_BY(simulator_queue_);
 };
 
 }  // namespace webrtc::video_timing_simulator
diff --git a/video/timing/simulator/rtc_event_log_driver_unittest.cc b/video/timing/simulator/rtc_event_log_driver_unittest.cc
index 5fef43ab5c..50ac1f2bef 100644
--- a/video/timing/simulator/rtc_event_log_driver_unittest.cc
+++ b/video/timing/simulator/rtc_event_log_driver_unittest.cc
@@ -119,6 +119,33 @@ TEST_F(RtcEventLogDriverTest, LoggedVideoRecvConfigCreatesStream) {
   EXPECT_EQ(stream_factory_.NumStreamsCreated(), 1);
 }
 
+TEST_F(RtcEventLogDriverTest,
+       LoggedVideoRecvConfigCreatesStreamIfIncludedInSsrcFilter) {
+  parsed_log_builder_.LogVideoRecvConfig(kSsrc1, kRtxSsrc1);
+  std::unique_ptr<ParsedRtcEventLog> parsed_log = parsed_log_builder_.Build();
+
+  RtcEventLogDriver driver(RtcEventLogDriver::Config{.ssrc_filter = {kSsrc1}},
+                           parsed_log.get(), kEmptyFieldTrialsString,
+                           BuildStreamFactory());
+  EXPECT_CALL(*stream_factory_.stream1_ptr_, Close());
+  driver.Simulate();
+
+  EXPECT_EQ(stream_factory_.NumStreamsCreated(), 1);
+}
+
+TEST_F(RtcEventLogDriverTest,
+       LoggedVideoRecvConfigDoesNotCreateStreamIfNotIncludedInSsrcFilter) {
+  parsed_log_builder_.LogVideoRecvConfig(kSsrc1, kRtxSsrc1);
+  std::unique_ptr<ParsedRtcEventLog> parsed_log = parsed_log_builder_.Build();
+
+  RtcEventLogDriver driver(RtcEventLogDriver::Config{.ssrc_filter = {kSsrc2}},
+                           parsed_log.get(), kEmptyFieldTrialsString,
+                           BuildStreamFactory());
+  driver.Simulate();
+
+  EXPECT_EQ(stream_factory_.NumStreamsCreated(), 0);
+}
+
 TEST_F(RtcEventLogDriverTest, LoggedVideoRecvConfigsCreateStreams) {
   parsed_log_builder_.LogVideoRecvConfig(kSsrc1, kRtxSsrc1);
   parsed_log_builder_.LogVideoRecvConfig(kSsrc2, kRtxSsrc2);

From 3d0277346241e8f9206a881d3ed500c4ce15cd85 Mon Sep 17 00:00:00 2001
From: Rasmus Brandt <brandtr@webrtc.org>
Date: Wed, 11 Feb 2026 14:37:17 +0100
Subject: [PATCH 021/472] Change logging severity to LS_VERBOSE for
 `RenderingTracker::OnDecodableFrameTimeout`
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This log line triggers every 10s whenever the video stream has been
deregistered.

Bug: b/384950328
Change-Id: I124746346c9f0a27f0319454c8c7cc6ca7694c67
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/448281
Auto-Submit: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46874}
---
 video/timing/simulator/rendering_tracker.cc | 5 ++++-
 1 file changed, 4 insertions(+), 1 deletion(-)

diff --git a/video/timing/simulator/rendering_tracker.cc b/video/timing/simulator/rendering_tracker.cc
index 86ccb85c0a..9741c5b0b6 100644
--- a/video/timing/simulator/rendering_tracker.cc
+++ b/video/timing/simulator/rendering_tracker.cc
@@ -158,7 +158,10 @@ void RenderingTracker::OnEncodedFrame(
 
 void RenderingTracker::OnDecodableFrameTimeout(TimeDelta wait_time) {
   RTC_DCHECK_RUN_ON(&sequence_checker_);
-  RTC_LOG(LS_WARNING) << "VideoStreamBufferController timed out after wait_ms="
+  // Log as verbose because this timeout will periodically trigger when the
+  // stream has been deregistered. Unfortunately there is no log event for the
+  // deregistration, so the simulator will keep expecting packets forever.
+  RTC_LOG(LS_VERBOSE) << "VideoStreamBufferController timed out after wait_ms="
                       << wait_time << " on ssrc=" << config_.ssrc
                       << " (simulated_ts=" << env_.clock().CurrentTime() << ")";
   // TODO: b/423646186 - Consider adding this as a callback event.

From cc688f55eb332c9e90f35a2db0ac571c91192698 Mon Sep 17 00:00:00 2001
From: Evan Shrubsole <eshr@webrtc.org>
Date: Wed, 11 Feb 2026 12:49:16 +0000
Subject: [PATCH 022/472] Use Clock and RunLoop in null_socket_server_unittest

Bug: webrtc:42223992,webrtc:469327588
Change-Id: I9b7646b9ba4809a924fbb8186594e2516a6a6964
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/448283
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46875}
---
 rtc_base/BUILD.gn                       |  1 +
 rtc_base/null_socket_server_unittest.cc | 27 ++++++++++++++-----------
 2 files changed, 16 insertions(+), 12 deletions(-)

diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
index 3a253296b2..d02a8341e1 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -2047,6 +2047,7 @@ if (rtc_include_tests) {
         "../test:create_test_environment",
         "../test:fileutils",
         "../test:near_matcher",
+        "../test:run_loop",
         "../test:test_support",
         "../test:wait_until",
         "system:file_wrapper",
diff --git a/rtc_base/null_socket_server_unittest.cc b/rtc_base/null_socket_server_unittest.cc
index 3b47acc5df..bd73efd3d8 100644
--- a/rtc_base/null_socket_server_unittest.cc
+++ b/rtc_base/null_socket_server_unittest.cc
@@ -10,41 +10,44 @@
 
 #include "rtc_base/null_socket_server.h"
 
-#include <cstdint>
 #include <memory>
 
-#include "api/test/rtc_error_matchers.h"
+#include "api/environment/environment.h"
 #include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
 #include "rtc_base/socket_server.h"
 #include "rtc_base/thread.h"
-#include "rtc_base/time_utils.h"
-#include "test/gmock.h"
+#include "test/create_test_environment.h"
 #include "test/gtest.h"
-#include "test/wait_until.h"
+#include "test/run_loop.h"
 
 namespace webrtc {
 
 TEST(NullSocketServerTest, WaitAndSet) {
-  AutoThread main_thread;
+  test::RunLoop run_loop;
   NullSocketServer ss;
   auto thread = Thread::Create();
   EXPECT_TRUE(thread->Start());
   thread->PostTask([&ss] { ss.WakeUp(); });
   // The process_io will be ignored.
   const bool process_io = true;
-  EXPECT_THAT(
-      WaitUntil([&] { return ss.Wait(SocketServer::kForever, process_io); },
-                ::testing::IsTrue(), {.timeout = TimeDelta::Millis(5'000)}),
-      IsRtcOk());
+  bool wait_result = false;
+  run_loop.PostTask([&] {
+    wait_result = ss.Wait(SocketServer::kForever, process_io);
+    run_loop.Quit();
+  });
+  run_loop.RunFor(TimeDelta::Seconds(5));
+  EXPECT_TRUE(wait_result);
 }
 
 TEST(NullSocketServerTest, TestWait) {
+  Environment env = CreateTestEnvironment();
   NullSocketServer ss;
-  int64_t start = TimeMillis();
+  Timestamp start = env.clock().CurrentTime();
   ss.Wait(TimeDelta::Millis(200), true);
   // The actual wait time is dependent on the resolution of the timer used by
   // the Event class. Allow for the event to signal ~20ms early.
-  EXPECT_GE(TimeSince(start), 180);
+  EXPECT_GE(env.clock().CurrentTime() - start, TimeDelta::Millis(180));
 }
 
 }  // namespace webrtc

From be9a88539033e2e245469705d335933058dd8bac Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Wed, 11 Feb 2026 08:03:23 -0800
Subject: [PATCH 023/472] Roll chromium_revision 974282c604..27f57c3b13
 (1583127:1583238)

Change log: https://chromium.googlesource.com/chromium/src/+log/974282c604..27f57c3b13
Full diff: https://chromium.googlesource.com/chromium/src/+/974282c604..27f57c3b13

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/d0790d1596..2b7fc786c8
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/66d45f56c2..115e1d8305
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/4b9107b4d6..c8c13c1ae7
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/0cee9b0c86..9f3cf55346
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/ec1c593db0..738fe6abe8
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/ab1d624455..785457deb3
* src/third_party/r8/cipd: TozMGRBtGJuZp31wJjyY7VkLaQ6SI8lsRnLVsSlmmogC..YvVZKKYTZDYMrvAXwMFbMGMUQaJJMb5LIirLrGcOLS0C
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/b503bfc821..39c5177180
DEPS diff: https://chromium.googlesource.com/chromium/src/+/974282c604..27f57c3b13/DEPS

BUG=None

Change-Id: I5cb6ec31f6b505955ba8638cbdce0efedfbb9cfb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/448400
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#46876}
---
 DEPS | 18 +++++++++---------
 1 file changed, 9 insertions(+), 9 deletions(-)

diff --git a/DEPS b/DEPS
index e573ffd0f3..61fb4ab1b6 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '974282c604825b4b9a3bcff508fa433e1b84dc5b',
+  'chromium_revision': '27f57c3b13c3c1d31dbbeee20c795eb5362cd15a',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@d0790d159683f591bcbea765f8f72f9c8f62f866',
+    'https://chromium.googlesource.com/chromium/src/build@2b7fc786c805ee613a9086e759fe23b3f2007770',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@6a18683f555b4ac8b05ac8395c29c84483ac9588',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@66d45f56c2145b9f0bbe28b26aae967c546bb94b',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@115e1d8305dd458457adad47faaede5ce5fa0c94',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@4b9107b4d692153ec821742de16cf049435318c7',
+    'https://chromium.googlesource.com/chromium/src/testing@c8c13c1ae7e3ca524bd6ff8bb3d19bb31e77d65d',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@0cee9b0c86b98cc0ce990de4d7bd8bf036e8f2df',
+    'https://chromium.googlesource.com/chromium/src/third_party@9f3cf553469c4138d78c1e90ab7470d49351ff56',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -488,7 +488,7 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@ec1c593db0ef36ed47634f9795022e102ec17f71',
+    'https://boringssl.googlesource.com/boringssl.git@738fe6abe82cf5f67cd324e384d095731ff5c323',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@79099fdf668ae097c9eca7052fd5c4c5de6c9098',
   'src/third_party/catapult':
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'ab1d624455b26a0094f946b88f5e772a0baa1d3a',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '785457deb38301b423ed5b0ea5424fe9b97de737',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -644,7 +644,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/r8',
-              'version': 'TozMGRBtGJuZp31wJjyY7VkLaQ6SI8lsRnLVsSlmmogC',
+              'version': 'YvVZKKYTZDYMrvAXwMFbMGMUQaJJMb5LIirLrGcOLS0C',
           },
       ],
       'condition': 'checkout_android',
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@b503bfc8210edd598e2006ceae2f9287869b533d',
+    'https://chromium.googlesource.com/chromium/src/tools@39c5177180a179097b62fc6d1f935f583e57b01b',
 
   'src/third_party/espresso': {
       'packages': [

From 0ab329d6d0100a05bcb9b172b622a5005fc7f2fd Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Wed, 11 Feb 2026 10:03:26 -0800
Subject: [PATCH 024/472] Roll chromium_revision 27f57c3b13..687500049d
 (1583238:1583344)

Change log: https://chromium.googlesource.com/chromium/src/+log/27f57c3b13..687500049d
Full diff: https://chromium.googlesource.com/chromium/src/+/27f57c3b13..687500049d

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/2b7fc786c8..436b1d9257
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/115e1d8305..23df6bf6f1
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/c8c13c1ae7..4a59c0f66d
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/9f3cf55346..adfbd21475
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/39c5177180..98da201eec
DEPS diff: https://chromium.googlesource.com/chromium/src/+/27f57c3b13..687500049d/DEPS

BUG=None

Change-Id: I5924f0c9c6a52da09e7d23325dcb383a0be2fc47
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/448440
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#46877}
---
 DEPS | 12 ++++++------
 1 file changed, 6 insertions(+), 6 deletions(-)

diff --git a/DEPS b/DEPS
index 61fb4ab1b6..6afa3b1824 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '27f57c3b13c3c1d31dbbeee20c795eb5362cd15a',
+  'chromium_revision': '687500049d83429ebd244bb3172838c577af8890',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@2b7fc786c805ee613a9086e759fe23b3f2007770',
+    'https://chromium.googlesource.com/chromium/src/build@436b1d92571250afe5bcbfb23e2d5f5340e6c02d',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@6a18683f555b4ac8b05ac8395c29c84483ac9588',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@115e1d8305dd458457adad47faaede5ce5fa0c94',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@23df6bf6f1fc0e4bac3b28e0900f9205b9a48071',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@c8c13c1ae7e3ca524bd6ff8bb3d19bb31e77d65d',
+    'https://chromium.googlesource.com/chromium/src/testing@4a59c0f66d754ed8031c21d83b2b56f83951fa0c',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@9f3cf553469c4138d78c1e90ab7470d49351ff56',
+    'https://chromium.googlesource.com/chromium/src/third_party@adfbd21475846f31ba34d6490171e2d319adef91',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@39c5177180a179097b62fc6d1f935f583e57b01b',
+    'https://chromium.googlesource.com/chromium/src/tools@98da201eec3bf5d025ad578773624ef69e0990f1',
 
   'src/third_party/espresso': {
       'packages': [

From a5a020fd315a5c55e79154fabfb9b1a084a73105 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Erik=20Spr=C3=A5ng?= <sprang@webrtc.org>
Date: Wed, 11 Feb 2026 16:56:34 +0100
Subject: [PATCH 025/472] Fix
 VideoStreamEncoderTest.QualityScalerTriggersOnFrameDropping on iOS
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

https://webrtc-review.git.corp.google.com/c/src/+/447001 introduced a
test case which verifies plumbing of encoder frame drop signals all
the way down to the quality scaler. Turns out that on iOS that feature
isn't used so the test will fail - and no currently enabled test bot
exercised that code path.

This CL checks for the existence of that feature and skips the test if
it is absent. Verified by running with a hacked local
`#define WEBRTC_IOS` and by force-including an extra try bot.

Bug: webrtc:467444018
Change-Id: I38c28a16553bcc22f1c907a24f0399322ede8d3d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/448340
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46878}
---
 video/BUILD.gn                         |  1 +
 video/video_stream_encoder_unittest.cc | 21 ++++++++++++++++++++-
 2 files changed, 21 insertions(+), 1 deletion(-)

diff --git a/video/BUILD.gn b/video/BUILD.gn
index 5c3217a8b7..5dca4b5928 100644
--- a/video/BUILD.gn
+++ b/video/BUILD.gn
@@ -973,6 +973,7 @@ if (rtc_include_tests) {
       "../rtc_base/containers:flat_map",
       "../rtc_base/experiments:alr_experiment",
       "../rtc_base/experiments:encoder_info_settings",
+      "../rtc_base/experiments:quality_scaling_experiment",
       "../rtc_base/experiments:rate_control_settings",
       "../rtc_base/synchronization:mutex",
       "../system_wrappers",
diff --git a/video/video_stream_encoder_unittest.cc b/video/video_stream_encoder_unittest.cc
index 016d3c2f09..dae6673a9f 100644
--- a/video/video_stream_encoder_unittest.cc
+++ b/video/video_stream_encoder_unittest.cc
@@ -96,6 +96,7 @@
 #include "rtc_base/checks.h"
 #include "rtc_base/event.h"
 #include "rtc_base/experiments/encoder_info_settings.h"
+#include "rtc_base/experiments/quality_scaling_experiment.h"
 #include "rtc_base/experiments/rate_control_settings.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/ref_counted_object.h"
@@ -2040,10 +2041,28 @@ TEST_F(VideoStreamEncoderTest, DropsFramesOnBitrateOvershoot) {
 }
 
 TEST_F(VideoStreamEncoderTest, QualityScalerTriggersOnFrameDropping) {
+  // Set the quality scaler checker frequency high to prevent flakiness.
+  FieldTrials trials = SetFieldTrial("WebRTC-Video-QualityScalerSettings",
+                                     "sampling_period_ms:250");
+
+  // If the quality scaler isn't configured to treat encoder frame drops
+  // as a signal for scaling, then this test doesn't make sense to run.
+  std::optional<QualityScalingExperiment::Config> config =
+      QualityScalingExperiment::GetConfig(trials);
+  if (!config.has_value() || !config->use_all_drop_reasons) {
+    video_stream_encoder_->Stop();
+    GTEST_SKIP() << "QualityScalerExperiment does not take encoder drop "
+                    "reasons into account, skipping "
+                    "QualityScalerTriggersOnFrameDropping.";
+  }
+
   // Set resolution high so there is a lower resolution to adapt to.
   codec_width_ = 1280;
   codec_height_ = 720;
-  ConfigureEncoder(video_encoder_config_.Copy());
+  ConfigureEncoder(
+      video_encoder_config_.Copy(),
+      VideoStreamEncoder::BitrateAllocationCallbackType::kVideoLayersAllocation,
+      /* num_cores= */ 1, &trials);
   video_stream_encoder_->OnBitrateUpdatedAndWaitForManagedResources(
       kTargetBitrate, kTargetBitrate, 0, 0, 0);
 

From 5700d8f30e6d976759d9e21bf93cc2792f022dd6 Mon Sep 17 00:00:00 2001
From: Jeremy Leconte <jleconte@webrtc.org>
Date: Wed, 11 Feb 2026 14:19:17 +0100
Subject: [PATCH 026/472] Re-enable ios debug simulator bot.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

* re-enable the presubmit bot
* re-enable the postsubmit bot for LKGR

Change-Id: Ibc1fd29b24df08973c1c9e876049c55efa8a4556
Bug: webrtc:460264453
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/448343
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#46879}
---
 infra/config/builders.star                     | 5 +----
 infra/config/generated/luci/commit-queue.cfg   | 6 ++++++
 infra/config/generated/luci/cr-buildbucket.cfg | 2 +-
 infra/config/generated/luci/luci-notify.cfg    | 5 +++++
 infra/config/generated/luci/project.cfg        | 2 +-
 5 files changed, 14 insertions(+), 6 deletions(-)

diff --git a/infra/config/builders.star b/infra/config/builders.star
index 8eb0a39d2a..52518ef480 100644
--- a/infra/config/builders.star
+++ b/infra/config/builders.star
@@ -20,8 +20,6 @@ WEBRTC_GIT = "https://webrtc.googlesource.com/src"
 # useful when a failure can be safely ignored while fixing it without
 # blocking the LKGR finder on it.
 skipped_lkgr_bots = [
-    # TODO: https://issues.webrtc.org/460264453 - Re-enable when reliable
-    "iOS Debug (simulator)",
 ]
 
 lkgr_builders = []
@@ -342,8 +340,7 @@ ios_builder("iOS64 Release", "iOS|arm64|rel")
 ios_try_job("ios_compile_arm64_rel")
 ios_builder("iOS Debug (simulator)", "iOS|x64|sim")
 
-# TODO: https://issues.webrtc.org/460264453 - Re-enable when reliable
-ios_try_job("ios_dbg_simulator", cq = None)
+ios_try_job("ios_dbg_simulator")
 ios_builder("iOS API Framework Builder", "iOS|fat|size", recipe = "ios_api_framework", prioritized = True)
 ios_try_job("ios_api_framework", recipe = "ios_api_framework")
 
diff --git a/infra/config/generated/luci/commit-queue.cfg b/infra/config/generated/luci/commit-queue.cfg
index 7834a2f9f6..b7c3a0c960 100644
--- a/infra/config/generated/luci/commit-queue.cfg
+++ b/infra/config/generated/luci/commit-queue.cfg
@@ -90,6 +90,9 @@ config_groups {
       builders {
         name: "webrtc/try/ios_compile_arm64_rel"
       }
+      builders {
+        name: "webrtc/try/ios_dbg_simulator"
+      }
       builders {
         name: "webrtc/try/iwyu_verifier"
       }
@@ -267,6 +270,9 @@ config_groups {
       builders {
         name: "webrtc/try/ios_compile_arm64_rel"
       }
+      builders {
+        name: "webrtc/try/ios_dbg_simulator"
+      }
       builders {
         name: "webrtc/try/iwyu_verifier"
       }
diff --git a/infra/config/generated/luci/cr-buildbucket.cfg b/infra/config/generated/luci/cr-buildbucket.cfg
index fb756f5ddb..04ad068f3c 100644
--- a/infra/config/generated/luci/cr-buildbucket.cfg
+++ b/infra/config/generated/luci/cr-buildbucket.cfg
@@ -1185,7 +1185,7 @@ buckets {
         cipd_package: "infra/recipe_bundles/chromium.googlesource.com/chromium/tools/build"
         cipd_version: "refs/heads/main"
         properties_j: "$recipe_engine/resultdb/test_presentation:{\"column_keys\":[],\"grouping_keys\":[\"status\",\"v.test_suite\"]}"
-        properties_j: "config:{\"allowed_gap\":150,\"allowed_lag\":9,\"buckets\":{\"chromium/webrtc.fyi\":{\"builders\":[\"WebRTC Chromium FYI Android Builder (dbg)\",\"WebRTC Chromium FYI Android Builder\",\"WebRTC Chromium FYI Android Tester\",\"WebRTC Chromium FYI Linux Builder (dbg)\",\"WebRTC Chromium FYI Linux Builder\",\"WebRTC Chromium FYI Linux Tester\",\"WebRTC Chromium FYI Mac Builder (dbg)\",\"WebRTC Chromium FYI Mac Builder\",\"WebRTC Chromium FYI Mac Tester\",\"WebRTC Chromium FYI Win Builder (dbg)\",\"WebRTC Chromium FYI Win Builder\",\"WebRTC Chromium FYI Win Tester\"]},\"webrtc/ci\":{\"builders\":[\"Android32\",\"Android32 (dbg)\",\"Android32 (more configs)\",\"Android32 Builder x86\",\"Android32 Builder x86 (dbg)\",\"Android64\",\"Android64 Builder x64 (dbg)\",\"Fuchsia Release\",\"Linux (more configs)\",\"Linux Asan\",\"Linux MSan\",\"Linux Tsan v2\",\"Linux UBSan\",\"Linux UBSan vptr\",\"Linux32 Debug\",\"Linux32 Debug (ARM)\",\"Linux32 Release\",\"Linux32 Release (ARM)\",\"Linux64 Debug\",\"Linux64 Debug (ARM)\",\"Linux64 Release\",\"Linux64 Release (ARM)\",\"Linux64 Release (Libfuzzer)\",\"Mac Asan\",\"Mac64 Debug\",\"Mac64 Release\",\"MacARM64 M1 Release\",\"Win (more configs)\",\"Win32 Debug (Clang)\",\"Win32 Release (Clang)\",\"Win64 ASan\",\"Win64 Debug (Clang)\",\"Win64 Release (Clang)\",\"iOS API Framework Builder\",\"iOS64 Debug\",\"iOS64 Release\"]}},\"project\":\"webrtc\",\"source_url\":\"https://webrtc.googlesource.com/src\",\"status_url\":\"https://webrtc-status.appspot.com\"}"
+        properties_j: "config:{\"allowed_gap\":150,\"allowed_lag\":9,\"buckets\":{\"chromium/webrtc.fyi\":{\"builders\":[\"WebRTC Chromium FYI Android Builder (dbg)\",\"WebRTC Chromium FYI Android Builder\",\"WebRTC Chromium FYI Android Tester\",\"WebRTC Chromium FYI Linux Builder (dbg)\",\"WebRTC Chromium FYI Linux Builder\",\"WebRTC Chromium FYI Linux Tester\",\"WebRTC Chromium FYI Mac Builder (dbg)\",\"WebRTC Chromium FYI Mac Builder\",\"WebRTC Chromium FYI Mac Tester\",\"WebRTC Chromium FYI Win Builder (dbg)\",\"WebRTC Chromium FYI Win Builder\",\"WebRTC Chromium FYI Win Tester\"]},\"webrtc/ci\":{\"builders\":[\"Android32\",\"Android32 (dbg)\",\"Android32 (more configs)\",\"Android32 Builder x86\",\"Android32 Builder x86 (dbg)\",\"Android64\",\"Android64 Builder x64 (dbg)\",\"Fuchsia Release\",\"Linux (more configs)\",\"Linux Asan\",\"Linux MSan\",\"Linux Tsan v2\",\"Linux UBSan\",\"Linux UBSan vptr\",\"Linux32 Debug\",\"Linux32 Debug (ARM)\",\"Linux32 Release\",\"Linux32 Release (ARM)\",\"Linux64 Debug\",\"Linux64 Debug (ARM)\",\"Linux64 Release\",\"Linux64 Release (ARM)\",\"Linux64 Release (Libfuzzer)\",\"Mac Asan\",\"Mac64 Debug\",\"Mac64 Release\",\"MacARM64 M1 Release\",\"Win (more configs)\",\"Win32 Debug (Clang)\",\"Win32 Release (Clang)\",\"Win64 ASan\",\"Win64 Debug (Clang)\",\"Win64 Release (Clang)\",\"iOS API Framework Builder\",\"iOS Debug (simulator)\",\"iOS64 Debug\",\"iOS64 Release\"]}},\"project\":\"webrtc\",\"source_url\":\"https://webrtc.googlesource.com/src\",\"status_url\":\"https://webrtc-status.appspot.com\"}"
         properties_j: "lkgr_status_gs_path:\"chromium-webrtc/lkgr-status\""
         properties_j: "project:\"webrtc\""
         properties_j: "ref:\"refs/heads/lkgr\""
diff --git a/infra/config/generated/luci/luci-notify.cfg b/infra/config/generated/luci/luci-notify.cfg
index 3cc8bc05bf..416b6f0f84 100644
--- a/infra/config/generated/luci/luci-notify.cfg
+++ b/infra/config/generated/luci/luci-notify.cfg
@@ -943,6 +943,11 @@ notifiers {
     name: "iOS Debug (simulator)"
     repository: "https://webrtc.googlesource.com/src"
   }
+  tree_closers {
+    tree_status_host: "webrtc-status.appspot.com"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp_exclude: ".*\\(experimental\\).*"
+  }
 }
 notifiers {
   notifications {
diff --git a/infra/config/generated/luci/project.cfg b/infra/config/generated/luci/project.cfg
index 6ec459c974..aee5e5ca68 100644
--- a/infra/config/generated/luci/project.cfg
+++ b/infra/config/generated/luci/project.cfg
@@ -7,7 +7,7 @@
 name: "webrtc"
 access: "group:all"
 lucicfg {
-  version: "1.46.1"
+  version: "1.46.2"
   package_name: "@webrtc-project"
   package_dir: "../.."
   config_dir: "generated/luci"

From ccd1d8875022db9a804600f01ee1084b9183856b Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Wed, 11 Feb 2026 14:02:29 -0800
Subject: [PATCH 027/472] Roll chromium_revision 687500049d..cabe765c34
 (1583344:1583505)

Change log: https://chromium.googlesource.com/chromium/src/+log/687500049d..cabe765c34
Full diff: https://chromium.googlesource.com/chromium/src/+/687500049d..cabe765c34

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/436b1d9257..664be89a22
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/23df6bf6f1..d68e57b862
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/4a59c0f66d..175e2a4da7
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/adfbd21475..6624d10a4b
* src/third_party/androidx/cipd: us_8XGxuKdmIoJEbqTnp_Ehw-VEaxJIVTzMy2h0fnhwC..Cl-ySzwc8_wt7xkWL9dKKHbvVRhOHTgGANeH3JHEB3cC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/738fe6abe8..932fb7389f
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/a476f554f8..4fe7ca3037
* src/third_party/fuzztest/src: https://chromium.googlesource.com/external/github.com/google/fuzztest.git/+log/54b2ccd592..fb29fbd005
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/3151eefe26..f9b8a88e84
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/785457deb3..7475cfd103
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/98da201eec..1f457fb242
DEPS diff: https://chromium.googlesource.com/chromium/src/+/687500049d..cabe765c34/DEPS

BUG=None

Change-Id: I6b3794af10a75dfe870f5d6650afce276d8a3ec1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/448520
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#46880}
---
 DEPS | 24 ++++++++++++------------
 1 file changed, 12 insertions(+), 12 deletions(-)

diff --git a/DEPS b/DEPS
index 6afa3b1824..5e688ce7c7 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '687500049d83429ebd244bb3172838c577af8890',
+  'chromium_revision': 'cabe765c3407b8d14986c515edcfe81ed96319e0',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@436b1d92571250afe5bcbfb23e2d5f5340e6c02d',
+    'https://chromium.googlesource.com/chromium/src/build@664be89a2294ef5491c87eb834443054385216b0',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@6a18683f555b4ac8b05ac8395c29c84483ac9588',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@23df6bf6f1fc0e4bac3b28e0900f9205b9a48071',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@d68e57b8626689925505a9d79d54aa841cb0a387',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@4a59c0f66d754ed8031c21d83b2b56f83951fa0c',
+    'https://chromium.googlesource.com/chromium/src/testing@175e2a4da7d46983fa2d21a56e3c597b333f3a99',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@adfbd21475846f31ba34d6490171e2d319adef91',
+    'https://chromium.googlesource.com/chromium/src/third_party@6624d10a4b14fab6670316f8e0033a2aea3fd7c7',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -488,11 +488,11 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@738fe6abe82cf5f67cd324e384d095731ff5c323',
+    'https://boringssl.googlesource.com/boringssl.git@932fb7389fbd5438227c05e53a345e3e4dfc2a9a',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@79099fdf668ae097c9eca7052fd5c4c5de6c9098',
   'src/third_party/catapult':
-    'https://chromium.googlesource.com/catapult.git@a476f554f8865b7d162ec9f1ad8aae1eab38e048',
+    'https://chromium.googlesource.com/catapult.git@4fe7ca3037f5f5e7227f0f0d0d6eae98d0ebf840',
   'src/third_party/ced/src': {
     'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5',
   },
@@ -511,7 +511,7 @@ deps = {
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@3151eefe26976c60f2ddd9bf74513dd0152136ea',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@f9b8a88e84441a237414dce860e3f6214a64f8bf',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -593,7 +593,7 @@ deps = {
   'src/third_party/libFuzzer/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt/lib/fuzzer.git@bea408a6e01f0f7e6c82a43121fe3af4506c932e',
   'src/third_party/fuzztest/src':
-    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@54b2ccd59262eaeab887c5798ae6b12b3bb7b04b',
+    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@fb29fbd0058c84ed0ae7b3b50ad2909a68fd0360',
   'src/third_party/libjpeg_turbo':
     'https://chromium.googlesource.com/chromium/deps/libjpeg_turbo.git@6bb85251a8382b5e07f635a981ac685cc5ab5053',
   'src/third_party/libsrtp':
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '785457deb38301b423ed5b0ea5424fe9b97de737',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '7475cfd1038ac63f0884e1f096fce054fa01c3b3',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@98da201eec3bf5d025ad578773624ef69e0990f1',
+    'https://chromium.googlesource.com/chromium/src/tools@1f457fb24256f08614f80b11a1764c5ebf5188a0',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'us_8XGxuKdmIoJEbqTnp_Ehw-VEaxJIVTzMy2h0fnhwC',
+          'version': 'Cl-ySzwc8_wt7xkWL9dKKHbvVRhOHTgGANeH3JHEB3cC',
       },
     ],
     'condition': 'checkout_android and non_git_source',

From a62566a25e16149f58d3c5fdd3da6715a45cae11 Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Wed, 11 Feb 2026 20:08:04 -0800
Subject: [PATCH 028/472] Update WebRTC code version (2026-02-12T04:07:43).

Bug: None
Change-Id: Ie6ce8d6dc397068ff859f3b89e472a44e5c353aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/448581
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#46881}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index 2e5114a394..4d2f88368d 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-02-11T04:02:53";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-02-12T04:07:42";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From 347fb3e8aa5ef57237a4555944bac3413edadaf8 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Thu, 12 Feb 2026 11:11:59 +0100
Subject: [PATCH 029/472] docs: Add pointer nullability annotation guidelines

Adds recommendations for explicitly annotating nullable pointers with
absl_nullable and absl_nonnull to prevent accidental use of invalid
null values.

No-try: true
Bug: none
Change-Id: Ic27cbfe4522c16fbd0a4206e93e0e7cf15bf0c0e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/448500
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46882}
---
 g3doc/implementation_basics.md | 17 +++++++++++++++++
 1 file changed, 17 insertions(+)

diff --git a/g3doc/implementation_basics.md b/g3doc/implementation_basics.md
index a1102a0979..6860638a94 100644
--- a/g3doc/implementation_basics.md
+++ b/g3doc/implementation_basics.md
@@ -157,3 +157,20 @@ Before uploading changes to gerrit for review, make sure to complete the followi
 * Ensure that a *full build* of all binaries passes. This will save time and resources.
 * If at any of the above steps changes had to be made, go back the first step and repeat
   until no changes are required at every step.
+
+## Pointer Nullability Annotations
+To ensure high-quality, machine-checkable C++ code in WebRTC, adhere to the following rules regarding pointer nullability.
+
+* **Core Principle**: Prevent accidental use of invalid null values by explicitly declaring the nullability contract of all raw and smart pointers.
+
+* **Classify Every Pointer**: Avoid leaving pointers in the "Unknown" (unannotated) state. Actively classify whether null is a valid semantic state for every pointer you write or modify.
+
+* **Annotate Nullable Exceptions (`absl_nullable`)**: When null is a valid and expected value at the point of use, you must explicitly annotate the pointer.
+    * Raw Pointers: Use the format `T* absl_nullable` (e.g., `Widget* absl_nullable widget`).
+    * Smart Pointers: Use the format `absl_nullable std::unique_ptr<T>` (e.g., `absl_nullable std::unique_ptr<Widget> widget`).
+
+* **Enforce Strict Requirements (`absl_nonnull`)**: When a pointer must fundamentally never be null, enforce this contract explicitly.
+    * Raw Pointers: Use the format `T* absl_nonnull` but for non-null raw pointers prefer to use a reference instead.
+    * Smart Pointers: Use the format `absl_nonnull std::unique_ptr<T>`.
+
+* **Leverage Machine Checking**: Treat these annotations as strict contracts. Rely on them to surface potential null-dereference errors or contract violations during static analysis and compilation.
\ No newline at end of file

From 20101b211bdf290d52bbf04822593364deaee443 Mon Sep 17 00:00:00 2001
From: AhmadDurrani579 <ahmaddurranitrg@gmail.com>
Date: Wed, 11 Feb 2026 21:03:18 +0000
Subject: [PATCH 030/472] Fix std::tmpnam deprecation in event_log_visualizer
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This CL replaces the deprecated std::tmpnam() function with
webrtc::test::TempFilename() to avoid compiler warnings on macOS.

This change requires:
1. Marking rtc_executable("event_log_visualizer") as testonly = true.
2. Adding dependencies on //test:test_support and //test:fileutils.
3. Removing the -Wno-deprecated-declarations suppression flag.

Bug: webrtc:14248
Change-Id: I8989a1839fabf70549843fd37d1c35df6c470efd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/448160
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46883}
---
 rtc_tools/BUILD.gn                         |  8 ++++----
 rtc_tools/rtc_event_log_visualizer/main.cc | 10 ++++++----
 2 files changed, 10 insertions(+), 8 deletions(-)

diff --git a/rtc_tools/BUILD.gn b/rtc_tools/BUILD.gn
index c14c29d018..1515936783 100644
--- a/rtc_tools/BUILD.gn
+++ b/rtc_tools/BUILD.gn
@@ -390,9 +390,9 @@ if (!build_with_chromium) {
       deps = [
         ":chart_proto",
         ":event_log_visualizer_bindings",
+        "../test:fileutils",
         "//rtc_base:protobuf_utils",
         "//rtc_base/system:file_wrapper",
-        "//test:fileutils",
         "//test:test_support",
         "//third_party/abseil-cpp/absl/strings:string_view",
       ]
@@ -444,9 +444,8 @@ if (rtc_include_tests) {
   if (!build_with_chromium) {
     if (rtc_enable_protobuf) {
       rtc_executable("event_log_visualizer") {
-        # TODO(bugs.webrtc.org/14248): Remove once usage of std::tmpnam
-        # is removed (in favor of in memory InputAudioFile.
-        cflags_cc = [ "-Wno-deprecated-declarations" ]
+        testonly = true
+
         sources = [
           "rtc_event_log_visualizer/conversational_speech_en.h",
           "rtc_event_log_visualizer/main.cc",
@@ -466,6 +465,7 @@ if (rtc_include_tests) {
           "../rtc_base:checks",
           "../rtc_base:logging",
           "../rtc_base:protobuf_utils",
+          "../test:fileutils",
           "//third_party/abseil-cpp/absl/algorithm:container",
           "//third_party/abseil-cpp/absl/flags:config",
           "//third_party/abseil-cpp/absl/flags:flag",
diff --git a/rtc_tools/rtc_event_log_visualizer/main.cc b/rtc_tools/rtc_event_log_visualizer/main.cc
index 03512b185b..6105e01f70 100644
--- a/rtc_tools/rtc_event_log_visualizer/main.cc
+++ b/rtc_tools/rtc_event_log_visualizer/main.cc
@@ -41,6 +41,7 @@
 #include "rtc_tools/rtc_event_log_visualizer/conversational_speech_en.h"
 #include "rtc_tools/rtc_event_log_visualizer/plot_base.h"
 #include "rtc_tools/rtc_event_log_visualizer/proto/chart.pb.h"
+#include "test/testsupport/file_utils.h"
 
 ABSL_FLAG(std::string,
           plot,
@@ -215,10 +216,11 @@ int main(int argc, char* argv[]) {
   if (!absl::GetFlag(FLAGS_wav_filename).empty()) {
     wav_path = absl::GetFlag(FLAGS_wav_filename);
   } else {
-    // TODO(bugs.webrtc.org/14248): Remove the need to generate a file
-    // and read the file directly from memory.
-    wav_path = std::tmpnam(nullptr);
-    std::ofstream out_wav_file(wav_path);
+    // TODO(bugs.webrtc.org/14248): Remove the need to generate a file.
+    wav_path = webrtc::test::TempFilename(webrtc::test::OutputPath(),
+                                          "temp_speech.wav");
+
+    std::ofstream out_wav_file(wav_path, std::ios::binary);
     out_wav_file.write(
         reinterpret_cast<char*>(&webrtc::conversational_speech_en_wav[0]),
         webrtc::conversational_speech_en_wav_len);

From 814e92fecd2858eece25c4ca7f640934d1c9711a Mon Sep 17 00:00:00 2001
From: Mirko Bonadei <mbonadei@webrtc.org>
Date: Thu, 12 Feb 2026 10:34:52 +0000
Subject: [PATCH 031/472] Switch to python3 as default script_executable.

Bug: b/483338006
Change-Id: I4437fd821f6ec534caf9ba7620bd74072ca6c644
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/448640
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46884}
---
 .gn | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/.gn b/.gn
index ae9661e2de..4706638e3e 100644
--- a/.gn
+++ b/.gn
@@ -12,8 +12,8 @@ import("//build/dotfile_settings.gni")
 buildconfig = "//build/config/BUILDCONFIG.gn"
 
 # The python interpreter to use by default. On Windows, this will look
-# for vpython3.exe and vpython3.bat.
-script_executable = "vpython3"
+# for python3.exe and python3.bat.
+script_executable = "python3"
 
 # The secondary source root is a parallel directory tree where
 # GN build files are placed when they can not be placed directly

From 1ccc59d5c94aafef555882b50d242a8efe12caee Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Mon, 9 Feb 2026 12:09:30 +0100
Subject: [PATCH 032/472] Spanify CopyOnWriteBuffer

Detach CopyOnWriteBuffer implementation from Buffer by creating a dedicated small type that has non-mutable capacity and doesn't have size. Each CoWBuffer object sharing the same buffer has its own members that specify which memory can be safe to write or read.
Delegate all pointer arithmetic to ArrayView silencing unsafe-buffer warnings.

Bug: webrtc:478086887
Change-Id: I2ee0c0fce6c4264c5d44e0f227908fc9c9afadc7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/444702
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46885}
---
 rtc_base/BUILD.gn                |   3 +
 rtc_base/copy_on_write_buffer.cc |  99 ++++++++++++++---------
 rtc_base/copy_on_write_buffer.h  | 135 ++++++++++++-------------------
 unsafe_buffers_paths.txt         |   2 -
 4 files changed, 115 insertions(+), 124 deletions(-)

diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
index d02a8341e1..962d992edd 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -185,8 +185,11 @@ rtc_library("copy_on_write_buffer") {
     ":checks",
     ":refcount",
     ":type_traits",
+    "../api:array_view",
+    "../api:make_ref_counted",
     "../api:scoped_refptr",
     "system:rtc_export",
+    "//third_party/abseil-cpp/absl/algorithm:container",
     "//third_party/abseil-cpp/absl/strings:string_view",
   ]
 }
diff --git a/rtc_base/copy_on_write_buffer.cc b/rtc_base/copy_on_write_buffer.cc
index 160ce94d05..be1fc302a6 100644
--- a/rtc_base/copy_on_write_buffer.cc
+++ b/rtc_base/copy_on_write_buffer.cc
@@ -12,14 +12,32 @@
 
 #include <algorithm>
 #include <cstddef>
+#include <cstdint>
 #include <cstring>
+#include <memory>
 #include <utility>
 
+#include "absl/algorithm/container.h"
 #include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "api/make_ref_counted.h"
+#include "api/scoped_refptr.h"
 #include "rtc_base/checks.h"
+#include "rtc_base/ref_counted_object.h"
 
 namespace webrtc {
 
+CopyOnWriteBuffer::RawBuffer::RawBuffer(size_t size)
+    : size_(size), data_(std::make_unique_for_overwrite<uint8_t[]>(size)) {}
+
+scoped_refptr<CopyOnWriteBuffer::RefCountedBuffer>
+CopyOnWriteBuffer::CreateBuffer(size_t capacity) {
+  if (capacity == 0) {
+    return nullptr;
+  }
+  return make_ref_counted<RefCountedBuffer>(capacity);
+}
+
 CopyOnWriteBuffer::CopyOnWriteBuffer() : offset_(0), size_(0) {
   RTC_DCHECK(IsConsistent());
 }
@@ -38,18 +56,13 @@ CopyOnWriteBuffer::CopyOnWriteBuffer(absl::string_view s)
     : CopyOnWriteBuffer(s.data(), s.length()) {}
 
 CopyOnWriteBuffer::CopyOnWriteBuffer(size_t size)
-    : buffer_(size > 0 ? new RefCountedBuffer(size, size) : nullptr),
-      // note - the RefCountedBuffer will be created uninitialized.
-      offset_(0),
-      size_(size) {
+    : buffer_(CreateBuffer(/*capacity=*/size)), offset_(0), size_(size) {
+  // note - the RefCountedBuffer will be created uninitialized.
   RTC_DCHECK(IsConsistent());
 }
 
 CopyOnWriteBuffer::CopyOnWriteBuffer(size_t size, size_t capacity)
-    : buffer_(size > 0 || capacity > 0 ? new RefCountedBuffer(size, capacity)
-                                       : nullptr),
-      offset_(0),
-      size_(size) {
+    : buffer_(CreateBuffer(std::max(size, capacity))), offset_(0), size_(size) {
   RTC_DCHECK(IsConsistent());
 }
 
@@ -65,39 +78,19 @@ bool CopyOnWriteBuffer::operator==(const CopyOnWriteBuffer& buf) const {
 
 void CopyOnWriteBuffer::SetSize(size_t size) {
   RTC_DCHECK(IsConsistent());
-  if (!buffer_) {
-    if (size > 0) {
-      buffer_ = new RefCountedBuffer(size, size);
-      // Note - the new buffer will be created uninitialized.
-      offset_ = 0;
-      size_ = size;
-    }
-    RTC_DCHECK(IsConsistent());
-    return;
-  }
-
   if (size <= size_) {
     size_ = size;
     return;
   }
 
   UnshareAndEnsureCapacity(std::max(capacity(), size));
-  buffer_->SetSize(size + offset_);
   size_ = size;
   RTC_DCHECK(IsConsistent());
 }
 
 void CopyOnWriteBuffer::EnsureCapacity(size_t new_capacity) {
   RTC_DCHECK(IsConsistent());
-  if (!buffer_) {
-    if (new_capacity > 0) {
-      buffer_ = new RefCountedBuffer(0, new_capacity);
-      offset_ = 0;
-      size_ = 0;
-    }
-    RTC_DCHECK(IsConsistent());
-    return;
-  } else if (new_capacity <= capacity()) {
+  if (new_capacity <= capacity()) {
     return;
   }
 
@@ -109,24 +102,56 @@ void CopyOnWriteBuffer::Clear() {
   if (!buffer_)
     return;
 
-  if (buffer_->HasOneRef()) {
-    buffer_->Clear();
-  } else {
-    buffer_ = new RefCountedBuffer(0, capacity());
+  if (!buffer_->HasOneRef()) {
+    buffer_ = CreateBuffer(capacity());
   }
   offset_ = 0;
   size_ = 0;
   RTC_DCHECK(IsConsistent());
 }
 
-void CopyOnWriteBuffer::UnshareAndEnsureCapacity(size_t new_capacity) {
-  if (buffer_->HasOneRef() && new_capacity <= capacity()) {
+void CopyOnWriteBuffer::Set(ArrayView<const uint8_t> data) {
+  RTC_DCHECK(IsConsistent());
+  if (data.empty()) {
+    offset_ = 0;
+    size_ = 0;
     return;
   }
 
-  buffer_ =
-      new RefCountedBuffer(buffer_->data() + offset_, size_, new_capacity);
+  if (buffer_ == nullptr || !buffer_->HasOneRef() ||
+      buffer_->capacity() < data.size()) {
+    buffer_ = CreateBuffer(std::max(data.size(), capacity()));
+  }
+  absl::c_copy(data, buffer_->data().begin());
   offset_ = 0;
+  size_ = data.size();
+
+  RTC_DCHECK(IsConsistent());
+}
+
+void CopyOnWriteBuffer::Append(ArrayView<const uint8_t> data) {
+  RTC_DCHECK(IsConsistent());
+  if (data.empty()) {
+    return;
+  }
+
+  UnshareAndEnsureCapacity(std::max(capacity(), size_ + data.size()));
+  absl::c_copy(data, buffer_->data().subspan(offset_ + size_).begin());
+  size_ += data.size();
+
+  RTC_DCHECK(IsConsistent());
+}
+
+void CopyOnWriteBuffer::UnshareAndEnsureCapacity(size_t new_capacity) {
+  if (buffer_ != nullptr && buffer_->HasOneRef() &&
+      new_capacity <= capacity()) {
+    return;
+  }
+
+  scoped_refptr<RefCountedBuffer> b = CreateBuffer(new_capacity);
+  absl::c_copy(AsConstSpan(), b->data().begin());
+  offset_ = 0;
+  buffer_ = std::move(b);
   RTC_DCHECK(IsConsistent());
 }
 
diff --git a/rtc_base/copy_on_write_buffer.h b/rtc_base/copy_on_write_buffer.h
index be8a261173..73aeb6d371 100644
--- a/rtc_base/copy_on_write_buffer.h
+++ b/rtc_base/copy_on_write_buffer.h
@@ -11,14 +11,14 @@
 #ifndef RTC_BASE_COPY_ON_WRITE_BUFFER_H_
 #define RTC_BASE_COPY_ON_WRITE_BUFFER_H_
 
-#include <stdint.h>
-
-#include <algorithm>
-#include <cstring>
+#include <cstddef>
+#include <cstdint>
+#include <memory>
 #include <type_traits>
 #include <utility>
 
 #include "absl/strings/string_view.h"
+#include "api/array_view.h"
 #include "api/scoped_refptr.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/checks.h"
@@ -34,8 +34,18 @@ class RTC_EXPORT CopyOnWriteBuffer {
   CopyOnWriteBuffer();
   // Share the data with an existing buffer.
   CopyOnWriteBuffer(const CopyOnWriteBuffer& buf);
+  CopyOnWriteBuffer& operator=(const CopyOnWriteBuffer& buf) = default;
+
   // Move contents from an existing buffer.
   CopyOnWriteBuffer(CopyOnWriteBuffer&& buf) noexcept;
+  CopyOnWriteBuffer& operator=(CopyOnWriteBuffer&& buf) {
+    RTC_DCHECK(IsConsistent());
+    RTC_DCHECK(buf.IsConsistent());
+    buffer_ = std::exchange(buf.buffer_, nullptr);
+    offset_ = std::exchange(buf.offset_, 0);
+    size_ = std::exchange(buf.size_, 0);
+    return *this;
+  }
 
   // Construct a buffer from a string, convenient for unittests.
   explicit CopyOnWriteBuffer(absl::string_view s);
@@ -56,11 +66,7 @@ class RTC_EXPORT CopyOnWriteBuffer {
                 internal::BufferCompat<uint8_t, T>::value>::type* = nullptr>
   CopyOnWriteBuffer(const T* data, size_t size, size_t capacity)
       : CopyOnWriteBuffer(size, capacity) {
-    if (buffer_) {
-      std::memcpy(buffer_->data(), data, size);
-      offset_ = 0;
-      size_ = size;
-    }
+    SetData(data, size);
   }
 
   // Construct a buffer from the contents of an array.
@@ -115,7 +121,7 @@ class RTC_EXPORT CopyOnWriteBuffer {
       return nullptr;
     }
     UnshareAndEnsureCapacity(capacity());
-    return buffer_->data<T>() + offset_;
+    return reinterpret_cast<T*>(buffer_->data().subspan(offset_).data());
   }
 
   // Get const pointer to the data. This will not create a copy of the
@@ -124,11 +130,7 @@ class RTC_EXPORT CopyOnWriteBuffer {
             typename std::enable_if<
                 internal::BufferCompat<uint8_t, T>::value>::type* = nullptr>
   const T* cdata() const {
-    RTC_DCHECK(IsConsistent());
-    if (!buffer_) {
-      return nullptr;
-    }
-    return buffer_->data<T>() + offset_;
+    return reinterpret_cast<const T*>(AsConstSpan().data());
   }
 
   bool empty() const { return size_ == 0; }
@@ -143,40 +145,14 @@ class RTC_EXPORT CopyOnWriteBuffer {
     return buffer_ ? buffer_->capacity() - offset_ : 0;
   }
 
-  const uint8_t* begin() const { return data(); }
-  const uint8_t* end() const { return data() + size_; }
-
-  CopyOnWriteBuffer& operator=(const CopyOnWriteBuffer& buf) {
-    RTC_DCHECK(IsConsistent());
-    RTC_DCHECK(buf.IsConsistent());
-    if (&buf != this) {
-      buffer_ = buf.buffer_;
-      offset_ = buf.offset_;
-      size_ = buf.size_;
-    }
-    return *this;
-  }
-
-  CopyOnWriteBuffer& operator=(CopyOnWriteBuffer&& buf) {
-    RTC_DCHECK(IsConsistent());
-    RTC_DCHECK(buf.IsConsistent());
-    buffer_ = std::move(buf.buffer_);
-    offset_ = buf.offset_;
-    size_ = buf.size_;
-    buf.offset_ = 0;
-    buf.size_ = 0;
-    return *this;
-  }
+  const uint8_t* begin() const { return AsConstSpan().begin(); }
+  const uint8_t* end() const { return AsConstSpan().end(); }
 
   bool operator==(const CopyOnWriteBuffer& buf) const;
 
-  bool operator!=(const CopyOnWriteBuffer& buf) const {
-    return !(*this == buf);
-  }
-
   uint8_t operator[](size_t index) const {
     RTC_DCHECK_LT(index, size());
-    return cdata()[index];
+    return AsConstSpan()[index];
   }
 
   // Replace the contents of the buffer. Accepts the same types as the
@@ -185,18 +161,7 @@ class RTC_EXPORT CopyOnWriteBuffer {
             typename std::enable_if<
                 internal::BufferCompat<uint8_t, T>::value>::type* = nullptr>
   void SetData(const T* data, size_t size) {
-    RTC_DCHECK(IsConsistent());
-    if (!buffer_) {
-      buffer_ = size > 0 ? new RefCountedBuffer(data, size) : nullptr;
-    } else if (!buffer_->HasOneRef()) {
-      buffer_ = new RefCountedBuffer(data, size, capacity());
-    } else {
-      buffer_->SetData(data, size);
-    }
-    offset_ = 0;
-    size_ = size;
-
-    RTC_DCHECK(IsConsistent());
+    Set(MakeArrayView(reinterpret_cast<const uint8_t*>(data), size));
   }
 
   template <typename T,
@@ -207,38 +172,14 @@ class RTC_EXPORT CopyOnWriteBuffer {
     SetData(array, N);
   }
 
-  void SetData(const CopyOnWriteBuffer& buf) {
-    RTC_DCHECK(IsConsistent());
-    RTC_DCHECK(buf.IsConsistent());
-    if (&buf != this) {
-      buffer_ = buf.buffer_;
-      offset_ = buf.offset_;
-      size_ = buf.size_;
-    }
-  }
+  void SetData(const CopyOnWriteBuffer& buf) { *this = buf; }
 
   // Append data to the buffer. Accepts the same types as the constructors.
   template <typename T,
             typename std::enable_if<
                 internal::BufferCompat<uint8_t, T>::value>::type* = nullptr>
   void AppendData(const T* data, size_t size) {
-    RTC_DCHECK(IsConsistent());
-    if (!buffer_) {
-      buffer_ = new RefCountedBuffer(data, size);
-      offset_ = 0;
-      size_ = size;
-      RTC_DCHECK(IsConsistent());
-      return;
-    }
-
-    UnshareAndEnsureCapacity(std::max(capacity(), size_ + size));
-
-    buffer_->SetSize(offset_ +
-                     size_);  // Remove data to the right of the slice.
-    buffer_->AppendData(data, size);
-    size_ += size;
-
-    RTC_DCHECK(IsConsistent());
+    Append(MakeArrayView(reinterpret_cast<const uint8_t*>(data), size));
   }
 
   template <typename T,
@@ -291,7 +232,32 @@ class RTC_EXPORT CopyOnWriteBuffer {
   }
 
  private:
-  using RefCountedBuffer = FinalRefCountedObject<Buffer>;
+  class RTC_EXPORT RawBuffer {
+   public:
+    explicit RawBuffer(size_t size);
+
+    ArrayView<uint8_t> data() { return MakeArrayView(data_.get(), size_); }
+    size_t capacity() const { return size_; }
+
+   private:
+    const size_t size_;
+    const std::unique_ptr<uint8_t[]> data_;
+  };
+  using RefCountedBuffer = FinalRefCountedObject<RawBuffer>;
+
+  static scoped_refptr<RefCountedBuffer> CreateBuffer(size_t capacity);
+
+  ArrayView<const uint8_t> AsConstSpan() const {
+    RTC_DCHECK(IsConsistent());
+    if (buffer_ == nullptr) {
+      return {};
+    }
+    return buffer_->data().subspan(offset_, size_);
+  }
+
+  void Set(ArrayView<const uint8_t> buffer);
+  void Append(ArrayView<const uint8_t> buffer);
+
   // Create a copy of the underlying data if it is referenced from other Buffer
   // objects or there is not enough capacity.
   void UnshareAndEnsureCapacity(size_t new_capacity);
@@ -299,8 +265,7 @@ class RTC_EXPORT CopyOnWriteBuffer {
   // Pre- and postcondition of all methods.
   bool IsConsistent() const {
     if (buffer_) {
-      return buffer_->capacity() > 0 && offset_ <= buffer_->size() &&
-             offset_ + size_ <= buffer_->size();
+      return buffer_->capacity() > 0 && offset_ + size_ <= buffer_->capacity();
     } else {
       return size_ == 0 && offset_ == 0;
     }
diff --git a/unsafe_buffers_paths.txt b/unsafe_buffers_paths.txt
index 64c5244c7b..59f9e0ac1d 100644
--- a/unsafe_buffers_paths.txt
+++ b/unsafe_buffers_paths.txt
@@ -614,8 +614,6 @@
 -rtc_base/checks.cc
 -rtc_base/containers/flat_tree.h
 -rtc_base/containers/flat_tree_unittest.cc
--rtc_base/copy_on_write_buffer.cc
--rtc_base/copy_on_write_buffer.h
 -rtc_base/copy_on_write_buffer_unittest.cc
 -rtc_base/cpu_info.cc
 -rtc_base/crc32.cc

From 4cff6ae4e72e209370057a0283d770ee7f7d9a03 Mon Sep 17 00:00:00 2001
From: Tomas Gunnarsson <tommi@webrtc.org>
Date: Thu, 12 Feb 2026 06:35:14 -0800
Subject: [PATCH 033/472] Revert "Fix std::tmpnam deprecation in
 event_log_visualizer"
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This reverts commit 20101b211bdf290d52bbf04822593364deaee443.

Reason for revert: Broken some downstream projects

Original change's description:
> Fix std::tmpnam deprecation in event_log_visualizer
>
> This CL replaces the deprecated std::tmpnam() function with
> webrtc::test::TempFilename() to avoid compiler warnings on macOS.
>
> This change requires:
> 1. Marking rtc_executable("event_log_visualizer") as testonly = true.
> 2. Adding dependencies on //test:test_support and //test:fileutils.
> 3. Removing the -Wno-deprecated-declarations suppression flag.
>
> Bug: webrtc:14248
> Change-Id: I8989a1839fabf70549843fd37d1c35df6c470efd
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/448160
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#46883}

Bug: webrtc:14248
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I150f32173a7c72e4190f0087aa5c417c18f31365
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/448286
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#46886}
---
 rtc_tools/BUILD.gn                         |  8 ++++----
 rtc_tools/rtc_event_log_visualizer/main.cc | 10 ++++------
 2 files changed, 8 insertions(+), 10 deletions(-)

diff --git a/rtc_tools/BUILD.gn b/rtc_tools/BUILD.gn
index 1515936783..c14c29d018 100644
--- a/rtc_tools/BUILD.gn
+++ b/rtc_tools/BUILD.gn
@@ -390,9 +390,9 @@ if (!build_with_chromium) {
       deps = [
         ":chart_proto",
         ":event_log_visualizer_bindings",
-        "../test:fileutils",
         "//rtc_base:protobuf_utils",
         "//rtc_base/system:file_wrapper",
+        "//test:fileutils",
         "//test:test_support",
         "//third_party/abseil-cpp/absl/strings:string_view",
       ]
@@ -444,8 +444,9 @@ if (rtc_include_tests) {
   if (!build_with_chromium) {
     if (rtc_enable_protobuf) {
       rtc_executable("event_log_visualizer") {
-        testonly = true
-
+        # TODO(bugs.webrtc.org/14248): Remove once usage of std::tmpnam
+        # is removed (in favor of in memory InputAudioFile.
+        cflags_cc = [ "-Wno-deprecated-declarations" ]
         sources = [
           "rtc_event_log_visualizer/conversational_speech_en.h",
           "rtc_event_log_visualizer/main.cc",
@@ -465,7 +466,6 @@ if (rtc_include_tests) {
           "../rtc_base:checks",
           "../rtc_base:logging",
           "../rtc_base:protobuf_utils",
-          "../test:fileutils",
           "//third_party/abseil-cpp/absl/algorithm:container",
           "//third_party/abseil-cpp/absl/flags:config",
           "//third_party/abseil-cpp/absl/flags:flag",
diff --git a/rtc_tools/rtc_event_log_visualizer/main.cc b/rtc_tools/rtc_event_log_visualizer/main.cc
index 6105e01f70..03512b185b 100644
--- a/rtc_tools/rtc_event_log_visualizer/main.cc
+++ b/rtc_tools/rtc_event_log_visualizer/main.cc
@@ -41,7 +41,6 @@
 #include "rtc_tools/rtc_event_log_visualizer/conversational_speech_en.h"
 #include "rtc_tools/rtc_event_log_visualizer/plot_base.h"
 #include "rtc_tools/rtc_event_log_visualizer/proto/chart.pb.h"
-#include "test/testsupport/file_utils.h"
 
 ABSL_FLAG(std::string,
           plot,
@@ -216,11 +215,10 @@ int main(int argc, char* argv[]) {
   if (!absl::GetFlag(FLAGS_wav_filename).empty()) {
     wav_path = absl::GetFlag(FLAGS_wav_filename);
   } else {
-    // TODO(bugs.webrtc.org/14248): Remove the need to generate a file.
-    wav_path = webrtc::test::TempFilename(webrtc::test::OutputPath(),
-                                          "temp_speech.wav");
-
-    std::ofstream out_wav_file(wav_path, std::ios::binary);
+    // TODO(bugs.webrtc.org/14248): Remove the need to generate a file
+    // and read the file directly from memory.
+    wav_path = std::tmpnam(nullptr);
+    std::ofstream out_wav_file(wav_path);
     out_wav_file.write(
         reinterpret_cast<char*>(&webrtc::conversational_speech_en_wav[0]),
         webrtc::conversational_speech_en_wav_len);

From 6fe2890e88be4119da5e3036035b009cc60a1ab4 Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Thu, 12 Feb 2026 12:02:45 +0000
Subject: [PATCH 034/472] Add payload_type.h to API

This change moves the PayloadType strong alias definition from
call/payload_type.h to api/payload_type.h to allow it to be used in
api/ interfaces without circular dependencies.

This is part of a larger refactoring to enforce type safety for payload types.

Bug: webrtc:360058654
Change-Id: If8ee7da21d1fc14904172a9d7a6fea2473ea6379
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/448660
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46887}
---
 api/BUILD.gn        | 13 +++++++++++++
 api/DEPS            |  5 +++++
 api/payload_type.h  | 44 ++++++++++++++++++++++++++++++++++++++++++++
 call/BUILD.gn       |  2 +-
 call/payload_type.h | 24 +-----------------------
 5 files changed, 64 insertions(+), 24 deletions(-)
 create mode 100644 api/payload_type.h

diff --git a/api/BUILD.gn b/api/BUILD.gn
index 7891d9adf1..537315fa5e 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -682,6 +682,7 @@ rtc_library("frame_transformer_interface") {
   deps = [
     ":array_view",
     ":make_ref_counted",
+    ":payload_type",
     ":ref_count",
     ":scoped_refptr",
     "../rtc_base:checks",
@@ -810,6 +811,15 @@ rtc_source_set("rtp_transceiver_direction") {
   sources = [ "rtp_transceiver_direction.h" ]
 }
 
+rtc_source_set("payload_type") {
+  visibility = [ "*" ]
+  sources = [ "payload_type.h" ]
+  deps = [
+    "../rtc_base:strong_alias",
+    "//third_party/abseil-cpp/absl/strings:str_format",
+  ]
+}
+
 rtc_library("priority") {
   visibility = [ "*" ]
   sources = [
@@ -1684,6 +1694,7 @@ if (rtc_include_tests) {
     deps = [
       ":array_view",
       ":frame_transformer_interface",
+      ":payload_type",
       "../test:test_support",
       "units:time_delta",
       "units:timestamp",
@@ -1758,6 +1769,7 @@ if (rtc_include_tests) {
     deps = [
       ":array_view",
       ":frame_transformer_interface",
+      ":payload_type",
       "../api/units:timestamp",
       "../test:test_support",
       "units:time_delta",
@@ -1772,6 +1784,7 @@ if (rtc_include_tests) {
     deps = [
       ":array_view",
       ":frame_transformer_interface",
+      ":payload_type",
       "../test:test_support",
       "units:time_delta",
       "units:timestamp",
diff --git a/api/DEPS b/api/DEPS
index b847c1151a..dc2d1f1cf3 100644
--- a/api/DEPS
+++ b/api/DEPS
@@ -139,6 +139,11 @@ specific_include_rules = {
     "+rtc_base/ssl_certificate.h",
   ],
 
+  "payload_type\.h": [
+    "+absl/strings/str_format.h",
+    "+rtc_base/strong_alias.h",
+  ],
+
   "turn_customizer\.h": [
     "+p2p/base/port_interface.h",
   ],
diff --git a/api/payload_type.h b/api/payload_type.h
new file mode 100644
index 0000000000..9d4a4434cf
--- /dev/null
+++ b/api/payload_type.h
@@ -0,0 +1,44 @@
+/*
+ *  Copyright 2024 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_PAYLOAD_TYPE_H_
+#define API_PAYLOAD_TYPE_H_
+
+#include <cstdint>
+
+#include "absl/strings/str_format.h"
+#include "rtc_base/strong_alias.h"
+
+namespace webrtc {
+
+class PayloadType : public StrongAlias<class PayloadTypeTag, uint8_t> {
+ public:
+  // Non-explicit conversions from and to ints are to be deprecated and
+  // removed once calling code is upgraded.
+  PayloadType(uint8_t pt) { value_ = pt; }                // NOLINT: explicit
+  constexpr operator uint8_t() const& { return value_; }  // NOLINT: Explicit
+  static bool IsValid(PayloadType id, bool rtcp_mux) {
+    // A payload type is a 7-bit value in the RTP header, so max = 127.
+    // If RTCP multiplexing is used, the numbers from 64 to 95 are reserved
+    // for RTCP packets.
+    if (rtcp_mux && (id > 63 && id < 96)) {
+      return false;
+    }
+    return id >= 0 && id <= 127;
+  }
+  template <typename Sink>
+  friend void AbslStringify(Sink& sink, const PayloadType pt) {
+    absl::Format(&sink, "%d", pt.value_);
+  }
+};
+
+}  // namespace webrtc
+
+#endif  // API_PAYLOAD_TYPE_H_
diff --git a/call/BUILD.gn b/call/BUILD.gn
index 2ad1a5422d..3a70a8f6ca 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -379,9 +379,9 @@ rtc_library("payload_type_picker") {
 rtc_source_set("payload_type") {
   sources = [ "payload_type.h" ]
   deps = [
+    "../api:payload_type",
     "../api:rtc_error",
     "../media:codec",
-    "../rtc_base:strong_alias",
     "//third_party/abseil-cpp/absl/strings:string_view",
   ]
 }
diff --git a/call/payload_type.h b/call/payload_type.h
index f67bb6b004..a3453f1e8c 100644
--- a/call/payload_type.h
+++ b/call/payload_type.h
@@ -11,38 +11,16 @@
 #ifndef CALL_PAYLOAD_TYPE_H_
 #define CALL_PAYLOAD_TYPE_H_
 
-#include <cstdint>
 
 #include "absl/strings/string_view.h"
+#include "api/payload_type.h"
 #include "api/rtc_error.h"
 #include "media/base/codec.h"
-#include "rtc_base/strong_alias.h"
 
 namespace webrtc {
 
 class PayloadTypePicker;
 
-class PayloadType : public StrongAlias<class PayloadTypeTag, uint8_t> {
- public:
-  // Non-explicit conversions from and to ints are to be deprecated and
-  // removed once calling code is upgraded.
-  PayloadType(uint8_t pt) { value_ = pt; }                // NOLINT: explicit
-  constexpr operator uint8_t() const& { return value_; }  // NOLINT: Explicit
-  static bool IsValid(PayloadType id, bool rtcp_mux) {
-    // A payload type is a 7-bit value in the RTP header, so max = 127.
-    // If RTCP multiplexing is used, the numbers from 64 to 95 are reserved
-    // for RTCP packets.
-    if (rtcp_mux && (id > 63 && id < 96)) {
-      return false;
-    }
-    return id >= 0 && id <= 127;
-  }
-  template <typename Sink>
-  friend void AbslStringify(Sink& sink, const PayloadType pt) {
-    absl::Format(&sink, "%d", pt.value_);
-  }
-};
-
 class PayloadTypeSuggester {
  public:
   virtual ~PayloadTypeSuggester() = default;

From e1357d9d8ae4775b80983a7ad486f8daf6946036 Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Thu, 12 Feb 2026 13:55:57 +0000
Subject: [PATCH 035/472] Declare rtc_base/buffer safe from
 unsafe_buffers_paths

This was skipped in https://webrtc-review.googlesource.com/c/src/+/444780
due to a bot issue.

Bug: webrtc:478086887
Change-Id: Icea3f111ba23878d81bf09a5676564a3676a9d31
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/448284
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46888}
---
 unsafe_buffers_paths.txt | 2 --
 1 file changed, 2 deletions(-)

diff --git a/unsafe_buffers_paths.txt b/unsafe_buffers_paths.txt
index 59f9e0ac1d..3e10db04fd 100644
--- a/unsafe_buffers_paths.txt
+++ b/unsafe_buffers_paths.txt
@@ -605,9 +605,7 @@
 -rtc_base/bounded_inline_vector.h
 -rtc_base/bounded_inline_vector_impl.h
 -rtc_base/bounded_inline_vector_unittest.cc
--rtc_base/buffer.h
 -rtc_base/buffer_queue_unittest.cc
--rtc_base/buffer_unittest.cc
 -rtc_base/byte_buffer.cc
 -rtc_base/byte_buffer.h
 -rtc_base/byte_order.h

From 191a6abd692a9596a2055526bb1aa972bd991cff Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Thu, 12 Feb 2026 13:58:11 +0100
Subject: [PATCH 036/472] Batch worker thread tasks in
 [Get]StopTransceiverProcedure

Refactor StopTransceiverProcedure and StopSendingAndReceiving to return
worker thread operations as invocable tasks instead of executing them
immediately via BlockingCall. This allows SdpOfferAnswerHandler to batch
operations with other worker thread tasks, reducing the number of
synchronous thread hops during SDP negotiation.

Key changes:
- Update RtpTransceiver to return worker thread tasks.
- Introduce ScopedOperationsBatcher in SdpOfferAnswerHandler to collect
  and execute tasks in a single worker thread block.
- Ensure StopInternal and StopStandard maintain the existing synchronous
  behavior when batching is not used.
- Remove unused parameters and update tests to accommodate the
  task-based workflow.

Bug: webrtc:42222804
Change-Id: Ib253d382ee46d3da17e3ce6324a77ed3d22b73dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/448460
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46889}
---
 pc/BUILD.gn                    |  1 +
 pc/rtp_transceiver.cc          | 61 +++++++++++++++++++++-------------
 pc/rtp_transceiver.h           | 11 ++++--
 pc/rtp_transceiver_unittest.cc |  5 ++-
 pc/sdp_offer_answer.cc         | 59 ++++++++++++++++++++++++--------
 5 files changed, 96 insertions(+), 41 deletions(-)

diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 3b692fcdc3..3c2aee65c5 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -1716,6 +1716,7 @@ rtc_library("rtp_transceiver") {
     "../rtc_base:threading",
     "../rtc_base/system:plan_b_only",
     "//third_party/abseil-cpp/absl/algorithm:container",
+    "//third_party/abseil-cpp/absl/base:nullability",
     "//third_party/abseil-cpp/absl/functional:any_invocable",
     "//third_party/abseil-cpp/absl/strings:string_view",
   ]
diff --git a/pc/rtp_transceiver.cc b/pc/rtp_transceiver.cc
index ae133e8e1b..608c1888bb 100644
--- a/pc/rtp_transceiver.cc
+++ b/pc/rtp_transceiver.cc
@@ -21,6 +21,7 @@
 #include <vector>
 
 #include "absl/algorithm/container.h"
+#include "absl/base/nullability.h"
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
 #include "api/array_view.h"
@@ -608,9 +609,9 @@ absl::AnyInvocable<void() &&> RtpTransceiver::GetDeleteChannelWorkerTask(
     return nullptr;
   }
 
-  std::vector<absl::AnyInvocable<void() &&>> stop;
+  std::vector<absl::AnyInvocable<void() &&>> stop_sender_actions;
   if (stop_senders) {
-    stop = DetachAndGetStopTasksForSenders(senders_);
+    stop_sender_actions = DetachAndGetStopTasksForSenders(senders_);
   }
 
   transport_name_ = std::nullopt;
@@ -618,9 +619,10 @@ absl::AnyInvocable<void() &&> RtpTransceiver::GetDeleteChannelWorkerTask(
   // Ensure that channel_ is not reachable via the transceiver, but is deleted
   // only after clearing the references in senders_ and receivers_.
   return [this, channel = std::move(channel_), senders = senders_,
-          receivers = receivers_, stop = std::move(stop)]() mutable {
+          receivers = receivers_,
+          stop_sender_actions = std::move(stop_sender_actions)]() mutable {
     RTC_DCHECK_RUN_ON(context()->worker_thread());
-    for (auto& task : stop) {
+    for (auto& task : stop_sender_actions) {
       std::move(task)();
     }
     ClearMediaChannelReferences();
@@ -917,14 +919,15 @@ void RtpTransceiver::set_receptive(bool receptive) {
   }
 }
 
-void RtpTransceiver::StopSendingAndReceiving() {
+absl_nonnull absl::AnyInvocable<void() &&>
+RtpTransceiver::GetStopSendingAndReceiving() {
   RTC_DCHECK_RUN_ON(thread_);
   RTC_DCHECK(!stopped_);
   RTC_DCHECK(!stopping_);
   // 1. Let sender be transceiver.[[Sender]].
   // 2. Let receiver be transceiver.[[Receiver]].
 
-  RTC_LOG_THREAD_BLOCK_COUNT();
+  RTC_DCHECK_DISALLOW_THREAD_BLOCKING_CALLS();
 
   // Signal to receiver sources that we're stopping.
   for (const auto& receiver : receivers_) {
@@ -936,25 +939,23 @@ void RtpTransceiver::StopSendingAndReceiving() {
   // worker thread to avoid each sender doing that within `Stop()`.
   // Senders will have already cleared send when the media channel was set to
   // nullptr.
-  std::vector<absl::AnyInvocable<void() &&>> stop =
+  std::vector<absl::AnyInvocable<void() &&>> stop_sender_actions =
       DetachAndGetStopTasksForSenders(senders_);
 
-  // 3. Send an RTCP BYE for each RTP stream that was being sent by sender, as
-  // specified in [RFC3550].
-  context()->worker_thread()->BlockingCall([&]() {
-    RTC_DCHECK_RUN_ON(context()->worker_thread());
-    for (auto& task : stop) {
-      std::move(task)();
-    }
-    ClearMediaChannelReferences();
-  });
-
-  RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(1);
-
   stopping_ = true;
   direction_ = RtpTransceiverDirection::kInactive;
 
-  RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(1);
+  // 3. Send an RTCP BYE for each RTP stream that was being sent by sender, as
+  // specified in [RFC3550].
+
+  return
+      [this, stop_sender_actions = std::move(stop_sender_actions)]() mutable {
+        RTC_DCHECK_RUN_ON(context()->worker_thread());
+        for (auto& task : stop_sender_actions) {
+          std::move(task)();
+        }
+        ClearMediaChannelReferences();
+      };
 }
 
 RTCError RtpTransceiver::StopStandard() {
@@ -980,7 +981,9 @@ RTCError RtpTransceiver::StopStandard() {
 
   // 5. Stop sending and receiving given transceiver, and update the
   // negotiation-needed flag for connection.
-  StopSendingAndReceiving();
+  auto stop_task = GetStopSendingAndReceiving();
+  context_->worker_thread()->BlockingCall(
+      [&]() mutable { std::move(stop_task)(); });
   on_negotiation_needed_();
 
   return RTCError::OK();
@@ -988,16 +991,24 @@ RTCError RtpTransceiver::StopStandard() {
 
 void RtpTransceiver::StopInternal() {
   RTC_DCHECK_RUN_ON(thread_);
-  StopTransceiverProcedure();
+  auto stop_task = GetStopTransceiverProcedure();
+  if (stop_task) {
+    context_->worker_thread()->BlockingCall(
+        [stop_task = std::move(stop_task)]() mutable {
+          std::move(stop_task)();
+        });
+  }
 }
 
-void RtpTransceiver::StopTransceiverProcedure() {
+absl_nullable absl::AnyInvocable<void() &&>
+RtpTransceiver::GetStopTransceiverProcedure() {
   RTC_DCHECK_RUN_ON(thread_);
   // As specified in the "Stop the RTCRtpTransceiver" procedure
   // 1. If transceiver.[[Stopping]] is false, stop sending and receiving given
   // transceiver.
+  absl::AnyInvocable<void() &&> stop_task;
   if (!stopping_)
-    StopSendingAndReceiving();
+    stop_task = GetStopSendingAndReceiving();
 
   // 2. Set transceiver.[[Stopped]] to true.
   stopped_ = true;
@@ -1007,6 +1018,8 @@ void RtpTransceiver::StopTransceiverProcedure() {
 
   // 4. Set transceiver.[[CurrentDirection]] to null.
   current_direction_ = std::nullopt;
+
+  return stop_task;
 }
 
 RTCError RtpTransceiver::SetCodecPreferences(
diff --git a/pc/rtp_transceiver.h b/pc/rtp_transceiver.h
index d47d563271..107e66e2f4 100644
--- a/pc/rtp_transceiver.h
+++ b/pc/rtp_transceiver.h
@@ -19,6 +19,7 @@
 #include <utility>
 #include <vector>
 
+#include "absl/base/nullability.h"
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
 #include "api/array_view.h"
@@ -318,7 +319,10 @@ class RtpTransceiver : public RtpTransceiverInterface {
 
   // Executes the "stop the RTCRtpTransceiver" procedure from
   // the webrtc-pc specification, described under the stop() method.
-  void StopTransceiverProcedure();
+  // The task must be executed on the worker thread.
+  // This is used by SdpOfferAnswerHandler to batch worker thread operations.
+  [[nodiscard]] absl_nullable absl::AnyInvocable<void() &&>
+  GetStopTransceiverProcedure();
 
   // RtpTransceiverInterface implementation.
   MediaType media_type() const override;
@@ -401,7 +405,10 @@ class RtpTransceiver : public RtpTransceiverInterface {
   void OnPacketReceived(scoped_refptr<PendingTaskSafetyFlag> safety)
       RTC_RUN_ON(context()->network_thread());
   void OnFirstPacketSent();
-  void StopSendingAndReceiving();
+  // Stops the receivers synchronously and returns a task that stops the
+  // senders. The returned task must be executed on the worker thread.
+  [[nodiscard]] absl_nonnull absl::AnyInvocable<void() &&>
+  GetStopSendingAndReceiving();
   // Tell the senders and receivers about possibly-new media channels
   // in a newly created `channel_`.
   void PushNewMediaChannel();
diff --git a/pc/rtp_transceiver_unittest.cc b/pc/rtp_transceiver_unittest.cc
index d2aa2cb271..1e42603ec2 100644
--- a/pc/rtp_transceiver_unittest.cc
+++ b/pc/rtp_transceiver_unittest.cc
@@ -271,7 +271,10 @@ TEST_F(RtpTransceiverUnifiedPlanTest, StopSetsDirection) {
   transceiver->StopStandard();
   EXPECT_EQ(RtpTransceiverDirection::kStopped, transceiver->direction());
   EXPECT_FALSE(transceiver->current_direction());
-  transceiver->StopTransceiverProcedure();
+  auto stop_task = transceiver->GetStopTransceiverProcedure();
+  if (stop_task) {
+    std::move(stop_task)();
+  }
   EXPECT_TRUE(transceiver->current_direction());
   EXPECT_EQ(RtpTransceiverDirection::kStopped, transceiver->direction());
   EXPECT_EQ(RtpTransceiverDirection::kStopped,
diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc
index 5242ec13d5..af366df54d 100644
--- a/pc/sdp_offer_answer.cc
+++ b/pc/sdp_offer_answer.cc
@@ -126,6 +126,39 @@ namespace webrtc {
 
 namespace {
 
+class ScopedOperationsBatcher {
+ public:
+  explicit ScopedOperationsBatcher(Thread* worker_thread)
+      : worker_thread_(worker_thread) {
+    RTC_DCHECK(worker_thread_);
+  }
+
+  ~ScopedOperationsBatcher() { Run(); }
+
+  void Run() {
+    if (!tasks_.empty()) {
+      worker_thread_->BlockingCall([tasks = std::move(tasks_)]() mutable {
+        for (auto& task : tasks) {
+          std::move(task)();
+        }
+      });
+      RTC_DCHECK(tasks_.empty());
+    }
+  }
+
+  // Queues non-nullptr tasks to be executed on the worker when the
+  // ScopedOperationsBatcher goes out of scope.
+  void push_back(absl::AnyInvocable<void() &&> task) {
+    if (task) {
+      tasks_.push_back(std::move(task));
+    }
+  }
+
+ private:
+  Thread* const worker_thread_;
+  std::vector<absl::AnyInvocable<void() &&>> tasks_;
+};
+
 struct DtlsTransportAndName {
   scoped_refptr<DtlsTransport> transport;
   std::optional<std::string> transport_name;
@@ -2349,7 +2382,7 @@ void SdpOfferAnswerHandler::ApplyRemoteDescriptionUpdateTransceiverState(
   std::vector<scoped_refptr<RtpTransceiverInterface>> remove_list;
   std::vector<scoped_refptr<MediaStreamInterface>> added_streams;
   std::vector<scoped_refptr<MediaStreamInterface>> removed_streams;
-  std::vector<absl::AnyInvocable<void() &&>> worker_tasks;
+  ScopedOperationsBatcher worker_tasks(context_->worker_thread());
   flat_map<std::string, DtlsTransportAndName> dtls_transports_by_mid =
       GetDtlsTransports(*transceivers(), context_->network_thread(),
                         transport_controller_s());
@@ -2440,7 +2473,7 @@ void SdpOfferAnswerHandler::ApplyRemoteDescriptionUpdateTransceiverState(
     if (content->rejected && !transceiver->stopped()) {
       RTC_LOG(LS_INFO) << "Stopping transceiver for MID=" << content->mid()
                        << " since the media section was rejected.";
-      transceiver->StopTransceiverProcedure();
+      worker_tasks.push_back(transceiver->GetStopTransceiverProcedure());
     }
     if (!content->rejected && RtpTransceiverDirectionHasRecv(local_direction)) {
       if (!media_desc->streams().empty() &&
@@ -2455,15 +2488,8 @@ void SdpOfferAnswerHandler::ApplyRemoteDescriptionUpdateTransceiverState(
     }
   }
 
-  if (!worker_tasks.empty()) {
-    context_->worker_thread()->BlockingCall([&worker_tasks] {
-      for (auto& task : worker_tasks) {
-        if (task) {
-          std::move(task)();
-        }
-      }
-    });
-  }
+  worker_tasks.Run();
+
   // Once all processing has finished, fire off callbacks.
   pc_->RunWithObserver([&](auto observer) {
     for (const auto& transceiver : now_receiving_transceivers) {
@@ -3457,6 +3483,7 @@ RTCError SdpOfferAnswerHandler::Rollback(SdpType desc_type) {
   std::vector<scoped_refptr<MediaStreamInterface>> all_added_streams;
   std::vector<scoped_refptr<MediaStreamInterface>> all_removed_streams;
   std::vector<scoped_refptr<RtpReceiverInterface>> removed_receivers;
+  ScopedOperationsBatcher worker_tasks(context_->worker_thread());
 
   for (auto&& transceivers_stable_state_pair : transceivers()->StableStates()) {
     auto transceiver = transceivers_stable_state_pair.first;
@@ -3518,7 +3545,8 @@ RTCError SdpOfferAnswerHandler::Rollback(SdpType desc_type) {
       if (transceiver->internal()->reused_for_addtrack()) {
         transceiver->internal()->set_created_by_addtrack(true);
       } else {
-        transceiver->internal()->StopTransceiverProcedure();
+        worker_tasks.push_back(
+            transceiver->internal()->GetStopTransceiverProcedure());
         transceivers()->Remove(transceiver);
       }
     }
@@ -4067,6 +4095,7 @@ RTCError SdpOfferAnswerHandler::UpdateTransceiversAndDataChannels(
     }
   }
 
+  ScopedOperationsBatcher worker_tasks(context_->worker_thread());
   const ContentInfos& new_contents = new_session.description()->contents();
   for (size_t i = 0; i < new_contents.size(); ++i) {
     const ContentInfo& new_content = new_contents[i];
@@ -4128,7 +4157,8 @@ RTCError SdpOfferAnswerHandler::UpdateTransceiversAndDataChannels(
           // is not already stopped, SDP munging has happened and we need to
           // ensure the transceiver is stopped.
           if (!transceiver->internal()->stopped()) {
-            transceiver->internal()->StopTransceiverProcedure();
+            worker_tasks.push_back(
+                transceiver->internal()->GetStopTransceiverProcedure());
           }
           RTC_DCHECK(transceiver->internal()->stopped());
         }
@@ -5716,8 +5746,9 @@ void SdpOfferAnswerHandler::GetMediaChannelTeardownTasks(
       if (auto task = transceiver->internal()->GetClearChannelNetworkTask())
         network_tasks.push_back(std::move(task));
       if (auto task = transceiver->internal()->GetDeleteChannelWorkerTask(
-              /*stop_senders=*/true))
+              /*stop_senders=*/true)) {
         worker_tasks.push_back(std::move(task));
+      }
     }
   }
   for (const auto& transceiver : list) {

From a7e7130d51ec87e49bff52edea3c7347cfe8f353 Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Thu, 12 Feb 2026 13:59:38 +0000
Subject: [PATCH 037/472] Deprecate the two-argument Buffer constructor

The two-argument Buffer constructor could create uninintialized data.
This refactor replaces it with CreateUninitializedWithSizeAndCapacity.

It also refactors the CopyOnWriteBuffer class to use explicit functions when creating uninitialized Buffers.

Also do some checking to ensure that SetData() always has room enough.

Bug: webrtc:42223681
Change-Id: I160c88ca41944c9888d7862e8ce46cd673417a69
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/445040
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46890}
---
 common_video/h264/sps_vui_rewriter.cc         |  4 +-
 .../codecs/red/audio_encoder_copy_red.cc      |  2 +-
 .../audio_coding/neteq/tools/audio_checksum.h |  2 +-
 rtc_base/buffer.h                             | 46 ++++++++++++-------
 rtc_base/buffer_queue.cc                      |  2 +-
 rtc_base/buffer_unittest.cc                   |  4 +-
 rtc_base/openssl_stream_adapter.cc            |  2 +-
 rtc_base/socket_unittest.cc                   |  6 +--
 rtc_base/ssl_fingerprint.cc                   |  2 +-
 rtc_base/ssl_identity_unittest.cc             |  4 +-
 rtc_base/ssl_stream_adapter_unittest.cc       |  8 ++--
 test/fuzzers/ssl_certificate_fuzzer.cc        |  2 +-
 12 files changed, 47 insertions(+), 37 deletions(-)

diff --git a/common_video/h264/sps_vui_rewriter.cc b/common_video/h264/sps_vui_rewriter.cc
index 4d23071e9c..4481afdd37 100644
--- a/common_video/h264/sps_vui_rewriter.cc
+++ b/common_video/h264/sps_vui_rewriter.cc
@@ -228,8 +228,8 @@ Buffer SpsVuiRewriter::ParseOutgoingBitstreamAndRewrite(
   std::vector<H264::NaluIndex> nalus = H264::FindNaluIndices(buffer);
 
   // Allocate some extra space for potentially adding a missing VUI.
-  Buffer output_buffer(/*size=*/0, /*capacity=*/buffer.size() +
-                                       nalus.size() * kMaxVuiSpsIncrease);
+  Buffer output_buffer = Buffer::CreateWithCapacity(
+      buffer.size() + nalus.size() * kMaxVuiSpsIncrease);
 
   for (const H264::NaluIndex& nalu_index : nalus) {
     // Copy NAL unit start code.
diff --git a/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc b/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
index b40a1918cb..a69c9203f2 100644
--- a/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
+++ b/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
@@ -64,7 +64,7 @@ size_t GetMaxRedundancyFromFieldTrial(const FieldTrialsView& field_trials) {
 AudioEncoderCopyRed::AudioEncoderCopyRed(Config&& config,
                                          const FieldTrialsView& field_trials)
     : speech_encoder_(std::move(config.speech_encoder)),
-      primary_encoded_(0, kAudioMaxRtpPacketLen),
+      primary_encoded_(Buffer::CreateWithCapacity(kAudioMaxRtpPacketLen)),
       max_packet_length_(kAudioMaxRtpPacketLen),
       red_payload_type_(config.payload_type) {
   RTC_CHECK(speech_encoder_) << "Speech encoder not provided.";
diff --git a/modules/audio_coding/neteq/tools/audio_checksum.h b/modules/audio_coding/neteq/tools/audio_checksum.h
index 845adf78d0..b7ce1493df 100644
--- a/modules/audio_coding/neteq/tools/audio_checksum.h
+++ b/modules/audio_coding/neteq/tools/audio_checksum.h
@@ -30,7 +30,7 @@ class AudioChecksum : public AudioSink {
  public:
   AudioChecksum()
       : checksum_(MessageDigestFactory::Create(DIGEST_MD5)),
-        checksum_result_(0, checksum_->Size()),
+        checksum_result_(Buffer::CreateWithCapacity(checksum_->Size())),
         finished_(false) {}
 
   AudioChecksum(const AudioChecksum&) = delete;
diff --git a/rtc_base/buffer.h b/rtc_base/buffer.h
index 322cf2536a..161cb76fe4 100644
--- a/rtc_base/buffer.h
+++ b/rtc_base/buffer.h
@@ -21,7 +21,6 @@
 #include <utility>
 
 #include "absl/algorithm/container.h"
-#include "absl/base/attributes.h"
 #include "absl/strings/string_view.h"
 #include "api/array_view.h"
 #include "rtc_base/checks.h"
@@ -71,6 +70,14 @@ class BufferT {
   using iterator = std::span<T>::iterator;
   using const_iterator = std::span<const T>::iterator;
 
+  // Static methods to construct a buffer with size and/or capacity.
+  static BufferT CreateWithCapacity(size_t capacity) {
+    return BufferT(InternalTag{}, 0, capacity);
+  }
+  static BufferT CreateUninitializedWithSize(size_t size) {
+    return BufferT(InternalTag{}, size, size);
+  }
+
   // An empty BufferT.
   BufferT() : size_(0), capacity_(0), data_(nullptr) {
     RTC_DCHECK(IsConsistent());
@@ -90,15 +97,15 @@ class BufferT {
   }
 
   // Construct a buffer with the specified number of uninitialized elements.
-  ABSL_DEPRECATED("Use CreateUninitializedWithSize()")
-  explicit BufferT(size_t size) : BufferT(size, size) {}
+  [[deprecated("Use CreateUninitializedWithSize()")]]
+  explicit BufferT(size_t size)
+      : BufferT(size, size) {}
 
+  // Construct a buffer with the specified number of uninitialized elements,
+  // and a possibly non-zero size()
+  [[deprecated("Use CreateWithCapacity() or CreateUninitializedWithSize()")]]
   BufferT(size_t size, size_t capacity)
-      : size_(size),
-        capacity_(std::max(size, capacity)),
-        data_(capacity_ > 0 ? new T[capacity_] : nullptr) {
-    RTC_DCHECK(IsConsistent());
-  }
+      : BufferT(InternalTag{}, size, capacity) {}
 
   // Construct a buffer and copy the specified number of elements into it.
   template <typename U,
@@ -109,10 +116,12 @@ class BufferT {
   template <typename U,
             typename std::enable_if<
                 internal::BufferCompat<T, U>::value>::type* = nullptr>
-  BufferT(U* data, size_t size, size_t capacity) : BufferT(size, capacity) {
+  BufferT(U* data, size_t size, size_t capacity)
+      : BufferT(InternalTag{}, size, capacity) {
     static_assert(sizeof(T) == sizeof(U), "");
     if (size > 0) {
       RTC_DCHECK(data);
+      RTC_DCHECK_LE(size, capacity);
       std::memcpy(data_.get(), data, size * sizeof(U));
     }
   }
@@ -139,14 +148,6 @@ class BufferT {
     return absl::string_view(data<char>(), size());
   }
 
-  // Static methods to construct a buffer with size and/or capacity.
-  static BufferT CreateWithCapacity(size_t capacity) {
-    return BufferT(0, capacity);
-  }
-  static BufferT CreateUninitializedWithSize(size_t size) {
-    return BufferT(size, size);
-  }
-
   // Get a pointer to the data. Just .data() will give you a (const) T*, but if
   // T is a byte-sized integer, you may also use .data<U>() for any other
   // byte-sized integer U.
@@ -392,6 +393,17 @@ class BufferT {
   }
 
  private:
+  // Internal constructor that allows uninitialized memory to be created.
+  // Used by CreateUninitialized* functions and by the deprecated constructors
+  // that are replaced by CreateUninitialized functions.
+  struct InternalTag {};
+  BufferT(InternalTag tag, size_t size, size_t capacity)
+      : size_(size),
+        capacity_(std::max(size, capacity)),
+        data_(capacity_ > 0 ? new T[capacity_] : nullptr) {
+    RTC_DCHECK(IsConsistent());
+  }
+
   void EnsureCapacityWithHeadroom(size_t capacity, bool extra_headroom) {
     RTC_DCHECK(IsConsistent());
     if (capacity <= capacity_)
diff --git a/rtc_base/buffer_queue.cc b/rtc_base/buffer_queue.cc
index e791e91322..5327dbd7a2 100644
--- a/rtc_base/buffer_queue.cc
+++ b/rtc_base/buffer_queue.cc
@@ -73,7 +73,7 @@ bool BufferQueue::WriteBack(const void* buffer,
     packet = free_list_.back();
     free_list_.pop_back();
   } else {
-    packet = new Buffer(bytes, default_size_);
+    packet = new Buffer(Buffer::CreateWithCapacity(default_size_));
   }
 
   packet->SetData(static_cast<const uint8_t*>(buffer), bytes);
diff --git a/rtc_base/buffer_unittest.cc b/rtc_base/buffer_unittest.cc
index 064320f96d..6e4f7e7a27 100644
--- a/rtc_base/buffer_unittest.cc
+++ b/rtc_base/buffer_unittest.cc
@@ -51,9 +51,7 @@ TEST(BufferTest, TestConstructEmpty) {
   TestBuf(Buffer(Buffer()), 0, 0);
   TestBuf(Buffer::CreateUninitializedWithSize(0), 0, 0);
 
-  // We can't use a literal 0 for the first argument, because C++ will allow
-  // that to be considered a null pointer, which makes the call ambiguous.
-  TestBuf(Buffer(0 + 0, 10), 0, 10);
+  TestBuf(Buffer::CreateWithCapacity(10), 0, 10);
 
   TestBuf(Buffer(kTestData.data(), 0), 0, 0);
   TestBuf(Buffer(kTestData.data(), 0, 20), 0, 20);
diff --git a/rtc_base/openssl_stream_adapter.cc b/rtc_base/openssl_stream_adapter.cc
index 1c0fa53d40..36bda6e0a6 100644
--- a/rtc_base/openssl_stream_adapter.cc
+++ b/rtc_base/openssl_stream_adapter.cc
@@ -1239,7 +1239,7 @@ bool OpenSSLStreamAdapter::VerifyPeerCertificate() {
     return false;
   }
 
-  Buffer computed_digest(0, EVP_MAX_MD_SIZE);
+  Buffer computed_digest = Buffer::CreateWithCapacity(EVP_MAX_MD_SIZE);
   if (!peer_cert_chain_->Get(0).ComputeDigest(
           peer_certificate_digest_algorithm_, computed_digest)) {
     RTC_LOG(LS_WARNING) << "Failed to compute peer cert digest.";
diff --git a/rtc_base/socket_unittest.cc b/rtc_base/socket_unittest.cc
index 66c933eead..ff3c0546f1 100644
--- a/rtc_base/socket_unittest.cc
+++ b/rtc_base/socket_unittest.cc
@@ -855,13 +855,13 @@ void SocketTest::TcpInternal(const IPAddress& loopback,
   EXPECT_EQ(sender->GetRemoteAddress(), receiver->GetLocalAddress());
 
   // Create test data.
-  Buffer send_buffer(0, data_size);
-  Buffer recv_buffer(0, data_size);
+  Buffer send_buffer(Buffer::CreateWithCapacity(data_size));
+  Buffer recv_buffer(Buffer::CreateWithCapacity(data_size));
   for (size_t i = 0; i < data_size; ++i) {
     char ch = static_cast<char>(i % 256);
     send_buffer.AppendData(&ch, sizeof(ch));
   }
-  Buffer recved_data(0, data_size);
+  Buffer recved_data(Buffer::CreateWithCapacity(data_size));
 
   // Send and receive a bunch of data.
   size_t sent_size = 0;
diff --git a/rtc_base/ssl_fingerprint.cc b/rtc_base/ssl_fingerprint.cc
index f3f6fa6228..52253630a7 100644
--- a/rtc_base/ssl_fingerprint.cc
+++ b/rtc_base/ssl_fingerprint.cc
@@ -43,7 +43,7 @@ std::unique_ptr<SSLFingerprint> SSLFingerprint::CreateUnique(
 std::unique_ptr<SSLFingerprint> SSLFingerprint::Create(
     absl::string_view algorithm,
     const SSLCertificate& cert) {
-  Buffer digest(0, MessageDigest::kMaxSize);
+  Buffer digest = Buffer::CreateWithCapacity(MessageDigest::kMaxSize);
   bool ret = cert.ComputeDigest(algorithm, digest);
   if (!ret) {
     return nullptr;
diff --git a/rtc_base/ssl_identity_unittest.cc b/rtc_base/ssl_identity_unittest.cc
index 136ef601fe..69d7b2a102 100644
--- a/rtc_base/ssl_identity_unittest.cc
+++ b/rtc_base/ssl_identity_unittest.cc
@@ -259,7 +259,7 @@ class SSLIdentityTest : public ::testing::Test {
     EXPECT_EQ(expected_len, digest.size());
 
     // Repeat digest computation for the identity as a sanity check.
-    Buffer digest1(0, MessageDigest::kMaxSize);
+    Buffer digest1 = Buffer::CreateWithCapacity(MessageDigest::kMaxSize);
     std::memset(digest1.data(), 0xff, expected_len);
     EXPECT_TRUE(identity->certificate().ComputeDigest(algorithm, digest1));
     EXPECT_EQ(expected_len, digest1.size());
@@ -291,7 +291,7 @@ class SSLIdentityTest : public ::testing::Test {
   void TestDigestForFixedCert(absl::string_view algorithm,
                               size_t expected_len,
                               const unsigned char* expected_digest) {
-    Buffer digest(0, MessageDigest::kMaxSize);
+    Buffer digest(Buffer::CreateWithCapacity(MessageDigest::kMaxSize));
 
     ASSERT_TRUE(expected_len <= digest.capacity());
 
diff --git a/rtc_base/ssl_stream_adapter_unittest.cc b/rtc_base/ssl_stream_adapter_unittest.cc
index 70b4c837bf..668a98c9da 100644
--- a/rtc_base/ssl_stream_adapter_unittest.cc
+++ b/rtc_base/ssl_stream_adapter_unittest.cc
@@ -519,8 +519,8 @@ class SSLStreamAdapterTestBase : public ::testing::Test {
   }
 
   void SetPeerIdentitiesByDigest(bool correct, bool expect_success) {
-    Buffer server_digest(0, EVP_MAX_MD_SIZE);
-    Buffer client_digest(0, EVP_MAX_MD_SIZE);
+    Buffer server_digest(Buffer::CreateWithCapacity(EVP_MAX_MD_SIZE));
+    Buffer client_digest(Buffer::CreateWithCapacity(EVP_MAX_MD_SIZE));
     SSLPeerCertificateDigestError err;
     SSLPeerCertificateDigestError expected_err =
         expect_success ? SSLPeerCertificateDigestError::NONE
@@ -687,8 +687,8 @@ class SSLStreamAdapterTestBase : public ::testing::Test {
 
     // Collect both of the certificate digests; needs to be done before calling
     // SetPeerCertificateDigest as that may reset the identity.
-    Buffer server_digest(0, EVP_MAX_MD_SIZE);
-    Buffer client_digest(0, EVP_MAX_MD_SIZE);
+    Buffer server_digest(Buffer::CreateWithCapacity(EVP_MAX_MD_SIZE));
+    Buffer client_digest(Buffer::CreateWithCapacity(EVP_MAX_MD_SIZE));
 
     ASSERT_THAT(server_identity(), NotNull());
     ASSERT_TRUE(server_identity()->certificate().ComputeDigest(
diff --git a/test/fuzzers/ssl_certificate_fuzzer.cc b/test/fuzzers/ssl_certificate_fuzzer.cc
index ea1f4ee8e1..f056994f8b 100644
--- a/test/fuzzers/ssl_certificate_fuzzer.cc
+++ b/test/fuzzers/ssl_certificate_fuzzer.cc
@@ -38,7 +38,7 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
   std::string algorithm;
   cert->GetSignatureDigestAlgorithm(&algorithm);
 
-  Buffer buffer(0, MessageDigest::kMaxSize);
+  Buffer buffer(Buffer::CreateWithCapacity(MessageDigest::kMaxSize));
   cert->ComputeDigest(algorithm, buffer);
 
   Buffer der_buffer;

From 68cb390aa5565bcf8a6c360f45177743c701bbda Mon Sep 17 00:00:00 2001
From: Bjorn Terelius <terelius@webrtc.org>
Date: Fri, 13 Feb 2026 10:05:39 +0100
Subject: [PATCH 038/472] Split analyzer.cc and change CreateFooGraph methods
 to free functions
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Bug: None
Change-Id: Iaf4a39ad3c9bc681b5ec3f078b328eed0c218942
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/446284
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46891}
---
 rtc_tools/BUILD.gn                            |    6 +
 .../rtc_event_log_visualizer/analyze_audio.cc |   90 +-
 .../rtc_event_log_visualizer/analyze_audio.h  |   13 +
 .../rtc_event_log_visualizer/analyze_bwe.cc   | 1221 +++++++++
 .../rtc_event_log_visualizer/analyze_bwe.h    |   81 +
 .../analyze_connectivity.cc                   |  285 ++
 .../analyze_connectivity.h                    |   38 +
 .../analyze_rtp_rtcp.cc                       |  809 ++++++
 .../analyze_rtp_rtcp.h                        |  100 +
 .../rtc_event_log_visualizer/analyzer.cc      | 2344 +----------------
 rtc_tools/rtc_event_log_visualizer/analyzer.h |   17 +-
 .../analyzer_bindings.cc                      |   13 +-
 .../analyzer_common.cc                        |   64 +
 .../analyzer_common.h                         |   27 +
 rtc_tools/rtc_event_log_visualizer/main.cc    |   12 +-
 15 files changed, 2846 insertions(+), 2274 deletions(-)
 create mode 100644 rtc_tools/rtc_event_log_visualizer/analyze_bwe.cc
 create mode 100644 rtc_tools/rtc_event_log_visualizer/analyze_bwe.h
 create mode 100644 rtc_tools/rtc_event_log_visualizer/analyze_connectivity.cc
 create mode 100644 rtc_tools/rtc_event_log_visualizer/analyze_connectivity.h
 create mode 100644 rtc_tools/rtc_event_log_visualizer/analyze_rtp_rtcp.cc
 create mode 100644 rtc_tools/rtc_event_log_visualizer/analyze_rtp_rtcp.h

diff --git a/rtc_tools/BUILD.gn b/rtc_tools/BUILD.gn
index c14c29d018..24c20bbb24 100644
--- a/rtc_tools/BUILD.gn
+++ b/rtc_tools/BUILD.gn
@@ -290,6 +290,12 @@ if (!build_with_chromium) {
         "rtc_event_log_visualizer/alerts.h",
         "rtc_event_log_visualizer/analyze_audio.cc",
         "rtc_event_log_visualizer/analyze_audio.h",
+        "rtc_event_log_visualizer/analyze_bwe.cc",
+        "rtc_event_log_visualizer/analyze_bwe.h",
+        "rtc_event_log_visualizer/analyze_connectivity.cc",
+        "rtc_event_log_visualizer/analyze_connectivity.h",
+        "rtc_event_log_visualizer/analyze_rtp_rtcp.cc",
+        "rtc_event_log_visualizer/analyze_rtp_rtcp.h",
         "rtc_event_log_visualizer/analyzer.cc",
         "rtc_event_log_visualizer/analyzer.h",
         "rtc_event_log_visualizer/analyzer_common.cc",
diff --git a/rtc_tools/rtc_event_log_visualizer/analyze_audio.cc b/rtc_tools/rtc_event_log_visualizer/analyze_audio.cc
index ed3629a787..f2ec478315 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyze_audio.cc
+++ b/rtc_tools/rtc_event_log_visualizer/analyze_audio.cc
@@ -30,7 +30,10 @@
 #include "api/neteq/neteq.h"
 #include "api/scoped_refptr.h"
 #include "api/units/timestamp.h"
+#include "logging/rtc_event_log/events/logged_rtp_rtcp.h"
 #include "logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h"
+#include "logging/rtc_event_log/events/rtc_event_audio_playout.h"
+#include "logging/rtc_event_log/events/rtc_event_neteq_set_minimum_delay.h"
 #include "logging/rtc_event_log/rtc_event_log_parser.h"
 #include "modules/audio_coding/neteq/tools/audio_sink.h"
 #include "modules/audio_coding/neteq/tools/fake_decode_from_file.h"
@@ -277,8 +280,8 @@ std::unique_ptr<test::NetEqStatsGetter> CreateNetEqTestAndRun(
 
   NetEq::Config config;
   test::NetEqTest test(config, decoder_factory, codecs, /*text_log=*/nullptr,
-                       /*factory=*/nullptr, std::move(input), std::move(output),
-                       callbacks, field_trials);
+                       /*neteq_factory=*/nullptr, std::move(input),
+                       std::move(output), callbacks, field_trials);
   test.Run();
   return neteq_stats_getter;
 }
@@ -426,4 +429,87 @@ void CreateNetEqLifetimeStatsGraph(
       stats_extractor, plot_name, plot);
 }
 
+// For each SSRC, plot the time between the consecutive playouts.
+void CreatePlayoutGraph(const ParsedRtcEventLog& parsed_log,
+                        const AnalyzerConfig& config,
+                        Plot* plot) {
+  for (const auto& playout_stream : parsed_log.audio_playout_events()) {
+    uint32_t ssrc = playout_stream.first;
+    if (!MatchingSsrc(ssrc, config.desired_ssrc_))
+      continue;
+    std::optional<int64_t> last_playout_ms;
+    TimeSeries time_series(SsrcToString(ssrc), LineStyle::kBar);
+    for (const auto& playout_event : playout_stream.second) {
+      float x = config.GetCallTimeSec(playout_event.log_time());
+      int64_t playout_time_ms = playout_event.log_time_ms();
+      // If there were no previous playouts, place the point on the x-axis.
+      float y = playout_time_ms - last_playout_ms.value_or(playout_time_ms);
+      time_series.points.push_back(TimeSeriesPoint(x, y));
+      last_playout_ms.emplace(playout_time_ms);
+    }
+    plot->AppendTimeSeries(std::move(time_series));
+  }
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Time since last playout (ms)", kBottomMargin,
+                          kTopMargin);
+  plot->SetTitle("Audio playout");
+}
+
+void CreateNetEqSetMinimumDelay(const ParsedRtcEventLog& parsed_log,
+                                const AnalyzerConfig& config,
+                                Plot* plot) {
+  for (const auto& playout_stream :
+       parsed_log.neteq_set_minimum_delay_events()) {
+    uint32_t ssrc = playout_stream.first;
+    if (!MatchingSsrc(ssrc, config.desired_ssrc_))
+      continue;
+
+    TimeSeries time_series(SsrcToString(ssrc), LineStyle::kStep,
+                           PointStyle::kHighlight);
+    for (const auto& event : playout_stream.second) {
+      float x = config.GetCallTimeSec(event.log_time());
+      float y = event.minimum_delay_ms;
+      time_series.points.push_back(TimeSeriesPoint(x, y));
+    }
+    plot->AppendTimeSeries(std::move(time_series));
+  }
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1000, "Minimum Delay (ms)", kBottomMargin,
+                          kTopMargin);
+  plot->SetTitle("Set Minimum Delay");
+}
+
+// For audio SSRCs, plot the audio level.
+void CreateAudioLevelGraph(const ParsedRtcEventLog& parsed_log,
+                           const AnalyzerConfig& config,
+                           PacketDirection direction,
+                           Plot* plot) {
+  for (const auto& stream : parsed_log.rtp_packets_by_ssrc(direction)) {
+    if (!IsAudioSsrc(parsed_log, direction, stream.ssrc))
+      continue;
+    TimeSeries time_series(GetStreamName(parsed_log, direction, stream.ssrc),
+                           LineStyle::kLine);
+    for (auto& packet : stream.packet_view) {
+      if (packet.header.extension.audio_level()) {
+        float x = config.GetCallTimeSec(packet.log_time());
+        // The audio level is stored in -dBov (so e.g. -10 dBov is stored as 10)
+        // Here we convert it to dBov.
+        float y =
+            static_cast<float>(-packet.header.extension.audio_level()->level());
+        time_series.points.emplace_back(TimeSeriesPoint(x, y));
+      }
+    }
+    plot->AppendTimeSeries(std::move(time_series));
+  }
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetYAxis(-127, 0, "Audio level (dBov)", kBottomMargin, kTopMargin);
+  plot->SetTitle(GetDirectionAsString(direction) + " audio level");
+}
+
 }  // namespace webrtc
diff --git a/rtc_tools/rtc_event_log_visualizer/analyze_audio.h b/rtc_tools/rtc_event_log_visualizer/analyze_audio.h
index 094f9d91b6..55040ae39f 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyze_audio.h
+++ b/rtc_tools/rtc_event_log_visualizer/analyze_audio.h
@@ -45,6 +45,19 @@ void CreateAudioEncoderNumChannelsGraph(const ParsedRtcEventLog& parsed_log,
                                         const AnalyzerConfig& config,
                                         Plot* plot);
 
+void CreatePlayoutGraph(const ParsedRtcEventLog& parsed_log,
+                        const AnalyzerConfig& config,
+                        Plot* plot);
+
+void CreateNetEqSetMinimumDelay(const ParsedRtcEventLog& parsed_log,
+                                const AnalyzerConfig& config,
+                                Plot* plot);
+
+void CreateAudioLevelGraph(const ParsedRtcEventLog& parsed_log,
+                           const AnalyzerConfig& config,
+                           PacketDirection direction,
+                           Plot* plot);
+
 using NetEqStatsGetterMap =
     std::map<uint32_t, std::unique_ptr<test::NetEqStatsGetter>>;
 NetEqStatsGetterMap SimulateNetEq(const ParsedRtcEventLog& parsed_log,
diff --git a/rtc_tools/rtc_event_log_visualizer/analyze_bwe.cc b/rtc_tools/rtc_event_log_visualizer/analyze_bwe.cc
new file mode 100644
index 0000000000..0480256576
--- /dev/null
+++ b/rtc_tools/rtc_event_log_visualizer/analyze_bwe.cc
@@ -0,0 +1,1221 @@
+/*
+ *  Copyright (c) 2026 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "rtc_tools/rtc_event_log_visualizer/analyze_bwe.h"
+
+#include <algorithm>
+#include <cmath>
+#include <cstddef>
+#include <cstdint>
+#include <limits>
+#include <map>
+#include <memory>
+#include <optional>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/algorithm/container.h"
+#include "absl/functional/bind_front.h"
+#include "absl/strings/string_view.h"
+#include "api/environment/environment.h"
+#include "api/environment/environment_factory.h"
+#include "api/field_trials.h"
+#include "api/function_view.h"
+#include "api/media_types.h"
+#include "api/rtp_headers.h"
+#include "api/transport/bandwidth_usage.h"
+#include "api/transport/goog_cc_factory.h"
+#include "api/transport/network_control.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "logging/rtc_event_log/events/logged_rtp_rtcp.h"
+#include "logging/rtc_event_log/rtc_event_log_parser.h"
+#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.h"
+#include "modules/congestion_controller/include/receive_side_congestion_controller.h"
+#include "modules/congestion_controller/rtp/transport_feedback_adapter.h"
+#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/remb.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/network/sent_packet.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
+#include "rtc_base/rate_statistics.h"
+#include "rtc_tools/rtc_event_log_visualizer/analyzer_common.h"
+#include "rtc_tools/rtc_event_log_visualizer/log_scream_simulation.h"
+#include "rtc_tools/rtc_event_log_visualizer/log_simulation.h"
+#include "rtc_tools/rtc_event_log_visualizer/plot_base.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+namespace {
+
+double AbsSendTimeToMicroseconds(int64_t abs_send_time) {
+  // The timestamp is a fixed point representation with 6 bits for seconds
+  // and 18 bits for fractions of a second. Thus, we divide by 2^18 to get the
+  // time in seconds and then multiply by kNumMicrosecsPerSec to convert to
+  // microseconds.
+  static constexpr double kTimestampToMicroSec =
+      static_cast<double>(kNumMicrosecsPerSec) / static_cast<double>(1ul << 18);
+  return abs_send_time * kTimestampToMicroSec;
+}
+
+// This is much more reliable for outgoing streams than for incoming streams.
+template <typename RtpPacketContainer>
+std::optional<uint32_t> EstimateRtpClockFrequency(
+    const RtpPacketContainer& packets,
+    int64_t end_time_us) {
+  RTC_CHECK(packets.size() >= 2);
+  SeqNumUnwrapper<uint32_t> unwrapper;
+  int64_t first_rtp_timestamp =
+      unwrapper.Unwrap(packets[0].rtp.header.timestamp);
+  int64_t first_log_timestamp = packets[0].log_time_us();
+  int64_t last_rtp_timestamp = first_rtp_timestamp;
+  int64_t last_log_timestamp = first_log_timestamp;
+  for (size_t i = 1; i < packets.size(); i++) {
+    if (packets[i].log_time_us() > end_time_us)
+      break;
+    last_rtp_timestamp = unwrapper.Unwrap(packets[i].rtp.header.timestamp);
+    last_log_timestamp = packets[i].log_time_us();
+  }
+  if (last_log_timestamp - first_log_timestamp < kNumMicrosecsPerSec) {
+    RTC_LOG(LS_WARNING)
+        << "Failed to estimate RTP clock frequency: Stream too short. ("
+        << packets.size() << " packets, "
+        << last_log_timestamp - first_log_timestamp << " us)";
+    return std::nullopt;
+  }
+  double duration =
+      static_cast<double>(last_log_timestamp - first_log_timestamp) /
+      kNumMicrosecsPerSec;
+  double estimated_frequency =
+      (last_rtp_timestamp - first_rtp_timestamp) / duration;
+  for (uint32_t f : {8000, 16000, 32000, 48000, 90000}) {
+    if (std::fabs(estimated_frequency - f) < 0.15 * f) {
+      return f;
+    }
+  }
+  RTC_LOG(LS_WARNING) << "Failed to estimate RTP clock frequency: Estimate "
+                      << estimated_frequency
+                      << " not close to any standard RTP frequency."
+                      << " Last timestamp " << last_rtp_timestamp
+                      << " first timestamp " << first_rtp_timestamp;
+  return std::nullopt;
+}
+
+std::optional<double> NetworkDelayDiff_AbsSendTime(
+    const LoggedRtpPacketIncoming& old_packet,
+    const LoggedRtpPacketIncoming& new_packet) {
+  if (old_packet.rtp.header.extension.hasAbsoluteSendTime &&
+      new_packet.rtp.header.extension.hasAbsoluteSendTime) {
+    int64_t send_time_diff = WrappingDifference(
+        new_packet.rtp.header.extension.absoluteSendTime,
+        old_packet.rtp.header.extension.absoluteSendTime, 1ul << 24);
+    int64_t recv_time_diff =
+        new_packet.log_time_us() - old_packet.log_time_us();
+    double delay_change_us =
+        recv_time_diff - AbsSendTimeToMicroseconds(send_time_diff);
+    return delay_change_us / 1000;
+  } else {
+    return std::nullopt;
+  }
+}
+
+std::optional<double> NetworkDelayDiff_CaptureTime(
+    const LoggedRtpPacketIncoming& old_packet,
+    const LoggedRtpPacketIncoming& new_packet,
+    const double sample_rate) {
+  int64_t send_time_diff =
+      WrappingDifference(new_packet.rtp.header.timestamp,
+                         old_packet.rtp.header.timestamp, 1ull << 32);
+  int64_t recv_time_diff = new_packet.log_time_us() - old_packet.log_time_us();
+
+  double delay_change =
+      static_cast<double>(recv_time_diff) / 1000 -
+      static_cast<double>(send_time_diff) / sample_rate * 1000;
+  if (delay_change < -10000 || 10000 < delay_change) {
+    RTC_LOG(LS_WARNING) << "Very large delay change. Timestamps correct?";
+    RTC_LOG(LS_WARNING) << "Old capture time "
+                        << old_packet.rtp.header.timestamp << ", received time "
+                        << old_packet.log_time_us();
+    RTC_LOG(LS_WARNING) << "New capture time "
+                        << new_packet.rtp.header.timestamp << ", received time "
+                        << new_packet.log_time_us();
+    RTC_LOG(LS_WARNING) << "Receive time difference " << recv_time_diff << " = "
+                        << static_cast<double>(recv_time_diff) /
+                               kNumMicrosecsPerSec
+                        << "s";
+    RTC_LOG(LS_WARNING) << "Send time difference " << send_time_diff << " = "
+                        << static_cast<double>(send_time_diff) / sample_rate
+                        << "s";
+  }
+  return delay_change;
+}
+
+struct FakeExtensionSmall {
+  static constexpr RTPExtensionType kId = kRtpExtensionMid;
+  static constexpr absl::string_view Uri() { return "fake-extension-small"; }
+};
+struct FakeExtensionLarge {
+  static constexpr RTPExtensionType kId = kRtpExtensionRtpStreamId;
+  static constexpr absl::string_view Uri() { return "fake-extension-large"; }
+};
+
+RtpPacketReceived RtpPacketForBWEFromHeader(const RTPHeader& header) {
+  RtpHeaderExtensionMap rtp_header_extensions(/*extmap_allow_mixed=*/true);
+  // ReceiveSideCongestionController doesn't need to know extensions ids as
+  // long as it able to get extensions by type. So any ids would work here.
+  rtp_header_extensions.Register<TransmissionOffset>(1);
+  rtp_header_extensions.Register<AbsoluteSendTime>(2);
+  rtp_header_extensions.Register<TransportSequenceNumber>(3);
+  rtp_header_extensions.Register<FakeExtensionSmall>(4);
+  // Use id > 14 to force two byte header per rtp header when this one is used.
+  rtp_header_extensions.Register<FakeExtensionLarge>(16);
+
+  RtpPacketReceived rtp_packet(&rtp_header_extensions);
+  // Set only fields that might be relevant for the bandwidth estimatior.
+  rtp_packet.SetSsrc(header.ssrc);
+  rtp_packet.SetTimestamp(header.timestamp);
+  size_t num_bwe_extensions = 0;
+  if (header.extension.hasTransmissionTimeOffset) {
+    rtp_packet.SetExtension<TransmissionOffset>(
+        header.extension.transmissionTimeOffset);
+    ++num_bwe_extensions;
+  }
+  if (header.extension.hasAbsoluteSendTime) {
+    rtp_packet.SetExtension<AbsoluteSendTime>(
+        header.extension.absoluteSendTime);
+    ++num_bwe_extensions;
+  }
+  if (header.extension.hasTransportSequenceNumber) {
+    rtp_packet.SetExtension<TransportSequenceNumber>(
+        header.extension.transportSequenceNumber);
+    ++num_bwe_extensions;
+  }
+
+  // All parts of the RTP header are 32bit aligned.
+  RTC_CHECK_EQ(header.headerLength % 4, 0);
+
+  // Original packet could have more extensions, there could be csrcs that are
+  // not propagated by the rtc event log, i.e. logged header size might be
+  // larger that rtp_packet.header_size(). Increase it by setting an extra fake
+  // extension.
+  RTC_CHECK_GE(header.headerLength, rtp_packet.headers_size());
+  size_t bytes_to_add = header.headerLength - rtp_packet.headers_size();
+  if (bytes_to_add > 0) {
+    if (bytes_to_add <= 16) {
+      // one-byte header rtp header extension allows to add up to 16 bytes.
+      rtp_packet.AllocateExtension(FakeExtensionSmall::kId, bytes_to_add - 1);
+    } else {
+      // two-byte header rtp header extension would also add one byte per
+      // already set extension.
+      rtp_packet.AllocateExtension(FakeExtensionLarge::kId,
+                                   bytes_to_add - 2 - num_bwe_extensions);
+    }
+  }
+  RTC_CHECK_EQ(rtp_packet.headers_size(), header.headerLength);
+
+  return rtp_packet;
+}
+
+}  // namespace
+
+class BitrateObserver : public RemoteBitrateObserver {
+ public:
+  BitrateObserver() : last_bitrate_bps_(0), bitrate_updated_(false) {}
+
+  void Update(NetworkControlUpdate update) {
+    if (update.target_rate) {
+      last_bitrate_bps_ = update.target_rate->target_rate.bps();
+      bitrate_updated_ = true;
+    }
+  }
+
+  void OnReceiveBitrateChanged(const std::vector<uint32_t>& ssrcs,
+                               uint32_t bitrate) override {}
+
+  uint32_t last_bitrate_bps() const { return last_bitrate_bps_; }
+  bool GetAndResetBitrateUpdated() {
+    bool bitrate_updated = bitrate_updated_;
+    bitrate_updated_ = false;
+    return bitrate_updated;
+  }
+
+ private:
+  uint32_t last_bitrate_bps_;
+  bool bitrate_updated_;
+};
+
+void CreateIncomingDelayGraph(const ParsedRtcEventLog& parsed_log,
+                              const AnalyzerConfig& config,
+                              Plot* plot) {
+  for (const auto& stream : parsed_log.incoming_rtp_packets_by_ssrc()) {
+    // Filter on SSRC.
+    if (!MatchingSsrc(stream.ssrc, config.desired_ssrc_) ||
+        IsRtxSsrc(parsed_log, kIncomingPacket, stream.ssrc)) {
+      continue;
+    }
+
+    const std::vector<LoggedRtpPacketIncoming>& packets =
+        stream.incoming_packets;
+    if (packets.size() < 100) {
+      RTC_LOG(LS_WARNING) << "Can't estimate the RTP clock frequency with "
+                          << packets.size() << " packets in the stream.";
+      continue;
+    }
+    int64_t segment_end_us = parsed_log.first_log_segment().stop_time_us();
+    std::optional<uint32_t> estimated_frequency =
+        EstimateRtpClockFrequency(packets, segment_end_us);
+    if (!estimated_frequency)
+      continue;
+    const double frequency_hz = *estimated_frequency;
+    if (IsVideoSsrc(parsed_log, kIncomingPacket, stream.ssrc) &&
+        frequency_hz != 90000) {
+      RTC_LOG(LS_WARNING)
+          << "Video stream should use a 90 kHz clock but appears to use "
+          << frequency_hz / 1000 << ". Discarding.";
+      continue;
+    }
+
+    auto ToCallTime = [&config](const LoggedRtpPacketIncoming& packet) {
+      return config.GetCallTimeSec(packet.log_time());
+    };
+    auto ToNetworkDelay = [frequency_hz](
+                              const LoggedRtpPacketIncoming& old_packet,
+                              const LoggedRtpPacketIncoming& new_packet) {
+      return NetworkDelayDiff_CaptureTime(old_packet, new_packet, frequency_hz);
+    };
+
+    TimeSeries capture_time_data(
+        GetStreamName(parsed_log, kIncomingPacket, stream.ssrc) +
+            " capture-time",
+        LineStyle::kLine);
+    AccumulatePairs<LoggedRtpPacketIncoming, double>(
+        ToCallTime, ToNetworkDelay, packets, &capture_time_data);
+    plot->AppendTimeSeries(std::move(capture_time_data));
+
+    TimeSeries send_time_data(
+        GetStreamName(parsed_log, kIncomingPacket, stream.ssrc) +
+            " abs-send-time",
+        LineStyle::kLine);
+    AccumulatePairs<LoggedRtpPacketIncoming, double>(
+        ToCallTime, NetworkDelayDiff_AbsSendTime, packets, &send_time_data);
+    plot->AppendTimeSeriesIfNotEmpty(std::move(send_time_data));
+  }
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Delay (ms)", kBottomMargin, kTopMargin);
+  plot->SetTitle("Incoming network delay (relative to first packet)");
+}
+
+// Plot the fraction of packets lost (as perceived by the loss-based BWE).
+void CreateFractionLossGraph(const ParsedRtcEventLog& parsed_log,
+                             const AnalyzerConfig& config,
+                             Plot* plot) {
+  TimeSeries time_series("Fraction lost", LineStyle::kLine,
+                         PointStyle::kHighlight);
+  for (auto& bwe_update : parsed_log.bwe_loss_updates()) {
+    float x = config.GetCallTimeSec(bwe_update.log_time());
+    float y = static_cast<float>(bwe_update.fraction_lost) / 255 * 100;
+    time_series.points.emplace_back(x, y);
+  }
+
+  plot->AppendTimeSeries(std::move(time_series));
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 10, "Loss rate (in %)", kBottomMargin, kTopMargin);
+  plot->SetTitle("Outgoing packet loss (as reported by BWE)");
+}
+
+// Plot the total bandwidth used by all RTP streams.
+void CreateTotalIncomingBitrateGraph(const ParsedRtcEventLog& parsed_log,
+                                     const AnalyzerConfig& config,
+                                     Plot* plot) {
+  // TODO(terelius): This could be provided by the parser.
+  std::multimap<Timestamp, size_t> packets_in_order;
+  for (const auto& stream : parsed_log.incoming_rtp_packets_by_ssrc()) {
+    for (const LoggedRtpPacketIncoming& packet : stream.incoming_packets)
+      packets_in_order.insert(
+          std::make_pair(packet.rtp.log_time(), packet.rtp.total_length));
+  }
+
+  auto window_begin = packets_in_order.begin();
+  auto window_end = packets_in_order.begin();
+  size_t bytes_in_window = 0;
+
+  if (!packets_in_order.empty()) {
+    // Calculate a moving average of the bitrate and store in a TimeSeries.
+    TimeSeries bitrate_series("Bitrate", LineStyle::kLine);
+    for (Timestamp time = config.begin_time_;
+         time < config.end_time_ + config.step_; time += config.step_) {
+      while (window_end != packets_in_order.end() && window_end->first < time) {
+        bytes_in_window += window_end->second;
+        ++window_end;
+      }
+      while (window_begin != packets_in_order.end() &&
+             window_begin->first < time - config.window_duration_) {
+        RTC_DCHECK_LE(window_begin->second, bytes_in_window);
+        bytes_in_window -= window_begin->second;
+        ++window_begin;
+      }
+      float window_duration_in_seconds =
+          static_cast<float>(config.window_duration_.us()) /
+          kNumMicrosecsPerSec;
+      float x = config.GetCallTimeSec(time);
+      float y = bytes_in_window * 8 / window_duration_in_seconds / 1000;
+      bitrate_series.points.emplace_back(x, y);
+    }
+    plot->AppendTimeSeries(std::move(bitrate_series));
+  }
+
+  // Overlay the outgoing REMB over incoming bitrate.
+  TimeSeries remb_series("Remb", LineStyle::kStep);
+  for (const auto& rtcp : parsed_log.rembs(kOutgoingPacket)) {
+    float x = config.GetCallTimeSec(rtcp.log_time());
+    float y = static_cast<float>(rtcp.remb.bitrate_bps()) / 1000;
+    remb_series.points.emplace_back(x, y);
+  }
+  plot->AppendTimeSeriesIfNotEmpty(std::move(remb_series));
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin);
+  plot->SetTitle("Incoming RTP bitrate");
+}
+
+// Plot the total bandwidth used by all RTP streams.
+void CreateTotalOutgoingBitrateGraph(const ParsedRtcEventLog& parsed_log,
+                                     const AnalyzerConfig& config,
+                                     Plot* plot,
+                                     bool show_detector_state,
+                                     bool show_alr_state,
+                                     bool show_link_capacity) {
+  // TODO(terelius): This could be provided by the parser.
+  std::multimap<Timestamp, size_t> packets_in_order;
+  for (const auto& stream : parsed_log.outgoing_rtp_packets_by_ssrc()) {
+    for (const LoggedRtpPacketOutgoing& packet : stream.outgoing_packets)
+      packets_in_order.insert(
+          std::make_pair(packet.rtp.log_time(), packet.rtp.total_length));
+  }
+
+  auto window_begin = packets_in_order.begin();
+  auto window_end = packets_in_order.begin();
+  size_t bytes_in_window = 0;
+
+  if (!packets_in_order.empty()) {
+    // Calculate a moving average of the bitrate and store in a TimeSeries.
+    TimeSeries bitrate_series("Bitrate", LineStyle::kLine);
+    for (Timestamp time = config.begin_time_;
+         time < config.end_time_ + config.step_; time += config.step_) {
+      while (window_end != packets_in_order.end() && window_end->first < time) {
+        bytes_in_window += window_end->second;
+        ++window_end;
+      }
+      while (window_begin != packets_in_order.end() &&
+             window_begin->first < time - config.window_duration_) {
+        RTC_DCHECK_LE(window_begin->second, bytes_in_window);
+        bytes_in_window -= window_begin->second;
+        ++window_begin;
+      }
+      float window_duration_in_seconds =
+          static_cast<float>(config.window_duration_.us()) /
+          kNumMicrosecsPerSec;
+      float x = config.GetCallTimeSec(time);
+      float y = bytes_in_window * 8 / window_duration_in_seconds / 1000;
+      bitrate_series.points.emplace_back(x, y);
+    }
+    plot->AppendTimeSeries(std::move(bitrate_series));
+  }
+
+  // Overlay the send-side bandwidth estimate over the outgoing bitrate.
+  TimeSeries loss_series("Loss-based estimate", LineStyle::kStep);
+  for (auto& loss_update : parsed_log.bwe_loss_updates()) {
+    float x = config.GetCallTimeSec(loss_update.log_time());
+    float y = static_cast<float>(loss_update.bitrate_bps) / 1000;
+    loss_series.points.emplace_back(x, y);
+  }
+
+  TimeSeries link_capacity_lower_series("Link-capacity-lower",
+                                        LineStyle::kStep);
+  TimeSeries link_capacity_upper_series("Link-capacity-upper",
+                                        LineStyle::kStep);
+  for (auto& remote_estimate_event : parsed_log.remote_estimate_events()) {
+    float x = config.GetCallTimeSec(remote_estimate_event.log_time());
+    if (remote_estimate_event.link_capacity_lower.has_value()) {
+      float link_capacity_lower = static_cast<float>(
+          remote_estimate_event.link_capacity_lower.value().kbps());
+      link_capacity_lower_series.points.emplace_back(x, link_capacity_lower);
+    }
+    if (remote_estimate_event.link_capacity_upper.has_value()) {
+      float link_capacity_upper = static_cast<float>(
+          remote_estimate_event.link_capacity_upper.value().kbps());
+      link_capacity_upper_series.points.emplace_back(x, link_capacity_upper);
+    }
+  }
+
+  TimeSeries delay_series("Delay-based estimate", LineStyle::kStep);
+  IntervalSeries overusing_series("Overusing", "#ff8e82",
+                                  IntervalSeries::kHorizontal);
+  IntervalSeries underusing_series("Underusing", "#5092fc",
+                                   IntervalSeries::kHorizontal);
+  IntervalSeries normal_series("Normal", "#c4ffc4",
+                               IntervalSeries::kHorizontal);
+  IntervalSeries* last_series = &normal_series;
+  float last_detector_switch = 0.0;
+
+  BandwidthUsage last_detector_state = BandwidthUsage::kBwNormal;
+
+  for (auto& delay_update : parsed_log.bwe_delay_updates()) {
+    float x = config.GetCallTimeSec(delay_update.log_time());
+    float y = static_cast<float>(delay_update.bitrate_bps) / 1000;
+
+    if (last_detector_state != delay_update.detector_state) {
+      last_series->intervals.emplace_back(last_detector_switch, x);
+      last_detector_state = delay_update.detector_state;
+      last_detector_switch = x;
+
+      switch (delay_update.detector_state) {
+        case BandwidthUsage::kBwNormal:
+          last_series = &normal_series;
+          break;
+        case BandwidthUsage::kBwUnderusing:
+          last_series = &underusing_series;
+          break;
+        case BandwidthUsage::kBwOverusing:
+          last_series = &overusing_series;
+          break;
+        case BandwidthUsage::kLast:
+          RTC_DCHECK_NOTREACHED();
+      }
+    }
+
+    delay_series.points.emplace_back(x, y);
+  }
+
+  RTC_CHECK(last_series);
+  last_series->intervals.emplace_back(last_detector_switch,
+                                      config.CallEndTimeSec());
+
+  TimeSeries scream_series("Scream target rate", LineStyle::kStep);
+  for (auto& scream_update : parsed_log.bwe_scream_updates()) {
+    float x = config.GetCallTimeSec(scream_update.log_time());
+    float y = static_cast<float>(scream_update.target_rate.kbps());
+    scream_series.points.emplace_back(x, y);
+  }
+
+  TimeSeries created_series("Probe cluster created.", LineStyle::kNone,
+                            PointStyle::kHighlight);
+  for (auto& cluster : parsed_log.bwe_probe_cluster_created_events()) {
+    float x = config.GetCallTimeSec(cluster.log_time());
+    float y = static_cast<float>(cluster.bitrate_bps) / 1000;
+    created_series.points.emplace_back(x, y);
+  }
+
+  TimeSeries result_series("Probing results.", LineStyle::kNone,
+                           PointStyle::kHighlight);
+  for (auto& result : parsed_log.bwe_probe_success_events()) {
+    float x = config.GetCallTimeSec(result.log_time());
+    float y = static_cast<float>(result.bitrate_bps) / 1000;
+    result_series.points.emplace_back(x, y);
+  }
+
+  TimeSeries probe_failures_series("Probe failed", LineStyle::kNone,
+                                   PointStyle::kHighlight);
+  for (auto& failure : parsed_log.bwe_probe_failure_events()) {
+    float x = config.GetCallTimeSec(failure.log_time());
+    probe_failures_series.points.emplace_back(x, 0);
+  }
+
+  IntervalSeries alr_state("ALR", "#555555", IntervalSeries::kHorizontal);
+  bool previously_in_alr = false;
+  Timestamp alr_start = Timestamp::Zero();
+  for (auto& alr : parsed_log.alr_state_events()) {
+    float y = config.GetCallTimeSec(alr.log_time());
+    if (!previously_in_alr && alr.in_alr) {
+      alr_start = alr.log_time();
+      previously_in_alr = true;
+    } else if (previously_in_alr && !alr.in_alr) {
+      float x = config.GetCallTimeSec(alr_start);
+      alr_state.intervals.emplace_back(x, y);
+      previously_in_alr = false;
+    }
+  }
+
+  if (previously_in_alr) {
+    float x = config.GetCallTimeSec(alr_start);
+    float y = config.GetCallTimeSec(config.end_time_);
+    alr_state.intervals.emplace_back(x, y);
+  }
+
+  if (show_detector_state) {
+    plot->AppendIntervalSeries(std::move(overusing_series));
+    plot->AppendIntervalSeries(std::move(underusing_series));
+    plot->AppendIntervalSeries(std::move(normal_series));
+  }
+
+  if (show_alr_state) {
+    plot->AppendIntervalSeries(std::move(alr_state));
+  }
+
+  if (show_link_capacity) {
+    plot->AppendTimeSeriesIfNotEmpty(std::move(link_capacity_lower_series));
+    plot->AppendTimeSeriesIfNotEmpty(std::move(link_capacity_upper_series));
+  }
+
+  plot->AppendTimeSeries(std::move(loss_series));
+  plot->AppendTimeSeriesIfNotEmpty(std::move(probe_failures_series));
+  plot->AppendTimeSeries(std::move(delay_series));
+  plot->AppendTimeSeriesIfNotEmpty(std::move(scream_series));
+  plot->AppendTimeSeries(std::move(created_series));
+  plot->AppendTimeSeries(std::move(result_series));
+
+  // Overlay the incoming REMB over the outgoing bitrate.
+  TimeSeries remb_series("Remb", LineStyle::kStep);
+  for (const auto& rtcp : parsed_log.rembs(kIncomingPacket)) {
+    float x = config.GetCallTimeSec(rtcp.log_time());
+    float y = static_cast<float>(rtcp.remb.bitrate_bps()) / 1000;
+    remb_series.points.emplace_back(x, y);
+  }
+  plot->AppendTimeSeriesIfNotEmpty(std::move(remb_series));
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin);
+  plot->SetTitle("Outgoing RTP bitrate");
+}
+
+void CreateGoogCcSimulationGraph(const ParsedRtcEventLog& parsed_log,
+                                 const AnalyzerConfig& config,
+                                 Plot* plot) {
+  TimeSeries target_rates("Simulated target rate", LineStyle::kStep,
+                          PointStyle::kHighlight);
+  TimeSeries delay_based("Logged delay-based estimate", LineStyle::kStep,
+                         PointStyle::kHighlight);
+  TimeSeries loss_based("Logged loss-based estimate", LineStyle::kStep,
+                        PointStyle::kHighlight);
+  TimeSeries probe_results("Logged probe success", LineStyle::kNone,
+                           PointStyle::kHighlight);
+
+  LogBasedNetworkControllerSimulation simulation(
+      config.env_, std::make_unique<GoogCcNetworkControllerFactory>(),
+      [&](const NetworkControlUpdate& update, Timestamp at_time) {
+        if (update.target_rate) {
+          target_rates.points.emplace_back(
+              config.GetCallTimeSec(at_time),
+              update.target_rate->target_rate.kbps<float>());
+        }
+      });
+
+  simulation.ProcessEventsInLog(parsed_log);
+  for (const auto& logged : parsed_log.bwe_delay_updates())
+    delay_based.points.emplace_back(config.GetCallTimeSec(logged.log_time()),
+                                    logged.bitrate_bps / 1000);
+  for (const auto& logged : parsed_log.bwe_probe_success_events())
+    probe_results.points.emplace_back(config.GetCallTimeSec(logged.log_time()),
+                                      logged.bitrate_bps / 1000);
+  for (const auto& logged : parsed_log.bwe_loss_updates())
+    loss_based.points.emplace_back(config.GetCallTimeSec(logged.log_time()),
+                                   logged.bitrate_bps / 1000);
+
+  plot->AppendTimeSeries(std::move(delay_based));
+  plot->AppendTimeSeries(std::move(loss_based));
+  plot->AppendTimeSeries(std::move(probe_results));
+  plot->AppendTimeSeries(std::move(target_rates));
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 10, "Bitrate (kbps)", kBottomMargin, kTopMargin);
+  plot->SetTitle("Simulated BWE behavior");
+}
+
+void CreateScreamSimulationBitrateGraph(const ParsedRtcEventLog& parsed_log,
+                                        const AnalyzerConfig& config,
+                                        Plot* plot) {
+  TimeSeries target_rate_series("Target rate", LineStyle::kStep);
+  TimeSeries pacing_rate_series("Pacing rate", LineStyle::kStep);
+  TimeSeries send_rate_series("Send rate", LineStyle::kStep);
+
+  LogScreamSimulation simulation({.rate_window = config.window_duration_},
+                                 config.env_);
+  simulation.ProcessEventsInLog(parsed_log);
+
+  for (const LogScreamSimulation::State& state : simulation.updates()) {
+    target_rate_series.points.emplace_back(config.GetCallTimeSec(state.time),
+                                           state.target_rate.bps() / 1000);
+    pacing_rate_series.points.emplace_back(config.GetCallTimeSec(state.time),
+                                           state.pacing_rate.bps() / 1000);
+    send_rate_series.points.emplace_back(config.GetCallTimeSec(state.time),
+                                         state.send_rate.bps() / 1000);
+  }
+  plot->AppendTimeSeries(std::move(target_rate_series));
+  plot->AppendTimeSeries(std::move(pacing_rate_series));
+  plot->AppendTimeSeries(std::move(send_rate_series));
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 100, "Kbps", kBottomMargin, kTopMargin);
+  plot->SetTitle("Simulated Scream rates");
+}
+
+void CreateScreamSimulationRefWindowGraph(const ParsedRtcEventLog& parsed_log,
+                                          const AnalyzerConfig& config,
+                                          Plot* plot) {
+  using SendWindowUsage = LogScreamSimulation::State::SendWindowUsage;
+  TimeSeries ref_window_series("RefWindow", LineStyle::kStep);
+  TimeSeries ref_window_i_series("RefWindowI", LineStyle::kStep);
+  TimeSeries max_data_in_flight("Max allowed data in flight", LineStyle::kStep);
+  TimeSeries data_in_flight("Data in flight", LineStyle::kStep);
+  IntervalSeries send_window_above_max_series(
+      "Data in flight > Max allowed", "#ff8e82", IntervalSeries::kHorizontal);
+  IntervalSeries send_window_below_ref_window_series(
+      "Data in flight < RefWindow", "#c5dff2", IntervalSeries::kHorizontal);
+  IntervalSeries send_window_above_ref_window_series(
+      "Data in flight >= RefWindow", "#b9fad8", IntervalSeries::kHorizontal);
+  IntervalSeries* last_series = &send_window_below_ref_window_series;
+
+  LogScreamSimulation simulation({.rate_window = config.window_duration_},
+                                 config.env_);
+  simulation.ProcessEventsInLog(parsed_log);
+  if (simulation.updates().empty()) {
+    RTC_LOG(LS_ERROR) << "Empty simulation.";
+    return;
+  }
+
+  float send_window_state_switch =
+      config.GetCallTimeSec(simulation.updates().front().time);
+  SendWindowUsage send_window_usage = SendWindowUsage::kBelowRefWindow;
+
+  for (const LogScreamSimulation::State& state : simulation.updates()) {
+    ref_window_series.points.emplace_back(config.GetCallTimeSec(state.time),
+                                          state.ref_window.bytes());
+    ref_window_i_series.points.emplace_back(config.GetCallTimeSec(state.time),
+                                            state.ref_window_i.bytes());
+    max_data_in_flight.points.emplace_back(config.GetCallTimeSec(state.time),
+                                           state.max_data_in_flight.bytes());
+    data_in_flight.points.emplace_back(config.GetCallTimeSec(state.time),
+                                       state.data_in_flight.bytes());
+    if (state.send_window_usage != send_window_usage) {
+      last_series->intervals.emplace_back(send_window_state_switch,
+                                          config.GetCallTimeSec(state.time));
+      send_window_usage = state.send_window_usage;
+      send_window_state_switch = config.GetCallTimeSec(state.time);
+      switch (send_window_usage) {
+        case SendWindowUsage::kAboveRefWindow:
+          last_series = &send_window_above_ref_window_series;
+          break;
+        case SendWindowUsage::kBelowRefWindow:
+          last_series = &send_window_below_ref_window_series;
+          break;
+        case SendWindowUsage::kAboveScreamMax:
+          last_series = &send_window_above_max_series;
+          break;
+      }
+    }
+  }
+  last_series->intervals.emplace_back(send_window_state_switch,
+                                      config.CallEndTimeSec());
+  plot->AppendTimeSeries(std::move(ref_window_series));
+  plot->AppendTimeSeries(std::move(ref_window_i_series));
+  plot->AppendTimeSeries(std::move(max_data_in_flight));
+  plot->AppendTimeSeries(std::move(data_in_flight));
+  plot->AppendIntervalSeries(std::move(send_window_above_max_series));
+  plot->AppendIntervalSeries(std::move(send_window_below_ref_window_series));
+  plot->AppendIntervalSeries(std::move(send_window_above_ref_window_series));
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 10, "Bytes", kBottomMargin, kTopMargin);
+  plot->SetTitle("Simulated Scream RefWindow");
+}
+
+void CreateScreamSimulationRatiosGraph(const ParsedRtcEventLog& parsed_log,
+                                       const AnalyzerConfig& config,
+                                       Plot* plot) {
+  TimeSeries queue_delay_dev_norm_series("QueueDelayDevNorm", LineStyle::kStep);
+  TimeSeries l4s_alpha_series("L4sAlpha", LineStyle::kStep);
+  TimeSeries l4s_alpha_v_series("L4sAlphaV", LineStyle::kStep);
+  TimeSeries ref_window_scale_factor_due_to_increased_delay(
+      "RefWindowScaleFactorDueToIncreasedDelay", LineStyle::kStep);
+  TimeSeries ref_window_scale_factor_due_to_delay_variation(
+      "RefWindowScaleFactorDueToDelayVariation", LineStyle::kStep);
+
+  LogScreamSimulation simulation({.rate_window = config.window_duration_},
+                                 config.env_);
+  simulation.ProcessEventsInLog(parsed_log);
+
+  for (const LogScreamSimulation::State& state : simulation.updates()) {
+    queue_delay_dev_norm_series.points.emplace_back(
+        config.GetCallTimeSec(state.time), state.queue_delay_dev_norm);
+    l4s_alpha_series.points.emplace_back(config.GetCallTimeSec(state.time),
+                                         state.l4s_alpha);
+    l4s_alpha_v_series.points.emplace_back(config.GetCallTimeSec(state.time),
+                                           state.l4s_alpha_v);
+    ref_window_scale_factor_due_to_increased_delay.points.emplace_back(
+        config.GetCallTimeSec(state.time),
+        state.ref_window_scale_factor_due_to_increased_delay);
+    ref_window_scale_factor_due_to_delay_variation.points.emplace_back(
+        config.GetCallTimeSec(state.time),
+        state.ref_window_scale_factor_due_to_delay_variation);
+  }
+  plot->AppendTimeSeries(std::move(queue_delay_dev_norm_series));
+  plot->AppendTimeSeries(std::move(l4s_alpha_series));
+  plot->AppendTimeSeries(std::move(l4s_alpha_v_series));
+  plot->AppendTimeSeries(
+      std::move(ref_window_scale_factor_due_to_increased_delay));
+  plot->AppendTimeSeries(
+      std::move(ref_window_scale_factor_due_to_delay_variation));
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Ratios", kBottomMargin, kTopMargin);
+  plot->SetTitle("Simulated Scream Ratios");
+}
+
+void CreateScreamRefWindowGraph(const ParsedRtcEventLog& parsed_log,
+                                const AnalyzerConfig& config,
+                                Plot* plot) {
+  TimeSeries ref_window_series("RefWindow", LineStyle::kStep);
+  for (auto& scream_update : parsed_log.bwe_scream_updates()) {
+    float x = config.GetCallTimeSec(scream_update.log_time());
+    float y = static_cast<float>(scream_update.ref_window.bytes());
+    ref_window_series.points.emplace_back(x, y);
+  }
+  plot->AppendTimeSeries(std::move(ref_window_series));
+
+  TimeSeries data_in_flight_series("Data in flight", LineStyle::kLine);
+  for (auto& scream_update : parsed_log.bwe_scream_updates()) {
+    float x = config.GetCallTimeSec(scream_update.log_time());
+    float y = static_cast<float>(scream_update.data_in_flight.bytes());
+    data_in_flight_series.points.emplace_back(x, y);
+  }
+  plot->AppendTimeSeries(std::move(data_in_flight_series));
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 3000, "Bytes", kBottomMargin, kTopMargin);
+  plot->SetTitle("Scream Ref Window");
+}
+
+void CreateScreamDelayEstimateGraph(const ParsedRtcEventLog& parsed_log,
+                                    const AnalyzerConfig& config,
+                                    Plot* plot) {
+  TimeSeries smoothed_rtt_series("Smoothed RTT", LineStyle::kStep);
+  TimeSeries avg_queue_delay_series("Avg queue delay", LineStyle::kStep);
+
+  for (auto& scream_update : parsed_log.bwe_scream_updates()) {
+    float x = config.GetCallTimeSec(scream_update.log_time());
+    float smoothed_rtt_ms = static_cast<float>(scream_update.smoothed_rtt.ms());
+    smoothed_rtt_series.points.emplace_back(x, smoothed_rtt_ms);
+    float avg_queue_delay_ms =
+        static_cast<float>(scream_update.avg_queue_delay.ms());
+    avg_queue_delay_series.points.emplace_back(x, avg_queue_delay_ms);
+  }
+
+  plot->AppendTimeSeries(std::move(smoothed_rtt_series));
+  plot->AppendTimeSeries(std::move(avg_queue_delay_series));
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 50, "Delay (ms)", kBottomMargin, kTopMargin);
+  plot->SetTitle("Scream delay estimates");
+}
+
+void CreateSendSideBweSimulationGraph(const ParsedRtcEventLog& parsed_log,
+                                      const AnalyzerConfig& config,
+                                      Plot* plot) {
+  using RtpPacketType = LoggedRtpPacketOutgoing;
+  using TransportFeedbackType = LoggedRtcpPacketTransportFeedback;
+
+  // TODO(terelius): The parser could provide a clearer view of the
+  // streams, so that we don't have to recalculate it.
+  std::multimap<int64_t, const RtpPacketType*> outgoing_rtp;
+  for (const auto& stream : parsed_log.outgoing_rtp_packets_by_ssrc()) {
+    for (const RtpPacketType& rtp_packet : stream.outgoing_packets)
+      outgoing_rtp.insert(
+          std::make_pair(rtp_packet.rtp.log_time_us(), &rtp_packet));
+  }
+
+  const std::vector<TransportFeedbackType>& incoming_rtcp =
+      parsed_log.transport_feedbacks(kIncomingPacket);
+
+  SimulatedClock clock(0);
+  BitrateObserver observer;
+  TransportFeedbackAdapter transport_feedback;
+  auto factory = GoogCcNetworkControllerFactory();
+  TimeDelta process_interval = factory.GetProcessInterval();
+  // TODO(holmer): Log the call config and use that here instead.
+  static const uint32_t kDefaultStartBitrateBps = 300000;
+  NetworkControllerConfig cc_config(config.env_);
+  cc_config.constraints.at_time = clock.CurrentTime();
+  cc_config.constraints.starting_rate =
+      DataRate::BitsPerSec(kDefaultStartBitrateBps);
+  auto goog_cc = factory.Create(cc_config);
+
+  TimeSeries time_series("Delay-based estimate", LineStyle::kStep,
+                         PointStyle::kHighlight);
+  TimeSeries acked_time_series("Raw acked bitrate", LineStyle::kLine,
+                               PointStyle::kHighlight);
+  TimeSeries robust_time_series("Robust throughput estimate", LineStyle::kLine,
+                                PointStyle::kHighlight);
+  TimeSeries acked_estimate_time_series("Ackednowledged bitrate estimate",
+                                        LineStyle::kLine,
+                                        PointStyle::kHighlight);
+
+  auto rtp_iterator = outgoing_rtp.begin();
+  auto rtcp_iterator = incoming_rtcp.begin();
+
+  auto NextRtpTime = [&]() {
+    if (rtp_iterator != outgoing_rtp.end())
+      return static_cast<int64_t>(rtp_iterator->first);
+    return std::numeric_limits<int64_t>::max();
+  };
+
+  auto NextRtcpTime = [&]() {
+    if (rtcp_iterator != incoming_rtcp.end())
+      return static_cast<int64_t>(rtcp_iterator->log_time_us());
+    return std::numeric_limits<int64_t>::max();
+  };
+  int64_t next_process_time_us_ = std::min({NextRtpTime(), NextRtcpTime()});
+
+  auto NextProcessTime = [&]() {
+    if (rtcp_iterator != incoming_rtcp.end() ||
+        rtp_iterator != outgoing_rtp.end()) {
+      return next_process_time_us_;
+    }
+    return std::numeric_limits<int64_t>::max();
+  };
+
+  RateStatistics raw_acked_bitrate(750, 8000);
+  FieldTrials throughput_config(
+      "WebRTC-Bwe-RobustThroughputEstimatorSettings/enabled:true/");
+  std::unique_ptr<AcknowledgedBitrateEstimatorInterface>
+      robust_throughput_estimator(
+          AcknowledgedBitrateEstimatorInterface::Create(&throughput_config));
+  FieldTrials acked_bitrate_config(
+      "WebRTC-Bwe-RobustThroughputEstimatorSettings/enabled:false/");
+  std::unique_ptr<AcknowledgedBitrateEstimatorInterface>
+      acknowledged_bitrate_estimator(
+          AcknowledgedBitrateEstimatorInterface::Create(&acked_bitrate_config));
+  int64_t time_us =
+      std::min({NextRtpTime(), NextRtcpTime(), NextProcessTime()});
+  int64_t last_update_us = 0;
+  while (time_us != std::numeric_limits<int64_t>::max()) {
+    clock.AdvanceTimeMicroseconds(time_us - clock.TimeInMicroseconds());
+    if (clock.TimeInMicroseconds() >= NextRtpTime()) {
+      RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtpTime());
+      const RtpPacketType& rtp_packet = *rtp_iterator->second;
+      if (rtp_packet.rtp.header.extension.hasTransportSequenceNumber) {
+        RtpPacketToSend send_packet(/*extensions=*/nullptr);
+        send_packet.set_transport_sequence_number(
+            rtp_packet.rtp.header.extension.transportSequenceNumber);
+        send_packet.SetSsrc(rtp_packet.rtp.header.ssrc);
+        send_packet.SetSequenceNumber(rtp_packet.rtp.header.sequenceNumber);
+        send_packet.SetPayloadSize(rtp_packet.rtp.total_length -
+                                   send_packet.headers_size());
+        RTC_DCHECK_EQ(send_packet.size(), rtp_packet.rtp.total_length);
+        if (IsRtxSsrc(parsed_log, PacketDirection::kOutgoingPacket,
+                      rtp_packet.rtp.header.ssrc)) {
+          // Don't set the optional media type as we don't know if it is
+          // a retransmission, FEC or padding.
+        } else if (IsVideoSsrc(parsed_log, PacketDirection::kOutgoingPacket,
+                               rtp_packet.rtp.header.ssrc)) {
+          send_packet.set_packet_type(RtpPacketMediaType::kVideo);
+        } else if (IsAudioSsrc(parsed_log, PacketDirection::kOutgoingPacket,
+                               rtp_packet.rtp.header.ssrc)) {
+          send_packet.set_packet_type(RtpPacketMediaType::kAudio);
+        }
+        transport_feedback.AddPacket(
+            send_packet, PacedPacketInfo(),
+            0u,  // Per packet overhead bytes.,
+            Timestamp::Micros(rtp_packet.rtp.log_time_us()));
+      }
+      SentPacketInfo sent_packet;
+      sent_packet.send_time_ms = rtp_packet.rtp.log_time_ms();
+      sent_packet.info.included_in_allocation = true;
+      sent_packet.info.packet_size_bytes = rtp_packet.rtp.total_length;
+      if (rtp_packet.rtp.header.extension.hasTransportSequenceNumber) {
+        sent_packet.packet_id =
+            rtp_packet.rtp.header.extension.transportSequenceNumber;
+        sent_packet.info.included_in_feedback = true;
+      }
+      auto sent_msg = transport_feedback.ProcessSentPacket(sent_packet);
+      if (sent_msg)
+        observer.Update(goog_cc->OnSentPacket(*sent_msg));
+      ++rtp_iterator;
+    }
+    if (clock.TimeInMicroseconds() >= NextRtcpTime()) {
+      RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtcpTime());
+
+      auto feedback_msg = transport_feedback.ProcessTransportFeedback(
+          rtcp_iterator->transport_feedback, clock.CurrentTime());
+      if (feedback_msg) {
+        observer.Update(goog_cc->OnTransportPacketsFeedback(*feedback_msg));
+        std::vector<PacketResult> feedback =
+            feedback_msg->SortedByReceiveTime();
+        if (!feedback.empty()) {
+          acknowledged_bitrate_estimator->IncomingPacketFeedbackVector(
+              feedback);
+          robust_throughput_estimator->IncomingPacketFeedbackVector(feedback);
+          for (const PacketResult& packet : feedback) {
+            raw_acked_bitrate.Update(packet.sent_packet.size.bytes(),
+                                     packet.receive_time.ms());
+          }
+          std::optional<uint32_t> raw_bitrate_bps =
+              raw_acked_bitrate.Rate(feedback.back().receive_time.ms());
+          float x = config.GetCallTimeSec(clock.CurrentTime());
+          if (raw_bitrate_bps) {
+            float y = raw_bitrate_bps.value() / 1000;
+            acked_time_series.points.emplace_back(x, y);
+          }
+          std::optional<DataRate> robust_estimate =
+              robust_throughput_estimator->bitrate();
+          if (robust_estimate) {
+            float y = robust_estimate.value().kbps();
+            robust_time_series.points.emplace_back(x, y);
+          }
+          std::optional<DataRate> acked_estimate =
+              acknowledged_bitrate_estimator->bitrate();
+          if (acked_estimate) {
+            float y = acked_estimate.value().kbps();
+            acked_estimate_time_series.points.emplace_back(x, y);
+          }
+        }
+      }
+      ++rtcp_iterator;
+    }
+    if (clock.TimeInMicroseconds() >= NextProcessTime()) {
+      RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextProcessTime());
+      ProcessInterval msg;
+      msg.at_time = clock.CurrentTime();
+      observer.Update(goog_cc->OnProcessInterval(msg));
+      next_process_time_us_ += process_interval.us();
+    }
+    if (observer.GetAndResetBitrateUpdated() ||
+        time_us - last_update_us >= 1e6) {
+      uint32_t y = observer.last_bitrate_bps() / 1000;
+      float x = config.GetCallTimeSec(clock.CurrentTime());
+      time_series.points.emplace_back(x, y);
+      last_update_us = time_us;
+    }
+    time_us = std::min({NextRtpTime(), NextRtcpTime(), NextProcessTime()});
+  }
+  // Add the data set to the plot.
+  plot->AppendTimeSeries(std::move(time_series));
+  plot->AppendTimeSeries(std::move(robust_time_series));
+  plot->AppendTimeSeries(std::move(acked_time_series));
+  plot->AppendTimeSeriesIfNotEmpty(std::move(acked_estimate_time_series));
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 10, "Bitrate (kbps)", kBottomMargin, kTopMargin);
+  plot->SetTitle("Simulated send-side BWE behavior");
+}
+
+void CreateReceiveSideBweSimulationGraph(const ParsedRtcEventLog& parsed_log,
+                                         const AnalyzerConfig& config,
+                                         Plot* plot) {
+  using RtpPacketType = LoggedRtpPacketIncoming;
+  class RembInterceptor {
+   public:
+    void SendRemb(uint32_t bitrate_bps, std::vector<uint32_t> ssrcs) {
+      last_bitrate_bps_ = bitrate_bps;
+      bitrate_updated_ = true;
+    }
+    uint32_t last_bitrate_bps() const { return last_bitrate_bps_; }
+    bool GetAndResetBitrateUpdated() {
+      bool bitrate_updated = bitrate_updated_;
+      bitrate_updated_ = false;
+      return bitrate_updated;
+    }
+
+   private:
+    // We don't know the start bitrate, but assume that it is the default 300
+    // kbps.
+    uint32_t last_bitrate_bps_ = 300000;
+    bool bitrate_updated_ = false;
+  };
+
+  std::multimap<int64_t, const RtpPacketType*> incoming_rtp;
+
+  for (const auto& stream : parsed_log.incoming_rtp_packets_by_ssrc()) {
+    if (IsVideoSsrc(parsed_log, kIncomingPacket, stream.ssrc)) {
+      for (const auto& rtp_packet : stream.incoming_packets)
+        incoming_rtp.insert(
+            std::make_pair(rtp_packet.rtp.log_time_us(), &rtp_packet));
+    }
+  }
+
+  SimulatedClock clock(0);
+  EnvironmentFactory env_factory(config.env_);
+  env_factory.Set(&clock);
+  RembInterceptor remb_interceptor;
+  ReceiveSideCongestionController rscc(
+      env_factory.Create(), [](auto...) {},
+      absl::bind_front(&RembInterceptor::SendRemb, &remb_interceptor));
+  // TODO(holmer): Log the call config and use that here instead.
+  // static const uint32_t kDefaultStartBitrateBps = 300000;
+  // rscc.SetBweBitrates(0, kDefaultStartBitrateBps, -1);
+
+  TimeSeries time_series("Receive side estimate", LineStyle::kLine,
+                         PointStyle::kHighlight);
+  TimeSeries acked_time_series("Received bitrate", LineStyle::kLine);
+
+  RateStatistics acked_bitrate(250, 8000);
+  int64_t last_update_us = 0;
+  for (const auto& kv : incoming_rtp) {
+    const RtpPacketType& packet = *kv.second;
+
+    RtpPacketReceived rtp_packet = RtpPacketForBWEFromHeader(packet.rtp.header);
+    rtp_packet.set_arrival_time(packet.rtp.log_time());
+    rtp_packet.SetPayloadSize(packet.rtp.total_length -
+                              rtp_packet.headers_size());
+
+    clock.AdvanceTime(rtp_packet.arrival_time() - clock.CurrentTime());
+    rscc.OnReceivedPacket(rtp_packet, MediaType::VIDEO);
+    int64_t arrival_time_ms = packet.rtp.log_time().ms();
+    acked_bitrate.Update(packet.rtp.total_length, arrival_time_ms);
+    std::optional<uint32_t> bitrate_bps = acked_bitrate.Rate(arrival_time_ms);
+    if (bitrate_bps) {
+      uint32_t y = *bitrate_bps / 1000;
+      float x = config.GetCallTimeSec(clock.CurrentTime());
+      acked_time_series.points.emplace_back(x, y);
+    }
+    if (remb_interceptor.GetAndResetBitrateUpdated() ||
+        clock.TimeInMicroseconds() - last_update_us >= 1e6) {
+      uint32_t y = remb_interceptor.last_bitrate_bps() / 1000;
+      float x = config.GetCallTimeSec(clock.CurrentTime());
+      time_series.points.emplace_back(x, y);
+      last_update_us = clock.TimeInMicroseconds();
+    }
+  }
+  // Add the data set to the plot.
+  plot->AppendTimeSeries(std::move(time_series));
+  plot->AppendTimeSeries(std::move(acked_time_series));
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 10, "Bitrate (kbps)", kBottomMargin, kTopMargin);
+  plot->SetTitle("Simulated receive-side BWE behavior");
+}
+
+void CreateNetworkDelayFeedbackGraph(const ParsedRtcEventLog& parsed_log,
+                                     const AnalyzerConfig& config,
+                                     Plot* plot) {
+  TimeSeries time_series("Network delay", LineStyle::kLine,
+                         PointStyle::kHighlight);
+  int64_t min_send_receive_diff_ms = std::numeric_limits<int64_t>::max();
+  int64_t min_rtt_ms = std::numeric_limits<int64_t>::max();
+
+  std::vector<MatchedSendArrivalTimes> matched_rtp_rtcp =
+      GetNetworkTrace(parsed_log);
+  absl::c_stable_sort(matched_rtp_rtcp, [](const MatchedSendArrivalTimes& a,
+                                           const MatchedSendArrivalTimes& b) {
+    return a.feedback_arrival_time_ms < b.feedback_arrival_time_ms ||
+           (a.feedback_arrival_time_ms == b.feedback_arrival_time_ms &&
+            a.arrival_time_ms < b.arrival_time_ms);
+  });
+  for (const auto& packet : matched_rtp_rtcp) {
+    if (packet.arrival_time_ms == MatchedSendArrivalTimes::kNotReceived)
+      continue;
+    float x = config.GetCallTimeSecFromMs(packet.feedback_arrival_time_ms);
+    int64_t y = packet.arrival_time_ms - packet.send_time_ms;
+    int64_t rtt_ms = packet.feedback_arrival_time_ms - packet.send_time_ms;
+    min_rtt_ms = std::min(rtt_ms, min_rtt_ms);
+    min_send_receive_diff_ms = std::min(y, min_send_receive_diff_ms);
+    time_series.points.emplace_back(x, y);
+  }
+
+  // We assume that the base network delay (w/o queues) is equal to half
+  // the minimum RTT. Therefore rescale the delays by subtracting the minimum
+  // observed 1-ways delay and add half the minimum RTT.
+  const int64_t estimated_clock_offset_ms =
+      min_send_receive_diff_ms - min_rtt_ms / 2;
+  for (TimeSeriesPoint& point : time_series.points)
+    point.y -= estimated_clock_offset_ms;
+
+  // Add the data set to the plot.
+  plot->AppendTimeSeriesIfNotEmpty(std::move(time_series));
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 10, "Delay (ms)", kBottomMargin, kTopMargin);
+  plot->SetTitle("Outgoing network delay (based on per-packet feedback)");
+}
+
+void CreatePacerDelayGraph(const ParsedRtcEventLog& parsed_log,
+                           const AnalyzerConfig& config,
+                           Plot* plot) {
+  for (const auto& stream : parsed_log.outgoing_rtp_packets_by_ssrc()) {
+    const std::vector<LoggedRtpPacketOutgoing>& packets =
+        stream.outgoing_packets;
+
+    if (IsRtxSsrc(parsed_log, kOutgoingPacket, stream.ssrc)) {
+      continue;
+    }
+
+    if (packets.size() < 2) {
+      RTC_LOG(LS_WARNING)
+          << "Can't estimate a the RTP clock frequency or the "
+             "pacer delay with less than 2 packets in the stream";
+      continue;
+    }
+    int64_t segment_end_us = parsed_log.first_log_segment().stop_time_us();
+    std::optional<uint32_t> estimated_frequency =
+        EstimateRtpClockFrequency(packets, segment_end_us);
+    if (!estimated_frequency)
+      continue;
+    if (IsVideoSsrc(parsed_log, kOutgoingPacket, stream.ssrc) &&
+        *estimated_frequency != 90000) {
+      RTC_LOG(LS_WARNING)
+          << "Video stream should use a 90 kHz clock but appears to use "
+          << *estimated_frequency / 1000 << ". Discarding.";
+      continue;
+    }
+
+    TimeSeries pacer_delay_series(
+        GetStreamName(parsed_log, kOutgoingPacket, stream.ssrc) + "(" +
+            std::to_string(*estimated_frequency / 1000) + " kHz)",
+        LineStyle::kLine, PointStyle::kHighlight);
+    SeqNumUnwrapper<uint32_t> timestamp_unwrapper;
+    uint64_t first_capture_timestamp =
+        timestamp_unwrapper.Unwrap(packets.front().rtp.header.timestamp);
+    uint64_t first_send_timestamp = packets.front().rtp.log_time_us();
+    for (const auto& packet : packets) {
+      double capture_time_ms = (static_cast<double>(timestamp_unwrapper.Unwrap(
+                                    packet.rtp.header.timestamp)) -
+                                first_capture_timestamp) /
+                               *estimated_frequency * 1000;
+      double send_time_ms =
+          static_cast<double>(packet.rtp.log_time_us() - first_send_timestamp) /
+          1000;
+      float x = config.GetCallTimeSec(packet.rtp.log_time());
+      float y = send_time_ms - capture_time_ms;
+      pacer_delay_series.points.emplace_back(x, y);
+    }
+    plot->AppendTimeSeries(std::move(pacer_delay_series));
+  }
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 10, "Pacer delay (ms)", kBottomMargin, kTopMargin);
+  plot->SetTitle(
+      "Delay from capture to send time. (First packet normalized to 0.)");
+}
+
+}  // namespace webrtc
diff --git a/rtc_tools/rtc_event_log_visualizer/analyze_bwe.h b/rtc_tools/rtc_event_log_visualizer/analyze_bwe.h
new file mode 100644
index 0000000000..fd4b041c78
--- /dev/null
+++ b/rtc_tools/rtc_event_log_visualizer/analyze_bwe.h
@@ -0,0 +1,81 @@
+/*
+ *  Copyright (c) 2026 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZE_BWE_H_
+#define RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZE_BWE_H_
+
+#include "logging/rtc_event_log/rtc_event_log_parser.h"
+#include "rtc_tools/rtc_event_log_visualizer/analyzer_common.h"
+#include "rtc_tools/rtc_event_log_visualizer/plot_base.h"
+
+namespace webrtc {
+
+void CreateIncomingDelayGraph(const ParsedRtcEventLog& parsed_log,
+                              const AnalyzerConfig& config,
+                              Plot* plot);
+
+void CreateFractionLossGraph(const ParsedRtcEventLog& parsed_log,
+                             const AnalyzerConfig& config,
+                             Plot* plot);
+
+void CreateTotalIncomingBitrateGraph(const ParsedRtcEventLog& parsed_log,
+                                     const AnalyzerConfig& config,
+                                     Plot* plot);
+
+void CreateTotalOutgoingBitrateGraph(const ParsedRtcEventLog& parsed_log,
+                                     const AnalyzerConfig& config,
+                                     Plot* plot,
+                                     bool show_detector_state = false,
+                                     bool show_alr_state = false,
+                                     bool show_link_capacity = false);
+
+void CreateGoogCcSimulationGraph(const ParsedRtcEventLog& parsed_log,
+                                 const AnalyzerConfig& config,
+                                 Plot* plot);
+
+void CreateScreamSimulationBitrateGraph(const ParsedRtcEventLog& parsed_log,
+                                        const AnalyzerConfig& config,
+                                        Plot* plot);
+
+void CreateScreamSimulationRefWindowGraph(const ParsedRtcEventLog& parsed_log,
+                                          const AnalyzerConfig& config,
+                                          Plot* plot);
+
+void CreateScreamSimulationRatiosGraph(const ParsedRtcEventLog& parsed_log,
+                                       const AnalyzerConfig& config,
+                                       Plot* plot);
+
+void CreateScreamRefWindowGraph(const ParsedRtcEventLog& parsed_log,
+                                const AnalyzerConfig& config,
+                                Plot* plot);
+
+void CreateScreamDelayEstimateGraph(const ParsedRtcEventLog& parsed_log,
+                                    const AnalyzerConfig& config,
+                                    Plot* plot);
+
+void CreateSendSideBweSimulationGraph(const ParsedRtcEventLog& parsed_log,
+                                      const AnalyzerConfig& config,
+                                      Plot* plot);
+
+void CreateReceiveSideBweSimulationGraph(const ParsedRtcEventLog& parsed_log,
+                                         const AnalyzerConfig& config,
+                                         Plot* plot);
+
+void CreateNetworkDelayFeedbackGraph(const ParsedRtcEventLog& parsed_log,
+                                     const AnalyzerConfig& config,
+                                     Plot* plot);
+
+void CreatePacerDelayGraph(const ParsedRtcEventLog& parsed_log,
+                           const AnalyzerConfig& config,
+                           Plot* plot);
+
+}  // namespace webrtc
+
+#endif  // RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZE_BWE_H_
diff --git a/rtc_tools/rtc_event_log_visualizer/analyze_connectivity.cc b/rtc_tools/rtc_event_log_visualizer/analyze_connectivity.cc
new file mode 100644
index 0000000000..b87a108b96
--- /dev/null
+++ b/rtc_tools/rtc_event_log_visualizer/analyze_connectivity.cc
@@ -0,0 +1,285 @@
+/*
+ *  Copyright (c) 2026 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "rtc_tools/rtc_event_log_visualizer/analyze_connectivity.h"
+
+#include <cstddef>
+#include <cstdint>
+#include <map>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/candidate.h"
+#include "api/dtls_transport_interface.h"
+#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair.h"
+#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.h"
+#include "logging/rtc_event_log/rtc_event_log_parser.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_tools/rtc_event_log_visualizer/analyzer_common.h"
+#include "rtc_tools/rtc_event_log_visualizer/plot_base.h"
+
+namespace webrtc {
+
+namespace {
+
+const char kUnknownEnumValue[] = "unknown";
+
+// TODO(tommi): This should be "host".
+const char kIceCandidateTypeLocal[] = "local";
+// TODO(tommi): This should be "srflx".
+const char kIceCandidateTypeStun[] = "stun";
+const char kIceCandidateTypePrflx[] = "prflx";
+const char kIceCandidateTypeRelay[] = "relay";
+
+const char kProtocolUdp[] = "udp";
+const char kProtocolTcp[] = "tcp";
+const char kProtocolSsltcp[] = "ssltcp";
+const char kProtocolTls[] = "tls";
+
+const char kAddressFamilyIpv4[] = "ipv4";
+const char kAddressFamilyIpv6[] = "ipv6";
+
+const char kNetworkTypeEthernet[] = "ethernet";
+const char kNetworkTypeLoopback[] = "loopback";
+const char kNetworkTypeWifi[] = "wifi";
+const char kNetworkTypeVpn[] = "vpn";
+const char kNetworkTypeCellular[] = "cellular";
+
+absl::string_view GetIceCandidateTypeAsString(IceCandidateType type) {
+  switch (type) {
+    case IceCandidateType::kHost:
+      return kIceCandidateTypeLocal;
+    case IceCandidateType::kSrflx:
+      return kIceCandidateTypeStun;
+    case IceCandidateType::kPrflx:
+      return kIceCandidateTypePrflx;
+    case IceCandidateType::kRelay:
+      return kIceCandidateTypeRelay;
+    default:
+      RTC_DCHECK_NOTREACHED();
+      return kUnknownEnumValue;
+  }
+}
+
+std::string GetProtocolAsString(IceCandidatePairProtocol protocol) {
+  switch (protocol) {
+    case IceCandidatePairProtocol::kUdp:
+      return kProtocolUdp;
+    case IceCandidatePairProtocol::kTcp:
+      return kProtocolTcp;
+    case IceCandidatePairProtocol::kSsltcp:
+      return kProtocolSsltcp;
+    case IceCandidatePairProtocol::kTls:
+      return kProtocolTls;
+    default:
+      return kUnknownEnumValue;
+  }
+}
+
+std::string GetAddressFamilyAsString(IceCandidatePairAddressFamily family) {
+  switch (family) {
+    case IceCandidatePairAddressFamily::kIpv4:
+      return kAddressFamilyIpv4;
+    case IceCandidatePairAddressFamily::kIpv6:
+      return kAddressFamilyIpv6;
+    default:
+      return kUnknownEnumValue;
+  }
+}
+
+std::string GetNetworkTypeAsString(IceCandidateNetworkType type) {
+  switch (type) {
+    case IceCandidateNetworkType::kEthernet:
+      return kNetworkTypeEthernet;
+    case IceCandidateNetworkType::kLoopback:
+      return kNetworkTypeLoopback;
+    case IceCandidateNetworkType::kWifi:
+      return kNetworkTypeWifi;
+    case IceCandidateNetworkType::kVpn:
+      return kNetworkTypeVpn;
+    case IceCandidateNetworkType::kCellular:
+      return kNetworkTypeCellular;
+    default:
+      return kUnknownEnumValue;
+  }
+}
+
+std::string GetCandidatePairLogDescriptionAsString(
+    const LoggedIceCandidatePairConfig& config) {
+  // Example: stun:wifi->relay(tcp):cellular@udp:ipv4
+  // represents a pair of a local server-reflexive candidate on a WiFi network
+  // and a remote relay candidate using TCP as the relay protocol on a cell
+  // network, when the candidate pair communicates over UDP using IPv4.
+  StringBuilder ss;
+  ss << GetIceCandidateTypeAsString(config.local_candidate_type);
+
+  if (config.local_candidate_type == IceCandidateType::kRelay) {
+    ss << "(" << GetProtocolAsString(config.local_relay_protocol) << ")";
+  }
+
+  ss << ":" << GetNetworkTypeAsString(config.local_network_type) << ":"
+     << GetAddressFamilyAsString(config.local_address_family) << "->"
+     << GetIceCandidateTypeAsString(config.remote_candidate_type) << ":"
+     << GetAddressFamilyAsString(config.remote_address_family) << "@"
+     << GetProtocolAsString(config.candidate_pair_protocol);
+  return ss.Release();
+}
+
+std::map<uint32_t, std::string> BuildCandidateIdLogDescriptionMap(
+    const std::vector<LoggedIceCandidatePairConfig>&
+        ice_candidate_pair_configs) {
+  std::map<uint32_t, std::string> candidate_pair_desc_by_id;
+  for (const auto& config : ice_candidate_pair_configs) {
+    // TODO(qingsi): Add the handling of the "Updated" config event after the
+    // visualization of property change for candidate pairs is introduced.
+    if (candidate_pair_desc_by_id.find(config.candidate_pair_id) ==
+        candidate_pair_desc_by_id.end()) {
+      const std::string candidate_pair_desc =
+          GetCandidatePairLogDescriptionAsString(config);
+      candidate_pair_desc_by_id[config.candidate_pair_id] = candidate_pair_desc;
+    }
+  }
+  return candidate_pair_desc_by_id;
+}
+
+}  // namespace
+
+void CreateIceCandidatePairConfigGraph(const ParsedRtcEventLog& parsed_log,
+                                       const AnalyzerConfig& config,
+                                       Plot* plot) {
+  std::map<uint32_t, TimeSeries> configs_by_cp_id;
+  for (const auto& config_item : parsed_log.ice_candidate_pair_configs()) {
+    if (configs_by_cp_id.find(config_item.candidate_pair_id) ==
+        configs_by_cp_id.end()) {
+      const std::string candidate_pair_desc =
+          GetCandidatePairLogDescriptionAsString(config_item);
+      configs_by_cp_id[config_item.candidate_pair_id] =
+          TimeSeries("[" + std::to_string(config_item.candidate_pair_id) + "]" +
+                         candidate_pair_desc,
+                     LineStyle::kNone, PointStyle::kHighlight);
+    }
+    float x = config.GetCallTimeSec(config_item.log_time());
+    float y = static_cast<float>(config_item.type);
+    configs_by_cp_id[config_item.candidate_pair_id].points.emplace_back(x, y);
+  }
+
+  // TODO(qingsi): There can be a large number of candidate pairs generated by
+  // certain calls and the frontend cannot render the chart in this case due
+  // to the failure of generating a palette with the same number of colors.
+  for (auto& kv : configs_by_cp_id) {
+    plot->AppendTimeSeries(std::move(kv.second));
+  }
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 3, "Config Type", kBottomMargin, kTopMargin);
+  plot->SetTitle("[IceEventLog] ICE candidate pair configs");
+  plot->SetYAxisTickLabels(
+      {{static_cast<float>(IceCandidatePairConfigType::kAdded), "ADDED"},
+       {static_cast<float>(IceCandidatePairConfigType::kUpdated), "UPDATED"},
+       {static_cast<float>(IceCandidatePairConfigType::kDestroyed),
+        "DESTROYED"},
+       {static_cast<float>(IceCandidatePairConfigType::kSelected),
+        "SELECTED"}});
+}
+
+void CreateIceConnectivityCheckGraph(const ParsedRtcEventLog& parsed_log,
+                                     const AnalyzerConfig& config,
+                                     Plot* plot) {
+  constexpr int kEventTypeOffset =
+      static_cast<int>(IceCandidatePairConfigType::kNumValues);
+  std::map<uint32_t, TimeSeries> checks_by_cp_id;
+  std::map<uint32_t, std::string> candidate_pair_desc_by_id =
+      BuildCandidateIdLogDescriptionMap(
+          parsed_log.ice_candidate_pair_configs());
+  for (const auto& event : parsed_log.ice_candidate_pair_events()) {
+    if (checks_by_cp_id.find(event.candidate_pair_id) ==
+        checks_by_cp_id.end()) {
+      checks_by_cp_id[event.candidate_pair_id] =
+          TimeSeries("[" + std::to_string(event.candidate_pair_id) + "]" +
+                         candidate_pair_desc_by_id[event.candidate_pair_id],
+                     LineStyle::kNone, PointStyle::kHighlight);
+    }
+    float x = config.GetCallTimeSec(event.log_time());
+    float y = static_cast<float>(event.type) + kEventTypeOffset;
+    checks_by_cp_id[event.candidate_pair_id].points.emplace_back(x, y);
+  }
+
+  // TODO(qingsi): The same issue as in CreateIceCandidatePairConfigGraph.
+  for (auto& kv : checks_by_cp_id) {
+    plot->AppendTimeSeries(std::move(kv.second));
+  }
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 4, "Connectivity State", kBottomMargin,
+                          kTopMargin);
+  plot->SetTitle("[IceEventLog] ICE connectivity checks");
+
+  plot->SetYAxisTickLabels(
+      {{static_cast<float>(IceCandidatePairEventType::kCheckSent) +
+            kEventTypeOffset,
+        "CHECK SENT"},
+       {static_cast<float>(IceCandidatePairEventType::kCheckReceived) +
+            kEventTypeOffset,
+        "CHECK RECEIVED"},
+       {static_cast<float>(IceCandidatePairEventType::kCheckResponseSent) +
+            kEventTypeOffset,
+        "RESPONSE SENT"},
+       {static_cast<float>(IceCandidatePairEventType::kCheckResponseReceived) +
+            kEventTypeOffset,
+        "RESPONSE RECEIVED"}});
+}
+
+void CreateDtlsTransportStateGraph(const ParsedRtcEventLog& parsed_log,
+                                   const AnalyzerConfig& config,
+                                   Plot* plot) {
+  TimeSeries states("DTLS Transport State", LineStyle::kNone,
+                    PointStyle::kHighlight);
+  for (const auto& event : parsed_log.dtls_transport_states()) {
+    float x = config.GetCallTimeSec(event.log_time());
+    float y = static_cast<float>(event.dtls_transport_state);
+    states.points.emplace_back(x, y);
+  }
+  plot->AppendTimeSeries(std::move(states));
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, static_cast<float>(DtlsTransportState::kNumValues),
+                          "Transport State", kBottomMargin, kTopMargin);
+  plot->SetTitle("DTLS Transport State");
+  plot->SetYAxisTickLabels(
+      {{static_cast<float>(DtlsTransportState::kNew), "NEW"},
+       {static_cast<float>(DtlsTransportState::kConnecting), "CONNECTING"},
+       {static_cast<float>(DtlsTransportState::kConnected), "CONNECTED"},
+       {static_cast<float>(DtlsTransportState::kClosed), "CLOSED"},
+       {static_cast<float>(DtlsTransportState::kFailed), "FAILED"}});
+}
+
+void CreateDtlsWritableStateGraph(const ParsedRtcEventLog& parsed_log,
+                                  const AnalyzerConfig& config,
+                                  Plot* plot) {
+  TimeSeries writable("DTLS Writable", LineStyle::kNone,
+                      PointStyle::kHighlight);
+  for (const auto& event : parsed_log.dtls_writable_states()) {
+    float x = config.GetCallTimeSec(event.log_time());
+    float y = static_cast<float>(event.writable);
+    writable.points.emplace_back(x, y);
+  }
+  plot->AppendTimeSeries(std::move(writable));
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Writable", kBottomMargin, kTopMargin);
+  plot->SetTitle("DTLS Writable State");
+}
+
+}  // namespace webrtc
diff --git a/rtc_tools/rtc_event_log_visualizer/analyze_connectivity.h b/rtc_tools/rtc_event_log_visualizer/analyze_connectivity.h
new file mode 100644
index 0000000000..1eac2f0c52
--- /dev/null
+++ b/rtc_tools/rtc_event_log_visualizer/analyze_connectivity.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2026 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZE_CONNECTIVITY_H_
+#define RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZE_CONNECTIVITY_H_
+
+#include "logging/rtc_event_log/rtc_event_log_parser.h"
+#include "rtc_tools/rtc_event_log_visualizer/analyzer_common.h"
+#include "rtc_tools/rtc_event_log_visualizer/plot_base.h"
+
+namespace webrtc {
+
+void CreateIceCandidatePairConfigGraph(const ParsedRtcEventLog& parsed_log,
+                                       const AnalyzerConfig& config,
+                                       Plot* plot);
+
+void CreateIceConnectivityCheckGraph(const ParsedRtcEventLog& parsed_log,
+                                     const AnalyzerConfig& config,
+                                     Plot* plot);
+
+void CreateDtlsTransportStateGraph(const ParsedRtcEventLog& parsed_log,
+                                   const AnalyzerConfig& config,
+                                   Plot* plot);
+
+void CreateDtlsWritableStateGraph(const ParsedRtcEventLog& parsed_log,
+                                  const AnalyzerConfig& config,
+                                  Plot* plot);
+
+}  // namespace webrtc
+
+#endif  // RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZE_CONNECTIVITY_H_
diff --git a/rtc_tools/rtc_event_log_visualizer/analyze_rtp_rtcp.cc b/rtc_tools/rtc_event_log_visualizer/analyze_rtp_rtcp.cc
new file mode 100644
index 0000000000..9d6c0668e8
--- /dev/null
+++ b/rtc_tools/rtc_event_log_visualizer/analyze_rtp_rtcp.cc
@@ -0,0 +1,809 @@
+/*
+ *  Copyright (c) 2026 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "rtc_tools/rtc_event_log_visualizer/analyze_rtp_rtcp.h"
+
+#include <algorithm>
+#include <cstddef>
+#include <cstdint>
+#include <deque>
+#include <map>
+#include <optional>
+#include <string>
+#include <tuple>
+#include <unordered_set>
+#include <utility>
+#include <vector>
+
+#include "api/function_view.h"
+#include "api/rtp_headers.h"
+#include "api/transport/ecn_marking.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "logging/rtc_event_log/events/logged_rtp_rtcp.h"
+#include "logging/rtc_event_log/rtc_event_log_parser.h"
+#include "logging/rtc_event_log/rtc_event_processor.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/congestion_control_feedback.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/target_bitrate.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
+#include "rtc_tools/rtc_event_log_visualizer/analyzer_common.h"
+#include "rtc_tools/rtc_event_log_visualizer/plot_base.h"
+
+namespace webrtc {
+
+namespace {
+
+template <typename T>
+TimeSeries CreateRtcpTypeTimeSeries(const std::vector<T>& rtcp_list,
+                                    AnalyzerConfig config,
+                                    std::string rtcp_name,
+                                    int category_id) {
+  TimeSeries time_series(rtcp_name, LineStyle::kNone, PointStyle::kHighlight);
+  for (const auto& rtcp : rtcp_list) {
+    float x = config.GetCallTimeSec(rtcp.timestamp);
+    float y = category_id;
+    time_series.points.emplace_back(x, y);
+  }
+  return time_series;
+}
+
+struct PacketLossSummary {
+  size_t num_packets = 0;
+  size_t num_lost_packets = 0;
+  Timestamp base_time = Timestamp::MinusInfinity();
+};
+
+}  // namespace
+
+float GetHighestSeqNumber(const rtcp::ReportBlock& block) {
+  return block.extended_high_seq_num();
+}
+
+float GetFractionLost(const rtcp::ReportBlock& block) {
+  return static_cast<double>(block.fraction_lost()) / 256 * 100;
+}
+
+float GetCumulativeLost(const rtcp::ReportBlock& block) {
+  return block.cumulative_lost();
+}
+
+float DelaySinceLastSr(const rtcp::ReportBlock& block) {
+  return static_cast<double>(block.delay_since_last_sr()) / 65536;
+}
+
+void CreatePacketGraph(PacketDirection direction,
+                       const ParsedRtcEventLog& parsed_log,
+                       const AnalyzerConfig& config,
+                       Plot* plot) {
+  for (const auto& stream : parsed_log.rtp_packets_by_ssrc(direction)) {
+    // Filter on SSRC.
+    if (!MatchingSsrc(stream.ssrc, config.desired_ssrc_)) {
+      continue;
+    }
+
+    TimeSeries time_series(GetStreamName(parsed_log, direction, stream.ssrc),
+                           LineStyle::kBar);
+    auto GetPacketSize = [](const LoggedRtpPacket& packet) {
+      return std::optional<float>(packet.total_length);
+    };
+    auto ToCallTime = [&config](const LoggedRtpPacket& packet) {
+      return config.GetCallTimeSec(packet.timestamp);
+    };
+    ProcessPoints<LoggedRtpPacket>(ToCallTime, GetPacketSize,
+                                   stream.packet_view, &time_series);
+    plot->AppendTimeSeries(std::move(time_series));
+  }
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Packet size (bytes)", kBottomMargin,
+                          kTopMargin);
+  plot->SetTitle(GetDirectionAsString(direction) + " RTP packets");
+}
+
+void CreateRtcpTypeGraph(PacketDirection direction,
+                         const ParsedRtcEventLog& parsed_log,
+                         const AnalyzerConfig& config,
+                         Plot* plot) {
+  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(
+      parsed_log.transport_feedbacks(direction), config, "TWCC", 1));
+  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(
+      parsed_log.congestion_feedback(direction), config, "CCFB", 2));
+  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(
+      parsed_log.receiver_reports(direction), config, "RR", 3));
+  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(
+      parsed_log.sender_reports(direction), config, "SR", 4));
+  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(
+      parsed_log.extended_reports(direction), config, "XR", 5));
+  plot->AppendTimeSeries(
+      CreateRtcpTypeTimeSeries(parsed_log.nacks(direction), config, "NACK", 6));
+  plot->AppendTimeSeries(
+      CreateRtcpTypeTimeSeries(parsed_log.rembs(direction), config, "REMB", 7));
+  plot->AppendTimeSeries(
+      CreateRtcpTypeTimeSeries(parsed_log.firs(direction), config, "FIR", 8));
+  plot->AppendTimeSeries(
+      CreateRtcpTypeTimeSeries(parsed_log.plis(direction), config, "PLI", 9));
+  plot->AppendTimeSeries(
+      CreateRtcpTypeTimeSeries(parsed_log.byes(direction), config, "BYE", 10));
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "RTCP type", kBottomMargin, kTopMargin);
+  plot->SetTitle(GetDirectionAsString(direction) + " RTCP packets");
+  plot->SetYAxisTickLabels({{1, "TWCC"},
+                            {2, "CCFB"},
+                            {3, "RR"},
+                            {4, "SR"},
+                            {5, "XR"},
+                            {6, "NACK"},
+                            {7, "REMB"},
+                            {8, "FIR"},
+                            {9, "PLI"},
+                            {10, "BYE"}});
+}
+
+template <typename IterableType>
+void CreateAccumulatedPacketsTimeSeries(Plot* plot,
+                                        const AnalyzerConfig& config,
+                                        const IterableType& packets,
+                                        const std::string& label) {
+  TimeSeries time_series(label, LineStyle::kStep);
+  for (size_t i = 0; i < packets.size(); i++) {
+    float x = config.GetCallTimeSec(packets[i].log_time());
+    time_series.points.emplace_back(x, i + 1);
+  }
+  plot->AppendTimeSeries(std::move(time_series));
+}
+
+void CreateAccumulatedPacketsGraph(PacketDirection direction,
+                                   const ParsedRtcEventLog& parsed_log,
+                                   const AnalyzerConfig& config,
+                                   Plot* plot) {
+  for (const auto& stream : parsed_log.rtp_packets_by_ssrc(direction)) {
+    if (!MatchingSsrc(stream.ssrc, config.desired_ssrc_))
+      continue;
+    std::string label =
+        std::string("RTP ") + GetStreamName(parsed_log, direction, stream.ssrc);
+    CreateAccumulatedPacketsTimeSeries(plot, config, stream.packet_view, label);
+  }
+  std::string label =
+      std::string("RTCP ") + "(" + GetDirectionAsShortString(direction) + ")";
+  if (direction == kIncomingPacket) {
+    CreateAccumulatedPacketsTimeSeries(
+        plot, config, parsed_log.incoming_rtcp_packets(), label);
+  } else {
+    CreateAccumulatedPacketsTimeSeries(
+        plot, config, parsed_log.outgoing_rtcp_packets(), label);
+  }
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Received Packets", kBottomMargin, kTopMargin);
+  plot->SetTitle(std::string("Accumulated ") + GetDirectionAsString(direction) +
+                 " RTP/RTCP packets");
+}
+
+void CreatePacketRateGraph(PacketDirection direction,
+                           const ParsedRtcEventLog& parsed_log,
+                           const AnalyzerConfig& config,
+                           Plot* plot) {
+  auto CountPackets = [](auto packet) { return 1.0; };
+  for (const auto& stream : parsed_log.rtp_packets_by_ssrc(direction)) {
+    // Filter on SSRC.
+    if (!MatchingSsrc(stream.ssrc, config.desired_ssrc_)) {
+      continue;
+    }
+    TimeSeries time_series(
+        std::string("RTP ") + GetStreamName(parsed_log, direction, stream.ssrc),
+        LineStyle::kLine);
+    MovingAverage<LoggedRtpPacket, double>(CountPackets, stream.packet_view,
+                                           config, &time_series);
+    plot->AppendTimeSeries(std::move(time_series));
+  }
+  TimeSeries time_series(
+      std::string("RTCP ") + "(" + GetDirectionAsShortString(direction) + ")",
+      LineStyle::kLine);
+  if (direction == kIncomingPacket) {
+    MovingAverage<LoggedRtcpPacketIncoming, double>(
+        CountPackets, parsed_log.incoming_rtcp_packets(), config, &time_series);
+  } else {
+    MovingAverage<LoggedRtcpPacketOutgoing, double>(
+        CountPackets, parsed_log.outgoing_rtcp_packets(), config, &time_series);
+  }
+  plot->AppendTimeSeries(std::move(time_series));
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Packet Rate (packets/s)", kBottomMargin,
+                          kTopMargin);
+  plot->SetTitle("Rate of " + GetDirectionAsString(direction) +
+                 " RTP/RTCP packets");
+}
+
+void CreateTotalPacketRateGraph(PacketDirection direction,
+                                const ParsedRtcEventLog& parsed_log,
+                                const AnalyzerConfig& config,
+                                Plot* plot) {
+  // Contains a log timestamp to enable counting logged events of different
+  // types using MovingAverage().
+  class LogTime {
+   public:
+    explicit LogTime(Timestamp log_time) : log_time_(log_time) {}
+    Timestamp log_time() const { return log_time_; }
+
+   private:
+    Timestamp log_time_;
+  };
+  std::vector<LogTime> packet_times;
+  auto handle_rtp = [&packet_times](const LoggedRtpPacket& packet) {
+    packet_times.emplace_back(packet.log_time());
+  };
+  RtcEventProcessor process;
+  for (const auto& stream : parsed_log.rtp_packets_by_ssrc(direction)) {
+    process.AddEvents(stream.packet_view, handle_rtp, direction);
+  }
+  if (direction == kIncomingPacket) {
+    auto handle_incoming_rtcp =
+        [&packet_times](const LoggedRtcpPacketIncoming& packet) {
+          packet_times.emplace_back(packet.log_time());
+        };
+    process.AddEvents(parsed_log.incoming_rtcp_packets(), handle_incoming_rtcp);
+  } else {
+    auto handle_outgoing_rtcp =
+        [&packet_times](const LoggedRtcpPacketOutgoing& packet) {
+          packet_times.emplace_back(packet.log_time());
+        };
+    process.AddEvents(parsed_log.outgoing_rtcp_packets(), handle_outgoing_rtcp);
+  }
+  process.ProcessEventsInOrder();
+  TimeSeries time_series(std::string("Total ") + "(" +
+                             GetDirectionAsShortString(direction) + ") packets",
+                         LineStyle::kLine);
+  MovingAverage<LogTime, uint64_t>([](auto packet) { return 1; }, packet_times,
+                                   config, &time_series);
+  plot->AppendTimeSeries(std::move(time_series));
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Packet Rate (packets/s)", kBottomMargin,
+                          kTopMargin);
+  plot->SetTitle("Rate of all " + GetDirectionAsString(direction) +
+                 " RTP/RTCP packets");
+}
+
+// For each SSRC, plot the sequence number difference between consecutive
+// incoming packets.
+void CreateSequenceNumberGraph(const ParsedRtcEventLog& parsed_log,
+                               const AnalyzerConfig& config,
+                               Plot* plot) {
+  for (const auto& stream : parsed_log.incoming_rtp_packets_by_ssrc()) {
+    // Filter on SSRC.
+    if (!MatchingSsrc(stream.ssrc, config.desired_ssrc_)) {
+      continue;
+    }
+
+    TimeSeries time_series(
+        GetStreamName(parsed_log, kIncomingPacket, stream.ssrc),
+        LineStyle::kBar);
+    auto GetSequenceNumberDiff = [](const LoggedRtpPacketIncoming& old_packet,
+                                    const LoggedRtpPacketIncoming& new_packet) {
+      int64_t diff =
+          WrappingDifference(new_packet.rtp.header.sequenceNumber,
+                             old_packet.rtp.header.sequenceNumber, 1ul << 16);
+      return diff;
+    };
+    auto ToCallTime = [&config](const LoggedRtpPacketIncoming& packet) {
+      return config.GetCallTimeSec(packet.log_time());
+    };
+    ProcessPairs<LoggedRtpPacketIncoming, float>(
+        ToCallTime, GetSequenceNumberDiff, stream.incoming_packets,
+        &time_series);
+    plot->AppendTimeSeries(std::move(time_series));
+  }
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Difference since last packet", kBottomMargin,
+                          kTopMargin);
+  plot->SetTitle("Incoming sequence number delta");
+}
+
+void CreateIncomingPacketLossGraph(const ParsedRtcEventLog& parsed_log,
+                                   const AnalyzerConfig& config,
+                                   Plot* plot) {
+  for (const auto& stream : parsed_log.incoming_rtp_packets_by_ssrc()) {
+    const std::vector<LoggedRtpPacketIncoming>& packets =
+        stream.incoming_packets;
+    // Filter on SSRC.
+    if (!MatchingSsrc(stream.ssrc, config.desired_ssrc_) || packets.empty()) {
+      continue;
+    }
+
+    TimeSeries time_series(
+        GetStreamName(parsed_log, kIncomingPacket, stream.ssrc),
+        LineStyle::kLine, PointStyle::kHighlight);
+    // TODO(terelius): Should the window and step size be read from the class
+    // instead?
+    const TimeDelta kWindow = TimeDelta::Millis(1000);
+    const TimeDelta kStep = TimeDelta::Millis(1000);
+    SeqNumUnwrapper<uint16_t> unwrapper_;
+    SeqNumUnwrapper<uint16_t> prior_unwrapper_;
+    size_t window_index_begin = 0;
+    size_t window_index_end = 0;
+    uint64_t highest_seq_number =
+        unwrapper_.Unwrap(packets[0].rtp.header.sequenceNumber) - 1;
+    uint64_t highest_prior_seq_number =
+        prior_unwrapper_.Unwrap(packets[0].rtp.header.sequenceNumber) - 1;
+
+    for (Timestamp t = config.begin_time_; t < config.end_time_ + kStep;
+         t += kStep) {
+      while (window_index_end < packets.size() &&
+             packets[window_index_end].rtp.log_time() < t) {
+        uint64_t sequence_number = unwrapper_.Unwrap(
+            packets[window_index_end].rtp.header.sequenceNumber);
+        highest_seq_number = std::max(highest_seq_number, sequence_number);
+        ++window_index_end;
+      }
+      while (window_index_begin < packets.size() &&
+             packets[window_index_begin].rtp.log_time() < t - kWindow) {
+        uint64_t sequence_number = prior_unwrapper_.Unwrap(
+            packets[window_index_begin].rtp.header.sequenceNumber);
+        highest_prior_seq_number =
+            std::max(highest_prior_seq_number, sequence_number);
+        ++window_index_begin;
+      }
+      float x = config.GetCallTimeSec(t);
+      uint64_t expected_packets = highest_seq_number - highest_prior_seq_number;
+      if (expected_packets > 0) {
+        int64_t received_packets = window_index_end - window_index_begin;
+        int64_t lost_packets = expected_packets - received_packets;
+        float y = static_cast<float>(lost_packets) / expected_packets * 100;
+        time_series.points.emplace_back(x, y);
+      }
+    }
+    plot->AppendTimeSeries(std::move(time_series));
+  }
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Loss rate (in %)", kBottomMargin, kTopMargin);
+  plot->SetTitle("Incoming packet loss (derived from incoming packets)");
+}
+
+// For each SSRC, plot the bandwidth used by that stream.
+void CreateStreamBitrateGraph(PacketDirection direction,
+                              const ParsedRtcEventLog& parsed_log,
+                              const AnalyzerConfig& config,
+                              Plot* plot) {
+  for (const auto& stream : parsed_log.rtp_packets_by_ssrc(direction)) {
+    // Filter on SSRC.
+    if (!MatchingSsrc(stream.ssrc, config.desired_ssrc_)) {
+      continue;
+    }
+
+    TimeSeries time_series(GetStreamName(parsed_log, direction, stream.ssrc),
+                           LineStyle::kLine);
+    auto GetPacketSizeKilobits = [](const LoggedRtpPacket& packet) {
+      return packet.total_length * 8.0 / 1000.0;
+    };
+    MovingAverage<LoggedRtpPacket, double>(
+        GetPacketSizeKilobits, stream.packet_view, config, &time_series);
+    plot->AppendTimeSeries(std::move(time_series));
+  }
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin);
+  plot->SetTitle(GetDirectionAsString(direction) + " bitrate per stream");
+}
+
+// Plot the bitrate allocation for each temporal and spatial layer.
+// Computed from RTCP XR target bitrate block, so the graph is only populated if
+// those are sent.
+void CreateBitrateAllocationGraph(PacketDirection direction,
+                                  const ParsedRtcEventLog& parsed_log,
+                                  const AnalyzerConfig& config,
+                                  Plot* plot) {
+  std::map<LayerDescription, TimeSeries> time_series;
+  const auto& xr_list = parsed_log.extended_reports(direction);
+  for (const auto& rtcp : xr_list) {
+    const std::optional<rtcp::TargetBitrate>& target_bitrate =
+        rtcp.xr.target_bitrate();
+    if (!target_bitrate.has_value())
+      continue;
+    for (const auto& bitrate_item : target_bitrate->GetTargetBitrates()) {
+      LayerDescription layer(rtcp.xr.sender_ssrc(), bitrate_item.spatial_layer,
+                             bitrate_item.temporal_layer);
+      auto time_series_it = time_series.find(layer);
+      if (time_series_it == time_series.end()) {
+        std::string layer_name = GetLayerName(layer);
+        bool inserted;
+        std::tie(time_series_it, inserted) = time_series.insert(
+            std::make_pair(layer, TimeSeries(layer_name, LineStyle::kStep)));
+        RTC_DCHECK(inserted);
+      }
+      float x = config.GetCallTimeSec(rtcp.log_time());
+      float y = bitrate_item.target_bitrate_kbps;
+      time_series_it->second.points.emplace_back(x, y);
+    }
+  }
+  for (auto& layer : time_series) {
+    plot->AppendTimeSeries(std::move(layer.second));
+  }
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin);
+  if (direction == kIncomingPacket)
+    plot->SetTitle("Target bitrate per incoming layer");
+  else
+    plot->SetTitle("Target bitrate per outgoing layer");
+}
+
+void CreateEcnFeedbackGraph(Plot* plot,
+                            PacketDirection direction,
+                            const ParsedRtcEventLog& parsed_log,
+                            const AnalyzerConfig& config) {
+  TimeSeries not_ect("Not ECN capable", LineStyle::kBar,
+                     PointStyle::kHighlight);
+  TimeSeries ect_1("ECN capable", LineStyle::kBar, PointStyle::kHighlight);
+  TimeSeries ce("Congestion experienced", LineStyle::kBar,
+                PointStyle::kHighlight);
+
+  for (const LoggedRtcpCongestionControlFeedback& feedback :
+       parsed_log.congestion_feedback(direction)) {
+    int ect_1_count = 0;
+    int not_ect_count = 0;
+    int ce_count = 0;
+
+    for (const rtcp::CongestionControlFeedback::PacketInfo& info :
+         feedback.congestion_feedback.packets()) {
+      switch (info.ecn) {
+        case EcnMarking::kNotEct:
+          ++not_ect_count;
+          break;
+        case EcnMarking::kEct1:
+          ++ect_1_count;
+          break;
+        case EcnMarking::kEct0:
+          RTC_LOG(LS_ERROR) << "unexpected ect(0)";
+          break;
+        case EcnMarking::kCe:
+          ++ce_count;
+          break;
+      }
+    }
+    ect_1.points.emplace_back(config.GetCallTimeSec(feedback.timestamp),
+                              ect_1_count);
+    not_ect.points.emplace_back(config.GetCallTimeSec(feedback.timestamp),
+                                not_ect_count);
+    ce.points.emplace_back(config.GetCallTimeSec(feedback.timestamp), ce_count);
+  }
+
+  plot->AppendTimeSeriesIfNotEmpty(std::move(ect_1));
+  plot->AppendTimeSeriesIfNotEmpty(std::move(not_ect));
+  plot->AppendTimeSeriesIfNotEmpty(std::move(ce));
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 10, "Count per feedback", kBottomMargin,
+                          kTopMargin);
+}
+
+void CreateOutgoingEcnFeedbackGraph(const ParsedRtcEventLog& parsed_log,
+                                    const AnalyzerConfig& config,
+                                    Plot* plot) {
+  CreateEcnFeedbackGraph(plot, kOutgoingPacket, parsed_log, config);
+  plot->SetTitle("Outgoing ECN count per feedback");
+}
+
+void CreateIncomingEcnFeedbackGraph(const ParsedRtcEventLog& parsed_log,
+                                    const AnalyzerConfig& config,
+                                    Plot* plot) {
+  CreateEcnFeedbackGraph(plot, kIncomingPacket, parsed_log, config);
+  plot->SetTitle("Incoming ECN count per feedback");
+}
+
+void CreateOutgoingLossRateGraph(const ParsedRtcEventLog& parsed_log,
+                                 const AnalyzerConfig& config,
+                                 Plot* plot) {
+  struct PacketLossPerFeedback {
+    Timestamp timestamp;              // Time when this feedback was received.
+    int num_packets_in_feedback = 0;  // Includes lost packets.
+    int num_lost_packets = 0;         // In this specific feedback.
+    int num_reordered_packets = 0;    // Packets received in this feedback, but
+                                      // was previously reported as lost.
+    int num_missing_feedback =
+        0;  // Packets missing feedback between this report and the previous.
+  };
+
+  class LossFeedbackBuilder {
+   public:
+    void AddPacket(uint16_t sequence_number, TimeDelta arrival_time_delta) {
+      last_unwrapped_sequence_number_ =
+          sequence_number_unwrapper_.Unwrap(sequence_number);
+      if (!first_sequence_number_.has_value()) {
+        first_sequence_number_ = last_unwrapped_sequence_number_;
+      }
+      ++num_packets_;
+      if (arrival_time_delta.IsInfinite()) {
+        lost_sequence_numbers_.insert(last_unwrapped_sequence_number_);
+      } else {
+        num_reordered_packets_ += previous_lost_sequence_numbers_.count(
+            last_unwrapped_sequence_number_);
+      }
+    }
+
+    void Update(PacketLossPerFeedback& feedback) {
+      feedback.num_packets_in_feedback += num_packets_;
+      feedback.num_lost_packets += lost_sequence_numbers_.size();
+      feedback.num_reordered_packets += num_reordered_packets_;
+      if (first_sequence_number_.has_value() &&
+          previous_feedback_highest_seq_number_.has_value()) {
+        feedback.num_missing_feedback +=
+            *first_sequence_number_ - *previous_feedback_highest_seq_number_ -
+            1;
+      }
+
+      // Prepare for next feedback.
+      first_sequence_number_ = std::nullopt;
+      previous_lost_sequence_numbers_.insert(lost_sequence_numbers_.begin(),
+                                             lost_sequence_numbers_.end());
+      previous_feedback_highest_seq_number_ = last_unwrapped_sequence_number_;
+      lost_sequence_numbers_.clear();
+      num_reordered_packets_ = 0;
+      num_packets_ = 0;
+    }
+
+   private:
+    int64_t last_unwrapped_sequence_number_ = 0;
+    int num_reordered_packets_ = 0;
+    int num_packets_ = 0;
+    std::optional<int64_t> first_sequence_number_;
+
+    std::unordered_set<int64_t> lost_sequence_numbers_;
+    std::unordered_set<int64_t> previous_lost_sequence_numbers_;
+    std::optional<int64_t> previous_feedback_highest_seq_number_;
+
+    RtpSequenceNumberUnwrapper sequence_number_unwrapper_;
+  };
+
+  TimeSeries loss_rate_series("Loss rate (from packet feedback)",
+                              LineStyle::kLine, PointStyle::kHighlight);
+  TimeSeries reordered_packets_between_feedback(
+      "Ratio of reordered packets from last feedback", LineStyle::kLine,
+      PointStyle::kHighlight);
+  TimeSeries average_loss_rate_series("Average loss rate last 5s",
+                                      LineStyle::kLine, PointStyle::kHighlight);
+  TimeSeries missing_feedback_series("Missing feedback", LineStyle::kNone,
+                                     PointStyle::kHighlight);
+
+  std::vector<PacketLossPerFeedback> loss_per_feedback;
+
+  if (!parsed_log.congestion_feedback(kIncomingPacket).empty()) {
+    plot->SetTitle("Outgoing loss rate (from CCFB)");
+
+    std::map</*ssrc*/ uint32_t, LossFeedbackBuilder> per_ssrc_builder;
+    for (const LoggedRtcpCongestionControlFeedback& feedback :
+         parsed_log.congestion_feedback(kIncomingPacket)) {
+      const rtcp::CongestionControlFeedback& transport_feedback =
+          feedback.congestion_feedback;
+
+      PacketLossPerFeedback packet_loss_per_feedback = {
+          .timestamp = feedback.log_time()};
+      for (const rtcp::CongestionControlFeedback::PacketInfo& packet :
+           transport_feedback.packets()) {
+        per_ssrc_builder[packet.ssrc].AddPacket(packet.sequence_number,
+                                                packet.arrival_time_offset);
+      }
+      for (auto& [ssrc, builder] : per_ssrc_builder) {
+        builder.Update(packet_loss_per_feedback);
+      }
+      loss_per_feedback.push_back(packet_loss_per_feedback);
+    }
+  } else if (!parsed_log.transport_feedbacks(kIncomingPacket).empty()) {
+    plot->SetTitle("Outgoing loss rate (from TWCC)");
+
+    LossFeedbackBuilder builder;
+    for (const LoggedRtcpPacketTransportFeedback& feedback :
+         parsed_log.transport_feedbacks(kIncomingPacket)) {
+      feedback.transport_feedback.ForAllPackets(
+          [&](uint16_t sequence_number, TimeDelta receive_time_delta) {
+            builder.AddPacket(sequence_number, receive_time_delta);
+          });
+      PacketLossPerFeedback packet_loss_per_feedback = {
+          .timestamp = feedback.log_time()};
+      builder.Update(packet_loss_per_feedback);
+      loss_per_feedback.push_back(packet_loss_per_feedback);
+    }
+  }
+
+  PacketLossSummary window_summary;
+  Timestamp last_observation_receive_time = Timestamp::Zero();
+
+  // Use loss based bwe 2 observation duration and observation window size.
+  constexpr TimeDelta kObservationDuration = TimeDelta::Millis(250);
+  constexpr uint32_t kObservationWindowSize = 20;
+  std::deque<PacketLossSummary> observations;
+  int previous_feedback_size = 0;
+  for (const PacketLossPerFeedback& feedback : loss_per_feedback) {
+    for (int64_t num = 0; num < feedback.num_missing_feedback; ++num) {
+      missing_feedback_series.points.emplace_back(
+          config.GetCallTimeSec(feedback.timestamp), 100 + num);
+    }
+
+    // Compute loss rate from the transport feedback.
+    float loss_rate = static_cast<float>(feedback.num_lost_packets * 100.0 /
+                                         feedback.num_packets_in_feedback);
+
+    loss_rate_series.points.emplace_back(
+        config.GetCallTimeSec(feedback.timestamp), loss_rate);
+    float reordered_rate =
+        previous_feedback_size == 0
+            ? 0
+            : static_cast<float>(feedback.num_reordered_packets * 100.0 /
+                                 previous_feedback_size);
+    previous_feedback_size = feedback.num_packets_in_feedback;
+    reordered_packets_between_feedback.points.emplace_back(
+        config.GetCallTimeSec(feedback.timestamp), reordered_rate);
+
+    // Compute loss rate in a window of kObservationWindowSize.
+    if (window_summary.num_packets == 0) {
+      window_summary.base_time = feedback.timestamp;
+    }
+    window_summary.num_packets += feedback.num_packets_in_feedback;
+    window_summary.num_lost_packets +=
+        feedback.num_lost_packets - feedback.num_reordered_packets;
+
+    const Timestamp last_received_time = feedback.timestamp;
+    const TimeDelta observation_duration =
+        window_summary.base_time == Timestamp::Zero()
+            ? TimeDelta::Zero()
+            : last_received_time - window_summary.base_time;
+    if (observation_duration > kObservationDuration) {
+      last_observation_receive_time = last_received_time;
+      observations.push_back(window_summary);
+      if (observations.size() > kObservationWindowSize) {
+        observations.pop_front();
+      }
+
+      // Compute average loss rate in a number of windows.
+      int total_packets = 0;
+      int total_loss = 0;
+      for (const auto& observation : observations) {
+        total_loss += observation.num_lost_packets;
+        total_packets += observation.num_packets;
+      }
+      if (total_packets > 0) {
+        float average_loss_rate = total_loss * 100.0 / total_packets;
+        average_loss_rate_series.points.emplace_back(
+            config.GetCallTimeSec(feedback.timestamp), average_loss_rate);
+      } else {
+        average_loss_rate_series.points.emplace_back(
+            config.GetCallTimeSec(feedback.timestamp), 0);
+      }
+      window_summary = PacketLossSummary();
+    }
+  }
+  // Add the data set to the plot.
+  plot->AppendTimeSeriesIfNotEmpty(std::move(loss_rate_series));
+  plot->AppendTimeSeriesIfNotEmpty(
+      std::move(reordered_packets_between_feedback));
+  plot->AppendTimeSeriesIfNotEmpty(std::move(average_loss_rate_series));
+  plot->AppendTimeSeriesIfNotEmpty(std::move(missing_feedback_series));
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 100, "Loss rate (percent)", kBottomMargin,
+                          kTopMargin);
+}
+
+void CreateTimestampGraph(PacketDirection direction,
+                          const ParsedRtcEventLog& parsed_log,
+                          const AnalyzerConfig& config,
+                          Plot* plot) {
+  for (const auto& stream : parsed_log.rtp_packets_by_ssrc(direction)) {
+    TimeSeries rtp_timestamps(
+        GetStreamName(parsed_log, direction, stream.ssrc) + " capture-time",
+        LineStyle::kLine, PointStyle::kHighlight);
+    for (const auto& packet : stream.packet_view) {
+      float x = config.GetCallTimeSec(packet.log_time());
+      float y = packet.header.timestamp;
+      rtp_timestamps.points.emplace_back(x, y);
+    }
+    plot->AppendTimeSeries(std::move(rtp_timestamps));
+
+    TimeSeries rtcp_timestamps(
+        GetStreamName(parsed_log, direction, stream.ssrc) +
+            " rtcp capture-time",
+        LineStyle::kLine, PointStyle::kHighlight);
+    // TODO(terelius): Why only sender reports?
+    const auto& sender_reports = parsed_log.sender_reports(direction);
+    for (const auto& rtcp : sender_reports) {
+      if (rtcp.sr.sender_ssrc() != stream.ssrc)
+        continue;
+      float x = config.GetCallTimeSec(rtcp.log_time());
+      float y = rtcp.sr.rtp_timestamp();
+      rtcp_timestamps.points.emplace_back(x, y);
+    }
+    plot->AppendTimeSeriesIfNotEmpty(std::move(rtcp_timestamps));
+  }
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "RTP timestamp", kBottomMargin, kTopMargin);
+  plot->SetTitle(GetDirectionAsString(direction) + " timestamps");
+}
+
+void CreateSenderAndReceiverReportPlot(
+    PacketDirection direction,
+    FunctionView<float(const rtcp::ReportBlock&)> fy,
+    std::string title,
+    std::string yaxis_label,
+    const ParsedRtcEventLog& parsed_log,
+    const AnalyzerConfig& config,
+    Plot* plot) {
+  std::map<uint32_t, TimeSeries> sr_reports_by_ssrc;
+  const auto& sender_reports = parsed_log.sender_reports(direction);
+  for (const auto& rtcp : sender_reports) {
+    float x = config.GetCallTimeSec(rtcp.log_time());
+    uint32_t ssrc = rtcp.sr.sender_ssrc();
+    for (const auto& block : rtcp.sr.report_blocks()) {
+      float y = fy(block);
+      auto sr_report_it = sr_reports_by_ssrc.find(ssrc);
+      bool inserted;
+      if (sr_report_it == sr_reports_by_ssrc.end()) {
+        std::tie(sr_report_it, inserted) = sr_reports_by_ssrc.emplace(
+            ssrc, TimeSeries(GetStreamName(parsed_log, direction, ssrc) +
+                                 " Sender Reports",
+                             LineStyle::kLine, PointStyle::kHighlight));
+      }
+      sr_report_it->second.points.emplace_back(x, y);
+    }
+  }
+  for (auto& kv : sr_reports_by_ssrc) {
+    plot->AppendTimeSeries(std::move(kv.second));
+  }
+
+  std::map<uint32_t, TimeSeries> rr_reports_by_ssrc;
+  const auto& receiver_reports = parsed_log.receiver_reports(direction);
+  for (const auto& rtcp : receiver_reports) {
+    float x = config.GetCallTimeSec(rtcp.log_time());
+    uint32_t ssrc = rtcp.rr.sender_ssrc();
+    for (const auto& block : rtcp.rr.report_blocks()) {
+      float y = fy(block);
+      auto rr_report_it = rr_reports_by_ssrc.find(ssrc);
+      bool inserted;
+      if (rr_report_it == rr_reports_by_ssrc.end()) {
+        std::tie(rr_report_it, inserted) = rr_reports_by_ssrc.emplace(
+            ssrc, TimeSeries(GetStreamName(parsed_log, direction, ssrc) +
+                                 " Receiver Reports",
+                             LineStyle::kLine, PointStyle::kHighlight));
+      }
+      rr_report_it->second.points.emplace_back(x, y);
+    }
+  }
+  for (auto& kv : rr_reports_by_ssrc) {
+    plot->AppendTimeSeries(std::move(kv.second));
+  }
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, yaxis_label, kBottomMargin, kTopMargin);
+  plot->SetTitle(title);
+}
+
+}  // namespace webrtc
diff --git a/rtc_tools/rtc_event_log_visualizer/analyze_rtp_rtcp.h b/rtc_tools/rtc_event_log_visualizer/analyze_rtp_rtcp.h
new file mode 100644
index 0000000000..2865023e60
--- /dev/null
+++ b/rtc_tools/rtc_event_log_visualizer/analyze_rtp_rtcp.h
@@ -0,0 +1,100 @@
+/*
+ *  Copyright (c) 2026 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZE_RTP_RTCP_H_
+#define RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZE_RTP_RTCP_H_
+
+#include <string>
+
+#include "api/function_view.h"
+#include "logging/rtc_event_log/rtc_event_log_parser.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
+#include "rtc_tools/rtc_event_log_visualizer/analyzer_common.h"
+#include "rtc_tools/rtc_event_log_visualizer/plot_base.h"
+
+namespace webrtc {
+
+float GetHighestSeqNumber(const rtcp::ReportBlock& block);
+float GetFractionLost(const rtcp::ReportBlock& block);
+float GetCumulativeLost(const rtcp::ReportBlock& block);
+float DelaySinceLastSr(const rtcp::ReportBlock& block);
+
+void CreatePacketGraph(PacketDirection direction,
+                       const ParsedRtcEventLog& parsed_log,
+                       const AnalyzerConfig& config,
+                       Plot* plot);
+
+void CreateRtcpTypeGraph(PacketDirection direction,
+                         const ParsedRtcEventLog& parsed_log,
+                         const AnalyzerConfig& config,
+                         Plot* plot);
+
+void CreateAccumulatedPacketsGraph(PacketDirection direction,
+                                   const ParsedRtcEventLog& parsed_log,
+                                   const AnalyzerConfig& config,
+                                   Plot* plot);
+
+void CreatePacketRateGraph(PacketDirection direction,
+                           const ParsedRtcEventLog& parsed_log,
+                           const AnalyzerConfig& config,
+                           Plot* plot);
+
+void CreateTotalPacketRateGraph(PacketDirection direction,
+                                const ParsedRtcEventLog& parsed_log,
+                                const AnalyzerConfig& config,
+                                Plot* plot);
+
+void CreateSequenceNumberGraph(const ParsedRtcEventLog& parsed_log,
+                               const AnalyzerConfig& config,
+                               Plot* plot);
+
+void CreateIncomingPacketLossGraph(const ParsedRtcEventLog& parsed_log,
+                                   const AnalyzerConfig& config,
+                                   Plot* plot);
+
+void CreateStreamBitrateGraph(PacketDirection direction,
+                              const ParsedRtcEventLog& parsed_log,
+                              const AnalyzerConfig& config,
+                              Plot* plot);
+
+void CreateBitrateAllocationGraph(PacketDirection direction,
+                                  const ParsedRtcEventLog& parsed_log,
+                                  const AnalyzerConfig& config,
+                                  Plot* plot);
+
+void CreateOutgoingEcnFeedbackGraph(const ParsedRtcEventLog& parsed_log,
+                                    const AnalyzerConfig& config,
+                                    Plot* plot);
+
+void CreateIncomingEcnFeedbackGraph(const ParsedRtcEventLog& parsed_log,
+                                    const AnalyzerConfig& config,
+                                    Plot* plot);
+
+void CreateOutgoingLossRateGraph(const ParsedRtcEventLog& parsed_log,
+                                 const AnalyzerConfig& config,
+                                 Plot* plot);
+
+void CreateTimestampGraph(PacketDirection direction,
+                          const ParsedRtcEventLog& parsed_log,
+                          const AnalyzerConfig& config,
+                          Plot* plot);
+
+void CreateSenderAndReceiverReportPlot(
+    PacketDirection direction,
+    FunctionView<float(const rtcp::ReportBlock&)> fy,
+    std::string title,
+    std::string yaxis_label,
+    const ParsedRtcEventLog& parsed_log,
+    const AnalyzerConfig& config,
+    Plot* plot);
+
+}  // namespace webrtc
+
+#endif  // RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZE_RTP_RTCP_H_
diff --git a/rtc_tools/rtc_event_log_visualizer/analyzer.cc b/rtc_tools/rtc_event_log_visualizer/analyzer.cc
index cc7a5457bd..73e1db7509 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyzer.cc
+++ b/rtc_tools/rtc_event_log_visualizer/analyzer.cc
@@ -10,476 +10,35 @@
 
 #include "rtc_tools/rtc_event_log_visualizer/analyzer.h"
 
-#include <algorithm>
-#include <cmath>
 #include <cstddef>
-#include <cstdint>
-#include <deque>
-#include <limits>
-#include <map>
-#include <memory>
-#include <optional>
 #include <string>
-#include <tuple>
-#include <unordered_set>
-#include <utility>
 #include <vector>
 
 #include "absl/algorithm/container.h"
-#include "absl/functional/bind_front.h"
 #include "absl/strings/string_view.h"
-#include "api/candidate.h"
-#include "api/dtls_transport_interface.h"
-#include "api/environment/environment.h"
 #include "api/environment/environment_factory.h"
-#include "api/field_trials.h"
 #include "api/function_view.h"
-#include "api/media_types.h"
-#include "api/rtp_headers.h"
-#include "api/transport/bandwidth_usage.h"
-#include "api/transport/ecn_marking.h"
-#include "api/transport/goog_cc_factory.h"
-#include "api/transport/network_control.h"
-#include "api/transport/network_types.h"
 #include "api/units/data_rate.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
-#include "logging/rtc_event_log/events/logged_rtp_rtcp.h"
-#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair.h"
-#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.h"
 #include "logging/rtc_event_log/rtc_event_log_parser.h"
-#include "logging/rtc_event_log/rtc_event_processor.h"
-#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.h"
-#include "modules/congestion_controller/include/receive_side_congestion_controller.h"
-#include "modules/congestion_controller/rtp/transport_feedback_adapter.h"
-#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
-#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
-#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
-#include "modules/rtp_rtcp/source/rtcp_packet/congestion_control_feedback.h"
-#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
-#include "modules/rtp_rtcp/source/rtcp_packet/remb.h"
 #include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
-#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
-#include "modules/rtp_rtcp/source/rtcp_packet/target_bitrate.h"
-#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
-#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
-#include "modules/rtp_rtcp/source/rtp_packet_received.h"
-#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
-#include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
-#include "rtc_base/network/sent_packet.h"
-#include "rtc_base/numerics/sequence_number_unwrapper.h"
-#include "rtc_base/rate_statistics.h"
-#include "rtc_base/strings/string_builder.h"
 #include "rtc_tools/rtc_event_log_visualizer/analyze_audio.h"
+#include "rtc_tools/rtc_event_log_visualizer/analyze_bwe.h"
+#include "rtc_tools/rtc_event_log_visualizer/analyze_connectivity.h"
+#include "rtc_tools/rtc_event_log_visualizer/analyze_rtp_rtcp.h"
 #include "rtc_tools/rtc_event_log_visualizer/analyzer_common.h"
-#include "rtc_tools/rtc_event_log_visualizer/log_scream_simulation.h"
-#include "rtc_tools/rtc_event_log_visualizer/log_simulation.h"
 #include "rtc_tools/rtc_event_log_visualizer/plot_base.h"
-#include "system_wrappers/include/clock.h"
 
 namespace webrtc {
 
-namespace {
-
-std::string SsrcToString(uint32_t ssrc) {
-  StringBuilder ss;
-  ss << "SSRC " << ssrc;
-  return ss.Release();
-}
-
-// Checks whether an SSRC is contained in the list of desired SSRCs.
-// Note that an empty SSRC list matches every SSRC.
-bool MatchingSsrc(uint32_t ssrc, const std::vector<uint32_t>& desired_ssrc) {
-  if (desired_ssrc.empty())
-    return true;
-  return std::find(desired_ssrc.begin(), desired_ssrc.end(), ssrc) !=
-         desired_ssrc.end();
-}
-
-double AbsSendTimeToMicroseconds(int64_t abs_send_time) {
-  // The timestamp is a fixed point representation with 6 bits for seconds
-  // and 18 bits for fractions of a second. Thus, we divide by 2^18 to get the
-  // time in seconds and then multiply by kNumMicrosecsPerSec to convert to
-  // microseconds.
-  static constexpr double kTimestampToMicroSec =
-      static_cast<double>(kNumMicrosecsPerSec) / static_cast<double>(1ul << 18);
-  return abs_send_time * kTimestampToMicroSec;
-}
-
-// Computes the difference `later` - `earlier` where `later` and `earlier`
-// are counters that wrap at `modulus`. The difference is chosen to have the
-// least absolute value. For example if `modulus` is 8, then the difference will
-// be chosen in the range [-3, 4]. If `modulus` is 9, then the difference will
-// be in [-4, 4].
-int64_t WrappingDifference(uint32_t later, uint32_t earlier, int64_t modulus) {
-  RTC_DCHECK_LE(1, modulus);
-  RTC_DCHECK_LT(later, modulus);
-  RTC_DCHECK_LT(earlier, modulus);
-  int64_t difference =
-      static_cast<int64_t>(later) - static_cast<int64_t>(earlier);
-  int64_t max_difference = modulus / 2;
-  int64_t min_difference = max_difference - modulus + 1;
-  if (difference > max_difference) {
-    difference -= modulus;
-  }
-  if (difference < min_difference) {
-    difference += modulus;
-  }
-  if (difference > max_difference / 2 || difference < min_difference / 2) {
-    RTC_LOG(LS_WARNING) << "Difference between" << later << " and " << earlier
-                        << " expected to be in the range ("
-                        << min_difference / 2 << "," << max_difference / 2
-                        << ") but is " << difference
-                        << ". Correct unwrapping is uncertain.";
-  }
-  return difference;
-}
-
-// This is much more reliable for outgoing streams than for incoming streams.
-template <typename RtpPacketContainer>
-std::optional<uint32_t> EstimateRtpClockFrequency(
-    const RtpPacketContainer& packets,
-    int64_t end_time_us) {
-  RTC_CHECK(packets.size() >= 2);
-  SeqNumUnwrapper<uint32_t> unwrapper;
-  int64_t first_rtp_timestamp =
-      unwrapper.Unwrap(packets[0].rtp.header.timestamp);
-  int64_t first_log_timestamp = packets[0].log_time_us();
-  int64_t last_rtp_timestamp = first_rtp_timestamp;
-  int64_t last_log_timestamp = first_log_timestamp;
-  for (size_t i = 1; i < packets.size(); i++) {
-    if (packets[i].log_time_us() > end_time_us)
-      break;
-    last_rtp_timestamp = unwrapper.Unwrap(packets[i].rtp.header.timestamp);
-    last_log_timestamp = packets[i].log_time_us();
-  }
-  if (last_log_timestamp - first_log_timestamp < kNumMicrosecsPerSec) {
-    RTC_LOG(LS_WARNING)
-        << "Failed to estimate RTP clock frequency: Stream too short. ("
-        << packets.size() << " packets, "
-        << last_log_timestamp - first_log_timestamp << " us)";
-    return std::nullopt;
-  }
-  double duration =
-      static_cast<double>(last_log_timestamp - first_log_timestamp) /
-      kNumMicrosecsPerSec;
-  double estimated_frequency =
-      (last_rtp_timestamp - first_rtp_timestamp) / duration;
-  for (uint32_t f : {8000, 16000, 32000, 48000, 90000}) {
-    if (std::fabs(estimated_frequency - f) < 0.15 * f) {
-      return f;
-    }
-  }
-  RTC_LOG(LS_WARNING) << "Failed to estimate RTP clock frequency: Estimate "
-                      << estimated_frequency
-                      << " not close to any standard RTP frequency."
-                      << " Last timestamp " << last_rtp_timestamp
-                      << " first timestamp " << first_rtp_timestamp;
-  return std::nullopt;
-}
-
-std::optional<double> NetworkDelayDiff_AbsSendTime(
-    const LoggedRtpPacketIncoming& old_packet,
-    const LoggedRtpPacketIncoming& new_packet) {
-  if (old_packet.rtp.header.extension.hasAbsoluteSendTime &&
-      new_packet.rtp.header.extension.hasAbsoluteSendTime) {
-    int64_t send_time_diff = WrappingDifference(
-        new_packet.rtp.header.extension.absoluteSendTime,
-        old_packet.rtp.header.extension.absoluteSendTime, 1ul << 24);
-    int64_t recv_time_diff =
-        new_packet.log_time_us() - old_packet.log_time_us();
-    double delay_change_us =
-        recv_time_diff - AbsSendTimeToMicroseconds(send_time_diff);
-    return delay_change_us / 1000;
-  } else {
-    return std::nullopt;
-  }
-}
-
-std::optional<double> NetworkDelayDiff_CaptureTime(
-    const LoggedRtpPacketIncoming& old_packet,
-    const LoggedRtpPacketIncoming& new_packet,
-    const double sample_rate) {
-  int64_t send_time_diff =
-      WrappingDifference(new_packet.rtp.header.timestamp,
-                         old_packet.rtp.header.timestamp, 1ull << 32);
-  int64_t recv_time_diff = new_packet.log_time_us() - old_packet.log_time_us();
-
-  double delay_change =
-      static_cast<double>(recv_time_diff) / 1000 -
-      static_cast<double>(send_time_diff) / sample_rate * 1000;
-  if (delay_change < -10000 || 10000 < delay_change) {
-    RTC_LOG(LS_WARNING) << "Very large delay change. Timestamps correct?";
-    RTC_LOG(LS_WARNING) << "Old capture time "
-                        << old_packet.rtp.header.timestamp << ", received time "
-                        << old_packet.log_time_us();
-    RTC_LOG(LS_WARNING) << "New capture time "
-                        << new_packet.rtp.header.timestamp << ", received time "
-                        << new_packet.log_time_us();
-    RTC_LOG(LS_WARNING) << "Receive time difference " << recv_time_diff << " = "
-                        << static_cast<double>(recv_time_diff) /
-                               kNumMicrosecsPerSec
-                        << "s";
-    RTC_LOG(LS_WARNING) << "Send time difference " << send_time_diff << " = "
-                        << static_cast<double>(send_time_diff) / sample_rate
-                        << "s";
-  }
-  return delay_change;
-}
-
-template <typename T>
-TimeSeries CreateRtcpTypeTimeSeries(const std::vector<T>& rtcp_list,
-                                    AnalyzerConfig config,
-                                    std::string rtcp_name,
-                                    int category_id) {
-  TimeSeries time_series(rtcp_name, LineStyle::kNone, PointStyle::kHighlight);
-  for (const auto& rtcp : rtcp_list) {
-    float x = config.GetCallTimeSec(rtcp.timestamp);
-    float y = category_id;
-    time_series.points.emplace_back(x, y);
-  }
-  return time_series;
-}
-
-const char kUnknownEnumValue[] = "unknown";
-
-// TODO(tommi): This should be "host".
-const char kIceCandidateTypeLocal[] = "local";
-// TODO(tommi): This should be "srflx".
-const char kIceCandidateTypeStun[] = "stun";
-const char kIceCandidateTypePrflx[] = "prflx";
-const char kIceCandidateTypeRelay[] = "relay";
-
-const char kProtocolUdp[] = "udp";
-const char kProtocolTcp[] = "tcp";
-const char kProtocolSsltcp[] = "ssltcp";
-const char kProtocolTls[] = "tls";
-
-const char kAddressFamilyIpv4[] = "ipv4";
-const char kAddressFamilyIpv6[] = "ipv6";
-
-const char kNetworkTypeEthernet[] = "ethernet";
-const char kNetworkTypeLoopback[] = "loopback";
-const char kNetworkTypeWifi[] = "wifi";
-const char kNetworkTypeVpn[] = "vpn";
-const char kNetworkTypeCellular[] = "cellular";
-
-absl::string_view GetIceCandidateTypeAsString(IceCandidateType type) {
-  switch (type) {
-    case IceCandidateType::kHost:
-      return kIceCandidateTypeLocal;
-    case IceCandidateType::kSrflx:
-      return kIceCandidateTypeStun;
-    case IceCandidateType::kPrflx:
-      return kIceCandidateTypePrflx;
-    case IceCandidateType::kRelay:
-      return kIceCandidateTypeRelay;
-    default:
-      RTC_DCHECK_NOTREACHED();
-      return kUnknownEnumValue;
-  }
-}
-
-std::string GetProtocolAsString(IceCandidatePairProtocol protocol) {
-  switch (protocol) {
-    case IceCandidatePairProtocol::kUdp:
-      return kProtocolUdp;
-    case IceCandidatePairProtocol::kTcp:
-      return kProtocolTcp;
-    case IceCandidatePairProtocol::kSsltcp:
-      return kProtocolSsltcp;
-    case IceCandidatePairProtocol::kTls:
-      return kProtocolTls;
-    default:
-      return kUnknownEnumValue;
-  }
-}
-
-std::string GetAddressFamilyAsString(IceCandidatePairAddressFamily family) {
-  switch (family) {
-    case IceCandidatePairAddressFamily::kIpv4:
-      return kAddressFamilyIpv4;
-    case IceCandidatePairAddressFamily::kIpv6:
-      return kAddressFamilyIpv6;
-    default:
-      return kUnknownEnumValue;
-  }
-}
-
-std::string GetNetworkTypeAsString(IceCandidateNetworkType type) {
-  switch (type) {
-    case IceCandidateNetworkType::kEthernet:
-      return kNetworkTypeEthernet;
-    case IceCandidateNetworkType::kLoopback:
-      return kNetworkTypeLoopback;
-    case IceCandidateNetworkType::kWifi:
-      return kNetworkTypeWifi;
-    case IceCandidateNetworkType::kVpn:
-      return kNetworkTypeVpn;
-    case IceCandidateNetworkType::kCellular:
-      return kNetworkTypeCellular;
-    default:
-      return kUnknownEnumValue;
-  }
-}
-
-std::string GetCandidatePairLogDescriptionAsString(
-    const LoggedIceCandidatePairConfig& config) {
-  // Example: stun:wifi->relay(tcp):cellular@udp:ipv4
-  // represents a pair of a local server-reflexive candidate on a WiFi network
-  // and a remote relay candidate using TCP as the relay protocol on a cell
-  // network, when the candidate pair communicates over UDP using IPv4.
-  StringBuilder ss;
-  ss << GetIceCandidateTypeAsString(config.local_candidate_type);
-
-  if (config.local_candidate_type == IceCandidateType::kRelay) {
-    ss << "(" << GetProtocolAsString(config.local_relay_protocol) << ")";
-  }
-
-  ss << ":" << GetNetworkTypeAsString(config.local_network_type) << ":"
-     << GetAddressFamilyAsString(config.local_address_family) << "->"
-     << GetIceCandidateTypeAsString(config.remote_candidate_type) << ":"
-     << GetAddressFamilyAsString(config.remote_address_family) << "@"
-     << GetProtocolAsString(config.candidate_pair_protocol);
-  return ss.Release();
-}
-
-std::string GetDirectionAsString(PacketDirection direction) {
-  if (direction == kIncomingPacket) {
-    return "Incoming";
-  } else {
-    return "Outgoing";
-  }
-}
-
-std::string GetDirectionAsShortString(PacketDirection direction) {
-  if (direction == kIncomingPacket) {
-    return "In";
-  } else {
-    return "Out";
-  }
-}
-
-struct FakeExtensionSmall {
-  static constexpr RTPExtensionType kId = kRtpExtensionMid;
-  static constexpr absl::string_view Uri() { return "fake-extension-small"; }
-};
-struct FakeExtensionLarge {
-  static constexpr RTPExtensionType kId = kRtpExtensionRtpStreamId;
-  static constexpr absl::string_view Uri() { return "fake-extension-large"; }
-};
-
-RtpPacketReceived RtpPacketForBWEFromHeader(const RTPHeader& header) {
-  RtpHeaderExtensionMap rtp_header_extensions(/*extmap_allow_mixed=*/true);
-  // ReceiveSideCongestionController doesn't need to know extensions ids as
-  // long as it able to get extensions by type. So any ids would work here.
-  rtp_header_extensions.Register<TransmissionOffset>(1);
-  rtp_header_extensions.Register<AbsoluteSendTime>(2);
-  rtp_header_extensions.Register<TransportSequenceNumber>(3);
-  rtp_header_extensions.Register<FakeExtensionSmall>(4);
-  // Use id > 14 to force two byte header per rtp header when this one is used.
-  rtp_header_extensions.Register<FakeExtensionLarge>(16);
-
-  RtpPacketReceived rtp_packet(&rtp_header_extensions);
-  // Set only fields that might be relevant for the bandwidth estimatior.
-  rtp_packet.SetSsrc(header.ssrc);
-  rtp_packet.SetTimestamp(header.timestamp);
-  size_t num_bwe_extensions = 0;
-  if (header.extension.hasTransmissionTimeOffset) {
-    rtp_packet.SetExtension<TransmissionOffset>(
-        header.extension.transmissionTimeOffset);
-    ++num_bwe_extensions;
-  }
-  if (header.extension.hasAbsoluteSendTime) {
-    rtp_packet.SetExtension<AbsoluteSendTime>(
-        header.extension.absoluteSendTime);
-    ++num_bwe_extensions;
-  }
-  if (header.extension.hasTransportSequenceNumber) {
-    rtp_packet.SetExtension<TransportSequenceNumber>(
-        header.extension.transportSequenceNumber);
-    ++num_bwe_extensions;
-  }
-
-  // All parts of the RTP header are 32bit aligned.
-  RTC_CHECK_EQ(header.headerLength % 4, 0);
-
-  // Original packet could have more extensions, there could be csrcs that are
-  // not propagated by the rtc event log, i.e. logged header size might be
-  // larger that rtp_packet.header_size(). Increase it by setting an extra fake
-  // extension.
-  RTC_CHECK_GE(header.headerLength, rtp_packet.headers_size());
-  size_t bytes_to_add = header.headerLength - rtp_packet.headers_size();
-  if (bytes_to_add > 0) {
-    if (bytes_to_add <= 16) {
-      // one-byte header rtp header extension allows to add up to 16 bytes.
-      rtp_packet.AllocateExtension(FakeExtensionSmall::kId, bytes_to_add - 1);
-    } else {
-      // two-byte header rtp header extension would also add one byte per
-      // already set extension.
-      rtp_packet.AllocateExtension(FakeExtensionLarge::kId,
-                                   bytes_to_add - 2 - num_bwe_extensions);
-    }
-  }
-  RTC_CHECK_EQ(rtp_packet.headers_size(), header.headerLength);
-
-  return rtp_packet;
-}
-
-struct PacketLossSummary {
-  size_t num_packets = 0;
-  size_t num_lost_packets = 0;
-  Timestamp base_time = Timestamp::MinusInfinity();
-};
-
-float GetHighestSeqNumber(const rtcp::ReportBlock& block) {
-  return block.extended_high_seq_num();
-}
-
-float GetFractionLost(const rtcp::ReportBlock& block) {
-  return static_cast<double>(block.fraction_lost()) / 256 * 100;
-}
-
-float GetCumulativeLost(const rtcp::ReportBlock& block) {
-  return block.cumulative_lost();
-}
-
-float DelaySinceLastSr(const rtcp::ReportBlock& block) {
-  return static_cast<double>(block.delay_since_last_sr()) / 65536;
-}
-
-std::map<uint32_t, std::string> BuildCandidateIdLogDescriptionMap(
-    const std::vector<LoggedIceCandidatePairConfig>&
-        ice_candidate_pair_configs) {
-  std::map<uint32_t, std::string> candidate_pair_desc_by_id;
-  for (const auto& config : ice_candidate_pair_configs) {
-    // TODO(qingsi): Add the handling of the "Updated" config event after the
-    // visualization of property change for candidate pairs is introduced.
-    if (candidate_pair_desc_by_id.find(config.candidate_pair_id) ==
-        candidate_pair_desc_by_id.end()) {
-      const std::string candidate_pair_desc =
-          GetCandidatePairLogDescriptionAsString(config);
-      candidate_pair_desc_by_id[config.candidate_pair_id] = candidate_pair_desc;
-    }
-  }
-  return candidate_pair_desc_by_id;
-}
-
-}  // namespace
-
-EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log,
+EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& parsed_log,
                                    bool normalize_time)
-    : env_(CreateEnvironment()), parsed_log_(log) {
+    : parsed_log_(parsed_log),
+      config_(CreateEnvironment(), parsed_log, normalize_time) {
   config_.window_duration_ = TimeDelta::Millis(250);
   config_.step_ = TimeDelta::Millis(10);
-  if (!log.start_log_events().empty()) {
-    config_.rtc_to_utc_offset_ = log.start_log_events()[0].utc_time() -
-                                 log.start_log_events()[0].log_time();
-  }
-  config_.normalize_time_ = normalize_time;
-  config_.begin_time_ = parsed_log_.first_timestamp();
-  config_.end_time_ = parsed_log_.last_timestamp();
   if (config_.end_time_ < config_.begin_time_) {
     RTC_LOG(LS_WARNING) << "No useful events in the log.";
     config_.begin_time_ = config_.end_time_ = Timestamp::Zero();
@@ -492,10 +51,9 @@ EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log,
                    << " seconds long.";
 }
 
-EventLogAnalyzer::EventLogAnalyzer(const Environment& env,
-                                   const ParsedRtcEventLog& log,
+EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& parsed_log,
                                    const AnalyzerConfig& config)
-    : env_(env), parsed_log_(log), config_(config) {
+    : parsed_log_(parsed_log), config_(config) {
   RTC_LOG(LS_INFO) << "Log is "
                    << (parsed_log_.last_timestamp().ms() -
                        parsed_log_.first_timestamp().ms()) /
@@ -503,31 +61,18 @@ EventLogAnalyzer::EventLogAnalyzer(const Environment& env,
                    << " seconds long.";
 }
 
-class BitrateObserver : public RemoteBitrateObserver {
- public:
-  BitrateObserver() : last_bitrate_bps_(0), bitrate_updated_(false) {}
-
-  void Update(NetworkControlUpdate update) {
-    if (update.target_rate) {
-      last_bitrate_bps_ = update.target_rate->target_rate.bps();
-      bitrate_updated_ = true;
+void EventLogAnalyzer::CreateGraphsByName(const std::vector<std::string>& names,
+                                          PlotCollection* collection) const {
+  for (absl::string_view name : names) {
+    auto plot = absl::c_find_if(plots_, [name](const PlotDeclaration& plot) {
+      return plot.label == name;
+    });
+    if (plot != plots_.end()) {
+      Plot* output = collection->AppendNewPlot(plot->label);
+      plot->plot_func(output);
     }
   }
-
-  void OnReceiveBitrateChanged(const std::vector<uint32_t>& ssrcs,
-                               uint32_t bitrate) override {}
-
-  uint32_t last_bitrate_bps() const { return last_bitrate_bps_; }
-  bool GetAndResetBitrateUpdated() {
-    bool bitrate_updated = bitrate_updated_;
-    bitrate_updated_ = false;
-    return bitrate_updated;
-  }
-
- private:
-  uint32_t last_bitrate_bps_;
-  bool bitrate_updated_;
-};
+}
 
 void EventLogAnalyzer::InitializeMapOfNamedGraphs(bool show_detector_state,
                                                   bool show_alr_state,
@@ -737,1664 +282,155 @@ void EventLogAnalyzer::InitializeMapOfNamedGraphs(bool show_detector_state,
     this->CreateDtlsWritableStateGraph(plot);
   });
 }
-
-void EventLogAnalyzer::CreateGraphsByName(const std::vector<std::string>& names,
-                                          PlotCollection* collection) const {
-  for (absl::string_view name : names) {
-    auto plot = absl::c_find_if(plots_, [name](const PlotDeclaration& plot) {
-      return plot.label == name;
-    });
-    if (plot != plots_.end()) {
-      Plot* output = collection->AppendNewPlot(plot->label);
-      plot->plot_func(output);
-    }
-  }
-}
-
-void EventLogAnalyzer::CreatePacketGraph(PacketDirection direction,
-                                         Plot* plot) const {
-  for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
-    // Filter on SSRC.
-    if (!MatchingSsrc(stream.ssrc, desired_ssrc_)) {
-      continue;
-    }
-
-    TimeSeries time_series(GetStreamName(parsed_log_, direction, stream.ssrc),
-                           LineStyle::kBar);
-    auto GetPacketSize = [](const LoggedRtpPacket& packet) {
-      return std::optional<float>(packet.total_length);
-    };
-    auto ToCallTime = [this](const LoggedRtpPacket& packet) {
-      return this->config_.GetCallTimeSec(packet.timestamp);
-    };
-    ProcessPoints<LoggedRtpPacket>(ToCallTime, GetPacketSize,
-                                   stream.packet_view, &time_series);
-    plot->AppendTimeSeries(std::move(time_series));
-  }
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Packet size (bytes)", kBottomMargin,
-                          kTopMargin);
-  plot->SetTitle(GetDirectionAsString(direction) + " RTP packets");
-}
-
-void EventLogAnalyzer::CreateRtcpTypeGraph(PacketDirection direction,
-                                           Plot* plot) const {
-  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(
-      parsed_log_.transport_feedbacks(direction), config_, "TWCC", 1));
-  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(
-      parsed_log_.congestion_feedback(direction), config_, "CCFB", 2));
-  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(
-      parsed_log_.receiver_reports(direction), config_, "RR", 3));
-  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(
-      parsed_log_.sender_reports(direction), config_, "SR", 4));
-  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(
-      parsed_log_.extended_reports(direction), config_, "XR", 5));
-  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(parsed_log_.nacks(direction),
-                                                  config_, "NACK", 6));
-  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(parsed_log_.rembs(direction),
-                                                  config_, "REMB", 7));
-  plot->AppendTimeSeries(
-      CreateRtcpTypeTimeSeries(parsed_log_.firs(direction), config_, "FIR", 8));
-  plot->AppendTimeSeries(
-      CreateRtcpTypeTimeSeries(parsed_log_.plis(direction), config_, "PLI", 9));
-  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(parsed_log_.byes(direction),
-                                                  config_, "BYE", 10));
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "RTCP type", kBottomMargin, kTopMargin);
-  plot->SetTitle(GetDirectionAsString(direction) + " RTCP packets");
-  plot->SetYAxisTickLabels({{1, "TWCC"},
-                            {2, "CCFB"},
-                            {3, "RR"},
-                            {4, "SR"},
-                            {5, "XR"},
-                            {6, "NACK"},
-                            {7, "REMB"},
-                            {8, "FIR"},
-                            {9, "PLI"},
-                            {10, "BYE"}});
-}
-
-template <typename IterableType>
-void EventLogAnalyzer::CreateAccumulatedPacketsTimeSeries(
-    Plot* plot,
-    const IterableType& packets,
-    const std::string& label) const {
-  TimeSeries time_series(label, LineStyle::kStep);
-  for (size_t i = 0; i < packets.size(); i++) {
-    float x = config_.GetCallTimeSec(packets[i].log_time());
-    time_series.points.emplace_back(x, i + 1);
-  }
-  plot->AppendTimeSeries(std::move(time_series));
-}
-
-void EventLogAnalyzer::CreateAccumulatedPacketsGraph(PacketDirection direction,
-                                                     Plot* plot) const {
-  for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
-    if (!MatchingSsrc(stream.ssrc, desired_ssrc_))
-      continue;
-    std::string label = std::string("RTP ") +
-                        GetStreamName(parsed_log_, direction, stream.ssrc);
-    CreateAccumulatedPacketsTimeSeries(plot, stream.packet_view, label);
-  }
-  std::string label =
-      std::string("RTCP ") + "(" + GetDirectionAsShortString(direction) + ")";
-  if (direction == kIncomingPacket) {
-    CreateAccumulatedPacketsTimeSeries(
-        plot, parsed_log_.incoming_rtcp_packets(), label);
-  } else {
-    CreateAccumulatedPacketsTimeSeries(
-        plot, parsed_log_.outgoing_rtcp_packets(), label);
-  }
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Received Packets", kBottomMargin, kTopMargin);
-  plot->SetTitle(std::string("Accumulated ") + GetDirectionAsString(direction) +
-                 " RTP/RTCP packets");
-}
-
-void EventLogAnalyzer::CreatePacketRateGraph(PacketDirection direction,
-                                             Plot* plot) const {
-  auto CountPackets = [](auto packet) { return 1.0; };
-  for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
-    // Filter on SSRC.
-    if (!MatchingSsrc(stream.ssrc, desired_ssrc_)) {
-      continue;
-    }
-    TimeSeries time_series(
-        std::string("RTP ") +
-            GetStreamName(parsed_log_, direction, stream.ssrc),
-        LineStyle::kLine);
-    MovingAverage<LoggedRtpPacket, double>(CountPackets, stream.packet_view,
-                                           config_, &time_series);
-    plot->AppendTimeSeries(std::move(time_series));
-  }
-  TimeSeries time_series(
-      std::string("RTCP ") + "(" + GetDirectionAsShortString(direction) + ")",
-      LineStyle::kLine);
-  if (direction == kIncomingPacket) {
-    MovingAverage<LoggedRtcpPacketIncoming, double>(
-        CountPackets, parsed_log_.incoming_rtcp_packets(), config_,
-        &time_series);
-  } else {
-    MovingAverage<LoggedRtcpPacketOutgoing, double>(
-        CountPackets, parsed_log_.outgoing_rtcp_packets(), config_,
-        &time_series);
-  }
-  plot->AppendTimeSeries(std::move(time_series));
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Packet Rate (packets/s)", kBottomMargin,
-                          kTopMargin);
-  plot->SetTitle("Rate of " + GetDirectionAsString(direction) +
-                 " RTP/RTCP packets");
-}
-
-void EventLogAnalyzer::CreateTotalPacketRateGraph(PacketDirection direction,
-                                                  Plot* plot) const {
-  // Contains a log timestamp to enable counting logged events of different
-  // types using MovingAverage().
-  class LogTime {
-   public:
-    explicit LogTime(Timestamp log_time) : log_time_(log_time) {}
-    Timestamp log_time() const { return log_time_; }
-
-   private:
-    Timestamp log_time_;
-  };
-  std::vector<LogTime> packet_times;
-  auto handle_rtp = [&packet_times](const LoggedRtpPacket& packet) {
-    packet_times.emplace_back(packet.log_time());
-  };
-  RtcEventProcessor process;
-  for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
-    process.AddEvents(stream.packet_view, handle_rtp, direction);
-  }
-  if (direction == kIncomingPacket) {
-    auto handle_incoming_rtcp =
-        [&packet_times](const LoggedRtcpPacketIncoming& packet) {
-          packet_times.emplace_back(packet.log_time());
-        };
-    process.AddEvents(parsed_log_.incoming_rtcp_packets(),
-                      handle_incoming_rtcp);
-  } else {
-    auto handle_outgoing_rtcp =
-        [&packet_times](const LoggedRtcpPacketOutgoing& packet) {
-          packet_times.emplace_back(packet.log_time());
-        };
-    process.AddEvents(parsed_log_.outgoing_rtcp_packets(),
-                      handle_outgoing_rtcp);
-  }
-  process.ProcessEventsInOrder();
-  TimeSeries time_series(std::string("Total ") + "(" +
-                             GetDirectionAsShortString(direction) + ") packets",
-                         LineStyle::kLine);
-  MovingAverage<LogTime, uint64_t>([](auto packet) { return 1; }, packet_times,
-                                   config_, &time_series);
-  plot->AppendTimeSeries(std::move(time_series));
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Packet Rate (packets/s)", kBottomMargin,
-                          kTopMargin);
-  plot->SetTitle("Rate of all " + GetDirectionAsString(direction) +
-                 " RTP/RTCP packets");
-}
-
-// For each SSRC, plot the time between the consecutive playouts.
 void EventLogAnalyzer::CreatePlayoutGraph(Plot* plot) const {
-  for (const auto& playout_stream : parsed_log_.audio_playout_events()) {
-    uint32_t ssrc = playout_stream.first;
-    if (!MatchingSsrc(ssrc, desired_ssrc_))
-      continue;
-    std::optional<int64_t> last_playout_ms;
-    TimeSeries time_series(SsrcToString(ssrc), LineStyle::kBar);
-    for (const auto& playout_event : playout_stream.second) {
-      float x = config_.GetCallTimeSec(playout_event.log_time());
-      int64_t playout_time_ms = playout_event.log_time_ms();
-      // If there were no previous playouts, place the point on the x-axis.
-      float y = playout_time_ms - last_playout_ms.value_or(playout_time_ms);
-      time_series.points.push_back(TimeSeriesPoint(x, y));
-      last_playout_ms.emplace(playout_time_ms);
-    }
-    plot->AppendTimeSeries(std::move(time_series));
-  }
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Time since last playout (ms)", kBottomMargin,
-                          kTopMargin);
-  plot->SetTitle("Audio playout");
+  webrtc::CreatePlayoutGraph(parsed_log_, config_, plot);
 }
 
 void EventLogAnalyzer::CreateNetEqSetMinimumDelay(Plot* plot) const {
-  for (const auto& playout_stream :
-       parsed_log_.neteq_set_minimum_delay_events()) {
-    uint32_t ssrc = playout_stream.first;
-    if (!MatchingSsrc(ssrc, desired_ssrc_))
-      continue;
-
-    TimeSeries time_series(SsrcToString(ssrc), LineStyle::kStep,
-                           PointStyle::kHighlight);
-    for (const auto& event : playout_stream.second) {
-      float x = config_.GetCallTimeSec(event.log_time());
-      float y = event.minimum_delay_ms;
-      time_series.points.push_back(TimeSeriesPoint(x, y));
-    }
-    plot->AppendTimeSeries(std::move(time_series));
-  }
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1000, "Minimum Delay (ms)", kBottomMargin,
-                          kTopMargin);
-  plot->SetTitle("Set Minimum Delay");
+  webrtc::CreateNetEqSetMinimumDelay(parsed_log_, config_, plot);
 }
 
-// For audio SSRCs, plot the audio level.
 void EventLogAnalyzer::CreateAudioLevelGraph(PacketDirection direction,
                                              Plot* plot) const {
-  for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
-    if (!IsAudioSsrc(parsed_log_, direction, stream.ssrc))
-      continue;
-    TimeSeries time_series(GetStreamName(parsed_log_, direction, stream.ssrc),
-                           LineStyle::kLine);
-    for (auto& packet : stream.packet_view) {
-      if (packet.header.extension.audio_level()) {
-        float x = config_.GetCallTimeSec(packet.log_time());
-        // The audio level is stored in -dBov (so e.g. -10 dBov is stored as 10)
-        // Here we convert it to dBov.
-        float y =
-            static_cast<float>(-packet.header.extension.audio_level()->level());
-        time_series.points.emplace_back(TimeSeriesPoint(x, y));
-      }
-    }
-    plot->AppendTimeSeries(std::move(time_series));
-  }
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetYAxis(-127, 0, "Audio level (dBov)", kBottomMargin, kTopMargin);
-  plot->SetTitle(GetDirectionAsString(direction) + " audio level");
-}
-
-// For each SSRC, plot the sequence number difference between consecutive
-// incoming packets.
-void EventLogAnalyzer::CreateSequenceNumberGraph(Plot* plot) const {
-  for (const auto& stream : parsed_log_.incoming_rtp_packets_by_ssrc()) {
-    // Filter on SSRC.
-    if (!MatchingSsrc(stream.ssrc, desired_ssrc_)) {
-      continue;
-    }
-
-    TimeSeries time_series(
-        GetStreamName(parsed_log_, kIncomingPacket, stream.ssrc),
-        LineStyle::kBar);
-    auto GetSequenceNumberDiff = [](const LoggedRtpPacketIncoming& old_packet,
-                                    const LoggedRtpPacketIncoming& new_packet) {
-      int64_t diff =
-          WrappingDifference(new_packet.rtp.header.sequenceNumber,
-                             old_packet.rtp.header.sequenceNumber, 1ul << 16);
-      return diff;
-    };
-    auto ToCallTime = [this](const LoggedRtpPacketIncoming& packet) {
-      return this->config_.GetCallTimeSec(packet.log_time());
-    };
-    ProcessPairs<LoggedRtpPacketIncoming, float>(
-        ToCallTime, GetSequenceNumberDiff, stream.incoming_packets,
-        &time_series);
-    plot->AppendTimeSeries(std::move(time_series));
-  }
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Difference since last packet", kBottomMargin,
-                          kTopMargin);
-  plot->SetTitle("Incoming sequence number delta");
-}
-
-void EventLogAnalyzer::CreateIncomingPacketLossGraph(Plot* plot) const {
-  for (const auto& stream : parsed_log_.incoming_rtp_packets_by_ssrc()) {
-    const std::vector<LoggedRtpPacketIncoming>& packets =
-        stream.incoming_packets;
-    // Filter on SSRC.
-    if (!MatchingSsrc(stream.ssrc, desired_ssrc_) || packets.empty()) {
-      continue;
-    }
-
-    TimeSeries time_series(
-        GetStreamName(parsed_log_, kIncomingPacket, stream.ssrc),
-        LineStyle::kLine, PointStyle::kHighlight);
-    // TODO(terelius): Should the window and step size be read from the class
-    // instead?
-    const TimeDelta kWindow = TimeDelta::Millis(1000);
-    const TimeDelta kStep = TimeDelta::Millis(1000);
-    SeqNumUnwrapper<uint16_t> unwrapper_;
-    SeqNumUnwrapper<uint16_t> prior_unwrapper_;
-    size_t window_index_begin = 0;
-    size_t window_index_end = 0;
-    uint64_t highest_seq_number =
-        unwrapper_.Unwrap(packets[0].rtp.header.sequenceNumber) - 1;
-    uint64_t highest_prior_seq_number =
-        prior_unwrapper_.Unwrap(packets[0].rtp.header.sequenceNumber) - 1;
-
-    for (Timestamp t = config_.begin_time_; t < config_.end_time_ + kStep;
-         t += kStep) {
-      while (window_index_end < packets.size() &&
-             packets[window_index_end].rtp.log_time() < t) {
-        uint64_t sequence_number = unwrapper_.Unwrap(
-            packets[window_index_end].rtp.header.sequenceNumber);
-        highest_seq_number = std::max(highest_seq_number, sequence_number);
-        ++window_index_end;
-      }
-      while (window_index_begin < packets.size() &&
-             packets[window_index_begin].rtp.log_time() < t - kWindow) {
-        uint64_t sequence_number = prior_unwrapper_.Unwrap(
-            packets[window_index_begin].rtp.header.sequenceNumber);
-        highest_prior_seq_number =
-            std::max(highest_prior_seq_number, sequence_number);
-        ++window_index_begin;
-      }
-      float x = config_.GetCallTimeSec(t);
-      uint64_t expected_packets = highest_seq_number - highest_prior_seq_number;
-      if (expected_packets > 0) {
-        int64_t received_packets = window_index_end - window_index_begin;
-        int64_t lost_packets = expected_packets - received_packets;
-        float y = static_cast<float>(lost_packets) / expected_packets * 100;
-        time_series.points.emplace_back(x, y);
-      }
-    }
-    plot->AppendTimeSeries(std::move(time_series));
-  }
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Loss rate (in %)", kBottomMargin, kTopMargin);
-  plot->SetTitle("Incoming packet loss (derived from incoming packets)");
+  webrtc::CreateAudioLevelGraph(parsed_log_, config_, direction, plot);
 }
 
 void EventLogAnalyzer::CreateIncomingDelayGraph(Plot* plot) const {
-  for (const auto& stream : parsed_log_.incoming_rtp_packets_by_ssrc()) {
-    // Filter on SSRC.
-    if (!MatchingSsrc(stream.ssrc, desired_ssrc_) ||
-        IsRtxSsrc(parsed_log_, kIncomingPacket, stream.ssrc)) {
-      continue;
-    }
-
-    const std::vector<LoggedRtpPacketIncoming>& packets =
-        stream.incoming_packets;
-    if (packets.size() < 100) {
-      RTC_LOG(LS_WARNING) << "Can't estimate the RTP clock frequency with "
-                          << packets.size() << " packets in the stream.";
-      continue;
-    }
-    int64_t segment_end_us = parsed_log_.first_log_segment().stop_time_us();
-    std::optional<uint32_t> estimated_frequency =
-        EstimateRtpClockFrequency(packets, segment_end_us);
-    if (!estimated_frequency)
-      continue;
-    const double frequency_hz = *estimated_frequency;
-    if (IsVideoSsrc(parsed_log_, kIncomingPacket, stream.ssrc) &&
-        frequency_hz != 90000) {
-      RTC_LOG(LS_WARNING)
-          << "Video stream should use a 90 kHz clock but appears to use "
-          << frequency_hz / 1000 << ". Discarding.";
-      continue;
-    }
-
-    auto ToCallTime = [this](const LoggedRtpPacketIncoming& packet) {
-      return this->config_.GetCallTimeSec(packet.log_time());
-    };
-    auto ToNetworkDelay = [frequency_hz](
-                              const LoggedRtpPacketIncoming& old_packet,
-                              const LoggedRtpPacketIncoming& new_packet) {
-      return NetworkDelayDiff_CaptureTime(old_packet, new_packet, frequency_hz);
-    };
-
-    TimeSeries capture_time_data(
-        GetStreamName(parsed_log_, kIncomingPacket, stream.ssrc) +
-            " capture-time",
-        LineStyle::kLine);
-    AccumulatePairs<LoggedRtpPacketIncoming, double>(
-        ToCallTime, ToNetworkDelay, packets, &capture_time_data);
-    plot->AppendTimeSeries(std::move(capture_time_data));
-
-    TimeSeries send_time_data(
-        GetStreamName(parsed_log_, kIncomingPacket, stream.ssrc) +
-            " abs-send-time",
-        LineStyle::kLine);
-    AccumulatePairs<LoggedRtpPacketIncoming, double>(
-        ToCallTime, NetworkDelayDiff_AbsSendTime, packets, &send_time_data);
-    plot->AppendTimeSeriesIfNotEmpty(std::move(send_time_data));
-  }
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Delay (ms)", kBottomMargin, kTopMargin);
-  plot->SetTitle("Incoming network delay (relative to first packet)");
+  webrtc::CreateIncomingDelayGraph(parsed_log_, config_, plot);
 }
 
-// Plot the fraction of packets lost (as perceived by the loss-based BWE).
 void EventLogAnalyzer::CreateFractionLossGraph(Plot* plot) const {
-  TimeSeries time_series("Fraction lost", LineStyle::kLine,
-                         PointStyle::kHighlight);
-  for (auto& bwe_update : parsed_log_.bwe_loss_updates()) {
-    float x = config_.GetCallTimeSec(bwe_update.log_time());
-    float y = static_cast<float>(bwe_update.fraction_lost) / 255 * 100;
-    time_series.points.emplace_back(x, y);
-  }
-
-  plot->AppendTimeSeries(std::move(time_series));
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 10, "Loss rate (in %)", kBottomMargin, kTopMargin);
-  plot->SetTitle("Outgoing packet loss (as reported by BWE)");
+  webrtc::CreateFractionLossGraph(parsed_log_, config_, plot);
 }
 
-// Plot the total bandwidth used by all RTP streams.
 void EventLogAnalyzer::CreateTotalIncomingBitrateGraph(Plot* plot) const {
-  // TODO(terelius): This could be provided by the parser.
-  std::multimap<Timestamp, size_t> packets_in_order;
-  for (const auto& stream : parsed_log_.incoming_rtp_packets_by_ssrc()) {
-    for (const LoggedRtpPacketIncoming& packet : stream.incoming_packets)
-      packets_in_order.insert(
-          std::make_pair(packet.rtp.log_time(), packet.rtp.total_length));
-  }
-
-  auto window_begin = packets_in_order.begin();
-  auto window_end = packets_in_order.begin();
-  size_t bytes_in_window = 0;
-
-  if (!packets_in_order.empty()) {
-    // Calculate a moving average of the bitrate and store in a TimeSeries.
-    TimeSeries bitrate_series("Bitrate", LineStyle::kLine);
-    for (Timestamp time = config_.begin_time_;
-         time < config_.end_time_ + config_.step_; time += config_.step_) {
-      while (window_end != packets_in_order.end() && window_end->first < time) {
-        bytes_in_window += window_end->second;
-        ++window_end;
-      }
-      while (window_begin != packets_in_order.end() &&
-             window_begin->first < time - config_.window_duration_) {
-        RTC_DCHECK_LE(window_begin->second, bytes_in_window);
-        bytes_in_window -= window_begin->second;
-        ++window_begin;
-      }
-      float window_duration_in_seconds =
-          static_cast<float>(config_.window_duration_.us()) /
-          kNumMicrosecsPerSec;
-      float x = config_.GetCallTimeSec(time);
-      float y = bytes_in_window * 8 / window_duration_in_seconds / 1000;
-      bitrate_series.points.emplace_back(x, y);
-    }
-    plot->AppendTimeSeries(std::move(bitrate_series));
-  }
-
-  // Overlay the outgoing REMB over incoming bitrate.
-  TimeSeries remb_series("Remb", LineStyle::kStep);
-  for (const auto& rtcp : parsed_log_.rembs(kOutgoingPacket)) {
-    float x = config_.GetCallTimeSec(rtcp.log_time());
-    float y = static_cast<float>(rtcp.remb.bitrate_bps()) / 1000;
-    remb_series.points.emplace_back(x, y);
-  }
-  plot->AppendTimeSeriesIfNotEmpty(std::move(remb_series));
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin);
-  plot->SetTitle("Incoming RTP bitrate");
+  webrtc::CreateTotalIncomingBitrateGraph(parsed_log_, config_, plot);
 }
 
-// Plot the total bandwidth used by all RTP streams.
 void EventLogAnalyzer::CreateTotalOutgoingBitrateGraph(
     Plot* plot,
     bool show_detector_state,
     bool show_alr_state,
     bool show_link_capacity) const {
-  // TODO(terelius): This could be provided by the parser.
-  std::multimap<Timestamp, size_t> packets_in_order;
-  for (const auto& stream : parsed_log_.outgoing_rtp_packets_by_ssrc()) {
-    for (const LoggedRtpPacketOutgoing& packet : stream.outgoing_packets)
-      packets_in_order.insert(
-          std::make_pair(packet.rtp.log_time(), packet.rtp.total_length));
-  }
-
-  auto window_begin = packets_in_order.begin();
-  auto window_end = packets_in_order.begin();
-  size_t bytes_in_window = 0;
-
-  if (!packets_in_order.empty()) {
-    // Calculate a moving average of the bitrate and store in a TimeSeries.
-    TimeSeries bitrate_series("Bitrate", LineStyle::kLine);
-    for (Timestamp time = config_.begin_time_;
-         time < config_.end_time_ + config_.step_; time += config_.step_) {
-      while (window_end != packets_in_order.end() && window_end->first < time) {
-        bytes_in_window += window_end->second;
-        ++window_end;
-      }
-      while (window_begin != packets_in_order.end() &&
-             window_begin->first < time - config_.window_duration_) {
-        RTC_DCHECK_LE(window_begin->second, bytes_in_window);
-        bytes_in_window -= window_begin->second;
-        ++window_begin;
-      }
-      float window_duration_in_seconds =
-          static_cast<float>(config_.window_duration_.us()) /
-          kNumMicrosecsPerSec;
-      float x = config_.GetCallTimeSec(time);
-      float y = bytes_in_window * 8 / window_duration_in_seconds / 1000;
-      bitrate_series.points.emplace_back(x, y);
-    }
-    plot->AppendTimeSeries(std::move(bitrate_series));
-  }
-
-  // Overlay the send-side bandwidth estimate over the outgoing bitrate.
-  TimeSeries loss_series("Loss-based estimate", LineStyle::kStep);
-  for (auto& loss_update : parsed_log_.bwe_loss_updates()) {
-    float x = config_.GetCallTimeSec(loss_update.log_time());
-    float y = static_cast<float>(loss_update.bitrate_bps) / 1000;
-    loss_series.points.emplace_back(x, y);
-  }
-
-  TimeSeries link_capacity_lower_series("Link-capacity-lower",
-                                        LineStyle::kStep);
-  TimeSeries link_capacity_upper_series("Link-capacity-upper",
-                                        LineStyle::kStep);
-  for (auto& remote_estimate_event : parsed_log_.remote_estimate_events()) {
-    float x = config_.GetCallTimeSec(remote_estimate_event.log_time());
-    if (remote_estimate_event.link_capacity_lower.has_value()) {
-      float link_capacity_lower = static_cast<float>(
-          remote_estimate_event.link_capacity_lower.value().kbps());
-      link_capacity_lower_series.points.emplace_back(x, link_capacity_lower);
-    }
-    if (remote_estimate_event.link_capacity_upper.has_value()) {
-      float link_capacity_upper = static_cast<float>(
-          remote_estimate_event.link_capacity_upper.value().kbps());
-      link_capacity_upper_series.points.emplace_back(x, link_capacity_upper);
-    }
-  }
-
-  TimeSeries delay_series("Delay-based estimate", LineStyle::kStep);
-  IntervalSeries overusing_series("Overusing", "#ff8e82",
-                                  IntervalSeries::kHorizontal);
-  IntervalSeries underusing_series("Underusing", "#5092fc",
-                                   IntervalSeries::kHorizontal);
-  IntervalSeries normal_series("Normal", "#c4ffc4",
-                               IntervalSeries::kHorizontal);
-  IntervalSeries* last_series = &normal_series;
-  float last_detector_switch = 0.0;
-
-  BandwidthUsage last_detector_state = BandwidthUsage::kBwNormal;
-
-  for (auto& delay_update : parsed_log_.bwe_delay_updates()) {
-    float x = config_.GetCallTimeSec(delay_update.log_time());
-    float y = static_cast<float>(delay_update.bitrate_bps) / 1000;
-
-    if (last_detector_state != delay_update.detector_state) {
-      last_series->intervals.emplace_back(last_detector_switch, x);
-      last_detector_state = delay_update.detector_state;
-      last_detector_switch = x;
-
-      switch (delay_update.detector_state) {
-        case BandwidthUsage::kBwNormal:
-          last_series = &normal_series;
-          break;
-        case BandwidthUsage::kBwUnderusing:
-          last_series = &underusing_series;
-          break;
-        case BandwidthUsage::kBwOverusing:
-          last_series = &overusing_series;
-          break;
-        case BandwidthUsage::kLast:
-          RTC_DCHECK_NOTREACHED();
-      }
-    }
-
-    delay_series.points.emplace_back(x, y);
-  }
-
-  RTC_CHECK(last_series);
-  last_series->intervals.emplace_back(last_detector_switch,
-                                      config_.CallEndTimeSec());
-
-  TimeSeries scream_series("Scream target rate", LineStyle::kStep);
-  for (auto& scream_update : parsed_log_.bwe_scream_updates()) {
-    float x = config_.GetCallTimeSec(scream_update.log_time());
-    float y = static_cast<float>(scream_update.target_rate.kbps());
-    scream_series.points.emplace_back(x, y);
-  }
-
-  TimeSeries created_series("Probe cluster created.", LineStyle::kNone,
-                            PointStyle::kHighlight);
-  for (auto& cluster : parsed_log_.bwe_probe_cluster_created_events()) {
-    float x = config_.GetCallTimeSec(cluster.log_time());
-    float y = static_cast<float>(cluster.bitrate_bps) / 1000;
-    created_series.points.emplace_back(x, y);
-  }
-
-  TimeSeries result_series("Probing results.", LineStyle::kNone,
-                           PointStyle::kHighlight);
-  for (auto& result : parsed_log_.bwe_probe_success_events()) {
-    float x = config_.GetCallTimeSec(result.log_time());
-    float y = static_cast<float>(result.bitrate_bps) / 1000;
-    result_series.points.emplace_back(x, y);
-  }
-
-  TimeSeries probe_failures_series("Probe failed", LineStyle::kNone,
-                                   PointStyle::kHighlight);
-  for (auto& failure : parsed_log_.bwe_probe_failure_events()) {
-    float x = config_.GetCallTimeSec(failure.log_time());
-    probe_failures_series.points.emplace_back(x, 0);
-  }
-
-  IntervalSeries alr_state("ALR", "#555555", IntervalSeries::kHorizontal);
-  bool previously_in_alr = false;
-  Timestamp alr_start = Timestamp::Zero();
-  for (auto& alr : parsed_log_.alr_state_events()) {
-    float y = config_.GetCallTimeSec(alr.log_time());
-    if (!previously_in_alr && alr.in_alr) {
-      alr_start = alr.log_time();
-      previously_in_alr = true;
-    } else if (previously_in_alr && !alr.in_alr) {
-      float x = config_.GetCallTimeSec(alr_start);
-      alr_state.intervals.emplace_back(x, y);
-      previously_in_alr = false;
-    }
-  }
-
-  if (previously_in_alr) {
-    float x = config_.GetCallTimeSec(alr_start);
-    float y = config_.GetCallTimeSec(config_.end_time_);
-    alr_state.intervals.emplace_back(x, y);
-  }
-
-  if (show_detector_state) {
-    plot->AppendIntervalSeries(std::move(overusing_series));
-    plot->AppendIntervalSeries(std::move(underusing_series));
-    plot->AppendIntervalSeries(std::move(normal_series));
-  }
-
-  if (show_alr_state) {
-    plot->AppendIntervalSeries(std::move(alr_state));
-  }
-
-  if (show_link_capacity) {
-    plot->AppendTimeSeriesIfNotEmpty(std::move(link_capacity_lower_series));
-    plot->AppendTimeSeriesIfNotEmpty(std::move(link_capacity_upper_series));
-  }
-
-  plot->AppendTimeSeries(std::move(loss_series));
-  plot->AppendTimeSeriesIfNotEmpty(std::move(probe_failures_series));
-  plot->AppendTimeSeries(std::move(delay_series));
-  plot->AppendTimeSeriesIfNotEmpty(std::move(scream_series));
-  plot->AppendTimeSeries(std::move(created_series));
-  plot->AppendTimeSeries(std::move(result_series));
-
-  // Overlay the incoming REMB over the outgoing bitrate.
-  TimeSeries remb_series("Remb", LineStyle::kStep);
-  for (const auto& rtcp : parsed_log_.rembs(kIncomingPacket)) {
-    float x = config_.GetCallTimeSec(rtcp.log_time());
-    float y = static_cast<float>(rtcp.remb.bitrate_bps()) / 1000;
-    remb_series.points.emplace_back(x, y);
-  }
-  plot->AppendTimeSeriesIfNotEmpty(std::move(remb_series));
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin);
-  plot->SetTitle("Outgoing RTP bitrate");
-}
-
-// For each SSRC, plot the bandwidth used by that stream.
-void EventLogAnalyzer::CreateStreamBitrateGraph(PacketDirection direction,
-                                                Plot* plot) const {
-  for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
-    // Filter on SSRC.
-    if (!MatchingSsrc(stream.ssrc, desired_ssrc_)) {
-      continue;
-    }
-
-    TimeSeries time_series(GetStreamName(parsed_log_, direction, stream.ssrc),
-                           LineStyle::kLine);
-    auto GetPacketSizeKilobits = [](const LoggedRtpPacket& packet) {
-      return packet.total_length * 8.0 / 1000.0;
-    };
-    MovingAverage<LoggedRtpPacket, double>(
-        GetPacketSizeKilobits, stream.packet_view, config_, &time_series);
-    plot->AppendTimeSeries(std::move(time_series));
-  }
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin);
-  plot->SetTitle(GetDirectionAsString(direction) + " bitrate per stream");
-}
-
-// Plot the bitrate allocation for each temporal and spatial layer.
-// Computed from RTCP XR target bitrate block, so the graph is only populated if
-// those are sent.
-void EventLogAnalyzer::CreateBitrateAllocationGraph(PacketDirection direction,
-                                                    Plot* plot) const {
-  std::map<LayerDescription, TimeSeries> time_series;
-  const auto& xr_list = parsed_log_.extended_reports(direction);
-  for (const auto& rtcp : xr_list) {
-    const std::optional<rtcp::TargetBitrate>& target_bitrate =
-        rtcp.xr.target_bitrate();
-    if (!target_bitrate.has_value())
-      continue;
-    for (const auto& bitrate_item : target_bitrate->GetTargetBitrates()) {
-      LayerDescription layer(rtcp.xr.sender_ssrc(), bitrate_item.spatial_layer,
-                             bitrate_item.temporal_layer);
-      auto time_series_it = time_series.find(layer);
-      if (time_series_it == time_series.end()) {
-        std::string layer_name = GetLayerName(layer);
-        bool inserted;
-        std::tie(time_series_it, inserted) = time_series.insert(
-            std::make_pair(layer, TimeSeries(layer_name, LineStyle::kStep)));
-        RTC_DCHECK(inserted);
-      }
-      float x = config_.GetCallTimeSec(rtcp.log_time());
-      float y = bitrate_item.target_bitrate_kbps;
-      time_series_it->second.points.emplace_back(x, y);
-    }
-  }
-  for (auto& layer : time_series) {
-    plot->AppendTimeSeries(std::move(layer.second));
-  }
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin);
-  if (direction == kIncomingPacket)
-    plot->SetTitle("Target bitrate per incoming layer");
-  else
-    plot->SetTitle("Target bitrate per outgoing layer");
+  webrtc::CreateTotalOutgoingBitrateGraph(parsed_log_, config_, plot,
+                                          show_detector_state, show_alr_state,
+                                          show_link_capacity);
 }
 
 void EventLogAnalyzer::CreateGoogCcSimulationGraph(Plot* plot) const {
-  TimeSeries target_rates("Simulated target rate", LineStyle::kStep,
-                          PointStyle::kHighlight);
-  TimeSeries delay_based("Logged delay-based estimate", LineStyle::kStep,
-                         PointStyle::kHighlight);
-  TimeSeries loss_based("Logged loss-based estimate", LineStyle::kStep,
-                        PointStyle::kHighlight);
-  TimeSeries probe_results("Logged probe success", LineStyle::kNone,
-                           PointStyle::kHighlight);
-
-  LogBasedNetworkControllerSimulation simulation(
-      env_, std::make_unique<GoogCcNetworkControllerFactory>(),
-      [&](const NetworkControlUpdate& update, Timestamp at_time) {
-        if (update.target_rate) {
-          target_rates.points.emplace_back(
-              config_.GetCallTimeSec(at_time),
-              update.target_rate->target_rate.kbps<float>());
-        }
-      });
-
-  simulation.ProcessEventsInLog(parsed_log_);
-  for (const auto& logged : parsed_log_.bwe_delay_updates())
-    delay_based.points.emplace_back(config_.GetCallTimeSec(logged.log_time()),
-                                    logged.bitrate_bps / 1000);
-  for (const auto& logged : parsed_log_.bwe_probe_success_events())
-    probe_results.points.emplace_back(config_.GetCallTimeSec(logged.log_time()),
-                                      logged.bitrate_bps / 1000);
-  for (const auto& logged : parsed_log_.bwe_loss_updates())
-    loss_based.points.emplace_back(config_.GetCallTimeSec(logged.log_time()),
-                                   logged.bitrate_bps / 1000);
-
-  plot->AppendTimeSeries(std::move(delay_based));
-  plot->AppendTimeSeries(std::move(loss_based));
-  plot->AppendTimeSeries(std::move(probe_results));
-  plot->AppendTimeSeries(std::move(target_rates));
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 10, "Bitrate (kbps)", kBottomMargin, kTopMargin);
-  plot->SetTitle("Simulated BWE behavior");
+  webrtc::CreateGoogCcSimulationGraph(parsed_log_, config_, plot);
 }
 
 void EventLogAnalyzer::CreateScreamSimulationBitrateGraph(Plot* plot) const {
-  TimeSeries target_rate_series("Target rate", LineStyle::kStep);
-  TimeSeries pacing_rate_series("Pacing rate", LineStyle::kStep);
-  TimeSeries send_rate_series("Send rate", LineStyle::kStep);
-
-  LogScreamSimulation simulation({.rate_window = config_.window_duration_},
-                                 env_);
-  simulation.ProcessEventsInLog(parsed_log_);
-
-  for (const LogScreamSimulation::State& state : simulation.updates()) {
-    target_rate_series.points.emplace_back(config_.GetCallTimeSec(state.time),
-                                           state.target_rate.bps() / 1000);
-    pacing_rate_series.points.emplace_back(config_.GetCallTimeSec(state.time),
-                                           state.pacing_rate.bps() / 1000);
-    send_rate_series.points.emplace_back(config_.GetCallTimeSec(state.time),
-                                         state.send_rate.bps() / 1000);
-  }
-  plot->AppendTimeSeries(std::move(target_rate_series));
-  plot->AppendTimeSeries(std::move(pacing_rate_series));
-  plot->AppendTimeSeries(std::move(send_rate_series));
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 100, "Kbps", kBottomMargin, kTopMargin);
-  plot->SetTitle("Simulated Scream rates");
+  webrtc::CreateScreamSimulationBitrateGraph(parsed_log_, config_, plot);
 }
 
 void EventLogAnalyzer::CreateScreamSimulationRefWindowGraph(Plot* plot) const {
-  using SendWindowUsage = LogScreamSimulation::State::SendWindowUsage;
-  TimeSeries ref_window_series("RefWindow", LineStyle::kStep);
-  TimeSeries ref_window_i_series("RefWindowI", LineStyle::kStep);
-  TimeSeries max_data_in_flight("Max allowed data in flight", LineStyle::kStep);
-  TimeSeries data_in_flight("Data in flight", LineStyle::kStep);
-  IntervalSeries send_window_above_max_series(
-      "Data in flight > Max allowed", "#ff8e82", IntervalSeries::kHorizontal);
-  IntervalSeries send_window_below_ref_window_series(
-      "Data in flight < RefWindow", "#c5dff2", IntervalSeries::kHorizontal);
-  IntervalSeries send_window_above_ref_window_series(
-      "Data in flight >= RefWindow", "#b9fad8", IntervalSeries::kHorizontal);
-  IntervalSeries* last_series = &send_window_below_ref_window_series;
-
-  LogScreamSimulation simulation({.rate_window = config_.window_duration_},
-                                 env_);
-  simulation.ProcessEventsInLog(parsed_log_);
-  if (simulation.updates().empty()) {
-    RTC_LOG(LS_ERROR) << "Empty simulation.";
-    return;
-  }
-
-  float send_window_state_switch =
-      config_.GetCallTimeSec(simulation.updates().front().time);
-  SendWindowUsage send_window_usage = SendWindowUsage::kBelowRefWindow;
-
-  for (const LogScreamSimulation::State& state : simulation.updates()) {
-    ref_window_series.points.emplace_back(config_.GetCallTimeSec(state.time),
-                                          state.ref_window.bytes());
-    ref_window_i_series.points.emplace_back(config_.GetCallTimeSec(state.time),
-                                            state.ref_window_i.bytes());
-    max_data_in_flight.points.emplace_back(config_.GetCallTimeSec(state.time),
-                                           state.max_data_in_flight.bytes());
-    data_in_flight.points.emplace_back(config_.GetCallTimeSec(state.time),
-                                       state.data_in_flight.bytes());
-    if (state.send_window_usage != send_window_usage) {
-      last_series->intervals.emplace_back(send_window_state_switch,
-                                          config_.GetCallTimeSec(state.time));
-      send_window_usage = state.send_window_usage;
-      send_window_state_switch = config_.GetCallTimeSec(state.time);
-      switch (send_window_usage) {
-        case SendWindowUsage::kAboveRefWindow:
-          last_series = &send_window_above_ref_window_series;
-          break;
-        case SendWindowUsage::kBelowRefWindow:
-          last_series = &send_window_below_ref_window_series;
-          break;
-        case SendWindowUsage::kAboveScreamMax:
-          last_series = &send_window_above_max_series;
-          break;
-      }
-    }
-  }
-  last_series->intervals.emplace_back(send_window_state_switch,
-                                      config_.CallEndTimeSec());
-  plot->AppendTimeSeries(std::move(ref_window_series));
-  plot->AppendTimeSeries(std::move(ref_window_i_series));
-  plot->AppendTimeSeries(std::move(max_data_in_flight));
-  plot->AppendTimeSeries(std::move(data_in_flight));
-  plot->AppendIntervalSeries(std::move(send_window_above_max_series));
-  plot->AppendIntervalSeries(std::move(send_window_below_ref_window_series));
-  plot->AppendIntervalSeries(std::move(send_window_above_ref_window_series));
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 10, "Bytes", kBottomMargin, kTopMargin);
-  plot->SetTitle("Simulated Scream RefWindow");
+  webrtc::CreateScreamSimulationRefWindowGraph(parsed_log_, config_, plot);
 }
 
 void EventLogAnalyzer::CreateScreamSimulationRatiosGraph(Plot* plot) const {
-  TimeSeries queue_delay_dev_norm_series("QueueDelayDevNorm", LineStyle::kStep);
-  TimeSeries l4s_alpha_series("L4sAlpha", LineStyle::kStep);
-  TimeSeries l4s_alpha_v_series("L4sAlphaV", LineStyle::kStep);
-  TimeSeries ref_window_scale_factor_due_to_increased_delay(
-      "RefWindowScaleFactorDueToIncreasedDelay", LineStyle::kStep);
-  TimeSeries ref_window_scale_factor_due_to_delay_variation(
-      "RefWindowScaleFactorDueToDelayVariation", LineStyle::kStep);
-
-  LogScreamSimulation simulation({.rate_window = config_.window_duration_},
-                                 env_);
-  simulation.ProcessEventsInLog(parsed_log_);
-
-  for (const LogScreamSimulation::State& state : simulation.updates()) {
-    queue_delay_dev_norm_series.points.emplace_back(
-        config_.GetCallTimeSec(state.time), state.queue_delay_dev_norm);
-    l4s_alpha_series.points.emplace_back(config_.GetCallTimeSec(state.time),
-                                         state.l4s_alpha);
-    l4s_alpha_v_series.points.emplace_back(config_.GetCallTimeSec(state.time),
-                                           state.l4s_alpha_v);
-    ref_window_scale_factor_due_to_increased_delay.points.emplace_back(
-        config_.GetCallTimeSec(state.time),
-        state.ref_window_scale_factor_due_to_increased_delay);
-    ref_window_scale_factor_due_to_delay_variation.points.emplace_back(
-        config_.GetCallTimeSec(state.time),
-        state.ref_window_scale_factor_due_to_delay_variation);
-  }
-  plot->AppendTimeSeries(std::move(queue_delay_dev_norm_series));
-  plot->AppendTimeSeries(std::move(l4s_alpha_series));
-  plot->AppendTimeSeries(std::move(l4s_alpha_v_series));
-  plot->AppendTimeSeries(
-      std::move(ref_window_scale_factor_due_to_increased_delay));
-  plot->AppendTimeSeries(
-      std::move(ref_window_scale_factor_due_to_delay_variation));
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Ratios", kBottomMargin, kTopMargin);
-  plot->SetTitle("Simulated Scream Ratios");
-}
-
-void EventLogAnalyzer::CreateOutgoingEcnFeedbackGraph(Plot* plot) const {
-  CreateEcnFeedbackGraph(plot, kOutgoingPacket);
-  plot->SetTitle("Outgoing ECN count per feedback");
-}
-
-void EventLogAnalyzer::CreateIncomingEcnFeedbackGraph(Plot* plot) const {
-  CreateEcnFeedbackGraph(plot, kIncomingPacket);
-  plot->SetTitle("Incoming ECN count per feedback");
+  webrtc::CreateScreamSimulationRatiosGraph(parsed_log_, config_, plot);
 }
 
 void EventLogAnalyzer::CreateScreamRefWindowGraph(Plot* plot) const {
-  TimeSeries ref_window_series("RefWindow", LineStyle::kStep);
-  for (auto& scream_update : parsed_log_.bwe_scream_updates()) {
-    float x = config_.GetCallTimeSec(scream_update.log_time());
-    float y = static_cast<float>(scream_update.ref_window.bytes());
-    ref_window_series.points.emplace_back(x, y);
-  }
-  plot->AppendTimeSeries(std::move(ref_window_series));
-
-  TimeSeries data_in_flight_series("Data in flight", LineStyle::kLine);
-  for (auto& scream_update : parsed_log_.bwe_scream_updates()) {
-    float x = config_.GetCallTimeSec(scream_update.log_time());
-    float y = static_cast<float>(scream_update.data_in_flight.bytes());
-    data_in_flight_series.points.emplace_back(x, y);
-  }
-  plot->AppendTimeSeries(std::move(data_in_flight_series));
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 3000, "Bytes", kBottomMargin, kTopMargin);
-  plot->SetTitle("Scream Ref Window");
+  webrtc::CreateScreamRefWindowGraph(parsed_log_, config_, plot);
 }
 
 void EventLogAnalyzer::CreateScreamDelayEstimateGraph(Plot* plot) const {
-  TimeSeries smoothed_rtt_series("Smoothed RTT", LineStyle::kStep);
-  TimeSeries avg_queue_delay_series("Avg queue delay", LineStyle::kStep);
-
-  for (auto& scream_update : parsed_log_.bwe_scream_updates()) {
-    float x = config_.GetCallTimeSec(scream_update.log_time());
-    float smoothed_rtt_ms = static_cast<float>(scream_update.smoothed_rtt.ms());
-    smoothed_rtt_series.points.emplace_back(x, smoothed_rtt_ms);
-    float avg_queue_delay_ms =
-        static_cast<float>(scream_update.avg_queue_delay.ms());
-    avg_queue_delay_series.points.emplace_back(x, avg_queue_delay_ms);
-  }
-
-  plot->AppendTimeSeries(std::move(smoothed_rtt_series));
-  plot->AppendTimeSeries(std::move(avg_queue_delay_series));
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 50, "Delay (ms)", kBottomMargin, kTopMargin);
-  plot->SetTitle("Scream delay estimates");
-}
-
-void EventLogAnalyzer::CreateEcnFeedbackGraph(Plot* plot,
-                                              PacketDirection direction) const {
-  TimeSeries not_ect("Not ECN capable", LineStyle::kBar,
-                     PointStyle::kHighlight);
-  TimeSeries ect_1("ECN capable", LineStyle::kBar, PointStyle::kHighlight);
-  TimeSeries ce("Congestion experienced", LineStyle::kBar,
-                PointStyle::kHighlight);
-
-  for (const LoggedRtcpCongestionControlFeedback& feedback :
-       parsed_log_.congestion_feedback(direction)) {
-    int ect_1_count = 0;
-    int not_ect_count = 0;
-    int ce_count = 0;
-
-    for (const rtcp::CongestionControlFeedback::PacketInfo& info :
-         feedback.congestion_feedback.packets()) {
-      switch (info.ecn) {
-        case EcnMarking::kNotEct:
-          ++not_ect_count;
-          break;
-        case EcnMarking::kEct1:
-          ++ect_1_count;
-          break;
-        case EcnMarking::kEct0:
-          RTC_LOG(LS_ERROR) << "unexpected ect(0)";
-          break;
-        case EcnMarking::kCe:
-          ++ce_count;
-          break;
-      }
-    }
-    ect_1.points.emplace_back(config_.GetCallTimeSec(feedback.timestamp),
-                              ect_1_count);
-    not_ect.points.emplace_back(config_.GetCallTimeSec(feedback.timestamp),
-                                not_ect_count);
-    ce.points.emplace_back(config_.GetCallTimeSec(feedback.timestamp),
-                           ce_count);
-  }
-
-  plot->AppendTimeSeriesIfNotEmpty(std::move(ect_1));
-  plot->AppendTimeSeriesIfNotEmpty(std::move(not_ect));
-  plot->AppendTimeSeriesIfNotEmpty(std::move(ce));
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 10, "Count per feedback", kBottomMargin,
-                          kTopMargin);
-}
-
-void EventLogAnalyzer::CreateOutgoingLossRateGraph(Plot* plot) const {
-  struct PacketLossPerFeedback {
-    Timestamp timestamp;              // Time when this feedback was received.
-    int num_packets_in_feedback = 0;  // Includes lost packets.
-    int num_lost_packets = 0;         // In this specific feedback.
-    int num_reordered_packets = 0;    // Packets received in this feedback, but
-                                      // was previously reported as lost.
-    int num_missing_feedback =
-        0;  // Packets missing feedback between this report and the previous.
-  };
-
-  class LossFeedbackBuilder {
-   public:
-    void AddPacket(uint16_t sequence_number, TimeDelta arrival_time_delta) {
-      last_unwrapped_sequence_number_ =
-          sequence_number_unwrapper_.Unwrap(sequence_number);
-      if (!first_sequence_number_.has_value()) {
-        first_sequence_number_ = last_unwrapped_sequence_number_;
-      }
-      ++num_packets_;
-      if (arrival_time_delta.IsInfinite()) {
-        lost_sequence_numbers_.insert(last_unwrapped_sequence_number_);
-      } else {
-        num_reordered_packets_ += previous_lost_sequence_numbers_.count(
-            last_unwrapped_sequence_number_);
-      }
-    }
-
-    void Update(PacketLossPerFeedback& feedback) {
-      feedback.num_packets_in_feedback += num_packets_;
-      feedback.num_lost_packets += lost_sequence_numbers_.size();
-      feedback.num_reordered_packets += num_reordered_packets_;
-      if (first_sequence_number_.has_value() &&
-          previous_feedback_highest_seq_number_.has_value()) {
-        feedback.num_missing_feedback +=
-            *first_sequence_number_ - *previous_feedback_highest_seq_number_ -
-            1;
-      }
-
-      // Prepare for next feedback.
-      first_sequence_number_ = std::nullopt;
-      previous_lost_sequence_numbers_.insert(lost_sequence_numbers_.begin(),
-                                             lost_sequence_numbers_.end());
-      previous_feedback_highest_seq_number_ = last_unwrapped_sequence_number_;
-      lost_sequence_numbers_.clear();
-      num_reordered_packets_ = 0;
-      num_packets_ = 0;
-    }
-
-   private:
-    int64_t last_unwrapped_sequence_number_ = 0;
-    int num_reordered_packets_ = 0;
-    int num_packets_ = 0;
-    std::optional<int64_t> first_sequence_number_;
-
-    std::unordered_set<int64_t> lost_sequence_numbers_;
-    std::unordered_set<int64_t> previous_lost_sequence_numbers_;
-    std::optional<int64_t> previous_feedback_highest_seq_number_;
-
-    RtpSequenceNumberUnwrapper sequence_number_unwrapper_;
-  };
-
-  TimeSeries loss_rate_series("Loss rate (from packet feedback)",
-                              LineStyle::kLine, PointStyle::kHighlight);
-  TimeSeries reordered_packets_between_feedback(
-      "Ratio of reordered packets from last feedback", LineStyle::kLine,
-      PointStyle::kHighlight);
-  TimeSeries average_loss_rate_series("Average loss rate last 5s",
-                                      LineStyle::kLine, PointStyle::kHighlight);
-  TimeSeries missing_feedback_series("Missing feedback", LineStyle::kNone,
-                                     PointStyle::kHighlight);
-
-  std::vector<PacketLossPerFeedback> loss_per_feedback;
-
-  if (!parsed_log_.congestion_feedback(kIncomingPacket).empty()) {
-    plot->SetTitle("Outgoing loss rate (from CCFB)");
-
-    std::map</*ssrc*/ uint32_t, LossFeedbackBuilder> per_ssrc_builder;
-    for (const LoggedRtcpCongestionControlFeedback& feedback :
-         parsed_log_.congestion_feedback(kIncomingPacket)) {
-      const rtcp::CongestionControlFeedback& transport_feedback =
-          feedback.congestion_feedback;
-
-      PacketLossPerFeedback packet_loss_per_feedback = {
-          .timestamp = feedback.log_time()};
-      for (const rtcp::CongestionControlFeedback::PacketInfo& packet :
-           transport_feedback.packets()) {
-        per_ssrc_builder[packet.ssrc].AddPacket(packet.sequence_number,
-                                                packet.arrival_time_offset);
-      }
-      for (auto& [ssrc, builder] : per_ssrc_builder) {
-        builder.Update(packet_loss_per_feedback);
-      }
-      loss_per_feedback.push_back(packet_loss_per_feedback);
-    }
-  } else if (!parsed_log_.transport_feedbacks(kIncomingPacket).empty()) {
-    plot->SetTitle("Outgoing loss rate (from TWCC)");
-
-    LossFeedbackBuilder builder;
-    for (const LoggedRtcpPacketTransportFeedback& feedback :
-         parsed_log_.transport_feedbacks(kIncomingPacket)) {
-      feedback.transport_feedback.ForAllPackets(
-          [&](uint16_t sequence_number, TimeDelta receive_time_delta) {
-            builder.AddPacket(sequence_number, receive_time_delta);
-          });
-      PacketLossPerFeedback packet_loss_per_feedback = {
-          .timestamp = feedback.log_time()};
-      builder.Update(packet_loss_per_feedback);
-      loss_per_feedback.push_back(packet_loss_per_feedback);
-    }
-  }
-
-  PacketLossSummary window_summary;
-  Timestamp last_observation_receive_time = Timestamp::Zero();
-
-  // Use loss based bwe 2 observation duration and observation window size.
-  constexpr TimeDelta kObservationDuration = TimeDelta::Millis(250);
-  constexpr uint32_t kObservationWindowSize = 20;
-  std::deque<PacketLossSummary> observations;
-  int previous_feedback_size = 0;
-  for (const PacketLossPerFeedback& feedback : loss_per_feedback) {
-    for (int64_t num = 0; num < feedback.num_missing_feedback; ++num) {
-      missing_feedback_series.points.emplace_back(
-          config_.GetCallTimeSec(feedback.timestamp), 100 + num);
-    }
-
-    // Compute loss rate from the transport feedback.
-    float loss_rate = static_cast<float>(feedback.num_lost_packets * 100.0 /
-                                         feedback.num_packets_in_feedback);
-
-    loss_rate_series.points.emplace_back(
-        config_.GetCallTimeSec(feedback.timestamp), loss_rate);
-    float reordered_rate =
-        previous_feedback_size == 0
-            ? 0
-            : static_cast<float>(feedback.num_reordered_packets * 100.0 /
-                                 previous_feedback_size);
-    previous_feedback_size = feedback.num_packets_in_feedback;
-    reordered_packets_between_feedback.points.emplace_back(
-        config_.GetCallTimeSec(feedback.timestamp), reordered_rate);
-
-    // Compute loss rate in a window of kObservationWindowSize.
-    if (window_summary.num_packets == 0) {
-      window_summary.base_time = feedback.timestamp;
-    }
-    window_summary.num_packets += feedback.num_packets_in_feedback;
-    window_summary.num_lost_packets +=
-        feedback.num_lost_packets - feedback.num_reordered_packets;
-
-    const Timestamp last_received_time = feedback.timestamp;
-    const TimeDelta observation_duration =
-        window_summary.base_time == Timestamp::Zero()
-            ? TimeDelta::Zero()
-            : last_received_time - window_summary.base_time;
-    if (observation_duration > kObservationDuration) {
-      last_observation_receive_time = last_received_time;
-      observations.push_back(window_summary);
-      if (observations.size() > kObservationWindowSize) {
-        observations.pop_front();
-      }
-
-      // Compute average loss rate in a number of windows.
-      int total_packets = 0;
-      int total_loss = 0;
-      for (const auto& observation : observations) {
-        total_loss += observation.num_lost_packets;
-        total_packets += observation.num_packets;
-      }
-      if (total_packets > 0) {
-        float average_loss_rate = total_loss * 100.0 / total_packets;
-        average_loss_rate_series.points.emplace_back(
-            config_.GetCallTimeSec(feedback.timestamp), average_loss_rate);
-      } else {
-        average_loss_rate_series.points.emplace_back(
-            config_.GetCallTimeSec(feedback.timestamp), 0);
-      }
-      window_summary = PacketLossSummary();
-    }
-  }
-  // Add the data set to the plot.
-  plot->AppendTimeSeriesIfNotEmpty(std::move(loss_rate_series));
-  plot->AppendTimeSeriesIfNotEmpty(
-      std::move(reordered_packets_between_feedback));
-  plot->AppendTimeSeriesIfNotEmpty(std::move(average_loss_rate_series));
-  plot->AppendTimeSeriesIfNotEmpty(std::move(missing_feedback_series));
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 100, "Loss rate (percent)", kBottomMargin,
-                          kTopMargin);
+  webrtc::CreateScreamDelayEstimateGraph(parsed_log_, config_, plot);
 }
 
 void EventLogAnalyzer::CreateSendSideBweSimulationGraph(Plot* plot) const {
-  using RtpPacketType = LoggedRtpPacketOutgoing;
-  using TransportFeedbackType = LoggedRtcpPacketTransportFeedback;
-
-  // TODO(terelius): This could be provided by the parser.
-  std::multimap<int64_t, const RtpPacketType*> outgoing_rtp;
-  for (const auto& stream : parsed_log_.outgoing_rtp_packets_by_ssrc()) {
-    for (const RtpPacketType& rtp_packet : stream.outgoing_packets)
-      outgoing_rtp.insert(
-          std::make_pair(rtp_packet.rtp.log_time_us(), &rtp_packet));
-  }
-
-  const std::vector<TransportFeedbackType>& incoming_rtcp =
-      parsed_log_.transport_feedbacks(kIncomingPacket);
-
-  SimulatedClock clock(0);
-  BitrateObserver observer;
-  TransportFeedbackAdapter transport_feedback;
-  auto factory = GoogCcNetworkControllerFactory();
-  TimeDelta process_interval = factory.GetProcessInterval();
-  // TODO(holmer): Log the call config and use that here instead.
-  static const uint32_t kDefaultStartBitrateBps = 300000;
-  NetworkControllerConfig cc_config(env_);
-  cc_config.constraints.at_time = clock.CurrentTime();
-  cc_config.constraints.starting_rate =
-      DataRate::BitsPerSec(kDefaultStartBitrateBps);
-  auto goog_cc = factory.Create(cc_config);
-
-  TimeSeries time_series("Delay-based estimate", LineStyle::kStep,
-                         PointStyle::kHighlight);
-  TimeSeries acked_time_series("Raw acked bitrate", LineStyle::kLine,
-                               PointStyle::kHighlight);
-  TimeSeries robust_time_series("Robust throughput estimate", LineStyle::kLine,
-                                PointStyle::kHighlight);
-  TimeSeries acked_estimate_time_series("Ackednowledged bitrate estimate",
-                                        LineStyle::kLine,
-                                        PointStyle::kHighlight);
-
-  auto rtp_iterator = outgoing_rtp.begin();
-  auto rtcp_iterator = incoming_rtcp.begin();
-
-  auto NextRtpTime = [&]() {
-    if (rtp_iterator != outgoing_rtp.end())
-      return static_cast<int64_t>(rtp_iterator->first);
-    return std::numeric_limits<int64_t>::max();
-  };
-
-  auto NextRtcpTime = [&]() {
-    if (rtcp_iterator != incoming_rtcp.end())
-      return static_cast<int64_t>(rtcp_iterator->log_time_us());
-    return std::numeric_limits<int64_t>::max();
-  };
-  int64_t next_process_time_us_ = std::min({NextRtpTime(), NextRtcpTime()});
-
-  auto NextProcessTime = [&]() {
-    if (rtcp_iterator != incoming_rtcp.end() ||
-        rtp_iterator != outgoing_rtp.end()) {
-      return next_process_time_us_;
-    }
-    return std::numeric_limits<int64_t>::max();
-  };
-
-  RateStatistics raw_acked_bitrate(750, 8000);
-  FieldTrials throughput_config(
-      "WebRTC-Bwe-RobustThroughputEstimatorSettings/enabled:true/");
-  std::unique_ptr<AcknowledgedBitrateEstimatorInterface>
-      robust_throughput_estimator(
-          AcknowledgedBitrateEstimatorInterface::Create(&throughput_config));
-  FieldTrials acked_bitrate_config(
-      "WebRTC-Bwe-RobustThroughputEstimatorSettings/enabled:false/");
-  std::unique_ptr<AcknowledgedBitrateEstimatorInterface>
-      acknowledged_bitrate_estimator(
-          AcknowledgedBitrateEstimatorInterface::Create(&acked_bitrate_config));
-  int64_t time_us =
-      std::min({NextRtpTime(), NextRtcpTime(), NextProcessTime()});
-  int64_t last_update_us = 0;
-  while (time_us != std::numeric_limits<int64_t>::max()) {
-    clock.AdvanceTimeMicroseconds(time_us - clock.TimeInMicroseconds());
-    if (clock.TimeInMicroseconds() >= NextRtpTime()) {
-      RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtpTime());
-      const RtpPacketType& rtp_packet = *rtp_iterator->second;
-      if (rtp_packet.rtp.header.extension.hasTransportSequenceNumber) {
-        RtpPacketToSend send_packet(/*extensions=*/nullptr);
-        send_packet.set_transport_sequence_number(
-            rtp_packet.rtp.header.extension.transportSequenceNumber);
-        send_packet.SetSsrc(rtp_packet.rtp.header.ssrc);
-        send_packet.SetSequenceNumber(rtp_packet.rtp.header.sequenceNumber);
-        send_packet.SetPayloadSize(rtp_packet.rtp.total_length -
-                                   send_packet.headers_size());
-        RTC_DCHECK_EQ(send_packet.size(), rtp_packet.rtp.total_length);
-        if (IsRtxSsrc(parsed_log_, PacketDirection::kOutgoingPacket,
-                      rtp_packet.rtp.header.ssrc)) {
-          // Don't set the optional media type as we don't know if it is
-          // a retransmission, FEC or padding.
-        } else if (IsVideoSsrc(parsed_log_, PacketDirection::kOutgoingPacket,
-                               rtp_packet.rtp.header.ssrc)) {
-          send_packet.set_packet_type(RtpPacketMediaType::kVideo);
-        } else if (IsAudioSsrc(parsed_log_, PacketDirection::kOutgoingPacket,
-                               rtp_packet.rtp.header.ssrc)) {
-          send_packet.set_packet_type(RtpPacketMediaType::kAudio);
-        }
-        transport_feedback.AddPacket(
-            send_packet, PacedPacketInfo(),
-            0u,  // Per packet overhead bytes.,
-            Timestamp::Micros(rtp_packet.rtp.log_time_us()));
-      }
-      SentPacketInfo sent_packet;
-      sent_packet.send_time_ms = rtp_packet.rtp.log_time_ms();
-      sent_packet.info.included_in_allocation = true;
-      sent_packet.info.packet_size_bytes = rtp_packet.rtp.total_length;
-      if (rtp_packet.rtp.header.extension.hasTransportSequenceNumber) {
-        sent_packet.packet_id =
-            rtp_packet.rtp.header.extension.transportSequenceNumber;
-        sent_packet.info.included_in_feedback = true;
-      }
-      auto sent_msg = transport_feedback.ProcessSentPacket(sent_packet);
-      if (sent_msg)
-        observer.Update(goog_cc->OnSentPacket(*sent_msg));
-      ++rtp_iterator;
-    }
-    if (clock.TimeInMicroseconds() >= NextRtcpTime()) {
-      RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtcpTime());
-
-      auto feedback_msg = transport_feedback.ProcessTransportFeedback(
-          rtcp_iterator->transport_feedback, clock.CurrentTime());
-      if (feedback_msg) {
-        observer.Update(goog_cc->OnTransportPacketsFeedback(*feedback_msg));
-        std::vector<PacketResult> feedback =
-            feedback_msg->SortedByReceiveTime();
-        if (!feedback.empty()) {
-          acknowledged_bitrate_estimator->IncomingPacketFeedbackVector(
-              feedback);
-          robust_throughput_estimator->IncomingPacketFeedbackVector(feedback);
-          for (const PacketResult& packet : feedback) {
-            raw_acked_bitrate.Update(packet.sent_packet.size.bytes(),
-                                     packet.receive_time.ms());
-          }
-          std::optional<uint32_t> raw_bitrate_bps =
-              raw_acked_bitrate.Rate(feedback.back().receive_time.ms());
-          float x = config_.GetCallTimeSec(clock.CurrentTime());
-          if (raw_bitrate_bps) {
-            float y = raw_bitrate_bps.value() / 1000;
-            acked_time_series.points.emplace_back(x, y);
-          }
-          std::optional<DataRate> robust_estimate =
-              robust_throughput_estimator->bitrate();
-          if (robust_estimate) {
-            float y = robust_estimate.value().kbps();
-            robust_time_series.points.emplace_back(x, y);
-          }
-          std::optional<DataRate> acked_estimate =
-              acknowledged_bitrate_estimator->bitrate();
-          if (acked_estimate) {
-            float y = acked_estimate.value().kbps();
-            acked_estimate_time_series.points.emplace_back(x, y);
-          }
-        }
-      }
-      ++rtcp_iterator;
-    }
-    if (clock.TimeInMicroseconds() >= NextProcessTime()) {
-      RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextProcessTime());
-      ProcessInterval msg;
-      msg.at_time = clock.CurrentTime();
-      observer.Update(goog_cc->OnProcessInterval(msg));
-      next_process_time_us_ += process_interval.us();
-    }
-    if (observer.GetAndResetBitrateUpdated() ||
-        time_us - last_update_us >= 1e6) {
-      uint32_t y = observer.last_bitrate_bps() / 1000;
-      float x = config_.GetCallTimeSec(clock.CurrentTime());
-      time_series.points.emplace_back(x, y);
-      last_update_us = time_us;
-    }
-    time_us = std::min({NextRtpTime(), NextRtcpTime(), NextProcessTime()});
-  }
-  // Add the data set to the plot.
-  plot->AppendTimeSeries(std::move(time_series));
-  plot->AppendTimeSeries(std::move(robust_time_series));
-  plot->AppendTimeSeries(std::move(acked_time_series));
-  plot->AppendTimeSeriesIfNotEmpty(std::move(acked_estimate_time_series));
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 10, "Bitrate (kbps)", kBottomMargin, kTopMargin);
-  plot->SetTitle("Simulated send-side BWE behavior");
+  webrtc::CreateSendSideBweSimulationGraph(parsed_log_, config_, plot);
 }
 
 void EventLogAnalyzer::CreateReceiveSideBweSimulationGraph(Plot* plot) const {
-  using RtpPacketType = LoggedRtpPacketIncoming;
-  class RembInterceptor {
-   public:
-    void SendRemb(uint32_t bitrate_bps, std::vector<uint32_t> ssrcs) {
-      last_bitrate_bps_ = bitrate_bps;
-      bitrate_updated_ = true;
-    }
-    uint32_t last_bitrate_bps() const { return last_bitrate_bps_; }
-    bool GetAndResetBitrateUpdated() {
-      bool bitrate_updated = bitrate_updated_;
-      bitrate_updated_ = false;
-      return bitrate_updated;
-    }
-
-   private:
-    // We don't know the start bitrate, but assume that it is the default 300
-    // kbps.
-    uint32_t last_bitrate_bps_ = 300000;
-    bool bitrate_updated_ = false;
-  };
-
-  std::multimap<int64_t, const RtpPacketType*> incoming_rtp;
-
-  for (const auto& stream : parsed_log_.incoming_rtp_packets_by_ssrc()) {
-    if (IsVideoSsrc(parsed_log_, kIncomingPacket, stream.ssrc)) {
-      for (const auto& rtp_packet : stream.incoming_packets)
-        incoming_rtp.insert(
-            std::make_pair(rtp_packet.rtp.log_time_us(), &rtp_packet));
-    }
-  }
-
-  SimulatedClock clock(0);
-  EnvironmentFactory env_factory(env_);
-  env_factory.Set(&clock);
-  RembInterceptor remb_interceptor;
-  ReceiveSideCongestionController rscc(
-      env_factory.Create(), [](auto...) {},
-      absl::bind_front(&RembInterceptor::SendRemb, &remb_interceptor));
-  // TODO(holmer): Log the call config and use that here instead.
-  // static const uint32_t kDefaultStartBitrateBps = 300000;
-  // rscc.SetBweBitrates(0, kDefaultStartBitrateBps, -1);
-
-  TimeSeries time_series("Receive side estimate", LineStyle::kLine,
-                         PointStyle::kHighlight);
-  TimeSeries acked_time_series("Received bitrate", LineStyle::kLine);
-
-  RateStatistics acked_bitrate(250, 8000);
-  int64_t last_update_us = 0;
-  for (const auto& kv : incoming_rtp) {
-    const RtpPacketType& packet = *kv.second;
-
-    RtpPacketReceived rtp_packet = RtpPacketForBWEFromHeader(packet.rtp.header);
-    rtp_packet.set_arrival_time(packet.rtp.log_time());
-    rtp_packet.SetPayloadSize(packet.rtp.total_length -
-                              rtp_packet.headers_size());
-
-    clock.AdvanceTime(rtp_packet.arrival_time() - clock.CurrentTime());
-    rscc.OnReceivedPacket(rtp_packet, MediaType::VIDEO);
-    int64_t arrival_time_ms = packet.rtp.log_time().ms();
-    acked_bitrate.Update(packet.rtp.total_length, arrival_time_ms);
-    std::optional<uint32_t> bitrate_bps = acked_bitrate.Rate(arrival_time_ms);
-    if (bitrate_bps) {
-      uint32_t y = *bitrate_bps / 1000;
-      float x = config_.GetCallTimeSec(clock.CurrentTime());
-      acked_time_series.points.emplace_back(x, y);
-    }
-    if (remb_interceptor.GetAndResetBitrateUpdated() ||
-        clock.TimeInMicroseconds() - last_update_us >= 1e6) {
-      uint32_t y = remb_interceptor.last_bitrate_bps() / 1000;
-      float x = config_.GetCallTimeSec(clock.CurrentTime());
-      time_series.points.emplace_back(x, y);
-      last_update_us = clock.TimeInMicroseconds();
-    }
-  }
-  // Add the data set to the plot.
-  plot->AppendTimeSeries(std::move(time_series));
-  plot->AppendTimeSeries(std::move(acked_time_series));
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 10, "Bitrate (kbps)", kBottomMargin, kTopMargin);
-  plot->SetTitle("Simulated receive-side BWE behavior");
+  webrtc::CreateReceiveSideBweSimulationGraph(parsed_log_, config_, plot);
 }
 
 void EventLogAnalyzer::CreateNetworkDelayFeedbackGraph(Plot* plot) const {
-  TimeSeries time_series("Network delay", LineStyle::kLine,
-                         PointStyle::kHighlight);
-  int64_t min_send_receive_diff_ms = std::numeric_limits<int64_t>::max();
-  int64_t min_rtt_ms = std::numeric_limits<int64_t>::max();
-
-  std::vector<MatchedSendArrivalTimes> matched_rtp_rtcp =
-      GetNetworkTrace(parsed_log_);
-  absl::c_stable_sort(matched_rtp_rtcp, [](const MatchedSendArrivalTimes& a,
-                                           const MatchedSendArrivalTimes& b) {
-    return a.feedback_arrival_time_ms < b.feedback_arrival_time_ms ||
-           (a.feedback_arrival_time_ms == b.feedback_arrival_time_ms &&
-            a.arrival_time_ms < b.arrival_time_ms);
-  });
-  for (const auto& packet : matched_rtp_rtcp) {
-    if (packet.arrival_time_ms == MatchedSendArrivalTimes::kNotReceived)
-      continue;
-    float x = config_.GetCallTimeSecFromMs(packet.feedback_arrival_time_ms);
-    int64_t y = packet.arrival_time_ms - packet.send_time_ms;
-    int64_t rtt_ms = packet.feedback_arrival_time_ms - packet.send_time_ms;
-    min_rtt_ms = std::min(rtt_ms, min_rtt_ms);
-    min_send_receive_diff_ms = std::min(y, min_send_receive_diff_ms);
-    time_series.points.emplace_back(x, y);
-  }
-
-  // We assume that the base network delay (w/o queues) is equal to half
-  // the minimum RTT. Therefore rescale the delays by subtracting the minimum
-  // observed 1-ways delay and add half the minimum RTT.
-  const int64_t estimated_clock_offset_ms =
-      min_send_receive_diff_ms - min_rtt_ms / 2;
-  for (TimeSeriesPoint& point : time_series.points)
-    point.y -= estimated_clock_offset_ms;
-
-  // Add the data set to the plot.
-  plot->AppendTimeSeriesIfNotEmpty(std::move(time_series));
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 10, "Delay (ms)", kBottomMargin, kTopMargin);
-  plot->SetTitle("Outgoing network delay (based on per-packet feedback)");
+  webrtc::CreateNetworkDelayFeedbackGraph(parsed_log_, config_, plot);
 }
 
 void EventLogAnalyzer::CreatePacerDelayGraph(Plot* plot) const {
-  for (const auto& stream : parsed_log_.outgoing_rtp_packets_by_ssrc()) {
-    const std::vector<LoggedRtpPacketOutgoing>& packets =
-        stream.outgoing_packets;
+  webrtc::CreatePacerDelayGraph(parsed_log_, config_, plot);
+}
 
-    if (IsRtxSsrc(parsed_log_, kOutgoingPacket, stream.ssrc)) {
-      continue;
-    }
+void EventLogAnalyzer::CreateIceCandidatePairConfigGraph(Plot* plot) const {
+  webrtc::CreateIceCandidatePairConfigGraph(parsed_log_, config_, plot);
+}
 
-    if (packets.size() < 2) {
-      RTC_LOG(LS_WARNING)
-          << "Can't estimate a the RTP clock frequency or the "
-             "pacer delay with less than 2 packets in the stream";
-      continue;
-    }
-    int64_t segment_end_us = parsed_log_.first_log_segment().stop_time_us();
-    std::optional<uint32_t> estimated_frequency =
-        EstimateRtpClockFrequency(packets, segment_end_us);
-    if (!estimated_frequency)
-      continue;
-    if (IsVideoSsrc(parsed_log_, kOutgoingPacket, stream.ssrc) &&
-        *estimated_frequency != 90000) {
-      RTC_LOG(LS_WARNING)
-          << "Video stream should use a 90 kHz clock but appears to use "
-          << *estimated_frequency / 1000 << ". Discarding.";
-      continue;
-    }
+void EventLogAnalyzer::CreateIceConnectivityCheckGraph(Plot* plot) const {
+  webrtc::CreateIceConnectivityCheckGraph(parsed_log_, config_, plot);
+}
 
-    TimeSeries pacer_delay_series(
-        GetStreamName(parsed_log_, kOutgoingPacket, stream.ssrc) + "(" +
-            std::to_string(*estimated_frequency / 1000) + " kHz)",
-        LineStyle::kLine, PointStyle::kHighlight);
-    SeqNumUnwrapper<uint32_t> timestamp_unwrapper;
-    uint64_t first_capture_timestamp =
-        timestamp_unwrapper.Unwrap(packets.front().rtp.header.timestamp);
-    uint64_t first_send_timestamp = packets.front().rtp.log_time_us();
-    for (const auto& packet : packets) {
-      double capture_time_ms = (static_cast<double>(timestamp_unwrapper.Unwrap(
-                                    packet.rtp.header.timestamp)) -
-                                first_capture_timestamp) /
-                               *estimated_frequency * 1000;
-      double send_time_ms =
-          static_cast<double>(packet.rtp.log_time_us() - first_send_timestamp) /
-          1000;
-      float x = config_.GetCallTimeSec(packet.rtp.log_time());
-      float y = send_time_ms - capture_time_ms;
-      pacer_delay_series.points.emplace_back(x, y);
-    }
-    plot->AppendTimeSeries(std::move(pacer_delay_series));
-  }
+void EventLogAnalyzer::CreateDtlsTransportStateGraph(Plot* plot) const {
+  webrtc::CreateDtlsTransportStateGraph(parsed_log_, config_, plot);
+}
 
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 10, "Pacer delay (ms)", kBottomMargin, kTopMargin);
-  plot->SetTitle(
-      "Delay from capture to send time. (First packet normalized to 0.)");
+void EventLogAnalyzer::CreateDtlsWritableStateGraph(Plot* plot) const {
+  webrtc::CreateDtlsWritableStateGraph(parsed_log_, config_, plot);
+}
+
+void EventLogAnalyzer::CreatePacketGraph(PacketDirection direction,
+                                         Plot* plot) const {
+  webrtc::CreatePacketGraph(direction, parsed_log_, config_, plot);
+}
+
+void EventLogAnalyzer::CreateRtcpTypeGraph(PacketDirection direction,
+                                           Plot* plot) const {
+  webrtc::CreateRtcpTypeGraph(direction, parsed_log_, config_, plot);
+}
+
+void EventLogAnalyzer::CreateAccumulatedPacketsGraph(PacketDirection direction,
+                                                     Plot* plot) const {
+  webrtc::CreateAccumulatedPacketsGraph(direction, parsed_log_, config_, plot);
+}
+
+void EventLogAnalyzer::CreatePacketRateGraph(PacketDirection direction,
+                                             Plot* plot) const {
+  webrtc::CreatePacketRateGraph(direction, parsed_log_, config_, plot);
+}
+
+void EventLogAnalyzer::CreateTotalPacketRateGraph(PacketDirection direction,
+                                                  Plot* plot) const {
+  webrtc::CreateTotalPacketRateGraph(direction, parsed_log_, config_, plot);
+}
+
+void EventLogAnalyzer::CreateSequenceNumberGraph(Plot* plot) const {
+  webrtc::CreateSequenceNumberGraph(parsed_log_, config_, plot);
+}
+
+void EventLogAnalyzer::CreateIncomingPacketLossGraph(Plot* plot) const {
+  webrtc::CreateIncomingPacketLossGraph(parsed_log_, config_, plot);
+}
+
+void EventLogAnalyzer::CreateStreamBitrateGraph(PacketDirection direction,
+                                                Plot* plot) const {
+  webrtc::CreateStreamBitrateGraph(direction, parsed_log_, config_, plot);
+}
+
+void EventLogAnalyzer::CreateBitrateAllocationGraph(PacketDirection direction,
+                                                    Plot* plot) const {
+  webrtc::CreateBitrateAllocationGraph(direction, parsed_log_, config_, plot);
+}
+
+void EventLogAnalyzer::CreateOutgoingEcnFeedbackGraph(Plot* plot) const {
+  webrtc::CreateOutgoingEcnFeedbackGraph(parsed_log_, config_, plot);
+}
+
+void EventLogAnalyzer::CreateIncomingEcnFeedbackGraph(Plot* plot) const {
+  webrtc::CreateIncomingEcnFeedbackGraph(parsed_log_, config_, plot);
+}
+
+void EventLogAnalyzer::CreateOutgoingLossRateGraph(Plot* plot) const {
+  webrtc::CreateOutgoingLossRateGraph(parsed_log_, config_, plot);
 }
 
 void EventLogAnalyzer::CreateTimestampGraph(PacketDirection direction,
                                             Plot* plot) const {
-  for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
-    TimeSeries rtp_timestamps(
-        GetStreamName(parsed_log_, direction, stream.ssrc) + " capture-time",
-        LineStyle::kLine, PointStyle::kHighlight);
-    for (const auto& packet : stream.packet_view) {
-      float x = config_.GetCallTimeSec(packet.log_time());
-      float y = packet.header.timestamp;
-      rtp_timestamps.points.emplace_back(x, y);
-    }
-    plot->AppendTimeSeries(std::move(rtp_timestamps));
-
-    TimeSeries rtcp_timestamps(
-        GetStreamName(parsed_log_, direction, stream.ssrc) +
-            " rtcp capture-time",
-        LineStyle::kLine, PointStyle::kHighlight);
-    // TODO(terelius): Why only sender reports?
-    const auto& sender_reports = parsed_log_.sender_reports(direction);
-    for (const auto& rtcp : sender_reports) {
-      if (rtcp.sr.sender_ssrc() != stream.ssrc)
-        continue;
-      float x = config_.GetCallTimeSec(rtcp.log_time());
-      float y = rtcp.sr.rtp_timestamp();
-      rtcp_timestamps.points.emplace_back(x, y);
-    }
-    plot->AppendTimeSeriesIfNotEmpty(std::move(rtcp_timestamps));
-  }
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "RTP timestamp", kBottomMargin, kTopMargin);
-  plot->SetTitle(GetDirectionAsString(direction) + " timestamps");
+  webrtc::CreateTimestampGraph(direction, parsed_log_, config_, plot);
 }
 
 void EventLogAnalyzer::CreateSenderAndReceiverReportPlot(
@@ -2403,174 +439,8 @@ void EventLogAnalyzer::CreateSenderAndReceiverReportPlot(
     std::string title,
     std::string yaxis_label,
     Plot* plot) const {
-  std::map<uint32_t, TimeSeries> sr_reports_by_ssrc;
-  const auto& sender_reports = parsed_log_.sender_reports(direction);
-  for (const auto& rtcp : sender_reports) {
-    float x = config_.GetCallTimeSec(rtcp.log_time());
-    uint32_t ssrc = rtcp.sr.sender_ssrc();
-    for (const auto& block : rtcp.sr.report_blocks()) {
-      float y = fy(block);
-      auto sr_report_it = sr_reports_by_ssrc.find(ssrc);
-      bool inserted;
-      if (sr_report_it == sr_reports_by_ssrc.end()) {
-        std::tie(sr_report_it, inserted) = sr_reports_by_ssrc.emplace(
-            ssrc, TimeSeries(GetStreamName(parsed_log_, direction, ssrc) +
-                                 " Sender Reports",
-                             LineStyle::kLine, PointStyle::kHighlight));
-      }
-      sr_report_it->second.points.emplace_back(x, y);
-    }
-  }
-  for (auto& kv : sr_reports_by_ssrc) {
-    plot->AppendTimeSeries(std::move(kv.second));
-  }
-
-  std::map<uint32_t, TimeSeries> rr_reports_by_ssrc;
-  const auto& receiver_reports = parsed_log_.receiver_reports(direction);
-  for (const auto& rtcp : receiver_reports) {
-    float x = config_.GetCallTimeSec(rtcp.log_time());
-    uint32_t ssrc = rtcp.rr.sender_ssrc();
-    for (const auto& block : rtcp.rr.report_blocks()) {
-      float y = fy(block);
-      auto rr_report_it = rr_reports_by_ssrc.find(ssrc);
-      bool inserted;
-      if (rr_report_it == rr_reports_by_ssrc.end()) {
-        std::tie(rr_report_it, inserted) = rr_reports_by_ssrc.emplace(
-            ssrc, TimeSeries(GetStreamName(parsed_log_, direction, ssrc) +
-                                 " Receiver Reports",
-                             LineStyle::kLine, PointStyle::kHighlight));
-      }
-      rr_report_it->second.points.emplace_back(x, y);
-    }
-  }
-  for (auto& kv : rr_reports_by_ssrc) {
-    plot->AppendTimeSeries(std::move(kv.second));
-  }
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, yaxis_label, kBottomMargin, kTopMargin);
-  plot->SetTitle(title);
-}
-
-void EventLogAnalyzer::CreateIceCandidatePairConfigGraph(Plot* plot) const {
-  std::map<uint32_t, TimeSeries> configs_by_cp_id;
-  for (const auto& config : parsed_log_.ice_candidate_pair_configs()) {
-    if (configs_by_cp_id.find(config.candidate_pair_id) ==
-        configs_by_cp_id.end()) {
-      const std::string candidate_pair_desc =
-          GetCandidatePairLogDescriptionAsString(config);
-      configs_by_cp_id[config.candidate_pair_id] =
-          TimeSeries("[" + std::to_string(config.candidate_pair_id) + "]" +
-                         candidate_pair_desc,
-                     LineStyle::kNone, PointStyle::kHighlight);
-    }
-    float x = config_.GetCallTimeSec(config.log_time());
-    float y = static_cast<float>(config.type);
-    configs_by_cp_id[config.candidate_pair_id].points.emplace_back(x, y);
-  }
-
-  // TODO(qingsi): There can be a large number of candidate pairs generated by
-  // certain calls and the frontend cannot render the chart in this case due
-  // to the failure of generating a palette with the same number of colors.
-  for (auto& kv : configs_by_cp_id) {
-    plot->AppendTimeSeries(std::move(kv.second));
-  }
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 3, "Config Type", kBottomMargin, kTopMargin);
-  plot->SetTitle("[IceEventLog] ICE candidate pair configs");
-  plot->SetYAxisTickLabels(
-      {{static_cast<float>(IceCandidatePairConfigType::kAdded), "ADDED"},
-       {static_cast<float>(IceCandidatePairConfigType::kUpdated), "UPDATED"},
-       {static_cast<float>(IceCandidatePairConfigType::kDestroyed),
-        "DESTROYED"},
-       {static_cast<float>(IceCandidatePairConfigType::kSelected),
-        "SELECTED"}});
-}
-
-void EventLogAnalyzer::CreateIceConnectivityCheckGraph(Plot* plot) const {
-  constexpr int kEventTypeOffset =
-      static_cast<int>(IceCandidatePairConfigType::kNumValues);
-  std::map<uint32_t, TimeSeries> checks_by_cp_id;
-  std::map<uint32_t, std::string> candidate_pair_desc_by_id =
-      BuildCandidateIdLogDescriptionMap(
-          parsed_log_.ice_candidate_pair_configs());
-  for (const auto& event : parsed_log_.ice_candidate_pair_events()) {
-    if (checks_by_cp_id.find(event.candidate_pair_id) ==
-        checks_by_cp_id.end()) {
-      checks_by_cp_id[event.candidate_pair_id] =
-          TimeSeries("[" + std::to_string(event.candidate_pair_id) + "]" +
-                         candidate_pair_desc_by_id[event.candidate_pair_id],
-                     LineStyle::kNone, PointStyle::kHighlight);
-    }
-    float x = config_.GetCallTimeSec(event.log_time());
-    float y = static_cast<float>(event.type) + kEventTypeOffset;
-    checks_by_cp_id[event.candidate_pair_id].points.emplace_back(x, y);
-  }
-
-  // TODO(qingsi): The same issue as in CreateIceCandidatePairConfigGraph.
-  for (auto& kv : checks_by_cp_id) {
-    plot->AppendTimeSeries(std::move(kv.second));
-  }
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 4, "Connectivity State", kBottomMargin,
-                          kTopMargin);
-  plot->SetTitle("[IceEventLog] ICE connectivity checks");
-
-  plot->SetYAxisTickLabels(
-      {{static_cast<float>(IceCandidatePairEventType::kCheckSent) +
-            kEventTypeOffset,
-        "CHECK SENT"},
-       {static_cast<float>(IceCandidatePairEventType::kCheckReceived) +
-            kEventTypeOffset,
-        "CHECK RECEIVED"},
-       {static_cast<float>(IceCandidatePairEventType::kCheckResponseSent) +
-            kEventTypeOffset,
-        "RESPONSE SENT"},
-       {static_cast<float>(IceCandidatePairEventType::kCheckResponseReceived) +
-            kEventTypeOffset,
-        "RESPONSE RECEIVED"}});
-}
-
-void EventLogAnalyzer::CreateDtlsTransportStateGraph(Plot* plot) const {
-  TimeSeries states("DTLS Transport State", LineStyle::kNone,
-                    PointStyle::kHighlight);
-  for (const auto& event : parsed_log_.dtls_transport_states()) {
-    float x = config_.GetCallTimeSec(event.log_time());
-    float y = static_cast<float>(event.dtls_transport_state);
-    states.points.emplace_back(x, y);
-  }
-  plot->AppendTimeSeries(std::move(states));
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, static_cast<float>(DtlsTransportState::kNumValues),
-                          "Transport State", kBottomMargin, kTopMargin);
-  plot->SetTitle("DTLS Transport State");
-  plot->SetYAxisTickLabels(
-      {{static_cast<float>(DtlsTransportState::kNew), "NEW"},
-       {static_cast<float>(DtlsTransportState::kConnecting), "CONNECTING"},
-       {static_cast<float>(DtlsTransportState::kConnected), "CONNECTED"},
-       {static_cast<float>(DtlsTransportState::kClosed), "CLOSED"},
-       {static_cast<float>(DtlsTransportState::kFailed), "FAILED"}});
-}
-
-void EventLogAnalyzer::CreateDtlsWritableStateGraph(Plot* plot) const {
-  TimeSeries writable("DTLS Writable", LineStyle::kNone,
-                      PointStyle::kHighlight);
-  for (const auto& event : parsed_log_.dtls_writable_states()) {
-    float x = config_.GetCallTimeSec(event.log_time());
-    float y = static_cast<float>(event.writable);
-    writable.points.emplace_back(x, y);
-  }
-  plot->AppendTimeSeries(std::move(writable));
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Writable", kBottomMargin, kTopMargin);
-  plot->SetTitle("DTLS Writable State");
+  webrtc::CreateSenderAndReceiverReportPlot(direction, fy, title, yaxis_label,
+                                            parsed_log_, config_, plot);
 }
 
 }  // namespace webrtc
diff --git a/rtc_tools/rtc_event_log_visualizer/analyzer.h b/rtc_tools/rtc_event_log_visualizer/analyzer.h
index ca729b3cbc..8ac90e3040 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyzer.h
+++ b/rtc_tools/rtc_event_log_visualizer/analyzer.h
@@ -11,13 +11,11 @@
 #ifndef RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZER_H_
 #define RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZER_H_
 
-#include <cstdint>
 #include <cstdio>
 #include <functional>
 #include <string>
 #include <vector>
 
-#include "api/environment/environment.h"
 #include "api/function_view.h"
 #include "logging/rtc_event_log/rtc_event_log_parser.h"
 #include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
@@ -62,9 +60,7 @@ class EventLogAnalyzer {
   // duration of its lifetime. The ParsedRtcEventLogNew must not be destroyed or
   // modified while the EventLogAnalyzer is being used.
   EventLogAnalyzer(const ParsedRtcEventLog& log, bool normalize_time);
-  EventLogAnalyzer(const Environment& env,
-                   const ParsedRtcEventLog& log,
-                   const AnalyzerConfig& config);
+  EventLogAnalyzer(const ParsedRtcEventLog& log, const AnalyzerConfig& config);
 
   void CreateGraphsByName(const std::vector<std::string>& names,
                           PlotCollection* collection) const;
@@ -149,19 +145,8 @@ class EventLogAnalyzer {
   void PrintNotifications(FILE* file) const;
 
  private:
-  template <typename IterableType>
-  void CreateAccumulatedPacketsTimeSeries(Plot* plot,
-                                          const IterableType& packets,
-                                          const std::string& label) const;
-  void CreateEcnFeedbackGraph(Plot* plot, PacketDirection direction) const;
-
-  const Environment env_;
   const ParsedRtcEventLog& parsed_log_;
 
-  // A list of SSRCs we are interested in analysing.
-  // If left empty, all SSRCs will be considered relevant.
-  std::vector<uint32_t> desired_ssrc_;
-
   AnalyzerConfig config_;
 
   PlotMap plots_;
diff --git a/rtc_tools/rtc_event_log_visualizer/analyzer_bindings.cc b/rtc_tools/rtc_event_log_visualizer/analyzer_bindings.cc
index fa93031ff1..4aed263a98 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyzer_bindings.cc
+++ b/rtc_tools/rtc_event_log_visualizer/analyzer_bindings.cc
@@ -64,16 +64,10 @@ void analyze_rtc_event_log(const char* log_contents,
     return;
   }
 
-  webrtc::AnalyzerConfig config;
+  webrtc::AnalyzerConfig config(webrtc::CreateEnvironment(), parsed_log,
+                                /*normalize_time=*/true);
   config.window_duration_ = webrtc::TimeDelta::Millis(250);
   config.step_ = webrtc::TimeDelta::Millis(10);
-  if (!parsed_log.start_log_events().empty()) {
-    config.rtc_to_utc_offset_ = parsed_log.start_log_events()[0].utc_time() -
-                                parsed_log.start_log_events()[0].log_time();
-  }
-  config.normalize_time_ = true;
-  config.begin_time_ = parsed_log.first_timestamp();
-  config.end_time_ = parsed_log.last_timestamp();
   if (config.end_time_ < config.begin_time_) {
     std::cerr << "Log end time " << config.end_time_.ms()
               << " not after begin time " << config.begin_time_.ms()
@@ -82,8 +76,7 @@ void analyze_rtc_event_log(const char* log_contents,
     return;
   }
 
-  webrtc::EventLogAnalyzer analyzer(webrtc::CreateEnvironment(), parsed_log,
-                                    config);
+  webrtc::EventLogAnalyzer analyzer(parsed_log, config);
   analyzer.InitializeMapOfNamedGraphs(/*show_detector_state=*/false,
                                       /*show_alr_state=*/false,
                                       /*show_link_capacity=*/false);
diff --git a/rtc_tools/rtc_event_log_visualizer/analyzer_common.cc b/rtc_tools/rtc_event_log_visualizer/analyzer_common.cc
index c1cf7dc6b7..559ad78b4a 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyzer_common.cc
+++ b/rtc_tools/rtc_event_log_visualizer/analyzer_common.cc
@@ -11,10 +11,14 @@
 
 #include "rtc_tools/rtc_event_log_visualizer/analyzer_common.h"
 
+#include <algorithm>
 #include <cstdint>
 #include <string>
+#include <vector>
 
 #include "logging/rtc_event_log/rtc_event_log_parser.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
 #include "rtc_base/strings/string_builder.h"
 
 namespace webrtc {
@@ -86,4 +90,64 @@ std::string GetLayerName(LayerDescription layer) {
   return name.str();
 }
 
+std::string SsrcToString(uint32_t ssrc) {
+  StringBuilder ss;
+  ss << "SSRC " << ssrc;
+  return ss.Release();
+}
+
+// Checks whether an SSRC is contained in the list of desired SSRCs.
+// Note that an empty SSRC list matches every SSRC.
+bool MatchingSsrc(uint32_t ssrc, const std::vector<uint32_t>& desired_ssrc) {
+  if (desired_ssrc.empty())
+    return true;
+  return std::find(desired_ssrc.begin(), desired_ssrc.end(), ssrc) !=
+         desired_ssrc.end();
+}
+
+// Computes the difference `later` - `earlier` where `later` and `earlier`
+// are counters that wrap at `modulus`. The difference is chosen to have the
+// least absolute value. For example if `modulus` is 8, then the difference will
+// be chosen in the range [-3, 4]. If `modulus` is 9, then the difference will
+// be in [-4, 4].
+int64_t WrappingDifference(uint32_t later, uint32_t earlier, int64_t modulus) {
+  RTC_DCHECK_LE(1, modulus);
+  RTC_DCHECK_LT(later, modulus);
+  RTC_DCHECK_LT(earlier, modulus);
+  int64_t difference =
+      static_cast<int64_t>(later) - static_cast<int64_t>(earlier);
+  int64_t max_difference = modulus / 2;
+  int64_t min_difference = max_difference - modulus + 1;
+  if (difference > max_difference) {
+    difference -= modulus;
+  }
+  if (difference < min_difference) {
+    difference += modulus;
+  }
+  if (difference > max_difference / 2 || difference < min_difference / 2) {
+    RTC_LOG(LS_WARNING) << "Difference between" << later << " and " << earlier
+                        << " expected to be in the range ("
+                        << min_difference / 2 << "," << max_difference / 2
+                        << ") but is " << difference
+                        << ". Correct unwrapping is uncertain.";
+  }
+  return difference;
+}
+
+std::string GetDirectionAsString(PacketDirection direction) {
+  if (direction == kIncomingPacket) {
+    return "Incoming";
+  } else {
+    return "Outgoing";
+  }
+}
+
+std::string GetDirectionAsShortString(PacketDirection direction) {
+  if (direction == kIncomingPacket) {
+    return "In";
+  } else {
+    return "Out";
+  }
+}
+
 }  // namespace webrtc
diff --git a/rtc_tools/rtc_event_log_visualizer/analyzer_common.h b/rtc_tools/rtc_event_log_visualizer/analyzer_common.h
index d2698f84b1..048d3d1578 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyzer_common.h
+++ b/rtc_tools/rtc_event_log_visualizer/analyzer_common.h
@@ -15,7 +15,9 @@
 #include <cstdint>
 #include <optional>
 #include <string>
+#include <vector>
 
+#include "api/environment/environment.h"
 #include "api/function_view.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
@@ -33,6 +35,18 @@ constexpr float kTopMargin = 0.05f;
 
 class AnalyzerConfig {
  public:
+  AnalyzerConfig(const Environment& env,
+                 const ParsedRtcEventLog& parsed_log,
+                 bool normalize_time)
+      : normalize_time_(normalize_time), env_(env) {
+    begin_time_ = parsed_log.first_timestamp();
+    end_time_ = parsed_log.last_timestamp();
+    if (!parsed_log.start_log_events().empty()) {
+      rtc_to_utc_offset_ = parsed_log.start_log_events().front().utc_time() -
+                           parsed_log.start_log_events().front().log_time();
+    }
+  }
+
   float GetCallTimeSec(Timestamp timestamp) const {
     Timestamp offset = normalize_time_ ? begin_time_ : Timestamp::Zero();
     return static_cast<float>((timestamp - offset).us()) / 1000000;
@@ -66,6 +80,8 @@ class AnalyzerConfig {
   Timestamp end_time_ = Timestamp::MinusInfinity();
   TimeDelta rtc_to_utc_offset_ = TimeDelta::Zero();
   bool normalize_time_;
+  std::vector<uint32_t> desired_ssrc_;
+  const Environment env_;
 };
 
 struct LayerDescription {
@@ -98,8 +114,19 @@ bool IsAudioSsrc(const ParsedRtcEventLog& parsed_log,
 std::string GetStreamName(const ParsedRtcEventLog& parsed_log,
                           PacketDirection direction,
                           uint32_t ssrc);
+std::string SsrcToString(uint32_t ssrc);
+
 std::string GetLayerName(LayerDescription layer);
 
+std::string GetDirectionAsString(PacketDirection direction);
+std::string GetDirectionAsShortString(PacketDirection direction);
+
+int64_t WrappingDifference(uint32_t later, uint32_t earlier, int64_t modulus);
+
+// Checks whether an SSRC is contained in the list of desired SSRCs.
+// Note that an empty SSRC list matches every SSRC.
+bool MatchingSsrc(uint32_t ssrc, const std::vector<uint32_t>& desired_ssrc);
+
 // For each element in data_view, use `f()` to extract a y-coordinate and
 // store the result in a TimeSeries.
 template <typename DataType, typename IterableType>
diff --git a/rtc_tools/rtc_event_log_visualizer/main.cc b/rtc_tools/rtc_event_log_visualizer/main.cc
index 03512b185b..a08588e8a1 100644
--- a/rtc_tools/rtc_event_log_visualizer/main.cc
+++ b/rtc_tools/rtc_event_log_visualizer/main.cc
@@ -192,17 +192,11 @@ int main(int argc, char* argv[]) {
     }
   }
 
-  webrtc::AnalyzerConfig config;
+  webrtc::AnalyzerConfig config(env, parsed_log,
+                                absl::GetFlag(FLAGS_normalize_time));
   config.window_duration_ =
       webrtc::TimeDelta::Millis(absl::GetFlag(FLAGS_averaging_window));
   config.step_ = webrtc::TimeDelta::Millis(absl::GetFlag(FLAGS_averaging_step));
-  if (!parsed_log.start_log_events().empty()) {
-    config.rtc_to_utc_offset_ = parsed_log.start_log_events()[0].utc_time() -
-                                parsed_log.start_log_events()[0].log_time();
-  }
-  config.normalize_time_ = absl::GetFlag(FLAGS_normalize_time);
-  config.begin_time_ = parsed_log.first_timestamp();
-  config.end_time_ = parsed_log.last_timestamp();
   if (config.end_time_ < config.begin_time_) {
     RTC_LOG(LS_WARNING) << "Log end time " << config.end_time_
                         << " not after begin time " << config.begin_time_
@@ -225,7 +219,7 @@ int main(int argc, char* argv[]) {
     has_generated_wav_file = true;
   }
 
-  webrtc::EventLogAnalyzer analyzer(env, parsed_log, config);
+  webrtc::EventLogAnalyzer analyzer(parsed_log, config);
   analyzer.InitializeMapOfNamedGraphs(absl::GetFlag(FLAGS_show_detector_state),
                                       absl::GetFlag(FLAGS_show_alr_state),
                                       absl::GetFlag(FLAGS_show_link_capacity));

From a6c289611eff119ff8e216d41657facd4718732f Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Bj=C3=B6rn=20Terelius?= <terelius@webrtc.org>
Date: Fri, 13 Feb 2026 03:25:59 -0800
Subject: [PATCH 039/472] Revert "Split analyzer.cc and change CreateFooGraph
 methods to free functions"
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This reverts commit 68cb390aa5565bcf8a6c360f45177743c701bbda.

Reason for revert: Breaks downstream code.

Original change's description:
> Split analyzer.cc and change CreateFooGraph methods to free functions
>
> Bug: None
> Change-Id: Iaf4a39ad3c9bc681b5ec3f078b328eed0c218942
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/446284
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#46891}

Bug: None
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: Ie0114c6569c80c6decd4a51c7cb1ea68ab102918
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/448981
Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46892}
---
 rtc_tools/BUILD.gn                            |    6 -
 .../rtc_event_log_visualizer/analyze_audio.cc |   90 +-
 .../rtc_event_log_visualizer/analyze_audio.h  |   13 -
 .../rtc_event_log_visualizer/analyze_bwe.cc   | 1221 ---------
 .../rtc_event_log_visualizer/analyze_bwe.h    |   81 -
 .../analyze_connectivity.cc                   |  285 --
 .../analyze_connectivity.h                    |   38 -
 .../analyze_rtp_rtcp.cc                       |  809 ------
 .../analyze_rtp_rtcp.h                        |  100 -
 .../rtc_event_log_visualizer/analyzer.cc      | 2364 ++++++++++++++++-
 rtc_tools/rtc_event_log_visualizer/analyzer.h |   17 +-
 .../analyzer_bindings.cc                      |   13 +-
 .../analyzer_common.cc                        |   64 -
 .../analyzer_common.h                         |   27 -
 rtc_tools/rtc_event_log_visualizer/main.cc    |   12 +-
 15 files changed, 2284 insertions(+), 2856 deletions(-)
 delete mode 100644 rtc_tools/rtc_event_log_visualizer/analyze_bwe.cc
 delete mode 100644 rtc_tools/rtc_event_log_visualizer/analyze_bwe.h
 delete mode 100644 rtc_tools/rtc_event_log_visualizer/analyze_connectivity.cc
 delete mode 100644 rtc_tools/rtc_event_log_visualizer/analyze_connectivity.h
 delete mode 100644 rtc_tools/rtc_event_log_visualizer/analyze_rtp_rtcp.cc
 delete mode 100644 rtc_tools/rtc_event_log_visualizer/analyze_rtp_rtcp.h

diff --git a/rtc_tools/BUILD.gn b/rtc_tools/BUILD.gn
index 24c20bbb24..c14c29d018 100644
--- a/rtc_tools/BUILD.gn
+++ b/rtc_tools/BUILD.gn
@@ -290,12 +290,6 @@ if (!build_with_chromium) {
         "rtc_event_log_visualizer/alerts.h",
         "rtc_event_log_visualizer/analyze_audio.cc",
         "rtc_event_log_visualizer/analyze_audio.h",
-        "rtc_event_log_visualizer/analyze_bwe.cc",
-        "rtc_event_log_visualizer/analyze_bwe.h",
-        "rtc_event_log_visualizer/analyze_connectivity.cc",
-        "rtc_event_log_visualizer/analyze_connectivity.h",
-        "rtc_event_log_visualizer/analyze_rtp_rtcp.cc",
-        "rtc_event_log_visualizer/analyze_rtp_rtcp.h",
         "rtc_event_log_visualizer/analyzer.cc",
         "rtc_event_log_visualizer/analyzer.h",
         "rtc_event_log_visualizer/analyzer_common.cc",
diff --git a/rtc_tools/rtc_event_log_visualizer/analyze_audio.cc b/rtc_tools/rtc_event_log_visualizer/analyze_audio.cc
index f2ec478315..ed3629a787 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyze_audio.cc
+++ b/rtc_tools/rtc_event_log_visualizer/analyze_audio.cc
@@ -30,10 +30,7 @@
 #include "api/neteq/neteq.h"
 #include "api/scoped_refptr.h"
 #include "api/units/timestamp.h"
-#include "logging/rtc_event_log/events/logged_rtp_rtcp.h"
 #include "logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h"
-#include "logging/rtc_event_log/events/rtc_event_audio_playout.h"
-#include "logging/rtc_event_log/events/rtc_event_neteq_set_minimum_delay.h"
 #include "logging/rtc_event_log/rtc_event_log_parser.h"
 #include "modules/audio_coding/neteq/tools/audio_sink.h"
 #include "modules/audio_coding/neteq/tools/fake_decode_from_file.h"
@@ -280,8 +277,8 @@ std::unique_ptr<test::NetEqStatsGetter> CreateNetEqTestAndRun(
 
   NetEq::Config config;
   test::NetEqTest test(config, decoder_factory, codecs, /*text_log=*/nullptr,
-                       /*neteq_factory=*/nullptr, std::move(input),
-                       std::move(output), callbacks, field_trials);
+                       /*factory=*/nullptr, std::move(input), std::move(output),
+                       callbacks, field_trials);
   test.Run();
   return neteq_stats_getter;
 }
@@ -429,87 +426,4 @@ void CreateNetEqLifetimeStatsGraph(
       stats_extractor, plot_name, plot);
 }
 
-// For each SSRC, plot the time between the consecutive playouts.
-void CreatePlayoutGraph(const ParsedRtcEventLog& parsed_log,
-                        const AnalyzerConfig& config,
-                        Plot* plot) {
-  for (const auto& playout_stream : parsed_log.audio_playout_events()) {
-    uint32_t ssrc = playout_stream.first;
-    if (!MatchingSsrc(ssrc, config.desired_ssrc_))
-      continue;
-    std::optional<int64_t> last_playout_ms;
-    TimeSeries time_series(SsrcToString(ssrc), LineStyle::kBar);
-    for (const auto& playout_event : playout_stream.second) {
-      float x = config.GetCallTimeSec(playout_event.log_time());
-      int64_t playout_time_ms = playout_event.log_time_ms();
-      // If there were no previous playouts, place the point on the x-axis.
-      float y = playout_time_ms - last_playout_ms.value_or(playout_time_ms);
-      time_series.points.push_back(TimeSeriesPoint(x, y));
-      last_playout_ms.emplace(playout_time_ms);
-    }
-    plot->AppendTimeSeries(std::move(time_series));
-  }
-
-  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
-                 kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Time since last playout (ms)", kBottomMargin,
-                          kTopMargin);
-  plot->SetTitle("Audio playout");
-}
-
-void CreateNetEqSetMinimumDelay(const ParsedRtcEventLog& parsed_log,
-                                const AnalyzerConfig& config,
-                                Plot* plot) {
-  for (const auto& playout_stream :
-       parsed_log.neteq_set_minimum_delay_events()) {
-    uint32_t ssrc = playout_stream.first;
-    if (!MatchingSsrc(ssrc, config.desired_ssrc_))
-      continue;
-
-    TimeSeries time_series(SsrcToString(ssrc), LineStyle::kStep,
-                           PointStyle::kHighlight);
-    for (const auto& event : playout_stream.second) {
-      float x = config.GetCallTimeSec(event.log_time());
-      float y = event.minimum_delay_ms;
-      time_series.points.push_back(TimeSeriesPoint(x, y));
-    }
-    plot->AppendTimeSeries(std::move(time_series));
-  }
-
-  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
-                 kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1000, "Minimum Delay (ms)", kBottomMargin,
-                          kTopMargin);
-  plot->SetTitle("Set Minimum Delay");
-}
-
-// For audio SSRCs, plot the audio level.
-void CreateAudioLevelGraph(const ParsedRtcEventLog& parsed_log,
-                           const AnalyzerConfig& config,
-                           PacketDirection direction,
-                           Plot* plot) {
-  for (const auto& stream : parsed_log.rtp_packets_by_ssrc(direction)) {
-    if (!IsAudioSsrc(parsed_log, direction, stream.ssrc))
-      continue;
-    TimeSeries time_series(GetStreamName(parsed_log, direction, stream.ssrc),
-                           LineStyle::kLine);
-    for (auto& packet : stream.packet_view) {
-      if (packet.header.extension.audio_level()) {
-        float x = config.GetCallTimeSec(packet.log_time());
-        // The audio level is stored in -dBov (so e.g. -10 dBov is stored as 10)
-        // Here we convert it to dBov.
-        float y =
-            static_cast<float>(-packet.header.extension.audio_level()->level());
-        time_series.points.emplace_back(TimeSeriesPoint(x, y));
-      }
-    }
-    plot->AppendTimeSeries(std::move(time_series));
-  }
-
-  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
-                 kLeftMargin, kRightMargin);
-  plot->SetYAxis(-127, 0, "Audio level (dBov)", kBottomMargin, kTopMargin);
-  plot->SetTitle(GetDirectionAsString(direction) + " audio level");
-}
-
 }  // namespace webrtc
diff --git a/rtc_tools/rtc_event_log_visualizer/analyze_audio.h b/rtc_tools/rtc_event_log_visualizer/analyze_audio.h
index 55040ae39f..094f9d91b6 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyze_audio.h
+++ b/rtc_tools/rtc_event_log_visualizer/analyze_audio.h
@@ -45,19 +45,6 @@ void CreateAudioEncoderNumChannelsGraph(const ParsedRtcEventLog& parsed_log,
                                         const AnalyzerConfig& config,
                                         Plot* plot);
 
-void CreatePlayoutGraph(const ParsedRtcEventLog& parsed_log,
-                        const AnalyzerConfig& config,
-                        Plot* plot);
-
-void CreateNetEqSetMinimumDelay(const ParsedRtcEventLog& parsed_log,
-                                const AnalyzerConfig& config,
-                                Plot* plot);
-
-void CreateAudioLevelGraph(const ParsedRtcEventLog& parsed_log,
-                           const AnalyzerConfig& config,
-                           PacketDirection direction,
-                           Plot* plot);
-
 using NetEqStatsGetterMap =
     std::map<uint32_t, std::unique_ptr<test::NetEqStatsGetter>>;
 NetEqStatsGetterMap SimulateNetEq(const ParsedRtcEventLog& parsed_log,
diff --git a/rtc_tools/rtc_event_log_visualizer/analyze_bwe.cc b/rtc_tools/rtc_event_log_visualizer/analyze_bwe.cc
deleted file mode 100644
index 0480256576..0000000000
--- a/rtc_tools/rtc_event_log_visualizer/analyze_bwe.cc
+++ /dev/null
@@ -1,1221 +0,0 @@
-/*
- *  Copyright (c) 2026 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "rtc_tools/rtc_event_log_visualizer/analyze_bwe.h"
-
-#include <algorithm>
-#include <cmath>
-#include <cstddef>
-#include <cstdint>
-#include <limits>
-#include <map>
-#include <memory>
-#include <optional>
-#include <string>
-#include <utility>
-#include <vector>
-
-#include "absl/algorithm/container.h"
-#include "absl/functional/bind_front.h"
-#include "absl/strings/string_view.h"
-#include "api/environment/environment.h"
-#include "api/environment/environment_factory.h"
-#include "api/field_trials.h"
-#include "api/function_view.h"
-#include "api/media_types.h"
-#include "api/rtp_headers.h"
-#include "api/transport/bandwidth_usage.h"
-#include "api/transport/goog_cc_factory.h"
-#include "api/transport/network_control.h"
-#include "api/transport/network_types.h"
-#include "api/units/data_rate.h"
-#include "api/units/time_delta.h"
-#include "api/units/timestamp.h"
-#include "logging/rtc_event_log/events/logged_rtp_rtcp.h"
-#include "logging/rtc_event_log/rtc_event_log_parser.h"
-#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.h"
-#include "modules/congestion_controller/include/receive_side_congestion_controller.h"
-#include "modules/congestion_controller/rtp/transport_feedback_adapter.h"
-#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
-#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
-#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
-#include "modules/rtp_rtcp/source/rtcp_packet/remb.h"
-#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
-#include "modules/rtp_rtcp/source/rtp_packet_received.h"
-#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
-#include "rtc_base/checks.h"
-#include "rtc_base/logging.h"
-#include "rtc_base/network/sent_packet.h"
-#include "rtc_base/numerics/sequence_number_unwrapper.h"
-#include "rtc_base/rate_statistics.h"
-#include "rtc_tools/rtc_event_log_visualizer/analyzer_common.h"
-#include "rtc_tools/rtc_event_log_visualizer/log_scream_simulation.h"
-#include "rtc_tools/rtc_event_log_visualizer/log_simulation.h"
-#include "rtc_tools/rtc_event_log_visualizer/plot_base.h"
-#include "system_wrappers/include/clock.h"
-
-namespace webrtc {
-
-namespace {
-
-double AbsSendTimeToMicroseconds(int64_t abs_send_time) {
-  // The timestamp is a fixed point representation with 6 bits for seconds
-  // and 18 bits for fractions of a second. Thus, we divide by 2^18 to get the
-  // time in seconds and then multiply by kNumMicrosecsPerSec to convert to
-  // microseconds.
-  static constexpr double kTimestampToMicroSec =
-      static_cast<double>(kNumMicrosecsPerSec) / static_cast<double>(1ul << 18);
-  return abs_send_time * kTimestampToMicroSec;
-}
-
-// This is much more reliable for outgoing streams than for incoming streams.
-template <typename RtpPacketContainer>
-std::optional<uint32_t> EstimateRtpClockFrequency(
-    const RtpPacketContainer& packets,
-    int64_t end_time_us) {
-  RTC_CHECK(packets.size() >= 2);
-  SeqNumUnwrapper<uint32_t> unwrapper;
-  int64_t first_rtp_timestamp =
-      unwrapper.Unwrap(packets[0].rtp.header.timestamp);
-  int64_t first_log_timestamp = packets[0].log_time_us();
-  int64_t last_rtp_timestamp = first_rtp_timestamp;
-  int64_t last_log_timestamp = first_log_timestamp;
-  for (size_t i = 1; i < packets.size(); i++) {
-    if (packets[i].log_time_us() > end_time_us)
-      break;
-    last_rtp_timestamp = unwrapper.Unwrap(packets[i].rtp.header.timestamp);
-    last_log_timestamp = packets[i].log_time_us();
-  }
-  if (last_log_timestamp - first_log_timestamp < kNumMicrosecsPerSec) {
-    RTC_LOG(LS_WARNING)
-        << "Failed to estimate RTP clock frequency: Stream too short. ("
-        << packets.size() << " packets, "
-        << last_log_timestamp - first_log_timestamp << " us)";
-    return std::nullopt;
-  }
-  double duration =
-      static_cast<double>(last_log_timestamp - first_log_timestamp) /
-      kNumMicrosecsPerSec;
-  double estimated_frequency =
-      (last_rtp_timestamp - first_rtp_timestamp) / duration;
-  for (uint32_t f : {8000, 16000, 32000, 48000, 90000}) {
-    if (std::fabs(estimated_frequency - f) < 0.15 * f) {
-      return f;
-    }
-  }
-  RTC_LOG(LS_WARNING) << "Failed to estimate RTP clock frequency: Estimate "
-                      << estimated_frequency
-                      << " not close to any standard RTP frequency."
-                      << " Last timestamp " << last_rtp_timestamp
-                      << " first timestamp " << first_rtp_timestamp;
-  return std::nullopt;
-}
-
-std::optional<double> NetworkDelayDiff_AbsSendTime(
-    const LoggedRtpPacketIncoming& old_packet,
-    const LoggedRtpPacketIncoming& new_packet) {
-  if (old_packet.rtp.header.extension.hasAbsoluteSendTime &&
-      new_packet.rtp.header.extension.hasAbsoluteSendTime) {
-    int64_t send_time_diff = WrappingDifference(
-        new_packet.rtp.header.extension.absoluteSendTime,
-        old_packet.rtp.header.extension.absoluteSendTime, 1ul << 24);
-    int64_t recv_time_diff =
-        new_packet.log_time_us() - old_packet.log_time_us();
-    double delay_change_us =
-        recv_time_diff - AbsSendTimeToMicroseconds(send_time_diff);
-    return delay_change_us / 1000;
-  } else {
-    return std::nullopt;
-  }
-}
-
-std::optional<double> NetworkDelayDiff_CaptureTime(
-    const LoggedRtpPacketIncoming& old_packet,
-    const LoggedRtpPacketIncoming& new_packet,
-    const double sample_rate) {
-  int64_t send_time_diff =
-      WrappingDifference(new_packet.rtp.header.timestamp,
-                         old_packet.rtp.header.timestamp, 1ull << 32);
-  int64_t recv_time_diff = new_packet.log_time_us() - old_packet.log_time_us();
-
-  double delay_change =
-      static_cast<double>(recv_time_diff) / 1000 -
-      static_cast<double>(send_time_diff) / sample_rate * 1000;
-  if (delay_change < -10000 || 10000 < delay_change) {
-    RTC_LOG(LS_WARNING) << "Very large delay change. Timestamps correct?";
-    RTC_LOG(LS_WARNING) << "Old capture time "
-                        << old_packet.rtp.header.timestamp << ", received time "
-                        << old_packet.log_time_us();
-    RTC_LOG(LS_WARNING) << "New capture time "
-                        << new_packet.rtp.header.timestamp << ", received time "
-                        << new_packet.log_time_us();
-    RTC_LOG(LS_WARNING) << "Receive time difference " << recv_time_diff << " = "
-                        << static_cast<double>(recv_time_diff) /
-                               kNumMicrosecsPerSec
-                        << "s";
-    RTC_LOG(LS_WARNING) << "Send time difference " << send_time_diff << " = "
-                        << static_cast<double>(send_time_diff) / sample_rate
-                        << "s";
-  }
-  return delay_change;
-}
-
-struct FakeExtensionSmall {
-  static constexpr RTPExtensionType kId = kRtpExtensionMid;
-  static constexpr absl::string_view Uri() { return "fake-extension-small"; }
-};
-struct FakeExtensionLarge {
-  static constexpr RTPExtensionType kId = kRtpExtensionRtpStreamId;
-  static constexpr absl::string_view Uri() { return "fake-extension-large"; }
-};
-
-RtpPacketReceived RtpPacketForBWEFromHeader(const RTPHeader& header) {
-  RtpHeaderExtensionMap rtp_header_extensions(/*extmap_allow_mixed=*/true);
-  // ReceiveSideCongestionController doesn't need to know extensions ids as
-  // long as it able to get extensions by type. So any ids would work here.
-  rtp_header_extensions.Register<TransmissionOffset>(1);
-  rtp_header_extensions.Register<AbsoluteSendTime>(2);
-  rtp_header_extensions.Register<TransportSequenceNumber>(3);
-  rtp_header_extensions.Register<FakeExtensionSmall>(4);
-  // Use id > 14 to force two byte header per rtp header when this one is used.
-  rtp_header_extensions.Register<FakeExtensionLarge>(16);
-
-  RtpPacketReceived rtp_packet(&rtp_header_extensions);
-  // Set only fields that might be relevant for the bandwidth estimatior.
-  rtp_packet.SetSsrc(header.ssrc);
-  rtp_packet.SetTimestamp(header.timestamp);
-  size_t num_bwe_extensions = 0;
-  if (header.extension.hasTransmissionTimeOffset) {
-    rtp_packet.SetExtension<TransmissionOffset>(
-        header.extension.transmissionTimeOffset);
-    ++num_bwe_extensions;
-  }
-  if (header.extension.hasAbsoluteSendTime) {
-    rtp_packet.SetExtension<AbsoluteSendTime>(
-        header.extension.absoluteSendTime);
-    ++num_bwe_extensions;
-  }
-  if (header.extension.hasTransportSequenceNumber) {
-    rtp_packet.SetExtension<TransportSequenceNumber>(
-        header.extension.transportSequenceNumber);
-    ++num_bwe_extensions;
-  }
-
-  // All parts of the RTP header are 32bit aligned.
-  RTC_CHECK_EQ(header.headerLength % 4, 0);
-
-  // Original packet could have more extensions, there could be csrcs that are
-  // not propagated by the rtc event log, i.e. logged header size might be
-  // larger that rtp_packet.header_size(). Increase it by setting an extra fake
-  // extension.
-  RTC_CHECK_GE(header.headerLength, rtp_packet.headers_size());
-  size_t bytes_to_add = header.headerLength - rtp_packet.headers_size();
-  if (bytes_to_add > 0) {
-    if (bytes_to_add <= 16) {
-      // one-byte header rtp header extension allows to add up to 16 bytes.
-      rtp_packet.AllocateExtension(FakeExtensionSmall::kId, bytes_to_add - 1);
-    } else {
-      // two-byte header rtp header extension would also add one byte per
-      // already set extension.
-      rtp_packet.AllocateExtension(FakeExtensionLarge::kId,
-                                   bytes_to_add - 2 - num_bwe_extensions);
-    }
-  }
-  RTC_CHECK_EQ(rtp_packet.headers_size(), header.headerLength);
-
-  return rtp_packet;
-}
-
-}  // namespace
-
-class BitrateObserver : public RemoteBitrateObserver {
- public:
-  BitrateObserver() : last_bitrate_bps_(0), bitrate_updated_(false) {}
-
-  void Update(NetworkControlUpdate update) {
-    if (update.target_rate) {
-      last_bitrate_bps_ = update.target_rate->target_rate.bps();
-      bitrate_updated_ = true;
-    }
-  }
-
-  void OnReceiveBitrateChanged(const std::vector<uint32_t>& ssrcs,
-                               uint32_t bitrate) override {}
-
-  uint32_t last_bitrate_bps() const { return last_bitrate_bps_; }
-  bool GetAndResetBitrateUpdated() {
-    bool bitrate_updated = bitrate_updated_;
-    bitrate_updated_ = false;
-    return bitrate_updated;
-  }
-
- private:
-  uint32_t last_bitrate_bps_;
-  bool bitrate_updated_;
-};
-
-void CreateIncomingDelayGraph(const ParsedRtcEventLog& parsed_log,
-                              const AnalyzerConfig& config,
-                              Plot* plot) {
-  for (const auto& stream : parsed_log.incoming_rtp_packets_by_ssrc()) {
-    // Filter on SSRC.
-    if (!MatchingSsrc(stream.ssrc, config.desired_ssrc_) ||
-        IsRtxSsrc(parsed_log, kIncomingPacket, stream.ssrc)) {
-      continue;
-    }
-
-    const std::vector<LoggedRtpPacketIncoming>& packets =
-        stream.incoming_packets;
-    if (packets.size() < 100) {
-      RTC_LOG(LS_WARNING) << "Can't estimate the RTP clock frequency with "
-                          << packets.size() << " packets in the stream.";
-      continue;
-    }
-    int64_t segment_end_us = parsed_log.first_log_segment().stop_time_us();
-    std::optional<uint32_t> estimated_frequency =
-        EstimateRtpClockFrequency(packets, segment_end_us);
-    if (!estimated_frequency)
-      continue;
-    const double frequency_hz = *estimated_frequency;
-    if (IsVideoSsrc(parsed_log, kIncomingPacket, stream.ssrc) &&
-        frequency_hz != 90000) {
-      RTC_LOG(LS_WARNING)
-          << "Video stream should use a 90 kHz clock but appears to use "
-          << frequency_hz / 1000 << ". Discarding.";
-      continue;
-    }
-
-    auto ToCallTime = [&config](const LoggedRtpPacketIncoming& packet) {
-      return config.GetCallTimeSec(packet.log_time());
-    };
-    auto ToNetworkDelay = [frequency_hz](
-                              const LoggedRtpPacketIncoming& old_packet,
-                              const LoggedRtpPacketIncoming& new_packet) {
-      return NetworkDelayDiff_CaptureTime(old_packet, new_packet, frequency_hz);
-    };
-
-    TimeSeries capture_time_data(
-        GetStreamName(parsed_log, kIncomingPacket, stream.ssrc) +
-            " capture-time",
-        LineStyle::kLine);
-    AccumulatePairs<LoggedRtpPacketIncoming, double>(
-        ToCallTime, ToNetworkDelay, packets, &capture_time_data);
-    plot->AppendTimeSeries(std::move(capture_time_data));
-
-    TimeSeries send_time_data(
-        GetStreamName(parsed_log, kIncomingPacket, stream.ssrc) +
-            " abs-send-time",
-        LineStyle::kLine);
-    AccumulatePairs<LoggedRtpPacketIncoming, double>(
-        ToCallTime, NetworkDelayDiff_AbsSendTime, packets, &send_time_data);
-    plot->AppendTimeSeriesIfNotEmpty(std::move(send_time_data));
-  }
-
-  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
-                 kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Delay (ms)", kBottomMargin, kTopMargin);
-  plot->SetTitle("Incoming network delay (relative to first packet)");
-}
-
-// Plot the fraction of packets lost (as perceived by the loss-based BWE).
-void CreateFractionLossGraph(const ParsedRtcEventLog& parsed_log,
-                             const AnalyzerConfig& config,
-                             Plot* plot) {
-  TimeSeries time_series("Fraction lost", LineStyle::kLine,
-                         PointStyle::kHighlight);
-  for (auto& bwe_update : parsed_log.bwe_loss_updates()) {
-    float x = config.GetCallTimeSec(bwe_update.log_time());
-    float y = static_cast<float>(bwe_update.fraction_lost) / 255 * 100;
-    time_series.points.emplace_back(x, y);
-  }
-
-  plot->AppendTimeSeries(std::move(time_series));
-  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
-                 kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 10, "Loss rate (in %)", kBottomMargin, kTopMargin);
-  plot->SetTitle("Outgoing packet loss (as reported by BWE)");
-}
-
-// Plot the total bandwidth used by all RTP streams.
-void CreateTotalIncomingBitrateGraph(const ParsedRtcEventLog& parsed_log,
-                                     const AnalyzerConfig& config,
-                                     Plot* plot) {
-  // TODO(terelius): This could be provided by the parser.
-  std::multimap<Timestamp, size_t> packets_in_order;
-  for (const auto& stream : parsed_log.incoming_rtp_packets_by_ssrc()) {
-    for (const LoggedRtpPacketIncoming& packet : stream.incoming_packets)
-      packets_in_order.insert(
-          std::make_pair(packet.rtp.log_time(), packet.rtp.total_length));
-  }
-
-  auto window_begin = packets_in_order.begin();
-  auto window_end = packets_in_order.begin();
-  size_t bytes_in_window = 0;
-
-  if (!packets_in_order.empty()) {
-    // Calculate a moving average of the bitrate and store in a TimeSeries.
-    TimeSeries bitrate_series("Bitrate", LineStyle::kLine);
-    for (Timestamp time = config.begin_time_;
-         time < config.end_time_ + config.step_; time += config.step_) {
-      while (window_end != packets_in_order.end() && window_end->first < time) {
-        bytes_in_window += window_end->second;
-        ++window_end;
-      }
-      while (window_begin != packets_in_order.end() &&
-             window_begin->first < time - config.window_duration_) {
-        RTC_DCHECK_LE(window_begin->second, bytes_in_window);
-        bytes_in_window -= window_begin->second;
-        ++window_begin;
-      }
-      float window_duration_in_seconds =
-          static_cast<float>(config.window_duration_.us()) /
-          kNumMicrosecsPerSec;
-      float x = config.GetCallTimeSec(time);
-      float y = bytes_in_window * 8 / window_duration_in_seconds / 1000;
-      bitrate_series.points.emplace_back(x, y);
-    }
-    plot->AppendTimeSeries(std::move(bitrate_series));
-  }
-
-  // Overlay the outgoing REMB over incoming bitrate.
-  TimeSeries remb_series("Remb", LineStyle::kStep);
-  for (const auto& rtcp : parsed_log.rembs(kOutgoingPacket)) {
-    float x = config.GetCallTimeSec(rtcp.log_time());
-    float y = static_cast<float>(rtcp.remb.bitrate_bps()) / 1000;
-    remb_series.points.emplace_back(x, y);
-  }
-  plot->AppendTimeSeriesIfNotEmpty(std::move(remb_series));
-
-  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
-                 kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin);
-  plot->SetTitle("Incoming RTP bitrate");
-}
-
-// Plot the total bandwidth used by all RTP streams.
-void CreateTotalOutgoingBitrateGraph(const ParsedRtcEventLog& parsed_log,
-                                     const AnalyzerConfig& config,
-                                     Plot* plot,
-                                     bool show_detector_state,
-                                     bool show_alr_state,
-                                     bool show_link_capacity) {
-  // TODO(terelius): This could be provided by the parser.
-  std::multimap<Timestamp, size_t> packets_in_order;
-  for (const auto& stream : parsed_log.outgoing_rtp_packets_by_ssrc()) {
-    for (const LoggedRtpPacketOutgoing& packet : stream.outgoing_packets)
-      packets_in_order.insert(
-          std::make_pair(packet.rtp.log_time(), packet.rtp.total_length));
-  }
-
-  auto window_begin = packets_in_order.begin();
-  auto window_end = packets_in_order.begin();
-  size_t bytes_in_window = 0;
-
-  if (!packets_in_order.empty()) {
-    // Calculate a moving average of the bitrate and store in a TimeSeries.
-    TimeSeries bitrate_series("Bitrate", LineStyle::kLine);
-    for (Timestamp time = config.begin_time_;
-         time < config.end_time_ + config.step_; time += config.step_) {
-      while (window_end != packets_in_order.end() && window_end->first < time) {
-        bytes_in_window += window_end->second;
-        ++window_end;
-      }
-      while (window_begin != packets_in_order.end() &&
-             window_begin->first < time - config.window_duration_) {
-        RTC_DCHECK_LE(window_begin->second, bytes_in_window);
-        bytes_in_window -= window_begin->second;
-        ++window_begin;
-      }
-      float window_duration_in_seconds =
-          static_cast<float>(config.window_duration_.us()) /
-          kNumMicrosecsPerSec;
-      float x = config.GetCallTimeSec(time);
-      float y = bytes_in_window * 8 / window_duration_in_seconds / 1000;
-      bitrate_series.points.emplace_back(x, y);
-    }
-    plot->AppendTimeSeries(std::move(bitrate_series));
-  }
-
-  // Overlay the send-side bandwidth estimate over the outgoing bitrate.
-  TimeSeries loss_series("Loss-based estimate", LineStyle::kStep);
-  for (auto& loss_update : parsed_log.bwe_loss_updates()) {
-    float x = config.GetCallTimeSec(loss_update.log_time());
-    float y = static_cast<float>(loss_update.bitrate_bps) / 1000;
-    loss_series.points.emplace_back(x, y);
-  }
-
-  TimeSeries link_capacity_lower_series("Link-capacity-lower",
-                                        LineStyle::kStep);
-  TimeSeries link_capacity_upper_series("Link-capacity-upper",
-                                        LineStyle::kStep);
-  for (auto& remote_estimate_event : parsed_log.remote_estimate_events()) {
-    float x = config.GetCallTimeSec(remote_estimate_event.log_time());
-    if (remote_estimate_event.link_capacity_lower.has_value()) {
-      float link_capacity_lower = static_cast<float>(
-          remote_estimate_event.link_capacity_lower.value().kbps());
-      link_capacity_lower_series.points.emplace_back(x, link_capacity_lower);
-    }
-    if (remote_estimate_event.link_capacity_upper.has_value()) {
-      float link_capacity_upper = static_cast<float>(
-          remote_estimate_event.link_capacity_upper.value().kbps());
-      link_capacity_upper_series.points.emplace_back(x, link_capacity_upper);
-    }
-  }
-
-  TimeSeries delay_series("Delay-based estimate", LineStyle::kStep);
-  IntervalSeries overusing_series("Overusing", "#ff8e82",
-                                  IntervalSeries::kHorizontal);
-  IntervalSeries underusing_series("Underusing", "#5092fc",
-                                   IntervalSeries::kHorizontal);
-  IntervalSeries normal_series("Normal", "#c4ffc4",
-                               IntervalSeries::kHorizontal);
-  IntervalSeries* last_series = &normal_series;
-  float last_detector_switch = 0.0;
-
-  BandwidthUsage last_detector_state = BandwidthUsage::kBwNormal;
-
-  for (auto& delay_update : parsed_log.bwe_delay_updates()) {
-    float x = config.GetCallTimeSec(delay_update.log_time());
-    float y = static_cast<float>(delay_update.bitrate_bps) / 1000;
-
-    if (last_detector_state != delay_update.detector_state) {
-      last_series->intervals.emplace_back(last_detector_switch, x);
-      last_detector_state = delay_update.detector_state;
-      last_detector_switch = x;
-
-      switch (delay_update.detector_state) {
-        case BandwidthUsage::kBwNormal:
-          last_series = &normal_series;
-          break;
-        case BandwidthUsage::kBwUnderusing:
-          last_series = &underusing_series;
-          break;
-        case BandwidthUsage::kBwOverusing:
-          last_series = &overusing_series;
-          break;
-        case BandwidthUsage::kLast:
-          RTC_DCHECK_NOTREACHED();
-      }
-    }
-
-    delay_series.points.emplace_back(x, y);
-  }
-
-  RTC_CHECK(last_series);
-  last_series->intervals.emplace_back(last_detector_switch,
-                                      config.CallEndTimeSec());
-
-  TimeSeries scream_series("Scream target rate", LineStyle::kStep);
-  for (auto& scream_update : parsed_log.bwe_scream_updates()) {
-    float x = config.GetCallTimeSec(scream_update.log_time());
-    float y = static_cast<float>(scream_update.target_rate.kbps());
-    scream_series.points.emplace_back(x, y);
-  }
-
-  TimeSeries created_series("Probe cluster created.", LineStyle::kNone,
-                            PointStyle::kHighlight);
-  for (auto& cluster : parsed_log.bwe_probe_cluster_created_events()) {
-    float x = config.GetCallTimeSec(cluster.log_time());
-    float y = static_cast<float>(cluster.bitrate_bps) / 1000;
-    created_series.points.emplace_back(x, y);
-  }
-
-  TimeSeries result_series("Probing results.", LineStyle::kNone,
-                           PointStyle::kHighlight);
-  for (auto& result : parsed_log.bwe_probe_success_events()) {
-    float x = config.GetCallTimeSec(result.log_time());
-    float y = static_cast<float>(result.bitrate_bps) / 1000;
-    result_series.points.emplace_back(x, y);
-  }
-
-  TimeSeries probe_failures_series("Probe failed", LineStyle::kNone,
-                                   PointStyle::kHighlight);
-  for (auto& failure : parsed_log.bwe_probe_failure_events()) {
-    float x = config.GetCallTimeSec(failure.log_time());
-    probe_failures_series.points.emplace_back(x, 0);
-  }
-
-  IntervalSeries alr_state("ALR", "#555555", IntervalSeries::kHorizontal);
-  bool previously_in_alr = false;
-  Timestamp alr_start = Timestamp::Zero();
-  for (auto& alr : parsed_log.alr_state_events()) {
-    float y = config.GetCallTimeSec(alr.log_time());
-    if (!previously_in_alr && alr.in_alr) {
-      alr_start = alr.log_time();
-      previously_in_alr = true;
-    } else if (previously_in_alr && !alr.in_alr) {
-      float x = config.GetCallTimeSec(alr_start);
-      alr_state.intervals.emplace_back(x, y);
-      previously_in_alr = false;
-    }
-  }
-
-  if (previously_in_alr) {
-    float x = config.GetCallTimeSec(alr_start);
-    float y = config.GetCallTimeSec(config.end_time_);
-    alr_state.intervals.emplace_back(x, y);
-  }
-
-  if (show_detector_state) {
-    plot->AppendIntervalSeries(std::move(overusing_series));
-    plot->AppendIntervalSeries(std::move(underusing_series));
-    plot->AppendIntervalSeries(std::move(normal_series));
-  }
-
-  if (show_alr_state) {
-    plot->AppendIntervalSeries(std::move(alr_state));
-  }
-
-  if (show_link_capacity) {
-    plot->AppendTimeSeriesIfNotEmpty(std::move(link_capacity_lower_series));
-    plot->AppendTimeSeriesIfNotEmpty(std::move(link_capacity_upper_series));
-  }
-
-  plot->AppendTimeSeries(std::move(loss_series));
-  plot->AppendTimeSeriesIfNotEmpty(std::move(probe_failures_series));
-  plot->AppendTimeSeries(std::move(delay_series));
-  plot->AppendTimeSeriesIfNotEmpty(std::move(scream_series));
-  plot->AppendTimeSeries(std::move(created_series));
-  plot->AppendTimeSeries(std::move(result_series));
-
-  // Overlay the incoming REMB over the outgoing bitrate.
-  TimeSeries remb_series("Remb", LineStyle::kStep);
-  for (const auto& rtcp : parsed_log.rembs(kIncomingPacket)) {
-    float x = config.GetCallTimeSec(rtcp.log_time());
-    float y = static_cast<float>(rtcp.remb.bitrate_bps()) / 1000;
-    remb_series.points.emplace_back(x, y);
-  }
-  plot->AppendTimeSeriesIfNotEmpty(std::move(remb_series));
-
-  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
-                 kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin);
-  plot->SetTitle("Outgoing RTP bitrate");
-}
-
-void CreateGoogCcSimulationGraph(const ParsedRtcEventLog& parsed_log,
-                                 const AnalyzerConfig& config,
-                                 Plot* plot) {
-  TimeSeries target_rates("Simulated target rate", LineStyle::kStep,
-                          PointStyle::kHighlight);
-  TimeSeries delay_based("Logged delay-based estimate", LineStyle::kStep,
-                         PointStyle::kHighlight);
-  TimeSeries loss_based("Logged loss-based estimate", LineStyle::kStep,
-                        PointStyle::kHighlight);
-  TimeSeries probe_results("Logged probe success", LineStyle::kNone,
-                           PointStyle::kHighlight);
-
-  LogBasedNetworkControllerSimulation simulation(
-      config.env_, std::make_unique<GoogCcNetworkControllerFactory>(),
-      [&](const NetworkControlUpdate& update, Timestamp at_time) {
-        if (update.target_rate) {
-          target_rates.points.emplace_back(
-              config.GetCallTimeSec(at_time),
-              update.target_rate->target_rate.kbps<float>());
-        }
-      });
-
-  simulation.ProcessEventsInLog(parsed_log);
-  for (const auto& logged : parsed_log.bwe_delay_updates())
-    delay_based.points.emplace_back(config.GetCallTimeSec(logged.log_time()),
-                                    logged.bitrate_bps / 1000);
-  for (const auto& logged : parsed_log.bwe_probe_success_events())
-    probe_results.points.emplace_back(config.GetCallTimeSec(logged.log_time()),
-                                      logged.bitrate_bps / 1000);
-  for (const auto& logged : parsed_log.bwe_loss_updates())
-    loss_based.points.emplace_back(config.GetCallTimeSec(logged.log_time()),
-                                   logged.bitrate_bps / 1000);
-
-  plot->AppendTimeSeries(std::move(delay_based));
-  plot->AppendTimeSeries(std::move(loss_based));
-  plot->AppendTimeSeries(std::move(probe_results));
-  plot->AppendTimeSeries(std::move(target_rates));
-
-  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
-                 kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 10, "Bitrate (kbps)", kBottomMargin, kTopMargin);
-  plot->SetTitle("Simulated BWE behavior");
-}
-
-void CreateScreamSimulationBitrateGraph(const ParsedRtcEventLog& parsed_log,
-                                        const AnalyzerConfig& config,
-                                        Plot* plot) {
-  TimeSeries target_rate_series("Target rate", LineStyle::kStep);
-  TimeSeries pacing_rate_series("Pacing rate", LineStyle::kStep);
-  TimeSeries send_rate_series("Send rate", LineStyle::kStep);
-
-  LogScreamSimulation simulation({.rate_window = config.window_duration_},
-                                 config.env_);
-  simulation.ProcessEventsInLog(parsed_log);
-
-  for (const LogScreamSimulation::State& state : simulation.updates()) {
-    target_rate_series.points.emplace_back(config.GetCallTimeSec(state.time),
-                                           state.target_rate.bps() / 1000);
-    pacing_rate_series.points.emplace_back(config.GetCallTimeSec(state.time),
-                                           state.pacing_rate.bps() / 1000);
-    send_rate_series.points.emplace_back(config.GetCallTimeSec(state.time),
-                                         state.send_rate.bps() / 1000);
-  }
-  plot->AppendTimeSeries(std::move(target_rate_series));
-  plot->AppendTimeSeries(std::move(pacing_rate_series));
-  plot->AppendTimeSeries(std::move(send_rate_series));
-
-  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
-                 kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 100, "Kbps", kBottomMargin, kTopMargin);
-  plot->SetTitle("Simulated Scream rates");
-}
-
-void CreateScreamSimulationRefWindowGraph(const ParsedRtcEventLog& parsed_log,
-                                          const AnalyzerConfig& config,
-                                          Plot* plot) {
-  using SendWindowUsage = LogScreamSimulation::State::SendWindowUsage;
-  TimeSeries ref_window_series("RefWindow", LineStyle::kStep);
-  TimeSeries ref_window_i_series("RefWindowI", LineStyle::kStep);
-  TimeSeries max_data_in_flight("Max allowed data in flight", LineStyle::kStep);
-  TimeSeries data_in_flight("Data in flight", LineStyle::kStep);
-  IntervalSeries send_window_above_max_series(
-      "Data in flight > Max allowed", "#ff8e82", IntervalSeries::kHorizontal);
-  IntervalSeries send_window_below_ref_window_series(
-      "Data in flight < RefWindow", "#c5dff2", IntervalSeries::kHorizontal);
-  IntervalSeries send_window_above_ref_window_series(
-      "Data in flight >= RefWindow", "#b9fad8", IntervalSeries::kHorizontal);
-  IntervalSeries* last_series = &send_window_below_ref_window_series;
-
-  LogScreamSimulation simulation({.rate_window = config.window_duration_},
-                                 config.env_);
-  simulation.ProcessEventsInLog(parsed_log);
-  if (simulation.updates().empty()) {
-    RTC_LOG(LS_ERROR) << "Empty simulation.";
-    return;
-  }
-
-  float send_window_state_switch =
-      config.GetCallTimeSec(simulation.updates().front().time);
-  SendWindowUsage send_window_usage = SendWindowUsage::kBelowRefWindow;
-
-  for (const LogScreamSimulation::State& state : simulation.updates()) {
-    ref_window_series.points.emplace_back(config.GetCallTimeSec(state.time),
-                                          state.ref_window.bytes());
-    ref_window_i_series.points.emplace_back(config.GetCallTimeSec(state.time),
-                                            state.ref_window_i.bytes());
-    max_data_in_flight.points.emplace_back(config.GetCallTimeSec(state.time),
-                                           state.max_data_in_flight.bytes());
-    data_in_flight.points.emplace_back(config.GetCallTimeSec(state.time),
-                                       state.data_in_flight.bytes());
-    if (state.send_window_usage != send_window_usage) {
-      last_series->intervals.emplace_back(send_window_state_switch,
-                                          config.GetCallTimeSec(state.time));
-      send_window_usage = state.send_window_usage;
-      send_window_state_switch = config.GetCallTimeSec(state.time);
-      switch (send_window_usage) {
-        case SendWindowUsage::kAboveRefWindow:
-          last_series = &send_window_above_ref_window_series;
-          break;
-        case SendWindowUsage::kBelowRefWindow:
-          last_series = &send_window_below_ref_window_series;
-          break;
-        case SendWindowUsage::kAboveScreamMax:
-          last_series = &send_window_above_max_series;
-          break;
-      }
-    }
-  }
-  last_series->intervals.emplace_back(send_window_state_switch,
-                                      config.CallEndTimeSec());
-  plot->AppendTimeSeries(std::move(ref_window_series));
-  plot->AppendTimeSeries(std::move(ref_window_i_series));
-  plot->AppendTimeSeries(std::move(max_data_in_flight));
-  plot->AppendTimeSeries(std::move(data_in_flight));
-  plot->AppendIntervalSeries(std::move(send_window_above_max_series));
-  plot->AppendIntervalSeries(std::move(send_window_below_ref_window_series));
-  plot->AppendIntervalSeries(std::move(send_window_above_ref_window_series));
-
-  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
-                 kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 10, "Bytes", kBottomMargin, kTopMargin);
-  plot->SetTitle("Simulated Scream RefWindow");
-}
-
-void CreateScreamSimulationRatiosGraph(const ParsedRtcEventLog& parsed_log,
-                                       const AnalyzerConfig& config,
-                                       Plot* plot) {
-  TimeSeries queue_delay_dev_norm_series("QueueDelayDevNorm", LineStyle::kStep);
-  TimeSeries l4s_alpha_series("L4sAlpha", LineStyle::kStep);
-  TimeSeries l4s_alpha_v_series("L4sAlphaV", LineStyle::kStep);
-  TimeSeries ref_window_scale_factor_due_to_increased_delay(
-      "RefWindowScaleFactorDueToIncreasedDelay", LineStyle::kStep);
-  TimeSeries ref_window_scale_factor_due_to_delay_variation(
-      "RefWindowScaleFactorDueToDelayVariation", LineStyle::kStep);
-
-  LogScreamSimulation simulation({.rate_window = config.window_duration_},
-                                 config.env_);
-  simulation.ProcessEventsInLog(parsed_log);
-
-  for (const LogScreamSimulation::State& state : simulation.updates()) {
-    queue_delay_dev_norm_series.points.emplace_back(
-        config.GetCallTimeSec(state.time), state.queue_delay_dev_norm);
-    l4s_alpha_series.points.emplace_back(config.GetCallTimeSec(state.time),
-                                         state.l4s_alpha);
-    l4s_alpha_v_series.points.emplace_back(config.GetCallTimeSec(state.time),
-                                           state.l4s_alpha_v);
-    ref_window_scale_factor_due_to_increased_delay.points.emplace_back(
-        config.GetCallTimeSec(state.time),
-        state.ref_window_scale_factor_due_to_increased_delay);
-    ref_window_scale_factor_due_to_delay_variation.points.emplace_back(
-        config.GetCallTimeSec(state.time),
-        state.ref_window_scale_factor_due_to_delay_variation);
-  }
-  plot->AppendTimeSeries(std::move(queue_delay_dev_norm_series));
-  plot->AppendTimeSeries(std::move(l4s_alpha_series));
-  plot->AppendTimeSeries(std::move(l4s_alpha_v_series));
-  plot->AppendTimeSeries(
-      std::move(ref_window_scale_factor_due_to_increased_delay));
-  plot->AppendTimeSeries(
-      std::move(ref_window_scale_factor_due_to_delay_variation));
-
-  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
-                 kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Ratios", kBottomMargin, kTopMargin);
-  plot->SetTitle("Simulated Scream Ratios");
-}
-
-void CreateScreamRefWindowGraph(const ParsedRtcEventLog& parsed_log,
-                                const AnalyzerConfig& config,
-                                Plot* plot) {
-  TimeSeries ref_window_series("RefWindow", LineStyle::kStep);
-  for (auto& scream_update : parsed_log.bwe_scream_updates()) {
-    float x = config.GetCallTimeSec(scream_update.log_time());
-    float y = static_cast<float>(scream_update.ref_window.bytes());
-    ref_window_series.points.emplace_back(x, y);
-  }
-  plot->AppendTimeSeries(std::move(ref_window_series));
-
-  TimeSeries data_in_flight_series("Data in flight", LineStyle::kLine);
-  for (auto& scream_update : parsed_log.bwe_scream_updates()) {
-    float x = config.GetCallTimeSec(scream_update.log_time());
-    float y = static_cast<float>(scream_update.data_in_flight.bytes());
-    data_in_flight_series.points.emplace_back(x, y);
-  }
-  plot->AppendTimeSeries(std::move(data_in_flight_series));
-
-  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
-                 kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 3000, "Bytes", kBottomMargin, kTopMargin);
-  plot->SetTitle("Scream Ref Window");
-}
-
-void CreateScreamDelayEstimateGraph(const ParsedRtcEventLog& parsed_log,
-                                    const AnalyzerConfig& config,
-                                    Plot* plot) {
-  TimeSeries smoothed_rtt_series("Smoothed RTT", LineStyle::kStep);
-  TimeSeries avg_queue_delay_series("Avg queue delay", LineStyle::kStep);
-
-  for (auto& scream_update : parsed_log.bwe_scream_updates()) {
-    float x = config.GetCallTimeSec(scream_update.log_time());
-    float smoothed_rtt_ms = static_cast<float>(scream_update.smoothed_rtt.ms());
-    smoothed_rtt_series.points.emplace_back(x, smoothed_rtt_ms);
-    float avg_queue_delay_ms =
-        static_cast<float>(scream_update.avg_queue_delay.ms());
-    avg_queue_delay_series.points.emplace_back(x, avg_queue_delay_ms);
-  }
-
-  plot->AppendTimeSeries(std::move(smoothed_rtt_series));
-  plot->AppendTimeSeries(std::move(avg_queue_delay_series));
-
-  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
-                 kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 50, "Delay (ms)", kBottomMargin, kTopMargin);
-  plot->SetTitle("Scream delay estimates");
-}
-
-void CreateSendSideBweSimulationGraph(const ParsedRtcEventLog& parsed_log,
-                                      const AnalyzerConfig& config,
-                                      Plot* plot) {
-  using RtpPacketType = LoggedRtpPacketOutgoing;
-  using TransportFeedbackType = LoggedRtcpPacketTransportFeedback;
-
-  // TODO(terelius): The parser could provide a clearer view of the
-  // streams, so that we don't have to recalculate it.
-  std::multimap<int64_t, const RtpPacketType*> outgoing_rtp;
-  for (const auto& stream : parsed_log.outgoing_rtp_packets_by_ssrc()) {
-    for (const RtpPacketType& rtp_packet : stream.outgoing_packets)
-      outgoing_rtp.insert(
-          std::make_pair(rtp_packet.rtp.log_time_us(), &rtp_packet));
-  }
-
-  const std::vector<TransportFeedbackType>& incoming_rtcp =
-      parsed_log.transport_feedbacks(kIncomingPacket);
-
-  SimulatedClock clock(0);
-  BitrateObserver observer;
-  TransportFeedbackAdapter transport_feedback;
-  auto factory = GoogCcNetworkControllerFactory();
-  TimeDelta process_interval = factory.GetProcessInterval();
-  // TODO(holmer): Log the call config and use that here instead.
-  static const uint32_t kDefaultStartBitrateBps = 300000;
-  NetworkControllerConfig cc_config(config.env_);
-  cc_config.constraints.at_time = clock.CurrentTime();
-  cc_config.constraints.starting_rate =
-      DataRate::BitsPerSec(kDefaultStartBitrateBps);
-  auto goog_cc = factory.Create(cc_config);
-
-  TimeSeries time_series("Delay-based estimate", LineStyle::kStep,
-                         PointStyle::kHighlight);
-  TimeSeries acked_time_series("Raw acked bitrate", LineStyle::kLine,
-                               PointStyle::kHighlight);
-  TimeSeries robust_time_series("Robust throughput estimate", LineStyle::kLine,
-                                PointStyle::kHighlight);
-  TimeSeries acked_estimate_time_series("Ackednowledged bitrate estimate",
-                                        LineStyle::kLine,
-                                        PointStyle::kHighlight);
-
-  auto rtp_iterator = outgoing_rtp.begin();
-  auto rtcp_iterator = incoming_rtcp.begin();
-
-  auto NextRtpTime = [&]() {
-    if (rtp_iterator != outgoing_rtp.end())
-      return static_cast<int64_t>(rtp_iterator->first);
-    return std::numeric_limits<int64_t>::max();
-  };
-
-  auto NextRtcpTime = [&]() {
-    if (rtcp_iterator != incoming_rtcp.end())
-      return static_cast<int64_t>(rtcp_iterator->log_time_us());
-    return std::numeric_limits<int64_t>::max();
-  };
-  int64_t next_process_time_us_ = std::min({NextRtpTime(), NextRtcpTime()});
-
-  auto NextProcessTime = [&]() {
-    if (rtcp_iterator != incoming_rtcp.end() ||
-        rtp_iterator != outgoing_rtp.end()) {
-      return next_process_time_us_;
-    }
-    return std::numeric_limits<int64_t>::max();
-  };
-
-  RateStatistics raw_acked_bitrate(750, 8000);
-  FieldTrials throughput_config(
-      "WebRTC-Bwe-RobustThroughputEstimatorSettings/enabled:true/");
-  std::unique_ptr<AcknowledgedBitrateEstimatorInterface>
-      robust_throughput_estimator(
-          AcknowledgedBitrateEstimatorInterface::Create(&throughput_config));
-  FieldTrials acked_bitrate_config(
-      "WebRTC-Bwe-RobustThroughputEstimatorSettings/enabled:false/");
-  std::unique_ptr<AcknowledgedBitrateEstimatorInterface>
-      acknowledged_bitrate_estimator(
-          AcknowledgedBitrateEstimatorInterface::Create(&acked_bitrate_config));
-  int64_t time_us =
-      std::min({NextRtpTime(), NextRtcpTime(), NextProcessTime()});
-  int64_t last_update_us = 0;
-  while (time_us != std::numeric_limits<int64_t>::max()) {
-    clock.AdvanceTimeMicroseconds(time_us - clock.TimeInMicroseconds());
-    if (clock.TimeInMicroseconds() >= NextRtpTime()) {
-      RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtpTime());
-      const RtpPacketType& rtp_packet = *rtp_iterator->second;
-      if (rtp_packet.rtp.header.extension.hasTransportSequenceNumber) {
-        RtpPacketToSend send_packet(/*extensions=*/nullptr);
-        send_packet.set_transport_sequence_number(
-            rtp_packet.rtp.header.extension.transportSequenceNumber);
-        send_packet.SetSsrc(rtp_packet.rtp.header.ssrc);
-        send_packet.SetSequenceNumber(rtp_packet.rtp.header.sequenceNumber);
-        send_packet.SetPayloadSize(rtp_packet.rtp.total_length -
-                                   send_packet.headers_size());
-        RTC_DCHECK_EQ(send_packet.size(), rtp_packet.rtp.total_length);
-        if (IsRtxSsrc(parsed_log, PacketDirection::kOutgoingPacket,
-                      rtp_packet.rtp.header.ssrc)) {
-          // Don't set the optional media type as we don't know if it is
-          // a retransmission, FEC or padding.
-        } else if (IsVideoSsrc(parsed_log, PacketDirection::kOutgoingPacket,
-                               rtp_packet.rtp.header.ssrc)) {
-          send_packet.set_packet_type(RtpPacketMediaType::kVideo);
-        } else if (IsAudioSsrc(parsed_log, PacketDirection::kOutgoingPacket,
-                               rtp_packet.rtp.header.ssrc)) {
-          send_packet.set_packet_type(RtpPacketMediaType::kAudio);
-        }
-        transport_feedback.AddPacket(
-            send_packet, PacedPacketInfo(),
-            0u,  // Per packet overhead bytes.,
-            Timestamp::Micros(rtp_packet.rtp.log_time_us()));
-      }
-      SentPacketInfo sent_packet;
-      sent_packet.send_time_ms = rtp_packet.rtp.log_time_ms();
-      sent_packet.info.included_in_allocation = true;
-      sent_packet.info.packet_size_bytes = rtp_packet.rtp.total_length;
-      if (rtp_packet.rtp.header.extension.hasTransportSequenceNumber) {
-        sent_packet.packet_id =
-            rtp_packet.rtp.header.extension.transportSequenceNumber;
-        sent_packet.info.included_in_feedback = true;
-      }
-      auto sent_msg = transport_feedback.ProcessSentPacket(sent_packet);
-      if (sent_msg)
-        observer.Update(goog_cc->OnSentPacket(*sent_msg));
-      ++rtp_iterator;
-    }
-    if (clock.TimeInMicroseconds() >= NextRtcpTime()) {
-      RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtcpTime());
-
-      auto feedback_msg = transport_feedback.ProcessTransportFeedback(
-          rtcp_iterator->transport_feedback, clock.CurrentTime());
-      if (feedback_msg) {
-        observer.Update(goog_cc->OnTransportPacketsFeedback(*feedback_msg));
-        std::vector<PacketResult> feedback =
-            feedback_msg->SortedByReceiveTime();
-        if (!feedback.empty()) {
-          acknowledged_bitrate_estimator->IncomingPacketFeedbackVector(
-              feedback);
-          robust_throughput_estimator->IncomingPacketFeedbackVector(feedback);
-          for (const PacketResult& packet : feedback) {
-            raw_acked_bitrate.Update(packet.sent_packet.size.bytes(),
-                                     packet.receive_time.ms());
-          }
-          std::optional<uint32_t> raw_bitrate_bps =
-              raw_acked_bitrate.Rate(feedback.back().receive_time.ms());
-          float x = config.GetCallTimeSec(clock.CurrentTime());
-          if (raw_bitrate_bps) {
-            float y = raw_bitrate_bps.value() / 1000;
-            acked_time_series.points.emplace_back(x, y);
-          }
-          std::optional<DataRate> robust_estimate =
-              robust_throughput_estimator->bitrate();
-          if (robust_estimate) {
-            float y = robust_estimate.value().kbps();
-            robust_time_series.points.emplace_back(x, y);
-          }
-          std::optional<DataRate> acked_estimate =
-              acknowledged_bitrate_estimator->bitrate();
-          if (acked_estimate) {
-            float y = acked_estimate.value().kbps();
-            acked_estimate_time_series.points.emplace_back(x, y);
-          }
-        }
-      }
-      ++rtcp_iterator;
-    }
-    if (clock.TimeInMicroseconds() >= NextProcessTime()) {
-      RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextProcessTime());
-      ProcessInterval msg;
-      msg.at_time = clock.CurrentTime();
-      observer.Update(goog_cc->OnProcessInterval(msg));
-      next_process_time_us_ += process_interval.us();
-    }
-    if (observer.GetAndResetBitrateUpdated() ||
-        time_us - last_update_us >= 1e6) {
-      uint32_t y = observer.last_bitrate_bps() / 1000;
-      float x = config.GetCallTimeSec(clock.CurrentTime());
-      time_series.points.emplace_back(x, y);
-      last_update_us = time_us;
-    }
-    time_us = std::min({NextRtpTime(), NextRtcpTime(), NextProcessTime()});
-  }
-  // Add the data set to the plot.
-  plot->AppendTimeSeries(std::move(time_series));
-  plot->AppendTimeSeries(std::move(robust_time_series));
-  plot->AppendTimeSeries(std::move(acked_time_series));
-  plot->AppendTimeSeriesIfNotEmpty(std::move(acked_estimate_time_series));
-
-  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
-                 kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 10, "Bitrate (kbps)", kBottomMargin, kTopMargin);
-  plot->SetTitle("Simulated send-side BWE behavior");
-}
-
-void CreateReceiveSideBweSimulationGraph(const ParsedRtcEventLog& parsed_log,
-                                         const AnalyzerConfig& config,
-                                         Plot* plot) {
-  using RtpPacketType = LoggedRtpPacketIncoming;
-  class RembInterceptor {
-   public:
-    void SendRemb(uint32_t bitrate_bps, std::vector<uint32_t> ssrcs) {
-      last_bitrate_bps_ = bitrate_bps;
-      bitrate_updated_ = true;
-    }
-    uint32_t last_bitrate_bps() const { return last_bitrate_bps_; }
-    bool GetAndResetBitrateUpdated() {
-      bool bitrate_updated = bitrate_updated_;
-      bitrate_updated_ = false;
-      return bitrate_updated;
-    }
-
-   private:
-    // We don't know the start bitrate, but assume that it is the default 300
-    // kbps.
-    uint32_t last_bitrate_bps_ = 300000;
-    bool bitrate_updated_ = false;
-  };
-
-  std::multimap<int64_t, const RtpPacketType*> incoming_rtp;
-
-  for (const auto& stream : parsed_log.incoming_rtp_packets_by_ssrc()) {
-    if (IsVideoSsrc(parsed_log, kIncomingPacket, stream.ssrc)) {
-      for (const auto& rtp_packet : stream.incoming_packets)
-        incoming_rtp.insert(
-            std::make_pair(rtp_packet.rtp.log_time_us(), &rtp_packet));
-    }
-  }
-
-  SimulatedClock clock(0);
-  EnvironmentFactory env_factory(config.env_);
-  env_factory.Set(&clock);
-  RembInterceptor remb_interceptor;
-  ReceiveSideCongestionController rscc(
-      env_factory.Create(), [](auto...) {},
-      absl::bind_front(&RembInterceptor::SendRemb, &remb_interceptor));
-  // TODO(holmer): Log the call config and use that here instead.
-  // static const uint32_t kDefaultStartBitrateBps = 300000;
-  // rscc.SetBweBitrates(0, kDefaultStartBitrateBps, -1);
-
-  TimeSeries time_series("Receive side estimate", LineStyle::kLine,
-                         PointStyle::kHighlight);
-  TimeSeries acked_time_series("Received bitrate", LineStyle::kLine);
-
-  RateStatistics acked_bitrate(250, 8000);
-  int64_t last_update_us = 0;
-  for (const auto& kv : incoming_rtp) {
-    const RtpPacketType& packet = *kv.second;
-
-    RtpPacketReceived rtp_packet = RtpPacketForBWEFromHeader(packet.rtp.header);
-    rtp_packet.set_arrival_time(packet.rtp.log_time());
-    rtp_packet.SetPayloadSize(packet.rtp.total_length -
-                              rtp_packet.headers_size());
-
-    clock.AdvanceTime(rtp_packet.arrival_time() - clock.CurrentTime());
-    rscc.OnReceivedPacket(rtp_packet, MediaType::VIDEO);
-    int64_t arrival_time_ms = packet.rtp.log_time().ms();
-    acked_bitrate.Update(packet.rtp.total_length, arrival_time_ms);
-    std::optional<uint32_t> bitrate_bps = acked_bitrate.Rate(arrival_time_ms);
-    if (bitrate_bps) {
-      uint32_t y = *bitrate_bps / 1000;
-      float x = config.GetCallTimeSec(clock.CurrentTime());
-      acked_time_series.points.emplace_back(x, y);
-    }
-    if (remb_interceptor.GetAndResetBitrateUpdated() ||
-        clock.TimeInMicroseconds() - last_update_us >= 1e6) {
-      uint32_t y = remb_interceptor.last_bitrate_bps() / 1000;
-      float x = config.GetCallTimeSec(clock.CurrentTime());
-      time_series.points.emplace_back(x, y);
-      last_update_us = clock.TimeInMicroseconds();
-    }
-  }
-  // Add the data set to the plot.
-  plot->AppendTimeSeries(std::move(time_series));
-  plot->AppendTimeSeries(std::move(acked_time_series));
-
-  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
-                 kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 10, "Bitrate (kbps)", kBottomMargin, kTopMargin);
-  plot->SetTitle("Simulated receive-side BWE behavior");
-}
-
-void CreateNetworkDelayFeedbackGraph(const ParsedRtcEventLog& parsed_log,
-                                     const AnalyzerConfig& config,
-                                     Plot* plot) {
-  TimeSeries time_series("Network delay", LineStyle::kLine,
-                         PointStyle::kHighlight);
-  int64_t min_send_receive_diff_ms = std::numeric_limits<int64_t>::max();
-  int64_t min_rtt_ms = std::numeric_limits<int64_t>::max();
-
-  std::vector<MatchedSendArrivalTimes> matched_rtp_rtcp =
-      GetNetworkTrace(parsed_log);
-  absl::c_stable_sort(matched_rtp_rtcp, [](const MatchedSendArrivalTimes& a,
-                                           const MatchedSendArrivalTimes& b) {
-    return a.feedback_arrival_time_ms < b.feedback_arrival_time_ms ||
-           (a.feedback_arrival_time_ms == b.feedback_arrival_time_ms &&
-            a.arrival_time_ms < b.arrival_time_ms);
-  });
-  for (const auto& packet : matched_rtp_rtcp) {
-    if (packet.arrival_time_ms == MatchedSendArrivalTimes::kNotReceived)
-      continue;
-    float x = config.GetCallTimeSecFromMs(packet.feedback_arrival_time_ms);
-    int64_t y = packet.arrival_time_ms - packet.send_time_ms;
-    int64_t rtt_ms = packet.feedback_arrival_time_ms - packet.send_time_ms;
-    min_rtt_ms = std::min(rtt_ms, min_rtt_ms);
-    min_send_receive_diff_ms = std::min(y, min_send_receive_diff_ms);
-    time_series.points.emplace_back(x, y);
-  }
-
-  // We assume that the base network delay (w/o queues) is equal to half
-  // the minimum RTT. Therefore rescale the delays by subtracting the minimum
-  // observed 1-ways delay and add half the minimum RTT.
-  const int64_t estimated_clock_offset_ms =
-      min_send_receive_diff_ms - min_rtt_ms / 2;
-  for (TimeSeriesPoint& point : time_series.points)
-    point.y -= estimated_clock_offset_ms;
-
-  // Add the data set to the plot.
-  plot->AppendTimeSeriesIfNotEmpty(std::move(time_series));
-
-  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
-                 kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 10, "Delay (ms)", kBottomMargin, kTopMargin);
-  plot->SetTitle("Outgoing network delay (based on per-packet feedback)");
-}
-
-void CreatePacerDelayGraph(const ParsedRtcEventLog& parsed_log,
-                           const AnalyzerConfig& config,
-                           Plot* plot) {
-  for (const auto& stream : parsed_log.outgoing_rtp_packets_by_ssrc()) {
-    const std::vector<LoggedRtpPacketOutgoing>& packets =
-        stream.outgoing_packets;
-
-    if (IsRtxSsrc(parsed_log, kOutgoingPacket, stream.ssrc)) {
-      continue;
-    }
-
-    if (packets.size() < 2) {
-      RTC_LOG(LS_WARNING)
-          << "Can't estimate a the RTP clock frequency or the "
-             "pacer delay with less than 2 packets in the stream";
-      continue;
-    }
-    int64_t segment_end_us = parsed_log.first_log_segment().stop_time_us();
-    std::optional<uint32_t> estimated_frequency =
-        EstimateRtpClockFrequency(packets, segment_end_us);
-    if (!estimated_frequency)
-      continue;
-    if (IsVideoSsrc(parsed_log, kOutgoingPacket, stream.ssrc) &&
-        *estimated_frequency != 90000) {
-      RTC_LOG(LS_WARNING)
-          << "Video stream should use a 90 kHz clock but appears to use "
-          << *estimated_frequency / 1000 << ". Discarding.";
-      continue;
-    }
-
-    TimeSeries pacer_delay_series(
-        GetStreamName(parsed_log, kOutgoingPacket, stream.ssrc) + "(" +
-            std::to_string(*estimated_frequency / 1000) + " kHz)",
-        LineStyle::kLine, PointStyle::kHighlight);
-    SeqNumUnwrapper<uint32_t> timestamp_unwrapper;
-    uint64_t first_capture_timestamp =
-        timestamp_unwrapper.Unwrap(packets.front().rtp.header.timestamp);
-    uint64_t first_send_timestamp = packets.front().rtp.log_time_us();
-    for (const auto& packet : packets) {
-      double capture_time_ms = (static_cast<double>(timestamp_unwrapper.Unwrap(
-                                    packet.rtp.header.timestamp)) -
-                                first_capture_timestamp) /
-                               *estimated_frequency * 1000;
-      double send_time_ms =
-          static_cast<double>(packet.rtp.log_time_us() - first_send_timestamp) /
-          1000;
-      float x = config.GetCallTimeSec(packet.rtp.log_time());
-      float y = send_time_ms - capture_time_ms;
-      pacer_delay_series.points.emplace_back(x, y);
-    }
-    plot->AppendTimeSeries(std::move(pacer_delay_series));
-  }
-
-  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
-                 kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 10, "Pacer delay (ms)", kBottomMargin, kTopMargin);
-  plot->SetTitle(
-      "Delay from capture to send time. (First packet normalized to 0.)");
-}
-
-}  // namespace webrtc
diff --git a/rtc_tools/rtc_event_log_visualizer/analyze_bwe.h b/rtc_tools/rtc_event_log_visualizer/analyze_bwe.h
deleted file mode 100644
index fd4b041c78..0000000000
--- a/rtc_tools/rtc_event_log_visualizer/analyze_bwe.h
+++ /dev/null
@@ -1,81 +0,0 @@
-/*
- *  Copyright (c) 2026 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZE_BWE_H_
-#define RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZE_BWE_H_
-
-#include "logging/rtc_event_log/rtc_event_log_parser.h"
-#include "rtc_tools/rtc_event_log_visualizer/analyzer_common.h"
-#include "rtc_tools/rtc_event_log_visualizer/plot_base.h"
-
-namespace webrtc {
-
-void CreateIncomingDelayGraph(const ParsedRtcEventLog& parsed_log,
-                              const AnalyzerConfig& config,
-                              Plot* plot);
-
-void CreateFractionLossGraph(const ParsedRtcEventLog& parsed_log,
-                             const AnalyzerConfig& config,
-                             Plot* plot);
-
-void CreateTotalIncomingBitrateGraph(const ParsedRtcEventLog& parsed_log,
-                                     const AnalyzerConfig& config,
-                                     Plot* plot);
-
-void CreateTotalOutgoingBitrateGraph(const ParsedRtcEventLog& parsed_log,
-                                     const AnalyzerConfig& config,
-                                     Plot* plot,
-                                     bool show_detector_state = false,
-                                     bool show_alr_state = false,
-                                     bool show_link_capacity = false);
-
-void CreateGoogCcSimulationGraph(const ParsedRtcEventLog& parsed_log,
-                                 const AnalyzerConfig& config,
-                                 Plot* plot);
-
-void CreateScreamSimulationBitrateGraph(const ParsedRtcEventLog& parsed_log,
-                                        const AnalyzerConfig& config,
-                                        Plot* plot);
-
-void CreateScreamSimulationRefWindowGraph(const ParsedRtcEventLog& parsed_log,
-                                          const AnalyzerConfig& config,
-                                          Plot* plot);
-
-void CreateScreamSimulationRatiosGraph(const ParsedRtcEventLog& parsed_log,
-                                       const AnalyzerConfig& config,
-                                       Plot* plot);
-
-void CreateScreamRefWindowGraph(const ParsedRtcEventLog& parsed_log,
-                                const AnalyzerConfig& config,
-                                Plot* plot);
-
-void CreateScreamDelayEstimateGraph(const ParsedRtcEventLog& parsed_log,
-                                    const AnalyzerConfig& config,
-                                    Plot* plot);
-
-void CreateSendSideBweSimulationGraph(const ParsedRtcEventLog& parsed_log,
-                                      const AnalyzerConfig& config,
-                                      Plot* plot);
-
-void CreateReceiveSideBweSimulationGraph(const ParsedRtcEventLog& parsed_log,
-                                         const AnalyzerConfig& config,
-                                         Plot* plot);
-
-void CreateNetworkDelayFeedbackGraph(const ParsedRtcEventLog& parsed_log,
-                                     const AnalyzerConfig& config,
-                                     Plot* plot);
-
-void CreatePacerDelayGraph(const ParsedRtcEventLog& parsed_log,
-                           const AnalyzerConfig& config,
-                           Plot* plot);
-
-}  // namespace webrtc
-
-#endif  // RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZE_BWE_H_
diff --git a/rtc_tools/rtc_event_log_visualizer/analyze_connectivity.cc b/rtc_tools/rtc_event_log_visualizer/analyze_connectivity.cc
deleted file mode 100644
index b87a108b96..0000000000
--- a/rtc_tools/rtc_event_log_visualizer/analyze_connectivity.cc
+++ /dev/null
@@ -1,285 +0,0 @@
-/*
- *  Copyright (c) 2026 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "rtc_tools/rtc_event_log_visualizer/analyze_connectivity.h"
-
-#include <cstddef>
-#include <cstdint>
-#include <map>
-#include <string>
-#include <utility>
-#include <vector>
-
-#include "absl/strings/string_view.h"
-#include "api/candidate.h"
-#include "api/dtls_transport_interface.h"
-#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair.h"
-#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.h"
-#include "logging/rtc_event_log/rtc_event_log_parser.h"
-#include "rtc_base/checks.h"
-#include "rtc_base/strings/string_builder.h"
-#include "rtc_tools/rtc_event_log_visualizer/analyzer_common.h"
-#include "rtc_tools/rtc_event_log_visualizer/plot_base.h"
-
-namespace webrtc {
-
-namespace {
-
-const char kUnknownEnumValue[] = "unknown";
-
-// TODO(tommi): This should be "host".
-const char kIceCandidateTypeLocal[] = "local";
-// TODO(tommi): This should be "srflx".
-const char kIceCandidateTypeStun[] = "stun";
-const char kIceCandidateTypePrflx[] = "prflx";
-const char kIceCandidateTypeRelay[] = "relay";
-
-const char kProtocolUdp[] = "udp";
-const char kProtocolTcp[] = "tcp";
-const char kProtocolSsltcp[] = "ssltcp";
-const char kProtocolTls[] = "tls";
-
-const char kAddressFamilyIpv4[] = "ipv4";
-const char kAddressFamilyIpv6[] = "ipv6";
-
-const char kNetworkTypeEthernet[] = "ethernet";
-const char kNetworkTypeLoopback[] = "loopback";
-const char kNetworkTypeWifi[] = "wifi";
-const char kNetworkTypeVpn[] = "vpn";
-const char kNetworkTypeCellular[] = "cellular";
-
-absl::string_view GetIceCandidateTypeAsString(IceCandidateType type) {
-  switch (type) {
-    case IceCandidateType::kHost:
-      return kIceCandidateTypeLocal;
-    case IceCandidateType::kSrflx:
-      return kIceCandidateTypeStun;
-    case IceCandidateType::kPrflx:
-      return kIceCandidateTypePrflx;
-    case IceCandidateType::kRelay:
-      return kIceCandidateTypeRelay;
-    default:
-      RTC_DCHECK_NOTREACHED();
-      return kUnknownEnumValue;
-  }
-}
-
-std::string GetProtocolAsString(IceCandidatePairProtocol protocol) {
-  switch (protocol) {
-    case IceCandidatePairProtocol::kUdp:
-      return kProtocolUdp;
-    case IceCandidatePairProtocol::kTcp:
-      return kProtocolTcp;
-    case IceCandidatePairProtocol::kSsltcp:
-      return kProtocolSsltcp;
-    case IceCandidatePairProtocol::kTls:
-      return kProtocolTls;
-    default:
-      return kUnknownEnumValue;
-  }
-}
-
-std::string GetAddressFamilyAsString(IceCandidatePairAddressFamily family) {
-  switch (family) {
-    case IceCandidatePairAddressFamily::kIpv4:
-      return kAddressFamilyIpv4;
-    case IceCandidatePairAddressFamily::kIpv6:
-      return kAddressFamilyIpv6;
-    default:
-      return kUnknownEnumValue;
-  }
-}
-
-std::string GetNetworkTypeAsString(IceCandidateNetworkType type) {
-  switch (type) {
-    case IceCandidateNetworkType::kEthernet:
-      return kNetworkTypeEthernet;
-    case IceCandidateNetworkType::kLoopback:
-      return kNetworkTypeLoopback;
-    case IceCandidateNetworkType::kWifi:
-      return kNetworkTypeWifi;
-    case IceCandidateNetworkType::kVpn:
-      return kNetworkTypeVpn;
-    case IceCandidateNetworkType::kCellular:
-      return kNetworkTypeCellular;
-    default:
-      return kUnknownEnumValue;
-  }
-}
-
-std::string GetCandidatePairLogDescriptionAsString(
-    const LoggedIceCandidatePairConfig& config) {
-  // Example: stun:wifi->relay(tcp):cellular@udp:ipv4
-  // represents a pair of a local server-reflexive candidate on a WiFi network
-  // and a remote relay candidate using TCP as the relay protocol on a cell
-  // network, when the candidate pair communicates over UDP using IPv4.
-  StringBuilder ss;
-  ss << GetIceCandidateTypeAsString(config.local_candidate_type);
-
-  if (config.local_candidate_type == IceCandidateType::kRelay) {
-    ss << "(" << GetProtocolAsString(config.local_relay_protocol) << ")";
-  }
-
-  ss << ":" << GetNetworkTypeAsString(config.local_network_type) << ":"
-     << GetAddressFamilyAsString(config.local_address_family) << "->"
-     << GetIceCandidateTypeAsString(config.remote_candidate_type) << ":"
-     << GetAddressFamilyAsString(config.remote_address_family) << "@"
-     << GetProtocolAsString(config.candidate_pair_protocol);
-  return ss.Release();
-}
-
-std::map<uint32_t, std::string> BuildCandidateIdLogDescriptionMap(
-    const std::vector<LoggedIceCandidatePairConfig>&
-        ice_candidate_pair_configs) {
-  std::map<uint32_t, std::string> candidate_pair_desc_by_id;
-  for (const auto& config : ice_candidate_pair_configs) {
-    // TODO(qingsi): Add the handling of the "Updated" config event after the
-    // visualization of property change for candidate pairs is introduced.
-    if (candidate_pair_desc_by_id.find(config.candidate_pair_id) ==
-        candidate_pair_desc_by_id.end()) {
-      const std::string candidate_pair_desc =
-          GetCandidatePairLogDescriptionAsString(config);
-      candidate_pair_desc_by_id[config.candidate_pair_id] = candidate_pair_desc;
-    }
-  }
-  return candidate_pair_desc_by_id;
-}
-
-}  // namespace
-
-void CreateIceCandidatePairConfigGraph(const ParsedRtcEventLog& parsed_log,
-                                       const AnalyzerConfig& config,
-                                       Plot* plot) {
-  std::map<uint32_t, TimeSeries> configs_by_cp_id;
-  for (const auto& config_item : parsed_log.ice_candidate_pair_configs()) {
-    if (configs_by_cp_id.find(config_item.candidate_pair_id) ==
-        configs_by_cp_id.end()) {
-      const std::string candidate_pair_desc =
-          GetCandidatePairLogDescriptionAsString(config_item);
-      configs_by_cp_id[config_item.candidate_pair_id] =
-          TimeSeries("[" + std::to_string(config_item.candidate_pair_id) + "]" +
-                         candidate_pair_desc,
-                     LineStyle::kNone, PointStyle::kHighlight);
-    }
-    float x = config.GetCallTimeSec(config_item.log_time());
-    float y = static_cast<float>(config_item.type);
-    configs_by_cp_id[config_item.candidate_pair_id].points.emplace_back(x, y);
-  }
-
-  // TODO(qingsi): There can be a large number of candidate pairs generated by
-  // certain calls and the frontend cannot render the chart in this case due
-  // to the failure of generating a palette with the same number of colors.
-  for (auto& kv : configs_by_cp_id) {
-    plot->AppendTimeSeries(std::move(kv.second));
-  }
-
-  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
-                 kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 3, "Config Type", kBottomMargin, kTopMargin);
-  plot->SetTitle("[IceEventLog] ICE candidate pair configs");
-  plot->SetYAxisTickLabels(
-      {{static_cast<float>(IceCandidatePairConfigType::kAdded), "ADDED"},
-       {static_cast<float>(IceCandidatePairConfigType::kUpdated), "UPDATED"},
-       {static_cast<float>(IceCandidatePairConfigType::kDestroyed),
-        "DESTROYED"},
-       {static_cast<float>(IceCandidatePairConfigType::kSelected),
-        "SELECTED"}});
-}
-
-void CreateIceConnectivityCheckGraph(const ParsedRtcEventLog& parsed_log,
-                                     const AnalyzerConfig& config,
-                                     Plot* plot) {
-  constexpr int kEventTypeOffset =
-      static_cast<int>(IceCandidatePairConfigType::kNumValues);
-  std::map<uint32_t, TimeSeries> checks_by_cp_id;
-  std::map<uint32_t, std::string> candidate_pair_desc_by_id =
-      BuildCandidateIdLogDescriptionMap(
-          parsed_log.ice_candidate_pair_configs());
-  for (const auto& event : parsed_log.ice_candidate_pair_events()) {
-    if (checks_by_cp_id.find(event.candidate_pair_id) ==
-        checks_by_cp_id.end()) {
-      checks_by_cp_id[event.candidate_pair_id] =
-          TimeSeries("[" + std::to_string(event.candidate_pair_id) + "]" +
-                         candidate_pair_desc_by_id[event.candidate_pair_id],
-                     LineStyle::kNone, PointStyle::kHighlight);
-    }
-    float x = config.GetCallTimeSec(event.log_time());
-    float y = static_cast<float>(event.type) + kEventTypeOffset;
-    checks_by_cp_id[event.candidate_pair_id].points.emplace_back(x, y);
-  }
-
-  // TODO(qingsi): The same issue as in CreateIceCandidatePairConfigGraph.
-  for (auto& kv : checks_by_cp_id) {
-    plot->AppendTimeSeries(std::move(kv.second));
-  }
-
-  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
-                 kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 4, "Connectivity State", kBottomMargin,
-                          kTopMargin);
-  plot->SetTitle("[IceEventLog] ICE connectivity checks");
-
-  plot->SetYAxisTickLabels(
-      {{static_cast<float>(IceCandidatePairEventType::kCheckSent) +
-            kEventTypeOffset,
-        "CHECK SENT"},
-       {static_cast<float>(IceCandidatePairEventType::kCheckReceived) +
-            kEventTypeOffset,
-        "CHECK RECEIVED"},
-       {static_cast<float>(IceCandidatePairEventType::kCheckResponseSent) +
-            kEventTypeOffset,
-        "RESPONSE SENT"},
-       {static_cast<float>(IceCandidatePairEventType::kCheckResponseReceived) +
-            kEventTypeOffset,
-        "RESPONSE RECEIVED"}});
-}
-
-void CreateDtlsTransportStateGraph(const ParsedRtcEventLog& parsed_log,
-                                   const AnalyzerConfig& config,
-                                   Plot* plot) {
-  TimeSeries states("DTLS Transport State", LineStyle::kNone,
-                    PointStyle::kHighlight);
-  for (const auto& event : parsed_log.dtls_transport_states()) {
-    float x = config.GetCallTimeSec(event.log_time());
-    float y = static_cast<float>(event.dtls_transport_state);
-    states.points.emplace_back(x, y);
-  }
-  plot->AppendTimeSeries(std::move(states));
-  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
-                 kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, static_cast<float>(DtlsTransportState::kNumValues),
-                          "Transport State", kBottomMargin, kTopMargin);
-  plot->SetTitle("DTLS Transport State");
-  plot->SetYAxisTickLabels(
-      {{static_cast<float>(DtlsTransportState::kNew), "NEW"},
-       {static_cast<float>(DtlsTransportState::kConnecting), "CONNECTING"},
-       {static_cast<float>(DtlsTransportState::kConnected), "CONNECTED"},
-       {static_cast<float>(DtlsTransportState::kClosed), "CLOSED"},
-       {static_cast<float>(DtlsTransportState::kFailed), "FAILED"}});
-}
-
-void CreateDtlsWritableStateGraph(const ParsedRtcEventLog& parsed_log,
-                                  const AnalyzerConfig& config,
-                                  Plot* plot) {
-  TimeSeries writable("DTLS Writable", LineStyle::kNone,
-                      PointStyle::kHighlight);
-  for (const auto& event : parsed_log.dtls_writable_states()) {
-    float x = config.GetCallTimeSec(event.log_time());
-    float y = static_cast<float>(event.writable);
-    writable.points.emplace_back(x, y);
-  }
-  plot->AppendTimeSeries(std::move(writable));
-  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
-                 kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Writable", kBottomMargin, kTopMargin);
-  plot->SetTitle("DTLS Writable State");
-}
-
-}  // namespace webrtc
diff --git a/rtc_tools/rtc_event_log_visualizer/analyze_connectivity.h b/rtc_tools/rtc_event_log_visualizer/analyze_connectivity.h
deleted file mode 100644
index 1eac2f0c52..0000000000
--- a/rtc_tools/rtc_event_log_visualizer/analyze_connectivity.h
+++ /dev/null
@@ -1,38 +0,0 @@
-/*
- *  Copyright (c) 2026 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZE_CONNECTIVITY_H_
-#define RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZE_CONNECTIVITY_H_
-
-#include "logging/rtc_event_log/rtc_event_log_parser.h"
-#include "rtc_tools/rtc_event_log_visualizer/analyzer_common.h"
-#include "rtc_tools/rtc_event_log_visualizer/plot_base.h"
-
-namespace webrtc {
-
-void CreateIceCandidatePairConfigGraph(const ParsedRtcEventLog& parsed_log,
-                                       const AnalyzerConfig& config,
-                                       Plot* plot);
-
-void CreateIceConnectivityCheckGraph(const ParsedRtcEventLog& parsed_log,
-                                     const AnalyzerConfig& config,
-                                     Plot* plot);
-
-void CreateDtlsTransportStateGraph(const ParsedRtcEventLog& parsed_log,
-                                   const AnalyzerConfig& config,
-                                   Plot* plot);
-
-void CreateDtlsWritableStateGraph(const ParsedRtcEventLog& parsed_log,
-                                  const AnalyzerConfig& config,
-                                  Plot* plot);
-
-}  // namespace webrtc
-
-#endif  // RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZE_CONNECTIVITY_H_
diff --git a/rtc_tools/rtc_event_log_visualizer/analyze_rtp_rtcp.cc b/rtc_tools/rtc_event_log_visualizer/analyze_rtp_rtcp.cc
deleted file mode 100644
index 9d6c0668e8..0000000000
--- a/rtc_tools/rtc_event_log_visualizer/analyze_rtp_rtcp.cc
+++ /dev/null
@@ -1,809 +0,0 @@
-/*
- *  Copyright (c) 2026 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "rtc_tools/rtc_event_log_visualizer/analyze_rtp_rtcp.h"
-
-#include <algorithm>
-#include <cstddef>
-#include <cstdint>
-#include <deque>
-#include <map>
-#include <optional>
-#include <string>
-#include <tuple>
-#include <unordered_set>
-#include <utility>
-#include <vector>
-
-#include "api/function_view.h"
-#include "api/rtp_headers.h"
-#include "api/transport/ecn_marking.h"
-#include "api/units/data_rate.h"
-#include "api/units/time_delta.h"
-#include "api/units/timestamp.h"
-#include "logging/rtc_event_log/events/logged_rtp_rtcp.h"
-#include "logging/rtc_event_log/rtc_event_log_parser.h"
-#include "logging/rtc_event_log/rtc_event_processor.h"
-#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
-#include "modules/rtp_rtcp/source/rtcp_packet/congestion_control_feedback.h"
-#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
-#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
-#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
-#include "modules/rtp_rtcp/source/rtcp_packet/target_bitrate.h"
-#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
-#include "rtc_base/checks.h"
-#include "rtc_base/logging.h"
-#include "rtc_base/numerics/sequence_number_unwrapper.h"
-#include "rtc_tools/rtc_event_log_visualizer/analyzer_common.h"
-#include "rtc_tools/rtc_event_log_visualizer/plot_base.h"
-
-namespace webrtc {
-
-namespace {
-
-template <typename T>
-TimeSeries CreateRtcpTypeTimeSeries(const std::vector<T>& rtcp_list,
-                                    AnalyzerConfig config,
-                                    std::string rtcp_name,
-                                    int category_id) {
-  TimeSeries time_series(rtcp_name, LineStyle::kNone, PointStyle::kHighlight);
-  for (const auto& rtcp : rtcp_list) {
-    float x = config.GetCallTimeSec(rtcp.timestamp);
-    float y = category_id;
-    time_series.points.emplace_back(x, y);
-  }
-  return time_series;
-}
-
-struct PacketLossSummary {
-  size_t num_packets = 0;
-  size_t num_lost_packets = 0;
-  Timestamp base_time = Timestamp::MinusInfinity();
-};
-
-}  // namespace
-
-float GetHighestSeqNumber(const rtcp::ReportBlock& block) {
-  return block.extended_high_seq_num();
-}
-
-float GetFractionLost(const rtcp::ReportBlock& block) {
-  return static_cast<double>(block.fraction_lost()) / 256 * 100;
-}
-
-float GetCumulativeLost(const rtcp::ReportBlock& block) {
-  return block.cumulative_lost();
-}
-
-float DelaySinceLastSr(const rtcp::ReportBlock& block) {
-  return static_cast<double>(block.delay_since_last_sr()) / 65536;
-}
-
-void CreatePacketGraph(PacketDirection direction,
-                       const ParsedRtcEventLog& parsed_log,
-                       const AnalyzerConfig& config,
-                       Plot* plot) {
-  for (const auto& stream : parsed_log.rtp_packets_by_ssrc(direction)) {
-    // Filter on SSRC.
-    if (!MatchingSsrc(stream.ssrc, config.desired_ssrc_)) {
-      continue;
-    }
-
-    TimeSeries time_series(GetStreamName(parsed_log, direction, stream.ssrc),
-                           LineStyle::kBar);
-    auto GetPacketSize = [](const LoggedRtpPacket& packet) {
-      return std::optional<float>(packet.total_length);
-    };
-    auto ToCallTime = [&config](const LoggedRtpPacket& packet) {
-      return config.GetCallTimeSec(packet.timestamp);
-    };
-    ProcessPoints<LoggedRtpPacket>(ToCallTime, GetPacketSize,
-                                   stream.packet_view, &time_series);
-    plot->AppendTimeSeries(std::move(time_series));
-  }
-
-  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
-                 kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Packet size (bytes)", kBottomMargin,
-                          kTopMargin);
-  plot->SetTitle(GetDirectionAsString(direction) + " RTP packets");
-}
-
-void CreateRtcpTypeGraph(PacketDirection direction,
-                         const ParsedRtcEventLog& parsed_log,
-                         const AnalyzerConfig& config,
-                         Plot* plot) {
-  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(
-      parsed_log.transport_feedbacks(direction), config, "TWCC", 1));
-  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(
-      parsed_log.congestion_feedback(direction), config, "CCFB", 2));
-  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(
-      parsed_log.receiver_reports(direction), config, "RR", 3));
-  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(
-      parsed_log.sender_reports(direction), config, "SR", 4));
-  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(
-      parsed_log.extended_reports(direction), config, "XR", 5));
-  plot->AppendTimeSeries(
-      CreateRtcpTypeTimeSeries(parsed_log.nacks(direction), config, "NACK", 6));
-  plot->AppendTimeSeries(
-      CreateRtcpTypeTimeSeries(parsed_log.rembs(direction), config, "REMB", 7));
-  plot->AppendTimeSeries(
-      CreateRtcpTypeTimeSeries(parsed_log.firs(direction), config, "FIR", 8));
-  plot->AppendTimeSeries(
-      CreateRtcpTypeTimeSeries(parsed_log.plis(direction), config, "PLI", 9));
-  plot->AppendTimeSeries(
-      CreateRtcpTypeTimeSeries(parsed_log.byes(direction), config, "BYE", 10));
-  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
-                 kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "RTCP type", kBottomMargin, kTopMargin);
-  plot->SetTitle(GetDirectionAsString(direction) + " RTCP packets");
-  plot->SetYAxisTickLabels({{1, "TWCC"},
-                            {2, "CCFB"},
-                            {3, "RR"},
-                            {4, "SR"},
-                            {5, "XR"},
-                            {6, "NACK"},
-                            {7, "REMB"},
-                            {8, "FIR"},
-                            {9, "PLI"},
-                            {10, "BYE"}});
-}
-
-template <typename IterableType>
-void CreateAccumulatedPacketsTimeSeries(Plot* plot,
-                                        const AnalyzerConfig& config,
-                                        const IterableType& packets,
-                                        const std::string& label) {
-  TimeSeries time_series(label, LineStyle::kStep);
-  for (size_t i = 0; i < packets.size(); i++) {
-    float x = config.GetCallTimeSec(packets[i].log_time());
-    time_series.points.emplace_back(x, i + 1);
-  }
-  plot->AppendTimeSeries(std::move(time_series));
-}
-
-void CreateAccumulatedPacketsGraph(PacketDirection direction,
-                                   const ParsedRtcEventLog& parsed_log,
-                                   const AnalyzerConfig& config,
-                                   Plot* plot) {
-  for (const auto& stream : parsed_log.rtp_packets_by_ssrc(direction)) {
-    if (!MatchingSsrc(stream.ssrc, config.desired_ssrc_))
-      continue;
-    std::string label =
-        std::string("RTP ") + GetStreamName(parsed_log, direction, stream.ssrc);
-    CreateAccumulatedPacketsTimeSeries(plot, config, stream.packet_view, label);
-  }
-  std::string label =
-      std::string("RTCP ") + "(" + GetDirectionAsShortString(direction) + ")";
-  if (direction == kIncomingPacket) {
-    CreateAccumulatedPacketsTimeSeries(
-        plot, config, parsed_log.incoming_rtcp_packets(), label);
-  } else {
-    CreateAccumulatedPacketsTimeSeries(
-        plot, config, parsed_log.outgoing_rtcp_packets(), label);
-  }
-
-  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
-                 kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Received Packets", kBottomMargin, kTopMargin);
-  plot->SetTitle(std::string("Accumulated ") + GetDirectionAsString(direction) +
-                 " RTP/RTCP packets");
-}
-
-void CreatePacketRateGraph(PacketDirection direction,
-                           const ParsedRtcEventLog& parsed_log,
-                           const AnalyzerConfig& config,
-                           Plot* plot) {
-  auto CountPackets = [](auto packet) { return 1.0; };
-  for (const auto& stream : parsed_log.rtp_packets_by_ssrc(direction)) {
-    // Filter on SSRC.
-    if (!MatchingSsrc(stream.ssrc, config.desired_ssrc_)) {
-      continue;
-    }
-    TimeSeries time_series(
-        std::string("RTP ") + GetStreamName(parsed_log, direction, stream.ssrc),
-        LineStyle::kLine);
-    MovingAverage<LoggedRtpPacket, double>(CountPackets, stream.packet_view,
-                                           config, &time_series);
-    plot->AppendTimeSeries(std::move(time_series));
-  }
-  TimeSeries time_series(
-      std::string("RTCP ") + "(" + GetDirectionAsShortString(direction) + ")",
-      LineStyle::kLine);
-  if (direction == kIncomingPacket) {
-    MovingAverage<LoggedRtcpPacketIncoming, double>(
-        CountPackets, parsed_log.incoming_rtcp_packets(), config, &time_series);
-  } else {
-    MovingAverage<LoggedRtcpPacketOutgoing, double>(
-        CountPackets, parsed_log.outgoing_rtcp_packets(), config, &time_series);
-  }
-  plot->AppendTimeSeries(std::move(time_series));
-
-  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
-                 kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Packet Rate (packets/s)", kBottomMargin,
-                          kTopMargin);
-  plot->SetTitle("Rate of " + GetDirectionAsString(direction) +
-                 " RTP/RTCP packets");
-}
-
-void CreateTotalPacketRateGraph(PacketDirection direction,
-                                const ParsedRtcEventLog& parsed_log,
-                                const AnalyzerConfig& config,
-                                Plot* plot) {
-  // Contains a log timestamp to enable counting logged events of different
-  // types using MovingAverage().
-  class LogTime {
-   public:
-    explicit LogTime(Timestamp log_time) : log_time_(log_time) {}
-    Timestamp log_time() const { return log_time_; }
-
-   private:
-    Timestamp log_time_;
-  };
-  std::vector<LogTime> packet_times;
-  auto handle_rtp = [&packet_times](const LoggedRtpPacket& packet) {
-    packet_times.emplace_back(packet.log_time());
-  };
-  RtcEventProcessor process;
-  for (const auto& stream : parsed_log.rtp_packets_by_ssrc(direction)) {
-    process.AddEvents(stream.packet_view, handle_rtp, direction);
-  }
-  if (direction == kIncomingPacket) {
-    auto handle_incoming_rtcp =
-        [&packet_times](const LoggedRtcpPacketIncoming& packet) {
-          packet_times.emplace_back(packet.log_time());
-        };
-    process.AddEvents(parsed_log.incoming_rtcp_packets(), handle_incoming_rtcp);
-  } else {
-    auto handle_outgoing_rtcp =
-        [&packet_times](const LoggedRtcpPacketOutgoing& packet) {
-          packet_times.emplace_back(packet.log_time());
-        };
-    process.AddEvents(parsed_log.outgoing_rtcp_packets(), handle_outgoing_rtcp);
-  }
-  process.ProcessEventsInOrder();
-  TimeSeries time_series(std::string("Total ") + "(" +
-                             GetDirectionAsShortString(direction) + ") packets",
-                         LineStyle::kLine);
-  MovingAverage<LogTime, uint64_t>([](auto packet) { return 1; }, packet_times,
-                                   config, &time_series);
-  plot->AppendTimeSeries(std::move(time_series));
-
-  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
-                 kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Packet Rate (packets/s)", kBottomMargin,
-                          kTopMargin);
-  plot->SetTitle("Rate of all " + GetDirectionAsString(direction) +
-                 " RTP/RTCP packets");
-}
-
-// For each SSRC, plot the sequence number difference between consecutive
-// incoming packets.
-void CreateSequenceNumberGraph(const ParsedRtcEventLog& parsed_log,
-                               const AnalyzerConfig& config,
-                               Plot* plot) {
-  for (const auto& stream : parsed_log.incoming_rtp_packets_by_ssrc()) {
-    // Filter on SSRC.
-    if (!MatchingSsrc(stream.ssrc, config.desired_ssrc_)) {
-      continue;
-    }
-
-    TimeSeries time_series(
-        GetStreamName(parsed_log, kIncomingPacket, stream.ssrc),
-        LineStyle::kBar);
-    auto GetSequenceNumberDiff = [](const LoggedRtpPacketIncoming& old_packet,
-                                    const LoggedRtpPacketIncoming& new_packet) {
-      int64_t diff =
-          WrappingDifference(new_packet.rtp.header.sequenceNumber,
-                             old_packet.rtp.header.sequenceNumber, 1ul << 16);
-      return diff;
-    };
-    auto ToCallTime = [&config](const LoggedRtpPacketIncoming& packet) {
-      return config.GetCallTimeSec(packet.log_time());
-    };
-    ProcessPairs<LoggedRtpPacketIncoming, float>(
-        ToCallTime, GetSequenceNumberDiff, stream.incoming_packets,
-        &time_series);
-    plot->AppendTimeSeries(std::move(time_series));
-  }
-
-  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
-                 kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Difference since last packet", kBottomMargin,
-                          kTopMargin);
-  plot->SetTitle("Incoming sequence number delta");
-}
-
-void CreateIncomingPacketLossGraph(const ParsedRtcEventLog& parsed_log,
-                                   const AnalyzerConfig& config,
-                                   Plot* plot) {
-  for (const auto& stream : parsed_log.incoming_rtp_packets_by_ssrc()) {
-    const std::vector<LoggedRtpPacketIncoming>& packets =
-        stream.incoming_packets;
-    // Filter on SSRC.
-    if (!MatchingSsrc(stream.ssrc, config.desired_ssrc_) || packets.empty()) {
-      continue;
-    }
-
-    TimeSeries time_series(
-        GetStreamName(parsed_log, kIncomingPacket, stream.ssrc),
-        LineStyle::kLine, PointStyle::kHighlight);
-    // TODO(terelius): Should the window and step size be read from the class
-    // instead?
-    const TimeDelta kWindow = TimeDelta::Millis(1000);
-    const TimeDelta kStep = TimeDelta::Millis(1000);
-    SeqNumUnwrapper<uint16_t> unwrapper_;
-    SeqNumUnwrapper<uint16_t> prior_unwrapper_;
-    size_t window_index_begin = 0;
-    size_t window_index_end = 0;
-    uint64_t highest_seq_number =
-        unwrapper_.Unwrap(packets[0].rtp.header.sequenceNumber) - 1;
-    uint64_t highest_prior_seq_number =
-        prior_unwrapper_.Unwrap(packets[0].rtp.header.sequenceNumber) - 1;
-
-    for (Timestamp t = config.begin_time_; t < config.end_time_ + kStep;
-         t += kStep) {
-      while (window_index_end < packets.size() &&
-             packets[window_index_end].rtp.log_time() < t) {
-        uint64_t sequence_number = unwrapper_.Unwrap(
-            packets[window_index_end].rtp.header.sequenceNumber);
-        highest_seq_number = std::max(highest_seq_number, sequence_number);
-        ++window_index_end;
-      }
-      while (window_index_begin < packets.size() &&
-             packets[window_index_begin].rtp.log_time() < t - kWindow) {
-        uint64_t sequence_number = prior_unwrapper_.Unwrap(
-            packets[window_index_begin].rtp.header.sequenceNumber);
-        highest_prior_seq_number =
-            std::max(highest_prior_seq_number, sequence_number);
-        ++window_index_begin;
-      }
-      float x = config.GetCallTimeSec(t);
-      uint64_t expected_packets = highest_seq_number - highest_prior_seq_number;
-      if (expected_packets > 0) {
-        int64_t received_packets = window_index_end - window_index_begin;
-        int64_t lost_packets = expected_packets - received_packets;
-        float y = static_cast<float>(lost_packets) / expected_packets * 100;
-        time_series.points.emplace_back(x, y);
-      }
-    }
-    plot->AppendTimeSeries(std::move(time_series));
-  }
-
-  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
-                 kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Loss rate (in %)", kBottomMargin, kTopMargin);
-  plot->SetTitle("Incoming packet loss (derived from incoming packets)");
-}
-
-// For each SSRC, plot the bandwidth used by that stream.
-void CreateStreamBitrateGraph(PacketDirection direction,
-                              const ParsedRtcEventLog& parsed_log,
-                              const AnalyzerConfig& config,
-                              Plot* plot) {
-  for (const auto& stream : parsed_log.rtp_packets_by_ssrc(direction)) {
-    // Filter on SSRC.
-    if (!MatchingSsrc(stream.ssrc, config.desired_ssrc_)) {
-      continue;
-    }
-
-    TimeSeries time_series(GetStreamName(parsed_log, direction, stream.ssrc),
-                           LineStyle::kLine);
-    auto GetPacketSizeKilobits = [](const LoggedRtpPacket& packet) {
-      return packet.total_length * 8.0 / 1000.0;
-    };
-    MovingAverage<LoggedRtpPacket, double>(
-        GetPacketSizeKilobits, stream.packet_view, config, &time_series);
-    plot->AppendTimeSeries(std::move(time_series));
-  }
-
-  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
-                 kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin);
-  plot->SetTitle(GetDirectionAsString(direction) + " bitrate per stream");
-}
-
-// Plot the bitrate allocation for each temporal and spatial layer.
-// Computed from RTCP XR target bitrate block, so the graph is only populated if
-// those are sent.
-void CreateBitrateAllocationGraph(PacketDirection direction,
-                                  const ParsedRtcEventLog& parsed_log,
-                                  const AnalyzerConfig& config,
-                                  Plot* plot) {
-  std::map<LayerDescription, TimeSeries> time_series;
-  const auto& xr_list = parsed_log.extended_reports(direction);
-  for (const auto& rtcp : xr_list) {
-    const std::optional<rtcp::TargetBitrate>& target_bitrate =
-        rtcp.xr.target_bitrate();
-    if (!target_bitrate.has_value())
-      continue;
-    for (const auto& bitrate_item : target_bitrate->GetTargetBitrates()) {
-      LayerDescription layer(rtcp.xr.sender_ssrc(), bitrate_item.spatial_layer,
-                             bitrate_item.temporal_layer);
-      auto time_series_it = time_series.find(layer);
-      if (time_series_it == time_series.end()) {
-        std::string layer_name = GetLayerName(layer);
-        bool inserted;
-        std::tie(time_series_it, inserted) = time_series.insert(
-            std::make_pair(layer, TimeSeries(layer_name, LineStyle::kStep)));
-        RTC_DCHECK(inserted);
-      }
-      float x = config.GetCallTimeSec(rtcp.log_time());
-      float y = bitrate_item.target_bitrate_kbps;
-      time_series_it->second.points.emplace_back(x, y);
-    }
-  }
-  for (auto& layer : time_series) {
-    plot->AppendTimeSeries(std::move(layer.second));
-  }
-  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
-                 kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin);
-  if (direction == kIncomingPacket)
-    plot->SetTitle("Target bitrate per incoming layer");
-  else
-    plot->SetTitle("Target bitrate per outgoing layer");
-}
-
-void CreateEcnFeedbackGraph(Plot* plot,
-                            PacketDirection direction,
-                            const ParsedRtcEventLog& parsed_log,
-                            const AnalyzerConfig& config) {
-  TimeSeries not_ect("Not ECN capable", LineStyle::kBar,
-                     PointStyle::kHighlight);
-  TimeSeries ect_1("ECN capable", LineStyle::kBar, PointStyle::kHighlight);
-  TimeSeries ce("Congestion experienced", LineStyle::kBar,
-                PointStyle::kHighlight);
-
-  for (const LoggedRtcpCongestionControlFeedback& feedback :
-       parsed_log.congestion_feedback(direction)) {
-    int ect_1_count = 0;
-    int not_ect_count = 0;
-    int ce_count = 0;
-
-    for (const rtcp::CongestionControlFeedback::PacketInfo& info :
-         feedback.congestion_feedback.packets()) {
-      switch (info.ecn) {
-        case EcnMarking::kNotEct:
-          ++not_ect_count;
-          break;
-        case EcnMarking::kEct1:
-          ++ect_1_count;
-          break;
-        case EcnMarking::kEct0:
-          RTC_LOG(LS_ERROR) << "unexpected ect(0)";
-          break;
-        case EcnMarking::kCe:
-          ++ce_count;
-          break;
-      }
-    }
-    ect_1.points.emplace_back(config.GetCallTimeSec(feedback.timestamp),
-                              ect_1_count);
-    not_ect.points.emplace_back(config.GetCallTimeSec(feedback.timestamp),
-                                not_ect_count);
-    ce.points.emplace_back(config.GetCallTimeSec(feedback.timestamp), ce_count);
-  }
-
-  plot->AppendTimeSeriesIfNotEmpty(std::move(ect_1));
-  plot->AppendTimeSeriesIfNotEmpty(std::move(not_ect));
-  plot->AppendTimeSeriesIfNotEmpty(std::move(ce));
-
-  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
-                 kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 10, "Count per feedback", kBottomMargin,
-                          kTopMargin);
-}
-
-void CreateOutgoingEcnFeedbackGraph(const ParsedRtcEventLog& parsed_log,
-                                    const AnalyzerConfig& config,
-                                    Plot* plot) {
-  CreateEcnFeedbackGraph(plot, kOutgoingPacket, parsed_log, config);
-  plot->SetTitle("Outgoing ECN count per feedback");
-}
-
-void CreateIncomingEcnFeedbackGraph(const ParsedRtcEventLog& parsed_log,
-                                    const AnalyzerConfig& config,
-                                    Plot* plot) {
-  CreateEcnFeedbackGraph(plot, kIncomingPacket, parsed_log, config);
-  plot->SetTitle("Incoming ECN count per feedback");
-}
-
-void CreateOutgoingLossRateGraph(const ParsedRtcEventLog& parsed_log,
-                                 const AnalyzerConfig& config,
-                                 Plot* plot) {
-  struct PacketLossPerFeedback {
-    Timestamp timestamp;              // Time when this feedback was received.
-    int num_packets_in_feedback = 0;  // Includes lost packets.
-    int num_lost_packets = 0;         // In this specific feedback.
-    int num_reordered_packets = 0;    // Packets received in this feedback, but
-                                      // was previously reported as lost.
-    int num_missing_feedback =
-        0;  // Packets missing feedback between this report and the previous.
-  };
-
-  class LossFeedbackBuilder {
-   public:
-    void AddPacket(uint16_t sequence_number, TimeDelta arrival_time_delta) {
-      last_unwrapped_sequence_number_ =
-          sequence_number_unwrapper_.Unwrap(sequence_number);
-      if (!first_sequence_number_.has_value()) {
-        first_sequence_number_ = last_unwrapped_sequence_number_;
-      }
-      ++num_packets_;
-      if (arrival_time_delta.IsInfinite()) {
-        lost_sequence_numbers_.insert(last_unwrapped_sequence_number_);
-      } else {
-        num_reordered_packets_ += previous_lost_sequence_numbers_.count(
-            last_unwrapped_sequence_number_);
-      }
-    }
-
-    void Update(PacketLossPerFeedback& feedback) {
-      feedback.num_packets_in_feedback += num_packets_;
-      feedback.num_lost_packets += lost_sequence_numbers_.size();
-      feedback.num_reordered_packets += num_reordered_packets_;
-      if (first_sequence_number_.has_value() &&
-          previous_feedback_highest_seq_number_.has_value()) {
-        feedback.num_missing_feedback +=
-            *first_sequence_number_ - *previous_feedback_highest_seq_number_ -
-            1;
-      }
-
-      // Prepare for next feedback.
-      first_sequence_number_ = std::nullopt;
-      previous_lost_sequence_numbers_.insert(lost_sequence_numbers_.begin(),
-                                             lost_sequence_numbers_.end());
-      previous_feedback_highest_seq_number_ = last_unwrapped_sequence_number_;
-      lost_sequence_numbers_.clear();
-      num_reordered_packets_ = 0;
-      num_packets_ = 0;
-    }
-
-   private:
-    int64_t last_unwrapped_sequence_number_ = 0;
-    int num_reordered_packets_ = 0;
-    int num_packets_ = 0;
-    std::optional<int64_t> first_sequence_number_;
-
-    std::unordered_set<int64_t> lost_sequence_numbers_;
-    std::unordered_set<int64_t> previous_lost_sequence_numbers_;
-    std::optional<int64_t> previous_feedback_highest_seq_number_;
-
-    RtpSequenceNumberUnwrapper sequence_number_unwrapper_;
-  };
-
-  TimeSeries loss_rate_series("Loss rate (from packet feedback)",
-                              LineStyle::kLine, PointStyle::kHighlight);
-  TimeSeries reordered_packets_between_feedback(
-      "Ratio of reordered packets from last feedback", LineStyle::kLine,
-      PointStyle::kHighlight);
-  TimeSeries average_loss_rate_series("Average loss rate last 5s",
-                                      LineStyle::kLine, PointStyle::kHighlight);
-  TimeSeries missing_feedback_series("Missing feedback", LineStyle::kNone,
-                                     PointStyle::kHighlight);
-
-  std::vector<PacketLossPerFeedback> loss_per_feedback;
-
-  if (!parsed_log.congestion_feedback(kIncomingPacket).empty()) {
-    plot->SetTitle("Outgoing loss rate (from CCFB)");
-
-    std::map</*ssrc*/ uint32_t, LossFeedbackBuilder> per_ssrc_builder;
-    for (const LoggedRtcpCongestionControlFeedback& feedback :
-         parsed_log.congestion_feedback(kIncomingPacket)) {
-      const rtcp::CongestionControlFeedback& transport_feedback =
-          feedback.congestion_feedback;
-
-      PacketLossPerFeedback packet_loss_per_feedback = {
-          .timestamp = feedback.log_time()};
-      for (const rtcp::CongestionControlFeedback::PacketInfo& packet :
-           transport_feedback.packets()) {
-        per_ssrc_builder[packet.ssrc].AddPacket(packet.sequence_number,
-                                                packet.arrival_time_offset);
-      }
-      for (auto& [ssrc, builder] : per_ssrc_builder) {
-        builder.Update(packet_loss_per_feedback);
-      }
-      loss_per_feedback.push_back(packet_loss_per_feedback);
-    }
-  } else if (!parsed_log.transport_feedbacks(kIncomingPacket).empty()) {
-    plot->SetTitle("Outgoing loss rate (from TWCC)");
-
-    LossFeedbackBuilder builder;
-    for (const LoggedRtcpPacketTransportFeedback& feedback :
-         parsed_log.transport_feedbacks(kIncomingPacket)) {
-      feedback.transport_feedback.ForAllPackets(
-          [&](uint16_t sequence_number, TimeDelta receive_time_delta) {
-            builder.AddPacket(sequence_number, receive_time_delta);
-          });
-      PacketLossPerFeedback packet_loss_per_feedback = {
-          .timestamp = feedback.log_time()};
-      builder.Update(packet_loss_per_feedback);
-      loss_per_feedback.push_back(packet_loss_per_feedback);
-    }
-  }
-
-  PacketLossSummary window_summary;
-  Timestamp last_observation_receive_time = Timestamp::Zero();
-
-  // Use loss based bwe 2 observation duration and observation window size.
-  constexpr TimeDelta kObservationDuration = TimeDelta::Millis(250);
-  constexpr uint32_t kObservationWindowSize = 20;
-  std::deque<PacketLossSummary> observations;
-  int previous_feedback_size = 0;
-  for (const PacketLossPerFeedback& feedback : loss_per_feedback) {
-    for (int64_t num = 0; num < feedback.num_missing_feedback; ++num) {
-      missing_feedback_series.points.emplace_back(
-          config.GetCallTimeSec(feedback.timestamp), 100 + num);
-    }
-
-    // Compute loss rate from the transport feedback.
-    float loss_rate = static_cast<float>(feedback.num_lost_packets * 100.0 /
-                                         feedback.num_packets_in_feedback);
-
-    loss_rate_series.points.emplace_back(
-        config.GetCallTimeSec(feedback.timestamp), loss_rate);
-    float reordered_rate =
-        previous_feedback_size == 0
-            ? 0
-            : static_cast<float>(feedback.num_reordered_packets * 100.0 /
-                                 previous_feedback_size);
-    previous_feedback_size = feedback.num_packets_in_feedback;
-    reordered_packets_between_feedback.points.emplace_back(
-        config.GetCallTimeSec(feedback.timestamp), reordered_rate);
-
-    // Compute loss rate in a window of kObservationWindowSize.
-    if (window_summary.num_packets == 0) {
-      window_summary.base_time = feedback.timestamp;
-    }
-    window_summary.num_packets += feedback.num_packets_in_feedback;
-    window_summary.num_lost_packets +=
-        feedback.num_lost_packets - feedback.num_reordered_packets;
-
-    const Timestamp last_received_time = feedback.timestamp;
-    const TimeDelta observation_duration =
-        window_summary.base_time == Timestamp::Zero()
-            ? TimeDelta::Zero()
-            : last_received_time - window_summary.base_time;
-    if (observation_duration > kObservationDuration) {
-      last_observation_receive_time = last_received_time;
-      observations.push_back(window_summary);
-      if (observations.size() > kObservationWindowSize) {
-        observations.pop_front();
-      }
-
-      // Compute average loss rate in a number of windows.
-      int total_packets = 0;
-      int total_loss = 0;
-      for (const auto& observation : observations) {
-        total_loss += observation.num_lost_packets;
-        total_packets += observation.num_packets;
-      }
-      if (total_packets > 0) {
-        float average_loss_rate = total_loss * 100.0 / total_packets;
-        average_loss_rate_series.points.emplace_back(
-            config.GetCallTimeSec(feedback.timestamp), average_loss_rate);
-      } else {
-        average_loss_rate_series.points.emplace_back(
-            config.GetCallTimeSec(feedback.timestamp), 0);
-      }
-      window_summary = PacketLossSummary();
-    }
-  }
-  // Add the data set to the plot.
-  plot->AppendTimeSeriesIfNotEmpty(std::move(loss_rate_series));
-  plot->AppendTimeSeriesIfNotEmpty(
-      std::move(reordered_packets_between_feedback));
-  plot->AppendTimeSeriesIfNotEmpty(std::move(average_loss_rate_series));
-  plot->AppendTimeSeriesIfNotEmpty(std::move(missing_feedback_series));
-
-  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
-                 kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 100, "Loss rate (percent)", kBottomMargin,
-                          kTopMargin);
-}
-
-void CreateTimestampGraph(PacketDirection direction,
-                          const ParsedRtcEventLog& parsed_log,
-                          const AnalyzerConfig& config,
-                          Plot* plot) {
-  for (const auto& stream : parsed_log.rtp_packets_by_ssrc(direction)) {
-    TimeSeries rtp_timestamps(
-        GetStreamName(parsed_log, direction, stream.ssrc) + " capture-time",
-        LineStyle::kLine, PointStyle::kHighlight);
-    for (const auto& packet : stream.packet_view) {
-      float x = config.GetCallTimeSec(packet.log_time());
-      float y = packet.header.timestamp;
-      rtp_timestamps.points.emplace_back(x, y);
-    }
-    plot->AppendTimeSeries(std::move(rtp_timestamps));
-
-    TimeSeries rtcp_timestamps(
-        GetStreamName(parsed_log, direction, stream.ssrc) +
-            " rtcp capture-time",
-        LineStyle::kLine, PointStyle::kHighlight);
-    // TODO(terelius): Why only sender reports?
-    const auto& sender_reports = parsed_log.sender_reports(direction);
-    for (const auto& rtcp : sender_reports) {
-      if (rtcp.sr.sender_ssrc() != stream.ssrc)
-        continue;
-      float x = config.GetCallTimeSec(rtcp.log_time());
-      float y = rtcp.sr.rtp_timestamp();
-      rtcp_timestamps.points.emplace_back(x, y);
-    }
-    plot->AppendTimeSeriesIfNotEmpty(std::move(rtcp_timestamps));
-  }
-
-  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
-                 kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "RTP timestamp", kBottomMargin, kTopMargin);
-  plot->SetTitle(GetDirectionAsString(direction) + " timestamps");
-}
-
-void CreateSenderAndReceiverReportPlot(
-    PacketDirection direction,
-    FunctionView<float(const rtcp::ReportBlock&)> fy,
-    std::string title,
-    std::string yaxis_label,
-    const ParsedRtcEventLog& parsed_log,
-    const AnalyzerConfig& config,
-    Plot* plot) {
-  std::map<uint32_t, TimeSeries> sr_reports_by_ssrc;
-  const auto& sender_reports = parsed_log.sender_reports(direction);
-  for (const auto& rtcp : sender_reports) {
-    float x = config.GetCallTimeSec(rtcp.log_time());
-    uint32_t ssrc = rtcp.sr.sender_ssrc();
-    for (const auto& block : rtcp.sr.report_blocks()) {
-      float y = fy(block);
-      auto sr_report_it = sr_reports_by_ssrc.find(ssrc);
-      bool inserted;
-      if (sr_report_it == sr_reports_by_ssrc.end()) {
-        std::tie(sr_report_it, inserted) = sr_reports_by_ssrc.emplace(
-            ssrc, TimeSeries(GetStreamName(parsed_log, direction, ssrc) +
-                                 " Sender Reports",
-                             LineStyle::kLine, PointStyle::kHighlight));
-      }
-      sr_report_it->second.points.emplace_back(x, y);
-    }
-  }
-  for (auto& kv : sr_reports_by_ssrc) {
-    plot->AppendTimeSeries(std::move(kv.second));
-  }
-
-  std::map<uint32_t, TimeSeries> rr_reports_by_ssrc;
-  const auto& receiver_reports = parsed_log.receiver_reports(direction);
-  for (const auto& rtcp : receiver_reports) {
-    float x = config.GetCallTimeSec(rtcp.log_time());
-    uint32_t ssrc = rtcp.rr.sender_ssrc();
-    for (const auto& block : rtcp.rr.report_blocks()) {
-      float y = fy(block);
-      auto rr_report_it = rr_reports_by_ssrc.find(ssrc);
-      bool inserted;
-      if (rr_report_it == rr_reports_by_ssrc.end()) {
-        std::tie(rr_report_it, inserted) = rr_reports_by_ssrc.emplace(
-            ssrc, TimeSeries(GetStreamName(parsed_log, direction, ssrc) +
-                                 " Receiver Reports",
-                             LineStyle::kLine, PointStyle::kHighlight));
-      }
-      rr_report_it->second.points.emplace_back(x, y);
-    }
-  }
-  for (auto& kv : rr_reports_by_ssrc) {
-    plot->AppendTimeSeries(std::move(kv.second));
-  }
-
-  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
-                 kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, yaxis_label, kBottomMargin, kTopMargin);
-  plot->SetTitle(title);
-}
-
-}  // namespace webrtc
diff --git a/rtc_tools/rtc_event_log_visualizer/analyze_rtp_rtcp.h b/rtc_tools/rtc_event_log_visualizer/analyze_rtp_rtcp.h
deleted file mode 100644
index 2865023e60..0000000000
--- a/rtc_tools/rtc_event_log_visualizer/analyze_rtp_rtcp.h
+++ /dev/null
@@ -1,100 +0,0 @@
-/*
- *  Copyright (c) 2026 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZE_RTP_RTCP_H_
-#define RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZE_RTP_RTCP_H_
-
-#include <string>
-
-#include "api/function_view.h"
-#include "logging/rtc_event_log/rtc_event_log_parser.h"
-#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
-#include "rtc_tools/rtc_event_log_visualizer/analyzer_common.h"
-#include "rtc_tools/rtc_event_log_visualizer/plot_base.h"
-
-namespace webrtc {
-
-float GetHighestSeqNumber(const rtcp::ReportBlock& block);
-float GetFractionLost(const rtcp::ReportBlock& block);
-float GetCumulativeLost(const rtcp::ReportBlock& block);
-float DelaySinceLastSr(const rtcp::ReportBlock& block);
-
-void CreatePacketGraph(PacketDirection direction,
-                       const ParsedRtcEventLog& parsed_log,
-                       const AnalyzerConfig& config,
-                       Plot* plot);
-
-void CreateRtcpTypeGraph(PacketDirection direction,
-                         const ParsedRtcEventLog& parsed_log,
-                         const AnalyzerConfig& config,
-                         Plot* plot);
-
-void CreateAccumulatedPacketsGraph(PacketDirection direction,
-                                   const ParsedRtcEventLog& parsed_log,
-                                   const AnalyzerConfig& config,
-                                   Plot* plot);
-
-void CreatePacketRateGraph(PacketDirection direction,
-                           const ParsedRtcEventLog& parsed_log,
-                           const AnalyzerConfig& config,
-                           Plot* plot);
-
-void CreateTotalPacketRateGraph(PacketDirection direction,
-                                const ParsedRtcEventLog& parsed_log,
-                                const AnalyzerConfig& config,
-                                Plot* plot);
-
-void CreateSequenceNumberGraph(const ParsedRtcEventLog& parsed_log,
-                               const AnalyzerConfig& config,
-                               Plot* plot);
-
-void CreateIncomingPacketLossGraph(const ParsedRtcEventLog& parsed_log,
-                                   const AnalyzerConfig& config,
-                                   Plot* plot);
-
-void CreateStreamBitrateGraph(PacketDirection direction,
-                              const ParsedRtcEventLog& parsed_log,
-                              const AnalyzerConfig& config,
-                              Plot* plot);
-
-void CreateBitrateAllocationGraph(PacketDirection direction,
-                                  const ParsedRtcEventLog& parsed_log,
-                                  const AnalyzerConfig& config,
-                                  Plot* plot);
-
-void CreateOutgoingEcnFeedbackGraph(const ParsedRtcEventLog& parsed_log,
-                                    const AnalyzerConfig& config,
-                                    Plot* plot);
-
-void CreateIncomingEcnFeedbackGraph(const ParsedRtcEventLog& parsed_log,
-                                    const AnalyzerConfig& config,
-                                    Plot* plot);
-
-void CreateOutgoingLossRateGraph(const ParsedRtcEventLog& parsed_log,
-                                 const AnalyzerConfig& config,
-                                 Plot* plot);
-
-void CreateTimestampGraph(PacketDirection direction,
-                          const ParsedRtcEventLog& parsed_log,
-                          const AnalyzerConfig& config,
-                          Plot* plot);
-
-void CreateSenderAndReceiverReportPlot(
-    PacketDirection direction,
-    FunctionView<float(const rtcp::ReportBlock&)> fy,
-    std::string title,
-    std::string yaxis_label,
-    const ParsedRtcEventLog& parsed_log,
-    const AnalyzerConfig& config,
-    Plot* plot);
-
-}  // namespace webrtc
-
-#endif  // RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZE_RTP_RTCP_H_
diff --git a/rtc_tools/rtc_event_log_visualizer/analyzer.cc b/rtc_tools/rtc_event_log_visualizer/analyzer.cc
index 73e1db7509..cc7a5457bd 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyzer.cc
+++ b/rtc_tools/rtc_event_log_visualizer/analyzer.cc
@@ -10,35 +10,476 @@
 
 #include "rtc_tools/rtc_event_log_visualizer/analyzer.h"
 
+#include <algorithm>
+#include <cmath>
 #include <cstddef>
+#include <cstdint>
+#include <deque>
+#include <limits>
+#include <map>
+#include <memory>
+#include <optional>
 #include <string>
+#include <tuple>
+#include <unordered_set>
+#include <utility>
 #include <vector>
 
 #include "absl/algorithm/container.h"
+#include "absl/functional/bind_front.h"
 #include "absl/strings/string_view.h"
+#include "api/candidate.h"
+#include "api/dtls_transport_interface.h"
+#include "api/environment/environment.h"
 #include "api/environment/environment_factory.h"
+#include "api/field_trials.h"
 #include "api/function_view.h"
+#include "api/media_types.h"
+#include "api/rtp_headers.h"
+#include "api/transport/bandwidth_usage.h"
+#include "api/transport/ecn_marking.h"
+#include "api/transport/goog_cc_factory.h"
+#include "api/transport/network_control.h"
+#include "api/transport/network_types.h"
 #include "api/units/data_rate.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
+#include "logging/rtc_event_log/events/logged_rtp_rtcp.h"
+#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair.h"
+#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.h"
 #include "logging/rtc_event_log/rtc_event_log_parser.h"
+#include "logging/rtc_event_log/rtc_event_processor.h"
+#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.h"
+#include "modules/congestion_controller/include/receive_side_congestion_controller.h"
+#include "modules/congestion_controller/rtp/transport_feedback_adapter.h"
+#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/congestion_control_feedback.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/remb.h"
 #include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/target_bitrate.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
+#include "rtc_base/network/sent_packet.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
+#include "rtc_base/rate_statistics.h"
+#include "rtc_base/strings/string_builder.h"
 #include "rtc_tools/rtc_event_log_visualizer/analyze_audio.h"
-#include "rtc_tools/rtc_event_log_visualizer/analyze_bwe.h"
-#include "rtc_tools/rtc_event_log_visualizer/analyze_connectivity.h"
-#include "rtc_tools/rtc_event_log_visualizer/analyze_rtp_rtcp.h"
 #include "rtc_tools/rtc_event_log_visualizer/analyzer_common.h"
+#include "rtc_tools/rtc_event_log_visualizer/log_scream_simulation.h"
+#include "rtc_tools/rtc_event_log_visualizer/log_simulation.h"
 #include "rtc_tools/rtc_event_log_visualizer/plot_base.h"
+#include "system_wrappers/include/clock.h"
 
 namespace webrtc {
 
-EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& parsed_log,
+namespace {
+
+std::string SsrcToString(uint32_t ssrc) {
+  StringBuilder ss;
+  ss << "SSRC " << ssrc;
+  return ss.Release();
+}
+
+// Checks whether an SSRC is contained in the list of desired SSRCs.
+// Note that an empty SSRC list matches every SSRC.
+bool MatchingSsrc(uint32_t ssrc, const std::vector<uint32_t>& desired_ssrc) {
+  if (desired_ssrc.empty())
+    return true;
+  return std::find(desired_ssrc.begin(), desired_ssrc.end(), ssrc) !=
+         desired_ssrc.end();
+}
+
+double AbsSendTimeToMicroseconds(int64_t abs_send_time) {
+  // The timestamp is a fixed point representation with 6 bits for seconds
+  // and 18 bits for fractions of a second. Thus, we divide by 2^18 to get the
+  // time in seconds and then multiply by kNumMicrosecsPerSec to convert to
+  // microseconds.
+  static constexpr double kTimestampToMicroSec =
+      static_cast<double>(kNumMicrosecsPerSec) / static_cast<double>(1ul << 18);
+  return abs_send_time * kTimestampToMicroSec;
+}
+
+// Computes the difference `later` - `earlier` where `later` and `earlier`
+// are counters that wrap at `modulus`. The difference is chosen to have the
+// least absolute value. For example if `modulus` is 8, then the difference will
+// be chosen in the range [-3, 4]. If `modulus` is 9, then the difference will
+// be in [-4, 4].
+int64_t WrappingDifference(uint32_t later, uint32_t earlier, int64_t modulus) {
+  RTC_DCHECK_LE(1, modulus);
+  RTC_DCHECK_LT(later, modulus);
+  RTC_DCHECK_LT(earlier, modulus);
+  int64_t difference =
+      static_cast<int64_t>(later) - static_cast<int64_t>(earlier);
+  int64_t max_difference = modulus / 2;
+  int64_t min_difference = max_difference - modulus + 1;
+  if (difference > max_difference) {
+    difference -= modulus;
+  }
+  if (difference < min_difference) {
+    difference += modulus;
+  }
+  if (difference > max_difference / 2 || difference < min_difference / 2) {
+    RTC_LOG(LS_WARNING) << "Difference between" << later << " and " << earlier
+                        << " expected to be in the range ("
+                        << min_difference / 2 << "," << max_difference / 2
+                        << ") but is " << difference
+                        << ". Correct unwrapping is uncertain.";
+  }
+  return difference;
+}
+
+// This is much more reliable for outgoing streams than for incoming streams.
+template <typename RtpPacketContainer>
+std::optional<uint32_t> EstimateRtpClockFrequency(
+    const RtpPacketContainer& packets,
+    int64_t end_time_us) {
+  RTC_CHECK(packets.size() >= 2);
+  SeqNumUnwrapper<uint32_t> unwrapper;
+  int64_t first_rtp_timestamp =
+      unwrapper.Unwrap(packets[0].rtp.header.timestamp);
+  int64_t first_log_timestamp = packets[0].log_time_us();
+  int64_t last_rtp_timestamp = first_rtp_timestamp;
+  int64_t last_log_timestamp = first_log_timestamp;
+  for (size_t i = 1; i < packets.size(); i++) {
+    if (packets[i].log_time_us() > end_time_us)
+      break;
+    last_rtp_timestamp = unwrapper.Unwrap(packets[i].rtp.header.timestamp);
+    last_log_timestamp = packets[i].log_time_us();
+  }
+  if (last_log_timestamp - first_log_timestamp < kNumMicrosecsPerSec) {
+    RTC_LOG(LS_WARNING)
+        << "Failed to estimate RTP clock frequency: Stream too short. ("
+        << packets.size() << " packets, "
+        << last_log_timestamp - first_log_timestamp << " us)";
+    return std::nullopt;
+  }
+  double duration =
+      static_cast<double>(last_log_timestamp - first_log_timestamp) /
+      kNumMicrosecsPerSec;
+  double estimated_frequency =
+      (last_rtp_timestamp - first_rtp_timestamp) / duration;
+  for (uint32_t f : {8000, 16000, 32000, 48000, 90000}) {
+    if (std::fabs(estimated_frequency - f) < 0.15 * f) {
+      return f;
+    }
+  }
+  RTC_LOG(LS_WARNING) << "Failed to estimate RTP clock frequency: Estimate "
+                      << estimated_frequency
+                      << " not close to any standard RTP frequency."
+                      << " Last timestamp " << last_rtp_timestamp
+                      << " first timestamp " << first_rtp_timestamp;
+  return std::nullopt;
+}
+
+std::optional<double> NetworkDelayDiff_AbsSendTime(
+    const LoggedRtpPacketIncoming& old_packet,
+    const LoggedRtpPacketIncoming& new_packet) {
+  if (old_packet.rtp.header.extension.hasAbsoluteSendTime &&
+      new_packet.rtp.header.extension.hasAbsoluteSendTime) {
+    int64_t send_time_diff = WrappingDifference(
+        new_packet.rtp.header.extension.absoluteSendTime,
+        old_packet.rtp.header.extension.absoluteSendTime, 1ul << 24);
+    int64_t recv_time_diff =
+        new_packet.log_time_us() - old_packet.log_time_us();
+    double delay_change_us =
+        recv_time_diff - AbsSendTimeToMicroseconds(send_time_diff);
+    return delay_change_us / 1000;
+  } else {
+    return std::nullopt;
+  }
+}
+
+std::optional<double> NetworkDelayDiff_CaptureTime(
+    const LoggedRtpPacketIncoming& old_packet,
+    const LoggedRtpPacketIncoming& new_packet,
+    const double sample_rate) {
+  int64_t send_time_diff =
+      WrappingDifference(new_packet.rtp.header.timestamp,
+                         old_packet.rtp.header.timestamp, 1ull << 32);
+  int64_t recv_time_diff = new_packet.log_time_us() - old_packet.log_time_us();
+
+  double delay_change =
+      static_cast<double>(recv_time_diff) / 1000 -
+      static_cast<double>(send_time_diff) / sample_rate * 1000;
+  if (delay_change < -10000 || 10000 < delay_change) {
+    RTC_LOG(LS_WARNING) << "Very large delay change. Timestamps correct?";
+    RTC_LOG(LS_WARNING) << "Old capture time "
+                        << old_packet.rtp.header.timestamp << ", received time "
+                        << old_packet.log_time_us();
+    RTC_LOG(LS_WARNING) << "New capture time "
+                        << new_packet.rtp.header.timestamp << ", received time "
+                        << new_packet.log_time_us();
+    RTC_LOG(LS_WARNING) << "Receive time difference " << recv_time_diff << " = "
+                        << static_cast<double>(recv_time_diff) /
+                               kNumMicrosecsPerSec
+                        << "s";
+    RTC_LOG(LS_WARNING) << "Send time difference " << send_time_diff << " = "
+                        << static_cast<double>(send_time_diff) / sample_rate
+                        << "s";
+  }
+  return delay_change;
+}
+
+template <typename T>
+TimeSeries CreateRtcpTypeTimeSeries(const std::vector<T>& rtcp_list,
+                                    AnalyzerConfig config,
+                                    std::string rtcp_name,
+                                    int category_id) {
+  TimeSeries time_series(rtcp_name, LineStyle::kNone, PointStyle::kHighlight);
+  for (const auto& rtcp : rtcp_list) {
+    float x = config.GetCallTimeSec(rtcp.timestamp);
+    float y = category_id;
+    time_series.points.emplace_back(x, y);
+  }
+  return time_series;
+}
+
+const char kUnknownEnumValue[] = "unknown";
+
+// TODO(tommi): This should be "host".
+const char kIceCandidateTypeLocal[] = "local";
+// TODO(tommi): This should be "srflx".
+const char kIceCandidateTypeStun[] = "stun";
+const char kIceCandidateTypePrflx[] = "prflx";
+const char kIceCandidateTypeRelay[] = "relay";
+
+const char kProtocolUdp[] = "udp";
+const char kProtocolTcp[] = "tcp";
+const char kProtocolSsltcp[] = "ssltcp";
+const char kProtocolTls[] = "tls";
+
+const char kAddressFamilyIpv4[] = "ipv4";
+const char kAddressFamilyIpv6[] = "ipv6";
+
+const char kNetworkTypeEthernet[] = "ethernet";
+const char kNetworkTypeLoopback[] = "loopback";
+const char kNetworkTypeWifi[] = "wifi";
+const char kNetworkTypeVpn[] = "vpn";
+const char kNetworkTypeCellular[] = "cellular";
+
+absl::string_view GetIceCandidateTypeAsString(IceCandidateType type) {
+  switch (type) {
+    case IceCandidateType::kHost:
+      return kIceCandidateTypeLocal;
+    case IceCandidateType::kSrflx:
+      return kIceCandidateTypeStun;
+    case IceCandidateType::kPrflx:
+      return kIceCandidateTypePrflx;
+    case IceCandidateType::kRelay:
+      return kIceCandidateTypeRelay;
+    default:
+      RTC_DCHECK_NOTREACHED();
+      return kUnknownEnumValue;
+  }
+}
+
+std::string GetProtocolAsString(IceCandidatePairProtocol protocol) {
+  switch (protocol) {
+    case IceCandidatePairProtocol::kUdp:
+      return kProtocolUdp;
+    case IceCandidatePairProtocol::kTcp:
+      return kProtocolTcp;
+    case IceCandidatePairProtocol::kSsltcp:
+      return kProtocolSsltcp;
+    case IceCandidatePairProtocol::kTls:
+      return kProtocolTls;
+    default:
+      return kUnknownEnumValue;
+  }
+}
+
+std::string GetAddressFamilyAsString(IceCandidatePairAddressFamily family) {
+  switch (family) {
+    case IceCandidatePairAddressFamily::kIpv4:
+      return kAddressFamilyIpv4;
+    case IceCandidatePairAddressFamily::kIpv6:
+      return kAddressFamilyIpv6;
+    default:
+      return kUnknownEnumValue;
+  }
+}
+
+std::string GetNetworkTypeAsString(IceCandidateNetworkType type) {
+  switch (type) {
+    case IceCandidateNetworkType::kEthernet:
+      return kNetworkTypeEthernet;
+    case IceCandidateNetworkType::kLoopback:
+      return kNetworkTypeLoopback;
+    case IceCandidateNetworkType::kWifi:
+      return kNetworkTypeWifi;
+    case IceCandidateNetworkType::kVpn:
+      return kNetworkTypeVpn;
+    case IceCandidateNetworkType::kCellular:
+      return kNetworkTypeCellular;
+    default:
+      return kUnknownEnumValue;
+  }
+}
+
+std::string GetCandidatePairLogDescriptionAsString(
+    const LoggedIceCandidatePairConfig& config) {
+  // Example: stun:wifi->relay(tcp):cellular@udp:ipv4
+  // represents a pair of a local server-reflexive candidate on a WiFi network
+  // and a remote relay candidate using TCP as the relay protocol on a cell
+  // network, when the candidate pair communicates over UDP using IPv4.
+  StringBuilder ss;
+  ss << GetIceCandidateTypeAsString(config.local_candidate_type);
+
+  if (config.local_candidate_type == IceCandidateType::kRelay) {
+    ss << "(" << GetProtocolAsString(config.local_relay_protocol) << ")";
+  }
+
+  ss << ":" << GetNetworkTypeAsString(config.local_network_type) << ":"
+     << GetAddressFamilyAsString(config.local_address_family) << "->"
+     << GetIceCandidateTypeAsString(config.remote_candidate_type) << ":"
+     << GetAddressFamilyAsString(config.remote_address_family) << "@"
+     << GetProtocolAsString(config.candidate_pair_protocol);
+  return ss.Release();
+}
+
+std::string GetDirectionAsString(PacketDirection direction) {
+  if (direction == kIncomingPacket) {
+    return "Incoming";
+  } else {
+    return "Outgoing";
+  }
+}
+
+std::string GetDirectionAsShortString(PacketDirection direction) {
+  if (direction == kIncomingPacket) {
+    return "In";
+  } else {
+    return "Out";
+  }
+}
+
+struct FakeExtensionSmall {
+  static constexpr RTPExtensionType kId = kRtpExtensionMid;
+  static constexpr absl::string_view Uri() { return "fake-extension-small"; }
+};
+struct FakeExtensionLarge {
+  static constexpr RTPExtensionType kId = kRtpExtensionRtpStreamId;
+  static constexpr absl::string_view Uri() { return "fake-extension-large"; }
+};
+
+RtpPacketReceived RtpPacketForBWEFromHeader(const RTPHeader& header) {
+  RtpHeaderExtensionMap rtp_header_extensions(/*extmap_allow_mixed=*/true);
+  // ReceiveSideCongestionController doesn't need to know extensions ids as
+  // long as it able to get extensions by type. So any ids would work here.
+  rtp_header_extensions.Register<TransmissionOffset>(1);
+  rtp_header_extensions.Register<AbsoluteSendTime>(2);
+  rtp_header_extensions.Register<TransportSequenceNumber>(3);
+  rtp_header_extensions.Register<FakeExtensionSmall>(4);
+  // Use id > 14 to force two byte header per rtp header when this one is used.
+  rtp_header_extensions.Register<FakeExtensionLarge>(16);
+
+  RtpPacketReceived rtp_packet(&rtp_header_extensions);
+  // Set only fields that might be relevant for the bandwidth estimatior.
+  rtp_packet.SetSsrc(header.ssrc);
+  rtp_packet.SetTimestamp(header.timestamp);
+  size_t num_bwe_extensions = 0;
+  if (header.extension.hasTransmissionTimeOffset) {
+    rtp_packet.SetExtension<TransmissionOffset>(
+        header.extension.transmissionTimeOffset);
+    ++num_bwe_extensions;
+  }
+  if (header.extension.hasAbsoluteSendTime) {
+    rtp_packet.SetExtension<AbsoluteSendTime>(
+        header.extension.absoluteSendTime);
+    ++num_bwe_extensions;
+  }
+  if (header.extension.hasTransportSequenceNumber) {
+    rtp_packet.SetExtension<TransportSequenceNumber>(
+        header.extension.transportSequenceNumber);
+    ++num_bwe_extensions;
+  }
+
+  // All parts of the RTP header are 32bit aligned.
+  RTC_CHECK_EQ(header.headerLength % 4, 0);
+
+  // Original packet could have more extensions, there could be csrcs that are
+  // not propagated by the rtc event log, i.e. logged header size might be
+  // larger that rtp_packet.header_size(). Increase it by setting an extra fake
+  // extension.
+  RTC_CHECK_GE(header.headerLength, rtp_packet.headers_size());
+  size_t bytes_to_add = header.headerLength - rtp_packet.headers_size();
+  if (bytes_to_add > 0) {
+    if (bytes_to_add <= 16) {
+      // one-byte header rtp header extension allows to add up to 16 bytes.
+      rtp_packet.AllocateExtension(FakeExtensionSmall::kId, bytes_to_add - 1);
+    } else {
+      // two-byte header rtp header extension would also add one byte per
+      // already set extension.
+      rtp_packet.AllocateExtension(FakeExtensionLarge::kId,
+                                   bytes_to_add - 2 - num_bwe_extensions);
+    }
+  }
+  RTC_CHECK_EQ(rtp_packet.headers_size(), header.headerLength);
+
+  return rtp_packet;
+}
+
+struct PacketLossSummary {
+  size_t num_packets = 0;
+  size_t num_lost_packets = 0;
+  Timestamp base_time = Timestamp::MinusInfinity();
+};
+
+float GetHighestSeqNumber(const rtcp::ReportBlock& block) {
+  return block.extended_high_seq_num();
+}
+
+float GetFractionLost(const rtcp::ReportBlock& block) {
+  return static_cast<double>(block.fraction_lost()) / 256 * 100;
+}
+
+float GetCumulativeLost(const rtcp::ReportBlock& block) {
+  return block.cumulative_lost();
+}
+
+float DelaySinceLastSr(const rtcp::ReportBlock& block) {
+  return static_cast<double>(block.delay_since_last_sr()) / 65536;
+}
+
+std::map<uint32_t, std::string> BuildCandidateIdLogDescriptionMap(
+    const std::vector<LoggedIceCandidatePairConfig>&
+        ice_candidate_pair_configs) {
+  std::map<uint32_t, std::string> candidate_pair_desc_by_id;
+  for (const auto& config : ice_candidate_pair_configs) {
+    // TODO(qingsi): Add the handling of the "Updated" config event after the
+    // visualization of property change for candidate pairs is introduced.
+    if (candidate_pair_desc_by_id.find(config.candidate_pair_id) ==
+        candidate_pair_desc_by_id.end()) {
+      const std::string candidate_pair_desc =
+          GetCandidatePairLogDescriptionAsString(config);
+      candidate_pair_desc_by_id[config.candidate_pair_id] = candidate_pair_desc;
+    }
+  }
+  return candidate_pair_desc_by_id;
+}
+
+}  // namespace
+
+EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log,
                                    bool normalize_time)
-    : parsed_log_(parsed_log),
-      config_(CreateEnvironment(), parsed_log, normalize_time) {
+    : env_(CreateEnvironment()), parsed_log_(log) {
   config_.window_duration_ = TimeDelta::Millis(250);
   config_.step_ = TimeDelta::Millis(10);
+  if (!log.start_log_events().empty()) {
+    config_.rtc_to_utc_offset_ = log.start_log_events()[0].utc_time() -
+                                 log.start_log_events()[0].log_time();
+  }
+  config_.normalize_time_ = normalize_time;
+  config_.begin_time_ = parsed_log_.first_timestamp();
+  config_.end_time_ = parsed_log_.last_timestamp();
   if (config_.end_time_ < config_.begin_time_) {
     RTC_LOG(LS_WARNING) << "No useful events in the log.";
     config_.begin_time_ = config_.end_time_ = Timestamp::Zero();
@@ -51,9 +492,10 @@ EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& parsed_log,
                    << " seconds long.";
 }
 
-EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& parsed_log,
+EventLogAnalyzer::EventLogAnalyzer(const Environment& env,
+                                   const ParsedRtcEventLog& log,
                                    const AnalyzerConfig& config)
-    : parsed_log_(parsed_log), config_(config) {
+    : env_(env), parsed_log_(log), config_(config) {
   RTC_LOG(LS_INFO) << "Log is "
                    << (parsed_log_.last_timestamp().ms() -
                        parsed_log_.first_timestamp().ms()) /
@@ -61,18 +503,31 @@ EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& parsed_log,
                    << " seconds long.";
 }
 
-void EventLogAnalyzer::CreateGraphsByName(const std::vector<std::string>& names,
-                                          PlotCollection* collection) const {
-  for (absl::string_view name : names) {
-    auto plot = absl::c_find_if(plots_, [name](const PlotDeclaration& plot) {
-      return plot.label == name;
-    });
-    if (plot != plots_.end()) {
-      Plot* output = collection->AppendNewPlot(plot->label);
-      plot->plot_func(output);
+class BitrateObserver : public RemoteBitrateObserver {
+ public:
+  BitrateObserver() : last_bitrate_bps_(0), bitrate_updated_(false) {}
+
+  void Update(NetworkControlUpdate update) {
+    if (update.target_rate) {
+      last_bitrate_bps_ = update.target_rate->target_rate.bps();
+      bitrate_updated_ = true;
     }
   }
-}
+
+  void OnReceiveBitrateChanged(const std::vector<uint32_t>& ssrcs,
+                               uint32_t bitrate) override {}
+
+  uint32_t last_bitrate_bps() const { return last_bitrate_bps_; }
+  bool GetAndResetBitrateUpdated() {
+    bool bitrate_updated = bitrate_updated_;
+    bitrate_updated_ = false;
+    return bitrate_updated;
+  }
+
+ private:
+  uint32_t last_bitrate_bps_;
+  bool bitrate_updated_;
+};
 
 void EventLogAnalyzer::InitializeMapOfNamedGraphs(bool show_detector_state,
                                                   bool show_alr_state,
@@ -282,155 +737,1664 @@ void EventLogAnalyzer::InitializeMapOfNamedGraphs(bool show_detector_state,
     this->CreateDtlsWritableStateGraph(plot);
   });
 }
+
+void EventLogAnalyzer::CreateGraphsByName(const std::vector<std::string>& names,
+                                          PlotCollection* collection) const {
+  for (absl::string_view name : names) {
+    auto plot = absl::c_find_if(plots_, [name](const PlotDeclaration& plot) {
+      return plot.label == name;
+    });
+    if (plot != plots_.end()) {
+      Plot* output = collection->AppendNewPlot(plot->label);
+      plot->plot_func(output);
+    }
+  }
+}
+
+void EventLogAnalyzer::CreatePacketGraph(PacketDirection direction,
+                                         Plot* plot) const {
+  for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
+    // Filter on SSRC.
+    if (!MatchingSsrc(stream.ssrc, desired_ssrc_)) {
+      continue;
+    }
+
+    TimeSeries time_series(GetStreamName(parsed_log_, direction, stream.ssrc),
+                           LineStyle::kBar);
+    auto GetPacketSize = [](const LoggedRtpPacket& packet) {
+      return std::optional<float>(packet.total_length);
+    };
+    auto ToCallTime = [this](const LoggedRtpPacket& packet) {
+      return this->config_.GetCallTimeSec(packet.timestamp);
+    };
+    ProcessPoints<LoggedRtpPacket>(ToCallTime, GetPacketSize,
+                                   stream.packet_view, &time_series);
+    plot->AppendTimeSeries(std::move(time_series));
+  }
+
+  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
+                 "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Packet size (bytes)", kBottomMargin,
+                          kTopMargin);
+  plot->SetTitle(GetDirectionAsString(direction) + " RTP packets");
+}
+
+void EventLogAnalyzer::CreateRtcpTypeGraph(PacketDirection direction,
+                                           Plot* plot) const {
+  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(
+      parsed_log_.transport_feedbacks(direction), config_, "TWCC", 1));
+  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(
+      parsed_log_.congestion_feedback(direction), config_, "CCFB", 2));
+  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(
+      parsed_log_.receiver_reports(direction), config_, "RR", 3));
+  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(
+      parsed_log_.sender_reports(direction), config_, "SR", 4));
+  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(
+      parsed_log_.extended_reports(direction), config_, "XR", 5));
+  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(parsed_log_.nacks(direction),
+                                                  config_, "NACK", 6));
+  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(parsed_log_.rembs(direction),
+                                                  config_, "REMB", 7));
+  plot->AppendTimeSeries(
+      CreateRtcpTypeTimeSeries(parsed_log_.firs(direction), config_, "FIR", 8));
+  plot->AppendTimeSeries(
+      CreateRtcpTypeTimeSeries(parsed_log_.plis(direction), config_, "PLI", 9));
+  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(parsed_log_.byes(direction),
+                                                  config_, "BYE", 10));
+  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
+                 "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "RTCP type", kBottomMargin, kTopMargin);
+  plot->SetTitle(GetDirectionAsString(direction) + " RTCP packets");
+  plot->SetYAxisTickLabels({{1, "TWCC"},
+                            {2, "CCFB"},
+                            {3, "RR"},
+                            {4, "SR"},
+                            {5, "XR"},
+                            {6, "NACK"},
+                            {7, "REMB"},
+                            {8, "FIR"},
+                            {9, "PLI"},
+                            {10, "BYE"}});
+}
+
+template <typename IterableType>
+void EventLogAnalyzer::CreateAccumulatedPacketsTimeSeries(
+    Plot* plot,
+    const IterableType& packets,
+    const std::string& label) const {
+  TimeSeries time_series(label, LineStyle::kStep);
+  for (size_t i = 0; i < packets.size(); i++) {
+    float x = config_.GetCallTimeSec(packets[i].log_time());
+    time_series.points.emplace_back(x, i + 1);
+  }
+  plot->AppendTimeSeries(std::move(time_series));
+}
+
+void EventLogAnalyzer::CreateAccumulatedPacketsGraph(PacketDirection direction,
+                                                     Plot* plot) const {
+  for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
+    if (!MatchingSsrc(stream.ssrc, desired_ssrc_))
+      continue;
+    std::string label = std::string("RTP ") +
+                        GetStreamName(parsed_log_, direction, stream.ssrc);
+    CreateAccumulatedPacketsTimeSeries(plot, stream.packet_view, label);
+  }
+  std::string label =
+      std::string("RTCP ") + "(" + GetDirectionAsShortString(direction) + ")";
+  if (direction == kIncomingPacket) {
+    CreateAccumulatedPacketsTimeSeries(
+        plot, parsed_log_.incoming_rtcp_packets(), label);
+  } else {
+    CreateAccumulatedPacketsTimeSeries(
+        plot, parsed_log_.outgoing_rtcp_packets(), label);
+  }
+
+  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
+                 "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Received Packets", kBottomMargin, kTopMargin);
+  plot->SetTitle(std::string("Accumulated ") + GetDirectionAsString(direction) +
+                 " RTP/RTCP packets");
+}
+
+void EventLogAnalyzer::CreatePacketRateGraph(PacketDirection direction,
+                                             Plot* plot) const {
+  auto CountPackets = [](auto packet) { return 1.0; };
+  for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
+    // Filter on SSRC.
+    if (!MatchingSsrc(stream.ssrc, desired_ssrc_)) {
+      continue;
+    }
+    TimeSeries time_series(
+        std::string("RTP ") +
+            GetStreamName(parsed_log_, direction, stream.ssrc),
+        LineStyle::kLine);
+    MovingAverage<LoggedRtpPacket, double>(CountPackets, stream.packet_view,
+                                           config_, &time_series);
+    plot->AppendTimeSeries(std::move(time_series));
+  }
+  TimeSeries time_series(
+      std::string("RTCP ") + "(" + GetDirectionAsShortString(direction) + ")",
+      LineStyle::kLine);
+  if (direction == kIncomingPacket) {
+    MovingAverage<LoggedRtcpPacketIncoming, double>(
+        CountPackets, parsed_log_.incoming_rtcp_packets(), config_,
+        &time_series);
+  } else {
+    MovingAverage<LoggedRtcpPacketOutgoing, double>(
+        CountPackets, parsed_log_.outgoing_rtcp_packets(), config_,
+        &time_series);
+  }
+  plot->AppendTimeSeries(std::move(time_series));
+
+  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
+                 "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Packet Rate (packets/s)", kBottomMargin,
+                          kTopMargin);
+  plot->SetTitle("Rate of " + GetDirectionAsString(direction) +
+                 " RTP/RTCP packets");
+}
+
+void EventLogAnalyzer::CreateTotalPacketRateGraph(PacketDirection direction,
+                                                  Plot* plot) const {
+  // Contains a log timestamp to enable counting logged events of different
+  // types using MovingAverage().
+  class LogTime {
+   public:
+    explicit LogTime(Timestamp log_time) : log_time_(log_time) {}
+    Timestamp log_time() const { return log_time_; }
+
+   private:
+    Timestamp log_time_;
+  };
+  std::vector<LogTime> packet_times;
+  auto handle_rtp = [&packet_times](const LoggedRtpPacket& packet) {
+    packet_times.emplace_back(packet.log_time());
+  };
+  RtcEventProcessor process;
+  for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
+    process.AddEvents(stream.packet_view, handle_rtp, direction);
+  }
+  if (direction == kIncomingPacket) {
+    auto handle_incoming_rtcp =
+        [&packet_times](const LoggedRtcpPacketIncoming& packet) {
+          packet_times.emplace_back(packet.log_time());
+        };
+    process.AddEvents(parsed_log_.incoming_rtcp_packets(),
+                      handle_incoming_rtcp);
+  } else {
+    auto handle_outgoing_rtcp =
+        [&packet_times](const LoggedRtcpPacketOutgoing& packet) {
+          packet_times.emplace_back(packet.log_time());
+        };
+    process.AddEvents(parsed_log_.outgoing_rtcp_packets(),
+                      handle_outgoing_rtcp);
+  }
+  process.ProcessEventsInOrder();
+  TimeSeries time_series(std::string("Total ") + "(" +
+                             GetDirectionAsShortString(direction) + ") packets",
+                         LineStyle::kLine);
+  MovingAverage<LogTime, uint64_t>([](auto packet) { return 1; }, packet_times,
+                                   config_, &time_series);
+  plot->AppendTimeSeries(std::move(time_series));
+
+  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
+                 "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Packet Rate (packets/s)", kBottomMargin,
+                          kTopMargin);
+  plot->SetTitle("Rate of all " + GetDirectionAsString(direction) +
+                 " RTP/RTCP packets");
+}
+
+// For each SSRC, plot the time between the consecutive playouts.
 void EventLogAnalyzer::CreatePlayoutGraph(Plot* plot) const {
-  webrtc::CreatePlayoutGraph(parsed_log_, config_, plot);
+  for (const auto& playout_stream : parsed_log_.audio_playout_events()) {
+    uint32_t ssrc = playout_stream.first;
+    if (!MatchingSsrc(ssrc, desired_ssrc_))
+      continue;
+    std::optional<int64_t> last_playout_ms;
+    TimeSeries time_series(SsrcToString(ssrc), LineStyle::kBar);
+    for (const auto& playout_event : playout_stream.second) {
+      float x = config_.GetCallTimeSec(playout_event.log_time());
+      int64_t playout_time_ms = playout_event.log_time_ms();
+      // If there were no previous playouts, place the point on the x-axis.
+      float y = playout_time_ms - last_playout_ms.value_or(playout_time_ms);
+      time_series.points.push_back(TimeSeriesPoint(x, y));
+      last_playout_ms.emplace(playout_time_ms);
+    }
+    plot->AppendTimeSeries(std::move(time_series));
+  }
+
+  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
+                 "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Time since last playout (ms)", kBottomMargin,
+                          kTopMargin);
+  plot->SetTitle("Audio playout");
 }
 
 void EventLogAnalyzer::CreateNetEqSetMinimumDelay(Plot* plot) const {
-  webrtc::CreateNetEqSetMinimumDelay(parsed_log_, config_, plot);
+  for (const auto& playout_stream :
+       parsed_log_.neteq_set_minimum_delay_events()) {
+    uint32_t ssrc = playout_stream.first;
+    if (!MatchingSsrc(ssrc, desired_ssrc_))
+      continue;
+
+    TimeSeries time_series(SsrcToString(ssrc), LineStyle::kStep,
+                           PointStyle::kHighlight);
+    for (const auto& event : playout_stream.second) {
+      float x = config_.GetCallTimeSec(event.log_time());
+      float y = event.minimum_delay_ms;
+      time_series.points.push_back(TimeSeriesPoint(x, y));
+    }
+    plot->AppendTimeSeries(std::move(time_series));
+  }
+
+  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
+                 "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1000, "Minimum Delay (ms)", kBottomMargin,
+                          kTopMargin);
+  plot->SetTitle("Set Minimum Delay");
 }
 
+// For audio SSRCs, plot the audio level.
 void EventLogAnalyzer::CreateAudioLevelGraph(PacketDirection direction,
                                              Plot* plot) const {
-  webrtc::CreateAudioLevelGraph(parsed_log_, config_, direction, plot);
+  for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
+    if (!IsAudioSsrc(parsed_log_, direction, stream.ssrc))
+      continue;
+    TimeSeries time_series(GetStreamName(parsed_log_, direction, stream.ssrc),
+                           LineStyle::kLine);
+    for (auto& packet : stream.packet_view) {
+      if (packet.header.extension.audio_level()) {
+        float x = config_.GetCallTimeSec(packet.log_time());
+        // The audio level is stored in -dBov (so e.g. -10 dBov is stored as 10)
+        // Here we convert it to dBov.
+        float y =
+            static_cast<float>(-packet.header.extension.audio_level()->level());
+        time_series.points.emplace_back(TimeSeriesPoint(x, y));
+      }
+    }
+    plot->AppendTimeSeries(std::move(time_series));
+  }
+
+  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
+                 "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetYAxis(-127, 0, "Audio level (dBov)", kBottomMargin, kTopMargin);
+  plot->SetTitle(GetDirectionAsString(direction) + " audio level");
+}
+
+// For each SSRC, plot the sequence number difference between consecutive
+// incoming packets.
+void EventLogAnalyzer::CreateSequenceNumberGraph(Plot* plot) const {
+  for (const auto& stream : parsed_log_.incoming_rtp_packets_by_ssrc()) {
+    // Filter on SSRC.
+    if (!MatchingSsrc(stream.ssrc, desired_ssrc_)) {
+      continue;
+    }
+
+    TimeSeries time_series(
+        GetStreamName(parsed_log_, kIncomingPacket, stream.ssrc),
+        LineStyle::kBar);
+    auto GetSequenceNumberDiff = [](const LoggedRtpPacketIncoming& old_packet,
+                                    const LoggedRtpPacketIncoming& new_packet) {
+      int64_t diff =
+          WrappingDifference(new_packet.rtp.header.sequenceNumber,
+                             old_packet.rtp.header.sequenceNumber, 1ul << 16);
+      return diff;
+    };
+    auto ToCallTime = [this](const LoggedRtpPacketIncoming& packet) {
+      return this->config_.GetCallTimeSec(packet.log_time());
+    };
+    ProcessPairs<LoggedRtpPacketIncoming, float>(
+        ToCallTime, GetSequenceNumberDiff, stream.incoming_packets,
+        &time_series);
+    plot->AppendTimeSeries(std::move(time_series));
+  }
+
+  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
+                 "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Difference since last packet", kBottomMargin,
+                          kTopMargin);
+  plot->SetTitle("Incoming sequence number delta");
+}
+
+void EventLogAnalyzer::CreateIncomingPacketLossGraph(Plot* plot) const {
+  for (const auto& stream : parsed_log_.incoming_rtp_packets_by_ssrc()) {
+    const std::vector<LoggedRtpPacketIncoming>& packets =
+        stream.incoming_packets;
+    // Filter on SSRC.
+    if (!MatchingSsrc(stream.ssrc, desired_ssrc_) || packets.empty()) {
+      continue;
+    }
+
+    TimeSeries time_series(
+        GetStreamName(parsed_log_, kIncomingPacket, stream.ssrc),
+        LineStyle::kLine, PointStyle::kHighlight);
+    // TODO(terelius): Should the window and step size be read from the class
+    // instead?
+    const TimeDelta kWindow = TimeDelta::Millis(1000);
+    const TimeDelta kStep = TimeDelta::Millis(1000);
+    SeqNumUnwrapper<uint16_t> unwrapper_;
+    SeqNumUnwrapper<uint16_t> prior_unwrapper_;
+    size_t window_index_begin = 0;
+    size_t window_index_end = 0;
+    uint64_t highest_seq_number =
+        unwrapper_.Unwrap(packets[0].rtp.header.sequenceNumber) - 1;
+    uint64_t highest_prior_seq_number =
+        prior_unwrapper_.Unwrap(packets[0].rtp.header.sequenceNumber) - 1;
+
+    for (Timestamp t = config_.begin_time_; t < config_.end_time_ + kStep;
+         t += kStep) {
+      while (window_index_end < packets.size() &&
+             packets[window_index_end].rtp.log_time() < t) {
+        uint64_t sequence_number = unwrapper_.Unwrap(
+            packets[window_index_end].rtp.header.sequenceNumber);
+        highest_seq_number = std::max(highest_seq_number, sequence_number);
+        ++window_index_end;
+      }
+      while (window_index_begin < packets.size() &&
+             packets[window_index_begin].rtp.log_time() < t - kWindow) {
+        uint64_t sequence_number = prior_unwrapper_.Unwrap(
+            packets[window_index_begin].rtp.header.sequenceNumber);
+        highest_prior_seq_number =
+            std::max(highest_prior_seq_number, sequence_number);
+        ++window_index_begin;
+      }
+      float x = config_.GetCallTimeSec(t);
+      uint64_t expected_packets = highest_seq_number - highest_prior_seq_number;
+      if (expected_packets > 0) {
+        int64_t received_packets = window_index_end - window_index_begin;
+        int64_t lost_packets = expected_packets - received_packets;
+        float y = static_cast<float>(lost_packets) / expected_packets * 100;
+        time_series.points.emplace_back(x, y);
+      }
+    }
+    plot->AppendTimeSeries(std::move(time_series));
+  }
+
+  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
+                 "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Loss rate (in %)", kBottomMargin, kTopMargin);
+  plot->SetTitle("Incoming packet loss (derived from incoming packets)");
 }
 
 void EventLogAnalyzer::CreateIncomingDelayGraph(Plot* plot) const {
-  webrtc::CreateIncomingDelayGraph(parsed_log_, config_, plot);
+  for (const auto& stream : parsed_log_.incoming_rtp_packets_by_ssrc()) {
+    // Filter on SSRC.
+    if (!MatchingSsrc(stream.ssrc, desired_ssrc_) ||
+        IsRtxSsrc(parsed_log_, kIncomingPacket, stream.ssrc)) {
+      continue;
+    }
+
+    const std::vector<LoggedRtpPacketIncoming>& packets =
+        stream.incoming_packets;
+    if (packets.size() < 100) {
+      RTC_LOG(LS_WARNING) << "Can't estimate the RTP clock frequency with "
+                          << packets.size() << " packets in the stream.";
+      continue;
+    }
+    int64_t segment_end_us = parsed_log_.first_log_segment().stop_time_us();
+    std::optional<uint32_t> estimated_frequency =
+        EstimateRtpClockFrequency(packets, segment_end_us);
+    if (!estimated_frequency)
+      continue;
+    const double frequency_hz = *estimated_frequency;
+    if (IsVideoSsrc(parsed_log_, kIncomingPacket, stream.ssrc) &&
+        frequency_hz != 90000) {
+      RTC_LOG(LS_WARNING)
+          << "Video stream should use a 90 kHz clock but appears to use "
+          << frequency_hz / 1000 << ". Discarding.";
+      continue;
+    }
+
+    auto ToCallTime = [this](const LoggedRtpPacketIncoming& packet) {
+      return this->config_.GetCallTimeSec(packet.log_time());
+    };
+    auto ToNetworkDelay = [frequency_hz](
+                              const LoggedRtpPacketIncoming& old_packet,
+                              const LoggedRtpPacketIncoming& new_packet) {
+      return NetworkDelayDiff_CaptureTime(old_packet, new_packet, frequency_hz);
+    };
+
+    TimeSeries capture_time_data(
+        GetStreamName(parsed_log_, kIncomingPacket, stream.ssrc) +
+            " capture-time",
+        LineStyle::kLine);
+    AccumulatePairs<LoggedRtpPacketIncoming, double>(
+        ToCallTime, ToNetworkDelay, packets, &capture_time_data);
+    plot->AppendTimeSeries(std::move(capture_time_data));
+
+    TimeSeries send_time_data(
+        GetStreamName(parsed_log_, kIncomingPacket, stream.ssrc) +
+            " abs-send-time",
+        LineStyle::kLine);
+    AccumulatePairs<LoggedRtpPacketIncoming, double>(
+        ToCallTime, NetworkDelayDiff_AbsSendTime, packets, &send_time_data);
+    plot->AppendTimeSeriesIfNotEmpty(std::move(send_time_data));
+  }
+
+  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
+                 "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Delay (ms)", kBottomMargin, kTopMargin);
+  plot->SetTitle("Incoming network delay (relative to first packet)");
 }
 
+// Plot the fraction of packets lost (as perceived by the loss-based BWE).
 void EventLogAnalyzer::CreateFractionLossGraph(Plot* plot) const {
-  webrtc::CreateFractionLossGraph(parsed_log_, config_, plot);
+  TimeSeries time_series("Fraction lost", LineStyle::kLine,
+                         PointStyle::kHighlight);
+  for (auto& bwe_update : parsed_log_.bwe_loss_updates()) {
+    float x = config_.GetCallTimeSec(bwe_update.log_time());
+    float y = static_cast<float>(bwe_update.fraction_lost) / 255 * 100;
+    time_series.points.emplace_back(x, y);
+  }
+
+  plot->AppendTimeSeries(std::move(time_series));
+  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
+                 "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 10, "Loss rate (in %)", kBottomMargin, kTopMargin);
+  plot->SetTitle("Outgoing packet loss (as reported by BWE)");
 }
 
+// Plot the total bandwidth used by all RTP streams.
 void EventLogAnalyzer::CreateTotalIncomingBitrateGraph(Plot* plot) const {
-  webrtc::CreateTotalIncomingBitrateGraph(parsed_log_, config_, plot);
+  // TODO(terelius): This could be provided by the parser.
+  std::multimap<Timestamp, size_t> packets_in_order;
+  for (const auto& stream : parsed_log_.incoming_rtp_packets_by_ssrc()) {
+    for (const LoggedRtpPacketIncoming& packet : stream.incoming_packets)
+      packets_in_order.insert(
+          std::make_pair(packet.rtp.log_time(), packet.rtp.total_length));
+  }
+
+  auto window_begin = packets_in_order.begin();
+  auto window_end = packets_in_order.begin();
+  size_t bytes_in_window = 0;
+
+  if (!packets_in_order.empty()) {
+    // Calculate a moving average of the bitrate and store in a TimeSeries.
+    TimeSeries bitrate_series("Bitrate", LineStyle::kLine);
+    for (Timestamp time = config_.begin_time_;
+         time < config_.end_time_ + config_.step_; time += config_.step_) {
+      while (window_end != packets_in_order.end() && window_end->first < time) {
+        bytes_in_window += window_end->second;
+        ++window_end;
+      }
+      while (window_begin != packets_in_order.end() &&
+             window_begin->first < time - config_.window_duration_) {
+        RTC_DCHECK_LE(window_begin->second, bytes_in_window);
+        bytes_in_window -= window_begin->second;
+        ++window_begin;
+      }
+      float window_duration_in_seconds =
+          static_cast<float>(config_.window_duration_.us()) /
+          kNumMicrosecsPerSec;
+      float x = config_.GetCallTimeSec(time);
+      float y = bytes_in_window * 8 / window_duration_in_seconds / 1000;
+      bitrate_series.points.emplace_back(x, y);
+    }
+    plot->AppendTimeSeries(std::move(bitrate_series));
+  }
+
+  // Overlay the outgoing REMB over incoming bitrate.
+  TimeSeries remb_series("Remb", LineStyle::kStep);
+  for (const auto& rtcp : parsed_log_.rembs(kOutgoingPacket)) {
+    float x = config_.GetCallTimeSec(rtcp.log_time());
+    float y = static_cast<float>(rtcp.remb.bitrate_bps()) / 1000;
+    remb_series.points.emplace_back(x, y);
+  }
+  plot->AppendTimeSeriesIfNotEmpty(std::move(remb_series));
+
+  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
+                 "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin);
+  plot->SetTitle("Incoming RTP bitrate");
 }
 
+// Plot the total bandwidth used by all RTP streams.
 void EventLogAnalyzer::CreateTotalOutgoingBitrateGraph(
     Plot* plot,
     bool show_detector_state,
     bool show_alr_state,
     bool show_link_capacity) const {
-  webrtc::CreateTotalOutgoingBitrateGraph(parsed_log_, config_, plot,
-                                          show_detector_state, show_alr_state,
-                                          show_link_capacity);
+  // TODO(terelius): This could be provided by the parser.
+  std::multimap<Timestamp, size_t> packets_in_order;
+  for (const auto& stream : parsed_log_.outgoing_rtp_packets_by_ssrc()) {
+    for (const LoggedRtpPacketOutgoing& packet : stream.outgoing_packets)
+      packets_in_order.insert(
+          std::make_pair(packet.rtp.log_time(), packet.rtp.total_length));
+  }
+
+  auto window_begin = packets_in_order.begin();
+  auto window_end = packets_in_order.begin();
+  size_t bytes_in_window = 0;
+
+  if (!packets_in_order.empty()) {
+    // Calculate a moving average of the bitrate and store in a TimeSeries.
+    TimeSeries bitrate_series("Bitrate", LineStyle::kLine);
+    for (Timestamp time = config_.begin_time_;
+         time < config_.end_time_ + config_.step_; time += config_.step_) {
+      while (window_end != packets_in_order.end() && window_end->first < time) {
+        bytes_in_window += window_end->second;
+        ++window_end;
+      }
+      while (window_begin != packets_in_order.end() &&
+             window_begin->first < time - config_.window_duration_) {
+        RTC_DCHECK_LE(window_begin->second, bytes_in_window);
+        bytes_in_window -= window_begin->second;
+        ++window_begin;
+      }
+      float window_duration_in_seconds =
+          static_cast<float>(config_.window_duration_.us()) /
+          kNumMicrosecsPerSec;
+      float x = config_.GetCallTimeSec(time);
+      float y = bytes_in_window * 8 / window_duration_in_seconds / 1000;
+      bitrate_series.points.emplace_back(x, y);
+    }
+    plot->AppendTimeSeries(std::move(bitrate_series));
+  }
+
+  // Overlay the send-side bandwidth estimate over the outgoing bitrate.
+  TimeSeries loss_series("Loss-based estimate", LineStyle::kStep);
+  for (auto& loss_update : parsed_log_.bwe_loss_updates()) {
+    float x = config_.GetCallTimeSec(loss_update.log_time());
+    float y = static_cast<float>(loss_update.bitrate_bps) / 1000;
+    loss_series.points.emplace_back(x, y);
+  }
+
+  TimeSeries link_capacity_lower_series("Link-capacity-lower",
+                                        LineStyle::kStep);
+  TimeSeries link_capacity_upper_series("Link-capacity-upper",
+                                        LineStyle::kStep);
+  for (auto& remote_estimate_event : parsed_log_.remote_estimate_events()) {
+    float x = config_.GetCallTimeSec(remote_estimate_event.log_time());
+    if (remote_estimate_event.link_capacity_lower.has_value()) {
+      float link_capacity_lower = static_cast<float>(
+          remote_estimate_event.link_capacity_lower.value().kbps());
+      link_capacity_lower_series.points.emplace_back(x, link_capacity_lower);
+    }
+    if (remote_estimate_event.link_capacity_upper.has_value()) {
+      float link_capacity_upper = static_cast<float>(
+          remote_estimate_event.link_capacity_upper.value().kbps());
+      link_capacity_upper_series.points.emplace_back(x, link_capacity_upper);
+    }
+  }
+
+  TimeSeries delay_series("Delay-based estimate", LineStyle::kStep);
+  IntervalSeries overusing_series("Overusing", "#ff8e82",
+                                  IntervalSeries::kHorizontal);
+  IntervalSeries underusing_series("Underusing", "#5092fc",
+                                   IntervalSeries::kHorizontal);
+  IntervalSeries normal_series("Normal", "#c4ffc4",
+                               IntervalSeries::kHorizontal);
+  IntervalSeries* last_series = &normal_series;
+  float last_detector_switch = 0.0;
+
+  BandwidthUsage last_detector_state = BandwidthUsage::kBwNormal;
+
+  for (auto& delay_update : parsed_log_.bwe_delay_updates()) {
+    float x = config_.GetCallTimeSec(delay_update.log_time());
+    float y = static_cast<float>(delay_update.bitrate_bps) / 1000;
+
+    if (last_detector_state != delay_update.detector_state) {
+      last_series->intervals.emplace_back(last_detector_switch, x);
+      last_detector_state = delay_update.detector_state;
+      last_detector_switch = x;
+
+      switch (delay_update.detector_state) {
+        case BandwidthUsage::kBwNormal:
+          last_series = &normal_series;
+          break;
+        case BandwidthUsage::kBwUnderusing:
+          last_series = &underusing_series;
+          break;
+        case BandwidthUsage::kBwOverusing:
+          last_series = &overusing_series;
+          break;
+        case BandwidthUsage::kLast:
+          RTC_DCHECK_NOTREACHED();
+      }
+    }
+
+    delay_series.points.emplace_back(x, y);
+  }
+
+  RTC_CHECK(last_series);
+  last_series->intervals.emplace_back(last_detector_switch,
+                                      config_.CallEndTimeSec());
+
+  TimeSeries scream_series("Scream target rate", LineStyle::kStep);
+  for (auto& scream_update : parsed_log_.bwe_scream_updates()) {
+    float x = config_.GetCallTimeSec(scream_update.log_time());
+    float y = static_cast<float>(scream_update.target_rate.kbps());
+    scream_series.points.emplace_back(x, y);
+  }
+
+  TimeSeries created_series("Probe cluster created.", LineStyle::kNone,
+                            PointStyle::kHighlight);
+  for (auto& cluster : parsed_log_.bwe_probe_cluster_created_events()) {
+    float x = config_.GetCallTimeSec(cluster.log_time());
+    float y = static_cast<float>(cluster.bitrate_bps) / 1000;
+    created_series.points.emplace_back(x, y);
+  }
+
+  TimeSeries result_series("Probing results.", LineStyle::kNone,
+                           PointStyle::kHighlight);
+  for (auto& result : parsed_log_.bwe_probe_success_events()) {
+    float x = config_.GetCallTimeSec(result.log_time());
+    float y = static_cast<float>(result.bitrate_bps) / 1000;
+    result_series.points.emplace_back(x, y);
+  }
+
+  TimeSeries probe_failures_series("Probe failed", LineStyle::kNone,
+                                   PointStyle::kHighlight);
+  for (auto& failure : parsed_log_.bwe_probe_failure_events()) {
+    float x = config_.GetCallTimeSec(failure.log_time());
+    probe_failures_series.points.emplace_back(x, 0);
+  }
+
+  IntervalSeries alr_state("ALR", "#555555", IntervalSeries::kHorizontal);
+  bool previously_in_alr = false;
+  Timestamp alr_start = Timestamp::Zero();
+  for (auto& alr : parsed_log_.alr_state_events()) {
+    float y = config_.GetCallTimeSec(alr.log_time());
+    if (!previously_in_alr && alr.in_alr) {
+      alr_start = alr.log_time();
+      previously_in_alr = true;
+    } else if (previously_in_alr && !alr.in_alr) {
+      float x = config_.GetCallTimeSec(alr_start);
+      alr_state.intervals.emplace_back(x, y);
+      previously_in_alr = false;
+    }
+  }
+
+  if (previously_in_alr) {
+    float x = config_.GetCallTimeSec(alr_start);
+    float y = config_.GetCallTimeSec(config_.end_time_);
+    alr_state.intervals.emplace_back(x, y);
+  }
+
+  if (show_detector_state) {
+    plot->AppendIntervalSeries(std::move(overusing_series));
+    plot->AppendIntervalSeries(std::move(underusing_series));
+    plot->AppendIntervalSeries(std::move(normal_series));
+  }
+
+  if (show_alr_state) {
+    plot->AppendIntervalSeries(std::move(alr_state));
+  }
+
+  if (show_link_capacity) {
+    plot->AppendTimeSeriesIfNotEmpty(std::move(link_capacity_lower_series));
+    plot->AppendTimeSeriesIfNotEmpty(std::move(link_capacity_upper_series));
+  }
+
+  plot->AppendTimeSeries(std::move(loss_series));
+  plot->AppendTimeSeriesIfNotEmpty(std::move(probe_failures_series));
+  plot->AppendTimeSeries(std::move(delay_series));
+  plot->AppendTimeSeriesIfNotEmpty(std::move(scream_series));
+  plot->AppendTimeSeries(std::move(created_series));
+  plot->AppendTimeSeries(std::move(result_series));
+
+  // Overlay the incoming REMB over the outgoing bitrate.
+  TimeSeries remb_series("Remb", LineStyle::kStep);
+  for (const auto& rtcp : parsed_log_.rembs(kIncomingPacket)) {
+    float x = config_.GetCallTimeSec(rtcp.log_time());
+    float y = static_cast<float>(rtcp.remb.bitrate_bps()) / 1000;
+    remb_series.points.emplace_back(x, y);
+  }
+  plot->AppendTimeSeriesIfNotEmpty(std::move(remb_series));
+
+  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
+                 "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin);
+  plot->SetTitle("Outgoing RTP bitrate");
+}
+
+// For each SSRC, plot the bandwidth used by that stream.
+void EventLogAnalyzer::CreateStreamBitrateGraph(PacketDirection direction,
+                                                Plot* plot) const {
+  for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
+    // Filter on SSRC.
+    if (!MatchingSsrc(stream.ssrc, desired_ssrc_)) {
+      continue;
+    }
+
+    TimeSeries time_series(GetStreamName(parsed_log_, direction, stream.ssrc),
+                           LineStyle::kLine);
+    auto GetPacketSizeKilobits = [](const LoggedRtpPacket& packet) {
+      return packet.total_length * 8.0 / 1000.0;
+    };
+    MovingAverage<LoggedRtpPacket, double>(
+        GetPacketSizeKilobits, stream.packet_view, config_, &time_series);
+    plot->AppendTimeSeries(std::move(time_series));
+  }
+
+  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
+                 "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin);
+  plot->SetTitle(GetDirectionAsString(direction) + " bitrate per stream");
+}
+
+// Plot the bitrate allocation for each temporal and spatial layer.
+// Computed from RTCP XR target bitrate block, so the graph is only populated if
+// those are sent.
+void EventLogAnalyzer::CreateBitrateAllocationGraph(PacketDirection direction,
+                                                    Plot* plot) const {
+  std::map<LayerDescription, TimeSeries> time_series;
+  const auto& xr_list = parsed_log_.extended_reports(direction);
+  for (const auto& rtcp : xr_list) {
+    const std::optional<rtcp::TargetBitrate>& target_bitrate =
+        rtcp.xr.target_bitrate();
+    if (!target_bitrate.has_value())
+      continue;
+    for (const auto& bitrate_item : target_bitrate->GetTargetBitrates()) {
+      LayerDescription layer(rtcp.xr.sender_ssrc(), bitrate_item.spatial_layer,
+                             bitrate_item.temporal_layer);
+      auto time_series_it = time_series.find(layer);
+      if (time_series_it == time_series.end()) {
+        std::string layer_name = GetLayerName(layer);
+        bool inserted;
+        std::tie(time_series_it, inserted) = time_series.insert(
+            std::make_pair(layer, TimeSeries(layer_name, LineStyle::kStep)));
+        RTC_DCHECK(inserted);
+      }
+      float x = config_.GetCallTimeSec(rtcp.log_time());
+      float y = bitrate_item.target_bitrate_kbps;
+      time_series_it->second.points.emplace_back(x, y);
+    }
+  }
+  for (auto& layer : time_series) {
+    plot->AppendTimeSeries(std::move(layer.second));
+  }
+  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
+                 "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin);
+  if (direction == kIncomingPacket)
+    plot->SetTitle("Target bitrate per incoming layer");
+  else
+    plot->SetTitle("Target bitrate per outgoing layer");
 }
 
 void EventLogAnalyzer::CreateGoogCcSimulationGraph(Plot* plot) const {
-  webrtc::CreateGoogCcSimulationGraph(parsed_log_, config_, plot);
+  TimeSeries target_rates("Simulated target rate", LineStyle::kStep,
+                          PointStyle::kHighlight);
+  TimeSeries delay_based("Logged delay-based estimate", LineStyle::kStep,
+                         PointStyle::kHighlight);
+  TimeSeries loss_based("Logged loss-based estimate", LineStyle::kStep,
+                        PointStyle::kHighlight);
+  TimeSeries probe_results("Logged probe success", LineStyle::kNone,
+                           PointStyle::kHighlight);
+
+  LogBasedNetworkControllerSimulation simulation(
+      env_, std::make_unique<GoogCcNetworkControllerFactory>(),
+      [&](const NetworkControlUpdate& update, Timestamp at_time) {
+        if (update.target_rate) {
+          target_rates.points.emplace_back(
+              config_.GetCallTimeSec(at_time),
+              update.target_rate->target_rate.kbps<float>());
+        }
+      });
+
+  simulation.ProcessEventsInLog(parsed_log_);
+  for (const auto& logged : parsed_log_.bwe_delay_updates())
+    delay_based.points.emplace_back(config_.GetCallTimeSec(logged.log_time()),
+                                    logged.bitrate_bps / 1000);
+  for (const auto& logged : parsed_log_.bwe_probe_success_events())
+    probe_results.points.emplace_back(config_.GetCallTimeSec(logged.log_time()),
+                                      logged.bitrate_bps / 1000);
+  for (const auto& logged : parsed_log_.bwe_loss_updates())
+    loss_based.points.emplace_back(config_.GetCallTimeSec(logged.log_time()),
+                                   logged.bitrate_bps / 1000);
+
+  plot->AppendTimeSeries(std::move(delay_based));
+  plot->AppendTimeSeries(std::move(loss_based));
+  plot->AppendTimeSeries(std::move(probe_results));
+  plot->AppendTimeSeries(std::move(target_rates));
+
+  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
+                 "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 10, "Bitrate (kbps)", kBottomMargin, kTopMargin);
+  plot->SetTitle("Simulated BWE behavior");
 }
 
 void EventLogAnalyzer::CreateScreamSimulationBitrateGraph(Plot* plot) const {
-  webrtc::CreateScreamSimulationBitrateGraph(parsed_log_, config_, plot);
+  TimeSeries target_rate_series("Target rate", LineStyle::kStep);
+  TimeSeries pacing_rate_series("Pacing rate", LineStyle::kStep);
+  TimeSeries send_rate_series("Send rate", LineStyle::kStep);
+
+  LogScreamSimulation simulation({.rate_window = config_.window_duration_},
+                                 env_);
+  simulation.ProcessEventsInLog(parsed_log_);
+
+  for (const LogScreamSimulation::State& state : simulation.updates()) {
+    target_rate_series.points.emplace_back(config_.GetCallTimeSec(state.time),
+                                           state.target_rate.bps() / 1000);
+    pacing_rate_series.points.emplace_back(config_.GetCallTimeSec(state.time),
+                                           state.pacing_rate.bps() / 1000);
+    send_rate_series.points.emplace_back(config_.GetCallTimeSec(state.time),
+                                         state.send_rate.bps() / 1000);
+  }
+  plot->AppendTimeSeries(std::move(target_rate_series));
+  plot->AppendTimeSeries(std::move(pacing_rate_series));
+  plot->AppendTimeSeries(std::move(send_rate_series));
+
+  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
+                 "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 100, "Kbps", kBottomMargin, kTopMargin);
+  plot->SetTitle("Simulated Scream rates");
 }
 
 void EventLogAnalyzer::CreateScreamSimulationRefWindowGraph(Plot* plot) const {
-  webrtc::CreateScreamSimulationRefWindowGraph(parsed_log_, config_, plot);
+  using SendWindowUsage = LogScreamSimulation::State::SendWindowUsage;
+  TimeSeries ref_window_series("RefWindow", LineStyle::kStep);
+  TimeSeries ref_window_i_series("RefWindowI", LineStyle::kStep);
+  TimeSeries max_data_in_flight("Max allowed data in flight", LineStyle::kStep);
+  TimeSeries data_in_flight("Data in flight", LineStyle::kStep);
+  IntervalSeries send_window_above_max_series(
+      "Data in flight > Max allowed", "#ff8e82", IntervalSeries::kHorizontal);
+  IntervalSeries send_window_below_ref_window_series(
+      "Data in flight < RefWindow", "#c5dff2", IntervalSeries::kHorizontal);
+  IntervalSeries send_window_above_ref_window_series(
+      "Data in flight >= RefWindow", "#b9fad8", IntervalSeries::kHorizontal);
+  IntervalSeries* last_series = &send_window_below_ref_window_series;
+
+  LogScreamSimulation simulation({.rate_window = config_.window_duration_},
+                                 env_);
+  simulation.ProcessEventsInLog(parsed_log_);
+  if (simulation.updates().empty()) {
+    RTC_LOG(LS_ERROR) << "Empty simulation.";
+    return;
+  }
+
+  float send_window_state_switch =
+      config_.GetCallTimeSec(simulation.updates().front().time);
+  SendWindowUsage send_window_usage = SendWindowUsage::kBelowRefWindow;
+
+  for (const LogScreamSimulation::State& state : simulation.updates()) {
+    ref_window_series.points.emplace_back(config_.GetCallTimeSec(state.time),
+                                          state.ref_window.bytes());
+    ref_window_i_series.points.emplace_back(config_.GetCallTimeSec(state.time),
+                                            state.ref_window_i.bytes());
+    max_data_in_flight.points.emplace_back(config_.GetCallTimeSec(state.time),
+                                           state.max_data_in_flight.bytes());
+    data_in_flight.points.emplace_back(config_.GetCallTimeSec(state.time),
+                                       state.data_in_flight.bytes());
+    if (state.send_window_usage != send_window_usage) {
+      last_series->intervals.emplace_back(send_window_state_switch,
+                                          config_.GetCallTimeSec(state.time));
+      send_window_usage = state.send_window_usage;
+      send_window_state_switch = config_.GetCallTimeSec(state.time);
+      switch (send_window_usage) {
+        case SendWindowUsage::kAboveRefWindow:
+          last_series = &send_window_above_ref_window_series;
+          break;
+        case SendWindowUsage::kBelowRefWindow:
+          last_series = &send_window_below_ref_window_series;
+          break;
+        case SendWindowUsage::kAboveScreamMax:
+          last_series = &send_window_above_max_series;
+          break;
+      }
+    }
+  }
+  last_series->intervals.emplace_back(send_window_state_switch,
+                                      config_.CallEndTimeSec());
+  plot->AppendTimeSeries(std::move(ref_window_series));
+  plot->AppendTimeSeries(std::move(ref_window_i_series));
+  plot->AppendTimeSeries(std::move(max_data_in_flight));
+  plot->AppendTimeSeries(std::move(data_in_flight));
+  plot->AppendIntervalSeries(std::move(send_window_above_max_series));
+  plot->AppendIntervalSeries(std::move(send_window_below_ref_window_series));
+  plot->AppendIntervalSeries(std::move(send_window_above_ref_window_series));
+
+  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
+                 "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 10, "Bytes", kBottomMargin, kTopMargin);
+  plot->SetTitle("Simulated Scream RefWindow");
 }
 
 void EventLogAnalyzer::CreateScreamSimulationRatiosGraph(Plot* plot) const {
-  webrtc::CreateScreamSimulationRatiosGraph(parsed_log_, config_, plot);
-}
+  TimeSeries queue_delay_dev_norm_series("QueueDelayDevNorm", LineStyle::kStep);
+  TimeSeries l4s_alpha_series("L4sAlpha", LineStyle::kStep);
+  TimeSeries l4s_alpha_v_series("L4sAlphaV", LineStyle::kStep);
+  TimeSeries ref_window_scale_factor_due_to_increased_delay(
+      "RefWindowScaleFactorDueToIncreasedDelay", LineStyle::kStep);
+  TimeSeries ref_window_scale_factor_due_to_delay_variation(
+      "RefWindowScaleFactorDueToDelayVariation", LineStyle::kStep);
 
-void EventLogAnalyzer::CreateScreamRefWindowGraph(Plot* plot) const {
-  webrtc::CreateScreamRefWindowGraph(parsed_log_, config_, plot);
-}
+  LogScreamSimulation simulation({.rate_window = config_.window_duration_},
+                                 env_);
+  simulation.ProcessEventsInLog(parsed_log_);
 
-void EventLogAnalyzer::CreateScreamDelayEstimateGraph(Plot* plot) const {
-  webrtc::CreateScreamDelayEstimateGraph(parsed_log_, config_, plot);
-}
+  for (const LogScreamSimulation::State& state : simulation.updates()) {
+    queue_delay_dev_norm_series.points.emplace_back(
+        config_.GetCallTimeSec(state.time), state.queue_delay_dev_norm);
+    l4s_alpha_series.points.emplace_back(config_.GetCallTimeSec(state.time),
+                                         state.l4s_alpha);
+    l4s_alpha_v_series.points.emplace_back(config_.GetCallTimeSec(state.time),
+                                           state.l4s_alpha_v);
+    ref_window_scale_factor_due_to_increased_delay.points.emplace_back(
+        config_.GetCallTimeSec(state.time),
+        state.ref_window_scale_factor_due_to_increased_delay);
+    ref_window_scale_factor_due_to_delay_variation.points.emplace_back(
+        config_.GetCallTimeSec(state.time),
+        state.ref_window_scale_factor_due_to_delay_variation);
+  }
+  plot->AppendTimeSeries(std::move(queue_delay_dev_norm_series));
+  plot->AppendTimeSeries(std::move(l4s_alpha_series));
+  plot->AppendTimeSeries(std::move(l4s_alpha_v_series));
+  plot->AppendTimeSeries(
+      std::move(ref_window_scale_factor_due_to_increased_delay));
+  plot->AppendTimeSeries(
+      std::move(ref_window_scale_factor_due_to_delay_variation));
 
-void EventLogAnalyzer::CreateSendSideBweSimulationGraph(Plot* plot) const {
-  webrtc::CreateSendSideBweSimulationGraph(parsed_log_, config_, plot);
-}
-
-void EventLogAnalyzer::CreateReceiveSideBweSimulationGraph(Plot* plot) const {
-  webrtc::CreateReceiveSideBweSimulationGraph(parsed_log_, config_, plot);
-}
-
-void EventLogAnalyzer::CreateNetworkDelayFeedbackGraph(Plot* plot) const {
-  webrtc::CreateNetworkDelayFeedbackGraph(parsed_log_, config_, plot);
-}
-
-void EventLogAnalyzer::CreatePacerDelayGraph(Plot* plot) const {
-  webrtc::CreatePacerDelayGraph(parsed_log_, config_, plot);
-}
-
-void EventLogAnalyzer::CreateIceCandidatePairConfigGraph(Plot* plot) const {
-  webrtc::CreateIceCandidatePairConfigGraph(parsed_log_, config_, plot);
-}
-
-void EventLogAnalyzer::CreateIceConnectivityCheckGraph(Plot* plot) const {
-  webrtc::CreateIceConnectivityCheckGraph(parsed_log_, config_, plot);
-}
-
-void EventLogAnalyzer::CreateDtlsTransportStateGraph(Plot* plot) const {
-  webrtc::CreateDtlsTransportStateGraph(parsed_log_, config_, plot);
-}
-
-void EventLogAnalyzer::CreateDtlsWritableStateGraph(Plot* plot) const {
-  webrtc::CreateDtlsWritableStateGraph(parsed_log_, config_, plot);
-}
-
-void EventLogAnalyzer::CreatePacketGraph(PacketDirection direction,
-                                         Plot* plot) const {
-  webrtc::CreatePacketGraph(direction, parsed_log_, config_, plot);
-}
-
-void EventLogAnalyzer::CreateRtcpTypeGraph(PacketDirection direction,
-                                           Plot* plot) const {
-  webrtc::CreateRtcpTypeGraph(direction, parsed_log_, config_, plot);
-}
-
-void EventLogAnalyzer::CreateAccumulatedPacketsGraph(PacketDirection direction,
-                                                     Plot* plot) const {
-  webrtc::CreateAccumulatedPacketsGraph(direction, parsed_log_, config_, plot);
-}
-
-void EventLogAnalyzer::CreatePacketRateGraph(PacketDirection direction,
-                                             Plot* plot) const {
-  webrtc::CreatePacketRateGraph(direction, parsed_log_, config_, plot);
-}
-
-void EventLogAnalyzer::CreateTotalPacketRateGraph(PacketDirection direction,
-                                                  Plot* plot) const {
-  webrtc::CreateTotalPacketRateGraph(direction, parsed_log_, config_, plot);
-}
-
-void EventLogAnalyzer::CreateSequenceNumberGraph(Plot* plot) const {
-  webrtc::CreateSequenceNumberGraph(parsed_log_, config_, plot);
-}
-
-void EventLogAnalyzer::CreateIncomingPacketLossGraph(Plot* plot) const {
-  webrtc::CreateIncomingPacketLossGraph(parsed_log_, config_, plot);
-}
-
-void EventLogAnalyzer::CreateStreamBitrateGraph(PacketDirection direction,
-                                                Plot* plot) const {
-  webrtc::CreateStreamBitrateGraph(direction, parsed_log_, config_, plot);
-}
-
-void EventLogAnalyzer::CreateBitrateAllocationGraph(PacketDirection direction,
-                                                    Plot* plot) const {
-  webrtc::CreateBitrateAllocationGraph(direction, parsed_log_, config_, plot);
+  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
+                 "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Ratios", kBottomMargin, kTopMargin);
+  plot->SetTitle("Simulated Scream Ratios");
 }
 
 void EventLogAnalyzer::CreateOutgoingEcnFeedbackGraph(Plot* plot) const {
-  webrtc::CreateOutgoingEcnFeedbackGraph(parsed_log_, config_, plot);
+  CreateEcnFeedbackGraph(plot, kOutgoingPacket);
+  plot->SetTitle("Outgoing ECN count per feedback");
 }
 
 void EventLogAnalyzer::CreateIncomingEcnFeedbackGraph(Plot* plot) const {
-  webrtc::CreateIncomingEcnFeedbackGraph(parsed_log_, config_, plot);
+  CreateEcnFeedbackGraph(plot, kIncomingPacket);
+  plot->SetTitle("Incoming ECN count per feedback");
+}
+
+void EventLogAnalyzer::CreateScreamRefWindowGraph(Plot* plot) const {
+  TimeSeries ref_window_series("RefWindow", LineStyle::kStep);
+  for (auto& scream_update : parsed_log_.bwe_scream_updates()) {
+    float x = config_.GetCallTimeSec(scream_update.log_time());
+    float y = static_cast<float>(scream_update.ref_window.bytes());
+    ref_window_series.points.emplace_back(x, y);
+  }
+  plot->AppendTimeSeries(std::move(ref_window_series));
+
+  TimeSeries data_in_flight_series("Data in flight", LineStyle::kLine);
+  for (auto& scream_update : parsed_log_.bwe_scream_updates()) {
+    float x = config_.GetCallTimeSec(scream_update.log_time());
+    float y = static_cast<float>(scream_update.data_in_flight.bytes());
+    data_in_flight_series.points.emplace_back(x, y);
+  }
+  plot->AppendTimeSeries(std::move(data_in_flight_series));
+
+  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
+                 "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 3000, "Bytes", kBottomMargin, kTopMargin);
+  plot->SetTitle("Scream Ref Window");
+}
+
+void EventLogAnalyzer::CreateScreamDelayEstimateGraph(Plot* plot) const {
+  TimeSeries smoothed_rtt_series("Smoothed RTT", LineStyle::kStep);
+  TimeSeries avg_queue_delay_series("Avg queue delay", LineStyle::kStep);
+
+  for (auto& scream_update : parsed_log_.bwe_scream_updates()) {
+    float x = config_.GetCallTimeSec(scream_update.log_time());
+    float smoothed_rtt_ms = static_cast<float>(scream_update.smoothed_rtt.ms());
+    smoothed_rtt_series.points.emplace_back(x, smoothed_rtt_ms);
+    float avg_queue_delay_ms =
+        static_cast<float>(scream_update.avg_queue_delay.ms());
+    avg_queue_delay_series.points.emplace_back(x, avg_queue_delay_ms);
+  }
+
+  plot->AppendTimeSeries(std::move(smoothed_rtt_series));
+  plot->AppendTimeSeries(std::move(avg_queue_delay_series));
+
+  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
+                 "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 50, "Delay (ms)", kBottomMargin, kTopMargin);
+  plot->SetTitle("Scream delay estimates");
+}
+
+void EventLogAnalyzer::CreateEcnFeedbackGraph(Plot* plot,
+                                              PacketDirection direction) const {
+  TimeSeries not_ect("Not ECN capable", LineStyle::kBar,
+                     PointStyle::kHighlight);
+  TimeSeries ect_1("ECN capable", LineStyle::kBar, PointStyle::kHighlight);
+  TimeSeries ce("Congestion experienced", LineStyle::kBar,
+                PointStyle::kHighlight);
+
+  for (const LoggedRtcpCongestionControlFeedback& feedback :
+       parsed_log_.congestion_feedback(direction)) {
+    int ect_1_count = 0;
+    int not_ect_count = 0;
+    int ce_count = 0;
+
+    for (const rtcp::CongestionControlFeedback::PacketInfo& info :
+         feedback.congestion_feedback.packets()) {
+      switch (info.ecn) {
+        case EcnMarking::kNotEct:
+          ++not_ect_count;
+          break;
+        case EcnMarking::kEct1:
+          ++ect_1_count;
+          break;
+        case EcnMarking::kEct0:
+          RTC_LOG(LS_ERROR) << "unexpected ect(0)";
+          break;
+        case EcnMarking::kCe:
+          ++ce_count;
+          break;
+      }
+    }
+    ect_1.points.emplace_back(config_.GetCallTimeSec(feedback.timestamp),
+                              ect_1_count);
+    not_ect.points.emplace_back(config_.GetCallTimeSec(feedback.timestamp),
+                                not_ect_count);
+    ce.points.emplace_back(config_.GetCallTimeSec(feedback.timestamp),
+                           ce_count);
+  }
+
+  plot->AppendTimeSeriesIfNotEmpty(std::move(ect_1));
+  plot->AppendTimeSeriesIfNotEmpty(std::move(not_ect));
+  plot->AppendTimeSeriesIfNotEmpty(std::move(ce));
+
+  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
+                 "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 10, "Count per feedback", kBottomMargin,
+                          kTopMargin);
 }
 
 void EventLogAnalyzer::CreateOutgoingLossRateGraph(Plot* plot) const {
-  webrtc::CreateOutgoingLossRateGraph(parsed_log_, config_, plot);
+  struct PacketLossPerFeedback {
+    Timestamp timestamp;              // Time when this feedback was received.
+    int num_packets_in_feedback = 0;  // Includes lost packets.
+    int num_lost_packets = 0;         // In this specific feedback.
+    int num_reordered_packets = 0;    // Packets received in this feedback, but
+                                      // was previously reported as lost.
+    int num_missing_feedback =
+        0;  // Packets missing feedback between this report and the previous.
+  };
+
+  class LossFeedbackBuilder {
+   public:
+    void AddPacket(uint16_t sequence_number, TimeDelta arrival_time_delta) {
+      last_unwrapped_sequence_number_ =
+          sequence_number_unwrapper_.Unwrap(sequence_number);
+      if (!first_sequence_number_.has_value()) {
+        first_sequence_number_ = last_unwrapped_sequence_number_;
+      }
+      ++num_packets_;
+      if (arrival_time_delta.IsInfinite()) {
+        lost_sequence_numbers_.insert(last_unwrapped_sequence_number_);
+      } else {
+        num_reordered_packets_ += previous_lost_sequence_numbers_.count(
+            last_unwrapped_sequence_number_);
+      }
+    }
+
+    void Update(PacketLossPerFeedback& feedback) {
+      feedback.num_packets_in_feedback += num_packets_;
+      feedback.num_lost_packets += lost_sequence_numbers_.size();
+      feedback.num_reordered_packets += num_reordered_packets_;
+      if (first_sequence_number_.has_value() &&
+          previous_feedback_highest_seq_number_.has_value()) {
+        feedback.num_missing_feedback +=
+            *first_sequence_number_ - *previous_feedback_highest_seq_number_ -
+            1;
+      }
+
+      // Prepare for next feedback.
+      first_sequence_number_ = std::nullopt;
+      previous_lost_sequence_numbers_.insert(lost_sequence_numbers_.begin(),
+                                             lost_sequence_numbers_.end());
+      previous_feedback_highest_seq_number_ = last_unwrapped_sequence_number_;
+      lost_sequence_numbers_.clear();
+      num_reordered_packets_ = 0;
+      num_packets_ = 0;
+    }
+
+   private:
+    int64_t last_unwrapped_sequence_number_ = 0;
+    int num_reordered_packets_ = 0;
+    int num_packets_ = 0;
+    std::optional<int64_t> first_sequence_number_;
+
+    std::unordered_set<int64_t> lost_sequence_numbers_;
+    std::unordered_set<int64_t> previous_lost_sequence_numbers_;
+    std::optional<int64_t> previous_feedback_highest_seq_number_;
+
+    RtpSequenceNumberUnwrapper sequence_number_unwrapper_;
+  };
+
+  TimeSeries loss_rate_series("Loss rate (from packet feedback)",
+                              LineStyle::kLine, PointStyle::kHighlight);
+  TimeSeries reordered_packets_between_feedback(
+      "Ratio of reordered packets from last feedback", LineStyle::kLine,
+      PointStyle::kHighlight);
+  TimeSeries average_loss_rate_series("Average loss rate last 5s",
+                                      LineStyle::kLine, PointStyle::kHighlight);
+  TimeSeries missing_feedback_series("Missing feedback", LineStyle::kNone,
+                                     PointStyle::kHighlight);
+
+  std::vector<PacketLossPerFeedback> loss_per_feedback;
+
+  if (!parsed_log_.congestion_feedback(kIncomingPacket).empty()) {
+    plot->SetTitle("Outgoing loss rate (from CCFB)");
+
+    std::map</*ssrc*/ uint32_t, LossFeedbackBuilder> per_ssrc_builder;
+    for (const LoggedRtcpCongestionControlFeedback& feedback :
+         parsed_log_.congestion_feedback(kIncomingPacket)) {
+      const rtcp::CongestionControlFeedback& transport_feedback =
+          feedback.congestion_feedback;
+
+      PacketLossPerFeedback packet_loss_per_feedback = {
+          .timestamp = feedback.log_time()};
+      for (const rtcp::CongestionControlFeedback::PacketInfo& packet :
+           transport_feedback.packets()) {
+        per_ssrc_builder[packet.ssrc].AddPacket(packet.sequence_number,
+                                                packet.arrival_time_offset);
+      }
+      for (auto& [ssrc, builder] : per_ssrc_builder) {
+        builder.Update(packet_loss_per_feedback);
+      }
+      loss_per_feedback.push_back(packet_loss_per_feedback);
+    }
+  } else if (!parsed_log_.transport_feedbacks(kIncomingPacket).empty()) {
+    plot->SetTitle("Outgoing loss rate (from TWCC)");
+
+    LossFeedbackBuilder builder;
+    for (const LoggedRtcpPacketTransportFeedback& feedback :
+         parsed_log_.transport_feedbacks(kIncomingPacket)) {
+      feedback.transport_feedback.ForAllPackets(
+          [&](uint16_t sequence_number, TimeDelta receive_time_delta) {
+            builder.AddPacket(sequence_number, receive_time_delta);
+          });
+      PacketLossPerFeedback packet_loss_per_feedback = {
+          .timestamp = feedback.log_time()};
+      builder.Update(packet_loss_per_feedback);
+      loss_per_feedback.push_back(packet_loss_per_feedback);
+    }
+  }
+
+  PacketLossSummary window_summary;
+  Timestamp last_observation_receive_time = Timestamp::Zero();
+
+  // Use loss based bwe 2 observation duration and observation window size.
+  constexpr TimeDelta kObservationDuration = TimeDelta::Millis(250);
+  constexpr uint32_t kObservationWindowSize = 20;
+  std::deque<PacketLossSummary> observations;
+  int previous_feedback_size = 0;
+  for (const PacketLossPerFeedback& feedback : loss_per_feedback) {
+    for (int64_t num = 0; num < feedback.num_missing_feedback; ++num) {
+      missing_feedback_series.points.emplace_back(
+          config_.GetCallTimeSec(feedback.timestamp), 100 + num);
+    }
+
+    // Compute loss rate from the transport feedback.
+    float loss_rate = static_cast<float>(feedback.num_lost_packets * 100.0 /
+                                         feedback.num_packets_in_feedback);
+
+    loss_rate_series.points.emplace_back(
+        config_.GetCallTimeSec(feedback.timestamp), loss_rate);
+    float reordered_rate =
+        previous_feedback_size == 0
+            ? 0
+            : static_cast<float>(feedback.num_reordered_packets * 100.0 /
+                                 previous_feedback_size);
+    previous_feedback_size = feedback.num_packets_in_feedback;
+    reordered_packets_between_feedback.points.emplace_back(
+        config_.GetCallTimeSec(feedback.timestamp), reordered_rate);
+
+    // Compute loss rate in a window of kObservationWindowSize.
+    if (window_summary.num_packets == 0) {
+      window_summary.base_time = feedback.timestamp;
+    }
+    window_summary.num_packets += feedback.num_packets_in_feedback;
+    window_summary.num_lost_packets +=
+        feedback.num_lost_packets - feedback.num_reordered_packets;
+
+    const Timestamp last_received_time = feedback.timestamp;
+    const TimeDelta observation_duration =
+        window_summary.base_time == Timestamp::Zero()
+            ? TimeDelta::Zero()
+            : last_received_time - window_summary.base_time;
+    if (observation_duration > kObservationDuration) {
+      last_observation_receive_time = last_received_time;
+      observations.push_back(window_summary);
+      if (observations.size() > kObservationWindowSize) {
+        observations.pop_front();
+      }
+
+      // Compute average loss rate in a number of windows.
+      int total_packets = 0;
+      int total_loss = 0;
+      for (const auto& observation : observations) {
+        total_loss += observation.num_lost_packets;
+        total_packets += observation.num_packets;
+      }
+      if (total_packets > 0) {
+        float average_loss_rate = total_loss * 100.0 / total_packets;
+        average_loss_rate_series.points.emplace_back(
+            config_.GetCallTimeSec(feedback.timestamp), average_loss_rate);
+      } else {
+        average_loss_rate_series.points.emplace_back(
+            config_.GetCallTimeSec(feedback.timestamp), 0);
+      }
+      window_summary = PacketLossSummary();
+    }
+  }
+  // Add the data set to the plot.
+  plot->AppendTimeSeriesIfNotEmpty(std::move(loss_rate_series));
+  plot->AppendTimeSeriesIfNotEmpty(
+      std::move(reordered_packets_between_feedback));
+  plot->AppendTimeSeriesIfNotEmpty(std::move(average_loss_rate_series));
+  plot->AppendTimeSeriesIfNotEmpty(std::move(missing_feedback_series));
+
+  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
+                 "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 100, "Loss rate (percent)", kBottomMargin,
+                          kTopMargin);
+}
+
+void EventLogAnalyzer::CreateSendSideBweSimulationGraph(Plot* plot) const {
+  using RtpPacketType = LoggedRtpPacketOutgoing;
+  using TransportFeedbackType = LoggedRtcpPacketTransportFeedback;
+
+  // TODO(terelius): This could be provided by the parser.
+  std::multimap<int64_t, const RtpPacketType*> outgoing_rtp;
+  for (const auto& stream : parsed_log_.outgoing_rtp_packets_by_ssrc()) {
+    for (const RtpPacketType& rtp_packet : stream.outgoing_packets)
+      outgoing_rtp.insert(
+          std::make_pair(rtp_packet.rtp.log_time_us(), &rtp_packet));
+  }
+
+  const std::vector<TransportFeedbackType>& incoming_rtcp =
+      parsed_log_.transport_feedbacks(kIncomingPacket);
+
+  SimulatedClock clock(0);
+  BitrateObserver observer;
+  TransportFeedbackAdapter transport_feedback;
+  auto factory = GoogCcNetworkControllerFactory();
+  TimeDelta process_interval = factory.GetProcessInterval();
+  // TODO(holmer): Log the call config and use that here instead.
+  static const uint32_t kDefaultStartBitrateBps = 300000;
+  NetworkControllerConfig cc_config(env_);
+  cc_config.constraints.at_time = clock.CurrentTime();
+  cc_config.constraints.starting_rate =
+      DataRate::BitsPerSec(kDefaultStartBitrateBps);
+  auto goog_cc = factory.Create(cc_config);
+
+  TimeSeries time_series("Delay-based estimate", LineStyle::kStep,
+                         PointStyle::kHighlight);
+  TimeSeries acked_time_series("Raw acked bitrate", LineStyle::kLine,
+                               PointStyle::kHighlight);
+  TimeSeries robust_time_series("Robust throughput estimate", LineStyle::kLine,
+                                PointStyle::kHighlight);
+  TimeSeries acked_estimate_time_series("Ackednowledged bitrate estimate",
+                                        LineStyle::kLine,
+                                        PointStyle::kHighlight);
+
+  auto rtp_iterator = outgoing_rtp.begin();
+  auto rtcp_iterator = incoming_rtcp.begin();
+
+  auto NextRtpTime = [&]() {
+    if (rtp_iterator != outgoing_rtp.end())
+      return static_cast<int64_t>(rtp_iterator->first);
+    return std::numeric_limits<int64_t>::max();
+  };
+
+  auto NextRtcpTime = [&]() {
+    if (rtcp_iterator != incoming_rtcp.end())
+      return static_cast<int64_t>(rtcp_iterator->log_time_us());
+    return std::numeric_limits<int64_t>::max();
+  };
+  int64_t next_process_time_us_ = std::min({NextRtpTime(), NextRtcpTime()});
+
+  auto NextProcessTime = [&]() {
+    if (rtcp_iterator != incoming_rtcp.end() ||
+        rtp_iterator != outgoing_rtp.end()) {
+      return next_process_time_us_;
+    }
+    return std::numeric_limits<int64_t>::max();
+  };
+
+  RateStatistics raw_acked_bitrate(750, 8000);
+  FieldTrials throughput_config(
+      "WebRTC-Bwe-RobustThroughputEstimatorSettings/enabled:true/");
+  std::unique_ptr<AcknowledgedBitrateEstimatorInterface>
+      robust_throughput_estimator(
+          AcknowledgedBitrateEstimatorInterface::Create(&throughput_config));
+  FieldTrials acked_bitrate_config(
+      "WebRTC-Bwe-RobustThroughputEstimatorSettings/enabled:false/");
+  std::unique_ptr<AcknowledgedBitrateEstimatorInterface>
+      acknowledged_bitrate_estimator(
+          AcknowledgedBitrateEstimatorInterface::Create(&acked_bitrate_config));
+  int64_t time_us =
+      std::min({NextRtpTime(), NextRtcpTime(), NextProcessTime()});
+  int64_t last_update_us = 0;
+  while (time_us != std::numeric_limits<int64_t>::max()) {
+    clock.AdvanceTimeMicroseconds(time_us - clock.TimeInMicroseconds());
+    if (clock.TimeInMicroseconds() >= NextRtpTime()) {
+      RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtpTime());
+      const RtpPacketType& rtp_packet = *rtp_iterator->second;
+      if (rtp_packet.rtp.header.extension.hasTransportSequenceNumber) {
+        RtpPacketToSend send_packet(/*extensions=*/nullptr);
+        send_packet.set_transport_sequence_number(
+            rtp_packet.rtp.header.extension.transportSequenceNumber);
+        send_packet.SetSsrc(rtp_packet.rtp.header.ssrc);
+        send_packet.SetSequenceNumber(rtp_packet.rtp.header.sequenceNumber);
+        send_packet.SetPayloadSize(rtp_packet.rtp.total_length -
+                                   send_packet.headers_size());
+        RTC_DCHECK_EQ(send_packet.size(), rtp_packet.rtp.total_length);
+        if (IsRtxSsrc(parsed_log_, PacketDirection::kOutgoingPacket,
+                      rtp_packet.rtp.header.ssrc)) {
+          // Don't set the optional media type as we don't know if it is
+          // a retransmission, FEC or padding.
+        } else if (IsVideoSsrc(parsed_log_, PacketDirection::kOutgoingPacket,
+                               rtp_packet.rtp.header.ssrc)) {
+          send_packet.set_packet_type(RtpPacketMediaType::kVideo);
+        } else if (IsAudioSsrc(parsed_log_, PacketDirection::kOutgoingPacket,
+                               rtp_packet.rtp.header.ssrc)) {
+          send_packet.set_packet_type(RtpPacketMediaType::kAudio);
+        }
+        transport_feedback.AddPacket(
+            send_packet, PacedPacketInfo(),
+            0u,  // Per packet overhead bytes.,
+            Timestamp::Micros(rtp_packet.rtp.log_time_us()));
+      }
+      SentPacketInfo sent_packet;
+      sent_packet.send_time_ms = rtp_packet.rtp.log_time_ms();
+      sent_packet.info.included_in_allocation = true;
+      sent_packet.info.packet_size_bytes = rtp_packet.rtp.total_length;
+      if (rtp_packet.rtp.header.extension.hasTransportSequenceNumber) {
+        sent_packet.packet_id =
+            rtp_packet.rtp.header.extension.transportSequenceNumber;
+        sent_packet.info.included_in_feedback = true;
+      }
+      auto sent_msg = transport_feedback.ProcessSentPacket(sent_packet);
+      if (sent_msg)
+        observer.Update(goog_cc->OnSentPacket(*sent_msg));
+      ++rtp_iterator;
+    }
+    if (clock.TimeInMicroseconds() >= NextRtcpTime()) {
+      RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtcpTime());
+
+      auto feedback_msg = transport_feedback.ProcessTransportFeedback(
+          rtcp_iterator->transport_feedback, clock.CurrentTime());
+      if (feedback_msg) {
+        observer.Update(goog_cc->OnTransportPacketsFeedback(*feedback_msg));
+        std::vector<PacketResult> feedback =
+            feedback_msg->SortedByReceiveTime();
+        if (!feedback.empty()) {
+          acknowledged_bitrate_estimator->IncomingPacketFeedbackVector(
+              feedback);
+          robust_throughput_estimator->IncomingPacketFeedbackVector(feedback);
+          for (const PacketResult& packet : feedback) {
+            raw_acked_bitrate.Update(packet.sent_packet.size.bytes(),
+                                     packet.receive_time.ms());
+          }
+          std::optional<uint32_t> raw_bitrate_bps =
+              raw_acked_bitrate.Rate(feedback.back().receive_time.ms());
+          float x = config_.GetCallTimeSec(clock.CurrentTime());
+          if (raw_bitrate_bps) {
+            float y = raw_bitrate_bps.value() / 1000;
+            acked_time_series.points.emplace_back(x, y);
+          }
+          std::optional<DataRate> robust_estimate =
+              robust_throughput_estimator->bitrate();
+          if (robust_estimate) {
+            float y = robust_estimate.value().kbps();
+            robust_time_series.points.emplace_back(x, y);
+          }
+          std::optional<DataRate> acked_estimate =
+              acknowledged_bitrate_estimator->bitrate();
+          if (acked_estimate) {
+            float y = acked_estimate.value().kbps();
+            acked_estimate_time_series.points.emplace_back(x, y);
+          }
+        }
+      }
+      ++rtcp_iterator;
+    }
+    if (clock.TimeInMicroseconds() >= NextProcessTime()) {
+      RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextProcessTime());
+      ProcessInterval msg;
+      msg.at_time = clock.CurrentTime();
+      observer.Update(goog_cc->OnProcessInterval(msg));
+      next_process_time_us_ += process_interval.us();
+    }
+    if (observer.GetAndResetBitrateUpdated() ||
+        time_us - last_update_us >= 1e6) {
+      uint32_t y = observer.last_bitrate_bps() / 1000;
+      float x = config_.GetCallTimeSec(clock.CurrentTime());
+      time_series.points.emplace_back(x, y);
+      last_update_us = time_us;
+    }
+    time_us = std::min({NextRtpTime(), NextRtcpTime(), NextProcessTime()});
+  }
+  // Add the data set to the plot.
+  plot->AppendTimeSeries(std::move(time_series));
+  plot->AppendTimeSeries(std::move(robust_time_series));
+  plot->AppendTimeSeries(std::move(acked_time_series));
+  plot->AppendTimeSeriesIfNotEmpty(std::move(acked_estimate_time_series));
+
+  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
+                 "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 10, "Bitrate (kbps)", kBottomMargin, kTopMargin);
+  plot->SetTitle("Simulated send-side BWE behavior");
+}
+
+void EventLogAnalyzer::CreateReceiveSideBweSimulationGraph(Plot* plot) const {
+  using RtpPacketType = LoggedRtpPacketIncoming;
+  class RembInterceptor {
+   public:
+    void SendRemb(uint32_t bitrate_bps, std::vector<uint32_t> ssrcs) {
+      last_bitrate_bps_ = bitrate_bps;
+      bitrate_updated_ = true;
+    }
+    uint32_t last_bitrate_bps() const { return last_bitrate_bps_; }
+    bool GetAndResetBitrateUpdated() {
+      bool bitrate_updated = bitrate_updated_;
+      bitrate_updated_ = false;
+      return bitrate_updated;
+    }
+
+   private:
+    // We don't know the start bitrate, but assume that it is the default 300
+    // kbps.
+    uint32_t last_bitrate_bps_ = 300000;
+    bool bitrate_updated_ = false;
+  };
+
+  std::multimap<int64_t, const RtpPacketType*> incoming_rtp;
+
+  for (const auto& stream : parsed_log_.incoming_rtp_packets_by_ssrc()) {
+    if (IsVideoSsrc(parsed_log_, kIncomingPacket, stream.ssrc)) {
+      for (const auto& rtp_packet : stream.incoming_packets)
+        incoming_rtp.insert(
+            std::make_pair(rtp_packet.rtp.log_time_us(), &rtp_packet));
+    }
+  }
+
+  SimulatedClock clock(0);
+  EnvironmentFactory env_factory(env_);
+  env_factory.Set(&clock);
+  RembInterceptor remb_interceptor;
+  ReceiveSideCongestionController rscc(
+      env_factory.Create(), [](auto...) {},
+      absl::bind_front(&RembInterceptor::SendRemb, &remb_interceptor));
+  // TODO(holmer): Log the call config and use that here instead.
+  // static const uint32_t kDefaultStartBitrateBps = 300000;
+  // rscc.SetBweBitrates(0, kDefaultStartBitrateBps, -1);
+
+  TimeSeries time_series("Receive side estimate", LineStyle::kLine,
+                         PointStyle::kHighlight);
+  TimeSeries acked_time_series("Received bitrate", LineStyle::kLine);
+
+  RateStatistics acked_bitrate(250, 8000);
+  int64_t last_update_us = 0;
+  for (const auto& kv : incoming_rtp) {
+    const RtpPacketType& packet = *kv.second;
+
+    RtpPacketReceived rtp_packet = RtpPacketForBWEFromHeader(packet.rtp.header);
+    rtp_packet.set_arrival_time(packet.rtp.log_time());
+    rtp_packet.SetPayloadSize(packet.rtp.total_length -
+                              rtp_packet.headers_size());
+
+    clock.AdvanceTime(rtp_packet.arrival_time() - clock.CurrentTime());
+    rscc.OnReceivedPacket(rtp_packet, MediaType::VIDEO);
+    int64_t arrival_time_ms = packet.rtp.log_time().ms();
+    acked_bitrate.Update(packet.rtp.total_length, arrival_time_ms);
+    std::optional<uint32_t> bitrate_bps = acked_bitrate.Rate(arrival_time_ms);
+    if (bitrate_bps) {
+      uint32_t y = *bitrate_bps / 1000;
+      float x = config_.GetCallTimeSec(clock.CurrentTime());
+      acked_time_series.points.emplace_back(x, y);
+    }
+    if (remb_interceptor.GetAndResetBitrateUpdated() ||
+        clock.TimeInMicroseconds() - last_update_us >= 1e6) {
+      uint32_t y = remb_interceptor.last_bitrate_bps() / 1000;
+      float x = config_.GetCallTimeSec(clock.CurrentTime());
+      time_series.points.emplace_back(x, y);
+      last_update_us = clock.TimeInMicroseconds();
+    }
+  }
+  // Add the data set to the plot.
+  plot->AppendTimeSeries(std::move(time_series));
+  plot->AppendTimeSeries(std::move(acked_time_series));
+
+  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
+                 "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 10, "Bitrate (kbps)", kBottomMargin, kTopMargin);
+  plot->SetTitle("Simulated receive-side BWE behavior");
+}
+
+void EventLogAnalyzer::CreateNetworkDelayFeedbackGraph(Plot* plot) const {
+  TimeSeries time_series("Network delay", LineStyle::kLine,
+                         PointStyle::kHighlight);
+  int64_t min_send_receive_diff_ms = std::numeric_limits<int64_t>::max();
+  int64_t min_rtt_ms = std::numeric_limits<int64_t>::max();
+
+  std::vector<MatchedSendArrivalTimes> matched_rtp_rtcp =
+      GetNetworkTrace(parsed_log_);
+  absl::c_stable_sort(matched_rtp_rtcp, [](const MatchedSendArrivalTimes& a,
+                                           const MatchedSendArrivalTimes& b) {
+    return a.feedback_arrival_time_ms < b.feedback_arrival_time_ms ||
+           (a.feedback_arrival_time_ms == b.feedback_arrival_time_ms &&
+            a.arrival_time_ms < b.arrival_time_ms);
+  });
+  for (const auto& packet : matched_rtp_rtcp) {
+    if (packet.arrival_time_ms == MatchedSendArrivalTimes::kNotReceived)
+      continue;
+    float x = config_.GetCallTimeSecFromMs(packet.feedback_arrival_time_ms);
+    int64_t y = packet.arrival_time_ms - packet.send_time_ms;
+    int64_t rtt_ms = packet.feedback_arrival_time_ms - packet.send_time_ms;
+    min_rtt_ms = std::min(rtt_ms, min_rtt_ms);
+    min_send_receive_diff_ms = std::min(y, min_send_receive_diff_ms);
+    time_series.points.emplace_back(x, y);
+  }
+
+  // We assume that the base network delay (w/o queues) is equal to half
+  // the minimum RTT. Therefore rescale the delays by subtracting the minimum
+  // observed 1-ways delay and add half the minimum RTT.
+  const int64_t estimated_clock_offset_ms =
+      min_send_receive_diff_ms - min_rtt_ms / 2;
+  for (TimeSeriesPoint& point : time_series.points)
+    point.y -= estimated_clock_offset_ms;
+
+  // Add the data set to the plot.
+  plot->AppendTimeSeriesIfNotEmpty(std::move(time_series));
+
+  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
+                 "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 10, "Delay (ms)", kBottomMargin, kTopMargin);
+  plot->SetTitle("Outgoing network delay (based on per-packet feedback)");
+}
+
+void EventLogAnalyzer::CreatePacerDelayGraph(Plot* plot) const {
+  for (const auto& stream : parsed_log_.outgoing_rtp_packets_by_ssrc()) {
+    const std::vector<LoggedRtpPacketOutgoing>& packets =
+        stream.outgoing_packets;
+
+    if (IsRtxSsrc(parsed_log_, kOutgoingPacket, stream.ssrc)) {
+      continue;
+    }
+
+    if (packets.size() < 2) {
+      RTC_LOG(LS_WARNING)
+          << "Can't estimate a the RTP clock frequency or the "
+             "pacer delay with less than 2 packets in the stream";
+      continue;
+    }
+    int64_t segment_end_us = parsed_log_.first_log_segment().stop_time_us();
+    std::optional<uint32_t> estimated_frequency =
+        EstimateRtpClockFrequency(packets, segment_end_us);
+    if (!estimated_frequency)
+      continue;
+    if (IsVideoSsrc(parsed_log_, kOutgoingPacket, stream.ssrc) &&
+        *estimated_frequency != 90000) {
+      RTC_LOG(LS_WARNING)
+          << "Video stream should use a 90 kHz clock but appears to use "
+          << *estimated_frequency / 1000 << ". Discarding.";
+      continue;
+    }
+
+    TimeSeries pacer_delay_series(
+        GetStreamName(parsed_log_, kOutgoingPacket, stream.ssrc) + "(" +
+            std::to_string(*estimated_frequency / 1000) + " kHz)",
+        LineStyle::kLine, PointStyle::kHighlight);
+    SeqNumUnwrapper<uint32_t> timestamp_unwrapper;
+    uint64_t first_capture_timestamp =
+        timestamp_unwrapper.Unwrap(packets.front().rtp.header.timestamp);
+    uint64_t first_send_timestamp = packets.front().rtp.log_time_us();
+    for (const auto& packet : packets) {
+      double capture_time_ms = (static_cast<double>(timestamp_unwrapper.Unwrap(
+                                    packet.rtp.header.timestamp)) -
+                                first_capture_timestamp) /
+                               *estimated_frequency * 1000;
+      double send_time_ms =
+          static_cast<double>(packet.rtp.log_time_us() - first_send_timestamp) /
+          1000;
+      float x = config_.GetCallTimeSec(packet.rtp.log_time());
+      float y = send_time_ms - capture_time_ms;
+      pacer_delay_series.points.emplace_back(x, y);
+    }
+    plot->AppendTimeSeries(std::move(pacer_delay_series));
+  }
+
+  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
+                 "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 10, "Pacer delay (ms)", kBottomMargin, kTopMargin);
+  plot->SetTitle(
+      "Delay from capture to send time. (First packet normalized to 0.)");
 }
 
 void EventLogAnalyzer::CreateTimestampGraph(PacketDirection direction,
                                             Plot* plot) const {
-  webrtc::CreateTimestampGraph(direction, parsed_log_, config_, plot);
+  for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
+    TimeSeries rtp_timestamps(
+        GetStreamName(parsed_log_, direction, stream.ssrc) + " capture-time",
+        LineStyle::kLine, PointStyle::kHighlight);
+    for (const auto& packet : stream.packet_view) {
+      float x = config_.GetCallTimeSec(packet.log_time());
+      float y = packet.header.timestamp;
+      rtp_timestamps.points.emplace_back(x, y);
+    }
+    plot->AppendTimeSeries(std::move(rtp_timestamps));
+
+    TimeSeries rtcp_timestamps(
+        GetStreamName(parsed_log_, direction, stream.ssrc) +
+            " rtcp capture-time",
+        LineStyle::kLine, PointStyle::kHighlight);
+    // TODO(terelius): Why only sender reports?
+    const auto& sender_reports = parsed_log_.sender_reports(direction);
+    for (const auto& rtcp : sender_reports) {
+      if (rtcp.sr.sender_ssrc() != stream.ssrc)
+        continue;
+      float x = config_.GetCallTimeSec(rtcp.log_time());
+      float y = rtcp.sr.rtp_timestamp();
+      rtcp_timestamps.points.emplace_back(x, y);
+    }
+    plot->AppendTimeSeriesIfNotEmpty(std::move(rtcp_timestamps));
+  }
+
+  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
+                 "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "RTP timestamp", kBottomMargin, kTopMargin);
+  plot->SetTitle(GetDirectionAsString(direction) + " timestamps");
 }
 
 void EventLogAnalyzer::CreateSenderAndReceiverReportPlot(
@@ -439,8 +2403,174 @@ void EventLogAnalyzer::CreateSenderAndReceiverReportPlot(
     std::string title,
     std::string yaxis_label,
     Plot* plot) const {
-  webrtc::CreateSenderAndReceiverReportPlot(direction, fy, title, yaxis_label,
-                                            parsed_log_, config_, plot);
+  std::map<uint32_t, TimeSeries> sr_reports_by_ssrc;
+  const auto& sender_reports = parsed_log_.sender_reports(direction);
+  for (const auto& rtcp : sender_reports) {
+    float x = config_.GetCallTimeSec(rtcp.log_time());
+    uint32_t ssrc = rtcp.sr.sender_ssrc();
+    for (const auto& block : rtcp.sr.report_blocks()) {
+      float y = fy(block);
+      auto sr_report_it = sr_reports_by_ssrc.find(ssrc);
+      bool inserted;
+      if (sr_report_it == sr_reports_by_ssrc.end()) {
+        std::tie(sr_report_it, inserted) = sr_reports_by_ssrc.emplace(
+            ssrc, TimeSeries(GetStreamName(parsed_log_, direction, ssrc) +
+                                 " Sender Reports",
+                             LineStyle::kLine, PointStyle::kHighlight));
+      }
+      sr_report_it->second.points.emplace_back(x, y);
+    }
+  }
+  for (auto& kv : sr_reports_by_ssrc) {
+    plot->AppendTimeSeries(std::move(kv.second));
+  }
+
+  std::map<uint32_t, TimeSeries> rr_reports_by_ssrc;
+  const auto& receiver_reports = parsed_log_.receiver_reports(direction);
+  for (const auto& rtcp : receiver_reports) {
+    float x = config_.GetCallTimeSec(rtcp.log_time());
+    uint32_t ssrc = rtcp.rr.sender_ssrc();
+    for (const auto& block : rtcp.rr.report_blocks()) {
+      float y = fy(block);
+      auto rr_report_it = rr_reports_by_ssrc.find(ssrc);
+      bool inserted;
+      if (rr_report_it == rr_reports_by_ssrc.end()) {
+        std::tie(rr_report_it, inserted) = rr_reports_by_ssrc.emplace(
+            ssrc, TimeSeries(GetStreamName(parsed_log_, direction, ssrc) +
+                                 " Receiver Reports",
+                             LineStyle::kLine, PointStyle::kHighlight));
+      }
+      rr_report_it->second.points.emplace_back(x, y);
+    }
+  }
+  for (auto& kv : rr_reports_by_ssrc) {
+    plot->AppendTimeSeries(std::move(kv.second));
+  }
+
+  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
+                 "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, yaxis_label, kBottomMargin, kTopMargin);
+  plot->SetTitle(title);
+}
+
+void EventLogAnalyzer::CreateIceCandidatePairConfigGraph(Plot* plot) const {
+  std::map<uint32_t, TimeSeries> configs_by_cp_id;
+  for (const auto& config : parsed_log_.ice_candidate_pair_configs()) {
+    if (configs_by_cp_id.find(config.candidate_pair_id) ==
+        configs_by_cp_id.end()) {
+      const std::string candidate_pair_desc =
+          GetCandidatePairLogDescriptionAsString(config);
+      configs_by_cp_id[config.candidate_pair_id] =
+          TimeSeries("[" + std::to_string(config.candidate_pair_id) + "]" +
+                         candidate_pair_desc,
+                     LineStyle::kNone, PointStyle::kHighlight);
+    }
+    float x = config_.GetCallTimeSec(config.log_time());
+    float y = static_cast<float>(config.type);
+    configs_by_cp_id[config.candidate_pair_id].points.emplace_back(x, y);
+  }
+
+  // TODO(qingsi): There can be a large number of candidate pairs generated by
+  // certain calls and the frontend cannot render the chart in this case due
+  // to the failure of generating a palette with the same number of colors.
+  for (auto& kv : configs_by_cp_id) {
+    plot->AppendTimeSeries(std::move(kv.second));
+  }
+
+  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
+                 "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 3, "Config Type", kBottomMargin, kTopMargin);
+  plot->SetTitle("[IceEventLog] ICE candidate pair configs");
+  plot->SetYAxisTickLabels(
+      {{static_cast<float>(IceCandidatePairConfigType::kAdded), "ADDED"},
+       {static_cast<float>(IceCandidatePairConfigType::kUpdated), "UPDATED"},
+       {static_cast<float>(IceCandidatePairConfigType::kDestroyed),
+        "DESTROYED"},
+       {static_cast<float>(IceCandidatePairConfigType::kSelected),
+        "SELECTED"}});
+}
+
+void EventLogAnalyzer::CreateIceConnectivityCheckGraph(Plot* plot) const {
+  constexpr int kEventTypeOffset =
+      static_cast<int>(IceCandidatePairConfigType::kNumValues);
+  std::map<uint32_t, TimeSeries> checks_by_cp_id;
+  std::map<uint32_t, std::string> candidate_pair_desc_by_id =
+      BuildCandidateIdLogDescriptionMap(
+          parsed_log_.ice_candidate_pair_configs());
+  for (const auto& event : parsed_log_.ice_candidate_pair_events()) {
+    if (checks_by_cp_id.find(event.candidate_pair_id) ==
+        checks_by_cp_id.end()) {
+      checks_by_cp_id[event.candidate_pair_id] =
+          TimeSeries("[" + std::to_string(event.candidate_pair_id) + "]" +
+                         candidate_pair_desc_by_id[event.candidate_pair_id],
+                     LineStyle::kNone, PointStyle::kHighlight);
+    }
+    float x = config_.GetCallTimeSec(event.log_time());
+    float y = static_cast<float>(event.type) + kEventTypeOffset;
+    checks_by_cp_id[event.candidate_pair_id].points.emplace_back(x, y);
+  }
+
+  // TODO(qingsi): The same issue as in CreateIceCandidatePairConfigGraph.
+  for (auto& kv : checks_by_cp_id) {
+    plot->AppendTimeSeries(std::move(kv.second));
+  }
+
+  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
+                 "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 4, "Connectivity State", kBottomMargin,
+                          kTopMargin);
+  plot->SetTitle("[IceEventLog] ICE connectivity checks");
+
+  plot->SetYAxisTickLabels(
+      {{static_cast<float>(IceCandidatePairEventType::kCheckSent) +
+            kEventTypeOffset,
+        "CHECK SENT"},
+       {static_cast<float>(IceCandidatePairEventType::kCheckReceived) +
+            kEventTypeOffset,
+        "CHECK RECEIVED"},
+       {static_cast<float>(IceCandidatePairEventType::kCheckResponseSent) +
+            kEventTypeOffset,
+        "RESPONSE SENT"},
+       {static_cast<float>(IceCandidatePairEventType::kCheckResponseReceived) +
+            kEventTypeOffset,
+        "RESPONSE RECEIVED"}});
+}
+
+void EventLogAnalyzer::CreateDtlsTransportStateGraph(Plot* plot) const {
+  TimeSeries states("DTLS Transport State", LineStyle::kNone,
+                    PointStyle::kHighlight);
+  for (const auto& event : parsed_log_.dtls_transport_states()) {
+    float x = config_.GetCallTimeSec(event.log_time());
+    float y = static_cast<float>(event.dtls_transport_state);
+    states.points.emplace_back(x, y);
+  }
+  plot->AppendTimeSeries(std::move(states));
+  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
+                 "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, static_cast<float>(DtlsTransportState::kNumValues),
+                          "Transport State", kBottomMargin, kTopMargin);
+  plot->SetTitle("DTLS Transport State");
+  plot->SetYAxisTickLabels(
+      {{static_cast<float>(DtlsTransportState::kNew), "NEW"},
+       {static_cast<float>(DtlsTransportState::kConnecting), "CONNECTING"},
+       {static_cast<float>(DtlsTransportState::kConnected), "CONNECTED"},
+       {static_cast<float>(DtlsTransportState::kClosed), "CLOSED"},
+       {static_cast<float>(DtlsTransportState::kFailed), "FAILED"}});
+}
+
+void EventLogAnalyzer::CreateDtlsWritableStateGraph(Plot* plot) const {
+  TimeSeries writable("DTLS Writable", LineStyle::kNone,
+                      PointStyle::kHighlight);
+  for (const auto& event : parsed_log_.dtls_writable_states()) {
+    float x = config_.GetCallTimeSec(event.log_time());
+    float y = static_cast<float>(event.writable);
+    writable.points.emplace_back(x, y);
+  }
+  plot->AppendTimeSeries(std::move(writable));
+  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
+                 "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Writable", kBottomMargin, kTopMargin);
+  plot->SetTitle("DTLS Writable State");
 }
 
 }  // namespace webrtc
diff --git a/rtc_tools/rtc_event_log_visualizer/analyzer.h b/rtc_tools/rtc_event_log_visualizer/analyzer.h
index 8ac90e3040..ca729b3cbc 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyzer.h
+++ b/rtc_tools/rtc_event_log_visualizer/analyzer.h
@@ -11,11 +11,13 @@
 #ifndef RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZER_H_
 #define RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZER_H_
 
+#include <cstdint>
 #include <cstdio>
 #include <functional>
 #include <string>
 #include <vector>
 
+#include "api/environment/environment.h"
 #include "api/function_view.h"
 #include "logging/rtc_event_log/rtc_event_log_parser.h"
 #include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
@@ -60,7 +62,9 @@ class EventLogAnalyzer {
   // duration of its lifetime. The ParsedRtcEventLogNew must not be destroyed or
   // modified while the EventLogAnalyzer is being used.
   EventLogAnalyzer(const ParsedRtcEventLog& log, bool normalize_time);
-  EventLogAnalyzer(const ParsedRtcEventLog& log, const AnalyzerConfig& config);
+  EventLogAnalyzer(const Environment& env,
+                   const ParsedRtcEventLog& log,
+                   const AnalyzerConfig& config);
 
   void CreateGraphsByName(const std::vector<std::string>& names,
                           PlotCollection* collection) const;
@@ -145,8 +149,19 @@ class EventLogAnalyzer {
   void PrintNotifications(FILE* file) const;
 
  private:
+  template <typename IterableType>
+  void CreateAccumulatedPacketsTimeSeries(Plot* plot,
+                                          const IterableType& packets,
+                                          const std::string& label) const;
+  void CreateEcnFeedbackGraph(Plot* plot, PacketDirection direction) const;
+
+  const Environment env_;
   const ParsedRtcEventLog& parsed_log_;
 
+  // A list of SSRCs we are interested in analysing.
+  // If left empty, all SSRCs will be considered relevant.
+  std::vector<uint32_t> desired_ssrc_;
+
   AnalyzerConfig config_;
 
   PlotMap plots_;
diff --git a/rtc_tools/rtc_event_log_visualizer/analyzer_bindings.cc b/rtc_tools/rtc_event_log_visualizer/analyzer_bindings.cc
index 4aed263a98..fa93031ff1 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyzer_bindings.cc
+++ b/rtc_tools/rtc_event_log_visualizer/analyzer_bindings.cc
@@ -64,10 +64,16 @@ void analyze_rtc_event_log(const char* log_contents,
     return;
   }
 
-  webrtc::AnalyzerConfig config(webrtc::CreateEnvironment(), parsed_log,
-                                /*normalize_time=*/true);
+  webrtc::AnalyzerConfig config;
   config.window_duration_ = webrtc::TimeDelta::Millis(250);
   config.step_ = webrtc::TimeDelta::Millis(10);
+  if (!parsed_log.start_log_events().empty()) {
+    config.rtc_to_utc_offset_ = parsed_log.start_log_events()[0].utc_time() -
+                                parsed_log.start_log_events()[0].log_time();
+  }
+  config.normalize_time_ = true;
+  config.begin_time_ = parsed_log.first_timestamp();
+  config.end_time_ = parsed_log.last_timestamp();
   if (config.end_time_ < config.begin_time_) {
     std::cerr << "Log end time " << config.end_time_.ms()
               << " not after begin time " << config.begin_time_.ms()
@@ -76,7 +82,8 @@ void analyze_rtc_event_log(const char* log_contents,
     return;
   }
 
-  webrtc::EventLogAnalyzer analyzer(parsed_log, config);
+  webrtc::EventLogAnalyzer analyzer(webrtc::CreateEnvironment(), parsed_log,
+                                    config);
   analyzer.InitializeMapOfNamedGraphs(/*show_detector_state=*/false,
                                       /*show_alr_state=*/false,
                                       /*show_link_capacity=*/false);
diff --git a/rtc_tools/rtc_event_log_visualizer/analyzer_common.cc b/rtc_tools/rtc_event_log_visualizer/analyzer_common.cc
index 559ad78b4a..c1cf7dc6b7 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyzer_common.cc
+++ b/rtc_tools/rtc_event_log_visualizer/analyzer_common.cc
@@ -11,14 +11,10 @@
 
 #include "rtc_tools/rtc_event_log_visualizer/analyzer_common.h"
 
-#include <algorithm>
 #include <cstdint>
 #include <string>
-#include <vector>
 
 #include "logging/rtc_event_log/rtc_event_log_parser.h"
-#include "rtc_base/checks.h"
-#include "rtc_base/logging.h"
 #include "rtc_base/strings/string_builder.h"
 
 namespace webrtc {
@@ -90,64 +86,4 @@ std::string GetLayerName(LayerDescription layer) {
   return name.str();
 }
 
-std::string SsrcToString(uint32_t ssrc) {
-  StringBuilder ss;
-  ss << "SSRC " << ssrc;
-  return ss.Release();
-}
-
-// Checks whether an SSRC is contained in the list of desired SSRCs.
-// Note that an empty SSRC list matches every SSRC.
-bool MatchingSsrc(uint32_t ssrc, const std::vector<uint32_t>& desired_ssrc) {
-  if (desired_ssrc.empty())
-    return true;
-  return std::find(desired_ssrc.begin(), desired_ssrc.end(), ssrc) !=
-         desired_ssrc.end();
-}
-
-// Computes the difference `later` - `earlier` where `later` and `earlier`
-// are counters that wrap at `modulus`. The difference is chosen to have the
-// least absolute value. For example if `modulus` is 8, then the difference will
-// be chosen in the range [-3, 4]. If `modulus` is 9, then the difference will
-// be in [-4, 4].
-int64_t WrappingDifference(uint32_t later, uint32_t earlier, int64_t modulus) {
-  RTC_DCHECK_LE(1, modulus);
-  RTC_DCHECK_LT(later, modulus);
-  RTC_DCHECK_LT(earlier, modulus);
-  int64_t difference =
-      static_cast<int64_t>(later) - static_cast<int64_t>(earlier);
-  int64_t max_difference = modulus / 2;
-  int64_t min_difference = max_difference - modulus + 1;
-  if (difference > max_difference) {
-    difference -= modulus;
-  }
-  if (difference < min_difference) {
-    difference += modulus;
-  }
-  if (difference > max_difference / 2 || difference < min_difference / 2) {
-    RTC_LOG(LS_WARNING) << "Difference between" << later << " and " << earlier
-                        << " expected to be in the range ("
-                        << min_difference / 2 << "," << max_difference / 2
-                        << ") but is " << difference
-                        << ". Correct unwrapping is uncertain.";
-  }
-  return difference;
-}
-
-std::string GetDirectionAsString(PacketDirection direction) {
-  if (direction == kIncomingPacket) {
-    return "Incoming";
-  } else {
-    return "Outgoing";
-  }
-}
-
-std::string GetDirectionAsShortString(PacketDirection direction) {
-  if (direction == kIncomingPacket) {
-    return "In";
-  } else {
-    return "Out";
-  }
-}
-
 }  // namespace webrtc
diff --git a/rtc_tools/rtc_event_log_visualizer/analyzer_common.h b/rtc_tools/rtc_event_log_visualizer/analyzer_common.h
index 048d3d1578..d2698f84b1 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyzer_common.h
+++ b/rtc_tools/rtc_event_log_visualizer/analyzer_common.h
@@ -15,9 +15,7 @@
 #include <cstdint>
 #include <optional>
 #include <string>
-#include <vector>
 
-#include "api/environment/environment.h"
 #include "api/function_view.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
@@ -35,18 +33,6 @@ constexpr float kTopMargin = 0.05f;
 
 class AnalyzerConfig {
  public:
-  AnalyzerConfig(const Environment& env,
-                 const ParsedRtcEventLog& parsed_log,
-                 bool normalize_time)
-      : normalize_time_(normalize_time), env_(env) {
-    begin_time_ = parsed_log.first_timestamp();
-    end_time_ = parsed_log.last_timestamp();
-    if (!parsed_log.start_log_events().empty()) {
-      rtc_to_utc_offset_ = parsed_log.start_log_events().front().utc_time() -
-                           parsed_log.start_log_events().front().log_time();
-    }
-  }
-
   float GetCallTimeSec(Timestamp timestamp) const {
     Timestamp offset = normalize_time_ ? begin_time_ : Timestamp::Zero();
     return static_cast<float>((timestamp - offset).us()) / 1000000;
@@ -80,8 +66,6 @@ class AnalyzerConfig {
   Timestamp end_time_ = Timestamp::MinusInfinity();
   TimeDelta rtc_to_utc_offset_ = TimeDelta::Zero();
   bool normalize_time_;
-  std::vector<uint32_t> desired_ssrc_;
-  const Environment env_;
 };
 
 struct LayerDescription {
@@ -114,19 +98,8 @@ bool IsAudioSsrc(const ParsedRtcEventLog& parsed_log,
 std::string GetStreamName(const ParsedRtcEventLog& parsed_log,
                           PacketDirection direction,
                           uint32_t ssrc);
-std::string SsrcToString(uint32_t ssrc);
-
 std::string GetLayerName(LayerDescription layer);
 
-std::string GetDirectionAsString(PacketDirection direction);
-std::string GetDirectionAsShortString(PacketDirection direction);
-
-int64_t WrappingDifference(uint32_t later, uint32_t earlier, int64_t modulus);
-
-// Checks whether an SSRC is contained in the list of desired SSRCs.
-// Note that an empty SSRC list matches every SSRC.
-bool MatchingSsrc(uint32_t ssrc, const std::vector<uint32_t>& desired_ssrc);
-
 // For each element in data_view, use `f()` to extract a y-coordinate and
 // store the result in a TimeSeries.
 template <typename DataType, typename IterableType>
diff --git a/rtc_tools/rtc_event_log_visualizer/main.cc b/rtc_tools/rtc_event_log_visualizer/main.cc
index a08588e8a1..03512b185b 100644
--- a/rtc_tools/rtc_event_log_visualizer/main.cc
+++ b/rtc_tools/rtc_event_log_visualizer/main.cc
@@ -192,11 +192,17 @@ int main(int argc, char* argv[]) {
     }
   }
 
-  webrtc::AnalyzerConfig config(env, parsed_log,
-                                absl::GetFlag(FLAGS_normalize_time));
+  webrtc::AnalyzerConfig config;
   config.window_duration_ =
       webrtc::TimeDelta::Millis(absl::GetFlag(FLAGS_averaging_window));
   config.step_ = webrtc::TimeDelta::Millis(absl::GetFlag(FLAGS_averaging_step));
+  if (!parsed_log.start_log_events().empty()) {
+    config.rtc_to_utc_offset_ = parsed_log.start_log_events()[0].utc_time() -
+                                parsed_log.start_log_events()[0].log_time();
+  }
+  config.normalize_time_ = absl::GetFlag(FLAGS_normalize_time);
+  config.begin_time_ = parsed_log.first_timestamp();
+  config.end_time_ = parsed_log.last_timestamp();
   if (config.end_time_ < config.begin_time_) {
     RTC_LOG(LS_WARNING) << "Log end time " << config.end_time_
                         << " not after begin time " << config.begin_time_
@@ -219,7 +225,7 @@ int main(int argc, char* argv[]) {
     has_generated_wav_file = true;
   }
 
-  webrtc::EventLogAnalyzer analyzer(parsed_log, config);
+  webrtc::EventLogAnalyzer analyzer(env, parsed_log, config);
   analyzer.InitializeMapOfNamedGraphs(absl::GetFlag(FLAGS_show_detector_state),
                                       absl::GetFlag(FLAGS_show_alr_state),
                                       absl::GetFlag(FLAGS_show_link_capacity));

From 48d38b6ea5546fd91db35e45d0dd37aeb2b3e907 Mon Sep 17 00:00:00 2001
From: henrika <henrika@google.com>
Date: Fri, 13 Feb 2026 11:40:50 +0100
Subject: [PATCH 040/472] Removes
 WebRTC.DesktopCapture.Win.WgcDirtyRegionSupport UMA

See also https://chromium-review.googlesource.com/c/chromium/src/+/7575113

Bug: chromium:483057570
Change-Id: I439d08504bc9c077608a814b0c4ebe03ed442a79
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/448980
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46893}
---
 .../desktop_capture/win/wgc_capturer_win.cc   | 40 -------------------
 1 file changed, 40 deletions(-)

diff --git a/modules/desktop_capture/win/wgc_capturer_win.cc b/modules/desktop_capture/win/wgc_capturer_win.cc
index f15eddc2a9..2f9abc07c1 100644
--- a/modules/desktop_capture/win/wgc_capturer_win.cc
+++ b/modules/desktop_capture/win/wgc_capturer_win.cc
@@ -49,7 +49,6 @@ constexpr wchar_t kCoreMessagingDll[] = L"CoreMessaging.dll";
 constexpr wchar_t kWgcSessionType[] =
     L"Windows.Graphics.Capture.GraphicsCaptureSession";
 constexpr wchar_t kApiContract[] = L"Windows.Foundation.UniversalApiContract";
-constexpr wchar_t kDirtyRegionMode[] = L"DirtyRegionMode";
 constexpr UINT16 kRequiredApiContractVersion = 8;
 
 enum class WgcCapturerResult {
@@ -70,44 +69,6 @@ void RecordWgcCapturerResult(WgcCapturerResult error) {
                             static_cast<int>(WgcCapturerResult::kMaxValue));
 }
 
-// Checks if the DirtyRegionMode property is present in GraphicsCaptureSession
-// and logs a boolean histogram with the result.
-// TODO(https://crbug.com/40259177): Detecting support for this property means
-// that the WGC API supports dirty regions and it can be utilized to improve
-// the capture performance and the existing zero-herz support.
-void LogDirtyRegionSupport() {
-  ComPtr<ABI::Windows::Foundation::Metadata::IApiInformationStatics>
-      api_info_statics;
-  HRESULT hr = GetActivationFactory<
-      ABI::Windows::Foundation::Metadata::IApiInformationStatics,
-      RuntimeClass_Windows_Foundation_Metadata_ApiInformation>(
-      &api_info_statics);
-  if (FAILED(hr)) {
-    return;
-  }
-
-  HSTRING dirty_region_mode;
-  hr = CreateHstring(kDirtyRegionMode, wcslen(kDirtyRegionMode),
-                     &dirty_region_mode);
-  if (FAILED(hr)) {
-    DeleteHstring(dirty_region_mode);
-    return;
-  }
-
-  HSTRING wgc_session_type;
-  hr = CreateHstring(kWgcSessionType, wcslen(kWgcSessionType),
-                     &wgc_session_type);
-  if (SUCCEEDED(hr)) {
-    boolean is_dirty_region_mode_supported =
-        api_info_statics->IsPropertyPresent(wgc_session_type, dirty_region_mode,
-                                            &is_dirty_region_mode_supported);
-    RTC_HISTOGRAM_BOOLEAN("WebRTC.DesktopCapture.Win.WgcDirtyRegionSupport",
-                          !!is_dirty_region_mode_supported);
-  }
-  DeleteHstring(dirty_region_mode);
-  DeleteHstring(wgc_session_type);
-}
-
 }  // namespace
 
 bool IsWgcSupported(CaptureType capture_type) {
@@ -217,7 +178,6 @@ WgcCapturerWin::WgcCapturerWin(
         reinterpret_cast<CreateDispatcherQueueControllerFunc>(GetProcAddress(
             core_messaging_library_, "CreateDispatcherQueueController"));
   }
-  LogDirtyRegionSupport();
 }
 
 WgcCapturerWin::~WgcCapturerWin() {

From a4e0209846980a90125adbbf02fee52574e1d8f5 Mon Sep 17 00:00:00 2001
From: Rasmus Brandt <brandtr@webrtc.org>
Date: Tue, 10 Feb 2026 14:38:36 +0100
Subject: [PATCH 041/472] Video timing simulator: Implement stream metrics.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This change implements aggregation of per-frame metrics into per-stream
metrics. There is some custom aggregation implemented, but the
aggregation of most metrics is delegated to `SamplesStatsCounter`.

I also took the opportunity to make the handling of the sentinel values
(infinite timestamps and time deltas) more robust.

Bug: b/423646186
Change-Id: Ia40cc4308f9fabdd137a7ac032194fed6a6a6964
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/448901
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46894}
---
 video/timing/simulator/BUILD.gn               |   1 +
 .../timing/simulator/decodability_simulator.h |  25 ++-
 .../decodability_simulator_unittest.cc        |  28 +++
 video/timing/simulator/frame_base.h           |   5 +-
 video/timing/simulator/frame_base_unittest.cc |   7 +-
 video/timing/simulator/rendering_simulator.h  | 186 +++++++++++++++---
 .../simulator/rendering_simulator_unittest.cc | 159 ++++++++++++---
 video/timing/simulator/rendering_tracker.h    |   2 +-
 video/timing/simulator/stream_base.h          | 181 +++++++++++++++--
 .../timing/simulator/stream_base_unittest.cc  |  97 ++++++++-
 10 files changed, 613 insertions(+), 78 deletions(-)

diff --git a/video/timing/simulator/BUILD.gn b/video/timing/simulator/BUILD.gn
index c0e282370b..8c5af06e95 100644
--- a/video/timing/simulator/BUILD.gn
+++ b/video/timing/simulator/BUILD.gn
@@ -107,6 +107,7 @@ if (rtc_enable_protobuf) {
         "../../../api:rtp_packet_info",
         "../../../api:sequence_checker",
         "../../../api/environment",
+        "../../../api/numerics",
         "../../../api/transport/rtp:dependency_descriptor",
         "../../../api/units:data_size",
         "../../../api/units:time_delta",
diff --git a/video/timing/simulator/decodability_simulator.h b/video/timing/simulator/decodability_simulator.h
index 5fc56f365e..7d1860865f 100644
--- a/video/timing/simulator/decodability_simulator.h
+++ b/video/timing/simulator/decodability_simulator.h
@@ -17,10 +17,12 @@
 
 #include "absl/algorithm/container.h"
 #include "api/array_view.h"
+#include "api/numerics/samples_stats_counter.h"
 #include "api/units/data_size.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
 #include "logging/rtc_event_log/rtc_event_log_parser.h"
+#include "rtc_base/checks.h"
 #include "video/timing/simulator/frame_base.h"
 #include "video/timing/simulator/results_base.h"
 #include "video/timing/simulator/stream_base.h"
@@ -46,14 +48,14 @@ class DecodabilitySimulator {
   struct Frame : public FrameBase<Frame> {
     // -- Values --
     // Frame information.
-    int num_packets = -1;
-    DataSize size = DataSize::Zero();
+    int num_packets = -1;              // Required.
+    DataSize size = DataSize::Zero();  // Required.
 
     // RTP header information.
-    int64_t unwrapped_rtp_timestamp = -1;
+    int64_t unwrapped_rtp_timestamp = -1;  // Required.
 
     // Frame timestamps.
-    Timestamp assembled_timestamp = Timestamp::PlusInfinity();
+    Timestamp assembled_timestamp = Timestamp::PlusInfinity();  // Required.
     Timestamp decodable_timestamp = Timestamp::PlusInfinity();
 
     // -- Populated values --
@@ -66,15 +68,28 @@ class DecodabilitySimulator {
     // -- Per-frame metrics --
     // Time spent waiting for reference frames to arrive.
     TimeDelta UndecodableDuration() const {
+      RTC_DCHECK(assembled_timestamp.IsFinite());
       return decodable_timestamp - assembled_timestamp;
     }
   };
 
   // All frames in one stream.
-  struct Stream : public StreamBase<Stream> {
+  struct Stream : public StreamBase<Stream, Frame> {
     Timestamp creation_timestamp = Timestamp::PlusInfinity();
     uint32_t ssrc = 0;
     std::vector<Frame> frames;
+
+    // -- Per-stream metrics --
+
+    // Total number of decodable frames.
+    int NumDecodableFrames() const {
+      return CountFiniteTimestamps(&Frame::decodable_timestamp);
+    }
+
+    // Samples of undecodable durations in ms.
+    SamplesStatsCounter UndecodableDurationMs() const {
+      return BuildSamplesMs(&Frame::UndecodableDuration);
+    }
   };
 
   // All streams.
diff --git a/video/timing/simulator/decodability_simulator_unittest.cc b/video/timing/simulator/decodability_simulator_unittest.cc
index a7e3047999..a95e0c9c47 100644
--- a/video/timing/simulator/decodability_simulator_unittest.cc
+++ b/video/timing/simulator/decodability_simulator_unittest.cc
@@ -13,7 +13,9 @@
 #include <memory>
 
 #include "absl/algorithm/container.h"
+#include "api/numerics/samples_stats_counter.h"
 #include "api/units/data_size.h"
+#include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
 #include "logging/rtc_event_log/rtc_event_log_parser.h"
 #include "test/gmock.h"
@@ -35,6 +37,32 @@ using ::testing::SizeIs;
 using Frame = DecodabilitySimulator::Frame;
 using Stream = DecodabilitySimulator::Stream;
 
+TEST(DecodabilitySimulatorFrameTest, UndecodableDuration) {
+  DecodabilitySimulator::Frame frame{
+      .assembled_timestamp = Timestamp::Millis(0),
+      .decodable_timestamp = Timestamp::Millis(5)};
+
+  EXPECT_EQ(frame.UndecodableDuration(), TimeDelta::Millis(5));
+}
+
+TEST(DecodabilitySimulatorStreamTest, NumDecodableFrames) {
+  DecodabilitySimulator::Stream stream{
+      .frames = {{}, {.decodable_timestamp = Timestamp::Millis(10)}}};
+
+  EXPECT_EQ(stream.NumDecodableFrames(), 1);
+}
+
+TEST(DecodabilitySimulatorStreamTest, UndecodableDurationMs) {
+  DecodabilitySimulator::Stream stream{
+      .frames = {{.assembled_timestamp = Timestamp::Millis(0)},
+                 {.assembled_timestamp = Timestamp::Millis(1),
+                  .decodable_timestamp = Timestamp::Millis(5)}}};
+
+  EXPECT_THAT(
+      stream.UndecodableDurationMs().GetTimedSamples(),
+      ElementsAre(Field(&SamplesStatsCounter::StatsSample::value, Eq(4))));
+}
+
 Matcher<const Frame&> EqualsFrame(const Frame& expected) {
   return AllOf(
       Field("num_packets", &Frame::num_packets, Eq(expected.num_packets)),
diff --git a/video/timing/simulator/frame_base.h b/video/timing/simulator/frame_base.h
index f46327e76b..939a8744cc 100644
--- a/video/timing/simulator/frame_base.h
+++ b/video/timing/simulator/frame_base.h
@@ -18,6 +18,7 @@
 #include "absl/algorithm/container.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
+#include "rtc_base/checks.h"
 
 namespace webrtc::video_timing_simulator {
 
@@ -35,9 +36,7 @@ struct FrameBase {
   // the derived class.
   Timestamp DepartureTimestamp(Timestamp offset = Timestamp::Zero()) const {
     int64_t unwrapped_rtp_timestamp = self().unwrapped_rtp_timestamp;
-    if (unwrapped_rtp_timestamp < 0) {
-      return Timestamp::PlusInfinity();
-    }
+    RTC_DCHECK_GE(unwrapped_rtp_timestamp, 0);
     constexpr int64_t kMicrosPerMillis = 1'000;
     constexpr int64_t kRtpVideoTicksPerMillis = 90;
     // Convert from RTP ticks to microseconds using integer division with
diff --git a/video/timing/simulator/frame_base_unittest.cc b/video/timing/simulator/frame_base_unittest.cc
index 63829f8e5e..ce4c0c1deb 100644
--- a/video/timing/simulator/frame_base_unittest.cc
+++ b/video/timing/simulator/frame_base_unittest.cc
@@ -29,11 +29,6 @@ struct TestFrame : public FrameBase<TestFrame> {
   Timestamp ArrivalTimestampInternal() const { return assembled_timestamp; }
 };
 
-TEST(FrameBaseTest, DepartureTimestampIsInvalidForUnsetRtpTimestamp) {
-  TestFrame frame;
-  EXPECT_FALSE(frame.DepartureTimestamp().IsFinite());
-}
-
 TEST(FrameBaseTest, DepartureTimestamp) {
   TestFrame frame{.unwrapped_rtp_timestamp = 3000};
   EXPECT_EQ(frame.DepartureTimestamp(), Timestamp::Micros(33333));
@@ -48,7 +43,7 @@ TEST(FrameBaseTest, DepartureTimestampWithOffset) {
             Timestamp::Micros(33333));
 }
 
-TEST(FrameBaseTest, ArrivalTimestampIsInvalidForUnsetRtpTimestamp) {
+TEST(FrameBaseTest, ArrivalTimestampIsInfiniteForUnsetRtpTimestamp) {
   TestFrame frame;
   EXPECT_FALSE(frame.ArrivalTimestamp().IsFinite());
 }
diff --git a/video/timing/simulator/rendering_simulator.h b/video/timing/simulator/rendering_simulator.h
index 83f3b46a36..6e6ea2dd9c 100644
--- a/video/timing/simulator/rendering_simulator.h
+++ b/video/timing/simulator/rendering_simulator.h
@@ -22,6 +22,7 @@
 #include "absl/algorithm/container.h"
 #include "api/array_view.h"
 #include "api/environment/environment.h"
+#include "api/numerics/samples_stats_counter.h"
 #include "api/units/data_size.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
@@ -62,13 +63,13 @@ class RenderingSimulator {
   struct Frame : public FrameBase<Frame> {
     // -- Values --
     // Frame information.
-    int num_packets = -1;
-    DataSize size = DataSize::Zero();
+    int num_packets = -1;              // Required.
+    DataSize size = DataSize::Zero();  // Required.
 
     // RTP header information.
     int payload_type = -1;
     uint32_t rtp_timestamp = 0;
-    int64_t unwrapped_rtp_timestamp = -1;
+    int64_t unwrapped_rtp_timestamp = -1;  // Required.
 
     // Dependency descriptor information.
     int64_t frame_id = -1;
@@ -76,28 +77,28 @@ class RenderingSimulator {
     int temporal_id = -1;
     int num_references = -1;
 
-    // Packet timestamps.
+    // Packet timestamps. Both are required.
     Timestamp first_packet_arrival_timestamp = Timestamp::PlusInfinity();
     Timestamp last_packet_arrival_timestamp = Timestamp::MinusInfinity();
 
     // Frame timestamps.
-    Timestamp assembled_timestamp = Timestamp::PlusInfinity();
+    Timestamp assembled_timestamp = Timestamp::PlusInfinity();  // Required.
     Timestamp render_timestamp = Timestamp::PlusInfinity();
     Timestamp decoded_timestamp = Timestamp::PlusInfinity();
     Timestamp rendered_timestamp = Timestamp::PlusInfinity();
 
     // Jitter buffer state at the time of this frame.
-    int frames_dropped = -1;
+    int frames_dropped = 0;
     // TODO: b/423646186 - Add `current_delay_ms`.
     // The `jitter_buffer_*` metrics below are recorded by the production code,
     // and should be compatible with the `webrtc-stats` definitions. One major
     // difference is that they are _not_ cumulative.
     // https://www.w3.org/TR/webrtc-stats/#dom-rtcinboundrtpstreamstats-jitterbufferminimumdelay
-    TimeDelta jitter_buffer_minimum_delay = TimeDelta::MinusInfinity();
+    TimeDelta jitter_buffer_minimum_delay = TimeDelta::PlusInfinity();
     // https://www.w3.org/TR/webrtc-stats/#dom-rtcinboundrtpstreamstats-jitterbuffertargetdelay
-    TimeDelta jitter_buffer_target_delay = TimeDelta::MinusInfinity();
+    TimeDelta jitter_buffer_target_delay = TimeDelta::PlusInfinity();
     // https://www.w3.org/TR/webrtc-stats/#dom-rtcinboundrtpstreamstats-jitterbufferdelay
-    TimeDelta jitter_buffer_delay = TimeDelta::MinusInfinity();
+    TimeDelta jitter_buffer_delay = TimeDelta::PlusInfinity();
 
     // -- Populated values --
     // One-way delay relative some baseline.
@@ -109,6 +110,8 @@ class RenderingSimulator {
     // -- Per-frame metrics --
     // Time spent being assembled (waiting for all packets to arrive).
     TimeDelta PacketBufferDuration() const {
+      RTC_DCHECK(assembled_timestamp.IsFinite());
+      RTC_DCHECK(first_packet_arrival_timestamp.IsFinite());
       return assembled_timestamp - first_packet_arrival_timestamp;
     }
 
@@ -118,11 +121,19 @@ class RenderingSimulator {
     // production code; and, 2) is anchored on `first_packet_arrival_timestamp`
     // rather than `assembled_timestamp`.
     TimeDelta FrameBufferDuration() const {
+      RTC_DCHECK(assembled_timestamp.IsFinite());
       return decoded_timestamp - assembled_timestamp;
     }
 
     // Time spent waiting to be rendered, after decode.
     TimeDelta RenderBufferDuration() const {
+      if (!decoded_timestamp.IsFinite()) {
+        RTC_DCHECK(!rendered_timestamp.IsFinite());
+        return TimeDelta::PlusInfinity();
+      }
+      if (!rendered_timestamp.IsFinite()) {
+        return TimeDelta::PlusInfinity();
+      }
       return rendered_timestamp - decoded_timestamp;
     }
 
@@ -150,12 +161,27 @@ class RenderingSimulator {
     // negative margin.
     // TODO: b/423646186 - Change this to be `DecodabilityMargin`.
     TimeDelta AssembledMargin() const {
+      if (!render_timestamp.IsFinite()) {
+        return TimeDelta::PlusInfinity();
+      }
+      RTC_DCHECK(assembled_timestamp.IsFinite());
       // Subtract `kRenderDelay`, since that is what `VCMTiming` and
       // `VideoRenderFrames` also do.
       return (render_timestamp - kRenderDelay) - assembled_timestamp;
     }
-    bool AssembledInTime() const {
-      return AssembledMargin() > kInTimeMarginThreshold;
+    std::optional<bool> AssembledInTime() const {
+      TimeDelta assembled_margin = AssembledMargin();
+      if (!assembled_margin.IsFinite()) {
+        return std::nullopt;
+      }
+      return assembled_margin > kInTimeMarginThreshold;
+    }
+    std::optional<bool> AssembledLate() const {
+      std::optional<bool> assembled_in_time = AssembledInTime();
+      if (!assembled_in_time.has_value()) {
+        return std::nullopt;
+      }
+      return !assembled_in_time.value();
     }
 
     // Split the assembled margin along zero: "excess margin" for positive
@@ -164,13 +190,22 @@ class RenderingSimulator {
     // (delayed frames/buffer underruns => video freezes), while also minimizing
     // the stream-level min/p10 of the excess margin (early frames spending a
     // long time in the buffer => high latency).
+    // Frames with `render_timestamp` unset are excluded from both.
     std::optional<TimeDelta> AssembledMarginExcess() const {
-      return AssembledInTime() ? std::optional<TimeDelta>(AssembledMargin())
-                               : std::nullopt;
+      std::optional<bool> assembled_in_time = AssembledInTime();
+      if (!assembled_in_time.has_value()) {
+        return std::nullopt;
+      }
+      return *assembled_in_time ? std::optional<TimeDelta>(AssembledMargin())
+                                : std::nullopt;
     }
     std::optional<TimeDelta> AssembledMarginDeficit() const {
-      return !AssembledInTime() ? std::optional<TimeDelta>(AssembledMargin())
-                                : std::nullopt;
+      std::optional<bool> assembled_late = AssembledLate();
+      if (!assembled_late.has_value()) {
+        return std::nullopt;
+      }
+      return *assembled_late ? std::optional<TimeDelta>(AssembledMargin())
+                             : std::nullopt;
     }
 
     // Post-buffer margin between render timestamp (target) and
@@ -178,30 +213,137 @@ class RenderingSimulator {
     //   * Frames should not be rendered early, if the timing works well.
     //   * A frame that is rendered on time w.r.t. the target has zero margin.
     //   * A frame that is rendered late w.r.t. the target has negative margin.
+    // Frames with `render_timestamp` or `rendered_timestamp` unset are excluded
+    // from all.
     TimeDelta RenderedMargin() const {
+      if (!render_timestamp.IsFinite()) {
+        RTC_DCHECK(!rendered_timestamp.IsFinite());
+        return TimeDelta::PlusInfinity();
+      }
+      if (!rendered_timestamp.IsFinite()) {
+        return TimeDelta::PlusInfinity();
+      }
       return (render_timestamp - kRenderDelay) - rendered_timestamp;
     }
-    bool RenderedInTime() const {
-      return RenderedMargin() > kInTimeMarginThreshold;
+    std::optional<bool> RenderedInTime() const {
+      TimeDelta rendered_margin = RenderedMargin();
+      if (!rendered_margin.IsFinite()) {
+        return std::nullopt;
+      }
+      return rendered_margin > kInTimeMarginThreshold;
+    }
+    std::optional<bool> RenderedLate() const {
+      std::optional<bool> rendered_in_time = RenderedInTime();
+      if (!rendered_in_time.has_value()) {
+        return std::nullopt;
+      }
+      return !rendered_in_time.value();
     }
 
     // A well-functioning jitter buffer would have very few (or non) frames with
     // a render margin excess. Render margin deficits can happen though.
+    // Frames with `render_timestamp` or `rendered_timestamp` unset are excluded
+    // from both.
     std::optional<TimeDelta> RenderedMarginExcess() const {
-      return RenderedInTime() ? std::optional<TimeDelta>(RenderedMargin())
-                              : std::nullopt;
+      std::optional<bool> rendered_in_time = RenderedInTime();
+      if (!rendered_in_time.has_value()) {
+        return std::nullopt;
+      }
+      return *rendered_in_time ? std::optional<TimeDelta>(RenderedMargin())
+                               : std::nullopt;
     }
     std::optional<TimeDelta> RenderedMarginDeficit() const {
-      return !RenderedInTime() ? std::optional<TimeDelta>(RenderedMargin())
-                               : std::nullopt;
+      std::optional<bool> rendered_late = RenderedLate();
+      if (!rendered_late.has_value()) {
+        return std::nullopt;
+      }
+      return *rendered_late ? std::optional<TimeDelta>(RenderedMargin())
+                            : std::nullopt;
     }
   };
 
   // All frames in one stream.
-  struct Stream : public StreamBase<Stream> {
+  struct Stream : public StreamBase<Stream, Frame> {
     Timestamp creation_timestamp = Timestamp::PlusInfinity();
     uint32_t ssrc = 0;
     std::vector<Frame> frames;
+
+    // -- Per-stream metrics --
+
+    // Total number of frames that were assembled in time or late.
+    int NumAssembledInTimeFrames() const {
+      return CountSetAndTrue(&Frame::AssembledInTime);
+    }
+    int NumAssembledLateFrames() const {
+      return CountSetAndTrue(&Frame::AssembledLate);
+    }
+
+    // Total number of decoded frames.
+    int NumDecodedFrames() const {
+      return CountFiniteTimestamps(&Frame::decoded_timestamp);
+    }
+
+    // Total number of rendered frames.
+    int NumRenderedFrames() const {
+      return CountFiniteTimestamps(&Frame::rendered_timestamp);
+    }
+
+    // Total number of frames that were rendered in time or late.
+    int NumRenderedInTimeFrames() const {
+      return CountSetAndTrue(&Frame::RenderedInTime);
+    }
+    int NumRenderedLateFrames() const {
+      return CountSetAndTrue(&Frame::RenderedLate);
+    }
+
+    // Total number of dropped frames in the decoder.
+    int NumDecoderDroppedFrames() const {
+      return SumNonNegativeIntField(&Frame::frames_dropped);
+    }
+
+    // Samples of webrtc-stats values in ms.
+    SamplesStatsCounter JitterBufferMinimumDelayMs() const {
+      return BuildSamplesMs(&Frame::jitter_buffer_minimum_delay);
+    }
+    SamplesStatsCounter JitterBufferDelayMs() const {
+      return BuildSamplesMs(&Frame::jitter_buffer_delay);
+    }
+
+    // Samples of buffer durations in ms.
+    SamplesStatsCounter PacketBufferDurationMs() const {
+      return BuildSamplesMs(&Frame::PacketBufferDuration);
+    }
+    SamplesStatsCounter FrameBufferDurationMs() const {
+      return BuildSamplesMs(&Frame::FrameBufferDuration);
+    }
+    SamplesStatsCounter RenderBufferDurationMs() const {
+      return BuildSamplesMs(&Frame::RenderBufferDuration);
+    }
+    SamplesStatsCounter TotalBufferDurationMs() const {
+      return BuildSamplesMs(&Frame::TotalBufferDuration);
+    }
+
+    // Samples of assembled margin in ms.
+    SamplesStatsCounter AssembledMarginMs() const {
+      return BuildSamplesMs(&Frame::AssembledMargin);
+    }
+    SamplesStatsCounter AssembledMarginExcessMs() const {
+      return BuildSamplesMs(&Frame::AssembledMarginExcess);
+    }
+    SamplesStatsCounter AssembledMarginDeficitMs() const {
+      return BuildSamplesMs(&Frame::AssembledMarginDeficit);
+    }
+
+    // Samples of render margin in ms.
+    SamplesStatsCounter RenderedMarginMs() const {
+      return BuildSamplesMs(&Frame::RenderedMargin);
+    }
+    SamplesStatsCounter RenderedMarginExcessMs() const {
+      return BuildSamplesMs(&Frame::RenderedMarginExcess);
+    }
+    SamplesStatsCounter RenderedMarginDeficitMs() const {
+      return BuildSamplesMs(&Frame::RenderedMarginDeficit);
+    }
   };
 
   // All streams.
diff --git a/video/timing/simulator/rendering_simulator_unittest.cc b/video/timing/simulator/rendering_simulator_unittest.cc
index debec90aee..e3f6d5904c 100644
--- a/video/timing/simulator/rendering_simulator_unittest.cc
+++ b/video/timing/simulator/rendering_simulator_unittest.cc
@@ -11,9 +11,11 @@
 #include "video/timing/simulator/rendering_simulator.h"
 
 #include <memory>
+#include <optional>
 
 #include "absl/algorithm/container.h"
 #include "absl/strings/string_view.h"
+#include "api/numerics/samples_stats_counter.h"
 #include "api/units/data_size.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
@@ -29,8 +31,10 @@ namespace {
 using ::testing::AllOf;
 using ::testing::Eq;
 using ::testing::Field;
+using ::testing::IsEmpty;
 using ::testing::Matcher;
 using ::testing::Ne;
+using ::testing::Optional;
 using ::testing::SizeIs;
 
 using Frame = RenderingSimulator::Frame;
@@ -46,20 +50,22 @@ TEST(RenderingSimulatorFrameTest, BufferDurationsAndMarginsForInTimeFrame) {
       .rendered_timestamp = Timestamp::Millis(10)};
 
   // Buffer durations.
-  EXPECT_EQ(frame.PacketBufferDuration(), TimeDelta::Millis(1));
-  EXPECT_EQ(frame.FrameBufferDuration(), TimeDelta::Millis(8));
-  EXPECT_EQ(frame.RenderBufferDuration(), TimeDelta::Millis(1));
-  EXPECT_EQ(frame.TotalBufferDuration(), TimeDelta::Millis(10));
+  EXPECT_THAT(frame.PacketBufferDuration(), Eq(TimeDelta::Millis(1)));
+  EXPECT_THAT(frame.FrameBufferDuration(), Eq(TimeDelta::Millis(8)));
+  EXPECT_THAT(frame.RenderBufferDuration(), Eq(TimeDelta::Millis(1)));
+  EXPECT_THAT(frame.TotalBufferDuration(), Eq(TimeDelta::Millis(10)));
 
   // Margins.
-  EXPECT_EQ(frame.AssembledMargin(), TimeDelta::Millis(9));
-  EXPECT_TRUE(frame.AssembledInTime());
-  EXPECT_TRUE(frame.AssembledMarginExcess().has_value());
-  EXPECT_FALSE(frame.AssembledMarginDeficit().has_value());
-  EXPECT_EQ(frame.RenderedMargin(), TimeDelta::Zero());
-  EXPECT_TRUE(frame.RenderedInTime());
-  EXPECT_TRUE(frame.RenderedMarginExcess().has_value());
-  EXPECT_FALSE(frame.RenderedMarginDeficit().has_value());
+  EXPECT_THAT(frame.AssembledMargin(), Eq(TimeDelta::Millis(9)));
+  EXPECT_THAT(frame.AssembledInTime(), Optional(true));
+  EXPECT_THAT(frame.AssembledLate(), Optional(false));
+  EXPECT_THAT(frame.AssembledMarginExcess(), Optional(TimeDelta::Millis(9)));
+  EXPECT_THAT(frame.AssembledMarginDeficit(), Eq(std::nullopt));
+  EXPECT_THAT(frame.RenderedMargin(), Eq(TimeDelta::Zero()));
+  EXPECT_THAT(frame.RenderedInTime(), Optional(true));
+  EXPECT_THAT(frame.RenderedLate(), Optional(false));
+  EXPECT_THAT(frame.RenderedMarginExcess(), Optional(TimeDelta::Zero()));
+  EXPECT_THAT(frame.RenderedMarginDeficit(), Eq(std::nullopt));
 }
 
 TEST(RenderingSimulatorFrameTest, BufferDurationsAndMarginsForDelayedFrame) {
@@ -72,20 +78,22 @@ TEST(RenderingSimulatorFrameTest, BufferDurationsAndMarginsForDelayedFrame) {
       .rendered_timestamp = Timestamp::Millis(12)};
 
   // Buffer durations.
-  EXPECT_EQ(frame.PacketBufferDuration(), TimeDelta::Zero());
-  EXPECT_EQ(frame.FrameBufferDuration(), TimeDelta::Zero());
-  EXPECT_EQ(frame.RenderBufferDuration(), TimeDelta::Zero());
-  EXPECT_EQ(frame.TotalBufferDuration(), TimeDelta::Zero());
+  EXPECT_THAT(frame.PacketBufferDuration(), Eq(TimeDelta::Zero()));
+  EXPECT_THAT(frame.FrameBufferDuration(), Eq(TimeDelta::Zero()));
+  EXPECT_THAT(frame.RenderBufferDuration(), Eq(TimeDelta::Zero()));
+  EXPECT_THAT(frame.TotalBufferDuration(), Eq(TimeDelta::Zero()));
 
   // Margins.
-  EXPECT_EQ(frame.AssembledMargin(), TimeDelta::Millis(-2));
-  EXPECT_FALSE(frame.AssembledInTime());
-  EXPECT_FALSE(frame.AssembledMarginExcess().has_value());
-  EXPECT_TRUE(frame.AssembledMarginDeficit().has_value());
-  EXPECT_EQ(frame.RenderedMargin(), TimeDelta::Millis(-2));
-  EXPECT_FALSE(frame.RenderedInTime());
-  EXPECT_FALSE(frame.RenderedMarginExcess().has_value());
-  EXPECT_TRUE(frame.RenderedMarginDeficit().has_value());
+  EXPECT_THAT(frame.AssembledMargin(), Eq(TimeDelta::Millis(-2)));
+  EXPECT_THAT(frame.AssembledInTime(), Optional(false));
+  EXPECT_THAT(frame.AssembledLate(), Optional(true));
+  EXPECT_THAT(frame.AssembledMarginExcess(), Eq(std::nullopt));
+  EXPECT_THAT(frame.AssembledMarginDeficit(), Optional(TimeDelta::Millis(-2)));
+  EXPECT_THAT(frame.RenderedMargin(), Eq(TimeDelta::Millis(-2)));
+  EXPECT_THAT(frame.AssembledInTime(), Optional(false));
+  EXPECT_THAT(frame.AssembledLate(), Optional(true));
+  EXPECT_THAT(frame.RenderedMarginExcess(), Eq(std::nullopt));
+  EXPECT_THAT(frame.RenderedMarginDeficit(), Optional(TimeDelta::Millis(-2)));
 }
 
 TEST(RenderingSimulatorFrameTest,
@@ -126,6 +134,109 @@ TEST(RenderingSimulatorFrameTest,
   EXPECT_TRUE(frame.RenderedInTime());
 }
 
+TEST(RenderingSimulatorStreamTest, NumAssembledInTimeAndLateFrames) {
+  RenderingSimulator::Stream stream{
+      .frames = {{.assembled_timestamp = Timestamp::Millis(0)},
+                 {.assembled_timestamp = Timestamp::Millis(10),
+                  .render_timestamp =
+                      Timestamp::Millis(10) + RenderingSimulator::kRenderDelay},
+                 {.assembled_timestamp = Timestamp::Millis(30),
+                  .render_timestamp = Timestamp::Millis(20) +
+                                      RenderingSimulator::kRenderDelay}}};
+
+  EXPECT_EQ(stream.NumAssembledFrames(), 3);
+  EXPECT_EQ(stream.NumAssembledInTimeFrames(), 1);
+  EXPECT_EQ(stream.NumAssembledLateFrames(), 1);
+}
+
+TEST(RenderingSimulatorStreamTest, NumDecodedFrames) {
+  RenderingSimulator::Stream stream{
+      .frames = {{}, {.decoded_timestamp = Timestamp::Millis(10)}}};
+
+  EXPECT_EQ(stream.NumDecodedFrames(), 1);
+}
+
+TEST(RenderingSimulatorStreamTest, NumRenderedInTimeAndLateFrames) {
+  RenderingSimulator::Stream stream{
+      .frames = {{},
+                 {.render_timestamp =
+                      Timestamp::Millis(10) + RenderingSimulator::kRenderDelay,
+                  .rendered_timestamp = Timestamp::Millis(10)},
+                 {.render_timestamp =
+                      Timestamp::Millis(20) + RenderingSimulator::kRenderDelay,
+                  .rendered_timestamp = Timestamp::Millis(30)}}};
+
+  EXPECT_EQ(stream.NumRenderedFrames(), 2);
+  EXPECT_EQ(stream.NumRenderedInTimeFrames(), 1);
+  EXPECT_EQ(stream.NumRenderedLateFrames(), 1);
+}
+
+TEST(RenderingSimulatorStreamTest, NumDecoderDroppedFrames) {
+  RenderingSimulator::Stream stream{
+      .frames = {{.frames_dropped = 1}, {.frames_dropped = 2}}};
+
+  EXPECT_EQ(stream.NumDecoderDroppedFrames(), 3);
+}
+
+TEST(RenderingSimulatorStreamTest, WebrtcStatsSamples) {
+  RenderingSimulator::Stream stream{
+      .frames = {{.rendered_timestamp = Timestamp::Zero(),
+                  .jitter_buffer_minimum_delay = TimeDelta::Millis(20),
+                  .jitter_buffer_delay = TimeDelta::Millis(30)}}};
+
+  EXPECT_THAT(
+      stream.JitterBufferMinimumDelayMs().GetTimedSamples(),
+      ElementsAre(Field(&SamplesStatsCounter::StatsSample::value, Eq(20))));
+  EXPECT_THAT(
+      stream.JitterBufferDelayMs().GetTimedSamples(),
+      ElementsAre(Field(&SamplesStatsCounter::StatsSample::value, Eq(30))));
+}
+
+TEST(RenderingSimulatorStreamTest, BufferDurations) {
+  RenderingSimulator::Stream stream{
+      .frames = {{.first_packet_arrival_timestamp = Timestamp::Millis(1),
+                  .assembled_timestamp = Timestamp::Millis(3),
+                  .decoded_timestamp = Timestamp::Millis(7),
+                  .rendered_timestamp = Timestamp::Millis(14)}}};
+
+  EXPECT_THAT(
+      stream.PacketBufferDurationMs().GetTimedSamples(),
+      ElementsAre(Field(&SamplesStatsCounter::StatsSample::value, Eq(2))));
+  EXPECT_THAT(
+      stream.FrameBufferDurationMs().GetTimedSamples(),
+      ElementsAre(Field(&SamplesStatsCounter::StatsSample::value, Eq(4))));
+  EXPECT_THAT(
+      stream.RenderBufferDurationMs().GetTimedSamples(),
+      ElementsAre(Field(&SamplesStatsCounter::StatsSample::value, Eq(7))));
+  EXPECT_THAT(
+      stream.TotalBufferDurationMs().GetTimedSamples(),
+      ElementsAre(Field(&SamplesStatsCounter::StatsSample::value, Eq(13))));
+}
+
+TEST(RenderingSimulatorStreamTest, Margins) {
+  RenderingSimulator::Stream stream{
+      .frames = {{.assembled_timestamp = Timestamp::Millis(0),
+                  .render_timestamp =
+                      Timestamp::Millis(10) + RenderingSimulator::kRenderDelay,
+                  .rendered_timestamp = Timestamp::Millis(15)}}};
+
+  EXPECT_THAT(
+      stream.AssembledMarginMs().GetTimedSamples(),
+      ElementsAre(Field(&SamplesStatsCounter::StatsSample::value, Eq(10))));
+  EXPECT_THAT(
+      stream.AssembledMarginExcessMs().GetTimedSamples(),
+      ElementsAre(Field(&SamplesStatsCounter::StatsSample::value, Eq(10))));
+  EXPECT_THAT(stream.AssembledMarginDeficitMs().GetTimedSamples(), IsEmpty());
+
+  EXPECT_THAT(
+      stream.RenderedMarginMs().GetTimedSamples(),
+      ElementsAre(Field(&SamplesStatsCounter::StatsSample::value, Eq(-5))));
+  EXPECT_THAT(stream.RenderedMarginExcessMs().GetTimedSamples(), IsEmpty());
+  EXPECT_THAT(
+      stream.RenderedMarginDeficitMs().GetTimedSamples(),
+      ElementsAre(Field(&SamplesStatsCounter::StatsSample::value, Eq(-5))));
+}
+
 Matcher<const Frame&> EqualsFrame(const Frame& expected) {
   return AllOf(
       Field("num_packets", &Frame::num_packets, Eq(expected.num_packets)),
diff --git a/video/timing/simulator/rendering_tracker.h b/video/timing/simulator/rendering_tracker.h
index 6639101a49..de33fae833 100644
--- a/video/timing/simulator/rendering_tracker.h
+++ b/video/timing/simulator/rendering_tracker.h
@@ -62,7 +62,7 @@ class RenderingTracker : public AssembledFrameCallback,
   struct Config {
     uint32_t ssrc = 0;
     // Fixed render delay term added to the render timestamps.
-    TimeDelta render_delay = TimeDelta::MinusInfinity();
+    TimeDelta render_delay = TimeDelta::PlusInfinity();
   };
 
   RenderingTracker(const Environment& env,
diff --git a/video/timing/simulator/stream_base.h b/video/timing/simulator/stream_base.h
index a097188618..2075bd3dca 100644
--- a/video/timing/simulator/stream_base.h
+++ b/video/timing/simulator/stream_base.h
@@ -8,14 +8,21 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <algorithm>
 #include <cstddef>
+#include <optional>
+#include <utility>
 #include <vector>
 
 #include "absl/algorithm/container.h"
+#include "absl/functional/any_invocable.h"
+#include "api/numerics/samples_stats_counter.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
+#include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/numerics/moving_percentile_filter.h"
+#include "video/timing/simulator/frame_base.h"
 
 #ifndef VIDEO_TIMING_SIMULATOR_STREAM_BASE_H_
 #define VIDEO_TIMING_SIMULATOR_STREAM_BASE_H_
@@ -23,7 +30,7 @@
 namespace webrtc::video_timing_simulator {
 
 // CRTP base struct for code reuse.
-template <typename StreamT>
+template <typename StreamT, typename FrameT>
 struct StreamBase {
   // Data members are defined in derived struct.
 
@@ -34,24 +41,17 @@ struct StreamBase {
   // -- Helpers --
   bool IsEmpty() const { return self().frames.empty(); }
 
-  // -- Metric population --
+  // -- Per-frame metric population --
   void PopulateFrameDelayVariations(float baseline_percentile = 0.0,
                                     size_t baseline_window_size = 300) {
-    auto& frames = self().frames;
-    if (frames.empty()) {
+    if (IsEmpty()) {
       return;
     }
 
-    // The baseline filter measures the minimum (by default) one-way delay
-    // seen over a window. The corresponding value is then used to anchor all
-    // other one-way delay measurements, creating the frame delay variation.
-    MovingPercentileFilter<TimeDelta> baseline_filter(baseline_percentile,
-                                                      baseline_window_size);
-
     // One-way delay measurement offsets.
     Timestamp arrival_offset = Timestamp::PlusInfinity();
     Timestamp departure_offset = Timestamp::PlusInfinity();
-    for (const auto& frame : frames) {
+    for (const auto& frame : self().frames) {
       Timestamp arrival = frame.ArrivalTimestamp();
       Timestamp departure = frame.DepartureTimestamp();
       if (arrival.IsFinite() && departure.IsFinite()) {
@@ -66,8 +66,14 @@ struct StreamBase {
       return;
     }
 
+    // The baseline filter measures the minimum (by default) one-way delay
+    // seen over a window. The corresponding value is then used to anchor all
+    // other one-way delay measurements, creating the frame delay variation.
+    MovingPercentileFilter<TimeDelta> baseline_filter(baseline_percentile,
+                                                      baseline_window_size);
+
     // Calculate frame delay variations relative the moving baseline.
-    for (auto& frame : frames) {
+    for (auto& frame : self().frames) {
       TimeDelta one_way_delay =
           frame.OneWayDelay(arrival_offset, departure_offset);
       baseline_filter.Insert(one_way_delay);
@@ -75,6 +81,157 @@ struct StreamBase {
           one_way_delay - baseline_filter.GetFilteredValue();
     }
   }
+
+  // -- Per stream-metric aggregation --
+
+  // Count number of set and true booleans accessed through `accessor`.
+  int CountSetAndTrue(absl::AnyInvocable<std::optional<bool>(const FrameT&)
+                                             const> accessor) const {
+    return absl::c_count_if(
+        self().frames, [accessor = std::move(accessor)](const auto& frame) {
+          std::optional<bool> value = accessor(frame);
+          return value.has_value() && *value;
+        });
+  }
+
+  // Count number of finite timestamps accessed through `accessor`.
+  int CountFiniteTimestamps(
+      absl::AnyInvocable<Timestamp(const FrameT&) const> accessor) const {
+    return absl::c_count_if(
+        self().frames, [accessor = std::move(accessor)](const auto& frame) {
+          return accessor(frame).IsFinite();
+        });
+  }
+
+  // Sum non-negative int field values accessed through `accessor`.
+  int SumNonNegativeIntField(
+      absl::AnyInvocable<int(const FrameT&) const> accessor) const {
+    return absl::c_accumulate(
+        self().frames, 0,
+        [accessor = std::move(accessor)](int sum, const auto& frame) {
+          int value = accessor(frame);
+          RTC_DCHECK_GE(value, 0);
+          return sum + value;
+        });
+  }
+
+  // Build samples of positive int field values accessed through `accessor`.
+  SamplesStatsCounter BuildSamplesPositiveInt(
+      absl::AnyInvocable<int(const FrameT&) const> accessor) const {
+    SamplesStatsCounter stats(self().frames.size());
+    for (const auto& frame : self().frames) {
+      int value = accessor(frame);
+      RTC_DCHECK_GT(value, 0);
+      stats.AddSample({.value = static_cast<double>(value),
+                       .time = Timestamp::PlusInfinity()});
+    }
+    return stats;
+  }
+
+  // Build samples of all set and finite TimeDelta field values accessed
+  // through `accessor`.
+  SamplesStatsCounter BuildSamplesMs(
+      absl::AnyInvocable<std::optional<TimeDelta>(const FrameT&) const>
+          accessor) const {
+    SamplesStatsCounter stats(self().frames.size());
+    for (const auto& frame : self().frames) {
+      std::optional<TimeDelta> value = accessor(frame);
+      if (!value.has_value() || !value->IsFinite()) {
+        continue;
+      }
+      stats.AddSample(
+          {.value = value->ms<double>(), .time = Timestamp::PlusInfinity()});
+    }
+    return stats;
+  }
+
+  // Build samples of all TimeDelta inter-calculated metrics provided by
+  // `calculator`.
+  SamplesStatsCounter BuildSamplesMs(
+      absl::AnyInvocable<TimeDelta(const FrameT&, const FrameT&) const>
+          calculator) const {
+    SamplesStatsCounter stats(self().frames.size());
+    for (size_t i = 1; i < self().frames.size(); ++i) {
+      const auto& cur = self().frames[i];
+      const auto& prev = self().frames[i - 1];
+      TimeDelta inter = calculator(cur, prev);
+      if (!inter.IsFinite()) {
+        continue;
+      }
+      stats.AddSample(
+          {.value = inter.ms<double>(), .time = Timestamp::PlusInfinity()});
+    }
+    return stats;
+  }
+
+  // Duration between min and max finite timestamps accessed through
+  // `accessor`.
+  TimeDelta MinMaxDuration(
+      absl::AnyInvocable<Timestamp(const FrameT&) const> accessor) const {
+    if (IsEmpty()) {
+      return TimeDelta::PlusInfinity();
+    }
+    Timestamp min_value = Timestamp::PlusInfinity();
+    Timestamp max_value = Timestamp::MinusInfinity();
+    for (const auto& frame : self().frames) {
+      Timestamp time = accessor(frame);
+      if (!time.IsFinite()) {
+        continue;
+      }
+      min_value = std::min(min_value, time);
+      max_value = std::max(time, max_value);
+    }
+    if (!min_value.IsFinite() || !max_value.IsFinite()) {
+      return TimeDelta::PlusInfinity();
+    }
+    RTC_DCHECK_LE(min_value, max_value);
+    return max_value - min_value;
+  }
+
+  // -- Per-stream metrics --
+
+  // Duration between first and last departed frames.
+  TimeDelta DepartureDuration() const {
+    return MinMaxDuration(
+        [](const auto& frame) { return frame.DepartureTimestamp(); });
+  }
+
+  // Duration between first and last arrived frames.
+  TimeDelta ArrivalDuration() const {
+    return MinMaxDuration(
+        [](const auto& frame) { return frame.ArrivalTimestamp(); });
+  }
+
+  // Total number of assembled frames.
+  int NumAssembledFrames() const {
+    int num_finite_timestamps =
+        CountFiniteTimestamps(&FrameT::assembled_timestamp);
+    RTC_DCHECK_EQ(num_finite_timestamps, self().frames.size());
+    return num_finite_timestamps;
+  }
+
+  // Samples of per-frame sizes.
+  SamplesStatsCounter NumPackets() const {
+    return BuildSamplesPositiveInt(&FrameT::num_packets);
+  }
+  SamplesStatsCounter SizeBytes() const {
+    return BuildSamplesPositiveInt(
+        [](const auto& frame) { return frame.size.bytes(); });
+  }
+
+  // Samples of per-frame delay variation metrics.
+  SamplesStatsCounter FrameDelayVariationMs() const {
+    return BuildSamplesMs(&FrameT::frame_delay_variation);
+  }
+  SamplesStatsCounter InterDepartureTimeMs() const {
+    return BuildSamplesMs(&InterDepartureTime<FrameT>);
+  }
+  SamplesStatsCounter InterArrivalTimeMs() const {
+    return BuildSamplesMs(&InterArrivalTime<FrameT>);
+  }
+  SamplesStatsCounter InterFrameDelayVariationMs() const {
+    return BuildSamplesMs(&InterFrameDelayVariation<FrameT>);
+  }
 };
 
 // -- Comparators and sorting --
diff --git a/video/timing/simulator/stream_base_unittest.cc b/video/timing/simulator/stream_base_unittest.cc
index ff1ec4d1d7..4c56b0b439 100644
--- a/video/timing/simulator/stream_base_unittest.cc
+++ b/video/timing/simulator/stream_base_unittest.cc
@@ -13,6 +13,8 @@
 #include <cstdint>
 #include <vector>
 
+#include "api/numerics/samples_stats_counter.h"
+#include "api/units/data_size.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
 #include "test/gmock.h"
@@ -22,18 +24,20 @@
 namespace webrtc::video_timing_simulator {
 namespace {
 
-using ::testing::ElementsAreArray;
+using ::testing::ElementsAre;
 using ::testing::Eq;
 using ::testing::Field;
 
 struct TestFrame : public FrameBase<TestFrame> {
+  int num_packets = -1;
+  DataSize size = DataSize::Zero();
   int64_t unwrapped_rtp_timestamp = -1;
   Timestamp assembled_timestamp = Timestamp::PlusInfinity();
   Timestamp ArrivalTimestampInternal() const { return assembled_timestamp; }
   TimeDelta frame_delay_variation = TimeDelta::PlusInfinity();
 };
 
-struct TestStream : public StreamBase<TestStream> {
+struct TestStream : public StreamBase<TestStream, TestFrame> {
   Timestamp creation_timestamp = Timestamp::PlusInfinity();
   uint32_t ssrc = 0;
   std::vector<TestFrame> frames;
@@ -66,15 +70,98 @@ TEST(StreamBaseTest, PopulateFrameDelayVariations) {
 
   EXPECT_THAT(
       stream.frames,
-      ElementsAreArray({
+      ElementsAre(
           Field(&TestFrame::frame_delay_variation, Eq(TimeDelta::Zero())),
           Field(&TestFrame::frame_delay_variation, Eq(TimeDelta::Micros(1000))),
           // Due to the non-integer 1000/90 factor in the timestamp
           // translation, we get a 33332us here instead of 33333us.
           Field(&TestFrame::frame_delay_variation,
                 Eq(TimeDelta::Micros(33332))),
-          Field(&TestFrame::frame_delay_variation, Eq(TimeDelta::Zero())),
-      }));
+          Field(&TestFrame::frame_delay_variation, Eq(TimeDelta::Zero()))));
+}
+
+TEST(StreamBaseTest, DepartureDuration) {
+  TestStream stream{.frames = {{.unwrapped_rtp_timestamp = 3000},
+                               {.unwrapped_rtp_timestamp = 6000}}};
+
+  EXPECT_EQ(stream.DepartureDuration(), TimeDelta::Micros(33333));
+}
+
+TEST(StreamBaseTest, ArrivalDuration) {
+  TestStream stream{
+      .frames = {{.assembled_timestamp = Timestamp::Micros(33333)},
+                 {.assembled_timestamp = Timestamp::Micros(66666 + 1000)}}};
+
+  EXPECT_EQ(stream.ArrivalDuration(), TimeDelta::Micros(34333));
+}
+
+TEST(StreamBaseTest, NumAssembledFrames) {
+  TestStream stream{
+      .frames = {{.assembled_timestamp = Timestamp::Micros(33333)},
+                 {.assembled_timestamp = Timestamp::Micros(66666 + 1000)}}};
+
+  EXPECT_EQ(stream.NumAssembledFrames(), 2);
+}
+
+TEST(StreamBaseTest, NumPackets) {
+  TestStream stream{.frames = {{.num_packets = 1}, {.num_packets = 2}}};
+
+  EXPECT_THAT(
+      stream.NumPackets().GetTimedSamples(),
+      ElementsAre(Field(&SamplesStatsCounter::StatsSample::value, Eq(1)),
+                  Field(&SamplesStatsCounter::StatsSample::value, Eq(2))));
+}
+
+TEST(StreamBaseTest, SizeBytes) {
+  TestStream stream{
+      .frames = {{.size = DataSize::Bytes(10)}, {.size = DataSize::Bytes(20)}}};
+
+  EXPECT_THAT(
+      stream.SizeBytes().GetTimedSamples(),
+      ElementsAre(Field(&SamplesStatsCounter::StatsSample::value, Eq(10)),
+                  Field(&SamplesStatsCounter::StatsSample::value, Eq(20))));
+}
+
+TEST(StreamBaseTest, FrameDelayVariationMs) {
+  TestStream stream{
+      .frames = {{.frame_delay_variation = TimeDelta::Millis(10)},
+                 {.frame_delay_variation = TimeDelta::Millis(20)}}};
+
+  EXPECT_THAT(
+      stream.FrameDelayVariationMs().GetTimedSamples(),
+      ElementsAre(Field(&SamplesStatsCounter::StatsSample::value, Eq(10)),
+                  Field(&SamplesStatsCounter::StatsSample::value, Eq(20))));
+}
+
+TEST(StreamBaseTest, InterDepartureTimeMs) {
+  TestStream stream{.frames = {{.unwrapped_rtp_timestamp = 3000},
+                               {.unwrapped_rtp_timestamp = 6000}}};
+
+  EXPECT_THAT(
+      stream.InterDepartureTimeMs().GetTimedSamples(),
+      ElementsAre(Field(&SamplesStatsCounter::StatsSample::value, Eq(33.333))));
+}
+
+TEST(StreamBaseTest, InterArrivalTimeMs) {
+  TestStream stream{
+      .frames = {{.assembled_timestamp = Timestamp::Micros(33333)},
+                 {.assembled_timestamp = Timestamp::Micros(66666 + 1000)}}};
+
+  EXPECT_THAT(
+      stream.InterArrivalTimeMs().GetTimedSamples(),
+      ElementsAre(Field(&SamplesStatsCounter::StatsSample::value, Eq(34.333))));
+}
+
+TEST(StreamBaseTest, InterFrameDelayVariationMs) {
+  TestStream stream{
+      .frames = {{.unwrapped_rtp_timestamp = 3000,
+                  .assembled_timestamp = Timestamp::Micros(33333)},
+                 {.unwrapped_rtp_timestamp = 6000,
+                  .assembled_timestamp = Timestamp::Micros(66666 + 1000)}}};
+
+  EXPECT_THAT(
+      stream.InterFrameDelayVariationMs().GetTimedSamples(),
+      ElementsAre(Field(&SamplesStatsCounter::StatsSample::value, Eq(1.0))));
 }
 
 }  // namespace

From cf7d2b1bf3127b8bfd399837e71b2f550cd8622c Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Erik=20Spr=C3=A5ng?= <sprang@webrtc.org>
Date: Fri, 13 Feb 2026 10:48:11 +0100
Subject: [PATCH 042/472] Add support for OnFrameDropped callback in
 FrameEncodeMetadataWriter.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This wires the new OnFrameDropped callback method up through the encoder
metadata writer. Similar to SimulcastEncoderAdapter, it attempts to
detect gaps in the callbacks to detect frame drops - so that we can
support legacy encoders in a best-effort manner. Down the line the new
behavior should however be made mandatory.

Bug: webrtc:467444018
Change-Id: I8436fa64f8b2f6286008ebf0183188c0d279d1de
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/448700
Auto-Submit: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46895}
---
 video/frame_encode_metadata_writer.cc         |  88 ++++-
 video/frame_encode_metadata_writer.h          |  10 +
 .../frame_encode_metadata_writer_unittest.cc  | 374 +++++++++++++++++-
 3 files changed, 467 insertions(+), 5 deletions(-)

diff --git a/video/frame_encode_metadata_writer.cc b/video/frame_encode_metadata_writer.cc
index 40b6ad048a..828ae790a8 100644
--- a/video/frame_encode_metadata_writer.cc
+++ b/video/frame_encode_metadata_writer.cc
@@ -136,9 +136,13 @@ void FrameEncodeMetadataWriter::OnEncodeStarted(const VideoFrame& frame) {
                  "warnings will be throttled.";
         }
       }
-      frame_drop_callback_->OnDroppedFrame(
-          EncodedImageCallback::DropReason::kDroppedByEncoder);
+
+      uint32_t dropped_rtp_timestamp =
+          timing_frames_info_[si].frames.front().rtp_timestamp;
       timing_frames_info_[si].frames.pop_front();
+      frame_drop_callback_->OnFrameDropped(
+          dropped_rtp_timestamp, si,
+          !IsFrameWithRtpTimestampPending(dropped_rtp_timestamp));
     }
     timing_frames_info_[si].frames.emplace_back(metadata);
   }
@@ -200,6 +204,45 @@ void FrameEncodeMetadataWriter::FillMetadataAndTimingInfo(
   } else {
     encoded_image->timing_.flags = VideoSendTiming::kInvalid;
   }
+
+  // Cancel timing frames for older layers.
+  if (encoded_image->is_end_of_temporal_unit().value_or(false)) {
+    DropOldOrEqualFrames(encoded_image->RtpTimestamp(), simulcast_svc_idx);
+  }
+}
+
+void FrameEncodeMetadataWriter::OnFrameDropped(uint32_t rtp_timestamp,
+                                               int spatial_id,
+                                               bool is_end_of_temporal_unit) {
+  MutexLock lock(&lock_);
+  RTC_DCHECK_LT(spatial_id, static_cast<int>(timing_frames_info_.size()));
+  bool found_timestamp = false;
+  while (!timing_frames_info_[spatial_id].frames.empty()) {
+    if (timing_frames_info_[spatial_id].frames.front().rtp_timestamp ==
+        rtp_timestamp) {
+      found_timestamp = true;
+      break;
+    } else if (IsNewerTimestamp(rtp_timestamp, timing_frames_info_[spatial_id]
+                                                   .frames.front()
+                                                   .rtp_timestamp)) {
+      uint32_t dropped_timestamp =
+          timing_frames_info_[spatial_id].frames.front().rtp_timestamp;
+      timing_frames_info_[spatial_id].frames.pop_front();
+      frame_drop_callback_->OnFrameDropped(
+          dropped_timestamp, spatial_id,
+          !IsFrameWithRtpTimestampPending(dropped_timestamp));
+      continue;
+    }
+    break;
+  }
+  if (found_timestamp) {
+    if (is_end_of_temporal_unit) {
+      DropOldOrEqualFrames(rtp_timestamp, spatial_id);
+    }
+    timing_frames_info_[spatial_id].frames.pop_front();
+    frame_drop_callback_->OnFrameDropped(rtp_timestamp, spatial_id,
+                                         is_end_of_temporal_unit);
+  }
 }
 
 void FrameEncodeMetadataWriter::UpdateBitstream(
@@ -245,9 +288,11 @@ FrameEncodeMetadataWriter::ExtractEncodeStartTimeAndFillMetadata(
     while (!metadata_list->empty() &&
            IsNewerTimestamp(encoded_image->RtpTimestamp(),
                             metadata_list->front().rtp_timestamp)) {
-      frame_drop_callback_->OnDroppedFrame(
-          EncodedImageCallback::DropReason::kDroppedByEncoder);
+      uint32_t dropped_timestamp = metadata_list->front().rtp_timestamp;
       metadata_list->pop_front();
+      frame_drop_callback_->OnFrameDropped(
+          dropped_timestamp, simulcast_svc_idx,
+          !IsFrameWithRtpTimestampPending(dropped_timestamp));
     }
 
     encoded_image->content_type_ =
@@ -286,4 +331,39 @@ FrameEncodeMetadataWriter::ExtractEncodeStartTimeAndFillMetadata(
   return result;
 }
 
+void FrameEncodeMetadataWriter::DropOldOrEqualFrames(uint32_t rtp_timestamp,
+                                                     int spatial_id_to_skip) {
+  for (size_t si = 0; si < num_spatial_layers_; ++si) {
+    if (spatial_id_to_skip == static_cast<int>(si)) {
+      continue;
+    }
+    auto metadata_list = &timing_frames_info_[si].frames;
+    while (!metadata_list->empty() &&
+           (metadata_list->front().rtp_timestamp == rtp_timestamp ||
+            IsNewerTimestamp(rtp_timestamp,
+                             metadata_list->front().rtp_timestamp))) {
+      uint32_t dropped_timestamp = metadata_list->front().rtp_timestamp;
+      metadata_list->pop_front();
+      frame_drop_callback_->OnFrameDropped(
+          dropped_timestamp, si,
+          !IsFrameWithRtpTimestampPending(dropped_timestamp));
+    }
+  }
+}
+
+bool FrameEncodeMetadataWriter::IsFrameWithRtpTimestampPending(
+    uint32_t rtp_timestamp) {
+  for (size_t si = 0; si < num_spatial_layers_; ++si) {
+    for (const FrameMetadata& metadata : timing_frames_info_[si].frames) {
+      if (metadata.rtp_timestamp == rtp_timestamp) {
+        return true;
+      }
+      if (IsNewerTimestamp(metadata.rtp_timestamp, rtp_timestamp)) {
+        break;
+      }
+    }
+  }
+  return false;
+}
+
 }  // namespace webrtc
diff --git a/video/frame_encode_metadata_writer.h b/video/frame_encode_metadata_writer.h
index f8df3ae259..aa400c4741 100644
--- a/video/frame_encode_metadata_writer.h
+++ b/video/frame_encode_metadata_writer.h
@@ -50,6 +50,10 @@ class FrameEncodeMetadataWriter {
   void UpdateBitstream(const CodecSpecificInfo* codec_specific_info,
                        EncodedImage* encoded_image);
 
+  void OnFrameDropped(uint32_t rtp_timestamp,
+                      int spatial_id,
+                      bool is_end_of_temporal_unit);
+
   void Reset();
 
  private:
@@ -58,6 +62,12 @@ class FrameEncodeMetadataWriter {
   std::optional<int64_t> ExtractEncodeStartTimeAndFillMetadata(
       size_t simulcast_svc_idx,
       EncodedImage* encoded_image) RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
+  void DropOldOrEqualFrames(uint32_t rtp_timestamp, int spatial_id_to_skip)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
+  // Returns true if any spatial layer contains a pending frame with the given
+  // RTP timestamp.
+  bool IsFrameWithRtpTimestampPending(uint32_t rtp_timestamp)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
 
   struct FrameMetadata {
     uint32_t rtp_timestamp;
diff --git a/video/frame_encode_metadata_writer_unittest.cc b/video/frame_encode_metadata_writer_unittest.cc
index 6a166a39c3..f7fef8075f 100644
--- a/video/frame_encode_metadata_writer_unittest.cc
+++ b/video/frame_encode_metadata_writer_unittest.cc
@@ -37,6 +37,11 @@
 #include "test/gmock.h"
 #include "test/gtest.h"
 
+using ::testing::AllOf;
+using ::testing::ElementsAreArray;
+using ::testing::Eq;
+using ::testing::Field;
+
 namespace webrtc {
 namespace test {
 namespace {
@@ -50,6 +55,12 @@ inline size_t FrameSize(const size_t& min_frame_size,
   return min_frame_size + (s + 1) * i % (max_frame_size - min_frame_size);
 }
 
+struct DroppedFrameInfo {
+  uint32_t rtp_timestamp;
+  int spatial_id;
+  bool is_end_of_temporal_unit;
+};
+
 class FakeEncodedImageCallback : public EncodedImageCallback {
  public:
   FakeEncodedImageCallback() : num_frames_dropped_(0) {}
@@ -57,11 +68,21 @@ class FakeEncodedImageCallback : public EncodedImageCallback {
                         const CodecSpecificInfo* codec_specific_info) override {
     return Result(Result::OK);
   }
-  void OnDroppedFrame(DropReason reason) override { ++num_frames_dropped_; }
+  void OnFrameDropped(uint32_t rtp_timestamp,
+                      int spatial_id,
+                      bool is_end_of_temporal_unit) override {
+    ++num_frames_dropped_;
+    dropped_frames_.push_back(
+        {rtp_timestamp, spatial_id, is_end_of_temporal_unit});
+  }
   size_t GetNumFramesDropped() { return num_frames_dropped_; }
+  const std::vector<DroppedFrameInfo>& GetDroppedFrames() const {
+    return dropped_frames_;
+  }
 
  private:
   size_t num_frames_dropped_;
+  std::vector<DroppedFrameInfo> dropped_frames_;
 };
 
 enum class FrameType {
@@ -616,6 +637,357 @@ TEST(FrameEncodeMetadataWriterTest, Av1SimulcastSpatialLayersCalculation) {
     EXPECT_TRUE(IsTimingFrame(image));
   }
 }
+TEST(FrameEncodeMetadataWriterTest, NotifiesAboutExplicitlyDroppedFrames) {
+  const int64_t kTimestampMs = 4772184;
 
+  FakeEncodedImageCallback sink;
+  FrameEncodeMetadataWriter encode_timer(CreateTestEnvironment(), &sink);
+  encode_timer.OnEncoderInit(VideoCodec());
+  VideoBitrateAllocation bitrate_allocation;
+  bitrate_allocation.SetBitrate(0, 0, 500000);
+  encode_timer.OnSetRates(bitrate_allocation, 30);
+
+  VideoFrame frame = VideoFrame::Builder()
+                         .set_rtp_timestamp(kTimestampMs * 90)
+                         .set_timestamp_ms(kTimestampMs)
+                         .set_video_frame_buffer(kFrameBuffer)
+                         .build();
+  encode_timer.OnEncodeStarted(frame);
+  encode_timer.OnFrameDropped(kTimestampMs * 90, 0, true);
+
+  EXPECT_THAT(
+      sink.GetDroppedFrames(),
+      ElementsAreArray({AllOf(
+          Field(&DroppedFrameInfo::rtp_timestamp, Eq(kTimestampMs * 90)),
+          Field(&DroppedFrameInfo::spatial_id, Eq(0)),
+          Field(&DroppedFrameInfo::is_end_of_temporal_unit, Eq(true)))}));
+}
+
+TEST(FrameEncodeMetadataWriterTest, NotifiesAboutImplicitDropsOnEncodedFrame) {
+  const int64_t kTimestampMs1 = 100000;
+  const int64_t kTimestampMs2 = 100010;
+
+  FakeEncodedImageCallback sink;
+  FrameEncodeMetadataWriter encode_timer(CreateTestEnvironment(), &sink);
+  encode_timer.OnEncoderInit(VideoCodec());
+  VideoBitrateAllocation bitrate_allocation;
+  bitrate_allocation.SetBitrate(0, 0, 500000);
+  encode_timer.OnSetRates(bitrate_allocation, 30);
+
+  VideoFrame frame1 = VideoFrame::Builder()
+                          .set_rtp_timestamp(kTimestampMs1 * 90)
+                          .set_timestamp_ms(kTimestampMs1)
+                          .set_video_frame_buffer(kFrameBuffer)
+                          .build();
+  encode_timer.OnEncodeStarted(frame1);
+
+  VideoFrame frame2 = VideoFrame::Builder()
+                          .set_rtp_timestamp(kTimestampMs2 * 90)
+                          .set_timestamp_ms(kTimestampMs2)
+                          .set_video_frame_buffer(kFrameBuffer)
+                          .build();
+  encode_timer.OnEncodeStarted(frame2);
+
+  // Fill metadata for frame 2 without having been given any feedback signal for
+  // frame 1 - implying it has been dropped.
+  EncodedImage image2;
+  image2.capture_time_ms_ = kTimestampMs2;
+  image2.SetRtpTimestamp(static_cast<uint32_t>(kTimestampMs2 * 90));
+  encode_timer.FillMetadataAndTimingInfo(0, &image2);
+
+  EXPECT_THAT(
+      sink.GetDroppedFrames(),
+      ElementsAreArray({AllOf(
+          Field(&DroppedFrameInfo::rtp_timestamp, Eq(kTimestampMs1 * 90)),
+          Field(&DroppedFrameInfo::spatial_id, Eq(0)),
+          Field(&DroppedFrameInfo::is_end_of_temporal_unit, Eq(true)))}));
+}
+
+TEST(FrameEncodeMetadataWriterTest, NotifiesAboutImplicitDropsOnExplicitSkip) {
+  const int64_t kTimestampMs1 = 100000;
+  const int64_t kTimestampMs2 = 100010;
+
+  FakeEncodedImageCallback sink;
+  FrameEncodeMetadataWriter encode_timer(CreateTestEnvironment(), &sink);
+  encode_timer.OnEncoderInit(VideoCodec());
+  VideoBitrateAllocation bitrate_allocation;
+  bitrate_allocation.SetBitrate(0, 0, 500000);
+  encode_timer.OnSetRates(bitrate_allocation, 30);
+
+  VideoFrame frame1 = VideoFrame::Builder()
+                          .set_rtp_timestamp(kTimestampMs1 * 90)
+                          .set_timestamp_ms(kTimestampMs1)
+                          .set_video_frame_buffer(kFrameBuffer)
+                          .build();
+  encode_timer.OnEncodeStarted(frame1);
+
+  VideoFrame frame2 = VideoFrame::Builder()
+                          .set_rtp_timestamp(kTimestampMs2 * 90)
+                          .set_timestamp_ms(kTimestampMs2)
+                          .set_video_frame_buffer(kFrameBuffer)
+                          .build();
+  encode_timer.OnEncodeStarted(frame2);
+
+  // Drop signal for frame 2 without having been given a feedback signal for
+  // frame 1 implying it has been dropped.
+  encode_timer.OnFrameDropped(kTimestampMs2 * 90, 0, true);
+
+  EXPECT_THAT(
+      sink.GetDroppedFrames(),
+      ElementsAreArray(
+          {AllOf(
+               Field(&DroppedFrameInfo::rtp_timestamp, Eq(kTimestampMs1 * 90)),
+               Field(&DroppedFrameInfo::spatial_id, Eq(0)),
+               Field(&DroppedFrameInfo::is_end_of_temporal_unit, Eq(true))),
+           AllOf(
+               Field(&DroppedFrameInfo::rtp_timestamp, Eq(kTimestampMs2 * 90)),
+               Field(&DroppedFrameInfo::spatial_id, Eq(0)),
+               Field(&DroppedFrameInfo::is_end_of_temporal_unit, Eq(true)))}));
+}
+
+TEST(FrameEncodeMetadataWriterTest, DroppingAlreadyDroppedFrameDoesNothing) {
+  const int64_t kTimestampMs1 = 100000;
+  const int64_t kTimestampMs2 = 100010;
+
+  FakeEncodedImageCallback sink;
+  FrameEncodeMetadataWriter encode_timer(CreateTestEnvironment(), &sink);
+  encode_timer.OnEncoderInit(VideoCodec());
+  VideoBitrateAllocation bitrate_allocation;
+  bitrate_allocation.SetBitrate(0, 0, 500000);
+  encode_timer.OnSetRates(bitrate_allocation, 30);
+
+  VideoFrame frame1 = VideoFrame::Builder()
+                          .set_rtp_timestamp(kTimestampMs1 * 90)
+                          .set_timestamp_ms(kTimestampMs1)
+                          .set_video_frame_buffer(kFrameBuffer)
+                          .build();
+  encode_timer.OnEncodeStarted(frame1);
+
+  VideoFrame frame2 = VideoFrame::Builder()
+                          .set_rtp_timestamp(kTimestampMs2 * 90)
+                          .set_timestamp_ms(kTimestampMs2)
+                          .set_video_frame_buffer(kFrameBuffer)
+                          .build();
+  encode_timer.OnEncodeStarted(frame2);
+
+  // Drop frame 1.
+  encode_timer.OnFrameDropped(kTimestampMs1 * 90, 0, true);
+
+  // Dropping an old already-dropped frame does nothing.
+  encode_timer.OnFrameDropped(kTimestampMs1 * 90, 0, true);
+
+  // Encoding an old already-dropped frame shouldn't crash, but it won't drop
+  // anything new.
+  EncodedImage image1;
+  image1.capture_time_ms_ = kTimestampMs1;
+  image1.SetRtpTimestamp(static_cast<uint32_t>(kTimestampMs1 * 90));
+  encode_timer.FillMetadataAndTimingInfo(0, &image1);
+
+  EXPECT_THAT(
+      sink.GetDroppedFrames(),
+      ElementsAreArray({AllOf(
+          Field(&DroppedFrameInfo::rtp_timestamp, Eq(kTimestampMs1 * 90)),
+          Field(&DroppedFrameInfo::spatial_id, Eq(0)),
+          Field(&DroppedFrameInfo::is_end_of_temporal_unit, Eq(true)))}));
+}
+
+TEST(FrameEncodeMetadataWriterTest, ExplicitDropsWithMultipleSpatialLayers) {
+  const int64_t kTimestampMs1 = 100000;
+  const int64_t kTimestampMs2 = 100010;
+
+  FakeEncodedImageCallback sink;
+  FrameEncodeMetadataWriter encode_timer(CreateTestEnvironment(), &sink);
+  VideoCodec codec_settings;
+  codec_settings.codecType = kVideoCodecVP8;
+  codec_settings.numberOfSimulcastStreams = 2;
+  encode_timer.OnEncoderInit(codec_settings);
+  VideoBitrateAllocation bitrate_allocation;
+  bitrate_allocation.SetBitrate(0, 0, 500000);
+  bitrate_allocation.SetBitrate(1, 0, 500000);
+  encode_timer.OnSetRates(bitrate_allocation, 30);
+
+  VideoFrame frame1 = VideoFrame::Builder()
+                          .set_rtp_timestamp(kTimestampMs1 * 90)
+                          .set_timestamp_ms(kTimestampMs1)
+                          .set_video_frame_buffer(kFrameBuffer)
+                          .build();
+  encode_timer.OnEncodeStarted(frame1);
+
+  VideoFrame frame2 = VideoFrame::Builder()
+                          .set_rtp_timestamp(kTimestampMs2 * 90)
+                          .set_timestamp_ms(kTimestampMs2)
+                          .set_video_frame_buffer(kFrameBuffer)
+                          .build();
+  encode_timer.OnEncodeStarted(frame2);
+
+  // Drop frame 1 on spatial layer 0.
+  encode_timer.OnFrameDropped(kTimestampMs1 * 90, /*spatial_id=*/0,
+                              /*is_end_of_temporal_unit=*/false);
+  EXPECT_THAT(
+      sink.GetDroppedFrames(),
+      ElementsAreArray({AllOf(
+          Field(&DroppedFrameInfo::rtp_timestamp, Eq(kTimestampMs1 * 90)),
+          Field(&DroppedFrameInfo::spatial_id, Eq(0)),
+          Field(&DroppedFrameInfo::is_end_of_temporal_unit, Eq(false)))}));
+
+  // Drop frame 2 on spatial layer 1.
+  encode_timer.OnFrameDropped(kTimestampMs2 * 90, /*spatial_id=*/1,
+                              /*is_end_of_temporal_unit=*/true);
+  // This implicitly drops frame 1 on layer 1, THEN explicitly drops frame 2 on
+  // layer 1.
+  EXPECT_THAT(
+      sink.GetDroppedFrames(),
+      ElementsAreArray(
+          {AllOf(
+               Field(&DroppedFrameInfo::rtp_timestamp, Eq(kTimestampMs1 * 90)),
+               Field(&DroppedFrameInfo::spatial_id, Eq(0)),
+               Field(&DroppedFrameInfo::is_end_of_temporal_unit, Eq(false))),
+           AllOf(
+               Field(&DroppedFrameInfo::rtp_timestamp, Eq(kTimestampMs1 * 90)),
+               Field(&DroppedFrameInfo::spatial_id, Eq(1)),
+               Field(&DroppedFrameInfo::is_end_of_temporal_unit, Eq(true))),
+           AllOf(
+               Field(&DroppedFrameInfo::rtp_timestamp, Eq(kTimestampMs2 * 90)),
+               Field(&DroppedFrameInfo::spatial_id, Eq(0)),
+               Field(&DroppedFrameInfo::is_end_of_temporal_unit, Eq(false))),
+           AllOf(
+               Field(&DroppedFrameInfo::rtp_timestamp, Eq(kTimestampMs2 * 90)),
+               Field(&DroppedFrameInfo::spatial_id, Eq(1)),
+               Field(&DroppedFrameInfo::is_end_of_temporal_unit, Eq(true)))}));
+}
+
+TEST(FrameEncodeMetadataWriterTest,
+     EndOfTemporalUnitOnDroppedFrameDropsOtherLayers) {
+  const int64_t kTimestampMs1 = 100000;
+  const int64_t kTimestampMs2 = 100010;
+
+  FakeEncodedImageCallback sink;
+  FrameEncodeMetadataWriter encode_timer(CreateTestEnvironment(), &sink);
+  VideoCodec codec_settings;
+  codec_settings.codecType = kVideoCodecVP8;
+  codec_settings.numberOfSimulcastStreams = 2;
+  encode_timer.OnEncoderInit(codec_settings);
+  VideoBitrateAllocation bitrate_allocation;
+  bitrate_allocation.SetBitrate(0, 0, 500000);
+  bitrate_allocation.SetBitrate(1, 0, 500000);
+  encode_timer.OnSetRates(bitrate_allocation, 30);
+
+  VideoFrame frame1 = VideoFrame::Builder()
+                          .set_rtp_timestamp(kTimestampMs1 * 90)
+                          .set_timestamp_ms(kTimestampMs1)
+                          .set_video_frame_buffer(kFrameBuffer)
+                          .build();
+  encode_timer.OnEncodeStarted(frame1);
+
+  VideoFrame frame2 = VideoFrame::Builder()
+                          .set_rtp_timestamp(kTimestampMs2 * 90)
+                          .set_timestamp_ms(kTimestampMs2)
+                          .set_video_frame_buffer(kFrameBuffer)
+                          .build();
+  encode_timer.OnEncodeStarted(frame2);
+
+  encode_timer.OnFrameDropped(kTimestampMs2 * 90, 0, true);
+
+  // Only the explicitly dropped frame (layer 0) should get
+  // is_end_of_temporal_unit=true. The implicitly dropped frame on layer 1
+  // should be pending on layer 0 at evaluation time and evaluate to false.
+  EXPECT_THAT(
+      sink.GetDroppedFrames(),
+      ElementsAreArray(
+          {AllOf(
+               Field(&DroppedFrameInfo::rtp_timestamp, Eq(kTimestampMs1 * 90)),
+               Field(&DroppedFrameInfo::spatial_id, Eq(0)),
+               Field(&DroppedFrameInfo::is_end_of_temporal_unit, Eq(false))),
+           AllOf(
+               Field(&DroppedFrameInfo::rtp_timestamp, Eq(kTimestampMs1 * 90)),
+               Field(&DroppedFrameInfo::spatial_id, Eq(1)),
+               Field(&DroppedFrameInfo::is_end_of_temporal_unit, Eq(true))),
+           AllOf(
+               Field(&DroppedFrameInfo::rtp_timestamp, Eq(kTimestampMs2 * 90)),
+               Field(&DroppedFrameInfo::spatial_id, Eq(1)),
+               Field(&DroppedFrameInfo::is_end_of_temporal_unit, Eq(false))),
+           AllOf(
+               Field(&DroppedFrameInfo::rtp_timestamp, Eq(kTimestampMs2 * 90)),
+               Field(&DroppedFrameInfo::spatial_id, Eq(0)),
+               Field(&DroppedFrameInfo::is_end_of_temporal_unit, Eq(true)))}));
+}
+
+TEST(FrameEncodeMetadataWriterTest, DropsDifferentTimingAcrossLayers) {
+  const int64_t kTimestampMs1 = 100000;
+  const int64_t kTimestampMs2 = 100010;
+  const int64_t kTimestampMs3 = 100020;
+
+  FakeEncodedImageCallback sink;
+  FrameEncodeMetadataWriter encode_timer(CreateTestEnvironment(), &sink);
+  VideoCodec codec_settings;
+  codec_settings.codecType = kVideoCodecVP8;
+  codec_settings.numberOfSimulcastStreams = 2;
+  encode_timer.OnEncoderInit(codec_settings);
+  VideoBitrateAllocation bitrate_allocation;
+  bitrate_allocation.SetBitrate(0, 0, 500000);
+  bitrate_allocation.SetBitrate(1, 0, 500000);
+  encode_timer.OnSetRates(bitrate_allocation, 30);
+
+  VideoFrame frame1 = VideoFrame::Builder()
+                          .set_rtp_timestamp(kTimestampMs1 * 90)
+                          .set_timestamp_ms(kTimestampMs1)
+                          .set_video_frame_buffer(kFrameBuffer)
+                          .build();
+  encode_timer.OnEncodeStarted(frame1);
+
+  VideoFrame frame2 = VideoFrame::Builder()
+                          .set_rtp_timestamp(kTimestampMs2 * 90)
+                          .set_timestamp_ms(kTimestampMs2)
+                          .set_video_frame_buffer(kFrameBuffer)
+                          .build();
+  encode_timer.OnEncodeStarted(frame2);
+
+  VideoFrame frame3 = VideoFrame::Builder()
+                          .set_rtp_timestamp(kTimestampMs3 * 90)
+                          .set_timestamp_ms(kTimestampMs3)
+                          .set_video_frame_buffer(kFrameBuffer)
+                          .build();
+  encode_timer.OnEncodeStarted(frame3);
+
+  // Fill metadata for frame 3 on spatial layer 0 without having given any
+  // feedback signal for frame 1 and 2 - implying they have been dropped for
+  // spatial layer 0.
+  EncodedImage image3_s0;
+  image3_s0.capture_time_ms_ = kTimestampMs3;
+  image3_s0.SetRtpTimestamp(static_cast<uint32_t>(kTimestampMs3 * 90));
+  encode_timer.FillMetadataAndTimingInfo(0, &image3_s0);
+
+  EXPECT_THAT(
+      sink.GetDroppedFrames(),
+      ElementsAreArray(
+          {AllOf(
+               Field(&DroppedFrameInfo::rtp_timestamp, Eq(kTimestampMs1 * 90)),
+               Field(&DroppedFrameInfo::spatial_id, Eq(0)),
+               Field(&DroppedFrameInfo::is_end_of_temporal_unit, Eq(false))),
+           AllOf(
+               Field(&DroppedFrameInfo::rtp_timestamp, Eq(kTimestampMs2 * 90)),
+               Field(&DroppedFrameInfo::spatial_id, Eq(0)),
+               Field(&DroppedFrameInfo::is_end_of_temporal_unit, Eq(false)))}));
+
+  // Now explicitly drop frame 1 on spatial layer 1.
+  encode_timer.OnFrameDropped(kTimestampMs1 * 90, /*spatial_id=*/1,
+                              /*is_end_of_temporal_unit=*/false);
+
+  EXPECT_THAT(
+      sink.GetDroppedFrames(),
+      ElementsAreArray(
+          {AllOf(
+               Field(&DroppedFrameInfo::rtp_timestamp, Eq(kTimestampMs1 * 90)),
+               Field(&DroppedFrameInfo::spatial_id, Eq(0)),
+               Field(&DroppedFrameInfo::is_end_of_temporal_unit, Eq(false))),
+           AllOf(
+               Field(&DroppedFrameInfo::rtp_timestamp, Eq(kTimestampMs2 * 90)),
+               Field(&DroppedFrameInfo::spatial_id, Eq(0)),
+               Field(&DroppedFrameInfo::is_end_of_temporal_unit, Eq(false))),
+           AllOf(
+               Field(&DroppedFrameInfo::rtp_timestamp, Eq(kTimestampMs1 * 90)),
+               Field(&DroppedFrameInfo::spatial_id, Eq(1)),
+               Field(&DroppedFrameInfo::is_end_of_temporal_unit, Eq(false)))}));
+}
 }  // namespace test
 }  // namespace webrtc

From 15fd6363943a47bd9bf16df856113e98959f55df Mon Sep 17 00:00:00 2001
From: Etienne Pierre-doray <etiennep@chromium.org>
Date: Tue, 10 Feb 2026 23:08:12 +0000
Subject: [PATCH 043/472] [tracing] Migrate legacy async macros in webrtc
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Migrating the following async event macros somewhat mechanically:
```
- TRACE_EVENT(_NESTABLE)?_ASYNC_BEGIN(_WITH_TIMESTAMP)?[012](<category_group>, <name>, <id>, ...)
-> TRACE_EVENT_BEGIN(<category_group>, <name>, <track>, ...)

- TRACE_EVENT(_NESTABLE)?_ASYNC_END(_WITH_TIMESTAMP)?[012](<category_group>, <name>, <id>, ...)
-> TRACE_EVENT_END(<category_group>, <track>, ...)

- TRACE_EVENT(_NESTABLE)?_ASYNC_INSTANT(_WITH_TIMESTAMP)?[012](<category_group>, <name>, <id>, ...)
-> TRACE_EVENT_INSTANT(<category_group>, <name>, <track>, ...)
```

Bug: chromium:432427382
Change-Id: I5d25d8dfd19613bcafa98c8d7a81a4050cd6534c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/435580
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Etienne Pierre-Doray <etiennep@chromium.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46896}
---
 video/video_stream_encoder.cc | 20 ++++++++++++--------
 1 file changed, 12 insertions(+), 8 deletions(-)

diff --git a/video/video_stream_encoder.cc b/video/video_stream_encoder.cc
index ad12271019..004b70a87c 100644
--- a/video/video_stream_encoder.cc
+++ b/video/video_stream_encoder.cc
@@ -868,11 +868,13 @@ void VideoStreamEncoder::AddAdaptationResource(
   // Map any externally added resources as kCpu for the sake of stats reporting.
   // TODO(hbos): Make the manager map any unknown resources to kCpu and get rid
   // of this MapResourceToReason() call.
-  TRACE_EVENT_ASYNC_BEGIN0(
-      "webrtc", "VideoStreamEncoder::AddAdaptationResource(latency)", this);
+  TRACE_EVENT_BEGIN("webrtc",
+                    "VideoStreamEncoder::AddAdaptationResource(latency)",
+                    perfetto::Track::FromPointer(this));
   encoder_queue_->PostTask([this, resource = std::move(resource)] {
-    TRACE_EVENT_ASYNC_END0(
-        "webrtc", "VideoStreamEncoder::AddAdaptationResource(latency)", this);
+    TRACE_EVENT_END("webrtc",
+                    /* VideoStreamEncoder::AddAdaptationResource(latency) */
+                    perfetto::Track::FromPointer(this));
     RTC_DCHECK_RUN_ON(encoder_queue_.get());
     additional_resources_.push_back(resource);
     stream_resource_manager_.AddResource(resource, VideoAdaptationReason::kCpu);
@@ -1729,7 +1731,8 @@ void VideoStreamEncoder::TraceFrameDropStart() {
   RTC_DCHECK_RUN_ON(encoder_queue_.get());
   // Start trace event only on the first frame after encoder is paused.
   if (!encoder_paused_and_dropped_frame_) {
-    TRACE_EVENT_ASYNC_BEGIN0("webrtc", "EncoderPaused", this);
+    TRACE_EVENT_BEGIN("webrtc", "EncoderPaused",
+                      perfetto::Track::FromPointer(this));
   }
   encoder_paused_and_dropped_frame_ = true;
 }
@@ -1738,7 +1741,8 @@ void VideoStreamEncoder::TraceFrameDropEnd() {
   RTC_DCHECK_RUN_ON(encoder_queue_.get());
   // End trace event on first frame after encoder resumes, if frame was dropped.
   if (encoder_paused_and_dropped_frame_) {
-    TRACE_EVENT_ASYNC_END0("webrtc", "EncoderPaused", this);
+    TRACE_EVENT_END("webrtc", /* EncoderPaused */
+                    perfetto::Track::FromPointer(this));
   }
   encoder_paused_and_dropped_frame_ = false;
 }
@@ -2197,8 +2201,8 @@ void VideoStreamEncoder::EncodeVideoFrame(const VideoFrame& video_frame,
   }
   accumulated_update_rect_is_valid_ = true;
 
-  TRACE_EVENT_ASYNC_STEP_INTO0("webrtc", "Video", video_frame.render_time_ms(),
-                               "Encode");
+  TRACE_EVENT_INSTANT("webrtc", "Encode", perfetto::Track::FromPointer(this),
+                      "render_time", video_frame.render_time_ms());
 
   stream_resource_manager_.OnEncodeStarted(out_frame, time_when_posted_us);
 

From 8ca627dd9b409e9c655a086cc3976a5a99b69e86 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Fri, 13 Feb 2026 13:39:20 +0100
Subject: [PATCH 044/472] Fix potential deadlock in RTCStatsCollector during
 shutdown.

When stats gathering on the network thread is cancelled (e.g., during PC
closure), the `SafeTask` evaluating the partial results on the network
thread may be dropped without executing. This causes its closures
(including `RtpTransceiverStatsInfo` objects) to be destructed on the
network thread.

Because `RtpTransceiverStatsInfo` holds `scoped_refptr` references to
`RtpTransceiver` and `RtpReceiverInternal` (which may respectively hold
`VideoTrackProxyWithInternal`), their destruction on the network thread
can trigger a blocking call back to the signaling thread (via the
proxies), potentially leading to a deadlock if the signaling thread is
issuing a blocking call to the network thread.

This change prevents clears the object references (`transceiver` and
`receivers`) on the signaling thread before posting to the network
thread. When the task is dropped on the network thread, the subsequent
destruction of `RtpTransceiverStatsInfo` will be a no-op that avoids
cross-thread blocking.

Bug: b/483026914
Change-Id: Ifc5c317e6a26cc0c97c9ca67101ef8b0cc90ba1e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/448940
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46897}
---
 pc/rtc_stats_collector.cc | 7 +++++++
 pc/rtc_stats_collector.h  | 2 +-
 2 files changed, 8 insertions(+), 1 deletion(-)

diff --git a/pc/rtc_stats_collector.cc b/pc/rtc_stats_collector.cc
index 93752928e7..8c8a865de3 100644
--- a/pc/rtc_stats_collector.cc
+++ b/pc/rtc_stats_collector.cc
@@ -1372,6 +1372,13 @@ void RTCStatsCollector::GetStatsReportInternal(
         transport_names.insert(*info.transport_name);
     }
 
+    // Clear references to signaling thread objects (and proxies) before
+    // attempting to post to the network thread. We're done with these objects.
+    for (RtpTransceiverStatsInfo& stats : results.transceiver_stats_infos) {
+      stats.transceiver = nullptr;
+      stats.receivers.clear();
+    }
+
     network_thread_->PostTask(SafeTask(
         network_safety_, [this, transport_names = std::move(transport_names),
                           timestamp, signaling_flag = signaling_safety_,
diff --git a/pc/rtc_stats_collector.h b/pc/rtc_stats_collector.h
index 7f9d9c8d51..4276db6348 100644
--- a/pc/rtc_stats_collector.h
+++ b/pc/rtc_stats_collector.h
@@ -61,7 +61,7 @@ class RtpReceiverInternal;
 // If a BaseChannel is not available (e.g., if signaling has not started),
 // then `mid` and `transport_name` will be null.
 struct RtpTransceiverStatsInfo {
-  const scoped_refptr<RtpTransceiver> transceiver;
+  scoped_refptr<RtpTransceiver> transceiver;
   const MediaType media_type;
   const std::optional<std::string> mid;
   std::optional<std::string> transport_name;

From 6d4ca8138fc122cb5871a3958ce7fe7be0535ad3 Mon Sep 17 00:00:00 2001
From: Bjorn Terelius <terelius@webrtc.org>
Date: Fri, 13 Feb 2026 12:47:30 +0100
Subject: [PATCH 045/472] Reland "Split analyzer.cc and change CreateFooGraph
 methods to free functions"
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Bug: None
Change-Id: I889b50706b7bb5eb7303d9fc32bea996713191f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449001
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46898}
---
 rtc_tools/BUILD.gn                            |    6 +
 .../rtc_event_log_visualizer/analyze_audio.cc |   90 +-
 .../rtc_event_log_visualizer/analyze_audio.h  |   13 +
 .../rtc_event_log_visualizer/analyze_bwe.cc   | 1221 +++++++++
 .../rtc_event_log_visualizer/analyze_bwe.h    |   81 +
 .../analyze_connectivity.cc                   |  285 ++
 .../analyze_connectivity.h                    |   38 +
 .../analyze_rtp_rtcp.cc                       |  809 ++++++
 .../analyze_rtp_rtcp.h                        |  100 +
 .../rtc_event_log_visualizer/analyzer.cc      | 2344 +----------------
 rtc_tools/rtc_event_log_visualizer/analyzer.h |   17 +-
 .../analyzer_bindings.cc                      |   13 +-
 .../analyzer_common.cc                        |   64 +
 .../analyzer_common.h                         |   30 +
 rtc_tools/rtc_event_log_visualizer/main.cc    |   12 +-
 15 files changed, 2849 insertions(+), 2274 deletions(-)
 create mode 100644 rtc_tools/rtc_event_log_visualizer/analyze_bwe.cc
 create mode 100644 rtc_tools/rtc_event_log_visualizer/analyze_bwe.h
 create mode 100644 rtc_tools/rtc_event_log_visualizer/analyze_connectivity.cc
 create mode 100644 rtc_tools/rtc_event_log_visualizer/analyze_connectivity.h
 create mode 100644 rtc_tools/rtc_event_log_visualizer/analyze_rtp_rtcp.cc
 create mode 100644 rtc_tools/rtc_event_log_visualizer/analyze_rtp_rtcp.h

diff --git a/rtc_tools/BUILD.gn b/rtc_tools/BUILD.gn
index c14c29d018..24c20bbb24 100644
--- a/rtc_tools/BUILD.gn
+++ b/rtc_tools/BUILD.gn
@@ -290,6 +290,12 @@ if (!build_with_chromium) {
         "rtc_event_log_visualizer/alerts.h",
         "rtc_event_log_visualizer/analyze_audio.cc",
         "rtc_event_log_visualizer/analyze_audio.h",
+        "rtc_event_log_visualizer/analyze_bwe.cc",
+        "rtc_event_log_visualizer/analyze_bwe.h",
+        "rtc_event_log_visualizer/analyze_connectivity.cc",
+        "rtc_event_log_visualizer/analyze_connectivity.h",
+        "rtc_event_log_visualizer/analyze_rtp_rtcp.cc",
+        "rtc_event_log_visualizer/analyze_rtp_rtcp.h",
         "rtc_event_log_visualizer/analyzer.cc",
         "rtc_event_log_visualizer/analyzer.h",
         "rtc_event_log_visualizer/analyzer_common.cc",
diff --git a/rtc_tools/rtc_event_log_visualizer/analyze_audio.cc b/rtc_tools/rtc_event_log_visualizer/analyze_audio.cc
index ed3629a787..f2ec478315 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyze_audio.cc
+++ b/rtc_tools/rtc_event_log_visualizer/analyze_audio.cc
@@ -30,7 +30,10 @@
 #include "api/neteq/neteq.h"
 #include "api/scoped_refptr.h"
 #include "api/units/timestamp.h"
+#include "logging/rtc_event_log/events/logged_rtp_rtcp.h"
 #include "logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h"
+#include "logging/rtc_event_log/events/rtc_event_audio_playout.h"
+#include "logging/rtc_event_log/events/rtc_event_neteq_set_minimum_delay.h"
 #include "logging/rtc_event_log/rtc_event_log_parser.h"
 #include "modules/audio_coding/neteq/tools/audio_sink.h"
 #include "modules/audio_coding/neteq/tools/fake_decode_from_file.h"
@@ -277,8 +280,8 @@ std::unique_ptr<test::NetEqStatsGetter> CreateNetEqTestAndRun(
 
   NetEq::Config config;
   test::NetEqTest test(config, decoder_factory, codecs, /*text_log=*/nullptr,
-                       /*factory=*/nullptr, std::move(input), std::move(output),
-                       callbacks, field_trials);
+                       /*neteq_factory=*/nullptr, std::move(input),
+                       std::move(output), callbacks, field_trials);
   test.Run();
   return neteq_stats_getter;
 }
@@ -426,4 +429,87 @@ void CreateNetEqLifetimeStatsGraph(
       stats_extractor, plot_name, plot);
 }
 
+// For each SSRC, plot the time between the consecutive playouts.
+void CreatePlayoutGraph(const ParsedRtcEventLog& parsed_log,
+                        const AnalyzerConfig& config,
+                        Plot* plot) {
+  for (const auto& playout_stream : parsed_log.audio_playout_events()) {
+    uint32_t ssrc = playout_stream.first;
+    if (!MatchingSsrc(ssrc, config.desired_ssrc_))
+      continue;
+    std::optional<int64_t> last_playout_ms;
+    TimeSeries time_series(SsrcToString(ssrc), LineStyle::kBar);
+    for (const auto& playout_event : playout_stream.second) {
+      float x = config.GetCallTimeSec(playout_event.log_time());
+      int64_t playout_time_ms = playout_event.log_time_ms();
+      // If there were no previous playouts, place the point on the x-axis.
+      float y = playout_time_ms - last_playout_ms.value_or(playout_time_ms);
+      time_series.points.push_back(TimeSeriesPoint(x, y));
+      last_playout_ms.emplace(playout_time_ms);
+    }
+    plot->AppendTimeSeries(std::move(time_series));
+  }
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Time since last playout (ms)", kBottomMargin,
+                          kTopMargin);
+  plot->SetTitle("Audio playout");
+}
+
+void CreateNetEqSetMinimumDelay(const ParsedRtcEventLog& parsed_log,
+                                const AnalyzerConfig& config,
+                                Plot* plot) {
+  for (const auto& playout_stream :
+       parsed_log.neteq_set_minimum_delay_events()) {
+    uint32_t ssrc = playout_stream.first;
+    if (!MatchingSsrc(ssrc, config.desired_ssrc_))
+      continue;
+
+    TimeSeries time_series(SsrcToString(ssrc), LineStyle::kStep,
+                           PointStyle::kHighlight);
+    for (const auto& event : playout_stream.second) {
+      float x = config.GetCallTimeSec(event.log_time());
+      float y = event.minimum_delay_ms;
+      time_series.points.push_back(TimeSeriesPoint(x, y));
+    }
+    plot->AppendTimeSeries(std::move(time_series));
+  }
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1000, "Minimum Delay (ms)", kBottomMargin,
+                          kTopMargin);
+  plot->SetTitle("Set Minimum Delay");
+}
+
+// For audio SSRCs, plot the audio level.
+void CreateAudioLevelGraph(const ParsedRtcEventLog& parsed_log,
+                           const AnalyzerConfig& config,
+                           PacketDirection direction,
+                           Plot* plot) {
+  for (const auto& stream : parsed_log.rtp_packets_by_ssrc(direction)) {
+    if (!IsAudioSsrc(parsed_log, direction, stream.ssrc))
+      continue;
+    TimeSeries time_series(GetStreamName(parsed_log, direction, stream.ssrc),
+                           LineStyle::kLine);
+    for (auto& packet : stream.packet_view) {
+      if (packet.header.extension.audio_level()) {
+        float x = config.GetCallTimeSec(packet.log_time());
+        // The audio level is stored in -dBov (so e.g. -10 dBov is stored as 10)
+        // Here we convert it to dBov.
+        float y =
+            static_cast<float>(-packet.header.extension.audio_level()->level());
+        time_series.points.emplace_back(TimeSeriesPoint(x, y));
+      }
+    }
+    plot->AppendTimeSeries(std::move(time_series));
+  }
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetYAxis(-127, 0, "Audio level (dBov)", kBottomMargin, kTopMargin);
+  plot->SetTitle(GetDirectionAsString(direction) + " audio level");
+}
+
 }  // namespace webrtc
diff --git a/rtc_tools/rtc_event_log_visualizer/analyze_audio.h b/rtc_tools/rtc_event_log_visualizer/analyze_audio.h
index 094f9d91b6..55040ae39f 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyze_audio.h
+++ b/rtc_tools/rtc_event_log_visualizer/analyze_audio.h
@@ -45,6 +45,19 @@ void CreateAudioEncoderNumChannelsGraph(const ParsedRtcEventLog& parsed_log,
                                         const AnalyzerConfig& config,
                                         Plot* plot);
 
+void CreatePlayoutGraph(const ParsedRtcEventLog& parsed_log,
+                        const AnalyzerConfig& config,
+                        Plot* plot);
+
+void CreateNetEqSetMinimumDelay(const ParsedRtcEventLog& parsed_log,
+                                const AnalyzerConfig& config,
+                                Plot* plot);
+
+void CreateAudioLevelGraph(const ParsedRtcEventLog& parsed_log,
+                           const AnalyzerConfig& config,
+                           PacketDirection direction,
+                           Plot* plot);
+
 using NetEqStatsGetterMap =
     std::map<uint32_t, std::unique_ptr<test::NetEqStatsGetter>>;
 NetEqStatsGetterMap SimulateNetEq(const ParsedRtcEventLog& parsed_log,
diff --git a/rtc_tools/rtc_event_log_visualizer/analyze_bwe.cc b/rtc_tools/rtc_event_log_visualizer/analyze_bwe.cc
new file mode 100644
index 0000000000..0480256576
--- /dev/null
+++ b/rtc_tools/rtc_event_log_visualizer/analyze_bwe.cc
@@ -0,0 +1,1221 @@
+/*
+ *  Copyright (c) 2026 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "rtc_tools/rtc_event_log_visualizer/analyze_bwe.h"
+
+#include <algorithm>
+#include <cmath>
+#include <cstddef>
+#include <cstdint>
+#include <limits>
+#include <map>
+#include <memory>
+#include <optional>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/algorithm/container.h"
+#include "absl/functional/bind_front.h"
+#include "absl/strings/string_view.h"
+#include "api/environment/environment.h"
+#include "api/environment/environment_factory.h"
+#include "api/field_trials.h"
+#include "api/function_view.h"
+#include "api/media_types.h"
+#include "api/rtp_headers.h"
+#include "api/transport/bandwidth_usage.h"
+#include "api/transport/goog_cc_factory.h"
+#include "api/transport/network_control.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "logging/rtc_event_log/events/logged_rtp_rtcp.h"
+#include "logging/rtc_event_log/rtc_event_log_parser.h"
+#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.h"
+#include "modules/congestion_controller/include/receive_side_congestion_controller.h"
+#include "modules/congestion_controller/rtp/transport_feedback_adapter.h"
+#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/remb.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/network/sent_packet.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
+#include "rtc_base/rate_statistics.h"
+#include "rtc_tools/rtc_event_log_visualizer/analyzer_common.h"
+#include "rtc_tools/rtc_event_log_visualizer/log_scream_simulation.h"
+#include "rtc_tools/rtc_event_log_visualizer/log_simulation.h"
+#include "rtc_tools/rtc_event_log_visualizer/plot_base.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+namespace {
+
+double AbsSendTimeToMicroseconds(int64_t abs_send_time) {
+  // The timestamp is a fixed point representation with 6 bits for seconds
+  // and 18 bits for fractions of a second. Thus, we divide by 2^18 to get the
+  // time in seconds and then multiply by kNumMicrosecsPerSec to convert to
+  // microseconds.
+  static constexpr double kTimestampToMicroSec =
+      static_cast<double>(kNumMicrosecsPerSec) / static_cast<double>(1ul << 18);
+  return abs_send_time * kTimestampToMicroSec;
+}
+
+// This is much more reliable for outgoing streams than for incoming streams.
+template <typename RtpPacketContainer>
+std::optional<uint32_t> EstimateRtpClockFrequency(
+    const RtpPacketContainer& packets,
+    int64_t end_time_us) {
+  RTC_CHECK(packets.size() >= 2);
+  SeqNumUnwrapper<uint32_t> unwrapper;
+  int64_t first_rtp_timestamp =
+      unwrapper.Unwrap(packets[0].rtp.header.timestamp);
+  int64_t first_log_timestamp = packets[0].log_time_us();
+  int64_t last_rtp_timestamp = first_rtp_timestamp;
+  int64_t last_log_timestamp = first_log_timestamp;
+  for (size_t i = 1; i < packets.size(); i++) {
+    if (packets[i].log_time_us() > end_time_us)
+      break;
+    last_rtp_timestamp = unwrapper.Unwrap(packets[i].rtp.header.timestamp);
+    last_log_timestamp = packets[i].log_time_us();
+  }
+  if (last_log_timestamp - first_log_timestamp < kNumMicrosecsPerSec) {
+    RTC_LOG(LS_WARNING)
+        << "Failed to estimate RTP clock frequency: Stream too short. ("
+        << packets.size() << " packets, "
+        << last_log_timestamp - first_log_timestamp << " us)";
+    return std::nullopt;
+  }
+  double duration =
+      static_cast<double>(last_log_timestamp - first_log_timestamp) /
+      kNumMicrosecsPerSec;
+  double estimated_frequency =
+      (last_rtp_timestamp - first_rtp_timestamp) / duration;
+  for (uint32_t f : {8000, 16000, 32000, 48000, 90000}) {
+    if (std::fabs(estimated_frequency - f) < 0.15 * f) {
+      return f;
+    }
+  }
+  RTC_LOG(LS_WARNING) << "Failed to estimate RTP clock frequency: Estimate "
+                      << estimated_frequency
+                      << " not close to any standard RTP frequency."
+                      << " Last timestamp " << last_rtp_timestamp
+                      << " first timestamp " << first_rtp_timestamp;
+  return std::nullopt;
+}
+
+std::optional<double> NetworkDelayDiff_AbsSendTime(
+    const LoggedRtpPacketIncoming& old_packet,
+    const LoggedRtpPacketIncoming& new_packet) {
+  if (old_packet.rtp.header.extension.hasAbsoluteSendTime &&
+      new_packet.rtp.header.extension.hasAbsoluteSendTime) {
+    int64_t send_time_diff = WrappingDifference(
+        new_packet.rtp.header.extension.absoluteSendTime,
+        old_packet.rtp.header.extension.absoluteSendTime, 1ul << 24);
+    int64_t recv_time_diff =
+        new_packet.log_time_us() - old_packet.log_time_us();
+    double delay_change_us =
+        recv_time_diff - AbsSendTimeToMicroseconds(send_time_diff);
+    return delay_change_us / 1000;
+  } else {
+    return std::nullopt;
+  }
+}
+
+std::optional<double> NetworkDelayDiff_CaptureTime(
+    const LoggedRtpPacketIncoming& old_packet,
+    const LoggedRtpPacketIncoming& new_packet,
+    const double sample_rate) {
+  int64_t send_time_diff =
+      WrappingDifference(new_packet.rtp.header.timestamp,
+                         old_packet.rtp.header.timestamp, 1ull << 32);
+  int64_t recv_time_diff = new_packet.log_time_us() - old_packet.log_time_us();
+
+  double delay_change =
+      static_cast<double>(recv_time_diff) / 1000 -
+      static_cast<double>(send_time_diff) / sample_rate * 1000;
+  if (delay_change < -10000 || 10000 < delay_change) {
+    RTC_LOG(LS_WARNING) << "Very large delay change. Timestamps correct?";
+    RTC_LOG(LS_WARNING) << "Old capture time "
+                        << old_packet.rtp.header.timestamp << ", received time "
+                        << old_packet.log_time_us();
+    RTC_LOG(LS_WARNING) << "New capture time "
+                        << new_packet.rtp.header.timestamp << ", received time "
+                        << new_packet.log_time_us();
+    RTC_LOG(LS_WARNING) << "Receive time difference " << recv_time_diff << " = "
+                        << static_cast<double>(recv_time_diff) /
+                               kNumMicrosecsPerSec
+                        << "s";
+    RTC_LOG(LS_WARNING) << "Send time difference " << send_time_diff << " = "
+                        << static_cast<double>(send_time_diff) / sample_rate
+                        << "s";
+  }
+  return delay_change;
+}
+
+struct FakeExtensionSmall {
+  static constexpr RTPExtensionType kId = kRtpExtensionMid;
+  static constexpr absl::string_view Uri() { return "fake-extension-small"; }
+};
+struct FakeExtensionLarge {
+  static constexpr RTPExtensionType kId = kRtpExtensionRtpStreamId;
+  static constexpr absl::string_view Uri() { return "fake-extension-large"; }
+};
+
+RtpPacketReceived RtpPacketForBWEFromHeader(const RTPHeader& header) {
+  RtpHeaderExtensionMap rtp_header_extensions(/*extmap_allow_mixed=*/true);
+  // ReceiveSideCongestionController doesn't need to know extensions ids as
+  // long as it able to get extensions by type. So any ids would work here.
+  rtp_header_extensions.Register<TransmissionOffset>(1);
+  rtp_header_extensions.Register<AbsoluteSendTime>(2);
+  rtp_header_extensions.Register<TransportSequenceNumber>(3);
+  rtp_header_extensions.Register<FakeExtensionSmall>(4);
+  // Use id > 14 to force two byte header per rtp header when this one is used.
+  rtp_header_extensions.Register<FakeExtensionLarge>(16);
+
+  RtpPacketReceived rtp_packet(&rtp_header_extensions);
+  // Set only fields that might be relevant for the bandwidth estimatior.
+  rtp_packet.SetSsrc(header.ssrc);
+  rtp_packet.SetTimestamp(header.timestamp);
+  size_t num_bwe_extensions = 0;
+  if (header.extension.hasTransmissionTimeOffset) {
+    rtp_packet.SetExtension<TransmissionOffset>(
+        header.extension.transmissionTimeOffset);
+    ++num_bwe_extensions;
+  }
+  if (header.extension.hasAbsoluteSendTime) {
+    rtp_packet.SetExtension<AbsoluteSendTime>(
+        header.extension.absoluteSendTime);
+    ++num_bwe_extensions;
+  }
+  if (header.extension.hasTransportSequenceNumber) {
+    rtp_packet.SetExtension<TransportSequenceNumber>(
+        header.extension.transportSequenceNumber);
+    ++num_bwe_extensions;
+  }
+
+  // All parts of the RTP header are 32bit aligned.
+  RTC_CHECK_EQ(header.headerLength % 4, 0);
+
+  // Original packet could have more extensions, there could be csrcs that are
+  // not propagated by the rtc event log, i.e. logged header size might be
+  // larger that rtp_packet.header_size(). Increase it by setting an extra fake
+  // extension.
+  RTC_CHECK_GE(header.headerLength, rtp_packet.headers_size());
+  size_t bytes_to_add = header.headerLength - rtp_packet.headers_size();
+  if (bytes_to_add > 0) {
+    if (bytes_to_add <= 16) {
+      // one-byte header rtp header extension allows to add up to 16 bytes.
+      rtp_packet.AllocateExtension(FakeExtensionSmall::kId, bytes_to_add - 1);
+    } else {
+      // two-byte header rtp header extension would also add one byte per
+      // already set extension.
+      rtp_packet.AllocateExtension(FakeExtensionLarge::kId,
+                                   bytes_to_add - 2 - num_bwe_extensions);
+    }
+  }
+  RTC_CHECK_EQ(rtp_packet.headers_size(), header.headerLength);
+
+  return rtp_packet;
+}
+
+}  // namespace
+
+class BitrateObserver : public RemoteBitrateObserver {
+ public:
+  BitrateObserver() : last_bitrate_bps_(0), bitrate_updated_(false) {}
+
+  void Update(NetworkControlUpdate update) {
+    if (update.target_rate) {
+      last_bitrate_bps_ = update.target_rate->target_rate.bps();
+      bitrate_updated_ = true;
+    }
+  }
+
+  void OnReceiveBitrateChanged(const std::vector<uint32_t>& ssrcs,
+                               uint32_t bitrate) override {}
+
+  uint32_t last_bitrate_bps() const { return last_bitrate_bps_; }
+  bool GetAndResetBitrateUpdated() {
+    bool bitrate_updated = bitrate_updated_;
+    bitrate_updated_ = false;
+    return bitrate_updated;
+  }
+
+ private:
+  uint32_t last_bitrate_bps_;
+  bool bitrate_updated_;
+};
+
+void CreateIncomingDelayGraph(const ParsedRtcEventLog& parsed_log,
+                              const AnalyzerConfig& config,
+                              Plot* plot) {
+  for (const auto& stream : parsed_log.incoming_rtp_packets_by_ssrc()) {
+    // Filter on SSRC.
+    if (!MatchingSsrc(stream.ssrc, config.desired_ssrc_) ||
+        IsRtxSsrc(parsed_log, kIncomingPacket, stream.ssrc)) {
+      continue;
+    }
+
+    const std::vector<LoggedRtpPacketIncoming>& packets =
+        stream.incoming_packets;
+    if (packets.size() < 100) {
+      RTC_LOG(LS_WARNING) << "Can't estimate the RTP clock frequency with "
+                          << packets.size() << " packets in the stream.";
+      continue;
+    }
+    int64_t segment_end_us = parsed_log.first_log_segment().stop_time_us();
+    std::optional<uint32_t> estimated_frequency =
+        EstimateRtpClockFrequency(packets, segment_end_us);
+    if (!estimated_frequency)
+      continue;
+    const double frequency_hz = *estimated_frequency;
+    if (IsVideoSsrc(parsed_log, kIncomingPacket, stream.ssrc) &&
+        frequency_hz != 90000) {
+      RTC_LOG(LS_WARNING)
+          << "Video stream should use a 90 kHz clock but appears to use "
+          << frequency_hz / 1000 << ". Discarding.";
+      continue;
+    }
+
+    auto ToCallTime = [&config](const LoggedRtpPacketIncoming& packet) {
+      return config.GetCallTimeSec(packet.log_time());
+    };
+    auto ToNetworkDelay = [frequency_hz](
+                              const LoggedRtpPacketIncoming& old_packet,
+                              const LoggedRtpPacketIncoming& new_packet) {
+      return NetworkDelayDiff_CaptureTime(old_packet, new_packet, frequency_hz);
+    };
+
+    TimeSeries capture_time_data(
+        GetStreamName(parsed_log, kIncomingPacket, stream.ssrc) +
+            " capture-time",
+        LineStyle::kLine);
+    AccumulatePairs<LoggedRtpPacketIncoming, double>(
+        ToCallTime, ToNetworkDelay, packets, &capture_time_data);
+    plot->AppendTimeSeries(std::move(capture_time_data));
+
+    TimeSeries send_time_data(
+        GetStreamName(parsed_log, kIncomingPacket, stream.ssrc) +
+            " abs-send-time",
+        LineStyle::kLine);
+    AccumulatePairs<LoggedRtpPacketIncoming, double>(
+        ToCallTime, NetworkDelayDiff_AbsSendTime, packets, &send_time_data);
+    plot->AppendTimeSeriesIfNotEmpty(std::move(send_time_data));
+  }
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Delay (ms)", kBottomMargin, kTopMargin);
+  plot->SetTitle("Incoming network delay (relative to first packet)");
+}
+
+// Plot the fraction of packets lost (as perceived by the loss-based BWE).
+void CreateFractionLossGraph(const ParsedRtcEventLog& parsed_log,
+                             const AnalyzerConfig& config,
+                             Plot* plot) {
+  TimeSeries time_series("Fraction lost", LineStyle::kLine,
+                         PointStyle::kHighlight);
+  for (auto& bwe_update : parsed_log.bwe_loss_updates()) {
+    float x = config.GetCallTimeSec(bwe_update.log_time());
+    float y = static_cast<float>(bwe_update.fraction_lost) / 255 * 100;
+    time_series.points.emplace_back(x, y);
+  }
+
+  plot->AppendTimeSeries(std::move(time_series));
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 10, "Loss rate (in %)", kBottomMargin, kTopMargin);
+  plot->SetTitle("Outgoing packet loss (as reported by BWE)");
+}
+
+// Plot the total bandwidth used by all RTP streams.
+void CreateTotalIncomingBitrateGraph(const ParsedRtcEventLog& parsed_log,
+                                     const AnalyzerConfig& config,
+                                     Plot* plot) {
+  // TODO(terelius): This could be provided by the parser.
+  std::multimap<Timestamp, size_t> packets_in_order;
+  for (const auto& stream : parsed_log.incoming_rtp_packets_by_ssrc()) {
+    for (const LoggedRtpPacketIncoming& packet : stream.incoming_packets)
+      packets_in_order.insert(
+          std::make_pair(packet.rtp.log_time(), packet.rtp.total_length));
+  }
+
+  auto window_begin = packets_in_order.begin();
+  auto window_end = packets_in_order.begin();
+  size_t bytes_in_window = 0;
+
+  if (!packets_in_order.empty()) {
+    // Calculate a moving average of the bitrate and store in a TimeSeries.
+    TimeSeries bitrate_series("Bitrate", LineStyle::kLine);
+    for (Timestamp time = config.begin_time_;
+         time < config.end_time_ + config.step_; time += config.step_) {
+      while (window_end != packets_in_order.end() && window_end->first < time) {
+        bytes_in_window += window_end->second;
+        ++window_end;
+      }
+      while (window_begin != packets_in_order.end() &&
+             window_begin->first < time - config.window_duration_) {
+        RTC_DCHECK_LE(window_begin->second, bytes_in_window);
+        bytes_in_window -= window_begin->second;
+        ++window_begin;
+      }
+      float window_duration_in_seconds =
+          static_cast<float>(config.window_duration_.us()) /
+          kNumMicrosecsPerSec;
+      float x = config.GetCallTimeSec(time);
+      float y = bytes_in_window * 8 / window_duration_in_seconds / 1000;
+      bitrate_series.points.emplace_back(x, y);
+    }
+    plot->AppendTimeSeries(std::move(bitrate_series));
+  }
+
+  // Overlay the outgoing REMB over incoming bitrate.
+  TimeSeries remb_series("Remb", LineStyle::kStep);
+  for (const auto& rtcp : parsed_log.rembs(kOutgoingPacket)) {
+    float x = config.GetCallTimeSec(rtcp.log_time());
+    float y = static_cast<float>(rtcp.remb.bitrate_bps()) / 1000;
+    remb_series.points.emplace_back(x, y);
+  }
+  plot->AppendTimeSeriesIfNotEmpty(std::move(remb_series));
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin);
+  plot->SetTitle("Incoming RTP bitrate");
+}
+
+// Plot the total bandwidth used by all RTP streams.
+void CreateTotalOutgoingBitrateGraph(const ParsedRtcEventLog& parsed_log,
+                                     const AnalyzerConfig& config,
+                                     Plot* plot,
+                                     bool show_detector_state,
+                                     bool show_alr_state,
+                                     bool show_link_capacity) {
+  // TODO(terelius): This could be provided by the parser.
+  std::multimap<Timestamp, size_t> packets_in_order;
+  for (const auto& stream : parsed_log.outgoing_rtp_packets_by_ssrc()) {
+    for (const LoggedRtpPacketOutgoing& packet : stream.outgoing_packets)
+      packets_in_order.insert(
+          std::make_pair(packet.rtp.log_time(), packet.rtp.total_length));
+  }
+
+  auto window_begin = packets_in_order.begin();
+  auto window_end = packets_in_order.begin();
+  size_t bytes_in_window = 0;
+
+  if (!packets_in_order.empty()) {
+    // Calculate a moving average of the bitrate and store in a TimeSeries.
+    TimeSeries bitrate_series("Bitrate", LineStyle::kLine);
+    for (Timestamp time = config.begin_time_;
+         time < config.end_time_ + config.step_; time += config.step_) {
+      while (window_end != packets_in_order.end() && window_end->first < time) {
+        bytes_in_window += window_end->second;
+        ++window_end;
+      }
+      while (window_begin != packets_in_order.end() &&
+             window_begin->first < time - config.window_duration_) {
+        RTC_DCHECK_LE(window_begin->second, bytes_in_window);
+        bytes_in_window -= window_begin->second;
+        ++window_begin;
+      }
+      float window_duration_in_seconds =
+          static_cast<float>(config.window_duration_.us()) /
+          kNumMicrosecsPerSec;
+      float x = config.GetCallTimeSec(time);
+      float y = bytes_in_window * 8 / window_duration_in_seconds / 1000;
+      bitrate_series.points.emplace_back(x, y);
+    }
+    plot->AppendTimeSeries(std::move(bitrate_series));
+  }
+
+  // Overlay the send-side bandwidth estimate over the outgoing bitrate.
+  TimeSeries loss_series("Loss-based estimate", LineStyle::kStep);
+  for (auto& loss_update : parsed_log.bwe_loss_updates()) {
+    float x = config.GetCallTimeSec(loss_update.log_time());
+    float y = static_cast<float>(loss_update.bitrate_bps) / 1000;
+    loss_series.points.emplace_back(x, y);
+  }
+
+  TimeSeries link_capacity_lower_series("Link-capacity-lower",
+                                        LineStyle::kStep);
+  TimeSeries link_capacity_upper_series("Link-capacity-upper",
+                                        LineStyle::kStep);
+  for (auto& remote_estimate_event : parsed_log.remote_estimate_events()) {
+    float x = config.GetCallTimeSec(remote_estimate_event.log_time());
+    if (remote_estimate_event.link_capacity_lower.has_value()) {
+      float link_capacity_lower = static_cast<float>(
+          remote_estimate_event.link_capacity_lower.value().kbps());
+      link_capacity_lower_series.points.emplace_back(x, link_capacity_lower);
+    }
+    if (remote_estimate_event.link_capacity_upper.has_value()) {
+      float link_capacity_upper = static_cast<float>(
+          remote_estimate_event.link_capacity_upper.value().kbps());
+      link_capacity_upper_series.points.emplace_back(x, link_capacity_upper);
+    }
+  }
+
+  TimeSeries delay_series("Delay-based estimate", LineStyle::kStep);
+  IntervalSeries overusing_series("Overusing", "#ff8e82",
+                                  IntervalSeries::kHorizontal);
+  IntervalSeries underusing_series("Underusing", "#5092fc",
+                                   IntervalSeries::kHorizontal);
+  IntervalSeries normal_series("Normal", "#c4ffc4",
+                               IntervalSeries::kHorizontal);
+  IntervalSeries* last_series = &normal_series;
+  float last_detector_switch = 0.0;
+
+  BandwidthUsage last_detector_state = BandwidthUsage::kBwNormal;
+
+  for (auto& delay_update : parsed_log.bwe_delay_updates()) {
+    float x = config.GetCallTimeSec(delay_update.log_time());
+    float y = static_cast<float>(delay_update.bitrate_bps) / 1000;
+
+    if (last_detector_state != delay_update.detector_state) {
+      last_series->intervals.emplace_back(last_detector_switch, x);
+      last_detector_state = delay_update.detector_state;
+      last_detector_switch = x;
+
+      switch (delay_update.detector_state) {
+        case BandwidthUsage::kBwNormal:
+          last_series = &normal_series;
+          break;
+        case BandwidthUsage::kBwUnderusing:
+          last_series = &underusing_series;
+          break;
+        case BandwidthUsage::kBwOverusing:
+          last_series = &overusing_series;
+          break;
+        case BandwidthUsage::kLast:
+          RTC_DCHECK_NOTREACHED();
+      }
+    }
+
+    delay_series.points.emplace_back(x, y);
+  }
+
+  RTC_CHECK(last_series);
+  last_series->intervals.emplace_back(last_detector_switch,
+                                      config.CallEndTimeSec());
+
+  TimeSeries scream_series("Scream target rate", LineStyle::kStep);
+  for (auto& scream_update : parsed_log.bwe_scream_updates()) {
+    float x = config.GetCallTimeSec(scream_update.log_time());
+    float y = static_cast<float>(scream_update.target_rate.kbps());
+    scream_series.points.emplace_back(x, y);
+  }
+
+  TimeSeries created_series("Probe cluster created.", LineStyle::kNone,
+                            PointStyle::kHighlight);
+  for (auto& cluster : parsed_log.bwe_probe_cluster_created_events()) {
+    float x = config.GetCallTimeSec(cluster.log_time());
+    float y = static_cast<float>(cluster.bitrate_bps) / 1000;
+    created_series.points.emplace_back(x, y);
+  }
+
+  TimeSeries result_series("Probing results.", LineStyle::kNone,
+                           PointStyle::kHighlight);
+  for (auto& result : parsed_log.bwe_probe_success_events()) {
+    float x = config.GetCallTimeSec(result.log_time());
+    float y = static_cast<float>(result.bitrate_bps) / 1000;
+    result_series.points.emplace_back(x, y);
+  }
+
+  TimeSeries probe_failures_series("Probe failed", LineStyle::kNone,
+                                   PointStyle::kHighlight);
+  for (auto& failure : parsed_log.bwe_probe_failure_events()) {
+    float x = config.GetCallTimeSec(failure.log_time());
+    probe_failures_series.points.emplace_back(x, 0);
+  }
+
+  IntervalSeries alr_state("ALR", "#555555", IntervalSeries::kHorizontal);
+  bool previously_in_alr = false;
+  Timestamp alr_start = Timestamp::Zero();
+  for (auto& alr : parsed_log.alr_state_events()) {
+    float y = config.GetCallTimeSec(alr.log_time());
+    if (!previously_in_alr && alr.in_alr) {
+      alr_start = alr.log_time();
+      previously_in_alr = true;
+    } else if (previously_in_alr && !alr.in_alr) {
+      float x = config.GetCallTimeSec(alr_start);
+      alr_state.intervals.emplace_back(x, y);
+      previously_in_alr = false;
+    }
+  }
+
+  if (previously_in_alr) {
+    float x = config.GetCallTimeSec(alr_start);
+    float y = config.GetCallTimeSec(config.end_time_);
+    alr_state.intervals.emplace_back(x, y);
+  }
+
+  if (show_detector_state) {
+    plot->AppendIntervalSeries(std::move(overusing_series));
+    plot->AppendIntervalSeries(std::move(underusing_series));
+    plot->AppendIntervalSeries(std::move(normal_series));
+  }
+
+  if (show_alr_state) {
+    plot->AppendIntervalSeries(std::move(alr_state));
+  }
+
+  if (show_link_capacity) {
+    plot->AppendTimeSeriesIfNotEmpty(std::move(link_capacity_lower_series));
+    plot->AppendTimeSeriesIfNotEmpty(std::move(link_capacity_upper_series));
+  }
+
+  plot->AppendTimeSeries(std::move(loss_series));
+  plot->AppendTimeSeriesIfNotEmpty(std::move(probe_failures_series));
+  plot->AppendTimeSeries(std::move(delay_series));
+  plot->AppendTimeSeriesIfNotEmpty(std::move(scream_series));
+  plot->AppendTimeSeries(std::move(created_series));
+  plot->AppendTimeSeries(std::move(result_series));
+
+  // Overlay the incoming REMB over the outgoing bitrate.
+  TimeSeries remb_series("Remb", LineStyle::kStep);
+  for (const auto& rtcp : parsed_log.rembs(kIncomingPacket)) {
+    float x = config.GetCallTimeSec(rtcp.log_time());
+    float y = static_cast<float>(rtcp.remb.bitrate_bps()) / 1000;
+    remb_series.points.emplace_back(x, y);
+  }
+  plot->AppendTimeSeriesIfNotEmpty(std::move(remb_series));
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin);
+  plot->SetTitle("Outgoing RTP bitrate");
+}
+
+void CreateGoogCcSimulationGraph(const ParsedRtcEventLog& parsed_log,
+                                 const AnalyzerConfig& config,
+                                 Plot* plot) {
+  TimeSeries target_rates("Simulated target rate", LineStyle::kStep,
+                          PointStyle::kHighlight);
+  TimeSeries delay_based("Logged delay-based estimate", LineStyle::kStep,
+                         PointStyle::kHighlight);
+  TimeSeries loss_based("Logged loss-based estimate", LineStyle::kStep,
+                        PointStyle::kHighlight);
+  TimeSeries probe_results("Logged probe success", LineStyle::kNone,
+                           PointStyle::kHighlight);
+
+  LogBasedNetworkControllerSimulation simulation(
+      config.env_, std::make_unique<GoogCcNetworkControllerFactory>(),
+      [&](const NetworkControlUpdate& update, Timestamp at_time) {
+        if (update.target_rate) {
+          target_rates.points.emplace_back(
+              config.GetCallTimeSec(at_time),
+              update.target_rate->target_rate.kbps<float>());
+        }
+      });
+
+  simulation.ProcessEventsInLog(parsed_log);
+  for (const auto& logged : parsed_log.bwe_delay_updates())
+    delay_based.points.emplace_back(config.GetCallTimeSec(logged.log_time()),
+                                    logged.bitrate_bps / 1000);
+  for (const auto& logged : parsed_log.bwe_probe_success_events())
+    probe_results.points.emplace_back(config.GetCallTimeSec(logged.log_time()),
+                                      logged.bitrate_bps / 1000);
+  for (const auto& logged : parsed_log.bwe_loss_updates())
+    loss_based.points.emplace_back(config.GetCallTimeSec(logged.log_time()),
+                                   logged.bitrate_bps / 1000);
+
+  plot->AppendTimeSeries(std::move(delay_based));
+  plot->AppendTimeSeries(std::move(loss_based));
+  plot->AppendTimeSeries(std::move(probe_results));
+  plot->AppendTimeSeries(std::move(target_rates));
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 10, "Bitrate (kbps)", kBottomMargin, kTopMargin);
+  plot->SetTitle("Simulated BWE behavior");
+}
+
+void CreateScreamSimulationBitrateGraph(const ParsedRtcEventLog& parsed_log,
+                                        const AnalyzerConfig& config,
+                                        Plot* plot) {
+  TimeSeries target_rate_series("Target rate", LineStyle::kStep);
+  TimeSeries pacing_rate_series("Pacing rate", LineStyle::kStep);
+  TimeSeries send_rate_series("Send rate", LineStyle::kStep);
+
+  LogScreamSimulation simulation({.rate_window = config.window_duration_},
+                                 config.env_);
+  simulation.ProcessEventsInLog(parsed_log);
+
+  for (const LogScreamSimulation::State& state : simulation.updates()) {
+    target_rate_series.points.emplace_back(config.GetCallTimeSec(state.time),
+                                           state.target_rate.bps() / 1000);
+    pacing_rate_series.points.emplace_back(config.GetCallTimeSec(state.time),
+                                           state.pacing_rate.bps() / 1000);
+    send_rate_series.points.emplace_back(config.GetCallTimeSec(state.time),
+                                         state.send_rate.bps() / 1000);
+  }
+  plot->AppendTimeSeries(std::move(target_rate_series));
+  plot->AppendTimeSeries(std::move(pacing_rate_series));
+  plot->AppendTimeSeries(std::move(send_rate_series));
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 100, "Kbps", kBottomMargin, kTopMargin);
+  plot->SetTitle("Simulated Scream rates");
+}
+
+void CreateScreamSimulationRefWindowGraph(const ParsedRtcEventLog& parsed_log,
+                                          const AnalyzerConfig& config,
+                                          Plot* plot) {
+  using SendWindowUsage = LogScreamSimulation::State::SendWindowUsage;
+  TimeSeries ref_window_series("RefWindow", LineStyle::kStep);
+  TimeSeries ref_window_i_series("RefWindowI", LineStyle::kStep);
+  TimeSeries max_data_in_flight("Max allowed data in flight", LineStyle::kStep);
+  TimeSeries data_in_flight("Data in flight", LineStyle::kStep);
+  IntervalSeries send_window_above_max_series(
+      "Data in flight > Max allowed", "#ff8e82", IntervalSeries::kHorizontal);
+  IntervalSeries send_window_below_ref_window_series(
+      "Data in flight < RefWindow", "#c5dff2", IntervalSeries::kHorizontal);
+  IntervalSeries send_window_above_ref_window_series(
+      "Data in flight >= RefWindow", "#b9fad8", IntervalSeries::kHorizontal);
+  IntervalSeries* last_series = &send_window_below_ref_window_series;
+
+  LogScreamSimulation simulation({.rate_window = config.window_duration_},
+                                 config.env_);
+  simulation.ProcessEventsInLog(parsed_log);
+  if (simulation.updates().empty()) {
+    RTC_LOG(LS_ERROR) << "Empty simulation.";
+    return;
+  }
+
+  float send_window_state_switch =
+      config.GetCallTimeSec(simulation.updates().front().time);
+  SendWindowUsage send_window_usage = SendWindowUsage::kBelowRefWindow;
+
+  for (const LogScreamSimulation::State& state : simulation.updates()) {
+    ref_window_series.points.emplace_back(config.GetCallTimeSec(state.time),
+                                          state.ref_window.bytes());
+    ref_window_i_series.points.emplace_back(config.GetCallTimeSec(state.time),
+                                            state.ref_window_i.bytes());
+    max_data_in_flight.points.emplace_back(config.GetCallTimeSec(state.time),
+                                           state.max_data_in_flight.bytes());
+    data_in_flight.points.emplace_back(config.GetCallTimeSec(state.time),
+                                       state.data_in_flight.bytes());
+    if (state.send_window_usage != send_window_usage) {
+      last_series->intervals.emplace_back(send_window_state_switch,
+                                          config.GetCallTimeSec(state.time));
+      send_window_usage = state.send_window_usage;
+      send_window_state_switch = config.GetCallTimeSec(state.time);
+      switch (send_window_usage) {
+        case SendWindowUsage::kAboveRefWindow:
+          last_series = &send_window_above_ref_window_series;
+          break;
+        case SendWindowUsage::kBelowRefWindow:
+          last_series = &send_window_below_ref_window_series;
+          break;
+        case SendWindowUsage::kAboveScreamMax:
+          last_series = &send_window_above_max_series;
+          break;
+      }
+    }
+  }
+  last_series->intervals.emplace_back(send_window_state_switch,
+                                      config.CallEndTimeSec());
+  plot->AppendTimeSeries(std::move(ref_window_series));
+  plot->AppendTimeSeries(std::move(ref_window_i_series));
+  plot->AppendTimeSeries(std::move(max_data_in_flight));
+  plot->AppendTimeSeries(std::move(data_in_flight));
+  plot->AppendIntervalSeries(std::move(send_window_above_max_series));
+  plot->AppendIntervalSeries(std::move(send_window_below_ref_window_series));
+  plot->AppendIntervalSeries(std::move(send_window_above_ref_window_series));
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 10, "Bytes", kBottomMargin, kTopMargin);
+  plot->SetTitle("Simulated Scream RefWindow");
+}
+
+void CreateScreamSimulationRatiosGraph(const ParsedRtcEventLog& parsed_log,
+                                       const AnalyzerConfig& config,
+                                       Plot* plot) {
+  TimeSeries queue_delay_dev_norm_series("QueueDelayDevNorm", LineStyle::kStep);
+  TimeSeries l4s_alpha_series("L4sAlpha", LineStyle::kStep);
+  TimeSeries l4s_alpha_v_series("L4sAlphaV", LineStyle::kStep);
+  TimeSeries ref_window_scale_factor_due_to_increased_delay(
+      "RefWindowScaleFactorDueToIncreasedDelay", LineStyle::kStep);
+  TimeSeries ref_window_scale_factor_due_to_delay_variation(
+      "RefWindowScaleFactorDueToDelayVariation", LineStyle::kStep);
+
+  LogScreamSimulation simulation({.rate_window = config.window_duration_},
+                                 config.env_);
+  simulation.ProcessEventsInLog(parsed_log);
+
+  for (const LogScreamSimulation::State& state : simulation.updates()) {
+    queue_delay_dev_norm_series.points.emplace_back(
+        config.GetCallTimeSec(state.time), state.queue_delay_dev_norm);
+    l4s_alpha_series.points.emplace_back(config.GetCallTimeSec(state.time),
+                                         state.l4s_alpha);
+    l4s_alpha_v_series.points.emplace_back(config.GetCallTimeSec(state.time),
+                                           state.l4s_alpha_v);
+    ref_window_scale_factor_due_to_increased_delay.points.emplace_back(
+        config.GetCallTimeSec(state.time),
+        state.ref_window_scale_factor_due_to_increased_delay);
+    ref_window_scale_factor_due_to_delay_variation.points.emplace_back(
+        config.GetCallTimeSec(state.time),
+        state.ref_window_scale_factor_due_to_delay_variation);
+  }
+  plot->AppendTimeSeries(std::move(queue_delay_dev_norm_series));
+  plot->AppendTimeSeries(std::move(l4s_alpha_series));
+  plot->AppendTimeSeries(std::move(l4s_alpha_v_series));
+  plot->AppendTimeSeries(
+      std::move(ref_window_scale_factor_due_to_increased_delay));
+  plot->AppendTimeSeries(
+      std::move(ref_window_scale_factor_due_to_delay_variation));
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Ratios", kBottomMargin, kTopMargin);
+  plot->SetTitle("Simulated Scream Ratios");
+}
+
+void CreateScreamRefWindowGraph(const ParsedRtcEventLog& parsed_log,
+                                const AnalyzerConfig& config,
+                                Plot* plot) {
+  TimeSeries ref_window_series("RefWindow", LineStyle::kStep);
+  for (auto& scream_update : parsed_log.bwe_scream_updates()) {
+    float x = config.GetCallTimeSec(scream_update.log_time());
+    float y = static_cast<float>(scream_update.ref_window.bytes());
+    ref_window_series.points.emplace_back(x, y);
+  }
+  plot->AppendTimeSeries(std::move(ref_window_series));
+
+  TimeSeries data_in_flight_series("Data in flight", LineStyle::kLine);
+  for (auto& scream_update : parsed_log.bwe_scream_updates()) {
+    float x = config.GetCallTimeSec(scream_update.log_time());
+    float y = static_cast<float>(scream_update.data_in_flight.bytes());
+    data_in_flight_series.points.emplace_back(x, y);
+  }
+  plot->AppendTimeSeries(std::move(data_in_flight_series));
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 3000, "Bytes", kBottomMargin, kTopMargin);
+  plot->SetTitle("Scream Ref Window");
+}
+
+void CreateScreamDelayEstimateGraph(const ParsedRtcEventLog& parsed_log,
+                                    const AnalyzerConfig& config,
+                                    Plot* plot) {
+  TimeSeries smoothed_rtt_series("Smoothed RTT", LineStyle::kStep);
+  TimeSeries avg_queue_delay_series("Avg queue delay", LineStyle::kStep);
+
+  for (auto& scream_update : parsed_log.bwe_scream_updates()) {
+    float x = config.GetCallTimeSec(scream_update.log_time());
+    float smoothed_rtt_ms = static_cast<float>(scream_update.smoothed_rtt.ms());
+    smoothed_rtt_series.points.emplace_back(x, smoothed_rtt_ms);
+    float avg_queue_delay_ms =
+        static_cast<float>(scream_update.avg_queue_delay.ms());
+    avg_queue_delay_series.points.emplace_back(x, avg_queue_delay_ms);
+  }
+
+  plot->AppendTimeSeries(std::move(smoothed_rtt_series));
+  plot->AppendTimeSeries(std::move(avg_queue_delay_series));
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 50, "Delay (ms)", kBottomMargin, kTopMargin);
+  plot->SetTitle("Scream delay estimates");
+}
+
+void CreateSendSideBweSimulationGraph(const ParsedRtcEventLog& parsed_log,
+                                      const AnalyzerConfig& config,
+                                      Plot* plot) {
+  using RtpPacketType = LoggedRtpPacketOutgoing;
+  using TransportFeedbackType = LoggedRtcpPacketTransportFeedback;
+
+  // TODO(terelius): The parser could provide a clearer view of the
+  // streams, so that we don't have to recalculate it.
+  std::multimap<int64_t, const RtpPacketType*> outgoing_rtp;
+  for (const auto& stream : parsed_log.outgoing_rtp_packets_by_ssrc()) {
+    for (const RtpPacketType& rtp_packet : stream.outgoing_packets)
+      outgoing_rtp.insert(
+          std::make_pair(rtp_packet.rtp.log_time_us(), &rtp_packet));
+  }
+
+  const std::vector<TransportFeedbackType>& incoming_rtcp =
+      parsed_log.transport_feedbacks(kIncomingPacket);
+
+  SimulatedClock clock(0);
+  BitrateObserver observer;
+  TransportFeedbackAdapter transport_feedback;
+  auto factory = GoogCcNetworkControllerFactory();
+  TimeDelta process_interval = factory.GetProcessInterval();
+  // TODO(holmer): Log the call config and use that here instead.
+  static const uint32_t kDefaultStartBitrateBps = 300000;
+  NetworkControllerConfig cc_config(config.env_);
+  cc_config.constraints.at_time = clock.CurrentTime();
+  cc_config.constraints.starting_rate =
+      DataRate::BitsPerSec(kDefaultStartBitrateBps);
+  auto goog_cc = factory.Create(cc_config);
+
+  TimeSeries time_series("Delay-based estimate", LineStyle::kStep,
+                         PointStyle::kHighlight);
+  TimeSeries acked_time_series("Raw acked bitrate", LineStyle::kLine,
+                               PointStyle::kHighlight);
+  TimeSeries robust_time_series("Robust throughput estimate", LineStyle::kLine,
+                                PointStyle::kHighlight);
+  TimeSeries acked_estimate_time_series("Ackednowledged bitrate estimate",
+                                        LineStyle::kLine,
+                                        PointStyle::kHighlight);
+
+  auto rtp_iterator = outgoing_rtp.begin();
+  auto rtcp_iterator = incoming_rtcp.begin();
+
+  auto NextRtpTime = [&]() {
+    if (rtp_iterator != outgoing_rtp.end())
+      return static_cast<int64_t>(rtp_iterator->first);
+    return std::numeric_limits<int64_t>::max();
+  };
+
+  auto NextRtcpTime = [&]() {
+    if (rtcp_iterator != incoming_rtcp.end())
+      return static_cast<int64_t>(rtcp_iterator->log_time_us());
+    return std::numeric_limits<int64_t>::max();
+  };
+  int64_t next_process_time_us_ = std::min({NextRtpTime(), NextRtcpTime()});
+
+  auto NextProcessTime = [&]() {
+    if (rtcp_iterator != incoming_rtcp.end() ||
+        rtp_iterator != outgoing_rtp.end()) {
+      return next_process_time_us_;
+    }
+    return std::numeric_limits<int64_t>::max();
+  };
+
+  RateStatistics raw_acked_bitrate(750, 8000);
+  FieldTrials throughput_config(
+      "WebRTC-Bwe-RobustThroughputEstimatorSettings/enabled:true/");
+  std::unique_ptr<AcknowledgedBitrateEstimatorInterface>
+      robust_throughput_estimator(
+          AcknowledgedBitrateEstimatorInterface::Create(&throughput_config));
+  FieldTrials acked_bitrate_config(
+      "WebRTC-Bwe-RobustThroughputEstimatorSettings/enabled:false/");
+  std::unique_ptr<AcknowledgedBitrateEstimatorInterface>
+      acknowledged_bitrate_estimator(
+          AcknowledgedBitrateEstimatorInterface::Create(&acked_bitrate_config));
+  int64_t time_us =
+      std::min({NextRtpTime(), NextRtcpTime(), NextProcessTime()});
+  int64_t last_update_us = 0;
+  while (time_us != std::numeric_limits<int64_t>::max()) {
+    clock.AdvanceTimeMicroseconds(time_us - clock.TimeInMicroseconds());
+    if (clock.TimeInMicroseconds() >= NextRtpTime()) {
+      RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtpTime());
+      const RtpPacketType& rtp_packet = *rtp_iterator->second;
+      if (rtp_packet.rtp.header.extension.hasTransportSequenceNumber) {
+        RtpPacketToSend send_packet(/*extensions=*/nullptr);
+        send_packet.set_transport_sequence_number(
+            rtp_packet.rtp.header.extension.transportSequenceNumber);
+        send_packet.SetSsrc(rtp_packet.rtp.header.ssrc);
+        send_packet.SetSequenceNumber(rtp_packet.rtp.header.sequenceNumber);
+        send_packet.SetPayloadSize(rtp_packet.rtp.total_length -
+                                   send_packet.headers_size());
+        RTC_DCHECK_EQ(send_packet.size(), rtp_packet.rtp.total_length);
+        if (IsRtxSsrc(parsed_log, PacketDirection::kOutgoingPacket,
+                      rtp_packet.rtp.header.ssrc)) {
+          // Don't set the optional media type as we don't know if it is
+          // a retransmission, FEC or padding.
+        } else if (IsVideoSsrc(parsed_log, PacketDirection::kOutgoingPacket,
+                               rtp_packet.rtp.header.ssrc)) {
+          send_packet.set_packet_type(RtpPacketMediaType::kVideo);
+        } else if (IsAudioSsrc(parsed_log, PacketDirection::kOutgoingPacket,
+                               rtp_packet.rtp.header.ssrc)) {
+          send_packet.set_packet_type(RtpPacketMediaType::kAudio);
+        }
+        transport_feedback.AddPacket(
+            send_packet, PacedPacketInfo(),
+            0u,  // Per packet overhead bytes.,
+            Timestamp::Micros(rtp_packet.rtp.log_time_us()));
+      }
+      SentPacketInfo sent_packet;
+      sent_packet.send_time_ms = rtp_packet.rtp.log_time_ms();
+      sent_packet.info.included_in_allocation = true;
+      sent_packet.info.packet_size_bytes = rtp_packet.rtp.total_length;
+      if (rtp_packet.rtp.header.extension.hasTransportSequenceNumber) {
+        sent_packet.packet_id =
+            rtp_packet.rtp.header.extension.transportSequenceNumber;
+        sent_packet.info.included_in_feedback = true;
+      }
+      auto sent_msg = transport_feedback.ProcessSentPacket(sent_packet);
+      if (sent_msg)
+        observer.Update(goog_cc->OnSentPacket(*sent_msg));
+      ++rtp_iterator;
+    }
+    if (clock.TimeInMicroseconds() >= NextRtcpTime()) {
+      RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtcpTime());
+
+      auto feedback_msg = transport_feedback.ProcessTransportFeedback(
+          rtcp_iterator->transport_feedback, clock.CurrentTime());
+      if (feedback_msg) {
+        observer.Update(goog_cc->OnTransportPacketsFeedback(*feedback_msg));
+        std::vector<PacketResult> feedback =
+            feedback_msg->SortedByReceiveTime();
+        if (!feedback.empty()) {
+          acknowledged_bitrate_estimator->IncomingPacketFeedbackVector(
+              feedback);
+          robust_throughput_estimator->IncomingPacketFeedbackVector(feedback);
+          for (const PacketResult& packet : feedback) {
+            raw_acked_bitrate.Update(packet.sent_packet.size.bytes(),
+                                     packet.receive_time.ms());
+          }
+          std::optional<uint32_t> raw_bitrate_bps =
+              raw_acked_bitrate.Rate(feedback.back().receive_time.ms());
+          float x = config.GetCallTimeSec(clock.CurrentTime());
+          if (raw_bitrate_bps) {
+            float y = raw_bitrate_bps.value() / 1000;
+            acked_time_series.points.emplace_back(x, y);
+          }
+          std::optional<DataRate> robust_estimate =
+              robust_throughput_estimator->bitrate();
+          if (robust_estimate) {
+            float y = robust_estimate.value().kbps();
+            robust_time_series.points.emplace_back(x, y);
+          }
+          std::optional<DataRate> acked_estimate =
+              acknowledged_bitrate_estimator->bitrate();
+          if (acked_estimate) {
+            float y = acked_estimate.value().kbps();
+            acked_estimate_time_series.points.emplace_back(x, y);
+          }
+        }
+      }
+      ++rtcp_iterator;
+    }
+    if (clock.TimeInMicroseconds() >= NextProcessTime()) {
+      RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextProcessTime());
+      ProcessInterval msg;
+      msg.at_time = clock.CurrentTime();
+      observer.Update(goog_cc->OnProcessInterval(msg));
+      next_process_time_us_ += process_interval.us();
+    }
+    if (observer.GetAndResetBitrateUpdated() ||
+        time_us - last_update_us >= 1e6) {
+      uint32_t y = observer.last_bitrate_bps() / 1000;
+      float x = config.GetCallTimeSec(clock.CurrentTime());
+      time_series.points.emplace_back(x, y);
+      last_update_us = time_us;
+    }
+    time_us = std::min({NextRtpTime(), NextRtcpTime(), NextProcessTime()});
+  }
+  // Add the data set to the plot.
+  plot->AppendTimeSeries(std::move(time_series));
+  plot->AppendTimeSeries(std::move(robust_time_series));
+  plot->AppendTimeSeries(std::move(acked_time_series));
+  plot->AppendTimeSeriesIfNotEmpty(std::move(acked_estimate_time_series));
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 10, "Bitrate (kbps)", kBottomMargin, kTopMargin);
+  plot->SetTitle("Simulated send-side BWE behavior");
+}
+
+void CreateReceiveSideBweSimulationGraph(const ParsedRtcEventLog& parsed_log,
+                                         const AnalyzerConfig& config,
+                                         Plot* plot) {
+  using RtpPacketType = LoggedRtpPacketIncoming;
+  class RembInterceptor {
+   public:
+    void SendRemb(uint32_t bitrate_bps, std::vector<uint32_t> ssrcs) {
+      last_bitrate_bps_ = bitrate_bps;
+      bitrate_updated_ = true;
+    }
+    uint32_t last_bitrate_bps() const { return last_bitrate_bps_; }
+    bool GetAndResetBitrateUpdated() {
+      bool bitrate_updated = bitrate_updated_;
+      bitrate_updated_ = false;
+      return bitrate_updated;
+    }
+
+   private:
+    // We don't know the start bitrate, but assume that it is the default 300
+    // kbps.
+    uint32_t last_bitrate_bps_ = 300000;
+    bool bitrate_updated_ = false;
+  };
+
+  std::multimap<int64_t, const RtpPacketType*> incoming_rtp;
+
+  for (const auto& stream : parsed_log.incoming_rtp_packets_by_ssrc()) {
+    if (IsVideoSsrc(parsed_log, kIncomingPacket, stream.ssrc)) {
+      for (const auto& rtp_packet : stream.incoming_packets)
+        incoming_rtp.insert(
+            std::make_pair(rtp_packet.rtp.log_time_us(), &rtp_packet));
+    }
+  }
+
+  SimulatedClock clock(0);
+  EnvironmentFactory env_factory(config.env_);
+  env_factory.Set(&clock);
+  RembInterceptor remb_interceptor;
+  ReceiveSideCongestionController rscc(
+      env_factory.Create(), [](auto...) {},
+      absl::bind_front(&RembInterceptor::SendRemb, &remb_interceptor));
+  // TODO(holmer): Log the call config and use that here instead.
+  // static const uint32_t kDefaultStartBitrateBps = 300000;
+  // rscc.SetBweBitrates(0, kDefaultStartBitrateBps, -1);
+
+  TimeSeries time_series("Receive side estimate", LineStyle::kLine,
+                         PointStyle::kHighlight);
+  TimeSeries acked_time_series("Received bitrate", LineStyle::kLine);
+
+  RateStatistics acked_bitrate(250, 8000);
+  int64_t last_update_us = 0;
+  for (const auto& kv : incoming_rtp) {
+    const RtpPacketType& packet = *kv.second;
+
+    RtpPacketReceived rtp_packet = RtpPacketForBWEFromHeader(packet.rtp.header);
+    rtp_packet.set_arrival_time(packet.rtp.log_time());
+    rtp_packet.SetPayloadSize(packet.rtp.total_length -
+                              rtp_packet.headers_size());
+
+    clock.AdvanceTime(rtp_packet.arrival_time() - clock.CurrentTime());
+    rscc.OnReceivedPacket(rtp_packet, MediaType::VIDEO);
+    int64_t arrival_time_ms = packet.rtp.log_time().ms();
+    acked_bitrate.Update(packet.rtp.total_length, arrival_time_ms);
+    std::optional<uint32_t> bitrate_bps = acked_bitrate.Rate(arrival_time_ms);
+    if (bitrate_bps) {
+      uint32_t y = *bitrate_bps / 1000;
+      float x = config.GetCallTimeSec(clock.CurrentTime());
+      acked_time_series.points.emplace_back(x, y);
+    }
+    if (remb_interceptor.GetAndResetBitrateUpdated() ||
+        clock.TimeInMicroseconds() - last_update_us >= 1e6) {
+      uint32_t y = remb_interceptor.last_bitrate_bps() / 1000;
+      float x = config.GetCallTimeSec(clock.CurrentTime());
+      time_series.points.emplace_back(x, y);
+      last_update_us = clock.TimeInMicroseconds();
+    }
+  }
+  // Add the data set to the plot.
+  plot->AppendTimeSeries(std::move(time_series));
+  plot->AppendTimeSeries(std::move(acked_time_series));
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 10, "Bitrate (kbps)", kBottomMargin, kTopMargin);
+  plot->SetTitle("Simulated receive-side BWE behavior");
+}
+
+void CreateNetworkDelayFeedbackGraph(const ParsedRtcEventLog& parsed_log,
+                                     const AnalyzerConfig& config,
+                                     Plot* plot) {
+  TimeSeries time_series("Network delay", LineStyle::kLine,
+                         PointStyle::kHighlight);
+  int64_t min_send_receive_diff_ms = std::numeric_limits<int64_t>::max();
+  int64_t min_rtt_ms = std::numeric_limits<int64_t>::max();
+
+  std::vector<MatchedSendArrivalTimes> matched_rtp_rtcp =
+      GetNetworkTrace(parsed_log);
+  absl::c_stable_sort(matched_rtp_rtcp, [](const MatchedSendArrivalTimes& a,
+                                           const MatchedSendArrivalTimes& b) {
+    return a.feedback_arrival_time_ms < b.feedback_arrival_time_ms ||
+           (a.feedback_arrival_time_ms == b.feedback_arrival_time_ms &&
+            a.arrival_time_ms < b.arrival_time_ms);
+  });
+  for (const auto& packet : matched_rtp_rtcp) {
+    if (packet.arrival_time_ms == MatchedSendArrivalTimes::kNotReceived)
+      continue;
+    float x = config.GetCallTimeSecFromMs(packet.feedback_arrival_time_ms);
+    int64_t y = packet.arrival_time_ms - packet.send_time_ms;
+    int64_t rtt_ms = packet.feedback_arrival_time_ms - packet.send_time_ms;
+    min_rtt_ms = std::min(rtt_ms, min_rtt_ms);
+    min_send_receive_diff_ms = std::min(y, min_send_receive_diff_ms);
+    time_series.points.emplace_back(x, y);
+  }
+
+  // We assume that the base network delay (w/o queues) is equal to half
+  // the minimum RTT. Therefore rescale the delays by subtracting the minimum
+  // observed 1-ways delay and add half the minimum RTT.
+  const int64_t estimated_clock_offset_ms =
+      min_send_receive_diff_ms - min_rtt_ms / 2;
+  for (TimeSeriesPoint& point : time_series.points)
+    point.y -= estimated_clock_offset_ms;
+
+  // Add the data set to the plot.
+  plot->AppendTimeSeriesIfNotEmpty(std::move(time_series));
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 10, "Delay (ms)", kBottomMargin, kTopMargin);
+  plot->SetTitle("Outgoing network delay (based on per-packet feedback)");
+}
+
+void CreatePacerDelayGraph(const ParsedRtcEventLog& parsed_log,
+                           const AnalyzerConfig& config,
+                           Plot* plot) {
+  for (const auto& stream : parsed_log.outgoing_rtp_packets_by_ssrc()) {
+    const std::vector<LoggedRtpPacketOutgoing>& packets =
+        stream.outgoing_packets;
+
+    if (IsRtxSsrc(parsed_log, kOutgoingPacket, stream.ssrc)) {
+      continue;
+    }
+
+    if (packets.size() < 2) {
+      RTC_LOG(LS_WARNING)
+          << "Can't estimate a the RTP clock frequency or the "
+             "pacer delay with less than 2 packets in the stream";
+      continue;
+    }
+    int64_t segment_end_us = parsed_log.first_log_segment().stop_time_us();
+    std::optional<uint32_t> estimated_frequency =
+        EstimateRtpClockFrequency(packets, segment_end_us);
+    if (!estimated_frequency)
+      continue;
+    if (IsVideoSsrc(parsed_log, kOutgoingPacket, stream.ssrc) &&
+        *estimated_frequency != 90000) {
+      RTC_LOG(LS_WARNING)
+          << "Video stream should use a 90 kHz clock but appears to use "
+          << *estimated_frequency / 1000 << ". Discarding.";
+      continue;
+    }
+
+    TimeSeries pacer_delay_series(
+        GetStreamName(parsed_log, kOutgoingPacket, stream.ssrc) + "(" +
+            std::to_string(*estimated_frequency / 1000) + " kHz)",
+        LineStyle::kLine, PointStyle::kHighlight);
+    SeqNumUnwrapper<uint32_t> timestamp_unwrapper;
+    uint64_t first_capture_timestamp =
+        timestamp_unwrapper.Unwrap(packets.front().rtp.header.timestamp);
+    uint64_t first_send_timestamp = packets.front().rtp.log_time_us();
+    for (const auto& packet : packets) {
+      double capture_time_ms = (static_cast<double>(timestamp_unwrapper.Unwrap(
+                                    packet.rtp.header.timestamp)) -
+                                first_capture_timestamp) /
+                               *estimated_frequency * 1000;
+      double send_time_ms =
+          static_cast<double>(packet.rtp.log_time_us() - first_send_timestamp) /
+          1000;
+      float x = config.GetCallTimeSec(packet.rtp.log_time());
+      float y = send_time_ms - capture_time_ms;
+      pacer_delay_series.points.emplace_back(x, y);
+    }
+    plot->AppendTimeSeries(std::move(pacer_delay_series));
+  }
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 10, "Pacer delay (ms)", kBottomMargin, kTopMargin);
+  plot->SetTitle(
+      "Delay from capture to send time. (First packet normalized to 0.)");
+}
+
+}  // namespace webrtc
diff --git a/rtc_tools/rtc_event_log_visualizer/analyze_bwe.h b/rtc_tools/rtc_event_log_visualizer/analyze_bwe.h
new file mode 100644
index 0000000000..fd4b041c78
--- /dev/null
+++ b/rtc_tools/rtc_event_log_visualizer/analyze_bwe.h
@@ -0,0 +1,81 @@
+/*
+ *  Copyright (c) 2026 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZE_BWE_H_
+#define RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZE_BWE_H_
+
+#include "logging/rtc_event_log/rtc_event_log_parser.h"
+#include "rtc_tools/rtc_event_log_visualizer/analyzer_common.h"
+#include "rtc_tools/rtc_event_log_visualizer/plot_base.h"
+
+namespace webrtc {
+
+void CreateIncomingDelayGraph(const ParsedRtcEventLog& parsed_log,
+                              const AnalyzerConfig& config,
+                              Plot* plot);
+
+void CreateFractionLossGraph(const ParsedRtcEventLog& parsed_log,
+                             const AnalyzerConfig& config,
+                             Plot* plot);
+
+void CreateTotalIncomingBitrateGraph(const ParsedRtcEventLog& parsed_log,
+                                     const AnalyzerConfig& config,
+                                     Plot* plot);
+
+void CreateTotalOutgoingBitrateGraph(const ParsedRtcEventLog& parsed_log,
+                                     const AnalyzerConfig& config,
+                                     Plot* plot,
+                                     bool show_detector_state = false,
+                                     bool show_alr_state = false,
+                                     bool show_link_capacity = false);
+
+void CreateGoogCcSimulationGraph(const ParsedRtcEventLog& parsed_log,
+                                 const AnalyzerConfig& config,
+                                 Plot* plot);
+
+void CreateScreamSimulationBitrateGraph(const ParsedRtcEventLog& parsed_log,
+                                        const AnalyzerConfig& config,
+                                        Plot* plot);
+
+void CreateScreamSimulationRefWindowGraph(const ParsedRtcEventLog& parsed_log,
+                                          const AnalyzerConfig& config,
+                                          Plot* plot);
+
+void CreateScreamSimulationRatiosGraph(const ParsedRtcEventLog& parsed_log,
+                                       const AnalyzerConfig& config,
+                                       Plot* plot);
+
+void CreateScreamRefWindowGraph(const ParsedRtcEventLog& parsed_log,
+                                const AnalyzerConfig& config,
+                                Plot* plot);
+
+void CreateScreamDelayEstimateGraph(const ParsedRtcEventLog& parsed_log,
+                                    const AnalyzerConfig& config,
+                                    Plot* plot);
+
+void CreateSendSideBweSimulationGraph(const ParsedRtcEventLog& parsed_log,
+                                      const AnalyzerConfig& config,
+                                      Plot* plot);
+
+void CreateReceiveSideBweSimulationGraph(const ParsedRtcEventLog& parsed_log,
+                                         const AnalyzerConfig& config,
+                                         Plot* plot);
+
+void CreateNetworkDelayFeedbackGraph(const ParsedRtcEventLog& parsed_log,
+                                     const AnalyzerConfig& config,
+                                     Plot* plot);
+
+void CreatePacerDelayGraph(const ParsedRtcEventLog& parsed_log,
+                           const AnalyzerConfig& config,
+                           Plot* plot);
+
+}  // namespace webrtc
+
+#endif  // RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZE_BWE_H_
diff --git a/rtc_tools/rtc_event_log_visualizer/analyze_connectivity.cc b/rtc_tools/rtc_event_log_visualizer/analyze_connectivity.cc
new file mode 100644
index 0000000000..b87a108b96
--- /dev/null
+++ b/rtc_tools/rtc_event_log_visualizer/analyze_connectivity.cc
@@ -0,0 +1,285 @@
+/*
+ *  Copyright (c) 2026 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "rtc_tools/rtc_event_log_visualizer/analyze_connectivity.h"
+
+#include <cstddef>
+#include <cstdint>
+#include <map>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/candidate.h"
+#include "api/dtls_transport_interface.h"
+#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair.h"
+#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.h"
+#include "logging/rtc_event_log/rtc_event_log_parser.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_tools/rtc_event_log_visualizer/analyzer_common.h"
+#include "rtc_tools/rtc_event_log_visualizer/plot_base.h"
+
+namespace webrtc {
+
+namespace {
+
+const char kUnknownEnumValue[] = "unknown";
+
+// TODO(tommi): This should be "host".
+const char kIceCandidateTypeLocal[] = "local";
+// TODO(tommi): This should be "srflx".
+const char kIceCandidateTypeStun[] = "stun";
+const char kIceCandidateTypePrflx[] = "prflx";
+const char kIceCandidateTypeRelay[] = "relay";
+
+const char kProtocolUdp[] = "udp";
+const char kProtocolTcp[] = "tcp";
+const char kProtocolSsltcp[] = "ssltcp";
+const char kProtocolTls[] = "tls";
+
+const char kAddressFamilyIpv4[] = "ipv4";
+const char kAddressFamilyIpv6[] = "ipv6";
+
+const char kNetworkTypeEthernet[] = "ethernet";
+const char kNetworkTypeLoopback[] = "loopback";
+const char kNetworkTypeWifi[] = "wifi";
+const char kNetworkTypeVpn[] = "vpn";
+const char kNetworkTypeCellular[] = "cellular";
+
+absl::string_view GetIceCandidateTypeAsString(IceCandidateType type) {
+  switch (type) {
+    case IceCandidateType::kHost:
+      return kIceCandidateTypeLocal;
+    case IceCandidateType::kSrflx:
+      return kIceCandidateTypeStun;
+    case IceCandidateType::kPrflx:
+      return kIceCandidateTypePrflx;
+    case IceCandidateType::kRelay:
+      return kIceCandidateTypeRelay;
+    default:
+      RTC_DCHECK_NOTREACHED();
+      return kUnknownEnumValue;
+  }
+}
+
+std::string GetProtocolAsString(IceCandidatePairProtocol protocol) {
+  switch (protocol) {
+    case IceCandidatePairProtocol::kUdp:
+      return kProtocolUdp;
+    case IceCandidatePairProtocol::kTcp:
+      return kProtocolTcp;
+    case IceCandidatePairProtocol::kSsltcp:
+      return kProtocolSsltcp;
+    case IceCandidatePairProtocol::kTls:
+      return kProtocolTls;
+    default:
+      return kUnknownEnumValue;
+  }
+}
+
+std::string GetAddressFamilyAsString(IceCandidatePairAddressFamily family) {
+  switch (family) {
+    case IceCandidatePairAddressFamily::kIpv4:
+      return kAddressFamilyIpv4;
+    case IceCandidatePairAddressFamily::kIpv6:
+      return kAddressFamilyIpv6;
+    default:
+      return kUnknownEnumValue;
+  }
+}
+
+std::string GetNetworkTypeAsString(IceCandidateNetworkType type) {
+  switch (type) {
+    case IceCandidateNetworkType::kEthernet:
+      return kNetworkTypeEthernet;
+    case IceCandidateNetworkType::kLoopback:
+      return kNetworkTypeLoopback;
+    case IceCandidateNetworkType::kWifi:
+      return kNetworkTypeWifi;
+    case IceCandidateNetworkType::kVpn:
+      return kNetworkTypeVpn;
+    case IceCandidateNetworkType::kCellular:
+      return kNetworkTypeCellular;
+    default:
+      return kUnknownEnumValue;
+  }
+}
+
+std::string GetCandidatePairLogDescriptionAsString(
+    const LoggedIceCandidatePairConfig& config) {
+  // Example: stun:wifi->relay(tcp):cellular@udp:ipv4
+  // represents a pair of a local server-reflexive candidate on a WiFi network
+  // and a remote relay candidate using TCP as the relay protocol on a cell
+  // network, when the candidate pair communicates over UDP using IPv4.
+  StringBuilder ss;
+  ss << GetIceCandidateTypeAsString(config.local_candidate_type);
+
+  if (config.local_candidate_type == IceCandidateType::kRelay) {
+    ss << "(" << GetProtocolAsString(config.local_relay_protocol) << ")";
+  }
+
+  ss << ":" << GetNetworkTypeAsString(config.local_network_type) << ":"
+     << GetAddressFamilyAsString(config.local_address_family) << "->"
+     << GetIceCandidateTypeAsString(config.remote_candidate_type) << ":"
+     << GetAddressFamilyAsString(config.remote_address_family) << "@"
+     << GetProtocolAsString(config.candidate_pair_protocol);
+  return ss.Release();
+}
+
+std::map<uint32_t, std::string> BuildCandidateIdLogDescriptionMap(
+    const std::vector<LoggedIceCandidatePairConfig>&
+        ice_candidate_pair_configs) {
+  std::map<uint32_t, std::string> candidate_pair_desc_by_id;
+  for (const auto& config : ice_candidate_pair_configs) {
+    // TODO(qingsi): Add the handling of the "Updated" config event after the
+    // visualization of property change for candidate pairs is introduced.
+    if (candidate_pair_desc_by_id.find(config.candidate_pair_id) ==
+        candidate_pair_desc_by_id.end()) {
+      const std::string candidate_pair_desc =
+          GetCandidatePairLogDescriptionAsString(config);
+      candidate_pair_desc_by_id[config.candidate_pair_id] = candidate_pair_desc;
+    }
+  }
+  return candidate_pair_desc_by_id;
+}
+
+}  // namespace
+
+void CreateIceCandidatePairConfigGraph(const ParsedRtcEventLog& parsed_log,
+                                       const AnalyzerConfig& config,
+                                       Plot* plot) {
+  std::map<uint32_t, TimeSeries> configs_by_cp_id;
+  for (const auto& config_item : parsed_log.ice_candidate_pair_configs()) {
+    if (configs_by_cp_id.find(config_item.candidate_pair_id) ==
+        configs_by_cp_id.end()) {
+      const std::string candidate_pair_desc =
+          GetCandidatePairLogDescriptionAsString(config_item);
+      configs_by_cp_id[config_item.candidate_pair_id] =
+          TimeSeries("[" + std::to_string(config_item.candidate_pair_id) + "]" +
+                         candidate_pair_desc,
+                     LineStyle::kNone, PointStyle::kHighlight);
+    }
+    float x = config.GetCallTimeSec(config_item.log_time());
+    float y = static_cast<float>(config_item.type);
+    configs_by_cp_id[config_item.candidate_pair_id].points.emplace_back(x, y);
+  }
+
+  // TODO(qingsi): There can be a large number of candidate pairs generated by
+  // certain calls and the frontend cannot render the chart in this case due
+  // to the failure of generating a palette with the same number of colors.
+  for (auto& kv : configs_by_cp_id) {
+    plot->AppendTimeSeries(std::move(kv.second));
+  }
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 3, "Config Type", kBottomMargin, kTopMargin);
+  plot->SetTitle("[IceEventLog] ICE candidate pair configs");
+  plot->SetYAxisTickLabels(
+      {{static_cast<float>(IceCandidatePairConfigType::kAdded), "ADDED"},
+       {static_cast<float>(IceCandidatePairConfigType::kUpdated), "UPDATED"},
+       {static_cast<float>(IceCandidatePairConfigType::kDestroyed),
+        "DESTROYED"},
+       {static_cast<float>(IceCandidatePairConfigType::kSelected),
+        "SELECTED"}});
+}
+
+void CreateIceConnectivityCheckGraph(const ParsedRtcEventLog& parsed_log,
+                                     const AnalyzerConfig& config,
+                                     Plot* plot) {
+  constexpr int kEventTypeOffset =
+      static_cast<int>(IceCandidatePairConfigType::kNumValues);
+  std::map<uint32_t, TimeSeries> checks_by_cp_id;
+  std::map<uint32_t, std::string> candidate_pair_desc_by_id =
+      BuildCandidateIdLogDescriptionMap(
+          parsed_log.ice_candidate_pair_configs());
+  for (const auto& event : parsed_log.ice_candidate_pair_events()) {
+    if (checks_by_cp_id.find(event.candidate_pair_id) ==
+        checks_by_cp_id.end()) {
+      checks_by_cp_id[event.candidate_pair_id] =
+          TimeSeries("[" + std::to_string(event.candidate_pair_id) + "]" +
+                         candidate_pair_desc_by_id[event.candidate_pair_id],
+                     LineStyle::kNone, PointStyle::kHighlight);
+    }
+    float x = config.GetCallTimeSec(event.log_time());
+    float y = static_cast<float>(event.type) + kEventTypeOffset;
+    checks_by_cp_id[event.candidate_pair_id].points.emplace_back(x, y);
+  }
+
+  // TODO(qingsi): The same issue as in CreateIceCandidatePairConfigGraph.
+  for (auto& kv : checks_by_cp_id) {
+    plot->AppendTimeSeries(std::move(kv.second));
+  }
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 4, "Connectivity State", kBottomMargin,
+                          kTopMargin);
+  plot->SetTitle("[IceEventLog] ICE connectivity checks");
+
+  plot->SetYAxisTickLabels(
+      {{static_cast<float>(IceCandidatePairEventType::kCheckSent) +
+            kEventTypeOffset,
+        "CHECK SENT"},
+       {static_cast<float>(IceCandidatePairEventType::kCheckReceived) +
+            kEventTypeOffset,
+        "CHECK RECEIVED"},
+       {static_cast<float>(IceCandidatePairEventType::kCheckResponseSent) +
+            kEventTypeOffset,
+        "RESPONSE SENT"},
+       {static_cast<float>(IceCandidatePairEventType::kCheckResponseReceived) +
+            kEventTypeOffset,
+        "RESPONSE RECEIVED"}});
+}
+
+void CreateDtlsTransportStateGraph(const ParsedRtcEventLog& parsed_log,
+                                   const AnalyzerConfig& config,
+                                   Plot* plot) {
+  TimeSeries states("DTLS Transport State", LineStyle::kNone,
+                    PointStyle::kHighlight);
+  for (const auto& event : parsed_log.dtls_transport_states()) {
+    float x = config.GetCallTimeSec(event.log_time());
+    float y = static_cast<float>(event.dtls_transport_state);
+    states.points.emplace_back(x, y);
+  }
+  plot->AppendTimeSeries(std::move(states));
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, static_cast<float>(DtlsTransportState::kNumValues),
+                          "Transport State", kBottomMargin, kTopMargin);
+  plot->SetTitle("DTLS Transport State");
+  plot->SetYAxisTickLabels(
+      {{static_cast<float>(DtlsTransportState::kNew), "NEW"},
+       {static_cast<float>(DtlsTransportState::kConnecting), "CONNECTING"},
+       {static_cast<float>(DtlsTransportState::kConnected), "CONNECTED"},
+       {static_cast<float>(DtlsTransportState::kClosed), "CLOSED"},
+       {static_cast<float>(DtlsTransportState::kFailed), "FAILED"}});
+}
+
+void CreateDtlsWritableStateGraph(const ParsedRtcEventLog& parsed_log,
+                                  const AnalyzerConfig& config,
+                                  Plot* plot) {
+  TimeSeries writable("DTLS Writable", LineStyle::kNone,
+                      PointStyle::kHighlight);
+  for (const auto& event : parsed_log.dtls_writable_states()) {
+    float x = config.GetCallTimeSec(event.log_time());
+    float y = static_cast<float>(event.writable);
+    writable.points.emplace_back(x, y);
+  }
+  plot->AppendTimeSeries(std::move(writable));
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Writable", kBottomMargin, kTopMargin);
+  plot->SetTitle("DTLS Writable State");
+}
+
+}  // namespace webrtc
diff --git a/rtc_tools/rtc_event_log_visualizer/analyze_connectivity.h b/rtc_tools/rtc_event_log_visualizer/analyze_connectivity.h
new file mode 100644
index 0000000000..1eac2f0c52
--- /dev/null
+++ b/rtc_tools/rtc_event_log_visualizer/analyze_connectivity.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2026 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZE_CONNECTIVITY_H_
+#define RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZE_CONNECTIVITY_H_
+
+#include "logging/rtc_event_log/rtc_event_log_parser.h"
+#include "rtc_tools/rtc_event_log_visualizer/analyzer_common.h"
+#include "rtc_tools/rtc_event_log_visualizer/plot_base.h"
+
+namespace webrtc {
+
+void CreateIceCandidatePairConfigGraph(const ParsedRtcEventLog& parsed_log,
+                                       const AnalyzerConfig& config,
+                                       Plot* plot);
+
+void CreateIceConnectivityCheckGraph(const ParsedRtcEventLog& parsed_log,
+                                     const AnalyzerConfig& config,
+                                     Plot* plot);
+
+void CreateDtlsTransportStateGraph(const ParsedRtcEventLog& parsed_log,
+                                   const AnalyzerConfig& config,
+                                   Plot* plot);
+
+void CreateDtlsWritableStateGraph(const ParsedRtcEventLog& parsed_log,
+                                  const AnalyzerConfig& config,
+                                  Plot* plot);
+
+}  // namespace webrtc
+
+#endif  // RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZE_CONNECTIVITY_H_
diff --git a/rtc_tools/rtc_event_log_visualizer/analyze_rtp_rtcp.cc b/rtc_tools/rtc_event_log_visualizer/analyze_rtp_rtcp.cc
new file mode 100644
index 0000000000..9d6c0668e8
--- /dev/null
+++ b/rtc_tools/rtc_event_log_visualizer/analyze_rtp_rtcp.cc
@@ -0,0 +1,809 @@
+/*
+ *  Copyright (c) 2026 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "rtc_tools/rtc_event_log_visualizer/analyze_rtp_rtcp.h"
+
+#include <algorithm>
+#include <cstddef>
+#include <cstdint>
+#include <deque>
+#include <map>
+#include <optional>
+#include <string>
+#include <tuple>
+#include <unordered_set>
+#include <utility>
+#include <vector>
+
+#include "api/function_view.h"
+#include "api/rtp_headers.h"
+#include "api/transport/ecn_marking.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "logging/rtc_event_log/events/logged_rtp_rtcp.h"
+#include "logging/rtc_event_log/rtc_event_log_parser.h"
+#include "logging/rtc_event_log/rtc_event_processor.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/congestion_control_feedback.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/target_bitrate.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
+#include "rtc_tools/rtc_event_log_visualizer/analyzer_common.h"
+#include "rtc_tools/rtc_event_log_visualizer/plot_base.h"
+
+namespace webrtc {
+
+namespace {
+
+template <typename T>
+TimeSeries CreateRtcpTypeTimeSeries(const std::vector<T>& rtcp_list,
+                                    AnalyzerConfig config,
+                                    std::string rtcp_name,
+                                    int category_id) {
+  TimeSeries time_series(rtcp_name, LineStyle::kNone, PointStyle::kHighlight);
+  for (const auto& rtcp : rtcp_list) {
+    float x = config.GetCallTimeSec(rtcp.timestamp);
+    float y = category_id;
+    time_series.points.emplace_back(x, y);
+  }
+  return time_series;
+}
+
+struct PacketLossSummary {
+  size_t num_packets = 0;
+  size_t num_lost_packets = 0;
+  Timestamp base_time = Timestamp::MinusInfinity();
+};
+
+}  // namespace
+
+float GetHighestSeqNumber(const rtcp::ReportBlock& block) {
+  return block.extended_high_seq_num();
+}
+
+float GetFractionLost(const rtcp::ReportBlock& block) {
+  return static_cast<double>(block.fraction_lost()) / 256 * 100;
+}
+
+float GetCumulativeLost(const rtcp::ReportBlock& block) {
+  return block.cumulative_lost();
+}
+
+float DelaySinceLastSr(const rtcp::ReportBlock& block) {
+  return static_cast<double>(block.delay_since_last_sr()) / 65536;
+}
+
+void CreatePacketGraph(PacketDirection direction,
+                       const ParsedRtcEventLog& parsed_log,
+                       const AnalyzerConfig& config,
+                       Plot* plot) {
+  for (const auto& stream : parsed_log.rtp_packets_by_ssrc(direction)) {
+    // Filter on SSRC.
+    if (!MatchingSsrc(stream.ssrc, config.desired_ssrc_)) {
+      continue;
+    }
+
+    TimeSeries time_series(GetStreamName(parsed_log, direction, stream.ssrc),
+                           LineStyle::kBar);
+    auto GetPacketSize = [](const LoggedRtpPacket& packet) {
+      return std::optional<float>(packet.total_length);
+    };
+    auto ToCallTime = [&config](const LoggedRtpPacket& packet) {
+      return config.GetCallTimeSec(packet.timestamp);
+    };
+    ProcessPoints<LoggedRtpPacket>(ToCallTime, GetPacketSize,
+                                   stream.packet_view, &time_series);
+    plot->AppendTimeSeries(std::move(time_series));
+  }
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Packet size (bytes)", kBottomMargin,
+                          kTopMargin);
+  plot->SetTitle(GetDirectionAsString(direction) + " RTP packets");
+}
+
+void CreateRtcpTypeGraph(PacketDirection direction,
+                         const ParsedRtcEventLog& parsed_log,
+                         const AnalyzerConfig& config,
+                         Plot* plot) {
+  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(
+      parsed_log.transport_feedbacks(direction), config, "TWCC", 1));
+  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(
+      parsed_log.congestion_feedback(direction), config, "CCFB", 2));
+  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(
+      parsed_log.receiver_reports(direction), config, "RR", 3));
+  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(
+      parsed_log.sender_reports(direction), config, "SR", 4));
+  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(
+      parsed_log.extended_reports(direction), config, "XR", 5));
+  plot->AppendTimeSeries(
+      CreateRtcpTypeTimeSeries(parsed_log.nacks(direction), config, "NACK", 6));
+  plot->AppendTimeSeries(
+      CreateRtcpTypeTimeSeries(parsed_log.rembs(direction), config, "REMB", 7));
+  plot->AppendTimeSeries(
+      CreateRtcpTypeTimeSeries(parsed_log.firs(direction), config, "FIR", 8));
+  plot->AppendTimeSeries(
+      CreateRtcpTypeTimeSeries(parsed_log.plis(direction), config, "PLI", 9));
+  plot->AppendTimeSeries(
+      CreateRtcpTypeTimeSeries(parsed_log.byes(direction), config, "BYE", 10));
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "RTCP type", kBottomMargin, kTopMargin);
+  plot->SetTitle(GetDirectionAsString(direction) + " RTCP packets");
+  plot->SetYAxisTickLabels({{1, "TWCC"},
+                            {2, "CCFB"},
+                            {3, "RR"},
+                            {4, "SR"},
+                            {5, "XR"},
+                            {6, "NACK"},
+                            {7, "REMB"},
+                            {8, "FIR"},
+                            {9, "PLI"},
+                            {10, "BYE"}});
+}
+
+template <typename IterableType>
+void CreateAccumulatedPacketsTimeSeries(Plot* plot,
+                                        const AnalyzerConfig& config,
+                                        const IterableType& packets,
+                                        const std::string& label) {
+  TimeSeries time_series(label, LineStyle::kStep);
+  for (size_t i = 0; i < packets.size(); i++) {
+    float x = config.GetCallTimeSec(packets[i].log_time());
+    time_series.points.emplace_back(x, i + 1);
+  }
+  plot->AppendTimeSeries(std::move(time_series));
+}
+
+void CreateAccumulatedPacketsGraph(PacketDirection direction,
+                                   const ParsedRtcEventLog& parsed_log,
+                                   const AnalyzerConfig& config,
+                                   Plot* plot) {
+  for (const auto& stream : parsed_log.rtp_packets_by_ssrc(direction)) {
+    if (!MatchingSsrc(stream.ssrc, config.desired_ssrc_))
+      continue;
+    std::string label =
+        std::string("RTP ") + GetStreamName(parsed_log, direction, stream.ssrc);
+    CreateAccumulatedPacketsTimeSeries(plot, config, stream.packet_view, label);
+  }
+  std::string label =
+      std::string("RTCP ") + "(" + GetDirectionAsShortString(direction) + ")";
+  if (direction == kIncomingPacket) {
+    CreateAccumulatedPacketsTimeSeries(
+        plot, config, parsed_log.incoming_rtcp_packets(), label);
+  } else {
+    CreateAccumulatedPacketsTimeSeries(
+        plot, config, parsed_log.outgoing_rtcp_packets(), label);
+  }
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Received Packets", kBottomMargin, kTopMargin);
+  plot->SetTitle(std::string("Accumulated ") + GetDirectionAsString(direction) +
+                 " RTP/RTCP packets");
+}
+
+void CreatePacketRateGraph(PacketDirection direction,
+                           const ParsedRtcEventLog& parsed_log,
+                           const AnalyzerConfig& config,
+                           Plot* plot) {
+  auto CountPackets = [](auto packet) { return 1.0; };
+  for (const auto& stream : parsed_log.rtp_packets_by_ssrc(direction)) {
+    // Filter on SSRC.
+    if (!MatchingSsrc(stream.ssrc, config.desired_ssrc_)) {
+      continue;
+    }
+    TimeSeries time_series(
+        std::string("RTP ") + GetStreamName(parsed_log, direction, stream.ssrc),
+        LineStyle::kLine);
+    MovingAverage<LoggedRtpPacket, double>(CountPackets, stream.packet_view,
+                                           config, &time_series);
+    plot->AppendTimeSeries(std::move(time_series));
+  }
+  TimeSeries time_series(
+      std::string("RTCP ") + "(" + GetDirectionAsShortString(direction) + ")",
+      LineStyle::kLine);
+  if (direction == kIncomingPacket) {
+    MovingAverage<LoggedRtcpPacketIncoming, double>(
+        CountPackets, parsed_log.incoming_rtcp_packets(), config, &time_series);
+  } else {
+    MovingAverage<LoggedRtcpPacketOutgoing, double>(
+        CountPackets, parsed_log.outgoing_rtcp_packets(), config, &time_series);
+  }
+  plot->AppendTimeSeries(std::move(time_series));
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Packet Rate (packets/s)", kBottomMargin,
+                          kTopMargin);
+  plot->SetTitle("Rate of " + GetDirectionAsString(direction) +
+                 " RTP/RTCP packets");
+}
+
+void CreateTotalPacketRateGraph(PacketDirection direction,
+                                const ParsedRtcEventLog& parsed_log,
+                                const AnalyzerConfig& config,
+                                Plot* plot) {
+  // Contains a log timestamp to enable counting logged events of different
+  // types using MovingAverage().
+  class LogTime {
+   public:
+    explicit LogTime(Timestamp log_time) : log_time_(log_time) {}
+    Timestamp log_time() const { return log_time_; }
+
+   private:
+    Timestamp log_time_;
+  };
+  std::vector<LogTime> packet_times;
+  auto handle_rtp = [&packet_times](const LoggedRtpPacket& packet) {
+    packet_times.emplace_back(packet.log_time());
+  };
+  RtcEventProcessor process;
+  for (const auto& stream : parsed_log.rtp_packets_by_ssrc(direction)) {
+    process.AddEvents(stream.packet_view, handle_rtp, direction);
+  }
+  if (direction == kIncomingPacket) {
+    auto handle_incoming_rtcp =
+        [&packet_times](const LoggedRtcpPacketIncoming& packet) {
+          packet_times.emplace_back(packet.log_time());
+        };
+    process.AddEvents(parsed_log.incoming_rtcp_packets(), handle_incoming_rtcp);
+  } else {
+    auto handle_outgoing_rtcp =
+        [&packet_times](const LoggedRtcpPacketOutgoing& packet) {
+          packet_times.emplace_back(packet.log_time());
+        };
+    process.AddEvents(parsed_log.outgoing_rtcp_packets(), handle_outgoing_rtcp);
+  }
+  process.ProcessEventsInOrder();
+  TimeSeries time_series(std::string("Total ") + "(" +
+                             GetDirectionAsShortString(direction) + ") packets",
+                         LineStyle::kLine);
+  MovingAverage<LogTime, uint64_t>([](auto packet) { return 1; }, packet_times,
+                                   config, &time_series);
+  plot->AppendTimeSeries(std::move(time_series));
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Packet Rate (packets/s)", kBottomMargin,
+                          kTopMargin);
+  plot->SetTitle("Rate of all " + GetDirectionAsString(direction) +
+                 " RTP/RTCP packets");
+}
+
+// For each SSRC, plot the sequence number difference between consecutive
+// incoming packets.
+void CreateSequenceNumberGraph(const ParsedRtcEventLog& parsed_log,
+                               const AnalyzerConfig& config,
+                               Plot* plot) {
+  for (const auto& stream : parsed_log.incoming_rtp_packets_by_ssrc()) {
+    // Filter on SSRC.
+    if (!MatchingSsrc(stream.ssrc, config.desired_ssrc_)) {
+      continue;
+    }
+
+    TimeSeries time_series(
+        GetStreamName(parsed_log, kIncomingPacket, stream.ssrc),
+        LineStyle::kBar);
+    auto GetSequenceNumberDiff = [](const LoggedRtpPacketIncoming& old_packet,
+                                    const LoggedRtpPacketIncoming& new_packet) {
+      int64_t diff =
+          WrappingDifference(new_packet.rtp.header.sequenceNumber,
+                             old_packet.rtp.header.sequenceNumber, 1ul << 16);
+      return diff;
+    };
+    auto ToCallTime = [&config](const LoggedRtpPacketIncoming& packet) {
+      return config.GetCallTimeSec(packet.log_time());
+    };
+    ProcessPairs<LoggedRtpPacketIncoming, float>(
+        ToCallTime, GetSequenceNumberDiff, stream.incoming_packets,
+        &time_series);
+    plot->AppendTimeSeries(std::move(time_series));
+  }
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Difference since last packet", kBottomMargin,
+                          kTopMargin);
+  plot->SetTitle("Incoming sequence number delta");
+}
+
+void CreateIncomingPacketLossGraph(const ParsedRtcEventLog& parsed_log,
+                                   const AnalyzerConfig& config,
+                                   Plot* plot) {
+  for (const auto& stream : parsed_log.incoming_rtp_packets_by_ssrc()) {
+    const std::vector<LoggedRtpPacketIncoming>& packets =
+        stream.incoming_packets;
+    // Filter on SSRC.
+    if (!MatchingSsrc(stream.ssrc, config.desired_ssrc_) || packets.empty()) {
+      continue;
+    }
+
+    TimeSeries time_series(
+        GetStreamName(parsed_log, kIncomingPacket, stream.ssrc),
+        LineStyle::kLine, PointStyle::kHighlight);
+    // TODO(terelius): Should the window and step size be read from the class
+    // instead?
+    const TimeDelta kWindow = TimeDelta::Millis(1000);
+    const TimeDelta kStep = TimeDelta::Millis(1000);
+    SeqNumUnwrapper<uint16_t> unwrapper_;
+    SeqNumUnwrapper<uint16_t> prior_unwrapper_;
+    size_t window_index_begin = 0;
+    size_t window_index_end = 0;
+    uint64_t highest_seq_number =
+        unwrapper_.Unwrap(packets[0].rtp.header.sequenceNumber) - 1;
+    uint64_t highest_prior_seq_number =
+        prior_unwrapper_.Unwrap(packets[0].rtp.header.sequenceNumber) - 1;
+
+    for (Timestamp t = config.begin_time_; t < config.end_time_ + kStep;
+         t += kStep) {
+      while (window_index_end < packets.size() &&
+             packets[window_index_end].rtp.log_time() < t) {
+        uint64_t sequence_number = unwrapper_.Unwrap(
+            packets[window_index_end].rtp.header.sequenceNumber);
+        highest_seq_number = std::max(highest_seq_number, sequence_number);
+        ++window_index_end;
+      }
+      while (window_index_begin < packets.size() &&
+             packets[window_index_begin].rtp.log_time() < t - kWindow) {
+        uint64_t sequence_number = prior_unwrapper_.Unwrap(
+            packets[window_index_begin].rtp.header.sequenceNumber);
+        highest_prior_seq_number =
+            std::max(highest_prior_seq_number, sequence_number);
+        ++window_index_begin;
+      }
+      float x = config.GetCallTimeSec(t);
+      uint64_t expected_packets = highest_seq_number - highest_prior_seq_number;
+      if (expected_packets > 0) {
+        int64_t received_packets = window_index_end - window_index_begin;
+        int64_t lost_packets = expected_packets - received_packets;
+        float y = static_cast<float>(lost_packets) / expected_packets * 100;
+        time_series.points.emplace_back(x, y);
+      }
+    }
+    plot->AppendTimeSeries(std::move(time_series));
+  }
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Loss rate (in %)", kBottomMargin, kTopMargin);
+  plot->SetTitle("Incoming packet loss (derived from incoming packets)");
+}
+
+// For each SSRC, plot the bandwidth used by that stream.
+void CreateStreamBitrateGraph(PacketDirection direction,
+                              const ParsedRtcEventLog& parsed_log,
+                              const AnalyzerConfig& config,
+                              Plot* plot) {
+  for (const auto& stream : parsed_log.rtp_packets_by_ssrc(direction)) {
+    // Filter on SSRC.
+    if (!MatchingSsrc(stream.ssrc, config.desired_ssrc_)) {
+      continue;
+    }
+
+    TimeSeries time_series(GetStreamName(parsed_log, direction, stream.ssrc),
+                           LineStyle::kLine);
+    auto GetPacketSizeKilobits = [](const LoggedRtpPacket& packet) {
+      return packet.total_length * 8.0 / 1000.0;
+    };
+    MovingAverage<LoggedRtpPacket, double>(
+        GetPacketSizeKilobits, stream.packet_view, config, &time_series);
+    plot->AppendTimeSeries(std::move(time_series));
+  }
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin);
+  plot->SetTitle(GetDirectionAsString(direction) + " bitrate per stream");
+}
+
+// Plot the bitrate allocation for each temporal and spatial layer.
+// Computed from RTCP XR target bitrate block, so the graph is only populated if
+// those are sent.
+void CreateBitrateAllocationGraph(PacketDirection direction,
+                                  const ParsedRtcEventLog& parsed_log,
+                                  const AnalyzerConfig& config,
+                                  Plot* plot) {
+  std::map<LayerDescription, TimeSeries> time_series;
+  const auto& xr_list = parsed_log.extended_reports(direction);
+  for (const auto& rtcp : xr_list) {
+    const std::optional<rtcp::TargetBitrate>& target_bitrate =
+        rtcp.xr.target_bitrate();
+    if (!target_bitrate.has_value())
+      continue;
+    for (const auto& bitrate_item : target_bitrate->GetTargetBitrates()) {
+      LayerDescription layer(rtcp.xr.sender_ssrc(), bitrate_item.spatial_layer,
+                             bitrate_item.temporal_layer);
+      auto time_series_it = time_series.find(layer);
+      if (time_series_it == time_series.end()) {
+        std::string layer_name = GetLayerName(layer);
+        bool inserted;
+        std::tie(time_series_it, inserted) = time_series.insert(
+            std::make_pair(layer, TimeSeries(layer_name, LineStyle::kStep)));
+        RTC_DCHECK(inserted);
+      }
+      float x = config.GetCallTimeSec(rtcp.log_time());
+      float y = bitrate_item.target_bitrate_kbps;
+      time_series_it->second.points.emplace_back(x, y);
+    }
+  }
+  for (auto& layer : time_series) {
+    plot->AppendTimeSeries(std::move(layer.second));
+  }
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin);
+  if (direction == kIncomingPacket)
+    plot->SetTitle("Target bitrate per incoming layer");
+  else
+    plot->SetTitle("Target bitrate per outgoing layer");
+}
+
+void CreateEcnFeedbackGraph(Plot* plot,
+                            PacketDirection direction,
+                            const ParsedRtcEventLog& parsed_log,
+                            const AnalyzerConfig& config) {
+  TimeSeries not_ect("Not ECN capable", LineStyle::kBar,
+                     PointStyle::kHighlight);
+  TimeSeries ect_1("ECN capable", LineStyle::kBar, PointStyle::kHighlight);
+  TimeSeries ce("Congestion experienced", LineStyle::kBar,
+                PointStyle::kHighlight);
+
+  for (const LoggedRtcpCongestionControlFeedback& feedback :
+       parsed_log.congestion_feedback(direction)) {
+    int ect_1_count = 0;
+    int not_ect_count = 0;
+    int ce_count = 0;
+
+    for (const rtcp::CongestionControlFeedback::PacketInfo& info :
+         feedback.congestion_feedback.packets()) {
+      switch (info.ecn) {
+        case EcnMarking::kNotEct:
+          ++not_ect_count;
+          break;
+        case EcnMarking::kEct1:
+          ++ect_1_count;
+          break;
+        case EcnMarking::kEct0:
+          RTC_LOG(LS_ERROR) << "unexpected ect(0)";
+          break;
+        case EcnMarking::kCe:
+          ++ce_count;
+          break;
+      }
+    }
+    ect_1.points.emplace_back(config.GetCallTimeSec(feedback.timestamp),
+                              ect_1_count);
+    not_ect.points.emplace_back(config.GetCallTimeSec(feedback.timestamp),
+                                not_ect_count);
+    ce.points.emplace_back(config.GetCallTimeSec(feedback.timestamp), ce_count);
+  }
+
+  plot->AppendTimeSeriesIfNotEmpty(std::move(ect_1));
+  plot->AppendTimeSeriesIfNotEmpty(std::move(not_ect));
+  plot->AppendTimeSeriesIfNotEmpty(std::move(ce));
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 10, "Count per feedback", kBottomMargin,
+                          kTopMargin);
+}
+
+void CreateOutgoingEcnFeedbackGraph(const ParsedRtcEventLog& parsed_log,
+                                    const AnalyzerConfig& config,
+                                    Plot* plot) {
+  CreateEcnFeedbackGraph(plot, kOutgoingPacket, parsed_log, config);
+  plot->SetTitle("Outgoing ECN count per feedback");
+}
+
+void CreateIncomingEcnFeedbackGraph(const ParsedRtcEventLog& parsed_log,
+                                    const AnalyzerConfig& config,
+                                    Plot* plot) {
+  CreateEcnFeedbackGraph(plot, kIncomingPacket, parsed_log, config);
+  plot->SetTitle("Incoming ECN count per feedback");
+}
+
+void CreateOutgoingLossRateGraph(const ParsedRtcEventLog& parsed_log,
+                                 const AnalyzerConfig& config,
+                                 Plot* plot) {
+  struct PacketLossPerFeedback {
+    Timestamp timestamp;              // Time when this feedback was received.
+    int num_packets_in_feedback = 0;  // Includes lost packets.
+    int num_lost_packets = 0;         // In this specific feedback.
+    int num_reordered_packets = 0;    // Packets received in this feedback, but
+                                      // was previously reported as lost.
+    int num_missing_feedback =
+        0;  // Packets missing feedback between this report and the previous.
+  };
+
+  class LossFeedbackBuilder {
+   public:
+    void AddPacket(uint16_t sequence_number, TimeDelta arrival_time_delta) {
+      last_unwrapped_sequence_number_ =
+          sequence_number_unwrapper_.Unwrap(sequence_number);
+      if (!first_sequence_number_.has_value()) {
+        first_sequence_number_ = last_unwrapped_sequence_number_;
+      }
+      ++num_packets_;
+      if (arrival_time_delta.IsInfinite()) {
+        lost_sequence_numbers_.insert(last_unwrapped_sequence_number_);
+      } else {
+        num_reordered_packets_ += previous_lost_sequence_numbers_.count(
+            last_unwrapped_sequence_number_);
+      }
+    }
+
+    void Update(PacketLossPerFeedback& feedback) {
+      feedback.num_packets_in_feedback += num_packets_;
+      feedback.num_lost_packets += lost_sequence_numbers_.size();
+      feedback.num_reordered_packets += num_reordered_packets_;
+      if (first_sequence_number_.has_value() &&
+          previous_feedback_highest_seq_number_.has_value()) {
+        feedback.num_missing_feedback +=
+            *first_sequence_number_ - *previous_feedback_highest_seq_number_ -
+            1;
+      }
+
+      // Prepare for next feedback.
+      first_sequence_number_ = std::nullopt;
+      previous_lost_sequence_numbers_.insert(lost_sequence_numbers_.begin(),
+                                             lost_sequence_numbers_.end());
+      previous_feedback_highest_seq_number_ = last_unwrapped_sequence_number_;
+      lost_sequence_numbers_.clear();
+      num_reordered_packets_ = 0;
+      num_packets_ = 0;
+    }
+
+   private:
+    int64_t last_unwrapped_sequence_number_ = 0;
+    int num_reordered_packets_ = 0;
+    int num_packets_ = 0;
+    std::optional<int64_t> first_sequence_number_;
+
+    std::unordered_set<int64_t> lost_sequence_numbers_;
+    std::unordered_set<int64_t> previous_lost_sequence_numbers_;
+    std::optional<int64_t> previous_feedback_highest_seq_number_;
+
+    RtpSequenceNumberUnwrapper sequence_number_unwrapper_;
+  };
+
+  TimeSeries loss_rate_series("Loss rate (from packet feedback)",
+                              LineStyle::kLine, PointStyle::kHighlight);
+  TimeSeries reordered_packets_between_feedback(
+      "Ratio of reordered packets from last feedback", LineStyle::kLine,
+      PointStyle::kHighlight);
+  TimeSeries average_loss_rate_series("Average loss rate last 5s",
+                                      LineStyle::kLine, PointStyle::kHighlight);
+  TimeSeries missing_feedback_series("Missing feedback", LineStyle::kNone,
+                                     PointStyle::kHighlight);
+
+  std::vector<PacketLossPerFeedback> loss_per_feedback;
+
+  if (!parsed_log.congestion_feedback(kIncomingPacket).empty()) {
+    plot->SetTitle("Outgoing loss rate (from CCFB)");
+
+    std::map</*ssrc*/ uint32_t, LossFeedbackBuilder> per_ssrc_builder;
+    for (const LoggedRtcpCongestionControlFeedback& feedback :
+         parsed_log.congestion_feedback(kIncomingPacket)) {
+      const rtcp::CongestionControlFeedback& transport_feedback =
+          feedback.congestion_feedback;
+
+      PacketLossPerFeedback packet_loss_per_feedback = {
+          .timestamp = feedback.log_time()};
+      for (const rtcp::CongestionControlFeedback::PacketInfo& packet :
+           transport_feedback.packets()) {
+        per_ssrc_builder[packet.ssrc].AddPacket(packet.sequence_number,
+                                                packet.arrival_time_offset);
+      }
+      for (auto& [ssrc, builder] : per_ssrc_builder) {
+        builder.Update(packet_loss_per_feedback);
+      }
+      loss_per_feedback.push_back(packet_loss_per_feedback);
+    }
+  } else if (!parsed_log.transport_feedbacks(kIncomingPacket).empty()) {
+    plot->SetTitle("Outgoing loss rate (from TWCC)");
+
+    LossFeedbackBuilder builder;
+    for (const LoggedRtcpPacketTransportFeedback& feedback :
+         parsed_log.transport_feedbacks(kIncomingPacket)) {
+      feedback.transport_feedback.ForAllPackets(
+          [&](uint16_t sequence_number, TimeDelta receive_time_delta) {
+            builder.AddPacket(sequence_number, receive_time_delta);
+          });
+      PacketLossPerFeedback packet_loss_per_feedback = {
+          .timestamp = feedback.log_time()};
+      builder.Update(packet_loss_per_feedback);
+      loss_per_feedback.push_back(packet_loss_per_feedback);
+    }
+  }
+
+  PacketLossSummary window_summary;
+  Timestamp last_observation_receive_time = Timestamp::Zero();
+
+  // Use loss based bwe 2 observation duration and observation window size.
+  constexpr TimeDelta kObservationDuration = TimeDelta::Millis(250);
+  constexpr uint32_t kObservationWindowSize = 20;
+  std::deque<PacketLossSummary> observations;
+  int previous_feedback_size = 0;
+  for (const PacketLossPerFeedback& feedback : loss_per_feedback) {
+    for (int64_t num = 0; num < feedback.num_missing_feedback; ++num) {
+      missing_feedback_series.points.emplace_back(
+          config.GetCallTimeSec(feedback.timestamp), 100 + num);
+    }
+
+    // Compute loss rate from the transport feedback.
+    float loss_rate = static_cast<float>(feedback.num_lost_packets * 100.0 /
+                                         feedback.num_packets_in_feedback);
+
+    loss_rate_series.points.emplace_back(
+        config.GetCallTimeSec(feedback.timestamp), loss_rate);
+    float reordered_rate =
+        previous_feedback_size == 0
+            ? 0
+            : static_cast<float>(feedback.num_reordered_packets * 100.0 /
+                                 previous_feedback_size);
+    previous_feedback_size = feedback.num_packets_in_feedback;
+    reordered_packets_between_feedback.points.emplace_back(
+        config.GetCallTimeSec(feedback.timestamp), reordered_rate);
+
+    // Compute loss rate in a window of kObservationWindowSize.
+    if (window_summary.num_packets == 0) {
+      window_summary.base_time = feedback.timestamp;
+    }
+    window_summary.num_packets += feedback.num_packets_in_feedback;
+    window_summary.num_lost_packets +=
+        feedback.num_lost_packets - feedback.num_reordered_packets;
+
+    const Timestamp last_received_time = feedback.timestamp;
+    const TimeDelta observation_duration =
+        window_summary.base_time == Timestamp::Zero()
+            ? TimeDelta::Zero()
+            : last_received_time - window_summary.base_time;
+    if (observation_duration > kObservationDuration) {
+      last_observation_receive_time = last_received_time;
+      observations.push_back(window_summary);
+      if (observations.size() > kObservationWindowSize) {
+        observations.pop_front();
+      }
+
+      // Compute average loss rate in a number of windows.
+      int total_packets = 0;
+      int total_loss = 0;
+      for (const auto& observation : observations) {
+        total_loss += observation.num_lost_packets;
+        total_packets += observation.num_packets;
+      }
+      if (total_packets > 0) {
+        float average_loss_rate = total_loss * 100.0 / total_packets;
+        average_loss_rate_series.points.emplace_back(
+            config.GetCallTimeSec(feedback.timestamp), average_loss_rate);
+      } else {
+        average_loss_rate_series.points.emplace_back(
+            config.GetCallTimeSec(feedback.timestamp), 0);
+      }
+      window_summary = PacketLossSummary();
+    }
+  }
+  // Add the data set to the plot.
+  plot->AppendTimeSeriesIfNotEmpty(std::move(loss_rate_series));
+  plot->AppendTimeSeriesIfNotEmpty(
+      std::move(reordered_packets_between_feedback));
+  plot->AppendTimeSeriesIfNotEmpty(std::move(average_loss_rate_series));
+  plot->AppendTimeSeriesIfNotEmpty(std::move(missing_feedback_series));
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 100, "Loss rate (percent)", kBottomMargin,
+                          kTopMargin);
+}
+
+void CreateTimestampGraph(PacketDirection direction,
+                          const ParsedRtcEventLog& parsed_log,
+                          const AnalyzerConfig& config,
+                          Plot* plot) {
+  for (const auto& stream : parsed_log.rtp_packets_by_ssrc(direction)) {
+    TimeSeries rtp_timestamps(
+        GetStreamName(parsed_log, direction, stream.ssrc) + " capture-time",
+        LineStyle::kLine, PointStyle::kHighlight);
+    for (const auto& packet : stream.packet_view) {
+      float x = config.GetCallTimeSec(packet.log_time());
+      float y = packet.header.timestamp;
+      rtp_timestamps.points.emplace_back(x, y);
+    }
+    plot->AppendTimeSeries(std::move(rtp_timestamps));
+
+    TimeSeries rtcp_timestamps(
+        GetStreamName(parsed_log, direction, stream.ssrc) +
+            " rtcp capture-time",
+        LineStyle::kLine, PointStyle::kHighlight);
+    // TODO(terelius): Why only sender reports?
+    const auto& sender_reports = parsed_log.sender_reports(direction);
+    for (const auto& rtcp : sender_reports) {
+      if (rtcp.sr.sender_ssrc() != stream.ssrc)
+        continue;
+      float x = config.GetCallTimeSec(rtcp.log_time());
+      float y = rtcp.sr.rtp_timestamp();
+      rtcp_timestamps.points.emplace_back(x, y);
+    }
+    plot->AppendTimeSeriesIfNotEmpty(std::move(rtcp_timestamps));
+  }
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "RTP timestamp", kBottomMargin, kTopMargin);
+  plot->SetTitle(GetDirectionAsString(direction) + " timestamps");
+}
+
+void CreateSenderAndReceiverReportPlot(
+    PacketDirection direction,
+    FunctionView<float(const rtcp::ReportBlock&)> fy,
+    std::string title,
+    std::string yaxis_label,
+    const ParsedRtcEventLog& parsed_log,
+    const AnalyzerConfig& config,
+    Plot* plot) {
+  std::map<uint32_t, TimeSeries> sr_reports_by_ssrc;
+  const auto& sender_reports = parsed_log.sender_reports(direction);
+  for (const auto& rtcp : sender_reports) {
+    float x = config.GetCallTimeSec(rtcp.log_time());
+    uint32_t ssrc = rtcp.sr.sender_ssrc();
+    for (const auto& block : rtcp.sr.report_blocks()) {
+      float y = fy(block);
+      auto sr_report_it = sr_reports_by_ssrc.find(ssrc);
+      bool inserted;
+      if (sr_report_it == sr_reports_by_ssrc.end()) {
+        std::tie(sr_report_it, inserted) = sr_reports_by_ssrc.emplace(
+            ssrc, TimeSeries(GetStreamName(parsed_log, direction, ssrc) +
+                                 " Sender Reports",
+                             LineStyle::kLine, PointStyle::kHighlight));
+      }
+      sr_report_it->second.points.emplace_back(x, y);
+    }
+  }
+  for (auto& kv : sr_reports_by_ssrc) {
+    plot->AppendTimeSeries(std::move(kv.second));
+  }
+
+  std::map<uint32_t, TimeSeries> rr_reports_by_ssrc;
+  const auto& receiver_reports = parsed_log.receiver_reports(direction);
+  for (const auto& rtcp : receiver_reports) {
+    float x = config.GetCallTimeSec(rtcp.log_time());
+    uint32_t ssrc = rtcp.rr.sender_ssrc();
+    for (const auto& block : rtcp.rr.report_blocks()) {
+      float y = fy(block);
+      auto rr_report_it = rr_reports_by_ssrc.find(ssrc);
+      bool inserted;
+      if (rr_report_it == rr_reports_by_ssrc.end()) {
+        std::tie(rr_report_it, inserted) = rr_reports_by_ssrc.emplace(
+            ssrc, TimeSeries(GetStreamName(parsed_log, direction, ssrc) +
+                                 " Receiver Reports",
+                             LineStyle::kLine, PointStyle::kHighlight));
+      }
+      rr_report_it->second.points.emplace_back(x, y);
+    }
+  }
+  for (auto& kv : rr_reports_by_ssrc) {
+    plot->AppendTimeSeries(std::move(kv.second));
+  }
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, yaxis_label, kBottomMargin, kTopMargin);
+  plot->SetTitle(title);
+}
+
+}  // namespace webrtc
diff --git a/rtc_tools/rtc_event_log_visualizer/analyze_rtp_rtcp.h b/rtc_tools/rtc_event_log_visualizer/analyze_rtp_rtcp.h
new file mode 100644
index 0000000000..2865023e60
--- /dev/null
+++ b/rtc_tools/rtc_event_log_visualizer/analyze_rtp_rtcp.h
@@ -0,0 +1,100 @@
+/*
+ *  Copyright (c) 2026 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZE_RTP_RTCP_H_
+#define RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZE_RTP_RTCP_H_
+
+#include <string>
+
+#include "api/function_view.h"
+#include "logging/rtc_event_log/rtc_event_log_parser.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
+#include "rtc_tools/rtc_event_log_visualizer/analyzer_common.h"
+#include "rtc_tools/rtc_event_log_visualizer/plot_base.h"
+
+namespace webrtc {
+
+float GetHighestSeqNumber(const rtcp::ReportBlock& block);
+float GetFractionLost(const rtcp::ReportBlock& block);
+float GetCumulativeLost(const rtcp::ReportBlock& block);
+float DelaySinceLastSr(const rtcp::ReportBlock& block);
+
+void CreatePacketGraph(PacketDirection direction,
+                       const ParsedRtcEventLog& parsed_log,
+                       const AnalyzerConfig& config,
+                       Plot* plot);
+
+void CreateRtcpTypeGraph(PacketDirection direction,
+                         const ParsedRtcEventLog& parsed_log,
+                         const AnalyzerConfig& config,
+                         Plot* plot);
+
+void CreateAccumulatedPacketsGraph(PacketDirection direction,
+                                   const ParsedRtcEventLog& parsed_log,
+                                   const AnalyzerConfig& config,
+                                   Plot* plot);
+
+void CreatePacketRateGraph(PacketDirection direction,
+                           const ParsedRtcEventLog& parsed_log,
+                           const AnalyzerConfig& config,
+                           Plot* plot);
+
+void CreateTotalPacketRateGraph(PacketDirection direction,
+                                const ParsedRtcEventLog& parsed_log,
+                                const AnalyzerConfig& config,
+                                Plot* plot);
+
+void CreateSequenceNumberGraph(const ParsedRtcEventLog& parsed_log,
+                               const AnalyzerConfig& config,
+                               Plot* plot);
+
+void CreateIncomingPacketLossGraph(const ParsedRtcEventLog& parsed_log,
+                                   const AnalyzerConfig& config,
+                                   Plot* plot);
+
+void CreateStreamBitrateGraph(PacketDirection direction,
+                              const ParsedRtcEventLog& parsed_log,
+                              const AnalyzerConfig& config,
+                              Plot* plot);
+
+void CreateBitrateAllocationGraph(PacketDirection direction,
+                                  const ParsedRtcEventLog& parsed_log,
+                                  const AnalyzerConfig& config,
+                                  Plot* plot);
+
+void CreateOutgoingEcnFeedbackGraph(const ParsedRtcEventLog& parsed_log,
+                                    const AnalyzerConfig& config,
+                                    Plot* plot);
+
+void CreateIncomingEcnFeedbackGraph(const ParsedRtcEventLog& parsed_log,
+                                    const AnalyzerConfig& config,
+                                    Plot* plot);
+
+void CreateOutgoingLossRateGraph(const ParsedRtcEventLog& parsed_log,
+                                 const AnalyzerConfig& config,
+                                 Plot* plot);
+
+void CreateTimestampGraph(PacketDirection direction,
+                          const ParsedRtcEventLog& parsed_log,
+                          const AnalyzerConfig& config,
+                          Plot* plot);
+
+void CreateSenderAndReceiverReportPlot(
+    PacketDirection direction,
+    FunctionView<float(const rtcp::ReportBlock&)> fy,
+    std::string title,
+    std::string yaxis_label,
+    const ParsedRtcEventLog& parsed_log,
+    const AnalyzerConfig& config,
+    Plot* plot);
+
+}  // namespace webrtc
+
+#endif  // RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZE_RTP_RTCP_H_
diff --git a/rtc_tools/rtc_event_log_visualizer/analyzer.cc b/rtc_tools/rtc_event_log_visualizer/analyzer.cc
index cc7a5457bd..73e1db7509 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyzer.cc
+++ b/rtc_tools/rtc_event_log_visualizer/analyzer.cc
@@ -10,476 +10,35 @@
 
 #include "rtc_tools/rtc_event_log_visualizer/analyzer.h"
 
-#include <algorithm>
-#include <cmath>
 #include <cstddef>
-#include <cstdint>
-#include <deque>
-#include <limits>
-#include <map>
-#include <memory>
-#include <optional>
 #include <string>
-#include <tuple>
-#include <unordered_set>
-#include <utility>
 #include <vector>
 
 #include "absl/algorithm/container.h"
-#include "absl/functional/bind_front.h"
 #include "absl/strings/string_view.h"
-#include "api/candidate.h"
-#include "api/dtls_transport_interface.h"
-#include "api/environment/environment.h"
 #include "api/environment/environment_factory.h"
-#include "api/field_trials.h"
 #include "api/function_view.h"
-#include "api/media_types.h"
-#include "api/rtp_headers.h"
-#include "api/transport/bandwidth_usage.h"
-#include "api/transport/ecn_marking.h"
-#include "api/transport/goog_cc_factory.h"
-#include "api/transport/network_control.h"
-#include "api/transport/network_types.h"
 #include "api/units/data_rate.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
-#include "logging/rtc_event_log/events/logged_rtp_rtcp.h"
-#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair.h"
-#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.h"
 #include "logging/rtc_event_log/rtc_event_log_parser.h"
-#include "logging/rtc_event_log/rtc_event_processor.h"
-#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.h"
-#include "modules/congestion_controller/include/receive_side_congestion_controller.h"
-#include "modules/congestion_controller/rtp/transport_feedback_adapter.h"
-#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
-#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
-#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
-#include "modules/rtp_rtcp/source/rtcp_packet/congestion_control_feedback.h"
-#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
-#include "modules/rtp_rtcp/source/rtcp_packet/remb.h"
 #include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
-#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
-#include "modules/rtp_rtcp/source/rtcp_packet/target_bitrate.h"
-#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
-#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
-#include "modules/rtp_rtcp/source/rtp_packet_received.h"
-#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
-#include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
-#include "rtc_base/network/sent_packet.h"
-#include "rtc_base/numerics/sequence_number_unwrapper.h"
-#include "rtc_base/rate_statistics.h"
-#include "rtc_base/strings/string_builder.h"
 #include "rtc_tools/rtc_event_log_visualizer/analyze_audio.h"
+#include "rtc_tools/rtc_event_log_visualizer/analyze_bwe.h"
+#include "rtc_tools/rtc_event_log_visualizer/analyze_connectivity.h"
+#include "rtc_tools/rtc_event_log_visualizer/analyze_rtp_rtcp.h"
 #include "rtc_tools/rtc_event_log_visualizer/analyzer_common.h"
-#include "rtc_tools/rtc_event_log_visualizer/log_scream_simulation.h"
-#include "rtc_tools/rtc_event_log_visualizer/log_simulation.h"
 #include "rtc_tools/rtc_event_log_visualizer/plot_base.h"
-#include "system_wrappers/include/clock.h"
 
 namespace webrtc {
 
-namespace {
-
-std::string SsrcToString(uint32_t ssrc) {
-  StringBuilder ss;
-  ss << "SSRC " << ssrc;
-  return ss.Release();
-}
-
-// Checks whether an SSRC is contained in the list of desired SSRCs.
-// Note that an empty SSRC list matches every SSRC.
-bool MatchingSsrc(uint32_t ssrc, const std::vector<uint32_t>& desired_ssrc) {
-  if (desired_ssrc.empty())
-    return true;
-  return std::find(desired_ssrc.begin(), desired_ssrc.end(), ssrc) !=
-         desired_ssrc.end();
-}
-
-double AbsSendTimeToMicroseconds(int64_t abs_send_time) {
-  // The timestamp is a fixed point representation with 6 bits for seconds
-  // and 18 bits for fractions of a second. Thus, we divide by 2^18 to get the
-  // time in seconds and then multiply by kNumMicrosecsPerSec to convert to
-  // microseconds.
-  static constexpr double kTimestampToMicroSec =
-      static_cast<double>(kNumMicrosecsPerSec) / static_cast<double>(1ul << 18);
-  return abs_send_time * kTimestampToMicroSec;
-}
-
-// Computes the difference `later` - `earlier` where `later` and `earlier`
-// are counters that wrap at `modulus`. The difference is chosen to have the
-// least absolute value. For example if `modulus` is 8, then the difference will
-// be chosen in the range [-3, 4]. If `modulus` is 9, then the difference will
-// be in [-4, 4].
-int64_t WrappingDifference(uint32_t later, uint32_t earlier, int64_t modulus) {
-  RTC_DCHECK_LE(1, modulus);
-  RTC_DCHECK_LT(later, modulus);
-  RTC_DCHECK_LT(earlier, modulus);
-  int64_t difference =
-      static_cast<int64_t>(later) - static_cast<int64_t>(earlier);
-  int64_t max_difference = modulus / 2;
-  int64_t min_difference = max_difference - modulus + 1;
-  if (difference > max_difference) {
-    difference -= modulus;
-  }
-  if (difference < min_difference) {
-    difference += modulus;
-  }
-  if (difference > max_difference / 2 || difference < min_difference / 2) {
-    RTC_LOG(LS_WARNING) << "Difference between" << later << " and " << earlier
-                        << " expected to be in the range ("
-                        << min_difference / 2 << "," << max_difference / 2
-                        << ") but is " << difference
-                        << ". Correct unwrapping is uncertain.";
-  }
-  return difference;
-}
-
-// This is much more reliable for outgoing streams than for incoming streams.
-template <typename RtpPacketContainer>
-std::optional<uint32_t> EstimateRtpClockFrequency(
-    const RtpPacketContainer& packets,
-    int64_t end_time_us) {
-  RTC_CHECK(packets.size() >= 2);
-  SeqNumUnwrapper<uint32_t> unwrapper;
-  int64_t first_rtp_timestamp =
-      unwrapper.Unwrap(packets[0].rtp.header.timestamp);
-  int64_t first_log_timestamp = packets[0].log_time_us();
-  int64_t last_rtp_timestamp = first_rtp_timestamp;
-  int64_t last_log_timestamp = first_log_timestamp;
-  for (size_t i = 1; i < packets.size(); i++) {
-    if (packets[i].log_time_us() > end_time_us)
-      break;
-    last_rtp_timestamp = unwrapper.Unwrap(packets[i].rtp.header.timestamp);
-    last_log_timestamp = packets[i].log_time_us();
-  }
-  if (last_log_timestamp - first_log_timestamp < kNumMicrosecsPerSec) {
-    RTC_LOG(LS_WARNING)
-        << "Failed to estimate RTP clock frequency: Stream too short. ("
-        << packets.size() << " packets, "
-        << last_log_timestamp - first_log_timestamp << " us)";
-    return std::nullopt;
-  }
-  double duration =
-      static_cast<double>(last_log_timestamp - first_log_timestamp) /
-      kNumMicrosecsPerSec;
-  double estimated_frequency =
-      (last_rtp_timestamp - first_rtp_timestamp) / duration;
-  for (uint32_t f : {8000, 16000, 32000, 48000, 90000}) {
-    if (std::fabs(estimated_frequency - f) < 0.15 * f) {
-      return f;
-    }
-  }
-  RTC_LOG(LS_WARNING) << "Failed to estimate RTP clock frequency: Estimate "
-                      << estimated_frequency
-                      << " not close to any standard RTP frequency."
-                      << " Last timestamp " << last_rtp_timestamp
-                      << " first timestamp " << first_rtp_timestamp;
-  return std::nullopt;
-}
-
-std::optional<double> NetworkDelayDiff_AbsSendTime(
-    const LoggedRtpPacketIncoming& old_packet,
-    const LoggedRtpPacketIncoming& new_packet) {
-  if (old_packet.rtp.header.extension.hasAbsoluteSendTime &&
-      new_packet.rtp.header.extension.hasAbsoluteSendTime) {
-    int64_t send_time_diff = WrappingDifference(
-        new_packet.rtp.header.extension.absoluteSendTime,
-        old_packet.rtp.header.extension.absoluteSendTime, 1ul << 24);
-    int64_t recv_time_diff =
-        new_packet.log_time_us() - old_packet.log_time_us();
-    double delay_change_us =
-        recv_time_diff - AbsSendTimeToMicroseconds(send_time_diff);
-    return delay_change_us / 1000;
-  } else {
-    return std::nullopt;
-  }
-}
-
-std::optional<double> NetworkDelayDiff_CaptureTime(
-    const LoggedRtpPacketIncoming& old_packet,
-    const LoggedRtpPacketIncoming& new_packet,
-    const double sample_rate) {
-  int64_t send_time_diff =
-      WrappingDifference(new_packet.rtp.header.timestamp,
-                         old_packet.rtp.header.timestamp, 1ull << 32);
-  int64_t recv_time_diff = new_packet.log_time_us() - old_packet.log_time_us();
-
-  double delay_change =
-      static_cast<double>(recv_time_diff) / 1000 -
-      static_cast<double>(send_time_diff) / sample_rate * 1000;
-  if (delay_change < -10000 || 10000 < delay_change) {
-    RTC_LOG(LS_WARNING) << "Very large delay change. Timestamps correct?";
-    RTC_LOG(LS_WARNING) << "Old capture time "
-                        << old_packet.rtp.header.timestamp << ", received time "
-                        << old_packet.log_time_us();
-    RTC_LOG(LS_WARNING) << "New capture time "
-                        << new_packet.rtp.header.timestamp << ", received time "
-                        << new_packet.log_time_us();
-    RTC_LOG(LS_WARNING) << "Receive time difference " << recv_time_diff << " = "
-                        << static_cast<double>(recv_time_diff) /
-                               kNumMicrosecsPerSec
-                        << "s";
-    RTC_LOG(LS_WARNING) << "Send time difference " << send_time_diff << " = "
-                        << static_cast<double>(send_time_diff) / sample_rate
-                        << "s";
-  }
-  return delay_change;
-}
-
-template <typename T>
-TimeSeries CreateRtcpTypeTimeSeries(const std::vector<T>& rtcp_list,
-                                    AnalyzerConfig config,
-                                    std::string rtcp_name,
-                                    int category_id) {
-  TimeSeries time_series(rtcp_name, LineStyle::kNone, PointStyle::kHighlight);
-  for (const auto& rtcp : rtcp_list) {
-    float x = config.GetCallTimeSec(rtcp.timestamp);
-    float y = category_id;
-    time_series.points.emplace_back(x, y);
-  }
-  return time_series;
-}
-
-const char kUnknownEnumValue[] = "unknown";
-
-// TODO(tommi): This should be "host".
-const char kIceCandidateTypeLocal[] = "local";
-// TODO(tommi): This should be "srflx".
-const char kIceCandidateTypeStun[] = "stun";
-const char kIceCandidateTypePrflx[] = "prflx";
-const char kIceCandidateTypeRelay[] = "relay";
-
-const char kProtocolUdp[] = "udp";
-const char kProtocolTcp[] = "tcp";
-const char kProtocolSsltcp[] = "ssltcp";
-const char kProtocolTls[] = "tls";
-
-const char kAddressFamilyIpv4[] = "ipv4";
-const char kAddressFamilyIpv6[] = "ipv6";
-
-const char kNetworkTypeEthernet[] = "ethernet";
-const char kNetworkTypeLoopback[] = "loopback";
-const char kNetworkTypeWifi[] = "wifi";
-const char kNetworkTypeVpn[] = "vpn";
-const char kNetworkTypeCellular[] = "cellular";
-
-absl::string_view GetIceCandidateTypeAsString(IceCandidateType type) {
-  switch (type) {
-    case IceCandidateType::kHost:
-      return kIceCandidateTypeLocal;
-    case IceCandidateType::kSrflx:
-      return kIceCandidateTypeStun;
-    case IceCandidateType::kPrflx:
-      return kIceCandidateTypePrflx;
-    case IceCandidateType::kRelay:
-      return kIceCandidateTypeRelay;
-    default:
-      RTC_DCHECK_NOTREACHED();
-      return kUnknownEnumValue;
-  }
-}
-
-std::string GetProtocolAsString(IceCandidatePairProtocol protocol) {
-  switch (protocol) {
-    case IceCandidatePairProtocol::kUdp:
-      return kProtocolUdp;
-    case IceCandidatePairProtocol::kTcp:
-      return kProtocolTcp;
-    case IceCandidatePairProtocol::kSsltcp:
-      return kProtocolSsltcp;
-    case IceCandidatePairProtocol::kTls:
-      return kProtocolTls;
-    default:
-      return kUnknownEnumValue;
-  }
-}
-
-std::string GetAddressFamilyAsString(IceCandidatePairAddressFamily family) {
-  switch (family) {
-    case IceCandidatePairAddressFamily::kIpv4:
-      return kAddressFamilyIpv4;
-    case IceCandidatePairAddressFamily::kIpv6:
-      return kAddressFamilyIpv6;
-    default:
-      return kUnknownEnumValue;
-  }
-}
-
-std::string GetNetworkTypeAsString(IceCandidateNetworkType type) {
-  switch (type) {
-    case IceCandidateNetworkType::kEthernet:
-      return kNetworkTypeEthernet;
-    case IceCandidateNetworkType::kLoopback:
-      return kNetworkTypeLoopback;
-    case IceCandidateNetworkType::kWifi:
-      return kNetworkTypeWifi;
-    case IceCandidateNetworkType::kVpn:
-      return kNetworkTypeVpn;
-    case IceCandidateNetworkType::kCellular:
-      return kNetworkTypeCellular;
-    default:
-      return kUnknownEnumValue;
-  }
-}
-
-std::string GetCandidatePairLogDescriptionAsString(
-    const LoggedIceCandidatePairConfig& config) {
-  // Example: stun:wifi->relay(tcp):cellular@udp:ipv4
-  // represents a pair of a local server-reflexive candidate on a WiFi network
-  // and a remote relay candidate using TCP as the relay protocol on a cell
-  // network, when the candidate pair communicates over UDP using IPv4.
-  StringBuilder ss;
-  ss << GetIceCandidateTypeAsString(config.local_candidate_type);
-
-  if (config.local_candidate_type == IceCandidateType::kRelay) {
-    ss << "(" << GetProtocolAsString(config.local_relay_protocol) << ")";
-  }
-
-  ss << ":" << GetNetworkTypeAsString(config.local_network_type) << ":"
-     << GetAddressFamilyAsString(config.local_address_family) << "->"
-     << GetIceCandidateTypeAsString(config.remote_candidate_type) << ":"
-     << GetAddressFamilyAsString(config.remote_address_family) << "@"
-     << GetProtocolAsString(config.candidate_pair_protocol);
-  return ss.Release();
-}
-
-std::string GetDirectionAsString(PacketDirection direction) {
-  if (direction == kIncomingPacket) {
-    return "Incoming";
-  } else {
-    return "Outgoing";
-  }
-}
-
-std::string GetDirectionAsShortString(PacketDirection direction) {
-  if (direction == kIncomingPacket) {
-    return "In";
-  } else {
-    return "Out";
-  }
-}
-
-struct FakeExtensionSmall {
-  static constexpr RTPExtensionType kId = kRtpExtensionMid;
-  static constexpr absl::string_view Uri() { return "fake-extension-small"; }
-};
-struct FakeExtensionLarge {
-  static constexpr RTPExtensionType kId = kRtpExtensionRtpStreamId;
-  static constexpr absl::string_view Uri() { return "fake-extension-large"; }
-};
-
-RtpPacketReceived RtpPacketForBWEFromHeader(const RTPHeader& header) {
-  RtpHeaderExtensionMap rtp_header_extensions(/*extmap_allow_mixed=*/true);
-  // ReceiveSideCongestionController doesn't need to know extensions ids as
-  // long as it able to get extensions by type. So any ids would work here.
-  rtp_header_extensions.Register<TransmissionOffset>(1);
-  rtp_header_extensions.Register<AbsoluteSendTime>(2);
-  rtp_header_extensions.Register<TransportSequenceNumber>(3);
-  rtp_header_extensions.Register<FakeExtensionSmall>(4);
-  // Use id > 14 to force two byte header per rtp header when this one is used.
-  rtp_header_extensions.Register<FakeExtensionLarge>(16);
-
-  RtpPacketReceived rtp_packet(&rtp_header_extensions);
-  // Set only fields that might be relevant for the bandwidth estimatior.
-  rtp_packet.SetSsrc(header.ssrc);
-  rtp_packet.SetTimestamp(header.timestamp);
-  size_t num_bwe_extensions = 0;
-  if (header.extension.hasTransmissionTimeOffset) {
-    rtp_packet.SetExtension<TransmissionOffset>(
-        header.extension.transmissionTimeOffset);
-    ++num_bwe_extensions;
-  }
-  if (header.extension.hasAbsoluteSendTime) {
-    rtp_packet.SetExtension<AbsoluteSendTime>(
-        header.extension.absoluteSendTime);
-    ++num_bwe_extensions;
-  }
-  if (header.extension.hasTransportSequenceNumber) {
-    rtp_packet.SetExtension<TransportSequenceNumber>(
-        header.extension.transportSequenceNumber);
-    ++num_bwe_extensions;
-  }
-
-  // All parts of the RTP header are 32bit aligned.
-  RTC_CHECK_EQ(header.headerLength % 4, 0);
-
-  // Original packet could have more extensions, there could be csrcs that are
-  // not propagated by the rtc event log, i.e. logged header size might be
-  // larger that rtp_packet.header_size(). Increase it by setting an extra fake
-  // extension.
-  RTC_CHECK_GE(header.headerLength, rtp_packet.headers_size());
-  size_t bytes_to_add = header.headerLength - rtp_packet.headers_size();
-  if (bytes_to_add > 0) {
-    if (bytes_to_add <= 16) {
-      // one-byte header rtp header extension allows to add up to 16 bytes.
-      rtp_packet.AllocateExtension(FakeExtensionSmall::kId, bytes_to_add - 1);
-    } else {
-      // two-byte header rtp header extension would also add one byte per
-      // already set extension.
-      rtp_packet.AllocateExtension(FakeExtensionLarge::kId,
-                                   bytes_to_add - 2 - num_bwe_extensions);
-    }
-  }
-  RTC_CHECK_EQ(rtp_packet.headers_size(), header.headerLength);
-
-  return rtp_packet;
-}
-
-struct PacketLossSummary {
-  size_t num_packets = 0;
-  size_t num_lost_packets = 0;
-  Timestamp base_time = Timestamp::MinusInfinity();
-};
-
-float GetHighestSeqNumber(const rtcp::ReportBlock& block) {
-  return block.extended_high_seq_num();
-}
-
-float GetFractionLost(const rtcp::ReportBlock& block) {
-  return static_cast<double>(block.fraction_lost()) / 256 * 100;
-}
-
-float GetCumulativeLost(const rtcp::ReportBlock& block) {
-  return block.cumulative_lost();
-}
-
-float DelaySinceLastSr(const rtcp::ReportBlock& block) {
-  return static_cast<double>(block.delay_since_last_sr()) / 65536;
-}
-
-std::map<uint32_t, std::string> BuildCandidateIdLogDescriptionMap(
-    const std::vector<LoggedIceCandidatePairConfig>&
-        ice_candidate_pair_configs) {
-  std::map<uint32_t, std::string> candidate_pair_desc_by_id;
-  for (const auto& config : ice_candidate_pair_configs) {
-    // TODO(qingsi): Add the handling of the "Updated" config event after the
-    // visualization of property change for candidate pairs is introduced.
-    if (candidate_pair_desc_by_id.find(config.candidate_pair_id) ==
-        candidate_pair_desc_by_id.end()) {
-      const std::string candidate_pair_desc =
-          GetCandidatePairLogDescriptionAsString(config);
-      candidate_pair_desc_by_id[config.candidate_pair_id] = candidate_pair_desc;
-    }
-  }
-  return candidate_pair_desc_by_id;
-}
-
-}  // namespace
-
-EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log,
+EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& parsed_log,
                                    bool normalize_time)
-    : env_(CreateEnvironment()), parsed_log_(log) {
+    : parsed_log_(parsed_log),
+      config_(CreateEnvironment(), parsed_log, normalize_time) {
   config_.window_duration_ = TimeDelta::Millis(250);
   config_.step_ = TimeDelta::Millis(10);
-  if (!log.start_log_events().empty()) {
-    config_.rtc_to_utc_offset_ = log.start_log_events()[0].utc_time() -
-                                 log.start_log_events()[0].log_time();
-  }
-  config_.normalize_time_ = normalize_time;
-  config_.begin_time_ = parsed_log_.first_timestamp();
-  config_.end_time_ = parsed_log_.last_timestamp();
   if (config_.end_time_ < config_.begin_time_) {
     RTC_LOG(LS_WARNING) << "No useful events in the log.";
     config_.begin_time_ = config_.end_time_ = Timestamp::Zero();
@@ -492,10 +51,9 @@ EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log,
                    << " seconds long.";
 }
 
-EventLogAnalyzer::EventLogAnalyzer(const Environment& env,
-                                   const ParsedRtcEventLog& log,
+EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& parsed_log,
                                    const AnalyzerConfig& config)
-    : env_(env), parsed_log_(log), config_(config) {
+    : parsed_log_(parsed_log), config_(config) {
   RTC_LOG(LS_INFO) << "Log is "
                    << (parsed_log_.last_timestamp().ms() -
                        parsed_log_.first_timestamp().ms()) /
@@ -503,31 +61,18 @@ EventLogAnalyzer::EventLogAnalyzer(const Environment& env,
                    << " seconds long.";
 }
 
-class BitrateObserver : public RemoteBitrateObserver {
- public:
-  BitrateObserver() : last_bitrate_bps_(0), bitrate_updated_(false) {}
-
-  void Update(NetworkControlUpdate update) {
-    if (update.target_rate) {
-      last_bitrate_bps_ = update.target_rate->target_rate.bps();
-      bitrate_updated_ = true;
+void EventLogAnalyzer::CreateGraphsByName(const std::vector<std::string>& names,
+                                          PlotCollection* collection) const {
+  for (absl::string_view name : names) {
+    auto plot = absl::c_find_if(plots_, [name](const PlotDeclaration& plot) {
+      return plot.label == name;
+    });
+    if (plot != plots_.end()) {
+      Plot* output = collection->AppendNewPlot(plot->label);
+      plot->plot_func(output);
     }
   }
-
-  void OnReceiveBitrateChanged(const std::vector<uint32_t>& ssrcs,
-                               uint32_t bitrate) override {}
-
-  uint32_t last_bitrate_bps() const { return last_bitrate_bps_; }
-  bool GetAndResetBitrateUpdated() {
-    bool bitrate_updated = bitrate_updated_;
-    bitrate_updated_ = false;
-    return bitrate_updated;
-  }
-
- private:
-  uint32_t last_bitrate_bps_;
-  bool bitrate_updated_;
-};
+}
 
 void EventLogAnalyzer::InitializeMapOfNamedGraphs(bool show_detector_state,
                                                   bool show_alr_state,
@@ -737,1664 +282,155 @@ void EventLogAnalyzer::InitializeMapOfNamedGraphs(bool show_detector_state,
     this->CreateDtlsWritableStateGraph(plot);
   });
 }
-
-void EventLogAnalyzer::CreateGraphsByName(const std::vector<std::string>& names,
-                                          PlotCollection* collection) const {
-  for (absl::string_view name : names) {
-    auto plot = absl::c_find_if(plots_, [name](const PlotDeclaration& plot) {
-      return plot.label == name;
-    });
-    if (plot != plots_.end()) {
-      Plot* output = collection->AppendNewPlot(plot->label);
-      plot->plot_func(output);
-    }
-  }
-}
-
-void EventLogAnalyzer::CreatePacketGraph(PacketDirection direction,
-                                         Plot* plot) const {
-  for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
-    // Filter on SSRC.
-    if (!MatchingSsrc(stream.ssrc, desired_ssrc_)) {
-      continue;
-    }
-
-    TimeSeries time_series(GetStreamName(parsed_log_, direction, stream.ssrc),
-                           LineStyle::kBar);
-    auto GetPacketSize = [](const LoggedRtpPacket& packet) {
-      return std::optional<float>(packet.total_length);
-    };
-    auto ToCallTime = [this](const LoggedRtpPacket& packet) {
-      return this->config_.GetCallTimeSec(packet.timestamp);
-    };
-    ProcessPoints<LoggedRtpPacket>(ToCallTime, GetPacketSize,
-                                   stream.packet_view, &time_series);
-    plot->AppendTimeSeries(std::move(time_series));
-  }
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Packet size (bytes)", kBottomMargin,
-                          kTopMargin);
-  plot->SetTitle(GetDirectionAsString(direction) + " RTP packets");
-}
-
-void EventLogAnalyzer::CreateRtcpTypeGraph(PacketDirection direction,
-                                           Plot* plot) const {
-  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(
-      parsed_log_.transport_feedbacks(direction), config_, "TWCC", 1));
-  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(
-      parsed_log_.congestion_feedback(direction), config_, "CCFB", 2));
-  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(
-      parsed_log_.receiver_reports(direction), config_, "RR", 3));
-  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(
-      parsed_log_.sender_reports(direction), config_, "SR", 4));
-  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(
-      parsed_log_.extended_reports(direction), config_, "XR", 5));
-  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(parsed_log_.nacks(direction),
-                                                  config_, "NACK", 6));
-  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(parsed_log_.rembs(direction),
-                                                  config_, "REMB", 7));
-  plot->AppendTimeSeries(
-      CreateRtcpTypeTimeSeries(parsed_log_.firs(direction), config_, "FIR", 8));
-  plot->AppendTimeSeries(
-      CreateRtcpTypeTimeSeries(parsed_log_.plis(direction), config_, "PLI", 9));
-  plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(parsed_log_.byes(direction),
-                                                  config_, "BYE", 10));
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "RTCP type", kBottomMargin, kTopMargin);
-  plot->SetTitle(GetDirectionAsString(direction) + " RTCP packets");
-  plot->SetYAxisTickLabels({{1, "TWCC"},
-                            {2, "CCFB"},
-                            {3, "RR"},
-                            {4, "SR"},
-                            {5, "XR"},
-                            {6, "NACK"},
-                            {7, "REMB"},
-                            {8, "FIR"},
-                            {9, "PLI"},
-                            {10, "BYE"}});
-}
-
-template <typename IterableType>
-void EventLogAnalyzer::CreateAccumulatedPacketsTimeSeries(
-    Plot* plot,
-    const IterableType& packets,
-    const std::string& label) const {
-  TimeSeries time_series(label, LineStyle::kStep);
-  for (size_t i = 0; i < packets.size(); i++) {
-    float x = config_.GetCallTimeSec(packets[i].log_time());
-    time_series.points.emplace_back(x, i + 1);
-  }
-  plot->AppendTimeSeries(std::move(time_series));
-}
-
-void EventLogAnalyzer::CreateAccumulatedPacketsGraph(PacketDirection direction,
-                                                     Plot* plot) const {
-  for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
-    if (!MatchingSsrc(stream.ssrc, desired_ssrc_))
-      continue;
-    std::string label = std::string("RTP ") +
-                        GetStreamName(parsed_log_, direction, stream.ssrc);
-    CreateAccumulatedPacketsTimeSeries(plot, stream.packet_view, label);
-  }
-  std::string label =
-      std::string("RTCP ") + "(" + GetDirectionAsShortString(direction) + ")";
-  if (direction == kIncomingPacket) {
-    CreateAccumulatedPacketsTimeSeries(
-        plot, parsed_log_.incoming_rtcp_packets(), label);
-  } else {
-    CreateAccumulatedPacketsTimeSeries(
-        plot, parsed_log_.outgoing_rtcp_packets(), label);
-  }
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Received Packets", kBottomMargin, kTopMargin);
-  plot->SetTitle(std::string("Accumulated ") + GetDirectionAsString(direction) +
-                 " RTP/RTCP packets");
-}
-
-void EventLogAnalyzer::CreatePacketRateGraph(PacketDirection direction,
-                                             Plot* plot) const {
-  auto CountPackets = [](auto packet) { return 1.0; };
-  for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
-    // Filter on SSRC.
-    if (!MatchingSsrc(stream.ssrc, desired_ssrc_)) {
-      continue;
-    }
-    TimeSeries time_series(
-        std::string("RTP ") +
-            GetStreamName(parsed_log_, direction, stream.ssrc),
-        LineStyle::kLine);
-    MovingAverage<LoggedRtpPacket, double>(CountPackets, stream.packet_view,
-                                           config_, &time_series);
-    plot->AppendTimeSeries(std::move(time_series));
-  }
-  TimeSeries time_series(
-      std::string("RTCP ") + "(" + GetDirectionAsShortString(direction) + ")",
-      LineStyle::kLine);
-  if (direction == kIncomingPacket) {
-    MovingAverage<LoggedRtcpPacketIncoming, double>(
-        CountPackets, parsed_log_.incoming_rtcp_packets(), config_,
-        &time_series);
-  } else {
-    MovingAverage<LoggedRtcpPacketOutgoing, double>(
-        CountPackets, parsed_log_.outgoing_rtcp_packets(), config_,
-        &time_series);
-  }
-  plot->AppendTimeSeries(std::move(time_series));
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Packet Rate (packets/s)", kBottomMargin,
-                          kTopMargin);
-  plot->SetTitle("Rate of " + GetDirectionAsString(direction) +
-                 " RTP/RTCP packets");
-}
-
-void EventLogAnalyzer::CreateTotalPacketRateGraph(PacketDirection direction,
-                                                  Plot* plot) const {
-  // Contains a log timestamp to enable counting logged events of different
-  // types using MovingAverage().
-  class LogTime {
-   public:
-    explicit LogTime(Timestamp log_time) : log_time_(log_time) {}
-    Timestamp log_time() const { return log_time_; }
-
-   private:
-    Timestamp log_time_;
-  };
-  std::vector<LogTime> packet_times;
-  auto handle_rtp = [&packet_times](const LoggedRtpPacket& packet) {
-    packet_times.emplace_back(packet.log_time());
-  };
-  RtcEventProcessor process;
-  for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
-    process.AddEvents(stream.packet_view, handle_rtp, direction);
-  }
-  if (direction == kIncomingPacket) {
-    auto handle_incoming_rtcp =
-        [&packet_times](const LoggedRtcpPacketIncoming& packet) {
-          packet_times.emplace_back(packet.log_time());
-        };
-    process.AddEvents(parsed_log_.incoming_rtcp_packets(),
-                      handle_incoming_rtcp);
-  } else {
-    auto handle_outgoing_rtcp =
-        [&packet_times](const LoggedRtcpPacketOutgoing& packet) {
-          packet_times.emplace_back(packet.log_time());
-        };
-    process.AddEvents(parsed_log_.outgoing_rtcp_packets(),
-                      handle_outgoing_rtcp);
-  }
-  process.ProcessEventsInOrder();
-  TimeSeries time_series(std::string("Total ") + "(" +
-                             GetDirectionAsShortString(direction) + ") packets",
-                         LineStyle::kLine);
-  MovingAverage<LogTime, uint64_t>([](auto packet) { return 1; }, packet_times,
-                                   config_, &time_series);
-  plot->AppendTimeSeries(std::move(time_series));
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Packet Rate (packets/s)", kBottomMargin,
-                          kTopMargin);
-  plot->SetTitle("Rate of all " + GetDirectionAsString(direction) +
-                 " RTP/RTCP packets");
-}
-
-// For each SSRC, plot the time between the consecutive playouts.
 void EventLogAnalyzer::CreatePlayoutGraph(Plot* plot) const {
-  for (const auto& playout_stream : parsed_log_.audio_playout_events()) {
-    uint32_t ssrc = playout_stream.first;
-    if (!MatchingSsrc(ssrc, desired_ssrc_))
-      continue;
-    std::optional<int64_t> last_playout_ms;
-    TimeSeries time_series(SsrcToString(ssrc), LineStyle::kBar);
-    for (const auto& playout_event : playout_stream.second) {
-      float x = config_.GetCallTimeSec(playout_event.log_time());
-      int64_t playout_time_ms = playout_event.log_time_ms();
-      // If there were no previous playouts, place the point on the x-axis.
-      float y = playout_time_ms - last_playout_ms.value_or(playout_time_ms);
-      time_series.points.push_back(TimeSeriesPoint(x, y));
-      last_playout_ms.emplace(playout_time_ms);
-    }
-    plot->AppendTimeSeries(std::move(time_series));
-  }
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Time since last playout (ms)", kBottomMargin,
-                          kTopMargin);
-  plot->SetTitle("Audio playout");
+  webrtc::CreatePlayoutGraph(parsed_log_, config_, plot);
 }
 
 void EventLogAnalyzer::CreateNetEqSetMinimumDelay(Plot* plot) const {
-  for (const auto& playout_stream :
-       parsed_log_.neteq_set_minimum_delay_events()) {
-    uint32_t ssrc = playout_stream.first;
-    if (!MatchingSsrc(ssrc, desired_ssrc_))
-      continue;
-
-    TimeSeries time_series(SsrcToString(ssrc), LineStyle::kStep,
-                           PointStyle::kHighlight);
-    for (const auto& event : playout_stream.second) {
-      float x = config_.GetCallTimeSec(event.log_time());
-      float y = event.minimum_delay_ms;
-      time_series.points.push_back(TimeSeriesPoint(x, y));
-    }
-    plot->AppendTimeSeries(std::move(time_series));
-  }
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1000, "Minimum Delay (ms)", kBottomMargin,
-                          kTopMargin);
-  plot->SetTitle("Set Minimum Delay");
+  webrtc::CreateNetEqSetMinimumDelay(parsed_log_, config_, plot);
 }
 
-// For audio SSRCs, plot the audio level.
 void EventLogAnalyzer::CreateAudioLevelGraph(PacketDirection direction,
                                              Plot* plot) const {
-  for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
-    if (!IsAudioSsrc(parsed_log_, direction, stream.ssrc))
-      continue;
-    TimeSeries time_series(GetStreamName(parsed_log_, direction, stream.ssrc),
-                           LineStyle::kLine);
-    for (auto& packet : stream.packet_view) {
-      if (packet.header.extension.audio_level()) {
-        float x = config_.GetCallTimeSec(packet.log_time());
-        // The audio level is stored in -dBov (so e.g. -10 dBov is stored as 10)
-        // Here we convert it to dBov.
-        float y =
-            static_cast<float>(-packet.header.extension.audio_level()->level());
-        time_series.points.emplace_back(TimeSeriesPoint(x, y));
-      }
-    }
-    plot->AppendTimeSeries(std::move(time_series));
-  }
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetYAxis(-127, 0, "Audio level (dBov)", kBottomMargin, kTopMargin);
-  plot->SetTitle(GetDirectionAsString(direction) + " audio level");
-}
-
-// For each SSRC, plot the sequence number difference between consecutive
-// incoming packets.
-void EventLogAnalyzer::CreateSequenceNumberGraph(Plot* plot) const {
-  for (const auto& stream : parsed_log_.incoming_rtp_packets_by_ssrc()) {
-    // Filter on SSRC.
-    if (!MatchingSsrc(stream.ssrc, desired_ssrc_)) {
-      continue;
-    }
-
-    TimeSeries time_series(
-        GetStreamName(parsed_log_, kIncomingPacket, stream.ssrc),
-        LineStyle::kBar);
-    auto GetSequenceNumberDiff = [](const LoggedRtpPacketIncoming& old_packet,
-                                    const LoggedRtpPacketIncoming& new_packet) {
-      int64_t diff =
-          WrappingDifference(new_packet.rtp.header.sequenceNumber,
-                             old_packet.rtp.header.sequenceNumber, 1ul << 16);
-      return diff;
-    };
-    auto ToCallTime = [this](const LoggedRtpPacketIncoming& packet) {
-      return this->config_.GetCallTimeSec(packet.log_time());
-    };
-    ProcessPairs<LoggedRtpPacketIncoming, float>(
-        ToCallTime, GetSequenceNumberDiff, stream.incoming_packets,
-        &time_series);
-    plot->AppendTimeSeries(std::move(time_series));
-  }
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Difference since last packet", kBottomMargin,
-                          kTopMargin);
-  plot->SetTitle("Incoming sequence number delta");
-}
-
-void EventLogAnalyzer::CreateIncomingPacketLossGraph(Plot* plot) const {
-  for (const auto& stream : parsed_log_.incoming_rtp_packets_by_ssrc()) {
-    const std::vector<LoggedRtpPacketIncoming>& packets =
-        stream.incoming_packets;
-    // Filter on SSRC.
-    if (!MatchingSsrc(stream.ssrc, desired_ssrc_) || packets.empty()) {
-      continue;
-    }
-
-    TimeSeries time_series(
-        GetStreamName(parsed_log_, kIncomingPacket, stream.ssrc),
-        LineStyle::kLine, PointStyle::kHighlight);
-    // TODO(terelius): Should the window and step size be read from the class
-    // instead?
-    const TimeDelta kWindow = TimeDelta::Millis(1000);
-    const TimeDelta kStep = TimeDelta::Millis(1000);
-    SeqNumUnwrapper<uint16_t> unwrapper_;
-    SeqNumUnwrapper<uint16_t> prior_unwrapper_;
-    size_t window_index_begin = 0;
-    size_t window_index_end = 0;
-    uint64_t highest_seq_number =
-        unwrapper_.Unwrap(packets[0].rtp.header.sequenceNumber) - 1;
-    uint64_t highest_prior_seq_number =
-        prior_unwrapper_.Unwrap(packets[0].rtp.header.sequenceNumber) - 1;
-
-    for (Timestamp t = config_.begin_time_; t < config_.end_time_ + kStep;
-         t += kStep) {
-      while (window_index_end < packets.size() &&
-             packets[window_index_end].rtp.log_time() < t) {
-        uint64_t sequence_number = unwrapper_.Unwrap(
-            packets[window_index_end].rtp.header.sequenceNumber);
-        highest_seq_number = std::max(highest_seq_number, sequence_number);
-        ++window_index_end;
-      }
-      while (window_index_begin < packets.size() &&
-             packets[window_index_begin].rtp.log_time() < t - kWindow) {
-        uint64_t sequence_number = prior_unwrapper_.Unwrap(
-            packets[window_index_begin].rtp.header.sequenceNumber);
-        highest_prior_seq_number =
-            std::max(highest_prior_seq_number, sequence_number);
-        ++window_index_begin;
-      }
-      float x = config_.GetCallTimeSec(t);
-      uint64_t expected_packets = highest_seq_number - highest_prior_seq_number;
-      if (expected_packets > 0) {
-        int64_t received_packets = window_index_end - window_index_begin;
-        int64_t lost_packets = expected_packets - received_packets;
-        float y = static_cast<float>(lost_packets) / expected_packets * 100;
-        time_series.points.emplace_back(x, y);
-      }
-    }
-    plot->AppendTimeSeries(std::move(time_series));
-  }
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Loss rate (in %)", kBottomMargin, kTopMargin);
-  plot->SetTitle("Incoming packet loss (derived from incoming packets)");
+  webrtc::CreateAudioLevelGraph(parsed_log_, config_, direction, plot);
 }
 
 void EventLogAnalyzer::CreateIncomingDelayGraph(Plot* plot) const {
-  for (const auto& stream : parsed_log_.incoming_rtp_packets_by_ssrc()) {
-    // Filter on SSRC.
-    if (!MatchingSsrc(stream.ssrc, desired_ssrc_) ||
-        IsRtxSsrc(parsed_log_, kIncomingPacket, stream.ssrc)) {
-      continue;
-    }
-
-    const std::vector<LoggedRtpPacketIncoming>& packets =
-        stream.incoming_packets;
-    if (packets.size() < 100) {
-      RTC_LOG(LS_WARNING) << "Can't estimate the RTP clock frequency with "
-                          << packets.size() << " packets in the stream.";
-      continue;
-    }
-    int64_t segment_end_us = parsed_log_.first_log_segment().stop_time_us();
-    std::optional<uint32_t> estimated_frequency =
-        EstimateRtpClockFrequency(packets, segment_end_us);
-    if (!estimated_frequency)
-      continue;
-    const double frequency_hz = *estimated_frequency;
-    if (IsVideoSsrc(parsed_log_, kIncomingPacket, stream.ssrc) &&
-        frequency_hz != 90000) {
-      RTC_LOG(LS_WARNING)
-          << "Video stream should use a 90 kHz clock but appears to use "
-          << frequency_hz / 1000 << ". Discarding.";
-      continue;
-    }
-
-    auto ToCallTime = [this](const LoggedRtpPacketIncoming& packet) {
-      return this->config_.GetCallTimeSec(packet.log_time());
-    };
-    auto ToNetworkDelay = [frequency_hz](
-                              const LoggedRtpPacketIncoming& old_packet,
-                              const LoggedRtpPacketIncoming& new_packet) {
-      return NetworkDelayDiff_CaptureTime(old_packet, new_packet, frequency_hz);
-    };
-
-    TimeSeries capture_time_data(
-        GetStreamName(parsed_log_, kIncomingPacket, stream.ssrc) +
-            " capture-time",
-        LineStyle::kLine);
-    AccumulatePairs<LoggedRtpPacketIncoming, double>(
-        ToCallTime, ToNetworkDelay, packets, &capture_time_data);
-    plot->AppendTimeSeries(std::move(capture_time_data));
-
-    TimeSeries send_time_data(
-        GetStreamName(parsed_log_, kIncomingPacket, stream.ssrc) +
-            " abs-send-time",
-        LineStyle::kLine);
-    AccumulatePairs<LoggedRtpPacketIncoming, double>(
-        ToCallTime, NetworkDelayDiff_AbsSendTime, packets, &send_time_data);
-    plot->AppendTimeSeriesIfNotEmpty(std::move(send_time_data));
-  }
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Delay (ms)", kBottomMargin, kTopMargin);
-  plot->SetTitle("Incoming network delay (relative to first packet)");
+  webrtc::CreateIncomingDelayGraph(parsed_log_, config_, plot);
 }
 
-// Plot the fraction of packets lost (as perceived by the loss-based BWE).
 void EventLogAnalyzer::CreateFractionLossGraph(Plot* plot) const {
-  TimeSeries time_series("Fraction lost", LineStyle::kLine,
-                         PointStyle::kHighlight);
-  for (auto& bwe_update : parsed_log_.bwe_loss_updates()) {
-    float x = config_.GetCallTimeSec(bwe_update.log_time());
-    float y = static_cast<float>(bwe_update.fraction_lost) / 255 * 100;
-    time_series.points.emplace_back(x, y);
-  }
-
-  plot->AppendTimeSeries(std::move(time_series));
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 10, "Loss rate (in %)", kBottomMargin, kTopMargin);
-  plot->SetTitle("Outgoing packet loss (as reported by BWE)");
+  webrtc::CreateFractionLossGraph(parsed_log_, config_, plot);
 }
 
-// Plot the total bandwidth used by all RTP streams.
 void EventLogAnalyzer::CreateTotalIncomingBitrateGraph(Plot* plot) const {
-  // TODO(terelius): This could be provided by the parser.
-  std::multimap<Timestamp, size_t> packets_in_order;
-  for (const auto& stream : parsed_log_.incoming_rtp_packets_by_ssrc()) {
-    for (const LoggedRtpPacketIncoming& packet : stream.incoming_packets)
-      packets_in_order.insert(
-          std::make_pair(packet.rtp.log_time(), packet.rtp.total_length));
-  }
-
-  auto window_begin = packets_in_order.begin();
-  auto window_end = packets_in_order.begin();
-  size_t bytes_in_window = 0;
-
-  if (!packets_in_order.empty()) {
-    // Calculate a moving average of the bitrate and store in a TimeSeries.
-    TimeSeries bitrate_series("Bitrate", LineStyle::kLine);
-    for (Timestamp time = config_.begin_time_;
-         time < config_.end_time_ + config_.step_; time += config_.step_) {
-      while (window_end != packets_in_order.end() && window_end->first < time) {
-        bytes_in_window += window_end->second;
-        ++window_end;
-      }
-      while (window_begin != packets_in_order.end() &&
-             window_begin->first < time - config_.window_duration_) {
-        RTC_DCHECK_LE(window_begin->second, bytes_in_window);
-        bytes_in_window -= window_begin->second;
-        ++window_begin;
-      }
-      float window_duration_in_seconds =
-          static_cast<float>(config_.window_duration_.us()) /
-          kNumMicrosecsPerSec;
-      float x = config_.GetCallTimeSec(time);
-      float y = bytes_in_window * 8 / window_duration_in_seconds / 1000;
-      bitrate_series.points.emplace_back(x, y);
-    }
-    plot->AppendTimeSeries(std::move(bitrate_series));
-  }
-
-  // Overlay the outgoing REMB over incoming bitrate.
-  TimeSeries remb_series("Remb", LineStyle::kStep);
-  for (const auto& rtcp : parsed_log_.rembs(kOutgoingPacket)) {
-    float x = config_.GetCallTimeSec(rtcp.log_time());
-    float y = static_cast<float>(rtcp.remb.bitrate_bps()) / 1000;
-    remb_series.points.emplace_back(x, y);
-  }
-  plot->AppendTimeSeriesIfNotEmpty(std::move(remb_series));
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin);
-  plot->SetTitle("Incoming RTP bitrate");
+  webrtc::CreateTotalIncomingBitrateGraph(parsed_log_, config_, plot);
 }
 
-// Plot the total bandwidth used by all RTP streams.
 void EventLogAnalyzer::CreateTotalOutgoingBitrateGraph(
     Plot* plot,
     bool show_detector_state,
     bool show_alr_state,
     bool show_link_capacity) const {
-  // TODO(terelius): This could be provided by the parser.
-  std::multimap<Timestamp, size_t> packets_in_order;
-  for (const auto& stream : parsed_log_.outgoing_rtp_packets_by_ssrc()) {
-    for (const LoggedRtpPacketOutgoing& packet : stream.outgoing_packets)
-      packets_in_order.insert(
-          std::make_pair(packet.rtp.log_time(), packet.rtp.total_length));
-  }
-
-  auto window_begin = packets_in_order.begin();
-  auto window_end = packets_in_order.begin();
-  size_t bytes_in_window = 0;
-
-  if (!packets_in_order.empty()) {
-    // Calculate a moving average of the bitrate and store in a TimeSeries.
-    TimeSeries bitrate_series("Bitrate", LineStyle::kLine);
-    for (Timestamp time = config_.begin_time_;
-         time < config_.end_time_ + config_.step_; time += config_.step_) {
-      while (window_end != packets_in_order.end() && window_end->first < time) {
-        bytes_in_window += window_end->second;
-        ++window_end;
-      }
-      while (window_begin != packets_in_order.end() &&
-             window_begin->first < time - config_.window_duration_) {
-        RTC_DCHECK_LE(window_begin->second, bytes_in_window);
-        bytes_in_window -= window_begin->second;
-        ++window_begin;
-      }
-      float window_duration_in_seconds =
-          static_cast<float>(config_.window_duration_.us()) /
-          kNumMicrosecsPerSec;
-      float x = config_.GetCallTimeSec(time);
-      float y = bytes_in_window * 8 / window_duration_in_seconds / 1000;
-      bitrate_series.points.emplace_back(x, y);
-    }
-    plot->AppendTimeSeries(std::move(bitrate_series));
-  }
-
-  // Overlay the send-side bandwidth estimate over the outgoing bitrate.
-  TimeSeries loss_series("Loss-based estimate", LineStyle::kStep);
-  for (auto& loss_update : parsed_log_.bwe_loss_updates()) {
-    float x = config_.GetCallTimeSec(loss_update.log_time());
-    float y = static_cast<float>(loss_update.bitrate_bps) / 1000;
-    loss_series.points.emplace_back(x, y);
-  }
-
-  TimeSeries link_capacity_lower_series("Link-capacity-lower",
-                                        LineStyle::kStep);
-  TimeSeries link_capacity_upper_series("Link-capacity-upper",
-                                        LineStyle::kStep);
-  for (auto& remote_estimate_event : parsed_log_.remote_estimate_events()) {
-    float x = config_.GetCallTimeSec(remote_estimate_event.log_time());
-    if (remote_estimate_event.link_capacity_lower.has_value()) {
-      float link_capacity_lower = static_cast<float>(
-          remote_estimate_event.link_capacity_lower.value().kbps());
-      link_capacity_lower_series.points.emplace_back(x, link_capacity_lower);
-    }
-    if (remote_estimate_event.link_capacity_upper.has_value()) {
-      float link_capacity_upper = static_cast<float>(
-          remote_estimate_event.link_capacity_upper.value().kbps());
-      link_capacity_upper_series.points.emplace_back(x, link_capacity_upper);
-    }
-  }
-
-  TimeSeries delay_series("Delay-based estimate", LineStyle::kStep);
-  IntervalSeries overusing_series("Overusing", "#ff8e82",
-                                  IntervalSeries::kHorizontal);
-  IntervalSeries underusing_series("Underusing", "#5092fc",
-                                   IntervalSeries::kHorizontal);
-  IntervalSeries normal_series("Normal", "#c4ffc4",
-                               IntervalSeries::kHorizontal);
-  IntervalSeries* last_series = &normal_series;
-  float last_detector_switch = 0.0;
-
-  BandwidthUsage last_detector_state = BandwidthUsage::kBwNormal;
-
-  for (auto& delay_update : parsed_log_.bwe_delay_updates()) {
-    float x = config_.GetCallTimeSec(delay_update.log_time());
-    float y = static_cast<float>(delay_update.bitrate_bps) / 1000;
-
-    if (last_detector_state != delay_update.detector_state) {
-      last_series->intervals.emplace_back(last_detector_switch, x);
-      last_detector_state = delay_update.detector_state;
-      last_detector_switch = x;
-
-      switch (delay_update.detector_state) {
-        case BandwidthUsage::kBwNormal:
-          last_series = &normal_series;
-          break;
-        case BandwidthUsage::kBwUnderusing:
-          last_series = &underusing_series;
-          break;
-        case BandwidthUsage::kBwOverusing:
-          last_series = &overusing_series;
-          break;
-        case BandwidthUsage::kLast:
-          RTC_DCHECK_NOTREACHED();
-      }
-    }
-
-    delay_series.points.emplace_back(x, y);
-  }
-
-  RTC_CHECK(last_series);
-  last_series->intervals.emplace_back(last_detector_switch,
-                                      config_.CallEndTimeSec());
-
-  TimeSeries scream_series("Scream target rate", LineStyle::kStep);
-  for (auto& scream_update : parsed_log_.bwe_scream_updates()) {
-    float x = config_.GetCallTimeSec(scream_update.log_time());
-    float y = static_cast<float>(scream_update.target_rate.kbps());
-    scream_series.points.emplace_back(x, y);
-  }
-
-  TimeSeries created_series("Probe cluster created.", LineStyle::kNone,
-                            PointStyle::kHighlight);
-  for (auto& cluster : parsed_log_.bwe_probe_cluster_created_events()) {
-    float x = config_.GetCallTimeSec(cluster.log_time());
-    float y = static_cast<float>(cluster.bitrate_bps) / 1000;
-    created_series.points.emplace_back(x, y);
-  }
-
-  TimeSeries result_series("Probing results.", LineStyle::kNone,
-                           PointStyle::kHighlight);
-  for (auto& result : parsed_log_.bwe_probe_success_events()) {
-    float x = config_.GetCallTimeSec(result.log_time());
-    float y = static_cast<float>(result.bitrate_bps) / 1000;
-    result_series.points.emplace_back(x, y);
-  }
-
-  TimeSeries probe_failures_series("Probe failed", LineStyle::kNone,
-                                   PointStyle::kHighlight);
-  for (auto& failure : parsed_log_.bwe_probe_failure_events()) {
-    float x = config_.GetCallTimeSec(failure.log_time());
-    probe_failures_series.points.emplace_back(x, 0);
-  }
-
-  IntervalSeries alr_state("ALR", "#555555", IntervalSeries::kHorizontal);
-  bool previously_in_alr = false;
-  Timestamp alr_start = Timestamp::Zero();
-  for (auto& alr : parsed_log_.alr_state_events()) {
-    float y = config_.GetCallTimeSec(alr.log_time());
-    if (!previously_in_alr && alr.in_alr) {
-      alr_start = alr.log_time();
-      previously_in_alr = true;
-    } else if (previously_in_alr && !alr.in_alr) {
-      float x = config_.GetCallTimeSec(alr_start);
-      alr_state.intervals.emplace_back(x, y);
-      previously_in_alr = false;
-    }
-  }
-
-  if (previously_in_alr) {
-    float x = config_.GetCallTimeSec(alr_start);
-    float y = config_.GetCallTimeSec(config_.end_time_);
-    alr_state.intervals.emplace_back(x, y);
-  }
-
-  if (show_detector_state) {
-    plot->AppendIntervalSeries(std::move(overusing_series));
-    plot->AppendIntervalSeries(std::move(underusing_series));
-    plot->AppendIntervalSeries(std::move(normal_series));
-  }
-
-  if (show_alr_state) {
-    plot->AppendIntervalSeries(std::move(alr_state));
-  }
-
-  if (show_link_capacity) {
-    plot->AppendTimeSeriesIfNotEmpty(std::move(link_capacity_lower_series));
-    plot->AppendTimeSeriesIfNotEmpty(std::move(link_capacity_upper_series));
-  }
-
-  plot->AppendTimeSeries(std::move(loss_series));
-  plot->AppendTimeSeriesIfNotEmpty(std::move(probe_failures_series));
-  plot->AppendTimeSeries(std::move(delay_series));
-  plot->AppendTimeSeriesIfNotEmpty(std::move(scream_series));
-  plot->AppendTimeSeries(std::move(created_series));
-  plot->AppendTimeSeries(std::move(result_series));
-
-  // Overlay the incoming REMB over the outgoing bitrate.
-  TimeSeries remb_series("Remb", LineStyle::kStep);
-  for (const auto& rtcp : parsed_log_.rembs(kIncomingPacket)) {
-    float x = config_.GetCallTimeSec(rtcp.log_time());
-    float y = static_cast<float>(rtcp.remb.bitrate_bps()) / 1000;
-    remb_series.points.emplace_back(x, y);
-  }
-  plot->AppendTimeSeriesIfNotEmpty(std::move(remb_series));
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin);
-  plot->SetTitle("Outgoing RTP bitrate");
-}
-
-// For each SSRC, plot the bandwidth used by that stream.
-void EventLogAnalyzer::CreateStreamBitrateGraph(PacketDirection direction,
-                                                Plot* plot) const {
-  for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
-    // Filter on SSRC.
-    if (!MatchingSsrc(stream.ssrc, desired_ssrc_)) {
-      continue;
-    }
-
-    TimeSeries time_series(GetStreamName(parsed_log_, direction, stream.ssrc),
-                           LineStyle::kLine);
-    auto GetPacketSizeKilobits = [](const LoggedRtpPacket& packet) {
-      return packet.total_length * 8.0 / 1000.0;
-    };
-    MovingAverage<LoggedRtpPacket, double>(
-        GetPacketSizeKilobits, stream.packet_view, config_, &time_series);
-    plot->AppendTimeSeries(std::move(time_series));
-  }
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin);
-  plot->SetTitle(GetDirectionAsString(direction) + " bitrate per stream");
-}
-
-// Plot the bitrate allocation for each temporal and spatial layer.
-// Computed from RTCP XR target bitrate block, so the graph is only populated if
-// those are sent.
-void EventLogAnalyzer::CreateBitrateAllocationGraph(PacketDirection direction,
-                                                    Plot* plot) const {
-  std::map<LayerDescription, TimeSeries> time_series;
-  const auto& xr_list = parsed_log_.extended_reports(direction);
-  for (const auto& rtcp : xr_list) {
-    const std::optional<rtcp::TargetBitrate>& target_bitrate =
-        rtcp.xr.target_bitrate();
-    if (!target_bitrate.has_value())
-      continue;
-    for (const auto& bitrate_item : target_bitrate->GetTargetBitrates()) {
-      LayerDescription layer(rtcp.xr.sender_ssrc(), bitrate_item.spatial_layer,
-                             bitrate_item.temporal_layer);
-      auto time_series_it = time_series.find(layer);
-      if (time_series_it == time_series.end()) {
-        std::string layer_name = GetLayerName(layer);
-        bool inserted;
-        std::tie(time_series_it, inserted) = time_series.insert(
-            std::make_pair(layer, TimeSeries(layer_name, LineStyle::kStep)));
-        RTC_DCHECK(inserted);
-      }
-      float x = config_.GetCallTimeSec(rtcp.log_time());
-      float y = bitrate_item.target_bitrate_kbps;
-      time_series_it->second.points.emplace_back(x, y);
-    }
-  }
-  for (auto& layer : time_series) {
-    plot->AppendTimeSeries(std::move(layer.second));
-  }
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin);
-  if (direction == kIncomingPacket)
-    plot->SetTitle("Target bitrate per incoming layer");
-  else
-    plot->SetTitle("Target bitrate per outgoing layer");
+  webrtc::CreateTotalOutgoingBitrateGraph(parsed_log_, config_, plot,
+                                          show_detector_state, show_alr_state,
+                                          show_link_capacity);
 }
 
 void EventLogAnalyzer::CreateGoogCcSimulationGraph(Plot* plot) const {
-  TimeSeries target_rates("Simulated target rate", LineStyle::kStep,
-                          PointStyle::kHighlight);
-  TimeSeries delay_based("Logged delay-based estimate", LineStyle::kStep,
-                         PointStyle::kHighlight);
-  TimeSeries loss_based("Logged loss-based estimate", LineStyle::kStep,
-                        PointStyle::kHighlight);
-  TimeSeries probe_results("Logged probe success", LineStyle::kNone,
-                           PointStyle::kHighlight);
-
-  LogBasedNetworkControllerSimulation simulation(
-      env_, std::make_unique<GoogCcNetworkControllerFactory>(),
-      [&](const NetworkControlUpdate& update, Timestamp at_time) {
-        if (update.target_rate) {
-          target_rates.points.emplace_back(
-              config_.GetCallTimeSec(at_time),
-              update.target_rate->target_rate.kbps<float>());
-        }
-      });
-
-  simulation.ProcessEventsInLog(parsed_log_);
-  for (const auto& logged : parsed_log_.bwe_delay_updates())
-    delay_based.points.emplace_back(config_.GetCallTimeSec(logged.log_time()),
-                                    logged.bitrate_bps / 1000);
-  for (const auto& logged : parsed_log_.bwe_probe_success_events())
-    probe_results.points.emplace_back(config_.GetCallTimeSec(logged.log_time()),
-                                      logged.bitrate_bps / 1000);
-  for (const auto& logged : parsed_log_.bwe_loss_updates())
-    loss_based.points.emplace_back(config_.GetCallTimeSec(logged.log_time()),
-                                   logged.bitrate_bps / 1000);
-
-  plot->AppendTimeSeries(std::move(delay_based));
-  plot->AppendTimeSeries(std::move(loss_based));
-  plot->AppendTimeSeries(std::move(probe_results));
-  plot->AppendTimeSeries(std::move(target_rates));
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 10, "Bitrate (kbps)", kBottomMargin, kTopMargin);
-  plot->SetTitle("Simulated BWE behavior");
+  webrtc::CreateGoogCcSimulationGraph(parsed_log_, config_, plot);
 }
 
 void EventLogAnalyzer::CreateScreamSimulationBitrateGraph(Plot* plot) const {
-  TimeSeries target_rate_series("Target rate", LineStyle::kStep);
-  TimeSeries pacing_rate_series("Pacing rate", LineStyle::kStep);
-  TimeSeries send_rate_series("Send rate", LineStyle::kStep);
-
-  LogScreamSimulation simulation({.rate_window = config_.window_duration_},
-                                 env_);
-  simulation.ProcessEventsInLog(parsed_log_);
-
-  for (const LogScreamSimulation::State& state : simulation.updates()) {
-    target_rate_series.points.emplace_back(config_.GetCallTimeSec(state.time),
-                                           state.target_rate.bps() / 1000);
-    pacing_rate_series.points.emplace_back(config_.GetCallTimeSec(state.time),
-                                           state.pacing_rate.bps() / 1000);
-    send_rate_series.points.emplace_back(config_.GetCallTimeSec(state.time),
-                                         state.send_rate.bps() / 1000);
-  }
-  plot->AppendTimeSeries(std::move(target_rate_series));
-  plot->AppendTimeSeries(std::move(pacing_rate_series));
-  plot->AppendTimeSeries(std::move(send_rate_series));
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 100, "Kbps", kBottomMargin, kTopMargin);
-  plot->SetTitle("Simulated Scream rates");
+  webrtc::CreateScreamSimulationBitrateGraph(parsed_log_, config_, plot);
 }
 
 void EventLogAnalyzer::CreateScreamSimulationRefWindowGraph(Plot* plot) const {
-  using SendWindowUsage = LogScreamSimulation::State::SendWindowUsage;
-  TimeSeries ref_window_series("RefWindow", LineStyle::kStep);
-  TimeSeries ref_window_i_series("RefWindowI", LineStyle::kStep);
-  TimeSeries max_data_in_flight("Max allowed data in flight", LineStyle::kStep);
-  TimeSeries data_in_flight("Data in flight", LineStyle::kStep);
-  IntervalSeries send_window_above_max_series(
-      "Data in flight > Max allowed", "#ff8e82", IntervalSeries::kHorizontal);
-  IntervalSeries send_window_below_ref_window_series(
-      "Data in flight < RefWindow", "#c5dff2", IntervalSeries::kHorizontal);
-  IntervalSeries send_window_above_ref_window_series(
-      "Data in flight >= RefWindow", "#b9fad8", IntervalSeries::kHorizontal);
-  IntervalSeries* last_series = &send_window_below_ref_window_series;
-
-  LogScreamSimulation simulation({.rate_window = config_.window_duration_},
-                                 env_);
-  simulation.ProcessEventsInLog(parsed_log_);
-  if (simulation.updates().empty()) {
-    RTC_LOG(LS_ERROR) << "Empty simulation.";
-    return;
-  }
-
-  float send_window_state_switch =
-      config_.GetCallTimeSec(simulation.updates().front().time);
-  SendWindowUsage send_window_usage = SendWindowUsage::kBelowRefWindow;
-
-  for (const LogScreamSimulation::State& state : simulation.updates()) {
-    ref_window_series.points.emplace_back(config_.GetCallTimeSec(state.time),
-                                          state.ref_window.bytes());
-    ref_window_i_series.points.emplace_back(config_.GetCallTimeSec(state.time),
-                                            state.ref_window_i.bytes());
-    max_data_in_flight.points.emplace_back(config_.GetCallTimeSec(state.time),
-                                           state.max_data_in_flight.bytes());
-    data_in_flight.points.emplace_back(config_.GetCallTimeSec(state.time),
-                                       state.data_in_flight.bytes());
-    if (state.send_window_usage != send_window_usage) {
-      last_series->intervals.emplace_back(send_window_state_switch,
-                                          config_.GetCallTimeSec(state.time));
-      send_window_usage = state.send_window_usage;
-      send_window_state_switch = config_.GetCallTimeSec(state.time);
-      switch (send_window_usage) {
-        case SendWindowUsage::kAboveRefWindow:
-          last_series = &send_window_above_ref_window_series;
-          break;
-        case SendWindowUsage::kBelowRefWindow:
-          last_series = &send_window_below_ref_window_series;
-          break;
-        case SendWindowUsage::kAboveScreamMax:
-          last_series = &send_window_above_max_series;
-          break;
-      }
-    }
-  }
-  last_series->intervals.emplace_back(send_window_state_switch,
-                                      config_.CallEndTimeSec());
-  plot->AppendTimeSeries(std::move(ref_window_series));
-  plot->AppendTimeSeries(std::move(ref_window_i_series));
-  plot->AppendTimeSeries(std::move(max_data_in_flight));
-  plot->AppendTimeSeries(std::move(data_in_flight));
-  plot->AppendIntervalSeries(std::move(send_window_above_max_series));
-  plot->AppendIntervalSeries(std::move(send_window_below_ref_window_series));
-  plot->AppendIntervalSeries(std::move(send_window_above_ref_window_series));
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 10, "Bytes", kBottomMargin, kTopMargin);
-  plot->SetTitle("Simulated Scream RefWindow");
+  webrtc::CreateScreamSimulationRefWindowGraph(parsed_log_, config_, plot);
 }
 
 void EventLogAnalyzer::CreateScreamSimulationRatiosGraph(Plot* plot) const {
-  TimeSeries queue_delay_dev_norm_series("QueueDelayDevNorm", LineStyle::kStep);
-  TimeSeries l4s_alpha_series("L4sAlpha", LineStyle::kStep);
-  TimeSeries l4s_alpha_v_series("L4sAlphaV", LineStyle::kStep);
-  TimeSeries ref_window_scale_factor_due_to_increased_delay(
-      "RefWindowScaleFactorDueToIncreasedDelay", LineStyle::kStep);
-  TimeSeries ref_window_scale_factor_due_to_delay_variation(
-      "RefWindowScaleFactorDueToDelayVariation", LineStyle::kStep);
-
-  LogScreamSimulation simulation({.rate_window = config_.window_duration_},
-                                 env_);
-  simulation.ProcessEventsInLog(parsed_log_);
-
-  for (const LogScreamSimulation::State& state : simulation.updates()) {
-    queue_delay_dev_norm_series.points.emplace_back(
-        config_.GetCallTimeSec(state.time), state.queue_delay_dev_norm);
-    l4s_alpha_series.points.emplace_back(config_.GetCallTimeSec(state.time),
-                                         state.l4s_alpha);
-    l4s_alpha_v_series.points.emplace_back(config_.GetCallTimeSec(state.time),
-                                           state.l4s_alpha_v);
-    ref_window_scale_factor_due_to_increased_delay.points.emplace_back(
-        config_.GetCallTimeSec(state.time),
-        state.ref_window_scale_factor_due_to_increased_delay);
-    ref_window_scale_factor_due_to_delay_variation.points.emplace_back(
-        config_.GetCallTimeSec(state.time),
-        state.ref_window_scale_factor_due_to_delay_variation);
-  }
-  plot->AppendTimeSeries(std::move(queue_delay_dev_norm_series));
-  plot->AppendTimeSeries(std::move(l4s_alpha_series));
-  plot->AppendTimeSeries(std::move(l4s_alpha_v_series));
-  plot->AppendTimeSeries(
-      std::move(ref_window_scale_factor_due_to_increased_delay));
-  plot->AppendTimeSeries(
-      std::move(ref_window_scale_factor_due_to_delay_variation));
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Ratios", kBottomMargin, kTopMargin);
-  plot->SetTitle("Simulated Scream Ratios");
-}
-
-void EventLogAnalyzer::CreateOutgoingEcnFeedbackGraph(Plot* plot) const {
-  CreateEcnFeedbackGraph(plot, kOutgoingPacket);
-  plot->SetTitle("Outgoing ECN count per feedback");
-}
-
-void EventLogAnalyzer::CreateIncomingEcnFeedbackGraph(Plot* plot) const {
-  CreateEcnFeedbackGraph(plot, kIncomingPacket);
-  plot->SetTitle("Incoming ECN count per feedback");
+  webrtc::CreateScreamSimulationRatiosGraph(parsed_log_, config_, plot);
 }
 
 void EventLogAnalyzer::CreateScreamRefWindowGraph(Plot* plot) const {
-  TimeSeries ref_window_series("RefWindow", LineStyle::kStep);
-  for (auto& scream_update : parsed_log_.bwe_scream_updates()) {
-    float x = config_.GetCallTimeSec(scream_update.log_time());
-    float y = static_cast<float>(scream_update.ref_window.bytes());
-    ref_window_series.points.emplace_back(x, y);
-  }
-  plot->AppendTimeSeries(std::move(ref_window_series));
-
-  TimeSeries data_in_flight_series("Data in flight", LineStyle::kLine);
-  for (auto& scream_update : parsed_log_.bwe_scream_updates()) {
-    float x = config_.GetCallTimeSec(scream_update.log_time());
-    float y = static_cast<float>(scream_update.data_in_flight.bytes());
-    data_in_flight_series.points.emplace_back(x, y);
-  }
-  plot->AppendTimeSeries(std::move(data_in_flight_series));
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 3000, "Bytes", kBottomMargin, kTopMargin);
-  plot->SetTitle("Scream Ref Window");
+  webrtc::CreateScreamRefWindowGraph(parsed_log_, config_, plot);
 }
 
 void EventLogAnalyzer::CreateScreamDelayEstimateGraph(Plot* plot) const {
-  TimeSeries smoothed_rtt_series("Smoothed RTT", LineStyle::kStep);
-  TimeSeries avg_queue_delay_series("Avg queue delay", LineStyle::kStep);
-
-  for (auto& scream_update : parsed_log_.bwe_scream_updates()) {
-    float x = config_.GetCallTimeSec(scream_update.log_time());
-    float smoothed_rtt_ms = static_cast<float>(scream_update.smoothed_rtt.ms());
-    smoothed_rtt_series.points.emplace_back(x, smoothed_rtt_ms);
-    float avg_queue_delay_ms =
-        static_cast<float>(scream_update.avg_queue_delay.ms());
-    avg_queue_delay_series.points.emplace_back(x, avg_queue_delay_ms);
-  }
-
-  plot->AppendTimeSeries(std::move(smoothed_rtt_series));
-  plot->AppendTimeSeries(std::move(avg_queue_delay_series));
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 50, "Delay (ms)", kBottomMargin, kTopMargin);
-  plot->SetTitle("Scream delay estimates");
-}
-
-void EventLogAnalyzer::CreateEcnFeedbackGraph(Plot* plot,
-                                              PacketDirection direction) const {
-  TimeSeries not_ect("Not ECN capable", LineStyle::kBar,
-                     PointStyle::kHighlight);
-  TimeSeries ect_1("ECN capable", LineStyle::kBar, PointStyle::kHighlight);
-  TimeSeries ce("Congestion experienced", LineStyle::kBar,
-                PointStyle::kHighlight);
-
-  for (const LoggedRtcpCongestionControlFeedback& feedback :
-       parsed_log_.congestion_feedback(direction)) {
-    int ect_1_count = 0;
-    int not_ect_count = 0;
-    int ce_count = 0;
-
-    for (const rtcp::CongestionControlFeedback::PacketInfo& info :
-         feedback.congestion_feedback.packets()) {
-      switch (info.ecn) {
-        case EcnMarking::kNotEct:
-          ++not_ect_count;
-          break;
-        case EcnMarking::kEct1:
-          ++ect_1_count;
-          break;
-        case EcnMarking::kEct0:
-          RTC_LOG(LS_ERROR) << "unexpected ect(0)";
-          break;
-        case EcnMarking::kCe:
-          ++ce_count;
-          break;
-      }
-    }
-    ect_1.points.emplace_back(config_.GetCallTimeSec(feedback.timestamp),
-                              ect_1_count);
-    not_ect.points.emplace_back(config_.GetCallTimeSec(feedback.timestamp),
-                                not_ect_count);
-    ce.points.emplace_back(config_.GetCallTimeSec(feedback.timestamp),
-                           ce_count);
-  }
-
-  plot->AppendTimeSeriesIfNotEmpty(std::move(ect_1));
-  plot->AppendTimeSeriesIfNotEmpty(std::move(not_ect));
-  plot->AppendTimeSeriesIfNotEmpty(std::move(ce));
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 10, "Count per feedback", kBottomMargin,
-                          kTopMargin);
-}
-
-void EventLogAnalyzer::CreateOutgoingLossRateGraph(Plot* plot) const {
-  struct PacketLossPerFeedback {
-    Timestamp timestamp;              // Time when this feedback was received.
-    int num_packets_in_feedback = 0;  // Includes lost packets.
-    int num_lost_packets = 0;         // In this specific feedback.
-    int num_reordered_packets = 0;    // Packets received in this feedback, but
-                                      // was previously reported as lost.
-    int num_missing_feedback =
-        0;  // Packets missing feedback between this report and the previous.
-  };
-
-  class LossFeedbackBuilder {
-   public:
-    void AddPacket(uint16_t sequence_number, TimeDelta arrival_time_delta) {
-      last_unwrapped_sequence_number_ =
-          sequence_number_unwrapper_.Unwrap(sequence_number);
-      if (!first_sequence_number_.has_value()) {
-        first_sequence_number_ = last_unwrapped_sequence_number_;
-      }
-      ++num_packets_;
-      if (arrival_time_delta.IsInfinite()) {
-        lost_sequence_numbers_.insert(last_unwrapped_sequence_number_);
-      } else {
-        num_reordered_packets_ += previous_lost_sequence_numbers_.count(
-            last_unwrapped_sequence_number_);
-      }
-    }
-
-    void Update(PacketLossPerFeedback& feedback) {
-      feedback.num_packets_in_feedback += num_packets_;
-      feedback.num_lost_packets += lost_sequence_numbers_.size();
-      feedback.num_reordered_packets += num_reordered_packets_;
-      if (first_sequence_number_.has_value() &&
-          previous_feedback_highest_seq_number_.has_value()) {
-        feedback.num_missing_feedback +=
-            *first_sequence_number_ - *previous_feedback_highest_seq_number_ -
-            1;
-      }
-
-      // Prepare for next feedback.
-      first_sequence_number_ = std::nullopt;
-      previous_lost_sequence_numbers_.insert(lost_sequence_numbers_.begin(),
-                                             lost_sequence_numbers_.end());
-      previous_feedback_highest_seq_number_ = last_unwrapped_sequence_number_;
-      lost_sequence_numbers_.clear();
-      num_reordered_packets_ = 0;
-      num_packets_ = 0;
-    }
-
-   private:
-    int64_t last_unwrapped_sequence_number_ = 0;
-    int num_reordered_packets_ = 0;
-    int num_packets_ = 0;
-    std::optional<int64_t> first_sequence_number_;
-
-    std::unordered_set<int64_t> lost_sequence_numbers_;
-    std::unordered_set<int64_t> previous_lost_sequence_numbers_;
-    std::optional<int64_t> previous_feedback_highest_seq_number_;
-
-    RtpSequenceNumberUnwrapper sequence_number_unwrapper_;
-  };
-
-  TimeSeries loss_rate_series("Loss rate (from packet feedback)",
-                              LineStyle::kLine, PointStyle::kHighlight);
-  TimeSeries reordered_packets_between_feedback(
-      "Ratio of reordered packets from last feedback", LineStyle::kLine,
-      PointStyle::kHighlight);
-  TimeSeries average_loss_rate_series("Average loss rate last 5s",
-                                      LineStyle::kLine, PointStyle::kHighlight);
-  TimeSeries missing_feedback_series("Missing feedback", LineStyle::kNone,
-                                     PointStyle::kHighlight);
-
-  std::vector<PacketLossPerFeedback> loss_per_feedback;
-
-  if (!parsed_log_.congestion_feedback(kIncomingPacket).empty()) {
-    plot->SetTitle("Outgoing loss rate (from CCFB)");
-
-    std::map</*ssrc*/ uint32_t, LossFeedbackBuilder> per_ssrc_builder;
-    for (const LoggedRtcpCongestionControlFeedback& feedback :
-         parsed_log_.congestion_feedback(kIncomingPacket)) {
-      const rtcp::CongestionControlFeedback& transport_feedback =
-          feedback.congestion_feedback;
-
-      PacketLossPerFeedback packet_loss_per_feedback = {
-          .timestamp = feedback.log_time()};
-      for (const rtcp::CongestionControlFeedback::PacketInfo& packet :
-           transport_feedback.packets()) {
-        per_ssrc_builder[packet.ssrc].AddPacket(packet.sequence_number,
-                                                packet.arrival_time_offset);
-      }
-      for (auto& [ssrc, builder] : per_ssrc_builder) {
-        builder.Update(packet_loss_per_feedback);
-      }
-      loss_per_feedback.push_back(packet_loss_per_feedback);
-    }
-  } else if (!parsed_log_.transport_feedbacks(kIncomingPacket).empty()) {
-    plot->SetTitle("Outgoing loss rate (from TWCC)");
-
-    LossFeedbackBuilder builder;
-    for (const LoggedRtcpPacketTransportFeedback& feedback :
-         parsed_log_.transport_feedbacks(kIncomingPacket)) {
-      feedback.transport_feedback.ForAllPackets(
-          [&](uint16_t sequence_number, TimeDelta receive_time_delta) {
-            builder.AddPacket(sequence_number, receive_time_delta);
-          });
-      PacketLossPerFeedback packet_loss_per_feedback = {
-          .timestamp = feedback.log_time()};
-      builder.Update(packet_loss_per_feedback);
-      loss_per_feedback.push_back(packet_loss_per_feedback);
-    }
-  }
-
-  PacketLossSummary window_summary;
-  Timestamp last_observation_receive_time = Timestamp::Zero();
-
-  // Use loss based bwe 2 observation duration and observation window size.
-  constexpr TimeDelta kObservationDuration = TimeDelta::Millis(250);
-  constexpr uint32_t kObservationWindowSize = 20;
-  std::deque<PacketLossSummary> observations;
-  int previous_feedback_size = 0;
-  for (const PacketLossPerFeedback& feedback : loss_per_feedback) {
-    for (int64_t num = 0; num < feedback.num_missing_feedback; ++num) {
-      missing_feedback_series.points.emplace_back(
-          config_.GetCallTimeSec(feedback.timestamp), 100 + num);
-    }
-
-    // Compute loss rate from the transport feedback.
-    float loss_rate = static_cast<float>(feedback.num_lost_packets * 100.0 /
-                                         feedback.num_packets_in_feedback);
-
-    loss_rate_series.points.emplace_back(
-        config_.GetCallTimeSec(feedback.timestamp), loss_rate);
-    float reordered_rate =
-        previous_feedback_size == 0
-            ? 0
-            : static_cast<float>(feedback.num_reordered_packets * 100.0 /
-                                 previous_feedback_size);
-    previous_feedback_size = feedback.num_packets_in_feedback;
-    reordered_packets_between_feedback.points.emplace_back(
-        config_.GetCallTimeSec(feedback.timestamp), reordered_rate);
-
-    // Compute loss rate in a window of kObservationWindowSize.
-    if (window_summary.num_packets == 0) {
-      window_summary.base_time = feedback.timestamp;
-    }
-    window_summary.num_packets += feedback.num_packets_in_feedback;
-    window_summary.num_lost_packets +=
-        feedback.num_lost_packets - feedback.num_reordered_packets;
-
-    const Timestamp last_received_time = feedback.timestamp;
-    const TimeDelta observation_duration =
-        window_summary.base_time == Timestamp::Zero()
-            ? TimeDelta::Zero()
-            : last_received_time - window_summary.base_time;
-    if (observation_duration > kObservationDuration) {
-      last_observation_receive_time = last_received_time;
-      observations.push_back(window_summary);
-      if (observations.size() > kObservationWindowSize) {
-        observations.pop_front();
-      }
-
-      // Compute average loss rate in a number of windows.
-      int total_packets = 0;
-      int total_loss = 0;
-      for (const auto& observation : observations) {
-        total_loss += observation.num_lost_packets;
-        total_packets += observation.num_packets;
-      }
-      if (total_packets > 0) {
-        float average_loss_rate = total_loss * 100.0 / total_packets;
-        average_loss_rate_series.points.emplace_back(
-            config_.GetCallTimeSec(feedback.timestamp), average_loss_rate);
-      } else {
-        average_loss_rate_series.points.emplace_back(
-            config_.GetCallTimeSec(feedback.timestamp), 0);
-      }
-      window_summary = PacketLossSummary();
-    }
-  }
-  // Add the data set to the plot.
-  plot->AppendTimeSeriesIfNotEmpty(std::move(loss_rate_series));
-  plot->AppendTimeSeriesIfNotEmpty(
-      std::move(reordered_packets_between_feedback));
-  plot->AppendTimeSeriesIfNotEmpty(std::move(average_loss_rate_series));
-  plot->AppendTimeSeriesIfNotEmpty(std::move(missing_feedback_series));
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 100, "Loss rate (percent)", kBottomMargin,
-                          kTopMargin);
+  webrtc::CreateScreamDelayEstimateGraph(parsed_log_, config_, plot);
 }
 
 void EventLogAnalyzer::CreateSendSideBweSimulationGraph(Plot* plot) const {
-  using RtpPacketType = LoggedRtpPacketOutgoing;
-  using TransportFeedbackType = LoggedRtcpPacketTransportFeedback;
-
-  // TODO(terelius): This could be provided by the parser.
-  std::multimap<int64_t, const RtpPacketType*> outgoing_rtp;
-  for (const auto& stream : parsed_log_.outgoing_rtp_packets_by_ssrc()) {
-    for (const RtpPacketType& rtp_packet : stream.outgoing_packets)
-      outgoing_rtp.insert(
-          std::make_pair(rtp_packet.rtp.log_time_us(), &rtp_packet));
-  }
-
-  const std::vector<TransportFeedbackType>& incoming_rtcp =
-      parsed_log_.transport_feedbacks(kIncomingPacket);
-
-  SimulatedClock clock(0);
-  BitrateObserver observer;
-  TransportFeedbackAdapter transport_feedback;
-  auto factory = GoogCcNetworkControllerFactory();
-  TimeDelta process_interval = factory.GetProcessInterval();
-  // TODO(holmer): Log the call config and use that here instead.
-  static const uint32_t kDefaultStartBitrateBps = 300000;
-  NetworkControllerConfig cc_config(env_);
-  cc_config.constraints.at_time = clock.CurrentTime();
-  cc_config.constraints.starting_rate =
-      DataRate::BitsPerSec(kDefaultStartBitrateBps);
-  auto goog_cc = factory.Create(cc_config);
-
-  TimeSeries time_series("Delay-based estimate", LineStyle::kStep,
-                         PointStyle::kHighlight);
-  TimeSeries acked_time_series("Raw acked bitrate", LineStyle::kLine,
-                               PointStyle::kHighlight);
-  TimeSeries robust_time_series("Robust throughput estimate", LineStyle::kLine,
-                                PointStyle::kHighlight);
-  TimeSeries acked_estimate_time_series("Ackednowledged bitrate estimate",
-                                        LineStyle::kLine,
-                                        PointStyle::kHighlight);
-
-  auto rtp_iterator = outgoing_rtp.begin();
-  auto rtcp_iterator = incoming_rtcp.begin();
-
-  auto NextRtpTime = [&]() {
-    if (rtp_iterator != outgoing_rtp.end())
-      return static_cast<int64_t>(rtp_iterator->first);
-    return std::numeric_limits<int64_t>::max();
-  };
-
-  auto NextRtcpTime = [&]() {
-    if (rtcp_iterator != incoming_rtcp.end())
-      return static_cast<int64_t>(rtcp_iterator->log_time_us());
-    return std::numeric_limits<int64_t>::max();
-  };
-  int64_t next_process_time_us_ = std::min({NextRtpTime(), NextRtcpTime()});
-
-  auto NextProcessTime = [&]() {
-    if (rtcp_iterator != incoming_rtcp.end() ||
-        rtp_iterator != outgoing_rtp.end()) {
-      return next_process_time_us_;
-    }
-    return std::numeric_limits<int64_t>::max();
-  };
-
-  RateStatistics raw_acked_bitrate(750, 8000);
-  FieldTrials throughput_config(
-      "WebRTC-Bwe-RobustThroughputEstimatorSettings/enabled:true/");
-  std::unique_ptr<AcknowledgedBitrateEstimatorInterface>
-      robust_throughput_estimator(
-          AcknowledgedBitrateEstimatorInterface::Create(&throughput_config));
-  FieldTrials acked_bitrate_config(
-      "WebRTC-Bwe-RobustThroughputEstimatorSettings/enabled:false/");
-  std::unique_ptr<AcknowledgedBitrateEstimatorInterface>
-      acknowledged_bitrate_estimator(
-          AcknowledgedBitrateEstimatorInterface::Create(&acked_bitrate_config));
-  int64_t time_us =
-      std::min({NextRtpTime(), NextRtcpTime(), NextProcessTime()});
-  int64_t last_update_us = 0;
-  while (time_us != std::numeric_limits<int64_t>::max()) {
-    clock.AdvanceTimeMicroseconds(time_us - clock.TimeInMicroseconds());
-    if (clock.TimeInMicroseconds() >= NextRtpTime()) {
-      RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtpTime());
-      const RtpPacketType& rtp_packet = *rtp_iterator->second;
-      if (rtp_packet.rtp.header.extension.hasTransportSequenceNumber) {
-        RtpPacketToSend send_packet(/*extensions=*/nullptr);
-        send_packet.set_transport_sequence_number(
-            rtp_packet.rtp.header.extension.transportSequenceNumber);
-        send_packet.SetSsrc(rtp_packet.rtp.header.ssrc);
-        send_packet.SetSequenceNumber(rtp_packet.rtp.header.sequenceNumber);
-        send_packet.SetPayloadSize(rtp_packet.rtp.total_length -
-                                   send_packet.headers_size());
-        RTC_DCHECK_EQ(send_packet.size(), rtp_packet.rtp.total_length);
-        if (IsRtxSsrc(parsed_log_, PacketDirection::kOutgoingPacket,
-                      rtp_packet.rtp.header.ssrc)) {
-          // Don't set the optional media type as we don't know if it is
-          // a retransmission, FEC or padding.
-        } else if (IsVideoSsrc(parsed_log_, PacketDirection::kOutgoingPacket,
-                               rtp_packet.rtp.header.ssrc)) {
-          send_packet.set_packet_type(RtpPacketMediaType::kVideo);
-        } else if (IsAudioSsrc(parsed_log_, PacketDirection::kOutgoingPacket,
-                               rtp_packet.rtp.header.ssrc)) {
-          send_packet.set_packet_type(RtpPacketMediaType::kAudio);
-        }
-        transport_feedback.AddPacket(
-            send_packet, PacedPacketInfo(),
-            0u,  // Per packet overhead bytes.,
-            Timestamp::Micros(rtp_packet.rtp.log_time_us()));
-      }
-      SentPacketInfo sent_packet;
-      sent_packet.send_time_ms = rtp_packet.rtp.log_time_ms();
-      sent_packet.info.included_in_allocation = true;
-      sent_packet.info.packet_size_bytes = rtp_packet.rtp.total_length;
-      if (rtp_packet.rtp.header.extension.hasTransportSequenceNumber) {
-        sent_packet.packet_id =
-            rtp_packet.rtp.header.extension.transportSequenceNumber;
-        sent_packet.info.included_in_feedback = true;
-      }
-      auto sent_msg = transport_feedback.ProcessSentPacket(sent_packet);
-      if (sent_msg)
-        observer.Update(goog_cc->OnSentPacket(*sent_msg));
-      ++rtp_iterator;
-    }
-    if (clock.TimeInMicroseconds() >= NextRtcpTime()) {
-      RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtcpTime());
-
-      auto feedback_msg = transport_feedback.ProcessTransportFeedback(
-          rtcp_iterator->transport_feedback, clock.CurrentTime());
-      if (feedback_msg) {
-        observer.Update(goog_cc->OnTransportPacketsFeedback(*feedback_msg));
-        std::vector<PacketResult> feedback =
-            feedback_msg->SortedByReceiveTime();
-        if (!feedback.empty()) {
-          acknowledged_bitrate_estimator->IncomingPacketFeedbackVector(
-              feedback);
-          robust_throughput_estimator->IncomingPacketFeedbackVector(feedback);
-          for (const PacketResult& packet : feedback) {
-            raw_acked_bitrate.Update(packet.sent_packet.size.bytes(),
-                                     packet.receive_time.ms());
-          }
-          std::optional<uint32_t> raw_bitrate_bps =
-              raw_acked_bitrate.Rate(feedback.back().receive_time.ms());
-          float x = config_.GetCallTimeSec(clock.CurrentTime());
-          if (raw_bitrate_bps) {
-            float y = raw_bitrate_bps.value() / 1000;
-            acked_time_series.points.emplace_back(x, y);
-          }
-          std::optional<DataRate> robust_estimate =
-              robust_throughput_estimator->bitrate();
-          if (robust_estimate) {
-            float y = robust_estimate.value().kbps();
-            robust_time_series.points.emplace_back(x, y);
-          }
-          std::optional<DataRate> acked_estimate =
-              acknowledged_bitrate_estimator->bitrate();
-          if (acked_estimate) {
-            float y = acked_estimate.value().kbps();
-            acked_estimate_time_series.points.emplace_back(x, y);
-          }
-        }
-      }
-      ++rtcp_iterator;
-    }
-    if (clock.TimeInMicroseconds() >= NextProcessTime()) {
-      RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextProcessTime());
-      ProcessInterval msg;
-      msg.at_time = clock.CurrentTime();
-      observer.Update(goog_cc->OnProcessInterval(msg));
-      next_process_time_us_ += process_interval.us();
-    }
-    if (observer.GetAndResetBitrateUpdated() ||
-        time_us - last_update_us >= 1e6) {
-      uint32_t y = observer.last_bitrate_bps() / 1000;
-      float x = config_.GetCallTimeSec(clock.CurrentTime());
-      time_series.points.emplace_back(x, y);
-      last_update_us = time_us;
-    }
-    time_us = std::min({NextRtpTime(), NextRtcpTime(), NextProcessTime()});
-  }
-  // Add the data set to the plot.
-  plot->AppendTimeSeries(std::move(time_series));
-  plot->AppendTimeSeries(std::move(robust_time_series));
-  plot->AppendTimeSeries(std::move(acked_time_series));
-  plot->AppendTimeSeriesIfNotEmpty(std::move(acked_estimate_time_series));
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 10, "Bitrate (kbps)", kBottomMargin, kTopMargin);
-  plot->SetTitle("Simulated send-side BWE behavior");
+  webrtc::CreateSendSideBweSimulationGraph(parsed_log_, config_, plot);
 }
 
 void EventLogAnalyzer::CreateReceiveSideBweSimulationGraph(Plot* plot) const {
-  using RtpPacketType = LoggedRtpPacketIncoming;
-  class RembInterceptor {
-   public:
-    void SendRemb(uint32_t bitrate_bps, std::vector<uint32_t> ssrcs) {
-      last_bitrate_bps_ = bitrate_bps;
-      bitrate_updated_ = true;
-    }
-    uint32_t last_bitrate_bps() const { return last_bitrate_bps_; }
-    bool GetAndResetBitrateUpdated() {
-      bool bitrate_updated = bitrate_updated_;
-      bitrate_updated_ = false;
-      return bitrate_updated;
-    }
-
-   private:
-    // We don't know the start bitrate, but assume that it is the default 300
-    // kbps.
-    uint32_t last_bitrate_bps_ = 300000;
-    bool bitrate_updated_ = false;
-  };
-
-  std::multimap<int64_t, const RtpPacketType*> incoming_rtp;
-
-  for (const auto& stream : parsed_log_.incoming_rtp_packets_by_ssrc()) {
-    if (IsVideoSsrc(parsed_log_, kIncomingPacket, stream.ssrc)) {
-      for (const auto& rtp_packet : stream.incoming_packets)
-        incoming_rtp.insert(
-            std::make_pair(rtp_packet.rtp.log_time_us(), &rtp_packet));
-    }
-  }
-
-  SimulatedClock clock(0);
-  EnvironmentFactory env_factory(env_);
-  env_factory.Set(&clock);
-  RembInterceptor remb_interceptor;
-  ReceiveSideCongestionController rscc(
-      env_factory.Create(), [](auto...) {},
-      absl::bind_front(&RembInterceptor::SendRemb, &remb_interceptor));
-  // TODO(holmer): Log the call config and use that here instead.
-  // static const uint32_t kDefaultStartBitrateBps = 300000;
-  // rscc.SetBweBitrates(0, kDefaultStartBitrateBps, -1);
-
-  TimeSeries time_series("Receive side estimate", LineStyle::kLine,
-                         PointStyle::kHighlight);
-  TimeSeries acked_time_series("Received bitrate", LineStyle::kLine);
-
-  RateStatistics acked_bitrate(250, 8000);
-  int64_t last_update_us = 0;
-  for (const auto& kv : incoming_rtp) {
-    const RtpPacketType& packet = *kv.second;
-
-    RtpPacketReceived rtp_packet = RtpPacketForBWEFromHeader(packet.rtp.header);
-    rtp_packet.set_arrival_time(packet.rtp.log_time());
-    rtp_packet.SetPayloadSize(packet.rtp.total_length -
-                              rtp_packet.headers_size());
-
-    clock.AdvanceTime(rtp_packet.arrival_time() - clock.CurrentTime());
-    rscc.OnReceivedPacket(rtp_packet, MediaType::VIDEO);
-    int64_t arrival_time_ms = packet.rtp.log_time().ms();
-    acked_bitrate.Update(packet.rtp.total_length, arrival_time_ms);
-    std::optional<uint32_t> bitrate_bps = acked_bitrate.Rate(arrival_time_ms);
-    if (bitrate_bps) {
-      uint32_t y = *bitrate_bps / 1000;
-      float x = config_.GetCallTimeSec(clock.CurrentTime());
-      acked_time_series.points.emplace_back(x, y);
-    }
-    if (remb_interceptor.GetAndResetBitrateUpdated() ||
-        clock.TimeInMicroseconds() - last_update_us >= 1e6) {
-      uint32_t y = remb_interceptor.last_bitrate_bps() / 1000;
-      float x = config_.GetCallTimeSec(clock.CurrentTime());
-      time_series.points.emplace_back(x, y);
-      last_update_us = clock.TimeInMicroseconds();
-    }
-  }
-  // Add the data set to the plot.
-  plot->AppendTimeSeries(std::move(time_series));
-  plot->AppendTimeSeries(std::move(acked_time_series));
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 10, "Bitrate (kbps)", kBottomMargin, kTopMargin);
-  plot->SetTitle("Simulated receive-side BWE behavior");
+  webrtc::CreateReceiveSideBweSimulationGraph(parsed_log_, config_, plot);
 }
 
 void EventLogAnalyzer::CreateNetworkDelayFeedbackGraph(Plot* plot) const {
-  TimeSeries time_series("Network delay", LineStyle::kLine,
-                         PointStyle::kHighlight);
-  int64_t min_send_receive_diff_ms = std::numeric_limits<int64_t>::max();
-  int64_t min_rtt_ms = std::numeric_limits<int64_t>::max();
-
-  std::vector<MatchedSendArrivalTimes> matched_rtp_rtcp =
-      GetNetworkTrace(parsed_log_);
-  absl::c_stable_sort(matched_rtp_rtcp, [](const MatchedSendArrivalTimes& a,
-                                           const MatchedSendArrivalTimes& b) {
-    return a.feedback_arrival_time_ms < b.feedback_arrival_time_ms ||
-           (a.feedback_arrival_time_ms == b.feedback_arrival_time_ms &&
-            a.arrival_time_ms < b.arrival_time_ms);
-  });
-  for (const auto& packet : matched_rtp_rtcp) {
-    if (packet.arrival_time_ms == MatchedSendArrivalTimes::kNotReceived)
-      continue;
-    float x = config_.GetCallTimeSecFromMs(packet.feedback_arrival_time_ms);
-    int64_t y = packet.arrival_time_ms - packet.send_time_ms;
-    int64_t rtt_ms = packet.feedback_arrival_time_ms - packet.send_time_ms;
-    min_rtt_ms = std::min(rtt_ms, min_rtt_ms);
-    min_send_receive_diff_ms = std::min(y, min_send_receive_diff_ms);
-    time_series.points.emplace_back(x, y);
-  }
-
-  // We assume that the base network delay (w/o queues) is equal to half
-  // the minimum RTT. Therefore rescale the delays by subtracting the minimum
-  // observed 1-ways delay and add half the minimum RTT.
-  const int64_t estimated_clock_offset_ms =
-      min_send_receive_diff_ms - min_rtt_ms / 2;
-  for (TimeSeriesPoint& point : time_series.points)
-    point.y -= estimated_clock_offset_ms;
-
-  // Add the data set to the plot.
-  plot->AppendTimeSeriesIfNotEmpty(std::move(time_series));
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 10, "Delay (ms)", kBottomMargin, kTopMargin);
-  plot->SetTitle("Outgoing network delay (based on per-packet feedback)");
+  webrtc::CreateNetworkDelayFeedbackGraph(parsed_log_, config_, plot);
 }
 
 void EventLogAnalyzer::CreatePacerDelayGraph(Plot* plot) const {
-  for (const auto& stream : parsed_log_.outgoing_rtp_packets_by_ssrc()) {
-    const std::vector<LoggedRtpPacketOutgoing>& packets =
-        stream.outgoing_packets;
+  webrtc::CreatePacerDelayGraph(parsed_log_, config_, plot);
+}
 
-    if (IsRtxSsrc(parsed_log_, kOutgoingPacket, stream.ssrc)) {
-      continue;
-    }
+void EventLogAnalyzer::CreateIceCandidatePairConfigGraph(Plot* plot) const {
+  webrtc::CreateIceCandidatePairConfigGraph(parsed_log_, config_, plot);
+}
 
-    if (packets.size() < 2) {
-      RTC_LOG(LS_WARNING)
-          << "Can't estimate a the RTP clock frequency or the "
-             "pacer delay with less than 2 packets in the stream";
-      continue;
-    }
-    int64_t segment_end_us = parsed_log_.first_log_segment().stop_time_us();
-    std::optional<uint32_t> estimated_frequency =
-        EstimateRtpClockFrequency(packets, segment_end_us);
-    if (!estimated_frequency)
-      continue;
-    if (IsVideoSsrc(parsed_log_, kOutgoingPacket, stream.ssrc) &&
-        *estimated_frequency != 90000) {
-      RTC_LOG(LS_WARNING)
-          << "Video stream should use a 90 kHz clock but appears to use "
-          << *estimated_frequency / 1000 << ". Discarding.";
-      continue;
-    }
+void EventLogAnalyzer::CreateIceConnectivityCheckGraph(Plot* plot) const {
+  webrtc::CreateIceConnectivityCheckGraph(parsed_log_, config_, plot);
+}
 
-    TimeSeries pacer_delay_series(
-        GetStreamName(parsed_log_, kOutgoingPacket, stream.ssrc) + "(" +
-            std::to_string(*estimated_frequency / 1000) + " kHz)",
-        LineStyle::kLine, PointStyle::kHighlight);
-    SeqNumUnwrapper<uint32_t> timestamp_unwrapper;
-    uint64_t first_capture_timestamp =
-        timestamp_unwrapper.Unwrap(packets.front().rtp.header.timestamp);
-    uint64_t first_send_timestamp = packets.front().rtp.log_time_us();
-    for (const auto& packet : packets) {
-      double capture_time_ms = (static_cast<double>(timestamp_unwrapper.Unwrap(
-                                    packet.rtp.header.timestamp)) -
-                                first_capture_timestamp) /
-                               *estimated_frequency * 1000;
-      double send_time_ms =
-          static_cast<double>(packet.rtp.log_time_us() - first_send_timestamp) /
-          1000;
-      float x = config_.GetCallTimeSec(packet.rtp.log_time());
-      float y = send_time_ms - capture_time_ms;
-      pacer_delay_series.points.emplace_back(x, y);
-    }
-    plot->AppendTimeSeries(std::move(pacer_delay_series));
-  }
+void EventLogAnalyzer::CreateDtlsTransportStateGraph(Plot* plot) const {
+  webrtc::CreateDtlsTransportStateGraph(parsed_log_, config_, plot);
+}
 
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 10, "Pacer delay (ms)", kBottomMargin, kTopMargin);
-  plot->SetTitle(
-      "Delay from capture to send time. (First packet normalized to 0.)");
+void EventLogAnalyzer::CreateDtlsWritableStateGraph(Plot* plot) const {
+  webrtc::CreateDtlsWritableStateGraph(parsed_log_, config_, plot);
+}
+
+void EventLogAnalyzer::CreatePacketGraph(PacketDirection direction,
+                                         Plot* plot) const {
+  webrtc::CreatePacketGraph(direction, parsed_log_, config_, plot);
+}
+
+void EventLogAnalyzer::CreateRtcpTypeGraph(PacketDirection direction,
+                                           Plot* plot) const {
+  webrtc::CreateRtcpTypeGraph(direction, parsed_log_, config_, plot);
+}
+
+void EventLogAnalyzer::CreateAccumulatedPacketsGraph(PacketDirection direction,
+                                                     Plot* plot) const {
+  webrtc::CreateAccumulatedPacketsGraph(direction, parsed_log_, config_, plot);
+}
+
+void EventLogAnalyzer::CreatePacketRateGraph(PacketDirection direction,
+                                             Plot* plot) const {
+  webrtc::CreatePacketRateGraph(direction, parsed_log_, config_, plot);
+}
+
+void EventLogAnalyzer::CreateTotalPacketRateGraph(PacketDirection direction,
+                                                  Plot* plot) const {
+  webrtc::CreateTotalPacketRateGraph(direction, parsed_log_, config_, plot);
+}
+
+void EventLogAnalyzer::CreateSequenceNumberGraph(Plot* plot) const {
+  webrtc::CreateSequenceNumberGraph(parsed_log_, config_, plot);
+}
+
+void EventLogAnalyzer::CreateIncomingPacketLossGraph(Plot* plot) const {
+  webrtc::CreateIncomingPacketLossGraph(parsed_log_, config_, plot);
+}
+
+void EventLogAnalyzer::CreateStreamBitrateGraph(PacketDirection direction,
+                                                Plot* plot) const {
+  webrtc::CreateStreamBitrateGraph(direction, parsed_log_, config_, plot);
+}
+
+void EventLogAnalyzer::CreateBitrateAllocationGraph(PacketDirection direction,
+                                                    Plot* plot) const {
+  webrtc::CreateBitrateAllocationGraph(direction, parsed_log_, config_, plot);
+}
+
+void EventLogAnalyzer::CreateOutgoingEcnFeedbackGraph(Plot* plot) const {
+  webrtc::CreateOutgoingEcnFeedbackGraph(parsed_log_, config_, plot);
+}
+
+void EventLogAnalyzer::CreateIncomingEcnFeedbackGraph(Plot* plot) const {
+  webrtc::CreateIncomingEcnFeedbackGraph(parsed_log_, config_, plot);
+}
+
+void EventLogAnalyzer::CreateOutgoingLossRateGraph(Plot* plot) const {
+  webrtc::CreateOutgoingLossRateGraph(parsed_log_, config_, plot);
 }
 
 void EventLogAnalyzer::CreateTimestampGraph(PacketDirection direction,
                                             Plot* plot) const {
-  for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
-    TimeSeries rtp_timestamps(
-        GetStreamName(parsed_log_, direction, stream.ssrc) + " capture-time",
-        LineStyle::kLine, PointStyle::kHighlight);
-    for (const auto& packet : stream.packet_view) {
-      float x = config_.GetCallTimeSec(packet.log_time());
-      float y = packet.header.timestamp;
-      rtp_timestamps.points.emplace_back(x, y);
-    }
-    plot->AppendTimeSeries(std::move(rtp_timestamps));
-
-    TimeSeries rtcp_timestamps(
-        GetStreamName(parsed_log_, direction, stream.ssrc) +
-            " rtcp capture-time",
-        LineStyle::kLine, PointStyle::kHighlight);
-    // TODO(terelius): Why only sender reports?
-    const auto& sender_reports = parsed_log_.sender_reports(direction);
-    for (const auto& rtcp : sender_reports) {
-      if (rtcp.sr.sender_ssrc() != stream.ssrc)
-        continue;
-      float x = config_.GetCallTimeSec(rtcp.log_time());
-      float y = rtcp.sr.rtp_timestamp();
-      rtcp_timestamps.points.emplace_back(x, y);
-    }
-    plot->AppendTimeSeriesIfNotEmpty(std::move(rtcp_timestamps));
-  }
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "RTP timestamp", kBottomMargin, kTopMargin);
-  plot->SetTitle(GetDirectionAsString(direction) + " timestamps");
+  webrtc::CreateTimestampGraph(direction, parsed_log_, config_, plot);
 }
 
 void EventLogAnalyzer::CreateSenderAndReceiverReportPlot(
@@ -2403,174 +439,8 @@ void EventLogAnalyzer::CreateSenderAndReceiverReportPlot(
     std::string title,
     std::string yaxis_label,
     Plot* plot) const {
-  std::map<uint32_t, TimeSeries> sr_reports_by_ssrc;
-  const auto& sender_reports = parsed_log_.sender_reports(direction);
-  for (const auto& rtcp : sender_reports) {
-    float x = config_.GetCallTimeSec(rtcp.log_time());
-    uint32_t ssrc = rtcp.sr.sender_ssrc();
-    for (const auto& block : rtcp.sr.report_blocks()) {
-      float y = fy(block);
-      auto sr_report_it = sr_reports_by_ssrc.find(ssrc);
-      bool inserted;
-      if (sr_report_it == sr_reports_by_ssrc.end()) {
-        std::tie(sr_report_it, inserted) = sr_reports_by_ssrc.emplace(
-            ssrc, TimeSeries(GetStreamName(parsed_log_, direction, ssrc) +
-                                 " Sender Reports",
-                             LineStyle::kLine, PointStyle::kHighlight));
-      }
-      sr_report_it->second.points.emplace_back(x, y);
-    }
-  }
-  for (auto& kv : sr_reports_by_ssrc) {
-    plot->AppendTimeSeries(std::move(kv.second));
-  }
-
-  std::map<uint32_t, TimeSeries> rr_reports_by_ssrc;
-  const auto& receiver_reports = parsed_log_.receiver_reports(direction);
-  for (const auto& rtcp : receiver_reports) {
-    float x = config_.GetCallTimeSec(rtcp.log_time());
-    uint32_t ssrc = rtcp.rr.sender_ssrc();
-    for (const auto& block : rtcp.rr.report_blocks()) {
-      float y = fy(block);
-      auto rr_report_it = rr_reports_by_ssrc.find(ssrc);
-      bool inserted;
-      if (rr_report_it == rr_reports_by_ssrc.end()) {
-        std::tie(rr_report_it, inserted) = rr_reports_by_ssrc.emplace(
-            ssrc, TimeSeries(GetStreamName(parsed_log_, direction, ssrc) +
-                                 " Receiver Reports",
-                             LineStyle::kLine, PointStyle::kHighlight));
-      }
-      rr_report_it->second.points.emplace_back(x, y);
-    }
-  }
-  for (auto& kv : rr_reports_by_ssrc) {
-    plot->AppendTimeSeries(std::move(kv.second));
-  }
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, yaxis_label, kBottomMargin, kTopMargin);
-  plot->SetTitle(title);
-}
-
-void EventLogAnalyzer::CreateIceCandidatePairConfigGraph(Plot* plot) const {
-  std::map<uint32_t, TimeSeries> configs_by_cp_id;
-  for (const auto& config : parsed_log_.ice_candidate_pair_configs()) {
-    if (configs_by_cp_id.find(config.candidate_pair_id) ==
-        configs_by_cp_id.end()) {
-      const std::string candidate_pair_desc =
-          GetCandidatePairLogDescriptionAsString(config);
-      configs_by_cp_id[config.candidate_pair_id] =
-          TimeSeries("[" + std::to_string(config.candidate_pair_id) + "]" +
-                         candidate_pair_desc,
-                     LineStyle::kNone, PointStyle::kHighlight);
-    }
-    float x = config_.GetCallTimeSec(config.log_time());
-    float y = static_cast<float>(config.type);
-    configs_by_cp_id[config.candidate_pair_id].points.emplace_back(x, y);
-  }
-
-  // TODO(qingsi): There can be a large number of candidate pairs generated by
-  // certain calls and the frontend cannot render the chart in this case due
-  // to the failure of generating a palette with the same number of colors.
-  for (auto& kv : configs_by_cp_id) {
-    plot->AppendTimeSeries(std::move(kv.second));
-  }
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 3, "Config Type", kBottomMargin, kTopMargin);
-  plot->SetTitle("[IceEventLog] ICE candidate pair configs");
-  plot->SetYAxisTickLabels(
-      {{static_cast<float>(IceCandidatePairConfigType::kAdded), "ADDED"},
-       {static_cast<float>(IceCandidatePairConfigType::kUpdated), "UPDATED"},
-       {static_cast<float>(IceCandidatePairConfigType::kDestroyed),
-        "DESTROYED"},
-       {static_cast<float>(IceCandidatePairConfigType::kSelected),
-        "SELECTED"}});
-}
-
-void EventLogAnalyzer::CreateIceConnectivityCheckGraph(Plot* plot) const {
-  constexpr int kEventTypeOffset =
-      static_cast<int>(IceCandidatePairConfigType::kNumValues);
-  std::map<uint32_t, TimeSeries> checks_by_cp_id;
-  std::map<uint32_t, std::string> candidate_pair_desc_by_id =
-      BuildCandidateIdLogDescriptionMap(
-          parsed_log_.ice_candidate_pair_configs());
-  for (const auto& event : parsed_log_.ice_candidate_pair_events()) {
-    if (checks_by_cp_id.find(event.candidate_pair_id) ==
-        checks_by_cp_id.end()) {
-      checks_by_cp_id[event.candidate_pair_id] =
-          TimeSeries("[" + std::to_string(event.candidate_pair_id) + "]" +
-                         candidate_pair_desc_by_id[event.candidate_pair_id],
-                     LineStyle::kNone, PointStyle::kHighlight);
-    }
-    float x = config_.GetCallTimeSec(event.log_time());
-    float y = static_cast<float>(event.type) + kEventTypeOffset;
-    checks_by_cp_id[event.candidate_pair_id].points.emplace_back(x, y);
-  }
-
-  // TODO(qingsi): The same issue as in CreateIceCandidatePairConfigGraph.
-  for (auto& kv : checks_by_cp_id) {
-    plot->AppendTimeSeries(std::move(kv.second));
-  }
-
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 4, "Connectivity State", kBottomMargin,
-                          kTopMargin);
-  plot->SetTitle("[IceEventLog] ICE connectivity checks");
-
-  plot->SetYAxisTickLabels(
-      {{static_cast<float>(IceCandidatePairEventType::kCheckSent) +
-            kEventTypeOffset,
-        "CHECK SENT"},
-       {static_cast<float>(IceCandidatePairEventType::kCheckReceived) +
-            kEventTypeOffset,
-        "CHECK RECEIVED"},
-       {static_cast<float>(IceCandidatePairEventType::kCheckResponseSent) +
-            kEventTypeOffset,
-        "RESPONSE SENT"},
-       {static_cast<float>(IceCandidatePairEventType::kCheckResponseReceived) +
-            kEventTypeOffset,
-        "RESPONSE RECEIVED"}});
-}
-
-void EventLogAnalyzer::CreateDtlsTransportStateGraph(Plot* plot) const {
-  TimeSeries states("DTLS Transport State", LineStyle::kNone,
-                    PointStyle::kHighlight);
-  for (const auto& event : parsed_log_.dtls_transport_states()) {
-    float x = config_.GetCallTimeSec(event.log_time());
-    float y = static_cast<float>(event.dtls_transport_state);
-    states.points.emplace_back(x, y);
-  }
-  plot->AppendTimeSeries(std::move(states));
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, static_cast<float>(DtlsTransportState::kNumValues),
-                          "Transport State", kBottomMargin, kTopMargin);
-  plot->SetTitle("DTLS Transport State");
-  plot->SetYAxisTickLabels(
-      {{static_cast<float>(DtlsTransportState::kNew), "NEW"},
-       {static_cast<float>(DtlsTransportState::kConnecting), "CONNECTING"},
-       {static_cast<float>(DtlsTransportState::kConnected), "CONNECTED"},
-       {static_cast<float>(DtlsTransportState::kClosed), "CLOSED"},
-       {static_cast<float>(DtlsTransportState::kFailed), "FAILED"}});
-}
-
-void EventLogAnalyzer::CreateDtlsWritableStateGraph(Plot* plot) const {
-  TimeSeries writable("DTLS Writable", LineStyle::kNone,
-                      PointStyle::kHighlight);
-  for (const auto& event : parsed_log_.dtls_writable_states()) {
-    float x = config_.GetCallTimeSec(event.log_time());
-    float y = static_cast<float>(event.writable);
-    writable.points.emplace_back(x, y);
-  }
-  plot->AppendTimeSeries(std::move(writable));
-  plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
-                 "Time (s)", kLeftMargin, kRightMargin);
-  plot->SetSuggestedYAxis(0, 1, "Writable", kBottomMargin, kTopMargin);
-  plot->SetTitle("DTLS Writable State");
+  webrtc::CreateSenderAndReceiverReportPlot(direction, fy, title, yaxis_label,
+                                            parsed_log_, config_, plot);
 }
 
 }  // namespace webrtc
diff --git a/rtc_tools/rtc_event_log_visualizer/analyzer.h b/rtc_tools/rtc_event_log_visualizer/analyzer.h
index ca729b3cbc..8ac90e3040 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyzer.h
+++ b/rtc_tools/rtc_event_log_visualizer/analyzer.h
@@ -11,13 +11,11 @@
 #ifndef RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZER_H_
 #define RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZER_H_
 
-#include <cstdint>
 #include <cstdio>
 #include <functional>
 #include <string>
 #include <vector>
 
-#include "api/environment/environment.h"
 #include "api/function_view.h"
 #include "logging/rtc_event_log/rtc_event_log_parser.h"
 #include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
@@ -62,9 +60,7 @@ class EventLogAnalyzer {
   // duration of its lifetime. The ParsedRtcEventLogNew must not be destroyed or
   // modified while the EventLogAnalyzer is being used.
   EventLogAnalyzer(const ParsedRtcEventLog& log, bool normalize_time);
-  EventLogAnalyzer(const Environment& env,
-                   const ParsedRtcEventLog& log,
-                   const AnalyzerConfig& config);
+  EventLogAnalyzer(const ParsedRtcEventLog& log, const AnalyzerConfig& config);
 
   void CreateGraphsByName(const std::vector<std::string>& names,
                           PlotCollection* collection) const;
@@ -149,19 +145,8 @@ class EventLogAnalyzer {
   void PrintNotifications(FILE* file) const;
 
  private:
-  template <typename IterableType>
-  void CreateAccumulatedPacketsTimeSeries(Plot* plot,
-                                          const IterableType& packets,
-                                          const std::string& label) const;
-  void CreateEcnFeedbackGraph(Plot* plot, PacketDirection direction) const;
-
-  const Environment env_;
   const ParsedRtcEventLog& parsed_log_;
 
-  // A list of SSRCs we are interested in analysing.
-  // If left empty, all SSRCs will be considered relevant.
-  std::vector<uint32_t> desired_ssrc_;
-
   AnalyzerConfig config_;
 
   PlotMap plots_;
diff --git a/rtc_tools/rtc_event_log_visualizer/analyzer_bindings.cc b/rtc_tools/rtc_event_log_visualizer/analyzer_bindings.cc
index fa93031ff1..4aed263a98 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyzer_bindings.cc
+++ b/rtc_tools/rtc_event_log_visualizer/analyzer_bindings.cc
@@ -64,16 +64,10 @@ void analyze_rtc_event_log(const char* log_contents,
     return;
   }
 
-  webrtc::AnalyzerConfig config;
+  webrtc::AnalyzerConfig config(webrtc::CreateEnvironment(), parsed_log,
+                                /*normalize_time=*/true);
   config.window_duration_ = webrtc::TimeDelta::Millis(250);
   config.step_ = webrtc::TimeDelta::Millis(10);
-  if (!parsed_log.start_log_events().empty()) {
-    config.rtc_to_utc_offset_ = parsed_log.start_log_events()[0].utc_time() -
-                                parsed_log.start_log_events()[0].log_time();
-  }
-  config.normalize_time_ = true;
-  config.begin_time_ = parsed_log.first_timestamp();
-  config.end_time_ = parsed_log.last_timestamp();
   if (config.end_time_ < config.begin_time_) {
     std::cerr << "Log end time " << config.end_time_.ms()
               << " not after begin time " << config.begin_time_.ms()
@@ -82,8 +76,7 @@ void analyze_rtc_event_log(const char* log_contents,
     return;
   }
 
-  webrtc::EventLogAnalyzer analyzer(webrtc::CreateEnvironment(), parsed_log,
-                                    config);
+  webrtc::EventLogAnalyzer analyzer(parsed_log, config);
   analyzer.InitializeMapOfNamedGraphs(/*show_detector_state=*/false,
                                       /*show_alr_state=*/false,
                                       /*show_link_capacity=*/false);
diff --git a/rtc_tools/rtc_event_log_visualizer/analyzer_common.cc b/rtc_tools/rtc_event_log_visualizer/analyzer_common.cc
index c1cf7dc6b7..559ad78b4a 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyzer_common.cc
+++ b/rtc_tools/rtc_event_log_visualizer/analyzer_common.cc
@@ -11,10 +11,14 @@
 
 #include "rtc_tools/rtc_event_log_visualizer/analyzer_common.h"
 
+#include <algorithm>
 #include <cstdint>
 #include <string>
+#include <vector>
 
 #include "logging/rtc_event_log/rtc_event_log_parser.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
 #include "rtc_base/strings/string_builder.h"
 
 namespace webrtc {
@@ -86,4 +90,64 @@ std::string GetLayerName(LayerDescription layer) {
   return name.str();
 }
 
+std::string SsrcToString(uint32_t ssrc) {
+  StringBuilder ss;
+  ss << "SSRC " << ssrc;
+  return ss.Release();
+}
+
+// Checks whether an SSRC is contained in the list of desired SSRCs.
+// Note that an empty SSRC list matches every SSRC.
+bool MatchingSsrc(uint32_t ssrc, const std::vector<uint32_t>& desired_ssrc) {
+  if (desired_ssrc.empty())
+    return true;
+  return std::find(desired_ssrc.begin(), desired_ssrc.end(), ssrc) !=
+         desired_ssrc.end();
+}
+
+// Computes the difference `later` - `earlier` where `later` and `earlier`
+// are counters that wrap at `modulus`. The difference is chosen to have the
+// least absolute value. For example if `modulus` is 8, then the difference will
+// be chosen in the range [-3, 4]. If `modulus` is 9, then the difference will
+// be in [-4, 4].
+int64_t WrappingDifference(uint32_t later, uint32_t earlier, int64_t modulus) {
+  RTC_DCHECK_LE(1, modulus);
+  RTC_DCHECK_LT(later, modulus);
+  RTC_DCHECK_LT(earlier, modulus);
+  int64_t difference =
+      static_cast<int64_t>(later) - static_cast<int64_t>(earlier);
+  int64_t max_difference = modulus / 2;
+  int64_t min_difference = max_difference - modulus + 1;
+  if (difference > max_difference) {
+    difference -= modulus;
+  }
+  if (difference < min_difference) {
+    difference += modulus;
+  }
+  if (difference > max_difference / 2 || difference < min_difference / 2) {
+    RTC_LOG(LS_WARNING) << "Difference between" << later << " and " << earlier
+                        << " expected to be in the range ("
+                        << min_difference / 2 << "," << max_difference / 2
+                        << ") but is " << difference
+                        << ". Correct unwrapping is uncertain.";
+  }
+  return difference;
+}
+
+std::string GetDirectionAsString(PacketDirection direction) {
+  if (direction == kIncomingPacket) {
+    return "Incoming";
+  } else {
+    return "Outgoing";
+  }
+}
+
+std::string GetDirectionAsShortString(PacketDirection direction) {
+  if (direction == kIncomingPacket) {
+    return "In";
+  } else {
+    return "Out";
+  }
+}
+
 }  // namespace webrtc
diff --git a/rtc_tools/rtc_event_log_visualizer/analyzer_common.h b/rtc_tools/rtc_event_log_visualizer/analyzer_common.h
index d2698f84b1..4ef2e676c1 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyzer_common.h
+++ b/rtc_tools/rtc_event_log_visualizer/analyzer_common.h
@@ -15,7 +15,10 @@
 #include <cstdint>
 #include <optional>
 #include <string>
+#include <vector>
 
+#include "api/environment/environment.h"
+#include "api/environment/environment_factory.h"
 #include "api/function_view.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
@@ -33,6 +36,20 @@ constexpr float kTopMargin = 0.05f;
 
 class AnalyzerConfig {
  public:
+  AnalyzerConfig() : normalize_time_(true), env_(CreateEnvironment()) {}
+
+  AnalyzerConfig(const Environment& env,
+                 const ParsedRtcEventLog& parsed_log,
+                 bool normalize_time)
+      : normalize_time_(normalize_time), env_(env) {
+    begin_time_ = parsed_log.first_timestamp();
+    end_time_ = parsed_log.last_timestamp();
+    if (!parsed_log.start_log_events().empty()) {
+      rtc_to_utc_offset_ = parsed_log.start_log_events().front().utc_time() -
+                           parsed_log.start_log_events().front().log_time();
+    }
+  }
+
   float GetCallTimeSec(Timestamp timestamp) const {
     Timestamp offset = normalize_time_ ? begin_time_ : Timestamp::Zero();
     return static_cast<float>((timestamp - offset).us()) / 1000000;
@@ -66,6 +83,8 @@ class AnalyzerConfig {
   Timestamp end_time_ = Timestamp::MinusInfinity();
   TimeDelta rtc_to_utc_offset_ = TimeDelta::Zero();
   bool normalize_time_;
+  std::vector<uint32_t> desired_ssrc_;
+  const Environment env_;
 };
 
 struct LayerDescription {
@@ -98,8 +117,19 @@ bool IsAudioSsrc(const ParsedRtcEventLog& parsed_log,
 std::string GetStreamName(const ParsedRtcEventLog& parsed_log,
                           PacketDirection direction,
                           uint32_t ssrc);
+std::string SsrcToString(uint32_t ssrc);
+
 std::string GetLayerName(LayerDescription layer);
 
+std::string GetDirectionAsString(PacketDirection direction);
+std::string GetDirectionAsShortString(PacketDirection direction);
+
+int64_t WrappingDifference(uint32_t later, uint32_t earlier, int64_t modulus);
+
+// Checks whether an SSRC is contained in the list of desired SSRCs.
+// Note that an empty SSRC list matches every SSRC.
+bool MatchingSsrc(uint32_t ssrc, const std::vector<uint32_t>& desired_ssrc);
+
 // For each element in data_view, use `f()` to extract a y-coordinate and
 // store the result in a TimeSeries.
 template <typename DataType, typename IterableType>
diff --git a/rtc_tools/rtc_event_log_visualizer/main.cc b/rtc_tools/rtc_event_log_visualizer/main.cc
index 03512b185b..a08588e8a1 100644
--- a/rtc_tools/rtc_event_log_visualizer/main.cc
+++ b/rtc_tools/rtc_event_log_visualizer/main.cc
@@ -192,17 +192,11 @@ int main(int argc, char* argv[]) {
     }
   }
 
-  webrtc::AnalyzerConfig config;
+  webrtc::AnalyzerConfig config(env, parsed_log,
+                                absl::GetFlag(FLAGS_normalize_time));
   config.window_duration_ =
       webrtc::TimeDelta::Millis(absl::GetFlag(FLAGS_averaging_window));
   config.step_ = webrtc::TimeDelta::Millis(absl::GetFlag(FLAGS_averaging_step));
-  if (!parsed_log.start_log_events().empty()) {
-    config.rtc_to_utc_offset_ = parsed_log.start_log_events()[0].utc_time() -
-                                parsed_log.start_log_events()[0].log_time();
-  }
-  config.normalize_time_ = absl::GetFlag(FLAGS_normalize_time);
-  config.begin_time_ = parsed_log.first_timestamp();
-  config.end_time_ = parsed_log.last_timestamp();
   if (config.end_time_ < config.begin_time_) {
     RTC_LOG(LS_WARNING) << "Log end time " << config.end_time_
                         << " not after begin time " << config.begin_time_
@@ -225,7 +219,7 @@ int main(int argc, char* argv[]) {
     has_generated_wav_file = true;
   }
 
-  webrtc::EventLogAnalyzer analyzer(env, parsed_log, config);
+  webrtc::EventLogAnalyzer analyzer(parsed_log, config);
   analyzer.InitializeMapOfNamedGraphs(absl::GetFlag(FLAGS_show_detector_state),
                                       absl::GetFlag(FLAGS_show_alr_state),
                                       absl::GetFlag(FLAGS_show_link_capacity));

From e27b7b67aa82f6c27279ffb2f1bbf24d6bc6041a Mon Sep 17 00:00:00 2001
From: Victor Hugo Vianna Silva <victorvianna@google.com>
Date: Fri, 13 Feb 2026 14:57:59 +0000
Subject: [PATCH 046/472] Remove reference to catapult's github repository

That repo was archived. Prefer using the corresponding chromium URL.

Bug: None
Change-Id: I5dcc18641a463c91da9aa06150a96d206380ecba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449120
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Auto-Submit: Victor Vianna <victorvianna@google.com>
Commit-Queue: Victor Vianna <victorvianna@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46899}
---
 test/testsupport/perf_test.h | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/test/testsupport/perf_test.h b/test/testsupport/perf_test.h
index f151340021..9773b7c873 100644
--- a/test/testsupport/perf_test.h
+++ b/test/testsupport/perf_test.h
@@ -93,7 +93,7 @@ void PrintResult(absl::string_view measurement,
 
 // Returns a string-encoded proto as described in
 // tracing/tracing/proto/histogram.proto in
-// https://github.com/catapult-project/catapult/blob/master/.
+// https://chromium.googlesource.com/catapult/+/refs/heads/main.
 // If you want to print the proto in human readable format, use
 // tracing/bin/proto2json from third_party/catapult in your WebRTC checkout.
 std::string GetPerfResults();

From f81c7f5156e54146b6b8de154300ccb8dc1e0a31 Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Fri, 13 Feb 2026 13:21:27 +0000
Subject: [PATCH 047/472] Spanify some files in pc/

Bug: webrtc:478086887
Change-Id: Id03a770fb9e752985e9854d061891a6b88416842
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/448820
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46900}
---
 pc/dtls_srtp_transport_unittest.cc       |  5 +-
 pc/ice_server_parsing.cc                 |  4 +-
 pc/jsep_transport_unittest.cc            |  3 +-
 pc/peer_connection_histogram_unittest.cc |  5 +-
 pc/srtp_transport_unittest.cc            |  5 +-
 pc/test/fake_audio_capture_module.cc     | 20 ++----
 pc/test/fake_audio_capture_module.h      |  5 +-
 pc/test/fake_rtc_certificate_generator.h | 83 ++++++++++++------------
 unsafe_buffers_paths.txt                 |  7 --
 9 files changed, 65 insertions(+), 72 deletions(-)

diff --git a/pc/dtls_srtp_transport_unittest.cc b/pc/dtls_srtp_transport_unittest.cc
index d6e259a47b..c3cd857ae5 100644
--- a/pc/dtls_srtp_transport_unittest.cc
+++ b/pc/dtls_srtp_transport_unittest.cc
@@ -18,6 +18,7 @@
 #include <vector>
 
 #include "absl/strings/string_view.h"
+#include "api/array_view.h"
 #include "api/field_trials.h"
 #include "api/make_ref_counted.h"
 #include "api/transport/ecn_marking.h"
@@ -137,7 +138,7 @@ class DtlsSrtpTransportTest : public ::testing::Test {
     memcpy(rtp_packet_data, kPcmuFrame, rtp_len);
     // In order to be able to run this test function multiple times we can not
     // use the same sequence number twice. Increase the sequence number by one.
-    SetBE16(reinterpret_cast<uint8_t*>(rtp_packet_data) + 2,
+    SetBE16(ArrayView<uint8_t>(rtp_packet_buffer).subspan(2).data(),
             ++sequence_number_);
     CopyOnWriteBuffer rtp_packet1to2(rtp_packet_data, rtp_len, packet_size);
     CopyOnWriteBuffer rtp_packet2to1(rtp_packet_data, rtp_len, packet_size);
@@ -210,7 +211,7 @@ class DtlsSrtpTransportTest : public ::testing::Test {
     memcpy(rtp_packet_data, kPcmuFrameWithExtensions, rtp_len);
     // In order to be able to run this test function multiple times we can not
     // use the same sequence number twice. Increase the sequence number by one.
-    SetBE16(reinterpret_cast<uint8_t*>(rtp_packet_data) + 2,
+    SetBE16(ArrayView<uint8_t>(rtp_packet_buffer).subspan(2).data(),
             ++sequence_number_);
     CopyOnWriteBuffer rtp_packet1to2(rtp_packet_data, rtp_len, packet_size);
     CopyOnWriteBuffer rtp_packet2to1(rtp_packet_data, rtp_len, packet_size);
diff --git a/pc/ice_server_parsing.cc b/pc/ice_server_parsing.cc
index 6b357cce5c..b83cc86783 100644
--- a/pc/ice_server_parsing.cc
+++ b/pc/ice_server_parsing.cc
@@ -10,6 +10,7 @@
 
 #include "pc/ice_server_parsing.h"
 
+#include <array>
 #include <cctype>
 #include <cstddef>
 #include <optional>
@@ -52,7 +53,8 @@ const char kRegNameCharacters[] =
     "!$&'()*+,;=";  // sub-delims
 
 // NOTE: Must be in the same order as the ServiceType enum.
-const char* kValidIceServiceTypes[] = {"stun", "stuns", "turn", "turns"};
+constexpr std::array<absl::string_view, 4> kValidIceServiceTypes{
+    "stun", "stuns", "turn", "turns"};
 
 // NOTE: A loop below assumes that the first value of this enum is 0 and all
 // other values are incremental.
diff --git a/pc/jsep_transport_unittest.cc b/pc/jsep_transport_unittest.cc
index c585844f88..0ee7519343 100644
--- a/pc/jsep_transport_unittest.cc
+++ b/pc/jsep_transport_unittest.cc
@@ -20,6 +20,7 @@
 #include <utility>
 #include <vector>
 
+#include "api/array_view.h"
 #include "api/candidate.h"
 #include "api/field_trials.h"
 #include "api/ice_transport_interface.h"
@@ -1172,7 +1173,7 @@ class JsepTransport2HeaderExtensionTest
     memcpy(rtp_packet_data, kPcmuFrameWithExtensions, rtp_len);
     // In order to be able to run this test function multiple times we can not
     // use the same sequence number twice. Increase the sequence number by one.
-    SetBE16(reinterpret_cast<uint8_t*>(rtp_packet_data) + 2,
+    SetBE16(ArrayView<uint8_t>(rtp_packet_buffer).subspan(2).data(),
             ++sequence_number_);
     CopyOnWriteBuffer rtp_packet(rtp_packet_data, rtp_len, packet_size);
 
diff --git a/pc/peer_connection_histogram_unittest.cc b/pc/peer_connection_histogram_unittest.cc
index 89a323259e..53b034c1ff 100644
--- a/pc/peer_connection_histogram_unittest.cc
+++ b/pc/peer_connection_histogram_unittest.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <array>
 #include <memory>
 #include <optional>
 #include <set>
@@ -74,8 +75,8 @@ a=rtpmap:101 fake_audio_codec/8000
 
 constexpr char kUsagePatternMetric[] = "WebRTC.PeerConnection.UsagePattern";
 constexpr TimeDelta kDefaultTimeout = TimeDelta::Millis(10000);
-const SocketAddress kLocalAddrs[2] = {SocketAddress("1.1.1.1", 0),
-                                      SocketAddress("2.2.2.2", 0)};
+const std::array kLocalAddrs{SocketAddress("1.1.1.1", 0),
+                             SocketAddress("2.2.2.2", 0)};
 const SocketAddress kPrivateLocalAddress("10.1.1.1", 0);
 const SocketAddress kPrivateIpv6LocalAddress("fd12:3456:789a:1::1", 0);
 
diff --git a/pc/srtp_transport_unittest.cc b/pc/srtp_transport_unittest.cc
index eb89889e50..f9dbeb7875 100644
--- a/pc/srtp_transport_unittest.cc
+++ b/pc/srtp_transport_unittest.cc
@@ -17,6 +17,7 @@
 #include <utility>
 #include <vector>
 
+#include "api/array_view.h"
 #include "api/field_trials.h"
 #include "api/transport/ecn_marking.h"
 #include "api/units/timestamp.h"
@@ -141,7 +142,7 @@ class SrtpTransportTest : public ::testing::Test {
     memcpy(rtp_packet_data, kPcmuFrame, rtp_len);
     // In order to be able to run this test function multiple times we can not
     // use the same sequence number twice. Increase the sequence number by one.
-    SetBE16(reinterpret_cast<uint8_t*>(rtp_packet_data) + 2,
+    SetBE16(ArrayView<uint8_t>(rtp_packet_buffer).subspan(2).data(),
             ++sequence_number_);
     CopyOnWriteBuffer rtp_packet1to2(rtp_packet_data, rtp_len, packet_size);
     CopyOnWriteBuffer rtp_packet2to1(rtp_packet_data, rtp_len, packet_size);
@@ -263,7 +264,7 @@ class SrtpTransportTest : public ::testing::Test {
     memcpy(rtp_packet_data, kPcmuFrameWithExtensions, rtp_len);
     // In order to be able to run this test function multiple times we can not
     // use the same sequence number twice. Increase the sequence number by one.
-    SetBE16(reinterpret_cast<uint8_t*>(rtp_packet_data) + 2,
+    SetBE16(ArrayView<uint8_t>(rtp_packet_buffer).subspan(2).data(),
             ++sequence_number_);
     CopyOnWriteBuffer rtp_packet1to2(rtp_packet_data, rtp_len, packet_size);
     CopyOnWriteBuffer rtp_packet2to1(rtp_packet_data, rtp_len, packet_size);
diff --git a/pc/test/fake_audio_capture_module.cc b/pc/test/fake_audio_capture_module.cc
index e55a8af65e..29b57ef63d 100644
--- a/pc/test/fake_audio_capture_module.cc
+++ b/pc/test/fake_audio_capture_module.cc
@@ -403,24 +403,16 @@ bool FakeAudioCaptureModule::Initialize() {
 }
 
 void FakeAudioCaptureModule::SetSendBuffer(int value) {
-  Sample* buffer_ptr = reinterpret_cast<Sample*>(send_buffer_);
-  const size_t buffer_size_in_samples =
-      sizeof(send_buffer_) / kNumberBytesPerSample;
-  for (size_t i = 0; i < buffer_size_in_samples; ++i) {
-    buffer_ptr[i] = value;
-  }
+  send_buffer_.fill(value);
 }
 
 void FakeAudioCaptureModule::ResetRecBuffer() {
-  memset(rec_buffer_, 0, sizeof(rec_buffer_));
+  rec_buffer_.fill(0);
 }
 
 bool FakeAudioCaptureModule::CheckRecBuffer(int value) {
-  const Sample* buffer_ptr = reinterpret_cast<const Sample*>(rec_buffer_);
-  const size_t buffer_size_in_samples =
-      sizeof(rec_buffer_) / kNumberBytesPerSample;
-  for (size_t i = 0; i < buffer_size_in_samples; ++i) {
-    if (buffer_ptr[i] >= value)
+  for (Sample sample : rec_buffer_) {
+    if (sample >= value)
       return true;
   }
   return false;
@@ -498,7 +490,7 @@ void FakeAudioCaptureModule::ReceiveFrameP() {
   int64_t ntp_time_ms = 0;
   if (audio_callback_->NeedMorePlayData(kNumberSamples, kNumberBytesPerSample,
                                         kNumberOfChannels, kSamplesPerSecond,
-                                        rec_buffer_, nSamplesOut,
+                                        rec_buffer_.data(), nSamplesOut,
                                         &elapsed_time_ms, &ntp_time_ms) != 0) {
     RTC_DCHECK_NOTREACHED();
   }
@@ -523,7 +515,7 @@ void FakeAudioCaptureModule::SendFrameP() {
   bool key_pressed = false;
   uint32_t current_mic_level = current_mic_level_;
   if (audio_callback_->RecordedDataIsAvailable(
-          send_buffer_, kNumberSamples, kNumberBytesPerSample,
+          send_buffer_.data(), kNumberSamples, kNumberBytesPerSample,
           kNumberOfChannels, kSamplesPerSecond, kTotalDelayMs, kClockDriftMs,
           current_mic_level, key_pressed, current_mic_level) != 0) {
     RTC_DCHECK_NOTREACHED();
diff --git a/pc/test/fake_audio_capture_module.h b/pc/test/fake_audio_capture_module.h
index 06a71c937e..0e5578d93d 100644
--- a/pc/test/fake_audio_capture_module.h
+++ b/pc/test/fake_audio_capture_module.h
@@ -23,6 +23,7 @@
 #include <stddef.h>
 #include <stdint.h>
 
+#include <array>
 #include <memory>
 #include <optional>
 
@@ -215,9 +216,9 @@ class FakeAudioCaptureModule : public webrtc::AudioDeviceModule {
   std::unique_ptr<webrtc::Thread> process_thread_;
 
   // Buffer for storing samples received from the webrtc::AudioTransport.
-  char rec_buffer_[kNumberSamples * kNumberBytesPerSample];
+  std::array<Sample, kNumberSamples> rec_buffer_;
   // Buffer for samples to send to the webrtc::AudioTransport.
-  char send_buffer_[kNumberSamples * kNumberBytesPerSample];
+  std::array<Sample, kNumberSamples> send_buffer_;
 
   // Counter of frames received that have samples of high enough amplitude to
   // indicate that the frames are not faked somewhere in the audio pipeline
diff --git a/pc/test/fake_rtc_certificate_generator.h b/pc/test/fake_rtc_certificate_generator.h
index 2253366e49..c17037900a 100644
--- a/pc/test/fake_rtc_certificate_generator.h
+++ b/pc/test/fake_rtc_certificate_generator.h
@@ -11,6 +11,7 @@
 #ifndef PC_TEST_FAKE_RTC_CERTIFICATE_GENERATOR_H_
 #define PC_TEST_FAKE_RTC_CERTIFICATE_GENERATOR_H_
 
+#include <array>
 #include <cstdint>
 #include <optional>
 #include <string>
@@ -25,9 +26,11 @@
 #include "rtc_base/rtc_certificate_generator.h"
 #include "rtc_base/ssl_identity.h"
 
+namespace webrtc {
+
 // RSA with mod size 1024, pub exp 0x10001.
-static const webrtc::RTCCertificatePEM kRsaPems[] = {
-    webrtc::RTCCertificatePEM(
+static const std::array kRsaPems{
+    RTCCertificatePEM(
         "-----BEGIN RSA PRI"  // Linebreak to avoid detection of private
         "VATE KEY-----\n"     // keys by linters.
         "MIICdwIBADANBgkqhkiG9w0BAQEFAASCAmEwggJdAgEAAoGBAMYRkbhmI7kVA/rM\n"
@@ -56,7 +59,7 @@ static const webrtc::RTCCertificatePEM kRsaPems[] = {
         "LJE/mGw3MyFHEqi81jh95J+ypl6xKW6Rm8jKLR87gUvCaVYn/Z4/P3AqcQTB7wOv\n"
         "UD0A8qfhfDM+LK6rPAnCsVN0NRDY3jvd6rzix9M=\n"
         "-----END CERTIFICATE-----\n"),
-    webrtc::RTCCertificatePEM(
+    RTCCertificatePEM(
         "-----BEGIN RSA PRI"  // Linebreak to avoid detection of private
         "VATE KEY-----\n"     // keys by linters.
         "MIICXQIBAAKBgQDeYqlyJ1wuiMsi905e3X81/WA/G3ym50PIDZBVtSwZi7JVQPgj\n"
@@ -94,8 +97,8 @@ static const webrtc::RTCCertificatePEM kRsaPems[] = {
 // `SSLIdentity::Create` and invoking `identity->PrivateKeyToPEMString()`,
 // `identity->PublicKeyToPEMString()` and
 // `identity->certificate().ToPEMString()`.
-static const webrtc::RTCCertificatePEM kEcdsaPems[] = {
-    webrtc::RTCCertificatePEM(
+static const std::array kEcdsaPems{
+    RTCCertificatePEM(
         "-----BEGIN PRI"   // Linebreak to avoid detection of private
         "VATE KEY-----\n"  // keys by linters.
         "MIGHAgEAMBMGByqGSM49AgEGCCqGSM49AwEHBG0wawIBAQQg+qaRsR5uHtqG689M\n"
@@ -110,7 +113,7 @@ static const webrtc::RTCCertificatePEM kEcdsaPems[] = {
         "vK0wCgYIKoZIzj0EAwIDSQAwRgIhAIIc3+CqfkZ9lLwTj1PvUtt3KhnqF2kD0War\n"
         "cCoTBbCxAiEAyp9Cn4vo2ZBhRIVDKyoxmwak8Z0PAVhJAQaWCgoY2D4=\n"
         "-----END CERTIFICATE-----\n"),
-    webrtc::RTCCertificatePEM(
+    RTCCertificatePEM(
         "-----BEGIN PRI"   // Linebreak to avoid detection of private
         "VATE KEY-----\n"  // keys by linters.
         "MIGHAgEAMBMGByqGSM49AgEGCCqGSM49AwEHBG0wawIBAQQghL/G4JRYnuDNbQuh\n"
@@ -126,8 +129,7 @@ static const webrtc::RTCCertificatePEM kEcdsaPems[] = {
         "1fE/g68CIQD7uoFfbiq6dTp8ZwzbwQ8jJf08KjriamqA9OW/4268Dw==\n"
         "-----END CERTIFICATE-----\n")};
 
-class FakeRTCCertificateGenerator
-    : public webrtc::RTCCertificateGeneratorInterface {
+class FakeRTCCertificateGenerator : public RTCCertificateGeneratorInterface {
  public:
   FakeRTCCertificateGenerator() : should_fail_(false), should_wait_(false) {}
 
@@ -143,7 +145,7 @@ class FakeRTCCertificateGenerator
   int generated_certificates() { return generated_certificates_; }
   int generated_failures() { return generated_failures_; }
 
-  void GenerateCertificateAsync(const webrtc::KeyParams& key_params,
+  void GenerateCertificateAsync(const KeyParams& key_params,
                                 const std::optional<uint64_t>& expires_ms,
                                 Callback callback) override {
     // The certificates are created from constant PEM strings and use its coded
@@ -151,27 +153,27 @@ class FakeRTCCertificateGenerator
     RTC_DCHECK(!expires_ms);
 
     // Only supports RSA-1024-0x10001 and ECDSA-P256.
-    if (key_params.type() == webrtc::KT_RSA) {
+    if (key_params.type() == KT_RSA) {
       RTC_DCHECK_EQ(key_params.rsa_params().mod_size, 1024);
       RTC_DCHECK_EQ(key_params.rsa_params().pub_exp, 0x10001);
     } else {
-      RTC_DCHECK_EQ(key_params.type(), webrtc::KT_ECDSA);
-      RTC_DCHECK_EQ(key_params.ec_curve(), webrtc::EC_NIST_P256);
+      RTC_DCHECK_EQ(key_params.type(), KT_ECDSA);
+      RTC_DCHECK_EQ(key_params.ec_curve(), EC_NIST_P256);
     }
-    webrtc::KeyType key_type = key_params.type();
-    webrtc::TaskQueueBase::Current()->PostTask(webrtc::SafeTask(
-        pending_delete_.flag(),
-        [this, key_type, callback = std::move(callback)]() mutable {
-          GenerateCertificate(key_type, std::move(callback));
-        }));
+    KeyType key_type = key_params.type();
+    TaskQueueBase::Current()->PostTask(
+        SafeTask(pending_delete_.flag(),
+                 [this, key_type, callback = std::move(callback)]() mutable {
+                   GenerateCertificate(key_type, std::move(callback));
+                 }));
   }
 
-  static webrtc::scoped_refptr<webrtc::RTCCertificate> GenerateCertificate() {
-    switch (webrtc::KT_DEFAULT) {
-      case webrtc::KT_RSA:
-        return webrtc::RTCCertificate::FromPEM(kRsaPems[0]);
-      case webrtc::KT_ECDSA:
-        return webrtc::RTCCertificate::FromPEM(kEcdsaPems[0]);
+  static scoped_refptr<RTCCertificate> GenerateCertificate() {
+    switch (KT_DEFAULT) {
+      case KT_RSA:
+        return RTCCertificate::FromPEM(kRsaPems[0]);
+      case KT_ECDSA:
+        return RTCCertificate::FromPEM(kEcdsaPems[0]);
       default:
         RTC_DCHECK_NOTREACHED();
         return nullptr;
@@ -179,52 +181,50 @@ class FakeRTCCertificateGenerator
   }
 
  private:
-  const webrtc::RTCCertificatePEM& get_pem(
-      const webrtc::KeyType& key_type) const {
+  const RTCCertificatePEM& get_pem(const KeyType& key_type) const {
     switch (key_type) {
-      case webrtc::KT_RSA:
+      case KT_RSA:
         return kRsaPems[key_index_];
-      case webrtc::KT_ECDSA:
+      case KT_ECDSA:
         return kEcdsaPems[key_index_];
       default:
         RTC_DCHECK_NOTREACHED();
         return kEcdsaPems[key_index_];
     }
   }
-  const std::string& get_key(const webrtc::KeyType& key_type) const {
+  const std::string& get_key(const KeyType& key_type) const {
     return get_pem(key_type).private_key();
   }
-  const std::string& get_cert(const webrtc::KeyType& key_type) const {
+  const std::string& get_cert(const KeyType& key_type) const {
     return get_pem(key_type).certificate();
   }
 
-  void GenerateCertificate(webrtc::KeyType key_type, Callback callback) {
+  void GenerateCertificate(KeyType key_type, Callback callback) {
     // If the certificate generation should be stalled, re-post this same
     // message to the queue with a small delay so as to wait in a loop until
     // set_should_wait(false) is called.
     if (should_wait_) {
-      webrtc::TaskQueueBase::Current()->PostDelayedTask(
-          webrtc::SafeTask(
-              pending_delete_.flag(),
-              [this, key_type, callback = std::move(callback)]() mutable {
-                GenerateCertificate(key_type, std::move(callback));
-              }),
-          webrtc::TimeDelta::Millis(1));
+      TaskQueueBase::Current()->PostDelayedTask(
+          SafeTask(pending_delete_.flag(),
+                   [this, key_type, callback = std::move(callback)]() mutable {
+                     GenerateCertificate(key_type, std::move(callback));
+                   }),
+          TimeDelta::Millis(1));
       return;
     }
     if (should_fail_) {
       ++generated_failures_;
       std::move(callback)(nullptr);
     } else {
-      webrtc::scoped_refptr<webrtc::RTCCertificate> certificate =
-          webrtc::RTCCertificate::FromPEM(get_pem(key_type));
+      scoped_refptr<RTCCertificate> certificate =
+          RTCCertificate::FromPEM(get_pem(key_type));
       RTC_DCHECK(certificate);
       ++generated_certificates_;
       std::move(callback)(std::move(certificate));
     }
   }
 
-  webrtc::ScopedTaskSafetyDetached pending_delete_;
+  ScopedTaskSafetyDetached pending_delete_;
 
   bool should_fail_;
   bool should_wait_;
@@ -233,4 +233,5 @@ class FakeRTCCertificateGenerator
   int generated_failures_ = 0;
 };
 
+}  // namespace webrtc
 #endif  // PC_TEST_FAKE_RTC_CERTIFICATE_GENERATOR_H_
diff --git a/unsafe_buffers_paths.txt b/unsafe_buffers_paths.txt
index 3e10db04fd..579df0fe36 100644
--- a/unsafe_buffers_paths.txt
+++ b/unsafe_buffers_paths.txt
@@ -587,17 +587,10 @@
 -p2p/test/nat_unittest.cc
 -p2p/test/turn_server.cc
 -pc/channel_unittest.cc
--pc/dtls_srtp_transport_unittest.cc
--pc/ice_server_parsing.cc
--pc/jsep_transport_unittest.cc
 -pc/media_session_unittest.cc
--pc/peer_connection_histogram_unittest.cc
 -pc/peer_connection_interface_unittest.cc
 -pc/rtc_stats_collector_unittest.cc
 -pc/srtp_session_unittest.cc
--pc/srtp_transport_unittest.cc
--pc/test/fake_audio_capture_module.cc
--pc/test/fake_rtc_certificate_generator.h
 -rtc_base/async_tcp_socket.cc
 -rtc_base/bit_buffer.cc
 -rtc_base/bitstream_reader.cc

From 60c186265f075bb792996200a4752354e438b37f Mon Sep 17 00:00:00 2001
From: Mirko Bonadei <mbonadei@webrtc.org>
Date: Fri, 13 Feb 2026 15:19:48 +0000
Subject: [PATCH 048/472] Fork Chromium's TestStatusReporter to avoid a
 dependency on //base.

This CL forks org.chromium.test.reporter.TestStatusReporter to prevent
WebRTC from taking a dependency on Chromium's //base. Since [1],
Chromium's TestStatusReporter started to depend on //base for logging.

[1] - https://chromium-review.googlesource.com/c/chromium/src/+/7555110

Bug: None
Change-Id: I921b6cc90ed1c1101724fa120e77eb103ff793ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/448902
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46901}
---
 test/BUILD.gn                                 |  2 +-
 .../webrtc/native_test/NativeTestWebrtc.java  |  2 +-
 .../native_test/TestStatusReporter.java       | 52 +++++++++++++++++++
 3 files changed, 54 insertions(+), 2 deletions(-)
 create mode 100644 test/android/org/webrtc/native_test/TestStatusReporter.java

diff --git a/test/BUILD.gn b/test/BUILD.gn
index cb8ae51843..13e8b6da96 100644
--- a/test/BUILD.gn
+++ b/test/BUILD.gn
@@ -1400,6 +1400,7 @@ if (!build_with_chromium && is_android) {
       "android/org/webrtc/native_test/RTCNativeUnitTest.java",
       "android/org/webrtc/native_test/RTCNativeUnitTestActivity.java",
       "android/org/webrtc/native_test/StrictModeContext.java",
+      "android/org/webrtc/native_test/TestStatusReporter.java",
     ]
     srcjar_deps = [ ":native_test_jni" ]
     deps = [
@@ -1407,7 +1408,6 @@ if (!build_with_chromium && is_android) {
       "../rtc_base:base_java",
       "//build/android:build_java",
       "//build/android/gtest_apk:native_test_instrumentation_test_runner_java",
-      "//testing/android/reporter:reporter_java",
       "//third_party/jni_zero:jni_zero_java",
     ]
   }
diff --git a/test/android/org/webrtc/native_test/NativeTestWebrtc.java b/test/android/org/webrtc/native_test/NativeTestWebrtc.java
index 2367c277e4..60cb538384 100644
--- a/test/android/org/webrtc/native_test/NativeTestWebrtc.java
+++ b/test/android/org/webrtc/native_test/NativeTestWebrtc.java
@@ -28,8 +28,8 @@ import org.jni_zero.JniType;
 import org.jni_zero.NativeMethods;
 
 import org.webrtc.native_test.StrictModeContext;
+import org.webrtc.native_test.TestStatusReporter;
 import org.chromium.build.gtest_apk.NativeTestIntent;
-import org.chromium.test.reporter.TestStatusReporter;
 
 import java.io.File;
 
diff --git a/test/android/org/webrtc/native_test/TestStatusReporter.java b/test/android/org/webrtc/native_test/TestStatusReporter.java
new file mode 100644
index 0000000000..ed1e3ff23c
--- /dev/null
+++ b/test/android/org/webrtc/native_test/TestStatusReporter.java
@@ -0,0 +1,52 @@
+/*
+ *  Copyright 2015 The Chromium Authors.
+ *  Copyright 2024 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc.native_test;
+
+import android.content.Context;
+import android.content.Intent;
+import android.util.Log;
+
+import org.chromium.build.gtest_apk.TestStatusIntent;
+
+/**
+ * Broadcasts test status to any listening {@link org.chromium.test.reporter.TestStatusReceiver}.
+ */
+public class TestStatusReporter {
+    private final Context mContext;
+
+    public TestStatusReporter(Context c) {
+        mContext = c;
+    }
+
+    public void testRunStarted(int pid) {
+        sendTestRunBroadcast(TestStatusIntent.ACTION_TEST_RUN_STARTED, pid);
+    }
+
+    public void testRunFinished(int pid) {
+        sendTestRunBroadcast(TestStatusIntent.ACTION_TEST_RUN_FINISHED, pid);
+    }
+
+    private void sendTestRunBroadcast(String action, int pid) {
+        Intent i = new Intent(action);
+        i.setType(TestStatusIntent.DATA_TYPE_RESULT);
+        i.putExtra(TestStatusIntent.EXTRA_PID, pid);
+        mContext.sendBroadcast(i);
+    }
+
+    public void uncaughtException(int pid, Throwable ex) {
+        Intent i = new Intent(TestStatusIntent.ACTION_UNCAUGHT_EXCEPTION);
+        i.setType(TestStatusIntent.DATA_TYPE_RESULT);
+        i.putExtra(TestStatusIntent.EXTRA_PID, pid);
+        i.putExtra(TestStatusIntent.EXTRA_STACK_TRACE, Log.getStackTraceString(ex));
+        mContext.sendBroadcast(i);
+    }
+}

From a3c1ddd476b0b4bcdbab03a942da129d674a346c Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Fri, 13 Feb 2026 13:30:45 +0100
Subject: [PATCH 049/472] Migrate from ArrayView::subview to subspan in audio
 code

subspan match behavior of the std::span::subspan
In particular it requires input parameters to produce valid range

All callers already ensure offset and size are within valid range.

Bug: webrtc:439801349
Change-Id: If03b148e559872fa16b0cd591bf100fe0f4193c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449000
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46902}
---
 modules/audio_coding/neteq/audio_vector.cc    |  4 ++--
 .../audio_device/include/test_audio_device.cc |  4 ++--
 modules/audio_processing/agc2/limiter.cc      |  2 +-
 .../agc2/rnn_vad/features_extraction.cc       |  3 ++-
 .../agc2/rnn_vad/pitch_search_internal.cc     |  8 +++----
 .../audio_processing/agc2/rnn_vad/rnn_fc.cc   |  2 +-
 .../audio_processing/agc2/rnn_vad/rnn_gru.cc  | 24 +++++++++----------
 .../agc2/rnn_vad/rnn_gru_unittest.cc          |  6 ++---
 .../audio_processing/rms_level_unittest.cc    |  4 ++--
 9 files changed, 29 insertions(+), 28 deletions(-)

diff --git a/modules/audio_coding/neteq/audio_vector.cc b/modules/audio_coding/neteq/audio_vector.cc
index 096c8c483f..088578c6bc 100644
--- a/modules/audio_coding/neteq/audio_vector.cc
+++ b/modules/audio_coding/neteq/audio_vector.cc
@@ -70,12 +70,12 @@ bool AudioVector::CopyTo(size_t position, MonoView<int16_t> dst) const {
   const size_t copy_index = (begin_index_ + position) % capacity_;
   const size_t first_chunk_length =
       std::min(dst.size(), capacity_ - copy_index);
-  auto first_chunk = dst.subview(0, first_chunk_length);
+  auto first_chunk = dst.subspan(0, first_chunk_length);
   CopySamples(first_chunk,
               MonoView<const int16_t>(&array_[copy_index], first_chunk_length));
   const size_t remaining_length = dst.size() - first_chunk_length;
   if (remaining_length > 0) {
-    auto second_chunk = dst.subview(first_chunk_length, remaining_length);
+    auto second_chunk = dst.subspan(first_chunk_length);
     CopySamples(second_chunk, MonoView<int16_t>(&array_[0], remaining_length));
   }
   return true;
diff --git a/modules/audio_device/include/test_audio_device.cc b/modules/audio_device/include/test_audio_device.cc
index 9e42c16490..328874fe55 100644
--- a/modules/audio_device/include/test_audio_device.cc
+++ b/modules/audio_device/include/test_audio_device.cc
@@ -126,7 +126,7 @@ class WavFileReader final : public TestAudioDeviceModule::Capturer {
             do {
               wav_reader_->Reset();
               size_t delta = wav_reader_->ReadSamples(
-                  data.size() - read, data.subview(read).data());
+                  data.size() - read, data.subspan(read).data());
               RTC_CHECK_GT(delta, 0) << "No new data read from file";
               read += delta;
             } while (read < data.size());
@@ -310,7 +310,7 @@ class RawFileReader final : public TestAudioDeviceModule::Capturer {
             do {
               input_file_.Rewind();
               size_t delta = input_file_.Read(
-                  read_buffer_view.subview(read).data(), size - read);
+                  read_buffer_view.subspan(read).data(), size - read);
               RTC_CHECK_GT(delta, 0) << "No new data to read from file";
               read += delta;
             } while (read < size);
diff --git a/modules/audio_processing/agc2/limiter.cc b/modules/audio_processing/agc2/limiter.cc
index ca3e359da8..faa252c2ae 100644
--- a/modules/audio_processing/agc2/limiter.cc
+++ b/modules/audio_processing/agc2/limiter.cc
@@ -60,7 +60,7 @@ void ComputePerSampleSubframeFactors(
   if (is_attack) {
     InterpolateFirstSubframe(
         scaling_factors[0], scaling_factors[1],
-        per_sample_scaling_factors.subview(0, subframe_size));
+        per_sample_scaling_factors.subspan(0, subframe_size));
   }
 
   for (size_t i = is_attack ? 1 : 0; i < num_subframes; ++i) {
diff --git a/modules/audio_processing/agc2/rnn_vad/features_extraction.cc b/modules/audio_processing/agc2/rnn_vad/features_extraction.cc
index 4efb8fe372..bc170f99dc 100644
--- a/modules/audio_processing/agc2/rnn_vad/features_extraction.cc
+++ b/modules/audio_processing/agc2/rnn_vad/features_extraction.cc
@@ -11,6 +11,7 @@
 #include "modules/audio_processing/agc2/rnn_vad/features_extraction.h"
 
 #include <array>
+#include <cstddef>
 
 #include "api/array_view.h"
 #include "modules/audio_processing/agc2/biquad_filter.h"
@@ -76,7 +77,7 @@ bool FeaturesExtractor::CheckSilenceComputeFeatures(
   feature_vector[kFeatureVectorSize - 2] = 0.01f * (pitch_period_48kHz_ - 300);
   // Extract lagged frames (according to the estimated pitch period).
   RTC_DCHECK_LE(pitch_period_48kHz_ / 2, kMaxPitch24kHz);
-  auto lagged_frame = pitch_buf_24kHz_view_.subview(
+  auto lagged_frame = pitch_buf_24kHz_view_.subspan(
       kMaxPitch24kHz - pitch_period_48kHz_ / 2, kFrameSize20ms24kHz);
   // Analyze reference and lagged frames checking if silence has been detected
   // and write the feature vector.
diff --git a/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc b/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc
index b7eb7f4a4e..019aedca17 100644
--- a/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc
+++ b/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc
@@ -33,8 +33,8 @@ float ComputeAutoCorrelation(int inverted_lag,
   RTC_DCHECK_LT(inverted_lag, kRefineNumLags24kHz);
   static_assert(kMaxPitch24kHz < kBufSize24kHz, "");
   return vector_math.DotProduct(
-      pitch_buffer.subview(/*offset=*/kMaxPitch24kHz),
-      pitch_buffer.subview(inverted_lag, kFrameSize20ms24kHz));
+      pitch_buffer.subspan(/*offset=*/kMaxPitch24kHz),
+      pitch_buffer.subspan(inverted_lag, kFrameSize20ms24kHz));
 }
 
 // Given an auto-correlation coefficient `curr_auto_correlation` and its
@@ -295,7 +295,7 @@ void ComputeSlidingFrameSquareEnergies24kHz(
     AvailableCpuFeatures cpu_features) {
   VectorMath vector_math(cpu_features);
   static_assert(kFrameSize20ms24kHz < kBufSize24kHz, "");
-  const auto frame_20ms_view = pitch_buffer.subview(0, kFrameSize20ms24kHz);
+  const auto frame_20ms_view = pitch_buffer.subspan(0, kFrameSize20ms24kHz);
   float yy = vector_math.DotProduct(frame_20ms_view, frame_20ms_view);
   y_energy[0] = yy;
   static_assert(kMaxPitch24kHz - 1 + kFrameSize20ms24kHz < kBufSize24kHz, "");
@@ -333,7 +333,7 @@ CandidatePitchPeriods ComputePitchPeriod12kHz(
 
   VectorMath vector_math(cpu_features);
   static_assert(kFrameSize20ms12kHz + 1 < kBufSize12kHz, "");
-  const auto frame_view = pitch_buffer.subview(0, kFrameSize20ms12kHz + 1);
+  const auto frame_view = pitch_buffer.subspan(0, kFrameSize20ms12kHz + 1);
   float denominator = 1.f + vector_math.DotProduct(frame_view, frame_view);
   // Search best and second best pitches by looking at the scaled
   // auto-correlation.
diff --git a/modules/audio_processing/agc2/rnn_vad/rnn_fc.cc b/modules/audio_processing/agc2/rnn_vad/rnn_fc.cc
index 3850a256fd..0c4b7bbe4f 100644
--- a/modules/audio_processing/agc2/rnn_vad/rnn_fc.cc
+++ b/modules/audio_processing/agc2/rnn_vad/rnn_fc.cc
@@ -101,7 +101,7 @@ void FullyConnectedLayer::ComputeOutput(ArrayView<const float> input) {
   for (int o = 0; o < output_size_; ++o) {
     output_[o] = activation_function_(
         bias_[o] + vector_math_.DotProduct(
-                       input, weights.subview(o * input_size_, input_size_)));
+                       input, weights.subspan(o * input_size_, input_size_)));
   }
 }
 
diff --git a/modules/audio_processing/agc2/rnn_vad/rnn_gru.cc b/modules/audio_processing/agc2/rnn_vad/rnn_gru.cc
index d679b614a8..5bd7119332 100644
--- a/modules/audio_processing/agc2/rnn_vad/rnn_gru.cc
+++ b/modules/audio_processing/agc2/rnn_vad/rnn_gru.cc
@@ -79,9 +79,9 @@ void ComputeUpdateResetGate(int input_size,
   for (int o = 0; o < output_size; ++o) {
     float x = bias[o];
     x += vector_math.DotProduct(input,
-                                weights.subview(o * input_size, input_size));
+                                weights.subspan(o * input_size, input_size));
     x += vector_math.DotProduct(
-        state, recurrent_weights.subview(o * output_size, output_size));
+        state, recurrent_weights.subspan(o * output_size, output_size));
     gate[o] = ::rnnoise::SigmoidApproximated(x);
   }
 }
@@ -121,10 +121,10 @@ void ComputeStateGate(int input_size,
   for (int o = 0; o < output_size; ++o) {
     float x = bias[o];
     x += vector_math.DotProduct(input,
-                                weights.subview(o * input_size, input_size));
+                                weights.subspan(o * input_size, input_size));
     x += vector_math.DotProduct(
         {reset_x_state.data(), static_cast<size_t>(output_size)},
-        recurrent_weights.subview(o * output_size, output_size));
+        recurrent_weights.subspan(o * output_size, output_size));
     state[o] = update[o] * state[o] + (1.f - update[o]) * std::max(0.f, x);
   }
 }
@@ -185,21 +185,21 @@ void GatedRecurrentLayer::ComputeOutput(ArrayView<const float> input) {
   std::array<float, kGruLayerMaxUnits> update;
   ComputeUpdateResetGate(
       input_size_, output_size_, vector_math_, input, state,
-      bias.subview(0, output_size_), weights.subview(0, stride_weights),
-      recurrent_weights.subview(0, stride_recurrent_weights), update);
+      bias.subspan(0, output_size_), weights.subspan(0, stride_weights),
+      recurrent_weights.subspan(0, stride_recurrent_weights), update);
   // Reset gate.
   std::array<float, kGruLayerMaxUnits> reset;
   ComputeUpdateResetGate(input_size_, output_size_, vector_math_, input, state,
-                         bias.subview(output_size_, output_size_),
-                         weights.subview(stride_weights, stride_weights),
-                         recurrent_weights.subview(stride_recurrent_weights,
+                         bias.subspan(output_size_, output_size_),
+                         weights.subspan(stride_weights, stride_weights),
+                         recurrent_weights.subspan(stride_recurrent_weights,
                                                    stride_recurrent_weights),
                          reset);
   // State gate.
   ComputeStateGate(input_size_, output_size_, vector_math_, input, update,
-                   reset, bias.subview(2 * output_size_, output_size_),
-                   weights.subview(2 * stride_weights, stride_weights),
-                   recurrent_weights.subview(2 * stride_recurrent_weights,
+                   reset, bias.subspan(2 * output_size_, output_size_),
+                   weights.subspan(2 * stride_weights, stride_weights),
+                   recurrent_weights.subspan(2 * stride_recurrent_weights,
                                              stride_recurrent_weights),
                    state);
 }
diff --git a/modules/audio_processing/agc2/rnn_vad/rnn_gru_unittest.cc b/modules/audio_processing/agc2/rnn_vad/rnn_gru_unittest.cc
index d49a116354..f6009e26d5 100644
--- a/modules/audio_processing/agc2/rnn_vad/rnn_gru_unittest.cc
+++ b/modules/audio_processing/agc2/rnn_vad/rnn_gru_unittest.cc
@@ -44,9 +44,9 @@ void TestGatedRecurrentLayer(GatedRecurrentLayer& gru,
   for (int i = 0; i < input_sequence_length; ++i) {
     SCOPED_TRACE(i);
     gru.ComputeOutput(
-        input_sequence.subview(i * gru.input_size(), gru.input_size()));
+        input_sequence.subspan(i * gru.input_size(), gru.input_size()));
     const auto expected_output =
-        expected_output_sequence.subview(i * gru.size(), gru.size());
+        expected_output_sequence.subspan(i * gru.size(), gru.size());
     ExpectNearAbsolute(expected_output, gru, 3e-6f);
   }
 }
@@ -150,7 +150,7 @@ TEST_P(RnnGruParametrization, DISABLED_BenchmarkGatedRecurrentLayer) {
     perf_timer.StartTimer();
     for (int i = 0; i < input_sequence_length; ++i) {
       gru.ComputeOutput(
-          input_sequence.subview(i * gru.input_size(), gru.input_size()));
+          input_sequence.subspan(i * gru.input_size(), gru.input_size()));
     }
     perf_timer.StopTimer();
   }
diff --git a/modules/audio_processing/rms_level_unittest.cc b/modules/audio_processing/rms_level_unittest.cc
index 9c95652705..ed9372dbb5 100644
--- a/modules/audio_processing/rms_level_unittest.cc
+++ b/modules/audio_processing/rms_level_unittest.cc
@@ -30,7 +30,7 @@ std::unique_ptr<RmsLevel> RunTest(ArrayView<const int16_t> input) {
   std::unique_ptr<RmsLevel> level(new RmsLevel);
   for (size_t n = 0; n + kBlockSizeSamples <= input.size();
        n += kBlockSizeSamples) {
-    level->Analyze(input.subview(n, kBlockSizeSamples));
+    level->Analyze(input.subspan(n, kBlockSizeSamples));
   }
   return level;
 }
@@ -39,7 +39,7 @@ std::unique_ptr<RmsLevel> RunTest(ArrayView<const float> input) {
   std::unique_ptr<RmsLevel> level(new RmsLevel);
   for (size_t n = 0; n + kBlockSizeSamples <= input.size();
        n += kBlockSizeSamples) {
-    level->Analyze(input.subview(n, kBlockSizeSamples));
+    level->Analyze(input.subspan(n, kBlockSizeSamples));
   }
   return level;
 }

From cc7094359ec1f23f234cf99324a41fa73476b524 Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Fri, 13 Feb 2026 21:38:53 +0000
Subject: [PATCH 050/472] Migrate internal usages of ByteOrder to ArrayView API

This change replaces internal usages of the legacy void* based ByteOrder
API with the new type-safe ArrayView overloads. All 19 identified
internal files have been updated.

Additionally:
- Updated rtc_base/byte_order.h to deprecate the legacy void* overloads
  using ABSL_DEPRECATE_AND_INLINE.
- Updated rtc_base/BUILD.gn to include necessary dependencies for the
  new ByteOrder API.
- Added documentation for the migration plan in g3doc/todo/byte_order_safety.md.

Bug: webrtc:484070270
Change-Id: I3836ff3a7d128d40e24dac1ec1cac2f289aaa6cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449061
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46903}
---
 api/transport/stun.cc                         |  47 ++++---
 api/transport/stun_unittest.cc                |  16 ++-
 g3doc/todo/byte_order_safety.md               |  35 +++++
 media/BUILD.gn                                |   1 +
 media/base/fake_network_interface.h           |   5 +-
 media/base/rtp_utils.cc                       |  14 +-
 media/base/turn_utils.cc                      |  12 +-
 media/engine/webrtc_voice_engine_unittest.cc  |  24 ++--
 .../codecs/red/audio_encoder_copy_red.cc      |   2 +-
 p2p/base/async_stun_tcp_socket.cc             |   8 +-
 p2p/base/turn_port.cc                         |   9 +-
 p2p/dtls/dtls_transport_unittest.cc           |  19 ++-
 p2p/test/turn_server.cc                       |  13 +-
 pc/channel_unittest.cc                        |   7 +-
 pc/dtls_srtp_transport_unittest.cc            |   4 +-
 pc/jsep_transport_unittest.cc                 |   2 +-
 pc/srtp_session_unittest.cc                   |  39 ++++--
 pc/srtp_transport_unittest.cc                 |   4 +-
 rtc_base/BUILD.gn                             |   7 +-
 rtc_base/async_tcp_socket.cc                  |   7 +-
 rtc_base/byte_buffer_unittest.cc              |  30 +++--
 rtc_base/byte_order.h                         | 124 ++++++++++++++++--
 rtc_base/byte_order_unittest.cc               |  35 ++---
 23 files changed, 341 insertions(+), 123 deletions(-)
 create mode 100644 g3doc/todo/byte_order_safety.md

diff --git a/api/transport/stun.cc b/api/transport/stun.cc
index 100b6c7e9b..6b9ca04d11 100644
--- a/api/transport/stun.cc
+++ b/api/transport/stun.cc
@@ -379,8 +379,11 @@ bool StunMessage::ValidateMessageIntegrityOfType(int mi_attr_type,
     return false;
   }
 
+  ArrayView<const uint8_t> data_view(reinterpret_cast<const uint8_t*>(data),
+                                     size);
+
   // Getting the message length from the STUN header.
-  uint16_t msg_length = GetBE16(&data[2]);
+  uint16_t msg_length = GetBE16(data_view.subspan(2, 2));
   if (size != (msg_length + kStunHeaderSize)) {
     return false;
   }
@@ -391,8 +394,9 @@ bool StunMessage::ValidateMessageIntegrityOfType(int mi_attr_type,
   while (current_pos + 4 <= size) {
     uint16_t attr_type, attr_length;
     // Getting attribute type and length.
-    attr_type = GetBE16(&data[current_pos]);
-    attr_length = GetBE16(&data[current_pos + sizeof(attr_type)]);
+    attr_type = GetBE16(data_view.subspan(current_pos, 2));
+    attr_length =
+        GetBE16(data_view.subspan(current_pos + sizeof(attr_type), 2));
 
     // If M-I, sanity check it, and break out.
     if (attr_type == mi_attr_type) {
@@ -433,7 +437,9 @@ bool StunMessage::ValidateMessageIntegrityOfType(int mi_attr_type,
     //     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     //     |0 0|     STUN Message Type     |         Message Length        |
     //     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-    SetBE16(temp_data.get() + 2, static_cast<uint16_t>(new_adjusted_len));
+    SetBE16(
+        ArrayView<uint8_t>(reinterpret_cast<uint8_t*>(temp_data.get() + 2), 2),
+        static_cast<uint16_t>(new_adjusted_len));
   }
 
   char hmac[kStunMessageIntegritySize];
@@ -504,22 +510,26 @@ bool StunMessage::ValidateFingerprint(const char* data, size_t size) {
   if (size % 4 != 0 || size < kStunHeaderSize + fingerprint_attr_size)
     return false;
 
+  ArrayView<const uint8_t> data_view(reinterpret_cast<const uint8_t*>(data),
+                                     size);
+
   // Skip the rest if the magic cookie isn't present.
-  const char* magic_cookie =
-      data + kStunTransactionIdOffset - kStunMagicCookieLength;
-  if (GetBE32(magic_cookie) != kStunMagicCookie)
+  size_t magic_cookie_offset =
+      kStunTransactionIdOffset - kStunMagicCookieLength;
+  if (GetBE32(data_view.subspan(magic_cookie_offset, 4)) != kStunMagicCookie)
     return false;
 
   // Check the fingerprint type and length.
-  const char* fingerprint_attr_data = data + size - fingerprint_attr_size;
-  if (GetBE16(fingerprint_attr_data) != STUN_ATTR_FINGERPRINT ||
-      GetBE16(fingerprint_attr_data + sizeof(uint16_t)) !=
-          StunUInt32Attribute::SIZE)
+  size_t fingerprint_attr_offset = size - fingerprint_attr_size;
+  if (GetBE16(data_view.subspan(fingerprint_attr_offset, 2)) !=
+          STUN_ATTR_FINGERPRINT ||
+      GetBE16(data_view.subspan(fingerprint_attr_offset + sizeof(uint16_t),
+                                2)) != StunUInt32Attribute::SIZE)
     return false;
 
   // Check the fingerprint value.
-  uint32_t fingerprint =
-      GetBE32(fingerprint_attr_data + kStunAttributeHeaderSize);
+  uint32_t fingerprint = GetBE32(
+      data_view.subspan(fingerprint_attr_offset + kStunAttributeHeaderSize, 4));
   return ((fingerprint ^ STUN_FINGERPRINT_XOR_VALUE) ==
           ComputeCrc32(data, size - fingerprint_attr_size));
 }
@@ -536,13 +546,16 @@ bool StunMessage::IsStunMethod(ArrayView<int> methods,
   if (size % 4 != 0 || size < kStunHeaderSize)
     return false;
 
+  ArrayView<const uint8_t> data_view(reinterpret_cast<const uint8_t*>(data),
+                                     size);
+
   // Skip the rest if the magic cookie isn't present.
-  const char* magic_cookie =
-      data + kStunTransactionIdOffset - kStunMagicCookieLength;
-  if (GetBE32(magic_cookie) != kStunMagicCookie)
+  size_t magic_cookie_offset =
+      kStunTransactionIdOffset - kStunMagicCookieLength;
+  if (GetBE32(data_view.subspan(magic_cookie_offset, 4)) != kStunMagicCookie)
     return false;
 
-  int method = GetBE16(data);
+  int method = GetBE16(data_view);
   for (int m : methods) {
     if (m == method) {
       return true;
diff --git a/api/transport/stun_unittest.cc b/api/transport/stun_unittest.cc
index 542a5bfd24..1d70f5fd8a 100644
--- a/api/transport/stun_unittest.cc
+++ b/api/transport/stun_unittest.cc
@@ -212,6 +212,10 @@ constexpr uint8_t kRtcpPacket[] = {
   0x00, 0x03, 0x73, 0x50,
 };
 
+const ArrayView<const uint8_t> kZeroLenView(kStunMessageWithZeroLength);
+const ArrayView<const uint8_t> kExcessLenView(kStunMessageWithExcessLength);
+const ArrayView<const uint8_t> kSmallLenView(kStunMessageWithSmallLength);
+
 
 // RFC5769 Test Vectors
 // Software name (request):  "STUN test client" (without quotes)
@@ -1197,15 +1201,15 @@ TEST_F(StunTest, ValidateMessageIntegrity) {
   // Again, but with the lengths matching what is claimed in the headers.
   EXPECT_FALSE(StunMessage::ValidateMessageIntegrityForTesting(
       reinterpret_cast<const char*>(kStunMessageWithZeroLength),
-      kStunHeaderSize + GetBE16(&kStunMessageWithZeroLength[2]),
+      kStunHeaderSize + GetBE16(kZeroLenView.subspan(2, 2)),
       kRfc5769SampleMsgPassword));
   EXPECT_FALSE(StunMessage::ValidateMessageIntegrityForTesting(
       reinterpret_cast<const char*>(kStunMessageWithExcessLength),
-      kStunHeaderSize + GetBE16(&kStunMessageWithExcessLength[2]),
+      kStunHeaderSize + GetBE16(kExcessLenView.subspan(2, 2)),
       kRfc5769SampleMsgPassword));
   EXPECT_FALSE(StunMessage::ValidateMessageIntegrityForTesting(
       reinterpret_cast<const char*>(kStunMessageWithSmallLength),
-      kStunHeaderSize + GetBE16(&kStunMessageWithSmallLength[2]),
+      kStunHeaderSize + GetBE16(kSmallLenView.subspan(2, 2)),
       kRfc5769SampleMsgPassword));
 
   // Check that a too-short HMAC doesn't cause buffer overflow.
@@ -1288,15 +1292,15 @@ TEST_F(StunTest, ValidateMessageIntegrity32) {
   // Again, but with the lengths matching what is claimed in the headers.
   EXPECT_FALSE(StunMessage::ValidateMessageIntegrity32ForTesting(
       reinterpret_cast<const char*>(kStunMessageWithZeroLength),
-      kStunHeaderSize + GetBE16(&kStunMessageWithZeroLength[2]),
+      kStunHeaderSize + GetBE16(kZeroLenView.subspan(2, 2)),
       kRfc5769SampleMsgPassword));
   EXPECT_FALSE(StunMessage::ValidateMessageIntegrity32ForTesting(
       reinterpret_cast<const char*>(kStunMessageWithExcessLength),
-      kStunHeaderSize + GetBE16(&kStunMessageWithExcessLength[2]),
+      kStunHeaderSize + GetBE16(kExcessLenView.subspan(2, 2)),
       kRfc5769SampleMsgPassword));
   EXPECT_FALSE(StunMessage::ValidateMessageIntegrity32ForTesting(
       reinterpret_cast<const char*>(kStunMessageWithSmallLength),
-      kStunHeaderSize + GetBE16(&kStunMessageWithSmallLength[2]),
+      kStunHeaderSize + GetBE16(kSmallLenView.subspan(2, 2)),
       kRfc5769SampleMsgPassword));
 
   // Check that a too-short HMAC doesn't cause buffer overflow.
diff --git a/g3doc/todo/byte_order_safety.md b/g3doc/todo/byte_order_safety.md
new file mode 100644
index 0000000000..ee774f8beb
--- /dev/null
+++ b/g3doc/todo/byte_order_safety.md
@@ -0,0 +1,35 @@
+# Objective
+Migrate the `rtc_base/byte_order.h` API from unsafe `void*` pointers to `webrtc::ArrayView<uint8_t>` to improve type safety and bounds checking. This migration follows a multi-phase approach to ensure backward compatibility for external consumers, provide clear deprecation warnings, and eventually remove the unsafe API.
+
+# Key Files & Context
+- `rtc_base/byte_order.h`: The target API for conversion.
+- `api/array_view.h`: The replacement type for buffer spans.
+- 19 identified files with internal usages of the legacy API.
+
+# Implementation Steps
+
+## Phase 0: Documentation
+- Create `g3doc/todo/byte_order_safety.md` and store this migration plan there for tracking.
+
+## Phase 1: Introduce New API
+- Modify `rtc_base/byte_order.h` to include `#include "api/array_view.h"`.
+- Implement `ArrayView` overloads for all `Get/Set` functions (`Get8`, `Set8`, `GetBE16`, etc.).
+- Ensure new implementations include `RTC_DCHECK_GE(data.size(), ...)` for safety.
+- **Note**: Do not add deprecation attributes yet.
+
+## Phase 2: Migrate Internal Usages
+- Replace all internal usages of the legacy `void*` API in the WebRTC codebase with the new `ArrayView` API.
+- This ensures that WebRTC itself does not trigger deprecation warnings/errors once the attributes are added.
+
+## Phase 3: Deprecation and External Notice
+- Add `ABSL_DEPRECATE_AND_INLINE()` to the legacy `void*` overloads in `rtc_base/byte_order.h`.
+- **External Action (Human)**: Send a deprecation notice to relevant mailing lists (e.g., `discuss-webrtc`) informing external consumers of the new `ArrayView` API and the impending removal of the `void*` API.
+- Allow for a transition period for external projects to update their code.
+
+## Phase 4: Final Deletion
+- After the transition period, remove the deprecated `void*` functions and the associated deprecation attributes, leaving only the `ArrayView` API.
+
+# Verification & Testing
+- **Compilation**: Perform a full build (`autoninja -C out/Default`) to ensure no internal usages of deprecated functions remain.
+- **Unit Tests**: Run `out/Default/rtc_unittests --gtest_filter=ByteOrderTest.*` and verify all tests pass with the new API.
+- **Presubmit**: Run `git cl format` and `git cl presubmit -u --force`.
diff --git a/media/BUILD.gn b/media/BUILD.gn
index f61e68589f..febe66aa1e 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -457,6 +457,7 @@ rtc_library("turn_utils") {
     "base/turn_utils.h",
   ]
   deps = [
+    "../api:array_view",
     "../api/transport:stun_types",
     "../rtc_base:byte_order",
     "../rtc_base/system:rtc_export",
diff --git a/media/base/fake_network_interface.h b/media/base/fake_network_interface.h
index 1967ff627c..fa758aa4af 100644
--- a/media/base/fake_network_interface.h
+++ b/media/base/fake_network_interface.h
@@ -18,6 +18,7 @@
 #include <utility>
 #include <vector>
 
+#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/task_queue/pending_task_safety_flag.h"
 #include "api/task_queue/task_queue_base.h"
@@ -190,7 +191,9 @@ class FakeNetworkInterface : public MediaChannelNetworkInterface {
  private:
   void SetRtpSsrc(uint32_t ssrc, CopyOnWriteBuffer& buffer) {
     RTC_CHECK_GE(buffer.size(), 12);
-    SetBE32(buffer.MutableData() + 8, ssrc);
+    SetBE32(
+        ArrayView<uint8_t>(buffer.MutableData(), buffer.size()).subspan(8, 4),
+        ssrc);
   }
 
   void GetNumRtpBytesAndPackets(uint32_t ssrc, int* bytes, int* packets) {
diff --git a/media/base/rtp_utils.cc b/media/base/rtp_utils.cc
index 1bb208ab31..f7cb5322c8 100644
--- a/media/base/rtp_utils.cc
+++ b/media/base/rtp_utils.cc
@@ -145,7 +145,8 @@ bool GetRtcpSsrc(const void* data, size_t len, uint32_t* value) {
   // SDES packet parsing is not supported.
   if (pl_type == kRtcpTypeSDES)
     return false;
-  *value = GetBE32(static_cast<const uint8_t*>(data) + 4);
+  *value = GetBE32(
+      ArrayView<const uint8_t>(static_cast<const uint8_t*>(data) + 4, 4));
   return true;
 }
 
@@ -215,7 +216,7 @@ bool ValidateRtpHeader(ArrayView<const uint8_t> rtp, size_t* header_length) {
   // Getting extension profile length.
   // Length is in 32 bit words.
   uint16_t extension_length_in_32bits =
-      GetBE16(&rtp[header_length_without_extension + 2]);
+      GetBE16(rtp.subspan(header_length_without_extension + 2, 2));
   size_t extension_length = extension_length_in_32bits * 4;
 
   size_t rtp_header_length = extension_length +
@@ -262,10 +263,13 @@ bool UpdateRtpAbsSendTimeExtension(ArrayView<uint8_t> packet,
   uint8_t* rtp = packet.data();
   rtp += header_length_without_extension;
 
+  ArrayView<const uint8_t> extension_header =
+      packet.subspan(header_length_without_extension, kRtpExtensionHeaderLen);
+
   // Getting extension profile ID and length.
-  uint16_t profile_id = GetBE16(rtp);
+  uint16_t profile_id = GetBE16(extension_header);
   // Length is in 32 bit words.
-  uint16_t extension_length_in_32bits = GetBE16(rtp + 2);
+  uint16_t extension_length_in_32bits = GetBE16(extension_header.subspan(2, 2));
   size_t extension_length = extension_length_in_32bits * 4;
 
   rtp += kRtpExtensionHeaderLen;  // Moving past extension header.
@@ -374,7 +378,7 @@ bool ApplyPacketOptions(ArrayView<uint8_t> data,
   }
 
   // Making sure we have a valid RTP packet at the end.
-  auto packet = data.subview(rtp_start_pos, rtp_length);
+  auto packet = data.subspan(rtp_start_pos, rtp_length);
   if (!IsRtpPacket(packet) || !ValidateRtpHeader(packet, nullptr)) {
     RTC_DCHECK_NOTREACHED();
     return false;
diff --git a/media/base/turn_utils.cc b/media/base/turn_utils.cc
index d006bb892d..77f0f18ae5 100644
--- a/media/base/turn_utils.cc
+++ b/media/base/turn_utils.cc
@@ -13,6 +13,7 @@
 #include <cstddef>
 #include <cstdint>
 
+#include "api/array_view.h"
 #include "api/transport/stun.h"
 #include "rtc_base/byte_order.h"
 
@@ -31,7 +32,7 @@ bool IsTurnSendIndicationPacket(const uint8_t* data, size_t length) {
     return false;
   }
 
-  uint16_t type = GetBE16(data);
+  uint16_t type = GetBE16(ArrayView<const uint8_t>(data, 2));
   return (type == TURN_SEND_INDICATION);
 }
 
@@ -41,6 +42,7 @@ bool UnwrapTurnPacket(const uint8_t* packet,
                       size_t packet_size,
                       size_t* content_position,
                       size_t* content_size) {
+  ArrayView<const uint8_t> data_view(packet, packet_size);
   if (IsTurnChannelData(packet, packet_size)) {
     // Turn Channel Message header format.
     //   0                   1                   2                   3
@@ -52,7 +54,7 @@ bool UnwrapTurnPacket(const uint8_t* packet,
     // /                       Application Data                        /
     // /                                                               /
     // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-    size_t length = GetBE16(&packet[2]);
+    size_t length = GetBE16(data_view.subspan(2, 2));
     if (length + kTurnChannelHeaderLength > packet_size) {
       return false;
     }
@@ -64,7 +66,7 @@ bool UnwrapTurnPacket(const uint8_t* packet,
 
   if (IsTurnSendIndicationPacket(packet, packet_size)) {
     // Validate STUN message length.
-    const size_t stun_message_length = GetBE16(&packet[2]);
+    const size_t stun_message_length = GetBE16(data_view.subspan(2, 2));
     if (stun_message_length + kStunHeaderSize != packet_size) {
       return false;
     }
@@ -94,8 +96,8 @@ bool UnwrapTurnPacket(const uint8_t* packet,
       }
 
       // Getting attribute type and length.
-      attr_type = GetBE16(&packet[pos]);
-      attr_length = GetBE16(&packet[pos + sizeof(attr_type)]);
+      attr_type = GetBE16(data_view.subspan(pos, 2));
+      attr_length = GetBE16(data_view.subspan(pos + sizeof(attr_type), 2));
 
       pos += kAttrHeaderLength;  // Skip STUN_DATA_ATTR header.
 
diff --git a/media/engine/webrtc_voice_engine_unittest.cc b/media/engine/webrtc_voice_engine_unittest.cc
index 88a6acf7a5..a257806a0d 100644
--- a/media/engine/webrtc_voice_engine_unittest.cc
+++ b/media/engine/webrtc_voice_engine_unittest.cc
@@ -2788,7 +2788,7 @@ TEST_P(WebRtcVoiceEngineTestFake, RecvWithMultipleStreams) {
   uint8_t packets[4][sizeof(kPcmuFrame)];
   for (size_t i = 0; i < std::size(packets); ++i) {
     memcpy(packets[i], kPcmuFrame, sizeof(kPcmuFrame));
-    SetBE32(packets[i] + 8, static_cast<uint32_t>(i));
+    SetBE32(ArrayView<uint8_t>(packets[i] + 8, 4), static_cast<uint32_t>(i));
   }
 
   const FakeAudioReceiveStream& s1 = GetRecvStream(ssrc1);
@@ -2877,9 +2877,9 @@ TEST_P(WebRtcVoiceEngineTestFake,
   // Deliver a couple packets with unsignaled SSRCs.
   uint8_t packet[sizeof(kPcmuFrame)];
   memcpy(packet, kPcmuFrame, sizeof(kPcmuFrame));
-  SetBE32(&packet[8], 0x1234);
+  SetBE32(ArrayView<uint8_t>(&packet[8], 4), 0x1234);
   DeliverPacket(packet);
-  SetBE32(&packet[8], 0x5678);
+  SetBE32(ArrayView<uint8_t>(&packet[8], 4), 0x5678);
   DeliverPacket(packet);
 
   // Verify that the receive streams were created.
@@ -2901,7 +2901,7 @@ TEST_P(WebRtcVoiceEngineTestFake, RecvMultipleUnsignaled) {
 
   // Note that SSRC = 0 is not supported.
   for (uint32_t ssrc = 1; ssrc < (1 + kMaxUnsignaledRecvStreams); ++ssrc) {
-    SetBE32(&packet[8], ssrc);
+    SetBE32(ArrayView<uint8_t>(&packet[8], 4), ssrc);
     DeliverPacket(packet);
 
     // Verify we have one new stream for each loop iteration.
@@ -2912,7 +2912,7 @@ TEST_P(WebRtcVoiceEngineTestFake, RecvMultipleUnsignaled) {
 
   // Sending on the same SSRCs again should not create new streams.
   for (uint32_t ssrc = 1; ssrc < (1 + kMaxUnsignaledRecvStreams); ++ssrc) {
-    SetBE32(&packet[8], ssrc);
+    SetBE32(ArrayView<uint8_t>(&packet[8], 4), ssrc);
     DeliverPacket(packet);
 
     EXPECT_EQ(kMaxUnsignaledRecvStreams, call_.GetAudioReceiveStreams().size());
@@ -2922,7 +2922,7 @@ TEST_P(WebRtcVoiceEngineTestFake, RecvMultipleUnsignaled) {
 
   // Send on another SSRC, the oldest unsignaled stream (SSRC=1) is replaced.
   constexpr uint32_t kAnotherSsrc = 667;
-  SetBE32(&packet[8], kAnotherSsrc);
+  SetBE32(ArrayView<uint8_t>(&packet[8], 4), kAnotherSsrc);
   DeliverPacket(packet);
 
   const auto& streams = call_.GetAudioReceiveStreams();
@@ -2947,7 +2947,7 @@ TEST_P(WebRtcVoiceEngineTestFake, RecvUnsignaledAfterSignaled) {
 
   // Add a known stream, send packet and verify we got it.
   const uint32_t signaled_ssrc = 1;
-  SetBE32(&packet[8], signaled_ssrc);
+  SetBE32(ArrayView<uint8_t>(&packet[8], 4), signaled_ssrc);
   EXPECT_TRUE(AddRecvStream(signaled_ssrc));
   DeliverPacket(packet);
   EXPECT_TRUE(GetRecvStream(signaled_ssrc).VerifyLastPacket(packet));
@@ -2956,7 +2956,7 @@ TEST_P(WebRtcVoiceEngineTestFake, RecvUnsignaledAfterSignaled) {
   // Note that the first unknown SSRC cannot be 0, because we only support
   // creating receive streams for SSRC!=0.
   const uint32_t unsignaled_ssrc = 7011;
-  SetBE32(&packet[8], unsignaled_ssrc);
+  SetBE32(ArrayView<uint8_t>(&packet[8], 4), unsignaled_ssrc);
   DeliverPacket(packet);
   EXPECT_TRUE(GetRecvStream(unsignaled_ssrc).VerifyLastPacket(packet));
   EXPECT_EQ(2u, call_.GetAudioReceiveStreams().size());
@@ -2964,7 +2964,7 @@ TEST_P(WebRtcVoiceEngineTestFake, RecvUnsignaledAfterSignaled) {
   DeliverPacket(packet);
   EXPECT_EQ(2, GetRecvStream(unsignaled_ssrc).received_packets());
 
-  SetBE32(&packet[8], signaled_ssrc);
+  SetBE32(ArrayView<uint8_t>(&packet[8], 4), signaled_ssrc);
   DeliverPacket(packet);
   EXPECT_EQ(2, GetRecvStream(signaled_ssrc).received_packets());
   EXPECT_EQ(2u, call_.GetAudioReceiveStreams().size());
@@ -3429,7 +3429,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetOutputVolumeUnsignaledRecvStream) {
   // Spawn an unsignaled stream by sending a packet - gain should be 2.
   uint8_t pcmuFrame2[sizeof(kPcmuFrame)];
   memcpy(pcmuFrame2, kPcmuFrame, sizeof(kPcmuFrame));
-  SetBE32(&pcmuFrame2[8], kSsrcX);
+  SetBE32(ArrayView<uint8_t>(&pcmuFrame2[8], 4), kSsrcX);
   DeliverPacket(pcmuFrame2);
   EXPECT_DOUBLE_EQ(2, GetRecvStream(kSsrcX).gain());
 
@@ -3491,7 +3491,7 @@ TEST_P(WebRtcVoiceEngineTestFake,
   // Spawn an unsignaled stream by sending a packet - delay should be 100.
   uint8_t pcmuFrame2[sizeof(kPcmuFrame)];
   memcpy(pcmuFrame2, kPcmuFrame, sizeof(kPcmuFrame));
-  SetBE32(&pcmuFrame2[8], kSsrcX);
+  SetBE32(ArrayView<uint8_t>(&pcmuFrame2[8], 4), kSsrcX);
   DeliverPacket(pcmuFrame2);
   EXPECT_EQ(
       100, receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrcX).value_or(-1));
@@ -3693,7 +3693,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetRawAudioSinkUnsignaledRecvStream) {
   // and the previous unsignaled stream should lose it.
   uint8_t pcmuFrame2[sizeof(kPcmuFrame)];
   memcpy(pcmuFrame2, kPcmuFrame, sizeof(kPcmuFrame));
-  SetBE32(&pcmuFrame2[8], kSsrcX);
+  SetBE32(ArrayView<uint8_t>(&pcmuFrame2[8], 4), kSsrcX);
   DeliverPacket(pcmuFrame2);
   if (kMaxUnsignaledRecvStreams > 1) {
     EXPECT_EQ(nullptr, GetRecvStream(kSsrc1).sink());
diff --git a/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc b/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
index a69c9203f2..b77f891c42 100644
--- a/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
+++ b/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
@@ -160,7 +160,7 @@ AudioEncoder::EncodedInfo AudioEncoderCopyRed::EncodeImpl(
     const uint32_t timestamp_delta =
         info.encoded_timestamp - it->first.encoded_timestamp;
     encoded->data()[header_offset] = it->first.payload_type | 0x80;
-    SetBE16(static_cast<uint8_t*>(encoded->data()) + header_offset + 1,
+    SetBE16(ArrayView<uint8_t>(*encoded).subspan(header_offset + 1, 2),
             (timestamp_delta << 2) | (it->first.encoded_bytes >> 8));
     encoded->data()[header_offset + 3] = it->first.encoded_bytes & 0xff;
     header_offset += kRedHeaderLength;
diff --git a/p2p/base/async_stun_tcp_socket.cc b/p2p/base/async_stun_tcp_socket.cc
index ff8d510570..aeb67141c1 100644
--- a/p2p/base/async_stun_tcp_socket.cc
+++ b/p2p/base/async_stun_tcp_socket.cc
@@ -126,13 +126,13 @@ size_t AsyncStunTCPSocket::ProcessInput(ArrayView<const uint8_t> data) {
 }
 
 size_t AsyncStunTCPSocket::GetExpectedLength(const void* data,
-                                             size_t /* len */,
+                                             size_t len,
                                              int* pad_bytes) {
   *pad_bytes = 0;
-  PacketLength pkt_len =
-      GetBE16(static_cast<const char*>(data) + kPacketLenOffset);
+  ArrayView<const uint8_t> view(static_cast<const uint8_t*>(data), len);
+  PacketLength pkt_len = GetBE16(view.subspan(kPacketLenOffset, 2));
   size_t expected_pkt_len;
-  uint16_t msg_type = GetBE16(data);
+  uint16_t msg_type = GetBE16(view);
   if (IsStunMessage(msg_type)) {
     // STUN message.
     expected_pkt_len = kStunHeaderSize + pkt_len;
diff --git a/p2p/base/turn_port.cc b/p2p/base/turn_port.cc
index 7eba2fa37d..9a22505231 100644
--- a/p2p/base/turn_port.cc
+++ b/p2p/base/turn_port.cc
@@ -789,7 +789,7 @@ bool TurnPort::HandleIncomingPacket(AsyncPacketSocket* socket,
   // Check the message type, to see if is a Channel Data message.
   // The message will either be channel data, a TURN data indication, or
   // a response to a previous request.
-  uint16_t msg_type = GetBE16(packet.payload().data());
+  uint16_t msg_type = GetBE16(packet.payload());
   if (IsTurnChannelData(msg_type)) {
     HandleChannelData(msg_type, packet);
     return true;
@@ -1118,8 +1118,9 @@ void TurnPort::HandleChannelData(uint16_t channel_id,
   //   +-------------------------------+
 
   // Extract header fields from the message.
-  uint16_t len = GetBE16(packet.payload().data() + 2);
-  if (len > packet.payload().size() - TURN_CHANNEL_HEADER_SIZE) {
+  ArrayView<const uint8_t> payload = packet.payload();
+  uint16_t len = GetBE16(payload.subspan(2, 2));
+  if (len > payload.size() - TURN_CHANNEL_HEADER_SIZE) {
     RTC_LOG(LS_WARNING) << ToString()
                         << ": Received TURN channel data message with "
                            "incorrect length, len: "
@@ -1137,7 +1138,7 @@ void TurnPort::HandleChannelData(uint16_t channel_id,
     return;
   }
   ReceivedIpPacket unwrapped_packet = ReceivedIpPacket(
-      packet.payload().subview(TURN_CHANNEL_HEADER_SIZE, len), entry->address(),
+      payload.subspan(TURN_CHANNEL_HEADER_SIZE, len), entry->address(),
       packet.arrival_time(), packet.ecn(), packet.decryption_info());
   DispatchPacket(unwrapped_packet, PROTO_UDP);
 }
diff --git a/p2p/dtls/dtls_transport_unittest.cc b/p2p/dtls/dtls_transport_unittest.cc
index 8975f7bd53..afc7d46726 100644
--- a/p2p/dtls/dtls_transport_unittest.cc
+++ b/p2p/dtls/dtls_transport_unittest.cc
@@ -277,7 +277,10 @@ class DtlsTestClient {
       // against, and make sure that it doesn't look like DTLS.
       memset(packet.get(), sent & 0xff, size);
       packet[0] = (srtp) ? kRtpLeadByte : 0x00;
-      SetBE32(packet.get() + kPacketNumOffset, static_cast<uint32_t>(sent));
+      SetBE32(
+          ArrayView<uint8_t>(
+              reinterpret_cast<uint8_t*>(packet.get() + kPacketNumOffset), 4),
+          static_cast<uint32_t>(sent));
 
       // Only set the bypass flag if we've activated DTLS.
       int flags = (certificate_ && srtp) ? PF_SRTP_BYPASS : 0;
@@ -316,7 +319,8 @@ class DtlsTestClient {
     if (size != packet_size_ || (data[0] != 0 && (data[0]) != 0x80)) {
       return false;
     }
-    uint32_t packet_num = GetBE32(data + kPacketNumOffset);
+    uint32_t packet_num =
+        GetBE32(ArrayView<const uint8_t>(data + kPacketNumOffset, 4));
     for (size_t i = kPacketHeaderLen; i < size; ++i) {
       if (data[i] != (packet_num & 0xff)) {
         return false;
@@ -333,7 +337,8 @@ class DtlsTestClient {
     if (size <= packet_size_) {
       return false;
     }
-    uint32_t packet_num = GetBE32(data + kPacketNumOffset);
+    uint32_t packet_num =
+        GetBE32(ArrayView<const uint8_t>(data + kPacketNumOffset, 4));
     int num_matches = 0;
     for (size_t i = kPacketNumOffset; i < size; ++i) {
       if (data[i] == (packet_num & 0xff)) {
@@ -742,7 +747,9 @@ TEST_F(DtlsTransportInternalImplTest, TestSendPacketWithOptions) {
   std::unique_ptr<char[]> packet(new char[size]);
   memset(packet.get(), 0, size);
   packet[0] = 0x00;
-  SetBE32(packet.get() + kPacketNumOffset, 0);
+  SetBE32(ArrayView<uint8_t>(
+              reinterpret_cast<uint8_t*>(packet.get() + kPacketNumOffset), 4),
+          0);
 
   AsyncSocketPacketOptions packet_options;
   packet_options.packet_id = kFakePacketId;
@@ -772,7 +779,9 @@ TEST_F(DtlsTransportInternalImplTest, TestSendSrtpBypassPacketWithOptions) {
   std::unique_ptr<char[]> packet(new char[size]);
   memset(packet.get(), 0, size);
   packet[0] = kRtpLeadByte;  // Make it look like an SRTP packet.
-  SetBE32(packet.get() + kPacketNumOffset, 0);
+  SetBE32(ArrayView<uint8_t>(
+              reinterpret_cast<uint8_t*>(packet.get() + kPacketNumOffset), 4),
+          0);
 
   AsyncSocketPacketOptions packet_options;
   packet_options.packet_id = kFakePacketId;
diff --git a/p2p/test/turn_server.cc b/p2p/test/turn_server.cc
index 154f22c1d2..a4fc03b819 100644
--- a/p2p/test/turn_server.cc
+++ b/p2p/test/turn_server.cc
@@ -173,24 +173,25 @@ void TurnServer::OnInternalPacket(AsyncPacketSocket* socket,
                                   const ReceivedIpPacket& packet) {
   RTC_DCHECK_RUN_ON(thread_);
   // Fail if the packet is too small to even contain a channel header.
-  if (packet.payload().size() < TURN_CHANNEL_HEADER_SIZE) {
+  ArrayView<const uint8_t> payload = packet.payload();
+  if (payload.size() < TURN_CHANNEL_HEADER_SIZE) {
     return;
   }
   auto iter = server_sockets_.find(socket);
   RTC_DCHECK(iter != server_sockets_.end());
   TurnServerConnection conn(packet.source_address(), iter->second, socket);
-  uint16_t msg_type = GetBE16(packet.payload().data());
+  uint16_t msg_type = GetBE16(payload);
   if (!IsTurnChannelData(msg_type)) {
     // This is a STUN message.
-    HandleStunMessage(&conn, packet.payload(), packet.ecn());
+    HandleStunMessage(&conn, payload, packet.ecn());
   } else {
     // This is a channel message; let the allocation handle it.
     TurnServerAllocation* allocation = FindAllocation(&conn);
     if (allocation) {
-      allocation->HandleChannelData(packet.payload(), packet.ecn());
+      allocation->HandleChannelData(payload, packet.ecn());
     }
     if (stun_message_observer_ != nullptr) {
-      stun_message_observer_->ReceivedChannelData(packet.payload());
+      stun_message_observer_->ReceivedChannelData(payload);
     }
   }
 }
@@ -786,7 +787,7 @@ void TurnServerAllocation::HandleChannelBindRequest(const TurnMessage* msg) {
 void TurnServerAllocation::HandleChannelData(ArrayView<const uint8_t> payload,
                                              EcnMarking ecn) {
   // Extract the channel number from the data.
-  uint16_t channel_id = GetBE16(payload.data());
+  uint16_t channel_id = GetBE16(payload);
   auto channel = FindChannel(channel_id);
   if (channel != channels_.end()) {
     // Send the data to the peer address.
diff --git a/pc/channel_unittest.cc b/pc/channel_unittest.cc
index e277fece3f..68c35f678c 100644
--- a/pc/channel_unittest.cc
+++ b/pc/channel_unittest.cc
@@ -528,10 +528,11 @@ class ChannelTest : public ::testing::Test {
                                int pl_type) {
     webrtc::Buffer data(rtp_packet_.data(), rtp_packet_.size());
     // Set SSRC in the rtp packet copy.
-    webrtc::SetBE32(data.data() + 8, ssrc);
-    webrtc::SetBE16(data.data() + 2, sequence_number);
+    webrtc::SetBE32(webrtc::ArrayView<uint8_t>(data).subspan(8, 4), ssrc);
+    webrtc::SetBE16(webrtc::ArrayView<uint8_t>(data).subspan(2, 2),
+                    sequence_number);
     if (pl_type >= 0) {
-      webrtc::Set8(data.data(), 1, static_cast<uint8_t>(pl_type));
+      webrtc::Set8(data, 1, static_cast<uint8_t>(pl_type));
     }
     return data;
   }
diff --git a/pc/dtls_srtp_transport_unittest.cc b/pc/dtls_srtp_transport_unittest.cc
index c3cd857ae5..3279b3124b 100644
--- a/pc/dtls_srtp_transport_unittest.cc
+++ b/pc/dtls_srtp_transport_unittest.cc
@@ -138,7 +138,7 @@ class DtlsSrtpTransportTest : public ::testing::Test {
     memcpy(rtp_packet_data, kPcmuFrame, rtp_len);
     // In order to be able to run this test function multiple times we can not
     // use the same sequence number twice. Increase the sequence number by one.
-    SetBE16(ArrayView<uint8_t>(rtp_packet_buffer).subspan(2).data(),
+    SetBE16(ArrayView<uint8_t>(rtp_packet_buffer).subspan(2),
             ++sequence_number_);
     CopyOnWriteBuffer rtp_packet1to2(rtp_packet_data, rtp_len, packet_size);
     CopyOnWriteBuffer rtp_packet2to1(rtp_packet_data, rtp_len, packet_size);
@@ -211,7 +211,7 @@ class DtlsSrtpTransportTest : public ::testing::Test {
     memcpy(rtp_packet_data, kPcmuFrameWithExtensions, rtp_len);
     // In order to be able to run this test function multiple times we can not
     // use the same sequence number twice. Increase the sequence number by one.
-    SetBE16(ArrayView<uint8_t>(rtp_packet_buffer).subspan(2).data(),
+    SetBE16(ArrayView<uint8_t>(rtp_packet_buffer).subspan(2),
             ++sequence_number_);
     CopyOnWriteBuffer rtp_packet1to2(rtp_packet_data, rtp_len, packet_size);
     CopyOnWriteBuffer rtp_packet2to1(rtp_packet_data, rtp_len, packet_size);
diff --git a/pc/jsep_transport_unittest.cc b/pc/jsep_transport_unittest.cc
index 0ee7519343..a9f4189fda 100644
--- a/pc/jsep_transport_unittest.cc
+++ b/pc/jsep_transport_unittest.cc
@@ -1173,7 +1173,7 @@ class JsepTransport2HeaderExtensionTest
     memcpy(rtp_packet_data, kPcmuFrameWithExtensions, rtp_len);
     // In order to be able to run this test function multiple times we can not
     // use the same sequence number twice. Increase the sequence number by one.
-    SetBE16(ArrayView<uint8_t>(rtp_packet_buffer).subspan(2).data(),
+    SetBE16(ArrayView<uint8_t>(rtp_packet_buffer).subspan(2),
             ++sequence_number_);
     CopyOnWriteBuffer rtp_packet(rtp_packet_data, rtp_len, packet_size);
 
diff --git a/pc/srtp_session_unittest.cc b/pc/srtp_session_unittest.cc
index 79a87ec9c5..a9ab7af918 100644
--- a/pc/srtp_session_unittest.cc
+++ b/pc/srtp_session_unittest.cc
@@ -15,6 +15,7 @@
 #include <limits>
 #include <vector>
 
+#include "api/array_view.h"
 #include "api/field_trials.h"
 #include "media/base/fake_rtp.h"
 #include "pc/test/srtp_test_util.h"
@@ -213,35 +214,50 @@ TEST_F(SrtpSessionTest, TestReplay) {
                              kEncryptedHeaderExtensionIds));
 
   // Initial sequence number.
-  SetBE16(rtp_packet_.MutableData<uint8_t>() + 2, seqnum_big);
+  SetBE16(
+      ArrayView<uint8_t>(rtp_packet_.MutableData<uint8_t>(), rtp_packet_.size())
+          .subspan(2, 2),
+      seqnum_big);
   EXPECT_TRUE(s1_.ProtectRtp(rtp_packet_));
   rtp_packet_.SetData(kPcmuFrame, sizeof(kPcmuFrame));
 
   // Replay within the 1024 window should succeed.
-  SetBE16(rtp_packet_.MutableData<uint8_t>() + 2,
-          seqnum_big - replay_window + 1);
+  SetBE16(
+      ArrayView<uint8_t>(rtp_packet_.MutableData<uint8_t>(), rtp_packet_.size())
+          .subspan(2, 2),
+      seqnum_big - replay_window + 1);
   EXPECT_TRUE(s1_.ProtectRtp(rtp_packet_));
   rtp_packet_.SetData(kPcmuFrame, sizeof(kPcmuFrame));
 
   // Replay out side of the 1024 window should fail.
-  SetBE16(rtp_packet_.MutableData<uint8_t>() + 2,
-          seqnum_big - replay_window - 1);
+  SetBE16(
+      ArrayView<uint8_t>(rtp_packet_.MutableData<uint8_t>(), rtp_packet_.size())
+          .subspan(2, 2),
+      seqnum_big - replay_window - 1);
   EXPECT_FALSE(s1_.ProtectRtp(rtp_packet_));
   rtp_packet_.SetData(kPcmuFrame, sizeof(kPcmuFrame));
 
   // Increment sequence number to a small number.
-  SetBE16(rtp_packet_.MutableData<uint8_t>() + 2, seqnum_small);
+  SetBE16(
+      ArrayView<uint8_t>(rtp_packet_.MutableData<uint8_t>(), rtp_packet_.size())
+          .subspan(2, 2),
+      seqnum_small);
   EXPECT_TRUE(s1_.ProtectRtp(rtp_packet_));
 
   // Replay around 0 but out side of the 1024 window should fail.
-  SetBE16(rtp_packet_.MutableData<uint8_t>() + 2,
-          kMaxSeqnum + seqnum_small - replay_window - 1);
+  SetBE16(
+      ArrayView<uint8_t>(rtp_packet_.MutableData<uint8_t>(), rtp_packet_.size())
+          .subspan(2, 2),
+      kMaxSeqnum + seqnum_small - replay_window - 1);
   EXPECT_FALSE(s1_.ProtectRtp(rtp_packet_));
   rtp_packet_.SetData(kPcmuFrame, sizeof(kPcmuFrame));
 
   // Replay around 0 but within the 1024 window should succeed.
   for (uint16_t seqnum = 65000; seqnum < 65003; ++seqnum) {
-    SetBE16(rtp_packet_.MutableData<uint8_t>() + 2, seqnum);
+    SetBE16(ArrayView<uint8_t>(rtp_packet_.MutableData<uint8_t>(),
+                               rtp_packet_.size())
+                .subspan(2, 2),
+            seqnum);
     EXPECT_TRUE(s1_.ProtectRtp(rtp_packet_));
     rtp_packet_.SetData(kPcmuFrame, sizeof(kPcmuFrame));
   }
@@ -251,7 +267,10 @@ TEST_F(SrtpSessionTest, TestReplay) {
   // without the fix, the loop above would keep incrementing local sequence
   // number in libsrtp, eventually the new sequence number would go out side
   // of the window.
-  SetBE16(rtp_packet_.MutableData<uint8_t>() + 2, seqnum_small + 1);
+  SetBE16(
+      ArrayView<uint8_t>(rtp_packet_.MutableData<uint8_t>(), rtp_packet_.size())
+          .subspan(2, 2),
+      seqnum_small + 1);
   EXPECT_TRUE(s1_.ProtectRtp(rtp_packet_));
 }
 
diff --git a/pc/srtp_transport_unittest.cc b/pc/srtp_transport_unittest.cc
index f9dbeb7875..e97260f31f 100644
--- a/pc/srtp_transport_unittest.cc
+++ b/pc/srtp_transport_unittest.cc
@@ -142,7 +142,7 @@ class SrtpTransportTest : public ::testing::Test {
     memcpy(rtp_packet_data, kPcmuFrame, rtp_len);
     // In order to be able to run this test function multiple times we can not
     // use the same sequence number twice. Increase the sequence number by one.
-    SetBE16(ArrayView<uint8_t>(rtp_packet_buffer).subspan(2).data(),
+    SetBE16(ArrayView<uint8_t>(rtp_packet_buffer).subspan(2),
             ++sequence_number_);
     CopyOnWriteBuffer rtp_packet1to2(rtp_packet_data, rtp_len, packet_size);
     CopyOnWriteBuffer rtp_packet2to1(rtp_packet_data, rtp_len, packet_size);
@@ -264,7 +264,7 @@ class SrtpTransportTest : public ::testing::Test {
     memcpy(rtp_packet_data, kPcmuFrameWithExtensions, rtp_len);
     // In order to be able to run this test function multiple times we can not
     // use the same sequence number twice. Increase the sequence number by one.
-    SetBE16(ArrayView<uint8_t>(rtp_packet_buffer).subspan(2).data(),
+    SetBE16(ArrayView<uint8_t>(rtp_packet_buffer).subspan(2),
             ++sequence_number_);
     CopyOnWriteBuffer rtp_packet1to2(rtp_packet_data, rtp_len, packet_size);
     CopyOnWriteBuffer rtp_packet2to1(rtp_packet_data, rtp_len, packet_size);
diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
index 962d992edd..d582c6275a 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -90,7 +90,12 @@ rtc_source_set("buffer") {
 rtc_source_set("byte_order") {
   visibility = [ "*" ]
   sources = [ "byte_order.h" ]
-  deps = [ "system:arch" ]
+  deps = [
+    ":checks",
+    "../api:array_view",
+    "system:arch",
+    "//third_party/abseil-cpp/absl/base:core_headers",
+  ]
 }
 
 rtc_source_set("mod_ops") {
diff --git a/rtc_base/async_tcp_socket.cc b/rtc_base/async_tcp_socket.cc
index 219b325003..9e2221c82a 100644
--- a/rtc_base/async_tcp_socket.cc
+++ b/rtc_base/async_tcp_socket.cc
@@ -139,7 +139,7 @@ int AsyncTCPSocketBase::FlushOutBuffer() {
       res = -1;
       break;
     }
-    view = view.subview(res);
+    view = view.subspan(res);
   }
   if (res > 0) {
     // The output buffer may have been written out over multiple partial Send(),
@@ -280,12 +280,13 @@ size_t AsyncTCPSocket::ProcessInput(ArrayView<const uint8_t> data) {
     if (bytes_left < kPacketLenSize)
       return processed_bytes;
 
-    PacketLength pkt_len = GetBE16(data.data() + processed_bytes);
+    PacketLength pkt_len =
+        GetBE16(data.subspan(processed_bytes, kPacketLenSize));
     if (bytes_left < kPacketLenSize + pkt_len)
       return processed_bytes;
 
     ReceivedIpPacket received_packet(
-        data.subview(processed_bytes + kPacketLenSize, pkt_len), remote_addr,
+        data.subspan(processed_bytes + kPacketLenSize, pkt_len), remote_addr,
         env_.clock().CurrentTime());
     NotifyPacketReceived(received_packet);
     processed_bytes += kPacketLenSize + pkt_len;
diff --git a/rtc_base/byte_buffer_unittest.cc b/rtc_base/byte_buffer_unittest.cc
index 07752b1092..cb4fdde41c 100644
--- a/rtc_base/byte_buffer_unittest.cc
+++ b/rtc_base/byte_buffer_unittest.cc
@@ -56,9 +56,12 @@ TEST(ByteBufferTest, TestByteOrder) {
     EXPECT_EQ(n64, HostToNetwork64(n64));
 
     // GetBE converts big endian to little endian here.
-    EXPECT_EQ(n16 >> 8, GetBE16(&n16));
-    EXPECT_EQ(n32 >> 24, GetBE32(&n32));
-    EXPECT_EQ(n64 >> 56, GetBE64(&n64));
+    EXPECT_EQ(n16 >> 8, GetBE16(ArrayView<const uint8_t>(
+                            reinterpret_cast<const uint8_t*>(&n16), 2)));
+    EXPECT_EQ(n32 >> 24, GetBE32(ArrayView<const uint8_t>(
+                             reinterpret_cast<const uint8_t*>(&n32), 4)));
+    EXPECT_EQ(n64 >> 56, GetBE64(ArrayView<const uint8_t>(
+                             reinterpret_cast<const uint8_t*>(&n64), 8)));
   } else {
     // The host is little endian.
     EXPECT_NE(n16, HostToNetwork16(n16));
@@ -66,14 +69,23 @@ TEST(ByteBufferTest, TestByteOrder) {
     EXPECT_NE(n64, HostToNetwork64(n64));
 
     // GetBE converts little endian to big endian here.
-    EXPECT_EQ(GetBE16(&n16), HostToNetwork16(n16));
-    EXPECT_EQ(GetBE32(&n32), HostToNetwork32(n32));
-    EXPECT_EQ(GetBE64(&n64), HostToNetwork64(n64));
+    EXPECT_EQ(GetBE16(ArrayView<const uint8_t>(
+                  reinterpret_cast<const uint8_t*>(&n16), 2)),
+              HostToNetwork16(n16));
+    EXPECT_EQ(GetBE32(ArrayView<const uint8_t>(
+                  reinterpret_cast<const uint8_t*>(&n32), 4)),
+              HostToNetwork32(n32));
+    EXPECT_EQ(GetBE64(ArrayView<const uint8_t>(
+                  reinterpret_cast<const uint8_t*>(&n64), 8)),
+              HostToNetwork64(n64));
 
     // GetBE converts little endian to big endian here.
-    EXPECT_EQ(n16 << 8, GetBE16(&n16));
-    EXPECT_EQ(n32 << 24, GetBE32(&n32));
-    EXPECT_EQ(n64 << 56, GetBE64(&n64));
+    EXPECT_EQ(n16 << 8, GetBE16(ArrayView<const uint8_t>(
+                            reinterpret_cast<const uint8_t*>(&n16), 2)));
+    EXPECT_EQ(n32 << 24, GetBE32(ArrayView<const uint8_t>(
+                             reinterpret_cast<const uint8_t*>(&n32), 4)));
+    EXPECT_EQ(n64 << 56, GetBE64(ArrayView<const uint8_t>(
+                             reinterpret_cast<const uint8_t*>(&n64), 8)));
   }
 }
 
diff --git a/rtc_base/byte_order.h b/rtc_base/byte_order.h
index 3a9bfe46e0..7f163980b5 100644
--- a/rtc_base/byte_order.h
+++ b/rtc_base/byte_order.h
@@ -15,6 +15,8 @@
 #include <cstdlib>
 #include <cstring>
 
+#include "api/array_view.h"
+#include "rtc_base/checks.h"
 #include "rtc_base/system/arch.h"  // IWYU pragma: keep
 
 #if defined(WEBRTC_POSIX)
@@ -92,75 +94,177 @@ namespace webrtc {
 
 // Reading and writing of little and big-endian numbers from memory
 
+inline void Set8(ArrayView<uint8_t> data, size_t offset, uint8_t v) {
+  data[offset] = v;
+}
+
+[[deprecated("Use version with ArrayView")]]
 inline void Set8(void* memory, size_t offset, uint8_t v) {
   static_cast<uint8_t*>(memory)[offset] = v;
 }
 
+inline uint8_t Get8(ArrayView<const uint8_t> data, size_t offset) {
+  return data[offset];
+}
+
+[[deprecated("Use version with ArrayView")]]
 inline uint8_t Get8(const void* memory, size_t offset) {
   return static_cast<const uint8_t*>(memory)[offset];
 }
 
-inline void SetBE16(void* memory, uint16_t v) {
+inline void SetBE16(ArrayView<uint8_t> data, uint16_t v) {
+  RTC_DCHECK_GE(data.size(), sizeof(v));
+  uint16_t val = htobe16(v);
+  memcpy(data.data(), &val, sizeof(val));
+}
+
+[[deprecated("Use version with ArrayView")]] inline void SetBE16(void* memory,
+                                                                 uint16_t v) {
   uint16_t val = htobe16(v);
   memcpy(memory, &val, sizeof(val));
 }
 
+inline void SetBE32(ArrayView<uint8_t> data, uint32_t v) {
+  RTC_DCHECK_GE(data.size(), sizeof(v));
+  uint32_t val = htobe32(v);
+  memcpy(data.data(), &val, sizeof(val));
+}
+
+// TODO: issues.webrtc.org/484070270 - deprecate once Chromium usage is gone.
+// [[deprecated("Use version with ArrayView")]]
 inline void SetBE32(void* memory, uint32_t v) {
   uint32_t val = htobe32(v);
   memcpy(memory, &val, sizeof(val));
 }
 
-inline void SetBE64(void* memory, uint64_t v) {
+inline void SetBE64(ArrayView<uint8_t> data, uint64_t v) {
+  RTC_DCHECK_GE(data.size(), sizeof(v));
+  uint64_t val = htobe64(v);
+  memcpy(data.data(), &val, sizeof(val));
+}
+
+[[deprecated("Use version with ArrayView")]] inline void SetBE64(void* memory,
+                                                                 uint64_t v) {
   uint64_t val = htobe64(v);
   memcpy(memory, &val, sizeof(val));
 }
 
-inline uint16_t GetBE16(const void* memory) {
+inline uint16_t GetBE16(ArrayView<const uint8_t> data) {
+  uint16_t val;
+  RTC_DCHECK_GE(data.size(), sizeof(val));
+  memcpy(&val, data.data(), sizeof(val));
+  return be16toh(val);
+}
+
+[[deprecated("Use version with ArrayView")]] inline uint16_t GetBE16(
+    const void* memory) {
   uint16_t val;
   memcpy(&val, memory, sizeof(val));
   return be16toh(val);
 }
 
+inline uint32_t GetBE32(ArrayView<const uint8_t> data) {
+  uint32_t val;
+  RTC_DCHECK_GE(data.size(), sizeof(val));
+  memcpy(&val, data.data(), sizeof(val));
+  return be32toh(val);
+}
+
+// TODO: issues.webrtc.org/484070270 - deprecate once Chromium usage is gone.
+// [[deprecated("Use version with ArrayView")]]
 inline uint32_t GetBE32(const void* memory) {
   uint32_t val;
   memcpy(&val, memory, sizeof(val));
   return be32toh(val);
 }
 
-inline uint64_t GetBE64(const void* memory) {
+inline uint64_t GetBE64(ArrayView<const uint8_t> data) {
+  uint64_t val;
+  RTC_DCHECK_GE(data.size(), sizeof(val));
+  memcpy(&val, data.data(), sizeof(val));
+  return be64toh(val);
+}
+
+[[deprecated("Use version with ArrayView")]] inline uint64_t GetBE64(
+    const void* memory) {
   uint64_t val;
   memcpy(&val, memory, sizeof(val));
   return be64toh(val);
 }
 
-inline void SetLE16(void* memory, uint16_t v) {
+inline void SetLE16(ArrayView<uint8_t> data, uint16_t v) {
+  RTC_DCHECK_GE(data.size(), sizeof(v));
+  uint16_t val = htole16(v);
+  memcpy(data.data(), &val, sizeof(val));
+}
+
+[[deprecated("Use version with ArrayView")]] inline void SetLE16(void* memory,
+                                                                 uint16_t v) {
   uint16_t val = htole16(v);
   memcpy(memory, &val, sizeof(val));
 }
 
-inline void SetLE32(void* memory, uint32_t v) {
+inline void SetLE32(ArrayView<uint8_t> data, uint32_t v) {
+  RTC_DCHECK_GE(data.size(), sizeof(v));
+  uint32_t val = htole32(v);
+  memcpy(data.data(), &val, sizeof(val));
+}
+
+[[deprecated("Use version with ArrayView")]] inline void SetLE32(void* memory,
+                                                                 uint32_t v) {
   uint32_t val = htole32(v);
   memcpy(memory, &val, sizeof(val));
 }
 
-inline void SetLE64(void* memory, uint64_t v) {
+inline void SetLE64(ArrayView<uint8_t> data, uint64_t v) {
+  RTC_DCHECK_GE(data.size(), sizeof(v));
+  uint64_t val = htole64(v);
+  memcpy(data.data(), &val, sizeof(val));
+}
+
+[[deprecated("Use version with ArrayView")]] inline void SetLE64(void* memory,
+                                                                 uint64_t v) {
   uint64_t val = htole64(v);
   memcpy(memory, &val, sizeof(val));
 }
 
-inline uint16_t GetLE16(const void* memory) {
+inline uint16_t GetLE16(ArrayView<const uint8_t> data) {
+  uint16_t val;
+  RTC_DCHECK_GE(data.size(), sizeof(val));
+  memcpy(&val, data.data(), sizeof(val));
+  return le16toh(val);
+}
+
+[[deprecated("Use version with ArrayView")]] inline uint16_t GetLE16(
+    const void* memory) {
   uint16_t val;
   memcpy(&val, memory, sizeof(val));
   return le16toh(val);
 }
 
-inline uint32_t GetLE32(const void* memory) {
+inline uint32_t GetLE32(ArrayView<const uint8_t> data) {
+  uint32_t val;
+  RTC_DCHECK_GE(data.size(), sizeof(val));
+  memcpy(&val, data.data(), sizeof(val));
+  return le32toh(val);
+}
+
+[[deprecated("Use version with ArrayView")]] inline uint32_t GetLE32(
+    const void* memory) {
   uint32_t val;
   memcpy(&val, memory, sizeof(val));
   return le32toh(val);
 }
 
-inline uint64_t GetLE64(const void* memory) {
+inline uint64_t GetLE64(ArrayView<const uint8_t> data) {
+  uint64_t val;
+  RTC_DCHECK_GE(data.size(), sizeof(val));
+  memcpy(&val, data.data(), sizeof(val));
+  return le64toh(val);
+}
+
+[[deprecated("Use version with ArrayView")]] inline uint64_t GetLE64(
+    const void* memory) {
   uint64_t val;
   memcpy(&val, memory, sizeof(val));
   return le64toh(val);
diff --git a/rtc_base/byte_order_unittest.cc b/rtc_base/byte_order_unittest.cc
index 8c1e10edf2..75a370c7b8 100644
--- a/rtc_base/byte_order_unittest.cc
+++ b/rtc_base/byte_order_unittest.cc
@@ -14,6 +14,7 @@
 
 #include <cstdint>
 
+#include "api/array_view.h"
 #include "test/gtest.h"
 
 namespace webrtc {
@@ -21,27 +22,28 @@ namespace webrtc {
 // Test memory set functions put values into memory in expected order.
 TEST(ByteOrderTest, TestSet) {
   uint8_t buf[8] = {0u, 0u, 0u, 0u, 0u, 0u, 0u, 0u};
-  Set8(buf, 0, 0xfb);
-  Set8(buf, 1, 0x12);
+  ArrayView<uint8_t> av(buf);
+  Set8(av, 0, 0xfb);
+  Set8(av, 1, 0x12);
   EXPECT_EQ(0xfb, buf[0]);
   EXPECT_EQ(0x12, buf[1]);
-  SetBE16(buf, 0x1234);
+  SetBE16(av, 0x1234);
   EXPECT_EQ(0x12, buf[0]);
   EXPECT_EQ(0x34, buf[1]);
-  SetLE16(buf, 0x1234);
+  SetLE16(av, 0x1234);
   EXPECT_EQ(0x34, buf[0]);
   EXPECT_EQ(0x12, buf[1]);
-  SetBE32(buf, 0x12345678);
+  SetBE32(av, 0x12345678);
   EXPECT_EQ(0x12, buf[0]);
   EXPECT_EQ(0x34, buf[1]);
   EXPECT_EQ(0x56, buf[2]);
   EXPECT_EQ(0x78, buf[3]);
-  SetLE32(buf, 0x12345678);
+  SetLE32(av, 0x12345678);
   EXPECT_EQ(0x78, buf[0]);
   EXPECT_EQ(0x56, buf[1]);
   EXPECT_EQ(0x34, buf[2]);
   EXPECT_EQ(0x12, buf[3]);
-  SetBE64(buf, UINT64_C(0x0123456789abcdef));
+  SetBE64(av, UINT64_C(0x0123456789abcdef));
   EXPECT_EQ(0x01, buf[0]);
   EXPECT_EQ(0x23, buf[1]);
   EXPECT_EQ(0x45, buf[2]);
@@ -50,7 +52,7 @@ TEST(ByteOrderTest, TestSet) {
   EXPECT_EQ(0xab, buf[5]);
   EXPECT_EQ(0xcd, buf[6]);
   EXPECT_EQ(0xef, buf[7]);
-  SetLE64(buf, UINT64_C(0x0123456789abcdef));
+  SetLE64(av, UINT64_C(0x0123456789abcdef));
   EXPECT_EQ(0xef, buf[0]);
   EXPECT_EQ(0xcd, buf[1]);
   EXPECT_EQ(0xab, buf[2]);
@@ -64,6 +66,7 @@ TEST(ByteOrderTest, TestSet) {
 // Test memory get functions get values from memory in expected order.
 TEST(ByteOrderTest, TestGet) {
   uint8_t buf[8];
+  ArrayView<const uint8_t> av(buf);
   buf[0] = 0x01u;
   buf[1] = 0x23u;
   buf[2] = 0x45u;
@@ -72,14 +75,14 @@ TEST(ByteOrderTest, TestGet) {
   buf[5] = 0xabu;
   buf[6] = 0xcdu;
   buf[7] = 0xefu;
-  EXPECT_EQ(0x01u, Get8(buf, 0));
-  EXPECT_EQ(0x23u, Get8(buf, 1));
-  EXPECT_EQ(0x0123u, GetBE16(buf));
-  EXPECT_EQ(0x2301u, GetLE16(buf));
-  EXPECT_EQ(0x01234567u, GetBE32(buf));
-  EXPECT_EQ(0x67452301u, GetLE32(buf));
-  EXPECT_EQ(UINT64_C(0x0123456789abcdef), GetBE64(buf));
-  EXPECT_EQ(UINT64_C(0xefcdab8967452301), GetLE64(buf));
+  EXPECT_EQ(0x01u, Get8(av, 0));
+  EXPECT_EQ(0x23u, Get8(av, 1));
+  EXPECT_EQ(0x0123u, GetBE16(av));
+  EXPECT_EQ(0x2301u, GetLE16(av));
+  EXPECT_EQ(0x01234567u, GetBE32(av));
+  EXPECT_EQ(0x67452301u, GetLE32(av));
+  EXPECT_EQ(UINT64_C(0x0123456789abcdef), GetBE64(av));
+  EXPECT_EQ(UINT64_C(0xefcdab8967452301), GetLE64(av));
 }
 
 }  // namespace webrtc

From e952c943e3ae1cf1ab8d96a4c349ffd1a570ee57 Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Fri, 13 Feb 2026 20:06:47 -0800
Subject: [PATCH 051/472] Update WebRTC code version (2026-02-14T04:06:46).

Bug: None
Change-Id: I68781ea3c77033a47a9d8cbe8f93523ac264348a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449280
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#46904}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index 4d2f88368d..d5df230d52 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-02-12T04:07:42";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-02-14T04:06:46";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From 5e1c8156699645f78141c4cbe5b30555a2b5a0d8 Mon Sep 17 00:00:00 2001
From: Bjorn Terelius <terelius@webrtc.org>
Date: Fri, 13 Feb 2026 22:27:39 +0100
Subject: [PATCH 052/472] Remove deprecated --no_auth flag from DEPS
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Bug: None
Change-Id: Ida529f202eba45935cb557ac70107c77c1056692
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449200
Auto-Submit: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#46905}
---
 DEPS | 7 -------
 1 file changed, 7 deletions(-)

diff --git a/DEPS b/DEPS
index 5e688ce7c7..ea80f58948 100644
--- a/DEPS
+++ b/DEPS
@@ -1145,7 +1145,6 @@ hooks = [
     'action': [ 'python3',
                 'src/third_party/depot_tools/download_from_google_storage.py',
                 '--no_resume',
-                '--no_auth',
                 '--bucket', 'chromium-browser-clang/ciopfs',
                 '-s', 'src/build/ciopfs.sha1',
     ]
@@ -1201,7 +1200,6 @@ hooks = [
     'action': [ 'python3',
                 'src/third_party/depot_tools/download_from_google_storage.py',
                 '--no_resume',
-                '--no_auth',
                 '--bucket', 'chromium-browser-clang',
                 '-s', 'src/tools/clang/dsymutil/bin/dsymutil.arm64.sha1',
                 '-o', 'src/tools/clang/dsymutil/bin/dsymutil',
@@ -1214,7 +1212,6 @@ hooks = [
     'action': [ 'python3',
                 'src/third_party/depot_tools/download_from_google_storage.py',
                 '--no_resume',
-                '--no_auth',
                 '--bucket', 'chromium-browser-clang',
                 '-s', 'src/tools/clang/dsymutil/bin/dsymutil.x64.sha1',
                 '-o', 'src/tools/clang/dsymutil/bin/dsymutil',
@@ -1228,7 +1225,6 @@ hooks = [
     'action': [ 'python3',
                 'src/third_party/depot_tools/download_from_google_storage.py',
                 '--no_resume',
-                '--no_auth',
                 '--bucket', 'chromium-browser-clang/rc',
                 '-s', 'src/build/toolchain/win/rc/win/rc.exe.sha1',
     ],
@@ -1240,7 +1236,6 @@ hooks = [
     'action': [ 'python3',
                 'src/third_party/depot_tools/download_from_google_storage.py',
                 '--no_resume',
-                '--no_auth',
                 '--bucket', 'chromium-browser-clang/rc',
                 '-s', 'src/build/toolchain/win/rc/mac/rc.sha1',
     ],
@@ -1252,7 +1247,6 @@ hooks = [
     'action': [ 'python3',
                 'src/third_party/depot_tools/download_from_google_storage.py',
                 '--no_resume',
-                '--no_auth',
                 '--bucket', 'chromium-browser-clang/rc',
                 '-s', 'src/build/toolchain/win/rc/linux64/rc.sha1',
     ],
@@ -1264,7 +1258,6 @@ hooks = [
                '--directory',
                '--recursive',
                '--num_threads=10',
-               '--no_auth',
                '--quiet',
                '--bucket', 'chromium-webrtc-resources',
                'src/resources'],

From 25713d97ffa88b4b478981c333a0d88444295b08 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Sat, 14 Feb 2026 11:10:32 +0100
Subject: [PATCH 053/472] Separate transceiver references from the stats info
 struct

Extract the transceiver and receivers members from the
RtpTransceiverStatsInfo struct into a new TransceiverReferences
struct.

The RtpTransceiverStatsInfo structure is used to transport statistics
data between the signaling, worker, and network threads. Because the
transceiver and receiver objects are signaling-thread-specific, they
previously had to be manually cleared from the struct before it was
posted to the network thread to avoid cross-thread access.

Separating these references into a distinct container makes the data
ownership and thread boundaries more explicit. This refactoring
removes the need for manual pointer nullification and reduces the
risk of accidental cross-thread access during the statistics
gathering process.

Bug: none
Change-Id: I29a5fe77074a139ee9437dbca52b5450c51fa673
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449040
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46906}
---
 pc/rtc_stats_collector.cc          | 93 ++++++++++++++++++------------
 pc/rtc_stats_collector.h           | 18 +++++-
 pc/rtc_stats_collector_unittest.cc |  1 +
 3 files changed, 72 insertions(+), 40 deletions(-)

diff --git a/pc/rtc_stats_collector.cc b/pc/rtc_stats_collector.cc
index 8c8a865de3..15cb864621 100644
--- a/pc/rtc_stats_collector.cc
+++ b/pc/rtc_stats_collector.cc
@@ -1355,11 +1355,14 @@ void RTCStatsCollector::GetStatsReportInternal(
   // `ProducePartialResultsOnSignalingThread`.
   auto worker_task = PrepareTransceiverStatsInfosAndCallStats_s_w();
 
-  auto signaling_task = [this](StatsGatheringResults results) mutable {
+  auto signaling_task = [this](WorkerThreadResult worker_result) mutable {
     RTC_DCHECK_RUN_ON(signaling_thread_);
     // Create the initial `partial_report_` for the gathering operation.
-    ProducePartialResultsOnSignalingThread(results.transceiver_stats_infos,
-                                           results.audio_device_stats);
+    ProducePartialResultsOnSignalingThread(
+        worker_result.results.transceiver_stats_infos,
+        worker_result.transceiver_references,
+        worker_result.results.audio_device_stats);
+
     Timestamp timestamp = collection_context_->partial_report->timestamp();
     std::set<std::string> transport_names;
     auto sctp_transport_name = pc_->sctp_transport_name();
@@ -1367,22 +1370,16 @@ void RTCStatsCollector::GetStatsReportInternal(
       transport_names.emplace(std::move(*sctp_transport_name));
     }
 
-    for (const auto& info : results.transceiver_stats_infos) {
+    for (const auto& info : worker_result.results.transceiver_stats_infos) {
       if (info.transport_name)
         transport_names.insert(*info.transport_name);
     }
 
-    // Clear references to signaling thread objects (and proxies) before
-    // attempting to post to the network thread. We're done with these objects.
-    for (RtpTransceiverStatsInfo& stats : results.transceiver_stats_infos) {
-      stats.transceiver = nullptr;
-      stats.receivers.clear();
-    }
-
     network_thread_->PostTask(SafeTask(
-        network_safety_, [this, transport_names = std::move(transport_names),
-                          timestamp, signaling_flag = signaling_safety_,
-                          results = std::move(results)]() mutable {
+        network_safety_,
+        [this, transport_names = std::move(transport_names), timestamp,
+         signaling_flag = signaling_safety_,
+         results = std::move(worker_result.results)]() mutable {
           ProducePartialResultsOnNetworkThread(
               std::move(signaling_flag), timestamp, std::move(transport_names),
               std::move(results));
@@ -1392,12 +1389,13 @@ void RTCStatsCollector::GetStatsReportInternal(
   worker_thread_->PostTask(SafeTask(
       worker_safety_, [this, worker_task = std::move(worker_task),
                        signaling_task = std::move(signaling_task)]() mutable {
-        auto results = std::move(worker_task)();
-        signaling_thread_->PostTask(SafeTask(
-            signaling_safety_, [signaling_task = std::move(signaling_task),
-                                results = std::move(results)]() mutable {
-              signaling_task(std::move(results));
-            }));
+        auto worker_result = std::move(worker_task)();
+        signaling_thread_->PostTask(
+            SafeTask(signaling_safety_,
+                     [signaling_task = std::move(signaling_task),
+                      worker_result = std::move(worker_result)]() mutable {
+                       signaling_task(std::move(worker_result));
+                     }));
       }));
 }
 
@@ -1424,6 +1422,7 @@ void RTCStatsCollector::CancelPendingRequestAndGetShutdownTasks(
 
 void RTCStatsCollector::ProducePartialResultsOnSignalingThread(
     const std::vector<RtpTransceiverStatsInfo>& transceiver_stats_infos,
+    const std::vector<TransceiverReferences>& transceiver_references,
     const std::optional<AudioDeviceModule::Stats>& audio_device_stats) {
   RTC_DCHECK_RUN_ON(signaling_thread_);
   RTC_DCHECK(collection_context_);
@@ -1432,16 +1431,19 @@ void RTCStatsCollector::ProducePartialResultsOnSignalingThread(
 
   ProducePartialResultsOnSignalingThreadImpl(
       collection_context_->partial_report->timestamp(), transceiver_stats_infos,
-      audio_device_stats, collection_context_->partial_report.get());
+      transceiver_references, audio_device_stats,
+      collection_context_->partial_report.get());
 }
 
 void RTCStatsCollector::ProducePartialResultsOnSignalingThreadImpl(
     Timestamp timestamp,
     const std::vector<RtpTransceiverStatsInfo>& transceiver_stats_infos,
+    const std::vector<TransceiverReferences>& transceiver_references,
     const std::optional<AudioDeviceModule::Stats>& audio_device_stats,
     RTCStatsReport* partial_report) {
   RTC_DCHECK_RUN_ON(signaling_thread_);
-  ProduceMediaSourceStats_s(timestamp, transceiver_stats_infos, partial_report);
+  ProduceMediaSourceStats_s(timestamp, transceiver_stats_infos,
+                            transceiver_references, partial_report);
   ProducePeerConnectionStats_s(timestamp, partial_report);
   ProduceAudioPlayoutStats_s(timestamp, audio_device_stats, partial_report);
 }
@@ -1711,12 +1713,17 @@ void RTCStatsCollector::ProduceIceCandidateAndPairStats_s(
 void RTCStatsCollector::ProduceMediaSourceStats_s(
     Timestamp timestamp,
     const std::vector<RtpTransceiverStatsInfo>& transceiver_stats_infos,
+    const std::vector<TransceiverReferences>& transceiver_references,
     RTCStatsReport* report) const {
   RTC_DCHECK_RUN_ON(signaling_thread_);
   Thread::ScopedDisallowBlockingCalls no_blocking_calls;
 
-  for (const RtpTransceiverStatsInfo& transceiver_stats_info :
-       transceiver_stats_infos) {
+  RTC_DCHECK_EQ(transceiver_stats_infos.size(), transceiver_references.size());
+  for (size_t i = 0; i < transceiver_stats_infos.size(); ++i) {
+    const RtpTransceiverStatsInfo& transceiver_stats_info =
+        transceiver_stats_infos[i];
+    const TransceiverReferences& refs = transceiver_references[i];
+
     // The transceiver will still exist but in a stopped state after pc.close().
     if (transceiver_stats_info.current_direction ==
         RtpTransceiverDirection::kStopped) {
@@ -1726,7 +1733,7 @@ void RTCStatsCollector::ProduceMediaSourceStats_s(
     const TrackMediaInfoMap& track_media_info_map =
         *transceiver_stats_info.track_media_info_map;
 
-    for (const auto& sender : transceiver_stats_info.transceiver->senders()) {
+    for (const auto& sender : refs.transceiver->senders()) {
       const auto& sender_internal = sender->internal();
       const auto& track = sender_internal->track();
       if (!track)
@@ -2254,11 +2261,12 @@ RTCStatsCollector::PrepareTransportCertificateStats_n(
   return transport_cert_stats;
 }
 
-absl::AnyInvocable<RTCStatsCollector::StatsGatheringResults()>
+absl::AnyInvocable<RTCStatsCollector::WorkerThreadResult()>
 RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w() {
   RTC_DCHECK_RUN_ON(signaling_thread_);
 
   std::vector<RtpTransceiverStatsInfo> transceiver_stats_infos;
+  std::vector<TransceiverReferences> transceiver_references;
   // These are used to invoke GetStats for all the media channels together in
   // one worker thread hop.
   std::map<VoiceMediaSendChannelInterface*, VoiceMediaSendInfo>
@@ -2276,13 +2284,15 @@ RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w() {
     RtpTransceiver* transceiver = transceiver_proxy->internal();
 
     RtpTransceiverStatsInfo stats{
-        .transceiver = scoped_refptr<RtpTransceiver>(transceiver),
         .media_type = transceiver->media_type(),
         .mid = transceiver->mid(),
         .transport_name = transceiver->transport_name(),
         .current_direction = transceiver->current_direction(),
         .has_channel = transceiver->HasChannel()};
 
+    TransceiverReferences refs{.transceiver =
+                                   scoped_refptr<RtpTransceiver>(transceiver)};
+
     for (const auto& sender : transceiver->senders()) {
       stats.sender_infos.push_back(
           {.ssrc = sender->ssrc(),
@@ -2294,10 +2304,10 @@ RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w() {
           {.track_id = receiver->track() ? receiver->track()->id() : "",
            .attachment_id = receiver->internal()->AttachmentId(),
            .media_type = receiver->media_type()});
-      stats.receivers.push_back(
+      refs.receivers.push_back(
           scoped_refptr<RtpReceiverInternal>(receiver->internal()));
     }
-    stats.has_receivers = !stats.receivers.empty();
+    stats.has_receivers = !refs.receivers.empty();
 
     if (stats.has_channel) {
       if (stats.media_type == MediaType::AUDIO) {
@@ -2316,6 +2326,7 @@ RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w() {
     }
 
     transceiver_stats_infos.push_back(std::move(stats));
+    transceiver_references.push_back(std::move(refs));
   }
 
   // Embed the collected information into this lambda which will run on the
@@ -2323,14 +2334,17 @@ RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w() {
   // as GetCallStats(). At the same time we construct the TrackMediaInfoMaps,
   // which also needs info from the worker thread.
   return [this, transceiver_stats_infos = std::move(transceiver_stats_infos),
+          transceiver_references = std::move(transceiver_references),
           voice_send_stats = std::move(voice_send_stats),
           voice_receive_stats = std::move(voice_receive_stats),
           video_send_stats = std::move(video_send_stats),
           video_receive_stats = std::move(video_receive_stats)]() mutable {
     Thread::ScopedDisallowBlockingCalls no_blocking_calls;
 
-    StatsGatheringResults results = {.transceiver_stats_infos =
-                                         std::move(transceiver_stats_infos)};
+    WorkerThreadResult worker_result;
+    worker_result.results.transceiver_stats_infos =
+        std::move(transceiver_stats_infos);
+    worker_result.transceiver_references = std::move(transceiver_references);
 
     for (auto& pair : voice_send_stats) {
       if (!pair.first->GetStats(&pair.second)) {
@@ -2357,7 +2371,12 @@ RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w() {
     // Create the TrackMediaInfoMap for each transceiver stats object
     // and keep track of whether we have at least one audio receiver.
     bool has_audio_receiver = false;
-    for (auto& stats : results.transceiver_stats_infos) {
+    RTC_DCHECK_EQ(worker_result.results.transceiver_stats_infos.size(),
+                  worker_result.transceiver_references.size());
+    for (size_t i = 0; i < worker_result.results.transceiver_stats_infos.size();
+         ++i) {
+      auto& stats = worker_result.results.transceiver_stats_infos[i];
+      auto& refs = worker_result.transceiver_references[i];
       // The transceiver will still exist but in a stopped state after
       // pc.close().
       if (stats.current_direction == RtpTransceiverDirection::kStopped) {
@@ -2365,14 +2384,14 @@ RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w() {
       }
 
       std::vector<RtpParameters> receiver_parameters;
-      for (const auto& receiver : stats.receivers) {
+      for (const auto& receiver : refs.receivers) {
         receiver_parameters.push_back(receiver->GetParameters());
       }
 
       std::optional<VoiceMediaInfo> voice_media_info;
       std::optional<VideoMediaInfo> video_media_info;
       if (stats.has_channel) {
-        auto& transceiver = stats.transceiver;
+        auto& transceiver = refs.transceiver;
         if (stats.media_type == MediaType::AUDIO) {
           auto voice_send_channel = transceiver->voice_media_send_channel();
           auto voice_receive_channel =
@@ -2399,10 +2418,10 @@ RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w() {
       }
     }
 
-    results.call_stats = pc_->GetCallStats();
-    results.audio_device_stats =
+    worker_result.results.call_stats = pc_->GetCallStats();
+    worker_result.results.audio_device_stats =
         has_audio_receiver ? pc_->GetAudioDeviceStats() : std::nullopt;
-    return results;
+    return worker_result;
   };
 }
 
diff --git a/pc/rtc_stats_collector.h b/pc/rtc_stats_collector.h
index 4276db6348..5e9bbfe463 100644
--- a/pc/rtc_stats_collector.h
+++ b/pc/rtc_stats_collector.h
@@ -61,19 +61,24 @@ class RtpReceiverInternal;
 // If a BaseChannel is not available (e.g., if signaling has not started),
 // then `mid` and `transport_name` will be null.
 struct RtpTransceiverStatsInfo {
-  scoped_refptr<RtpTransceiver> transceiver;
   const MediaType media_type;
   const std::optional<std::string> mid;
   std::optional<std::string> transport_name;
   std::vector<TrackMediaInfoMap::RtpSenderSignalInfo> sender_infos;
   std::vector<TrackMediaInfoMap::RtpReceiverSignalInfo> receiver_infos;
-  std::vector<scoped_refptr<RtpReceiverInternal>> receivers;
   std::unique_ptr<TrackMediaInfoMap> track_media_info_map;
   const std::optional<RtpTransceiverDirection> current_direction;
   bool has_receivers = false;
   const bool has_channel;
 };
 
+// References to objects used on the signaling and worker threads for populating
+// RtpTransceiverStatsInfo but must always be released on the signaling thread
+struct TransceiverReferences {
+  scoped_refptr<RtpTransceiver> transceiver;
+  std::vector<scoped_refptr<RtpReceiverInternal>> receivers;
+};
+
 // All public methods of the collector are to be called on the signaling thread.
 // Stats are gathered on the signaling, worker and network threads
 // asynchronously. The callback is invoked on the signaling thread. Resulting
@@ -134,6 +139,7 @@ class RTCStatsCollector {
   virtual void ProducePartialResultsOnSignalingThreadImpl(
       Timestamp timestamp,
       const std::vector<RtpTransceiverStatsInfo>& transceiver_stats_infos,
+      const std::vector<TransceiverReferences>& transceiver_references,
       const std::optional<AudioDeviceModule::Stats>& audio_device_stats,
       RTCStatsReport* partial_report);
 
@@ -153,6 +159,10 @@ class RTCStatsCollector {
     std::optional<AudioDeviceModule::Stats> audio_device_stats;
   };
 
+  struct WorkerThreadResult {
+    StatsGatheringResults results;
+    std::vector<TransceiverReferences> transceiver_references;
+  };
   struct CollectionContext;
   class RequestInfo {
    public:
@@ -221,6 +231,7 @@ class RTCStatsCollector {
   void ProduceMediaSourceStats_s(
       Timestamp timestamp,
       const std::vector<RtpTransceiverStatsInfo>& transceiver_stats_infos,
+      const std::vector<TransceiverReferences>& transceiver_references,
       RTCStatsReport* report) const;
   // Produces `RTCPeerConnectionStats`.
   void ProducePeerConnectionStats_s(Timestamp timestamp,
@@ -267,12 +278,13 @@ class RTCStatsCollector {
   // Prepares the transceiver stats infos and call stats.
   // Returns a callback that should be executed on the worker thread to populate
   // the stats.
-  absl::AnyInvocable<StatsGatheringResults()>
+  absl::AnyInvocable<WorkerThreadResult()>
   PrepareTransceiverStatsInfosAndCallStats_s_w();
 
   // Stats gathering on a particular thread.
   void ProducePartialResultsOnSignalingThread(
       const std::vector<RtpTransceiverStatsInfo>& transceiver_stats_infos,
+      const std::vector<TransceiverReferences>& transceiver_references,
       const std::optional<AudioDeviceModule::Stats>& audio_device_stats);
   void ProducePartialResultsOnNetworkThread(
       scoped_refptr<PendingTaskSafetyFlag> signaling_safety,
diff --git a/pc/rtc_stats_collector_unittest.cc b/pc/rtc_stats_collector_unittest.cc
index 3fb8e03a73..480ab705b4 100644
--- a/pc/rtc_stats_collector_unittest.cc
+++ b/pc/rtc_stats_collector_unittest.cc
@@ -3980,6 +3980,7 @@ class FakeRTCStatsCollector final : public RTCStatsCollector {
   void ProducePartialResultsOnSignalingThreadImpl(
       Timestamp timestamp,
       const std::vector<RtpTransceiverStatsInfo>& transceiver_stats_infos,
+      const std::vector<TransceiverReferences>& transceiver_references,
       const std::optional<AudioDeviceModule::Stats>& audio_device_stats,
       RTCStatsReport* partial_report) override {
     EXPECT_TRUE(signaling_thread_->IsCurrent());

From f382b0d3841321a9f9dba425046f17c85bc3666b Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Sat, 14 Feb 2026 20:07:51 -0800
Subject: [PATCH 054/472] Update WebRTC code version (2026-02-15T04:07:17).

Bug: None
Change-Id: I994533c9ead96bc9c88428e2f36b65468576dbd8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449400
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#46907}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index d5df230d52..7d05117193 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-02-14T04:06:46";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-02-15T04:07:17";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From d9ace3fe414023e109e876d23f7280270cabdc6d Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Sat, 14 Feb 2026 21:40:46 +0100
Subject: [PATCH 055/472] Deprecated ArrayView::subview

Replace remaining usages of subview with subspan

Bug: webrtc:439801349
Change-Id: I205a7de55c24b2f77356095ee736d7863c70c2ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449360
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46908}
---
 api/array_view.h                              | 12 +++----
 api/array_view_unittest.cc                    |  6 ++++
 call/rtx_receive_stream.cc                    |  2 +-
 call/rtx_receive_stream_unittest.cc           |  2 +-
 .../scalability_structure_key_svc_unittest.cc |  4 +--
 p2p/base/async_stun_tcp_socket.cc             |  2 +-
 rtc_base/buffer.h                             |  4 +--
 rtc_base/buffer_unittest.cc                   |  4 +--
 test/fuzzers/fuzz_data_helper.h               |  2 +-
 .../bwe_integration_tests/scream_test.cc      | 36 +++++++++----------
 test/video_codec_tester.cc                    |  2 +-
 11 files changed, 40 insertions(+), 36 deletions(-)

diff --git a/api/array_view.h b/api/array_view.h
index 0b6e8649af..7ff065f892 100644
--- a/api/array_view.h
+++ b/api/array_view.h
@@ -298,17 +298,15 @@ class ArrayView final : public array_view_internal::ArrayViewBase<T, Size> {
     return ArrayView<T>(this->data() + offset, count);
   }
 
-  // Prefer to use `subspan` instead of `subview` in new code.
-  // TODO: bugs.webrtc.org/439801349 - make deprecated when all usages in
-  // WebRTC and chromium are migrated to subspan.
+  [[deprecated("Use subspan")]]
   ArrayView<T> subview(size_t offset, size_t size) const {
-    return offset < this->size()
-               ? ArrayView<T>(this->data() + offset,
-                              std::min(size, this->size() - offset))
+    return offset <= this->size()
+               ? subspan(offset, std::min(size, this->size() - offset))
                : ArrayView<T>();
   }
+  [[deprecated("Use subspan")]]
   ArrayView<T> subview(size_t offset) const {
-    return subview(offset, this->size());
+    return offset <= this->size() ? subspan(offset) : ArrayView<T>();
   }
 };
 
diff --git a/api/array_view_unittest.cc b/api/array_view_unittest.cc
index da361a85ac..1ab7b6e981 100644
--- a/api/array_view_unittest.cc
+++ b/api/array_view_unittest.cc
@@ -575,6 +575,8 @@ TEST(ArrayViewTest, TestCompare) {
   EXPECT_NE((ArrayView<int, 3>(a)), (ArrayView<int, 2>(a, 2)));
 }
 
+#pragma clang diagnostic push
+#pragma clang diagnostic ignored "-Wdeprecated-declarations"
 TEST(ArrayViewTest, TestSubViewVariable) {
   int a[] = {1, 2, 3};
   ArrayView<int> av(a);
@@ -591,6 +593,7 @@ TEST(ArrayViewTest, TestSubViewVariable) {
   EXPECT_THAT(av.subview(1, 2), ElementsAre(2, 3));
   EXPECT_THAT(av.subview(1, 3), ElementsAre(2, 3));
 }
+#pragma clang diagnostic pop
 
 TEST(ArrayViewTest, TestSubSpanVariable) {
   int a[] = {1, 2, 3};
@@ -613,6 +616,8 @@ TEST(ArrayViewTest, TestSubSpanWithInvalidInput) {
   EXPECT_DEATH_IF_SUPPORTED(av.subspan(1, 3), "");
 }
 
+#pragma clang diagnostic push
+#pragma clang diagnostic ignored "-Wdeprecated-declarations"
 TEST(ArrayViewTest, TestSubViewFixed) {
   int a[] = {1, 2, 3};
   ArrayView<int, 3> av(a);
@@ -629,6 +634,7 @@ TEST(ArrayViewTest, TestSubViewFixed) {
   EXPECT_THAT(av.subview(1, 2), ElementsAre(2, 3));
   EXPECT_THAT(av.subview(1, 3), ElementsAre(2, 3));
 }
+#pragma clang diagnostic pop
 
 TEST(ArrayViewTest, TestReinterpretCastFixedSize) {
   uint8_t bytes[] = {1, 2, 3};
diff --git a/call/rtx_receive_stream.cc b/call/rtx_receive_stream.cc
index a6625e866f..d725d285da 100644
--- a/call/rtx_receive_stream.cc
+++ b/call/rtx_receive_stream.cc
@@ -111,7 +111,7 @@ void RtxReceiveStream::OnRtpPacket(const RtpPacketReceived& rtx_packet) {
   media_packet.set_arrival_time(rtx_packet.arrival_time());
 
   // Skip the RTX payload header.
-  media_packet.SetPayload(payload.subview(kRtxHeaderSize));
+  media_packet.SetPayload(payload.subspan(kRtxHeaderSize));
 
   media_sink_->OnRtpPacket(media_packet);
 }
diff --git a/call/rtx_receive_stream_unittest.cc b/call/rtx_receive_stream_unittest.cc
index 5c53098810..80fd8c96ca 100644
--- a/call/rtx_receive_stream_unittest.cc
+++ b/call/rtx_receive_stream_unittest.cc
@@ -124,7 +124,7 @@ std::map<int, int> PayloadTypeMapping() {
 
 template <typename T>
 ArrayView<T> Truncate(ArrayView<T> a, size_t drop) {
-  return a.subview(0, a.size() - drop);
+  return a.subspan(0, a.size() - drop);
 }
 
 }  // namespace
diff --git a/modules/video_coding/svc/scalability_structure_key_svc_unittest.cc b/modules/video_coding/svc/scalability_structure_key_svc_unittest.cc
index 175b8fc277..145a278234 100644
--- a/modules/video_coding/svc/scalability_structure_key_svc_unittest.cc
+++ b/modules/video_coding/svc/scalability_structure_key_svc_unittest.cc
@@ -235,9 +235,9 @@ TEST(ScalabilityStructureL3T3KeyTest, ReenablingSpatialLayerTriggersKeyFrame) {
   EXPECT_EQ(frames[14].temporal_id, 0);
   EXPECT_EQ(frames[15].temporal_id, 0);
   auto all_frames = MakeArrayView(frames.data(), frames.size());
-  EXPECT_TRUE(wrapper.FrameReferencesAreValid(all_frames.subview(0, 13)));
+  EXPECT_TRUE(wrapper.FrameReferencesAreValid(all_frames.subspan(0, 13)));
   // Frames starting from the frame#13 should not reference any earlier frames.
-  EXPECT_TRUE(wrapper.FrameReferencesAreValid(all_frames.subview(13)));
+  EXPECT_TRUE(wrapper.FrameReferencesAreValid(all_frames.subspan(13)));
 }
 
 }  // namespace
diff --git a/p2p/base/async_stun_tcp_socket.cc b/p2p/base/async_stun_tcp_socket.cc
index aeb67141c1..3344f5a3b1 100644
--- a/p2p/base/async_stun_tcp_socket.cc
+++ b/p2p/base/async_stun_tcp_socket.cc
@@ -118,7 +118,7 @@ size_t AsyncStunTCPSocket::ProcessInput(ArrayView<const uint8_t> data) {
     }
 
     ReceivedIpPacket received_packet(
-        data.subview(processed_bytes, expected_pkt_len), remote_addr,
+        data.subspan(processed_bytes, expected_pkt_len), remote_addr,
         env_.clock().CurrentTime());
     NotifyPacketReceived(received_packet);
     processed_bytes += actual_length;
diff --git a/rtc_base/buffer.h b/rtc_base/buffer.h
index 161cb76fe4..81d84eefe6 100644
--- a/rtc_base/buffer.h
+++ b/rtc_base/buffer.h
@@ -300,7 +300,7 @@ class BufferT {
     static_assert(sizeof(T) == sizeof(U), "");
     ArrayView<const U> source(data, size);
     ArrayView<T> destination =
-        ArrayView<T>(data_.get(), capacity_).subview(size_, size);
+        ArrayView<T>(data_.get(), capacity_).subspan(size_, size);
     absl::c_copy(source, destination.begin());
     size_ = new_size;
     RTC_DCHECK(IsConsistent());
@@ -442,7 +442,7 @@ class BufferT {
   void ZeroTrailingData(size_t count) {
     RTC_DCHECK(IsConsistent());
     RTC_DCHECK_LE(count, capacity_ - size_);
-    ExplicitZeroMemory(MakeArrayView(data(), capacity_).subview(size_));
+    ExplicitZeroMemory(MakeArrayView(data(), capacity_).subspan(size_));
   }
 
   // Precondition for all methods except Clear, operator= and the destructor.
diff --git a/rtc_base/buffer_unittest.cc b/rtc_base/buffer_unittest.cc
index 6e4f7e7a27..58857f486c 100644
--- a/rtc_base/buffer_unittest.cc
+++ b/rtc_base/buffer_unittest.cc
@@ -36,7 +36,7 @@ auto kTestData =
                             0xa, 0xb, 0xc, 0xd, 0xe, 0xf});
 
 uint8_t* TestDataAtIndex(int index) {
-  return ArrayView<uint8_t>(kTestData).subview(index).data();
+  return ArrayView<uint8_t>(kTestData).subspan(index).data();
 }
 
 void TestBuf(const Buffer& b1, size_t size, size_t capacity) {
@@ -127,7 +127,7 @@ TEST(BufferTest, TestSetAndAppendWithUnknownArg) {
   buf.AppendData(TestDataContainer());
   EXPECT_EQ(6u, buf.size());
   EXPECT_EQ(0, memcmp(buf.data(), kTestData.data(), 3));
-  EXPECT_EQ(0, memcmp(ArrayView<uint8_t>(buf).subview(3).data(),
+  EXPECT_EQ(0, memcmp(ArrayView<uint8_t>(buf).subspan(3).data(),
                       kTestData.data(), 3));
   EXPECT_THAT(buf, ElementsAre(0, 1, 2, 0, 1, 2));
 }
diff --git a/test/fuzzers/fuzz_data_helper.h b/test/fuzzers/fuzz_data_helper.h
index c095e19989..b76cb2d93a 100644
--- a/test/fuzzers/fuzz_data_helper.h
+++ b/test/fuzzers/fuzz_data_helper.h
@@ -90,7 +90,7 @@ class FuzzDataHelper {
     }
     const size_t index_to_return = data_ix_;
     data_ix_ += bytes;
-    return data_.subview(index_to_return, bytes);
+    return data_.subspan(index_to_return, bytes);
   }
 
   // If sizeof(T) > BytesLeft then the remaining bytes will be used and the rest
diff --git a/test/peer_scenario/bwe_integration_tests/scream_test.cc b/test/peer_scenario/bwe_integration_tests/scream_test.cc
index 9d33588dbd..a6767305e8 100644
--- a/test/peer_scenario/bwe_integration_tests/scream_test.cc
+++ b/test/peer_scenario/bwe_integration_tests/scream_test.cc
@@ -258,7 +258,7 @@ TEST(ScreamTest, MaybeTest(LinkCapacity600KbpsRtt100msNoEcn)) {
       CreateNetworkPath(s, /*use_dual_pi= */ false,
                         DataRate::KilobitsPerSec(600), TimeDelta::Millis(50));
   SendMediaTestResult result = SendMediaInOneDirection(std::move(params), s);
-  EXPECT_THAT(result.caller().subview(1), Each(AvailableSendBitrateIsBetween(
+  EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
                                               DataRate::KilobitsPerSec(250),
                                               DataRate::KilobitsPerSec(700))));
 }
@@ -275,11 +275,11 @@ TEST(ScreamTest,
   SendMediaTestResult result = SendMediaInOneDirection(std::move(params), s);
   // Stats 2-5s
   EXPECT_THAT(
-      result.caller().subview(1, 3),
+      result.caller().subspan(1, 3),
       Each(AvailableSendBitrateIsBetween(DataRate::KilobitsPerSec(200),
                                          DataRate::KilobitsPerSec(600))));
   // Stats after 9s
-  EXPECT_THAT(result.caller().subview(9), Each(AvailableSendBitrateIsBetween(
+  EXPECT_THAT(result.caller().subspan(9), Each(AvailableSendBitrateIsBetween(
                                               DataRate::KilobitsPerSec(1200),
                                               DataRate::KilobitsPerSec(5000))));
 }
@@ -296,7 +296,7 @@ TEST(ScreamTest, MaybeTest(LinkCapacity600KbpsRtt20msNoEcn)) {
                         DataRate::KilobitsPerSec(600), TimeDelta::Millis(10));
 
   SendMediaTestResult result = SendMediaInOneDirection(std::move(params), s);
-  EXPECT_THAT(result.caller().subview(1), Each(AvailableSendBitrateIsBetween(
+  EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
                                               DataRate::KilobitsPerSec(300),
                                               DataRate::KilobitsPerSec(700))));
 }
@@ -313,7 +313,7 @@ TEST(ScreamTest, MaybeTest(LinkCapacity600KbpsRtt100msEcn)) {
 
   SendMediaTestResult result = SendMediaInOneDirection(std::move(params), s);
   // Allow rampup to take 3s.
-  EXPECT_THAT(result.caller().subview(3), Each(AvailableSendBitrateIsBetween(
+  EXPECT_THAT(result.caller().subspan(3), Each(AvailableSendBitrateIsBetween(
                                               DataRate::KilobitsPerSec(350),
                                               DataRate::KilobitsPerSec(660))));
 }
@@ -382,7 +382,7 @@ TEST(ScreamTest, MaybeTest(LinkCapacity1000KbpsRtt100msEcn)) {
 
   SendMediaTestResult result = SendMediaInOneDirection(std::move(params), s);
   // Ignore result first 2s since ramp up is rather slow at higher RTT.
-  EXPECT_THAT(result.caller().subview(2), Each(AvailableSendBitrateIsBetween(
+  EXPECT_THAT(result.caller().subspan(2), Each(AvailableSendBitrateIsBetween(
                                               DataRate::KilobitsPerSec(600),
                                               DataRate::KilobitsPerSec(1000))));
 }
@@ -399,7 +399,7 @@ TEST(ScreamTest, MaybeTest(LinkCapacity1500KbpsRtt30msNoEcn)) {
                         DataRate::KilobitsPerSec(1500), TimeDelta::Millis(15));
 
   SendMediaTestResult result = SendMediaInOneDirection(std::move(params), s);
-  EXPECT_THAT(result.caller().subview(1), Each(AvailableSendBitrateIsBetween(
+  EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
                                               DataRate::KilobitsPerSec(800),
                                               DataRate::KilobitsPerSec(1900))));
 }
@@ -416,7 +416,7 @@ TEST(ScreamTest, MaybeTest(LinkCapacity2MbpsRtt50msNoEcn)) {
                         DataRate::KilobitsPerSec(2000), TimeDelta::Millis(25));
 
   SendMediaTestResult result = SendMediaInOneDirection(std::move(params), s);
-  EXPECT_THAT(result.caller().subview(1), Each(AvailableSendBitrateIsBetween(
+  EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
                                               DataRate::KilobitsPerSec(1500),
                                               DataRate::KilobitsPerSec(2300))));
 }
@@ -434,7 +434,7 @@ TEST(ScreamTest, MaybeTest(LinkCapacity2MbpsRtt50msEcn)) {
 
   SendMediaTestResult result = SendMediaInOneDirection(std::move(params), s);
 
-  EXPECT_THAT(result.caller().subview(1), Each(AvailableSendBitrateIsBetween(
+  EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
                                               DataRate::KilobitsPerSec(1300),
                                               DataRate::KilobitsPerSec(2100))));
 }
@@ -477,7 +477,7 @@ TEST(ScreamTest, MaybeTest(LinkCapacity2MbpsRtt50msNoEcnWithTwcc)) {
   SendMediaTestResult result = SendMediaInOneDirection(std::move(params), s);
   // BWE rampup is quite slow since feedback is only sent every 90ms
   // approximately.
-  EXPECT_THAT(result.caller().subview(5), Each(AvailableSendBitrateIsBetween(
+  EXPECT_THAT(result.caller().subspan(5), Each(AvailableSendBitrateIsBetween(
                                               DataRate::KilobitsPerSec(1200),
                                               DataRate::KilobitsPerSec(2300))));
 }
@@ -559,7 +559,7 @@ TEST(ScreamTest, MaybeTest(ScreencastSlideChange2Mbit50msRttNoEcn)) {
   SendMediaTestResult result = SendMediaInOneDirection(std::move(params), s);
 
   // Ignore estimate during rampup.
-  EXPECT_THAT(result.caller().subview(1), Each(AvailableSendBitrateIsBetween(
+  EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
                                               DataRate::KilobitsPerSec(1200),
                                               DataRate::KilobitsPerSec(2600))));
 }
@@ -581,7 +581,7 @@ TEST(ScreamTest, MaybeTest(ScreencastSlideChangeRepeatedDelaySpikes)) {
   // Ignore estimate during rampup.
   // TODO: bugs.webrtc.org/447037083 - consider improving resilience by limiting
   // queue delay from one feedback.
-  EXPECT_THAT(result.caller().subview(1), Each(AvailableSendBitrateIsBetween(
+  EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
                                               DataRate::KilobitsPerSec(100),
                                               DataRate::KilobitsPerSec(4000))));
 }
@@ -598,7 +598,7 @@ TEST(ScreamTest, MaybeTest(LinkCapacity1MbitRtt50msWithShortQueuesNoEcn)) {
 
   SendMediaTestResult result = SendMediaInOneDirection(std::move(params), s);
 
-  EXPECT_THAT(result.caller().subview(1), Each(AvailableSendBitrateIsBetween(
+  EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
                                               DataRate::KilobitsPerSec(200),
                                               DataRate::KilobitsPerSec(1100))));
 }
@@ -619,7 +619,7 @@ TEST(ScreamTest,
   ASSERT_GE(GetPacketsLost(result.callee_stats.back()),
             0.05 * GetPacketsSent(result.caller_stats.back()));
   // Ignore estimate during rampup.
-  EXPECT_THAT(result.caller().subview(1), Each(AvailableSendBitrateIsBetween(
+  EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
                                               DataRate::KilobitsPerSec(200),
                                               DataRate::KilobitsPerSec(1100))));
 }
@@ -642,7 +642,7 @@ TEST(ScreamTest, MaybeTest(ReturnLinkWithBurstLoss)) {
   // feedback is eventually received even if feedback packets are lost.
   EXPECT_GT(GetPacketsSent(result.caller_stats.back()),
             GetPacketsSent(result.caller_stats[5]));
-  EXPECT_THAT(result.caller().subview(1), Each(AvailableSendBitrateIsBetween(
+  EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
                                               DataRate::KilobitsPerSec(300),
                                               DataRate::KilobitsPerSec(1100))));
 }
@@ -667,7 +667,7 @@ TEST(ScreamTest, MaybeTest(SendVideoOnlyReturnLinkWithBurstLoss)) {
   // packets to fast if queued too long, BWE drop to a very low value.
   EXPECT_GT(GetPacketsSent(result.caller_stats.back()),
             GetPacketsSent(result.caller_stats[5]));
-  EXPECT_THAT(result.caller().subview(1), Each(AvailableSendBitrateIsBetween(
+  EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
                                               DataRate::KilobitsPerSec(50),
                                               DataRate::KilobitsPerSec(1100))));
 }
@@ -693,7 +693,7 @@ TEST(ScreamTest, MaybeTest(LinkCapacity5MbitPolicedTo256Kbit)) {
   // TODO: bugs.webrtc.org/447037083 - Improve Scream at low bitrates with
   // policed networks. Make it less aggressive to ramp up after backdown due to
   // loss. Consider lowering min ref window.
-  EXPECT_THAT(result.caller().subview(1), Each(AvailableSendBitrateIsBetween(
+  EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
                                               DataRate::KilobitsPerSec(50),
                                               DataRate::KilobitsPerSec(1100))));
 }
@@ -734,7 +734,7 @@ TEST(ScreamTest, MaybeTest(LinkCapacity5MbitWithCrossTrafficNoEcn)) {
   int index_where_available_bitrate_should_have_recovered =
       (tcp_message_delivered_time - start_time).seconds<int>() + 10;
   EXPECT_THAT(
-      result.caller().subview(
+      result.caller().subspan(
           index_where_available_bitrate_should_have_recovered),
       Each(AvailableSendBitrateIsBetween(DataRate::KilobitsPerSec(1500),
                                          DataRate::KilobitsPerSec(5000))));
diff --git a/test/video_codec_tester.cc b/test/video_codec_tester.cc
index edcc234f10..773528ab27 100644
--- a/test/video_codec_tester.cc
+++ b/test/video_codec_tester.cc
@@ -587,7 +587,7 @@ class VideoCodecAnalyzer : public VideoCodecTester::VideoCodecStats {
 
       Frame superframe = subframes.back();
       for (const Frame& frame :
-           ArrayView<Frame>(subframes).subview(0, subframes.size() - 1)) {
+           ArrayView<Frame>(subframes).subspan(0, subframes.size() - 1)) {
         superframe.decoded |= frame.decoded;
         superframe.encoded |= frame.encoded;
         superframe.frame_size += frame.frame_size;

From 66fb2492586d8bf74550541513bc7cc0e7b4c80b Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Sun, 15 Feb 2026 20:05:17 -0800
Subject: [PATCH 056/472] Update WebRTC code version (2026-02-16T04:04:48).

Bug: None
Change-Id: I93fa90bb40bd3c88dd5522184de9602ce222d001
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449581
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#46909}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index 7d05117193..70c2be6283 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-02-15T04:07:17";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-02-16T04:04:48";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From 49900f4cf11dc97310624fa418872879fd83fdc1 Mon Sep 17 00:00:00 2001
From: Rasmus Brandt <brandtr@webrtc.org>
Date: Mon, 16 Feb 2026 10:16:24 +0100
Subject: [PATCH 057/472] Video timing simulator: Add some missing 'inter'
 stream metrics
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Bug: b/423646186
Change-Id: Icf646886a4563f4883cfc31a056ea6066a6a6964
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449620
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46910}
---
 video/timing/simulator/rendering_simulator.cc | 16 +++++++++++++++
 video/timing/simulator/rendering_simulator.h  |  6 ++++++
 .../simulator/rendering_simulator_unittest.cc | 20 +++++++++++++++++++
 video/timing/simulator/stream_base.h          | 13 +++++++++---
 .../timing/simulator/stream_base_unittest.cc  |  9 +++++++++
 5 files changed, 61 insertions(+), 3 deletions(-)

diff --git a/video/timing/simulator/rendering_simulator.cc b/video/timing/simulator/rendering_simulator.cc
index e2101bd9f3..66b1e4854b 100644
--- a/video/timing/simulator/rendering_simulator.cc
+++ b/video/timing/simulator/rendering_simulator.cc
@@ -19,6 +19,7 @@
 #include "absl/base/nullability.h"
 #include "absl/container/flat_hash_map.h"
 #include "api/environment/environment.h"
+#include "api/numerics/samples_stats_counter.h"
 #include "api/sequence_checker.h"
 #include "api/units/data_size.h"
 #include "api/units/time_delta.h"
@@ -250,4 +251,19 @@ RenderingSimulator::Results RenderingSimulator::Simulate(
   return results;
 }
 
+SamplesStatsCounter RenderingSimulator::Stream::InterRenderTimeMs() {
+  SortByRenderOrder(frames);
+  return BuildSamplesMs(&InterRenderTime);
+}
+
+SamplesStatsCounter RenderingSimulator::Stream::InterDecodedTimeMs() {
+  SortByDecodedOrder(frames);
+  return BuildSamplesMs(&InterDecodedTime);
+}
+
+SamplesStatsCounter RenderingSimulator::Stream::InterRenderedTimeMs() {
+  SortByRenderedOrder(frames);
+  return BuildSamplesMs(&InterRenderedTime);
+}
+
 }  // namespace webrtc::video_timing_simulator
diff --git a/video/timing/simulator/rendering_simulator.h b/video/timing/simulator/rendering_simulator.h
index 6e6ea2dd9c..2be7dea597 100644
--- a/video/timing/simulator/rendering_simulator.h
+++ b/video/timing/simulator/rendering_simulator.h
@@ -301,6 +301,12 @@ class RenderingSimulator {
       return SumNonNegativeIntField(&Frame::frames_dropped);
     }
 
+    // Samples of per-frame delay variation metrics. Not `const` because they
+    // sort `frames`. Defined in `.cc` for circular include order reasons.
+    SamplesStatsCounter InterRenderTimeMs();
+    SamplesStatsCounter InterDecodedTimeMs();
+    SamplesStatsCounter InterRenderedTimeMs();
+
     // Samples of webrtc-stats values in ms.
     SamplesStatsCounter JitterBufferMinimumDelayMs() const {
       return BuildSamplesMs(&Frame::jitter_buffer_minimum_delay);
diff --git a/video/timing/simulator/rendering_simulator_unittest.cc b/video/timing/simulator/rendering_simulator_unittest.cc
index e3f6d5904c..a5f805cb02 100644
--- a/video/timing/simulator/rendering_simulator_unittest.cc
+++ b/video/timing/simulator/rendering_simulator_unittest.cc
@@ -178,6 +178,26 @@ TEST(RenderingSimulatorStreamTest, NumDecoderDroppedFrames) {
   EXPECT_EQ(stream.NumDecoderDroppedFrames(), 3);
 }
 
+TEST(RenderingSimulatorStreamTest, InterFrameDurations) {
+  RenderingSimulator::Stream stream{
+      .frames = {{.render_timestamp = Timestamp::Millis(1),
+                  .decoded_timestamp = Timestamp::Millis(2),
+                  .rendered_timestamp = Timestamp::Millis(3)},
+                 {.render_timestamp = Timestamp::Millis(11),
+                  .decoded_timestamp = Timestamp::Millis(102),
+                  .rendered_timestamp = Timestamp::Millis(1003)}}};
+
+  EXPECT_THAT(
+      stream.InterRenderTimeMs().GetTimedSamples(),
+      ElementsAre(Field(&SamplesStatsCounter::StatsSample::value, Eq(10))));
+  EXPECT_THAT(
+      stream.InterDecodedTimeMs().GetTimedSamples(),
+      ElementsAre(Field(&SamplesStatsCounter::StatsSample::value, Eq(100))));
+  EXPECT_THAT(
+      stream.InterRenderedTimeMs().GetTimedSamples(),
+      ElementsAre(Field(&SamplesStatsCounter::StatsSample::value, Eq(1000))));
+}
+
 TEST(RenderingSimulatorStreamTest, WebrtcStatsSamples) {
   RenderingSimulator::Stream stream{
       .frames = {{.rendered_timestamp = Timestamp::Zero(),
diff --git a/video/timing/simulator/stream_base.h b/video/timing/simulator/stream_base.h
index 2075bd3dca..6f545116f0 100644
--- a/video/timing/simulator/stream_base.h
+++ b/video/timing/simulator/stream_base.h
@@ -223,15 +223,22 @@ struct StreamBase {
   SamplesStatsCounter FrameDelayVariationMs() const {
     return BuildSamplesMs(&FrameT::frame_delay_variation);
   }
-  SamplesStatsCounter InterDepartureTimeMs() const {
+  SamplesStatsCounter InterDepartureTimeMs() {
+    SortByDepartureOrder(self().frames);
     return BuildSamplesMs(&InterDepartureTime<FrameT>);
   }
-  SamplesStatsCounter InterArrivalTimeMs() const {
+  SamplesStatsCounter InterArrivalTimeMs() {
+    SortByArrivalOrder(self().frames);
     return BuildSamplesMs(&InterArrivalTime<FrameT>);
   }
-  SamplesStatsCounter InterFrameDelayVariationMs() const {
+  SamplesStatsCounter InterFrameDelayVariationMs() {
+    SortByArrivalOrder(self().frames);
     return BuildSamplesMs(&InterFrameDelayVariation<FrameT>);
   }
+  SamplesStatsCounter InterAssembledTimeMs() {
+    SortByAssembledOrder(self().frames);
+    return BuildSamplesMs(&InterAssembledTime<FrameT>);
+  }
 };
 
 // -- Comparators and sorting --
diff --git a/video/timing/simulator/stream_base_unittest.cc b/video/timing/simulator/stream_base_unittest.cc
index 4c56b0b439..94a95ae713 100644
--- a/video/timing/simulator/stream_base_unittest.cc
+++ b/video/timing/simulator/stream_base_unittest.cc
@@ -164,5 +164,14 @@ TEST(StreamBaseTest, InterFrameDelayVariationMs) {
       ElementsAre(Field(&SamplesStatsCounter::StatsSample::value, Eq(1.0))));
 }
 
+TEST(StreamBaseTest, InterAssembledTimeMs) {
+  TestStream stream{
+      .frames = {{.assembled_timestamp = Timestamp::Micros(33333)},
+                 {.assembled_timestamp = Timestamp::Micros(66666 + 1000)}}};
+
+  EXPECT_THAT(
+      stream.InterAssembledTimeMs().GetTimedSamples(),
+      ElementsAre(Field(&SamplesStatsCounter::StatsSample::value, Eq(34.333))));
+}
 }  // namespace
 }  // namespace webrtc::video_timing_simulator

From 649af90f56e154c975f66d2df41fc62a363ce45e Mon Sep 17 00:00:00 2001
From: Sergey Silkin <ssilkin@webrtc.org>
Date: Mon, 16 Feb 2026 11:51:43 +0000
Subject: [PATCH 058/472] Set num_cores in encode-only mode
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

num_cores was added in
https://webrtc-review.googlesource.com/c/src/+/424420 and passed to the tester in encode-decode mode, but not in encode-only mode.

Bug: webrtc:42225151
Change-Id: Ie0123ce2aec522dfb0287cf9c6e966415073713a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449680
Reviewed-by: Erik Språng <sprang@webrtc.org>
Auto-Submit: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46911}
---
 modules/video_coding/codecs/test/video_codec_test.cc | 1 +
 1 file changed, 1 insertion(+)

diff --git a/modules/video_coding/codecs/test/video_codec_test.cc b/modules/video_coding/codecs/test/video_codec_test.cc
index aee9bd2300..9cbfd71df7 100644
--- a/modules/video_coding/codecs/test/video_codec_test.cc
+++ b/modules/video_coding/codecs/test/video_codec_test.cc
@@ -292,6 +292,7 @@ std::unique_ptr<VideoCodecStats> RunEncodeTest(
   if (absl::GetFlag(FLAGS_dump_encoder_output)) {
     encoder_settings.encoder_output_base_path = output_path + "_enc_output";
   }
+  encoder_settings.num_cores = absl::GetFlag(FLAGS_num_cores);
 
   return VideoCodecTester::RunEncodeTest(env, source_settings,
                                          encoder_factory.get(),

From 38839df1bfdc18b12f47aaabae52bdf19429db8c Mon Sep 17 00:00:00 2001
From: Evan Shrubsole <eshr@webrtc.org>
Date: Thu, 12 Feb 2026 14:05:47 +0000
Subject: [PATCH 059/472] Use environment clock in audio_device_unittest
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Bug: webrtc:42223992
Change-Id: Icd5ba9d7ceb8203e5ab4bdcd8e88069f6a6a6964
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/448285
Commit-Queue: Zoé Lepaul <zlep@webrtc.org>
Reviewed-by: Zoé Lepaul <zlep@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46912}
---
 sdk/android/BUILD.gn                           |  1 +
 .../audio_device/audio_device_unittest.cc      | 18 ++++++++++--------
 2 files changed, 11 insertions(+), 8 deletions(-)

diff --git a/sdk/android/BUILD.gn b/sdk/android/BUILD.gn
index 21687fa76c..1673cb90bb 100644
--- a/sdk/android/BUILD.gn
+++ b/sdk/android/BUILD.gn
@@ -1792,6 +1792,7 @@ if (is_android) {
       "../../rtc_base:timeutils",
       "../../rtc_base/synchronization:mutex",
       "../../rtc_base/system:inline",
+      "../../test:create_test_environment",
       "../../test:create_test_field_trials",
       "../../test:fileutils",
       "../../test:test_support",
diff --git a/sdk/android/native_unittests/audio_device/audio_device_unittest.cc b/sdk/android/native_unittests/audio_device/audio_device_unittest.cc
index a86fb1e225..107343789a 100644
--- a/sdk/android/native_unittests/audio_device/audio_device_unittest.cc
+++ b/sdk/android/native_unittests/audio_device/audio_device_unittest.cc
@@ -35,7 +35,6 @@
 #include "modules/audio_device/include/mock_audio_transport.h"
 #include "rtc_base/event.h"
 #include "rtc_base/synchronization/mutex.h"
-#include "rtc_base/time_utils.h"
 #include "sdk/android/generated_native_unittests_jni/BuildInfo_jni.h"
 #include "sdk/android/native_api/audio_device_module/audio_device_android.h"
 #include "sdk/android/native_api/jni/application_context_provider.h"
@@ -44,6 +43,7 @@
 #include "sdk/android/src/jni/audio_device/audio_device_module.h"
 #include "sdk/android/src/jni/audio_device/opensles_common.h"
 #include "sdk/android/src/jni/jvm.h"
+#include "test/create_test_environment.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
 #include "test/testsupport/file_utils.h"
@@ -258,8 +258,9 @@ class FifoAudioStream : public AudioStreamInterface {
 // See http://source.android.com/devices/audio/loopback.html for details.
 class LatencyMeasuringAudioStream : public AudioStreamInterface {
  public:
-  explicit LatencyMeasuringAudioStream(size_t frames_per_buffer)
-      : frames_per_buffer_(frames_per_buffer),
+  LatencyMeasuringAudioStream(size_t frames_per_buffer, Environment env)
+      : env_(env),
+        frames_per_buffer_(frames_per_buffer),
         bytes_per_buffer_(frames_per_buffer_ * sizeof(int16_t)),
         play_count_(0),
         rec_count_(0),
@@ -275,7 +276,7 @@ class LatencyMeasuringAudioStream : public AudioStreamInterface {
     memset(destination, 0, bytes_per_buffer_);
     if (play_count_ % (kNumCallbacksPerSecond / kImpulseFrequencyInHz) == 0) {
       if (pulse_time_ == 0) {
-        pulse_time_ = webrtc::TimeMillis();
+        pulse_time_ = env_.clock().TimeInMilliseconds();
       }
       PRINT(".");
       const int16_t impulse = std::numeric_limits<int16_t>::max();
@@ -305,7 +306,7 @@ class LatencyMeasuringAudioStream : public AudioStreamInterface {
         std::distance(vec.begin(), std::find(vec.begin(), vec.end(), max));
     if (max > kImpulseThreshold) {
       PRINTD("(%d,%d)", max, index_of_max);
-      int64_t now_time = webrtc::TimeMillis();
+      int64_t now_time = env_.clock().TimeInMilliseconds();
       int extra_delay = IndexToMilliseconds(static_cast<double>(index_of_max));
       PRINTD("[%d]", static_cast<int>(now_time - pulse_time_));
       PRINTD("[%d]", extra_delay);
@@ -358,6 +359,7 @@ class LatencyMeasuringAudioStream : public AudioStreamInterface {
   }
 
  private:
+  const Environment env_;
   const size_t frames_per_buffer_;
   const size_t bytes_per_buffer_;
   size_t play_count_;
@@ -685,7 +687,7 @@ class AudioDeviceTest : public ::testing::Test {
   }
 
   JNIEnv* jni_;
-  const Environment webrtc_env_ = CreateEnvironment();
+  const Environment webrtc_env_ = CreateTestEnvironment();
   ScopedJavaLocalRef<jobject> context_;
   webrtc::Event test_is_done_;
   webrtc::scoped_refptr<AudioDeviceModule> audio_device_;
@@ -1163,8 +1165,8 @@ TEST_F(AudioDeviceTest, DISABLED_MeasureLoopbackLatency) {
   EXPECT_EQ(record_channels(), playout_channels());
   EXPECT_EQ(record_sample_rate(), playout_sample_rate());
   NiceMock<MockAudioTransportAndroid> mock(kPlayout | kRecording);
-  std::unique_ptr<LatencyMeasuringAudioStream> latency_audio_stream(
-      new LatencyMeasuringAudioStream(playout_frames_per_10ms_buffer()));
+  auto latency_audio_stream = std::make_unique<LatencyMeasuringAudioStream>(
+      playout_frames_per_10ms_buffer(), webrtc_env_);
   mock.HandleCallbacks(&test_is_done_, latency_audio_stream.get(),
                        kMeasureLatencyTime.seconds() * kNumCallbacksPerSecond);
   EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));

From a7c29a690e22a2c0edd6f543e580c7f601362061 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Erik=20Spr=C3=A5ng?= <sprang@webrtc.org>
Date: Mon, 16 Feb 2026 13:33:24 +0100
Subject: [PATCH 060/472] Move VideoQualityAnalyzer to OnFrameDropped callback.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This deprecates usage of the EncodedImageCallback::DropReason type,
which was never actually read anyway, and moves this code to the new
callback method.

Bug: webrtc:467444018
Change-Id: I105e44237e6f1bf02bd5664cc6a098a2e734c6e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449640
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46913}
---
 api/test/video_quality_analyzer_interface.h         |  9 +++++++--
 .../video/default_video_quality_analyzer.cc         |  5 +----
 .../analyzer/video/default_video_quality_analyzer.h |  4 +---
 .../video/example_video_quality_analyzer.cc         |  5 +----
 .../analyzer/video/example_video_quality_analyzer.h |  4 +---
 .../video/quality_analyzing_video_encoder.cc        | 13 ++++++++++++-
 .../video/quality_analyzing_video_encoder.h         |  4 ++++
 7 files changed, 27 insertions(+), 17 deletions(-)

diff --git a/api/test/video_quality_analyzer_interface.h b/api/test/video_quality_analyzer_interface.h
index 48685b9936..43223364ec 100644
--- a/api/test/video_quality_analyzer_interface.h
+++ b/api/test/video_quality_analyzer_interface.h
@@ -112,8 +112,13 @@ class VideoQualityAnalyzerInterface
                               bool /* discarded */) {}
   // Will be called for each frame dropped by encoder.
   // `peer_name` is name of the peer on which side frame drop was detected.
-  virtual void OnFrameDropped(absl::string_view /* peer_name */,
-                              EncodedImageCallback::DropReason /* reason */) {}
+  virtual void OnFrameDropped(absl::string_view /* peer_name */) {}
+  // TODO: webrtc:467444018 - Remove when downstream usage is gone.
+  [[deprecated("Use callback without DropReason parameter")]]
+  virtual void OnFrameDropped(absl::string_view peer_name,
+                              EncodedImageCallback::DropReason /* reason */) {
+    OnFrameDropped(peer_name);
+  }
   // Will be called before calling the decoder.
   // `peer_name` is name of the peer on which side frame was received.
   virtual void OnFramePreDecode(absl::string_view /* peer_name */,
diff --git a/test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc b/test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc
index f5c322b407..5130dd6a3c 100644
--- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc
+++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc
@@ -33,7 +33,6 @@
 #include "api/units/timestamp.h"
 #include "api/video/encoded_image.h"
 #include "api/video/video_frame.h"
-#include "api/video_codecs/video_encoder.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/strings/string_builder.h"
@@ -385,9 +384,7 @@ void DefaultVideoQualityAnalyzer::OnFrameEncoded(
   }
 }
 
-void DefaultVideoQualityAnalyzer::OnFrameDropped(
-    absl::string_view peer_name,
-    EncodedImageCallback::DropReason reason) {
+void DefaultVideoQualityAnalyzer::OnFrameDropped(absl::string_view peer_name) {
   // Here we do nothing, because we will see this drop on renderer side.
 }
 
diff --git a/test/pc/e2e/analyzer/video/default_video_quality_analyzer.h b/test/pc/e2e/analyzer/video/default_video_quality_analyzer.h
index 0e041431ed..6b056d07e1 100644
--- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer.h
+++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer.h
@@ -29,7 +29,6 @@
 #include "api/units/timestamp.h"
 #include "api/video/encoded_image.h"
 #include "api/video/video_frame.h"
-#include "api/video_codecs/video_encoder.h"
 #include "rtc_base/synchronization/mutex.h"
 #include "rtc_base/thread_annotations.h"
 #include "system_wrappers/include/clock.h"
@@ -64,8 +63,7 @@ class DefaultVideoQualityAnalyzer : public VideoQualityAnalyzerInterface {
                       const EncodedImage& encoded_image,
                       const EncoderStats& stats,
                       bool discarded) override;
-  void OnFrameDropped(absl::string_view peer_name,
-                      EncodedImageCallback::DropReason reason) override;
+  void OnFrameDropped(absl::string_view peer_name) override;
   void OnFramePreDecode(absl::string_view peer_name,
                         uint16_t frame_id,
                         const EncodedImage& input_image) override;
diff --git a/test/pc/e2e/analyzer/video/example_video_quality_analyzer.cc b/test/pc/e2e/analyzer/video/example_video_quality_analyzer.cc
index d68db34033..8bd9453335 100644
--- a/test/pc/e2e/analyzer/video/example_video_quality_analyzer.cc
+++ b/test/pc/e2e/analyzer/video/example_video_quality_analyzer.cc
@@ -17,7 +17,6 @@
 #include "api/array_view.h"
 #include "api/video/encoded_image.h"
 #include "api/video/video_frame.h"
-#include "api/video_codecs/video_encoder.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/synchronization/mutex.h"
@@ -75,9 +74,7 @@ void ExampleVideoQualityAnalyzer::OnFrameEncoded(
   ++frames_encoded_;
 }
 
-void ExampleVideoQualityAnalyzer::OnFrameDropped(
-    absl::string_view peer_name,
-    EncodedImageCallback::DropReason reason) {
+void ExampleVideoQualityAnalyzer::OnFrameDropped(absl::string_view peer_name) {
   RTC_LOG(LS_INFO) << "Frame dropped by encoder";
   MutexLock lock(&lock_);
   ++frames_dropped_;
diff --git a/test/pc/e2e/analyzer/video/example_video_quality_analyzer.h b/test/pc/e2e/analyzer/video/example_video_quality_analyzer.h
index e347524439..59b056a74c 100644
--- a/test/pc/e2e/analyzer/video/example_video_quality_analyzer.h
+++ b/test/pc/e2e/analyzer/video/example_video_quality_analyzer.h
@@ -21,7 +21,6 @@
 #include "api/test/video_quality_analyzer_interface.h"
 #include "api/video/encoded_image.h"
 #include "api/video/video_frame.h"
-#include "api/video_codecs/video_encoder.h"
 #include "rtc_base/synchronization/mutex.h"
 #include "rtc_base/thread_annotations.h"
 
@@ -49,8 +48,7 @@ class ExampleVideoQualityAnalyzer : public VideoQualityAnalyzerInterface {
                       const EncodedImage& encoded_image,
                       const EncoderStats& stats,
                       bool discarded) override;
-  void OnFrameDropped(absl::string_view peer_name,
-                      EncodedImageCallback::DropReason reason) override;
+  void OnFrameDropped(absl::string_view peer_name) override;
   void OnFramePreDecode(absl::string_view peer_name,
                         uint16_t frame_id,
                         const EncodedImage& encoded_image) override;
diff --git a/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.cc b/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.cc
index 6516eead14..01138cf25a 100644
--- a/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.cc
+++ b/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.cc
@@ -339,11 +339,22 @@ EncodedImageCallback::Result QualityAnalyzingVideoEncoder::OnEncodedImage(
 void QualityAnalyzingVideoEncoder::OnDroppedFrame(
     EncodedImageCallback::DropReason reason) {
   MutexLock lock(&mutex_);
-  analyzer_->OnFrameDropped(peer_name_, reason);
+  analyzer_->OnFrameDropped(peer_name_);
   RTC_DCHECK(delegate_callback_);
   delegate_callback_->OnDroppedFrame(reason);
 }
 
+void QualityAnalyzingVideoEncoder::OnFrameDropped(
+    uint32_t rtp_timestamp,
+    int spatial_id,
+    bool is_end_of_temporal_unit) {
+  MutexLock lock(&mutex_);
+  analyzer_->OnFrameDropped(peer_name_);
+  RTC_DCHECK(delegate_callback_);
+  delegate_callback_->OnFrameDropped(rtp_timestamp, spatial_id,
+                                     is_end_of_temporal_unit);
+}
+
 bool QualityAnalyzingVideoEncoder::ShouldDiscard(
     uint16_t frame_id,
     const EncodedImage& encoded_image) {
diff --git a/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.h b/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.h
index 2be4bdffc7..bfa7cc9cf0 100644
--- a/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.h
+++ b/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.h
@@ -88,7 +88,11 @@ class QualityAnalyzingVideoEncoder : public VideoEncoder,
   EncodedImageCallback::Result OnEncodedImage(
       const EncodedImage& encoded_image,
       const CodecSpecificInfo* codec_specific_info) override;
+  // TODO: webrtc:467444018 - Remove after usage shifts to OnFrameDropped.
   void OnDroppedFrame(DropReason reason) override;
+  void OnFrameDropped(uint32_t rtp_timestamp,
+                      int spatial_id,
+                      bool is_end_of_temporal_unit) override;
 
  private:
   enum SimulcastMode {

From d5d0f8d60ca43dec4273ee014c2c3b9a73a04dfa Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Erik=20Spr=C3=A5ng?= <sprang@webrtc.org>
Date: Mon, 16 Feb 2026 06:07:23 -0800
Subject: [PATCH 061/472] Revert "Move VideoQualityAnalyzer to OnFrameDropped
 callback."
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This reverts commit a7c29a690e22a2c0edd6f543e580c7f601362061.

Reason for revert: Breaks downstream project.

Original change's description:
> Move VideoQualityAnalyzer to OnFrameDropped callback.
>
> This deprecates usage of the EncodedImageCallback::DropReason type,
> which was never actually read anyway, and moves this code to the new
> callback method.
>
> Bug: webrtc:467444018
> Change-Id: I105e44237e6f1bf02bd5664cc6a098a2e734c6e9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449640
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#46913}

Bug: webrtc:467444018
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I7630db5a5e46d320d0d1e385bede09570533ca0a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449720
Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46914}
---
 api/test/video_quality_analyzer_interface.h         |  9 ++-------
 .../video/default_video_quality_analyzer.cc         |  5 ++++-
 .../analyzer/video/default_video_quality_analyzer.h |  4 +++-
 .../video/example_video_quality_analyzer.cc         |  5 ++++-
 .../analyzer/video/example_video_quality_analyzer.h |  4 +++-
 .../video/quality_analyzing_video_encoder.cc        | 13 +------------
 .../video/quality_analyzing_video_encoder.h         |  4 ----
 7 files changed, 17 insertions(+), 27 deletions(-)

diff --git a/api/test/video_quality_analyzer_interface.h b/api/test/video_quality_analyzer_interface.h
index 43223364ec..48685b9936 100644
--- a/api/test/video_quality_analyzer_interface.h
+++ b/api/test/video_quality_analyzer_interface.h
@@ -112,13 +112,8 @@ class VideoQualityAnalyzerInterface
                               bool /* discarded */) {}
   // Will be called for each frame dropped by encoder.
   // `peer_name` is name of the peer on which side frame drop was detected.
-  virtual void OnFrameDropped(absl::string_view /* peer_name */) {}
-  // TODO: webrtc:467444018 - Remove when downstream usage is gone.
-  [[deprecated("Use callback without DropReason parameter")]]
-  virtual void OnFrameDropped(absl::string_view peer_name,
-                              EncodedImageCallback::DropReason /* reason */) {
-    OnFrameDropped(peer_name);
-  }
+  virtual void OnFrameDropped(absl::string_view /* peer_name */,
+                              EncodedImageCallback::DropReason /* reason */) {}
   // Will be called before calling the decoder.
   // `peer_name` is name of the peer on which side frame was received.
   virtual void OnFramePreDecode(absl::string_view /* peer_name */,
diff --git a/test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc b/test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc
index 5130dd6a3c..f5c322b407 100644
--- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc
+++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc
@@ -33,6 +33,7 @@
 #include "api/units/timestamp.h"
 #include "api/video/encoded_image.h"
 #include "api/video/video_frame.h"
+#include "api/video_codecs/video_encoder.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/strings/string_builder.h"
@@ -384,7 +385,9 @@ void DefaultVideoQualityAnalyzer::OnFrameEncoded(
   }
 }
 
-void DefaultVideoQualityAnalyzer::OnFrameDropped(absl::string_view peer_name) {
+void DefaultVideoQualityAnalyzer::OnFrameDropped(
+    absl::string_view peer_name,
+    EncodedImageCallback::DropReason reason) {
   // Here we do nothing, because we will see this drop on renderer side.
 }
 
diff --git a/test/pc/e2e/analyzer/video/default_video_quality_analyzer.h b/test/pc/e2e/analyzer/video/default_video_quality_analyzer.h
index 6b056d07e1..0e041431ed 100644
--- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer.h
+++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer.h
@@ -29,6 +29,7 @@
 #include "api/units/timestamp.h"
 #include "api/video/encoded_image.h"
 #include "api/video/video_frame.h"
+#include "api/video_codecs/video_encoder.h"
 #include "rtc_base/synchronization/mutex.h"
 #include "rtc_base/thread_annotations.h"
 #include "system_wrappers/include/clock.h"
@@ -63,7 +64,8 @@ class DefaultVideoQualityAnalyzer : public VideoQualityAnalyzerInterface {
                       const EncodedImage& encoded_image,
                       const EncoderStats& stats,
                       bool discarded) override;
-  void OnFrameDropped(absl::string_view peer_name) override;
+  void OnFrameDropped(absl::string_view peer_name,
+                      EncodedImageCallback::DropReason reason) override;
   void OnFramePreDecode(absl::string_view peer_name,
                         uint16_t frame_id,
                         const EncodedImage& input_image) override;
diff --git a/test/pc/e2e/analyzer/video/example_video_quality_analyzer.cc b/test/pc/e2e/analyzer/video/example_video_quality_analyzer.cc
index 8bd9453335..d68db34033 100644
--- a/test/pc/e2e/analyzer/video/example_video_quality_analyzer.cc
+++ b/test/pc/e2e/analyzer/video/example_video_quality_analyzer.cc
@@ -17,6 +17,7 @@
 #include "api/array_view.h"
 #include "api/video/encoded_image.h"
 #include "api/video/video_frame.h"
+#include "api/video_codecs/video_encoder.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/synchronization/mutex.h"
@@ -74,7 +75,9 @@ void ExampleVideoQualityAnalyzer::OnFrameEncoded(
   ++frames_encoded_;
 }
 
-void ExampleVideoQualityAnalyzer::OnFrameDropped(absl::string_view peer_name) {
+void ExampleVideoQualityAnalyzer::OnFrameDropped(
+    absl::string_view peer_name,
+    EncodedImageCallback::DropReason reason) {
   RTC_LOG(LS_INFO) << "Frame dropped by encoder";
   MutexLock lock(&lock_);
   ++frames_dropped_;
diff --git a/test/pc/e2e/analyzer/video/example_video_quality_analyzer.h b/test/pc/e2e/analyzer/video/example_video_quality_analyzer.h
index 59b056a74c..e347524439 100644
--- a/test/pc/e2e/analyzer/video/example_video_quality_analyzer.h
+++ b/test/pc/e2e/analyzer/video/example_video_quality_analyzer.h
@@ -21,6 +21,7 @@
 #include "api/test/video_quality_analyzer_interface.h"
 #include "api/video/encoded_image.h"
 #include "api/video/video_frame.h"
+#include "api/video_codecs/video_encoder.h"
 #include "rtc_base/synchronization/mutex.h"
 #include "rtc_base/thread_annotations.h"
 
@@ -48,7 +49,8 @@ class ExampleVideoQualityAnalyzer : public VideoQualityAnalyzerInterface {
                       const EncodedImage& encoded_image,
                       const EncoderStats& stats,
                       bool discarded) override;
-  void OnFrameDropped(absl::string_view peer_name) override;
+  void OnFrameDropped(absl::string_view peer_name,
+                      EncodedImageCallback::DropReason reason) override;
   void OnFramePreDecode(absl::string_view peer_name,
                         uint16_t frame_id,
                         const EncodedImage& encoded_image) override;
diff --git a/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.cc b/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.cc
index 01138cf25a..6516eead14 100644
--- a/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.cc
+++ b/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.cc
@@ -339,22 +339,11 @@ EncodedImageCallback::Result QualityAnalyzingVideoEncoder::OnEncodedImage(
 void QualityAnalyzingVideoEncoder::OnDroppedFrame(
     EncodedImageCallback::DropReason reason) {
   MutexLock lock(&mutex_);
-  analyzer_->OnFrameDropped(peer_name_);
+  analyzer_->OnFrameDropped(peer_name_, reason);
   RTC_DCHECK(delegate_callback_);
   delegate_callback_->OnDroppedFrame(reason);
 }
 
-void QualityAnalyzingVideoEncoder::OnFrameDropped(
-    uint32_t rtp_timestamp,
-    int spatial_id,
-    bool is_end_of_temporal_unit) {
-  MutexLock lock(&mutex_);
-  analyzer_->OnFrameDropped(peer_name_);
-  RTC_DCHECK(delegate_callback_);
-  delegate_callback_->OnFrameDropped(rtp_timestamp, spatial_id,
-                                     is_end_of_temporal_unit);
-}
-
 bool QualityAnalyzingVideoEncoder::ShouldDiscard(
     uint16_t frame_id,
     const EncodedImage& encoded_image) {
diff --git a/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.h b/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.h
index bfa7cc9cf0..2be4bdffc7 100644
--- a/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.h
+++ b/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.h
@@ -88,11 +88,7 @@ class QualityAnalyzingVideoEncoder : public VideoEncoder,
   EncodedImageCallback::Result OnEncodedImage(
       const EncodedImage& encoded_image,
       const CodecSpecificInfo* codec_specific_info) override;
-  // TODO: webrtc:467444018 - Remove after usage shifts to OnFrameDropped.
   void OnDroppedFrame(DropReason reason) override;
-  void OnFrameDropped(uint32_t rtp_timestamp,
-                      int spatial_id,
-                      bool is_end_of_temporal_unit) override;
 
  private:
   enum SimulcastMode {

From 7522ef394bd531d80f58aef64db57d5b0d46a27f Mon Sep 17 00:00:00 2001
From: Evan Shrubsole <eshr@webrtc.org>
Date: Fri, 13 Feb 2026 13:55:26 +0000
Subject: [PATCH 062/472] Add support for main thread socket server in
 GlobalSimulatedTimeController

This CL updates virtual_socket_unittest accordingly. The goal of this
change is to remove the need for AutoSocketServerThread when simulated
time is used.

Drive by fix all unsafe buffer usages in this test.

Bug: webrtc:469327588,webrtc:381524905
Change-Id: I4297e0efc1e7254294219c372eac7a216a6a6964
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449121
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46915}
---
 rtc_base/BUILD.gn                             |   2 +
 rtc_base/virtual_socket_unittest.cc           | 170 ++++++++++--------
 test/time_controller/BUILD.gn                 |   1 +
 test/time_controller/simulated_thread.cc      |  26 ++-
 test/time_controller/simulated_thread.h       |  10 +-
 .../simulated_time_controller.cc              |  10 +-
 .../simulated_time_controller.h               |   3 +-
 .../simulated_time_controller_impl.cc         |  12 +-
 .../simulated_time_controller_impl.h          |   3 +-
 unsafe_buffers_paths.txt                      |   1 -
 10 files changed, 150 insertions(+), 88 deletions(-)

diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
index d582c6275a..a2022711f8 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -2183,12 +2183,14 @@ if (rtc_include_tests) {
         "../test:create_test_environment",
         "../test:fileutils",
         "../test:test_support",
+        "../test/time_controller",
         "containers:flat_map",
         "containers:unittests",
         "memory:unittests",
         "network:received_packet",
         "synchronization:mutex",
         "task_utils:repeating_task",
+        "//third_party/abseil-cpp/absl/algorithm:container",
         "//third_party/abseil-cpp/absl/base:core_headers",
         "//third_party/abseil-cpp/absl/memory",
         "//third_party/abseil-cpp/absl/numeric:bits",
diff --git a/rtc_base/virtual_socket_unittest.cc b/rtc_base/virtual_socket_unittest.cc
index 9fadd190dd..9246372671 100644
--- a/rtc_base/virtual_socket_unittest.cc
+++ b/rtc_base/virtual_socket_unittest.cc
@@ -9,22 +9,27 @@
  */
 
 #include <algorithm>
+#include <array>
 #include <cmath>
+#include <cstddef>
 #include <cstdint>
 #include <cstdlib>
 #include <cstring>
 #include <ctime>
 #include <memory>
+#include <span>
 #include <utility>
+#include <vector>
 
+#include "absl/algorithm/container.h"
 #include "absl/memory/memory.h"
+#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/transport/ecn_marking.h"
 #include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
 #include "rtc_base/async_packet_socket.h"
 #include "rtc_base/async_udp_socket.h"
-#include "rtc_base/fake_clock.h"
-#include "rtc_base/gunit.h"
 #include "rtc_base/ip_address.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/net_helpers.h"
@@ -39,11 +44,15 @@
 #include "test/create_test_environment.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/time_controller/simulated_time_controller.h"
 
 namespace webrtc {
 namespace {
 
+using ::testing::ContainerEq;
+using ::testing::Eq;
 using ::testing::NotNull;
+using ::testing::Pointwise;
 using testing::SSE_CLOSE;
 using testing::SSE_ERROR;
 using testing::SSE_OPEN;
@@ -70,8 +79,10 @@ struct Sender {
       uint32_t size =
           std::clamp<uint32_t>(rate * delay / 1000, sizeof(uint32_t), 4096);
       count += size;
-      memcpy(dummy, &cur_time, sizeof(cur_time));
-      socket->Send(dummy, size, options);
+      auto cur_time_bytes = std::as_bytes(std::span{&cur_time, 1});
+      absl::c_transform(cur_time_bytes, dummy.begin(),
+                        [](std::byte b) { return static_cast<char>(b); });
+      socket->Send(dummy.data(), size, options);
 
       last_send = cur_time;
       return NextDelay();
@@ -91,7 +102,7 @@ struct Sender {
   uint32_t rate;  // bytes per second
   uint32_t count;
   int64_t last_send;
-  char dummy[4096];
+  std::array<char, 4096> dummy;
 };
 
 struct Receiver {
@@ -158,8 +169,9 @@ struct Receiver {
 class VirtualSocketServerTest : public ::testing::Test {
  public:
   VirtualSocketServerTest()
-      : ss_(&fake_clock_),
-        thread_(&ss_),
+      : time_controller_(Timestamp::Millis(1000), &ss_),
+        env_(CreateTestEnvironment(
+            CreateTestEnvironmentOptions{.time = &time_controller_})),
         kIPv4AnyAddress(IPAddress(INADDR_ANY), 0),
         kIPv6AnyAddress(IPAddress(in6addr_any), 0) {}
 
@@ -176,9 +188,10 @@ class VirtualSocketServerTest : public ::testing::Test {
     } else if (post_ip.family() == AF_INET6) {
       in6_addr post_ip6 = post_ip.ipv6_address();
       in6_addr pre_ip6 = pre_ip.ipv6_address();
-      uint32_t* post_as_ints = reinterpret_cast<uint32_t*>(&post_ip6.s6_addr);
-      uint32_t* pre_as_ints = reinterpret_cast<uint32_t*>(&pre_ip6.s6_addr);
-      EXPECT_EQ(post_as_ints[3], pre_as_ints[3]);
+      ArrayView<uint8_t, 16> post_bytes = post_ip6.s6_addr;
+      ArrayView<uint8_t, 16> pre_bytes = pre_ip6.s6_addr;
+      EXPECT_THAT(post_bytes.subspan(12, 4),
+                  Pointwise(Eq(), pre_bytes.subspan(12, 4)));
     }
   }
 
@@ -196,7 +209,7 @@ class VirtualSocketServerTest : public ::testing::Test {
     EXPECT_TRUE(client1_any_addr.IsAnyIP());
     TestClient client1(
         std::make_unique<AsyncUDPSocket>(env_, std::move(socket)),
-        &fake_clock_);
+        &time_controller_);
 
     // Create client2 bound to the address route.
     std::unique_ptr<Socket> socket2 =
@@ -206,7 +219,7 @@ class VirtualSocketServerTest : public ::testing::Test {
     EXPECT_FALSE(client2_addr.IsAnyIP());
     TestClient client2(
         std::make_unique<AsyncUDPSocket>(env_, std::move(socket2)),
-        &fake_clock_);
+        &time_controller_);
 
     // Client1 sends to client2, client2 should see the default address as
     // client1's address.
@@ -231,14 +244,14 @@ class VirtualSocketServerTest : public ::testing::Test {
 
     TestClient client1(
         std::make_unique<AsyncUDPSocket>(env_, std::move(socket)),
-        &fake_clock_);
+        &time_controller_);
     SocketAddress client2_addr;
     {
       std::unique_ptr<Socket> socket2 =
           ss_.Create(initial_addr.family(), SOCK_DGRAM);
       TestClient client2(
           std::make_unique<AsyncUDPSocket>(env_, std::move(socket2)),
-          &fake_clock_);
+          &time_controller_);
 
       EXPECT_EQ(3, client2.SendTo("foo", 3, server_addr));
       EXPECT_TRUE(client1.CheckNextPacket("foo", 3, &client2_addr));
@@ -252,7 +265,7 @@ class VirtualSocketServerTest : public ::testing::Test {
     SocketAddress empty = EmptySocketAddressWithFamily(initial_addr.family());
     for (int i = 0; i < 10; i++) {
       TestClient client2(AsyncUDPSocket::Create(env_, empty, ss_),
-                         &fake_clock_);
+                         &time_controller_);
 
       SocketAddress next_client2_addr;
       EXPECT_EQ(3, client2.SendTo("foo", 3, server_addr));
@@ -309,7 +322,7 @@ class VirtualSocketServerTest : public ::testing::Test {
     EXPECT_FALSE(sink.Check(client.get(), SSE_OPEN));
     EXPECT_FALSE(sink.Check(client.get(), SSE_CLOSE));
 
-    ss_.ProcessMessagesUntilIdle();
+    time_controller_.AdvanceTime(TimeDelta::Zero());
 
     // Client still connecting
     EXPECT_EQ(client->GetState(), Socket::CS_CONNECTING);
@@ -328,7 +341,7 @@ class VirtualSocketServerTest : public ::testing::Test {
     EXPECT_EQ(accepted->GetLocalAddress(), server->GetLocalAddress());
     EXPECT_EQ(accepted->GetRemoteAddress(), client->GetLocalAddress());
 
-    ss_.ProcessMessagesUntilIdle();
+    time_controller_.AdvanceTime(TimeDelta::Zero());
 
     // Client has connected
     EXPECT_EQ(client->GetState(), Socket::CS_CONNECTED);
@@ -359,7 +372,7 @@ class VirtualSocketServerTest : public ::testing::Test {
     // Attempt connect to non-listening socket
     EXPECT_EQ(0, client->Connect(server->GetLocalAddress()));
 
-    ss_.ProcessMessagesUntilIdle();
+    time_controller_.AdvanceTime(TimeDelta::Zero());
 
     // No pending server connections
     EXPECT_FALSE(sink.Check(server.get(), SSE_READ));
@@ -398,7 +411,7 @@ class VirtualSocketServerTest : public ::testing::Test {
     EXPECT_FALSE(sink.Check(server.get(), SSE_READ));
     server->Close();
 
-    ss_.ProcessMessagesUntilIdle();
+    time_controller_.AdvanceTime(TimeDelta::Zero());
 
     // Result: connection failed
     EXPECT_EQ(client->GetState(), Socket::CS_CLOSED);
@@ -414,13 +427,13 @@ class VirtualSocketServerTest : public ::testing::Test {
     EXPECT_EQ(0, server->Listen(5));
     EXPECT_EQ(0, client->Connect(server->GetLocalAddress()));
 
-    ss_.ProcessMessagesUntilIdle();
+    time_controller_.AdvanceTime(TimeDelta::Zero());
 
     // Server close while socket is in accept queue
     EXPECT_TRUE(sink.Check(server.get(), SSE_READ));
     server->Close();
 
-    ss_.ProcessMessagesUntilIdle();
+    time_controller_.AdvanceTime(TimeDelta::Zero());
 
     // Result: connection failed
     EXPECT_EQ(client->GetState(), Socket::CS_CLOSED);
@@ -437,7 +450,7 @@ class VirtualSocketServerTest : public ::testing::Test {
     EXPECT_EQ(0, server->Listen(5));
     EXPECT_EQ(0, client->Connect(server->GetLocalAddress()));
 
-    ss_.ProcessMessagesUntilIdle();
+    time_controller_.AdvanceTime(TimeDelta::Zero());
 
     // Server accepts connection
     EXPECT_TRUE(sink.Check(server.get(), SSE_READ));
@@ -452,7 +465,7 @@ class VirtualSocketServerTest : public ::testing::Test {
     EXPECT_EQ(client->GetState(), Socket::CS_CONNECTING);
     client->Close();
 
-    ss_.ProcessMessagesUntilIdle();
+    time_controller_.AdvanceTime(TimeDelta::Zero());
 
     // Result: accepted socket closes
     EXPECT_EQ(accepted->GetState(), Socket::CS_CLOSED);
@@ -478,7 +491,7 @@ class VirtualSocketServerTest : public ::testing::Test {
     EXPECT_EQ(0, a->Connect(b->GetLocalAddress()));
     EXPECT_EQ(0, b->Connect(a->GetLocalAddress()));
 
-    ss_.ProcessMessagesUntilIdle();
+    time_controller_.AdvanceTime(TimeDelta::Zero());
 
     EXPECT_TRUE(sink.Check(a.get(), SSE_OPEN));
     EXPECT_EQ(a->GetState(), Socket::CS_CONNECTED);
@@ -492,7 +505,7 @@ class VirtualSocketServerTest : public ::testing::Test {
     b->Close();
     EXPECT_EQ(1, a->Send("b", 1));
 
-    ss_.ProcessMessagesUntilIdle();
+    time_controller_.AdvanceTime(TimeDelta::Zero());
 
     char buffer[10];
     EXPECT_FALSE(sink.Check(b.get(), SSE_READ));
@@ -526,59 +539,62 @@ class VirtualSocketServerTest : public ::testing::Test {
     EXPECT_EQ(0, a->Connect(b->GetLocalAddress()));
     EXPECT_EQ(0, b->Connect(a->GetLocalAddress()));
 
-    ss_.ProcessMessagesUntilIdle();
+    time_controller_.AdvanceTime(TimeDelta::Zero());
 
     const size_t kBufferSize = 2000;
     ss_.set_send_buffer_capacity(kBufferSize);
     ss_.set_recv_buffer_capacity(kBufferSize);
 
     const size_t kDataSize = 5000;
-    char send_buffer[kDataSize], recv_buffer[kDataSize];
+    std::vector<char> send_buffer(kDataSize);
+    std::vector<char> recv_buffer(kDataSize);
     for (size_t i = 0; i < kDataSize; ++i)
       send_buffer[i] = static_cast<char>(i % 256);
-    memset(recv_buffer, 0, sizeof(recv_buffer));
+    std::fill(recv_buffer.begin(), recv_buffer.end(), 0);
     size_t send_pos = 0, recv_pos = 0;
 
     // Can't send more than send buffer in one write
-    int result = a->Send(send_buffer + send_pos, kDataSize - send_pos);
+    int result = a->Send(&send_buffer[send_pos], kDataSize - send_pos);
     EXPECT_EQ(static_cast<int>(kBufferSize), result);
     send_pos += result;
 
-    ss_.ProcessMessagesUntilIdle();
+    time_controller_.AdvanceTime(TimeDelta::Zero());
     EXPECT_FALSE(sink.Check(a.get(), SSE_WRITE));
     EXPECT_TRUE(sink.Check(b.get(), SSE_READ));
 
     // Receive buffer is already filled, fill send buffer again
-    result = a->Send(send_buffer + send_pos, kDataSize - send_pos);
+    result = a->Send(&send_buffer[send_pos], kDataSize - send_pos);
     EXPECT_EQ(static_cast<int>(kBufferSize), result);
     send_pos += result;
 
-    ss_.ProcessMessagesUntilIdle();
+    time_controller_.AdvanceTime(TimeDelta::Zero());
     EXPECT_FALSE(sink.Check(a.get(), SSE_WRITE));
     EXPECT_FALSE(sink.Check(b.get(), SSE_READ));
 
     // No more room in send or receive buffer
-    result = a->Send(send_buffer + send_pos, kDataSize - send_pos);
+    result = a->Send(&send_buffer[send_pos], kDataSize - send_pos);
     EXPECT_EQ(-1, result);
     EXPECT_TRUE(a->IsBlocking());
 
     // Read a subset of the data
-    result = b->Recv(recv_buffer + recv_pos, 500, nullptr);
+    result = b->Recv(&recv_buffer[recv_pos], 500, nullptr);
     EXPECT_EQ(500, result);
     recv_pos += result;
 
-    ss_.ProcessMessagesUntilIdle();
+    time_controller_.AdvanceTime(TimeDelta::Zero());
     EXPECT_TRUE(sink.Check(a.get(), SSE_WRITE));
     EXPECT_TRUE(sink.Check(b.get(), SSE_READ));
 
     // Room for more on the sending side
-    result = a->Send(send_buffer + send_pos, kDataSize - send_pos);
+    result = a->Send(&send_buffer[send_pos], kDataSize - send_pos);
     EXPECT_EQ(500, result);
     send_pos += result;
 
     // Empty the recv buffer
     while (true) {
-      result = b->Recv(recv_buffer + recv_pos, kDataSize - recv_pos, nullptr);
+      if (recv_pos >= kDataSize)
+        break;
+      result = b->Recv(&recv_buffer[recv_pos], kDataSize - recv_pos, nullptr);
       if (result < 0) {
         EXPECT_EQ(-1, result);
         EXPECT_TRUE(b->IsBlocking());
@@ -587,12 +603,14 @@ class VirtualSocketServerTest : public ::testing::Test {
       recv_pos += result;
     }
 
-    ss_.ProcessMessagesUntilIdle();
+    time_controller_.AdvanceTime(TimeDelta::Zero());
     EXPECT_TRUE(sink.Check(b.get(), SSE_READ));
 
     // Continue to empty the recv buffer
     while (true) {
-      result = b->Recv(recv_buffer + recv_pos, kDataSize - recv_pos, nullptr);
+      if (recv_pos >= kDataSize)
+        break;
+      result = b->Recv(&recv_buffer[recv_pos], kDataSize - recv_pos, nullptr);
       if (result < 0) {
         EXPECT_EQ(-1, result);
         EXPECT_TRUE(b->IsBlocking());
@@ -602,16 +620,18 @@ class VirtualSocketServerTest : public ::testing::Test {
     }
 
     // Send last of the data
-    result = a->Send(send_buffer + send_pos, kDataSize - send_pos);
+    result = a->Send(&send_buffer[send_pos], kDataSize - send_pos);
     EXPECT_EQ(500, result);
     send_pos += result;
 
-    ss_.ProcessMessagesUntilIdle();
+    time_controller_.AdvanceTime(TimeDelta::Zero());
     EXPECT_TRUE(sink.Check(b.get(), SSE_READ));
 
     // Receive the last of the data
     while (true) {
-      result = b->Recv(recv_buffer + recv_pos, kDataSize - recv_pos, nullptr);
+      if (recv_pos >= kDataSize)
+        break;
+      result = b->Recv(&recv_buffer[recv_pos], kDataSize - recv_pos, nullptr);
       if (result < 0) {
         EXPECT_EQ(-1, result);
         EXPECT_TRUE(b->IsBlocking());
@@ -620,13 +640,13 @@ class VirtualSocketServerTest : public ::testing::Test {
       recv_pos += result;
     }
 
-    ss_.ProcessMessagesUntilIdle();
+    time_controller_.AdvanceTime(TimeDelta::Zero());
     EXPECT_FALSE(sink.Check(b.get(), SSE_READ));
 
     // The received data matches the sent data
     EXPECT_EQ(kDataSize, send_pos);
     EXPECT_EQ(kDataSize, recv_pos);
-    EXPECT_EQ(0, memcmp(recv_buffer, send_buffer, kDataSize));
+    EXPECT_EQ(recv_buffer, send_buffer);
   }
 
   void TcpSendsPacketsInOrderTest(const SocketAddress& initial_addr) {
@@ -643,20 +663,20 @@ class VirtualSocketServerTest : public ::testing::Test {
 
     EXPECT_EQ(0, a->Connect(b->GetLocalAddress()));
     EXPECT_EQ(0, b->Connect(a->GetLocalAddress()));
-    ss_.ProcessMessagesUntilIdle();
+    time_controller_.AdvanceTime(TimeDelta::Zero());
 
     // First, deliver all packets in 0 ms.
-    char buffer[2] = {0, 0};
+    std::array<char, 2> buffer = {0, 0};
     const char cNumPackets = 10;
     for (char i = 0; i < cNumPackets; ++i) {
       buffer[0] = '0' + i;
-      EXPECT_EQ(1, a->Send(buffer, 1));
+      EXPECT_EQ(1, a->Send(buffer.data(), 1));
     }
 
-    ss_.ProcessMessagesUntilIdle();
+    time_controller_.AdvanceTime(TimeDelta::Zero());
 
     for (char i = 0; i < cNumPackets; ++i) {
-      EXPECT_EQ(1, b->Recv(buffer, sizeof(buffer), nullptr));
+      EXPECT_EQ(1, b->Recv(buffer.data(), buffer.size(), nullptr));
       EXPECT_EQ(static_cast<char>('0' + i), buffer[0]);
     }
 
@@ -670,13 +690,13 @@ class VirtualSocketServerTest : public ::testing::Test {
 
     for (char i = 0; i < cNumPackets; ++i) {
       buffer[0] = 'A' + i;
-      EXPECT_EQ(1, a->Send(buffer, 1));
+      EXPECT_EQ(1, a->Send(buffer.data(), 1));
     }
 
-    ss_.ProcessMessagesUntilIdle();
+    time_controller_.AdvanceTime(TimeDelta::Seconds(2));
 
     for (char i = 0; i < cNumPackets; ++i) {
-      EXPECT_EQ(1, b->Recv(buffer, sizeof(buffer), nullptr));
+      EXPECT_EQ(1, b->Recv(buffer.data(), buffer.size(), nullptr));
       EXPECT_EQ(static_cast<char>('A' + i), buffer[0]);
     }
   }
@@ -698,15 +718,15 @@ class VirtualSocketServerTest : public ::testing::Test {
     uint32_t bandwidth = 64 * 1024;
     ss_.set_bandwidth(bandwidth);
 
-    Thread* pthMain = Thread::Current();
+    Thread* pthMain = time_controller_.GetMainThread();
     Sender sender(env_, pthMain, std::move(send_socket), 80 * 1024);
     Receiver receiver(env_, pthMain, std::move(recv_socket), bandwidth);
 
     // Allow the sender to run for 5 (simulated) seconds, then be stopped for 5
     // seconds.
-    SIMULATED_WAIT(false, 5000, fake_clock_);
+    time_controller_.AdvanceTime(TimeDelta::Millis(5000));
     sender.periodic.Stop();
-    SIMULATED_WAIT(false, 5000, fake_clock_);
+    time_controller_.AdvanceTime(TimeDelta::Millis(5000));
 
     // Ensure the observed bandwidth fell within a reasonable margin of error.
     EXPECT_TRUE(receiver.count >= 5 * 3 * bandwidth / 4);
@@ -740,7 +760,7 @@ class VirtualSocketServerTest : public ::testing::Test {
     EXPECT_EQ(recv_socket->GetLocalAddress().family(), initial_addr.family());
     ASSERT_EQ(0, send_socket->Connect(recv_socket->GetLocalAddress()));
 
-    Thread* pthMain = Thread::Current();
+    Thread* pthMain = time_controller_.GetMainThread();
     // Avg packet size is 2K, so at 200KB/s for 10s, we should see about
     // 1000 packets, which is necessary to get a good distribution.
     Sender sender(env_, pthMain, std::move(send_socket), 100 * 2 * 1024);
@@ -748,10 +768,10 @@ class VirtualSocketServerTest : public ::testing::Test {
 
     // Simulate 10 seconds of packets being sent, then check the observed delay
     // distribution.
-    SIMULATED_WAIT(false, 10000, fake_clock_);
+    time_controller_.AdvanceTime(TimeDelta::Millis(10000));
     sender.periodic.Stop();
     receiver.periodic.Stop();
-    ss_.ProcessMessagesUntilIdle();
+    time_controller_.AdvanceTime(TimeDelta::Zero());
 
     const double sample_mean = receiver.sum / receiver.samples;
     double num =
@@ -800,19 +820,19 @@ class VirtualSocketServerTest : public ::testing::Test {
 
     if (shouldSucceed) {
       EXPECT_EQ(0, client->Connect(server->GetLocalAddress()));
-      ss_.ProcessMessagesUntilIdle();
+      time_controller_.AdvanceTime(TimeDelta::Zero());
       EXPECT_TRUE(sink.Check(server.get(), SSE_READ));
       std::unique_ptr<Socket> accepted =
           absl::WrapUnique(server->Accept(&accept_address));
       EXPECT_TRUE(nullptr != accepted);
       EXPECT_NE(kEmptyAddr, accept_address);
-      ss_.ProcessMessagesUntilIdle();
+      time_controller_.AdvanceTime(TimeDelta::Zero());
       EXPECT_TRUE(sink.Check(client.get(), SSE_OPEN));
       EXPECT_EQ(client->GetRemoteAddress(), server->GetLocalAddress());
     } else {
       // Check that the connection failed.
       EXPECT_EQ(-1, client->Connect(server->GetLocalAddress()));
-      ss_.ProcessMessagesUntilIdle();
+      time_controller_.AdvanceTime(TimeDelta::Zero());
 
       EXPECT_FALSE(sink.Check(server.get(), SSE_READ));
       EXPECT_TRUE(nullptr == server->Accept(&accept_address));
@@ -834,13 +854,13 @@ class VirtualSocketServerTest : public ::testing::Test {
     SocketAddress bound_server_addr = socket->GetLocalAddress();
     TestClient client1(
         std::make_unique<AsyncUDPSocket>(env_, std::move(socket)),
-        &fake_clock_);
+        &time_controller_);
 
     std::unique_ptr<Socket> socket2 = ss_.Create(AF_INET, SOCK_DGRAM);
     socket2->Bind(client_addr);
     TestClient client2(
         std::make_unique<AsyncUDPSocket>(env_, std::move(socket2)),
-        &fake_clock_);
+        &time_controller_);
     SocketAddress client2_addr;
 
     if (shouldSucceed) {
@@ -864,14 +884,14 @@ class VirtualSocketServerTest : public ::testing::Test {
 
     TestClient client1(
         std::make_unique<AsyncUDPSocket>(env_, std::move(socket)),
-        &fake_clock_);
+        &time_controller_);
 
     SocketAddress client2_addr;
     std::unique_ptr<Socket> socket2 =
         ss_.Create(initial_addr.family(), SOCK_DGRAM);
     TestClient client2(
         std::make_unique<AsyncUDPSocket>(env_, std::move(socket2)),
-        &fake_clock_);
+        &time_controller_);
 
     client2.SendTo("foo", 3, server_addr);
     std::unique_ptr<TestClient::Packet> packet_1 = client1.NextPacket();
@@ -892,10 +912,9 @@ class VirtualSocketServerTest : public ::testing::Test {
   }
 
  protected:
-  ScopedFakeClock fake_clock_;
-  const Environment env_ = CreateTestEnvironment();
   VirtualSocketServer ss_;
-  AutoSocketServerThread thread_;
+  GlobalSimulatedTimeController time_controller_;
+  const Environment env_;
   const SocketAddress kIPv4AnyAddress;
   const SocketAddress kIPv6AnyAddress;
 };
@@ -1087,7 +1106,7 @@ TEST_F(VirtualSocketServerTest, SetSendingBlockedWithUdpSocket) {
   socket1->Bind(kIPv4AnyAddress);
   socket2->Bind(kIPv4AnyAddress);
   TestClient client1(std::make_unique<AsyncUDPSocket>(env_, std::move(socket1)),
-                     &fake_clock_);
+                     &time_controller_);
 
   ss_.SetSendingBlocked(true);
   EXPECT_EQ(-1, client1.SendTo("foo", 3, socket2->GetLocalAddress()));
@@ -1117,20 +1136,21 @@ TEST_F(VirtualSocketServerTest, SetSendingBlockedWithTcpSocket) {
   // Connect sockets.
   EXPECT_EQ(0, socket1->Connect(socket2->GetLocalAddress()));
   EXPECT_EQ(0, socket2->Connect(socket1->GetLocalAddress()));
-  ss_.ProcessMessagesUntilIdle();
+  time_controller_.AdvanceTime(TimeDelta::Zero());
 
-  char data[kBufferSize] = {};
+  std::vector<char> data(kBufferSize);
 
   // First Send call will fill the send buffer but not send anything.
   ss_.SetSendingBlocked(true);
-  EXPECT_EQ(static_cast<int>(kBufferSize), socket1->Send(data, kBufferSize));
-  ss_.ProcessMessagesUntilIdle();
+  EXPECT_EQ(static_cast<int>(kBufferSize),
+            socket1->Send(data.data(), kBufferSize));
+  time_controller_.AdvanceTime(TimeDelta::Zero());
   EXPECT_FALSE(sink.Check(socket1.get(), SSE_WRITE));
   EXPECT_FALSE(sink.Check(socket2.get(), SSE_READ));
   EXPECT_FALSE(socket1->IsBlocking());
 
   // Since the send buffer is full, next Send will result in EWOULDBLOCK.
-  EXPECT_EQ(-1, socket1->Send(data, kBufferSize));
+  EXPECT_EQ(-1, socket1->Send(data.data(), kBufferSize));
   EXPECT_FALSE(sink.Check(socket1.get(), SSE_WRITE));
   EXPECT_FALSE(sink.Check(socket2.get(), SSE_READ));
   EXPECT_TRUE(socket1->IsBlocking());
@@ -1138,7 +1158,7 @@ TEST_F(VirtualSocketServerTest, SetSendingBlockedWithTcpSocket) {
   // When sending is unblocked, the buffered data should be sent and
   // SignalWriteEvent should fire.
   ss_.SetSendingBlocked(false);
-  ss_.ProcessMessagesUntilIdle();
+  time_controller_.AdvanceTime(TimeDelta::Zero());
   EXPECT_TRUE(sink.Check(socket1.get(), SSE_WRITE));
   EXPECT_TRUE(sink.Check(socket2.get(), SSE_READ));
 }
diff --git a/test/time_controller/BUILD.gn b/test/time_controller/BUILD.gn
index 4f9fe256d4..99f2e21706 100644
--- a/test/time_controller/BUILD.gn
+++ b/test/time_controller/BUILD.gn
@@ -60,6 +60,7 @@ rtc_library("time_controller_impl") {
     "../../rtc_base:timeutils",
     "../../rtc_base/synchronization:mutex",
     "../../rtc_base/synchronization:yield_policy",
+    "//third_party/abseil-cpp/absl/base:nullability",
     "//third_party/abseil-cpp/absl/functional:any_invocable",
     "//third_party/abseil-cpp/absl/strings:string_view",
   ]
diff --git a/test/time_controller/simulated_thread.cc b/test/time_controller/simulated_thread.cc
index ebf98cc018..f89c0335b8 100644
--- a/test/time_controller/simulated_thread.cc
+++ b/test/time_controller/simulated_thread.cc
@@ -13,6 +13,7 @@
 #include <memory>
 #include <utility>
 
+#include "absl/base/nullability.h"
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
 #include "api/function_view.h"
@@ -49,17 +50,29 @@ class DummySocketServer : public SocketServer {
 
 }  // namespace
 
+SimulatedThread::SimulatedThread(
+    sim_time_impl::SimulatedTimeControllerImpl* handler,
+    absl::string_view name)
+    : SimulatedThread(handler, name, std::make_unique<DummySocketServer>()) {}
+
 SimulatedThread::SimulatedThread(
     sim_time_impl::SimulatedTimeControllerImpl* handler,
     absl::string_view name,
-    std::unique_ptr<SocketServer> socket_server)
-    : Thread(socket_server ? std::move(socket_server)
-                           : std::make_unique<DummySocketServer>()),
+    std::unique_ptr<SocketServer> absl_nonnull socket_server)
+    : Thread(std::move(socket_server)),
       handler_(handler),
       name_(new char[name.size()]) {
   std::copy_n(name.begin(), name.size(), name_);
 }
 
+SimulatedThread::SimulatedThread(
+    sim_time_impl::SimulatedTimeControllerImpl* handler,
+    absl::string_view name,
+    SocketServer* absl_nonnull socket_server)
+    : Thread(socket_server), handler_(handler), name_(new char[name.size()]) {
+  std::copy_n(name.begin(), name.size(), name_);
+}
+
 SimulatedThread::~SimulatedThread() {
   handler_->Unregister(this);
   delete[] name_;
@@ -118,7 +131,12 @@ void SimulatedThread::Stop() {
 
 SimulatedMainThread::SimulatedMainThread(
     sim_time_impl::SimulatedTimeControllerImpl* handler)
-    : SimulatedThread(handler, "main", nullptr), current_setter_(this) {}
+    : SimulatedThread(handler, "main"), current_setter_(this) {}
+
+SimulatedMainThread::SimulatedMainThread(
+    sim_time_impl::SimulatedTimeControllerImpl* handler,
+    SocketServer* socket_server)
+    : SimulatedThread(handler, "main", socket_server), current_setter_(this) {}
 
 SimulatedMainThread::~SimulatedMainThread() {
   // Removes pending tasks in case they keep shared pointer references to
diff --git a/test/time_controller/simulated_thread.h b/test/time_controller/simulated_thread.h
index 5772b83b67..d9f01e4d86 100644
--- a/test/time_controller/simulated_thread.h
+++ b/test/time_controller/simulated_thread.h
@@ -12,6 +12,7 @@
 
 #include <memory>
 
+#include "absl/base/nullability.h"
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
 #include "api/function_view.h"
@@ -31,9 +32,14 @@ class SimulatedThread : public Thread,
                         public sim_time_impl::SimulatedSequenceRunner {
  public:
   using CurrentThreadSetter = CurrentThreadSetter;
+  SimulatedThread(sim_time_impl::SimulatedTimeControllerImpl* handler,
+                  absl::string_view name);
   SimulatedThread(sim_time_impl::SimulatedTimeControllerImpl* handler,
                   absl::string_view name,
-                  std::unique_ptr<SocketServer> socket_server);
+                  std::unique_ptr<SocketServer> absl_nonnull socket_server);
+  SimulatedThread(sim_time_impl::SimulatedTimeControllerImpl* handler,
+                  absl::string_view name,
+                  SocketServer* absl_nonnull socket_server);
   ~SimulatedThread() override;
 
   void RunReady(Timestamp at_time) override;
@@ -70,6 +76,8 @@ class SimulatedMainThread : public SimulatedThread {
  public:
   explicit SimulatedMainThread(
       sim_time_impl::SimulatedTimeControllerImpl* handler);
+  SimulatedMainThread(sim_time_impl::SimulatedTimeControllerImpl* handler,
+                      SocketServer* socket_server);
   ~SimulatedMainThread() override;
 
  private:
diff --git a/test/time_controller/simulated_time_controller.cc b/test/time_controller/simulated_time_controller.cc
index afd1ed517b..c32efb1dd3 100644
--- a/test/time_controller/simulated_time_controller.cc
+++ b/test/time_controller/simulated_time_controller.cc
@@ -28,10 +28,16 @@
 namespace webrtc {
 
 GlobalSimulatedTimeController::GlobalSimulatedTimeController(
-    Timestamp start_time)
+    Timestamp start_time,
+    SocketServer* socket_server)
     : sim_clock_(start_time.us()), impl_(start_time), yield_policy_(&impl_) {
   global_clock_.SetTime(start_time);
-  auto main_thread = std::make_unique<SimulatedMainThread>(&impl_);
+  std::unique_ptr<SimulatedMainThread> main_thread;
+  if (socket_server) {
+    main_thread = std::make_unique<SimulatedMainThread>(&impl_, socket_server);
+  } else {
+    main_thread = std::make_unique<SimulatedMainThread>(&impl_);
+  }
   impl_.Register(main_thread.get());
   main_thread_ = std::move(main_thread);
 }
diff --git a/test/time_controller/simulated_time_controller.h b/test/time_controller/simulated_time_controller.h
index eff805fe26..0d583ffcee 100644
--- a/test/time_controller/simulated_time_controller.h
+++ b/test/time_controller/simulated_time_controller.h
@@ -33,7 +33,8 @@ namespace webrtc {
 // safe since it modifies global state.
 class GlobalSimulatedTimeController : public TimeController {
  public:
-  explicit GlobalSimulatedTimeController(Timestamp start_time);
+  explicit GlobalSimulatedTimeController(Timestamp start_time,
+                                         SocketServer* socket_server = nullptr);
   ~GlobalSimulatedTimeController() override;
 
   Clock* GetClock() override;
diff --git a/test/time_controller/simulated_time_controller_impl.cc b/test/time_controller/simulated_time_controller_impl.cc
index 10e1dc6649..08f5cdc7d4 100644
--- a/test/time_controller/simulated_time_controller_impl.cc
+++ b/test/time_controller/simulated_time_controller_impl.cc
@@ -15,6 +15,7 @@
 #include <string>
 #include <utility>
 
+#include "absl/base/nullability.h"
 #include "absl/strings/string_view.h"
 #include "api/task_queue/task_queue_base.h"
 #include "api/task_queue/task_queue_factory.h"
@@ -62,9 +63,14 @@ SimulatedTimeControllerImpl::CreateTaskQueue(
 
 std::unique_ptr<Thread> SimulatedTimeControllerImpl::CreateThread(
     const std::string& name,
-    std::unique_ptr<SocketServer> socket_server) {
-  auto thread =
-      std::make_unique<SimulatedThread>(this, name, std::move(socket_server));
+    std::unique_ptr<SocketServer> absl_nullable socket_server) {
+  std::unique_ptr<SimulatedThread> thread;
+  if (socket_server == nullptr) {
+    thread = std::make_unique<SimulatedThread>(this, name);
+  } else {
+    thread =
+        std::make_unique<SimulatedThread>(this, name, std::move(socket_server));
+  }
   Register(thread.get());
   return thread;
 }
diff --git a/test/time_controller/simulated_time_controller_impl.h b/test/time_controller/simulated_time_controller_impl.h
index 5f920d8a01..21db3bf4b9 100644
--- a/test/time_controller/simulated_time_controller_impl.h
+++ b/test/time_controller/simulated_time_controller_impl.h
@@ -17,6 +17,7 @@
 #include <unordered_set>
 #include <vector>
 
+#include "absl/base/nullability.h"
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
 #include "api/location.h"
@@ -67,7 +68,7 @@ class SimulatedTimeControllerImpl : public TaskQueueFactory,
   // Create thread using provided `socket_server`.
   std::unique_ptr<Thread> CreateThread(
       const std::string& name,
-      std::unique_ptr<SocketServer> socket_server)
+      std::unique_ptr<SocketServer> absl_nullable socket_server)
       RTC_LOCKS_EXCLUDED(time_lock_, lock_);
 
   // Runs all runners in `runners_` that has tasks or modules ready for
diff --git a/unsafe_buffers_paths.txt b/unsafe_buffers_paths.txt
index 579df0fe36..ad7cbb7728 100644
--- a/unsafe_buffers_paths.txt
+++ b/unsafe_buffers_paths.txt
@@ -646,7 +646,6 @@
 -rtc_base/time_utils_unittest.cc
 -rtc_base/timestamp_aligner_unittest.cc
 -rtc_base/virtual_socket_server.cc
--rtc_base/virtual_socket_unittest.cc
 -rtc_base/win32.cc
 -rtc_base/zero_memory_unittest.cc
 -rtc_tools/data_channel_benchmark/data_channel_benchmark.cc

From af4890e3474a7021b724db0ca8ef8ae7cb72cab6 Mon Sep 17 00:00:00 2001
From: Per K <perkj@webrtc.org>
Date: Mon, 16 Feb 2026 13:33:52 +0000
Subject: [PATCH 063/472] Add more scream tests
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

LinkCapacity5MbitRepeatedDelaySpikesNoEcn
Tests BWE when every 200ms link capacity is zero for 100ms.

RampupFastOnLinkCapacity50Mbit20MsRttNoEcn
Tests that BWE ramp up fast on good connections

LinkCapacity100Kbit50msRttNoEcn
Test that BWE can adapt to low capacity without causing to large delay.

LinkCapacityIncreaseFrom80KbitTo5MbpsAfter5sVideoCaptureStopedNoEcn
Test that BWE can ramp up from a low capacity to high capacity  even if the video capturer is stopped as long as video encoding max rate is set.

Bug: webrtc:447037083
Change-Id: I2eaadd46fc9b6a5841696ad0b500f4dbe107aa8d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/448120
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46916}
---
 .../bwe_integration_tests/scream_test.cc      | 137 ++++++++++++++++--
 .../bwe_integration_tests/stats_utilities.cc  |  10 ++
 .../bwe_integration_tests/stats_utilities.h   |   3 +
 3 files changed, 140 insertions(+), 10 deletions(-)

diff --git a/test/peer_scenario/bwe_integration_tests/scream_test.cc b/test/peer_scenario/bwe_integration_tests/scream_test.cc
index a6767305e8..a625f7d367 100644
--- a/test/peer_scenario/bwe_integration_tests/scream_test.cc
+++ b/test/peer_scenario/bwe_integration_tests/scream_test.cc
@@ -17,6 +17,7 @@
 
 #include "api/array_view.h"
 #include "api/audio_options.h"
+#include "api/rtp_parameters.h"
 #include "api/scoped_refptr.h"
 #include "api/stats/rtc_stats_report.h"
 #include "api/test/network_emulation/dual_pi2_network_queue.h"
@@ -42,6 +43,7 @@ namespace webrtc {
 namespace {
 
 using test::GetAvailableSendBitrate;
+using test::GetCurrentRoundTripTime;
 using test::GetFirstReportAtOrAfter;
 using test::GetPacketsLost;
 using test::GetPacketsReceived;
@@ -75,6 +77,18 @@ MATCHER_P2(AvailableSendBitrateIsBetween, low, high, "") {
   return false;
 }
 
+MATCHER_P2(CurrentRoundTripTimeIsBetween, low, high, "") {
+  TimeDelta rtt = GetCurrentRoundTripTime(arg);
+  if (rtt > low && rtt < high) {
+    return true;
+  }
+  *result_listener << "the current round trip time is " << rtt.ms()
+                   << "ms, which is not between " << low.ms() << "ms and "
+                   << high.ms() << " ms";
+
+  return false;
+}
+
 std::vector<EmulatedNetworkNode*> CreateNetworkPath(
     NetworkEmulationManager::SimulatedNetworkNode::Builder& network_builder,
     bool use_dual_pi) {
@@ -118,19 +132,17 @@ std::vector<EmulatedNetworkNode*> CreateNetworkPathWithPauseBetween3sAnd6s(
 std::vector<EmulatedNetworkNode*> CreateNetworkPathWithRepeatedPause(
     PeerScenario& s,
     DataRate link_capacity,
-    TimeDelta pause_duration) {
+    TimeDelta pause_duration,
+    TimeDelta repeat_pause_interval) {
   network_behaviour::NetworkConfigSchedule schedule;
   auto initial_config = schedule.add_item();
   initial_config->set_link_capacity_kbps(link_capacity.kbps());
   initial_config->set_queue_delay_ms(10);
   auto updated_capacity = schedule.add_item();
-  updated_capacity->set_time_since_first_sent_packet_ms(1000);
+  updated_capacity->set_time_since_first_sent_packet_ms(
+      repeat_pause_interval.ms());
   updated_capacity->set_link_capacity_kbps(0);
-  updated_capacity = schedule.add_item();
-  updated_capacity->set_time_since_first_sent_packet_ms(1000 +
-                                                        pause_duration.ms());
-  updated_capacity->set_link_capacity_kbps(link_capacity.kbps());
-  schedule.set_repeat_schedule_after_last_ms(4000 + pause_duration.ms());
+  schedule.set_repeat_schedule_after_last_ms(pause_duration.ms());
   SchedulableNetworkNodeBuilder schedulable_builder(*s.net(),
                                                     std::move(schedule));
   return {schedulable_builder.Build()};
@@ -189,7 +201,15 @@ struct SendMediaTestParams {
       {"WebRTC-RFC8888CongestionControlFeedback", "Enabled,offer:true"},
       {"WebRTC-Bwe-ScreamV2", "Enabled"}};
 
+  // Audio is negotiated and sent.
   bool send_audio = true;
+  // Video capturer is producing frames.
+  bool video_capture_enabled = true;
+
+  // Configure and set the max video encoding bitrate. If not set, the max
+  // default per codec and resolution is used.
+  std::optional<DataRate> max_video_bitrate;
+
   PeerScenarioClient::VideoSendTrackConfig caller_video_conf = {
       .generator = {.squares_video =
                         test::FrameGeneratorCapturerConfig::SquaresVideo{
@@ -199,6 +219,7 @@ struct SendMediaTestParams {
                         }}};
 
   TimeDelta test_duration = TimeDelta::Seconds(10);
+  TimeDelta stats_interval = TimeDelta::Seconds(1);
 };
 
 // Sends audio and video from a caller to a callee with symmetric
@@ -215,7 +236,18 @@ SendMediaTestResult SendMediaInOneDirection(SendMediaTestParams params,
   if (params.send_audio) {
     caller->CreateAudio("AUDIO_1", {});
   }
-  caller->CreateVideo("VIDEO_1", params.caller_video_conf);
+  test::PeerScenarioClient::VideoSendTrack video_track =
+      caller->CreateVideo("VIDEO_1", params.caller_video_conf);
+  if (!params.video_capture_enabled) {
+    // Stop the capturer.
+    video_track.source->Stop();
+  }
+  if (params.max_video_bitrate.has_value()) {
+    RtpParameters rtp_parameters = video_track.sender->GetParameters();
+    rtp_parameters.encodings[0].max_bitrate_bps =
+        params.max_video_bitrate->bps();
+    video_track.sender->SetParameters(rtp_parameters);
+  }
 
   s.SimpleConnection(caller, callee, std::move(params.caller_to_callee_path),
                      std::move(params.callee_to_caller_path));
@@ -223,7 +255,7 @@ SendMediaTestResult SendMediaInOneDirection(SendMediaTestParams params,
 
   Timestamp end_time = s.net()->Now() + params.test_duration;
   while (s.net()->Now() < end_time) {
-    s.ProcessMessages(TimeDelta::Seconds(1));
+    s.ProcessMessages(params.stats_interval);
     result.caller_stats.push_back(GetStatsAndProcess(s, caller));
     result.callee_stats.push_back(GetStatsAndProcess(s, callee));
   }
@@ -284,6 +316,36 @@ TEST(ScreamTest,
                                               DataRate::KilobitsPerSec(5000))));
 }
 
+TEST(ScreamTest,
+     MaybeTest(
+         LinkCapacityIncreaseFrom80KbitTo5MbpsAfter5sVideoCaptureStopedNoEcn)) {
+  PeerScenario s(*testing::UnitTest::GetInstance()->current_test_info());
+  SendMediaTestParams params;
+  params.video_capture_enabled = false;
+  params.max_video_bitrate = DataRate::KilobitsPerSec(3000);
+  params.caller_to_callee_path = CreateNetworkPathWithChangedCapacityAfter5s(
+      s, DataRate::KilobitsPerSec(80), DataRate::KilobitsPerSec(5000));
+  params.callee_to_caller_path =
+      CreateNetworkPath(s, /*use_dual_pi= */ false,
+                        DataRate::KilobitsPerSec(3000), TimeDelta::Millis(25));
+  SendMediaTestResult result = SendMediaInOneDirection(std::move(params), s);
+  // Stats 2-5s
+  EXPECT_THAT(
+      result.caller().subspan(1, 3),
+      Each(AvailableSendBitrateIsBetween(DataRate::KilobitsPerSec(10),
+                                         DataRate::KilobitsPerSec(100))));
+  // TODO: bugs.webrtc.org/447037083 - Improve Scream at low bitrates. Consider
+  // lowering min ref window.
+  EXPECT_THAT(result.caller()[3],
+              CurrentRoundTripTimeIsBetween(TimeDelta::Millis(40),
+                                            TimeDelta::Millis(500)));
+  // Stats after 9s
+  // TODO: bugs.webrtc.org/447037083 - Ensure BWE does ramp up.
+  EXPECT_THAT(result.caller_stats.back(),
+              AvailableSendBitrateIsBetween(DataRate::KilobitsPerSec(200),
+                                            DataRate::KilobitsPerSec(5000)));
+}
+
 TEST(ScreamTest, MaybeTest(LinkCapacity600KbpsRtt20msNoEcn)) {
   PeerScenario s(*testing::UnitTest::GetInstance()->current_test_info());
   SendMediaTestParams params;
@@ -568,7 +630,8 @@ TEST(ScreamTest, MaybeTest(ScreencastSlideChangeRepeatedDelaySpikes)) {
   PeerScenario s(*testing::UnitTest::GetInstance()->current_test_info());
   SendMediaTestParams params{.test_duration = TimeDelta::Seconds(20)};
   params.caller_to_callee_path = CreateNetworkPathWithRepeatedPause(
-      s, DataRate::KilobitsPerSec(4000), TimeDelta::Millis(500));
+      s, DataRate::KilobitsPerSec(4000), TimeDelta::Millis(500),
+      TimeDelta::Seconds(4));
   params.callee_to_caller_path =
       CreateNetworkPath(s, /*use_dual_pi= */ false,
                         DataRate::KilobitsPerSec(2000), TimeDelta::Millis(25));
@@ -586,6 +649,60 @@ TEST(ScreamTest, MaybeTest(ScreencastSlideChangeRepeatedDelaySpikes)) {
                                               DataRate::KilobitsPerSec(4000))));
 }
 
+TEST(ScreamTest, MaybeTest(LinkCapacity5MbitRepeatedDelaySpikesNoEcn)) {
+  PeerScenario s(*testing::UnitTest::GetInstance()->current_test_info());
+  SendMediaTestParams params{.test_duration = TimeDelta::Seconds(20)};
+  params.caller_to_callee_path = CreateNetworkPathWithRepeatedPause(
+      s, DataRate::KilobitsPerSec(5000), TimeDelta::Millis(100),
+      TimeDelta::Millis(200));
+  params.callee_to_caller_path = CreateNetworkPathWithRepeatedPause(
+      s, DataRate::KilobitsPerSec(5000), TimeDelta::Millis(100),
+      TimeDelta::Millis(200));
+  SendMediaTestResult result = SendMediaInOneDirection(std::move(params), s);
+
+  // TODO: bugs.webrtc.org/447037083 - consider improving resilience by limiting
+  // queue delay from one feedback.
+  EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
+                                              DataRate::KilobitsPerSec(500),
+                                              DataRate::KilobitsPerSec(5000))));
+}
+
+TEST(ScreamTest, MaybeTest(RampupFastOnLinkCapacity50Mbit20MsRttNoEcn)) {
+  PeerScenario s(*testing::UnitTest::GetInstance()->current_test_info());
+  SendMediaTestParams params{.test_duration = TimeDelta::Millis(300),
+                             .stats_interval = TimeDelta::Millis(100)};
+  params.caller_to_callee_path =
+      CreateNetworkPath(s, /*use_dual_pi= */ false,
+                        DataRate::KilobitsPerSec(50000), TimeDelta::Millis(10));
+  params.callee_to_caller_path =
+      CreateNetworkPath(s, /*use_dual_pi= */ false,
+                        DataRate::KilobitsPerSec(50000), TimeDelta::Millis(10));
+  SendMediaTestResult result = SendMediaInOneDirection(std::move(params), s);
+
+  // TODO: bugs.webrtc.org/447037083 - On a good network, we should be able to
+  // let the bitrate increase by 50% every RTT.
+  EXPECT_THAT(result.caller_stats.back(),
+              AvailableSendBitrateIsBetween(DataRate::KilobitsPerSec(500),
+                                            DataRate::KilobitsPerSec(5000)));
+}
+
+TEST(ScreamTest, MaybeTest(LinkCapacity100Kbit50msRttNoEcn)) {
+  PeerScenario s(*testing::UnitTest::GetInstance()->current_test_info());
+  SendMediaTestParams params;
+  params.caller_to_callee_path =
+      CreateNetworkPath(s, /*use_dual_pi= */ false,
+                        DataRate::KilobitsPerSec(100), TimeDelta::Millis(25));
+  params.callee_to_caller_path =
+      CreateNetworkPath(s, /*use_dual_pi= */ false,
+                        DataRate::KilobitsPerSec(100), TimeDelta::Millis(25));
+  SendMediaTestResult result = SendMediaInOneDirection(std::move(params), s);
+
+  // TODO: bugs.webrtc.org/447037083 - Investigate how the delay can be reduced.
+  EXPECT_THAT(result.caller(), Each(AvailableSendBitrateIsBetween(
+                                   DataRate::KilobitsPerSec(10),
+                                   DataRate::KilobitsPerSec(150))));
+}
+
 TEST(ScreamTest, MaybeTest(LinkCapacity1MbitRtt50msWithShortQueuesNoEcn)) {
   PeerScenario s(*testing::UnitTest::GetInstance()->current_test_info());
   SendMediaTestParams params;
diff --git a/test/peer_scenario/bwe_integration_tests/stats_utilities.cc b/test/peer_scenario/bwe_integration_tests/stats_utilities.cc
index 6a7dfa874b..49a6ca2fae 100644
--- a/test/peer_scenario/bwe_integration_tests/stats_utilities.cc
+++ b/test/peer_scenario/bwe_integration_tests/stats_utilities.cc
@@ -74,6 +74,16 @@ TimeDelta GetAverageRoundTripTime(
                             *stats[0]->responses_received);
 }
 
+TimeDelta GetCurrentRoundTripTime(
+    const scoped_refptr<const RTCStatsReport>& report) {
+  auto stats = report->GetStatsOfType<RTCIceCandidatePairStats>();
+  if (stats.empty() || !stats[0]->current_round_trip_time.has_value()) {
+    return TimeDelta::PlusInfinity();
+  }
+
+  return TimeDelta::Seconds(*stats[0]->current_round_trip_time);
+}
+
 int64_t GetPacketsSentWithEct1(
     const scoped_refptr<const RTCStatsReport>& report) {
   auto stats = report->GetStatsOfType<RTCOutboundRtpStreamStats>();
diff --git a/test/peer_scenario/bwe_integration_tests/stats_utilities.h b/test/peer_scenario/bwe_integration_tests/stats_utilities.h
index 83128424ec..ed34399114 100644
--- a/test/peer_scenario/bwe_integration_tests/stats_utilities.h
+++ b/test/peer_scenario/bwe_integration_tests/stats_utilities.h
@@ -40,6 +40,9 @@ DataRate GetAvailableSendBitrate(
 TimeDelta GetAverageRoundTripTime(
     const scoped_refptr<const RTCStatsReport>& report);
 
+TimeDelta GetCurrentRoundTripTime(
+    const scoped_refptr<const RTCStatsReport>& report);
+
 int64_t GetPacketsSentWithEct1(
     const scoped_refptr<const RTCStatsReport>& report);
 

From 2761c47ae118cdb0d95f2667f4cbf15a0a71e3c7 Mon Sep 17 00:00:00 2001
From: Evan Shrubsole <eshr@webrtc.org>
Date: Wed, 11 Feb 2026 12:21:45 +0000
Subject: [PATCH 064/472] Use injected clock in dcsctp_socket_network_test

Bug: webrtc:42223992
Change-Id: I45f662b6d9c123aff297fb87122697486a6a6964
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/448341
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46917}
---
 net/dcsctp/socket/BUILD.gn                    |  2 +
 net/dcsctp/socket/DEPS                        |  1 +
 .../socket/dcsctp_socket_network_test.cc      | 48 ++++++++++++-------
 3 files changed, 35 insertions(+), 16 deletions(-)

diff --git a/net/dcsctp/socket/BUILD.gn b/net/dcsctp/socket/BUILD.gn
index d69e2d56f8..951bfffabe 100644
--- a/net/dcsctp/socket/BUILD.gn
+++ b/net/dcsctp/socket/BUILD.gn
@@ -222,6 +222,7 @@ if (rtc_include_tests) {
       "../../../rtc_base:stringutils",
       "../../../rtc_base:threading",
       "../../../rtc_base:timeutils",
+      "../../../system_wrappers",
       "../../../test:test_support",
       "../common:handover_testing",
       "../common:internal_types",
@@ -243,6 +244,7 @@ if (rtc_include_tests) {
       "../timer:task_queue_timeout",
       "../tx:mock_send_queue",
       "../tx:retransmission_queue",
+      "//third_party/abseil-cpp/absl/base:nullability",
       "//third_party/abseil-cpp/absl/flags:flag",
       "//third_party/abseil-cpp/absl/memory",
       "//third_party/abseil-cpp/absl/strings:string_view",
diff --git a/net/dcsctp/socket/DEPS b/net/dcsctp/socket/DEPS
index d4966290e3..fa1def17eb 100644
--- a/net/dcsctp/socket/DEPS
+++ b/net/dcsctp/socket/DEPS
@@ -1,5 +1,6 @@
 specific_include_rules = {
   "dcsctp_socket_network_test.cc": [
     "+call",
+    "+system_wrappers/include/clock.h",
   ]
 }
diff --git a/net/dcsctp/socket/dcsctp_socket_network_test.cc b/net/dcsctp/socket/dcsctp_socket_network_test.cc
index ff8d68ae6c..41b71ba475 100644
--- a/net/dcsctp/socket/dcsctp_socket_network_test.cc
+++ b/net/dcsctp/socket/dcsctp_socket_network_test.cc
@@ -16,6 +16,7 @@
 #include <utility>
 #include <vector>
 
+#include "absl/base/nullability.h"
 #include "absl/strings/string_view.h"
 #include "api/array_view.h"
 #include "api/task_queue/pending_task_safety_flag.h"
@@ -42,7 +43,7 @@
 #include "rtc_base/strings/string_builder.h"
 #include "rtc_base/strings/string_format.h"
 #include "rtc_base/thread.h"
-#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/clock.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
 
@@ -147,10 +148,12 @@ class SctpActor : public DcSctpSocketCallbacks {
  public:
   SctpActor(absl::string_view name,
             BoundSocket& emulated_socket,
-            const DcSctpOptions& sctp_options)
+            const DcSctpOptions& sctp_options,
+            webrtc::Clock* absl_nonnull clock)
       : log_prefix_(std::string(name) + ": "),
         thread_(webrtc::Thread::Current()),
         emulated_socket_(emulated_socket),
+        clock_(*clock),
         timeout_factory_(
             *thread_,
             [this]() { return TimeMs(Now().ms()); },
@@ -197,7 +200,7 @@ class SctpActor : public DcSctpSocketCallbacks {
     return timeout_factory_.CreateTimeout(precision);
   }
 
-  Timestamp Now() override { return Timestamp::Millis(webrtc::TimeMillis()); }
+  Timestamp Now() override { return clock_.CurrentTime(); }
 
   uint32_t GetRandomInt(uint32_t low, uint32_t high) override {
     return random_.Rand(low, high);
@@ -313,7 +316,7 @@ class SctpActor : public DcSctpSocketCallbacks {
   std::string log_prefix() const {
     webrtc::StringBuilder sb;
     sb << log_prefix_;
-    sb << webrtc::TimeMillis();
+    sb << clock_.CurrentTime().ms();
     sb << ": ";
     return sb.Release();
   }
@@ -322,6 +325,7 @@ class SctpActor : public DcSctpSocketCallbacks {
   const std::string log_prefix_;
   webrtc::Thread* thread_;
   BoundSocket& emulated_socket_;
+  webrtc::Clock& clock_;
   TaskQueueTimeoutFactory timeout_factory_;
   webrtc::Random random_;
   DcSctpSocket sctp_socket_;
@@ -376,8 +380,10 @@ TEST_F(DcSctpSocketNetworkTest, CanConnectAndShutdown) {
   webrtc::BuiltInNetworkBehaviorConfig pipe_config;
   MakeNetwork(pipe_config);
 
-  SctpActor sender("A", emulated_socket_a_, options_);
-  SctpActor receiver("Z", emulated_socket_z_, options_);
+  SctpActor sender("A", emulated_socket_a_, options_,
+                   emulation_->time_controller()->GetClock());
+  SctpActor receiver("Z", emulated_socket_z_, options_,
+                     emulation_->time_controller()->GetClock());
   EXPECT_THAT(sender.sctp_socket().state(), SocketState::kClosed);
 
   sender.sctp_socket().Connect();
@@ -394,8 +400,10 @@ TEST_F(DcSctpSocketNetworkTest, CanSendLargeMessage) {
   pipe_config.queue_delay_ms = 30;
   MakeNetwork(pipe_config);
 
-  SctpActor sender("A", emulated_socket_a_, options_);
-  SctpActor receiver("Z", emulated_socket_z_, options_);
+  SctpActor sender("A", emulated_socket_a_, options_,
+                   emulation_->time_controller()->GetClock());
+  SctpActor receiver("Z", emulated_socket_z_, options_,
+                     emulation_->time_controller()->GetClock());
   sender.sctp_socket().Connect();
 
   constexpr size_t kPayloadSize = 100 * 1024;
@@ -421,8 +429,10 @@ TEST_F(DcSctpSocketNetworkTest, CanSendMessagesReliablyWithLowBandwidth) {
   pipe_config.link_capacity = DataRate::KilobitsPerSec(1000);
   MakeNetwork(pipe_config);
 
-  SctpActor sender("A", emulated_socket_a_, options_);
-  SctpActor receiver("Z", emulated_socket_z_, options_);
+  SctpActor sender("A", emulated_socket_a_, options_,
+                   emulation_->time_controller()->GetClock());
+  SctpActor receiver("Z", emulated_socket_z_, options_,
+                     emulation_->time_controller()->GetClock());
   sender.sctp_socket().Connect();
 
   sender.SetActorMode(ActorMode::kThroughputSender);
@@ -450,8 +460,10 @@ TEST_F(DcSctpSocketNetworkTest,
   pipe_config.link_capacity = DataRate::KilobitsPerSec(18000);
   MakeNetwork(pipe_config);
 
-  SctpActor sender("A", emulated_socket_a_, options_);
-  SctpActor receiver("Z", emulated_socket_z_, options_);
+  SctpActor sender("A", emulated_socket_a_, options_,
+                   emulation_->time_controller()->GetClock());
+  SctpActor receiver("Z", emulated_socket_z_, options_,
+                     emulation_->time_controller()->GetClock());
   sender.sctp_socket().Connect();
 
   sender.SetActorMode(ActorMode::kThroughputSender);
@@ -478,8 +490,10 @@ TEST_F(DcSctpSocketNetworkTest, CanSendMessagesReliablyWithMuchPacketLoss) {
   config.loss_percent = 1;
   MakeNetwork(config);
 
-  SctpActor sender("A", emulated_socket_a_, options_);
-  SctpActor receiver("Z", emulated_socket_z_, options_);
+  SctpActor sender("A", emulated_socket_a_, options_,
+                   emulation_->time_controller()->GetClock());
+  SctpActor receiver("Z", emulated_socket_z_, options_,
+                     emulation_->time_controller()->GetClock());
   sender.sctp_socket().Connect();
 
   sender.SetActorMode(ActorMode::kThroughputSender);
@@ -507,8 +521,10 @@ TEST_F(DcSctpSocketNetworkTest, DCSCTP_NDEBUG_TEST(HasHighBandwidth)) {
   pipe_config.queue_delay_ms = 30;
   MakeNetwork(pipe_config);
 
-  SctpActor sender("A", emulated_socket_a_, options_);
-  SctpActor receiver("Z", emulated_socket_z_, options_);
+  SctpActor sender("A", emulated_socket_a_, options_,
+                   emulation_->time_controller()->GetClock());
+  SctpActor receiver("Z", emulated_socket_z_, options_,
+                     emulation_->time_controller()->GetClock());
   sender.sctp_socket().Connect();
 
   sender.SetActorMode(ActorMode::kThroughputSender);

From 91ebeb8408b69cbda1bed3e86f5ff18637f703ab Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Erik=20Spr=C3=A5ng?= <sprang@webrtc.org>
Date: Mon, 16 Feb 2026 13:21:55 +0100
Subject: [PATCH 065/472] Add OnFrameDropped support to classes in video/
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This CL adds simple plumbing/pass through of the new OnFrameDropped
callback in the classes in the video/ directory.

Bug: webrtc:467444018
Change-Id: Iad9d558740124841c552a5ff3ad836db6c7e0ea4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449685
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46918}
---
 .../evaluation/webrtc_picture_pair_provider.cc           | 9 +++++++++
 .../evaluation/webrtc_picture_pair_provider.h            | 3 +++
 video/frame_dumping_encoder.cc                           | 6 ++++++
 video/video_quality_test.cc                              | 6 ++++++
 4 files changed, 24 insertions(+)

diff --git a/video/corruption_detection/evaluation/webrtc_picture_pair_provider.cc b/video/corruption_detection/evaluation/webrtc_picture_pair_provider.cc
index b5b82be9e8..509157aca9 100644
--- a/video/corruption_detection/evaluation/webrtc_picture_pair_provider.cc
+++ b/video/corruption_detection/evaluation/webrtc_picture_pair_provider.cc
@@ -359,6 +359,15 @@ void WebRtcEncoderDecoderPicturePairProvider::OnDroppedFrame(
   encoded_image_.reset();
 }
 
+void WebRtcEncoderDecoderPicturePairProvider::OnFrameDropped(
+    uint32_t rtp_timestamp,
+    int spatial_id,
+    bool is_end_of_temporal_unit) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+
+  encoded_image_.reset();
+}
+
 int32_t WebRtcEncoderDecoderPicturePairProvider::Decoded(
     VideoFrame& decoded_image) {
   RTC_DCHECK_RUN_ON(&sequence_checker_);
diff --git a/video/corruption_detection/evaluation/webrtc_picture_pair_provider.h b/video/corruption_detection/evaluation/webrtc_picture_pair_provider.h
index 5b9b80b4a6..ea5cd37092 100644
--- a/video/corruption_detection/evaluation/webrtc_picture_pair_provider.h
+++ b/video/corruption_detection/evaluation/webrtc_picture_pair_provider.h
@@ -70,6 +70,9 @@ class WebRtcEncoderDecoderPicturePairProvider : public PicturePairProvider,
   Result OnEncodedImage(const EncodedImage& encoded_image,
                         const CodecSpecificInfo*) override;
   void OnDroppedFrame(DropReason reason) override;
+  void OnFrameDropped(uint32_t rtp_timestamp,
+                      int spatial_id,
+                      bool is_end_of_temporal_unit) override;
 
   // Overridden from DecodedImageCallback.
   int32_t Decoded(VideoFrame& decoded_image) override;
diff --git a/video/frame_dumping_encoder.cc b/video/frame_dumping_encoder.cc
index ff16231bff..3377989198 100644
--- a/video/frame_dumping_encoder.cc
+++ b/video/frame_dumping_encoder.cc
@@ -98,6 +98,12 @@ class FrameDumpingEncoder : public VideoEncoder, public EncodedImageCallback {
   void OnDroppedFrame(DropReason reason) override {
     callback_->OnDroppedFrame(reason);
   }
+  void OnFrameDropped(uint32_t rtp_timestamp,
+                      int spatial_id,
+                      bool is_end_of_temporal_unit) override {
+    callback_->OnFrameDropped(rtp_timestamp, spatial_id,
+                              is_end_of_temporal_unit);
+  }
 
  private:
   std::string FilenameFromSimulcastIndex(int index)
diff --git a/video/video_quality_test.cc b/video/video_quality_test.cc
index 0094e2cbd4..b219817964 100644
--- a/video/video_quality_test.cc
+++ b/video/video_quality_test.cc
@@ -298,6 +298,12 @@ class QualityTestVideoEncoder : public VideoEncoder,
   void OnDroppedFrame(DropReason reason) override {
     callback_->OnDroppedFrame(reason);
   }
+  void OnFrameDropped(uint32_t rtp_timestamp,
+                      int spatial_id,
+                      bool is_end_of_temporal_unit) override {
+    callback_->OnFrameDropped(rtp_timestamp, spatial_id,
+                              is_end_of_temporal_unit);
+  }
 
   const std::unique_ptr<VideoEncoder> encoder_;
   const double overshoot_factor_;

From 190c6680d95cdf03728bf328fd41ef8aa603e827 Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Mon, 16 Feb 2026 15:03:26 +0000
Subject: [PATCH 066/472] Define RTC_HARDENING_ASSERT and use in rtc::ArrayView
 bounds check

This will assert on out-of-bounds access if RTC_HARDENING is set
This allows RTC_HARDENING to control whether these checks are treated as CHECK or as DCHECK

If RTC_HARDENING is set, the assert will be checked, both in debug and production builds.

Bug: webrtc:367205686
Change-Id: I456edabc7107a5b6edb82183fa86d160c83642d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362960
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46919}
---
 api/array_view.h  |  7 +++++--
 rtc_base/checks.h | 12 ++++++++++++
 2 files changed, 17 insertions(+), 2 deletions(-)

diff --git a/api/array_view.h b/api/array_view.h
index 7ff065f892..6c2e4938c7 100644
--- a/api/array_view.h
+++ b/api/array_view.h
@@ -266,8 +266,11 @@ class ArrayView final : public array_view_internal::ArrayViewBase<T, Size> {
   // const, because the ArrayView doesn't own the array. (To prevent mutation,
   // use a const element type.)
   T& operator[](size_t idx) const {
-    RTC_DCHECK_LT(idx, this->size());
-    RTC_DCHECK(this->data());
+    // RTC_HARDENING_ASSERT is used because this is tunable at compile time
+    // to be on or off in production, depending on the choices of the
+    // embedder.
+    RTC_HARDENING_ASSERT(idx < this->size());
+    RTC_HARDENING_ASSERT(this->data());
     return this->data()[idx];
   }
   T* begin() const { return this->data(); }
diff --git a/rtc_base/checks.h b/rtc_base/checks.h
index 992c98d4da..26f5da2754 100644
--- a/rtc_base/checks.h
+++ b/rtc_base/checks.h
@@ -526,4 +526,16 @@ inline T CheckedDivExact(T a, T b) {
 
 #endif  // __cplusplus
 
+// Hardening assert. This assert can be turned on and off independently
+// of debug mode / non debug mode.
+// If enabled, it turns into RTC_CHECK.
+// If disabled, it turns into RTC_DCHECK.
+// This resembles the behavior of ABSL_HARDENING_ASSERT,
+// but is not affected by NDEBUG.
+#ifdef RTC_HARDENING
+#define RTC_HARDENING_ASSERT(x) RTC_CHECK(x)
+#else
+#define RTC_HARDENING_ASSERT(x) RTC_DCHECK(x)
+#endif
+
 #endif  // RTC_BASE_CHECKS_H_

From e56ca1554b1ecdc7376d2f1ea997e040ffb2ccff Mon Sep 17 00:00:00 2001
From: Per K <perkj@webrtc.org>
Date: Tue, 17 Feb 2026 07:08:45 +0000
Subject: [PATCH 067/472] Relax expectation in test
 LinkCapacityIncreaseFrom80KbitTo5MbpsAfter5sVideoCaptureStopedNoEcn

Done to avoid flake on IOS debug.

Bug: webrtc:447037083
Change-Id: Ie880538890fa35319620de22c80c06f67a9ed35d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449840
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46920}
---
 test/peer_scenario/bwe_integration_tests/scream_test.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/test/peer_scenario/bwe_integration_tests/scream_test.cc b/test/peer_scenario/bwe_integration_tests/scream_test.cc
index a625f7d367..9adbf0b90f 100644
--- a/test/peer_scenario/bwe_integration_tests/scream_test.cc
+++ b/test/peer_scenario/bwe_integration_tests/scream_test.cc
@@ -333,7 +333,7 @@ TEST(ScreamTest,
   EXPECT_THAT(
       result.caller().subspan(1, 3),
       Each(AvailableSendBitrateIsBetween(DataRate::KilobitsPerSec(10),
-                                         DataRate::KilobitsPerSec(100))));
+                                         DataRate::KilobitsPerSec(150))));
   // TODO: bugs.webrtc.org/447037083 - Improve Scream at low bitrates. Consider
   // lowering min ref window.
   EXPECT_THAT(result.caller()[3],

From 770ba4d4c86a257ae62f578cf345f186c75d5549 Mon Sep 17 00:00:00 2001
From: Evan Shrubsole <eshr@webrtc.org>
Date: Thu, 5 Feb 2026 16:35:00 +0000
Subject: [PATCH 068/472] Reland "Use injected clock in
 android_video_track_source"
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This is a reland of commit 3b4a327fe1f6531050b6dc50c6573f20b587c91a

Safe to reland: Keeps the old definition of CreateJavaVideoSource.
Tested internal compilation.

Original change's description:
> Use injected clock in android_video_track_source
>
> This adds Environment to the OwnedFactoryAndThreads for propagation to
> utilities created by the PeerConnectionFactory.
>
> Bug: webrtc:42223992
> Change-Id: I08550f7e06f77cdeeb911ec7fb3b94d26a6a6964
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/446602
> Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
> Reviewed-by: Zoé Lepaul <zlep@webrtc.org>
> Commit-Queue: Zoé Lepaul <zlep@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#46842}

Bug: webrtc:42223992
Change-Id: Id2224db3107b006c51f5cf90d03635585eaba764
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/447002
Reviewed-by: Zoé Lepaul <zlep@webrtc.org>
Commit-Queue: Zoé Lepaul <zlep@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46921}
---
 examples/androidnativeapi/BUILD.gn            |  2 ++
 .../jni/android_call_client.cc                | 15 +++++++-----
 .../jni/android_call_client.h                 |  2 ++
 sdk/android/BUILD.gn                          |  5 ++++
 .../peerconnection/peer_connection_factory.cc |  6 +++--
 .../peerconnection/peer_connection_factory.h  |  4 +++-
 sdk/android/native_api/video/video_source.cc  | 18 +++++++++------
 sdk/android/native_api/video/video_source.h   | 19 +++++++++++++--
 .../peer_connection_factory_unittest.cc       | 15 ++++++------
 .../video/video_source_unittest.cc            | 19 ++++++++++-----
 .../src/jni/android_video_track_source.cc     | 23 +++++++++++++++----
 .../src/jni/android_video_track_source.h      | 10 ++++++++
 .../src/jni/pc/owned_factory_and_threads.cc   | 16 +++++++++++++
 .../src/jni/pc/owned_factory_and_threads.h    | 13 ++++++++++-
 .../src/jni/pc/peer_connection_factory.cc     | 18 ++++++++-------
 .../src/jni/pc/peer_connection_factory.h      |  4 +++-
 sdk/android/src/jni/pc/video.cc               |  6 +++--
 sdk/android/src/jni/pc/video.h                |  6 ++++-
 18 files changed, 153 insertions(+), 48 deletions(-)

diff --git a/examples/androidnativeapi/BUILD.gn b/examples/androidnativeapi/BUILD.gn
index adc9a332a0..ff4458e79f 100644
--- a/examples/androidnativeapi/BUILD.gn
+++ b/examples/androidnativeapi/BUILD.gn
@@ -54,6 +54,8 @@ if (is_android) {
       "../../api:rtc_error",
       "../../api:scoped_refptr",
       "../../api:sequence_checker",
+      "../../api/environment",
+      "../../api/environment:environment_factory",
       "../../api/video:video_frame",
       "../../rtc_base:checks",
       "../../rtc_base:logging",
diff --git a/examples/androidnativeapi/jni/android_call_client.cc b/examples/androidnativeapi/jni/android_call_client.cc
index e2fa4e79b4..b81d159f05 100644
--- a/examples/androidnativeapi/jni/android_call_client.cc
+++ b/examples/androidnativeapi/jni/android_call_client.cc
@@ -19,12 +19,12 @@
 #include "api/create_modular_peer_connection_factory.h"
 #include "api/data_channel_interface.h"
 #include "api/enable_media_with_defaults.h"
+#include "api/environment/environment_factory.h"
 #include "api/jsep.h"
 #include "api/make_ref_counted.h"
 #include "api/media_stream_interface.h"
 #include "api/peer_connection_interface.h"
 #include "api/rtc_error.h"
-#include "api/rtc_event_log/rtc_event_log_factory.h"
 #include "api/rtp_transceiver_interface.h"
 #include "api/scoped_refptr.h"
 #include "api/sequence_checker.h"
@@ -93,7 +93,9 @@ class SetLocalSessionDescriptionObserver
 }  // namespace
 
 AndroidCallClient::AndroidCallClient()
-    : call_started_(false), pc_observer_(std::make_unique<PCObserver>(this)) {
+    : env_(webrtc::CreateEnvironment()),
+      call_started_(false),
+      pc_observer_(std::make_unique<PCObserver>(this)) {
   thread_checker_.Detach();
   CreatePeerConnectionFactory();
 }
@@ -115,9 +117,10 @@ void AndroidCallClient::Call(JNIEnv* env,
   local_sink_ = webrtc::JavaToNativeVideoSink(env, local_sink.obj());
   remote_sink_ = webrtc::JavaToNativeVideoSink(env, remote_sink.obj());
 
-  video_source_ = webrtc::CreateJavaVideoSource(env, signaling_thread_.get(),
-                                                /* is_screencast= */ false,
-                                                /* align_timestamps= */ true);
+  video_source_ =
+      webrtc::CreateJavaVideoSource(env, signaling_thread_.get(),
+                                    /* is_screencast= */ false,
+                                    /* align_timestamps= */ true, env_);
 
   CreatePeerConnection();
   Connect();
@@ -168,10 +171,10 @@ void AndroidCallClient::CreatePeerConnectionFactory() {
   RTC_CHECK(signaling_thread_->Start()) << "Failed to start thread";
 
   webrtc::PeerConnectionFactoryDependencies pcf_deps;
+  pcf_deps.env = env_;
   pcf_deps.network_thread = network_thread_.get();
   pcf_deps.worker_thread = worker_thread_.get();
   pcf_deps.signaling_thread = signaling_thread_.get();
-  pcf_deps.event_log_factory = std::make_unique<webrtc::RtcEventLogFactory>();
 
   pcf_deps.video_encoder_factory =
       std::make_unique<webrtc::InternalEncoderFactory>();
diff --git a/examples/androidnativeapi/jni/android_call_client.h b/examples/androidnativeapi/jni/android_call_client.h
index 6fffdada8e..5853ae622a 100644
--- a/examples/androidnativeapi/jni/android_call_client.h
+++ b/examples/androidnativeapi/jni/android_call_client.h
@@ -15,6 +15,7 @@
 
 #include <memory>
 
+#include "api/environment/environment.h"
 #include "api/peer_connection_interface.h"
 #include "api/scoped_refptr.h"
 #include "api/sequence_checker.h"
@@ -49,6 +50,7 @@ class AndroidCallClient {
   void CreatePeerConnection() RTC_RUN_ON(thread_checker_);
   void Connect() RTC_RUN_ON(thread_checker_);
 
+  const webrtc::Environment env_;
   webrtc::SequenceChecker thread_checker_;
 
   bool call_started_ RTC_GUARDED_BY(thread_checker_);
diff --git a/sdk/android/BUILD.gn b/sdk/android/BUILD.gn
index 1673cb90bb..35d1ea4176 100644
--- a/sdk/android/BUILD.gn
+++ b/sdk/android/BUILD.gn
@@ -617,6 +617,7 @@ if (current_os == "linux" || is_android) {
       "../../api:scoped_refptr",
       "../../api:sequence_checker",
       "../../api/audio:audio_processing",
+      "../../api/environment",
       "../../api/task_queue:pending_task_safety_flag",
       "../../rtc_base:checks",
       "../../rtc_base:ip_address",
@@ -741,6 +742,7 @@ if (current_os == "linux" || is_android) {
       "../../rtc_base:timestamp_aligner",
       "../../rtc_base:timeutils",
       "../../rtc_base/synchronization:mutex",
+      "../../system_wrappers",
       "//third_party/abseil-cpp/absl/memory",
       "//third_party/jni_zero",
       "//third_party/libyuv",
@@ -1147,6 +1149,7 @@ if (current_os == "linux" || is_android) {
       "../../api:jsep",
       "../../api:libjingle_peerconnection_api",
       "../../api:scoped_refptr",
+      "../../api/environment",
       "../../api/video_codecs:video_codecs_api",
       "../../rtc_base:socket_factory",
       "../../rtc_base:threading",
@@ -1189,11 +1192,13 @@ if (current_os == "linux" || is_android) {
       "../../api:make_ref_counted",
       "../../api:media_stream_interface",
       "../../api:scoped_refptr",
+      "../../api/environment",
       "../../api/video:recordable_encoded_frame",
       "../../api/video:video_frame",
       "../../api/video:video_rtp_headers",
       "../../rtc_base:refcount",
       "../../rtc_base:threading",
+      "//third_party/abseil-cpp/absl/base:core_headers",
       "//third_party/jni_zero",
     ]
   }
diff --git a/sdk/android/native_api/peerconnection/peer_connection_factory.cc b/sdk/android/native_api/peerconnection/peer_connection_factory.cc
index 74ac367453..20a4ee7cf0 100644
--- a/sdk/android/native_api/peerconnection/peer_connection_factory.cc
+++ b/sdk/android/native_api/peerconnection/peer_connection_factory.cc
@@ -14,6 +14,7 @@
 #include <memory>
 #include <utility>
 
+#include "api/environment/environment.h"
 #include "api/peer_connection_interface.h"
 #include "api/scoped_refptr.h"
 #include "rtc_base/socket_factory.h"
@@ -28,10 +29,11 @@ jobject NativeToJavaPeerConnectionFactory(
     std::unique_ptr<SocketFactory> socket_factory,
     std::unique_ptr<Thread> network_thread,
     std::unique_ptr<Thread> worker_thread,
-    std::unique_ptr<Thread> signaling_thread) {
+    std::unique_ptr<Thread> signaling_thread,
+    const Environment& env) {
   return jni::NativeToJavaPeerConnectionFactory(
       jni, pcf, std::move(socket_factory), std::move(network_thread),
-      std::move(worker_thread), std::move(signaling_thread));
+      std::move(worker_thread), std::move(signaling_thread), env);
 }
 
 }  // namespace webrtc
diff --git a/sdk/android/native_api/peerconnection/peer_connection_factory.h b/sdk/android/native_api/peerconnection/peer_connection_factory.h
index 1f5d845fd5..1252626ccf 100644
--- a/sdk/android/native_api/peerconnection/peer_connection_factory.h
+++ b/sdk/android/native_api/peerconnection/peer_connection_factory.h
@@ -15,6 +15,7 @@
 
 #include <memory>
 
+#include "api/environment/environment.h"
 #include "api/peer_connection_interface.h"
 #include "api/scoped_refptr.h"
 #include "rtc_base/socket_factory.h"
@@ -29,7 +30,8 @@ jobject NativeToJavaPeerConnectionFactory(
     std::unique_ptr<SocketFactory> socket_factory,
     std::unique_ptr<Thread> network_thread,
     std::unique_ptr<Thread> worker_thread,
-    std::unique_ptr<Thread> signaling_thread);
+    std::unique_ptr<Thread> signaling_thread,
+    const Environment& env);
 
 }  // namespace webrtc
 
diff --git a/sdk/android/native_api/video/video_source.cc b/sdk/android/native_api/video/video_source.cc
index 3b1a3996c2..a4f1ee7be8 100644
--- a/sdk/android/native_api/video/video_source.cc
+++ b/sdk/android/native_api/video/video_source.cc
@@ -14,6 +14,7 @@
 
 #include <optional>
 
+#include "api/environment/environment.h"
 #include "api/make_ref_counted.h"
 #include "api/media_stream_interface.h"
 #include "api/scoped_refptr.h"
@@ -36,17 +37,19 @@ namespace {
 // VideoTrackSourceInterface.
 class JavaVideoTrackSourceImpl : public JavaVideoTrackSourceInterface {
  public:
-  JavaVideoTrackSourceImpl(JNIEnv* env,
+  JavaVideoTrackSourceImpl(JNIEnv* jni,
                            Thread* signaling_thread,
                            bool is_screencast,
-                           bool align_timestamps)
+                           bool align_timestamps,
+                           std::optional<Environment> env)
       : android_video_track_source_(
             make_ref_counted<jni::AndroidVideoTrackSource>(signaling_thread,
-                                                           env,
+                                                           jni,
                                                            is_screencast,
-                                                           align_timestamps)),
+                                                           align_timestamps,
+                                                           env)),
         native_capturer_observer_(jni::CreateJavaNativeCapturerObserver(
-            env,
+            jni,
             android_video_track_source_)) {}
 
   ScopedJavaLocalRef<jobject> GetJavaVideoCapturerObserver(
@@ -119,9 +122,10 @@ scoped_refptr<JavaVideoTrackSourceInterface> CreateJavaVideoSource(
     JNIEnv* jni,
     Thread* signaling_thread,
     bool is_screencast,
-    bool align_timestamps) {
+    bool align_timestamps,
+    std::optional<Environment> env) {
   return make_ref_counted<JavaVideoTrackSourceImpl>(
-      jni, signaling_thread, is_screencast, align_timestamps);
+      jni, signaling_thread, is_screencast, align_timestamps, env);
 }
 
 }  // namespace webrtc
diff --git a/sdk/android/native_api/video/video_source.h b/sdk/android/native_api/video/video_source.h
index 7f6779d458..21f5de1b47 100644
--- a/sdk/android/native_api/video/video_source.h
+++ b/sdk/android/native_api/video/video_source.h
@@ -13,6 +13,10 @@
 
 #include <jni.h>
 
+#include <optional>
+
+#include "absl/base/macros.h"
+#include "api/environment/environment.h"
 #include "api/media_stream_interface.h"
 #include "api/scoped_refptr.h"
 #include "rtc_base/thread.h"
@@ -32,10 +36,21 @@ class JavaVideoTrackSourceInterface : public VideoTrackSourceInterface {
 
 // Creates an instance of JavaVideoTrackSourceInterface,
 scoped_refptr<JavaVideoTrackSourceInterface> CreateJavaVideoSource(
-    JNIEnv* env,
+    JNIEnv* jni,
     Thread* signaling_thread,
     bool is_screencast,
-    bool align_timestamps);
+    bool align_timestamps,
+    std::optional<Environment> env);
+
+ABSL_DEPRECATE_AND_INLINE()
+inline scoped_refptr<JavaVideoTrackSourceInterface> CreateJavaVideoSource(
+    JNIEnv* jni,
+    Thread* signaling_thread,
+    bool is_screencast,
+    bool align_timestamps) {
+  return CreateJavaVideoSource(jni, signaling_thread, is_screencast,
+                               align_timestamps, std::nullopt);
+}
 
 }  // namespace webrtc
 
diff --git a/sdk/android/native_unittests/peerconnection/peer_connection_factory_unittest.cc b/sdk/android/native_unittests/peerconnection/peer_connection_factory_unittest.cc
index d789828628..51c4fc07e3 100644
--- a/sdk/android/native_unittests/peerconnection/peer_connection_factory_unittest.cc
+++ b/sdk/android/native_unittests/peerconnection/peer_connection_factory_unittest.cc
@@ -17,9 +17,8 @@
 
 #include "api/create_modular_peer_connection_factory.h"
 #include "api/enable_media_with_defaults.h"
-#include "api/environment/environment_factory.h"
+#include "api/environment/environment.h"
 #include "api/peer_connection_interface.h"
-#include "api/rtc_event_log/rtc_event_log_factory.h"
 #include "api/scoped_refptr.h"
 #include "media/engine/internal_decoder_factory.h"
 #include "media/engine/internal_encoder_factory.h"
@@ -31,6 +30,7 @@
 #include "sdk/android/native_api/audio_device_module/audio_device_android.h"
 #include "sdk/android/native_api/jni/application_context_provider.h"
 #include "sdk/android/native_api/jni/jvm.h"
+#include "test/create_test_environment.h"
 #include "test/gtest.h"
 
 namespace webrtc {
@@ -42,7 +42,8 @@ webrtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> CreateTestPCF(
     JNIEnv* jni,
     webrtc::Thread* network_thread,
     webrtc::Thread* worker_thread,
-    webrtc::Thread* signaling_thread) {
+    webrtc::Thread* signaling_thread,
+    const Environment& env) {
   // talk/ assumes pretty widely that the current Thread is ThreadManager'd, but
   // ThreadManager only WrapCurrentThread()s the thread where it is first
   // created.  Since the semantics around when auto-wrapping happens in
@@ -54,8 +55,7 @@ webrtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> CreateTestPCF(
   pcf_deps.network_thread = network_thread;
   pcf_deps.worker_thread = worker_thread;
   pcf_deps.signaling_thread = signaling_thread;
-  pcf_deps.event_log_factory = std::make_unique<RtcEventLogFactory>();
-  pcf_deps.env = CreateEnvironment();
+  pcf_deps.env = env;
 
   pcf_deps.adm =
       CreateJavaAudioDeviceModule(jni, *pcf_deps.env, GetAppContext(jni).obj());
@@ -96,13 +96,14 @@ TEST(PeerConnectionFactoryTest, NativeToJavaPeerConnectionFactory) {
   signaling_thread->SetName("signaling_thread", NULL);
   RTC_CHECK(signaling_thread->Start()) << "Failed to start thread";
 
+  const Environment env = CreateTestEnvironment();
   webrtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> factory =
       CreateTestPCF(jni, network_thread.get(), worker_thread.get(),
-                    signaling_thread.get());
+                    signaling_thread.get(), env);
 
   jobject java_factory = NativeToJavaPeerConnectionFactory(
       jni, factory, std::move(socket_server), std::move(network_thread),
-      std::move(worker_thread), std::move(signaling_thread));
+      std::move(worker_thread), std::move(signaling_thread), env);
 
   RTC_LOG(LS_INFO) << java_factory;
 
diff --git a/sdk/android/native_unittests/video/video_source_unittest.cc b/sdk/android/native_unittests/video/video_source_unittest.cc
index 0dce960054..9a5144bfc6 100644
--- a/sdk/android/native_unittests/video/video_source_unittest.cc
+++ b/sdk/android/native_unittests/video/video_source_unittest.cc
@@ -23,6 +23,7 @@
 #include "rtc_base/thread.h"
 #include "sdk/android/generated_native_unittests_jni/JavaVideoSourceTestHelper_jni.h"
 #include "sdk/android/native_api/jni/jvm.h"
+#include "test/create_test_environment.h"
 #include "test/gtest.h"
 
 namespace webrtc {
@@ -52,7 +53,8 @@ TEST(JavaVideoSourceTest, CreateJavaVideoSource) {
   webrtc::scoped_refptr<JavaVideoTrackSourceInterface> video_track_source =
       CreateJavaVideoSource(
           env, webrtc::ThreadManager::Instance()->CurrentThread(),
-          false /* is_screencast */, true /* align_timestamps */);
+          false /* is_screencast */, true /* align_timestamps */,
+          CreateTestEnvironment());
 
   ASSERT_NE(nullptr, video_track_source);
   EXPECT_NE(nullptr,
@@ -69,7 +71,8 @@ TEST(JavaVideoSourceTest, OnFrameCapturedFrameIsDeliveredToSink) {
   webrtc::scoped_refptr<JavaVideoTrackSourceInterface> video_track_source =
       CreateJavaVideoSource(
           env, webrtc::ThreadManager::Instance()->CurrentThread(),
-          false /* is_screencast */, true /* align_timestamps */);
+          false /* is_screencast */, true /* align_timestamps */,
+          CreateTestEnvironment());
   video_track_source->AddOrUpdateSink(&test_video_sink,
                                       webrtc::VideoSinkWants());
 
@@ -103,7 +106,8 @@ TEST(JavaVideoSourceTest,
   webrtc::scoped_refptr<JavaVideoTrackSourceInterface> video_track_source =
       CreateJavaVideoSource(
           env, webrtc::ThreadManager::Instance()->CurrentThread(),
-          false /* is_screencast */, false /* align_timestamps */);
+          false /* is_screencast */, false /* align_timestamps */,
+          CreateTestEnvironment());
   video_track_source->AddOrUpdateSink(&test_video_sink,
                                       webrtc::VideoSinkWants());
 
@@ -135,7 +139,8 @@ TEST(JavaVideoSourceTest, CapturerStartedSuccessStateBecomesLive) {
   webrtc::scoped_refptr<JavaVideoTrackSourceInterface> video_track_source =
       CreateJavaVideoSource(
           env, webrtc::ThreadManager::Instance()->CurrentThread(),
-          false /* is_screencast */, true /* align_timestamps */);
+          false /* is_screencast */, true /* align_timestamps */,
+          CreateTestEnvironment());
 
   jni::Java_JavaVideoSourceTestHelper_startCapture(
       env, video_track_source->GetJavaVideoCapturerObserver(env),
@@ -153,7 +158,8 @@ TEST(JavaVideoSourceTest, CapturerStartedFailureStateBecomesEnded) {
   webrtc::scoped_refptr<JavaVideoTrackSourceInterface> video_track_source =
       CreateJavaVideoSource(
           env, webrtc::ThreadManager::Instance()->CurrentThread(),
-          false /* is_screencast */, true /* align_timestamps */);
+          false /* is_screencast */, true /* align_timestamps */,
+          CreateTestEnvironment());
 
   jni::Java_JavaVideoSourceTestHelper_startCapture(
       env, video_track_source->GetJavaVideoCapturerObserver(env),
@@ -171,7 +177,8 @@ TEST(JavaVideoSourceTest, CapturerStoppedStateBecomesEnded) {
   webrtc::scoped_refptr<JavaVideoTrackSourceInterface> video_track_source =
       CreateJavaVideoSource(
           env, webrtc::ThreadManager::Instance()->CurrentThread(),
-          false /* is_screencast */, true /* align_timestamps */);
+          false /* is_screencast */, true /* align_timestamps */,
+          CreateTestEnvironment());
 
   jni::Java_JavaVideoSourceTestHelper_startCapture(
       env, video_track_source->GetJavaVideoCapturerObserver(env),
diff --git a/sdk/android/src/jni/android_video_track_source.cc b/sdk/android/src/jni/android_video_track_source.cc
index bfa493fe90..36c48e86ae 100644
--- a/sdk/android/src/jni/android_video_track_source.cc
+++ b/sdk/android/src/jni/android_video_track_source.cc
@@ -16,6 +16,7 @@
 #include <optional>
 #include <utility>
 
+#include "api/environment/environment.h"
 #include "api/media_stream_interface.h"
 #include "api/scoped_refptr.h"
 #include "api/task_queue/pending_task_safety_flag.h"
@@ -31,6 +32,7 @@
 #include "sdk/android/native_api/jni/java_types.h"
 #include "sdk/android/native_api/jni/scoped_java_ref.h"
 #include "sdk/android/src/jni/video_frame.h"
+#include "system_wrappers/include/clock.h"
 
 namespace webrtc {
 namespace jni {
@@ -58,7 +60,20 @@ AndroidVideoTrackSource::AndroidVideoTrackSource(Thread* signaling_thread,
                                                  JNIEnv* jni,
                                                  bool is_screencast,
                                                  bool align_timestamps)
+    : AndroidVideoTrackSource(signaling_thread,
+                              jni,
+                              is_screencast,
+                              align_timestamps,
+                              std::nullopt) {}
+
+AndroidVideoTrackSource::AndroidVideoTrackSource(Thread* signaling_thread,
+                                                 JNIEnv* jni,
+                                                 bool is_screencast,
+                                                 bool align_timestamps,
+                                                 std::optional<Environment> env)
     : AdaptedVideoTrackSource(kRequiredResolutionAlignment),
+      env_(env),
+      clock_(env_.has_value() ? env_->clock() : *Clock::GetRealTimeClock()),
       signaling_thread_(signaling_thread),
       is_screencast_(is_screencast),
       align_timestamps_(align_timestamps),
@@ -110,10 +125,10 @@ ScopedJavaLocalRef<jobject> AndroidVideoTrackSource::AdaptFrame(
 
   const int64_t camera_time_us = j_timestamp_ns / kNumNanosecsPerMicrosec;
   const int64_t aligned_timestamp_ns =
-      align_timestamps_
-          ? kNumNanosecsPerMicrosec * timestamp_aligner_.TranslateTimestamp(
-                                          camera_time_us, TimeMicros())
-          : j_timestamp_ns;
+      align_timestamps_ ? kNumNanosecsPerMicrosec *
+                              timestamp_aligner_.TranslateTimestamp(
+                                  camera_time_us, clock_.TimeInMicroseconds())
+                        : j_timestamp_ns;
 
   int adapted_width = 0;
   int adapted_height = 0;
diff --git a/sdk/android/src/jni/android_video_track_source.h b/sdk/android/src/jni/android_video_track_source.h
index 3ea1e71259..ce0a0c1df1 100644
--- a/sdk/android/src/jni/android_video_track_source.h
+++ b/sdk/android/src/jni/android_video_track_source.h
@@ -16,12 +16,14 @@
 #include <atomic>
 #include <optional>
 
+#include "api/environment/environment.h"
 #include "api/scoped_refptr.h"
 #include "api/task_queue/pending_task_safety_flag.h"
 #include "media/base/adapted_video_track_source.h"
 #include "rtc_base/thread.h"
 #include "rtc_base/timestamp_aligner.h"
 #include "sdk/android/native_api/jni/scoped_java_ref.h"
+#include "system_wrappers/include/clock.h"
 
 namespace webrtc {
 namespace jni {
@@ -33,10 +35,16 @@ namespace jni {
 // order to onFrameCaptured().
 class AndroidVideoTrackSource : public AdaptedVideoTrackSource {
  public:
+  [[deprecated("Provide a webrtc::Environment")]]
   AndroidVideoTrackSource(Thread* signaling_thread,
                           JNIEnv* jni,
                           bool is_screencast,
                           bool align_timestamps);
+  AndroidVideoTrackSource(Thread* signaling_thread,
+                          JNIEnv* jni,
+                          bool is_screencast,
+                          bool align_timestamps,
+                          std::optional<Environment> env);
   ~AndroidVideoTrackSource() override;
 
   bool is_screencast() const override;
@@ -87,6 +95,8 @@ class AndroidVideoTrackSource : public AdaptedVideoTrackSource {
   void SetIsScreencast(JNIEnv* env, jboolean j_is_screencast);
 
  private:
+  std::optional<Environment> env_;
+  Clock& clock_;
   Thread* signaling_thread_;
   std::atomic<SourceState> state_;
   std::atomic<bool> is_screencast_;
diff --git a/sdk/android/src/jni/pc/owned_factory_and_threads.cc b/sdk/android/src/jni/pc/owned_factory_and_threads.cc
index ccf360c9b2..69b81d01a6 100644
--- a/sdk/android/src/jni/pc/owned_factory_and_threads.cc
+++ b/sdk/android/src/jni/pc/owned_factory_and_threads.cc
@@ -13,6 +13,7 @@
 #include <memory>
 #include <utility>
 
+#include "api/environment/environment.h"
 #include "api/peer_connection_interface.h"
 #include "api/scoped_refptr.h"
 #include "rtc_base/socket_factory.h"
@@ -26,11 +27,26 @@ OwnedFactoryAndThreads::OwnedFactoryAndThreads(
     std::unique_ptr<Thread> network_thread,
     std::unique_ptr<Thread> worker_thread,
     std::unique_ptr<Thread> signaling_thread,
+    const Environment& env,
     const scoped_refptr<PeerConnectionFactoryInterface>& factory)
     : socket_factory_(std::move(socket_factory)),
       network_thread_(std::move(network_thread)),
       worker_thread_(std::move(worker_thread)),
       signaling_thread_(std::move(signaling_thread)),
+      env_(env),
+      factory_(factory) {}
+
+OwnedFactoryAndThreads::OwnedFactoryAndThreads(
+    std::unique_ptr<SocketFactory> socket_factory,
+    std::unique_ptr<Thread> network_thread,
+    std::unique_ptr<Thread> worker_thread,
+    std::unique_ptr<Thread> signaling_thread,
+    const scoped_refptr<PeerConnectionFactoryInterface>& factory)
+    : socket_factory_(std::move(socket_factory)),
+      network_thread_(std::move(network_thread)),
+      worker_thread_(std::move(worker_thread)),
+      signaling_thread_(std::move(signaling_thread)),
+      env_(std::nullopt),
       factory_(factory) {}
 
 }  // namespace jni
diff --git a/sdk/android/src/jni/pc/owned_factory_and_threads.h b/sdk/android/src/jni/pc/owned_factory_and_threads.h
index 4b22cb95ad..1235da9c07 100644
--- a/sdk/android/src/jni/pc/owned_factory_and_threads.h
+++ b/sdk/android/src/jni/pc/owned_factory_and_threads.h
@@ -11,9 +11,9 @@
 #ifndef SDK_ANDROID_SRC_JNI_PC_OWNED_FACTORY_AND_THREADS_H_
 #define SDK_ANDROID_SRC_JNI_PC_OWNED_FACTORY_AND_THREADS_H_
 
-
 #include <memory>
 
+#include "api/environment/environment.h"
 #include "api/peer_connection_interface.h"
 #include "api/scoped_refptr.h"
 #include "rtc_base/socket_factory.h"
@@ -30,6 +30,15 @@ namespace jni {
 // single thing for Java to hold and eventually free.
 class OwnedFactoryAndThreads {
  public:
+  OwnedFactoryAndThreads(
+      std::unique_ptr<SocketFactory> socket_factory,
+      std::unique_ptr<Thread> network_thread,
+      std::unique_ptr<Thread> worker_thread,
+      std::unique_ptr<Thread> signaling_thread,
+      const Environment& env,
+      const scoped_refptr<PeerConnectionFactoryInterface>& factory);
+
+  [[deprecated("Use webrtc::Environment constructor.")]]
   OwnedFactoryAndThreads(
       std::unique_ptr<SocketFactory> socket_factory,
       std::unique_ptr<Thread> network_thread,
@@ -44,6 +53,7 @@ class OwnedFactoryAndThreads {
   Thread* network_thread() { return network_thread_.get(); }
   Thread* signaling_thread() { return signaling_thread_.get(); }
   Thread* worker_thread() { return worker_thread_.get(); }
+  std::optional<Environment> env() const { return env_; }
 
  private:
   // Usually implemented by the SocketServer associated with the network thread,
@@ -52,6 +62,7 @@ class OwnedFactoryAndThreads {
   const std::unique_ptr<Thread> network_thread_;
   const std::unique_ptr<Thread> worker_thread_;
   const std::unique_ptr<Thread> signaling_thread_;
+  const std::optional<Environment> env_;
   const scoped_refptr<PeerConnectionFactoryInterface> factory_;
 };
 
diff --git a/sdk/android/src/jni/pc/peer_connection_factory.cc b/sdk/android/src/jni/pc/peer_connection_factory.cc
index bd26b84516..4fae2fb6d0 100644
--- a/sdk/android/src/jni/pc/peer_connection_factory.cc
+++ b/sdk/android/src/jni/pc/peer_connection_factory.cc
@@ -147,10 +147,11 @@ ScopedJavaLocalRef<jobject> NativeToScopedJavaPeerConnectionFactory(
     std::unique_ptr<SocketFactory> socket_factory,
     std::unique_ptr<Thread> network_thread,
     std::unique_ptr<Thread> worker_thread,
-    std::unique_ptr<Thread> signaling_thread) {
+    std::unique_ptr<Thread> signaling_thread,
+    const Environment& webrtc_env) {
   OwnedFactoryAndThreads* owned_factory = new OwnedFactoryAndThreads(
       std::move(socket_factory), std::move(network_thread),
-      std::move(worker_thread), std::move(signaling_thread), pcf);
+      std::move(worker_thread), std::move(signaling_thread), webrtc_env, pcf);
 
   jni_zero::ScopedJavaLocalRef<jobject> j_pcf =
       Java_PeerConnectionFactory_Constructor(
@@ -187,10 +188,11 @@ jobject NativeToJavaPeerConnectionFactory(
     std::unique_ptr<SocketFactory> socket_factory,
     std::unique_ptr<Thread> network_thread,
     std::unique_ptr<Thread> worker_thread,
-    std::unique_ptr<Thread> signaling_thread) {
+    std::unique_ptr<Thread> signaling_thread,
+    const Environment& env) {
   return NativeToScopedJavaPeerConnectionFactory(
              jni, pcf, std::move(socket_factory), std::move(network_thread),
-             std::move(worker_thread), std::move(signaling_thread))
+             std::move(worker_thread), std::move(signaling_thread), env)
       .Release();
 }
 
@@ -338,7 +340,7 @@ ScopedJavaLocalRef<jobject> CreatePeerConnectionFactoryForJava(
 
   return NativeToScopedJavaPeerConnectionFactory(
       jni, factory, std::move(socket_server), std::move(network_thread),
-      std::move(worker_thread), std::move(signaling_thread));
+      std::move(worker_thread), std::move(signaling_thread), env);
 }
 
 static jni_zero::ScopedJavaLocalRef<jobject>
@@ -524,9 +526,9 @@ static jlong JNI_PeerConnectionFactory_CreateVideoSource(
     jboolean align_timestamps) {
   OwnedFactoryAndThreads* factory =
       reinterpret_cast<OwnedFactoryAndThreads*>(native_factory);
-  return jlongFromPointer(CreateVideoSource(jni, factory->signaling_thread(),
-                                            factory->worker_thread(),
-                                            is_screencast, align_timestamps));
+  return jlongFromPointer(CreateVideoSource(
+      jni, factory->signaling_thread(), factory->worker_thread(), is_screencast,
+      align_timestamps, factory->env()));
 }
 
 static jlong JNI_PeerConnectionFactory_CreateVideoTrack(
diff --git a/sdk/android/src/jni/pc/peer_connection_factory.h b/sdk/android/src/jni/pc/peer_connection_factory.h
index f09c904e4e..2ed3e306e2 100644
--- a/sdk/android/src/jni/pc/peer_connection_factory.h
+++ b/sdk/android/src/jni/pc/peer_connection_factory.h
@@ -15,6 +15,7 @@
 
 #include <memory>
 
+#include "api/environment/environment.h"
 #include "api/peer_connection_interface.h"
 #include "api/scoped_refptr.h"
 #include "rtc_base/socket_factory.h"
@@ -30,7 +31,8 @@ jobject NativeToJavaPeerConnectionFactory(
     std::unique_ptr<SocketFactory> socket_factory,
     std::unique_ptr<Thread> network_thread,
     std::unique_ptr<Thread> worker_thread,
-    std::unique_ptr<Thread> signaling_thread);
+    std::unique_ptr<Thread> signaling_thread,
+    const Environment& env);
 
 }  // namespace jni
 }  // namespace webrtc
diff --git a/sdk/android/src/jni/pc/video.cc b/sdk/android/src/jni/pc/video.cc
index 38fb1bedc9..5a6d2e0202 100644
--- a/sdk/android/src/jni/pc/video.cc
+++ b/sdk/android/src/jni/pc/video.cc
@@ -12,6 +12,7 @@
 
 #include <jni.h>
 
+#include "api/environment/environment.h"
 #include "api/make_ref_counted.h"
 #include "api/video_codecs/video_decoder_factory.h"
 #include "api/video_codecs/video_encoder_factory.h"
@@ -45,9 +46,10 @@ void* CreateVideoSource(JNIEnv* env,
                         Thread* signaling_thread,
                         Thread* worker_thread,
                         jboolean is_screencast,
-                        jboolean align_timestamps) {
+                        jboolean align_timestamps,
+                        std::optional<Environment> env_webrtc) {
   auto source = make_ref_counted<AndroidVideoTrackSource>(
-      signaling_thread, env, is_screencast, align_timestamps);
+      signaling_thread, env, is_screencast, align_timestamps, env_webrtc);
   return source.release();
 }
 
diff --git a/sdk/android/src/jni/pc/video.h b/sdk/android/src/jni/pc/video.h
index 3386cb9288..e3744d74c9 100644
--- a/sdk/android/src/jni/pc/video.h
+++ b/sdk/android/src/jni/pc/video.h
@@ -13,6 +13,9 @@
 
 #include <jni.h>
 
+#include <optional>
+
+#include "api/environment/environment.h"
 #include "rtc_base/thread.h"
 #include "sdk/android/native_api/jni/scoped_java_ref.h"
 
@@ -36,7 +39,8 @@ void* CreateVideoSource(JNIEnv* env,
                         Thread* signaling_thread,
                         Thread* worker_thread,
                         jboolean is_screencast,
-                        jboolean align_timestamps);
+                        jboolean align_timestamps,
+                        std::optional<Environment> env_webrtc = std::nullopt);
 
 }  // namespace jni
 }  // namespace webrtc

From c445a96208c51d143e828990c5743fa5db1bd2a9 Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Mon, 16 Feb 2026 17:58:27 +0100
Subject: [PATCH 069/472] Deprecate conversion from nullptr to an empty
 ArrayView

Use default constructor instead.

Bug: webrtc:439801349
Change-Id: I68d459865f4e7d7eb598f47b37ab21bc7c1eb133
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449740
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46922}
---
 api/array_view.h                              |  1 +
 api/audio_codecs/audio_encoder.cc             |  2 +-
 audio/channel_receive.cc                      |  4 ++--
 audio/channel_send.cc                         |  2 +-
 .../aec3/decimator_unittest.cc                |  2 +-
 .../aec3/fft_data_unittest.cc                 |  2 +-
 ...lute_capture_time_interpolator_unittest.cc |  2 +-
 modules/rtp_rtcp/source/rtp_packet.cc         | 20 +++++++++----------
 .../video_coding/packet_buffer_unittest.cc    |  4 ++--
 rtc_base/network_unittest.cc                  |  2 +-
 10 files changed, 21 insertions(+), 20 deletions(-)

diff --git a/api/array_view.h b/api/array_view.h
index 6c2e4938c7..c12bc5d15b 100644
--- a/api/array_view.h
+++ b/api/array_view.h
@@ -171,6 +171,7 @@ class ArrayView final : public array_view_internal::ArrayViewBase<T, Size> {
   // Construct an empty ArrayView. Note that fixed-size ArrayViews of size > 0
   // cannot be empty.
   ArrayView() : ArrayView(nullptr, 0) {}
+  ABSL_DEPRECATE_AND_INLINE()
   ArrayView(std::nullptr_t)  // NOLINT
       : ArrayView() {}
   ArrayView(std::nullptr_t, size_t size)
diff --git a/api/audio_codecs/audio_encoder.cc b/api/audio_codecs/audio_encoder.cc
index ece18fe046..cb86adcb35 100644
--- a/api/audio_codecs/audio_encoder.cc
+++ b/api/audio_codecs/audio_encoder.cc
@@ -79,7 +79,7 @@ void AudioEncoder::SetTargetBitrate(int /* target_bps */) {}
 
 ArrayView<std::unique_ptr<AudioEncoder>>
 AudioEncoder::ReclaimContainedEncoders() {
-  return nullptr;
+  return {};
 }
 
 bool AudioEncoder::EnableAudioNetworkAdaptor(absl::string_view /*config*/) {
diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index 1709ccf12b..007d3cfabb 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -736,8 +736,8 @@ void ChannelReceive::ReceivePacket(const uint8_t* packet,
     const FrameDecryptorInterface::Result decrypt_result =
         frame_decryptor_->Decrypt(
             MediaType::AUDIO, csrcs,
-            /*additional_data=*/
-            nullptr, ArrayView<const uint8_t>(payload, payload_data_length),
+            /*additional_data=*/{},
+            ArrayView<const uint8_t>(payload, payload_data_length),
             decrypted_audio_payload);
 
     if (decrypt_result.IsOk()) {
diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index a280c7e476..68fd2f62d9 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -441,7 +441,7 @@ int32_t ChannelSend::SendRtpAudio(AudioFrameType frameType,
       size_t bytes_written = 0;
       int encrypt_status =
           frame_encryptor_->Encrypt(MediaType::AUDIO, rtp_rtcp_->SSRC(),
-                                    /*additional_data=*/nullptr, payload,
+                                    /*additional_data=*/{}, payload,
                                     encrypted_audio_payload, &bytes_written);
       if (encrypt_status != 0) {
         RTC_DLOG(LS_ERROR)
diff --git a/modules/audio_processing/aec3/decimator_unittest.cc b/modules/audio_processing/aec3/decimator_unittest.cc
index 2adb46f64c..8424c36cc7 100644
--- a/modules/audio_processing/aec3/decimator_unittest.cc
+++ b/modules/audio_processing/aec3/decimator_unittest.cc
@@ -114,7 +114,7 @@ TEST(DecimatorDeathTest, WrongInputSize) {
 TEST(DecimatorDeathTest, NullOutput) {
   Decimator decimator(4);
   std::vector<float> x(kBlockSize, 0.f);
-  EXPECT_DEATH(decimator.Decimate(x, nullptr), "");
+  EXPECT_DEATH(decimator.Decimate(x, {}), "");
 }
 
 // Verifies the check for the output size.
diff --git a/modules/audio_processing/aec3/fft_data_unittest.cc b/modules/audio_processing/aec3/fft_data_unittest.cc
index a089c7b7bc..acf42a5e3d 100644
--- a/modules/audio_processing/aec3/fft_data_unittest.cc
+++ b/modules/audio_processing/aec3/fft_data_unittest.cc
@@ -78,7 +78,7 @@ TEST(FftDataDeathTest, NonNullCopyToPackedArrayOutput) {
 
 // Verifies the check for null output in Spectrum.
 TEST(FftDataDeathTest, NonNullSpectrumOutput) {
-  EXPECT_DEATH(FftData().Spectrum(Aec3Optimization::kNone, nullptr), "");
+  EXPECT_DEATH(FftData().Spectrum(Aec3Optimization::kNone, {}), "");
 }
 
 #endif
diff --git a/modules/rtp_rtcp/source/absolute_capture_time_interpolator_unittest.cc b/modules/rtp_rtcp/source/absolute_capture_time_interpolator_unittest.cc
index a99d06a7a1..a8531c6e52 100644
--- a/modules/rtp_rtcp/source/absolute_capture_time_interpolator_unittest.cc
+++ b/modules/rtp_rtcp/source/absolute_capture_time_interpolator_unittest.cc
@@ -31,7 +31,7 @@ using testing::Le;
 TEST(AbsoluteCaptureTimeInterpolatorTest, GetSourceWithoutCsrcs) {
   constexpr uint32_t kSsrc = 12;
 
-  EXPECT_EQ(AbsoluteCaptureTimeInterpolator::GetSource(kSsrc, nullptr), kSsrc);
+  EXPECT_EQ(AbsoluteCaptureTimeInterpolator::GetSource(kSsrc, {}), kSsrc);
 }
 
 TEST(AbsoluteCaptureTimeInterpolatorTest, GetSourceWithCsrcs) {
diff --git a/modules/rtp_rtcp/source/rtp_packet.cc b/modules/rtp_rtcp/source/rtp_packet.cc
index 3515521fcd..963519dd0c 100644
--- a/modules/rtp_rtcp/source/rtp_packet.cc
+++ b/modules/rtp_rtcp/source/rtp_packet.cc
@@ -247,17 +247,17 @@ ArrayView<uint8_t> RtpPacket::AllocateRawExtension(int id, size_t length) {
                       << ": expected "
                       << static_cast<int>(extension_entry->length)
                       << ". received " << length;
-    return nullptr;
+    return {};
   }
   if (payload_size_ > 0) {
     RTC_LOG(LS_ERROR) << "Can't add new extension id " << id
                       << " after payload was set.";
-    return nullptr;
+    return {};
   }
   if (padding_size_ > 0) {
     RTC_LOG(LS_ERROR) << "Can't add new extension id " << id
                       << " after padding was set.";
-    return nullptr;
+    return {};
   }
 
   const size_t num_csrc = data()[0] & 0x0F;
@@ -286,7 +286,7 @@ ArrayView<uint8_t> RtpPacket::AllocateRawExtension(int id, size_t length) {
             << "Extension cannot be registered: Not enough space left in "
                "buffer to change to two-byte header extension and add new "
                "extension.";
-        return nullptr;
+        return {};
       }
       // Promote already written data to two-byte header format.
       PromoteToTwoByteHeaderExtension();
@@ -307,7 +307,7 @@ ArrayView<uint8_t> RtpPacket::AllocateRawExtension(int id, size_t length) {
   if (extensions_offset + new_extensions_size > capacity()) {
     RTC_LOG(LS_ERROR)
         << "Extension cannot be registered: Not enough space left in buffer.";
-    return nullptr;
+    return {};
   }
 
   // All checks passed, write down the extension headers.
@@ -598,11 +598,11 @@ ArrayView<const uint8_t> RtpPacket::FindExtension(ExtensionType type) const {
   uint8_t id = extensions_.GetId(type);
   if (id == ExtensionManager::kInvalidId) {
     // Extension not registered.
-    return nullptr;
+    return {};
   }
   ExtensionInfo const* extension_info = FindExtensionInfo(id);
   if (extension_info == nullptr) {
-    return nullptr;
+    return {};
   }
   return MakeArrayView(data() + extension_info->offset, extension_info->length);
 }
@@ -613,17 +613,17 @@ ArrayView<uint8_t> RtpPacket::AllocateExtension(ExtensionType type,
   if (length == 0 || length > RtpExtension::kMaxValueSize ||
       (!extensions_.ExtmapAllowMixed() &&
        length > RtpExtension::kOneByteHeaderExtensionMaxValueSize)) {
-    return nullptr;
+    return {};
   }
 
   uint8_t id = extensions_.GetId(type);
   if (id == ExtensionManager::kInvalidId) {
     // Extension not registered.
-    return nullptr;
+    return {};
   }
   if (!extensions_.ExtmapAllowMixed() &&
       id > RtpExtension::kOneByteHeaderExtensionMaxId) {
-    return nullptr;
+    return {};
   }
   return AllocateRawExtension(id, length);
 }
diff --git a/modules/video_coding/packet_buffer_unittest.cc b/modules/video_coding/packet_buffer_unittest.cc
index 961a259409..af994c1aed 100644
--- a/modules/video_coding/packet_buffer_unittest.cc
+++ b/modules/video_coding/packet_buffer_unittest.cc
@@ -862,9 +862,9 @@ TEST_F(PacketBufferH264FrameGap,
 
 TEST_F(PacketBufferH264FrameGap, DoesntCrashWhenTryToClearBefore1stPacket) {
   // Test scenario copied from the https://issues.chromium.org/370689424
-  InsertH264(41087, kKeyFrame, kNotFirst, kNotLast, 123, nullptr, 0, false);
+  InsertH264(41087, kKeyFrame, kNotFirst, kNotLast, 123, {}, 0, false);
   packet_buffer_.ClearTo(30896);
-  InsertH264(32896, kKeyFrame, kFirst, kLast, 123, nullptr, 0, false);
+  InsertH264(32896, kKeyFrame, kFirst, kLast, 123, {}, 0, false);
 }
 
 }  // namespace
diff --git a/rtc_base/network_unittest.cc b/rtc_base/network_unittest.cc
index e0015c121c..f9438b06c6 100644
--- a/rtc_base/network_unittest.cc
+++ b/rtc_base/network_unittest.cc
@@ -1748,7 +1748,7 @@ TEST_F(NetworkTest, HardcodedVpn) {
       NetworkManagerBase::IsVpnMacAddress(ArrayView<const uint8_t>(cisco, 5)));
   EXPECT_FALSE(NetworkManagerBase::IsVpnMacAddress(five_bytes));
   EXPECT_FALSE(NetworkManagerBase::IsVpnMacAddress(unknown));
-  EXPECT_FALSE(NetworkManagerBase::IsVpnMacAddress(nullptr));
+  EXPECT_FALSE(NetworkManagerBase::IsVpnMacAddress({}));
 }
 
 TEST(CompareNetworks, IrreflexivityTest) {

From 2d76be9907269fa1ba75e777c6e4876e570df1bf Mon Sep 17 00:00:00 2001
From: Per K <perkj@webrtc.org>
Date: Tue, 17 Feb 2026 09:37:35 +0000
Subject: [PATCH 070/472] Preserve packet order in
 ParsedRtcEventLog::GetPacketInfos if CCFB is used
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Use log feedback time to preserve packet order if log time is equal to ensure packets sent in simulation at least are ordered correctly between feedback.

Bug: webrtc:483936087
Change-Id: I61195f12774fb4e7b3cd467e4c7e794e86ab1e05
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449841
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46923}
---
 logging/rtc_event_log/rtc_event_log_parser.cc | 19 +++++++++++++++++++
 1 file changed, 19 insertions(+)

diff --git a/logging/rtc_event_log/rtc_event_log_parser.cc b/logging/rtc_event_log/rtc_event_log_parser.cc
index 79720d36bf..2c4439b965 100644
--- a/logging/rtc_event_log/rtc_event_log_parser.cc
+++ b/logging/rtc_event_log/rtc_event_log_parser.cc
@@ -2658,6 +2658,25 @@ std::vector<LoggedPacketInfo> ParsedRtcEventLog::GetPacketInfos(
                       PacketDirection::kOutgoingPacket);
   }
   process.ProcessEventsInOrder();
+
+  if (!ccfb_indices.empty()) {
+    // The log stores RTP packets by stream (with millisecond timestamps), but
+    // doesn't guarantee the order between packets sent or received at the same
+    // time on different SSRCs. We don't have transport sequence numbers when
+    // CCFB is used, so the order can't be reconstructed using those.
+    // process.ProcessEventsInOrder() will set log feedback time per packet
+    // based on when feedback was originally received/sent for a packet, and we
+    // can use that to ensure packets are at least ordered as originally seen in
+    // feedback.
+    std::stable_sort(packets.begin(), packets.end(),
+                     [](const LoggedPacketInfo& a, const LoggedPacketInfo& b) {
+                       if (a.log_packet_time == b.log_packet_time) {
+                         return a.log_feedback_time < b.log_feedback_time;
+                       }
+                       return a.log_packet_time < b.log_packet_time;
+                     });
+  }
+
   return packets;
 }
 

From 2d71ccad18e407459ce231e565e196932617abf2 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Tue, 17 Feb 2026 11:12:58 +0100
Subject: [PATCH 071/472] Refactor RTP receiver tests to use dedicated worker
 threads

Bug: none
Change-Id: If5b66fdc74a345efc10ab1ea8621c3d508ab808a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449320
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46924}
---
 pc/audio_rtp_receiver_unittest.cc | 51 +++++++++++++++++++------------
 pc/video_rtp_receiver_unittest.cc | 15 +++++----
 2 files changed, 39 insertions(+), 27 deletions(-)

diff --git a/pc/audio_rtp_receiver_unittest.cc b/pc/audio_rtp_receiver_unittest.cc
index a1c5e37ce7..6a69cd66de 100644
--- a/pc/audio_rtp_receiver_unittest.cc
+++ b/pc/audio_rtp_receiver_unittest.cc
@@ -12,6 +12,7 @@
 
 #include <atomic>
 #include <cstdint>
+#include <memory>
 #include <string>
 #include <utility>
 #include <vector>
@@ -20,6 +21,7 @@
 #include "api/scoped_refptr.h"
 #include "api/test/rtc_error_matchers.h"
 #include "api/units/time_delta.h"
+#include "media/base/media_channel.h"
 #include "pc/test/mock_voice_media_receive_channel_interface.h"
 #include "rtc_base/thread.h"
 #include "test/gmock.h"
@@ -41,22 +43,28 @@ namespace webrtc {
 class AudioRtpReceiverTest : public ::testing::Test {
  protected:
   AudioRtpReceiverTest()
-      : worker_(Thread::Current()),
+      : worker_thread_(Thread::Create()),
         receiver_(
-            make_ref_counted<AudioRtpReceiver>(worker_,
+            make_ref_counted<AudioRtpReceiver>(worker_thread_.get(),
                                                std::string(),
                                                std::vector<std::string>())) {
+    worker_thread_->Start();
     EXPECT_CALL(receive_channel_, SetRawAudioSink(kSsrc, _));
     EXPECT_CALL(receive_channel_, SetBaseMinimumPlayoutDelayMs(kSsrc, _));
   }
 
   ~AudioRtpReceiverTest() override {
     EXPECT_CALL(receive_channel_, SetOutputVolume(kSsrc, kVolumeMuted));
-    receiver_->SetMediaChannel(nullptr);
+    SetMediaChannel(nullptr);
   }
 
-  AutoThread main_thread_;
-  Thread* worker_;
+  void SetMediaChannel(MediaReceiveChannelInterface* media_channel) {
+    worker_thread_->BlockingCall(
+        [&]() { receiver_->SetMediaChannel(media_channel); });
+  }
+
+  test::RunLoop loop_;
+  std::unique_ptr<Thread> worker_thread_;
   scoped_refptr<AudioRtpReceiver> receiver_;
   MockVoiceMediaReceiveChannelInterface receive_channel_;
 };
@@ -72,10 +80,10 @@ TEST_F(AudioRtpReceiverTest, SetOutputVolumeIsCalled) {
 
   receiver_->track();
   receiver_->track()->set_enabled(true);
-  receiver_->SetMediaChannel(&receive_channel_);
+  SetMediaChannel(&receive_channel_);
   EXPECT_CALL(receive_channel_, SetDefaultRawAudioSink(_)).Times(0);
   auto setup_task = receiver_->GetSetupForMediaChannel(kSsrc);
-  std::move(setup_task)();
+  worker_thread_->BlockingCall([&]() { std::move(setup_task)(); });
 
   EXPECT_CALL(receive_channel_, SetOutputVolume(kSsrc, kVolume))
       .WillOnce(InvokeWithoutArgs([&] {
@@ -95,14 +103,14 @@ TEST_F(AudioRtpReceiverTest, VolumesSetBeforeStartingAreRespected) {
   receiver_->OnSetVolume(kVolume);
 
   receiver_->track()->set_enabled(true);
-  receiver_->SetMediaChannel(&receive_channel_);
+  SetMediaChannel(&receive_channel_);
 
   // The previosly set initial volume should be propagated to the provided
   // media_channel_ as soon as GetSetupForMediaChannel is called.
   EXPECT_CALL(receive_channel_, SetOutputVolume(kSsrc, kVolume));
 
   auto setup_task = receiver_->GetSetupForMediaChannel(kSsrc);
-  std::move(setup_task)();
+  worker_thread_->BlockingCall([&]() { std::move(setup_task)(); });
 }
 
 // Tests that OnChanged notifications are processed correctly on the worker
@@ -110,28 +118,33 @@ TEST_F(AudioRtpReceiverTest, VolumesSetBeforeStartingAreRespected) {
 // constructor.
 TEST(AudioRtpReceiver, OnChangedNotificationsAfterConstruction) {
   test::RunLoop loop;
-  auto* thread = Thread::Current();  // Points to loop's thread.
+
+  std::unique_ptr<Thread> worker_thread = Thread::Create();
+  worker_thread->Start();
   MockVoiceMediaReceiveChannelInterface receive_channel;
   auto receiver = make_ref_counted<AudioRtpReceiver>(
-      thread, std::string(), std::vector<std::string>(), &receive_channel);
+      worker_thread.get(), std::string(), std::vector<std::string>(),
+      &receive_channel);
 
   EXPECT_CALL(receive_channel, SetDefaultRawAudioSink(_)).Times(1);
   EXPECT_CALL(receive_channel, SetDefaultOutputVolume(kDefaultVolume)).Times(1);
   auto setup_task = receiver->GetSetupForUnsignaledMediaChannel();
-  std::move(setup_task)();
+  worker_thread->BlockingCall([&]() { std::move(setup_task)(); });
+
+  // When the track is marked as disabled, an async notification is queued
+  // for the worker thread. This notification should trigger the volume
+  // of the media channel to be set to kVolumeMuted.
+  // Set the expectation first for the call, before changing the track state.
+  EXPECT_CALL(receive_channel, SetDefaultOutputVolume(kVolumeMuted)).Times(1);
 
   // Mark the track as disabled.
   receiver->track()->set_enabled(false);
 
-  // When the track was marked as disabled, an async notification was queued
-  // for the worker thread. This notification should trigger the volume
-  // of the media channel to be set to kVolumeMuted.
-  // Flush the worker thread, but set the expectation first for the call.
-  EXPECT_CALL(receive_channel, SetDefaultOutputVolume(kVolumeMuted)).Times(1);
-  loop.Flush();
+  // Flush the worker thread.
+  worker_thread->BlockingCall([] {});
 
   EXPECT_CALL(receive_channel, SetDefaultOutputVolume(kVolumeMuted)).Times(1);
-  receiver->SetMediaChannel(nullptr);
+  worker_thread->BlockingCall([&]() { receiver->SetMediaChannel(nullptr); });
 }
 
 }  // namespace webrtc
diff --git a/pc/video_rtp_receiver_unittest.cc b/pc/video_rtp_receiver_unittest.cc
index 8756c10b76..ed73fac800 100644
--- a/pc/video_rtp_receiver_unittest.cc
+++ b/pc/video_rtp_receiver_unittest.cc
@@ -26,10 +26,10 @@
 #include "api/video/video_sink_interface.h"
 #include "media/base/fake_media_engine.h"
 #include "media/base/media_channel.h"
-#include "rtc_base/task_queue_for_test.h"
 #include "rtc_base/thread.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 
 using ::testing::_;
 using ::testing::AnyNumber;
@@ -97,15 +97,15 @@ class VideoRtpReceiverTest : public testing::Test {
   }
 
   void SetMediaChannel(MediaReceiveChannelInterface* media_channel) {
-    SendTask(worker_thread_.get(),
-             [&]() { receiver_->SetMediaChannel(media_channel); });
+    worker_thread_->BlockingCall(
+        [&]() { receiver_->SetMediaChannel(media_channel); });
   }
 
   VideoTrackSourceInterface* Source() {
     return receiver_->streams()[0]->FindVideoTrack("receiver")->GetSource();
   }
 
-  AutoThread main_thread_;
+  test::RunLoop loop_;
   std::unique_ptr<Thread> worker_thread_;
   NiceMock<MockVideoMediaReceiveChannel> channel_;
   scoped_refptr<VideoRtpReceiver> receiver_;
@@ -198,7 +198,7 @@ TEST_F(VideoRtpReceiverTest, BroadcastsEncodedFramesWhenEnabled) {
   EXPECT_CALL(sink, OnFrame).Times(2);
   MockRecordableEncodedFrame frame;
   broadcast(frame);
-  SendTask(worker_thread_.get(), [&] { broadcast(frame); });
+  worker_thread_->BlockingCall([&] { broadcast(frame); });
 }
 
 TEST_F(VideoRtpReceiverTest, EnablesEncodedOutputOnChannelRestart) {
@@ -207,12 +207,11 @@ TEST_F(VideoRtpReceiverTest, EnablesEncodedOutputOnChannelRestart) {
   Source()->AddEncodedSink(&sink);
   EXPECT_CALL(channel_, SetRecordableEncodedFrameCallback(4711, _));
   auto setup_media_channel = receiver_->GetSetupForMediaChannel(4711);
-  worker_thread_->BlockingCall(
-      [task = std::move(setup_media_channel)]() mutable { std::move(task)(); });
+  worker_thread_->BlockingCall([&]() { std::move(setup_media_channel)(); });
   EXPECT_CALL(channel_, ClearRecordableEncodedFrameCallback(4711));
   EXPECT_CALL(channel_, SetRecordableEncodedFrameCallback(0, _));
   auto setup_task = receiver_->GetSetupForUnsignaledMediaChannel();
-  worker_thread_->BlockingCall([&]() mutable { std::move(setup_task)(); });
+  worker_thread_->BlockingCall([&]() { std::move(setup_task)(); });
 }
 
 }  // namespace

From ecaf12aa9a0406bfffbd62db5d11dc8dc2289225 Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Tue, 17 Feb 2026 11:42:13 +0100
Subject: [PATCH 072/472] Deprecate ArrayView::operator==

To align with std::span that doesn't have such operator to avoid
confusion if comparison should be deep or shallow.

operator!= is deleted because it is auto-generated from operator== since
c++20

Bug: webrtc:439801349
Change-Id: I1ac35029d819c3abc52531b33c6ccbc43dc78319
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449686
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46925}
---
 api/array_view.h                              |  5 +---
 api/array_view_unittest.cc                    | 26 ++-----------------
 api/audio/test/audio_view_unittest.cc         | 11 +++++---
 modules/audio_processing/aec3/echo_remover.cc |  7 +++--
 .../agc2/rnn_vad/ring_buffer_unittest.cc      |  2 +-
 .../audio_frame_view_unittest.cc              |  6 +++--
 ...deo_frame_transformer_delegate_unittest.cc | 16 ++++++------
 7 files changed, 26 insertions(+), 47 deletions(-)

diff --git a/api/array_view.h b/api/array_view.h
index c12bc5d15b..8d367e4fa3 100644
--- a/api/array_view.h
+++ b/api/array_view.h
@@ -317,13 +317,10 @@ class ArrayView final : public array_view_internal::ArrayViewBase<T, Size> {
 // Comparing two ArrayViews compares their (pointer,size) pairs; it does *not*
 // dereference the pointers.
 template <typename T, std::ptrdiff_t Size1, std::ptrdiff_t Size2>
+[[deprecated("issues.webrtc.org/439801349")]]
 bool operator==(const ArrayView<T, Size1>& a, const ArrayView<T, Size2>& b) {
   return a.data() == b.data() && a.size() == b.size();
 }
-template <typename T, std::ptrdiff_t Size1, std::ptrdiff_t Size2>
-bool operator!=(const ArrayView<T, Size1>& a, const ArrayView<T, Size2>& b) {
-  return !(a == b);
-}
 
 // Variable-size ArrayViews are the size of two pointers; fixed-size ArrayViews
 // are the size of one pointer. (And as a special case, fixed-size ArrayViews
diff --git a/api/array_view_unittest.cc b/api/array_view_unittest.cc
index 1ab7b6e981..c4b1385909 100644
--- a/api/array_view_unittest.cc
+++ b/api/array_view_unittest.cc
@@ -555,34 +555,13 @@ TEST(ArrayViewTest, TestEmpty) {
   static_assert(!ArrayView<int, 3>::empty(), "");
 }
 
-TEST(ArrayViewTest, TestCompare) {
-  int a[] = {1, 2, 3};
-  int b[] = {1, 2, 3};
-
-  EXPECT_EQ(ArrayView<int>(a), ArrayView<int>(a));
-  EXPECT_EQ((ArrayView<int, 3>(a)), (ArrayView<int, 3>(a)));
-  EXPECT_EQ(ArrayView<int>(a), (ArrayView<int, 3>(a)));
-  EXPECT_EQ(ArrayView<int>(), ArrayView<int>());
-  EXPECT_EQ(ArrayView<int>(), ArrayView<int>(a, 0));
-  EXPECT_EQ(ArrayView<int>(a, 0), ArrayView<int>(b, 0));
-  EXPECT_EQ((ArrayView<int, 0>(a, 0)), ArrayView<int>());
-
-  EXPECT_NE(ArrayView<int>(a), ArrayView<int>(b));
-  EXPECT_NE((ArrayView<int, 3>(a)), (ArrayView<int, 3>(b)));
-  EXPECT_NE((ArrayView<int, 3>(a)), ArrayView<int>(b));
-  EXPECT_NE(ArrayView<int>(a), ArrayView<int>());
-  EXPECT_NE(ArrayView<int>(a), ArrayView<int>(a, 2));
-  EXPECT_NE((ArrayView<int, 3>(a)), (ArrayView<int, 2>(a, 2)));
-}
-
 #pragma clang diagnostic push
 #pragma clang diagnostic ignored "-Wdeprecated-declarations"
 TEST(ArrayViewTest, TestSubViewVariable) {
   int a[] = {1, 2, 3};
   ArrayView<int> av(a);
 
-  EXPECT_EQ(av.subview(0), av);
-
+  EXPECT_THAT(av.subview(0), ElementsAre(1, 2, 3));
   EXPECT_THAT(av.subview(1), ElementsAre(2, 3));
   EXPECT_THAT(av.subview(2), ElementsAre(3));
   EXPECT_THAT(av.subview(3), IsEmpty());
@@ -622,8 +601,7 @@ TEST(ArrayViewTest, TestSubViewFixed) {
   int a[] = {1, 2, 3};
   ArrayView<int, 3> av(a);
 
-  EXPECT_EQ(av.subview(0), av);
-
+  EXPECT_THAT(av.subview(0), ElementsAre(1, 2, 3));
   EXPECT_THAT(av.subview(1), ElementsAre(2, 3));
   EXPECT_THAT(av.subview(2), ElementsAre(3));
   EXPECT_THAT(av.subview(3), IsEmpty());
diff --git a/api/audio/test/audio_view_unittest.cc b/api/audio/test/audio_view_unittest.cc
index e28c73ca11..57567b1965 100644
--- a/api/audio/test/audio_view_unittest.cc
+++ b/api/audio/test/audio_view_unittest.cc
@@ -75,9 +75,10 @@ TEST(AudioViewTest, InterleavedView) {
   EXPECT_EQ(NumChannels(interleaved), 1u);
   EXPECT_TRUE(IsMono(interleaved));
   EXPECT_EQ(SamplesPerChannel(interleaved), kArraySize);
-  EXPECT_EQ(interleaved.AsMono().size(), kArraySize);
-  EXPECT_EQ(&interleaved.AsMono()[0], &arr[0]);
-  EXPECT_EQ(interleaved.AsMono(), interleaved.data());
+  EXPECT_EQ(interleaved.AsMono().size(), std::size(arr));
+  EXPECT_EQ(interleaved.AsMono().data(), std::data(arr));
+  EXPECT_EQ(interleaved.AsMono().size(), interleaved.data().size());
+  EXPECT_EQ(interleaved.AsMono().data(), interleaved.data().data());
 
   // Basic iterator test.
   int i = 0;
@@ -121,7 +122,9 @@ TEST(AudioViewTest, DeinterleavedView) {
   auto mono_ch = di.AsMono();
   EXPECT_EQ(NumChannels(mono_ch), 1u);
   EXPECT_EQ(SamplesPerChannel(mono_ch), 10u);
-  EXPECT_EQ(di[0], mono_ch);  // first channel should be same as mono.
+  // first channel should be same as mono.
+  EXPECT_EQ(di[0].data(), mono_ch.data());
+  EXPECT_EQ(di[0].size(), mono_ch.size());
 
   di = DeinterleavedView<int16_t>(arr, 50, 2);
   // Test assignment.
diff --git a/modules/audio_processing/aec3/echo_remover.cc b/modules/audio_processing/aec3/echo_remover.cc
index c720982193..b6f3a7b24f 100644
--- a/modules/audio_processing/aec3/echo_remover.cc
+++ b/modules/audio_processing/aec3/echo_remover.cc
@@ -77,15 +77,14 @@ void LinearEchoPower(const FftData& E,
 void SignalTransition(ArrayView<const float> from,
                       ArrayView<const float> to,
                       ArrayView<float> out) {
-  if (from == to) {
-    RTC_DCHECK_EQ(to.size(), out.size());
+  RTC_DCHECK_EQ(from.size(), to.size());
+  RTC_DCHECK_EQ(from.size(), out.size());
+  if (from.data() == to.data()) {
     std::copy(to.begin(), to.end(), out.begin());
   } else {
     constexpr size_t kTransitionSize = 30;
     constexpr float kOneByTransitionSizePlusOne = 1.f / (kTransitionSize + 1);
 
-    RTC_DCHECK_EQ(from.size(), to.size());
-    RTC_DCHECK_EQ(from.size(), out.size());
     RTC_DCHECK_LE(kTransitionSize, out.size());
 
     for (size_t k = 0; k < kTransitionSize; ++k) {
diff --git a/modules/audio_processing/agc2/rnn_vad/ring_buffer_unittest.cc b/modules/audio_processing/agc2/rnn_vad/ring_buffer_unittest.cc
index 7300121bdb..e223c94479 100644
--- a/modules/audio_processing/agc2/rnn_vad/ring_buffer_unittest.cc
+++ b/modules/audio_processing/agc2/rnn_vad/ring_buffer_unittest.cc
@@ -84,7 +84,7 @@ TEST(RnnVadTest, RingBufferArrayViews) {
       for (int j = i + 1; j < n; ++j) {
         SCOPED_TRACE(j);
         auto view_j = ring_buf.GetArrayView(j);
-        EXPECT_NE(view_i, view_j);
+        EXPECT_NE(view_i.data(), view_j.data());
       }
     }
     ring_buf.Push(pushed_array);
diff --git a/modules/audio_processing/audio_frame_view_unittest.cc b/modules/audio_processing/audio_frame_view_unittest.cc
index e8aee63c26..7952acea83 100644
--- a/modules/audio_processing/audio_frame_view_unittest.cc
+++ b/modules/audio_processing/audio_frame_view_unittest.cc
@@ -86,8 +86,10 @@ TEST(AudioFrameTest, FromDeinterleavedView) {
   AudioFrameView<float> frame_view(view);
   EXPECT_EQ(static_cast<size_t>(frame_view.num_channels()),
             view.num_channels());
-  EXPECT_EQ(frame_view[0], view[0]);
-  EXPECT_EQ(frame_view[1], view[1]);
+  EXPECT_EQ(frame_view[0].size(), view[0].size());
+  EXPECT_EQ(frame_view[0].data(), view[0].data());
+  EXPECT_EQ(frame_view[1].size(), view[1].size());
+  EXPECT_EQ(frame_view[1].data(), view[1].data());
 }
 
 }  // namespace webrtc
diff --git a/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate_unittest.cc
index 1c4a58e5b9..eac237eaac 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate_unittest.cc
@@ -10,6 +10,7 @@
 
 #include "modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.h"
 
+#include <array>
 #include <cstddef>
 #include <cstdint>
 #include <memory>
@@ -276,6 +277,7 @@ TEST_F(RtpSenderVideoFrameTransformerDelegateTest,
   const uint8_t payload_type = 1;
   const uint32_t timestamp = 2;
   const std::vector<uint32_t> frame_csrcs = {123, 456, 789};
+  const std::array<uint8_t, 3> buffer = {3, 2, 1};
 
   auto mock_receiver_frame =
       std::make_unique<NiceMock<MockTransformableVideoFrame>>();
@@ -286,8 +288,6 @@ TEST_F(RtpSenderVideoFrameTransformerDelegateTest,
   metadata.SetRTPVideoHeaderCodecSpecifics(RTPVideoHeaderVP8());
   metadata.SetCsrcs(frame_csrcs);
   ON_CALL(*mock_receiver_frame, Metadata).WillByDefault(Return(metadata));
-  ArrayView<const uint8_t> buffer =
-      (ArrayView<const uint8_t>)*EncodedImageBuffer::Create(1);
   ON_CALL(*mock_receiver_frame, GetData).WillByDefault(Return(buffer));
   ON_CALL(*mock_receiver_frame, GetPayloadType)
       .WillByDefault(Return(payload_type));
@@ -300,12 +300,12 @@ TEST_F(RtpSenderVideoFrameTransformerDelegateTest,
   ASSERT_TRUE(callback);
 
   Event event;
-  EXPECT_CALL(
-      test_sender_,
-      SendVideo(payload_type, kVideoCodecVP8, timestamp,
-                /*capture_time=*/Timestamp::MinusInfinity(), buffer, _, _,
-                /*expected_retransmission_time=*/TimeDelta::Millis(10),
-                frame_csrcs))
+  EXPECT_CALL(test_sender_,
+              SendVideo(payload_type, kVideoCodecVP8, timestamp,
+                        /*capture_time=*/Timestamp::MinusInfinity(),
+                        ElementsAreArray(buffer), _, _,
+                        /*expected_retransmission_time=*/TimeDelta::Millis(10),
+                        frame_csrcs))
       .WillOnce(WithoutArgs([&] {
         event.Set();
         return true;

From ffcb479ca44ae27f7f67cf6d8d2e3b62ba05a2f5 Mon Sep 17 00:00:00 2001
From: Jonas Oreland <jonaso@webrtc.org>
Date: Tue, 17 Feb 2026 06:11:26 -0800
Subject: [PATCH 073/472] Add testcase showing what happens if loosing "last"
 dtls handshake message

BUG=webrtc:367395350

Change-Id: I71630fbb21932aa132642edfc8f46d111a3e4f04
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449762
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46926}
---
 p2p/dtls/dtls_transport_unittest.cc | 250 ++++++++++++++++++++++++++++
 1 file changed, 250 insertions(+)

diff --git a/p2p/dtls/dtls_transport_unittest.cc b/p2p/dtls/dtls_transport_unittest.cc
index afc7d46726..b50ea7a42e 100644
--- a/p2p/dtls/dtls_transport_unittest.cc
+++ b/p2p/dtls/dtls_transport_unittest.cc
@@ -644,13 +644,17 @@ class DtlsTransportInternalImplTestBase {
         IsRtcOk());
   }
 
+  int client1_recv_packets = 0;
+  int client2_recv_packets = 0;
   void AddPacketLogging() {
     client1_.fake_ice_transport()->set_packet_recv_filter(
         [&](auto packet, auto timestamp_us) {
+          client1_recv_packets++;
           return LogRecv(client1_.name(), packet);
         });
     client2_.fake_ice_transport()->set_packet_recv_filter(
         [&](auto packet, auto timestamp_us) {
+          client2_recv_packets++;
           return LogRecv(client2_.name(), packet);
         });
     client1_.set_writable_callback([&]() {});
@@ -698,6 +702,23 @@ class DtlsTransportInternalImplTestBase {
     return drop;
   }
 
+  bool LogBlock(absl::string_view name,
+                bool block,
+                const char* data,
+                size_t len) {
+    auto timestamp_ms = (fake_clock_.TimeNanos() - start_time_ns_) / 1000000;
+    if (block) {
+      RTC_LOG(LS_INFO) << "time=" << timestamp_ms << " : " << name
+                       << ": blocking packet len=" << len
+                       << ", data[0]: " << static_cast<uint8_t>(data[0]);
+    } else {
+      RTC_LOG(LS_INFO) << "time=" << timestamp_ms << " : " << name
+                       << ": SendPacket, len=" << len
+                       << ", data[0]: " << static_cast<uint8_t>(data[0]);
+    }
+    return block;
+  }
+
   template <typename Fn>
   bool WaitUntil(Fn func) {
     return webrtc::WaitUntil(
@@ -2198,5 +2219,234 @@ TEST_P(DtlsInStunTest, OptimalDtls13Handshake) {
 
   ClearPacketFilters();
 }
+
+EndpointConfig Vanilla12(bool client1) {
+  return {
+      .max_protocol_version = SSL_PROTOCOL_DTLS_12,
+      .dtls_in_stun = false,
+      .ice_role = client1 ? ICEROLE_CONTROLLING : ICEROLE_CONTROLLED,
+      .ssl_role = client1 ? SSL_CLIENT : SSL_SERVER,
+      .pqc = false,
+  };
+}
+
+EndpointConfig Vanilla13(bool client1) {
+  return {
+      .max_protocol_version = SSL_PROTOCOL_DTLS_13,
+      .dtls_in_stun = false,
+      .ice_role = client1 ? ICEROLE_CONTROLLING : ICEROLE_CONTROLLED,
+      .ssl_role = client1 ? SSL_CLIENT : SSL_SERVER,
+      .pqc = false,
+  };
+}
+
+struct DtlsTransportLastHandshakePacketTest
+    : public DtlsTransportInternalImplVersionTest {
+  int client1_packet_num = 0;
+  int client2_packet_num = 0;
+
+  int client1_packet_to_block = -1;
+  int client1_packet_size = 0;
+  std::unique_ptr<char[]> client1_packet = nullptr;
+
+  int client2_packet_to_block = -1;
+  int client2_packet_size = 0;
+  std::unique_ptr<char[]> client2_packet = nullptr;
+
+  void InstallBlock() {
+    AddPacketLogging();
+    client1_.fake_ice_transport()->set_packet_send_filter(
+        [&](auto data, auto len, auto options, auto flags) {
+          // Always allow STUN.
+          bool stun = data[0] == 0 || data[0] == 1;
+
+          if (stun) {
+            return false;
+          }
+          if (client1_packet_num++ != client1_packet_to_block) {
+            return LogBlock(client1_.name(), /* block=*/false, data, len);
+          }
+          client1_packet_size = len;
+          client1_packet = std::unique_ptr<char[]>(new char[len]);
+          memcpy(client1_packet.get(), data, len);
+          return LogBlock(client1_.name(), /* block=*/true, data, len);
+        });
+
+    client2_.fake_ice_transport()->set_packet_send_filter(
+        [&](auto data, auto len, auto options, auto flags) {
+          // Always allow STUN.
+          bool stun = data[0] == 0 || data[0] == 1;
+
+          if (stun) {
+            return false;
+          }
+          if (client2_packet_num++ != client2_packet_to_block) {
+            return LogBlock(client2_.name(), /* block=*/false, data, len);
+            return false;
+          }
+          client2_packet_size = len;
+          client2_packet = std::unique_ptr<char[]>(new char[len]);
+          memcpy(client2_packet.get(), data, len);
+          return LogBlock(client2_.name(), /* block=*/true, data, len);
+        });
+  }
+
+  void resend_blocked() {
+    int flags = 0;
+    AsyncSocketPacketOptions packet_options;
+    if (client1_packet != nullptr) {
+      int size = client1_packet_size;
+      RTC_LOG(LS_INFO) << "client1 => client2: RESEND "
+                       << client1_packet.get()[0] << " len= " << size;
+      int rv = client1_.fake_ice_transport()->SendPacket(
+          client1_packet.get(), size, packet_options, flags);
+      ASSERT_EQ(size, rv);
+      client1_packet = nullptr;
+    }
+    if (client2_packet != nullptr) {
+      int size = client2_packet_size;
+      RTC_LOG(LS_INFO) << "client2 => client1: RESEND "
+                       << client2_packet.get()[0] << " len= " << size;
+      int rv = client2_.fake_ice_transport()->SendPacket(
+          client2_packet.get(), size, packet_options, flags);
+      ASSERT_EQ(size, rv);
+      client2_packet = nullptr;
+    }
+  }
+};
+
+TEST_P(DtlsTransportLastHandshakePacketTest,
+       VerifyThatDtls12ClientCanNotDecodeBeforeBecomingWritable) {
+  if (!SSLStreamAdapter::IsBoringSsl()) {
+    GTEST_SKIP() << "Needs boringssl.";
+  }
+
+  if (std::get<0>(GetParam()).max_protocol_version != SSL_PROTOCOL_DTLS_12) {
+    GTEST_SKIP() << "This is a DTLS 1.2 test.";
+  }
+
+  Prepare(false);
+  EXPECT_TRUE(client1_.ConnectIceTransport(&client2_));
+
+  // Block the last handshake message server to client.
+  client2_packet_to_block = 1;
+  InstallBlock();
+
+  client1_.SendIcePing();
+  client2_.SendIcePing();
+  ASSERT_TRUE(WaitUntil([&] {
+    return client1_.fake_ice_transport()->GetCountOfReceivedStunMessages(
+               STUN_BINDING_REQUEST) == 1 &&
+           client2_.fake_ice_transport()->GetCountOfReceivedStunMessages(
+               STUN_BINDING_REQUEST) == 1;
+  }));
+  client1_.SendIcePingConf();
+  client2_.SendIcePingConf();
+  ASSERT_TRUE(WaitUntil([&] {
+    return client1_.fake_ice_transport()->GetCountOfReceivedStunMessages(
+               STUN_BINDING_RESPONSE) == 1 &&
+           client2_.fake_ice_transport()->GetCountOfReceivedStunMessages(
+               STUN_BINDING_RESPONSE) == 1;
+  }));
+
+  ASSERT_TRUE(WaitUntil([&] { return client2_packet != nullptr; }));
+
+  // Now everything except the last message from server to client is exchanged.
+  ASSERT_FALSE(client1_.dtls_transport()->writable());
+  ASSERT_TRUE(client2_.dtls_transport()->writable());
+
+  // Verify that messages sent from client2 can NOT be decoded at client1.
+  int size = 100;
+  int count = 1;
+  client1_recv_packets = 0;
+  client1_.ExpectPackets(size);
+  client2_.SendPackets(size, count, /* srtp= */ false);
+  ASSERT_TRUE(WaitUntil([&] { return client1_recv_packets == 1; }));
+
+  // Last received packet was not decodable!
+  ASSERT_EQ(client1_.NumPacketsReceived(), 0u);
+
+  // Resend the blocked packet.
+  client1_recv_packets = 0;
+  resend_blocked();
+  ASSERT_TRUE(WaitUntil([&] { return client1_recv_packets == 1; }));
+  ASSERT_TRUE(client1_.dtls_transport()->writable());
+  ClearPacketFilters();
+}
+
+INSTANTIATE_TEST_SUITE_P(DtlsTransportLastHandshakePacket12Test,
+                         DtlsTransportLastHandshakePacketTest,
+                         ::testing::Values(std::make_tuple(Vanilla12(true),
+                                                           Vanilla12(false))));
+
+TEST_P(DtlsTransportLastHandshakePacketTest,
+       VerifyThatDtls13ServerCanNotDecodeBeforeBecomingWritable) {
+  if (!SSLStreamAdapter::IsBoringSsl()) {
+    GTEST_SKIP() << "Needs boringssl.";
+  }
+
+  if (std::get<0>(GetParam()).max_protocol_version != SSL_PROTOCOL_DTLS_13) {
+    GTEST_SKIP() << "This is a DTLS 1.3 test.";
+  }
+
+  Prepare(false);
+  EXPECT_TRUE(client1_.ConnectIceTransport(&client2_));
+
+  // Block the ACK from client to server.
+  client1_packet_to_block = 1;
+  InstallBlock();
+
+  client1_.SendIcePing();
+  client2_.SendIcePing();
+  ASSERT_TRUE(WaitUntil([&] {
+    return client1_.fake_ice_transport()->GetCountOfReceivedStunMessages(
+               STUN_BINDING_REQUEST) == 1 &&
+           client2_.fake_ice_transport()->GetCountOfReceivedStunMessages(
+               STUN_BINDING_REQUEST) == 1;
+  }));
+  client1_.SendIcePingConf();
+  client2_.SendIcePingConf();
+  ASSERT_TRUE(WaitUntil([&] {
+    return client1_.fake_ice_transport()->GetCountOfReceivedStunMessages(
+               STUN_BINDING_RESPONSE) == 1 &&
+           client2_.fake_ice_transport()->GetCountOfReceivedStunMessages(
+               STUN_BINDING_RESPONSE) == 1;
+  }));
+
+  ASSERT_TRUE(WaitUntil([&] { return client1_packet != nullptr; }));
+
+  // Now everything except the last message from client to server is exchanged.
+  ASSERT_TRUE(client1_.dtls_transport()->writable());
+  ASSERT_FALSE(client2_.dtls_transport()->writable());
+
+  // Verify that messages sent from client1 can NOT be decoded at server.
+  int size = 100;
+  int count = 1;
+  client2_recv_packets = 0;
+  client2_.ExpectPackets(size);
+  client1_.SendPackets(size, count, /* srtp= */ false);
+  ASSERT_TRUE(WaitUntil([&] { return client2_recv_packets == 1; }));
+
+  // Last received packet was NOT decodable!
+  ASSERT_EQ(client2_.NumPacketsReceived(), 0u);
+
+  // Resend the blocked packet.
+  client2_recv_packets = 0;
+  resend_blocked();
+  ASSERT_TRUE(WaitUntil([&] { return client2_recv_packets == 1; }));
+  ASSERT_TRUE(client2_.dtls_transport()->writable());
+
+  client2_recv_packets = 0;
+  client1_.SendPackets(size, count, /* srtp= */ false);
+  ASSERT_TRUE(WaitUntil([&] { return client2_recv_packets == 1; }));
+  ASSERT_EQ(client2_.NumPacketsReceived(), 1u);
+  ClearPacketFilters();
+}
+
+INSTANTIATE_TEST_SUITE_P(DtlsTransportLastHandshakePacket13Test,
+                         DtlsTransportLastHandshakePacketTest,
+                         ::testing::Values(std::make_tuple(Vanilla13(true),
+                                                           Vanilla13(false))));
+
 }  // namespace
 }  // namespace webrtc

From e0915ee2899b961d94d3eec096f3fa6cfa4552d8 Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Tue, 17 Feb 2026 11:59:12 +0000
Subject: [PATCH 074/472] Stop using Buffer::SetSize in some simple cases.

SetSize() can allow access to uninitialized memory, so should be
avoided. This CL addresses some simple cases, but does not yet
deprecate SetSize().

It also introduces Truncate() as a variant known not to expose
uninitialized memory.

Bug: webrtc:42223681
Change-Id: I136d2394139c7cb271686d634bf6a01edfd5691b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449140
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46927}
---
 rtc_base/BUILD.gn                  |  2 ++
 rtc_base/async_tcp_socket.cc       |  9 +++++----
 rtc_base/buffer.h                  | 22 ++++++++++++++++++----
 rtc_base/physical_socket_server.cc | 17 ++++++++++++-----
 rtc_base/socket.cc                 | 13 +++++++++----
 5 files changed, 46 insertions(+), 17 deletions(-)

diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
index a2022711f8..4a0c8cb9c5 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -1040,6 +1040,7 @@ rtc_library("threading") {
     ":socket_address",
     ":socket_server",
     ":timeutils",
+    "../api:array_view",
     "../api:async_dns_resolver",
     "../api:function_view",
     "../api:location",
@@ -1107,6 +1108,7 @@ rtc_library("socket") {
     ":ip_address",
     ":net_helpers",
     ":socket_address",
+    "../api:array_view",
     "../api/transport:ecn_marking",
     "../api/units:timestamp",
     "system:rtc_export",
diff --git a/rtc_base/async_tcp_socket.cc b/rtc_base/async_tcp_socket.cc
index 9e2221c82a..ad2867ed22 100644
--- a/rtc_base/async_tcp_socket.cc
+++ b/rtc_base/async_tcp_socket.cc
@@ -156,8 +156,7 @@ int AsyncTCPSocketBase::FlushOutBuffer() {
       res = outbuf_.size() - view.size();
     }
     if (view.size() < outbuf_.size()) {
-      memmove(outbuf_.data(), view.data(), view.size());
-      outbuf_.SetSize(view.size());
+      outbuf_.SetData(view);
     }
   }
   return res;
@@ -212,9 +211,11 @@ void AsyncTCPSocketBase::OnReadEvent(Socket* socket) {
     inbuf_.Clear();
   } else {
     if (bytes_remaining > 0) {
-      memmove(inbuf_.data(), inbuf_.data() + processed, bytes_remaining);
+      // Move remaining bytes to beginning of buffer.
+      inbuf_.SetData(ArrayView<uint8_t>(inbuf_).subspan(processed));
+    } else {
+      inbuf_.Clear();
     }
-    inbuf_.SetSize(bytes_remaining);
   }
 }
 
diff --git a/rtc_base/buffer.h b/rtc_base/buffer.h
index 81d84eefe6..f030f6b525 100644
--- a/rtc_base/buffer.h
+++ b/rtc_base/buffer.h
@@ -344,7 +344,7 @@ class BufferT {
   size_t AppendData(size_t max_elements, F&& setter) {
     RTC_DCHECK(IsConsistent());
     const size_t old_size = size_;
-    SetSize(old_size + max_elements);
+    SetSizeInternal(old_size + max_elements);
     size_t written_elements =
         setter(ArrayView<U>(data<U>(), size()).subspan(old_size));
 
@@ -358,9 +358,15 @@ class BufferT {
   // buffer contents will be kept but truncated; if the new size is greater,
   // the existing contents will be kept and the new space will be
   // uninitialized.
-  void SetSize(size_t size) {
-    const size_t old_size = size_;
-    EnsureCapacityWithHeadroom(size, true);
+  // TODO: issues.webrtc.org/42223681 - deprecate and remove the ability to
+  // create uninitialized buffer space.
+  // When we know that the new size is smaller than the old, use Truncate().
+  void SetSize(size_t size) { SetSizeInternal(size); }
+
+  // Truncate the buffer. The buffer contents will be kept but truncated.
+  void Truncate(size_t size) {
+    RTC_DCHECK_LE(size, size_);
+    size_t old_size = size_;
     size_ = size;
     if (ZeroOnFree && size_ < old_size) {
       ZeroTrailingData(old_size - size_);
@@ -404,6 +410,14 @@ class BufferT {
     RTC_DCHECK(IsConsistent());
   }
 
+  void SetSizeInternal(size_t size) {
+    const size_t old_size = size_;
+    EnsureCapacityWithHeadroom(size, true);
+    size_ = size;
+    if (ZeroOnFree && size_ < old_size) {
+      ZeroTrailingData(old_size - size_);
+    }
+  }
   void EnsureCapacityWithHeadroom(size_t capacity, bool extra_headroom) {
     RTC_DCHECK(IsConsistent());
     if (capacity <= capacity_)
diff --git a/rtc_base/physical_socket_server.cc b/rtc_base/physical_socket_server.cc
index 4d986f1752..e11f630271 100644
--- a/rtc_base/physical_socket_server.cc
+++ b/rtc_base/physical_socket_server.cc
@@ -9,6 +9,9 @@
  */
 #include "rtc_base/physical_socket_server.h"
 
+#include <stdio.h>
+
+#include <algorithm>
 #include <array>
 #include <cerrno>
 #include <cstdint>
@@ -16,6 +19,7 @@
 #include <memory>
 #include <utility>
 
+#include "api/array_view.h"
 #include "api/async_dns_resolver.h"
 #include "api/transport/ecn_marking.h"
 #include "api/units/time_delta.h"
@@ -487,11 +491,14 @@ int PhysicalSocket::RecvFrom(ReceiveBuffer& buffer) {
   int64_t timestamp = -1;
   static constexpr int BUF_SIZE = 64 * 1024;
   buffer.payload.EnsureCapacity(BUF_SIZE);
-
-  int received = DoReadFromSocket(
-      buffer.payload.data(), buffer.payload.capacity(), &buffer.source_address,
-      &timestamp, ecn_ ? &buffer.ecn : nullptr);
-  buffer.payload.SetSize(received > 0 ? received : 0);
+  int received;
+  buffer.payload.SetData(BUF_SIZE, [&](ArrayView<uint8_t> payload) {
+    received =
+        DoReadFromSocket(payload.data(), payload.size(), &buffer.source_address,
+                         &timestamp, ecn_ ? &buffer.ecn : nullptr);
+    // return 0 if received is -1, indicating error.
+    return std::max(received, 0);
+  });
   if (received > 0 && timestamp != -1) {
     buffer.arrival_time = Timestamp::Micros(timestamp);
   }
diff --git a/rtc_base/socket.cc b/rtc_base/socket.cc
index 639e78bb80..dee6caa320 100644
--- a/rtc_base/socket.cc
+++ b/rtc_base/socket.cc
@@ -10,8 +10,10 @@
 
 #include "rtc_base/socket.h"
 
+#include <algorithm>
 #include <cstdint>
 
+#include "api/array_view.h"
 #include "api/units/timestamp.h"
 #include "rtc_base/buffer.h"
 
@@ -20,11 +22,14 @@ namespace webrtc {
 int Socket::RecvFrom(ReceiveBuffer& buffer) {
   static constexpr int BUF_SIZE = 64 * 1024;
   int64_t timestamp = -1;
+  int len;
   buffer.payload.EnsureCapacity(BUF_SIZE);
-  int len = RecvFrom(buffer.payload.data(), buffer.payload.capacity(),
-                     &buffer.source_address, &timestamp);
-  buffer.payload.SetSize(len > 0 ? len : 0);
-  if (len > 0 && timestamp != -1) {
+  buffer.payload.SetData(BUF_SIZE, [&](ArrayView<uint8_t> payload) {
+    len = RecvFrom(payload.data(), payload.size(), &buffer.source_address,
+                   &timestamp);
+    return std::max(len, 0);
+  });
+  if (!buffer.payload.empty() && timestamp != -1) {
     buffer.arrival_time = Timestamp::Micros(timestamp);
   }
 

From 82f6bca4a641e60e035e01c6375288d565bc5724 Mon Sep 17 00:00:00 2001
From: Jeremy Leconte <jleconte@webrtc.org>
Date: Tue, 17 Feb 2026 15:23:52 +0100
Subject: [PATCH 075/472] Add DIR_METADATA for tools_webrtc

* remove empty tools_webrtc/chromiumos folder
* move sanitizer suppression files under the sanitizer directory
* clean up OWNERS_INFRA file

Change-Id: Ic7c19039c17068909f7ba9465ff6d6d062e723ea
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449880
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46928}
---
 OWNERS_INFRA                                    | 17 +++++++++--------
 build_overrides/build.gni                       |  9 ++++++---
 tools_webrtc/DIR_METADATA                       |  7 +++++++
 tools_webrtc/chromiumos/OWNERS                  |  3 ---
 .../msan_suppressions.txt}                      |  0
 .../ubsan_suppressions.txt}                     |  0
 .../ubsan_vptr_suppressions.txt}                |  0
 7 files changed, 22 insertions(+), 14 deletions(-)
 create mode 100644 tools_webrtc/DIR_METADATA
 delete mode 100644 tools_webrtc/chromiumos/OWNERS
 rename tools_webrtc/{msan/suppressions.txt => sanitizers/msan_suppressions.txt} (100%)
 rename tools_webrtc/{ubsan/suppressions.txt => sanitizers/ubsan_suppressions.txt} (100%)
 rename tools_webrtc/{ubsan/vptr_suppressions.txt => sanitizers/ubsan_vptr_suppressions.txt} (100%)

diff --git a/OWNERS_INFRA b/OWNERS_INFRA
index 0a192b9c36..3c7098037d 100644
--- a/OWNERS_INFRA
+++ b/OWNERS_INFRA
@@ -1,20 +1,21 @@
-#Owners for infra and repo related files
+# Owners for infra and repo related files
 per-file .gitignore=*
+per-file AUTHORS=*
+per-file DEPS=*
+per-file WATCHLISTS=*
+per-file whitespace.txt=*
+
 per-file .gn=mbonadei@webrtc.org,jansson@webrtc.org,jleconte@webrtc.org
 per-file BUILD.gn=mbonadei@webrtc.org,jansson@webrtc.org,jleconte@webrtc.org
 per-file .../BUILD.gn=mbonadei@webrtc.org,jansson@webrtc.org,jleconte@webrtc.org
 per-file *.gni=mbonadei@webrtc.org,jansson@webrtc.org,jleconte@webrtc.org
 per-file .../*.gni=mbonadei@webrtc.org,jansson@webrtc.org,jleconte@webrtc.org
+
 per-file .vpython=mbonadei@webrtc.org,jansson@webrtc.org,jleconte@webrtc.org
 per-file .vpython3=mbonadei@webrtc.org,jansson@webrtc.org,jleconte@webrtc.org
-per-file AUTHORS=*
-per-file DEPS=*
 per-file pylintrc=mbonadei@webrtc.org,jansson@webrtc.org,jleconte@webrtc.org
-per-file .rustfmt.toml=boivie@webrtc.org,mbonadei@webrtc.org,jleconte@webrtc.org
 per-file pylintrc_old_style=mbonadei@webrtc.org,jansson@webrtc.org,jleconte@webrtc.org
-per-file WATCHLISTS=*
-per-file whitespace.txt=*
-per-file native-api.md=mbonadei@webrtc.org
-per-file ....lua=titovartem@webrtc.org
 per-file .style.yapf=jleconte@webrtc.org
 per-file *.py=mbonadei@webrtc.org,jansson@webrtc.org,jleconte@webrtc.org
+
+per-file .rustfmt.toml=boivie@webrtc.org,mbonadei@webrtc.org,jleconte@webrtc.org
diff --git a/build_overrides/build.gni b/build_overrides/build.gni
index e15bc95707..3abfcb71d0 100644
--- a/build_overrides/build.gni
+++ b/build_overrides/build.gni
@@ -21,11 +21,14 @@ asan_suppressions_file = "//build/sanitizers/asan_suppressions.cc"
 lsan_suppressions_file = "//tools_webrtc/sanitizers/lsan_suppressions_webrtc.cc"
 tsan_suppressions_file = "//tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc"
 msan_ignorelist_path =
-    rebase_path("//tools_webrtc/msan/suppressions.txt", root_build_dir)
+    rebase_path("//tools_webrtc/sanitizers/msan_suppressions.txt",
+                root_build_dir)
 ubsan_ignorelist_path =
-    rebase_path("//tools_webrtc/ubsan/suppressions.txt", root_build_dir)
+    rebase_path("//tools_webrtc/sanitizers/ubsan_suppressions.txt",
+                root_build_dir)
 ubsan_vptr_ignorelist_path =
-    rebase_path("//tools_webrtc/ubsan/vptr_suppressions.txt", root_build_dir)
+    rebase_path("//tools_webrtc/sanitizers/ubsan_vptr_suppressions.txt",
+                root_build_dir)
 
 # For Chromium, Android 32-bit non-component, non-clang builds hit a 4GiB size
 # limit, making them requiring symbol_level=2. WebRTC doesn't hit that problem
diff --git a/tools_webrtc/DIR_METADATA b/tools_webrtc/DIR_METADATA
new file mode 100644
index 0000000000..469e068491
--- /dev/null
+++ b/tools_webrtc/DIR_METADATA
@@ -0,0 +1,7 @@
+team_email: "webrtc-infra@google.com"
+monorail: {
+  component: "WebRTC>Tools"
+}
+buganizer_public: {
+  component_id: 1565968
+}
diff --git a/tools_webrtc/chromiumos/OWNERS b/tools_webrtc/chromiumos/OWNERS
deleted file mode 100644
index a15b0e6155..0000000000
--- a/tools_webrtc/chromiumos/OWNERS
+++ /dev/null
@@ -1,3 +0,0 @@
-aaronyu@google.com
-pteerapong@google.com
-mbonadei@webrtc.org
diff --git a/tools_webrtc/msan/suppressions.txt b/tools_webrtc/sanitizers/msan_suppressions.txt
similarity index 100%
rename from tools_webrtc/msan/suppressions.txt
rename to tools_webrtc/sanitizers/msan_suppressions.txt
diff --git a/tools_webrtc/ubsan/suppressions.txt b/tools_webrtc/sanitizers/ubsan_suppressions.txt
similarity index 100%
rename from tools_webrtc/ubsan/suppressions.txt
rename to tools_webrtc/sanitizers/ubsan_suppressions.txt
diff --git a/tools_webrtc/ubsan/vptr_suppressions.txt b/tools_webrtc/sanitizers/ubsan_vptr_suppressions.txt
similarity index 100%
rename from tools_webrtc/ubsan/vptr_suppressions.txt
rename to tools_webrtc/sanitizers/ubsan_vptr_suppressions.txt

From a075f1298c454f77be141c8aa2d8ad66f1fa03e7 Mon Sep 17 00:00:00 2001
From: Mirko Bonadei <mbonadei@webrtc.org>
Date: Tue, 17 Feb 2026 14:30:01 +0000
Subject: [PATCH 076/472] Update style guide to mention AI coding.

Bug: None
No-Try: True
Change-Id: Ia6f2d07aebf59389a3a3ca4b23fd3b9526d9fae5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449920
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46929}
---
 g3doc/style-guide.md | 4 ++++
 1 file changed, 4 insertions(+)

diff --git a/g3doc/style-guide.md b/g3doc/style-guide.md
index 1fb03bcc66..afc886e118 100644
--- a/g3doc/style-guide.md
+++ b/g3doc/style-guide.md
@@ -9,6 +9,10 @@ Some older parts of the code violate the style guide in various ways.
 If making large changes to such code, consider first cleaning it up in a
   separate CL.
 
+WebRTC's policy about AI coding matches [Chromium's policy][ai-policy].
+
+[ai-policy]: https://chromium.googlesource.com/chromium/src.git/+/HEAD/agents/ai_policy.md
+
 ## C++
 
 WebRTC follows the [Chromium C++ style guide][chr-style] and the

From a20f01e98afbf9660b92d6180e1095b1f5abd2f6 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Erik=20Spr=C3=A5ng?= <sprang@webrtc.org>
Date: Mon, 16 Feb 2026 15:23:11 +0100
Subject: [PATCH 077/472] SimulcastEncoderAdapter: Avoid holding lock while
 calling callback.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

To reduce the risk of deadlocks, avoid holding mutexes (e.g.
`pending_frames_mutex_`) while performing callbacks. Even if it is
just a frame drop callback.

Bug: webrtc:467444018
Change-Id: Iddf89f8bc2546ac24f36417005d292ed91cc80df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449687
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46930}
---
 media/engine/simulcast_encoder_adapter.cc | 57 +++++++++++++----------
 1 file changed, 33 insertions(+), 24 deletions(-)

diff --git a/media/engine/simulcast_encoder_adapter.cc b/media/engine/simulcast_encoder_adapter.cc
index c55919c114..f290a1c266 100644
--- a/media/engine/simulcast_encoder_adapter.cc
+++ b/media/engine/simulcast_encoder_adapter.cc
@@ -566,35 +566,44 @@ int SimulcastEncoderAdapter::Encode(
     // not reporting the completion of frames via either OnEncodedFrame or
     // OnFrameDropped.
     constexpr size_t kMaxPendingFrames = 15;
-    MutexLock lock(&pending_frames_mutex_);
-    if (pending_frames_.size() >= kMaxPendingFrames) {
-      // Drop the oldest frame.
-      PendingFrame& dropped_frame = pending_frames_.front();
-      RTC_LOG(LS_WARNING) << "Pending frames queue full. Dropping frame with "
-                             "rtp_timestamp="
-                          << dropped_frame.rtp_timestamp;
-      for (size_t stream_idx = 0; stream_idx < kMaxSimulcastStreams;
-           ++stream_idx) {
-        if (dropped_frame.expected_layer_index.test(stream_idx)) {
-          dropped_frame.expected_layer_index.reset(stream_idx);
-          bool is_last = dropped_frame.expected_layer_index.none();
-          encoded_complete_callback_->OnFrameDropped(
-              dropped_frame.rtp_timestamp, stream_idx, is_last);
+    std::vector<std::tuple<uint32_t, int, bool>> frame_drops;
+    {
+      MutexLock lock(&pending_frames_mutex_);
+      if (pending_frames_.size() >= kMaxPendingFrames) {
+        // Drop the oldest frame.
+        PendingFrame& dropped_frame = pending_frames_.front();
+        RTC_LOG(LS_WARNING) << "Pending frames queue full. Dropping frame with "
+                               "rtp_timestamp="
+                            << dropped_frame.rtp_timestamp;
+        for (size_t stream_idx = 0; stream_idx < kMaxSimulcastStreams;
+             ++stream_idx) {
+          if (dropped_frame.expected_layer_index.test(stream_idx)) {
+            dropped_frame.expected_layer_index.reset(stream_idx);
+            bool is_last = dropped_frame.expected_layer_index.none();
+            frame_drops.emplace_back(dropped_frame.rtp_timestamp, stream_idx,
+                                     is_last);
+          }
+        }
+        pending_frames_.pop_front();
+      }
+
+      std::bitset<kMaxSimulcastStreams> expected_layer_index;
+      for (const auto& layer : stream_contexts_) {
+        if (!layer.is_paused()) {
+          expected_layer_index.set(layer.stream_idx());
         }
       }
-      pending_frames_.pop_front();
+      pending_frames_.push_back({
+          .rtp_timestamp = input_image.rtp_timestamp(),
+          .expected_layer_index = expected_layer_index,
+      });
     }
 
-    std::bitset<kMaxSimulcastStreams> expected_layer_index;
-    for (const auto& layer : stream_contexts_) {
-      if (!layer.is_paused()) {
-        expected_layer_index.set(layer.stream_idx());
-      }
+    // Make frame drop callbacks outside of the lock.
+    for (const auto& [rtp_timestamp, stream_idx, is_last] : frame_drops) {
+      encoded_complete_callback_->OnFrameDropped(rtp_timestamp, stream_idx,
+                                                 is_last);
     }
-    pending_frames_.push_back({
-        .rtp_timestamp = input_image.rtp_timestamp(),
-        .expected_layer_index = expected_layer_index,
-    });
   }
 
   for (auto& layer : stream_contexts_) {

From 114ea00794ce9f723238887eee6f6974bb30ecc9 Mon Sep 17 00:00:00 2001
From: Jeremy Leconte <jleconte@webrtc.org>
Date: Tue, 17 Feb 2026 17:50:07 +0100
Subject: [PATCH 078/472] Fix malformed DEPS files in WebRTC.

Bug: webrtc:447469714
Change-Id: I4a18d0d609d175f83942583d36eb7453e2507983
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449981
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46931}
---
 DEPS                                  |   2 +-
 api/DEPS                              | 104 +++++++++++++-------------
 api/neteq/DEPS                        |   8 +-
 api/numerics/DEPS                     |   2 +-
 api/task_queue/DEPS                   |   6 +-
 api/test/DEPS                         |  30 ++++----
 api/test/metrics/DEPS                 |   8 +-
 api/test/pclf/DEPS                    |   2 +-
 api/test/video/DEPS                   |   2 +-
 api/transport/DEPS                    |   4 +-
 api/transport/rtp/DEPS                |   2 +-
 api/video/DEPS                        |  36 ++++-----
 api/voip/DEPS                         |   2 +-
 audio/DEPS                            |   4 +-
 call/DEPS                             |  34 ++++-----
 media/DEPS                            |  16 ++--
 modules/audio_coding/codecs/opus/DEPS |   2 +-
 modules/audio_device/DEPS             |   4 +-
 modules/congestion_controller/DEPS    |   2 +-
 modules/desktop_capture/DEPS          |  10 +--
 modules/video_coding/DEPS             |   4 +-
 net/dcsctp/socket/DEPS                |   2 +-
 pc/DEPS                               |   2 +-
 rtc_base/DEPS                         |  28 +++----
 rtc_base/experiments/DEPS             |   2 +-
 rtc_base/synchronization/DEPS         |   6 +-
 rtc_base/system/DEPS                  |   2 +-
 rtc_tools/DEPS                        |   8 +-
 system_wrappers/source/DEPS           |   2 +-
 test/DEPS                             |  34 ++++-----
 30 files changed, 185 insertions(+), 185 deletions(-)

diff --git a/DEPS b/DEPS
index ea80f58948..7f04a3fcff 100644
--- a/DEPS
+++ b/DEPS
@@ -1373,7 +1373,7 @@ include_rules = [
 ]
 
 specific_include_rules = {
-  "webrtc_lib_link_test\.cc": [
+  "webrtc_lib_link_test\\.cc": [
     "+media/engine",
     "+modules/audio_device",
     "+modules/audio_processing",
diff --git a/api/DEPS b/api/DEPS
index dc2d1f1cf3..98f9f35cbe 100644
--- a/api/DEPS
+++ b/api/DEPS
@@ -45,110 +45,110 @@ include_rules = [
 specific_include_rules = {
   # Some internal headers are allowed even in API headers:
 
-  ".*\.h": [
+  ".*\\.h": [
     "+rtc_base/checks.h",
     "+rtc_base/system/rtc_export.h",
     "+rtc_base/system/rtc_export_template.h",
     "+rtc_base/units/unit_base.h",
   ],
 
-  "array_view\.h": [
+  "array_view\\.h": [
     "+rtc_base/type_traits.h",
   ],
 
   # Needed because AudioEncoderOpus is in the wrong place for
   # backwards compatibilty reasons. See
   # https://bugs.chromium.org/p/webrtc/issues/detail?id=7847
-  "audio_encoder_opus\.h": [
+  "audio_encoder_opus\\.h": [
     "+modules/audio_coding/codecs/opus/audio_encoder_opus.h",
   ],
 
-  "async_resolver_factory\.h": [
+  "async_resolver_factory\\.h": [
     "+rtc_base/async_resolver_interface.h",
   ],
 
-  "async_dns_resolver\.h": [
+  "async_dns_resolver\\.h": [
     "+rtc_base/socket_address.h",
   ],
 
-  "audio_device_defines\.h": [
+  "audio_device_defines\\.h": [
     "+rtc_base/strings/string_builder.h",
   ],
 
-  "audio_format\.h": [
+  "audio_format\\.h": [
     "+rtc_base/strings/string_builder.h",
   ],
 
-  "candidate\.h": [
+  "candidate\\.h": [
     "+rtc_base/network_constants.h",
     "+rtc_base/socket_address.h",
   ],
 
-  "create_peerconnection_factory\.h": [
+  "create_peerconnection_factory\\.h": [
     "+rtc_base/thread.h",
   ],
 
-  "data_channel_interface\.h": [
+  "data_channel_interface\\.h": [
     "+rtc_base/copy_on_write_buffer.h",
   ],
 
-  "data_channel_transport_interface\.h": [
+  "data_channel_transport_interface\\.h": [
     "+rtc_base/copy_on_write_buffer.h",
   ],
 
-  "datagram_connection\.h": [
+  "datagram_connection\\.h": [
     "+p2p/base/transport_description.h",
   ],
 
-  "mock_datagram_connection\.h": [
+  "mock_datagram_connection\\.h": [
     "+p2p/base/transport_description.h",
   ],
 
-  "datagram_connection_factory\.h": [
+  "datagram_connection_factory\\.h": [
     "+p2p/base/port_allocator.h",
     "+rtc_base/rtc_certificate.h",
   ],
 
-  "dtls_transport_interface\.h": [
+  "dtls_transport_interface\\.h": [
     "+rtc_base/ssl_certificate.h",
   ],
 
-  "fec_controller\.h": [
+  "fec_controller\\.h": [
     "+modules/include/module_fec_types.h",
   ],
 
-  "ice_server_parsing\.h": [
+  "ice_server_parsing\\.h": [
     "+p2p/base/port_allocator.h",
     "+rtc_base/socket_address.h",
   ],
 
-  "jsep\.h": [
+  "jsep\\.h": [
     "+absl/strings/has_absl_stringify.h",
     "+absl/strings/str_format.h",
     "+rtc_base/system/no_unique_address.h",
     "+rtc_base/thread_annotations.h",
   ],
 
-  "local_network_access_permission\.h": [
+  "local_network_access_permission\\.h": [
     "+rtc_base/socket_address.h",
   ],
 
-  "packet_socket_factory\.h": [
+  "packet_socket_factory\\.h": [
     "+rtc_base/async_packet_socket.h",
     "+rtc_base/socket_address.h",
     "+rtc_base/ssl_certificate.h",
   ],
 
-  "payload_type\.h": [
+  "payload_type\\.h": [
     "+absl/strings/str_format.h",
     "+rtc_base/strong_alias.h",
   ],
 
-  "turn_customizer\.h": [
+  "turn_customizer\\.h": [
     "+p2p/base/port_interface.h",
   ],
 
-  "peer_connection_interface\.h": [
+  "peer_connection_interface\\.h": [
     "+media/base/media_config.h",
     "+media/base/media_engine.h",
     "+p2p/base/port.h",
@@ -167,125 +167,125 @@ specific_include_rules = {
     "+rtc_base/thread.h",
   ],
 
-  "proxy\.h": [
+  "proxy\\.h": [
     "+rtc_base/event.h",
     "+rtc_base/message_handler.h",  # Inherits from it.
     "+rtc_base/thread.h",
   ],
 
-  "ref_counted_base\.h": [
+  "ref_counted_base\\.h": [
     "+rtc_base/ref_counter.h",
   ],
 
-  "rtc_error\.h": [
+  "rtc_error\\.h": [
     "+rtc_base/logging.h",
     "+rtc_base/strings/string_builder.h",
     "+absl/strings/has_absl_stringify.h",
     "+absl/strings/str_format.h",
   ],
-  "rtc_event_log_output_file.h": [
+  "rtc_event_log_output_file\\.h": [
     # For private member and constructor.
     "+rtc_base/system/file_wrapper.h",
   ],
 
-  "legacy_stats_types\.h": [
+  "legacy_stats_types\\.h": [
     "+rtc_base/thread_annotations.h",
     "+rtc_base/thread_checker.h",
   ],
 
-  "audio_decoder\.h": [
+  "audio_decoder\\.h": [
     "+rtc_base/buffer.h",
   ],
 
-  "audio_encoder\.h": [
+  "audio_encoder\\.h": [
     "+rtc_base/buffer.h",
   ],
 
-  "make_ref_counted\.h": [
+  "make_ref_counted\\.h": [
     "+rtc_base/ref_counted_object.h",
   ],
 
-  "mock.*\.h": [
+  "mock.*\\.h": [
     "+test/gmock.h",
   ],
 
-  "mock_peerconnectioninterface\.h": [
+  "mock_peerconnectioninterface\\.h": [
     "+rtc_base/ref_counted_object.h",
   ],
 
-  "mock_video_track\.h": [
+  "mock_video_track\\.h": [
     "+rtc_base/ref_counted_object.h",
   ],
 
-  "notifier\.h": [
+  "notifier\\.h": [
     "+rtc_base/system/no_unique_address.h",
     "+rtc_base/thread_annotations.h",
   ],
 
-  "priority\.h": [
+  "priority\\.h": [
     "+rtc_base/strong_alias.h",
   ],
 
-  "sctp_transport_interface\.h": [
+  "sctp_transport_interface\\.h": [
     "+absl/strings/str_format.h",
   ],
 
-  "simulated_network\.h": [
+  "simulated_network\\.h": [
     "+rtc_base/random.h",
     "+rtc_base/thread_annotations.h",
   ],
 
-  "time_controller\.h": [
+  "time_controller\\.h": [
     "+rtc_base/thread.h",
   ],
 
-  "videocodec_test_fixture\.h": [
+  "videocodec_test_fixture\\.h": [
     "+modules/video_coding/include/video_codec_interface.h"
   ],
 
-  "rtp_parameters\.h": [
+  "rtp_parameters\\.h": [
     "+absl/strings/str_format.h",
   ],
 
-  "sequence_checker\.h": [
+  "sequence_checker\\.h": [
     "+rtc_base/synchronization/sequence_checker_internal.h",
     "+rtc_base/thread_annotations.h",
   ],
 
-  "video_encoder_factory_template.*\.h": [
+  "video_encoder_factory_template.*\\.h": [
     "+modules/video_coding",
   ],
 
-  "video_encoder_factory_interface\.h": [
+  "video_encoder_factory_interface\\.h": [
     "+rtc_base/numerics",
   ],
 
-  "video_encoder_interface\.h": [
+  "video_encoder_interface\\.h": [
     "+rtc_base/numerics",
   ],
 
-  "simple_encoder_wrapper\.h": [
+  "simple_encoder_wrapper\\.h": [
     "+common_video",
     "+modules",
   ],
 
-  "video_decoder_factory_template.*\.h": [
+  "video_decoder_factory_template.*\\.h": [
     "+modules/video_coding",
   ],
 
-  "field_trials\.h": [
+  "field_trials\\.h": [
     "+rtc_base/containers/flat_map.h",
   ],
 
-  "video_track_source_proxy_factory.h": [
+  "video_track_source_proxy_factory\\.h": [
     "+rtc_base/thread.h",
   ],
 
-  "field_trials_registry\.h": [
+  "field_trials_registry\\.h": [
     "+rtc_base/containers/flat_set.h",
   ],
 
-  "ice_transport_factory\.h": [
+  "ice_transport_factory\\.h": [
     "+p2p/base/port_allocator.h",
   ],
 
@@ -293,7 +293,7 @@ specific_include_rules = {
   # so we re-add all the top-level directories here. (That's because .h
   # files leak their #includes to whoever's #including them, but .cc files
   # do not since no one #includes them.)
-  ".*\.cc": [
+  ".*\\.cc": [
     "+audio",
     "+call",
     "+common_audio",
diff --git a/api/neteq/DEPS b/api/neteq/DEPS
index 6c1c602b42..4eb72f0567 100644
--- a/api/neteq/DEPS
+++ b/api/neteq/DEPS
@@ -1,14 +1,14 @@
 specific_include_rules = {
-  "custom_neteq_factory\.h": [
+  "custom_neteq_factory\\.h": [
     "+system_wrappers/include/clock.h",
   ],
-  "default_neteq_factory\.h": [
+  "default_neteq_factory\\.h": [
     "+system_wrappers/include/clock.h",
   ],
-  "neteq_controller\.h": [
+  "neteq_controller\\.h": [
     "+system_wrappers/include/clock.h",
   ],
-  "neteq_factory\.h": [
+  "neteq_factory\\.h": [
     "+system_wrappers/include/clock.h",
   ],
 }
diff --git a/api/numerics/DEPS b/api/numerics/DEPS
index 2d89d57557..02b9be2669 100644
--- a/api/numerics/DEPS
+++ b/api/numerics/DEPS
@@ -1,6 +1,6 @@
 specific_include_rules = {
   # Some internal headers are allowed even in API headers:
-  "samples_stats_counter\.h": [
+  "samples_stats_counter\\.h": [
     "+rtc_base/numerics/running_statistics.h",
   ]
 }
diff --git a/api/task_queue/DEPS b/api/task_queue/DEPS
index 1365edb504..a56f855c60 100644
--- a/api/task_queue/DEPS
+++ b/api/task_queue/DEPS
@@ -1,13 +1,13 @@
 specific_include_rules = {
-  "task_queue_base\.h": [
+  "task_queue_base\\.h": [
     # Make TaskQueueBase RTC_LOCKABALE to allow annotate variables are only
     # accessed on specific task queue.
     "+rtc_base/thread_annotations.h",
   ],
-  "task_queue_test\.h": [
+  "task_queue_test\\.h": [
     "+test/gtest.h",
   ],
-  "pending_task_safety_flag.h": [
+  "pending_task_safety_flag\\.h": [
     "+rtc_base/checks.h",
     "+rtc_base/system/no_unique_address.h",
   ],
diff --git a/api/test/DEPS b/api/test/DEPS
index b3504bd021..2fbfb6d4f1 100644
--- a/api/test/DEPS
+++ b/api/test/DEPS
@@ -8,23 +8,23 @@ specific_include_rules = {
   ".*": [
     "+rtc_base/ref_counted_object.h",
   ],
-  "dummy_peer_connection\.h": [
+  "dummy_peer_connection\\.h": [
     "+rtc_base/ref_counted_object.h",
   ],
-  "neteq_factory_with_codecs\.h": [
+  "neteq_factory_with_codecs\\.h": [
     "+system_wrappers/include/clock.h",
   ],
-  "network_emulation_manager\.h": [
+  "network_emulation_manager\\.h": [
     "+rtc_base/network_constants.h",
     "+rtc_base/ip_address.h",
     "+rtc_base/socket_address.h",
   ],
-  "peer_network_dependencies\.h": [
+  "peer_network_dependencies\\.h": [
     "+rtc_base/socket_factory.h",
     "+rtc_base/network.h",
     "+rtc_base/thread.h",
   ],
-  "peerconnection_quality_test_fixture\.h": [
+  "peerconnection_quality_test_fixture\\.h": [
     "+logging/rtc_event_log/rtc_event_log_factory_interface.h",
     "+rtc_base/network.h",
     "+rtc_base/rtc_certificate_generator.h",
@@ -33,38 +33,38 @@ specific_include_rules = {
     "+media/base/media_constants.h",
     "+modules/audio_processing/include/audio_processing.h",
   ],
-  "time_controller\.h": [
+  "time_controller\\.h": [
     "+rtc_base/synchronization/yield_policy.h",
     "+system_wrappers/include/clock.h",
     "+rtc_base/socket_server.h",
   ],
-  "create_frame_generator\.h": [
+  "create_frame_generator\\.h": [
     "+system_wrappers/include/clock.h",
   ],
-  "mock_async_dns_resolver\.h": [
+  "mock_async_dns_resolver\\.h": [
     "+rtc_base/socket_address.h",
   ],
-  "mock_local_network_access_permission\.h": [
+  "mock_local_network_access_permission\\.h": [
     "+rtc_base/socket_address.h",
   ],
-  "mock_packet_socket_factory\.h": [
+  "mock_packet_socket_factory\\.h": [
     "+rtc_base/async_packet_socket.h",
     "+rtc_base/socket_address.h",
   ],
-  "mock_peerconnectioninterface\.h": [
+  "mock_peerconnectioninterface\\.h": [
     "+rtc_base/thread.h",
   ],
-  "mock_peer_connection_factory_interface\.h": [
+  "mock_peer_connection_factory_interface\\.h": [
     "+p2p/base/port_allocator.h",
     "+rtc_base/rtc_certificate_generator.h",
   ],
-  "mock_peerconnectioninterface\.h": [
+  "mock_peerconnectioninterface\\.h": [
     "+rtc_base/thread.h",
   ],
-  "videocodec_test_fixture\.h": [
+  "videocodec_test_fixture\\.h": [
     "+modules/video_coding/codecs/h264/include/h264_globals.h",
   ],
-  "rtc_error_matchers\.h": [
+  "rtc_error_matchers\\.h": [
     "+test/gmock.h",
   ],
 }
diff --git a/api/test/metrics/DEPS b/api/test/metrics/DEPS
index 9edcc370ed..d7929a5852 100644
--- a/api/test/metrics/DEPS
+++ b/api/test/metrics/DEPS
@@ -1,18 +1,18 @@
 specific_include_rules = {
-  "metrics_logger_and_exporter\.h": [
+  "metrics_logger_and_exporter\\.h": [
     "+rtc_base/synchronization/mutex.h",
     "+system_wrappers/include/clock.h",
   ],
-  "metrics_logger\.h": [
+  "metrics_logger\\.h": [
     "+rtc_base/synchronization/mutex.h",
     "+rtc_base/thread_annotations.h",
     "+system_wrappers/include/clock.h",
   ],
-  "metrics_accumulator\.h": [
+  "metrics_accumulator\\.h": [
     "+rtc_base/synchronization/mutex.h",
     "+rtc_base/thread_annotations.h",
   ],
-  "chrome_perf_dashboard_metrics_exporter_test.cc": [
+  "chrome_perf_dashboard_metrics_exporter_test\\.cc": [
     "+tracing/tracing/proto/histogram.pb.h"
   ],
 }
diff --git a/api/test/pclf/DEPS b/api/test/pclf/DEPS
index ec2451b05b..088237970a 100644
--- a/api/test/pclf/DEPS
+++ b/api/test/pclf/DEPS
@@ -8,7 +8,7 @@ specific_include_rules = {
     "+rtc_base/socket_factory.h",
     "+rtc_base/thread.h",
   ],
-  "media_quality_test_params\.h": [
+  "media_quality_test_params\\.h": [
     "+p2p/base/port_allocator.h",
     "+rtc_base/socket_factory.h",
   ],
diff --git a/api/test/video/DEPS b/api/test/video/DEPS
index 2256b34db2..63d42aab75 100644
--- a/api/test/video/DEPS
+++ b/api/test/video/DEPS
@@ -1,5 +1,5 @@
 specific_include_rules = {
-  "test_video_track_source\.h": [
+  "test_video_track_source\\.h": [
     "+rtc_base/thread_annotations.h",
     "+rtc_base/system/no_unique_address.h",
   ],
diff --git a/api/transport/DEPS b/api/transport/DEPS
index 2202424f06..7da3887844 100644
--- a/api/transport/DEPS
+++ b/api/transport/DEPS
@@ -1,11 +1,11 @@
 specific_include_rules = {
-  "stun\.h": [
+  "stun\\.h": [
     "+rtc_base/byte_buffer.h",
     "+rtc_base/ip_address.h",
     "+rtc_base/net_helpers.h",
     "+rtc_base/socket_address.h",
   ],
-  "data_channel_transport_interface\.h": [
+  "data_channel_transport_interface\\.h": [
     "+rtc_base/ssl_stream_adapter.h",
   ],
 }
diff --git a/api/transport/rtp/DEPS b/api/transport/rtp/DEPS
index 97daa2bb67..562e1b6deb 100644
--- a/api/transport/rtp/DEPS
+++ b/api/transport/rtp/DEPS
@@ -1,5 +1,5 @@
 specific_include_rules = {
-  "rtp_source\.h": [
+  "rtp_source\\.h": [
     "+absl/strings/str_format.h",
   ],
 }
diff --git a/api/video/DEPS b/api/video/DEPS
index fd3ad70862..741e79a0e5 100644
--- a/api/video/DEPS
+++ b/api/video/DEPS
@@ -1,77 +1,77 @@
 specific_include_rules = {
-  "encoded_frame.h" : [
+  "encoded_frame\\.h" : [
     "+modules/rtp_rtcp/source/rtp_video_header.h",
     "+modules/video_coding/include/video_codec_interface.h",
     "+modules/video_coding/include/video_coding_defines.h",
     "+common_video/frame_instrumentation_data.h",
   ],
-  "encoded_image\.h" : [
+  "encoded_image\\.h" : [
     "+rtc_base/buffer.h",
     "+rtc_base/ref_count.h",
   ],
 
-  "i010_buffer\.h": [
+  "i010_buffer\\.h": [
     "+rtc_base/memory/aligned_malloc.h",
   ],
 
-  "i210_buffer\.h": [
+  "i210_buffer\\.h": [
     "+rtc_base/memory/aligned_malloc.h",
   ],
 
-  "i410_buffer\.h": [
+  "i410_buffer\\.h": [
     "+rtc_base/memory/aligned_malloc.h",
   ],
 
-  "i420_buffer\.h": [
+  "i420_buffer\\.h": [
     "+rtc_base/memory/aligned_malloc.h",
   ],
 
-  "i422_buffer\.h": [
+  "i422_buffer\\.h": [
     "+rtc_base/memory/aligned_malloc.h",
   ],
 
-  "i444_buffer\.h": [
+  "i444_buffer\\.h": [
     "+rtc_base/memory/aligned_malloc.h",
   ],
 
-  "nv12_buffer\.h": [
+  "nv12_buffer\\.h": [
     "+rtc_base/memory/aligned_malloc.h",
   ],
 
-  "recordable_encoded_frame\.h": [
+  "recordable_encoded_frame\\.h": [
     "+rtc_base/ref_count.h",
   ],
 
-  "video_frame\.h": [
+  "video_frame\\.h": [
   ],
 
-  "video_frame_buffer\.h": [
+  "video_frame_buffer\\.h": [
     "+rtc_base/ref_count.h",
   ],
 
-  "video_frame_metadata\.h": [
+  "video_frame_metadata\\.h": [
     "+modules/video_coding/codecs/h264/include/h264_globals.h",
     "+modules/video_coding/codecs/vp8/include/vp8_globals.h",
     "+modules/video_coding/codecs/vp9/include/vp9_globals.h",
   ],
 
-  "video_stream_decoder_create.cc": [
+  "video_stream_decoder_create\\.cc": [
     "+video/video_stream_decoder_impl.h",
   ],
 
-  "video_stream_encoder_create.cc": [
+  "video_stream_encoder_create\\.cc": [
     "+video/video_stream_encoder.h",
   ],
 
-  "rtp_video_frame_assembler.h": [
+  "rtp_video_frame_assembler\\.h": [
     "+modules/rtp_rtcp/source/rtp_packet_received.h",
   ],
 
-  "frame_buffer.h": [
+  "frame_buffer\\.h": [
     "+modules/video_coding/utility/decoded_frames_history.h",
   ],
 
-  "video_frame_matchers\.h": [
+  "video_frame_matchers\\.h": [
     "+test/gmock.h",
   ],
 }
diff --git a/api/voip/DEPS b/api/voip/DEPS
index a36df41a9c..3994acffd8 100644
--- a/api/voip/DEPS
+++ b/api/voip/DEPS
@@ -1,5 +1,5 @@
 specific_include_rules = {
-  "voip_engine_factory.h": [
+  "voip_engine_factory\\.h": [
     "+modules/audio_device/include/audio_device.h",
     "+modules/audio_processing/include/audio_processing.h",
   ],
diff --git a/audio/DEPS b/audio/DEPS
index 7a0c7e7ce6..43d71c4854 100644
--- a/audio/DEPS
+++ b/audio/DEPS
@@ -18,10 +18,10 @@ include_rules = [
 ]
 
 specific_include_rules = {
-  "audio_send_stream.cc": [
+  "audio_send_stream\\.cc": [
     "+modules/audio_coding/codecs/cng/audio_encoder_cng.h",
   ],
-  "audio_transport_impl.h": [
+  "audio_transport_impl\\.h": [
     "+modules/audio_processing/typing_detection.h",
   ]
 }
diff --git a/call/DEPS b/call/DEPS
index 98a8a4b68d..292a8309be 100644
--- a/call/DEPS
+++ b/call/DEPS
@@ -17,61 +17,61 @@ include_rules = [
 ]
 
 specific_include_rules = {
-  "video_receive_stream\.h": [
+  "video_receive_stream\\.h": [
     "+common_video/frame_counts.h",
   ],
-  "video_send_stream\.h": [
+  "video_send_stream\\.h": [
     "+common_video",
   ],
-  "rtp_transport_controller_send_interface\.h": [
+  "rtp_transport_controller_send_interface\\.h": [
     "+common_video/frame_counts.h",
   ],
-  "call_perf_tests\.cc": [
+  "call_perf_tests\\.cc": [
     "+media/engine",
   ],
-  "simulated_network\.h": [
+  "simulated_network\\.h": [
     "+test/network/simulated_network.h",
   ],
-  "rtp_payload_params\.cc": [
+  "rtp_payload_params\\.cc": [
     "+common_video/generic_frame_descriptor",
   ],
-  "rtp_payload_params\.h": [
+  "rtp_payload_params\\.h": [
     "+common_video/generic_frame_descriptor",
   ],
-  "rtp_payload_params_unittest\.cc": [
+  "rtp_payload_params_unittest\\.cc": [
     "+common_video/generic_frame_descriptor",
   ],
-  "rtp_video_sender\.cc": [
+  "rtp_video_sender\\.cc": [
     "+common_video/frame_counts.h",
     "+common_video/generic_frame_descriptor",
   ],
-  "rtp_video_sender.h": [
+  "rtp_video_sender\\.h": [
     "+common_video/frame_counts.h",
   ],
-  "rtp_video_sender_unittest.cc": [
+  "rtp_video_sender_unittest\\.cc": [
     "+common_video/frame_counts.h",
     "+common_video/generic_frame_descriptor",
   ],
-  "payload_type\.h": [
+  "payload_type\\.h": [
     "+media/base/codec.h",
   ],
-  "payload_type_picker\.h": [
+  "payload_type_picker\\.h": [
     "+media/base/codec.h",
     "+media/base/media_constants.h",
   ],
-  "payload_type_picker\.cc": [
+  "payload_type_picker\\.cc": [
     "+media/base/codec.h",
     "+media/base/codec_comparators.h",
     "+media/base/media_constants.h",
   ],
-  "payload_type_picker_unittest\.cc": [
+  "payload_type_picker_unittest\\.cc": [
     "+media/base/codec.h",
     "+media/base/media_constants.h",
   ],
-  "call\.cc": [
+  "call\\.cc": [
     "+media/base/codec.h",
   ],
-  "fake_payload_type_suggester": [
+  "fake_payload_type_suggester\\.h": [
     "+media/base/codec.h",
   ]
 }
diff --git a/media/DEPS b/media/DEPS
index 10636cc5af..60c0fb7472 100644
--- a/media/DEPS
+++ b/media/DEPS
@@ -19,28 +19,28 @@ include_rules = [
 ]
 
 specific_include_rules = {
-  "win32devicemanager\.cc": [
+  "win32devicemanager\\.cc": [
     "+third_party/logitech/files/logitechquickcam.h",
   ],
-  ".*webrtc_video_engine\.h": [
+  ".*webrtc_video_engine\\.h": [
     "+video/config",
   ],
-  ".*webrtc_video_engine\.cc": [
+  ".*webrtc_video_engine\\.cc": [
     "+video/config",
   ],
-  ".*media_channel\.h": [
+  ".*media_channel\\.h": [
     "+video/config",
   ],
-  ".*webrtc_video_engine_unittest\.cc": [
+  ".*webrtc_video_engine_unittest\\.cc": [
     "+video/config",
   ],
-  ".*fake_webrtc_call\.cc": [
+  ".*fake_webrtc_call\\.cc": [
     "+video/config",
   ],
-  ".*fake_webrtc_call\.h": [
+  ".*fake_webrtc_call\\.h": [
     "+video/config",
   ],
-  ".*codec\.h": [
+  ".*codec\\.h": [
     "+absl/strings/str_format.h",
   ],
 }
diff --git a/modules/audio_coding/codecs/opus/DEPS b/modules/audio_coding/codecs/opus/DEPS
index c2530726ad..ea59f62649 100644
--- a/modules/audio_coding/codecs/opus/DEPS
+++ b/modules/audio_coding/codecs/opus/DEPS
@@ -1,5 +1,5 @@
 specific_include_rules = {
-  "opus_inst\.h": [
+  "opus_inst\\.h": [
     "+third_party/opus",
   ],
 }
diff --git a/modules/audio_device/DEPS b/modules/audio_device/DEPS
index b0571deb0e..380f3fd928 100644
--- a/modules/audio_device/DEPS
+++ b/modules/audio_device/DEPS
@@ -4,10 +4,10 @@ include_rules = [
 ]
 
 specific_include_rules = {
-  "ensure_initialized\.cc": [
+  "ensure_initialized\\.cc": [
     "+sdk/android",
   ],
-  "audio_device_impl\.cc": [
+  "audio_device_impl\\.cc": [
     "+sdk/objc",
     "+sdk/android",
   ],
diff --git a/modules/congestion_controller/DEPS b/modules/congestion_controller/DEPS
index 4bb4026c37..2ba1ad5a24 100644
--- a/modules/congestion_controller/DEPS
+++ b/modules/congestion_controller/DEPS
@@ -4,7 +4,7 @@ include_rules = [
   "+video",
 ]
 specific_include_rules = {
-  "goog_cc_network_control_unittest.cc": [
+  "goog_cc_network_control_unittest\\.cc": [
     "+call/video_receive_stream.h",
   ],
 }
diff --git a/modules/desktop_capture/DEPS b/modules/desktop_capture/DEPS
index b1847ad32e..74327f736e 100644
--- a/modules/desktop_capture/DEPS
+++ b/modules/desktop_capture/DEPS
@@ -5,19 +5,19 @@ include_rules = [
 ]
 
 specific_include_rules = {
-  "desktop_frame_cgimage\.h": [
+  "desktop_frame_cgimage\\.h": [
     "+sdk/objc",
   ],
-  "desktop_frame_iosurface\.h": [
+  "desktop_frame_iosurface\\.h": [
     "+sdk/objc",
   ],
-  "desktop_frame_provider\.h": [
+  "desktop_frame_provider\\.h": [
     "+sdk/objc",
   ],
-  "screen_capturer_mac\.mm": [
+  "screen_capturer_mac\\.mm": [
     "+sdk/objc",
   ],
-  "screen_capturer_sck\.mm": [
+  "screen_capturer_sck\\.mm": [
     "+absl/strings/str_format.h",
     "+sdk/objc",
   ],
diff --git a/modules/video_coding/DEPS b/modules/video_coding/DEPS
index 8794491681..e71c53f96c 100644
--- a/modules/video_coding/DEPS
+++ b/modules/video_coding/DEPS
@@ -11,10 +11,10 @@ include_rules = [
 ]
 
 specific_include_rules = {
-  "multiplex_encoder_adapter\.cc": [
+  "multiplex_encoder_adapter\\.cc": [
     "+media/base",
   ],
-  ".*test.*\.cc": [
+  ".*test.*\\.cc": [
      "+media/base",
      "+media/engine",
      "+video/config",
diff --git a/net/dcsctp/socket/DEPS b/net/dcsctp/socket/DEPS
index fa1def17eb..07eee1dd57 100644
--- a/net/dcsctp/socket/DEPS
+++ b/net/dcsctp/socket/DEPS
@@ -1,5 +1,5 @@
 specific_include_rules = {
-  "dcsctp_socket_network_test.cc": [
+  "dcsctp_socket_network_test\\.cc": [
     "+call",
     "+system_wrappers/include/clock.h",
   ]
diff --git a/pc/DEPS b/pc/DEPS
index 53ef6e45ee..aa4bbbe665 100644
--- a/pc/DEPS
+++ b/pc/DEPS
@@ -18,7 +18,7 @@ include_rules = [
 ]
 
 specific_include_rules = {
-  "rtc_stats_collector_unittest.cc": [
+  "rtc_stats_collector_unittest\\.cc": [
     "+json/reader.h",
     "+json/value.h",
   ],
diff --git a/rtc_base/DEPS b/rtc_base/DEPS
index 3861de1bb6..01756f3c99 100644
--- a/rtc_base/DEPS
+++ b/rtc_base/DEPS
@@ -6,46 +6,46 @@ include_rules = [
 ]
 
 specific_include_rules = {
-  "checks\.h": [
+  "checks\\.h": [
     "+absl/strings/has_absl_stringify.h",
   ],
-  "string_builder\.h": [
+  "string_builder\\.h": [
     "+absl/strings/has_absl_stringify.h",
   ],
-  "protobuf_utils\.h": [
+  "protobuf_utils\\.h": [
     "+third_party/protobuf",
   ],
-  "gunit\.h": [
+  "gunit\\.h": [
     "+testing/base/public/gunit.h"
   ],
-  "trace_categories\.h": [
+  "trace_categories\\.h": [
     "+third_party/perfetto",
   ],
-  "event_tracer\.cc": [
+  "event_tracer\\.cc": [
     "+third_party/perfetto",
   ],
-  "logging.h": [
+  "logging\\.h": [
     "+absl/strings/has_absl_stringify.h",
   ],
-  "trace_event\.h": [
+  "trace_event\\.h": [
     "+third_party/perfetto",
   ],
-  "openssl_adapter.cc": [
+  "openssl_adapter\\.cc": [
     "+openssl",
   ],
-  "openssl_stream_adapter.cc": [
+  "openssl_stream_adapter\\.cc": [
     "+openssl",
   ],
-  "openssl_stream_adapter.h": [
+  "openssl_stream_adapter\\.h": [
     "+openssl",
   ],
-  "openssl_session_cache\.h": [
+  "openssl_session_cache\\.h": [
     "+openssl",
   ],
-  "base64_benchmark\.cc": [
+  "base64_benchmark\\.cc": [
     "+benchmark",
   ],
-  "base64_rust\.cc": [
+  "base64_rust\\.cc": [
     "+third_party/rust/chromium_crates_io/vendor/cxx-v1/include/cxx.h",
   ],
 }
diff --git a/rtc_base/experiments/DEPS b/rtc_base/experiments/DEPS
index cb997ad8ab..855824f071 100644
--- a/rtc_base/experiments/DEPS
+++ b/rtc_base/experiments/DEPS
@@ -1,5 +1,5 @@
 specific_include_rules = {
-  ".*rate_control_settings.*": [
+  "rate_control_settings.*": [
     "+video/config",
   ],
 }
diff --git a/rtc_base/synchronization/DEPS b/rtc_base/synchronization/DEPS
index 4ed1f2444b..f3806553a7 100644
--- a/rtc_base/synchronization/DEPS
+++ b/rtc_base/synchronization/DEPS
@@ -1,11 +1,11 @@
 specific_include_rules = {
-  "mutex_abseil\.h": [
+  "mutex_abseil\\.h": [
     "+absl/synchronization"
   ],
-  ".*_benchmark\.cc": [
+  ".*_benchmark\\.cc": [
     "+benchmark",
   ],
-  ".*_unittest\.cc": [
+  ".*_unittest\\.cc": [
     "+benchmark",
   ]
 }
diff --git a/rtc_base/system/DEPS b/rtc_base/system/DEPS
index ab9449f70a..eaf2d4c2be 100644
--- a/rtc_base/system/DEPS
+++ b/rtc_base/system/DEPS
@@ -1,5 +1,5 @@
 specific_include_rules = {
-  "warn_current_thread_is_deadlocked\.cc": [
+  "warn_current_thread_is_deadlocked\\.cc": [
     "+sdk/android/native_api/stacktrace/stacktrace.h",
   ],
 }
diff --git a/rtc_tools/DEPS b/rtc_tools/DEPS
index 49c1d01f1c..2a5e29e31e 100644
--- a/rtc_tools/DEPS
+++ b/rtc_tools/DEPS
@@ -21,7 +21,7 @@ include_rules = [
 ]
 
 specific_include_rules = {
-  ".*ivf_converter\.cc": [
+  ".*ivf_converter\\.cc": [
     "+absl/debugging/failure_signal_handler.h",
     "+absl/debugging/symbolize.h",
     "+modules/video_coding/codecs/vp8/include/vp8.h",
@@ -30,19 +30,19 @@ specific_include_rules = {
     "+modules/video_coding/utility/ivf_file_writer.h",
     "+modules/video_coding/codecs/h264/include/h264.h",
   ],
-  ".*video_replay\.cc": [
+  ".*video_replay\\.cc": [
     "+absl/strings/str_format.h",
     "+absl/strings/str_split.h",
     "+modules/video_coding/include/video_error_codes.h",
     "+modules/video_coding/utility/ivf_file_writer.h",
   ],
-  ".*video_encoder\.cc": [
+  ".*video_encoder\\.cc": [
     "+modules/video_coding/codecs/av1/av1_svc_config.h",
     "+modules/video_coding/include/video_codec_interface.h",
     "+modules/video_coding/include/video_error_codes.h",
     "+modules/video_coding/svc/scalability_mode_util.h",
   ],
-  ".*encoded_image_file_writer\.(cc|h)": [
+  ".*encoded_image_file_writer\\.(cc|h)": [
     "+modules/video_coding/include/video_codec_interface.h",
     "+modules/video_coding/svc/scalability_mode_util.h",
     "+modules/video_coding/utility/ivf_file_writer.h",
diff --git a/system_wrappers/source/DEPS b/system_wrappers/source/DEPS
index ac7f5a234f..5395c9d280 100644
--- a/system_wrappers/source/DEPS
+++ b/system_wrappers/source/DEPS
@@ -1,6 +1,6 @@
 specific_include_rules = {
   # TODO(bugs.webrtc.org/10335): Remove rule when global string is removed.
-  "field_trial\.cc": [
+  "field_trial\\.cc": [
     "+experiments/registered_field_trials.h",
   ],
 }
diff --git a/test/DEPS b/test/DEPS
index 331fbadcbc..df7b9e6fea 100644
--- a/test/DEPS
+++ b/test/DEPS
@@ -28,64 +28,64 @@ include_rules = [
 ]
 
 specific_include_rules = {
-  "gmock\.h": [
+  "gmock\\.h": [
     "+testing/gmock/include/gmock",
   ],
-  "gtest\.h": [
+  "gtest\\.h": [
     "+testing/gtest/include/gtest",
   ],
-  ".*congestion_controller_feedback_fuzzer\.cc": [
+  ".*congestion_controller_feedback_fuzzer\\.cc": [
     "+modules/congestion_controller/include/receive_side_congestion_controller.h",
     "+modules/pacing/packet_router.h",
     "+modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h",
   ],
-  ".*mdns_parser_fuzzer\.cc": [
+  ".*mdns_parser_fuzzer\\.cc": [
     "+p2p/base/mdns_message.h",
   ],
-  ".*stun_parser_fuzzer\.cc": [
+  ".*stun_parser_fuzzer\\.cc": [
     "+p2p/base/stun.h",
   ],
-  ".*stun_validator_fuzzer\.cc": [
+  ".*stun_validator_fuzzer\\.cc": [
     "+p2p/base/stun.h",
   ],
-  ".*test_main_lib\.cc": [
+  ".*test_main_lib\\.cc": [
     "+third_party/perfetto",
   ],
-  ".*test_main\.cc": [
+  ".*test_main\\.cc": [
     "+absl/debugging/failure_signal_handler.h",
     "+absl/debugging/symbolize.h",
   ],
-  ".*test_peer\.(h|cc)": [
+  ".*test_peer\\.(h|cc)": [
     "+pc",
     "+p2p",
   ],
-  ".*test_peer_factory\.(h|cc)": [
+  ".*test_peer_factory\\.(h|cc)": [
     "+pc",
     "+p2p",
   ],
-  ".*network_emulation_pc_unittest\.cc": [
+  ".*network_emulation_pc_unittest\\.cc": [
     "+p2p/base/port_allocator.h",
     "+pc/peer_connection_wrapper.h",
     "+pc/test/mock_peer_connection_observers.h",
   ],
-  ".*peer_connection_quality_test\.(h|cc)": [
+  ".*peer_connection_quality_test\\.(h|cc)": [
     "+pc",
   ],
-  ".*sdp_changer\.(h|cc)": [
+  ".*sdp_changer\\.(h|cc)": [
     "+pc",
     "+p2p",
   ],
-  ".*test_video_capturer_video_track_source.h": [
+  ".*test_video_capturer_video_track_source\\.h": [
     "+pc",
   ],
-  "benchmark_main\.cc": [
+  "benchmark_main\\.cc": [
     "+benchmark",
   ],
-  "emulated_turn_server\.(h|cc)": [
+  "emulated_turn_server\\.(h|cc)": [
      "+p2p/test/turn_server.h",
      "+p2p/base/port_interface.h",
   ],
-  "wait_until\.cc": [
+  "wait_until\\.cc": [
     "+absl/functional/overload.h",
   ],
 }

From 125c674e170eb878bbe00c985fdc0bb36f42e5e7 Mon Sep 17 00:00:00 2001
From: Jonas Oreland <jonaso@webrtc.org>
Date: Tue, 17 Feb 2026 08:53:30 -0800
Subject: [PATCH 079/472] Add jonaso@ as OWNER in rtc_base

Currently lots of stuff only used by p2p/ is located in rtc_base/
(e.g. DTLS). I think such stuff should be moved into p2p.

Bug: None
Change-Id: Ia0cc1125ac1e4f6c7b294022d5d33d7fa89819d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/450020
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46932}
---
 rtc_base/OWNERS | 1 +
 1 file changed, 1 insertion(+)

diff --git a/rtc_base/OWNERS b/rtc_base/OWNERS
index 3a420a088e..680815b65f 100644
--- a/rtc_base/OWNERS
+++ b/rtc_base/OWNERS
@@ -2,6 +2,7 @@ hta@webrtc.org
 mflodman@webrtc.org
 tommi@webrtc.org
 mbonadei@webrtc.org
+jonaso@webrtc.org
 
 per-file rate_statistics*=sprang@webrtc.org
 per-file rate_statistics*=stefan@webrtc.org

From ee527ba4c9e45d5c9d9beb8d08fba1b491022b9c Mon Sep 17 00:00:00 2001
From: Nfrederiksen <nicholas.frederiksen@eyevinn.se>
Date: Wed, 11 Feb 2026 10:58:40 +0100
Subject: [PATCH 080/472] video: add EncoderInfo CPU overuse flag and aggregate
 in simulcast

enable_cpu_overuse_detection=true by default, even when all sub-encoders
disabled it.
sub-encoder's value, so CPU overuse detection only runs when at least
one encoder opts in.

Before: In multi-encoder simulcast, SimulcastEncoderAdapter returned
After: The adapter initializes the flag to false and ORs each
Bug: webrtc:42225459
Change-Id: Ib5ae02408854507364bb2c5041e3aea61c3b0ad8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/443381
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46933}
---
 api/video_codecs/video_encoder.cc             |   5 +-
 api/video_codecs/video_encoder.h              |   3 +
 media/engine/simulcast_encoder_adapter.cc     |   7 ++
 .../simulcast_encoder_adapter_unittest.cc     |  28 +++++
 .../video_stream_encoder_resource_manager.cc  |   9 ++
 video/video_stream_encoder_unittest.cc        | 116 ++++++++++++++++++
 6 files changed, 167 insertions(+), 1 deletion(-)

diff --git a/api/video_codecs/video_encoder.cc b/api/video_codecs/video_encoder.cc
index 0c25a2f6d7..e4a2177079 100644
--- a/api/video_codecs/video_encoder.cc
+++ b/api/video_codecs/video_encoder.cc
@@ -102,6 +102,7 @@ VideoEncoder::EncoderInfo::EncoderInfo()
       implementation_name("unknown"),
       has_trusted_rate_controller(false),
       is_hardware_accelerated(true),
+      enable_cpu_overuse_detection(true),
       fps_allocation{absl::InlinedVector<uint8_t, kMaxTemporalStreams>(
           1,
           kMaxFramerateFraction)},
@@ -133,6 +134,7 @@ std::string VideoEncoder::EncoderInfo::ToString() const {
          ", has_trusted_rate_controller = "
       << has_trusted_rate_controller
       << ", is_hardware_accelerated = " << is_hardware_accelerated
+      << ", enable_cpu_overuse_detection = " << enable_cpu_overuse_detection
       << ", fps_allocation = [";
   size_t num_spatial_layer_with_fps_allocation = 0;
   for (size_t i = 0; i < kMaxSpatialLayers; ++i) {
@@ -217,7 +219,8 @@ bool VideoEncoder::EncoderInfo::operator==(const EncoderInfo& rhs) const {
   if (supports_native_handle != rhs.supports_native_handle ||
       implementation_name != rhs.implementation_name ||
       has_trusted_rate_controller != rhs.has_trusted_rate_controller ||
-      is_hardware_accelerated != rhs.is_hardware_accelerated) {
+      is_hardware_accelerated != rhs.is_hardware_accelerated ||
+      enable_cpu_overuse_detection != rhs.enable_cpu_overuse_detection) {
     return false;
   }
 
diff --git a/api/video_codecs/video_encoder.h b/api/video_codecs/video_encoder.h
index d8f4b35d5f..1110b44e51 100644
--- a/api/video_codecs/video_encoder.h
+++ b/api/video_codecs/video_encoder.h
@@ -230,6 +230,9 @@ class RTC_EXPORT VideoEncoder {
     // thresholds will be used in CPU adaptation.
     bool is_hardware_accelerated;
 
+    // If this field is true, this encoder opts into CPU overuse detection.
+    bool enable_cpu_overuse_detection;
+
     // For each spatial layer (simulcast stream or SVC layer), represented as an
     // element in `fps_allocation` a vector indicates how many temporal layers
     // the encoder is using for that spatial layer.
diff --git a/media/engine/simulcast_encoder_adapter.cc b/media/engine/simulcast_encoder_adapter.cc
index f290a1c266..3e10c92aec 100644
--- a/media/engine/simulcast_encoder_adapter.cc
+++ b/media/engine/simulcast_encoder_adapter.cc
@@ -1142,6 +1142,10 @@ VideoEncoder::EncoderInfo SimulcastEncoderAdapter::GetEncoderInfo() const {
     return encoder_info;
   }
 
+  // Default is to not enable CPU overuse detection. Aggregate across
+  // sub-encoders so we enable it if any encoder opts in.
+  encoder_info.enable_cpu_overuse_detection = false;
+
   encoder_info.scaling_settings = VideoEncoder::ScalingSettings::kOff;
   std::vector<std::string> encoder_names;
 
@@ -1191,6 +1195,9 @@ VideoEncoder::EncoderInfo SimulcastEncoderAdapter::GetEncoderInfo() const {
           encoder_info.is_qp_trusted.value_or(true) &&
           encoder_impl_info.is_qp_trusted.value_or(true);
     }
+    // If any encoder wants CPU overuse detection, enable it for all of them
+    encoder_info.enable_cpu_overuse_detection |=
+        encoder_impl_info.enable_cpu_overuse_detection;
     encoder_info.fps_allocation[i] = encoder_impl_info.fps_allocation[0];
     encoder_info.requested_resolution_alignment =
         std::lcm(encoder_info.requested_resolution_alignment,
diff --git a/media/engine/simulcast_encoder_adapter_unittest.cc b/media/engine/simulcast_encoder_adapter_unittest.cc
index 68c0b820c2..ca0bb2bded 100644
--- a/media/engine/simulcast_encoder_adapter_unittest.cc
+++ b/media/engine/simulcast_encoder_adapter_unittest.cc
@@ -288,6 +288,7 @@ class MockVideoEncoder : public VideoEncoder {
         apply_alignment_to_all_simulcast_layers_;
     info.has_trusted_rate_controller = has_trusted_rate_controller_;
     info.is_hardware_accelerated = is_hardware_accelerated_;
+    info.enable_cpu_overuse_detection = enable_cpu_overuse_detection_;
     info.fps_allocation[0] = fps_allocation_;
     info.supports_simulcast = supports_simulcast_;
     info.is_qp_trusted = is_qp_trusted_;
@@ -355,6 +356,10 @@ class MockVideoEncoder : public VideoEncoder {
     is_hardware_accelerated_ = is_hardware_accelerated;
   }
 
+  void set_enable_cpu_overuse_detection(bool enable) {
+    enable_cpu_overuse_detection_ = enable;
+  }
+
   void set_fps_allocation(const FramerateFractions& fps_allocation) {
     fps_allocation_ = fps_allocation;
   }
@@ -391,6 +396,7 @@ class MockVideoEncoder : public VideoEncoder {
   bool apply_alignment_to_all_simulcast_layers_ = false;
   bool has_trusted_rate_controller_ = false;
   bool is_hardware_accelerated_ = false;
+  bool enable_cpu_overuse_detection_ = true;
   int32_t init_encode_return_value_ = 0;
   std::optional<int32_t> fallback_from_simulcast_;
   VideoEncoder::RateControlParameters last_set_rates_;
@@ -2026,6 +2032,28 @@ TEST_F(TestSimulcastEncoderAdapterFake, ReportsHardwareAccelerated) {
   EXPECT_TRUE(adapter_->GetEncoderInfo().is_hardware_accelerated);
 }
 
+TEST_F(TestSimulcastEncoderAdapterFake, ReportsEnableCpuOveruseDetection) {
+  SimulcastTestFixtureImpl::DefaultSettings(
+      &codec_, static_cast<const int*>(kTestTemporalLayerProfile),
+      kVideoCodecVP8);
+  codec_.numberOfSimulcastStreams = 3;
+  adapter_->RegisterEncodeCompleteCallback(this);
+  EXPECT_EQ(0, adapter_->InitEncode(&codec_, kSettings));
+  ASSERT_EQ(3u, helper_->factory()->encoders().size());
+
+  // All encoders opt out, so simulcast adapter should report false.
+  for (MockVideoEncoder* encoder : helper_->factory()->encoders()) {
+    encoder->set_enable_cpu_overuse_detection(false);
+  }
+  EXPECT_EQ(0, adapter_->InitEncode(&codec_, kSettings));
+  EXPECT_FALSE(adapter_->GetEncoderInfo().enable_cpu_overuse_detection);
+
+  // One encoder opts in, so simulcast adapter should report true.
+  helper_->factory()->encoders()[1]->set_enable_cpu_overuse_detection(true);
+  EXPECT_EQ(0, adapter_->InitEncode(&codec_, kSettings));
+  EXPECT_TRUE(adapter_->GetEncoderInfo().enable_cpu_overuse_detection);
+}
+
 TEST_F(TestSimulcastEncoderAdapterFake,
        ReportsLeastCommonMultipleOfRequestedResolutionAlignments) {
   SimulcastTestFixtureImpl::DefaultSettings(
diff --git a/video/adaptation/video_stream_encoder_resource_manager.cc b/video/adaptation/video_stream_encoder_resource_manager.cc
index b79a186012..ea817269c8 100644
--- a/video/adaptation/video_stream_encoder_resource_manager.cc
+++ b/video/adaptation/video_stream_encoder_resource_manager.cc
@@ -341,6 +341,15 @@ VideoStreamEncoderResourceManager::degradation_preference() const {
 void VideoStreamEncoderResourceManager::ConfigureEncodeUsageResource() {
   RTC_DCHECK_RUN_ON(encoder_queue_);
   RTC_DCHECK(encoder_settings_.has_value());
+  if (!encoder_settings_->encoder_info().enable_cpu_overuse_detection) {
+    if (encode_usage_resource_->is_started()) {
+      encode_usage_resource_->StopCheckForOveruse();
+    }
+    if (resources_.find(encode_usage_resource_) != resources_.end()) {
+      RemoveResource(encode_usage_resource_);
+    }
+    return;
+  }
   if (encode_usage_resource_->is_started()) {
     encode_usage_resource_->StopCheckForOveruse();
   } else {
diff --git a/video/video_stream_encoder_unittest.cc b/video/video_stream_encoder_unittest.cc
index dae6673a9f..adfcbb5c26 100644
--- a/video/video_stream_encoder_unittest.cc
+++ b/video/video_stream_encoder_unittest.cc
@@ -1148,6 +1148,7 @@ class VideoStreamEncoderTest : public ::testing::Test {
       info.apply_alignment_to_all_simulcast_layers =
           apply_alignment_to_all_simulcast_layers_;
       info.preferred_pixel_formats = preferred_pixel_formats_;
+      info.enable_cpu_overuse_detection = enable_cpu_overuse_detection_;
       if (is_qp_trusted_.has_value()) {
         info.is_qp_trusted = is_qp_trusted_;
       }
@@ -1194,6 +1195,11 @@ class VideoStreamEncoderTest : public ::testing::Test {
       is_hardware_accelerated_ = is_hardware_accelerated;
     }
 
+    void SetEnableCpuOveruseDetection(bool enable) {
+      MutexLock lock(&local_mutex_);
+      enable_cpu_overuse_detection_ = enable;
+    }
+
     void SetTemporalLayersSupported(size_t spatial_idx, bool supported) {
       RTC_DCHECK_LT(spatial_idx, kMaxSpatialLayers);
       MutexLock lock(&local_mutex_);
@@ -1440,6 +1446,7 @@ class VideoStreamEncoderTest : public ::testing::Test {
     bool apply_alignment_to_all_simulcast_layers_ RTC_GUARDED_BY(local_mutex_) =
         false;
     bool is_hardware_accelerated_ RTC_GUARDED_BY(local_mutex_) = false;
+    bool enable_cpu_overuse_detection_ RTC_GUARDED_BY(local_mutex_) = true;
     scoped_refptr<EncodedImageBufferInterface> encoded_image_data_
         RTC_GUARDED_BY(local_mutex_);
     std::unique_ptr<Vp8FrameBufferController> frame_buffer_controller_
@@ -5886,6 +5893,115 @@ TEST_F(VideoStreamEncoderTest, VerifyBitrateAllocationForTwoStreams) {
   video_stream_encoder_->Stop();
 }
 
+TEST_F(VideoStreamEncoderTest,
+       EncodeUsageResourceDisabledWhenCpuOveruseDetectionDisabled) {
+  const int kFrameWidth = 1280;
+  const int kFrameHeight = 720;
+
+  fake_encoder_.SetEnableCpuOveruseDetection(false);
+
+  VideoEncoderConfig video_encoder_config;
+  test::FillEncoderConfiguration(kVideoCodecVP8, 1, &video_encoder_config);
+  video_encoder_config.max_bitrate_bps = kTargetBitrate.bps();
+  ConfigureEncoder(std::move(video_encoder_config));
+
+  video_source_.IncomingCapturedFrame(
+      CreateFrame(1, kFrameWidth, kFrameHeight));
+  video_stream_encoder_->WaitUntilTaskQueueIsIdle();
+
+  EXPECT_EQ(
+      video_stream_encoder_->overuse_detector_proxy_->GetLastTargetFramerate(),
+      -1);
+
+  video_stream_encoder_->Stop();
+}
+
+TEST_F(VideoStreamEncoderTest,
+       EncodeUsageResourceStoppedWhenCpuOveruseDetectionDisabled) {
+  const int kFrameWidth = 1280;
+  const int kFrameHeight = 720;
+
+  video_stream_encoder_->OnBitrateUpdatedAndWaitForManagedResources(
+      kTargetBitrate, kTargetBitrate, 0, 0, 0);
+
+  video_source_.IncomingCapturedFrame(
+      CreateFrame(1, kFrameWidth, kFrameHeight));
+  video_stream_encoder_->WaitUntilTaskQueueIsIdle();
+
+  const int last_framerate =
+      video_stream_encoder_->overuse_detector_proxy_->GetLastTargetFramerate();
+  EXPECT_NE(last_framerate, -1);
+
+  fake_encoder_.SetEnableCpuOveruseDetection(false);
+  VideoEncoderConfig video_encoder_config = video_encoder_config_.Copy();
+  video_stream_encoder_->ConfigureEncoder(std::move(video_encoder_config),
+                                          kMaxPayloadLength);
+  video_stream_encoder_->WaitUntilTaskQueueIsIdle();
+
+  video_stream_encoder_->overuse_detector_proxy_->framerate_updated_event()
+      ->Reset();
+  video_source_.IncomingCapturedFrame(
+      CreateFrame(2, kFrameWidth, kFrameHeight));
+  video_stream_encoder_->WaitUntilTaskQueueIsIdle();
+
+  EXPECT_FALSE(
+      video_stream_encoder_->overuse_detector_proxy_->framerate_updated_event()
+          ->Wait(TimeDelta::Millis(10)));
+
+  video_stream_encoder_->Stop();
+}
+
+TEST_F(VideoStreamEncoderTest,
+       EncodeUsageResourceRestartsWhenCpuOveruseDetectionReEnabled) {
+  const int kFrameWidth = 1280;
+  const int kFrameHeight = 720;
+
+  video_stream_encoder_->OnBitrateUpdatedAndWaitForManagedResources(
+      kTargetBitrate, kTargetBitrate, 0, 0, 0);
+
+  video_source_.IncomingCapturedFrame(
+      CreateFrame(1, kFrameWidth, kFrameHeight));
+  video_stream_encoder_->WaitUntilTaskQueueIsIdle();
+
+  EXPECT_NE(
+      video_stream_encoder_->overuse_detector_proxy_->GetLastTargetFramerate(),
+      -1);
+
+  fake_encoder_.SetEnableCpuOveruseDetection(false);
+  VideoEncoderConfig video_encoder_config = video_encoder_config_.Copy();
+  video_stream_encoder_->ConfigureEncoder(std::move(video_encoder_config),
+                                          kMaxPayloadLength);
+  video_stream_encoder_->WaitUntilTaskQueueIsIdle();
+
+  video_stream_encoder_->overuse_detector_proxy_->framerate_updated_event()
+      ->Reset();
+  video_source_.IncomingCapturedFrame(
+      CreateFrame(2, kFrameWidth, kFrameHeight));
+  video_stream_encoder_->WaitUntilTaskQueueIsIdle();
+
+  EXPECT_FALSE(
+      video_stream_encoder_->overuse_detector_proxy_->framerate_updated_event()
+          ->Wait(TimeDelta::Millis(10)));
+
+  fake_encoder_.SetEnableCpuOveruseDetection(true);
+  video_encoder_config = video_encoder_config_.Copy();
+  video_stream_encoder_->ConfigureEncoder(std::move(video_encoder_config),
+                                          kMaxPayloadLength);
+  video_stream_encoder_->WaitUntilTaskQueueIsIdle();
+
+  video_stream_encoder_->overuse_detector_proxy_->framerate_updated_event()
+      ->Reset();
+  video_source_.IncomingCapturedFrame(
+      CreateFrame(3, kFrameWidth, kFrameHeight));
+  video_stream_encoder_->WaitUntilTaskQueueIsIdle();
+
+  EXPECT_TRUE(
+      video_stream_encoder_->overuse_detector_proxy_->framerate_updated_event()
+          ->Wait(TimeDelta::Millis(10)));
+
+  video_stream_encoder_->Stop();
+}
+
 TEST_F(VideoStreamEncoderTest, OveruseDetectorUpdatedOnReconfigureAndAdaption) {
   const int kFrameWidth = 1280;
   const int kFrameHeight = 720;

From e7872b6cda87d18eddc97acf2a6417ad2aab9fed Mon Sep 17 00:00:00 2001
From: Jeremy Leconte <jleconte@webrtc.org>
Date: Tue, 17 Feb 2026 20:25:18 +0100
Subject: [PATCH 081/472] Fix checkdeps and add a presubmit check.

ERROR in webrtc/src/api/test/video_quality_test_fixture.h
  Illegal include: "video/config/video_encoder_config.h"
    Because of "-video" from api's include_rules.

ERROR in webrtc/src/net/dcsctp/packet/crc32c.cc
  Illegal include: "third_party/crc32c/src/include/crc32c/crc32c.h"
    Because of no rule applying.

Change-Id: I0e546c8fea0082b7fb936e6c5e130f63173380df
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/450060
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#46934}
---
 PRESUBMIT.py           | 18 ++++++++++++++++++
 api/DEPS               |  4 ++++
 net/dcsctp/packet/DEPS |  3 +++
 3 files changed, 25 insertions(+)
 create mode 100644 net/dcsctp/packet/DEPS

diff --git a/PRESUBMIT.py b/PRESUBMIT.py
index fa6888cda9..cfb9220c3d 100755
--- a/PRESUBMIT.py
+++ b/PRESUBMIT.py
@@ -662,6 +662,23 @@ def CheckGnGen(input_api, output_api):
     return []
 
 
+def CheckDeps(input_api, output_api):
+    """Runs checkdeps """
+    repo_root = input_api.change.RepositoryRoot()
+    checkdeps_path = input_api.os_path.join(repo_root, 'buildtools',
+                                            'checkdeps')
+    with _AddToPath(checkdeps_path):
+        import checkdeps
+
+    deps_checker = checkdeps.DepsChecker(input_api.PresubmitLocalPath())
+    deps_checker.CheckDirectory(input_api.PresubmitLocalPath())
+    results = []
+    if deps_checker.results_formatter.GetResults():
+        results.append(
+            output_api.PresubmitError('\n'.join(
+                deps_checker.results_formatter.GetResults())))
+    return results
+
 def CheckUnwantedDependencies(input_api, output_api, source_file_filter):
     """Runs checkdeps on #include statements added in this
   change. Breaking - rules is an error, breaking ! rules is a
@@ -1000,6 +1017,7 @@ def CommonChecks(input_api, output_api):
     results.extend(
         input_api.canned_checks.CheckPatchFormatted(input_api, output_api))
     results.extend(CheckNativeApiHeaderChanges(input_api, output_api))
+    results.extend(CheckDeps(input_api, output_api))
     results.extend(
         CheckNoIOStreamInHeaders(input_api,
                                  output_api,
diff --git a/api/DEPS b/api/DEPS
index 98f9f35cbe..1656c3f916 100644
--- a/api/DEPS
+++ b/api/DEPS
@@ -264,6 +264,10 @@ specific_include_rules = {
     "+rtc_base/numerics",
   ],
 
+  "video_quality_test_fixture\\.h": [
+    "+video/config/video_encoder_config.h",
+  ],
+
   "simple_encoder_wrapper\\.h": [
     "+common_video",
     "+modules",
diff --git a/net/dcsctp/packet/DEPS b/net/dcsctp/packet/DEPS
new file mode 100644
index 0000000000..206b405b82
--- /dev/null
+++ b/net/dcsctp/packet/DEPS
@@ -0,0 +1,3 @@
+include_rules = [
+    "+third_party/crc32c",
+]

From a4dd38a556f11e9157dd1ac360506d6775253cc5 Mon Sep 17 00:00:00 2001
From: Jonas Oreland <jonaso@webrtc.org>
Date: Tue, 17 Feb 2026 07:42:16 -0800
Subject: [PATCH 082/472] Add a CompleteCallback to DtlsStunPiggybackController

This give the controller greater control over when it
shall be uninstalled. This will allow for the controller
to be active even if "local dtls" is COMPLETE.

Note: There "should" be a completecallback when
we detect that peer does not support the feature.
But adding that causes a bunch of test failures
so I instead added a TODO, one more :(

This cl is a NOP but allows for future modifications
(read for fippo) to become smaller.

This cl/ also modifies dtls_transport.cc so that
only BoringSSL packets injected into the DtlsStunPiggybackController.
In the current version this doesn't really matter,
but if we in future version delay "completing" of DtlsInStun
it means that it will not be polluted with "user" packets.

BUG=webrtc:367395350

Change-Id: I813539a8c3d883a3fa0d8d363954c3c6872ea137
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449561
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46935}
---
 p2p/dtls/dtls_ice_integration_test.cc         |  2 +
 p2p/dtls/dtls_stun_piggyback_controller.cc    | 25 ++++++++++--
 p2p/dtls/dtls_stun_piggyback_controller.h     | 12 ++++--
 ...dtls_stun_piggyback_controller_unittest.cc | 24 +++++++++--
 p2p/dtls/dtls_transport.cc                    | 40 ++++++++++++++-----
 p2p/dtls/dtls_transport.h                     |  2 +
 6 files changed, 85 insertions(+), 20 deletions(-)

diff --git a/p2p/dtls/dtls_ice_integration_test.cc b/p2p/dtls/dtls_ice_integration_test.cc
index 0ff6f24c7d..20b91f400e 100644
--- a/p2p/dtls/dtls_ice_integration_test.cc
+++ b/p2p/dtls/dtls_ice_integration_test.cc
@@ -461,6 +461,8 @@ TEST_P(DtlsIceIntegrationPerformanceTest, ConnectTime) {
     }
   }
 
+  RTC_LOG(LS_INFO) << "START: " << GetParam();
+
   int iter = 50;
   // Use a fixed seed to get consistent behavior.
   Random rand(/* seed= */ 77);
diff --git a/p2p/dtls/dtls_stun_piggyback_controller.cc b/p2p/dtls/dtls_stun_piggyback_controller.cc
index 646a866026..2abdcbc7f1 100644
--- a/p2p/dtls/dtls_stun_piggyback_controller.cc
+++ b/p2p/dtls/dtls_stun_piggyback_controller.cc
@@ -30,10 +30,16 @@
 namespace webrtc {
 
 DtlsStunPiggybackController::DtlsStunPiggybackController(
-    absl::AnyInvocable<void(ArrayView<const uint8_t>)> dtls_data_callback)
-    : dtls_data_callback_(std::move(dtls_data_callback)) {}
+    absl::AnyInvocable<void(ArrayView<const uint8_t>)> dtls_data_callback,
+    absl::AnyInvocable<void()> complete_callback)
+    : dtls_data_callback_(std::move(dtls_data_callback)),
+      complete_callback_(std::move(complete_callback)) {}
 
-DtlsStunPiggybackController::~DtlsStunPiggybackController() {}
+DtlsStunPiggybackController::~DtlsStunPiggybackController() {
+  complete_callback_ = [&]() {
+    RTC_DCHECK(false) << "Calling CompleteCallback after destructor!";
+  };
+}
 
 void DtlsStunPiggybackController::SetDtlsHandshakeComplete(bool is_dtls_client,
                                                            bool is_dtls13) {
@@ -66,6 +72,7 @@ void DtlsStunPiggybackController::SetDtlsFailed() {
         << "DTLS-STUN piggybacking DTLS failed during negotiation.";
   }
   state_ = State::OFF;
+  CallCompleteCallback();
 }
 
 void DtlsStunPiggybackController::CapturePacket(ArrayView<const uint8_t> data) {
@@ -158,6 +165,9 @@ void DtlsStunPiggybackController::ReportDataPiggybacked(
   if (state_ == State::TENTATIVE && !data.has_value() && !acks.has_value()) {
     RTC_LOG(LS_INFO) << "DTLS-STUN piggybacking not supported by peer.";
     state_ = State::OFF;
+    // TODO jonaso, webrtc:367395350: We should call CallCompleteCallback here
+    // but this causes a slew of failed tests. Investigate why!
+    // CallCompleteCallback();
     return;
   }
 
@@ -168,6 +178,7 @@ void DtlsStunPiggybackController::ReportDataPiggybacked(
     state_ = State::COMPLETE;
     pending_packets_.clear();
     handshake_messages_received_.clear();
+    CallCompleteCallback();
     return;
   }
 
@@ -206,6 +217,7 @@ void DtlsStunPiggybackController::ReportDataPiggybacked(
     state_ = State::COMPLETE;
     pending_packets_.clear();
     handshake_messages_received_.clear();
+    CallCompleteCallback();
     return;
   }
 
@@ -250,4 +262,11 @@ void DtlsStunPiggybackController::ReportDtlsPacket(
   }
 }
 
+void DtlsStunPiggybackController::CallCompleteCallback() {
+  complete_callback_();
+  complete_callback_ = [&]() {
+    RTC_DCHECK(false) << "CompleteCallback called twice!";
+  };
+}
+
 }  // namespace webrtc
diff --git a/p2p/dtls/dtls_stun_piggyback_controller.h b/p2p/dtls/dtls_stun_piggyback_controller.h
index 6c11a377cf..3741bd985e 100644
--- a/p2p/dtls/dtls_stun_piggyback_controller.h
+++ b/p2p/dtls/dtls_stun_piggyback_controller.h
@@ -36,7 +36,9 @@ class DtlsStunPiggybackController {
   // dtls_data_callback will be called with any DTLS packets received
   // piggybacked.
   DtlsStunPiggybackController(
-      absl::AnyInvocable<void(ArrayView<const uint8_t>)> dtls_data_callback);
+      absl::AnyInvocable<void(ArrayView<const uint8_t>)> dtls_data_callback,
+      absl::AnyInvocable<void()> complete_callback);
+
   ~DtlsStunPiggybackController();
 
   enum class State {
@@ -60,8 +62,10 @@ class DtlsStunPiggybackController {
     return state_;
   }
 
-  // Called by DtlsTransport when the handshake is complete.
+  // Called by DtlsTransport when the handshake is complete per our local DTLS
+  // agent.
   void SetDtlsHandshakeComplete(bool is_dtls_client, bool is_dtls13);
+
   // Called by DtlsTransport when DTLS failed.
   void SetDtlsFailed();
 
@@ -98,7 +102,7 @@ class DtlsStunPiggybackController {
   bool writing_packets_ RTC_GUARDED_BY(sequence_checker_) = false;
   PacketStash pending_packets_ RTC_GUARDED_BY(sequence_checker_);
   absl::AnyInvocable<void(ArrayView<const uint8_t>)> dtls_data_callback_;
-  absl::AnyInvocable<void()> disable_piggybacking_callback_;
+  absl::AnyInvocable<void()> complete_callback_;
 
   std::vector<uint32_t> handshake_messages_received_
       RTC_GUARDED_BY(sequence_checker_);
@@ -106,6 +110,8 @@ class DtlsStunPiggybackController {
   // Count of embedded data attributes received.
   int data_recv_count_ = 0;
 
+  void CallCompleteCallback();
+
   // In practice this will be the network thread.
   RTC_NO_UNIQUE_ADDRESS SequenceChecker sequence_checker_;
 };
diff --git a/p2p/dtls/dtls_stun_piggyback_controller_unittest.cc b/p2p/dtls/dtls_stun_piggyback_controller_unittest.cc
index 49894be85c..7f12b65ecf 100644
--- a/p2p/dtls/dtls_stun_piggyback_controller_unittest.cc
+++ b/p2p/dtls/dtls_stun_piggyback_controller_unittest.cc
@@ -107,10 +107,11 @@ class DtlsStunPiggybackControllerTest : public ::testing::Test {
  protected:
   DtlsStunPiggybackControllerTest()
       : client_(
-            [this](ArrayView<const uint8_t> data) { ClientPacketSink(data); }),
-        server_([this](ArrayView<const uint8_t> data) {
-          ServerPacketSink(data);
-        }) {}
+            [this](ArrayView<const uint8_t> data) { ClientPacketSink(data); },
+            [this]() { ClientCompleteCallback(); }),
+        server_(
+            [this](ArrayView<const uint8_t> data) { ServerPacketSink(data); },
+            [this]() { ServerCompleteCallback(); }) {}
 
   // Send from client to server embedded in STUN.
   void SendClientToServerEmbedded(const std::vector<uint8_t>& packet,
@@ -193,6 +194,9 @@ class DtlsStunPiggybackControllerTest : public ::testing::Test {
   MOCK_METHOD(void, ClientPacketSink, (ArrayView<const uint8_t>));
   MOCK_METHOD(void, ServerPacketSink, (ArrayView<const uint8_t>));
 
+  MOCK_METHOD(void, ClientCompleteCallback, ());
+  MOCK_METHOD(void, ServerCompleteCallback, ());
+
  private:
   void MaybeSetHandshakeComplete(std::vector<uint8_t> packet) {
     // Note: this assumes DTLS 1.2
@@ -220,7 +224,9 @@ TEST_F(DtlsStunPiggybackControllerTest, BasicHandshake) {
   EXPECT_EQ(client_.state(), State::PENDING);
 
   // Post-handshake ACK
+  EXPECT_CALL(*this, ClientCompleteCallback);
   SendServerToClientEmbedded(empty, STUN_BINDING_REQUEST);
+  EXPECT_CALL(*this, ServerCompleteCallback);
   SendClientToServerEmbedded(empty, STUN_BINDING_RESPONSE);
   EXPECT_EQ(server_.state(), State::COMPLETE);
   EXPECT_EQ(client_.state(), State::COMPLETE);
@@ -242,11 +248,13 @@ TEST_F(DtlsStunPiggybackControllerTest, FirstClientPacketLost) {
 
   // Flight 4
   SendServerToClientEmbedded(dtls_flight4, STUN_BINDING_REQUEST);
+  EXPECT_CALL(*this, ServerCompleteCallback);
   SendClientToServerEmbedded(empty, STUN_BINDING_RESPONSE);
   EXPECT_EQ(server_.state(), State::COMPLETE);
   EXPECT_EQ(client_.state(), State::PENDING);
 
   // Post-handshake ACK
+  EXPECT_CALL(*this, ClientCompleteCallback);
   SendServerToClientEmbedded(empty, STUN_BINDING_REQUEST);
   EXPECT_EQ(client_.state(), State::COMPLETE);
 }
@@ -297,7 +305,9 @@ TEST_F(DtlsStunPiggybackControllerTest, SomeRequestsDoNotGoThrough) {
   EXPECT_EQ(client_.state(), State::PENDING);
 
   // Post-handshake ACK
+  EXPECT_CALL(*this, ServerCompleteCallback);
   SendClientToServerEmbedded(empty, STUN_BINDING_REQUEST);
+  EXPECT_CALL(*this, ClientCompleteCallback);
   SendServerToClientEmbedded(empty, STUN_BINDING_RESPONSE);
   EXPECT_EQ(server_.state(), State::COMPLETE);
   EXPECT_EQ(client_.state(), State::COMPLETE);
@@ -317,7 +327,9 @@ TEST_F(DtlsStunPiggybackControllerTest, LossOnPostHandshakeAck) {
   EXPECT_EQ(client_.state(), State::PENDING);
 
   // Post-handshake ACK. Client to server gets lost
+  EXPECT_CALL(*this, ClientCompleteCallback);
   SendServerToClientEmbedded(empty, STUN_BINDING_REQUEST);
+  EXPECT_CALL(*this, ServerCompleteCallback);
   SendClientToServerEmbedded(empty, STUN_BINDING_RESPONSE);
   EXPECT_EQ(server_.state(), State::COMPLETE);
   EXPECT_EQ(client_.state(), State::COMPLETE);
@@ -364,7 +376,9 @@ TEST_F(DtlsStunPiggybackControllerTest, BasicHandshakeAckData) {
               }));
 
   // Post-handshake ACK
+  EXPECT_CALL(*this, ClientCompleteCallback);
   SendServerToClientEmbedded(empty, STUN_BINDING_REQUEST);
+  EXPECT_CALL(*this, ServerCompleteCallback);
   SendClientToServerEmbedded(empty, STUN_BINDING_RESPONSE);
   EXPECT_EQ(server_.state(), State::COMPLETE);
   EXPECT_EQ(client_.state(), State::COMPLETE);
@@ -401,7 +415,9 @@ TEST_F(DtlsStunPiggybackControllerTest, UnwrappedHandshakeAckData) {
               }));
 
   // Post-handshake ACK
+  EXPECT_CALL(*this, ClientCompleteCallback);
   SendServerToClientEmbedded(empty, STUN_BINDING_REQUEST);
+  EXPECT_CALL(*this, ServerCompleteCallback);
   SendClientToServerEmbedded(empty, STUN_BINDING_RESPONSE);
   EXPECT_EQ(server_.state(), State::COMPLETE);
   EXPECT_EQ(client_.state(), State::COMPLETE);
diff --git a/p2p/dtls/dtls_transport.cc b/p2p/dtls/dtls_transport.cc
index 9d4b91caa8..2fc2ab28a7 100644
--- a/p2p/dtls/dtls_transport.cc
+++ b/p2p/dtls/dtls_transport.cc
@@ -182,17 +182,18 @@ StreamResult StreamInterfaceChannel::Write(ArrayView<const uint8_t> data,
                                            int& /* error */) {
   RTC_DCHECK_RUN_ON(&callback_sequence_);
 
-  // If we use DTLS-in-STUN, DTLS packets will be sent as part of STUN
-  // packets, they are captured by the controller.
-  if (dtls_stun_piggyback_controller_) {
-    dtls_stun_piggyback_controller_->CapturePacket(data);
-  }
-
   AsyncSocketPacketOptions packet_options;
   if (next_packet_options_) {
     packet_options = std::move(*next_packet_options_);
     next_packet_options_.reset();
+  } else if (dtls_stun_piggyback_controller_) {
+    // Note: packets with `next_packet_options_` are user packets.
+    // Packets without `next_packet_options_` are generated by BoringSSL.
+    // The DtlsStunPiggybackController captures BoringSSL packets.
+    RTC_DCHECK(IsDtlsPacket(data));
+    dtls_stun_piggyback_controller_->CapturePacket(data);
   }
+
   ice_transport_->SendPacket(reinterpret_cast<const char*>(data.data()),
                              data.size(), packet_options);
   written = data.size();
@@ -266,7 +267,8 @@ DtlsTransportInternalImpl::DtlsTransportInternalImpl(
             piggybacked_dtls_callback_(
                 this,
                 ReceivedIpPacket(piggybacked_dtls_packet, SocketAddress()));
-          }) {
+          },
+          [this]() { CompleteDtlsInStun(); }) {
   RTC_DCHECK(ice_transport_);
   ConnectToIceTransport();
   dtls_in_stun_ = env_.field_trials().IsEnabled("WebRTC-IceHandshakeDtls");
@@ -286,8 +288,24 @@ DtlsTransportInternalImpl::DtlsTransportInternalImpl(
           ssl_stream_factory) {}
 
 DtlsTransportInternalImpl::~DtlsTransportInternalImpl() {
-  ice_transport()->ResetDtlsStunPiggybackCallbacks();
-  ice_transport()->DeregisterReceivedPacketCallback(this);
+  if (downward_) {
+    downward_->SetDtlsStunPiggybackController(nullptr);
+  }
+  auto ice_transport = this->ice_transport();
+  if (ice_transport) {
+    ice_transport->ResetDtlsStunPiggybackCallbacks();
+    ice_transport->DeregisterReceivedPacketCallback(this);
+  }
+}
+
+void DtlsTransportInternalImpl::CompleteDtlsInStun() {
+  if (downward_) {
+    downward_->SetDtlsStunPiggybackController(nullptr);
+  }
+  auto ice_transport = this->ice_transport();
+  if (ice_transport) {
+    ice_transport->ResetDtlsStunPiggybackCallbacks();
+  }
 }
 
 DtlsTransportState DtlsTransportInternalImpl::dtls_state() const {
@@ -502,6 +520,7 @@ bool DtlsTransportInternalImpl::SetupDtls() {
   RTC_DCHECK(dtls_role_);
 
   dtls_in_stun_ = ice_transport()->config().dtls_handshake_in_stun;
+
   {
     auto downward = std::make_unique<StreamInterfaceChannel>(ice_transport());
     StreamInterfaceChannel* downward_ptr = downward.get();
@@ -976,7 +995,6 @@ void DtlsTransportInternalImpl::OnDtlsEvent(int sig, int err) {
       RTC_DCHECK(ret);
       dtls_stun_piggyback_controller_.SetDtlsHandshakeComplete(
           dtls_role_ == SSL_CLIENT, ssl_version_bytes == kDtls13VersionBytes);
-      downward_->SetDtlsStunPiggybackController(nullptr);
       set_dtls_state(DtlsTransportState::kConnected);
       set_writable(true);
     }
@@ -1004,6 +1022,7 @@ void DtlsTransportInternalImpl::OnDtlsEvent(int sig, int err) {
         set_writable(false);
         set_dtls_state(DtlsTransportState::kClosed);
         NotifyOnClose();
+        CompleteDtlsInStun();
       } else if (ret == SR_ERROR) {
         // Remote peer shut down the association with an error.
         RTC_LOG(LS_INFO)
@@ -1013,6 +1032,7 @@ void DtlsTransportInternalImpl::OnDtlsEvent(int sig, int err) {
         set_writable(false);
         set_dtls_state(DtlsTransportState::kFailed);
         NotifyOnClose();
+        CompleteDtlsInStun();
       }
     } while (ret == SR_SUCCESS);
   }
diff --git a/p2p/dtls/dtls_transport.h b/p2p/dtls/dtls_transport.h
index 66b3953043..c642525889 100644
--- a/p2p/dtls/dtls_transport.h
+++ b/p2p/dtls/dtls_transport.h
@@ -302,6 +302,8 @@ class DtlsTransportInternalImpl : public DtlsTransportInternal {
   void set_writable(bool writable);
   // Sets the DTLS state, signaling if necessary.
   void set_dtls_state(DtlsTransportState state);
+
+  void CompleteDtlsInStun();
   void SetPiggybackDtlsDataCallback(
       absl::AnyInvocable<void(PacketTransportInternal* transport,
                               const ReceivedIpPacket& packet)> callback);

From 1d5f5e30f65f09c98fbb74c6c7b0eed36ed17fbc Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Tue, 17 Feb 2026 20:08:19 -0800
Subject: [PATCH 083/472] Update WebRTC code version (2026-02-18T04:08:18).

Bug: None
Change-Id: I455e4d430189d343013ea884b60c77a41ebee948
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/450140
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#46936}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index 70c2be6283..96c187e869 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-02-16T04:04:48";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-02-18T04:08:18";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From 5ce1ea76934628685d6f6d97df188a854ce22459 Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Wed, 18 Feb 2026 10:08:56 +0100
Subject: [PATCH 084/472] Remove inactive OWNERS in network related folders

Ensure each file has at least two OWNERS.
Add DIR_METADATA files to document associated component for changed
folders.

Bug: webrtc:483056241
Change-Id: Ie056a6ff27926172c9bfb107310a93801f8f711e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449101
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46937}
---
 api/transport/DIR_METADATA                    | 3 +++
 api/transport/OWNERS                          | 2 +-
 call/DIR_METADATA                             | 0
 call/OWNERS                                   | 2 --
 modules/congestion_controller/DIR_METADATA    | 3 +++
 modules/congestion_controller/OWNERS          | 2 --
 modules/pacing/DIR_METADATA                   | 3 +++
 modules/pacing/OWNERS                         | 2 --
 modules/remote_bitrate_estimator/DIR_METADATA | 3 +++
 modules/remote_bitrate_estimator/OWNERS       | 2 --
 modules/rtp_rtcp/DIR_METADATA                 | 3 +++
 modules/rtp_rtcp/OWNERS                       | 2 --
 net/dcsctp/DIR_METADATA                       | 3 +++
 net/dcsctp/OWNERS                             | 2 +-
 p2p/DIR_METADATA                              | 3 +++
 p2p/OWNERS                                    | 2 --
 16 files changed, 23 insertions(+), 14 deletions(-)
 create mode 100644 api/transport/DIR_METADATA
 create mode 100644 call/DIR_METADATA
 create mode 100644 modules/congestion_controller/DIR_METADATA
 create mode 100644 modules/pacing/DIR_METADATA
 create mode 100644 modules/remote_bitrate_estimator/DIR_METADATA
 create mode 100644 modules/rtp_rtcp/DIR_METADATA
 create mode 100644 net/dcsctp/DIR_METADATA
 create mode 100644 p2p/DIR_METADATA

diff --git a/api/transport/DIR_METADATA b/api/transport/DIR_METADATA
new file mode 100644
index 0000000000..783c9b8958
--- /dev/null
+++ b/api/transport/DIR_METADATA
@@ -0,0 +1,3 @@
+buganizer_public: {
+  component_id: 1565889  # Network
+}
\ No newline at end of file
diff --git a/api/transport/OWNERS b/api/transport/OWNERS
index 5991f6fc56..eef3b09a0f 100644
--- a/api/transport/OWNERS
+++ b/api/transport/OWNERS
@@ -1,2 +1,2 @@
-srte@webrtc.org
+perkj@webrtc.org
 terelius@webrtc.org
diff --git a/call/DIR_METADATA b/call/DIR_METADATA
new file mode 100644
index 0000000000..e69de29bb2
diff --git a/call/OWNERS b/call/OWNERS
index d37ac06b3c..0c771a0c1c 100644
--- a/call/OWNERS
+++ b/call/OWNERS
@@ -2,8 +2,6 @@ sprang@webrtc.org
 danilchap@webrtc.org
 brandtr@webrtc.org
 tommi@webrtc.org
-mflodman@webrtc.org
-stefan@webrtc.org
 perkj@webrtc.org
 
 per-file version.cc=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com
diff --git a/modules/congestion_controller/DIR_METADATA b/modules/congestion_controller/DIR_METADATA
new file mode 100644
index 0000000000..a904cc9a09
--- /dev/null
+++ b/modules/congestion_controller/DIR_METADATA
@@ -0,0 +1,3 @@
+buganizer_public: {
+  component_id: 1565318  # BWE
+}
\ No newline at end of file
diff --git a/modules/congestion_controller/OWNERS b/modules/congestion_controller/OWNERS
index 9a836bad06..7a424948a6 100644
--- a/modules/congestion_controller/OWNERS
+++ b/modules/congestion_controller/OWNERS
@@ -1,7 +1,5 @@
 danilchap@webrtc.org
 linderborg@webrtc.org
-stefan@webrtc.org
 terelius@webrtc.org
-mflodman@webrtc.org
 yinwa@webrtc.org
 perkj@webrtc.org
diff --git a/modules/pacing/DIR_METADATA b/modules/pacing/DIR_METADATA
new file mode 100644
index 0000000000..a904cc9a09
--- /dev/null
+++ b/modules/pacing/DIR_METADATA
@@ -0,0 +1,3 @@
+buganizer_public: {
+  component_id: 1565318  # BWE
+}
\ No newline at end of file
diff --git a/modules/pacing/OWNERS b/modules/pacing/OWNERS
index f709476d43..fa476a42c6 100644
--- a/modules/pacing/OWNERS
+++ b/modules/pacing/OWNERS
@@ -1,5 +1,3 @@
-stefan@webrtc.org
-mflodman@webrtc.org
 philipel@webrtc.org
 sprang@webrtc.org
 perkj@webrtc.org
diff --git a/modules/remote_bitrate_estimator/DIR_METADATA b/modules/remote_bitrate_estimator/DIR_METADATA
new file mode 100644
index 0000000000..a904cc9a09
--- /dev/null
+++ b/modules/remote_bitrate_estimator/DIR_METADATA
@@ -0,0 +1,3 @@
+buganizer_public: {
+  component_id: 1565318  # BWE
+}
\ No newline at end of file
diff --git a/modules/remote_bitrate_estimator/OWNERS b/modules/remote_bitrate_estimator/OWNERS
index 993e2fd77b..9db73ffca9 100644
--- a/modules/remote_bitrate_estimator/OWNERS
+++ b/modules/remote_bitrate_estimator/OWNERS
@@ -1,5 +1,3 @@
 danilchap@webrtc.org
-stefan@webrtc.org
 terelius@webrtc.org
-mflodman@webrtc.org
 perkj@webrtc.org
diff --git a/modules/rtp_rtcp/DIR_METADATA b/modules/rtp_rtcp/DIR_METADATA
new file mode 100644
index 0000000000..783c9b8958
--- /dev/null
+++ b/modules/rtp_rtcp/DIR_METADATA
@@ -0,0 +1,3 @@
+buganizer_public: {
+  component_id: 1565889  # Network
+}
\ No newline at end of file
diff --git a/modules/rtp_rtcp/OWNERS b/modules/rtp_rtcp/OWNERS
index 47d12c401f..bc1dea8ba4 100644
--- a/modules/rtp_rtcp/OWNERS
+++ b/modules/rtp_rtcp/OWNERS
@@ -1,6 +1,4 @@
-stefan@webrtc.org
 henrik.lundin@webrtc.org
-mflodman@webrtc.org
 asapersson@webrtc.org
 danilchap@webrtc.org
 sprang@webrtc.org
diff --git a/net/dcsctp/DIR_METADATA b/net/dcsctp/DIR_METADATA
new file mode 100644
index 0000000000..afb0cad4ee
--- /dev/null
+++ b/net/dcsctp/DIR_METADATA
@@ -0,0 +1,3 @@
+buganizer_public: {
+  component_id: 1565733  # DataChannel
+}
\ No newline at end of file
diff --git a/net/dcsctp/OWNERS b/net/dcsctp/OWNERS
index 06a0f86179..af58d19247 100644
--- a/net/dcsctp/OWNERS
+++ b/net/dcsctp/OWNERS
@@ -1,2 +1,2 @@
 boivie@webrtc.org
-orphis@webrtc.org
+danilchap@webrtc.org
diff --git a/p2p/DIR_METADATA b/p2p/DIR_METADATA
new file mode 100644
index 0000000000..783c9b8958
--- /dev/null
+++ b/p2p/DIR_METADATA
@@ -0,0 +1,3 @@
+buganizer_public: {
+  component_id: 1565889  # Network
+}
\ No newline at end of file
diff --git a/p2p/OWNERS b/p2p/OWNERS
index 5f4ceb6015..eb01ed8625 100644
--- a/p2p/OWNERS
+++ b/p2p/OWNERS
@@ -1,6 +1,4 @@
 hta@webrtc.org
-mflodman@webrtc.org
 perkj@webrtc.org
-sergeyu@chromium.org
 tommi@webrtc.org
 jonaso@webrtc.org

From 259c1ca4c51a1dbd05e9d45d5e6e64479a03d557 Mon Sep 17 00:00:00 2001
From: Ludvig Noring <ludvig.noring@eyevinn.se>
Date: Tue, 27 Jan 2026 08:16:29 +0100
Subject: [PATCH 085/472] Clamp timer interval in AudioDeviceBuffer::LogStats
 to handle time jumps

The assertion that time_to_wait_ms > 0 can fail when the process is
paused (e.g., under a debugger).

Bug: webrtc:478779824
Change-Id: I92a96ce621ce766e012f1e2c9fbdde46fd12e345
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/443462
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46938}
---
 modules/audio_device/audio_device_buffer.cc | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

diff --git a/modules/audio_device/audio_device_buffer.cc b/modules/audio_device/audio_device_buffer.cc
index ba064ed399..c27ba37666 100644
--- a/modules/audio_device/audio_device_buffer.cc
+++ b/modules/audio_device/audio_device_buffer.cc
@@ -10,6 +10,7 @@
 
 #include "modules/audio_device/audio_device_buffer.h"
 
+#include <algorithm>
 #include <cmath>
 #include <cstddef>
 #include <cstdint>
@@ -512,9 +513,8 @@ void AudioDeviceBuffer::LogStats(LogState state) {
   }
   last_stats_ = stats;
 
-  int64_t time_to_wait_ms =
-      next_callback_time - env_.clock().TimeInMilliseconds();
-  RTC_DCHECK_GT(time_to_wait_ms, 0) << "Invalid timer interval";
+  int64_t time_to_wait_ms = std::max<int64_t>(
+      next_callback_time - env_.clock().TimeInMilliseconds(), 0);
 
   // Keep posting new (delayed) tasks until state is changed to kLogStop.
   task_queue_->PostDelayedTask(

From 796906eb332893b2c861338e23ea5a61b4782ca5 Mon Sep 17 00:00:00 2001
From: AhmadDurrani579 <ahmaddurranitrg@gmail.com>
Date: Mon, 16 Feb 2026 12:34:03 +0000
Subject: [PATCH 086/472] Fix potential OOM and file handle leaks in
 OpenY4mFile

The OpenY4mFile function reads the file header without a size limit,
which could lead to an infinite loop or OOM crash if the file
does not contain a newline character.

Additionally, file handles were not being closed on several error
paths, leading to resource leaks.

This CL:
1. Adds a check to abort reading if the header exceeds 64 bytes.
2. Fixes multiple file handle leaks where fclose() was missing.

Bug: None
Test: tools_unittests --gtest_filter="*OpenY4mFileAbortsOnLargeHeader*"
Change-Id: I0d4ce793713fc420915bcda16c1fd754c04eb797
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/448082
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46939}
---
 rtc_tools/BUILD.gn                      |  1 +
 rtc_tools/video_file_reader.cc          | 18 ++++++++++++++++--
 rtc_tools/video_file_reader_unittest.cc | 18 ++++++++++++++++++
 3 files changed, 35 insertions(+), 2 deletions(-)

diff --git a/rtc_tools/BUILD.gn b/rtc_tools/BUILD.gn
index 24c20bbb24..0f4c54c9c4 100644
--- a/rtc_tools/BUILD.gn
+++ b/rtc_tools/BUILD.gn
@@ -62,6 +62,7 @@ rtc_library("video_file_reader") {
     "../rtc_base:logging",
     "../rtc_base:refcount",
     "../rtc_base:stringutils",
+    "//third_party/abseil-cpp/absl/cleanup",
     "//third_party/abseil-cpp/absl/strings",
   ]
 }
diff --git a/rtc_tools/video_file_reader.cc b/rtc_tools/video_file_reader.cc
index cacee666ab..ce33bb2560 100644
--- a/rtc_tools/video_file_reader.cc
+++ b/rtc_tools/video_file_reader.cc
@@ -16,8 +16,10 @@
 #include <cstring>
 #include <optional>
 #include <string>
+#include <utility>
 #include <vector>
 
+#include "absl/cleanup/cleanup.h"
 #include "absl/strings/match.h"
 #include "api/make_ref_counted.h"
 #include "api/scoped_refptr.h"
@@ -33,6 +35,11 @@ namespace test {
 
 namespace {
 
+// A limit to prevent infinite loops or OOM if a Y4M file has a malformed
+// header. 256 bytes is plenty for any valid Y4M header but small enough
+// to be safe.
+constexpr int kMaxHeaderSize = 256;
+
 bool ReadBytes(uint8_t* dst, size_t n, FILE* file) {
   return fread(reinterpret_cast<char*>(dst), /* size= */ 1, n, file) == n;
 }
@@ -128,6 +135,8 @@ scoped_refptr<Video> OpenY4mFile(const std::string& file_name) {
     return nullptr;
   }
 
+  absl::Cleanup file_closer = [file] { fclose(file); };
+
   int parse_file_header_result = -1;
   if (fscanf(file, "YUV4MPEG2 %n", &parse_file_header_result) != 0 ||
       parse_file_header_result == -1) {
@@ -145,6 +154,10 @@ scoped_refptr<Video> OpenY4mFile(const std::string& file_name) {
     }
     if (c == '\n')
       break;
+    if (header_line.size() >= kMaxHeaderSize) {
+      RTC_LOG(LS_ERROR) << "Header line too long, file might be corrupted.";
+      return nullptr;
+    }
     header_line.push_back(static_cast<char>(c));
   }
 
@@ -230,7 +243,7 @@ scoped_refptr<Video> OpenY4mFile(const std::string& file_name) {
     return nullptr;
   }
   RTC_LOG(LS_INFO) << "Video has " << frame_positions.size() << " frames";
-
+  std::move(file_closer).Cancel();
   return make_ref_counted<VideoFile>(*width, *height, frame_positions, file);
 }
 
@@ -246,12 +259,12 @@ scoped_refptr<Video> OpenYuvFile(const std::string& file_name,
     RTC_LOG(LS_ERROR) << "Could not open input file for reading: " << file_name;
     return nullptr;
   }
+  absl::Cleanup file_closer = [file] { fclose(file); };
 
   if (width % 2 != 0 || height % 2 != 0) {
     RTC_LOG(LS_ERROR)
         << "Only supports even width/height so that chroma size is a "
            "whole number.";
-    fclose(file);
     return nullptr;
   }
 
@@ -277,6 +290,7 @@ scoped_refptr<Video> OpenYuvFile(const std::string& file_name,
   }
   RTC_LOG(LS_INFO) << "Video has " << frame_positions.size() << " frames";
 
+  std::move(file_closer).Cancel();
   return make_ref_counted<VideoFile>(width, height, frame_positions, file);
 }
 
diff --git a/rtc_tools/video_file_reader_unittest.cc b/rtc_tools/video_file_reader_unittest.cc
index 3a5bb630be..57d42a51d3 100644
--- a/rtc_tools/video_file_reader_unittest.cc
+++ b/rtc_tools/video_file_reader_unittest.cc
@@ -124,5 +124,23 @@ TEST_F(YuvFileReaderTest, TestPixelContent) {
   }
 }
 
+TEST(VideoFileReaderTest, OpenY4mFileAbortsOnLargeHeader) {
+  const std::string file_name =
+      TempFilename(test::OutputPath(), "temp_large_header.y4m");
+
+  FILE* file = fopen(file_name.c_str(), "wb");
+  ASSERT_TRUE(file != nullptr);
+
+  fprintf(file, "YUV4MPEG2 ");
+
+  // Write 2000 'A's without a newline (Simulate the attack)
+  for (int i = 0; i < 2000; ++i) {
+    fputc('A', file);
+  }
+  fclose(file);
+  EXPECT_EQ(nullptr, OpenY4mFile(file_name));
+
+  remove(file_name.c_str());
+}
 }  // namespace test
 }  // namespace webrtc

From 726c2f056cc37d3eb8e491439150bb2bc21546d2 Mon Sep 17 00:00:00 2001
From: Rasmus Brandt <brandtr@webrtc.org>
Date: Wed, 18 Feb 2026 11:18:16 +0100
Subject: [PATCH 087/472] Video timing simulator: improve robustness of
 `PopulateFrameDelayVariations`
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

We should explicitly sort the frames before performing the sliding
window calculations.

Bug: b/423646186
Change-Id: I6eaf96cacc0b85288210c958471bf28e6a6a6964
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/450200
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46940}
---
 video/timing/simulator/stream_base.h | 2 ++
 1 file changed, 2 insertions(+)

diff --git a/video/timing/simulator/stream_base.h b/video/timing/simulator/stream_base.h
index 6f545116f0..e3191bcb1c 100644
--- a/video/timing/simulator/stream_base.h
+++ b/video/timing/simulator/stream_base.h
@@ -48,6 +48,8 @@ struct StreamBase {
       return;
     }
 
+    SortByArrivalOrder(self().frames);
+
     // One-way delay measurement offsets.
     Timestamp arrival_offset = Timestamp::PlusInfinity();
     Timestamp departure_offset = Timestamp::PlusInfinity();

From 7a8eed0780d804e6d7dfdace14513f3a0e092971 Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Wed, 18 Feb 2026 10:47:59 +0100
Subject: [PATCH 088/472] Deprecate ArrayView(nullptr_t, size) constructor

Use default constructor instead.
std::span doesn't have matching constructor.

Bug: webrtc:439801349
Change-Id: Ic228c128835a86296970ef0cfc6cc3af770caaad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/450180
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46941}
---
 api/array_view.h                      | 6 +++---
 rtc_base/bitstream_reader_unittest.cc | 6 +++---
 rtc_base/byte_buffer_unittest.cc      | 2 +-
 test/fuzzers/fuzz_data_helper.h       | 2 +-
 4 files changed, 8 insertions(+), 8 deletions(-)

diff --git a/api/array_view.h b/api/array_view.h
index 8d367e4fa3..c4a487b59b 100644
--- a/api/array_view.h
+++ b/api/array_view.h
@@ -170,12 +170,12 @@ class ArrayView final : public array_view_internal::ArrayViewBase<T, Size> {
 
   // Construct an empty ArrayView. Note that fixed-size ArrayViews of size > 0
   // cannot be empty.
-  ArrayView() : ArrayView(nullptr, 0) {}
+  ArrayView() : ArrayView(static_cast<T*>(nullptr), 0) {}
   ABSL_DEPRECATE_AND_INLINE()
   ArrayView(std::nullptr_t)  // NOLINT
       : ArrayView() {}
-  ArrayView(std::nullptr_t, size_t size)
-      : ArrayView(static_cast<T*>(nullptr), size) {
+  ABSL_DEPRECATE_AND_INLINE()
+  ArrayView(std::nullptr_t, size_t size) : ArrayView() {
     static_assert(Size == 0 || Size == array_view_internal::kArrayViewVarSize,
                   "");
     RTC_DCHECK_EQ(0, size);
diff --git a/rtc_base/bitstream_reader_unittest.cc b/rtc_base/bitstream_reader_unittest.cc
index 4d76546f69..4ad6624db4 100644
--- a/rtc_base/bitstream_reader_unittest.cc
+++ b/rtc_base/bitstream_reader_unittest.cc
@@ -197,14 +197,14 @@ TEST(BitstreamReaderTest, ReadBits) {
 }
 
 TEST(BitstreamReaderTest, ReadZeroBits) {
-  BitstreamReader reader(ArrayView<const uint8_t>(nullptr, 0));
+  BitstreamReader reader(ArrayView<const uint8_t>{});
 
   EXPECT_EQ(reader.ReadBits(0), 0u);
   EXPECT_TRUE(reader.Ok());
 }
 
 TEST(BitstreamReaderTest, ReadBitFromEmptyArray) {
-  BitstreamReader reader(ArrayView<const uint8_t>(nullptr, 0));
+  BitstreamReader reader(ArrayView<const uint8_t>{});
 
   // Trying to read from the empty array shouldn't dereference the pointer,
   // i.e. shouldn't crash.
@@ -213,7 +213,7 @@ TEST(BitstreamReaderTest, ReadBitFromEmptyArray) {
 }
 
 TEST(BitstreamReaderTest, ReadBitsFromEmptyArray) {
-  BitstreamReader reader(ArrayView<const uint8_t>(nullptr, 0));
+  BitstreamReader reader(ArrayView<const uint8_t>{});
 
   // Trying to read from the empty array shouldn't dereference the pointer,
   // i.e. shouldn't crash.
diff --git a/rtc_base/byte_buffer_unittest.cc b/rtc_base/byte_buffer_unittest.cc
index cb4fdde41c..267f107ee6 100644
--- a/rtc_base/byte_buffer_unittest.cc
+++ b/rtc_base/byte_buffer_unittest.cc
@@ -117,7 +117,7 @@ TEST(ByteBufferTest, TestBufferLength) {
 
 TEST(ByteBufferTest, TestReadWriteBuffer) {
   ByteBufferWriter buffer;
-  ByteBufferReader read_buf(ArrayView<const uint8_t>(nullptr, 0));
+  ByteBufferReader read_buf(ArrayView<const uint8_t>{});
   uint8_t ru8;
   EXPECT_FALSE(read_buf.ReadUInt8(&ru8));
 
diff --git a/test/fuzzers/fuzz_data_helper.h b/test/fuzzers/fuzz_data_helper.h
index b76cb2d93a..42a469c74b 100644
--- a/test/fuzzers/fuzz_data_helper.h
+++ b/test/fuzzers/fuzz_data_helper.h
@@ -86,7 +86,7 @@ class FuzzDataHelper {
 
   ArrayView<const uint8_t> ReadByteArray(size_t bytes) {
     if (!CanReadBytes(bytes)) {
-      return ArrayView<const uint8_t>(nullptr, 0);
+      return {};
     }
     const size_t index_to_return = data_ix_;
     data_ix_ += bytes;

From 71c87155e7fbe0b67499295f288fd0f6ec94e1d2 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Erik=20Spr=C3=A5ng?= <sprang@webrtc.org>
Date: Mon, 16 Feb 2026 13:33:24 +0100
Subject: [PATCH 089/472] Reland "Move VideoQualityAnalyzer to OnFrameDropped
 callback."
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This is a reland of commit a7c29a690e22a2c0edd6f543e580c7f601362061
Problems seen in previous attempt should be addressed as of
https://webrtc-review.git.corp.google.com/c/src/+/449687

Original change's description:
> Move VideoQualityAnalyzer to OnFrameDropped callback.
>
> This deprecates usage of the EncodedImageCallback::DropReason type,
> which was never actually read anyway, and moves this code to the new
> callback method.
>
> Bug: webrtc:467444018
> Change-Id: I105e44237e6f1bf02bd5664cc6a098a2e734c6e9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449640
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#46913}

Bug: webrtc:467444018
Change-Id: Id61ad42cbaf4d1dc00fbe1338f9d3c203837332e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/450040
Auto-Submit: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46942}
---
 api/test/video_quality_analyzer_interface.h         |  9 +++++++--
 .../video/default_video_quality_analyzer.cc         |  5 +----
 .../analyzer/video/default_video_quality_analyzer.h |  4 +---
 .../video/example_video_quality_analyzer.cc         |  5 +----
 .../analyzer/video/example_video_quality_analyzer.h |  4 +---
 .../video/quality_analyzing_video_encoder.cc        | 13 ++++++++++++-
 .../video/quality_analyzing_video_encoder.h         |  4 ++++
 7 files changed, 27 insertions(+), 17 deletions(-)

diff --git a/api/test/video_quality_analyzer_interface.h b/api/test/video_quality_analyzer_interface.h
index 48685b9936..43223364ec 100644
--- a/api/test/video_quality_analyzer_interface.h
+++ b/api/test/video_quality_analyzer_interface.h
@@ -112,8 +112,13 @@ class VideoQualityAnalyzerInterface
                               bool /* discarded */) {}
   // Will be called for each frame dropped by encoder.
   // `peer_name` is name of the peer on which side frame drop was detected.
-  virtual void OnFrameDropped(absl::string_view /* peer_name */,
-                              EncodedImageCallback::DropReason /* reason */) {}
+  virtual void OnFrameDropped(absl::string_view /* peer_name */) {}
+  // TODO: webrtc:467444018 - Remove when downstream usage is gone.
+  [[deprecated("Use callback without DropReason parameter")]]
+  virtual void OnFrameDropped(absl::string_view peer_name,
+                              EncodedImageCallback::DropReason /* reason */) {
+    OnFrameDropped(peer_name);
+  }
   // Will be called before calling the decoder.
   // `peer_name` is name of the peer on which side frame was received.
   virtual void OnFramePreDecode(absl::string_view /* peer_name */,
diff --git a/test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc b/test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc
index f5c322b407..5130dd6a3c 100644
--- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc
+++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc
@@ -33,7 +33,6 @@
 #include "api/units/timestamp.h"
 #include "api/video/encoded_image.h"
 #include "api/video/video_frame.h"
-#include "api/video_codecs/video_encoder.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/strings/string_builder.h"
@@ -385,9 +384,7 @@ void DefaultVideoQualityAnalyzer::OnFrameEncoded(
   }
 }
 
-void DefaultVideoQualityAnalyzer::OnFrameDropped(
-    absl::string_view peer_name,
-    EncodedImageCallback::DropReason reason) {
+void DefaultVideoQualityAnalyzer::OnFrameDropped(absl::string_view peer_name) {
   // Here we do nothing, because we will see this drop on renderer side.
 }
 
diff --git a/test/pc/e2e/analyzer/video/default_video_quality_analyzer.h b/test/pc/e2e/analyzer/video/default_video_quality_analyzer.h
index 0e041431ed..6b056d07e1 100644
--- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer.h
+++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer.h
@@ -29,7 +29,6 @@
 #include "api/units/timestamp.h"
 #include "api/video/encoded_image.h"
 #include "api/video/video_frame.h"
-#include "api/video_codecs/video_encoder.h"
 #include "rtc_base/synchronization/mutex.h"
 #include "rtc_base/thread_annotations.h"
 #include "system_wrappers/include/clock.h"
@@ -64,8 +63,7 @@ class DefaultVideoQualityAnalyzer : public VideoQualityAnalyzerInterface {
                       const EncodedImage& encoded_image,
                       const EncoderStats& stats,
                       bool discarded) override;
-  void OnFrameDropped(absl::string_view peer_name,
-                      EncodedImageCallback::DropReason reason) override;
+  void OnFrameDropped(absl::string_view peer_name) override;
   void OnFramePreDecode(absl::string_view peer_name,
                         uint16_t frame_id,
                         const EncodedImage& input_image) override;
diff --git a/test/pc/e2e/analyzer/video/example_video_quality_analyzer.cc b/test/pc/e2e/analyzer/video/example_video_quality_analyzer.cc
index d68db34033..8bd9453335 100644
--- a/test/pc/e2e/analyzer/video/example_video_quality_analyzer.cc
+++ b/test/pc/e2e/analyzer/video/example_video_quality_analyzer.cc
@@ -17,7 +17,6 @@
 #include "api/array_view.h"
 #include "api/video/encoded_image.h"
 #include "api/video/video_frame.h"
-#include "api/video_codecs/video_encoder.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/synchronization/mutex.h"
@@ -75,9 +74,7 @@ void ExampleVideoQualityAnalyzer::OnFrameEncoded(
   ++frames_encoded_;
 }
 
-void ExampleVideoQualityAnalyzer::OnFrameDropped(
-    absl::string_view peer_name,
-    EncodedImageCallback::DropReason reason) {
+void ExampleVideoQualityAnalyzer::OnFrameDropped(absl::string_view peer_name) {
   RTC_LOG(LS_INFO) << "Frame dropped by encoder";
   MutexLock lock(&lock_);
   ++frames_dropped_;
diff --git a/test/pc/e2e/analyzer/video/example_video_quality_analyzer.h b/test/pc/e2e/analyzer/video/example_video_quality_analyzer.h
index e347524439..59b056a74c 100644
--- a/test/pc/e2e/analyzer/video/example_video_quality_analyzer.h
+++ b/test/pc/e2e/analyzer/video/example_video_quality_analyzer.h
@@ -21,7 +21,6 @@
 #include "api/test/video_quality_analyzer_interface.h"
 #include "api/video/encoded_image.h"
 #include "api/video/video_frame.h"
-#include "api/video_codecs/video_encoder.h"
 #include "rtc_base/synchronization/mutex.h"
 #include "rtc_base/thread_annotations.h"
 
@@ -49,8 +48,7 @@ class ExampleVideoQualityAnalyzer : public VideoQualityAnalyzerInterface {
                       const EncodedImage& encoded_image,
                       const EncoderStats& stats,
                       bool discarded) override;
-  void OnFrameDropped(absl::string_view peer_name,
-                      EncodedImageCallback::DropReason reason) override;
+  void OnFrameDropped(absl::string_view peer_name) override;
   void OnFramePreDecode(absl::string_view peer_name,
                         uint16_t frame_id,
                         const EncodedImage& encoded_image) override;
diff --git a/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.cc b/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.cc
index 6516eead14..01138cf25a 100644
--- a/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.cc
+++ b/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.cc
@@ -339,11 +339,22 @@ EncodedImageCallback::Result QualityAnalyzingVideoEncoder::OnEncodedImage(
 void QualityAnalyzingVideoEncoder::OnDroppedFrame(
     EncodedImageCallback::DropReason reason) {
   MutexLock lock(&mutex_);
-  analyzer_->OnFrameDropped(peer_name_, reason);
+  analyzer_->OnFrameDropped(peer_name_);
   RTC_DCHECK(delegate_callback_);
   delegate_callback_->OnDroppedFrame(reason);
 }
 
+void QualityAnalyzingVideoEncoder::OnFrameDropped(
+    uint32_t rtp_timestamp,
+    int spatial_id,
+    bool is_end_of_temporal_unit) {
+  MutexLock lock(&mutex_);
+  analyzer_->OnFrameDropped(peer_name_);
+  RTC_DCHECK(delegate_callback_);
+  delegate_callback_->OnFrameDropped(rtp_timestamp, spatial_id,
+                                     is_end_of_temporal_unit);
+}
+
 bool QualityAnalyzingVideoEncoder::ShouldDiscard(
     uint16_t frame_id,
     const EncodedImage& encoded_image) {
diff --git a/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.h b/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.h
index 2be4bdffc7..bfa7cc9cf0 100644
--- a/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.h
+++ b/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.h
@@ -88,7 +88,11 @@ class QualityAnalyzingVideoEncoder : public VideoEncoder,
   EncodedImageCallback::Result OnEncodedImage(
       const EncodedImage& encoded_image,
       const CodecSpecificInfo* codec_specific_info) override;
+  // TODO: webrtc:467444018 - Remove after usage shifts to OnFrameDropped.
   void OnDroppedFrame(DropReason reason) override;
+  void OnFrameDropped(uint32_t rtp_timestamp,
+                      int spatial_id,
+                      bool is_end_of_temporal_unit) override;
 
  private:
   enum SimulcastMode {

From 310bd0755b008881f1469c2de111b92705940a78 Mon Sep 17 00:00:00 2001
From: Ludvig Noring <ludvig.noring@eyevinn.se>
Date: Tue, 3 Feb 2026 09:26:26 +0100
Subject: [PATCH 090/472] Linux: Race condition in ALSA/PulseAudio symbol table
 loading

- Use std::call_once in GetAlsaSymbolTable() and GetPulseSymbolTable() to
  ensure Load() is called exactly once
- Change Init()/InitPulseAudio() to check IsLoaded() instead of calling Load()

Bug: webrtc:481087594
Change-Id: I936dd7919ed3aa7f5d66d586cfd3f0ef3f7cd63f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/445820
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46943}
---
 modules/audio_device/linux/audio_device_alsa_linux.cc  | 8 ++++++--
 modules/audio_device/linux/audio_device_pulse_linux.cc | 9 ++++++---
 2 files changed, 12 insertions(+), 5 deletions(-)

diff --git a/modules/audio_device/linux/audio_device_alsa_linux.cc b/modules/audio_device/linux/audio_device_alsa_linux.cc
index 4e504f354d..731d5b161d 100644
--- a/modules/audio_device/linux/audio_device_alsa_linux.cc
+++ b/modules/audio_device/linux/audio_device_alsa_linux.cc
@@ -34,7 +34,11 @@
 #endif
 
 WebRTCAlsaSymbolTable* GetAlsaSymbolTable() {
-  static WebRTCAlsaSymbolTable* alsa_symbol_table = new WebRTCAlsaSymbolTable();
+  static WebRTCAlsaSymbolTable* alsa_symbol_table = []() {
+    auto* table = new WebRTCAlsaSymbolTable();
+    table->Load();
+    return table;
+  }();
   return alsa_symbol_table;
 }
 
@@ -161,7 +165,7 @@ AudioDeviceGeneric::InitStatus AudioDeviceLinuxALSA::Init() {
   MutexLock lock(&mutex_);
 
   // Load libasound
-  if (!GetAlsaSymbolTable()->Load()) {
+  if (!GetAlsaSymbolTable()->IsLoaded()) {
     // Alsa is not installed on this system
     RTC_LOG(LS_ERROR) << "failed to load symbol table";
     return InitStatus::OTHER_ERROR;
diff --git a/modules/audio_device/linux/audio_device_pulse_linux.cc b/modules/audio_device/linux/audio_device_pulse_linux.cc
index a098e908ff..f1121c1a9a 100644
--- a/modules/audio_device/linux/audio_device_pulse_linux.cc
+++ b/modules/audio_device/linux/audio_device_pulse_linux.cc
@@ -31,8 +31,11 @@
 #endif
 
 WebRTCPulseSymbolTable* GetPulseSymbolTable() {
-  static WebRTCPulseSymbolTable* pulse_symbol_table =
-      new WebRTCPulseSymbolTable();
+  static WebRTCPulseSymbolTable* pulse_symbol_table = []() {
+    auto* table = new WebRTCPulseSymbolTable();
+    table->Load();
+    return table;
+  }();
   return pulse_symbol_table;
 }
 
@@ -1555,7 +1558,7 @@ int32_t AudioDeviceLinuxPulse::InitPulseAudio() {
   int retVal = 0;
 
   // Load libpulse
-  if (!GetPulseSymbolTable()->Load()) {
+  if (!GetPulseSymbolTable()->IsLoaded()) {
     // Most likely the Pulse library and sound server are not installed on
     // this system
     RTC_LOG(LS_ERROR) << "failed to load symbol table";

From 8ffca5923aa0f09812b970ae4f6b53d68bc85972 Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Mon, 16 Feb 2026 12:11:03 +0000
Subject: [PATCH 091/472] Switch ArrayView, CopyOnWriteBuffer and AudioView to
 iterators

These functions previously returned a T*, which meant that they
had to be manipulated with pointer arithmetic. Using iterators
gives opportunities for bounds checking.

Bug: webrtc:478086887, webrtc:439801349
Change-Id: If5b91920105b43e72cf0114f003914bd39d145c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449660
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46944}
---
 api/array_view.h                              | 27 ++++++++++++-------
 api/array_view_unittest.cc                    |  8 +++---
 api/audio/audio_frame.cc                      |  2 +-
 api/audio/audio_view.h                        | 20 ++++++++------
 api/audio/test/audio_frame_unittest.cc        |  2 +-
 .../audio_device/include/test_audio_device.cc | 12 ++++-----
 .../aec3/reverb_decay_estimator.cc            |  3 +--
 .../aec_dump/aec_dump_impl.cc                 |  4 +--
 .../aec_dump/capture_stream_info.cc           |  4 +--
 .../agc2/rnn_vad/sequence_buffer_unittest.cc  |  4 +--
 .../wavreader_factory.cc                      |  4 +--
 .../source/rtcp_packet/remote_estimate.cc     |  4 +--
 modules/rtp_rtcp/source/rtcp_receiver.cc      |  9 ++++---
 modules/rtp_rtcp/source/rtp_sender.cc         |  2 +-
 rtc_base/copy_on_write_buffer.h               |  6 +++--
 .../fuzzers/audio_processing_fuzzer_helper.cc |  2 +-
 16 files changed, 64 insertions(+), 49 deletions(-)

diff --git a/api/array_view.h b/api/array_view.h
index c4a487b59b..d03fbb0e76 100644
--- a/api/array_view.h
+++ b/api/array_view.h
@@ -156,7 +156,8 @@ class ArrayView final : public array_view_internal::ArrayViewBase<T, Size> {
   using const_reference = const value_type&;
   using pointer = value_type*;
   using const_pointer = const value_type*;
-  using const_iterator = const T*;
+  using iterator = typename std::span<T>::iterator;
+  using const_iterator = typename std::span<const T>::iterator;
 
   // Construct an ArrayView from a pointer and a length.
   template <typename U>
@@ -274,20 +275,28 @@ class ArrayView final : public array_view_internal::ArrayViewBase<T, Size> {
     RTC_HARDENING_ASSERT(this->data());
     return this->data()[idx];
   }
-  T* begin() const { return this->data(); }
-  T* end() const { return this->data() + this->size(); }
-  const T* cbegin() const { return this->data(); }
-  const T* cend() const { return this->data() + this->size(); }
-  std::reverse_iterator<T*> rbegin() const {
+  iterator begin() const {
+    return std::span<T>(this->data(), this->size()).begin();
+  }
+  iterator end() const {
+    return std::span<T>(this->data(), this->size()).end();
+  }
+  const_iterator cbegin() const {
+    return std::span<const T>(this->data(), this->size()).begin();
+  }
+  const_iterator cend() const {
+    return std::span<const T>(this->data(), this->size()).end();
+  }
+  std::reverse_iterator<iterator> rbegin() const {
     return std::make_reverse_iterator(end());
   }
-  std::reverse_iterator<T*> rend() const {
+  std::reverse_iterator<iterator> rend() const {
     return std::make_reverse_iterator(begin());
   }
-  std::reverse_iterator<const T*> crbegin() const {
+  std::reverse_iterator<const_iterator> crbegin() const {
     return std::make_reverse_iterator(cend());
   }
-  std::reverse_iterator<const T*> crend() const {
+  std::reverse_iterator<const_iterator> crend() const {
     return std::make_reverse_iterator(cbegin());
   }
 
diff --git a/api/array_view_unittest.cc b/api/array_view_unittest.cc
index c4b1385909..93cd508c9d 100644
--- a/api/array_view_unittest.cc
+++ b/api/array_view_unittest.cc
@@ -495,12 +495,12 @@ TEST(ArrayViewTest, TestReverseIterationVariable) {
   EXPECT_EQ('A', *(av.rend() - 1));
   EXPECT_EQ('A', *(av.crend() - 1));
 
-  const char* cit = av.cend() - 1;
+  auto cit = av.cend() - 1;
   for (auto crit = av.crbegin(); crit != av.crend(); ++crit, --cit) {
     EXPECT_EQ(*cit, *crit);
   }
 
-  char* it = av.end() - 1;
+  auto it = av.end() - 1;
   for (auto rit = av.rbegin(); rit != av.rend(); ++rit, --it) {
     EXPECT_EQ(*it, *rit);
   }
@@ -535,12 +535,12 @@ TEST(ArrayViewTest, TestReverseIterationFixed) {
   EXPECT_EQ('A', *(av.rend() - 1));
   EXPECT_EQ('A', *(av.crend() - 1));
 
-  const char* cit = av.cend() - 1;
+  auto cit = av.cend() - 1;
   for (auto crit = av.crbegin(); crit != av.crend(); ++crit, --cit) {
     EXPECT_EQ(*cit, *crit);
   }
 
-  char* it = av.end() - 1;
+  auto it = av.end() - 1;
   for (auto rit = av.rbegin(); rit != av.rend(); ++rit, --it) {
     EXPECT_EQ(*it, *rit);
   }
diff --git a/api/audio/audio_frame.cc b/api/audio/audio_frame.cc
index 04282394a4..b99d8304d3 100644
--- a/api/audio/audio_frame.cc
+++ b/api/audio/audio_frame.cc
@@ -126,7 +126,7 @@ void AudioFrame::CopyFrom(const AudioFrame& src) {
 }
 
 const int16_t* AudioFrame::data() const {
-  return muted_ ? zeroed_data().begin() : data_.data();
+  return muted_ ? zeroed_data().data() : data_.data();
 }
 
 InterleavedView<const int16_t> AudioFrame::data_view() const {
diff --git a/api/audio/audio_view.h b/api/audio/audio_view.h
index 7b7e1bd7af..163f03d7c2 100644
--- a/api/audio/audio_view.h
+++ b/api/audio/audio_view.h
@@ -112,14 +112,18 @@ class InterleavedView {
   }
 
   T& operator[](size_t idx) const { return data_[idx]; }
-  T* begin() const { return data_.begin(); }
-  T* end() const { return data_.end(); }
-  const T* cbegin() const { return data_.cbegin(); }
-  const T* cend() const { return data_.cend(); }
-  std::reverse_iterator<T*> rbegin() const { return data_.rbegin(); }
-  std::reverse_iterator<T*> rend() const { return data_.rend(); }
-  std::reverse_iterator<const T*> crbegin() const { return data_.crbegin(); }
-  std::reverse_iterator<const T*> crend() const { return data_.crend(); }
+  using iterator = typename ArrayView<T>::iterator;
+  using const_iterator = typename ArrayView<T>::const_iterator;
+  iterator begin() const { return data_.begin(); }
+  iterator end() const { return data_.end(); }
+  const_iterator cbegin() const { return data_.cbegin(); }
+  const_iterator cend() const { return data_.cend(); }
+  std::reverse_iterator<iterator> rbegin() const { return data_.rbegin(); }
+  std::reverse_iterator<iterator> rend() const { return data_.rend(); }
+  std::reverse_iterator<const_iterator> crbegin() const {
+    return data_.crbegin();
+  }
+  std::reverse_iterator<const_iterator> crend() const { return data_.crend(); }
 
  private:
   // TODO(tommi): Consider having these both be stored as uint16_t to
diff --git a/api/audio/test/audio_frame_unittest.cc b/api/audio/test/audio_frame_unittest.cc
index 9ee422eea1..a7c0166330 100644
--- a/api/audio/test/audio_frame_unittest.cc
+++ b/api/audio/test/audio_frame_unittest.cc
@@ -88,7 +88,7 @@ TEST(AudioFrameTest, UnmutedFrameIsInitiallyZeroed) {
 TEST(AudioFrameTest, MutedFrameBufferIsZeroed) {
   AudioFrame frame;
   int16_t* frame_data =
-      frame.mutable_data(kSamplesPerChannel, kNumChannelsMono).begin();
+      frame.mutable_data(kSamplesPerChannel, kNumChannelsMono).data().data();
   EXPECT_FALSE(frame.muted());
   // Fill the reserved buffer with non-zero data.
   for (size_t i = 0; i < frame.max_16bit_samples(); i++) {
diff --git a/modules/audio_device/include/test_audio_device.cc b/modules/audio_device/include/test_audio_device.cc
index 328874fe55..132128e00a 100644
--- a/modules/audio_device/include/test_audio_device.cc
+++ b/modules/audio_device/include/test_audio_device.cc
@@ -210,8 +210,8 @@ class BoundedWavFileWriter : public TestAudioDeviceModule::Renderer {
   bool Render(ArrayView<const int16_t> data) override {
     const int16_t kAmplitudeThreshold = 5;
 
-    const int16_t* begin = data.begin();
-    const int16_t* end = data.end();
+    const int16_t* begin = data.data();
+    const int16_t* end = data.data() + data.size();
     if (!started_writing_) {
       // Cut off silence at the beginning.
       while (begin < end) {
@@ -242,7 +242,7 @@ class BoundedWavFileWriter : public TestAudioDeviceModule::Renderer {
         wav_writer_.WriteSamples(begin, end - begin);
       }
       // Save the number of zeros we skipped in case this needs to be restored.
-      trailing_zeros_ += data.end() - end;
+      trailing_zeros_ += (data.data() + data.size()) - end;
     }
     return true;
   }
@@ -363,8 +363,8 @@ class RawFileWriter : public TestAudioDeviceModule::Renderer {
   bool Render(ArrayView<const int16_t> data) override {
     const int16_t kAmplitudeThreshold = 5;
 
-    const int16_t* begin = data.begin();
-    const int16_t* end = data.end();
+    const int16_t* begin = data.data();
+    const int16_t* end = data.data() + data.size();
     if (!started_writing_) {
       // Cut off silence at the beginning.
       while (begin < end) {
@@ -395,7 +395,7 @@ class RawFileWriter : public TestAudioDeviceModule::Renderer {
         WriteInt16(begin, end);
       }
       // Save the number of zeros we skipped in case this needs to be restored.
-      trailing_zeros_ += data.end() - end;
+      trailing_zeros_ += (data.data() + data.size()) - end;
     }
     return true;
   }
diff --git a/modules/audio_processing/aec3/reverb_decay_estimator.cc b/modules/audio_processing/aec3/reverb_decay_estimator.cc
index bd7c4251ed..dbedfc79e0 100644
--- a/modules/audio_processing/aec3/reverb_decay_estimator.cc
+++ b/modules/audio_processing/aec3/reverb_decay_estimator.cc
@@ -224,8 +224,7 @@ void ReverbDecayEstimator::EstimateDecay(ArrayView<const float> filter,
 }
 
 void ReverbDecayEstimator::AnalyzeFilter(ArrayView<const float> filter) {
-  auto h = ArrayView<const float>(
-      filter.begin() + block_to_analyze_ * kFftLengthBy2, kFftLengthBy2);
+  auto h = filter.subspan(block_to_analyze_ * kFftLengthBy2, kFftLengthBy2);
 
   // Compute squared filter coeffiecients for the block to analyze_;
   std::array<float, kFftLengthBy2> h2;
diff --git a/modules/audio_processing/aec_dump/aec_dump_impl.cc b/modules/audio_processing/aec_dump/aec_dump_impl.cc
index db3057c490..76de803b8f 100644
--- a/modules/audio_processing/aec_dump/aec_dump_impl.cc
+++ b/modules/audio_processing/aec_dump/aec_dump_impl.cc
@@ -169,7 +169,7 @@ void AecDumpImpl::WriteRenderStreamMessage(
 
   for (int i = 0; i < src.num_channels(); ++i) {
     const auto& channel_view = src.channel(i);
-    msg->add_channel(channel_view.begin(), sizeof(float) * channel_view.size());
+    msg->add_channel(channel_view.data(), sizeof(float) * channel_view.size());
   }
 
   PostWriteToFileTask(std::move(event));
@@ -185,7 +185,7 @@ void AecDumpImpl::WriteRenderStreamMessage(const float* const* data,
 
   for (int i = 0; i < num_channels; ++i) {
     MonoView<const float> channel_view(data[i], samples_per_channel);
-    msg->add_channel(channel_view.begin(), sizeof(float) * channel_view.size());
+    msg->add_channel(channel_view.data(), sizeof(float) * channel_view.size());
   }
 
   PostWriteToFileTask(std::move(event));
diff --git a/modules/audio_processing/aec_dump/capture_stream_info.cc b/modules/audio_processing/aec_dump/capture_stream_info.cc
index 993386658e..47933212c8 100644
--- a/modules/audio_processing/aec_dump/capture_stream_info.cc
+++ b/modules/audio_processing/aec_dump/capture_stream_info.cc
@@ -27,7 +27,7 @@ void CaptureStreamInfo::AddInput(const AudioFrameView<const float>& src) {
 
 void CaptureStreamInfo::AddInputChannel(MonoView<const float> channel) {
   auto* stream = event_->mutable_stream();
-  stream->add_input_channel(channel.begin(), sizeof(float) * channel.size());
+  stream->add_input_channel(channel.data(), sizeof(float) * channel.size());
 }
 
 void CaptureStreamInfo::AddOutput(const AudioFrameView<const float>& src) {
@@ -38,7 +38,7 @@ void CaptureStreamInfo::AddOutput(const AudioFrameView<const float>& src) {
 
 void CaptureStreamInfo::AddOutputChannel(MonoView<const float> channel) {
   auto* stream = event_->mutable_stream();
-  stream->add_output_channel(channel.begin(), sizeof(float) * channel.size());
+  stream->add_output_channel(channel.data(), sizeof(float) * channel.size());
 }
 
 void CaptureStreamInfo::AddInput(const int16_t* const data,
diff --git a/modules/audio_processing/agc2/rnn_vad/sequence_buffer_unittest.cc b/modules/audio_processing/agc2/rnn_vad/sequence_buffer_unittest.cc
index c43f99dc08..ff87e3bd9d 100644
--- a/modules/audio_processing/agc2/rnn_vad/sequence_buffer_unittest.cc
+++ b/modules/audio_processing/agc2/rnn_vad/sequence_buffer_unittest.cc
@@ -37,12 +37,12 @@ void TestSequenceBufferPushOp() {
     SCOPED_TRACE(i);
     seq_buf.Push(chunk);
     // Still in the buffer.
-    const auto* m = std::max_element(seq_buf_view.begin(), seq_buf_view.end());
+    auto m = std::max_element(seq_buf_view.begin(), seq_buf_view.end());
     EXPECT_EQ(1, *m);
   }
   // Gone after another push.
   seq_buf.Push(chunk);
-  const auto* m = std::max_element(seq_buf_view.begin(), seq_buf_view.end());
+  auto m = std::max_element(seq_buf_view.begin(), seq_buf_view.end());
   EXPECT_EQ(0, *m);
 
   // Check that the last item moves left by N positions after a push op.
diff --git a/modules/audio_processing/test/conversational_speech/wavreader_factory.cc b/modules/audio_processing/test/conversational_speech/wavreader_factory.cc
index 0a24f4f2e0..4cd7d096e2 100644
--- a/modules/audio_processing/test/conversational_speech/wavreader_factory.cc
+++ b/modules/audio_processing/test/conversational_speech/wavreader_factory.cc
@@ -32,11 +32,11 @@ class WavReaderAdaptor final : public WavReaderInterface {
   ~WavReaderAdaptor() override = default;
 
   size_t ReadFloatSamples(ArrayView<float> samples) override {
-    return wav_reader_.ReadSamples(samples.size(), samples.begin());
+    return wav_reader_.ReadSamples(samples.size(), samples.data());
   }
 
   size_t ReadInt16Samples(ArrayView<int16_t> samples) override {
-    return wav_reader_.ReadSamples(samples.size(), samples.begin());
+    return wav_reader_.ReadSamples(samples.size(), samples.data());
   }
 
   int SampleRate() const override { return wav_reader_.sample_rate(); }
diff --git a/modules/rtp_rtcp/source/rtcp_packet/remote_estimate.cc b/modules/rtp_rtcp/source/rtcp_packet/remote_estimate.cc
index 8855aa5dc9..708aafc973 100644
--- a/modules/rtp_rtcp/source/rtcp_packet/remote_estimate.cc
+++ b/modules/rtp_rtcp/source/rtcp_packet/remote_estimate.cc
@@ -100,8 +100,8 @@ class RemoteEstimateSerializerImpl : public RemoteEstimateSerializer {
     if (src.size() % kFieldSize != 0)
       return false;
     RTC_DCHECK_EQ(src.size() % kFieldSize, 0);
-    for (const uint8_t* data_ptr = src.data(); data_ptr < src.end();
-         data_ptr += kFieldSize) {
+    for (const uint8_t* data_ptr = src.data();
+         data_ptr < src.data() + src.size(); data_ptr += kFieldSize) {
       uint8_t field_id = ByteReader<uint8_t>::ReadBigEndian(data_ptr);
       for (const auto& field : fields_) {
         if (field.id() == field_id) {
diff --git a/modules/rtp_rtcp/source/rtcp_receiver.cc b/modules/rtp_rtcp/source/rtcp_receiver.cc
index 2bc9014a33..8f5146d61f 100644
--- a/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -363,12 +363,13 @@ bool RTCPReceiver::ParseCompoundPacket(ArrayView<const uint8_t> packet,
   // block.
   flat_map<uint32_t, RtcpReceivedBlock> received_blocks;
   bool valid = true;
-  for (const uint8_t* next_block = packet.begin();
-       valid && next_block != packet.end();
-       next_block = rtcp_block.NextPacket()) {
+  for (auto next_block = packet.begin(); valid && next_block != packet.end();
+       next_block =
+           packet.begin() + (rtcp_block.NextPacket() - packet.data())) {
     ptrdiff_t remaining_blocks_size = packet.end() - next_block;
     RTC_DCHECK_GT(remaining_blocks_size, 0);
-    if (!rtcp_block.Parse(next_block, remaining_blocks_size)) {
+    if (!rtcp_block.Parse(std::to_address(next_block),
+                          static_cast<size_t>(remaining_blocks_size))) {
       valid = false;
       break;
     }
diff --git a/modules/rtp_rtcp/source/rtp_sender.cc b/modules/rtp_rtcp/source/rtp_sender.cc
index ef9d0250f6..670b97d47e 100644
--- a/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/modules/rtp_rtcp/source/rtp_sender.cc
@@ -662,7 +662,7 @@ static void CopyHeaderAndExtensionsToRtxPacket(const RtpPacketToSend& packet,
       continue;
     }
 
-    std::memcpy(destination.begin(), source.begin(), destination.size());
+    std::memcpy(destination.data(), source.data(), destination.size());
   }
 }
 
diff --git a/rtc_base/copy_on_write_buffer.h b/rtc_base/copy_on_write_buffer.h
index 73aeb6d371..14554ed2e7 100644
--- a/rtc_base/copy_on_write_buffer.h
+++ b/rtc_base/copy_on_write_buffer.h
@@ -30,6 +30,8 @@ namespace webrtc {
 
 class RTC_EXPORT CopyOnWriteBuffer {
  public:
+  using const_iterator = ArrayView<const uint8_t>::iterator;
+
   // An empty buffer.
   CopyOnWriteBuffer();
   // Share the data with an existing buffer.
@@ -145,8 +147,8 @@ class RTC_EXPORT CopyOnWriteBuffer {
     return buffer_ ? buffer_->capacity() - offset_ : 0;
   }
 
-  const uint8_t* begin() const { return AsConstSpan().begin(); }
-  const uint8_t* end() const { return AsConstSpan().end(); }
+  const_iterator begin() const { return AsConstSpan().begin(); }
+  const_iterator end() const { return AsConstSpan().end(); }
 
   bool operator==(const CopyOnWriteBuffer& buf) const;
 
diff --git a/test/fuzzers/audio_processing_fuzzer_helper.cc b/test/fuzzers/audio_processing_fuzzer_helper.cc
index 6c716463c0..bfb82881c3 100644
--- a/test/fuzzers/audio_processing_fuzzer_helper.cc
+++ b/test/fuzzers/audio_processing_fuzzer_helper.cc
@@ -44,7 +44,7 @@ void GenerateFloatFrame(test::FuzzDataHelper* fuzz_data,
     if (fuzz_data->CanReadBytes(read_bytes)) {
       ArrayView<const uint8_t> byte_array =
           fuzz_data->ReadByteArray(read_bytes);
-      memmove(float_frames[i], byte_array.begin(), read_bytes);
+      memmove(float_frames[i], byte_array.data(), read_bytes);
     }
 
     // Sanitize input.

From 61fd3239f7f6e5f7946dd4c5ef0c7a225fdb0f3c Mon Sep 17 00:00:00 2001
From: Per K <perkj@webrtc.org>
Date: Wed, 18 Feb 2026 11:19:28 +0000
Subject: [PATCH 092/472] Add more detailed logging of ref window increase to
 analyzer
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Bug: webrtc:447037083
Change-Id: Idec75596c20eb01ff169e2526674603e3a5e6092
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449701
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46945}
---
 .../congestion_controller/scream/scream_v2.cc | 58 ++++++++++---------
 .../congestion_controller/scream/scream_v2.h  | 25 ++++++--
 .../rtc_event_log_visualizer/analyze_bwe.cc   | 26 ++++++++-
 .../log_scream_simulation.cc                  |  5 ++
 .../log_scream_simulation.h                   |  5 +-
 .../bwe_integration_tests/scream_test.cc      |  2 +-
 6 files changed, 85 insertions(+), 36 deletions(-)

diff --git a/modules/congestion_controller/scream/scream_v2.cc b/modules/congestion_controller/scream/scream_v2.cc
index cc45ca1887..ba647d340d 100644
--- a/modules/congestion_controller/scream/scream_v2.cc
+++ b/modules/congestion_controller/scream/scream_v2.cc
@@ -224,8 +224,7 @@ void ScreamV2::UpdateRefWindow(const TransportPacketsFeedback& msg) {
     // Just because there is a CE event, does not mean we send too much. At
     // rates close to the capacity, it is quite likely that one packet is CE
     // marked in every feedback.
-    DataSize increase =
-        DataUnitsAckedAndNotMarked(msg) * ref_window_mss_ratio();
+    double increase_scale_factor = ref_window_mss_ratio();
 
     // Limit increase for small RTTs
     if (delay_based_congestion_control_.rtt() + feedback_hold_time_ <
@@ -233,29 +232,32 @@ void ScreamV2::UpdateRefWindow(const TransportPacketsFeedback& msg) {
       double rtt_ratio =
           (delay_based_congestion_control_.rtt() + feedback_hold_time_) /
           params_.virtual_rtt.Get();
-      increase = increase * (rtt_ratio * rtt_ratio);
+      increase_scale_factor = increase_scale_factor * (rtt_ratio * rtt_ratio);
     }
 
     // Limit increase when close to the last inflection point.
-    increase = increase *
-               std::max(0.25, ref_window_scale_factor_close_to_ref_window_i());
+    increase_scale_factor =
+        increase_scale_factor *
+        std::max(0.25, ref_window_scale_factor_close_to_ref_window_i());
 
     // Limit increase when the reference window close to max segment size.
-    increase = increase * std::max(0.5, 1.0 - ref_window_mss_ratio());
+    increase_scale_factor =
+        increase_scale_factor * std::max(0.5, 1.0 - ref_window_mss_ratio());
 
     // Limit increase if L4S not enabled and queue delay is increased.
     if (l4s_alpha_ < 0.0001) {
-      increase =
-          increase * delay_based_congestion_control_
-                         .ref_window_scale_factor_due_to_increased_delay();
+      increase_scale_factor =
+          increase_scale_factor *
+          delay_based_congestion_control_
+              .ref_window_scale_factor_due_to_increased_delay();
     }
 
     // Put a additional restriction on reference window growth if rtt varies a
     // lot.
     // Better to enforce a slow increase in reference window and get
     // a more stable bitrate.
-    increase =
-        increase *
+    increase_scale_factor =
+        increase_scale_factor *
         std::max(0.1, delay_based_congestion_control_
                           .ref_window_scale_factor_due_to_delay_variation(
                               ref_window_mss_ratio()));
@@ -274,22 +276,14 @@ void ScreamV2::UpdateRefWindow(const TransportPacketsFeedback& msg) {
                   post_congestion_scale *
                   ref_window_scale_factor_close_to_ref_window_i();
     RTC_DCHECK_GE(multiplicative_scale, 1.0);
-    increase = increase * multiplicative_scale;
+    increase_scale_factor = increase_scale_factor * multiplicative_scale;
 
-    // Increase ref_window only if bytes in flight is large enough.
-    // Quite a lot of slack is allowed here to avoid that bitrate locks to low
-    // values. Increase is inhibited if max target bitrate is reached.
-    DataSize max_allowed_ref_window =
-        std::max(params_.max_segment_size.Get() +
-                     std::max(max_data_in_flight_this_rtt_,
-                              max_data_in_flight_prev_rtt_) *
-                         params_.bytes_in_flight_head_room.Get(),
-                 params_.min_ref_window.Get());
-
-    if (ref_window_ < max_allowed_ref_window) {
-      ref_window_ =
-          std::clamp(ref_window_ + increase, params_.min_ref_window.Get(),
-                     max_allowed_ref_window);
+    DataSize increase = DataUnitsAckedAndNotMarked(msg) * increase_scale_factor;
+    last_ref_window_increase_scale_factor_ = increase_scale_factor;
+    DataSize max_ref_window = max_allowed_ref_window();
+    if (ref_window_ < max_ref_window) {
+      ref_window_ = std::clamp(ref_window_ + increase,
+                               params_.min_ref_window.Get(), max_ref_window);
     }
   }
 
@@ -331,6 +325,18 @@ DataSize ScreamV2::max_data_in_flight() const {
   return ref_window_ * ref_window_overhead;
 }
 
+DataSize ScreamV2::max_allowed_ref_window() const {
+  // 4.2.2.2.
+  // Increase ref_window only if bytes in flight is large enough.
+  // Quite a lot of slack is allowed here to avoid that bitrate locks to low
+  // values.
+  return std::max(
+      params_.max_segment_size.Get() +
+          std::max(max_data_in_flight_this_rtt_, max_data_in_flight_prev_rtt_) *
+              params_.bytes_in_flight_head_room.Get(),
+      params_.min_ref_window.Get());
+}
+
 void ScreamV2::UpdateFeedbackHoldTime(const TransportPacketsFeedback& msg) {
   const TimeDelta feedback_hold_time = FeedbackHoldTime(msg);
   if (feedback_hold_time_.IsZero() &&
diff --git a/modules/congestion_controller/scream/scream_v2.h b/modules/congestion_controller/scream/scream_v2.h
index a8bc336165..28eb67276e 100644
--- a/modules/congestion_controller/scream/scream_v2.h
+++ b/modules/congestion_controller/scream/scream_v2.h
@@ -48,6 +48,7 @@ class ScreamV2 {
   DataRate pacing_rate() const {
     return target_rate_ * params_.pacing_factor.Get();
   }
+
   TimeDelta rtt() const { return delay_based_congestion_control_.rtt(); }
 
   // Max data in flight before the send window is full.
@@ -60,6 +61,10 @@ class ScreamV2 {
   // Last inflection point where ref_window started to decrease.
   DataSize ref_window_i() const { return ref_window_i_; }
 
+  // Returns the maximum allowed reference window based on data in flight during
+  // the last RTT.
+  DataSize max_allowed_ref_window() const;
+
   // Returns the average fraction of ECN-CE marked data units per RTT.
   double l4s_alpha() const { return l4s_alpha_; }
 
@@ -67,6 +72,10 @@ class ScreamV2 {
     return last_ref_window_decrease_time_;
   }
 
+  Timestamp last_reaction_to_congestion_time() const {
+    return last_reaction_to_congestion_time_;
+  }
+
   // Exposed for easier logging.
   const DelayBasedCongestionControl& delay_based_congestion_control() const {
     return delay_based_congestion_control_;
@@ -80,11 +89,9 @@ class ScreamV2 {
     return std::min(1.0, params_.max_segment_size.Get() / ref_window_);
   }
 
- private:
-  void UpdateL4SAlpha(const TransportPacketsFeedback& msg);
-  void UpdateRefWindow(const TransportPacketsFeedback& msg);
-  void UpdateFeedbackHoldTime(const TransportPacketsFeedback& msg);
-  void UpdateTargetRate(const TransportPacketsFeedback& msg);
+  double last_ref_window_increase_scale_factor() const {
+    return last_ref_window_increase_scale_factor_;
+  }
 
   // Scaling factor for reference window adjustment
   // when close to the last known inflection point.
@@ -106,6 +113,12 @@ class ScreamV2 {
                      params_.max_segment_size.Get();
   }
 
+ private:
+  void UpdateL4SAlpha(const TransportPacketsFeedback& msg);
+  void UpdateRefWindow(const TransportPacketsFeedback& msg);
+  void UpdateFeedbackHoldTime(const TransportPacketsFeedback& msg);
+  void UpdateTargetRate(const TransportPacketsFeedback& msg);
+
   const Environment env_;
   const ScreamV2Parameters params_;
 
@@ -124,6 +137,8 @@ class ScreamV2 {
   // since `ref_window_i_` was last set.
   bool allow_ref_window_i_update_ = true;
 
+  double last_ref_window_increase_scale_factor_ = 1.0;
+
   // `l4s_alpha_` tracks the average fraction of ECN-CE marked data units per
   // Round-Trip Time.
   double l4s_alpha_ = 0.0;
diff --git a/rtc_tools/rtc_event_log_visualizer/analyze_bwe.cc b/rtc_tools/rtc_event_log_visualizer/analyze_bwe.cc
index 0480256576..14fc0255e9 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyze_bwe.cc
+++ b/rtc_tools/rtc_event_log_visualizer/analyze_bwe.cc
@@ -680,6 +680,8 @@ void CreateScreamSimulationRefWindowGraph(const ParsedRtcEventLog& parsed_log,
   TimeSeries ref_window_series("RefWindow", LineStyle::kStep);
   TimeSeries ref_window_i_series("RefWindowI", LineStyle::kStep);
   TimeSeries max_data_in_flight("Max allowed data in flight", LineStyle::kStep);
+  TimeSeries max_allowed_ref_window_series("Max allowed ref window",
+                                           LineStyle::kStep);
   TimeSeries data_in_flight("Data in flight", LineStyle::kStep);
   IntervalSeries send_window_above_max_series(
       "Data in flight > Max allowed", "#ff8e82", IntervalSeries::kHorizontal);
@@ -700,7 +702,7 @@ void CreateScreamSimulationRefWindowGraph(const ParsedRtcEventLog& parsed_log,
   float send_window_state_switch =
       config.GetCallTimeSec(simulation.updates().front().time);
   SendWindowUsage send_window_usage = SendWindowUsage::kBelowRefWindow;
-
+  float last_time = config.CallBeginTimeSec();
   for (const LogScreamSimulation::State& state : simulation.updates()) {
     ref_window_series.points.emplace_back(config.GetCallTimeSec(state.time),
                                           state.ref_window.bytes());
@@ -708,8 +710,11 @@ void CreateScreamSimulationRefWindowGraph(const ParsedRtcEventLog& parsed_log,
                                             state.ref_window_i.bytes());
     max_data_in_flight.points.emplace_back(config.GetCallTimeSec(state.time),
                                            state.max_data_in_flight.bytes());
-    data_in_flight.points.emplace_back(config.GetCallTimeSec(state.time),
-                                       state.data_in_flight.bytes());
+    max_allowed_ref_window_series.points.emplace_back(
+        config.GetCallTimeSec(state.time),
+        state.max_allowed_ref_window.bytes());
+    // Plot the max data in flight before the feedback.
+    data_in_flight.points.emplace_back(last_time, state.data_in_flight.bytes());
     if (state.send_window_usage != send_window_usage) {
       last_series->intervals.emplace_back(send_window_state_switch,
                                           config.GetCallTimeSec(state.time));
@@ -727,12 +732,14 @@ void CreateScreamSimulationRefWindowGraph(const ParsedRtcEventLog& parsed_log,
           break;
       }
     }
+    last_time = config.GetCallTimeSec(state.time);
   }
   last_series->intervals.emplace_back(send_window_state_switch,
                                       config.CallEndTimeSec());
   plot->AppendTimeSeries(std::move(ref_window_series));
   plot->AppendTimeSeries(std::move(ref_window_i_series));
   plot->AppendTimeSeries(std::move(max_data_in_flight));
+  plot->AppendTimeSeries(std::move(max_allowed_ref_window_series));
   plot->AppendTimeSeries(std::move(data_in_flight));
   plot->AppendIntervalSeries(std::move(send_window_above_max_series));
   plot->AppendIntervalSeries(std::move(send_window_below_ref_window_series));
@@ -754,6 +761,10 @@ void CreateScreamSimulationRatiosGraph(const ParsedRtcEventLog& parsed_log,
       "RefWindowScaleFactorDueToIncreasedDelay", LineStyle::kStep);
   TimeSeries ref_window_scale_factor_due_to_delay_variation(
       "RefWindowScaleFactorDueToDelayVariation", LineStyle::kStep);
+  TimeSeries ref_window_scale_factor_close_to_ref_window_i(
+      "RefWindowScaleFactorCloseToRefWindowI", LineStyle::kStep);
+  TimeSeries ref_window_combined_increase_scale_factor(
+      "RefWindowCombinedIncreaseScaleFactor", LineStyle::kStep);
 
   LogScreamSimulation simulation({.rate_window = config.window_duration_},
                                  config.env_);
@@ -772,6 +783,12 @@ void CreateScreamSimulationRatiosGraph(const ParsedRtcEventLog& parsed_log,
     ref_window_scale_factor_due_to_delay_variation.points.emplace_back(
         config.GetCallTimeSec(state.time),
         state.ref_window_scale_factor_due_to_delay_variation);
+    ref_window_scale_factor_close_to_ref_window_i.points.emplace_back(
+        config.GetCallTimeSec(state.time),
+        state.ref_window_scale_factor_close_to_ref_window_i);
+    ref_window_combined_increase_scale_factor.points.emplace_back(
+        config.GetCallTimeSec(state.time),
+        state.ref_window_combined_increase_scale_factor);
   }
   plot->AppendTimeSeries(std::move(queue_delay_dev_norm_series));
   plot->AppendTimeSeries(std::move(l4s_alpha_series));
@@ -780,6 +797,9 @@ void CreateScreamSimulationRatiosGraph(const ParsedRtcEventLog& parsed_log,
       std::move(ref_window_scale_factor_due_to_increased_delay));
   plot->AppendTimeSeries(
       std::move(ref_window_scale_factor_due_to_delay_variation));
+  plot->AppendTimeSeries(
+      std::move(ref_window_scale_factor_close_to_ref_window_i));
+  plot->AppendTimeSeries(std::move(ref_window_combined_increase_scale_factor));
 
   plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
                  kLeftMargin, kRightMargin);
diff --git a/rtc_tools/rtc_event_log_visualizer/log_scream_simulation.cc b/rtc_tools/rtc_event_log_visualizer/log_scream_simulation.cc
index 2d07a59428..5e97896aa5 100644
--- a/rtc_tools/rtc_event_log_visualizer/log_scream_simulation.cc
+++ b/rtc_tools/rtc_event_log_visualizer/log_scream_simulation.cc
@@ -168,6 +168,7 @@ void LogScreamSimulation::LogState(const TransportPacketsFeedback& msg) {
           send_rate_tracker_.Rate(msg.feedback_time).value_or(DataRate::Zero()),
       .ref_window = scream_->ref_window(),
       .ref_window_i = scream_->ref_window_i(),
+      .max_allowed_ref_window = scream_->max_allowed_ref_window(),
       .max_data_in_flight = scream_->max_data_in_flight(),
       .data_in_flight = data_in_flight_,
       .send_window_usage = send_window_usage_,
@@ -182,6 +183,10 @@ void LogScreamSimulation::LogState(const TransportPacketsFeedback& msg) {
           scream_->delay_based_congestion_control()
               .ref_window_scale_factor_due_to_delay_variation(
                   scream_->ref_window_mss_ratio()),
+      .ref_window_scale_factor_close_to_ref_window_i =
+          scream_->ref_window_scale_factor_close_to_ref_window_i(),
+      .ref_window_combined_increase_scale_factor =
+          scream_->last_ref_window_increase_scale_factor(),
       .l4s_alpha = scream_->l4s_alpha(),
       .l4s_alpha_v = scream_->delay_based_congestion_control().l4s_alpha_v(),
   });
diff --git a/rtc_tools/rtc_event_log_visualizer/log_scream_simulation.h b/rtc_tools/rtc_event_log_visualizer/log_scream_simulation.h
index cc738981a6..9641f77eb7 100644
--- a/rtc_tools/rtc_event_log_visualizer/log_scream_simulation.h
+++ b/rtc_tools/rtc_event_log_visualizer/log_scream_simulation.h
@@ -47,6 +47,7 @@ class LogScreamSimulation {
 
     DataSize ref_window = DataSize::Zero();
     DataSize ref_window_i = DataSize::Zero();
+    DataSize max_allowed_ref_window = DataSize::Zero();
     DataSize max_data_in_flight = DataSize::Zero();
     // Data in flight after last packet was sent before the state was captured.
     DataSize data_in_flight = DataSize::Zero();
@@ -56,7 +57,9 @@ class LogScreamSimulation {
 
     double queue_delay_dev_norm = 0;
     double ref_window_scale_factor_due_to_increased_delay = 0.0;
-    double ref_window_scale_factor_due_to_delay_variation = 0;
+    double ref_window_scale_factor_due_to_delay_variation = 0.0;
+    double ref_window_scale_factor_close_to_ref_window_i = 0.0;
+    double ref_window_combined_increase_scale_factor = 0.0;
     double l4s_alpha = 0.0;
     double l4s_alpha_v = 0.0;
   };
diff --git a/test/peer_scenario/bwe_integration_tests/scream_test.cc b/test/peer_scenario/bwe_integration_tests/scream_test.cc
index 9adbf0b90f..5bc1df3110 100644
--- a/test/peer_scenario/bwe_integration_tests/scream_test.cc
+++ b/test/peer_scenario/bwe_integration_tests/scream_test.cc
@@ -479,7 +479,7 @@ TEST(ScreamTest, MaybeTest(LinkCapacity2MbpsRtt50msNoEcn)) {
 
   SendMediaTestResult result = SendMediaInOneDirection(std::move(params), s);
   EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
-                                              DataRate::KilobitsPerSec(1500),
+                                              DataRate::KilobitsPerSec(1300),
                                               DataRate::KilobitsPerSec(2300))));
 }
 

From 627bc15aa6cd2ca4c0ef5c02188a1885942516df Mon Sep 17 00:00:00 2001
From: Bjorn Terelius <terelius@webrtc.org>
Date: Wed, 18 Feb 2026 18:06:21 +0100
Subject: [PATCH 093/472] Pass field trials as FieldTrialView* to NetEqTest
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Bug: None
Change-Id: Ie4dca2e37464e1c747d778694b9a7b38a426b51e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/450261
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46946}
---
 modules/audio_coding/BUILD.gn                      |  2 ++
 modules/audio_coding/neteq/tools/neteq_test.cc     | 10 +++++-----
 modules/audio_coding/neteq/tools/neteq_test.h      |  4 ++--
 .../audio_coding/neteq/tools/neteq_test_factory.cc |  9 +++++----
 rtc_tools/BUILD.gn                                 |  1 +
 .../rtc_event_log_visualizer/analyze_audio.cc      |  5 +++--
 rtc_tools/rtc_event_log_visualizer/analyze_audio.h |  3 ++-
 rtc_tools/rtc_event_log_visualizer/main.cc         | 14 +++++++-------
 8 files changed, 27 insertions(+), 21 deletions(-)

diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index 1f94c30355..01a10e3dac 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -577,6 +577,7 @@ rtc_library("neteq_tools_minimal") {
     ":neteq",
     "../../api:array_view",
     "../../api:field_trials",
+    "../../api:field_trials_view",
     "../../api:neteq_simulator_api",
     "../../api:rtp_headers",
     "../../api:scoped_refptr",
@@ -1047,6 +1048,7 @@ if (rtc_include_tests) {
         ":neteq_test_tools",
         ":neteq_tools",
         ":neteq_tools_minimal",
+        "../../api:field_trials",
         "../../api:make_ref_counted",
         "../../api:rtp_headers",
         "../../api:scoped_refptr",
diff --git a/modules/audio_coding/neteq/tools/neteq_test.cc b/modules/audio_coding/neteq/tools/neteq_test.cc
index 57782b1826..bab757fae5 100644
--- a/modules/audio_coding/neteq/tools/neteq_test.cc
+++ b/modules/audio_coding/neteq/tools/neteq_test.cc
@@ -19,14 +19,13 @@
 #include <optional>
 #include <utility>
 
-#include "absl/strings/string_view.h"
 #include "api/array_view.h"
 #include "api/audio/audio_frame.h"
 #include "api/audio_codecs/audio_decoder_factory.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/environment/environment.h"
 #include "api/environment/environment_factory.h"
-#include "api/field_trials.h"
+#include "api/field_trials_view.h"
 #include "api/neteq/default_neteq_factory.h"
 #include "api/neteq/neteq.h"
 #include "api/neteq/neteq_factory.h"
@@ -91,11 +90,12 @@ NetEqTest::NetEqTest(const NetEq::Config& config,
                      std::unique_ptr<NetEqInput> input,
                      std::unique_ptr<AudioSink> output,
                      Callbacks callbacks,
-                     absl::string_view field_trials)
+                     const FieldTrialsView* field_trials)
     : input_(std::move(input)),
       clock_(Timestamp::Millis(input_->NextEventTime().value_or(0))),
-      env_(CreateEnvironment(&clock_,
-                             std::make_unique<FieldTrials>(field_trials))),
+      env_(CreateEnvironment(
+          &clock_,
+          field_trials ? field_trials->CreateCopy() : nullptr)),
       neteq_(
           neteq_factory
               ? neteq_factory->Create(env_, config, std::move(decoder_factory))
diff --git a/modules/audio_coding/neteq/tools/neteq_test.h b/modules/audio_coding/neteq/tools/neteq_test.h
index e0cf777fd1..0fddb2a25d 100644
--- a/modules/audio_coding/neteq/tools/neteq_test.h
+++ b/modules/audio_coding/neteq/tools/neteq_test.h
@@ -17,10 +17,10 @@
 #include <memory>
 #include <optional>
 
-#include "absl/strings/string_view.h"
 #include "api/audio_codecs/audio_decoder_factory.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/environment/environment.h"
+#include "api/field_trials_view.h"
 #include "api/neteq/neteq.h"
 #include "api/neteq/neteq_factory.h"
 #include "api/scoped_refptr.h"
@@ -92,7 +92,7 @@ class NetEqTest : public NetEqSimulator {
             std::unique_ptr<NetEqInput> input,
             std::unique_ptr<AudioSink> output,
             Callbacks callbacks,
-            absl::string_view field_trials = "");
+            const FieldTrialsView* field_trials = nullptr);
 
   ~NetEqTest() override;
 
diff --git a/modules/audio_coding/neteq/tools/neteq_test_factory.cc b/modules/audio_coding/neteq/tools/neteq_test_factory.cc
index 8b6f14a5e3..ced6030f4d 100644
--- a/modules/audio_coding/neteq/tools/neteq_test_factory.cc
+++ b/modules/audio_coding/neteq/tools/neteq_test_factory.cc
@@ -28,6 +28,7 @@
 #include "api/audio_codecs/audio_format.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "api/environment/environment.h"
+#include "api/field_trials.h"
 #include "api/make_ref_counted.h"
 #include "api/neteq/neteq.h"
 #include "api/neteq/neteq_factory.h"
@@ -347,10 +348,10 @@ std::unique_ptr<NetEqTest> NetEqTestFactory::InitializeTest(
   neteq_config.sample_rate_hz = *sample_rate_hz;
   neteq_config.max_packets_in_buffer = config.max_nr_packets_in_buffer;
   neteq_config.enable_fast_accelerate = config.enable_fast_accelerate;
-  return std::make_unique<NetEqTest>(neteq_config, decoder_factory, codecs,
-                                     std::move(text_log), factory,
-                                     std::move(input), std::move(output),
-                                     callbacks, config.field_trial_string);
+  FieldTrials field_trials_view(config.field_trial_string);
+  return std::make_unique<NetEqTest>(
+      neteq_config, decoder_factory, codecs, std::move(text_log), factory,
+      std::move(input), std::move(output), callbacks, &field_trials_view);
 }
 
 }  // namespace test
diff --git a/rtc_tools/BUILD.gn b/rtc_tools/BUILD.gn
index 0f4c54c9c4..f490928b83 100644
--- a/rtc_tools/BUILD.gn
+++ b/rtc_tools/BUILD.gn
@@ -313,6 +313,7 @@ if (!build_with_chromium) {
         "../api:candidate",
         "../api:dtls_transport_interface",
         "../api:field_trials",
+        "../api:field_trials_view",
         "../api:function_view",
         "../api:make_ref_counted",
         "../api:rtp_headers",
diff --git a/rtc_tools/rtc_event_log_visualizer/analyze_audio.cc b/rtc_tools/rtc_event_log_visualizer/analyze_audio.cc
index f2ec478315..c7a5242db0 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyze_audio.cc
+++ b/rtc_tools/rtc_event_log_visualizer/analyze_audio.cc
@@ -25,6 +25,7 @@
 #include "api/audio_codecs/audio_decoder_factory.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/environment/environment.h"
+#include "api/field_trials_view.h"
 #include "api/function_view.h"
 #include "api/make_ref_counted.h"
 #include "api/neteq/neteq.h"
@@ -246,7 +247,7 @@ std::unique_ptr<test::NetEqStatsGetter> CreateNetEqTestAndRun(
     uint32_t ssrc,
     absl::string_view replacement_file_name,
     int file_sample_rate_hz,
-    absl::string_view field_trials) {
+    const FieldTrialsView* field_trials) {
   std::unique_ptr<test::NetEqInput> input =
       test::CreateNetEqEventLogInput(parsed_log, ssrc);
   if (!input) {
@@ -291,7 +292,7 @@ NetEqStatsGetterMap SimulateNetEq(const ParsedRtcEventLog& parsed_log,
                                   const AnalyzerConfig& config,
                                   absl::string_view replacement_file_name,
                                   int file_sample_rate_hz,
-                                  absl::string_view field_trials) {
+                                  const FieldTrialsView* field_trials) {
   NetEqStatsGetterMap neteq_stats;
   for (uint32_t ssrc : parsed_log.incoming_audio_ssrcs()) {
     std::unique_ptr<test::NetEqStatsGetter> stats =
diff --git a/rtc_tools/rtc_event_log_visualizer/analyze_audio.h b/rtc_tools/rtc_event_log_visualizer/analyze_audio.h
index 55040ae39f..6b6ff7ed88 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyze_audio.h
+++ b/rtc_tools/rtc_event_log_visualizer/analyze_audio.h
@@ -17,6 +17,7 @@
 #include <string>
 
 #include "absl/strings/string_view.h"
+#include "api/field_trials_view.h"
 #include "api/function_view.h"
 #include "api/neteq/neteq.h"
 #include "logging/rtc_event_log/rtc_event_log_parser.h"
@@ -64,7 +65,7 @@ NetEqStatsGetterMap SimulateNetEq(const ParsedRtcEventLog& parsed_log,
                                   const AnalyzerConfig& config,
                                   absl::string_view replacement_file_name,
                                   int file_sample_rate_hz,
-                                  absl::string_view field_trials);
+                                  const FieldTrialsView* field_trials);
 
 void CreateAudioJitterBufferGraph(const ParsedRtcEventLog& parsed_log,
                                   const AnalyzerConfig& config,
diff --git a/rtc_tools/rtc_event_log_visualizer/main.cc b/rtc_tools/rtc_event_log_visualizer/main.cc
index a08588e8a1..1a775a8ee8 100644
--- a/rtc_tools/rtc_event_log_visualizer/main.cc
+++ b/rtc_tools/rtc_event_log_visualizer/main.cc
@@ -340,7 +340,7 @@ int main(int argc, char* argv[]) {
       plot_names.end()) {
     if (!neteq_stats) {
       neteq_stats = webrtc::SimulateNetEq(parsed_log, config, wav_path, 48000,
-                                          field_trials);
+                                          &env.field_trials());
     }
     webrtc::CreateNetEqNetworkStatsGraph(
         parsed_log, config, *neteq_stats,
@@ -353,7 +353,7 @@ int main(int argc, char* argv[]) {
       plot_names.end()) {
     if (!neteq_stats) {
       neteq_stats = webrtc::SimulateNetEq(parsed_log, config, wav_path, 48000,
-                                          field_trials);
+                                          &env.field_trials());
     }
     webrtc::CreateNetEqNetworkStatsGraph(
         parsed_log, config, *neteq_stats,
@@ -367,7 +367,7 @@ int main(int argc, char* argv[]) {
       plot_names.end()) {
     if (!neteq_stats) {
       neteq_stats = webrtc::SimulateNetEq(parsed_log, config, wav_path, 48000,
-                                          field_trials);
+                                          &env.field_trials());
     }
     webrtc::CreateNetEqNetworkStatsGraph(
         parsed_log, config, *neteq_stats,
@@ -381,7 +381,7 @@ int main(int argc, char* argv[]) {
       plot_names.end()) {
     if (!neteq_stats) {
       neteq_stats = webrtc::SimulateNetEq(parsed_log, config, wav_path, 48000,
-                                          field_trials);
+                                          &env.field_trials());
     }
     webrtc::CreateNetEqNetworkStatsGraph(
         parsed_log, config, *neteq_stats,
@@ -395,7 +395,7 @@ int main(int argc, char* argv[]) {
       plot_names.end()) {
     if (!neteq_stats) {
       neteq_stats = webrtc::SimulateNetEq(parsed_log, config, wav_path, 48000,
-                                          field_trials);
+                                          &env.field_trials());
     }
     webrtc::CreateNetEqLifetimeStatsGraph(
         parsed_log, config, *neteq_stats,
@@ -409,7 +409,7 @@ int main(int argc, char* argv[]) {
       plot_names.end()) {
     if (!neteq_stats) {
       neteq_stats = webrtc::SimulateNetEq(parsed_log, config, wav_path, 48000,
-                                          field_trials);
+                                          &env.field_trials());
     }
     webrtc::CreateNetEqNetworkStatsGraph(
         parsed_log, config, *neteq_stats,
@@ -429,7 +429,7 @@ int main(int argc, char* argv[]) {
       plot_names.end()) {
     if (!neteq_stats) {
       neteq_stats = webrtc::SimulateNetEq(parsed_log, config, wav_path, 48000,
-                                          field_trials);
+                                          &env.field_trials());
     }
     for (auto it = neteq_stats->cbegin(); it != neteq_stats->cend(); ++it) {
       webrtc::CreateAudioJitterBufferGraph(

From 754f9d3550d8f77599c7de0f3a350e63c9b4844c Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Wed, 18 Feb 2026 20:04:53 -0800
Subject: [PATCH 094/472] Update WebRTC code version (2026-02-19T04:04:25).

Bug: None
Change-Id: Ide24c401e574ed78b5139283c7dc83e28c4246af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/450400
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#46947}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index 96c187e869..1a32e3a4e1 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-02-18T04:08:18";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-02-19T04:04:25";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From 5459d3bd20d506128b3ae89309439ff39eb459bd Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Erik=20Spr=C3=A5ng?= <sprang@webrtc.org>
Date: Wed, 18 Feb 2026 14:23:00 +0100
Subject: [PATCH 095/472] Wire up usage on OnFrameDropped callback in VSE and
 VSS.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This CL makes sure the new frame drop signal is propagated to the right
places in VideoStreamEncoder and VideoSendStream.

Bug: webrtc:467444018
Change-Id: Ia42d2c099bf780d770936d4336a07a6f12ec727f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/450281
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46948}
---
 video/video_send_stream_impl.cc          |  6 +++
 video/video_send_stream_impl.h           |  3 ++
 video/video_send_stream_impl_unittest.cc | 21 ++++++++
 video/video_stream_encoder.cc            | 11 ++++
 video/video_stream_encoder.h             |  3 ++
 video/video_stream_encoder_unittest.cc   | 66 ++++++++++++++++++++----
 6 files changed, 100 insertions(+), 10 deletions(-)

diff --git a/video/video_send_stream_impl.cc b/video/video_send_stream_impl.cc
index d4e0c06190..b62da58183 100644
--- a/video/video_send_stream_impl.cc
+++ b/video/video_send_stream_impl.cc
@@ -907,6 +907,12 @@ void VideoSendStreamImpl::OnDroppedFrame(
   activity_ = true;
 }
 
+void VideoSendStreamImpl::OnFrameDropped(uint32_t rtp_timestamp,
+                                         int spatial_id,
+                                         bool is_end_of_temporal_unit) {
+  activity_ = true;
+}
+
 std::map<uint32_t, RtpState> VideoSendStreamImpl::GetRtpStates() const {
   return rtp_video_sender_->GetRtpStates();
 }
diff --git a/video/video_send_stream_impl.h b/video/video_send_stream_impl.h
index 9885bd83f1..c78994f799 100644
--- a/video/video_send_stream_impl.h
+++ b/video/video_send_stream_impl.h
@@ -186,6 +186,9 @@ class VideoSendStreamImpl : public webrtc::VideoSendStream,
 
   // Implements EncodedImageCallback.
   void OnDroppedFrame(EncodedImageCallback::DropReason reason) override;
+  void OnFrameDropped(uint32_t rtp_timestamp,
+                      int spatial_id,
+                      bool is_end_of_temporal_unit) override;
 
   // Starts monitoring and sends a keyframe.
   void StartupVideoSendStream();
diff --git a/video/video_send_stream_impl_unittest.cc b/video/video_send_stream_impl_unittest.cc
index ea150ebbb2..673da8d073 100644
--- a/video/video_send_stream_impl_unittest.cc
+++ b/video/video_send_stream_impl_unittest.cc
@@ -1119,6 +1119,27 @@ TEST_F(VideoSendStreamImplTest, KeepAliveOnDroppedFrame) {
   vss_impl->Stop();
 }
 
+TEST_F(VideoSendStreamImplTest, KeepAliveOnFrameDropped) {
+  auto vss_impl = CreateVideoSendStreamImpl(TestVideoEncoderConfig());
+  EXPECT_CALL(bitrate_allocator_, RemoveObserver(vss_impl.get())).Times(0);
+  vss_impl->Start();
+  const uint32_t kBitrateBps = 100000;
+  EXPECT_CALL(rtp_video_sender_, GetPayloadBitrateBps())
+      .Times(1)
+      .WillOnce(Return(kBitrateBps));
+  static_cast<BitrateAllocatorObserver*>(vss_impl.get())
+      ->OnBitrateUpdated(CreateAllocation(kBitrateBps));
+  encoder_queue_->PostTask([&] {
+    // Keep the stream from deallocating by dropping a frame.
+    static_cast<EncodedImageCallback*>(vss_impl.get())
+        ->OnFrameDropped(/*rtp_timestamp=*/0, /*spatial_id=*/0,
+                         /*is_end_of_temporal_unit=*/true);
+  });
+  time_controller_.AdvanceTime(TimeDelta::Seconds(2));
+  testing::Mock::VerifyAndClearExpectations(&bitrate_allocator_);
+  vss_impl->Stop();
+}
+
 TEST_F(VideoSendStreamImplTest, ConfiguresBitratesForSvc) {
   struct TestConfig {
     bool screenshare = false;
diff --git a/video/video_stream_encoder.cc b/video/video_stream_encoder.cc
index 004b70a87c..228e9fe2e9 100644
--- a/video/video_stream_encoder.cc
+++ b/video/video_stream_encoder.cc
@@ -2461,6 +2461,17 @@ void VideoStreamEncoder::OnDroppedFrame(DropReason reason) {
   });
 }
 
+void VideoStreamEncoder::OnFrameDropped(uint32_t rtp_timestamp,
+                                        int spatial_id,
+                                        bool is_end_of_temporal_unit) {
+  sink_->OnFrameDropped(rtp_timestamp, spatial_id, is_end_of_temporal_unit);
+  encoder_queue_->PostTask([this] {
+    RTC_DCHECK_RUN_ON(encoder_queue_.get());
+    stream_resource_manager_.OnFrameDropped(
+        VideoStreamEncoderObserver::DropReason::kEncoder);
+  });
+}
+
 DataRate VideoStreamEncoder::UpdateTargetBitrate(DataRate target_bitrate,
                                                  double cwnd_reduce_ratio) {
   RTC_DCHECK_RUN_ON(encoder_queue_.get());
diff --git a/video/video_stream_encoder.h b/video/video_stream_encoder.h
index cc6e541a2f..2e28008864 100644
--- a/video/video_stream_encoder.h
+++ b/video/video_stream_encoder.h
@@ -263,6 +263,9 @@ class VideoStreamEncoder : public VideoStreamEncoderInterface,
       const CodecSpecificInfo* codec_specific_info) override;
 
   void OnDroppedFrame(EncodedImageCallback::DropReason reason) override;
+  void OnFrameDropped(uint32_t rtp_timestamp,
+                      int spatial_id,
+                      bool is_end_of_temporal_unit) override;
 
   bool EncoderPaused() const;
   void TraceFrameDropStart();
diff --git a/video/video_stream_encoder_unittest.cc b/video/video_stream_encoder_unittest.cc
index adfcbb5c26..7cb806c03d 100644
--- a/video/video_stream_encoder_unittest.cc
+++ b/video/video_stream_encoder_unittest.cc
@@ -1612,7 +1612,19 @@ class VideoStreamEncoderTest : public ::testing::Test {
       last_frame_instrumentation_data_.reset();
     }
 
+    int number_of_dropped_frames() const {
+      MutexLock lock(&mutex_);
+      return number_of_dropped_frames_;
+    }
+
    private:
+    void OnFrameDropped(uint32_t /*rtp_timestamp*/,
+                        int /*spatial_id*/,
+                        bool /*is_end_of_temporal_unit*/) override {
+      MutexLock lock(&mutex_);
+      ++number_of_dropped_frames_;
+    }
+
     Result OnEncodedImage(
         const EncodedImage& encoded_image,
         const CodecSpecificInfo* codec_specific_info) override {
@@ -1700,9 +1712,10 @@ class VideoStreamEncoderTest : public ::testing::Test {
     VideoBitrateAllocation last_bitrate_allocation_ RTC_GUARDED_BY(&mutex_);
     int number_of_bitrate_allocations_ RTC_GUARDED_BY(&mutex_) = 0;
     VideoLayersAllocation last_layers_allocation_ RTC_GUARDED_BY(&mutex_);
-    int number_of_layers_allocations_ RTC_GUARDED_BY(&mutex_) = 0;
+    int number_of_layers_allocations_ RTC_GUARDED_BY(mutex_) = 0;
+    int number_of_dropped_frames_ RTC_GUARDED_BY(mutex_) = 0;
     std::optional<FrameInstrumentationData> last_frame_instrumentation_data_
-        RTC_GUARDED_BY(&mutex_);
+        RTC_GUARDED_BY(mutex_);
   };
 
   class VideoBitrateAllocatorProxyFactory
@@ -8086,15 +8099,48 @@ TEST_F(VideoStreamEncoderTest, AccumulatesUpdateRectOnDroppedFrames) {
   EXPECT_EQ(rect.width, 10);
   EXPECT_EQ(rect.height, 1);
 
-  source.IncomingCapturedFrame(CreateFrameWithUpdatedPixel(4, nullptr, 0));
-  WaitForEncodedFrame(4);
-  // Previous frame was encoded, so no accumulation should happen.
-  rect = fake_encoder_.GetLastUpdateRect();
-  EXPECT_EQ(rect.offset_x, 0);
-  EXPECT_EQ(rect.offset_y, 0);
-  EXPECT_EQ(rect.width, 1);
-  EXPECT_EQ(rect.height, 1);
+  video_stream_encoder_->Stop();
+}
 
+TEST_F(VideoStreamEncoderTest, ForwardsOnFrameDroppedToSink) {
+  NiceMock<MockVideoEncoder> video_encoder;
+  test::VideoEncoderProxyFactory encoder_factory(&video_encoder);
+  video_send_config_.encoder_settings.encoder_factory = &encoder_factory;
+
+  EncodedImageCallback* callback = nullptr;
+  ON_CALL(video_encoder, RegisterEncodeCompleteCallback)
+      .WillByDefault([&](EncodedImageCallback* cb) {
+        callback = cb;
+        return WEBRTC_VIDEO_CODEC_OK;
+      });
+
+  ON_CALL(video_encoder, Encode)
+      .WillByDefault(
+          [&](const VideoFrame&, const std::vector<VideoFrameType>*) {
+            EXPECT_TRUE(callback != nullptr);
+            if (callback) {
+              callback->OnFrameDropped(/*rtp_timestamp=*/0, /*spatial_id=*/0,
+                                       /*is_end_of_temporal_unit=*/true);
+            }
+            return WEBRTC_VIDEO_CODEC_OK;
+          });
+
+  // Reset encoder for new configuration to take effect.
+  ConfigureEncoder(video_encoder_config_.Copy());
+
+  video_stream_encoder_->OnBitrateUpdatedAndWaitForManagedResources(
+      kTargetBitrate, kTargetBitrate, 0, 0, 0);
+
+  // Trigger Encode by sending a frame.
+  video_source_.IncomingCapturedFrame(CreateFrame(1, 1280, 720));
+
+  // Wait for the encoder queue to process the drop.
+  AdvanceTime(TimeDelta::Millis(1));
+
+  EXPECT_EQ(sink_.number_of_dropped_frames(), 1);
+
+  video_stream_encoder_->SetSource(
+      nullptr, webrtc::DegradationPreference::MAINTAIN_FRAMERATE);
   video_stream_encoder_->Stop();
 }
 

From abec0c51cc761fba91e27ccd6294ccd65fbab84e Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Thu, 19 Feb 2026 08:12:16 +0000
Subject: [PATCH 096/472] Check tag length before comparing tags

Bug: None
Change-Id: I8403b5aaa05d623e3df3b21678142457e3f1077f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/450460
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46949}
---
 media/base/rtp_utils.cc | 2 ++
 1 file changed, 2 insertions(+)

diff --git a/media/base/rtp_utils.cc b/media/base/rtp_utils.cc
index f7cb5322c8..a6448d762c 100644
--- a/media/base/rtp_utils.cc
+++ b/media/base/rtp_utils.cc
@@ -92,6 +92,8 @@ void UpdateRtpAuthTag(ArrayView<uint8_t> rtp,
   uint8_t* auth_tag = rtp.data() + (rtp.size() - tag_length);
 
   // We should have a fake HMAC value @ auth_tag.
+  // Tag length must be no bigger than kFakeAuthTag size (currently 10).
+  RTC_DCHECK_LE(tag_length, sizeof(kFakeAuthTag));
   RTC_DCHECK_EQ(0, memcmp(auth_tag, kFakeAuthTag, tag_length));
 
   // Copy ROC after end of rtp packet.

From bf4fd8bccf85db2abc25cd92b2749ce7a4e8d893 Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Wed, 18 Feb 2026 16:31:47 +0100
Subject: [PATCH 097/472] Deprecated accessors returning
 ArrayView::const_iterator

Until c++23 std::span doesn't have such accessors.
Within WebRTC these accessors are used only for ArrayView<const T> where
non-const iterator also can't modify data.

Bug: webrtc:439801349
Change-Id: I1e361e99e3477af467b8e8368f51e935dad5e31c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/450000
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46950}
---
 api/array_view.h                              | 18 +++++----------
 api/array_view_unittest.cc                    | 22 -------------------
 api/audio/audio_view.h                        | 12 +++++-----
 .../rtp/corruption_detection_message.cc       |  2 +-
 .../fixed_length_encoding_parameters_v3.cc    |  2 +-
 .../codecs/cng/audio_encoder_cng.cc           |  2 +-
 .../audio_encoder_multi_channel_opus_impl.cc  |  2 +-
 .../codecs/opus/audio_encoder_opus.cc         |  2 +-
 .../default_output_rate_calculator.cc         |  2 +-
 .../audio_processing/aec3/echo_audibility.cc  |  2 +-
 .../neural_feature_extractor.cc               | 12 +++++-----
 .../agc2/rnn_vad/lp_residual.cc               |  7 +++---
 .../transport_feedback_adapter_unittest.cc    |  2 +-
 .../source/corruption_detection_extension.cc  |  4 ++--
 .../source/rtp_packetizer_av1_unittest.cc     |  2 +-
 15 files changed, 32 insertions(+), 61 deletions(-)

diff --git a/api/array_view.h b/api/array_view.h
index d03fbb0e76..96960d258a 100644
--- a/api/array_view.h
+++ b/api/array_view.h
@@ -281,24 +281,18 @@ class ArrayView final : public array_view_internal::ArrayViewBase<T, Size> {
   iterator end() const {
     return std::span<T>(this->data(), this->size()).end();
   }
-  const_iterator cbegin() const {
-    return std::span<const T>(this->data(), this->size()).begin();
-  }
-  const_iterator cend() const {
-    return std::span<const T>(this->data(), this->size()).end();
-  }
+  ABSL_DEPRECATE_AND_INLINE() const_iterator cbegin() const { return begin(); }
+  ABSL_DEPRECATE_AND_INLINE() const_iterator cend() const { return end(); }
   std::reverse_iterator<iterator> rbegin() const {
     return std::make_reverse_iterator(end());
   }
   std::reverse_iterator<iterator> rend() const {
     return std::make_reverse_iterator(begin());
   }
-  std::reverse_iterator<const_iterator> crbegin() const {
-    return std::make_reverse_iterator(cend());
-  }
-  std::reverse_iterator<const_iterator> crend() const {
-    return std::make_reverse_iterator(cbegin());
-  }
+  ABSL_DEPRECATE_AND_INLINE()
+  std::reverse_iterator<const_iterator> crbegin() const { return rbegin(); }
+  ABSL_DEPRECATE_AND_INLINE()
+  std::reverse_iterator<const_iterator> crend() const { return rend(); }
 
   constexpr ArrayView<T> subspan(size_t offset,
                                  size_t count = std::dynamic_extent) const {
diff --git a/api/array_view_unittest.cc b/api/array_view_unittest.cc
index 93cd508c9d..d4fa00b3aa 100644
--- a/api/array_view_unittest.cc
+++ b/api/array_view_unittest.cc
@@ -436,7 +436,6 @@ TEST(ArrayViewTest, TestIterationEmpty) {
   // Variable-size.
   ArrayView<std::vector<std::vector<std::vector<std::string>>>> av;
   EXPECT_EQ(av.begin(), av.end());
-  EXPECT_EQ(av.cbegin(), av.cend());
   for (auto& e : av) {
     EXPECT_TRUE(false);
     EXPECT_EQ(42u, e.size());  // Dummy use of e to prevent unused var warning.
@@ -445,7 +444,6 @@ TEST(ArrayViewTest, TestIterationEmpty) {
   // Fixed-size.
   ArrayView<std::vector<std::vector<std::vector<std::string>>>, 0> af;
   EXPECT_EQ(af.begin(), af.end());
-  EXPECT_EQ(af.cbegin(), af.cend());
   for (auto& e : af) {
     EXPECT_TRUE(false);
     EXPECT_EQ(42u, e.size());  // Dummy use of e to prevent unused var warning.
@@ -456,13 +454,11 @@ TEST(ArrayViewTest, TestReverseIterationEmpty) {
   // Variable-size.
   ArrayView<std::vector<std::vector<std::vector<std::string>>>> av;
   EXPECT_EQ(av.rbegin(), av.rend());
-  EXPECT_EQ(av.crbegin(), av.crend());
   EXPECT_TRUE(av.empty());
 
   // Fixed-size.
   ArrayView<std::vector<std::vector<std::vector<std::string>>>, 0> af;
   EXPECT_EQ(af.begin(), af.end());
-  EXPECT_EQ(af.cbegin(), af.cend());
   EXPECT_TRUE(af.empty());
 }
 
@@ -470,9 +466,7 @@ TEST(ArrayViewTest, TestIterationVariable) {
   char arr[] = "Arrr!";
   ArrayView<char> av(arr);
   EXPECT_EQ('A', *av.begin());
-  EXPECT_EQ('A', *av.cbegin());
   EXPECT_EQ('\0', *(av.end() - 1));
-  EXPECT_EQ('\0', *(av.cend() - 1));
   char i = 0;
   for (auto& e : av) {
     EXPECT_EQ(arr + i, &e);
@@ -491,14 +485,7 @@ TEST(ArrayViewTest, TestReverseIterationVariable) {
   char arr[] = "Arrr!";
   ArrayView<char> av(arr);
   EXPECT_EQ('\0', *av.rbegin());
-  EXPECT_EQ('\0', *av.crbegin());
   EXPECT_EQ('A', *(av.rend() - 1));
-  EXPECT_EQ('A', *(av.crend() - 1));
-
-  auto cit = av.cend() - 1;
-  for (auto crit = av.crbegin(); crit != av.crend(); ++crit, --cit) {
-    EXPECT_EQ(*cit, *crit);
-  }
 
   auto it = av.end() - 1;
   for (auto rit = av.rbegin(); rit != av.rend(); ++rit, --it) {
@@ -510,9 +497,7 @@ TEST(ArrayViewTest, TestIterationFixed) {
   char arr[] = "Arrr!";
   ArrayView<char, 6> av(arr);
   EXPECT_EQ('A', *av.begin());
-  EXPECT_EQ('A', *av.cbegin());
   EXPECT_EQ('\0', *(av.end() - 1));
-  EXPECT_EQ('\0', *(av.cend() - 1));
   char i = 0;
   for (auto& e : av) {
     EXPECT_EQ(arr + i, &e);
@@ -531,14 +516,7 @@ TEST(ArrayViewTest, TestReverseIterationFixed) {
   char arr[] = "Arrr!";
   ArrayView<char, 6> av(arr);
   EXPECT_EQ('\0', *av.rbegin());
-  EXPECT_EQ('\0', *av.crbegin());
   EXPECT_EQ('A', *(av.rend() - 1));
-  EXPECT_EQ('A', *(av.crend() - 1));
-
-  auto cit = av.cend() - 1;
-  for (auto crit = av.crbegin(); crit != av.crend(); ++crit, --cit) {
-    EXPECT_EQ(*cit, *crit);
-  }
 
   auto it = av.end() - 1;
   for (auto rit = av.rbegin(); rit != av.rend(); ++rit, --it) {
diff --git a/api/audio/audio_view.h b/api/audio/audio_view.h
index 163f03d7c2..c91e11fbf6 100644
--- a/api/audio/audio_view.h
+++ b/api/audio/audio_view.h
@@ -59,6 +59,8 @@ template <typename T>
 class InterleavedView {
  public:
   using value_type = T;
+  using iterator = typename ArrayView<T>::iterator;
+  using const_iterator = typename ArrayView<const T>::iterator;
 
   InterleavedView() = default;
 
@@ -112,18 +114,16 @@ class InterleavedView {
   }
 
   T& operator[](size_t idx) const { return data_[idx]; }
-  using iterator = typename ArrayView<T>::iterator;
-  using const_iterator = typename ArrayView<T>::const_iterator;
   iterator begin() const { return data_.begin(); }
   iterator end() const { return data_.end(); }
-  const_iterator cbegin() const { return data_.cbegin(); }
-  const_iterator cend() const { return data_.cend(); }
+  const_iterator cbegin() const { return data_.begin(); }
+  const_iterator cend() const { return data_.end(); }
   std::reverse_iterator<iterator> rbegin() const { return data_.rbegin(); }
   std::reverse_iterator<iterator> rend() const { return data_.rend(); }
   std::reverse_iterator<const_iterator> crbegin() const {
-    return data_.crbegin();
+    return data_.rbegin();
   }
-  std::reverse_iterator<const_iterator> crend() const { return data_.crend(); }
+  std::reverse_iterator<const_iterator> crend() const { return data_.rend(); }
 
  private:
   // TODO(tommi): Consider having these both be stored as uint16_t to
diff --git a/api/transport/rtp/corruption_detection_message.cc b/api/transport/rtp/corruption_detection_message.cc
index d812f9ab16..68e1ca66ec 100644
--- a/api/transport/rtp/corruption_detection_message.cc
+++ b/api/transport/rtp/corruption_detection_message.cc
@@ -117,7 +117,7 @@ CorruptionDetectionMessage::Builder::WithChromaErrorThreshold(
 CorruptionDetectionMessage::Builder&
 CorruptionDetectionMessage::Builder::WithSampleValues(
     const ArrayView<const double>& sample_values) {
-  message_.sample_values_.assign(sample_values.cbegin(), sample_values.cend());
+  message_.sample_values_.assign(sample_values.begin(), sample_values.end());
   return *this;
 }
 
diff --git a/logging/rtc_event_log/events/fixed_length_encoding_parameters_v3.cc b/logging/rtc_event_log/events/fixed_length_encoding_parameters_v3.cc
index fa3fadde9b..e2ef9b9425 100644
--- a/logging/rtc_event_log/events/fixed_length_encoding_parameters_v3.cc
+++ b/logging/rtc_event_log/events/fixed_length_encoding_parameters_v3.cc
@@ -34,7 +34,7 @@ FixedLengthEncodingParametersV3::CalculateParameters(
     bool values_optional) {
   // As a special case, if all of the elements are identical to the base
   // we just encode the base value with a special delta header.
-  if (std::all_of(values.cbegin(), values.cend(),
+  if (std::all_of(values.begin(), values.end(),
                   [base](uint64_t val) { return val == base; })) {
     // Delta header with signed=true and delta_bitwidth=64
     return FixedLengthEncodingParametersV3(/*delta_bit_width=*/64,
diff --git a/modules/audio_coding/codecs/cng/audio_encoder_cng.cc b/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
index e93887606f..fa7006fa75 100644
--- a/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
+++ b/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
@@ -132,7 +132,7 @@ AudioEncoder::EncodedInfo AudioEncoderCng::EncodeImpl(
                rtp_timestamps_.size() * samples_per_10ms_frame);
   rtp_timestamps_.push_back(rtp_timestamp);
   RTC_DCHECK_EQ(samples_per_10ms_frame, audio.size());
-  speech_buffer_.insert(speech_buffer_.end(), audio.cbegin(), audio.cend());
+  speech_buffer_.insert(speech_buffer_.end(), audio.begin(), audio.end());
   const size_t frames_to_encode = speech_encoder_->Num10MsFramesInNextPacket();
   if (rtp_timestamps_.size() < frames_to_encode) {
     return EncodedInfo();
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc b/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc
index d5188f51a2..be55406ecb 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc
@@ -344,7 +344,7 @@ AudioEncoder::EncodedInfo AudioEncoderMultiChannelOpusImpl::EncodeImpl(
   if (input_buffer_.empty())
     first_timestamp_in_buffer_ = rtp_timestamp;
 
-  input_buffer_.insert(input_buffer_.end(), audio.cbegin(), audio.cend());
+  input_buffer_.insert(input_buffer_.end(), audio.begin(), audio.end());
   if (input_buffer_.size() <
       (Num10msFramesPerPacket() * SamplesPer10msFrame())) {
     return EncodedInfo();
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index ff994fba57..a8e1bff186 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -590,7 +590,7 @@ AudioEncoder::EncodedInfo AudioEncoderOpusImpl::EncodeImpl(
   if (input_buffer_.empty())
     first_timestamp_in_buffer_ = rtp_timestamp;
 
-  input_buffer_.insert(input_buffer_.end(), audio.cbegin(), audio.cend());
+  input_buffer_.insert(input_buffer_.end(), audio.begin(), audio.end());
   if (input_buffer_.size() <
       (Num10msFramesPerPacket() * SamplesPer10msFrame())) {
     return EncodedInfo();
diff --git a/modules/audio_mixer/default_output_rate_calculator.cc b/modules/audio_mixer/default_output_rate_calculator.cc
index 6885875cae..be9819c027 100644
--- a/modules/audio_mixer/default_output_rate_calculator.cc
+++ b/modules/audio_mixer/default_output_rate_calculator.cc
@@ -26,7 +26,7 @@ int DefaultOutputRateCalculator::CalculateOutputRateFromRange(
   }
   using NativeRate = AudioProcessing::NativeRate;
   const int maximal_frequency = *std::max_element(
-      preferred_sample_rates.cbegin(), preferred_sample_rates.cend());
+      preferred_sample_rates.begin(), preferred_sample_rates.end());
 
   RTC_DCHECK_LE(NativeRate::kSampleRate8kHz, maximal_frequency);
   RTC_DCHECK_GE(NativeRate::kSampleRate48kHz, maximal_frequency);
diff --git a/modules/audio_processing/aec3/echo_audibility.cc b/modules/audio_processing/aec3/echo_audibility.cc
index 6853f51126..b7a9430290 100644
--- a/modules/audio_processing/aec3/echo_audibility.cc
+++ b/modules/audio_processing/aec3/echo_audibility.cc
@@ -103,7 +103,7 @@ bool EchoAudibility::IsRenderTooLow(const BlockBuffer& block_buffer) {
       for (int ch = 0; ch < num_render_channels; ++ch) {
         ArrayView<const float, kBlockSize> block =
             block_buffer.buffer[idx].View(/*band=*/0, /*channel=*/ch);
-        auto r = std::minmax_element(block.cbegin(), block.cend());
+        auto r = std::minmax_element(block.begin(), block.end());
         float max_abs_channel =
             std::max(std::fabs(*r.first), std::fabs(*r.second));
         max_abs_over_channels =
diff --git a/modules/audio_processing/aec3/neural_residual_echo_estimator/neural_feature_extractor.cc b/modules/audio_processing/aec3/neural_residual_echo_estimator/neural_feature_extractor.cc
index daf7ab71d6..45c08e1c70 100644
--- a/modules/audio_processing/aec3/neural_residual_echo_estimator/neural_feature_extractor.cc
+++ b/modules/audio_processing/aec3/neural_residual_echo_estimator/neural_feature_extractor.cc
@@ -101,8 +101,8 @@ void TimeDomainFeatureExtractor::UpdateBuffers(
   std::vector<float>& input_buffer =
       input_buffer_[static_cast<size_t>(input_type)];
   std::array<float, kBlockSize> summed_block = AverageAllChannels(all_channels);
-  input_buffer.insert(input_buffer.end(), summed_block.cbegin(),
-                      summed_block.cend());
+  input_buffer.insert(input_buffer.end(), summed_block.begin(),
+                      summed_block.end());
 }
 
 void TimeDomainFeatureExtractor::PrepareModelInput(ArrayView<float> model_input,
@@ -113,9 +113,9 @@ void TimeDomainFeatureExtractor::PrepareModelInput(ArrayView<float> model_input,
   std::vector<float>& input_buffer =
       input_buffer_[static_cast<size_t>(input_type)];
   RTC_CHECK_EQ(input_buffer.size(), step_size_);
-  std::copy(model_input.cbegin() + step_size_, model_input.cend(),
+  std::copy(model_input.begin() + step_size_, model_input.end(),
             model_input.begin());
-  std::copy(input_buffer.cbegin(), input_buffer.cend(),
+  std::copy(input_buffer.begin(), input_buffer.end(),
             model_input.end() - step_size_);
   input_buffer.clear();
 }
@@ -213,8 +213,8 @@ void FrequencyDomainFeatureExtractor::UpdateBuffers(
   for (size_t ch = 0; ch < all_channels.size(); ++ch) {
     const ArrayView<const float, kBlockSize>& frame_in = all_channels[ch];
     std::vector<float>& input_buffer_ch = input_buffer[ch];
-    input_buffer_ch.insert(input_buffer_ch.end(), frame_in.cbegin(),
-                           frame_in.cend());
+    input_buffer_ch.insert(input_buffer_ch.end(), frame_in.begin(),
+                           frame_in.end());
   }
 }
 void FrequencyDomainFeatureExtractor::PrepareModelInput(
diff --git a/modules/audio_processing/agc2/rnn_vad/lp_residual.cc b/modules/audio_processing/agc2/rnn_vad/lp_residual.cc
index 6568f93e4d..46cb1eb299 100644
--- a/modules/audio_processing/agc2/rnn_vad/lp_residual.cc
+++ b/modules/audio_processing/agc2/rnn_vad/lp_residual.cc
@@ -124,14 +124,13 @@ void ComputeLpResidual(ArrayView<const float, kNumLpcCoefficients> lpc_coeffs,
   // Edge case: i < kNumLpcCoefficients.
   y[0] = x[0];
   for (int i = 1; i < kNumLpcCoefficients; ++i) {
-    y[i] =
-        std::inner_product(x.crend() - i, x.crend(), lpc_coeffs.cbegin(), x[i]);
+    y[i] = std::inner_product(x.rend() - i, x.rend(), lpc_coeffs.begin(), x[i]);
   }
   // Regular case.
-  auto last = x.crend();
+  auto last = x.rend();
   for (int i = kNumLpcCoefficients; SafeLt(i, y.size()); ++i, --last) {
     y[i] = std::inner_product(last - kNumLpcCoefficients, last,
-                              lpc_coeffs.cbegin(), x[i]);
+                              lpc_coeffs.begin(), x[i]);
   }
 }
 
diff --git a/modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc b/modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc
index 52bdd43f1c..1abf22f5f4 100644
--- a/modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc
+++ b/modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc
@@ -163,7 +163,7 @@ rtcp::CongestionControlFeedback BuildRtcpCongestionControlFeedbackPacket(
     ArrayView<const PacketTemplate> packets) {
   // Assume the feedback was sent when the last packet was received.
   Timestamp feedback_sent_time = Timestamp::MinusInfinity();
-  for (auto it = packets.crbegin(); it != packets.crend(); ++it) {
+  for (auto it = packets.rbegin(); it != packets.rend(); ++it) {
     if (it->receive_timestamp.IsFinite()) {
       feedback_sent_time = it->receive_timestamp;
       break;
diff --git a/modules/rtp_rtcp/source/corruption_detection_extension.cc b/modules/rtp_rtcp/source/corruption_detection_extension.cc
index 2b49f4b782..7c173a2b67 100644
--- a/modules/rtp_rtcp/source/corruption_detection_extension.cc
+++ b/modules/rtp_rtcp/source/corruption_detection_extension.cc
@@ -49,8 +49,8 @@ bool CorruptionDetectionExtension::Parse(ArrayView<const uint8_t> data,
   uint8_t channel_error_thresholds = data[2];
   message->luma_error_threshold_ = channel_error_thresholds >> 4;
   message->chroma_error_threshold_ = channel_error_thresholds & 0xF;
-  message->sample_values_.assign(data.cbegin() + kConfigurationBytes,
-                                 data.cend());
+  message->sample_values_.assign(data.begin() + kConfigurationBytes,
+                                 data.end());
   return true;
 }
 
diff --git a/modules/rtp_rtcp/source/rtp_packetizer_av1_unittest.cc b/modules/rtp_rtcp/source/rtp_packetizer_av1_unittest.cc
index a7e65134d6..38b7f583ad 100644
--- a/modules/rtp_rtcp/source/rtp_packetizer_av1_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_packetizer_av1_unittest.cc
@@ -41,7 +41,7 @@ constexpr uint8_t kNewCodedVideoSequenceBit = 0b00'00'1000;
 // Wrapper around rtp_packet to make it look like container of payload bytes.
 struct RtpPayload {
   using value_type = ArrayView<const uint8_t>::value_type;
-  using const_iterator = ArrayView<const uint8_t>::const_iterator;
+  using const_iterator = ArrayView<const uint8_t>::iterator;
 
   RtpPayload() : rtp_packet(/*extensions=*/nullptr) {}
   RtpPayload& operator=(RtpPayload&&) = default;

From 35107c60123adee839ba4824a0e113fae1fc8286 Mon Sep 17 00:00:00 2001
From: Bjorn Terelius <terelius@webrtc.org>
Date: Thu, 19 Feb 2026 13:40:19 +0100
Subject: [PATCH 098/472] Move NetEq simulation plots into event log analyzer
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Bug: None
Change-Id: I4dc6a2ff8641b6c1a759025a7722701e51f8a13f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449980
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46951}
---
 .../rtc_event_log_visualizer/analyze_audio.cc |  20 +++
 .../rtc_event_log_visualizer/analyze_audio.h  |  24 +++-
 .../rtc_event_log_visualizer/analyzer.cc      |  86 +++++++++++-
 rtc_tools/rtc_event_log_visualizer/analyzer.h |  24 +++-
 rtc_tools/rtc_event_log_visualizer/main.cc    | 128 +-----------------
 5 files changed, 148 insertions(+), 134 deletions(-)

diff --git a/rtc_tools/rtc_event_log_visualizer/analyze_audio.cc b/rtc_tools/rtc_event_log_visualizer/analyze_audio.cc
index c7a5242db0..3fa1f53dd7 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyze_audio.cc
+++ b/rtc_tools/rtc_event_log_visualizer/analyze_audio.cc
@@ -513,4 +513,24 @@ void CreateAudioLevelGraph(const ParsedRtcEventLog& parsed_log,
   plot->SetTitle(GetDirectionAsString(direction) + " audio level");
 }
 
+LazyNetEqSimulator::LazyNetEqSimulator(const ParsedRtcEventLog& parsed_log,
+                                       const AnalyzerConfig& config)
+    : parsed_log_(parsed_log), config_(config) {}
+
+void LazyNetEqSimulator::SetReplacementAudioFile(
+    absl::string_view replacement_file_name,
+    int file_sample_rate_hz) {
+  replacement_file_name_ = std::string(replacement_file_name);
+  file_sample_rate_hz_ = file_sample_rate_hz;
+}
+
+const NetEqStatsGetterMap& LazyNetEqSimulator::GetStats() const {
+  if (!neteq_stats_) {
+    neteq_stats_ =
+        SimulateNetEq(parsed_log_, config_, replacement_file_name_,
+                      file_sample_rate_hz_, &config_.env_.field_trials());
+  }
+  return *neteq_stats_;
+}
+
 }  // namespace webrtc
diff --git a/rtc_tools/rtc_event_log_visualizer/analyze_audio.h b/rtc_tools/rtc_event_log_visualizer/analyze_audio.h
index 6b6ff7ed88..271e1d8c18 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyze_audio.h
+++ b/rtc_tools/rtc_event_log_visualizer/analyze_audio.h
@@ -14,10 +14,10 @@
 #include <cstdint>
 #include <map>
 #include <memory>
+#include <optional>
 #include <string>
 
 #include "absl/strings/string_view.h"
-#include "api/field_trials_view.h"
 #include "api/function_view.h"
 #include "api/neteq/neteq.h"
 #include "logging/rtc_event_log/rtc_event_log_parser.h"
@@ -61,11 +61,29 @@ void CreateAudioLevelGraph(const ParsedRtcEventLog& parsed_log,
 
 using NetEqStatsGetterMap =
     std::map<uint32_t, std::unique_ptr<test::NetEqStatsGetter>>;
+
+class LazyNetEqSimulator {
+ public:
+  LazyNetEqSimulator(const ParsedRtcEventLog& parsed_log,
+                     const AnalyzerConfig& config);
+
+  void SetReplacementAudioFile(absl::string_view replacement_file_name,
+                               int file_sample_rate_hz);
+
+  const NetEqStatsGetterMap& GetStats() const;
+
+ private:
+  const ParsedRtcEventLog& parsed_log_;
+  const AnalyzerConfig config_;
+  std::string replacement_file_name_;
+  int file_sample_rate_hz_ = 48000;
+  mutable std::optional<NetEqStatsGetterMap> neteq_stats_;
+};
+
 NetEqStatsGetterMap SimulateNetEq(const ParsedRtcEventLog& parsed_log,
                                   const AnalyzerConfig& config,
                                   absl::string_view replacement_file_name,
-                                  int file_sample_rate_hz,
-                                  const FieldTrialsView* field_trials);
+                                  int file_sample_rate_hz);
 
 void CreateAudioJitterBufferGraph(const ParsedRtcEventLog& parsed_log,
                                   const AnalyzerConfig& config,
diff --git a/rtc_tools/rtc_event_log_visualizer/analyzer.cc b/rtc_tools/rtc_event_log_visualizer/analyzer.cc
index 73e1db7509..72ef110236 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyzer.cc
+++ b/rtc_tools/rtc_event_log_visualizer/analyzer.cc
@@ -11,6 +11,7 @@
 #include "rtc_tools/rtc_event_log_visualizer/analyzer.h"
 
 #include <cstddef>
+#include <memory>
 #include <string>
 #include <vector>
 
@@ -18,6 +19,7 @@
 #include "absl/strings/string_view.h"
 #include "api/environment/environment_factory.h"
 #include "api/function_view.h"
+#include "api/neteq/neteq.h"
 #include "api/units/data_rate.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
@@ -43,6 +45,7 @@ EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& parsed_log,
     RTC_LOG(LS_WARNING) << "No useful events in the log.";
     config_.begin_time_ = config_.end_time_ = Timestamp::Zero();
   }
+  neteq_simulator_ = std::make_unique<LazyNetEqSimulator>(parsed_log_, config_);
 
   RTC_LOG(LS_INFO) << "Log is "
                    << (parsed_log_.last_timestamp().ms() -
@@ -54,6 +57,7 @@ EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& parsed_log,
 EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& parsed_log,
                                    const AnalyzerConfig& config)
     : parsed_log_(parsed_log), config_(config) {
+  neteq_simulator_ = std::make_unique<LazyNetEqSimulator>(parsed_log_, config_);
   RTC_LOG(LS_INFO) << "Log is "
                    << (parsed_log_.last_timestamp().ms() -
                        parsed_log_.first_timestamp().ms()) /
@@ -61,6 +65,15 @@ EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& parsed_log,
                    << " seconds long.";
 }
 
+EventLogAnalyzer::~EventLogAnalyzer() = default;
+
+void EventLogAnalyzer::SetNetEqReplacementFile(
+    absl::string_view replacement_file_name,
+    int file_sample_rate_hz) {
+  neteq_simulator_->SetReplacementAudioFile(replacement_file_name,
+                                            file_sample_rate_hz);
+}
+
 void EventLogAnalyzer::CreateGraphsByName(const std::vector<std::string>& names,
                                           PlotCollection* collection) const {
   for (absl::string_view name : names) {
@@ -68,8 +81,7 @@ void EventLogAnalyzer::CreateGraphsByName(const std::vector<std::string>& names,
       return plot.label == name;
     });
     if (plot != plots_.end()) {
-      Plot* output = collection->AppendNewPlot(plot->label);
-      plot->plot_func(output);
+      plot->plot_func(collection);
     }
   }
 }
@@ -281,6 +293,76 @@ void EventLogAnalyzer::InitializeMapOfNamedGraphs(bool show_detector_state,
   plots_.RegisterPlot("dtls_writable_state", [this](Plot* plot) {
     this->CreateDtlsWritableStateGraph(plot);
   });
+  plots_.RegisterPlot(
+      "simulated_neteq_expand_rate", [this](PlotCollection* collection) {
+        CreateNetEqNetworkStatsGraph(
+            parsed_log_, config_, neteq_simulator_->GetStats(),
+            [](const NetEqNetworkStatistics& stats) {
+              return stats.expand_rate / 16384.f;
+            },
+            "Expand rate",
+            collection->AppendNewPlot("simulated_neteq_expand_rate"));
+      });
+  plots_.RegisterPlot(
+      "simulated_neteq_speech_expand_rate", [this](PlotCollection* collection) {
+        CreateNetEqNetworkStatsGraph(
+            parsed_log_, config_, neteq_simulator_->GetStats(),
+            [](const NetEqNetworkStatistics& stats) {
+              return stats.speech_expand_rate / 16384.f;
+            },
+            "Speech expand rate",
+            collection->AppendNewPlot("simulated_neteq_speech_expand_rate"));
+      });
+  plots_.RegisterPlot(
+      "simulated_neteq_accelerate_rate", [this](PlotCollection* collection) {
+        CreateNetEqNetworkStatsGraph(
+            parsed_log_, config_, neteq_simulator_->GetStats(),
+            [](const NetEqNetworkStatistics& stats) {
+              return stats.accelerate_rate / 16384.f;
+            },
+            "Accelerate rate",
+            collection->AppendNewPlot("simulated_neteq_accelerate_rate"));
+      });
+  plots_.RegisterPlot(
+      "simulated_neteq_preemptive_rate", [this](PlotCollection* collection) {
+        CreateNetEqNetworkStatsGraph(
+            parsed_log_, config_, neteq_simulator_->GetStats(),
+            [](const NetEqNetworkStatistics& stats) {
+              return stats.preemptive_rate / 16384.f;
+            },
+            "Preemptive rate",
+            collection->AppendNewPlot("simulated_neteq_preemptive_rate"));
+      });
+  plots_.RegisterPlot(
+      "simulated_neteq_concealment_events", [this](PlotCollection* collection) {
+        CreateNetEqLifetimeStatsGraph(
+            parsed_log_, config_, neteq_simulator_->GetStats(),
+            [](const NetEqLifetimeStatistics& stats) {
+              return static_cast<float>(stats.concealment_events);
+            },
+            "Concealment events",
+            collection->AppendNewPlot("simulated_neteq_concealment_events"));
+      });
+  plots_.RegisterPlot(
+      "simulated_neteq_preferred_buffer_size",
+      [this](PlotCollection* collection) {
+        CreateNetEqNetworkStatsGraph(
+            parsed_log_, config_, neteq_simulator_->GetStats(),
+            [](const NetEqNetworkStatistics& stats) {
+              return stats.preferred_buffer_size_ms;
+            },
+            "Preferred buffer size (ms)",
+            collection->AppendNewPlot("simulated_neteq_preferred_buffer_size"));
+      });
+  plots_.RegisterPlot(
+      "simulated_neteq_jitter_buffer_delay",
+      [this](PlotCollection* collection) {
+        for (const auto& st : neteq_simulator_->GetStats()) {
+          CreateAudioJitterBufferGraph(
+              parsed_log_, config_, st.first, st.second.get(),
+              collection->AppendNewPlot("simulated_neteq_jitter_buffer_delay"));
+        }
+      });
 }
 void EventLogAnalyzer::CreatePlayoutGraph(Plot* plot) const {
   webrtc::CreatePlayoutGraph(parsed_log_, config_, plot);
diff --git a/rtc_tools/rtc_event_log_visualizer/analyzer.h b/rtc_tools/rtc_event_log_visualizer/analyzer.h
index 8ac90e3040..790c9c597c 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyzer.h
+++ b/rtc_tools/rtc_event_log_visualizer/analyzer.h
@@ -13,9 +13,11 @@
 
 #include <cstdio>
 #include <functional>
+#include <memory>
 #include <string>
 #include <vector>
 
+#include "absl/strings/string_view.h"
 #include "api/function_view.h"
 #include "logging/rtc_event_log/rtc_event_log_parser.h"
 #include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
@@ -25,18 +27,22 @@
 
 namespace webrtc {
 
+class LazyNetEqSimulator;
+
 class EventLogAnalyzer {
   struct PlotDeclaration {
-    PlotDeclaration(const std::string& label, std::function<void(Plot*)> f)
+    PlotDeclaration(const std::string& label,
+                    std::function<void(PlotCollection*)> f)
         : label(label), plot_func(f) {}
     const std::string label;
     // TODO(terelius): Add a help text/explanation.
-    const std::function<void(Plot*)> plot_func;
+    const std::function<void(PlotCollection*)> plot_func;
   };
 
   class PlotMap {
    public:
-    void RegisterPlot(const std::string& label, std::function<void(Plot*)> f) {
+    void RegisterPlot(const std::string& label,
+                      std::function<void(PlotCollection*)> f) {
       for (const auto& plot : plots_) {
         RTC_DCHECK(plot.label != label)
             << "Can't use the same label for multiple plots";
@@ -44,6 +50,12 @@ class EventLogAnalyzer {
       plots_.push_back({label, f});
     }
 
+    void RegisterPlot(const std::string& label, std::function<void(Plot*)> f) {
+      RegisterPlot(label, [f, label](PlotCollection* collection) {
+        f(collection->AppendNewPlot(label));
+      });
+    }
+
     std::vector<PlotDeclaration>::const_iterator begin() const {
       return plots_.begin();
     }
@@ -61,6 +73,7 @@ class EventLogAnalyzer {
   // modified while the EventLogAnalyzer is being used.
   EventLogAnalyzer(const ParsedRtcEventLog& log, bool normalize_time);
   EventLogAnalyzer(const ParsedRtcEventLog& log, const AnalyzerConfig& config);
+  ~EventLogAnalyzer();
 
   void CreateGraphsByName(const std::vector<std::string>& names,
                           PlotCollection* collection) const;
@@ -144,12 +157,17 @@ class EventLogAnalyzer {
   void CreateTriageNotifications() const;
   void PrintNotifications(FILE* file) const;
 
+  void SetNetEqReplacementFile(absl::string_view replacement_file_name,
+                               int file_sample_rate_hz);
+
  private:
   const ParsedRtcEventLog& parsed_log_;
 
   AnalyzerConfig config_;
 
   PlotMap plots_;
+
+  std::unique_ptr<LazyNetEqSimulator> neteq_simulator_;
 };
 
 }  // namespace webrtc
diff --git a/rtc_tools/rtc_event_log_visualizer/main.cc b/rtc_tools/rtc_event_log_visualizer/main.cc
index 1a775a8ee8..67ac267d80 100644
--- a/rtc_tools/rtc_event_log_visualizer/main.cc
+++ b/rtc_tools/rtc_event_log_visualizer/main.cc
@@ -8,15 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-
 #include <cstdio>
 #include <fstream>
 #include <iostream>
 #include <map>
 #include <memory>
-#include <optional>
 #include <string>
-#include <utility>
 #include <vector>
 
 #include "absl/algorithm/container.h"
@@ -29,13 +26,11 @@
 #include "api/environment/environment.h"
 #include "api/environment/environment_factory.h"
 #include "api/field_trials.h"
-#include "api/neteq/neteq.h"
 #include "api/units/time_delta.h"
 #include "logging/rtc_event_log/rtc_event_log_parser.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
 #include "rtc_tools/rtc_event_log_visualizer/alerts.h"
-#include "rtc_tools/rtc_event_log_visualizer/analyze_audio.h"
 #include "rtc_tools/rtc_event_log_visualizer/analyzer.h"
 #include "rtc_tools/rtc_event_log_visualizer/analyzer_common.h"
 #include "rtc_tools/rtc_event_log_visualizer/conversational_speech_en.h"
@@ -223,6 +218,7 @@ int main(int argc, char* argv[]) {
   analyzer.InitializeMapOfNamedGraphs(absl::GetFlag(FLAGS_show_detector_state),
                                       absl::GetFlag(FLAGS_show_alr_state),
                                       absl::GetFlag(FLAGS_show_link_capacity));
+  analyzer.SetNetEqReplacementFile(wav_path, 48000);
 
   // Flag replacements
   std::map<std::string, std::vector<std::string>> flag_aliases = {
@@ -266,11 +262,6 @@ int main(int argc, char* argv[]) {
       // TODO(terelius): Also print a help text.
       std::cerr << "  " << plot_name;
     }
-    // The following flags don't fit the model used for the other plots.
-    for (const auto& plot_name : flag_aliases["simulated_neteq_stats"]) {
-      std::cerr << " " << plot_name;
-    }
-    std::cerr << std::endl;
 
     std::cerr << "List of plot aliases (for use with the --plot flag):"
               << std::endl;
@@ -295,13 +286,7 @@ int main(int argc, char* argv[]) {
   std::vector<std::string> plot_flags =
       StrSplit(absl::GetFlag(FLAGS_plot), ",");
   std::vector<std::string> plot_names;
-  const std::vector<std::string> known_analyzer_plots =
-      analyzer.GetGraphNames();
-  const std::vector<std::string> known_neteq_plots =
-      flag_aliases["simulated_neteq_stats"];
-  std::vector<std::string> all_known_plots = known_analyzer_plots;
-  all_known_plots.insert(all_known_plots.end(), known_neteq_plots.begin(),
-                         known_neteq_plots.end());
+  const std::vector<std::string> all_known_plots = analyzer.GetGraphNames();
   for (const std::string& flag : plot_flags) {
     if (flag == "all") {
       plot_names = all_known_plots;
@@ -329,115 +314,6 @@ int main(int argc, char* argv[]) {
   webrtc::PlotCollection collection;
   analyzer.CreateGraphsByName(plot_names, &collection);
 
-  // The simulated neteq charts are treated separately because they have a
-  // different behavior compared to all other plots. In particular, the neteq
-  // plots
-  //  * cache the simulation results between different plots
-  //  * open and read files
-  //  * dont have a 1-to-1 mapping between IDs and charts.
-  std::optional<webrtc::NetEqStatsGetterMap> neteq_stats;
-  if (absl::c_find(plot_names, "simulated_neteq_expand_rate") !=
-      plot_names.end()) {
-    if (!neteq_stats) {
-      neteq_stats = webrtc::SimulateNetEq(parsed_log, config, wav_path, 48000,
-                                          &env.field_trials());
-    }
-    webrtc::CreateNetEqNetworkStatsGraph(
-        parsed_log, config, *neteq_stats,
-        [](const webrtc::NetEqNetworkStatistics& stats) {
-          return stats.expand_rate / 16384.f;
-        },
-        "Expand rate", collection.AppendNewPlot("simulated_neteq_expand_rate"));
-  }
-  if (absl::c_find(plot_names, "simulated_neteq_speech_expand_rate") !=
-      plot_names.end()) {
-    if (!neteq_stats) {
-      neteq_stats = webrtc::SimulateNetEq(parsed_log, config, wav_path, 48000,
-                                          &env.field_trials());
-    }
-    webrtc::CreateNetEqNetworkStatsGraph(
-        parsed_log, config, *neteq_stats,
-        [](const webrtc::NetEqNetworkStatistics& stats) {
-          return stats.speech_expand_rate / 16384.f;
-        },
-        "Speech expand rate",
-        collection.AppendNewPlot("simulated_neteq_speech_expand_rate"));
-  }
-  if (absl::c_find(plot_names, "simulated_neteq_accelerate_rate") !=
-      plot_names.end()) {
-    if (!neteq_stats) {
-      neteq_stats = webrtc::SimulateNetEq(parsed_log, config, wav_path, 48000,
-                                          &env.field_trials());
-    }
-    webrtc::CreateNetEqNetworkStatsGraph(
-        parsed_log, config, *neteq_stats,
-        [](const webrtc::NetEqNetworkStatistics& stats) {
-          return stats.accelerate_rate / 16384.f;
-        },
-        "Accelerate rate",
-        collection.AppendNewPlot("simulated_neteq_accelerate_rate"));
-  }
-  if (absl::c_find(plot_names, "simulated_neteq_preemptive_rate") !=
-      plot_names.end()) {
-    if (!neteq_stats) {
-      neteq_stats = webrtc::SimulateNetEq(parsed_log, config, wav_path, 48000,
-                                          &env.field_trials());
-    }
-    webrtc::CreateNetEqNetworkStatsGraph(
-        parsed_log, config, *neteq_stats,
-        [](const webrtc::NetEqNetworkStatistics& stats) {
-          return stats.preemptive_rate / 16384.f;
-        },
-        "Preemptive rate",
-        collection.AppendNewPlot("simulated_neteq_preemptive_rate"));
-  }
-  if (absl::c_find(plot_names, "simulated_neteq_concealment_events") !=
-      plot_names.end()) {
-    if (!neteq_stats) {
-      neteq_stats = webrtc::SimulateNetEq(parsed_log, config, wav_path, 48000,
-                                          &env.field_trials());
-    }
-    webrtc::CreateNetEqLifetimeStatsGraph(
-        parsed_log, config, *neteq_stats,
-        [](const webrtc::NetEqLifetimeStatistics& stats) {
-          return static_cast<float>(stats.concealment_events);
-        },
-        "Concealment events",
-        collection.AppendNewPlot("simulated_neteq_concealment_events"));
-  }
-  if (absl::c_find(plot_names, "simulated_neteq_preferred_buffer_size") !=
-      plot_names.end()) {
-    if (!neteq_stats) {
-      neteq_stats = webrtc::SimulateNetEq(parsed_log, config, wav_path, 48000,
-                                          &env.field_trials());
-    }
-    webrtc::CreateNetEqNetworkStatsGraph(
-        parsed_log, config, *neteq_stats,
-        [](const webrtc::NetEqNetworkStatistics& stats) {
-          return stats.preferred_buffer_size_ms;
-        },
-        "Preferred buffer size (ms)",
-        collection.AppendNewPlot("simulated_neteq_preferred_buffer_size"));
-  }
-
-  // The model we use for registering plots assumes that the each plot label
-  // can be mapped to a lambda that will produce exactly one plot. The
-  // simulated_neteq_jitter_buffer_delay plot doesn't fit this model since it
-  // creates multiple plots, and would need some state kept between the lambda
-  // calls.
-  if (absl::c_find(plot_names, "simulated_neteq_jitter_buffer_delay") !=
-      plot_names.end()) {
-    if (!neteq_stats) {
-      neteq_stats = webrtc::SimulateNetEq(parsed_log, config, wav_path, 48000,
-                                          &env.field_trials());
-    }
-    for (auto it = neteq_stats->cbegin(); it != neteq_stats->cend(); ++it) {
-      webrtc::CreateAudioJitterBufferGraph(
-          parsed_log, config, it->first, it->second.get(),
-          collection.AppendNewPlot("simulated_neteq_jitter_buffer_delay"));
-    }
-  }
-
   collection.SetCallTimeToUtcOffsetMs(config.CallTimeToUtcOffsetMs());
 
   if (absl::GetFlag(FLAGS_protobuf_output)) {

From 58016c0d9ccaee5fcd1f1301b6fcdf1365b76d0b Mon Sep 17 00:00:00 2001
From: Bjorn Terelius <terelius@webrtc.org>
Date: Thu, 19 Feb 2026 13:34:43 +0100
Subject: [PATCH 099/472] Update some RTC event log parser comments
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Bug: None
Change-Id: I6a3ef4e4cc2226f9577980ab82163d1b59387b12
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/450560
Auto-Submit: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46952}
---
 logging/rtc_event_log/rtc_event_log_parser.h  | 2 +-
 logging/rtc_event_log/rtc_event_processor.h   | 2 +-
 rtc_tools/rtc_event_log_visualizer/analyzer.h | 4 ++--
 3 files changed, 4 insertions(+), 4 deletions(-)

diff --git a/logging/rtc_event_log/rtc_event_log_parser.h b/logging/rtc_event_log/rtc_event_log_parser.h
index 4e97d91057..1b81e3c80f 100644
--- a/logging/rtc_event_log/rtc_event_log_parser.h
+++ b/logging/rtc_event_log/rtc_event_log_parser.h
@@ -388,7 +388,7 @@ class ParsedRtcEventLog {
 
   ~ParsedRtcEventLog();
 
-  // Clears previously parsed events and resets the ParsedRtcEventLogNew to an
+  // Clears previously parsed events and resets the ParsedRtcEventLog to an
   // empty state.
   void Clear();
 
diff --git a/logging/rtc_event_log/rtc_event_processor.h b/logging/rtc_event_log/rtc_event_processor.h
index 846fbb842c..66fbd7ae61 100644
--- a/logging/rtc_event_log/rtc_event_processor.h
+++ b/logging/rtc_event_log/rtc_event_processor.h
@@ -104,7 +104,7 @@ class ProcessableEventList : public ProcessableEventListInterface {
 // the common base class.
 //
 // Usage example:
-// ParsedRtcEventLogNew log;
+// ParsedRtcEventLog log;
 // auto incoming_handler = [] (LoggedRtcpPacketIncoming elem) { ... };
 // auto outgoing_handler = [] (LoggedRtcpPacketOutgoing elem) { ... };
 //
diff --git a/rtc_tools/rtc_event_log_visualizer/analyzer.h b/rtc_tools/rtc_event_log_visualizer/analyzer.h
index 790c9c597c..511d002036 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyzer.h
+++ b/rtc_tools/rtc_event_log_visualizer/analyzer.h
@@ -68,8 +68,8 @@ class EventLogAnalyzer {
   };
 
  public:
-  // The EventLogAnalyzer keeps a reference to the ParsedRtcEventLogNew for the
-  // duration of its lifetime. The ParsedRtcEventLogNew must not be destroyed or
+  // The EventLogAnalyzer keeps a reference to the ParsedRtcEventLog for the
+  // duration of its lifetime. The ParsedRtcEventLog must not be destroyed or
   // modified while the EventLogAnalyzer is being used.
   EventLogAnalyzer(const ParsedRtcEventLog& log, bool normalize_time);
   EventLogAnalyzer(const ParsedRtcEventLog& log, const AnalyzerConfig& config);

From de07f737b18064d8cf77c0e9f76ddb273b04a461 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Thu, 19 Feb 2026 08:07:39 -0800
Subject: [PATCH 100/472] Roll chromium_revision cabe765c34..1063bfbb8a
 (1583505:1587147)

Change log: https://chromium.googlesource.com/chromium/src/+log/cabe765c34..1063bfbb8a
Full diff: https://chromium.googlesource.com/chromium/src/+/cabe765c34..1063bfbb8a

Changed dependencies
* siso_version: git_revision:2599cc0037d82d9187beea5c13349ce05716817b..git_revision:54be8bcbdd069237406d476362aeb261d6deb8c0
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/664be89a22..94f11bf92f
* src/buildtools: https://chromium.googlesource.com/chromium/src/buildtools/+log/6a18683f55..451e71432a
* src/buildtools/linux64: git_revision:304bbef6c7e9a86630c12986b99c8654eb7fe648..git_revision:318ec1d2db43c1dc1e1810afb8e7fe524d7a64f2
* src/buildtools/mac: git_revision:304bbef6c7e9a86630c12986b99c8654eb7fe648..git_revision:318ec1d2db43c1dc1e1810afb8e7fe524d7a64f2
* src/buildtools/win: git_revision:304bbef6c7e9a86630c12986b99c8654eb7fe648..git_revision:318ec1d2db43c1dc1e1810afb8e7fe524d7a64f2
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/d68e57b862..e2c585cccd
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/175e2a4da7..c5cf2d34d5
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/6624d10a4b..e794b55e51
* src/third_party/android_build_tools/manifest_merger/cipd: xxaa38R0ED95Wxo8P29BC-mB7bjFPi2HKMpiIJn-UhAC..yPYRRYty6hqUJSOO_UdrYg31nwC-gaiPQTBBMXV8Y-wC
* src/third_party/android_deps/autorolled/cipd: RiKmV-EpMhg5ibpMbPpENQmfU4IPLpo8cxer4F3NSUgC..Lskjo0vFA222MYDpjYl71PwoWG_9mb-F_v3RTHvE-LkC
* src/third_party/androidx/cipd: Cl-ySzwc8_wt7xkWL9dKKHbvVRhOHTgGANeH3JHEB3cC..XpThiLpVJwtbiGf8RwS7hzulYyS7Q29MXabSOIrsIX4C
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/932fb7389f..17b020f428
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/4fe7ca3037..172c8e1ba5
* src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/7a0968b56b..57d67767dc
* src/third_party/cpuinfo/src: https://chromium.googlesource.com/external/github.com/pytorch/cpuinfo.git/+log/84818a41e0..7364b490b5
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/6c6a6572eb..5f1a0a82b5
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/23fcc1c6c9..bdec88009d
* src/third_party/ffmpeg: https://chromium.googlesource.com/chromium/third_party/ffmpeg.git/+log/ae11d2ba5c..946d97db8d
* src/third_party/fuzztest/src: https://chromium.googlesource.com/external/github.com/google/fuzztest.git/+log/fb29fbd005..4aaf69f8f8
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/f9b8a88e84..e6e7934770
* src/third_party/kotlin_stdlib/cipd: FfVJUc4q9S6JB4uqVGPJQ3anrc1Y_W0O-oGK_2MJA5gC..JcCPEPsUevwiZ519BkXHvRR8J-9XEcwj_plAWjVIR7MC
* src/third_party/libaom/source/libaom: https://aomedia.googlesource.com/aom.git/+log/37b2db4334..a27d804206
* src/third_party/libunwind/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git/+log/ba19d93d6d..17ccf7d110
* src/third_party/libyuv: https://chromium.googlesource.com/libyuv/libyuv.git/+log/917276084a..6067afde56
* src/third_party/llvm-build/Release+Asserts: Linux_x64/clang-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Linux_x64/clang-tidy-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Linux_x64/clangd-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Linux_x64/llvm-code-coverage-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Linux_x64/llvmobjdump-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Mac/clang-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Mac/clang-mac-runtime-library-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Mac/clang-tidy-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Mac/clangd-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Mac/llvm-code-coverage-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Mac/llvmobjdump-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Mac_arm64/clang-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Mac_arm64/clang-tidy-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Mac_arm64/clangd-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Mac_arm64/llvm-code-coverage-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Mac_arm64/llvmobjdump-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Win/clang-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Win/clang-tidy-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Win/clang-win-runtime-library-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Win/clangd-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Win/llvm-code-coverage-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz,Win/llvmobjdump-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz..Linux_x64/clang-llvmorg-23-init-3706-gfc648683-1.tar.xz,Linux_x64/clang-tidy-llvmorg-23-init-3706-gfc648683-1.tar.xz,Linux_x64/clangd-llvmorg-23-init-3706-gfc648683-1.tar.xz,Linux_x64/llvm-code-coverage-llvmorg-23-init-3706-gfc648683-1.tar.xz,Linux_x64/llvmobjdump-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac/clang-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac/clang-mac-runtime-library-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac/clang-tidy-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac/clangd-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac/llvm-code-coverage-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac/llvmobjdump-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac_arm64/clang-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac_arm64/clang-tidy-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac_arm64/clangd-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac_arm64/llvm-code-coverage-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac_arm64/llvmobjdump-llvmorg-23-init-3706-gfc648683-1.tar.xz,Win/clang-llvmorg-23-init-3706-gfc648683-1.tar.xz,Win/clang-tidy-llvmorg-23-init-3706-gfc648683-1.tar.xz,Win/clang-win-runtime-library-llvmorg-23-init-3706-gfc648683-1.tar.xz,Win/clangd-llvmorg-23-init-3706-gfc648683-1.tar.xz,Win/llvm-code-coverage-llvmorg-23-init-3706-gfc648683-1.tar.xz,Win/llvmobjdump-llvmorg-23-init-3706-gfc648683-1.tar.xz
* src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/2f7df07563..7061b80a01
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/7475cfd103..d374841f93
* src/third_party/r8/cipd: YvVZKKYTZDYMrvAXwMFbMGMUQaJJMb5LIirLrGcOLS0C..dKavKsYt7Uj8fRmoAALj-uumULBl43j-NHRUZ1YvnxUC
* src/third_party/r8/d8/cipd: a4fVqbIycCDqs1714SLRqxEdz6P-sH-z1QT_eeeF0PcC..dKavKsYt7Uj8fRmoAALj-uumULBl43j-NHRUZ1YvnxUC
* src/third_party/rust-toolchain: Linux_x64/rust-toolchain-7d8ebe3128fc87f3da1ad64240e63ccf07b8f0bd-3-llvmorg-23-init-2224-g5bd8dadb.tar.xz,Mac/rust-toolchain-7d8ebe3128fc87f3da1ad64240e63ccf07b8f0bd-3-llvmorg-23-init-2224-g5bd8dadb.tar.xz,Mac_arm64/rust-toolchain-7d8ebe3128fc87f3da1ad64240e63ccf07b8f0bd-3-llvmorg-23-init-2224-g5bd8dadb.tar.xz,Win/rust-toolchain-7d8ebe3128fc87f3da1ad64240e63ccf07b8f0bd-3-llvmorg-23-init-2224-g5bd8dadb.tar.xz..Linux_x64/rust-toolchain-7dc2e92b83be02dc07f87be7e94266d6e48e5ca5-1-llvmorg-23-init-3706-gfc648683.tar.xz,Mac/rust-toolchain-7dc2e92b83be02dc07f87be7e94266d6e48e5ca5-1-llvmorg-23-init-3706-gfc648683.tar.xz,Mac_arm64/rust-toolchain-7dc2e92b83be02dc07f87be7e94266d6e48e5ca5-1-llvmorg-23-init-3706-gfc648683.tar.xz,Win/rust-toolchain-7dc2e92b83be02dc07f87be7e94266d6e48e5ca5-1-llvmorg-23-init-3706-gfc648683.tar.xz
* src/third_party/ruy/src: https://chromium.googlesource.com/external/github.com/google/ruy.git/+log/576e020f06..2af8886361
* src/third_party/siso/cipd: git_revision:2599cc0037d82d9187beea5c13349ce05716817b..git_revision:54be8bcbdd069237406d476362aeb261d6deb8c0
* src/third_party/tflite/src: https://chromium.googlesource.com/external/github.com/tensorflow/tensorflow.git/+log/01e030d23d..38dd095516
* src/third_party/xnnpack/src: https://chromium.googlesource.com/external/github.com/google/XNNPACK.git/+log/4574c4d9b0..e8aad3cc4f
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/1f457fb242..f7ad2349e5
* src/tools/luci-go: git_revision:072101cbfec3372b812ff510df8547d7b4187bea..git_revision:4d646a5cce1959bdfb88c838b1044354c37f0655
* src/tools/luci-go: git_revision:072101cbfec3372b812ff510df8547d7b4187bea..git_revision:4d646a5cce1959bdfb88c838b1044354c37f0655
* src/tools/luci-go: git_revision:072101cbfec3372b812ff510df8547d7b4187bea..git_revision:4d646a5cce1959bdfb88c838b1044354c37f0655
DEPS diff: https://chromium.googlesource.com/chromium/src/+/cabe765c34..1063bfbb8a/DEPS

BUG=None

Change-Id: Ic35ee1079aa706ec68da703a2ea1158bccbadea7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/450620
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#46953}
---
 DEPS | 282 +++++++++++++++++++++++++++++------------------------------
 1 file changed, 141 insertions(+), 141 deletions(-)

diff --git a/DEPS b/DEPS
index 7f04a3fcff..29501525ec 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': 'cabe765c3407b8d14986c515edcfe81ed96319e0',
+  'chromium_revision': '1063bfbb8aea8b53ec93218deecbf3fc00aff8af',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -50,7 +50,7 @@ vars = {
   # reclient CIPD package version
   'reclient_version': 're_client_version:0.185.0.db415f21-gomaip',
   # siso CIPD package version.
-  'siso_version': 'git_revision:2599cc0037d82d9187beea5c13349ce05716817b',
+  'siso_version': 'git_revision:54be8bcbdd069237406d476362aeb261d6deb8c0',
 
   # ninja CIPD package.
   'ninja_package': 'infra/3pp/tools/ninja/',
@@ -69,28 +69,28 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@664be89a2294ef5491c87eb834443054385216b0',
+    'https://chromium.googlesource.com/chromium/src/build@94f11bf92fb777c6153a331d756f6d4cf2ee3e9f',
   'src/buildtools':
-    'https://chromium.googlesource.com/chromium/src/buildtools@6a18683f555b4ac8b05ac8395c29c84483ac9588',
+    'https://chromium.googlesource.com/chromium/src/buildtools@451e71432ac9f6a91d4d5604c29e0a5078e8c2e8',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
   'src/examples/androidtests/third_party/gradle': {
     'url': 'https://chromium.googlesource.com/external/github.com/gradle/gradle.git@f2d1fb54a951d8b11d25748e4711bec8d128d7e3',
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@d68e57b8626689925505a9d79d54aa841cb0a387',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@e2c585cccd76198491dbc6dcb65471aad488acca',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@175e2a4da7d46983fa2d21a56e3c597b333f3a99',
+    'https://chromium.googlesource.com/chromium/src/testing@c5cf2d34d56c6bc6177cf85439578638bb3535c9',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@6624d10a4b14fab6670316f8e0033a2aea3fd7c7',
+    'https://chromium.googlesource.com/chromium/src/third_party@e794b55e5118051f0676641942e16c62686d7bc7',
 
   'src/buildtools/linux64': {
     'packages': [
       {
         'package': 'gn/gn/linux-${{arch}}',
-        'version': 'git_revision:304bbef6c7e9a86630c12986b99c8654eb7fe648',
+        'version': 'git_revision:318ec1d2db43c1dc1e1810afb8e7fe524d7a64f2',
       }
     ],
     'dep_type': 'cipd',
@@ -100,7 +100,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/mac-${{arch}}',
-        'version': 'git_revision:304bbef6c7e9a86630c12986b99c8654eb7fe648',
+        'version': 'git_revision:318ec1d2db43c1dc1e1810afb8e7fe524d7a64f2',
       }
     ],
     'dep_type': 'cipd',
@@ -110,7 +110,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/windows-amd64',
-        'version': 'git_revision:304bbef6c7e9a86630c12986b99c8654eb7fe648',
+        'version': 'git_revision:318ec1d2db43c1dc1e1810afb8e7fe524d7a64f2',
       }
     ],
     'dep_type': 'cipd',
@@ -136,157 +136,157 @@ deps = {
     'objects': [
       {
         # The Android libclang_rt.builtins libraries are currently only included in the Linux clang package.
-        'object_name': 'Linux_x64/clang-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
-        'sha256sum': 'bab27d04b9740f772fb6f50461ba4c2cf96c57231eb83943f785775c0bc274a6',
-        'size_bytes': 57583584,
-        'generation': 1770400171446681,
+        'object_name': 'Linux_x64/clang-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': '9a5dfd23f6d37b21709327aa39562225c01a058cc209d1c629448d6d6d874ab2',
+        'size_bytes': 57775784,
+        'generation': 1770870606052091,
         'condition': '(host_os == "linux" or checkout_android) and non_git_source',
       },
       {
-        'object_name': 'Linux_x64/clang-tidy-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
-        'sha256sum': '00633aec3bf29aba53206cb1c58256367954670e1eb4665bd08c401a4c9e538d',
-        'size_bytes': 14383508,
-        'generation': 1770400171915898,
+        'object_name': 'Linux_x64/clang-tidy-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': 'cef6e9f2876cc28956213fd1cfb1431ed6bd9aca135b12ebd33e9d393595009d',
+        'size_bytes': 14411824,
+        'generation': 1770870606077571,
         'condition': 'host_os == "linux" and checkout_clang_tidy and non_git_source',
       },
       {
-        'object_name': 'Linux_x64/clangd-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
-        'sha256sum': 'a67ff56b3a3feb84a204952b6e517686d70cc9eb3744694f8a4ee4e62538efd8',
-        'size_bytes': 14612000,
-        'generation': 1770400171913400,
+        'object_name': 'Linux_x64/clangd-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': 'efccb1da6950b4a83857ad5243196dd9d4cefee5869362abcc9d9422b059fff2',
+        'size_bytes': 14638444,
+        'generation': 1770870606182500,
         'condition': 'host_os == "linux" and checkout_clangd and non_git_source',
       },
       {
-        'object_name': 'Linux_x64/llvm-code-coverage-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
-        'sha256sum': '8920f2faeb5c1620e2df2ebfc17afb1a7405a9ce0681560cd95caa72dee9348d',
-        'size_bytes': 2312556,
-        'generation': 1770400172534077,
+        'object_name': 'Linux_x64/llvm-code-coverage-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': '7d1648506f753035e0c96f9727abf58f0a275d8057be6fd3c231bf80ac60b2ac',
+        'size_bytes': 2326952,
+        'generation': 1770870606310079,
         'condition': 'host_os == "linux" and checkout_clang_coverage_tools and non_git_source',
       },
       {
-        'object_name': 'Linux_x64/llvmobjdump-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
-        'sha256sum': '5fa2372512142aba2824aadd652d8a5416d75806df046876be344b53c1c7d983',
-        'size_bytes': 5785056,
-        'generation': 1770400172080602,
+        'object_name': 'Linux_x64/llvmobjdump-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': '103ba738fb6ebcd290e38458fdec8fa7b95bc3216af1e60ad260bc065b49d447',
+        'size_bytes': 5798432,
+        'generation': 1770870606263889,
         'condition': '((checkout_linux or checkout_mac or checkout_android) and host_os == "linux") and non_git_source',
       },
       {
-        'object_name': 'Mac/clang-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
-        'sha256sum': '89bf5f88dd0e919f1906eb3d3d1c3c7268dc8163d924f492dac4d2627f2c8d94',
-        'size_bytes': 54752956,
-        'generation': 1770400174327435,
+        'object_name': 'Mac/clang-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': '0230c7c9a625ceb2d2c650d995fb9cb5cdb6ab7c0c05042f560cc0cda441a2d0',
+        'size_bytes': 54682532,
+        'generation': 1770870608103936,
         'condition': 'host_os == "mac" and host_cpu == "x64"',
       },
       {
-        'object_name': 'Mac/clang-mac-runtime-library-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
-        'sha256sum': '2c5a55301498d63286a4d249531254e586017c889e16d5d91fdb9db7fdc61451',
-        'size_bytes': 1014688,
-        'generation': 1770400193401638,
+        'object_name': 'Mac/clang-mac-runtime-library-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': 'a7af5d119637f67bff7f390923a78a712b1067157c019aebf2dcddbc4ed7b640',
+        'size_bytes': 1015272,
+        'generation': 1770870616072374,
         'condition': 'checkout_mac and not host_os == "mac"',
       },
       {
-        'object_name': 'Mac/clang-tidy-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
-        'sha256sum': 'fb67062f7fb7edc2ac5a2a7a698a29446f93b8e8c091258b5194aa2b8c166ce1',
-        'size_bytes': 14495128,
-        'generation': 1770400174657643,
+        'object_name': 'Mac/clang-tidy-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': '89bd1e39dc4114183a8cd931a9f5b3dc4fff5f697a133d91da2ecd2f82221774',
+        'size_bytes': 14261468,
+        'generation': 1770870608107521,
         'condition': 'host_os == "mac" and host_cpu == "x64" and checkout_clang_tidy',
       },
       {
-        'object_name': 'Mac/clangd-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
-        'sha256sum': '7d61e0cbca5ef8e9e2f2295097cb996abda3df85e003f9d64c54715b28312b12',
-        'size_bytes': 16448964,
-        'generation': 1770400174847386,
+        'object_name': 'Mac/clangd-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': 'd8c9c8f9a4e62422f2167b283da1e32d39978804543f7eaa08b903d4a0c580a2',
+        'size_bytes': 15425576,
+        'generation': 1770870608113120,
         'condition': 'host_os == "mac" and host_cpu == "x64" and checkout_clangd',
       },
       {
-        'object_name': 'Mac/llvm-code-coverage-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
-        'sha256sum': '35003b14528499ad7f51a5f0388f39708bc779502d289d884532275c3161f1fa',
-        'size_bytes': 2356044,
-        'generation': 1770400175337551,
+        'object_name': 'Mac/llvm-code-coverage-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': 'a705f42d6e441950594986c0afb3f7b2e65cafa10eb1a5f40dc43c0a5d127af6',
+        'size_bytes': 2372548,
+        'generation': 1770870608249587,
         'condition': 'host_os == "mac" and host_cpu == "x64" and checkout_clang_coverage_tools',
       },
       {
-        'object_name': 'Mac/llvmobjdump-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
-        'sha256sum': '72ed5ca9220026efb8a4fabea6b9b2ad39ded51b1bdac8e2290602497bbd74aa',
-        'size_bytes': 5695864,
-        'generation': 1770400175016502,
+        'object_name': 'Mac/llvmobjdump-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': '556b8abf3ec08c42769986ea8940ac7f09fa2cfe49328a2ccccee4641efce6c4',
+        'size_bytes': 5703228,
+        'generation': 1770870608115133,
         'condition': 'host_os == "mac" and host_cpu == "x64"',
       },
       {
-        'object_name': 'Mac_arm64/clang-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
-        'sha256sum': 'eb6086a89ae63808cb85a3b274b5c3870b832385f81b607f966d28b09971191f',
-        'size_bytes': 45747156,
-        'generation': 1770400195193634,
+        'object_name': 'Mac_arm64/clang-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': '34102ff942cb0d7e2163fada5aeb53f5c2438d07d8caf5ed79970b133adc04be',
+        'size_bytes': 45849908,
+        'generation': 1770870617609650,
         'condition': 'host_os == "mac" and host_cpu == "arm64"',
       },
       {
-        'object_name': 'Mac_arm64/clang-tidy-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
-        'sha256sum': '77b5cdea3c51876a87c08a826297c1d113dd5dc298f973af12e98fee3afebd5a',
-        'size_bytes': 12451868,
-        'generation': 1770400195530884,
+        'object_name': 'Mac_arm64/clang-tidy-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': '11dfe174b262142ce8d2d9e810ff2a5c5ac5e44ccb135876825fc3ff24be1b35',
+        'size_bytes': 12473944,
+        'generation': 1770870617763889,
         'condition': 'host_os == "mac" and host_cpu == "arm64" and checkout_clang_tidy',
       },
       {
-        'object_name': 'Mac_arm64/clangd-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
-        'sha256sum': '032f8b1cac0e03703890fca0d0438515b92f36b438fd7d125e2b7bdb05216b3e',
-        'size_bytes': 12855596,
-        'generation': 1770400195729193,
+        'object_name': 'Mac_arm64/clangd-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': 'a9d8847208d6411087464fb9d2f07d1a9a7617821ade0b6e00fb31e29934e12f',
+        'size_bytes': 12883052,
+        'generation': 1770870617857412,
         'condition': 'host_os == "mac" and host_cpu == "arm64" and checkout_clangd',
       },
       {
-        'object_name': 'Mac_arm64/llvm-code-coverage-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
-        'sha256sum': '7d696cdde5bdacde2eafcb3aaf5f7b3123f30998a75b50f84ce99aae57c59977',
-        'size_bytes': 1982584,
-        'generation': 1770400196199062,
+        'object_name': 'Mac_arm64/llvm-code-coverage-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': 'bf15d654a27132d07c5a03d6bb174f79449a88a34a08b9024e72a1b88127bdd0',
+        'size_bytes': 1991372,
+        'generation': 1770870618097417,
         'condition': 'host_os == "mac" and host_cpu == "arm64" and checkout_clang_coverage_tools',
       },
       {
-        'object_name': 'Mac_arm64/llvmobjdump-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
-        'sha256sum': '2656c91e724e5efaf844fdde1b2e14aeaf31bcb0766a7e28aa58f59a09b2b6b2',
-        'size_bytes': 5431708,
-        'generation': 1770400195856360,
+        'object_name': 'Mac_arm64/llvmobjdump-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': 'aea68efe7aeab8eb4d2371e79e7dcb021e1448d8edd0d4851fa34907b93205a6',
+        'size_bytes': 5446112,
+        'generation': 1770870617921094,
         'condition': 'host_os == "mac" and host_cpu == "arm64"',
       },
       {
-        'object_name': 'Win/clang-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
-        'sha256sum': '36d195fd31754af28bb5e9203ea582b67813b2f5a7772f86d5d567ec887067a0',
-        'size_bytes': 49100488,
-        'generation': 1770400217126755,
+        'object_name': 'Win/clang-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': 'cca114f624db433c2ba6458a6de34e6e474bea379f483817dd5dc0a02c59df3c',
+        'size_bytes': 49244312,
+        'generation': 1770870628650736,
         'condition': 'host_os == "win"',
       },
       {
-        'object_name': 'Win/clang-tidy-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
-        'sha256sum': '533a1233533bd97b3cf9b80bf06124b49f981bdde28a8d302b4fb49eb94ba889',
-        'size_bytes': 14441820,
-        'generation': 1770400217463907,
+        'object_name': 'Win/clang-tidy-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': 'de37a5f6e43f7d4ff7c7c8fe0ba7cd89bdd0cbe0370b171263f36ed372a7e3d0',
+        'size_bytes': 14464644,
+        'generation': 1770870628637623,
         'condition': 'host_os == "win" and checkout_clang_tidy',
       },
       {
-        'object_name': 'Win/clang-win-runtime-library-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
-        'sha256sum': 'bed1dee4a3ab4e7cbfad5d33aeb3f5c4caed676de5ba0eb22a8eea69bfc0c670',
-        'size_bytes': 2590736,
-        'generation': 1770400239673699,
+        'object_name': 'Win/clang-win-runtime-library-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': '3f2ad3f2adb92384ecdcb33385aa8beacd7f38d27bed6179e39b2e971469d028',
+        'size_bytes': 2595252,
+        'generation': 1770870636929948,
         'condition': 'checkout_win and not host_os == "win"',
       },
       {
-        'object_name': 'Win/clangd-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
-        'sha256sum': '8c25b517a03977dfaf09e2b5d3ddf8e4602165ce692e43413e3d728e4618ace6',
-        'size_bytes': 14853072,
-        'generation': 1770400217655297,
+        'object_name': 'Win/clangd-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': 'b9f868e6be509a1900e7a8157cd29235245e0847804abb8d20fbc37772fbf802',
+        'size_bytes': 14883160,
+        'generation': 1770870628678782,
         'condition': 'host_os == "win" and checkout_clangd',
       },
       {
-        'object_name': 'Win/llvm-code-coverage-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
-        'sha256sum': '8a6317407a98e3d3f99a30255c8559ff146d605b2d44e0e22d314cbd0c7f0636',
-        'size_bytes': 2466000,
-        'generation': 1770400218121410,
+        'object_name': 'Win/llvm-code-coverage-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': '30cb9e52d92731ed24cf08775aeb5eb0e4d2c1eb871daed1f344f6b4e635fe23',
+        'size_bytes': 2477228,
+        'generation': 1770870628999104,
         'condition': 'host_os == "win" and checkout_clang_coverage_tools',
       },
       {
-        'object_name': 'Win/llvmobjdump-llvmorg-23-init-2224-g5bd8dadb-3.tar.xz',
-        'sha256sum': '9a78b48d45ea671e1eca8a9c35a1e89015505e0cbd33ccf2c29f9740ce1c0957',
-        'size_bytes': 5845536,
-        'generation': 1770400217790811,
+        'object_name': 'Win/llvmobjdump-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': '7baba8d463b92d16c510d67be60eed89897ba76898121ea709264cbb41066af1',
+        'size_bytes': 5859968,
+        'generation': 1770870628750202,
         'condition': '(checkout_linux or checkout_mac or checkout_android) and host_os == "win"',
       },
     ]
@@ -298,31 +298,31 @@ deps = {
     'bucket': 'chromium-browser-clang',
     'objects': [
       {
-        'object_name': 'Linux_x64/rust-toolchain-7d8ebe3128fc87f3da1ad64240e63ccf07b8f0bd-3-llvmorg-23-init-2224-g5bd8dadb.tar.xz',
-        'sha256sum': 'dc8b9057f54cceea940301d0bc8db14565953e9f63b59c4ef0b5e45c14c26a13',
-        'size_bytes': 266591660,
-        'generation': 1770147297953244,
+        'object_name': 'Linux_x64/rust-toolchain-7dc2e92b83be02dc07f87be7e94266d6e48e5ca5-1-llvmorg-23-init-3706-gfc648683.tar.xz',
+        'sha256sum': 'c14a211a9fcef604df768db22a4422553c791c4ad7e3f982cf00b9c0c745ccaa',
+        'size_bytes': 266703680,
+        'generation': 1770870599310471,
         'condition': 'host_os == "linux" and non_git_source',
       },
       {
-        'object_name': 'Mac/rust-toolchain-7d8ebe3128fc87f3da1ad64240e63ccf07b8f0bd-3-llvmorg-23-init-2224-g5bd8dadb.tar.xz',
-        'sha256sum': '89c9d0f5655228059c7b97ca52dc91809b40b664bf5f778ccda1851602da2c36',
-        'size_bytes': 254144048,
-        'generation': 1770147299703374,
+        'object_name': 'Mac/rust-toolchain-7dc2e92b83be02dc07f87be7e94266d6e48e5ca5-1-llvmorg-23-init-3706-gfc648683.tar.xz',
+        'sha256sum': 'dfd5f890b8d84eeee4cf5aa3ba9b9e2e070bb5bece134f979060c35b9286ed00',
+        'size_bytes': 254634176,
+        'generation': 1770870600934341,
         'condition': 'host_os == "mac" and host_cpu == "x64"',
       },
       {
-        'object_name': 'Mac_arm64/rust-toolchain-7d8ebe3128fc87f3da1ad64240e63ccf07b8f0bd-3-llvmorg-23-init-2224-g5bd8dadb.tar.xz',
-        'sha256sum': '86945a58fd938e54eb21528b37502eaa0d6c9877c76d594d123341df8dc7d018',
-        'size_bytes': 237685480,
-        'generation': 1770147301452825,
+        'object_name': 'Mac_arm64/rust-toolchain-7dc2e92b83be02dc07f87be7e94266d6e48e5ca5-1-llvmorg-23-init-3706-gfc648683.tar.xz',
+        'sha256sum': '4e0a8de0ef6e65d17f723193c44c7dc6495f2274e7a7818e9b84e9645281be7e',
+        'size_bytes': 238113184,
+        'generation': 1770870602823865,
         'condition': 'host_os == "mac" and host_cpu == "arm64"',
       },
       {
-        'object_name': 'Win/rust-toolchain-7d8ebe3128fc87f3da1ad64240e63ccf07b8f0bd-3-llvmorg-23-init-2224-g5bd8dadb.tar.xz',
-        'sha256sum': 'c65ee3f45c289decb58d2ae1863c9db72921518e5b9063c3271b5edbd131d76b',
-        'size_bytes': 402682036,
-        'generation': 1770152314303019,
+        'object_name': 'Win/rust-toolchain-7dc2e92b83be02dc07f87be7e94266d6e48e5ca5-1-llvmorg-23-init-3706-gfc648683.tar.xz',
+        'sha256sum': '1c783f691498d98769211f427fb3c6ef483ec1b710c6e661614f9bcec6a71c3d',
+        'size_bytes': 404347608,
+        'generation': 1770870604323594,
         'condition': 'host_os == "win"',
       },
     ],
@@ -331,15 +331,15 @@ deps = {
   'src/third_party/clang-format/script':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@c2725e0622e1a86d55f14514f2177a39efea4a0e',
   'src/third_party/compiler-rt/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@7a0968b56bfaafaf09775f0b2f06a79fc47c83b2',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@57d67767dc005a7edaba2b1714c436393e41e4b9',
   'src/third_party/libc++/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@7ab65651aed6802d2599dcb7a73b1f82d5179d05',
   'src/third_party/libc++abi/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@8f11bb1d4438d0239d0dfc1bd9456a9f31629dda',
   'src/third_party/llvm-libc/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@2f7df07563d93f442b05a711769e2613715ad279',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@7061b80a014afdad13ce434000f4810a6e803eb0',
   'src/third_party/libunwind/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@ba19d93d6d4f467fba11ff20fe2fc7c056f79345',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@17ccf7d110c5526cb77e93cfd8330f491fb2bf18',
 
   'src/third_party/test_fonts/test_fonts': {
       'dep_type': 'gcs',
@@ -488,11 +488,11 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@932fb7389fbd5438227c05e53a345e3e4dfc2a9a',
+    'https://boringssl.googlesource.com/boringssl.git@17b020f428de76ce2ce0dafbb119f8e539c0a877',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@79099fdf668ae097c9eca7052fd5c4c5de6c9098',
   'src/third_party/catapult':
-    'https://chromium.googlesource.com/catapult.git@4fe7ca3037f5f5e7227f0f0d0d6eae98d0ebf840',
+    'https://chromium.googlesource.com/catapult.git@172c8e1ba5c324616c708f1ae82b37621365c22f',
   'src/third_party/ced/src': {
     'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5',
   },
@@ -505,13 +505,13 @@ deps = {
   'src/third_party/crc32c/src':
     'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558',
   'src/third_party/depot_tools':
-    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@6c6a6572eb0e27efd41f8724b4631ac2aa9df460',
+    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@5f1a0a82b50523fb6ec388b4287a4e26781ed928',
   'src/third_party/ffmpeg':
-    'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@ae11d2ba5c835b822a61d6a99eeb853ca30d41d8',
+    'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@946d97db8d906277085e361892b7efda5152e2f1',
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@f9b8a88e84441a237414dce860e3f6214a64f8bf',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@e6e7934770072cf0e97202de5b991ed580804118',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -573,7 +573,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/kotlin_stdlib',
-              'version': 'FfVJUc4q9S6JB4uqVGPJQ3anrc1Y_W0O-oGK_2MJA5gC',
+              'version': 'JcCPEPsUevwiZ519BkXHvRR8J-9XEcwj_plAWjVIR7MC',
           },
       ],
       'condition': 'checkout_android',
@@ -593,7 +593,7 @@ deps = {
   'src/third_party/libFuzzer/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt/lib/fuzzer.git@bea408a6e01f0f7e6c82a43121fe3af4506c932e',
   'src/third_party/fuzztest/src':
-    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@fb29fbd0058c84ed0ae7b3b50ad2909a68fd0360',
+    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@4aaf69f8f81667bbd4a3b09e99d289fb52fd13ce',
   'src/third_party/libjpeg_turbo':
     'https://chromium.googlesource.com/chromium/deps/libjpeg_turbo.git@6bb85251a8382b5e07f635a981ac685cc5ab5053',
   'src/third_party/libsrtp':
@@ -601,7 +601,7 @@ deps = {
   'src/third_party/dav1d/libdav1d':
     'https://chromium.googlesource.com/external/github.com/videolan/dav1d.git@b546257f770768b2c88258c533da38b91a06f737',
   'src/third_party/libaom/source/libaom':
-    'https://aomedia.googlesource.com/aom.git@37b2db4334d3b90c3b3b6e188659a2214a411907',
+    'https://aomedia.googlesource.com/aom.git@a27d804206a8972cb29c48d5903d308a52bc3138',
   'src/third_party/libgav1/src':
     Var('chromium_git') + '/codecs/libgav1.git' + '@' + '40f58ed32ff39071c3f2a51056dbc49a070af0dc',
   'src/third_party/libunwindstack': {
@@ -609,13 +609,13 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '7475cfd1038ac63f0884e1f096fce054fa01c3b3',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'd374841f939d443e7d5ad3cde2c7060e55524cb2',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
     'https://chromium.googlesource.com/webm/libvpx.git@a0ac06f5d3c83f890cb502c80470ef0502343fec',
   'src/third_party/libyuv':
-    'https://chromium.googlesource.com/libyuv/libyuv.git@917276084a49be726c90292ff0a6b0a3d571a6af',
+    'https://chromium.googlesource.com/libyuv/libyuv.git@6067afde563c3946eebd94f146b3824ab7a97a9c',
   'src/third_party/lss': {
     'url': 'https://chromium.googlesource.com/linux-syscall-support.git@29164a80da4d41134950d76d55199ea33fbb9613',
     'condition': 'checkout_android or checkout_linux',
@@ -644,7 +644,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/r8',
-              'version': 'YvVZKKYTZDYMrvAXwMFbMGMUQaJJMb5LIirLrGcOLS0C',
+              'version': 'dKavKsYt7Uj8fRmoAALj-uumULBl43j-NHRUZ1YvnxUC',
           },
       ],
       'condition': 'checkout_android',
@@ -657,7 +657,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/r8',
-              'version': 'a4fVqbIycCDqs1714SLRqxEdz6P-sH-z1QT_eeeF0PcC',
+              'version': 'dKavKsYt7Uj8fRmoAALj-uumULBl43j-NHRUZ1YvnxUC',
           },
       ],
       'condition': 'checkout_android',
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@1f457fb24256f08614f80b11a1764c5ebf5188a0',
+    'https://chromium.googlesource.com/chromium/src/tools@f7ad2349e57f3fed9217fca06a1764a5766b6658',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'Cl-ySzwc8_wt7xkWL9dKKHbvVRhOHTgGANeH3JHEB3cC',
+          'version': 'XpThiLpVJwtbiGf8RwS7hzulYyS7Q29MXabSOIrsIX4C',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -718,7 +718,7 @@ deps = {
       'packages': [
           {
                'package': 'chromium/third_party/android_build_tools/manifest_merger',
-               'version': 'xxaa38R0ED95Wxo8P29BC-mB7bjFPi2HKMpiIJn-UhAC',
+               'version': 'yPYRRYty6hqUJSOO_UdrYg31nwC-gaiPQTBBMXV8Y-wC',
           },
       ],
       'condition': 'checkout_android',
@@ -786,7 +786,7 @@ deps = {
   },
 
   'src/third_party/tflite/src':
-    Var('chromium_git') + '/external/github.com/tensorflow/tensorflow.git' + '@' + '01e030d23d3b904d98cbf908da74d63b3c186949',
+    Var('chromium_git') + '/external/github.com/tensorflow/tensorflow.git' + '@' + '38dd09551686ad00bc837f383b3a1db282079498',
 
   'src/third_party/turbine/cipd': {
       'packages': [
@@ -808,15 +808,15 @@ deps = {
       'packages': [
         {
           'package': 'infra/tools/luci/cas/${{platform}}',
-          'version': 'git_revision:072101cbfec3372b812ff510df8547d7b4187bea',
+          'version': 'git_revision:4d646a5cce1959bdfb88c838b1044354c37f0655',
         },
         {
           'package': 'infra/tools/luci/isolate/${{platform}}',
-          'version': 'git_revision:072101cbfec3372b812ff510df8547d7b4187bea',
+          'version': 'git_revision:4d646a5cce1959bdfb88c838b1044354c37f0655',
         },
         {
           'package': 'infra/tools/luci/swarming/${{platform}}',
-          'version': 'git_revision:072101cbfec3372b812ff510df8547d7b4187bea',
+          'version': 'git_revision:4d646a5cce1959bdfb88c838b1044354c37f0655',
         }
       ],
       'dep_type': 'cipd',
@@ -842,7 +842,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/android_deps/autorolled',
-              'version': 'RiKmV-EpMhg5ibpMbPpENQmfU4IPLpo8cxer4F3NSUgC',
+              'version': 'Lskjo0vFA222MYDpjYl71PwoWG_9mb-F_v3RTHvE-LkC',
           },
       ],
       'condition': 'checkout_android and non_git_source',
@@ -852,19 +852,19 @@ deps = {
     Var('chromium_git') + '/external/github.com/google/pthreadpool.git' + '@' + '9003ee6c137cea3b94161bd5c614fb43be523ee1',
 
   'src/third_party/xnnpack/src':
-    Var('chromium_git') + '/external/github.com/google/XNNPACK.git' + '@' + '4574c4d9b00703c15f2218634ddf101597b22b18',
+    Var('chromium_git') + '/external/github.com/google/XNNPACK.git' + '@' + 'e8aad3cc4ffb7259c4aab6dfb922bfbef0ef8be0',
 
   'src/third_party/farmhash/src':
     Var('chromium_git') + '/external/github.com/google/farmhash.git' + '@' + '816a4ae622e964763ca0862d9dbd19324a1eaf45',
 
   'src/third_party/ruy/src':
-    Var('chromium_git') + '/external/github.com/google/ruy.git' + '@' + '576e020f06334118994496b45f9796ed7fda3280',
+    Var('chromium_git') + '/external/github.com/google/ruy.git' + '@' + '2af88863614a8298689cc52b1a47b3fcad7be835',
 
   'src/third_party/cpuinfo/src':
-    Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '84818a41e074779dbb00521a4731d3e14160ff15',
+    Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7364b490b5f78d58efe23ea76e74210fd6c3c76f',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '23fcc1c6c9726ef212701971103b18cd1936ae9b',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + 'bdec88009d3124c7924b70a3fe4a1a6d33d58966',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From 6713f183a8736461d3d515eb49d9523305a7f0d9 Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Thu, 19 Feb 2026 13:58:36 +0100
Subject: [PATCH 101/472] Delete ClearSamples helper in favor or absl::c_fill

This function doesn't add much on top a more standard function. This
function may cause a compilation failure when it is called before
defined and ArrayView is replaced with std::span as currently it works
because uses subtle argument dependent lookup rules.

Bug: webrtc:439801349
Change-Id: I0627c53594b5e9eb85446f9050dfbc5e807e25fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/450580
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46954}
---
 api/audio/BUILD.gn                         |  1 +
 api/audio/audio_frame.cc                   |  7 ++---
 api/audio/audio_view.h                     | 19 ++------------
 api/audio/test/audio_view_unittest.cc      | 30 ----------------------
 modules/audio_coding/BUILD.gn              |  1 +
 modules/audio_coding/acm2/acm_resampler.cc |  3 ++-
 6 files changed, 10 insertions(+), 51 deletions(-)

diff --git a/api/audio/BUILD.gn b/api/audio/BUILD.gn
index a5e6ce506d..db356e6b56 100644
--- a/api/audio/BUILD.gn
+++ b/api/audio/BUILD.gn
@@ -53,6 +53,7 @@ rtc_library("audio_frame_api") {
     "../../rtc_base:checks",
     "../../rtc_base:logging",
     "../../rtc_base:timeutils",
+    "//third_party/abseil-cpp/absl/algorithm:container",
   ]
 }
 
diff --git a/api/audio/audio_frame.cc b/api/audio/audio_frame.cc
index b99d8304d3..b894673ebf 100644
--- a/api/audio/audio_frame.cc
+++ b/api/audio/audio_frame.cc
@@ -14,6 +14,7 @@
 #include <cstring>
 #include <optional>
 
+#include "absl/algorithm/container.h"
 #include "api/array_view.h"
 #include "api/audio/audio_view.h"
 #include "api/audio/channel_layout.h"
@@ -102,7 +103,7 @@ void AudioFrame::CopyFrom(const AudioFrame& src) {
     // copying over new values. If we don't, msan might complain in some tests.
     // Consider locking down construction, avoiding the default constructor and
     // prefering construction that initializes all state.
-    ClearSamples(data_);
+    absl::c_fill(data_, 0);
   }
 
   timestamp_ = src.timestamp_;
@@ -147,7 +148,7 @@ int16_t* AudioFrame::mutable_data() {
   // Consider instead if we should rather zero the buffer when `muted_` is set
   // to `true`.
   if (muted_) {
-    ClearSamples(data_);
+    absl::c_fill(data_, 0);
     muted_ = false;
   }
   return &data_[0];
@@ -170,7 +171,7 @@ InterleavedView<int16_t> AudioFrame::mutable_data(size_t samples_per_channel,
   // Consider instead if we should rather zero the whole buffer when `muted_` is
   // set to `true`.
   if (muted_) {
-    ClearSamples(data_, total_samples);
+    absl::c_fill_n(data_, total_samples, 0);
     muted_ = false;
   }
   samples_per_channel_ = samples_per_channel;
diff --git a/api/audio/audio_view.h b/api/audio/audio_view.h
index c91e11fbf6..b35c95e5a5 100644
--- a/api/audio/audio_view.h
+++ b/api/audio/audio_view.h
@@ -16,6 +16,7 @@
 #include <variant>
 #include <vector>
 
+#include "absl/algorithm/container.h"
 #include "api/array_view.h"
 #include "rtc_base/checks.h"
 
@@ -211,7 +212,7 @@ class DeinterleavedView {
   void Clear() {
     for (size_t i = 0u; i < num_channels_; ++i) {
       MonoView<T> view = (*this)[i];
-      ClearSamples(view);
+      absl::c_fill(view, 0);
     }
   }
 
@@ -308,22 +309,6 @@ void CopySamples(D& destination, const S& source) {
          source.size() * sizeof(typename S::value_type));
 }
 
-// Sets all the samples in a view to 0. This template function is a simple
-// wrapper around `memset()` but adds the benefit of automatically calculating
-// the byte size from the number of samples and sample type.
-template <typename T>
-void ClearSamples(T& view) {
-  memset(&view[0], 0, view.size() * sizeof(typename T::value_type));
-}
-
-// Same as `ClearSamples()` above but allows for clearing only the first
-// `sample_count` number of samples.
-template <typename T>
-void ClearSamples(T& view, size_t sample_count) {
-  RTC_DCHECK_LE(sample_count, view.size());
-  memset(&view[0], 0, sample_count * sizeof(typename T::value_type));
-}
-
 }  // namespace webrtc
 
 #endif  // API_AUDIO_AUDIO_VIEW_H_
diff --git a/api/audio/test/audio_view_unittest.cc b/api/audio/test/audio_view_unittest.cc
index 57567b1965..4ac914f1bc 100644
--- a/api/audio/test/audio_view_unittest.cc
+++ b/api/audio/test/audio_view_unittest.cc
@@ -165,36 +165,6 @@ TEST(AudioViewTest, CopySamples) {
   }
 }
 
-TEST(AudioViewTest, ClearSamples) {
-  std::array<int16_t, 100u> samples = {};
-  FillBuffer(ArrayView<int16_t>(samples));
-  ASSERT_NE(samples[0], 0);
-  ClearSamples(samples);
-  for (const auto s : samples) {
-    ASSERT_EQ(s, 0);
-  }
-
-  std::array<float, 100u> samples_f = {};
-  FillBuffer(ArrayView<float>(samples_f));
-  ASSERT_NE(samples_f[0], 0.0);
-  ClearSamples(samples_f);
-  for (const auto s : samples_f) {
-    ASSERT_EQ(s, 0.0);
-  }
-
-  // Clear only half of the buffer
-  FillBuffer(ArrayView<int16_t>(samples));
-  const auto half_way = samples.size() / 2;
-  ClearSamples(samples, half_way);
-  for (size_t i = 0u; i < samples.size(); ++i) {
-    if (i < half_way) {
-      ASSERT_EQ(samples[i], 0);
-    } else {
-      ASSERT_NE(samples[i], 0);
-    }
-  }
-}
-
 TEST(AudioViewTest, DeinterleavedViewPointerArray) {
   // Create vectors of varying sizes to guarantee that they don't end up
   // aligned in memory.
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index 01a10e3dac..016485c4e8 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -57,6 +57,7 @@ rtc_library("audio_coding") {
     "../../rtc_base/synchronization:mutex",
     "../../system_wrappers",
     "../../system_wrappers:metrics",
+    "//third_party/abseil-cpp/absl/algorithm:container",
     "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/strings:string_view",
   ]
diff --git a/modules/audio_coding/acm2/acm_resampler.cc b/modules/audio_coding/acm2/acm_resampler.cc
index 9daa04d14b..a8c4ba8ef0 100644
--- a/modules/audio_coding/acm2/acm_resampler.cc
+++ b/modules/audio_coding/acm2/acm_resampler.cc
@@ -13,6 +13,7 @@
 #include <array>
 #include <cstdint>
 
+#include "absl/algorithm/container.h"
 #include "api/audio/audio_frame.h"
 #include "api/audio/audio_view.h"
 #include "rtc_base/checks.h"
@@ -21,7 +22,7 @@ namespace webrtc {
 namespace acm2 {
 
 ResamplerHelper::ResamplerHelper() {
-  ClearSamples(last_audio_buffer_);
+  absl::c_fill(last_audio_buffer_, 0);
 }
 
 bool ResamplerHelper::MaybeResample(int desired_sample_rate_hz,

From 781f126b3044788a7d31d2ce68d638a169128f5e Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Thu, 19 Feb 2026 12:12:54 -0800
Subject: [PATCH 102/472] Roll chromium_revision 1063bfbb8a..7fd541abbf
 (1587147:1587331)

Change log: https://chromium.googlesource.com/chromium/src/+log/1063bfbb8a..7fd541abbf
Full diff: https://chromium.googlesource.com/chromium/src/+/1063bfbb8a..7fd541abbf

Changed dependencies
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/e2c585cccd..d99506c125
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/c5cf2d34d5..0d019ad639
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/e794b55e51..0fa89b56df
* src/third_party/android_build_tools/manifest_merger/cipd: yPYRRYty6hqUJSOO_UdrYg31nwC-gaiPQTBBMXV8Y-wC..lHLzbS87smMbtV1VPJa5UWm59SS7t7Q1PRsEjIKbl1sC
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/e6e7934770..ce8dc971ec
* src/third_party/kotlinc/current: KZWh3uhuenRgsoGqNYgSo82FO3sMwZjvmuR9TBvanTcC..1wGbWZPhnUbDtQCvcw74lP1g0ceq55pEkvO5hZ3DRLcC
* src/third_party/libaom/source/libaom: https://aomedia.googlesource.com/aom.git/+log/a27d804206..39606bf4ba
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/d374841f93..ea00d77a04
* src/third_party/r8/cipd: dKavKsYt7Uj8fRmoAALj-uumULBl43j-NHRUZ1YvnxUC..ygoUIlS4O_2F8NuFV3YNhN0oBwVMSgi9mipaKA9uaysC
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/f7ad2349e5..06ff38fb8a
DEPS diff: https://chromium.googlesource.com/chromium/src/+/1063bfbb8a..7fd541abbf/DEPS

BUG=None

Change-Id: I0d6280321a8520820045a5528dc6fb6a478cce6d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/450700
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#46955}
---
 DEPS | 22 +++++++++++-----------
 1 file changed, 11 insertions(+), 11 deletions(-)

diff --git a/DEPS b/DEPS
index 29501525ec..98037d08c4 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '1063bfbb8aea8b53ec93218deecbf3fc00aff8af',
+  'chromium_revision': '7fd541abbffce94b7885d5500fba1ff5a6ed8dd0',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@e2c585cccd76198491dbc6dcb65471aad488acca',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@d99506c125af14f013cf39eb802a7221d016898f',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@c5cf2d34d56c6bc6177cf85439578638bb3535c9',
+    'https://chromium.googlesource.com/chromium/src/testing@0d019ad63957abf507eaf4dd0c365a55bdba5c12',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@e794b55e5118051f0676641942e16c62686d7bc7',
+    'https://chromium.googlesource.com/chromium/src/third_party@0fa89b56dfed6afb8e6bcceb52823bec462b7014',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -511,7 +511,7 @@ deps = {
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@e6e7934770072cf0e97202de5b991ed580804118',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@ce8dc971ecd9451f8a462bd3d91b5c4a652a1f5c',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -584,7 +584,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/kotlinc',
-              'version': 'KZWh3uhuenRgsoGqNYgSo82FO3sMwZjvmuR9TBvanTcC',
+              'version': '1wGbWZPhnUbDtQCvcw74lP1g0ceq55pEkvO5hZ3DRLcC',
           },
       ],
       'condition': 'checkout_android',
@@ -601,7 +601,7 @@ deps = {
   'src/third_party/dav1d/libdav1d':
     'https://chromium.googlesource.com/external/github.com/videolan/dav1d.git@b546257f770768b2c88258c533da38b91a06f737',
   'src/third_party/libaom/source/libaom':
-    'https://aomedia.googlesource.com/aom.git@a27d804206a8972cb29c48d5903d308a52bc3138',
+    'https://aomedia.googlesource.com/aom.git@39606bf4bae2befef8c32bcdfab402226bb33783',
   'src/third_party/libgav1/src':
     Var('chromium_git') + '/codecs/libgav1.git' + '@' + '40f58ed32ff39071c3f2a51056dbc49a070af0dc',
   'src/third_party/libunwindstack': {
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'd374841f939d443e7d5ad3cde2c7060e55524cb2',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'ea00d77a04e88bff7613469a0bec5e7d7b899663',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -644,7 +644,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/r8',
-              'version': 'dKavKsYt7Uj8fRmoAALj-uumULBl43j-NHRUZ1YvnxUC',
+              'version': 'ygoUIlS4O_2F8NuFV3YNhN0oBwVMSgi9mipaKA9uaysC',
           },
       ],
       'condition': 'checkout_android',
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@f7ad2349e57f3fed9217fca06a1764a5766b6658',
+    'https://chromium.googlesource.com/chromium/src/tools@06ff38fb8a89653b1a6144e3f93ded9335ceaac2',
 
   'src/third_party/espresso': {
       'packages': [
@@ -718,7 +718,7 @@ deps = {
       'packages': [
           {
                'package': 'chromium/third_party/android_build_tools/manifest_merger',
-               'version': 'yPYRRYty6hqUJSOO_UdrYg31nwC-gaiPQTBBMXV8Y-wC',
+               'version': 'lHLzbS87smMbtV1VPJa5UWm59SS7t7Q1PRsEjIKbl1sC',
           },
       ],
       'condition': 'checkout_android',

From 225f036418a4a783e996e8e8ff7720a490f95c48 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Thu, 19 Feb 2026 16:03:36 -0800
Subject: [PATCH 103/472] Roll chromium_revision 7fd541abbf..3d3530d666
 (1587331:1587490)

Change log: https://chromium.googlesource.com/chromium/src/+log/7fd541abbf..3d3530d666
Full diff: https://chromium.googlesource.com/chromium/src/+/7fd541abbf..3d3530d666

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/94f11bf92f..0b1d3a05db
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/d99506c125..647c989972
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/0d019ad639..78a04c4e96
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/0fa89b56df..a01b284c6f
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/17b020f428..14fa4996cb
* src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/57d67767dc..6813553ab8
* src/third_party/fuzztest/src: https://chromium.googlesource.com/external/github.com/google/fuzztest.git/+log/4aaf69f8f8..7636e7cbf8
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/ce8dc971ec..f7931aae14
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/ea00d77a04..8395cf4efe
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/06ff38fb8a..c257779d8a
DEPS diff: https://chromium.googlesource.com/chromium/src/+/7fd541abbf..3d3530d666/DEPS

BUG=None

Change-Id: Ibe3f2d3e7ea7345b342aa8ab5855a9fefbbdd7bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/450740
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#46956}
---
 DEPS | 22 +++++++++++-----------
 1 file changed, 11 insertions(+), 11 deletions(-)

diff --git a/DEPS b/DEPS
index 98037d08c4..06f8b9a1a4 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '7fd541abbffce94b7885d5500fba1ff5a6ed8dd0',
+  'chromium_revision': '3d3530d6663a37252ef3705b0f4080a06d298862',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@94f11bf92fb777c6153a331d756f6d4cf2ee3e9f',
+    'https://chromium.googlesource.com/chromium/src/build@0b1d3a05db11e2e95622e80980cf5d4dd8029805',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@451e71432ac9f6a91d4d5604c29e0a5078e8c2e8',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@d99506c125af14f013cf39eb802a7221d016898f',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@647c9899724bcf191376b47106bc980402079f75',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@0d019ad63957abf507eaf4dd0c365a55bdba5c12',
+    'https://chromium.googlesource.com/chromium/src/testing@78a04c4e96cd034b9334e0b4e94e97dbf9737d75',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@0fa89b56dfed6afb8e6bcceb52823bec462b7014',
+    'https://chromium.googlesource.com/chromium/src/third_party@a01b284c6fe4bc2648e911f1018e0b08f2b8c9cb',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -331,7 +331,7 @@ deps = {
   'src/third_party/clang-format/script':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@c2725e0622e1a86d55f14514f2177a39efea4a0e',
   'src/third_party/compiler-rt/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@57d67767dc005a7edaba2b1714c436393e41e4b9',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@6813553ab8756887eb113da9eba4fb2163f5b8a5',
   'src/third_party/libc++/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@7ab65651aed6802d2599dcb7a73b1f82d5179d05',
   'src/third_party/libc++abi/src':
@@ -488,7 +488,7 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@17b020f428de76ce2ce0dafbb119f8e539c0a877',
+    'https://boringssl.googlesource.com/boringssl.git@14fa4996cb794af518b18d3a41733153f5f5f2f8',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@79099fdf668ae097c9eca7052fd5c4c5de6c9098',
   'src/third_party/catapult':
@@ -511,7 +511,7 @@ deps = {
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@ce8dc971ecd9451f8a462bd3d91b5c4a652a1f5c',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@f7931aae144a330cd38912315eae2417de1fb614',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -593,7 +593,7 @@ deps = {
   'src/third_party/libFuzzer/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt/lib/fuzzer.git@bea408a6e01f0f7e6c82a43121fe3af4506c932e',
   'src/third_party/fuzztest/src':
-    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@4aaf69f8f81667bbd4a3b09e99d289fb52fd13ce',
+    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@7636e7cbf8281e049950f7c229fce46e52011c21',
   'src/third_party/libjpeg_turbo':
     'https://chromium.googlesource.com/chromium/deps/libjpeg_turbo.git@6bb85251a8382b5e07f635a981ac685cc5ab5053',
   'src/third_party/libsrtp':
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'ea00d77a04e88bff7613469a0bec5e7d7b899663',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '8395cf4efec05c454b9f60f080aab5ce5277a730',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@06ff38fb8a89653b1a6144e3f93ded9335ceaac2',
+    'https://chromium.googlesource.com/chromium/src/tools@c257779d8af09f8470d6a4e711f1e0d64b131d1f',
 
   'src/third_party/espresso': {
       'packages': [

From 23e3744c0565477a80cd4f1ae11b5b5f98d8dc3e Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Thu, 19 Feb 2026 20:05:16 -0800
Subject: [PATCH 104/472] Update WebRTC code version (2026-02-20T04:04:46).

Bug: None
Change-Id: I144bc301cfec9c9353d7366b8c8163f4b32ab29e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/450780
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#46957}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index 1a32e3a4e1..6bf3db7608 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-02-19T04:04:25";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-02-20T04:04:46";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From 33d3cea2ce5725d9fac224e9456794e56af35187 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Erik=20Spr=C3=A5ng?= <sprang@webrtc.org>
Date: Wed, 18 Feb 2026 13:49:25 +0100
Subject: [PATCH 105/472] Add OnFrameDropped stubs for remaining
 EncodedImageCallbacks.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This adds noop implementations for the `OnFrameDropped` method in all
implementations of EncodedImageCallback in libwebrtc. This in in
preparation of eventually making it pure virtual.

Bug: webrtc:467444018
Change-Id: Ie45255bec2dbe001de9dcab24b0222fee0514738
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/450260
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46958}
---
 .../test/video_encoder_software_fallback_wrapper_unittest.cc | 4 ++++
 call/rtp_video_sender.cc                                     | 4 ++++
 call/rtp_video_sender.h                                      | 4 ++++
 .../video_coding/codecs/test/encoded_video_frame_producer.cc | 4 ++++
 modules/video_coding/codecs/test/video_codec_unittest.h      | 4 ++++
 modules/video_coding/codecs/test/videoprocessor.h            | 4 ++++
 modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc    | 4 ++++
 modules/video_coding/utility/simulcast_test_fixture_impl.cc  | 5 +++++
 rtc_tools/video_encoder/video_encoder.cc                     | 4 ++++
 test/testsupport/ivf_video_frame_generator_unittest.cc       | 4 ++++
 test/video_codec_tester.cc                                   | 4 ++++
 11 files changed, 45 insertions(+)

diff --git a/api/video_codecs/test/video_encoder_software_fallback_wrapper_unittest.cc b/api/video_codecs/test/video_encoder_software_fallback_wrapper_unittest.cc
index bdb388e6ec..542b194407 100644
--- a/api/video_codecs/test/video_encoder_software_fallback_wrapper_unittest.cc
+++ b/api/video_codecs/test/video_encoder_software_fallback_wrapper_unittest.cc
@@ -90,6 +90,10 @@ class FakeEncodedImageCallback : public EncodedImageCallback {
     ++callback_count_;
     return Result(Result::OK, callback_count_);
   }
+
+  void OnFrameDropped(uint32_t /*rtp_timestamp*/,
+                      int /*spatial_id*/,
+                      bool /*is_end_of_temporal_unit*/) override {}
   int callback_count_ = 0;
 };
 
diff --git a/call/rtp_video_sender.cc b/call/rtp_video_sender.cc
index 9044105539..c25fe194bf 100644
--- a/call/rtp_video_sender.cc
+++ b/call/rtp_video_sender.cc
@@ -636,6 +636,10 @@ EncodedImageCallback::Result RtpVideoSender::OnEncodedImage(
   return Result(Result::OK, rtp_timestamp);
 }
 
+void RtpVideoSender::OnFrameDropped(uint32_t /*rtp_timestamp*/,
+                                    int /*spatial_id*/,
+                                    bool /*is_end_of_temporal_unit*/) {}
+
 void RtpVideoSender::OnBitrateAllocationUpdated(
     const VideoBitrateAllocation& bitrate) {
   RTC_DCHECK_RUN_ON(&transport_checker_);
diff --git a/call/rtp_video_sender.h b/call/rtp_video_sender.h
index 77f60ef7a6..ca7d0989d1 100644
--- a/call/rtp_video_sender.h
+++ b/call/rtp_video_sender.h
@@ -138,6 +138,10 @@ class RtpVideoSender : public RtpVideoSenderInterface,
       const CodecSpecificInfo* codec_specific_info)
       RTC_LOCKS_EXCLUDED(mutex_) override;
 
+  void OnFrameDropped(uint32_t rtp_timestamp,
+                      int spatial_id,
+                      bool is_end_of_temporal_unit) override;
+
   void OnBitrateAllocationUpdated(const VideoBitrateAllocation& bitrate)
       RTC_LOCKS_EXCLUDED(mutex_) override;
   void OnVideoLayersAllocationUpdated(
diff --git a/modules/video_coding/codecs/test/encoded_video_frame_producer.cc b/modules/video_coding/codecs/test/encoded_video_frame_producer.cc
index 312605c1dc..289b61a792 100644
--- a/modules/video_coding/codecs/test/encoded_video_frame_producer.cc
+++ b/modules/video_coding/codecs/test/encoded_video_frame_producer.cc
@@ -42,6 +42,10 @@ class EncoderCallback : public EncodedImageCallback {
     return Result(Result::Error::OK);
   }
 
+  void OnFrameDropped(uint32_t /*rtp_timestamp*/,
+                      int /*spatial_id*/,
+                      bool /*is_end_of_temporal_unit*/) override {}
+
   std::vector<EncodedVideoFrameProducer::EncodedFrame>& output_frames_;
 };
 
diff --git a/modules/video_coding/codecs/test/video_codec_unittest.h b/modules/video_coding/codecs/test/video_codec_unittest.h
index fea4ba0d72..e24ecf0046 100644
--- a/modules/video_coding/codecs/test/video_codec_unittest.h
+++ b/modules/video_coding/codecs/test/video_codec_unittest.h
@@ -56,6 +56,10 @@ class VideoCodecUnitTest : public ::testing::Test {
         const EncodedImage& frame,
         const CodecSpecificInfo* codec_specific_info) override;
 
+    void OnFrameDropped(uint32_t /*rtp_timestamp*/,
+                        int /*spatial_id*/,
+                        bool /*is_end_of_temporal_unit*/) override {}
+
    private:
     VideoCodecUnitTest* const test_;
   };
diff --git a/modules/video_coding/codecs/test/videoprocessor.h b/modules/video_coding/codecs/test/videoprocessor.h
index 200d5bb154..ed64172008 100644
--- a/modules/video_coding/codecs/test/videoprocessor.h
+++ b/modules/video_coding/codecs/test/videoprocessor.h
@@ -120,6 +120,10 @@ class VideoProcessor {
       return Result(Result::OK, 0);
     }
 
+    void OnFrameDropped(uint32_t /*rtp_timestamp*/,
+                        int /*spatial_id*/,
+                        bool /*is_end_of_temporal_unit*/) override {}
+
    private:
     VideoProcessor* const video_processor_;
     TaskQueueBase* const task_queue_;
diff --git a/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc b/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
index 58d77fe15b..9e0ceb3152 100644
--- a/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
+++ b/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
@@ -949,6 +949,10 @@ TEST_P(TestVp8ImplWithMaxFrameDropTrial, EnforcesMaxFrameDropInterval) {
       return Result(Result::Error::OK);
     }
 
+    void OnFrameDropped(uint32_t /*rtp_timestamp*/,
+                        int /*spatial_id*/,
+                        bool /*is_end_of_temporal_unit*/) override {}
+
    private:
     std::vector<TimeDelta> callback_deltas_;
     Timestamp last_callback_;
diff --git a/modules/video_coding/utility/simulcast_test_fixture_impl.cc b/modules/video_coding/utility/simulcast_test_fixture_impl.cc
index 4abaf99cc4..8ea24fe3ea 100644
--- a/modules/video_coding/utility/simulcast_test_fixture_impl.cc
+++ b/modules/video_coding/utility/simulcast_test_fixture_impl.cc
@@ -124,6 +124,11 @@ class SimulcastTestFixtureImpl::TestEncodedImageCallback
     }
     return Result(Result::OK, encoded_image.RtpTimestamp());
   }
+
+  void OnFrameDropped(uint32_t /*rtp_timestamp*/,
+                      int /*spatial_id*/,
+                      bool /*is_end_of_temporal_unit*/) override {}
+
   // This method only makes sense for VP8.
   void GetLastEncodedFrameInfo(int* temporal_layer,
                                bool* layer_sync,
diff --git a/rtc_tools/video_encoder/video_encoder.cc b/rtc_tools/video_encoder/video_encoder.cc
index abadce0f43..bab90e522c 100644
--- a/rtc_tools/video_encoder/video_encoder.cc
+++ b/rtc_tools/video_encoder/video_encoder.cc
@@ -342,6 +342,10 @@ class BitstreamProcessor final : public EncodedImageCallback,
     return Result(Result::Error::OK);
   }
 
+  void OnFrameDropped(uint32_t /*rtp_timestamp*/,
+                      int /*spatial_id*/,
+                      bool /*is_end_of_temporal_unit*/) override {}
+
   VideoCodec video_codec_setting_;
   int32_t frames_ = 0;
 
diff --git a/test/testsupport/ivf_video_frame_generator_unittest.cc b/test/testsupport/ivf_video_frame_generator_unittest.cc
index eb9b9400e4..229d904daf 100644
--- a/test/testsupport/ivf_video_frame_generator_unittest.cc
+++ b/test/testsupport/ivf_video_frame_generator_unittest.cc
@@ -95,6 +95,10 @@ class IvfFileWriterEncodedCallback : public EncodedImageCallback {
     return Result(Result::Error::OK);
   }
 
+  void OnFrameDropped(uint32_t /*rtp_timestamp*/,
+                      int /*spatial_id*/,
+                      bool /*is_end_of_temporal_unit*/) override {}
+
   bool WaitForExpectedFramesReceived(TimeDelta timeout) {
     return expected_frames_count_received_.Wait(timeout);
   }
diff --git a/test/video_codec_tester.cc b/test/video_codec_tester.cc
index 773528ab27..78b632f0a0 100644
--- a/test/video_codec_tester.cc
+++ b/test/video_codec_tester.cc
@@ -1086,6 +1086,10 @@ class Encoder : public EncodedImageCallback {
     ScalabilityMode scalability_mode;
   };
 
+  void OnFrameDropped(uint32_t /*rtp_timestamp*/,
+                      int /*spatial_id*/,
+                      bool /*is_end_of_temporal_unit*/) override {}
+
   Result OnEncodedImage(const EncodedImage& encoded_frame,
                         const CodecSpecificInfo* codec_specific_info) override {
     analyzer_->FinishEncode(encoded_frame);

From 173784f5e554220c15912aaf33c1ed7f6c12ec07 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Fri, 20 Feb 2026 00:02:53 -0800
Subject: [PATCH 106/472] Roll chromium_revision 3d3530d666..085c3ffcf5
 (1587490:1587653)

Change log: https://chromium.googlesource.com/chromium/src/+log/3d3530d666..085c3ffcf5
Full diff: https://chromium.googlesource.com/chromium/src/+/3d3530d666..085c3ffcf5

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/0b1d3a05db..33652fea89
* src/buildtools/linux64: git_revision:318ec1d2db43c1dc1e1810afb8e7fe524d7a64f2..git_revision:2d1bf87c34a9ec952aac98bc6d88b978abfa9981
* src/buildtools/mac: git_revision:318ec1d2db43c1dc1e1810afb8e7fe524d7a64f2..git_revision:2d1bf87c34a9ec952aac98bc6d88b978abfa9981
* src/buildtools/win: git_revision:318ec1d2db43c1dc1e1810afb8e7fe524d7a64f2..git_revision:2d1bf87c34a9ec952aac98bc6d88b978abfa9981
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/647c989972..f209c46741
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/78a04c4e96..8565454576
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/a01b284c6f..6b4f14845b
* src/third_party/android_build_tools/manifest_merger/cipd: lHLzbS87smMbtV1VPJa5UWm59SS7t7Q1PRsEjIKbl1sC..7f2hBQelhgttzGk5TKRjsOLsTW3KYFbL66HKIXfh9tsC
* src/third_party/androidx/cipd: XpThiLpVJwtbiGf8RwS7hzulYyS7Q29MXabSOIrsIX4C..YCBAzxNcEKbZRoyf2ZC-GMvC8Vq7krnrNilMlfCFoUwC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/14fa4996cb..6a584e1d58
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/5f1a0a82b5..6d153ac3a6
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/bdec88009d..2c898e8b95
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/f7931aae14..00dc0eec59
* src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/7061b80a01..14776ea33c
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/c257779d8a..b5740713e2
DEPS diff: https://chromium.googlesource.com/chromium/src/+/3d3530d666..085c3ffcf5/DEPS

BUG=None

Change-Id: I8e2ea4e14fb845273e6f9d816ae734cbf7b1704f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/450840
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#46959}
---
 DEPS | 32 ++++++++++++++++----------------
 1 file changed, 16 insertions(+), 16 deletions(-)

diff --git a/DEPS b/DEPS
index 06f8b9a1a4..88acce6341 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '3d3530d6663a37252ef3705b0f4080a06d298862',
+  'chromium_revision': '085c3ffcf54d2f49fe40b07acd0e8f00797e91c9',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@0b1d3a05db11e2e95622e80980cf5d4dd8029805',
+    'https://chromium.googlesource.com/chromium/src/build@33652fea89b45afa80ca33a8bd25dff641800b05',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@451e71432ac9f6a91d4d5604c29e0a5078e8c2e8',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,19 +78,19 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@647c9899724bcf191376b47106bc980402079f75',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@f209c46741434444f50272ba95fa2832c3d582de',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@78a04c4e96cd034b9334e0b4e94e97dbf9737d75',
+    'https://chromium.googlesource.com/chromium/src/testing@856545457697ffd9411702caaa0467163f9f41b2',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@a01b284c6fe4bc2648e911f1018e0b08f2b8c9cb',
+    'https://chromium.googlesource.com/chromium/src/third_party@6b4f14845bed28bede3ee5b4a3449b909861615c',
 
   'src/buildtools/linux64': {
     'packages': [
       {
         'package': 'gn/gn/linux-${{arch}}',
-        'version': 'git_revision:318ec1d2db43c1dc1e1810afb8e7fe524d7a64f2',
+        'version': 'git_revision:2d1bf87c34a9ec952aac98bc6d88b978abfa9981',
       }
     ],
     'dep_type': 'cipd',
@@ -100,7 +100,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/mac-${{arch}}',
-        'version': 'git_revision:318ec1d2db43c1dc1e1810afb8e7fe524d7a64f2',
+        'version': 'git_revision:2d1bf87c34a9ec952aac98bc6d88b978abfa9981',
       }
     ],
     'dep_type': 'cipd',
@@ -110,7 +110,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/windows-amd64',
-        'version': 'git_revision:318ec1d2db43c1dc1e1810afb8e7fe524d7a64f2',
+        'version': 'git_revision:2d1bf87c34a9ec952aac98bc6d88b978abfa9981',
       }
     ],
     'dep_type': 'cipd',
@@ -337,7 +337,7 @@ deps = {
   'src/third_party/libc++abi/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@8f11bb1d4438d0239d0dfc1bd9456a9f31629dda',
   'src/third_party/llvm-libc/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@7061b80a014afdad13ce434000f4810a6e803eb0',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@14776ea33c0b64c3cec0ce80ff3eae7ddae9d977',
   'src/third_party/libunwind/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@17ccf7d110c5526cb77e93cfd8330f491fb2bf18',
 
@@ -488,7 +488,7 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@14fa4996cb794af518b18d3a41733153f5f5f2f8',
+    'https://boringssl.googlesource.com/boringssl.git@6a584e1d58d1fa95bc47d6c92cc9a8cb49d29635',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@79099fdf668ae097c9eca7052fd5c4c5de6c9098',
   'src/third_party/catapult':
@@ -505,13 +505,13 @@ deps = {
   'src/third_party/crc32c/src':
     'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558',
   'src/third_party/depot_tools':
-    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@5f1a0a82b50523fb6ec388b4287a4e26781ed928',
+    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@6d153ac3a600dea6c2ec320c9eb4df06aa3dca04',
   'src/third_party/ffmpeg':
     'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@946d97db8d906277085e361892b7efda5152e2f1',
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@f7931aae144a330cd38912315eae2417de1fb614',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@00dc0eec595569823703a31772c4a0466b719f4a',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@c257779d8af09f8470d6a4e711f1e0d64b131d1f',
+    'https://chromium.googlesource.com/chromium/src/tools@b5740713e25566e63b72b3df8bee3064b15069d5',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'XpThiLpVJwtbiGf8RwS7hzulYyS7Q29MXabSOIrsIX4C',
+          'version': 'YCBAzxNcEKbZRoyf2ZC-GMvC8Vq7krnrNilMlfCFoUwC',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -718,7 +718,7 @@ deps = {
       'packages': [
           {
                'package': 'chromium/third_party/android_build_tools/manifest_merger',
-               'version': 'lHLzbS87smMbtV1VPJa5UWm59SS7t7Q1PRsEjIKbl1sC',
+               'version': '7f2hBQelhgttzGk5TKRjsOLsTW3KYFbL66HKIXfh9tsC',
           },
       ],
       'condition': 'checkout_android',
@@ -864,7 +864,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7364b490b5f78d58efe23ea76e74210fd6c3c76f',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + 'bdec88009d3124c7924b70a3fe4a1a6d33d58966',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '2c898e8b95b11960092aeab247d3e016f5021098',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From 3a67d08077123d1a1a64d7bc73ade3817455f591 Mon Sep 17 00:00:00 2001
From: Bjorn Terelius <terelius@webrtc.org>
Date: Thu, 19 Feb 2026 13:15:08 +0100
Subject: [PATCH 107/472] Remove deprecated methods from event log visualizer
 Plot
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Bug: None
Change-Id: Ib0e8d306b0878b28ed63236e6c6f811f7bb07341
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/450540
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46960}
---
 .../rtc_event_log_visualizer/plot_base.h      | 22 +++----------------
 1 file changed, 3 insertions(+), 19 deletions(-)

diff --git a/rtc_tools/rtc_event_log_visualizer/plot_base.h b/rtc_tools/rtc_event_log_visualizer/plot_base.h
index df7a2cb23a..99160d88a8 100644
--- a/rtc_tools/rtc_event_log_visualizer/plot_base.h
+++ b/rtc_tools/rtc_event_log_visualizer/plot_base.h
@@ -17,7 +17,6 @@
 #include <utility>
 #include <vector>
 
-#include "absl/base/attributes.h"
 #include "absl/strings/string_view.h"
 
 // Generated at build-time by the protobuf compiler.
@@ -96,15 +95,9 @@ struct IntervalSeries {
 };
 
 // A container that represents a general graph, with axes, title and one or
-// more data series. A subclass should define the output format by overriding
-// the Draw() method.
+// more data series.
 class Plot {
  public:
-  virtual ~Plot() {}
-
-  ABSL_DEPRECATED("Use PrintPythonCode() or ExportProtobuf() instead.")
-  virtual void Draw() {}
-
   // Sets the lower x-axis limit to min_value (if left_margin == 0).
   // Sets the upper x-axis limit to max_value (if right_margin == 0).
   // The margins are measured as fractions of the interval
@@ -167,12 +160,10 @@ class Plot {
   // Otherwise, the call has no effect and the timeseries is destroyed.
   void AppendTimeSeriesIfNotEmpty(TimeSeries&& time_series);
 
-  // Replaces PythonPlot::Draw()
   void PrintPythonCode(
       bool show_grid = false,
       absl::string_view figure_output_path = absl::string_view()) const;
 
-  // Replaces ProtobufPlot::Draw()
   void ExportProtobuf(analytics::Chart* chart) const;
 
  protected:
@@ -191,26 +182,19 @@ class Plot {
 
 class PlotCollection {
  public:
-  virtual ~PlotCollection() {}
+  Plot* AppendNewPlot();
 
-  ABSL_DEPRECATED("Use PrintPythonCode() or ExportProtobuf() instead.")
-  virtual void Draw() {}
-
-  virtual Plot* AppendNewPlot();
-
-  virtual Plot* AppendNewPlot(absl::string_view);
+  Plot* AppendNewPlot(absl::string_view);
 
   void SetCallTimeToUtcOffsetMs(int64_t calltime_to_utc_ms) {
     calltime_to_utc_ms_ = calltime_to_utc_ms;
   }
 
-  // Replaces PythonPlotCollection::Draw()
   void PrintPythonCode(
       bool shared_xaxis,
       bool show_grid_on_all_plots,
       absl::string_view figure_output_path = absl::string_view()) const;
 
-  // Replaces ProtobufPlotCollections::Draw()
   void ExportProtobuf(analytics::ChartCollection* collection) const;
 
  protected:

From 93c5ab776a752b386493d135e67f67b717307f1e Mon Sep 17 00:00:00 2001
From: Rasmus Brandt <brandtr@webrtc.org>
Date: Wed, 18 Feb 2026 13:33:34 +0100
Subject: [PATCH 108/472] Video timing simulator: Avoid "inf-inf" type
 arithmetic.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

We should avoid this type of ill-defined calculations. Furthermore, they
fail in debug mode.

Bug: b/423646186
Change-Id: Id81b3066861f0fbb02de4714496d99836a6a6964
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/450240
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46961}
---
 video/timing/simulator/decodability_simulator.h |  4 ++++
 video/timing/simulator/frame_base.h             | 16 ++++++++++++++--
 video/timing/simulator/rendering_simulator.h    | 10 ++++++++++
 3 files changed, 28 insertions(+), 2 deletions(-)

diff --git a/video/timing/simulator/decodability_simulator.h b/video/timing/simulator/decodability_simulator.h
index 7d1860865f..3aaabe9996 100644
--- a/video/timing/simulator/decodability_simulator.h
+++ b/video/timing/simulator/decodability_simulator.h
@@ -123,6 +123,10 @@ inline void SortByDecodableOrder(
 // Difference in decodable time between two frames.
 inline TimeDelta InterDecodableTime(const DecodabilitySimulator::Frame& cur,
                                     const DecodabilitySimulator::Frame& prev) {
+  if (!cur.decodable_timestamp.IsFinite() &&
+      !prev.decodable_timestamp.IsFinite()) {
+    return TimeDelta::PlusInfinity();
+  }
   return cur.decodable_timestamp - prev.decodable_timestamp;
 }
 
diff --git a/video/timing/simulator/frame_base.h b/video/timing/simulator/frame_base.h
index 939a8744cc..6faa673ee2 100644
--- a/video/timing/simulator/frame_base.h
+++ b/video/timing/simulator/frame_base.h
@@ -121,24 +121,36 @@ std::optional<int64_t> InterFrameSizeBytes(const FrameT& cur,
 // Difference in departure timestamp between two frames.
 template <typename FrameT>
 TimeDelta InterDepartureTime(const FrameT& cur, const FrameT& prev) {
+  RTC_DCHECK(cur.DepartureTimestamp().IsFinite());
+  RTC_DCHECK(prev.DepartureTimestamp().IsFinite());
   return cur.DepartureTimestamp() - prev.DepartureTimestamp();
 }
 
 // Difference in arrival timestamp between two frames.
 template <typename FrameT>
 TimeDelta InterArrivalTime(const FrameT& cur, const FrameT& prev) {
-  return cur.ArrivalTimestamp() - prev.ArrivalTimestamp();
+  Timestamp cur_arrival = cur.ArrivalTimestamp();
+  Timestamp prev_arrival = prev.ArrivalTimestamp();
+  if (!cur_arrival.IsFinite() && !prev_arrival.IsFinite()) {
+    return TimeDelta::PlusInfinity();
+  }
+  return cur_arrival - prev_arrival;
 }
 
 // https://datatracker.ietf.org/doc/html/rfc5481#section-1
 template <typename FrameT>
 TimeDelta InterFrameDelayVariation(const FrameT& cur, const FrameT& prev) {
-  return InterArrivalTime(cur, prev) - InterDepartureTime(cur, prev);
+  TimeDelta iat = InterArrivalTime(cur, prev);
+  TimeDelta idt = InterDepartureTime(cur, prev);
+  RTC_DCHECK(idt.IsFinite());
+  return iat - idt;
 }
 
 // Difference in assembled timestamp between two frames.
 template <typename FrameT>
 TimeDelta InterAssembledTime(const FrameT& cur, const FrameT& prev) {
+  RTC_DCHECK(cur.assembled_timestamp.IsFinite());
+  RTC_DCHECK(prev.assembled_timestamp.IsFinite());
   return cur.assembled_timestamp - prev.assembled_timestamp;
 }
 
diff --git a/video/timing/simulator/rendering_simulator.h b/video/timing/simulator/rendering_simulator.h
index 2be7dea597..aedb5508fa 100644
--- a/video/timing/simulator/rendering_simulator.h
+++ b/video/timing/simulator/rendering_simulator.h
@@ -413,18 +413,28 @@ inline void SortByRenderedOrder(ArrayView<RenderingSimulator::Frame> frames) {
 // Difference in render time (target) between two frames.
 inline TimeDelta InterRenderTime(const RenderingSimulator::Frame& cur,
                                  const RenderingSimulator::Frame& prev) {
+  if (!cur.render_timestamp.IsFinite() && !prev.render_timestamp.IsFinite()) {
+    return TimeDelta::PlusInfinity();
+  }
   return cur.render_timestamp - prev.render_timestamp;
 }
 
 // Difference in decoded time (actual) between two frames.
 inline TimeDelta InterDecodedTime(const RenderingSimulator::Frame& cur,
                                   const RenderingSimulator::Frame& prev) {
+  if (!cur.decoded_timestamp.IsFinite() && !prev.decoded_timestamp.IsFinite()) {
+    return TimeDelta::PlusInfinity();
+  }
   return cur.decoded_timestamp - prev.decoded_timestamp;
 }
 
 // Difference in rendered time (actual) between two frames.
 inline TimeDelta InterRenderedTime(const RenderingSimulator::Frame& cur,
                                    const RenderingSimulator::Frame& prev) {
+  if (!cur.rendered_timestamp.IsFinite() &&
+      !prev.rendered_timestamp.IsFinite()) {
+    return TimeDelta::PlusInfinity();
+  }
   return cur.rendered_timestamp - prev.rendered_timestamp;
 }
 

From 453d89919e78631d5eb43584a91e5a60a1b2494d Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Thu, 19 Feb 2026 20:01:39 +0100
Subject: [PATCH 109/472] Add ArrayView::supspan variants to create spans with
 static extent

This functions match std::span api and provide concise way to create
spans with static extent. Note that most std::span constructors for
spans with static extent are explicit and thus can't be used through
braced initialization.

This change includes examples where those new functions are most
helpful.

Bug: webrtc:439801349
Change-Id: I4158ba12f28d0271fe9f9577533b2b36f3cd9222
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/450680
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46962}
---
 api/array_view.h                              | 33 +++++++++++
 api/array_view_unittest.cc                    | 57 +++++++++++++++++++
 .../agc2/rnn_vad/features_extraction.cc       | 12 ++--
 .../rnn_vad/features_extraction_unittest.cc   |  2 +-
 4 files changed, 97 insertions(+), 7 deletions(-)

diff --git a/api/array_view.h b/api/array_view.h
index 96960d258a..5d7a5beb6b 100644
--- a/api/array_view.h
+++ b/api/array_view.h
@@ -159,6 +159,11 @@ class ArrayView final : public array_view_internal::ArrayViewBase<T, Size> {
   using iterator = typename std::span<T>::iterator;
   using const_iterator = typename std::span<const T>::iterator;
 
+  static constexpr size_t extent =
+      Size == array_view_internal::kArrayViewVarSize
+          ? std::dynamic_extent
+          : static_cast<size_t>(Size);
+
   // Construct an ArrayView from a pointer and a length.
   template <typename U>
   ArrayView(U* data, size_t size)
@@ -294,6 +299,34 @@ class ArrayView final : public array_view_internal::ArrayViewBase<T, Size> {
   ABSL_DEPRECATE_AND_INLINE()
   std::reverse_iterator<const_iterator> crend() const { return rend(); }
 
+  template <size_t count>
+  constexpr ArrayView<T, count> first() const {
+    RTC_HARDENING_ASSERT(count <= this->size());
+    return ArrayView<T, count>(this->data(), count);
+  }
+
+  template <size_t count>
+  constexpr ArrayView<T, count> last() const {
+    RTC_HARDENING_ASSERT(count <= this->size());
+    return ArrayView<T, count>(this->data() + this->size() - count, count);
+  }
+
+  template <size_t offset, size_t count = std::dynamic_extent>
+  constexpr auto subspan() const {
+    RTC_HARDENING_ASSERT(offset <= this->size());
+    if constexpr (count == std::dynamic_extent) {
+      constexpr ptrdiff_t final_extent =
+          Size != array_view_internal::kArrayViewVarSize
+              ? Size - offset
+              : array_view_internal::kArrayViewVarSize;
+      return ArrayView<T, final_extent>(this->data() + offset,
+                                        this->size() - offset);
+    } else {
+      RTC_HARDENING_ASSERT(count <= this->size() - offset);
+      return ArrayView<T, count>(this->data() + offset, count);
+    }
+  }
+
   constexpr ArrayView<T> subspan(size_t offset,
                                  size_t count = std::dynamic_extent) const {
     ABSL_HARDENING_ASSERT(offset <= this->size());
diff --git a/api/array_view_unittest.cc b/api/array_view_unittest.cc
index d4fa00b3aa..4efc19f0a5 100644
--- a/api/array_view_unittest.cc
+++ b/api/array_view_unittest.cc
@@ -13,6 +13,7 @@
 #include <array>
 #include <cstddef>
 #include <cstdint>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
@@ -592,6 +593,62 @@ TEST(ArrayViewTest, TestSubViewFixed) {
 }
 #pragma clang diagnostic pop
 
+TEST(ArrayViewTest, SubspanFixed) {
+  std::array<int, 3> a = {1, 2, 3};
+  ArrayView<int, 3> av(a);
+
+  EXPECT_THAT(av.subspan<0>(), ElementsAre(1, 2, 3));
+  EXPECT_THAT(av.subspan<1>(), ElementsAre(2, 3));
+  EXPECT_THAT(av.subspan<2>(), ElementsAre(3));
+  EXPECT_THAT(av.subspan<3>(), IsEmpty());
+
+  EXPECT_THAT((av.subspan<1, 0>()), IsEmpty());
+  EXPECT_THAT((av.subspan<1, 1>()), ElementsAre(2));
+  EXPECT_THAT((av.subspan<1, 2>()), ElementsAre(2, 3));
+
+  EXPECT_EQ(av.subspan<1>().extent, 2u);
+  EXPECT_EQ((av.subspan<1, 1>().extent), 1u);
+
+  ArrayView<int> dynamic_av(a);
+  EXPECT_THAT(dynamic_av.subspan<0>(), ElementsAre(1, 2, 3));
+  EXPECT_THAT(dynamic_av.subspan<1>(), ElementsAre(2, 3));
+  EXPECT_THAT(dynamic_av.subspan<2>(), ElementsAre(3));
+  EXPECT_THAT(dynamic_av.subspan<3>(), IsEmpty());
+
+  EXPECT_THAT((dynamic_av.subspan<1, 0>()), IsEmpty());
+  EXPECT_THAT((dynamic_av.subspan<1, 1>()), ElementsAre(2));
+  EXPECT_THAT((dynamic_av.subspan<1, 2>()), ElementsAre(2, 3));
+
+  EXPECT_EQ(dynamic_av.subspan<1>().extent, std::dynamic_extent);
+  EXPECT_EQ((dynamic_av.subspan<1, 1>()).extent, 1u);
+}
+
+TEST(ArrayViewTest, FirstFixed) {
+  std::array<int, 3> a = {1, 2, 3};
+  ArrayView<int> av(a);
+
+  EXPECT_THAT(av.first<0>(), IsEmpty());
+  EXPECT_THAT(av.first<1>(), ElementsAre(1));
+  EXPECT_THAT(av.first<2>(), ElementsAre(1, 2));
+  EXPECT_THAT(av.first<3>(), ElementsAre(1, 2, 3));
+
+  EXPECT_EQ((av.first<0>()).extent, 0u);
+  EXPECT_EQ((av.first<2>()).extent, 2u);
+}
+
+TEST(ArrayViewTest, LastFixed) {
+  std::array<int, 3> a = {1, 2, 3};
+  ArrayView<int> av(a);
+
+  EXPECT_THAT(av.last<0>(), IsEmpty());
+  EXPECT_THAT(av.last<1>(), ElementsAre(3));
+  EXPECT_THAT(av.last<2>(), ElementsAre(2, 3));
+  EXPECT_THAT(av.last<3>(), ElementsAre(1, 2, 3));
+
+  EXPECT_EQ((av.last<0>()).extent, 0u);
+  EXPECT_EQ((av.last<2>()).extent, 2u);
+}
+
 TEST(ArrayViewTest, TestReinterpretCastFixedSize) {
   uint8_t bytes[] = {1, 2, 3};
   ArrayView<uint8_t, 3> uint8_av(bytes);
diff --git a/modules/audio_processing/agc2/rnn_vad/features_extraction.cc b/modules/audio_processing/agc2/rnn_vad/features_extraction.cc
index bc170f99dc..1dbfa1a841 100644
--- a/modules/audio_processing/agc2/rnn_vad/features_extraction.cc
+++ b/modules/audio_processing/agc2/rnn_vad/features_extraction.cc
@@ -82,12 +82,12 @@ bool FeaturesExtractor::CheckSilenceComputeFeatures(
   // Analyze reference and lagged frames checking if silence has been detected
   // and write the feature vector.
   return spectral_features_extractor_.CheckSilenceComputeFeatures(
-      reference_frame_view_, {lagged_frame.data(), kFrameSize20ms24kHz},
-      {feature_vector.data() + kNumLowerBands, kNumBands - kNumLowerBands},
-      {feature_vector.data(), kNumLowerBands},
-      {feature_vector.data() + kNumBands, kNumLowerBands},
-      {feature_vector.data() + kNumBands + kNumLowerBands, kNumLowerBands},
-      {feature_vector.data() + kNumBands + 2 * kNumLowerBands, kNumLowerBands},
+      reference_frame_view_, lagged_frame.first<kFrameSize20ms24kHz>(),
+      feature_vector.subspan<kNumLowerBands, kNumBands - kNumLowerBands>(),
+      feature_vector.first<kNumLowerBands>(),
+      feature_vector.subspan<kNumBands, kNumLowerBands>(),
+      feature_vector.subspan<kNumBands + kNumLowerBands, kNumLowerBands>(),
+      feature_vector.subspan<kNumBands + 2 * kNumLowerBands, kNumLowerBands>(),
       &feature_vector[kFeatureVectorSize - 1]);
 }
 
diff --git a/modules/audio_processing/agc2/rnn_vad/features_extraction_unittest.cc b/modules/audio_processing/agc2/rnn_vad/features_extraction_unittest.cc
index 9f5b4d37ff..7c0b575217 100644
--- a/modules/audio_processing/agc2/rnn_vad/features_extraction_unittest.cc
+++ b/modules/audio_processing/agc2/rnn_vad/features_extraction_unittest.cc
@@ -61,7 +61,7 @@ bool FeedTestData(FeaturesExtractor& features_extractor,
   const int num_frames = samples.size() / kFrameSize10ms24kHz;
   for (int i = 0; i < num_frames; ++i) {
     is_silence = features_extractor.CheckSilenceComputeFeatures(
-        {samples.data() + i * kFrameSize10ms24kHz, kFrameSize10ms24kHz},
+        samples.subspan(i * kFrameSize10ms24kHz).first<kFrameSize10ms24kHz>(),
         feature_vector);
   }
   return is_silence;

From f083c099b3b6409315e94257e8b2032a0f678235 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Fri, 20 Feb 2026 06:03:50 -0800
Subject: [PATCH 110/472] Roll chromium_revision 085c3ffcf5..1c9161b9bb
 (1587653:1587773)

Change log: https://chromium.googlesource.com/chromium/src/+log/085c3ffcf5..1c9161b9bb
Full diff: https://chromium.googlesource.com/chromium/src/+/085c3ffcf5..1c9161b9bb

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/33652fea89..39d4202659
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/f209c46741..f16867329d
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/8565454576..0e43b2634d
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/6b4f14845b..54440292c8
* src/third_party/androidx/cipd: YCBAzxNcEKbZRoyf2ZC-GMvC8Vq7krnrNilMlfCFoUwC..jcsrqbC9ptL2-ce_XhN-xUE53nHvM345DTcMXbVabSAC
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/2c898e8b95..374fe225bf
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/00dc0eec59..f2b7408057
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/8395cf4efe..a3e107ec80
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/b5740713e2..8e77480101
DEPS diff: https://chromium.googlesource.com/chromium/src/+/085c3ffcf5..1c9161b9bb/DEPS

BUG=None

Change-Id: Iaf22f495c8698554500e9159f256d2eb6f3b0d57
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/450940
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#46963}
---
 DEPS | 20 ++++++++++----------
 1 file changed, 10 insertions(+), 10 deletions(-)

diff --git a/DEPS b/DEPS
index 88acce6341..ac3521069b 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '085c3ffcf54d2f49fe40b07acd0e8f00797e91c9',
+  'chromium_revision': '1c9161b9bb60dab9783399d2178561168cba0090',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@33652fea89b45afa80ca33a8bd25dff641800b05',
+    'https://chromium.googlesource.com/chromium/src/build@39d420265929393ffd3080ae3b1740aab1443955',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@451e71432ac9f6a91d4d5604c29e0a5078e8c2e8',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@f209c46741434444f50272ba95fa2832c3d582de',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@f16867329de93739d9a0984a7ee0bee03064fadd',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@856545457697ffd9411702caaa0467163f9f41b2',
+    'https://chromium.googlesource.com/chromium/src/testing@0e43b2634d95bedd2aee0f6b824153daf76aa8ee',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@6b4f14845bed28bede3ee5b4a3449b909861615c',
+    'https://chromium.googlesource.com/chromium/src/third_party@54440292c873c5f38c4353fb91f5d23a88f6f213',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -511,7 +511,7 @@ deps = {
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@00dc0eec595569823703a31772c4a0466b719f4a',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@f2b7408057b44429dabb721a69cb55ef23661da8',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '8395cf4efec05c454b9f60f080aab5ce5277a730',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'a3e107ec803c876a870205f89c1e37742184b598',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@b5740713e25566e63b72b3df8bee3064b15069d5',
+    'https://chromium.googlesource.com/chromium/src/tools@8e77480101fc44debdfe5e1dec41293bacf9b852',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'YCBAzxNcEKbZRoyf2ZC-GMvC8Vq7krnrNilMlfCFoUwC',
+          'version': 'jcsrqbC9ptL2-ce_XhN-xUE53nHvM345DTcMXbVabSAC',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -864,7 +864,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7364b490b5f78d58efe23ea76e74210fd6c3c76f',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '2c898e8b95b11960092aeab247d3e016f5021098',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '374fe225bfadbad6c172b6ef06e21ac2fad90bc2',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From 1d38dc748507d5603e5fa50ffd061eb59dbb8c94 Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Fri, 20 Feb 2026 13:07:15 +0100
Subject: [PATCH 111/472] Avoid unsafe buffer warnings when ArrayView is
 replaces with std::span

While currently this code doesn't produced warnings when used with
ArrayView, using std::span::data same way does issue warnings.

Bug: webrtc:439801349
Change-Id: Icb8dc92f2d5fd20c38b364e7ea5218f0171e0b8e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/450880
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46964}
---
 pc/external_hmac.cc                 | 15 ++++++++-------
 rtc_base/thread_unittest.cc         | 10 ++++++----
 rtc_base/virtual_socket_unittest.cc | 16 +++++++---------
 3 files changed, 21 insertions(+), 20 deletions(-)

diff --git a/pc/external_hmac.cc b/pc/external_hmac.cc
index 195f7ba107..e65d2b7073 100644
--- a/pc/external_hmac.cc
+++ b/pc/external_hmac.cc
@@ -13,8 +13,8 @@
 #include <cstdint>
 #include <cstdlib>
 #include <cstring>
+#include <memory>
 
-#include "api/array_view.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/zero_memory.h"
 #include "third_party/libsrtp/crypto/include/auth.h"
@@ -77,19 +77,20 @@ srtp_err_status_t external_hmac_alloc(srtp_auth_t** a,
     return srtp_err_status_bad_param;
 
   // Allocate memory for auth and hmac_ctx_t structures.
-  size_t needed_size = sizeof(ExternalHmacContext) + sizeof(srtp_auth_t);
-  uint8_t* raw_pointer = new uint8_t[needed_size];
-  if (raw_pointer == nullptr)
+  auto raw_pointer = std::make_unique_for_overwrite<uint8_t[]>(
+      sizeof(srtp_auth_t) + sizeof(ExternalHmacContext));
+  if (raw_pointer == nullptr) {
     return srtp_err_status_alloc_fail;
-  ArrayView<uint8_t> pointer(raw_pointer, needed_size);
+  }
+  uint8_t* state_memory = &raw_pointer[sizeof(srtp_auth_t)];
 
   // Set pointers
-  *a = reinterpret_cast<srtp_auth_t*>(pointer.data());
+  *a = reinterpret_cast<srtp_auth_t*>(raw_pointer.release());
   // `external_hmac` is const and libsrtp expects `type` to be non-const.
   // const conversion is required. `external_hmac` is constant because we don't
   // want to increase global count in Chrome.
   (*a)->type = const_cast<srtp_auth_type_t*>(&external_hmac);
-  (*a)->state = &pointer[sizeof(srtp_auth_t)];
+  (*a)->state = state_memory;
   (*a)->out_len = out_len;
   (*a)->key_len = key_len;
   (*a)->prefix_len = 0;
diff --git a/rtc_base/thread_unittest.cc b/rtc_base/thread_unittest.cc
index f8b031523a..3fe9eda6d8 100644
--- a/rtc_base/thread_unittest.cc
+++ b/rtc_base/thread_unittest.cc
@@ -10,6 +10,7 @@
 
 #include "rtc_base/thread.h"
 
+#include <array>
 #include <atomic>
 #include <cstddef>
 #include <cstdint>
@@ -26,6 +27,7 @@
 #include "api/units/time_delta.h"
 #include "rtc_base/async_packet_socket.h"
 #include "rtc_base/async_udp_socket.h"
+#include "rtc_base/byte_order.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/event.h"
 #include "rtc_base/fake_clock.h"
@@ -81,8 +83,9 @@ class MessageClient : public TestGenerator {
   ~MessageClient() { delete socket_; }
 
   void OnValue(int value) {
-    int result = Next(value);
-    EXPECT_GE(socket_->Send(&result, sizeof(result)), 0);
+    std::array<uint8_t, sizeof(uint32_t)> octets;
+    SetLE32(octets, Next(value));
+    EXPECT_GE(socket_->Send(octets.data(), octets.size()), 0);
   }
 
  private:
@@ -113,8 +116,7 @@ class SocketClient : public TestGenerator {
 
   void OnPacket(AsyncPacketSocket* socket, const ReceivedIpPacket& packet) {
     EXPECT_EQ(packet.payload().size(), sizeof(uint32_t));
-    uint32_t prev =
-        reinterpret_cast<const uint32_t*>(packet.payload().data())[0];
+    uint32_t prev = GetLE32(packet.payload());
     uint32_t result = Next(prev);
 
     post_thread_->PostDelayedTask([post_handler_ = post_handler_,
diff --git a/rtc_base/virtual_socket_unittest.cc b/rtc_base/virtual_socket_unittest.cc
index 9246372671..8a7c806263 100644
--- a/rtc_base/virtual_socket_unittest.cc
+++ b/rtc_base/virtual_socket_unittest.cc
@@ -17,11 +17,9 @@
 #include <cstring>
 #include <ctime>
 #include <memory>
-#include <span>
 #include <utility>
 #include <vector>
 
-#include "absl/algorithm/container.h"
 #include "absl/memory/memory.h"
 #include "api/array_view.h"
 #include "api/environment/environment.h"
@@ -30,6 +28,7 @@
 #include "api/units/timestamp.h"
 #include "rtc_base/async_packet_socket.h"
 #include "rtc_base/async_udp_socket.h"
+#include "rtc_base/byte_order.h"
 #include "rtc_base/ip_address.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/net_helpers.h"
@@ -79,9 +78,8 @@ struct Sender {
       uint32_t size =
           std::clamp<uint32_t>(rate * delay / 1000, sizeof(uint32_t), 4096);
       count += size;
-      auto cur_time_bytes = std::as_bytes(std::span{&cur_time, 1});
-      absl::c_transform(cur_time_bytes, dummy.begin(),
-                        [](std::byte b) { return static_cast<char>(b); });
+
+      SetLE64(dummy, static_cast<uint64_t>(cur_time));
       socket->Send(dummy.data(), size, options);
 
       last_send = cur_time;
@@ -102,7 +100,7 @@ struct Sender {
   uint32_t rate;  // bytes per second
   uint32_t count;
   int64_t last_send;
-  std::array<char, 4096> dummy;
+  std::array<uint8_t, 4096> dummy;
 };
 
 struct Receiver {
@@ -144,9 +142,9 @@ struct Receiver {
     count += packet.payload().size();
     sec_count += packet.payload().size();
 
-    uint32_t send_time =
-        *reinterpret_cast<const uint32_t*>(packet.payload().data());
-    uint32_t recv_time = env.clock().TimeInMilliseconds();
+    uint32_t send_time = GetLE32(packet.payload());
+    uint32_t recv_time =
+        static_cast<uint32_t>(env.clock().TimeInMilliseconds());
     uint32_t delay = recv_time - send_time;
     sum += delay;
     sum_sq += delay * delay;

From 0c37d5c76fdff3f7eae040d65f43755d15d3f381 Mon Sep 17 00:00:00 2001
From: Bjorn Terelius <terelius@webrtc.org>
Date: Fri, 20 Feb 2026 16:01:49 +0100
Subject: [PATCH 112/472] Remove specs for Perf Mac 11 bot
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Bug: b/441483429
Change-Id: I33dbb5c63bb3f15fa4bf39ff04f42316e9e6111c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/450960
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46965}
---
 infra/config/builders.star                    |  2 -
 .../config/generated/luci/cr-buildbucket.cfg  | 58 ---------------
 infra/config/generated/luci/luci-milo.cfg     |  9 ---
 infra/config/generated/luci/luci-notify.cfg   | 49 -------------
 .../config/generated/luci/luci-scheduler.cfg  | 23 ------
 infra/config/generated/luci/realms.cfg        |  1 -
 infra/specs/client.webrtc.json                |  1 -
 infra/specs/client.webrtc.perf.json           | 71 -------------------
 infra/specs/mixins.pyl                        |  7 --
 infra/specs/mixins_webrtc.pyl                 |  7 --
 infra/specs/waterfalls.pyl                    | 12 ----
 tools_webrtc/mb/mb_config.pyl                 |  2 -
 12 files changed, 242 deletions(-)

diff --git a/infra/config/builders.star b/infra/config/builders.star
index 52518ef480..8816ef06fa 100644
--- a/infra/config/builders.star
+++ b/infra/config/builders.star
@@ -394,9 +394,7 @@ try_builder("mac_compile_dbg")
 ci_builder("Mac64 Release", "Mac|x64|rel")
 try_builder("mac_rel")
 try_builder("mac_compile_rel", cq = None)
-ci_builder("Mac64 Builder", ci_cat = None, perf_cat = "Mac|x64|Builder|")
 ci_builder("MacArm64 Builder", ci_cat = None, perf_cat = "Mac|arm64|Builder|")
-perf_builder("Perf Mac 11", "Mac|x64|Tester|11", triggered_by = ["Mac64 Builder"])
 perf_builder("Perf Mac M1 Arm64 12", "Mac|arm64|Tester|12", triggered_by = ["MacArm64 Builder"])
 ci_builder("Mac Asan", "Mac|x64|asan")
 try_builder("mac_asan")
diff --git a/infra/config/generated/luci/cr-buildbucket.cfg b/infra/config/generated/luci/cr-buildbucket.cfg
index 04ad068f3c..43bf68c18b 100644
--- a/infra/config/generated/luci/cr-buildbucket.cfg
+++ b/infra/config/generated/luci/cr-buildbucket.cfg
@@ -1358,36 +1358,6 @@ buckets {
         }
       }
     }
-    builders {
-      name: "Mac64 Builder"
-      swarming_host: "chromium-swarm.appspot.com"
-      swarming_tags: "vpython:native-python-wrapper"
-      dimensions: "builderless:1"
-      dimensions: "cpu:x86-64"
-      dimensions: "os:Mac"
-      dimensions: "pool:luci.webrtc.ci"
-      recipe {
-        name: "webrtc/standalone"
-        cipd_package: "infra/recipe_bundles/chromium.googlesource.com/chromium/tools/build"
-        cipd_version: "refs/heads/main"
-        properties_j: "$build/siso:{\"configs\":[\"builder\"],\"enable_cloud_profiler\":true,\"enable_cloud_trace\":true,\"enable_monitoring\":true,\"project\":\"rbe-webrtc-trusted\"}"
-        properties_j: "$recipe_engine/resultdb/test_presentation:{\"column_keys\":[],\"grouping_keys\":[\"status\",\"v.test_suite\"]}"
-        properties_j: "builder_group:\"client.webrtc\""
-      }
-      priority: 30
-      execution_timeout_secs: 7200
-      build_numbers: YES
-      service_account: "webrtc-ci-builder@chops-service-accounts.iam.gserviceaccount.com"
-      resultdb {
-        enable: true
-        bq_exports {
-          project: "webrtc-ci"
-          dataset: "resultdb"
-          table: "perf_test_results"
-          test_results {}
-        }
-      }
-    }
     builders {
       name: "MacArm64 Builder"
       swarming_host: "chromium-swarm.appspot.com"
@@ -1530,34 +1500,6 @@ buckets {
         }
       }
     }
-    builders {
-      name: "Perf Mac 11"
-      swarming_host: "chromium-swarm.appspot.com"
-      swarming_tags: "vpython:native-python-wrapper"
-      dimensions: "os:Linux"
-      dimensions: "pool:luci.webrtc.perf"
-      recipe {
-        name: "webrtc/standalone"
-        cipd_package: "infra/recipe_bundles/chromium.googlesource.com/chromium/tools/build"
-        cipd_version: "refs/heads/main"
-        properties_j: "$build/siso:{\"configs\":[\"builder\"],\"enable_cloud_profiler\":true,\"enable_cloud_trace\":true,\"enable_monitoring\":true,\"project\":\"rbe-webrtc-trusted\"}"
-        properties_j: "$recipe_engine/resultdb/test_presentation:{\"column_keys\":[],\"grouping_keys\":[\"status\",\"v.test_suite\"]}"
-        properties_j: "builder_group:\"client.webrtc.perf\""
-      }
-      priority: 30
-      execution_timeout_secs: 10800
-      build_numbers: YES
-      service_account: "webrtc-ci-builder@chops-service-accounts.iam.gserviceaccount.com"
-      resultdb {
-        enable: true
-        bq_exports {
-          project: "webrtc-ci"
-          dataset: "resultdb"
-          table: "perf_test_results"
-          test_results {}
-        }
-      }
-    }
     builders {
       name: "Perf Mac M1 Arm64 12"
       swarming_host: "chromium-swarm.appspot.com"
diff --git a/infra/config/generated/luci/luci-milo.cfg b/infra/config/generated/luci/luci-milo.cfg
index 4dff326fdf..a2311a61f8 100644
--- a/infra/config/generated/luci/luci-milo.cfg
+++ b/infra/config/generated/luci/luci-milo.cfg
@@ -313,19 +313,10 @@ consoles {
     name: "buildbucket/luci.webrtc.perf/Perf Fuchsia"
     category: "Fuchsia|x64|Tester"
   }
-  builders {
-    name: "buildbucket/luci.webrtc.perf/Mac64 Builder"
-    category: "Mac|x64|Builder"
-  }
   builders {
     name: "buildbucket/luci.webrtc.perf/MacArm64 Builder"
     category: "Mac|arm64|Builder"
   }
-  builders {
-    name: "buildbucket/luci.webrtc.perf/Perf Mac 11"
-    category: "Mac|x64|Tester"
-    short_name: "11"
-  }
   builders {
     name: "buildbucket/luci.webrtc.perf/Perf Mac M1 Arm64 12"
     category: "Mac|arm64|Tester"
diff --git a/infra/config/generated/luci/luci-notify.cfg b/infra/config/generated/luci/luci-notify.cfg
index 416b6f0f84..76a1902228 100644
--- a/infra/config/generated/luci/luci-notify.cfg
+++ b/infra/config/generated/luci/luci-notify.cfg
@@ -1153,33 +1153,6 @@ notifiers {
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
-notifiers {
-  notifications {
-    on_new_status: INFRA_FAILURE
-    email {
-      recipients: "webrtc-troopers-robots@google.com"
-    }
-    template: "infra_failure"
-  }
-  notifications {
-    on_new_status: FAILURE
-    email {
-      recipients: "webrtc-troopers-robots@google.com"
-    }
-    template: "build_failure"
-    notify_blamelist {}
-  }
-  builders {
-    bucket: "perf"
-    name: "Mac64 Builder"
-    repository: "https://webrtc.googlesource.com/src"
-  }
-  tree_closers {
-    tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
-    failed_step_regexp_exclude: ".*\\(experimental\\).*"
-  }
-}
 notifiers {
   notifications {
     on_new_status: INFRA_FAILURE
@@ -1295,28 +1268,6 @@ notifiers {
     repository: "https://webrtc.googlesource.com/src"
   }
 }
-notifiers {
-  notifications {
-    on_new_status: INFRA_FAILURE
-    email {
-      recipients: "webrtc-troopers-robots@google.com"
-    }
-    template: "infra_failure"
-  }
-  notifications {
-    on_new_status: FAILURE
-    email {
-      recipients: "webrtc-troopers-robots@google.com"
-    }
-    template: "build_failure"
-    notify_blamelist {}
-  }
-  builders {
-    bucket: "perf"
-    name: "Perf Mac 11"
-    repository: "https://webrtc.googlesource.com/src"
-  }
-}
 notifiers {
   notifications {
     on_new_status: INFRA_FAILURE
diff --git a/infra/config/generated/luci/luci-scheduler.cfg b/infra/config/generated/luci/luci-scheduler.cfg
index 6787a55f0d..dee7bc52fa 100644
--- a/infra/config/generated/luci/luci-scheduler.cfg
+++ b/infra/config/generated/luci/luci-scheduler.cfg
@@ -286,15 +286,6 @@ job {
     builder: "Mac Asan"
   }
 }
-job {
-  id: "Mac64 Builder"
-  realm: "perf"
-  buildbucket {
-    server: "cr-buildbucket.appspot.com"
-    bucket: "perf"
-    builder: "Mac64 Builder"
-  }
-}
 job {
   id: "Mac64 Debug"
   realm: "ci"
@@ -383,19 +374,6 @@ job {
     builder: "Perf Linux"
   }
 }
-job {
-  id: "Perf Mac 11"
-  realm: "perf"
-  triggering_policy {
-    kind: LOGARITHMIC_BATCHING
-    log_base: 1.7
-  }
-  buildbucket {
-    server: "cr-buildbucket.appspot.com"
-    bucket: "perf"
-    builder: "Perf Mac 11"
-  }
-}
 job {
   id: "Perf Mac M1 Arm64 12"
   realm: "perf"
@@ -590,7 +568,6 @@ trigger {
   triggers: "Android64 Builder arm64"
   triggers: "Fuchsia Builder"
   triggers: "Linux64 Builder"
-  triggers: "Mac64 Builder"
   triggers: "MacArm64 Builder"
   triggers: "Win64 Builder (Clang)"
   gitiles {
diff --git a/infra/config/generated/luci/realms.cfg b/infra/config/generated/luci/realms.cfg
index ebd33bd4ed..34d7f60948 100644
--- a/infra/config/generated/luci/realms.cfg
+++ b/infra/config/generated/luci/realms.cfg
@@ -128,7 +128,6 @@ realms {
         values: "Perf Android64 (R Pixel5)"
         values: "Perf Fuchsia"
         values: "Perf Linux"
-        values: "Perf Mac 11"
         values: "Perf Mac M1 Arm64 12"
         values: "Perf Win 10"
       }
diff --git a/infra/specs/client.webrtc.json b/infra/specs/client.webrtc.json
index edeb0bf87e..a5e80532b8 100644
--- a/infra/specs/client.webrtc.json
+++ b/infra/specs/client.webrtc.json
@@ -6830,7 +6830,6 @@
       }
     ]
   },
-  "Mac64 Builder": {},
   "Mac64 Debug": {
     "isolated_scripts": [
       {
diff --git a/infra/specs/client.webrtc.perf.json b/infra/specs/client.webrtc.perf.json
index 7be9b25a20..f70010c56e 100644
--- a/infra/specs/client.webrtc.perf.json
+++ b/infra/specs/client.webrtc.perf.json
@@ -293,77 +293,6 @@
       }
     ]
   },
-  "Perf Mac 11": {
-    "isolated_scripts": [
-      {
-        "args": [
-          "--gtest_output=json:${ISOLATED_OUTDIR}/gtest_output.json"
-        ],
-        "merge": {
-          "args": [
-            "--test-suite",
-            "video_codec_perf_tests"
-          ],
-          "script": "//tools_webrtc/perf/process_perf_results.py"
-        },
-        "module_name": "//modules/video_coding:video_codec_perf_tests",
-        "name": "video_codec_perf_tests",
-        "resultdb": {
-          "result_file": "${ISOLATED_OUTDIR}/gtest_output.json",
-          "result_format": "gtest_json"
-        },
-        "swarming": {
-          "dimensions": {
-            "cpu": "x86-64",
-            "gce": "0",
-            "os": "Mac-11",
-            "pool": "WebRTC-perf"
-          },
-          "expiration": 10800,
-          "hard_timeout": 10800,
-          "idempotent": false,
-          "io_timeout": 10800
-        },
-        "test": "video_codec_perf_tests",
-        "test_id_prefix": "ninja://modules/video_coding:video_codec_perf_tests/"
-      },
-      {
-        "args": [
-          "--test_artifacts_dir=${ISOLATED_OUTDIR}",
-          "--save_worst_frame",
-          "--nologs",
-          "--gtest_output=json:${ISOLATED_OUTDIR}/gtest_output.json"
-        ],
-        "merge": {
-          "args": [
-            "--test-suite",
-            "webrtc_perf_tests"
-          ],
-          "script": "//tools_webrtc/perf/process_perf_results.py"
-        },
-        "module_name": "//:webrtc_perf_tests",
-        "name": "webrtc_perf_tests",
-        "resultdb": {
-          "result_file": "${ISOLATED_OUTDIR}/gtest_output.json",
-          "result_format": "gtest_json"
-        },
-        "swarming": {
-          "dimensions": {
-            "cpu": "x86-64",
-            "gce": "0",
-            "os": "Mac-11",
-            "pool": "WebRTC-perf"
-          },
-          "expiration": 10800,
-          "hard_timeout": 10800,
-          "idempotent": false,
-          "io_timeout": 10800
-        },
-        "test": "webrtc_perf_tests",
-        "test_id_prefix": "ninja://:webrtc_perf_tests/"
-      }
-    ]
-  },
   "Perf Mac M1 Arm64 12": {
     "isolated_scripts": [
       {
diff --git a/infra/specs/mixins.pyl b/infra/specs/mixins.pyl
index 095aa1223f..ec75d88c8f 100644
--- a/infra/specs/mixins.pyl
+++ b/infra/specs/mixins.pyl
@@ -122,13 +122,6 @@
       }
     }
   },
-  'mac11': {
-    'swarming': {
-      'dimensions': {
-        'os': 'Mac-11'
-      }
-    }
-  },
   'mac_12_arm64': {
     'swarming': {
       'dimensions': {
diff --git a/infra/specs/mixins_webrtc.pyl b/infra/specs/mixins_webrtc.pyl
index 4ff80cec96..064f6fafec 100644
--- a/infra/specs/mixins_webrtc.pyl
+++ b/infra/specs/mixins_webrtc.pyl
@@ -125,13 +125,6 @@
       }
     }
   },
-  'mac11': {
-    'swarming': {
-      'dimensions': {
-        'os': 'Mac-11'
-      }
-    }
-  },
   'perf-output': {
     'args': [
       '--isolated-script-test-perf-output=${ISOLATED_OUTDIR}/perftest-output.pb',
diff --git a/infra/specs/waterfalls.pyl b/infra/specs/waterfalls.pyl
index bddb19c0fb..48283ff153 100644
--- a/infra/specs/waterfalls.pyl
+++ b/infra/specs/waterfalls.pyl
@@ -160,7 +160,6 @@
           'isolated_scripts': 'desktop_tests',
         },
       },
-      'Mac64 Builder': {},
       'Mac64 Debug': {
         'os_type': 'mac',
         'mixins': ['mac_15_x64', 'cores-12', 'resultdb-json-format'],
@@ -277,17 +276,6 @@
           'isolated_scripts': 'perf_tests_save_worst_frame',
         },
       },
-      'Perf Mac 11': {
-        'os_type':
-        'mac',
-        'mixins': [
-          'mac11', 'x86-64', 'perf-pool', 'timeout-3h',
-          'resultdb-gtest-json-format'
-        ],
-        'test_suites': {
-          'isolated_scripts': 'perf_tests_save_worst_frame',
-        },
-      },
       'Perf Mac M1 Arm64 12': {
         'os_type':
         'mac',
diff --git a/tools_webrtc/mb/mb_config.pyl b/tools_webrtc/mb/mb_config.pyl
index 5de23c9bc5..463b063fe8 100644
--- a/tools_webrtc/mb/mb_config.pyl
+++ b/tools_webrtc/mb/mb_config.pyl
@@ -84,7 +84,6 @@
 
       # Mac
       'Mac Asan': 'mac_asan_clang_release_bot_x64',
-      'Mac64 Builder': 'pure_release_bot_x64',
       'Mac64 Debug': 'debug_bot_x64',
       'Mac64 Release': 'release_bot_x64',
       'MacARM64 M1 Release': 'release_bot_arm64',
@@ -132,7 +131,6 @@
       'Perf Android64 (R Pixel5)': 'release_bot_x64',
       'Perf Fuchsia': 'release_bot_x64_fuchsia',
       'Perf Linux': 'release_bot_x64',
-      'Perf Mac 11': 'release_bot_x64',
       'Perf Mac M1 Arm64 12': 'release_bot_x64',
       'Perf Win 10': 'release_bot_x64',
     },

From d7b28ebba1bb063e00d203fe831ec435af223813 Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Fri, 20 Feb 2026 14:39:08 +0000
Subject: [PATCH 113/472] Move all the receiver SSRC choices to the
 AudioChannel/VideoChannel

This makes the mediachannels not have to care about reportig SSRCs,
these are a concern of the RTP/RTCP module and the Channel only.

This allows deleting ~300 lines of code propagating and caching
SSRCs for sending RTCP reports that were following the wrong abstraction.

Bug: webrtc:41480926
Change-Id: I56eee4628011a13613ed8d977f3ef91ea912e4fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/442881
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46966}
---
 audio/audio_receive_stream.cc                 | 16 +-----
 audio/audio_receive_stream.h                  |  4 --
 audio/audio_receive_stream_unittest.cc        |  4 +-
 audio/channel_receive.cc                      | 25 ++++----
 audio/channel_receive.h                       |  3 -
 audio/channel_receive_unittest.cc             |  5 +-
 audio/mock_voe_channel_proxy.h                |  1 -
 call/BUILD.gn                                 |  1 +
 call/bitrate_estimator_tests.cc               |  4 --
 call/call.cc                                  | 30 +---------
 call/call.h                                   |  9 ---
 call/call_perf_tests.cc                       |  2 -
 call/call_unittest.cc                         |  1 -
 call/flexfec_receive_stream.h                 |  3 -
 call/flexfec_receive_stream_impl.cc           | 25 ++++----
 call/flexfec_receive_stream_impl.h            |  6 +-
 call/flexfec_receive_stream_unittest.cc       |  3 +-
 call/rampup_tests.cc                          |  1 -
 call/receive_stream.h                         |  9 ++-
 call/video_receive_stream.cc                  |  1 -
 media/BUILD.gn                                |  1 +
 media/base/fake_media_engine.h                |  2 -
 media/base/media_channel.h                    |  2 -
 media/engine/fake_webrtc_call.cc              | 18 ------
 media/engine/fake_webrtc_call.h               | 16 ------
 media/engine/webrtc_video_engine.cc           | 52 +----------------
 media/engine/webrtc_video_engine.h            | 10 ----
 media/engine/webrtc_video_engine_unittest.cc  | 57 -------------------
 media/engine/webrtc_voice_engine.cc           | 28 +--------
 media/engine/webrtc_voice_engine.h            |  7 ---
 media/engine/webrtc_voice_engine_unittest.cc  | 35 ------------
 modules/pacing/packet_router.cc               | 12 ++++
 modules/pacing/packet_router.h                |  4 ++
 modules/rtp_rtcp/include/rtp_rtcp_defines.h   |  5 ++
 modules/rtp_rtcp/source/rtcp_packet/dlrr.cc   |  1 -
 modules/rtp_rtcp/source/rtcp_sender.cc        | 56 +++++++++++-------
 modules/rtp_rtcp/source/rtcp_sender.h         | 18 +++++-
 modules/rtp_rtcp/source/rtp_rtcp_impl2.cc     | 19 ++++++-
 modules/rtp_rtcp/source/rtp_rtcp_impl2.h      | 25 ++++++--
 pc/rtp_transceiver.cc                         |  8 ---
 ...ck_voice_media_receive_channel_interface.h |  5 --
 rtc_tools/video_replay.cc                     |  3 -
 test/BUILD.gn                                 |  5 +-
 test/DEPS                                     |  3 +
 test/call_config_utils.cc                     |  2 -
 test/call_config_utils_unittest.cc            |  2 -
 test/call_test.cc                             |  3 -
 test/fuzzers/vp8_replay_fuzzer.cc             |  1 -
 test/fuzzers/vp9_replay_fuzzer.cc             |  1 -
 test/scenario/audio_stream.cc                 |  1 -
 test/scenario/video_stream.cc                 |  6 +-
 test/video_test_constants.h                   |  7 ++-
 video/end_to_end_tests/bandwidth_tests.cc     |  6 +-
 video/end_to_end_tests/fec_tests.cc           | 33 ++++-------
 video/end_to_end_tests/multi_stream_tester.cc |  3 -
 .../end_to_end_tests/retransmission_tests.cc  |  1 -
 video/end_to_end_tests/rtp_rtcp_tests.cc      |  2 -
 video/rtp_video_stream_receiver2.cc           | 27 +++++----
 video/rtp_video_stream_receiver2_unittest.cc  |  1 -
 video/timing/simulator/assembler.cc           |  3 -
 video/video_receive_stream2.cc                | 10 ----
 video/video_receive_stream2.h                 |  4 --
 video/video_receive_stream2_unittest.cc       |  2 -
 video/video_send_stream_tests.cc              |  9 ++-
 64 files changed, 195 insertions(+), 474 deletions(-)

diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
index 741f7e9f80..4f31b489ea 100644
--- a/audio/audio_receive_stream.cc
+++ b/audio/audio_receive_stream.cc
@@ -55,7 +55,6 @@ std::string AudioReceiveStreamInterface::Config::Rtp::ToString() const {
   char ss_buf[1024];
   SimpleStringBuilder ss(ss_buf);
   ss << "{remote_ssrc: " << remote_ssrc;
-  ss << ", local_ssrc: " << local_ssrc;
   ss << ", nack: " << nack.ToString();
   ss << ", rtcp: "
      << (rtcp_mode == RtcpMode::kCompound
@@ -89,7 +88,7 @@ std::unique_ptr<voe::ChannelReceiveInterface> CreateChannelReceive(
       static_cast<internal::AudioState*>(audio_state);
   return voe::CreateChannelReceive(
       env, neteq_factory, internal_audio_state->audio_device_module(),
-      config.rtcp_send_transport, config.rtp.local_ssrc, config.rtp.remote_ssrc,
+      config.rtcp_send_transport, config.rtp.remote_ssrc,
       config.jitter_buffer_max_packets, config.jitter_buffer_fast_accelerate,
       config.jitter_buffer_min_delay_ms, config.enable_non_sender_rtt,
       config.decoder_factory, std::move(config.frame_decryptor),
@@ -168,7 +167,6 @@ void AudioReceiveStreamImpl::ReconfigureForTesting(
 
   // SSRC can't be changed mid-stream.
   RTC_DCHECK_EQ(remote_ssrc(), config.rtp.remote_ssrc);
-  RTC_DCHECK_EQ(local_ssrc(), config.rtp.local_ssrc);
 
   // Configuration parameters which cannot be changed.
   RTC_DCHECK_EQ(config_.rtcp_send_transport, config.rtcp_send_transport);
@@ -458,18 +456,6 @@ void AudioReceiveStreamImpl::SetSyncGroup(absl::string_view sync_group) {
   config_.sync_group = std::string(sync_group);
 }
 
-void AudioReceiveStreamImpl::SetLocalSsrc(uint32_t local_ssrc) {
-  RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
-  // TODO(tommi): Consider storing local_ssrc in one place.
-  config_.rtp.local_ssrc = local_ssrc;
-  channel_receive_->OnLocalSsrcChange(local_ssrc);
-}
-
-uint32_t AudioReceiveStreamImpl::local_ssrc() const {
-  RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
-  return config_.rtp.local_ssrc;
-}
-
 const std::string& AudioReceiveStreamImpl::sync_group() const {
   RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
   return config_.sync_group;
diff --git a/audio/audio_receive_stream.h b/audio/audio_receive_stream.h
index cba510ad5d..111cd898dc 100644
--- a/audio/audio_receive_stream.h
+++ b/audio/audio_receive_stream.h
@@ -132,10 +132,6 @@ class AudioReceiveStreamImpl final : public webrtc::AudioReceiveStreamInterface,
 
   void SetSyncGroup(absl::string_view sync_group);
 
-  void SetLocalSsrc(uint32_t local_ssrc);
-
-  uint32_t local_ssrc() const;
-
   uint32_t remote_ssrc() const override {
     // The remote_ssrc member variable of config_ will never change and can be
     // considered const.
diff --git a/audio/audio_receive_stream_unittest.cc b/audio/audio_receive_stream_unittest.cc
index 652224db58..84b5407c82 100644
--- a/audio/audio_receive_stream_unittest.cc
+++ b/audio/audio_receive_stream_unittest.cc
@@ -148,7 +148,6 @@ struct ConfigHelper {
           EXPECT_THAT(codecs, ::testing::IsEmpty());
         });
 
-    stream_config_.rtp.local_ssrc = kLocalSsrc;
     stream_config_.rtp.remote_ssrc = kRemoteSsrc;
     stream_config_.rtp.nack.rtp_history_ms = 300;
     stream_config_.rtcp_send_transport = &rtcp_send_transport_;
@@ -219,10 +218,9 @@ std::vector<uint8_t> CreateRtcpSenderReport() {
 TEST(AudioReceiveStreamTest, ConfigToString) {
   AudioReceiveStreamInterface::Config config;
   config.rtp.remote_ssrc = kRemoteSsrc;
-  config.rtp.local_ssrc = kLocalSsrc;
   config.rtp.rtcp_mode = RtcpMode::kOff;
   EXPECT_EQ(
-      "{rtp: {remote_ssrc: 1234, local_ssrc: 5678, nack: "
+      "{rtp: {remote_ssrc: 1234, nack: "
       "{rtp_history_ms: 0}, rtcp: off}, "
       "rtcp_send_transport: null}",
       config.ToString());
diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index 007d3cfabb..255880c24f 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -121,7 +121,6 @@ class ChannelReceive : public ChannelReceiveInterface,
                  NetEqFactory* neteq_factory,
                  AudioDeviceModule* audio_device_module,
                  Transport* rtcp_send_transport,
-                 uint32_t local_ssrc,
                  uint32_t remote_ssrc,
                  size_t jitter_buffer_max_packets,
                  bool jitter_buffer_fast_playout,
@@ -207,8 +206,6 @@ class ChannelReceive : public ChannelReceiveInterface,
   void SetFrameDecryptor(
       scoped_refptr<FrameDecryptorInterface> frame_decryptor) override;
 
-  void OnLocalSsrcChange(uint32_t local_ssrc) override;
-
   void RtcpPacketTypesCounterUpdated(
       uint32_t ssrc,
       const RtcpPacketTypeCounter& packet_counter) override;
@@ -556,7 +553,6 @@ ChannelReceive::ChannelReceive(
     NetEqFactory* neteq_factory,
     AudioDeviceModule* audio_device_module,
     Transport* rtcp_send_transport,
-    uint32_t local_ssrc,
     uint32_t remote_ssrc,
     size_t jitter_buffer_max_packets,
     bool jitter_buffer_fast_playout,
@@ -594,14 +590,20 @@ ChannelReceive::ChannelReceive(
   configuration.receiver_only = true;
   configuration.outgoing_transport = rtcp_send_transport;
   configuration.receive_statistics = rtp_receive_statistics_.get();
-  configuration.local_media_ssrc = local_ssrc;
   configuration.rtcp_packet_type_counter_observer = this;
   configuration.non_sender_rtt_measurement = enable_non_sender_rtt;
 
   if (frame_transformer)
     InitFrameTransformerDelegate(std::move(frame_transformer));
 
-  rtp_rtcp_ = ModuleRtpRtcpImpl2::CreateReceiveModule(env_, configuration);
+  rtp_rtcp_ =
+      ModuleRtpRtcpImpl2::CreateReceiveModule(env_, configuration, [this] {
+        if (packet_router_ == nullptr) {
+          return kFallbackRtcpSsrcForAudio;
+        }
+        return packet_router_->SsrcOfFirstSender().value_or(
+            kFallbackRtcpSsrcForAudio);
+      });
   rtp_rtcp_->SetRemoteSSRC(remote_ssrc_);
 
   // Ensure that RTCP is enabled for the created channel.
@@ -774,7 +776,6 @@ void ChannelReceive::ReceivePacket(const uint8_t* packet,
 
 void ChannelReceive::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
-
   // Store playout timestamp for the received RTCP packet
   UpdatePlayoutTimestamp(true, env_.clock().CurrentTime());
 
@@ -967,11 +968,6 @@ void ChannelReceive::SetFrameDecryptor(
   frame_decryptor_ = std::move(frame_decryptor);
 }
 
-void ChannelReceive::OnLocalSsrcChange(uint32_t local_ssrc) {
-  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
-  rtp_rtcp_->SetLocalSsrc(local_ssrc);
-}
-
 NetworkStatistics ChannelReceive::GetNetworkStatistics(
     bool get_and_clear_legacy_stats) const {
   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
@@ -1207,7 +1203,6 @@ std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive(
     NetEqFactory* neteq_factory,
     AudioDeviceModule* audio_device_module,
     Transport* rtcp_send_transport,
-    uint32_t local_ssrc,
     uint32_t remote_ssrc,
     size_t jitter_buffer_max_packets,
     bool jitter_buffer_fast_playout,
@@ -1218,8 +1213,8 @@ std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive(
     const CryptoOptions& crypto_options,
     scoped_refptr<FrameTransformerInterface> frame_transformer) {
   return std::make_unique<ChannelReceive>(
-      env, neteq_factory, audio_device_module, rtcp_send_transport, local_ssrc,
-      remote_ssrc, jitter_buffer_max_packets, jitter_buffer_fast_playout,
+      env, neteq_factory, audio_device_module, rtcp_send_transport, remote_ssrc,
+      jitter_buffer_max_packets, jitter_buffer_fast_playout,
       jitter_buffer_min_delay_ms, enable_non_sender_rtt, decoder_factory,
       std::move(frame_decryptor), crypto_options, std::move(frame_transformer));
 }
diff --git a/audio/channel_receive.h b/audio/channel_receive.h
index 2191878612..dd4188898a 100644
--- a/audio/channel_receive.h
+++ b/audio/channel_receive.h
@@ -166,8 +166,6 @@ class ChannelReceiveInterface : public RtpPacketSinkInterface {
 
   virtual void SetFrameDecryptor(
       scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) = 0;
-
-  virtual void OnLocalSsrcChange(uint32_t local_ssrc) = 0;
 };
 
 std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive(
@@ -175,7 +173,6 @@ std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive(
     NetEqFactory* neteq_factory,
     AudioDeviceModule* audio_device_module,
     Transport* rtcp_send_transport,
-    uint32_t local_ssrc,
     uint32_t remote_ssrc,
     size_t jitter_buffer_max_packets,
     bool jitter_buffer_fast_playout,
diff --git a/audio/channel_receive_unittest.cc b/audio/channel_receive_unittest.cc
index 06b82c25e3..72e45c3958 100644
--- a/audio/channel_receive_unittest.cc
+++ b/audio/channel_receive_unittest.cc
@@ -31,6 +31,7 @@
 #include "api/units/timestamp.h"
 #include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
 #include "modules/audio_device/include/mock_audio_device.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/ntp_time_util.h"
 #include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
 #include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
@@ -53,7 +54,7 @@ using ::testing::NotNull;
 using ::testing::Return;
 using ::testing::Test;
 
-constexpr uint32_t kLocalSsrc = 1111;
+constexpr uint32_t kLocalSsrc = kFallbackRtcpSsrcForAudio;
 constexpr uint32_t kRemoteSsrc = 2222;
 // We run RTP data with 8 kHz PCMA (fixed payload type 8).
 constexpr char kPayloadName[] = "PCMA";
@@ -74,7 +75,7 @@ class ChannelReceiveTest : public Test {
     auto channel = CreateChannelReceive(
         CreateEnvironment(time_controller_.GetClock(), &log_),
         /* neteq_factory= */ nullptr, audio_device_module_.get(), &transport_,
-        kLocalSsrc, kRemoteSsrc,
+        kRemoteSsrc,
         /* jitter_buffer_max_packets= */ 0,
         /* jitter_buffer_fast_playout= */ false,
         /* jitter_buffer_min_delay_ms= */ 0,
diff --git a/audio/mock_voe_channel_proxy.h b/audio/mock_voe_channel_proxy.h
index 407b537229..db4d73adaf 100644
--- a/audio/mock_voe_channel_proxy.h
+++ b/audio/mock_voe_channel_proxy.h
@@ -128,7 +128,6 @@ class MockChannelReceive : public voe::ChannelReceiveInterface {
       SetFrameDecryptor,
       (webrtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor),
       (override));
-  MOCK_METHOD(void, OnLocalSsrcChange, (uint32_t local_ssrc), (override));
   MOCK_METHOD(void, SetMaximumBufferPackets, (size_t max_packets), (override));
   MOCK_METHOD(void, SetFastAccelerate, (bool enable), (override));
 };
diff --git a/call/BUILD.gn b/call/BUILD.gn
index 3a70a8f6ca..dd7cbac1ac 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -329,6 +329,7 @@ rtc_library("call") {
     "../logging:rtc_event_video",
     "../logging:rtc_stream_config",
     "../modules/congestion_controller",
+    "../modules/pacing",
     "../modules/rtp_rtcp",
     "../modules/rtp_rtcp:rtp_rtcp_format",
     "../modules/video_coding",
diff --git a/call/bitrate_estimator_tests.cc b/call/bitrate_estimator_tests.cc
index 58cbaea044..7e71efbb76 100644
--- a/call/bitrate_estimator_tests.cc
+++ b/call/bitrate_estimator_tests.cc
@@ -7,7 +7,6 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#include <cstddef>
 #include <functional>
 #include <list>
 #include <memory>
@@ -152,8 +151,6 @@ class BitrateEstimatorTest : public test::CallTest {
           VideoReceiveStreamInterface::Config(receive_transport_.get());
       // receive_config_.decoders will be set by every stream separately.
       receive_config_.rtp.remote_ssrc = GetVideoSendConfig()->rtp.ssrcs[0];
-      receive_config_.rtp.local_ssrc =
-          test::VideoTestConstants::kReceiverLocalVideoSsrc;
     });
   }
 
@@ -209,7 +206,6 @@ class BitrateEstimatorTest : public test::CallTest {
       test_->receive_config_.decoders.push_back(decoder);
       test_->receive_config_.rtp.remote_ssrc =
           test_->GetVideoSendConfig()->rtp.ssrcs[0];
-      test_->receive_config_.rtp.local_ssrc++;
       test_->receive_config_.renderer = &test->fake_renderer_;
       video_receive_stream_ = test_->receiver_call_->CreateVideoReceiveStream(
           test_->receive_config_.Copy());
diff --git a/call/call.cc b/call/call.cc
index 2fedbb3268..fc9c544cbb 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -112,7 +112,6 @@ std::unique_ptr<rtclog::StreamConfig> CreateRtcLogStreamConfig(
     const VideoReceiveStreamInterface::Config& config) {
   auto rtclog_config = std::make_unique<rtclog::StreamConfig>();
   rtclog_config->remote_ssrc = config.rtp.remote_ssrc;
-  rtclog_config->local_ssrc = config.rtp.local_ssrc;
   rtclog_config->rtx_ssrc = config.rtp.rtx_ssrc;
   rtclog_config->rtcp_mode = config.rtp.rtcp_mode;
 
@@ -129,7 +128,6 @@ std::unique_ptr<rtclog::StreamConfig> CreateRtcLogStreamConfig(
     const VideoSendStream::Config& config,
     size_t ssrc_index) {
   auto rtclog_config = std::make_unique<rtclog::StreamConfig>();
-  rtclog_config->local_ssrc = config.rtp.ssrcs[ssrc_index];
   if (ssrc_index < config.rtp.rtx.ssrcs.size()) {
     rtclog_config->rtx_ssrc = config.rtp.rtx.ssrcs[ssrc_index];
   }
@@ -146,7 +144,6 @@ std::unique_ptr<rtclog::StreamConfig> CreateRtcLogStreamConfig(
     const AudioReceiveStreamInterface::Config& config) {
   auto rtclog_config = std::make_unique<rtclog::StreamConfig>();
   rtclog_config->remote_ssrc = config.rtp.remote_ssrc;
-  rtclog_config->local_ssrc = config.rtp.local_ssrc;
   return rtclog_config;
 }
 
@@ -271,13 +268,6 @@ class Call final : public webrtc::Call,
   void OnAudioTransportOverheadChanged(
       int transport_overhead_per_packet) override;
 
-  void OnLocalSsrcUpdated(webrtc::AudioReceiveStreamInterface& stream,
-                          uint32_t local_ssrc) override;
-  void OnLocalSsrcUpdated(VideoReceiveStreamInterface& stream,
-                          uint32_t local_ssrc) override;
-  void OnLocalSsrcUpdated(FlexfecReceiveStream& stream,
-                          uint32_t local_ssrc) override;
-
   void OnUpdateSyncGroup(webrtc::AudioReceiveStreamInterface& stream,
                          absl::string_view sync_group) override;
 
@@ -1053,7 +1043,7 @@ FlexfecReceiveStream* Call::CreateFlexfecReceiveStream(
   // in a valid state, since OnRtpPacket runs on the same thread.
   FlexfecReceiveStreamImpl* receive_stream = new FlexfecReceiveStreamImpl(
       env_, std::move(config), &video_receiver_controller_,
-      call_stats_->AsRtcpRttStats());
+      transport_send_->packet_router(), call_stats_->AsRtcpRttStats());
 
   // TODO(bugs.webrtc.org/11993): Set this up asynchronously on the network
   // thread.
@@ -1220,24 +1210,6 @@ void Call::UpdateAggregateNetworkState() {
   transport_send_->OnNetworkAvailability(aggregate_network_up);
 }
 
-void Call::OnLocalSsrcUpdated(webrtc::AudioReceiveStreamInterface& stream,
-                              uint32_t local_ssrc) {
-  RTC_DCHECK_RUN_ON(worker_thread_);
-  static_cast<webrtc::AudioReceiveStreamImpl&>(stream).SetLocalSsrc(local_ssrc);
-}
-
-void Call::OnLocalSsrcUpdated(VideoReceiveStreamInterface& stream,
-                              uint32_t local_ssrc) {
-  RTC_DCHECK_RUN_ON(worker_thread_);
-  static_cast<VideoReceiveStream2&>(stream).SetLocalSsrc(local_ssrc);
-}
-
-void Call::OnLocalSsrcUpdated(FlexfecReceiveStream& stream,
-                              uint32_t local_ssrc) {
-  RTC_DCHECK_RUN_ON(worker_thread_);
-  static_cast<FlexfecReceiveStreamImpl&>(stream).SetLocalSsrc(local_ssrc);
-}
-
 void Call::OnUpdateSyncGroup(webrtc::AudioReceiveStreamInterface& stream,
                              absl::string_view sync_group) {
   RTC_DCHECK_RUN_ON(worker_thread_);
diff --git a/call/call.h b/call/call.h
index 902b60d306..8f58370194 100644
--- a/call/call.h
+++ b/call/call.h
@@ -130,15 +130,6 @@ class Call {
   virtual void OnAudioTransportOverheadChanged(
       int transport_overhead_per_packet) = 0;
 
-  // Called when a receive stream's local ssrc has changed and association with
-  // send streams needs to be updated.
-  virtual void OnLocalSsrcUpdated(AudioReceiveStreamInterface& stream,
-                                  uint32_t local_ssrc) = 0;
-  virtual void OnLocalSsrcUpdated(VideoReceiveStreamInterface& stream,
-                                  uint32_t local_ssrc) = 0;
-  virtual void OnLocalSsrcUpdated(FlexfecReceiveStream& stream,
-                                  uint32_t local_ssrc) = 0;
-
   virtual void OnUpdateSyncGroup(AudioReceiveStreamInterface& stream,
                                  absl::string_view sync_group) = 0;
 
diff --git a/call/call_perf_tests.cc b/call/call_perf_tests.cc
index af4229f9dd..9cd29d8d1c 100644
--- a/call/call_perf_tests.cc
+++ b/call/call_perf_tests.cc
@@ -212,7 +212,6 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
                                       absl::string_view test_label) {
   const char* kSyncGroup = "av_sync";
   const uint32_t kAudioSendSsrc = 1234;
-  const uint32_t kAudioRecvSsrc = 5678;
 
   BuiltInNetworkBehaviorConfig audio_net_config;
   audio_net_config.queue_delay_ms = 500;
@@ -324,7 +323,6 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
 
     AudioReceiveStreamInterface::Config audio_recv_config;
     audio_recv_config.rtp.remote_ssrc = kAudioSendSsrc;
-    audio_recv_config.rtp.local_ssrc = kAudioRecvSsrc;
     audio_recv_config.rtcp_send_transport = receive_transport.get();
     audio_recv_config.sync_group = kSyncGroup;
     audio_recv_config.decoder_factory = audio_decoder_factory_;
diff --git a/call/call_unittest.cc b/call/call_unittest.cc
index b96455ade4..1a4f1e5bb7 100644
--- a/call/call_unittest.cc
+++ b/call/call_unittest.cc
@@ -510,7 +510,6 @@ class CallRtcEventLogTest : public ::testing::Test {
     video_config.decoders.emplace_back(SdpVideoFormat("VP8"),
                                        /*payload_type=*/96);
     video_config.decoder_factory = &video_decoder_factory_;
-    video_config.rtp.local_ssrc = kVideoSsrc + 1;
     video_config.renderer = &renderer_;
     video_stream_ = call_->CreateVideoReceiveStream(std::move(video_config));
 
diff --git a/call/flexfec_receive_stream.h b/call/flexfec_receive_stream.h
index d78481a4bd..95cf9eb2ae 100644
--- a/call/flexfec_receive_stream.h
+++ b/call/flexfec_receive_stream.h
@@ -48,9 +48,6 @@ class FlexfecReceiveStream : public RtpPacketSinkInterface,
     // post initialization.
     uint32_t remote_ssrc = 0;
 
-    // Sender SSRC used for sending RTCP (such as receiver reports).
-    uint32_t local_ssrc = 0;
-
     // Vector containing a single element, corresponding to the SSRC of the
     // media stream being protected by this FlexFEC stream. The vector MUST have
     // size 1.
diff --git a/call/flexfec_receive_stream_impl.cc b/call/flexfec_receive_stream_impl.cc
index d10492d081..8b16d4cbc8 100644
--- a/call/flexfec_receive_stream_impl.cc
+++ b/call/flexfec_receive_stream_impl.cc
@@ -11,7 +11,6 @@
 #include "call/flexfec_receive_stream_impl.h"
 
 #include <cstddef>
-#include <cstdint>
 #include <memory>
 #include <string>
 
@@ -21,7 +20,9 @@
 #include "call/flexfec_receive_stream.h"
 #include "call/rtp_stream_receiver_controller_interface.h"
 #include "logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.h"
+#include "modules/pacing/packet_router.h"
 #include "modules/rtp_rtcp/include/flexfec_receiver.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
 #include "rtc_base/checks.h"
@@ -35,7 +36,6 @@ std::string FlexfecReceiveStream::Config::ToString() const {
   SimpleStringBuilder ss(buf);
   ss << "{payload_type: " << payload_type;
   ss << ", remote_ssrc: " << remote_ssrc;
-  ss << ", local_ssrc: " << local_ssrc;
   ss << ", protected_media_ssrcs: [";
   size_t i = 0;
   for (; i + 1 < protected_media_ssrcs.size(); ++i)
@@ -107,6 +107,7 @@ FlexfecReceiveStreamImpl::FlexfecReceiveStreamImpl(
     const Environment& env,
     Config config,
     RecoveredPacketReceiver* recovered_packet_receiver,
+    PacketRouter* packet_router,
     RtcpRttStats* rtt_stats)
     : env_(env),
       remote_ssrc_(config.remote_ssrc),
@@ -121,8 +122,16 @@ FlexfecReceiveStreamImpl::FlexfecReceiveStreamImpl(
            .receiver_only = true,
            .receive_statistics = rtp_receive_statistics_.get(),
            .outgoing_transport = config.rtcp_send_transport,
-           .rtt_stats = rtt_stats,
-           .local_media_ssrc = config.local_ssrc})) {
+           .rtt_stats = rtt_stats},
+          [packet_router] {
+            // Use the same logic as for the video receiver.
+            if (packet_router != nullptr) {
+              return packet_router->SsrcOfFirstSender().value_or(
+                  kFallbackRtcpSsrcForVideo);
+            } else {
+              return kFallbackRtcpSsrcForVideo;
+            }
+          })) {
   RTC_LOG(LS_INFO) << "FlexfecReceiveStreamImpl: " << config.ToString();
   RTC_DCHECK_GE(payload_type_, -1);
 
@@ -182,12 +191,4 @@ int FlexfecReceiveStreamImpl::payload_type() const {
   return payload_type_;
 }
 
-void FlexfecReceiveStreamImpl::SetLocalSsrc(uint32_t local_ssrc) {
-  RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
-  if (local_ssrc == rtp_rtcp_->local_media_ssrc())
-    return;
-
-  rtp_rtcp_->SetLocalSsrc(local_ssrc);
-}
-
 }  // namespace webrtc
diff --git a/call/flexfec_receive_stream_impl.h b/call/flexfec_receive_stream_impl.h
index a437a6459c..211cd66238 100644
--- a/call/flexfec_receive_stream_impl.h
+++ b/call/flexfec_receive_stream_impl.h
@@ -19,6 +19,7 @@
 #include "api/sequence_checker.h"
 #include "call/flexfec_receive_stream.h"
 #include "call/rtp_packet_sink_interface.h"
+#include "modules/pacing/packet_router.h"
 #include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
 #include "rtc_base/system/no_unique_address.h"
 #include "rtc_base/thread_annotations.h"
@@ -38,6 +39,7 @@ class FlexfecReceiveStreamImpl : public FlexfecReceiveStream {
   FlexfecReceiveStreamImpl(const Environment& env,
                            Config config,
                            RecoveredPacketReceiver* recovered_packet_receiver,
+                           PacketRouter* packet_router,
                            RtcpRttStats* rtt_stats);
   // Destruction happens on the worker thread. Prior to destruction the caller
   // must ensure that a registration with the transport has been cleared. See
@@ -61,10 +63,6 @@ class FlexfecReceiveStreamImpl : public FlexfecReceiveStream {
   void SetPayloadType(int payload_type) override;
   int payload_type() const override;
 
-  // Updates the `rtp_video_stream_receiver_`'s `local_ssrc` when the default
-  // sender has been created, changed or removed.
-  void SetLocalSsrc(uint32_t local_ssrc);
-
   uint32_t remote_ssrc() const { return remote_ssrc_; }
 
   void SetRtcpMode(RtcpMode mode) override {
diff --git a/call/flexfec_receive_stream_unittest.cc b/call/flexfec_receive_stream_unittest.cc
index 0ea635ef2b..be769db5d1 100644
--- a/call/flexfec_receive_stream_unittest.cc
+++ b/call/flexfec_receive_stream_unittest.cc
@@ -64,7 +64,6 @@ TEST(FlexfecReceiveStreamConfigTest, IsCompleteAndEnabled) {
   MockTransport rtcp_send_transport;
   FlexfecReceiveStream::Config config(&rtcp_send_transport);
 
-  config.local_ssrc = 18374743;
   config.rtcp_mode = RtcpMode::kCompound;
   EXPECT_FALSE(config.IsCompleteAndEnabled());
 
@@ -87,7 +86,7 @@ class FlexfecReceiveStreamTest : public ::testing::Test {
       : config_(CreateDefaultConfig(&rtcp_send_transport_)) {
     receive_stream_ = std::make_unique<FlexfecReceiveStreamImpl>(
         CreateEnvironment(&log_), config_, &recovered_packet_receiver_,
-        &rtt_stats_);
+        /* packet_router= */ nullptr, &rtt_stats_);
     receive_stream_->RegisterWithTransport(&rtp_stream_receiver_controller_);
   }
 
diff --git a/call/rampup_tests.cc b/call/rampup_tests.cc
index 5dd1283efb..621597afb4 100644
--- a/call/rampup_tests.cc
+++ b/call/rampup_tests.cc
@@ -281,7 +281,6 @@ void RampUpTester::ModifyFlexfecConfigs(
   (*receive_configs)[0].remote_ssrc =
       test::VideoTestConstants::kFlexfecSendSsrc;
   (*receive_configs)[0].protected_media_ssrcs = {video_ssrcs_[0]};
-  (*receive_configs)[0].local_ssrc = video_ssrcs_[0];
 }
 
 void RampUpTester::OnCallsCreated(Call* sender_call,
diff --git a/call/receive_stream.h b/call/receive_stream.h
index 32678cbd8d..231f448056 100644
--- a/call/receive_stream.h
+++ b/call/receive_stream.h
@@ -34,11 +34,10 @@ class ReceiveStreamInterface {
     // This member will not change mid-stream and can be assumed to be const
     // post initialization.
     uint32_t remote_ssrc = 0;
-
-    // Sender SSRC used for sending RTCP (such as receiver reports).
-    // This value may change mid-stream and must be done on the same thread
-    // that the value is read on (i.e. packet delivery).
-    uint32_t local_ssrc = 0;
+    // This member is no longer used by WebRTC, but retained in order to
+    // allow downstream code to compile.
+    // TODO: issues.webrtc.org/41480926 - Delete when downstream changed.
+    [[deprecated("No longer used")]] uint32_t local_ssrc = 0;
   };
 
  protected:
diff --git a/call/video_receive_stream.cc b/call/video_receive_stream.cc
index c16cd59b4d..5a1c168f29 100644
--- a/call/video_receive_stream.cc
+++ b/call/video_receive_stream.cc
@@ -142,7 +142,6 @@ VideoReceiveStreamInterface::Config::Rtp::~Rtp() = default;
 std::string VideoReceiveStreamInterface::Config::Rtp::ToString() const {
   StringBuilder ss;
   ss << "{remote_ssrc: " << remote_ssrc;
-  ss << ", local_ssrc: " << local_ssrc;
   ss << ", rtcp_mode: "
      << (rtcp_mode == RtcpMode::kCompound ? "RtcpMode::kCompound"
                                           : "RtcpMode::kReducedSize");
diff --git a/media/BUILD.gn b/media/BUILD.gn
index febe66aa1e..a3075d9db0 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -601,6 +601,7 @@ rtc_library("rtc_audio_video") {
     "../api:frame_transformer_interface",
     "../api:make_ref_counted",
     "../api:media_stream_interface",
+    "../api:payload_type",
     "../api:priority",
     "../api:rtc_error",
     "../api:rtp_headers",
diff --git a/media/base/fake_media_engine.h b/media/base/fake_media_engine.h
index 3478d831be..69e9ef27cd 100644
--- a/media/base/fake_media_engine.h
+++ b/media/base/fake_media_engine.h
@@ -128,8 +128,6 @@ class RtpReceiveChannelHelper : public Base, public MediaChannelUtil {
   std::optional<uint32_t> GetUnsignaledSsrc() const override {
     return std::nullopt;
   }
-  void ChooseReceiverReportSsrc(
-      const std::set<uint32_t>& /* choices */) override {}
 
   virtual bool SetLocalSsrc(const StreamParams& /* sp */) { return true; }
   void OnDemuxerCriteriaUpdatePending() override {}
diff --git a/media/base/media_channel.h b/media/base/media_channel.h
index 6d45080926..496726501a 100644
--- a/media/base/media_channel.h
+++ b/media/base/media_channel.h
@@ -268,8 +268,6 @@ class MediaReceiveChannelInterface {
   virtual void OnPacketReceived(const RtpPacketReceived& packet) = 0;
   // Gets the current unsignaled receive stream's SSRC, if there is one.
   virtual std::optional<uint32_t> GetUnsignaledSsrc() const = 0;
-  // Sets the local SSRC for listening to incoming RTCP reports.
-  virtual void ChooseReceiverReportSsrc(const std::set<uint32_t>& choices) = 0;
   // This is currently a workaround because of the demuxer state being managed
   // across two separate threads. Once the state is consistently managed on
   // the same thread (network), this workaround can be removed.
diff --git a/media/engine/fake_webrtc_call.cc b/media/engine/fake_webrtc_call.cc
index b8a28858a9..cc537a4667 100644
--- a/media/engine/fake_webrtc_call.cc
+++ b/media/engine/fake_webrtc_call.cc
@@ -740,24 +740,6 @@ void FakeCall::SignalChannelNetworkState(MediaType media, NetworkState state) {
 void FakeCall::OnAudioTransportOverheadChanged(
     int /* transport_overhead_per_packet */) {}
 
-void FakeCall::OnLocalSsrcUpdated(AudioReceiveStreamInterface& stream,
-                                  uint32_t local_ssrc) {
-  auto& fake_stream = static_cast<FakeAudioReceiveStream&>(stream);
-  fake_stream.SetLocalSsrc(local_ssrc);
-}
-
-void FakeCall::OnLocalSsrcUpdated(VideoReceiveStreamInterface& stream,
-                                  uint32_t local_ssrc) {
-  auto& fake_stream = static_cast<FakeVideoReceiveStream&>(stream);
-  fake_stream.SetLocalSsrc(local_ssrc);
-}
-
-void FakeCall::OnLocalSsrcUpdated(FlexfecReceiveStream& stream,
-                                  uint32_t local_ssrc) {
-  auto& fake_stream = static_cast<FakeFlexfecReceiveStream&>(stream);
-  fake_stream.SetLocalSsrc(local_ssrc);
-}
-
 void FakeCall::OnUpdateSyncGroup(AudioReceiveStreamInterface& stream,
                                  absl::string_view sync_group) {
   auto& fake_stream = static_cast<FakeAudioReceiveStream&>(stream);
diff --git a/media/engine/fake_webrtc_call.h b/media/engine/fake_webrtc_call.h
index 892963eb23..ac12bd53f0 100644
--- a/media/engine/fake_webrtc_call.h
+++ b/media/engine/fake_webrtc_call.h
@@ -132,10 +132,6 @@ class FakeAudioReceiveStream final : public AudioReceiveStreamInterface {
     return base_mininum_playout_delay_ms_;
   }
 
-  void SetLocalSsrc(uint32_t local_ssrc) {
-    config_.rtp.local_ssrc = local_ssrc;
-  }
-
   void SetSyncGroup(absl::string_view sync_group) {
     config_.sync_group = std::string(sync_group);
   }
@@ -287,10 +283,6 @@ class FakeVideoReceiveStream final : public VideoReceiveStreamInterface {
     return base_mininum_playout_delay_ms_;
   }
 
-  void SetLocalSsrc(uint32_t local_ssrc) {
-    config_.rtp.local_ssrc = local_ssrc;
-  }
-
   void UpdateRtxSsrc(uint32_t ssrc) override { config_.rtp.rtx_ssrc = ssrc; }
 
   void SetFrameDecryptor(scoped_refptr<FrameDecryptorInterface>
@@ -364,8 +356,6 @@ class FakeFlexfecReceiveStream final : public FlexfecReceiveStream {
  public:
   explicit FakeFlexfecReceiveStream(const FlexfecReceiveStream::Config config);
 
-  void SetLocalSsrc(uint32_t local_ssrc) { config_.local_ssrc = local_ssrc; }
-
   void SetRtcpMode(RtcpMode mode) override { config_.rtcp_mode = mode; }
 
   int payload_type() const override { return config_.payload_type; }
@@ -492,12 +482,6 @@ class FakeCall final : public Call, public PacketReceiver {
   void SignalChannelNetworkState(MediaType media, NetworkState state) override;
   void OnAudioTransportOverheadChanged(
       int transport_overhead_per_packet) override;
-  void OnLocalSsrcUpdated(AudioReceiveStreamInterface& stream,
-                          uint32_t local_ssrc) override;
-  void OnLocalSsrcUpdated(VideoReceiveStreamInterface& stream,
-                          uint32_t local_ssrc) override;
-  void OnLocalSsrcUpdated(FlexfecReceiveStream& stream,
-                          uint32_t local_ssrc) override;
   void OnUpdateSyncGroup(AudioReceiveStreamInterface& stream,
                          absl::string_view sync_group) override;
   void OnSentPacket(const SentPacketInfo& sent_packet) override;
diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc
index c83a23e5a2..38d6909ea2 100644
--- a/media/engine/webrtc_video_engine.cc
+++ b/media/engine/webrtc_video_engine.cc
@@ -38,6 +38,7 @@
 #include "api/make_ref_counted.h"
 #include "api/media_stream_interface.h"
 #include "api/media_types.h"
+#include "api/payload_type.h"
 #include "api/priority.h"
 #include "api/rtc_error.h"
 #include "api/rtp_headers.h"
@@ -111,8 +112,6 @@ constexpr int64_t kUnsignaledSsrcCooldownMs = kNumMillisecsPerSec / 2;
 // duration hasn't been implemented.
 const int kNackHistoryMs = 1000;
 
-const int kDefaultRtcpReceiverReportSsrc = 1;
-
 // Minimum time interval for logging stats.
 const int64_t kStatsLogIntervalMs = 10000;
 
@@ -2809,7 +2808,6 @@ WebRtcVideoReceiveChannel::WebRtcVideoReceiveChannel(
       crypto_options_(crypto_options),
       receive_buffer_size_(ParseReceiveBufferSize(env_.field_trials())) {
   RTC_DCHECK_RUN_ON(&thread_checker_);
-  rtcp_receiver_report_ssrc_ = kDefaultRtcpReceiverReportSsrc;
   // Crash if MapCodecs fails.
   recv_codecs_ = MapCodecs(GetPayloadTypesAndDefaultCodecs(
                                decoder_factory_, /*is_decoder_factory=*/true,
@@ -2963,32 +2961,6 @@ bool WebRtcVideoReceiveChannel::SetReceiverParameters(
   return true;
 }
 
-void WebRtcVideoReceiveChannel::SetReceiverReportSsrc(uint32_t ssrc) {
-  RTC_DCHECK_RUN_ON(&thread_checker_);
-  if (ssrc == rtcp_receiver_report_ssrc_)
-    return;
-
-  rtcp_receiver_report_ssrc_ = ssrc;
-  for (auto& [unused, receive_stream] : receive_streams_)
-    receive_stream->SetLocalSsrc(ssrc);
-}
-
-void WebRtcVideoReceiveChannel::ChooseReceiverReportSsrc(
-    const std::set<uint32_t>& choices) {
-  RTC_DCHECK_RUN_ON(&thread_checker_);
-  // If we can continue using the current receiver report, do so.
-  if (choices.find(rtcp_receiver_report_ssrc_) != choices.end()) {
-    return;
-  }
-  // Go back to the default if list has been emptied.
-  if (choices.empty()) {
-    SetReceiverReportSsrc(kDefaultRtcpReceiverReportSsrc);
-    return;
-  }
-  // Any number is as good as any other.
-  SetReceiverReportSsrc(*choices.begin());
-}
-
 void WebRtcVideoReceiveChannel::SetReceive(bool receive) {
   RTC_DCHECK_RUN_ON(&thread_checker_);
   TRACE_EVENT0("webrtc", "WebRtcVideoReceiveChannel::SetReceive");
@@ -3094,19 +3066,6 @@ void WebRtcVideoReceiveChannel::ConfigureReceiverRtp(
   uint32_t ssrc = sp.first_ssrc();
 
   config->rtp.remote_ssrc = ssrc;
-  config->rtp.local_ssrc = rtcp_receiver_report_ssrc_;
-
-  // TODO(pbos): This protection is against setting the same local ssrc as
-  // remote which is not permitted by the lower-level API. RTCP requires a
-  // corresponding sender SSRC. Figure out what to do when we don't have
-  // (receive-only) or know a good local SSRC.
-  if (config->rtp.remote_ssrc == config->rtp.local_ssrc) {
-    if (config->rtp.local_ssrc != kDefaultRtcpReceiverReportSsrc) {
-      config->rtp.local_ssrc = kDefaultRtcpReceiverReportSsrc;
-    } else {
-      config->rtp.local_ssrc = kDefaultRtcpReceiverReportSsrc + 1;
-    }
-  }
 
   // The mode and rtx time is determined by a call to the configuration
   // function.
@@ -3119,7 +3078,6 @@ void WebRtcVideoReceiveChannel::ConfigureReceiverRtp(
   if (!env_.field_trials().IsDisabled("WebRTC-FlexFEC-03-Advertised") &&
       sp.GetFecFrSsrc(ssrc, &flexfec_config->remote_ssrc)) {
     flexfec_config->protected_media_ssrcs = {ssrc};
-    flexfec_config->local_ssrc = config->rtp.local_ssrc;
     flexfec_config->rtcp_mode = config->rtp.rtcp_mode;
   }
 }
@@ -3997,14 +3955,6 @@ void WebRtcVideoReceiveChannel::WebRtcVideoReceiveStream::
     stream_->SetDepacketizerToDecoderFrameTransformer(frame_transformer);
 }
 
-void WebRtcVideoReceiveChannel::WebRtcVideoReceiveStream::SetLocalSsrc(
-    uint32_t ssrc) {
-  config_.rtp.local_ssrc = ssrc;
-  call_->OnLocalSsrcUpdated(stream(), ssrc);
-  if (flexfec_stream_)
-    call_->OnLocalSsrcUpdated(*flexfec_stream_, ssrc);
-}
-
 void WebRtcVideoReceiveChannel::WebRtcVideoReceiveStream::UpdateRtxSsrc(
     uint32_t ssrc) {
   stream_->UpdateRtxSsrc(ssrc);
diff --git a/media/engine/webrtc_video_engine.h b/media/engine/webrtc_video_engine.h
index 0acd1c514d..759a6e4039 100644
--- a/media/engine/webrtc_video_engine.h
+++ b/media/engine/webrtc_video_engine.h
@@ -549,10 +549,6 @@ class WebRtcVideoReceiveChannel : public MediaChannelUtil,
 
   std::optional<int> GetBaseMinimumPlayoutDelayMs(uint32_t ssrc) const override;
 
-  // Choose one of the available SSRCs (or default if none) as the current
-  // receiver report SSRC.
-  void ChooseReceiverReportSsrc(const std::set<uint32_t>& choices) override;
-
   // E2E Encrypted Video Frame API
   // Set a frame decryptor to a particular ssrc that will intercept all
   // incoming video frames and attempt to decrypt them before forwarding the
@@ -610,10 +606,6 @@ class WebRtcVideoReceiveChannel : public MediaChannelUtil,
   void DeleteReceiveStream(WebRtcVideoReceiveStream* stream)
       RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
 
-  // Called when the local ssrc changes. Sets `rtcp_receiver_report_ssrc_` and
-  // updates the receive streams.
-  void SetReceiverReportSsrc(uint32_t ssrc) RTC_RUN_ON(&thread_checker_);
-
   // Wrapper for the receiver part, contains configs etc. that are needed to
   // reconstruct the underlying VideoReceiveStreamInterface.
   class WebRtcVideoReceiveStream : public VideoSinkInterface<VideoFrame> {
@@ -669,7 +661,6 @@ class WebRtcVideoReceiveChannel : public MediaChannelUtil,
     void SetDepacketizerToDecoderFrameTransformer(
         scoped_refptr<FrameTransformerInterface> frame_transformer);
 
-    void SetLocalSsrc(uint32_t local_ssrc);
     void UpdateRtxSsrc(uint32_t ssrc);
     void StartReceiveStream();
     void StopReceiveStream();
@@ -734,7 +725,6 @@ class WebRtcVideoReceiveChannel : public MediaChannelUtil,
       SequenceChecker::kDetached};
   RTC_NO_UNIQUE_ADDRESS SequenceChecker thread_checker_;
 
-  uint32_t rtcp_receiver_report_ssrc_ RTC_GUARDED_BY(thread_checker_);
   bool receiving_ RTC_GUARDED_BY(&thread_checker_);
   Call* const call_;
 
diff --git a/media/engine/webrtc_video_engine_unittest.cc b/media/engine/webrtc_video_engine_unittest.cc
index e1c263d4fd..e7b2aa342c 100644
--- a/media/engine/webrtc_video_engine_unittest.cc
+++ b/media/engine/webrtc_video_engine_unittest.cc
@@ -2749,11 +2749,6 @@ class WebRtcVideoChannelTest : public WebRtcVideoEngineTest {
     receive_channel_ =
         engine_->CreateReceiveChannel(env_, fake_call_.get(), GetMediaConfig(),
                                       VideoOptions(), CryptoOptions());
-    send_channel_->SetSsrcListChangedCallback(
-        [receive_channel =
-             receive_channel_.get()](const std::set<uint32_t>& choices) {
-          receive_channel->ChooseReceiverReportSsrc(choices);
-        });
     send_channel_->OnReadyToSend(true);
     receive_channel_->SetReceive(true);
     last_ssrc_ = 123;
@@ -4751,8 +4746,6 @@ TEST_F(WebRtcVideoChannelFlexfecRecvTest, SetRecvCodecsWithFec) {
   const FakeVideoReceiveStream* video_stream = video_streams.front();
   const VideoReceiveStreamInterface::Config& video_stream_config =
       video_stream->GetConfig();
-  EXPECT_EQ(video_stream_config.rtp.local_ssrc,
-            flexfec_stream_config.local_ssrc);
   EXPECT_EQ(video_stream_config.rtp.rtcp_mode, flexfec_stream_config.rtcp_mode);
   EXPECT_EQ(video_stream_config.rtcp_send_transport,
             flexfec_stream_config.rtcp_send_transport);
@@ -9585,56 +9578,6 @@ TEST_F(WebRtcVideoChannelTest,
   EXPECT_TRUE(receive_channel_->AddRecvStream(params));
 }
 
-void WebRtcVideoChannelTest::TestReceiverLocalSsrcConfiguration(
-    bool receiver_first) {
-  EXPECT_TRUE(send_channel_->SetSenderParameters(send_parameters_));
-
-  const uint32_t kSenderSsrc = 0xC0FFEE;
-  const uint32_t kSecondSenderSsrc = 0xBADCAFE;
-  const uint32_t kReceiverSsrc = 0x4711;
-  const uint32_t kExpectedDefaultReceiverSsrc = 1;
-
-  if (receiver_first) {
-    AddRecvStream(StreamParams::CreateLegacy(kReceiverSsrc));
-    std::vector<FakeVideoReceiveStream*> receive_streams =
-        fake_call_->GetVideoReceiveStreams();
-    ASSERT_EQ(1u, receive_streams.size());
-    // Default local SSRC when we have no sender.
-    EXPECT_EQ(kExpectedDefaultReceiverSsrc,
-              receive_streams[0]->GetConfig().rtp.local_ssrc);
-  }
-  AddSendStream(StreamParams::CreateLegacy(kSenderSsrc));
-  if (!receiver_first)
-    AddRecvStream(StreamParams::CreateLegacy(kReceiverSsrc));
-  std::vector<FakeVideoReceiveStream*> receive_streams =
-      fake_call_->GetVideoReceiveStreams();
-  ASSERT_EQ(1u, receive_streams.size());
-  EXPECT_EQ(kSenderSsrc, receive_streams[0]->GetConfig().rtp.local_ssrc);
-
-  // Removing first sender should fall back to another (in this case the second)
-  // local send stream's SSRC.
-  AddSendStream(StreamParams::CreateLegacy(kSecondSenderSsrc));
-  ASSERT_TRUE(send_channel_->RemoveSendStream(kSenderSsrc));
-  receive_streams = fake_call_->GetVideoReceiveStreams();
-  ASSERT_EQ(1u, receive_streams.size());
-  EXPECT_EQ(kSecondSenderSsrc, receive_streams[0]->GetConfig().rtp.local_ssrc);
-
-  // Removing the last sender should fall back to default local SSRC.
-  ASSERT_TRUE(send_channel_->RemoveSendStream(kSecondSenderSsrc));
-  receive_streams = fake_call_->GetVideoReceiveStreams();
-  ASSERT_EQ(1u, receive_streams.size());
-  EXPECT_EQ(kExpectedDefaultReceiverSsrc,
-            receive_streams[0]->GetConfig().rtp.local_ssrc);
-}
-
-TEST_F(WebRtcVideoChannelTest, ConfiguresLocalSsrc) {
-  TestReceiverLocalSsrcConfiguration(false);
-}
-
-TEST_F(WebRtcVideoChannelTest, ConfiguresLocalSsrcOnExistingReceivers) {
-  TestReceiverLocalSsrcConfiguration(true);
-}
-
 TEST_F(WebRtcVideoChannelTest, Simulcast_QualityScalingNotAllowed) {
   FakeVideoSendStream* stream = SetUpSimulcast(true, /*with_rtx=*/true);
   EXPECT_FALSE(stream->GetEncoderConfig().is_quality_scaling_allowed);
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index 9067545dfb..766205203a 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -286,7 +286,6 @@ struct AdaptivePtimeConfig {
 // Move some of this boiler plate code into the config structs themselves.
 AudioReceiveStreamInterface::Config BuildReceiveStreamConfig(
     uint32_t remote_ssrc,
-    uint32_t local_ssrc,
     bool use_nack,
     bool enable_non_sender_rtt,
     RtcpMode rtcp_mode,
@@ -303,7 +302,6 @@ AudioReceiveStreamInterface::Config BuildReceiveStreamConfig(
     scoped_refptr<FrameTransformerInterface> frame_transformer) {
   AudioReceiveStreamInterface::Config config;
   config.rtp.remote_ssrc = remote_ssrc;
-  config.rtp.local_ssrc = local_ssrc;
   config.rtp.nack.rtp_history_ms = use_nack ? kNackRtpHistoryMs : 0;
   config.rtp.rtcp_mode = rtcp_mode;
   if (!stream_ids.empty()) {
@@ -1664,10 +1662,6 @@ bool WebRtcVoiceSendChannel::RemoveSendStream(uint32_t ssrc) {
 
   it->second->SetSend(false);
 
-  // TODO(solenberg): If we're removing the receiver_reports_ssrc_ stream, find
-  // the first active send stream and use that instead, reassociating receive
-  // streams.
-
   delete it->second;
   send_streams_.erase(it);
   if (send_streams_.empty()) {
@@ -2447,8 +2441,8 @@ bool WebRtcVoiceReceiveChannel::AddRecvStream(const StreamParams& sp) {
 
   // Create a new channel for receiving audio data.
   auto config = BuildReceiveStreamConfig(
-      ssrc, receiver_reports_ssrc_, recv_nack_enabled_, enable_non_sender_rtt_,
-      recv_rtcp_mode_, sp.stream_ids(), recv_rtp_extensions_, transport(),
+      ssrc, recv_nack_enabled_, enable_non_sender_rtt_, recv_rtcp_mode_,
+      sp.stream_ids(), recv_rtp_extensions_, transport(),
       engine()->decoder_factory_, decoder_map_,
       engine()->audio_jitter_buffer_max_packets_,
       engine()->audio_jitter_buffer_fast_accelerate_,
@@ -2503,24 +2497,6 @@ std::optional<uint32_t> WebRtcVoiceReceiveChannel::GetUnsignaledSsrc() const {
   return unsignaled_recv_ssrcs_.back();
 }
 
-void WebRtcVoiceReceiveChannel::ChooseReceiverReportSsrc(
-    const std::set<uint32_t>& choices) {
-  RTC_DCHECK_RUN_ON(worker_thread_);
-  // Don't change SSRC if set is empty. Note that this differs from
-  // the behavior of video.
-  if (choices.empty()) {
-    return;
-  }
-  if (choices.find(receiver_reports_ssrc_) != choices.end()) {
-    return;
-  }
-  uint32_t ssrc = *(choices.begin());
-  receiver_reports_ssrc_ = ssrc;
-  for (auto& kv : recv_streams_) {
-    call_->OnLocalSsrcUpdated(kv.second->stream(), ssrc);
-  }
-}
-
 // Not implemented.
 // TODO(https://crbug.com/webrtc/12676): Implement a fix for the unsignalled
 // SSRC race that can happen when an m= section goes from receiving to not
diff --git a/media/engine/webrtc_voice_engine.h b/media/engine/webrtc_voice_engine.h
index 1034111db6..0a7b5eb2a5 100644
--- a/media/engine/webrtc_voice_engine.h
+++ b/media/engine/webrtc_voice_engine.h
@@ -356,8 +356,6 @@ class WebRtcVoiceReceiveChannel final
   void ResetUnsignaledRecvStream() override;
   std::optional<uint32_t> GetUnsignaledSsrc() const override;
 
-  void ChooseReceiverReportSsrc(const std::set<uint32_t>& choices) override;
-
   void OnDemuxerCriteriaUpdatePending() override;
   void OnDemuxerCriteriaUpdateComplete() override;
 
@@ -457,11 +455,6 @@ class WebRtcVoiceReceiveChannel final
   // Sink for latest unsignaled stream - may be set before the stream exists.
   std::unique_ptr<AudioSinkInterface> default_sink_
       RTC_GUARDED_BY(worker_thread_);
-  // Default SSRC to use for RTCP receiver reports in case of no signaled
-  // send streams. See: https://code.google.com/p/webrtc/issues/detail?id=4740
-  // and https://code.google.com/p/chromium/issues/detail?id=547661
-  uint32_t receiver_reports_ssrc_ RTC_GUARDED_BY(worker_thread_) = 0xFA17FA17u;
-
   std::string mid_ RTC_GUARDED_BY(worker_thread_);
 
   class WebRtcAudioReceiveStream;
diff --git a/media/engine/webrtc_voice_engine_unittest.cc b/media/engine/webrtc_voice_engine_unittest.cc
index a257806a0d..1361eb6afe 100644
--- a/media/engine/webrtc_voice_engine_unittest.cc
+++ b/media/engine/webrtc_voice_engine_unittest.cc
@@ -21,7 +21,6 @@
 #include <utility>
 #include <vector>
 
-#include "absl/functional/any_invocable.h"
 #include "absl/strings/match.h"
 #include "absl/strings/string_view.h"
 #include "api/array_view.h"
@@ -115,7 +114,6 @@ constexpr uint32_t kSsrc1 = 1;
 constexpr uint32_t kSsrcX = 0x99;
 constexpr uint32_t kSsrcY = 0x17;
 constexpr uint32_t kSsrcZ = 0x42;
-constexpr uint32_t kSsrcW = 0x02;
 constexpr uint32_t kSsrcs4[] = {11, 200, 30, 44};
 
 constexpr int kRtpHistoryMs = 5000;
@@ -297,11 +295,6 @@ class WebRtcVoiceEngineTestFake : public ::testing::TestWithParam<bool> {
                                                AudioOptions(), CryptoOptions());
     receive_channel_ = engine_->CreateReceiveChannel(
         env_, &call_, MediaConfig(), AudioOptions(), CryptoOptions());
-    send_channel_->SetSsrcListChangedCallback(
-        [receive_channel =
-             receive_channel_.get()](const std::set<uint32_t>& choices) {
-          receive_channel->ChooseReceiverReportSsrc(choices);
-        });
     return true;
   }
 
@@ -958,7 +951,6 @@ TEST_P(WebRtcVoiceEngineTestFake, CreateRecvStream) {
   const AudioReceiveStreamInterface::Config& config =
       GetRecvStreamConfig(kSsrcX);
   EXPECT_EQ(kSsrcX, config.rtp.remote_ssrc);
-  EXPECT_EQ(0xFA17FA17, config.rtp.local_ssrc);
   EXPECT_EQ(ReceiveImpl()->transport(), config.rtcp_send_transport);
   EXPECT_EQ("", config.sync_group);
 }
@@ -2753,7 +2745,6 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendSsrcWithMultipleStreams) {
   EXPECT_TRUE(SetupSendStream());
   EXPECT_TRUE(call_.GetAudioSendStream(kSsrcX));
   EXPECT_TRUE(AddRecvStream(kSsrcY));
-  EXPECT_EQ(kSsrcX, GetRecvStreamConfig(kSsrcY).rtp.local_ssrc);
 }
 
 // Test that the local SSRC is the same on sending and receiving channels if the
@@ -2763,7 +2754,6 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendSsrcAfterCreatingReceiveChannel) {
   EXPECT_TRUE(AddRecvStream(kSsrcY));
   EXPECT_TRUE(send_channel_->AddSendStream(StreamParams::CreateLegacy(kSsrcX)));
   EXPECT_TRUE(call_.GetAudioSendStream(kSsrcX));
-  EXPECT_EQ(kSsrcX, GetRecvStreamConfig(kSsrcY).rtp.local_ssrc);
 }
 
 // Test that we can properly receive packets.
@@ -3617,31 +3607,6 @@ TEST_P(WebRtcVoiceEngineTestFake, DeliverAudioPacket_Call) {
   EXPECT_EQ(1, s->received_packets());
 }
 
-// All receive channels should be associated with the first send channel,
-// since they do not send RTCP SR.
-TEST_P(WebRtcVoiceEngineTestFake, AssociateFirstSendChannel_SendCreatedFirst) {
-  EXPECT_TRUE(SetupSendStream());
-  EXPECT_TRUE(AddRecvStream(kSsrcY));
-  EXPECT_EQ(kSsrcX, GetRecvStreamConfig(kSsrcY).rtp.local_ssrc);
-  EXPECT_TRUE(send_channel_->AddSendStream(StreamParams::CreateLegacy(kSsrcZ)));
-  EXPECT_EQ(kSsrcX, GetRecvStreamConfig(kSsrcY).rtp.local_ssrc);
-  EXPECT_TRUE(AddRecvStream(kSsrcW));
-  EXPECT_EQ(kSsrcX, GetRecvStreamConfig(kSsrcW).rtp.local_ssrc);
-}
-
-TEST_P(WebRtcVoiceEngineTestFake, AssociateFirstSendChannel_RecvCreatedFirst) {
-  EXPECT_TRUE(SetupRecvStream());
-  EXPECT_EQ(0xFA17FA17u, GetRecvStreamConfig(kSsrcX).rtp.local_ssrc);
-  EXPECT_TRUE(send_channel_->AddSendStream(StreamParams::CreateLegacy(kSsrcY)));
-  EXPECT_EQ(kSsrcY, GetRecvStreamConfig(kSsrcX).rtp.local_ssrc);
-  EXPECT_TRUE(AddRecvStream(kSsrcZ));
-  EXPECT_EQ(kSsrcY, GetRecvStreamConfig(kSsrcZ).rtp.local_ssrc);
-  EXPECT_TRUE(send_channel_->AddSendStream(StreamParams::CreateLegacy(kSsrcW)));
-
-  EXPECT_EQ(kSsrcY, GetRecvStreamConfig(kSsrcX).rtp.local_ssrc);
-  EXPECT_EQ(kSsrcY, GetRecvStreamConfig(kSsrcZ).rtp.local_ssrc);
-}
-
 TEST_P(WebRtcVoiceEngineTestFake, SetRawAudioSink) {
   EXPECT_TRUE(SetupChannel());
   std::unique_ptr<FakeAudioSink> fake_sink_1(new FakeAudioSink());
diff --git a/modules/pacing/packet_router.cc b/modules/pacing/packet_router.cc
index eb6b524082..67c3a63fad 100644
--- a/modules/pacing/packet_router.cc
+++ b/modules/pacing/packet_router.cc
@@ -342,6 +342,18 @@ void PacketRouter::SendCombinedRtcpPacket(
   rtcp_sender->SendCombinedRtcpPacket(std::move(packets));
 }
 
+std::optional<uint32_t> PacketRouter::SsrcOfFirstSender() {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  // Prefer send modules.
+  for (RtpRtcpInterface* rtp_module : send_modules_list_) {
+    if (rtp_module->RTCP() == RtcpMode::kOff) {
+      continue;
+    }
+    return rtp_module->SSRC();
+  }
+  return std::nullopt;
+}
+
 void PacketRouter::AddRembModuleCandidate(
     RtcpFeedbackSenderInterface* candidate_module,
     bool media_sender) {
diff --git a/modules/pacing/packet_router.h b/modules/pacing/packet_router.h
index 6321f2006f..2a7336ab91 100644
--- a/modules/pacing/packet_router.h
+++ b/modules/pacing/packet_router.h
@@ -90,6 +90,10 @@ class PacketRouter : public PacingController::PacketSender {
   void SendCombinedRtcpPacket(
       std::vector<std::unique_ptr<rtcp::RtcpPacket>> packets);
 
+  // Gives the SSRC of a sender associated with this packet router that
+  // can be used for sending RTCP messages. std::nullopt if no sender exists.
+  std::optional<uint32_t> SsrcOfFirstSender();
+
  private:
   void AddRembModuleCandidate(RtcpFeedbackSenderInterface* candidate_module,
                               bool media_sender);
diff --git a/modules/rtp_rtcp/include/rtp_rtcp_defines.h b/modules/rtp_rtcp/include/rtp_rtcp_defines.h
index 72b27865d2..f6d317f928 100644
--- a/modules/rtp_rtcp/include/rtp_rtcp_defines.h
+++ b/modules/rtp_rtcp/include/rtp_rtcp_defines.h
@@ -128,6 +128,11 @@ enum RtxMode {
 
 const size_t kRtxHeaderSize = 2;
 
+// Special values for SSRC, used when sending receiver reports from
+// receive-only endpoints.
+inline constexpr uint32_t kFallbackRtcpSsrcForVideo = 1;
+inline constexpr uint32_t kFallbackRtcpSsrcForAudio = 0xFA17FA17;
+
 struct RtpState {
   uint16_t sequence_number = 0;
   uint32_t start_timestamp = 0;
diff --git a/modules/rtp_rtcp/source/rtcp_packet/dlrr.cc b/modules/rtp_rtcp/source/rtcp_packet/dlrr.cc
index 87ac475ac7..e244599d93 100644
--- a/modules/rtp_rtcp/source/rtcp_packet/dlrr.cc
+++ b/modules/rtp_rtcp/source/rtcp_packet/dlrr.cc
@@ -52,7 +52,6 @@ bool Dlrr::Parse(const uint8_t* buffer, uint16_t block_length_32bits) {
     RTC_LOG(LS_WARNING) << "Invalid size for dlrr block.";
     return false;
   }
-
   size_t blocks_count = block_length_32bits / 3;
   const uint8_t* read_at = buffer + kBlockHeaderLength;
   sub_blocks_.resize(blocks_count);
diff --git a/modules/rtp_rtcp/source/rtcp_sender.cc b/modules/rtp_rtcp/source/rtcp_sender.cc
index 589b387d11..d7a0c520c3 100644
--- a/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -126,8 +126,10 @@ class RTCPSender::RtcpContext {
 
 RTCPSender::RTCPSender(const Environment& env, Configuration config)
     : env_(env),
+      is_rtp_sender_(config.recv_ssrc_callback == nullptr),
       audio_(config.audio),
-      ssrc_(config.local_media_ssrc),
+      send_ssrc_(config.local_media_ssrc),
+      recv_ssrc_callback_(std::move(config.recv_ssrc_callback)),
       random_(env_.clock().TimeInMicroseconds()),
       method_(RtcpMode::kOff),
       transport_(config.outgoing_transport),
@@ -152,6 +154,9 @@ RTCPSender::RTCPSender(const Environment& env, Configuration config)
   RTC_CHECK(schedule_next_rtcp_send_evaluation_);
   RTC_CHECK_GT(report_interval_, TimeDelta::Zero());
   RTC_DCHECK(transport_ != nullptr);
+  // We don't want to see local_media_ssrc if we're not an RTP sender.
+  // It's likely a configuration error.
+  RTC_DCHECK(is_rtp_sender_ || send_ssrc_ == 0);
 
   builders_[kRtcpSr] = &RTCPSender::BuildSR;
   builders_[kRtcpRr] = &RTCPSender::BuildRR;
@@ -194,6 +199,8 @@ bool RTCPSender::Sending() const {
 void RTCPSender::SetSendingStatus(const FeedbackState& /* feedback_state */,
                                   bool sending) {
   MutexLock lock(&mutex_rtcp_sender_);
+  RTC_DCHECK(is_rtp_sender_ || !sending)
+      << "Only senders can have sending turned on";
   sending_ = sending;
 }
 
@@ -304,12 +311,12 @@ void RTCPSender::SetRtpClockRate(int8_t payload_type, int rtp_clock_rate_hz) {
 
 uint32_t RTCPSender::SSRC() const {
   MutexLock lock(&mutex_rtcp_sender_);
-  return ssrc_;
+  return ComputeSsrc();
 }
 
 void RTCPSender::SetSsrc(uint32_t ssrc) {
   MutexLock lock(&mutex_rtcp_sender_);
-  ssrc_ = ssrc;
+  send_ssrc_ = ssrc;
 }
 
 void RTCPSender::SetRemoteSSRC(uint32_t ssrc) {
@@ -364,7 +371,7 @@ void RTCPSender::BuildSR(const RtcpContext& ctx, PacketSender& sender) {
           rtp_rate;
 
   rtcp::SenderReport report;
-  report.SetSenderSsrc(ssrc_);
+  report.SetSenderSsrc(ComputeSsrc());
   report.SetNtp(env_.clock().ConvertTimestampToNtpTime(ctx.now_));
   report.SetRtpTimestamp(rtp_timestamp);
   report.SetPacketCount(ctx.feedback_state_.packets_sent);
@@ -378,13 +385,13 @@ void RTCPSender::BuildSDES(const RtcpContext& /* ctx */, PacketSender& sender) {
   RTC_CHECK_LT(length_cname, RTCP_CNAME_SIZE);
 
   rtcp::Sdes sdes;
-  sdes.AddCName(ssrc_, cname_);
+  sdes.AddCName(ComputeSsrc(), cname_);
   sender.AppendPacket(sdes);
 }
 
 void RTCPSender::BuildRR(const RtcpContext& ctx, PacketSender& sender) {
   rtcp::ReceiverReport report;
-  report.SetSenderSsrc(ssrc_);
+  report.SetSenderSsrc(ComputeSsrc());
   report.SetReportBlocks(CreateReportBlocks(ctx.feedback_state_));
   if (method_ == RtcpMode::kCompound || !report.report_blocks().empty()) {
     sender.AppendPacket(report);
@@ -393,7 +400,7 @@ void RTCPSender::BuildRR(const RtcpContext& ctx, PacketSender& sender) {
 
 void RTCPSender::BuildPLI(const RtcpContext& /* ctx */, PacketSender& sender) {
   rtcp::Pli pli;
-  pli.SetSenderSsrc(ssrc_);
+  pli.SetSenderSsrc(ComputeSsrc());
   pli.SetMediaSsrc(remote_ssrc_);
 
   ++packet_type_counter_.pli_packets;
@@ -404,7 +411,7 @@ void RTCPSender::BuildFIR(const RtcpContext& /* ctx */, PacketSender& sender) {
   ++sequence_number_fir_;
 
   rtcp::Fir fir;
-  fir.SetSenderSsrc(ssrc_);
+  fir.SetSenderSsrc(ComputeSsrc());
   fir.AddRequestTo(remote_ssrc_, sequence_number_fir_);
 
   ++packet_type_counter_.fir_packets;
@@ -413,7 +420,7 @@ void RTCPSender::BuildFIR(const RtcpContext& /* ctx */, PacketSender& sender) {
 
 void RTCPSender::BuildREMB(const RtcpContext& /* ctx */, PacketSender& sender) {
   rtcp::Remb remb;
-  remb.SetSenderSsrc(ssrc_);
+  remb.SetSenderSsrc(ComputeSsrc());
   remb.SetBitrateBps(remb_bitrate_);
   remb.SetSsrcs(remb_ssrcs_);
   sender.AppendPacket(remb);
@@ -452,12 +459,12 @@ void RTCPSender::BuildTMMBR(const RtcpContext& ctx, PacketSender& sender) {
     if (!tmmbr_owner) {
       // Use received bounding set as candidate set.
       // Add current tuple.
-      candidates.emplace_back(ssrc_, tmmbr_send_bps_, packet_oh_send_);
+      candidates.emplace_back(ComputeSsrc(), tmmbr_send_bps_, packet_oh_send_);
 
       // Find bounding set.
       std::vector<rtcp::TmmbItem> bounding =
           TMMBRHelp::FindBoundingSet(std::move(candidates));
-      tmmbr_owner = TMMBRHelp::IsOwner(bounding, ssrc_);
+      tmmbr_owner = TMMBRHelp::IsOwner(bounding, ComputeSsrc());
       if (!tmmbr_owner) {
         // Did not enter bounding set, no meaning to send this request.
         return;
@@ -469,7 +476,7 @@ void RTCPSender::BuildTMMBR(const RtcpContext& ctx, PacketSender& sender) {
     return;
 
   rtcp::Tmmbr tmmbr;
-  tmmbr.SetSenderSsrc(ssrc_);
+  tmmbr.SetSenderSsrc(ComputeSsrc());
   rtcp::TmmbItem request;
   request.set_ssrc(remote_ssrc_);
   request.set_bitrate_bps(tmmbr_send_bps_);
@@ -481,7 +488,7 @@ void RTCPSender::BuildTMMBR(const RtcpContext& ctx, PacketSender& sender) {
 void RTCPSender::BuildTMMBN(const RtcpContext& /* ctx */,
                             PacketSender& sender) {
   rtcp::Tmmbn tmmbn;
-  tmmbn.SetSenderSsrc(ssrc_);
+  tmmbn.SetSenderSsrc(ComputeSsrc());
   for (const rtcp::TmmbItem& tmmbr : tmmbn_to_send_) {
     if (tmmbr.bitrate_bps() > 0) {
       tmmbn.AddTmmbr(tmmbr);
@@ -492,20 +499,20 @@ void RTCPSender::BuildTMMBN(const RtcpContext& /* ctx */,
 
 void RTCPSender::BuildAPP(const RtcpContext& /* ctx */, PacketSender& sender) {
   rtcp::App app;
-  app.SetSenderSsrc(ssrc_);
+  app.SetSenderSsrc(ComputeSsrc());
   sender.AppendPacket(app);
 }
 
 void RTCPSender::BuildLossNotification(const RtcpContext& /* ctx */,
                                        PacketSender& sender) {
-  loss_notification_.SetSenderSsrc(ssrc_);
+  loss_notification_.SetSenderSsrc(ComputeSsrc());
   loss_notification_.SetMediaSsrc(remote_ssrc_);
   sender.AppendPacket(loss_notification_);
 }
 
 void RTCPSender::BuildNACK(const RtcpContext& ctx, PacketSender& sender) {
   rtcp::Nack nack;
-  nack.SetSenderSsrc(ssrc_);
+  nack.SetSenderSsrc(ComputeSsrc());
   nack.SetMediaSsrc(remote_ssrc_);
   nack.SetPacketIds(ctx.nacks_.data(), ctx.nacks_.size());
 
@@ -522,7 +529,7 @@ void RTCPSender::BuildNACK(const RtcpContext& ctx, PacketSender& sender) {
 
 void RTCPSender::BuildBYE(const RtcpContext& /* ctx */, PacketSender& sender) {
   rtcp::Bye bye;
-  bye.SetSenderSsrc(ssrc_);
+  bye.SetSenderSsrc(ComputeSsrc());
   bye.SetCsrcs(csrcs_);
   sender.AppendPacket(bye);
 }
@@ -530,7 +537,7 @@ void RTCPSender::BuildBYE(const RtcpContext& /* ctx */, PacketSender& sender) {
 void RTCPSender::BuildExtendedReports(const RtcpContext& ctx,
                                       PacketSender& sender) {
   rtcp::ExtendedReports xr;
-  xr.SetSenderSsrc(ssrc_);
+  xr.SetSenderSsrc(ComputeSsrc());
 
   if (!sending_ && xr_send_receiver_reference_time_enabled_) {
     rtcp::Rrtr rrtr;
@@ -586,6 +593,15 @@ int32_t RTCPSender::SendRTCP(const FeedbackState& feedback_state,
   return error_code;
 }
 
+uint32_t RTCPSender::ComputeSsrc() const {
+  if (is_rtp_sender_) {
+    return send_ssrc_;
+  } else {
+    RTC_DCHECK(recv_ssrc_callback_);
+    return recv_ssrc_callback_();
+  }
+}
+
 std::optional<int32_t> RTCPSender::ComputeCompoundRTCPPacket(
     const FeedbackState& feedback_state,
     RTCPPacketType packet_type,
@@ -810,7 +826,7 @@ void RTCPSender::SetVideoBitrateAllocation(
       CheckAndUpdateLayerStructure(bitrate);
   if (new_bitrate) {
     video_bitrate_allocation_ = *new_bitrate;
-    RTC_LOG(LS_INFO) << "Emitting TargetBitrate XR for SSRC " << ssrc_
+    RTC_LOG(LS_INFO) << "Emitting TargetBitrate XR for SSRC " << ComputeSsrc()
                      << " with new layers enabled/disabled: "
                      << video_bitrate_allocation_.ToString();
     SetNextRtcpSendEvaluationDuration(TimeDelta::Zero());
@@ -857,7 +873,7 @@ void RTCPSender::SendCombinedRtcpPacket(
     }
 
     max_packet_size = max_packet_size_;
-    ssrc = ssrc_;
+    ssrc = ComputeSsrc();
   }
   RTC_DCHECK_LE(max_packet_size, IP_PACKET_SIZE);
   auto callback = [&](ArrayView<const uint8_t> packet) {
diff --git a/modules/rtp_rtcp/source/rtcp_sender.h b/modules/rtp_rtcp/source/rtcp_sender.h
index b2ad982f28..85d25f5fd7 100644
--- a/modules/rtp_rtcp/source/rtcp_sender.h
+++ b/modules/rtp_rtcp/source/rtcp_sender.h
@@ -53,9 +53,12 @@ class RTCPSender final {
     // True for a audio version of the RTP/RTCP module object false will create
     // a video version.
     bool audio = false;
-    // SSRCs for media and retransmission, respectively.
-    // FlexFec SSRC is fetched from `flexfec_sender`.
+    // SSRC for sending media.
     uint32_t local_media_ssrc = 0;
+    // Function for fetching SSRC for sending RTCP reports when this object
+    // belongs to a ModuleRtpRtcp2 that is not used for sending RTP.
+    absl::AnyInvocable<uint32_t() const> recv_ssrc_callback;
+    // FlexFec SSRC is fetched from `flexfec_sender`.
 
     // Transport object that will be called when packets are ready to be sent
     // out on the network.
@@ -175,6 +178,8 @@ class RTCPSender final {
   class RtcpContext;
   class PacketSender;
 
+  uint32_t ComputeSsrc() const RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
+
   std::optional<int32_t> ComputeCompoundRTCPPacket(
       const FeedbackState& feedback_state,
       RTCPPacketType packet_type,
@@ -224,12 +229,19 @@ class RTCPSender final {
       RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
 
   const Environment env_;
+  const bool is_rtp_sender_;
   const bool audio_;
   // TODO(bugs.webrtc.org/11581): `mutex_rtcp_sender_` shouldn't be required if
   // we consistently run network related operations on the network thread.
   // This is currently not possible due to callbacks from the process thread in
   // ModuleRtpRtcpImpl2.
-  uint32_t ssrc_ RTC_GUARDED_BY(mutex_rtcp_sender_);
+  // The SSRC used for sending when is_rtp_sender_ is true
+  // and sending_ is true.
+  uint32_t send_ssrc_ RTC_GUARDED_BY(mutex_rtcp_sender_);
+  // The function used for getting the right SSRC to send from
+  // when the RTCPSender is used with a ModuleRtpRtcp that is not
+  // configured for sending RTP (is_rtp_sender_ is false).
+  absl::AnyInvocable<uint32_t() const> recv_ssrc_callback_;
   Random random_ RTC_GUARDED_BY(mutex_rtcp_sender_);
   RtcpMode method_ RTC_GUARDED_BY(mutex_rtcp_sender_);
 
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
index e552033d0a..dd4610ac55 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
@@ -76,14 +76,22 @@ ModuleRtpRtcpImpl2::RtpSenderContext::RtpSenderContext(
           &packet_history,
           config.paced_sender ? config.paced_sender : &non_paced_sender) {}
 
-ModuleRtpRtcpImpl2::ModuleRtpRtcpImpl2(const Environment& env,
-                                       const Configuration& configuration)
+ModuleRtpRtcpImpl2::ModuleRtpRtcpImpl2(
+    const Environment& env,
+    const Configuration& configuration,
+    absl::AnyInvocable<uint32_t(void) const> recv_ssrc_callback)
     : env_(env),
       worker_queue_(TaskQueueBase::Current()),
+      recv_ssrc_callback_(std::move(recv_ssrc_callback)),
       rtcp_sender_(
           env_,
           {.audio = configuration.audio,
            .local_media_ssrc = configuration.local_media_ssrc,
+           .recv_ssrc_callback =
+               (recv_ssrc_callback_ != nullptr)
+                   ? absl::AnyInvocable<uint32_t() const>(
+                         [this] { return RtcpSenderSourceSsrc(); })
+                   : nullptr,
            .outgoing_transport = configuration.outgoing_transport,
            .non_sender_rtt_measurement =
                configuration.non_sender_rtt_measurement,
@@ -832,4 +840,11 @@ void ModuleRtpRtcpImpl2::ScheduleMaybeSendRtcpAtOrAfterTimestamp(
       duration.RoundUpTo(TimeDelta::Millis(1)));
 }
 
+uint32_t ModuleRtpRtcpImpl2::RtcpSenderSourceSsrc() {
+  uint32_t ssrc = recv_ssrc_callback_();
+  // Inform the RtcpReceiver that this is now the SSRC to listen for
+  rtcp_receiver_.set_local_media_ssrc(ssrc);
+  return ssrc;
+}
+
 }  // namespace webrtc
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl2.h b/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
index b787fd5a39..86a40da740 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
@@ -16,8 +16,10 @@
 
 #include <memory>
 #include <optional>
+#include <utility>
 #include <vector>
 
+#include "absl/functional/any_invocable.h"
 #include "absl/memory/memory.h"
 #include "absl/strings/string_view.h"
 #include "api/array_view.h"
@@ -62,13 +64,17 @@ class ModuleRtpRtcpImpl2 final : public RtpRtcpInterface,
       const Environment& env,
       const RtpRtcpInterface::Configuration& configuration) {
     RTC_DCHECK(!configuration.receiver_only);
-    return absl::WrapUnique(new ModuleRtpRtcpImpl2(env, configuration));
+    return absl::WrapUnique(
+        new ModuleRtpRtcpImpl2(env, configuration, nullptr));
   }
   static std::unique_ptr<ModuleRtpRtcpImpl2> CreateReceiveModule(
       const Environment& env,
-      const RtpRtcpInterface::Configuration& configuration) {
+      const RtpRtcpInterface::Configuration& configuration,
+      absl::AnyInvocable<uint32_t() const> recv_ssrc_callback) {
     RTC_DCHECK(configuration.receiver_only);
-    return absl::WrapUnique(new ModuleRtpRtcpImpl2(env, configuration));
+    RTC_DCHECK(recv_ssrc_callback);
+    return absl::WrapUnique(new ModuleRtpRtcpImpl2(
+        env, configuration, std::move(recv_ssrc_callback)));
   }
   ~ModuleRtpRtcpImpl2() override;
 
@@ -286,7 +292,8 @@ class ModuleRtpRtcpImpl2 final : public RtpRtcpInterface,
 
   // Private constructor, to enforce sender/receiver separation.
   ModuleRtpRtcpImpl2(const Environment& env,
-                     const RtpRtcpInterface::Configuration& configuration);
+                     const RtpRtcpInterface::Configuration& configuration,
+                     absl::AnyInvocable<uint32_t() const> recv_ssrc_callback);
 
   void set_rtt_ms(int64_t rtt_ms);
   int64_t rtt_ms() const;
@@ -317,10 +324,20 @@ class ModuleRtpRtcpImpl2 final : public RtpRtcpInterface,
   void ScheduleMaybeSendRtcpAtOrAfterTimestamp(Timestamp execution_time,
                                                TimeDelta duration);
 
+  // This function is called by the RtcpSender when the module is configured
+  // for not sending RTP to query for the local SSRC, which may change over
+  // time. As a side effect, it informs the RtcpReceiver of the currently
+  // used SSRC.
+  uint32_t RtcpSenderSourceSsrc();
+
   const Environment env_;
   TaskQueueBase* const worker_queue_;
   RTC_NO_UNIQUE_ADDRESS SequenceChecker rtcp_thread_checker_;
 
+  // The function for getting the right SSRC for sending RTCP reports
+  // Must outlive rtcp_sender_, so placed before it.
+  absl::AnyInvocable<uint32_t() const> recv_ssrc_callback_;
+
   std::unique_ptr<RtpSenderContext> rtp_sender_;
   RTCPSender rtcp_sender_;
   RTCPReceiver rtcp_receiver_;
diff --git a/pc/rtp_transceiver.cc b/pc/rtp_transceiver.cc
index 608c1888bb..77d99a14e4 100644
--- a/pc/rtp_transceiver.cc
+++ b/pc/rtp_transceiver.cc
@@ -15,7 +15,6 @@
 #include <iterator>
 #include <memory>
 #include <optional>
-#include <set>
 #include <string>
 #include <utility>
 #include <vector>
@@ -477,13 +476,6 @@ RTCError RtpTransceiver::CreateChannel(
       media_receive_channel = std::move(channels.second);
       SetMediaChannels(media_send_channel.get(), media_receive_channel.get());
     }
-    // Note that this is safe because both sending and
-    // receiving channels will be deleted at the same time.
-    media_send_channel->SetSsrcListChangedCallback(
-        [receive_channel =
-             media_receive_channel.get()](const std::set<uint32_t>& choices) {
-          receive_channel->ChooseReceiverReportSsrc(choices);
-        });
 
     if (media_type() == MediaType::AUDIO) {
       new_channel =
diff --git a/pc/test/mock_voice_media_receive_channel_interface.h b/pc/test/mock_voice_media_receive_channel_interface.h
index cb7334f2b9..b38405e258 100644
--- a/pc/test/mock_voice_media_receive_channel_interface.h
+++ b/pc/test/mock_voice_media_receive_channel_interface.h
@@ -13,7 +13,6 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
-#include <set>
 #include <type_traits>
 #include <vector>
 
@@ -109,10 +108,6 @@ class MockVoiceMediaReceiveChannelInterface
               GetUnsignaledSsrc,
               (),
               (const, override));
-  MOCK_METHOD(void,
-              ChooseReceiverReportSsrc,
-              (const std::set<uint32_t>& choices),
-              (override));
   MOCK_METHOD(void, OnDemuxerCriteriaUpdatePending, (), (override));
   MOCK_METHOD(void, OnDemuxerCriteriaUpdateComplete, (), (override));
   MOCK_METHOD(void,
diff --git a/rtc_tools/video_replay.cc b/rtc_tools/video_replay.cc
index dbf01ec36a..ceb13849df 100644
--- a/rtc_tools/video_replay.cc
+++ b/rtc_tools/video_replay.cc
@@ -229,8 +229,6 @@ std::string ReadFileToString(const std::string& file_path) {
 namespace webrtc {
 namespace {
 
-const uint32_t kReceiverLocalSsrc = 0x123456;
-
 class NullRenderer : public VideoSinkInterface<VideoFrame> {
  public:
   void OnFrame(const VideoFrame& frame) override {}
@@ -419,7 +417,6 @@ std::unique_ptr<StreamState> ConfigureFromFlags(
   VideoReceiveStreamInterface::Config receive_config(
       &(stream_state->transport));
   receive_config.rtp.remote_ssrc = absl::GetFlag(FLAGS_ssrc);
-  receive_config.rtp.local_ssrc = kReceiverLocalSsrc;
   receive_config.rtp.rtx_ssrc = absl::GetFlag(FLAGS_ssrc_rtx);
   receive_config.rtp.rtx_associated_payload_types[absl::GetFlag(
       FLAGS_media_payload_type_rtx)] = absl::GetFlag(FLAGS_media_payload_type);
diff --git a/test/BUILD.gn b/test/BUILD.gn
index 13e8b6da96..cf3493505f 100644
--- a/test/BUILD.gn
+++ b/test/BUILD.gn
@@ -1113,7 +1113,10 @@ rtc_library("run_loop") {
 rtc_library("video_test_constants") {
   testonly = true
   sources = [ "video_test_constants.h" ]
-  deps = [ "../api/units:time_delta" ]
+  deps = [
+    "../api/units:time_delta",
+    "../modules/rtp_rtcp:rtp_rtcp_format",
+  ]
 }
 
 rtc_library("mock_transport") {
diff --git a/test/DEPS b/test/DEPS
index df7b9e6fea..59db2cbc33 100644
--- a/test/DEPS
+++ b/test/DEPS
@@ -85,6 +85,9 @@ specific_include_rules = {
      "+p2p/test/turn_server.h",
      "+p2p/base/port_interface.h",
   ],
+  "video_test_constants\\.h": [
+     "+modules/rtp_rtcp/include/rtp_rtcp_defines.h",
+  ],
   "wait_until\\.cc": [
     "+absl/functional/overload.h",
   ],
diff --git a/test/call_config_utils.cc b/test/call_config_utils.cc
index c6349bb622..5ef65eaa86 100644
--- a/test/call_config_utils.cc
+++ b/test/call_config_utils.cc
@@ -45,7 +45,6 @@ VideoReceiveStreamInterface::Config ParseVideoReceiveStreamJsonConfig(
   }
   receive_config.render_delay_ms = json["render_delay_ms"].asInt64();
   receive_config.rtp.remote_ssrc = json["rtp"]["remote_ssrc"].asInt64();
-  receive_config.rtp.local_ssrc = json["rtp"]["local_ssrc"].asInt64();
   receive_config.rtp.rtcp_mode =
       json["rtp"]["rtcp_mode"].asString() == "RtcpMode::kCompound"
           ? RtcpMode::kCompound
@@ -89,7 +88,6 @@ Json::Value GenerateVideoReceiveStreamJsonConfig(
 
   Json::Value rtp_json;
   rtp_json["remote_ssrc"] = config.rtp.remote_ssrc;
-  rtp_json["local_ssrc"] = config.rtp.local_ssrc;
   rtp_json["rtcp_mode"] = config.rtp.rtcp_mode == RtcpMode::kCompound
                               ? "RtcpMode::kCompound"
                               : "RtcpMode::kReducedSize";
diff --git a/test/call_config_utils_unittest.cc b/test/call_config_utils_unittest.cc
index 8c9598e41b..dd3534afdf 100644
--- a/test/call_config_utils_unittest.cc
+++ b/test/call_config_utils_unittest.cc
@@ -27,7 +27,6 @@ TEST(CallConfigUtils, MarshalUnmarshalProcessSameObject) {
   recv_config.decoders.push_back(decoder);
   recv_config.render_delay_ms = 10;
   recv_config.rtp.remote_ssrc = 100;
-  recv_config.rtp.local_ssrc = 101;
   recv_config.rtp.rtcp_mode = RtcpMode::kCompound;
   recv_config.rtp.lntf.enabled = false;
   recv_config.rtp.nack.rtp_history_ms = 150;
@@ -47,7 +46,6 @@ TEST(CallConfigUtils, MarshalUnmarshalProcessSameObject) {
             unmarshaled_config.decoders[0].video_format.parameters);
   EXPECT_EQ(recv_config.render_delay_ms, unmarshaled_config.render_delay_ms);
   EXPECT_EQ(recv_config.rtp.remote_ssrc, unmarshaled_config.rtp.remote_ssrc);
-  EXPECT_EQ(recv_config.rtp.local_ssrc, unmarshaled_config.rtp.local_ssrc);
   EXPECT_EQ(recv_config.rtp.rtcp_mode, unmarshaled_config.rtp.rtcp_mode);
   EXPECT_EQ(recv_config.rtp.lntf.enabled, unmarshaled_config.rtp.lntf.enabled);
   EXPECT_EQ(recv_config.rtp.nack.rtp_history_ms,
diff --git a/test/call_test.cc b/test/call_test.cc
index 994756c9ff..c25a6c828c 100644
--- a/test/call_test.cc
+++ b/test/call_test.cc
@@ -438,7 +438,6 @@ void CallTest::AddMatchingVideoReceiveConfigs(
     int rtp_history_ms) {
   RTC_DCHECK(!video_send_config.rtp.ssrcs.empty());
   VideoReceiveStreamInterface::Config default_config(rtcp_send_transport);
-  default_config.rtp.local_ssrc = VideoTestConstants::kReceiverLocalVideoSsrc;
   default_config.rtp.nack.rtp_history_ms = rtp_history_ms;
   // Enable RTT calculation so NTP time estimator will work.
   default_config.rtp.rtcp_xr.receiver_reference_time_report =
@@ -497,7 +496,6 @@ AudioReceiveStreamInterface::Config CallTest::CreateMatchingAudioConfig(
     Transport* transport,
     std::string sync_group) {
   AudioReceiveStreamInterface::Config audio_config;
-  audio_config.rtp.local_ssrc = VideoTestConstants::kReceiverLocalAudioSsrc;
   audio_config.rtcp_send_transport = transport;
   audio_config.rtp.remote_ssrc = send_config.rtp.ssrc;
   audio_config.decoder_factory = audio_decoder_factory;
@@ -514,7 +512,6 @@ void CallTest::CreateMatchingFecConfig(
   config.payload_type = send_config.rtp.flexfec.payload_type;
   config.remote_ssrc = send_config.rtp.flexfec.ssrc;
   config.protected_media_ssrcs = send_config.rtp.flexfec.protected_media_ssrcs;
-  config.local_ssrc = VideoTestConstants::kReceiverLocalVideoSsrc;
   if (!video_receive_configs_.empty()) {
     video_receive_configs_[0].rtp.protected_by_flexfec = true;
     video_receive_configs_[0].rtp.packet_sink_ = this;
diff --git a/test/fuzzers/vp8_replay_fuzzer.cc b/test/fuzzers/vp8_replay_fuzzer.cc
index 01a0ff9e8e..3d872962dc 100644
--- a/test/fuzzers/vp8_replay_fuzzer.cc
+++ b/test/fuzzers/vp8_replay_fuzzer.cc
@@ -30,7 +30,6 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
   vp8_decoder.payload_type = 125;
   vp8_config.decoders.push_back(std::move(vp8_decoder));
 
-  vp8_config.rtp.local_ssrc = 7731;
   vp8_config.rtp.remote_ssrc = 1337;
   vp8_config.rtp.rtx_ssrc = 100;
   vp8_config.rtp.nack.rtp_history_ms = 1000;
diff --git a/test/fuzzers/vp9_replay_fuzzer.cc b/test/fuzzers/vp9_replay_fuzzer.cc
index e55aa952f8..1bde2b5148 100644
--- a/test/fuzzers/vp9_replay_fuzzer.cc
+++ b/test/fuzzers/vp9_replay_fuzzer.cc
@@ -30,7 +30,6 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
   vp9_decoder.payload_type = 124;
   vp9_config.decoders.push_back(std::move(vp9_decoder));
 
-  vp9_config.rtp.local_ssrc = 7731;
   vp9_config.rtp.remote_ssrc = 1337;
   vp9_config.rtp.rtx_ssrc = 100;
   vp9_config.rtp.nack.rtp_history_ms = 1000;
diff --git a/test/scenario/audio_stream.cc b/test/scenario/audio_stream.cc
index 6b0922e5b6..78d6b0641c 100644
--- a/test/scenario/audio_stream.cc
+++ b/test/scenario/audio_stream.cc
@@ -202,7 +202,6 @@ ReceiveAudioStream::ReceiveAudioStream(
     Transport* feedback_transport)
     : receiver_(receiver), config_(config) {
   AudioReceiveStreamInterface::Config recv_config;
-  recv_config.rtp.local_ssrc = receiver_->GetNextAudioLocalSsrc();
   recv_config.rtcp_send_transport = feedback_transport;
   recv_config.rtp.remote_ssrc = send_stream->ssrc_;
   receiver->ssrc_media_types_[recv_config.rtp.remote_ssrc] = MediaType::AUDIO;
diff --git a/test/scenario/video_stream.cc b/test/scenario/video_stream.cc
index ccbb4b87d8..8cffa689c4 100644
--- a/test/scenario/video_stream.cc
+++ b/test/scenario/video_stream.cc
@@ -319,11 +319,9 @@ VideoReceiveStreamInterface::Config CreateVideoReceiveStreamConfig(
     VideoDecoderFactory* decoder_factory,
     VideoReceiveStreamInterface::Decoder decoder,
     VideoSinkInterface<VideoFrame>* renderer,
-    uint32_t local_ssrc,
     uint32_t ssrc,
     uint32_t rtx_ssrc) {
   VideoReceiveStreamInterface::Config recv(feedback_transport);
-  recv.rtp.local_ssrc = local_ssrc;
 
   RTC_DCHECK(!config.stream.use_rtx ||
              config.stream.nack_history_time > TimeDelta::Zero());
@@ -571,15 +569,13 @@ ReceiveVideoStream::ReceiveVideoStream(CallClient* receiver,
     }
     auto recv_config = CreateVideoReceiveStreamConfig(
         config, feedback_transport, decoder_factory_.get(), decoder, renderer,
-        receiver_->GetNextVideoLocalSsrc(), send_stream->ssrcs_[i],
-        send_stream->rtx_ssrcs_[i]);
+        send_stream->ssrcs_[i], send_stream->rtx_ssrcs_[i]);
     if (config.stream.use_flexfec) {
       RTC_DCHECK(num_streams == 1);
       FlexfecReceiveStream::Config flexfec(feedback_transport);
       flexfec.payload_type = VideoTestConstants::kFlexfecPayloadType;
       flexfec.remote_ssrc = VideoTestConstants::kFlexfecSendSsrc;
       flexfec.protected_media_ssrcs = send_stream->rtx_ssrcs_;
-      flexfec.local_ssrc = recv_config.rtp.local_ssrc;
       receiver_->ssrc_media_types_[flexfec.remote_ssrc] = MediaType::VIDEO;
 
       receiver_->SendTask([this, &flexfec] {
diff --git a/test/video_test_constants.h b/test/video_test_constants.h
index f3bf34f229..186a7bac9f 100644
--- a/test/video_test_constants.h
+++ b/test/video_test_constants.h
@@ -14,6 +14,7 @@
 #include <cstdint>
 
 #include "api/units/time_delta.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 namespace test {
@@ -48,8 +49,10 @@ class VideoTestConstants {
       0xC0FFED, 0xC0FFEE, 0xC0FFEF, 0xC0FFF0, 0xC0FFF1, 0xC0FFF2};
   static constexpr uint32_t kAudioSendSsrc = 0xDEADBEEF;
   static constexpr uint32_t kFlexfecSendSsrc = 0xBADBEEF;
-  static constexpr uint32_t kReceiverLocalVideoSsrc = 0x123456;
-  static constexpr uint32_t kReceiverLocalAudioSsrc = 0x1234567;
+  // When there is no send stream, the SSRC of receiver RTCP reports
+  // will always send from these two values.
+  static constexpr uint32_t kReceiverLocalVideoSsrc = kFallbackRtcpSsrcForVideo;
+  static constexpr uint32_t kReceiverLocalAudioSsrc = kFallbackRtcpSsrcForAudio;
   static constexpr int kNackRtpHistoryMs = 1000;
 
  private:
diff --git a/video/end_to_end_tests/bandwidth_tests.cc b/video/end_to_end_tests/bandwidth_tests.cc
index c02d6f8d9e..e4d32432e4 100644
--- a/video/end_to_end_tests/bandwidth_tests.cc
+++ b/video/end_to_end_tests/bandwidth_tests.cc
@@ -235,7 +235,6 @@ TEST_F(BandwidthEndToEndTest, RembWithSendSideBwe) {
       encoder_config->max_bitrate_bps = 2000000;
 
       ASSERT_EQ(1u, receive_configs->size());
-      remb_sender_local_ssrc_ = (*receive_configs)[0].rtp.local_ssrc;
       remb_sender_remote_ssrc_ = (*receive_configs)[0].rtp.remote_ssrc;
     }
 
@@ -254,9 +253,9 @@ TEST_F(BandwidthEndToEndTest, RembWithSendSideBwe) {
       config.receiver_only = true;
       config.outgoing_transport = to_sender;
       config.retransmission_rate_limiter = &retransmission_rate_limiter_;
-      config.local_media_ssrc = remb_sender_local_ssrc_;
 
-      rtp_rtcp_ = ModuleRtpRtcpImpl2::CreateReceiveModule(env_, config);
+      rtp_rtcp_ = ModuleRtpRtcpImpl2::CreateReceiveModule(
+          env_, config, [&] { return kFallbackRtcpSsrcForVideo; });
       rtp_rtcp_->SetRemoteSSRC(remb_sender_remote_ssrc_);
       rtp_rtcp_->SetRTCPStatus(RtcpMode::kReducedSize);
     }
@@ -309,7 +308,6 @@ TEST_F(BandwidthEndToEndTest, RembWithSendSideBwe) {
     Call* sender_call_;
     const Environment env_;
     uint32_t sender_ssrc_;
-    uint32_t remb_sender_local_ssrc_ = 0;
     uint32_t remb_sender_remote_ssrc_ = 0;
     int remb_bitrate_bps_;
     std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp_;
diff --git a/video/end_to_end_tests/fec_tests.cc b/video/end_to_end_tests/fec_tests.cc
index 91546a294b..dc29416813 100644
--- a/video/end_to_end_tests/fec_tests.cc
+++ b/video/end_to_end_tests/fec_tests.cc
@@ -26,13 +26,12 @@
 #include "api/video/video_sink_interface.h"
 #include "api/video_codecs/sdp_video_format.h"
 #include "api/video_codecs/video_encoder.h"
-#include "call/flexfec_receive_stream.h"
 #include "call/video_receive_stream.h"
 #include "call/video_send_stream.h"
 #include "media/engine/internal_decoder_factory.h"
 #include "modules/include/module_common_types_public.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/byte_io.h"
-#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
 #include "modules/rtp_rtcp/source/rtp_packet.h"
 #include "modules/video_coding/codecs/vp8/include/vp8.h"
 #include "rtc_base/random.h"
@@ -191,10 +190,7 @@ TEST_F(FecEndToEndTest, ReceivesUlpfec) {
 class FlexfecRenderObserver : public test::EndToEndTest,
                               public VideoSinkInterface<VideoFrame> {
  public:
-  static constexpr uint32_t kVideoLocalSsrc = 123;
-  static constexpr uint32_t kFlexfecLocalSsrc = 456;
-
-  explicit FlexfecRenderObserver(bool enable_nack, bool expect_flexfec_rtcp)
+  FlexfecRenderObserver(bool enable_nack, bool expect_flexfec_rtcp)
       : test::EndToEndTest(test::VideoTestConstants::kLongTimeout),
         enable_nack_(enable_nack),
         expect_flexfec_rtcp_(expect_flexfec_rtcp),
@@ -282,17 +278,14 @@ class FlexfecRenderObserver : public test::EndToEndTest,
     test::RtcpPacketParser parser;
 
     parser.Parse(data);
-    if (parser.sender_ssrc() == kFlexfecLocalSsrc) {
-      EXPECT_EQ(1, parser.receiver_report()->num_packets());
-      const std::vector<rtcp::ReportBlock>& report_blocks =
-          parser.receiver_report()->report_blocks();
-      if (!report_blocks.empty()) {
-        EXPECT_EQ(1U, report_blocks.size());
-        EXPECT_EQ(test::VideoTestConstants::kFlexfecSendSsrc,
-                  report_blocks[0].source_ssrc());
-        MutexLock lock(&mutex_);
-        received_flexfec_rtcp_ = true;
-      }
+    // RTCP for flexfec is distinguished from RTCP for video by looking
+    // at the report_blocks()[0].source_ssrc.
+    if (parser.receiver_report()->num_packets() == 1 &&
+        parser.receiver_report()->report_blocks().size() == 1 &&
+        parser.receiver_report()->report_blocks()[0].source_ssrc() ==
+            test::VideoTestConstants::kFlexfecSendSsrc) {
+      MutexLock lock(&mutex_);
+      received_flexfec_rtcp_ = true;
     }
 
     return SEND_PACKET;
@@ -324,7 +317,6 @@ class FlexfecRenderObserver : public test::EndToEndTest,
       VideoSendStream::Config* send_config,
       std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
       VideoEncoderConfig* encoder_config) override {
-    (*receive_configs)[0].rtp.local_ssrc = kVideoLocalSsrc;
     (*receive_configs)[0].renderer = this;
 
     if (enable_nack_) {
@@ -350,11 +342,6 @@ class FlexfecRenderObserver : public test::EndToEndTest,
     frame_generator_capturer->SetFakeRotation(kVideoRotation_90);
   }
 
-  void ModifyFlexfecConfigs(
-      std::vector<FlexfecReceiveStream::Config>* receive_configs) override {
-    (*receive_configs)[0].local_ssrc = kFlexfecLocalSsrc;
-  }
-
   void PerformTest() override {
     EXPECT_TRUE(Wait())
         << "Timed out waiting for dropped frames to be rendered.";
diff --git a/video/end_to_end_tests/multi_stream_tester.cc b/video/end_to_end_tests/multi_stream_tester.cc
index c718f1bf4f..2c68ad78ce 100644
--- a/video/end_to_end_tests/multi_stream_tester.cc
+++ b/video/end_to_end_tests/multi_stream_tester.cc
@@ -41,7 +41,6 @@
 #include "test/encoder_settings.h"
 #include "test/frame_generator_capturer.h"
 #include "test/network/simulated_network.h"
-#include "test/video_test_constants.h"
 #include "video/config/video_encoder_config.h"
 
 namespace webrtc {
@@ -115,8 +114,6 @@ void MultiStreamTester::RunTest() {
       VideoReceiveStreamInterface::Config receive_config(
           receiver_transport.get());
       receive_config.rtp.remote_ssrc = ssrc;
-      receive_config.rtp.local_ssrc =
-          test::VideoTestConstants::kReceiverLocalVideoSsrc;
       receive_config.decoder_factory = &decoder_factory;
       VideoReceiveStreamInterface::Decoder decoder =
           test::CreateMatchingDecoder(send_config);
diff --git a/video/end_to_end_tests/retransmission_tests.cc b/video/end_to_end_tests/retransmission_tests.cc
index 836a4a4598..3da96e6c49 100644
--- a/video/end_to_end_tests/retransmission_tests.cc
+++ b/video/end_to_end_tests/retransmission_tests.cc
@@ -198,7 +198,6 @@ TEST_F(RetransmissionEndToEndTest, ReceivesNackAndRetransmitsAudio) {
                                 receive_configs) override {
       (*receive_configs)[0].rtp.nack.rtp_history_ms =
           test::VideoTestConstants::kNackRtpHistoryMs;
-      local_ssrc_ = (*receive_configs)[0].rtp.local_ssrc;
       remote_ssrc_ = (*receive_configs)[0].rtp.remote_ssrc;
       receive_transport_ = (*receive_configs)[0].rtcp_send_transport;
     }
diff --git a/video/end_to_end_tests/rtp_rtcp_tests.cc b/video/end_to_end_tests/rtp_rtcp_tests.cc
index 89d1a24549..3407ba982a 100644
--- a/video/end_to_end_tests/rtp_rtcp_tests.cc
+++ b/video/end_to_end_tests/rtp_rtcp_tests.cc
@@ -545,8 +545,6 @@ TEST_F(RtpRtcpEndToEndTest, DISABLED_TestFlexfecRtpStatePreservation) {
     flexfec_receive_config.remote_ssrc = GetVideoSendConfig()->rtp.flexfec.ssrc;
     flexfec_receive_config.protected_media_ssrcs =
         GetVideoSendConfig()->rtp.flexfec.protected_media_ssrcs;
-    flexfec_receive_config.local_ssrc =
-        test::VideoTestConstants::kReceiverLocalVideoSsrc;
     flexfec_receive_configs_.push_back(flexfec_receive_config);
 
     CreateFlexfecStreams();
diff --git a/video/rtp_video_stream_receiver2.cc b/video/rtp_video_stream_receiver2.cc
index 8e74f37870..753325675c 100644
--- a/video/rtp_video_stream_receiver2.cc
+++ b/video/rtp_video_stream_receiver2.cc
@@ -126,8 +126,8 @@ std::unique_ptr<ModuleRtpRtcpImpl2> CreateRtpRtcpModule(
     RtcpRttStats* rtt_stats,
     RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer,
     RtcpCnameCallback* rtcp_cname_callback,
-    bool non_sender_rtt_measurement,
-    uint32_t local_ssrc) {
+    PacketRouter* packet_router,
+    bool non_sender_rtt_measurement) {
   RtpRtcpInterface::Configuration configuration;
   configuration.audio = false;
   configuration.receiver_only = true;
@@ -137,10 +137,17 @@ std::unique_ptr<ModuleRtpRtcpImpl2> CreateRtpRtcpModule(
   configuration.rtcp_packet_type_counter_observer =
       rtcp_packet_type_counter_observer;
   configuration.rtcp_cname_callback = rtcp_cname_callback;
-  configuration.local_media_ssrc = local_ssrc;
   configuration.non_sender_rtt_measurement = non_sender_rtt_measurement;
 
-  auto rtp_rtcp = ModuleRtpRtcpImpl2::CreateReceiveModule(env, configuration);
+  auto rtp_rtcp = ModuleRtpRtcpImpl2::CreateReceiveModule(
+      env, configuration, [packet_router]() {
+        if (packet_router) {
+          return packet_router->SsrcOfFirstSender().value_or(
+              kFallbackRtcpSsrcForVideo);
+        } else {  // This happens at least in some test configurations.
+          return kFallbackRtcpSsrcForVideo;
+        }
+      });
   rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
 
   return rtp_rtcp;
@@ -310,8 +317,8 @@ RtpVideoStreamReceiver2::RtpVideoStreamReceiver2(
           rtt_stats,
           rtcp_packet_type_counter_observer,
           rtcp_cname_callback,
-          config_.rtp.rtcp_xr.receiver_reference_time_report,
-          config_.rtp.local_ssrc)),
+          packet_router,
+          config_.rtp.rtcp_xr.receiver_reference_time_report)),
       nack_periodic_processor_(nack_periodic_processor),
       complete_frame_callback_(complete_frame_callback),
       keyframe_request_method_(config_.rtp.keyframe_method),
@@ -341,9 +348,6 @@ RtpVideoStreamReceiver2::RtpVideoStreamReceiver2(
   RTC_DCHECK(config_.rtp.rtcp_mode != RtcpMode::kOff)
       << "A stream should not be configured with RTCP disabled. This value is "
          "reserved for internal usage.";
-  // TODO(pbos): What's an appropriate local_ssrc for receive-only streams?
-  RTC_DCHECK(config_.rtp.local_ssrc != 0);
-  RTC_DCHECK(config_.rtp.remote_ssrc != config_.rtp.local_ssrc);
 
   rtp_rtcp_->SetRTCPStatus(config_.rtp.rtcp_mode);
   rtp_rtcp_->SetRemoteSSRC(config_.rtp.remote_ssrc);
@@ -1041,11 +1045,6 @@ void RtpVideoStreamReceiver2::UpdateRtt(int64_t max_rtt_ms) {
     nack_module_->UpdateRtt(max_rtt_ms);
 }
 
-void RtpVideoStreamReceiver2::OnLocalSsrcChange(uint32_t local_ssrc) {
-  RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
-  rtp_rtcp_->SetLocalSsrc(local_ssrc);
-}
-
 void RtpVideoStreamReceiver2::SetRtcpMode(RtcpMode mode) {
   RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
   rtp_rtcp_->SetRTCPStatus(mode);
diff --git a/video/rtp_video_stream_receiver2_unittest.cc b/video/rtp_video_stream_receiver2_unittest.cc
index 3363609c1c..962ad0fb15 100644
--- a/video/rtp_video_stream_receiver2_unittest.cc
+++ b/video/rtp_video_stream_receiver2_unittest.cc
@@ -263,7 +263,6 @@ class RtpVideoStreamReceiver2Test : public ::testing::Test,
   VideoReceiveStreamInterface::Config CreateConfig() {
     VideoReceiveStreamInterface::Config config(nullptr);
     config.rtp.remote_ssrc = 1111;
-    config.rtp.local_ssrc = 2222;
     config.rtp.red_payload_type = kRedPayloadType;
     config.rtp.packet_sink_ = this;
     return config;
diff --git a/video/timing/simulator/assembler.cc b/video/timing/simulator/assembler.cc
index bd5f8714fc..af881d0944 100644
--- a/video/timing/simulator/assembler.cc
+++ b/video/timing/simulator/assembler.cc
@@ -36,9 +36,6 @@ VideoReceiveStreamInterface::Config CreateVideoReceiveStreamConfig(
   // From `kNackHistoryMs` in webrtc_video_engine.cc. This enables creating the
   // `NackRequester`.
   config.rtp.nack.rtp_history_ms = 1000;
-  // The value of `local_ssrc` is not really used, but we need to set it to
-  // _something_ due to an RTC_DCHECK in rtp_video_stream_receiver2.cc.
-  config.rtp.local_ssrc = ssrc + 1;
   return config;
 }
 
diff --git a/video/video_receive_stream2.cc b/video/video_receive_stream2.cc
index 7d29d3d5d2..478b1af212 100644
--- a/video/video_receive_stream2.cc
+++ b/video/video_receive_stream2.cc
@@ -359,16 +359,6 @@ void VideoReceiveStream2::SetSync(Syncable* audio_syncable) {
   rtp_stream_sync_.ConfigureSync(audio_syncable);
 }
 
-void VideoReceiveStream2::SetLocalSsrc(uint32_t local_ssrc) {
-  RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
-  if (config_.rtp.local_ssrc == local_ssrc)
-    return;
-
-  // TODO(tommi): Make sure we don't rely on local_ssrc via the config struct.
-  const_cast<uint32_t&>(config_.rtp.local_ssrc) = local_ssrc;
-  rtp_video_stream_receiver_.OnLocalSsrcChange(local_ssrc);
-}
-
 void VideoReceiveStream2::Start() {
   RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
 
diff --git a/video/video_receive_stream2.h b/video/video_receive_stream2.h
index 8122f6d39e..109886cd2a 100644
--- a/video/video_receive_stream2.h
+++ b/video/video_receive_stream2.h
@@ -154,10 +154,6 @@ class VideoReceiveStream2
 
   void SetSync(Syncable* audio_syncable);
 
-  // Updates the `rtp_video_stream_receiver_`'s `local_ssrc` when the default
-  // sender has been created, changed or removed.
-  void SetLocalSsrc(uint32_t local_ssrc);
-
   // Implements webrtc::VideoReceiveStreamInterface.
   void Start() override;
   void Stop() override;
diff --git a/video/video_receive_stream2_unittest.cc b/video/video_receive_stream2_unittest.cc
index e29d6faab3..1b86860b6a 100644
--- a/video/video_receive_stream2_unittest.cc
+++ b/video/video_receive_stream2_unittest.cc
@@ -107,7 +107,6 @@ constexpr uint8_t kH264PayloadType = 99;
 constexpr uint8_t kH265PayloadType = 100;
 constexpr uint8_t kAv1PayloadType = 101;
 constexpr uint32_t kRemoteSsrc = 1111;
-constexpr uint32_t kLocalSsrc = 2222;
 
 class FakeVideoRenderer : public VideoSinkInterface<VideoFrame> {
  public:
@@ -228,7 +227,6 @@ class VideoReceiveStream2Test : public ::testing::TestWithParam<bool> {
 
   void SetUp() override {
     config_.rtp.remote_ssrc = kRemoteSsrc;
-    config_.rtp.local_ssrc = kLocalSsrc;
     config_.renderer = &fake_renderer_;
     VideoReceiveStreamInterface::Decoder h264_decoder;
     h264_decoder.payload_type = kH264PayloadType;
diff --git a/video/video_send_stream_tests.cc b/video/video_send_stream_tests.cc
index e5813e0b4f..3873bb360d 100644
--- a/video/video_send_stream_tests.cc
+++ b/video/video_send_stream_tests.cc
@@ -1077,8 +1077,9 @@ void VideoSendStreamTest::TestNackRetransmission(
         RTCPSender::Configuration config;
         config.outgoing_transport = transport_adapter_.get();
         config.rtcp_report_interval = TimeDelta::Millis(kRtcpIntervalMs);
-        config.local_media_ssrc =
-            test::VideoTestConstants::kReceiverLocalVideoSsrc;
+        config.recv_ssrc_callback = [] {
+          return test::VideoTestConstants::kReceiverLocalVideoSsrc;
+        };
         config.schedule_next_rtcp_send_evaluation = [](TimeDelta) {};
         RTCPSender rtcp_sender(env_, std::move(config));
 
@@ -1281,7 +1282,9 @@ void VideoSendStreamTest::TestPacketFragmentationSize(TestVideoFormat format,
         config.receive_statistics = &lossy_receive_stats;
         config.outgoing_transport = transport_adapter_.get();
         config.rtcp_report_interval = TimeDelta::Millis(kRtcpIntervalMs);
-        config.local_media_ssrc = test::VideoTestConstants::kVideoSendSsrcs[0];
+        config.recv_ssrc_callback = [] {
+          return test::VideoTestConstants::kVideoSendSsrcs[0];
+        };
         config.schedule_next_rtcp_send_evaluation = [](TimeDelta) {};
         RTCPSender rtcp_sender(env_, std::move(config));
 

From b5cb024fe9b7e94db920ce1e2360345715540fc4 Mon Sep 17 00:00:00 2001
From: Per K <perkj@webrtc.org>
Date: Fri, 20 Feb 2026 11:59:08 +0000
Subject: [PATCH 114/472] Improve Scream on low bitrate
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Reduce min_ref_window to 1000 bytes

Use stricter pacing for 15s after last congestion event regardless of event

Remove scaling close to mss since it reduce rampup and adaption speed.

Ensure ref_window_i_ is updated to the ref_window_ before a reduction. Continue to allow it to be changed after every time ref window increase and then decrease.

Bug: webrtc:447037083
Change-Id: I5a6cafa35c09d20b06bb9f05a22d97f687002fde
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449921
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46967}
---
 .../scream/delay_based_congestion_control.cc  |  2 -
 .../scream/scream_network_controller.cc       | 13 ++---
 .../scream_network_controller_unittest.cc     | 53 ++++++++++++++++---
 .../congestion_controller/scream/scream_v2.cc | 15 ++----
 .../scream/scream_v2_parameters.cc            | 10 ++--
 .../scream/scream_v2_parameters.h             |  6 +++
 .../bwe_integration_tests/scream_test.cc      | 51 ++++++++++--------
 7 files changed, 99 insertions(+), 51 deletions(-)

diff --git a/modules/congestion_controller/scream/delay_based_congestion_control.cc b/modules/congestion_controller/scream/delay_based_congestion_control.cc
index a760551750..4cc5e0d31b 100644
--- a/modules/congestion_controller/scream/delay_based_congestion_control.cc
+++ b/modules/congestion_controller/scream/delay_based_congestion_control.cc
@@ -139,8 +139,6 @@ DataSize DelayBasedCongestionControl::UpdateReferenceWindow(
 
   double backoff = l4s_alpha_v() / 2.0;  // Reduce by 50% if l4s_alpha_v = 1.0;
   backoff /= std::max(1.0, last_smoothed_rtt_ / params_.virtual_rtt);
-  backoff *= std::max(0.5, 1.0 - ref_window_mss_ratio);
-
   return std::max(min_allowed_reference_window, (1 - backoff) * ref_window);
 }
 
diff --git a/modules/congestion_controller/scream/scream_network_controller.cc b/modules/congestion_controller/scream/scream_network_controller.cc
index 2037658be2..d50ccc4d5d 100644
--- a/modules/congestion_controller/scream/scream_network_controller.cc
+++ b/modules/congestion_controller/scream/scream_network_controller.cc
@@ -207,13 +207,14 @@ NetworkControlUpdate ScreamNetworkController::CreateUpdate(Timestamp now) {
 }
 
 std::optional<PacerConfig> ScreamNetworkController::MaybeCreatePacerConfig() {
-  // Allow sending packets in larger bursts if data in flight is lower than
-  // reference window.
+  // Allow sending packets in larger bursts if some time has passed since last
+  // congestion event.
   TimeDelta pacing_window =
-      (scream_->delay_based_congestion_control().IsQueueDelayDetected() ||
-       scream_->l4s_alpha() > 0.001)
-          ? TimeDelta::Millis(10)
-          : default_pacing_window_;
+      (env_.clock().CurrentTime() -
+           scream_->last_reaction_to_congestion_time() >
+       params_.allow_large_pacing_bursts_after_congestion_time.Get())
+          ? default_pacing_window_
+          : TimeDelta::Millis(10);
   DataRate target_rate = scream_->target_rate();
   Timestamp now = env_.clock().CurrentTime();
   DataRate padding_rate = DataRate::Zero();
diff --git a/modules/congestion_controller/scream/scream_network_controller_unittest.cc b/modules/congestion_controller/scream/scream_network_controller_unittest.cc
index 409399706c..b3a0d90419 100644
--- a/modules/congestion_controller/scream/scream_network_controller_unittest.cc
+++ b/modules/congestion_controller/scream/scream_network_controller_unittest.cc
@@ -221,7 +221,10 @@ TEST(ScreamControllerTest, PacingWindowReducedIfCeCongestedStreamsConfigured) {
   DataRate send_rate = DataRate::KilobitsPerSec(500);
   for (int i = 0; i < 20; ++i) {
     TransportPacketsFeedback feedback =
-        feedback_generator.ProcessUntilNextFeedback(send_rate, clock);
+        feedback_generator.ProcessUntilNextFeedback(
+            send_rate, clock, [&](const SentPacket& packet) {
+              scream_controller.OnSentPacket(packet);
+            });
     update = scream_controller.OnTransportPacketsFeedback(feedback);
     if (update.target_rate.has_value()) {
       send_rate = update.target_rate->target_rate;
@@ -233,7 +236,7 @@ TEST(ScreamControllerTest, PacingWindowReducedIfCeCongestedStreamsConfigured) {
 }
 
 TEST(ScreamControllerTest,
-     PacingWindowNotReducedIfDelayCongestedStreamsConfigured) {
+     PacingWindowReducedIfDelayCongestedStreamsConfigured) {
   SimulatedClock clock(Timestamp::Seconds(1'234));
   Environment env = CreateTestEnvironment({.time = &clock});
   CcFeedbackGenerator feedback_generator(
@@ -249,9 +252,44 @@ TEST(ScreamControllerTest,
 
   NetworkControlUpdate update;
   DataRate send_rate = DataRate::KilobitsPerSec(500);
-  for (int i = 0; i < 20; ++i) {
+  for (int i = 0; i < 30; ++i) {
     TransportPacketsFeedback feedback =
-        feedback_generator.ProcessUntilNextFeedback(send_rate, clock);
+        feedback_generator.ProcessUntilNextFeedback(
+            send_rate, clock, [&](const SentPacket& packet) {
+              scream_controller.OnSentPacket(packet);
+            });
+    update = scream_controller.OnTransportPacketsFeedback(feedback);
+    if (update.target_rate.has_value()) {
+      send_rate = update.target_rate->target_rate;
+    }
+  }
+  EXPECT_THAT(update.pacer_config,
+              Optional(Field(&PacerConfig::time_window,
+                             Lt(PacerConfig::kDefaultTimeInterval))));
+}
+
+TEST(ScreamControllerTest, PacingWindowNotReducedIfNotCongested) {
+  SimulatedClock clock(Timestamp::Seconds(1'234));
+  Environment env = CreateTestEnvironment({.time = &clock});
+  CcFeedbackGenerator feedback_generator(
+      {.network_config = {.link_capacity = DataRate::KilobitsPerSec(9000)},
+       .send_as_ect1 = false});  // Scream will react to delay increase, not CE.
+
+  NetworkControllerConfig config(env);
+  ScreamNetworkController scream_controller(config);
+
+  StreamsConfig streams_config;
+  streams_config.max_total_allocated_bitrate = DataRate::KilobitsPerSec(1000);
+  scream_controller.OnStreamsConfig(streams_config);
+
+  NetworkControlUpdate update;
+  DataRate send_rate = DataRate::KilobitsPerSec(500);
+  for (int i = 0; i < 30; ++i) {
+    TransportPacketsFeedback feedback =
+        feedback_generator.ProcessUntilNextFeedback(
+            send_rate, clock, [&](const SentPacket& packet) {
+              scream_controller.OnSentPacket(packet);
+            });
     update = scream_controller.OnTransportPacketsFeedback(feedback);
     if (update.target_rate.has_value()) {
       send_rate = update.target_rate->target_rate;
@@ -382,7 +420,7 @@ TEST(ScreamControllerTest, PaddingStopIfNetworkCongested) {
   PaddingTestResult result = ProcessUntilPaddingStartAndStop(
       clock, scream_controller, feedback_generator);
 
-  EXPECT_LT(result.target_rate, DataRate::KilobitsPerSec(600));
+  EXPECT_LT(result.target_rate, DataRate::KilobitsPerSec(700));
   // Padding should stop when congestion is detected.
   EXPECT_LT(result.padding_stop - result.padding_start, TimeDelta::Seconds(1));
 }
@@ -411,10 +449,11 @@ TEST(ScreamControllerTest, PeriodicallyAllowPadding) {
   TimeDelta padding_duration = result_1.padding_stop - result_1.padding_start;
   TimeDelta time_between_padding =
       result_2.padding_start - result_1.padding_stop;
-  EXPECT_GT(padding_duration, TimeDelta::Millis(2900));
+  EXPECT_GT(padding_duration, TimeDelta::Millis(2800));
   EXPECT_LT(padding_duration, TimeDelta::Millis(3100));
+
   EXPECT_GT(time_between_padding, TimeDelta::Millis(2900));
-  EXPECT_LT(time_between_padding, TimeDelta::Millis(3200));
+  EXPECT_LT(time_between_padding, TimeDelta::Millis(3300));
 }
 
 TEST(ScreamControllerTest, PadsToMinOf2xCurrentMaxAndEverSeenMax) {
diff --git a/modules/congestion_controller/scream/scream_v2.cc b/modules/congestion_controller/scream/scream_v2.cc
index ba647d340d..3836c939ca 100644
--- a/modules/congestion_controller/scream/scream_v2.cc
+++ b/modules/congestion_controller/scream/scream_v2.cc
@@ -168,8 +168,6 @@ void ScreamV2::UpdateRefWindow(const TransportPacketsFeedback& msg) {
       // per RTT
       backoff /= std::max(
           1.0, delay_based_congestion_control_.rtt() / params_.virtual_rtt);
-      //  Increase stability for very small ref_wnd
-      backoff *= std::max(0.5, 1.0 - ref_window_mss_ratio());
 
       if (!delay_based_congestion_control_.IsQueueDelayDetected()) {
         // Scale down backoff if close to the last known max reference window
@@ -208,11 +206,6 @@ void ScreamV2::UpdateRefWindow(const TransportPacketsFeedback& msg) {
       ref_window_ = delay_based_congestion_control_.UpdateReferenceWindow(
           ref_window_, ref_window_mss_ratio());
     }
-
-    if (allow_ref_window_i_update_) {
-      ref_window_i_ = ref_window_;
-      allow_ref_window_i_update_ = false;
-    }
   }
 
   // Increase ref_window.
@@ -240,10 +233,6 @@ void ScreamV2::UpdateRefWindow(const TransportPacketsFeedback& msg) {
         increase_scale_factor *
         std::max(0.25, ref_window_scale_factor_close_to_ref_window_i());
 
-    // Limit increase when the reference window close to max segment size.
-    increase_scale_factor =
-        increase_scale_factor * std::max(0.5, 1.0 - ref_window_mss_ratio());
-
     // Limit increase if L4S not enabled and queue delay is increased.
     if (l4s_alpha_ < 0.0001) {
       increase_scale_factor =
@@ -295,6 +284,10 @@ void ScreamV2::UpdateRefWindow(const TransportPacketsFeedback& msg) {
   }
   if (previous_ref_window > ref_window_) {
     last_ref_window_decrease_time_ = msg.feedback_time;
+    if (allow_ref_window_i_update_) {
+      ref_window_i_ = previous_ref_window;
+      allow_ref_window_i_update_ = false;
+    }
   }
 
   RTC_LOG_IF(LS_VERBOSE, previous_ref_window != ref_window_)
diff --git a/modules/congestion_controller/scream/scream_v2_parameters.cc b/modules/congestion_controller/scream/scream_v2_parameters.cc
index 54ad648335..16c91b2d4d 100644
--- a/modules/congestion_controller/scream/scream_v2_parameters.cc
+++ b/modules/congestion_controller/scream/scream_v2_parameters.cc
@@ -18,7 +18,7 @@
 namespace webrtc {
 
 ScreamV2Parameters::ScreamV2Parameters(const FieldTrialsView& trials)
-    : min_ref_window("MinRefWindow", DataSize::Bytes(3000)),
+    : min_ref_window("MinRefWindow", DataSize::Bytes(1000)),
       l4s_avg_g_up("L4sAvgGUp", 1.0 / 8.0),
       l4s_avg_g_down("L4sAvgGDown", 1.0 / 128.0),
       smoothed_rtt_avg_g_up("SmoothedRttAvgGUp", 1.0 / 8.0),
@@ -56,7 +56,10 @@ ScreamV2Parameters::ScreamV2Parameters(const FieldTrialsView& trials)
           "AllowPaddingAfterLastCongestionTimeout",
           TimeDelta::Seconds(1)),
       pacing_factor("PacingFactor", 1.1),
-      feedback_hold_time_avg_g("FeedbackHoldTimeAvgG", 1.0 / 8.0) {
+      feedback_hold_time_avg_g("FeedbackHoldTimeAvgG", 1.0 / 8.0),
+      allow_large_pacing_bursts_after_congestion_time(
+          "AllowLargePacingBurstsAfterCongestionTime",
+          TimeDelta::Seconds(15)) {
   ParseFieldTrial({&min_ref_window,
                    &l4s_avg_g_up,
                    &l4s_avg_g_down,
@@ -85,7 +88,8 @@ ScreamV2Parameters::ScreamV2Parameters(const FieldTrialsView& trials)
                    &periodic_padding_duration,
                    &allow_padding_after_last_congestion_time,
                    &pacing_factor,
-                   &feedback_hold_time_avg_g},
+                   &feedback_hold_time_avg_g,
+                   &allow_large_pacing_bursts_after_congestion_time},
                   trials.Lookup("WebRTC-Bwe-ScreamV2"));
 }
 
diff --git a/modules/congestion_controller/scream/scream_v2_parameters.h b/modules/congestion_controller/scream/scream_v2_parameters.h
index 23a87caa01..53c8296039 100644
--- a/modules/congestion_controller/scream/scream_v2_parameters.h
+++ b/modules/congestion_controller/scream/scream_v2_parameters.h
@@ -119,6 +119,12 @@ struct ScreamV2Parameters {
   // time feedback is delayed by the receiver. I.e the time from a packet is
   // received until feedback is sent. If zero, this delay is ignored.
   FieldTrialParameter<double> feedback_hold_time_avg_g;
+
+  // If the time since last reaction to congestion is larger than this, the
+  // pacing window is increased. I.e. packets are allowed to be sent in larger
+  // bursts.
+  FieldTrialParameter<TimeDelta>
+      allow_large_pacing_bursts_after_congestion_time;
 };
 
 }  // namespace webrtc
diff --git a/test/peer_scenario/bwe_integration_tests/scream_test.cc b/test/peer_scenario/bwe_integration_tests/scream_test.cc
index 5bc1df3110..09f762941d 100644
--- a/test/peer_scenario/bwe_integration_tests/scream_test.cc
+++ b/test/peer_scenario/bwe_integration_tests/scream_test.cc
@@ -54,8 +54,11 @@ using test::GetPacketsSentWithEct1;
 using test::GetStatsAndProcess;
 using test::PeerScenario;
 using test::PeerScenarioClient;
+using ::testing::AllOf;
 using ::testing::Each;
+using ::testing::Gt;
 using ::testing::HasSubstr;
+using ::testing::Lt;
 using ::testing::TestWithParam;
 
 #if defined(ADDRESS_SANITIZER)
@@ -316,9 +319,10 @@ TEST(ScreamTest,
                                               DataRate::KilobitsPerSec(5000))));
 }
 
-TEST(ScreamTest,
-     MaybeTest(
-         LinkCapacityIncreaseFrom80KbitTo5MbpsAfter5sVideoCaptureStopedNoEcn)) {
+TEST(
+    ScreamTest,
+    MaybeTest(
+        LinkCapacityIncreaseFrom80KbitTo5MbpsAfter5sVideoCaptureStoppedNoEcn)) {
   PeerScenario s(*testing::UnitTest::GetInstance()->current_test_info());
   SendMediaTestParams params;
   params.video_capture_enabled = false;
@@ -333,16 +337,16 @@ TEST(ScreamTest,
   EXPECT_THAT(
       result.caller().subspan(1, 3),
       Each(AvailableSendBitrateIsBetween(DataRate::KilobitsPerSec(10),
-                                         DataRate::KilobitsPerSec(150))));
-  // TODO: bugs.webrtc.org/447037083 - Improve Scream at low bitrates. Consider
-  // lowering min ref window.
+                                         DataRate::KilobitsPerSec(100))));
   EXPECT_THAT(result.caller()[3],
               CurrentRoundTripTimeIsBetween(TimeDelta::Millis(40),
-                                            TimeDelta::Millis(500)));
+                                            TimeDelta::Millis(200)));
   // Stats after 9s
-  // TODO: bugs.webrtc.org/447037083 - Ensure BWE does ramp up.
+  // TODO: bugs.webrtc.org/447037083 - Figure out why setting max_video_bitrate
+  // does not force available send bitrate to increase despite that the camera
+  // is disabled.
   EXPECT_THAT(result.caller_stats.back(),
-              AvailableSendBitrateIsBetween(DataRate::KilobitsPerSec(200),
+              AvailableSendBitrateIsBetween(DataRate::KilobitsPerSec(100),
                                             DataRate::KilobitsPerSec(5000)));
 }
 
@@ -498,7 +502,7 @@ TEST(ScreamTest, MaybeTest(LinkCapacity2MbpsRtt50msEcn)) {
 
   EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
                                               DataRate::KilobitsPerSec(1300),
-                                              DataRate::KilobitsPerSec(2100))));
+                                              DataRate::KilobitsPerSec(2300))));
 }
 
 TEST(ScreamTest, MaybeTest(LinkCapacity2MbpsRtt50msNoEcnWithGoogCC)) {
@@ -696,8 +700,9 @@ TEST(ScreamTest, MaybeTest(LinkCapacity100Kbit50msRttNoEcn)) {
       CreateNetworkPath(s, /*use_dual_pi= */ false,
                         DataRate::KilobitsPerSec(100), TimeDelta::Millis(25));
   SendMediaTestResult result = SendMediaInOneDirection(std::move(params), s);
-
-  // TODO: bugs.webrtc.org/447037083 - Investigate how the delay can be reduced.
+  EXPECT_THAT(result.caller().subspan(4),
+              Each(CurrentRoundTripTimeIsBetween(TimeDelta::Millis(40),
+                                                 TimeDelta::Millis(200))));
   EXPECT_THAT(result.caller(), Each(AvailableSendBitrateIsBetween(
                                    DataRate::KilobitsPerSec(10),
                                    DataRate::KilobitsPerSec(150))));
@@ -718,6 +723,10 @@ TEST(ScreamTest, MaybeTest(LinkCapacity1MbitRtt50msWithShortQueuesNoEcn)) {
   EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
                                               DataRate::KilobitsPerSec(200),
                                               DataRate::KilobitsPerSec(1100))));
+  EXPECT_THAT(
+      GetPacketsLost(result.callee_stats.back()) /
+          static_cast<double>(GetPacketsSent(result.caller_stats.back())),
+      Lt(0.05));
 }
 
 TEST(ScreamTest,
@@ -737,7 +746,7 @@ TEST(ScreamTest,
             0.05 * GetPacketsSent(result.caller_stats.back()));
   // Ignore estimate during rampup.
   EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
-                                              DataRate::KilobitsPerSec(200),
+                                              DataRate::KilobitsPerSec(100),
                                               DataRate::KilobitsPerSec(1100))));
 }
 
@@ -785,7 +794,7 @@ TEST(ScreamTest, MaybeTest(SendVideoOnlyReturnLinkWithBurstLoss)) {
   EXPECT_GT(GetPacketsSent(result.caller_stats.back()),
             GetPacketsSent(result.caller_stats[5]));
   EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
-                                              DataRate::KilobitsPerSec(50),
+                                              DataRate::KilobitsPerSec(10),
                                               DataRate::KilobitsPerSec(1100))));
 }
 
@@ -807,12 +816,13 @@ TEST(ScreamTest, MaybeTest(LinkCapacity5MbitPolicedTo256Kbit)) {
 
   SendMediaTestResult result = SendMediaInOneDirection(std::move(params), s);
 
-  // TODO: bugs.webrtc.org/447037083 - Improve Scream at low bitrates with
-  // policed networks. Make it less aggressive to ramp up after backdown due to
-  // loss. Consider lowering min ref window.
   EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
-                                              DataRate::KilobitsPerSec(50),
-                                              DataRate::KilobitsPerSec(1100))));
+                                              DataRate::KilobitsPerSec(150),
+                                              DataRate::KilobitsPerSec(900))));
+  EXPECT_THAT(
+      GetPacketsLost(result.callee_stats.back()) /
+          static_cast<double>(GetPacketsSent(result.caller_stats.back())),
+      AllOf(Lt(0.08), Gt(0.01)));
 }
 
 TEST(ScreamTest, MaybeTest(LinkCapacity5MbitWithCrossTrafficNoEcn)) {
@@ -845,9 +855,6 @@ TEST(ScreamTest, MaybeTest(LinkCapacity5MbitWithCrossTrafficNoEcn)) {
   SendMediaTestResult result = SendMediaInOneDirection(std::move(params), s);
   ASSERT_TRUE(tcp_message_delivered_time.IsFinite());
 
-  // TODO: bugs.webrtc.org/447037083 - Consider if Scream can ramp up faster.
-  // Currently it is slow due to that `queue_delay_dev_norm` is high after the
-  // cross traffic.
   int index_where_available_bitrate_should_have_recovered =
       (tcp_message_delivered_time - start_time).seconds<int>() + 10;
   EXPECT_THAT(

From 700ef22406d9e03e7a36464efb3d210dc79e8de7 Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Fri, 20 Feb 2026 13:54:33 +0100
Subject: [PATCH 115/472] Mark one of the constructor explicit for ArrayView
 with static extent

to match similar std::span constructor restriction.

Update users of that constructor to use it explicitly, or rely on other
means to create an ArrayView with static extent

Bug: webrtc:439801349
Change-Id: If5836af3d42c6eb43e109f9418a460332d78d744
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/450920
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46968}
---
 api/array_view.h                              |  3 +-
 .../agc2/rnn_vad/pitch_search_unittest.cc     |  7 +-
 .../agc2/rnn_vad/ring_buffer.h                |  2 +-
 .../agc2/rnn_vad/rnn_vad_unittest.cc          | 19 +++---
 .../agc2/rnn_vad/sequence_buffer.h            |  6 +-
 .../rnn_vad/spectral_features_unittest.cc     | 67 +++++++++----------
 .../symmetric_matrix_buffer_unittest.cc       |  5 +-
 modules/audio_processing/agc2/vad_wrapper.cc  |  2 +-
 8 files changed, 57 insertions(+), 54 deletions(-)

diff --git a/api/array_view.h b/api/array_view.h
index 5d7a5beb6b..f67b8b971b 100644
--- a/api/array_view.h
+++ b/api/array_view.h
@@ -166,7 +166,8 @@ class ArrayView final : public array_view_internal::ArrayViewBase<T, Size> {
 
   // Construct an ArrayView from a pointer and a length.
   template <typename U>
-  ArrayView(U* data, size_t size)
+  explicit(Size != array_view_internal::kArrayViewVarSize)  //
+      ArrayView(U* data, size_t size)
       : array_view_internal::ArrayViewBase<T, Size>::ArrayViewBase(data, size) {
     RTC_DCHECK_EQ(size == 0 ? nullptr : data, this->data());
     RTC_DCHECK_EQ(size, this->size());
diff --git a/modules/audio_processing/agc2/rnn_vad/pitch_search_unittest.cc b/modules/audio_processing/agc2/rnn_vad/pitch_search_unittest.cc
index 4cddd566f7..633a9e5466 100644
--- a/modules/audio_processing/agc2/rnn_vad/pitch_search_unittest.cc
+++ b/modules/audio_processing/agc2/rnn_vad/pitch_search_unittest.cc
@@ -11,7 +11,7 @@
 #include "modules/audio_processing/agc2/rnn_vad/pitch_search.h"
 
 #include <algorithm>
-#include <vector>
+#include <array>
 
 #include "modules/audio_processing/agc2/cpu_features.h"
 #include "modules/audio_processing/agc2/rnn_vad/common.h"
@@ -28,7 +28,7 @@ namespace rnn_vad {
 TEST(RnnVadTest, PitchSearchWithinTolerance) {
   ChunksFileReader reader = CreateLpResidualAndPitchInfoReader();
   const int num_frames = std::min(reader.num_chunks, 300);  // Max 3 s.
-  std::vector<float> lp_residual(kBufSize24kHz);
+  std::array<float, kBufSize24kHz> lp_residual = {};
   float expected_pitch_period, expected_pitch_strength;
   const AvailableCpuFeatures cpu_features = GetAvailableCpuFeatures();
   PitchEstimator pitch_estimator(cpu_features);
@@ -40,8 +40,7 @@ TEST(RnnVadTest, PitchSearchWithinTolerance) {
       ASSERT_TRUE(reader.reader->ReadChunk(lp_residual));
       ASSERT_TRUE(reader.reader->ReadValue(expected_pitch_period));
       ASSERT_TRUE(reader.reader->ReadValue(expected_pitch_strength));
-      int pitch_period =
-          pitch_estimator.Estimate({lp_residual.data(), kBufSize24kHz});
+      int pitch_period = pitch_estimator.Estimate(lp_residual);
       EXPECT_EQ(expected_pitch_period, pitch_period);
       EXPECT_NEAR(expected_pitch_strength,
                   pitch_estimator.GetLastPitchStrengthForTesting(), 15e-6f);
diff --git a/modules/audio_processing/agc2/rnn_vad/ring_buffer.h b/modules/audio_processing/agc2/rnn_vad/ring_buffer.h
index 0f12b39805..61ecc380e2 100644
--- a/modules/audio_processing/agc2/rnn_vad/ring_buffer.h
+++ b/modules/audio_processing/agc2/rnn_vad/ring_buffer.h
@@ -52,7 +52,7 @@ class RingBuffer {
     int offset = tail_ - 1 - delay;
     if (offset < 0)
       offset += N;
-    return {buffer_.data() + S * offset, S};
+    return ArrayView<const T, S>(buffer_.data() + S * offset, S);
   }
 
  private:
diff --git a/modules/audio_processing/agc2/rnn_vad/rnn_vad_unittest.cc b/modules/audio_processing/agc2/rnn_vad/rnn_vad_unittest.cc
index 5015957f69..37e319743c 100644
--- a/modules/audio_processing/agc2/rnn_vad/rnn_vad_unittest.cc
+++ b/modules/audio_processing/agc2/rnn_vad/rnn_vad_unittest.cc
@@ -13,6 +13,7 @@
 #include <memory>
 #include <vector>
 
+#include "api/array_view.h"
 #include "common_audio/resampler/push_sinc_resampler.h"
 #include "modules/audio_processing/agc2/cpu_features.h"
 #include "modules/audio_processing/agc2/rnn_vad/common.h"
@@ -74,9 +75,9 @@ TEST_P(RnnVadProbabilityParametrization, RnnVadProbabilityWithinTolerance) {
   const int num_frames = samples_reader->size() / kFrameSize10ms48kHz;
 
   // Init buffers.
-  std::vector<float> samples_48k(kFrameSize10ms48kHz);
-  std::vector<float> samples_24k(kFrameSize10ms24kHz);
-  std::vector<float> feature_vector(kFeatureVectorSize);
+  std::array<float, kFrameSize10ms48kHz> samples_48k = {};
+  std::array<float, kFrameSize10ms24kHz> samples_24k = {};
+  std::array<float, kFeatureVectorSize> feature_vector = {};
   std::vector<float> computed_vad_prob(num_frames);
   std::vector<float> expected_vad_prob(num_frames);
 
@@ -90,10 +91,9 @@ TEST_P(RnnVadProbabilityParametrization, RnnVadProbabilityWithinTolerance) {
     decimator.Resample(samples_48k.data(), samples_48k.size(),
                        samples_24k.data(), samples_24k.size());
     bool is_silence = features_extractor.CheckSilenceComputeFeatures(
-        {samples_24k.data(), kFrameSize10ms24kHz},
-        {feature_vector.data(), kFeatureVectorSize});
-    computed_vad_prob[i] = rnn_vad.ComputeVadProbability(
-        {feature_vector.data(), kFeatureVectorSize}, is_silence);
+        samples_24k, feature_vector);
+    computed_vad_prob[i] =
+        rnn_vad.ComputeVadProbability(feature_vector, is_silence);
     EXPECT_NEAR(computed_vad_prob[i], expected_vad_prob[i], 1e-3f);
     cumulative_error += std::abs(computed_vad_prob[i] - expected_vad_prob[i]);
   }
@@ -141,8 +141,9 @@ TEST_P(RnnVadProbabilityParametrization, DISABLED_RnnVadPerformance) {
     perf_timer.StartTimer();
     for (int i = 0; i < num_frames; ++i) {
       bool is_silence = features_extractor.CheckSilenceComputeFeatures(
-          {&prefetched_decimated_samples[i * kFrameSize10ms24kHz],
-           kFrameSize10ms24kHz},
+          ArrayView<const float, kFrameSize10ms24kHz>(
+              &prefetched_decimated_samples[i * kFrameSize10ms24kHz],
+              kFrameSize10ms24kHz),
           feature_vector);
       rnn_vad.ComputeVadProbability(feature_vector, is_silence);
     }
diff --git a/modules/audio_processing/agc2/rnn_vad/sequence_buffer.h b/modules/audio_processing/agc2/rnn_vad/sequence_buffer.h
index c88683a2da..6870f951d1 100644
--- a/modules/audio_processing/agc2/rnn_vad/sequence_buffer.h
+++ b/modules/audio_processing/agc2/rnn_vad/sequence_buffer.h
@@ -50,13 +50,15 @@ class SequenceBuffer {
   // Sets the sequence buffer values to zero.
   void Reset() { std::fill(buffer_.begin(), buffer_.end(), 0); }
   // Returns a view on the whole buffer.
-  ArrayView<const T, S> GetBufferView() const { return {buffer_.data(), S}; }
+  ArrayView<const T, S> GetBufferView() const {
+    return ArrayView<const T, S>(buffer_.data(), S);
+  }
   // Returns a view on the M most recent values of the buffer.
   ArrayView<const T, M> GetMostRecentValuesView() const {
     static_assert(M <= S,
                   "The number of most recent values cannot be larger than the "
                   "sequence buffer size.");
-    return {buffer_.data() + S - M, M};
+    return ArrayView<const T, M>(buffer_.data() + S - M, M);
   }
   // Shifts left the buffer by N items and add new N items at the end.
   void Push(ArrayView<const T, N> new_values) {
diff --git a/modules/audio_processing/agc2/rnn_vad/spectral_features_unittest.cc b/modules/audio_processing/agc2/rnn_vad/spectral_features_unittest.cc
index 3f416bf642..7c9403391d 100644
--- a/modules/audio_processing/agc2/rnn_vad/spectral_features_unittest.cc
+++ b/modules/audio_processing/agc2/rnn_vad/spectral_features_unittest.cc
@@ -35,34 +35,34 @@ void WriteTestData(ArrayView<float> samples) {
 }
 
 ArrayView<float, kNumBands - kNumLowerBands> GetHigherBandsSpectrum(
-    std::array<float, kTestFeatureVectorSize>* feature_vector) {
-  return {feature_vector->data() + kNumLowerBands, kNumBands - kNumLowerBands};
+    ArrayView<float, kTestFeatureVectorSize> feature_vector) {
+  return feature_vector.subspan<kNumLowerBands, kNumBands - kNumLowerBands>();
 }
 
 ArrayView<float, kNumLowerBands> GetAverage(
-    std::array<float, kTestFeatureVectorSize>* feature_vector) {
-  return {feature_vector->data(), kNumLowerBands};
+    ArrayView<float, kTestFeatureVectorSize> feature_vector) {
+  return feature_vector.first<kNumLowerBands>();
 }
 
 ArrayView<float, kNumLowerBands> GetFirstDerivative(
-    std::array<float, kTestFeatureVectorSize>* feature_vector) {
-  return {feature_vector->data() + kNumBands, kNumLowerBands};
+    ArrayView<float, kTestFeatureVectorSize> feature_vector) {
+  return feature_vector.subspan<kNumBands, kNumLowerBands>();
 }
 
 ArrayView<float, kNumLowerBands> GetSecondDerivative(
-    std::array<float, kTestFeatureVectorSize>* feature_vector) {
-  return {feature_vector->data() + kNumBands + kNumLowerBands, kNumLowerBands};
+    ArrayView<float, kTestFeatureVectorSize> feature_vector) {
+  return feature_vector.subspan<kNumBands + kNumLowerBands, kNumLowerBands>();
 }
 
 ArrayView<float, kNumLowerBands> GetCepstralCrossCorrelation(
-    std::array<float, kTestFeatureVectorSize>* feature_vector) {
-  return {feature_vector->data() + kNumBands + 2 * kNumLowerBands,
-          kNumLowerBands};
+    ArrayView<float, kTestFeatureVectorSize> feature_vector) {
+  return feature_vector
+      .subspan<kNumBands + 2 * kNumLowerBands, kNumLowerBands>();
 }
 
 float* GetCepstralVariability(
-    std::array<float, kTestFeatureVectorSize>* feature_vector) {
-  return feature_vector->data() + kNumBands + 3 * kNumLowerBands;
+    ArrayView<float, kTestFeatureVectorSize> feature_vector) {
+  return &feature_vector[kNumBands + 3 * kNumLowerBands];
 }
 
 constexpr float kInitialFeatureVal = -9999.f;
@@ -86,11 +86,11 @@ TEST(RnnVadTest, SpectralFeaturesWithAndWithoutSilence) {
   // With silence.
   std::fill(samples.begin(), samples.end(), 0.f);
   is_silence = sfe.CheckSilenceComputeFeatures(
-      samples_view, samples_view, GetHigherBandsSpectrum(&feature_vector),
-      GetAverage(&feature_vector), GetFirstDerivative(&feature_vector),
-      GetSecondDerivative(&feature_vector),
-      GetCepstralCrossCorrelation(&feature_vector),
-      GetCepstralVariability(&feature_vector));
+      samples_view, samples_view, GetHigherBandsSpectrum(feature_vector),
+      GetAverage(feature_vector), GetFirstDerivative(feature_vector),
+      GetSecondDerivative(feature_vector),
+      GetCepstralCrossCorrelation(feature_vector),
+      GetCepstralVariability(feature_vector));
   // Silence is expected, the output won't be overwritten.
   EXPECT_TRUE(is_silence);
   EXPECT_TRUE(std::all_of(feature_vector.begin(), feature_vector.end(),
@@ -99,11 +99,11 @@ TEST(RnnVadTest, SpectralFeaturesWithAndWithoutSilence) {
   // With no silence.
   WriteTestData(samples);
   is_silence = sfe.CheckSilenceComputeFeatures(
-      samples_view, samples_view, GetHigherBandsSpectrum(&feature_vector),
-      GetAverage(&feature_vector), GetFirstDerivative(&feature_vector),
-      GetSecondDerivative(&feature_vector),
-      GetCepstralCrossCorrelation(&feature_vector),
-      GetCepstralVariability(&feature_vector));
+      samples_view, samples_view, GetHigherBandsSpectrum(feature_vector),
+      GetAverage(feature_vector), GetFirstDerivative(feature_vector),
+      GetSecondDerivative(feature_vector),
+      GetCepstralCrossCorrelation(feature_vector),
+      GetCepstralVariability(feature_vector));
   // Silence is not expected, the output will be overwritten.
   EXPECT_FALSE(is_silence);
   EXPECT_FALSE(std::all_of(feature_vector.begin(), feature_vector.end(),
@@ -125,22 +125,21 @@ TEST(RnnVadTest, CepstralFeaturesConstantAverageZeroDerivative) {
   std::array<float, kTestFeatureVectorSize> feature_vector;
   for (int i = 0; i < kCepstralCoeffsHistorySize; ++i) {
     sfe.CheckSilenceComputeFeatures(
-        samples_view, samples_view, GetHigherBandsSpectrum(&feature_vector),
-        GetAverage(&feature_vector), GetFirstDerivative(&feature_vector),
-        GetSecondDerivative(&feature_vector),
-        GetCepstralCrossCorrelation(&feature_vector),
-        GetCepstralVariability(&feature_vector));
+        samples_view, samples_view, GetHigherBandsSpectrum(feature_vector),
+        GetAverage(feature_vector), GetFirstDerivative(feature_vector),
+        GetSecondDerivative(feature_vector),
+        GetCepstralCrossCorrelation(feature_vector),
+        GetCepstralVariability(feature_vector));
   }
 
   // Feed the test data one last time but using a different output vector.
   std::array<float, kTestFeatureVectorSize> feature_vector_last;
   sfe.CheckSilenceComputeFeatures(
-      samples_view, samples_view, GetHigherBandsSpectrum(&feature_vector_last),
-      GetAverage(&feature_vector_last),
-      GetFirstDerivative(&feature_vector_last),
-      GetSecondDerivative(&feature_vector_last),
-      GetCepstralCrossCorrelation(&feature_vector_last),
-      GetCepstralVariability(&feature_vector_last));
+      samples_view, samples_view, GetHigherBandsSpectrum(feature_vector_last),
+      GetAverage(feature_vector_last), GetFirstDerivative(feature_vector_last),
+      GetSecondDerivative(feature_vector_last),
+      GetCepstralCrossCorrelation(feature_vector_last),
+      GetCepstralVariability(feature_vector_last));
 
   // Average is unchanged.
   ExpectEqualFloatArray({feature_vector.data(), kNumLowerBands},
diff --git a/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer_unittest.cc b/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer_unittest.cc
index b041acba58..43ca270586 100644
--- a/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer_unittest.cc
+++ b/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer_unittest.cc
@@ -14,6 +14,7 @@
 #include <array>
 #include <utility>
 
+#include "api/array_view.h"
 #include "modules/audio_processing/agc2/rnn_vad/ring_buffer.h"
 #include "test/gtest.h"
 
@@ -61,7 +62,7 @@ TEST(RnnVadTest, SymmetricMatrixBufferUseCase) {
   for (int t = 1; t <= 100; ++t) {  // Evolution steps.
     SCOPED_TRACE(t);
     const int t_removed = ring_buf.GetArrayView(kRingBufSize - 1)[0];
-    ring_buf.Push({&t, 1});
+    ring_buf.Push(ArrayView<int, 1>(&t, 1));
     // The head of the ring buffer is `t`.
     ASSERT_EQ(t, ring_buf.GetArrayView(0)[0]);
     // Create the comparisons between `t` and the older elements in the ring
@@ -77,7 +78,7 @@ TEST(RnnVadTest, SymmetricMatrixBufferUseCase) {
       new_comparions[i].second = t;
     }
     // Push the new comparisons in the symmetric matrix buffer.
-    sym_matrix_buf.Push({new_comparions.data(), new_comparions.size()});
+    sym_matrix_buf.Push(new_comparions);
     // Tests.
     CheckSymmetry(&sym_matrix_buf);
     // Check that the pairs resulting from the content in the ring buffer are
diff --git a/modules/audio_processing/agc2/vad_wrapper.cc b/modules/audio_processing/agc2/vad_wrapper.cc
index 0d77e7fb25..0d77181a16 100644
--- a/modules/audio_processing/agc2/vad_wrapper.cc
+++ b/modules/audio_processing/agc2/vad_wrapper.cc
@@ -42,7 +42,7 @@ class MonoVadImpl : public VoiceActivityDetectorWrapper::MonoVad {
     RTC_DCHECK_EQ(frame.size(), rnn_vad::kFrameSize10ms24kHz);
     std::array<float, rnn_vad::kFeatureVectorSize> feature_vector;
     const bool is_silence = features_extractor_.CheckSilenceComputeFeatures(
-        /*samples=*/{frame.data(), rnn_vad::kFrameSize10ms24kHz},
+        /*samples=*/frame.first<rnn_vad::kFrameSize10ms24kHz>(),
         feature_vector);
     return rnn_vad_.ComputeVadProbability(feature_vector, is_silence);
   }

From 25ffb6278cee46a19c6d73800422b39d5f11f994 Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Fri, 20 Feb 2026 20:07:08 -0800
Subject: [PATCH 116/472] Update WebRTC code version (2026-02-21T04:06:53).

Bug: None
Change-Id: Ied1ee914e0734c25724653e1ecdd1c89b8ab9582
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/451080
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#46969}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index 6bf3db7608..3a89c1d3c3 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-02-20T04:04:46";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-02-21T04:06:53";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From ede6975505e8ebd3f077ecc190eecbdd58e26f12 Mon Sep 17 00:00:00 2001
From: Per K <perkj@webrtc.org>
Date: Sat, 21 Feb 2026 15:58:00 +0000
Subject: [PATCH 117/472] Remove flaky expectation on RTT in Scream test
 LinkCapacity100Kbit50msRttNoEcn

Bug: webrtc:447037083
Change-Id: I68551d6ac9492d248bf8a1e0fea6d49ec2c6f2b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/451180
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46970}
---
 test/peer_scenario/bwe_integration_tests/scream_test.cc | 3 ---
 1 file changed, 3 deletions(-)

diff --git a/test/peer_scenario/bwe_integration_tests/scream_test.cc b/test/peer_scenario/bwe_integration_tests/scream_test.cc
index 09f762941d..dac4e2ff6f 100644
--- a/test/peer_scenario/bwe_integration_tests/scream_test.cc
+++ b/test/peer_scenario/bwe_integration_tests/scream_test.cc
@@ -700,9 +700,6 @@ TEST(ScreamTest, MaybeTest(LinkCapacity100Kbit50msRttNoEcn)) {
       CreateNetworkPath(s, /*use_dual_pi= */ false,
                         DataRate::KilobitsPerSec(100), TimeDelta::Millis(25));
   SendMediaTestResult result = SendMediaInOneDirection(std::move(params), s);
-  EXPECT_THAT(result.caller().subspan(4),
-              Each(CurrentRoundTripTimeIsBetween(TimeDelta::Millis(40),
-                                                 TimeDelta::Millis(200))));
   EXPECT_THAT(result.caller(), Each(AvailableSendBitrateIsBetween(
                                    DataRate::KilobitsPerSec(10),
                                    DataRate::KilobitsPerSec(150))));

From 57aaff451c07445bd0fb05da43b8f5df5e7fd6c0 Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Sat, 21 Feb 2026 21:57:56 +0000
Subject: [PATCH 118/472] Make rtp_sender pointer const in ModuleRtpRtcpImpl2

This makes it obvious that the member doesn't need a thread guard.
Also add documentation of the locking/threading model for the module
and its subcomponents.

Bug: None
Change-Id: Id2886d1f48f52c274e087521d3199e565604cc0d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/451260
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46971}
---
 modules/rtp_rtcp/source/rtp_rtcp_impl2.cc |  7 +++++--
 modules/rtp_rtcp/source/rtp_rtcp_impl2.h  | 18 +++++++++++++++++-
 2 files changed, 22 insertions(+), 3 deletions(-)

diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
index dd4610ac55..8e8b2e036c 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
@@ -83,6 +83,11 @@ ModuleRtpRtcpImpl2::ModuleRtpRtcpImpl2(
     : env_(env),
       worker_queue_(TaskQueueBase::Current()),
       recv_ssrc_callback_(std::move(recv_ssrc_callback)),
+      rtp_sender_(configuration.receiver_only
+                      ? nullptr
+                      : std::make_unique<RtpSenderContext>(env_,
+                                                           *worker_queue_,
+                                                           configuration)),
       rtcp_sender_(
           env_,
           {.audio = configuration.audio,
@@ -116,8 +121,6 @@ ModuleRtpRtcpImpl2::ModuleRtpRtcpImpl2(
   RTC_DCHECK(worker_queue_);
   rtcp_thread_checker_.Detach();
   if (!configuration.receiver_only) {
-    rtp_sender_ =
-        std::make_unique<RtpSenderContext>(env_, *worker_queue_, configuration);
     rtp_sender_->sequencing_checker.Detach();
     // Make sure rtcp sender use same timestamp offset as rtp sender.
     rtcp_sender_.SetTimestampOffset(
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl2.h b/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
index 86a40da740..f5939b6787 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
@@ -57,6 +57,22 @@ namespace webrtc {
 struct PacedPacketInfo;
 struct RTPVideoHeader;
 
+// This class and its members handles sending and receiving of RTP and RTCP
+// on behalf of a single media stream (incoming or outgoing).
+//
+// Threading and locking model:
+// Instances of ModuleRtpRtcpImpl2 are created and destroyed on the worker
+// queue.
+// The RTP sender lives on the RTP packet producing thread,
+// identified by rtp_sender_->sequencing_checker; calls that are forwarded
+// to the rtp_sender_ need to be called on this thread.
+// The RTCPSender allows multithread access using its `mutex_rtcp_sender_`.
+// The RTCPReceiver does the same using `mutex_receiver_lock_`.
+// These three objects are therefore not marked with guarding in this class.
+// Incoming RTCP packets are processed on the thread bound to
+// `rtcp_thread_checker_`.
+// `mutex_rtt_` allows multi-thread access to the "rtt" member.
+
 class ModuleRtpRtcpImpl2 final : public RtpRtcpInterface,
                                  public RTCPReceiver::ModuleRtpRtcp {
  public:
@@ -338,7 +354,7 @@ class ModuleRtpRtcpImpl2 final : public RtpRtcpInterface,
   // Must outlive rtcp_sender_, so placed before it.
   absl::AnyInvocable<uint32_t() const> recv_ssrc_callback_;
 
-  std::unique_ptr<RtpSenderContext> rtp_sender_;
+  const std::unique_ptr<RtpSenderContext> rtp_sender_;
   RTCPSender rtcp_sender_;
   RTCPReceiver rtcp_receiver_;
 

From a7a1467ad4fcaba4f6ae49ad01cc0acbb543ef28 Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Sat, 21 Feb 2026 20:11:01 -0800
Subject: [PATCH 119/472] Update WebRTC code version (2026-02-22T04:10:27).

Bug: None
Change-Id: I1516f97eb8df4190a3d6a93d799b4435e0d7f10c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/451341
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#46972}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index 3a89c1d3c3..6b3f56aff6 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-02-21T04:06:53";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-02-22T04:10:27";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From 7d6c068013049f656b836fe46fdd8bba940d23ce Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Sun, 22 Feb 2026 20:05:36 -0800
Subject: [PATCH 120/472] Update WebRTC code version (2026-02-23T04:05:06).

Bug: None
Change-Id: I5e8ae36d3c85d6a2b0d05a50385de3589a48e3da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/451560
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#46973}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index 6b3f56aff6..94a24f266f 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-02-22T04:10:27";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-02-23T04:05:06";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From 1be85893f6c07ceb617f3b71e774d3c259adcdbc Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Mon, 23 Feb 2026 08:02:42 -0800
Subject: [PATCH 121/472] Roll chromium_revision 1c9161b9bb..80fdbfe134
 (1587773:1588711)

Change log: https://chromium.googlesource.com/chromium/src/+log/1c9161b9bb..80fdbfe134
Full diff: https://chromium.googlesource.com/chromium/src/+/1c9161b9bb..80fdbfe134

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/39d4202659..31ac3ef412
* src/buildtools/linux64: git_revision:2d1bf87c34a9ec952aac98bc6d88b978abfa9981..git_revision:2c775ed71ed9c7ceb11823ced47a1a2640883403
* src/buildtools/mac: git_revision:2d1bf87c34a9ec952aac98bc6d88b978abfa9981..git_revision:2c775ed71ed9c7ceb11823ced47a1a2640883403
* src/buildtools/win: git_revision:2d1bf87c34a9ec952aac98bc6d88b978abfa9981..git_revision:2c775ed71ed9c7ceb11823ced47a1a2640883403
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/f16867329d..c68ca0692e
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/0e43b2634d..45ac53e274
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/54440292c8..7361a78642
* src/third_party/android_build_tools/manifest_merger/cipd: 7f2hBQelhgttzGk5TKRjsOLsTW3KYFbL66HKIXfh9tsC..vWr5VRbtPlKQeEMNOksUpsAqeH2yaPhBRf3C59yVhzQC
* src/third_party/androidx/cipd: jcsrqbC9ptL2-ce_XhN-xUE53nHvM345DTcMXbVabSAC..DLUFuZ1z4MFDK54yAzN2lScwHZlUoJssu_cnrpDbqwYC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/6a584e1d58..3d72439f88
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/172c8e1ba5..9fdb1d4c9d
* src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/6813553ab8..02ae8c1e75
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/6d153ac3a6..8fc02b1d3e
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/374fe225bf..667cabe3aa
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/f2b7408057..c485bf9b09
* src/third_party/kotlin_stdlib/cipd: JcCPEPsUevwiZ519BkXHvRR8J-9XEcwj_plAWjVIR7MC..MQrhfTOsQoBh8QajYKNMkPBvzh7bqyv31z7HPbnbo_QC
* src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/14776ea33c..a91d484f4c
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/a3e107ec80..601cac04b0
* src/third_party/r8/cipd: ygoUIlS4O_2F8NuFV3YNhN0oBwVMSgi9mipaKA9uaysC..OdXvkYaHnRN6eudfIbRYWbc4JTiLo6bJKvftjyuQt2UC
* src/third_party/turbine/cipd: BMHNhxMhr7uGz1rh_Od_JE4kAdP9K5MXr6GN2R9tQkAC..ymsdRx1sSajlZKbTHsqqquqR8lYueFUJCPwDCXmIfMEC
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/8e77480101..6580288c8d
* src/tools/luci-go: git_revision:4d646a5cce1959bdfb88c838b1044354c37f0655..git_revision:21504d3595ac903ef60bac732d885cd8e6ed92ba
* src/tools/luci-go: git_revision:4d646a5cce1959bdfb88c838b1044354c37f0655..git_revision:21504d3595ac903ef60bac732d885cd8e6ed92ba
* src/tools/luci-go: git_revision:4d646a5cce1959bdfb88c838b1044354c37f0655..git_revision:21504d3595ac903ef60bac732d885cd8e6ed92ba
DEPS diff: https://chromium.googlesource.com/chromium/src/+/1c9161b9bb..80fdbfe134/DEPS

BUG=None

Change-Id: Ib8ed92c90b4bd40b44e063424964eaed17ac6ca7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/451701
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#46974}
---
 DEPS | 50 +++++++++++++++++++++++++-------------------------
 1 file changed, 25 insertions(+), 25 deletions(-)

diff --git a/DEPS b/DEPS
index ac3521069b..e535f97b5b 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '1c9161b9bb60dab9783399d2178561168cba0090',
+  'chromium_revision': '80fdbfe13417d357d71d5cd8c24ccf4af6e1df70',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@39d420265929393ffd3080ae3b1740aab1443955',
+    'https://chromium.googlesource.com/chromium/src/build@31ac3ef412f947513c552c7e81f860feb4de2ce0',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@451e71432ac9f6a91d4d5604c29e0a5078e8c2e8',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,19 +78,19 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@f16867329de93739d9a0984a7ee0bee03064fadd',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@c68ca0692ee0da564f287410ab28ce926586445d',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@0e43b2634d95bedd2aee0f6b824153daf76aa8ee',
+    'https://chromium.googlesource.com/chromium/src/testing@45ac53e274c7fae5f221cca2b3d75e7328f071a2',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@54440292c873c5f38c4353fb91f5d23a88f6f213',
+    'https://chromium.googlesource.com/chromium/src/third_party@7361a7864218287d9a9e29af059e7a7d6c6356cc',
 
   'src/buildtools/linux64': {
     'packages': [
       {
         'package': 'gn/gn/linux-${{arch}}',
-        'version': 'git_revision:2d1bf87c34a9ec952aac98bc6d88b978abfa9981',
+        'version': 'git_revision:2c775ed71ed9c7ceb11823ced47a1a2640883403',
       }
     ],
     'dep_type': 'cipd',
@@ -100,7 +100,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/mac-${{arch}}',
-        'version': 'git_revision:2d1bf87c34a9ec952aac98bc6d88b978abfa9981',
+        'version': 'git_revision:2c775ed71ed9c7ceb11823ced47a1a2640883403',
       }
     ],
     'dep_type': 'cipd',
@@ -110,7 +110,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/windows-amd64',
-        'version': 'git_revision:2d1bf87c34a9ec952aac98bc6d88b978abfa9981',
+        'version': 'git_revision:2c775ed71ed9c7ceb11823ced47a1a2640883403',
       }
     ],
     'dep_type': 'cipd',
@@ -331,13 +331,13 @@ deps = {
   'src/third_party/clang-format/script':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@c2725e0622e1a86d55f14514f2177a39efea4a0e',
   'src/third_party/compiler-rt/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@6813553ab8756887eb113da9eba4fb2163f5b8a5',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@02ae8c1e75be8825a6c8c1cc74b4ea93933fb189',
   'src/third_party/libc++/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@7ab65651aed6802d2599dcb7a73b1f82d5179d05',
   'src/third_party/libc++abi/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@8f11bb1d4438d0239d0dfc1bd9456a9f31629dda',
   'src/third_party/llvm-libc/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@14776ea33c0b64c3cec0ce80ff3eae7ddae9d977',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@a91d484f4c303b81ea50cbd396d200c08ebdcab3',
   'src/third_party/libunwind/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@17ccf7d110c5526cb77e93cfd8330f491fb2bf18',
 
@@ -488,11 +488,11 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@6a584e1d58d1fa95bc47d6c92cc9a8cb49d29635',
+    'https://boringssl.googlesource.com/boringssl.git@3d72439f882dd6a9e8ddbc6d62b7c9557d9aac72',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@79099fdf668ae097c9eca7052fd5c4c5de6c9098',
   'src/third_party/catapult':
-    'https://chromium.googlesource.com/catapult.git@172c8e1ba5c324616c708f1ae82b37621365c22f',
+    'https://chromium.googlesource.com/catapult.git@9fdb1d4c9d0f278ae34891b069d96ef89ea9cdf2',
   'src/third_party/ced/src': {
     'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5',
   },
@@ -505,13 +505,13 @@ deps = {
   'src/third_party/crc32c/src':
     'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558',
   'src/third_party/depot_tools':
-    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@6d153ac3a600dea6c2ec320c9eb4df06aa3dca04',
+    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@8fc02b1d3e48470d9d7fd0e6a68d42f0e38b3bac',
   'src/third_party/ffmpeg':
     'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@946d97db8d906277085e361892b7efda5152e2f1',
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@f2b7408057b44429dabb721a69cb55ef23661da8',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@c485bf9b093bd32670b6af8a3a20eae1e2604602',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -573,7 +573,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/kotlin_stdlib',
-              'version': 'JcCPEPsUevwiZ519BkXHvRR8J-9XEcwj_plAWjVIR7MC',
+              'version': 'MQrhfTOsQoBh8QajYKNMkPBvzh7bqyv31z7HPbnbo_QC',
           },
       ],
       'condition': 'checkout_android',
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'a3e107ec803c876a870205f89c1e37742184b598',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '601cac04b0757f1132c64ace99205522d5cabb2d',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -644,7 +644,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/r8',
-              'version': 'ygoUIlS4O_2F8NuFV3YNhN0oBwVMSgi9mipaKA9uaysC',
+              'version': 'OdXvkYaHnRN6eudfIbRYWbc4JTiLo6bJKvftjyuQt2UC',
           },
       ],
       'condition': 'checkout_android',
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@8e77480101fc44debdfe5e1dec41293bacf9b852',
+    'https://chromium.googlesource.com/chromium/src/tools@6580288c8d453357aaf4e3388b62aa423b0bf53c',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'jcsrqbC9ptL2-ce_XhN-xUE53nHvM345DTcMXbVabSAC',
+          'version': 'DLUFuZ1z4MFDK54yAzN2lScwHZlUoJssu_cnrpDbqwYC',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -718,7 +718,7 @@ deps = {
       'packages': [
           {
                'package': 'chromium/third_party/android_build_tools/manifest_merger',
-               'version': '7f2hBQelhgttzGk5TKRjsOLsTW3KYFbL66HKIXfh9tsC',
+               'version': 'vWr5VRbtPlKQeEMNOksUpsAqeH2yaPhBRf3C59yVhzQC',
           },
       ],
       'condition': 'checkout_android',
@@ -792,7 +792,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/turbine',
-              'version': 'BMHNhxMhr7uGz1rh_Od_JE4kAdP9K5MXr6GN2R9tQkAC',
+              'version': 'ymsdRx1sSajlZKbTHsqqquqR8lYueFUJCPwDCXmIfMEC',
           },
       ],
       'condition': 'checkout_android',
@@ -808,15 +808,15 @@ deps = {
       'packages': [
         {
           'package': 'infra/tools/luci/cas/${{platform}}',
-          'version': 'git_revision:4d646a5cce1959bdfb88c838b1044354c37f0655',
+          'version': 'git_revision:21504d3595ac903ef60bac732d885cd8e6ed92ba',
         },
         {
           'package': 'infra/tools/luci/isolate/${{platform}}',
-          'version': 'git_revision:4d646a5cce1959bdfb88c838b1044354c37f0655',
+          'version': 'git_revision:21504d3595ac903ef60bac732d885cd8e6ed92ba',
         },
         {
           'package': 'infra/tools/luci/swarming/${{platform}}',
-          'version': 'git_revision:4d646a5cce1959bdfb88c838b1044354c37f0655',
+          'version': 'git_revision:21504d3595ac903ef60bac732d885cd8e6ed92ba',
         }
       ],
       'dep_type': 'cipd',
@@ -864,7 +864,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7364b490b5f78d58efe23ea76e74210fd6c3c76f',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '374fe225bfadbad6c172b6ef06e21ac2fad90bc2',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '667cabe3aa54d53d6c542950f6d3876c7f0ba58f',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From 30cfc99e522d62646651e5760bc890cb6368bdc1 Mon Sep 17 00:00:00 2001
From: Jonas Oreland <jonaso@webrtc.org>
Date: Mon, 23 Feb 2026 07:36:05 -0800
Subject: [PATCH 122/472] Add (rudimentary) handling of delta too big...i.e.
 don't send it.

A more advanced version could e.g. split an attribute,
or skip an attribute. But I can't see a useful case for this
now. This change however does have a real world use case
in that we currently hard code a max limit that is way lower
just to be safe.

BUG=webrtc:42225723

Change-Id: I9ec751514089eb8774d0deec85926c6130040cc9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/451681
Reviewed-by: Sameer Vijaykar <samvi@google.com>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46975}
---
 p2p/base/connection.cc          | 11 ++++--
 p2p/base/connection_unittest.cc | 62 ++++++++++++++++++++++++++++++++-
 2 files changed, 70 insertions(+), 3 deletions(-)

diff --git a/p2p/base/connection.cc b/p2p/base/connection.cc
index 6603e3f67f..0e3b2a9c74 100644
--- a/p2p/base/connection.cc
+++ b/p2p/base/connection.cc
@@ -1187,8 +1187,15 @@ std::unique_ptr<IceMessage> Connection::BuildPingRequest(
 
   if (delta) {
     RTC_DCHECK(delta->type() == STUN_ATTR_GOOG_DELTA);
-    RTC_LOG(LS_INFO) << "Sending GOOG_DELTA: len: " << delta->length();
-    message->AddAttribute(std::move(delta));
+    size_t msg_length = message->length();
+    if (msg_length + kStunAttributeHeaderSize + delta->length() <
+        kMaxStunBindingLength) {
+      RTC_LOG(LS_INFO) << "Sending GOOG_DELTA: len: " << delta->length();
+      message->AddAttribute(std::move(delta));
+    } else {
+      RTC_LOG(LS_WARNING) << "Not sending GOOG_DELTA, request full: len: "
+                          << delta->length() << " msg_length: " << msg_length;
+    }
   }
 
   MaybeAddDtlsPiggybackingAttributes(message.get());
diff --git a/p2p/base/connection_unittest.cc b/p2p/base/connection_unittest.cc
index d7db86fc3d..1e08606c28 100644
--- a/p2p/base/connection_unittest.cc
+++ b/p2p/base/connection_unittest.cc
@@ -16,6 +16,7 @@
 #include <list>
 #include <memory>
 #include <optional>
+#include <string>
 #include <utility>
 #include <vector>
 
@@ -300,7 +301,6 @@ TEST_F(ConnectionTest, SendReceiveGoogDelta) {
   ASSERT_GT(rport_->last_stun_buf().size(), 0u);
   lconn->OnReadPacket(
       ReceivedIpPacket(rport_->last_stun_buf(), SocketAddress(), std::nullopt));
-
   EXPECT_TRUE(received_goog_delta_ack);
 }
 
@@ -346,5 +346,65 @@ TEST_F(ConnectionTest, SendGoogDeltaNoReply) {
   EXPECT_TRUE(received_goog_delta_ack_error);
 }
 
+// Test that if StunBinding is sufficiently full, aka DELTA is too big, it is
+// not sent.
+TEST_F(ConnectionTest, TooBigDeltaIsNotSent) {
+  constexpr int64_t ms = 10;
+  Connection* lconn = CreateConnection(ICEROLE_CONTROLLING);
+  Connection* rconn = CreateConnection(ICEROLE_CONTROLLED);
+
+  std::string a_long_string(1200, 'a');
+  std::unique_ptr<StunByteStringAttribute> delta =
+      absl::WrapUnique(new StunByteStringAttribute(STUN_ATTR_GOOG_DELTA));
+  delta->CopyBytes(a_long_string);
+
+  std::unique_ptr<StunAttribute> delta_ack =
+      absl::WrapUnique(new StunUInt64Attribute(STUN_ATTR_GOOG_DELTA_ACK, 133));
+
+  bool received_goog_delta = false;
+  bool received_goog_delta_ack = false;
+  lconn->SetStunDictConsumer(
+      // DeltaReceived
+      [](const StunByteStringAttribute* delta)
+          -> std::unique_ptr<StunAttribute> { return nullptr; },
+      // DeltaAckReceived
+      [&](RTCErrorOr<const StunUInt64Attribute*> error_or_ack) {
+        received_goog_delta_ack = true;
+        EXPECT_TRUE(error_or_ack.ok());
+        EXPECT_EQ(error_or_ack.value()->value(), 133ull);
+      });
+
+  rconn->SetStunDictConsumer(
+      // DeltaReceived
+      [&](const StunByteStringAttribute* delta)
+          -> std::unique_ptr<StunAttribute> {
+        received_goog_delta = true;
+        EXPECT_EQ(delta->string_view(), "DELTA");
+        return std::move(delta_ack);
+      },
+      // DeltaAckReceived
+      [](RTCErrorOr<const StunUInt64Attribute*> error_or__ack) {});
+
+  lconn->Ping(env().clock().CurrentTime(), std::move(delta));
+  ASSERT_THAT(WaitUntil([&] { return lport_->last_stun_msg(); }, IsTrue(),
+                        {.timeout = TimeDelta::Millis(kDefaultTimeout),
+                         .clock = &time_controller_}),
+              IsRtcOk());
+  ASSERT_GT(lport_->last_stun_buf().size(), 0u);
+  rconn->OnReadPacket(
+      ReceivedIpPacket(lport_->last_stun_buf(), SocketAddress(), std::nullopt));
+  EXPECT_FALSE(received_goog_delta);
+
+  time_controller_.SkipForwardBy(TimeDelta::Millis(ms));
+  ASSERT_TRUE(WaitUntil([&] { return rport_->last_stun_msg(); },
+                        {.timeout = TimeDelta::Millis(kDefaultTimeout),
+                         .clock = &time_controller_}));
+  ASSERT_GT(rport_->last_stun_buf().size(), 0u);
+  lconn->OnReadPacket(
+      ReceivedIpPacket(rport_->last_stun_buf(), SocketAddress(), std::nullopt));
+
+  EXPECT_FALSE(received_goog_delta_ack);
+}
+
 }  // namespace
 }  // namespace webrtc

From 5c7cca0ff961fe01df697430b453da12a68ecb5b Mon Sep 17 00:00:00 2001
From: Jonas Oreland <jonaso@webrtc.org>
Date: Mon, 23 Feb 2026 08:03:58 -0800
Subject: [PATCH 123/472] dtls-in-stun: fix DTLS 1.x resend/acking behavior

This patch does "the same" as https://webrtc-review.git.corp.google.com/c/src/+/450983
with the following noticeable changes
- handle both 1.2 & 1.3 !
- add unit test
- use dtls/srtp as a signal for then peer is also writable
as show by https://webrtc-review.git.corp.google.com/c/src/+/449762,
(note: it use existing callbacks rather than adding a new!).

Bonus!
1) fix bug in callback_list when removing from list
during callback, add new testcases that failed before change in this.
2) address most of comments added late to https://webrtc-review.git.corp.google.com/c/src/+/449561

Bug: webrtc:367395350
Change-Id: I131ba3ac920187bc9dec004ba55e297540ddf27c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/451521
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46976}
---
 p2p/BUILD.gn                                  |  1 +
 p2p/dtls/dtls_stun_piggyback_controller.cc    | 57 ++++++++++---------
 p2p/dtls/dtls_stun_piggyback_controller.h     | 17 ++++--
 ...dtls_stun_piggyback_controller_unittest.cc | 35 +++++++++++-
 p2p/dtls/dtls_transport.cc                    | 26 ++++-----
 p2p/dtls/dtls_transport.h                     |  3 +
 rtc_base/callback_list.cc                     |  6 ++
 rtc_base/callback_list_unittest.cc            | 48 ++++++++++++++++
 8 files changed, 149 insertions(+), 44 deletions(-)

diff --git a/p2p/BUILD.gn b/p2p/BUILD.gn
index 107919d6b5..f08c9eb7f5 100644
--- a/p2p/BUILD.gn
+++ b/p2p/BUILD.gn
@@ -678,6 +678,7 @@ rtc_library("dtls_stun_piggyback_controller") {
     "../rtc_base:logging",
     "../rtc_base:macromagic",
     "../rtc_base:stringutils",
+    "../rtc_base/network:received_packet",
     "../rtc_base/system:no_unique_address",
     "//third_party/abseil-cpp/absl/container:flat_hash_set",
     "//third_party/abseil-cpp/absl/functional:any_invocable",
diff --git a/p2p/dtls/dtls_stun_piggyback_controller.cc b/p2p/dtls/dtls_stun_piggyback_controller.cc
index 2abdcbc7f1..19a6c9a8a7 100644
--- a/p2p/dtls/dtls_stun_piggyback_controller.cc
+++ b/p2p/dtls/dtls_stun_piggyback_controller.cc
@@ -25,6 +25,7 @@
 #include "p2p/dtls/dtls_utils.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
+#include "rtc_base/network/received_packet.h"
 #include "rtc_base/strings/str_join.h"
 
 namespace webrtc {
@@ -36,25 +37,12 @@ DtlsStunPiggybackController::DtlsStunPiggybackController(
       complete_callback_(std::move(complete_callback)) {}
 
 DtlsStunPiggybackController::~DtlsStunPiggybackController() {
-  complete_callback_ = [&]() {
-    RTC_DCHECK(false) << "Calling CompleteCallback after destructor!";
-  };
 }
 
 void DtlsStunPiggybackController::SetDtlsHandshakeComplete(bool is_dtls_client,
                                                            bool is_dtls13) {
   RTC_DCHECK_RUN_ON(&sequence_checker_);
 
-  // As DTLS 1.2 server we need to keep the last flight around until
-  // we receive the post-handshake acknowledgment.
-  // As DTLS 1.2 client we have nothing more to send at this point
-  // but will continue to send ACK attributes until receiving
-  // the last flight from the server.
-  // For DTLS 1.3 this is reversed since the handshake has one round trip less.
-  if ((is_dtls_client && !is_dtls13) || (!is_dtls_client && is_dtls13)) {
-    pending_packets_.clear();
-  }
-
   // Peer does not support this so fallback to a normal DTLS handshake
   // happened.
   if (state_ == State::OFF) {
@@ -63,6 +51,23 @@ void DtlsStunPiggybackController::SetDtlsHandshakeComplete(bool is_dtls_client,
   state_ = State::PENDING;
 }
 
+void DtlsStunPiggybackController::DecryptedPacketReceived(
+    const ReceivedIpPacket& packet) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+
+  if (state_ == State::OFF) {
+    return;
+  }
+
+  RTC_DCHECK(packet.decryption_info() == ReceivedIpPacket::kDtlsDecrypted ||
+             packet.decryption_info() == ReceivedIpPacket::kSrtpEncrypted);
+
+  // We should be writable before this to happen.
+  RTC_DCHECK(state_ == State::PENDING);
+  state_ = State::COMPLETE;
+  CallCompleteCallback();
+}
+
 void DtlsStunPiggybackController::SetDtlsFailed() {
   RTC_DCHECK_RUN_ON(&sequence_checker_);
 
@@ -111,9 +116,6 @@ DtlsStunPiggybackController::GetDataToPiggyback(
   RTC_DCHECK(stun_message_type == STUN_BINDING_REQUEST ||
              stun_message_type == STUN_BINDING_RESPONSE);
 
-  // No longer writing packets...since we're now about to send them.
-  RTC_DCHECK(!writing_packets_);
-
   if (state_ == State::COMPLETE) {
     return std::nullopt;
   }
@@ -122,6 +124,9 @@ DtlsStunPiggybackController::GetDataToPiggyback(
     return std::nullopt;
   }
 
+  // No longer writing packets...since we're now about to send them.
+  RTC_DCHECK(!writing_packets_);
+
   if (pending_packets_.empty()) {
     return std::nullopt;
   }
@@ -165,8 +170,8 @@ void DtlsStunPiggybackController::ReportDataPiggybacked(
   if (state_ == State::TENTATIVE && !data.has_value() && !acks.has_value()) {
     RTC_LOG(LS_INFO) << "DTLS-STUN piggybacking not supported by peer.";
     state_ = State::OFF;
-    // TODO jonaso, webrtc:367395350: We should call CallCompleteCallback here
-    // but this causes a slew of failed tests. Investigate why!
+    // TODO: bugs.webrtc.org/367395350 - We should call CallCompleteCallback
+    // here but this causes a slew of failed tests. Investigate why!
     // CallCompleteCallback();
     return;
   }
@@ -176,8 +181,6 @@ void DtlsStunPiggybackController::ReportDataPiggybacked(
   if (state_ == State::PENDING && !data.has_value() && !acks.has_value()) {
     RTC_LOG(LS_INFO) << "DTLS-STUN piggybacking complete.";
     state_ = State::COMPLETE;
-    pending_packets_.clear();
-    handshake_messages_received_.clear();
     CallCompleteCallback();
     return;
   }
@@ -215,8 +218,6 @@ void DtlsStunPiggybackController::ReportDataPiggybacked(
   if (!data.has_value() && acks.has_value() && state_ == State::PENDING) {
     RTC_LOG(LS_INFO) << "DTLS-STUN piggybacking complete.";
     state_ = State::COMPLETE;
-    pending_packets_.clear();
-    handshake_messages_received_.clear();
     CallCompleteCallback();
     return;
   }
@@ -263,10 +264,14 @@ void DtlsStunPiggybackController::ReportDtlsPacket(
 }
 
 void DtlsStunPiggybackController::CallCompleteCallback() {
-  complete_callback_();
-  complete_callback_ = [&]() {
-    RTC_DCHECK(false) << "CompleteCallback called twice!";
-  };
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  pending_packets_.clear();
+  handshake_messages_received_.clear();
+  if (!complete_callback_) {
+    RTC_DCHECK_NOTREACHED() << "CompleteCallback called twice!";
+    return;
+  }
+  std::move(complete_callback_)();
 }
 
 }  // namespace webrtc
diff --git a/p2p/dtls/dtls_stun_piggyback_controller.h b/p2p/dtls/dtls_stun_piggyback_controller.h
index 3741bd985e..1d034dc1e0 100644
--- a/p2p/dtls/dtls_stun_piggyback_controller.h
+++ b/p2p/dtls/dtls_stun_piggyback_controller.h
@@ -21,6 +21,7 @@
 #include "api/sequence_checker.h"
 #include "api/transport/stun.h"
 #include "p2p/dtls/dtls_utils.h"
+#include "rtc_base/network/received_packet.h"
 #include "rtc_base/system/no_unique_address.h"
 #include "rtc_base/thread_annotations.h"
 
@@ -62,10 +63,16 @@ class DtlsStunPiggybackController {
     return state_;
   }
 
-  // Called by DtlsTransport when the handshake is complete per our local DTLS
-  // agent.
+  // Called by DtlsTransport when the handshake is complete "locally",
+  // i.e. we can send encrypted packets to peer (but we don't strictly know
+  // that peer can decode them).
   void SetDtlsHandshakeComplete(bool is_dtls_client, bool is_dtls13);
 
+  // Called by DtlsTransport when a packet has been received and decoded.
+  // This means that dtls is writable for the peer and that we have enough
+  // information to decode their packets.
+  void DecryptedPacketReceived(const ReceivedIpPacket& packet);
+
   // Called by DtlsTransport when DTLS failed.
   void SetDtlsFailed();
 
@@ -101,8 +108,10 @@ class DtlsStunPiggybackController {
   State state_ RTC_GUARDED_BY(sequence_checker_) = State::TENTATIVE;
   bool writing_packets_ RTC_GUARDED_BY(sequence_checker_) = false;
   PacketStash pending_packets_ RTC_GUARDED_BY(sequence_checker_);
-  absl::AnyInvocable<void(ArrayView<const uint8_t>)> dtls_data_callback_;
-  absl::AnyInvocable<void()> complete_callback_;
+  absl::AnyInvocable<void(ArrayView<const uint8_t>)> dtls_data_callback_
+      RTC_GUARDED_BY(sequence_checker_);
+  absl::AnyInvocable<void() &&> complete_callback_
+      RTC_GUARDED_BY(sequence_checker_);
 
   std::vector<uint32_t> handshake_messages_received_
       RTC_GUARDED_BY(sequence_checker_);
diff --git a/p2p/dtls/dtls_stun_piggyback_controller_unittest.cc b/p2p/dtls/dtls_stun_piggyback_controller_unittest.cc
index 7f12b65ecf..3a18f2a7b1 100644
--- a/p2p/dtls/dtls_stun_piggyback_controller_unittest.cc
+++ b/p2p/dtls/dtls_stun_piggyback_controller_unittest.cc
@@ -22,6 +22,8 @@
 #include "p2p/dtls/dtls_utils.h"
 #include "rtc_base/byte_buffer.h"
 #include "rtc_base/checks.h"
+#include "rtc_base/network/received_packet.h"
+#include "rtc_base/socket_address.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
 
@@ -57,6 +59,8 @@ const std::vector<uint8_t> dtls_flight4 = {
 
 const std::vector<uint8_t> empty = {};
 
+const std::vector<uint8_t> kPayload = {0x1, 0x2, 0x3};
+
 std::vector<uint32_t> FromAckAttribute(webrtc::ArrayView<uint8_t> attr) {
   webrtc::ByteBufferReader ack_reader(attr);
   std::vector<uint32_t> values;
@@ -111,7 +115,8 @@ class DtlsStunPiggybackControllerTest : public ::testing::Test {
             [this]() { ClientCompleteCallback(); }),
         server_(
             [this](ArrayView<const uint8_t> data) { ServerPacketSink(data); },
-            [this]() { ServerCompleteCallback(); }) {}
+            [this]() { ServerCompleteCallback(); }),
+        packet_(kPayload, SocketAddress(), std::nullopt) {}
 
   // Send from client to server embedded in STUN.
   void SendClientToServerEmbedded(const std::vector<uint8_t>& packet,
@@ -197,6 +202,8 @@ class DtlsStunPiggybackControllerTest : public ::testing::Test {
   MOCK_METHOD(void, ClientCompleteCallback, ());
   MOCK_METHOD(void, ServerCompleteCallback, ());
 
+  ReceivedIpPacket packet_;
+
  private:
   void MaybeSetHandshakeComplete(std::vector<uint8_t> packet) {
     // Note: this assumes DTLS 1.2
@@ -232,6 +239,32 @@ TEST_F(DtlsStunPiggybackControllerTest, BasicHandshake) {
   EXPECT_EQ(client_.state(), State::COMPLETE);
 }
 
+TEST_F(DtlsStunPiggybackControllerTest,
+       BasicHandshakeCompleteWithDecryptedPacket) {
+  // Flight 1+2
+  SendClientToServerEmbedded(dtls_flight1, STUN_BINDING_REQUEST);
+  EXPECT_EQ(server_.state(), State::CONFIRMED);
+  SendServerToClientEmbedded(dtls_flight2, STUN_BINDING_RESPONSE);
+  EXPECT_EQ(client_.state(), State::CONFIRMED);
+
+  // Flight 3+4
+  SendClientToServerEmbedded(dtls_flight3, STUN_BINDING_REQUEST);
+  SendServerToClientEmbedded(dtls_flight4, STUN_BINDING_RESPONSE);
+  EXPECT_EQ(server_.state(), State::PENDING);
+  EXPECT_EQ(client_.state(), State::PENDING);
+
+  // Post-handshake ACK
+  EXPECT_CALL(*this, ClientCompleteCallback);
+  client_.DecryptedPacketReceived(
+      packet_.CopyAndSet(ReceivedIpPacket::kDtlsDecrypted));
+  EXPECT_EQ(client_.state(), State::COMPLETE);
+
+  EXPECT_CALL(*this, ServerCompleteCallback);
+  server_.DecryptedPacketReceived(
+      packet_.CopyAndSet(ReceivedIpPacket::kSrtpEncrypted));
+  EXPECT_EQ(server_.state(), State::COMPLETE);
+}
+
 TEST_F(DtlsStunPiggybackControllerTest, FirstClientPacketLost) {
   // Client to server got lost (or arrives late)
   // Flight 1
diff --git a/p2p/dtls/dtls_transport.cc b/p2p/dtls/dtls_transport.cc
index 2fc2ab28a7..eda8219d3f 100644
--- a/p2p/dtls/dtls_transport.cc
+++ b/p2p/dtls/dtls_transport.cc
@@ -288,24 +288,17 @@ DtlsTransportInternalImpl::DtlsTransportInternalImpl(
           ssl_stream_factory) {}
 
 DtlsTransportInternalImpl::~DtlsTransportInternalImpl() {
-  if (downward_) {
-    downward_->SetDtlsStunPiggybackController(nullptr);
-  }
-  auto ice_transport = this->ice_transport();
-  if (ice_transport) {
-    ice_transport->ResetDtlsStunPiggybackCallbacks();
-    ice_transport->DeregisterReceivedPacketCallback(this);
-  }
+  CompleteDtlsInStun();
 }
 
 void DtlsTransportInternalImpl::CompleteDtlsInStun() {
+  dtls_in_stun_complete_ = true;
   if (downward_) {
     downward_->SetDtlsStunPiggybackController(nullptr);
   }
-  auto ice_transport = this->ice_transport();
-  if (ice_transport) {
-    ice_transport->ResetDtlsStunPiggybackCallbacks();
-  }
+  ice_transport()->ResetDtlsStunPiggybackCallbacks();
+
+  DeregisterReceivedPacketCallback(&dtls_stun_piggyback_controller_);
 }
 
 DtlsTransportState DtlsTransportInternalImpl::dtls_state() const {
@@ -525,9 +518,16 @@ bool DtlsTransportInternalImpl::SetupDtls() {
     auto downward = std::make_unique<StreamInterfaceChannel>(ice_transport());
     StreamInterfaceChannel* downward_ptr = downward.get();
 
-    if (dtls_in_stun_) {
+    if (dtls_in_stun_ && !dtls_in_stun_complete_) {
       downward_ptr->SetDtlsStunPiggybackController(
           &dtls_stun_piggyback_controller_);
+
+      RegisterReceivedPacketCallback(
+          &dtls_stun_piggyback_controller_,
+          [this](webrtc::PacketTransportInternal* transport,
+                 const ReceivedIpPacket& packet) {
+            dtls_stun_piggyback_controller_.DecryptedPacketReceived(packet);
+          });
     }
     if (ssl_stream_factory_) {
       dtls_ = ssl_stream_factory_(
diff --git a/p2p/dtls/dtls_transport.h b/p2p/dtls/dtls_transport.h
index c642525889..7d92c71971 100644
--- a/p2p/dtls/dtls_transport.h
+++ b/p2p/dtls/dtls_transport.h
@@ -357,6 +357,9 @@ class DtlsTransportInternalImpl : public DtlsTransportInternal {
   // (so that we return PIGGYBACK_ACK to client if we get STUN_BINDING_REQUEST
   // directly). Maybe disabled in SetupDtls has been called.
   bool dtls_in_stun_ = false;
+  // Has DtlsInStun Complete been run?
+  // This variable is used to prevent reinitializing after dtls-restart.
+  bool dtls_in_stun_complete_ = false;
 
   // A controller for piggybacking DTLS in STUN.
   DtlsStunPiggybackController dtls_stun_piggyback_controller_;
diff --git a/rtc_base/callback_list.cc b/rtc_base/callback_list.cc
index 8c108a5208..68adf9f229 100644
--- a/rtc_base/callback_list.cc
+++ b/rtc_base/callback_list.cc
@@ -62,6 +62,12 @@ void CallbackListReceivers::RemoveReceivers(const void* removal_tag) {
       ++first_todo;
       RTC_DCHECK_EQ(receivers_[first_remove - 1].removal_tag, removal_tag);
       --first_remove;
+    } else {
+      // When send_in_progress_ we really only iterate the list and
+      // mark elements with pending_removal_tag().
+      RTC_DCHECK(send_in_progress_);
+      RTC_DCHECK_EQ(receivers_[first_todo].removal_tag, removal_tag);
+      receivers_[first_todo++].removal_tag = pending_removal_tag();
     }
   }
 
diff --git a/rtc_base/callback_list_unittest.cc b/rtc_base/callback_list_unittest.cc
index 77e4c73b79..7c576ea39c 100644
--- a/rtc_base/callback_list_unittest.cc
+++ b/rtc_base/callback_list_unittest.cc
@@ -267,6 +267,54 @@ TEST(CallbackList, RemoveFromSend) {
   c.Send();
   EXPECT_EQ(removal_tag, 1);
 }
+
+TEST(CallbackList, RemoveFromSendFirst) {
+  int removal_tag0 = 0;
+  int removal_tag1 = 0;
+  CallbackList<> c;
+  c.AddReceiver(&removal_tag0, [&] {
+    c.RemoveReceivers(&removal_tag0);
+    // Do after RemoveReceivers to make sure the lambda is still valid.
+    ++removal_tag0;
+  });
+  c.AddReceiver(&removal_tag1, [&] {});
+  c.Send();
+  c.Send();
+  EXPECT_EQ(removal_tag0, 1);
+}
+
+TEST(CallbackList, RemoveLastFromSend) {
+  int removal_tag0 = 0;
+  int removal_tag1 = 0;
+  CallbackList<> c;
+  c.AddReceiver(&removal_tag0, [] {});
+  c.AddReceiver(&removal_tag1, [&] {
+    c.RemoveReceivers(&removal_tag1);
+    // Do after RemoveReceivers to make sure the lambda is still valid.
+    ++removal_tag1;
+  });
+  c.Send();
+  c.Send();
+  EXPECT_EQ(removal_tag1, 1);
+}
+
+TEST(CallbackList, RemoveMiddleFromSend) {
+  int removal_tag0 = 0;
+  int removal_tag1 = 0;
+  int removal_tag2 = 0;
+  CallbackList<> c;
+  c.AddReceiver(&removal_tag0, [&] {});
+  c.AddReceiver(&removal_tag1, [&] {
+    c.RemoveReceivers(&removal_tag1);
+    // Do after RemoveReceivers to make sure the lambda is still valid.
+    ++removal_tag1;
+  });
+  c.AddReceiver(&removal_tag2, [&] {});
+  c.Send();
+  c.Send();
+  EXPECT_EQ(removal_tag1, 1);
+}
+
 #pragma clang diagnostic pop
 
 }  // namespace

From 08a2baf20dfd6d5bee92aa5ee30ac3ed253d00b2 Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Mon, 23 Feb 2026 20:07:47 -0800
Subject: [PATCH 124/472] Update WebRTC code version (2026-02-24T04:07:11).

Bug: None
Change-Id: Idee74ef8b3e3556fe0e5b90150bf23b69e384d75
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/451761
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#46977}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index 94a24f266f..033d646f25 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-02-23T04:05:06";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-02-24T04:07:11";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From c60f75df0faa49fe7fa5805500f91ebd549ebec4 Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Tue, 24 Feb 2026 08:21:57 +0000
Subject: [PATCH 125/472] Delete deprecated ProtectRtp and related functions

Bug: webrtc:42224904
Change-Id: I82db82189874e252d1fceb7aef9c17a4d983be04
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/451802
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46978}
---
 pc/srtp_session.cc | 132 ---------------------------------------------
 pc/srtp_session.h  |  25 +--------
 2 files changed, 1 insertion(+), 156 deletions(-)

diff --git a/pc/srtp_session.cc b/pc/srtp_session.cc
index d1c533c91f..fa4eb89877 100644
--- a/pc/srtp_session.cc
+++ b/pc/srtp_session.cc
@@ -231,41 +231,6 @@ bool SrtpSession::ProtectRtp(CopyOnWriteBuffer& buffer) {
   return true;
 }
 
-bool SrtpSession::ProtectRtp(void* p, int in_len, int max_len, int* out_len) {
-  RTC_DCHECK(thread_checker_.IsCurrent());
-  if (!session_) {
-    RTC_LOG(LS_WARNING) << "Failed to protect SRTP packet: no SRTP Session";
-    return false;
-  }
-
-  // Note: the need_len differs from the libsrtp recommendatіon to ensure
-  // SRTP_MAX_TRAILER_LEN bytes of free space after the data. WebRTC
-  // never includes a MKI, therefore the amount of bytes added by the
-  // srtp_protect call is known in advance and depends on the cipher suite.
-  int need_len = in_len + rtp_auth_tag_len_;  // NOLINT
-  if (max_len < need_len) {
-    RTC_LOG(LS_WARNING) << "Failed to protect SRTP packet: The buffer length "
-                        << max_len << " is less than the needed " << need_len;
-    return false;
-  }
-  if (dump_plain_rtp_) {
-    DumpPacket(p, in_len, /*outbound=*/true);
-  }
-
-  *out_len = in_len;
-  int err = srtp_protect(session_, p, out_len);
-  int seq_num = ParseRtpSequenceNumber(
-      MakeArrayView(reinterpret_cast<const uint8_t*>(p), in_len));
-  if (err != srtp_err_status_ok) {
-    RTC_LOG(LS_WARNING) << "Failed to protect SRTP packet, seqnum=" << seq_num
-                        << ", err=" << err
-                        << ", last seqnum=" << last_send_seq_num_;
-    return false;
-  }
-  last_send_seq_num_ = seq_num;
-  return true;
-}
-
 bool SrtpSession::ProtectRtp(CopyOnWriteBuffer& buffer, int64_t* index) {
   if (!ProtectRtp(buffer)) {
     return false;
@@ -273,19 +238,6 @@ bool SrtpSession::ProtectRtp(CopyOnWriteBuffer& buffer, int64_t* index) {
   return (index) ? GetSendStreamPacketIndex(buffer, index) : true;
 }
 
-bool SrtpSession::ProtectRtp(void* data,
-                             int in_len,
-                             int max_len,
-                             int* out_len,
-                             int64_t* index) {
-  CopyOnWriteBuffer buffer(static_cast<uint8_t*>(data), in_len, max_len);
-  if (!ProtectRtp(buffer)) {
-    return false;
-  }
-  *out_len = buffer.size();
-  return (index) ? GetSendStreamPacketIndex(buffer, index) : true;
-}
-
 bool SrtpSession::ProtectRtcp(CopyOnWriteBuffer& buffer) {
   RTC_DCHECK(thread_checker_.IsCurrent());
   if (!session_) {
@@ -319,35 +271,6 @@ bool SrtpSession::ProtectRtcp(CopyOnWriteBuffer& buffer) {
   return true;
 }
 
-bool SrtpSession::ProtectRtcp(void* p, int in_len, int max_len, int* out_len) {
-  RTC_DCHECK(thread_checker_.IsCurrent());
-  if (!session_) {
-    RTC_LOG(LS_WARNING) << "Failed to protect SRTCP packet: no SRTP Session";
-    return false;
-  }
-
-  // Note: the need_len differs from the libsrtp recommendatіon to ensure
-  // SRTP_MAX_TRAILER_LEN bytes of free space after the data. WebRTC
-  // never includes a MKI, therefore the amount of bytes added by the
-  // srtp_protect_rtp call is known in advance and depends on the cipher suite.
-  int need_len = in_len + sizeof(uint32_t) + rtcp_auth_tag_len_;  // NOLINT
-  if (max_len < need_len) {
-    RTC_LOG(LS_WARNING) << "Failed to protect SRTCP packet: The buffer length "
-                        << max_len << " is less than the needed " << need_len;
-    return false;
-  }
-  if (dump_plain_rtp_) {
-    DumpPacket(p, in_len, /*outbound=*/true);
-  }
-
-  *out_len = in_len;
-  int err = srtp_protect_rtcp(session_, p, out_len);
-  if (err != srtp_err_status_ok) {
-    RTC_LOG(LS_WARNING) << "Failed to protect SRTCP packet, err=" << err;
-    return false;
-  }
-  return true;
-}
 
 bool SrtpSession::UnprotectRtp(CopyOnWriteBuffer& buffer) {
   RTC_DCHECK(thread_checker_.IsCurrent());
@@ -379,35 +302,6 @@ bool SrtpSession::UnprotectRtp(CopyOnWriteBuffer& buffer) {
   return true;
 }
 
-bool SrtpSession::UnprotectRtp(void* p, int in_len, int* out_len) {
-  RTC_DCHECK(thread_checker_.IsCurrent());
-  if (!session_) {
-    RTC_LOG(LS_WARNING) << "Failed to unprotect SRTP packet: no SRTP Session";
-    return false;
-  }
-
-  *out_len = in_len;
-  int err = srtp_unprotect(session_, p, out_len);
-  if (err != srtp_err_status_ok) {
-    // Limit the error logging to avoid excessive logs when there are lots of
-    // bad packets.
-    const int kFailureLogThrottleCount = 100;
-    if (decryption_failure_count_ % kFailureLogThrottleCount == 0) {
-      RTC_LOG(LS_WARNING) << "Failed to unprotect SRTP packet, err=" << err
-                          << ", previous failure count: "
-                          << decryption_failure_count_;
-    }
-    ++decryption_failure_count_;
-    RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.SrtpUnprotectError",
-                              static_cast<int>(err), kSrtpErrorCodeBoundary);
-    return false;
-  }
-  if (dump_plain_rtp_) {
-    DumpPacket(p, *out_len, /*outbound=*/false);
-  }
-  return true;
-}
-
 bool SrtpSession::UnprotectRtcp(CopyOnWriteBuffer& buffer) {
   RTC_DCHECK(thread_checker_.IsCurrent());
   if (!session_) {
@@ -430,27 +324,6 @@ bool SrtpSession::UnprotectRtcp(CopyOnWriteBuffer& buffer) {
   return true;
 }
 
-bool SrtpSession::UnprotectRtcp(void* p, int in_len, int* out_len) {
-  RTC_DCHECK(thread_checker_.IsCurrent());
-  if (!session_) {
-    RTC_LOG(LS_WARNING) << "Failed to unprotect SRTCP packet: no SRTP Session";
-    return false;
-  }
-
-  *out_len = in_len;
-  int err = srtp_unprotect_rtcp(session_, p, out_len);
-  if (err != srtp_err_status_ok) {
-    RTC_LOG(LS_WARNING) << "Failed to unprotect SRTCP packet, err=" << err;
-    RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.SrtcpUnprotectError",
-                              static_cast<int>(err), kSrtpErrorCodeBoundary);
-    return false;
-  }
-  if (dump_plain_rtp_) {
-    DumpPacket(p, *out_len, /*outbound=*/false);
-  }
-  return true;
-}
-
 bool SrtpSession::GetRtpAuthParams(uint8_t** key, int* key_len, int* tag_len) {
   RTC_DCHECK(thread_checker_.IsCurrent());
   RTC_DCHECK(IsExternalAuthActive());
@@ -685,9 +558,4 @@ void SrtpSession::DumpPacket(const CopyOnWriteBuffer& buffer, bool outbound) {
       << " # RTP_DUMP";
 }
 
-void SrtpSession::DumpPacket(const void* buf, int len, bool outbound) {
-  const CopyOnWriteBuffer buffer(static_cast<const uint8_t*>(buf), len, len);
-  DumpPacket(buffer, outbound);
-}
-
 }  // namespace webrtc
diff --git a/pc/srtp_session.h b/pc/srtp_session.h
index 18918549f1..af47b18f17 100644
--- a/pc/srtp_session.h
+++ b/pc/srtp_session.h
@@ -62,33 +62,13 @@ class SrtpSession {
 
   // Encrypts/signs an individual RTP/RTCP packet, in-place.
   // If an HMAC is used, this will increase the packet size.
-  [[deprecated("Pass CopyOnWriteBuffer")]] bool ProtectRtp(void* data,
-                                                           int in_len,
-                                                           int max_len,
-                                                           int* out_len);
-  bool ProtectRtp(CopyOnWriteBuffer& buffer);
-  // Overloaded version, outputs packet index.
-  [[deprecated("Pass CopyOnWriteBuffer")]] bool ProtectRtp(void* data,
-                                                           int in_len,
-                                                           int max_len,
-                                                           int* out_len,
-                                                           int64_t* index);
   bool ProtectRtp(CopyOnWriteBuffer& buffer, int64_t* index);
+  bool ProtectRtp(CopyOnWriteBuffer& buffer);
 
-  [[deprecated("Pass CopyOnWriteBuffer")]] bool ProtectRtcp(void* data,
-                                                            int in_len,
-                                                            int max_len,
-                                                            int* out_len);
   bool ProtectRtcp(CopyOnWriteBuffer& buffer);
   // Decrypts/verifies an invidiual RTP/RTCP packet.
   // If an HMAC is used, this will decrease the packet size.
-  [[deprecated("Pass CopyOnWriteBuffer")]] bool UnprotectRtp(void* data,
-                                                             int in_len,
-                                                             int* out_len);
   bool UnprotectRtp(CopyOnWriteBuffer& buffer);
-  [[deprecated("Pass CopyOnWriteBuffer")]] bool UnprotectRtcp(void* data,
-                                                              int in_len,
-                                                              int* out_len);
   bool UnprotectRtcp(CopyOnWriteBuffer& buffer);
 
   // Helper method to get authentication params.
@@ -136,9 +116,6 @@ class SrtpSession {
   // Writes unencrypted packets in text2pcap format to the log file
   // for debugging.
   void DumpPacket(const CopyOnWriteBuffer& buffer, bool outbound);
-  [[deprecated("Pass CopyOnWriteBuffer")]] void DumpPacket(const void* buf,
-                                                           int len,
-                                                           bool outbound);
 
   void HandleEvent(const srtp_event_data_t* ev);
   static void HandleEventThunk(srtp_event_data_t* ev);

From 26b8f63bdc6b599bd98eab16b394a424268e3ba0 Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Mon, 23 Feb 2026 15:26:28 +0000
Subject: [PATCH 126/472] Guard the remaining members of ModuleRtpRtcp2

This documents the threading properties of these members.

Bug: webrtc:486665180
Change-Id: I153630e2bb635b2c223939bc297d7fc2b652ee7a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/451300
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46979}
---
 modules/rtp_rtcp/source/rtp_rtcp_impl2.cc |  4 ++++
 modules/rtp_rtcp/source/rtp_rtcp_impl2.h  | 16 +++++++++++-----
 2 files changed, 15 insertions(+), 5 deletions(-)

diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
index 8e8b2e036c..ad76e51269 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
@@ -479,6 +479,7 @@ size_t ModuleRtpRtcpImpl2::MaxRtpPacketSize() const {
 }
 
 void ModuleRtpRtcpImpl2::SetMaxRtpPacketSize(size_t rtp_packet_size) {
+  RTC_DCHECK_RUN_ON(&rtcp_module_checker_);
   RTC_DCHECK_LE(rtp_packet_size, IP_PACKET_SIZE)
       << "rtp packet size too large: " << rtp_packet_size;
   RTC_DCHECK_GT(rtp_packet_size, packet_overhead_)
@@ -595,6 +596,7 @@ void ModuleRtpRtcpImpl2::SetTmmbn(std::vector<rtcp::TmmbItem> bounding_set) {
 // Send a Negative acknowledgment packet.
 int32_t ModuleRtpRtcpImpl2::SendNACK(const uint16_t* nack_list,
                                      const uint16_t size) {
+  RTC_DCHECK_RUN_ON(&rtcp_module_checker_);
   uint16_t nack_length = size;
   uint16_t start_id = 0;
   int64_t now_ms = env_.clock().TimeInMilliseconds();
@@ -634,6 +636,7 @@ void ModuleRtpRtcpImpl2::SendNack(
 }
 
 bool ModuleRtpRtcpImpl2::TimeToSendFullNackList(int64_t now) const {
+  RTC_DCHECK_RUN_ON(&rtcp_module_checker_);
   // Use RTT from RtcpRttStats class if provided.
   int64_t rtt = rtt_ms();
   if (rtt == 0) {
@@ -844,6 +847,7 @@ void ModuleRtpRtcpImpl2::ScheduleMaybeSendRtcpAtOrAfterTimestamp(
 }
 
 uint32_t ModuleRtpRtcpImpl2::RtcpSenderSourceSsrc() {
+  RTC_DCHECK_RUN_ON(&rtcp_module_checker_);
   uint32_t ssrc = recv_ssrc_callback_();
   // Inform the RtcpReceiver that this is now the SSRC to listen for
   rtcp_receiver_.set_local_media_ssrc(ssrc);
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl2.h b/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
index f5939b6787..8096112e84 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
@@ -348,23 +348,29 @@ class ModuleRtpRtcpImpl2 final : public RtpRtcpInterface,
 
   const Environment env_;
   TaskQueueBase* const worker_queue_;
+  // Thread checker used for guarding the IncomingRtcpPacket call.
   RTC_NO_UNIQUE_ADDRESS SequenceChecker rtcp_thread_checker_;
+  // Thread checker used for guarding member variables in this class.
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker rtcp_module_checker_;
+  // TODO: issues.webrtc.org/48665180 - figure out why these are different.
 
   // The function for getting the right SSRC for sending RTCP reports
   // Must outlive rtcp_sender_, so placed before it.
-  absl::AnyInvocable<uint32_t() const> recv_ssrc_callback_;
+  absl::AnyInvocable<uint32_t() const> recv_ssrc_callback_
+      RTC_GUARDED_BY(rtcp_module_checker_);
 
+  // These three classes contain their own thread checking.
   const std::unique_ptr<RtpSenderContext> rtp_sender_;
   RTCPSender rtcp_sender_;
   RTCPReceiver rtcp_receiver_;
 
-  uint16_t packet_overhead_;
+  uint16_t packet_overhead_ RTC_GUARDED_BY(rtcp_module_checker_);
 
   // Send side
-  int64_t nack_last_time_sent_full_ms_;
-  uint16_t nack_last_seq_number_sent_;
+  int64_t nack_last_time_sent_full_ms_ RTC_GUARDED_BY(rtcp_module_checker_);
+  uint16_t nack_last_seq_number_sent_ RTC_GUARDED_BY(rtcp_module_checker_);
 
-  RtcpRttStats* const rtt_stats_;
+  RtcpRttStats* const rtt_stats_ RTC_PT_GUARDED_BY(worker_queue_);
   RepeatingTaskHandle rtt_update_task_ RTC_GUARDED_BY(worker_queue_);
 
   // The processed RTT from RtcpRttStats.

From b4ba35203bae9b12806aa2b40d5bc3c97de31e2b Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Sat, 21 Feb 2026 19:47:13 +0100
Subject: [PATCH 127/472] Restrict ArrayView constructor from genertic range

Add constrains matching similar std::span constructor

Update classes that don't fulfill that constraints:
Classes that represent a single range got `begin` and `end` accessors.
Classes that represent more than a single range got single accessor
returning desired ArrayView.

Bug: webrtc:439801349
Change-Id: Ic0e6b4964582d91e0904ef814ce446be5e0ead2d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/451220
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46980}
---
 api/array_view.h                                           | 3 +++
 api/video/encoded_image.h                                  | 5 +++++
 .../audio_processing/agc2/rnn_vad/pitch_search_internal.cc | 7 +++++++
 modules/audio_processing/agc2/rnn_vad/rnn.cc               | 4 ++--
 modules/audio_processing/agc2/rnn_vad/rnn_fc.h             | 5 +++++
 modules/audio_processing/agc2/rnn_vad/rnn_fc_unittest.cc   | 2 +-
 modules/audio_processing/agc2/rnn_vad/rnn_gru.h            | 5 +++++
 modules/audio_processing/agc2/rnn_vad/rnn_gru_unittest.cc  | 2 +-
 modules/rtp_rtcp/source/rtp_packet.h                       | 1 +
 modules/rtp_rtcp/source/rtp_sender_egress.cc               | 2 +-
 test/layer_filtering_transport.cc                          | 2 +-
 11 files changed, 32 insertions(+), 6 deletions(-)

diff --git a/api/array_view.h b/api/array_view.h
index f67b8b971b..67976fe2de 100644
--- a/api/array_view.h
+++ b/api/array_view.h
@@ -15,6 +15,7 @@
 #include <array>
 #include <cstddef>
 #include <iterator>
+#include <ranges>
 #include <span>
 #include <type_traits>
 
@@ -259,6 +260,7 @@ class ArrayView final : public array_view_internal::ArrayViewBase<T, Size> {
                 !std::is_same_v<ArrayView, std::remove_reference_t<U>> &&
                 Size == array_view_internal::kArrayViewVarSize &&
                 HasDataAndSize<U, T>::value>* = nullptr>
+    requires(std::ranges::contiguous_range<U> && std::ranges::sized_range<U>)
   ArrayView(U& u)  // NOLINT
       : ArrayView(u.data(), u.size()) {}
 
@@ -267,6 +269,7 @@ class ArrayView final : public array_view_internal::ArrayViewBase<T, Size> {
                 !std::is_same_v<ArrayView, std::remove_reference_t<U>> &&
                 Size == array_view_internal::kArrayViewVarSize &&
                 HasDataAndSize<U, T>::value>* = nullptr>
+    requires(std::ranges::contiguous_range<U> && std::ranges::sized_range<U>)
   ArrayView(const U& u)  // NOLINT(runtime/explicit)
       : ArrayView(u.data(), u.size()) {}
 
diff --git a/api/video/encoded_image.h b/api/video/encoded_image.h
index ff25b75334..a934347c29 100644
--- a/api/video/encoded_image.h
+++ b/api/video/encoded_image.h
@@ -52,6 +52,8 @@ class EncodedImageBufferInterface : public RefCountInterface {
 // Basic implementation of EncodedImageBufferInterface.
 class RTC_EXPORT EncodedImageBuffer : public EncodedImageBufferInterface {
  public:
+  using iterator = Buffer::iterator;
+
   static scoped_refptr<EncodedImageBuffer> Create() { return Create(0); }
   static scoped_refptr<EncodedImageBuffer> Create(size_t size);
   static scoped_refptr<EncodedImageBuffer> Create(const uint8_t* data,
@@ -63,6 +65,9 @@ class RTC_EXPORT EncodedImageBuffer : public EncodedImageBufferInterface {
   size_t size() const override;
   void Realloc(size_t t);
 
+  iterator begin() { return buffer_.begin(); }
+  iterator end() { return buffer_.end(); }
+
  protected:
   explicit EncodedImageBuffer(size_t size);
   EncodedImageBuffer(const uint8_t* data, size_t size);
diff --git a/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc b/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc
index 019aedca17..084eb21b8f 100644
--- a/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc
+++ b/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc
@@ -130,7 +130,11 @@ constexpr int kMaxPitchPeriods24kHz =
 
 // Collection of inverted lags.
 class InvertedLagsIndex {
+  using Storage = std::array<int, kMaxPitchPeriods24kHz>;
+
  public:
+  using const_iterator = Storage::const_iterator;
+
   InvertedLagsIndex() : num_entries_(0) {}
   // Adds an inverted lag to the index. Cannot add more than
   // `kMaxPitchPeriods24kHz` values.
@@ -141,6 +145,9 @@ class InvertedLagsIndex {
   const int* data() const { return inverted_lags_.data(); }
   int size() const { return num_entries_; }
 
+  const_iterator begin() const { return inverted_lags_.begin(); }
+  const_iterator end() const { return begin() + size(); }
+
  private:
   std::array<int, kMaxPitchPeriods24kHz> inverted_lags_;
   int num_entries_;
diff --git a/modules/audio_processing/agc2/rnn_vad/rnn.cc b/modules/audio_processing/agc2/rnn_vad/rnn.cc
index 3b88554880..775ea86343 100644
--- a/modules/audio_processing/agc2/rnn_vad/rnn.cc
+++ b/modules/audio_processing/agc2/rnn_vad/rnn.cc
@@ -86,8 +86,8 @@ float RnnVad::ComputeVadProbability(
     return 0.f;
   }
   input_.ComputeOutput(feature_vector);
-  hidden_.ComputeOutput(input_);
-  output_.ComputeOutput(hidden_);
+  hidden_.ComputeOutput(input_.output());
+  output_.ComputeOutput(hidden_.output());
   RTC_DCHECK_EQ(output_.size(), 1);
   return output_.data()[0];
 }
diff --git a/modules/audio_processing/agc2/rnn_vad/rnn_fc.h b/modules/audio_processing/agc2/rnn_vad/rnn_fc.h
index 8f3e9b43bb..12ed2f5453 100644
--- a/modules/audio_processing/agc2/rnn_vad/rnn_fc.h
+++ b/modules/audio_processing/agc2/rnn_vad/rnn_fc.h
@@ -53,6 +53,11 @@ class FullyConnectedLayer {
   // Returns the size of the output buffer.
   int size() const { return output_size_; }
 
+  // Returns the output buffer.
+  ArrayView<const float> output() const {
+    return MakeArrayView(output_.data(), output_size_);
+  }
+
   // Computes the fully-connected layer output.
   void ComputeOutput(ArrayView<const float> input);
 
diff --git a/modules/audio_processing/agc2/rnn_vad/rnn_fc_unittest.cc b/modules/audio_processing/agc2/rnn_vad/rnn_fc_unittest.cc
index 8f86b27881..705f43aa2e 100644
--- a/modules/audio_processing/agc2/rnn_vad/rnn_fc_unittest.cc
+++ b/modules/audio_processing/agc2/rnn_vad/rnn_fc_unittest.cc
@@ -57,7 +57,7 @@ TEST_P(RnnFcParametrization, CheckFullyConnectedLayerOutput) {
                          /*cpu_features=*/GetParam(),
                          /*layer_name=*/"FC");
   fc.ComputeOutput(kFullyConnectedInputVector);
-  ExpectNearAbsolute(kFullyConnectedExpectedOutput, fc, 1e-5f);
+  ExpectNearAbsolute(kFullyConnectedExpectedOutput, fc.output(), 1e-5f);
 }
 
 TEST_P(RnnFcParametrization, DISABLED_BenchmarkFullyConnectedLayer) {
diff --git a/modules/audio_processing/agc2/rnn_vad/rnn_gru.h b/modules/audio_processing/agc2/rnn_vad/rnn_gru.h
index 5712ee4fa9..367f3cda6b 100644
--- a/modules/audio_processing/agc2/rnn_vad/rnn_gru.h
+++ b/modules/audio_processing/agc2/rnn_vad/rnn_gru.h
@@ -49,6 +49,11 @@ class GatedRecurrentLayer {
   // Returns the size of the output buffer.
   int size() const { return output_size_; }
 
+  // Returns the output buffer.
+  ArrayView<const float> output() const {
+    return MakeArrayView(state_.data(), output_size_);
+  }
+
   // Resets the GRU state.
   void Reset();
   // Computes the recurrent layer output and updates the status.
diff --git a/modules/audio_processing/agc2/rnn_vad/rnn_gru_unittest.cc b/modules/audio_processing/agc2/rnn_vad/rnn_gru_unittest.cc
index f6009e26d5..738c2219e0 100644
--- a/modules/audio_processing/agc2/rnn_vad/rnn_gru_unittest.cc
+++ b/modules/audio_processing/agc2/rnn_vad/rnn_gru_unittest.cc
@@ -47,7 +47,7 @@ void TestGatedRecurrentLayer(GatedRecurrentLayer& gru,
         input_sequence.subspan(i * gru.input_size(), gru.input_size()));
     const auto expected_output =
         expected_output_sequence.subspan(i * gru.size(), gru.size());
-    ExpectNearAbsolute(expected_output, gru, 3e-6f);
+    ExpectNearAbsolute(expected_output, gru.output(), 3e-6f);
   }
 }
 
diff --git a/modules/rtp_rtcp/source/rtp_packet.h b/modules/rtp_rtcp/source/rtp_packet.h
index 3d27520b10..9abdbc6591 100644
--- a/modules/rtp_rtcp/source/rtp_packet.h
+++ b/modules/rtp_rtcp/source/rtp_packet.h
@@ -91,6 +91,7 @@ class RtpPacket {
     return payload_offset_ + payload_size_ + padding_size_;
   }
   const uint8_t* data() const { return buffer_.cdata(); }
+  ArrayView<const uint8_t> buffer() const { return buffer_; }
   size_t FreeCapacity() const { return capacity() - size(); }
   size_t MaxPayloadSize() const { return capacity() - headers_size(); }
 
diff --git a/modules/rtp_rtcp/source/rtp_sender_egress.cc b/modules/rtp_rtcp/source/rtp_sender_egress.cc
index 2d62e194e5..c0a0ee7703 100644
--- a/modules/rtp_rtcp/source/rtp_sender_egress.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_egress.cc
@@ -441,7 +441,7 @@ bool RtpSenderEgress::SendPacketToNetwork(const RtpPacketToSend& packet,
                                           const PacketOptions& options,
                                           const PacedPacketInfo& pacing_info) {
   RTC_DCHECK_RUN_ON(worker_queue_);
-  if (transport_ == nullptr || !transport_->SendRtp(packet, options)) {
+  if (transport_ == nullptr || !transport_->SendRtp(packet.buffer(), options)) {
     RTC_LOG(LS_WARNING) << "Transport failed to send packet.";
     return false;
   }
diff --git a/test/layer_filtering_transport.cc b/test/layer_filtering_transport.cc
index 7f4dac202f..9a2996a335 100644
--- a/test/layer_filtering_transport.cc
+++ b/test/layer_filtering_transport.cc
@@ -190,7 +190,7 @@ bool LayerFilteringTransport::SendRtp(ArrayView<const uint8_t> packet,
     }
   }
 
-  return test::DirectTransport::SendRtp(rtp_packet, options);
+  return test::DirectTransport::SendRtp(rtp_packet.buffer(), options);
 }
 
 }  // namespace test

From cbb623e509ae232a36f8856992726dc5edd4ce6b Mon Sep 17 00:00:00 2001
From: Bjorn Terelius <terelius@webrtc.org>
Date: Mon, 23 Feb 2026 08:21:58 +0100
Subject: [PATCH 128/472] Fix minor mistakes in event log analyzer
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Bug: None
Change-Id: I7a0833d46f0df30d796c43822d4b7153fef81ad9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/451620
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Auto-Submit: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46981}
---
 rtc_tools/rtc_event_log_visualizer/analyze_audio.h | 4 +++-
 rtc_tools/rtc_event_log_visualizer/analyze_bwe.cc  | 4 ++--
 2 files changed, 5 insertions(+), 3 deletions(-)

diff --git a/rtc_tools/rtc_event_log_visualizer/analyze_audio.h b/rtc_tools/rtc_event_log_visualizer/analyze_audio.h
index 271e1d8c18..db306f813f 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyze_audio.h
+++ b/rtc_tools/rtc_event_log_visualizer/analyze_audio.h
@@ -18,6 +18,7 @@
 #include <string>
 
 #include "absl/strings/string_view.h"
+#include "api/field_trials_view.h"
 #include "api/function_view.h"
 #include "api/neteq/neteq.h"
 #include "logging/rtc_event_log/rtc_event_log_parser.h"
@@ -83,7 +84,8 @@ class LazyNetEqSimulator {
 NetEqStatsGetterMap SimulateNetEq(const ParsedRtcEventLog& parsed_log,
                                   const AnalyzerConfig& config,
                                   absl::string_view replacement_file_name,
-                                  int file_sample_rate_hz);
+                                  int file_sample_rate_hz,
+                                  const FieldTrialsView* field_trials);
 
 void CreateAudioJitterBufferGraph(const ParsedRtcEventLog& parsed_log,
                                   const AnalyzerConfig& config,
diff --git a/rtc_tools/rtc_event_log_visualizer/analyze_bwe.cc b/rtc_tools/rtc_event_log_visualizer/analyze_bwe.cc
index 14fc0255e9..d0cf941983 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyze_bwe.cc
+++ b/rtc_tools/rtc_event_log_visualizer/analyze_bwe.cc
@@ -893,7 +893,7 @@ void CreateSendSideBweSimulationGraph(const ParsedRtcEventLog& parsed_log,
                                PointStyle::kHighlight);
   TimeSeries robust_time_series("Robust throughput estimate", LineStyle::kLine,
                                 PointStyle::kHighlight);
-  TimeSeries acked_estimate_time_series("Ackednowledged bitrate estimate",
+  TimeSeries acked_estimate_time_series("Acknowledged bitrate estimate",
                                         LineStyle::kLine,
                                         PointStyle::kHighlight);
 
@@ -1191,7 +1191,7 @@ void CreatePacerDelayGraph(const ParsedRtcEventLog& parsed_log,
 
     if (packets.size() < 2) {
       RTC_LOG(LS_WARNING)
-          << "Can't estimate a the RTP clock frequency or the "
+          << "Can't estimate the RTP clock frequency or the "
              "pacer delay with less than 2 packets in the stream";
       continue;
     }

From 10e303c4f26cb3f2790893205a44b52d3fd539df Mon Sep 17 00:00:00 2001
From: Markus Handell <handellm@webrtc.org>
Date: Tue, 24 Feb 2026 11:15:09 +0100
Subject: [PATCH 129/472] Add and populate content type in VideoFrame.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This change adds a content type optional field to the VideoFrame.
It's populated in generic decoder to make the content type
available in Chromium's MediaStreamRemoteVideoSource, for the
purposes of identifying screenshare content.

Bug: chromium:470337728
Change-Id: I3ceed08c9903376372c929e9eb3b1c68abd6a097
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/451840
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46982}
---
 api/video/video_frame.cc                      | 10 ++++-
 api/video/video_frame.h                       | 17 +++++++-
 modules/video_coding/generic_decoder.cc       |  2 +-
 .../video_coding/generic_decoder_unittest.cc  | 40 +++++++++++++++++++
 4 files changed, 65 insertions(+), 4 deletions(-)

diff --git a/api/video/video_frame.cc b/api/video/video_frame.cc
index 85816ed930..ec97aff57b 100644
--- a/api/video/video_frame.cc
+++ b/api/video/video_frame.cc
@@ -19,6 +19,7 @@
 #include "api/scoped_refptr.h"
 #include "api/units/timestamp.h"
 #include "api/video/color_space.h"
+#include "api/video/video_content_type.h"
 #include "api/video/video_frame_buffer.h"
 #include "api/video/video_rotation.h"
 #include "rtc_base/checks.h"
@@ -174,7 +175,8 @@ VideoFrame VideoFrame::Builder::build() {
   return VideoFrame(id_, video_frame_buffer_, timestamp_us_,
                     presentation_timestamp_, reference_time_, timestamp_rtp_,
                     ntp_time_ms_, rotation_, color_space_, render_parameters_,
-                    update_rect_, packet_infos_, is_repeat_frame_);
+                    update_rect_, packet_infos_, is_repeat_frame_,
+                    content_type_);
 }
 
 VideoFrame::Builder& VideoFrame::Builder::set_video_frame_buffer(
@@ -270,6 +272,12 @@ VideoFrame::Builder& VideoFrame::Builder::set_is_repeat_frame(
   return *this;
 }
 
+VideoFrame::Builder& VideoFrame::Builder::set_content_type(
+    VideoContentType content_type) {
+  content_type_ = content_type;
+  return *this;
+}
+
 VideoFrame::VideoFrame(const scoped_refptr<VideoFrameBuffer>& buffer,
                        VideoRotation rotation,
                        int64_t timestamp_us)
diff --git a/api/video/video_frame.h b/api/video/video_frame.h
index ef054ba407..268d5e7e0b 100644
--- a/api/video/video_frame.h
+++ b/api/video/video_frame.h
@@ -21,6 +21,7 @@
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
 #include "api/video/color_space.h"
+#include "api/video/video_content_type.h"
 #include "api/video/video_frame_buffer.h"
 #include "api/video/video_rotation.h"
 #include "rtc_base/checks.h"
@@ -125,6 +126,7 @@ class RTC_EXPORT VideoFrame {
     Builder& set_update_rect(const std::optional<UpdateRect>& update_rect);
     Builder& set_packet_infos(RtpPacketInfos packet_infos);
     Builder& set_is_repeat_frame(bool is_repeat_frame);
+    Builder& set_content_type(VideoContentType content_type);
 
    private:
     uint16_t id_ = kNotSetId;
@@ -140,6 +142,7 @@ class RTC_EXPORT VideoFrame {
     std::optional<UpdateRect> update_rect_;
     RtpPacketInfos packet_infos_;
     bool is_repeat_frame_ = false;
+    std::optional<VideoContentType> content_type_;
   };
 
   // To be deprecated. Migrate all use to Builder.
@@ -291,6 +294,11 @@ class RTC_EXPORT VideoFrame {
     is_repeat_frame_ = is_repeat_frame;
   }
 
+  std::optional<VideoContentType> content_type() const { return content_type_; }
+  void set_content_type(std::optional<VideoContentType> content_type) {
+    content_type_ = content_type;
+  }
+
  private:
   VideoFrame(uint16_t id,
              const scoped_refptr<VideoFrameBuffer>& buffer,
@@ -304,7 +312,8 @@ class RTC_EXPORT VideoFrame {
              const RenderParameters& render_parameters,
              const std::optional<UpdateRect>& update_rect,
              RtpPacketInfos packet_infos,
-             bool is_repeat_frame)
+             bool is_repeat_frame,
+             std::optional<VideoContentType> content_type)
       : id_(id),
         video_frame_buffer_(buffer),
         timestamp_rtp_(timestamp_rtp),
@@ -317,7 +326,8 @@ class RTC_EXPORT VideoFrame {
         render_parameters_(render_parameters),
         update_rect_(update_rect),
         packet_infos_(std::move(packet_infos)),
-        is_repeat_frame_(is_repeat_frame) {}
+        is_repeat_frame_(is_repeat_frame),
+        content_type_(content_type) {}
 
   uint16_t id_;
   // An opaque reference counted handle that stores the pixel data.
@@ -355,6 +365,9 @@ class RTC_EXPORT VideoFrame {
   // in cases where a capturer is using a variable frame rate and stops
   // producing frames when nothing has changed.
   bool is_repeat_frame_;
+  // The content type of the video frame. This represents the mode in which
+  // the video frame was encoded by the remote peer (if signaled).
+  std::optional<VideoContentType> content_type_;
 };
 
 }  // namespace webrtc
diff --git a/modules/video_coding/generic_decoder.cc b/modules/video_coding/generic_decoder.cc
index 0cf98a40f8..2320dfdffc 100644
--- a/modules/video_coding/generic_decoder.cc
+++ b/modules/video_coding/generic_decoder.cc
@@ -237,9 +237,9 @@ void VCMDecodedFrameCallback::Decoded(VideoFrame& decodedImage,
       "WebRTC.Video.GenericDecoder.DecodeDelay",
       timing_frame_info.decode_finish_ms - timing_frame_info.decode_start_ms);
   timing_->SetTimingFrameInfo(timing_frame_info);
-
   decodedImage.set_timestamp_us(
       frame_info->render_time ? frame_info->render_time->us() : -1);
+  decodedImage.set_content_type(frame_info->content_type);
   receive_callback_->OnFrameToRender({.video_frame = decodedImage,
                                       .qp = qp,
                                       .decode_time = decode_time,
diff --git a/modules/video_coding/generic_decoder_unittest.cc b/modules/video_coding/generic_decoder_unittest.cc
index 081c9ba4fa..7c083459c7 100644
--- a/modules/video_coding/generic_decoder_unittest.cc
+++ b/modules/video_coding/generic_decoder_unittest.cc
@@ -273,6 +273,46 @@ TEST_F(GenericDecoderTest, UsesMappedColorSpaceIfSet) {
   EXPECT_EQ(decoded_frame->color_space(), kMappedColorSpace);
 }
 
+TEST_F(GenericDecoderTest, SetsScreenshareContentTypeIfSetInFrameInfo) {
+  constexpr uint32_t kRtpTimestamp = 1;
+  FrameInfo frame_info;
+  frame_info.rtp_timestamp = kRtpTimestamp;
+  frame_info.decode_start = Timestamp::Zero();
+  frame_info.content_type = VideoContentType::SCREENSHARE;
+  frame_info.frame_type = VideoFrameType::kVideoFrameKey;
+
+  VideoFrame video_frame = VideoFrame::Builder()
+                               .set_video_frame_buffer(I420Buffer::Create(5, 5))
+                               .set_rtp_timestamp(kRtpTimestamp)
+                               .build();
+  vcm_callback_.Map(std::move(frame_info));
+  vcm_callback_.Decoded(video_frame);
+
+  std::optional<VideoFrame> decoded_frame = user_callback_.PopLastFrame();
+  ASSERT_TRUE(decoded_frame.has_value());
+  EXPECT_EQ(decoded_frame->content_type(), VideoContentType::SCREENSHARE);
+}
+
+TEST_F(GenericDecoderTest, SetsUnspecifiedContentTypeIfSetInFrameInfo) {
+  constexpr uint32_t kRtpTimestamp = 1;
+  FrameInfo frame_info;
+  frame_info.rtp_timestamp = kRtpTimestamp;
+  frame_info.decode_start = Timestamp::Zero();
+  frame_info.content_type = VideoContentType::UNSPECIFIED;
+  frame_info.frame_type = VideoFrameType::kVideoFrameKey;
+
+  VideoFrame video_frame = VideoFrame::Builder()
+                               .set_video_frame_buffer(I420Buffer::Create(5, 5))
+                               .set_rtp_timestamp(kRtpTimestamp)
+                               .build();
+  vcm_callback_.Map(std::move(frame_info));
+  vcm_callback_.Decoded(video_frame);
+
+  std::optional<VideoFrame> decoded_frame = user_callback_.PopLastFrame();
+  ASSERT_TRUE(decoded_frame.has_value());
+  EXPECT_EQ(decoded_frame->content_type(), VideoContentType::UNSPECIFIED);
+}
+
 TEST_F(GenericDecoderTest, UsesDecoderColorSpaceIfNoneMapped) {
   constexpr uint32_t kRtpTimestamp = 1;
   const ColorSpace kDecoderColorSpace(

From 1185f13f75eb78d1074994d72d346af1051982fa Mon Sep 17 00:00:00 2001
From: Philipp Hancke <philipp.hancke@googlemail.com>
Date: Tue, 24 Feb 2026 12:06:14 +0100
Subject: [PATCH 130/472] rtc_base: add text2pcap helper function

which dumps the content of an array view in text2pcap format so it can
be extracted from the logs and viewed with wireshark.

The existing variants in net/dcsctp/ and pc/srtp_session are updated to
use this helper.

Bug: None
Change-Id: Ibdf3c2c7a4e67c19028610d8551415d24f5de8ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/451380
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46983}
---
 net/dcsctp/public/BUILD.gn                    |  1 +
 .../public/text_pcap_packet_observer.cc       | 36 ++++------------
 pc/BUILD.gn                                   |  1 +
 pc/srtp_session.cc                            | 21 +++-------
 rtc_base/BUILD.gn                             | 16 +++++++
 rtc_base/text2pcap.cc                         | 42 +++++++++++++++++++
 rtc_base/text2pcap.h                          | 38 +++++++++++++++++
 7 files changed, 112 insertions(+), 43 deletions(-)
 create mode 100644 rtc_base/text2pcap.cc
 create mode 100644 rtc_base/text2pcap.h

diff --git a/net/dcsctp/public/BUILD.gn b/net/dcsctp/public/BUILD.gn
index 7e946a5f13..a6eb3df4f1 100644
--- a/net/dcsctp/public/BUILD.gn
+++ b/net/dcsctp/public/BUILD.gn
@@ -76,6 +76,7 @@ rtc_library("utils") {
     "../../../api:array_view",
     "../../../rtc_base:logging",
     "../../../rtc_base:stringutils",
+    "../../../rtc_base:text2pcap",
     "//third_party/abseil-cpp/absl/strings:string_view",
   ]
   sources = [
diff --git a/net/dcsctp/public/text_pcap_packet_observer.cc b/net/dcsctp/public/text_pcap_packet_observer.cc
index 467112dd9a..7fed9717c6 100644
--- a/net/dcsctp/public/text_pcap_packet_observer.cc
+++ b/net/dcsctp/public/text_pcap_packet_observer.cc
@@ -11,47 +11,29 @@
 
 #include <cstdint>
 
-#include "absl/strings/string_view.h"
 #include "api/array_view.h"
 #include "net/dcsctp/public/types.h"
 #include "rtc_base/logging.h"
-#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/text2pcap.h"
 
 namespace dcsctp {
 
 void TextPcapPacketObserver::OnSentPacket(
     dcsctp::TimeMs now,
     webrtc::ArrayView<const uint8_t> payload) {
-  PrintPacket("O ", name_, now, payload);
+  RTC_LOG(LS_VERBOSE) << "\n"
+                      << webrtc::Text2Pcap::DumpPacket(/*outbound=*/true,
+                                                       payload, *now)
+                      << " # SCTP_PACKET " << name_;
 }
 
 void TextPcapPacketObserver::OnReceivedPacket(
     dcsctp::TimeMs now,
     webrtc::ArrayView<const uint8_t> payload) {
-  PrintPacket("I ", name_, now, payload);
-}
-
-void TextPcapPacketObserver::PrintPacket(
-    absl::string_view prefix,
-    absl::string_view socket_name,
-    dcsctp::TimeMs now,
-    webrtc::ArrayView<const uint8_t> payload) {
-  webrtc::StringBuilder s;
-  s << "\n" << prefix;
-  int64_t remaining = *now % (24 * 60 * 60 * 1000);
-  int hours = remaining / (60 * 60 * 1000);
-  remaining = remaining % (60 * 60 * 1000);
-  int minutes = remaining / (60 * 1000);
-  remaining = remaining % (60 * 1000);
-  int seconds = remaining / 1000;
-  int ms = remaining % 1000;
-  s.AppendFormat("%02d:%02d:%02d.%03d", hours, minutes, seconds, ms);
-  s << " 0000";
-  for (uint8_t byte : payload) {
-    s.AppendFormat(" %02x", byte);
-  }
-  s << " # SCTP_PACKET " << socket_name;
-  RTC_LOG(LS_VERBOSE) << s.str();
+  RTC_LOG(LS_VERBOSE) << "\n"
+                      << webrtc::Text2Pcap::DumpPacket(/*outbound=*/false,
+                                                       payload, *now)
+                      << " # SCTP_PACKET " << name_;
 }
 
 }  // namespace dcsctp
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 3c2aee65c5..2c187836fc 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -710,6 +710,7 @@ rtc_library("srtp_session") {
     "../rtc_base:macromagic",
     "../rtc_base:ssl_adapter",
     "../rtc_base:stringutils",
+    "../rtc_base:text2pcap",
     "../rtc_base:timeutils",
     "../rtc_base/synchronization:mutex",
     "../system_wrappers:metrics",
diff --git a/pc/srtp_session.cc b/pc/srtp_session.cc
index fa4eb89877..59b840fd5d 100644
--- a/pc/srtp_session.cc
+++ b/pc/srtp_session.cc
@@ -12,7 +12,6 @@
 
 #include <cstdint>
 #include <cstring>
-#include <iomanip>
 #include <vector>
 
 #include "absl/strings/string_view.h"
@@ -27,8 +26,8 @@
 #include "rtc_base/ip_address.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/ssl_stream_adapter.h"
-#include "rtc_base/string_encode.h"
 #include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/text2pcap.h"
 #include "rtc_base/thread_annotations.h"
 #include "rtc_base/time_utils.h"
 #include "system_wrappers/include/metrics.h"
@@ -542,20 +541,10 @@ void SrtpSession::HandleEventThunk(srtp_event_data_t* ev) {
 // The resulting file can be replayed using the WebRTC video_replay tool and
 // be inspected in Wireshark using the RTP, VP8 and H264 dissectors.
 void SrtpSession::DumpPacket(const CopyOnWriteBuffer& buffer, bool outbound) {
-  int64_t time_of_day = TimeUTCMillis() % (24 * 3600 * 1000);
-  int64_t hours = time_of_day / (3600 * 1000);
-  int64_t minutes = (time_of_day / (60 * 1000)) % 60;
-  int64_t seconds = (time_of_day / 1000) % 60;
-  int64_t millis = time_of_day % 1000;
-  RTC_LOG(LS_VERBOSE)
-      << "\n"
-      << (outbound ? "O" : "I") << " " << std::setfill('0') << std::setw(2)
-      << hours << ":" << std::setfill('0') << std::setw(2) << minutes << ":"
-      << std::setfill('0') << std::setw(2) << seconds << "."
-      << std::setfill('0') << std::setw(3) << millis << " " << "000000 "
-      << hex_encode_with_delimiter(
-             absl::string_view(buffer.data<char>(), buffer.size()), ' ')
-      << " # RTP_DUMP";
+  RTC_LOG(LS_VERBOSE) << "\n"
+                      << Text2Pcap::DumpPacket(outbound, buffer,
+                                               TimeUTCMillis())
+                      << " # RTP_DUMP";
 }
 
 }  // namespace webrtc
diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
index 4a0c8cb9c5..7e1bcc362c 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -650,6 +650,22 @@ rtc_library("stringutils") {
   ]
 }
 
+rtc_library("text2pcap") {
+  visibility = [ "*" ]
+  sources = [
+    "text2pcap.cc",
+    "text2pcap.h",
+  ]
+  deps = [
+    ":logging",
+    ":stringutils",
+    ":timeutils",
+    "../api:array_view",
+    "system:rtc_export",
+    "//third_party/abseil-cpp/absl/strings:string_view",
+  ]
+}
+
 rtc_source_set("type_traits") {
   sources = [ "type_traits.h" ]
 }
diff --git a/rtc_base/text2pcap.cc b/rtc_base/text2pcap.cc
new file mode 100644
index 0000000000..74f8448faa
--- /dev/null
+++ b/rtc_base/text2pcap.cc
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2026 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "rtc_base/text2pcap.h"
+
+#include <cstdint>
+#include <string>
+
+#include "api/array_view.h"
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/time_utils.h"
+
+namespace webrtc {
+
+std::string Text2Pcap::DumpPacket(bool outbound,
+                                  ArrayView<const uint8_t> payload,
+                                  int64_t timestamp_ms) {
+  webrtc::StringBuilder s;
+  s << "\n" << (outbound ? "O " : "I ");
+
+  int64_t remaining = timestamp_ms % (24 * 60 * 60 * kNumMillisecsPerSec);
+  int hours = remaining / (24 * 60 * 60 * kNumMillisecsPerSec);
+  remaining = remaining % (60 * 60 * kNumMillisecsPerSec);
+  int minutes = remaining / (60 * 60 * kNumMillisecsPerSec);
+  remaining = remaining % (60 * kNumMillisecsPerSec);
+  int seconds = remaining / kNumMillisecsPerSec;
+  int ms = remaining % kNumMillisecsPerSec;
+  s.AppendFormat("%02d:%02d:%02d.%03d", hours, minutes, seconds, ms);
+  s << " 0000";
+  for (uint8_t byte : payload) {
+    s.AppendFormat(" %02x", byte);
+  }
+  return s.str();
+}
+
+}  // namespace webrtc
diff --git a/rtc_base/text2pcap.h b/rtc_base/text2pcap.h
new file mode 100644
index 0000000000..1f384ec7ed
--- /dev/null
+++ b/rtc_base/text2pcap.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2026 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef RTC_BASE_TEXT2PCAP_H_
+#define RTC_BASE_TEXT2PCAP_H_
+
+#include <cstdint>
+#include <string>
+
+#include "api/array_view.h"
+
+namespace webrtc {
+
+class Text2Pcap {
+ public:
+  // Dumps the packet in text2pcap format, returning the formatted string.
+  // The format is described on
+  // https://www.wireshark.org/docs/man-pages/text2pcap.html
+  // and resulting logs can be turned into a PCAP file that can be opened
+  // with the Wireshark tool using a command line along the lines off
+  //   text2pcap -D -u 1000,2000 -t %H:%M:%S.%f log.txt out.pcap
+  // Returns the text2pcap formatted log which is typically prefixed with a
+  // newline and has a grep-able suffix (e.g. ` # SCTP_PACKET` or ` # RTP_DUMP`)
+  // for easy extraction from logs.
+  static std::string DumpPacket(bool outbound,
+                                ArrayView<const uint8_t> payload,
+                                int64_t timestamp_ms);
+};
+
+}  // namespace webrtc
+
+#endif  // RTC_BASE_TEXT2PCAP_H_

From 92ec44dcdf15cb24d34b3f761965b39d2b44cf56 Mon Sep 17 00:00:00 2001
From: Jakob Ivarsson <jakobi@webrtc.org>
Date: Tue, 24 Feb 2026 12:17:51 +0000
Subject: [PATCH 131/472] Make GetNackList take optional TimeDelta.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Bug: webrtc:42223979
Change-Id: I35c1b3a270f5b689bd1eb4ec46d7eae004f87e42
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/451920
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46984}
---
 audio/channel_receive.cc       |  4 +--
 audio/nack_tracker.cc          | 12 +++----
 audio/nack_tracker.h           |  4 +--
 audio/nack_tracker_unittest.cc | 66 +++++++++++++++++-----------------
 4 files changed, 43 insertions(+), 43 deletions(-)

diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index 255880c24f..2eb4a6202a 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -686,12 +686,10 @@ void ChannelReceive::OnRtpPacket(const RtpPacketReceived& packet) {
   packet_copy.set_payload_type_frequency(it->second);
   if (nack_tracker_) {
     nack_tracker_->UpdateSampleRate(it->second);
-    TimeDelta round_trip_time =
-        rtp_rtcp_->LastRtt().value_or(TimeDelta::Zero());
     nack_tracker_->UpdateLastReceivedPacket(packet.SequenceNumber(),
                                             packet.Timestamp());
     std::vector<uint16_t> nack_list =
-        nack_tracker_->GetNackList(round_trip_time.ms());
+        nack_tracker_->GetNackList(rtp_rtcp_->LastRtt());
     if (!nack_list.empty()) {
       rtp_rtcp_->SendNACK(nack_list.data(), nack_list.size());
     }
diff --git a/audio/nack_tracker.cc b/audio/nack_tracker.cc
index d0e51c39a5..9878cd0292 100644
--- a/audio/nack_tracker.cc
+++ b/audio/nack_tracker.cc
@@ -208,19 +208,19 @@ int64_t NackTracker::TimeToPlay(uint32_t timestamp) const {
 }
 
 // We don't erase elements with time-to-play shorter than round-trip-time.
-std::vector<uint16_t> NackTracker::GetNackList(int64_t round_trip_time_ms) {
-  RTC_DCHECK_GE(round_trip_time_ms, 0);
+std::vector<uint16_t> NackTracker::GetNackList(
+    std::optional<TimeDelta> round_trip_time) {
   std::vector<uint16_t> sequence_numbers;
-  if (round_trip_time_ms == 0) {
+  if (!round_trip_time.has_value()) {
     if (config_.require_valid_rtt) {
-      return sequence_numbers;
+      return {};
     } else {
-      round_trip_time_ms = config_.default_rtt_ms;
+      round_trip_time = TimeDelta::Millis(config_.default_rtt_ms);
     }
   }
   for (NackList::const_iterator it = nack_list_.begin(); it != nack_list_.end();
        ++it) {
-    if (Nack(it->second, round_trip_time_ms)) {
+    if (Nack(it->second, round_trip_time->ms())) {
       sequence_numbers.push_back(it->first);
     }
   }
diff --git a/audio/nack_tracker.h b/audio/nack_tracker.h
index 63a90cfffc..81772eaaaf 100644
--- a/audio/nack_tracker.h
+++ b/audio/nack_tracker.h
@@ -78,11 +78,11 @@ class NackTracker {
   void UpdateLastReceivedPacket(uint16_t sequence_number, uint32_t timestamp);
 
   // Get a list of "missing" packets which have expected time-to-play larger
-  // than the given round-trip-time (in milliseconds).
+  // than the given round-trip-time.
   // Note: Late packets are not included.
   // Calling this method multiple times may give different results, since the
   // internal nack list may get flushed if never_nack_multiple_times_ is true.
-  std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms);
+  std::vector<uint16_t> GetNackList(std::optional<TimeDelta> round_trip_time);
 
   // Reset to default values. The NACK list is cleared.
   // `max_nack_list_size_` preserves its value.
diff --git a/audio/nack_tracker_unittest.cc b/audio/nack_tracker_unittest.cc
index 9c4699329d..1ca22f70bd 100644
--- a/audio/nack_tracker_unittest.cc
+++ b/audio/nack_tracker_unittest.cc
@@ -14,9 +14,11 @@
 #include <cstddef>
 #include <cstdint>
 #include <memory>
+#include <optional>
 #include <vector>
 
 #include "api/field_trials.h"
+#include "api/units/time_delta.h"
 #include "test/create_test_field_trials.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
@@ -30,7 +32,7 @@ using ::testing::IsEmpty;
 constexpr int kSampleRateHz = 16000;
 constexpr int kPacketSizeMs = 30;
 constexpr uint32_t kTimestampIncrement = 480;  // 30 ms.
-constexpr int64_t kShortRoundTripTimeMs = 1;
+constexpr TimeDelta kShortRoundTripTime = TimeDelta::Zero();
 
 bool IsNackListCorrect(const std::vector<uint16_t>& nack_list,
                        const uint16_t* lost_sequence_numbers,
@@ -69,10 +71,10 @@ TEST(NackTrackerTest, EmptyListWhenNoPacketLoss) {
   std::vector<uint16_t> nack_list;
   for (int n = 0; n < 100; n++) {
     nack.UpdateLastReceivedPacket(seq_num, timestamp);
-    nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+    nack_list = nack.GetNackList(kShortRoundTripTime);
     seq_num++;
     timestamp += kTimestampIncrement;
-    nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+    nack_list = nack.GetNackList(kShortRoundTripTime);
     EXPECT_TRUE(nack_list.empty());
   }
 }
@@ -99,7 +101,7 @@ TEST(NackTrackerTest, LatePacketsMovedToNackThenNackListDoesNotChange) {
     std::vector<uint16_t> nack_list;
 
     nack.UpdateLastReceivedPacket(seq_num, timestamp);
-    nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+    nack_list = nack.GetNackList(kShortRoundTripTime);
     EXPECT_TRUE(nack_list.empty());
 
     seq_num = sequence_num_lost_packets[kNumAllLostPackets - 1] + 1;
@@ -107,7 +109,7 @@ TEST(NackTrackerTest, LatePacketsMovedToNackThenNackListDoesNotChange) {
     int num_lost_packets = std::max(0, kNumAllLostPackets);
 
     nack.UpdateLastReceivedPacket(seq_num, timestamp);
-    nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+    nack_list = nack.GetNackList(kShortRoundTripTime);
     EXPECT_TRUE(IsNackListCorrect(nack_list, sequence_num_lost_packets,
                                   num_lost_packets));
     seq_num++;
@@ -116,7 +118,7 @@ TEST(NackTrackerTest, LatePacketsMovedToNackThenNackListDoesNotChange) {
 
     for (int n = 0; n < 20; ++n) {
       nack.UpdateLastReceivedPacket(seq_num, timestamp);
-      nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+      nack_list = nack.GetNackList(kShortRoundTripTime);
       EXPECT_TRUE(IsNackListCorrect(nack_list, sequence_num_lost_packets,
                                     kNumAllLostPackets));
       seq_num++;
@@ -145,7 +147,7 @@ TEST(NackTrackerTest, ArrivedPacketsAreRemovedFromNackList) {
     uint32_t timestamp = 0;
 
     nack.UpdateLastReceivedPacket(seq_num, timestamp);
-    std::vector<uint16_t> nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+    std::vector<uint16_t> nack_list = nack.GetNackList(kShortRoundTripTime);
     EXPECT_TRUE(nack_list.empty());
 
     size_t index_retransmitted_rtp = 0;
@@ -162,7 +164,7 @@ TEST(NackTrackerTest, ArrivedPacketsAreRemovedFromNackList) {
         num_lost_packets--;
 
       nack.UpdateLastReceivedPacket(seq_num, timestamp);
-      nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+      nack_list = nack.GetNackList(kShortRoundTripTime);
       EXPECT_TRUE(IsNackListCorrect(
           nack_list, &sequence_num_lost_packets[index_retransmitted_rtp],
           num_lost_packets));
@@ -176,7 +178,7 @@ TEST(NackTrackerTest, ArrivedPacketsAreRemovedFromNackList) {
       index_retransmitted_rtp++;
       timestamp_retransmitted_rtp += kTimestampIncrement;
 
-      nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+      nack_list = nack.GetNackList(kShortRoundTripTime);
       EXPECT_TRUE(IsNackListCorrect(
           nack_list, &sequence_num_lost_packets[index_retransmitted_rtp],
           num_lost_packets - 1));  // One less lost packet in the list.
@@ -273,21 +275,21 @@ TEST(NackTrackerTest,
                                   seq_num * kTimestampIncrement);
 
     const size_t kExpectedListSize = kNumLostPackets;
-    std::vector<uint16_t> nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+    std::vector<uint16_t> nack_list = nack.GetNackList(kShortRoundTripTime);
     EXPECT_EQ(kExpectedListSize, nack_list.size());
 
     for (int k = 0; k < 2; ++k) {
       // Decoding of the first and the second arrived packets.
       for (int n = 0; n < kPacketSizeMs / 10; ++n) {
         nack.UpdateLastDecodedPacket(k * kTimestampIncrement);
-        nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+        nack_list = nack.GetNackList(kShortRoundTripTime);
         EXPECT_EQ(kExpectedListSize, nack_list.size());
       }
     }
 
     // Decoding of the last received packet.
     nack.UpdateLastDecodedPacket(seq_num * kTimestampIncrement);
-    nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+    nack_list = nack.GetNackList(kShortRoundTripTime);
     EXPECT_TRUE(nack_list.empty());
 
     // Make sure list of late packets is also empty. To check that, push few
@@ -297,7 +299,7 @@ TEST(NackTrackerTest,
       seq_num++;
       nack.UpdateLastReceivedPacket(seq_num_offset + seq_num,
                                     seq_num * kTimestampIncrement);
-      nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+      nack_list = nack.GetNackList(kShortRoundTripTime);
       EXPECT_TRUE(nack_list.empty());
     }
   }
@@ -320,11 +322,11 @@ TEST(NackTrackerTest, Reset) {
   nack.UpdateLastReceivedPacket(seq_num, seq_num * kTimestampIncrement);
 
   const size_t kExpectedListSize = kNumLostPackets;
-  std::vector<uint16_t> nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+  std::vector<uint16_t> nack_list = nack.GetNackList(kShortRoundTripTime);
   EXPECT_EQ(kExpectedListSize, nack_list.size());
 
   nack.Reset();
-  nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+  nack_list = nack.GetNackList(kShortRoundTripTime);
   EXPECT_TRUE(nack_list.empty());
 }
 
@@ -348,7 +350,7 @@ TEST(NackTrackerTest, ListSizeAppliedFromBeginning) {
     timestamp += (num_lost_packets + 1) * kTimestampIncrement;
     nack.UpdateLastReceivedPacket(seq_num, timestamp);
 
-    std::vector<uint16_t> nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+    std::vector<uint16_t> nack_list = nack.GetNackList(kShortRoundTripTime);
     EXPECT_EQ(kNackListSize, nack_list.size());
   }
 }
@@ -378,12 +380,12 @@ TEST(NackTrackerTest, ChangeOfListSizeAppliedAndOldElementsRemoved) {
     nack.UpdateLastReceivedPacket(seq_num, timestamp);
     size_t expected_size = num_lost_packets;
 
-    std::vector<uint16_t> nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+    std::vector<uint16_t> nack_list = nack.GetNackList(kShortRoundTripTime);
     EXPECT_EQ(expected_size, nack_list.size());
 
     nack.SetMaxNackListSize(kNackListSize);
     expected_size = kNackListSize;
-    nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+    nack_list = nack.GetNackList(kShortRoundTripTime);
     EXPECT_TRUE(IsNackListCorrect(
         nack_list, &seq_num_lost[num_lost_packets - kNackListSize],
         expected_size));
@@ -394,7 +396,7 @@ TEST(NackTrackerTest, ChangeOfListSizeAppliedAndOldElementsRemoved) {
       timestamp += kTimestampIncrement;
       nack.UpdateLastReceivedPacket(seq_num, timestamp);
       --expected_size;
-      nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+      nack_list = nack.GetNackList(kShortRoundTripTime);
       EXPECT_TRUE(IsNackListCorrect(
           nack_list, &seq_num_lost[num_lost_packets - kNackListSize + n],
           expected_size));
@@ -404,7 +406,7 @@ TEST(NackTrackerTest, ChangeOfListSizeAppliedAndOldElementsRemoved) {
     ++seq_num;
     timestamp += kTimestampIncrement;
     nack.UpdateLastReceivedPacket(seq_num, timestamp);
-    nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+    nack_list = nack.GetNackList(kShortRoundTripTime);
     EXPECT_TRUE(nack_list.empty());
   }
 }
@@ -434,7 +436,7 @@ TEST(NackTrackerTest, RoudTripTimeIsApplied) {
   // sequence number:  1,  2,  3,   4,   5
   // time-to-play:    20, 50, 80, 110, 140
   //
-  std::vector<uint16_t> nack_list = nack.GetNackList(100);
+  std::vector<uint16_t> nack_list = nack.GetNackList(TimeDelta::Millis(100));
   ASSERT_EQ(2u, nack_list.size());
   EXPECT_EQ(4, nack_list[0]);
   EXPECT_EQ(5, nack_list[1]);
@@ -462,13 +464,13 @@ TEST(NackTrackerTest, DoNotNackMultipleTimes) {
   timestamp += (1 + kNumLostPackets) * kTimestampIncrement;
   nack.UpdateLastReceivedPacket(seq_num, timestamp);
 
-  std::vector<uint16_t> nack_list = nack.GetNackList(10);
+  std::vector<uint16_t> nack_list = nack.GetNackList(kShortRoundTripTime);
   ASSERT_EQ(3u, nack_list.size());
   EXPECT_EQ(1, nack_list[0]);
   EXPECT_EQ(2, nack_list[1]);
   EXPECT_EQ(3, nack_list[2]);
   // When we get the nack list again, it should be empty.
-  std::vector<uint16_t> nack_list2 = nack.GetNackList(10);
+  std::vector<uint16_t> nack_list2 = nack.GetNackList(kShortRoundTripTime);
   EXPECT_TRUE(nack_list2.empty());
 }
 
@@ -509,11 +511,11 @@ TEST(NackTrackerTest, DoNotNackAfterDtx) {
   uint16_t seq_num = 0;
   uint32_t timestamp = 0x87654321;
   nack.UpdateLastReceivedPacket(seq_num, timestamp);
-  EXPECT_TRUE(nack.GetNackList(0).empty());
+  EXPECT_TRUE(nack.GetNackList(std::nullopt).empty());
   constexpr int kDtxPeriod = 400;
   nack.UpdateLastReceivedPacket(seq_num + 2,
                                 timestamp + kDtxPeriod * kSampleRateHz / 1000);
-  EXPECT_TRUE(nack.GetNackList(0).empty());
+  EXPECT_TRUE(nack.GetNackList(std::nullopt).empty());
 }
 
 TEST(NackTrackerTest, DoNotNackIfLossRateIsTooHigh) {
@@ -538,7 +540,7 @@ TEST(NackTrackerTest, DoNotNackIfLossRateIsTooHigh) {
   }
   // Expect 50% loss rate which is higher that the configured maximum 40%.
   EXPECT_NEAR(nack.GetPacketLossRateForTest(), 1 << 29, (1 << 30) / 100);
-  EXPECT_TRUE(nack.GetNackList(0).empty());
+  EXPECT_TRUE(nack.GetNackList(std::nullopt).empty());
 }
 
 TEST(NackTrackerTest, OnlyNackIfRttIsValid) {
@@ -560,8 +562,8 @@ TEST(NackTrackerTest, OnlyNackIfRttIsValid) {
   add_packet(true);
   add_packet(false);
   add_packet(true);
-  EXPECT_TRUE(nack.GetNackList(0).empty());
-  EXPECT_FALSE(nack.GetNackList(10).empty());
+  EXPECT_TRUE(nack.GetNackList(std::nullopt).empty());
+  EXPECT_FALSE(nack.GetNackList(kShortRoundTripTime).empty());
 }
 
 TEST(NackTrackerTest, FixedDelayMode) {
@@ -575,18 +577,18 @@ TEST(NackTrackerTest, FixedDelayMode) {
   nack.UpdateLastReceivedPacket(seq_num, timestamp);
   nack.UpdateLastReceivedPacket(seq_num + 2,
                                 timestamp + 2 * kTimestampIncrement);
-  EXPECT_THAT(nack.GetNackList(10), ElementsAre(seq_num + 1));
+  EXPECT_THAT(nack.GetNackList(kShortRoundTripTime), ElementsAre(seq_num + 1));
   // The RTT is larger than the fixed delay, so no packets should be NACKed.
-  EXPECT_THAT(nack.GetNackList(1000), IsEmpty());
+  EXPECT_THAT(nack.GetNackList(TimeDelta::Seconds(1)), IsEmpty());
   // Decoding a packet should not affect the NACK list in fixed delay mode.
   nack.UpdateLastDecodedPacket(timestamp + 2 * kTimestampIncrement);
-  EXPECT_THAT(nack.GetNackList(10), ElementsAre(seq_num + 1));
+  EXPECT_THAT(nack.GetNackList(kShortRoundTripTime), ElementsAre(seq_num + 1));
 
   // Update the latest received packet such that the lost packet is older than
   // the fixed delay.
   nack.UpdateLastReceivedPacket(
       seq_num + 3, timestamp + 2 * kTimestampIncrement + kSampleRateHz);
-  EXPECT_THAT(nack.GetNackList(10), IsEmpty());
+  EXPECT_THAT(nack.GetNackList(kShortRoundTripTime), IsEmpty());
 }
 
 }  // namespace webrtc

From f5440d383dfaad680587ce342d47517d4d11532a Mon Sep 17 00:00:00 2001
From: Philipp Hancke <philipp.hancke@googlemail.com>
Date: Tue, 24 Feb 2026 13:20:36 +0100
Subject: [PATCH 132/472] dtls: return number of acked packets

which allows counting non-duplicate acks in the unit tests

Bug: webrtc:367395350
Change-Id: If637fce9fc08fe03901878608491c34a33c6c177
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/450981
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#46985}
---
 p2p/dtls/dtls_utils.cc          | 6 ++++--
 p2p/dtls/dtls_utils.h           | 3 ++-
 p2p/dtls/dtls_utils_unittest.cc | 3 ++-
 3 files changed, 8 insertions(+), 4 deletions(-)

diff --git a/p2p/dtls/dtls_utils.cc b/p2p/dtls/dtls_utils.cc
index 28f9826274..8744d6ef1e 100644
--- a/p2p/dtls/dtls_utils.cc
+++ b/p2p/dtls/dtls_utils.cc
@@ -10,6 +10,7 @@
 
 #include "p2p/dtls/dtls_utils.h"
 
+#include <cstddef>
 #include <cstdint>
 #include <memory>
 #include <vector>
@@ -77,9 +78,9 @@ void PacketStash::Add(ArrayView<const uint8_t> packet) {
        .buffer = std::make_unique<Buffer>(packet.data(), packet.size())});
 }
 
-void PacketStash::Prune(const absl::flat_hash_set<uint32_t>& hashes) {
+size_t PacketStash::Prune(const absl::flat_hash_set<uint32_t>& hashes) {
   if (hashes.empty()) {
-    return;
+    return 0;
   }
   uint32_t before = packets_.size();
   std::erase_if(packets_,
@@ -92,6 +93,7 @@ void PacketStash::Prune(const absl::flat_hash_set<uint32_t>& hashes) {
   if (pos_ >= packets_.size()) {
     pos_ = packets_.empty() ? 0 : packets_.size() - 1;
   }
+  return removed;
 }
 
 void PacketStash::Prune(uint32_t max_size) {
diff --git a/p2p/dtls/dtls_utils.h b/p2p/dtls/dtls_utils.h
index c433d28f1c..3a60159562 100644
--- a/p2p/dtls/dtls_utils.h
+++ b/p2p/dtls/dtls_utils.h
@@ -37,7 +37,8 @@ class PacketStash {
 
   void Add(ArrayView<const uint8_t> packet);
   bool AddIfUnique(ArrayView<const uint8_t> packet);
-  void Prune(const absl::flat_hash_set<uint32_t>& packet_hashes);
+  // Returns number of elements that were removed.
+  size_t Prune(const absl::flat_hash_set<uint32_t>& packet_hashes);
   void Prune(uint32_t max_size);
   ArrayView<const uint8_t> GetNext();
   std::vector<ArrayView<const uint8_t>> GetAll() const;
diff --git a/p2p/dtls/dtls_utils_unittest.cc b/p2p/dtls/dtls_utils_unittest.cc
index 19c1f57bbe..5cd3011fad 100644
--- a/p2p/dtls/dtls_utils_unittest.cc
+++ b/p2p/dtls/dtls_utils_unittest.cc
@@ -175,7 +175,8 @@ TEST(PacketStash, PruneSome) {
   absl::flat_hash_set<uint32_t> remove;
   remove.insert(PacketStash::Hash(packet1));
   remove.insert(PacketStash::Hash(packet2));
-  stash.Prune(remove);
+  EXPECT_EQ(stash.Prune(remove), 2u);
+  EXPECT_EQ(stash.Prune(remove), 0u);
 
   EXPECT_EQ(ToVector(stash.GetNext()), packet3);
 }

From ab0115bacd8136c14f82417c93e4a8103d602da9 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Tue, 24 Feb 2026 06:07:06 -0800
Subject: [PATCH 133/472] Roll chromium_revision 80fdbfe134..acca47dc0a
 (1588711:1589354)

Change log: https://chromium.googlesource.com/chromium/src/+log/80fdbfe134..acca47dc0a
Full diff: https://chromium.googlesource.com/chromium/src/+/80fdbfe134..acca47dc0a

Changed dependencies
* siso_version: git_revision:54be8bcbdd069237406d476362aeb261d6deb8c0..git_revision:8aacae89cf77656164b64fd9d24b3edb884b88ac
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/31ac3ef412..9002579929
* src/buildtools/linux64: git_revision:2c775ed71ed9c7ceb11823ced47a1a2640883403..git_revision:bfffe163a23e55b944a3b4d5b9ce0ded126f414a
* src/buildtools/mac: git_revision:2c775ed71ed9c7ceb11823ced47a1a2640883403..git_revision:bfffe163a23e55b944a3b4d5b9ce0ded126f414a
* src/buildtools/win: git_revision:2c775ed71ed9c7ceb11823ced47a1a2640883403..git_revision:bfffe163a23e55b944a3b4d5b9ce0ded126f414a
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/c68ca0692e..918d9fd6ed
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/45ac53e274..3ca5ab0699
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/7361a78642..1feaf237f7
* src/third_party/androidx/cipd: DLUFuZ1z4MFDK54yAzN2lScwHZlUoJssu_cnrpDbqwYC..OwcKmOzxpFqUmowRcCL4njbWYzvLYwF19przW4elBvQC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/3d72439f88..5e15537e65
* src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/02ae8c1e75..f0b7ec7780
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/8fc02b1d3e..8aad286386
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/667cabe3aa..00cc497d32
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/c485bf9b09..305f951f5f
* src/third_party/libvpx/source/libvpx: https://chromium.googlesource.com/webm/libvpx.git/+log/a0ac06f5d3..db312f06fb
* src/third_party/llvm-build/Release+Asserts: Linux_x64/clang-llvmorg-23-init-3706-gfc648683-1.tar.xz,Linux_x64/clang-tidy-llvmorg-23-init-3706-gfc648683-1.tar.xz,Linux_x64/clangd-llvmorg-23-init-3706-gfc648683-1.tar.xz,Linux_x64/llvm-code-coverage-llvmorg-23-init-3706-gfc648683-1.tar.xz,Linux_x64/llvmobjdump-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac/clang-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac/clang-mac-runtime-library-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac/clang-tidy-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac/clangd-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac/llvm-code-coverage-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac/llvmobjdump-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac_arm64/clang-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac_arm64/clang-tidy-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac_arm64/clangd-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac_arm64/llvm-code-coverage-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac_arm64/llvmobjdump-llvmorg-23-init-3706-gfc648683-1.tar.xz,Win/clang-llvmorg-23-init-3706-gfc648683-1.tar.xz,Win/clang-tidy-llvmorg-23-init-3706-gfc648683-1.tar.xz,Win/clang-win-runtime-library-llvmorg-23-init-3706-gfc648683-1.tar.xz,Win/clangd-llvmorg-23-init-3706-gfc648683-1.tar.xz,Win/llvm-code-coverage-llvmorg-23-init-3706-gfc648683-1.tar.xz,Win/llvmobjdump-llvmorg-23-init-3706-gfc648683-1.tar.xz..Linux_x64/clang-llvmorg-23-init-4965-g686acf63-1.tar.xz,Linux_x64/clang-tidy-llvmorg-23-init-4965-g686acf63-1.tar.xz,Linux_x64/clangd-llvmorg-23-init-4965-g686acf63-1.tar.xz,Linux_x64/llvm-code-coverage-llvmorg-23-init-4965-g686acf63-1.tar.xz,Linux_x64/llvmobjdump-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac/clang-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac/clang-mac-runtime-library-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac/clang-tidy-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac/clangd-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac/llvm-code-coverage-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac/llvmobjdump-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac_arm64/clang-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac_arm64/clang-tidy-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac_arm64/clangd-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac_arm64/llvm-code-coverage-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac_arm64/llvmobjdump-llvmorg-23-init-4965-g686acf63-1.tar.xz,Win/clang-llvmorg-23-init-4965-g686acf63-1.tar.xz,Win/clang-tidy-llvmorg-23-init-4965-g686acf63-1.tar.xz,Win/clang-win-runtime-library-llvmorg-23-init-4965-g686acf63-1.tar.xz,Win/clangd-llvmorg-23-init-4965-g686acf63-1.tar.xz,Win/llvm-code-coverage-llvmorg-23-init-4965-g686acf63-1.tar.xz,Win/llvmobjdump-llvmorg-23-init-4965-g686acf63-1.tar.xz
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/601cac04b0..84f98435cf
* src/third_party/rust-toolchain: Linux_x64/rust-toolchain-7dc2e92b83be02dc07f87be7e94266d6e48e5ca5-1-llvmorg-23-init-3706-gfc648683.tar.xz,Mac/rust-toolchain-7dc2e92b83be02dc07f87be7e94266d6e48e5ca5-1-llvmorg-23-init-3706-gfc648683.tar.xz,Mac_arm64/rust-toolchain-7dc2e92b83be02dc07f87be7e94266d6e48e5ca5-1-llvmorg-23-init-3706-gfc648683.tar.xz,Win/rust-toolchain-7dc2e92b83be02dc07f87be7e94266d6e48e5ca5-1-llvmorg-23-init-3706-gfc648683.tar.xz..Linux_x64/rust-toolchain-c78a29473a68f07012904af11c92ecffa68fcc75-1-llvmorg-23-init-4965-g686acf63.tar.xz,Mac/rust-toolchain-c78a29473a68f07012904af11c92ecffa68fcc75-1-llvmorg-23-init-4965-g686acf63.tar.xz,Mac_arm64/rust-toolchain-c78a29473a68f07012904af11c92ecffa68fcc75-1-llvmorg-23-init-4965-g686acf63.tar.xz,Win/rust-toolchain-c78a29473a68f07012904af11c92ecffa68fcc75-1-llvmorg-23-init-4965-g686acf63.tar.xz
* src/third_party/siso/cipd: git_revision:54be8bcbdd069237406d476362aeb261d6deb8c0..git_revision:8aacae89cf77656164b64fd9d24b3edb884b88ac
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/6580288c8d..5dfffd1f75
DEPS diff: https://chromium.googlesource.com/chromium/src/+/80fdbfe134..acca47dc0a/DEPS

BUG=None

Change-Id: Ic461fc51b71d1e4b0400d0bee2dc9c472e87ae1e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/451960
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#46986}
---
 DEPS | 244 +++++++++++++++++++++++++++++------------------------------
 1 file changed, 122 insertions(+), 122 deletions(-)

diff --git a/DEPS b/DEPS
index e535f97b5b..45f6efa106 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '80fdbfe13417d357d71d5cd8c24ccf4af6e1df70',
+  'chromium_revision': 'acca47dc0a9e6971232cf29c4482cc410740f278',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -50,7 +50,7 @@ vars = {
   # reclient CIPD package version
   'reclient_version': 're_client_version:0.185.0.db415f21-gomaip',
   # siso CIPD package version.
-  'siso_version': 'git_revision:54be8bcbdd069237406d476362aeb261d6deb8c0',
+  'siso_version': 'git_revision:8aacae89cf77656164b64fd9d24b3edb884b88ac',
 
   # ninja CIPD package.
   'ninja_package': 'infra/3pp/tools/ninja/',
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@31ac3ef412f947513c552c7e81f860feb4de2ce0',
+    'https://chromium.googlesource.com/chromium/src/build@9002579929e246c1ff22c7015a2bdd6170c968a5',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@451e71432ac9f6a91d4d5604c29e0a5078e8c2e8',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,19 +78,19 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@c68ca0692ee0da564f287410ab28ce926586445d',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@918d9fd6eddba5b7c380a1635e1baabd1a02134f',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@45ac53e274c7fae5f221cca2b3d75e7328f071a2',
+    'https://chromium.googlesource.com/chromium/src/testing@3ca5ab0699d20e6260f10f07e65f3e7b1b83cc86',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@7361a7864218287d9a9e29af059e7a7d6c6356cc',
+    'https://chromium.googlesource.com/chromium/src/third_party@1feaf237f73d66a536361b817cacd59b58a4df55',
 
   'src/buildtools/linux64': {
     'packages': [
       {
         'package': 'gn/gn/linux-${{arch}}',
-        'version': 'git_revision:2c775ed71ed9c7ceb11823ced47a1a2640883403',
+        'version': 'git_revision:bfffe163a23e55b944a3b4d5b9ce0ded126f414a',
       }
     ],
     'dep_type': 'cipd',
@@ -100,7 +100,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/mac-${{arch}}',
-        'version': 'git_revision:2c775ed71ed9c7ceb11823ced47a1a2640883403',
+        'version': 'git_revision:bfffe163a23e55b944a3b4d5b9ce0ded126f414a',
       }
     ],
     'dep_type': 'cipd',
@@ -110,7 +110,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/windows-amd64',
-        'version': 'git_revision:2c775ed71ed9c7ceb11823ced47a1a2640883403',
+        'version': 'git_revision:bfffe163a23e55b944a3b4d5b9ce0ded126f414a',
       }
     ],
     'dep_type': 'cipd',
@@ -136,157 +136,157 @@ deps = {
     'objects': [
       {
         # The Android libclang_rt.builtins libraries are currently only included in the Linux clang package.
-        'object_name': 'Linux_x64/clang-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': '9a5dfd23f6d37b21709327aa39562225c01a058cc209d1c629448d6d6d874ab2',
-        'size_bytes': 57775784,
-        'generation': 1770870606052091,
+        'object_name': 'Linux_x64/clang-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': '6d4796631177f3c070923242601a4e14425ffdc2ee84c95ce630c0198d755637',
+        'size_bytes': 57997052,
+        'generation': 1771832273904291,
         'condition': '(host_os == "linux" or checkout_android) and non_git_source',
       },
       {
-        'object_name': 'Linux_x64/clang-tidy-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': 'cef6e9f2876cc28956213fd1cfb1431ed6bd9aca135b12ebd33e9d393595009d',
-        'size_bytes': 14411824,
-        'generation': 1770870606077571,
+        'object_name': 'Linux_x64/clang-tidy-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': '250f932ba1195eaa1ab8fd6e50f3c650ff3b74d28127d982ee79b96c4844bfb7',
+        'size_bytes': 14392496,
+        'generation': 1771832274267259,
         'condition': 'host_os == "linux" and checkout_clang_tidy and non_git_source',
       },
       {
-        'object_name': 'Linux_x64/clangd-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': 'efccb1da6950b4a83857ad5243196dd9d4cefee5869362abcc9d9422b059fff2',
-        'size_bytes': 14638444,
-        'generation': 1770870606182500,
+        'object_name': 'Linux_x64/clangd-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': '39258f20af9d459b8b032ee6bd38a79ae764fd950b79bace3227b8965d58d189',
+        'size_bytes': 14618644,
+        'generation': 1771832274470726,
         'condition': 'host_os == "linux" and checkout_clangd and non_git_source',
       },
       {
-        'object_name': 'Linux_x64/llvm-code-coverage-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': '7d1648506f753035e0c96f9727abf58f0a275d8057be6fd3c231bf80ac60b2ac',
-        'size_bytes': 2326952,
-        'generation': 1770870606310079,
+        'object_name': 'Linux_x64/llvm-code-coverage-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': 'cf2fd97359cc982f56b0beb7e9abff5a4799a12ff8c3e22728ddf06e1053b867',
+        'size_bytes': 2330268,
+        'generation': 1771832275108482,
         'condition': 'host_os == "linux" and checkout_clang_coverage_tools and non_git_source',
       },
       {
-        'object_name': 'Linux_x64/llvmobjdump-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': '103ba738fb6ebcd290e38458fdec8fa7b95bc3216af1e60ad260bc065b49d447',
-        'size_bytes': 5798432,
-        'generation': 1770870606263889,
+        'object_name': 'Linux_x64/llvmobjdump-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': '720a9cbb8745d4ac4c3f887bc1f004fed44cfccb851adebaa765a1517531c952',
+        'size_bytes': 5786008,
+        'generation': 1771832274626197,
         'condition': '((checkout_linux or checkout_mac or checkout_android) and host_os == "linux") and non_git_source',
       },
       {
-        'object_name': 'Mac/clang-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': '0230c7c9a625ceb2d2c650d995fb9cb5cdb6ab7c0c05042f560cc0cda441a2d0',
-        'size_bytes': 54682532,
-        'generation': 1770870608103936,
+        'object_name': 'Mac/clang-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': '189a5bca90f83e2dc9671fef5f149386452c215a532e94cdc98cc13461411fb2',
+        'size_bytes': 54754892,
+        'generation': 1771832277222794,
         'condition': 'host_os == "mac" and host_cpu == "x64"',
       },
       {
-        'object_name': 'Mac/clang-mac-runtime-library-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': 'a7af5d119637f67bff7f390923a78a712b1067157c019aebf2dcddbc4ed7b640',
-        'size_bytes': 1015272,
-        'generation': 1770870616072374,
+        'object_name': 'Mac/clang-mac-runtime-library-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': '3a0177f1cf5f88f00e86006cb9be42f5d86a8dc461d03daf2919dee2c632f619',
+        'size_bytes': 1012728,
+        'generation': 1771832301108033,
         'condition': 'checkout_mac and not host_os == "mac"',
       },
       {
-        'object_name': 'Mac/clang-tidy-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': '89bd1e39dc4114183a8cd931a9f5b3dc4fff5f697a133d91da2ecd2f82221774',
-        'size_bytes': 14261468,
-        'generation': 1770870608107521,
+        'object_name': 'Mac/clang-tidy-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': 'c8a3f088c4d8bbb92eba2036bb68d1640c9c54ef7ec4686b4e2a5c49ff9b6bc3',
+        'size_bytes': 14282996,
+        'generation': 1771832277199632,
         'condition': 'host_os == "mac" and host_cpu == "x64" and checkout_clang_tidy',
       },
       {
-        'object_name': 'Mac/clangd-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': 'd8c9c8f9a4e62422f2167b283da1e32d39978804543f7eaa08b903d4a0c580a2',
-        'size_bytes': 15425576,
-        'generation': 1770870608113120,
+        'object_name': 'Mac/clangd-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': '4746a11e14fe96a00024a7b7d7b4ee1d61c8960171578efad263d1c54cfa6901',
+        'size_bytes': 15449448,
+        'generation': 1771832277304839,
         'condition': 'host_os == "mac" and host_cpu == "x64" and checkout_clangd',
       },
       {
-        'object_name': 'Mac/llvm-code-coverage-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': 'a705f42d6e441950594986c0afb3f7b2e65cafa10eb1a5f40dc43c0a5d127af6',
-        'size_bytes': 2372548,
-        'generation': 1770870608249587,
+        'object_name': 'Mac/llvm-code-coverage-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': 'fbaa78a2d70a0872c2e0fd94cbe9f13c61b3077c6ba73f76a34413510a420ed0',
+        'size_bytes': 2373980,
+        'generation': 1771832277813116,
         'condition': 'host_os == "mac" and host_cpu == "x64" and checkout_clang_coverage_tools',
       },
       {
-        'object_name': 'Mac/llvmobjdump-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': '556b8abf3ec08c42769986ea8940ac7f09fa2cfe49328a2ccccee4641efce6c4',
-        'size_bytes': 5703228,
-        'generation': 1770870608115133,
+        'object_name': 'Mac/llvmobjdump-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': 'f091b4127cd59792c2f278535b1a30ae2e0953c6e896f31a5b5c7beff707e388',
+        'size_bytes': 5693600,
+        'generation': 1771832277787419,
         'condition': 'host_os == "mac" and host_cpu == "x64"',
       },
       {
-        'object_name': 'Mac_arm64/clang-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': '34102ff942cb0d7e2163fada5aeb53f5c2438d07d8caf5ed79970b133adc04be',
-        'size_bytes': 45849908,
-        'generation': 1770870617609650,
+        'object_name': 'Mac_arm64/clang-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': '4a091561d8ee6cfff59efffecbfc0122d233d23427e1cdbcee5080253bfc81f9',
+        'size_bytes': 45966912,
+        'generation': 1771832302845435,
         'condition': 'host_os == "mac" and host_cpu == "arm64"',
       },
       {
-        'object_name': 'Mac_arm64/clang-tidy-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': '11dfe174b262142ce8d2d9e810ff2a5c5ac5e44ccb135876825fc3ff24be1b35',
-        'size_bytes': 12473944,
-        'generation': 1770870617763889,
+        'object_name': 'Mac_arm64/clang-tidy-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': '9248646928a75768d682b484e3a83f481152b8edce5f71fa6e8052d9ae210398',
+        'size_bytes': 12473436,
+        'generation': 1771832303183995,
         'condition': 'host_os == "mac" and host_cpu == "arm64" and checkout_clang_tidy',
       },
       {
-        'object_name': 'Mac_arm64/clangd-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': 'a9d8847208d6411087464fb9d2f07d1a9a7617821ade0b6e00fb31e29934e12f',
-        'size_bytes': 12883052,
-        'generation': 1770870617857412,
+        'object_name': 'Mac_arm64/clangd-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': 'e5fde137e7372fedc8e2c4a8e4f9fe718c3bcc63c76acaa0c4a859b04e3f4762',
+        'size_bytes': 12882364,
+        'generation': 1771832303644261,
         'condition': 'host_os == "mac" and host_cpu == "arm64" and checkout_clangd',
       },
       {
-        'object_name': 'Mac_arm64/llvm-code-coverage-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': 'bf15d654a27132d07c5a03d6bb174f79449a88a34a08b9024e72a1b88127bdd0',
-        'size_bytes': 1991372,
-        'generation': 1770870618097417,
+        'object_name': 'Mac_arm64/llvm-code-coverage-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': 'd4e9d19346ef021be42a27b367efcdcfee03cc19bbceddb9ea4f92619ef8b06c',
+        'size_bytes': 1993468,
+        'generation': 1771832304157004,
         'condition': 'host_os == "mac" and host_cpu == "arm64" and checkout_clang_coverage_tools',
       },
       {
-        'object_name': 'Mac_arm64/llvmobjdump-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': 'aea68efe7aeab8eb4d2371e79e7dcb021e1448d8edd0d4851fa34907b93205a6',
-        'size_bytes': 5446112,
-        'generation': 1770870617921094,
+        'object_name': 'Mac_arm64/llvmobjdump-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': 'b01bce55530dbca2d6d25fa14ddbac8259184fdca1cc63365130953b9ebb517f',
+        'size_bytes': 5442884,
+        'generation': 1771832303592136,
         'condition': 'host_os == "mac" and host_cpu == "arm64"',
       },
       {
-        'object_name': 'Win/clang-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': 'cca114f624db433c2ba6458a6de34e6e474bea379f483817dd5dc0a02c59df3c',
-        'size_bytes': 49244312,
-        'generation': 1770870628650736,
+        'object_name': 'Win/clang-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': '84c13e593b1532c0cf0b8fe0928482e6eea96408846a679ce68e53c90911c33e',
+        'size_bytes': 49494200,
+        'generation': 1771832332047169,
         'condition': 'host_os == "win"',
       },
       {
-        'object_name': 'Win/clang-tidy-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': 'de37a5f6e43f7d4ff7c7c8fe0ba7cd89bdd0cbe0370b171263f36ed372a7e3d0',
-        'size_bytes': 14464644,
-        'generation': 1770870628637623,
+        'object_name': 'Win/clang-tidy-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': '2a5072bfa4a3eb9dd60321d955d6d4744b7b9f2f657f32dbc706774282787e9a',
+        'size_bytes': 14455216,
+        'generation': 1771832332462256,
         'condition': 'host_os == "win" and checkout_clang_tidy',
       },
       {
-        'object_name': 'Win/clang-win-runtime-library-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': '3f2ad3f2adb92384ecdcb33385aa8beacd7f38d27bed6179e39b2e971469d028',
-        'size_bytes': 2595252,
-        'generation': 1770870636929948,
+        'object_name': 'Win/clang-win-runtime-library-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': 'a9ddbb29b61eaac9f777b1fb532eebb49ab9e546bcd680129fcc7a2fd648cd27',
+        'size_bytes': 2596776,
+        'generation': 1771832355675158,
         'condition': 'checkout_win and not host_os == "win"',
       },
       {
-        'object_name': 'Win/clangd-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': 'b9f868e6be509a1900e7a8157cd29235245e0847804abb8d20fbc37772fbf802',
-        'size_bytes': 14883160,
-        'generation': 1770870628678782,
+        'object_name': 'Win/clangd-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': '5dad22f5da22e268e43476a28689ec7b9fb8d41ae55e53911a26e9c25a9c4c4b',
+        'size_bytes': 14884728,
+        'generation': 1771832332829491,
         'condition': 'host_os == "win" and checkout_clangd',
       },
       {
-        'object_name': 'Win/llvm-code-coverage-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': '30cb9e52d92731ed24cf08775aeb5eb0e4d2c1eb871daed1f344f6b4e635fe23',
-        'size_bytes': 2477228,
-        'generation': 1770870628999104,
+        'object_name': 'Win/llvm-code-coverage-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': 'f5a46dbdf979e31d35dd12db37d714bc06a36957883f3c11d727d56382465079',
+        'size_bytes': 2478920,
+        'generation': 1771832333093363,
         'condition': 'host_os == "win" and checkout_clang_coverage_tools',
       },
       {
-        'object_name': 'Win/llvmobjdump-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': '7baba8d463b92d16c510d67be60eed89897ba76898121ea709264cbb41066af1',
-        'size_bytes': 5859968,
-        'generation': 1770870628750202,
+        'object_name': 'Win/llvmobjdump-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': '0519107fdef9c730bb4f073c87dbd03995fd2da074aabef86347f62fb2b784f3',
+        'size_bytes': 5838628,
+        'generation': 1771832332695387,
         'condition': '(checkout_linux or checkout_mac or checkout_android) and host_os == "win"',
       },
     ]
@@ -298,31 +298,31 @@ deps = {
     'bucket': 'chromium-browser-clang',
     'objects': [
       {
-        'object_name': 'Linux_x64/rust-toolchain-7dc2e92b83be02dc07f87be7e94266d6e48e5ca5-1-llvmorg-23-init-3706-gfc648683.tar.xz',
-        'sha256sum': 'c14a211a9fcef604df768db22a4422553c791c4ad7e3f982cf00b9c0c745ccaa',
-        'size_bytes': 266703680,
-        'generation': 1770870599310471,
+        'object_name': 'Linux_x64/rust-toolchain-c78a29473a68f07012904af11c92ecffa68fcc75-1-llvmorg-23-init-4965-g686acf63.tar.xz',
+        'sha256sum': 'e3f380e162218de33aeea559dd7696431fd505b47b9669422faea83b47b16de4',
+        'size_bytes': 270144912,
+        'generation': 1771832265918901,
         'condition': 'host_os == "linux" and non_git_source',
       },
       {
-        'object_name': 'Mac/rust-toolchain-7dc2e92b83be02dc07f87be7e94266d6e48e5ca5-1-llvmorg-23-init-3706-gfc648683.tar.xz',
-        'sha256sum': 'dfd5f890b8d84eeee4cf5aa3ba9b9e2e070bb5bece134f979060c35b9286ed00',
-        'size_bytes': 254634176,
-        'generation': 1770870600934341,
+        'object_name': 'Mac/rust-toolchain-c78a29473a68f07012904af11c92ecffa68fcc75-1-llvmorg-23-init-4965-g686acf63.tar.xz',
+        'sha256sum': '01a4c3b5a3be52b5ae2a99e78541c512cd198b1a87baa1bc1c0b4697428d8fb2',
+        'size_bytes': 257817220,
+        'generation': 1771832267985133,
         'condition': 'host_os == "mac" and host_cpu == "x64"',
       },
       {
-        'object_name': 'Mac_arm64/rust-toolchain-7dc2e92b83be02dc07f87be7e94266d6e48e5ca5-1-llvmorg-23-init-3706-gfc648683.tar.xz',
-        'sha256sum': '4e0a8de0ef6e65d17f723193c44c7dc6495f2274e7a7818e9b84e9645281be7e',
-        'size_bytes': 238113184,
-        'generation': 1770870602823865,
+        'object_name': 'Mac_arm64/rust-toolchain-c78a29473a68f07012904af11c92ecffa68fcc75-1-llvmorg-23-init-4965-g686acf63.tar.xz',
+        'sha256sum': 'e26455a6b906d08f848c1569a3c4d9b41d0c57c65f069e28a285409f6f8aca30',
+        'size_bytes': 241326948,
+        'generation': 1771832269754008,
         'condition': 'host_os == "mac" and host_cpu == "arm64"',
       },
       {
-        'object_name': 'Win/rust-toolchain-7dc2e92b83be02dc07f87be7e94266d6e48e5ca5-1-llvmorg-23-init-3706-gfc648683.tar.xz',
-        'sha256sum': '1c783f691498d98769211f427fb3c6ef483ec1b710c6e661614f9bcec6a71c3d',
-        'size_bytes': 404347608,
-        'generation': 1770870604323594,
+        'object_name': 'Win/rust-toolchain-c78a29473a68f07012904af11c92ecffa68fcc75-1-llvmorg-23-init-4965-g686acf63.tar.xz',
+        'sha256sum': '50f23d9770fc930d2d7fd8eae8b1ff21573fe7d83f71e014a7e674dc8884cdc5',
+        'size_bytes': 408636940,
+        'generation': 1771832271552765,
         'condition': 'host_os == "win"',
       },
     ],
@@ -331,7 +331,7 @@ deps = {
   'src/third_party/clang-format/script':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@c2725e0622e1a86d55f14514f2177a39efea4a0e',
   'src/third_party/compiler-rt/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@02ae8c1e75be8825a6c8c1cc74b4ea93933fb189',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@f0b7ec7780a7a8cd070536cc49dfcdafbfd3c11e',
   'src/third_party/libc++/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@7ab65651aed6802d2599dcb7a73b1f82d5179d05',
   'src/third_party/libc++abi/src':
@@ -488,7 +488,7 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@3d72439f882dd6a9e8ddbc6d62b7c9557d9aac72',
+    'https://boringssl.googlesource.com/boringssl.git@5e15537e658fbe7b9e06a0a885e67dc0af489ee3',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@79099fdf668ae097c9eca7052fd5c4c5de6c9098',
   'src/third_party/catapult':
@@ -505,13 +505,13 @@ deps = {
   'src/third_party/crc32c/src':
     'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558',
   'src/third_party/depot_tools':
-    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@8fc02b1d3e48470d9d7fd0e6a68d42f0e38b3bac',
+    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@8aad2863865f0d0bde8b2eb804b389412f04a38b',
   'src/third_party/ffmpeg':
     'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@946d97db8d906277085e361892b7efda5152e2f1',
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@c485bf9b093bd32670b6af8a3a20eae1e2604602',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@305f951f5fcd319876c94fa8c8e0ff359cb30631',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -609,11 +609,11 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '601cac04b0757f1132c64ace99205522d5cabb2d',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '84f98435cfd3858a96a3b9efe96d64207f7d4a02',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
-    'https://chromium.googlesource.com/webm/libvpx.git@a0ac06f5d3c83f890cb502c80470ef0502343fec',
+    'https://chromium.googlesource.com/webm/libvpx.git@db312f06fb50089e699f13a9c516daf44ce0f7d8',
   'src/third_party/libyuv':
     'https://chromium.googlesource.com/libyuv/libyuv.git@6067afde563c3946eebd94f146b3824ab7a97a9c',
   'src/third_party/lss': {
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@6580288c8d453357aaf4e3388b62aa423b0bf53c',
+    'https://chromium.googlesource.com/chromium/src/tools@5dfffd1f75a7312b3ecc5cd9bcf8a2f582e3c648',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'DLUFuZ1z4MFDK54yAzN2lScwHZlUoJssu_cnrpDbqwYC',
+          'version': 'OwcKmOzxpFqUmowRcCL4njbWYzvLYwF19przW4elBvQC',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -864,7 +864,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7364b490b5f78d58efe23ea76e74210fd6c3c76f',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '667cabe3aa54d53d6c542950f6d3876c7f0ba58f',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '00cc497d320ba6b0c6e19b556f5f864570ee9980',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From cda00269ac6dc7353b8a05f4fb427928ae267364 Mon Sep 17 00:00:00 2001
From: Philipp Hancke <philipp.hancke@googlemail.com>
Date: Tue, 24 Feb 2026 13:00:18 +0100
Subject: [PATCH 134/472] build: ignore unittest targets in dependency removal

since the unittests are linked into the test binary in ways
which are beyond the tool and accidentally removed targets will
not be run, decreasing test coverage.

Applied to
  modules/congestion_controller/BUILD.gn
as an example.

Bug: webrtc:467294026
Change-Id: Ic44a8eb3e2250c857bcea0145bf6f2d4ac995113
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/451881
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#46987}
---
 modules/congestion_controller/BUILD.gn | 5 -----
 tools_webrtc/gn_check_autofix.py       | 9 ++++++---
 2 files changed, 6 insertions(+), 8 deletions(-)

diff --git a/modules/congestion_controller/BUILD.gn b/modules/congestion_controller/BUILD.gn
index f45e2e59cf..dc115682b1 100644
--- a/modules/congestion_controller/BUILD.gn
+++ b/modules/congestion_controller/BUILD.gn
@@ -22,9 +22,7 @@ rtc_library("congestion_controller") {
     "../../api:rtp_parameters",
     "../../api:sequence_checker",
     "../../api/environment",
-    "../../api/transport:network_control",
     "../../api/units:data_rate",
-    "../../api/units:data_size",
     "../../api/units:time_delta",
     "../../api/units:timestamp",
     "../../rtc_base:logging",
@@ -33,7 +31,6 @@ rtc_library("congestion_controller") {
     "../../rtc_base/experiments:field_trial_parser",
     "../../rtc_base/synchronization:mutex",
     "../../system_wrappers",
-    "../pacing",
     "../remote_bitrate_estimator",
     "../remote_bitrate_estimator:congestion_control_feedback_generator",
     "../remote_bitrate_estimator:transport_sequence_number_feedback_generator",
@@ -68,9 +65,7 @@ if (rtc_include_tests) {
       "../../test:create_test_field_trials",
       "../../test:test_support",
       "../../test/scenario",
-      "../pacing",
       "../rtp_rtcp:rtp_rtcp_format",
-      "goog_cc:estimators",
       "goog_cc:goog_cc_unittests",
       "goog_cc_scream_network_controller:goog_cc_scream_network_controller_unittests",
       "pcc:pcc_unittests",
diff --git a/tools_webrtc/gn_check_autofix.py b/tools_webrtc/gn_check_autofix.py
index 3b67b7b752..99d57dfe70 100755
--- a/tools_webrtc/gn_check_autofix.py
+++ b/tools_webrtc/gn_check_autofix.py
@@ -166,16 +166,19 @@ def rebase(base_path, dependency_path, dependency):
 
 def _update_deps(new_lines, current_deps_lines_content, block_type):
     """Helper to finalize and append a processed deps block to new_lines."""
-    absolute_deps = []
+    deps_to_keep = []
     for dep_line in current_deps_lines_content:
-        absolute_deps.extend(re.findall(r'"(//[^"]*)"', dep_line))
+        # Keep absolute dependencies.
+        deps_to_keep.extend(re.findall(r'"(//[^"]*)"', dep_line))
+        # Keep test targets.
+        deps_to_keep.extend(re.findall(r'"([^"]*_unittests)"', dep_line))
 
     if block_type == 'deps_set':
         new_lines.append('deps = [\n')
     elif block_type == 'deps_add':
         new_lines.append('deps += [\n')
 
-    for dep in absolute_deps:
+    for dep in deps_to_keep:
         new_lines.append('"' + dep + '",\n')
     new_lines.append(']\n')
 

From 89561f13f6c5531cc7e890d0bfbac20540f42e38 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Erik=20Spr=C3=A5ng?= <sprang@webrtc.org>
Date: Tue, 24 Feb 2026 14:25:44 +0100
Subject: [PATCH 135/472] Remove frame references in corruption detection on
 frame drops.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

The `FrameInstrumentationGenerator` needs both the raw input frame (in
order to do sampling) and the corresponding encoded frame (in order to
map the QP to a filter size). For that reason it holds on to a small
queue of the last two input frames. This was needed in order to deal
with asynchronous encoders, especially in simulcast mode.

In webrtc:467444018 a new callback scheme was introduced where encoders
always announce when all spatial/simulcast layers are done with a frame
either through the `EncodeImage` or through a new `OnFrameDropped`
callback method.

This CL wires the new callback up through `VideoStreamEncoder` to the
`FrameInstrumentationGenerator` so that raw frame handles can be
released as soon as possible.

Bug: webrtc:467444018, webrtc:358039777
Change-Id: Icdb57bdf9c0c8efc2a3153ae204cc852e8f6cd42
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/451880
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46988}
---
 .../frame_instrumentation_generator.h         |  3 +
 video/BUILD.gn                                |  1 +
 .../frame_instrumentation_generator_impl.cc   | 16 ++++
 .../frame_instrumentation_generator_impl.h    |  2 +
 ...rame_instrumentation_generator_unittest.cc | 89 +++++++++++++++++++
 video/video_stream_encoder.cc                 |  9 +-
 video/video_stream_encoder_unittest.cc        | 42 +++++++++
 7 files changed, 160 insertions(+), 2 deletions(-)

diff --git a/api/video/corruption_detection/frame_instrumentation_generator.h b/api/video/corruption_detection/frame_instrumentation_generator.h
index f4ce90546e..d5833d3579 100644
--- a/api/video/corruption_detection/frame_instrumentation_generator.h
+++ b/api/video/corruption_detection/frame_instrumentation_generator.h
@@ -11,6 +11,7 @@
 #ifndef API_VIDEO_CORRUPTION_DETECTION_FRAME_INSTRUMENTATION_GENERATOR_H_
 #define API_VIDEO_CORRUPTION_DETECTION_FRAME_INSTRUMENTATION_GENERATOR_H_
 
+#include <cstdint>
 #include <memory>
 #include <optional>
 
@@ -38,6 +39,8 @@ class FrameInstrumentationGenerator {
   virtual void OnCapturedFrame(VideoFrame frame) = 0;
   virtual std::optional<FrameInstrumentationData> OnEncodedImage(
       const EncodedImage& encoded_image) = 0;
+  // Indicates that all encoding operations for this frame has been completed.
+  virtual void OnFrameReleased(uint32_t rtp_timestamp) = 0;
 
   // Returns `std::nullopt` if there is no context for the given layer.
   // The layer id is the simulcast id or SVC spatial layer id depending on
diff --git a/video/BUILD.gn b/video/BUILD.gn
index 5dca4b5928..b9e2d4497f 100644
--- a/video/BUILD.gn
+++ b/video/BUILD.gn
@@ -904,6 +904,7 @@ if (rtc_include_tests) {
       "../api/video:video_rtp_headers",
       "../api/video:video_stream_encoder",
       "../api/video/corruption_detection:frame_instrumentation_data",
+      "../api/video/corruption_detection:frame_instrumentation_generator_header",
       "../api/video/test:video_frame_matchers",
       "../api/video_codecs:scalability_mode",
       "../api/video_codecs:video_codecs_api",
diff --git a/video/corruption_detection/frame_instrumentation_generator_impl.cc b/video/corruption_detection/frame_instrumentation_generator_impl.cc
index 2dc0bd2621..e1c9f22755 100644
--- a/video/corruption_detection/frame_instrumentation_generator_impl.cc
+++ b/video/corruption_detection/frame_instrumentation_generator_impl.cc
@@ -143,6 +143,10 @@ FrameInstrumentationGeneratorImpl::OnEncodedImage(
     }
     captured_frame = captured_frames_.front();
 
+    if (encoded_image.is_end_of_temporal_unit()) {
+      captured_frames_.pop();
+    }
+
     layer_id = GetSpatialLayerId(encoded_image);
 
     bool is_key_frame = encoded_image.IsKey();
@@ -242,6 +246,18 @@ FrameInstrumentationGeneratorImpl::OnEncodedImage(
   return data;
 }
 
+void FrameInstrumentationGeneratorImpl::OnFrameReleased(
+    uint32_t rtp_timestamp) {
+  MutexLock lock(&mutex_);
+  // Remove the frame and any preceding ones from the queue.
+  while (!captured_frames_.empty() &&
+         (IsNewerTimestamp(rtp_timestamp,
+                           captured_frames_.front().rtp_timestamp()) ||
+          rtp_timestamp == captured_frames_.front().rtp_timestamp())) {
+    captured_frames_.pop();
+  }
+}
+
 std::optional<int> FrameInstrumentationGeneratorImpl::GetHaltonSequenceIndex(
     int layer_id) const {
   MutexLock lock(&mutex_);
diff --git a/video/corruption_detection/frame_instrumentation_generator_impl.h b/video/corruption_detection/frame_instrumentation_generator_impl.h
index e84035ac0b..2544d07ddc 100644
--- a/video/corruption_detection/frame_instrumentation_generator_impl.h
+++ b/video/corruption_detection/frame_instrumentation_generator_impl.h
@@ -48,6 +48,8 @@ class FrameInstrumentationGeneratorImpl : public FrameInstrumentationGenerator {
   void OnCapturedFrame(VideoFrame frame) override RTC_LOCKS_EXCLUDED(mutex_);
   std::optional<FrameInstrumentationData> OnEncodedImage(
       const EncodedImage& encoded_image) override RTC_LOCKS_EXCLUDED(mutex_);
+  void OnFrameReleased(uint32_t rtp_timestamp) override
+      RTC_LOCKS_EXCLUDED(mutex_);
 
   std::optional<int> GetHaltonSequenceIndex(int layer_id) const override
       RTC_LOCKS_EXCLUDED(mutex_);
diff --git a/video/corruption_detection/frame_instrumentation_generator_unittest.cc b/video/corruption_detection/frame_instrumentation_generator_unittest.cc
index 5f0ab33044..d60176fc2d 100644
--- a/video/corruption_detection/frame_instrumentation_generator_unittest.cc
+++ b/video/corruption_detection/frame_instrumentation_generator_unittest.cc
@@ -809,5 +809,94 @@ TEST(FrameInstrumentationGeneratorTest, UsesFrameSelectorWhenEnabled) {
   EXPECT_TRUE(generator.OnEncodedImage(encoded_image2).has_value());
 }
 
+TEST(FrameInstrumentationGeneratorTest, FrameReleasedRemovesFramesFromQueue) {
+  const Environment env = CreateTestEnvironment();
+  auto generator = std::make_unique<FrameInstrumentationGeneratorImpl>(
+      &env, VideoCodecType::kVideoCodecVP8, ScalabilityMode::kL1T1);
+
+  bool frames_destroyed[2] = {};
+  class TestBuffer : public I420Buffer {
+   public:
+    TestBuffer(int width, int height, bool* frame_destroyed_indicator)
+        : I420Buffer(width, height),
+          frame_destroyed_indicator_(frame_destroyed_indicator) {
+      SetBlack(this);
+    }
+
+   private:
+    friend class RefCountedObject<TestBuffer>;
+    ~TestBuffer() override { *frame_destroyed_indicator_ = true; }
+
+    bool* frame_destroyed_indicator_;
+  };
+
+  generator->OnCapturedFrame(
+      VideoFrame::Builder()
+          .set_video_frame_buffer(make_ref_counted<TestBuffer>(
+              kDefaultScaledWidth, kDefaultScaledHeight, &frames_destroyed[0]))
+          .set_rtp_timestamp(1)
+          .build());
+  generator->OnCapturedFrame(
+      VideoFrame::Builder()
+          .set_video_frame_buffer(make_ref_counted<TestBuffer>(
+              kDefaultScaledWidth, kDefaultScaledHeight, &frames_destroyed[1]))
+          .set_rtp_timestamp(2)
+          .build());
+
+  EXPECT_FALSE(frames_destroyed[0]);
+  EXPECT_FALSE(frames_destroyed[1]);
+
+  // Releasing the first frame should destroy it.
+  generator->OnFrameReleased(1);
+  EXPECT_TRUE(frames_destroyed[0]);
+  EXPECT_FALSE(frames_destroyed[1]);
+
+  // Releasing the second frame should destroy it.
+  generator->OnFrameReleased(2);
+  EXPECT_TRUE(frames_destroyed[1]);
+}
+
+TEST(FrameInstrumentationGeneratorTest,
+     EndOfTemporalUnitRemovesFrameFromQueue) {
+  const Environment env = CreateTestEnvironment();
+  auto generator = std::make_unique<FrameInstrumentationGeneratorImpl>(
+      &env, VideoCodecType::kVideoCodecVP8, ScalabilityMode::kL1T1);
+
+  bool frame_destroyed = false;
+  class TestBuffer : public I420Buffer {
+   public:
+    TestBuffer(int width, int height, bool* frame_destroyed_indicator)
+        : I420Buffer(width, height),
+          frame_destroyed_indicator_(frame_destroyed_indicator) {
+      SetBlack(this);
+    }
+
+   private:
+    friend class RefCountedObject<TestBuffer>;
+    ~TestBuffer() override { *frame_destroyed_indicator_ = true; }
+
+    bool* frame_destroyed_indicator_;
+  };
+
+  generator->OnCapturedFrame(
+      VideoFrame::Builder()
+          .set_video_frame_buffer(make_ref_counted<TestBuffer>(
+              kDefaultScaledWidth, kDefaultScaledHeight, &frame_destroyed))
+          .set_rtp_timestamp(1)
+          .build());
+
+  EncodedImage encoded_image;
+  encoded_image.SetRtpTimestamp(1);
+  encoded_image.set_frame_type(VideoFrameType::kVideoFrameKey);
+  encoded_image.qp_ = 10;
+  encoded_image._encodedWidth = kDefaultScaledWidth;
+  encoded_image._encodedHeight = kDefaultScaledHeight;
+  encoded_image.set_end_of_temporal_unit(true);
+
+  EXPECT_FALSE(frame_destroyed);
+  ASSERT_TRUE(generator->OnEncodedImage(encoded_image).has_value());
+  EXPECT_TRUE(frame_destroyed);
+}
+
 }  // namespace
 }  // namespace webrtc
diff --git a/video/video_stream_encoder.cc b/video/video_stream_encoder.cc
index 228e9fe2e9..3a56e77630 100644
--- a/video/video_stream_encoder.cc
+++ b/video/video_stream_encoder.cc
@@ -2465,10 +2465,15 @@ void VideoStreamEncoder::OnFrameDropped(uint32_t rtp_timestamp,
                                         int spatial_id,
                                         bool is_end_of_temporal_unit) {
   sink_->OnFrameDropped(rtp_timestamp, spatial_id, is_end_of_temporal_unit);
-  encoder_queue_->PostTask([this] {
+  encoder_queue_->PostTask([this, rtp_timestamp, is_end_of_temporal_unit] {
     RTC_DCHECK_RUN_ON(encoder_queue_.get());
     stream_resource_manager_.OnFrameDropped(
         VideoStreamEncoderObserver::DropReason::kEncoder);
+    // If this is the end of the temporal unit, signal frame instrumentation
+    // that any reference to this frame can be released.
+    if (frame_instrumentation_generator_ && is_end_of_temporal_unit) {
+      frame_instrumentation_generator_->OnFrameReleased(rtp_timestamp);
+    }
   });
 }
 
@@ -2701,7 +2706,7 @@ void VideoStreamEncoder::ReleaseEncoder() {
   }
   encoder_->Release();
   encoder_initialized_ = false;
-  frame_instrumentation_generator_ = nullptr;
+  frame_instrumentation_generator_.reset();
   TRACE_EVENT0("webrtc", "VCMGenericEncoder::Release");
 }
 
diff --git a/video/video_stream_encoder_unittest.cc b/video/video_stream_encoder_unittest.cc
index 7cb806c03d..f41cfeaca6 100644
--- a/video/video_stream_encoder_unittest.cc
+++ b/video/video_stream_encoder_unittest.cc
@@ -1824,6 +1824,48 @@ TEST_F(VideoStreamEncoderTest,
   video_stream_encoder_->Stop();
 }
 
+TEST_F(VideoStreamEncoderTest,
+       FrameInstrumentationGeneratorFrameReleasedOnFrameDropped) {
+  NiceMock<MockVideoEncoder> video_encoder;
+  test::VideoEncoderProxyFactory encoder_factory(&video_encoder);
+  video_send_config_.encoder_settings.encoder_factory = &encoder_factory;
+  video_send_config_.encoder_settings.enable_frame_instrumentation_generator =
+      true;
+
+  EncodedImageCallback* callback = nullptr;
+  ON_CALL(video_encoder, RegisterEncodeCompleteCallback)
+      .WillByDefault([&](EncodedImageCallback* cb) {
+        callback = cb;
+        return WEBRTC_VIDEO_CODEC_OK;
+      });
+
+  ON_CALL(video_encoder, Encode)
+      .WillByDefault(
+          [&](const VideoFrame& frame, const std::vector<VideoFrameType>*) {
+            EXPECT_TRUE(callback != nullptr);
+            if (callback) {
+              callback->OnFrameDropped(frame.rtp_timestamp(), /*spatial_id=*/0,
+                                       /*is_end_of_temporal_unit=*/true);
+            }
+            return WEBRTC_VIDEO_CODEC_OK;
+          });
+
+  ConfigureEncoder(video_encoder_config_.Copy());
+  video_stream_encoder_->OnBitrateUpdatedAndWaitForManagedResources(
+      kTargetBitrate, kTargetBitrate, 0, 0, 0);
+
+  Event frame_destroyed_event;
+  video_source_.IncomingCapturedFrame(CreateFrame(1, &frame_destroyed_event));
+
+  // Wait for the encoder queue to process the drop.
+  AdvanceTime(TimeDelta::Millis(1));
+
+  // Verify that the frame was released (destruction event triggered).
+  EXPECT_TRUE(frame_destroyed_event.Wait(kDefaultTimeout));
+
+  video_stream_encoder_->Stop();
+}
+
 TEST_F(VideoStreamEncoderTest,
        FrameInstrumentationGeneratorHandlesQueuedFrames) {
   video_send_config_.encoder_settings.enable_frame_instrumentation_generator =

From a95fc3fa2c38ca2eaaa5157eea64faa09d83d4f8 Mon Sep 17 00:00:00 2001
From: Jeremy Leconte <jleconte@webrtc.org>
Date: Tue, 24 Feb 2026 15:49:32 +0100
Subject: [PATCH 136/472] Remove ios internal bots.

Change-Id: Ic75d7d5d40fdffec9e90bec986913a1374f8615e
Bug: b:487237998
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/451942
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Christoffer Dewerin <jansson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46989}
---
 infra/specs/internal.client.webrtc.json | 248 ------------------------
 infra/specs/mixins.pyl                  |  36 ----
 infra/specs/mixins_webrtc.pyl           |  31 ---
 infra/specs/test_suites.pyl             |  33 ----
 infra/specs/waterfalls.pyl              |  39 ----
 tools_webrtc/mb/mb_config.pyl           |  36 ----
 6 files changed, 423 deletions(-)

diff --git a/infra/specs/internal.client.webrtc.json b/infra/specs/internal.client.webrtc.json
index 772c38727d..fb25df4748 100644
--- a/infra/specs/internal.client.webrtc.json
+++ b/infra/specs/internal.client.webrtc.json
@@ -3,130 +3,6 @@
   "AAAAA2 See generate_buildbot_json.py to make changes": {},
   "iOS64 Debug": {
     "isolated_scripts": [
-      {
-        "args": [
-          "--xctest",
-          "--xcode-build-version",
-          "17a400",
-          "--out-dir",
-          "${ISOLATED_OUTDIR}"
-        ],
-        "merge": {
-          "script": "//testing/merge_scripts/standard_isolated_script_merge.py"
-        },
-        "module_name": "//common_audio:common_audio_unittests",
-        "name": "common_audio_unittests",
-        "resultdb": {
-          "enable": true,
-          "has_native_resultdb_integration": true
-        },
-        "swarming": {
-          "cipd_packages": [
-            {
-              "cipd_package": "infra/tools/mac_toolchain/${platform}",
-              "location": ".",
-              "revision": "git_revision:4c7290150d1c360cecc6a93c0214dc531585c3ab"
-            }
-          ],
-          "dimensions": {
-            "os": "iOS-18",
-            "pool": "chrome.tests"
-          },
-          "named_caches": [
-            {
-              "name": "xcode_ios_17a400",
-              "path": "Xcode.app"
-            }
-          ],
-          "service_account": "chrome-tester@chops-service-accounts.iam.gserviceaccount.com"
-        },
-        "test": "common_audio_unittests",
-        "test_id_prefix": "ninja://common_audio:common_audio_unittests/"
-      },
-      {
-        "args": [
-          "--xctest",
-          "--xcode-build-version",
-          "17a400",
-          "--out-dir",
-          "${ISOLATED_OUTDIR}"
-        ],
-        "merge": {
-          "script": "//testing/merge_scripts/standard_isolated_script_merge.py"
-        },
-        "module_name": "//common_video:common_video_unittests",
-        "name": "common_video_unittests",
-        "resultdb": {
-          "enable": true,
-          "has_native_resultdb_integration": true
-        },
-        "swarming": {
-          "cipd_packages": [
-            {
-              "cipd_package": "infra/tools/mac_toolchain/${platform}",
-              "location": ".",
-              "revision": "git_revision:4c7290150d1c360cecc6a93c0214dc531585c3ab"
-            }
-          ],
-          "dimensions": {
-            "os": "iOS-18",
-            "pool": "chrome.tests"
-          },
-          "named_caches": [
-            {
-              "name": "xcode_ios_17a400",
-              "path": "Xcode.app"
-            }
-          ],
-          "service_account": "chrome-tester@chops-service-accounts.iam.gserviceaccount.com"
-        },
-        "test": "common_video_unittests",
-        "test_id_prefix": "ninja://common_video:common_video_unittests/"
-      },
-      {
-        "args": [
-          "--readline-timeout=1200",
-          "--xctest",
-          "--xcode-build-version",
-          "17a400",
-          "--out-dir",
-          "${ISOLATED_OUTDIR}"
-        ],
-        "merge": {
-          "script": "//testing/merge_scripts/standard_isolated_script_merge.py"
-        },
-        "module_name": "//modules:modules_tests",
-        "name": "modules_tests",
-        "resultdb": {
-          "enable": true,
-          "has_native_resultdb_integration": true
-        },
-        "swarming": {
-          "cipd_packages": [
-            {
-              "cipd_package": "infra/tools/mac_toolchain/${platform}",
-              "location": ".",
-              "revision": "git_revision:4c7290150d1c360cecc6a93c0214dc531585c3ab"
-            }
-          ],
-          "dimensions": {
-            "os": "iOS-18",
-            "pool": "chrome.tests"
-          },
-          "hard_timeout": 7200,
-          "io_timeout": 7200,
-          "named_caches": [
-            {
-              "name": "xcode_ios_17a400",
-              "path": "Xcode.app"
-            }
-          ],
-          "service_account": "chrome-tester@chops-service-accounts.iam.gserviceaccount.com",
-          "shards": 2
-        },
-        "test": "modules_tests",
-        "test_id_prefix": "ninja://modules:modules_tests/"
-      },
       {
         "args": [
           "--xctest",
@@ -507,130 +383,6 @@
   },
   "iOS64 Release": {
     "isolated_scripts": [
-      {
-        "args": [
-          "--xctest",
-          "--xcode-build-version",
-          "17a400",
-          "--out-dir",
-          "${ISOLATED_OUTDIR}"
-        ],
-        "merge": {
-          "script": "//testing/merge_scripts/standard_isolated_script_merge.py"
-        },
-        "module_name": "//common_audio:common_audio_unittests",
-        "name": "common_audio_unittests",
-        "resultdb": {
-          "enable": true,
-          "has_native_resultdb_integration": true
-        },
-        "swarming": {
-          "cipd_packages": [
-            {
-              "cipd_package": "infra/tools/mac_toolchain/${platform}",
-              "location": ".",
-              "revision": "git_revision:4c7290150d1c360cecc6a93c0214dc531585c3ab"
-            }
-          ],
-          "dimensions": {
-            "os": "iOS-18",
-            "pool": "chrome.tests"
-          },
-          "named_caches": [
-            {
-              "name": "xcode_ios_17a400",
-              "path": "Xcode.app"
-            }
-          ],
-          "service_account": "chrome-tester@chops-service-accounts.iam.gserviceaccount.com"
-        },
-        "test": "common_audio_unittests",
-        "test_id_prefix": "ninja://common_audio:common_audio_unittests/"
-      },
-      {
-        "args": [
-          "--xctest",
-          "--xcode-build-version",
-          "17a400",
-          "--out-dir",
-          "${ISOLATED_OUTDIR}"
-        ],
-        "merge": {
-          "script": "//testing/merge_scripts/standard_isolated_script_merge.py"
-        },
-        "module_name": "//common_video:common_video_unittests",
-        "name": "common_video_unittests",
-        "resultdb": {
-          "enable": true,
-          "has_native_resultdb_integration": true
-        },
-        "swarming": {
-          "cipd_packages": [
-            {
-              "cipd_package": "infra/tools/mac_toolchain/${platform}",
-              "location": ".",
-              "revision": "git_revision:4c7290150d1c360cecc6a93c0214dc531585c3ab"
-            }
-          ],
-          "dimensions": {
-            "os": "iOS-18",
-            "pool": "chrome.tests"
-          },
-          "named_caches": [
-            {
-              "name": "xcode_ios_17a400",
-              "path": "Xcode.app"
-            }
-          ],
-          "service_account": "chrome-tester@chops-service-accounts.iam.gserviceaccount.com"
-        },
-        "test": "common_video_unittests",
-        "test_id_prefix": "ninja://common_video:common_video_unittests/"
-      },
-      {
-        "args": [
-          "--readline-timeout=1200",
-          "--xctest",
-          "--xcode-build-version",
-          "17a400",
-          "--out-dir",
-          "${ISOLATED_OUTDIR}"
-        ],
-        "merge": {
-          "script": "//testing/merge_scripts/standard_isolated_script_merge.py"
-        },
-        "module_name": "//modules:modules_tests",
-        "name": "modules_tests",
-        "resultdb": {
-          "enable": true,
-          "has_native_resultdb_integration": true
-        },
-        "swarming": {
-          "cipd_packages": [
-            {
-              "cipd_package": "infra/tools/mac_toolchain/${platform}",
-              "location": ".",
-              "revision": "git_revision:4c7290150d1c360cecc6a93c0214dc531585c3ab"
-            }
-          ],
-          "dimensions": {
-            "os": "iOS-18",
-            "pool": "chrome.tests"
-          },
-          "hard_timeout": 7200,
-          "io_timeout": 7200,
-          "named_caches": [
-            {
-              "name": "xcode_ios_17a400",
-              "path": "Xcode.app"
-            }
-          ],
-          "service_account": "chrome-tester@chops-service-accounts.iam.gserviceaccount.com",
-          "shards": 2
-        },
-        "test": "modules_tests",
-        "test_id_prefix": "ninja://modules:modules_tests/"
-      },
       {
         "args": [
           "--xctest",
diff --git a/infra/specs/mixins.pyl b/infra/specs/mixins.pyl
index ec75d88c8f..d69dd1ee0e 100644
--- a/infra/specs/mixins.pyl
+++ b/infra/specs/mixins.pyl
@@ -18,19 +18,6 @@
       }
     }
   },
-  'arm64': {
-    'swarming': {
-      'dimensions': {
-        'cpu': 'arm64'
-      }
-    }
-  },
-  'chrome-tester-service-account': {
-    'swarming': {
-      'service_account':
-      'chrome-tester@chops-service-accounts.iam.gserviceaccount.com'
-    }
-  },
   'chromium-tester-service-account': {
     'fail_if_unused': False,
     'swarming': {
@@ -66,23 +53,6 @@
       'has_native_resultdb_integration': True
     }
   },
-  'ios-device-18': {
-    'swarming': {
-      'dimensions': {
-        'os': 'iOS-18',
-        'pool': 'chrome.tests'
-      }
-    }
-  },
-  'ios-device-perf': {
-    'swarming': {
-      'idempotent': False,
-      'dimensions': {
-        'os': 'iOS-17.6.1',
-        'pool': 'WebRTC'
-      }
-    }
-  },
   'ios_runtime_cache_17_5': {
     'swarming': {
       'named_caches': [{
@@ -240,12 +210,6 @@
       'shards': 8
     }
   },
-  'timeout-2h': {
-    'swarming': {
-      'hard_timeout': 7200,
-      'io_timeout': 7200
-    }
-  },
   'timeout-3h': {
     'swarming': {
       'hard_timeout': 10800,
diff --git a/infra/specs/mixins_webrtc.pyl b/infra/specs/mixins_webrtc.pyl
index 064f6fafec..5ff53a3424 100644
--- a/infra/specs/mixins_webrtc.pyl
+++ b/infra/specs/mixins_webrtc.pyl
@@ -14,13 +14,6 @@
       },
     },
   },
-  'arm64': {
-    'swarming': {
-      'dimensions': {
-        'cpu': 'arm64'
-      }
-    }
-  },
   'cores-12': {
     'swarming': {
       'dimensions': {
@@ -45,24 +38,6 @@
       '--test-arg=--undefok=test_launcher_summary_output'
     ],
   },
-  'ios-device-18': {
-    'swarming': {
-      'dimensions': {
-        'os': 'iOS-18',
-        'pool': 'chrome.tests',
-      }
-    }
-  },
-  'ios-device-perf': {
-    'swarming': {
-      'idempotent': False,
-      'dimensions': {
-        'os': 'iOS-17.6.1',
-        'pool': 'WebRTC',
-        #'device_status': 'available'
-      },
-    },
-  },
   'ios_runtime_cache_17_5': {
     'swarming': {
       'named_caches': [
@@ -212,12 +187,6 @@
       'shards': 8,
     },
   },
-  'timeout-2h': {
-    'swarming': {
-      'hard_timeout': 7200,
-      'io_timeout': 7200,
-    },
-  },
   'timeout-3h': {
     'swarming': {
       'hard_timeout': 10800,
diff --git a/infra/specs/test_suites.pyl b/infra/specs/test_suites.pyl
index 0e827a63df..b3645f5f36 100644
--- a/infra/specs/test_suites.pyl
+++ b/infra/specs/test_suites.pyl
@@ -144,34 +144,6 @@
       # 'tools_unittests': {},
       #
     },
-    'ios_device_tests': {
-      # TODO(bugs.webrtc.org/11362): Real XCTests fail to start on devices.
-      #'apprtcmobile_tests': {'mixins': ['xcodebuild-device-runner']},
-      'common_audio_unittests': {},
-      'common_video_unittests': {},
-      'modules_tests': {
-        'mixins': ['shards-2', 'timeout-2h'],
-        'args': [
-          # Some tests exceed the default 180 seconds readline timeout.
-          '--readline-timeout=1200',
-        ]
-      },
-      'modules_unittests': {
-        'mixins': ['shards-6'],
-      },
-      'rtc_p2p_unittests': {},
-      'rtc_pc_unittests': {},
-      'rtc_stats_unittests': {},
-      # TODO(bugs.webrtc.org/11362): Real XCTests fail to start on devices.
-      #'sdk_framework_unittests': {'mixins': ['xcodebuild-device-runner']},
-      #'sdk_unittests': {'mixins': ['xcodebuild-device-runner']},
-      'system_wrappers_unittests': {},
-      'test_support_unittests': {},
-      'tools_unittests': {},
-      'video_engine_tests': {
-        'mixins': ['shards-4'],
-      },
-    },
     'ios_simulator_tests': {
       'apprtcmobile_tests': {
         'mixins': ['xcodebuild_sim_runner']
@@ -229,11 +201,6 @@
         'mixins': ['perf-webrtc-perf-tests'],
       },
     },
-    'perf_tests_no_video_codec': {
-      'webrtc_perf_tests': {
-        'mixins': ['perf-webrtc-perf-tests'],
-      },
-    },
     'perf_tests_save_worst_frame': {
       'video_codec_perf_tests': {
         'mixins': ['perf-video-codec-perf-tests'],
diff --git a/infra/specs/waterfalls.pyl b/infra/specs/waterfalls.pyl
index 48283ff153..a716f5e96f 100644
--- a/infra/specs/waterfalls.pyl
+++ b/infra/specs/waterfalls.pyl
@@ -298,45 +298,6 @@
       },
     },
   },
-  {
-    'name': 'internal.client.webrtc',
-    'mixins': [],
-    'machines': {
-      'iOS64 Debug': {
-        'mixins': [
-          'ios-device-18', 'webrtc-xctest', 'chrome-tester-service-account',
-          'xcode_26_main', 'mac_toolchain', 'has_native_resultdb_integration',
-          'out_dir_arg'
-        ],
-        'test_suites': {
-          'isolated_scripts': 'ios_device_tests',
-        },
-      },
-      'iOS64 Perf': {
-        'mixins': [
-          'arm64', 'ios-device-perf', 'webrtc-xctest', 'timeout-3h',
-          'chrome-tester-service-account', 'xcode_26_main', 'mac_toolchain',
-          'has_native_resultdb_integration', 'out_dir_arg'
-        ],
-        'test_suites': {
-          'isolated_scripts': 'perf_tests_no_video_codec',
-        },
-        'args': [
-          '--write_perf_output_on_ios',
-        ],
-      },
-      'iOS64 Release': {
-        'mixins': [
-          'ios-device-18', 'webrtc-xctest', 'chrome-tester-service-account',
-          'xcode_26_main', 'mac_toolchain', 'has_native_resultdb_integration',
-          'out_dir_arg'
-        ],
-        'test_suites': {
-          'isolated_scripts': 'ios_device_tests',
-        },
-      },
-    },
-  },
   {
     'name': 'tryserver.webrtc',
     'mixins': [],
diff --git a/tools_webrtc/mb/mb_config.pyl b/tools_webrtc/mb/mb_config.pyl
index 463b063fe8..a2bff5c2ca 100644
--- a/tools_webrtc/mb/mb_config.pyl
+++ b/tools_webrtc/mb/mb_config.pyl
@@ -134,11 +134,6 @@
       'Perf Mac M1 Arm64 12': 'release_bot_x64',
       'Perf Win 10': 'release_bot_x64',
     },
-    'internal.client.webrtc': {
-      'iOS64 Debug': 'ios_internal_debug_bot_arm64',
-      'iOS64 Perf': 'ios_internal_pure_release_bot_arm64',
-      'iOS64 Release': 'ios_internal_release_bot_arm64',
-    },
     'tryserver.webrtc': {
       # Android
       'android_arm64_dbg': 'android_release_bot_arm64',
@@ -353,31 +348,6 @@
       'arm64',
       'xctest',
     ],
-    'ios_internal_debug_bot_arm64': [
-      'ios',
-      'ios_device',
-      'debug_bot',
-      'arm64',
-      'ios_code_signing_identity_description',
-      'xctest',
-    ],
-    'ios_internal_pure_release_bot_arm64': [
-      'ios',
-      'ios_device',
-      'pure_release_bot',
-      'arm64',
-      'ios_code_signing_identity_description',
-      'xctest',
-      'rtc_objc_test_prefix',
-    ],
-    'ios_internal_release_bot_arm64': [
-      'ios',
-      'ios_device',
-      'release_bot',
-      'arm64',
-      'ios_code_signing_identity_description',
-      'xctest',
-    ],
     'ios_release_bot_arm64': [
       'ios',
       'ios_simulator',
@@ -569,12 +539,6 @@
     'ios': {
       'gn_args': 'target_os="ios"',
     },
-    'ios_code_signing_identity_description': {
-      'gn_args': 'ios_code_signing_identity_description="Apple Development"',
-    },
-    'ios_device': {
-      'gn_args': 'target_environment="device"'
-    },
     'ios_simulator': {
       'gn_args': 'target_environment="simulator"'
     },

From 7a52ff95be5953e9ecf61ffaf505feedbfc6ecb7 Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Tue, 24 Feb 2026 20:08:47 -0800
Subject: [PATCH 137/472] Update WebRTC code version (2026-02-25T04:08:12).

Bug: None
Change-Id: Ie365dcbecfba0e3bbe1612136535bc935d393bf5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452101
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#46990}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index 033d646f25..b3caa829ce 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-02-24T04:07:11";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-02-25T04:08:12";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From a736d08f7c3f154786336cc75801f42093b364bd Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Tue, 24 Feb 2026 18:43:44 +0100
Subject: [PATCH 138/472] Avoid integer overflow in UniqueTimestampCounter
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Bug: chromium:486498791
Change-Id: I676101c3eab7199282141b7ea8cd1d98f8a0eaba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/451980
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46991}
---
 video/unique_timestamp_counter.cc | 4 ++++
 1 file changed, 4 insertions(+)

diff --git a/video/unique_timestamp_counter.cc b/video/unique_timestamp_counter.cc
index 14cc039ec9..9e70920373 100644
--- a/video/unique_timestamp_counter.cc
+++ b/video/unique_timestamp_counter.cc
@@ -11,6 +11,7 @@
 #include "video/unique_timestamp_counter.h"
 
 #include <cstdint>
+#include <limits>
 #include <memory>
 #include <set>
 
@@ -25,6 +26,9 @@ UniqueTimestampCounter::UniqueTimestampCounter()
     : latest_(std::make_unique<uint32_t[]>(kMaxHistory)) {}
 
 void UniqueTimestampCounter::Add(uint32_t value) {
+  if (unique_seen_ == std::numeric_limits<decltype(unique_seen_)>::max()) {
+    return;
+  }
   if (value == last_ || !search_index_.insert(value).second) {
     // Already known.
     return;

From 6f7ad28e168d903245c59c2b3d2462f437259cd8 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Wed, 25 Feb 2026 12:07:30 +0100
Subject: [PATCH 139/472] Add AbslStringify to RtpParameters, disable DCHECK

* Temporarily disables the `RTC_DCHECK(cached_filtered == result)`
  in `RtpSenderBase::GetParameters` while tracking down the exact
  sequence of events triggering it in a downstream project.
* Add `AbslStringify` overloads for `RtpParameters`,
  `RtpEncodingParameters`, `RtpCodecParameters`, and
  `RtcpParameters` to facilitate detailed logging.
* Log an error in `rtp_sender.cc` when the cache doesn't seem to be
  correct.

Bug: b/478050997
Change-Id: I7f2fbcf8ffdffcaf0c0a4a5aa19e023542022fed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452000
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Sameer Vijaykar <samvi@google.com>
Cr-Commit-Position: refs/heads/main@{#46992}
---
 api/DEPS             |  1 +
 api/rtp_parameters.h | 95 +++++++++++++++++++++++++++++++++++++++++++-
 pc/rtp_sender.cc     | 12 +++++-
 3 files changed, 105 insertions(+), 3 deletions(-)

diff --git a/api/DEPS b/api/DEPS
index 1656c3f916..01d7057f7d 100644
--- a/api/DEPS
+++ b/api/DEPS
@@ -245,6 +245,7 @@ specific_include_rules = {
 
   "rtp_parameters\\.h": [
     "+absl/strings/str_format.h",
+    "+rtc_base/strings/str_join.h"
   ],
 
   "sequence_checker\\.h": [
diff --git a/api/rtp_parameters.h b/api/rtp_parameters.h
index ab268104f4..ef57479371 100644
--- a/api/rtp_parameters.h
+++ b/api/rtp_parameters.h
@@ -13,6 +13,7 @@
 
 #include <stdint.h>
 
+#include <cstddef>
 #include <map>
 #include <optional>
 #include <string>
@@ -27,6 +28,7 @@
 #include "api/rtp_transceiver_direction.h"
 #include "api/video/resolution.h"
 #include "api/video_codecs/scalability_mode.h"
+#include "rtc_base/strings/str_join.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
@@ -227,6 +229,30 @@ struct RTC_EXPORT RtpCodec {
   bool operator!=(const RtpCodec& o) const { return !(*this == o); }
   bool IsResiliencyCodec() const;
   bool IsMediaCodec() const;
+
+  template <typename Sink>
+  friend void AbslStringify(Sink& sink, const RtpCodec& c) {
+    absl::Format(&sink, "{mime_type: %s", c.mime_type());
+    if (c.clock_rate) {
+      absl::Format(&sink, ", clock_rate: %d", *c.clock_rate);
+    }
+    if (c.num_channels) {
+      absl::Format(&sink, ", num_channels: %d", *c.num_channels);
+    }
+    if (!c.parameters.empty()) {
+      sink.Append(", parameters: {");
+      bool first = true;
+      for (const auto& kv : c.parameters) {
+        if (!first) {
+          sink.Append(", ");
+        }
+        absl::Format(&sink, "%s: %s", kv.first, kv.second);
+        first = false;
+      }
+      sink.Append("}");
+    }
+    sink.Append("}");
+  }
 };
 
 // RtpCodecCapability is to RtpCodecParameters as RtpCapabilities is to
@@ -642,6 +668,44 @@ struct RTC_EXPORT RtpEncodingParameters {
   bool operator!=(const RtpEncodingParameters& o) const {
     return !(*this == o);
   }
+
+  template <typename Sink>
+  friend void AbslStringify(Sink& sink, const RtpEncodingParameters& p) {
+    sink.Append("{");
+    if (p.ssrc)
+      absl::Format(&sink, "ssrc: %u, ", *p.ssrc);
+    absl::Format(&sink, "active: %s, ", p.active ? "true" : "false");
+    absl::Format(&sink, "rid: '%s', ", p.rid);
+    if (p.max_bitrate_bps) {
+      absl::Format(&sink, "max_bitrate_bps: %d, ", *p.max_bitrate_bps);
+    }
+    if (p.min_bitrate_bps) {
+      absl::Format(&sink, "min_bitrate_bps: %d, ", *p.min_bitrate_bps);
+    }
+    if (p.max_framerate) {
+      absl::Format(&sink, "max_framerate: %.2f, ", *p.max_framerate);
+    }
+    if (p.num_temporal_layers) {
+      absl::Format(&sink, "num_temporal_layers: %d, ", *p.num_temporal_layers);
+    }
+    if (p.scale_resolution_down_by) {
+      absl::Format(&sink, "scale_resolution_down_by: %.2f, ",
+                   *p.scale_resolution_down_by);
+    }
+    if (p.scale_resolution_down_to) {
+      absl::Format(&sink, "scale_resolution_down_to: %dx%d, ",
+                   p.scale_resolution_down_to->width,
+                   p.scale_resolution_down_to->height);
+    }
+    if (p.scalability_mode) {
+      absl::Format(&sink, "scalability_mode: '%s', ", *p.scalability_mode);
+    }
+    if (p.codec) {
+      absl::Format(&sink, "codec: %v, ", *p.codec);
+    }
+    absl::Format(&sink, "adaptive_ptime: %s}",
+                 p.adaptive_ptime ? "true" : "false");
+  }
 };
 
 struct RTC_EXPORT RtpCodecParameters : public RtpCodec {
@@ -660,7 +724,8 @@ struct RTC_EXPORT RtpCodecParameters : public RtpCodec {
   bool operator!=(const RtpCodecParameters& o) const { return !(*this == o); }
   template <typename Sink>
   friend void AbslStringify(Sink& sink, const RtpCodecParameters& p) {
-    absl::Format(&sink, "[%d: %s]", p.payload_type, p.mime_type());
+    absl::Format(&sink, "{payload_type: %d, codec: %v}", p.payload_type,
+                 static_cast<const RtpCodec&>(p));
   }
 };
 
@@ -721,6 +786,17 @@ struct RtcpParameters final {
            reduced_size == o.reduced_size && mux == o.mux;
   }
   bool operator!=(const RtcpParameters& o) const { return !(*this == o); }
+
+  template <typename Sink>
+  friend void AbslStringify(Sink& sink, const RtcpParameters& p) {
+    sink.Append("{");
+    if (p.ssrc)
+      absl::Format(&sink, "ssrc: %u, ", *p.ssrc);
+    absl::Format(&sink, "cname: '%s', ", p.cname);
+    absl::Format(&sink, "reduced_size: %s, ",
+                 p.reduced_size ? "true" : "false");
+    absl::Format(&sink, "mux: %s}", p.mux ? "true" : "false");
+  }
 };
 
 struct RTC_EXPORT RtpParameters {
@@ -767,6 +843,23 @@ struct RTC_EXPORT RtpParameters {
   // If at least two active encodings have different codec values
   // (including one being unset and another set), this is considered mixed.
   bool IsMixedCodec() const;
+
+  template <typename Sink>
+  friend void AbslStringify(Sink& sink, const RtpParameters& p) {
+    sink.Append("{");
+    absl::Format(&sink, "transaction_id: '%s', ", p.transaction_id);
+    absl::Format(&sink, "mid: '%s', ", p.mid);
+    absl::Format(&sink, "codecs: [%v], ", StrJoin(p.codecs, ", "));
+    absl::Format(&sink, "header_extensions: [%v], ",
+                 StrJoin(p.header_extensions, ", "));
+    absl::Format(&sink, "encodings: [%v], ", StrJoin(p.encodings, ", "));
+    absl::Format(&sink, "rtcp: %v", p.rtcp);
+    if (p.degradation_preference) {
+      absl::Format(&sink, ", degradation_preference: %s",
+                   DegradationPreferenceToString(*p.degradation_preference));
+    }
+    sink.Append("}");
+  }
 };
 
 }  // namespace webrtc
diff --git a/pc/rtp_sender.cc b/pc/rtp_sender.cc
index 66c7931909..5330733536 100644
--- a/pc/rtp_sender.cc
+++ b/pc/rtp_sender.cc
@@ -357,8 +357,16 @@ RtpParameters RtpSenderBase::GetParameters() const {
   if (cached && !stopped_ && ssrc_ != 0 && cached_parameters_) {
     RtpParameters cached_filtered = *cached;
     RemoveEncodingLayers(disabled_rids_, &cached_filtered.encodings);
-    RTC_DCHECK(cached_filtered == result)
-        << "The cached value should have been equal (filtered)";
+    if (cached_filtered != result) {
+      RTC_LOG(LS_ERROR)
+          << "Cached send params not equal to worker thread state.\n"
+          << "Cached: " << cached_filtered << "\n"
+          << "Result: " << result;
+    }
+    // TODO(b/478050997): Re-enable this check once the downstream issue is
+    // resolved.
+    // RTC_DCHECK(cached_filtered == result)
+    //     << "The cached value should have been equal (filtered)";
   }
 #endif
   return result;

From aab29cd134655f820ded7c709ca599c33882921d Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Wed, 25 Feb 2026 06:12:15 -0800
Subject: [PATCH 140/472] Roll chromium_revision acca47dc0a..25a5bc7242
 (1589354:1590100)

Change log: https://chromium.googlesource.com/chromium/src/+log/acca47dc0a..25a5bc7242
Full diff: https://chromium.googlesource.com/chromium/src/+/acca47dc0a..25a5bc7242

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/9002579929..cdd1fdc6cf
* src/buildtools/linux64: git_revision:bfffe163a23e55b944a3b4d5b9ce0ded126f414a..git_revision:47b40ef49e678319c0dac767a95a363cc3f1f6cb
* src/buildtools/mac: git_revision:bfffe163a23e55b944a3b4d5b9ce0ded126f414a..git_revision:47b40ef49e678319c0dac767a95a363cc3f1f6cb
* src/buildtools/win: git_revision:bfffe163a23e55b944a3b4d5b9ce0ded126f414a..git_revision:47b40ef49e678319c0dac767a95a363cc3f1f6cb
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/918d9fd6ed..8b97cccd9c
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/3ca5ab0699..56f1e9e797
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/1feaf237f7..f3dd4165f6
* src/third_party/androidx/cipd: OwcKmOzxpFqUmowRcCL4njbWYzvLYwF19przW4elBvQC..xPOOM_O7RM7zSKvQL5dce7wU0aMURYvUGE6_Kvjxal8C
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/5e15537e65..7d88bb1bf3
* src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/f0b7ec7780..efa78c1067
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/00cc497d32..c4c704e5dd
* src/third_party/fuzztest/src: https://chromium.googlesource.com/external/github.com/google/fuzztest.git/+log/7636e7cbf8..362a279f0a
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/305f951f5f..8c3ea8f13c
* src/third_party/libjpeg_turbo: https://chromium.googlesource.com/chromium/deps/libjpeg_turbo.git/+log/6bb85251a8..d1f5f2393e
* src/third_party/libsrtp: https://chromium.googlesource.com/chromium/deps/libsrtp.git/+log/a52756acb1..e8383771af
* src/third_party/llvm-build/Release+Asserts: Linux_x64/clang-llvmorg-23-init-4965-g686acf63-1.tar.xz,Linux_x64/clang-tidy-llvmorg-23-init-4965-g686acf63-1.tar.xz,Linux_x64/clangd-llvmorg-23-init-4965-g686acf63-1.tar.xz,Linux_x64/llvm-code-coverage-llvmorg-23-init-4965-g686acf63-1.tar.xz,Linux_x64/llvmobjdump-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac/clang-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac/clang-mac-runtime-library-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac/clang-tidy-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac/clangd-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac/llvm-code-coverage-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac/llvmobjdump-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac_arm64/clang-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac_arm64/clang-tidy-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac_arm64/clangd-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac_arm64/llvm-code-coverage-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac_arm64/llvmobjdump-llvmorg-23-init-4965-g686acf63-1.tar.xz,Win/clang-llvmorg-23-init-4965-g686acf63-1.tar.xz,Win/clang-tidy-llvmorg-23-init-4965-g686acf63-1.tar.xz,Win/clang-win-runtime-library-llvmorg-23-init-4965-g686acf63-1.tar.xz,Win/clangd-llvmorg-23-init-4965-g686acf63-1.tar.xz,Win/llvm-code-coverage-llvmorg-23-init-4965-g686acf63-1.tar.xz,Win/llvmobjdump-llvmorg-23-init-4965-g686acf63-1.tar.xz..Linux_x64/clang-llvmorg-23-init-3706-gfc648683-1.tar.xz,Linux_x64/clang-tidy-llvmorg-23-init-3706-gfc648683-1.tar.xz,Linux_x64/clangd-llvmorg-23-init-3706-gfc648683-1.tar.xz,Linux_x64/llvm-code-coverage-llvmorg-23-init-3706-gfc648683-1.tar.xz,Linux_x64/llvmobjdump-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac/clang-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac/clang-mac-runtime-library-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac/clang-tidy-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac/clangd-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac/llvm-code-coverage-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac/llvmobjdump-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac_arm64/clang-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac_arm64/clang-tidy-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac_arm64/clangd-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac_arm64/llvm-code-coverage-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac_arm64/llvmobjdump-llvmorg-23-init-3706-gfc648683-1.tar.xz,Win/clang-llvmorg-23-init-3706-gfc648683-1.tar.xz,Win/clang-tidy-llvmorg-23-init-3706-gfc648683-1.tar.xz,Win/clang-win-runtime-library-llvmorg-23-init-3706-gfc648683-1.tar.xz,Win/clangd-llvmorg-23-init-3706-gfc648683-1.tar.xz,Win/llvm-code-coverage-llvmorg-23-init-3706-gfc648683-1.tar.xz,Win/llvmobjdump-llvmorg-23-init-3706-gfc648683-1.tar.xz
* src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/a91d484f4c..5a08c449f7
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/84f98435cf..c82fb577c5
* src/third_party/rust-toolchain: Linux_x64/rust-toolchain-c78a29473a68f07012904af11c92ecffa68fcc75-1-llvmorg-23-init-4965-g686acf63.tar.xz,Mac/rust-toolchain-c78a29473a68f07012904af11c92ecffa68fcc75-1-llvmorg-23-init-4965-g686acf63.tar.xz,Mac_arm64/rust-toolchain-c78a29473a68f07012904af11c92ecffa68fcc75-1-llvmorg-23-init-4965-g686acf63.tar.xz,Win/rust-toolchain-c78a29473a68f07012904af11c92ecffa68fcc75-1-llvmorg-23-init-4965-g686acf63.tar.xz..Linux_x64/rust-toolchain-7dc2e92b83be02dc07f87be7e94266d6e48e5ca5-1-llvmorg-23-init-3706-gfc648683.tar.xz,Mac/rust-toolchain-7dc2e92b83be02dc07f87be7e94266d6e48e5ca5-1-llvmorg-23-init-3706-gfc648683.tar.xz,Mac_arm64/rust-toolchain-7dc2e92b83be02dc07f87be7e94266d6e48e5ca5-1-llvmorg-23-init-3706-gfc648683.tar.xz,Win/rust-toolchain-7dc2e92b83be02dc07f87be7e94266d6e48e5ca5-1-llvmorg-23-init-3706-gfc648683.tar.xz
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/5dfffd1f75..21fbb49679
DEPS diff: https://chromium.googlesource.com/chromium/src/+/acca47dc0a..25a5bc7242/DEPS

BUG=None

Change-Id: I2f1a8ad75856c01c5833646294ed0d22440dd444
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452200
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#46993}
---
 DEPS | 246 +++++++++++++++++++++++++++++------------------------------
 1 file changed, 123 insertions(+), 123 deletions(-)

diff --git a/DEPS b/DEPS
index 45f6efa106..2d9d6d2553 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': 'acca47dc0a9e6971232cf29c4482cc410740f278',
+  'chromium_revision': '25a5bc7242a972e00204085eca1bc300b9861102',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@9002579929e246c1ff22c7015a2bdd6170c968a5',
+    'https://chromium.googlesource.com/chromium/src/build@cdd1fdc6cfb4c17f19886eec62a28bf18727a7c7',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@451e71432ac9f6a91d4d5604c29e0a5078e8c2e8',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,19 +78,19 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@918d9fd6eddba5b7c380a1635e1baabd1a02134f',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@8b97cccd9ca6ad4d613dc48178afffeda9e978a2',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@3ca5ab0699d20e6260f10f07e65f3e7b1b83cc86',
+    'https://chromium.googlesource.com/chromium/src/testing@56f1e9e7979880aa686a78709013ffc112256444',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@1feaf237f73d66a536361b817cacd59b58a4df55',
+    'https://chromium.googlesource.com/chromium/src/third_party@f3dd4165f6ff7a8e156969d3770967ddc9baf606',
 
   'src/buildtools/linux64': {
     'packages': [
       {
         'package': 'gn/gn/linux-${{arch}}',
-        'version': 'git_revision:bfffe163a23e55b944a3b4d5b9ce0ded126f414a',
+        'version': 'git_revision:47b40ef49e678319c0dac767a95a363cc3f1f6cb',
       }
     ],
     'dep_type': 'cipd',
@@ -100,7 +100,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/mac-${{arch}}',
-        'version': 'git_revision:bfffe163a23e55b944a3b4d5b9ce0ded126f414a',
+        'version': 'git_revision:47b40ef49e678319c0dac767a95a363cc3f1f6cb',
       }
     ],
     'dep_type': 'cipd',
@@ -110,7 +110,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/windows-amd64',
-        'version': 'git_revision:bfffe163a23e55b944a3b4d5b9ce0ded126f414a',
+        'version': 'git_revision:47b40ef49e678319c0dac767a95a363cc3f1f6cb',
       }
     ],
     'dep_type': 'cipd',
@@ -136,157 +136,157 @@ deps = {
     'objects': [
       {
         # The Android libclang_rt.builtins libraries are currently only included in the Linux clang package.
-        'object_name': 'Linux_x64/clang-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': '6d4796631177f3c070923242601a4e14425ffdc2ee84c95ce630c0198d755637',
-        'size_bytes': 57997052,
-        'generation': 1771832273904291,
+        'object_name': 'Linux_x64/clang-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': '9a5dfd23f6d37b21709327aa39562225c01a058cc209d1c629448d6d6d874ab2',
+        'size_bytes': 57775784,
+        'generation': 1770870606052091,
         'condition': '(host_os == "linux" or checkout_android) and non_git_source',
       },
       {
-        'object_name': 'Linux_x64/clang-tidy-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': '250f932ba1195eaa1ab8fd6e50f3c650ff3b74d28127d982ee79b96c4844bfb7',
-        'size_bytes': 14392496,
-        'generation': 1771832274267259,
+        'object_name': 'Linux_x64/clang-tidy-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': 'cef6e9f2876cc28956213fd1cfb1431ed6bd9aca135b12ebd33e9d393595009d',
+        'size_bytes': 14411824,
+        'generation': 1770870606077571,
         'condition': 'host_os == "linux" and checkout_clang_tidy and non_git_source',
       },
       {
-        'object_name': 'Linux_x64/clangd-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': '39258f20af9d459b8b032ee6bd38a79ae764fd950b79bace3227b8965d58d189',
-        'size_bytes': 14618644,
-        'generation': 1771832274470726,
+        'object_name': 'Linux_x64/clangd-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': 'efccb1da6950b4a83857ad5243196dd9d4cefee5869362abcc9d9422b059fff2',
+        'size_bytes': 14638444,
+        'generation': 1770870606182500,
         'condition': 'host_os == "linux" and checkout_clangd and non_git_source',
       },
       {
-        'object_name': 'Linux_x64/llvm-code-coverage-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': 'cf2fd97359cc982f56b0beb7e9abff5a4799a12ff8c3e22728ddf06e1053b867',
-        'size_bytes': 2330268,
-        'generation': 1771832275108482,
+        'object_name': 'Linux_x64/llvm-code-coverage-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': '7d1648506f753035e0c96f9727abf58f0a275d8057be6fd3c231bf80ac60b2ac',
+        'size_bytes': 2326952,
+        'generation': 1770870606310079,
         'condition': 'host_os == "linux" and checkout_clang_coverage_tools and non_git_source',
       },
       {
-        'object_name': 'Linux_x64/llvmobjdump-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': '720a9cbb8745d4ac4c3f887bc1f004fed44cfccb851adebaa765a1517531c952',
-        'size_bytes': 5786008,
-        'generation': 1771832274626197,
+        'object_name': 'Linux_x64/llvmobjdump-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': '103ba738fb6ebcd290e38458fdec8fa7b95bc3216af1e60ad260bc065b49d447',
+        'size_bytes': 5798432,
+        'generation': 1770870606263889,
         'condition': '((checkout_linux or checkout_mac or checkout_android) and host_os == "linux") and non_git_source',
       },
       {
-        'object_name': 'Mac/clang-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': '189a5bca90f83e2dc9671fef5f149386452c215a532e94cdc98cc13461411fb2',
-        'size_bytes': 54754892,
-        'generation': 1771832277222794,
+        'object_name': 'Mac/clang-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': '0230c7c9a625ceb2d2c650d995fb9cb5cdb6ab7c0c05042f560cc0cda441a2d0',
+        'size_bytes': 54682532,
+        'generation': 1770870608103936,
         'condition': 'host_os == "mac" and host_cpu == "x64"',
       },
       {
-        'object_name': 'Mac/clang-mac-runtime-library-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': '3a0177f1cf5f88f00e86006cb9be42f5d86a8dc461d03daf2919dee2c632f619',
-        'size_bytes': 1012728,
-        'generation': 1771832301108033,
+        'object_name': 'Mac/clang-mac-runtime-library-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': 'a7af5d119637f67bff7f390923a78a712b1067157c019aebf2dcddbc4ed7b640',
+        'size_bytes': 1015272,
+        'generation': 1770870616072374,
         'condition': 'checkout_mac and not host_os == "mac"',
       },
       {
-        'object_name': 'Mac/clang-tidy-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': 'c8a3f088c4d8bbb92eba2036bb68d1640c9c54ef7ec4686b4e2a5c49ff9b6bc3',
-        'size_bytes': 14282996,
-        'generation': 1771832277199632,
+        'object_name': 'Mac/clang-tidy-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': '89bd1e39dc4114183a8cd931a9f5b3dc4fff5f697a133d91da2ecd2f82221774',
+        'size_bytes': 14261468,
+        'generation': 1770870608107521,
         'condition': 'host_os == "mac" and host_cpu == "x64" and checkout_clang_tidy',
       },
       {
-        'object_name': 'Mac/clangd-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': '4746a11e14fe96a00024a7b7d7b4ee1d61c8960171578efad263d1c54cfa6901',
-        'size_bytes': 15449448,
-        'generation': 1771832277304839,
+        'object_name': 'Mac/clangd-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': 'd8c9c8f9a4e62422f2167b283da1e32d39978804543f7eaa08b903d4a0c580a2',
+        'size_bytes': 15425576,
+        'generation': 1770870608113120,
         'condition': 'host_os == "mac" and host_cpu == "x64" and checkout_clangd',
       },
       {
-        'object_name': 'Mac/llvm-code-coverage-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': 'fbaa78a2d70a0872c2e0fd94cbe9f13c61b3077c6ba73f76a34413510a420ed0',
-        'size_bytes': 2373980,
-        'generation': 1771832277813116,
+        'object_name': 'Mac/llvm-code-coverage-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': 'a705f42d6e441950594986c0afb3f7b2e65cafa10eb1a5f40dc43c0a5d127af6',
+        'size_bytes': 2372548,
+        'generation': 1770870608249587,
         'condition': 'host_os == "mac" and host_cpu == "x64" and checkout_clang_coverage_tools',
       },
       {
-        'object_name': 'Mac/llvmobjdump-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': 'f091b4127cd59792c2f278535b1a30ae2e0953c6e896f31a5b5c7beff707e388',
-        'size_bytes': 5693600,
-        'generation': 1771832277787419,
+        'object_name': 'Mac/llvmobjdump-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': '556b8abf3ec08c42769986ea8940ac7f09fa2cfe49328a2ccccee4641efce6c4',
+        'size_bytes': 5703228,
+        'generation': 1770870608115133,
         'condition': 'host_os == "mac" and host_cpu == "x64"',
       },
       {
-        'object_name': 'Mac_arm64/clang-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': '4a091561d8ee6cfff59efffecbfc0122d233d23427e1cdbcee5080253bfc81f9',
-        'size_bytes': 45966912,
-        'generation': 1771832302845435,
+        'object_name': 'Mac_arm64/clang-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': '34102ff942cb0d7e2163fada5aeb53f5c2438d07d8caf5ed79970b133adc04be',
+        'size_bytes': 45849908,
+        'generation': 1770870617609650,
         'condition': 'host_os == "mac" and host_cpu == "arm64"',
       },
       {
-        'object_name': 'Mac_arm64/clang-tidy-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': '9248646928a75768d682b484e3a83f481152b8edce5f71fa6e8052d9ae210398',
-        'size_bytes': 12473436,
-        'generation': 1771832303183995,
+        'object_name': 'Mac_arm64/clang-tidy-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': '11dfe174b262142ce8d2d9e810ff2a5c5ac5e44ccb135876825fc3ff24be1b35',
+        'size_bytes': 12473944,
+        'generation': 1770870617763889,
         'condition': 'host_os == "mac" and host_cpu == "arm64" and checkout_clang_tidy',
       },
       {
-        'object_name': 'Mac_arm64/clangd-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': 'e5fde137e7372fedc8e2c4a8e4f9fe718c3bcc63c76acaa0c4a859b04e3f4762',
-        'size_bytes': 12882364,
-        'generation': 1771832303644261,
+        'object_name': 'Mac_arm64/clangd-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': 'a9d8847208d6411087464fb9d2f07d1a9a7617821ade0b6e00fb31e29934e12f',
+        'size_bytes': 12883052,
+        'generation': 1770870617857412,
         'condition': 'host_os == "mac" and host_cpu == "arm64" and checkout_clangd',
       },
       {
-        'object_name': 'Mac_arm64/llvm-code-coverage-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': 'd4e9d19346ef021be42a27b367efcdcfee03cc19bbceddb9ea4f92619ef8b06c',
-        'size_bytes': 1993468,
-        'generation': 1771832304157004,
+        'object_name': 'Mac_arm64/llvm-code-coverage-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': 'bf15d654a27132d07c5a03d6bb174f79449a88a34a08b9024e72a1b88127bdd0',
+        'size_bytes': 1991372,
+        'generation': 1770870618097417,
         'condition': 'host_os == "mac" and host_cpu == "arm64" and checkout_clang_coverage_tools',
       },
       {
-        'object_name': 'Mac_arm64/llvmobjdump-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': 'b01bce55530dbca2d6d25fa14ddbac8259184fdca1cc63365130953b9ebb517f',
-        'size_bytes': 5442884,
-        'generation': 1771832303592136,
+        'object_name': 'Mac_arm64/llvmobjdump-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': 'aea68efe7aeab8eb4d2371e79e7dcb021e1448d8edd0d4851fa34907b93205a6',
+        'size_bytes': 5446112,
+        'generation': 1770870617921094,
         'condition': 'host_os == "mac" and host_cpu == "arm64"',
       },
       {
-        'object_name': 'Win/clang-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': '84c13e593b1532c0cf0b8fe0928482e6eea96408846a679ce68e53c90911c33e',
-        'size_bytes': 49494200,
-        'generation': 1771832332047169,
+        'object_name': 'Win/clang-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': 'cca114f624db433c2ba6458a6de34e6e474bea379f483817dd5dc0a02c59df3c',
+        'size_bytes': 49244312,
+        'generation': 1770870628650736,
         'condition': 'host_os == "win"',
       },
       {
-        'object_name': 'Win/clang-tidy-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': '2a5072bfa4a3eb9dd60321d955d6d4744b7b9f2f657f32dbc706774282787e9a',
-        'size_bytes': 14455216,
-        'generation': 1771832332462256,
+        'object_name': 'Win/clang-tidy-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': 'de37a5f6e43f7d4ff7c7c8fe0ba7cd89bdd0cbe0370b171263f36ed372a7e3d0',
+        'size_bytes': 14464644,
+        'generation': 1770870628637623,
         'condition': 'host_os == "win" and checkout_clang_tidy',
       },
       {
-        'object_name': 'Win/clang-win-runtime-library-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': 'a9ddbb29b61eaac9f777b1fb532eebb49ab9e546bcd680129fcc7a2fd648cd27',
-        'size_bytes': 2596776,
-        'generation': 1771832355675158,
+        'object_name': 'Win/clang-win-runtime-library-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': '3f2ad3f2adb92384ecdcb33385aa8beacd7f38d27bed6179e39b2e971469d028',
+        'size_bytes': 2595252,
+        'generation': 1770870636929948,
         'condition': 'checkout_win and not host_os == "win"',
       },
       {
-        'object_name': 'Win/clangd-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': '5dad22f5da22e268e43476a28689ec7b9fb8d41ae55e53911a26e9c25a9c4c4b',
-        'size_bytes': 14884728,
-        'generation': 1771832332829491,
+        'object_name': 'Win/clangd-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': 'b9f868e6be509a1900e7a8157cd29235245e0847804abb8d20fbc37772fbf802',
+        'size_bytes': 14883160,
+        'generation': 1770870628678782,
         'condition': 'host_os == "win" and checkout_clangd',
       },
       {
-        'object_name': 'Win/llvm-code-coverage-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': 'f5a46dbdf979e31d35dd12db37d714bc06a36957883f3c11d727d56382465079',
-        'size_bytes': 2478920,
-        'generation': 1771832333093363,
+        'object_name': 'Win/llvm-code-coverage-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': '30cb9e52d92731ed24cf08775aeb5eb0e4d2c1eb871daed1f344f6b4e635fe23',
+        'size_bytes': 2477228,
+        'generation': 1770870628999104,
         'condition': 'host_os == "win" and checkout_clang_coverage_tools',
       },
       {
-        'object_name': 'Win/llvmobjdump-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': '0519107fdef9c730bb4f073c87dbd03995fd2da074aabef86347f62fb2b784f3',
-        'size_bytes': 5838628,
-        'generation': 1771832332695387,
+        'object_name': 'Win/llvmobjdump-llvmorg-23-init-3706-gfc648683-1.tar.xz',
+        'sha256sum': '7baba8d463b92d16c510d67be60eed89897ba76898121ea709264cbb41066af1',
+        'size_bytes': 5859968,
+        'generation': 1770870628750202,
         'condition': '(checkout_linux or checkout_mac or checkout_android) and host_os == "win"',
       },
     ]
@@ -298,31 +298,31 @@ deps = {
     'bucket': 'chromium-browser-clang',
     'objects': [
       {
-        'object_name': 'Linux_x64/rust-toolchain-c78a29473a68f07012904af11c92ecffa68fcc75-1-llvmorg-23-init-4965-g686acf63.tar.xz',
-        'sha256sum': 'e3f380e162218de33aeea559dd7696431fd505b47b9669422faea83b47b16de4',
-        'size_bytes': 270144912,
-        'generation': 1771832265918901,
+        'object_name': 'Linux_x64/rust-toolchain-7dc2e92b83be02dc07f87be7e94266d6e48e5ca5-1-llvmorg-23-init-3706-gfc648683.tar.xz',
+        'sha256sum': 'c14a211a9fcef604df768db22a4422553c791c4ad7e3f982cf00b9c0c745ccaa',
+        'size_bytes': 266703680,
+        'generation': 1770870599310471,
         'condition': 'host_os == "linux" and non_git_source',
       },
       {
-        'object_name': 'Mac/rust-toolchain-c78a29473a68f07012904af11c92ecffa68fcc75-1-llvmorg-23-init-4965-g686acf63.tar.xz',
-        'sha256sum': '01a4c3b5a3be52b5ae2a99e78541c512cd198b1a87baa1bc1c0b4697428d8fb2',
-        'size_bytes': 257817220,
-        'generation': 1771832267985133,
+        'object_name': 'Mac/rust-toolchain-7dc2e92b83be02dc07f87be7e94266d6e48e5ca5-1-llvmorg-23-init-3706-gfc648683.tar.xz',
+        'sha256sum': 'dfd5f890b8d84eeee4cf5aa3ba9b9e2e070bb5bece134f979060c35b9286ed00',
+        'size_bytes': 254634176,
+        'generation': 1770870600934341,
         'condition': 'host_os == "mac" and host_cpu == "x64"',
       },
       {
-        'object_name': 'Mac_arm64/rust-toolchain-c78a29473a68f07012904af11c92ecffa68fcc75-1-llvmorg-23-init-4965-g686acf63.tar.xz',
-        'sha256sum': 'e26455a6b906d08f848c1569a3c4d9b41d0c57c65f069e28a285409f6f8aca30',
-        'size_bytes': 241326948,
-        'generation': 1771832269754008,
+        'object_name': 'Mac_arm64/rust-toolchain-7dc2e92b83be02dc07f87be7e94266d6e48e5ca5-1-llvmorg-23-init-3706-gfc648683.tar.xz',
+        'sha256sum': '4e0a8de0ef6e65d17f723193c44c7dc6495f2274e7a7818e9b84e9645281be7e',
+        'size_bytes': 238113184,
+        'generation': 1770870602823865,
         'condition': 'host_os == "mac" and host_cpu == "arm64"',
       },
       {
-        'object_name': 'Win/rust-toolchain-c78a29473a68f07012904af11c92ecffa68fcc75-1-llvmorg-23-init-4965-g686acf63.tar.xz',
-        'sha256sum': '50f23d9770fc930d2d7fd8eae8b1ff21573fe7d83f71e014a7e674dc8884cdc5',
-        'size_bytes': 408636940,
-        'generation': 1771832271552765,
+        'object_name': 'Win/rust-toolchain-7dc2e92b83be02dc07f87be7e94266d6e48e5ca5-1-llvmorg-23-init-3706-gfc648683.tar.xz',
+        'sha256sum': '1c783f691498d98769211f427fb3c6ef483ec1b710c6e661614f9bcec6a71c3d',
+        'size_bytes': 404347608,
+        'generation': 1770870604323594,
         'condition': 'host_os == "win"',
       },
     ],
@@ -331,13 +331,13 @@ deps = {
   'src/third_party/clang-format/script':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@c2725e0622e1a86d55f14514f2177a39efea4a0e',
   'src/third_party/compiler-rt/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@f0b7ec7780a7a8cd070536cc49dfcdafbfd3c11e',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@efa78c10673ca0ea0ab5c6c6165c4173117e6ed3',
   'src/third_party/libc++/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@7ab65651aed6802d2599dcb7a73b1f82d5179d05',
   'src/third_party/libc++abi/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@8f11bb1d4438d0239d0dfc1bd9456a9f31629dda',
   'src/third_party/llvm-libc/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@a91d484f4c303b81ea50cbd396d200c08ebdcab3',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@5a08c449f7cfea83a178a0eb3418b611570ca0bb',
   'src/third_party/libunwind/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@17ccf7d110c5526cb77e93cfd8330f491fb2bf18',
 
@@ -488,7 +488,7 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@5e15537e658fbe7b9e06a0a885e67dc0af489ee3',
+    'https://boringssl.googlesource.com/boringssl.git@7d88bb1bf3372bda1134ad8bf624b25b88e0db86',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@79099fdf668ae097c9eca7052fd5c4c5de6c9098',
   'src/third_party/catapult':
@@ -511,7 +511,7 @@ deps = {
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@305f951f5fcd319876c94fa8c8e0ff359cb30631',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@8c3ea8f13cec6729b5c0fed5812b3f54980498e6',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -593,11 +593,11 @@ deps = {
   'src/third_party/libFuzzer/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt/lib/fuzzer.git@bea408a6e01f0f7e6c82a43121fe3af4506c932e',
   'src/third_party/fuzztest/src':
-    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@7636e7cbf8281e049950f7c229fce46e52011c21',
+    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@362a279f0ad018574278e72c4e98e2b99d3991bb',
   'src/third_party/libjpeg_turbo':
-    'https://chromium.googlesource.com/chromium/deps/libjpeg_turbo.git@6bb85251a8382b5e07f635a981ac685cc5ab5053',
+    'https://chromium.googlesource.com/chromium/deps/libjpeg_turbo.git@d1f5f2393e0d51f840207342ae86e55a86443288',
   'src/third_party/libsrtp':
-    'https://chromium.googlesource.com/chromium/deps/libsrtp.git@a52756acb1c5e133089c798736dd171567df11f5',
+    'https://chromium.googlesource.com/chromium/deps/libsrtp.git@e8383771af8aa4096f5bcfe3743a5ea128f88a9a',
   'src/third_party/dav1d/libdav1d':
     'https://chromium.googlesource.com/external/github.com/videolan/dav1d.git@b546257f770768b2c88258c533da38b91a06f737',
   'src/third_party/libaom/source/libaom':
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '84f98435cfd3858a96a3b9efe96d64207f7d4a02',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'c82fb577c5b0e0100da838161757d37851514ffb',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@5dfffd1f75a7312b3ecc5cd9bcf8a2f582e3c648',
+    'https://chromium.googlesource.com/chromium/src/tools@21fbb49679d1252d98599ecfd36a0511195eaf1e',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'OwcKmOzxpFqUmowRcCL4njbWYzvLYwF19przW4elBvQC',
+          'version': 'xPOOM_O7RM7zSKvQL5dce7wU0aMURYvUGE6_Kvjxal8C',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -864,7 +864,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7364b490b5f78d58efe23ea76e74210fd6c3c76f',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '00cc497d320ba6b0c6e19b556f5f864570ee9980',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + 'c4c704e5ddca35c10cca625807a6919654753459',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From bde09d5bd568a34fda3c6c3fbc5f4ee4efdcd48f Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Wed, 25 Feb 2026 10:06:35 -0800
Subject: [PATCH 141/472] Roll chromium_revision 25a5bc7242..d2eedb0120
 (1590100:1590259)

Change log: https://chromium.googlesource.com/chromium/src/+log/25a5bc7242..d2eedb0120
Full diff: https://chromium.googlesource.com/chromium/src/+/25a5bc7242..d2eedb0120

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/cdd1fdc6cf..781cb9adb9
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/8b97cccd9c..6ba00cac4e
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/56f1e9e797..8b0ad5e09f
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/f3dd4165f6..43bfc7dafa
* src/third_party/androidx/cipd: xPOOM_O7RM7zSKvQL5dce7wU0aMURYvUGE6_Kvjxal8C..xaK45BLOgiQ39L2Uk63LH-VFziubwYJ15XzCCwHKAo0C
* src/third_party/breakpad/breakpad: https://chromium.googlesource.com/breakpad/breakpad.git/+log/79099fdf66..9f26d631e4
* src/third_party/icu: https://chromium.googlesource.com/chromium/deps/icu.git/+log/a86a32e67b..7971660ba6
* src/third_party/llvm-build/Release+Asserts: Linux_x64/clang-llvmorg-23-init-3706-gfc648683-1.tar.xz,Linux_x64/clang-tidy-llvmorg-23-init-3706-gfc648683-1.tar.xz,Linux_x64/clangd-llvmorg-23-init-3706-gfc648683-1.tar.xz,Linux_x64/llvm-code-coverage-llvmorg-23-init-3706-gfc648683-1.tar.xz,Linux_x64/llvmobjdump-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac/clang-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac/clang-mac-runtime-library-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac/clang-tidy-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac/clangd-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac/llvm-code-coverage-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac/llvmobjdump-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac_arm64/clang-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac_arm64/clang-tidy-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac_arm64/clangd-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac_arm64/llvm-code-coverage-llvmorg-23-init-3706-gfc648683-1.tar.xz,Mac_arm64/llvmobjdump-llvmorg-23-init-3706-gfc648683-1.tar.xz,Win/clang-llvmorg-23-init-3706-gfc648683-1.tar.xz,Win/clang-tidy-llvmorg-23-init-3706-gfc648683-1.tar.xz,Win/clang-win-runtime-library-llvmorg-23-init-3706-gfc648683-1.tar.xz,Win/clangd-llvmorg-23-init-3706-gfc648683-1.tar.xz,Win/llvm-code-coverage-llvmorg-23-init-3706-gfc648683-1.tar.xz,Win/llvmobjdump-llvmorg-23-init-3706-gfc648683-1.tar.xz..Linux_x64/clang-llvmorg-23-init-4965-g686acf63-1.tar.xz,Linux_x64/clang-tidy-llvmorg-23-init-4965-g686acf63-1.tar.xz,Linux_x64/clangd-llvmorg-23-init-4965-g686acf63-1.tar.xz,Linux_x64/llvm-code-coverage-llvmorg-23-init-4965-g686acf63-1.tar.xz,Linux_x64/llvmobjdump-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac/clang-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac/clang-mac-runtime-library-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac/clang-tidy-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac/clangd-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac/llvm-code-coverage-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac/llvmobjdump-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac_arm64/clang-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac_arm64/clang-tidy-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac_arm64/clangd-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac_arm64/llvm-code-coverage-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac_arm64/llvmobjdump-llvmorg-23-init-4965-g686acf63-1.tar.xz,Win/clang-llvmorg-23-init-4965-g686acf63-1.tar.xz,Win/clang-tidy-llvmorg-23-init-4965-g686acf63-1.tar.xz,Win/clang-win-runtime-library-llvmorg-23-init-4965-g686acf63-1.tar.xz,Win/clangd-llvmorg-23-init-4965-g686acf63-1.tar.xz,Win/llvm-code-coverage-llvmorg-23-init-4965-g686acf63-1.tar.xz,Win/llvmobjdump-llvmorg-23-init-4965-g686acf63-1.tar.xz
* src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/5a08c449f7..f4940a069b
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/c82fb577c5..ea8d0aaeb1
* src/third_party/rust-toolchain: Linux_x64/rust-toolchain-7dc2e92b83be02dc07f87be7e94266d6e48e5ca5-1-llvmorg-23-init-3706-gfc648683.tar.xz,Mac/rust-toolchain-7dc2e92b83be02dc07f87be7e94266d6e48e5ca5-1-llvmorg-23-init-3706-gfc648683.tar.xz,Mac_arm64/rust-toolchain-7dc2e92b83be02dc07f87be7e94266d6e48e5ca5-1-llvmorg-23-init-3706-gfc648683.tar.xz,Win/rust-toolchain-7dc2e92b83be02dc07f87be7e94266d6e48e5ca5-1-llvmorg-23-init-3706-gfc648683.tar.xz..Linux_x64/rust-toolchain-c78a29473a68f07012904af11c92ecffa68fcc75-1-llvmorg-23-init-4965-g686acf63.tar.xz,Mac/rust-toolchain-c78a29473a68f07012904af11c92ecffa68fcc75-1-llvmorg-23-init-4965-g686acf63.tar.xz,Mac_arm64/rust-toolchain-c78a29473a68f07012904af11c92ecffa68fcc75-1-llvmorg-23-init-4965-g686acf63.tar.xz,Win/rust-toolchain-c78a29473a68f07012904af11c92ecffa68fcc75-1-llvmorg-23-init-4965-g686acf63.tar.xz
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/21fbb49679..b68ae3255c
DEPS diff: https://chromium.googlesource.com/chromium/src/+/25a5bc7242..d2eedb0120/DEPS

BUG=None

Change-Id: I1412a3c7b936c223a51a0f4e345ba148bf835ec3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452241
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#46994}
---
 DEPS | 230 +++++++++++++++++++++++++++++------------------------------
 1 file changed, 115 insertions(+), 115 deletions(-)

diff --git a/DEPS b/DEPS
index 2d9d6d2553..7f28df1359 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '25a5bc7242a972e00204085eca1bc300b9861102',
+  'chromium_revision': 'd2eedb0120845815f8e8d4d4dc69bbf4c0671216',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@cdd1fdc6cfb4c17f19886eec62a28bf18727a7c7',
+    'https://chromium.googlesource.com/chromium/src/build@781cb9adb9dbb5feb7b54605415d68f2ffe8409d',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@451e71432ac9f6a91d4d5604c29e0a5078e8c2e8',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@8b97cccd9ca6ad4d613dc48178afffeda9e978a2',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@6ba00cac4ea78e220d0bccafe54d1c3d0ceff99e',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@56f1e9e7979880aa686a78709013ffc112256444',
+    'https://chromium.googlesource.com/chromium/src/testing@8b0ad5e09f566d0b2709abef4d2f0b06af4af5b7',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@f3dd4165f6ff7a8e156969d3770967ddc9baf606',
+    'https://chromium.googlesource.com/chromium/src/third_party@43bfc7dafafdc106a23bdb0f40a02ac4d70e76fb',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -136,157 +136,157 @@ deps = {
     'objects': [
       {
         # The Android libclang_rt.builtins libraries are currently only included in the Linux clang package.
-        'object_name': 'Linux_x64/clang-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': '9a5dfd23f6d37b21709327aa39562225c01a058cc209d1c629448d6d6d874ab2',
-        'size_bytes': 57775784,
-        'generation': 1770870606052091,
+        'object_name': 'Linux_x64/clang-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': '6d4796631177f3c070923242601a4e14425ffdc2ee84c95ce630c0198d755637',
+        'size_bytes': 57997052,
+        'generation': 1771832273904291,
         'condition': '(host_os == "linux" or checkout_android) and non_git_source',
       },
       {
-        'object_name': 'Linux_x64/clang-tidy-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': 'cef6e9f2876cc28956213fd1cfb1431ed6bd9aca135b12ebd33e9d393595009d',
-        'size_bytes': 14411824,
-        'generation': 1770870606077571,
+        'object_name': 'Linux_x64/clang-tidy-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': '250f932ba1195eaa1ab8fd6e50f3c650ff3b74d28127d982ee79b96c4844bfb7',
+        'size_bytes': 14392496,
+        'generation': 1771832274267259,
         'condition': 'host_os == "linux" and checkout_clang_tidy and non_git_source',
       },
       {
-        'object_name': 'Linux_x64/clangd-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': 'efccb1da6950b4a83857ad5243196dd9d4cefee5869362abcc9d9422b059fff2',
-        'size_bytes': 14638444,
-        'generation': 1770870606182500,
+        'object_name': 'Linux_x64/clangd-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': '39258f20af9d459b8b032ee6bd38a79ae764fd950b79bace3227b8965d58d189',
+        'size_bytes': 14618644,
+        'generation': 1771832274470726,
         'condition': 'host_os == "linux" and checkout_clangd and non_git_source',
       },
       {
-        'object_name': 'Linux_x64/llvm-code-coverage-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': '7d1648506f753035e0c96f9727abf58f0a275d8057be6fd3c231bf80ac60b2ac',
-        'size_bytes': 2326952,
-        'generation': 1770870606310079,
+        'object_name': 'Linux_x64/llvm-code-coverage-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': 'cf2fd97359cc982f56b0beb7e9abff5a4799a12ff8c3e22728ddf06e1053b867',
+        'size_bytes': 2330268,
+        'generation': 1771832275108482,
         'condition': 'host_os == "linux" and checkout_clang_coverage_tools and non_git_source',
       },
       {
-        'object_name': 'Linux_x64/llvmobjdump-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': '103ba738fb6ebcd290e38458fdec8fa7b95bc3216af1e60ad260bc065b49d447',
-        'size_bytes': 5798432,
-        'generation': 1770870606263889,
+        'object_name': 'Linux_x64/llvmobjdump-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': '720a9cbb8745d4ac4c3f887bc1f004fed44cfccb851adebaa765a1517531c952',
+        'size_bytes': 5786008,
+        'generation': 1771832274626197,
         'condition': '((checkout_linux or checkout_mac or checkout_android) and host_os == "linux") and non_git_source',
       },
       {
-        'object_name': 'Mac/clang-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': '0230c7c9a625ceb2d2c650d995fb9cb5cdb6ab7c0c05042f560cc0cda441a2d0',
-        'size_bytes': 54682532,
-        'generation': 1770870608103936,
+        'object_name': 'Mac/clang-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': '189a5bca90f83e2dc9671fef5f149386452c215a532e94cdc98cc13461411fb2',
+        'size_bytes': 54754892,
+        'generation': 1771832277222794,
         'condition': 'host_os == "mac" and host_cpu == "x64"',
       },
       {
-        'object_name': 'Mac/clang-mac-runtime-library-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': 'a7af5d119637f67bff7f390923a78a712b1067157c019aebf2dcddbc4ed7b640',
-        'size_bytes': 1015272,
-        'generation': 1770870616072374,
+        'object_name': 'Mac/clang-mac-runtime-library-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': '3a0177f1cf5f88f00e86006cb9be42f5d86a8dc461d03daf2919dee2c632f619',
+        'size_bytes': 1012728,
+        'generation': 1771832301108033,
         'condition': 'checkout_mac and not host_os == "mac"',
       },
       {
-        'object_name': 'Mac/clang-tidy-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': '89bd1e39dc4114183a8cd931a9f5b3dc4fff5f697a133d91da2ecd2f82221774',
-        'size_bytes': 14261468,
-        'generation': 1770870608107521,
+        'object_name': 'Mac/clang-tidy-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': 'c8a3f088c4d8bbb92eba2036bb68d1640c9c54ef7ec4686b4e2a5c49ff9b6bc3',
+        'size_bytes': 14282996,
+        'generation': 1771832277199632,
         'condition': 'host_os == "mac" and host_cpu == "x64" and checkout_clang_tidy',
       },
       {
-        'object_name': 'Mac/clangd-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': 'd8c9c8f9a4e62422f2167b283da1e32d39978804543f7eaa08b903d4a0c580a2',
-        'size_bytes': 15425576,
-        'generation': 1770870608113120,
+        'object_name': 'Mac/clangd-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': '4746a11e14fe96a00024a7b7d7b4ee1d61c8960171578efad263d1c54cfa6901',
+        'size_bytes': 15449448,
+        'generation': 1771832277304839,
         'condition': 'host_os == "mac" and host_cpu == "x64" and checkout_clangd',
       },
       {
-        'object_name': 'Mac/llvm-code-coverage-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': 'a705f42d6e441950594986c0afb3f7b2e65cafa10eb1a5f40dc43c0a5d127af6',
-        'size_bytes': 2372548,
-        'generation': 1770870608249587,
+        'object_name': 'Mac/llvm-code-coverage-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': 'fbaa78a2d70a0872c2e0fd94cbe9f13c61b3077c6ba73f76a34413510a420ed0',
+        'size_bytes': 2373980,
+        'generation': 1771832277813116,
         'condition': 'host_os == "mac" and host_cpu == "x64" and checkout_clang_coverage_tools',
       },
       {
-        'object_name': 'Mac/llvmobjdump-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': '556b8abf3ec08c42769986ea8940ac7f09fa2cfe49328a2ccccee4641efce6c4',
-        'size_bytes': 5703228,
-        'generation': 1770870608115133,
+        'object_name': 'Mac/llvmobjdump-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': 'f091b4127cd59792c2f278535b1a30ae2e0953c6e896f31a5b5c7beff707e388',
+        'size_bytes': 5693600,
+        'generation': 1771832277787419,
         'condition': 'host_os == "mac" and host_cpu == "x64"',
       },
       {
-        'object_name': 'Mac_arm64/clang-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': '34102ff942cb0d7e2163fada5aeb53f5c2438d07d8caf5ed79970b133adc04be',
-        'size_bytes': 45849908,
-        'generation': 1770870617609650,
+        'object_name': 'Mac_arm64/clang-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': '4a091561d8ee6cfff59efffecbfc0122d233d23427e1cdbcee5080253bfc81f9',
+        'size_bytes': 45966912,
+        'generation': 1771832302845435,
         'condition': 'host_os == "mac" and host_cpu == "arm64"',
       },
       {
-        'object_name': 'Mac_arm64/clang-tidy-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': '11dfe174b262142ce8d2d9e810ff2a5c5ac5e44ccb135876825fc3ff24be1b35',
-        'size_bytes': 12473944,
-        'generation': 1770870617763889,
+        'object_name': 'Mac_arm64/clang-tidy-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': '9248646928a75768d682b484e3a83f481152b8edce5f71fa6e8052d9ae210398',
+        'size_bytes': 12473436,
+        'generation': 1771832303183995,
         'condition': 'host_os == "mac" and host_cpu == "arm64" and checkout_clang_tidy',
       },
       {
-        'object_name': 'Mac_arm64/clangd-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': 'a9d8847208d6411087464fb9d2f07d1a9a7617821ade0b6e00fb31e29934e12f',
-        'size_bytes': 12883052,
-        'generation': 1770870617857412,
+        'object_name': 'Mac_arm64/clangd-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': 'e5fde137e7372fedc8e2c4a8e4f9fe718c3bcc63c76acaa0c4a859b04e3f4762',
+        'size_bytes': 12882364,
+        'generation': 1771832303644261,
         'condition': 'host_os == "mac" and host_cpu == "arm64" and checkout_clangd',
       },
       {
-        'object_name': 'Mac_arm64/llvm-code-coverage-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': 'bf15d654a27132d07c5a03d6bb174f79449a88a34a08b9024e72a1b88127bdd0',
-        'size_bytes': 1991372,
-        'generation': 1770870618097417,
+        'object_name': 'Mac_arm64/llvm-code-coverage-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': 'd4e9d19346ef021be42a27b367efcdcfee03cc19bbceddb9ea4f92619ef8b06c',
+        'size_bytes': 1993468,
+        'generation': 1771832304157004,
         'condition': 'host_os == "mac" and host_cpu == "arm64" and checkout_clang_coverage_tools',
       },
       {
-        'object_name': 'Mac_arm64/llvmobjdump-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': 'aea68efe7aeab8eb4d2371e79e7dcb021e1448d8edd0d4851fa34907b93205a6',
-        'size_bytes': 5446112,
-        'generation': 1770870617921094,
+        'object_name': 'Mac_arm64/llvmobjdump-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': 'b01bce55530dbca2d6d25fa14ddbac8259184fdca1cc63365130953b9ebb517f',
+        'size_bytes': 5442884,
+        'generation': 1771832303592136,
         'condition': 'host_os == "mac" and host_cpu == "arm64"',
       },
       {
-        'object_name': 'Win/clang-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': 'cca114f624db433c2ba6458a6de34e6e474bea379f483817dd5dc0a02c59df3c',
-        'size_bytes': 49244312,
-        'generation': 1770870628650736,
+        'object_name': 'Win/clang-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': '84c13e593b1532c0cf0b8fe0928482e6eea96408846a679ce68e53c90911c33e',
+        'size_bytes': 49494200,
+        'generation': 1771832332047169,
         'condition': 'host_os == "win"',
       },
       {
-        'object_name': 'Win/clang-tidy-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': 'de37a5f6e43f7d4ff7c7c8fe0ba7cd89bdd0cbe0370b171263f36ed372a7e3d0',
-        'size_bytes': 14464644,
-        'generation': 1770870628637623,
+        'object_name': 'Win/clang-tidy-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': '2a5072bfa4a3eb9dd60321d955d6d4744b7b9f2f657f32dbc706774282787e9a',
+        'size_bytes': 14455216,
+        'generation': 1771832332462256,
         'condition': 'host_os == "win" and checkout_clang_tidy',
       },
       {
-        'object_name': 'Win/clang-win-runtime-library-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': '3f2ad3f2adb92384ecdcb33385aa8beacd7f38d27bed6179e39b2e971469d028',
-        'size_bytes': 2595252,
-        'generation': 1770870636929948,
+        'object_name': 'Win/clang-win-runtime-library-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': 'a9ddbb29b61eaac9f777b1fb532eebb49ab9e546bcd680129fcc7a2fd648cd27',
+        'size_bytes': 2596776,
+        'generation': 1771832355675158,
         'condition': 'checkout_win and not host_os == "win"',
       },
       {
-        'object_name': 'Win/clangd-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': 'b9f868e6be509a1900e7a8157cd29235245e0847804abb8d20fbc37772fbf802',
-        'size_bytes': 14883160,
-        'generation': 1770870628678782,
+        'object_name': 'Win/clangd-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': '5dad22f5da22e268e43476a28689ec7b9fb8d41ae55e53911a26e9c25a9c4c4b',
+        'size_bytes': 14884728,
+        'generation': 1771832332829491,
         'condition': 'host_os == "win" and checkout_clangd',
       },
       {
-        'object_name': 'Win/llvm-code-coverage-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': '30cb9e52d92731ed24cf08775aeb5eb0e4d2c1eb871daed1f344f6b4e635fe23',
-        'size_bytes': 2477228,
-        'generation': 1770870628999104,
+        'object_name': 'Win/llvm-code-coverage-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': 'f5a46dbdf979e31d35dd12db37d714bc06a36957883f3c11d727d56382465079',
+        'size_bytes': 2478920,
+        'generation': 1771832333093363,
         'condition': 'host_os == "win" and checkout_clang_coverage_tools',
       },
       {
-        'object_name': 'Win/llvmobjdump-llvmorg-23-init-3706-gfc648683-1.tar.xz',
-        'sha256sum': '7baba8d463b92d16c510d67be60eed89897ba76898121ea709264cbb41066af1',
-        'size_bytes': 5859968,
-        'generation': 1770870628750202,
+        'object_name': 'Win/llvmobjdump-llvmorg-23-init-4965-g686acf63-1.tar.xz',
+        'sha256sum': '0519107fdef9c730bb4f073c87dbd03995fd2da074aabef86347f62fb2b784f3',
+        'size_bytes': 5838628,
+        'generation': 1771832332695387,
         'condition': '(checkout_linux or checkout_mac or checkout_android) and host_os == "win"',
       },
     ]
@@ -298,31 +298,31 @@ deps = {
     'bucket': 'chromium-browser-clang',
     'objects': [
       {
-        'object_name': 'Linux_x64/rust-toolchain-7dc2e92b83be02dc07f87be7e94266d6e48e5ca5-1-llvmorg-23-init-3706-gfc648683.tar.xz',
-        'sha256sum': 'c14a211a9fcef604df768db22a4422553c791c4ad7e3f982cf00b9c0c745ccaa',
-        'size_bytes': 266703680,
-        'generation': 1770870599310471,
+        'object_name': 'Linux_x64/rust-toolchain-c78a29473a68f07012904af11c92ecffa68fcc75-1-llvmorg-23-init-4965-g686acf63.tar.xz',
+        'sha256sum': 'e3f380e162218de33aeea559dd7696431fd505b47b9669422faea83b47b16de4',
+        'size_bytes': 270144912,
+        'generation': 1771832265918901,
         'condition': 'host_os == "linux" and non_git_source',
       },
       {
-        'object_name': 'Mac/rust-toolchain-7dc2e92b83be02dc07f87be7e94266d6e48e5ca5-1-llvmorg-23-init-3706-gfc648683.tar.xz',
-        'sha256sum': 'dfd5f890b8d84eeee4cf5aa3ba9b9e2e070bb5bece134f979060c35b9286ed00',
-        'size_bytes': 254634176,
-        'generation': 1770870600934341,
+        'object_name': 'Mac/rust-toolchain-c78a29473a68f07012904af11c92ecffa68fcc75-1-llvmorg-23-init-4965-g686acf63.tar.xz',
+        'sha256sum': '01a4c3b5a3be52b5ae2a99e78541c512cd198b1a87baa1bc1c0b4697428d8fb2',
+        'size_bytes': 257817220,
+        'generation': 1771832267985133,
         'condition': 'host_os == "mac" and host_cpu == "x64"',
       },
       {
-        'object_name': 'Mac_arm64/rust-toolchain-7dc2e92b83be02dc07f87be7e94266d6e48e5ca5-1-llvmorg-23-init-3706-gfc648683.tar.xz',
-        'sha256sum': '4e0a8de0ef6e65d17f723193c44c7dc6495f2274e7a7818e9b84e9645281be7e',
-        'size_bytes': 238113184,
-        'generation': 1770870602823865,
+        'object_name': 'Mac_arm64/rust-toolchain-c78a29473a68f07012904af11c92ecffa68fcc75-1-llvmorg-23-init-4965-g686acf63.tar.xz',
+        'sha256sum': 'e26455a6b906d08f848c1569a3c4d9b41d0c57c65f069e28a285409f6f8aca30',
+        'size_bytes': 241326948,
+        'generation': 1771832269754008,
         'condition': 'host_os == "mac" and host_cpu == "arm64"',
       },
       {
-        'object_name': 'Win/rust-toolchain-7dc2e92b83be02dc07f87be7e94266d6e48e5ca5-1-llvmorg-23-init-3706-gfc648683.tar.xz',
-        'sha256sum': '1c783f691498d98769211f427fb3c6ef483ec1b710c6e661614f9bcec6a71c3d',
-        'size_bytes': 404347608,
-        'generation': 1770870604323594,
+        'object_name': 'Win/rust-toolchain-c78a29473a68f07012904af11c92ecffa68fcc75-1-llvmorg-23-init-4965-g686acf63.tar.xz',
+        'sha256sum': '50f23d9770fc930d2d7fd8eae8b1ff21573fe7d83f71e014a7e674dc8884cdc5',
+        'size_bytes': 408636940,
+        'generation': 1771832271552765,
         'condition': 'host_os == "win"',
       },
     ],
@@ -337,7 +337,7 @@ deps = {
   'src/third_party/libc++abi/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@8f11bb1d4438d0239d0dfc1bd9456a9f31629dda',
   'src/third_party/llvm-libc/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@5a08c449f7cfea83a178a0eb3418b611570ca0bb',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@f4940a069bf76a358d7cc7f525e1fa74b85f6ba0',
   'src/third_party/libunwind/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@17ccf7d110c5526cb77e93cfd8330f491fb2bf18',
 
@@ -490,7 +490,7 @@ deps = {
   'src/third_party/boringssl/src':
     'https://boringssl.googlesource.com/boringssl.git@7d88bb1bf3372bda1134ad8bf624b25b88e0db86',
   'src/third_party/breakpad/breakpad':
-    'https://chromium.googlesource.com/breakpad/breakpad.git@79099fdf668ae097c9eca7052fd5c4c5de6c9098',
+    'https://chromium.googlesource.com/breakpad/breakpad.git@9f26d631e4063c0af3e8ee3bc8826486115372a8',
   'src/third_party/catapult':
     'https://chromium.googlesource.com/catapult.git@9fdb1d4c9d0f278ae34891b069d96ef89ea9cdf2',
   'src/third_party/ced/src': {
@@ -537,7 +537,7 @@ deps = {
   'src/third_party/googletest/src':
     'https://chromium.googlesource.com/external/github.com/google/googletest.git@4fe3307fb2d9f86d19777c7eb0e4809e9694dde7',
   'src/third_party/icu': {
-    'url': 'https://chromium.googlesource.com/chromium/deps/icu.git@a86a32e67b8d1384b33f8fa48c83a6079b86f8cd',
+    'url': 'https://chromium.googlesource.com/chromium/deps/icu.git@7971660ba6306a4cc8e6f872905138f59de5bc1d',
   },
   'src/third_party/jdk/current': {
       'packages': [
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'c82fb577c5b0e0100da838161757d37851514ffb',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'ea8d0aaeb1e22f03fa0498c42e3f86f3daf753f2',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@21fbb49679d1252d98599ecfd36a0511195eaf1e',
+    'https://chromium.googlesource.com/chromium/src/tools@b68ae3255c6d8930b40faaf51ca8624c15eeabf1',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'xPOOM_O7RM7zSKvQL5dce7wU0aMURYvUGE6_Kvjxal8C',
+          'version': 'xaK45BLOgiQ39L2Uk63LH-VFziubwYJ15XzCCwHKAo0C',
       },
     ],
     'condition': 'checkout_android and non_git_source',

From af2a2ae44eb338f7cf4608e14bb1e1a0d6cb24ac Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Wed, 25 Feb 2026 16:05:38 -0800
Subject: [PATCH 142/472] Roll chromium_revision d2eedb0120..33dcdf76ee
 (1590259:1590492)

Change log: https://chromium.googlesource.com/chromium/src/+log/d2eedb0120..33dcdf76ee
Full diff: https://chromium.googlesource.com/chromium/src/+/d2eedb0120..33dcdf76ee

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/781cb9adb9..07efd23b7d
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/6ba00cac4e..14067ecb4b
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/8b0ad5e09f..3b1ff75cc0
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/43bfc7dafa..b000765734
* src/third_party/androidx/cipd: xaK45BLOgiQ39L2Uk63LH-VFziubwYJ15XzCCwHKAo0C..mvV9ZifgIOHqOM9xEhhdmQJUTmj0fBmZciH8DOyvG88C
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/7d88bb1bf3..626a2578ac
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/9fdb1d4c9d..1ec586469a
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/8aad286386..044f01ae03
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/c4c704e5dd..d0d70a9527
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/8c3ea8f13c..fad7939709
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/ea8d0aaeb1..c9609d1e65
* src/third_party/tflite/src: https://chromium.googlesource.com/external/github.com/tensorflow/tensorflow.git/+log/38dd095516..204f897f9a
* src/third_party/xnnpack/src: https://chromium.googlesource.com/external/github.com/google/XNNPACK.git/+log/e8aad3cc4f..066e3e72b0
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/b68ae3255c..5cb7e32d33
* src/tools/luci-go: git_revision:21504d3595ac903ef60bac732d885cd8e6ed92ba..git_revision:f769e04338c2db24c3697f00ede9cadbd3ee7582
* src/tools/luci-go: git_revision:21504d3595ac903ef60bac732d885cd8e6ed92ba..git_revision:f769e04338c2db24c3697f00ede9cadbd3ee7582
* src/tools/luci-go: git_revision:21504d3595ac903ef60bac732d885cd8e6ed92ba..git_revision:f769e04338c2db24c3697f00ede9cadbd3ee7582
DEPS diff: https://chromium.googlesource.com/chromium/src/+/d2eedb0120..33dcdf76ee/DEPS

BUG=None

Change-Id: I62c65d4ebaec65652ce40902844903770c3bcbb2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452320
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#46995}
---
 DEPS | 36 ++++++++++++++++++------------------
 1 file changed, 18 insertions(+), 18 deletions(-)

diff --git a/DEPS b/DEPS
index 7f28df1359..e0eff8484c 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': 'd2eedb0120845815f8e8d4d4dc69bbf4c0671216',
+  'chromium_revision': '33dcdf76eebc97e39d04746dbde27bf2c06ccd9e',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@781cb9adb9dbb5feb7b54605415d68f2ffe8409d',
+    'https://chromium.googlesource.com/chromium/src/build@07efd23b7d9cf9b6a59076d14f5a68cd0c4c1fa7',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@451e71432ac9f6a91d4d5604c29e0a5078e8c2e8',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@6ba00cac4ea78e220d0bccafe54d1c3d0ceff99e',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@14067ecb4b74487149d4eaa74f0ae40df9ee6d6d',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@8b0ad5e09f566d0b2709abef4d2f0b06af4af5b7',
+    'https://chromium.googlesource.com/chromium/src/testing@3b1ff75cc08b882e4fc18a461b17c0a44f5b815e',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@43bfc7dafafdc106a23bdb0f40a02ac4d70e76fb',
+    'https://chromium.googlesource.com/chromium/src/third_party@b000765734c7b82c2ca5a3ee40dc274dcafbef31',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -488,11 +488,11 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@7d88bb1bf3372bda1134ad8bf624b25b88e0db86',
+    'https://boringssl.googlesource.com/boringssl.git@626a2578ac3e2840d4664562de39d25b1e710a19',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@9f26d631e4063c0af3e8ee3bc8826486115372a8',
   'src/third_party/catapult':
-    'https://chromium.googlesource.com/catapult.git@9fdb1d4c9d0f278ae34891b069d96ef89ea9cdf2',
+    'https://chromium.googlesource.com/catapult.git@1ec586469aedb45f1e0809017409346ddf96e210',
   'src/third_party/ced/src': {
     'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5',
   },
@@ -505,13 +505,13 @@ deps = {
   'src/third_party/crc32c/src':
     'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558',
   'src/third_party/depot_tools':
-    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@8aad2863865f0d0bde8b2eb804b389412f04a38b',
+    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@044f01ae0351fb8a97efce042c93f99d3e95240c',
   'src/third_party/ffmpeg':
     'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@946d97db8d906277085e361892b7efda5152e2f1',
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@8c3ea8f13cec6729b5c0fed5812b3f54980498e6',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@fad79397092a7029152f6ecb0712a4e00c15e51a',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'ea8d0aaeb1e22f03fa0498c42e3f86f3daf753f2',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'c9609d1e65bcd21014a708c0c8b8df094cd89a0b',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@b68ae3255c6d8930b40faaf51ca8624c15eeabf1',
+    'https://chromium.googlesource.com/chromium/src/tools@5cb7e32d338c947a9bd1a8db7659c07089c1ac52',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'xaK45BLOgiQ39L2Uk63LH-VFziubwYJ15XzCCwHKAo0C',
+          'version': 'mvV9ZifgIOHqOM9xEhhdmQJUTmj0fBmZciH8DOyvG88C',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -786,7 +786,7 @@ deps = {
   },
 
   'src/third_party/tflite/src':
-    Var('chromium_git') + '/external/github.com/tensorflow/tensorflow.git' + '@' + '38dd09551686ad00bc837f383b3a1db282079498',
+    Var('chromium_git') + '/external/github.com/tensorflow/tensorflow.git' + '@' + '204f897f9a511e62242fcbaac6be5fbbe9e2e245',
 
   'src/third_party/turbine/cipd': {
       'packages': [
@@ -808,15 +808,15 @@ deps = {
       'packages': [
         {
           'package': 'infra/tools/luci/cas/${{platform}}',
-          'version': 'git_revision:21504d3595ac903ef60bac732d885cd8e6ed92ba',
+          'version': 'git_revision:f769e04338c2db24c3697f00ede9cadbd3ee7582',
         },
         {
           'package': 'infra/tools/luci/isolate/${{platform}}',
-          'version': 'git_revision:21504d3595ac903ef60bac732d885cd8e6ed92ba',
+          'version': 'git_revision:f769e04338c2db24c3697f00ede9cadbd3ee7582',
         },
         {
           'package': 'infra/tools/luci/swarming/${{platform}}',
-          'version': 'git_revision:21504d3595ac903ef60bac732d885cd8e6ed92ba',
+          'version': 'git_revision:f769e04338c2db24c3697f00ede9cadbd3ee7582',
         }
       ],
       'dep_type': 'cipd',
@@ -852,7 +852,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/google/pthreadpool.git' + '@' + '9003ee6c137cea3b94161bd5c614fb43be523ee1',
 
   'src/third_party/xnnpack/src':
-    Var('chromium_git') + '/external/github.com/google/XNNPACK.git' + '@' + 'e8aad3cc4ffb7259c4aab6dfb922bfbef0ef8be0',
+    Var('chromium_git') + '/external/github.com/google/XNNPACK.git' + '@' + '066e3e72b05e56441aebaa57c57fad9280d869c8',
 
   'src/third_party/farmhash/src':
     Var('chromium_git') + '/external/github.com/google/farmhash.git' + '@' + '816a4ae622e964763ca0862d9dbd19324a1eaf45',
@@ -864,7 +864,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7364b490b5f78d58efe23ea76e74210fd6c3c76f',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + 'c4c704e5ddca35c10cca625807a6919654753459',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + 'd0d70a952766f6b02f449da040141529748ecd30',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From ebafa8817345b3a574a89ae765ef63d0c1e2eab8 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Wed, 25 Feb 2026 20:03:15 -0800
Subject: [PATCH 143/472] Roll chromium_revision 33dcdf76ee..da78729057
 (1590492:1590614)

Change log: https://chromium.googlesource.com/chromium/src/+log/33dcdf76ee..da78729057
Full diff: https://chromium.googlesource.com/chromium/src/+/33dcdf76ee..da78729057

Changed dependencies
* src/buildtools/linux64: git_revision:47b40ef49e678319c0dac767a95a363cc3f1f6cb..git_revision:6767e99aea9e7c32b6fc421ad3271ab99a82ca78
* src/buildtools/mac: git_revision:47b40ef49e678319c0dac767a95a363cc3f1f6cb..git_revision:6767e99aea9e7c32b6fc421ad3271ab99a82ca78
* src/buildtools/win: git_revision:47b40ef49e678319c0dac767a95a363cc3f1f6cb..git_revision:6767e99aea9e7c32b6fc421ad3271ab99a82ca78
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/14067ecb4b..ff68cbebda
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/3b1ff75cc0..87d06a6fe4
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/b000765734..842a66cb4d
* src/third_party/android_deps/autorolled/cipd: Lskjo0vFA222MYDpjYl71PwoWG_9mb-F_v3RTHvE-LkC..pmna6eE_w31koBiBbnzTBXKHTCuDkgeg1ezshh73OnEC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/626a2578ac..fa93d5c99f
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/d0d70a9527..a31de4778d
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/fad7939709..beeadce947
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/c9609d1e65..1364a55a55
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/5cb7e32d33..7562f57738
DEPS diff: https://chromium.googlesource.com/chromium/src/+/33dcdf76ee..da78729057/DEPS

BUG=None

Change-Id: I905546c58a20a8f8f236412fb5cd5843ec92e763
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452302
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#46996}
---
 DEPS | 26 +++++++++++++-------------
 1 file changed, 13 insertions(+), 13 deletions(-)

diff --git a/DEPS b/DEPS
index e0eff8484c..8367bba2fd 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '33dcdf76eebc97e39d04746dbde27bf2c06ccd9e',
+  'chromium_revision': 'da7872905751e7ad109b4aab6f564b650c2fae20',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -78,19 +78,19 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@14067ecb4b74487149d4eaa74f0ae40df9ee6d6d',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@ff68cbebdac7c38b5a04e5bfefd6341835dffe96',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@3b1ff75cc08b882e4fc18a461b17c0a44f5b815e',
+    'https://chromium.googlesource.com/chromium/src/testing@87d06a6fe4843026375a21de14bcd14e58859c01',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@b000765734c7b82c2ca5a3ee40dc274dcafbef31',
+    'https://chromium.googlesource.com/chromium/src/third_party@842a66cb4d7746c780df4806331b519323097967',
 
   'src/buildtools/linux64': {
     'packages': [
       {
         'package': 'gn/gn/linux-${{arch}}',
-        'version': 'git_revision:47b40ef49e678319c0dac767a95a363cc3f1f6cb',
+        'version': 'git_revision:6767e99aea9e7c32b6fc421ad3271ab99a82ca78',
       }
     ],
     'dep_type': 'cipd',
@@ -100,7 +100,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/mac-${{arch}}',
-        'version': 'git_revision:47b40ef49e678319c0dac767a95a363cc3f1f6cb',
+        'version': 'git_revision:6767e99aea9e7c32b6fc421ad3271ab99a82ca78',
       }
     ],
     'dep_type': 'cipd',
@@ -110,7 +110,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/windows-amd64',
-        'version': 'git_revision:47b40ef49e678319c0dac767a95a363cc3f1f6cb',
+        'version': 'git_revision:6767e99aea9e7c32b6fc421ad3271ab99a82ca78',
       }
     ],
     'dep_type': 'cipd',
@@ -488,7 +488,7 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@626a2578ac3e2840d4664562de39d25b1e710a19',
+    'https://boringssl.googlesource.com/boringssl.git@fa93d5c99f36f2c00fa341fb4f53d47a6f302da4',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@9f26d631e4063c0af3e8ee3bc8826486115372a8',
   'src/third_party/catapult':
@@ -511,7 +511,7 @@ deps = {
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@fad79397092a7029152f6ecb0712a4e00c15e51a',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@beeadce9470442e10465305f56cfbc5d13ec78bc',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'c9609d1e65bcd21014a708c0c8b8df094cd89a0b',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '1364a55a55c8bd81eeb32e9bd3bf91b10c461137',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@5cb7e32d338c947a9bd1a8db7659c07089c1ac52',
+    'https://chromium.googlesource.com/chromium/src/tools@7562f577389c3af5ef014c226897f46efd2b4062',
 
   'src/third_party/espresso': {
       'packages': [
@@ -842,7 +842,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/android_deps/autorolled',
-              'version': 'Lskjo0vFA222MYDpjYl71PwoWG_9mb-F_v3RTHvE-LkC',
+              'version': 'pmna6eE_w31koBiBbnzTBXKHTCuDkgeg1ezshh73OnEC',
           },
       ],
       'condition': 'checkout_android and non_git_source',
@@ -864,7 +864,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7364b490b5f78d58efe23ea76e74210fd6c3c76f',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + 'd0d70a952766f6b02f449da040141529748ecd30',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + 'a31de4778de4911ce4838a4cc042ae0276451499',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From 5984bc14283160f0a9927c9cd63586839903068e Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Wed, 25 Feb 2026 20:05:32 -0800
Subject: [PATCH 144/472] Update WebRTC code version (2026-02-26T04:05:18).

Bug: None
Change-Id: If6825c7a1f178a2093da00ee9761a1ff3b27bb05
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452303
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#46997}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index b3caa829ce..053366f854 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-02-25T04:08:12";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-02-26T04:05:18";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From 4a1f349771ded52d43bb61b16f580c8381033c11 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Thu, 26 Feb 2026 04:03:39 -0800
Subject: [PATCH 145/472] Roll chromium_revision da78729057..d49ece3195
 (1590614:1590759)

Change log: https://chromium.googlesource.com/chromium/src/+log/da78729057..d49ece3195
Full diff: https://chromium.googlesource.com/chromium/src/+/da78729057..d49ece3195

Changed dependencies
* src/buildtools: https://chromium.googlesource.com/chromium/src/buildtools/+log/451e71432a..136da69a12
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/ff68cbebda..87538efd40
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/87d06a6fe4..61da74e717
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/842a66cb4d..f309db1d4c
* src/third_party/androidx/cipd: mvV9ZifgIOHqOM9xEhhdmQJUTmj0fBmZciH8DOyvG88C..Pjbeih307Di9V8AT-ZSnNf8A118x4hWnhdLLideqj5wC
* src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/efa78c1067..ecffa351c7
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/a31de4778d..a95440de17
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/beeadce947..03052d67d5
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/1364a55a55..597706b827
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/7562f57738..c9c2c6b9fb
DEPS diff: https://chromium.googlesource.com/chromium/src/+/da78729057..d49ece3195/DEPS

BUG=None

Change-Id: I666e2338d899faa6daf9466c76e0417db7a666f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452380
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#46998}
---
 DEPS | 22 +++++++++++-----------
 1 file changed, 11 insertions(+), 11 deletions(-)

diff --git a/DEPS b/DEPS
index 8367bba2fd..e4f7d111c7 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': 'da7872905751e7ad109b4aab6f564b650c2fae20',
+  'chromium_revision': 'd49ece31956a31fb839d437b39921f63fbc4c285',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -71,20 +71,20 @@ deps = {
   'src/build':
     'https://chromium.googlesource.com/chromium/src/build@07efd23b7d9cf9b6a59076d14f5a68cd0c4c1fa7',
   'src/buildtools':
-    'https://chromium.googlesource.com/chromium/src/buildtools@451e71432ac9f6a91d4d5604c29e0a5078e8c2e8',
+    'https://chromium.googlesource.com/chromium/src/buildtools@136da69a1267b8db487354b96d44d0cc8add5aeb',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
   'src/examples/androidtests/third_party/gradle': {
     'url': 'https://chromium.googlesource.com/external/github.com/gradle/gradle.git@f2d1fb54a951d8b11d25748e4711bec8d128d7e3',
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@ff68cbebdac7c38b5a04e5bfefd6341835dffe96',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@87538efd40bd6fdc454e45e4aa22aed769a1d13d',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@87d06a6fe4843026375a21de14bcd14e58859c01',
+    'https://chromium.googlesource.com/chromium/src/testing@61da74e7174a660e6975d2143461ac55c68fbc8a',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@842a66cb4d7746c780df4806331b519323097967',
+    'https://chromium.googlesource.com/chromium/src/third_party@f309db1d4c0522b790e9b4f6f0b3672ac68490ae',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -331,7 +331,7 @@ deps = {
   'src/third_party/clang-format/script':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@c2725e0622e1a86d55f14514f2177a39efea4a0e',
   'src/third_party/compiler-rt/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@efa78c10673ca0ea0ab5c6c6165c4173117e6ed3',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@ecffa351c7a4c9e792a9a940c7c7bb3c4523eebe',
   'src/third_party/libc++/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@7ab65651aed6802d2599dcb7a73b1f82d5179d05',
   'src/third_party/libc++abi/src':
@@ -511,7 +511,7 @@ deps = {
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@beeadce9470442e10465305f56cfbc5d13ec78bc',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@03052d67d5d9fa1c3a4b003f6bbfd031aab803a3',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '1364a55a55c8bd81eeb32e9bd3bf91b10c461137',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '597706b827f0807c7b82180b1a636024224d8ce4',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@7562f577389c3af5ef014c226897f46efd2b4062',
+    'https://chromium.googlesource.com/chromium/src/tools@c9c2c6b9fbb76ace51494fa3bb6fd94bec3c253c',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'mvV9ZifgIOHqOM9xEhhdmQJUTmj0fBmZciH8DOyvG88C',
+          'version': 'Pjbeih307Di9V8AT-ZSnNf8A118x4hWnhdLLideqj5wC',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -864,7 +864,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7364b490b5f78d58efe23ea76e74210fd6c3c76f',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + 'a31de4778de4911ce4838a4cc042ae0276451499',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + 'a95440de174e5e4e0f979e829a56eda35af9e38f',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From 165c8e88cbb21a344f1c4128bddb567cc4870001 Mon Sep 17 00:00:00 2001
From: Philipp Hancke <philipp.hancke@googlemail.com>
Date: Thu, 26 Feb 2026 10:13:07 +0100
Subject: [PATCH 146/472] srtp: move GetRtpAuthParams to other external auth
 methods
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Bug: webrtc:357776213
Change-Id: I7829b6fe424699891494aaf5883509c867273a83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452360
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#46999}
---
 pc/srtp_session.cc | 33 +++++++++++++++++----------------
 pc/srtp_session.h  |  6 ++----
 2 files changed, 19 insertions(+), 20 deletions(-)

diff --git a/pc/srtp_session.cc b/pc/srtp_session.cc
index 59b840fd5d..5c64265d0f 100644
--- a/pc/srtp_session.cc
+++ b/pc/srtp_session.cc
@@ -323,6 +323,23 @@ bool SrtpSession::UnprotectRtcp(CopyOnWriteBuffer& buffer) {
   return true;
 }
 
+int SrtpSession::GetSrtpOverhead() const {
+  return rtp_auth_tag_len_;
+}
+
+void SrtpSession::EnableExternalAuth() {
+  RTC_DCHECK(!session_);
+  external_auth_enabled_ = true;
+}
+
+bool SrtpSession::IsExternalAuthEnabled() const {
+  return external_auth_enabled_;
+}
+
+bool SrtpSession::IsExternalAuthActive() const {
+  return external_auth_active_;
+}
+
 bool SrtpSession::GetRtpAuthParams(uint8_t** key, int* key_len, int* tag_len) {
   RTC_DCHECK(thread_checker_.IsCurrent());
   RTC_DCHECK(IsExternalAuthActive());
@@ -351,22 +368,6 @@ bool SrtpSession::GetRtpAuthParams(uint8_t** key, int* key_len, int* tag_len) {
   return true;
 }
 
-int SrtpSession::GetSrtpOverhead() const {
-  return rtp_auth_tag_len_;
-}
-
-void SrtpSession::EnableExternalAuth() {
-  RTC_DCHECK(!session_);
-  external_auth_enabled_ = true;
-}
-
-bool SrtpSession::IsExternalAuthEnabled() const {
-  return external_auth_enabled_;
-}
-
-bool SrtpSession::IsExternalAuthActive() const {
-  return external_auth_active_;
-}
 
 bool SrtpSession::RemoveSsrcFromSession(uint32_t ssrc) {
   RTC_DCHECK(session_);
diff --git a/pc/srtp_session.h b/pc/srtp_session.h
index af47b18f17..059e715eed 100644
--- a/pc/srtp_session.h
+++ b/pc/srtp_session.h
@@ -71,9 +71,6 @@ class SrtpSession {
   bool UnprotectRtp(CopyOnWriteBuffer& buffer);
   bool UnprotectRtcp(CopyOnWriteBuffer& buffer);
 
-  // Helper method to get authentication params.
-  bool GetRtpAuthParams(uint8_t** key, int* key_len, int* tag_len);
-
   int GetSrtpOverhead() const;
 
   // If external auth is enabled, SRTP will write a dummy auth tag that then
@@ -83,7 +80,8 @@ class SrtpSession {
   // have been set.
   void EnableExternalAuth();
   bool IsExternalAuthEnabled() const;
-
+  // Helper method to get authentication params.
+  bool GetRtpAuthParams(uint8_t** key, int* key_len, int* tag_len);
   // A SRTP session supports external creation of the auth tag if a non-GCM
   // cipher is used. This method is only valid after the RTP params have
   // been set.

From e7ad41ca4a59807cb14a9e378eaa7e1d687e6dd9 Mon Sep 17 00:00:00 2001
From: Jonas Oreland <jonaso@webrtc.org>
Date: Thu, 26 Feb 2026 07:05:02 -0800
Subject: [PATCH 147/472] DtlsInStun: Modify termination by application data

This is a follow up to https://webrtc-review.git.corp.google.com/c/src/+/45152
which I think(?) address the comment in
https://webrtc-review.git.corp.google.com/c/src/+/451521/comment/0e8a1639_699fdc3a

I.e. encrypted srtp is only pass though by the dtls layer, meaning we
need to check if we are writable before concluding that both are
writable.

Update method name to ApplicationPacketReceived to highlight
that it doesn't have to be decrypted.

Bug: webrtc:367395350
Change-Id: I37f2d0c2fdeb6cb33c1a042b30d2afdfe1524032
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452280
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Sameer Vijaykar <samvi@google.com>
Cr-Commit-Position: refs/heads/main@{#47000}
---
 p2p/dtls/dtls_stun_piggyback_controller.cc    | 15 ++++++--
 p2p/dtls/dtls_stun_piggyback_controller.h     |  8 ++--
 ...dtls_stun_piggyback_controller_unittest.cc | 38 ++++++++++++++++++-
 p2p/dtls/dtls_transport.cc                    |  2 +-
 4 files changed, 53 insertions(+), 10 deletions(-)

diff --git a/p2p/dtls/dtls_stun_piggyback_controller.cc b/p2p/dtls/dtls_stun_piggyback_controller.cc
index 19a6c9a8a7..2f056c9886 100644
--- a/p2p/dtls/dtls_stun_piggyback_controller.cc
+++ b/p2p/dtls/dtls_stun_piggyback_controller.cc
@@ -51,7 +51,7 @@ void DtlsStunPiggybackController::SetDtlsHandshakeComplete(bool is_dtls_client,
   state_ = State::PENDING;
 }
 
-void DtlsStunPiggybackController::DecryptedPacketReceived(
+void DtlsStunPiggybackController::ApplicationPacketReceived(
     const ReceivedIpPacket& packet) {
   RTC_DCHECK_RUN_ON(&sequence_checker_);
 
@@ -62,8 +62,17 @@ void DtlsStunPiggybackController::DecryptedPacketReceived(
   RTC_DCHECK(packet.decryption_info() == ReceivedIpPacket::kDtlsDecrypted ||
              packet.decryption_info() == ReceivedIpPacket::kSrtpEncrypted);
 
-  // We should be writable before this to happen.
-  RTC_DCHECK(state_ == State::PENDING);
+  if (packet.decryption_info() == ReceivedIpPacket::kDtlsDecrypted) {
+    // We should be writable before this to happen.
+    RTC_DCHECK(state_ == State::PENDING);
+  } else if (packet.decryption_info() == ReceivedIpPacket::kSrtpEncrypted) {
+    // Peer sending encrypted srtp mean that it must be writable,
+    // but we don't necessarily know that it's decodable. However, if
+    // we are also dtls-writable (PENDING) this means that we are complete.
+    if (state_ != State::PENDING) {
+      return;
+    }
+  }
   state_ = State::COMPLETE;
   CallCompleteCallback();
 }
diff --git a/p2p/dtls/dtls_stun_piggyback_controller.h b/p2p/dtls/dtls_stun_piggyback_controller.h
index 1d034dc1e0..72a2e124bd 100644
--- a/p2p/dtls/dtls_stun_piggyback_controller.h
+++ b/p2p/dtls/dtls_stun_piggyback_controller.h
@@ -68,10 +68,10 @@ class DtlsStunPiggybackController {
   // that peer can decode them).
   void SetDtlsHandshakeComplete(bool is_dtls_client, bool is_dtls13);
 
-  // Called by DtlsTransport when a packet has been received and decoded.
-  // This means that dtls is writable for the peer and that we have enough
-  // information to decode their packets.
-  void DecryptedPacketReceived(const ReceivedIpPacket& packet);
+  // Called by DtlsTransport when a packet has been received and passed
+  // to layers above us. This means that dtls is writable for the peer,
+  // and maybe we are complete.
+  void ApplicationPacketReceived(const ReceivedIpPacket& packet);
 
   // Called by DtlsTransport when DTLS failed.
   void SetDtlsFailed();
diff --git a/p2p/dtls/dtls_stun_piggyback_controller_unittest.cc b/p2p/dtls/dtls_stun_piggyback_controller_unittest.cc
index 3a18f2a7b1..8409202414 100644
--- a/p2p/dtls/dtls_stun_piggyback_controller_unittest.cc
+++ b/p2p/dtls/dtls_stun_piggyback_controller_unittest.cc
@@ -255,12 +255,46 @@ TEST_F(DtlsStunPiggybackControllerTest,
 
   // Post-handshake ACK
   EXPECT_CALL(*this, ClientCompleteCallback);
-  client_.DecryptedPacketReceived(
+  client_.ApplicationPacketReceived(
       packet_.CopyAndSet(ReceivedIpPacket::kDtlsDecrypted));
   EXPECT_EQ(client_.state(), State::COMPLETE);
 
   EXPECT_CALL(*this, ServerCompleteCallback);
-  server_.DecryptedPacketReceived(
+  server_.ApplicationPacketReceived(
+      packet_.CopyAndSet(ReceivedIpPacket::kSrtpEncrypted));
+  EXPECT_EQ(server_.state(), State::COMPLETE);
+}
+
+TEST_F(DtlsStunPiggybackControllerTest,
+       BasicHandshakeEarlySrtpDoesNotComplete) {
+  // Flight 1+2
+  SendClientToServerEmbedded(dtls_flight1, STUN_BINDING_REQUEST);
+  EXPECT_EQ(server_.state(), State::CONFIRMED);
+  SendServerToClientEmbedded(dtls_flight2, STUN_BINDING_RESPONSE);
+  EXPECT_EQ(client_.state(), State::CONFIRMED);
+
+  // Flight 3
+  SendClientToServerEmbedded(dtls_flight3, STUN_BINDING_REQUEST);
+  EXPECT_EQ(server_.state(), State::CONFIRMED);
+
+  // An srtp packet arriving before reaching PENDING state.
+  server_.ApplicationPacketReceived(
+      packet_.CopyAndSet(ReceivedIpPacket::kSrtpEncrypted));
+  EXPECT_EQ(server_.state(), State::CONFIRMED);
+
+  // Flight 4
+  SendServerToClientEmbedded(dtls_flight4, STUN_BINDING_RESPONSE);
+  EXPECT_EQ(server_.state(), State::PENDING);
+  EXPECT_EQ(client_.state(), State::PENDING);
+
+  // Post-handshake ACK
+  EXPECT_CALL(*this, ClientCompleteCallback);
+  client_.ApplicationPacketReceived(
+      packet_.CopyAndSet(ReceivedIpPacket::kDtlsDecrypted));
+  EXPECT_EQ(client_.state(), State::COMPLETE);
+
+  EXPECT_CALL(*this, ServerCompleteCallback);
+  server_.ApplicationPacketReceived(
       packet_.CopyAndSet(ReceivedIpPacket::kSrtpEncrypted));
   EXPECT_EQ(server_.state(), State::COMPLETE);
 }
diff --git a/p2p/dtls/dtls_transport.cc b/p2p/dtls/dtls_transport.cc
index eda8219d3f..d18c6f8581 100644
--- a/p2p/dtls/dtls_transport.cc
+++ b/p2p/dtls/dtls_transport.cc
@@ -526,7 +526,7 @@ bool DtlsTransportInternalImpl::SetupDtls() {
           &dtls_stun_piggyback_controller_,
           [this](webrtc::PacketTransportInternal* transport,
                  const ReceivedIpPacket& packet) {
-            dtls_stun_piggyback_controller_.DecryptedPacketReceived(packet);
+            dtls_stun_piggyback_controller_.ApplicationPacketReceived(packet);
           });
     }
     if (ssl_stream_factory_) {

From 12d717eb0cffc501d08e604752d58cf2075b10e5 Mon Sep 17 00:00:00 2001
From: Yuwei Huang <yuweih@chromium.org>
Date: Tue, 24 Feb 2026 14:59:32 -0800
Subject: [PATCH 148/472] Add SharedScreenCastStream::SetSharedMemoryFactory

Currently Chrome Remote Desktop needs to copy the BasicDesktopFrame into
a SharedMemoryDesktopFrame in order to pass it between processes. This
CL adds a new SetSharedMemoryFactory() method to SharedScreenCastStream,
which allows it to create a SharedMemoryDesktopFrame and copy the
content from the DMA buffer directly into the shared memory frame,
eliminating an unnecessary copy.

See http://crrev.com/c/7603221 for CRD's usage in Chromium.

Bug: chromium:475611769
Change-Id: I470b04241d3198ca9feb12e4fc502f96e7a4c13c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452020
Commit-Queue: Yuwei Huang <yuweih@chromium.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Auto-Submit: Yuwei Huang <yuweih@chromium.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47001}
---
 .../linux/wayland/shared_screencast_stream.cc | 21 +++++++++++++++++--
 .../linux/wayland/shared_screencast_stream.h  |  2 ++
 2 files changed, 21 insertions(+), 2 deletions(-)

diff --git a/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc b/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc
index f420038f09..44eab8fa33 100644
--- a/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc
+++ b/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc
@@ -98,6 +98,10 @@ class SharedScreenCastStreamPrivate {
   void SetObserver(SharedScreenCastStream::Observer* observer) {
     observer_ = observer;
   }
+  void SetSharedMemoryFactory(
+      std::unique_ptr<webrtc::SharedMemoryFactory> shared_memory_factory) {
+    shared_memory_factory_ = std::move(shared_memory_factory);
+  }
   void StopScreenCastStream();
   std::unique_ptr<SharedDesktopFrame> CaptureFrame();
   std::unique_ptr<MouseCursor> CaptureCursor();
@@ -129,6 +133,7 @@ class SharedScreenCastStreamPrivate {
 
   int64_t modifier_;
   std::unique_ptr<EglDmaBuf> egl_dmabuf_;
+  std::unique_ptr<SharedMemoryFactory> shared_memory_factory_;
 
   // PipeWire types
   std::unique_ptr<PipeWireInitializer> pw_initializer_;
@@ -891,8 +896,15 @@ void SharedScreenCastStreamPrivate::ProcessBuffer(pw_buffer* buffer) {
 
   if (!queue_.current_frame() ||
       !queue_.current_frame()->size().equals(frame_size_)) {
-    std::unique_ptr<DesktopFrame> frame(new BasicDesktopFrame(
-        DesktopSize(frame_size_.width(), frame_size_.height()), FOURCC_ARGB));
+    std::unique_ptr<DesktopFrame> frame;
+    if (shared_memory_factory_) {
+      frame = SharedMemoryDesktopFrame::Create(
+          DesktopSize(frame_size_.width(), frame_size_.height()), FOURCC_ARGB,
+          shared_memory_factory_.get());
+    } else {
+      frame = std::make_unique<BasicDesktopFrame>(
+          DesktopSize(frame_size_.width(), frame_size_.height()), FOURCC_ARGB);
+    }
     queue_.ReplaceCurrentFrame(SharedDesktopFrame::Wrap(std::move(frame)));
   }
 
@@ -1119,6 +1131,11 @@ void SharedScreenCastStream::SetObserver(
   private_->SetObserver(observer);
 }
 
+void SharedScreenCastStream::SetSharedMemoryFactory(
+    std::unique_ptr<webrtc::SharedMemoryFactory> shared_memory_factory) {
+  private_->SetSharedMemoryFactory(std::move(shared_memory_factory));
+}
+
 void SharedScreenCastStream::StopScreenCastStream() {
   private_->StopScreenCastStream();
 }
diff --git a/modules/desktop_capture/linux/wayland/shared_screencast_stream.h b/modules/desktop_capture/linux/wayland/shared_screencast_stream.h
index 529803dec5..48d5004bb1 100644
--- a/modules/desktop_capture/linux/wayland/shared_screencast_stream.h
+++ b/modules/desktop_capture/linux/wayland/shared_screencast_stream.h
@@ -70,6 +70,8 @@ class RTC_EXPORT SharedScreenCastStream
   void UpdateScreenCastStreamFrameRate(uint32_t frame_rate);
   void SetUseDamageRegion(bool use_damage_region);
   void SetObserver(SharedScreenCastStream::Observer* observer);
+  void SetSharedMemoryFactory(
+      std::unique_ptr<webrtc::SharedMemoryFactory> shared_memory_factory);
   void StopScreenCastStream();
 
   // Below functions return the most recent information we get from a

From f5d7e1fa1c743742d8be4dc8ffec63ab3c3d63ab Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Thu, 26 Feb 2026 20:08:42 -0800
Subject: [PATCH 149/472] Update WebRTC code version (2026-02-27T04:08:14).

Bug: None
Change-Id: If140faaa0192687fc532dfb9a27d55696d934835
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452521
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47002}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index 053366f854..b402f3879c 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-02-26T04:05:18";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-02-27T04:08:14";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From aa6e12a248f96ac725f445f248142268c9bc15ff Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Thu, 26 Feb 2026 09:56:36 +0000
Subject: [PATCH 150/472] [Plan B] Remove global deprecation suppression from
 rtc_test

and apply local markup.

This is part of the transition to ensure all Plan B usage in tests is
explicitly marked.

Modified:
- webrtc.gni: Removed automatic addition of suppress_plan_b_deprecation to rtc_test.
- pc/*_unittests.cc: Added suitable markup
- test/testsupport: Fixed an inappropriately suppressed deprecation

Bug: webrtc:467984837
Change-Id: I18197155d7e521634cd0738357e8504567bbdebd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452160
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47003}
---
 p2p/dtls/fake_dtls_transport.h                |   2 +-
 pc/BUILD.gn                                   |   2 +
 pc/legacy_stats_collector_unittest.cc         |  66 +++++++++-
 pc/media_session_unittest.cc                  |   3 +
 pc/peer_connection_integrationtest.cc         |   5 +
 pc/peer_connection_interface_unittest.cc      |  58 ++++++++-
 pc/peer_connection_rtp_unittest.cc            |  17 +++
 pc/rtc_stats_collector_unittest.cc            | 113 ++++++++++++++++++
 .../ivf_video_frame_generator_unittest.cc     |   4 +-
 webrtc.gni                                    |   3 -
 10 files changed, 262 insertions(+), 11 deletions(-)

diff --git a/p2p/dtls/fake_dtls_transport.h b/p2p/dtls/fake_dtls_transport.h
index aed27ce4b2..6444bf7be9 100644
--- a/p2p/dtls/fake_dtls_transport.h
+++ b/p2p/dtls/fake_dtls_transport.h
@@ -56,7 +56,7 @@ class FakeDtlsTransport : public DtlsTransportInternal {
             ice_transport_ref_->internal())),
         transport_name_(ice_transport_->transport_name()),
         component_(ice_transport_->component()),
-        dtls_fingerprint_("", nullptr) {
+        dtls_fingerprint_("", ArrayView<const uint8_t>()) {
     RTC_DCHECK(ice_transport_);
     ice_transport_->RegisterReceivedPacketCallback(
         this, [&](PacketTransportInternal* transport,
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 2c187836fc..0ac4ec4cf0 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -2328,6 +2328,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "../rtc_base/containers:flat_set",
       "../rtc_base/network:received_packet",
       "../rtc_base/network:sent_packet",
+      "../rtc_base/system:plan_b_only",
       "../system_wrappers:metrics",
       "../test:create_test_environment",
       "../test:create_test_field_trials",
@@ -2730,6 +2731,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "../rtc_base:weak_ptr",
       "../rtc_base/containers:flat_map",
       "../rtc_base/synchronization:mutex",
+      "../rtc_base/system:plan_b_only",
       "../system_wrappers",
       "../system_wrappers:metrics",
       "../test:audio_codec_mocks",
diff --git a/pc/legacy_stats_collector_unittest.cc b/pc/legacy_stats_collector_unittest.cc
index 7c860a8508..676b2eec49 100644
--- a/pc/legacy_stats_collector_unittest.cc
+++ b/pc/legacy_stats_collector_unittest.cc
@@ -59,6 +59,7 @@
 #include "rtc_base/socket_address.h"
 #include "rtc_base/ssl_identity.h"
 #include "rtc_base/ssl_stream_adapter.h"
+#include "rtc_base/system/plan_b_only.h"
 #include "rtc_base/thread.h"
 #include "system_wrappers/include/clock.h"
 #include "system_wrappers/include/ntp_time.h"
@@ -661,7 +662,9 @@ class LegacyStatsCollectorTest : public ::testing::Test {
     auto pc = CreatePeerConnection();
     auto stats = CreateStatsCollector(pc.get());
 
+    RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
     pc->AddVoiceChannel(/*mid=*/"audio", kTransportName);
+    RTC_ALLOW_PLAN_B_DEPRECATION_END();
 
     // Fake stats to process.
     TransportChannelStats channel_stats;
@@ -809,7 +812,9 @@ class StatsCollectorTrackTest : public LegacyStatsCollectorTest,
     } else {
       stats->AddTrack(video_track_.get());
     }
+    RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
     pc->AddSender(CreateMockSender(video_track_, kSsrcOfTrack));
+    RTC_ALLOW_PLAN_B_DEPRECATION_END();
   }
 
   // Adds a incoming video track with a given SSRC into the stats.
@@ -824,7 +829,9 @@ class StatsCollectorTrackTest : public LegacyStatsCollectorTest,
     } else {
       stats->AddTrack(video_track_.get());
     }
+    RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
     pc->AddReceiver(CreateMockReceiver(video_track_, kSsrcOfTrack));
+    RTC_ALLOW_PLAN_B_DEPRECATION_END();
   }
 
   // Adds a outgoing audio track with a given SSRC into the stats,
@@ -843,7 +850,10 @@ class StatsCollectorTrackTest : public LegacyStatsCollectorTest,
     } else {
       stats->AddTrack(audio_track_.get());
     }
-    return pc->AddSender(CreateMockSender(audio_track_, kSsrcOfTrack));
+    RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
+    auto sender = pc->AddSender(CreateMockSender(audio_track_, kSsrcOfTrack));
+    RTC_ALLOW_PLAN_B_DEPRECATION_END();
+    return sender;
   }
 
   // Adds a incoming audio track with a given SSRC into the stats.
@@ -858,7 +868,9 @@ class StatsCollectorTrackTest : public LegacyStatsCollectorTest,
     } else {
       stats->AddTrack(audio_track_.get());
     }
+    RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
     pc->AddReceiver(CreateMockReceiver(audio_track_, kSsrcOfTrack));
+    RTC_ALLOW_PLAN_B_DEPRECATION_END();
   }
 
   scoped_refptr<AudioTrackInterface> audio_track() { return audio_track_; }
@@ -995,7 +1007,9 @@ TEST_P(StatsCollectorTrackTest, BytesCounterHandles64Bits) {
   VideoMediaInfo video_info;
   video_info.aggregated_senders.push_back(video_sender_info);
 
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc->AddVideoChannel("video", "transport", video_info);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
 
   AddOutgoingVideoTrack(pc.get(), stats.get());
 
@@ -1027,7 +1041,9 @@ TEST_P(StatsCollectorTrackTest, AudioBandwidthEstimationInfoIsReported) {
   VoiceMediaInfo voice_info;
   voice_info.senders.push_back(voice_sender_info);
 
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   auto voice_media_channels = pc->AddVoiceChannel("audio", "transport");
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   voice_media_channels.first->SetStats(voice_info);
   voice_media_channels.second->SetStats(voice_info);
 
@@ -1078,7 +1094,9 @@ TEST_P(StatsCollectorTrackTest, VideoBandwidthEstimationInfoIsReported) {
   VideoMediaInfo video_info;
   video_info.aggregated_senders.push_back(video_sender_info);
 
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc->AddVideoChannel("video", "transport", video_info);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
 
   AddOutgoingVideoTrack(pc.get(), stats.get());
 
@@ -1144,7 +1162,9 @@ TEST_P(StatsCollectorTrackTest, TrackObjectExistsWithoutUpdateStats) {
   auto pc = CreatePeerConnection();
   auto stats = CreateStatsCollector(pc.get());
 
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc->AddVideoChannel("video", "transport");
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   AddOutgoingVideoTrack(pc.get(), stats.get());
 
   // Verfies the existence of the track report.
@@ -1175,7 +1195,9 @@ TEST_P(StatsCollectorTrackTest, TrackAndSsrcObjectExistAfterUpdateSsrcStats) {
   VideoMediaInfo video_info;
   video_info.aggregated_senders.push_back(video_sender_info);
 
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc->AddVideoChannel("video", "transport", video_info);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
 
   AddOutgoingVideoTrack(pc.get(), stats.get());
 
@@ -1229,7 +1251,9 @@ TEST_P(StatsCollectorTrackTest, TransportObjectLinkedFromSsrcObject) {
   VideoMediaInfo video_info;
   video_info.aggregated_senders.push_back(video_sender_info);
 
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc->AddVideoChannel("video", "transport", video_info);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
 
   AddOutgoingVideoTrack(pc.get(), stats.get());
 
@@ -1264,7 +1288,9 @@ TEST_P(StatsCollectorTrackTest, RemoteSsrcInfoIsAbsent) {
   auto pc = CreatePeerConnection();
   auto stats = CreateStatsCollector(pc.get());
 
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc->AddVideoChannel("video", "transport");
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   AddOutgoingVideoTrack(pc.get(), stats.get());
 
   stats->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
@@ -1291,7 +1317,9 @@ TEST_P(StatsCollectorTrackTest, RemoteSsrcInfoIsPresent) {
   VideoMediaInfo video_info;
   video_info.aggregated_senders.push_back(video_sender_info);
 
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc->AddVideoChannel("video", "transport", video_info);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
 
   AddOutgoingVideoTrack(pc.get(), stats.get());
 
@@ -1319,7 +1347,9 @@ TEST_P(StatsCollectorTrackTest, ReportsFromRemoteTrack) {
   VideoMediaInfo video_info;
   video_info.receivers.push_back(video_receiver_info);
 
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc->AddVideoChannel("video", "transport", video_info);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
 
   AddIncomingVideoTrack(pc.get(), stats.get());
 
@@ -1384,7 +1414,9 @@ TEST_F(LegacyStatsCollectorTest, IceCandidateReport) {
   TransportChannelStats channel_stats;
   channel_stats.ice_transport_stats.connection_infos.push_back(connection_info);
 
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc->AddVoiceChannel("audio", kTransportName);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   pc->SetTransportStats(kTransportName, channel_stats);
 
   stats->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
@@ -1500,7 +1532,9 @@ TEST_F(LegacyStatsCollectorTest, NoTransport) {
 
   // This will cause the fake PeerConnection to generate a TransportStats entry
   // but with only a single dummy TransportChannelStats.
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc->AddVoiceChannel("audio", "transport");
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
 
   stats->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
   StatsReports reports;
@@ -1562,7 +1596,9 @@ TEST_P(StatsCollectorTrackTest, FilterOutNegativeInitialValues) {
   auto local_track =
       make_ref_counted<FakeAudioTrackWithInitValue>(kLocalTrackId);
   stream_->AddTrack(scoped_refptr<AudioTrackInterface>(local_track.get()));
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc->AddSender(CreateMockSender(local_track, kSsrcOfTrack));
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   if (GetParam()) {
     stats->AddStream(stream_.get());
   }
@@ -1574,7 +1610,9 @@ TEST_P(StatsCollectorTrackTest, FilterOutNegativeInitialValues) {
   scoped_refptr<AudioTrackInterface> remote_track =
       make_ref_counted<FakeAudioTrackWithInitValue>(kRemoteTrackId);
   remote_stream->AddTrack(remote_track);
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc->AddReceiver(CreateMockReceiver(remote_track, kSsrcOfTrack));
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   if (GetParam()) {
     stats->AddStream(remote_stream.get());
   }
@@ -1602,7 +1640,9 @@ TEST_P(StatsCollectorTrackTest, FilterOutNegativeInitialValues) {
   voice_info.senders.push_back(voice_sender_info);
   voice_info.receivers.push_back(voice_receiver_info);
 
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   auto voice_media_channels = pc->AddVoiceChannel("voice", "transport");
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   voice_media_channels.first->SetStats(voice_info);
   voice_media_channels.second->SetStats(voice_info);
 
@@ -1655,7 +1695,9 @@ TEST_P(StatsCollectorTrackTest, GetStatsFromLocalAudioTrack) {
   VoiceMediaInfo voice_info;
   voice_info.senders.push_back(voice_sender_info);
 
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   auto voice_media_channels = pc->AddVoiceChannel("audio", "transport");
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   voice_media_channels.first->SetStats(voice_info);
   voice_media_channels.second->SetStats(voice_info);
 
@@ -1683,7 +1725,9 @@ TEST_P(StatsCollectorTrackTest, GetStatsFromRemoteStream) {
   VoiceMediaInfo voice_info;
   voice_info.receivers.push_back(voice_receiver_info);
 
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   auto voice_media_channels = pc->AddVoiceChannel("audio", "transport");
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   voice_media_channels.first->SetStats(voice_info);
   voice_media_channels.second->SetStats(voice_info);
 
@@ -1705,7 +1749,9 @@ TEST_P(StatsCollectorTrackTest, GetStatsAfterRemoveAudioStream) {
   VoiceMediaInfo voice_info;
   voice_info.senders.push_back(voice_sender_info);
 
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   auto voice_media_channels = pc->AddVoiceChannel("audio", "transport");
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   voice_media_channels.first->SetStats(voice_info);
   voice_media_channels.second->SetStats(voice_info);
 
@@ -1748,7 +1794,9 @@ TEST_P(StatsCollectorTrackTest, LocalAndRemoteTracksWithSameSsrc) {
       MediaStream::Create("remotestreamid"));
   scoped_refptr<AudioTrackInterface> remote_track =
       make_ref_counted<FakeAudioTrack>(kRemoteTrackId);
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc->AddReceiver(CreateMockReceiver(remote_track, kSsrcOfTrack));
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   remote_stream->AddTrack(remote_track);
   stats->AddStream(remote_stream.get());
 
@@ -1768,7 +1816,9 @@ TEST_P(StatsCollectorTrackTest, LocalAndRemoteTracksWithSameSsrc) {
   voice_info.receivers.push_back(voice_receiver_info);
 
   // Instruct the session to return stats containing the transport channel.
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   auto voice_media_channels = pc->AddVoiceChannel("audio", "transport");
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   voice_media_channels.first->SetStats(voice_info);
   voice_media_channels.second->SetStats(voice_info);
 
@@ -1825,7 +1875,9 @@ TEST_P(StatsCollectorTrackTest, TwoLocalTracksWithSameSsrc) {
   VoiceMediaInfo voice_info;
   voice_info.senders.push_back(voice_sender_info);
 
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   auto voice_media_channels = pc->AddVoiceChannel("voice", "transport");
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   voice_media_channels.first->SetStats(voice_info);
   voice_media_channels.second->SetStats(voice_info);
 
@@ -1835,12 +1887,16 @@ TEST_P(StatsCollectorTrackTest, TwoLocalTracksWithSameSsrc) {
   // Remove the previous audio track from the stream.
   stream_->RemoveTrack(audio_track());
   stats->RemoveLocalAudioTrack(audio_track_.get(), kSsrcOfTrack);
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc->RemoveSender(sender);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
 
   // Create a new audio track and adds it to the stream and stats.
   static const std::string kNewTrackId = "new_track_id";
   auto new_audio_track = make_ref_counted<FakeAudioTrack>(kNewTrackId);
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc->AddSender(CreateMockSender(new_audio_track, kSsrcOfTrack));
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   stream_->AddTrack(scoped_refptr<AudioTrackInterface>(new_audio_track.get()));
 
   stats->AddLocalAudioTrack(new_audio_track.get(), kSsrcOfTrack);
@@ -1872,9 +1928,11 @@ TEST_P(StatsCollectorTrackTest, TwoLocalSendersWithSameTrack) {
 
   auto local_track =
       make_ref_counted<FakeAudioTrackWithInitValue>(kLocalTrackId);
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc->AddSender(CreateMockSender(local_track, kFirstSsrc));
   stats->AddLocalAudioTrack(local_track.get(), kFirstSsrc);
   pc->AddSender(CreateMockSender(local_track, kSecondSsrc));
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   stats->AddLocalAudioTrack(local_track.get(), kSecondSsrc);
 
   VoiceSenderInfo first_sender_info;
@@ -1891,7 +1949,9 @@ TEST_P(StatsCollectorTrackTest, TwoLocalSendersWithSameTrack) {
   voice_info.senders.push_back(first_sender_info);
   voice_info.senders.push_back(second_sender_info);
 
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   auto voice_media_channels = pc->AddVoiceChannel("voice", "transport");
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   voice_media_channels.first->SetStats(voice_info);
   voice_media_channels.second->SetStats(voice_info);
 
@@ -1940,7 +2000,9 @@ TEST_P(StatsCollectorTrackTest, VerifyVideoSendSsrcStats) {
   VideoMediaInfo video_info;
   video_info.aggregated_senders.push_back(video_sender_info);
 
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc->AddVideoChannel("video", "transport", video_info);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
 
   stats->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
   StatsReports reports;
@@ -1967,7 +2029,9 @@ TEST_P(StatsCollectorTrackTest, VerifyVideoReceiveSsrcStatsNew) {
   VideoMediaInfo video_info;
   video_info.receivers.push_back(video_receiver_info);
 
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc->AddVideoChannel("video", "transport", video_info);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
 
   stats->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
   StatsReports reports;
diff --git a/pc/media_session_unittest.cc b/pc/media_session_unittest.cc
index 095912fae0..3c97818fe1 100644
--- a/pc/media_session_unittest.cc
+++ b/pc/media_session_unittest.cc
@@ -61,6 +61,7 @@
 #include "rtc_base/ssl_identity.h"
 #include "rtc_base/string_encode.h"
 #include "rtc_base/strings/string_builder.h"
+#include "rtc_base/system/plan_b_only.h"
 #include "rtc_base/unique_id_generator.h"
 #include "test/create_test_field_trials.h"
 #include "test/gmock.h"
@@ -546,6 +547,7 @@ void AttachSenderToMediaDescriptionOptions(
     int num_sim_layer,
     MediaSessionOptions* session_options) {
   auto it = FindFirstMediaDescriptionByMid(mid, session_options);
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   switch (type) {
     case webrtc::MediaType::AUDIO:
       it->AddAudioSender(track_id, stream_ids);
@@ -557,6 +559,7 @@ void AttachSenderToMediaDescriptionOptions(
     default:
       RTC_DCHECK_NOTREACHED();
   }
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
 }
 
 void AttachSenderToMediaDescriptionOptions(
diff --git a/pc/peer_connection_integrationtest.cc b/pc/peer_connection_integrationtest.cc
index da06cb4181..579e34d4a3 100644
--- a/pc/peer_connection_integrationtest.cc
+++ b/pc/peer_connection_integrationtest.cc
@@ -86,6 +86,7 @@
 #include "rtc_base/ssl_identity.h"
 #include "rtc_base/ssl_stream_adapter.h"
 #include "rtc_base/string_encode.h"
+#include "rtc_base/system/plan_b_only.h"
 #include "rtc_base/task_queue_for_test.h"
 #include "rtc_base/test_certificate_verifier.h"
 #include "rtc_base/virtual_socket_server.h"
@@ -2451,6 +2452,7 @@ TEST_P(PeerConnectionIntegrationTest,
 // received end-to-end.
 TEST_F(PeerConnectionIntegrationTestPlanB,
        MediaFlowsAfterEarlyWarmupWithCreateSender) {
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   auto caller_audio_sender =
@@ -2461,6 +2463,7 @@ TEST_F(PeerConnectionIntegrationTestPlanB,
       callee()->pc()->CreateSender("audio", "callee_stream");
   auto callee_video_sender =
       callee()->pc()->CreateSender("video", "callee_stream");
+  RTC_ALLOW_PLAN_B_DEPRECATION_END()
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_THAT(WaitUntil([&] { return SignalingStateStable(); }, IsTrue(),
                         {.timeout = kMaxWaitForActivation}),
@@ -2542,6 +2545,7 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
 // from the caller to the callee, rather than being forwarded to a third
 // PeerConnection.
 TEST_F(PeerConnectionIntegrationTestPlanB, CanSendRemoteVideoTrack) {
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   // Just send a video track from the caller.
@@ -2555,6 +2559,7 @@ TEST_F(PeerConnectionIntegrationTestPlanB, CanSendRemoteVideoTrack) {
   // Echo the stream back, and do a new offer/anwer (initiated by callee this
   // time).
   callee()->pc()->AddStream(callee()->remote_streams()->at(0));
+  RTC_ALLOW_PLAN_B_DEPRECATION_END()
   callee()->CreateAndSetAndSignalOffer();
   ASSERT_THAT(WaitUntil([&] { return SignalingStateStable(); }, IsTrue(),
                         {.timeout = kMaxWaitForActivation}),
diff --git a/pc/peer_connection_interface_unittest.cc b/pc/peer_connection_interface_unittest.cc
index 36c3e9ea43..15f4d319a5 100644
--- a/pc/peer_connection_interface_unittest.cc
+++ b/pc/peer_connection_interface_unittest.cc
@@ -86,6 +86,7 @@
 #include "rtc_base/rtc_certificate_generator.h"
 #include "rtc_base/socket_address.h"
 #include "rtc_base/socket_server.h"
+#include "rtc_base/system/plan_b_only.h"
 #include "rtc_base/thread.h"
 #include "rtc_base/virtual_socket_server.h"
 #include "rtc_base/weak_ptr.h"
@@ -826,7 +827,9 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test {
     scoped_refptr<MediaStreamInterface> stream(
         pc_factory_->CreateLocalMediaStream(label));
     stream->AddTrack(CreateVideoTrack(label + "v0"));
+    RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
     ASSERT_TRUE(pc_->AddStream(stream.get()));
+    RTC_ALLOW_PLAN_B_DEPRECATION_END();
   }
 
   scoped_refptr<AudioTrackInterface> CreateAudioTrack(
@@ -845,7 +848,9 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test {
     scoped_refptr<MediaStreamInterface> stream(
         pc_factory_->CreateLocalMediaStream(label));
     stream->AddTrack(CreateAudioTrack(label + "a0"));
+    RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
     ASSERT_TRUE(pc_->AddStream(stream.get()));
+    RTC_ALLOW_PLAN_B_DEPRECATION_END();
   }
 
   void AddAudioVideoStream(const std::string& stream_id,
@@ -856,7 +861,9 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test {
         pc_factory_->CreateLocalMediaStream(stream_id));
     stream->AddTrack(CreateAudioTrack(audio_track_label));
     stream->AddTrack(CreateVideoTrack(video_track_label));
+    RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
     ASSERT_TRUE(pc_->AddStream(stream.get()));
+    RTC_ALLOW_PLAN_B_DEPRECATION_END();
   }
 
   scoped_refptr<RtpReceiverInterface> GetFirstReceiverOfType(
@@ -929,8 +936,12 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test {
   // be required.
   bool DoGetStats(MediaStreamTrackInterface* track) {
     auto observer = make_ref_counted<MockStatsObserver>();
-    if (!pc_->GetStats(observer.get(), track,
-                       PeerConnectionInterface::kStatsOutputLevelStandard))
+    RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
+    bool result =
+        pc_->GetStats(observer.get(), track,
+                      PeerConnectionInterface::kStatsOutputLevelStandard);
+    RTC_ALLOW_PLAN_B_DEPRECATION_END();
+    if (!result)
       return false;
     EXPECT_THAT(WaitUntil([&] { return observer->called(); }, IsTrue(),
                           {.timeout = TimeDelta::Millis(kTimeout)}),
@@ -1088,6 +1099,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test {
   // Waits until a remote stream with the given id is signaled. This helper
   // function will verify both OnAddTrack and OnAddStream (Plan B only) are
   // called with the given stream id and expected number of tracks.
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
   void WaitAndVerifyOnAddStream(const std::string& stream_id,
                                 int expected_num_tracks) {
     // Verify that both OnAddStream and OnAddTrack are called.
@@ -1101,6 +1113,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test {
             Eq(expected_num_tracks), {.timeout = TimeDelta::Millis(kTimeout)}),
         IsRtcOk());
   }
+  RTC_ALLOW_PLAN_B_DEPRECATION_END()
 
   // Creates an offer and applies it as a local session description.
   // Creates an answer with the same SDP an the offer but removes all lines
@@ -1297,12 +1310,17 @@ class PeerConnectionInterfaceTest
   PeerConnectionInterfaceTest() : PeerConnectionInterfaceBaseTest(GetParam()) {}
 };
 
-class PeerConnectionInterfaceTestPlanB
+RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
+// Plan B specific tests.
+// Note - since the class is PLAN_B_ONLY, instances must be bracketed by
+// RTC_ALLOW_PLAN_B_DEPRECATION macros.
+class PLAN_B_ONLY PeerConnectionInterfaceTestPlanB
     : public PeerConnectionInterfaceBaseTest {
  protected:
   PeerConnectionInterfaceTestPlanB()
       : PeerConnectionInterfaceBaseTest(SdpSemantics::kPlanB_DEPRECATED) {}
 };
+RTC_ALLOW_PLAN_B_DEPRECATION_END()
 
 // Generate different CNAMEs when PeerConnections are created.
 // The CNAMEs are expected to be generated randomly. It is possible
@@ -1473,6 +1491,7 @@ TEST_P(PeerConnectionInterfaceTest, SetConfigurationFailsAfterClose) {
       pc_->SetConfiguration(PeerConnectionInterface::RTCConfiguration()).ok());
 }
 
+RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
 TEST_F(PeerConnectionInterfaceTestPlanB, AddStreams) {
   CreatePeerConnectionWithoutDtls();
   AddVideoStream(kStreamId1);
@@ -1648,6 +1667,7 @@ TEST_F(PeerConnectionInterfaceTestPlanB, AddTrackWithSendEncodings) {
   EXPECT_TRUE(pc_->RemoveTrackOrError(audio_sender).ok());
   EXPECT_TRUE(pc_->RemoveTrackOrError(video_sender).ok());
 }
+RTC_ALLOW_PLAN_B_DEPRECATION_END()
 
 // Test creating senders without a stream specified,
 // expecting a random stream ID to be generated.
@@ -1711,6 +1731,7 @@ TEST_P(PeerConnectionInterfaceTest, AttachmentIdIsSetOnAddTrack) {
   EXPECT_NE(0, video_sender_proxy->internal()->AttachmentId());
 }
 
+RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
 // Don't run under Unified Plan since the stream API is not available.
 TEST_F(PeerConnectionInterfaceTestPlanB, AttachmentIdIsSetOnAddStream) {
   CreatePeerConnection();
@@ -1722,6 +1743,7 @@ TEST_F(PeerConnectionInterfaceTestPlanB, AttachmentIdIsSetOnAddStream) {
           senders[0].get());
   EXPECT_NE(0, sender_proxy->internal()->AttachmentId());
 }
+RTC_ALLOW_PLAN_B_DEPRECATION_END()
 
 TEST_P(PeerConnectionInterfaceTest, CreateOfferReceiveAnswer) {
   InitiateCall();
@@ -1760,6 +1782,7 @@ TEST_P(PeerConnectionInterfaceTest, ReceiveOfferCreatePrAnswerAndAnswer) {
   WaitAndVerifyOnAddStream(kStreamId1, 1);
 }
 
+RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
 // Don't run under Unified Plan since the stream API is not available.
 TEST_F(PeerConnectionInterfaceTestPlanB, Renegotiate) {
   InitiateCall();
@@ -1784,6 +1807,7 @@ TEST_F(PeerConnectionInterfaceTestPlanB, RenegotiateAudioOnly) {
   CreateOfferReceiveAnswer();
   EXPECT_EQ(0u, pc_->remote_streams()->count());
 }
+RTC_ALLOW_PLAN_B_DEPRECATION_END()
 
 // Test that candidates are generated and that we can parse our own candidates.
 TEST_P(PeerConnectionInterfaceTest, IceCandidates) {
@@ -1812,6 +1836,7 @@ TEST_P(PeerConnectionInterfaceTest, IceCandidates) {
   EXPECT_TRUE(pc_->AddIceCandidate(observer_.last_candidate()));
 }
 
+RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
 // Test that CreateOffer and CreateAnswer will fail if the track labels are
 // not unique.
 TEST_F(PeerConnectionInterfaceTestPlanB, CreateOfferAnswerWithInvalidStream) {
@@ -1833,6 +1858,7 @@ TEST_F(PeerConnectionInterfaceTestPlanB, CreateOfferAnswerWithInvalidStream) {
   std::unique_ptr<SessionDescriptionInterface> answer;
   EXPECT_FALSE(DoCreateAnswer(&answer, nullptr));
 }
+RTC_ALLOW_PLAN_B_DEPRECATION_END()
 
 // Test that we will get different SSRCs for each tracks in the offer and answer
 // we created.
@@ -1866,6 +1892,7 @@ TEST_P(PeerConnectionInterfaceTest, SsrcInOfferAnswer) {
   EXPECT_NE(audio_ssrc, video_ssrc);
 }
 
+RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
 // Test that it's possible to call AddTrack on a MediaStream after adding
 // the stream to a PeerConnection.
 // TODO(deadbeef): Remove this test once this behavior is no longer supported.
@@ -1926,6 +1953,7 @@ TEST_F(PeerConnectionInterfaceTestPlanB, CreateSenderWithStream) {
   ASSERT_EQ(1u, video_desc->streams().size());
   EXPECT_EQ(kStreamId1, video_desc->streams()[0].first_stream_id());
 }
+RTC_ALLOW_PLAN_B_DEPRECATION_END()
 
 // Test that we can specify a certain track that we want statistics about.
 TEST_P(PeerConnectionInterfaceTest, GetStatsForSpecificTrack) {
@@ -2588,8 +2616,10 @@ TEST_P(PeerConnectionInterfaceTest, CloseAndTestStreamsAndStates) {
   // With Plan B, verify the stream count. The analog with Unified Plan is the
   // RtpTransceiver count.
   if (sdp_semantics_ == SdpSemantics::kPlanB_DEPRECATED) {
+    RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
     ASSERT_EQ(1u, pc_->local_streams()->count());
     ASSERT_EQ(1u, pc_->remote_streams()->count());
+    RTC_ALLOW_PLAN_B_DEPRECATION_END();
   } else {
     ASSERT_EQ(2u, pc_->GetTransceivers().size());
   }
@@ -2603,8 +2633,10 @@ TEST_P(PeerConnectionInterfaceTest, CloseAndTestStreamsAndStates) {
             pc_->ice_gathering_state());
 
   if (sdp_semantics_ == SdpSemantics::kPlanB_DEPRECATED) {
+    RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
     EXPECT_EQ(1u, pc_->local_streams()->count());
     EXPECT_EQ(1u, pc_->remote_streams()->count());
+    RTC_ALLOW_PLAN_B_DEPRECATION_END();
   } else {
     // Verify that the RtpTransceivers are still returned.
     EXPECT_EQ(2u, pc_->GetTransceivers().size());
@@ -2630,6 +2662,7 @@ TEST_P(PeerConnectionInterfaceTest, CloseAndTestStreamsAndStates) {
   }
 }
 
+RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
 // Test that PeerConnection methods fails gracefully after
 // PeerConnection::Close has been called.
 // Don't run under Unified Plan since the stream API is not available.
@@ -2668,6 +2701,7 @@ TEST_F(PeerConnectionInterfaceTestPlanB, CloseAndTestMethods) {
       CreateSessionDescription(SdpType::kOffer, sdp));
   EXPECT_FALSE(DoSetLocalDescription(std::move(local_offer)));
 }
+RTC_ALLOW_PLAN_B_DEPRECATION_END()
 
 // Test that GetStats can still be called after PeerConnection::Close.
 TEST_P(PeerConnectionInterfaceTest, CloseAndGetStats) {
@@ -2690,9 +2724,11 @@ TEST_P(PeerConnectionInterfaceTest, UpdateRemoteStreams) {
 
   scoped_refptr<StreamCollection> reference(
       CreateStreamCollection(1, 1, worker_thread_.get()));
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
   EXPECT_TRUE(
       CompareStreamCollections(observer_.remote_streams(), reference.get()));
   MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   EXPECT_TRUE(remote_stream->GetVideoTracks()[0]->GetSource() != nullptr);
 
   // Create a session description based on another SDP with another
@@ -2701,10 +2737,13 @@ TEST_P(PeerConnectionInterfaceTest, UpdateRemoteStreams) {
 
   scoped_refptr<StreamCollection> reference2(
       CreateStreamCollection(2, 1, worker_thread_.get()));
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   EXPECT_TRUE(
       CompareStreamCollections(observer_.remote_streams(), reference2.get()));
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
 }
 
+RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
 // This test verifies that when remote tracks are added/removed from SDP, the
 // created remote streams are updated appropriately.
 // Don't run under Unified Plan since this test uses Plan B SDP to test Plan B
@@ -2753,6 +2792,7 @@ TEST_F(PeerConnectionInterfaceTestPlanB,
                         {.timeout = TimeDelta::Millis(kTimeout)}),
               IsRtcOk());
 }
+RTC_ALLOW_PLAN_B_DEPRECATION_END()
 
 // This tests that remote tracks are ended if a local session description is set
 // that rejects the media content type.
@@ -2805,6 +2845,7 @@ TEST_P(PeerConnectionInterfaceTest, RejectMediaContent) {
               IsRtcOk());
 }
 
+RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
 // This tests that we won't crash if the remote track has been removed outside
 // of PeerConnection and then PeerConnection tries to reject the track.
 // Don't run under Unified Plan since the stream API is not available.
@@ -2829,6 +2870,7 @@ TEST_F(PeerConnectionInterfaceTestPlanB, RemoveTrackThenRejectMediaContent) {
 
   // No crash is a pass.
 }
+RTC_ALLOW_PLAN_B_DEPRECATION_END()
 
 // This tests that if a recvonly remote description is set, no remote streams
 // will be created, even if the description contains SSRCs/MSIDs.
@@ -2841,9 +2883,12 @@ TEST_P(PeerConnectionInterfaceTest, RecvonlyDescriptionDoesntCreateStream) {
   absl::StrReplaceAll({{kSendrecv, kRecvonly}}, &recvonly_offer);
   CreateAndSetRemoteOffer(recvonly_offer);
 
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   EXPECT_EQ(0u, observer_.remote_streams()->count());
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
 }
 
+RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
 // This tests that a default MediaStream is created if a remote session
 // description doesn't contain any streams and no MSID support.
 // It also tests that the default stream is updated if a video m-line is added
@@ -3094,6 +3139,7 @@ TEST_F(PeerConnectionInterfaceTestPlanB,
   EXPECT_TRUE(ContainsSender(senders, kAudioTracks[1]));
   EXPECT_TRUE(ContainsSender(senders, kVideoTracks[1]));
 }
+RTC_ALLOW_PLAN_B_DEPRECATION_END()
 
 // This tests that the expected behavior occurs if the SSRC on a local track is
 // changed when SetLocalDescription is called.
@@ -3145,6 +3191,7 @@ TEST_P(PeerConnectionInterfaceTest,
   // changed.
 }
 
+RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
 // This tests that the expected behavior occurs if a new session description is
 // set with the same tracks, but on a different MediaStream.
 // Don't run under Unified Plan since the stream API is not available.
@@ -3183,6 +3230,7 @@ TEST_F(PeerConnectionInterfaceTestPlanB,
   EXPECT_TRUE(ContainsSender(new_senders, kAudioTracks[0], kStreams[1]));
   EXPECT_TRUE(ContainsSender(new_senders, kVideoTracks[0], kStreams[1]));
 }
+RTC_ALLOW_PLAN_B_DEPRECATION_END()
 
 // This tests that PeerConnectionObserver::OnAddTrack is correctly called.
 TEST_P(PeerConnectionInterfaceTest, OnAddTrackCallback) {
@@ -3499,6 +3547,7 @@ TEST_P(PeerConnectionInterfaceTest, CreateOfferWithOfferToReceiveConstraints) {
   EXPECT_FALSE(video->rejected);
 }
 
+RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
 // Test that if CreateAnswer is called with the deprecated "offer to receive
 // audio/video" constraints, they're processed and can be used to reject an
 // offered m= section just as can be done with RTCOfferAnswerOptions;
@@ -3530,6 +3579,7 @@ TEST_F(PeerConnectionInterfaceTestPlanB,
   EXPECT_TRUE(audio->rejected);
   EXPECT_TRUE(video->rejected);
 }
+RTC_ALLOW_PLAN_B_DEPRECATION_END()
 
 // Test that negotiation can succeed with a data channel only, and with the max
 // bundle policy. Previously there was a bug that prevented this.
@@ -3782,6 +3832,7 @@ TEST_P(PeerConnectionInterfaceTest, CreateOfferWithRtpMux) {
   EXPECT_FALSE(offer->description()->HasGroup(GROUP_TYPE_BUNDLE));
 }
 
+RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
 // This test ensures OnRenegotiationNeeded is called when we add track with
 // MediaStream -> AddTrack in the same way it is called when we add track with
 // PeerConnection -> AddTrack.
@@ -3825,6 +3876,7 @@ TEST_F(PeerConnectionInterfaceTestPlanB,
               IsRtcOk());
   observer_.renegotiation_needed_ = false;
 }
+RTC_ALLOW_PLAN_B_DEPRECATION_END()
 
 // Tests that an error is returned if a description is applied that has fewer
 // media sections than the existing description.
diff --git a/pc/peer_connection_rtp_unittest.cc b/pc/peer_connection_rtp_unittest.cc
index d235e50418..ddb011b923 100644
--- a/pc/peer_connection_rtp_unittest.cc
+++ b/pc/peer_connection_rtp_unittest.cc
@@ -53,6 +53,7 @@
 #include "pc/test/integration_test_helpers.h"
 #include "pc/test/mock_peer_connection_observers.h"
 #include "rtc_base/checks.h"
+#include "rtc_base/system/plan_b_only.h"
 #include "rtc_base/thread.h"
 #include "system_wrappers/include/metrics.h"
 #include "test/gmock.h"
@@ -162,11 +163,13 @@ class PeerConnectionRtpTest
   PeerConnectionRtpTest() : PeerConnectionRtpBaseTest(GetParam()) {}
 };
 
+RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
 class PeerConnectionRtpTestPlanB : public PeerConnectionRtpBaseTest {
  protected:
   PeerConnectionRtpTestPlanB()
       : PeerConnectionRtpBaseTest(SdpSemantics::kPlanB_DEPRECATED) {}
 };
+RTC_ALLOW_PLAN_B_DEPRECATION_END()
 
 class PeerConnectionRtpTestUnifiedPlan : public PeerConnectionRtpBaseTest {
  protected:
@@ -269,7 +272,9 @@ TEST_P(PeerConnectionRtpTest, RemoveTrackWithStreamFiresOnRemoveTrack) {
   ASSERT_EQ(callee->observer()->add_track_events_.size(), 1u);
   EXPECT_EQ(callee->observer()->GetAddTrackReceivers(),
             callee->observer()->remove_track_events_);
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   EXPECT_EQ(0u, callee->observer()->remote_streams()->count());
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
 }
 
 TEST_P(PeerConnectionRtpTest, RemoveTrackWithSharedStreamFiresOnRemoveTrack) {
@@ -292,7 +297,9 @@ TEST_P(PeerConnectionRtpTest, RemoveTrackWithSharedStreamFiresOnRemoveTrack) {
       std::vector<scoped_refptr<RtpReceiverInterface>>{
           callee->observer()->add_track_events_[0].receiver},
       callee->observer()->remove_track_events_);
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   ASSERT_EQ(1u, callee->observer()->remote_streams()->count());
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   ASSERT_TRUE(
       caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
 
@@ -302,11 +309,14 @@ TEST_P(PeerConnectionRtpTest, RemoveTrackWithSharedStreamFiresOnRemoveTrack) {
   ASSERT_EQ(callee->observer()->add_track_events_.size(), 2u);
   EXPECT_EQ(callee->observer()->GetAddTrackReceivers(),
             callee->observer()->remove_track_events_);
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   EXPECT_EQ(0u, callee->observer()->remote_streams()->count());
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
 }
 
 // Tests the edge case that if a stream ID changes for a given track that both
 // OnRemoveTrack and OnAddTrack is fired.
+RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
 TEST_F(PeerConnectionRtpTestPlanB,
        RemoteStreamIdChangesFiresOnRemoveAndOnAddTrack) {
   auto caller = CreatePeerConnection();
@@ -333,6 +343,7 @@ TEST_F(PeerConnectionRtpTestPlanB,
   EXPECT_EQ(callee->observer()->remove_track_events_[0]->streams()[0]->id(),
             kStreamId1);
 }
+RTC_ALLOW_PLAN_B_DEPRECATION_END()
 
 // Tests that setting a remote description with sending transceivers will fire
 // the OnTrack callback for each transceiver and setting a remote description
@@ -705,6 +716,7 @@ TEST_P(PeerConnectionRtpTest, VideoGetParametersHasHeaderExtensions) {
 // calls and examine the state of the peer connection inside the callbacks to
 // ensure that the second call does not occur prematurely, contaminating the
 // state of the peer connection of the first callback.
+RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
 TEST_F(PeerConnectionRtpTestPlanB,
        StatesCorrelateWithSetRemoteDescriptionCall) {
   auto caller = CreatePeerConnection();
@@ -759,6 +771,7 @@ TEST_F(PeerConnectionRtpTestPlanB,
       WaitUntil([&] { return srd2_callback_called; }, ::testing::IsTrue()),
       IsRtcOk());
 }
+RTC_ALLOW_PLAN_B_DEPRECATION_END()
 
 // Tests that a remote track is created with the signaled MSIDs when they are
 // communicated with a=msid and no SSRCs are signaled at all (i.e., no a=ssrc
@@ -883,6 +896,7 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan,
 // remote audio senders, FindSenderInfo didn't find them as unique, because
 // it looked up by StreamParam.id, which none had. This meant only one
 // AudioRtpReceiver was created, as opposed to one for each remote sender.
+RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
 TEST_F(PeerConnectionRtpTestPlanB,
        MultipleRemoteSendersWithoutStreamParamIdAddsMultipleReceivers) {
   auto caller = CreatePeerConnection();
@@ -910,6 +924,7 @@ TEST_F(PeerConnectionRtpTestPlanB,
   ASSERT_EQ(receivers[1]->streams().size(), 1u);
   EXPECT_EQ(kStreamId2, receivers[1]->streams()[0]->id());
 }
+RTC_ALLOW_PLAN_B_DEPRECATION_END()
 
 // Tests for the legacy SetRemoteDescription() function signature.
 
@@ -1982,9 +1997,11 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan,
   EXPECT_EQ("stream5", callee_streams[2]->id());
 }
 
+RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
 INSTANTIATE_TEST_SUITE_P(PeerConnectionRtpTest,
                          PeerConnectionRtpTest,
                          Values(SdpSemantics::kPlanB_DEPRECATED,
                                 SdpSemantics::kUnifiedPlan));
+RTC_ALLOW_PLAN_B_DEPRECATION_END()
 
 }  // namespace webrtc
diff --git a/pc/rtc_stats_collector_unittest.cc b/pc/rtc_stats_collector_unittest.cc
index 480ab705b4..4c69afd636 100644
--- a/pc/rtc_stats_collector_unittest.cc
+++ b/pc/rtc_stats_collector_unittest.cc
@@ -90,6 +90,7 @@
 #include "rtc_base/ssl_identity.h"
 #include "rtc_base/ssl_stream_adapter.h"
 #include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/system/plan_b_only.h"
 #include "rtc_base/thread.h"
 #include "rtc_base/time_utils.h"
 #include "system_wrappers/include/clock.h"
@@ -468,7 +469,9 @@ class RTCStatsCollectorWrapper {
     });
     EXPECT_CALL(*sender, AttachmentId()).WillRepeatedly(Return(attachment_id));
     EXPECT_CALL(*sender, Stop()).WillRepeatedly(Return());
+    RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
     pc_->AddSender(sender);
+    RTC_ALLOW_PLAN_B_DEPRECATION_END();
     return sender;
   }
 
@@ -499,7 +502,9 @@ class RTCStatsCollectorWrapper {
         .WillRepeatedly(Return(
             std::vector<scoped_refptr<MediaStreamInterface>>({remote_stream})));
     EXPECT_CALL(*receiver, SetMediaChannel(_)).WillRepeatedly(Return());
+    RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
     pc_->AddReceiver(receiver);
+    RTC_ALLOW_PLAN_B_DEPRECATION_END();
     return receiver;
   }
 
@@ -545,7 +550,9 @@ class RTCStatsCollectorWrapper {
         return nullptr;
       });
       EXPECT_CALL(*rtp_sender, SetSendCodecs(_));
+      RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
       pc_->AddSender(rtp_sender);
+      RTC_ALLOW_PLAN_B_DEPRECATION_END();
     }
 
     // Remote audio tracks and voice receiver infos
@@ -563,7 +570,9 @@ class RTCStatsCollectorWrapper {
       EXPECT_CALL(*rtp_receiver, streams())
           .WillRepeatedly(Return(remote_streams));
       EXPECT_CALL(*rtp_receiver, SetMediaChannel(_)).WillRepeatedly(Return());
+      RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
       pc_->AddReceiver(rtp_receiver);
+      RTC_ALLOW_PLAN_B_DEPRECATION_END();
     }
 
     // Local video tracks and video sender infos
@@ -585,7 +594,9 @@ class RTCStatsCollectorWrapper {
         return nullptr;
       });
       EXPECT_CALL(*rtp_sender, SetSendCodecs(_));
+      RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
       pc_->AddSender(rtp_sender);
+      RTC_ALLOW_PLAN_B_DEPRECATION_END();
     }
 
     // Remote video tracks and video receiver infos
@@ -603,11 +614,15 @@ class RTCStatsCollectorWrapper {
       EXPECT_CALL(*rtp_receiver, streams())
           .WillRepeatedly(Return(remote_streams));
       EXPECT_CALL(*rtp_receiver, SetMediaChannel(_)).WillRepeatedly(Return());
+      RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
       pc_->AddReceiver(rtp_receiver);
+      RTC_ALLOW_PLAN_B_DEPRECATION_END();
     }
 
+    RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
     pc_->AddVoiceChannel("audio", "transport", voice_media_info);
     pc_->AddVideoChannel("video", "transport", video_media_info);
+    RTC_ALLOW_PLAN_B_DEPRECATION_END();
   }
 
  private:
@@ -722,7 +737,9 @@ class RTCStatsCollectorTestBase : public ::testing::Test {
     video_media_info.receivers[0].packets_received = 123;
     // transport
     graph.transport_id = "TTransportName1";
+    RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
     pc_->AddVideoChannel("VideoMid", "TransportName", video_media_info);
+    RTC_ALLOW_PLAN_B_DEPRECATION_END();
     // outbound-rtp's sender
     graph.sender = stats_->SetupLocalTrackAndSender(
         MediaType::VIDEO, "LocalVideoTrackID", 3, false, 50);
@@ -826,7 +843,9 @@ class RTCStatsCollectorTestBase : public ::testing::Test {
     }
     // transport
     graph.transport_id = "TTransportName1";
+    RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
     pc_->AddVoiceChannel("VoiceMid", "TransportName", media_info);
+    RTC_ALLOW_PLAN_B_DEPRECATION_END();
     // outbound-rtp's sender
     graph.sender = stats_->SetupLocalTrackAndSender(
         MediaType::AUDIO, "LocalAudioTrackID", kLocalSsrc, false, 50);
@@ -986,7 +1005,9 @@ TEST_P(RTCStatsCollectorTest, ToJsonProducesParseableJson) {
 TEST_P(RTCStatsCollectorTest, CollectRTCCertificateStatsSingle) {
   const char kTransportName[] = "transport";
 
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddVoiceChannel("audio", kTransportName);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
 
   std::unique_ptr<CertificateInfo> local_certinfo =
       CreateFakeCertificateAndInfoFromDers(
@@ -1016,14 +1037,18 @@ TEST_P(RTCStatsCollectorTest, ValidSsrcCollisionDoesNotCrash) {
   mid1_info.receivers[0].packets_received = 123;
   mid1_info.senders.emplace_back();
   mid1_info.senders[0].add_ssrc(2);
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddVoiceChannel("Mid1", "Transport1", mid1_info);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   VideoMediaInfo mid2_info;
   mid2_info.receivers.emplace_back();
   mid2_info.receivers[0].add_ssrc(3);
   mid2_info.receivers[0].packets_received = 123;
   mid2_info.senders.emplace_back();
   mid2_info.senders[0].add_ssrc(4);
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddVideoChannel("Mid2", "Transport1", mid2_info);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   // Now create a second BUNDLE group with SSRCs colliding with the first group
   // (but again no collisions within the group).
   VoiceMediaInfo mid3_info;
@@ -1032,14 +1057,18 @@ TEST_P(RTCStatsCollectorTest, ValidSsrcCollisionDoesNotCrash) {
   mid3_info.receivers[0].packets_received = 123;
   mid3_info.senders.emplace_back();
   mid3_info.senders[0].add_ssrc(2);
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddVoiceChannel("Mid3", "Transport2", mid3_info);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   VideoMediaInfo mid4_info;
   mid4_info.receivers.emplace_back();
   mid4_info.receivers[0].add_ssrc(3);
   mid4_info.receivers[0].packets_received = 123;
   mid4_info.senders.emplace_back();
   mid4_info.senders[0].add_ssrc(4);
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddVideoChannel("Mid4", "Transport2", mid4_info);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
 
   // This should not crash (https://crbug.com/1361612).
   scoped_refptr<const RTCStatsReport> report =
@@ -1060,25 +1089,33 @@ TEST_P(RTCStatsCollectorTest, InvalidSsrcCollisionDoesNotCrash) {
   mid1_info.receivers[0].add_ssrc(1);
   mid1_info.senders.emplace_back();
   mid1_info.senders[0].add_ssrc(1);
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddVoiceChannel("Mid1", "BundledTransport", mid1_info);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   VideoMediaInfo mid2_info;
   mid2_info.receivers.emplace_back();
   mid2_info.receivers[0].add_ssrc(1);
   mid2_info.senders.emplace_back();
   mid2_info.senders[0].add_ssrc(1);
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddVideoChannel("Mid2", "BundledTransport", mid2_info);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   VoiceMediaInfo mid3_info;
   mid3_info.receivers.emplace_back();
   mid3_info.receivers[0].add_ssrc(1);
   mid3_info.senders.emplace_back();
   mid3_info.senders[0].add_ssrc(1);
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddVoiceChannel("Mid3", "BundledTransport", mid3_info);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   VideoMediaInfo mid4_info;
   mid4_info.receivers.emplace_back();
   mid4_info.receivers[0].add_ssrc(1);
   mid4_info.senders.emplace_back();
   mid4_info.senders[0].add_ssrc(1);
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddVideoChannel("Mid4", "BundledTransport", mid4_info);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
 
   // This should not crash (https://crbug.com/1361612).
   stats_->GetStatsReport(main_thread_);
@@ -1154,10 +1191,12 @@ TEST_P(RTCStatsCollectorTest, CollectRTCCodecStatsOnlyIfReferenced) {
   outbound_video_info.codec_payload_type = 4;
   video_media_info.senders.push_back(outbound_video_info);
 
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   auto audio_channels =
       pc_->AddVoiceChannel("AudioMid", "TransportName", voice_media_info);
   auto video_channels =
       pc_->AddVideoChannel("VideoMid", "TransportName", video_media_info);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
 
   scoped_refptr<const RTCStatsReport> report =
       stats_->GetStatsReport(main_thread_);
@@ -1282,9 +1321,11 @@ TEST_P(RTCStatsCollectorTest, CodecStatsAreCollectedPerTransport) {
       outbound_codec_pt11.payload_type;
 
   // First two mids contain subsets, the third one contains all PTs.
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddVideoChannel("Mid1", "FirstTransport", info_pt10);
   pc_->AddVideoChannel("Mid2", "FirstTransport", info_pt11);
   pc_->AddVideoChannel("Mid3", "FirstTransport", info_pt10_pt11);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
 
   // There should be no duplicate codecs because all codec references are on the
   // same transport.
@@ -1295,7 +1336,9 @@ TEST_P(RTCStatsCollectorTest, CodecStatsAreCollectedPerTransport) {
 
   // If a second transport is added with the same PT information, this does
   // count as different codec objects.
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddVideoChannel("Mid4", "SecondTransport", info_pt10_pt11);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   stats_->stats_collector().ClearCachedStatsReport();
   report = stats_->GetStatsReport(main_thread_);
   codec_stats = report->GetStatsOfType<RTCCodecStats>();
@@ -1339,8 +1382,10 @@ TEST_P(RTCStatsCollectorTest, SamePayloadTypeButDifferentFmtpLines) {
       inbound_codec_pt111_fec.payload_type;
 
   // First two mids contain subsets, the third one contains all PTs.
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddVideoChannel("Mid1", "BundledTransport", info_nofec);
   pc_->AddVideoChannel("Mid2", "BundledTransport", info_fec);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
 
   // Despite having the same PT we should see two codec stats because their FMTP
   // lines are different.
@@ -1355,8 +1400,10 @@ TEST_P(RTCStatsCollectorTest, SamePayloadTypeButDifferentFmtpLines) {
   info_fec.receivers[0].local_stats[0].ssrc = 21;
   // Adding more m= sections that does have the same FMTP lines does not result
   // in duplicates.
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddVideoChannel("Mid3", "BundledTransport", info_nofec);
   pc_->AddVideoChannel("Mid4", "BundledTransport", info_fec);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   stats_->stats_collector().ClearCachedStatsReport();
   report = stats_->GetStatsReport(main_thread_);
   codec_stats = report->GetStatsOfType<RTCCodecStats>();
@@ -1379,7 +1426,9 @@ TEST_P(RTCStatsCollectorTest, SamePayloadTypeButDifferentFmtpLines) {
   info_fec_pt112.receivers[0].packets_received = 123;
   info_fec_pt112.receivers[0].codec_payload_type =
       inbound_codec_pt112_fec.payload_type;
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddVideoChannel("Mid5", "BundledTransport", info_fec_pt112);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   stats_->stats_collector().ClearCachedStatsReport();
   report = stats_->GetStatsReport(main_thread_);
   codec_stats = report->GetStatsOfType<RTCCodecStats>();
@@ -1390,7 +1439,9 @@ TEST_P(RTCStatsCollectorTest, CollectRTCCertificateStatsMultiple) {
   const char kAudioTransport[] = "audio";
   const char kVideoTransport[] = "video";
 
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddVoiceChannel("audio", kAudioTransport);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
 
   std::unique_ptr<CertificateInfo> audio_local_certinfo =
       CreateFakeCertificateAndInfoFromDers(
@@ -1403,7 +1454,9 @@ TEST_P(RTCStatsCollectorTest, CollectRTCCertificateStatsMultiple) {
       kAudioTransport,
       audio_remote_certinfo->certificate->GetSSLCertificateChain().Clone());
 
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddVideoChannel("video", kVideoTransport);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   std::unique_ptr<CertificateInfo> video_local_certinfo =
       CreateFakeCertificateAndInfoFromDers(
           std::vector<std::string>({"(local) video"}));
@@ -1426,7 +1479,9 @@ TEST_P(RTCStatsCollectorTest, CollectRTCCertificateStatsMultiple) {
 TEST_P(RTCStatsCollectorTest, CollectRTCCertificateStatsChain) {
   const char kTransportName[] = "transport";
 
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddVoiceChannel("audio", kTransportName);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
 
   std::unique_ptr<CertificateInfo> local_certinfo =
       CreateFakeCertificateAndInfoFromDers(
@@ -1450,7 +1505,9 @@ TEST_P(RTCStatsCollectorTest, CollectRTCCertificateStatsChain) {
 TEST_P(RTCStatsCollectorTest, CertificateStatsCache) {
   const char kTransportName[] = "transport";
 
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddVoiceChannel("audio", kTransportName);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
 
   // Set local and remote cerificates.
   std::unique_ptr<CertificateInfo> initial_local_certinfo =
@@ -1867,7 +1924,9 @@ TEST_P(RTCStatsCollectorTest, CollectRTCIceCandidateStats) {
   a_transport_channel_stats.ice_transport_stats.candidate_stats_list.push_back(
       CandidateStats(*a_local_host_not_paired));
 
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddVoiceChannel("audio", "a");
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   pc_->SetTransportStats("a", a_transport_channel_stats);
 
   TransportChannelStats b_transport_channel_stats;
@@ -1878,7 +1937,9 @@ TEST_P(RTCStatsCollectorTest, CollectRTCIceCandidateStats) {
   b_transport_channel_stats.ice_transport_stats.connection_infos[0]
       .remote_candidate = *b_remote;
 
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddVideoChannel("video", "b");
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   pc_->SetTransportStats("b", b_transport_channel_stats);
 
   scoped_refptr<const RTCStatsReport> report =
@@ -1965,7 +2026,9 @@ TEST_P(RTCStatsCollectorTest, CollectRTCIceCandidatePairStats) {
   transport_channel_stats.ice_transport_stats.connection_infos.push_back(
       connection_info);
 
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddVideoChannel("video", kTransportName);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   pc_->SetTransportStats(kTransportName, transport_channel_stats);
 
   scoped_refptr<const RTCStatsReport> report =
@@ -2235,8 +2298,10 @@ TEST_P(RTCStatsCollectorTest, CollectRTCInboundRtpStreamStats_Audio) {
       .num_packets_reported_lost = 222, .num_packets_reported_recovered = 200};
   pc_->SetCallStats(call_stats);
 
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   auto voice_media_channels =
       pc_->AddVoiceChannel("AudioMid", "TransportName", voice_media_info);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   stats_->SetupRemoteTrackAndReceiver(MediaType::AUDIO, "RemoteAudioTrackID",
                                       "RemoteStreamId", 1);
 
@@ -2322,7 +2387,9 @@ TEST_P(RTCStatsCollectorTest, CollectRTCInboundRtpStreamStats_Audio_PlayoutId) {
   voice_media_info.receivers[0].local_stats[0].ssrc = 1;
   voice_media_info.receivers[0].packets_received = 123;
 
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddVoiceChannel("AudioMid", "TransportName", voice_media_info);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   stats_->SetupRemoteTrackAndReceiver(MediaType::AUDIO, "RemoteAudioTrackID",
                                       "RemoteStreamId", 1);
   // Needed for playoutId to be populated.
@@ -2425,8 +2492,10 @@ TEST_P(RTCStatsCollectorTest, CollectRTCInboundRtpStreamStats_Video) {
       .num_packets_reported_lost = 222, .num_packets_reported_recovered = 200};
   pc_->SetCallStats(call_stats);
 
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   auto video_media_channels =
       pc_->AddVideoChannel("VideoMid", "TransportName", video_media_info);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   stats_->SetupRemoteTrackAndReceiver(MediaType::VIDEO, "RemoteVideoTrackID",
                                       "RemoteStreamId", 1);
 
@@ -2532,7 +2601,9 @@ TEST_P(RTCStatsCollectorTest, CollectRTCAudioPlayoutStats) {
   audio_device_stats.total_playout_delay_s = 5;
   pc_->SetAudioDeviceStats(audio_device_stats);
 
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddVoiceChannel("AudioMid", "TransportName", {});
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   stats_->SetupRemoteTrackAndReceiver(MediaType::AUDIO, "RemoteAudioTrackID",
                                       "RemoteStreamId", 1);
 
@@ -2578,7 +2649,9 @@ TEST_P(RTCStatsCollectorTest, CollectGoogTimingFrameInfo) {
   timing_frame_info.flags = 14;
   video_media_info.receivers[0].timing_frame_info = timing_frame_info;
 
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddVideoChannel("Mid0", "Transport0", video_media_info);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   stats_->SetupRemoteTrackAndReceiver(MediaType::VIDEO, "RemoteVideoTrackID",
                                       "RemoteStreamId", 1);
 
@@ -2617,7 +2690,9 @@ TEST_P(RTCStatsCollectorTest, CollectRTCOutboundRtpStreamStats_Audio) {
   voice_media_info.send_codecs.insert(
       std::make_pair(codec_parameters.payload_type, codec_parameters));
 
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddVoiceChannel("AudioMid", "TransportName", voice_media_info);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   stats_->SetupLocalTrackAndSender(MediaType::AUDIO, "LocalAudioTrackID", 1,
                                    true,
                                    /*attachment_id=*/50);
@@ -2716,8 +2791,10 @@ TEST_P(RTCStatsCollectorTest, CollectRTCOutboundRtpStreamStats_Video) {
   video_media_info.senders[1].rid = "h";
   video_media_info.senders[1].encoding_index = 1;
 
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   auto video_media_channels =
       pc_->AddVideoChannel("VideoMid", "TransportName", video_media_info);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   stats_->SetupLocalTrackAndSender(MediaType::VIDEO, "LocalVideoTrackID", 1,
                                    true,
                                    /*attachment_id=*/50);
@@ -2815,7 +2892,9 @@ TEST_P(RTCStatsCollectorTest, CollectRTCOutboundRtpStreamStats_Video) {
 TEST_P(RTCStatsCollectorTest, CollectRTCTransportStats) {
   const char kTransportName[] = "transport";
 
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddVoiceChannel("audio", kTransportName);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
 
   std::unique_ptr<Candidate> rtp_local_candidate =
       CreateFakeCandidate("42.42.42.42", 42, "protocol", ADAPTER_TYPE_WIFI,
@@ -2985,7 +3064,9 @@ TEST_P(RTCStatsCollectorTest, CollectRTCTransportStats) {
 TEST_P(RTCStatsCollectorTest, CollectRTCTransportStatsWithCrypto) {
   const char kTransportName[] = "transport";
 
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddVoiceChannel("audio", kTransportName);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
 
   std::unique_ptr<Candidate> rtp_local_candidate =
       CreateFakeCandidate("42.42.42.42", 42, "protocol", ADAPTER_TYPE_WIFI,
@@ -3086,7 +3167,9 @@ TEST_P(RTCStatsCollectorTest, CollectNoStreamRTCOutboundRtpStreamStats_Audio) {
       std::make_pair(codec_parameters.payload_type, codec_parameters));
 
   // Emulates the case where AddTrack is used without an associated MediaStream
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddVoiceChannel("AudioMid", "TransportName", voice_media_info);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   stats_->SetupLocalTrackAndSender(MediaType::AUDIO, "LocalAudioTrackID", 1,
                                    false,
                                    /*attachment_id=*/50);
@@ -3134,7 +3217,9 @@ TEST_P(RTCStatsCollectorTest, RTCAudioSourceStatsCollectedForSenderWithTrack) {
   voice_media_info.senders[0].apm_statistics.echo_return_loss = 42.0;
   voice_media_info.senders[0].apm_statistics.echo_return_loss_enhancement =
       52.0;
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddVoiceChannel("AudioMid", "TransportName", voice_media_info);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   stats_->SetupLocalTrackAndSender(MediaType::AUDIO, "LocalAudioTrackID", kSsrc,
                                    false, kAttachmentId);
 
@@ -3172,7 +3257,9 @@ TEST_P(RTCStatsCollectorTest, RTCVideoSourceStatsCollectedForSenderWithTrack) {
   video_media_info.aggregated_senders[0].local_stats[0].ssrc = kSsrc;
   video_media_info.aggregated_senders[0].framerate_input = 29.0;
   video_media_info.aggregated_senders[0].frames = 10001;
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddVideoChannel("VideoMid", "TransportName", video_media_info);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
 
   auto video_source = FakeVideoTrackSourceForStats::Create(kVideoSourceWidth,
                                                            kVideoSourceHeight);
@@ -3185,7 +3272,9 @@ TEST_P(RTCStatsCollectorTest, RTCVideoSourceStatsCollectedForSenderWithTrack) {
   });
   EXPECT_CALL(*sender, SetMediaChannel(_)).WillRepeatedly(Return());
   EXPECT_CALL(*sender, SetSendCodecs(_));
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddSender(sender);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
 
   scoped_refptr<const RTCStatsReport> report =
       stats_->GetStatsReport(main_thread_);
@@ -3220,7 +3309,9 @@ TEST_P(RTCStatsCollectorTest,
   video_media_info.senders.push_back(VideoSenderInfo());
   video_media_info.senders[0].local_stats.push_back(SsrcSenderInfo());
   video_media_info.senders[0].framerate_input = 29.0;
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddVideoChannel("VideoMid", "TransportName", video_media_info);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
 
   auto video_source = FakeVideoTrackSourceForStats::Create(kVideoSourceWidth,
                                                            kVideoSourceHeight);
@@ -3233,7 +3324,9 @@ TEST_P(RTCStatsCollectorTest,
   });
   EXPECT_CALL(*sender, SetMediaChannel(_)).WillRepeatedly(Return());
   EXPECT_CALL(*sender, SetSendCodecs(_));
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddSender(sender);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
 
   scoped_refptr<const RTCStatsReport> report =
       stats_->GetStatsReport(main_thread_);
@@ -3255,7 +3348,9 @@ TEST_P(RTCStatsCollectorTest,
   video_media_info.senders[0].local_stats.push_back(SsrcSenderInfo());
   video_media_info.senders[0].local_stats[0].ssrc = kSsrc;
   video_media_info.senders[0].framerate_input = 29.0;
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddVideoChannel("VideoMid", "TransportName", video_media_info);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
 
   auto video_track = FakeVideoTrackForStats::Create(
       "LocalVideoTrackID", MediaStreamTrackInterface::kLive,
@@ -3267,7 +3362,9 @@ TEST_P(RTCStatsCollectorTest,
   });
   EXPECT_CALL(*sender, SetMediaChannel(_)).WillRepeatedly(Return());
   EXPECT_CALL(*sender, SetSendCodecs(_));
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddSender(sender);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
 
   scoped_refptr<const RTCStatsReport> report =
       stats_->GetStatsReport(main_thread_);
@@ -3286,7 +3383,9 @@ TEST_P(RTCStatsCollectorTest,
   voice_media_info.senders.push_back(VoiceSenderInfo());
   voice_media_info.senders[0].local_stats.push_back(SsrcSenderInfo());
   voice_media_info.senders[0].local_stats[0].ssrc = kSsrc;
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddVoiceChannel("AudioMid", "TransportName", voice_media_info);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
   scoped_refptr<MockRtpSenderInternal> sender = CreateMockSender(
       MediaType::AUDIO, /*track=*/nullptr, kSsrc, kAttachmentId, {});
   EXPECT_CALL(*sender, DetachTrackAndGetStopTask()).WillRepeatedly([] {
@@ -3294,7 +3393,9 @@ TEST_P(RTCStatsCollectorTest,
   });
   EXPECT_CALL(*sender, SetMediaChannel(_)).WillRepeatedly(Return());
   EXPECT_CALL(*sender, SetSendCodecs(_));
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddSender(sender);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
 
   scoped_refptr<const RTCStatsReport> report =
       stats_->GetStatsReport(main_thread_);
@@ -3356,7 +3457,9 @@ class RTCStatsCollectorTestWithParamKind
           sender.report_block_datas.push_back(report_block_data);
           voice_media_info.senders.push_back(sender);
         }
+        RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
         pc_->AddVoiceChannel("mid", transport_name, voice_media_info);
+        RTC_ALLOW_PLAN_B_DEPRECATION_END();
         return;
       }
       case MediaType::VIDEO: {
@@ -3374,7 +3477,9 @@ class RTCStatsCollectorTestWithParamKind
           video_media_info.aggregated_senders.push_back(sender);
           video_media_info.senders.push_back(sender);
         }
+        RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
         pc_->AddVideoChannel("mid", transport_name, video_media_info);
+        RTC_ALLOW_PLAN_B_DEPRECATION_END();
         return;
       }
       case MediaType::DATA:
@@ -3686,7 +3791,9 @@ TEST_P(RTCStatsCollectorTest,
   video_media_info.senders[0].local_stats.push_back(SsrcSenderInfo());
   video_media_info.senders[0].local_stats[0].ssrc = kSsrc;
   video_media_info.senders[0].framerate_input = 29.0;
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddVideoChannel("VideoMid", "TransportName", video_media_info);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
 
   scoped_refptr<MockRtpSenderInternal> sender = CreateMockSender(
       MediaType::VIDEO, /*track=*/nullptr, kSsrc, kAttachmentId, {});
@@ -3695,7 +3802,9 @@ TEST_P(RTCStatsCollectorTest,
   });
   EXPECT_CALL(*sender, SetMediaChannel(_)).WillRepeatedly(Return());
   EXPECT_CALL(*sender, SetSendCodecs(_));
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddSender(sender);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
 
   scoped_refptr<const RTCStatsReport> report =
       stats_->GetStatsReport(main_thread_);
@@ -3831,7 +3940,9 @@ TEST_P(RTCStatsCollectorTest, RtpIsMissingWhileSsrcIsZero) {
     return nullptr;
   });
   EXPECT_CALL(*sender, SetSendCodecs(_));
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddSender(sender);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
 
   scoped_refptr<const RTCStatsReport> report =
       stats_->GetStatsReport(main_thread_);
@@ -3851,7 +3962,9 @@ TEST_P(RTCStatsCollectorTest, DoNotCrashIfSsrcIsKnownButInfosAreStillMissing) {
     return nullptr;
   });
   EXPECT_CALL(*sender, SetSendCodecs(_));
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc_->AddSender(sender);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
 
   // We do not generate any matching voice_sender_info stats.
   scoped_refptr<const RTCStatsReport> report =
diff --git a/test/testsupport/ivf_video_frame_generator_unittest.cc b/test/testsupport/ivf_video_frame_generator_unittest.cc
index 229d904daf..e79b725b95 100644
--- a/test/testsupport/ivf_video_frame_generator_unittest.cc
+++ b/test/testsupport/ivf_video_frame_generator_unittest.cc
@@ -37,7 +37,6 @@
 #include "modules/video_coding/utility/ivf_file_writer.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/event.h"
-#include "rtc_base/system/file_wrapper.h"
 #include "rtc_base/thread_annotations.h"
 #include "test/create_test_environment.h"
 #include "test/gtest.h"
@@ -72,8 +71,7 @@ class IvfFileWriterEncodedCallback : public EncodedImageCallback {
   IvfFileWriterEncodedCallback(const std::string& file_name,
                                VideoCodecType video_codec_type,
                                int expected_frames_count)
-      : file_writer_(
-            IvfFileWriter::Wrap(FileWrapper::OpenWriteOnly(file_name), 0)),
+      : file_writer_(IvfFileWriter::Wrap(file_name, 0)),
         video_codec_type_(video_codec_type),
         expected_frames_count_(expected_frames_count) {
     EXPECT_TRUE(file_writer_.get());
diff --git a/webrtc.gni b/webrtc.gni
index 64a739ca92..c78fc8a0b0 100644
--- a/webrtc.gni
+++ b/webrtc.gni
@@ -518,9 +518,6 @@ template("rtc_test") {
     if (defined(invoker.public_configs)) {
       public_configs += invoker.public_configs
     }
-    if (deprecate_plan_b) {
-      configs += [ webrtc_root + ":suppress_plan_b_deprecation" ]
-    }
     if (!build_with_chromium && is_android) {
       use_default_launcher = false
       android_manifest = webrtc_root + "test/android/AndroidManifest.xml"

From 1e24c9e308e4a6d27db872ac8b15c47af8111c41 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Thu, 26 Feb 2026 22:15:32 +0100
Subject: [PATCH 151/472] Replace AutoThread with test::RunLoop in
 PeerConnectionRtpBaseTest

Bug: webrtc:469327588
Change-Id: I27f54786fb7520e728fa70d83325f8a7822d2c5f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452502
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47004}
---
 pc/peer_connection_rtp_unittest.cc | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

diff --git a/pc/peer_connection_rtp_unittest.cc b/pc/peer_connection_rtp_unittest.cc
index ddb011b923..f82ff87ec6 100644
--- a/pc/peer_connection_rtp_unittest.cc
+++ b/pc/peer_connection_rtp_unittest.cc
@@ -58,6 +58,7 @@
 #include "system_wrappers/include/metrics.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 #include "test/wait_until.h"
 
 // This file contains tests for RTP Media API-related behavior of
@@ -137,6 +138,7 @@ class PeerConnectionRtpBaseTest : public ::testing::Test {
 
  protected:
   const SdpSemantics sdp_semantics_;
+  test::RunLoop run_loop_;
   scoped_refptr<PeerConnectionFactoryInterface> pc_factory_;
 
  private:
@@ -152,8 +154,6 @@ class PeerConnectionRtpBaseTest : public ::testing::Test {
     return std::make_unique<PeerConnectionWrapper>(
         pc_factory_, result.MoveValue(), std::move(observer));
   }
-
-  AutoThread main_thread_;
 };
 
 class PeerConnectionRtpTest
@@ -953,7 +953,7 @@ TEST_P(PeerConnectionRtpTest,
       caller->CreateOfferAndSetAsLocal();
   callee->pc()->SetRemoteDescription(observer.get(), offer.release());
   callee = nullptr;
-  Thread::Current()->ProcessMessages(0);
+  run_loop_.Flush();
   EXPECT_FALSE(observer->called());
 }
 

From c06fbd7aa8048df323ad8a90b3dacf52e35e142f Mon Sep 17 00:00:00 2001
From: Jeremy Leconte <jleconte@webrtc.org>
Date: Fri, 27 Feb 2026 11:52:23 +0100
Subject: [PATCH 152/472] Update the tree closing steps.

Change-Id: I7cdd861fbfeef87b72464bade63569fcdfbec78e
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452620
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Christoffer Dewerin <jansson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47005}
---
 infra/config/config.star                    | 24 ++----
 infra/config/generated/luci/luci-notify.cfg | 86 ++++++++++-----------
 2 files changed, 51 insertions(+), 59 deletions(-)

diff --git a/infra/config/config.star b/infra/config/config.star
index 592e2d4521..7204e1b5e7 100755
--- a/infra/config/config.star
+++ b/infra/config/config.star
@@ -18,6 +18,13 @@ load("@chromium-luci//recipe_experiments.star", "register_recipe_experiments")
 WEBRTC_GIT = "https://webrtc.googlesource.com/src"
 WEBRTC_GERRIT = "https://webrtc-review.googlesource.com/src"
 WEBRTC_TROOPER_EMAIL = "webrtc-troopers-robots@google.com"
+TREE_CLOSING_STEPS_REGEXP = [
+    "bot_update",
+    "compile",
+    "gclient runhooks",
+    "generate_build_files",
+    "update",
+]
 
 # Use LUCI Scheduler BBv2 names and add Scheduler realms configs.
 lucicfg.enable_experiment("crbug.com/1182002")
@@ -423,22 +430,7 @@ luci.buildbucket_notification_topic(
 luci.tree_closer(
     name = "webrtc_tree_closer",
     tree_status_host = "webrtc-status.appspot.com",
-    # TODO: These step filters are copied verbatim from Gatekeeper, for testing
-    # that LUCI-Notify would take the exact same actions. Once we've switched
-    # over, this should be updated - several of these steps don't exist in
-    # WebRTC recipes.
-    failed_step_regexp = [
-        "bot_update",
-        "compile",
-        "gclient runhooks",
-        "runhooks",
-        "update",
-        "extract build",
-        "cleanup_temp",
-        "taskkill",
-        "compile",
-        "gn",
-    ],
+    failed_step_regexp = TREE_CLOSING_STEPS_REGEXP,
     failed_step_regexp_exclude = ".*\\(experimental\\).*",
 )
 
diff --git a/infra/config/generated/luci/luci-notify.cfg b/infra/config/generated/luci/luci-notify.cfg
index 76a1902228..309b46196d 100644
--- a/infra/config/generated/luci/luci-notify.cfg
+++ b/infra/config/generated/luci/luci-notify.cfg
@@ -27,7 +27,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -54,7 +54,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -81,7 +81,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -108,7 +108,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -135,7 +135,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -162,7 +162,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -189,7 +189,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -216,7 +216,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -243,7 +243,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -270,7 +270,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -297,7 +297,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -324,7 +324,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -351,7 +351,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -378,7 +378,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -405,7 +405,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -432,7 +432,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -459,7 +459,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -486,7 +486,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -513,7 +513,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -540,7 +540,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -567,7 +567,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -594,7 +594,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -621,7 +621,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -648,7 +648,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -675,7 +675,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -702,7 +702,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -729,7 +729,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -756,7 +756,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -783,7 +783,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -810,7 +810,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -837,7 +837,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -864,7 +864,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -891,7 +891,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -918,7 +918,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -945,7 +945,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -972,7 +972,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -999,7 +999,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -1068,7 +1068,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -1095,7 +1095,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -1122,7 +1122,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -1149,7 +1149,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -1176,7 +1176,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }
@@ -1335,7 +1335,7 @@ notifiers {
   }
   tree_closers {
     tree_status_host: "webrtc-status.appspot.com"
-    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|generate_build_files|update"
     failed_step_regexp_exclude: ".*\\(experimental\\).*"
   }
 }

From bb1e977e069a37adc3f834de97039ad9081e76e9 Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Fri, 27 Feb 2026 11:06:27 +0100
Subject: [PATCH 153/472] Fix remaining differences between ArrayView and
 std::span usage

In tests explicitly ignore arguments std::span marks as no discard.
Add missing includes that currently rely on array_view.h transitive
includes.
Remove assumption 2nd template parameter of the ArrayView is signed
integer, instead migrate test helper ExpectEq to a standard matcher.
Remote assumption empty ArrayView::data() is nullptr.
Deprecate reinterpret_array_view as unused in WebRTC.

Bug: webrtc:439801349
Change-Id: I70c020d5ef6129396ed118ae7d32c5dc18101d9d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452500
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47006}
---
 api/array_view.h                              | 26 +++++----
 api/array_view_unittest.cc                    | 56 ++++---------------
 .../agc2/rnn_vad/ring_buffer_unittest.cc      | 20 +++----
 .../full_screen_mac_application_handler.cc    |  4 ++
 .../video_frame_buffer/RTCNativeI420Buffer.mm |  3 +
 5 files changed, 40 insertions(+), 69 deletions(-)

diff --git a/api/array_view.h b/api/array_view.h
index 67976fe2de..01593e0d02 100644
--- a/api/array_view.h
+++ b/api/array_view.h
@@ -100,11 +100,11 @@ class ArrayViewBase {
   static_assert(Size > 0, "ArrayView size must be variable or non-negative");
 
  public:
-  ArrayViewBase(T* data, size_t /* size */) : data_(data) {}
+  constexpr ArrayViewBase(T* data, size_t /* size */) : data_(data) {}
 
   static constexpr size_t size() { return Size; }
   static constexpr bool empty() { return false; }
-  T* data() const { return data_; }
+  constexpr T* data() const { return data_; }
 
  protected:
   static constexpr bool fixed_size() { return true; }
@@ -117,11 +117,11 @@ class ArrayViewBase {
 template <typename T>
 class ArrayViewBase<T, 0> {
  public:
-  explicit ArrayViewBase(T* /* data */, size_t /* size */) {}
+  explicit constexpr ArrayViewBase(T* /* data */, size_t /* size */) {}
 
   static constexpr size_t size() { return 0; }
   static constexpr bool empty() { return true; }
-  T* data() const { return nullptr; }
+  constexpr T* data() const { return nullptr; }
 
  protected:
   static constexpr bool fixed_size() { return true; }
@@ -131,12 +131,12 @@ class ArrayViewBase<T, 0> {
 template <typename T>
 class ArrayViewBase<T, array_view_internal::kArrayViewVarSize> {
  public:
-  ArrayViewBase(T* data, size_t size)
+  constexpr ArrayViewBase(T* data, size_t size)
       : data_(size == 0 ? nullptr : data), size_(size) {}
 
   constexpr size_t size() const { return size_; }
   constexpr bool empty() const { return size_ == 0; }
-  T* data() const { return data_; }
+  constexpr T* data() const { return data_; }
 
  protected:
   static constexpr bool fixed_size() { return false; }
@@ -168,7 +168,7 @@ class ArrayView final : public array_view_internal::ArrayViewBase<T, Size> {
   // Construct an ArrayView from a pointer and a length.
   template <typename U>
   explicit(Size != array_view_internal::kArrayViewVarSize)  //
-      ArrayView(U* data, size_t size)
+      constexpr ArrayView(U* data, size_t size)
       : array_view_internal::ArrayViewBase<T, Size>::ArrayViewBase(data, size) {
     RTC_DCHECK_EQ(size == 0 ? nullptr : data, this->data());
     RTC_DCHECK_EQ(size, this->size());
@@ -178,7 +178,7 @@ class ArrayView final : public array_view_internal::ArrayViewBase<T, Size> {
 
   // Construct an empty ArrayView. Note that fixed-size ArrayViews of size > 0
   // cannot be empty.
-  ArrayView() : ArrayView(static_cast<T*>(nullptr), 0) {}
+  constexpr ArrayView() : ArrayView(static_cast<T*>(nullptr), 0) {}
   ABSL_DEPRECATE_AND_INLINE()
   ArrayView(std::nullptr_t)  // NOLINT
       : ArrayView() {}
@@ -276,7 +276,7 @@ class ArrayView final : public array_view_internal::ArrayViewBase<T, Size> {
   // Indexing and iteration. These allow mutation even if the ArrayView is
   // const, because the ArrayView doesn't own the array. (To prevent mutation,
   // use a const element type.)
-  T& operator[](size_t idx) const {
+  [[nodiscard]] T& operator[](size_t idx) const {
     // RTC_HARDENING_ASSERT is used because this is tunable at compile time
     // to be on or off in production, depending on the choices of the
     // embedder.
@@ -331,8 +331,9 @@ class ArrayView final : public array_view_internal::ArrayViewBase<T, Size> {
     }
   }
 
-  constexpr ArrayView<T> subspan(size_t offset,
-                                 size_t count = std::dynamic_extent) const {
+  [[nodiscard]] constexpr ArrayView<T> subspan(
+      size_t offset,
+      size_t count = std::dynamic_extent) const {
     ABSL_HARDENING_ASSERT(offset <= this->size());
     if (count == std::dynamic_extent) {
       count = this->size() - offset;
@@ -380,7 +381,8 @@ inline ArrayView<T> MakeArrayView(T* data, size_t size) {
 // Template arguments order is (U, T, Size) to allow deduction of the template
 // arguments in client calls: reinterpret_array_view<target_type>(array_view).
 template <typename U, typename T, std::ptrdiff_t Size>
-inline ArrayView<U, Size> reinterpret_array_view(ArrayView<T, Size> view) {
+[[deprecated]] ArrayView<U, Size> reinterpret_array_view(
+    ArrayView<T, Size> view) {
   static_assert(sizeof(U) == sizeof(T) && alignof(U) == alignof(T),
                 "ArrayView reinterpret_cast is only supported for casting "
                 "between views that represent the same chunk of memory.");
diff --git a/api/array_view_unittest.cc b/api/array_view_unittest.cc
index 4efc19f0a5..bd7e491984 100644
--- a/api/array_view_unittest.cc
+++ b/api/array_view_unittest.cc
@@ -71,10 +71,10 @@ TEST(ArrayViewDeathTest, TestConstructFromPtrAndArray) {
   EXPECT_EQ(arr, wf.data());
   ArrayView<char> q(arr, 0);
   EXPECT_EQ(0u, q.size());
-  EXPECT_EQ(nullptr, q.data());
+  EXPECT_TRUE(q.empty());
   ArrayView<char, 0> qf(arr, 0);
   static_assert(qf.size() == 0, "");
-  EXPECT_EQ(nullptr, qf.data());
+  EXPECT_TRUE(qf.empty());
 #if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
   // DCHECK error (nullptr with nonzero size).
   EXPECT_DEATH(ArrayView<int>(static_cast<int*>(nullptr), 5), "");
@@ -429,7 +429,8 @@ TEST(ArrayViewDeathTest, TestIndexing) {
   EXPECT_EQ('Y', y[2]);
   EXPECT_EQ('X', z[3]);
 #if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
-  EXPECT_DEATH(z[8], "");  // DCHECK error (index out of bounds).
+  // DCHECK error (index out of bounds).
+  EXPECT_DEATH(std::ignore = z[8], "");
 #endif
 }
 
@@ -530,29 +531,9 @@ TEST(ArrayViewTest, TestEmpty) {
   const int a[] = {1, 2, 3};
   EXPECT_FALSE(ArrayView<const int>(a).empty());
 
-  static_assert(ArrayView<int, 0>::empty(), "");
-  static_assert(!ArrayView<int, 3>::empty(), "");
+  static_assert(ArrayView<int, 0>().empty());
 }
 
-#pragma clang diagnostic push
-#pragma clang diagnostic ignored "-Wdeprecated-declarations"
-TEST(ArrayViewTest, TestSubViewVariable) {
-  int a[] = {1, 2, 3};
-  ArrayView<int> av(a);
-
-  EXPECT_THAT(av.subview(0), ElementsAre(1, 2, 3));
-  EXPECT_THAT(av.subview(1), ElementsAre(2, 3));
-  EXPECT_THAT(av.subview(2), ElementsAre(3));
-  EXPECT_THAT(av.subview(3), IsEmpty());
-  EXPECT_THAT(av.subview(4), IsEmpty());
-
-  EXPECT_THAT(av.subview(1, 0), IsEmpty());
-  EXPECT_THAT(av.subview(1, 1), ElementsAre(2));
-  EXPECT_THAT(av.subview(1, 2), ElementsAre(2, 3));
-  EXPECT_THAT(av.subview(1, 3), ElementsAre(2, 3));
-}
-#pragma clang diagnostic pop
-
 TEST(ArrayViewTest, TestSubSpanVariable) {
   int a[] = {1, 2, 3};
   ArrayView<int> av(a);
@@ -570,29 +551,10 @@ TEST(ArrayViewTest, TestSubSpanVariable) {
 TEST(ArrayViewTest, TestSubSpanWithInvalidInput) {
   int a[] = {1, 2, 3};
   ArrayView<int> av(a);
-  EXPECT_DEATH_IF_SUPPORTED(av.subspan(4), "");
-  EXPECT_DEATH_IF_SUPPORTED(av.subspan(1, 3), "");
+  EXPECT_DEATH_IF_SUPPORTED(std::ignore = av.subspan(4), "");
+  EXPECT_DEATH_IF_SUPPORTED(std::ignore = av.subspan(1, 3), "");
 }
 
-#pragma clang diagnostic push
-#pragma clang diagnostic ignored "-Wdeprecated-declarations"
-TEST(ArrayViewTest, TestSubViewFixed) {
-  int a[] = {1, 2, 3};
-  ArrayView<int, 3> av(a);
-
-  EXPECT_THAT(av.subview(0), ElementsAre(1, 2, 3));
-  EXPECT_THAT(av.subview(1), ElementsAre(2, 3));
-  EXPECT_THAT(av.subview(2), ElementsAre(3));
-  EXPECT_THAT(av.subview(3), IsEmpty());
-  EXPECT_THAT(av.subview(4), IsEmpty());
-
-  EXPECT_THAT(av.subview(1, 0), IsEmpty());
-  EXPECT_THAT(av.subview(1, 1), ElementsAre(2));
-  EXPECT_THAT(av.subview(1, 2), ElementsAre(2, 3));
-  EXPECT_THAT(av.subview(1, 3), ElementsAre(2, 3));
-}
-#pragma clang diagnostic pop
-
 TEST(ArrayViewTest, SubspanFixed) {
   std::array<int, 3> a = {1, 2, 3};
   ArrayView<int, 3> av(a);
@@ -649,6 +611,8 @@ TEST(ArrayViewTest, LastFixed) {
   EXPECT_EQ((av.last<2>()).extent, 2u);
 }
 
+#pragma clang diagnostic push
+#pragma clang diagnostic ignored "-Wdeprecated-declarations"
 TEST(ArrayViewTest, TestReinterpretCastFixedSize) {
   uint8_t bytes[] = {1, 2, 3};
   ArrayView<uint8_t, 3> uint8_av(bytes);
@@ -668,4 +632,6 @@ TEST(ArrayViewTest, TestReinterpretCastVariableSize) {
   EXPECT_EQ(uint8_av[1], 2);
   EXPECT_EQ(uint8_av[2], 3);
 }
+#pragma clang diagnostic pop
+
 }  // namespace webrtc
diff --git a/modules/audio_processing/agc2/rnn_vad/ring_buffer_unittest.cc b/modules/audio_processing/agc2/rnn_vad/ring_buffer_unittest.cc
index e223c94479..82bd01f591 100644
--- a/modules/audio_processing/agc2/rnn_vad/ring_buffer_unittest.cc
+++ b/modules/audio_processing/agc2/rnn_vad/ring_buffer_unittest.cc
@@ -15,20 +15,14 @@
 #include <cstdint>
 
 #include "api/array_view.h"
+#include "test/gmock.h"
 #include "test/gtest.h"
 
 namespace webrtc {
 namespace rnn_vad {
 namespace {
 
-// Compare the elements of two given array views.
-template <typename T, std::ptrdiff_t S>
-void ExpectEq(ArrayView<const T, S> a, ArrayView<const T, S> b) {
-  for (int i = 0; i < S; ++i) {
-    SCOPED_TRACE(i);
-    EXPECT_EQ(a[i], b[i]);
-  }
-}
+using ::testing::ElementsAreArray;
 
 // Test push/read sequences.
 template <typename T, int S, int N>
@@ -44,7 +38,7 @@ void TestRingBuffer() {
   ring_buf.GetArrayView(0);
   pushed_array.fill(0);
   ring_buf.Push(pushed_array_view);
-  ExpectEq(pushed_array_view, ring_buf.GetArrayView(0));
+  EXPECT_THAT(ring_buf.GetArrayView(0), ElementsAreArray(pushed_array_view));
 
   // Push N times and check most recent and second most recent.
   for (T v = 1; v <= static_cast<T>(N); ++v) {
@@ -52,10 +46,11 @@ void TestRingBuffer() {
     prev_pushed_array = pushed_array;
     pushed_array.fill(v);
     ring_buf.Push(pushed_array_view);
-    ExpectEq(pushed_array_view, ring_buf.GetArrayView(0));
+    EXPECT_THAT(ring_buf.GetArrayView(0), ElementsAreArray(pushed_array_view));
     if (N > 1) {
       pushed_array.fill(v - 1);
-      ExpectEq(pushed_array_view, ring_buf.GetArrayView(1));
+      EXPECT_THAT(ring_buf.GetArrayView(1),
+                  ElementsAreArray(pushed_array_view));
     }
   }
 
@@ -64,7 +59,8 @@ void TestRingBuffer() {
     SCOPED_TRACE(delay);
     T expected_value = N - static_cast<T>(delay);
     pushed_array.fill(expected_value);
-    ExpectEq(pushed_array_view, ring_buf.GetArrayView(delay));
+    EXPECT_THAT(ring_buf.GetArrayView(delay),
+                ElementsAreArray(pushed_array_view));
   }
 }
 
diff --git a/modules/desktop_capture/mac/full_screen_mac_application_handler.cc b/modules/desktop_capture/mac/full_screen_mac_application_handler.cc
index a43797f8a6..ce7e4d4db8 100644
--- a/modules/desktop_capture/mac/full_screen_mac_application_handler.cc
+++ b/modules/desktop_capture/mac/full_screen_mac_application_handler.cc
@@ -13,7 +13,11 @@
 #include <libproc.h>
 
 #include <algorithm>
+#include <cstdint>
+#include <cstring>
 #include <functional>
+#include <iterator>
+#include <memory>
 #include <string>
 
 #include "absl/strings/match.h"
diff --git a/sdk/objc/api/video_frame_buffer/RTCNativeI420Buffer.mm b/sdk/objc/api/video_frame_buffer/RTCNativeI420Buffer.mm
index 5eb5862ca5..4682bf810f 100644
--- a/sdk/objc/api/video_frame_buffer/RTCNativeI420Buffer.mm
+++ b/sdk/objc/api/video_frame_buffer/RTCNativeI420Buffer.mm
@@ -10,7 +10,10 @@
 
 #import "RTCNativeI420Buffer+Private.h"
 
+#include <cstdint>
+
 #include "api/video/i420_buffer.h"
+#include "rtc_base/checks.h"
 
 #if !defined(NDEBUG) && defined(WEBRTC_IOS)
 #import <UIKit/UIKit.h>

From 051601162f002d7f911189a2e2ef3b6c147d44de Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Fri, 27 Feb 2026 10:24:18 +0000
Subject: [PATCH 154/472] Restrict RtpTransceiver senders/receivers access to
 PLAN_B_ONLY.

This change restricts access to the full lists of senders and receivers
in RtpTransceiver to code paths that explicitly support Plan B.
Unified Plan code paths are migrated to use new helpers that assume
at most one sender/receiver.

- Mark RtpTransceiver::senders() and receivers() as PLAN_B_ONLY.
- Mark RtpTransmissionManager::FindReceiverById() as PLAN_B_ONLY.
- Update callers to use new helpers or wrap legacy access in deprecation allowance blocks.
- Add missing plan_b_only dependencies in pc/BUILD.gn.

Bug: webrtc:467984837
Change-Id: I5a99da1d25d8c96c066542bac7160fe39249c2b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/451400
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47007}
---
 g3doc/todo/plan_b_senders_receivers.md   | 31 ++++++++++++++++++++++++
 pc/BUILD.gn                              |  3 +++
 pc/legacy_stats_collector.cc             |  5 ++++
 pc/peer_connection.cc                    |  4 +++
 pc/rtc_stats_collector.cc                |  7 +++++-
 pc/rtp_transceiver.h                     | 13 ++++++----
 pc/rtp_transmission_manager.cc           | 14 ++++++++++-
 pc/rtp_transmission_manager.h            |  2 +-
 pc/sdp_offer_answer.cc                   |  2 +-
 pc/test/fake_peer_connection_for_stats.h |  4 +++
 10 files changed, 76 insertions(+), 9 deletions(-)
 create mode 100644 g3doc/todo/plan_b_senders_receivers.md

diff --git a/g3doc/todo/plan_b_senders_receivers.md b/g3doc/todo/plan_b_senders_receivers.md
new file mode 100644
index 0000000000..dcae1de7ea
--- /dev/null
+++ b/g3doc/todo/plan_b_senders_receivers.md
@@ -0,0 +1,31 @@
+<!-- go/cmark -->
+# Plan: Restrict RtpTransceiver sender/receiver access to PLAN_B_ONLY
+
+## Objective
+Restrict access to `RtpTransceiver::senders_` and `RtpTransceiver::receivers_` such that access to all elements except the first one is only possible via functions marked `PLAN_B_ONLY`. This aligns the implementation with Unified Plan requirements where only one sender and one receiver are used, while preserving legacy Plan B functionality.
+
+## Proposed Changes
+
+### 1. Header Refactoring (`pc/rtp_transceiver.h`)
+*   **Deprecate Vector Accessors**: Mark `senders()` and `receivers()` with the `PLAN_B_ONLY` macro. This triggers deprecation warnings for any code accessing the full lists, making Plan B dependencies explicit.
+*   **Add Unified Plan Accessors**:
+    *   Add `internal_first_sender()` and `internal_first_receiver()` private helpers.
+    *   These helpers will return the first element (or `nullptr`) and will be used by Unified Plan paths.
+
+### 2. Implementation Update (`pc/rtp_transceiver.cc`)
+*   **Shared Logic Branching**: Methods that currently iterate over `senders_` or `receivers_` (e.g., `SetMediaChannels`, `GetStopSendingAndReceiving`, `GetDeleteChannelWorkerTask`, `OnNegotiationUpdate`) must be updated:
+    *   **Unified Plan**: Use `internal_first_...` helpers to process only the primary element.
+    *   **Plan B**: Wrap iterations in `RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN` and `RTC_ALLOW_PLAN_B_DEPRECATION_END`.
+*   **Protect Index Access**: Ensure `senders_[0]` or `receivers_[0]` are only used in Unified Plan paths (guarded by `RTC_DCHECK(unified_plan_)`).
+
+### 3. External Caller Migration
+*   Update components that iterate over senders/receivers:
+    *   `pc/legacy_stats_collector.cc`
+    *   `pc/peer_connection.cc`
+    *   `pc/rtc_stats_collector.cc`
+    *   `pc/rtp_transmission_manager.cc`
+*   Wrap these iterations in `RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN/END` to acknowledge the legacy dependency.
+
+## Verification Plan
+*   **Compilation**: Ensure all targets compile with and without `WEBRTC_DEPRECATE_PLAN_B` defined.
+*   **Unit Tests**: Run `pc_unittests` and `peerconnection_unittests` to ensure no regression in both Unified Plan and Plan B modes.
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 0ac4ec4cf0..d1d436c1d7 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -1065,6 +1065,7 @@ rtc_library("rtc_stats_collector") {
     "../rtc_base:threading",
     "../rtc_base:timeutils",
     "../rtc_base/containers:flat_set",
+    "../rtc_base/system:plan_b_only",
     "//third_party/abseil-cpp/absl/functional:any_invocable",
     "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/strings:string_view",
@@ -1151,6 +1152,7 @@ rtc_library("sdp_offer_answer") {
     "../api:jsep",
     "../api:make_ref_counted",
     "../api:media_stream_interface",
+    "../api:payload_type",
     "../api:peer_connection_interface",
     "../api:rtc_error",
     "../api:rtp_parameters",
@@ -1452,6 +1454,7 @@ rtc_library("legacy_stats_collector") {
     "../rtc_base:ssl_adapter",
     "../rtc_base:threading",
     "../rtc_base:timeutils",
+    "../rtc_base/system:plan_b_only",
     "../system_wrappers",
     "//third_party/abseil-cpp/absl/base:nullability",
     "//third_party/abseil-cpp/absl/functional:any_invocable",
diff --git a/pc/legacy_stats_collector.cc b/pc/legacy_stats_collector.cc
index fdb1f99412..f41de1cded 100644
--- a/pc/legacy_stats_collector.cc
+++ b/pc/legacy_stats_collector.cc
@@ -61,6 +61,7 @@
 #include "rtc_base/socket_address.h"
 #include "rtc_base/ssl_certificate.h"
 #include "rtc_base/ssl_stream_adapter.h"
+#include "rtc_base/system/plan_b_only.h"
 #include "rtc_base/thread.h"
 #include "rtc_base/trace_event.h"
 #include "system_wrappers/include/clock.h"
@@ -1302,12 +1303,14 @@ void LegacyStatsCollector::ExtractMediaInfo(
       if (it != transport_names_by_mid.end()) {
         gatherer->transport_name = it->second;
       }
+      RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
       for (const auto& sender : transceiver->internal()->senders()) {
         auto track = sender->track();
         std::string track_id = (track ? track->id() : "");
         gatherer->sender_track_id_by_ssrc.insert(
             std::make_pair(sender->ssrc(), track_id));
       }
+      RTC_ALLOW_PLAN_B_DEPRECATION_END()
 
       // Populating `receiver_track_id_by_ssrc` will be done on the worker
       // thread as the `ssrc` property of the receiver needs to be accessed
@@ -1325,10 +1328,12 @@ void LegacyStatsCollector::ExtractMediaInfo(
       if (!transceiver->internal()->HasChannel())
         continue;
       ChannelStatsGatherer* gatherer = gatherers[i++].get();
+      RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
       for (const auto& receiver : transceiver->internal()->receivers()) {
         gatherer->receiver_track_id_by_ssrc.insert(std::make_pair(
             receiver->internal()->ssrc().value_or(0), receiver->track()->id()));
       }
+      RTC_ALLOW_PLAN_B_DEPRECATION_END()
     }
 
     for (auto it = gatherers.begin(); it != gatherers.end();
diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc
index 9c34f6d07e..760bc8175c 100644
--- a/pc/peer_connection.cc
+++ b/pc/peer_connection.cc
@@ -1383,6 +1383,7 @@ void PeerConnection::GetStats(
   if (selector) {
     for (const auto& proxy_transceiver :
          rtp_manager()->transceivers()->List()) {
+      RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
       for (const auto& proxy_sender :
            proxy_transceiver->internal()->senders()) {
         if (proxy_sender == selector) {
@@ -1390,6 +1391,7 @@ void PeerConnection::GetStats(
           break;
         }
       }
+      RTC_ALLOW_PLAN_B_DEPRECATION_END()
       if (internal_sender)
         break;
     }
@@ -1413,6 +1415,7 @@ void PeerConnection::GetStats(
   if (selector) {
     for (const auto& proxy_transceiver :
          rtp_manager()->transceivers()->List()) {
+      RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
       for (const auto& proxy_receiver :
            proxy_transceiver->internal()->receivers()) {
         if (proxy_receiver == selector) {
@@ -1420,6 +1423,7 @@ void PeerConnection::GetStats(
           break;
         }
       }
+      RTC_ALLOW_PLAN_B_DEPRECATION_END()
       if (internal_receiver)
         break;
     }
diff --git a/pc/rtc_stats_collector.cc b/pc/rtc_stats_collector.cc
index 15cb864621..19582ad274 100644
--- a/pc/rtc_stats_collector.cc
+++ b/pc/rtc_stats_collector.cc
@@ -75,6 +75,7 @@
 #include "rtc_base/ssl_certificate.h"
 #include "rtc_base/ssl_stream_adapter.h"
 #include "rtc_base/strings/string_builder.h"
+#include "rtc_base/system/plan_b_only.h"
 #include "rtc_base/thread.h"
 #include "rtc_base/time_utils.h"
 #include "rtc_base/trace_event.h"
@@ -1733,7 +1734,9 @@ void RTCStatsCollector::ProduceMediaSourceStats_s(
     const TrackMediaInfoMap& track_media_info_map =
         *transceiver_stats_info.track_media_info_map;
 
+    RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
     for (const auto& sender : refs.transceiver->senders()) {
+      RTC_ALLOW_PLAN_B_DEPRECATION_END()
       const auto& sender_internal = sender->internal();
       const auto& track = sender_internal->track();
       if (!track)
@@ -1746,7 +1749,7 @@ void RTCStatsCollector::ProduceMediaSourceStats_s(
       // levels are moved to the corresponding audio track/source object), don't
       // create separate media source stats objects on a per-attachment basis.
       std::unique_ptr<RTCMediaSourceStats> media_source_stats;
-      if (track->kind() == MediaStreamTrackInterface::kAudioKind) {
+      if (sender_internal->media_type() == MediaType::AUDIO) {
         AudioTrackInterface* audio_track =
             static_cast<AudioTrackInterface*>(track.get());
         auto audio_source_stats = std::make_unique<RTCAudioSourceStats>(
@@ -2293,6 +2296,7 @@ RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w() {
     TransceiverReferences refs{.transceiver =
                                    scoped_refptr<RtpTransceiver>(transceiver)};
 
+    RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
     for (const auto& sender : transceiver->senders()) {
       stats.sender_infos.push_back(
           {.ssrc = sender->ssrc(),
@@ -2307,6 +2311,7 @@ RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w() {
       refs.receivers.push_back(
           scoped_refptr<RtpReceiverInternal>(receiver->internal()));
     }
+    RTC_ALLOW_PLAN_B_DEPRECATION_END()
     stats.has_receivers = !refs.receivers.empty();
 
     if (stats.has_channel) {
diff --git a/pc/rtp_transceiver.h b/pc/rtp_transceiver.h
index 107e66e2f4..431e4cffde 100644
--- a/pc/rtp_transceiver.h
+++ b/pc/rtp_transceiver.h
@@ -223,8 +223,9 @@ class RtpTransceiver : public RtpTransceiverInterface {
   PLAN_B_ONLY bool RemoveSenderPlanB(RtpSenderInterface* sender);
 
   // Returns a vector of the senders owned by this transceiver.
-  std::vector<scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>
-  senders() const {
+  PLAN_B_ONLY const
+      std::vector<scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>&
+      senders() const {
     return senders_;
   }
 
@@ -239,7 +240,8 @@ class RtpTransceiver : public RtpTransceiverInterface {
   PLAN_B_ONLY bool RemoveReceiverPlanB(RtpReceiverInterface* receiver);
 
   // Returns a vector of the receivers owned by this transceiver.
-  std::vector<scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>>
+  PLAN_B_ONLY const std::vector<
+      scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>>&
   receivers() const {
     return receivers_;
   }
@@ -327,8 +329,9 @@ class RtpTransceiver : public RtpTransceiverInterface {
   // RtpTransceiverInterface implementation.
   MediaType media_type() const override;
   std::optional<std::string> mid() const override;
-  scoped_refptr<RtpSenderInterface> sender() const override;
-  scoped_refptr<RtpReceiverInterface> receiver() const override;
+  absl_nonnull scoped_refptr<RtpSenderInterface> sender() const override;
+  absl_nonnull scoped_refptr<RtpReceiverInterface> receiver() const override;
+
   bool stopped() const override;
   bool stopping() const override;
   RtpTransceiverDirection direction() const override;
diff --git a/pc/rtp_transmission_manager.cc b/pc/rtp_transmission_manager.cc
index 25f1360c34..dc6e4bc6f4 100644
--- a/pc/rtp_transmission_manager.cc
+++ b/pc/rtp_transmission_manager.cc
@@ -349,7 +349,9 @@ RtpTransmissionManager::GetSendersInternal() const {
     if (IsUnifiedPlan() && transceiver->internal()->stopped())
       continue;
 
+    RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
     auto senders = transceiver->internal()->senders();
+    RTC_ALLOW_PLAN_B_DEPRECATION_END()
     all_senders.insert(all_senders.end(), senders.begin(), senders.end());
   }
   return all_senders;
@@ -364,7 +366,9 @@ RtpTransmissionManager::GetReceiversInternal() const {
     if (IsUnifiedPlan() && transceiver->internal()->stopped())
       continue;
 
+    RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
     auto receivers = transceiver->internal()->receivers();
+    RTC_ALLOW_PLAN_B_DEPRECATION_END()
     all_receivers.insert(all_receivers.end(), receivers.begin(),
                          receivers.end());
   }
@@ -656,11 +660,13 @@ RtpTransmissionManager::FindSenderForTrack(
     MediaStreamTrackInterface* track) const {
   RTC_DCHECK_RUN_ON(signaling_thread());
   for (const auto& transceiver : transceivers_.List()) {
+    RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
     for (auto sender : transceiver->internal()->senders()) {
       if (sender->track() == track) {
         return sender;
       }
     }
+    RTC_ALLOW_PLAN_B_DEPRECATION_END()
   }
   return nullptr;
 }
@@ -669,16 +675,22 @@ scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>
 RtpTransmissionManager::FindSenderById(absl::string_view sender_id) const {
   RTC_DCHECK_RUN_ON(signaling_thread());
   for (const auto& transceiver : transceivers_.List()) {
+    RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
+    // Under Unified Plan, senders() always has exactly one entry,
+    // and one can use sender() not senders().
+    // Since this function is used both in Plan B and Unified, this is
+    // left as-is for now.
     for (auto sender : transceiver->internal()->senders()) {
       if (sender->id() == sender_id) {
         return sender;
       }
     }
+    RTC_ALLOW_PLAN_B_DEPRECATION_END();
   }
   return nullptr;
 }
 
-scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
+PLAN_B_ONLY scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
 RtpTransmissionManager::FindReceiverById(absl::string_view receiver_id) const {
   RTC_DCHECK_RUN_ON(signaling_thread());
   for (const auto& transceiver : transceivers_.List()) {
diff --git a/pc/rtp_transmission_manager.h b/pc/rtp_transmission_manager.h
index 1bb5795b58..cd2d914ce5 100644
--- a/pc/rtp_transmission_manager.h
+++ b/pc/rtp_transmission_manager.h
@@ -180,7 +180,7 @@ class RtpTransmissionManager : public RtpSenderBase::SetStreamsObserver {
       absl::string_view sender_id) const;
 
   // Return the RtpReceiver with the given id, or null if none exists.
-  scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
+  PLAN_B_ONLY scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
   FindReceiverById(absl::string_view receiver_id) const;
 
   TransceiverList* transceivers() { return &transceivers_; }
diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc
index af366df54d..aa6fe33e34 100644
--- a/pc/sdp_offer_answer.cc
+++ b/pc/sdp_offer_answer.cc
@@ -39,6 +39,7 @@
 #include "api/make_ref_counted.h"
 #include "api/media_stream_interface.h"
 #include "api/media_types.h"
+#include "api/payload_type.h"
 #include "api/peer_connection_interface.h"
 #include "api/rtc_error.h"
 #include "api/rtp_parameters.h"
@@ -53,7 +54,6 @@
 #include "api/uma_metrics.h"
 #include "api/video/builtin_video_bitrate_allocator_factory.h"
 #include "api/video/video_codec_constants.h"
-#include "call/payload_type.h"
 #include "media/base/codec.h"
 #include "media/base/codec_comparators.h"
 #include "media/base/media_constants.h"
diff --git a/pc/test/fake_peer_connection_for_stats.h b/pc/test/fake_peer_connection_for_stats.h
index 94075fda6d..1e380820d6 100644
--- a/pc/test/fake_peer_connection_for_stats.h
+++ b/pc/test/fake_peer_connection_for_stats.h
@@ -504,9 +504,11 @@ class FakePeerConnectionForStats : public FakePeerConnectionBase,
   std::vector<scoped_refptr<RtpSenderInterface>> GetSenders() const override {
     std::vector<scoped_refptr<RtpSenderInterface>> senders;
     for (auto transceiver : transceivers_) {
+      RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
       for (auto sender : transceiver->internal()->senders()) {
         senders.push_back(sender);
       }
+      RTC_ALLOW_PLAN_B_DEPRECATION_END()
     }
     return senders;
   }
@@ -515,9 +517,11 @@ class FakePeerConnectionForStats : public FakePeerConnectionBase,
       const override {
     std::vector<scoped_refptr<RtpReceiverInterface>> receivers;
     for (auto transceiver : transceivers_) {
+      RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
       for (auto receiver : transceiver->internal()->receivers()) {
         receivers.push_back(receiver);
       }
+      RTC_ALLOW_PLAN_B_DEPRECATION_END()
     }
     return receivers;
   }

From ab408f48e234bb9c6f9d8aa264e3e8d496b5821a Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Fri, 27 Feb 2026 04:07:24 -0800
Subject: [PATCH 155/472] Roll chromium_revision d49ece3195..972b43f675
 (1590759:1591474)

Change log: https://chromium.googlesource.com/chromium/src/+log/d49ece3195..972b43f675
Full diff: https://chromium.googlesource.com/chromium/src/+/d49ece3195..972b43f675

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/07efd23b7d..371581ad06
* src/buildtools/linux64: git_revision:6767e99aea9e7c32b6fc421ad3271ab99a82ca78..git_revision:1a310e88443018837759c952b113846b0096f65b
* src/buildtools/mac: git_revision:6767e99aea9e7c32b6fc421ad3271ab99a82ca78..git_revision:1a310e88443018837759c952b113846b0096f65b
* src/buildtools/win: git_revision:6767e99aea9e7c32b6fc421ad3271ab99a82ca78..git_revision:1a310e88443018837759c952b113846b0096f65b
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/87538efd40..0b4e87c548
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/61da74e717..4f57914322
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/f309db1d4c..ac4bea0343
* src/third_party/android_build_tools/manifest_merger/cipd: vWr5VRbtPlKQeEMNOksUpsAqeH2yaPhBRf3C59yVhzQC..tjC-RIoLTs8HsZHxeRq-7GN90bYNYSQTx2yzmzL8hUcC
* src/third_party/android_deps/autorolled/cipd: pmna6eE_w31koBiBbnzTBXKHTCuDkgeg1ezshh73OnEC..cA8nRMj8uhS3aXj_1a5gxG7mzG-iKRo3ndtBCv5qszEC
* src/third_party/androidx/cipd: Pjbeih307Di9V8AT-ZSnNf8A118x4hWnhdLLideqj5wC..Di9v_vOtvM11yYkWUaYwjOdOqjM73OgEkVvRg0vXJXMC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/fa93d5c99f..80053a9bb7
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/1ec586469a..e507931d68
* src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/ecffa351c7..bc0a39e5b3
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/044f01ae03..3969ba2efb
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/a95440de17..f64d1e0acc
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/03052d67d5..78c96746a3
* src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/f4940a069b..c365236fa3
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/597706b827..bb0a61b409
* src/third_party/r8/cipd: OdXvkYaHnRN6eudfIbRYWbc4JTiLo6bJKvftjyuQt2UC..giX-zHOTxdkwnAiuPSCUktVPPlw_zCiSGvkGQKfFmNgC
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/c9c2c6b9fb..58325a01ba
* src/tools/luci-go: git_revision:f769e04338c2db24c3697f00ede9cadbd3ee7582..git_revision:1959c0f345fc4fc01903ce8437d507d9cf94e390
* src/tools/luci-go: git_revision:f769e04338c2db24c3697f00ede9cadbd3ee7582..git_revision:1959c0f345fc4fc01903ce8437d507d9cf94e390
* src/tools/luci-go: git_revision:f769e04338c2db24c3697f00ede9cadbd3ee7582..git_revision:1959c0f345fc4fc01903ce8437d507d9cf94e390
DEPS diff: https://chromium.googlesource.com/chromium/src/+/d49ece3195..972b43f675/DEPS

BUG=None

Change-Id: Ib76a76167bdc7a2265120bbf4fd0858d03bf61ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452640
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47008}
---
 DEPS | 48 ++++++++++++++++++++++++------------------------
 1 file changed, 24 insertions(+), 24 deletions(-)

diff --git a/DEPS b/DEPS
index e4f7d111c7..95033a2c07 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': 'd49ece31956a31fb839d437b39921f63fbc4c285',
+  'chromium_revision': '972b43f675102cd8b620038c64d00789c20924a5',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@07efd23b7d9cf9b6a59076d14f5a68cd0c4c1fa7',
+    'https://chromium.googlesource.com/chromium/src/build@371581ad0695c16f70aa1d7c060f5f51a188abb5',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@136da69a1267b8db487354b96d44d0cc8add5aeb',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,19 +78,19 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@87538efd40bd6fdc454e45e4aa22aed769a1d13d',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@0b4e87c5486f59384da7bcc11fd2a1d230b6f0d1',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@61da74e7174a660e6975d2143461ac55c68fbc8a',
+    'https://chromium.googlesource.com/chromium/src/testing@4f57914322cd5778334524da56b5fed52602b471',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@f309db1d4c0522b790e9b4f6f0b3672ac68490ae',
+    'https://chromium.googlesource.com/chromium/src/third_party@ac4bea03435519c8e20b21f9a20cd77af66bfbd3',
 
   'src/buildtools/linux64': {
     'packages': [
       {
         'package': 'gn/gn/linux-${{arch}}',
-        'version': 'git_revision:6767e99aea9e7c32b6fc421ad3271ab99a82ca78',
+        'version': 'git_revision:1a310e88443018837759c952b113846b0096f65b',
       }
     ],
     'dep_type': 'cipd',
@@ -100,7 +100,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/mac-${{arch}}',
-        'version': 'git_revision:6767e99aea9e7c32b6fc421ad3271ab99a82ca78',
+        'version': 'git_revision:1a310e88443018837759c952b113846b0096f65b',
       }
     ],
     'dep_type': 'cipd',
@@ -110,7 +110,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/windows-amd64',
-        'version': 'git_revision:6767e99aea9e7c32b6fc421ad3271ab99a82ca78',
+        'version': 'git_revision:1a310e88443018837759c952b113846b0096f65b',
       }
     ],
     'dep_type': 'cipd',
@@ -331,13 +331,13 @@ deps = {
   'src/third_party/clang-format/script':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@c2725e0622e1a86d55f14514f2177a39efea4a0e',
   'src/third_party/compiler-rt/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@ecffa351c7a4c9e792a9a940c7c7bb3c4523eebe',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@bc0a39e5b30af5cc47dd0df097a9b817e80ee3aa',
   'src/third_party/libc++/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@7ab65651aed6802d2599dcb7a73b1f82d5179d05',
   'src/third_party/libc++abi/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@8f11bb1d4438d0239d0dfc1bd9456a9f31629dda',
   'src/third_party/llvm-libc/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@f4940a069bf76a358d7cc7f525e1fa74b85f6ba0',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@c365236fa3dbdaff431a582f4eb04f6772d1e3da',
   'src/third_party/libunwind/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@17ccf7d110c5526cb77e93cfd8330f491fb2bf18',
 
@@ -488,11 +488,11 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@fa93d5c99f36f2c00fa341fb4f53d47a6f302da4',
+    'https://boringssl.googlesource.com/boringssl.git@80053a9bb7260a6979d09c19157136e59d97db76',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@9f26d631e4063c0af3e8ee3bc8826486115372a8',
   'src/third_party/catapult':
-    'https://chromium.googlesource.com/catapult.git@1ec586469aedb45f1e0809017409346ddf96e210',
+    'https://chromium.googlesource.com/catapult.git@e507931d6883f0216da761d51e165a0f775ec1a8',
   'src/third_party/ced/src': {
     'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5',
   },
@@ -505,13 +505,13 @@ deps = {
   'src/third_party/crc32c/src':
     'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558',
   'src/third_party/depot_tools':
-    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@044f01ae0351fb8a97efce042c93f99d3e95240c',
+    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@3969ba2efbbc9894f0b4140df211b70f073a5b35',
   'src/third_party/ffmpeg':
     'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@946d97db8d906277085e361892b7efda5152e2f1',
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@03052d67d5d9fa1c3a4b003f6bbfd031aab803a3',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@78c96746a3a149db4cf71de9a3882800fac04248',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '597706b827f0807c7b82180b1a636024224d8ce4',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'bb0a61b40921991a23ab2aec62e9528d9904f52e',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -644,7 +644,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/r8',
-              'version': 'OdXvkYaHnRN6eudfIbRYWbc4JTiLo6bJKvftjyuQt2UC',
+              'version': 'giX-zHOTxdkwnAiuPSCUktVPPlw_zCiSGvkGQKfFmNgC',
           },
       ],
       'condition': 'checkout_android',
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@c9c2c6b9fbb76ace51494fa3bb6fd94bec3c253c',
+    'https://chromium.googlesource.com/chromium/src/tools@58325a01ba56d27ad4c04a58148c40687af92647',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'Pjbeih307Di9V8AT-ZSnNf8A118x4hWnhdLLideqj5wC',
+          'version': 'Di9v_vOtvM11yYkWUaYwjOdOqjM73OgEkVvRg0vXJXMC',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -718,7 +718,7 @@ deps = {
       'packages': [
           {
                'package': 'chromium/third_party/android_build_tools/manifest_merger',
-               'version': 'vWr5VRbtPlKQeEMNOksUpsAqeH2yaPhBRf3C59yVhzQC',
+               'version': 'tjC-RIoLTs8HsZHxeRq-7GN90bYNYSQTx2yzmzL8hUcC',
           },
       ],
       'condition': 'checkout_android',
@@ -808,15 +808,15 @@ deps = {
       'packages': [
         {
           'package': 'infra/tools/luci/cas/${{platform}}',
-          'version': 'git_revision:f769e04338c2db24c3697f00ede9cadbd3ee7582',
+          'version': 'git_revision:1959c0f345fc4fc01903ce8437d507d9cf94e390',
         },
         {
           'package': 'infra/tools/luci/isolate/${{platform}}',
-          'version': 'git_revision:f769e04338c2db24c3697f00ede9cadbd3ee7582',
+          'version': 'git_revision:1959c0f345fc4fc01903ce8437d507d9cf94e390',
         },
         {
           'package': 'infra/tools/luci/swarming/${{platform}}',
-          'version': 'git_revision:f769e04338c2db24c3697f00ede9cadbd3ee7582',
+          'version': 'git_revision:1959c0f345fc4fc01903ce8437d507d9cf94e390',
         }
       ],
       'dep_type': 'cipd',
@@ -842,7 +842,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/android_deps/autorolled',
-              'version': 'pmna6eE_w31koBiBbnzTBXKHTCuDkgeg1ezshh73OnEC',
+              'version': 'cA8nRMj8uhS3aXj_1a5gxG7mzG-iKRo3ndtBCv5qszEC',
           },
       ],
       'condition': 'checkout_android and non_git_source',
@@ -864,7 +864,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7364b490b5f78d58efe23ea76e74210fd6c3c76f',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + 'a95440de174e5e4e0f979e829a56eda35af9e38f',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + 'f64d1e0acc2c2c33e325b8dd7b2b4673de2b9f14',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From 012ba18eab91ff6ec5991b7135c952ce7be94a5c Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Fri, 27 Feb 2026 11:46:29 +0000
Subject: [PATCH 156/472] Delete external authentication code.

The callers to this functionality seem to have disappeared over the 5 years since it was last looked at.

Bug: webrtc:487743907
Change-Id: I4f6b8d2ecaec4c0d97d9e1b44cb42870f0190de6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452400
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47009}
---
 BUILD.gn                        |   4 -
 api/rtp_parameters.cc           |   8 --
 pc/BUILD.gn                     |  17 ----
 pc/external_hmac.cc             | 154 --------------------------------
 pc/external_hmac.h              |  75 ----------------
 pc/jsep_transport_controller.cc |   7 --
 pc/jsep_transport_controller.h  |   1 -
 pc/peer_connection.cc           |   3 -
 pc/srtp_session.cc              |  62 -------------
 pc/srtp_session.h               |  16 ----
 pc/srtp_transport.cc            |  70 +--------------
 pc/srtp_transport.h             |  18 ----
 pc/srtp_transport_unittest.cc   | 122 ++++++-------------------
 webrtc.gni                      |   4 -
 14 files changed, 27 insertions(+), 534 deletions(-)
 delete mode 100644 pc/external_hmac.cc
 delete mode 100644 pc/external_hmac.h

diff --git a/BUILD.gn b/BUILD.gn
index 3b1c52b6ad..654077f303 100644
--- a/BUILD.gn
+++ b/BUILD.gn
@@ -332,10 +332,6 @@ config("common_config") {
     defines += [ "WEBRTC_HAVE_SCTP" ]
   }
 
-  if (rtc_enable_external_auth) {
-    defines += [ "ENABLE_EXTERNAL_AUTH" ]
-  }
-
   if (rtc_use_h264) {
     defines += [ "WEBRTC_USE_H264" ]
   }
diff --git a/api/rtp_parameters.cc b/api/rtp_parameters.cc
index 7ca5481bca..3a36bfe00a 100644
--- a/api/rtp_parameters.cc
+++ b/api/rtp_parameters.cc
@@ -248,14 +248,6 @@ bool RtpExtension::IsSupportedForVideo(absl::string_view uri) {
 
 bool RtpExtension::IsEncryptionSupported(absl::string_view uri) {
   return
-#if defined(ENABLE_EXTERNAL_AUTH)
-      // TODO(jbauch): Figure out a way to always allow "kAbsSendTimeUri"
-      // here and filter out later if external auth is really used in
-      // srtpfilter. External auth is used by Chromium and replaces the
-      // extension header value of "kAbsSendTimeUri", so it must not be
-      // encrypted (which can't be done by Chromium).
-      uri != RtpExtension::kAbsSendTimeUri &&
-#endif
       uri != RtpExtension::kEncryptHeaderExtensionsUri;
 }
 
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index d1d436c1d7..0d67ff1119 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -170,22 +170,6 @@ rtc_library("dtls_transport") {
   ]
 }
 
-rtc_library("external_hmac") {
-  visibility = [ ":*" ]
-  sources = [
-    "external_hmac.cc",
-    "external_hmac.h",
-  ]
-  deps = [
-    "../api:array_view",
-    "../rtc_base:logging",
-    "../rtc_base:zero_memory",
-  ]
-  if (rtc_build_libsrtp) {
-    deps += [ "//third_party/libsrtp" ]
-  }
-}
-
 rtc_library("ice_transport") {
   visibility = [ ":*" ]
   sources = [
@@ -696,7 +680,6 @@ rtc_library("srtp_session") {
     "srtp_session.h",
   ]
   deps = [
-    ":external_hmac",
     "../api:array_view",
     "../api:field_trials_view",
     "../api:sequence_checker",
diff --git a/pc/external_hmac.cc b/pc/external_hmac.cc
deleted file mode 100644
index e65d2b7073..0000000000
--- a/pc/external_hmac.cc
+++ /dev/null
@@ -1,154 +0,0 @@
-/*
- *  Copyright 2014 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "pc/external_hmac.h"
-
-#include <cstdint>
-#include <cstdlib>
-#include <cstring>
-#include <memory>
-
-#include "rtc_base/logging.h"
-#include "rtc_base/zero_memory.h"
-#include "third_party/libsrtp/crypto/include/auth.h"
-#include "third_party/libsrtp/include/srtp.h"
-
-namespace webrtc {
-namespace {
-
-// Begin test case 0 */
-constexpr uint8_t kExternalHmacTestCase0Key[20] = {
-    0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b,
-    0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b};
-
-constexpr uint8_t kExternalHmacTestCase0Data[8] = {
-    0x48, 0x69, 0x20, 0x54, 0x68, 0x65, 0x72, 0x65  // "Hi There"
-};
-
-constexpr uint8_t kExternalHmacFakeTag[10] = {0xba, 0xdd, 0xba, 0xdd, 0xba,
-                                              0xdd, 0xba, 0xdd, 0xba, 0xdd};
-
-constexpr srtp_auth_test_case_t kExternalHmacTestCase0 = {
-    .key_length_octets = 20,                                   // Octets in key
-    .key = const_cast<uint8_t*>(kExternalHmacTestCase0Key),    // Key
-    .data_length_octets = 8,                                   // Octets in data
-    .data = const_cast<uint8_t*>(kExternalHmacTestCase0Data),  // Data
-    .tag_length_octets = 10,                                   // Octets in tag
-    .tag = const_cast<uint8_t*>(kExternalHmacFakeTag),         // Tag
-    .next_test_case = nullptr  // Pointer to next
-                               // testcase
-};
-
-constexpr char kExternalHmacDescription[] =
-    "external hmac sha-1 authentication";
-
-// srtp_auth_type_t external_hmac is the hmac metaobject
-
-constexpr srtp_auth_type_t external_hmac = {
-    .alloc = external_hmac_alloc,
-    .dealloc = external_hmac_dealloc,
-    .init = external_hmac_init,
-    .compute = external_hmac_compute,
-    .update = external_hmac_update,
-    .start = external_hmac_start,
-    .description = const_cast<char*>(kExternalHmacDescription),
-    .test_data = const_cast<srtp_auth_test_case_t*>(&kExternalHmacTestCase0),
-    .id = EXTERNAL_HMAC_SHA1};
-
-}  // namespace
-
-srtp_err_status_t external_hmac_alloc(srtp_auth_t** a,
-                                      int key_len,
-                                      int out_len) {
-  // Check key length - note that we don't support keys larger
-  // than 20 bytes yet
-  if (key_len > 20)
-    return srtp_err_status_bad_param;
-
-  // Check output length - should be less than 20 bytes/
-  if (out_len > 20)
-    return srtp_err_status_bad_param;
-
-  // Allocate memory for auth and hmac_ctx_t structures.
-  auto raw_pointer = std::make_unique_for_overwrite<uint8_t[]>(
-      sizeof(srtp_auth_t) + sizeof(ExternalHmacContext));
-  if (raw_pointer == nullptr) {
-    return srtp_err_status_alloc_fail;
-  }
-  uint8_t* state_memory = &raw_pointer[sizeof(srtp_auth_t)];
-
-  // Set pointers
-  *a = reinterpret_cast<srtp_auth_t*>(raw_pointer.release());
-  // `external_hmac` is const and libsrtp expects `type` to be non-const.
-  // const conversion is required. `external_hmac` is constant because we don't
-  // want to increase global count in Chrome.
-  (*a)->type = const_cast<srtp_auth_type_t*>(&external_hmac);
-  (*a)->state = state_memory;
-  (*a)->out_len = out_len;
-  (*a)->key_len = key_len;
-  (*a)->prefix_len = 0;
-
-  return srtp_err_status_ok;
-}
-
-srtp_err_status_t external_hmac_dealloc(srtp_auth_t* a) {
-  ExplicitZeroMemory(a, sizeof(ExternalHmacContext) + sizeof(srtp_auth_t));
-
-  // Free memory
-  delete[] reinterpret_cast<uint8_t*>(a);
-
-  return srtp_err_status_ok;
-}
-
-srtp_err_status_t external_hmac_init(void* state,
-                                     const uint8_t* key,
-                                     int key_len) {
-  if (key_len > HMAC_KEY_LENGTH)
-    return srtp_err_status_bad_param;
-
-  ExternalHmacContext* context = static_cast<ExternalHmacContext*>(state);
-  memcpy(context->key, key, key_len);
-  context->key_length = key_len;
-  return srtp_err_status_ok;
-}
-
-srtp_err_status_t external_hmac_start(void* /*state*/) {
-  return srtp_err_status_ok;
-}
-
-srtp_err_status_t external_hmac_update(void* /*state*/,
-                                       const uint8_t* /*message*/,
-                                       int /*msg_octets*/) {
-  return srtp_err_status_ok;
-}
-
-srtp_err_status_t external_hmac_compute(void* /*state*/,
-                                        const uint8_t* /*message*/,
-                                        int /*msg_octets*/,
-                                        int tag_len,
-                                        uint8_t* result) {
-  memcpy(result, kExternalHmacFakeTag, tag_len);
-  return srtp_err_status_ok;
-}
-
-srtp_err_status_t external_crypto_init() {
-  // `external_hmac` is const. const_cast is required as libsrtp expects
-  // non-const.
-  srtp_err_status_t status = srtp_replace_auth_type(
-      const_cast<srtp_auth_type_t*>(&external_hmac), EXTERNAL_HMAC_SHA1);
-  if (status) {
-    RTC_LOG(LS_ERROR) << "Error in replacing default auth module, error: "
-                      << status;
-    return srtp_err_status_fail;
-  }
-  return srtp_err_status_ok;
-}
-
-}  // namespace webrtc
diff --git a/pc/external_hmac.h b/pc/external_hmac.h
deleted file mode 100644
index 04c9d3b504..0000000000
--- a/pc/external_hmac.h
+++ /dev/null
@@ -1,75 +0,0 @@
-/*
- *  Copyright 2014 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef PC_EXTERNAL_HMAC_H_
-#define PC_EXTERNAL_HMAC_H_
-
-// External libsrtp HMAC auth module which implements methods defined in
-// auth_type_t.
-// The default auth module will be replaced only when the ENABLE_EXTERNAL_AUTH
-// flag is enabled. This allows us to access to authentication keys,
-// as the default auth implementation doesn't provide access and avoids
-// hashing each packet twice.
-
-// How will libsrtp select this module?
-// Libsrtp defines authentication function types identified by an unsigned
-// integer, e.g. SRTP_HMAC_SHA1 is 3. Using authentication ids, the
-// application can plug any desired authentication modules into libsrtp.
-// libsrtp also provides a mechanism to select different auth functions for
-// individual streams. This can be done by setting the right value in
-// the auth_type of srtp_policy_t. The application must first register auth
-// functions and the corresponding authentication id using
-// crypto_kernel_replace_auth_type function.
-
-#include <stdint.h>
-
-#include "third_party/libsrtp/crypto/include/auth.h"
-#include "third_party/libsrtp/crypto/include/crypto_types.h"
-#include "third_party/libsrtp/include/srtp.h"
-
-namespace webrtc {
-
-#define EXTERNAL_HMAC_SHA1 SRTP_HMAC_SHA1 + 1
-#define HMAC_KEY_LENGTH 20
-
-// The HMAC context structure used to store authentication keys.
-// The pointer to the key  will be allocated in the external_hmac_init function.
-// This pointer is owned by srtp_t in a template context.
-struct ExternalHmacContext {
-  uint8_t key[HMAC_KEY_LENGTH];
-  int key_length;
-};
-
-srtp_err_status_t external_hmac_alloc(srtp_auth_t** a,
-                                      int key_len,
-                                      int out_len);
-
-srtp_err_status_t external_hmac_dealloc(srtp_auth_t* a);
-
-srtp_err_status_t external_hmac_init(void* state,
-                                     const uint8_t* key,
-                                     int key_len);
-
-srtp_err_status_t external_hmac_start(void* state);
-
-srtp_err_status_t external_hmac_update(void* state,
-                                       const uint8_t* message,
-                                       int msg_octets);
-
-srtp_err_status_t external_hmac_compute(void* state,
-                                        const uint8_t* message,
-                                        int msg_octets,
-                                        int tag_len,
-                                        uint8_t* result);
-
-srtp_err_status_t external_crypto_init();
-}  // namespace webrtc
-
-#endif  // PC_EXTERNAL_HMAC_H_
diff --git a/pc/jsep_transport_controller.cc b/pc/jsep_transport_controller.cc
index 61dfe73de7..d90cab3f4d 100644
--- a/pc/jsep_transport_controller.cc
+++ b/pc/jsep_transport_controller.cc
@@ -554,9 +554,6 @@ JsepTransportController::CreateDtlsSrtpTransport(
   RTC_DCHECK_RUN_ON(network_thread_);
   auto dtls_srtp_transport = std::make_unique<DtlsSrtpTransport>(
       rtcp_dtls_transport == nullptr, env_.field_trials());
-  if (config_.enable_external_auth) {
-    dtls_srtp_transport->EnableExternalAuth();
-  }
 
   dtls_srtp_transport->SetDtlsTransportsOwned(std::move(rtp_dtls_transport),
                                               std::move(rtcp_dtls_transport));
@@ -1072,10 +1069,6 @@ JsepTransportController::MergeEncryptedHeaderExtensionIdsForBundles(
 
 int JsepTransportController::GetRtpAbsSendTimeHeaderExtensionId(
     const ContentInfo& content_info) {
-  if (!config_.enable_external_auth) {
-    return -1;
-  }
-
   const MediaContentDescription* content_desc =
       content_info.media_description();
 
diff --git a/pc/jsep_transport_controller.h b/pc/jsep_transport_controller.h
index 260cc34ee9..ff86840532 100644
--- a/pc/jsep_transport_controller.h
+++ b/pc/jsep_transport_controller.h
@@ -107,7 +107,6 @@ class JsepTransportController final {
     PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy =
         PeerConnectionInterface::kRtcpMuxPolicyRequire;
     bool disable_encryption = false;
-    bool enable_external_auth = false;
     // Used to inject the ICE/DTLS/SRTP transports created externally.
     IceTransportFactory* ice_transport_factory = nullptr;
     DtlsTransportFactory* dtls_transport_factory = nullptr;
diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc
index 760bc8175c..57812745ea 100644
--- a/pc/peer_connection.cc
+++ b/pc/peer_connection.cc
@@ -744,9 +744,6 @@ JsepTransportController* PeerConnection::InitializeNetworkThread(
   config.transport_observer = this;
   config.rtcp_handler = InitializeRtcpCallback();
   config.un_demuxable_packet_handler = InitializeUnDemuxablePacketHandler();
-#if defined(ENABLE_EXTERNAL_AUTH)
-  config.enable_external_auth = true;
-#endif
 
   // DTLS has to be enabled to use SCTP.
   if (dtls_enabled_) {
diff --git a/pc/srtp_session.cc b/pc/srtp_session.cc
index 5c64265d0f..361c924485 100644
--- a/pc/srtp_session.cc
+++ b/pc/srtp_session.cc
@@ -18,21 +18,18 @@
 #include "api/array_view.h"
 #include "api/field_trials_view.h"
 #include "modules/rtp_rtcp/source/rtp_util.h"
-#include "pc/external_hmac.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/byte_order.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/copy_on_write_buffer.h"
 #include "rtc_base/ip_address.h"
 #include "rtc_base/logging.h"
-#include "rtc_base/ssl_stream_adapter.h"
 #include "rtc_base/synchronization/mutex.h"
 #include "rtc_base/text2pcap.h"
 #include "rtc_base/thread_annotations.h"
 #include "rtc_base/time_utils.h"
 #include "system_wrappers/include/metrics.h"
 #include "third_party/libsrtp/include/srtp.h"
-#include "third_party/libsrtp/include/srtp_priv.h"
 
 #ifndef SRTP_SRCTP_INDEX_LEN
 #define SRTP_SRCTP_INDEX_LEN 4
@@ -119,12 +116,6 @@ bool LibSrtpInitializer::IncrementLibsrtpUsageCountAndMaybeInit(
       RTC_LOG(LS_ERROR) << "Failed to install SRTP event handler, err=" << err;
       return false;
     }
-
-    err = external_crypto_init();
-    if (err != srtp_err_status_ok) {
-      RTC_LOG(LS_ERROR) << "Failed to initialize fake auth, err=" << err;
-      return false;
-    }
   }
   ++usage_count_;
   return true;
@@ -327,48 +318,6 @@ int SrtpSession::GetSrtpOverhead() const {
   return rtp_auth_tag_len_;
 }
 
-void SrtpSession::EnableExternalAuth() {
-  RTC_DCHECK(!session_);
-  external_auth_enabled_ = true;
-}
-
-bool SrtpSession::IsExternalAuthEnabled() const {
-  return external_auth_enabled_;
-}
-
-bool SrtpSession::IsExternalAuthActive() const {
-  return external_auth_active_;
-}
-
-bool SrtpSession::GetRtpAuthParams(uint8_t** key, int* key_len, int* tag_len) {
-  RTC_DCHECK(thread_checker_.IsCurrent());
-  RTC_DCHECK(IsExternalAuthActive());
-  if (!IsExternalAuthActive()) {
-    return false;
-  }
-
-  ExternalHmacContext* external_hmac = nullptr;
-  // stream_template will be the reference context for other streams.
-  // Let's use it for getting the keys.
-  srtp_stream_ctx_t* srtp_context = session_->stream_template;
-  if (srtp_context && srtp_context->session_keys &&
-      srtp_context->session_keys->rtp_auth) {
-    external_hmac = reinterpret_cast<ExternalHmacContext*>(
-        srtp_context->session_keys->rtp_auth->state);
-  }
-
-  if (!external_hmac) {
-    RTC_LOG(LS_ERROR) << "Failed to get auth keys from libsrtp!.";
-    return false;
-  }
-
-  *key = external_hmac->key;
-  *key_len = external_hmac->key_length;
-  *tag_len = rtp_auth_tag_len_;
-  return true;
-}
-
-
 bool SrtpSession::RemoveSsrcFromSession(uint32_t ssrc) {
   RTC_DCHECK(session_);
   // libSRTP expects the SSRC to be in network byte order.
@@ -424,16 +373,6 @@ bool SrtpSession::DoSetKey(int type,
   // TODO(astor) parse window size from WSH session-param
   policy.window_size = 1024;
   policy.allow_repeat_tx = 1;
-  // If external authentication option is enabled, supply custom auth module
-  // id EXTERNAL_HMAC_SHA1 in the policy structure.
-  // We want to set this option only for rtp packets.
-  // By default policy structure is initialized to HMAC_SHA1.
-  // Enable external HMAC authentication only for outgoing streams and only
-  // for cipher suites that support it (i.e. only non-GCM cipher suites).
-  if (type == ssrc_any_outbound && IsExternalAuthEnabled() &&
-      !IsGcmCryptoSuite(crypto_suite)) {
-    policy.rtp.auth_type = EXTERNAL_HMAC_SHA1;
-  }
   if (!extension_ids.empty()) {
     policy.enc_xtn_hdr = const_cast<int*>(&extension_ids[0]);
     policy.enc_xtn_hdr_count = static_cast<int>(extension_ids.size());
@@ -458,7 +397,6 @@ bool SrtpSession::DoSetKey(int type,
 
   rtp_auth_tag_len_ = policy.rtp.auth_tag_len;
   rtcp_auth_tag_len_ = policy.rtcp.auth_tag_len;
-  external_auth_active_ = (policy.rtp.auth_type == EXTERNAL_HMAC_SHA1);
   return true;
 }
 
diff --git a/pc/srtp_session.h b/pc/srtp_session.h
index 059e715eed..3e4581ca83 100644
--- a/pc/srtp_session.h
+++ b/pc/srtp_session.h
@@ -73,20 +73,6 @@ class SrtpSession {
 
   int GetSrtpOverhead() const;
 
-  // If external auth is enabled, SRTP will write a dummy auth tag that then
-  // later must get replaced before the packet is sent out. Only supported for
-  // non-GCM cipher suites and can be checked through "IsExternalAuthActive"
-  // if it is actually used. This method is only valid before the RTP params
-  // have been set.
-  void EnableExternalAuth();
-  bool IsExternalAuthEnabled() const;
-  // Helper method to get authentication params.
-  bool GetRtpAuthParams(uint8_t** key, int* key_len, int* tag_len);
-  // A SRTP session supports external creation of the auth tag if a non-GCM
-  // cipher is used. This method is only valid after the RTP params have
-  // been set.
-  bool IsExternalAuthActive() const;
-
   // Removes a SSRC from the underlying libSRTP session.
   // Note: this should only be done for SSRCs that are received.
   // Removing SSRCs that were sent and then reusing them leads to
@@ -130,8 +116,6 @@ class SrtpSession {
 
   bool inited_ = false;
   int last_send_seq_num_ = -1;
-  bool external_auth_active_ = false;
-  bool external_auth_enabled_ = false;
   int decryption_failure_count_ = 0;
   bool dump_plain_rtp_ = false;
 };
diff --git a/pc/srtp_transport.cc b/pc/srtp_transport.cc
index 5c2c3c930a..3eaf514fd3 100644
--- a/pc/srtp_transport.cc
+++ b/pc/srtp_transport.cc
@@ -49,39 +49,8 @@ bool SrtpTransport::SendRtpPacket(CopyOnWriteBuffer* packet,
         << "Failed to send the packet because SRTP transport is inactive.";
     return false;
   }
-  AsyncSocketPacketOptions updated_options = options;
   TRACE_EVENT0("webrtc", "SRTP Encode");
-  // If ENABLE_EXTERNAL_AUTH flag is on then packet authentication is not done
-  // inside libsrtp for a RTP packet. A external HMAC module will be writing
-  // a fake HMAC value. This is ONLY done for a RTP packet.
-  // Socket layer will update rtp sendtime extension header if present in
-  // packet with current time before updating the HMAC.
-  bool res;
-#if !defined(ENABLE_EXTERNAL_AUTH)
-  res = ProtectRtp(*packet);
-#else
-  if (!IsExternalAuthActive()) {
-    res = ProtectRtp(*packet);
-  } else {
-    updated_options.packet_time_params.rtp_sendtime_extension_id =
-        rtp_abs_sendtime_extn_id_;
-    res = ProtectRtp(*packet,
-                     &updated_options.packet_time_params.srtp_packet_index);
-    // If protection succeeds, let's get auth params from srtp.
-    if (res) {
-      uint8_t* auth_key = nullptr;
-      int key_len = 0;
-      res = GetRtpAuthParams(
-          &auth_key, &key_len,
-          &updated_options.packet_time_params.srtp_auth_tag_len);
-      if (res) {
-        updated_options.packet_time_params.srtp_auth_key.resize(key_len);
-        updated_options.packet_time_params.srtp_auth_key.assign(
-            auth_key, auth_key + key_len);
-      }
-    }
-  }
-#endif
+  bool res = ProtectRtp(*packet);
   if (!res) {
     uint16_t seq_num = ParseRtpSequenceNumber(*packet);
     uint32_t ssrc = ParseRtpSsrc(*packet);
@@ -90,7 +59,7 @@ bool SrtpTransport::SendRtpPacket(CopyOnWriteBuffer* packet,
     return false;
   }
 
-  return RtpTransport::SendRtpPacket(packet, updated_options, flags);
+  return RtpTransport::SendRtpPacket(packet, options, flags);
 }
 
 bool SrtpTransport::SendRtcpPacket(CopyOnWriteBuffer* packet,
@@ -276,9 +245,6 @@ void SrtpTransport::ResetParams() {
 void SrtpTransport::CreateSrtpSessions() {
   send_session_.reset(new SrtpSession(field_trials_));
   recv_session_.reset(new SrtpSession(field_trials_));
-  if (external_auth_enabled_) {
-    send_session_->EnableExternalAuth();
-  }
 }
 
 bool SrtpTransport::ProtectRtp(CopyOnWriteBuffer& buffer) {
@@ -334,18 +300,6 @@ bool SrtpTransport::UnprotectRtcp(CopyOnWriteBuffer& buffer) {
   }
 }
 
-bool SrtpTransport::GetRtpAuthParams(uint8_t** key,
-                                     int* key_len,
-                                     int* tag_len) {
-  if (!IsSrtpActive()) {
-    RTC_LOG(LS_WARNING) << "Failed to GetRtpAuthParams: SRTP not active";
-    return false;
-  }
-
-  RTC_CHECK(send_session_);
-  return send_session_->GetRtpAuthParams(key, key_len, tag_len);
-}
-
 bool SrtpTransport::GetSrtpOverhead(int* srtp_overhead) const {
   if (!IsSrtpActive()) {
     RTC_LOG(LS_WARNING) << "Failed to GetSrtpOverhead: SRTP not active";
@@ -357,26 +311,6 @@ bool SrtpTransport::GetSrtpOverhead(int* srtp_overhead) const {
   return true;
 }
 
-void SrtpTransport::EnableExternalAuth() {
-  RTC_DCHECK(!IsSrtpActive());
-  external_auth_enabled_ = true;
-}
-
-bool SrtpTransport::IsExternalAuthEnabled() const {
-  return external_auth_enabled_;
-}
-
-bool SrtpTransport::IsExternalAuthActive() const {
-  if (!IsSrtpActive()) {
-    RTC_LOG(LS_WARNING)
-        << "Failed to check IsExternalAuthActive: SRTP not active";
-    return false;
-  }
-
-  RTC_CHECK(send_session_);
-  return send_session_->IsExternalAuthActive();
-}
-
 void SrtpTransport::MaybeUpdateWritableState() {
   bool writable = IsWritable(/*rtcp=*/true) && IsWritable(/*rtcp=*/false);
   // Only fire the signal if the writable state changes.
diff --git a/pc/srtp_transport.h b/pc/srtp_transport.h
index e14ebb4120..4f8e23a2c5 100644
--- a/pc/srtp_transport.h
+++ b/pc/srtp_transport.h
@@ -77,25 +77,9 @@ class SrtpTransport : public RtpTransport {
 
   void ResetParams();
 
-  // If external auth is enabled, SRTP will write a dummy auth tag that then
-  // later must get replaced before the packet is sent out. Only supported for
-  // non-GCM crypto suites and can be checked through "IsExternalAuthActive"
-  // if it is actually used. This method is only valid before the RTP params
-  // have been set.
-  void EnableExternalAuth();
-  bool IsExternalAuthEnabled() const;
-
-  // A SrtpTransport supports external creation of the auth tag if a non-GCM
-  // cipher is used. This method is only valid after the RTP params have
-  // been set.
-  bool IsExternalAuthActive() const;
-
   // Returns srtp overhead for rtp packets.
   bool GetSrtpOverhead(int* srtp_overhead) const;
 
-  // Returns rtp auth params from srtp context.
-  bool GetRtpAuthParams(uint8_t** key, int* key_len, int* tag_len);
-
   // Cache RTP Absoulute SendTime extension header ID. This is only used when
   // external authentication is enabled.
   void CacheRtpAbsSendTimeHeaderExtension(int rtp_abs_sendtime_extn_id) {
@@ -146,8 +130,6 @@ class SrtpTransport : public RtpTransport {
 
   bool writable_ = false;
 
-  bool external_auth_enabled_ = false;
-
   int rtp_abs_sendtime_extn_id_ = -1;
 
   int decryption_failure_count_ = 0;
diff --git a/pc/srtp_transport_unittest.cc b/pc/srtp_transport_unittest.cc
index e97260f31f..3ef3545876 100644
--- a/pc/srtp_transport_unittest.cc
+++ b/pc/srtp_transport_unittest.cc
@@ -30,7 +30,6 @@
 #include "rtc_base/async_packet_socket.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/byte_order.h"
-#include "rtc_base/checks.h"
 #include "rtc_base/containers/flat_set.h"
 #include "rtc_base/copy_on_write_buffer.h"
 #include "rtc_base/ssl_stream_adapter.h"
@@ -106,34 +105,6 @@ class SrtpTransportTest : public ::testing::Test {
     }
   }
 
-  // With external auth enabled, SRTP doesn't write the auth tag and
-  // unprotect would fail. Check accessing the information about the
-  // tag instead, similar to what the actual code would do that relies
-  // on external auth.
-  void TestRtpAuthParams(SrtpTransport* transport, int crypto_suite) {
-    int overhead;
-    EXPECT_TRUE(transport->GetSrtpOverhead(&overhead));
-    switch (crypto_suite) {
-      case kSrtpAes128CmSha1_32:
-        EXPECT_EQ(32 / 8, overhead);  // 32-bit tag.
-        break;
-      case kSrtpAes128CmSha1_80:
-        EXPECT_EQ(80 / 8, overhead);  // 80-bit tag.
-        break;
-      default:
-        RTC_DCHECK_NOTREACHED();
-        break;
-    }
-
-    uint8_t* auth_key = nullptr;
-    int key_len = 0;
-    int tag_len = 0;
-    EXPECT_TRUE(transport->GetRtpAuthParams(&auth_key, &key_len, &tag_len));
-    EXPECT_NE(nullptr, auth_key);
-    EXPECT_EQ(160 / 8, key_len);  // Length of SHA-1 is 160 bits.
-    EXPECT_EQ(overhead, tag_len);
-  }
-
   void TestSendRecvRtpPacket(int crypto_suite) {
     size_t rtp_len = sizeof(kPcmuFrame);
     size_t packet_size = rtp_len + rtp_auth_tag_len(crypto_suite);
@@ -155,34 +126,26 @@ class SrtpTransportTest : public ::testing::Test {
     // that the packet can be successfully received and decrypted.
     ASSERT_TRUE(srtp_transport1_->SendRtpPacket(&rtp_packet1to2, options,
                                                 PF_SRTP_BYPASS));
-    if (srtp_transport1_->IsExternalAuthActive()) {
-      TestRtpAuthParams(srtp_transport1_.get(), crypto_suite);
-    } else {
-      ASSERT_TRUE(rtp_sink2_.last_recv_rtp_packet().data());
-      EXPECT_EQ(0, memcmp(rtp_sink2_.last_recv_rtp_packet().data(),
-                          original_rtp_data, rtp_len));
-      // Get the encrypted packet from underneath packet transport and verify
-      // the data is actually encrypted.
-      auto fake_rtp_packet_transport = static_cast<FakePacketTransport*>(
-          srtp_transport1_->rtp_packet_transport());
-      EXPECT_NE(0, memcmp(fake_rtp_packet_transport->last_sent_packet()->data(),
-                          original_rtp_data, rtp_len));
-    }
+    ASSERT_TRUE(rtp_sink2_.last_recv_rtp_packet().data());
+    EXPECT_EQ(0, memcmp(rtp_sink2_.last_recv_rtp_packet().data(),
+                        original_rtp_data, rtp_len));
+    // Get the encrypted packet from underneath packet transport and verify
+    // the data is actually encrypted.
+    auto fake_rtp_packet_transport = static_cast<FakePacketTransport*>(
+        srtp_transport1_->rtp_packet_transport());
+    EXPECT_NE(0, memcmp(fake_rtp_packet_transport->last_sent_packet()->data(),
+                        original_rtp_data, rtp_len));
 
     // Do the same thing in the opposite direction;
     ASSERT_TRUE(srtp_transport2_->SendRtpPacket(&rtp_packet2to1, options,
                                                 PF_SRTP_BYPASS));
-    if (srtp_transport2_->IsExternalAuthActive()) {
-      TestRtpAuthParams(srtp_transport2_.get(), crypto_suite);
-    } else {
-      ASSERT_TRUE(rtp_sink1_.last_recv_rtp_packet().data());
-      EXPECT_EQ(0, memcmp(rtp_sink1_.last_recv_rtp_packet().data(),
-                          original_rtp_data, rtp_len));
-      auto fake_rtp_packet_transport = static_cast<FakePacketTransport*>(
-          srtp_transport2_->rtp_packet_transport());
-      EXPECT_NE(0, memcmp(fake_rtp_packet_transport->last_sent_packet()->data(),
-                          original_rtp_data, rtp_len));
-    }
+    ASSERT_TRUE(rtp_sink1_.last_recv_rtp_packet().data());
+    EXPECT_EQ(0, memcmp(rtp_sink1_.last_recv_rtp_packet().data(),
+                        original_rtp_data, rtp_len));
+    fake_rtp_packet_transport = static_cast<FakePacketTransport*>(
+        srtp_transport2_->rtp_packet_transport());
+    EXPECT_NE(0, memcmp(fake_rtp_packet_transport->last_sent_packet()->data(),
+                        original_rtp_data, rtp_len));
   }
 
   void TestSendRecvRtcpPacket(int crypto_suite) {
@@ -223,15 +186,10 @@ class SrtpTransportTest : public ::testing::Test {
                         rtcp_packet_data, rtcp_len));
   }
 
-  void TestSendRecvPacket(bool enable_external_auth,
-                          int crypto_suite,
+  void TestSendRecvPacket(int crypto_suite,
                           const ZeroOnFreeBuffer<uint8_t>& key1,
                           const ZeroOnFreeBuffer<uint8_t>& key2) {
     EXPECT_EQ(key1.size(), key2.size());
-    if (enable_external_auth) {
-      srtp_transport1_->EnableExternalAuth();
-      srtp_transport2_->EnableExternalAuth();
-    }
     std::vector<int> extension_ids;
     EXPECT_TRUE(srtp_transport1_->SetRtpParams(
         crypto_suite, key1, extension_ids, crypto_suite, key2, extension_ids));
@@ -243,13 +201,6 @@ class SrtpTransportTest : public ::testing::Test {
         crypto_suite, key2, extension_ids, crypto_suite, key1, extension_ids));
     EXPECT_TRUE(srtp_transport1_->IsSrtpActive());
     EXPECT_TRUE(srtp_transport2_->IsSrtpActive());
-    if (IsGcmCryptoSuite(crypto_suite)) {
-      EXPECT_FALSE(srtp_transport1_->IsExternalAuthActive());
-      EXPECT_FALSE(srtp_transport2_->IsExternalAuthActive());
-    } else if (enable_external_auth) {
-      EXPECT_TRUE(srtp_transport1_->IsExternalAuthActive());
-      EXPECT_TRUE(srtp_transport2_->IsExternalAuthActive());
-    }
     TestSendRecvRtpPacket(crypto_suite);
     TestSendRecvRtcpPacket(crypto_suite);
   }
@@ -326,8 +277,6 @@ class SrtpTransportTest : public ::testing::Test {
                                                key1, encrypted_headers));
     EXPECT_TRUE(srtp_transport1_->IsSrtpActive());
     EXPECT_TRUE(srtp_transport2_->IsSrtpActive());
-    EXPECT_FALSE(srtp_transport1_->IsExternalAuthActive());
-    EXPECT_FALSE(srtp_transport2_->IsExternalAuthActive());
     TestSendRecvPacketWithEncryptedHeaderExtension(crypto_suite,
                                                    encrypted_headers);
   }
@@ -346,15 +295,8 @@ class SrtpTransportTest : public ::testing::Test {
   FieldTrials field_trials_ = CreateTestFieldTrials();
 };
 
-class SrtpTransportTestWithExternalAuth
-    : public SrtpTransportTest,
-      public ::testing::WithParamInterface<bool> {};
-
-TEST_P(SrtpTransportTestWithExternalAuth,
-       SendAndRecvPacket_AES_CM_128_HMAC_SHA1_80) {
-  bool enable_external_auth = GetParam();
-  TestSendRecvPacket(enable_external_auth, kSrtpAes128CmSha1_80, kTestKey1,
-                     kTestKey2);
+TEST_F(SrtpTransportTest, SendAndRecvPacket_AES_CM_128_HMAC_SHA1_80) {
+  TestSendRecvPacket(kSrtpAes128CmSha1_80, kTestKey1, kTestKey2);
 }
 
 TEST_F(SrtpTransportTest,
@@ -363,11 +305,8 @@ TEST_F(SrtpTransportTest,
                                        kTestKey2);
 }
 
-TEST_P(SrtpTransportTestWithExternalAuth,
-       SendAndRecvPacket_AES_CM_128_HMAC_SHA1_32) {
-  bool enable_external_auth = GetParam();
-  TestSendRecvPacket(enable_external_auth, kSrtpAes128CmSha1_32, kTestKey1,
-                     kTestKey2);
+TEST_F(SrtpTransportTest, SendAndRecvPacket_AES_CM_128_HMAC_SHA1_32) {
+  TestSendRecvPacket(kSrtpAes128CmSha1_32, kTestKey1, kTestKey2);
 }
 
 TEST_F(SrtpTransportTest,
@@ -376,11 +315,8 @@ TEST_F(SrtpTransportTest,
                                        kTestKey2);
 }
 
-TEST_P(SrtpTransportTestWithExternalAuth,
-       SendAndRecvPacket_kSrtpAeadAes128Gcm) {
-  bool enable_external_auth = GetParam();
-  TestSendRecvPacket(enable_external_auth, kSrtpAeadAes128Gcm, kTestKeyGcm128_1,
-                     kTestKeyGcm128_2);
+TEST_F(SrtpTransportTest, SendAndRecvPacket_kSrtpAeadAes128Gcm) {
+  TestSendRecvPacket(kSrtpAeadAes128Gcm, kTestKeyGcm128_1, kTestKeyGcm128_2);
 }
 
 TEST_F(SrtpTransportTest,
@@ -389,11 +325,8 @@ TEST_F(SrtpTransportTest,
                                        kTestKeyGcm128_2);
 }
 
-TEST_P(SrtpTransportTestWithExternalAuth,
-       SendAndRecvPacket_kSrtpAeadAes256Gcm) {
-  bool enable_external_auth = GetParam();
-  TestSendRecvPacket(enable_external_auth, kSrtpAeadAes256Gcm, kTestKeyGcm256_1,
-                     kTestKeyGcm256_2);
+TEST_F(SrtpTransportTest, SendAndRecvPacket_kSrtpAeadAes256Gcm) {
+  TestSendRecvPacket(kSrtpAeadAes256Gcm, kTestKeyGcm256_1, kTestKeyGcm256_2);
 }
 
 TEST_F(SrtpTransportTest,
@@ -402,11 +335,6 @@ TEST_F(SrtpTransportTest,
                                        kTestKeyGcm256_2);
 }
 
-// Run all tests both with and without external auth enabled.
-INSTANTIATE_TEST_SUITE_P(ExternalAuth,
-                         SrtpTransportTestWithExternalAuth,
-                         ::testing::Values(true, false));
-
 // Test directly setting the params with bogus keys.
 TEST_F(SrtpTransportTest, TestSetParamsKeyTooShort) {
   std::vector<int> extension_ids;
diff --git a/webrtc.gni b/webrtc.gni
index c78fc8a0b0..6eb16cf188 100644
--- a/webrtc.gni
+++ b/webrtc.gni
@@ -104,10 +104,6 @@ declare_args() {
   # library that comes with WebRTC (i.e. rtc_build_ssl == 0).
   rtc_ssl_root = ""
 
-  # Enable when an external authentication mechanism is used for performing
-  # packet authentication for RTP packets instead of libsrtp.
-  rtc_enable_external_auth = build_with_chromium
-
   # Selects whether debug dumps for the audio processing module
   # should be generated.
   apm_debug_dump = false

From 19e62b7f55673239e1f0045339e432fdd01c1eda Mon Sep 17 00:00:00 2001
From: Philipp Hancke <philipp.hancke@googlemail.com>
Date: Fri, 27 Feb 2026 11:54:08 +0100
Subject: [PATCH 157/472] dtls-in-stun: add status to piggyback complete
 callback

which allows the transport and unit tests to know whether
the handshake was completed successfully.

Also make the callback moveable (and curse cpplint)

Bug: webrtc:367395350
Change-Id: I59d67246ad5e7aba8c2b23ddddea292414c2af12
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/450982
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#47010}
---
 p2p/dtls/dtls_ice_integration_test.cc         |  2 --
 p2p/dtls/dtls_stun_piggyback_controller.cc    | 23 +++++++++-------
 p2p/dtls/dtls_stun_piggyback_controller.h     |  8 +++---
 ...dtls_stun_piggyback_controller_unittest.cc | 27 ++++++++++---------
 p2p/dtls/dtls_transport.cc                    | 14 ++++++----
 p2p/dtls/dtls_transport.h                     |  2 +-
 6 files changed, 43 insertions(+), 33 deletions(-)

diff --git a/p2p/dtls/dtls_ice_integration_test.cc b/p2p/dtls/dtls_ice_integration_test.cc
index 20b91f400e..0ff6f24c7d 100644
--- a/p2p/dtls/dtls_ice_integration_test.cc
+++ b/p2p/dtls/dtls_ice_integration_test.cc
@@ -461,8 +461,6 @@ TEST_P(DtlsIceIntegrationPerformanceTest, ConnectTime) {
     }
   }
 
-  RTC_LOG(LS_INFO) << "START: " << GetParam();
-
   int iter = 50;
   // Use a fixed seed to get consistent behavior.
   Random rand(/* seed= */ 77);
diff --git a/p2p/dtls/dtls_stun_piggyback_controller.cc b/p2p/dtls/dtls_stun_piggyback_controller.cc
index 2f056c9886..42e4be6ed7 100644
--- a/p2p/dtls/dtls_stun_piggyback_controller.cc
+++ b/p2p/dtls/dtls_stun_piggyback_controller.cc
@@ -32,11 +32,14 @@ namespace webrtc {
 
 DtlsStunPiggybackController::DtlsStunPiggybackController(
     absl::AnyInvocable<void(ArrayView<const uint8_t>)> dtls_data_callback,
-    absl::AnyInvocable<void()> complete_callback)
+    // NOLINTNEXTLINE(readability/casting) - not a cast; false positive!
+    absl::AnyInvocable<void(bool) &&> piggyback_complete_callback)
     : dtls_data_callback_(std::move(dtls_data_callback)),
-      complete_callback_(std::move(complete_callback)) {}
+      piggyback_complete_callback_(std::move(piggyback_complete_callback)) {}
 
 DtlsStunPiggybackController::~DtlsStunPiggybackController() {
+  RTC_DCHECK(dtls_data_callback_);
+  RTC_DCHECK(piggyback_complete_callback_);
 }
 
 void DtlsStunPiggybackController::SetDtlsHandshakeComplete(bool is_dtls_client,
@@ -74,7 +77,7 @@ void DtlsStunPiggybackController::ApplicationPacketReceived(
     }
   }
   state_ = State::COMPLETE;
-  CallCompleteCallback();
+  CallCompleteCallback(/*success=*/true);
 }
 
 void DtlsStunPiggybackController::SetDtlsFailed() {
@@ -86,7 +89,7 @@ void DtlsStunPiggybackController::SetDtlsFailed() {
         << "DTLS-STUN piggybacking DTLS failed during negotiation.";
   }
   state_ = State::OFF;
-  CallCompleteCallback();
+  CallCompleteCallback(/*success=*/false);
 }
 
 void DtlsStunPiggybackController::CapturePacket(ArrayView<const uint8_t> data) {
@@ -181,7 +184,7 @@ void DtlsStunPiggybackController::ReportDataPiggybacked(
     state_ = State::OFF;
     // TODO: bugs.webrtc.org/367395350 - We should call CallCompleteCallback
     // here but this causes a slew of failed tests. Investigate why!
-    // CallCompleteCallback();
+    // CallCompleteCallback(/*success=*/false);
     return;
   }
 
@@ -190,7 +193,7 @@ void DtlsStunPiggybackController::ReportDataPiggybacked(
   if (state_ == State::PENDING && !data.has_value() && !acks.has_value()) {
     RTC_LOG(LS_INFO) << "DTLS-STUN piggybacking complete.";
     state_ = State::COMPLETE;
-    CallCompleteCallback();
+    CallCompleteCallback(/*success=*/true);
     return;
   }
 
@@ -227,7 +230,7 @@ void DtlsStunPiggybackController::ReportDataPiggybacked(
   if (!data.has_value() && acks.has_value() && state_ == State::PENDING) {
     RTC_LOG(LS_INFO) << "DTLS-STUN piggybacking complete.";
     state_ = State::COMPLETE;
-    CallCompleteCallback();
+    CallCompleteCallback(/*success=*/true);
     return;
   }
 
@@ -272,15 +275,15 @@ void DtlsStunPiggybackController::ReportDtlsPacket(
   }
 }
 
-void DtlsStunPiggybackController::CallCompleteCallback() {
+void DtlsStunPiggybackController::CallCompleteCallback(bool success) {
   RTC_DCHECK_RUN_ON(&sequence_checker_);
   pending_packets_.clear();
   handshake_messages_received_.clear();
-  if (!complete_callback_) {
+  if (!piggyback_complete_callback_) {
     RTC_DCHECK_NOTREACHED() << "CompleteCallback called twice!";
     return;
   }
-  std::move(complete_callback_)();
+  std::move(piggyback_complete_callback_)(success);
 }
 
 }  // namespace webrtc
diff --git a/p2p/dtls/dtls_stun_piggyback_controller.h b/p2p/dtls/dtls_stun_piggyback_controller.h
index 72a2e124bd..89cd27be1c 100644
--- a/p2p/dtls/dtls_stun_piggyback_controller.h
+++ b/p2p/dtls/dtls_stun_piggyback_controller.h
@@ -38,7 +38,8 @@ class DtlsStunPiggybackController {
   // piggybacked.
   DtlsStunPiggybackController(
       absl::AnyInvocable<void(ArrayView<const uint8_t>)> dtls_data_callback,
-      absl::AnyInvocable<void()> complete_callback);
+      // NOLINTNEXTLINE(readability/casting) - not a cast; false positive!
+      absl::AnyInvocable<void(bool) &&> piggyback_complete_callback);
 
   ~DtlsStunPiggybackController();
 
@@ -110,7 +111,8 @@ class DtlsStunPiggybackController {
   PacketStash pending_packets_ RTC_GUARDED_BY(sequence_checker_);
   absl::AnyInvocable<void(ArrayView<const uint8_t>)> dtls_data_callback_
       RTC_GUARDED_BY(sequence_checker_);
-  absl::AnyInvocable<void() &&> complete_callback_
+  // NOLINTNEXTLINE(readability/casting) - not a cast; false positive!
+  absl::AnyInvocable<void(bool) &&> piggyback_complete_callback_
       RTC_GUARDED_BY(sequence_checker_);
 
   std::vector<uint32_t> handshake_messages_received_
@@ -119,7 +121,7 @@ class DtlsStunPiggybackController {
   // Count of embedded data attributes received.
   int data_recv_count_ = 0;
 
-  void CallCompleteCallback();
+  void CallCompleteCallback(bool success);
 
   // In practice this will be the network thread.
   RTC_NO_UNIQUE_ADDRESS SequenceChecker sequence_checker_;
diff --git a/p2p/dtls/dtls_stun_piggyback_controller_unittest.cc b/p2p/dtls/dtls_stun_piggyback_controller_unittest.cc
index 8409202414..66f5d3a354 100644
--- a/p2p/dtls/dtls_stun_piggyback_controller_unittest.cc
+++ b/p2p/dtls/dtls_stun_piggyback_controller_unittest.cc
@@ -112,10 +112,10 @@ class DtlsStunPiggybackControllerTest : public ::testing::Test {
   DtlsStunPiggybackControllerTest()
       : client_(
             [this](ArrayView<const uint8_t> data) { ClientPacketSink(data); },
-            [this]() { ClientCompleteCallback(); }),
+            [this](bool success) { ClientCompleteCallback(success); }),
         server_(
             [this](ArrayView<const uint8_t> data) { ServerPacketSink(data); },
-            [this]() { ServerCompleteCallback(); }),
+            [this](bool success) { ServerCompleteCallback(success); }),
         packet_(kPayload, SocketAddress(), std::nullopt) {}
 
   // Send from client to server embedded in STUN.
@@ -199,8 +199,8 @@ class DtlsStunPiggybackControllerTest : public ::testing::Test {
   MOCK_METHOD(void, ClientPacketSink, (ArrayView<const uint8_t>));
   MOCK_METHOD(void, ServerPacketSink, (ArrayView<const uint8_t>));
 
-  MOCK_METHOD(void, ClientCompleteCallback, ());
-  MOCK_METHOD(void, ServerCompleteCallback, ());
+  MOCK_METHOD(void, ClientCompleteCallback, (bool));
+  MOCK_METHOD(void, ServerCompleteCallback, (bool));
 
   ReceivedIpPacket packet_;
 
@@ -231,9 +231,9 @@ TEST_F(DtlsStunPiggybackControllerTest, BasicHandshake) {
   EXPECT_EQ(client_.state(), State::PENDING);
 
   // Post-handshake ACK
-  EXPECT_CALL(*this, ClientCompleteCallback);
+  EXPECT_CALL(*this, ClientCompleteCallback(true));
   SendServerToClientEmbedded(empty, STUN_BINDING_REQUEST);
-  EXPECT_CALL(*this, ServerCompleteCallback);
+  EXPECT_CALL(*this, ServerCompleteCallback(true));
   SendClientToServerEmbedded(empty, STUN_BINDING_RESPONSE);
   EXPECT_EQ(server_.state(), State::COMPLETE);
   EXPECT_EQ(client_.state(), State::COMPLETE);
@@ -315,13 +315,13 @@ TEST_F(DtlsStunPiggybackControllerTest, FirstClientPacketLost) {
 
   // Flight 4
   SendServerToClientEmbedded(dtls_flight4, STUN_BINDING_REQUEST);
-  EXPECT_CALL(*this, ServerCompleteCallback);
+  EXPECT_CALL(*this, ServerCompleteCallback(true));
   SendClientToServerEmbedded(empty, STUN_BINDING_RESPONSE);
   EXPECT_EQ(server_.state(), State::COMPLETE);
   EXPECT_EQ(client_.state(), State::PENDING);
 
   // Post-handshake ACK
-  EXPECT_CALL(*this, ClientCompleteCallback);
+  EXPECT_CALL(*this, ClientCompleteCallback(true));
   SendServerToClientEmbedded(empty, STUN_BINDING_REQUEST);
   EXPECT_EQ(client_.state(), State::COMPLETE);
 }
@@ -331,6 +331,9 @@ TEST_F(DtlsStunPiggybackControllerTest, NotSupportedByServer) {
 
   // Flight 1
   SendClientToServerEmbedded(dtls_flight1, STUN_BINDING_REQUEST);
+  // TODO: bugs.webrtc.org/367395350 - assert when calling the complete
+  // callback in this case which currently causes a sleuth of test failures.
+  // EXPECT_CALL(*this, ClientCompleteCallback());
   SendServerToClientEmbedded(empty, STUN_BINDING_RESPONSE);
   EXPECT_EQ(client_.state(), State::OFF);
 }
@@ -372,9 +375,9 @@ TEST_F(DtlsStunPiggybackControllerTest, SomeRequestsDoNotGoThrough) {
   EXPECT_EQ(client_.state(), State::PENDING);
 
   // Post-handshake ACK
-  EXPECT_CALL(*this, ServerCompleteCallback);
+  EXPECT_CALL(*this, ServerCompleteCallback(true));
   SendClientToServerEmbedded(empty, STUN_BINDING_REQUEST);
-  EXPECT_CALL(*this, ClientCompleteCallback);
+  EXPECT_CALL(*this, ClientCompleteCallback(true));
   SendServerToClientEmbedded(empty, STUN_BINDING_RESPONSE);
   EXPECT_EQ(server_.state(), State::COMPLETE);
   EXPECT_EQ(client_.state(), State::COMPLETE);
@@ -394,9 +397,9 @@ TEST_F(DtlsStunPiggybackControllerTest, LossOnPostHandshakeAck) {
   EXPECT_EQ(client_.state(), State::PENDING);
 
   // Post-handshake ACK. Client to server gets lost
-  EXPECT_CALL(*this, ClientCompleteCallback);
+  EXPECT_CALL(*this, ClientCompleteCallback(true));
   SendServerToClientEmbedded(empty, STUN_BINDING_REQUEST);
-  EXPECT_CALL(*this, ServerCompleteCallback);
+  EXPECT_CALL(*this, ServerCompleteCallback(true));
   SendClientToServerEmbedded(empty, STUN_BINDING_RESPONSE);
   EXPECT_EQ(server_.state(), State::COMPLETE);
   EXPECT_EQ(client_.state(), State::COMPLETE);
diff --git a/p2p/dtls/dtls_transport.cc b/p2p/dtls/dtls_transport.cc
index d18c6f8581..98e901c849 100644
--- a/p2p/dtls/dtls_transport.cc
+++ b/p2p/dtls/dtls_transport.cc
@@ -268,7 +268,7 @@ DtlsTransportInternalImpl::DtlsTransportInternalImpl(
                 this,
                 ReceivedIpPacket(piggybacked_dtls_packet, SocketAddress()));
           },
-          [this]() { CompleteDtlsInStun(); }) {
+          [this](bool success) { CompleteDtlsInStun(success); }) {
   RTC_DCHECK(ice_transport_);
   ConnectToIceTransport();
   dtls_in_stun_ = env_.field_trials().IsEnabled("WebRTC-IceHandshakeDtls");
@@ -288,10 +288,14 @@ DtlsTransportInternalImpl::DtlsTransportInternalImpl(
           ssl_stream_factory) {}
 
 DtlsTransportInternalImpl::~DtlsTransportInternalImpl() {
-  CompleteDtlsInStun();
+  if (dtls_in_stun_) {
+    CompleteDtlsInStun(/*success=*/false);
+  }
 }
 
-void DtlsTransportInternalImpl::CompleteDtlsInStun() {
+void DtlsTransportInternalImpl::CompleteDtlsInStun(bool success) {
+  RTC_LOG(LS_INFO) << "DTLS-STUN piggyback complete with success=" << success;
+
   dtls_in_stun_complete_ = true;
   if (downward_) {
     downward_->SetDtlsStunPiggybackController(nullptr);
@@ -1022,7 +1026,7 @@ void DtlsTransportInternalImpl::OnDtlsEvent(int sig, int err) {
         set_writable(false);
         set_dtls_state(DtlsTransportState::kClosed);
         NotifyOnClose();
-        CompleteDtlsInStun();
+        CompleteDtlsInStun(/*success=*/false);
       } else if (ret == SR_ERROR) {
         // Remote peer shut down the association with an error.
         RTC_LOG(LS_INFO)
@@ -1032,7 +1036,7 @@ void DtlsTransportInternalImpl::OnDtlsEvent(int sig, int err) {
         set_writable(false);
         set_dtls_state(DtlsTransportState::kFailed);
         NotifyOnClose();
-        CompleteDtlsInStun();
+        CompleteDtlsInStun(/*success=*/false);
       }
     } while (ret == SR_SUCCESS);
   }
diff --git a/p2p/dtls/dtls_transport.h b/p2p/dtls/dtls_transport.h
index 7d92c71971..7176ccefad 100644
--- a/p2p/dtls/dtls_transport.h
+++ b/p2p/dtls/dtls_transport.h
@@ -303,7 +303,7 @@ class DtlsTransportInternalImpl : public DtlsTransportInternal {
   // Sets the DTLS state, signaling if necessary.
   void set_dtls_state(DtlsTransportState state);
 
-  void CompleteDtlsInStun();
+  void CompleteDtlsInStun(bool success);
   void SetPiggybackDtlsDataCallback(
       absl::AnyInvocable<void(PacketTransportInternal* transport,
                               const ReceivedIpPacket& packet)> callback);

From 1b08071e7a4747ae7614aa7898d0d393bbe297a0 Mon Sep 17 00:00:00 2001
From: Victor Boivie <boivie@webrtc.org>
Date: Fri, 27 Feb 2026 14:38:26 +0100
Subject: [PATCH 158/472] dcsctp: Remove assignment of redundant metrics

The DcSctpSocket::GetMetrics() returns a Metrics struct that contains
both constant values calculated when the connection was established,
such as capabilities, and also dynamic information that change for each
incoming or outgoing packet.

The constant values are stored within "metrics_" when the TCB is created
and copied into the returned metrics during GetMetrics() call. The
dynamic values are then only set on the local copy before returning.

Before this CL, the "negotiated_maximum_incoming_streams" metric, which
is a constant metric, was additionally set - twice - in GetMetrics(),
which could make one believe that the
"negotiated_maximum_outgoing_streams" was never set at all, but it was,
and this was also verified with the
DcSctpSocketParametrizedTest::ExposesTheNumberOfNegotiatedStreams test
case.

This change removes the redundant assignment, but please note that the
returned Metrics have always been correct.

Bug: webrtc:487768780
Change-Id: If29f5c28dc5499bb9dac8b461bdc15462671b9b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452700
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47011}
---
 net/dcsctp/socket/dcsctp_socket.cc | 5 +----
 1 file changed, 1 insertion(+), 4 deletions(-)

diff --git a/net/dcsctp/socket/dcsctp_socket.cc b/net/dcsctp/socket/dcsctp_socket.cc
index 5b27f9a8bf..dd54f75a8b 100644
--- a/net/dcsctp/socket/dcsctp_socket.cc
+++ b/net/dcsctp/socket/dcsctp_socket.cc
@@ -678,6 +678,7 @@ std::optional<Metrics> DcSctpSocket::GetMetrics() const {
     return std::nullopt;
   }
 
+  // Note: `metrics_` has some pre-calculated (negotiated) values set already.
   Metrics metrics = metrics_;
   metrics.cwnd_bytes = tcb_->cwnd();
   metrics.srtt_ms = tcb_->current_srtt().ms();
@@ -688,10 +689,6 @@ std::optional<Metrics> DcSctpSocket::GetMetrics() const {
       (send_queue_.total_buffered_amount() + packet_payload_size - 1) /
           packet_payload_size;
   metrics.peer_rwnd_bytes = tcb_->retransmission_queue().rwnd();
-  metrics.negotiated_maximum_incoming_streams =
-      tcb_->capabilities().negotiated_maximum_incoming_streams;
-  metrics.negotiated_maximum_incoming_streams =
-      tcb_->capabilities().negotiated_maximum_incoming_streams;
   metrics.rtx_packets_count = tcb_->retransmission_queue().rtx_packets_count();
   metrics.rtx_bytes_count = tcb_->retransmission_queue().rtx_bytes_count();
 

From bf3812b3a37a7589a6d8a65aecdae2d59f8173bb Mon Sep 17 00:00:00 2001
From: Jonas Oreland <jonaso@webrtc.org>
Date: Fri, 27 Feb 2026 07:30:31 -0800
Subject: [PATCH 159/472] dtls-in-stun: Enable new BoringSSL update-timeout
 feature

Enable usage of
https://boringssl-review.git.corp.google.com/c/boringssl/+/86167
for DTLS handshake.

This has the following positive effects:
- The very hard ConnectTime test now works for all scenarios, yay!
- The (not automatically run) _bench_test now has zero outliers.

I.e. this is very close to something launchable.

However the following problems still remain
1) When client.dtls_in_stun != server.dtls_in_stun there are spurious
re-transmissions in unit tests. I know what the problem is and is working on a follow up.

2) There is a problem in the AlmostFull test case with dtls_in_stun,
causing retransmissions for unknown reason.

Bug: webrtc:367395350
Change-Id: Ia9cfc42e353856d072fc52c4a0076873eefa0c16
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/451001
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47012}
---
 g3doc/dtls_stun_piggyback.md          | 47 +++++++++++++++++++++
 p2p/dtls/dtls_ice_integration_test.cc | 61 +++++++++++++++------------
 p2p/dtls/dtls_transport_unittest.cc   | 32 ++++++++------
 rtc_base/openssl_stream_adapter.cc    |  6 +--
 4 files changed, 101 insertions(+), 45 deletions(-)
 create mode 100644 g3doc/dtls_stun_piggyback.md

diff --git a/g3doc/dtls_stun_piggyback.md b/g3doc/dtls_stun_piggyback.md
new file mode 100644
index 0000000000..cb9e396240
--- /dev/null
+++ b/g3doc/dtls_stun_piggyback.md
@@ -0,0 +1,47 @@
+Piggybacking DTLS on Stun is a mechanism whereby we send the initial DTLS
+handshake over the STUN exchange that opens up the data path, thereby saving round
+trips.
+
+The implementation is in dtls_stun_piggyback_controller.{h,cc} and is tested by
+dtls_ice_integration_test.cc.
+
+The state machine is as follows:
+
+Initial state:
+CLIENT: TENTATIVE
+SERVER: TENTATIVE
+
+A "flight" is a set of one or more DTLS messages (see resp. RFC).
+
+*** DTLS1.2 (RFC 6347) ***
+
+Flight 1: CLIENT => SERVER
+SERVER when receiving Flight 1: TENTATIVE => CONFIRMED
+
+Flight 2: SERVER => CLIENT
+CLIENT when receiving Flight 2: TENTATIVE => CONFIRMED
+
+Flight 3: CLIENT => SERVER
+SERVER when receiving Flight 3: CONFIRMED => PENDING (dtls writable)
+
+Flight 4: SERVER => CLIENT
+Client when receiving Flight 4: CONFIRMED => COMPLETE (dtls writable)
+
+SERVER will switch to COMPLETE when one of
+- Flight 4 is acked
+- It receives decryptable data from CLIENT (i.e. it learns that CLIENT is dtls writable)
+
+*** DTLS1.3 (RFC 9147) ***
+
+Flight 1: CLIENT => SERVER
+SERVER when receiving Flight 1: TENTATIVE => CONFIRMED
+
+Flight 2: SERVER => CLIENT
+CLIENT when receiving Flight 2: TENTATIVE => PENDING (dtls writable)
+
+Flight 3: CLIENT => SERVER
+SERVER when receiving Flight 3: CONFIRMED => COMPLETE (dtls writable)
+
+CLIENT will switch to COMPLETE when one of
+- Flight 3 is acked
+- It receives decryptable data from SERVER (i.e. it learns that SERVER is dtls writable)
diff --git a/p2p/dtls/dtls_ice_integration_test.cc b/p2p/dtls/dtls_ice_integration_test.cc
index 0ff6f24c7d..083c4add85 100644
--- a/p2p/dtls/dtls_ice_integration_test.cc
+++ b/p2p/dtls/dtls_ice_integration_test.cc
@@ -84,6 +84,27 @@ class DtlsIceIntegrationTest : public dtls_ice_integration_fixture::Base,
     return CountConnectionsWithFilter(ice,
                                       [](auto con) { return con.writable; });
   }
+
+  void CheckRetransmissions() {
+    if (client_.config.dtls_in_stun != server_.config.dtls_in_stun) {
+      // If peers does not have same setting for DTLS in STUN
+      // the current implementation does sometimes introduce a DTLS
+      // retransmit.
+      //
+      // TODO: bugs.webrtc.org/367395350 - It would be nice to remove these!
+      EXPECT_LE(client_.dtls->GetRetransmissionCount(), 1);
+      EXPECT_LE(server_.dtls->GetRetransmissionCount(), 1);
+      return;
+    }
+
+    EXPECT_EQ(client_.dtls->GetRetransmissionCount(), 0);
+    if (!server_.config.ice_lite) {
+      // TODO: bugs.webrtc.org/367395350 - We allow retransmissions
+      // from the fake ice-lite agent, but it would be great if
+      // we can remove them.
+      EXPECT_EQ(server_.dtls->GetRetransmissionCount(), 0);
+    }
+  }
 };
 
 TEST_P(DtlsIceIntegrationTest, SmokeTest) {
@@ -117,8 +138,7 @@ TEST_P(DtlsIceIntegrationTest, SmokeTest) {
     EXPECT_GE(dtls_server().dtls->GetStunDataCount(), 0);
   }
 
-  EXPECT_EQ(client_.dtls->GetRetransmissionCount(), 0);
-  EXPECT_EQ(server_.dtls->GetRetransmissionCount(), 0);
+  CheckRetransmissions();
 }
 
 TEST_P(DtlsIceIntegrationTest, AddCandidates) {
@@ -195,8 +215,7 @@ TEST_P(DtlsIceIntegrationTest, ClientLateCertificate) {
               client_.config.dtls_in_stun && server_.config.dtls_in_stun);
   });
 
-  EXPECT_EQ(client_.dtls->GetRetransmissionCount(), 0);
-  EXPECT_EQ(server_.dtls->GetRetransmissionCount(), 0);
+  CheckRetransmissions();
 }
 
 TEST_P(DtlsIceIntegrationTest, TestWithPacketLoss) {
@@ -204,12 +223,6 @@ TEST_P(DtlsIceIntegrationTest, TestWithPacketLoss) {
     GTEST_SKIP() << "Needs boringssl.";
   }
 
-  if (client_.config.dtls_in_stun != server_.config.dtls_in_stun) {
-    // TODO jonaso, webrtc:404763475 : re-enable once
-    // boringssl has been merged and test cases updated.
-    GTEST_SKIP() << "TODO jonaso.";
-  }
-
   ConfigureEmulatedNetwork();
   Prepare();
 
@@ -242,12 +255,6 @@ TEST_P(DtlsIceIntegrationTest, LongRunningTestWithPacketLoss) {
     GTEST_SKIP() << "Needs boringssl.";
   }
 
-  if (client_.config.dtls_in_stun != server_.config.dtls_in_stun) {
-    // TODO jonaso, webrtc:404763475 : re-enable once
-    // boringssl has been merged and test cases updated.
-    GTEST_SKIP() << "TODO jonaso.";
-  }
-
   int seed = absl::GetFlag(FLAGS_long_running_seed);
   if (seed == 0) {
     seed = 1 + time(0);
@@ -381,8 +388,16 @@ TEST_P(DtlsIceIntegrationTest, AlmostFullSTUN_BINDING) {
               client_.config.dtls_in_stun && server_.config.dtls_in_stun);
   });
 
-  EXPECT_EQ(client_.dtls->GetRetransmissionCount(), 0);
-  EXPECT_EQ(server_.dtls->GetRetransmissionCount(), 0);
+  if (client_.config.dtls_in_stun && server_.config.dtls_in_stun &&
+      (client_.config.pqc || server_.config.pqc)) {
+    // TODO: bugs.webrtc.org/367395350 - Investigate why there is
+    // retransmissions in the scenario where the STUN packet is almost full,
+    // e.g. will the issue be solved by our effort to smooth BoringSSL packets ?
+    EXPECT_LE(client_.dtls->GetRetransmissionCount(), 1);
+    EXPECT_LE(server_.dtls->GetRetransmissionCount(), 1);
+  } else {
+    CheckRetransmissions();
+  }
 }
 
 INSTANTIATE_TEST_SUITE_P(DtlsStunPiggybackingIntegrationTest,
@@ -451,16 +466,6 @@ TEST_P(DtlsIceIntegrationPerformanceTest, ConnectTime) {
     GTEST_SKIP() << "Needs boringssl.";
   }
 
-  {
-    TestConfig config = GetParam();
-    if (config.client_config.pqc == 1 && config.server_config.pqc &&
-        config.server_config.ice_lite) {
-      // TODO jonaso, webrtc:404763475 : re-enable once
-      // boringssl has been merged and test cases updated.
-      GTEST_SKIP() << "TODO jonaso.";
-    }
-  }
-
   int iter = 50;
   // Use a fixed seed to get consistent behavior.
   Random rand(/* seed= */ 77);
diff --git a/p2p/dtls/dtls_transport_unittest.cc b/p2p/dtls/dtls_transport_unittest.cc
index b50ea7a42e..a987d71330 100644
--- a/p2p/dtls/dtls_transport_unittest.cc
+++ b/p2p/dtls/dtls_transport_unittest.cc
@@ -1819,6 +1819,22 @@ class DtlsTransportInternalImplDtlsInStunTest
     : public DtlsTransportInternalImplVersionTest {
  public:
   DtlsTransportInternalImplDtlsInStunTest() {}
+
+  void CheckRetransmissions() {
+    if (std::get<0>(GetParam()).dtls_in_stun ==
+        std::get<1>(GetParam()).dtls_in_stun) {
+      EXPECT_EQ(client1_.dtls_transport()->GetRetransmissionCount(), 0);
+      EXPECT_EQ(client2_.dtls_transport()->GetRetransmissionCount(), 0);
+    } else {
+      // If peers does not have same setting for DTLS in STUN
+      // current implementation does sometimes introduce a DTLS
+      // retransmit.
+      //
+      // TODO: bugs.webrtc.org/367395350 - It would be nice to remove these!
+      EXPECT_LE(client1_.dtls_transport()->GetRetransmissionCount(), 1);
+      EXPECT_LE(client2_.dtls_transport()->GetRetransmissionCount(), 1);
+    }
+  }
 };
 
 std::vector<std::tuple<EndpointConfig, EndpointConfig>> AllEndpointVariants() {
@@ -1904,9 +1920,7 @@ TEST_P(DtlsTransportInternalImplDtlsInStunTest, Handshake1) {
   EXPECT_TRUE(client1_.dtls_transport()->writable());
   EXPECT_TRUE(client2_.dtls_transport()->writable());
 
-  EXPECT_EQ(client1_.dtls_transport()->GetRetransmissionCount(), 0);
-  EXPECT_EQ(client2_.dtls_transport()->GetRetransmissionCount(), 0);
-
+  CheckRetransmissions();
   ClearPacketFilters();
 }
 
@@ -1955,9 +1969,7 @@ TEST_P(DtlsTransportInternalImplDtlsInStunTest, Handshake2) {
   EXPECT_TRUE(client1_.dtls_transport()->writable());
   EXPECT_TRUE(client2_.dtls_transport()->writable());
 
-  EXPECT_EQ(client1_.dtls_transport()->GetRetransmissionCount(), 0);
-  EXPECT_EQ(client2_.dtls_transport()->GetRetransmissionCount(), 0);
-
+  CheckRetransmissions();
   ClearPacketFilters();
 }
 
@@ -2012,9 +2024,7 @@ TEST_P(DtlsTransportInternalImplDtlsInStunTest, PartiallyPiggybacked) {
   EXPECT_TRUE(client1_.dtls_transport()->writable());
   EXPECT_TRUE(client2_.dtls_transport()->writable());
 
-  EXPECT_EQ(client1_.dtls_transport()->GetRetransmissionCount(), 0);
-  EXPECT_EQ(client2_.dtls_transport()->GetRetransmissionCount(), 0);
-
+  CheckRetransmissions();
   ClearPacketFilters();
 }
 
@@ -2214,9 +2224,7 @@ TEST_P(DtlsInStunTest, OptimalDtls13Handshake) {
   }));
   EXPECT_TRUE(client2_.dtls_transport()->writable());
 
-  EXPECT_EQ(client1_.dtls_transport()->GetRetransmissionCount(), 0);
-  EXPECT_EQ(client2_.dtls_transport()->GetRetransmissionCount(), 0);
-
+  CheckRetransmissions();
   ClearPacketFilters();
 }
 
diff --git a/rtc_base/openssl_stream_adapter.cc b/rtc_base/openssl_stream_adapter.cc
index 36bda6e0a6..0a410f8e80 100644
--- a/rtc_base/openssl_stream_adapter.cc
+++ b/rtc_base/openssl_stream_adapter.cc
@@ -646,11 +646,7 @@ void OpenSSLStreamAdapter::UpdateRetransmissionTimeout(int timeout_ms) {
   dtls_handshake_timeout_ms_ = timeout_ms;
 #ifdef OPENSSL_IS_BORINGSSL
   if (ssl_ctx_ != nullptr && ssl_mode_ == SSL_MODE_DTLS) {
-    // TODO jonaso, webrtc:404763475 : Now that this is actually
-    // implemented in BORING_SSL we need to update test cases that
-    // will otherwise start to fail.
-    // DTLSv1_set_initial_timeout_duration(ssl_, dtls_handshake_timeout_ms_);
-
+    DTLSv1_set_initial_timeout_duration(ssl_, dtls_handshake_timeout_ms_);
     // Clear the DTLS timer
     timeout_task_.Stop();
     MaybeSetTimeout();

From 1d33367dd0cc566386fce69d2b9b5d84e8aeea0e Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Fri, 27 Feb 2026 12:07:20 -0800
Subject: [PATCH 160/472] Roll chromium_revision 972b43f675..500ba22722
 (1591474:1591721)

Change log: https://chromium.googlesource.com/chromium/src/+log/972b43f675..500ba22722
Full diff: https://chromium.googlesource.com/chromium/src/+/972b43f675..500ba22722

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/371581ad06..8cfed3408e
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/0b4e87c548..0ef252d508
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/4f57914322..62e165dd5e
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/ac4bea0343..158f73d26e
* src/third_party/androidx/cipd: Di9v_vOtvM11yYkWUaYwjOdOqjM73OgEkVvRg0vXJXMC..yjDOkdvC492M3KA9EiBGFBmpTkSe0XvVXtHqQqB1gZYC
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/78c96746a3..53d9ac8086
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/bb0a61b409..ca4508e8f2
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/58325a01ba..eb23e9a93e
DEPS diff: https://chromium.googlesource.com/chromium/src/+/972b43f675..500ba22722/DEPS

BUG=None

Change-Id: Id405ea76bc7a1ce6c122be2a653e21725241a460
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452721
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47013}
---
 DEPS | 18 +++++++++---------
 1 file changed, 9 insertions(+), 9 deletions(-)

diff --git a/DEPS b/DEPS
index 95033a2c07..fd3815fe3f 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '972b43f675102cd8b620038c64d00789c20924a5',
+  'chromium_revision': '500ba2272263e5185769d20f3b0ac5fa85b169df',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@371581ad0695c16f70aa1d7c060f5f51a188abb5',
+    'https://chromium.googlesource.com/chromium/src/build@8cfed3408ec11d1780abcd66e9e0d8ad8b21ce75',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@136da69a1267b8db487354b96d44d0cc8add5aeb',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@0b4e87c5486f59384da7bcc11fd2a1d230b6f0d1',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@0ef252d508527fe07de83ca32e42512d6c8f6e99',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@4f57914322cd5778334524da56b5fed52602b471',
+    'https://chromium.googlesource.com/chromium/src/testing@62e165dd5e5e8d69ab43fd05b860930487abec78',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@ac4bea03435519c8e20b21f9a20cd77af66bfbd3',
+    'https://chromium.googlesource.com/chromium/src/third_party@158f73d26e1d3eedf5840546138ac5ea3c2acc1b',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -511,7 +511,7 @@ deps = {
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@78c96746a3a149db4cf71de9a3882800fac04248',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@53d9ac808651af1dc26efaffc25f724f036214ff',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'bb0a61b40921991a23ab2aec62e9528d9904f52e',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'ca4508e8f261cdaf3d5b48c9381a504b99b2669a',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@58325a01ba56d27ad4c04a58148c40687af92647',
+    'https://chromium.googlesource.com/chromium/src/tools@eb23e9a93e7cd293c8a7b6b8ef790c37acb771a0',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'Di9v_vOtvM11yYkWUaYwjOdOqjM73OgEkVvRg0vXJXMC',
+          'version': 'yjDOkdvC492M3KA9EiBGFBmpTkSe0XvVXtHqQqB1gZYC',
       },
     ],
     'condition': 'checkout_android and non_git_source',

From f209e0176785b8807e20c82a075f6c021c4d5d01 Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Fri, 27 Feb 2026 12:35:07 +0000
Subject: [PATCH 161/472] Remove special treatment for the AbsSendTime header

This seems to have only been intended for use with external
authentication, and that code seems to have been unused for a long time.

Its value is still set later than the value of other headers.

Bug: webrtc:487743907
Change-Id: Id400f147b9aeef9e9b324d9bdb0117632a580346
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452681
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47014}
---
 pc/jsep_transport.cc            |  6 ------
 pc/jsep_transport.h             |  2 --
 pc/jsep_transport_controller.cc | 23 ++---------------------
 pc/jsep_transport_controller.h  |  5 +----
 pc/srtp_transport.h             |  8 --------
 5 files changed, 3 insertions(+), 41 deletions(-)

diff --git a/pc/jsep_transport.cc b/pc/jsep_transport.cc
index c1972c6bf2..82471c7181 100644
--- a/pc/jsep_transport.cc
+++ b/pc/jsep_transport.cc
@@ -57,18 +57,15 @@ JsepTransportDescription::JsepTransportDescription() {}
 JsepTransportDescription::JsepTransportDescription(
     bool rtcp_mux_enabled,
     const std::vector<int>& encrypted_header_extension_ids,
-    int rtp_abs_sendtime_extn_id,
     const TransportDescription& transport_desc)
     : rtcp_mux_enabled(rtcp_mux_enabled),
       encrypted_header_extension_ids(encrypted_header_extension_ids),
-      rtp_abs_sendtime_extn_id(rtp_abs_sendtime_extn_id),
       transport_desc(transport_desc) {}
 
 JsepTransportDescription::JsepTransportDescription(
     const JsepTransportDescription& from)
     : rtcp_mux_enabled(from.rtcp_mux_enabled),
       encrypted_header_extension_ids(from.encrypted_header_extension_ids),
-      rtp_abs_sendtime_extn_id(from.rtp_abs_sendtime_extn_id),
       transport_desc(from.transport_desc) {}
 
 JsepTransportDescription::~JsepTransportDescription() = default;
@@ -80,7 +77,6 @@ JsepTransportDescription& JsepTransportDescription::operator=(
   }
   rtcp_mux_enabled = from.rtcp_mux_enabled;
   encrypted_header_extension_ids = from.encrypted_header_extension_ids;
-  rtp_abs_sendtime_extn_id = from.rtp_abs_sendtime_extn_id;
   transport_desc = from.transport_desc;
 
   return *this;
@@ -222,8 +218,6 @@ RTCError JsepTransport::SetRemoteJsepTransportDescription(
   if (auto* dtls_srtp_transport = rtp_transport_->AsDtlsSrtpTransport()) {
     dtls_srtp_transport->UpdateSendEncryptedHeaderExtensionIds(
         jsep_description.encrypted_header_extension_ids);
-    dtls_srtp_transport->CacheRtpAbsSendTimeHeaderExtension(
-        jsep_description.rtp_abs_sendtime_extn_id);
   }
 
   remote_description_.reset(new JsepTransportDescription(jsep_description));
diff --git a/pc/jsep_transport.h b/pc/jsep_transport.h
index 6475a27177..9ec919b9b7 100644
--- a/pc/jsep_transport.h
+++ b/pc/jsep_transport.h
@@ -48,7 +48,6 @@ struct JsepTransportDescription {
   JsepTransportDescription(
       bool rtcp_mux_enabled,
       const std::vector<int>& encrypted_header_extension_ids,
-      int rtp_abs_sendtime_extn_id,
       const TransportDescription& transport_description);
   JsepTransportDescription(const JsepTransportDescription& from);
   ~JsepTransportDescription();
@@ -57,7 +56,6 @@ struct JsepTransportDescription {
 
   bool rtcp_mux_enabled = true;
   std::vector<int> encrypted_header_extension_ids;
-  int rtp_abs_sendtime_extn_id = -1;
   // TODO(zhihuang): Add the ICE and DTLS related variables and methods from
   // TransportDescription and remove this extra layer of abstraction.
   TransportDescription transport_desc;
diff --git a/pc/jsep_transport_controller.cc b/pc/jsep_transport_controller.cc
index d90cab3f4d..2dd7f4f857 100644
--- a/pc/jsep_transport_controller.cc
+++ b/pc/jsep_transport_controller.cc
@@ -731,13 +731,10 @@ RTCError JsepTransportController::ApplyDescription_n(
       extension_ids = GetEncryptedHeaderExtensionIds(content_info);
     }
 
-    int rtp_abs_sendtime_extn_id =
-        GetRtpAbsSendTimeHeaderExtensionId(content_info);
-
     SetIceRole_n(DetermineIceRole(transport, transport_info, type, local));
 
     JsepTransportDescription jsep_description = CreateJsepTransportDescription(
-        content_info, transport_info, extension_ids, rtp_abs_sendtime_extn_id);
+        content_info, transport_info, extension_ids);
     if (local) {
       error =
           transport->SetLocalJsepTransportDescription(jsep_description, type);
@@ -1003,8 +1000,7 @@ JsepTransportDescription
 JsepTransportController::CreateJsepTransportDescription(
     const ContentInfo& content_info,
     const TransportInfo& transport_info,
-    const std::vector<int>& encrypted_extension_ids,
-    int rtp_abs_sendtime_extn_id) {
+    const std::vector<int>& encrypted_extension_ids) {
   TRACE_EVENT0("webrtc",
                "JsepTransportController::CreateJsepTransportDescription");
   const MediaContentDescription* content_desc =
@@ -1015,7 +1011,6 @@ JsepTransportController::CreateJsepTransportDescription(
                               : content_desc->rtcp_mux();
 
   return JsepTransportDescription(rtcp_mux_enabled, encrypted_extension_ids,
-                                  rtp_abs_sendtime_extn_id,
                                   transport_info.description);
 }
 
@@ -1067,20 +1062,6 @@ JsepTransportController::MergeEncryptedHeaderExtensionIdsForBundles(
   return merged_encrypted_extension_ids_by_bundle;
 }
 
-int JsepTransportController::GetRtpAbsSendTimeHeaderExtensionId(
-    const ContentInfo& content_info) {
-  const MediaContentDescription* content_desc =
-      content_info.media_description();
-
-  const RtpExtension* send_time_extension =
-      RtpExtension::FindHeaderExtensionByUri(
-          content_desc->rtp_header_extensions(), RtpExtension::kAbsSendTimeUri,
-          config_.crypto_options.srtp.enable_encrypted_rtp_header_extensions
-              ? RtpExtension::kPreferEncryptedExtension
-              : RtpExtension::kDiscardEncryptedExtension);
-  return send_time_extension ? send_time_extension->id : -1;
-}
-
 const JsepTransport* JsepTransportController::GetJsepTransportForMid(
     absl::string_view mid) const {
   return transports_.GetTransportForMid(mid);
diff --git a/pc/jsep_transport_controller.h b/pc/jsep_transport_controller.h
index ff86840532..e2c95b3561 100644
--- a/pc/jsep_transport_controller.h
+++ b/pc/jsep_transport_controller.h
@@ -316,8 +316,7 @@ class JsepTransportController final {
   JsepTransportDescription CreateJsepTransportDescription(
       const ContentInfo& content_info,
       const TransportInfo& transport_info,
-      const std::vector<int>& encrypted_extension_ids,
-      int rtp_abs_sendtime_extn_id);
+      const std::vector<int>& encrypted_extension_ids);
 
   std::map<const ContentGroup*, std::vector<int>>
   MergeEncryptedHeaderExtensionIdsForBundles(
@@ -325,8 +324,6 @@ class JsepTransportController final {
   std::vector<int> GetEncryptedHeaderExtensionIds(
       const ContentInfo& content_info);
 
-  int GetRtpAbsSendTimeHeaderExtensionId(const ContentInfo& content_info);
-
   // This method takes the BUNDLE group into account. If the JsepTransport is
   // destroyed because of BUNDLE, it would return the transport which other
   // transports are bundled on (In current implementation, it is the first
diff --git a/pc/srtp_transport.h b/pc/srtp_transport.h
index 4f8e23a2c5..b157220dc7 100644
--- a/pc/srtp_transport.h
+++ b/pc/srtp_transport.h
@@ -80,12 +80,6 @@ class SrtpTransport : public RtpTransport {
   // Returns srtp overhead for rtp packets.
   bool GetSrtpOverhead(int* srtp_overhead) const;
 
-  // Cache RTP Absoulute SendTime extension header ID. This is only used when
-  // external authentication is enabled.
-  void CacheRtpAbsSendTimeHeaderExtension(int rtp_abs_sendtime_extn_id) {
-    rtp_abs_sendtime_extn_id_ = rtp_abs_sendtime_extn_id;
-  }
-
   // In addition to unregistering the sink, the SRTP transport
   // disassociates all SSRCs of the sink from libSRTP.
   bool UnregisterRtpDemuxerSink(RtpPacketSinkInterface* sink) override;
@@ -130,8 +124,6 @@ class SrtpTransport : public RtpTransport {
 
   bool writable_ = false;
 
-  int rtp_abs_sendtime_extn_id_ = -1;
-
   int decryption_failure_count_ = 0;
 
   const FieldTrialsView& field_trials_;

From 6f90f4ff7a7fb27e8e6785bf40c5ca46b6254162 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Fri, 27 Feb 2026 16:03:51 -0800
Subject: [PATCH 162/472] Roll chromium_revision 500ba22722..a0f393d920
 (1591721:1591883)

Change log: https://chromium.googlesource.com/chromium/src/+log/500ba22722..a0f393d920
Full diff: https://chromium.googlesource.com/chromium/src/+/500ba22722..a0f393d920

Changed dependencies
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/0ef252d508..b75b9c3af2
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/62e165dd5e..b39365f684
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/158f73d26e..786f7b88d6
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/e507931d68..87c9af3b7c
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/3969ba2efb..55659c2bf4
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/53d9ac8086..4b296e3278
* src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/c365236fa3..bfe85bb036
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/ca4508e8f2..dcb5e98f9f
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/eb23e9a93e..61a4a20ac4
DEPS diff: https://chromium.googlesource.com/chromium/src/+/500ba22722..a0f393d920/DEPS

BUG=None

Change-Id: Ie568f4b786e5dcb9732ab66b490b78ee866f097b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452741
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47015}
---
 DEPS | 20 ++++++++++----------
 1 file changed, 10 insertions(+), 10 deletions(-)

diff --git a/DEPS b/DEPS
index fd3815fe3f..b8d5cb4065 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '500ba2272263e5185769d20f3b0ac5fa85b169df',
+  'chromium_revision': 'a0f393d920ee0ffde8f21326fd8e77fbfdd17956',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@0ef252d508527fe07de83ca32e42512d6c8f6e99',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@b75b9c3af27174d15a0d9f46370a17d696e941cf',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@62e165dd5e5e8d69ab43fd05b860930487abec78',
+    'https://chromium.googlesource.com/chromium/src/testing@b39365f68470bf9cca757d4980d1aad23c133b02',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@158f73d26e1d3eedf5840546138ac5ea3c2acc1b',
+    'https://chromium.googlesource.com/chromium/src/third_party@786f7b88d63c0c3d72f833bc88c6263159874d32',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -337,7 +337,7 @@ deps = {
   'src/third_party/libc++abi/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@8f11bb1d4438d0239d0dfc1bd9456a9f31629dda',
   'src/third_party/llvm-libc/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@c365236fa3dbdaff431a582f4eb04f6772d1e3da',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@bfe85bb0366b861995d6bad528c1426ce72a0518',
   'src/third_party/libunwind/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@17ccf7d110c5526cb77e93cfd8330f491fb2bf18',
 
@@ -492,7 +492,7 @@ deps = {
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@9f26d631e4063c0af3e8ee3bc8826486115372a8',
   'src/third_party/catapult':
-    'https://chromium.googlesource.com/catapult.git@e507931d6883f0216da761d51e165a0f775ec1a8',
+    'https://chromium.googlesource.com/catapult.git@87c9af3b7c6495ab0a0ba86a3eb6a58ff84eca85',
   'src/third_party/ced/src': {
     'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5',
   },
@@ -505,13 +505,13 @@ deps = {
   'src/third_party/crc32c/src':
     'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558',
   'src/third_party/depot_tools':
-    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@3969ba2efbbc9894f0b4140df211b70f073a5b35',
+    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@55659c2bf434a4fb508799fa93550822c5309ed5',
   'src/third_party/ffmpeg':
     'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@946d97db8d906277085e361892b7efda5152e2f1',
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@53d9ac808651af1dc26efaffc25f724f036214ff',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@4b296e3278b656c8baf20041bc8f7e1f410f67b9',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'ca4508e8f261cdaf3d5b48c9381a504b99b2669a',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'dcb5e98f9f6e9ba61a95b0a3fef56671b0b6ba83',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@eb23e9a93e7cd293c8a7b6b8ef790c37acb771a0',
+    'https://chromium.googlesource.com/chromium/src/tools@61a4a20ac4df97f75b46a20bb5d393ee65526dde',
 
   'src/third_party/espresso': {
       'packages': [

From ae6b59c4b385132e5f819176aec432ce3e768c8a Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Fri, 27 Feb 2026 20:02:45 -0800
Subject: [PATCH 163/472] Update WebRTC code version (2026-02-28T04:02:22).

Bug: None
Change-Id: I774acfbc439854e18c76466be42ae69a894647d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452780
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47016}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index b402f3879c..8d5d9c74f8 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-02-27T04:08:14";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-02-28T04:02:22";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From 8d0a4663fa35d79fc5a923d4f45d53856bdaa90b Mon Sep 17 00:00:00 2001
From: Jonas Oreland <jonaso@webrtc.org>
Date: Fri, 27 Feb 2026 12:28:41 -0800
Subject: [PATCH 164/472] dtls-in-stun: Simplify PENDING => COMPLETE condition

We are COMPLETE when we are writable(PENDING) and we have no packets
that has not been acked by peer (i.e. pending packet is empty).

This change passes all existing test and re-enables 32 tests, yay!
I.e. makes fewer tests skipped by adding conditions to the skip-if.

Also, remove incorrect observations that we can clear ack
if getting no data, add unit test for this which explain
why this is not true which is
a) the peer might not have gotten all it needs to send any
data (pqc).
b) there might be arbitrary reorders, and we con't now it
the stun-binding (request/response) we are processing is
really old.

BUG=webrtc:367395350

Change-Id: Ic2999587c08f970ceb75efd4575ade4adf65eee8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/451741
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47017}
---
 p2p/dtls/dtls_stun_piggyback_controller.cc    | 76 +++++++------------
 ...dtls_stun_piggyback_controller_unittest.cc | 20 +++++
 2 files changed, 49 insertions(+), 47 deletions(-)

diff --git a/p2p/dtls/dtls_stun_piggyback_controller.cc b/p2p/dtls/dtls_stun_piggyback_controller.cc
index 42e4be6ed7..f0b84c62a8 100644
--- a/p2p/dtls/dtls_stun_piggyback_controller.cc
+++ b/p2p/dtls/dtls_stun_piggyback_controller.cc
@@ -176,29 +176,20 @@ void DtlsStunPiggybackController::ReportDataPiggybacked(
     return;
   }
 
-  // We sent dtls piggybacked but got nothing in return or
-  // we received a stun request with neither attribute set
-  // => peer does not support.
-  if (state_ == State::TENTATIVE && !data.has_value() && !acks.has_value()) {
-    RTC_LOG(LS_INFO) << "DTLS-STUN piggybacking not supported by peer.";
-    state_ = State::OFF;
-    // TODO: bugs.webrtc.org/367395350 - We should call CallCompleteCallback
-    // here but this causes a slew of failed tests. Investigate why!
-    // CallCompleteCallback(/*success=*/false);
-    return;
-  }
-
-  // In PENDING state the peer may have stopped sending the ack
-  // when it moved to the COMPLETE state. Move to the same state.
-  if (state_ == State::PENDING && !data.has_value() && !acks.has_value()) {
-    RTC_LOG(LS_INFO) << "DTLS-STUN piggybacking complete.";
-    state_ = State::COMPLETE;
-    CallCompleteCallback(/*success=*/true);
-    return;
-  }
-
-  // We sent dtls piggybacked and got something in return => peer does support.
   if (state_ == State::TENTATIVE) {
+    if (!data.has_value() && !acks.has_value()) {
+      // We sent dtls piggybacked but got nothing in return or
+      // we received a stun request with neither attribute set
+      // => peer does not support.
+      RTC_LOG(LS_INFO) << "DTLS-STUN piggybacking not supported by peer.";
+      state_ = State::OFF;
+      // TODO: bugs.webrtc.org/367395350 - We should call CallCompleteCallback
+      // here but this causes a slew of failed tests. Investigate why!
+      // CallCompleteCallback(/*success=*/false);
+      return;
+    }
+    // We sent dtls piggybacked and got something in return => peer does
+    // support.
     state_ = State::CONFIRMED;
   }
 
@@ -217,38 +208,29 @@ void DtlsStunPiggybackController::ReportDataPiggybacked(
     }
   }
 
-  if (!data.has_value()) {
-    // If we receive msg w/o any data, that means that the peer
-    // is not retransmitting, so we don't need to ACK anything.
-    handshake_messages_received_.clear();
+  if (data.has_value() && !data->empty()) {
+    // Drop non-DTLS packets.
+    if (!IsDtlsPacket(*data)) {
+      RTC_LOG(LS_WARNING) << "Dropping non-DTLS data.";
+      return;
+    }
+    ++data_recv_count_;
+    ReportDtlsPacket(*data);
+
+    // Forwards the data to the DTLS layer. Note that this will call
+    // ProcessDtlsPacket() again which does not change the state.
+    dtls_data_callback_(*data);
   }
 
-  // The response to the final flight of the handshake will not contain
-  // the DTLS data but will contain an ack.
-  // Must not happen on the initial server to client packet which
-  // has no DTLS data yet.
-  if (!data.has_value() && acks.has_value() && state_ == State::PENDING) {
+  if (state_ == State::PENDING && pending_packets_.empty()) {
+    // We are writeable(PENDING) and have no pending packets, i.e.
+    // peer has acked everything we sent, this means that we
+    // are complete.
     RTC_LOG(LS_INFO) << "DTLS-STUN piggybacking complete.";
     state_ = State::COMPLETE;
     CallCompleteCallback(/*success=*/true);
     return;
   }
-
-  if (!data.has_value() || data->empty()) {
-    return;
-  }
-  // Drop non-DTLS packets.
-  if (!IsDtlsPacket(*data)) {
-    RTC_LOG(LS_WARNING) << "Dropping non-DTLS data.";
-    return;
-  }
-  data_recv_count_++;
-
-  ReportDtlsPacket(*data);
-
-  // Forwards the data to the DTLS layer. Note that this will call
-  // ProcessDtlsPacket() again which does not change the state.
-  dtls_data_callback_(*data);
 }
 
 void DtlsStunPiggybackController::ReportDtlsPacket(
diff --git a/p2p/dtls/dtls_stun_piggyback_controller_unittest.cc b/p2p/dtls/dtls_stun_piggyback_controller_unittest.cc
index 66f5d3a354..05887a8bfe 100644
--- a/p2p/dtls/dtls_stun_piggyback_controller_unittest.cc
+++ b/p2p/dtls/dtls_stun_piggyback_controller_unittest.cc
@@ -587,6 +587,26 @@ TEST_F(DtlsStunPiggybackControllerTest, LimitAckSize) {
               }));
 }
 
+TEST_F(DtlsStunPiggybackControllerTest, EmptyDataDoesNotClearAck) {
+  std::vector<uint8_t> dtls_flight5 = FakeDtlsPacket(0x5487);
+
+  server_.ReportDataPiggybacked(
+      WrapInStun(STUN_ATTR_META_DTLS_IN_STUN, dtls_flight1)->array_view(),
+      std::nullopt);
+  EXPECT_EQ(server_.GetAckToPiggyback(STUN_BINDING_REQUEST)->size(), 1u);
+
+  // The fact that we don't get any data does not mean that
+  // we can clear the ack list.
+  // a) packets can be arbitrary reordered.
+  // b) the peer might be needing 2 packets (ie. pqc) to produce
+  // a return packet and only one of them has arrived.
+  server_.ReportDataPiggybacked(
+      std::nullopt,
+      std::vector<uint32_t>({ComputeDtlsPacketHash(dtls_flight1)}));
+
+  EXPECT_EQ(server_.GetAckToPiggyback(STUN_BINDING_REQUEST)->size(), 1u);
+}
+
 TEST_F(DtlsStunPiggybackControllerTest, MultiPacketRoundRobin) {
   // Let's pretend that a flight is 3 packets...
   server_.CapturePacket(dtls_flight1);

From ac3d3319930edccfaa13a7e58d9a183ac4934689 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Sat, 28 Feb 2026 02:03:30 -0800
Subject: [PATCH 165/472] Roll chromium_revision a0f393d920..d5da27240e
 (1591883:1591990)

Change log: https://chromium.googlesource.com/chromium/src/+log/a0f393d920..d5da27240e
Full diff: https://chromium.googlesource.com/chromium/src/+/a0f393d920..d5da27240e

Changed dependencies
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/b75b9c3af2..06e0cff019
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/b39365f684..0a8ae42116
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/786f7b88d6..a93b072843
* src/third_party/android_deps/autorolled/cipd: cA8nRMj8uhS3aXj_1a5gxG7mzG-iKRo3ndtBCv5qszEC..9DXI8FnmXkL2tti6GWrRX5mKZzOJzOVKFbiQLuE4ThYC
* src/third_party/androidx/cipd: yjDOkdvC492M3KA9EiBGFBmpTkSe0XvVXtHqQqB1gZYC.._wBJ2hBwWyOSj4dkL0Ttoxv3fb4wX0ZrTBMEDITnWXwC
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/87c9af3b7c..0f9739da7f
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/f64d1e0acc..25dba492e3
* src/third_party/fuzztest/src: https://chromium.googlesource.com/external/github.com/google/fuzztest.git/+log/362a279f0a..f67a6fe9ca
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/dcb5e98f9f..fa49f16ad9
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/61a4a20ac4..58559285e3
DEPS diff: https://chromium.googlesource.com/chromium/src/+/a0f393d920..d5da27240e/DEPS

BUG=None

Change-Id: I56a51a40a4f09f54cba2ca372c0f6e37b9be8816
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452840
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47018}
---
 DEPS | 22 +++++++++++-----------
 1 file changed, 11 insertions(+), 11 deletions(-)

diff --git a/DEPS b/DEPS
index b8d5cb4065..379672f915 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': 'a0f393d920ee0ffde8f21326fd8e77fbfdd17956',
+  'chromium_revision': 'd5da27240ef4e2938af4795cb9fa9d351bdf1f5b',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@b75b9c3af27174d15a0d9f46370a17d696e941cf',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@06e0cff019f4c65b5aec03bf9dce1c9bc6002575',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@b39365f68470bf9cca757d4980d1aad23c133b02',
+    'https://chromium.googlesource.com/chromium/src/testing@0a8ae42116bf528b9fd7e71529a01dfef04613ed',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@786f7b88d63c0c3d72f833bc88c6263159874d32',
+    'https://chromium.googlesource.com/chromium/src/third_party@a93b072843ac210f36fdd17de27f441e026f0ce7',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -492,7 +492,7 @@ deps = {
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@9f26d631e4063c0af3e8ee3bc8826486115372a8',
   'src/third_party/catapult':
-    'https://chromium.googlesource.com/catapult.git@87c9af3b7c6495ab0a0ba86a3eb6a58ff84eca85',
+    'https://chromium.googlesource.com/catapult.git@0f9739da7f76bbb5bc9c108986880d5cfb9b9cf6',
   'src/third_party/ced/src': {
     'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5',
   },
@@ -593,7 +593,7 @@ deps = {
   'src/third_party/libFuzzer/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt/lib/fuzzer.git@bea408a6e01f0f7e6c82a43121fe3af4506c932e',
   'src/third_party/fuzztest/src':
-    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@362a279f0ad018574278e72c4e98e2b99d3991bb',
+    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@f67a6fe9ca3afc4d71997e75e8d87c571f0c30cb',
   'src/third_party/libjpeg_turbo':
     'https://chromium.googlesource.com/chromium/deps/libjpeg_turbo.git@d1f5f2393e0d51f840207342ae86e55a86443288',
   'src/third_party/libsrtp':
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'dcb5e98f9f6e9ba61a95b0a3fef56671b0b6ba83',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'fa49f16ad958de750da23698f12aebb9f1ae662f',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@61a4a20ac4df97f75b46a20bb5d393ee65526dde',
+    'https://chromium.googlesource.com/chromium/src/tools@58559285e323d5e5d6645861d788cda8d3030ef9',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'yjDOkdvC492M3KA9EiBGFBmpTkSe0XvVXtHqQqB1gZYC',
+          'version': '_wBJ2hBwWyOSj4dkL0Ttoxv3fb4wX0ZrTBMEDITnWXwC',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -842,7 +842,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/android_deps/autorolled',
-              'version': 'cA8nRMj8uhS3aXj_1a5gxG7mzG-iKRo3ndtBCv5qszEC',
+              'version': '9DXI8FnmXkL2tti6GWrRX5mKZzOJzOVKFbiQLuE4ThYC',
           },
       ],
       'condition': 'checkout_android and non_git_source',
@@ -864,7 +864,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7364b490b5f78d58efe23ea76e74210fd6c3c76f',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + 'f64d1e0acc2c2c33e325b8dd7b2b4673de2b9f14',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '25dba492e30e19dce3d4bd4cd38af2f5c88c387c',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From 2b2c9a50fef5263a79359c91da3b1ba2677f5e7e Mon Sep 17 00:00:00 2001
From: Jonas Oreland <jonaso@webrtc.org>
Date: Fri, 27 Feb 2026 12:40:47 -0800
Subject: [PATCH 166/472] dtls-in-stun : Fix spurious DTLS retransmit

For the scenario where the SSL_CLIENT send an STUN_BINDING
and discovers that peer does not support dtls-in-stun.

Bug: webrtc:367395350
Change-Id: I0568ab3064576636d67efeda5e71d48c65147fe3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452220
Reviewed-by: Sameer Vijaykar <samvi@google.com>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Cr-Commit-Position: refs/heads/main@{#47019}
---
 p2p/dtls/dtls_ice_integration_test.cc | 34 ++++++++++++++++-----------
 p2p/dtls/dtls_transport.cc            | 20 +++++++++++++++-
 p2p/dtls/dtls_transport_unittest.cc   | 22 ++++++++++-------
 3 files changed, 52 insertions(+), 24 deletions(-)

diff --git a/p2p/dtls/dtls_ice_integration_test.cc b/p2p/dtls/dtls_ice_integration_test.cc
index 083c4add85..7f8d51a97c 100644
--- a/p2p/dtls/dtls_ice_integration_test.cc
+++ b/p2p/dtls/dtls_ice_integration_test.cc
@@ -31,6 +31,7 @@
 #include "rtc_base/logging.h"
 #include "rtc_base/random.h"
 #include "rtc_base/socket_address.h"
+#include "rtc_base/ssl_stream_adapter.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
 #include "test/wait_until.h"
@@ -86,24 +87,29 @@ class DtlsIceIntegrationTest : public dtls_ice_integration_fixture::Base,
   }
 
   void CheckRetransmissions() {
-    if (client_.config.dtls_in_stun != server_.config.dtls_in_stun) {
-      // If peers does not have same setting for DTLS in STUN
-      // the current implementation does sometimes introduce a DTLS
-      // retransmit.
-      //
-      // TODO: bugs.webrtc.org/367395350 - It would be nice to remove these!
-      EXPECT_LE(client_.dtls->GetRetransmissionCount(), 1);
-      EXPECT_LE(server_.dtls->GetRetransmissionCount(), 1);
+    if (!server_.config.ice_lite) {
+      EXPECT_EQ(client_.dtls->GetRetransmissionCount(), 0);
+      EXPECT_EQ(server_.dtls->GetRetransmissionCount(), 0);
+      return;
+    }
+    if (client_.config.ssl_role == SSL_CLIENT) {
+      EXPECT_EQ(client_.dtls->GetRetransmissionCount(), 0);
+      EXPECT_EQ(server_.dtls->GetRetransmissionCount(), 0);
       return;
     }
 
-    EXPECT_EQ(client_.dtls->GetRetransmissionCount(), 0);
-    if (!server_.config.ice_lite) {
-      // TODO: bugs.webrtc.org/367395350 - We allow retransmissions
-      // from the fake ice-lite agent, but it would be great if
-      // we can remove them.
-      EXPECT_EQ(server_.dtls->GetRetransmissionCount(), 0);
+    // TODO: bugs.webrtc.org/367395350 - Investigate retransmissions
+    // from in fake ice lite scenarios. Very few remaining!
+    // - server is (fake) ICE lite but SSL_CLIENT
+    if (client_.config.dtls_in_stun == server_.config.dtls_in_stun) {
+      const bool pqc = client_.config.pqc && server_.config.pqc;
+      EXPECT_EQ(client_.dtls->GetRetransmissionCount(), 0);
+      EXPECT_LE(server_.dtls->GetRetransmissionCount(), !pqc ? 0 : 1);
+      return;
     }
+
+    EXPECT_LE(client_.dtls->GetRetransmissionCount(), 1);
+    EXPECT_LE(server_.dtls->GetRetransmissionCount(), 1);
   }
 };
 
diff --git a/p2p/dtls/dtls_transport.cc b/p2p/dtls/dtls_transport.cc
index 98e901c849..ae516ad704 100644
--- a/p2p/dtls/dtls_transport.cc
+++ b/p2p/dtls/dtls_transport.cc
@@ -1235,8 +1235,19 @@ void DtlsTransportInternalImpl::ConfigureHandshakeTimeout() {
 void DtlsTransportInternalImpl::UpdateHandshakeTimeout() {
   RTC_DCHECK(dtls_);
   const auto rtt_ms = ice_transport()->GetRttEstimate();
-  const int delay_ms = ComputeRetransmissionTimeout(
+  int delay_ms = ComputeRetransmissionTimeout(
       rtt_ms.value_or(kDefaultHandshakeEstimateRttMs));
+  if (dtls_stun_piggyback_controller_.state() ==
+          DtlsStunPiggybackController::State::OFF &&
+      dtls_role_ == SSL_CLIENT) {
+    // We sent one STUN BINDING request with an embedded DTLS packet and
+    // discovered that peer does not support DtlsInStun. The DTLS packet will be
+    // sent by PeriodicRetransmitDtlsPacketUntilDtlsConnected and that will
+    // incur one more RTT. Increase slightly timeout to avoid unneeded DTLS
+    // retranmission.
+    delay_ms = (delay_ms * 133) / 100;
+  }
+
   RTC_LOG(LS_INFO) << ToString() << ": Update DTLS handshake timeout to "
                    << delay_ms << "ms based on ICE RTT "
                    << (rtt_ms ? std::to_string(*rtt_ms) : "<unset>");
@@ -1296,6 +1307,13 @@ void DtlsTransportInternalImpl::
     }
   }
 
+  if (dtls_stun_piggyback_controller_.state() ==
+      DtlsStunPiggybackController::State::OFF) {
+    // Peer does not support DTLS in STUN. We have now retransmitted the packet
+    // once, and let DTLS handle further retransmits.
+    return;
+  }
+
   const auto rtt_ms = ice_transport()->GetRttEstimate().value_or(
       kDefaultHandshakeEstimateRttMs);
   const int delay_ms = ComputeRetransmissionTimeout(rtt_ms);
diff --git a/p2p/dtls/dtls_transport_unittest.cc b/p2p/dtls/dtls_transport_unittest.cc
index a987d71330..b73a0db9fd 100644
--- a/p2p/dtls/dtls_transport_unittest.cc
+++ b/p2p/dtls/dtls_transport_unittest.cc
@@ -1821,19 +1821,23 @@ class DtlsTransportInternalImplDtlsInStunTest
   DtlsTransportInternalImplDtlsInStunTest() {}
 
   void CheckRetransmissions() {
+    EXPECT_EQ(client1_.dtls_transport()->GetRetransmissionCount(), 0);
     if (std::get<0>(GetParam()).dtls_in_stun ==
         std::get<1>(GetParam()).dtls_in_stun) {
-      EXPECT_EQ(client1_.dtls_transport()->GetRetransmissionCount(), 0);
       EXPECT_EQ(client2_.dtls_transport()->GetRetransmissionCount(), 0);
-    } else {
-      // If peers does not have same setting for DTLS in STUN
-      // current implementation does sometimes introduce a DTLS
-      // retransmit.
-      //
-      // TODO: bugs.webrtc.org/367395350 - It would be nice to remove these!
-      EXPECT_LE(client1_.dtls_transport()->GetRetransmissionCount(), 1);
-      EXPECT_LE(client2_.dtls_transport()->GetRetransmissionCount(), 1);
+      return;
     }
+    if (std::get<0>(GetParam()).ssl_role == SSL_CLIENT) {
+      EXPECT_EQ(client2_.dtls_transport()->GetRetransmissionCount(), 0);
+      return;
+    }
+
+    // If peers does not have same setting for DTLS in STUN
+    // current implementation does sometimes introduce a DTLS
+    // retransmit on client2 (activing as SSL_CLIENT).
+    //
+    // TODO: bugs.webrtc.org/367395350 - It would be nice to remove these!
+    EXPECT_LE(client2_.dtls_transport()->GetRetransmissionCount(), 1);
   }
 };
 

From cb6cfeaccaed6f0f994c4cad611937b55aabdaba Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Mon, 2 Mar 2026 10:23:26 +0100
Subject: [PATCH 167/472] Update WebRtcVoiceEngineTestFake tests to use
 test::RunLoop

Bug: webrtc:469327588
Change-Id: Ic4e29b02390359d8afadfebe391b0c5bc858fa35
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452961
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47020}
---
 media/engine/webrtc_voice_engine_unittest.cc | 20 ++++++++++----------
 1 file changed, 10 insertions(+), 10 deletions(-)

diff --git a/media/engine/webrtc_voice_engine_unittest.cc b/media/engine/webrtc_voice_engine_unittest.cc
index 1361eb6afe..78e4f2ae78 100644
--- a/media/engine/webrtc_voice_engine_unittest.cc
+++ b/media/engine/webrtc_voice_engine_unittest.cc
@@ -73,12 +73,12 @@
 #include "rtc_base/copy_on_write_buffer.h"
 #include "rtc_base/dscp.h"
 #include "rtc_base/numerics/safe_conversions.h"
-#include "rtc_base/thread.h"
 #include "test/create_test_field_trials.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
 #include "test/mock_audio_decoder_factory.h"
 #include "test/mock_audio_encoder_factory.h"
+#include "test/run_loop.h"
 
 namespace webrtc {
 namespace {
@@ -340,7 +340,7 @@ class WebRtcVoiceEngineTestFake : public ::testing::TestWithParam<bool> {
     RtpPacketReceived packet;
     packet.Parse(data);
     receive_channel_->OnPacketReceived(packet);
-    Thread::Current()->ProcessMessages(0);
+    run_loop_.Flush();
   }
 
   const FakeAudioSendStream& GetSendStream(uint32_t ssrc) {
@@ -886,7 +886,7 @@ class WebRtcVoiceEngineTestFake : public ::testing::TestWithParam<bool> {
   }
 
  protected:
-  AutoThread main_thread_;
+  test::RunLoop run_loop_;
   const bool use_null_apm_;
   FieldTrials field_trials_;
   const Environment env_;
@@ -1608,7 +1608,7 @@ TEST_P(WebRtcVoiceEngineTestFake, OnPacketReceivedIdentifiesExtensions) {
   ASSERT_TRUE(received_packet.Parse(reference_packet.Buffer()));
 
   receive_channel_->OnPacketReceived(received_packet);
-  Thread::Current()->ProcessMessages(0);
+  run_loop_.Flush();
 
   AudioLevel audio_level;
   EXPECT_TRUE(
@@ -3602,7 +3602,7 @@ TEST_P(WebRtcVoiceEngineTestFake, DeliverAudioPacket_Call) {
   RtpPacketReceived parsed_packet;
   RTC_CHECK(parsed_packet.Parse(kPcmuPacket));
   receive_channel_->OnPacketReceived(parsed_packet);
-  Thread::Current()->ProcessMessages(0);
+  run_loop_.Flush();
 
   EXPECT_EQ(1, s->received_packets());
 }
@@ -3736,7 +3736,7 @@ TEST_P(WebRtcVoiceEngineTestFake, GetSourcesWithNonExistingSsrc) {
 
 // Tests that the library initializes and shuts down properly.
 TEST(WebRtcVoiceEngineTest, StartupShutdown) {
-  AutoThread main_thread;
+  test::RunLoop run_loop;
   for (bool use_null_apm : {false, true}) {
     // If the VoiceEngine wants to gather available codecs early, that's fine
     // but we never want it to create a decoder at this stage.
@@ -3763,7 +3763,7 @@ TEST(WebRtcVoiceEngineTest, StartupShutdown) {
 
 // Tests that reference counting on the external ADM is correct.
 TEST(WebRtcVoiceEngineTest, StartupShutdownWithExternalADM) {
-  AutoThread main_thread;
+  test::RunLoop run_loop;
   for (bool use_null_apm : {false, true}) {
     Environment env = CreateEnvironment();
     auto adm =
@@ -3840,7 +3840,7 @@ TEST(WebRtcVoiceEngineTest, HasCorrectPayloadTypeMapping) {
 
 // Tests that VoE supports at least 32 channels
 TEST(WebRtcVoiceEngineTest, Has32Channels) {
-  AutoThread main_thread;
+  test::RunLoop run_loop;
   for (bool use_null_apm : {false, true}) {
     Environment env = CreateEnvironment();
     scoped_refptr<test::MockAudioDeviceModule> adm =
@@ -3869,7 +3869,7 @@ TEST(WebRtcVoiceEngineTest, Has32Channels) {
 
 // Test that we set our preferred codecs properly.
 TEST(WebRtcVoiceEngineTest, SetRecvCodecs) {
-  AutoThread main_thread;
+  test::RunLoop run_loop;
   for (bool use_null_apm : {false, true}) {
     Environment env = CreateEnvironment();
     // TODO(ossu): I'm not sure of the intent of this test. It's either:
@@ -3898,7 +3898,7 @@ TEST(WebRtcVoiceEngineTest, SetRecvCodecs) {
 }
 
 TEST(WebRtcVoiceEngineTest, SetRtpSendParametersMaxBitrate) {
-  AutoThread main_thread;
+  test::RunLoop run_loop;
   Environment env = CreateEnvironment();
   scoped_refptr<test::MockAudioDeviceModule> adm =
       test::MockAudioDeviceModule::CreateNice();

From 03e07825487c7e853c06ffa161e158cb02ac3a8b Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Sun, 1 Mar 2026 22:04:11 -0800
Subject: [PATCH 168/472] Roll chromium_revision d5da27240e..0fe94aae05
 (1591990:1592255)

Change log: https://chromium.googlesource.com/chromium/src/+log/d5da27240e..0fe94aae05
Full diff: https://chromium.googlesource.com/chromium/src/+/d5da27240e..0fe94aae05

Changed dependencies
* siso_version: git_revision:8aacae89cf77656164b64fd9d24b3edb884b88ac..git_revision:7495bf92ea44f7eed89273b5aea7bd72b702ed1c
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/06e0cff019..dc5d07285e
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/0a8ae42116..3a29ca982c
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/a93b072843..647ef08e52
* src/third_party/android_build_tools/manifest_merger/cipd: tjC-RIoLTs8HsZHxeRq-7GN90bYNYSQTx2yzmzL8hUcC..tu2pGdrcOsdBnUbmZgzOgxIrZh-YJhlAL7ICIZRTjH4C
* src/third_party/androidx/cipd: _wBJ2hBwWyOSj4dkL0Ttoxv3fb4wX0ZrTBMEDITnWXwC..OD2jbM01RuaeYoMBn_c20OTgrQTMyZc6HCvDmfzda0UC
* src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/bc0a39e5b3..36b7ba36ef
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/25dba492e3..662d5c21ff
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/4b296e3278..95f56ff92d
* src/third_party/kotlin_stdlib/cipd: MQrhfTOsQoBh8QajYKNMkPBvzh7bqyv31z7HPbnbo_QC..r1jin7oxmCEIuca5n9GqpxpPk_aV2Dr0CLAmWSV0tEAC
* src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/bfe85bb036..7434d317bd
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/fa49f16ad9..4490d5b468
* src/third_party/siso/cipd: git_revision:8aacae89cf77656164b64fd9d24b3edb884b88ac..git_revision:7495bf92ea44f7eed89273b5aea7bd72b702ed1c
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/58559285e3..8fb7050d8d
DEPS diff: https://chromium.googlesource.com/chromium/src/+/d5da27240e..0fe94aae05/DEPS

BUG=None

Change-Id: Icb8118f4218ff71434d9c60e0bea9213eb14a621
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452920
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47021}
---
 DEPS | 28 ++++++++++++++--------------
 1 file changed, 14 insertions(+), 14 deletions(-)

diff --git a/DEPS b/DEPS
index 379672f915..6d408e903a 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': 'd5da27240ef4e2938af4795cb9fa9d351bdf1f5b',
+  'chromium_revision': '0fe94aae0587d2116579a2cdeb9f630c02aa60b1',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -50,7 +50,7 @@ vars = {
   # reclient CIPD package version
   'reclient_version': 're_client_version:0.185.0.db415f21-gomaip',
   # siso CIPD package version.
-  'siso_version': 'git_revision:8aacae89cf77656164b64fd9d24b3edb884b88ac',
+  'siso_version': 'git_revision:7495bf92ea44f7eed89273b5aea7bd72b702ed1c',
 
   # ninja CIPD package.
   'ninja_package': 'infra/3pp/tools/ninja/',
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@06e0cff019f4c65b5aec03bf9dce1c9bc6002575',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@dc5d07285ede7e88bbbcec05e4300cf9b8562d21',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@0a8ae42116bf528b9fd7e71529a01dfef04613ed',
+    'https://chromium.googlesource.com/chromium/src/testing@3a29ca982cd7b33629f7a07a63212c13375fb0b2',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@a93b072843ac210f36fdd17de27f441e026f0ce7',
+    'https://chromium.googlesource.com/chromium/src/third_party@647ef08e52f851b369be8b5b49a5007b2add6ca3',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -331,13 +331,13 @@ deps = {
   'src/third_party/clang-format/script':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@c2725e0622e1a86d55f14514f2177a39efea4a0e',
   'src/third_party/compiler-rt/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@bc0a39e5b30af5cc47dd0df097a9b817e80ee3aa',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@36b7ba36ef48cbb211135173c1285063df3f73c3',
   'src/third_party/libc++/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@7ab65651aed6802d2599dcb7a73b1f82d5179d05',
   'src/third_party/libc++abi/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@8f11bb1d4438d0239d0dfc1bd9456a9f31629dda',
   'src/third_party/llvm-libc/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@bfe85bb0366b861995d6bad528c1426ce72a0518',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@7434d317bd47832fe6b30fcdf33796ddcc02a7d4',
   'src/third_party/libunwind/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@17ccf7d110c5526cb77e93cfd8330f491fb2bf18',
 
@@ -511,7 +511,7 @@ deps = {
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@4b296e3278b656c8baf20041bc8f7e1f410f67b9',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@95f56ff92dd300ca2626451679c9a2ee7ea69007',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -573,7 +573,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/kotlin_stdlib',
-              'version': 'MQrhfTOsQoBh8QajYKNMkPBvzh7bqyv31z7HPbnbo_QC',
+              'version': 'r1jin7oxmCEIuca5n9GqpxpPk_aV2Dr0CLAmWSV0tEAC',
           },
       ],
       'condition': 'checkout_android',
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'fa49f16ad958de750da23698f12aebb9f1ae662f',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '4490d5b468a6898635169adb3a5372bfa570c0c2',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@58559285e323d5e5d6645861d788cda8d3030ef9',
+    'https://chromium.googlesource.com/chromium/src/tools@8fb7050d8debd17ccab49d987ac823c21ece1cf0',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': '_wBJ2hBwWyOSj4dkL0Ttoxv3fb4wX0ZrTBMEDITnWXwC',
+          'version': 'OD2jbM01RuaeYoMBn_c20OTgrQTMyZc6HCvDmfzda0UC',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -718,7 +718,7 @@ deps = {
       'packages': [
           {
                'package': 'chromium/third_party/android_build_tools/manifest_merger',
-               'version': 'tjC-RIoLTs8HsZHxeRq-7GN90bYNYSQTx2yzmzL8hUcC',
+               'version': 'tu2pGdrcOsdBnUbmZgzOgxIrZh-YJhlAL7ICIZRTjH4C',
           },
       ],
       'condition': 'checkout_android',
@@ -864,7 +864,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7364b490b5f78d58efe23ea76e74210fd6c3c76f',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '25dba492e30e19dce3d4bd4cd38af2f5c88c387c',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '662d5c21ff84359e433799b72738f96063cc60f2',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From 556f930bf915abd3d6fccae3416d08b563bfe88f Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Sat, 28 Feb 2026 20:05:23 -0800
Subject: [PATCH 169/472] Update WebRTC code version (2026-03-01T04:04:55).

Bug: None
Change-Id: I921dc754f0780579b28b0255f28d0c0f20d4fa09
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452880
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47022}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index 8d5d9c74f8..e9f968dd36 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-02-28T04:02:22";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-01T04:04:55";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From c1436d3676d2117357565eea2e3c09d50250c98e Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Mon, 2 Mar 2026 04:11:39 -0800
Subject: [PATCH 170/472] Roll chromium_revision 0fe94aae05..e4c5820c11
 (1592255:1592359)

Change log: https://chromium.googlesource.com/chromium/src/+log/0fe94aae05..e4c5820c11
Full diff: https://chromium.googlesource.com/chromium/src/+/0fe94aae05..e4c5820c11

Changed dependencies
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/dc5d07285e..f7ca1085c6
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/3a29ca982c..39d5c89737
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/647ef08e52..cee93ade03
* src/third_party/androidx/cipd: OD2jbM01RuaeYoMBn_c20OTgrQTMyZc6HCvDmfzda0UC..ouJMUlKeLTumklZ_7dBIDOk5sUXRmhSpw5PThCegRVsC
* src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/36b7ba36ef..64c252741e
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/662d5c21ff..57b1de2330
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/95f56ff92d..034b681167
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/4490d5b468..7616314b39
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/8fb7050d8d..8fcdba46a3
* src/tools/luci-go: git_revision:1959c0f345fc4fc01903ce8437d507d9cf94e390..git_revision:b9d7d1a7774833e136aaa5084a0d150cd5ecb641
* src/tools/luci-go: git_revision:1959c0f345fc4fc01903ce8437d507d9cf94e390..git_revision:b9d7d1a7774833e136aaa5084a0d150cd5ecb641
* src/tools/luci-go: git_revision:1959c0f345fc4fc01903ce8437d507d9cf94e390..git_revision:b9d7d1a7774833e136aaa5084a0d150cd5ecb641
DEPS diff: https://chromium.googlesource.com/chromium/src/+/0fe94aae05..e4c5820c11/DEPS

BUG=None

Change-Id: Iee5a16cf71a0fa4e51b35716d2961ccee87163d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452980
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47023}
---
 DEPS | 26 +++++++++++++-------------
 1 file changed, 13 insertions(+), 13 deletions(-)

diff --git a/DEPS b/DEPS
index 6d408e903a..4869afa5e3 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '0fe94aae0587d2116579a2cdeb9f630c02aa60b1',
+  'chromium_revision': 'e4c5820c11dc2bdc6f2eeb3586836f3d59c6a415',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@dc5d07285ede7e88bbbcec05e4300cf9b8562d21',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@f7ca1085c6a04bfa0887448317557e841eb2574f',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@3a29ca982cd7b33629f7a07a63212c13375fb0b2',
+    'https://chromium.googlesource.com/chromium/src/testing@39d5c89737938f96f2dbc23aafc6916bffc52bf3',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@647ef08e52f851b369be8b5b49a5007b2add6ca3',
+    'https://chromium.googlesource.com/chromium/src/third_party@cee93ade030bf334229c9930b53da2cf2e55f2a6',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -331,7 +331,7 @@ deps = {
   'src/third_party/clang-format/script':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@c2725e0622e1a86d55f14514f2177a39efea4a0e',
   'src/third_party/compiler-rt/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@36b7ba36ef48cbb211135173c1285063df3f73c3',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@64c252741e3d8d9906473b1d09733cff93b91f50',
   'src/third_party/libc++/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@7ab65651aed6802d2599dcb7a73b1f82d5179d05',
   'src/third_party/libc++abi/src':
@@ -511,7 +511,7 @@ deps = {
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@95f56ff92dd300ca2626451679c9a2ee7ea69007',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@034b6811670e7edf0694bf21f4e2ad44f497b545',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '4490d5b468a6898635169adb3a5372bfa570c0c2',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '7616314b391aa5fe69aa57705216c40727eb0160',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@8fb7050d8debd17ccab49d987ac823c21ece1cf0',
+    'https://chromium.googlesource.com/chromium/src/tools@8fcdba46a3a3897d9bec19c7da16b0c1b8b2e201',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'OD2jbM01RuaeYoMBn_c20OTgrQTMyZc6HCvDmfzda0UC',
+          'version': 'ouJMUlKeLTumklZ_7dBIDOk5sUXRmhSpw5PThCegRVsC',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -808,15 +808,15 @@ deps = {
       'packages': [
         {
           'package': 'infra/tools/luci/cas/${{platform}}',
-          'version': 'git_revision:1959c0f345fc4fc01903ce8437d507d9cf94e390',
+          'version': 'git_revision:b9d7d1a7774833e136aaa5084a0d150cd5ecb641',
         },
         {
           'package': 'infra/tools/luci/isolate/${{platform}}',
-          'version': 'git_revision:1959c0f345fc4fc01903ce8437d507d9cf94e390',
+          'version': 'git_revision:b9d7d1a7774833e136aaa5084a0d150cd5ecb641',
         },
         {
           'package': 'infra/tools/luci/swarming/${{platform}}',
-          'version': 'git_revision:1959c0f345fc4fc01903ce8437d507d9cf94e390',
+          'version': 'git_revision:b9d7d1a7774833e136aaa5084a0d150cd5ecb641',
         }
       ],
       'dep_type': 'cipd',
@@ -864,7 +864,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7364b490b5f78d58efe23ea76e74210fd6c3c76f',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '662d5c21ff84359e433799b72738f96063cc60f2',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '57b1de2330d0660f353037a743fada210bde41ce',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From bfd03d6f2518ffad67cf4b5371025df6ec9e7551 Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Tue, 17 Feb 2026 22:02:50 +0000
Subject: [PATCH 171/472] Delete deprecated functions in rtc_base/byte_order.h

This can be submitted when there are no more users - no earlier than March 1 according to the PSA sent out to discuss-webrtc on Feb 17.

Bug: webrtc:484070270
Change-Id: I4a3c5f3db949242761311d7b4c3799ff995d7cd4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449361
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47024}
---
 rtc_base/byte_order.h | 90 -------------------------------------------
 1 file changed, 90 deletions(-)

diff --git a/rtc_base/byte_order.h b/rtc_base/byte_order.h
index 7f163980b5..8a31c6307f 100644
--- a/rtc_base/byte_order.h
+++ b/rtc_base/byte_order.h
@@ -98,57 +98,28 @@ inline void Set8(ArrayView<uint8_t> data, size_t offset, uint8_t v) {
   data[offset] = v;
 }
 
-[[deprecated("Use version with ArrayView")]]
-inline void Set8(void* memory, size_t offset, uint8_t v) {
-  static_cast<uint8_t*>(memory)[offset] = v;
-}
-
 inline uint8_t Get8(ArrayView<const uint8_t> data, size_t offset) {
   return data[offset];
 }
 
-[[deprecated("Use version with ArrayView")]]
-inline uint8_t Get8(const void* memory, size_t offset) {
-  return static_cast<const uint8_t*>(memory)[offset];
-}
-
 inline void SetBE16(ArrayView<uint8_t> data, uint16_t v) {
   RTC_DCHECK_GE(data.size(), sizeof(v));
   uint16_t val = htobe16(v);
   memcpy(data.data(), &val, sizeof(val));
 }
 
-[[deprecated("Use version with ArrayView")]] inline void SetBE16(void* memory,
-                                                                 uint16_t v) {
-  uint16_t val = htobe16(v);
-  memcpy(memory, &val, sizeof(val));
-}
-
 inline void SetBE32(ArrayView<uint8_t> data, uint32_t v) {
   RTC_DCHECK_GE(data.size(), sizeof(v));
   uint32_t val = htobe32(v);
   memcpy(data.data(), &val, sizeof(val));
 }
 
-// TODO: issues.webrtc.org/484070270 - deprecate once Chromium usage is gone.
-// [[deprecated("Use version with ArrayView")]]
-inline void SetBE32(void* memory, uint32_t v) {
-  uint32_t val = htobe32(v);
-  memcpy(memory, &val, sizeof(val));
-}
-
 inline void SetBE64(ArrayView<uint8_t> data, uint64_t v) {
   RTC_DCHECK_GE(data.size(), sizeof(v));
   uint64_t val = htobe64(v);
   memcpy(data.data(), &val, sizeof(val));
 }
 
-[[deprecated("Use version with ArrayView")]] inline void SetBE64(void* memory,
-                                                                 uint64_t v) {
-  uint64_t val = htobe64(v);
-  memcpy(memory, &val, sizeof(val));
-}
-
 inline uint16_t GetBE16(ArrayView<const uint8_t> data) {
   uint16_t val;
   RTC_DCHECK_GE(data.size(), sizeof(val));
@@ -156,13 +127,6 @@ inline uint16_t GetBE16(ArrayView<const uint8_t> data) {
   return be16toh(val);
 }
 
-[[deprecated("Use version with ArrayView")]] inline uint16_t GetBE16(
-    const void* memory) {
-  uint16_t val;
-  memcpy(&val, memory, sizeof(val));
-  return be16toh(val);
-}
-
 inline uint32_t GetBE32(ArrayView<const uint8_t> data) {
   uint32_t val;
   RTC_DCHECK_GE(data.size(), sizeof(val));
@@ -170,14 +134,6 @@ inline uint32_t GetBE32(ArrayView<const uint8_t> data) {
   return be32toh(val);
 }
 
-// TODO: issues.webrtc.org/484070270 - deprecate once Chromium usage is gone.
-// [[deprecated("Use version with ArrayView")]]
-inline uint32_t GetBE32(const void* memory) {
-  uint32_t val;
-  memcpy(&val, memory, sizeof(val));
-  return be32toh(val);
-}
-
 inline uint64_t GetBE64(ArrayView<const uint8_t> data) {
   uint64_t val;
   RTC_DCHECK_GE(data.size(), sizeof(val));
@@ -185,49 +141,24 @@ inline uint64_t GetBE64(ArrayView<const uint8_t> data) {
   return be64toh(val);
 }
 
-[[deprecated("Use version with ArrayView")]] inline uint64_t GetBE64(
-    const void* memory) {
-  uint64_t val;
-  memcpy(&val, memory, sizeof(val));
-  return be64toh(val);
-}
-
 inline void SetLE16(ArrayView<uint8_t> data, uint16_t v) {
   RTC_DCHECK_GE(data.size(), sizeof(v));
   uint16_t val = htole16(v);
   memcpy(data.data(), &val, sizeof(val));
 }
 
-[[deprecated("Use version with ArrayView")]] inline void SetLE16(void* memory,
-                                                                 uint16_t v) {
-  uint16_t val = htole16(v);
-  memcpy(memory, &val, sizeof(val));
-}
-
 inline void SetLE32(ArrayView<uint8_t> data, uint32_t v) {
   RTC_DCHECK_GE(data.size(), sizeof(v));
   uint32_t val = htole32(v);
   memcpy(data.data(), &val, sizeof(val));
 }
 
-[[deprecated("Use version with ArrayView")]] inline void SetLE32(void* memory,
-                                                                 uint32_t v) {
-  uint32_t val = htole32(v);
-  memcpy(memory, &val, sizeof(val));
-}
-
 inline void SetLE64(ArrayView<uint8_t> data, uint64_t v) {
   RTC_DCHECK_GE(data.size(), sizeof(v));
   uint64_t val = htole64(v);
   memcpy(data.data(), &val, sizeof(val));
 }
 
-[[deprecated("Use version with ArrayView")]] inline void SetLE64(void* memory,
-                                                                 uint64_t v) {
-  uint64_t val = htole64(v);
-  memcpy(memory, &val, sizeof(val));
-}
-
 inline uint16_t GetLE16(ArrayView<const uint8_t> data) {
   uint16_t val;
   RTC_DCHECK_GE(data.size(), sizeof(val));
@@ -235,13 +166,6 @@ inline uint16_t GetLE16(ArrayView<const uint8_t> data) {
   return le16toh(val);
 }
 
-[[deprecated("Use version with ArrayView")]] inline uint16_t GetLE16(
-    const void* memory) {
-  uint16_t val;
-  memcpy(&val, memory, sizeof(val));
-  return le16toh(val);
-}
-
 inline uint32_t GetLE32(ArrayView<const uint8_t> data) {
   uint32_t val;
   RTC_DCHECK_GE(data.size(), sizeof(val));
@@ -249,13 +173,6 @@ inline uint32_t GetLE32(ArrayView<const uint8_t> data) {
   return le32toh(val);
 }
 
-[[deprecated("Use version with ArrayView")]] inline uint32_t GetLE32(
-    const void* memory) {
-  uint32_t val;
-  memcpy(&val, memory, sizeof(val));
-  return le32toh(val);
-}
-
 inline uint64_t GetLE64(ArrayView<const uint8_t> data) {
   uint64_t val;
   RTC_DCHECK_GE(data.size(), sizeof(val));
@@ -263,13 +180,6 @@ inline uint64_t GetLE64(ArrayView<const uint8_t> data) {
   return le64toh(val);
 }
 
-[[deprecated("Use version with ArrayView")]] inline uint64_t GetLE64(
-    const void* memory) {
-  uint64_t val;
-  memcpy(&val, memory, sizeof(val));
-  return le64toh(val);
-}
-
 // Check if the current host is big endian.
 inline bool IsHostBigEndian() {
 #if defined(WEBRTC_ARCH_BIG_ENDIAN)

From 2f5a2659b28d1aed4b301b4b137db98bd618e287 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Mon, 2 Mar 2026 08:04:01 -0800
Subject: [PATCH 172/472] Roll chromium_revision e4c5820c11..faab630875
 (1592359:1592487)

Change log: https://chromium.googlesource.com/chromium/src/+log/e4c5820c11..faab630875
Full diff: https://chromium.googlesource.com/chromium/src/+/e4c5820c11..faab630875

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/8cfed3408e..b62d394bff
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/f7ca1085c6..e72cef5d25
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/39d5c89737..747c2fa02b
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/cee93ade03..64fce8e53a
* src/third_party/androidx/cipd: ouJMUlKeLTumklZ_7dBIDOk5sUXRmhSpw5PThCegRVsC..F33f66w1lXogUL7EN68PuqtiS1I2B8ptEm0Zq-MhYnIC
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/0f9739da7f..27800c3890
* src/third_party/freetype/src: https://chromium.googlesource.com/chromium/src/third_party/freetype2.git/+log/e3a0652b6d..bf1f34d75e
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/7616314b39..6391c13617
* src/third_party/turbine/cipd: ymsdRx1sSajlZKbTHsqqquqR8lYueFUJCPwDCXmIfMEC..0A4lFRLjqycR4-EvoBz1w2FxPEzzG94cFvwu8v39DRYC
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/8fcdba46a3..a381c66832
DEPS diff: https://chromium.googlesource.com/chromium/src/+/e4c5820c11..faab630875/DEPS

BUG=None

Change-Id: I344c415b3af6a25aeda6a0011e394c8032e7565d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453060
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47025}
---
 DEPS | 22 +++++++++++-----------
 1 file changed, 11 insertions(+), 11 deletions(-)

diff --git a/DEPS b/DEPS
index 4869afa5e3..57bb580ac8 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': 'e4c5820c11dc2bdc6f2eeb3586836f3d59c6a415',
+  'chromium_revision': 'faab630875af0fdba81c9ce794f28b87532adfea',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@8cfed3408ec11d1780abcd66e9e0d8ad8b21ce75',
+    'https://chromium.googlesource.com/chromium/src/build@b62d394bff1cca126e4e8b7a711da7829003d003',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@136da69a1267b8db487354b96d44d0cc8add5aeb',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@f7ca1085c6a04bfa0887448317557e841eb2574f',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@e72cef5d25de4912a4380b6903a1def08d81aacf',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@39d5c89737938f96f2dbc23aafc6916bffc52bf3',
+    'https://chromium.googlesource.com/chromium/src/testing@747c2fa02b992061c8b773a3927de086fb2378b9',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@cee93ade030bf334229c9930b53da2cf2e55f2a6',
+    'https://chromium.googlesource.com/chromium/src/third_party@64fce8e53a91fdae0d404332980555ca6514cf25',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -492,7 +492,7 @@ deps = {
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@9f26d631e4063c0af3e8ee3bc8826486115372a8',
   'src/third_party/catapult':
-    'https://chromium.googlesource.com/catapult.git@0f9739da7f76bbb5bc9c108986880d5cfb9b9cf6',
+    'https://chromium.googlesource.com/catapult.git@27800c3890b2aaf0bdbefc8ba7b0a0a394cb7e72',
   'src/third_party/ced/src': {
     'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5',
   },
@@ -519,7 +519,7 @@ deps = {
       'condition': 'checkout_linux',
   },
   'src/third_party/freetype/src':
-    'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@e3a0652b6d706ee1ce77d4dda606b6597dd8b5c4',
+    'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@bf1f34d75ed4a36358e7a4ae6b21082d0d435b0e',
   'src/third_party/harfbuzz-ng/src':
     'https://chromium.googlesource.com/external/github.com/harfbuzz/harfbuzz.git@fa2908bf16d2ccd6623f4d575455fea72a1a722b',
   'src/third_party/google_benchmark/src': {
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '7616314b391aa5fe69aa57705216c40727eb0160',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '6391c1361787a9be611fc207447d0af2ddb00f5e',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@8fcdba46a3a3897d9bec19c7da16b0c1b8b2e201',
+    'https://chromium.googlesource.com/chromium/src/tools@a381c6683255f0d7d80d94f4bd72f22199b27ef2',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'ouJMUlKeLTumklZ_7dBIDOk5sUXRmhSpw5PThCegRVsC',
+          'version': 'F33f66w1lXogUL7EN68PuqtiS1I2B8ptEm0Zq-MhYnIC',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -792,7 +792,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/turbine',
-              'version': 'ymsdRx1sSajlZKbTHsqqquqR8lYueFUJCPwDCXmIfMEC',
+              'version': '0A4lFRLjqycR4-EvoBz1w2FxPEzzG94cFvwu8v39DRYC',
           },
       ],
       'condition': 'checkout_android',

From 427b9202ad1ce6f9ec57c13e6712728444f0291c Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Mon, 2 Mar 2026 14:56:52 +0100
Subject: [PATCH 173/472] Allow calls to DeliverRtpPacket from the network
 thread

Update the pre-decoding part of the RTP packet receive pipeline to allow
DeliverRtpPacket to be called from either the network or worker thread
(network thread preferred moving forward).

This is a step towards moving the full packet processing pipeline to the
network thread. In practical terms, this removes one PostTask for audio
packets when the network and worker threads are configured to be the
same. Architecturally, this moves the PostTask from network to worker,
out of the voice/video receive channel classes and into call (by 1
step).

Key modifications:
* Update Call::DeliverRtpPacket to automatically dispatch tasks to the
  worker thread if it is invoked from a different context.
* Shift RtpPacketReceived processing in WebRtcVideoReceiveChannel and
  WebRtcVoiceReceiveChannel to the network thread.
* Relocate recv_rtp_extension_map_ to be guarded by the network thread
  checker, ensuring header extension identification occurs immediately
  upon packet receipt.
* Synchronize extension map updates from the worker thread to the
  network thread using a safe task-posting mechanism.
* Modify OnPacketReceived signatures to accept packets by value,
  enabling efficient moves across thread boundaries.

Bug: webrtc:11993, chromium:40242052
Change-Id: I2bf80a67632cee6bd553009ea9523499527bc93e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452900
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47026}
---
 call/call.cc                                  |  9 +++
 call/packet_receiver.h                        |  2 +-
 media/base/fake_media_engine.h                |  2 +-
 media/base/media_channel.h                    |  2 +-
 media/engine/webrtc_video_engine.cc           | 72 +++++++++----------
 media/engine/webrtc_video_engine.h            |  9 ++-
 media/engine/webrtc_video_engine_unittest.cc  |  9 ++-
 media/engine/webrtc_voice_engine.cc           | 51 +++++++------
 media/engine/webrtc_voice_engine.h            |  9 ++-
 media/engine/webrtc_voice_engine_unittest.cc  |  1 +
 ...ck_voice_media_receive_channel_interface.h |  5 +-
 11 files changed, 94 insertions(+), 77 deletions(-)

diff --git a/call/call.cc b/call/call.cc
index fc9c544cbb..052c8a76fe 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -1358,6 +1358,15 @@ void Call::DeliverRtpPacket(
     MediaType media_type,
     RtpPacketReceived packet,
     OnUndemuxablePacketHandler undemuxable_packet_handler) {
+  if (!worker_thread_->IsCurrent()) {
+    worker_thread_->PostTask(SafeTask(
+        task_safety_.flag(),
+        [this, media_type, packet = std::move(packet),
+         handler = std::move(undemuxable_packet_handler)]() mutable {
+          DeliverRtpPacket(media_type, std::move(packet), std::move(handler));
+        }));
+    return;
+  }
   RTC_DCHECK_RUN_ON(worker_thread_);
   RTC_DCHECK(packet.arrival_time().IsFinite());
 
diff --git a/call/packet_receiver.h b/call/packet_receiver.h
index c149f8e3cc..a1b5f66c13 100644
--- a/call/packet_receiver.h
+++ b/call/packet_receiver.h
@@ -27,7 +27,7 @@ class PacketReceiver {
   using OnUndemuxablePacketHandler =
       absl::AnyInvocable<bool(const RtpPacketReceived& parsed_packet)>;
 
-  // Must be called on the worker thread.
+  // Can be called on the network thread or the worker thread.
   // If `media_type` is not Audio or Video, packets may be used for BWE
   // calculations but are not demuxed.
   virtual void DeliverRtpPacket(
diff --git a/media/base/fake_media_engine.h b/media/base/fake_media_engine.h
index 69e9ef27cd..75e204d361 100644
--- a/media/base/fake_media_engine.h
+++ b/media/base/fake_media_engine.h
@@ -211,7 +211,7 @@ class RtpReceiveChannelHelper : public Base, public MediaChannelUtil {
       const MediaChannelParameters::RtcpParameters& params) {
     recv_rtcp_parameters_ = params;
   }
-  void OnPacketReceived(const RtpPacketReceived& packet) override {
+  void OnPacketReceived(RtpPacketReceived packet) override {
     rtp_packets_.push_back(
         std::string(packet.Buffer().cdata<char>(), packet.size()));
   }
diff --git a/media/base/media_channel.h b/media/base/media_channel.h
index 496726501a..a41a12dac2 100644
--- a/media/base/media_channel.h
+++ b/media/base/media_channel.h
@@ -265,7 +265,7 @@ class MediaReceiveChannelInterface {
   // Sets the abstract interface class for sending RTP/RTCP data.
   virtual void SetInterface(MediaChannelNetworkInterface* iface) = 0;
   // Called on the network when an RTP packet is received.
-  virtual void OnPacketReceived(const RtpPacketReceived& packet) = 0;
+  virtual void OnPacketReceived(RtpPacketReceived packet) = 0;
   // Gets the current unsignaled receive stream's SSRC, if there is one.
   virtual std::optional<uint32_t> GetUnsignaledSsrc() const = 0;
   // This is currently a workaround because of the demuxer state being managed
diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc
index 38d6909ea2..c6d876c517 100644
--- a/media/engine/webrtc_video_engine.cc
+++ b/media/engine/webrtc_video_engine.cc
@@ -2797,6 +2797,9 @@ WebRtcVideoReceiveChannel::WebRtcVideoReceiveChannel(
     : MediaChannelUtil(call->network_thread(), config.enable_dscp),
       env_(env),
       worker_thread_(call->worker_thread()),
+      network_thread_safety_(PendingTaskSafetyFlag::CreateAttachedToTaskQueue(
+          /*alive=*/true,
+          call->network_thread())),
       receiving_(false),
       call_(call),
       default_sink_(nullptr),
@@ -2944,7 +2947,13 @@ bool WebRtcVideoReceiveChannel::SetReceiverParameters(
   }
   if (changed_params.rtp_header_extensions) {
     recv_rtp_extensions_ = *changed_params.rtp_header_extensions;
-    recv_rtp_extension_map_ = RtpHeaderExtensionMap(recv_rtp_extensions_);
+    call_->network_thread()->PostTask(SafeTask(
+        network_thread_safety_,
+        [this, recv_rtp_extension_map =
+                   RtpHeaderExtensionMap(recv_rtp_extensions_)]() mutable {
+          RTC_DCHECK_RUN_ON(&network_thread_checker_);
+          recv_rtp_extension_map_ = std::move(recv_rtp_extension_map);
+        }));
   }
   if (changed_params.codec_settings) {
     RTC_DLOG(LS_INFO) << "Changing recv codecs from "
@@ -3210,25 +3219,32 @@ void WebRtcVideoReceiveChannel::FillReceiveCodecStats(
   }
 }
 
-void WebRtcVideoReceiveChannel::OnPacketReceived(
-    const RtpPacketReceived& packet) {
+void WebRtcVideoReceiveChannel::OnPacketReceived(RtpPacketReceived packet) {
   // Note: the network_thread_checker may refer to the worker thread if the two
   // threads are combined, but this is either always true or always false
   // depending on configuration set at object initialization.
   RTC_DCHECK_RUN_ON(&network_thread_checker_);
 
-  // TODO(crbug.com/1373439): Stop posting to the worker thread when the
-  // combined network/worker project launches.
-  if (TaskQueueBase::Current() != worker_thread_) {
-    worker_thread_->PostTask(
-        SafeTask(task_safety_.flag(), [this, packet = packet]() mutable {
-          RTC_DCHECK_RUN_ON(&thread_checker_);
-          ProcessReceivedPacket(std::move(packet));
-        }));
-  } else {
-    RTC_DCHECK_RUN_ON(&thread_checker_);
-    ProcessReceivedPacket(packet);
+  // TODO(bugs.webrtc.org/7135): The mapping of extension ID (on-packet
+  // representation) to C++ class object (internal representation) is currently
+  // associated with the per-channel object (e.g. WebRtcVideoReceiveChannel).
+  // However, this mapping is more logically associated with per-media-stream
+  // objects (like ModuleRtpRtcpImpl2), which handle specific streams flowing
+  // in one direction (usually a single SSRC).
+  // This mapping should be moved out of the media engines altogether, storing
+  // it instead at the stream level initialized by the Transport. RTP packets
+  // consumed by media channels would thus have header extensions distinguished
+  // by C++ class rather than negotiated header extension ID.
+  packet.IdentifyExtensions(recv_rtp_extension_map_);
+  packet.set_payload_type_frequency(kVideoPayloadTypeFrequency);
+  if (!packet.arrival_time().IsFinite()) {
+    packet.set_arrival_time(env_.clock().CurrentTime());
   }
+
+  call_->Receiver()->DeliverRtpPacket(
+      MediaType::VIDEO, std::move(packet),
+      absl::bind_front(
+          &WebRtcVideoReceiveChannel::MaybeCreateDefaultReceiveStream, this));
 }
 
 bool WebRtcVideoReceiveChannel::MaybeCreateDefaultReceiveStream(
@@ -3338,6 +3354,8 @@ void WebRtcVideoReceiveChannel::ReCreateDefaultReceiveStream(
 void WebRtcVideoReceiveChannel::SetInterface(
     MediaChannelNetworkInterface* iface) {
   RTC_DCHECK_RUN_ON(&network_thread_checker_);
+  iface ? network_thread_safety_->SetAlive()
+        : network_thread_safety_->SetNotAlive();
   MediaChannelUtil::SetInterface(iface);
   // Set the RTP recv/send buffer to a bigger size.
   MediaChannelUtil::SetOption(MediaChannelNetworkInterface::ST_RTP,
@@ -3975,32 +3993,6 @@ WebRtcVideoReceiveChannel::FindReceiveStream(uint32_t ssrc) {
   return nullptr;
 }
 
-// RTC_RUN_ON(worker_thread_)
-void WebRtcVideoReceiveChannel::ProcessReceivedPacket(
-    RtpPacketReceived packet) {
-  // TODO(bugs.webrtc.org/11993): This code is very similar to what
-  // WebRtcVoiceMediaChannel::OnPacketReceived does. For maintainability and
-  // consistency it would be good to move the interaction with call_->Receiver()
-  // to a common implementation and provide a callback on the worker thread
-  // for the exception case (DELIVERY_UNKNOWN_SSRC) and how retry is attempted.
-  // TODO(bugs.webrtc.org/7135): extensions in `packet` is currently set
-  // in RtpTransport and does not neccessarily include extensions specific
-  // to this channel/MID. Also see comment in
-  // BaseChannel::MaybeUpdateDemuxerAndRtpExtensions_w.
-  // It would likely be good if extensions where merged per BUNDLE and
-  // applied directly in RtpTransport::DemuxPacket;
-  packet.IdentifyExtensions(recv_rtp_extension_map_);
-  packet.set_payload_type_frequency(kVideoPayloadTypeFrequency);
-  if (!packet.arrival_time().IsFinite()) {
-    packet.set_arrival_time(env_.clock().CurrentTime());
-  }
-
-  call_->Receiver()->DeliverRtpPacket(
-      MediaType::VIDEO, std::move(packet),
-      absl::bind_front(
-          &WebRtcVideoReceiveChannel::MaybeCreateDefaultReceiveStream, this));
-}
-
 void WebRtcVideoReceiveChannel::SetRecordableEncodedFrameCallback(
     uint32_t ssrc,
     std::function<void(const RecordableEncodedFrame&)> callback) {
diff --git a/media/engine/webrtc_video_engine.h b/media/engine/webrtc_video_engine.h
index 759a6e4039..6306b91e4f 100644
--- a/media/engine/webrtc_video_engine.h
+++ b/media/engine/webrtc_video_engine.h
@@ -544,7 +544,7 @@ class WebRtcVideoReceiveChannel : public MediaChannelUtil,
   bool SetSink(uint32_t ssrc, VideoSinkInterface<VideoFrame>* sink) override;
   void SetDefaultSink(VideoSinkInterface<VideoFrame>* sink) override;
   bool GetStats(VideoMediaReceiveInfo* info) override;
-  void OnPacketReceived(const RtpPacketReceived& packet) override;
+  void OnPacketReceived(RtpPacketReceived packet) override;
   bool SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, int delay_ms) override;
 
   std::optional<int> GetBaseMinimumPlayoutDelayMs(uint32_t ssrc) const override;
@@ -584,8 +584,6 @@ class WebRtcVideoReceiveChannel : public MediaChannelUtil,
   WebRtcVideoReceiveStream* FindReceiveStream(uint32_t ssrc)
       RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
 
-  void ProcessReceivedPacket(RtpPacketReceived packet)
-      RTC_RUN_ON(thread_checker_);
 
   // Expected to be invoked once per packet that belongs to this channel that
   // can not be demuxed.
@@ -720,7 +718,7 @@ class WebRtcVideoReceiveChannel : public MediaChannelUtil,
   // Variables.
   const Environment env_;
   TaskQueueBase* const worker_thread_;
-  ScopedTaskSafety task_safety_;
+  scoped_refptr<PendingTaskSafetyFlag> network_thread_safety_;
   RTC_NO_UNIQUE_ADDRESS SequenceChecker network_thread_checker_{
       SequenceChecker::kDetached};
   RTC_NO_UNIQUE_ADDRESS SequenceChecker thread_checker_;
@@ -757,7 +755,8 @@ class WebRtcVideoReceiveChannel : public MediaChannelUtil,
 
   VideoDecoderFactory* const decoder_factory_ RTC_GUARDED_BY(thread_checker_);
   std::vector<VideoCodecSettings> recv_codecs_ RTC_GUARDED_BY(thread_checker_);
-  RtpHeaderExtensionMap recv_rtp_extension_map_ RTC_GUARDED_BY(thread_checker_);
+  RtpHeaderExtensionMap recv_rtp_extension_map_
+      RTC_GUARDED_BY(network_thread_checker_);
   std::vector<RtpExtension> recv_rtp_extensions_
       RTC_GUARDED_BY(thread_checker_);
   // See reason for keeping track of the FlexFEC payload type separately in
diff --git a/media/engine/webrtc_video_engine_unittest.cc b/media/engine/webrtc_video_engine_unittest.cc
index e7b2aa342c..e1f3b7ef4a 100644
--- a/media/engine/webrtc_video_engine_unittest.cc
+++ b/media/engine/webrtc_video_engine_unittest.cc
@@ -122,6 +122,7 @@
 #include "test/gmock.h"
 #include "test/gtest.h"
 #include "test/rtcp_packet_parser.h"
+#include "test/run_loop.h"
 #include "test/time_controller/simulated_time_controller.h"
 #include "video/config/encoder_stream_factory.h"
 #include "video/config/simulcast.h"
@@ -3410,6 +3411,7 @@ TEST_F(WebRtcVideoChannelTest, OnPacketReceivedIdentifiesExtensions) {
   parameters.extensions.push_back(
       RtpExtension(RtpExtension::kVideoRotationUri, /*id=*/1));
   ASSERT_TRUE(receive_channel_->SetReceiverParameters(parameters));
+  time_controller_.AdvanceTime(TimeDelta::Zero());
   RtpHeaderExtensionMap extension_map(parameters.extensions);
   RtpPacketReceived reference_packet(&extension_map);
   reference_packet.SetExtension<VideoOrientation>(
@@ -5333,6 +5335,7 @@ TEST_F(WebRtcVideoChannelTest, DuplicateUlpfecCodecIsDropped) {
   parameters.codecs.push_back(
       CreateVideoCodec(kSecondUlpfecPayloadType, kUlpfecCodecName));
   ASSERT_TRUE(receive_channel_->SetReceiverParameters(parameters));
+  time_controller_.AdvanceTime(TimeDelta::Zero());
 
   FakeVideoReceiveStream* recv_stream = AddRecvStream();
   EXPECT_EQ(kFirstUlpfecPayloadType,
@@ -5350,6 +5353,7 @@ TEST_F(WebRtcVideoChannelTest, DuplicateRedCodecIsDropped) {
   parameters.codecs.push_back(
       CreateVideoCodec(kSecondRedPayloadType, kRedCodecName));
   ASSERT_TRUE(receive_channel_->SetReceiverParameters(parameters));
+  time_controller_.AdvanceTime(TimeDelta::Zero());
 
   FakeVideoReceiveStream* recv_stream = AddRecvStream();
   EXPECT_EQ(kFirstRedPayloadType,
@@ -7665,6 +7669,7 @@ TEST_F(WebRtcVideoChannelTest,
   VideoReceiverParameters parameters;
   parameters.codecs.push_back(GetEngineCodec("VP8"));
   ASSERT_TRUE(receive_channel_->SetReceiverParameters(parameters));
+  time_controller_.AdvanceTime(TimeDelta::Zero());
 
   // No streams signaled and no packets received, so we should not have any
   // stream objects created yet.
@@ -9703,7 +9708,8 @@ TEST_F(WebRtcVideoChannelTest, GenerateKeyFrameSimulcast) {
 class WebRtcVideoChannelSimulcastTest : public ::testing::Test {
  public:
   WebRtcVideoChannelSimulcastTest()
-      : env_(CreateTestEnvironment()),
+      : run_loop_(),
+        env_(CreateTestEnvironment()),
         fake_call_(env_),
         encoder_factory_(new FakeWebRtcVideoEncoderFactory),
         decoder_factory_(new FakeWebRtcVideoDecoderFactory),
@@ -9869,6 +9875,7 @@ class WebRtcVideoChannelSimulcastTest : public ::testing::Test {
     return streams[streams.size() - 1];
   }
 
+  test::RunLoop run_loop_;
   Environment env_;
   FakeCall fake_call_;
   FakeWebRtcVideoEncoderFactory* encoder_factory_;
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index 766205203a..dd04d4ec7a 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -2147,6 +2147,9 @@ WebRtcVoiceReceiveChannel::WebRtcVoiceReceiveChannel(
     : MediaChannelUtil(call->network_thread(), config.enable_dscp),
       env_(env),
       worker_thread_(call->worker_thread()),
+      network_thread_safety_(PendingTaskSafetyFlag::CreateAttachedToTaskQueue(
+          /*alive=*/true,
+          call->network_thread())),
       engine_(engine),
       call_(call),
       audio_config_(config.audio),
@@ -2190,7 +2193,13 @@ bool WebRtcVoiceReceiveChannel::SetReceiverParameters(
                           false, env_.field_trials());
   if (recv_rtp_extensions_ != filtered_extensions) {
     recv_rtp_extensions_.swap(filtered_extensions);
-    recv_rtp_extension_map_ = RtpHeaderExtensionMap(recv_rtp_extensions_);
+    call_->network_thread()->PostTask(SafeTask(
+        network_thread_safety_,
+        [this, recv_rtp_extension_map =
+                   RtpHeaderExtensionMap(recv_rtp_extensions_)]() mutable {
+          RTC_DCHECK_RUN_ON(&network_thread_checker_);
+          recv_rtp_extension_map_ = std::move(recv_rtp_extension_map);
+        }));
   }
   // RTCP mode, NACK, and receive-side RTT are not configured here because they
   // enable send functionality in the receive channels. This functionality is
@@ -2591,8 +2600,7 @@ void WebRtcVoiceReceiveChannel::SetFrameDecryptor(
   }
 }
 
-void WebRtcVoiceReceiveChannel::OnPacketReceived(
-    const RtpPacketReceived& packet) {
+void WebRtcVoiceReceiveChannel::OnPacketReceived(RtpPacketReceived packet) {
   RTC_DCHECK_RUN_ON(&network_thread_checker_);
 
   // TODO(bugs.webrtc.org/11993): This code is very similar to what
@@ -2601,27 +2609,26 @@ void WebRtcVoiceReceiveChannel::OnPacketReceived(
   // call_->Receiver() to a common implementation and provide a callback on
   // the worker thread for the exception case (DELIVERY_UNKNOWN_SSRC) and
   // how retry is attempted.
-  worker_thread_->PostTask(
-      SafeTask(task_safety_.flag(), [this, packet = packet]() mutable {
-        RTC_DCHECK_RUN_ON(worker_thread_);
 
-        // TODO(bugs.webrtc.org/7135): extensions in `packet` is currently set
-        // in RtpTransport and does not necessarily include extensions specific
-        // to this channel/MID. Also see comment in
-        // BaseChannel::MaybeUpdateDemuxerAndRtpExtensions_w.
-        // It would likely be good if extensions where merged per BUNDLE and
-        // applied directly in RtpTransport::DemuxPacket;
-        packet.IdentifyExtensions(recv_rtp_extension_map_);
-        if (!packet.arrival_time().IsFinite()) {
-          packet.set_arrival_time(env_.clock().CurrentTime());
-        }
+  // TODO(bugs.webrtc.org/7135): The mapping of extension ID (on-packet
+  // representation) to C++ class object (internal representation) is currently
+  // associated with the per-channel object (e.g. WebRtcVoiceReceiveChannel).
+  // However, this mapping is more logically associated with per-media-stream
+  // objects (like ModuleRtpRtcpImpl2), which handle specific streams flowing
+  // in one direction (usually a single SSRC).
+  // This mapping should be moved out of the media engines altogether, storing
+  // it instead at the stream level initialized by the Transport. RTP packets
+  // consumed by media channels would thus have header extensions distinguished
+  // by C++ class rather than negotiated header extension ID.
+  packet.IdentifyExtensions(recv_rtp_extension_map_);
+  if (!packet.arrival_time().IsFinite()) {
+    packet.set_arrival_time(env_.clock().CurrentTime());
+  }
 
-        call_->Receiver()->DeliverRtpPacket(
-            MediaType::AUDIO, std::move(packet),
-            absl::bind_front(
-                &WebRtcVoiceReceiveChannel::MaybeCreateDefaultReceiveStream,
-                this));
-      }));
+  call_->Receiver()->DeliverRtpPacket(
+      MediaType::AUDIO, std::move(packet),
+      absl::bind_front(
+          &WebRtcVoiceReceiveChannel::MaybeCreateDefaultReceiveStream, this));
 }
 
 bool WebRtcVoiceReceiveChannel::MaybeCreateDefaultReceiveStream(
diff --git a/media/engine/webrtc_voice_engine.h b/media/engine/webrtc_voice_engine.h
index 0a7b5eb2a5..c4facf5890 100644
--- a/media/engine/webrtc_voice_engine.h
+++ b/media/engine/webrtc_voice_engine.h
@@ -344,6 +344,9 @@ class WebRtcVoiceReceiveChannel final
   const AudioOptions& options() const { return options_; }
 
   void SetInterface(MediaChannelNetworkInterface* iface) override {
+    RTC_DCHECK_RUN_ON(&network_thread_checker_);
+    iface ? network_thread_safety_->SetAlive()
+          : network_thread_safety_->SetNotAlive();
     MediaChannelUtil::SetInterface(iface);
   }
   bool SetReceiverParameters(const AudioReceiverParameters& params) override;
@@ -374,7 +377,7 @@ class WebRtcVoiceReceiveChannel final
   bool SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, int delay_ms) override;
   std::optional<int> GetBaseMinimumPlayoutDelayMs(uint32_t ssrc) const override;
 
-  void OnPacketReceived(const RtpPacketReceived& packet) override;
+  void OnPacketReceived(RtpPacketReceived packet) override;
   bool GetStats(VoiceMediaReceiveInfo* info,
                 bool get_and_clear_legacy_stats) override;
 
@@ -418,6 +421,7 @@ class WebRtcVoiceReceiveChannel final
   const Environment env_;
   TaskQueueBase* const worker_thread_;
   ScopedTaskSafety task_safety_;
+  scoped_refptr<PendingTaskSafetyFlag> network_thread_safety_;
   SequenceChecker network_thread_checker_{SequenceChecker::kDetached};
 
   WebRtcVoiceEngine* const engine_ = nullptr;
@@ -461,7 +465,8 @@ class WebRtcVoiceReceiveChannel final
   std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_
       RTC_GUARDED_BY(worker_thread_);
   std::vector<RtpExtension> recv_rtp_extensions_ RTC_GUARDED_BY(worker_thread_);
-  RtpHeaderExtensionMap recv_rtp_extension_map_ RTC_GUARDED_BY(worker_thread_);
+  RtpHeaderExtensionMap recv_rtp_extension_map_
+      RTC_GUARDED_BY(network_thread_checker_);
 
   std::optional<AudioSendStream::Config::SendCodecSpec> send_codec_spec_
       RTC_GUARDED_BY(worker_thread_);
diff --git a/media/engine/webrtc_voice_engine_unittest.cc b/media/engine/webrtc_voice_engine_unittest.cc
index 78e4f2ae78..8329713ca1 100644
--- a/media/engine/webrtc_voice_engine_unittest.cc
+++ b/media/engine/webrtc_voice_engine_unittest.cc
@@ -1598,6 +1598,7 @@ TEST_P(WebRtcVoiceEngineTestFake, OnPacketReceivedIdentifiesExtensions) {
   parameters.extensions.push_back(
       RtpExtension(RtpExtension::kAudioLevelUri, /*id=*/1));
   ASSERT_TRUE(receive_channel_->SetReceiverParameters(parameters));
+  run_loop_.Flush();
   RtpHeaderExtensionMap extension_map(parameters.extensions);
   RtpPacketReceived reference_packet(&extension_map);
   constexpr uint8_t kAudioLevel = 123;
diff --git a/pc/test/mock_voice_media_receive_channel_interface.h b/pc/test/mock_voice_media_receive_channel_interface.h
index b38405e258..1c1f5e3211 100644
--- a/pc/test/mock_voice_media_receive_channel_interface.h
+++ b/pc/test/mock_voice_media_receive_channel_interface.h
@@ -100,10 +100,7 @@ class MockVoiceMediaReceiveChannelInterface
               SetInterface,
               (webrtc::MediaChannelNetworkInterface * iface),
               (override));
-  MOCK_METHOD(void,
-              OnPacketReceived,
-              (const RtpPacketReceived& packet),
-              (override));
+  MOCK_METHOD(void, OnPacketReceived, (RtpPacketReceived packet), (override));
   MOCK_METHOD(std::optional<uint32_t>,
               GetUnsignaledSsrc,
               (),

From 47efaeafa6ebe589e240054f506886adcb5040b3 Mon Sep 17 00:00:00 2001
From: Jan Grulich <grulja@gmail.com>
Date: Mon, 2 Mar 2026 14:59:32 +0100
Subject: [PATCH 174/472] PipeWire capture: disconnect stream callbacks before
 destroying it

Disconnect PipeWire stream callbacks before destroying it to prevent
OnStreamParamChanged() callback from being invoked during destruction.
pw_stream_destroy() can trigger the callback on the PipeWire thread
while StopCapture() runs on the API thread, causing
RTC_CHECK_RUNS_SERIALIZED to detect cross-thread access violation.

Bug: webrtc:487161137
Change-Id: Ib53054922579c60d793a6cc8dcd7c42e316cb757
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/448042
Reviewed-by: Andreas Pehrson <apehrson@mozilla.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Cr-Commit-Position: refs/heads/main@{#47027}
---
 modules/video_capture/linux/video_capture_pipewire.cc | 4 ++++
 modules/video_capture/linux/video_capture_pipewire.h  | 2 +-
 2 files changed, 5 insertions(+), 1 deletion(-)

diff --git a/modules/video_capture/linux/video_capture_pipewire.cc b/modules/video_capture/linux/video_capture_pipewire.cc
index 5ae2b6f565..58021f9f69 100644
--- a/modules/video_capture/linux/video_capture_pipewire.cc
+++ b/modules/video_capture/linux/video_capture_pipewire.cc
@@ -254,11 +254,15 @@ int32_t VideoCaptureModulePipeWire::StopCapture() {
   RTC_DCHECK_RUN_ON(&api_checker_);
 
   PipeWireThreadLoopLock thread_loop_lock(session_->pw_main_loop_);
+
   // PipeWireSession is guarded by API checker so just make sure we do
   // race detection when the PipeWire loop is locked/stopped to not run
   // any callback at this point.
   RTC_CHECK_RUNS_SERIALIZED(&capture_checker_);
   if (stream_) {
+    // Removing the listener first guarantees no callbacks will fire after this
+    // point.
+    spa_hook_remove(&stream_listener_);
     pw_stream_destroy(stream_);
     stream_ = nullptr;
   }
diff --git a/modules/video_capture/linux/video_capture_pipewire.h b/modules/video_capture/linux/video_capture_pipewire.h
index e285573035..9be497940a 100644
--- a/modules/video_capture/linux/video_capture_pipewire.h
+++ b/modules/video_capture/linux/video_capture_pipewire.h
@@ -63,7 +63,7 @@ class VideoCaptureModulePipeWire : public VideoCaptureImpl {
       RTC_GUARDED_BY(capture_checker_);
 
   struct pw_stream* stream_ RTC_GUARDED_BY(capture_checker_) = nullptr;
-  struct spa_hook stream_listener_ RTC_GUARDED_BY(capture_checker_);
+  struct spa_hook stream_listener_ RTC_GUARDED_BY(api_checker_) = {};
 };
 }  // namespace videocapturemodule
 }  // namespace webrtc

From b88b9f23009483df52db141956ebd33d9cc8c181 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Mon, 2 Mar 2026 12:03:52 -0800
Subject: [PATCH 175/472] Roll chromium_revision faab630875..460b91592c
 (1592487:1592626)

Change log: https://chromium.googlesource.com/chromium/src/+log/faab630875..460b91592c
Full diff: https://chromium.googlesource.com/chromium/src/+/faab630875..460b91592c

Changed dependencies
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/e72cef5d25..79bcc4f8dd
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/747c2fa02b..0371d548fc
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/64fce8e53a..d25c43455e
* src/third_party/fuzztest/src: https://chromium.googlesource.com/external/github.com/google/fuzztest.git/+log/f67a6fe9ca..6a217df9ed
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/034b681167..73b070473d
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/a381c66832..88a9f66c07
DEPS diff: https://chromium.googlesource.com/chromium/src/+/faab630875..460b91592c/DEPS

BUG=None

Change-Id: I0ba09d4f248a92e9b05594f163f790f73d6056aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453080
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47028}
---
 DEPS | 14 +++++++-------
 1 file changed, 7 insertions(+), 7 deletions(-)

diff --git a/DEPS b/DEPS
index 57bb580ac8..69a631ceba 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': 'faab630875af0fdba81c9ce794f28b87532adfea',
+  'chromium_revision': '460b91592c227585c769c7154f0cb643605d44c2',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@e72cef5d25de4912a4380b6903a1def08d81aacf',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@79bcc4f8dd01e209ee6b9a096275c900b3e8c0be',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@747c2fa02b992061c8b773a3927de086fb2378b9',
+    'https://chromium.googlesource.com/chromium/src/testing@0371d548fc3ec99bf6c6ff889c9afe488c3cde1b',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@64fce8e53a91fdae0d404332980555ca6514cf25',
+    'https://chromium.googlesource.com/chromium/src/third_party@d25c43455e0c50b53b799f5c4bf2a4867f3dbfb7',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -511,7 +511,7 @@ deps = {
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@034b6811670e7edf0694bf21f4e2ad44f497b545',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@73b070473d87f8943e67deddd8d3fc4722ed18aa',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -593,7 +593,7 @@ deps = {
   'src/third_party/libFuzzer/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt/lib/fuzzer.git@bea408a6e01f0f7e6c82a43121fe3af4506c932e',
   'src/third_party/fuzztest/src':
-    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@f67a6fe9ca3afc4d71997e75e8d87c571f0c30cb',
+    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@6a217df9edbe5705885457f10968ebe80aec9504',
   'src/third_party/libjpeg_turbo':
     'https://chromium.googlesource.com/chromium/deps/libjpeg_turbo.git@d1f5f2393e0d51f840207342ae86e55a86443288',
   'src/third_party/libsrtp':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@a381c6683255f0d7d80d94f4bd72f22199b27ef2',
+    'https://chromium.googlesource.com/chromium/src/tools@88a9f66c07407b3f53761aff51625cdb60b8f2e0',
 
   'src/third_party/espresso': {
       'packages': [

From fe9e67d3620cc005c1bbe686b4d114bd8d7c877b Mon Sep 17 00:00:00 2001
From: Jeremy Leconte <jleconte@webrtc.org>
Date: Mon, 2 Mar 2026 12:00:18 +0100
Subject: [PATCH 176/472] Allow ci scheduler to retry build on cron and perf
 bucket.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Change-Id: I81d28314cc47e2ba93a8fa9cc9f507370b6841b4
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452941
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#47029}
---
 infra/config/config.star               | 6 ++++++
 infra/config/generated/luci/realms.cfg | 5 +++++
 2 files changed, 11 insertions(+)

diff --git a/infra/config/config.star b/infra/config/config.star
index 7204e1b5e7..49718cf05f 100755
--- a/infra/config/config.star
+++ b/infra/config/config.star
@@ -308,6 +308,7 @@ luci.bucket(
             "webrtc-ci-builder@chops-service-accounts.iam.gserviceaccount.com",
         ]),
         acl.entry(acl.BUILDBUCKET_TRIGGERER, groups = [
+            "project-webrtc-ci-schedulers",
             # Allow Pinpoint to trigger builds for bisection
             "service-account-chromeperf",
         ]),
@@ -320,6 +321,11 @@ luci.bucket(
 
 luci.bucket(
     name = "cron",
+    acls = [
+        acl.entry(acl.BUILDBUCKET_TRIGGERER, groups = [
+            "project-webrtc-ci-schedulers",
+        ]),
+    ],
 )
 
 # Commit queue definitions:
diff --git a/infra/config/generated/luci/realms.cfg b/infra/config/generated/luci/realms.cfg
index 34d7f60948..7540cb1312 100644
--- a/infra/config/generated/luci/realms.cfg
+++ b/infra/config/generated/luci/realms.cfg
@@ -82,6 +82,10 @@ realms {
     principals: "user:chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com"
     principals: "user:webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com"
   }
+  bindings {
+    role: "role/buildbucket.triggerer"
+    principals: "group:project-webrtc-ci-schedulers"
+  }
 }
 realms {
   name: "debug-bot-acls"
@@ -111,6 +115,7 @@ realms {
   }
   bindings {
     role: "role/buildbucket.triggerer"
+    principals: "group:project-webrtc-ci-schedulers"
     principals: "group:service-account-chromeperf"
     principals: "user:webrtc-ci-builder@chops-service-accounts.iam.gserviceaccount.com"
   }

From 7ec1333a0b9fa55178c3fa7835a8924bc826f8b5 Mon Sep 17 00:00:00 2001
From: Jeremy Leconte <jleconte@webrtc.org>
Date: Tue, 3 Mar 2026 15:12:34 +0100
Subject: [PATCH 177/472] Disable android_chromium_compile from CQ.

Change-Id: Iaa9095b5a7f2cce02a59590a29e13cb0c06fa620
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453261
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47030}
---
 infra/config/builders.star                   | 4 +++-
 infra/config/generated/luci/commit-queue.cfg | 3 ---
 2 files changed, 3 insertions(+), 4 deletions(-)

diff --git a/infra/config/builders.star b/infra/config/builders.star
index 8816ef06fa..9e9a92fac0 100644
--- a/infra/config/builders.star
+++ b/infra/config/builders.star
@@ -332,7 +332,9 @@ ci_builder("Android32 Builder x86", "Android|x86|rel")
 try_builder("android_compile_x86_rel")
 ci_builder("Android32 (more configs)", "Android|arm|more")
 try_builder("android_arm_more_configs")
-chromium_try_builder("android_chromium_compile")
+
+# TODO(jleconte): re-enable when the issue is fixed in Chromium.
+chromium_try_builder("android_chromium_compile", cq = None)
 
 ios_builder("iOS64 Debug", "iOS|arm64|dbg")
 ios_try_job("ios_compile_arm64_dbg")
diff --git a/infra/config/generated/luci/commit-queue.cfg b/infra/config/generated/luci/commit-queue.cfg
index b7c3a0c960..48d7850133 100644
--- a/infra/config/generated/luci/commit-queue.cfg
+++ b/infra/config/generated/luci/commit-queue.cfg
@@ -56,9 +56,6 @@ config_groups {
       builders {
         name: "webrtc/try/android_arm_rel"
       }
-      builders {
-        name: "webrtc/try/android_chromium_compile"
-      }
       builders {
         name: "webrtc/try/android_compile_arm64_rel"
       }

From cc65e24a638ba40d8edf968e99a257734e8a68a8 Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Tue, 3 Mar 2026 14:16:34 +0100
Subject: [PATCH 178/472] Refactor webrtc fuzzer api

Instead of feeding each fuzzer with pointer + size, provide FuzzDataHelper
class to simplify and unify producing structured fuzz data

Bug: webrtc:42225170
Change-Id: Ide0d02450c685a33c733952a5114a87a20bdbbc6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/411501
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47031}
---
 test/fuzzers/BUILD.gn                         | 29 ++++++++---
 test/fuzzers/aec3_config_json_fuzzer.cc       | 10 ++--
 test/fuzzers/aec3_fuzzer.cc                   |  9 ++--
 test/fuzzers/agc_fuzzer.cc                    | 12 ++---
 test/fuzzers/audio_decoder_fuzzer.cc          | 10 ++--
 test/fuzzers/audio_decoder_fuzzer.h           |  5 +-
 test/fuzzers/audio_decoder_g722_fuzzer.cc     |  9 ++--
 .../audio_decoder_multistream_opus_fuzzer.cc  | 12 ++---
 test/fuzzers/audio_decoder_opus_fuzzer.cc     |  7 +--
 .../audio_decoder_opus_redundant_fuzzer.cc    |  7 +--
 test/fuzzers/audio_decoder_pcm16b_fuzzer.cc   | 15 +++---
 test/fuzzers/audio_decoder_pcm_fuzzer.cc      | 15 +++---
 test/fuzzers/audio_encoder_fuzzer.cc          |  5 +-
 test/fuzzers/audio_encoder_fuzzer.h           |  5 +-
 test/fuzzers/audio_encoder_opus_fuzzer.cc     |  6 +--
 .../audio_processing_configs_fuzzer.cc        | 12 ++---
 .../fuzzers/audio_processing_fuzzer_helper.cc |  6 +--
 test/fuzzers/audio_processing_fuzzer_helper.h |  2 +-
 .../audio_processing_sample_rate_fuzzer.cc    | 10 ++--
 test/fuzzers/base64_decode_fuzzer.cc          |  8 ++-
 test/fuzzers/base64_encode_fuzzer.cc          |  9 ++--
 test/fuzzers/comfort_noise_decoder_fuzzer.cc  |  6 +--
 test/fuzzers/dcsctp_packet_fuzzer.cc          |  9 ++--
 test/fuzzers/dcsctp_socket_fuzzer.cc          |  8 ++-
 test/fuzzers/field_trial_fuzzer.cc            |  8 ++-
 test/fuzzers/flexfec_header_reader_fuzzer.cc  | 10 ++--
 test/fuzzers/flexfec_receiver_fuzzer.cc       | 49 ++++++++-----------
 test/fuzzers/flexfec_sender_fuzzer.cc         | 36 +++++++-------
 .../forward_error_correction_fuzzer.cc        |  7 +--
 test/fuzzers/frame_buffer_fuzzer.cc           | 21 ++++----
 test/fuzzers/fuzz_data_helper.cc              | 23 ---------
 test/fuzzers/fuzz_data_helper.h               | 27 +++++++---
 test/fuzzers/h264_bitstream_parser_fuzzer.cc  |  7 ++-
 test/fuzzers/h264_depacketizer_fuzzer.cc      |  8 +--
 test/fuzzers/h265_bitstream_parser_fuzzer.cc  |  7 ++-
 test/fuzzers/h265_depacketizer_fuzzer.cc      |  8 +--
 test/fuzzers/neteq_rtp_fuzzer.cc              | 12 ++---
 test/fuzzers/neteq_signal_fuzzer.cc           | 10 ++--
 ...ceive_side_congestion_controller_fuzzer.cc | 20 +++-----
 test/fuzzers/residual_echo_detector_fuzzer.cc | 18 +++----
 test/fuzzers/rtcp_receiver_fuzzer.cc          |  7 +--
 ..._depacketizer_av1_assemble_frame_fuzzer.cc | 13 +++--
 .../rtp_dependency_descriptor_fuzzer.cc       |  3 +-
 test/fuzzers/rtp_format_h264_fuzzer.cc        | 13 ++---
 test/fuzzers/rtp_format_vp8_fuzzer.cc         | 12 ++---
 test/fuzzers/rtp_format_vp9_fuzzer.cc         | 12 ++---
 .../rtp_frame_reference_finder_fuzzer.cc      | 37 +++++---------
 test/fuzzers/rtp_packet_fuzzer.cc             | 11 ++---
 test/fuzzers/rtp_packetizer_av1_fuzzer.cc     | 12 ++---
 test/fuzzers/rtp_packetizer_h265_fuzzer.cc    | 11 ++---
 .../rtp_video_frame_assembler_fuzzer.cc       | 15 +++---
 .../rtp_video_layers_allocation_fuzzer.cc     |  7 +--
 test/fuzzers/sctp_utils_fuzzer.cc             |  5 +-
 test/fuzzers/sdp_integration_fuzzer.cc        |  8 +--
 test/fuzzers/sdp_parser_fuzzer.cc             |  7 +--
 test/fuzzers/ssl_certificate_fuzzer.cc        |  7 ++-
 test/fuzzers/string_to_number_fuzzer.cc       |  8 +--
 test/fuzzers/stun_parser_fuzzer.cc            |  6 +--
 test/fuzzers/stun_validator_fuzzer.cc         | 11 +++--
 test/fuzzers/turn_unwrap_fuzzer.cc            |  8 ++-
 test/fuzzers/ulpfec_generator_fuzzer.cc       | 29 +++++------
 test/fuzzers/ulpfec_header_reader_fuzzer.cc   | 10 ++--
 test/fuzzers/ulpfec_receiver_fuzzer.cc        | 14 +++---
 test/fuzzers/utils/BUILD.gn                   |  1 +
 test/fuzzers/utils/validate_rtp_packetizer.cc |  5 +-
 test/fuzzers/utils/validate_rtp_packetizer.h  |  2 +-
 test/fuzzers/vp8_depacketizer_fuzzer.cc       |  7 ++-
 test/fuzzers/vp8_qp_parser_fuzzer.cc          |  7 ++-
 test/fuzzers/vp8_replay_fuzzer.cc             |  7 ++-
 test/fuzzers/vp9_depacketizer_fuzzer.cc       |  7 ++-
 test/fuzzers/vp9_encoder_references_fuzzer.cc | 14 ++----
 test/fuzzers/vp9_qp_parser_fuzzer.cc          |  7 ++-
 test/fuzzers/vp9_replay_fuzzer.cc             |  7 ++-
 test/fuzzers/webrtc_fuzzer_main.cc            |  8 +--
 test/fuzzers/windowed_min_filter_fuzzer.cc    |  7 +--
 75 files changed, 394 insertions(+), 448 deletions(-)
 delete mode 100644 test/fuzzers/fuzz_data_helper.cc

diff --git a/test/fuzzers/BUILD.gn b/test/fuzzers/BUILD.gn
index f05995da70..f44ec6acc1 100644
--- a/test/fuzzers/BUILD.gn
+++ b/test/fuzzers/BUILD.gn
@@ -14,6 +14,8 @@ rtc_library("webrtc_fuzzer_main") {
   sources = [ "webrtc_fuzzer_main.cc" ]
   testonly = true
   deps = [
+    ":fuzz_data_helper",
+    "../../api:array_view",
     "../../rtc_base:logging",
     "//testing/libfuzzer:fuzzing_engine_main",
   ]
@@ -30,14 +32,11 @@ rtc_library("webrtc_fuzzer_main") {
 
 rtc_library("fuzz_data_helper") {
   testonly = true
-  sources = [
-    "fuzz_data_helper.cc",
-    "fuzz_data_helper.h",
-  ]
+  sources = [ "fuzz_data_helper.h" ]
   deps = [
-    "../../api:array_view",
     "../../modules/rtp_rtcp:rtp_rtcp_format",
     "../../rtc_base:checks",
+    "//third_party/abseil-cpp/absl/strings:string_view",
   ]
 
   # Only fuzzer targets can depend on this.
@@ -113,6 +112,7 @@ webrtc_fuzzer_test("vp9_depacketizer_fuzzer") {
 webrtc_fuzzer_test("vp8_qp_parser_fuzzer") {
   sources = [ "vp8_qp_parser_fuzzer.cc" ]
   deps = [
+    "../../api:array_view",
     "../../modules/video_coding:video_coding_utility",
     "../../modules/video_coding/",
   ]
@@ -170,6 +170,7 @@ webrtc_fuzzer_test("forward_error_correction_fuzzer") {
 webrtc_fuzzer_test("flexfec_header_reader_fuzzer") {
   sources = [ "flexfec_header_reader_fuzzer.cc" ]
   deps = [
+    "../../api:array_view",
     "../../api:scoped_refptr",
     "../../modules/rtp_rtcp",
     "../../modules/rtp_rtcp:rtp_rtcp_format",
@@ -192,6 +193,7 @@ webrtc_fuzzer_test("flexfec_sender_fuzzer") {
 webrtc_fuzzer_test("ulpfec_header_reader_fuzzer") {
   sources = [ "ulpfec_header_reader_fuzzer.cc" ]
   deps = [
+    "../../api:array_view",
     "../../api:scoped_refptr",
     "../../modules/rtp_rtcp",
     "../../modules/rtp_rtcp:fec_test_helper",
@@ -343,6 +345,8 @@ rtc_library("audio_decoder_fuzzer") {
     "audio_decoder_fuzzer.h",
   ]
   deps = [
+    ":fuzz_data_helper",
+    "../../api:array_view",
     "../../api/audio_codecs:audio_codecs_api",
     "../../modules/rtp_rtcp:rtp_rtcp_format",
     "../../rtc_base:checks",
@@ -422,6 +426,7 @@ webrtc_fuzzer_test("audio_encoder_opus_fuzzer") {
   sources = [ "audio_encoder_opus_fuzzer.cc" ]
   deps = [
     ":audio_encoder_fuzzer",
+    ":fuzz_data_helper",
     "../../api/audio_codecs/opus:audio_encoder_opus",
     "../../api/environment",
     "../../api/environment:environment_factory",
@@ -432,6 +437,7 @@ webrtc_fuzzer_test("audio_encoder_opus_fuzzer") {
 webrtc_fuzzer_test("turn_unwrap_fuzzer") {
   sources = [ "turn_unwrap_fuzzer.cc" ]
   deps = [
+    "../../api:array_view",
     "../../media",
     "../../media:turn_utils",
   ]
@@ -531,7 +537,10 @@ webrtc_fuzzer_test("stun_parser_fuzzer") {
 
 webrtc_fuzzer_test("stun_validator_fuzzer") {
   sources = [ "stun_validator_fuzzer.cc" ]
-  deps = [ "../../api/transport:stun_types" ]
+  deps = [
+    "../../api/transport:stun_types",
+    "//third_party/abseil-cpp/absl/strings:string_view",
+  ]
   seed_corpus = "corpora/stun-corpus"
   dict = "corpora/stun.tokens"
 }
@@ -615,6 +624,7 @@ webrtc_fuzzer_test("aec3_config_json_fuzzer") {
     ":fuzz_data_helper",
     "../../api/audio:aec3_config",
     "../../modules/audio_processing:aec3_config_json",
+    "//third_party/abseil-cpp/absl/strings:string_view",
   ]
   dict = "//testing/libfuzzer/fuzzers/dicts/json.dict"
   seed_corpus = "corpora/aec3-config-json-corpus"
@@ -725,7 +735,10 @@ webrtc_fuzzer_test("field_trial_fuzzer") {
 
 webrtc_fuzzer_test("string_to_number_fuzzer") {
   sources = [ "string_to_number_fuzzer.cc" ]
-  deps = [ "../../rtc_base:stringutils" ]
+  deps = [
+    "../../rtc_base:stringutils",
+    "//third_party/abseil-cpp/absl/strings:string_view",
+  ]
   seed_corpus = "corpora/string_to_number-corpus"
 }
 
@@ -798,7 +811,6 @@ if (rtc_build_libvpx) {
       "//third_party/abseil-cpp/absl/algorithm:container",
       "//third_party/abseil-cpp/absl/container:inlined_vector",
       "//third_party/abseil-cpp/absl/strings:string_view",
-      rtc_libvpx_dir,
     ]
     seed_corpus = "corpora/vp9-encoder-references-corpus"
     defines = [ "RTC_ENABLE_VP9" ]
@@ -808,6 +820,7 @@ if (rtc_build_libvpx) {
 webrtc_fuzzer_test("vp9_replay_fuzzer") {
   sources = [ "vp9_replay_fuzzer.cc" ]
   deps = [
+    "../../api:array_view",
     "../../api/video_codecs:video_codecs_api",
     "../../call:video_receive_stream_api",
     "utils:rtp_replayer",
diff --git a/test/fuzzers/aec3_config_json_fuzzer.cc b/test/fuzzers/aec3_config_json_fuzzer.cc
index a42e3fa89d..0eb19e9c1c 100644
--- a/test/fuzzers/aec3_config_json_fuzzer.cc
+++ b/test/fuzzers/aec3_config_json_fuzzer.cc
@@ -9,18 +9,18 @@
  */
 
 #include <cstddef>
-#include <cstdint>
-#include <string>
 
+#include "absl/strings/string_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "modules/audio_processing/test/echo_canceller3_config_json.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  if (size > 10000) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
+  if (fuzz_data.size() > 10'000) {
     return;
   }
-  std::string config_json(reinterpret_cast<const char*>(data), size);
+  absl::string_view config_json = fuzz_data.ReadString();
 
   EchoCanceller3Config config;
   bool success;
diff --git a/test/fuzzers/aec3_fuzzer.cc b/test/fuzzers/aec3_fuzzer.cc
index fde7d56f61..7435f70ad9 100644
--- a/test/fuzzers/aec3_fuzzer.cc
+++ b/test/fuzzers/aec3_fuzzer.cc
@@ -12,7 +12,6 @@
 #include <cstdint>
 #include <optional>
 
-#include "api/array_view.h"
 #include "api/audio/audio_processing.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "api/environment/environment_factory.h"
@@ -25,7 +24,7 @@ namespace {
 using SampleRate = ::webrtc::AudioProcessing::NativeRate;
 
 void PrepareAudioBuffer(int sample_rate_hz,
-                        test::FuzzDataHelper* fuzz_data,
+                        FuzzDataHelper* fuzz_data,
                         AudioBuffer* buffer) {
   float* const* channels = buffer->channels_f();
   for (size_t i = 0; i < buffer->num_channels(); ++i) {
@@ -41,13 +40,11 @@ void PrepareAudioBuffer(int sample_rate_hz,
 
 }  // namespace
 
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  if (size > 200000) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
+  if (fuzz_data.size() > 200'000) {
     return;
   }
 
-  test::FuzzDataHelper fuzz_data(webrtc::ArrayView<const uint8_t>(data, size));
-
   constexpr int kSampleRates[] = {16000, 32000, 48000};
   const int sample_rate_hz =
       static_cast<size_t>(fuzz_data.SelectOneOf(kSampleRates));
diff --git a/test/fuzzers/agc_fuzzer.cc b/test/fuzzers/agc_fuzzer.cc
index cf436255ae..7b6aca1198 100644
--- a/test/fuzzers/agc_fuzzer.cc
+++ b/test/fuzzers/agc_fuzzer.cc
@@ -25,7 +25,7 @@ namespace webrtc {
 namespace {
 
 void FillAudioBuffer(size_t sample_rate_hz,
-                     test::FuzzDataHelper* fuzz_data,
+                     FuzzDataHelper* fuzz_data,
                      AudioBuffer* buffer) {
   float* const* channels = buffer->channels_f();
   for (size_t i = 0; i < buffer->num_channels(); ++i) {
@@ -42,8 +42,7 @@ void FillAudioBuffer(size_t sample_rate_hz,
 
 // This function calls the GainControl functions that are overriden as private
 // in GainControlInterface.
-void FuzzGainControllerConfig(test::FuzzDataHelper* fuzz_data,
-                              GainControl* gc) {
+void FuzzGainControllerConfig(FuzzDataHelper* fuzz_data, GainControl* gc) {
   GainControl::Mode modes[] = {GainControl::Mode::kAdaptiveAnalog,
                                GainControl::Mode::kAdaptiveDigital,
                                GainControl::Mode::kFixedDigital};
@@ -82,7 +81,7 @@ void FuzzGainControllerConfig(test::FuzzDataHelper* fuzz_data,
   static_cast<void>(gc->is_limiter_enabled());
 }
 
-void FuzzGainController(test::FuzzDataHelper* fuzz_data, GainControlImpl* gci) {
+void FuzzGainController(FuzzDataHelper* fuzz_data, GainControlImpl* gci) {
   using Rate = ::webrtc::AudioProcessing::NativeRate;
   const Rate rate_kinds[] = {Rate::kSampleRate16kHz, Rate::kSampleRate32kHz,
                              Rate::kSampleRate48kHz};
@@ -117,11 +116,10 @@ void FuzzGainController(test::FuzzDataHelper* fuzz_data, GainControlImpl* gci) {
 
 }  // namespace
 
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  if (size > 200000) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
+  if (fuzz_data.size() > 200'000) {
     return;
   }
-  test::FuzzDataHelper fuzz_data(webrtc::ArrayView<const uint8_t>(data, size));
   auto gci = std::make_unique<GainControlImpl>();
   FuzzGainController(&fuzz_data, gci.get());
 }
diff --git a/test/fuzzers/audio_decoder_fuzzer.cc b/test/fuzzers/audio_decoder_fuzzer.cc
index 3da93721d7..84d84171db 100644
--- a/test/fuzzers/audio_decoder_fuzzer.cc
+++ b/test/fuzzers/audio_decoder_fuzzer.cc
@@ -14,8 +14,10 @@
 #include <cstdint>
 #include <limits>
 
+#include "api/array_view.h"
 #include "api/audio_codecs/audio_decoder.h"
 #include "modules/rtp_rtcp/source/byte_io.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
 namespace {
@@ -42,14 +44,14 @@ bool ParseInt(const uint8_t** data, size_t* remaining_size, T* value) {
 // AudioDecoder::DecodeRedundant is used, depending on the value of
 // `decode_type`.
 void FuzzAudioDecoder(DecoderFunctionType decode_type,
-                      const uint8_t* data,
-                      size_t size,
+                      FuzzDataHelper& fuzz_data,
                       AudioDecoder* decoder,
                       int sample_rate_hz,
                       size_t max_decoded_bytes,
                       int16_t* decoded) {
-  const uint8_t* data_ptr = data;
-  size_t remaining_size = size;
+  ArrayView<const uint8_t> data = fuzz_data.ReadRemaining();
+  const uint8_t* data_ptr = data.data();
+  size_t remaining_size = data.size();
   size_t packet_len;
   constexpr size_t kMaxNumFuzzedPackets = 200;
   for (size_t num_packets = 0; num_packets < kMaxNumFuzzedPackets;
diff --git a/test/fuzzers/audio_decoder_fuzzer.h b/test/fuzzers/audio_decoder_fuzzer.h
index 8c5246bb5e..b9bbc14c14 100644
--- a/test/fuzzers/audio_decoder_fuzzer.h
+++ b/test/fuzzers/audio_decoder_fuzzer.h
@@ -14,6 +14,8 @@
 #include <stddef.h>
 #include <stdint.h>
 
+#include "test/fuzzers/fuzz_data_helper.h"
+
 namespace webrtc {
 
 class AudioDecoder;
@@ -24,8 +26,7 @@ enum class DecoderFunctionType {
 };
 
 void FuzzAudioDecoder(DecoderFunctionType decode_type,
-                      const uint8_t* data,
-                      size_t size,
+                      FuzzDataHelper& fuzz_data,
                       AudioDecoder* decoder,
                       int sample_rate_hz,
                       size_t max_decoded_bytes,
diff --git a/test/fuzzers/audio_decoder_g722_fuzzer.cc b/test/fuzzers/audio_decoder_g722_fuzzer.cc
index 0e75baf095..c60b8ed66a 100644
--- a/test/fuzzers/audio_decoder_g722_fuzzer.cc
+++ b/test/fuzzers/audio_decoder_g722_fuzzer.cc
@@ -14,16 +14,17 @@
 
 #include "modules/audio_coding/codecs/g722/audio_decoder_g722.h"
 #include "test/fuzzers/audio_decoder_fuzzer.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  if (size > 10000 || size < 1) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
+  if (fuzz_data.size() > 10'000 || fuzz_data.size() < 1) {
     return;
   }
 
   std::unique_ptr<AudioDecoder> dec;
   size_t num_channels;
-  if (data[0] % 2) {
+  if (fuzz_data.Read<uint8_t>() % 2) {
     dec = std::make_unique<AudioDecoderG722Impl>();
     num_channels = 1;
   } else {
@@ -37,7 +38,7 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
   std::unique_ptr<int16_t[]> output =
       std::make_unique<int16_t[]>(allocated_ouput_size_samples);
   FuzzAudioDecoder(
-      DecoderFunctionType::kNormalDecode, data, size, dec.get(), sample_rate_hz,
+      DecoderFunctionType::kNormalDecode, fuzz_data, dec.get(), sample_rate_hz,
       allocated_ouput_size_samples * sizeof(int16_t), output.get());
 }
 }  // namespace webrtc
diff --git a/test/fuzzers/audio_decoder_multistream_opus_fuzzer.cc b/test/fuzzers/audio_decoder_multistream_opus_fuzzer.cc
index 1bcb1b156b..b52355373d 100644
--- a/test/fuzzers/audio_decoder_multistream_opus_fuzzer.cc
+++ b/test/fuzzers/audio_decoder_multistream_opus_fuzzer.cc
@@ -13,7 +13,6 @@
 #include <memory>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio_codecs/opus/audio_decoder_multi_channel_opus.h"
 #include "api/audio_codecs/opus/audio_decoder_multi_channel_opus_config.h"
 #include "rtc_base/checks.h"
@@ -22,8 +21,6 @@
 
 namespace webrtc {
 
-using test::FuzzDataHelper;
-
 AudioDecoderMultiChannelOpusConfig MakeDecoderConfig(
     int num_channels,
     int num_streams,
@@ -37,8 +34,8 @@ AudioDecoderMultiChannelOpusConfig MakeDecoderConfig(
   return config;
 }
 
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  AudioDecoderMultiChannelOpusConfig surround_configs[] = {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
+  const AudioDecoderMultiChannelOpusConfig kSurroundConfigs[] = {
       MakeDecoderConfig(1, 1, 0, {0}),  // Mono
 
       MakeDecoderConfig(2, 2, 0, {0, 0}),  // Copy the first (of
@@ -54,9 +51,8 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
       MakeDecoderConfig(8, 5, 3, {0, 6, 1, 2, 3, 4, 5, 7})  // 7.1
   };
 
-  FuzzDataHelper helper(MakeArrayView(data, size));
+  const auto config = fuzz_data.SelectOneOf(kSurroundConfigs);
 
-  auto config = helper.MoveOneOf(surround_configs);
   RTC_CHECK(config.IsOk());
   std::unique_ptr<AudioDecoder> dec =
       AudioDecoderMultiChannelOpus::MakeAudioDecoder(std::move(config));
@@ -67,7 +63,7 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
                                // for the stereo Opus codec. It should be enough
                                // for 8 channels.
   int16_t output[kAllocatedOuputSizeSamples];
-  FuzzAudioDecoder(DecoderFunctionType::kNormalDecode, data, size, dec.get(),
+  FuzzAudioDecoder(DecoderFunctionType::kNormalDecode, fuzz_data, dec.get(),
                    kSampleRateHz, sizeof(output), output);
 }
 }  // namespace webrtc
diff --git a/test/fuzzers/audio_decoder_opus_fuzzer.cc b/test/fuzzers/audio_decoder_opus_fuzzer.cc
index bfca95512f..9afd6e20b6 100644
--- a/test/fuzzers/audio_decoder_opus_fuzzer.cc
+++ b/test/fuzzers/audio_decoder_opus_fuzzer.cc
@@ -14,15 +14,16 @@
 #include "api/field_trials.h"
 #include "modules/audio_coding/codecs/opus/audio_decoder_opus.h"
 #include "test/fuzzers/audio_decoder_fuzzer.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  const size_t channels = (size % 2) + 1;  // 1 or 2 channels.
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
+  const size_t channels = (fuzz_data.size() % 2) + 1;  // 1 or 2 channels.
   const int kSampleRateHz = 48'000;
   AudioDecoderOpusImpl dec(FieldTrials(""), channels, kSampleRateHz);
   const size_t kAllocatedOuputSizeSamples = kSampleRateHz / 10;  // 100 ms.
   int16_t output[kAllocatedOuputSizeSamples];
-  FuzzAudioDecoder(DecoderFunctionType::kNormalDecode, data, size, &dec,
+  FuzzAudioDecoder(DecoderFunctionType::kNormalDecode, fuzz_data, &dec,
                    kSampleRateHz, sizeof(output), output);
 }
 }  // namespace webrtc
diff --git a/test/fuzzers/audio_decoder_opus_redundant_fuzzer.cc b/test/fuzzers/audio_decoder_opus_redundant_fuzzer.cc
index 1c2d56efa0..742dce943c 100644
--- a/test/fuzzers/audio_decoder_opus_redundant_fuzzer.cc
+++ b/test/fuzzers/audio_decoder_opus_redundant_fuzzer.cc
@@ -14,15 +14,16 @@
 #include "api/field_trials.h"
 #include "modules/audio_coding/codecs/opus/audio_decoder_opus.h"
 #include "test/fuzzers/audio_decoder_fuzzer.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  const size_t channels = (size % 2) + 1;  // 1 or 2 channels.
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
+  const size_t channels = (fuzz_data.size() % 2) + 1;  // 1 or 2 channels.
   const int kSampleRateHz = 48'000;
   AudioDecoderOpusImpl dec(FieldTrials(""), channels, kSampleRateHz);
   const size_t kAllocatedOuputSizeSamples = kSampleRateHz / 10;  // 100 ms.
   int16_t output[kAllocatedOuputSizeSamples];
-  FuzzAudioDecoder(DecoderFunctionType::kRedundantDecode, data, size, &dec,
+  FuzzAudioDecoder(DecoderFunctionType::kRedundantDecode, fuzz_data, &dec,
                    kSampleRateHz, sizeof(output), output);
 }
 }  // namespace webrtc
diff --git a/test/fuzzers/audio_decoder_pcm16b_fuzzer.cc b/test/fuzzers/audio_decoder_pcm16b_fuzzer.cc
index 90001465e5..463c981279 100644
--- a/test/fuzzers/audio_decoder_pcm16b_fuzzer.cc
+++ b/test/fuzzers/audio_decoder_pcm16b_fuzzer.cc
@@ -15,15 +15,16 @@
 #include "modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
 #include "rtc_base/checks.h"
 #include "test/fuzzers/audio_decoder_fuzzer.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  if (size > 10000 || size < 2) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
+  if (fuzz_data.size() > 10'000 || fuzz_data.size() < 2) {
     return;
   }
 
   int sample_rate_hz;
-  switch (data[0] % 4) {
+  switch (fuzz_data.Read<uint8_t>() % 4) {
     case 0:
       sample_rate_hz = 8000;
       break;
@@ -40,11 +41,7 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
       RTC_DCHECK_NOTREACHED();
       return;
   }
-  const size_t num_channels = data[1] % 16 + 1;
-
-  // Two first bytes of the data are used. Move forward.
-  data += 2;
-  size -= 2;
+  const size_t num_channels = fuzz_data.Read<uint8_t>() % 16 + 1;
 
   AudioDecoderPcm16B dec(sample_rate_hz, num_channels);
   // Allocate a maximum output size of 100 ms.
@@ -53,7 +50,7 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
   std::unique_ptr<int16_t[]> output =
       std::make_unique<int16_t[]>(allocated_ouput_size_samples);
   FuzzAudioDecoder(
-      DecoderFunctionType::kNormalDecode, data, size, &dec, sample_rate_hz,
+      DecoderFunctionType::kNormalDecode, fuzz_data, &dec, sample_rate_hz,
       allocated_ouput_size_samples * sizeof(int16_t), output.get());
 }
 }  // namespace webrtc
diff --git a/test/fuzzers/audio_decoder_pcm_fuzzer.cc b/test/fuzzers/audio_decoder_pcm_fuzzer.cc
index efd723493e..41e3472c6a 100644
--- a/test/fuzzers/audio_decoder_pcm_fuzzer.cc
+++ b/test/fuzzers/audio_decoder_pcm_fuzzer.cc
@@ -14,32 +14,29 @@
 
 #include "modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
 #include "test/fuzzers/audio_decoder_fuzzer.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  if (size > 10000 || size < 2) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
+  if (fuzz_data.size() > 10'000 || fuzz_data.size() < 2) {
     return;
   }
 
-  const size_t num_channels = data[0] % 16 + 1;
+  const size_t num_channels = fuzz_data.Read<uint8_t>() % 16 + 1;
 
   std::unique_ptr<AudioDecoder> dec;
-  if (data[1] % 2) {
+  if (fuzz_data.Read<uint8_t>() % 2) {
     dec = std::make_unique<AudioDecoderPcmU>(num_channels);
   } else {
     dec = std::make_unique<AudioDecoderPcmA>(num_channels);
   }
 
-  // Two first bytes of the data are used. Move forward.
-  data += 2;
-  size -= 2;
-
   // Allocate a maximum output size of 100 ms.
   const size_t allocated_ouput_size_samples =
       dec->SampleRateHz() * num_channels / 10;
   std::unique_ptr<int16_t[]> output =
       std::make_unique<int16_t[]>(allocated_ouput_size_samples);
-  FuzzAudioDecoder(DecoderFunctionType::kNormalDecode, data, size, dec.get(),
+  FuzzAudioDecoder(DecoderFunctionType::kNormalDecode, fuzz_data, dec.get(),
                    dec->SampleRateHz(),
                    allocated_ouput_size_samples * sizeof(int16_t),
                    output.get());
diff --git a/test/fuzzers/audio_encoder_fuzzer.cc b/test/fuzzers/audio_encoder_fuzzer.cc
index 010e519bde..a0bdcfd596 100644
--- a/test/fuzzers/audio_encoder_fuzzer.cc
+++ b/test/fuzzers/audio_encoder_fuzzer.cc
@@ -25,13 +25,12 @@ namespace webrtc {
 // This function reads bytes from `data_view`, interprets them as RTP timestamp
 // and input samples, and sends them for encoding. The process continues until
 // no more data is available.
-void FuzzAudioEncoder(ArrayView<const uint8_t> data_view,
+void FuzzAudioEncoder(FuzzDataHelper data,
                       std::unique_ptr<AudioEncoder> encoder) {
-  test::FuzzDataHelper data(data_view);
   const size_t block_size_samples =
       encoder->SampleRateHz() / 100 * encoder->NumChannels();
   const size_t block_size_bytes = block_size_samples * sizeof(int16_t);
-  if (data_view.size() / block_size_bytes > 1000) {
+  if (data.BytesLeft() / block_size_bytes > 1000) {
     // If the size of the fuzzer data is more than 1000 input blocks (i.e., more
     // than 10 seconds), then don't fuzz at all for the fear of timing out.
     return;
diff --git a/test/fuzzers/audio_encoder_fuzzer.h b/test/fuzzers/audio_encoder_fuzzer.h
index de0609dff8..257401b3b8 100644
--- a/test/fuzzers/audio_encoder_fuzzer.h
+++ b/test/fuzzers/audio_encoder_fuzzer.h
@@ -11,15 +11,14 @@
 #ifndef TEST_FUZZERS_AUDIO_ENCODER_FUZZER_H_
 #define TEST_FUZZERS_AUDIO_ENCODER_FUZZER_H_
 
-#include <cstdint>
 #include <memory>
 
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_encoder.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
 
-void FuzzAudioEncoder(ArrayView<const uint8_t> data_view,
+void FuzzAudioEncoder(FuzzDataHelper data,
                       std::unique_ptr<AudioEncoder> encoder);
 
 }  // namespace webrtc
diff --git a/test/fuzzers/audio_encoder_opus_fuzzer.cc b/test/fuzzers/audio_encoder_opus_fuzzer.cc
index f47161a06c..e87689e4b3 100644
--- a/test/fuzzers/audio_encoder_opus_fuzzer.cc
+++ b/test/fuzzers/audio_encoder_opus_fuzzer.cc
@@ -9,17 +9,17 @@
  */
 
 #include <cstddef>
-#include <cstdint>
 
 #include "api/audio_codecs/opus/audio_encoder_opus.h"
 #include "api/environment/environment.h"
 #include "api/environment/environment_factory.h"
 #include "rtc_base/checks.h"
 #include "test/fuzzers/audio_encoder_fuzzer.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
 
-void FuzzOneInput(const uint8_t* data, size_t size) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
   // Create Environment once because creating it for each input noticably
   // reduces the speed of the fuzzer.
   static const Environment* const env = new Environment(CreateEnvironment());
@@ -29,7 +29,7 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
   RTC_CHECK(config.IsOk());
 
   FuzzAudioEncoder(
-      /*data_view=*/{data, size},
+      /*data=*/fuzz_data,
       /*encoder=*/AudioEncoderOpus::MakeAudioEncoder(*env, std::move(config),
                                                      {.payload_type = 100}));
 }
diff --git a/test/fuzzers/audio_processing_configs_fuzzer.cc b/test/fuzzers/audio_processing_configs_fuzzer.cc
index 7541529862..614cc66b37 100644
--- a/test/fuzzers/audio_processing_configs_fuzzer.cc
+++ b/test/fuzzers/audio_processing_configs_fuzzer.cc
@@ -17,7 +17,6 @@
 
 #include "absl/base/nullability.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio/audio_processing.h"
 #include "api/audio/builtin_audio_processing_builder.h"
 #include "api/audio/echo_canceller3_factory.h"
@@ -47,9 +46,9 @@ const Environment& GetEnvironment() {
   return *env;
 }
 
-webrtc::scoped_refptr<AudioProcessing> CreateApm(
-    test::FuzzDataHelper* fuzz_data,
-    TaskQueueBase* absl_nonnull worker_queue) {
+webrtc::scoped_refptr<AudioProcessing> CreateApm(FuzzDataHelper* fuzz_data,
+                                                 TaskQueueBase* absl_nonnull
+                                                     worker_queue) {
   // Parse boolean values for optionally enabling different
   // configurable public components of APM.
   bool use_ts = fuzz_data->ReadOrDefaultValue(true);
@@ -135,11 +134,10 @@ webrtc::scoped_refptr<AudioProcessing> CreateApm(
 
 }  // namespace
 
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  if (size > 400000) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
+  if (fuzz_data.size() > 400'000) {
     return;
   }
-  test::FuzzDataHelper fuzz_data(webrtc::ArrayView<const uint8_t>(data, size));
 
   std::unique_ptr<TaskQueueBase, TaskQueueDeleter> worker_queue =
       GetEnvironment().task_queue_factory().CreateTaskQueue(
diff --git a/test/fuzzers/audio_processing_fuzzer_helper.cc b/test/fuzzers/audio_processing_fuzzer_helper.cc
index bfb82881c3..dffe40d1fa 100644
--- a/test/fuzzers/audio_processing_fuzzer_helper.cc
+++ b/test/fuzzers/audio_processing_fuzzer_helper.cc
@@ -31,7 +31,7 @@ bool ValidForApm(float x) {
   return std::isfinite(x) && -1.0f <= x && x <= 1.0f;
 }
 
-void GenerateFloatFrame(test::FuzzDataHelper* fuzz_data,
+void GenerateFloatFrame(FuzzDataHelper* fuzz_data,
                         int input_rate,
                         int num_channels,
                         float* const* float_frames) {
@@ -56,7 +56,7 @@ void GenerateFloatFrame(test::FuzzDataHelper* fuzz_data,
   }
 }
 
-void GenerateFixedFrame(test::FuzzDataHelper* fuzz_data,
+void GenerateFixedFrame(FuzzDataHelper* fuzz_data,
                         int input_rate,
                         int num_channels,
                         AudioFrame* fixed_frame) {
@@ -75,7 +75,7 @@ void GenerateFixedFrame(test::FuzzDataHelper* fuzz_data,
 }
 }  // namespace
 
-void FuzzAudioProcessing(test::FuzzDataHelper* fuzz_data,
+void FuzzAudioProcessing(FuzzDataHelper* fuzz_data,
                          scoped_refptr<AudioProcessing> apm) {
   AudioFrame fixed_frame;
   // Normal usage is up to 8 channels. Allowing to fuzz one beyond this allows
diff --git a/test/fuzzers/audio_processing_fuzzer_helper.h b/test/fuzzers/audio_processing_fuzzer_helper.h
index fc216f2e45..59c6161459 100644
--- a/test/fuzzers/audio_processing_fuzzer_helper.h
+++ b/test/fuzzers/audio_processing_fuzzer_helper.h
@@ -17,7 +17,7 @@
 #include "test/fuzzers/fuzz_data_helper.h"
 namespace webrtc {
 
-void FuzzAudioProcessing(test::FuzzDataHelper* fuzz_data,
+void FuzzAudioProcessing(FuzzDataHelper* fuzz_data,
                          scoped_refptr<AudioProcessing> apm);
 
 }  // namespace webrtc
diff --git a/test/fuzzers/audio_processing_sample_rate_fuzzer.cc b/test/fuzzers/audio_processing_sample_rate_fuzzer.cc
index 5c0dedfa45..8ce3d059b0 100644
--- a/test/fuzzers/audio_processing_sample_rate_fuzzer.cc
+++ b/test/fuzzers/audio_processing_sample_rate_fuzzer.cc
@@ -16,7 +16,6 @@
 #include <string>
 #include <utility>
 
-#include "api/array_view.h"
 #include "api/audio/audio_processing.h"
 #include "api/audio/builtin_audio_processing_builder.h"
 #include "api/environment/environment_factory.h"
@@ -32,7 +31,7 @@ constexpr int kMaxNumChannels = 2;
 constexpr int kMaxSampleRateHz = 400000;
 constexpr int kMaxSamplesPerChannel = kMaxSampleRateHz / 100;
 
-void GenerateFloatFrame(test::FuzzDataHelper& fuzz_data,
+void GenerateFloatFrame(FuzzDataHelper& fuzz_data,
                         int input_rate,
                         int num_channels,
                         float* const* float_frames) {
@@ -47,7 +46,7 @@ void GenerateFloatFrame(test::FuzzDataHelper& fuzz_data,
   }
 }
 
-void GenerateFixedFrame(test::FuzzDataHelper& fuzz_data,
+void GenerateFixedFrame(FuzzDataHelper& fuzz_data,
                         int input_rate,
                         int num_channels,
                         int16_t* fixed_frames) {
@@ -81,11 +80,10 @@ class NoopCustomProcessing : public CustomProcessing {
 // of APM. For example, the sample rate 22050 Hz is processed by APM in frames
 // of floor(22050/100) = 220 samples. This is not exactly 10 ms of audio
 // content, and may break assumptions commonly made on the APM frame size.
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  if (size > 100) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
+  if (fuzz_data.size() > 100) {
     return;
   }
-  test::FuzzDataHelper fuzz_data(webrtc::ArrayView<const uint8_t>(data, size));
 
   std::unique_ptr<CustomProcessing> capture_processor =
       fuzz_data.ReadOrDefaultValue(true)
diff --git a/test/fuzzers/base64_decode_fuzzer.cc b/test/fuzzers/base64_decode_fuzzer.cc
index a986b3d240..0b96642942 100644
--- a/test/fuzzers/base64_decode_fuzzer.cc
+++ b/test/fuzzers/base64_decode_fuzzer.cc
@@ -8,17 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include <cstdint>
 #include <cstring>
-#include <string>
 
 #include "rtc_base/base64.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
 
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  std::string str(reinterpret_cast<const char*>(data), size);
-  Base64Decode(str);
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
+  Base64Decode(fuzz_data.ReadString());
 }
 
 }  // namespace webrtc
diff --git a/test/fuzzers/base64_encode_fuzzer.cc b/test/fuzzers/base64_encode_fuzzer.cc
index a8cb6213c2..5edaa6ede9 100644
--- a/test/fuzzers/base64_encode_fuzzer.cc
+++ b/test/fuzzers/base64_encode_fuzzer.cc
@@ -16,14 +16,17 @@
 #include "api/array_view.h"
 #include "rtc_base/base64.h"
 #include "rtc_base/checks.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
 
-void FuzzOneInput(const uint8_t* data, size_t size) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
+  ArrayView<const uint8_t> raw = fuzz_data.ReadRemaining();
   std::optional<std::string> decoded_encoded_data =
-      Base64Decode(Base64Encode(webrtc::MakeArrayView(data, size)));
+      Base64Decode(Base64Encode(raw));
   RTC_CHECK(decoded_encoded_data.has_value());
-  RTC_CHECK_EQ(std::memcmp(data, decoded_encoded_data->data(), size), 0);
+  RTC_CHECK_EQ(
+      std::memcmp(raw.data(), decoded_encoded_data->data(), raw.size()), 0);
 }
 
 }  // namespace webrtc
diff --git a/test/fuzzers/comfort_noise_decoder_fuzzer.cc b/test/fuzzers/comfort_noise_decoder_fuzzer.cc
index 8820dfcb1b..6ee89e2eec 100644
--- a/test/fuzzers/comfort_noise_decoder_fuzzer.cc
+++ b/test/fuzzers/comfort_noise_decoder_fuzzer.cc
@@ -55,11 +55,11 @@ void FuzzOneInputTest(webrtc::ArrayView<const uint8_t> data) {
 }  // namespace
 }  // namespace test
 
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  if (size > 5000) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
+  if (fuzz_data.size() > 5'000) {
     return;
   }
-  test::FuzzOneInputTest(webrtc::ArrayView<const uint8_t>(data, size));
+  test::FuzzOneInputTest(fuzz_data.ReadRemaining());
 }
 
 }  // namespace webrtc
diff --git a/test/fuzzers/dcsctp_packet_fuzzer.cc b/test/fuzzers/dcsctp_packet_fuzzer.cc
index e6d1d25139..a65c8f20b0 100644
--- a/test/fuzzers/dcsctp_packet_fuzzer.cc
+++ b/test/fuzzers/dcsctp_packet_fuzzer.cc
@@ -8,20 +8,19 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #include <cstddef>
-#include <cstdint>
 #include <optional>
 
 #include "api/array_view.h"
 #include "net/dcsctp/packet/chunk/chunk.h"
 #include "net/dcsctp/packet/sctp_packet.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
 using dcsctp::SctpPacket;
 
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  std::optional<SctpPacket> c =
-      SctpPacket::Parse(webrtc::ArrayView<const uint8_t>(data, size),
-                        {.disable_checksum_verification = true});
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
+  std::optional<SctpPacket> c = SctpPacket::Parse(
+      fuzz_data.ReadRemaining(), {.disable_checksum_verification = true});
 
   if (!c.has_value()) {
     return;
diff --git a/test/fuzzers/dcsctp_socket_fuzzer.cc b/test/fuzzers/dcsctp_socket_fuzzer.cc
index fb2556faad..235d17f50a 100644
--- a/test/fuzzers/dcsctp_socket_fuzzer.cc
+++ b/test/fuzzers/dcsctp_socket_fuzzer.cc
@@ -8,22 +8,20 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #include <cstddef>
-#include <cstdint>
 
-#include "api/array_view.h"
 #include "net/dcsctp/fuzzers/dcsctp_fuzzers.h"
 #include "net/dcsctp/public/dcsctp_options.h"
 #include "net/dcsctp/socket/dcsctp_socket.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
 
-void FuzzOneInput(const uint8_t* data, size_t size) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
   dcsctp::dcsctp_fuzzers::FuzzerCallbacks cb;
   dcsctp::DcSctpOptions options;
   options.disable_checksum_verification = true;
   dcsctp::DcSctpSocket socket("A", cb, nullptr, options);
 
-  dcsctp::dcsctp_fuzzers::FuzzSocket(
-      socket, cb, webrtc::ArrayView<const uint8_t>(data, size));
+  dcsctp::dcsctp_fuzzers::FuzzSocket(socket, cb, fuzz_data.ReadRemaining());
 }
 }  // namespace webrtc
diff --git a/test/fuzzers/field_trial_fuzzer.cc b/test/fuzzers/field_trial_fuzzer.cc
index ee9d845608..a031c0e202 100644
--- a/test/fuzzers/field_trial_fuzzer.cc
+++ b/test/fuzzers/field_trial_fuzzer.cc
@@ -9,19 +9,17 @@
  */
 
 #include <cstddef>
-#include <cstdint>
 
-#include "absl/strings/string_view.h"
 #include "api/field_trials.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
 
-void FuzzOneInput(const uint8_t* data, size_t size) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
   // FieldTrials constructor crashes on invalid input.
   // FieldTrials::Create validates input and returns nullptr when it is invalid,
   // but should never crash.
-  FieldTrials::Create(
-      absl::string_view(reinterpret_cast<const char*>(data), size));
+  FieldTrials::Create(fuzz_data.ReadString());
 }
 
 }  // namespace webrtc
diff --git a/test/fuzzers/flexfec_header_reader_fuzzer.cc b/test/fuzzers/flexfec_header_reader_fuzzer.cc
index d268537b03..1f397f29c1 100644
--- a/test/fuzzers/flexfec_header_reader_fuzzer.cc
+++ b/test/fuzzers/flexfec_header_reader_fuzzer.cc
@@ -13,24 +13,26 @@
 #include <cstdint>
 #include <cstring>
 
+#include "api/array_view.h"
 #include "api/scoped_refptr.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/flexfec_header_reader_writer.h"
 #include "modules/rtp_rtcp/source/forward_error_correction.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
 
 using Packet = ForwardErrorCorrection::Packet;
 using ReceivedFecPacket = ForwardErrorCorrection::ReceivedFecPacket;
 
-void FuzzOneInput(const uint8_t* data, size_t size) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
   ReceivedFecPacket packet;
   packet.pkt = webrtc::scoped_refptr<Packet>(new Packet());
   const size_t packet_size =
-      std::min(size, static_cast<size_t>(IP_PACKET_SIZE));
-  packet.pkt->data.SetSize(packet_size);
+      std::min(fuzz_data.size(), static_cast<size_t>(IP_PACKET_SIZE));
+  ArrayView<const uint8_t> raw = fuzz_data.ReadByteArray(packet_size);
   packet.pkt->data.EnsureCapacity(IP_PACKET_SIZE);
-  memcpy(packet.pkt->data.MutableData(), data, packet_size);
+  packet.pkt->data.SetData(raw.data(), raw.size());
 
   FlexfecHeaderReader flexfec_reader;
   flexfec_reader.ReadFecHeader(&packet);
diff --git a/test/fuzzers/flexfec_receiver_fuzzer.cc b/test/fuzzers/flexfec_receiver_fuzzer.cc
index a35e0bd80d..5b201df67b 100644
--- a/test/fuzzers/flexfec_receiver_fuzzer.cc
+++ b/test/fuzzers/flexfec_receiver_fuzzer.cc
@@ -11,13 +11,12 @@
 #include <cstddef>
 #include <cstdint>
 #include <cstring>
-#include <memory>
 
 #include "modules/rtp_rtcp/include/flexfec_receiver.h"
 #include "modules/rtp_rtcp/include/recovered_packet_receiver.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
-#include "modules/rtp_rtcp/source/byte_io.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
 
@@ -27,48 +26,40 @@ class DummyCallback : public RecoveredPacketReceiver {
 };
 }  // namespace
 
-void FuzzOneInput(const uint8_t* data, size_t size) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
   constexpr size_t kMinDataNeeded = 12;
-  if (size < kMinDataNeeded || size > 2000) {
+  if (fuzz_data.size() < kMinDataNeeded || fuzz_data.size() > 2'000) {
     return;
   }
 
-  uint32_t flexfec_ssrc;
-  memcpy(&flexfec_ssrc, data + 0, 4);
-  uint16_t flexfec_seq_num;
-  memcpy(&flexfec_seq_num, data + 4, 2);
-  uint32_t media_ssrc;
-  memcpy(&media_ssrc, data + 6, 4);
-  uint16_t media_seq_num;
-  memcpy(&media_seq_num, data + 10, 2);
+  uint32_t flexfec_ssrc = fuzz_data.Read<uint32_t>();
+  uint16_t flexfec_seq_num = fuzz_data.Read<uint16_t>();
+  uint32_t media_ssrc = fuzz_data.Read<uint32_t>();
+  uint16_t media_seq_num = fuzz_data.Read<uint16_t>();
 
   DummyCallback callback;
   FlexfecReceiver receiver(flexfec_ssrc, media_ssrc, &callback);
 
-  std::unique_ptr<uint8_t[]> packet;
   size_t packet_length;
-  size_t i = kMinDataNeeded;
-  while (i < size) {
-    packet_length = kRtpHeaderSize + data[i++];
-    packet = std::unique_ptr<uint8_t[]>(new uint8_t[packet_length]);
-    if (i + packet_length >= size) {
+  while (fuzz_data.BytesLeft() > 0) {
+    packet_length = kRtpHeaderSize + fuzz_data.Read<uint8_t>();
+    if (fuzz_data.BytesLeft() < packet_length + 1) {
       break;
     }
-    memcpy(packet.get(), data + i, packet_length);
-    i += packet_length;
-    if (i < size && data[i++] % 2 == 0) {
+    RtpPacketReceived parsed_packet;
+    if (!parsed_packet.Parse(fuzz_data.ReadByteArray(packet_length))) {
+      continue;
+    }
+    if (fuzz_data.Read<uint8_t>() % 2 == 0) {
       // Simulate FlexFEC packet.
-      ByteWriter<uint16_t>::WriteBigEndian(packet.get() + 2, flexfec_seq_num++);
-      ByteWriter<uint32_t>::WriteBigEndian(packet.get() + 8, flexfec_ssrc);
+      parsed_packet.SetSequenceNumber(flexfec_seq_num++);
+      parsed_packet.SetSsrc(flexfec_ssrc);
     } else {
       // Simulate media packet.
-      ByteWriter<uint16_t>::WriteBigEndian(packet.get() + 2, media_seq_num++);
-      ByteWriter<uint32_t>::WriteBigEndian(packet.get() + 8, media_ssrc);
-    }
-    RtpPacketReceived parsed_packet;
-    if (parsed_packet.Parse(packet.get(), packet_length)) {
-      receiver.OnRtpPacket(parsed_packet);
+      parsed_packet.SetSequenceNumber(media_seq_num++);
+      parsed_packet.SetSsrc(media_ssrc);
     }
+    receiver.OnRtpPacket(parsed_packet);
   }
 }
 
diff --git a/test/fuzzers/flexfec_sender_fuzzer.cc b/test/fuzzers/flexfec_sender_fuzzer.cc
index e7bf7427cc..6738a38915 100644
--- a/test/fuzzers/flexfec_sender_fuzzer.cc
+++ b/test/fuzzers/flexfec_sender_fuzzer.cc
@@ -11,7 +11,6 @@
 #include <cstddef>
 #include <cstdint>
 #include <cstring>
-#include <memory>
 #include <vector>
 
 #include "api/environment/environment.h"
@@ -20,10 +19,10 @@
 #include "modules/include/module_fec_types.h"
 #include "modules/rtp_rtcp/include/flexfec_sender.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
-#include "modules/rtp_rtcp/source/byte_io.h"
 #include "modules/rtp_rtcp/source/rtp_header_extension_size.h"
 #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
 #include "system_wrappers/include/clock.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
 
@@ -38,43 +37,42 @@ const std::vector<RtpExtensionSize> kNoRtpHeaderExtensionSizes;
 
 }  // namespace
 
-void FuzzOneInput(const uint8_t* data, size_t size) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
   // Create Environment once because creating it for each input noticably
   // reduces the speed of the fuzzer.
   static SimulatedClock* const clock = new SimulatedClock(1);
   static const Environment* const env =
       new Environment(CreateEnvironment(clock));
 
-  size_t i = 0;
-  if (size < 5 || size > 200) {
+  if (fuzz_data.size() < 5 || fuzz_data.size() > 200) {
     return;
   }
   // Set time to (1 + data[i++]);
-  clock->AdvanceTimeMicroseconds(1 + data[i++] - clock->TimeInMicroseconds());
+  clock->AdvanceTimeMicroseconds(1 + fuzz_data.Read<uint8_t>() -
+                                 clock->TimeInMicroseconds());
   FlexfecSender sender(*env, kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc,
                        kNoMid, kNoRtpHeaderExtensions,
                        kNoRtpHeaderExtensionSizes, nullptr /* rtp_state */);
   FecProtectionParams params = {
-      data[i++], static_cast<int>(data[i++] % 100),
-      data[i++] <= 127 ? kFecMaskRandom : kFecMaskBursty};
+      .fec_rate = fuzz_data.Read<uint8_t>(),
+      .max_fec_frames = static_cast<int>(fuzz_data.Read<uint8_t>() % 100),
+      .fec_mask_type =
+          fuzz_data.Read<uint8_t>() <= 127 ? kFecMaskRandom : kFecMaskBursty};
   sender.SetProtectionParameters(params, params);
-  uint16_t seq_num = data[i++];
+  uint16_t seq_num = fuzz_data.Read<uint8_t>();
 
-  while (i + 1 < size) {
+  while (fuzz_data.BytesLeft() >= 1) {
     // Everything past the base RTP header (12 bytes) is payload,
     // from the perspective of FlexFEC.
-    size_t payload_size = data[i++];
-    if (i + kRtpHeaderSize + payload_size >= size)
+    size_t payload_size = fuzz_data.Read<uint8_t>();
+    if (fuzz_data.BytesLeft() <= kRtpHeaderSize + payload_size)
       break;
-    std::unique_ptr<uint8_t[]> packet(
-        new uint8_t[kRtpHeaderSize + payload_size]);
-    memcpy(packet.get(), &data[i], kRtpHeaderSize + payload_size);
-    i += kRtpHeaderSize + payload_size;
-    ByteWriter<uint16_t>::WriteBigEndian(&packet[2], seq_num++);
-    ByteWriter<uint32_t>::WriteBigEndian(&packet[8], kMediaSsrc);
     RtpPacketToSend rtp_packet(nullptr);
-    if (!rtp_packet.Parse(packet.get(), kRtpHeaderSize + payload_size))
+    if (!rtp_packet.Parse(
+            fuzz_data.ReadByteArray(kRtpHeaderSize + payload_size)))
       break;
+    rtp_packet.SetSequenceNumber(seq_num++);
+    rtp_packet.SetSsrc(kMediaSsrc);
     sender.AddPacketAndGenerateFec(rtp_packet);
     sender.GetFecPackets();
   }
diff --git a/test/fuzzers/forward_error_correction_fuzzer.cc b/test/fuzzers/forward_error_correction_fuzzer.cc
index 6997dc3523..588fbdf139 100644
--- a/test/fuzzers/forward_error_correction_fuzzer.cc
+++ b/test/fuzzers/forward_error_correction_fuzzer.cc
@@ -19,6 +19,7 @@
 #include "modules/rtp_rtcp/source/byte_io.h"
 #include "modules/rtp_rtcp/source/forward_error_correction.h"
 #include "rtc_base/byte_buffer.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
 
@@ -30,8 +31,8 @@ constexpr size_t kPacketSize = 50;
 constexpr size_t kMaxPacketsInBuffer = 48;
 }  // namespace
 
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  if (size > 5000) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
+  if (fuzz_data.size() > 5'000) {
     return;
   }
   // Object under test.
@@ -39,7 +40,7 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
       ForwardErrorCorrection::CreateFlexfec(kFecSsrc, kMediaSsrc);
 
   // Entropy from fuzzer.
-  webrtc::ByteBufferReader fuzz_buffer(webrtc::MakeArrayView(data, size));
+  webrtc::ByteBufferReader fuzz_buffer(fuzz_data.ReadRemaining());
 
   // Initial stream state.
   uint16_t media_seqnum;
diff --git a/test/fuzzers/frame_buffer_fuzzer.cc b/test/fuzzers/frame_buffer_fuzzer.cc
index 98e8d5a550..f2164c1616 100644
--- a/test/fuzzers/frame_buffer_fuzzer.cc
+++ b/test/fuzzers/frame_buffer_fuzzer.cc
@@ -13,7 +13,6 @@
 #include <memory>
 #include <utility>
 
-#include "api/array_view.h"
 #include "api/field_trials.h"
 #include "api/video/encoded_frame.h"
 #include "api/video/frame_buffer.h"
@@ -32,19 +31,18 @@ constexpr int kFrameIdLength = 1 << 15;
 
 }  // namespace
 
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  if (size > 10000) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
+  if (fuzz_data.size() > 10'000) {
     return;
   }
 
   FieldTrials field_trials("");
   FrameBuffer buffer(/*max_frame_slots=*/100, /*max_decode_history=*/1000,
                      field_trials);
-  test::FuzzDataHelper helper(webrtc::MakeArrayView(data, size));
   SeqNumUnwrapper<uint16_t, kFrameIdLength> unwrapper;
 
-  while (helper.BytesLeft() > 0) {
-    int action = helper.ReadOrDefaultValue<uint8_t>(0) % 6;
+  while (fuzz_data.BytesLeft() > 0) {
+    int action = fuzz_data.ReadOrDefaultValue<uint8_t>(0) % 6;
 
     switch (action) {
       case 0: {
@@ -69,17 +67,18 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
       }
       case 5: {
         auto frame = std::make_unique<FuzzyFrameObject>();
-        frame->SetRtpTimestamp(helper.ReadOrDefaultValue<uint32_t>(0));
+        frame->SetRtpTimestamp(fuzz_data.ReadOrDefaultValue<uint32_t>(0));
         int64_t wire_id =
-            helper.ReadOrDefaultValue<uint16_t>(0) & (kFrameIdLength - 1);
+            fuzz_data.ReadOrDefaultValue<uint16_t>(0) & (kFrameIdLength - 1);
         frame->SetId(unwrapper.Unwrap(wire_id));
-        frame->is_last_spatial_layer = helper.ReadOrDefaultValue<bool>(false);
+        frame->is_last_spatial_layer =
+            fuzz_data.ReadOrDefaultValue<bool>(false);
 
-        frame->num_references = helper.ReadOrDefaultValue<uint8_t>(0) %
+        frame->num_references = fuzz_data.ReadOrDefaultValue<uint8_t>(0) %
                                 EncodedFrame::kMaxFrameReferences;
 
         for (uint8_t i = 0; i < frame->num_references; ++i) {
-          frame->references[i] = helper.ReadOrDefaultValue<int64_t>(0);
+          frame->references[i] = fuzz_data.ReadOrDefaultValue<int64_t>(0);
         }
 
         buffer.InsertFrame(std::move(frame));
diff --git a/test/fuzzers/fuzz_data_helper.cc b/test/fuzzers/fuzz_data_helper.cc
deleted file mode 100644
index 5e8019d46d..0000000000
--- a/test/fuzzers/fuzz_data_helper.cc
+++ /dev/null
@@ -1,23 +0,0 @@
-/*
- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "test/fuzzers/fuzz_data_helper.h"
-
-#include <cstdint>
-
-#include "api/array_view.h"
-
-namespace webrtc {
-namespace test {
-
-FuzzDataHelper::FuzzDataHelper(ArrayView<const uint8_t> data) : data_(data) {}
-
-}  // namespace test
-}  // namespace webrtc
diff --git a/test/fuzzers/fuzz_data_helper.h b/test/fuzzers/fuzz_data_helper.h
index 42a469c74b..0e6486903b 100644
--- a/test/fuzzers/fuzz_data_helper.h
+++ b/test/fuzzers/fuzz_data_helper.h
@@ -15,20 +15,20 @@
 #include <cstddef>
 #include <cstdint>
 #include <limits>
+#include <span>
 #include <type_traits>
 
-#include "api/array_view.h"
+#include "absl/strings/string_view.h"
 #include "modules/rtp_rtcp/source/byte_io.h"
 #include "rtc_base/checks.h"
 
 namespace webrtc {
-namespace test {
 
 // Helper class to take care of the fuzzer input, read from it, and keep track
 // of when the end of the data has been reached.
 class FuzzDataHelper {
  public:
-  explicit FuzzDataHelper(ArrayView<const uint8_t> data);
+  explicit FuzzDataHelper(std::span<const uint8_t> data) : data_(data) {}
 
   // Returns true if n bytes can be read.
   bool CanReadBytes(size_t n) const { return data_ix_ + n <= data_.size(); }
@@ -84,7 +84,7 @@ class FuzzDataHelper {
     return std::move(select_from[index]);
   }
 
-  ArrayView<const uint8_t> ReadByteArray(size_t bytes) {
+  std::span<const uint8_t> ReadByteArray(size_t bytes) {
     if (!CanReadBytes(bytes)) {
       return {};
     }
@@ -93,6 +93,20 @@ class FuzzDataHelper {
     return data_.subspan(index_to_return, bytes);
   }
 
+  // Returns all unused fuzzing bytes. May return an empty view.
+  std::span<const uint8_t> ReadRemaining() {
+    std::span<const uint8_t> result = data_.subspan(data_ix_);
+    data_ix_ = data_.size();
+    return result;
+  }
+
+  // Returns all unused fuzzing bytes as a string.
+  absl::string_view ReadString() {
+    std::span<const uint8_t> raw = ReadRemaining();
+    return absl::string_view(reinterpret_cast<const char*>(raw.data()),
+                             raw.size());
+  }
+
   // If sizeof(T) > BytesLeft then the remaining bytes will be used and the rest
   // of the object will be zero initialized.
   template <typename T>
@@ -108,12 +122,13 @@ class FuzzDataHelper {
 
   size_t BytesLeft() const { return data_.size() - data_ix_; }
 
+  size_t size() const { return data_.size(); }
+
  private:
-  ArrayView<const uint8_t> data_;
+  std::span<const uint8_t> data_;
   size_t data_ix_ = 0;
 };
 
-}  // namespace test
 }  // namespace webrtc
 
 #endif  // TEST_FUZZERS_FUZZ_DATA_HELPER_H_
diff --git a/test/fuzzers/h264_bitstream_parser_fuzzer.cc b/test/fuzzers/h264_bitstream_parser_fuzzer.cc
index 4122fea08b..c0dabaf58d 100644
--- a/test/fuzzers/h264_bitstream_parser_fuzzer.cc
+++ b/test/fuzzers/h264_bitstream_parser_fuzzer.cc
@@ -11,14 +11,13 @@
 
 #include <cstddef>
 
-#include "api/array_view.h"
 #include "common_video/h264/h264_bitstream_parser.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
-void FuzzOneInput(const uint8_t* data, size_t size) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
   H264BitstreamParser h264_bitstream_parser;
-  h264_bitstream_parser.ParseBitstream(
-      webrtc::ArrayView<const uint8_t>(data, size));
+  h264_bitstream_parser.ParseBitstream(fuzz_data.ReadRemaining());
   h264_bitstream_parser.GetLastSliceQp();
 }
 }  // namespace webrtc
diff --git a/test/fuzzers/h264_depacketizer_fuzzer.cc b/test/fuzzers/h264_depacketizer_fuzzer.cc
index 59e232e889..23075a797f 100644
--- a/test/fuzzers/h264_depacketizer_fuzzer.cc
+++ b/test/fuzzers/h264_depacketizer_fuzzer.cc
@@ -8,16 +8,16 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #include <cstddef>
-#include <cstdint>
 
 #include "modules/rtp_rtcp/source/video_rtp_depacketizer_h264.h"
 #include "rtc_base/copy_on_write_buffer.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  if (size > 200000)
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
+  if (fuzz_data.size() > 200'000)
     return;
   VideoRtpDepacketizerH264 depacketizer;
-  depacketizer.Parse(webrtc::CopyOnWriteBuffer(data, size));
+  depacketizer.Parse(webrtc::CopyOnWriteBuffer(fuzz_data.ReadRemaining()));
 }
 }  // namespace webrtc
diff --git a/test/fuzzers/h265_bitstream_parser_fuzzer.cc b/test/fuzzers/h265_bitstream_parser_fuzzer.cc
index 83ec007bb5..2b5a3d42b0 100644
--- a/test/fuzzers/h265_bitstream_parser_fuzzer.cc
+++ b/test/fuzzers/h265_bitstream_parser_fuzzer.cc
@@ -11,14 +11,13 @@
 
 #include <cstddef>
 
-#include "api/array_view.h"
 #include "common_video/h265/h265_bitstream_parser.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
-void FuzzOneInput(const uint8_t* data, size_t size) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
   H265BitstreamParser h265_bitstream_parser;
-  h265_bitstream_parser.ParseBitstream(
-      webrtc::ArrayView<const uint8_t>(data, size));
+  h265_bitstream_parser.ParseBitstream(fuzz_data.ReadRemaining());
   h265_bitstream_parser.GetLastSliceQp();
 }
 }  // namespace webrtc
diff --git a/test/fuzzers/h265_depacketizer_fuzzer.cc b/test/fuzzers/h265_depacketizer_fuzzer.cc
index 22835e0fa1..76d8ba8c8b 100644
--- a/test/fuzzers/h265_depacketizer_fuzzer.cc
+++ b/test/fuzzers/h265_depacketizer_fuzzer.cc
@@ -8,16 +8,16 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #include <cstddef>
-#include <cstdint>
 
 #include "modules/rtp_rtcp/source/video_rtp_depacketizer_h265.h"
 #include "rtc_base/copy_on_write_buffer.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  if (size > 200000)
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
+  if (fuzz_data.size() > 200'000)
     return;
   VideoRtpDepacketizerH265 depacketizer;
-  depacketizer.Parse(webrtc::CopyOnWriteBuffer(data, size));
+  depacketizer.Parse(CopyOnWriteBuffer(fuzz_data.ReadRemaining()));
 }
 }  // namespace webrtc
diff --git a/test/fuzzers/neteq_rtp_fuzzer.cc b/test/fuzzers/neteq_rtp_fuzzer.cc
index f7ba929a49..b62e580650 100644
--- a/test/fuzzers/neteq_rtp_fuzzer.cc
+++ b/test/fuzzers/neteq_rtp_fuzzer.cc
@@ -31,6 +31,7 @@
 #include "modules/rtp_rtcp/source/byte_io.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "rtc_base/checks.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
 namespace test {
@@ -171,9 +172,8 @@ class FuzzRtpInput : public NetEqInput {
 };
 }  // namespace
 
-void FuzzOneInputTest(const uint8_t* data, size_t size) {
-  std::unique_ptr<FuzzRtpInput> input(
-      new FuzzRtpInput(webrtc::ArrayView<const uint8_t>(data, size)));
+void FuzzOneInputTest(FuzzDataHelper fuzz_data) {
+  auto input = std::make_unique<FuzzRtpInput>(fuzz_data.ReadRemaining());
   std::unique_ptr<AudioChecksum> output(new AudioChecksum);
   NetEqTest::Callbacks callbacks;
   NetEq::Config config;
@@ -193,11 +193,11 @@ void FuzzOneInputTest(const uint8_t* data, size_t size) {
 
 }  // namespace test
 
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  if (size > 70000) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
+  if (fuzz_data.size() > 70'000) {
     return;
   }
-  test::FuzzOneInputTest(data, size);
+  test::FuzzOneInputTest(fuzz_data);
 }
 
 }  // namespace webrtc
diff --git a/test/fuzzers/neteq_signal_fuzzer.cc b/test/fuzzers/neteq_signal_fuzzer.cc
index 8bd00dc7ca..477227c386 100644
--- a/test/fuzzers/neteq_signal_fuzzer.cc
+++ b/test/fuzzers/neteq_signal_fuzzer.cc
@@ -171,13 +171,11 @@ bool MapHas(const std::map<int, T>& m, int key, const T& value) {
 
 }  // namespace
 
-void FuzzOneInputTest(const uint8_t* data, size_t size) {
-  if (size < 1 || size > 65000) {
+void FuzzOneInputTest(FuzzDataHelper fuzz_data) {
+  if (fuzz_data.size() < 1 || fuzz_data.size() > 65'000) {
     return;
   }
 
-  FuzzDataHelper fuzz_data(webrtc::ArrayView<const uint8_t>(data, size));
-
   // Allowed sample rates and payload types used in the test.
   std::pair<int, uint8_t> rate_types[] = {
       {8000, 93}, {16000, 94}, {32000, 95}, {48000, 96}};
@@ -217,8 +215,8 @@ void FuzzOneInputTest(const uint8_t* data, size_t size) {
 
 }  // namespace test
 
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  test::FuzzOneInputTest(data, size);
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
+  test::FuzzOneInputTest(fuzz_data);
 }
 
 }  // namespace webrtc
diff --git a/test/fuzzers/receive_side_congestion_controller_fuzzer.cc b/test/fuzzers/receive_side_congestion_controller_fuzzer.cc
index 94c5f1deab..c3120f592b 100644
--- a/test/fuzzers/receive_side_congestion_controller_fuzzer.cc
+++ b/test/fuzzers/receive_side_congestion_controller_fuzzer.cc
@@ -12,21 +12,20 @@
 #include <cstddef>
 #include <cstdint>
 
-#include "api/array_view.h"
 #include "api/environment/environment_factory.h"
 #include "api/media_types.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
 #include "modules/congestion_controller/include/receive_side_congestion_controller.h"
 #include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
-#include "modules/rtp_rtcp/source/byte_io.h"
 #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "system_wrappers/include/clock.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
 
-void FuzzOneInput(const uint8_t* data, size_t size) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
   Timestamp arrival_time = Timestamp::Micros(123'456'789);
   SimulatedClock clock(arrival_time);
   ReceiveSideCongestionController cc(
@@ -41,17 +40,12 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
   RtpPacketReceived rtp_packet(&extensions);
 
   constexpr int kMinPacketSize = sizeof(uint16_t) + sizeof(uint8_t) + 12;
-  const uint8_t* const end_data = data + size;
-  while (end_data - data >= kMinPacketSize) {
-    size_t packet_size = ByteReader<uint16_t>::ReadBigEndian(data) % 1500;
-    data += sizeof(uint16_t);
-    arrival_time += TimeDelta::Millis(ByteReader<uint8_t>::ReadBigEndian(data));
-    data += sizeof(uint8_t);
-    packet_size = std::min<size_t>(end_data - data, packet_size);
-    auto raw_packet = webrtc::MakeArrayView(data, packet_size);
-    data += packet_size;
+  while (fuzz_data.BytesLeft() >= kMinPacketSize) {
+    size_t packet_size = fuzz_data.Read<uint16_t>() % 1500;
+    arrival_time += TimeDelta::Millis(fuzz_data.Read<uint8_t>());
+    packet_size = std::min<size_t>(fuzz_data.BytesLeft(), packet_size);
 
-    if (!rtp_packet.Parse(raw_packet)) {
+    if (!rtp_packet.Parse(fuzz_data.ReadByteArray(packet_size))) {
       continue;
     }
     rtp_packet.set_arrival_time(arrival_time);
diff --git a/test/fuzzers/residual_echo_detector_fuzzer.cc b/test/fuzzers/residual_echo_detector_fuzzer.cc
index 18b48e0f8f..b191a647d7 100644
--- a/test/fuzzers/residual_echo_detector_fuzzer.cc
+++ b/test/fuzzers/residual_echo_detector_fuzzer.cc
@@ -13,16 +13,15 @@
 
 #include <bitset>
 #include <cstdint>
-#include <vector>
 
 #include "api/audio/audio_processing.h"
 #include "api/audio/echo_detector_creator.h"
 #include "api/scoped_refptr.h"
-#include "rtc_base/checks.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
 
-void FuzzOneInput(const uint8_t* data, size_t size) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
   // Number of times to update the echo detector.
   constexpr size_t kNrOfUpdates = 7;
   // Each round of updates requires a call to both AnalyzeRender and
@@ -34,25 +33,20 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
   // to represent an audio frame in this test, so it should have a maximum value
   // equal to the square root of that value.
   const float maxFuzzedValue = sqrtf(20 * 48) * 32768;
-  if (size < kNrOfNeededInputBytes) {
+  if (fuzz_data.size() < kNrOfNeededInputBytes) {
     return;
   }
-  size_t read_idx = 0;
   // Use the first two bytes to choose the call order.
-  uint16_t call_order_int;
-  memcpy(&call_order_int, &data[read_idx], 2);
-  read_idx += 2;
+  uint16_t call_order_int = fuzz_data.Read<uint16_t>();
   std::bitset<16> call_order(call_order_int);
 
   webrtc::scoped_refptr<EchoDetector> echo_detector = CreateEchoDetector();
-  std::vector<float> input(1);
+  float input[1];
   // Call AnalyzeCaptureAudio once to prevent the flushing of the buffer.
   echo_detector->AnalyzeCaptureAudio(input);
   for (size_t i = 0; i < 2 * kNrOfUpdates; ++i) {
     // Convert 4 input bytes to a float.
-    RTC_DCHECK_LE(read_idx + sizeof(float), size);
-    memcpy(input.data(), &data[read_idx], sizeof(float));
-    read_idx += sizeof(float);
+    fuzz_data.CopyTo(&input[0]);
     if (!isfinite(input[0]) || fabs(input[0]) > maxFuzzedValue) {
       // Ignore infinity, nan values and values that are unrealistically large.
       continue;
diff --git a/test/fuzzers/rtcp_receiver_fuzzer.cc b/test/fuzzers/rtcp_receiver_fuzzer.cc
index 98fdbfc45b..9eebc417ce 100644
--- a/test/fuzzers/rtcp_receiver_fuzzer.cc
+++ b/test/fuzzers/rtcp_receiver_fuzzer.cc
@@ -20,6 +20,7 @@
 #include "modules/rtp_rtcp/source/rtcp_receiver.h"
 #include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
 #include "system_wrappers/include/clock.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
 namespace {
@@ -41,8 +42,8 @@ class NullModuleRtpRtcp : public RTCPReceiver::ModuleRtpRtcp {
 
 }  // namespace
 
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  if (size > kMaxInputLenBytes) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
+  if (fuzz_data.size() > kMaxInputLenBytes) {
     return;
   }
   FieldTrials field_trials("WebRTC-RFC8888CongestionControlFeedback/Enabled/");
@@ -56,6 +57,6 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
   RTCPReceiver receiver(CreateEnvironment(&clock, &field_trials), config,
                         &rtp_rtcp_module);
 
-  receiver.IncomingPacket(webrtc::MakeArrayView(data, size));
+  receiver.IncomingPacket(fuzz_data.ReadRemaining());
 }
 }  // namespace webrtc
diff --git a/test/fuzzers/rtp_depacketizer_av1_assemble_frame_fuzzer.cc b/test/fuzzers/rtp_depacketizer_av1_assemble_frame_fuzzer.cc
index f59a6ce2dc..d4049b93a0 100644
--- a/test/fuzzers/rtp_depacketizer_av1_assemble_frame_fuzzer.cc
+++ b/test/fuzzers/rtp_depacketizer_av1_assemble_frame_fuzzer.cc
@@ -17,20 +17,19 @@
 #include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
-void FuzzOneInput(const uint8_t* data, size_t size) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
   std::vector<webrtc::ArrayView<const uint8_t>> rtp_payloads;
 
   // Convert plain array of bytes into array of array bytes.
-  test::FuzzDataHelper fuzz_input(webrtc::MakeArrayView(data, size));
-  while (fuzz_input.CanReadBytes(sizeof(uint16_t))) {
+  while (fuzz_data.CanReadBytes(sizeof(uint16_t))) {
     // In practice one rtp payload can be up to ~1200 - 1500 bytes. Majority
     // of the payload is just copied. To make fuzzing more efficient limit the
     // size of rtp payload to realistic value.
-    uint16_t next_size = fuzz_input.Read<uint16_t>() % 1200;
-    if (next_size > fuzz_input.BytesLeft()) {
-      next_size = fuzz_input.BytesLeft();
+    uint16_t next_size = fuzz_data.Read<uint16_t>() % 1200;
+    if (next_size > fuzz_data.BytesLeft()) {
+      next_size = fuzz_data.BytesLeft();
     }
-    rtp_payloads.push_back(fuzz_input.ReadByteArray(next_size));
+    rtp_payloads.push_back(fuzz_data.ReadByteArray(next_size));
   }
   // Run code under test.
   VideoRtpDepacketizerAv1().AssembleFrame(rtp_payloads);
diff --git a/test/fuzzers/rtp_dependency_descriptor_fuzzer.cc b/test/fuzzers/rtp_dependency_descriptor_fuzzer.cc
index a96d80045d..55d6ba8a37 100644
--- a/test/fuzzers/rtp_dependency_descriptor_fuzzer.cc
+++ b/test/fuzzers/rtp_dependency_descriptor_fuzzer.cc
@@ -22,13 +22,12 @@
 
 namespace webrtc {
 
-void FuzzOneInput(const uint8_t* data, size_t size) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
   FrameDependencyStructure structure1;
   // nullptr during 1st while loop, after that should point to structure1.
   const FrameDependencyStructure* structure1_ptr = nullptr;
   std::unique_ptr<const FrameDependencyStructure> structure2;
 
-  test::FuzzDataHelper fuzz_data(webrtc::MakeArrayView(data, size));
   while (fuzz_data.CanReadBytes(1)) {
     // Treat next byte as size of the next extension. That aligns how
     // two-byte rtp header extension sizes are written.
diff --git a/test/fuzzers/rtp_format_h264_fuzzer.cc b/test/fuzzers/rtp_format_h264_fuzzer.cc
index c5137e8c21..84b643184f 100644
--- a/test/fuzzers/rtp_format_h264_fuzzer.cc
+++ b/test/fuzzers/rtp_format_h264_fuzzer.cc
@@ -10,7 +10,6 @@
 #include <stddef.h>
 #include <stdint.h>
 
-#include "api/array_view.h"
 #include "modules/rtp_rtcp/source/rtp_format.h"
 #include "modules/rtp_rtcp/source/rtp_format_h264.h"
 #include "modules/video_coding/codecs/h264/include/h264_globals.h"
@@ -18,21 +17,19 @@
 #include "test/fuzzers/utils/validate_rtp_packetizer.h"
 
 namespace webrtc {
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  test::FuzzDataHelper fuzz_input(MakeArrayView(data, size));
-
-  RtpPacketizer::PayloadSizeLimits limits = ReadPayloadSizeLimits(fuzz_input);
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
+  RtpPacketizer::PayloadSizeLimits limits = ReadPayloadSizeLimits(fuzz_data);
 
   const H264PacketizationMode kPacketizationModes[] = {
       H264PacketizationMode::NonInterleaved,
       H264PacketizationMode::SingleNalUnit};
 
   H264PacketizationMode packetization_mode =
-      fuzz_input.SelectOneOf(kPacketizationModes);
+      fuzz_data.SelectOneOf(kPacketizationModes);
 
   // Main function under test: RtpPacketizerH264's constructor.
-  RtpPacketizerH264 packetizer(fuzz_input.ReadByteArray(fuzz_input.BytesLeft()),
-                               limits, packetization_mode);
+  RtpPacketizerH264 packetizer(fuzz_data.ReadRemaining(), limits,
+                               packetization_mode);
 
   ValidateRtpPacketizer(limits, packetizer);
 }
diff --git a/test/fuzzers/rtp_format_vp8_fuzzer.cc b/test/fuzzers/rtp_format_vp8_fuzzer.cc
index b4998ed7f3..37f7cc82ba 100644
--- a/test/fuzzers/rtp_format_vp8_fuzzer.cc
+++ b/test/fuzzers/rtp_format_vp8_fuzzer.cc
@@ -10,7 +10,6 @@
 #include <stddef.h>
 #include <stdint.h>
 
-#include "api/array_view.h"
 #include "modules/rtp_rtcp/source/rtp_format.h"
 #include "modules/rtp_rtcp/source/rtp_format_vp8.h"
 #include "modules/video_coding/codecs/interface/common_constants.h"
@@ -19,20 +18,17 @@
 #include "test/fuzzers/utils/validate_rtp_packetizer.h"
 
 namespace webrtc {
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  test::FuzzDataHelper fuzz_input(webrtc::MakeArrayView(data, size));
-
-  RtpPacketizer::PayloadSizeLimits limits = ReadPayloadSizeLimits(fuzz_input);
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
+  RtpPacketizer::PayloadSizeLimits limits = ReadPayloadSizeLimits(fuzz_data);
 
   RTPVideoHeaderVP8 hdr_info;
   hdr_info.InitRTPVideoHeaderVP8();
-  uint16_t picture_id = fuzz_input.ReadOrDefaultValue<uint16_t>(0);
+  uint16_t picture_id = fuzz_data.ReadOrDefaultValue<uint16_t>(0);
   hdr_info.pictureId =
       picture_id >= 0x8000 ? kNoPictureId : picture_id & 0x7fff;
 
   // Main function under test: RtpPacketizerVp8's constructor.
-  RtpPacketizerVp8 packetizer(fuzz_input.ReadByteArray(fuzz_input.BytesLeft()),
-                              limits, hdr_info);
+  RtpPacketizerVp8 packetizer(fuzz_data.ReadRemaining(), limits, hdr_info);
 
   ValidateRtpPacketizer(limits, packetizer);
 }
diff --git a/test/fuzzers/rtp_format_vp9_fuzzer.cc b/test/fuzzers/rtp_format_vp9_fuzzer.cc
index 5f0f664486..eda3b7d53d 100644
--- a/test/fuzzers/rtp_format_vp9_fuzzer.cc
+++ b/test/fuzzers/rtp_format_vp9_fuzzer.cc
@@ -10,7 +10,6 @@
 #include <stddef.h>
 #include <stdint.h>
 
-#include "api/array_view.h"
 #include "modules/rtp_rtcp/source/rtp_format.h"
 #include "modules/rtp_rtcp/source/rtp_format_vp9.h"
 #include "modules/video_coding/codecs/interface/common_constants.h"
@@ -19,20 +18,17 @@
 #include "test/fuzzers/utils/validate_rtp_packetizer.h"
 
 namespace webrtc {
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  test::FuzzDataHelper fuzz_input(MakeArrayView(data, size));
-
-  RtpPacketizer::PayloadSizeLimits limits = ReadPayloadSizeLimits(fuzz_input);
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
+  RtpPacketizer::PayloadSizeLimits limits = ReadPayloadSizeLimits(fuzz_data);
 
   RTPVideoHeaderVP9 hdr_info;
   hdr_info.InitRTPVideoHeaderVP9();
-  uint16_t picture_id = fuzz_input.ReadOrDefaultValue<uint16_t>(0);
+  uint16_t picture_id = fuzz_data.ReadOrDefaultValue<uint16_t>(0);
   hdr_info.picture_id =
       picture_id >= 0x8000 ? kNoPictureId : picture_id & 0x7fff;
 
   // Main function under test: RtpPacketizerVp9's constructor.
-  RtpPacketizerVp9 packetizer(fuzz_input.ReadByteArray(fuzz_input.BytesLeft()),
-                              limits, hdr_info);
+  RtpPacketizerVp9 packetizer(fuzz_data.ReadRemaining(), limits, hdr_info);
 
   ValidateRtpPacketizer(limits, packetizer);
 }
diff --git a/test/fuzzers/rtp_frame_reference_finder_fuzzer.cc b/test/fuzzers/rtp_frame_reference_finder_fuzzer.cc
index c8c0f6fd7b..4e1da6040d 100644
--- a/test/fuzzers/rtp_frame_reference_finder_fuzzer.cc
+++ b/test/fuzzers/rtp_frame_reference_finder_fuzzer.cc
@@ -8,7 +8,6 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include <algorithm>
 #include <cstddef>
 #include <cstdint>
 #include <cstring>
@@ -29,47 +28,33 @@
 #include "modules/video_coding/codecs/vp8/include/vp8_globals.h"
 #include "modules/video_coding/codecs/vp9/include/vp9_globals.h"
 #include "modules/video_coding/rtp_frame_reference_finder.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
 
 namespace {
 class DataReader {
  public:
-  DataReader(const uint8_t* data, size_t size) : data_(data), size_(size) {}
+  DataReader(FuzzDataHelper fuzz_data) : data_(fuzz_data) {}
 
   template <typename T>
   void CopyTo(T* object) {
-    static_assert(std::is_trivial_v<T> && std::is_standard_layout_v<T>, "");
-    uint8_t* destination = reinterpret_cast<uint8_t*>(object);
-    size_t object_size = sizeof(T);
-    size_t num_bytes = std::min(size_ - offset_, object_size);
-    memcpy(destination, data_ + offset_, num_bytes);
-    offset_ += num_bytes;
-
-    // If we did not have enough data, fill the rest with 0.
-    object_size -= num_bytes;
-    memset(destination + num_bytes, 0, object_size);
+    return data_.CopyTo(object);
   }
 
   template <typename T>
   T GetNum() {
-    T res;
-    if (offset_ + sizeof(res) < size_) {
-      memcpy(&res, data_ + offset_, sizeof(res));
-      offset_ += sizeof(res);
-      return res;
+    T t = {};
+    if (data_.BytesLeft() >= sizeof(T)) {
+      CopyTo(&t);
     }
-
-    offset_ = size_;
-    return T(0);
+    return t;
   }
 
-  bool MoreToRead() { return offset_ < size_; }
+  bool MoreToRead() { return data_.BytesLeft() > 0; }
 
  private:
-  const uint8_t* data_;
-  size_t size_;
-  size_t offset_ = 0;
+  FuzzDataHelper data_;
 };
 
 RTPVideoHeaderH264 GenerateRTPVideoHeaderH264(DataReader* reader) {
@@ -111,8 +96,8 @@ GenerateGenericFrameDependencies(DataReader* reader) {
 }
 }  // namespace
 
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  DataReader reader(data, size);
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
+  DataReader reader(fuzz_data);
   RtpFrameReferenceFinder reference_finder;
 
   auto codec = static_cast<VideoCodecType>(reader.GetNum<uint8_t>() % 5);
diff --git a/test/fuzzers/rtp_packet_fuzzer.cc b/test/fuzzers/rtp_packet_fuzzer.cc
index 2b9c95a5b7..aa2f819934 100644
--- a/test/fuzzers/rtp_packet_fuzzer.cc
+++ b/test/fuzzers/rtp_packet_fuzzer.cc
@@ -28,6 +28,7 @@
 #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "modules/rtp_rtcp/source/rtp_video_layers_allocation_extension.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
 // We decide which header extensions to register by reading four bytes
@@ -37,14 +38,12 @@ static_assert(kRtpExtensionNumberOfExtensions <= 32,
               "Insufficient bits read to configure all header extensions. Add "
               "an extra byte and update the switches.");
 
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  if (size <= 4)
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
+  if (fuzz_data.size() <= 4)
     return;
 
   // Don't use the configuration byte as part of the packet.
-  std::bitset<32> extensionMask(*reinterpret_cast<const uint32_t*>(data));
-  data += 4;
-  size -= 4;
+  std::bitset<32> extensionMask(fuzz_data.Read<uint32_t>());
 
   RtpPacketReceived::ExtensionManager extensions(/*extmap_allow_mixed=*/true);
   // Start at local_id = 1 since 0 is an invalid extension id.
@@ -61,7 +60,7 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
   }
 
   RtpPacketReceived packet(&extensions);
-  packet.Parse(data, size);
+  packet.Parse(fuzz_data.ReadRemaining());
 
   // Call packet accessors because they have extra checks.
   packet.Marker();
diff --git a/test/fuzzers/rtp_packetizer_av1_fuzzer.cc b/test/fuzzers/rtp_packetizer_av1_fuzzer.cc
index 99ba7d58df..8bfad07ab3 100644
--- a/test/fuzzers/rtp_packetizer_av1_fuzzer.cc
+++ b/test/fuzzers/rtp_packetizer_av1_fuzzer.cc
@@ -10,7 +10,6 @@
 #include <stddef.h>
 #include <stdint.h>
 
-#include "api/array_view.h"
 #include "api/video/video_frame_type.h"
 #include "modules/rtp_rtcp/source/rtp_format.h"
 #include "modules/rtp_rtcp/source/rtp_packetizer_av1.h"
@@ -18,18 +17,15 @@
 #include "test/fuzzers/utils/validate_rtp_packetizer.h"
 
 namespace webrtc {
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  test::FuzzDataHelper fuzz_input(MakeArrayView(data, size));
-
-  RtpPacketizer::PayloadSizeLimits limits = ReadPayloadSizeLimits(fuzz_input);
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
+  RtpPacketizer::PayloadSizeLimits limits = ReadPayloadSizeLimits(fuzz_data);
 
   const VideoFrameType kFrameTypes[] = {VideoFrameType::kVideoFrameKey,
                                         VideoFrameType::kVideoFrameDelta};
-  VideoFrameType frame_type = fuzz_input.SelectOneOf(kFrameTypes);
+  VideoFrameType frame_type = fuzz_data.SelectOneOf(kFrameTypes);
 
   // Main function under test: RtpPacketizerAv1's constructor.
-  RtpPacketizerAv1 packetizer(fuzz_input.ReadByteArray(fuzz_input.BytesLeft()),
-                              limits, frame_type,
+  RtpPacketizerAv1 packetizer(fuzz_data.ReadRemaining(), limits, frame_type,
                               /*is_last_frame_in_picture=*/true);
 
   ValidateRtpPacketizer(limits, packetizer);
diff --git a/test/fuzzers/rtp_packetizer_h265_fuzzer.cc b/test/fuzzers/rtp_packetizer_h265_fuzzer.cc
index a65318da6e..6deb7cf807 100644
--- a/test/fuzzers/rtp_packetizer_h265_fuzzer.cc
+++ b/test/fuzzers/rtp_packetizer_h265_fuzzer.cc
@@ -8,22 +8,19 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #include <cstddef>
-#include <cstdint>
 
+#include "modules/rtp_rtcp/source/rtp_format.h"
 #include "modules/rtp_rtcp/source/rtp_packetizer_h265.h"
 #include "test/fuzzers/fuzz_data_helper.h"
 #include "test/fuzzers/utils/validate_rtp_packetizer.h"
 
 namespace webrtc {
 
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  test::FuzzDataHelper fuzz_input(MakeArrayView(data, size));
-
-  RtpPacketizer::PayloadSizeLimits limits = ReadPayloadSizeLimits(fuzz_input);
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
+  RtpPacketizer::PayloadSizeLimits limits = ReadPayloadSizeLimits(fuzz_data);
 
   // Main function under test: RtpPacketizerH265's constructor.
-  RtpPacketizerH265 packetizer(fuzz_input.ReadByteArray(fuzz_input.BytesLeft()),
-                               limits);
+  RtpPacketizerH265 packetizer(fuzz_data.ReadRemaining(), limits);
 
   ValidateRtpPacketizer(limits, packetizer);
 }
diff --git a/test/fuzzers/rtp_video_frame_assembler_fuzzer.cc b/test/fuzzers/rtp_video_frame_assembler_fuzzer.cc
index 6ab6b9a905..27819e4a3d 100644
--- a/test/fuzzers/rtp_video_frame_assembler_fuzzer.cc
+++ b/test/fuzzers/rtp_video_frame_assembler_fuzzer.cc
@@ -17,11 +17,12 @@
 #include "modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h"
 #include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
 
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  if (size == 0) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
+  if (fuzz_data.size() == 0) {
     return;
   }
   RtpHeaderExtensionMap extensions;
@@ -30,14 +31,14 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
   RtpPacketReceived rtp_packet(&extensions);
 
   RtpVideoFrameAssembler assembler(
-      static_cast<RtpVideoFrameAssembler::PayloadFormat>(data[0] % 6));
+      static_cast<RtpVideoFrameAssembler::PayloadFormat>(
+          fuzz_data.Read<uint8_t>() % 6));
 
-  for (size_t i = 1; i < size;) {
-    size_t packet_size = std::min<size_t>(size - i, 300);
-    if (rtp_packet.Parse(data + i, packet_size)) {
+  while (fuzz_data.BytesLeft() > 0) {
+    size_t packet_size = std::min<size_t>(fuzz_data.BytesLeft(), 300);
+    if (rtp_packet.Parse(fuzz_data.ReadByteArray(packet_size))) {
       assembler.InsertPacket(rtp_packet);
     }
-    i += packet_size;
   }
 }
 
diff --git a/test/fuzzers/rtp_video_layers_allocation_fuzzer.cc b/test/fuzzers/rtp_video_layers_allocation_fuzzer.cc
index c6f557f145..cf5b4fe3fb 100644
--- a/test/fuzzers/rtp_video_layers_allocation_fuzzer.cc
+++ b/test/fuzzers/rtp_video_layers_allocation_fuzzer.cc
@@ -16,19 +16,20 @@
 #include "api/video/video_layers_allocation.h"
 #include "modules/rtp_rtcp/source/rtp_video_layers_allocation_extension.h"
 #include "rtc_base/checks.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
 
-void FuzzOneInput(const uint8_t* data, size_t size) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
   // Video layers allocation is an rtp header extension.
   // Per https://datatracker.ietf.org/doc/html/rfc8285#section-4.3
   // rtp header extension uses up to one byte to store the size, i.e.
   // maximum size of any rtp header extension is 255 bytes.
   constexpr int kMaxSize = std::numeric_limits<uint8_t>::max();
-  if (size > kMaxSize) {
+  if (fuzz_data.size() > kMaxSize) {
     return;
   }
-  auto raw = webrtc::MakeArrayView(data, size);
+  ArrayView<const uint8_t> raw = fuzz_data.ReadRemaining();
 
   VideoLayersAllocation allocation1;
   if (!RtpVideoLayersAllocationExtension::Parse(raw, &allocation1)) {
diff --git a/test/fuzzers/sctp_utils_fuzzer.cc b/test/fuzzers/sctp_utils_fuzzer.cc
index 90d107ad33..a867bc8e51 100644
--- a/test/fuzzers/sctp_utils_fuzzer.cc
+++ b/test/fuzzers/sctp_utils_fuzzer.cc
@@ -16,11 +16,12 @@
 #include "api/data_channel_interface.h"
 #include "pc/sctp_utils.h"
 #include "rtc_base/copy_on_write_buffer.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
 
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  webrtc::CopyOnWriteBuffer payload(data, size);
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
+  CopyOnWriteBuffer payload(fuzz_data.ReadRemaining());
   std::string label;
   DataChannelInit config;
   IsOpenMessage(payload);
diff --git a/test/fuzzers/sdp_integration_fuzzer.cc b/test/fuzzers/sdp_integration_fuzzer.cc
index c3ff73e0af..002e194e44 100644
--- a/test/fuzzers/sdp_integration_fuzzer.cc
+++ b/test/fuzzers/sdp_integration_fuzzer.cc
@@ -23,6 +23,7 @@
 #include "pc/test/integration_test_helpers.h"
 #include "pc/test/mock_peer_connection_observers.h"
 #include "rtc_base/checks.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 #include "test/gmock.h"
 #include "test/wait_until.h"
 
@@ -69,14 +70,13 @@ class FuzzerTest : public PeerConnectionIntegrationBaseTest {
   void TestBody() override {}
 };
 
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  if (size > 16384) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
+  if (fuzz_data.size() > 16384) {
     return;
   }
 
   FuzzerTest test;
-  test.RunNegotiateCycle(
-      absl::string_view(reinterpret_cast<const char*>(data), size));
+  test.RunNegotiateCycle(fuzz_data.ReadString());
 }
 
 }  // namespace webrtc
diff --git a/test/fuzzers/sdp_parser_fuzzer.cc b/test/fuzzers/sdp_parser_fuzzer.cc
index 04ee4af7bb..bf36e90918 100644
--- a/test/fuzzers/sdp_parser_fuzzer.cc
+++ b/test/fuzzers/sdp_parser_fuzzer.cc
@@ -15,13 +15,14 @@
 #include <string>
 
 #include "api/jsep.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  if (size > 16384) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
+  if (fuzz_data.size() > 16384) {
     return;
   }
-  std::string message(reinterpret_cast<const char*>(data), size);
+  std::string message(fuzz_data.ReadString());
   webrtc::SdpParseError error;
 
   std::unique_ptr<webrtc::SessionDescriptionInterface> sdp =
diff --git a/test/fuzzers/ssl_certificate_fuzzer.cc b/test/fuzzers/ssl_certificate_fuzzer.cc
index f056994f8b..f89bd16ea1 100644
--- a/test/fuzzers/ssl_certificate_fuzzer.cc
+++ b/test/fuzzers/ssl_certificate_fuzzer.cc
@@ -17,14 +17,13 @@
 #include "rtc_base/buffer.h"
 #include "rtc_base/message_digest.h"
 #include "rtc_base/ssl_certificate.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
 
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  std::string pem_certificate(reinterpret_cast<const char*>(data), size);
-
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
   std::unique_ptr<SSLCertificate> cert =
-      SSLCertificate::FromPEMString(pem_certificate);
+      SSLCertificate::FromPEMString(fuzz_data.ReadString());
 
   if (cert == nullptr) {
     return;
diff --git a/test/fuzzers/string_to_number_fuzzer.cc b/test/fuzzers/string_to_number_fuzzer.cc
index 4d619c5a5e..96ef2e6070 100644
--- a/test/fuzzers/string_to_number_fuzzer.cc
+++ b/test/fuzzers/string_to_number_fuzzer.cc
@@ -11,14 +11,14 @@
 #include <stddef.h>
 #include <stdint.h>
 
-#include <string>
-
+#include "absl/strings/string_view.h"
 #include "rtc_base/string_to_number.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
 
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  std::string number_to_parse(reinterpret_cast<const char*>(data), size);
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
+  absl::string_view number_to_parse = fuzz_data.ReadString();
   webrtc::StringToNumber<int8_t>(number_to_parse);
   webrtc::StringToNumber<int16_t>(number_to_parse);
   webrtc::StringToNumber<int32_t>(number_to_parse);
diff --git a/test/fuzzers/stun_parser_fuzzer.cc b/test/fuzzers/stun_parser_fuzzer.cc
index 5a3188d201..88659519b5 100644
--- a/test/fuzzers/stun_parser_fuzzer.cc
+++ b/test/fuzzers/stun_parser_fuzzer.cc
@@ -13,18 +13,18 @@
 
 #include <memory>
 
-#include "api/array_view.h"
 #include "api/transport/stun.h"
 #include "rtc_base/byte_buffer.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
-void FuzzOneInput(const uint8_t* data, size_t size) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
   // Normally we'd check the integrity first, but those checks are
   // fuzzed separately in stun_validator_fuzzer.cc. We still want to
   // fuzz this target since the integrity checks could be forged by a
   // malicious adversary who receives a call.
   std::unique_ptr<webrtc::IceMessage> stun_msg(new webrtc::IceMessage());
-  webrtc::ByteBufferReader buf(webrtc::MakeArrayView(data, size));
+  webrtc::ByteBufferReader buf(fuzz_data.ReadRemaining());
   stun_msg->Read(&buf);
   stun_msg->ValidateMessageIntegrity("");
 }
diff --git a/test/fuzzers/stun_validator_fuzzer.cc b/test/fuzzers/stun_validator_fuzzer.cc
index 84c900a7e0..ec1892afde 100644
--- a/test/fuzzers/stun_validator_fuzzer.cc
+++ b/test/fuzzers/stun_validator_fuzzer.cc
@@ -11,13 +11,16 @@
 #include <stddef.h>
 #include <stdint.h>
 
+#include "absl/strings/string_view.h"
 #include "api/transport/stun.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  const char* message = reinterpret_cast<const char*>(data);
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
+  absl::string_view message = fuzz_data.ReadString();
 
-  webrtc::StunMessage::ValidateFingerprint(message, size);
-  webrtc::StunMessage::ValidateMessageIntegrityForTesting(message, size, "");
+  webrtc::StunMessage::ValidateFingerprint(message.data(), message.size());
+  webrtc::StunMessage::ValidateMessageIntegrityForTesting(message.data(),
+                                                          message.size(), "");
 }
 }  // namespace webrtc
diff --git a/test/fuzzers/turn_unwrap_fuzzer.cc b/test/fuzzers/turn_unwrap_fuzzer.cc
index 9da8e7addc..3f3670edc8 100644
--- a/test/fuzzers/turn_unwrap_fuzzer.cc
+++ b/test/fuzzers/turn_unwrap_fuzzer.cc
@@ -11,12 +11,16 @@
 #include <stddef.h>
 #include <stdint.h>
 
+#include "api/array_view.h"
 #include "media/base/turn_utils.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
-void FuzzOneInput(const uint8_t* data, size_t size) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
   size_t content_position;
   size_t content_size;
-  webrtc::UnwrapTurnPacket(data, size, &content_position, &content_size);
+  ArrayView<const uint8_t> raw = fuzz_data.ReadRemaining();
+  webrtc::UnwrapTurnPacket(raw.data(), raw.size(), &content_position,
+                           &content_size);
 }
 }  // namespace webrtc
diff --git a/test/fuzzers/ulpfec_generator_fuzzer.cc b/test/fuzzers/ulpfec_generator_fuzzer.cc
index e250c8ca0c..6ad54f5f17 100644
--- a/test/fuzzers/ulpfec_generator_fuzzer.cc
+++ b/test/fuzzers/ulpfec_generator_fuzzer.cc
@@ -24,6 +24,7 @@
 #include "rtc_base/checks.h"
 #include "rtc_base/copy_on_write_buffer.h"
 #include "system_wrappers/include/clock.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
 
@@ -32,36 +33,36 @@ constexpr uint8_t kFecPayloadType = 96;
 constexpr uint8_t kRedPayloadType = 97;
 }  // namespace
 
-void FuzzOneInput(const uint8_t* data, size_t size) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
   // Create Environment once because creating it for each input noticably
   // reduces the speed of the fuzzer.
   static const Environment* const env =
       new Environment(CreateEnvironment(std::make_unique<SimulatedClock>(1)));
 
   UlpfecGenerator generator(*env, kRedPayloadType, kFecPayloadType);
-  size_t i = 0;
-  if (size < 4)
+  if (fuzz_data.size() < 4)
     return;
   FecProtectionParams params = {
-      data[i++] % 128, static_cast<int>(data[i++] % 10), kFecMaskBursty};
+      .fec_rate = fuzz_data.Read<uint8_t>() % 128,
+      .max_fec_frames = static_cast<int>(fuzz_data.Read<uint8_t>() % 10),
+      .fec_mask_type = kFecMaskBursty};
   generator.SetProtectionParameters(params, params);
-  uint16_t seq_num = data[i++];
+  uint16_t seq_num = fuzz_data.Read<uint8_t>();
   uint16_t prev_seq_num = 0;
-  while (i + 3 < size) {
-    size_t rtp_header_length = data[i++] % 10 + 12;
-    size_t payload_size = data[i++] % 10;
-    if (i + payload_size + rtp_header_length + 2 > size)
+  while (fuzz_data.BytesLeft() > 3) {
+    size_t rtp_header_length = fuzz_data.Read<uint8_t>() % 10 + 12;
+    size_t payload_size = fuzz_data.Read<uint8_t>() % 10;
+    if (fuzz_data.BytesLeft() < payload_size + rtp_header_length + 2)
       break;
-    webrtc::CopyOnWriteBuffer packet(&data[i],
-                                     payload_size + rtp_header_length);
-    packet.EnsureCapacity(IP_PACKET_SIZE);
+    CopyOnWriteBuffer packet(
+        fuzz_data.ReadByteArray(payload_size + rtp_header_length),
+        IP_PACKET_SIZE);
     // Write a valid parsable header (version = 2, no padding, no extensions,
     // no CSRCs).
     ByteWriter<uint8_t>::WriteBigEndian(packet.MutableData(), 2 << 6);
     // Make sure sequence numbers are increasing.
     ByteWriter<uint16_t>::WriteBigEndian(packet.MutableData() + 2, seq_num++);
-    i += payload_size + rtp_header_length;
-    const bool protect = data[i++] % 2 == 1;
+    const bool protect = fuzz_data.Read<uint8_t>() % 2 == 1;
 
     // Check the sequence numbers are monotonic. In rare case the packets number
     // may loop around and in the same FEC-protected group the packet sequence
diff --git a/test/fuzzers/ulpfec_header_reader_fuzzer.cc b/test/fuzzers/ulpfec_header_reader_fuzzer.cc
index 16f123a850..aa7a0027a3 100644
--- a/test/fuzzers/ulpfec_header_reader_fuzzer.cc
+++ b/test/fuzzers/ulpfec_header_reader_fuzzer.cc
@@ -13,24 +13,26 @@
 #include <cstdint>
 #include <cstring>
 
+#include "api/array_view.h"
 #include "api/scoped_refptr.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/forward_error_correction.h"
 #include "modules/rtp_rtcp/source/ulpfec_header_reader_writer.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
 
 using Packet = ForwardErrorCorrection::Packet;
 using ReceivedFecPacket = ForwardErrorCorrection::ReceivedFecPacket;
 
-void FuzzOneInput(const uint8_t* data, size_t size) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
   ReceivedFecPacket packet;
   packet.pkt = webrtc::scoped_refptr<Packet>(new Packet());
   const size_t packet_size =
-      std::min(size, static_cast<size_t>(IP_PACKET_SIZE));
-  packet.pkt->data.SetSize(packet_size);
+      std::min(fuzz_data.size(), static_cast<size_t>(IP_PACKET_SIZE));
+  ArrayView<const uint8_t> raw = fuzz_data.ReadByteArray(packet_size);
   packet.pkt->data.EnsureCapacity(IP_PACKET_SIZE);
-  memcpy(packet.pkt->data.MutableData(), data, packet_size);
+  packet.pkt->data.SetData(raw.data(), raw.size());
 
   UlpfecHeaderReader ulpfec_reader;
   ulpfec_reader.ReadFecHeader(&packet);
diff --git a/test/fuzzers/ulpfec_receiver_fuzzer.cc b/test/fuzzers/ulpfec_receiver_fuzzer.cc
index b73651a69c..6e70da957b 100644
--- a/test/fuzzers/ulpfec_receiver_fuzzer.cc
+++ b/test/fuzzers/ulpfec_receiver_fuzzer.cc
@@ -14,7 +14,6 @@
 #include "api/array_view.h"
 #include "modules/rtp_rtcp/include/recovered_packet_receiver.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
-#include "modules/rtp_rtcp/source/byte_io.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "modules/rtp_rtcp/source/ulpfec_receiver.h"
 #include "system_wrappers/include/clock.h"
@@ -28,21 +27,20 @@ class DummyCallback : public RecoveredPacketReceiver {
 };
 }  // namespace
 
-void FuzzOneInput(const uint8_t* data, size_t size) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
   constexpr size_t kMinDataNeeded = 12;
-  if (size < kMinDataNeeded || size > 2000) {
+  if (fuzz_data.size() < kMinDataNeeded || fuzz_data.size() > 2'000) {
     return;
   }
 
-  uint32_t ulpfec_ssrc = ByteReader<uint32_t>::ReadLittleEndian(data + 0);
-  uint16_t ulpfec_seq_num = ByteReader<uint16_t>::ReadLittleEndian(data + 4);
-  uint32_t media_ssrc = ByteReader<uint32_t>::ReadLittleEndian(data + 6);
-  uint16_t media_seq_num = ByteReader<uint16_t>::ReadLittleEndian(data + 10);
+  uint32_t ulpfec_ssrc = fuzz_data.Read<uint32_t>();
+  uint16_t ulpfec_seq_num = fuzz_data.Read<uint16_t>();
+  uint32_t media_ssrc = fuzz_data.Read<uint32_t>();
+  uint16_t media_seq_num = fuzz_data.Read<uint16_t>();
 
   DummyCallback callback;
   UlpfecReceiver receiver(ulpfec_ssrc, 0, &callback, Clock::GetRealTimeClock());
 
-  test::FuzzDataHelper fuzz_data(webrtc::MakeArrayView(data, size));
   while (fuzz_data.CanReadBytes(kMinDataNeeded)) {
     size_t packet_length = kRtpHeaderSize + fuzz_data.Read<uint8_t>();
     auto raw_packet = fuzz_data.ReadByteArray(packet_length);
diff --git a/test/fuzzers/utils/BUILD.gn b/test/fuzzers/utils/BUILD.gn
index 6d1c867f92..dabdab56d1 100644
--- a/test/fuzzers/utils/BUILD.gn
+++ b/test/fuzzers/utils/BUILD.gn
@@ -62,5 +62,6 @@ rtc_library("validate_rtp_packetizer") {
     "..:fuzz_data_helper",
     "../../../modules/rtp_rtcp",
     "../../../modules/rtp_rtcp:rtp_rtcp_format",
+    "../../../rtc_base:checks",
   ]
 }
diff --git a/test/fuzzers/utils/validate_rtp_packetizer.cc b/test/fuzzers/utils/validate_rtp_packetizer.cc
index fbc6781bd8..1687479c4d 100644
--- a/test/fuzzers/utils/validate_rtp_packetizer.cc
+++ b/test/fuzzers/utils/validate_rtp_packetizer.cc
@@ -11,14 +11,17 @@
 #include "test/fuzzers/utils/validate_rtp_packetizer.h"
 
 #include <cstddef>
+#include <cstdint>
 
 #include "modules/rtp_rtcp/source/rtp_format.h"
 #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "rtc_base/checks.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
 
 RtpPacketizer::PayloadSizeLimits ReadPayloadSizeLimits(
-    test::FuzzDataHelper& fuzz_input) {
+    FuzzDataHelper& fuzz_input) {
   RtpPacketizer::PayloadSizeLimits limits;
   limits.max_payload_len = 1200;
 
diff --git a/test/fuzzers/utils/validate_rtp_packetizer.h b/test/fuzzers/utils/validate_rtp_packetizer.h
index 9c4000e5c0..59db5f14bd 100644
--- a/test/fuzzers/utils/validate_rtp_packetizer.h
+++ b/test/fuzzers/utils/validate_rtp_packetizer.h
@@ -18,7 +18,7 @@ namespace webrtc {
 
 // Generates valid `RtpPacketizer::PayloadSizeLimits` from the `fuzz_input`.
 RtpPacketizer::PayloadSizeLimits ReadPayloadSizeLimits(
-    test::FuzzDataHelper& fuzz_input);
+    FuzzDataHelper& fuzz_input);
 
 // RTC_CHECKs if `rtp_packetizer` created packets with respect to the `limits`.
 void ValidateRtpPacketizer(const RtpPacketizer::PayloadSizeLimits& limits,
diff --git a/test/fuzzers/vp8_depacketizer_fuzzer.cc b/test/fuzzers/vp8_depacketizer_fuzzer.cc
index 345fe5cefe..ede53a7b5e 100644
--- a/test/fuzzers/vp8_depacketizer_fuzzer.cc
+++ b/test/fuzzers/vp8_depacketizer_fuzzer.cc
@@ -8,16 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #include <cstddef>
-#include <cstdint>
 
-#include "api/array_view.h"
 #include "modules/rtp_rtcp/source/rtp_video_header.h"
 #include "modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
-void FuzzOneInput(const uint8_t* data, size_t size) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
   RTPVideoHeader video_header;
-  VideoRtpDepacketizerVp8::ParseRtpPayload(webrtc::MakeArrayView(data, size),
+  VideoRtpDepacketizerVp8::ParseRtpPayload(fuzz_data.ReadRemaining(),
                                            &video_header);
 }
 }  // namespace webrtc
diff --git a/test/fuzzers/vp8_qp_parser_fuzzer.cc b/test/fuzzers/vp8_qp_parser_fuzzer.cc
index 597e1f492f..7c771e905f 100644
--- a/test/fuzzers/vp8_qp_parser_fuzzer.cc
+++ b/test/fuzzers/vp8_qp_parser_fuzzer.cc
@@ -10,11 +10,14 @@
 #include <cstddef>
 #include <cstdint>
 
+#include "api/array_view.h"
 #include "modules/video_coding/utility/vp8_header_parser.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
-void FuzzOneInput(const uint8_t* data, size_t size) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
   int qp;
-  vp8::GetQp(data, size, &qp);
+  ArrayView<const uint8_t> raw = fuzz_data.ReadRemaining();
+  vp8::GetQp(raw.data(), raw.size(), &qp);
 }
 }  // namespace webrtc
diff --git a/test/fuzzers/vp8_replay_fuzzer.cc b/test/fuzzers/vp8_replay_fuzzer.cc
index 3d872962dc..8f95050b53 100644
--- a/test/fuzzers/vp8_replay_fuzzer.cc
+++ b/test/fuzzers/vp8_replay_fuzzer.cc
@@ -15,13 +15,15 @@
 #include <utility>
 #include <vector>
 
+#include "api/array_view.h"
 #include "api/video_codecs/sdp_video_format.h"
 #include "call/video_receive_stream.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 #include "test/fuzzers/utils/rtp_replayer.h"
 
 namespace webrtc {
 
-void FuzzOneInput(const uint8_t* data, size_t size) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
   auto stream_state = std::make_unique<test::RtpReplayer::StreamState>();
   VideoReceiveStreamInterface::Config vp8_config(&(stream_state->transport));
 
@@ -38,8 +40,9 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
   std::vector<VideoReceiveStreamInterface::Config> replay_configs;
   replay_configs.push_back(std::move(vp8_config));
 
+  ArrayView<const uint8_t> raw = fuzz_data.ReadRemaining();
   test::RtpReplayer::Replay(std::move(stream_state), std::move(replay_configs),
-                            data, size);
+                            raw.data(), raw.size());
 }
 
 }  // namespace webrtc
diff --git a/test/fuzzers/vp9_depacketizer_fuzzer.cc b/test/fuzzers/vp9_depacketizer_fuzzer.cc
index 68758e8b7e..3ea2a9f94f 100644
--- a/test/fuzzers/vp9_depacketizer_fuzzer.cc
+++ b/test/fuzzers/vp9_depacketizer_fuzzer.cc
@@ -8,16 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #include <cstddef>
-#include <cstdint>
 
-#include "api/array_view.h"
 #include "modules/rtp_rtcp/source/rtp_video_header.h"
 #include "modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
-void FuzzOneInput(const uint8_t* data, size_t size) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
   RTPVideoHeader video_header;
-  VideoRtpDepacketizerVp9::ParseRtpPayload(webrtc::MakeArrayView(data, size),
+  VideoRtpDepacketizerVp9::ParseRtpPayload(fuzz_data.ReadRemaining(),
                                            &video_header);
 }
 }  // namespace webrtc
diff --git a/test/fuzzers/vp9_encoder_references_fuzzer.cc b/test/fuzzers/vp9_encoder_references_fuzzer.cc
index 4f41cafb21..35b58628a4 100644
--- a/test/fuzzers/vp9_encoder_references_fuzzer.cc
+++ b/test/fuzzers/vp9_encoder_references_fuzzer.cc
@@ -47,8 +47,6 @@
 namespace webrtc {
 namespace {
 
-using test::FuzzDataHelper;
-
 constexpr int kBitrateEnabledBps = 100'000;
 
 class FrameValidator : public EncodedImageCallback {
@@ -542,18 +540,16 @@ static_assert(DropBelow(0b1101, /*sid=*/3, 4) == false, "");
 
 }  // namespace
 
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  FuzzDataHelper helper(webrtc::MakeArrayView(data, size));
-
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
   FrameValidator validator;
-  FieldTrials field_trials(helper);
+  FieldTrials field_trials(fuzz_data);
   // Setup call callbacks for the fake
   LibvpxState state;
 
   // Initialize encoder
   LibvpxVp9Encoder encoder(CreateEnvironment(&field_trials), {},
                            std::make_unique<StubLibvpx>(&state));
-  VideoCodec codec = CodecSettings(helper);
+  VideoCodec codec = CodecSettings(fuzz_data);
   if (encoder.InitEncode(&codec, EncoderSettings()) != WEBRTC_VIDEO_CODEC_OK) {
     return;
   }
@@ -579,8 +575,8 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
                               .build();
 
   // Start producing frames at random.
-  while (helper.CanReadBytes(1)) {
-    uint8_t action = helper.Read<uint8_t>();
+  while (fuzz_data.CanReadBytes(1)) {
+    uint8_t action = fuzz_data.Read<uint8_t>();
     switch (action & 0b11) {
       case kEncode: {
         // bitmask of the action: SSSS-K00, where
diff --git a/test/fuzzers/vp9_qp_parser_fuzzer.cc b/test/fuzzers/vp9_qp_parser_fuzzer.cc
index a65cee54cf..350fdea81c 100644
--- a/test/fuzzers/vp9_qp_parser_fuzzer.cc
+++ b/test/fuzzers/vp9_qp_parser_fuzzer.cc
@@ -9,13 +9,12 @@
  */
 
 #include <cstddef>
-#include <cstdint>
 
-#include "api/array_view.h"
 #include "modules/video_coding/utility/vp9_uncompressed_header_parser.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  ParseUncompressedVp9Header(webrtc::MakeArrayView(data, size));
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
+  ParseUncompressedVp9Header(fuzz_data.ReadRemaining());
 }
 }  // namespace webrtc
diff --git a/test/fuzzers/vp9_replay_fuzzer.cc b/test/fuzzers/vp9_replay_fuzzer.cc
index 1bde2b5148..b996326da7 100644
--- a/test/fuzzers/vp9_replay_fuzzer.cc
+++ b/test/fuzzers/vp9_replay_fuzzer.cc
@@ -15,13 +15,15 @@
 #include <utility>
 #include <vector>
 
+#include "api/array_view.h"
 #include "api/video_codecs/sdp_video_format.h"
 #include "call/video_receive_stream.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 #include "test/fuzzers/utils/rtp_replayer.h"
 
 namespace webrtc {
 
-void FuzzOneInput(const uint8_t* data, size_t size) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
   auto stream_state = std::make_unique<test::RtpReplayer::StreamState>();
   VideoReceiveStreamInterface::Config vp9_config(&(stream_state->transport));
 
@@ -37,8 +39,9 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
   std::vector<VideoReceiveStreamInterface::Config> replay_configs;
   replay_configs.push_back(std::move(vp9_config));
 
+  ArrayView<const uint8_t> raw = fuzz_data.ReadRemaining();
   test::RtpReplayer::Replay(std::move(stream_state), std::move(replay_configs),
-                            data, size);
+                            raw.data(), raw.size());
 }
 
 }  // namespace webrtc
diff --git a/test/fuzzers/webrtc_fuzzer_main.cc b/test/fuzzers/webrtc_fuzzer_main.cc
index 830a9bead2..bd277c748a 100644
--- a/test/fuzzers/webrtc_fuzzer_main.cc
+++ b/test/fuzzers/webrtc_fuzzer_main.cc
@@ -13,9 +13,10 @@
 // fuzzing efficiency.
 
 #include <cstddef>
-#include <cstdint>
 
+#include "api/array_view.h"
 #include "rtc_base/logging.h"
+#include "test/fuzzers/fuzz_data_helper.h"
 
 namespace {
 bool g_initialized = false;
@@ -34,11 +35,12 @@ void InitializeWebRtcFuzzDefaults() {
 }  // namespace
 
 namespace webrtc {
-extern void FuzzOneInput(const uint8_t* data, size_t size);
+extern void FuzzOneInput(FuzzDataHelper fuzz_data);
 }  // namespace webrtc
 
 extern "C" int LLVMFuzzerTestOneInput(const unsigned char* data, size_t size) {
   InitializeWebRtcFuzzDefaults();
-  webrtc::FuzzOneInput(data, size);
+  webrtc::FuzzDataHelper fuzz_data(webrtc::MakeArrayView(data, size));
+  webrtc::FuzzOneInput(fuzz_data);
   return 0;
 }
diff --git a/test/fuzzers/windowed_min_filter_fuzzer.cc b/test/fuzzers/windowed_min_filter_fuzzer.cc
index a79f6bc1c6..d381db2c40 100644
--- a/test/fuzzers/windowed_min_filter_fuzzer.cc
+++ b/test/fuzzers/windowed_min_filter_fuzzer.cc
@@ -8,15 +8,17 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <cstddef>
 #include <deque>
+
 #include "absl/algorithm/container.h"
-#include "rtc_base/numerics/windowed_min_filter.h"
 #include "rtc_base/checks.h"
+#include "rtc_base/numerics/windowed_min_filter.h"
 #include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
 
-void FuzzOneInput(const uint8_t* data, size_t size) {
+void FuzzOneInput(FuzzDataHelper fuzz_data) {
   class ReferenceFilter {
    public:
     explicit ReferenceFilter(int window_length) : max_size_(window_length) {}
@@ -35,7 +37,6 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
 
   ReferenceFilter reference_filter(/*window_length=*/10);
   WindowedMinFilter<int> filter(/*window_length=*/10);
-  test::FuzzDataHelper fuzz_data(MakeArrayView(data, size));
 
   while (fuzz_data.CanReadBytes(sizeof(int))) {
     int value = fuzz_data.Read<int>();

From 1d1753600d60dc85035355353bb9ae23de73e73e Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Tue, 3 Mar 2026 10:27:19 +0100
Subject: [PATCH 179/472] Alias ArrayView to be std::span

Bug: webrtc:439801349
Change-Id: I63dc91abfc8c3aee0eb95f31934246def2c16d87
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/451941
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47032}
---
 api/BUILD.gn     |   5 -
 api/array_view.h | 357 +----------------------------------------------
 2 files changed, 4 insertions(+), 358 deletions(-)

diff --git a/api/BUILD.gn b/api/BUILD.gn
index 537315fa5e..4759de57c8 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -1250,11 +1250,6 @@ rtc_source_set("network_state_predictor_api") {
 rtc_source_set("array_view") {
   visibility = [ "*" ]
   sources = [ "array_view.h" ]
-  deps = [
-    "../rtc_base:checks",
-    "../rtc_base:type_traits",
-    "//third_party/abseil-cpp/absl/base:core_headers",
-  ]
 }
 
 rtc_source_set("refcountedbase") {
diff --git a/api/array_view.h b/api/array_view.h
index 01593e0d02..c670314852 100644
--- a/api/array_view.h
+++ b/api/array_view.h
@@ -11,365 +11,16 @@
 #ifndef API_ARRAY_VIEW_H_
 #define API_ARRAY_VIEW_H_
 
-#include <algorithm>
-#include <array>
 #include <cstddef>
-#include <iterator>
-#include <ranges>
 #include <span>
 #include <type_traits>
 
-#include "absl/base/macros.h"
-#include "rtc_base/checks.h"
-#include "rtc_base/type_traits.h"
-
 namespace webrtc {
 
-// tl;dr: ArrayView is the same thing as gsl::span from the Guideline
-//        Support Library.
-//
-// Many functions read from or write to arrays. The obvious way to do this is
-// to use two arguments, a pointer to the first element and an element count:
-//
-//   bool Contains17(const int* arr, size_t size) {
-//     for (size_t i = 0; i < size; ++i) {
-//       if (arr[i] == 17)
-//         return true;
-//     }
-//     return false;
-//   }
-//
-// This is flexible, since it doesn't matter how the array is stored (C array,
-// std::vector, Buffer, ...), but it's error-prone because the caller
-// has to correctly specify the array length:
-//
-//   Contains17(arr, std::size(arr));     // C array
-//   Contains17(arr.data(), arr.size());  // std::vector
-//   Contains17(arr, size);               // pointer + size
-//   ...
-//
-// It's also kind of messy to have two separate arguments for what is
-// conceptually a single thing.
-//
-// Enter ArrayView<T>. It contains a T pointer (to an array it doesn't
-// own) and a count, and supports the basic things you'd expect, such as
-// indexing and iteration. It allows us to write our function like this:
-//
-//   bool Contains17(ArrayView<const int> arr) {
-//     for (auto e : arr) {
-//       if (e == 17)
-//         return true;
-//     }
-//     return false;
-//   }
-//
-// And even better, because a bunch of things will implicitly convert to
-// ArrayView, we can call it like this:
-//
-//   Contains17(arr);                             // C array
-//   Contains17(arr);                             // std::vector
-//   Contains17(ArrayView<int>(arr, size));  // pointer + size
-//   Contains17(nullptr);                         // nullptr -> empty ArrayView
-//   ...
-//
-// ArrayView<T> stores both a pointer and a size, but you may also use
-// ArrayView<T, N>, which has a size that's fixed at compile time (which means
-// it only has to store the pointer).
-//
-// One important point is that ArrayView<T> and ArrayView<const T> are
-// different types, which allow and don't allow mutation of the array elements,
-// respectively. The implicit conversions work just like you'd hope, so that
-// e.g. vector<int> will convert to either ArrayView<int> or ArrayView<const
-// int>, but const vector<int> will convert only to ArrayView<const int>.
-// (ArrayView itself can be the source type in such conversions, so
-// ArrayView<int> will convert to ArrayView<const int>.)
-//
-// Note: ArrayView is tiny (just a pointer and a count if variable-sized, just
-// a pointer if fix-sized) and trivially copyable, so it's probably cheaper to
-// pass it by value than by const reference.
-
-namespace array_view_internal {
-
-// Magic constant for indicating that the size of an ArrayView is variable
-// instead of fixed.
-enum : std::ptrdiff_t { kArrayViewVarSize = -4711 };
-
-// Base class for ArrayViews of fixed nonzero size.
-template <typename T, std::ptrdiff_t Size>
-class ArrayViewBase {
-  static_assert(Size > 0, "ArrayView size must be variable or non-negative");
-
- public:
-  constexpr ArrayViewBase(T* data, size_t /* size */) : data_(data) {}
-
-  static constexpr size_t size() { return Size; }
-  static constexpr bool empty() { return false; }
-  constexpr T* data() const { return data_; }
-
- protected:
-  static constexpr bool fixed_size() { return true; }
-
- private:
-  T* data_;
-};
-
-// Specialized base class for ArrayViews of fixed zero size.
-template <typename T>
-class ArrayViewBase<T, 0> {
- public:
-  explicit constexpr ArrayViewBase(T* /* data */, size_t /* size */) {}
-
-  static constexpr size_t size() { return 0; }
-  static constexpr bool empty() { return true; }
-  constexpr T* data() const { return nullptr; }
-
- protected:
-  static constexpr bool fixed_size() { return true; }
-};
-
-// Specialized base class for ArrayViews of variable size.
-template <typename T>
-class ArrayViewBase<T, array_view_internal::kArrayViewVarSize> {
- public:
-  constexpr ArrayViewBase(T* data, size_t size)
-      : data_(size == 0 ? nullptr : data), size_(size) {}
-
-  constexpr size_t size() const { return size_; }
-  constexpr bool empty() const { return size_ == 0; }
-  constexpr T* data() const { return data_; }
-
- protected:
-  static constexpr bool fixed_size() { return false; }
-
- private:
-  T* data_;
-  size_t size_;
-};
-
-}  // namespace array_view_internal
-
-template <typename T,
-          std::ptrdiff_t Size = array_view_internal::kArrayViewVarSize>
-class ArrayView final : public array_view_internal::ArrayViewBase<T, Size> {
- public:
-  using value_type = T;
-  using reference = value_type&;
-  using const_reference = const value_type&;
-  using pointer = value_type*;
-  using const_pointer = const value_type*;
-  using iterator = typename std::span<T>::iterator;
-  using const_iterator = typename std::span<const T>::iterator;
-
-  static constexpr size_t extent =
-      Size == array_view_internal::kArrayViewVarSize
-          ? std::dynamic_extent
-          : static_cast<size_t>(Size);
-
-  // Construct an ArrayView from a pointer and a length.
-  template <typename U>
-  explicit(Size != array_view_internal::kArrayViewVarSize)  //
-      constexpr ArrayView(U* data, size_t size)
-      : array_view_internal::ArrayViewBase<T, Size>::ArrayViewBase(data, size) {
-    RTC_DCHECK_EQ(size == 0 ? nullptr : data, this->data());
-    RTC_DCHECK_EQ(size, this->size());
-    RTC_DCHECK_EQ(!this->data(),
-                  this->size() == 0);  // data is null iff size == 0.
-  }
-
-  // Construct an empty ArrayView. Note that fixed-size ArrayViews of size > 0
-  // cannot be empty.
-  constexpr ArrayView() : ArrayView(static_cast<T*>(nullptr), 0) {}
-  ABSL_DEPRECATE_AND_INLINE()
-  ArrayView(std::nullptr_t)  // NOLINT
-      : ArrayView() {}
-  ABSL_DEPRECATE_AND_INLINE()
-  ArrayView(std::nullptr_t, size_t size) : ArrayView() {
-    static_assert(Size == 0 || Size == array_view_internal::kArrayViewVarSize,
-                  "");
-    RTC_DCHECK_EQ(0, size);
-  }
-
-  // Construct an ArrayView from a C-style array.
-  template <typename U, size_t N>
-  ArrayView(U (&array)[N])  // NOLINT
-      : ArrayView(array, N) {
-    static_assert(Size == N || Size == array_view_internal::kArrayViewVarSize,
-                  "Array size must match ArrayView size");
-  }
-
-  // (Only if size is fixed.) Construct a fixed size ArrayView<T, N> from a
-  // non-const std::array instance. For an ArrayView with variable size, the
-  // used ctor is ArrayView(U& u) instead.
-  template <typename U,
-            size_t N,
-            typename std::enable_if<
-                Size == static_cast<std::ptrdiff_t>(N)>::type* = nullptr>
-  ArrayView(std::array<U, N>& u)  // NOLINT
-      : ArrayView(u.data(), u.size()) {}
-
-  // (Only if size is fixed.) Construct a fixed size ArrayView<T, N> where T is
-  // const from a const(expr) std::array instance. For an ArrayView with
-  // variable size, the used ctor is ArrayView(U& u) instead.
-  template <typename U,
-            size_t N,
-            typename std::enable_if<
-                Size == static_cast<std::ptrdiff_t>(N)>::type* = nullptr>
-  ArrayView(const std::array<U, N>& u)  // NOLINT
-      : ArrayView(u.data(), u.size()) {}
-
-  // (Only if size is fixed.) Construct an ArrayView from any type U that has a
-  // static constexpr size() method whose return value is equal to Size, and a
-  // data() method whose return value converts implicitly to T*. In particular,
-  // this means we allow conversion from ArrayView<T, N> to ArrayView<const T,
-  // N>, but not the other way around. We also don't allow conversion from
-  // ArrayView<T> to ArrayView<T, N>, or from ArrayView<T, M> to ArrayView<T,
-  // N> when M != N.
-  template <typename U,
-            typename std::enable_if_t<
-                !std::is_same_v<ArrayView, std::remove_reference_t<U>> &&
-                Size != array_view_internal::kArrayViewVarSize &&
-                HasDataAndSize<U, T>::value>* = nullptr>
-  ArrayView(U& u)  // NOLINT
-      : ArrayView(u.data(), u.size()) {
-    static_assert(U::size() == Size, "Sizes must match exactly");
-  }
-
-  template <typename U,
-            typename std::enable_if_t<
-                !std::is_same_v<ArrayView, std::remove_reference_t<U>> &&
-                Size != array_view_internal::kArrayViewVarSize &&
-                HasDataAndSize<U, T>::value>* = nullptr>
-  ArrayView(const U& u)  // NOLINT(runtime/explicit)
-      : ArrayView(u.data(), u.size()) {
-    static_assert(U::size() == Size, "Sizes must match exactly");
-  }
-
-  // (Only if size is variable.) Construct an ArrayView from any type U that
-  // has a size() method whose return value converts implicitly to size_t, and
-  // a data() method whose return value converts implicitly to T*. In
-  // particular, this means we allow conversion from ArrayView<T> to
-  // ArrayView<const T>, but not the other way around. Other allowed
-  // conversions include
-  // ArrayView<T, N> to ArrayView<T> or ArrayView<const T>,
-  // std::vector<T> to ArrayView<T> or ArrayView<const T>,
-  // const std::vector<T> to ArrayView<const T>,
-  // Buffer to ArrayView<uint8_t> or ArrayView<const uint8_t>, and
-  // const Buffer to ArrayView<const uint8_t>.
-  template <typename U,
-            typename std::enable_if_t<
-                !std::is_same_v<ArrayView, std::remove_reference_t<U>> &&
-                Size == array_view_internal::kArrayViewVarSize &&
-                HasDataAndSize<U, T>::value>* = nullptr>
-    requires(std::ranges::contiguous_range<U> && std::ranges::sized_range<U>)
-  ArrayView(U& u)  // NOLINT
-      : ArrayView(u.data(), u.size()) {}
-
-  template <typename U,
-            typename std::enable_if_t<
-                !std::is_same_v<ArrayView, std::remove_reference_t<U>> &&
-                Size == array_view_internal::kArrayViewVarSize &&
-                HasDataAndSize<U, T>::value>* = nullptr>
-    requires(std::ranges::contiguous_range<U> && std::ranges::sized_range<U>)
-  ArrayView(const U& u)  // NOLINT(runtime/explicit)
-      : ArrayView(u.data(), u.size()) {}
-
-  // Indexing and iteration. These allow mutation even if the ArrayView is
-  // const, because the ArrayView doesn't own the array. (To prevent mutation,
-  // use a const element type.)
-  [[nodiscard]] T& operator[](size_t idx) const {
-    // RTC_HARDENING_ASSERT is used because this is tunable at compile time
-    // to be on or off in production, depending on the choices of the
-    // embedder.
-    RTC_HARDENING_ASSERT(idx < this->size());
-    RTC_HARDENING_ASSERT(this->data());
-    return this->data()[idx];
-  }
-  iterator begin() const {
-    return std::span<T>(this->data(), this->size()).begin();
-  }
-  iterator end() const {
-    return std::span<T>(this->data(), this->size()).end();
-  }
-  ABSL_DEPRECATE_AND_INLINE() const_iterator cbegin() const { return begin(); }
-  ABSL_DEPRECATE_AND_INLINE() const_iterator cend() const { return end(); }
-  std::reverse_iterator<iterator> rbegin() const {
-    return std::make_reverse_iterator(end());
-  }
-  std::reverse_iterator<iterator> rend() const {
-    return std::make_reverse_iterator(begin());
-  }
-  ABSL_DEPRECATE_AND_INLINE()
-  std::reverse_iterator<const_iterator> crbegin() const { return rbegin(); }
-  ABSL_DEPRECATE_AND_INLINE()
-  std::reverse_iterator<const_iterator> crend() const { return rend(); }
-
-  template <size_t count>
-  constexpr ArrayView<T, count> first() const {
-    RTC_HARDENING_ASSERT(count <= this->size());
-    return ArrayView<T, count>(this->data(), count);
-  }
-
-  template <size_t count>
-  constexpr ArrayView<T, count> last() const {
-    RTC_HARDENING_ASSERT(count <= this->size());
-    return ArrayView<T, count>(this->data() + this->size() - count, count);
-  }
-
-  template <size_t offset, size_t count = std::dynamic_extent>
-  constexpr auto subspan() const {
-    RTC_HARDENING_ASSERT(offset <= this->size());
-    if constexpr (count == std::dynamic_extent) {
-      constexpr ptrdiff_t final_extent =
-          Size != array_view_internal::kArrayViewVarSize
-              ? Size - offset
-              : array_view_internal::kArrayViewVarSize;
-      return ArrayView<T, final_extent>(this->data() + offset,
-                                        this->size() - offset);
-    } else {
-      RTC_HARDENING_ASSERT(count <= this->size() - offset);
-      return ArrayView<T, count>(this->data() + offset, count);
-    }
-  }
-
-  [[nodiscard]] constexpr ArrayView<T> subspan(
-      size_t offset,
-      size_t count = std::dynamic_extent) const {
-    ABSL_HARDENING_ASSERT(offset <= this->size());
-    if (count == std::dynamic_extent) {
-      count = this->size() - offset;
-    } else {
-      ABSL_HARDENING_ASSERT(count <= this->size() - offset);
-    }
-    return ArrayView<T>(this->data() + offset, count);
-  }
-
-  [[deprecated("Use subspan")]]
-  ArrayView<T> subview(size_t offset, size_t size) const {
-    return offset <= this->size()
-               ? subspan(offset, std::min(size, this->size() - offset))
-               : ArrayView<T>();
-  }
-  [[deprecated("Use subspan")]]
-  ArrayView<T> subview(size_t offset) const {
-    return offset <= this->size() ? subspan(offset) : ArrayView<T>();
-  }
-};
-
-// Comparing two ArrayViews compares their (pointer,size) pairs; it does *not*
-// dereference the pointers.
-template <typename T, std::ptrdiff_t Size1, std::ptrdiff_t Size2>
-[[deprecated("issues.webrtc.org/439801349")]]
-bool operator==(const ArrayView<T, Size1>& a, const ArrayView<T, Size2>& b) {
-  return a.data() == b.data() && a.size() == b.size();
-}
-
-// Variable-size ArrayViews are the size of two pointers; fixed-size ArrayViews
-// are the size of one pointer. (And as a special case, fixed-size ArrayViews
-// of size 0 require no storage.)
-static_assert(sizeof(ArrayView<int>) == 2 * sizeof(int*), "");
-static_assert(sizeof(ArrayView<int, 17>) == sizeof(int*), "");
-static_assert(std::is_empty<ArrayView<int, 0>>::value, "");
+template <typename T, size_t extent = std::dynamic_extent>
+using ArrayView = std::span<T, extent>;
 
+// TODO: bugs.webrtc.org/439801349 - deprecate when unused by WebRTC
 template <typename T>
 inline ArrayView<T> MakeArrayView(T* data, size_t size) {
   return ArrayView<T>(data, size);
@@ -380,7 +31,7 @@ inline ArrayView<T> MakeArrayView(T* data, size_t size) {
 // same size.
 // Template arguments order is (U, T, Size) to allow deduction of the template
 // arguments in client calls: reinterpret_array_view<target_type>(array_view).
-template <typename U, typename T, std::ptrdiff_t Size>
+template <typename U, typename T, size_t Size>
 [[deprecated]] ArrayView<U, Size> reinterpret_array_view(
     ArrayView<T, Size> view) {
   static_assert(sizeof(U) == sizeof(T) && alignof(U) == alignof(T),

From fcc3382dba18ae8591c452c63b7b1759653dcecf Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Tue, 3 Mar 2026 08:03:52 -0800
Subject: [PATCH 180/472] Roll chromium_revision 460b91592c..890f13186f
 (1592626:1593243)

Change log: https://chromium.googlesource.com/chromium/src/+log/460b91592c..890f13186f
Full diff: https://chromium.googlesource.com/chromium/src/+/460b91592c..890f13186f

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/b62d394bff..bc0d205afa
* src/buildtools: https://chromium.googlesource.com/chromium/src/buildtools/+log/136da69a12..fef8d6947f
* src/buildtools/linux64: git_revision:1a310e88443018837759c952b113846b0096f65b..git_revision:a155c8c78af4030169a8da4886dc0c6b170535de
* src/buildtools/mac: git_revision:1a310e88443018837759c952b113846b0096f65b..git_revision:a155c8c78af4030169a8da4886dc0c6b170535de
* src/buildtools/win: git_revision:1a310e88443018837759c952b113846b0096f65b..git_revision:a155c8c78af4030169a8da4886dc0c6b170535de
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/79bcc4f8dd..b40bc07ded
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/0371d548fc..edc9633604
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/d25c43455e..82dd665e17
* src/third_party/android_deps/autorolled/cipd: 9DXI8FnmXkL2tti6GWrRX5mKZzOJzOVKFbiQLuE4ThYC..gOFPZuLmUt7S4rnhusfD5p-Fhy3qllnFyNIwDbThlusC
* src/third_party/androidx/cipd: F33f66w1lXogUL7EN68PuqtiS1I2B8ptEm0Zq-MhYnIC..CW8b7YlBGZSbouGqDIHzMn5HRGQAqAM8d0kEcogTqJMC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/80053a9bb7..d0fcf49574
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/27800c3890..109c6d7bab
* src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/64c252741e..0cd354e178
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/55659c2bf4..d6aae1009f
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/57b1de2330..661cdb227f
* src/third_party/fuzztest/src: https://chromium.googlesource.com/external/github.com/google/fuzztest.git/+log/6a217df9ed..2774d423cc
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/73b070473d..060a217622
* src/third_party/icu: https://chromium.googlesource.com/chromium/deps/icu.git/+log/7971660ba6..ee5f27adc2
* src/third_party/kotlin_stdlib/cipd: r1jin7oxmCEIuca5n9GqpxpPk_aV2Dr0CLAmWSV0tEAC..9ldUaVxyYClDj9QDPGLyxhrrLE40VxHjm65wiU_4xaoC
* src/third_party/libvpx/source/libvpx: https://chromium.googlesource.com/webm/libvpx.git/+log/db312f06fb..4fcebeabe5
* src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/7434d317bd..dacaf47cea
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/6391c13617..f884650a15
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/88a9f66c07..a3ead8a269
* src/tools/luci-go: git_revision:b9d7d1a7774833e136aaa5084a0d150cd5ecb641..git_revision:a61d4d67348b452a5e7bf09541f808e8c1101e76
* src/tools/luci-go: git_revision:b9d7d1a7774833e136aaa5084a0d150cd5ecb641..git_revision:a61d4d67348b452a5e7bf09541f808e8c1101e76
* src/tools/luci-go: git_revision:b9d7d1a7774833e136aaa5084a0d150cd5ecb641..git_revision:a61d4d67348b452a5e7bf09541f808e8c1101e76
DEPS diff: https://chromium.googlesource.com/chromium/src/+/460b91592c..890f13186f/DEPS

BUG=None

Change-Id: Id89fa11e2fe87a6bfce61683320b28bdd6b525f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453281
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47033}
---
 DEPS | 54 +++++++++++++++++++++++++++---------------------------
 1 file changed, 27 insertions(+), 27 deletions(-)

diff --git a/DEPS b/DEPS
index 69a631ceba..98f788ed71 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '460b91592c227585c769c7154f0cb643605d44c2',
+  'chromium_revision': '890f13186fc2d261eeb6cc3d78448dfc58cf03da',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,28 +69,28 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@b62d394bff1cca126e4e8b7a711da7829003d003',
+    'https://chromium.googlesource.com/chromium/src/build@bc0d205afa3133b2ee54cef32d6d2b5693b5eca8',
   'src/buildtools':
-    'https://chromium.googlesource.com/chromium/src/buildtools@136da69a1267b8db487354b96d44d0cc8add5aeb',
+    'https://chromium.googlesource.com/chromium/src/buildtools@fef8d6947fa54202e8a00c58d5ac8fd15780bc33',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
   'src/examples/androidtests/third_party/gradle': {
     'url': 'https://chromium.googlesource.com/external/github.com/gradle/gradle.git@f2d1fb54a951d8b11d25748e4711bec8d128d7e3',
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@79bcc4f8dd01e209ee6b9a096275c900b3e8c0be',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@b40bc07ded3eb0479ce5e9c1e470f3c9a921ae54',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@0371d548fc3ec99bf6c6ff889c9afe488c3cde1b',
+    'https://chromium.googlesource.com/chromium/src/testing@edc96336049d435f9b6d47df1a213a33b4ff6694',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@d25c43455e0c50b53b799f5c4bf2a4867f3dbfb7',
+    'https://chromium.googlesource.com/chromium/src/third_party@82dd665e1743938f6e75cdcebdb7e84603a9c048',
 
   'src/buildtools/linux64': {
     'packages': [
       {
         'package': 'gn/gn/linux-${{arch}}',
-        'version': 'git_revision:1a310e88443018837759c952b113846b0096f65b',
+        'version': 'git_revision:a155c8c78af4030169a8da4886dc0c6b170535de',
       }
     ],
     'dep_type': 'cipd',
@@ -100,7 +100,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/mac-${{arch}}',
-        'version': 'git_revision:1a310e88443018837759c952b113846b0096f65b',
+        'version': 'git_revision:a155c8c78af4030169a8da4886dc0c6b170535de',
       }
     ],
     'dep_type': 'cipd',
@@ -110,7 +110,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/windows-amd64',
-        'version': 'git_revision:1a310e88443018837759c952b113846b0096f65b',
+        'version': 'git_revision:a155c8c78af4030169a8da4886dc0c6b170535de',
       }
     ],
     'dep_type': 'cipd',
@@ -331,13 +331,13 @@ deps = {
   'src/third_party/clang-format/script':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@c2725e0622e1a86d55f14514f2177a39efea4a0e',
   'src/third_party/compiler-rt/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@64c252741e3d8d9906473b1d09733cff93b91f50',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@0cd354e178fec2af0c516235f945ceca8611b337',
   'src/third_party/libc++/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@7ab65651aed6802d2599dcb7a73b1f82d5179d05',
   'src/third_party/libc++abi/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@8f11bb1d4438d0239d0dfc1bd9456a9f31629dda',
   'src/third_party/llvm-libc/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@7434d317bd47832fe6b30fcdf33796ddcc02a7d4',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@dacaf47ceaeddc37b35b9a65eb2ba817bb110069',
   'src/third_party/libunwind/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@17ccf7d110c5526cb77e93cfd8330f491fb2bf18',
 
@@ -488,11 +488,11 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@80053a9bb7260a6979d09c19157136e59d97db76',
+    'https://boringssl.googlesource.com/boringssl.git@d0fcf49574b06de7090cba4ad8124df638a0134e',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@9f26d631e4063c0af3e8ee3bc8826486115372a8',
   'src/third_party/catapult':
-    'https://chromium.googlesource.com/catapult.git@27800c3890b2aaf0bdbefc8ba7b0a0a394cb7e72',
+    'https://chromium.googlesource.com/catapult.git@109c6d7babfc9afbb19af677a0d21fe3a281f2c1',
   'src/third_party/ced/src': {
     'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5',
   },
@@ -505,13 +505,13 @@ deps = {
   'src/third_party/crc32c/src':
     'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558',
   'src/third_party/depot_tools':
-    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@55659c2bf434a4fb508799fa93550822c5309ed5',
+    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@d6aae1009f694a7292d28a5db9a3f7bcb15bad4d',
   'src/third_party/ffmpeg':
     'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@946d97db8d906277085e361892b7efda5152e2f1',
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@73b070473d87f8943e67deddd8d3fc4722ed18aa',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@060a21762295ba058c599a4e879996363bc8ca46',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -537,7 +537,7 @@ deps = {
   'src/third_party/googletest/src':
     'https://chromium.googlesource.com/external/github.com/google/googletest.git@4fe3307fb2d9f86d19777c7eb0e4809e9694dde7',
   'src/third_party/icu': {
-    'url': 'https://chromium.googlesource.com/chromium/deps/icu.git@7971660ba6306a4cc8e6f872905138f59de5bc1d',
+    'url': 'https://chromium.googlesource.com/chromium/deps/icu.git@ee5f27adc28bd3f15b2c293f726d14d2e336cbd5',
   },
   'src/third_party/jdk/current': {
       'packages': [
@@ -573,7 +573,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/kotlin_stdlib',
-              'version': 'r1jin7oxmCEIuca5n9GqpxpPk_aV2Dr0CLAmWSV0tEAC',
+              'version': '9ldUaVxyYClDj9QDPGLyxhrrLE40VxHjm65wiU_4xaoC',
           },
       ],
       'condition': 'checkout_android',
@@ -593,7 +593,7 @@ deps = {
   'src/third_party/libFuzzer/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt/lib/fuzzer.git@bea408a6e01f0f7e6c82a43121fe3af4506c932e',
   'src/third_party/fuzztest/src':
-    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@6a217df9edbe5705885457f10968ebe80aec9504',
+    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@2774d423cc62d44035ad8dd1651d15e4756a163f',
   'src/third_party/libjpeg_turbo':
     'https://chromium.googlesource.com/chromium/deps/libjpeg_turbo.git@d1f5f2393e0d51f840207342ae86e55a86443288',
   'src/third_party/libsrtp':
@@ -609,11 +609,11 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '6391c1361787a9be611fc207447d0af2ddb00f5e',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'f884650a1523d2e48fbc98980d9a5a4a56d883e2',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
-    'https://chromium.googlesource.com/webm/libvpx.git@db312f06fb50089e699f13a9c516daf44ce0f7d8',
+    'https://chromium.googlesource.com/webm/libvpx.git@4fcebeabe58e79255d291acb4cead4ed7953149e',
   'src/third_party/libyuv':
     'https://chromium.googlesource.com/libyuv/libyuv.git@6067afde563c3946eebd94f146b3824ab7a97a9c',
   'src/third_party/lss': {
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@88a9f66c07407b3f53761aff51625cdb60b8f2e0',
+    'https://chromium.googlesource.com/chromium/src/tools@a3ead8a269e96840406ab5aa34a527d6f0d8dcfb',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'F33f66w1lXogUL7EN68PuqtiS1I2B8ptEm0Zq-MhYnIC',
+          'version': 'CW8b7YlBGZSbouGqDIHzMn5HRGQAqAM8d0kEcogTqJMC',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -808,15 +808,15 @@ deps = {
       'packages': [
         {
           'package': 'infra/tools/luci/cas/${{platform}}',
-          'version': 'git_revision:b9d7d1a7774833e136aaa5084a0d150cd5ecb641',
+          'version': 'git_revision:a61d4d67348b452a5e7bf09541f808e8c1101e76',
         },
         {
           'package': 'infra/tools/luci/isolate/${{platform}}',
-          'version': 'git_revision:b9d7d1a7774833e136aaa5084a0d150cd5ecb641',
+          'version': 'git_revision:a61d4d67348b452a5e7bf09541f808e8c1101e76',
         },
         {
           'package': 'infra/tools/luci/swarming/${{platform}}',
-          'version': 'git_revision:b9d7d1a7774833e136aaa5084a0d150cd5ecb641',
+          'version': 'git_revision:a61d4d67348b452a5e7bf09541f808e8c1101e76',
         }
       ],
       'dep_type': 'cipd',
@@ -842,7 +842,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/android_deps/autorolled',
-              'version': '9DXI8FnmXkL2tti6GWrRX5mKZzOJzOVKFbiQLuE4ThYC',
+              'version': 'gOFPZuLmUt7S4rnhusfD5p-Fhy3qllnFyNIwDbThlusC',
           },
       ],
       'condition': 'checkout_android and non_git_source',
@@ -864,7 +864,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7364b490b5f78d58efe23ea76e74210fd6c3c76f',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '57b1de2330d0660f353037a743fada210bde41ce',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '661cdb227f2adc2a649b48518bceb88ea4126e3a',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From 74721a160c40395d83cc487dea9f0a9a5121d656 Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Wed, 25 Feb 2026 15:09:06 +0100
Subject: [PATCH 181/472] Delegate crc32 calculation to abseil

To reduce number of third_party dependencies of the dcsctp library.

Bug: None
Change-Id: I251d24cc214860342abcc73cdadaf61e3cd02941
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452180
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47034}
---
 DEPS                        |  1 +
 g3doc/abseil-in-webrtc.md   |  1 +
 net/dcsctp/packet/BUILD.gn  |  4 +++-
 net/dcsctp/packet/DEPS      |  3 ---
 net/dcsctp/packet/crc32c.cc | 16 ++++++----------
 5 files changed, 11 insertions(+), 14 deletions(-)
 delete mode 100644 net/dcsctp/packet/DEPS

diff --git a/DEPS b/DEPS
index 98f788ed71..8451e1b5b2 100644
--- a/DEPS
+++ b/DEPS
@@ -1352,6 +1352,7 @@ include_rules = [
   "+absl/cleanup/cleanup.h",
   "+absl/container",
   "-absl/container/fixed_array.h",
+  "+absl/crc",
   "+absl/functional/any_invocable.h",
   "+absl/functional/bind_front.h",
   "+absl/memory/memory.h",
diff --git a/g3doc/abseil-in-webrtc.md b/g3doc/abseil-in-webrtc.md
index af305cb1c5..fd74182176 100644
--- a/g3doc/abseil-in-webrtc.md
+++ b/g3doc/abseil-in-webrtc.md
@@ -40,6 +40,7 @@ on a monolithic Abseil build target that will generate a shared library.
 * The macros in `absl/base/attributes.h`, `absl/base/config.h` and
   `absl/base/macros.h`.
 * `absl/numeric/bits.h`
+* `absl/crc`
 * Single argument `absl::StrCat`
 
 * ABSL_FLAG is allowed in tests and tools, but disallowed in in non-test code.
diff --git a/net/dcsctp/packet/BUILD.gn b/net/dcsctp/packet/BUILD.gn
index e92489329f..852788b3f5 100644
--- a/net/dcsctp/packet/BUILD.gn
+++ b/net/dcsctp/packet/BUILD.gn
@@ -47,7 +47,9 @@ rtc_source_set("data") {
 rtc_library("crc32c") {
   deps = [
     "../../../api:array_view",
-    "//third_party/crc32c",
+    "//third_party/abseil-cpp/absl/crc:crc32c",
+    "//third_party/abseil-cpp/absl/numeric:bits",
+    "//third_party/abseil-cpp/absl/strings:string_view",
   ]
   sources = [
     "crc32c.cc",
diff --git a/net/dcsctp/packet/DEPS b/net/dcsctp/packet/DEPS
deleted file mode 100644
index 206b405b82..0000000000
--- a/net/dcsctp/packet/DEPS
+++ /dev/null
@@ -1,3 +0,0 @@
-include_rules = [
-    "+third_party/crc32c",
-]
diff --git a/net/dcsctp/packet/crc32c.cc b/net/dcsctp/packet/crc32c.cc
index 8318b0dc75..a55749b2bd 100644
--- a/net/dcsctp/packet/crc32c.cc
+++ b/net/dcsctp/packet/crc32c.cc
@@ -11,20 +11,16 @@
 
 #include <cstdint>
 
+#include "absl/crc/crc32c.h"
+#include "absl/numeric/bits.h"
+#include "absl/strings/string_view.h"
 #include "api/array_view.h"
-#include "third_party/crc32c/src/include/crc32c/crc32c.h"
 
 namespace dcsctp {
 
 uint32_t GenerateCrc32C(webrtc::ArrayView<const uint8_t> data) {
-  uint32_t crc32c = crc32c_value(data.data(), data.size());
-
-  // Byte swapping for little endian byte order:
-  uint8_t byte0 = crc32c;
-  uint8_t byte1 = crc32c >> 8;
-  uint8_t byte2 = crc32c >> 16;
-  uint8_t byte3 = crc32c >> 24;
-  crc32c = ((byte0 << 24) | (byte1 << 16) | (byte2 << 8) | byte3);
-  return crc32c;
+  absl::crc32c_t crc32 = absl::ComputeCrc32c(absl::string_view(
+      reinterpret_cast<const char*>(data.data()), data.size()));
+  return absl::byteswap(static_cast<uint32_t>(crc32));
 }
 }  // namespace dcsctp

From bfc36bfabcaccde6b1f8ae2596dc92f4d5eaebef Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Tue, 3 Mar 2026 17:43:48 +0100
Subject: [PATCH 182/472] Generalize FuzzDataHelper::Read function

To make it easier to use in fuzzers:
- change it not to crash when there is not enough fuzz data left.
- change it to read any trivial object, not just integers

Bug: None
Change-Id: I5dfb47d3accc3c856dfc5d4376676354953d3f27
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/411660
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47035}
---
 test/BUILD.gn                                 |  1 +
 test/fuzzers/BUILD.gn                         | 15 ++--
 .../audio_processing_sample_rate_fuzzer.cc    |  3 +-
 test/fuzzers/fuzz_data_helper.h               | 45 ++++++++---
 test/fuzzers/fuzz_data_helper_unittest.cc     | 75 +++++++++++++++++++
 test/fuzzers/residual_echo_detector_fuzzer.cc |  2 +-
 .../rtp_frame_reference_finder_fuzzer.cc      | 15 ++--
 7 files changed, 126 insertions(+), 30 deletions(-)
 create mode 100644 test/fuzzers/fuzz_data_helper_unittest.cc

diff --git a/test/BUILD.gn b/test/BUILD.gn
index cf3493505f..96da3e9f11 100644
--- a/test/BUILD.gn
+++ b/test/BUILD.gn
@@ -782,6 +782,7 @@ if (rtc_include_tests) {
         "../rtc_base/synchronization:mutex",
         "../rtc_base/system:file_wrapper",
         "../system_wrappers",
+        "fuzzers:fuzz_data_helper_unittest",
         "jitter:jitter_unittests",
         "pc/e2e:e2e_unittests",
         "pc/e2e/analyzer/video:video_analyzer_unittests",
diff --git a/test/fuzzers/BUILD.gn b/test/fuzzers/BUILD.gn
index f44ec6acc1..2a78afab87 100644
--- a/test/fuzzers/BUILD.gn
+++ b/test/fuzzers/BUILD.gn
@@ -33,11 +33,7 @@ rtc_library("webrtc_fuzzer_main") {
 rtc_library("fuzz_data_helper") {
   testonly = true
   sources = [ "fuzz_data_helper.h" ]
-  deps = [
-    "../../modules/rtp_rtcp:rtp_rtcp_format",
-    "../../rtc_base:checks",
-    "//third_party/abseil-cpp/absl/strings:string_view",
-  ]
+  deps = [ "//third_party/abseil-cpp/absl/strings:string_view" ]
 
   # Only fuzzer targets can depend on this.
   visibility = [
@@ -46,6 +42,15 @@ rtc_library("fuzz_data_helper") {
   ]
 }
 
+rtc_library("fuzz_data_helper_unittest") {
+  testonly = true
+  sources = [ "fuzz_data_helper_unittest.cc" ]
+  deps = [
+    ":fuzz_data_helper",
+    "..:test_support",
+  ]
+}
+
 set_defaults("webrtc_fuzzer_test") {
   configs = rtc_add_configs
 }
diff --git a/test/fuzzers/audio_processing_sample_rate_fuzzer.cc b/test/fuzzers/audio_processing_sample_rate_fuzzer.cc
index 8ce3d059b0..3f7bec7e4c 100644
--- a/test/fuzzers/audio_processing_sample_rate_fuzzer.cc
+++ b/test/fuzzers/audio_processing_sample_rate_fuzzer.cc
@@ -39,8 +39,7 @@ void GenerateFloatFrame(FuzzDataHelper& fuzz_data,
       AudioProcessing::GetFrameSize(input_rate);
   RTC_DCHECK_LE(samples_per_input_channel, kMaxSamplesPerChannel);
   for (int i = 0; i < num_channels; ++i) {
-    float channel_value;
-    fuzz_data.CopyTo<float>(&channel_value);
+    float channel_value = fuzz_data.Read<float>();
     std::fill(float_frames[i], float_frames[i] + samples_per_input_channel,
               channel_value);
   }
diff --git a/test/fuzzers/fuzz_data_helper.h b/test/fuzzers/fuzz_data_helper.h
index 0e6486903b..d33f233268 100644
--- a/test/fuzzers/fuzz_data_helper.h
+++ b/test/fuzzers/fuzz_data_helper.h
@@ -19,8 +19,6 @@
 #include <type_traits>
 
 #include "absl/strings/string_view.h"
-#include "modules/rtp_rtcp/source/byte_io.h"
-#include "rtc_base/checks.h"
 
 namespace webrtc {
 
@@ -35,16 +33,13 @@ class FuzzDataHelper {
 
   // Reads and returns data of type T.
   template <typename T>
-  T Read() {
-    RTC_CHECK(CanReadBytes(sizeof(T)));
-    T x = ByteReader<T>::ReadLittleEndian(&data_[data_ix_]);
-    data_ix_ += sizeof(T);
-    return x;
-  }
+    requires(std::is_trivial_v<T>)
+  T Read();
 
   // Reads and returns data of type T. Returns default_value if not enough
   // fuzzer input remains to read a T.
   template <typename T>
+    requires(std::is_trivial_v<T>)
   T ReadOrDefaultValue(T default_value) {
     if (!CanReadBytes(sizeof(T))) {
       return default_value;
@@ -54,8 +49,8 @@ class FuzzDataHelper {
 
   // Like ReadOrDefaultValue, but replaces the value 0 with default_value.
   template <typename T>
+    requires(std::is_integral_v<T>)
   T ReadOrDefaultValueNotZero(T default_value) {
-    static_assert(std::is_integral<T>::value, "");
     T x = ReadOrDefaultValue(default_value);
     return x == 0 ? default_value : x;
   }
@@ -110,11 +105,15 @@ class FuzzDataHelper {
   // If sizeof(T) > BytesLeft then the remaining bytes will be used and the rest
   // of the object will be zero initialized.
   template <typename T>
-  void CopyTo(T* object) {
-    memset(object, 0, sizeof(T));
+    requires(std::is_trivial_v<T>)
+  void CopyTo(T& object) {
+    std::span<uint8_t, sizeof(T)> object_memory(
+        reinterpret_cast<uint8_t*>(&object), sizeof(T));
 
     size_t bytes_to_copy = std::min(BytesLeft(), sizeof(T));
-    memcpy(object, data_.data() + data_ix_, bytes_to_copy);
+    std::ranges::copy(data_.subspan(data_ix_, bytes_to_copy),
+                      object_memory.begin());
+    std::ranges::fill(object_memory.subspan(bytes_to_copy), uint8_t{0});
     data_ix_ += bytes_to_copy;
   }
 
@@ -129,6 +128,28 @@ class FuzzDataHelper {
   size_t data_ix_ = 0;
 };
 
+template <typename T>
+  requires(std::is_trivial_v<T>)
+T FuzzDataHelper::Read() {
+  if constexpr (sizeof(T) == 1) {
+    if (BytesLeft() == 0) {
+      return {};
+    } else {
+      return static_cast<T>(data_[data_ix_++]);
+    }
+  }
+
+  T value;
+  CopyTo(value);
+  return value;
+}
+
+template <>
+inline bool FuzzDataHelper::Read<bool>() {
+  // Return `true' or 'false' with 50% chance each.
+  return (Read<uint8_t>() & 0b1) != 0;
+}
+
 }  // namespace webrtc
 
 #endif  // TEST_FUZZERS_FUZZ_DATA_HELPER_H_
diff --git a/test/fuzzers/fuzz_data_helper_unittest.cc b/test/fuzzers/fuzz_data_helper_unittest.cc
new file mode 100644
index 0000000000..b384b6c13e
--- /dev/null
+++ b/test/fuzzers/fuzz_data_helper_unittest.cc
@@ -0,0 +1,75 @@
+/*
+ *  Copyright (c) 2026 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/fuzzers/fuzz_data_helper.h"
+
+#include <cstdint>
+
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc::test {
+namespace {
+
+using ::testing::Each;
+using ::testing::ElementsAre;
+
+TEST(FuzzDataHelperTest, ReadsIntegers) {
+  const uint8_t kData[] = {0xFE, 0x01, 0x02, 0x03, 0x04, 0x05};
+  FuzzDataHelper fuzz_data(kData);
+  ASSERT_EQ(fuzz_data.BytesLeft(), 6u);
+
+  EXPECT_EQ(fuzz_data.Read<int8_t>(), -2);
+  EXPECT_EQ(fuzz_data.BytesLeft(), 5u);
+
+  EXPECT_EQ(fuzz_data.Read<uint8_t>(), 0x01);
+  EXPECT_EQ(fuzz_data.BytesLeft(), 4u);
+
+  EXPECT_EQ(fuzz_data.Read<uint16_t>(), 0x0302);
+  EXPECT_EQ(fuzz_data.BytesLeft(), 2u);
+
+  // Read 4 bytes while only two left in `kData` - expect that succeed and
+  // returns predictable results (fills missing data with zeros).
+  EXPECT_EQ(fuzz_data.Read<uint32_t>(), 0x0000'0504u);
+  EXPECT_EQ(fuzz_data.BytesLeft(), 0u);
+}
+
+struct PlainOldData {
+  uint32_t value;
+  uint8_t array[3];
+};
+
+TEST(FuzzDataHelperTest, ReadsAnyType) {
+  const uint8_t kData[] = {0x01, 0x01, 0x02, 0x03, 0x04, 0x05};
+  FuzzDataHelper fuzz_data(kData);
+
+  PlainOldData pod = fuzz_data.Read<PlainOldData>();
+  EXPECT_EQ(pod.value, 0x03020101u);
+  EXPECT_THAT(pod.array, ElementsAre(0x04, 0x05, 0));
+}
+
+TEST(FuzzDataHelperTest, ReadsZerosWhenNoDataLeft) {
+  const uint8_t kData[] = {0xFE};
+  FuzzDataHelper fuzz_data(kData);
+  ASSERT_EQ(fuzz_data.BytesLeft(), 1u);
+  fuzz_data.Read<uint8_t>();
+
+  ASSERT_EQ(fuzz_data.BytesLeft(), 0u);
+
+  EXPECT_EQ(fuzz_data.Read<int8_t>(), 0);
+  EXPECT_EQ(fuzz_data.Read<uint8_t>(), 0u);
+  EXPECT_EQ(fuzz_data.Read<uint16_t>(), 0u);
+  PlainOldData pod = fuzz_data.Read<PlainOldData>();
+  EXPECT_EQ(pod.value, 0u);
+  EXPECT_THAT(pod.array, Each(0u));
+}
+
+}  // namespace
+}  // namespace webrtc::test
diff --git a/test/fuzzers/residual_echo_detector_fuzzer.cc b/test/fuzzers/residual_echo_detector_fuzzer.cc
index b191a647d7..43845bc015 100644
--- a/test/fuzzers/residual_echo_detector_fuzzer.cc
+++ b/test/fuzzers/residual_echo_detector_fuzzer.cc
@@ -46,7 +46,7 @@ void FuzzOneInput(FuzzDataHelper fuzz_data) {
   echo_detector->AnalyzeCaptureAudio(input);
   for (size_t i = 0; i < 2 * kNrOfUpdates; ++i) {
     // Convert 4 input bytes to a float.
-    fuzz_data.CopyTo(&input[0]);
+    fuzz_data.CopyTo(input[0]);
     if (!isfinite(input[0]) || fabs(input[0]) > maxFuzzedValue) {
       // Ignore infinity, nan values and values that are unrealistically large.
       continue;
diff --git a/test/fuzzers/rtp_frame_reference_finder_fuzzer.cc b/test/fuzzers/rtp_frame_reference_finder_fuzzer.cc
index 4e1da6040d..bbfc288121 100644
--- a/test/fuzzers/rtp_frame_reference_finder_fuzzer.cc
+++ b/test/fuzzers/rtp_frame_reference_finder_fuzzer.cc
@@ -38,19 +38,14 @@ class DataReader {
   DataReader(FuzzDataHelper fuzz_data) : data_(fuzz_data) {}
 
   template <typename T>
-  void CopyTo(T* object) {
+  void CopyTo(T& object) {
     return data_.CopyTo(object);
   }
 
   template <typename T>
   T GetNum() {
-    T t = {};
-    if (data_.BytesLeft() >= sizeof(T)) {
-      CopyTo(&t);
-    }
-    return t;
+    return data_.Read<T>();
   }
-
   bool MoreToRead() { return data_.BytesLeft() > 0; }
 
  private:
@@ -63,7 +58,7 @@ RTPVideoHeaderH264 GenerateRTPVideoHeaderH264(DataReader* reader) {
   result.packetization_type = reader->GetNum<H264PacketizationTypes>();
   int nalus_length = reader->GetNum<uint8_t>();
   for (int i = 0; i < nalus_length; ++i) {
-    reader->CopyTo(&result.nalus.emplace_back());
+    reader->CopyTo(result.nalus.emplace_back());
   }
   result.packetization_mode = reader->GetNum<H264PacketizationMode>();
   return result;
@@ -123,11 +118,11 @@ void FuzzOneInput(FuzzDataHelper fuzz_data) {
     switch (codec) {
       case kVideoCodecVP8:
         reader.CopyTo(
-            &video_header.video_type_header.emplace<RTPVideoHeaderVP8>());
+            video_header.video_type_header.emplace<RTPVideoHeaderVP8>());
         break;
       case kVideoCodecVP9:
         reader.CopyTo(
-            &video_header.video_type_header.emplace<RTPVideoHeaderVP9>());
+            video_header.video_type_header.emplace<RTPVideoHeaderVP9>());
         break;
       case kVideoCodecH264:
         video_header.video_type_header = GenerateRTPVideoHeaderH264(&reader);

From 7915b363491c027c034afa183d51c7f07d76cd9f Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Mon, 2 Mar 2026 20:15:28 +0100
Subject: [PATCH 183/472] Invalidate RtpSender parameter cache on codec
 fallback/switch
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

The RtpSender maintains a cache of RTP parameters on the signaling
thread to improve performance and avoid frequent blocking calls to the
worker thread. However, some parameter changes—such as encoder
fallbacks or autonomous codec switches—occur directly on the worker
thread.

These out-of-band updates previously left the signaling thread cache
in a stale state, leading to consistency check failures and potential
logic errors.

This change introduces a callback mechanism in the MediaSendChannel to
notify the RtpSender when parameters change autonomously. Upon
receiving this notification, the RtpSender invalidates its cached
parameters and transaction ID on the signaling thread, ensuring
subsequent requests fetch fresh data from the worker thread.

Additionally, this re-enables the consistency RTC_DCHECK in
GetParametersInternal that was previously disabled due to these cache
mismatches.

Bug: b/478050997
Change-Id: Icd57e04670d7af33844db4d58685c02314a2f531
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452480
Reviewed-by: Sameer Vijaykar <samvi@google.com>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47036}
---
 media/base/media_channel.h          |   4 +
 media/engine/webrtc_video_engine.cc |   8 +-
 media/engine/webrtc_video_engine.h  |  10 ++
 pc/BUILD.gn                         |   3 +
 pc/peer_connection_rtp_unittest.cc  | 206 ++++++++++++++++++++++++++++
 pc/rtp_sender.cc                    |  25 +++-
 pc/rtp_sender.h                     |   8 ++
 7 files changed, 261 insertions(+), 3 deletions(-)

diff --git a/media/base/media_channel.h b/media/base/media_channel.h
index a41a12dac2..3cc8f7ba2f 100644
--- a/media/base/media_channel.h
+++ b/media/base/media_channel.h
@@ -241,6 +241,10 @@ class MediaSendChannelInterface {
   // Called whenever the list of sending SSRCs changes.
   virtual void SetSsrcListChangedCallback(
       absl::AnyInvocable<void(const std::set<uint32_t>&)> callback) = 0;
+  // Called whenever the parameters change autonomously on the worker thread.
+  // This is used for cache invalidation on the signaling thread.
+  virtual void SetParametersChangedCallback(
+      absl::AnyInvocable<void()> callback) {}
 };
 
 class MediaReceiveChannelInterface {
diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc
index c6d876c517..540b3a12ef 100644
--- a/media/engine/webrtc_video_engine.cc
+++ b/media/engine/webrtc_video_engine.cc
@@ -1257,7 +1257,9 @@ void WebRtcVideoSendChannel::RequestEncoderFallback() {
   params.negotiated_codecs = negotiated_codecs_;
   params.negotiated_codecs->erase(params.negotiated_codecs->begin());
   params.send_codec = params.negotiated_codecs->front();
-  ApplyChangedParams(params);
+  if (ApplyChangedParams(params) && parameters_changed_callback_) {
+    parameters_changed_callback_();
+  }
 }
 
 void WebRtcVideoSendChannel::RequestEncoderSwitch(const SdpVideoFormat& format,
@@ -1287,7 +1289,9 @@ void WebRtcVideoSendChannel::RequestEncoderSwitch(const SdpVideoFormat& format,
 
       ChangedSenderParameters params;
       params.send_codec = new_codec_setting;
-      ApplyChangedParams(params);
+      if (ApplyChangedParams(params) && parameters_changed_callback_) {
+        parameters_changed_callback_();
+      }
       return;
     }
   }
diff --git a/media/engine/webrtc_video_engine.h b/media/engine/webrtc_video_engine.h
index 6306b91e4f..3e2e7f2e0b 100644
--- a/media/engine/webrtc_video_engine.h
+++ b/media/engine/webrtc_video_engine.h
@@ -233,6 +233,12 @@ class WebRtcVideoSendChannel : public MediaChannelUtil,
     ssrc_list_changed_callback_ = std::move(callback);
   }
 
+  void SetParametersChangedCallback(
+      absl::AnyInvocable<void()> callback) override {
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    parameters_changed_callback_ = std::move(callback);
+  }
+
   // Implemented for VideoMediaChannelTest.
   bool sending() const {
     RTC_DCHECK_RUN_ON(&thread_checker_);
@@ -501,6 +507,10 @@ class WebRtcVideoSendChannel : public MediaChannelUtil,
   // Callback invoked whenever the list of SSRCs changes.
   absl::AnyInvocable<void(const std::set<uint32_t>&)>
       ssrc_list_changed_callback_;
+
+  // Callback invoked whenever the sender parameters change.
+  absl::AnyInvocable<void()> parameters_changed_callback_
+      RTC_GUARDED_BY(&thread_checker_);
 };
 
 class WebRtcVideoReceiveChannel : public MediaChannelUtil,
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 0d67ff1119..9cd20428be 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -2641,10 +2641,12 @@ if (rtc_include_tests && !build_with_chromium) {
       "../api/video:builtin_video_bitrate_allocator_factory",
       "../api/video:encoded_image",
       "../api/video:recordable_encoded_frame",
+      "../api/video:render_resolution",
       "../api/video:video_bitrate_allocator_factory",
       "../api/video:video_codec_constants",
       "../api/video:video_frame",
       "../api/video:video_rtp_headers",
+      "../api/video:video_stream_encoder",
       "../api/video_codecs:builtin_video_decoder_factory",
       "../api/video_codecs:builtin_video_encoder_factory",
       "../api/video_codecs:scalability_mode",
@@ -2669,6 +2671,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "../media:media_constants",
       "../media:media_engine",
       "../media:rid_description",
+      "../media:rtc_audio_video",
       "../media:rtc_data_sctp_transport_internal",
       "../media:rtc_media_config",
       "../media:rtc_media_tests_utils",
diff --git a/pc/peer_connection_rtp_unittest.cc b/pc/peer_connection_rtp_unittest.cc
index f82ff87ec6..d0d83520a5 100644
--- a/pc/peer_connection_rtp_unittest.cc
+++ b/pc/peer_connection_rtp_unittest.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <algorithm>
 #include <cstddef>
 #include <cstdint>
 #include <memory>
@@ -33,6 +34,9 @@
 #include "api/scoped_refptr.h"
 #include "api/set_remote_description_observer_interface.h"
 #include "api/test/rtc_error_matchers.h"
+#include "api/units/data_rate.h"
+#include "api/video/render_resolution.h"
+#include "api/video/video_stream_encoder_settings.h"
 #include "api/video_codecs/sdp_video_format.h"
 #include "api/video_codecs/video_decoder_factory_template.h"
 #include "api/video_codecs/video_decoder_factory_template_dav1d_adapter.h"
@@ -45,9 +49,12 @@
 #include "api/video_codecs/video_encoder_factory_template_libvpx_vp9_adapter.h"
 #include "api/video_codecs/video_encoder_factory_template_open_h264_adapter.h"
 #include "media/base/codec.h"
+#include "media/base/media_channel.h"
 #include "media/base/stream_params.h"
+#include "media/engine/webrtc_video_engine.h"
 #include "pc/media_session.h"
 #include "pc/peer_connection_wrapper.h"
+#include "pc/rtp_transceiver.h"
 #include "pc/session_description.h"
 #include "pc/test/fake_audio_capture_module.h"
 #include "pc/test/integration_test_helpers.h"
@@ -1997,6 +2004,205 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan,
   EXPECT_EQ("stream5", callee_streams[2]->id());
 }
 
+TEST_F(PeerConnectionRtpTestUnifiedPlan,
+       SendParamsCorrectWhenSendCodecChanges) {
+  auto caller = CreatePeerConnection();
+  auto callee = CreatePeerConnection();
+
+  auto sender = caller->AddVideoTrack("video_track");
+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
+
+  // Call GetParameters() to populate the cache.
+  auto parameters1 = sender->GetParameters();
+  ASSERT_GT(parameters1.codecs.size(), 0u);
+
+  // Force renegotiation to change the negotiated codec.
+  auto transceiver = callee->pc()->GetTransceivers()[0];
+  auto capabilities =
+      caller->pc_factory()->GetRtpReceiverCapabilities(MediaType::VIDEO);
+  std::vector<RtpCodecCapability> codecs = capabilities.codecs;
+  // Remove the currently negotiated codec from the receiver preferences
+  codecs.erase(std::remove_if(codecs.begin(), codecs.end(),
+                              [&](const RtpCodecCapability& c) {
+                                return c.name == parameters1.codecs[0].name;
+                              }),
+               codecs.end());
+
+  auto error = transceiver->SetCodecPreferences(codecs);
+  EXPECT_TRUE(error.ok());
+
+  // Exchange offer/answer again. This forces WebRtcVideoSendChannel to
+  // change the send codec without notifying RtpSender!
+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
+
+  // Call GetParameters(). The DCHECK evaluates cached_parameters_ against
+  // what's actually on the worker thread. Since we bypassed SetParameters,
+  // this should crash if the bug is present, or return the right values.
+  auto parameters2 = sender->GetParameters();
+  ASSERT_GT(parameters2.codecs.size(), 0u);
+  EXPECT_NE(parameters1.codecs[0].name, parameters2.codecs[0].name);
+}
+
+TEST_F(PeerConnectionRtpTestUnifiedPlan,
+       SendParamsCorrectWhenContentHintChanges) {
+  auto caller = CreatePeerConnection();
+  auto callee = CreatePeerConnection();
+
+  auto track = caller->CreateVideoTrack("video");
+
+  RtpTransceiverInit init;
+  init.direction = RtpTransceiverDirection::kSendRecv;
+  RtpEncodingParameters encoding;
+  encoding.active = true;
+  // Set an unsupported scalability mode so that it gets mutated
+  // by FallbackToDefaultScalabilityModeIfNotSupported later.
+  encoding.scalability_mode = "L3T3_KEY";
+  init.send_encodings.push_back(encoding);
+
+  auto transceiver = caller->AddTransceiver(track, init);
+  ASSERT_TRUE(transceiver != nullptr);
+  auto sender = transceiver->sender();
+
+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
+
+  // Get initial parameters to populate the signaling thread cache.
+  auto parameters = sender->GetParameters();
+
+  // Change content hint, which triggers VideoRtpSender::OnChanged() ->
+  // SetVideoSend() -> WebRtcVideoSendStream::SetOptions() ->
+  // ReconfigureEncoder() -> FallbackToDefaultScalabilityModeIfNotSupported()
+  // on the worker thread.
+  track->set_content_hint(VideoTrackInterface::ContentHint::kDetailed);
+
+  // Calling GetParameters will fetch the new parameters from the worker thread
+  // and compare them to the signaling thread cache, triggering the DCHECK if
+  // the cache is stale.
+  auto parameters_after = sender->GetParameters();
+
+  // If the DCHECK is not fatal, we manually verify the mismatch happened.
+  EXPECT_EQ(parameters_after.encodings.size(), 1u);
+}
+
+TEST_F(PeerConnectionRtpTestUnifiedPlan,
+       CacheMismatchWhenEncoderSelectorChanges) {
+  auto caller = CreatePeerConnection();
+  auto callee = CreatePeerConnection();
+
+  auto track = caller->CreateVideoTrack("video");
+
+  auto transceiver = caller->AddTransceiver(track);
+  ASSERT_TRUE(transceiver != nullptr);
+  auto sender = transceiver->sender();
+
+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
+
+  // Get initial parameters to populate the signaling thread cache.
+  auto parameters = sender->GetParameters();
+
+  class MockEncoderSelector
+      : public VideoEncoderFactory::EncoderSelectorInterface {
+   public:
+    void OnCurrentEncoder(const SdpVideoFormat& format) override {}
+    std::optional<SdpVideoFormat> OnAvailableBitrate(
+        const DataRate& rate) override {
+      return SdpVideoFormat("VP8");
+    }
+    std::optional<SdpVideoFormat> OnEncoderBroken() override {
+      return std::nullopt;
+    }
+    std::optional<SdpVideoFormat> OnResolutionChange(
+        const RenderResolution& resolution) override {
+      return std::nullopt;
+    }
+  };
+
+  auto encoder_selector = std::make_unique<MockEncoderSelector>();
+  auto format_to_switch_to =
+      *encoder_selector->OnAvailableBitrate(DataRate::Zero());
+
+  // Change encoder selector
+  sender->SetEncoderSelector(std::move(encoder_selector));
+
+  auto transceivers =
+      caller->GetInternalPeerConnection()->GetTransceiversInternal();
+  ASSERT_FALSE(transceivers.empty());
+  auto* transceiver_internal = transceivers[0]->internal();
+  ASSERT_TRUE(transceiver_internal != nullptr);
+
+  auto* media_channel = transceiver_internal->media_send_channel();
+  ASSERT_TRUE(media_channel != nullptr);
+
+  auto* worker_thread = caller->GetInternalPeerConnection()->worker_thread();
+  ASSERT_TRUE(worker_thread != nullptr);
+
+  worker_thread->BlockingCall([&] {
+    // Simulate VideoStreamEncoder calling RequestEncoderSwitch after getting
+    // the format from the EncoderSelector.
+    auto* switch_callback = static_cast<webrtc::EncoderSwitchRequestCallback*>(
+        static_cast<webrtc::WebRtcVideoSendChannel*>(
+            media_channel->AsVideoSendChannel()));
+    switch_callback->RequestEncoderSwitch(format_to_switch_to,
+                                          /*allow_default_fallback=*/false);
+  });
+
+  // Allow the signaling thread to process the cache invalidation task posted
+  // by RequestEncoderSwitch.
+  run_loop_.Flush();
+
+  // Calling GetParameters will fetch the new parameters from the worker thread
+  // and compare them to the signaling thread cache, triggering the DCHECK if
+  // the cache is stale.
+  auto parameters_after = sender->GetParameters();
+
+  EXPECT_EQ(parameters_after.encodings.size(), 1u);
+}
+
+TEST_F(PeerConnectionRtpTestUnifiedPlan, SendParamsCorrectWhenEncoderFallback) {
+  auto caller = CreatePeerConnection();
+  auto callee = CreatePeerConnection();
+
+  auto track = caller->CreateVideoTrack("video");
+
+  auto transceiver = caller->AddTransceiver(track);
+  ASSERT_TRUE(transceiver != nullptr);
+  auto sender = transceiver->sender();
+
+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
+
+  // Get initial parameters to populate the signaling thread cache.
+  auto parameters = sender->GetParameters();
+
+  auto transceivers =
+      caller->GetInternalPeerConnection()->GetTransceiversInternal();
+  ASSERT_FALSE(transceivers.empty());
+  auto* transceiver_internal = transceivers[0]->internal();
+  ASSERT_TRUE(transceiver_internal != nullptr);
+
+  auto* media_channel = transceiver_internal->media_send_channel();
+  ASSERT_TRUE(media_channel != nullptr);
+
+  auto* worker_thread = caller->GetInternalPeerConnection()->worker_thread();
+  ASSERT_TRUE(worker_thread != nullptr);
+
+  worker_thread->BlockingCall([&] {
+    // Simulate an encoder fallback on the worker thread.
+    auto* fallback_callback =
+        static_cast<webrtc::EncoderSwitchRequestCallback*>(
+            static_cast<webrtc::WebRtcVideoSendChannel*>(
+                media_channel->AsVideoSendChannel()));
+    fallback_callback->RequestEncoderFallback();
+  });
+
+  // Allow the signaling thread to process the cache invalidation task posted
+  // by RequestEncoderFallback.
+  run_loop_.Flush();
+
+  // Call GetParameters. This triggers an internal consistency check in the
+  // `GetParameters()` implementation.
+  auto parameters_after = sender->GetParameters();
+  EXPECT_EQ(parameters_after.codecs.size(), parameters.codecs.size());
+}
+
 RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
 INSTANTIATE_TEST_SUITE_P(PeerConnectionRtpTest,
                          PeerConnectionRtpTest,
diff --git a/pc/rtp_sender.cc b/pc/rtp_sender.cc
index 5330733536..ca3b131fb3 100644
--- a/pc/rtp_sender.cc
+++ b/pc/rtp_sender.cc
@@ -253,6 +253,7 @@ void RtpSenderBase::SetEncoderSelectorOnChannel() {
   if (stopped_ || ssrc_ == 0) {
     return;
   }
+
   worker_thread_->BlockingCall([&, ssrc = ssrc_] {
     RTC_DCHECK_RUN_ON(worker_thread_);
     if (media_channel_)
@@ -272,14 +273,33 @@ void RtpSenderBase::SetMediaChannel(MediaSendChannelInterface* media_channel) {
     return;
   }
 
+  if (media_channel_) {
+    media_channel_->SetParametersChangedCallback(nullptr);
+  }
+
   // Note that setting the media_channel_ to nullptr and clearing the send state
   // via ClearSend_w, are separate operations. Stopping the actual send
   // operation, needs to be done via any of the paths that end up with a call to
   // ClearSend_w(), such as DetachTrackAndGetStopTask().
   media_channel_ = media_channel;
+  if (media_channel_) {
+    media_channel_->SetParametersChangedCallback(
+        [this] { OnParametersChanged(); });
+  }
   media_channel_ ? worker_safety_->SetAlive() : worker_safety_->SetNotAlive();
 }
 
+void RtpSenderBase::OnParametersChanged() {
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  RTC_LOG(LS_INFO) << "RtpSender: OnParametersChanged signaled.";
+  signaling_thread_->PostTask(SafeTask(signaling_safety_.flag(), [this] {
+    RTC_DCHECK_RUN_ON(signaling_thread_);
+    cached_parameters_.reset();
+    last_transaction_id_.reset();
+    RTC_LOG(LS_INFO) << "RtpSender: OnParametersChanged cache cleared.";
+  }));
+}
+
 RtpParameters RtpSenderBase::GetParametersInternal(bool may_use_cache,
                                                    bool with_all_layers) const {
   RTC_DCHECK_RUN_ON(signaling_thread_);
@@ -366,7 +386,7 @@ RtpParameters RtpSenderBase::GetParameters() const {
     // TODO(b/478050997): Re-enable this check once the downstream issue is
     // resolved.
     // RTC_DCHECK(cached_filtered == result)
-    //     << "The cached value should have been equal (filtered)";
+    //    << "The cached value should have been equal (filtered)";
   }
 #endif
   return result;
@@ -646,6 +666,7 @@ void RtpSenderBase::SetStreams(const std::vector<std::string>& stream_ids) {
 bool RtpSenderBase::SetTrack(MediaStreamTrackInterface* track) {
   RTC_DCHECK_RUN_ON(signaling_thread_);
   TRACE_EVENT0("webrtc", "RtpSenderBase::SetTrack");
+
   if (stopped_) {
     RTC_LOG(LS_ERROR) << "SetTrack can't be called on a stopped RtpSender.";
     return false;
@@ -1065,6 +1086,7 @@ void AudioRtpSender::SetSend() {
   // `track_->enabled()` hops to the signaling thread, so call it before we hop
   // to the worker thread or else it will deadlock.
   bool track_enabled = track_->enabled();
+  InvalidateCache();
   bool success = worker_thread_->BlockingCall([&, ssrc = ssrc_] {
     RTC_DCHECK_RUN_ON(worker_thread_);
     return media_channel_
@@ -1206,6 +1228,7 @@ void VideoRtpSender::SetSend() {
       break;
   }
   auto* video_track = static_cast<VideoTrackInterface*>(track_.get());
+  InvalidateCache();
   bool success = worker_thread_->BlockingCall([&, ssrc = ssrc_] {
     RTC_DCHECK_RUN_ON(worker_thread_);
     return media_channel_ ? video_media_channel()->SetVideoSend(ssrc, &options,
diff --git a/pc/rtp_sender.h b/pc/rtp_sender.h
index b08aca1f60..dc9591b0e2 100644
--- a/pc/rtp_sender.h
+++ b/pc/rtp_sender.h
@@ -239,6 +239,14 @@ class RtpSenderBase : public RtpSenderInternal, public ObserverInterface {
   void SetObserver(RtpSenderObserverInterface* observer) override;
 
  protected:
+  void InvalidateCache() {
+    RTC_DCHECK_RUN_ON(signaling_thread_);
+    cached_parameters_.reset();
+  }
+
+  // Called by the media channel when parameters change autonomously on the
+  // worker thread (e.g., encoder fallback).
+  void OnParametersChanged();
   // If `set_streams_observer` is not null, it is invoked when SetStreams()
   // is called. `set_streams_observer` is not owned by this object. If not
   // null, it must be valid at least until this sender becomes stopped.

From 0767de4b45d7efda5464397bea7d353e526974ad Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Tue, 3 Mar 2026 12:03:12 -0800
Subject: [PATCH 184/472] Roll chromium_revision 890f13186f..7d23b500bc
 (1593243:1593409)

Change log: https://chromium.googlesource.com/chromium/src/+log/890f13186f..7d23b500bc
Full diff: https://chromium.googlesource.com/chromium/src/+/890f13186f..7d23b500bc

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/bc0d205afa..ccf052a78f
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/b40bc07ded..2e881a3ab0
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/edc9633604..bbfbb6b660
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/82dd665e17..98739a948a
* src/third_party/breakpad/breakpad: https://chromium.googlesource.com/breakpad/breakpad.git/+log/9f26d631e4..8be0e31146
* src/third_party/kotlinc/current: 1wGbWZPhnUbDtQCvcw74lP1g0ceq55pEkvO5hZ3DRLcC..JEwTOoywRjDr5YoLIeG8Qzb3nT7OauqHdhfL6tIbyP0C
* src/third_party/llvm-build/Release+Asserts: Linux_x64/clang-llvmorg-23-init-4965-g686acf63-1.tar.xz,Linux_x64/clang-tidy-llvmorg-23-init-4965-g686acf63-1.tar.xz,Linux_x64/clangd-llvmorg-23-init-4965-g686acf63-1.tar.xz,Linux_x64/llvm-code-coverage-llvmorg-23-init-4965-g686acf63-1.tar.xz,Linux_x64/llvmobjdump-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac/clang-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac/clang-mac-runtime-library-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac/clang-tidy-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac/clangd-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac/llvm-code-coverage-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac/llvmobjdump-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac_arm64/clang-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac_arm64/clang-tidy-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac_arm64/clangd-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac_arm64/llvm-code-coverage-llvmorg-23-init-4965-g686acf63-1.tar.xz,Mac_arm64/llvmobjdump-llvmorg-23-init-4965-g686acf63-1.tar.xz,Win/clang-llvmorg-23-init-4965-g686acf63-1.tar.xz,Win/clang-tidy-llvmorg-23-init-4965-g686acf63-1.tar.xz,Win/clang-win-runtime-library-llvmorg-23-init-4965-g686acf63-1.tar.xz,Win/clangd-llvmorg-23-init-4965-g686acf63-1.tar.xz,Win/llvm-code-coverage-llvmorg-23-init-4965-g686acf63-1.tar.xz,Win/llvmobjdump-llvmorg-23-init-4965-g686acf63-1.tar.xz..Linux_x64/clang-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz,Linux_x64/clang-tidy-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz,Linux_x64/clangd-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz,Linux_x64/llvm-code-coverage-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz,Linux_x64/llvmobjdump-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz,Mac/clang-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz,Mac/clang-mac-runtime-library-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz,Mac/clang-tidy-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz,Mac/clangd-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz,Mac/llvm-code-coverage-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz,Mac/llvmobjdump-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz,Mac_arm64/clang-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz,Mac_arm64/clang-tidy-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz,Mac_arm64/clangd-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz,Mac_arm64/llvm-code-coverage-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz,Mac_arm64/llvmobjdump-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz,Win/clang-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz,Win/clang-tidy-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz,Win/clang-win-runtime-library-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz,Win/clangd-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz,Win/llvm-code-coverage-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz,Win/llvmobjdump-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/f884650a15..21c77783aa
* src/third_party/rust-toolchain: Linux_x64/rust-toolchain-c78a29473a68f07012904af11c92ecffa68fcc75-1-llvmorg-23-init-4965-g686acf63.tar.xz,Mac/rust-toolchain-c78a29473a68f07012904af11c92ecffa68fcc75-1-llvmorg-23-init-4965-g686acf63.tar.xz,Mac_arm64/rust-toolchain-c78a29473a68f07012904af11c92ecffa68fcc75-1-llvmorg-23-init-4965-g686acf63.tar.xz,Win/rust-toolchain-c78a29473a68f07012904af11c92ecffa68fcc75-1-llvmorg-23-init-4965-g686acf63.tar.xz..Linux_x64/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-1-llvmorg-23-init-5669-g8a0be0bc.tar.xz,Mac/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-1-llvmorg-23-init-5669-g8a0be0bc.tar.xz,Mac_arm64/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-1-llvmorg-23-init-5669-g8a0be0bc.tar.xz,Win/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-1-llvmorg-23-init-5669-g8a0be0bc.tar.xz
* src/third_party/tflite/src: https://chromium.googlesource.com/external/github.com/tensorflow/tensorflow.git/+log/204f897f9a..f920077085
* src/third_party/xnnpack/src: https://chromium.googlesource.com/external/github.com/google/XNNPACK.git/+log/066e3e72b0..26feacc3eb
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/a3ead8a269..2a795a769f
DEPS diff: https://chromium.googlesource.com/chromium/src/+/890f13186f..7d23b500bc/DEPS

BUG=None

Change-Id: Ie743d8b31cbb7e06890f804322c3d361a7e486f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453420
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47037}
---
 DEPS | 230 +++++++++++++++++++++++++++++------------------------------
 1 file changed, 115 insertions(+), 115 deletions(-)

diff --git a/DEPS b/DEPS
index 8451e1b5b2..af7f14d1a4 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '890f13186fc2d261eeb6cc3d78448dfc58cf03da',
+  'chromium_revision': '7d23b500bc25e5ee75029930c83faac0f23ce867',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@bc0d205afa3133b2ee54cef32d6d2b5693b5eca8',
+    'https://chromium.googlesource.com/chromium/src/build@ccf052a78f4ce8c71f12f9d7c9bb02ae41b21a9b',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@fef8d6947fa54202e8a00c58d5ac8fd15780bc33',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@b40bc07ded3eb0479ce5e9c1e470f3c9a921ae54',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@2e881a3ab00b1a3cdddd68c67a95c53a64d37eca',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@edc96336049d435f9b6d47df1a213a33b4ff6694',
+    'https://chromium.googlesource.com/chromium/src/testing@bbfbb6b660bcb592aee5795dd20e405c625ff6b9',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@82dd665e1743938f6e75cdcebdb7e84603a9c048',
+    'https://chromium.googlesource.com/chromium/src/third_party@98739a948af1ad0b0197a3a64453f844228114fc',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -136,157 +136,157 @@ deps = {
     'objects': [
       {
         # The Android libclang_rt.builtins libraries are currently only included in the Linux clang package.
-        'object_name': 'Linux_x64/clang-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': '6d4796631177f3c070923242601a4e14425ffdc2ee84c95ce630c0198d755637',
-        'size_bytes': 57997052,
-        'generation': 1771832273904291,
+        'object_name': 'Linux_x64/clang-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz',
+        'sha256sum': '750b331006635281d7d90696629f67db748ba62004c46675eccb8af144141847',
+        'size_bytes': 58029996,
+        'generation': 1772218390302503,
         'condition': '(host_os == "linux" or checkout_android) and non_git_source',
       },
       {
-        'object_name': 'Linux_x64/clang-tidy-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': '250f932ba1195eaa1ab8fd6e50f3c650ff3b74d28127d982ee79b96c4844bfb7',
-        'size_bytes': 14392496,
-        'generation': 1771832274267259,
+        'object_name': 'Linux_x64/clang-tidy-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz',
+        'sha256sum': 'd53439bb6ac13c8d2c30c20555ded434039802f70d4119c0138bd77d03552223',
+        'size_bytes': 14392856,
+        'generation': 1772218390323510,
         'condition': 'host_os == "linux" and checkout_clang_tidy and non_git_source',
       },
       {
-        'object_name': 'Linux_x64/clangd-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': '39258f20af9d459b8b032ee6bd38a79ae764fd950b79bace3227b8965d58d189',
-        'size_bytes': 14618644,
-        'generation': 1771832274470726,
+        'object_name': 'Linux_x64/clangd-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz',
+        'sha256sum': 'a24613fb7afce42c076bb95d1b671ac028746b379e88070c126f0aab17a4c34e',
+        'size_bytes': 14635272,
+        'generation': 1772218390330947,
         'condition': 'host_os == "linux" and checkout_clangd and non_git_source',
       },
       {
-        'object_name': 'Linux_x64/llvm-code-coverage-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': 'cf2fd97359cc982f56b0beb7e9abff5a4799a12ff8c3e22728ddf06e1053b867',
-        'size_bytes': 2330268,
-        'generation': 1771832275108482,
+        'object_name': 'Linux_x64/llvm-code-coverage-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz',
+        'sha256sum': '8dcd816a83361b7924093ccba92dfe6bd29af2cf8af58bf7ce785b38c5027a8b',
+        'size_bytes': 2328908,
+        'generation': 1772218390452408,
         'condition': 'host_os == "linux" and checkout_clang_coverage_tools and non_git_source',
       },
       {
-        'object_name': 'Linux_x64/llvmobjdump-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': '720a9cbb8745d4ac4c3f887bc1f004fed44cfccb851adebaa765a1517531c952',
-        'size_bytes': 5786008,
-        'generation': 1771832274626197,
+        'object_name': 'Linux_x64/llvmobjdump-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz',
+        'sha256sum': '0a15d6b8c2b774b0706618d2afa123b9c87af2ec12e74dc44346df4c4690b670',
+        'size_bytes': 5780116,
+        'generation': 1772218390340688,
         'condition': '((checkout_linux or checkout_mac or checkout_android) and host_os == "linux") and non_git_source',
       },
       {
-        'object_name': 'Mac/clang-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': '189a5bca90f83e2dc9671fef5f149386452c215a532e94cdc98cc13461411fb2',
-        'size_bytes': 54754892,
-        'generation': 1771832277222794,
+        'object_name': 'Mac/clang-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz',
+        'sha256sum': '2661847eb275079358ab186eaf7f85d6139d44c7413a731dfac7f5ed1ec34a01',
+        'size_bytes': 54827776,
+        'generation': 1772218392155773,
         'condition': 'host_os == "mac" and host_cpu == "x64"',
       },
       {
-        'object_name': 'Mac/clang-mac-runtime-library-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': '3a0177f1cf5f88f00e86006cb9be42f5d86a8dc461d03daf2919dee2c632f619',
-        'size_bytes': 1012728,
-        'generation': 1771832301108033,
+        'object_name': 'Mac/clang-mac-runtime-library-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz',
+        'sha256sum': '69918295c163ec5a20aede81d4100bbd41e01142d32e0555366bba05141f7bf2',
+        'size_bytes': 1010608,
+        'generation': 1772218399449599,
         'condition': 'checkout_mac and not host_os == "mac"',
       },
       {
-        'object_name': 'Mac/clang-tidy-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': 'c8a3f088c4d8bbb92eba2036bb68d1640c9c54ef7ec4686b4e2a5c49ff9b6bc3',
-        'size_bytes': 14282996,
-        'generation': 1771832277199632,
+        'object_name': 'Mac/clang-tidy-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz',
+        'sha256sum': 'b8013fe5d2410db4f365ec8779972415d1d0a08042a3a43f823a0da712108cff',
+        'size_bytes': 14280488,
+        'generation': 1772218392176137,
         'condition': 'host_os == "mac" and host_cpu == "x64" and checkout_clang_tidy',
       },
       {
-        'object_name': 'Mac/clangd-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': '4746a11e14fe96a00024a7b7d7b4ee1d61c8960171578efad263d1c54cfa6901',
-        'size_bytes': 15449448,
-        'generation': 1771832277304839,
+        'object_name': 'Mac/clangd-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz',
+        'sha256sum': '508098b26e74bd7f5cdcc40a2ed2db24e2bdde15e0f1c14ce94f685f991b3dd6',
+        'size_bytes': 15455912,
+        'generation': 1772218392186146,
         'condition': 'host_os == "mac" and host_cpu == "x64" and checkout_clangd',
       },
       {
-        'object_name': 'Mac/llvm-code-coverage-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': 'fbaa78a2d70a0872c2e0fd94cbe9f13c61b3077c6ba73f76a34413510a420ed0',
-        'size_bytes': 2373980,
-        'generation': 1771832277813116,
+        'object_name': 'Mac/llvm-code-coverage-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz',
+        'sha256sum': '46c33f13a68fc14005560c01a91215b5cab54c07e920a714264352e46af1350c',
+        'size_bytes': 2376304,
+        'generation': 1772218392292978,
         'condition': 'host_os == "mac" and host_cpu == "x64" and checkout_clang_coverage_tools',
       },
       {
-        'object_name': 'Mac/llvmobjdump-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': 'f091b4127cd59792c2f278535b1a30ae2e0953c6e896f31a5b5c7beff707e388',
-        'size_bytes': 5693600,
-        'generation': 1771832277787419,
+        'object_name': 'Mac/llvmobjdump-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz',
+        'sha256sum': '6a92e3f21b3a035f406313d24688bb1b312a9a0ec423ff808752b6638104aff3',
+        'size_bytes': 5699700,
+        'generation': 1772218392189830,
         'condition': 'host_os == "mac" and host_cpu == "x64"',
       },
       {
-        'object_name': 'Mac_arm64/clang-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': '4a091561d8ee6cfff59efffecbfc0122d233d23427e1cdbcee5080253bfc81f9',
-        'size_bytes': 45966912,
-        'generation': 1771832302845435,
+        'object_name': 'Mac_arm64/clang-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz',
+        'sha256sum': '909be0f896bcf140c710548ccda4673c0aea2480e28d10803c19b1689b36acd5',
+        'size_bytes': 45847044,
+        'generation': 1772218401088162,
         'condition': 'host_os == "mac" and host_cpu == "arm64"',
       },
       {
-        'object_name': 'Mac_arm64/clang-tidy-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': '9248646928a75768d682b484e3a83f481152b8edce5f71fa6e8052d9ae210398',
-        'size_bytes': 12473436,
-        'generation': 1771832303183995,
+        'object_name': 'Mac_arm64/clang-tidy-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz',
+        'sha256sum': '83dc8d90529730ae503e684ea0047a0baec2b0c4a81941d1bb4196feea6ba264',
+        'size_bytes': 12444972,
+        'generation': 1772218401143017,
         'condition': 'host_os == "mac" and host_cpu == "arm64" and checkout_clang_tidy',
       },
       {
-        'object_name': 'Mac_arm64/clangd-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': 'e5fde137e7372fedc8e2c4a8e4f9fe718c3bcc63c76acaa0c4a859b04e3f4762',
-        'size_bytes': 12882364,
-        'generation': 1771832303644261,
+        'object_name': 'Mac_arm64/clangd-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz',
+        'sha256sum': '3b7ff06ccd41b0a1fb165e182a35bcd74ae49172f1720cd276eb5feac0e3dd9f',
+        'size_bytes': 12816980,
+        'generation': 1772218401144631,
         'condition': 'host_os == "mac" and host_cpu == "arm64" and checkout_clangd',
       },
       {
-        'object_name': 'Mac_arm64/llvm-code-coverage-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': 'd4e9d19346ef021be42a27b367efcdcfee03cc19bbceddb9ea4f92619ef8b06c',
-        'size_bytes': 1993468,
-        'generation': 1771832304157004,
+        'object_name': 'Mac_arm64/llvm-code-coverage-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz',
+        'sha256sum': '67148555d00427a3eaa8aeefb8c4c4e1271d585315bdbf0d28d20fd78957e309',
+        'size_bytes': 1988008,
+        'generation': 1772218401224240,
         'condition': 'host_os == "mac" and host_cpu == "arm64" and checkout_clang_coverage_tools',
       },
       {
-        'object_name': 'Mac_arm64/llvmobjdump-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': 'b01bce55530dbca2d6d25fa14ddbac8259184fdca1cc63365130953b9ebb517f',
-        'size_bytes': 5442884,
-        'generation': 1771832303592136,
+        'object_name': 'Mac_arm64/llvmobjdump-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz',
+        'sha256sum': 'a31075e7f46ed77c62ecec424722bec8335ef306a4701660f19b713229c49afa',
+        'size_bytes': 5421552,
+        'generation': 1772218401116635,
         'condition': 'host_os == "mac" and host_cpu == "arm64"',
       },
       {
-        'object_name': 'Win/clang-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': '84c13e593b1532c0cf0b8fe0928482e6eea96408846a679ce68e53c90911c33e',
-        'size_bytes': 49494200,
-        'generation': 1771832332047169,
+        'object_name': 'Win/clang-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz',
+        'sha256sum': 'f2c9d2a8accf7ed2e3c19b3f67fb94e60365411a536fb9d71391dd2d4e7e14bb',
+        'size_bytes': 49546756,
+        'generation': 1772218410442709,
         'condition': 'host_os == "win"',
       },
       {
-        'object_name': 'Win/clang-tidy-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': '2a5072bfa4a3eb9dd60321d955d6d4744b7b9f2f657f32dbc706774282787e9a',
-        'size_bytes': 14455216,
-        'generation': 1771832332462256,
+        'object_name': 'Win/clang-tidy-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz',
+        'sha256sum': '99e00bbb404557db32df4e7a183ac520c526fe0e143ca380dfb2d0c33a2025b5',
+        'size_bytes': 14462056,
+        'generation': 1772218410470169,
         'condition': 'host_os == "win" and checkout_clang_tidy',
       },
       {
-        'object_name': 'Win/clang-win-runtime-library-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': 'a9ddbb29b61eaac9f777b1fb532eebb49ab9e546bcd680129fcc7a2fd648cd27',
-        'size_bytes': 2596776,
-        'generation': 1771832355675158,
+        'object_name': 'Win/clang-win-runtime-library-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz',
+        'sha256sum': '62e9c022223e0fa6ff855c25dcee524818f04c570127ed7e74895b320a10100a',
+        'size_bytes': 2597584,
+        'generation': 1772218417651221,
         'condition': 'checkout_win and not host_os == "win"',
       },
       {
-        'object_name': 'Win/clangd-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': '5dad22f5da22e268e43476a28689ec7b9fb8d41ae55e53911a26e9c25a9c4c4b',
-        'size_bytes': 14884728,
-        'generation': 1771832332829491,
+        'object_name': 'Win/clangd-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz',
+        'sha256sum': '6a3ab3afb8d2e7f4a04eecd8073993586665ede3929308a0fa0119d9382b1e2d',
+        'size_bytes': 14887416,
+        'generation': 1772218410483998,
         'condition': 'host_os == "win" and checkout_clangd',
       },
       {
-        'object_name': 'Win/llvm-code-coverage-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': 'f5a46dbdf979e31d35dd12db37d714bc06a36957883f3c11d727d56382465079',
-        'size_bytes': 2478920,
-        'generation': 1771832333093363,
+        'object_name': 'Win/llvm-code-coverage-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz',
+        'sha256sum': '4bd610d2fbcc6e2bd8fd2df8d8c23a915373f8c987701d295314e8b33d457075',
+        'size_bytes': 2479300,
+        'generation': 1772218410570017,
         'condition': 'host_os == "win" and checkout_clang_coverage_tools',
       },
       {
-        'object_name': 'Win/llvmobjdump-llvmorg-23-init-4965-g686acf63-1.tar.xz',
-        'sha256sum': '0519107fdef9c730bb4f073c87dbd03995fd2da074aabef86347f62fb2b784f3',
-        'size_bytes': 5838628,
-        'generation': 1771832332695387,
+        'object_name': 'Win/llvmobjdump-llvmorg-23-init-5669-g8a0be0bc-1.tar.xz',
+        'sha256sum': '2ee77b6240b76353840439b38e7009d9f1fb8e97930dbbef3b1ff805ee981c5f',
+        'size_bytes': 5846184,
+        'generation': 1772218410487302,
         'condition': '(checkout_linux or checkout_mac or checkout_android) and host_os == "win"',
       },
     ]
@@ -298,31 +298,31 @@ deps = {
     'bucket': 'chromium-browser-clang',
     'objects': [
       {
-        'object_name': 'Linux_x64/rust-toolchain-c78a29473a68f07012904af11c92ecffa68fcc75-1-llvmorg-23-init-4965-g686acf63.tar.xz',
-        'sha256sum': 'e3f380e162218de33aeea559dd7696431fd505b47b9669422faea83b47b16de4',
-        'size_bytes': 270144912,
-        'generation': 1771832265918901,
+        'object_name': 'Linux_x64/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-1-llvmorg-23-init-5669-g8a0be0bc.tar.xz',
+        'sha256sum': 'ced6078ebd927bc45de379e645b0f0a8ad61ef751c06eb107ac4bc083daab9b4',
+        'size_bytes': 270674956,
+        'generation': 1772218383559628,
         'condition': 'host_os == "linux" and non_git_source',
       },
       {
-        'object_name': 'Mac/rust-toolchain-c78a29473a68f07012904af11c92ecffa68fcc75-1-llvmorg-23-init-4965-g686acf63.tar.xz',
-        'sha256sum': '01a4c3b5a3be52b5ae2a99e78541c512cd198b1a87baa1bc1c0b4697428d8fb2',
-        'size_bytes': 257817220,
-        'generation': 1771832267985133,
+        'object_name': 'Mac/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-1-llvmorg-23-init-5669-g8a0be0bc.tar.xz',
+        'sha256sum': 'ec05e6e8d011c8f576df228b7c09d169f6eb8627b0241e17e779b0fd1ee9a869',
+        'size_bytes': 258393452,
+        'generation': 1772218385203632,
         'condition': 'host_os == "mac" and host_cpu == "x64"',
       },
       {
-        'object_name': 'Mac_arm64/rust-toolchain-c78a29473a68f07012904af11c92ecffa68fcc75-1-llvmorg-23-init-4965-g686acf63.tar.xz',
-        'sha256sum': 'e26455a6b906d08f848c1569a3c4d9b41d0c57c65f069e28a285409f6f8aca30',
-        'size_bytes': 241326948,
-        'generation': 1771832269754008,
+        'object_name': 'Mac_arm64/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-1-llvmorg-23-init-5669-g8a0be0bc.tar.xz',
+        'sha256sum': '6d7cde3b7c7a5286a87f9a60aa8c9f0928555a39c8b71b9d039cbf12a805fe2b',
+        'size_bytes': 241760428,
+        'generation': 1772218386869177,
         'condition': 'host_os == "mac" and host_cpu == "arm64"',
       },
       {
-        'object_name': 'Win/rust-toolchain-c78a29473a68f07012904af11c92ecffa68fcc75-1-llvmorg-23-init-4965-g686acf63.tar.xz',
-        'sha256sum': '50f23d9770fc930d2d7fd8eae8b1ff21573fe7d83f71e014a7e674dc8884cdc5',
-        'size_bytes': 408636940,
-        'generation': 1771832271552765,
+        'object_name': 'Win/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-1-llvmorg-23-init-5669-g8a0be0bc.tar.xz',
+        'sha256sum': '8146753e3451a66e1d4dc9e73af29305b5ea2107fae88739880ddf7a7635ffb6',
+        'size_bytes': 409426524,
+        'generation': 1772218388518369,
         'condition': 'host_os == "win"',
       },
     ],
@@ -490,7 +490,7 @@ deps = {
   'src/third_party/boringssl/src':
     'https://boringssl.googlesource.com/boringssl.git@d0fcf49574b06de7090cba4ad8124df638a0134e',
   'src/third_party/breakpad/breakpad':
-    'https://chromium.googlesource.com/breakpad/breakpad.git@9f26d631e4063c0af3e8ee3bc8826486115372a8',
+    'https://chromium.googlesource.com/breakpad/breakpad.git@8be0e3114685fcc1589561067282edf75ea1259a',
   'src/third_party/catapult':
     'https://chromium.googlesource.com/catapult.git@109c6d7babfc9afbb19af677a0d21fe3a281f2c1',
   'src/third_party/ced/src': {
@@ -584,7 +584,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/kotlinc',
-              'version': '1wGbWZPhnUbDtQCvcw74lP1g0ceq55pEkvO5hZ3DRLcC',
+              'version': 'JEwTOoywRjDr5YoLIeG8Qzb3nT7OauqHdhfL6tIbyP0C',
           },
       ],
       'condition': 'checkout_android',
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'f884650a1523d2e48fbc98980d9a5a4a56d883e2',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '21c77783aa90ba9b46e79949c797ca6bc516c124',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@a3ead8a269e96840406ab5aa34a527d6f0d8dcfb',
+    'https://chromium.googlesource.com/chromium/src/tools@2a795a769f4727be367a4f704489cd2f7f5ec576',
 
   'src/third_party/espresso': {
       'packages': [
@@ -786,7 +786,7 @@ deps = {
   },
 
   'src/third_party/tflite/src':
-    Var('chromium_git') + '/external/github.com/tensorflow/tensorflow.git' + '@' + '204f897f9a511e62242fcbaac6be5fbbe9e2e245',
+    Var('chromium_git') + '/external/github.com/tensorflow/tensorflow.git' + '@' + 'f9200770858f5721b7fa95db7aa3e4cd712fe9df',
 
   'src/third_party/turbine/cipd': {
       'packages': [
@@ -852,7 +852,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/google/pthreadpool.git' + '@' + '9003ee6c137cea3b94161bd5c614fb43be523ee1',
 
   'src/third_party/xnnpack/src':
-    Var('chromium_git') + '/external/github.com/google/XNNPACK.git' + '@' + '066e3e72b05e56441aebaa57c57fad9280d869c8',
+    Var('chromium_git') + '/external/github.com/google/XNNPACK.git' + '@' + '26feacc3ebe30d1f3a82e1ff3938ef9859d74640',
 
   'src/third_party/farmhash/src':
     Var('chromium_git') + '/external/github.com/google/farmhash.git' + '@' + '816a4ae622e964763ca0862d9dbd19324a1eaf45',

From 2f069781037ac1f9965d39152c6c97ea98ce6aff Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Tue, 3 Mar 2026 12:55:09 +0000
Subject: [PATCH 185/472] Revert^2 "Default enable
 WebRTC-HeaderExtensionNegotiateMemory"

This reverts commit 74d988fd57f8f1b6722d8ac6f49c686b9f2bc196.

Reland "Default enable WebRTC-HeaderExtensionNegotiateMemory"

This reverts commit 74d988fd57f8f1b6722d8ac6f49c686b9f2bc196.

Reason for revert: Added code to skip stopped transceivers

Original change's description:
> Revert "Default enable WebRTC-HeaderExtensionNegotiateMemory"
>
> This reverts commit 4ecf297a4c0af68ff8af8c3a8e6b0832bddd14d5.
>
> Reason for revert: Impacted unexpected application usages
>
> Original change's description:
> > Default enable WebRTC-HeaderExtensionNegotiateMemory
> >
> > Spec change: https://github.com/w3c/webrtc-extensions/pull/238
> >
> > Chromestatus entry for change: https://chromestatus.com/feature/5135528638939136
> >
> > Bug: webrtc:439514253
> > Change-Id: I296f2f06dd69b4e30be419b44c926acfd2722e2d
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/407821
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#45993}
>
> Bug: webrtc:439514253, chromium:467164231
> Change-Id: I95dbc8cbaf1b769ffaea291801684bc26e4b9ccb
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/433102
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#46456}

Bug: webrtc:439514253, chromium:467164231
Change-Id: I2df85ba25016374a52abb13037cdca74fb1a395d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453240
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47038}
---
 pc/rtp_transceiver.cc          | 6 +++---
 pc/rtp_transmission_manager.cc | 3 ++-
 2 files changed, 5 insertions(+), 4 deletions(-)

diff --git a/pc/rtp_transceiver.cc b/pc/rtp_transceiver.cc
index 77d99a14e4..0ccad43623 100644
--- a/pc/rtp_transceiver.cc
+++ b/pc/rtp_transceiver.cc
@@ -1219,19 +1219,19 @@ void RtpTransceiver::OnNegotiationUpdate(
   RTC_DCHECK(content);
   if (sdp_type == SdpType::kAnswer || sdp_type == SdpType::kPrAnswer) {
     negotiated_header_extensions_ = content->rtp_header_extensions();
-    if (env_.field_trials().IsEnabled(
+    if (!env_.field_trials().IsDisabled(
             "WebRTC-HeaderExtensionNegotiateMemory")) {
       header_extensions_to_negotiate_ = GetNegotiatedHeaderExtensions();
     }
   } else if (sdp_type == SdpType::kOffer) {
-    if (env_.field_trials().IsEnabled(
+    if (!env_.field_trials().IsDisabled(
             "WebRTC-HeaderExtensionNegotiateMemory")) {
       header_extensions_for_rollback_ = header_extensions_to_negotiate_;
       header_extensions_to_negotiate_ =
           GetOfferedAndImplementedHeaderExtensions(content);
     }
   } else if (sdp_type == SdpType::kRollback) {
-    if (env_.field_trials().IsEnabled(
+    if (!env_.field_trials().IsDisabled(
             "WebRTC-HeaderExtensionNegotiateMemory")) {
       RTC_CHECK(!header_extensions_for_rollback_.empty());
       header_extensions_to_negotiate_ = header_extensions_for_rollback_;
diff --git a/pc/rtp_transmission_manager.cc b/pc/rtp_transmission_manager.cc
index dc6e4bc6f4..b67a3fe9e7 100644
--- a/pc/rtp_transmission_manager.cc
+++ b/pc/rtp_transmission_manager.cc
@@ -284,7 +284,8 @@ RtpTransmissionManager::CreateAndAddTransceiver(
   RTC_DCHECK(!FindSenderById(sender_id));
   std::vector<RtpHeaderExtensionCapability> header_extensions =
       std::move(header_extensions_to_negotiate);
-  if (env_.field_trials().IsEnabled("WebRTC-HeaderExtensionNegotiateMemory")) {
+  if (!env_.field_trials().IsDisabled(
+          "WebRTC-HeaderExtensionNegotiateMemory")) {
     // If we have already negotiated header extensions for this type,
     // and it is not stopped,
     // reuse the negotiated state for new transceivers of the same type.

From 9ccc4d0d73b99acbc17aa381efe1f74d356bea0d Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Tue, 3 Mar 2026 16:04:01 -0800
Subject: [PATCH 186/472] Roll chromium_revision 7d23b500bc..dfcdfec090
 (1593409:1593592)

Change log: https://chromium.googlesource.com/chromium/src/+log/7d23b500bc..dfcdfec090
Full diff: https://chromium.googlesource.com/chromium/src/+/7d23b500bc..dfcdfec090

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/ccf052a78f..3f9eea60de
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/2e881a3ab0..c440dd7d83
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/bbfbb6b660..28af9063ca
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/98739a948a..e5e1c9982f
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/d0fcf49574..16d1a81e47
* src/third_party/libvpx/source/libvpx: https://chromium.googlesource.com/webm/libvpx.git/+log/4fcebeabe5..9a2d3d1f46
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/21c77783aa..05b8b9cf93
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/2a795a769f..3ec1854646
DEPS diff: https://chromium.googlesource.com/chromium/src/+/7d23b500bc..dfcdfec090/DEPS

BUG=None

Change-Id: I5823f479ee84a718d51fa717d64dc8f441d36c1c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453460
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47039}
---
 DEPS | 18 +++++++++---------
 1 file changed, 9 insertions(+), 9 deletions(-)

diff --git a/DEPS b/DEPS
index af7f14d1a4..71945968aa 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '7d23b500bc25e5ee75029930c83faac0f23ce867',
+  'chromium_revision': 'dfcdfec090dc9c8551eaa68a13eddd8b8882c8c5',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@ccf052a78f4ce8c71f12f9d7c9bb02ae41b21a9b',
+    'https://chromium.googlesource.com/chromium/src/build@3f9eea60de6e62a108714cbc1fa99d29e7642fa5',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@fef8d6947fa54202e8a00c58d5ac8fd15780bc33',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@2e881a3ab00b1a3cdddd68c67a95c53a64d37eca',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@c440dd7d837ff45a1189fd1bf8aca67bc692971e',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@bbfbb6b660bcb592aee5795dd20e405c625ff6b9',
+    'https://chromium.googlesource.com/chromium/src/testing@28af9063ca04139abd9fbd6381862e64910167d3',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@98739a948af1ad0b0197a3a64453f844228114fc',
+    'https://chromium.googlesource.com/chromium/src/third_party@e5e1c9982f4b46c5b2404f7c4e779c968e725af0',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -488,7 +488,7 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@d0fcf49574b06de7090cba4ad8124df638a0134e',
+    'https://boringssl.googlesource.com/boringssl.git@16d1a81e475be0b3f859eb2264642bc1b6501f6e',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@8be0e3114685fcc1589561067282edf75ea1259a',
   'src/third_party/catapult':
@@ -609,11 +609,11 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '21c77783aa90ba9b46e79949c797ca6bc516c124',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '05b8b9cf93e16e960e71f38bf2e41944b4611f22',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
-    'https://chromium.googlesource.com/webm/libvpx.git@4fcebeabe58e79255d291acb4cead4ed7953149e',
+    'https://chromium.googlesource.com/webm/libvpx.git@9a2d3d1f46afbdfa9b9820a9fd3aacb084e65e2f',
   'src/third_party/libyuv':
     'https://chromium.googlesource.com/libyuv/libyuv.git@6067afde563c3946eebd94f146b3824ab7a97a9c',
   'src/third_party/lss': {
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@2a795a769f4727be367a4f704489cd2f7f5ec576',
+    'https://chromium.googlesource.com/chromium/src/tools@3ec1854646a040c58fe19f4dc28c5bd39b52e130',
 
   'src/third_party/espresso': {
       'packages': [

From f95a37a99bd16d4c8fea189c0be411c89ffc3645 Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Tue, 3 Mar 2026 20:04:11 -0800
Subject: [PATCH 187/472] Update WebRTC code version (2026-03-04T04:04:09).

Bug: None
Change-Id: Iccf18ddfc0d485d2466e7def60736659998c3ad0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453500
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47040}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index e9f968dd36..f5540b9e92 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-01T04:04:55";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-04T04:04:09";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From 3e52afec17d0b0dc616f1c62c59c60b214ac545d Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Tue, 3 Mar 2026 22:03:59 -0800
Subject: [PATCH 188/472] Roll chromium_revision dfcdfec090..04f26ef6ec
 (1593592:1593719)

Change log: https://chromium.googlesource.com/chromium/src/+log/dfcdfec090..04f26ef6ec
Full diff: https://chromium.googlesource.com/chromium/src/+/dfcdfec090..04f26ef6ec

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/3f9eea60de..80d98a1ed9
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/c440dd7d83..f3dfca4f79
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/28af9063ca..6a8f14f59f
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/e5e1c9982f..a3bc0f1a84
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/16d1a81e47..fbf01ffae1
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/d6aae1009f..27f8f7c421
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/661cdb227f..b0ebf966a5
* src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/dacaf47cea..e8faec51e3
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/3ec1854646..0d31bdf20c
DEPS diff: https://chromium.googlesource.com/chromium/src/+/dfcdfec090..04f26ef6ec/DEPS

BUG=None

Change-Id: I4010f93315c096cf0238a46c4b0f87f8f30df570
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453540
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47041}
---
 DEPS | 20 ++++++++++----------
 1 file changed, 10 insertions(+), 10 deletions(-)

diff --git a/DEPS b/DEPS
index 71945968aa..2e74a57e69 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': 'dfcdfec090dc9c8551eaa68a13eddd8b8882c8c5',
+  'chromium_revision': '04f26ef6ecf70adc1a82bb46e041f494a7f392c5',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@3f9eea60de6e62a108714cbc1fa99d29e7642fa5',
+    'https://chromium.googlesource.com/chromium/src/build@80d98a1ed98c47c52033553b5a10a5050291c2bc',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@fef8d6947fa54202e8a00c58d5ac8fd15780bc33',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@c440dd7d837ff45a1189fd1bf8aca67bc692971e',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@f3dfca4f794abc1f048f43f161b6e961ed70050f',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@28af9063ca04139abd9fbd6381862e64910167d3',
+    'https://chromium.googlesource.com/chromium/src/testing@6a8f14f59fd1e116cac6c7c201677f39a7483925',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@e5e1c9982f4b46c5b2404f7c4e779c968e725af0',
+    'https://chromium.googlesource.com/chromium/src/third_party@a3bc0f1a8407981ffb95feef7b09e8ecd10b99e8',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -337,7 +337,7 @@ deps = {
   'src/third_party/libc++abi/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@8f11bb1d4438d0239d0dfc1bd9456a9f31629dda',
   'src/third_party/llvm-libc/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@dacaf47ceaeddc37b35b9a65eb2ba817bb110069',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@e8faec51e35d65209d5ba660d711893ad3790735',
   'src/third_party/libunwind/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@17ccf7d110c5526cb77e93cfd8330f491fb2bf18',
 
@@ -488,7 +488,7 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@16d1a81e475be0b3f859eb2264642bc1b6501f6e',
+    'https://boringssl.googlesource.com/boringssl.git@fbf01ffae10929d863b6744bb505f0cae6c9c712',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@8be0e3114685fcc1589561067282edf75ea1259a',
   'src/third_party/catapult':
@@ -505,7 +505,7 @@ deps = {
   'src/third_party/crc32c/src':
     'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558',
   'src/third_party/depot_tools':
-    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@d6aae1009f694a7292d28a5db9a3f7bcb15bad4d',
+    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@27f8f7c4211120fabd8b6c186b13a77732551ae4',
   'src/third_party/ffmpeg':
     'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@946d97db8d906277085e361892b7efda5152e2f1',
   'src/third_party/flatbuffers/src':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@3ec1854646a040c58fe19f4dc28c5bd39b52e130',
+    'https://chromium.googlesource.com/chromium/src/tools@0d31bdf20c5d30ddeb83f0645ee74648f0356673',
 
   'src/third_party/espresso': {
       'packages': [
@@ -864,7 +864,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7364b490b5f78d58efe23ea76e74210fd6c3c76f',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '661cdb227f2adc2a649b48518bceb88ea4126e3a',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + 'b0ebf966a58d9169739cb12f3a6e5e7a908b2bfc',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From 6f9db9ffc646b2c74a7b6684db4e81dddb1f684a Mon Sep 17 00:00:00 2001
From: Jeremy Leconte <jleconte@google.com>
Date: Tue, 3 Mar 2026 11:14:53 -0800
Subject: [PATCH 189/472] Revert "Disable android_chromium_compile from CQ."

This reverts commit 7ec1333a0b9fa55178c3fa7835a8924bc826f8b5.

Reason for revert: issue seems to be fixed

Original change's description:
> Disable android_chromium_compile from CQ.
>
> Change-Id: Iaa9095b5a7f2cce02a59590a29e13cb0c06fa620
> Bug: None
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453261
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Auto-Submit: Jeremy Leconte <jleconte@google.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#47030}

Bug: None
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I090fa2a5cb339228d48edd73280f1f619ebbc3f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453400
Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
Auto-Submit: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#47042}
---
 infra/config/builders.star                   | 4 +---
 infra/config/generated/luci/commit-queue.cfg | 3 +++
 2 files changed, 4 insertions(+), 3 deletions(-)

diff --git a/infra/config/builders.star b/infra/config/builders.star
index 9e9a92fac0..8816ef06fa 100644
--- a/infra/config/builders.star
+++ b/infra/config/builders.star
@@ -332,9 +332,7 @@ ci_builder("Android32 Builder x86", "Android|x86|rel")
 try_builder("android_compile_x86_rel")
 ci_builder("Android32 (more configs)", "Android|arm|more")
 try_builder("android_arm_more_configs")
-
-# TODO(jleconte): re-enable when the issue is fixed in Chromium.
-chromium_try_builder("android_chromium_compile", cq = None)
+chromium_try_builder("android_chromium_compile")
 
 ios_builder("iOS64 Debug", "iOS|arm64|dbg")
 ios_try_job("ios_compile_arm64_dbg")
diff --git a/infra/config/generated/luci/commit-queue.cfg b/infra/config/generated/luci/commit-queue.cfg
index 48d7850133..b7c3a0c960 100644
--- a/infra/config/generated/luci/commit-queue.cfg
+++ b/infra/config/generated/luci/commit-queue.cfg
@@ -56,6 +56,9 @@ config_groups {
       builders {
         name: "webrtc/try/android_arm_rel"
       }
+      builders {
+        name: "webrtc/try/android_chromium_compile"
+      }
       builders {
         name: "webrtc/try/android_compile_arm64_rel"
       }

From 7de2447ebdf115f1437bfd5df87cc8362cd31860 Mon Sep 17 00:00:00 2001
From: Yuval Atia <yuval@island.io>
Date: Wed, 18 Feb 2026 13:06:25 +0200
Subject: [PATCH 190/472] [WGC] don't leak resources when capture session ends

To free resources related with the capture session, users must `Close`
the frame pool, otherwise WGC's internal worker thread never exits, and
references to some WinRT objects are never released and so those objects
also leak.

Also, construction of the frame pool may fail, and since it is
constructed after `WgcCaptureSession` has been constructed (as part of
`WgcCaptureSession::StartCapture`, the destructor will be invoked and
call `WgcCaptureSession::RemoveFrameArrivedEventHandler`, which will
`DCHECK` in debug builds and crash. In release builds this isn't an
issue since `frame_arrived_token_` being not null also guarantees that
`frame_pool_` is not null.

Bug: chromium:482343989
Change-Id: I790b2f4fd1399bbf9b4e9fd0e93977103efabab5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/447560
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47043}
---
 AUTHORS                                         |  1 +
 .../desktop_capture/win/wgc_capture_session.cc  | 17 ++++++++++++++++-
 2 files changed, 17 insertions(+), 1 deletion(-)

diff --git a/AUTHORS b/AUTHORS
index 6d700c9c80..e9f6f9b90c 100644
--- a/AUTHORS
+++ b/AUTHORS
@@ -189,6 +189,7 @@ Hopin Ltd. <*@hopin.to>
 HyperConnect Inc. <*@hpcnt.com>
 Igalia S.L. <*@igalia.com>
 Intel Corporation <*@intel.com>
+Island Technology, Inc. <*@island.io>
 LG Electronics, Inc. <*@lge.com>
 Life On Air Inc. <*@lifeonair.com>
 LiveKit <*@livekit.io>
diff --git a/modules/desktop_capture/win/wgc_capture_session.cc b/modules/desktop_capture/win/wgc_capture_session.cc
index 3969654bce..7d02d41ce1 100644
--- a/modules/desktop_capture/win/wgc_capture_session.cc
+++ b/modules/desktop_capture/win/wgc_capture_session.cc
@@ -39,6 +39,7 @@
 
 using Microsoft::WRL::ComPtr;
 namespace WGC = ABI::Windows::Graphics::Capture;
+using IClosable = ABI::Windows::Foundation::IClosable;
 
 namespace webrtc {
 namespace {
@@ -166,6 +167,18 @@ WgcCaptureSession::WgcCaptureSession(intptr_t source_id,
 
 WgcCaptureSession::~WgcCaptureSession() {
   RemoveEventHandlers();
+  if (frame_pool_) {
+    ComPtr<IClosable> closable;
+    HRESULT hr = frame_pool_.As(&closable);
+    if (FAILED(hr)) {
+      RTC_LOG(LS_WARNING) << "Failed to query frame pool as IClosable: " << hr;
+      return;
+    }
+    hr = closable->Close();
+    if (FAILED(hr)) {
+      RTC_LOG(LS_WARNING) << "Failed to close frame pool: " << hr;
+    }
+  }
 }
 
 HRESULT WgcCaptureSession::StartCapture(const DesktopCaptureOptions& options) {
@@ -748,7 +761,9 @@ HRESULT WgcCaptureSession::OnItemClosed(WGC::IGraphicsCaptureItem* sender,
 
 void WgcCaptureSession::RemoveEventHandlers() {
   RemoveItemClosedEventHandler();
-  RemoveFrameArrivedEventHandler();
+  if (frame_pool_) {
+    RemoveFrameArrivedEventHandler();
+  }
 }
 
 void WgcCaptureSession::RemoveItemClosedEventHandler() {

From 797288367b89303c313b97e7c4217262334798ed Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Mon, 2 Mar 2026 16:15:04 +0100
Subject: [PATCH 191/472] Remove usage of TimeMicros in FrameSelector
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Bug: webrtc:42223992
Change-Id: I001bfbc8dc2f673c53231a0cdf8f14485ac6cdcd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453040
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47044}
---
 video/corruption_detection/BUILD.gn           |  5 ++--
 .../frame_instrumentation_generator_impl.cc   |  2 +-
 video/corruption_detection/frame_selector.cc  |  7 +++--
 video/corruption_detection/frame_selector.h   |  4 ++-
 .../frame_selector_unittest.cc                | 29 ++++++++++---------
 5 files changed, 25 insertions(+), 22 deletions(-)

diff --git a/video/corruption_detection/BUILD.gn b/video/corruption_detection/BUILD.gn
index 97bc769994..53596d6317 100644
--- a/video/corruption_detection/BUILD.gn
+++ b/video/corruption_detection/BUILD.gn
@@ -95,19 +95,17 @@ rtc_library("frame_selector") {
     "frame_selector.h",
   ]
   deps = [
+    "../../api/environment",
     "../../api/units:time_delta",
     "../../api/units:timestamp",
     "../../api/video:encoded_image",
     "../../api/video:video_frame",
-    "../../api/video:video_frame_type",
     "../../api/video_codecs:scalability_mode",
     "../../api/video_codecs:video_codecs_api",
     "../../modules/rtp_rtcp:rtp_rtcp_format",
     "../../modules/video_coding/svc:scalability_mode_util",
     "../../rtc_base:checks",
-    "../../rtc_base:logging",
     "../../rtc_base:random",
-    "../../rtc_base:timeutils",
   ]
 }
 
@@ -239,6 +237,7 @@ if (rtc_include_tests) {
       "../../api/video:video_frame_type",
       "../../api/video:video_rtp_headers",
       "../../api/video_codecs:scalability_mode",
+      "../../test:create_test_environment",
       "../../test:test_support",
     ]
   }
diff --git a/video/corruption_detection/frame_instrumentation_generator_impl.cc b/video/corruption_detection/frame_instrumentation_generator_impl.cc
index e1c9f22755..40bbfcac24 100644
--- a/video/corruption_detection/frame_instrumentation_generator_impl.cc
+++ b/video/corruption_detection/frame_instrumentation_generator_impl.cc
@@ -92,7 +92,7 @@ std::unique_ptr<FrameSelector> CreateFrameSelector(
     return nullptr;
   }
   return std::make_unique<FrameSelector>(
-      scalability_mode.value_or(ScalabilityMode::kL1T1),
+      *environment, scalability_mode.value_or(ScalabilityMode::kL1T1),
       FrameSelector::Timespan{
           .lower_bound = settings.low_overhead_lower_bound(),
           .upper_bound = settings.low_overhead_upper_bound()},
diff --git a/video/corruption_detection/frame_selector.cc b/video/corruption_detection/frame_selector.cc
index 6cad99708c..eceab9dae1 100644
--- a/video/corruption_detection/frame_selector.cc
+++ b/video/corruption_detection/frame_selector.cc
@@ -13,6 +13,7 @@
 #include <algorithm>
 #include <cstdint>
 
+#include "api/environment/environment.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
 #include "api/video/encoded_image.h"
@@ -23,7 +24,6 @@
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/video_coding/svc/scalability_mode_util.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/time_utils.h"
 
 namespace webrtc {
 
@@ -42,14 +42,15 @@ bool CanNativelyHandleFormat(VideoFrameBuffer::Type type) {
 
 }  // namespace
 
-FrameSelector::FrameSelector(ScalabilityMode scalability_mode,
+FrameSelector::FrameSelector(const Environment& env,
+                             ScalabilityMode scalability_mode,
                              Timespan low_overhead_frame_span,
                              Timespan high_overhead_frame_span)
     : inter_layer_pred_mode_(
           ScalabilityModeToInterLayerPredMode(scalability_mode)),
       low_overhead_frame_span_(low_overhead_frame_span),
       high_overhead_frame_span_(high_overhead_frame_span),
-      random_(TimeMicros()) {
+      random_(env.clock().TimeInMicroseconds()) {
   RTC_DCHECK_GE(low_overhead_frame_span.upper_bound,
                 low_overhead_frame_span.lower_bound);
   RTC_DCHECK_GE(high_overhead_frame_span.upper_bound,
diff --git a/video/corruption_detection/frame_selector.h b/video/corruption_detection/frame_selector.h
index bc31cdf138..92107d2e1d 100644
--- a/video/corruption_detection/frame_selector.h
+++ b/video/corruption_detection/frame_selector.h
@@ -13,6 +13,7 @@
 
 #include <map>
 
+#include "api/environment/environment.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
 #include "api/video/encoded_image.h"
@@ -46,7 +47,8 @@ class FrameSelector {
     TimeDelta upper_bound;
   };
 
-  FrameSelector(ScalabilityMode scalability_mode,
+  FrameSelector(const Environment& env,
+                ScalabilityMode scalability_mode,
                 Timespan low_overhead_frame_span,
                 Timespan high_overhead_frame_span);
 
diff --git a/video/corruption_detection/frame_selector_unittest.cc b/video/corruption_detection/frame_selector_unittest.cc
index 29cbc5c562..265c25aa27 100644
--- a/video/corruption_detection/frame_selector_unittest.cc
+++ b/video/corruption_detection/frame_selector_unittest.cc
@@ -24,6 +24,7 @@
 #include "api/video/video_frame_type.h"
 #include "api/video/video_rotation.h"
 #include "api/video_codecs/scalability_mode.h"
+#include "test/create_test_environment.h"
 #include "test/gtest.h"
 
 namespace webrtc {
@@ -85,8 +86,8 @@ EncodedImage CreateEncodedImage(VideoFrameType frame_type,
 }
 
 TEST(FrameSelectorTest, AlwaysSelectsKeyFrames) {
-  FrameSelector selector(ScalabilityMode::kL1T1, kLowOverheadSpan,
-                         kHighOverheadSpan);
+  FrameSelector selector(CreateTestEnvironment(), ScalabilityMode::kL1T1,
+                         kLowOverheadSpan, kHighOverheadSpan);
   // Even before any threshold, keyframes should be selected.
   EXPECT_TRUE(selector.ShouldInstrumentFrame(
       CreateLowOverheadFrame(Timestamp::Millis(100)),
@@ -95,8 +96,8 @@ TEST(FrameSelectorTest, AlwaysSelectsKeyFrames) {
 }
 
 TEST(FrameSelectorTest, SelectsBasedOnLowOverheadSpan) {
-  FrameSelector selector(ScalabilityMode::kL1T1, kLowOverheadSpan,
-                         kHighOverheadSpan);
+  FrameSelector selector(CreateTestEnvironment(), ScalabilityMode::kL1T1,
+                         kLowOverheadSpan, kHighOverheadSpan);
   // First frame selected (to init).
   EXPECT_TRUE(selector.ShouldInstrumentFrame(
       CreateLowOverheadFrame(Timestamp::Millis(1000)),
@@ -117,8 +118,8 @@ TEST(FrameSelectorTest, SelectsBasedOnLowOverheadSpan) {
 }
 
 TEST(FrameSelectorTest, SelectsBasedOnHighOverheadSpan) {
-  FrameSelector selector(ScalabilityMode::kL1T1, kLowOverheadSpan,
-                         kHighOverheadSpan);
+  FrameSelector selector(CreateTestEnvironment(), ScalabilityMode::kL1T1,
+                         kLowOverheadSpan, kHighOverheadSpan);
   // First frame selected (to init).
   EXPECT_TRUE(selector.ShouldInstrumentFrame(
       CreateHighOverheadFrame(Timestamp::Millis(1000)),
@@ -139,8 +140,8 @@ TEST(FrameSelectorTest, SelectsBasedOnHighOverheadSpan) {
 }
 
 TEST(FrameSelectorTest, IndependentKeyframesWithSimulcast) {
-  FrameSelector selector(ScalabilityMode::kS2T2, kLowOverheadSpan,
-                         kHighOverheadSpan);
+  FrameSelector selector(CreateTestEnvironment(), ScalabilityMode::kS2T2,
+                         kLowOverheadSpan, kHighOverheadSpan);
   // Initial keyframe.
   EXPECT_TRUE(selector.ShouldInstrumentFrame(
       CreateLowOverheadFrame(Timestamp::Millis(1000)),
@@ -164,8 +165,8 @@ TEST(FrameSelectorTest, IndependentKeyframesWithSimulcast) {
 }
 
 TEST(FrameSelectorTest, TreatsDeltaAsKeyframeWithInterLayerPrediction) {
-  FrameSelector selector(ScalabilityMode::kL2T2, kLowOverheadSpan,
-                         kHighOverheadSpan);
+  FrameSelector selector(CreateTestEnvironment(), ScalabilityMode::kL2T2,
+                         kLowOverheadSpan, kHighOverheadSpan);
   // Initial keyframe.
   EXPECT_TRUE(selector.ShouldInstrumentFrame(
       CreateLowOverheadFrame(Timestamp::Millis(1000)),
@@ -190,8 +191,8 @@ TEST(FrameSelectorTest, TreatsDeltaAsKeyframeWithInterLayerPrediction) {
 }
 
 TEST(FrameSelector, SelectsAboutHalfInMiddleOfSpan) {
-  FrameSelector selector(ScalabilityMode::kL1T1, kLowOverheadSpan,
-                         kHighOverheadSpan);
+  FrameSelector selector(CreateTestEnvironment(), ScalabilityMode::kL1T1,
+                         kLowOverheadSpan, kHighOverheadSpan);
   Timestamp timestamp = Timestamp::Millis(1000);  // Arbitrary start timestamp;
   const TimeDelta p50_delta =
       (kLowOverheadSpan.lower_bound + kLowOverheadSpan.upper_bound) / 2;
@@ -217,8 +218,8 @@ TEST(FrameSelector, SelectsAboutHalfInMiddleOfSpan) {
 }
 
 TEST(FrameSelectorTest, FallbackToRtpTimestamp) {
-  FrameSelector selector(ScalabilityMode::kL1T1, kLowOverheadSpan,
-                         kHighOverheadSpan);
+  FrameSelector selector(CreateTestEnvironment(), ScalabilityMode::kL1T1,
+                         kLowOverheadSpan, kHighOverheadSpan);
   // Frame with 0 capture time but valid RTP.
   // 90kHz clock. 1 second = 90000.
   // Start at 0.

From 5e13499a3812d3ae0511574f26ee86bb431b0fc2 Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Wed, 4 Mar 2026 10:19:57 +0100
Subject: [PATCH 192/472] Update documentation to encourage std::span instead
 of ArrayView

No-try: true
Bug: webrtc:439801349
Change-Id: Ia1e7bb64b5de7565d3a4266716f8c05a9a6cd930
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453340
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47045}
---
 g3doc/abseil-in-webrtc.md |  9 +++------
 g3doc/style-guide.md      | 24 ++++++++++++++++--------
 2 files changed, 19 insertions(+), 14 deletions(-)

diff --git a/g3doc/abseil-in-webrtc.md b/g3doc/abseil-in-webrtc.md
index fd74182176..7a2f8deb10 100644
--- a/g3doc/abseil-in-webrtc.md
+++ b/g3doc/abseil-in-webrtc.md
@@ -63,13 +63,10 @@ on a monolithic Abseil build target that will generate a shared library.
 
 ### `absl::Span`
 
-*Use `webrtc::ArrayView` instead.*
+*Use `std::span` instead.*
 
-`absl::Span` differs from `webrtc::ArrayView` on several points, and both
-of them differ from the `std::span` introduced in C++20. We should just keep
-using `webrtc::ArrayView` and avoid `absl::Span`. Note that we are planning
-to replace `webrtc::ArrayView` with `std::span` rather than with `absl::Span`,
-see https://bugs.webrtc.org/439801349
+`absl::Span` differs from `std::span` on several points, in particular lacks
+static extent template parameter that WebRTC relies on.
 
 ### `absl::StrCat`, `absl::StrAppend`, `absl::StrJoin`, `absl::StrSplit`
 
diff --git a/g3doc/style-guide.md b/g3doc/style-guide.md
index afc886e118..b44b416892 100644
--- a/g3doc/style-guide.md
+++ b/g3doc/style-guide.md
@@ -130,9 +130,11 @@ call it using the `DEPRECATED_`-prefixed name don't get the warning.
 [DEPRECATED]: https://en.cppreference.com/w/cpp/language/attributes/deprecated
 [ABSL_DEPRECATE_AND_INLINE]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/abseil-cpp/absl/base/macros.h?q=ABSL_DEPRECATE_AND_INLINE
 
-### ArrayView
+### std::span
 
-When passing an array of values to a function, use `ArrayView`
+std::span is allowed and encouraged in WebRTC.
+
+When passing an array of values to a function, use `std::span`
 whenever possible—that is, whenever you're not passing ownership of
 the array, and don't allow the callee to change the array size.
 
@@ -140,14 +142,20 @@ For example,
 
 | instead of                          | use                  |
 |-------------------------------------|----------------------|
-| `const std::vector<T>&`             | `ArrayView<const T>` |
-| `const T* ptr, size_t num_elements` | `ArrayView<const T>` |
-| `T* ptr, size_t num_elements`       | `ArrayView<T>`       |
+| `const std::vector<T>&`             | `std::span<const T>` |
+| `const T* ptr, size_t num_elements` | `std::span<const T>` |
+| `T* ptr, size_t num_elements`       | `std::span<T>`       |
 
-See the [source code for `ArrayView`][ArrayView] for more detailed
-docs.
+See the [cpp reference][span] for more detailed docs.
 
-[ArrayView]: https://webrtc.googlesource.com/src/+/refs/heads/main/api/array_view.h
+std::span represents the same concept as base::span in Chromium and absl::Span
+in Abseil. WebRTC can't use base::span from Chromium, and prefers std::span over absl::Span.
+
+<!-- TODO: bugs.webrtc.org/439801349 - Delete this note after 2027-03-04 -->
+In the past WebRTC used own ArrayView type to represent a span, however that
+type has been migrated to std::span.
+
+[span]: https://en.cppreference.com/w/cpp/container/span.html
 
 ### Strings
 

From 230782283fa75d52d6e13da186140fcc708fbfdc Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Wed, 4 Mar 2026 04:11:24 -0800
Subject: [PATCH 193/472] Roll chromium_revision 04f26ef6ec..7ff1b6e256
 (1593719:1593841)

Change log: https://chromium.googlesource.com/chromium/src/+log/04f26ef6ec..7ff1b6e256
Full diff: https://chromium.googlesource.com/chromium/src/+/04f26ef6ec..7ff1b6e256

Changed dependencies
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/f3dfca4f79..147d613cc9
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/6a8f14f59f..b775d5f45c
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/a3bc0f1a84..64e303a7fc
* src/third_party/androidx/cipd: CW8b7YlBGZSbouGqDIHzMn5HRGQAqAM8d0kEcogTqJMC..6muHdqoZTRpe7RidORu60nYuGnBP5ylKApzvMhIx7LMC
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/b0ebf966a5..0269c017aa
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/05b8b9cf93..75904c77e1
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/0d31bdf20c..cee9ada040
DEPS diff: https://chromium.googlesource.com/chromium/src/+/04f26ef6ec..7ff1b6e256/DEPS

BUG=None

Change-Id: If174f0ab15ed080e0e69dab2a534cb225b739aff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453620
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47046}
---
 DEPS | 16 ++++++++--------
 1 file changed, 8 insertions(+), 8 deletions(-)

diff --git a/DEPS b/DEPS
index 2e74a57e69..c0e924963a 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '04f26ef6ecf70adc1a82bb46e041f494a7f392c5',
+  'chromium_revision': '7ff1b6e25600e3b16fe650a9b019b09e73a74a94',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@f3dfca4f794abc1f048f43f161b6e961ed70050f',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@147d613cc9abf67cd1a64a9ed0fa2ed80d196a98',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@6a8f14f59fd1e116cac6c7c201677f39a7483925',
+    'https://chromium.googlesource.com/chromium/src/testing@b775d5f45c707c7e249691021463375cc9cd7322',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@a3bc0f1a8407981ffb95feef7b09e8ecd10b99e8',
+    'https://chromium.googlesource.com/chromium/src/third_party@64e303a7fcb43f4ba6e30ad9b89b9cae34f43d1a',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '05b8b9cf93e16e960e71f38bf2e41944b4611f22',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '75904c77e1cce25a8d3f32658444fe1687586550',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@0d31bdf20c5d30ddeb83f0645ee74648f0356673',
+    'https://chromium.googlesource.com/chromium/src/tools@cee9ada0409ec8ec1321d9a96c7092ae397c37be',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'CW8b7YlBGZSbouGqDIHzMn5HRGQAqAM8d0kEcogTqJMC',
+          'version': '6muHdqoZTRpe7RidORu60nYuGnBP5ylKApzvMhIx7LMC',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -864,7 +864,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7364b490b5f78d58efe23ea76e74210fd6c3c76f',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + 'b0ebf966a58d9169739cb12f3a6e5e7a908b2bfc',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '0269c017aa20e30a35346f8552efe558e4ebfce4',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From cc9f553304940a2336b1ff207a691176c36c905e Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Wed, 4 Mar 2026 12:16:33 +0000
Subject: [PATCH 194/472] Regularize wildcard handling for rtcp-fb attributes.

Make the code indifferent to whether rtcp-fb section-wide
attributes are signalled using * or repeated for all codecs.

Bug: webrtc:489633228
Change-Id: I965a881361256f184cefd68ae1f41adeee2a5a70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453640
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47047}
---
 api/webrtc_sdp.cc          | 22 +++++++++++++++++++---
 api/webrtc_sdp_unittest.cc | 38 +++++++++++++++++++++++++++++++++++---
 2 files changed, 54 insertions(+), 6 deletions(-)

diff --git a/api/webrtc_sdp.cc b/api/webrtc_sdp.cc
index ca180ed0ad..f404b90eb7 100644
--- a/api/webrtc_sdp.cc
+++ b/api/webrtc_sdp.cc
@@ -2437,11 +2437,26 @@ void UpdateFromWildcardCodecs(MediaContentDescription* desc) {
   for (auto& codec : codecs) {
     AddFeedbackParameters(wildcard_codec->feedback_params, &codec);
   }
-  // Special treatment for transport-wide feedback params.
-  if (wildcard_codec->feedback_params.Has({"ack", "ccfb"})) {
+  desc->set_codecs(codecs);
+}
+
+// Update settings that are only valid if media-section-wide, but
+// are represented as per-codecs settings.
+// Note that this is done after UpdateFromWildcardCodecs, so settings
+// that are set with wildcard will also be picked up here.
+void UpdateFromMediaSectionWideSettings(MediaContentDescription* desc) {
+  if (desc->codecs().empty()) {
+    return;
+  }
+  bool all_codecs_have_ack_ccfb = true;
+  for (const auto& codec : desc->codecs()) {
+    if (!codec.feedback_params.Has({"ack", "ccfb"})) {
+      all_codecs_have_ack_ccfb = false;
+    }
+  }
+  if (all_codecs_have_ack_ccfb) {
     desc->set_rtcp_fb_ack_ccfb(true);
   }
-  desc->set_codecs(codecs);
 }
 
 void AddAudioAttribute(const std::string& name,
@@ -2862,6 +2877,7 @@ bool ParseContent(absl::string_view message,
   }
 
   UpdateFromWildcardCodecs(media_desc);
+  UpdateFromMediaSectionWideSettings(media_desc);
   // Codec has not been populated correctly unless the name has been set. This
   // can happen if an SDP has an fmtp or rtcp-fb with a payload type but doesn't
   // have a corresponding "rtpmap" line. This should lead to a parse error.
diff --git a/api/webrtc_sdp_unittest.cc b/api/webrtc_sdp_unittest.cc
index 8cdf977f92..cc58f50bce 100644
--- a/api/webrtc_sdp_unittest.cc
+++ b/api/webrtc_sdp_unittest.cc
@@ -1927,7 +1927,8 @@ class WebRtcSdpTest : public ::testing::Test {
 
   void TestDeserializeRtcpFb(
       std::unique_ptr<SessionDescriptionInterface>& jdesc_output,
-      bool use_wildcard) {
+      bool use_wildcard,
+      bool use_ccfb) {
     std::string sdp_session_and_audio =
         "v=0\r\n"
         "o=- 18446744069414584320 18446462598732840960 IN IP4 127.0.0.1\r\n"
@@ -1949,8 +1950,14 @@ class WebRtcSdpTest : public ::testing::Test {
     std::ostringstream os;
     os << sdp_session_and_audio;
     os << "a=rtcp-fb:" << (use_wildcard ? "*" : "111") << " nack\r\n";
+    if (use_ccfb) {
+      os << "a=rtcp-fb:" << (use_wildcard ? "*" : "111") << " ack ccfb\r\n";
+    }
     os << sdp_video;
     os << "a=rtcp-fb:" << (use_wildcard ? "*" : "101") << " ccm fir\r\n";
+    if (use_ccfb) {
+      os << "a=rtcp-fb:" << (use_wildcard ? "*" : "101") << " ack ccfb\r\n";
+    }
     std::string sdp = os.str();
     // Deserialize
     SdpParseError error;
@@ -1982,6 +1989,21 @@ class WebRtcSdpTest : public ::testing::Test {
         FeedbackParam(kRtcpFbParamRemb, kParamValueEmpty)));
     EXPECT_TRUE(vp8.HasFeedbackParam(
         FeedbackParam(kRtcpFbParamCcm, kRtcpFbCcmParamFir)));
+    MediaContentDescription* video_media =
+        GetFirstMediaContent(jdesc_output->description(), MediaType::VIDEO)
+            ->media_description();
+    MediaContentDescription* audio_media =
+        GetFirstMediaContent(jdesc_output->description(), MediaType::AUDIO)
+            ->media_description();
+    if (use_ccfb) {
+      EXPECT_TRUE(video_media->rtcp_fb_ack_ccfb())
+          << "Wildcard: " << use_wildcard;
+      EXPECT_TRUE(audio_media->rtcp_fb_ack_ccfb())
+          << "Wildcard: " << use_wildcard;
+    } else {
+      EXPECT_FALSE(video_media->rtcp_fb_ack_ccfb());
+      EXPECT_FALSE(audio_media->rtcp_fb_ack_ccfb());
+    }
   }
 
   // Two SDP messages can mean the same thing but be different strings, e.g.
@@ -3225,14 +3247,24 @@ TEST_F(WebRtcSdpTest, DeserializeSerializeCodecParams) {
 TEST_F(WebRtcSdpTest, DeserializeSerializeRtcpFb) {
   const bool kUseWildcard = false;
   std::unique_ptr<SessionDescriptionInterface> jdesc_output;
-  TestDeserializeRtcpFb(jdesc_output, kUseWildcard);
+  TestDeserializeRtcpFb(jdesc_output, kUseWildcard, /* use_ccfb= */ false);
   TestSerialize(jdesc_output);
 }
 
 TEST_F(WebRtcSdpTest, DeserializeSerializeRtcpFbWildcard) {
   const bool kUseWildcard = true;
   std::unique_ptr<SessionDescriptionInterface> jdesc_output;
-  TestDeserializeRtcpFb(jdesc_output, kUseWildcard);
+  TestDeserializeRtcpFb(jdesc_output, kUseWildcard, /* use_ccfb= */ false);
+  TestSerialize(jdesc_output);
+}
+
+TEST_F(WebRtcSdpTest, DeserializeSerializeRtcpWithCcfb) {
+  std::unique_ptr<SessionDescriptionInterface> jdesc_output;
+  TestDeserializeRtcpFb(jdesc_output, /* use_wildcard= */ false,
+                        /* use_ccfb= */ true);
+  TestSerialize(jdesc_output);
+  TestDeserializeRtcpFb(jdesc_output, /* use_wildcard= */ true,
+                        /* use_ccfb= */ true);
   TestSerialize(jdesc_output);
 }
 

From 8ec8cf9c14279a4d1c3338d9df5075a133abe7f3 Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Wed, 4 Mar 2026 13:38:22 +0100
Subject: [PATCH 195/472] Fix configuring initial bwe in scream controller

Set it when initial probing is configured to true, rather than std::optional initial probing is set to some value.

Bug: None
Change-Id: I326c9a12399f2364f48544b45a150fecb3e8d575
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453320
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47048}
---
 api/transport/network_types.h                              | 2 +-
 .../goog_cc/goog_cc_network_control.cc                     | 7 ++-----
 2 files changed, 3 insertions(+), 6 deletions(-)

diff --git a/api/transport/network_types.h b/api/transport/network_types.h
index fa0617743a..831e8a3670 100644
--- a/api/transport/network_types.h
+++ b/api/transport/network_types.h
@@ -52,7 +52,7 @@ struct RTC_EXPORT StreamsConfig {
   // If `enable_repeated_initial_probing` is set to true, Probes are sent
   // periodically every 1s during the first 5s after the network becomes
   // available. The probes ignores max_total_allocated_bitrate.
-  std::optional<bool> enable_repeated_initial_probing;
+  bool enable_repeated_initial_probing = false;
   std::optional<double> pacing_factor;
 
   // TODO(srte): Use BitrateAllocationLimits here.
diff --git a/modules/congestion_controller/goog_cc/goog_cc_network_control.cc b/modules/congestion_controller/goog_cc/goog_cc_network_control.cc
index 1b486e769c..2ba1aa6fdb 100644
--- a/modules/congestion_controller/goog_cc/goog_cc_network_control.cc
+++ b/modules/congestion_controller/goog_cc/goog_cc_network_control.cc
@@ -200,11 +200,8 @@ NetworkControlUpdate GoogCcNetworkController::OnProcessInterval(
       probe_controller_->EnablePeriodicAlrProbing(
           *initial_config_->stream_based_config.requests_alr_probing);
     }
-    if (initial_config_->stream_based_config.enable_repeated_initial_probing) {
-      probe_controller_->EnableRepeatedInitialProbing(
-          *initial_config_->stream_based_config
-               .enable_repeated_initial_probing);
-    }
+    probe_controller_->EnableRepeatedInitialProbing(
+        initial_config_->stream_based_config.enable_repeated_initial_probing);
     std::optional<DataRate> total_bitrate =
         initial_config_->stream_based_config.max_total_allocated_bitrate;
     if (total_bitrate) {

From 9b43041fb76ae6f895f8c721165a3fdedbcfa550 Mon Sep 17 00:00:00 2001
From: Philipp Hancke <philipp.hancke@googlemail.com>
Date: Mon, 2 Mar 2026 12:09:33 +0100
Subject: [PATCH 196/472] dtls-in-stun: log the benchmark name in the result

which makes it easier to identify a particular benchmark and run it with
  --gtest_filter

Bug: webrtc:367395350
Change-Id: I1416a7141faa29e2bf9789f3127990499195f917
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452942
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47049}
---
 p2p/dtls/dtls_ice_integration_bench.cc | 12 +++++++-----
 1 file changed, 7 insertions(+), 5 deletions(-)

diff --git a/p2p/dtls/dtls_ice_integration_bench.cc b/p2p/dtls/dtls_ice_integration_bench.cc
index 0171e2fac7..73bc6f3330 100644
--- a/p2p/dtls/dtls_ice_integration_bench.cc
+++ b/p2p/dtls/dtls_ice_integration_bench.cc
@@ -114,11 +114,13 @@ TEST_P(DtlsIceIntegrationBenchmark, Benchmark) {
   RTC_LOG(LS_INFO) << GetParam() << " START";
 
   SamplesStatsCounter stats = RunBenchmark(iter);
-  RTC_LOG(LS_INFO) << GetParam() << " RESULT:"
-                   << " p10: " << stats.GetPercentile(0.10)
-                   << " p50: " << stats.GetPercentile(0.50)
-                   << " avg: " << stats.GetAverage()
-                   << " p95: " << stats.GetPercentile(0.95);
+  RTC_LOG(LS_INFO)
+      << GetParam() << " RESULT:"
+      << " p10: " << stats.GetPercentile(0.10)
+      << " p50: " << stats.GetPercentile(0.50)
+      << " avg: " << stats.GetAverage()  // Keep the format.
+      << " p95: " << stats.GetPercentile(0.95) << " "
+      << ::testing::UnitTest::GetInstance()->current_test_info()->name();
 }
 
 }  // namespace

From db4c793a2e36fc70de7e924fec63bf13c1cd856b Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Wed, 4 Mar 2026 10:03:09 -0800
Subject: [PATCH 197/472] Roll chromium_revision 7ff1b6e256..0ae5bdcc6c
 (1593841:1594043)

Change log: https://chromium.googlesource.com/chromium/src/+log/7ff1b6e256..0ae5bdcc6c
Full diff: https://chromium.googlesource.com/chromium/src/+/7ff1b6e256..0ae5bdcc6c

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/80d98a1ed9..d863633554
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/147d613cc9..d4fc3d25e8
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/b775d5f45c..144384ca41
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/64e303a7fc..059e0423cc
* src/third_party/androidx/cipd: 6muHdqoZTRpe7RidORu60nYuGnBP5ylKApzvMhIx7LMC..Y9ecsp20FCn4bBgTNBLDqMM8XRz45d17SEsZ6tqD8XsC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/fbf01ffae1..e16f17a59e
* src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/0cd354e178..a18fb0c5f8
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/0269c017aa..abc3d6014d
* src/third_party/freetype/src: https://chromium.googlesource.com/chromium/src/third_party/freetype2.git/+log/bf1f34d75e..d262bd978c
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/75904c77e1..f6077f1db9
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/cee9ada040..89ee4151f9
DEPS diff: https://chromium.googlesource.com/chromium/src/+/7ff1b6e256..0ae5bdcc6c/DEPS

BUG=None

Change-Id: I9eb77c98e29db326d552da37a2dcd4e840945356
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453700
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47050}
---
 DEPS | 24 ++++++++++++------------
 1 file changed, 12 insertions(+), 12 deletions(-)

diff --git a/DEPS b/DEPS
index c0e924963a..c374d8db8a 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '7ff1b6e25600e3b16fe650a9b019b09e73a74a94',
+  'chromium_revision': '0ae5bdcc6cd044d004e52aabc554eb0fe169fad9',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@80d98a1ed98c47c52033553b5a10a5050291c2bc',
+    'https://chromium.googlesource.com/chromium/src/build@d863633554de2574eefb9a2f2b86d0bfa8d92850',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@fef8d6947fa54202e8a00c58d5ac8fd15780bc33',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@147d613cc9abf67cd1a64a9ed0fa2ed80d196a98',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@d4fc3d25e8e0fb261bcd0e5ddbe5633034292ea2',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@b775d5f45c707c7e249691021463375cc9cd7322',
+    'https://chromium.googlesource.com/chromium/src/testing@144384ca4106a72eea2d3737ba359480902dc366',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@64e303a7fcb43f4ba6e30ad9b89b9cae34f43d1a',
+    'https://chromium.googlesource.com/chromium/src/third_party@059e0423cc1e905b99be381e60396a4538211093',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -331,7 +331,7 @@ deps = {
   'src/third_party/clang-format/script':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@c2725e0622e1a86d55f14514f2177a39efea4a0e',
   'src/third_party/compiler-rt/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@0cd354e178fec2af0c516235f945ceca8611b337',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@a18fb0c5f8c7b5fc6996dfc77b3fb3df92ff549d',
   'src/third_party/libc++/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@7ab65651aed6802d2599dcb7a73b1f82d5179d05',
   'src/third_party/libc++abi/src':
@@ -488,7 +488,7 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@fbf01ffae10929d863b6744bb505f0cae6c9c712',
+    'https://boringssl.googlesource.com/boringssl.git@e16f17a59eab34110d011291ff97fbc84d776205',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@8be0e3114685fcc1589561067282edf75ea1259a',
   'src/third_party/catapult':
@@ -519,7 +519,7 @@ deps = {
       'condition': 'checkout_linux',
   },
   'src/third_party/freetype/src':
-    'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@bf1f34d75ed4a36358e7a4ae6b21082d0d435b0e',
+    'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@d262bd978c3ea303289153dba1ae8a6dc4ac747a',
   'src/third_party/harfbuzz-ng/src':
     'https://chromium.googlesource.com/external/github.com/harfbuzz/harfbuzz.git@fa2908bf16d2ccd6623f4d575455fea72a1a722b',
   'src/third_party/google_benchmark/src': {
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '75904c77e1cce25a8d3f32658444fe1687586550',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'f6077f1db9bc56d81cdd1874e2310ff21438060c',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@cee9ada0409ec8ec1321d9a96c7092ae397c37be',
+    'https://chromium.googlesource.com/chromium/src/tools@89ee4151f96f34df6525307d12af38eadc46ba4e',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': '6muHdqoZTRpe7RidORu60nYuGnBP5ylKApzvMhIx7LMC',
+          'version': 'Y9ecsp20FCn4bBgTNBLDqMM8XRz45d17SEsZ6tqD8XsC',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -864,7 +864,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7364b490b5f78d58efe23ea76e74210fd6c3c76f',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '0269c017aa20e30a35346f8552efe558e4ebfce4',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + 'abc3d6014d4bebb05e2a537e40ce36acfcf9ee26',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From fcea1cf20ab03a15d2ca702629208e965edb3990 Mon Sep 17 00:00:00 2001
From: Philip Eliasson <philipel@webrtc.org>
Date: Wed, 4 Mar 2026 13:51:43 +0100
Subject: [PATCH 198/472] Let dav1d retain references to wrapped bitstream
 buffers.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Bug: chromium:486421953
Change-Id: I855616d8206711b371df05a306730468ec8d23e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453200
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47051}
---
 .../video_coding/codecs/av1/dav1d_decoder.cc  | 25 ++++++--
 .../codecs/av1/dav1d_decoder_unittest.cc      | 59 ++++++++++++++-----
 2 files changed, 64 insertions(+), 20 deletions(-)

diff --git a/modules/video_coding/codecs/av1/dav1d_decoder.cc b/modules/video_coding/codecs/av1/dav1d_decoder.cc
index 7a52dfa1a3..9356dcb545 100644
--- a/modules/video_coding/codecs/av1/dav1d_decoder.cc
+++ b/modules/video_coding/codecs/av1/dav1d_decoder.cc
@@ -82,9 +82,6 @@ class ScopedDav1dPicture : public RefCountedNonVirtual<ScopedDav1dPicture> {
 
 constexpr char kDav1dName[] = "dav1d";
 
-// Calling `dav1d_data_wrap` requires a `free_callback` to be registered.
-void NullFreeCallback(const uint8_t* /* buffer */, void* /* opaque */) {}
-
 Dav1dDecoder::Dav1dDecoder() = default;
 
 Dav1dDecoder::Dav1dDecoder(const Environment& env)
@@ -142,9 +139,25 @@ int32_t Dav1dDecoder::Decode(const EncodedImage& encoded_image,
 
   ScopedDav1dData scoped_dav1d_data;
   Dav1dData& dav1d_data = scoped_dav1d_data.Data();
-  dav1d_data_wrap(&dav1d_data, encoded_image.data(), encoded_image.size(),
-                  /*free_callback=*/&NullFreeCallback,
-                  /*user_data=*/nullptr);
+
+  // Calling GetEncodedData will create a new `scoped_refptr` and increment the
+  // ref count. By simply releasing we are now responsible for decrementing
+  // the ref count when appropriate, which is when dav1d calls the
+  // `free_callback` to indicate that the buffer is no longer needed.
+  EncodedImageBufferInterface* bitstream_buffer =
+      encoded_image.GetEncodedData().release();
+
+  // Note that the `bitstream_buffer` can have a higher capacity than what is
+  // actually being used, so `encoded_image.size()` should be used to get the
+  // actual size of the bitstream.
+  dav1d_data_wrap(
+      &dav1d_data, encoded_image.data(), encoded_image.size(),
+      /*free_callback=*/
+      [](const uint8_t* /* buffer */, void* user_data) {
+        auto* bb = static_cast<EncodedImageBufferInterface*>(user_data);
+        bb->Release();
+      },
+      /*user_data=*/bitstream_buffer);
 
   if (int decode_res = dav1d_send_data(context_, &dav1d_data)) {
     RTC_LOG(LS_WARNING)
diff --git a/modules/video_coding/codecs/av1/dav1d_decoder_unittest.cc b/modules/video_coding/codecs/av1/dav1d_decoder_unittest.cc
index 1c6477bac1..217edc1048 100644
--- a/modules/video_coding/codecs/av1/dav1d_decoder_unittest.cc
+++ b/modules/video_coding/codecs/av1/dav1d_decoder_unittest.cc
@@ -67,13 +67,9 @@ class TestAv1Decoder : public DecodedImageCallback {
   TestAv1Decoder(const TestAv1Decoder&) = delete;
   TestAv1Decoder& operator=(const TestAv1Decoder&) = delete;
 
-  void Decode(const EncodedImage& image) {
-    ASSERT_THAT(decoder_, NotNull());
+  int32_t Decode(const EncodedImage& image) {
     decoded_frame_ = std::nullopt;
-    int32_t error =
-        decoder_->Decode(image, /*render_time_ms=*/image.capture_time_ms_);
-    ASSERT_EQ(error, WEBRTC_VIDEO_CODEC_OK);
-    ASSERT_THAT(decoded_frame_, Not(Eq(std::nullopt)));
+    return decoder_->Decode(image, /*render_time_ms=*/image.capture_time_ms_);
   }
 
   VideoFrame& decoded_frame() { return *decoded_frame_; }
@@ -95,8 +91,9 @@ class TestAv1Decoder : public DecodedImageCallback {
 
 TEST(Dav1dDecoderTest, KeepsDecodedResolutionByDefault) {
   TestAv1Decoder decoder(CreateEnvironment());
-  decoder.Decode(
-      CreateEncodedImage(kAv1FrameWith36x20EncodededAnd32x16RenderResolution));
+  EXPECT_EQ(decoder.Decode(CreateEncodedImage(
+                kAv1FrameWith36x20EncodededAnd32x16RenderResolution)),
+            WEBRTC_VIDEO_CODEC_OK);
   EXPECT_EQ(decoder.decoded_frame().width(), 36);
   EXPECT_EQ(decoder.decoded_frame().height(), 20);
 }
@@ -104,8 +101,9 @@ TEST(Dav1dDecoderTest, KeepsDecodedResolutionByDefault) {
 TEST(Dav1dDecoderTest, CropsToRenderResolutionWhenCropIsEnabled) {
   TestAv1Decoder decoder(CreateEnvironment(CreateTestFieldTrialsPtr(
       "WebRTC-Dav1dDecoder-CropToRenderResolution/Enabled/")));
-  decoder.Decode(
-      CreateEncodedImage(kAv1FrameWith36x20EncodededAnd32x16RenderResolution));
+  EXPECT_EQ(decoder.Decode(CreateEncodedImage(
+                kAv1FrameWith36x20EncodededAnd32x16RenderResolution)),
+            WEBRTC_VIDEO_CODEC_OK);
   EXPECT_EQ(decoder.decoded_frame().width(), 32);
   EXPECT_EQ(decoder.decoded_frame().height(), 16);
 }
@@ -113,16 +111,18 @@ TEST(Dav1dDecoderTest, CropsToRenderResolutionWhenCropIsEnabled) {
 TEST(Dav1dDecoderTest, DoesNotCropToRenderResolutionWhenCropIsDisabled) {
   TestAv1Decoder decoder(CreateEnvironment(CreateTestFieldTrialsPtr(
       "WebRTC-Dav1dDecoder-CropToRenderResolution/Disabled/")));
-  decoder.Decode(
-      CreateEncodedImage(kAv1FrameWith36x20EncodededAnd32x16RenderResolution));
+  EXPECT_EQ(decoder.Decode(CreateEncodedImage(
+                kAv1FrameWith36x20EncodededAnd32x16RenderResolution)),
+            WEBRTC_VIDEO_CODEC_OK);
   EXPECT_EQ(decoder.decoded_frame().width(), 36);
   EXPECT_EQ(decoder.decoded_frame().height(), 20);
 }
 
 TEST(Dav1dDecoderTest, SetsColorSpaceOnDecodedFrame) {
   TestAv1Decoder decoder(CreateEnvironment());
-  decoder.Decode(
-      CreateEncodedImage(kAv1FrameWithBT709FullRangeColorSpace));
+  EXPECT_EQ(
+      decoder.Decode(CreateEncodedImage(kAv1FrameWithBT709FullRangeColorSpace)),
+      WEBRTC_VIDEO_CODEC_OK);
   auto color_space = decoder.decoded_frame().color_space();
   EXPECT_TRUE(color_space.has_value());
   EXPECT_EQ(color_space->primaries(), ColorSpace::PrimaryID::kBT709);
@@ -131,6 +131,37 @@ TEST(Dav1dDecoderTest, SetsColorSpaceOnDecodedFrame) {
   EXPECT_EQ(color_space->range(), ColorSpace::RangeID::kFull);
 }
 
+TEST(Dav1dDecoderTest, FailDecoderOnTiles) {
+  // This test tests buffer management, any failures would be detected
+  // by running this test under ASAN.
+  static constexpr uint8_t kTile0[] = {
+      0x0a, 0x0a, 0x00, 0x00, 0x00, 0x04, 0x3c, 0xff, 0xbc, 0x01, 0xa0, 0x08,
+      0x32, 0xbd, 0x01, 0x10, 0x60, 0xb0, 0x01, 0xa6, 0x9a, 0x68, 0x50, 0x91,
+      0xb0, 0x80, 0xb0, 0xd9, 0x8a, 0x40, 0xe4, 0x3e, 0xfb, 0x06, 0x1f, 0x0e,
+      0xcc, 0xdc, 0xcb, 0x8c, 0x2d, 0x5c, 0xb0, 0x52, 0xe0, 0xb3, 0x89, 0x2b,
+      0x6b, 0x8b, 0x56, 0xd3, 0x0a, 0x70, 0xad, 0x26, 0xf0, 0xca, 0xfa, 0xad,
+      0x27, 0xcc, 0x5b, 0x8c, 0x0c, 0x72, 0x76, 0xa7, 0x42, 0x69, 0x4c, 0x47,
+      0xf4, 0x2b, 0xa9, 0x5e, 0x13, 0x81, 0x69, 0x47, 0x40, 0x13, 0x3e, 0xac,
+      0xef, 0x0f, 0x74, 0x02, 0x79, 0xa7, 0x50, 0xf4, 0x50, 0x82, 0x4d, 0x4f,
+      0x70, 0x02, 0x9c, 0x0f, 0x61, 0xbc, 0x1c, 0x22, 0xc2, 0xac, 0x9f, 0x88,
+      0xbd, 0x2a, 0xdc, 0xdb, 0xf0, 0xbe, 0x95, 0x54, 0x6e, 0xda, 0x08, 0x76,
+      0x2c, 0x86, 0xd2, 0x0f, 0x86, 0xc1, 0x86, 0xcb, 0x98, 0x35, 0xfc, 0x2c,
+      0xe7, 0x51, 0x44, 0x62, 0xfe, 0xf1, 0x97, 0x1f, 0xb0, 0x7f, 0x14, 0xc4,
+      0xef, 0xb0, 0x01, 0x7c, 0x07, 0x22, 0x3e, 0xda, 0x2b, 0xef, 0x59, 0xcc,
+      0x57, 0x65, 0xc5, 0x22, 0xe9, 0x61, 0x99, 0x0c, 0xd6, 0x07, 0x68, 0x1a,
+      0x26, 0xad, 0x9b, 0x1e, 0x91, 0x2c, 0xee, 0xe2, 0xe4, 0x7c, 0x4d, 0x07,
+      0x51, 0x72, 0x89, 0x51, 0xbf, 0x17, 0x67, 0xc0, 0x6d, 0x02, 0x69, 0x92,
+      0x20, 0x7b, 0xeb, 0x11, 0xea, 0xc1, 0x43, 0x59, 0x4f, 0x75, 0xbb, 0x20};
+  static constexpr uint8_t kTile1[] = {
+      0x22, 0x1e, 0xe0, 0xc1, 0xb6, 0x2f, 0x07, 0xd0, 0x0b, 0xaf, 0x14,
+      0xbf, 0x51, 0xa3, 0x0d, 0xf4, 0xb6, 0x45, 0xd8, 0x16, 0xda, 0x8c,
+      0xb0, 0xbf, 0x29, 0x39, 0xdf, 0x9a, 0x9e, 0x9c, 0x69, 0x80};
+
+  TestAv1Decoder decoder(CreateEnvironment());
+  decoder.Decode(CreateEncodedImage(kTile0));
+  decoder.Decode(CreateEncodedImage(kTile1));
+}
+
 }  // namespace
 }  // namespace test
 }  // namespace webrtc

From d0cbd0d804f49741532079844eb08a2f03665c45 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Wed, 4 Mar 2026 14:02:26 -0800
Subject: [PATCH 199/472] Roll chromium_revision 0ae5bdcc6c..bd3cb736a1
 (1594043:1594229)

Change log: https://chromium.googlesource.com/chromium/src/+log/0ae5bdcc6c..bd3cb736a1
Full diff: https://chromium.googlesource.com/chromium/src/+/0ae5bdcc6c..bd3cb736a1

Changed dependencies
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/d4fc3d25e8..88f3263876
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/144384ca41..ec8af4bd7b
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/059e0423cc..0aef453c8c
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/e16f17a59e..5774eca600
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/27f8f7c421..2167887d78
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/abc3d6014d..dd826edb42
* src/third_party/fuzztest/src: https://chromium.googlesource.com/external/github.com/google/fuzztest.git/+log/2774d423cc..1f7726d61f
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/060a217622..7ffb3786c6
* src/third_party/libunwind/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git/+log/17ccf7d110..78884e23fe
* src/third_party/libyuv: https://chromium.googlesource.com/libyuv/libyuv.git/+log/6067afde56..30809ff64a
* src/third_party/r8/cipd: giX-zHOTxdkwnAiuPSCUktVPPlw_zCiSGvkGQKfFmNgC..CQBWQVHNMmNBTkKhqA-Rnel-94ivC4c9jRROa8bvIUkC
* src/third_party/r8/d8/cipd: dKavKsYt7Uj8fRmoAALj-uumULBl43j-NHRUZ1YvnxUC..CQBWQVHNMmNBTkKhqA-Rnel-94ivC4c9jRROa8bvIUkC
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/89ee4151f9..baa30f6db6
DEPS diff: https://chromium.googlesource.com/chromium/src/+/0ae5bdcc6c..bd3cb736a1/DEPS

BUG=None

Change-Id: Ia38e4db3b58c6b2623f1217ba627924c77461be1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453740
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47052}
---
 DEPS | 28 ++++++++++++++--------------
 1 file changed, 14 insertions(+), 14 deletions(-)

diff --git a/DEPS b/DEPS
index c374d8db8a..4f550500a1 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '0ae5bdcc6cd044d004e52aabc554eb0fe169fad9',
+  'chromium_revision': 'bd3cb736a165e43ad71292b163a5fec194e6981e',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@d4fc3d25e8e0fb261bcd0e5ddbe5633034292ea2',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@88f3263876611ef82b25349577c721e13bc402fa',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@144384ca4106a72eea2d3737ba359480902dc366',
+    'https://chromium.googlesource.com/chromium/src/testing@ec8af4bd7b568f7d395097721597776495655bdd',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@059e0423cc1e905b99be381e60396a4538211093',
+    'https://chromium.googlesource.com/chromium/src/third_party@0aef453c8c27ade8985e3aeb50d2ff322d259471',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -339,7 +339,7 @@ deps = {
   'src/third_party/llvm-libc/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@e8faec51e35d65209d5ba660d711893ad3790735',
   'src/third_party/libunwind/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@17ccf7d110c5526cb77e93cfd8330f491fb2bf18',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@78884e23fe39cf5cc6987ea188a9b802d65a21c9',
 
   'src/third_party/test_fonts/test_fonts': {
       'dep_type': 'gcs',
@@ -488,7 +488,7 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@e16f17a59eab34110d011291ff97fbc84d776205',
+    'https://boringssl.googlesource.com/boringssl.git@5774eca6004ed7c7467bd644c057797ca96b65f2',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@8be0e3114685fcc1589561067282edf75ea1259a',
   'src/third_party/catapult':
@@ -505,13 +505,13 @@ deps = {
   'src/third_party/crc32c/src':
     'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558',
   'src/third_party/depot_tools':
-    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@27f8f7c4211120fabd8b6c186b13a77732551ae4',
+    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@2167887d7880c480873ca7d9a8621d89b72c7130',
   'src/third_party/ffmpeg':
     'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@946d97db8d906277085e361892b7efda5152e2f1',
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@060a21762295ba058c599a4e879996363bc8ca46',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@7ffb3786c67639e1300af11bcc51e61db087f954',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -593,7 +593,7 @@ deps = {
   'src/third_party/libFuzzer/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt/lib/fuzzer.git@bea408a6e01f0f7e6c82a43121fe3af4506c932e',
   'src/third_party/fuzztest/src':
-    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@2774d423cc62d44035ad8dd1651d15e4756a163f',
+    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@1f7726d61f7afa9aca1198a9395ede472ed70366',
   'src/third_party/libjpeg_turbo':
     'https://chromium.googlesource.com/chromium/deps/libjpeg_turbo.git@d1f5f2393e0d51f840207342ae86e55a86443288',
   'src/third_party/libsrtp':
@@ -615,7 +615,7 @@ deps = {
   'src/third_party/libvpx/source/libvpx':
     'https://chromium.googlesource.com/webm/libvpx.git@9a2d3d1f46afbdfa9b9820a9fd3aacb084e65e2f',
   'src/third_party/libyuv':
-    'https://chromium.googlesource.com/libyuv/libyuv.git@6067afde563c3946eebd94f146b3824ab7a97a9c',
+    'https://chromium.googlesource.com/libyuv/libyuv.git@30809ff64a9ca5e45f86439c0d474c2d3eef3d05',
   'src/third_party/lss': {
     'url': 'https://chromium.googlesource.com/linux-syscall-support.git@29164a80da4d41134950d76d55199ea33fbb9613',
     'condition': 'checkout_android or checkout_linux',
@@ -644,7 +644,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/r8',
-              'version': 'giX-zHOTxdkwnAiuPSCUktVPPlw_zCiSGvkGQKfFmNgC',
+              'version': 'CQBWQVHNMmNBTkKhqA-Rnel-94ivC4c9jRROa8bvIUkC',
           },
       ],
       'condition': 'checkout_android',
@@ -657,7 +657,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/r8',
-              'version': 'dKavKsYt7Uj8fRmoAALj-uumULBl43j-NHRUZ1YvnxUC',
+              'version': 'CQBWQVHNMmNBTkKhqA-Rnel-94ivC4c9jRROa8bvIUkC',
           },
       ],
       'condition': 'checkout_android',
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@89ee4151f96f34df6525307d12af38eadc46ba4e',
+    'https://chromium.googlesource.com/chromium/src/tools@baa30f6db60e23ac66e93dba5062cbe590540612',
 
   'src/third_party/espresso': {
       'packages': [
@@ -864,7 +864,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7364b490b5f78d58efe23ea76e74210fd6c3c76f',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + 'abc3d6014d4bebb05e2a537e40ce36acfcf9ee26',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + 'dd826edb42520814dab9239ec81f32ab91a4afe4',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From 3bff42b6de47957d86c86c646ba690a6f0d679c3 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Erik=20Spr=C3=A5ng?= <sprang@webrtc.org>
Date: Wed, 4 Mar 2026 15:05:37 +0100
Subject: [PATCH 200/472] Fix stale adaptation flag in video send stream stats
 on reconfiguration
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This CL makes sure we reset the CPU/BW adaptation state in the video
send stream stats when the stream was torn down and reconfigured.

Otherwise we may report the wrong value until an adaptation change
happens again.

Bug: webrtc:489633230
Change-Id: I97f122ab33b8d9ee078bdecaf61ae80feabb9606
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453660
Auto-Submit: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47053}
---
 media/BUILD.gn                               |  1 +
 media/engine/webrtc_video_engine.cc          | 10 ++++
 media/engine/webrtc_video_engine_unittest.cc | 48 ++++++++++++++++++++
 3 files changed, 59 insertions(+)

diff --git a/media/BUILD.gn b/media/BUILD.gn
index a3075d9db0..47e6d7b403 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -642,6 +642,7 @@ rtc_library("rtc_audio_video") {
     "../call:rtp_interfaces",
     "../call:video_receive_stream_api",
     "../call:video_send_stream_api",
+    "../common_video",
     "../common_video:frame_counts",
     "../modules/async_audio_processing",
     "../modules/audio_mixer:audio_mixer_impl",
diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc
index 540b3a12ef..640abe6386 100644
--- a/media/engine/webrtc_video_engine.cc
+++ b/media/engine/webrtc_video_engine.cc
@@ -73,6 +73,7 @@
 #include "call/video_receive_stream.h"
 #include "call/video_send_stream.h"
 #include "common_video/frame_counts.h"
+#include "common_video/include/quality_limitation_reason.h"
 #include "media/base/codec.h"
 #include "media/base/codec_comparators.h"
 #include "media/base/media_channel.h"
@@ -2731,6 +2732,15 @@ void WebRtcVideoSendChannel::WebRtcVideoSendStream::RecreateWebRtcStream() {
     call_->DestroyVideoSendStream(stream_);
     stream_ = call_->CreateVideoSendStream(std::move(config),
                                            parameters_.encoder_config.Copy());
+
+    // A new stream is created without any scaling or limitations, so these
+    // flags don't apply until the new stream experiences an adaptation event.
+    stats.bw_limited_resolution = false;
+    stats.bw_limited_framerate = false;
+    stats.cpu_limited_resolution = false;
+    stats.cpu_limited_framerate = false;
+    stats.quality_limitation_reason = QualityLimitationReason::kNone;
+
     stream_->SetStats(stats);
   } else {
     stream_ = call_->CreateVideoSendStream(std::move(config),
diff --git a/media/engine/webrtc_video_engine_unittest.cc b/media/engine/webrtc_video_engine_unittest.cc
index e1f3b7ef4a..d208e45521 100644
--- a/media/engine/webrtc_video_engine_unittest.cc
+++ b/media/engine/webrtc_video_engine_unittest.cc
@@ -5669,6 +5669,54 @@ TEST_F(WebRtcVideoChannelTest, ReceiveStreamReceivingByDefault) {
   EXPECT_TRUE(AddRecvStream()->IsReceiving());
 }
 
+TEST_F(WebRtcVideoChannelTest,
+       SetSendParametersRecreatesStreamAndClearsAdaptationStats) {
+  FakeVideoSendStream* stream = AddSendStream();
+  webrtc::VideoSendStream::Stats stats;
+  // Set cumulative stats that should be retained.
+  stats.number_of_cpu_adapt_changes = 2;
+  stats.number_of_quality_adapt_changes = 3;
+  stats.quality_limitation_durations_ms[webrtc::QualityLimitationReason::kCpu] =
+      100;
+
+  // Set ephemeral flags that should be cleared by RecreateWebRtcStream.
+  stats.bw_limited_resolution = true;
+  stats.bw_limited_framerate = true;
+  stats.cpu_limited_resolution = true;
+  stats.cpu_limited_framerate = true;
+  stats.quality_limitation_reason = webrtc::QualityLimitationReason::kBandwidth;
+
+  stream->SetStats(stats);
+
+  // Trigger stream recreation by changing construction-time parameters.
+  EXPECT_EQ(1, fake_call_->GetNumCreatedSendStreams());
+  VideoSenderParameters parameters;
+  parameters.codecs.push_back(GetEngineCodec("VP8"));
+  parameters.extmap_allow_mixed = true;  // This forces recreation.
+  EXPECT_TRUE(send_channel_->SetSenderParameters(parameters));
+  EXPECT_EQ(2, fake_call_->GetNumCreatedSendStreams());
+
+  ASSERT_EQ(1U, fake_call_->GetVideoSendStreams().size());
+  FakeVideoSendStream* new_stream = fake_call_->GetVideoSendStreams()[0];
+
+  webrtc::VideoSendStream::Stats new_stats =
+      static_cast<webrtc::VideoSendStream*>(new_stream)->GetStats();
+
+  // Check that ephemeral flags were cleared.
+  EXPECT_FALSE(new_stats.bw_limited_resolution);
+  EXPECT_FALSE(new_stats.bw_limited_framerate);
+  EXPECT_FALSE(new_stats.cpu_limited_resolution);
+  EXPECT_FALSE(new_stats.cpu_limited_framerate);
+  EXPECT_EQ(webrtc::QualityLimitationReason::kNone,
+            new_stats.quality_limitation_reason);
+
+  // Check that cumulative stats were retained.
+  EXPECT_EQ(2, new_stats.number_of_cpu_adapt_changes);
+  EXPECT_EQ(3, new_stats.number_of_quality_adapt_changes);
+  EXPECT_EQ(100, new_stats.quality_limitation_durations_ms
+                     [webrtc::QualityLimitationReason::kCpu]);
+}
+
 TEST_F(WebRtcVideoChannelTest, SetSend) {
   FakeVideoSendStream* stream = AddSendStream();
   EXPECT_FALSE(stream->IsSending());

From 48166af96728ceee9580884fbd7f850b205d52e5 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Wed, 4 Mar 2026 18:02:33 -0800
Subject: [PATCH 201/472] Roll chromium_revision bd3cb736a1..4b7d3263af
 (1594229:1594384)

Change log: https://chromium.googlesource.com/chromium/src/+log/bd3cb736a1..4b7d3263af
Full diff: https://chromium.googlesource.com/chromium/src/+/bd3cb736a1..4b7d3263af

Changed dependencies
* siso_version: git_revision:7495bf92ea44f7eed89273b5aea7bd72b702ed1c..git_revision:a4a219a47e8a8d1a4a2fe306ffa70d65ab4b23b8
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/88f3263876..2b015a25fa
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/ec8af4bd7b..01aef719ba
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/0aef453c8c..4338dc154d
* src/third_party/androidx/cipd: Y9ecsp20FCn4bBgTNBLDqMM8XRz45d17SEsZ6tqD8XsC..4fkujxcHswgZM_Abba388nGkoYd7fgfTF_0jCmWzodMC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/5774eca600..745d3ebf40
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/dd826edb42..eea4d31f58
* src/third_party/siso/cipd: git_revision:7495bf92ea44f7eed89273b5aea7bd72b702ed1c..git_revision:a4a219a47e8a8d1a4a2fe306ffa70d65ab4b23b8
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/baa30f6db6..1d12bf19da
DEPS diff: https://chromium.googlesource.com/chromium/src/+/bd3cb736a1..4b7d3263af/DEPS

BUG=None

Change-Id: Ibf26612b3870a19a156e3f29682d3e417ca4c893
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453722
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47054}
---
 DEPS | 18 +++++++++---------
 1 file changed, 9 insertions(+), 9 deletions(-)

diff --git a/DEPS b/DEPS
index 4f550500a1..f145aade35 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': 'bd3cb736a165e43ad71292b163a5fec194e6981e',
+  'chromium_revision': '4b7d3263af689b12659cb08ad78de8f6fe1b7642',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -50,7 +50,7 @@ vars = {
   # reclient CIPD package version
   'reclient_version': 're_client_version:0.185.0.db415f21-gomaip',
   # siso CIPD package version.
-  'siso_version': 'git_revision:7495bf92ea44f7eed89273b5aea7bd72b702ed1c',
+  'siso_version': 'git_revision:a4a219a47e8a8d1a4a2fe306ffa70d65ab4b23b8',
 
   # ninja CIPD package.
   'ninja_package': 'infra/3pp/tools/ninja/',
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@88f3263876611ef82b25349577c721e13bc402fa',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@2b015a25fa7e9b762b5863e8616fad4b0b92f093',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@ec8af4bd7b568f7d395097721597776495655bdd',
+    'https://chromium.googlesource.com/chromium/src/testing@01aef719bace483ebbf6cd8f1f1b96f65aa1a322',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@0aef453c8c27ade8985e3aeb50d2ff322d259471',
+    'https://chromium.googlesource.com/chromium/src/third_party@4338dc154dcdc373a388f92d98786171fdec72c8',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -488,7 +488,7 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@5774eca6004ed7c7467bd644c057797ca96b65f2',
+    'https://boringssl.googlesource.com/boringssl.git@745d3ebf4064a4ea7c0a115ecbba5affa2609bcd',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@8be0e3114685fcc1589561067282edf75ea1259a',
   'src/third_party/catapult':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@baa30f6db60e23ac66e93dba5062cbe590540612',
+    'https://chromium.googlesource.com/chromium/src/tools@1d12bf19dae8ead34b3a328c79e5af5b23416dfa',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'Y9ecsp20FCn4bBgTNBLDqMM8XRz45d17SEsZ6tqD8XsC',
+          'version': '4fkujxcHswgZM_Abba388nGkoYd7fgfTF_0jCmWzodMC',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -864,7 +864,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7364b490b5f78d58efe23ea76e74210fd6c3c76f',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + 'dd826edb42520814dab9239ec81f32ab91a4afe4',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + 'eea4d31f58572591bdaa3b654227e48efb495b96',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From 1a631fec6c8438b207c0a7206acf22e7cfc2fee9 Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Wed, 4 Mar 2026 20:07:01 -0800
Subject: [PATCH 202/472] Update WebRTC code version (2026-03-05T04:06:26).

Bug: None
Change-Id: I7f1d9534e9f8fd136c0caf4d000fb1c4992fd59d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453761
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47055}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index f5540b9e92..78f8f0461b 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-04T04:04:09";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-05T04:06:26";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From a83408136758a6350d53893c687e68c51db75ed9 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Tue, 3 Mar 2026 15:16:11 +0100
Subject: [PATCH 203/472] Implement cooperative yielding and high-priority
 tasks in Thread

Introduce mechanisms within the Thread class to support cooperative
multitasking. This allows long-running tasks to voluntarily yield
execution to pending high-priority work, improving responsiveness and
mitigating latency issues like audio glitches that can occur during
heavy operations such as session negotiation.

Key modifications:
- Add IsYieldRequested() to allow tasks to check for pending yield
  requests.
- Implement PostHighPriorityTask() to insert urgent tasks into a
  dedicated queue and signal a yield request on the target thread.
- Track nested synchronous blocking calls to catch when yielding in
  unsafe contexts, enforced via RTC_DCHECK.
- Ensure high-priority tasks are processed before regular tasks when the
  thread resumes execution.

# no-try: persistent webrtc_linux_chromium infra failure

No-try: true
Bug: webrtc:42223005
Change-Id: I093b7e04d665baef19085e72bc0548447fd6768c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/450561
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47056}
---
 api/task_queue/task_queue_test.cc |  27 ++++++
 rtc_base/BUILD.gn                 |   1 +
 rtc_base/thread.cc                |  48 ++++++++--
 rtc_base/thread.h                 |  29 +++++-
 rtc_base/thread_unittest.cc       | 151 ++++++++++++++++++++++++++++++
 5 files changed, 249 insertions(+), 7 deletions(-)

diff --git a/api/task_queue/task_queue_test.cc b/api/task_queue/task_queue_test.cc
index cf9ff886df..ba07e39f9f 100644
--- a/api/task_queue/task_queue_test.cc
+++ b/api/task_queue/task_queue_test.cc
@@ -15,6 +15,7 @@
 
 #include <cstdint>
 #include <memory>
+#include <string>
 #include <utility>
 #include <vector>
 
@@ -38,11 +39,14 @@
 #include "rtc_base/event.h"
 #include "rtc_base/ref_counter.h"
 #include "rtc_base/time_utils.h"
+#include "test/gmock.h"
 #include "test/gtest.h"
 
 namespace webrtc {
 namespace {
 
+using ::testing::ElementsAre;
+
 // Avoids a dependency to system_wrappers.
 void SleepFor(TimeDelta duration) {
   ScopedAllowBaseSyncPrimitivesForTesting allow;
@@ -81,6 +85,29 @@ TEST_P(TaskQueueTest, PostAndCheckCurrent) {
   EXPECT_TRUE(event.Wait(TimeDelta::Seconds(1)));
 }
 
+// Verifies that a task can post a new task from within the task
+// and that the new one eventually runs.
+TEST_P(TaskQueueTest, TaskCanPostContinuationTask) {
+  std::unique_ptr<TaskQueueFactory> factory = GetParam()(nullptr);
+  auto queue = CreateTaskQueue(factory, "TaskCanPostContinuationTask");
+
+  std::vector<std::string> events;
+  Event done;
+
+  queue->PostTask([&events, &done, queue = queue.get()] {
+    events.push_back("Start");
+    queue->PostTask([&events, &done] {
+      events.push_back("Continue");
+      done.Set();
+    });
+    events.push_back("FirstDone");
+  });
+
+  EXPECT_TRUE(done.Wait(TimeDelta::Seconds(1)));
+
+  EXPECT_THAT(events, ElementsAre("Start", "FirstDone", "Continue"));
+}
+
 TEST_P(TaskQueueTest, PostCustomTask) {
   std::unique_ptr<TaskQueueFactory> factory = GetParam()(nullptr);
   Event ran;
diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
index 7e1bcc362c..b589a35ec1 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -2380,6 +2380,7 @@ if (rtc_include_tests) {
         "../test:create_test_field_trials",
         "../test:fileutils",
         "../test:rtc_expect_death",
+        "../test:run_loop",
         "../test:test_support",
         "../test:wait_until",
         "memory:fifo_buffer",
diff --git a/rtc_base/thread.cc b/rtc_base/thread.cc
index 4d11e7af9c..b3c541ffa8 100644
--- a/rtc_base/thread.cc
+++ b/rtc_base/thread.cc
@@ -17,6 +17,7 @@
 #include <cstdint>
 #include <deque>
 #include <memory>
+#include <queue>
 #include <string>
 #include <vector>
 
@@ -90,7 +91,6 @@ class ScopedAutoReleasePool {
 
 namespace webrtc {
 
-
 ThreadManager* ThreadManager::Instance() {
   static ThreadManager* const thread_manager = new ThreadManager();
   return thread_manager;
@@ -389,7 +389,8 @@ void Thread::DoDestroy() {
   ThreadManager::Remove(this);
   // Clear.
   CurrentTaskQueueSetter set_current(this);
-  messages_ = {};
+  high_priority_messages_.clear();
+  messages_.clear();
   delayed_messages_ = {};
 }
 
@@ -436,13 +437,20 @@ absl::AnyInvocable<void() &&> Thread::Get(int cmsWait) {
               TimeDiff(delayed_messages_.top().run_time_ms, msCurrent);
           break;
         }
-        messages_.push(std::move(delayed_messages_.top().functor));
+        messages_.push_back(std::move(delayed_messages_.top().functor));
         delayed_messages_.pop();
       }
+      // Pull a high priority message off the queue, if available.
+      if (!high_priority_messages_.empty()) {
+        absl::AnyInvocable<void() &&> task =
+            std::move(high_priority_messages_.front());
+        high_priority_messages_.pop_front();
+        return task;
+      }
       // Pull a message off the message queue, if available.
       if (!messages_.empty()) {
         absl::AnyInvocable<void() &&> task = std::move(messages_.front());
-        messages_.pop();
+        messages_.pop_front();
         return task;
       }
     }
@@ -484,6 +492,15 @@ absl::AnyInvocable<void() &&> Thread::Get(int cmsWait) {
 void Thread::PostTaskImpl(absl::AnyInvocable<void() &&> task,
                           const PostTaskTraits& /* traits */,
                           const Location& /* location */) {
+  PostTaskInternal(std::move(task), /*high_priority=*/false);
+}
+
+void Thread::PostHighPriorityTask(absl::AnyInvocable<void() &&> task) {
+  PostTaskInternal(std::move(task), /*high_priority=*/true);
+}
+
+void Thread::PostTaskInternal(absl::AnyInvocable<void() &&> task,
+                              bool high_priority) {
   if (IsQuitting()) {
     return;
   }
@@ -494,7 +511,11 @@ void Thread::PostTaskImpl(absl::AnyInvocable<void() &&> task,
 
   {
     MutexLock lock(&mutex_);
-    messages_.push(std::move(task));
+    if (high_priority) {
+      high_priority_messages_.push_back(std::move(task));
+    } else {
+      messages_.push_back(std::move(task));
+    }
     WakeUpSocketServer();
   }
 }
@@ -531,7 +552,7 @@ void Thread::PostDelayedTaskImpl(absl::AnyInvocable<void() &&> task,
 int Thread::GetDelay() {
   MutexLock lock(&mutex_);
 
-  if (!messages_.empty())
+  if (!high_priority_messages_.empty() || !messages_.empty())
     return 0;
 
   if (!delayed_messages_.empty()) {
@@ -762,8 +783,12 @@ void Thread::BlockingCallImpl(FunctionView<void()> functor,
     RTC_DCHECK(this->IsInvokeToThreadAllowed(this));
     RTC_DCHECK_RUN_ON(this);
     could_be_blocking_call_count_++;
+    ++running_synchronous_blocking_call_count_;
 #endif
     functor();
+#if RTC_DCHECK_IS_ON
+    --running_synchronous_blocking_call_count_;
+#endif
     return;
   }
 
@@ -963,4 +988,15 @@ AutoSocketServerThread::~AutoSocketServerThread() {
   }
 }
 
+bool Thread::IsYieldRequested() const {
+  RTC_DCHECK_RUN_ON(this);
+#if RTC_DCHECK_IS_ON
+  // If you've hit this, then there's a cooperative task running from inside a
+  // blocking call.
+  RTC_DCHECK_EQ(running_synchronous_blocking_call_count_, 0);
+#endif
+  MutexLock lock(&mutex_);
+  return !high_priority_messages_.empty();
+}
+
 }  // namespace webrtc
diff --git a/rtc_base/thread.h b/rtc_base/thread.h
index feb8cf78c7..0629461366 100644
--- a/rtc_base/thread.h
+++ b/rtc_base/thread.h
@@ -14,6 +14,7 @@
 #include <atomic>
 #include <cstddef>
 #include <cstdint>
+#include <deque>
 #include <map>
 #include <memory>
 #include <queue>
@@ -316,6 +317,12 @@ class RTC_LOCKABLE RTC_EXPORT Thread : public TaskQueueBase {
   // ProcessMessages occasionally.
   virtual void Run();
 
+  // Called from cooperative tasks to know if they should yield.
+  // If a task yields, it is up to the task itself how or if to
+  // continue the ongoing operation. Typically this can be handled
+  // by using PostTask() to queue up a continuation task.
+  bool IsYieldRequested() const;
+
   // Convenience method to invoke a functor on another thread.
   // Blocks the current thread until execution is complete.
   // Ex: thread.BlockingCall([&] { result = MyFunctionReturningBool(); });
@@ -338,6 +345,12 @@ class RTC_LOCKABLE RTC_EXPORT Thread : public TaskQueueBase {
     return result;
   }
 
+  // Posts a task that jumps the queue (gets added as the next task to execute,
+  // ahead of other pending tasks). An internal yielding request flag is also
+  // raised that cooperative aware tasks can check by calling
+  // `IsYieldRequested()` and immediately exit.
+  void PostHighPriorityTask(absl::AnyInvocable<void() &&> task);
+
   // Allows BlockingCall to specified `thread`. Thread never will be
   // dereferenced and will be used only for reference-based comparison, so
   // instance can be safely deleted. If NDEBUG is defined and RTC_DCHECK_IS_ON
@@ -463,6 +476,8 @@ class RTC_LOCKABLE RTC_EXPORT Thread : public TaskQueueBase {
  private:
   static const int kSlowDispatchLoggingThreshold = 50;  // 50 ms
 
+  void PostTaskInternal(absl::AnyInvocable<void() &&> task, bool high_priority);
+
   // Get() will process I/O until:
   //  1) A task is available (returns it)
   //  2) cmsWait seconds have elapsed (returns empty task)
@@ -497,7 +512,9 @@ class RTC_LOCKABLE RTC_EXPORT Thread : public TaskQueueBase {
   // Called by the ThreadManager when being unset as the current thread.
   void ClearCurrentTaskQueue();
 
-  std::queue<absl::AnyInvocable<void() &&>> messages_ RTC_GUARDED_BY(mutex_);
+  std::deque<absl::AnyInvocable<void() &&>> messages_ RTC_GUARDED_BY(mutex_);
+  std::deque<absl::AnyInvocable<void() &&>> high_priority_messages_
+      RTC_GUARDED_BY(mutex_);
   std::priority_queue<DelayedMessage> delayed_messages_ RTC_GUARDED_BY(mutex_);
   uint32_t delayed_next_num_ RTC_GUARDED_BY(mutex_);
 #if RTC_DCHECK_IS_ON
@@ -545,6 +562,16 @@ class RTC_LOCKABLE RTC_EXPORT Thread : public TaskQueueBase {
   friend class ThreadManager;
 
   int dispatch_warning_ms_ RTC_GUARDED_BY(this) = kSlowDispatchLoggingThreshold;
+
+#if RTC_DCHECK_IS_ON
+  // This is used to catch if a cooperative task ends up being called from
+  // within another task. If that happens, the risk is that a full yield won't
+  // actually happen, so this is to help with ensuring we catch when things
+  // don't run as expected since webrtc can be configured in many ways and
+  // sometimes virtual thread concepts such as worker and network threads, can
+  // map to the same thread object.
+  int running_synchronous_blocking_call_count_ RTC_GUARDED_BY(this) = 0;
+#endif
 };
 
 // AutoThread automatically installs itself at construction
diff --git a/rtc_base/thread_unittest.cc b/rtc_base/thread_unittest.cc
index 3fe9eda6d8..97ed808817 100644
--- a/rtc_base/thread_unittest.cc
+++ b/rtc_base/thread_unittest.cc
@@ -15,6 +15,7 @@
 #include <cstddef>
 #include <cstdint>
 #include <memory>
+#include <string>
 #include <utility>
 #include <vector>
 
@@ -44,6 +45,7 @@
 #include "test/create_test_environment.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 #include "test/testsupport/rtc_expect_death.h"
 #include "test/wait_until.h"
 
@@ -871,6 +873,155 @@ TEST(ThreadPostDelayedTaskTest, IsCurrentTaskQueue) {
   EXPECT_EQ(TaskQueueBase::Current(), current_tq);
 }
 
+// Verifies that posting a high-priority task sets the "yield requested" flag
+// on the target thread, allowing a cooperative task to observe it.
+TEST(ThreadCooperativeTest, HighPriorityTaskTriggersYieldRequest) {
+  test::RunLoop loop;
+  auto thread = Thread::Create();
+  thread->Start();
+
+  bool yield_requested = false;
+  Event task_started;
+
+  // Post a long running task that checks for yield
+  thread->PostTask(
+      [&yield_requested, &loop, &task_started, thread = thread.get()] {
+        task_started.Set();
+        while (!thread->IsYieldRequested()) {
+          // Busy loop/simulated work
+        }
+        loop.PostTask([&yield_requested, &loop] {
+          yield_requested = true;
+          loop.Quit();
+        });
+      });
+
+  // Wait for the task to start to ensure that the high priority task doesn't
+  // run first.
+  task_started.Wait(Event::kForever);
+
+  // Post a high priority task that interrupts the busy loop.
+  thread->PostHighPriorityTask([] {});
+
+  loop.Run();
+  EXPECT_TRUE(yield_requested);
+}
+
+// Verifies that a high-priority task is executed before a regular priority task
+// that was posted earlier, if the thread was busy.
+TEST(ThreadCooperativeTest, HighPriorityTaskRunsFirst) {
+  test::RunLoop loop;
+  auto thread = Thread::Create();
+  thread->Start();
+
+  std::vector<std::string> execution_order;
+  Event task1_started;
+  Event continue_execution;
+
+  // Task 1: Blocks the thread until signaled.
+  thread->PostTask([&] {
+    task1_started.Set();
+    continue_execution.Wait(Event::kForever);
+    loop.PostTask([&] { execution_order.push_back("Task1"); });
+  });
+
+  task1_started.Wait(Event::kForever);
+
+  // Task 2: Regular priority.
+  thread->PostTask(
+      [&] { loop.PostTask([&] { execution_order.push_back("Task2"); }); });
+
+  // Task 3: High priority. High priority tasks should run before "Task2".
+  thread->PostHighPriorityTask(
+      [&] { loop.PostTask([&] { execution_order.push_back("Task3"); }); });
+
+  // Signal Task 1 to continue.
+  continue_execution.Set();
+
+  // Now post a task to quit the loop.
+  thread->PostTask([&] { loop.PostTask([&] { loop.Quit(); }); });
+
+  loop.Run();
+
+  // Expect Task1 (resuming), Task3 (High Prio), Task2 (Low Prio - pushed to
+  // back).
+  EXPECT_THAT(execution_order, ElementsAre("Task1", "Task3", "Task2"));
+}
+
+// Verifies that multiple high-priority tasks execute in the sequence they were
+// posted (FIFO order).
+TEST(ThreadCooperativeTest, HighPriorityTasksExecuteInSequence) {
+  test::RunLoop loop;
+  auto thread = Thread::Create();
+  thread->Start();
+
+  std::vector<std::string> execution_order;
+  Event task1_started;
+  Event continue_execution;
+
+  // Block the thread.
+  thread->PostTask([&] {
+    task1_started.Set();
+    continue_execution.Wait(Event::kForever);
+  });
+
+  task1_started.Wait(Event::kForever);
+
+  // Post a normal priority task first to ensure high priority tasks run before
+  // it.
+  thread->PostTask(
+      [&] { loop.PostTask([&] { execution_order.push_back("Normal1"); }); });
+
+  // Post multiple high priority tasks while the thread is blocked.
+  thread->PostHighPriorityTask(
+      [&] { loop.PostTask([&] { execution_order.push_back("HP1"); }); });
+  thread->PostHighPriorityTask(
+      [&] { loop.PostTask([&] { execution_order.push_back("HP2"); }); });
+  thread->PostHighPriorityTask(
+      [&] { loop.PostTask([&] { execution_order.push_back("HP3"); }); });
+
+  // Unblock the thread.
+  continue_execution.Set();
+
+  // Quit after all tasks.
+  thread->PostTask([&] { loop.PostTask([&] { loop.Quit(); }); });
+
+  loop.Run();
+
+  EXPECT_THAT(execution_order, ElementsAre("HP1", "HP2", "HP3", "Normal1"));
+}
+
+TEST(ThreadCooperativeTest, YieldRequestedClearedAfterHighPriorityTask) {
+  std::unique_ptr<Thread> thread(Thread::Create());
+  thread->Start();
+
+  // Initially false.
+  thread->BlockingCall([&] { EXPECT_FALSE(thread->IsYieldRequested()); });
+
+  // Post high priority task.
+  thread->PostHighPriorityTask([&] {});
+
+  // Post normal task.
+  thread->BlockingCall([&] { EXPECT_FALSE(thread->IsYieldRequested()); });
+}
+
+#if GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID) && RTC_DCHECK_IS_ON
+TEST(ThreadCooperativeDeathTest, YieldInSynchronousBlockingCallTriggersDeath) {
+  std::unique_ptr<Thread> background_thread = Thread::Create();
+  background_thread->Start();
+  background_thread->BlockingCall([&] {
+    // Yielding from a top-level synchronous blocking call is allowed.
+    background_thread->IsYieldRequested();
+
+    // Call a nested BlockingCall.
+    background_thread->BlockingCall([&] {
+      // This violates the "no yielding in blocking call" rule.
+      EXPECT_DEATH(background_thread->IsYieldRequested(), "");
+    });
+  });
+}
+#endif
+
 class ThreadFactory : public TaskQueueFactory {
  public:
   std::unique_ptr<TaskQueueBase, TaskQueueDeleter> CreateTaskQueue(

From 28dde62b904a7f39f2310f3d41f47dde2b8e53a7 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Thu, 5 Mar 2026 00:03:18 -0800
Subject: [PATCH 204/472] Roll chromium_revision 4b7d3263af..9d278b8335
 (1594384:1594505)

Change log: https://chromium.googlesource.com/chromium/src/+log/4b7d3263af..9d278b8335
Full diff: https://chromium.googlesource.com/chromium/src/+/4b7d3263af..9d278b8335

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/d863633554..51fd5e3b61
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/01aef719ba..7d23eff99a
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/4338dc154d..aa674c67b9
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/109c6d7bab..ab750b7590
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/2167887d78..a1e573faed
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/7ffb3786c6..4a1e0fb594
* src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/e8faec51e3..e490aa627e
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/1d12bf19da..1ddf8dac42
* src/tools/luci-go: git_revision:a61d4d67348b452a5e7bf09541f808e8c1101e76..git_revision:eefaa865c43930859313a63d426740cdd0e22673
* src/tools/luci-go: git_revision:a61d4d67348b452a5e7bf09541f808e8c1101e76..git_revision:eefaa865c43930859313a63d426740cdd0e22673
* src/tools/luci-go: git_revision:a61d4d67348b452a5e7bf09541f808e8c1101e76..git_revision:eefaa865c43930859313a63d426740cdd0e22673
DEPS diff: https://chromium.googlesource.com/chromium/src/+/4b7d3263af..9d278b8335/DEPS

BUG=None

Change-Id: I6efcb322102bfaf63d417b97af2c02b039bbc901
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453780
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47057}
---
 DEPS | 24 ++++++++++++------------
 1 file changed, 12 insertions(+), 12 deletions(-)

diff --git a/DEPS b/DEPS
index f145aade35..8ff1c56eff 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '4b7d3263af689b12659cb08ad78de8f6fe1b7642',
+  'chromium_revision': '9d278b833580deddd27a32636a6f56ccf5c17611',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@d863633554de2574eefb9a2f2b86d0bfa8d92850',
+    'https://chromium.googlesource.com/chromium/src/build@51fd5e3b61c86e0601a4803fabe0f1baf209260b',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@fef8d6947fa54202e8a00c58d5ac8fd15780bc33',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -82,9 +82,9 @@ deps = {
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@01aef719bace483ebbf6cd8f1f1b96f65aa1a322',
+    'https://chromium.googlesource.com/chromium/src/testing@7d23eff99a735cea4f99711d0ddde40e714af6df',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@4338dc154dcdc373a388f92d98786171fdec72c8',
+    'https://chromium.googlesource.com/chromium/src/third_party@aa674c67b9b66dc8c48d86d609f681140bbe11d7',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -337,7 +337,7 @@ deps = {
   'src/third_party/libc++abi/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@8f11bb1d4438d0239d0dfc1bd9456a9f31629dda',
   'src/third_party/llvm-libc/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@e8faec51e35d65209d5ba660d711893ad3790735',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@e490aa627ee45c36557b933ac659a8f04a26c758',
   'src/third_party/libunwind/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@78884e23fe39cf5cc6987ea188a9b802d65a21c9',
 
@@ -492,7 +492,7 @@ deps = {
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@8be0e3114685fcc1589561067282edf75ea1259a',
   'src/third_party/catapult':
-    'https://chromium.googlesource.com/catapult.git@109c6d7babfc9afbb19af677a0d21fe3a281f2c1',
+    'https://chromium.googlesource.com/catapult.git@ab750b7590e913a2dcf493fae9c93d0b649f5552',
   'src/third_party/ced/src': {
     'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5',
   },
@@ -505,13 +505,13 @@ deps = {
   'src/third_party/crc32c/src':
     'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558',
   'src/third_party/depot_tools':
-    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@2167887d7880c480873ca7d9a8621d89b72c7130',
+    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@a1e573faed3fd7e783fa340ff4af4e4193574d7b',
   'src/third_party/ffmpeg':
     'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@946d97db8d906277085e361892b7efda5152e2f1',
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@7ffb3786c67639e1300af11bcc51e61db087f954',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@4a1e0fb594588a81e11187d0c34507a22a141e42',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@1d12bf19dae8ead34b3a328c79e5af5b23416dfa',
+    'https://chromium.googlesource.com/chromium/src/tools@1ddf8dac421f5a8acdb162781e21723e597692fd',
 
   'src/third_party/espresso': {
       'packages': [
@@ -808,15 +808,15 @@ deps = {
       'packages': [
         {
           'package': 'infra/tools/luci/cas/${{platform}}',
-          'version': 'git_revision:a61d4d67348b452a5e7bf09541f808e8c1101e76',
+          'version': 'git_revision:eefaa865c43930859313a63d426740cdd0e22673',
         },
         {
           'package': 'infra/tools/luci/isolate/${{platform}}',
-          'version': 'git_revision:a61d4d67348b452a5e7bf09541f808e8c1101e76',
+          'version': 'git_revision:eefaa865c43930859313a63d426740cdd0e22673',
         },
         {
           'package': 'infra/tools/luci/swarming/${{platform}}',
-          'version': 'git_revision:a61d4d67348b452a5e7bf09541f808e8c1101e76',
+          'version': 'git_revision:eefaa865c43930859313a63d426740cdd0e22673',
         }
       ],
       'dep_type': 'cipd',

From 800efefa25ab4dac87575ef439b3c737ca21af9a Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Wed, 4 Mar 2026 18:06:29 +0100
Subject: [PATCH 205/472] Centralize RTP header extension parsing in
 RtpTransport

Centralize the management and parsing of RTP header extensions in
RtpTransport, move it out of the media channels
(WebRtcVideoReceiveChannel and WebRtcVoiceReceiveChannel). In this first
phase moving extension identification to the transport level, but
leaving in checks for consistency with the previous behavior in the
channel classes.

Previously, header extension maps were not consistently merged in BUNDLE
scenarios, potentially causing mismatches between the transport and
media layers. This update addresses the issue by introducing centralized
registration and unregistration of extension maps keyed by the MID.

Key modifications include:

* Implement RegisterRtpHeaderExtensionMap and
  UnregisterRtpHeaderExtensionMap in RtpTransport to maintain a merged
  view of extensions across all m-sections.
* Add runtime verification in video and voice receive channels to
  ensure local maps remain synchronized with the transport-level map.
* Ensure that the most recently registered extension IDs take
  precedence during BUNDLE negotiation to handle transient states per
  RFC 8843.

Bug: webrtc:42232224
Change-Id: Ifb3a1617f58d736a111f709b8c3c370a0b1c1da1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453160
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47058}
---
 media/engine/webrtc_video_engine.cc          | 31 ++++++++
 media/engine/webrtc_video_engine_unittest.cc |  4 +-
 media/engine/webrtc_voice_engine.cc          | 30 ++++++++
 media/engine/webrtc_voice_engine_unittest.cc |  2 +-
 pc/BUILD.gn                                  |  6 ++
 pc/channel.cc                                | 44 ++++++++---
 pc/rtp_transport.cc                          | 71 ++++++++++++++++-
 pc/rtp_transport.h                           | 33 +++++++-
 pc/rtp_transport_internal.h                  |  6 +-
 pc/rtp_transport_unittest.cc                 | 81 ++++++++++++++++++++
 10 files changed, 292 insertions(+), 16 deletions(-)

diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc
index 640abe6386..7ce882c7ad 100644
--- a/media/engine/webrtc_video_engine.cc
+++ b/media/engine/webrtc_video_engine.cc
@@ -3249,7 +3249,38 @@ void WebRtcVideoReceiveChannel::OnPacketReceived(RtpPacketReceived packet) {
   // it instead at the stream level initialized by the Transport. RTP packets
   // consumed by media channels would thus have header extensions distinguished
   // by C++ class rather than negotiated header extension ID.
+#if RTC_DCHECK_IS_ON
+  RtpPacketReceived original_packet = packet;
+#endif
   packet.IdentifyExtensions(recv_rtp_extension_map_);
+#if RTC_DCHECK_IS_ON
+  for (int i = kRtpExtensionNone + 1; i < kRtpExtensionNumberOfExtensions;
+       ++i) {
+    RTPExtensionType type = static_cast<RTPExtensionType>(i);
+    // TODO(bugs.webrtc.org/7135): Only verify if the extension
+    // is registered in either map to avoid unnecessary checks.
+    if (recv_rtp_extension_map_.IsRegistered(type)) {
+      // We only verify equality when the IDs match. This check is
+      // intentionally weak because the transport map (from `original_packet`)
+      // contains the merged extensions for all MIDs in a BUNDLE group,
+      // while the local map only contains extensions for this specific MID.
+      // Therefore, the transport map might contain extensions not present
+      // in the local map, but if both maps contain an ID for a specific
+      // extension type, they must both parse it consistently.
+      if (original_packet.extension_manager().GetId(type) ==
+          recv_rtp_extension_map_.GetId(type)) {
+        bool transport_has = original_packet.HasExtension(type);
+        bool local_has = packet.HasExtension(type);
+        RTC_DCHECK_EQ(transport_has, local_has)
+            << "Extension mapping mismatch for type " << type << ". "
+            << "Transport map has: " << transport_has << " (ID from transport: "
+            << static_cast<int>(original_packet.extension_manager().GetId(type))
+            << "). Local map has: " << local_has << " (ID from local: "
+            << static_cast<int>(recv_rtp_extension_map_.GetId(type)) << ").";
+      }
+    }
+  }
+#endif
   packet.set_payload_type_frequency(kVideoPayloadTypeFrequency);
   if (!packet.arrival_time().IsFinite()) {
     packet.set_arrival_time(env_.clock().CurrentTime());
diff --git a/media/engine/webrtc_video_engine_unittest.cc b/media/engine/webrtc_video_engine_unittest.cc
index d208e45521..03f25b5a4f 100644
--- a/media/engine/webrtc_video_engine_unittest.cc
+++ b/media/engine/webrtc_video_engine_unittest.cc
@@ -3417,7 +3417,9 @@ TEST_F(WebRtcVideoChannelTest, OnPacketReceivedIdentifiesExtensions) {
   reference_packet.SetExtension<VideoOrientation>(
       VideoRotation::kVideoRotation_270);
   // Create a packet without the extension map but with the same content.
-  RtpPacketReceived received_packet;
+  // We need to parse it using the extension map to identify extensions properly
+  // since RtpTransport is effectively skipped in this test.
+  RtpPacketReceived received_packet(&extension_map);
   ASSERT_TRUE(received_packet.Parse(reference_packet.Buffer()));
 
   receive_channel_->OnPacketReceived(received_packet);
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index dd04d4ec7a..857f6dbef9 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -85,6 +85,7 @@
 #include "modules/async_audio_processing/async_audio_processing.h"
 #include "modules/audio_mixer/audio_mixer_impl.h"
 #include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/dscp.h"
@@ -2620,7 +2621,36 @@ void WebRtcVoiceReceiveChannel::OnPacketReceived(RtpPacketReceived packet) {
   // it instead at the stream level initialized by the Transport. RTP packets
   // consumed by media channels would thus have header extensions distinguished
   // by C++ class rather than negotiated header extension ID.
+#if RTC_DCHECK_IS_ON
+  RtpPacketReceived original_packet = packet;
+#endif
   packet.IdentifyExtensions(recv_rtp_extension_map_);
+#if RTC_DCHECK_IS_ON
+  for (int i = kRtpExtensionNone + 1; i < kRtpExtensionNumberOfExtensions;
+       ++i) {
+    RTPExtensionType type = static_cast<RTPExtensionType>(i);
+    if (recv_rtp_extension_map_.IsRegistered(type)) {
+      // We only verify equality when the IDs match. This check is
+      // intentionally weak because the transport map (from `original_packet`)
+      // contains the merged extensions for all MIDs in a BUNDLE group,
+      // while the local map only contains extensions for this specific MID.
+      // Therefore, the transport map might contain extensions not present
+      // in the local map, but if both maps agree on an ID for a specific
+      // extension type, they must both parse it consistently.
+      if (original_packet.extension_manager().GetId(type) ==
+          recv_rtp_extension_map_.GetId(type)) {
+        bool transport_has = original_packet.HasExtension(type);
+        bool local_has = packet.HasExtension(type);
+        RTC_DCHECK_EQ(transport_has, local_has)
+            << "Extension mapping mismatch for type " << type << ". "
+            << "Transport map has: " << transport_has << " (ID from transport: "
+            << static_cast<int>(original_packet.extension_manager().GetId(type))
+            << "). Local map has: " << local_has << " (ID from local: "
+            << static_cast<int>(recv_rtp_extension_map_.GetId(type)) << ").";
+      }
+    }
+  }
+#endif
   if (!packet.arrival_time().IsFinite()) {
     packet.set_arrival_time(env_.clock().CurrentTime());
   }
diff --git a/media/engine/webrtc_voice_engine_unittest.cc b/media/engine/webrtc_voice_engine_unittest.cc
index 8329713ca1..831ca66c92 100644
--- a/media/engine/webrtc_voice_engine_unittest.cc
+++ b/media/engine/webrtc_voice_engine_unittest.cc
@@ -1605,7 +1605,7 @@ TEST_P(WebRtcVoiceEngineTestFake, OnPacketReceivedIdentifiesExtensions) {
   reference_packet.SetExtension<AudioLevelExtension>(
       AudioLevel(/*voice_activity=*/true, kAudioLevel));
   //  Create a packet without the extension map but with the same content.
-  RtpPacketReceived received_packet;
+  RtpPacketReceived received_packet(&extension_map);
   ASSERT_TRUE(received_packet.Parse(reference_packet.Buffer()));
 
   receive_channel_->OnPacketReceived(received_packet);
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 9cd20428be..2a641da2c2 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -585,6 +585,7 @@ rtc_library("rtp_transport") {
     ":session_description",
     "../api:array_view",
     "../api:field_trials_view",
+    "../api:sequence_checker",
     "../api/task_queue",
     "../api/task_queue:pending_task_safety_flag",
     "../api/transport:ecn_marking",
@@ -599,11 +600,15 @@ rtc_library("rtp_transport") {
     "../rtc_base:copy_on_write_buffer",
     "../rtc_base:event_tracer",
     "../rtc_base:logging",
+    "../rtc_base:macromagic",
     "../rtc_base:network_route",
     "../rtc_base:socket",
+    "../rtc_base/containers:flat_map",
     "../rtc_base/containers:flat_set",
     "../rtc_base/network:received_packet",
     "../rtc_base/network:sent_packet",
+    "../rtc_base/system:no_unique_address",
+    "//third_party/abseil-cpp/absl/strings:string_view",
   ]
 }
 
@@ -626,6 +631,7 @@ rtc_source_set("rtp_transport_internal") {
     "../rtc_base:socket",
     "../rtc_base/network:sent_packet",
     "//third_party/abseil-cpp/absl/functional:any_invocable",
+    "//third_party/abseil-cpp/absl/strings:string_view",
   ]
 }
 
diff --git a/pc/channel.cc b/pc/channel.cc
index 5fed0b364e..0cea1a549e 100644
--- a/pc/channel.cc
+++ b/pc/channel.cc
@@ -576,11 +576,9 @@ bool BaseChannel::MaybeUpdateDemuxerAndRtpExtensions_w(
       return false;
     }
 
-    // NOTE: This doesn't take the BUNDLE case in account meaning the RTP header
-    // extension maps are not merged when BUNDLE is enabled. This is fine
-    // because the ID for MID should be consistent among all the RTP transports.
-    if (extensions)
-      rtp_transport_->UpdateRtpHeaderExtensionMap(*extensions);
+    if (extensions) {
+      rtp_transport_->RegisterRtpHeaderExtensionMap(mid(), *extensions);
+    }
 
     if (!update_demuxer)
       return true;
@@ -1034,7 +1032,7 @@ bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content,
   bool success = MaybeUpdateDemuxerAndRtpExtensions_w(
       criteria_modified,
       update_header_extensions
-          ? std::optional<RtpHeaderExtensions>(std::move(header_extensions))
+          ? std::optional<RtpHeaderExtensions>(recv_params.extensions)
           : std::nullopt,
       error_desc);
 
@@ -1089,7 +1087,21 @@ bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content,
       media_send_channel()->SenderNonSenderRttEnabled());
   last_send_params_ = send_params;
 
-  return UpdateRemoteStreams_w(content, type, error_desc);
+  if (!UpdateRemoteStreams_w(content, type, error_desc)) {
+    return false;
+  }
+
+  if (type == SdpType::kAnswer || type == SdpType::kPrAnswer) {
+    bool success = MaybeUpdateDemuxerAndRtpExtensions_w(
+        /*update_demuxer=*/false,
+        std::optional<RtpHeaderExtensions>(last_recv_params_.extensions),
+        error_desc);
+    if (!success) {
+      return false;
+    }
+  }
+
+  return true;
 }
 
 VideoChannel::VideoChannel(
@@ -1209,7 +1221,7 @@ bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
   bool success = MaybeUpdateDemuxerAndRtpExtensions_w(
       criteria_modified,
       update_header_extensions
-          ? std::optional<RtpHeaderExtensions>(std::move(header_extensions))
+          ? std::optional<RtpHeaderExtensions>(recv_params.extensions)
           : std::nullopt,
       error_desc);
 
@@ -1265,7 +1277,21 @@ bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content,
   }
   last_send_params_ = send_params;
 
-  return UpdateRemoteStreams_w(content, type, error_desc);
+  if (!UpdateRemoteStreams_w(content, type, error_desc)) {
+    return false;
+  }
+
+  if (type == SdpType::kAnswer || type == SdpType::kPrAnswer) {
+    bool success = MaybeUpdateDemuxerAndRtpExtensions_w(
+        /*update_demuxer=*/false,
+        std::optional<RtpHeaderExtensions>(last_recv_params_.extensions),
+        error_desc);
+    if (!success) {
+      return false;
+    }
+  }
+
+  return true;
 }
 
 }  // namespace webrtc
diff --git a/pc/rtp_transport.cc b/pc/rtp_transport.cc
index d49678c7ba..5dcea00104 100644
--- a/pc/rtp_transport.cc
+++ b/pc/rtp_transport.cc
@@ -10,13 +10,17 @@
 
 #include "pc/rtp_transport.h"
 
+#include <algorithm>
 #include <cstdint>
 #include <memory>
 #include <optional>
 #include <string>
 #include <utility>
+#include <vector>
 
+#include "absl/strings/string_view.h"
 #include "api/array_view.h"
+#include "api/sequence_checker.h"
 #include "api/task_queue/pending_task_safety_flag.h"
 #include "api/task_queue/task_queue_base.h"
 #include "api/transport/ecn_marking.h"
@@ -24,6 +28,7 @@
 #include "call/rtp_demuxer.h"
 #include "media/base/rtp_utils.h"
 #include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "p2p/base/packet_transport_internal.h"
 #include "pc/session_description.h"
@@ -39,6 +44,19 @@
 #include "rtc_base/trace_event.h"
 
 namespace webrtc {
+namespace {
+
+void RemoveExtensionMapForMid(
+    absl::string_view mid,
+    std::vector<std::pair<std::string, RtpHeaderExtensions>>& extensions) {
+  auto it = std::find_if(extensions.begin(), extensions.end(),
+                         [mid](const auto& kv) { return kv.first == mid; });
+  if (it != extensions.end()) {
+    extensions.erase(it);
+  }
+}
+
+}  // namespace
 
 void RtpTransport::SetRtcpMuxEnabled(bool enable) {
   rtcp_mux_enabled_ = enable;
@@ -185,9 +203,58 @@ bool RtpTransport::SendPacket(bool rtcp,
   return true;
 }
 
-void RtpTransport::UpdateRtpHeaderExtensionMap(
+void RtpTransport::RegisterRtpHeaderExtensionMap(
+    absl::string_view mid,
     const RtpHeaderExtensions& header_extensions) {
-  header_extension_map_ = RtpHeaderExtensionMap(header_extensions);
+  RTC_DCHECK_RUN_ON(&network_thread_checker_);
+  RemoveExtensionMapForMid(mid, header_extensions_by_mid_);
+  header_extensions_by_mid_.emplace_back(std::string(mid), header_extensions);
+
+  RebuildMergedMap();
+}
+
+void RtpTransport::UnregisterRtpHeaderExtensionMap(absl::string_view mid) {
+  RTC_DCHECK_RUN_ON(&network_thread_checker_);
+  RemoveExtensionMapForMid(mid, header_extensions_by_mid_);
+
+  RebuildMergedMap();
+}
+
+void RtpTransport::RebuildMergedMap() {
+  RTC_DCHECK_RUN_ON(&network_thread_checker_);
+  RtpHeaderExtensionMap merged_map;
+
+  // RFC 8843 (BUNDLE) Section 7.1.3 requires that the same local identifier
+  // MUST be used for a given RTP header extension across all m-sections in a
+  // BUNDLE group.
+  //
+  // However, during negotiation, we may encounter transient states with
+  // conflicting IDs. This can occur with buggy endpoints or during "forked"
+  // signaling (e.g., an Offer followed by a PR-Answer from one endpoint,
+  // then a final Answer from a different endpoint). While most browsers
+  // send identical extension sets, different endpoints ringing simultaneously
+  // could theoretically provide differing maps.
+  //
+  // To handle this gracefully, we merge the maps. Because
+  // `header_extensions_by_mid_` preserves registration order, we iterate in
+  // reverse (newest first). This ensures tie-breaking is deterministic:
+  // more recently registered or updated MIDs (like those in a final Answer)
+  // take precedence over older or provisional ones.
+
+  for (auto rit = header_extensions_by_mid_.rbegin();
+       rit != header_extensions_by_mid_.rend(); ++rit) {
+    for (const auto& extension : rit->second) {
+      if (extension.id == RtpHeaderExtensionMap::kInvalidId) {
+        continue;
+      }
+      // Only register if the ID is not already in use.
+      RTPExtensionType type = merged_map.GetType(extension.id);
+      if (type == kRtpExtensionNone) {
+        merged_map.RegisterByUri(extension.id, extension.uri);
+      }
+    }
+  }
+  header_extension_map_ = std::move(merged_map);
 }
 
 bool RtpTransport::RegisterRtpDemuxerSink(const RtpDemuxerCriteria& criteria,
diff --git a/pc/rtp_transport.h b/pc/rtp_transport.h
index 2ea5b00339..4e1fd93ec3 100644
--- a/pc/rtp_transport.h
+++ b/pc/rtp_transport.h
@@ -17,8 +17,12 @@
 #include <memory>
 #include <optional>
 #include <string>
+#include <utility>
+#include <vector>
 
+#include "absl/strings/string_view.h"
 #include "api/field_trials_view.h"
+#include "api/sequence_checker.h"
 #include "api/task_queue/pending_task_safety_flag.h"
 #include "api/transport/ecn_marking.h"
 #include "api/units/timestamp.h"
@@ -35,6 +39,8 @@
 #include "rtc_base/network/sent_packet.h"
 #include "rtc_base/network_route.h"
 #include "rtc_base/socket.h"
+#include "rtc_base/system/no_unique_address.h"
+#include "rtc_base/thread_annotations.h"
 
 namespace webrtc {
 
@@ -86,9 +92,17 @@ class RtpTransport : public RtpTransportInternal {
 
   bool IsSrtpActive() const override { return false; }
 
-  void UpdateRtpHeaderExtensionMap(
+  void RegisterRtpHeaderExtensionMap(
+      absl::string_view mid,
       const RtpHeaderExtensions& header_extensions) override;
 
+  // Currently only used for testing. In production, unregistration isn't needed
+  // because leaving the registered extensions in `RtpTransport` is harmless
+  // when a channel/transceiver is stopped or disconnected. The negotiated
+  // extension IDs are typically stable for the lifetime of the transport, and
+  // the transport itself will be destroyed when the PeerConnection is closed.
+  void UnregisterRtpHeaderExtensionMap(absl::string_view mid) override;
+
   bool RegisterRtpDemuxerSink(const RtpDemuxerCriteria& criteria,
                               RtpPacketSinkInterface* sink) override;
 
@@ -131,6 +145,8 @@ class RtpTransport : public RtpTransportInternal {
 
   bool IsTransportWritable();
 
+  void RebuildMergedMap() RTC_RUN_ON(network_thread_checker_);
+
   bool rtcp_mux_enabled_;
 
   PacketTransportInternal* rtp_packet_transport_ = nullptr;
@@ -146,9 +162,22 @@ class RtpTransport : public RtpTransportInternal {
   RtpDemuxer rtp_demuxer_;
 
   // Used for identifying the MID for RtpDemuxer.
-  RtpHeaderExtensionMap header_extension_map_;
+  RtpHeaderExtensionMap header_extension_map_
+      RTC_GUARDED_BY(network_thread_checker_);
+  // Stores the registered RTP header extensions by MID.
+  // We use a std::vector to preserve the chronological registration order.
+  // In BUNDLE scenarios, RFC 8843 requires consistent extension IDs across
+  // all MIDs. If different MIDs request the same ID for different URIs,
+  // we resolve the conflict by giving precedence to the most recently
+  // registered MID. Preserving the insertion order allows
+  // UnregisterRtpHeaderExtensionMap to correctly fall back to the newest
+  // remaining registration when rebuilding the map.
+  std::vector<std::pair<std::string, RtpHeaderExtensions>>
+      header_extensions_by_mid_ RTC_GUARDED_BY(network_thread_checker_);
+
   // Guard against recursive "ready to send" signals
   bool processing_ready_to_send_ = false;
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker network_thread_checker_;
   ScopedTaskSafety safety_;
 };
 
diff --git a/pc/rtp_transport_internal.h b/pc/rtp_transport_internal.h
index a6df696f31..a2c9967545 100644
--- a/pc/rtp_transport_internal.h
+++ b/pc/rtp_transport_internal.h
@@ -16,6 +16,7 @@
 #include <utility>
 
 #include "absl/functional/any_invocable.h"
+#include "absl/strings/string_view.h"
 #include "api/task_queue/pending_task_safety_flag.h"
 #include "api/transport/ecn_marking.h"
 #include "api/units/timestamp.h"
@@ -133,9 +134,12 @@ class RtpTransportInternal {
   //   UpdateSendEncryptedHeaderExtensionIds,
   //   UpdateRecvEncryptedHeaderExtensionIds,
   //   CacheRtpAbsSendTimeHeaderExtension,
-  virtual void UpdateRtpHeaderExtensionMap(
+  virtual void RegisterRtpHeaderExtensionMap(
+      absl::string_view mid,
       const RtpHeaderExtensions& header_extensions) = 0;
 
+  virtual void UnregisterRtpHeaderExtensionMap(absl::string_view mid) = 0;
+
   virtual bool IsSrtpActive() const = 0;
 
   virtual bool RegisterRtpDemuxerSink(const RtpDemuxerCriteria& criteria,
diff --git a/pc/rtp_transport_unittest.cc b/pc/rtp_transport_unittest.cc
index 4b1b4099d6..4c2aca61af 100644
--- a/pc/rtp_transport_unittest.cc
+++ b/pc/rtp_transport_unittest.cc
@@ -14,17 +14,21 @@
 #include <cstdint>
 #include <optional>
 
+#include "api/rtp_parameters.h"
 #include "api/test/rtc_error_matchers.h"
 #include "api/transport/ecn_marking.h"
 #include "api/units/time_delta.h"
 #include "call/rtp_demuxer.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "p2p/base/packet_transport_internal.h"
 #include "p2p/test/fake_packet_transport.h"
+#include "pc/session_description.h"
 #include "pc/test/rtp_transport_test_util.h"
 #include "rtc_base/async_packet_socket.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/containers/flat_set.h"
 #include "rtc_base/copy_on_write_buffer.h"
+#include "rtc_base/logging.h"
 #include "rtc_base/network/sent_packet.h"
 #include "rtc_base/network_route.h"
 #include "rtc_base/thread.h"
@@ -284,6 +288,83 @@ TEST(RtpTransportTest, ChangingReadyToSendStateOnlySignalsWhenChanged) {
   EXPECT_EQ(observer.ready_to_send_signal_count(), 2);
 }
 
+TEST(RtpTransportTest, RegisterAndUnregisterRtpHeaderExtensionMap) {
+  AutoThread thread;
+  RtpTransport transport(kMuxDisabled, CreateTestFieldTrials());
+  RtpHeaderExtensions extensions1 = {
+      RtpExtension("urn:ietf:params:rtp-hdrext:ssrc-audio-level", 1)};
+  RtpHeaderExtensions extensions2 = {
+      RtpExtension("urn:ietf:params:rtp-hdrext:ssrc-audio-level", 1),
+      RtpExtension("http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time",
+                   2)};
+
+  // Register the first map.
+  transport.RegisterRtpHeaderExtensionMap("audio", extensions1);
+
+  // Parse a packet with an extension from the first map.
+  const unsigned char kRtpData1[] = {0x90, 0x11, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
+                                     0xBE, 0xDE, 0, 1,
+                                     // ID=1, len=0 (1 byte payload)
+                                     0x10, 0x00, 0x00, 0x00};
+  CopyOnWriteBuffer rtp_packet1(kRtpData1, sizeof(kRtpData1));
+  RtpPacketReceived parsed_packet1;
+  FakePacketTransport fake_rtp("fake_rtp");
+  fake_rtp.SetDestination(&fake_rtp, true);
+  transport.SetRtpPacketTransport(&fake_rtp);
+  TransportObserver observer(&transport);
+
+  // Register sink so the packet can be "demuxed" and surfaced to the observer.
+  RtpDemuxerCriteria demuxer_criteria;
+  demuxer_criteria.payload_types().insert(0x11);
+  transport.RegisterRtpDemuxerSink(demuxer_criteria, &observer);
+
+  // Send the packet.
+  fake_rtp.SendPacket(rtp_packet1.data<char>(), rtp_packet1.size(),
+                      AsyncSocketPacketOptions(), 0);
+  EXPECT_THAT(WaitUntil([&] { return observer.rtp_count(); }, Eq(1)),
+              IsRtcOk());
+  RTC_LOG(LS_INFO) << "Packet 1 received: " << observer.rtp_count();
+  RTC_LOG(LS_INFO)
+      << "Packet 1 has AudioLevelExtension: "
+      << observer.last_recv_rtp_packet().HasExtension<AudioLevelExtension>();
+  EXPECT_TRUE(
+      observer.last_recv_rtp_packet().HasExtension<AudioLevelExtension>());
+
+  // Register the second map (simulating BUNDLE).
+  transport.RegisterRtpHeaderExtensionMap("video", extensions2);
+
+  // Parse a packet with an extension from the second map.
+  const unsigned char kRtpData2[] = {0x90, 0x11, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
+                                     0xBE, 0xDE, 0, 1,
+                                     // ID=2, len=2 (3 byte payload)
+                                     0x22, 0x01, 0x02, 0x03};
+  CopyOnWriteBuffer rtp_packet2(kRtpData2, sizeof(kRtpData2));
+
+  fake_rtp.SendPacket(rtp_packet2.data<char>(), rtp_packet2.size(),
+                      AsyncSocketPacketOptions(), 0);
+  EXPECT_THAT(WaitUntil([&] { return observer.rtp_count(); }, Eq(2)),
+              IsRtcOk());
+  RTC_LOG(LS_INFO) << "Packet 2 received: " << observer.rtp_count();
+  RTC_LOG(LS_INFO)
+      << "Packet 2 has AbsoluteSendTime: "
+      << observer.last_recv_rtp_packet().HasExtension<AbsoluteSendTime>();
+  EXPECT_TRUE(observer.last_recv_rtp_packet().HasExtension<AbsoluteSendTime>());
+
+  // Unregister the second map.
+  transport.UnregisterRtpHeaderExtensionMap("video");
+
+  // A packet with the second map's extension should no longer parse the
+  // extension.
+  fake_rtp.SendPacket(rtp_packet2.data<char>(), rtp_packet2.size(),
+                      AsyncSocketPacketOptions(), 0);
+  EXPECT_THAT(WaitUntil([&] { return observer.rtp_count(); }, Eq(3)),
+              IsRtcOk());
+  EXPECT_FALSE(
+      observer.last_recv_rtp_packet().HasExtension<AbsoluteSendTime>());
+
+  transport.UnregisterRtpDemuxerSink(&observer);
+}
+
 // Test that SignalPacketReceived fires with rtcp=true when a RTCP packet is
 // received.
 TEST(RtpTransportTest, SignalDemuxedRtcp) {

From 359fa9ef4f5301b7a9d2ec9013b6dfe34d4fca17 Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Tue, 3 Mar 2026 19:24:42 +0100
Subject: [PATCH 206/472] Replace ArrayView with std::span in modules/rtp_rtcp

This change generated using mostly search and replace:
MakeArrayView replaced with std::span
ArrayView replaced with std::span
Run include cleaner
Removed dependency on array_view target

Bug: webrtc:439801349
Change-Id: I2726c523284723be23ee2e17325a741b4ef39795
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453380
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47059}
---
 modules/rtp_rtcp/BUILD.gn                     |  7 --
 modules/rtp_rtcp/include/flexfec_sender.h     |  4 +-
 .../include/rtp_header_extension_map.h        |  6 +-
 modules/rtp_rtcp/include/rtp_rtcp_defines.h   |  4 +-
 .../mocks/mock_network_link_rtcp_observer.h   |  5 +-
 modules/rtp_rtcp/mocks/mock_rtp_rtcp.h        | 10 +--
 .../absolute_capture_time_interpolator.cc     |  4 +-
 .../absolute_capture_time_interpolator.h      |  4 +-
 .../source/absolute_capture_time_sender.cc    |  4 +-
 .../source/absolute_capture_time_sender.h     |  4 +-
 .../source/active_decode_targets_helper.cc    | 10 +--
 .../source/active_decode_targets_helper.h     |  7 +-
 .../active_decode_targets_helper_unittest.cc  |  6 +-
 .../source/corruption_detection_extension.cc  |  8 +-
 .../source/corruption_detection_extension.h   |  6 +-
 .../fec_private_tables_bursty_unittest.cc     |  8 +-
 .../source/flexfec_03_header_reader_writer.cc |  4 +-
 .../source/flexfec_03_header_reader_writer.h  |  5 +-
 .../source/flexfec_header_reader_writer.cc    |  4 +-
 .../source/flexfec_header_reader_writer.h     |  5 +-
 .../flexfec_header_reader_writer_unittest.cc  | 28 +++----
 modules/rtp_rtcp/source/flexfec_sender.cc     |  4 +-
 .../source/forward_error_correction.h         |  6 +-
 .../forward_error_correction_internal.cc      | 14 ++--
 .../forward_error_correction_internal.h       |  7 +-
 .../frame_transformer_factory_unittest.cc     |  6 +-
 modules/rtp_rtcp/source/leb128_unittest.cc    | 12 ++-
 modules/rtp_rtcp/source/nack_rtx_unittest.cc  |  6 +-
 modules/rtp_rtcp/source/rtcp_packet.cc        |  6 +-
 modules/rtp_rtcp/source/rtcp_packet.h         |  5 +-
 .../rtcp_packet/compound_packet_unittest.cc   | 12 +--
 .../congestion_control_feedback.cc            |  6 +-
 .../rtcp_packet/congestion_control_feedback.h |  4 +-
 .../congestion_control_feedback_unittest.cc   |  6 +-
 .../source/rtcp_packet/nack_unittest.cc       | 10 +--
 .../source/rtcp_packet/remote_estimate.cc     |  6 +-
 .../source/rtcp_packet/remote_estimate.h      |  4 +-
 .../transport_feedback_unittest.cc            | 10 +--
 .../rtp_rtcp/source/rtcp_packet_unittest.cc   |  4 +-
 modules/rtp_rtcp/source/rtcp_receiver.cc      |  6 +-
 modules/rtp_rtcp/source/rtcp_receiver.h       |  8 +-
 .../rtp_rtcp/source/rtcp_receiver_unittest.cc |  4 +-
 modules/rtp_rtcp/source/rtcp_sender.cc        | 18 ++---
 modules/rtp_rtcp/source/rtcp_sender.h         |  6 +-
 .../rtp_rtcp/source/rtcp_sender_unittest.cc   |  8 +-
 .../rtp_rtcp/source/rtcp_transceiver_config.h |  6 +-
 .../rtp_rtcp/source/rtcp_transceiver_impl.cc  | 18 ++---
 .../rtp_rtcp/source/rtcp_transceiver_impl.h   | 12 +--
 .../source/rtcp_transceiver_impl_unittest.cc  | 25 +++---
 .../source/rtcp_transceiver_unittest.cc       | 25 +++---
 .../rtp_dependency_descriptor_extension.cc    |  8 +-
 .../rtp_dependency_descriptor_extension.h     | 10 +--
 ...ependency_descriptor_extension_unittest.cc |  4 +-
 .../rtp_dependency_descriptor_reader.cc       |  4 +-
 .../source/rtp_dependency_descriptor_reader.h |  4 +-
 .../rtp_dependency_descriptor_writer.cc       |  4 +-
 .../source/rtp_dependency_descriptor_writer.h |  4 +-
 modules/rtp_rtcp/source/rtp_format.cc         |  4 +-
 modules/rtp_rtcp/source/rtp_format.h          |  4 +-
 modules/rtp_rtcp/source/rtp_format_h264.cc    | 16 ++--
 modules/rtp_rtcp/source/rtp_format_h264.h     | 10 +--
 .../source/rtp_format_h264_unittest.cc        | 10 +--
 .../source/rtp_format_video_generic.cc        |  6 +-
 .../source/rtp_format_video_generic.h         |  8 +-
 .../rtp_format_video_generic_unittest.cc      | 10 +--
 modules/rtp_rtcp/source/rtp_format_vp8.cc     |  4 +-
 modules/rtp_rtcp/source/rtp_format_vp8.h      |  6 +-
 .../source/rtp_format_vp8_test_helper.cc      | 14 ++--
 .../source/rtp_format_vp8_test_helper.h       | 14 ++--
 modules/rtp_rtcp/source/rtp_format_vp9.cc     |  8 +-
 modules/rtp_rtcp/source/rtp_format_vp9.h      |  8 +-
 .../source/rtp_format_vp9_unittest.cc         |  8 +-
 .../source/rtp_generic_frame_descriptor.cc    |  6 +-
 .../source/rtp_generic_frame_descriptor.h     |  5 +-
 .../rtp_generic_frame_descriptor_extension.cc |  8 +-
 .../rtp_generic_frame_descriptor_extension.h  |  7 +-
 .../source/rtp_header_extension_map.cc        |  7 +-
 .../source/rtp_header_extension_size.cc       |  5 +-
 .../source/rtp_header_extension_size.h        |  5 +-
 .../rtp_rtcp/source/rtp_header_extensions.cc  | 76 +++++++++----------
 .../rtp_rtcp/source/rtp_header_extensions.h   | 76 +++++++++----------
 modules/rtp_rtcp/source/rtp_packet.cc         | 20 ++---
 modules/rtp_rtcp/source/rtp_packet.h          | 30 ++++----
 modules/rtp_rtcp/source/rtp_packet_history.cc |  4 +-
 modules/rtp_rtcp/source/rtp_packet_history.h  |  4 +-
 .../rtp_rtcp/source/rtp_packet_unittest.cc    | 16 ++--
 modules/rtp_rtcp/source/rtp_packetizer_av1.cc | 16 ++--
 modules/rtp_rtcp/source/rtp_packetizer_av1.h  | 12 +--
 .../source/rtp_packetizer_av1_unittest.cc     | 12 +--
 .../rtp_rtcp/source/rtp_packetizer_h265.cc    | 12 +--
 modules/rtp_rtcp/source/rtp_packetizer_h265.h |  8 +-
 .../source/rtp_packetizer_h265_unittest.cc    | 10 +--
 modules/rtp_rtcp/source/rtp_rtcp_impl2.cc     | 17 ++---
 modules/rtp_rtcp/source/rtp_rtcp_impl2.h      | 12 +--
 .../source/rtp_rtcp_impl2_unittest.cc         |  6 +-
 modules/rtp_rtcp/source/rtp_rtcp_interface.h  | 10 +--
 modules/rtp_rtcp/source/rtp_sender.cc         | 14 ++--
 modules/rtp_rtcp/source/rtp_sender.h          | 10 +--
 modules/rtp_rtcp/source/rtp_sender_audio.h    |  6 +-
 .../source/rtp_sender_audio_unittest.cc       |  6 +-
 modules/rtp_rtcp/source/rtp_sender_egress.cc  |  6 +-
 modules/rtp_rtcp/source/rtp_sender_egress.h   |  6 +-
 .../source/rtp_sender_egress_unittest.cc      | 10 +--
 modules/rtp_rtcp/source/rtp_sender_video.cc   |  4 +-
 modules/rtp_rtcp/source/rtp_sender_video.h    |  4 +-
 ...sender_video_frame_transformer_delegate.cc |  6 +-
 ..._sender_video_frame_transformer_delegate.h |  4 +-
 ...deo_frame_transformer_delegate_unittest.cc |  4 +-
 .../source/rtp_sender_video_unittest.cc       | 14 ++--
 modules/rtp_rtcp/source/rtp_util.cc           | 14 ++--
 modules/rtp_rtcp/source/rtp_util.h            | 13 ++--
 .../rtp_video_layers_allocation_extension.cc  |  6 +-
 .../rtp_video_layers_allocation_extension.h   |  6 +-
 ...eo_layers_allocation_extension_unittest.cc | 22 +++---
 ...eam_receiver_frame_transformer_delegate.cc |  8 +-
 ...ver_frame_transformer_delegate_unittest.cc |  4 +-
 .../source/ulpfec_header_reader_writer.cc     |  4 +-
 .../source/ulpfec_header_reader_writer.h      |  5 +-
 .../rtp_rtcp/source/video_rtp_depacketizer.cc |  8 +-
 .../rtp_rtcp/source/video_rtp_depacketizer.h  |  4 +-
 .../source/video_rtp_depacketizer_av1.cc      | 10 +--
 .../source/video_rtp_depacketizer_av1.h       |  4 +-
 .../video_rtp_depacketizer_av1_unittest.cc    | 62 +++++++--------
 .../source/video_rtp_depacketizer_h264.cc     | 18 ++---
 .../video_rtp_depacketizer_h264_unittest.cc   | 32 ++++----
 .../source/video_rtp_depacketizer_h265.cc     |  6 +-
 .../video_rtp_depacketizer_h265_unittest.cc   | 68 ++++++++---------
 .../source/video_rtp_depacketizer_vp8.cc      |  6 +-
 .../source/video_rtp_depacketizer_vp8.h       |  4 +-
 .../video_rtp_depacketizer_vp8_unittest.cc    |  4 +-
 .../source/video_rtp_depacketizer_vp9.cc      |  4 +-
 .../source/video_rtp_depacketizer_vp9.h       |  4 +-
 .../video_rtp_depacketizer_vp9_unittest.cc    |  4 +-
 .../test/testFec/test_packet_masks_metrics.cc |  4 +-
 134 files changed, 685 insertions(+), 692 deletions(-)

diff --git a/modules/rtp_rtcp/BUILD.gn b/modules/rtp_rtcp/BUILD.gn
index ebaaa9fec4..5eda096822 100644
--- a/modules/rtp_rtcp/BUILD.gn
+++ b/modules/rtp_rtcp/BUILD.gn
@@ -113,7 +113,6 @@ rtc_library("rtp_rtcp_format") {
   deps = [
     ":leb128",
     "..:module_api_public",
-    "../../api:array_view",
     "../../api:function_view",
     "../../api:refcountedbase",
     "../../api:rtp_headers",
@@ -276,7 +275,6 @@ rtc_library("rtp_rtcp") {
     ":rtp_video_header",
     "..:module_api_public",
     "..:module_fec_api",
-    "../../api:array_view",
     "../../api:field_trials_view",
     "../../api:frame_transformer_interface",
     "../../api:function_view",
@@ -387,7 +385,6 @@ rtc_library("rtcp_transceiver") {
     ":ntp_time_util",
     ":rtp_rtcp",
     ":rtp_rtcp_format",
-    "../../api:array_view",
     "../../api:rtp_headers",
     "../../api/task_queue",
     "../../api/units:data_rate",
@@ -472,7 +469,6 @@ rtc_library("mock_rtp_rtcp") {
     ":rtp_rtcp",
     ":rtp_rtcp_format",
     "..:module_fec_api",
-    "../../api:array_view",
     "../../api:rtp_headers",
     "../../api/transport:network_control",
     "../../api/units:data_rate",
@@ -505,7 +501,6 @@ if (rtc_include_tests) {
       deps = [
         ":rtp_rtcp",
         "..:module_fec_api",
-        "../../api:array_view",
         "../../rtc_base:checks",
         "../../test:fileutils",
         "../../test:test_main",
@@ -637,7 +632,6 @@ if (rtc_include_tests) {
       ":rtp_video_header",
       ":rtp_video_header_unittest",
       "..:module_fec_api",
-      "../../api:array_view",
       "../../api:create_time_controller",
       "../../api:field_trials_view",
       "../../api:frame_transformer_factory",
@@ -714,7 +708,6 @@ if (rtc_include_tests) {
     testonly = true
     sources = [ "source/frame_transformer_factory_unittest.cc" ]
     deps = [
-      "../../api:array_view",
       "../../api:frame_transformer_factory",
       "../../api:mock_transformable_audio_frame",
       "../../api:mock_transformable_video_frame",
diff --git a/modules/rtp_rtcp/include/flexfec_sender.h b/modules/rtp_rtcp/include/flexfec_sender.h
index 75006612f6..ff1855d634 100644
--- a/modules/rtp_rtcp/include/flexfec_sender.h
+++ b/modules/rtp_rtcp/include/flexfec_sender.h
@@ -15,11 +15,11 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/rtp_parameters.h"
 #include "api/units/data_rate.h"
@@ -50,7 +50,7 @@ class FlexfecSender : public VideoFecGenerator {
                 uint32_t protected_media_ssrc,
                 absl::string_view mid,
                 const std::vector<RtpExtension>& rtp_header_extensions,
-                ArrayView<const RtpExtensionSize> extension_sizes,
+                std::span<const RtpExtensionSize> extension_sizes,
                 const RtpState* rtp_state);
   ~FlexfecSender() override;
 
diff --git a/modules/rtp_rtcp/include/rtp_header_extension_map.h b/modules/rtp_rtcp/include/rtp_header_extension_map.h
index 28e33a7b72..9ebfcf7ec6 100644
--- a/modules/rtp_rtcp/include/rtp_header_extension_map.h
+++ b/modules/rtp_rtcp/include/rtp_header_extension_map.h
@@ -13,9 +13,9 @@
 
 #include <array>
 #include <cstdint>
+#include <span>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtp_parameters.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "rtc_base/checks.h"
@@ -29,9 +29,9 @@ class RtpHeaderExtensionMap {
 
   RtpHeaderExtensionMap();
   explicit RtpHeaderExtensionMap(bool extmap_allow_mixed);
-  explicit RtpHeaderExtensionMap(ArrayView<const RtpExtension> extensions);
+  explicit RtpHeaderExtensionMap(std::span<const RtpExtension> extensions);
 
-  void Reset(ArrayView<const RtpExtension> extensions);
+  void Reset(std::span<const RtpExtension> extensions);
 
   template <typename Extension>
   bool Register(int id) {
diff --git a/modules/rtp_rtcp/include/rtp_rtcp_defines.h b/modules/rtp_rtcp/include/rtp_rtcp_defines.h
index f6d317f928..a8b235401c 100644
--- a/modules/rtp_rtcp/include/rtp_rtcp_defines.h
+++ b/modules/rtp_rtcp/include/rtp_rtcp_defines.h
@@ -18,11 +18,11 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <vector>
 
 #include "absl/algorithm/container.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/transport/network_types.h"
 #include "api/units/data_rate.h"
 #include "api/units/time_delta.h"
@@ -183,7 +183,7 @@ class NetworkLinkRtcpObserver {
   // Called on an RTCP packet with sender or receiver reports with non zero
   // report blocks. Report blocks are combined from all reports into one array.
   virtual void OnReport(Timestamp /* receive_time */,
-                        ArrayView<const ReportBlockData> /* report_blocks */) {}
+                        std::span<const ReportBlockData> /* report_blocks */) {}
   virtual void OnRttUpdate(Timestamp /* receive_time */, TimeDelta /* rtt */) {}
 };
 
diff --git a/modules/rtp_rtcp/mocks/mock_network_link_rtcp_observer.h b/modules/rtp_rtcp/mocks/mock_network_link_rtcp_observer.h
index f2775b79d0..80dc5e971c 100644
--- a/modules/rtp_rtcp/mocks/mock_network_link_rtcp_observer.h
+++ b/modules/rtp_rtcp/mocks/mock_network_link_rtcp_observer.h
@@ -11,7 +11,8 @@
 #ifndef MODULES_RTP_RTCP_MOCKS_MOCK_NETWORK_LINK_RTCP_OBSERVER_H_
 #define MODULES_RTP_RTCP_MOCKS_MOCK_NETWORK_LINK_RTCP_OBSERVER_H_
 
-#include "api/array_view.h"
+#include <span>
+
 #include "api/units/data_rate.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
@@ -45,7 +46,7 @@ class MockNetworkLinkRtcpObserver : public NetworkLinkRtcpObserver {
   MOCK_METHOD(void,
               OnReport,
               (Timestamp receive_time,
-               ArrayView<const ReportBlockData> report_blocks),
+               std::span<const ReportBlockData> report_blocks),
               (override));
 };
 
diff --git a/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
index 46163f7f35..dbfc386ebf 100644
--- a/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
+++ b/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
@@ -15,10 +15,10 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtp_headers.h"
 #include "api/transport/network_types.h"
 #include "api/units/time_delta.h"
@@ -38,7 +38,7 @@ class MockRtpRtcpInterface : public RtpRtcpInterface {
  public:
   MOCK_METHOD(void,
               IncomingRtcpPacket,
-              (ArrayView<const uint8_t> packet),
+              (std::span<const uint8_t> packet),
               (override));
   MOCK_METHOD(void, SetRemoteSSRC, (uint32_t ssrc), (override));
   MOCK_METHOD(void, SetLocalSsrc, (uint32_t ssrc), (override));
@@ -120,11 +120,11 @@ class MockRtpRtcpInterface : public RtpRtcpInterface {
               (override));
   MOCK_METHOD(void,
               OnAbortedRetransmissions,
-              (ArrayView<const uint16_t>),
+              (std::span<const uint16_t>),
               (override));
   MOCK_METHOD(void,
               OnPacketsAcknowledged,
-              (ArrayView<const uint16_t>),
+              (std::span<const uint16_t>),
               (override));
   MOCK_METHOD(std::vector<std::unique_ptr<RtpPacketToSend>>,
               GeneratePadding,
@@ -132,7 +132,7 @@ class MockRtpRtcpInterface : public RtpRtcpInterface {
               (override));
   MOCK_METHOD(std::vector<RtpSequenceNumberMap::Info>,
               GetSentRtpPacketInfos,
-              (ArrayView<const uint16_t> sequence_numbers),
+              (std::span<const uint16_t> sequence_numbers),
               (const, override));
   MOCK_METHOD(size_t, ExpectedPerPacketOverhead, (), (const, override));
   MOCK_METHOD(void, OnPacketSendingThreadSwitched, (), (override));
diff --git a/modules/rtp_rtcp/source/absolute_capture_time_interpolator.cc b/modules/rtp_rtcp/source/absolute_capture_time_interpolator.cc
index ef7aa4ed6c..5745a31666 100644
--- a/modules/rtp_rtcp/source/absolute_capture_time_interpolator.cc
+++ b/modules/rtp_rtcp/source/absolute_capture_time_interpolator.cc
@@ -13,8 +13,8 @@
 #include <cstdint>
 #include <cstdlib>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/rtp_headers.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
@@ -31,7 +31,7 @@ AbsoluteCaptureTimeInterpolator::AbsoluteCaptureTimeInterpolator(Clock* clock)
 
 uint32_t AbsoluteCaptureTimeInterpolator::GetSource(
     uint32_t ssrc,
-    ArrayView<const uint32_t> csrcs) {
+    std::span<const uint32_t> csrcs) {
   if (csrcs.empty()) {
     return ssrc;
   }
diff --git a/modules/rtp_rtcp/source/absolute_capture_time_interpolator.h b/modules/rtp_rtcp/source/absolute_capture_time_interpolator.h
index c6212a8646..b9a5c49761 100644
--- a/modules/rtp_rtcp/source/absolute_capture_time_interpolator.h
+++ b/modules/rtp_rtcp/source/absolute_capture_time_interpolator.h
@@ -13,8 +13,8 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/rtp_headers.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
@@ -43,7 +43,7 @@ class AbsoluteCaptureTimeInterpolator {
   explicit AbsoluteCaptureTimeInterpolator(Clock* clock);
 
   // Returns the source (i.e. SSRC or CSRC) of the capture system.
-  static uint32_t GetSource(uint32_t ssrc, ArrayView<const uint32_t> csrcs);
+  static uint32_t GetSource(uint32_t ssrc, std::span<const uint32_t> csrcs);
 
   // Returns a received header extension, an interpolated header extension, or
   // `std::nullopt` if it's not possible to interpolate a header extension.
diff --git a/modules/rtp_rtcp/source/absolute_capture_time_sender.cc b/modules/rtp_rtcp/source/absolute_capture_time_sender.cc
index 61e321f6bb..bf621fe454 100644
--- a/modules/rtp_rtcp/source/absolute_capture_time_sender.cc
+++ b/modules/rtp_rtcp/source/absolute_capture_time_sender.cc
@@ -13,8 +13,8 @@
 #include <algorithm>
 #include <cstdint>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/rtp_headers.h"
 #include "api/units/timestamp.h"
 #include "modules/rtp_rtcp/source/absolute_capture_time_interpolator.h"
@@ -32,7 +32,7 @@ AbsoluteCaptureTimeSender::AbsoluteCaptureTimeSender(Clock* clock)
     : clock_(clock) {}
 
 uint32_t AbsoluteCaptureTimeSender::GetSource(uint32_t ssrc,
-                                              ArrayView<const uint32_t> csrcs) {
+                                              std::span<const uint32_t> csrcs) {
   return AbsoluteCaptureTimeInterpolator::GetSource(ssrc, csrcs);
 }
 
diff --git a/modules/rtp_rtcp/source/absolute_capture_time_sender.h b/modules/rtp_rtcp/source/absolute_capture_time_sender.h
index 352754c7b7..b7f885f430 100644
--- a/modules/rtp_rtcp/source/absolute_capture_time_sender.h
+++ b/modules/rtp_rtcp/source/absolute_capture_time_sender.h
@@ -13,8 +13,8 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/rtp_headers.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
@@ -48,7 +48,7 @@ class AbsoluteCaptureTimeSender {
   explicit AbsoluteCaptureTimeSender(Clock* clock);
 
   // Returns the source (i.e. SSRC or CSRC) of the capture system.
-  static uint32_t GetSource(uint32_t ssrc, ArrayView<const uint32_t> csrcs);
+  static uint32_t GetSource(uint32_t ssrc, std::span<const uint32_t> csrcs);
 
   // Returns value to write into AbsoluteCaptureTime RTP header extension to be
   // sent, or `std::nullopt` if the header extension shouldn't be attached to
diff --git a/modules/rtp_rtcp/source/active_decode_targets_helper.cc b/modules/rtp_rtcp/source/active_decode_targets_helper.cc
index c1150a571b..901ed6ec08 100644
--- a/modules/rtp_rtcp/source/active_decode_targets_helper.cc
+++ b/modules/rtp_rtcp/source/active_decode_targets_helper.cc
@@ -13,8 +13,8 @@
 #include <bitset>
 #include <cstddef>
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
 
@@ -26,7 +26,7 @@ namespace {
 // missing. That assumptions allows a simple detection when previous frame is
 // part of a chain.
 std::bitset<32> LastSendOnChain(int frame_diff,
-                                ArrayView<const int> chain_diffs) {
+                                std::span<const int> chain_diffs) {
   std::bitset<32> bitmask = 0;
   for (size_t i = 0; i < chain_diffs.size(); ++i) {
     if (frame_diff == chain_diffs[i]) {
@@ -44,7 +44,7 @@ std::bitset<32> AllActive(size_t num) {
 
 // Returns bitmask of chains that protect at least one active decode target.
 std::bitset<32> ActiveChains(
-    ArrayView<const int> decode_target_protected_by_chain,
+    std::span<const int> decode_target_protected_by_chain,
     int num_chains,
     std::bitset<32> active_decode_targets) {
   std::bitset<32> active_chains = 0;
@@ -62,11 +62,11 @@ std::bitset<32> ActiveChains(
 }  // namespace
 
 void ActiveDecodeTargetsHelper::OnFrame(
-    ArrayView<const int> decode_target_protected_by_chain,
+    std::span<const int> decode_target_protected_by_chain,
     std::bitset<32> active_decode_targets,
     bool is_keyframe,
     int64_t frame_id,
-    ArrayView<const int> chain_diffs) {
+    std::span<const int> chain_diffs) {
   const int num_chains = chain_diffs.size();
   if (num_chains == 0) {
     // Avoid printing the warning
diff --git a/modules/rtp_rtcp/source/active_decode_targets_helper.h b/modules/rtp_rtcp/source/active_decode_targets_helper.h
index 293abafb73..8e432ee859 100644
--- a/modules/rtp_rtcp/source/active_decode_targets_helper.h
+++ b/modules/rtp_rtcp/source/active_decode_targets_helper.h
@@ -15,8 +15,7 @@
 
 #include <bitset>
 #include <optional>
-
-#include "api/array_view.h"
+#include <span>
 
 namespace webrtc {
 
@@ -34,11 +33,11 @@ class ActiveDecodeTargetsHelper {
 
   // Decides if active decode target bitmask should be attached to the frame
   // that is about to be sent.
-  void OnFrame(ArrayView<const int> decode_target_protected_by_chain,
+  void OnFrame(std::span<const int> decode_target_protected_by_chain,
                std::bitset<32> active_decode_targets,
                bool is_keyframe,
                int64_t frame_id,
-               ArrayView<const int> chain_diffs);
+               std::span<const int> chain_diffs);
 
   // Returns active decode target to attach to the dependency descriptor.
   std::optional<uint32_t> ActiveDecodeTargetsBitmask() const {
diff --git a/modules/rtp_rtcp/source/active_decode_targets_helper_unittest.cc b/modules/rtp_rtcp/source/active_decode_targets_helper_unittest.cc
index e87ae1719d..5062a075d1 100644
--- a/modules/rtp_rtcp/source/active_decode_targets_helper_unittest.cc
+++ b/modules/rtp_rtcp/source/active_decode_targets_helper_unittest.cc
@@ -13,9 +13,9 @@
 #include <bitset>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "test/gtest.h"
 
 namespace webrtc {
@@ -228,8 +228,8 @@ TEST(ActiveDecodeTargetsHelperTest, ReturnsBitmaskWhenChanged) {
 }
 
 TEST(ActiveDecodeTargetsHelperTest, ReturnsNulloptWhenChainsAreNotUsed) {
-  const ArrayView<const int> kDecodeTargetProtectedByChain;
-  const ArrayView<const int> kNoChainDiffs;
+  const std::span<const int> kDecodeTargetProtectedByChain;
+  const std::span<const int> kNoChainDiffs;
 
   ActiveDecodeTargetsHelper helper;
   helper.OnFrame(kDecodeTargetProtectedByChain, /*active_decode_targets=*/kAll,
diff --git a/modules/rtp_rtcp/source/corruption_detection_extension.cc b/modules/rtp_rtcp/source/corruption_detection_extension.cc
index 7c173a2b67..4913dd1c9f 100644
--- a/modules/rtp_rtcp/source/corruption_detection_extension.cc
+++ b/modules/rtp_rtcp/source/corruption_detection_extension.cc
@@ -13,9 +13,9 @@
 #include <cmath>
 #include <cstddef>
 #include <cstdint>
+#include <span>
 
 #include "absl/container/inlined_vector.h"
-#include "api/array_view.h"
 #include "api/transport/rtp/corruption_detection_message.h"
 
 namespace webrtc {
@@ -30,7 +30,7 @@ constexpr double kMaxValueForStdDev = 40.0;
 // A description of the extension can be found at
 // http://www.webrtc.org/experiments/rtp-hdrext/corruption-detection
 
-bool CorruptionDetectionExtension::Parse(ArrayView<const uint8_t> data,
+bool CorruptionDetectionExtension::Parse(std::span<const uint8_t> data,
                                          CorruptionDetectionMessage* message) {
   if (message == nullptr) {
     return false;
@@ -55,7 +55,7 @@ bool CorruptionDetectionExtension::Parse(ArrayView<const uint8_t> data,
 }
 
 bool CorruptionDetectionExtension::Write(
-    ArrayView<uint8_t> data,
+    std::span<uint8_t> data,
     const CorruptionDetectionMessage& message) {
   if (data.size() != ValueSize(message) || data.size() > kMaxValueSizeBytes) {
     return false;
@@ -72,7 +72,7 @@ bool CorruptionDetectionExtension::Write(
       std::round(message.std_dev() / kMaxValueForStdDev * 255.0));
   data[2] = (message.luma_error_threshold() << 4) |
             (message.chroma_error_threshold() & 0xF);
-  ArrayView<uint8_t> sample_values = data.subspan(kConfigurationBytes);
+  std::span<uint8_t> sample_values = data.subspan(kConfigurationBytes);
   for (size_t i = 0; i < message.sample_values().size(); ++i) {
     sample_values[i] = std::floor(message.sample_values()[i]);
   }
diff --git a/modules/rtp_rtcp/source/corruption_detection_extension.h b/modules/rtp_rtcp/source/corruption_detection_extension.h
index 4dc6c55b39..ee3e14b822 100644
--- a/modules/rtp_rtcp/source/corruption_detection_extension.h
+++ b/modules/rtp_rtcp/source/corruption_detection_extension.h
@@ -13,9 +13,9 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtp_parameters.h"
 #include "api/transport/rtp/corruption_detection_message.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
@@ -38,9 +38,9 @@ class CorruptionDetectionExtension {
   static constexpr absl::string_view Uri() {
     return RtpExtension::kCorruptionDetectionUri;
   }
-  static bool Parse(ArrayView<const uint8_t> data,
+  static bool Parse(std::span<const uint8_t> data,
                     CorruptionDetectionMessage* message);
-  static bool Write(ArrayView<uint8_t> data,
+  static bool Write(std::span<uint8_t> data,
                     const CorruptionDetectionMessage& message);
   // Size of the header extension in bytes.
   static size_t ValueSize(const CorruptionDetectionMessage& message);
diff --git a/modules/rtp_rtcp/source/fec_private_tables_bursty_unittest.cc b/modules/rtp_rtcp/source/fec_private_tables_bursty_unittest.cc
index d85b450bdc..b14e2fb0ff 100644
--- a/modules/rtp_rtcp/source/fec_private_tables_bursty_unittest.cc
+++ b/modules/rtp_rtcp/source/fec_private_tables_bursty_unittest.cc
@@ -12,8 +12,8 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/include/module_fec_types.h"
 #include "modules/rtp_rtcp/source/fec_private_tables_random.h"
 #include "modules/rtp_rtcp/source/forward_error_correction_internal.h"
@@ -30,7 +30,7 @@ namespace fec_private_tables {
 using internal::LookUpInFecTable;
 
 TEST(FecTable, TestBurstyLookup) {
-  ArrayView<const uint8_t> result;
+  std::span<const uint8_t> result;
   result = LookUpInFecTable(&kPacketMaskBurstyTbl[0], 0, 0);
   // Should match kMaskBursty1_1.
   EXPECT_EQ(2u, result.size());
@@ -56,7 +56,7 @@ TEST(FecTable, TestBurstyLookup) {
 }
 
 TEST(FecTable, TestRandomLookup) {
-  ArrayView<const uint8_t> result;
+  std::span<const uint8_t> result;
   result = LookUpInFecTable(&kPacketMaskRandomTbl[0], 0, 0);
   EXPECT_EQ(2u, result.size());
   EXPECT_EQ(0x80u, result[0]);
@@ -75,7 +75,7 @@ TEST(FecTable, TestRandomGenerated) {
   int num_fec_packets = 6;
   size_t mask_size = sizeof(kMaskRandom15_6) / sizeof(uint8_t);
   internal::PacketMaskTable mask_table(fec_mask_type, num_media_packets);
-  ArrayView<const uint8_t> mask =
+  std::span<const uint8_t> mask =
       mask_table.LookUp(num_media_packets, num_fec_packets);
   EXPECT_EQ(mask.size(), mask_size);
   for (size_t i = 0; i < mask_size; ++i) {
diff --git a/modules/rtp_rtcp/source/flexfec_03_header_reader_writer.cc b/modules/rtp_rtcp/source/flexfec_03_header_reader_writer.cc
index f05f926847..5f4c32ab3e 100644
--- a/modules/rtp_rtcp/source/flexfec_03_header_reader_writer.cc
+++ b/modules/rtp_rtcp/source/flexfec_03_header_reader_writer.cc
@@ -12,8 +12,8 @@
 
 #include <cstdint>
 #include <cstring>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/scoped_refptr.h"
 #include "modules/rtp_rtcp/source/byte_io.h"
 #include "modules/rtp_rtcp/source/forward_error_correction.h"
@@ -249,7 +249,7 @@ size_t Flexfec03HeaderWriter::FecHeaderSize(size_t packet_mask_size) const {
 // FecHeaderSize(), so in this function we can be sure that we are
 // writing in space that is intended for the header.
 void Flexfec03HeaderWriter::FinalizeFecHeader(
-    ArrayView<const ProtectedStream> protected_streams,
+    std::span<const ProtectedStream> protected_streams,
     ForwardErrorCorrection::Packet& fec_packet) const {
   RTC_CHECK_EQ(protected_streams.size(), 1);
   uint32_t media_ssrc = protected_streams[0].ssrc;
diff --git a/modules/rtp_rtcp/source/flexfec_03_header_reader_writer.h b/modules/rtp_rtcp/source/flexfec_03_header_reader_writer.h
index 04dcf31ee1..30be1754ce 100644
--- a/modules/rtp_rtcp/source/flexfec_03_header_reader_writer.h
+++ b/modules/rtp_rtcp/source/flexfec_03_header_reader_writer.h
@@ -14,7 +14,8 @@
 #include <stddef.h>
 #include <stdint.h>
 
-#include "api/array_view.h"
+#include <span>
+
 #include "modules/rtp_rtcp/source/forward_error_correction.h"
 
 namespace webrtc {
@@ -77,7 +78,7 @@ class Flexfec03HeaderWriter : public FecHeaderWriter {
   size_t FecHeaderSize(size_t packet_mask_row_size) const override;
 
   void FinalizeFecHeader(
-      ArrayView<const ProtectedStream> protected_streams,
+      std::span<const ProtectedStream> protected_streams,
       ForwardErrorCorrection::Packet& fec_packet) const override;
 };
 
diff --git a/modules/rtp_rtcp/source/flexfec_header_reader_writer.cc b/modules/rtp_rtcp/source/flexfec_header_reader_writer.cc
index d467d53203..d79b4449f1 100644
--- a/modules/rtp_rtcp/source/flexfec_header_reader_writer.cc
+++ b/modules/rtp_rtcp/source/flexfec_header_reader_writer.cc
@@ -12,8 +12,8 @@
 
 #include <cstdint>
 #include <cstring>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/scoped_refptr.h"
 #include "modules/rtp_rtcp/source/byte_io.h"
 #include "modules/rtp_rtcp/source/forward_error_correction.h"
@@ -250,7 +250,7 @@ size_t FlexfecHeaderWriter::FecHeaderSize(size_t packet_mask_size) const {
 // TODO(brandtr): Update this function when we support offset-based masks
 // and retransmissions.
 void FlexfecHeaderWriter::FinalizeFecHeader(
-    ArrayView<const ProtectedStream> protected_streams,
+    std::span<const ProtectedStream> protected_streams,
     ForwardErrorCorrection::Packet& fec_packet) const {
   uint8_t* data = fec_packet.data.MutableData();
   *data &= 0x7f;  // Clear R bit.
diff --git a/modules/rtp_rtcp/source/flexfec_header_reader_writer.h b/modules/rtp_rtcp/source/flexfec_header_reader_writer.h
index 1f5202cd18..7018c856e5 100644
--- a/modules/rtp_rtcp/source/flexfec_header_reader_writer.h
+++ b/modules/rtp_rtcp/source/flexfec_header_reader_writer.h
@@ -14,7 +14,8 @@
 #include <stddef.h>
 #include <stdint.h>
 
-#include "api/array_view.h"
+#include <span>
+
 #include "modules/rtp_rtcp/source/forward_error_correction.h"
 
 namespace webrtc {
@@ -59,7 +60,7 @@ class FlexfecHeaderWriter : public FecHeaderWriter {
   size_t FecHeaderSize(size_t packet_mask_row_size) const override;
 
   void FinalizeFecHeader(
-      ArrayView<const ProtectedStream> protected_streams,
+      std::span<const ProtectedStream> protected_streams,
       ForwardErrorCorrection::Packet& fec_packet) const override;
 };
 
diff --git a/modules/rtp_rtcp/source/flexfec_header_reader_writer_unittest.cc b/modules/rtp_rtcp/source/flexfec_header_reader_writer_unittest.cc
index baeea9ddf3..1bc056fe8c 100644
--- a/modules/rtp_rtcp/source/flexfec_header_reader_writer_unittest.cc
+++ b/modules/rtp_rtcp/source/flexfec_header_reader_writer_unittest.cc
@@ -12,9 +12,9 @@
 
 #include <cstdint>
 #include <cstring>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/make_ref_counted.h"
 #include "modules/rtp_rtcp/source/byte_io.h"
 #include "modules/rtp_rtcp/source/forward_error_correction.h"
@@ -54,13 +54,13 @@ constexpr uint8_t kPayloadBits = 0x00;
 
 struct FecPacketStreamReadProperties {
   ProtectedStream stream;
-  ArrayView<const uint8_t> mask;
+  std::span<const uint8_t> mask;
 };
 
 struct FecPacketStreamWriteProperties {
   size_t byte_index;
   uint16_t seq_num_base;
-  ArrayView<const uint8_t> mask;
+  std::span<const uint8_t> mask;
 };
 
 Packet WritePacket(
@@ -93,9 +93,9 @@ void VerifyReadHeaders(size_t expected_fec_header_size,
     EXPECT_EQ(protected_stream.packet_mask_size,
               expected[i].stream.packet_mask_size);
     // Ensure that the K-bits are removed and the packet mask has been packed.
-    EXPECT_THAT(MakeArrayView(read_packet.pkt->data.cdata() +
-                                  protected_stream.packet_mask_offset,
-                              protected_stream.packet_mask_size),
+    EXPECT_THAT(std::span(read_packet.pkt->data.cdata() +
+                              protected_stream.packet_mask_offset,
+                          protected_stream.packet_mask_size),
                 ElementsAreArray(expected[i].mask));
   }
   EXPECT_EQ(read_packet.pkt->data.size() - expected_fec_header_size,
@@ -115,9 +115,9 @@ void VerifyFinalizedHeaders(
         ByteReader<uint16_t>::ReadBigEndian(packet + expected[i].byte_index),
         expected[i].seq_num_base);
     // Verify mask.
-    EXPECT_THAT(MakeArrayView(packet + expected[i].byte_index + 2,
-                              expected[i].mask.size()),
-                ElementsAreArray(expected[i].mask));
+    EXPECT_THAT(
+        std::span(packet + expected[i].byte_index + 2, expected[i].mask.size()),
+        ElementsAreArray(expected[i].mask));
   }
 }
 
@@ -162,19 +162,19 @@ void VerifyWrittenAndReadHeaders(
         read_packet.pkt->data.cdata() +
         read_packet.protected_streams[i].packet_mask_offset;
     // Verify actual mask bits.
-    EXPECT_THAT(MakeArrayView(read_mask_ptr, mask_write_size),
+    EXPECT_THAT(std::span(read_mask_ptr, mask_write_size),
                 ElementsAreArray(write_protected_streams[i].packet_mask));
     // If read mask size is larger than written mask size, verify all other bits
     // are 0.
-    EXPECT_THAT(MakeArrayView(read_mask_ptr + mask_write_size,
-                              expected_mask_read_size - mask_write_size),
+    EXPECT_THAT(std::span(read_mask_ptr + mask_write_size,
+                          expected_mask_read_size - mask_write_size),
                 Each(0));
   }
 
   // Verify that the call to ReadFecHeader did not tamper with the payload.
   EXPECT_THAT(
-      MakeArrayView(read_packet.pkt->data.cdata() + read_packet.fec_header_size,
-                    read_packet.pkt->data.size() - read_packet.fec_header_size),
+      std::span(read_packet.pkt->data.cdata() + read_packet.fec_header_size,
+                read_packet.pkt->data.size() - read_packet.fec_header_size),
       ElementsAreArray(written_packet.data.cdata() + expected_header_size,
                        written_packet.data.size() - expected_header_size));
 }
diff --git a/modules/rtp_rtcp/source/flexfec_sender.cc b/modules/rtp_rtcp/source/flexfec_sender.cc
index 32328fb511..5a87f3fbce 100644
--- a/modules/rtp_rtcp/source/flexfec_sender.cc
+++ b/modules/rtp_rtcp/source/flexfec_sender.cc
@@ -15,11 +15,11 @@
 #include <list>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/rtp_parameters.h"
 #include "api/units/data_rate.h"
@@ -89,7 +89,7 @@ FlexfecSender::FlexfecSender(
     uint32_t protected_media_ssrc,
     absl::string_view mid,
     const std::vector<RtpExtension>& rtp_header_extensions,
-    ArrayView<const RtpExtensionSize> extension_sizes,
+    std::span<const RtpExtensionSize> extension_sizes,
     const RtpState* rtp_state)
     : env_(env),
       random_(env_.clock().TimeInMicroseconds()),
diff --git a/modules/rtp_rtcp/source/forward_error_correction.h b/modules/rtp_rtcp/source/forward_error_correction.h
index b4e1a50f6e..fcfa641204 100644
--- a/modules/rtp_rtcp/source/forward_error_correction.h
+++ b/modules/rtp_rtcp/source/forward_error_correction.h
@@ -16,10 +16,10 @@
 
 #include <list>
 #include <memory>
+#include <span>
 #include <vector>
 
 #include "absl/container/inlined_vector.h"
-#include "api/array_view.h"
 #include "api/scoped_refptr.h"
 #include "modules/include/module_fec_types.h"
 #include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
@@ -398,7 +398,7 @@ class FecHeaderWriter {
   struct ProtectedStream {
     uint32_t ssrc = 0;
     uint16_t seq_num_base = 0;
-    ArrayView<const uint8_t> packet_mask;
+    std::span<const uint8_t> packet_mask;
   };
 
   virtual ~FecHeaderWriter();
@@ -424,7 +424,7 @@ class FecHeaderWriter {
 
   // Writes FEC header.
   virtual void FinalizeFecHeader(
-      ArrayView<const ProtectedStream> protected_streams,
+      std::span<const ProtectedStream> protected_streams,
       ForwardErrorCorrection::Packet& fec_packet) const = 0;
 
  protected:
diff --git a/modules/rtp_rtcp/source/forward_error_correction_internal.cc b/modules/rtp_rtcp/source/forward_error_correction_internal.cc
index d2f3b59ec4..bc749b6297 100644
--- a/modules/rtp_rtcp/source/forward_error_correction_internal.cc
+++ b/modules/rtp_rtcp/source/forward_error_correction_internal.cc
@@ -13,8 +13,8 @@
 #include <algorithm>
 #include <cstdint>
 #include <cstring>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/include/module_fec_types.h"
 #include "modules/rtp_rtcp/source/fec_private_tables_bursty.h"
 #include "modules/rtp_rtcp/source/fec_private_tables_random.h"
@@ -150,7 +150,7 @@ PacketMaskTable::PacketMaskTable(FecMaskType fec_mask_type,
 
 PacketMaskTable::~PacketMaskTable() = default;
 
-ArrayView<const uint8_t> PacketMaskTable::LookUp(int num_media_packets,
+std::span<const uint8_t> PacketMaskTable::LookUp(int num_media_packets,
                                                  int num_fec_packets) {
   RTC_DCHECK_GT(num_media_packets, 0);
   RTC_DCHECK_GT(num_fec_packets, 0);
@@ -246,7 +246,7 @@ void RemainingPacketProtection(int num_media_packets,
         PacketMaskSize(num_media_packets - num_fec_for_imp_packets);
 
     auto end_row = (num_fec_for_imp_packets + num_fec_remaining);
-    ArrayView<const uint8_t> packet_mask_sub_21 = mask_table->LookUp(
+    std::span<const uint8_t> packet_mask_sub_21 = mask_table->LookUp(
         num_media_packets - num_fec_for_imp_packets, num_fec_remaining);
 
     ShiftFitSubMask(num_mask_bytes, res_mask_bytes, num_fec_for_imp_packets,
@@ -254,7 +254,7 @@ void RemainingPacketProtection(int num_media_packets,
 
   } else if (mode == kModeOverlap || mode == kModeBiasFirstPacket) {
     // sub_mask22
-    ArrayView<const uint8_t> packet_mask_sub_22 =
+    std::span<const uint8_t> packet_mask_sub_22 =
         mask_table->LookUp(num_media_packets, num_fec_remaining);
 
     FitSubMask(num_mask_bytes, num_mask_bytes, num_fec_remaining,
@@ -281,7 +281,7 @@ void ImportantPacketProtection(int num_fec_for_imp_packets,
   const int num_imp_mask_bytes = PacketMaskSize(num_imp_packets);
 
   // Get sub_mask1 from table
-  ArrayView<const uint8_t> packet_mask_sub_1 =
+  std::span<const uint8_t> packet_mask_sub_1 =
       mask_table->LookUp(num_imp_packets, num_fec_for_imp_packets);
 
   FitSubMask(num_mask_bytes, num_imp_mask_bytes, num_fec_for_imp_packets,
@@ -410,7 +410,7 @@ void UnequalProtectionMask(int num_media_packets,
 //     * For all entries: 2 * fec index (1 based)
 //   * Size for kPacketMaskBurstyTbl: 2 bytes.
 //     * For all entries: 2 * fec index (1 based)
-ArrayView<const uint8_t> LookUpInFecTable(const uint8_t* table,
+std::span<const uint8_t> LookUpInFecTable(const uint8_t* table,
                                           int media_packet_index,
                                           int fec_index) {
   RTC_DCHECK_LT(media_packet_index, table[0]);
@@ -466,7 +466,7 @@ void GeneratePacketMasks(int num_media_packets,
     // Retrieve corresponding mask table directly:for equal-protection case.
     // Mask = (k,n-k), with protection factor = (n-k)/k,
     // where k = num_media_packets, n=total#packets, (n-k)=num_fec_packets.
-    ArrayView<const uint8_t> mask =
+    std::span<const uint8_t> mask =
         mask_table->LookUp(num_media_packets, num_fec_packets);
     memcpy(packet_mask, &mask[0], mask.size());
   } else {  // UEP case
diff --git a/modules/rtp_rtcp/source/forward_error_correction_internal.h b/modules/rtp_rtcp/source/forward_error_correction_internal.h
index e02ba499d8..1a9674c9f5 100644
--- a/modules/rtp_rtcp/source/forward_error_correction_internal.h
+++ b/modules/rtp_rtcp/source/forward_error_correction_internal.h
@@ -14,7 +14,8 @@
 #include <stddef.h>
 #include <stdint.h>
 
-#include "api/array_view.h"
+#include <span>
+
 #include "modules/include/module_fec_types.h"
 
 namespace webrtc {
@@ -43,7 +44,7 @@ class PacketMaskTable {
   PacketMaskTable(FecMaskType fec_mask_type, int num_media_packets);
   ~PacketMaskTable();
 
-  ArrayView<const uint8_t> LookUp(int num_media_packets, int num_fec_packets);
+  std::span<const uint8_t> LookUp(int num_media_packets, int num_fec_packets);
 
  private:
   static const uint8_t* PickTable(FecMaskType fec_mask_type,
@@ -52,7 +53,7 @@ class PacketMaskTable {
   uint8_t fec_packet_mask_[kFECPacketMaskMaxSize];
 };
 
-ArrayView<const uint8_t> LookUpInFecTable(const uint8_t* table,
+std::span<const uint8_t> LookUpInFecTable(const uint8_t* table,
                                           int media_packet_index,
                                           int fec_index);
 
diff --git a/modules/rtp_rtcp/source/frame_transformer_factory_unittest.cc b/modules/rtp_rtcp/source/frame_transformer_factory_unittest.cc
index f6a923f022..0e909c7bc9 100644
--- a/modules/rtp_rtcp/source/frame_transformer_factory_unittest.cc
+++ b/modules/rtp_rtcp/source/frame_transformer_factory_unittest.cc
@@ -13,9 +13,9 @@
 #include <algorithm>
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/test/mock_transformable_audio_frame.h"
 #include "api/test/mock_transformable_video_frame.h"
 #include "api/video/video_frame_metadata.h"
@@ -34,7 +34,7 @@ TEST(FrameTransformerFactory, CloneAudioFrame) {
   NiceMock<MockTransformableAudioFrame> original_frame;
   uint8_t data[10];
   std::fill_n(data, 10, 5);
-  ArrayView<uint8_t> data_view(data);
+  std::span<uint8_t> data_view(data);
   ON_CALL(original_frame, GetData()).WillByDefault(Return(data_view));
   auto cloned_frame = CloneAudioFrame(&original_frame);
 
@@ -45,7 +45,7 @@ TEST(FrameTransformerFactory, CloneVideoFrame) {
   NiceMock<MockTransformableVideoFrame> original_frame;
   uint8_t data[10];
   std::fill_n(data, 10, 5);
-  ArrayView<uint8_t> data_view(data);
+  std::span<uint8_t> data_view(data);
   EXPECT_CALL(original_frame, GetData()).WillRepeatedly(Return(data_view));
   VideoFrameMetadata metadata;
   std::vector<uint32_t> csrcs{123, 321};
diff --git a/modules/rtp_rtcp/source/leb128_unittest.cc b/modules/rtp_rtcp/source/leb128_unittest.cc
index c58b8bc776..3512dbd1c2 100644
--- a/modules/rtp_rtcp/source/leb128_unittest.cc
+++ b/modules/rtp_rtcp/source/leb128_unittest.cc
@@ -13,8 +13,8 @@
 #include <cstdint>
 #include <iterator>
 #include <limits>
+#include <span>
 
-#include "api/array_view.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
 
@@ -121,17 +121,15 @@ TEST(Leb128Test, WriteTwoByteValue) {
 TEST(Leb128Test, WriteNearlyMaxValue) {
   uint8_t buffer[16];
   EXPECT_EQ(WriteLeb128(0x7fff'ffff'ffff'ffff, buffer), 9);
-  EXPECT_THAT(
-      MakeArrayView(buffer, 9),
-      ElementsAre(0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0x7f));
+  EXPECT_THAT(std::span(buffer, 9), ElementsAre(0xff, 0xff, 0xff, 0xff, 0xff,
+                                                0xff, 0xff, 0xff, 0x7f));
 }
 
 TEST(Leb128Test, WriteMaxValue) {
   uint8_t buffer[16];
   EXPECT_EQ(WriteLeb128(0xffff'ffff'ffff'ffff, buffer), 10);
-  EXPECT_THAT(
-      MakeArrayView(buffer, 10),
-      ElementsAre(0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0x01));
+  EXPECT_THAT(std::span(buffer, 10), ElementsAre(0xff, 0xff, 0xff, 0xff, 0xff,
+                                                 0xff, 0xff, 0xff, 0xff, 0x01));
 }
 
 }  // namespace
diff --git a/modules/rtp_rtcp/source/nack_rtx_unittest.cc b/modules/rtp_rtcp/source/nack_rtx_unittest.cc
index 09ac8a88b5..a94dd69b38 100644
--- a/modules/rtp_rtcp/source/nack_rtx_unittest.cc
+++ b/modules/rtp_rtcp/source/nack_rtx_unittest.cc
@@ -15,9 +15,9 @@
 #include <map>
 #include <memory>
 #include <set>
+#include <span>
 
 #include "absl/algorithm/container.h"
-#include "api/array_view.h"
 #include "api/call/transport.h"
 #include "api/environment/environment.h"
 #include "api/environment/environment_factory.h"
@@ -90,7 +90,7 @@ class RtxLoopBackTransport : public Transport {
     packet_loss_ = 0;
   }
 
-  bool SendRtp(ArrayView<const uint8_t> data,
+  bool SendRtp(std::span<const uint8_t> data,
                const PacketOptions& /* options */) override {
     count_++;
     RtpPacketReceived packet;
@@ -115,7 +115,7 @@ class RtxLoopBackTransport : public Transport {
     return true;
   }
 
-  bool SendRtcp(ArrayView<const uint8_t> data,
+  bool SendRtcp(std::span<const uint8_t> data,
                 const PacketOptions& /* options */) override {
     module_->IncomingRtcpPacket(data);
     return true;
diff --git a/modules/rtp_rtcp/source/rtcp_packet.cc b/modules/rtp_rtcp/source/rtcp_packet.cc
index 242ec075f3..66cde51a0e 100644
--- a/modules/rtp_rtcp/source/rtcp_packet.cc
+++ b/modules/rtp_rtcp/source/rtcp_packet.cc
@@ -12,8 +12,8 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/checks.h"
@@ -24,7 +24,7 @@ namespace rtcp {
 Buffer RtcpPacket::Build() const {
   Buffer packet = Buffer::CreateWithCapacity(BlockLength());
 
-  packet.AppendData(BlockLength(), [&](ArrayView<uint8_t> data) {
+  packet.AppendData(BlockLength(), [&](std::span<uint8_t> data) {
     size_t length = 0;
     bool created = Create(data.data(), &length, data.size(), nullptr);
     RTC_DCHECK(created) << "Invalid packet is not supported.";
@@ -50,7 +50,7 @@ bool RtcpPacket::OnBufferFull(uint8_t* packet,
   if (*index == 0)
     return false;
   RTC_DCHECK(callback) << "Fragmentation not supported.";
-  callback(ArrayView<const uint8_t>(packet, *index));
+  callback(std::span<const uint8_t>(packet, *index));
   *index = 0;
   return true;
 }
diff --git a/modules/rtp_rtcp/source/rtcp_packet.h b/modules/rtp_rtcp/source/rtcp_packet.h
index 079414694c..988d49f751 100644
--- a/modules/rtp_rtcp/source/rtcp_packet.h
+++ b/modules/rtp_rtcp/source/rtcp_packet.h
@@ -14,7 +14,8 @@
 #include <stddef.h>
 #include <stdint.h>
 
-#include "api/array_view.h"
+#include <span>
+
 #include "api/function_view.h"
 #include "rtc_base/buffer.h"
 
@@ -54,7 +55,7 @@ class RtcpPacket {
   // max_length bytes, it will be fragmented and multiple calls to this
   // callback will be made.
   using PacketReadyCallback =
-      FunctionView<void(ArrayView<const uint8_t> packet)>;
+      FunctionView<void(std::span<const uint8_t> packet)>;
 
   virtual ~RtcpPacket() = default;
 
diff --git a/modules/rtp_rtcp/source/rtcp_packet/compound_packet_unittest.cc b/modules/rtp_rtcp/source/rtcp_packet/compound_packet_unittest.cc
index e31c330231..5c4bfe3f0b 100644
--- a/modules/rtp_rtcp/source/rtcp_packet/compound_packet_unittest.cc
+++ b/modules/rtp_rtcp/source/rtcp_packet/compound_packet_unittest.cc
@@ -13,9 +13,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <utility>
 
-#include "api/array_view.h"
 #include "modules/rtp_rtcp/source/rtcp_packet.h"
 #include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
 #include "modules/rtp_rtcp/source/rtcp_packet/fir.h"
@@ -108,8 +108,8 @@ TEST(RtcpCompoundPacketTest, BuildWithInputBuffer) {
   const size_t kFirLength = 20;
 
   const size_t kBufferSize = kRrLength + kReportBlockLength + kFirLength;
-  MockFunction<void(ArrayView<const uint8_t>)> callback;
-  EXPECT_CALL(callback, Call(_)).WillOnce([&](ArrayView<const uint8_t> packet) {
+  MockFunction<void(std::span<const uint8_t>)> callback;
+  EXPECT_CALL(callback, Call(_)).WillOnce([&](std::span<const uint8_t> packet) {
     RtcpPacketParser parser;
     parser.Parse(packet);
     EXPECT_EQ(1, parser.receiver_report()->num_packets());
@@ -135,16 +135,16 @@ TEST(RtcpCompoundPacketTest, BuildWithTooSmallBuffer_FragmentedSend) {
   const size_t kReportBlockLength = 24;
 
   const size_t kBufferSize = kRrLength + kReportBlockLength;
-  MockFunction<void(ArrayView<const uint8_t>)> callback;
+  MockFunction<void(std::span<const uint8_t>)> callback;
   EXPECT_CALL(callback, Call(_))
-      .WillOnce([&](ArrayView<const uint8_t> packet) {
+      .WillOnce([&](std::span<const uint8_t> packet) {
         RtcpPacketParser parser;
         parser.Parse(packet);
         EXPECT_EQ(1, parser.receiver_report()->num_packets());
         EXPECT_EQ(1U, parser.receiver_report()->report_blocks().size());
         EXPECT_EQ(0, parser.fir()->num_packets());
       })
-      .WillOnce([&](ArrayView<const uint8_t> packet) {
+      .WillOnce([&](std::span<const uint8_t> packet) {
         RtcpPacketParser parser;
         parser.Parse(packet);
         EXPECT_EQ(0, parser.receiver_report()->num_packets());
diff --git a/modules/rtp_rtcp/source/rtcp_packet/congestion_control_feedback.cc b/modules/rtp_rtcp/source/rtcp_packet/congestion_control_feedback.cc
index 68722aec27..abe858f7bd 100644
--- a/modules/rtp_rtcp/source/rtcp_packet/congestion_control_feedback.cc
+++ b/modules/rtp_rtcp/source/rtcp_packet/congestion_control_feedback.cc
@@ -13,10 +13,10 @@
 #include <algorithm>
 #include <cstddef>
 #include <cstdint>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/transport/ecn_marking.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
@@ -166,7 +166,7 @@ bool CongestionControlFeedback::Create(uint8_t* buffer,
   //   |R|ECN|  Arrival time offset    | ...                           .
   //   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   //   .                                                               .
-  auto write_report_for_ssrc = [&](ArrayView<const PacketInfo> packets) {
+  auto write_report_for_ssrc = [&](std::span<const PacketInfo> packets) {
     // SSRC of nth RTP stream.
     ByteWriter<uint32_t>::WriteBigEndian(&buffer[*position], packets[0].ssrc);
     *position += 4;
@@ -211,7 +211,7 @@ bool CongestionControlFeedback::Create(uint8_t* buffer,
     }
   };
 
-  ArrayView<const PacketInfo> remaining(packets_);
+  std::span<const PacketInfo> remaining(packets_);
   while (!remaining.empty()) {
     int number_of_packets_for_ssrc = 0;
     uint32_t ssrc = remaining[0].ssrc;
diff --git a/modules/rtp_rtcp/source/rtcp_packet/congestion_control_feedback.h b/modules/rtp_rtcp/source/rtcp_packet/congestion_control_feedback.h
index da9beb9e00..2d2b0771b5 100644
--- a/modules/rtp_rtcp/source/rtcp_packet/congestion_control_feedback.h
+++ b/modules/rtp_rtcp/source/rtcp_packet/congestion_control_feedback.h
@@ -12,9 +12,9 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/transport/ecn_marking.h"
 #include "api/units/time_delta.h"
 #include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
@@ -50,7 +50,7 @@ class CongestionControlFeedback : public Rtpfb {
 
   bool Parse(const CommonHeader& packet);
 
-  ArrayView<const PacketInfo> packets() const { return packets_; }
+  std::span<const PacketInfo> packets() const { return packets_; }
 
   uint32_t report_timestamp_compact_ntp() const {
     return report_timestamp_compact_ntp_;
diff --git a/modules/rtp_rtcp/source/rtcp_packet/congestion_control_feedback_unittest.cc b/modules/rtp_rtcp/source/rtcp_packet/congestion_control_feedback_unittest.cc
index e7b023b265..ea085cd747 100644
--- a/modules/rtp_rtcp/source/rtcp_packet/congestion_control_feedback_unittest.cc
+++ b/modules/rtp_rtcp/source/rtcp_packet/congestion_control_feedback_unittest.cc
@@ -12,10 +12,10 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/function_view.h"
 #include "api/transport/ecn_marking.h"
 #include "api/units/time_delta.h"
@@ -130,9 +130,9 @@ TEST(CongestionControlFeedbackTest, CreateReturnsTrueForBasicPacket) {
   CongestionControlFeedback fb(std::move(packets), /*compact_ntp_timestamp=*/1);
 
   Buffer buf = Buffer::CreateWithCapacity(fb.BlockLength());
-  buf.AppendData(fb.BlockLength(), [&](ArrayView<uint8_t> buf_view) {
+  buf.AppendData(fb.BlockLength(), [&](std::span<uint8_t> buf_view) {
     size_t position = 0;
-    FunctionView<void(ArrayView<const uint8_t> packet)> callback;
+    FunctionView<void(std::span<const uint8_t> packet)> callback;
     EXPECT_TRUE(
         fb.Create(buf_view.data(), &position, buf_view.size(), callback));
     return position;
diff --git a/modules/rtp_rtcp/source/rtcp_packet/nack_unittest.cc b/modules/rtp_rtcp/source/rtcp_packet/nack_unittest.cc
index 3f00d5d16d..3530bbc9c6 100644
--- a/modules/rtp_rtcp/source/rtcp_packet/nack_unittest.cc
+++ b/modules/rtp_rtcp/source/rtcp_packet/nack_unittest.cc
@@ -13,8 +13,8 @@
 #include <cstddef>
 #include <cstdint>
 #include <iterator>
+#include <span>
 
-#include "api/array_view.h"
 #include "rtc_base/buffer.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
@@ -133,16 +133,16 @@ TEST(RtcpPacketNackTest, CreateFragmented) {
 
   const size_t kBufferSize = 12 + (3 * 4);  // Fits common header + 3 nack items
 
-  MockFunction<void(ArrayView<const uint8_t>)> callback;
+  MockFunction<void(std::span<const uint8_t>)> callback;
   EXPECT_CALL(callback, Call(_))
-      .WillOnce([&](ArrayView<const uint8_t> packet) {
+      .WillOnce([&](std::span<const uint8_t> packet) {
         Nack nack;
         EXPECT_TRUE(test::ParseSinglePacket(packet, &nack));
         EXPECT_EQ(kSenderSsrc, nack.sender_ssrc());
         EXPECT_EQ(kRemoteSsrc, nack.media_ssrc());
         EXPECT_THAT(nack.packet_ids(), ElementsAre(1, 100, 200));
       })
-      .WillOnce([&](ArrayView<const uint8_t> packet) {
+      .WillOnce([&](std::span<const uint8_t> packet) {
         Nack nack;
         EXPECT_TRUE(test::ParseSinglePacket(packet, &nack));
         EXPECT_EQ(kSenderSsrc, nack.sender_ssrc());
@@ -161,7 +161,7 @@ TEST(RtcpPacketNackTest, CreateFailsWithTooSmallBuffer) {
   nack.SetMediaSsrc(kRemoteSsrc);
   nack.SetPacketIds(kSmallList, std::size(kSmallList));
 
-  MockFunction<void(ArrayView<const uint8_t>)> callback;
+  MockFunction<void(std::span<const uint8_t>)> callback;
   EXPECT_CALL(callback, Call(_)).Times(0);
   EXPECT_FALSE(nack.Build(kMinNackBlockSize - 1, callback.AsStdFunction()));
 }
diff --git a/modules/rtp_rtcp/source/rtcp_packet/remote_estimate.cc b/modules/rtp_rtcp/source/rtcp_packet/remote_estimate.cc
index 708aafc973..dd58f5b96a 100644
--- a/modules/rtp_rtcp/source/rtcp_packet/remote_estimate.cc
+++ b/modules/rtp_rtcp/source/rtcp_packet/remote_estimate.cc
@@ -12,10 +12,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <functional>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/transport/network_types.h"
 #include "api/units/data_rate.h"
 #include "modules/rtp_rtcp/source/byte_io.h"
@@ -88,14 +88,14 @@ class RemoteEstimateSerializerImpl : public RemoteEstimateSerializer {
     size_t max_size = fields_.size() * kFieldSize;
     Buffer buf = Buffer::CreateWithCapacity(max_size);
     for (const auto& field : fields_) {
-      buf.AppendData(kFieldSize, [&](ArrayView<uint8_t> dst) {
+      buf.AppendData(kFieldSize, [&](std::span<uint8_t> dst) {
         return field.Write(src, dst.data()) ? kFieldSize : 0;
       });
     }
     return buf;
   }
 
-  bool Parse(ArrayView<const uint8_t> src,
+  bool Parse(std::span<const uint8_t> src,
              NetworkStateEstimate* target) const override {
     if (src.size() % kFieldSize != 0)
       return false;
diff --git a/modules/rtp_rtcp/source/rtcp_packet/remote_estimate.h b/modules/rtp_rtcp/source/rtcp_packet/remote_estimate.h
index 4c8c29405f..ed596f4d30 100644
--- a/modules/rtp_rtcp/source/rtcp_packet/remote_estimate.h
+++ b/modules/rtp_rtcp/source/rtcp_packet/remote_estimate.h
@@ -11,8 +11,8 @@
 #define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REMOTE_ESTIMATE_H_
 
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/transport/network_types.h"
 #include "api/units/time_delta.h"
 #include "modules/rtp_rtcp/source/rtcp_packet/app.h"
@@ -24,7 +24,7 @@ namespace rtcp {
 class CommonHeader;
 class RemoteEstimateSerializer {
  public:
-  virtual bool Parse(ArrayView<const uint8_t> src,
+  virtual bool Parse(std::span<const uint8_t> src,
                      NetworkStateEstimate* target) const = 0;
   virtual Buffer Serialize(const NetworkStateEstimate& src) const = 0;
   virtual ~RemoteEstimateSerializer() = default;
diff --git a/modules/rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc b/modules/rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc
index deda519ade..2894aab98e 100644
--- a/modules/rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc
+++ b/modules/rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc
@@ -16,10 +16,10 @@
 #include <limits>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
 #include "modules/rtp_rtcp/source/byte_io.h"
@@ -62,7 +62,7 @@ MATCHER(IsValidFeedback, "") {
          feedback.Parse(rtcp_header);
 }
 
-TransportFeedback Parse(ArrayView<const uint8_t> buffer) {
+TransportFeedback Parse(std::span<const uint8_t> buffer) {
   rtcp::CommonHeader header;
   EXPECT_TRUE(header.Parse(buffer.data(), buffer.size()));
   EXPECT_EQ(header.type(), TransportFeedback::kPacketType);
@@ -86,8 +86,8 @@ class FeedbackTester {
 
   void WithDefaultDelta(TimeDelta delta) { default_delta_ = delta; }
 
-  void WithInput(ArrayView<const uint16_t> received_seq,
-                 ArrayView<const Timestamp> received_ts = {}) {
+  void WithInput(std::span<const uint16_t> received_seq,
+                 std::span<const Timestamp> received_ts = {}) {
     std::vector<Timestamp> temp_timestamps;
     if (received_ts.empty()) {
       temp_timestamps = GenerateReceiveTimestamps(received_seq);
@@ -153,7 +153,7 @@ class FeedbackTester {
   }
 
   std::vector<Timestamp> GenerateReceiveTimestamps(
-      ArrayView<const uint16_t> seq_nums) {
+      std::span<const uint16_t> seq_nums) {
     RTC_CHECK(!seq_nums.empty());
     uint16_t last_seq = seq_nums[0];
     Timestamp time = Timestamp::Zero();
diff --git a/modules/rtp_rtcp/source/rtcp_packet_unittest.cc b/modules/rtp_rtcp/source/rtcp_packet_unittest.cc
index 793535f0e4..101d61b8b3 100644
--- a/modules/rtp_rtcp/source/rtcp_packet_unittest.cc
+++ b/modules/rtp_rtcp/source/rtcp_packet_unittest.cc
@@ -12,8 +12,8 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
 #include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
 #include "test/gmock.h"
@@ -38,7 +38,7 @@ TEST(RtcpPacketTest, BuildWithTooSmallBuffer) {
   const size_t kReportBlockLength = 24;
 
   // No packet.
-  MockFunction<void(webrtc::ArrayView<const uint8_t>)> callback;
+  MockFunction<void(std::span<const uint8_t>)> callback;
   EXPECT_CALL(callback, Call(_)).Times(0);
   const size_t kBufferSize = kRrLength + kReportBlockLength - 1;
   EXPECT_FALSE(rr.Build(kBufferSize, callback.AsStdFunction()));
diff --git a/modules/rtp_rtcp/source/rtcp_receiver.cc b/modules/rtp_rtcp/source/rtcp_receiver.cc
index 8f5146d61f..72b5c35ffa 100644
--- a/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -19,12 +19,12 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
 #include "absl/algorithm/container.h"
 #include "absl/base/attributes.h"
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/field_trials_view.h"
 #include "api/sequence_checker.h"
@@ -184,7 +184,7 @@ RTCPReceiver::RTCPReceiver(const Environment& env,
 
 RTCPReceiver::~RTCPReceiver() {}
 
-void RTCPReceiver::IncomingPacket(ArrayView<const uint8_t> packet) {
+void RTCPReceiver::IncomingPacket(std::span<const uint8_t> packet) {
   if (packet.empty()) {
     RTC_LOG(LS_WARNING) << "Incoming empty RTCP packet";
     return;
@@ -347,7 +347,7 @@ std::vector<ReportBlockData> RTCPReceiver::GetLatestReportBlockData() const {
   return result;
 }
 
-bool RTCPReceiver::ParseCompoundPacket(ArrayView<const uint8_t> packet,
+bool RTCPReceiver::ParseCompoundPacket(std::span<const uint8_t> packet,
                                        PacketInformation* packet_information) {
   MutexLock lock(&rtcp_receiver_lock_);
 
diff --git a/modules/rtp_rtcp/source/rtcp_receiver.h b/modules/rtp_rtcp/source/rtcp_receiver.h
index 555269ae24..09b6f587d1 100644
--- a/modules/rtp_rtcp/source/rtcp_receiver.h
+++ b/modules/rtp_rtcp/source/rtcp_receiver.h
@@ -16,10 +16,10 @@
 #include <list>
 #include <map>
 #include <optional>
+#include <span>
 #include <vector>
 
 #include "absl/container/inlined_vector.h"
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/sequence_checker.h"
 #include "api/units/time_delta.h"
@@ -58,7 +58,7 @@ class RTCPReceiver final {
     virtual void OnReceivedNack(
         const std::vector<uint16_t>& nack_sequence_numbers) = 0;
     virtual void OnReceivedRtcpReportBlocks(
-        ArrayView<const ReportBlockData> report_blocks) = 0;
+        std::span<const ReportBlockData> report_blocks) = 0;
 
    protected:
     virtual ~ModuleRtpRtcp() = default;
@@ -99,7 +99,7 @@ class RTCPReceiver final {
 
   ~RTCPReceiver();
 
-  void IncomingPacket(ArrayView<const uint8_t> packet);
+  void IncomingPacket(std::span<const uint8_t> packet);
 
   int64_t LastReceivedReportBlockMs() const;
 
@@ -231,7 +231,7 @@ class RTCPReceiver final {
     size_t num_rtts_ = 0;
   };
 
-  bool ParseCompoundPacket(ArrayView<const uint8_t> packet,
+  bool ParseCompoundPacket(std::span<const uint8_t> packet,
                            PacketInformation* packet_information);
 
   void TriggerCallbacksFromRtcpPacket(
diff --git a/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc b/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
index 4c91dc46b9..cd724e6b83 100644
--- a/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
+++ b/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
@@ -15,12 +15,12 @@
 #include <memory>
 #include <optional>
 #include <set>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/environment/environment_factory.h"
 #include "api/transport/network_types.h"
 #include "api/units/data_rate.h"
@@ -126,7 +126,7 @@ class MockModuleRtpRtcp : public RTCPReceiver::ModuleRtpRtcp {
   MOCK_METHOD(void, OnReceivedNack, (const std::vector<uint16_t>&), (override));
   MOCK_METHOD(void,
               OnReceivedRtcpReportBlocks,
-              (ArrayView<const ReportBlockData>),
+              (std::span<const ReportBlockData>),
               (override));
 };
 
diff --git a/modules/rtp_rtcp/source/rtcp_sender.cc b/modules/rtp_rtcp/source/rtcp_sender.cc
index d7a0c520c3..c26d3659a7 100644
--- a/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -15,12 +15,12 @@
 #include <cstring>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/call/transport.h"
 #include "api/environment/environment.h"
 #include "api/rtc_event_log/rtc_event_log.h"
@@ -87,7 +87,7 @@ class RTCPSender::PacketSender {
   // Sends pending rtcp packet.
   void Send() {
     if (index_ > 0) {
-      callback_(ArrayView<const uint8_t>(buffer_, index_));
+      callback_(std::span<const uint8_t>(buffer_, index_));
       index_ = 0;
     }
   }
@@ -115,12 +115,12 @@ RTCPSender::FeedbackState::~FeedbackState() = default;
 class RTCPSender::RtcpContext {
  public:
   RtcpContext(const FeedbackState& feedback_state,
-              ArrayView<const uint16_t> nacks,
+              std::span<const uint16_t> nacks,
               Timestamp now)
       : feedback_state_(feedback_state), nacks_(nacks), now_(now) {}
 
   const FeedbackState& feedback_state_;
-  const ArrayView<const uint16_t> nacks_;
+  const std::span<const uint16_t> nacks_;
   const Timestamp now_;
 };
 
@@ -215,7 +215,7 @@ int32_t RTCPSender::SendLossNotification(const FeedbackState& feedback_state,
                                          bool decodability_flag,
                                          bool buffering_allowed) {
   int32_t error_code = -1;
-  auto callback = [&](ArrayView<const uint8_t> packet) {
+  auto callback = [&](std::span<const uint8_t> packet) {
     transport_->SendRtcp(packet, /*packet_options=*/{});
     error_code = 0;
     env_.event_log().Log(std::make_unique<RtcEventRtcpPacketOutgoing>(packet));
@@ -569,9 +569,9 @@ void RTCPSender::BuildExtendedReports(const RtcpContext& ctx,
 
 int32_t RTCPSender::SendRTCP(const FeedbackState& feedback_state,
                              RTCPPacketType packet_type,
-                             ArrayView<const uint16_t> nacks) {
+                             std::span<const uint16_t> nacks) {
   int32_t error_code = -1;
-  auto callback = [&](ArrayView<const uint8_t> packet) {
+  auto callback = [&](std::span<const uint8_t> packet) {
     if (transport_->SendRtcp(packet, /*packet_options=*/{})) {
       error_code = 0;
       env_.event_log().Log(
@@ -605,7 +605,7 @@ uint32_t RTCPSender::ComputeSsrc() const {
 std::optional<int32_t> RTCPSender::ComputeCompoundRTCPPacket(
     const FeedbackState& feedback_state,
     RTCPPacketType packet_type,
-    ArrayView<const uint16_t> nacks,
+    std::span<const uint16_t> nacks,
     PacketSender& sender) {
   if (method_ == RtcpMode::kOff) {
     RTC_LOG(LS_WARNING) << "Can't send RTCP if it is disabled.";
@@ -876,7 +876,7 @@ void RTCPSender::SendCombinedRtcpPacket(
     ssrc = ComputeSsrc();
   }
   RTC_DCHECK_LE(max_packet_size, IP_PACKET_SIZE);
-  auto callback = [&](ArrayView<const uint8_t> packet) {
+  auto callback = [&](std::span<const uint8_t> packet) {
     if (transport_->SendRtcp(packet, /*packet_options=*/{})) {
       env_.event_log().Log(
           std::make_unique<RtcEventRtcpPacketOutgoing>(packet));
diff --git a/modules/rtp_rtcp/source/rtcp_sender.h b/modules/rtp_rtcp/source/rtcp_sender.h
index 85d25f5fd7..7fa3ce9a33 100644
--- a/modules/rtp_rtcp/source/rtcp_sender.h
+++ b/modules/rtp_rtcp/source/rtcp_sender.h
@@ -17,12 +17,12 @@
 #include <memory>
 #include <optional>
 #include <set>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/call/transport.h"
 #include "api/environment/environment.h"
 #include "api/rtp_headers.h"
@@ -140,7 +140,7 @@ class RTCPSender final {
 
   int32_t SendRTCP(const FeedbackState& feedback_state,
                    RTCPPacketType packetType,
-                   ArrayView<const uint16_t> nacks = {})
+                   std::span<const uint16_t> nacks = {})
       RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
 
   int32_t SendLossNotification(const FeedbackState& feedback_state,
@@ -183,7 +183,7 @@ class RTCPSender final {
   std::optional<int32_t> ComputeCompoundRTCPPacket(
       const FeedbackState& feedback_state,
       RTCPPacketType packet_type,
-      ArrayView<const uint16_t> nacks,
+      std::span<const uint16_t> nacks,
       PacketSender& sender) RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
 
   TimeDelta ComputeTimeUntilNextReport(DataRate send_bitrate)
diff --git a/modules/rtp_rtcp/source/rtcp_sender_unittest.cc b/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
index 0b4d30a7ee..abe4761bd0 100644
--- a/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
+++ b/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
@@ -14,10 +14,10 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/call/transport.h"
 #include "api/environment/environment.h"
 #include "api/environment/environment_factory.h"
@@ -75,11 +75,11 @@ class TestTransport : public Transport {
  public:
   TestTransport() {}
 
-  bool SendRtp(ArrayView<const uint8_t> /*data*/,
+  bool SendRtp(std::span<const uint8_t> /*data*/,
                const PacketOptions& /* options */) override {
     return false;
   }
-  bool SendRtcp(ArrayView<const uint8_t> data,
+  bool SendRtcp(std::span<const uint8_t> data,
                 const PacketOptions& options) override {
     EXPECT_FALSE(options.is_media);
     parser_.Parse(data);
@@ -669,7 +669,7 @@ TEST_F(RtcpSenderTest, SendsTmmbnIfSetAndEmpty) {
 TEST_F(RtcpSenderTest, ByeMustBeLast) {
   MockTransport mock_transport;
   EXPECT_CALL(mock_transport, SendRtcp(_, _))
-      .WillOnce([](ArrayView<const uint8_t> data, ::testing::Unused) {
+      .WillOnce([](std::span<const uint8_t> data, ::testing::Unused) {
         const uint8_t* next_packet = data.data();
         const uint8_t* const packet_end = data.data() + data.size();
         rtcp::CommonHeader packet;
diff --git a/modules/rtp_rtcp/source/rtcp_transceiver_config.h b/modules/rtp_rtcp/source/rtcp_transceiver_config.h
index 37500fb13f..8fc7d6374a 100644
--- a/modules/rtp_rtcp/source/rtcp_transceiver_config.h
+++ b/modules/rtp_rtcp/source/rtcp_transceiver_config.h
@@ -14,9 +14,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <functional>
+#include <span>
 #include <string>
 
-#include "api/array_view.h"
 #include "api/rtp_headers.h"
 #include "api/task_queue/task_queue_base.h"
 #include "api/units/time_delta.h"
@@ -81,7 +81,7 @@ class RtpStreamRtcpHandler {
   virtual RtpStats SentStats() = 0;
 
   virtual void OnNack(uint32_t /* sender_ssrc */,
-                      ArrayView<const uint16_t> /* sequence_numbers */) {}
+                      std::span<const uint16_t> /* sequence_numbers */) {}
   virtual void OnFir(uint32_t /* sender_ssrc */) {}
   virtual void OnPli(uint32_t /* sender_ssrc */) {}
 
@@ -117,7 +117,7 @@ struct RtcpTransceiverConfig {
   Clock* clock = nullptr;
 
   // Transport to send RTCP packets to.
-  std::function<void(ArrayView<const uint8_t>)> rtcp_transport;
+  std::function<void(std::span<const uint8_t>)> rtcp_transport;
 
   // Queue for scheduling delayed tasks, e.g. sending periodic compound packets.
   TaskQueueBase* task_queue = nullptr;
diff --git a/modules/rtp_rtcp/source/rtcp_transceiver_impl.cc b/modules/rtp_rtcp/source/rtcp_transceiver_impl.cc
index e854511462..41144156a6 100644
--- a/modules/rtp_rtcp/source/rtcp_transceiver_impl.cc
+++ b/modules/rtp_rtcp/source/rtcp_transceiver_impl.cc
@@ -17,12 +17,12 @@
 #include <iterator>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/algorithm/container.h"
-#include "api/array_view.h"
 #include "api/rtp_headers.h"
 #include "api/task_queue/task_queue_base.h"
 #include "api/units/data_rate.h"
@@ -70,7 +70,7 @@ struct SenderReportTimes {
   NtpTime remote_sent_time;
 };
 
-std::function<void(ArrayView<const uint8_t>)> GetRtcpTransport(
+std::function<void(std::span<const uint8_t>)> GetRtcpTransport(
     const RtcpTransceiverConfig& config) {
   if (config.rtcp_transport != nullptr) {
     return config.rtcp_transport;
@@ -78,7 +78,7 @@ std::function<void(ArrayView<const uint8_t>)> GetRtcpTransport(
 
   bool first = true;
   std::string log_prefix = config.debug_id;
-  return [first, log_prefix](ArrayView<const uint8_t> /* packet */) mutable {
+  return [first, log_prefix](std::span<const uint8_t> /* packet */) mutable {
     if (first) {
       RTC_LOG(LS_ERROR) << log_prefix << "Sending RTCP packets is disabled.";
       first = false;
@@ -126,7 +126,7 @@ class RtcpTransceiverImpl::PacketSender {
   // Sends pending rtcp compound packet.
   void Send() {
     if (index_ > 0) {
-      callback_(ArrayView<const uint8_t>(buffer_, index_));
+      callback_(std::span<const uint8_t>(buffer_, index_));
       index_ = 0;
     }
   }
@@ -215,7 +215,7 @@ void RtcpTransceiverImpl::SetReadyToSend(bool ready) {
   ready_to_send_ = ready;
 }
 
-void RtcpTransceiverImpl::ReceivePacket(ArrayView<const uint8_t> packet,
+void RtcpTransceiverImpl::ReceivePacket(std::span<const uint8_t> packet,
                                         Timestamp now) {
   // Report blocks may be spread across multiple sender and receiver reports.
   std::vector<ReportBlockData> report_blocks;
@@ -288,7 +288,7 @@ void RtcpTransceiverImpl::SendPictureLossIndication(uint32_t ssrc) {
   SendImmediateFeedback(pli);
 }
 
-void RtcpTransceiverImpl::SendFullIntraRequest(ArrayView<const uint32_t> ssrcs,
+void RtcpTransceiverImpl::SendFullIntraRequest(std::span<const uint32_t> ssrcs,
                                                bool new_request) {
   RTC_DCHECK(!ssrcs.empty());
   if (!ready_to_send_)
@@ -377,7 +377,7 @@ void RtcpTransceiverImpl::HandleReceiverReport(
 void RtcpTransceiverImpl::HandleReportBlocks(
     uint32_t sender_ssrc,
     Timestamp now,
-    ArrayView<const rtcp::ReportBlock> rtcp_report_blocks,
+    std::span<const rtcp::ReportBlock> rtcp_report_blocks,
     std::vector<ReportBlockData>& report_blocks) {
   if (rtcp_report_blocks.empty()) {
     return;
@@ -575,7 +575,7 @@ void RtcpTransceiverImpl::HandleDlrr(const rtcp::Dlrr& dlrr, Timestamp now) {
 
 void RtcpTransceiverImpl::ProcessReportBlocks(
     Timestamp now,
-    ArrayView<const ReportBlockData> report_blocks) {
+    std::span<const ReportBlockData> report_blocks) {
   RTC_DCHECK(!report_blocks.empty());
   if (config_.network_link_observer == nullptr) {
     return;
@@ -837,7 +837,7 @@ void RtcpTransceiverImpl::CreateCompoundPacket(Timestamp now,
     sender.AppendPacket(xr_with_rrtr);
   }
   if (xr_with_dlrr.has_value()) {
-    ArrayView<const uint32_t> ssrcs(&sender_ssrc, 1);
+    std::span<const uint32_t> ssrcs(&sender_ssrc, 1);
     if (config_.reply_to_non_sender_rtt_mesaurments_on_all_ssrcs &&
         !sender_ssrcs.empty()) {
       ssrcs = sender_ssrcs;
diff --git a/modules/rtp_rtcp/source/rtcp_transceiver_impl.h b/modules/rtp_rtcp/source/rtcp_transceiver_impl.h
index adc9cc5283..c7249d30c5 100644
--- a/modules/rtp_rtcp/source/rtcp_transceiver_impl.h
+++ b/modules/rtp_rtcp/source/rtcp_transceiver_impl.h
@@ -17,9 +17,9 @@
 #include <list>
 #include <memory>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
 #include "modules/rtp_rtcp/include/report_block_data.h"
@@ -59,7 +59,7 @@ class RtcpTransceiverImpl {
 
   void SetReadyToSend(bool ready);
 
-  void ReceivePacket(ArrayView<const uint8_t> packet, Timestamp now);
+  void ReceivePacket(std::span<const uint8_t> packet, Timestamp now);
 
   void SendCompoundPacket();
 
@@ -71,7 +71,7 @@ class RtcpTransceiverImpl {
   void SendPictureLossIndication(uint32_t ssrc);
   // If new_request is true then requested sequence no. will increase for each
   // requested ssrc.
-  void SendFullIntraRequest(ArrayView<const uint32_t> ssrcs, bool new_request);
+  void SendFullIntraRequest(std::span<const uint32_t> ssrcs, bool new_request);
 
   // SendCombinedRtcpPacket ignores rtcp mode and does not send a compound
   // message. https://tools.ietf.org/html/rfc4585#section-3.1
@@ -103,7 +103,7 @@ class RtcpTransceiverImpl {
                             std::vector<ReportBlockData>& report_blocks);
   void HandleReportBlocks(uint32_t sender_ssrc,
                           Timestamp now,
-                          ArrayView<const rtcp::ReportBlock> rtcp_report_blocks,
+                          std::span<const rtcp::ReportBlock> rtcp_report_blocks,
                           std::vector<ReportBlockData>& report_blocks);
   void HandlePayloadSpecificFeedback(
       const rtcp::CommonHeader& rtcp_packet_header,
@@ -126,7 +126,7 @@ class RtcpTransceiverImpl {
   void HandleTargetBitrate(const rtcp::TargetBitrate& target_bitrate,
                            uint32_t remote_ssrc);
   void ProcessReportBlocks(Timestamp now,
-                           ArrayView<const ReportBlockData> report_blocks);
+                           std::span<const ReportBlockData> report_blocks);
 
   void ReschedulePeriodicCompoundPackets();
   void SchedulePeriodicCompoundPackets(TimeDelta delay);
@@ -156,7 +156,7 @@ class RtcpTransceiverImpl {
                                                     size_t num_max_blocks);
 
   const RtcpTransceiverConfig config_;
-  std::function<void(ArrayView<const uint8_t>)> rtcp_transport_;
+  std::function<void(std::span<const uint8_t>)> rtcp_transport_;
 
   bool ready_to_send_;
   std::optional<rtcp::Remb> remb_;
diff --git a/modules/rtp_rtcp/source/rtcp_transceiver_impl_unittest.cc b/modules/rtp_rtcp/source/rtcp_transceiver_impl_unittest.cc
index aeec95fa19..4f108781f8 100644
--- a/modules/rtp_rtcp/source/rtcp_transceiver_impl_unittest.cc
+++ b/modules/rtp_rtcp/source/rtcp_transceiver_impl_unittest.cc
@@ -15,10 +15,10 @@
 #include <functional>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/rtp_headers.h"
 #include "api/task_queue/task_queue_base.h"
 #include "api/task_queue/task_queue_factory.h"
@@ -106,10 +106,7 @@ class MockRtpStreamRtcpHandler : public RtpStreamRtcpHandler {
   }
 
   MOCK_METHOD(RtpStats, SentStats, (), (override));
-  MOCK_METHOD(void,
-              OnNack,
-              (uint32_t, ArrayView<const uint16_t>),
-              (override));
+  MOCK_METHOD(void, OnNack, (uint32_t, std::span<const uint16_t>), (override));
   MOCK_METHOD(void, OnFir, (uint32_t), (override));
   MOCK_METHOD(void, OnPli, (uint32_t), (override));
   MOCK_METHOD(void, OnReport, (const ReportBlockData&), (override));
@@ -131,8 +128,8 @@ class FakeRtcpTransport {
  public:
   explicit FakeRtcpTransport(TimeController& time) : time_(time) {}
 
-  std::function<void(ArrayView<const uint8_t>)> AsStdFunction() {
-    return [this](ArrayView<const uint8_t>) { sent_rtcp_ = true; };
+  std::function<void(std::span<const uint8_t>)> AsStdFunction() {
+    return [this](std::span<const uint8_t>) { sent_rtcp_ = true; };
   }
 
   // Returns true when packet was received by the transport.
@@ -148,9 +145,9 @@ class FakeRtcpTransport {
   bool sent_rtcp_ = false;
 };
 
-std::function<void(ArrayView<const uint8_t>)> RtcpParserTransport(
+std::function<void(std::span<const uint8_t>)> RtcpParserTransport(
     RtcpPacketParser& parser) {
-  return [&parser](ArrayView<const uint8_t> packet) {
+  return [&parser](std::span<const uint8_t> packet) {
     return parser.Parse(packet);
   };
 }
@@ -343,7 +340,7 @@ TEST_F(RtcpTransceiverImplTest, SendCompoundPacketDelaysPeriodicSendPackets) {
 }
 
 TEST_F(RtcpTransceiverImplTest, SendsNoRtcpWhenNetworkStateIsDown) {
-  MockFunction<void(ArrayView<const uint8_t>)> mock_transport;
+  MockFunction<void(std::span<const uint8_t>)> mock_transport;
   RtcpTransceiverConfig config = DefaultTestConfig();
   config.initial_ready_to_send = false;
   config.rtcp_transport = mock_transport.AsStdFunction();
@@ -360,7 +357,7 @@ TEST_F(RtcpTransceiverImplTest, SendsNoRtcpWhenNetworkStateIsDown) {
 }
 
 TEST_F(RtcpTransceiverImplTest, SendsRtcpWhenNetworkStateIsUp) {
-  MockFunction<void(ArrayView<const uint8_t>)> mock_transport;
+  MockFunction<void(std::span<const uint8_t>)> mock_transport;
   RtcpTransceiverConfig config = DefaultTestConfig();
   config.initial_ready_to_send = false;
   config.rtcp_transport = mock_transport.AsStdFunction();
@@ -426,7 +423,7 @@ TEST_F(RtcpTransceiverImplTest, SendsMinimalCompoundPacket) {
 }
 
 TEST_F(RtcpTransceiverImplTest, AvoidsEmptyPacketsInReducedMode) {
-  MockFunction<void(ArrayView<const uint8_t>)> transport;
+  MockFunction<void(std::span<const uint8_t>)> transport;
   EXPECT_CALL(transport, Call).Times(0);
   NiceMock<MockReceiveStatisticsProvider> receive_statistics;
 
@@ -1632,10 +1629,10 @@ TEST_F(RtcpTransceiverImplTest, RotatesSendersWhenAllSenderReportDoNotFit) {
     rtcp_receiver.AddMediaReceiverRtcpObserver(kSenderSsrc[i], &receiver[i]);
   }
 
-  MockFunction<void(ArrayView<const uint8_t>)> transport;
+  MockFunction<void(std::span<const uint8_t>)> transport;
   EXPECT_CALL(transport, Call)
       .Times(kNumSenders)
-      .WillRepeatedly([&](ArrayView<const uint8_t> packet) {
+      .WillRepeatedly([&](std::span<const uint8_t> packet) {
         rtcp_receiver.ReceivePacket(packet, CurrentTime());
         return true;
       });
diff --git a/modules/rtp_rtcp/source/rtcp_transceiver_unittest.cc b/modules/rtp_rtcp/source/rtcp_transceiver_unittest.cc
index b5de513c68..51cc368419 100644
--- a/modules/rtp_rtcp/source/rtcp_transceiver_unittest.cc
+++ b/modules/rtp_rtcp/source/rtcp_transceiver_unittest.cc
@@ -12,10 +12,10 @@
 
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
 #include "modules/rtp_rtcp/source/rtcp_packet.h"
@@ -69,7 +69,7 @@ void WaitPostedTasks(TaskQueueForTest* queue) {
 
 TEST(RtcpTransceiverTest, SendsRtcpOnTaskQueueWhenCreatedOffTaskQueue) {
   SimulatedClock clock(0);
-  MockFunction<void(webrtc::ArrayView<const uint8_t>)> outgoing_transport;
+  MockFunction<void(std::span<const uint8_t>)> outgoing_transport;
   TaskQueueForTest queue("rtcp");
   RtcpTransceiverConfig config;
   config.clock = &clock;
@@ -87,7 +87,7 @@ TEST(RtcpTransceiverTest, SendsRtcpOnTaskQueueWhenCreatedOffTaskQueue) {
 
 TEST(RtcpTransceiverTest, SendsRtcpOnTaskQueueWhenCreatedOnTaskQueue) {
   SimulatedClock clock(0);
-  MockFunction<void(webrtc::ArrayView<const uint8_t>)> outgoing_transport;
+  MockFunction<void(std::span<const uint8_t>)> outgoing_transport;
   TaskQueueForTest queue("rtcp");
   RtcpTransceiverConfig config;
   config.clock = &clock;
@@ -108,7 +108,7 @@ TEST(RtcpTransceiverTest, SendsRtcpOnTaskQueueWhenCreatedOnTaskQueue) {
 
 TEST(RtcpTransceiverTest, CanBeDestroyedOnTaskQueue) {
   SimulatedClock clock(0);
-  MockFunction<void(webrtc::ArrayView<const uint8_t>)> outgoing_transport;
+  MockFunction<void(std::span<const uint8_t>)> outgoing_transport;
   TaskQueueForTest queue("rtcp");
   RtcpTransceiverConfig config;
   config.clock = &clock;
@@ -148,7 +148,7 @@ TEST(RtcpTransceiverTest, CanBeDestroyedWithoutBlocking) {
 TEST(RtcpTransceiverTest, MaySendPacketsAfterDestructor) {  // i.e. Be careful!
   SimulatedClock clock(0);
   // Must outlive queue below.
-  NiceMock<MockFunction<void(webrtc::ArrayView<const uint8_t>)>> transport;
+  NiceMock<MockFunction<void(std::span<const uint8_t>)>> transport;
   TaskQueueForTest queue("rtcp");
   RtcpTransceiverConfig config;
   config.clock = &clock;
@@ -232,7 +232,7 @@ TEST(RtcpTransceiverTest, RemoveMediaReceiverRtcpObserverIsNonBlocking) {
 
 TEST(RtcpTransceiverTest, CanCallSendCompoundPacketFromAnyThread) {
   SimulatedClock clock(0);
-  MockFunction<void(webrtc::ArrayView<const uint8_t>)> outgoing_transport;
+  MockFunction<void(std::span<const uint8_t>)> outgoing_transport;
   TaskQueueForTest queue("rtcp");
   RtcpTransceiverConfig config;
   config.clock = &clock;
@@ -263,8 +263,7 @@ TEST(RtcpTransceiverTest, CanCallSendCompoundPacketFromAnyThread) {
 
 TEST(RtcpTransceiverTest, DoesntSendPacketsAfterStopCallback) {
   SimulatedClock clock(0);
-  NiceMock<MockFunction<void(webrtc::ArrayView<const uint8_t>)>>
-      outgoing_transport;
+  NiceMock<MockFunction<void(std::span<const uint8_t>)>> outgoing_transport;
   TaskQueueForTest queue("rtcp");
   RtcpTransceiverConfig config;
   config.clock = &clock;
@@ -287,7 +286,7 @@ TEST(RtcpTransceiverTest, SendsCombinedRtcpPacketOnTaskQueue) {
   static constexpr uint32_t kSenderSsrc = 12345;
 
   SimulatedClock clock(0);
-  MockFunction<void(webrtc::ArrayView<const uint8_t>)> outgoing_transport;
+  MockFunction<void(std::span<const uint8_t>)> outgoing_transport;
   TaskQueueForTest queue("rtcp");
   RtcpTransceiverConfig config;
   config.clock = &clock;
@@ -298,7 +297,7 @@ TEST(RtcpTransceiverTest, SendsCombinedRtcpPacketOnTaskQueue) {
   RtcpTransceiver rtcp_transceiver(config);
 
   EXPECT_CALL(outgoing_transport, Call)
-      .WillOnce([&](webrtc::ArrayView<const uint8_t> buffer) {
+      .WillOnce([&](std::span<const uint8_t> buffer) {
         EXPECT_TRUE(queue.IsCurrent());
         RtcpPacketParser rtcp_parser;
         rtcp_parser.Parse(buffer);
@@ -326,7 +325,7 @@ TEST(RtcpTransceiverTest, SendFrameIntraRequestDefaultsToNewRequest) {
   static constexpr uint32_t kSenderSsrc = 12345;
 
   SimulatedClock clock(0);
-  MockFunction<void(webrtc::ArrayView<const uint8_t>)> outgoing_transport;
+  MockFunction<void(std::span<const uint8_t>)> outgoing_transport;
   TaskQueueForTest queue("rtcp");
   RtcpTransceiverConfig config;
   config.clock = &clock;
@@ -338,7 +337,7 @@ TEST(RtcpTransceiverTest, SendFrameIntraRequestDefaultsToNewRequest) {
 
   uint8_t first_seq_nr;
   EXPECT_CALL(outgoing_transport, Call)
-      .WillOnce([&](webrtc::ArrayView<const uint8_t> buffer) {
+      .WillOnce([&](std::span<const uint8_t> buffer) {
         EXPECT_TRUE(queue.IsCurrent());
         RtcpPacketParser rtcp_parser;
         rtcp_parser.Parse(buffer);
@@ -346,7 +345,7 @@ TEST(RtcpTransceiverTest, SendFrameIntraRequestDefaultsToNewRequest) {
         first_seq_nr = rtcp_parser.fir()->requests()[0].seq_nr;
         return true;
       })
-      .WillOnce([&](webrtc::ArrayView<const uint8_t> buffer) {
+      .WillOnce([&](std::span<const uint8_t> buffer) {
         EXPECT_TRUE(queue.IsCurrent());
         RtcpPacketParser rtcp_parser;
         rtcp_parser.Parse(buffer);
diff --git a/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.cc b/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.cc
index 7a8b25d881..5b92c6466e 100644
--- a/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.cc
+++ b/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.cc
@@ -13,8 +13,8 @@
 #include <bitset>
 #include <cstddef>
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/transport/rtp/dependency_descriptor.h"
 #include "modules/rtp_rtcp/source/rtp_dependency_descriptor_reader.h"
 #include "modules/rtp_rtcp/source/rtp_dependency_descriptor_writer.h"
@@ -23,7 +23,7 @@
 namespace webrtc {
 
 bool RtpDependencyDescriptorExtension::Parse(
-    ArrayView<const uint8_t> data,
+    std::span<const uint8_t> data,
     const FrameDependencyStructure* structure,
     DependencyDescriptor* descriptor) {
   RtpDependencyDescriptorReader reader(data, structure, descriptor);
@@ -40,7 +40,7 @@ size_t RtpDependencyDescriptorExtension::ValueSize(
 }
 
 bool RtpDependencyDescriptorExtension::Write(
-    ArrayView<uint8_t> data,
+    std::span<uint8_t> data,
     const FrameDependencyStructure& structure,
     std::bitset<32> active_chains,
     const DependencyDescriptor& descriptor) {
@@ -50,7 +50,7 @@ bool RtpDependencyDescriptorExtension::Write(
 }
 
 bool RtpDependencyDescriptorExtension::Parse(
-    ArrayView<const uint8_t> data,
+    std::span<const uint8_t> data,
     DependencyDescriptorMandatory* descriptor) {
   if (data.size() < 3) {
     return false;
diff --git a/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h b/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h
index 97d421eb4d..c757e40d64 100644
--- a/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h
+++ b/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h
@@ -13,9 +13,9 @@
 #include <bitset>
 #include <cstddef>
 #include <cstdint>
+#include <span>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtp_parameters.h"
 #include "api/transport/rtp/dependency_descriptor.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
@@ -30,13 +30,13 @@ class RtpDependencyDescriptorExtension {
     return RtpExtension::kDependencyDescriptorUri;
   }
 
-  static bool Parse(ArrayView<const uint8_t> data,
+  static bool Parse(std::span<const uint8_t> data,
                     const FrameDependencyStructure* structure,
                     DependencyDescriptor* descriptor);
 
   // Reads the mandatory part of the descriptor.
   // Such read is stateless, i.e., doesn't require `FrameDependencyStructure`.
-  static bool Parse(ArrayView<const uint8_t> data,
+  static bool Parse(std::span<const uint8_t> data,
                     DependencyDescriptorMandatory* descriptor);
 
   static size_t ValueSize(const FrameDependencyStructure& structure,
@@ -46,12 +46,12 @@ class RtpDependencyDescriptorExtension {
   static size_t ValueSize(const FrameDependencyStructure& structure,
                           std::bitset<32> active_chains,
                           const DependencyDescriptor& descriptor);
-  static bool Write(ArrayView<uint8_t> data,
+  static bool Write(std::span<uint8_t> data,
                     const FrameDependencyStructure& structure,
                     const DependencyDescriptor& descriptor) {
     return Write(data, structure, kAllChainsAreActive, descriptor);
   }
-  static bool Write(ArrayView<uint8_t> data,
+  static bool Write(std::span<uint8_t> data,
                     const FrameDependencyStructure& structure,
                     std::bitset<32> active_chains,
                     const DependencyDescriptor& descriptor);
diff --git a/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension_unittest.cc b/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension_unittest.cc
index ec64e819af..9b054e7dc6 100644
--- a/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension_unittest.cc
@@ -14,8 +14,8 @@
 #include <cstddef>
 #include <cstdint>
 #include <cstring>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/transport/rtp/dependency_descriptor.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
@@ -64,7 +64,7 @@ TEST(RtpDependencyDescriptorExtensionTest, WriteZeroInUnusedBits) {
   const uint8_t* unused_bytes = buffer + value_size;
   size_t num_unused_bytes = buffer + sizeof(buffer) - unused_bytes;
   // Check remaining bytes are zeroed.
-  EXPECT_THAT(MakeArrayView(unused_bytes, num_unused_bytes), Each(0));
+  EXPECT_THAT(std::span(unused_bytes, num_unused_bytes), Each(0));
 }
 
 // In practice chain diff for inactive chain will grow uboundly because no
diff --git a/modules/rtp_rtcp/source/rtp_dependency_descriptor_reader.cc b/modules/rtp_rtcp/source/rtp_dependency_descriptor_reader.cc
index 2d1f06df15..ad20ce321e 100644
--- a/modules/rtp_rtcp/source/rtp_dependency_descriptor_reader.cc
+++ b/modules/rtp_rtcp/source/rtp_dependency_descriptor_reader.cc
@@ -13,10 +13,10 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/transport/rtp/dependency_descriptor.h"
 #include "rtc_base/bitstream_reader.h"
 #include "rtc_base/checks.h"
@@ -24,7 +24,7 @@
 namespace webrtc {
 
 RtpDependencyDescriptorReader::RtpDependencyDescriptorReader(
-    ArrayView<const uint8_t> raw_data,
+    std::span<const uint8_t> raw_data,
     const FrameDependencyStructure* structure,
     DependencyDescriptor* descriptor)
     : descriptor_(descriptor), buffer_(raw_data) {
diff --git a/modules/rtp_rtcp/source/rtp_dependency_descriptor_reader.h b/modules/rtp_rtcp/source/rtp_dependency_descriptor_reader.h
index a350428db0..b493fb6660 100644
--- a/modules/rtp_rtcp/source/rtp_dependency_descriptor_reader.h
+++ b/modules/rtp_rtcp/source/rtp_dependency_descriptor_reader.h
@@ -11,8 +11,8 @@
 #define MODULES_RTP_RTCP_SOURCE_RTP_DEPENDENCY_DESCRIPTOR_READER_H_
 
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/transport/rtp/dependency_descriptor.h"
 #include "rtc_base/bitstream_reader.h"
 
@@ -21,7 +21,7 @@ namespace webrtc {
 class RtpDependencyDescriptorReader {
  public:
   // Parses the dependency descriptor.
-  RtpDependencyDescriptorReader(ArrayView<const uint8_t> raw_data,
+  RtpDependencyDescriptorReader(std::span<const uint8_t> raw_data,
                                 const FrameDependencyStructure* structure,
                                 DependencyDescriptor* descriptor);
   RtpDependencyDescriptorReader(const RtpDependencyDescriptorReader&) = delete;
diff --git a/modules/rtp_rtcp/source/rtp_dependency_descriptor_writer.cc b/modules/rtp_rtcp/source/rtp_dependency_descriptor_writer.cc
index d7b309d5c6..3d12605004 100644
--- a/modules/rtp_rtcp/source/rtp_dependency_descriptor_writer.cc
+++ b/modules/rtp_rtcp/source/rtp_dependency_descriptor_writer.cc
@@ -14,10 +14,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <iterator>
+#include <span>
 #include <vector>
 
 #include "absl/algorithm/container.h"
-#include "api/array_view.h"
 #include "api/transport/rtp/dependency_descriptor.h"
 #include "api/video/render_resolution.h"
 #include "rtc_base/bit_buffer.h"
@@ -57,7 +57,7 @@ NextLayerIdc GetNextLayerIdc(const FrameDependencyTemplate& previous,
 }  // namespace
 
 RtpDependencyDescriptorWriter::RtpDependencyDescriptorWriter(
-    ArrayView<uint8_t> data,
+    std::span<uint8_t> data,
     const FrameDependencyStructure& structure,
     std::bitset<32> active_chains,
     const DependencyDescriptor& descriptor)
diff --git a/modules/rtp_rtcp/source/rtp_dependency_descriptor_writer.h b/modules/rtp_rtcp/source/rtp_dependency_descriptor_writer.h
index 4a6b362ade..1feb21ec61 100644
--- a/modules/rtp_rtcp/source/rtp_dependency_descriptor_writer.h
+++ b/modules/rtp_rtcp/source/rtp_dependency_descriptor_writer.h
@@ -13,9 +13,9 @@
 #include <bitset>
 #include <cstddef>
 #include <cstdint>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/transport/rtp/dependency_descriptor.h"
 #include "rtc_base/bit_buffer.h"
 
@@ -24,7 +24,7 @@ class RtpDependencyDescriptorWriter {
  public:
   // Assumes `structure` and `descriptor` are valid and
   // `descriptor` matches the `structure`.
-  RtpDependencyDescriptorWriter(ArrayView<uint8_t> data,
+  RtpDependencyDescriptorWriter(std::span<uint8_t> data,
                                 const FrameDependencyStructure& structure,
                                 std::bitset<32> active_chains,
                                 const DependencyDescriptor& descriptor);
diff --git a/modules/rtp_rtcp/source/rtp_format.cc b/modules/rtp_rtcp/source/rtp_format.cc
index fda60ffac9..5d3c880aff 100644
--- a/modules/rtp_rtcp/source/rtp_format.cc
+++ b/modules/rtp_rtcp/source/rtp_format.cc
@@ -12,9 +12,9 @@
 
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/rtp_rtcp/source/rtp_format_h264.h"
 #include "modules/rtp_rtcp/source/rtp_format_video_generic.h"
 #include "modules/rtp_rtcp/source/rtp_format_vp8.h"
@@ -32,7 +32,7 @@ namespace webrtc {
 
 std::unique_ptr<RtpPacketizer> RtpPacketizer::Create(
     PacketizationFormat format,
-    ArrayView<const uint8_t> payload,
+    std::span<const uint8_t> payload,
     PayloadSizeLimits limits,
     // Codec-specific details.
     const RTPVideoHeader& rtp_video_header) {
diff --git a/modules/rtp_rtcp/source/rtp_format.h b/modules/rtp_rtcp/source/rtp_format.h
index 6b850cda43..e329109e82 100644
--- a/modules/rtp_rtcp/source/rtp_format.h
+++ b/modules/rtp_rtcp/source/rtp_format.h
@@ -15,9 +15,9 @@
 
 #include <cstddef>
 #include <memory>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/rtp_rtcp/source/rtp_video_header.h"
 
 namespace webrtc {
@@ -45,7 +45,7 @@ class RtpPacketizer {
   };
   static std::unique_ptr<RtpPacketizer> Create(
       PacketizationFormat format,
-      ArrayView<const uint8_t> payload,
+      std::span<const uint8_t> payload,
       PayloadSizeLimits limits,
       // Codec-specific details.
       const RTPVideoHeader& rtp_video_header);
diff --git a/modules/rtp_rtcp/source/rtp_format_h264.cc b/modules/rtp_rtcp/source/rtp_format_h264.cc
index da2639dbd6..ba4f4feb17 100644
--- a/modules/rtp_rtcp/source/rtp_format_h264.cc
+++ b/modules/rtp_rtcp/source/rtp_format_h264.cc
@@ -13,10 +13,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <cstring>
+#include <span>
 #include <vector>
 
 #include "absl/algorithm/container.h"
-#include "api/array_view.h"
 #include "common_video/h264/h264_common.h"
 #include "modules/rtp_rtcp/source/byte_io.h"
 #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
@@ -33,7 +33,7 @@ constexpr size_t kLengthFieldSize = 2;
 
 }  // namespace
 
-RtpPacketizerH264::RtpPacketizerH264(ArrayView<const uint8_t> payload,
+RtpPacketizerH264::RtpPacketizerH264(std::span<const uint8_t> payload,
                                      PayloadSizeLimits limits,
                                      H264PacketizationMode packetization_mode)
     : limits_(limits), num_packets_left_(0) {
@@ -47,7 +47,7 @@ RtpPacketizerH264::RtpPacketizerH264(ArrayView<const uint8_t> payload,
   }
   bool has_empty_fragments = absl::c_any_of(
       input_fragments_,
-      [](const ArrayView<const uint8_t> fragment) { return fragment.empty(); });
+      [](const std::span<const uint8_t> fragment) { return fragment.empty(); });
   if (has_empty_fragments || !GeneratePackets(packetization_mode)) {
     // If empty fragments were found or we failed to generate all the packets,
     // discard already generated packets in case the caller would ignore the
@@ -100,7 +100,7 @@ bool RtpPacketizerH264::GeneratePackets(
 
 bool RtpPacketizerH264::PacketizeFuA(size_t fragment_index) {
   // Fragment payload into packets (FU-A).
-  ArrayView<const uint8_t> fragment = input_fragments_[fragment_index];
+  std::span<const uint8_t> fragment = input_fragments_[fragment_index];
 
   PayloadSizeLimits limits = limits_;
   // Leave room for the FU-A header.
@@ -151,7 +151,7 @@ size_t RtpPacketizerH264::PacketizeStapA(size_t fragment_index) {
   size_t payload_size_left = limits_.max_payload_len;
   int aggregated_fragments = 0;
   size_t fragment_headers_length = 0;
-  ArrayView<const uint8_t> fragment = input_fragments_[fragment_index];
+  std::span<const uint8_t> fragment = input_fragments_[fragment_index];
   RTC_CHECK_GE(payload_size_left, fragment.size());
   ++num_packets_left_;
 
@@ -205,7 +205,7 @@ bool RtpPacketizerH264::PacketizeSingleNalu(size_t fragment_index) {
     payload_size_left -= limits_.first_packet_reduction_len;
   else if (fragment_index + 1 == input_fragments_.size())
     payload_size_left -= limits_.last_packet_reduction_len;
-  ArrayView<const uint8_t> fragment = input_fragments_[fragment_index];
+  std::span<const uint8_t> fragment = input_fragments_[fragment_index];
   if (payload_size_left < fragment.size()) {
     RTC_LOG(LS_ERROR) << "Failed to fit a fragment to packet in SingleNalu "
                          "packetization mode. Payload size left "
@@ -257,7 +257,7 @@ void RtpPacketizerH264::NextAggregatePacket(RtpPacketToSend* rtp_packet) {
   size_t index = kNalHeaderSize;
   bool is_last_fragment = packet->last_fragment;
   while (packet->aggregated) {
-    ArrayView<const uint8_t> fragment = packet->source_fragment;
+    std::span<const uint8_t> fragment = packet->source_fragment;
     RTC_CHECK_LE(index + kLengthFieldSize + fragment.size(), payload_capacity);
     // Add NAL unit length field.
     ByteWriter<uint16_t>::WriteBigEndian(&buffer[index], fragment.size());
@@ -290,7 +290,7 @@ void RtpPacketizerH264::NextFragmentPacket(RtpPacketToSend* rtp_packet) {
   fu_header |= (packet->last_fragment ? kH264EBit : 0);
   uint8_t type = packet->header & kH264TypeMask;
   fu_header |= type;
-  ArrayView<const uint8_t> fragment = packet->source_fragment;
+  std::span<const uint8_t> fragment = packet->source_fragment;
   uint8_t* buffer =
       rtp_packet->AllocatePayload(kFuAHeaderSize + fragment.size());
   buffer[0] = fu_indicator;
diff --git a/modules/rtp_rtcp/source/rtp_format_h264.h b/modules/rtp_rtcp/source/rtp_format_h264.h
index def175aaf9..0af5ba51cb 100644
--- a/modules/rtp_rtcp/source/rtp_format_h264.h
+++ b/modules/rtp_rtcp/source/rtp_format_h264.h
@@ -16,8 +16,8 @@
 
 #include <deque>
 #include <queue>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/rtp_rtcp/source/rtp_format.h"
 #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
 #include "modules/video_coding/codecs/h264/include/h264_globals.h"
@@ -38,7 +38,7 @@ class RtpPacketizerH264 : public RtpPacketizer {
  public:
   // Initialize with payload from encoder.
   // The payload_data must be exactly one encoded H264 frame.
-  RtpPacketizerH264(ArrayView<const uint8_t> payload,
+  RtpPacketizerH264(std::span<const uint8_t> payload,
                     PayloadSizeLimits limits,
                     H264PacketizationMode packetization_mode);
 
@@ -62,7 +62,7 @@ class RtpPacketizerH264 : public RtpPacketizer {
   // packet unit may represent a single NAL unit or a STAP-A packet, of which
   // there may be multiple in a single RTP packet (if so, aggregated = true).
   struct PacketUnit {
-    PacketUnit(ArrayView<const uint8_t> source_fragment,
+    PacketUnit(std::span<const uint8_t> source_fragment,
                bool first_fragment,
                bool last_fragment,
                bool aggregated,
@@ -73,7 +73,7 @@ class RtpPacketizerH264 : public RtpPacketizer {
           aggregated(aggregated),
           header(header) {}
 
-    ArrayView<const uint8_t> source_fragment;
+    std::span<const uint8_t> source_fragment;
     bool first_fragment;
     bool last_fragment;
     bool aggregated;
@@ -90,7 +90,7 @@ class RtpPacketizerH264 : public RtpPacketizer {
 
   const PayloadSizeLimits limits_;
   size_t num_packets_left_;
-  std::deque<ArrayView<const uint8_t>> input_fragments_;
+  std::deque<std::span<const uint8_t>> input_fragments_;
   std::queue<PacketUnit> packets_;
 };
 }  // namespace webrtc
diff --git a/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc b/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
index 3da925d32e..15e6817155 100644
--- a/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
@@ -14,10 +14,10 @@
 #include <cstdint>
 #include <cstring>
 #include <initializer_list>
+#include <span>
 #include <vector>
 
 #include "absl/algorithm/container.h"
-#include "api/array_view.h"
 #include "modules/rtp_rtcp/source/rtp_format.h"
 #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
 #include "modules/video_coding/codecs/h264/include/h264_globals.h"
@@ -58,7 +58,7 @@ constexpr size_t kFuAHeaderSize = 2;
 Buffer GenerateNalUnit(size_t size) {
   RTC_CHECK_GT(size, 0);
   Buffer buffer = Buffer::CreateWithCapacity(size);
-  buffer.AppendData(size, [](ArrayView<uint8_t> buffer_view) {
+  buffer.AppendData(size, [](std::span<uint8_t> buffer_view) {
     // Set some valid header.
     buffer_view[0] = kSlice;
     for (size_t i = 1; i < buffer_view.size(); ++i) {
@@ -78,7 +78,7 @@ Buffer CreateFrame(std::initializer_list<size_t> nalu_sizes) {
   size_t size =
       absl::c_accumulate(nalu_sizes, 0) + kStartCodeSize * nalu_sizes.size();
   Buffer frame = Buffer::CreateWithCapacity(size);
-  frame.AppendData(size, [&](ArrayView<uint8_t> frame_view) {
+  frame.AppendData(size, [&](std::span<uint8_t> frame_view) {
     size_t offset = 0;
     for (size_t nalu_size : nalu_sizes) {
       EXPECT_GE(nalu_size, 1u);
@@ -101,14 +101,14 @@ Buffer CreateFrame(std::initializer_list<size_t> nalu_sizes) {
 }
 
 // Create frame consisting of given nalus.
-Buffer CreateFrame(ArrayView<const Buffer> nalus) {
+Buffer CreateFrame(std::span<const Buffer> nalus) {
   static constexpr int kStartCodeSize = 3;
   int frame_size = 0;
   for (const Buffer& nalu : nalus) {
     frame_size += (kStartCodeSize + nalu.size());
   }
   Buffer frame = Buffer::CreateWithCapacity(frame_size);
-  frame.AppendData(frame_size, [&](ArrayView<uint8_t> frame_view) {
+  frame.AppendData(frame_size, [&](std::span<uint8_t> frame_view) {
     size_t offset = 0;
     for (const Buffer& nalu : nalus) {
       // Insert nalu start code
diff --git a/modules/rtp_rtcp/source/rtp_format_video_generic.cc b/modules/rtp_rtcp/source/rtp_format_video_generic.cc
index 5df035a8c2..c44d7ca636 100644
--- a/modules/rtp_rtcp/source/rtp_format_video_generic.cc
+++ b/modules/rtp_rtcp/source/rtp_format_video_generic.cc
@@ -12,9 +12,9 @@
 
 #include <cstdint>
 #include <cstring>
+#include <span>
 #include <variant>
 
-#include "api/array_view.h"
 #include "api/video/video_frame_type.h"
 #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
 #include "modules/rtp_rtcp/source/rtp_video_header.h"
@@ -29,7 +29,7 @@ constexpr size_t kExtendedHeaderLength = 2;
 }  // namespace
 
 RtpPacketizerGeneric::RtpPacketizerGeneric(
-    ArrayView<const uint8_t> payload,
+    std::span<const uint8_t> payload,
     PayloadSizeLimits limits,
     const RTPVideoHeader& rtp_video_header)
     : remaining_payload_(payload) {
@@ -40,7 +40,7 @@ RtpPacketizerGeneric::RtpPacketizerGeneric(
   current_packet_ = payload_sizes_.begin();
 }
 
-RtpPacketizerGeneric::RtpPacketizerGeneric(ArrayView<const uint8_t> payload,
+RtpPacketizerGeneric::RtpPacketizerGeneric(std::span<const uint8_t> payload,
                                            PayloadSizeLimits limits)
     : header_size_(0), remaining_payload_(payload) {
   payload_sizes_ = SplitAboutEqually(payload.size(), limits);
diff --git a/modules/rtp_rtcp/source/rtp_format_video_generic.h b/modules/rtp_rtcp/source/rtp_format_video_generic.h
index 3e0c1e3f05..cc9a6fb058 100644
--- a/modules/rtp_rtcp/source/rtp_format_video_generic.h
+++ b/modules/rtp_rtcp/source/rtp_format_video_generic.h
@@ -13,9 +13,9 @@
 #include <stdint.h>
 
 #include <cstddef>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/rtp_rtcp/source/rtp_format.h"
 
 namespace webrtc {
@@ -36,14 +36,14 @@ class RtpPacketizerGeneric : public RtpPacketizer {
   // Initialize with payload from encoder.
   // The payload_data must be exactly one encoded generic frame.
   // Packets returned by `NextPacket` will contain the generic payload header.
-  RtpPacketizerGeneric(ArrayView<const uint8_t> payload,
+  RtpPacketizerGeneric(std::span<const uint8_t> payload,
                        PayloadSizeLimits limits,
                        const RTPVideoHeader& rtp_video_header);
   // Initialize with payload from encoder.
   // The payload_data must be exactly one encoded generic frame.
   // Packets returned by `NextPacket` will contain raw payload without the
   // generic payload header.
-  RtpPacketizerGeneric(ArrayView<const uint8_t> payload,
+  RtpPacketizerGeneric(std::span<const uint8_t> payload,
                        PayloadSizeLimits limits);
 
   ~RtpPacketizerGeneric() override;
@@ -64,7 +64,7 @@ class RtpPacketizerGeneric : public RtpPacketizer {
 
   uint8_t header_[3];
   size_t header_size_;
-  ArrayView<const uint8_t> remaining_payload_;
+  std::span<const uint8_t> remaining_payload_;
   std::vector<int> payload_sizes_;
   std::vector<int>::const_iterator current_packet_;
 };
diff --git a/modules/rtp_rtcp/source/rtp_format_video_generic_unittest.cc b/modules/rtp_rtcp/source/rtp_format_video_generic_unittest.cc
index e927038f04..2a200dd434 100644
--- a/modules/rtp_rtcp/source/rtp_format_video_generic_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_format_video_generic_unittest.cc
@@ -12,9 +12,9 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/video/video_frame_type.h"
 #include "modules/rtp_rtcp/source/rtp_format.h"
 #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
@@ -82,7 +82,7 @@ TEST(RtpPacketizerVideoGeneric, WritesExtendedHeaderWhenPictureIdIsSet) {
   RtpPacketToSend packet(nullptr);
   ASSERT_TRUE(packetizer.NextPacket(&packet));
 
-  ArrayView<const uint8_t> payload = packet.payload();
+  std::span<const uint8_t> payload = packet.payload();
   EXPECT_EQ(payload.size(), 3 + kPayloadSize);
   EXPECT_TRUE(payload[0] & 0x04);  // Extended header bit is set.
   // Frame id is 37.
@@ -136,7 +136,7 @@ TEST(RtpPacketizerVideoGeneric, FrameIdOver15bitsWrapsAround) {
   RtpPacketToSend packet(nullptr);
   ASSERT_TRUE(packetizer.NextPacket(&packet));
 
-  ArrayView<const uint8_t> payload = packet.payload();
+  std::span<const uint8_t> payload = packet.payload();
   EXPECT_TRUE(payload[0] & 0x04);  // Extended header bit is set.
   // Frame id is 0x137.
   EXPECT_EQ(0x01u, payload[1]);
@@ -152,7 +152,7 @@ TEST(RtpPacketizerVideoGeneric, NoFrameIdDoesNotWriteExtendedHeader) {
   RtpPacketToSend packet(nullptr);
   ASSERT_TRUE(packetizer.NextPacket(&packet));
 
-  ArrayView<const uint8_t> payload = packet.payload();
+  std::span<const uint8_t> payload = packet.payload();
   EXPECT_FALSE(payload[0] & 0x04);
 }
 
@@ -164,7 +164,7 @@ TEST(RtpPacketizerVideoGeneric, DoesNotWriteHeaderForRawPayload) {
   RtpPacketToSend packet(nullptr);
   ASSERT_TRUE(packetizer.NextPacket(&packet));
 
-  ArrayView<const uint8_t> payload = packet.payload();
+  std::span<const uint8_t> payload = packet.payload();
   EXPECT_THAT(payload, ElementsAreArray(kPayload));
 }
 
diff --git a/modules/rtp_rtcp/source/rtp_format_vp8.cc b/modules/rtp_rtcp/source/rtp_format_vp8.cc
index 18bb4eb844..7727438b0d 100644
--- a/modules/rtp_rtcp/source/rtp_format_vp8.cc
+++ b/modules/rtp_rtcp/source/rtp_format_vp8.cc
@@ -12,9 +12,9 @@
 
 #include <cstdint>
 #include <cstring>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
 #include "modules/video_coding/codecs/interface/common_constants.h"
 #include "modules/video_coding/codecs/vp8/include/vp8_globals.h"
@@ -57,7 +57,7 @@ bool ValidateHeader(const RTPVideoHeaderVP8& hdr_info) {
 
 }  // namespace
 
-RtpPacketizerVp8::RtpPacketizerVp8(ArrayView<const uint8_t> payload,
+RtpPacketizerVp8::RtpPacketizerVp8(std::span<const uint8_t> payload,
                                    PayloadSizeLimits limits,
                                    const RTPVideoHeaderVP8& hdr_info)
     : hdr_(BuildHeader(hdr_info)), remaining_payload_(payload) {
diff --git a/modules/rtp_rtcp/source/rtp_format_vp8.h b/modules/rtp_rtcp/source/rtp_format_vp8.h
index 36f4b8b3b3..034ad449a5 100644
--- a/modules/rtp_rtcp/source/rtp_format_vp8.h
+++ b/modules/rtp_rtcp/source/rtp_format_vp8.h
@@ -28,10 +28,10 @@
 #include <stddef.h>
 
 #include <cstdint>
+#include <span>
 #include <vector>
 
 #include "absl/container/inlined_vector.h"
-#include "api/array_view.h"
 #include "modules/rtp_rtcp/source/rtp_format.h"
 #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
 #include "modules/video_coding/codecs/vp8/include/vp8_globals.h"
@@ -43,7 +43,7 @@ class RtpPacketizerVp8 : public RtpPacketizer {
  public:
   // Initialize with payload from encoder.
   // The payload_data must be exactly one encoded VP8 frame.
-  RtpPacketizerVp8(ArrayView<const uint8_t> payload,
+  RtpPacketizerVp8(std::span<const uint8_t> payload,
                    PayloadSizeLimits limits,
                    const RTPVideoHeaderVP8& hdr_info);
 
@@ -65,7 +65,7 @@ class RtpPacketizerVp8 : public RtpPacketizer {
   static RawHeader BuildHeader(const RTPVideoHeaderVP8& header);
 
   RawHeader hdr_;
-  ArrayView<const uint8_t> remaining_payload_;
+  std::span<const uint8_t> remaining_payload_;
   std::vector<int> payload_sizes_;
   std::vector<int>::const_iterator current_packet_;
 };
diff --git a/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.cc b/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.cc
index 30cc3b7c7a..f55bbfb2dd 100644
--- a/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.cc
+++ b/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.cc
@@ -13,8 +13,8 @@
 #include <cstddef>
 #include <cstdint>
 #include <iterator>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/rtp_rtcp/source/rtp_format_vp8.h"
 #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
 #include "modules/video_coding/codecs/interface/common_constants.h"
@@ -57,7 +57,7 @@ int Bit(uint8_t byte, int position) {
 RtpFormatVp8TestHelper::RtpFormatVp8TestHelper(const RTPVideoHeaderVP8* hdr,
                                                size_t payload_len)
     : hdr_info_(hdr), payload_(Buffer::CreateWithCapacity(payload_len)) {
-  payload_.AppendData(payload_len, [](ArrayView<uint8_t> payload_view) {
+  payload_.AppendData(payload_len, [](std::span<uint8_t> payload_view) {
     for (size_t i = 0; i < payload_view.size(); ++i) {
       payload_view[i] = i;
     }
@@ -69,7 +69,7 @@ RtpFormatVp8TestHelper::~RtpFormatVp8TestHelper() = default;
 
 void RtpFormatVp8TestHelper::GetAllPacketsAndCheck(
     RtpPacketizerVp8* packetizer,
-    ArrayView<const size_t> expected_sizes) {
+    std::span<const size_t> expected_sizes) {
   EXPECT_EQ(packetizer->NumPackets(), expected_sizes.size());
   Buffer::const_iterator data_it = payload_.begin();
   RtpPacketToSend packet(kNoExtensions);
@@ -91,7 +91,7 @@ void RtpFormatVp8TestHelper::GetAllPacketsAndCheck(
   EXPECT_EQ(payload_.end(), data_it);
 }
 
-int RtpFormatVp8TestHelper::CheckHeader(ArrayView<const uint8_t> buffer,
+int RtpFormatVp8TestHelper::CheckHeader(std::span<const uint8_t> buffer,
                                         bool first) {
   int x_bit = Bit(buffer[0], 7);
   EXPECT_EQ(Bit(buffer[0], 6), 0);  // Reserved.
@@ -118,7 +118,7 @@ int RtpFormatVp8TestHelper::CheckHeader(ArrayView<const uint8_t> buffer,
 
 // Verify that the I bit and the PictureID field are both set in accordance
 // with the information in hdr_info_->pictureId.
-void RtpFormatVp8TestHelper::CheckPictureID(ArrayView<const uint8_t> buffer,
+void RtpFormatVp8TestHelper::CheckPictureID(std::span<const uint8_t> buffer,
                                             int* offset) {
   int i_bit = Bit(buffer[1], 7);
   if (hdr_info_->pictureId != kNoPictureId) {
@@ -135,7 +135,7 @@ void RtpFormatVp8TestHelper::CheckPictureID(ArrayView<const uint8_t> buffer,
 
 // Verify that the L bit and the TL0PICIDX field are both set in accordance
 // with the information in hdr_info_->tl0PicIdx.
-void RtpFormatVp8TestHelper::CheckTl0PicIdx(ArrayView<const uint8_t> buffer,
+void RtpFormatVp8TestHelper::CheckTl0PicIdx(std::span<const uint8_t> buffer,
                                             int* offset) {
   int l_bit = Bit(buffer[1], 6);
   if (hdr_info_->tl0PicIdx != kNoTl0PicIdx) {
@@ -150,7 +150,7 @@ void RtpFormatVp8TestHelper::CheckTl0PicIdx(ArrayView<const uint8_t> buffer,
 // Verify that the T bit and the TL0PICIDX field, and the K bit and KEYIDX
 // field are all set in accordance with the information in
 // hdr_info_->temporalIdx and hdr_info_->keyIdx, respectively.
-void RtpFormatVp8TestHelper::CheckTIDAndKeyIdx(ArrayView<const uint8_t> buffer,
+void RtpFormatVp8TestHelper::CheckTIDAndKeyIdx(std::span<const uint8_t> buffer,
                                                int* offset) {
   int t_bit = Bit(buffer[1], 5);
   int k_bit = Bit(buffer[1], 4);
diff --git a/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h b/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h
index a364df0294..3e744d9b20 100644
--- a/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h
+++ b/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h
@@ -19,8 +19,8 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/rtp_rtcp/source/rtp_format_vp8.h"
 #include "modules/video_coding/codecs/vp8/include/vp8_globals.h"
 #include "rtc_base/buffer.h"
@@ -36,17 +36,17 @@ class RtpFormatVp8TestHelper {
   RtpFormatVp8TestHelper& operator=(const RtpFormatVp8TestHelper&) = delete;
 
   void GetAllPacketsAndCheck(RtpPacketizerVp8* packetizer,
-                             ArrayView<const size_t> expected_sizes);
+                             std::span<const size_t> expected_sizes);
 
-  ArrayView<const uint8_t> payload() const { return payload_; }
+  std::span<const uint8_t> payload() const { return payload_; }
   size_t payload_size() const { return payload_.size(); }
 
  private:
   // Returns header size, i.e. payload offset.
-  int CheckHeader(ArrayView<const uint8_t> rtp_payload, bool first);
-  void CheckPictureID(ArrayView<const uint8_t> rtp_payload, int* offset);
-  void CheckTl0PicIdx(ArrayView<const uint8_t> rtp_payload, int* offset);
-  void CheckTIDAndKeyIdx(ArrayView<const uint8_t> rtp_payload, int* offset);
+  int CheckHeader(std::span<const uint8_t> rtp_payload, bool first);
+  void CheckPictureID(std::span<const uint8_t> rtp_payload, int* offset);
+  void CheckTl0PicIdx(std::span<const uint8_t> rtp_payload, int* offset);
+  void CheckTIDAndKeyIdx(std::span<const uint8_t> rtp_payload, int* offset);
   void CheckPayload(const uint8_t* data_ptr);
 
   const RTPVideoHeaderVP8* const hdr_info_;
diff --git a/modules/rtp_rtcp/source/rtp_format_vp9.cc b/modules/rtp_rtcp/source/rtp_format_vp9.cc
index a5523c6210..77935f9f05 100644
--- a/modules/rtp_rtcp/source/rtp_format_vp9.cc
+++ b/modules/rtp_rtcp/source/rtp_format_vp9.cc
@@ -12,8 +12,8 @@
 
 #include <cstdint>
 #include <cstring>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
 #include "modules/video_coding/codecs/interface/common_constants.h"
 #include "modules/video_coding/codecs/vp9/include/vp9_globals.h"
@@ -307,7 +307,7 @@ RTPVideoHeaderVP9 RemoveInactiveSpatialLayers(
 }
 }  // namespace
 
-RtpPacketizerVp9::RtpPacketizerVp9(ArrayView<const uint8_t> payload,
+RtpPacketizerVp9::RtpPacketizerVp9(std::span<const uint8_t> payload,
                                    PayloadSizeLimits limits,
                                    const RTPVideoHeaderVP9& hdr)
     : hdr_(RemoveInactiveSpatialLayers(hdr)),
@@ -349,7 +349,7 @@ bool RtpPacketizerVp9::NextPacket(RtpPacketToSend* packet) {
   uint8_t* buffer = packet->AllocatePayload(header_size + packet_payload_len);
   RTC_CHECK(buffer);
 
-  if (!WriteHeader(layer_begin, layer_end, MakeArrayView(buffer, header_size)))
+  if (!WriteHeader(layer_begin, layer_end, std::span(buffer, header_size)))
     return false;
 
   memcpy(buffer + header_size, remaining_payload_.data(), packet_payload_len);
@@ -401,7 +401,7 @@ bool RtpPacketizerVp9::NextPacket(RtpPacketToSend* packet) {
 //      +-+-+-+-+-+-+-+-+
 bool RtpPacketizerVp9::WriteHeader(bool layer_begin,
                                    bool layer_end,
-                                   ArrayView<uint8_t> buffer) const {
+                                   std::span<uint8_t> buffer) const {
   // Required payload descriptor byte.
   bool i_bit = PictureIdPresent(hdr_);
   bool p_bit = hdr_.inter_pic_predicted;
diff --git a/modules/rtp_rtcp/source/rtp_format_vp9.h b/modules/rtp_rtcp/source/rtp_format_vp9.h
index 80c43f35b0..59af6383c4 100644
--- a/modules/rtp_rtcp/source/rtp_format_vp9.h
+++ b/modules/rtp_rtcp/source/rtp_format_vp9.h
@@ -24,9 +24,9 @@
 #include <stddef.h>
 #include <stdint.h>
 
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/rtp_rtcp/source/rtp_format.h"
 #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
 #include "modules/video_coding/codecs/vp9/include/vp9_globals.h"
@@ -36,7 +36,7 @@ namespace webrtc {
 class RtpPacketizerVp9 : public RtpPacketizer {
  public:
   // The `payload` must be one encoded VP9 layer frame.
-  RtpPacketizerVp9(ArrayView<const uint8_t> payload,
+  RtpPacketizerVp9(std::span<const uint8_t> payload,
                    PayloadSizeLimits limits,
                    const RTPVideoHeaderVP9& hdr);
 
@@ -58,12 +58,12 @@ class RtpPacketizerVp9 : public RtpPacketizer {
   // the layer frame. Returns false on failure.
   bool WriteHeader(bool layer_begin,
                    bool layer_end,
-                   ArrayView<uint8_t> rtp_payload) const;
+                   std::span<uint8_t> rtp_payload) const;
 
   const RTPVideoHeaderVP9 hdr_;
   const int header_size_;
   const int first_packet_extra_header_size_;
-  ArrayView<const uint8_t> remaining_payload_;
+  std::span<const uint8_t> remaining_payload_;
   std::vector<int> payload_sizes_;
   std::vector<int>::const_iterator current_packet_;
 };
diff --git a/modules/rtp_rtcp/source/rtp_format_vp9_unittest.cc b/modules/rtp_rtcp/source/rtp_format_vp9_unittest.cc
index 58789da14f..9c59ef788f 100644
--- a/modules/rtp_rtcp/source/rtp_format_vp9_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_format_vp9_unittest.cc
@@ -13,9 +13,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/video/video_codec_type.h"
 #include "modules/rtp_rtcp/source/rtp_format.h"
 #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
@@ -78,7 +78,7 @@ void ParseAndCheckPacket(const uint8_t* packet,
                          size_t expected_length) {
   RTPVideoHeader video_header;
   EXPECT_EQ(VideoRtpDepacketizerVp9::ParseRtpPayload(
-                MakeArrayView(packet, expected_length), &video_header),
+                std::span(packet, expected_length), &video_header),
             expected_hdr_length);
   EXPECT_EQ(kVideoCodecVP9, video_header.codec);
   auto& vp9_header =
@@ -154,8 +154,8 @@ class RtpPacketizerVp9Test : public ::testing::Test {
     EXPECT_EQ(last, payload_pos_ == payload_.size());
   }
 
-  void CreateParseAndCheckPackets(ArrayView<const size_t> expected_hdr_sizes,
-                                  ArrayView<const size_t> expected_sizes) {
+  void CreateParseAndCheckPackets(std::span<const size_t> expected_hdr_sizes,
+                                  std::span<const size_t> expected_sizes) {
     ASSERT_EQ(expected_hdr_sizes.size(), expected_sizes.size());
     ASSERT_TRUE(packetizer_ != nullptr);
     EXPECT_EQ(expected_sizes.size(), num_packets_);
diff --git a/modules/rtp_rtcp/source/rtp_generic_frame_descriptor.cc b/modules/rtp_rtcp/source/rtp_generic_frame_descriptor.cc
index 9bb88987b7..44124f3b46 100644
--- a/modules/rtp_rtcp/source/rtp_generic_frame_descriptor.cc
+++ b/modules/rtp_rtcp/source/rtp_generic_frame_descriptor.cc
@@ -11,8 +11,8 @@
 #include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h"
 
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "rtc_base/checks.h"
 
 namespace webrtc {
@@ -75,10 +75,10 @@ void RtpGenericFrameDescriptor::SetFrameId(uint16_t frame_id) {
   frame_id_ = frame_id;
 }
 
-ArrayView<const uint16_t> RtpGenericFrameDescriptor::FrameDependenciesDiffs()
+std::span<const uint16_t> RtpGenericFrameDescriptor::FrameDependenciesDiffs()
     const {
   RTC_DCHECK(FirstPacketInSubFrame());
-  return MakeArrayView(frame_deps_id_diffs_, num_frame_deps_);
+  return std::span(frame_deps_id_diffs_, num_frame_deps_);
 }
 
 bool RtpGenericFrameDescriptor::AddFrameDependencyDiff(uint16_t fdiff) {
diff --git a/modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h b/modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h
index 7ecdb2a69e..4075bfef0e 100644
--- a/modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h
+++ b/modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h
@@ -13,8 +13,7 @@
 #include <stddef.h>
 #include <stdint.h>
 
-
-#include "api/array_view.h"
+#include <span>
 
 namespace webrtc {
 
@@ -54,7 +53,7 @@ class RtpGenericFrameDescriptor {
   uint16_t FrameId() const;
   void SetFrameId(uint16_t frame_id);
 
-  ArrayView<const uint16_t> FrameDependenciesDiffs() const;
+  std::span<const uint16_t> FrameDependenciesDiffs() const;
   void ClearFrameDependencies() { num_frame_deps_ = 0; }
   // Returns false on failure, i.e. number of dependencies is too large.
   bool AddFrameDependencyDiff(uint16_t fdiff);
diff --git a/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.cc b/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.cc
index e8ca3dd3b6..511b93e591 100644
--- a/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.cc
+++ b/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.cc
@@ -12,8 +12,8 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h"
 #include "rtc_base/checks.h"
 
@@ -63,7 +63,7 @@ constexpr uint8_t kFlageXtendedOffset = 0x02;
 //      +-+-+-+-+-+-+-+-+
 
 bool RtpGenericFrameDescriptorExtension00::Parse(
-    ArrayView<const uint8_t> data,
+    std::span<const uint8_t> data,
     RtpGenericFrameDescriptor* descriptor) {
   if (data.empty()) {
     return false;
@@ -131,7 +131,7 @@ size_t RtpGenericFrameDescriptorExtension00::ValueSize(
 }
 
 bool RtpGenericFrameDescriptorExtension00::Write(
-    ArrayView<uint8_t> data,
+    std::span<uint8_t> data,
     const RtpGenericFrameDescriptor& descriptor) {
   RTC_CHECK_EQ(data.size(), ValueSize(descriptor));
   uint8_t base_header =
@@ -152,7 +152,7 @@ bool RtpGenericFrameDescriptorExtension00::Write(
   uint16_t frame_id = descriptor.FrameId();
   data[2] = frame_id & 0xff;
   data[3] = frame_id >> 8;
-  ArrayView<const uint16_t> fdiffs = descriptor.FrameDependenciesDiffs();
+  std::span<const uint16_t> fdiffs = descriptor.FrameDependenciesDiffs();
   size_t offset = 4;
   if (descriptor.FirstPacketInSubFrame() && fdiffs.empty() &&
       descriptor.Width() > 0 && descriptor.Height() > 0) {
diff --git a/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h b/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h
index 8c6555564c..c213f52310 100644
--- a/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h
+++ b/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h
@@ -13,8 +13,9 @@
 #include <stddef.h>
 #include <stdint.h>
 
+#include <span>
+
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtp_parameters.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h"
@@ -33,10 +34,10 @@ class RtpGenericFrameDescriptorExtension00 {
   }
   static constexpr int kMaxSizeBytes = 16;
 
-  static bool Parse(ArrayView<const uint8_t> data,
+  static bool Parse(std::span<const uint8_t> data,
                     RtpGenericFrameDescriptor* descriptor);
   static size_t ValueSize(const RtpGenericFrameDescriptor& descriptor);
-  static bool Write(ArrayView<uint8_t> data,
+  static bool Write(std::span<uint8_t> data,
                     const RtpGenericFrameDescriptor& descriptor);
 };
 
diff --git a/modules/rtp_rtcp/source/rtp_header_extension_map.cc b/modules/rtp_rtcp/source/rtp_header_extension_map.cc
index f170ab9a7a..1a468a405a 100644
--- a/modules/rtp_rtcp/source/rtp_header_extension_map.cc
+++ b/modules/rtp_rtcp/source/rtp_header_extension_map.cc
@@ -11,9 +11,10 @@
 #include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
 
 #include <cstdint>
+#include <iterator>
+#include <span>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtp_parameters.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/corruption_detection_extension.h"
@@ -78,13 +79,13 @@ RtpHeaderExtensionMap::RtpHeaderExtensionMap(bool extmap_allow_mixed)
 }
 
 RtpHeaderExtensionMap::RtpHeaderExtensionMap(
-    ArrayView<const RtpExtension> extensions)
+    std::span<const RtpExtension> extensions)
     : RtpHeaderExtensionMap(false) {
   for (const RtpExtension& extension : extensions)
     RegisterByUri(extension.id, extension.uri);
 }
 
-void RtpHeaderExtensionMap::Reset(ArrayView<const RtpExtension> extensions) {
+void RtpHeaderExtensionMap::Reset(std::span<const RtpExtension> extensions) {
   for (auto& id : ids_)
     id = kInvalidId;
   for (const RtpExtension& extension : extensions)
diff --git a/modules/rtp_rtcp/source/rtp_header_extension_size.cc b/modules/rtp_rtcp/source/rtp_header_extension_size.cc
index b9016de16e..dd7ff9be2a 100644
--- a/modules/rtp_rtcp/source/rtp_header_extension_size.cc
+++ b/modules/rtp_rtcp/source/rtp_header_extension_size.cc
@@ -10,13 +10,14 @@
 
 #include "modules/rtp_rtcp/source/rtp_header_extension_size.h"
 
-#include "api/array_view.h"
+#include <span>
+
 #include "api/rtp_parameters.h"
 #include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
 
 namespace webrtc {
 
-int RtpHeaderExtensionSize(ArrayView<const RtpExtensionSize> extensions,
+int RtpHeaderExtensionSize(std::span<const RtpExtensionSize> extensions,
                            const RtpHeaderExtensionMap& registered_extensions) {
   // RFC3550 Section 5.3.1
   static constexpr int kExtensionBlockHeaderSize = 4;
diff --git a/modules/rtp_rtcp/source/rtp_header_extension_size.h b/modules/rtp_rtcp/source/rtp_header_extension_size.h
index b7fd960da1..48a180812e 100644
--- a/modules/rtp_rtcp/source/rtp_header_extension_size.h
+++ b/modules/rtp_rtcp/source/rtp_header_extension_size.h
@@ -10,7 +10,8 @@
 #ifndef MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSION_SIZE_H_
 #define MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSION_SIZE_H_
 
-#include "api/array_view.h"
+#include <span>
+
 #include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
@@ -24,7 +25,7 @@ struct RtpExtensionSize {
 // Calculates rtp header extension size in bytes assuming packet contain
 // all `extensions` with provided `value_size`.
 // Counts only extensions present among `registered_extensions`.
-int RtpHeaderExtensionSize(ArrayView<const RtpExtensionSize> extensions,
+int RtpHeaderExtensionSize(std::span<const RtpExtensionSize> extensions,
                            const RtpHeaderExtensionMap& registered_extensions);
 
 }  // namespace webrtc
diff --git a/modules/rtp_rtcp/source/rtp_header_extensions.cc b/modules/rtp_rtcp/source/rtp_header_extensions.cc
index 3db85794bc..15884c814a 100644
--- a/modules/rtp_rtcp/source/rtp_header_extensions.cc
+++ b/modules/rtp_rtcp/source/rtp_header_extensions.cc
@@ -18,11 +18,11 @@
 #include <cstring>
 #include <limits>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtp_headers.h"
 #include "api/units/time_delta.h"
 #include "api/video/color_space.h"
@@ -50,7 +50,7 @@ namespace webrtc {
 //   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 //   |  ID   | len=2 |              absolute send time               |
 //   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-bool AbsoluteSendTime::Parse(ArrayView<const uint8_t> data,
+bool AbsoluteSendTime::Parse(std::span<const uint8_t> data,
                              uint32_t* time_24bits) {
   if (data.size() != 3)
     return false;
@@ -58,7 +58,7 @@ bool AbsoluteSendTime::Parse(ArrayView<const uint8_t> data,
   return true;
 }
 
-bool AbsoluteSendTime::Write(ArrayView<uint8_t> data, uint32_t time_24bits) {
+bool AbsoluteSendTime::Write(std::span<uint8_t> data, uint32_t time_24bits) {
   RTC_DCHECK_EQ(data.size(), 3);
   RTC_DCHECK_LE(time_24bits, 0x00FFFFFF);
   ByteWriter<uint32_t, 3>::WriteBigEndian(data.data(), time_24bits);
@@ -100,7 +100,7 @@ bool AbsoluteSendTime::Write(ArrayView<uint8_t> data, uint32_t time_24bits) {
 //   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 //   |  ... (56-63)  |
 //   +-+-+-+-+-+-+-+-+
-bool AbsoluteCaptureTimeExtension::Parse(ArrayView<const uint8_t> data,
+bool AbsoluteCaptureTimeExtension::Parse(std::span<const uint8_t> data,
                                          AbsoluteCaptureTime* extension) {
   if (data.size() != kValueSizeBytes &&
       data.size() != kValueSizeBytesWithoutEstimatedCaptureClockOffset) {
@@ -127,7 +127,7 @@ size_t AbsoluteCaptureTimeExtension::ValueSize(
   }
 }
 
-bool AbsoluteCaptureTimeExtension::Write(ArrayView<uint8_t> data,
+bool AbsoluteCaptureTimeExtension::Write(std::span<uint8_t> data,
                                          const AbsoluteCaptureTime& extension) {
   RTC_DCHECK_EQ(data.size(), ValueSize(extension));
 
@@ -161,7 +161,7 @@ bool AbsoluteCaptureTimeExtension::Write(ArrayView<uint8_t> data,
 // |      ID       |     len=1     |V|    level    |
 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 // Sample Audio Level Encoding Using the Two-Byte Header Format
-bool AudioLevelExtension::Parse(ArrayView<const uint8_t> data,
+bool AudioLevelExtension::Parse(std::span<const uint8_t> data,
                                 AudioLevel* extension) {
   // One-byte and two-byte format share the same data definition.
   if (data.size() != 1)
@@ -172,7 +172,7 @@ bool AudioLevelExtension::Parse(ArrayView<const uint8_t> data,
   return true;
 }
 
-bool AudioLevelExtension::Write(ArrayView<uint8_t> data,
+bool AudioLevelExtension::Write(std::span<uint8_t> data,
                                 const AudioLevel& extension) {
   // One-byte and two-byte format share the same data definition.
   RTC_DCHECK_EQ(data.size(), 1);
@@ -203,7 +203,7 @@ bool AudioLevelExtension::Write(ArrayView<uint8_t> data,
 // |0|   level 3   |    0 (pad)    |               ...             |
 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 // Sample Audio Level Encoding Using the Two-Byte Header Format
-bool CsrcAudioLevel::Parse(ArrayView<const uint8_t> data,
+bool CsrcAudioLevel::Parse(std::span<const uint8_t> data,
                            std::vector<uint8_t>* csrc_audio_levels) {
   if (data.size() > kRtpCsrcSize) {
     return false;
@@ -215,12 +215,12 @@ bool CsrcAudioLevel::Parse(ArrayView<const uint8_t> data,
   return true;
 }
 
-size_t CsrcAudioLevel::ValueSize(ArrayView<const uint8_t> csrc_audio_levels) {
+size_t CsrcAudioLevel::ValueSize(std::span<const uint8_t> csrc_audio_levels) {
   return csrc_audio_levels.size();
 }
 
-bool CsrcAudioLevel::Write(ArrayView<uint8_t> data,
-                           ArrayView<const uint8_t> csrc_audio_levels) {
+bool CsrcAudioLevel::Write(std::span<uint8_t> data,
+                           std::span<const uint8_t> csrc_audio_levels) {
   RTC_CHECK_LE(csrc_audio_levels.size(), kRtpCsrcSize);
   if (csrc_audio_levels.size() != data.size()) {
     return false;
@@ -247,7 +247,7 @@ bool CsrcAudioLevel::Write(ArrayView<uint8_t> data,
 //   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 //   |  ID   | len=2 |              transmission offset              |
 //   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-bool TransmissionOffset::Parse(ArrayView<const uint8_t> data,
+bool TransmissionOffset::Parse(std::span<const uint8_t> data,
                                int32_t* rtp_time) {
   if (data.size() != 3)
     return false;
@@ -255,7 +255,7 @@ bool TransmissionOffset::Parse(ArrayView<const uint8_t> data,
   return true;
 }
 
-bool TransmissionOffset::Write(ArrayView<uint8_t> data, int32_t rtp_time) {
+bool TransmissionOffset::Write(std::span<uint8_t> data, int32_t rtp_time) {
   RTC_DCHECK_EQ(data.size(), 3);
   RTC_DCHECK_LE(rtp_time, 0x00ffffff);
   ByteWriter<int32_t, 3>::WriteBigEndian(data.data(), rtp_time);
@@ -269,7 +269,7 @@ bool TransmissionOffset::Write(ArrayView<uint8_t> data, int32_t rtp_time) {
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 //  |  ID   | L=1   |transport-wide sequence number |
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-bool TransportSequenceNumber::Parse(ArrayView<const uint8_t> data,
+bool TransportSequenceNumber::Parse(std::span<const uint8_t> data,
                                     uint16_t* transport_sequence_number) {
   if (data.size() != kValueSizeBytes)
     return false;
@@ -277,7 +277,7 @@ bool TransportSequenceNumber::Parse(ArrayView<const uint8_t> data,
   return true;
 }
 
-bool TransportSequenceNumber::Write(ArrayView<uint8_t> data,
+bool TransportSequenceNumber::Write(std::span<uint8_t> data,
                                     uint16_t transport_sequence_number) {
   RTC_DCHECK_EQ(data.size(), ValueSize(transport_sequence_number));
   ByteWriter<uint16_t>::WriteBigEndian(data.data(), transport_sequence_number);
@@ -302,7 +302,7 @@ bool TransportSequenceNumber::Write(ArrayView<uint8_t> data,
 // cover including the current packet. If `seq_count` is zero no feedback is
 // requested.
 bool TransportSequenceNumberV2::Parse(
-    ArrayView<const uint8_t> data,
+    std::span<const uint8_t> data,
     uint16_t* transport_sequence_number,
     std::optional<FeedbackRequest>* feedback_request) {
   if (data.size() != kValueSizeBytes &&
@@ -329,7 +329,7 @@ bool TransportSequenceNumberV2::Parse(
 }
 
 bool TransportSequenceNumberV2::Write(
-    ArrayView<uint8_t> data,
+    std::span<uint8_t> data,
     uint16_t transport_sequence_number,
     const std::optional<FeedbackRequest>& feedback_request) {
   RTC_DCHECK_EQ(data.size(),
@@ -359,7 +359,7 @@ bool TransportSequenceNumberV2::Write(
 //   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 //   |  ID   | len=0 |0 0 0 0 C F R R|
 //   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-bool VideoOrientation::Parse(ArrayView<const uint8_t> data,
+bool VideoOrientation::Parse(std::span<const uint8_t> data,
                              VideoRotation* rotation) {
   if (data.size() != 1)
     return false;
@@ -367,20 +367,20 @@ bool VideoOrientation::Parse(ArrayView<const uint8_t> data,
   return true;
 }
 
-bool VideoOrientation::Write(ArrayView<uint8_t> data, VideoRotation rotation) {
+bool VideoOrientation::Write(std::span<uint8_t> data, VideoRotation rotation) {
   RTC_DCHECK_EQ(data.size(), 1);
   data[0] = ConvertVideoRotationToCVOByte(rotation);
   return true;
 }
 
-bool VideoOrientation::Parse(ArrayView<const uint8_t> data, uint8_t* value) {
+bool VideoOrientation::Parse(std::span<const uint8_t> data, uint8_t* value) {
   if (data.size() != 1)
     return false;
   *value = data[0];
   return true;
 }
 
-bool VideoOrientation::Write(ArrayView<uint8_t> data, uint8_t value) {
+bool VideoOrientation::Write(std::span<uint8_t> data, uint8_t value) {
   RTC_DCHECK_EQ(data.size(), 1);
   data[0] = value;
   return true;
@@ -391,7 +391,7 @@ bool VideoOrientation::Write(ArrayView<uint8_t> data, uint8_t value) {
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 //  |  ID   | len=2 |   MIN delay           |   MAX delay           |
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-bool PlayoutDelayLimits::Parse(ArrayView<const uint8_t> data,
+bool PlayoutDelayLimits::Parse(std::span<const uint8_t> data,
                                VideoPlayoutDelay* playout_delay) {
   RTC_DCHECK(playout_delay);
   if (data.size() != 3)
@@ -402,7 +402,7 @@ bool PlayoutDelayLimits::Parse(ArrayView<const uint8_t> data,
   return playout_delay->Set(min_raw * kGranularity, max_raw * kGranularity);
 }
 
-bool PlayoutDelayLimits::Write(ArrayView<uint8_t> data,
+bool PlayoutDelayLimits::Write(std::span<uint8_t> data,
                                const VideoPlayoutDelay& playout_delay) {
   RTC_DCHECK_EQ(data.size(), 3);
 
@@ -431,7 +431,7 @@ bool PlayoutDelayLimits::Write(ArrayView<uint8_t> data,
 //   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 //   |  ID   | len=0 | Content type  |
 //   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-bool VideoContentTypeExtension::Parse(ArrayView<const uint8_t> data,
+bool VideoContentTypeExtension::Parse(std::span<const uint8_t> data,
                                       VideoContentType* content_type) {
   if (data.size() == 1 &&
       videocontenttypehelpers::IsValidContentType(data[0])) {
@@ -445,7 +445,7 @@ bool VideoContentTypeExtension::Parse(ArrayView<const uint8_t> data,
   return false;
 }
 
-bool VideoContentTypeExtension::Write(ArrayView<uint8_t> data,
+bool VideoContentTypeExtension::Write(std::span<uint8_t> data,
                                       VideoContentType content_type) {
   RTC_DCHECK_EQ(data.size(), 1);
   data[0] = static_cast<uint8_t>(content_type);
@@ -473,7 +473,7 @@ bool VideoContentTypeExtension::Write(ArrayView<uint8_t> data,
 //   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 //   |  network2 timestamp ms delta  |
 //   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-bool VideoTimingExtension::Parse(ArrayView<const uint8_t> data,
+bool VideoTimingExtension::Parse(std::span<const uint8_t> data,
                                  VideoSendTiming* timing) {
   RTC_DCHECK(timing);
   // TODO(sprang): Deprecate support for old wire format.
@@ -505,7 +505,7 @@ bool VideoTimingExtension::Parse(ArrayView<const uint8_t> data,
   return true;
 }
 
-bool VideoTimingExtension::Write(ArrayView<uint8_t> data,
+bool VideoTimingExtension::Write(std::span<uint8_t> data,
                                  const VideoSendTiming& timing) {
   RTC_DCHECK_EQ(data.size(), 1 + 2 * 6);
   ByteWriter<uint8_t>::WriteBigEndian(data.data() + kFlagsOffset, timing.flags);
@@ -527,7 +527,7 @@ bool VideoTimingExtension::Write(ArrayView<uint8_t> data,
   return true;
 }
 
-bool VideoTimingExtension::Write(ArrayView<uint8_t> data,
+bool VideoTimingExtension::Write(std::span<uint8_t> data,
                                  uint16_t time_delta_ms,
                                  uint8_t offset) {
   RTC_DCHECK_GE(data.size(), offset + 2);
@@ -571,7 +571,7 @@ bool VideoTimingExtension::Write(ArrayView<uint8_t> data,
 //   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 //   |range+chr.sit. |
 //   +-+-+-+-+-+-+-+-+
-bool ColorSpaceExtension::Parse(ArrayView<const uint8_t> data,
+bool ColorSpaceExtension::Parse(std::span<const uint8_t> data,
                                 ColorSpace* color_space) {
   RTC_DCHECK(color_space);
   if (data.size() != kValueSizeBytes &&
@@ -611,7 +611,7 @@ bool ColorSpaceExtension::Parse(ArrayView<const uint8_t> data,
   return true;
 }
 
-bool ColorSpaceExtension::Write(ArrayView<uint8_t> data,
+bool ColorSpaceExtension::Write(std::span<uint8_t> data,
                                 const ColorSpace& color_space) {
   RTC_DCHECK_EQ(data.size(), ValueSize(color_space));
   size_t offset = 0;
@@ -649,7 +649,7 @@ uint8_t ColorSpaceExtension::CombineRangeAndChromaSiting(
          static_cast<uint8_t>(chroma_siting_vertical);
 }
 
-size_t ColorSpaceExtension::ParseHdrMetadata(ArrayView<const uint8_t> data,
+size_t ColorSpaceExtension::ParseHdrMetadata(std::span<const uint8_t> data,
                                              HdrMetadata* hdr_metadata) {
   RTC_DCHECK_EQ(data.size(),
                 kValueSizeBytes - kValueSizeBytesWithoutHdrMetadata);
@@ -696,7 +696,7 @@ size_t ColorSpaceExtension::ParseLuminance(const uint8_t* data,
   return 2;  // Return number of bytes read.
 }
 
-size_t ColorSpaceExtension::WriteHdrMetadata(ArrayView<uint8_t> data,
+size_t ColorSpaceExtension::WriteHdrMetadata(std::span<uint8_t> data,
                                              const HdrMetadata& hdr_metadata) {
   RTC_DCHECK_EQ(data.size(),
                 kValueSizeBytes - kValueSizeBytesWithoutHdrMetadata);
@@ -750,7 +750,7 @@ size_t ColorSpaceExtension::WriteLuminance(uint8_t* data,
   return 2;  // Return number of bytes written.
 }
 
-bool BaseRtpStringExtension::Parse(ArrayView<const uint8_t> data,
+bool BaseRtpStringExtension::Parse(std::span<const uint8_t> data,
                                    std::string* str) {
   if (data.empty() || data[0] == 0)  // Valid string extension can't be empty.
     return false;
@@ -762,7 +762,7 @@ bool BaseRtpStringExtension::Parse(ArrayView<const uint8_t> data,
   return true;
 }
 
-bool BaseRtpStringExtension::Write(ArrayView<uint8_t> data,
+bool BaseRtpStringExtension::Write(std::span<uint8_t> data,
                                    absl::string_view str) {
   if (str.size() > kMaxValueSizeBytes) {
     return false;
@@ -790,7 +790,7 @@ bool BaseRtpStringExtension::Write(ArrayView<uint8_t> data,
 // |      ID       |     len=1     |N|    level    |
 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 // Sample Audio Level Encoding Using the Two-Byte Header Format
-bool InbandComfortNoiseExtension::Parse(ArrayView<const uint8_t> data,
+bool InbandComfortNoiseExtension::Parse(std::span<const uint8_t> data,
                                         std::optional<uint8_t>* level) {
   if (data.size() != kValueSizeBytes)
     return false;
@@ -800,7 +800,7 @@ bool InbandComfortNoiseExtension::Parse(ArrayView<const uint8_t> data,
   return true;
 }
 
-bool InbandComfortNoiseExtension::Write(ArrayView<uint8_t> data,
+bool InbandComfortNoiseExtension::Write(std::span<uint8_t> data,
                                         std::optional<uint8_t> level) {
   RTC_DCHECK_EQ(data.size(), kValueSizeBytes);
   data[0] = 0b0000'0000;
@@ -820,7 +820,7 @@ bool InbandComfortNoiseExtension::Write(ArrayView<uint8_t> data,
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 //  |  ID   | L=1   |    video-frame-tracking-id    |
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-bool VideoFrameTrackingIdExtension::Parse(ArrayView<const uint8_t> data,
+bool VideoFrameTrackingIdExtension::Parse(std::span<const uint8_t> data,
                                           uint16_t* video_frame_tracking_id) {
   if (data.size() != kValueSizeBytes) {
     return false;
@@ -829,7 +829,7 @@ bool VideoFrameTrackingIdExtension::Parse(ArrayView<const uint8_t> data,
   return true;
 }
 
-bool VideoFrameTrackingIdExtension::Write(ArrayView<uint8_t> data,
+bool VideoFrameTrackingIdExtension::Write(std::span<uint8_t> data,
                                           uint16_t video_frame_tracking_id) {
   RTC_DCHECK_EQ(data.size(), kValueSizeBytes);
   ByteWriter<uint16_t>::WriteBigEndian(data.data(), video_frame_tracking_id);
diff --git a/modules/rtp_rtcp/source/rtp_header_extensions.h b/modules/rtp_rtcp/source/rtp_header_extensions.h
index 8fb26e5522..03f846a905 100644
--- a/modules/rtp_rtcp/source/rtp_header_extensions.h
+++ b/modules/rtp_rtcp/source/rtp_header_extensions.h
@@ -15,11 +15,11 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtp_headers.h"
 #include "api/rtp_parameters.h"
 #include "api/units/time_delta.h"
@@ -51,11 +51,11 @@ class AbsoluteSendTime {
     return RtpExtension::kAbsSendTimeUri;
   }
 
-  static bool Parse(ArrayView<const uint8_t> data, uint32_t* time_24bits);
+  static bool Parse(std::span<const uint8_t> data, uint32_t* time_24bits);
   static size_t ValueSize(uint32_t /* time_24bits */) {
     return kValueSizeBytes;
   }
-  static bool Write(ArrayView<uint8_t> data, uint32_t time_24bits);
+  static bool Write(std::span<uint8_t> data, uint32_t time_24bits);
 
   static constexpr uint32_t To24Bits(Timestamp time) {
     int64_t time_us = time.us() % (int64_t{1 << 6} * 1'000'000);
@@ -87,10 +87,10 @@ class AbsoluteCaptureTimeExtension {
     return RtpExtension::kAbsoluteCaptureTimeUri;
   }
 
-  static bool Parse(ArrayView<const uint8_t> data,
+  static bool Parse(std::span<const uint8_t> data,
                     AbsoluteCaptureTime* extension);
   static size_t ValueSize(const AbsoluteCaptureTime& extension);
-  static bool Write(ArrayView<uint8_t> data,
+  static bool Write(std::span<uint8_t> data,
                     const AbsoluteCaptureTime& extension);
 };
 
@@ -103,11 +103,11 @@ class AudioLevelExtension {
     return RtpExtension::kAudioLevelUri;
   }
 
-  static bool Parse(ArrayView<const uint8_t> data, AudioLevel* extension);
+  static bool Parse(std::span<const uint8_t> data, AudioLevel* extension);
   static size_t ValueSize(const AudioLevel& /* extension */) {
     return kValueSizeBytes;
   }
-  static bool Write(ArrayView<uint8_t> data, const AudioLevel& extension);
+  static bool Write(std::span<uint8_t> data, const AudioLevel& extension);
 };
 
 class CsrcAudioLevel {
@@ -118,11 +118,11 @@ class CsrcAudioLevel {
     return RtpExtension::kCsrcAudioLevelsUri;
   }
 
-  static bool Parse(ArrayView<const uint8_t> data,
+  static bool Parse(std::span<const uint8_t> data,
                     std::vector<uint8_t>* csrc_audio_levels);
-  static size_t ValueSize(ArrayView<const uint8_t> csrc_audio_levels);
-  static bool Write(ArrayView<uint8_t> data,
-                    ArrayView<const uint8_t> csrc_audio_levels);
+  static size_t ValueSize(std::span<const uint8_t> csrc_audio_levels);
+  static bool Write(std::span<uint8_t> data,
+                    std::span<const uint8_t> csrc_audio_levels);
 };
 
 class TransmissionOffset {
@@ -134,9 +134,9 @@ class TransmissionOffset {
     return RtpExtension::kTimestampOffsetUri;
   }
 
-  static bool Parse(ArrayView<const uint8_t> data, int32_t* rtp_time);
+  static bool Parse(std::span<const uint8_t> data, int32_t* rtp_time);
   static size_t ValueSize(int32_t /* rtp_time */) { return kValueSizeBytes; }
-  static bool Write(ArrayView<uint8_t> data, int32_t rtp_time);
+  static bool Write(std::span<uint8_t> data, int32_t rtp_time);
 };
 
 class TransportSequenceNumber {
@@ -148,12 +148,12 @@ class TransportSequenceNumber {
     return RtpExtension::kTransportSequenceNumberUri;
   }
 
-  static bool Parse(ArrayView<const uint8_t> data,
+  static bool Parse(std::span<const uint8_t> data,
                     uint16_t* transport_sequence_number);
   static size_t ValueSize(uint16_t /*transport_sequence_number*/) {
     return kValueSizeBytes;
   }
-  static bool Write(ArrayView<uint8_t> data,
+  static bool Write(std::span<uint8_t> data,
                     uint16_t transport_sequence_number);
 };
 
@@ -167,7 +167,7 @@ class TransportSequenceNumberV2 {
     return RtpExtension::kTransportSequenceNumberV2Uri;
   }
 
-  static bool Parse(ArrayView<const uint8_t> data,
+  static bool Parse(std::span<const uint8_t> data,
                     uint16_t* transport_sequence_number,
                     std::optional<FeedbackRequest>* feedback_request);
   static size_t ValueSize(
@@ -176,7 +176,7 @@ class TransportSequenceNumberV2 {
     return feedback_request ? kValueSizeBytes
                             : kValueSizeBytesWithoutFeedbackRequest;
   }
-  static bool Write(ArrayView<uint8_t> data,
+  static bool Write(std::span<uint8_t> data,
                     uint16_t transport_sequence_number,
                     const std::optional<FeedbackRequest>& feedback_request);
 
@@ -193,12 +193,12 @@ class VideoOrientation {
     return RtpExtension::kVideoRotationUri;
   }
 
-  static bool Parse(ArrayView<const uint8_t> data, VideoRotation* value);
+  static bool Parse(std::span<const uint8_t> data, VideoRotation* value);
   static size_t ValueSize(VideoRotation) { return kValueSizeBytes; }
-  static bool Write(ArrayView<uint8_t> data, VideoRotation value);
-  static bool Parse(ArrayView<const uint8_t> data, uint8_t* value);
+  static bool Write(std::span<uint8_t> data, VideoRotation value);
+  static bool Parse(std::span<const uint8_t> data, uint8_t* value);
   static size_t ValueSize(uint8_t /* value */) { return kValueSizeBytes; }
-  static bool Write(ArrayView<uint8_t> data, uint8_t value);
+  static bool Write(std::span<uint8_t> data, uint8_t value);
 };
 
 class PlayoutDelayLimits {
@@ -217,10 +217,10 @@ class PlayoutDelayLimits {
   // Maximum playout delay value in milliseconds.
   static constexpr TimeDelta kMax = 0xfff * kGranularity;  // 40950.
 
-  static bool Parse(ArrayView<const uint8_t> data,
+  static bool Parse(std::span<const uint8_t> data,
                     VideoPlayoutDelay* playout_delay);
   static size_t ValueSize(const VideoPlayoutDelay&) { return kValueSizeBytes; }
-  static bool Write(ArrayView<uint8_t> data,
+  static bool Write(std::span<uint8_t> data,
                     const VideoPlayoutDelay& playout_delay);
 };
 
@@ -233,10 +233,10 @@ class VideoContentTypeExtension {
     return RtpExtension::kVideoContentTypeUri;
   }
 
-  static bool Parse(ArrayView<const uint8_t> data,
+  static bool Parse(std::span<const uint8_t> data,
                     VideoContentType* content_type);
   static size_t ValueSize(VideoContentType) { return kValueSizeBytes; }
-  static bool Write(ArrayView<uint8_t> data, VideoContentType content_type);
+  static bool Write(std::span<uint8_t> data, VideoContentType content_type);
 };
 
 class VideoTimingExtension {
@@ -258,15 +258,15 @@ class VideoTimingExtension {
   static constexpr uint8_t kNetworkTimestampDeltaOffset = 9;
   static constexpr uint8_t kNetwork2TimestampDeltaOffset = 11;
 
-  static bool Parse(ArrayView<const uint8_t> data, VideoSendTiming* timing);
+  static bool Parse(std::span<const uint8_t> data, VideoSendTiming* timing);
   static size_t ValueSize(const VideoSendTiming&) { return kValueSizeBytes; }
-  static bool Write(ArrayView<uint8_t> data, const VideoSendTiming& timing);
+  static bool Write(std::span<uint8_t> data, const VideoSendTiming& timing);
 
   static size_t ValueSize(uint16_t /* time_delta_ms */, uint8_t /* idx */) {
     return kValueSizeBytes;
   }
   // Writes only single time delta to position idx.
-  static bool Write(ArrayView<uint8_t> data,
+  static bool Write(std::span<uint8_t> data,
                     uint16_t time_delta_ms,
                     uint8_t offset);
 };
@@ -281,12 +281,12 @@ class ColorSpaceExtension {
     return RtpExtension::kColorSpaceUri;
   }
 
-  static bool Parse(ArrayView<const uint8_t> data, ColorSpace* color_space);
+  static bool Parse(std::span<const uint8_t> data, ColorSpace* color_space);
   static size_t ValueSize(const ColorSpace& color_space) {
     return color_space.hdr_metadata() ? kValueSizeBytes
                                       : kValueSizeBytesWithoutHdrMetadata;
   }
-  static bool Write(ArrayView<uint8_t> data, const ColorSpace& color_space);
+  static bool Write(std::span<uint8_t> data, const ColorSpace& color_space);
 
  private:
   static constexpr int kChromaticityDenominator = 50000;  // 0.00002 resolution.
@@ -297,12 +297,12 @@ class ColorSpaceExtension {
       ColorSpace::RangeID range,
       ColorSpace::ChromaSiting chroma_siting_horizontal,
       ColorSpace::ChromaSiting chroma_siting_vertical);
-  static size_t ParseHdrMetadata(ArrayView<const uint8_t> data,
+  static size_t ParseHdrMetadata(std::span<const uint8_t> data,
                                  HdrMetadata* hdr_metadata);
   static size_t ParseChromaticity(const uint8_t* data,
                                   HdrMasteringMetadata::Chromaticity* p);
   static size_t ParseLuminance(const uint8_t* data, float* f, int denominator);
-  static size_t WriteHdrMetadata(ArrayView<uint8_t> data,
+  static size_t WriteHdrMetadata(std::span<uint8_t> data,
                                  const HdrMetadata& hdr_metadata);
   static size_t WriteChromaticity(uint8_t* data,
                                   const HdrMasteringMetadata::Chromaticity& p);
@@ -318,9 +318,9 @@ class BaseRtpStringExtension {
   // maximum length that can be encoded with one-byte header extensions.
   static constexpr uint8_t kMaxValueSizeBytes = 16;
 
-  static bool Parse(ArrayView<const uint8_t> data, std::string* str);
+  static bool Parse(std::span<const uint8_t> data, std::string* str);
   static size_t ValueSize(absl::string_view str) { return str.size(); }
-  static bool Write(ArrayView<uint8_t> data, absl::string_view str);
+  static bool Write(std::span<uint8_t> data, absl::string_view str);
 };
 
 class RtpStreamId : public BaseRtpStringExtension {
@@ -353,12 +353,12 @@ class InbandComfortNoiseExtension {
       "http://www.webrtc.org/experiments/rtp-hdrext/inband-cn";
   static constexpr absl::string_view Uri() { return kUri; }
 
-  static bool Parse(ArrayView<const uint8_t> data,
+  static bool Parse(std::span<const uint8_t> data,
                     std::optional<uint8_t>* level);
   static size_t ValueSize(std::optional<uint8_t> /* level */) {
     return kValueSizeBytes;
   }
-  static bool Write(ArrayView<uint8_t> data, std::optional<uint8_t> level);
+  static bool Write(std::span<uint8_t> data, std::optional<uint8_t> level);
 };
 
 class VideoFrameTrackingIdExtension {
@@ -370,12 +370,12 @@ class VideoFrameTrackingIdExtension {
     return RtpExtension::kVideoFrameTrackingIdUri;
   }
 
-  static bool Parse(ArrayView<const uint8_t> data,
+  static bool Parse(std::span<const uint8_t> data,
                     uint16_t* video_frame_tracking_id);
   static size_t ValueSize(uint16_t /*video_frame_tracking_id*/) {
     return kValueSizeBytes;
   }
-  static bool Write(ArrayView<uint8_t> data, uint16_t video_frame_tracking_id);
+  static bool Write(std::span<uint8_t> data, uint16_t video_frame_tracking_id);
 };
 
 }  // namespace webrtc
diff --git a/modules/rtp_rtcp/source/rtp_packet.cc b/modules/rtp_rtcp/source/rtp_packet.cc
index 963519dd0c..97f6b54148 100644
--- a/modules/rtp_rtcp/source/rtp_packet.cc
+++ b/modules/rtp_rtcp/source/rtp_packet.cc
@@ -12,11 +12,11 @@
 
 #include <cstdint>
 #include <cstring>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/rtp_parameters.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/byte_io.h"
@@ -93,7 +93,7 @@ bool RtpPacket::Parse(const uint8_t* buffer, size_t buffer_size) {
   return true;
 }
 
-bool RtpPacket::Parse(ArrayView<const uint8_t> packet) {
+bool RtpPacket::Parse(std::span<const uint8_t> packet) {
   return Parse(packet.data(), packet.size());
 }
 
@@ -216,7 +216,7 @@ void RtpPacket::ZeroMutableExtensions() {
   }
 }
 
-void RtpPacket::SetCsrcs(ArrayView<const uint32_t> csrcs) {
+void RtpPacket::SetCsrcs(std::span<const uint32_t> csrcs) {
   RTC_DCHECK_EQ(extensions_size_, 0);
   RTC_DCHECK_EQ(payload_size_, 0);
   RTC_DCHECK_EQ(padding_size_, 0);
@@ -232,7 +232,7 @@ void RtpPacket::SetCsrcs(ArrayView<const uint32_t> csrcs) {
   buffer_.SetSize(payload_offset_);
 }
 
-ArrayView<uint8_t> RtpPacket::AllocateRawExtension(int id, size_t length) {
+std::span<uint8_t> RtpPacket::AllocateRawExtension(int id, size_t length) {
   RTC_DCHECK_GE(id, RtpExtension::kMinId);
   RTC_DCHECK_LE(id, RtpExtension::kMaxId);
   RTC_DCHECK_GE(length, 1);
@@ -241,7 +241,7 @@ ArrayView<uint8_t> RtpPacket::AllocateRawExtension(int id, size_t length) {
   if (extension_entry != nullptr) {
     // Extension already reserved. Check if same length is used.
     if (extension_entry->length == length)
-      return MakeArrayView(WriteAt(extension_entry->offset), length);
+      return std::span(WriteAt(extension_entry->offset), length);
 
     RTC_LOG(LS_ERROR) << "Length mismatch for extension id " << id
                       << ": expected "
@@ -342,7 +342,7 @@ ArrayView<uint8_t> RtpPacket::AllocateRawExtension(int id, size_t length) {
       SetExtensionLengthMaybeAddZeroPadding(extensions_offset);
   payload_offset_ = extensions_offset + extensions_size_padded;
   buffer_.SetSize(payload_offset_);
-  return MakeArrayView(WriteAt(extension_info_offset), extension_info_length);
+  return std::span(WriteAt(extension_info_offset), extension_info_length);
 }
 
 void RtpPacket::PromoteToTwoByteHeaderExtension() {
@@ -402,7 +402,7 @@ uint8_t* RtpPacket::AllocatePayload(size_t size_bytes) {
   return SetPayloadSize(size_bytes);
 }
 
-void RtpPacket::SetPayload(ArrayView<const uint8_t> payload) {
+void RtpPacket::SetPayload(std::span<const uint8_t> payload) {
   if (payload.empty()) {
     SetPayloadSize(0);
     return;
@@ -594,7 +594,7 @@ RtpPacket::ExtensionInfo& RtpPacket::FindOrCreateExtensionInfo(int id) {
   return extension_entries_.back();
 }
 
-ArrayView<const uint8_t> RtpPacket::FindExtension(ExtensionType type) const {
+std::span<const uint8_t> RtpPacket::FindExtension(ExtensionType type) const {
   uint8_t id = extensions_.GetId(type);
   if (id == ExtensionManager::kInvalidId) {
     // Extension not registered.
@@ -604,10 +604,10 @@ ArrayView<const uint8_t> RtpPacket::FindExtension(ExtensionType type) const {
   if (extension_info == nullptr) {
     return {};
   }
-  return MakeArrayView(data() + extension_info->offset, extension_info->length);
+  return std::span(data() + extension_info->offset, extension_info->length);
 }
 
-ArrayView<uint8_t> RtpPacket::AllocateExtension(ExtensionType type,
+std::span<uint8_t> RtpPacket::AllocateExtension(ExtensionType type,
                                                 size_t length) {
   // TODO(webrtc:7990): Add support for empty extensions (length==0).
   if (length == 0 || length > RtpExtension::kMaxValueSize ||
diff --git a/modules/rtp_rtcp/source/rtp_packet.h b/modules/rtp_rtcp/source/rtp_packet.h
index 9abdbc6591..1a9f91799b 100644
--- a/modules/rtp_rtcp/source/rtp_packet.h
+++ b/modules/rtp_rtcp/source/rtp_packet.h
@@ -14,11 +14,11 @@
 #include <cstdint>
 #include <cstring>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "rtc_base/copy_on_write_buffer.h"
@@ -52,7 +52,7 @@ class RtpPacket {
   // read or allocate extensions in methods GetExtension, AllocateExtension,
   // etc.)
   bool Parse(const uint8_t* buffer, size_t size);
-  bool Parse(ArrayView<const uint8_t> packet);
+  bool Parse(std::span<const uint8_t> packet);
 
   // Parse and move given buffer into Packet.
   bool Parse(CopyOnWriteBuffer packet);
@@ -77,8 +77,8 @@ class RtpPacket {
   size_t payload_size() const { return payload_size_; }
   bool has_padding() const { return buffer_[0] & 0x20; }
   size_t padding_size() const { return padding_size_; }
-  ArrayView<const uint8_t> payload() const {
-    return MakeArrayView(data() + payload_offset_, payload_size_);
+  std::span<const uint8_t> payload() const {
+    return std::span(data() + payload_offset_, payload_size_);
   }
   CopyOnWriteBuffer PayloadBuffer() const {
     return buffer_.Slice(payload_offset_, payload_size_);
@@ -91,7 +91,7 @@ class RtpPacket {
     return payload_offset_ + payload_size_ + padding_size_;
   }
   const uint8_t* data() const { return buffer_.cdata(); }
-  ArrayView<const uint8_t> buffer() const { return buffer_; }
+  std::span<const uint8_t> buffer() const { return buffer_; }
   size_t FreeCapacity() const { return capacity() - size(); }
   size_t MaxPayloadSize() const { return capacity() - headers_size(); }
 
@@ -119,7 +119,7 @@ class RtpPacket {
   // Writes csrc list. Assumes:
   // a) There is enough room left in buffer.
   // b) Extension headers, payload or padding data has not already been added.
-  void SetCsrcs(ArrayView<const uint32_t> csrcs);
+  void SetCsrcs(std::span<const uint32_t> csrcs);
 
   // Header extensions.
   template <typename Extension>
@@ -139,24 +139,24 @@ class RtpPacket {
 
   // Returns view of the raw extension or empty view on failure.
   template <typename Extension>
-  ArrayView<const uint8_t> GetRawExtension() const;
+  std::span<const uint8_t> GetRawExtension() const;
 
   template <typename Extension, typename... Values>
   bool SetExtension(const Values&...);
 
   template <typename Extension>
-  bool SetRawExtension(ArrayView<const uint8_t> data);
+  bool SetRawExtension(std::span<const uint8_t> data);
 
   template <typename Extension>
   bool ReserveExtension();
 
   // Find or allocate an extension `type`. Returns view of size `length`
   // to write raw extension to or an empty view on failure.
-  ArrayView<uint8_t> AllocateExtension(ExtensionType type, size_t length);
+  std::span<uint8_t> AllocateExtension(ExtensionType type, size_t length);
 
   // Find an extension `type`.
   // Returns view of the raw extension or empty view on failure.
-  ArrayView<const uint8_t> FindExtension(ExtensionType type) const;
+  std::span<const uint8_t> FindExtension(ExtensionType type) const;
 
   // Returns pointer to the payload of size at least `size_bytes`.
   // Keeps original payload, if any. If `size_bytes` is larger than current
@@ -167,7 +167,7 @@ class RtpPacket {
   uint8_t* AllocatePayload(size_t size_bytes);
 
   // Sets payload size to `payload.size()` and copies `payload`.
-  void SetPayload(ArrayView<const uint8_t> payload);
+  void SetPayload(std::span<const uint8_t> payload);
 
   bool SetPadding(size_t padding_size);
 
@@ -198,7 +198,7 @@ class RtpPacket {
 
   // Allocates and returns place to store rtp header extension.
   // Returns empty arrayview on failure.
-  ArrayView<uint8_t> AllocateRawExtension(int id, size_t length);
+  std::span<uint8_t> AllocateRawExtension(int id, size_t length);
 
   // Promotes existing one-byte header extensions to two-byte header extensions
   // by rewriting the data and updates the corresponding extension offsets.
@@ -257,7 +257,7 @@ std::optional<typename Extension::value_type> RtpPacket::GetExtension() const {
 }
 
 template <typename Extension>
-ArrayView<const uint8_t> RtpPacket::GetRawExtension() const {
+std::span<const uint8_t> RtpPacket::GetRawExtension() const {
   return FindExtension(Extension::kId);
 }
 
@@ -271,8 +271,8 @@ bool RtpPacket::SetExtension(const Values&... values) {
 }
 
 template <typename Extension>
-bool RtpPacket::SetRawExtension(ArrayView<const uint8_t> data) {
-  ArrayView<uint8_t> buffer = AllocateExtension(Extension::kId, data.size());
+bool RtpPacket::SetRawExtension(std::span<const uint8_t> data) {
+  std::span<uint8_t> buffer = AllocateExtension(Extension::kId, data.size());
   if (buffer.empty()) {
     return false;
   }
diff --git a/modules/rtp_rtcp/source/rtp_packet_history.cc b/modules/rtp_rtcp/source/rtp_packet_history.cc
index 9b2db996dd..f2154240fd 100644
--- a/modules/rtp_rtcp/source/rtp_packet_history.cc
+++ b/modules/rtp_rtcp/source/rtp_packet_history.cc
@@ -16,9 +16,9 @@
 #include <limits>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/function_view.h"
 #include "api/units/time_delta.h"
@@ -299,7 +299,7 @@ std::unique_ptr<RtpPacketToSend> RtpPacketHistory::GetPayloadPaddingPacket(
 }
 
 void RtpPacketHistory::CullAcknowledgedPackets(
-    ArrayView<const uint16_t> sequence_numbers) {
+    std::span<const uint16_t> sequence_numbers) {
   MutexLock lock(&lock_);
   for (uint16_t sequence_number : sequence_numbers) {
     int packet_index = GetPacketIndex(sequence_number);
diff --git a/modules/rtp_rtcp/source/rtp_packet_history.h b/modules/rtp_rtcp/source/rtp_packet_history.h
index ebd2f9119a..9a8c8825fd 100644
--- a/modules/rtp_rtcp/source/rtp_packet_history.h
+++ b/modules/rtp_rtcp/source/rtp_packet_history.h
@@ -16,8 +16,8 @@
 #include <deque>
 #include <memory>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/function_view.h"
 #include "api/units/time_delta.h"
@@ -116,7 +116,7 @@ class RtpPacketHistory {
           encapsulate);
 
   // Cull packets that have been acknowledged as received by the remote end.
-  void CullAcknowledgedPackets(ArrayView<const uint16_t> sequence_numbers);
+  void CullAcknowledgedPackets(std::span<const uint16_t> sequence_numbers);
 
   // Remove all pending packets from the history, but keep storage mode and
   // capacity.
diff --git a/modules/rtp_rtcp/source/rtp_packet_unittest.cc b/modules/rtp_rtcp/source/rtp_packet_unittest.cc
index 4f696985a6..37b5f62d66 100644
--- a/modules/rtp_rtcp/source/rtp_packet_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_packet_unittest.cc
@@ -13,12 +13,12 @@
 #include <cstdint>
 #include <cstring>
 #include <optional>
+#include <span>
 #include <string>
 #include <type_traits>
 #include <utility>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtp_headers.h"
 #include "api/units/time_delta.h"
 #include "api/video/color_space.h"
@@ -437,7 +437,7 @@ TEST(RtpPacketTest, SetReservedExtensionsAfterPayload) {
 }
 
 TEST(RtpPacketTest, SetEmptyPayload) {
-  ArrayView<const uint8_t> empty_payload;
+  std::span<const uint8_t> empty_payload;
   RtpPacket packet;
   packet.SetPayload(empty_payload);
 
@@ -446,7 +446,7 @@ TEST(RtpPacketTest, SetEmptyPayload) {
 
 TEST(RtpPacketTest, SetEmptyPayloadOverwritesExistingPayload) {
   const uint8_t payload[] = {1, 2, 3, 4, 2, 0, 42};
-  ArrayView<const uint8_t> empty_payload;
+  std::span<const uint8_t> empty_payload;
   RtpPacket packet;
 
   packet.SetPayload(payload);
@@ -505,7 +505,7 @@ TEST(RtpPacketTest, UsesZerosForPadding) {
   RtpPacket packet;
 
   EXPECT_TRUE(packet.SetPadding(kPaddingSize));
-  EXPECT_THAT(MakeArrayView(packet.data() + 12, kPaddingSize - 1), Each(0));
+  EXPECT_THAT(std::span(packet.data() + 12, kPaddingSize - 1), Each(0));
 }
 
 TEST(RtpPacketTest, CreateOneBytePadding) {
@@ -920,11 +920,11 @@ struct UncopyableExtension {
   static constexpr absl::string_view Uri() { return "uri"; }
 
   static size_t ValueSize(const UncopyableValue& /* value */) { return 1; }
-  static bool Write(ArrayView<uint8_t> /* data */,
+  static bool Write(std::span<uint8_t> /* data */,
                     const UncopyableValue& /* value */) {
     return true;
   }
-  static bool Parse(ArrayView<const uint8_t> /* data */,
+  static bool Parse(std::span<const uint8_t> /* data */,
                     UncopyableValue* /* value */) {
     return true;
   }
@@ -957,12 +957,12 @@ struct ParseByReferenceExtension {
   static size_t ValueSize(uint8_t /* value1 */, uint8_t /* value2 */) {
     return 2;
   }
-  static bool Write(ArrayView<uint8_t> data, uint8_t value1, uint8_t value2) {
+  static bool Write(std::span<uint8_t> data, uint8_t value1, uint8_t value2) {
     data[0] = value1;
     data[1] = value2;
     return true;
   }
-  static bool Parse(ArrayView<const uint8_t> data,
+  static bool Parse(std::span<const uint8_t> data,
                     uint8_t& value1,
                     uint8_t& value2) {
     value1 = data[0];
diff --git a/modules/rtp_rtcp/source/rtp_packetizer_av1.cc b/modules/rtp_rtcp/source/rtp_packetizer_av1.cc
index e96460e53b..e9c61a8972 100644
--- a/modules/rtp_rtcp/source/rtp_packetizer_av1.cc
+++ b/modules/rtp_rtcp/source/rtp_packetizer_av1.cc
@@ -13,9 +13,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <cstring>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/video/video_frame_type.h"
 #include "modules/rtp_rtcp/source/leb128.h"
 #include "modules/rtp_rtcp/source/rtp_format.h"
@@ -70,7 +70,7 @@ int MaxFragmentSize(int remaining_bytes) {
 
 }  // namespace
 
-RtpPacketizerAv1::RtpPacketizerAv1(ArrayView<const uint8_t> payload,
+RtpPacketizerAv1::RtpPacketizerAv1(std::span<const uint8_t> payload,
                                    RtpPacketizer::PayloadSizeLimits limits,
                                    VideoFrameType frame_type,
                                    bool is_last_frame_in_picture)
@@ -80,7 +80,7 @@ RtpPacketizerAv1::RtpPacketizerAv1(ArrayView<const uint8_t> payload,
       is_last_frame_in_picture_(is_last_frame_in_picture) {}
 
 std::vector<RtpPacketizerAv1::Obu> RtpPacketizerAv1::ParseObus(
-    ArrayView<const uint8_t> payload) {
+    std::span<const uint8_t> payload) {
   std::vector<Obu> result;
   ByteBufferReader payload_reader(payload);
   while (payload_reader.Length() > 0) {
@@ -99,8 +99,8 @@ std::vector<RtpPacketizerAv1::Obu> RtpPacketizerAv1::ParseObus(
     }
     if (!ObuHasSize(obu.header)) {
       obu.payload =
-          MakeArrayView(reinterpret_cast<const uint8_t*>(payload_reader.Data()),
-                        payload_reader.Length());
+          std::span(reinterpret_cast<const uint8_t*>(payload_reader.Data()),
+                    payload_reader.Length());
       payload_reader.Consume(payload_reader.Length());
     } else {
       uint64_t size = 0;
@@ -111,7 +111,7 @@ std::vector<RtpPacketizerAv1::Obu> RtpPacketizerAv1::ParseObus(
                            << payload_reader.Length();
         return {};
       }
-      obu.payload = MakeArrayView(
+      obu.payload = std::span(
           reinterpret_cast<const uint8_t*>(payload_reader.Data()), size);
       payload_reader.Consume(size);
     }
@@ -147,7 +147,7 @@ int RtpPacketizerAv1::AdditionalBytesForPreviousObuElement(
 }
 
 std::vector<RtpPacketizerAv1::Packet> RtpPacketizerAv1::PacketizeInternal(
-    ArrayView<const Obu> obus,
+    std::span<const Obu> obus,
     PayloadSizeLimits limits) {
   std::vector<Packet> packets;
   if (obus.empty()) {
@@ -300,7 +300,7 @@ std::vector<RtpPacketizerAv1::Packet> RtpPacketizerAv1::PacketizeInternal(
 }
 
 std::vector<RtpPacketizerAv1::Packet> RtpPacketizerAv1::Packetize(
-    ArrayView<const Obu> obus,
+    std::span<const Obu> obus,
     PayloadSizeLimits limits) {
   std::vector<Packet> packets = PacketizeInternal(obus, limits);
   if (packets.size() <= 1) {
diff --git a/modules/rtp_rtcp/source/rtp_packetizer_av1.h b/modules/rtp_rtcp/source/rtp_packetizer_av1.h
index b64418aace..c121877ff8 100644
--- a/modules/rtp_rtcp/source/rtp_packetizer_av1.h
+++ b/modules/rtp_rtcp/source/rtp_packetizer_av1.h
@@ -14,9 +14,9 @@
 #include <stddef.h>
 #include <stdint.h>
 
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/video/video_frame_type.h"
 #include "modules/rtp_rtcp/source/rtp_format.h"
 
@@ -24,7 +24,7 @@ namespace webrtc {
 
 class RtpPacketizerAv1 : public RtpPacketizer {
  public:
-  RtpPacketizerAv1(ArrayView<const uint8_t> payload,
+  RtpPacketizerAv1(std::span<const uint8_t> payload,
                    PayloadSizeLimits limits,
                    VideoFrameType frame_type,
                    bool is_last_frame_in_picture);
@@ -37,7 +37,7 @@ class RtpPacketizerAv1 : public RtpPacketizer {
   struct Obu {
     uint8_t header;
     uint8_t extension_header;  // undefined if (header & kXbit) == 0
-    ArrayView<const uint8_t> payload;
+    std::span<const uint8_t> payload;
     int size;  // size of the header and payload combined.
   };
   struct Packet {
@@ -53,14 +53,14 @@ class RtpPacketizerAv1 : public RtpPacketizer {
   };
 
   // Parses the payload into serie of OBUs.
-  static std::vector<Obu> ParseObus(ArrayView<const uint8_t> payload);
+  static std::vector<Obu> ParseObus(std::span<const uint8_t> payload);
   // Returns the number of additional bytes needed to store the previous OBU
   // element if an additonal OBU element is added to the packet.
   static int AdditionalBytesForPreviousObuElement(const Packet& packet);
-  static std::vector<Packet> PacketizeInternal(ArrayView<const Obu> obus,
+  static std::vector<Packet> PacketizeInternal(std::span<const Obu> obus,
                                                PayloadSizeLimits limits);
   // Packetize and try to distribute the payload evenly across packets.
-  static std::vector<Packet> Packetize(ArrayView<const Obu> obus,
+  static std::vector<Packet> Packetize(std::span<const Obu> obus,
                                        PayloadSizeLimits limits);
 
   uint8_t AggregationHeader() const;
diff --git a/modules/rtp_rtcp/source/rtp_packetizer_av1_unittest.cc b/modules/rtp_rtcp/source/rtp_packetizer_av1_unittest.cc
index 38b7f583ad..2a5a8cadc4 100644
--- a/modules/rtp_rtcp/source/rtp_packetizer_av1_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_packetizer_av1_unittest.cc
@@ -13,10 +13,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <initializer_list>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/scoped_refptr.h"
 #include "api/video/encoded_image.h"
 #include "api/video/video_frame_type.h"
@@ -40,8 +40,8 @@ constexpr uint8_t kNewCodedVideoSequenceBit = 0b00'00'1000;
 
 // Wrapper around rtp_packet to make it look like container of payload bytes.
 struct RtpPayload {
-  using value_type = ArrayView<const uint8_t>::value_type;
-  using const_iterator = ArrayView<const uint8_t>::iterator;
+  using value_type = std::span<const uint8_t>::value_type;
+  using const_iterator = std::span<const uint8_t>::iterator;
 
   RtpPayload() : rtp_packet(/*extensions=*/nullptr) {}
   RtpPayload& operator=(RtpPayload&&) = default;
@@ -77,7 +77,7 @@ class Av1Frame {
 };
 
 std::vector<RtpPayload> Packetize(
-    ArrayView<const uint8_t> payload,
+    std::span<const uint8_t> payload,
     RtpPacketizer::PayloadSizeLimits limits,
     VideoFrameType frame_type = VideoFrameType::kVideoFrameDelta,
     bool is_last_frame_in_picture = true) {
@@ -92,8 +92,8 @@ std::vector<RtpPayload> Packetize(
   return result;
 }
 
-Av1Frame ReassembleFrame(ArrayView<const RtpPayload> rtp_payloads) {
-  std::vector<ArrayView<const uint8_t>> payloads(rtp_payloads.size());
+Av1Frame ReassembleFrame(std::span<const RtpPayload> rtp_payloads) {
+  std::vector<std::span<const uint8_t>> payloads(rtp_payloads.size());
   for (size_t i = 0; i < rtp_payloads.size(); ++i) {
     payloads[i] = rtp_payloads[i];
   }
diff --git a/modules/rtp_rtcp/source/rtp_packetizer_h265.cc b/modules/rtp_rtcp/source/rtp_packetizer_h265.cc
index f848126031..0311754394 100644
--- a/modules/rtp_rtcp/source/rtp_packetizer_h265.cc
+++ b/modules/rtp_rtcp/source/rtp_packetizer_h265.cc
@@ -14,9 +14,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <cstring>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "common_video/h264/h264_common.h"
 #include "common_video/h265/h265_common.h"
 #include "modules/rtp_rtcp/source/byte_io.h"
@@ -26,7 +26,7 @@
 
 namespace webrtc {
 
-RtpPacketizerH265::RtpPacketizerH265(ArrayView<const uint8_t> payload,
+RtpPacketizerH265::RtpPacketizerH265(std::span<const uint8_t> payload,
                                      PayloadSizeLimits limits)
     : limits_(limits), num_packets_left_(0) {
   for (const H264::NaluIndex& nalu : H264::FindNaluIndices(payload)) {
@@ -84,7 +84,7 @@ bool RtpPacketizerH265::GeneratePackets() {
 bool RtpPacketizerH265::PacketizeFu(size_t fragment_index) {
   // Fragment payload into packets (FU).
   // Strip out the original header and leave room for the FU header.
-  ArrayView<const uint8_t> fragment = input_fragments_[fragment_index];
+  std::span<const uint8_t> fragment = input_fragments_[fragment_index];
   PayloadSizeLimits limits = limits_;
   // Refer to section 4.4.3 in RFC7798, each FU fragment will have a 2-bytes
   // payload header and a one-byte FU header. DONL is not supported so ignore
@@ -144,7 +144,7 @@ int RtpPacketizerH265::PacketizeAp(size_t fragment_index) {
   size_t payload_size_left = limits_.max_payload_len;
   int aggregated_fragments = 0;
   size_t fragment_headers_length = 0;
-  ArrayView<const uint8_t> fragment = input_fragments_[fragment_index];
+  std::span<const uint8_t> fragment = input_fragments_[fragment_index];
   RTC_CHECK_GE(payload_size_left, fragment.size());
   ++num_packets_left_;
 
@@ -247,7 +247,7 @@ void RtpPacketizerH265::NextAggregatePacket(RtpPacketToSend* rtp_packet) {
   uint8_t temporal_id_min = kH265MaxTemporalId;
   while (packet->aggregated) {
     // Add NAL unit length field.
-    ArrayView<const uint8_t> fragment = packet->source_fragment;
+    std::span<const uint8_t> fragment = packet->source_fragment;
     uint8_t layer_id = ((fragment[0] & kH265LayerIDHMask) << 5) |
                        ((fragment[1] & kH265LayerIDLMask) >> 3);
     layer_id_min = std::min(layer_id_min, layer_id);
@@ -310,7 +310,7 @@ void RtpPacketizerH265::NextFragmentPacket(RtpPacketToSend* rtp_packet) {
   // Now update payload_hdr_h with FU type.
   payload_hdr_h = (payload_hdr_h & kH265TypeMaskN) |
                   (H265::NaluType::kFu << 1) | layer_id_h;
-  ArrayView<const uint8_t> fragment = packet->source_fragment;
+  std::span<const uint8_t> fragment = packet->source_fragment;
   uint8_t* buffer = rtp_packet->AllocatePayload(
       kH265FuHeaderSizeBytes + kH265PayloadHeaderSizeBytes + fragment.size());
   RTC_CHECK(buffer);
diff --git a/modules/rtp_rtcp/source/rtp_packetizer_h265.h b/modules/rtp_rtcp/source/rtp_packetizer_h265.h
index 9c57fd887d..d3fca2e584 100644
--- a/modules/rtp_rtcp/source/rtp_packetizer_h265.h
+++ b/modules/rtp_rtcp/source/rtp_packetizer_h265.h
@@ -15,8 +15,8 @@
 #include <cstdint>
 #include <deque>
 #include <queue>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/rtp_rtcp/source/rtp_format.h"
 #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
 
@@ -27,7 +27,7 @@ class RtpPacketizerH265 : public RtpPacketizer {
   // Initialize with payload from encoder.
   // The payload_data must be exactly one encoded H.265 frame.
   // For H265 we only support tx-mode SRST.
-  RtpPacketizerH265(ArrayView<const uint8_t> payload, PayloadSizeLimits limits);
+  RtpPacketizerH265(std::span<const uint8_t> payload, PayloadSizeLimits limits);
 
   RtpPacketizerH265(const RtpPacketizerH265&) = delete;
   RtpPacketizerH265& operator=(const RtpPacketizerH265&) = delete;
@@ -43,13 +43,13 @@ class RtpPacketizerH265 : public RtpPacketizer {
 
  private:
   struct PacketUnit {
-    ArrayView<const uint8_t> source_fragment;
+    std::span<const uint8_t> source_fragment;
     bool first_fragment = false;
     bool last_fragment = false;
     bool aggregated = false;
     uint16_t header = 0;
   };
-  std::deque<ArrayView<const uint8_t>> input_fragments_;
+  std::deque<std::span<const uint8_t>> input_fragments_;
   std::queue<PacketUnit> packets_;
 
   bool GeneratePackets();
diff --git a/modules/rtp_rtcp/source/rtp_packetizer_h265_unittest.cc b/modules/rtp_rtcp/source/rtp_packetizer_h265_unittest.cc
index 5460166896..73985c3752 100644
--- a/modules/rtp_rtcp/source/rtp_packetizer_h265_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_packetizer_h265_unittest.cc
@@ -14,10 +14,10 @@
 #include <cstdint>
 #include <cstring>
 #include <initializer_list>
+#include <span>
 #include <vector>
 
 #include "absl/algorithm/container.h"
-#include "api/array_view.h"
 #include "common_video/h265/h265_common.h"
 #include "modules/rtp_rtcp/source/rtp_format.h"
 #include "modules/rtp_rtcp/source/rtp_packet_h265_common.h"
@@ -55,7 +55,7 @@ struct NalUnitHeader {
 Buffer GenerateNalUnit(NalUnitHeader header, size_t size) {
   RTC_CHECK_GT(size, 0);
   Buffer buffer = Buffer::CreateWithCapacity(size);
-  buffer.AppendData(size, [&](ArrayView<uint8_t> buffer_view) {
+  buffer.AppendData(size, [&](std::span<uint8_t> buffer_view) {
     buffer_view[0] = (header.nal_unit_type << 1) | (header.nuh_layer_id >> 5);
     buffer_view[1] = (header.nuh_layer_id << 3) | header.nuh_temporal_id_plus1;
     for (size_t i = 2; i < size; ++i) {
@@ -75,7 +75,7 @@ Buffer CreateFrame(std::initializer_list<size_t> nalu_sizes) {
   size_t size = absl::c_accumulate(nalu_sizes, size_t{0}) +
                 kStartCodeSize * nalu_sizes.size();
   Buffer frame = Buffer::CreateWithCapacity(size);
-  frame.AppendData(size, [&](ArrayView<uint8_t> frame_view) {
+  frame.AppendData(size, [&](std::span<uint8_t> frame_view) {
     size_t offset = 0;
     for (size_t nalu_size : nalu_sizes) {
       EXPECT_GE(nalu_size, 1u);
@@ -97,14 +97,14 @@ Buffer CreateFrame(std::initializer_list<size_t> nalu_sizes) {
 }
 
 // Create frame consisting of given nalus.
-Buffer CreateFrame(ArrayView<const Buffer> nalus) {
+Buffer CreateFrame(std::span<const Buffer> nalus) {
   static constexpr int kStartCodeSize = 3;
   int frame_size = 0;
   for (const Buffer& nalu : nalus) {
     frame_size += (kStartCodeSize + nalu.size());
   }
   Buffer frame = Buffer::CreateWithCapacity(frame_size);
-  frame.AppendData(frame_size, [&](ArrayView<uint8_t> frame_view) {
+  frame.AppendData(frame_size, [&](std::span<uint8_t> frame_view) {
     size_t offset = 0;
     for (const Buffer& nalu : nalus) {
       // Insert nalu start code
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
index ad76e51269..c10b86620c 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
@@ -14,12 +14,12 @@
 #include <cstring>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/rtp_headers.h"
 #include "api/sequence_checker.h"
@@ -172,7 +172,7 @@ std::optional<uint32_t> ModuleRtpRtcpImpl2::FlexfecSsrc() const {
 }
 
 void ModuleRtpRtcpImpl2::IncomingRtcpPacket(
-    ArrayView<const uint8_t> rtcp_packet) {
+    std::span<const uint8_t> rtcp_packet) {
   RTC_DCHECK_RUN_ON(&rtcp_thread_checker_);
   rtcp_receiver_.IncomingPacket(rtcp_packet);
 }
@@ -422,14 +422,14 @@ ModuleRtpRtcpImpl2::FetchFecPackets() {
 }
 
 void ModuleRtpRtcpImpl2::OnAbortedRetransmissions(
-    ArrayView<const uint16_t> sequence_numbers) {
+    std::span<const uint16_t> sequence_numbers) {
   RTC_DCHECK(rtp_sender_);
   RTC_DCHECK_RUN_ON(&rtp_sender_->sequencing_checker);
   rtp_sender_->packet_sender.OnAbortedRetransmissions(sequence_numbers);
 }
 
 void ModuleRtpRtcpImpl2::OnPacketsAcknowledged(
-    ArrayView<const uint16_t> sequence_numbers) {
+    std::span<const uint16_t> sequence_numbers) {
   RTC_DCHECK(rtp_sender_);
   rtp_sender_->packet_history.CullAcknowledgedPackets(sequence_numbers);
 }
@@ -456,7 +456,7 @@ ModuleRtpRtcpImpl2::GeneratePadding(size_t target_size_bytes) {
 
 std::vector<RtpSequenceNumberMap::Info>
 ModuleRtpRtcpImpl2::GetSentRtpPacketInfos(
-    ArrayView<const uint16_t> sequence_numbers) const {
+    std::span<const uint16_t> sequence_numbers) const {
   RTC_DCHECK(rtp_sender_);
   return rtp_sender_->packet_sender.GetSentRtpPacketInfos(sequence_numbers);
 }
@@ -625,9 +625,8 @@ int32_t ModuleRtpRtcpImpl2::SendNACK(const uint16_t* nack_list,
   }
   nack_last_seq_number_sent_ = nack_list[start_id + nack_length - 1];
 
-  return rtcp_sender_.SendRTCP(
-      GetFeedbackState(), kRtcpNack,
-      MakeArrayView(&nack_list[start_id], nack_length));
+  return rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpNack,
+                               std::span(&nack_list[start_id], nack_length));
 }
 
 void ModuleRtpRtcpImpl2::SendNack(
@@ -723,7 +722,7 @@ void ModuleRtpRtcpImpl2::OnReceivedNack(
 }
 
 void ModuleRtpRtcpImpl2::OnReceivedRtcpReportBlocks(
-    ArrayView<const ReportBlockData> report_blocks) {
+    std::span<const ReportBlockData> report_blocks) {
   if (rtp_sender_) {
     uint32_t ssrc = SSRC();
     std::optional<uint32_t> rtx_ssrc;
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl2.h b/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
index 8096112e84..3c74f9489c 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
@@ -16,13 +16,13 @@
 
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
 #include "absl/functional/any_invocable.h"
 #include "absl/memory/memory.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/rtp_headers.h"
 #include "api/sequence_checker.h"
@@ -97,7 +97,7 @@ class ModuleRtpRtcpImpl2 final : public RtpRtcpInterface,
   // Receiver part.
 
   // Called when we receive an RTCP packet.
-  void IncomingRtcpPacket(ArrayView<const uint8_t> incoming_packet) override;
+  void IncomingRtcpPacket(std::span<const uint8_t> incoming_packet) override;
 
   void SetRemoteSSRC(uint32_t ssrc) override;
 
@@ -191,16 +191,16 @@ class ModuleRtpRtcpImpl2 final : public RtpRtcpInterface,
   std::vector<std::unique_ptr<RtpPacketToSend>> FetchFecPackets() override;
 
   void OnAbortedRetransmissions(
-      ArrayView<const uint16_t> sequence_numbers) override;
+      std::span<const uint16_t> sequence_numbers) override;
 
   void OnPacketsAcknowledged(
-      ArrayView<const uint16_t> sequence_numbers) override;
+      std::span<const uint16_t> sequence_numbers) override;
 
   std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
       size_t target_size_bytes) override;
 
   std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos(
-      ArrayView<const uint16_t> sequence_numbers) const override;
+      std::span<const uint16_t> sequence_numbers) const override;
 
   size_t ExpectedPerPacketOverhead() const override;
 
@@ -275,7 +275,7 @@ class ModuleRtpRtcpImpl2 final : public RtpRtcpInterface,
   void OnReceivedNack(
       const std::vector<uint16_t>& nack_sequence_numbers) override;
   void OnReceivedRtcpReportBlocks(
-      ArrayView<const ReportBlockData> report_blocks) override;
+      std::span<const ReportBlockData> report_blocks) override;
   void OnRequestSendReport() override;
 
   void SetVideoBitrateAllocation(
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl2_unittest.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl2_unittest.cc
index 2b7197a27a..e29f98d21c 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl2_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl2_unittest.cc
@@ -15,11 +15,11 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/call/transport.h"
 #include "api/environment/environment.h"
 #include "api/environment/environment_factory.h"
@@ -122,13 +122,13 @@ class SendTransport : public Transport {
 
   void SetRtpRtcpModule(ModuleRtpRtcpImpl2* receiver) { receiver_ = receiver; }
   void SimulateNetworkDelay(TimeDelta delay) { delay_ = delay; }
-  bool SendRtp(ArrayView<const uint8_t> data,
+  bool SendRtp(std::span<const uint8_t> data,
                const PacketOptions& /* options */) override {
     EXPECT_TRUE(last_packet_.Parse(data));
     ++rtp_packets_sent_;
     return true;
   }
-  bool SendRtcp(ArrayView<const uint8_t> data,
+  bool SendRtcp(std::span<const uint8_t> data,
                 const PacketOptions& /* options */) override {
     test::RtcpPacketParser parser;
     parser.Parse(data);
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_interface.h b/modules/rtp_rtcp/source/rtp_rtcp_interface.h
index 5d5b44274e..630fa94163 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_interface.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_interface.h
@@ -15,11 +15,11 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/frame_transformer_interface.h"
 #include "api/rtp_headers.h"
 #include "api/rtp_packet_sender.h"
@@ -180,7 +180,7 @@ class RtpRtcpInterface : public RtcpFeedbackSenderInterface {
   // Receiver functions
   // **************************************************************************
 
-  virtual void IncomingRtcpPacket(ArrayView<const uint8_t> incoming_packet) = 0;
+  virtual void IncomingRtcpPacket(std::span<const uint8_t> incoming_packet) = 0;
 
   virtual void SetRemoteSSRC(uint32_t ssrc) = 0;
 
@@ -340,16 +340,16 @@ class RtpRtcpInterface : public RtcpFeedbackSenderInterface {
   virtual std::vector<std::unique_ptr<RtpPacketToSend>> FetchFecPackets() = 0;
 
   virtual void OnAbortedRetransmissions(
-      ArrayView<const uint16_t> sequence_numbers) = 0;
+      std::span<const uint16_t> sequence_numbers) = 0;
 
   virtual void OnPacketsAcknowledged(
-      ArrayView<const uint16_t> sequence_numbers) = 0;
+      std::span<const uint16_t> sequence_numbers) = 0;
 
   virtual std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
       size_t target_size_bytes) = 0;
 
   virtual std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos(
-      ArrayView<const uint16_t> sequence_numbers) const = 0;
+      std::span<const uint16_t> sequence_numbers) const = 0;
 
   // Returns an expected per packet overhead representing the main RTP header,
   // any CSRCs, and the registered header extensions that are expected on all
diff --git a/modules/rtp_rtcp/source/rtp_sender.cc b/modules/rtp_rtcp/source/rtp_sender.cc
index 670b97d47e..0e78f982a9 100644
--- a/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/modules/rtp_rtcp/source/rtp_sender.cc
@@ -16,12 +16,12 @@
 #include <cstring>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/rtp_headers.h"
 #include "api/rtp_packet_sender.h"
@@ -198,15 +198,15 @@ RTPSender::~RTPSender() {
   // to understand performance attributes and possibly remove locks.
 }
 
-ArrayView<const RtpExtensionSize> RTPSender::FecExtensionSizes() {
+std::span<const RtpExtensionSize> RTPSender::FecExtensionSizes() {
   return kFecOrPaddingExtensionSizes;
 }
 
-ArrayView<const RtpExtensionSize> RTPSender::VideoExtensionSizes() {
+std::span<const RtpExtensionSize> RTPSender::VideoExtensionSizes() {
   return kVideoExtensionSizes;
 }
 
-ArrayView<const RtpExtensionSize> RTPSender::AudioExtensionSizes() {
+std::span<const RtpExtensionSize> RTPSender::AudioExtensionSizes() {
   return kAudioExtensionSizes;
 }
 
@@ -513,7 +513,7 @@ size_t RTPSender::ExpectedPerPacketOverhead() const {
 }
 
 std::unique_ptr<RtpPacketToSend> RTPSender::AllocatePacket(
-    ArrayView<const uint32_t> csrcs) {
+    std::span<const uint32_t> csrcs) {
   MutexLock lock(&send_mutex_);
   RTC_DCHECK_LE(csrcs.size(), kRtpCsrcSize);
   if (csrcs.size() > max_num_csrcs_) {
@@ -649,9 +649,9 @@ static void CopyHeaderAndExtensionsToRtxPacket(const RtpPacketToSend& packet,
       continue;
     }
 
-    ArrayView<const uint8_t> source = packet.FindExtension(extension);
+    std::span<const uint8_t> source = packet.FindExtension(extension);
 
-    ArrayView<uint8_t> destination =
+    std::span<uint8_t> destination =
         rtx_packet->AllocateExtension(extension, source.size());
 
     // Could happen if any:
diff --git a/modules/rtp_rtcp/source/rtp_sender.h b/modules/rtp_rtcp/source/rtp_sender.h
index 5535a4061c..c649036090 100644
--- a/modules/rtp_rtcp/source/rtp_sender.h
+++ b/modules/rtp_rtcp/source/rtp_sender.h
@@ -16,11 +16,11 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/rtp_packet_sender.h"
 #include "modules/rtp_rtcp/include/flexfec_sender.h"
@@ -114,21 +114,21 @@ class RTPSender {
       RTC_LOCKS_EXCLUDED(send_mutex_);
 
   // Size info for header extensions used by FEC packets.
-  static ArrayView<const RtpExtensionSize> FecExtensionSizes()
+  static std::span<const RtpExtensionSize> FecExtensionSizes()
       RTC_LOCKS_EXCLUDED(send_mutex_);
 
   // Size info for header extensions used by video packets.
-  static ArrayView<const RtpExtensionSize> VideoExtensionSizes()
+  static std::span<const RtpExtensionSize> VideoExtensionSizes()
       RTC_LOCKS_EXCLUDED(send_mutex_);
 
   // Size info for header extensions used by audio packets.
-  static ArrayView<const RtpExtensionSize> AudioExtensionSizes()
+  static std::span<const RtpExtensionSize> AudioExtensionSizes()
       RTC_LOCKS_EXCLUDED(send_mutex_);
 
   // Create empty packet, fills ssrc, csrcs and reserve place for header
   // extensions RtpSender updates before sending.
   std::unique_ptr<RtpPacketToSend> AllocatePacket(
-      ArrayView<const uint32_t> csrcs = {}) RTC_LOCKS_EXCLUDED(send_mutex_);
+      std::span<const uint32_t> csrcs = {}) RTC_LOCKS_EXCLUDED(send_mutex_);
 
   // Maximum header overhead per fec/padding packet.
   size_t FecOrPaddingPacketMaxRtpHeaderLength() const
diff --git a/modules/rtp_rtcp/source/rtp_sender_audio.h b/modules/rtp_rtcp/source/rtp_sender_audio.h
index 6a87d358cb..b206ff2f5a 100644
--- a/modules/rtp_rtcp/source/rtp_sender_audio.h
+++ b/modules/rtp_rtcp/source/rtp_sender_audio.h
@@ -15,9 +15,9 @@
 #include <stdint.h>
 
 #include <optional>
+#include <span>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/units/timestamp.h"
 #include "modules/audio_coding/include/audio_coding_module_typedefs.h"
 #include "modules/rtp_rtcp/source/absolute_capture_time_sender.h"
@@ -48,7 +48,7 @@ class RTPSenderAudio {
 
   struct RtpAudioFrame {
     AudioFrameType type = AudioFrameType::kAudioFrameSpeech;
-    ArrayView<const uint8_t> payload;
+    std::span<const uint8_t> payload;
 
     // Payload id to write to the payload type field of the rtp packet.
     int payload_id = -1;
@@ -65,7 +65,7 @@ class RTPSenderAudio {
     std::optional<int> audio_level_dbov;
 
     // Contributing sources list.
-    ArrayView<const uint32_t> csrcs;
+    std::span<const uint32_t> csrcs;
   };
   bool SendAudio(const RtpAudioFrame& frame);
 
diff --git a/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
index 94d4d26cd9..6962e0fab4 100644
--- a/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
@@ -12,9 +12,9 @@
 
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/call/transport.h"
 #include "api/environment/environment.h"
 #include "api/environment/environment_factory.h"
@@ -53,13 +53,13 @@ class LoopbackTransportTest : public Transport {
         kAbsoluteCaptureTimeExtensionId);
   }
 
-  bool SendRtp(ArrayView<const uint8_t> data,
+  bool SendRtp(std::span<const uint8_t> data,
                const PacketOptions& /*options*/) override {
     sent_packets_.push_back(RtpPacketReceived(&receivers_extensions_));
     EXPECT_TRUE(sent_packets_.back().Parse(data));
     return true;
   }
-  bool SendRtcp(ArrayView<const uint8_t> /* data */,
+  bool SendRtcp(std::span<const uint8_t> /* data */,
                 const PacketOptions& /* options */) override {
     return false;
   }
diff --git a/modules/rtp_rtcp/source/rtp_sender_egress.cc b/modules/rtp_rtcp/source/rtp_sender_egress.cc
index c0a0ee7703..d4d39c2bdd 100644
--- a/modules/rtp_rtcp/source/rtp_sender_egress.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_egress.cc
@@ -15,10 +15,10 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/call/transport.h"
 #include "api/environment/environment.h"
 #include "api/rtc_event_log/rtc_event_log.h"
@@ -370,7 +370,7 @@ void RtpSenderEgress::SetTimestampOffset(uint32_t timestamp) {
 }
 
 std::vector<RtpSequenceNumberMap::Info> RtpSenderEgress::GetSentRtpPacketInfos(
-    ArrayView<const uint16_t> sequence_numbers) const {
+    std::span<const uint16_t> sequence_numbers) const {
   RTC_DCHECK_RUN_ON(worker_queue_);
   RTC_DCHECK(!sequence_numbers.empty());
   if (!need_rtp_packet_infos_) {
@@ -410,7 +410,7 @@ RtpSenderEgress::FetchFecPackets() {
 }
 
 void RtpSenderEgress::OnAbortedRetransmissions(
-    ArrayView<const uint16_t> sequence_numbers) {
+    std::span<const uint16_t> sequence_numbers) {
   RTC_DCHECK_RUN_ON(worker_queue_);
   // Mark aborted retransmissions as sent, rather than leaving them in
   // a 'pending' state - otherwise they can not be requested again and
diff --git a/modules/rtp_rtcp/source/rtp_sender_egress.h b/modules/rtp_rtcp/source/rtp_sender_egress.h
index df07a6d346..35779a4bbd 100644
--- a/modules/rtp_rtcp/source/rtp_sender_egress.h
+++ b/modules/rtp_rtcp/source/rtp_sender_egress.h
@@ -15,10 +15,10 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/call/transport.h"
 #include "api/environment/environment.h"
 #include "api/rtp_packet_sender.h"
@@ -93,14 +93,14 @@ class RtpSenderEgress {
   // recalled, return a vector with all of them (in corresponding order).
   // If any could not be recalled, return an empty vector.
   std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos(
-      ArrayView<const uint16_t> sequence_numbers) const;
+      std::span<const uint16_t> sequence_numbers) const;
 
   void SetFecProtectionParameters(const FecProtectionParams& delta_params,
                                   const FecProtectionParams& key_params);
   std::vector<std::unique_ptr<RtpPacketToSend>> FetchFecPackets();
 
   // Clears pending status for these sequence numbers in the packet history.
-  void OnAbortedRetransmissions(ArrayView<const uint16_t> sequence_numbers);
+  void OnAbortedRetransmissions(std::span<const uint16_t> sequence_numbers);
 
  private:
   struct Packet {
diff --git a/modules/rtp_rtcp/source/rtp_sender_egress_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_egress_unittest.cc
index 1f073070d4..1c58e13e9f 100644
--- a/modules/rtp_rtcp/source/rtp_sender_egress_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_egress_unittest.cc
@@ -15,9 +15,9 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 
-#include "api/array_view.h"
 #include "api/call/transport.h"
 #include "api/environment/environment.h"
 #include "api/environment/environment_factory.h"
@@ -92,7 +92,7 @@ class MockStreamDataCountersCallback : public StreamDataCountersCallback {
 };
 
 struct TransmittedPacket {
-  TransmittedPacket(ArrayView<const uint8_t> data,
+  TransmittedPacket(std::span<const uint8_t> data,
                     const PacketOptions& packet_options,
                     RtpHeaderExtensionMap* extensions)
       : packet(extensions), options(packet_options) {
@@ -107,7 +107,7 @@ class TestTransport : public Transport {
   explicit TestTransport(RtpHeaderExtensionMap* extensions)
       : total_data_sent_(DataSize::Zero()), extensions_(extensions) {}
   MOCK_METHOD(void, SentRtp, (const PacketOptions& options), ());
-  bool SendRtp(ArrayView<const uint8_t> packet,
+  bool SendRtp(std::span<const uint8_t> packet,
                const PacketOptions& options) override {
     total_data_sent_ += DataSize::Bytes(packet.size());
     last_packet_.emplace(packet, options, extensions_);
@@ -115,7 +115,7 @@ class TestTransport : public Transport {
     return true;
   }
 
-  bool SendRtcp(ArrayView<const uint8_t> /* packet */,
+  bool SendRtcp(std::span<const uint8_t> /* packet */,
                 const PacketOptions& /* options */) override {
     RTC_CHECK_NOTREACHED();
   }
@@ -852,7 +852,7 @@ TEST_F(RtpSenderEgressTest,
 TEST_F(RtpSenderEgressTest, SendPacketUpdatesStats) {
   const size_t kPayloadSize = 1000;
 
-  const ArrayView<const RtpExtensionSize> kNoRtpHeaderExtensionSizes;
+  const std::span<const RtpExtensionSize> kNoRtpHeaderExtensionSizes;
   FlexfecSender flexfec(env_, kFlexfectPayloadType, kFlexFecSsrc, kSsrc,
                         /*mid=*/"",
                         /*rtp_header_extensions=*/{},
diff --git a/modules/rtp_rtcp/source/rtp_sender_video.cc b/modules/rtp_rtcp/source/rtp_sender_video.cc
index 63d934b527..d5d2fa0f73 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -15,13 +15,13 @@
 #include <cstring>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <variant>
 #include <vector>
 
 #include "absl/algorithm/container.h"
 #include "absl/memory/memory.h"
-#include "api/array_view.h"
 #include "api/crypto/frame_encryptor_interface.h"
 #include "api/field_trials_view.h"
 #include "api/make_ref_counted.h"
@@ -522,7 +522,7 @@ bool RTPSenderVideo::SendVideo(int payload_type,
                                VideoCodecType codec_type,
                                uint32_t rtp_timestamp,
                                Timestamp capture_time,
-                               ArrayView<const uint8_t> payload,
+                               std::span<const uint8_t> payload,
                                size_t encoder_output_size,
                                RTPVideoHeader video_header,
                                TimeDelta expected_retransmission_time,
diff --git a/modules/rtp_rtcp/source/rtp_sender_video.h b/modules/rtp_rtcp/source/rtp_sender_video.h
index b3c33d6317..396591a402 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video.h
+++ b/modules/rtp_rtcp/source/rtp_sender_video.h
@@ -16,9 +16,9 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/field_trials_view.h"
 #include "api/frame_transformer_interface.h"
 #include "api/scoped_refptr.h"
@@ -106,7 +106,7 @@ class RTPSenderVideo : public RTPVideoFrameSenderInterface {
                  VideoCodecType codec_type,
                  uint32_t rtp_timestamp,
                  Timestamp capture_time,
-                 ArrayView<const uint8_t> payload,
+                 std::span<const uint8_t> payload,
                  size_t encoder_output_size,
                  RTPVideoHeader video_header,
                  TimeDelta expected_retransmission_time,
diff --git a/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc b/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc
index 33ea5f253e..c54f9f8991 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc
@@ -14,11 +14,11 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/frame_transformer_interface.h"
 #include "api/scoped_refptr.h"
 #include "api/sequence_checker.h"
@@ -78,9 +78,9 @@ class TransformableVideoSenderFrame : public TransformableVideoFrameInterface {
   ~TransformableVideoSenderFrame() override = default;
 
   // Implements TransformableVideoFrameInterface.
-  ArrayView<const uint8_t> GetData() const override { return *encoded_data_; }
+  std::span<const uint8_t> GetData() const override { return *encoded_data_; }
 
-  void SetData(ArrayView<const uint8_t> data) override {
+  void SetData(std::span<const uint8_t> data) override {
     encoded_data_ = EncodedImageBuffer::Create(data.data(), data.size());
   }
 
diff --git a/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.h b/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.h
index 4defcb8bfb..ad799a0408 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.h
+++ b/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.h
@@ -14,10 +14,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/frame_transformer_interface.h"
 #include "api/scoped_refptr.h"
 #include "api/sequence_checker.h"
@@ -43,7 +43,7 @@ class RTPVideoFrameSenderInterface {
                          VideoCodecType codec_type,
                          uint32_t rtp_timestamp,
                          Timestamp capture_time,
-                         ArrayView<const uint8_t> payload,
+                         std::span<const uint8_t> payload,
                          size_t encoder_output_size,
                          RTPVideoHeader video_header,
                          TimeDelta expected_retransmission_time,
diff --git a/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate_unittest.cc
index eac237eaac..332b53b7d7 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate_unittest.cc
@@ -15,11 +15,11 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/frame_transformer_interface.h"
 #include "api/make_ref_counted.h"
 #include "api/scoped_refptr.h"
@@ -58,7 +58,7 @@ class MockRTPVideoFrameSenderInterface : public RTPVideoFrameSenderInterface {
                VideoCodecType codec_type,
                uint32_t rtp_timestamp,
                Timestamp capture_time,
-               ArrayView<const uint8_t> payload,
+               std::span<const uint8_t> payload,
                size_t encoder_output_size,
                RTPVideoHeader video_header,
                TimeDelta expected_retransmission_time,
diff --git a/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
index 743224fc02..fb8c98faf4 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
@@ -15,11 +15,11 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
 #include "absl/memory/memory.h"
-#include "api/array_view.h"
 #include "api/call/transport.h"
 #include "api/environment/environment.h"
 #include "api/environment/environment_factory.h"
@@ -141,13 +141,13 @@ class LoopbackTransportTest : public Transport {
         kCorruptionDetectionExtensionId);
   }
 
-  bool SendRtp(ArrayView<const uint8_t> data,
+  bool SendRtp(std::span<const uint8_t> data,
                const PacketOptions& /* options */) override {
     sent_packets_.push_back(RtpPacketReceived(&receivers_extensions_));
     EXPECT_TRUE(sent_packets_.back().Parse(data));
     return true;
   }
-  bool SendRtcp(ArrayView<const uint8_t> /* data */,
+  bool SendRtcp(std::span<const uint8_t> /* data */,
                 const PacketOptions& /* options */) override {
     return false;
   }
@@ -626,7 +626,7 @@ TEST_F(RtpSenderVideoTest,
   // when rtx packet barely fit.
   for (size_t frame_size = 800; frame_size < kMaxPacketSize; ++frame_size) {
     SCOPED_TRACE(frame_size);
-    ArrayView<const uint8_t> payload(kPayload, frame_size);
+    std::span<const uint8_t> payload(kPayload, frame_size);
 
     EXPECT_TRUE(rtp_sender_video_->SendVideo(
         kMediaPayloadId, /*codec_type=*/kVideoCodecVP8, /*rtp_timestamp=*/0,
@@ -929,7 +929,7 @@ TEST_F(RtpSenderVideoTest,
   ON_CALL(*encryptor, GetMaxCiphertextByteSize).WillByDefault(ReturnArg<1>());
   ON_CALL(*encryptor, Encrypt)
       .WillByDefault(WithArgs<3, 5>(
-          [](ArrayView<const uint8_t> frame, size_t* bytes_written) {
+          [](std::span<const uint8_t> frame, size_t* bytes_written) {
             *bytes_written = frame.size();
             return 0;
           }));
@@ -1504,7 +1504,7 @@ TEST_F(RtpSenderVideoTest, SendGenericVideo) {
       kPayloadTypeGeneric, kCodecType, 1234, fake_clock_.CurrentTime(),
       kPayload, sizeof(kPayload), video_header, TimeDelta::PlusInfinity(), {}));
 
-  ArrayView<const uint8_t> sent_payload =
+  std::span<const uint8_t> sent_payload =
       transport_.last_sent_packet().payload();
   uint8_t generic_header = sent_payload[0];
   EXPECT_TRUE(generic_header & RtpFormatVideoGeneric::kKeyFrameBit);
@@ -1543,7 +1543,7 @@ TEST_F(RtpSenderVideoRawPacketizationTest, SendRawVideo) {
       fake_clock_.CurrentTime(), kPayload, sizeof(kPayload), video_header,
       TimeDelta::PlusInfinity(), {}));
 
-  ArrayView<const uint8_t> sent_payload =
+  std::span<const uint8_t> sent_payload =
       transport_.last_sent_packet().payload();
   EXPECT_THAT(sent_payload, ElementsAreArray(kPayload));
 }
diff --git a/modules/rtp_rtcp/source/rtp_util.cc b/modules/rtp_rtcp/source/rtp_util.cc
index 4d802b6308..0c62a31b6b 100644
--- a/modules/rtp_rtcp/source/rtp_util.cc
+++ b/modules/rtp_rtcp/source/rtp_util.cc
@@ -12,8 +12,8 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/rtp_rtcp/source/byte_io.h"
 #include "rtc_base/checks.h"
 
@@ -24,7 +24,7 @@ constexpr uint8_t kRtpVersion = 2;
 constexpr size_t kMinRtpPacketLen = 12;
 constexpr size_t kMinRtcpPacketLen = 4;
 
-bool HasCorrectRtpVersion(ArrayView<const uint8_t> packet) {
+bool HasCorrectRtpVersion(std::span<const uint8_t> packet) {
   return packet[0] >> 6 == kRtpVersion;
 }
 
@@ -35,27 +35,27 @@ bool PayloadTypeIsReservedForRtcp(uint8_t payload_type) {
 
 }  // namespace
 
-bool IsRtpPacket(ArrayView<const uint8_t> packet) {
+bool IsRtpPacket(std::span<const uint8_t> packet) {
   return packet.size() >= kMinRtpPacketLen && HasCorrectRtpVersion(packet) &&
          !PayloadTypeIsReservedForRtcp(packet[1] & 0x7F);
 }
 
-bool IsRtcpPacket(ArrayView<const uint8_t> packet) {
+bool IsRtcpPacket(std::span<const uint8_t> packet) {
   return packet.size() >= kMinRtcpPacketLen && HasCorrectRtpVersion(packet) &&
          PayloadTypeIsReservedForRtcp(packet[1] & 0x7F);
 }
 
-int ParseRtpPayloadType(ArrayView<const uint8_t> rtp_packet) {
+int ParseRtpPayloadType(std::span<const uint8_t> rtp_packet) {
   RTC_DCHECK(IsRtpPacket(rtp_packet));
   return rtp_packet[1] & 0x7F;
 }
 
-uint16_t ParseRtpSequenceNumber(ArrayView<const uint8_t> rtp_packet) {
+uint16_t ParseRtpSequenceNumber(std::span<const uint8_t> rtp_packet) {
   RTC_DCHECK(IsRtpPacket(rtp_packet));
   return ByteReader<uint16_t>::ReadBigEndian(rtp_packet.data() + 2);
 }
 
-uint32_t ParseRtpSsrc(ArrayView<const uint8_t> rtp_packet) {
+uint32_t ParseRtpSsrc(std::span<const uint8_t> rtp_packet) {
   RTC_DCHECK(IsRtpPacket(rtp_packet));
   return ByteReader<uint32_t>::ReadBigEndian(rtp_packet.data() + 8);
 }
diff --git a/modules/rtp_rtcp/source/rtp_util.h b/modules/rtp_rtcp/source/rtp_util.h
index a183e0e3df..90de009add 100644
--- a/modules/rtp_rtcp/source/rtp_util.h
+++ b/modules/rtp_rtcp/source/rtp_util.h
@@ -12,19 +12,18 @@
 #define MODULES_RTP_RTCP_SOURCE_RTP_UTIL_H_
 
 #include <cstdint>
-
-#include "api/array_view.h"
+#include <span>
 
 namespace webrtc {
 
-bool IsRtcpPacket(ArrayView<const uint8_t> packet);
-bool IsRtpPacket(ArrayView<const uint8_t> packet);
+bool IsRtcpPacket(std::span<const uint8_t> packet);
+bool IsRtpPacket(std::span<const uint8_t> packet);
 
 // Returns base rtp header fields of the rtp packet.
 // Behaviour is undefined when `!IsRtpPacket(rtp_packet)`.
-int ParseRtpPayloadType(ArrayView<const uint8_t> rtp_packet);
-uint16_t ParseRtpSequenceNumber(ArrayView<const uint8_t> rtp_packet);
-uint32_t ParseRtpSsrc(ArrayView<const uint8_t> rtp_packet);
+int ParseRtpPayloadType(std::span<const uint8_t> rtp_packet);
+uint16_t ParseRtpSequenceNumber(std::span<const uint8_t> rtp_packet);
+uint32_t ParseRtpSsrc(std::span<const uint8_t> rtp_packet);
 
 }  // namespace webrtc
 
diff --git a/modules/rtp_rtcp/source/rtp_video_layers_allocation_extension.cc b/modules/rtp_rtcp/source/rtp_video_layers_allocation_extension.cc
index d50902023a..ad91e35e5c 100644
--- a/modules/rtp_rtcp/source/rtp_video_layers_allocation_extension.cc
+++ b/modules/rtp_rtcp/source/rtp_video_layers_allocation_extension.cc
@@ -12,10 +12,10 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 #include <tuple>
 
 #include "absl/algorithm/container.h"
-#include "api/array_view.h"
 #include "api/units/data_rate.h"
 #include "api/video/video_layers_allocation.h"
 #include "modules/rtp_rtcp/source/byte_io.h"
@@ -109,7 +109,7 @@ SpatialLayersBitmasks SpatialLayersBitmasksPerRtpStream(
 // for the description of the format.
 
 bool RtpVideoLayersAllocationExtension::Write(
-    ArrayView<uint8_t> data,
+    std::span<uint8_t> data,
     const VideoLayersAllocation& allocation) {
   RTC_DCHECK(AllocationIsValid(allocation));
   RTC_DCHECK_GE(data.size(), ValueSize(allocation));
@@ -178,7 +178,7 @@ bool RtpVideoLayersAllocationExtension::Write(
 }
 
 bool RtpVideoLayersAllocationExtension::Parse(
-    ArrayView<const uint8_t> data,
+    std::span<const uint8_t> data,
     VideoLayersAllocation* allocation) {
   if (data.empty() || allocation == nullptr) {
     return false;
diff --git a/modules/rtp_rtcp/source/rtp_video_layers_allocation_extension.h b/modules/rtp_rtcp/source/rtp_video_layers_allocation_extension.h
index 520a16371b..d74a7cd539 100644
--- a/modules/rtp_rtcp/source/rtp_video_layers_allocation_extension.h
+++ b/modules/rtp_rtcp/source/rtp_video_layers_allocation_extension.h
@@ -13,9 +13,9 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtp_parameters.h"
 #include "api/video/video_layers_allocation.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
@@ -30,10 +30,10 @@ class RtpVideoLayersAllocationExtension {
     return RtpExtension::kVideoLayersAllocationUri;
   }
 
-  static bool Parse(ArrayView<const uint8_t> data,
+  static bool Parse(std::span<const uint8_t> data,
                     VideoLayersAllocation* allocation);
   static size_t ValueSize(const VideoLayersAllocation& allocation);
-  static bool Write(ArrayView<uint8_t> data,
+  static bool Write(std::span<uint8_t> data,
                     const VideoLayersAllocation& allocation);
 };
 
diff --git a/modules/rtp_rtcp/source/rtp_video_layers_allocation_extension_unittest.cc b/modules/rtp_rtcp/source/rtp_video_layers_allocation_extension_unittest.cc
index c8463a3caf..d80cc38ebe 100644
--- a/modules/rtp_rtcp/source/rtp_video_layers_allocation_extension_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_video_layers_allocation_extension_unittest.cc
@@ -12,8 +12,8 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/units/data_rate.h"
 #include "api/video/video_layers_allocation.h"
 #include "rtc_base/buffer.h"
@@ -28,7 +28,7 @@ TEST(RtpVideoLayersAllocationExtension, WriteEmptyLayersAllocationReturnsTrue) {
       RtpVideoLayersAllocationExtension::ValueSize(written_allocation));
   buffer.AppendData(
       RtpVideoLayersAllocationExtension::ValueSize(written_allocation),
-      [&](ArrayView<uint8_t> buffer_view) {
+      [&](std::span<uint8_t> buffer_view) {
         EXPECT_TRUE(RtpVideoLayersAllocationExtension::Write(
             buffer_view, written_allocation));
         return buffer_view.size();
@@ -46,7 +46,7 @@ TEST(RtpVideoLayersAllocationExtension,
       RtpVideoLayersAllocationExtension::ValueSize(written_allocation));
   buffer.AppendData(
       RtpVideoLayersAllocationExtension::ValueSize(written_allocation),
-      [&](ArrayView<uint8_t> buffer_view) {
+      [&](std::span<uint8_t> buffer_view) {
         EXPECT_TRUE(RtpVideoLayersAllocationExtension::Write(
             buffer_view, written_allocation));
         return buffer_view.size();
@@ -88,7 +88,7 @@ TEST(RtpVideoLayersAllocationExtension,
       RtpVideoLayersAllocationExtension::ValueSize(written_allocation));
   buffer.AppendData(
       RtpVideoLayersAllocationExtension::ValueSize(written_allocation),
-      [&](ArrayView<uint8_t> buffer_view) {
+      [&](std::span<uint8_t> buffer_view) {
         EXPECT_TRUE(RtpVideoLayersAllocationExtension::Write(
             buffer_view, written_allocation));
         return buffer_view.size();
@@ -130,7 +130,7 @@ TEST(RtpVideoLayersAllocationExtension,
       RtpVideoLayersAllocationExtension::ValueSize(written_allocation));
   buffer.AppendData(
       RtpVideoLayersAllocationExtension::ValueSize(written_allocation),
-      [&](ArrayView<uint8_t> buffer_view) {
+      [&](std::span<uint8_t> buffer_view) {
         EXPECT_TRUE(RtpVideoLayersAllocationExtension::Write(
             buffer_view, written_allocation));
         return buffer_view.size();
@@ -165,7 +165,7 @@ TEST(RtpVideoLayersAllocationExtension,
       RtpVideoLayersAllocationExtension::ValueSize(written_allocation));
   buffer.AppendData(
       RtpVideoLayersAllocationExtension::ValueSize(written_allocation),
-      [&](ArrayView<uint8_t> buffer_view) {
+      [&](std::span<uint8_t> buffer_view) {
         EXPECT_TRUE(RtpVideoLayersAllocationExtension::Write(
             buffer_view, written_allocation));
         return buffer_view.size();
@@ -200,7 +200,7 @@ TEST(RtpVideoLayersAllocationExtension,
       RtpVideoLayersAllocationExtension::ValueSize(written_allocation));
   buffer.AppendData(
       RtpVideoLayersAllocationExtension::ValueSize(written_allocation),
-      [&](ArrayView<uint8_t> buffer_view) {
+      [&](std::span<uint8_t> buffer_view) {
         EXPECT_TRUE(RtpVideoLayersAllocationExtension::Write(
             buffer_view, written_allocation));
         return buffer_view.size();
@@ -240,7 +240,7 @@ TEST(RtpVideoLayersAllocationExtension,
       RtpVideoLayersAllocationExtension::ValueSize(written_allocation));
   buffer.AppendData(
       RtpVideoLayersAllocationExtension::ValueSize(written_allocation),
-      [&](ArrayView<uint8_t> buffer_view) {
+      [&](std::span<uint8_t> buffer_view) {
         EXPECT_TRUE(RtpVideoLayersAllocationExtension::Write(
             buffer_view, written_allocation));
         return buffer_view.size();
@@ -283,7 +283,7 @@ TEST(RtpVideoLayersAllocationExtension,
       RtpVideoLayersAllocationExtension::ValueSize(written_allocation));
   buffer.AppendData(
       RtpVideoLayersAllocationExtension::ValueSize(written_allocation),
-      [&](ArrayView<uint8_t> buffer_view) {
+      [&](std::span<uint8_t> buffer_view) {
         EXPECT_TRUE(RtpVideoLayersAllocationExtension::Write(
             buffer_view, written_allocation));
         return buffer_view.size();
@@ -302,7 +302,7 @@ TEST(RtpVideoLayersAllocationExtension,
       RtpVideoLayersAllocationExtension::ValueSize(written_allocation));
   buffer.AppendData(
       RtpVideoLayersAllocationExtension::ValueSize(written_allocation),
-      [&](ArrayView<uint8_t> buffer_view) {
+      [&](std::span<uint8_t> buffer_view) {
         EXPECT_TRUE(RtpVideoLayersAllocationExtension::Write(
             buffer_view, written_allocation));
         return buffer_view.size();
@@ -336,7 +336,7 @@ TEST(RtpVideoLayersAllocationExtension, DiscardsInvalidHeight) {
       RtpVideoLayersAllocationExtension::ValueSize(written_allocation));
   size_t written = buffer.AppendData(
       RtpVideoLayersAllocationExtension::ValueSize(written_allocation),
-      [&](ArrayView<uint8_t> buffer_view) {
+      [&](std::span<uint8_t> buffer_view) {
         bool result = RtpVideoLayersAllocationExtension::Write(
             buffer_view, written_allocation);
         EXPECT_TRUE(result);
diff --git a/modules/rtp_rtcp/source/rtp_video_stream_receiver_frame_transformer_delegate.cc b/modules/rtp_rtcp/source/rtp_video_stream_receiver_frame_transformer_delegate.cc
index f805d19ca3..9e7f09939e 100644
--- a/modules/rtp_rtcp/source/rtp_video_stream_receiver_frame_transformer_delegate.cc
+++ b/modules/rtp_rtcp/source/rtp_video_stream_receiver_frame_transformer_delegate.cc
@@ -13,12 +13,12 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/memory/memory.h"
-#include "api/array_view.h"
 #include "api/frame_transformer_interface.h"
 #include "api/rtp_packet_infos.h"
 #include "api/scoped_refptr.h"
@@ -54,11 +54,11 @@ class TransformableVideoReceiverFrame
   ~TransformableVideoReceiverFrame() override = default;
 
   // Implements TransformableVideoFrameInterface.
-  ArrayView<const uint8_t> GetData() const override {
+  std::span<const uint8_t> GetData() const override {
     return *frame_->GetEncodedData();
   }
 
-  void SetData(ArrayView<const uint8_t> data) override {
+  void SetData(std::span<const uint8_t> data) override {
     frame_->SetEncodedData(
         EncodedImageBuffer::Create(data.data(), data.size()));
   }
@@ -232,7 +232,7 @@ void RtpVideoStreamReceiverFrameTransformerDelegate::ManageFrame(
     VideoFrameMetadata metadata = transformed_frame->Metadata();
     RTPVideoHeader video_header = RTPVideoHeader::FromMetadata(metadata);
     VideoSendTiming timing;
-    ArrayView<const uint8_t> data = transformed_frame->GetData();
+    std::span<const uint8_t> data = transformed_frame->GetData();
     int64_t receive_time = clock_->CurrentTime().ms();
     receiver_->ManageFrame(std::make_unique<RtpFrameObject>(
         /*first_seq_num=*/metadata.GetFrameId().value_or(0),
diff --git a/modules/rtp_rtcp/source/rtp_video_stream_receiver_frame_transformer_delegate_unittest.cc b/modules/rtp_rtcp/source/rtp_video_stream_receiver_frame_transformer_delegate_unittest.cc
index 2e4c1b6e08..989beddcba 100644
--- a/modules/rtp_rtcp/source/rtp_video_stream_receiver_frame_transformer_delegate_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_video_stream_receiver_frame_transformer_delegate_unittest.cc
@@ -13,11 +13,11 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
 #include "absl/memory/memory.h"
-#include "api/array_view.h"
 #include "api/frame_transformer_interface.h"
 #include "api/make_ref_counted.h"
 #include "api/rtp_headers.h"
@@ -342,7 +342,7 @@ TEST(RtpVideoStreamReceiverFrameTransformerDelegateTest,
   scoped_refptr<EncodedImageBufferInterface> buffer =
       EncodedImageBuffer::Create(1);
   ON_CALL(*mock_sender_frame, GetData)
-      .WillByDefault(Return(ArrayView<const uint8_t>(*buffer)));
+      .WillByDefault(Return(std::span<const uint8_t>(*buffer)));
 
   scoped_refptr<TransformedFrameCallback> callback;
   EXPECT_CALL(*mock_frame_transformer, RegisterTransformedFrameSinkCallback)
diff --git a/modules/rtp_rtcp/source/ulpfec_header_reader_writer.cc b/modules/rtp_rtcp/source/ulpfec_header_reader_writer.cc
index 95f3a2a9f1..b4202f2b6c 100644
--- a/modules/rtp_rtcp/source/ulpfec_header_reader_writer.cc
+++ b/modules/rtp_rtcp/source/ulpfec_header_reader_writer.cc
@@ -12,8 +12,8 @@
 
 #include <cstdint>
 #include <cstring>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/scoped_refptr.h"
 #include "modules/rtp_rtcp/source/byte_io.h"
 #include "modules/rtp_rtcp/source/forward_error_correction.h"
@@ -111,7 +111,7 @@ size_t UlpfecHeaderWriter::FecHeaderSize(size_t packet_mask_size) const {
 }
 
 void UlpfecHeaderWriter::FinalizeFecHeader(
-    ArrayView<const ProtectedStream> protected_streams,
+    std::span<const ProtectedStream> protected_streams,
     ForwardErrorCorrection::Packet& fec_packet) const {
   RTC_CHECK_EQ(protected_streams.size(), 1);
   uint16_t seq_num_base = protected_streams[0].seq_num_base;
diff --git a/modules/rtp_rtcp/source/ulpfec_header_reader_writer.h b/modules/rtp_rtcp/source/ulpfec_header_reader_writer.h
index b08725eebf..6c196fa5eb 100644
--- a/modules/rtp_rtcp/source/ulpfec_header_reader_writer.h
+++ b/modules/rtp_rtcp/source/ulpfec_header_reader_writer.h
@@ -14,7 +14,8 @@
 #include <stddef.h>
 #include <stdint.h>
 
-#include "api/array_view.h"
+#include <span>
+
 #include "modules/rtp_rtcp/source/forward_error_correction.h"
 
 namespace webrtc {
@@ -58,7 +59,7 @@ class UlpfecHeaderWriter : public FecHeaderWriter {
   size_t FecHeaderSize(size_t packet_mask_row_size) const override;
 
   void FinalizeFecHeader(
-      ArrayView<const ProtectedStream> protected_streams,
+      std::span<const ProtectedStream> protected_streams,
       ForwardErrorCorrection::Packet& fec_packet) const override;
 };
 
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer.cc b/modules/rtp_rtcp/source/video_rtp_depacketizer.cc
index d44e38f03d..6afc427537 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer.cc
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer.cc
@@ -13,8 +13,8 @@
 #include <cstddef>
 #include <cstdint>
 #include <cstring>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/scoped_refptr.h"
 #include "api/video/encoded_image.h"
 #include "rtc_base/checks.h"
@@ -22,9 +22,9 @@
 namespace webrtc {
 
 scoped_refptr<EncodedImageBuffer> VideoRtpDepacketizer::AssembleFrame(
-    ArrayView<const ArrayView<const uint8_t>> rtp_payloads) {
+    std::span<const std::span<const uint8_t>> rtp_payloads) {
   size_t frame_size = 0;
-  for (ArrayView<const uint8_t> payload : rtp_payloads) {
+  for (std::span<const uint8_t> payload : rtp_payloads) {
     frame_size += payload.size();
   }
 
@@ -32,7 +32,7 @@ scoped_refptr<EncodedImageBuffer> VideoRtpDepacketizer::AssembleFrame(
       EncodedImageBuffer::Create(frame_size);
 
   uint8_t* write_at = bitstream->data();
-  for (ArrayView<const uint8_t> payload : rtp_payloads) {
+  for (std::span<const uint8_t> payload : rtp_payloads) {
     memcpy(write_at, payload.data(), payload.size());
     write_at += payload.size();
   }
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer.h b/modules/rtp_rtcp/source/video_rtp_depacketizer.h
index 487be38f62..2a051238ad 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer.h
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer.h
@@ -14,8 +14,8 @@
 #include <stdint.h>
 
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/scoped_refptr.h"
 #include "api/video/encoded_image.h"
 #include "modules/rtp_rtcp/source/rtp_video_header.h"
@@ -34,7 +34,7 @@ class VideoRtpDepacketizer {
   virtual std::optional<ParsedRtpPayload> Parse(
       CopyOnWriteBuffer rtp_payload) = 0;
   virtual scoped_refptr<EncodedImageBuffer> AssembleFrame(
-      ArrayView<const ArrayView<const uint8_t>> rtp_payloads);
+      std::span<const std::span<const uint8_t>> rtp_payloads);
 };
 
 }  // namespace webrtc
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.cc b/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.cc
index ed7434e88d..625c98e961 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.cc
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.cc
@@ -16,10 +16,10 @@
 #include <cstring>
 #include <iterator>
 #include <optional>
+#include <span>
 #include <utility>
 
 #include "absl/container/inlined_vector.h"
-#include "api/array_view.h"
 #include "api/scoped_refptr.h"
 #include "api/video/encoded_image.h"
 #include "api/video/video_codec_type.h"
@@ -93,7 +93,7 @@ class ArrayOfArrayViews {
   }
 
  private:
-  using Storage = absl::InlinedVector<ArrayView<const uint8_t>, 2>;
+  using Storage = absl::InlinedVector<std::span<const uint8_t>, 2>;
 
   size_t size_ = 0;
   Storage data_;
@@ -195,10 +195,10 @@ int RtpStartsNewCodedVideoSequence(uint8_t aggregation_header) {
 // fills ObuInfo::data field.
 // Returns empty vector on error.
 VectorObuInfo ParseObus(
-    ArrayView<const ArrayView<const uint8_t>> rtp_payloads) {
+    std::span<const std::span<const uint8_t>> rtp_payloads) {
   VectorObuInfo obu_infos;
   bool expect_continues_obu = false;
-  for (ArrayView<const uint8_t> rtp_payload : rtp_payloads) {
+  for (std::span<const uint8_t> rtp_payload : rtp_payloads) {
     ByteBufferReader payload(rtp_payload);
     uint8_t aggregation_header;
     if (!payload.ReadUInt8(&aggregation_header)) {
@@ -337,7 +337,7 @@ bool CalculateObuSizes(ObuInfo* obu_info) {
 }  // namespace
 
 scoped_refptr<EncodedImageBuffer> VideoRtpDepacketizerAv1::AssembleFrame(
-    ArrayView<const ArrayView<const uint8_t>> rtp_payloads) {
+    std::span<const std::span<const uint8_t>> rtp_payloads) {
   VectorObuInfo obu_infos = ParseObus(rtp_payloads);
   if (obu_infos.empty()) {
     return nullptr;
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.h b/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.h
index 4acf209d9c..446b819fef 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.h
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.h
@@ -15,8 +15,8 @@
 #include <stdint.h>
 
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/scoped_refptr.h"
 #include "api/video/encoded_image.h"
 #include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
@@ -32,7 +32,7 @@ class VideoRtpDepacketizerAv1 : public VideoRtpDepacketizer {
   ~VideoRtpDepacketizerAv1() override = default;
 
   scoped_refptr<EncodedImageBuffer> AssembleFrame(
-      ArrayView<const ArrayView<const uint8_t>> rtp_payloads) override;
+      std::span<const std::span<const uint8_t>> rtp_payloads) override;
 
   std::optional<ParsedRtpPayload> Parse(CopyOnWriteBuffer rtp_payload) override;
 };
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_av1_unittest.cc b/modules/rtp_rtcp/source/video_rtp_depacketizer_av1_unittest.cc
index aa3a735fca..17c7f48162 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_av1_unittest.cc
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_av1_unittest.cc
@@ -13,8 +13,8 @@
 #include <cstdint>
 #include <cstring>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/video/video_frame_type.h"
 #include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
 #include "rtc_base/copy_on_write_buffer.h"
@@ -125,10 +125,10 @@ TEST(VideoRtpDepacketizerAv1Test, AssembleFrameSetsOBUPayloadSizeWhenAbsent) {
   const uint8_t payload1[] = {0b00'01'0000,  // aggregation header
                               0b0'0110'000,  // /  Frame
                               20, 30, 40};   // \  OBU
-  ArrayView<const uint8_t> payloads[] = {payload1};
+  std::span<const uint8_t> payloads[] = {payload1};
   auto frame = VideoRtpDepacketizerAv1().AssembleFrame(payloads);
   ASSERT_TRUE(frame);
-  ArrayView<const uint8_t> frame_view(*frame);
+  std::span<const uint8_t> frame_view(*frame);
   EXPECT_TRUE(frame_view[0] & kObuHeaderHasSize);
   EXPECT_EQ(frame_view[1], 3);
 }
@@ -140,10 +140,10 @@ TEST(VideoRtpDepacketizerAv1Test, AssembleFrameSetsOBUPayloadSizeWhenPresent) {
                               20,
                               30,
                               40};  // \  obu_payload
-  ArrayView<const uint8_t> payloads[] = {payload1};
+  std::span<const uint8_t> payloads[] = {payload1};
   auto frame = VideoRtpDepacketizerAv1().AssembleFrame(payloads);
   ASSERT_TRUE(frame);
-  ArrayView<const uint8_t> frame_view(*frame);
+  std::span<const uint8_t> frame_view(*frame);
   EXPECT_TRUE(frame_view[0] & kObuHeaderHasSize);
   EXPECT_EQ(frame_view[1], 3);
 }
@@ -154,10 +154,10 @@ TEST(VideoRtpDepacketizerAv1Test,
                               0b0'0110'100,           // /  Frame
                               0b010'01'000,           // | extension_header
                               20,           30, 40};  // \  OBU
-  ArrayView<const uint8_t> payloads[] = {payload1};
+  std::span<const uint8_t> payloads[] = {payload1};
   auto frame = VideoRtpDepacketizerAv1().AssembleFrame(payloads);
   ASSERT_TRUE(frame);
-  ArrayView<const uint8_t> frame_view(*frame);
+  std::span<const uint8_t> frame_view(*frame);
   EXPECT_TRUE(frame_view[0] & kObuHeaderHasSize);
   EXPECT_EQ(frame_view[2], 3);
 }
@@ -171,10 +171,10 @@ TEST(VideoRtpDepacketizerAv1Test,
                               20,
                               30,
                               40};  // \  obu_payload
-  ArrayView<const uint8_t> payloads[] = {payload1};
+  std::span<const uint8_t> payloads[] = {payload1};
   auto frame = VideoRtpDepacketizerAv1().AssembleFrame(payloads);
   ASSERT_TRUE(frame);
-  ArrayView<const uint8_t> frame_view(*frame);
+  std::span<const uint8_t> frame_view(*frame);
   EXPECT_TRUE(frame_view[0] & kObuHeaderHasSize);
   EXPECT_EQ(frame_view[2], 3);
 }
@@ -183,10 +183,10 @@ TEST(VideoRtpDepacketizerAv1Test, AssembleFrameFromOnePacketWithOneObu) {
   const uint8_t payload1[] = {0b00'01'0000,  // aggregation header
                               0b0'0110'000,  // /  Frame
                               20};           // \  OBU
-  ArrayView<const uint8_t> payloads[] = {payload1};
+  std::span<const uint8_t> payloads[] = {payload1};
   auto frame = VideoRtpDepacketizerAv1().AssembleFrame(payloads);
   ASSERT_TRUE(frame);
-  EXPECT_THAT(ArrayView<const uint8_t>(*frame),
+  EXPECT_THAT(std::span<const uint8_t>(*frame),
               ElementsAre(0b0'0110'010, 1, 20));
 }
 
@@ -197,10 +197,10 @@ TEST(VideoRtpDepacketizerAv1Test, AssembleFrameFromOnePacketWithTwoObus) {
                               10,            // \  OBU
                               0b0'0110'000,  // /  Frame
                               20};           // \  OBU
-  ArrayView<const uint8_t> payloads[] = {payload1};
+  std::span<const uint8_t> payloads[] = {payload1};
   auto frame = VideoRtpDepacketizerAv1().AssembleFrame(payloads);
   ASSERT_TRUE(frame);
-  EXPECT_THAT(ArrayView<const uint8_t>(*frame),
+  EXPECT_THAT(std::span<const uint8_t>(*frame),
               ElementsAre(0b0'0001'010, 1, 10,    // Sequence Header OBU
                           0b0'0110'010, 1, 20));  // Frame OBU
 }
@@ -210,10 +210,10 @@ TEST(VideoRtpDepacketizerAv1Test, AssembleFrameFromTwoPacketsWithOneObu) {
                               0b0'0110'000, 20, 30};
   const uint8_t payload2[] = {0b10'01'0000,  // aggregation header
                               40};
-  ArrayView<const uint8_t> payloads[] = {payload1, payload2};
+  std::span<const uint8_t> payloads[] = {payload1, payload2};
   auto frame = VideoRtpDepacketizerAv1().AssembleFrame(payloads);
   ASSERT_TRUE(frame);
-  EXPECT_THAT(ArrayView<const uint8_t>(*frame),
+  EXPECT_THAT(std::span<const uint8_t>(*frame),
               ElementsAre(0b0'0110'010, 3, 20, 30, 40));
 }
 
@@ -227,10 +227,10 @@ TEST(VideoRtpDepacketizerAv1Test, AssembleFrameFromTwoPacketsWithTwoObu) {
                               30};           //
   const uint8_t payload2[] = {0b10'01'0000,  // aggregation header
                               40};           //
-  ArrayView<const uint8_t> payloads[] = {payload1, payload2};
+  std::span<const uint8_t> payloads[] = {payload1, payload2};
   auto frame = VideoRtpDepacketizerAv1().AssembleFrame(payloads);
   ASSERT_TRUE(frame);
-  EXPECT_THAT(ArrayView<const uint8_t>(*frame),
+  EXPECT_THAT(std::span<const uint8_t>(*frame),
               ElementsAre(0b0'0001'010, 1, 10,            // SH
                           0b0'0110'010, 3, 20, 30, 40));  // Frame
 }
@@ -258,10 +258,10 @@ TEST(VideoRtpDepacketizerAv1Test,
   const uint8_t payload2[] = {0b10'01'0000,  // aggregation header
                               70, 80, 90};   // \  tail of the frame OBU
 
-  ArrayView<const uint8_t> payloads[] = {payload1, payload2};
+  std::span<const uint8_t> payloads[] = {payload1, payload2};
   auto frame = VideoRtpDepacketizerAv1().AssembleFrame(payloads);
   ASSERT_TRUE(frame);
-  EXPECT_THAT(ArrayView<const uint8_t>(*frame),
+  EXPECT_THAT(std::span<const uint8_t>(*frame),
               ElementsAre(  // Sequence header OBU
                   0b0'0001'010, 1, 10,
                   // Metadata OBU without extension
@@ -282,11 +282,11 @@ TEST(VideoRtpDepacketizerAv1Test, AssembleFrameWithOneObuFromManyPackets) {
   const uint8_t payload4[] = {0b10'01'0000,  // aggregation header
                               18};
 
-  ArrayView<const uint8_t> payloads[] = {payload1, payload2, payload3,
+  std::span<const uint8_t> payloads[] = {payload1, payload2, payload3,
                                          payload4};
   auto frame = VideoRtpDepacketizerAv1().AssembleFrame(payloads);
   ASSERT_TRUE(frame);
-  EXPECT_THAT(ArrayView<const uint8_t>(*frame),
+  EXPECT_THAT(std::span<const uint8_t>(*frame),
               ElementsAre(0b0'0110'010, 8, 11, 12, 13, 14, 15, 16, 17, 18));
 }
 
@@ -314,11 +314,11 @@ TEST(VideoRtpDepacketizerAv1Test,
                               32};
   const uint8_t payload4[] = {0b10'01'0000,  // aggregation header
                               33, 34, 35, 36};
-  ArrayView<const uint8_t> payloads[] = {payload1, payload2, payload3,
+  std::span<const uint8_t> payloads[] = {payload1, payload2, payload3,
                                          payload4};
   auto frame = VideoRtpDepacketizerAv1().AssembleFrame(payloads);
   ASSERT_TRUE(frame);
-  EXPECT_THAT(ArrayView<const uint8_t>(*frame),
+  EXPECT_THAT(std::span<const uint8_t>(*frame),
               ElementsAre(0b0'0011'010, 2, 11, 12,  // Frame header
                           0b0'0100'010, 7, 21, 22, 23, 24, 25, 26, 27,  //
                           0b0'0111'010, 2, 11, 12,                      //
@@ -334,11 +334,11 @@ TEST(VideoRtpDepacketizerAv1Test,
   payload1[2] = 0x01;          // in two bytes
   payload1[3] = 0b0'0110'000;  // obu_header with size and extension bits unset.
   payload1[4 + 42] = 0x42;
-  ArrayView<const uint8_t> payloads[] = {payload1};
+  std::span<const uint8_t> payloads[] = {payload1};
   auto frame = VideoRtpDepacketizerAv1().AssembleFrame(payloads);
   ASSERT_TRUE(frame);
   EXPECT_EQ(frame->size(), 2 + 127u);
-  ArrayView<const uint8_t> frame_view(*frame);
+  std::span<const uint8_t> frame_view(*frame);
   EXPECT_EQ(frame_view[0], 0b0'0110'010);  // obu_header with size bit set.
   EXPECT_EQ(frame_view[1], 127);  // obu payload size, 1 byte enough to encode.
   // Check 'random' byte from the payload is at the same 'random' offset.
@@ -359,11 +359,11 @@ TEST(VideoRtpDepacketizerAv1Test,
   payload2[1] = 96;            // leb128 encoded size of 96 bytes in one byte
   payload2[2 + 20] = 0x20;
 
-  ArrayView<const uint8_t> payloads[] = {payload1, payload2};
+  std::span<const uint8_t> payloads[] = {payload1, payload2};
   auto frame = VideoRtpDepacketizerAv1().AssembleFrame(payloads);
   ASSERT_TRUE(frame);
   EXPECT_EQ(frame->size(), 3 + 128u);
-  ArrayView<const uint8_t> frame_view(*frame);
+  std::span<const uint8_t> frame_view(*frame);
   EXPECT_EQ(frame_view[0], 0b0'0110'010);  // obu_header with size bit set.
   EXPECT_EQ(frame_view[1], 0x80);          // obu payload size of 128 bytes.
   EXPECT_EQ(frame_view[2], 0x01);          // encoded in two byes
@@ -376,11 +376,11 @@ TEST(VideoRtpDepacketizerAv1Test,
      AssembleFrameFromAlmostEmptyPacketStartingAnOBU) {
   const uint8_t payload1[] = {0b01'01'0000};
   const uint8_t payload2[] = {0b10'01'0000, 0b0'0110'000, 10, 20, 30};
-  ArrayView<const uint8_t> payloads[] = {payload1, payload2};
+  std::span<const uint8_t> payloads[] = {payload1, payload2};
 
   auto frame = VideoRtpDepacketizerAv1().AssembleFrame(payloads);
   ASSERT_TRUE(frame);
-  EXPECT_THAT(ArrayView<const uint8_t>(*frame),
+  EXPECT_THAT(std::span<const uint8_t>(*frame),
               ElementsAre(0b0'0110'010, 3, 10, 20, 30));
 }
 
@@ -388,11 +388,11 @@ TEST(VideoRtpDepacketizerAv1Test,
      AssembleFrameFromAlmostEmptyPacketFinishingAnOBU) {
   const uint8_t payload1[] = {0b01'01'0000, 0b0'0110'000, 10, 20, 30};
   const uint8_t payload2[] = {0b10'01'0000};
-  ArrayView<const uint8_t> payloads[] = {payload1, payload2};
+  std::span<const uint8_t> payloads[] = {payload1, payload2};
 
   auto frame = VideoRtpDepacketizerAv1().AssembleFrame(payloads);
   ASSERT_TRUE(frame);
-  EXPECT_THAT(ArrayView<const uint8_t>(*frame),
+  EXPECT_THAT(std::span<const uint8_t>(*frame),
               ElementsAre(0b0'0110'010, 3, 10, 20, 30));
 }
 
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_h264.cc b/modules/rtp_rtcp/source/video_rtp_depacketizer_h264.cc
index a30429ddc3..4e5a005d2e 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_h264.cc
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_h264.cc
@@ -13,10 +13,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/video/video_codec_type.h"
 #include "api/video/video_frame_type.h"
 #include "common_video/h264/h264_common.h"
@@ -40,9 +40,9 @@ constexpr size_t kNalHeaderSize = 1;
 constexpr size_t kFuAHeaderSize = 2;
 constexpr size_t kLengthFieldSize = 2;
 
-std::vector<ArrayView<const uint8_t>> ParseStapA(
-    ArrayView<const uint8_t> data) {
-  std::vector<ArrayView<const uint8_t>> nal_units;
+std::vector<std::span<const uint8_t>> ParseStapA(
+    std::span<const uint8_t> data) {
+  std::vector<std::span<const uint8_t>> nal_units;
   ByteBufferReader reader(data);
   if (!reader.Consume(kNalHeaderSize)) {
     return nal_units;
@@ -64,7 +64,7 @@ std::vector<ArrayView<const uint8_t>> ParseStapA(
 
 std::optional<VideoRtpDepacketizer::ParsedRtpPayload> ProcessStapAOrSingleNalu(
     CopyOnWriteBuffer rtp_payload) {
-  ArrayView<const uint8_t> payload_data(rtp_payload);
+  std::span<const uint8_t> payload_data(rtp_payload);
   std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed_payload(
       std::in_place);
   bool modified_buffer = false;
@@ -79,7 +79,7 @@ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> ProcessStapAOrSingleNalu(
                           .emplace<RTPVideoHeaderH264>();
 
   uint8_t nal_type = payload_data[0] & kH264TypeMask;
-  std::vector<ArrayView<const uint8_t>> nal_units;
+  std::vector<std::span<const uint8_t>> nal_units;
   if (nal_type == H264::NaluType::kStapA) {
     nal_units = ParseStapA(payload_data);
     if (nal_units.empty()) {
@@ -96,12 +96,12 @@ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> ProcessStapAOrSingleNalu(
 
   parsed_payload->video_header.frame_type = VideoFrameType::kVideoFrameDelta;
 
-  for (const ArrayView<const uint8_t>& nal_unit : nal_units) {
+  for (const std::span<const uint8_t>& nal_unit : nal_units) {
     NaluInfo nalu;
     nalu.type = nal_unit[0] & kH264TypeMask;
     nalu.sps_id = -1;
     nalu.pps_id = -1;
-    ArrayView<const uint8_t> nalu_data = nal_unit.subspan(H264::kNaluTypeSize);
+    std::span<const uint8_t> nalu_data = nal_unit.subspan(H264::kNaluTypeSize);
 
     if (nalu_data.empty()) {
       RTC_LOG(LS_WARNING) << "Skipping empty NAL unit.";
@@ -246,7 +246,7 @@ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> ParseFuaNalu(
     if (original_nal_type == H264::NaluType::kIdr ||
         original_nal_type == H264::NaluType::kSlice) {
       std::optional<PpsParser::SliceHeader> slice_header =
-          PpsParser::ParseSliceHeader(ArrayView<const uint8_t>(rtp_payload)
+          PpsParser::ParseSliceHeader(std::span<const uint8_t>(rtp_payload)
                                           .subspan(2 * kNalHeaderSize));
       if (slice_header) {
         nalu.pps_id = slice_header->pic_parameter_set_id;
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_h264_unittest.cc b/modules/rtp_rtcp/source/video_rtp_depacketizer_h264_unittest.cc
index 5739ec019b..b2cc10c3ee 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_h264_unittest.cc
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_h264_unittest.cc
@@ -12,9 +12,9 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/video/video_codec_type.h"
 #include "api/video/video_frame_type.h"
 #include "common_video/h264/h264_common.h"
@@ -228,9 +228,9 @@ TEST(VideoRtpDepacketizerH264Test, DepacketizeWithRewriting) {
   VideoRtpDepacketizerH264 depacketizer;
   auto parsed = depacketizer.Parse(in_buffer);
   ASSERT_TRUE(parsed);
-  EXPECT_THAT(MakeArrayView(parsed->video_payload.cdata(),
-                            parsed->video_payload.size()),
-              ElementsAreArray(out_buffer));
+  EXPECT_THAT(
+      std::span(parsed->video_payload.cdata(), parsed->video_payload.size()),
+      ElementsAreArray(out_buffer));
 }
 
 TEST(VideoRtpDepacketizerH264Test, DepacketizeWithDoubleRewriting) {
@@ -273,9 +273,9 @@ TEST(VideoRtpDepacketizerH264Test, DepacketizeWithDoubleRewriting) {
   ASSERT_TRUE(parsed);
   std::vector<uint8_t> expected_packet_payload(
       out_buffer.data(), &out_buffer.data()[out_buffer.size()]);
-  EXPECT_THAT(MakeArrayView(parsed->video_payload.cdata(),
-                            parsed->video_payload.size()),
-              ElementsAreArray(out_buffer));
+  EXPECT_THAT(
+      std::span(parsed->video_payload.cdata(), parsed->video_payload.size()),
+      ElementsAreArray(out_buffer));
 }
 
 TEST(VideoRtpDepacketizerH264Test, StapADelta) {
@@ -343,9 +343,9 @@ TEST(VideoRtpDepacketizerH264Test, FuA) {
   ASSERT_TRUE(parsed1);
   // We expect that the first packet is one byte shorter since the FU-A header
   // has been replaced by the original nal header.
-  EXPECT_THAT(MakeArrayView(parsed1->video_payload.cdata(),
-                            parsed1->video_payload.size()),
-              ElementsAreArray(kExpected1));
+  EXPECT_THAT(
+      std::span(parsed1->video_payload.cdata(), parsed1->video_payload.size()),
+      ElementsAreArray(kExpected1));
   EXPECT_EQ(parsed1->video_header.frame_type, VideoFrameType::kVideoFrameKey);
   EXPECT_EQ(parsed1->video_header.codec, kVideoCodecH264);
   EXPECT_TRUE(parsed1->video_header.is_first_packet_in_frame);
@@ -363,9 +363,9 @@ TEST(VideoRtpDepacketizerH264Test, FuA) {
   // Following packets will be 2 bytes shorter since they will only be appended
   // onto the first packet.
   auto parsed2 = depacketizer.Parse(CopyOnWriteBuffer(kPayload2));
-  EXPECT_THAT(MakeArrayView(parsed2->video_payload.cdata(),
-                            parsed2->video_payload.size()),
-              ElementsAreArray(kExpected2));
+  EXPECT_THAT(
+      std::span(parsed2->video_payload.cdata(), parsed2->video_payload.size()),
+      ElementsAreArray(kExpected2));
   EXPECT_FALSE(parsed2->video_header.is_first_packet_in_frame);
   EXPECT_EQ(parsed2->video_header.codec, kVideoCodecH264);
   {
@@ -378,9 +378,9 @@ TEST(VideoRtpDepacketizerH264Test, FuA) {
   }
 
   auto parsed3 = depacketizer.Parse(CopyOnWriteBuffer(kPayload3));
-  EXPECT_THAT(MakeArrayView(parsed3->video_payload.cdata(),
-                            parsed3->video_payload.size()),
-              ElementsAreArray(kExpected3));
+  EXPECT_THAT(
+      std::span(parsed3->video_payload.cdata(), parsed3->video_payload.size()),
+      ElementsAreArray(kExpected3));
   EXPECT_FALSE(parsed3->video_header.is_first_packet_in_frame);
   EXPECT_EQ(parsed3->video_header.codec, kVideoCodecH264);
   {
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_h265.cc b/modules/rtp_rtcp/source/video_rtp_depacketizer_h265.cc
index 4af35dc2c4..7babd6f298 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_h265.cc
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_h265.cc
@@ -13,11 +13,11 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
 #include "absl/base/attributes.h"
-#include "api/array_view.h"
 #include "api/video/video_codec_type.h"
 #include "api/video/video_frame_type.h"
 #include "common_video/h265/h265_bitstream_parser.h"
@@ -118,7 +118,7 @@ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> ProcessApOrSingleNalu(
 
     uint8_t nalu_type = (payload_data[start_offset] & kH265TypeMask) >> 1;
     start_offset += kH265NalHeaderSizeBytes;
-    ArrayView<const uint8_t> nalu_data(&payload_data[start_offset],
+    std::span<const uint8_t> nalu_data(&payload_data[start_offset],
                                        end_offset - start_offset);
     switch (nalu_type) {
       case H265::NaluType::kBlaWLp:
@@ -213,7 +213,7 @@ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> ParseFuNalu(
           kH265FuHeaderSizeBytes + kH265PayloadHeaderSizeBytes;
       std::optional<bool> first_slice_segment_in_pic_flag =
           H265BitstreamParser::IsFirstSliceSegmentInPic(
-              ArrayView<const uint8_t>(rtp_payload.cdata() + slice_offset,
+              std::span<const uint8_t>(rtp_payload.cdata() + slice_offset,
                                        rtp_payload.size() - slice_offset));
       if (first_slice_segment_in_pic_flag.value_or(false)) {
         is_first_packet_in_frame = true;
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_h265_unittest.cc b/modules/rtp_rtcp/source/video_rtp_depacketizer_h265_unittest.cc
index 0e21d00139..25ee05b7b2 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_h265_unittest.cc
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_h265_unittest.cc
@@ -12,8 +12,8 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/video/video_codec_type.h"
 #include "api/video/video_frame_type.h"
 #include "common_video/h265/h265_common.h"
@@ -39,9 +39,9 @@ TEST(VideoRtpDepacketizerH265Test, SingleNalu) {
       depacketizer.Parse(rtp_payload);
   ASSERT_TRUE(parsed);
 
-  EXPECT_THAT(MakeArrayView(parsed->video_payload.cdata(),
-                            parsed->video_payload.size()),
-              ElementsAreArray(expected_packet));
+  EXPECT_THAT(
+      std::span(parsed->video_payload.cdata(), parsed->video_payload.size()),
+      ElementsAreArray(expected_packet));
   EXPECT_EQ(parsed->video_header.frame_type, VideoFrameType::kVideoFrameKey);
   EXPECT_EQ(parsed->video_header.codec, kVideoCodecH265);
   EXPECT_TRUE(parsed->video_header.is_first_packet_in_frame);
@@ -73,9 +73,9 @@ TEST(VideoRtpDepacketizerH265Test, SingleNaluSpsWithResolution) {
       depacketizer.Parse(rtp_payload);
   ASSERT_TRUE(parsed);
 
-  EXPECT_THAT(MakeArrayView(parsed->video_payload.cdata(),
-                            parsed->video_payload.size()),
-              ElementsAreArray(expected_packet));
+  EXPECT_THAT(
+      std::span(parsed->video_payload.cdata(), parsed->video_payload.size()),
+      ElementsAreArray(expected_packet));
   EXPECT_EQ(parsed->video_header.codec, kVideoCodecH265);
   EXPECT_TRUE(parsed->video_header.is_first_packet_in_frame);
   EXPECT_EQ(parsed->video_header.width, 1280u);
@@ -159,9 +159,9 @@ TEST(VideoRtpDepacketizerH265Test, ApKey) {
       depacketizer.Parse(rtp_payload);
   ASSERT_TRUE(parsed);
 
-  EXPECT_THAT(MakeArrayView(parsed->video_payload.cdata(),
-                            parsed->video_payload.size()),
-              ElementsAreArray(expected_packet));
+  EXPECT_THAT(
+      std::span(parsed->video_payload.cdata(), parsed->video_payload.size()),
+      ElementsAreArray(expected_packet));
   EXPECT_EQ(parsed->video_header.frame_type, VideoFrameType::kVideoFrameKey);
   EXPECT_EQ(parsed->video_header.codec, kVideoCodecH265);
   EXPECT_TRUE(parsed->video_header.is_first_packet_in_frame);
@@ -218,9 +218,9 @@ TEST(VideoRtpDepacketizerH265Test, ApNaluSpsWithResolution) {
       depacketizer.Parse(rtp_payload);
   ASSERT_TRUE(parsed);
 
-  EXPECT_THAT(MakeArrayView(parsed->video_payload.cdata(),
-                            parsed->video_payload.size()),
-              ElementsAreArray(expected_packet));
+  EXPECT_THAT(
+      std::span(parsed->video_payload.cdata(), parsed->video_payload.size()),
+      ElementsAreArray(expected_packet));
   EXPECT_EQ(parsed->video_header.frame_type, VideoFrameType::kVideoFrameKey);
   EXPECT_EQ(parsed->video_header.codec, kVideoCodecH265);
   EXPECT_TRUE(parsed->video_header.is_first_packet_in_frame);
@@ -267,9 +267,9 @@ TEST(VideoRtpDepacketizerH265Test, ApDelta) {
       depacketizer.Parse(rtp_payload);
   ASSERT_TRUE(parsed);
 
-  EXPECT_THAT(MakeArrayView(parsed->video_payload.cdata(),
-                            parsed->video_payload.size()),
-              ElementsAreArray(expected_packet));
+  EXPECT_THAT(
+      std::span(parsed->video_payload.cdata(), parsed->video_payload.size()),
+      ElementsAreArray(expected_packet));
 
   EXPECT_EQ(parsed->video_header.frame_type, VideoFrameType::kVideoFrameDelta);
   EXPECT_EQ(parsed->video_header.codec, kVideoCodecH265);
@@ -309,9 +309,9 @@ TEST(VideoRtpDepacketizerH265Test, Fu) {
   ASSERT_TRUE(parsed1);
   // We expect that the first packet is one byte shorter since the FU header
   // has been replaced by the original nal header.
-  EXPECT_THAT(MakeArrayView(parsed1->video_payload.cdata(),
-                            parsed1->video_payload.size()),
-              ElementsAreArray(kExpected1));
+  EXPECT_THAT(
+      std::span(parsed1->video_payload.cdata(), parsed1->video_payload.size()),
+      ElementsAreArray(kExpected1));
   EXPECT_EQ(parsed1->video_header.frame_type, VideoFrameType::kVideoFrameKey);
   EXPECT_EQ(parsed1->video_header.codec, kVideoCodecH265);
   EXPECT_TRUE(parsed1->video_header.is_first_packet_in_frame);
@@ -319,17 +319,17 @@ TEST(VideoRtpDepacketizerH265Test, Fu) {
   // Following packets will be 2 bytes shorter since they will only be appended
   // onto the first packet.
   auto parsed2 = depacketizer.Parse(CopyOnWriteBuffer(packet2));
-  EXPECT_THAT(MakeArrayView(parsed2->video_payload.cdata(),
-                            parsed2->video_payload.size()),
-              ElementsAreArray(kExpected2));
+  EXPECT_THAT(
+      std::span(parsed2->video_payload.cdata(), parsed2->video_payload.size()),
+      ElementsAreArray(kExpected2));
   EXPECT_FALSE(parsed2->video_header.is_first_packet_in_frame);
   EXPECT_EQ(parsed2->video_header.frame_type, VideoFrameType::kVideoFrameKey);
   EXPECT_EQ(parsed2->video_header.codec, kVideoCodecH265);
 
   auto parsed3 = depacketizer.Parse(CopyOnWriteBuffer(packet3));
-  EXPECT_THAT(MakeArrayView(parsed3->video_payload.cdata(),
-                            parsed3->video_payload.size()),
-              ElementsAreArray(kExpected3));
+  EXPECT_THAT(
+      std::span(parsed3->video_payload.cdata(), parsed3->video_payload.size()),
+      ElementsAreArray(kExpected3));
   EXPECT_FALSE(parsed3->video_header.is_first_packet_in_frame);
   EXPECT_EQ(parsed3->video_header.frame_type, VideoFrameType::kVideoFrameKey);
   EXPECT_EQ(parsed3->video_header.codec, kVideoCodecH265);
@@ -448,9 +448,9 @@ TEST(VideoRtpDepacketizerH265Test, ApVpsSpsPpsMultiIdrSlices) {
       depacketizer.Parse(rtp_payload);
   ASSERT_TRUE(parsed.has_value());
 
-  EXPECT_THAT(MakeArrayView(parsed->video_payload.cdata(),
-                            parsed->video_payload.size()),
-              ElementsAreArray(expected_packet));
+  EXPECT_THAT(
+      std::span(parsed->video_payload.cdata(), parsed->video_payload.size()),
+      ElementsAreArray(expected_packet));
   EXPECT_EQ(parsed->video_header.frame_type, VideoFrameType::kVideoFrameKey);
   EXPECT_TRUE(parsed->video_header.is_first_packet_in_frame);
 }
@@ -484,9 +484,9 @@ TEST(VideoRtpDepacketizerH265Test, ApMultiNonFirstSlicesFromSingleNonIdrFrame) {
       depacketizer.Parse(rtp_payload);
   ASSERT_TRUE(parsed.has_value());
 
-  EXPECT_THAT(MakeArrayView(parsed->video_payload.cdata(),
-                            parsed->video_payload.size()),
-              ElementsAreArray(expected_packet));
+  EXPECT_THAT(
+      std::span(parsed->video_payload.cdata(), parsed->video_payload.size()),
+      ElementsAreArray(expected_packet));
   EXPECT_EQ(parsed->video_header.frame_type, VideoFrameType::kVideoFrameDelta);
   EXPECT_FALSE(parsed->video_header.is_first_packet_in_frame);
 }
@@ -520,9 +520,9 @@ TEST(VideoRtpDepacketizerH265Test, ApFirstTwoSlicesFromSingleNonIdrFrame) {
       depacketizer.Parse(rtp_payload);
   ASSERT_TRUE(parsed.has_value());
 
-  EXPECT_THAT(MakeArrayView(parsed->video_payload.cdata(),
-                            parsed->video_payload.size()),
-              ElementsAreArray(expected_packet));
+  EXPECT_THAT(
+      std::span(parsed->video_payload.cdata(), parsed->video_payload.size()),
+      ElementsAreArray(expected_packet));
   EXPECT_EQ(parsed->video_header.frame_type, VideoFrameType::kVideoFrameDelta);
   EXPECT_TRUE(parsed->video_header.is_first_packet_in_frame);
 }
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.cc b/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.cc
index f3f4fb5ad3..9a792962cb 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.cc
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.cc
@@ -13,9 +13,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <utility>
 
-#include "api/array_view.h"
 #include "api/video/video_codec_type.h"
 #include "api/video/video_frame_type.h"
 #include "modules/rtp_rtcp/source/rtp_video_header.h"
@@ -139,7 +139,7 @@ int ParseVP8Descriptor(RTPVideoHeaderVP8* vp8,
 
 std::optional<VideoRtpDepacketizer::ParsedRtpPayload>
 VideoRtpDepacketizerVp8::Parse(CopyOnWriteBuffer rtp_payload) {
-  ArrayView<const uint8_t> payload(rtp_payload.cdata(), rtp_payload.size());
+  std::span<const uint8_t> payload(rtp_payload.cdata(), rtp_payload.size());
   std::optional<ParsedRtpPayload> result(std::in_place);
   int offset = ParseRtpPayload(payload, &result->video_header);
   if (offset == kFailedToParse)
@@ -151,7 +151,7 @@ VideoRtpDepacketizerVp8::Parse(CopyOnWriteBuffer rtp_payload) {
 }
 
 int VideoRtpDepacketizerVp8::ParseRtpPayload(
-    ArrayView<const uint8_t> rtp_payload,
+    std::span<const uint8_t> rtp_payload,
     RTPVideoHeader* video_header) {
   RTC_DCHECK(video_header);
   if (rtp_payload.empty()) {
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.h b/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.h
index a01db9a04c..663ffe6799 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.h
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.h
@@ -13,8 +13,8 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/rtp_rtcp/source/rtp_video_header.h"
 #include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
 #include "rtc_base/copy_on_write_buffer.h"
@@ -30,7 +30,7 @@ class VideoRtpDepacketizerVp8 : public VideoRtpDepacketizer {
 
   // Parses vp8 rtp payload descriptor.
   // Returns zero on error or vp8 payload header offset on success.
-  static int ParseRtpPayload(ArrayView<const uint8_t> rtp_payload,
+  static int ParseRtpPayload(std::span<const uint8_t> rtp_payload,
                              RTPVideoHeader* video_header);
 
   std::optional<ParsedRtpPayload> Parse(CopyOnWriteBuffer rtp_payload) override;
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8_unittest.cc b/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8_unittest.cc
index 9fad3a1d2d..e92ee91f4d 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8_unittest.cc
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8_unittest.cc
@@ -13,8 +13,8 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/video/video_codec_type.h"
 #include "api/video/video_frame_type.h"
 #include "modules/rtp_rtcp/source/rtp_format_vp8.h"
@@ -244,7 +244,7 @@ TEST(VideoRtpDepacketizerVp8Test, ReferencesInputCopyOnWriteBuffer) {
 }
 
 TEST(VideoRtpDepacketizerVp8Test, FailsOnEmptyPayload) {
-  ArrayView<const uint8_t> empty;
+  std::span<const uint8_t> empty;
   RTPVideoHeader video_header;
   EXPECT_EQ(VideoRtpDepacketizerVp8::ParseRtpPayload(empty, &video_header), 0);
 }
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.cc b/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.cc
index c69f48f952..75ada1863b 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.cc
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.cc
@@ -13,9 +13,9 @@
 #include <cstdint>
 #include <cstring>
 #include <optional>
+#include <span>
 #include <utility>
 
-#include "api/array_view.h"
 #include "api/video/video_codec_constants.h"
 #include "api/video/video_codec_type.h"
 #include "api/video/video_frame_type.h"
@@ -167,7 +167,7 @@ VideoRtpDepacketizerVp9::Parse(CopyOnWriteBuffer rtp_payload) {
 }
 
 int VideoRtpDepacketizerVp9::ParseRtpPayload(
-    ArrayView<const uint8_t> rtp_payload,
+    std::span<const uint8_t> rtp_payload,
     RTPVideoHeader* video_header) {
   RTC_DCHECK(video_header);
   // Parse mandatory first byte of payload descriptor.
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.h b/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.h
index d5d45ec2c4..8416936378 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.h
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.h
@@ -13,8 +13,8 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/rtp_rtcp/source/rtp_video_header.h"
 #include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
 #include "rtc_base/copy_on_write_buffer.h"
@@ -30,7 +30,7 @@ class VideoRtpDepacketizerVp9 : public VideoRtpDepacketizer {
 
   // Parses vp9 rtp payload descriptor.
   // Returns zero on error or vp9 payload header offset on success.
-  static int ParseRtpPayload(ArrayView<const uint8_t> rtp_payload,
+  static int ParseRtpPayload(std::span<const uint8_t> rtp_payload,
                              RTPVideoHeader* video_header);
 
   std::optional<ParsedRtpPayload> Parse(CopyOnWriteBuffer rtp_payload) override;
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9_unittest.cc b/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9_unittest.cc
index 91bd7f5143..de7674c79f 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9_unittest.cc
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9_unittest.cc
@@ -13,8 +13,8 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/video/video_frame_type.h"
 #include "modules/rtp_rtcp/source/rtp_video_header.h"
 #include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
@@ -322,7 +322,7 @@ TEST(VideoRtpDepacketizerVp9Test, ParseResolution) {
 }
 
 TEST(VideoRtpDepacketizerVp9Test, ParseFailsForNoPayloadLength) {
-  ArrayView<const uint8_t> empty;
+  std::span<const uint8_t> empty;
 
   RTPVideoHeader video_header;
   EXPECT_EQ(VideoRtpDepacketizerVp9::ParseRtpPayload(empty, &video_header), 0);
diff --git a/modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc b/modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc
index 2dc4eef06a..e0278abaf9 100644
--- a/modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc
+++ b/modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc
@@ -48,9 +48,9 @@
 #include <cstdio>
 #include <cstring>
 #include <memory>
+#include <span>
 #include <string>
 
-#include "api/array_view.h"
 #include "modules/include/module_fec_types.h"
 #include "modules/rtp_rtcp/source/forward_error_correction_internal.h"
 #include "modules/rtp_rtcp/test/testFec/average_residual_loss_xor_codes.h"
@@ -731,7 +731,7 @@ class FecPacketMaskMetricsTest : public ::testing::Test {
       for (int num_fec_packets = 1; num_fec_packets <= num_media_packets;
            num_fec_packets++) {
         memset(packet_mask.get(), 0, num_media_packets * mask_bytes_fec_packet);
-        ArrayView<const uint8_t> mask =
+        std::span<const uint8_t> mask =
             mask_table.LookUp(num_media_packets, num_fec_packets);
         memcpy(packet_mask.get(), &mask[0], mask.size());
         // Convert to bit mask.

From 7aa71b935a455f8cb437ad6b7f83936cf533aa13 Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Thu, 5 Mar 2026 11:08:08 +0100
Subject: [PATCH 207/472] Replace ArrayView with std::span in video/

This change generated using mostly search and replace:
- MakeArrayView replaced with std::span
- ArrayView replaced with std::span
- Run include cleaner
- Removed dependency on array_view target

Bug: webrtc:439801349
Change-Id: I4a99476d9f56878d4af6155b20716ec369a0f025
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453800
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47060}
---
 video/BUILD.gn                                |  3 -
 video/adaptation/BUILD.gn                     |  1 -
 .../overuse_frame_detector_unittest.cc        |  4 +-
 video/buffered_frame_decryptor.cc             |  4 +-
 video/config/BUILD.gn                         |  1 -
 video/config/simulcast.cc                     |  6 +-
 video/config/simulcast.h                      |  4 +-
 video/corruption_detection/BUILD.gn           |  2 -
 .../corruption_classifier.cc                  | 10 ++--
 .../corruption_classifier.h                   | 10 ++--
 .../corruption_detection/evaluation/BUILD.gn  |  3 -
 .../evaluation/av1_qp_parser.cc               |  4 +-
 .../evaluation/av1_qp_parser.h                |  4 +-
 .../evaluation/qp_parser.cc                   |  4 +-
 .../evaluation/qp_parser.h                    |  4 +-
 .../corruption_detection/evaluation/utils.cc  |  4 +-
 video/corruption_detection/evaluation/utils.h |  4 +-
 .../halton_frame_sampler_unittest.cc          |  6 +-
 video/end_to_end_tests/bandwidth_tests.cc     |  6 +-
 .../extended_reports_tests.cc                 |  6 +-
 video/end_to_end_tests/fec_tests.cc           | 12 ++--
 video/end_to_end_tests/histogram_tests.cc     |  4 +-
 .../multi_codec_receive_tests.cc              |  4 +-
 video/end_to_end_tests/network_state_tests.cc | 18 +++---
 .../end_to_end_tests/retransmission_tests.cc  | 16 ++---
 video/end_to_end_tests/rtp_rtcp_tests.cc      | 12 ++--
 video/end_to_end_tests/ssrc_tests.cc          |  8 +--
 video/end_to_end_tests/stats_tests.cc         | 18 +++---
 .../transport_feedback_tests.cc               | 22 +++----
 video/picture_id_tests.cc                     |  4 +-
 video/quality_scaling_tests.cc                |  4 +-
 video/rtp_video_stream_receiver2.cc           |  8 +--
 video/rtp_video_stream_receiver2.h            |  4 +-
 video/rtp_video_stream_receiver2_unittest.cc  |  6 +-
 video/timing/simulator/BUILD.gn               |  1 -
 video/timing/simulator/assembler.h            |  6 +-
 .../timing/simulator/decodability_simulator.h |  4 +-
 video/timing/simulator/rendering_simulator.h  |  8 +--
 video/transport_adapter.cc                    |  6 +-
 video/transport_adapter.h                     |  6 +-
 video/video_analyzer.cc                       |  6 +-
 video/video_analyzer.h                        |  6 +-
 video/video_receive_stream2.cc                |  4 +-
 video/video_receive_stream2.h                 |  4 +-
 video/video_receive_stream2_unittest.cc       |  4 +-
 video/video_send_stream_impl.cc               |  6 +-
 video/video_send_stream_impl.h                |  6 +-
 video/video_send_stream_impl_unittest.cc      |  8 +--
 video/video_send_stream_tests.cc              | 58 +++++++++----------
 video/video_stream_encoder_unittest.cc        |  4 +-
 50 files changed, 177 insertions(+), 190 deletions(-)

diff --git a/video/BUILD.gn b/video/BUILD.gn
index b9e2d4497f..f7f773e62b 100644
--- a/video/BUILD.gn
+++ b/video/BUILD.gn
@@ -78,7 +78,6 @@ rtc_library("video") {
     ":video_stream_buffer_controller",
     ":video_stream_encoder_impl",
     ":video_stream_encoder_interface",
-    "../api:array_view",
     "../api:bitrate_allocation",
     "../api:fec_controller_api",
     "../api:field_trials_view",
@@ -528,7 +527,6 @@ if (rtc_include_tests) {
       ]
       deps = [
         ":frame_dumping_decoder",
-        "../api:array_view",
         "../api:create_frame_generator",
         "../api:fec_controller_api",
         "../api:field_trials_view",
@@ -836,7 +834,6 @@ if (rtc_include_tests) {
       ":video_stream_buffer_controller",
       ":video_stream_encoder_impl",
       ":video_stream_encoder_interface",
-      "../api:array_view",
       "../api:bitrate_allocation",
       "../api:create_frame_generator",
       "../api:fake_frame_decryptor",
diff --git a/video/adaptation/BUILD.gn b/video/adaptation/BUILD.gn
index fee4c2e494..04a00da3e0 100644
--- a/video/adaptation/BUILD.gn
+++ b/video/adaptation/BUILD.gn
@@ -85,7 +85,6 @@ if (rtc_include_tests) {
     deps = [
       ":video_adaptation",
       "..:video_stream_encoder_interface",
-      "../../api:array_view",
       "../../api:field_trials",
       "../../api:rtp_parameters",
       "../../api:scoped_refptr",
diff --git a/video/adaptation/overuse_frame_detector_unittest.cc b/video/adaptation/overuse_frame_detector_unittest.cc
index d474e9776d..a3948eb7d8 100644
--- a/video/adaptation/overuse_frame_detector_unittest.cc
+++ b/video/adaptation/overuse_frame_detector_unittest.cc
@@ -14,8 +14,8 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
@@ -128,7 +128,7 @@ class OveruseFrameDetectorTest : public ::testing::Test,
       int width,
       int height,
       // One element per layer
-      ArrayView<const TimeDelta> delays_us) {
+      std::span<const TimeDelta> delays_us) {
     VideoFrame frame =
         VideoFrame::Builder()
             .set_video_frame_buffer(I420Buffer::Create(width, height))
diff --git a/video/buffered_frame_decryptor.cc b/video/buffered_frame_decryptor.cc
index 03fc2eb567..4c635cb364 100644
--- a/video/buffered_frame_decryptor.cc
+++ b/video/buffered_frame_decryptor.cc
@@ -13,10 +13,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/crypto/frame_decryptor_interface.h"
 #include "api/field_trials_view.h"
 #include "api/media_types.h"
@@ -77,7 +77,7 @@ BufferedFrameDecryptor::FrameDecision BufferedFrameDecryptor::DecryptFrame(
                                                 frame->size());
   RTC_CHECK_LE(max_plaintext_byte_size, frame->size());
   // Place the decrypted frame inline into the existing frame.
-  ArrayView<uint8_t> inline_decrypted_bitstream(frame->mutable_data(),
+  std::span<uint8_t> inline_decrypted_bitstream(frame->mutable_data(),
                                                 max_plaintext_byte_size);
 
   // Enable authenticating the header if the field trial isn't disabled.
diff --git a/video/config/BUILD.gn b/video/config/BUILD.gn
index 6647485498..0112dd8796 100644
--- a/video/config/BUILD.gn
+++ b/video/config/BUILD.gn
@@ -18,7 +18,6 @@ rtc_library("streams_config") {
 
   deps = [
     ":encoder_config",
-    "../../api:array_view",
     "../../api:field_trials_view",
     "../../api/units:data_rate",
     "../../api/video:resolution",
diff --git a/video/config/simulcast.cc b/video/config/simulcast.cc
index 8c4ed15574..01c9115037 100644
--- a/video/config/simulcast.cc
+++ b/video/config/simulcast.cc
@@ -14,10 +14,10 @@
 #include <cstdint>
 #include <cstdio>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/field_trials_view.h"
 #include "api/units/data_rate.h"
 #include "api/video/resolution.h"
@@ -296,7 +296,7 @@ SimulcastFormat InterpolateSimulcastFormat(
 }
 
 std::vector<VideoStream> GetNormalSimulcastLayers(
-    ArrayView<const Resolution> resolutions,
+    std::span<const Resolution> resolutions,
     bool temporal_layers_supported,
     bool base_heavy_tl3_rate_alloc,
     const FieldTrialsView& trials,
@@ -469,7 +469,7 @@ DataRate GetTotalMaxBitrate(const std::vector<VideoStream>& layers) {
 }
 
 std::vector<VideoStream> GetSimulcastConfig(
-    ArrayView<const Resolution> resolutions,
+    std::span<const Resolution> resolutions,
     bool is_screenshare_with_conference_mode,
     bool temporal_layers_supported,
     const FieldTrialsView& trials,
diff --git a/video/config/simulcast.h b/video/config/simulcast.h
index 0e36cddb91..59711922be 100644
--- a/video/config/simulcast.h
+++ b/video/config/simulcast.h
@@ -13,9 +13,9 @@
 
 #include <stddef.h>
 
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/field_trials_view.h"
 #include "api/units/data_rate.h"
 #include "api/video/resolution.h"
@@ -44,7 +44,7 @@ size_t LimitSimulcastLayerCount(size_t min_num_layers,
 
 // Gets simulcast settings.
 std::vector<VideoStream> GetSimulcastConfig(
-    ArrayView<const Resolution> resolutions,
+    std::span<const Resolution> resolutions,
     bool is_screenshare_with_conference_mode,
     bool temporal_layers_supported,
     const FieldTrialsView& trials,
diff --git a/video/corruption_detection/BUILD.gn b/video/corruption_detection/BUILD.gn
index 53596d6317..824de16840 100644
--- a/video/corruption_detection/BUILD.gn
+++ b/video/corruption_detection/BUILD.gn
@@ -15,7 +15,6 @@ rtc_library("corruption_classifier") {
   ]
   deps = [
     ":halton_frame_sampler",
-    "../../api:array_view",
     "../../rtc_base:checks",
     "../../rtc_base:logging",
   ]
@@ -196,7 +195,6 @@ if (rtc_include_tests) {
     deps = [
       ":halton_frame_sampler",
       ":video_frame_sampler",
-      "../../api:array_view",
       "../../api:scoped_refptr",
       "../../api/video:video_frame",
       "../../test:test_support",
diff --git a/video/corruption_detection/corruption_classifier.cc b/video/corruption_detection/corruption_classifier.cc
index 0c37dae5b9..2b5729c6f5 100644
--- a/video/corruption_detection/corruption_classifier.cc
+++ b/video/corruption_detection/corruption_classifier.cc
@@ -12,9 +12,9 @@
 
 #include <algorithm>
 #include <cmath>
+#include <span>
 #include <variant>
 
-#include "api/array_view.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
 #include "video/corruption_detection/halton_frame_sampler.h"
@@ -39,8 +39,8 @@ CorruptionClassifier::CorruptionClassifier(float growth_rate, float midpoint)
 }
 
 double CorruptionClassifier::CalculateCorruptionProbability(
-    ArrayView<const FilteredSample> filtered_original_samples,
-    ArrayView<const FilteredSample> filtered_compressed_samples,
+    std::span<const FilteredSample> filtered_original_samples,
+    std::span<const FilteredSample> filtered_compressed_samples,
     int luma_threshold,
     int chroma_threshold) const {
   double loss = GetScore(filtered_original_samples, filtered_compressed_samples,
@@ -67,8 +67,8 @@ double CorruptionClassifier::CalculateCorruptionProbability(
 // either `luma_threshold` or `chroma_threshold` depending on whether the
 // sample is luma or chroma.
 double CorruptionClassifier::GetScore(
-    ArrayView<const FilteredSample> filtered_original_samples,
-    ArrayView<const FilteredSample> filtered_compressed_samples,
+    std::span<const FilteredSample> filtered_original_samples,
+    std::span<const FilteredSample> filtered_compressed_samples,
     int luma_threshold,
     int chroma_threshold) const {
   RTC_DCHECK_GE(luma_threshold, 0);
diff --git a/video/corruption_detection/corruption_classifier.h b/video/corruption_detection/corruption_classifier.h
index 552f315c86..1ce356a1c2 100644
--- a/video/corruption_detection/corruption_classifier.h
+++ b/video/corruption_detection/corruption_classifier.h
@@ -11,9 +11,9 @@
 #ifndef VIDEO_CORRUPTION_DETECTION_CORRUPTION_CLASSIFIER_H_
 #define VIDEO_CORRUPTION_DETECTION_CORRUPTION_CLASSIFIER_H_
 
+#include <span>
 #include <variant>
 
-#include "api/array_view.h"
 #include "video/corruption_detection/halton_frame_sampler.h"
 
 namespace webrtc {
@@ -42,8 +42,8 @@ class CorruptionClassifier {
   // or by applying a logistic function to the loss. The method is chosen
   // depending on the used constructor.
   double CalculateCorruptionProbability(
-      ArrayView<const FilteredSample> filtered_original_samples,
-      ArrayView<const FilteredSample> filtered_compressed_samples,
+      std::span<const FilteredSample> filtered_original_samples,
+      std::span<const FilteredSample> filtered_compressed_samples,
       int luma_threshold,
       int chroma_threshold) const;
 
@@ -61,8 +61,8 @@ class CorruptionClassifier {
 
   // Returns the non-normalized score between the original and the compressed
   // frames' samples.
-  double GetScore(ArrayView<const FilteredSample> filtered_original_samples,
-                  ArrayView<const FilteredSample> filtered_compressed_samples,
+  double GetScore(std::span<const FilteredSample> filtered_original_samples,
+                  std::span<const FilteredSample> filtered_compressed_samples,
                   int luma_threshold,
                   int chroma_threshold) const;
 
diff --git a/video/corruption_detection/evaluation/BUILD.gn b/video/corruption_detection/evaluation/BUILD.gn
index 87076f4464..abe28247c3 100644
--- a/video/corruption_detection/evaluation/BUILD.gn
+++ b/video/corruption_detection/evaluation/BUILD.gn
@@ -15,7 +15,6 @@ rtc_library("av1_qp_parser") {
     "av1_qp_parser.h",
   ]
   deps = [
-    "../../../api:array_view",
     "../../../api/video:video_frame",
     "../../../api/video:video_frame_type",
     "//third_party/libgav1:libgav1_parser",
@@ -108,7 +107,6 @@ rtc_library("qp_parser") {
   ]
   deps = [
     ":av1_qp_parser",
-    "../../../api:array_view",
     "../../../api/video:video_frame",
     "../../../modules/video_coding:video_coding_utility",
     "../../../rtc_base:checks",
@@ -138,7 +136,6 @@ rtc_library("utils") {
     "utils.h",
   ]
   deps = [
-    "../../../api:array_view",
     "../../../api/video:video_frame",
     "../../../api/video_codecs:video_codecs_api",
     "../../../rtc_base:checks",
diff --git a/video/corruption_detection/evaluation/av1_qp_parser.cc b/video/corruption_detection/evaluation/av1_qp_parser.cc
index 42ef4ae1e0..4ee4416d41 100644
--- a/video/corruption_detection/evaluation/av1_qp_parser.cc
+++ b/video/corruption_detection/evaluation/av1_qp_parser.cc
@@ -12,15 +12,15 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "third_party/libgav1/src/src/buffer_pool.h"
 #include "third_party/libgav1/src/src/gav1/status_code.h"
 #include "third_party/libgav1/src/src/obu_parser.h"
 
 namespace webrtc {
 
-std::optional<uint32_t> Av1QpParser::Parse(ArrayView<const uint8_t> frame_data,
+std::optional<uint32_t> Av1QpParser::Parse(std::span<const uint8_t> frame_data,
                                            int operating_point) {
   libgav1::RefCountedBufferPtr curr_frame;
   libgav1::ObuParser parser(frame_data.data(), frame_data.size(),
diff --git a/video/corruption_detection/evaluation/av1_qp_parser.h b/video/corruption_detection/evaluation/av1_qp_parser.h
index 6d1fe2aa3c..fcd4a5151d 100644
--- a/video/corruption_detection/evaluation/av1_qp_parser.h
+++ b/video/corruption_detection/evaluation/av1_qp_parser.h
@@ -13,8 +13,8 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "third_party/libgav1/src/src/buffer_pool.h"
 #include "third_party/libgav1/src/src/decoder_state.h"
 #include "third_party/libgav1/src/src/obu_parser.h"
@@ -44,7 +44,7 @@ class Av1QpParser {
                      /*release_frame_buffer=*/nullptr,
                      /*callback_private_data=*/nullptr) {}
 
-  std::optional<uint32_t> Parse(ArrayView<const uint8_t> frame_data,
+  std::optional<uint32_t> Parse(std::span<const uint8_t> frame_data,
                                 int operating_point = 0);
 
  private:
diff --git a/video/corruption_detection/evaluation/qp_parser.cc b/video/corruption_detection/evaluation/qp_parser.cc
index b50434475e..64e9aaec67 100644
--- a/video/corruption_detection/evaluation/qp_parser.cc
+++ b/video/corruption_detection/evaluation/qp_parser.cc
@@ -13,8 +13,8 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/video/video_codec_type.h"
 #include "rtc_base/checks.h"
 
@@ -23,7 +23,7 @@ namespace webrtc {
 std::optional<uint32_t> GenericQpParser::Parse(
     VideoCodecType codec_type,
     size_t spatial_idx,
-    ArrayView<const uint8_t> frame_data,
+    std::span<const uint8_t> frame_data,
     int operating_point) {
   if (codec_type != kVideoCodecAV1) {
     return non_av1_parsers_.Parse(codec_type, spatial_idx, frame_data.data(),
diff --git a/video/corruption_detection/evaluation/qp_parser.h b/video/corruption_detection/evaluation/qp_parser.h
index cf7d6ebe80..676d0eec15 100644
--- a/video/corruption_detection/evaluation/qp_parser.h
+++ b/video/corruption_detection/evaluation/qp_parser.h
@@ -14,8 +14,8 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/video/video_codec_type.h"
 #include "modules/video_coding/utility/qp_parser.h"
 #include "video/corruption_detection/evaluation/av1_qp_parser.h"
@@ -29,7 +29,7 @@ class GenericQpParser {
  public:
   std::optional<uint32_t> Parse(VideoCodecType codec_type,
                                 size_t spatial_idx,
-                                ArrayView<const uint8_t> frame_data,
+                                std::span<const uint8_t> frame_data,
                                 int operating_point = 0);
 
  private:
diff --git a/video/corruption_detection/evaluation/utils.cc b/video/corruption_detection/evaluation/utils.cc
index 247e39237b..9c011b26e7 100644
--- a/video/corruption_detection/evaluation/utils.cc
+++ b/video/corruption_detection/evaluation/utils.cc
@@ -12,10 +12,10 @@
 
 #include <cstdint>
 #include <cstdio>
+#include <span>
 #include <string>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/video/video_codec_type.h"
 #include "api/video_codecs/sdp_video_format.h"
 #include "api/video_codecs/video_codec.h"
@@ -51,7 +51,7 @@ TempY4mFileCreator::~TempY4mFileCreator() {
 }
 
 void TempY4mFileCreator::CreateTempY4mFile(
-    ArrayView<const uint8_t> file_content) {
+    std::span<const uint8_t> file_content) {
   RTC_CHECK_EQ(file_content.size() % frame_size_, 0)
       << "Content size is not a multiple of frame size. Probably some data is "
          "missing.";
diff --git a/video/corruption_detection/evaluation/utils.h b/video/corruption_detection/evaluation/utils.h
index 65af4cf8e1..373ce89363 100644
--- a/video/corruption_detection/evaluation/utils.h
+++ b/video/corruption_detection/evaluation/utils.h
@@ -12,10 +12,10 @@
 #define VIDEO_CORRUPTION_DETECTION_EVALUATION_UTILS_H_
 
 #include <cstdint>
+#include <span>
 #include <string>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/video/video_codec_type.h"
 #include "api/video_codecs/sdp_video_format.h"
 #include "rtc_base/system/file_wrapper.h"
@@ -37,7 +37,7 @@ class TempY4mFileCreator {
   // format.
   //
   // The number of frames depends on the size of `file_content`.
-  void CreateTempY4mFile(ArrayView<const uint8_t> file_content);
+  void CreateTempY4mFile(std::span<const uint8_t> file_content);
 
   absl::string_view y4m_filepath() const { return y4m_filepath_; }
 
diff --git a/video/corruption_detection/halton_frame_sampler_unittest.cc b/video/corruption_detection/halton_frame_sampler_unittest.cc
index 9ee5ff0670..b36a212330 100644
--- a/video/corruption_detection/halton_frame_sampler_unittest.cc
+++ b/video/corruption_detection/halton_frame_sampler_unittest.cc
@@ -13,9 +13,9 @@
 #include <cstdint>
 #include <cstring>
 #include <memory>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/scoped_refptr.h"
 #include "api/video/i420_buffer.h"
 #include "api/video/video_frame.h"
@@ -67,9 +67,7 @@ VideoFrame MakeDefaultFrame() {
       .build();
 }
 
-VideoFrame MakeFrame(int width,
-                     int height,
-                     webrtc::ArrayView<const uint8_t> data) {
+VideoFrame MakeFrame(int width, int height, std::span<const uint8_t> data) {
   scoped_refptr<I420Buffer> buffer = I420Buffer::Create(width, height);
   memcpy(buffer->MutableDataY(), data.data(), width * height);
   return VideoFrame::Builder().set_video_frame_buffer(buffer).build();
diff --git a/video/end_to_end_tests/bandwidth_tests.cc b/video/end_to_end_tests/bandwidth_tests.cc
index e4d32432e4..ad4a42f790 100644
--- a/video/end_to_end_tests/bandwidth_tests.cc
+++ b/video/end_to_end_tests/bandwidth_tests.cc
@@ -10,9 +10,9 @@
 
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/environment/environment_factory.h"
 #include "api/rtp_headers.h"
@@ -72,7 +72,7 @@ TEST_F(BandwidthEndToEndTest, ReceiveStreamSendsRemb) {
           RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
     }
 
-    Action OnReceiveRtcp(ArrayView<const uint8_t> packet) override {
+    Action OnReceiveRtcp(std::span<const uint8_t> packet) override {
       test::RtcpPacketParser parser;
       EXPECT_TRUE(parser.Parse(packet));
 
@@ -141,7 +141,7 @@ class BandwidthStatsTest : public test::EndToEndTest {
   }
 
   // Called on the pacer thread.
-  Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+  Action OnSendRtp(std::span<const uint8_t> packet) override {
     // Stats need to be fetched on the thread where the caller objects were
     // constructed.
     task_queue_->PostTask([this]() {
diff --git a/video/end_to_end_tests/extended_reports_tests.cc b/video/end_to_end_tests/extended_reports_tests.cc
index e193a14bd7..fd14163414 100644
--- a/video/end_to_end_tests/extended_reports_tests.cc
+++ b/video/end_to_end_tests/extended_reports_tests.cc
@@ -11,10 +11,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/rtp_headers.h"
 #include "api/rtp_parameters.h"
 #include "api/test/simulated_network.h"
@@ -78,7 +78,7 @@ class RtcpXrObserver : public test::EndToEndTest {
 
  private:
   // Receive stream should send RR packets (and RRTR packets if enabled).
-  Action OnReceiveRtcp(ArrayView<const uint8_t> packet) override {
+  Action OnReceiveRtcp(std::span<const uint8_t> packet) override {
     MutexLock lock(&mutex_);
     test::RtcpPacketParser parser;
     EXPECT_TRUE(parser.Parse(packet));
@@ -95,7 +95,7 @@ class RtcpXrObserver : public test::EndToEndTest {
     return SEND_PACKET;
   }
   // Send stream should send SR packets (and DLRR packets if enabled).
-  Action OnSendRtcp(ArrayView<const uint8_t> packet) override {
+  Action OnSendRtcp(std::span<const uint8_t> packet) override {
     MutexLock lock(&mutex_);
     test::RtcpPacketParser parser;
     EXPECT_TRUE(parser.Parse(packet));
diff --git a/video/end_to_end_tests/fec_tests.cc b/video/end_to_end_tests/fec_tests.cc
index dc29416813..628237a175 100644
--- a/video/end_to_end_tests/fec_tests.cc
+++ b/video/end_to_end_tests/fec_tests.cc
@@ -12,9 +12,9 @@
 #include <cstdint>
 #include <memory>
 #include <set>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/rtp_parameters.h"
 #include "api/test/simulated_network.h"
@@ -82,7 +82,7 @@ TEST_F(FecEndToEndTest, ReceivesUlpfec) {
           num_packets_sent_(0) {}
 
    private:
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       MutexLock lock(&mutex_);
       RtpPacket rtp_packet;
       EXPECT_TRUE(rtp_packet.Parse(packet));
@@ -201,7 +201,7 @@ class FlexfecRenderObserver : public test::EndToEndTest,
   size_t GetNumFlexfecStreams() const override { return 1; }
 
  private:
-  Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+  Action OnSendRtp(std::span<const uint8_t> packet) override {
     MutexLock lock(&mutex_);
     RtpPacket rtp_packet;
     EXPECT_TRUE(rtp_packet.Parse(packet));
@@ -274,7 +274,7 @@ class FlexfecRenderObserver : public test::EndToEndTest,
     return SEND_PACKET;
   }
 
-  Action OnReceiveRtcp(ArrayView<const uint8_t> data) override {
+  Action OnReceiveRtcp(std::span<const uint8_t> data) override {
     test::RtcpPacketParser parser;
 
     parser.Parse(data);
@@ -388,7 +388,7 @@ TEST_F(FecEndToEndTest, ReceivedUlpfecPacketsNotNacked) {
               }) {}
 
    private:
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       MutexLock lock_(&mutex_);
       RtpPacket rtp_packet;
       EXPECT_TRUE(rtp_packet.Parse(packet));
@@ -456,7 +456,7 @@ TEST_F(FecEndToEndTest, ReceivedUlpfecPacketsNotNacked) {
       return SEND_PACKET;
     }
 
-    Action OnReceiveRtcp(ArrayView<const uint8_t> packet) override {
+    Action OnReceiveRtcp(std::span<const uint8_t> packet) override {
       MutexLock lock_(&mutex_);
       if (state_ == kVerifyUlpfecPacketNotInNackList) {
         test::RtcpPacketParser rtcp_parser;
diff --git a/video/end_to_end_tests/histogram_tests.cc b/video/end_to_end_tests/histogram_tests.cc
index 2fb2eb54bf..b85f7ee847 100644
--- a/video/end_to_end_tests/histogram_tests.cc
+++ b/video/end_to_end_tests/histogram_tests.cc
@@ -10,10 +10,10 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/rtp_parameters.h"
 #include "api/test/video/function_video_encoder_factory.h"
@@ -87,7 +87,7 @@ void HistogramTest::VerifyHistogramStats(bool use_rtx,
       }
     }
 
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       if (MinMetricRunTimePassed() && MinNumberOfFramesReceived())
         observation_complete_.Set();
 
diff --git a/video/end_to_end_tests/multi_codec_receive_tests.cc b/video/end_to_end_tests/multi_codec_receive_tests.cc
index bfd5e88f12..d8aa99dbc1 100644
--- a/video/end_to_end_tests/multi_codec_receive_tests.cc
+++ b/video/end_to_end_tests/multi_codec_receive_tests.cc
@@ -13,10 +13,10 @@
 #include <memory>
 #include <optional>
 #include <set>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/test/simulated_network.h"
 #include "api/test/video/function_video_decoder_factory.h"
@@ -96,7 +96,7 @@ class FrameObserver : public test::RtpRtcpObserver,
 
  private:
   // Sends kFramesToObserve.
-  Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+  Action OnSendRtp(std::span<const uint8_t> packet) override {
     MutexLock lock(&mutex_);
 
     RtpPacket rtp_packet;
diff --git a/video/end_to_end_tests/network_state_tests.cc b/video/end_to_end_tests/network_state_tests.cc
index f29646160e..cabd9d425e 100644
--- a/video/end_to_end_tests/network_state_tests.cc
+++ b/video/end_to_end_tests/network_state_tests.cc
@@ -10,9 +10,9 @@
 
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/call/transport.h"
 #include "api/environment/environment.h"
 #include "api/media_types.h"
@@ -52,13 +52,13 @@ class NetworkStateEndToEndTest : public test::CallTest {
  protected:
   class UnusedTransport : public Transport {
    private:
-    bool SendRtp(ArrayView<const uint8_t> packet,
+    bool SendRtp(std::span<const uint8_t> packet,
                  const PacketOptions& options) override {
       ADD_FAILURE() << "Unexpected RTP sent.";
       return false;
     }
 
-    bool SendRtcp(ArrayView<const uint8_t> packet,
+    bool SendRtcp(std::span<const uint8_t> packet,
                   const PacketOptions& /* options */) override {
       ADD_FAILURE() << "Unexpected RTCP sent.";
       return false;
@@ -78,14 +78,14 @@ class NetworkStateEndToEndTest : public test::CallTest {
     }
 
    private:
-    bool SendRtp(ArrayView<const uint8_t> packet,
+    bool SendRtp(std::span<const uint8_t> packet,
                  const PacketOptions& options) override {
       MutexLock lock(&mutex_);
       need_rtp_ = false;
       return true;
     }
 
-    bool SendRtcp(ArrayView<const uint8_t> packet,
+    bool SendRtcp(std::span<const uint8_t> packet,
                   const PacketOptions& /* options */) override {
       MutexLock lock(&mutex_);
       need_rtcp_ = false;
@@ -190,7 +190,7 @@ TEST_F(NetworkStateEndToEndTest, RespectsNetworkState) {
           receiver_rtcp_(0),
           down_frames_(0) {}
 
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       MutexLock lock(&test_mutex_);
       RtpPacket rtp_packet;
       EXPECT_TRUE(rtp_packet.Parse(packet));
@@ -201,19 +201,19 @@ TEST_F(NetworkStateEndToEndTest, RespectsNetworkState) {
       return SEND_PACKET;
     }
 
-    Action OnSendRtcp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtcp(std::span<const uint8_t> packet) override {
       MutexLock lock(&test_mutex_);
       ++sender_rtcp_;
       packet_event_.Set();
       return SEND_PACKET;
     }
 
-    Action OnReceiveRtp(ArrayView<const uint8_t> packet) override {
+    Action OnReceiveRtp(std::span<const uint8_t> packet) override {
       ADD_FAILURE() << "Unexpected receiver RTP, should not be sending.";
       return SEND_PACKET;
     }
 
-    Action OnReceiveRtcp(ArrayView<const uint8_t> packet) override {
+    Action OnReceiveRtcp(std::span<const uint8_t> packet) override {
       MutexLock lock(&test_mutex_);
       ++receiver_rtcp_;
       packet_event_.Set();
diff --git a/video/end_to_end_tests/retransmission_tests.cc b/video/end_to_end_tests/retransmission_tests.cc
index 3da96e6c49..343c4069af 100644
--- a/video/end_to_end_tests/retransmission_tests.cc
+++ b/video/end_to_end_tests/retransmission_tests.cc
@@ -12,10 +12,10 @@
 #include <cstdint>
 #include <optional>
 #include <set>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/call/transport.h"
 #include "api/environment/environment.h"
 #include "api/rtp_parameters.h"
@@ -80,7 +80,7 @@ TEST_F(RetransmissionEndToEndTest, ReceivesAndRetransmitsNack) {
           nacks_left_(kNumberOfNacksToObserve) {}
 
    private:
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       MutexLock lock(&mutex_);
       RtpPacket rtp_packet;
       EXPECT_TRUE(rtp_packet.Parse(packet));
@@ -117,7 +117,7 @@ TEST_F(RetransmissionEndToEndTest, ReceivesAndRetransmitsNack) {
       return SEND_PACKET;
     }
 
-    Action OnReceiveRtcp(ArrayView<const uint8_t> packet) override {
+    Action OnReceiveRtcp(std::span<const uint8_t> packet) override {
       MutexLock lock(&mutex_);
       test::RtcpPacketParser parser;
       EXPECT_TRUE(parser.Parse(packet));
@@ -165,7 +165,7 @@ TEST_F(RetransmissionEndToEndTest, ReceivesNackAndRetransmitsAudio) {
     size_t GetNumVideoStreams() const override { return 0; }
     size_t GetNumAudioStreams() const override { return 1; }
 
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       RtpPacket rtp_packet;
       EXPECT_TRUE(rtp_packet.Parse(packet));
 
@@ -232,7 +232,7 @@ TEST_F(RetransmissionEndToEndTest,
       receive_stream_ = receive_streams[0];
     }
 
-    Action OnReceiveRtcp(ArrayView<const uint8_t> packet) override {
+    Action OnReceiveRtcp(std::span<const uint8_t> packet) override {
       test::RtcpPacketParser parser;
       EXPECT_TRUE(parser.Parse(packet));
       if (parser.pli()->num_packets() > 0)
@@ -304,7 +304,7 @@ void RetransmissionEndToEndTest::ReceivesPliAndRecovers(int rtp_history_ms) {
           received_pli_(false) {}
 
    private:
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       MutexLock lock(&mutex_);
       RtpPacket rtp_packet;
       EXPECT_TRUE(rtp_packet.Parse(packet));
@@ -322,7 +322,7 @@ void RetransmissionEndToEndTest::ReceivesPliAndRecovers(int rtp_history_ms) {
       return SEND_PACKET;
     }
 
-    Action OnReceiveRtcp(ArrayView<const uint8_t> packet) override {
+    Action OnReceiveRtcp(std::span<const uint8_t> packet) override {
       MutexLock lock(&mutex_);
       test::RtcpPacketParser parser;
       EXPECT_TRUE(parser.Parse(packet));
@@ -400,7 +400,7 @@ void RetransmissionEndToEndTest::DecodesRetransmittedFrame(bool enable_rtx,
           retransmitted_timestamp_(0) {}
 
    private:
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       MutexLock lock(&mutex_);
       RtpPacket rtp_packet;
       EXPECT_TRUE(rtp_packet.Parse(packet));
diff --git a/video/end_to_end_tests/rtp_rtcp_tests.cc b/video/end_to_end_tests/rtp_rtcp_tests.cc
index 3407ba982a..f0685d4a37 100644
--- a/video/end_to_end_tests/rtp_rtcp_tests.cc
+++ b/video/end_to_end_tests/rtp_rtcp_tests.cc
@@ -15,9 +15,9 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/call/transport.h"
 #include "api/environment/environment.h"
 #include "api/field_trials_view.h"
@@ -74,7 +74,7 @@ void RtpRtcpEndToEndTest::RespectsRtcpMode(RtcpMode rtcp_mode) {
           sent_rtcp_(0) {}
 
    private:
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       MutexLock lock(&mutex_);
       if (++sent_rtp_ % 3 == 0)
         return DROP_PACKET;
@@ -82,7 +82,7 @@ void RtpRtcpEndToEndTest::RespectsRtcpMode(RtcpMode rtcp_mode) {
       return SEND_PACKET;
     }
 
-    Action OnReceiveRtcp(ArrayView<const uint8_t> packet) override {
+    Action OnReceiveRtcp(std::span<const uint8_t> packet) override {
       MutexLock lock(&mutex_);
       ++sent_rtcp_;
       test::RtcpPacketParser parser;
@@ -238,7 +238,7 @@ void RtpRtcpEndToEndTest::TestRtpStatePreservation(
       }
     }
 
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       RtpPacket rtp_packet;
       EXPECT_TRUE(rtp_packet.Parse(packet));
       const uint32_t ssrc = rtp_packet.Ssrc();
@@ -288,7 +288,7 @@ void RtpRtcpEndToEndTest::TestRtpStatePreservation(
       return SEND_PACKET;
     }
 
-    Action OnSendRtcp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtcp(std::span<const uint8_t> packet) override {
       test::RtcpPacketParser rtcp_parser;
       rtcp_parser.Parse(packet);
       if (rtcp_parser.sender_report()->num_packets() > 0) {
@@ -432,7 +432,7 @@ TEST_F(RtpRtcpEndToEndTest, DISABLED_TestFlexfecRtpStatePreservation) {
     }
 
    private:
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       MutexLock lock(&mutex_);
 
       RtpPacket rtp_packet;
diff --git a/video/end_to_end_tests/ssrc_tests.cc b/video/end_to_end_tests/ssrc_tests.cc
index 2f5d27b345..ddd1f89c53 100644
--- a/video/end_to_end_tests/ssrc_tests.cc
+++ b/video/end_to_end_tests/ssrc_tests.cc
@@ -12,10 +12,10 @@
 #include <cstdint>
 #include <map>
 #include <memory>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/rtp_parameters.h"
 #include "api/task_queue/task_queue_base.h"
 #include "api/test/simulated_network.h"
@@ -55,7 +55,7 @@ TEST_F(SsrcEndToEndTest, ReceiverUsesLocalSsrc) {
     SyncRtcpObserver()
         : EndToEndTest(test::VideoTestConstants::kDefaultTimeout) {}
 
-    Action OnReceiveRtcp(ArrayView<const uint8_t> packet) override {
+    Action OnReceiveRtcp(std::span<const uint8_t> packet) override {
       test::RtcpPacketParser parser;
       EXPECT_TRUE(parser.Parse(packet));
       EXPECT_EQ(test::VideoTestConstants::kReceiverLocalVideoSsrc,
@@ -177,7 +177,7 @@ void SsrcEndToEndTest::TestSendsSetSsrcs(size_t num_ssrcs,
     }
 
    private:
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       RtpPacket rtp_packet;
       EXPECT_TRUE(rtp_packet.Parse(packet));
 
@@ -285,7 +285,7 @@ TEST_F(SsrcEndToEndTest, DISABLED_RedundantPayloadsTransmittedOnAllSsrcs) {
     }
 
    private:
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       RtpPacket rtp_packet;
       EXPECT_TRUE(rtp_packet.Parse(packet));
 
diff --git a/video/end_to_end_tests/stats_tests.cc b/video/end_to_end_tests/stats_tests.cc
index 8afd8d3bd2..9c50e4bb43 100644
--- a/video/end_to_end_tests/stats_tests.cc
+++ b/video/end_to_end_tests/stats_tests.cc
@@ -14,12 +14,12 @@
 #include <memory>
 #include <optional>
 #include <set>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/algorithm/container.h"
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/rtp_headers.h"
 #include "api/rtp_parameters.h"
@@ -87,7 +87,7 @@ TEST_F(StatsEndToEndTest, GetStats) {
               }) {}
 
    private:
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       // Drop every 25th packet => 4% loss.
       static const int kPacketLossFrac = 25;
       RtpPacket header;
@@ -101,17 +101,17 @@ TEST_F(StatsEndToEndTest, GetStats) {
       return SEND_PACKET;
     }
 
-    Action OnSendRtcp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtcp(std::span<const uint8_t> packet) override {
       check_stats_event_.Set();
       return SEND_PACKET;
     }
 
-    Action OnReceiveRtp(ArrayView<const uint8_t> packet) override {
+    Action OnReceiveRtp(std::span<const uint8_t> packet) override {
       check_stats_event_.Set();
       return SEND_PACKET;
     }
 
-    Action OnReceiveRtcp(ArrayView<const uint8_t> packet) override {
+    Action OnReceiveRtcp(std::span<const uint8_t> packet) override {
       check_stats_event_.Set();
       return SEND_PACKET;
     }
@@ -458,7 +458,7 @@ TEST_F(StatsEndToEndTest, TestReceivedRtpPacketStats) {
 
     void OnStreamsStopped() override { task_safety_flag_->SetNotAlive(); }
 
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       if (sent_rtp_ >= kNumRtpPacketsToSend) {
         // Need to check the stats on the correct thread.
         task_queue_->PostTask(SafeTask(task_safety_flag_, [this]() {
@@ -517,7 +517,7 @@ TEST_F(StatsEndToEndTest, MAYBE_ContentTypeSwitches) {
       }
     }
 
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       if (MinNumberOfFramesReceived())
         observation_complete_.Set();
       return SEND_PACKET;
@@ -631,7 +631,7 @@ TEST_F(StatsEndToEndTest, VerifyNackStats) {
           task_queue_(task_queue) {}
 
    private:
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       {
         MutexLock lock(&mutex_);
         if (++sent_rtp_packets_ == kPacketNumberToDrop) {
@@ -646,7 +646,7 @@ TEST_F(StatsEndToEndTest, VerifyNackStats) {
       return SEND_PACKET;
     }
 
-    Action OnReceiveRtcp(ArrayView<const uint8_t> packet) override {
+    Action OnReceiveRtcp(std::span<const uint8_t> packet) override {
       MutexLock lock(&mutex_);
       test::RtcpPacketParser rtcp_parser;
       rtcp_parser.Parse(packet);
diff --git a/video/end_to_end_tests/transport_feedback_tests.cc b/video/end_to_end_tests/transport_feedback_tests.cc
index 9d807552a7..66e475d82a 100644
--- a/video/end_to_end_tests/transport_feedback_tests.cc
+++ b/video/end_to_end_tests/transport_feedback_tests.cc
@@ -13,9 +13,9 @@
 #include <map>
 #include <memory>
 #include <set>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/call/transport.h"
 #include "api/environment/environment.h"
 #include "api/rtp_parameters.h"
@@ -72,8 +72,8 @@ TEST(TransportFeedbackMultiStreamTest, AssignsTransportSequenceNumbers) {
         Call* sender_call,
         const std::map<uint32_t, uint32_t>& ssrc_map,
         const std::map<uint8_t, MediaType>& payload_type_map,
-        ArrayView<const RtpExtension> audio_extensions,
-        ArrayView<const RtpExtension> video_extensions)
+        std::span<const RtpExtension> audio_extensions,
+        std::span<const RtpExtension> video_extensions)
         : DirectTransport(env,
                           task_queue,
                           std::make_unique<FakeNetworkPipe>(
@@ -93,7 +93,7 @@ TEST(TransportFeedbackMultiStreamTest, AssignsTransportSequenceNumbers) {
     }
     ~RtpExtensionHeaderObserver() override {}
 
-    bool SendRtp(ArrayView<const uint8_t> data,
+    bool SendRtp(std::span<const uint8_t> data,
                  const PacketOptions& options) override {
       {
         MutexLock lock(&lock_);
@@ -285,18 +285,18 @@ class TransportFeedbackTester : public test::EndToEndTest {
   }
 
  protected:
-  Action OnSendRtcp(ArrayView<const uint8_t> data) override {
+  Action OnSendRtcp(std::span<const uint8_t> data) override {
     EXPECT_FALSE(HasTransportFeedback(data));
     return SEND_PACKET;
   }
 
-  Action OnReceiveRtcp(ArrayView<const uint8_t> data) override {
+  Action OnReceiveRtcp(std::span<const uint8_t> data) override {
     if (HasTransportFeedback(data))
       observation_complete_.Set();
     return SEND_PACKET;
   }
 
-  bool HasTransportFeedback(ArrayView<const uint8_t> data) const {
+  bool HasTransportFeedback(std::span<const uint8_t> data) const {
     test::RtcpPacketParser parser;
     EXPECT_TRUE(parser.Parse(data));
     return parser.transport_feedback()->num_packets() > 0;
@@ -362,7 +362,7 @@ TEST_F(TransportFeedbackEndToEndTest,
     }
 
    protected:
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       RtpPacket rtp_packet;
       EXPECT_TRUE(rtp_packet.Parse(packet));
       const bool only_padding = rtp_packet.payload_size() == 0;
@@ -385,7 +385,7 @@ TEST_F(TransportFeedbackEndToEndTest,
       return SEND_PACKET;
     }
 
-    Action OnReceiveRtcp(ArrayView<const uint8_t> data) override {
+    Action OnReceiveRtcp(std::span<const uint8_t> data) override {
       MutexLock lock(&mutex_);
       // To fill up the congestion window we drop feedback on packets after 20
       // packets have been sent. This means that any packets that has not yet
@@ -402,7 +402,7 @@ TEST_F(TransportFeedbackEndToEndTest,
       return SEND_PACKET;
     }
 
-    bool HasTransportFeedback(ArrayView<const uint8_t> data) const {
+    bool HasTransportFeedback(std::span<const uint8_t> data) const {
       test::RtcpPacketParser parser;
       EXPECT_TRUE(parser.Parse(data));
       return parser.transport_feedback()->num_packets() > 0;
@@ -456,7 +456,7 @@ TEST_F(TransportFeedbackEndToEndTest, TransportSeqNumOnAudioAndVideo) {
                        kTransportSequenceNumberExtensionId));
     }
 
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       RtpPacket rtp_packet(&extensions_);
       EXPECT_TRUE(rtp_packet.Parse(packet));
       uint16_t transport_sequence_number = 0;
diff --git a/video/picture_id_tests.cc b/video/picture_id_tests.cc
index 5db58a924d..0b29a3f7f1 100644
--- a/video/picture_id_tests.cc
+++ b/video/picture_id_tests.cc
@@ -14,11 +14,11 @@
 #include <memory>
 #include <optional>
 #include <set>
+#include <span>
 #include <string>
 #include <variant>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/test/simulated_network.h"
 #include "api/test/video/function_video_encoder_factory.h"
@@ -197,7 +197,7 @@ class PictureIdObserver : public test::RtpRtcpObserver {
     }
   }
 
-  Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+  Action OnSendRtp(std::span<const uint8_t> packet) override {
     MutexLock lock(&mutex_);
 
     ParsedPacket parsed;
diff --git a/video/quality_scaling_tests.cc b/video/quality_scaling_tests.cc
index cd7e2d7c15..895bfa0fac 100644
--- a/video/quality_scaling_tests.cc
+++ b/video/quality_scaling_tests.cc
@@ -14,10 +14,10 @@
 #include <limits>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/make_ref_counted.h"
 #include "api/rtp_parameters.h"
@@ -191,7 +191,7 @@ class ScalingObserver : public test::SendTest {
                        test_params_.size());
   }
 
-  Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+  Action OnSendRtp(std::span<const uint8_t> packet) override {
     // The tests are expected to send at the configured start bitrate. Do not
     // send any packets to avoid receiving REMB and possibly go down in target
     // bitrate. A low bitrate estimate could result in downgrading due to other
diff --git a/video/rtp_video_stream_receiver2.cc b/video/rtp_video_stream_receiver2.cc
index 753325675c..2a86c55e21 100644
--- a/video/rtp_video_stream_receiver2.cc
+++ b/video/rtp_video_stream_receiver2.cc
@@ -16,12 +16,12 @@
 #include <limits>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/algorithm/container.h"
-#include "api/array_view.h"
 #include "api/crypto/frame_decryptor_interface.h"
 #include "api/environment/environment.h"
 #include "api/field_trials_view.h"
@@ -719,7 +719,7 @@ bool RtpVideoStreamReceiver2::OnReceivedPayloadData(
       !UseH26xPacketBuffer(packet->codec())) {
     video_coding::H264SpsPpsTracker::FixedBitstream fixed =
         tracker_.CopyAndFixBitstream(
-            MakeArrayView(codec_payload.cdata(), codec_payload.size()),
+            std::span(codec_payload.cdata(), codec_payload.size()),
             &packet->video_header);
 
     switch (fixed.action) {
@@ -828,7 +828,7 @@ void RtpVideoStreamReceiver2::OnInsertedPacket(
   int64_t min_recv_time;
   int64_t max_recv_time;
   std::optional<int64_t> absolute_capture_time_ms;
-  std::vector<ArrayView<const uint8_t>> payloads;
+  std::vector<std::span<const uint8_t>> payloads;
   RtpPacketInfos::vector_type packet_infos;
 
   bool skip_frame = false;
@@ -1283,7 +1283,7 @@ void RtpVideoStreamReceiver2::NotifyReceiverOfEmptyPacket(
 }
 
 bool RtpVideoStreamReceiver2::DeliverRtcp(
-    ArrayView<const uint8_t> rtcp_packet) {
+    std::span<const uint8_t> rtcp_packet) {
   RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
 
   if (!receiving_) {
diff --git a/video/rtp_video_stream_receiver2.h b/video/rtp_video_stream_receiver2.h
index 2ca52615a7..3fc2febca6 100644
--- a/video/rtp_video_stream_receiver2.h
+++ b/video/rtp_video_stream_receiver2.h
@@ -17,9 +17,9 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/crypto/frame_decryptor_interface.h"
 #include "api/environment/environment.h"
 #include "api/frame_transformer_interface.h"
@@ -131,7 +131,7 @@ class RtpVideoStreamReceiver2 : public LossNotificationSender,
   // Produces the transport-related timestamps; current_delay_ms is left unset.
   std::optional<Syncable::Info> GetSyncInfo() const;
 
-  bool DeliverRtcp(ArrayView<const uint8_t> rtcp_packet);
+  bool DeliverRtcp(std::span<const uint8_t> rtcp_packet);
 
   void FrameContinuous(int64_t seq_num);
 
diff --git a/video/rtp_video_stream_receiver2_unittest.cc b/video/rtp_video_stream_receiver2_unittest.cc
index 962ad0fb15..e195ac9c7e 100644
--- a/video/rtp_video_stream_receiver2_unittest.cc
+++ b/video/rtp_video_stream_receiver2_unittest.cc
@@ -15,10 +15,10 @@
 #include <cstring>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/environment/environment_factory.h"
 #include "api/frame_transformer_interface.h"
@@ -155,7 +155,7 @@ class MockOnCompleteFrameCallback
   void ClearExpectedBitstream() { buffer_.Clear(); }
 
   void AppendExpectedBitstream(const uint8_t data[], size_t size_in_bytes) {
-    buffer_.Write(ArrayView<const uint8_t>(data, size_in_bytes));
+    buffer_.Write(std::span<const uint8_t>(data, size_in_bytes));
   }
   ByteBufferWriter buffer_;
 };
@@ -206,7 +206,7 @@ class RtpVideoStreamReceiver2Test : public ::testing::Test,
                                                 /*raw_payload=*/false);
     ON_CALL(mock_transport_, SendRtcp)
         .WillByDefault(
-            [this](ArrayView<const uint8_t> packet, ::testing::Unused) {
+            [this](std::span<const uint8_t> packet, ::testing::Unused) {
               return rtcp_packet_parser_.Parse(packet);
             });
   }
diff --git a/video/timing/simulator/BUILD.gn b/video/timing/simulator/BUILD.gn
index 8c5af06e95..57f1770481 100644
--- a/video/timing/simulator/BUILD.gn
+++ b/video/timing/simulator/BUILD.gn
@@ -35,7 +35,6 @@ if (rtc_enable_protobuf) {
       "stream_base.h",
     ]
     deps = [
-      "../../../api:array_view",
       "../../../api:field_trials",
       "../../../api:rtp_headers",
       "../../../api:rtp_packet_info",
diff --git a/video/timing/simulator/assembler.h b/video/timing/simulator/assembler.h
index 0f83fc058d..2e3fe110d7 100644
--- a/video/timing/simulator/assembler.h
+++ b/video/timing/simulator/assembler.h
@@ -13,10 +13,10 @@
 
 #include <cstdint>
 #include <memory>
+#include <span>
 
 #include "absl/base/nullability.h"
 #include "absl/container/flat_hash_set.h"
-#include "api/array_view.h"
 #include "api/call/transport.h"
 #include "api/environment/environment.h"
 #include "api/sequence_checker.h"
@@ -83,10 +83,10 @@ class Assembler : public ReceivedRtpPacketCallback,
  private:
   // Trivially implements `Transport`.
   // We need to implement this due to an RTC_DCHECK in rtcp_sender.cc.
-  bool SendRtp(ArrayView<const uint8_t>, const PacketOptions&) override {
+  bool SendRtp(std::span<const uint8_t>, const PacketOptions&) override {
     return true;
   }
-  bool SendRtcp(ArrayView<const uint8_t>, const PacketOptions&) override {
+  bool SendRtcp(std::span<const uint8_t>, const PacketOptions&) override {
     return true;
   }
 
diff --git a/video/timing/simulator/decodability_simulator.h b/video/timing/simulator/decodability_simulator.h
index 3aaabe9996..71b77d08b4 100644
--- a/video/timing/simulator/decodability_simulator.h
+++ b/video/timing/simulator/decodability_simulator.h
@@ -13,10 +13,10 @@
 
 #include <cstdint>
 #include <set>
+#include <span>
 #include <vector>
 
 #include "absl/algorithm/container.h"
-#include "api/array_view.h"
 #include "api/numerics/samples_stats_counter.h"
 #include "api/units/data_size.h"
 #include "api/units/time_delta.h"
@@ -115,7 +115,7 @@ inline bool DecodableOrder(const DecodabilitySimulator::Frame& a,
   return a.decodable_timestamp < b.decodable_timestamp;
 }
 inline void SortByDecodableOrder(
-    ArrayView<DecodabilitySimulator::Frame> frames) {
+    std::span<DecodabilitySimulator::Frame> frames) {
   absl::c_stable_sort(frames, DecodableOrder);
 }
 
diff --git a/video/timing/simulator/rendering_simulator.h b/video/timing/simulator/rendering_simulator.h
index aedb5508fa..5367e66bf0 100644
--- a/video/timing/simulator/rendering_simulator.h
+++ b/video/timing/simulator/rendering_simulator.h
@@ -16,11 +16,11 @@
 #include <memory>
 #include <optional>
 #include <set>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/algorithm/container.h"
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/numerics/samples_stats_counter.h"
 #include "api/units/data_size.h"
@@ -389,7 +389,7 @@ inline bool RenderOrder(const RenderingSimulator::Frame& a,
                         const RenderingSimulator::Frame& b) {
   return a.render_timestamp < b.render_timestamp;
 }
-inline void SortByRenderOrder(ArrayView<RenderingSimulator::Frame> frames) {
+inline void SortByRenderOrder(std::span<RenderingSimulator::Frame> frames) {
   absl::c_stable_sort(frames, RenderOrder);
 }
 
@@ -397,7 +397,7 @@ inline bool DecodedOrder(const RenderingSimulator::Frame& a,
                          const RenderingSimulator::Frame& b) {
   return a.decoded_timestamp < b.decoded_timestamp;
 }
-inline void SortByDecodedOrder(ArrayView<RenderingSimulator::Frame> frames) {
+inline void SortByDecodedOrder(std::span<RenderingSimulator::Frame> frames) {
   absl::c_stable_sort(frames, DecodedOrder);
 }
 
@@ -405,7 +405,7 @@ inline bool RenderedOrder(const RenderingSimulator::Frame& a,
                           const RenderingSimulator::Frame& b) {
   return a.rendered_timestamp < b.rendered_timestamp;
 }
-inline void SortByRenderedOrder(ArrayView<RenderingSimulator::Frame> frames) {
+inline void SortByRenderedOrder(std::span<RenderingSimulator::Frame> frames) {
   absl::c_stable_sort(frames, RenderedOrder);
 }
 
diff --git a/video/transport_adapter.cc b/video/transport_adapter.cc
index 95e430f00e..149d73830b 100644
--- a/video/transport_adapter.cc
+++ b/video/transport_adapter.cc
@@ -11,8 +11,8 @@
 #include "video/transport_adapter.h"
 
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/call/transport.h"
 #include "rtc_base/checks.h"
 
@@ -26,7 +26,7 @@ TransportAdapter::TransportAdapter(Transport* transport)
 
 TransportAdapter::~TransportAdapter() = default;
 
-bool TransportAdapter::SendRtp(ArrayView<const uint8_t> packet,
+bool TransportAdapter::SendRtp(std::span<const uint8_t> packet,
                                const PacketOptions& options) {
   if (!enabled_.load())
     return false;
@@ -34,7 +34,7 @@ bool TransportAdapter::SendRtp(ArrayView<const uint8_t> packet,
   return transport_->SendRtp(packet, options);
 }
 
-bool TransportAdapter::SendRtcp(ArrayView<const uint8_t> packet,
+bool TransportAdapter::SendRtcp(std::span<const uint8_t> packet,
                                 const PacketOptions& options) {
   if (!enabled_.load())
     return false;
diff --git a/video/transport_adapter.h b/video/transport_adapter.h
index 1752851500..1a3615ff18 100644
--- a/video/transport_adapter.h
+++ b/video/transport_adapter.h
@@ -14,8 +14,8 @@
 #include <stdint.h>
 
 #include <atomic>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/call/transport.h"
 
 namespace webrtc {
@@ -26,9 +26,9 @@ class TransportAdapter : public Transport {
   explicit TransportAdapter(Transport* transport);
   ~TransportAdapter() override;
 
-  bool SendRtp(ArrayView<const uint8_t> packet,
+  bool SendRtp(std::span<const uint8_t> packet,
                const PacketOptions& options) override;
-  bool SendRtcp(ArrayView<const uint8_t> packet,
+  bool SendRtcp(std::span<const uint8_t> packet,
                 const PacketOptions& options) override;
 
   void Enable();
diff --git a/video/video_analyzer.cc b/video/video_analyzer.cc
index 8aff4b24bb..cd8c2e6e92 100644
--- a/video/video_analyzer.cc
+++ b/video/video_analyzer.cc
@@ -16,13 +16,13 @@
 #include <cstdint>
 #include <cstdio>
 #include <cstring>
+#include <span>
 #include <string>
 #include <utility>
 
 #include "absl/algorithm/container.h"
 #include "absl/flags/flag.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/call/transport.h"
 #include "api/media_types.h"
 #include "api/numerics/samples_stats_counter.h"
@@ -303,7 +303,7 @@ void VideoAnalyzer::PostEncodeOnFrame(size_t stream_id, uint32_t timestamp) {
   }
 }
 
-bool VideoAnalyzer::SendRtp(ArrayView<const uint8_t> packet,
+bool VideoAnalyzer::SendRtp(std::span<const uint8_t> packet,
                             const PacketOptions& options) {
   RtpPacket rtp_packet;
   rtp_packet.Parse(packet);
@@ -336,7 +336,7 @@ bool VideoAnalyzer::SendRtp(ArrayView<const uint8_t> packet,
   return result;
 }
 
-bool VideoAnalyzer::SendRtcp(ArrayView<const uint8_t> packet,
+bool VideoAnalyzer::SendRtcp(std::span<const uint8_t> packet,
                              const PacketOptions& options) {
   return transport_->SendRtcp(packet, options);
 }
diff --git a/video/video_analyzer.h b/video/video_analyzer.h
index 7e8c41102a..160069de3d 100644
--- a/video/video_analyzer.h
+++ b/video/video_analyzer.h
@@ -17,11 +17,11 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/call/transport.h"
 #include "api/media_types.h"
 #include "api/numerics/samples_stats_counter.h"
@@ -95,10 +95,10 @@ class VideoAnalyzer : public PacketReceiver,
   void PreEncodeOnFrame(const VideoFrame& video_frame);
   void PostEncodeOnFrame(size_t stream_id, uint32_t timestamp);
 
-  bool SendRtp(ArrayView<const uint8_t> packet,
+  bool SendRtp(std::span<const uint8_t> packet,
                const PacketOptions& options) override;
 
-  bool SendRtcp(ArrayView<const uint8_t> packet,
+  bool SendRtcp(std::span<const uint8_t> packet,
                 const PacketOptions& options) override;
   void OnFrame(const VideoFrame& video_frame) override;
   void Wait();
diff --git a/video/video_receive_stream2.cc b/video/video_receive_stream2.cc
index 478b1af212..93ceb84ef3 100644
--- a/video/video_receive_stream2.cc
+++ b/video/video_receive_stream2.cc
@@ -18,13 +18,13 @@
 #include <memory>
 #include <optional>
 #include <set>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/algorithm/container.h"
 #include "absl/strings/str_cat.h"
-#include "api/array_view.h"
 #include "api/crypto/frame_decryptor_interface.h"
 #include "api/environment/environment.h"
 #include "api/field_trials_view.h"
@@ -349,7 +349,7 @@ void VideoReceiveStream2::SignalNetworkState(NetworkState state) {
   rtp_video_stream_receiver_.SignalNetworkState(state);
 }
 
-bool VideoReceiveStream2::DeliverRtcp(ArrayView<const uint8_t> packet) {
+bool VideoReceiveStream2::DeliverRtcp(std::span<const uint8_t> packet) {
   RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
   return rtp_video_stream_receiver_.DeliverRtcp(packet);
 }
diff --git a/video/video_receive_stream2.h b/video/video_receive_stream2.h
index 109886cd2a..c90527ed9c 100644
--- a/video/video_receive_stream2.h
+++ b/video/video_receive_stream2.h
@@ -17,10 +17,10 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/crypto/frame_decryptor_interface.h"
 #include "api/environment/environment.h"
 #include "api/frame_transformer_interface.h"
@@ -150,7 +150,7 @@ class VideoReceiveStream2
   }
 
   void SignalNetworkState(NetworkState state);
-  bool DeliverRtcp(ArrayView<const uint8_t> packet);
+  bool DeliverRtcp(std::span<const uint8_t> packet);
 
   void SetSync(Syncable* audio_syncable);
 
diff --git a/video/video_receive_stream2_unittest.cc b/video/video_receive_stream2_unittest.cc
index 1b86860b6a..a499b0e881 100644
--- a/video/video_receive_stream2_unittest.cc
+++ b/video/video_receive_stream2_unittest.cc
@@ -16,11 +16,11 @@
 #include <limits>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
 #include "absl/memory/memory.h"
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/environment/environment_factory.h"
 #include "api/metronome/test/fake_metronome.h"
@@ -212,7 +212,7 @@ class VideoReceiveStream2Test : public ::testing::TestWithParam<bool> {
         .WillByDefault(Invoke(&fake_decoder_, &test::FakeDecoder::Release));
     ON_CALL(mock_transport_, SendRtcp)
         .WillByDefault(
-            [this](ArrayView<const uint8_t> packet, ::testing::Unused) {
+            [this](std::span<const uint8_t> packet, ::testing::Unused) {
               return rtcp_packet_parser_.Parse(packet);
             });
   }
diff --git a/video/video_send_stream_impl.cc b/video/video_send_stream_impl.cc
index b62da58183..d6ff87ee85 100644
--- a/video/video_send_stream_impl.cc
+++ b/video/video_send_stream_impl.cc
@@ -15,13 +15,13 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/algorithm/container.h"
 #include "api/adaptation/resource.h"
-#include "api/array_view.h"
 #include "api/call/bitrate_allocation.h"
 #include "api/crypto/crypto_options.h"
 #include "api/environment/environment.h"
@@ -583,7 +583,7 @@ void VideoSendStreamImpl::SetStats(const Stats& stats) {
   stats_proxy_.SetStats(stats);
 }
 
-void VideoSendStreamImpl::SetCsrcs(ArrayView<const uint32_t> csrcs) {
+void VideoSendStreamImpl::SetCsrcs(std::span<const uint32_t> csrcs) {
   RTC_DCHECK_RUN_ON(&thread_checker_);
   rtp_video_sender_->SetCsrcs(csrcs);
 }
@@ -630,7 +630,7 @@ void VideoSendStreamImpl::GenerateKeyFrame(
   }
 }
 
-void VideoSendStreamImpl::DeliverRtcp(ArrayView<const uint8_t> packet) {
+void VideoSendStreamImpl::DeliverRtcp(std::span<const uint8_t> packet) {
   RTC_DCHECK_RUN_ON(&thread_checker_);
   rtp_video_sender_->DeliverRtcp(packet);
 }
diff --git a/video/video_send_stream_impl.h b/video/video_send_stream_impl.h
index c78994f799..9ed23081fa 100644
--- a/video/video_send_stream_impl.h
+++ b/video/video_send_stream_impl.h
@@ -17,11 +17,11 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "api/adaptation/resource.h"
-#include "api/array_view.h"
 #include "api/call/bitrate_allocation.h"
 #include "api/environment/environment.h"
 #include "api/fec_controller.h"
@@ -104,7 +104,7 @@ class VideoSendStreamImpl : public webrtc::VideoSendStream,
                           video_stream_encoder_for_test = nullptr);
   ~VideoSendStreamImpl() override;
 
-  void DeliverRtcp(ArrayView<const uint8_t> packet);
+  void DeliverRtcp(std::span<const uint8_t> packet);
 
   // webrtc::VideoSendStream implementation.
   void Start() override;
@@ -123,7 +123,7 @@ class VideoSendStreamImpl : public webrtc::VideoSendStream,
   Stats GetStats() override;
   void SetStats(const Stats& stats) override;
 
-  void SetCsrcs(ArrayView<const uint32_t> csrcs) override;
+  void SetCsrcs(std::span<const uint32_t> csrcs) override;
 
   void StopPermanentlyAndGetRtpStates(RtpStateMap* rtp_state_map,
                                       RtpPayloadStateMap* payload_state_map);
diff --git a/video/video_send_stream_impl_unittest.cc b/video/video_send_stream_impl_unittest.cc
index 673da8d073..1c300242e5 100644
--- a/video/video_send_stream_impl_unittest.cc
+++ b/video/video_send_stream_impl_unittest.cc
@@ -16,12 +16,12 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/call/bitrate_allocation.h"
 #include "api/environment/environment_factory.h"
 #include "api/field_trials.h"
@@ -102,7 +102,7 @@ class MockRtpVideoSender : public RtpVideoSenderInterface {
               GetRtpPayloadStates,
               (),
               (const, override));
-  MOCK_METHOD(void, DeliverRtcp, (ArrayView<const uint8_t>), (override));
+  MOCK_METHOD(void, DeliverRtcp, (std::span<const uint8_t>), (override));
   MOCK_METHOD(void,
               OnBitrateAllocationUpdated,
               (const VideoBitrateAllocation&),
@@ -123,10 +123,10 @@ class MockRtpVideoSender : public RtpVideoSenderInterface {
   MOCK_METHOD(uint32_t, GetPayloadBitrateBps, (), (const, override));
   MOCK_METHOD(uint32_t, GetProtectionBitrateBps, (), (const, override));
   MOCK_METHOD(void, SetEncodingData, (size_t, size_t, size_t), (override));
-  MOCK_METHOD(void, SetCsrcs, (ArrayView<const uint32_t> csrcs), (override));
+  MOCK_METHOD(void, SetCsrcs, (std::span<const uint32_t> csrcs), (override));
   MOCK_METHOD(std::vector<RtpSequenceNumberMap::Info>,
               GetSentRtpPacketInfos,
-              (uint32_t ssrc, ArrayView<const uint16_t> sequence_numbers),
+              (uint32_t ssrc, std::span<const uint16_t> sequence_numbers),
               (const, override));
 
   MOCK_METHOD(void, SetFecAllowed, (bool fec_allowed), (override));
diff --git a/video/video_send_stream_tests.cc b/video/video_send_stream_tests.cc
index 3873bb360d..86366875e7 100644
--- a/video/video_send_stream_tests.cc
+++ b/video/video_send_stream_tests.cc
@@ -15,6 +15,7 @@
 #include <memory>
 #include <numeric>
 #include <optional>
+#include <span>
 #include <string>
 #include <tuple>
 #include <utility>
@@ -23,7 +24,6 @@
 
 #include "absl/strings/match.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/environment/environment_factory.h"
 #include "api/fec_controller_override.h"
@@ -239,7 +239,7 @@ TEST_F(VideoSendStreamTest, SupportsCName) {
     CNameObserver() : SendTest(test::VideoTestConstants::kDefaultTimeout) {}
 
    private:
-    Action OnSendRtcp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtcp(std::span<const uint8_t> packet) override {
       test::RtcpPacketParser parser;
       EXPECT_TRUE(parser.Parse(packet));
       if (parser.sdes()->num_packets() > 0) {
@@ -275,7 +275,7 @@ TEST_F(VideoSendStreamTest, SupportsAbsoluteSendTime) {
       extensions_.Register<AbsoluteSendTime>(kAbsSendTimeExtensionId);
     }
 
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       RtpPacket rtp_packet(&extensions_);
       EXPECT_TRUE(rtp_packet.Parse(packet));
 
@@ -331,7 +331,7 @@ TEST_F(VideoSendStreamTest, SupportsTransmissionTimeOffset) {
     }
 
    private:
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       RtpPacket rtp_packet(&extensions_);
       EXPECT_TRUE(rtp_packet.Parse(packet));
 
@@ -379,7 +379,7 @@ TEST_F(VideoSendStreamTest, SupportsTransportWideSequenceNumbers) {
     }
 
    private:
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       RtpPacket rtp_packet(&extensions_);
       EXPECT_TRUE(rtp_packet.Parse(packet));
 
@@ -418,7 +418,7 @@ TEST_F(VideoSendStreamTest, SupportsVideoRotation) {
       extensions_.Register<VideoOrientation>(kVideoRotationExtensionId);
     }
 
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       RtpPacket rtp_packet(&extensions_);
       EXPECT_TRUE(rtp_packet.Parse(packet));
       // Only the last packet of the frame is required to have the extension.
@@ -464,7 +464,7 @@ TEST_F(VideoSendStreamTest, SupportsVideoContentType) {
           kVideoContentTypeExtensionId);
     }
 
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       RtpPacket rtp_packet(&extensions_);
       EXPECT_TRUE(rtp_packet.Parse(packet));
       // Only the last packet of the key-frame must have extension.
@@ -510,7 +510,7 @@ TEST_F(VideoSendStreamTest, SupportsVideoTimingFrames) {
       extensions_.Register<VideoTimingExtension>(kVideoTimingExtensionId);
     }
 
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       RtpPacket rtp_packet(&extensions_);
       EXPECT_TRUE(rtp_packet.Parse(packet));
       // Only the last packet of the frame must have extension.
@@ -591,7 +591,7 @@ class UlpfecObserver : public test::EndToEndTest {
   }
 
  private:
-  Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+  Action OnSendRtp(std::span<const uint8_t> packet) override {
     RtpPacket rtp_packet(&extensions_);
     EXPECT_TRUE(rtp_packet.Parse(packet));
 
@@ -829,7 +829,7 @@ class FlexfecObserver : public test::EndToEndTest {
   size_t GetNumVideoStreams() const override { return num_video_streams_; }
 
  private:
-  Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+  Action OnSendRtp(std::span<const uint8_t> packet) override {
     RtpPacket rtp_packet(&extensions_);
     EXPECT_TRUE(rtp_packet.Parse(packet));
 
@@ -1016,7 +1016,7 @@ void VideoSendStreamTest::TestNackRetransmission(
           retransmit_payload_type_(retransmit_payload_type) {}
 
    private:
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       RtpPacket rtp_packet;
       EXPECT_TRUE(rtp_packet.Parse(packet));
 
@@ -1025,7 +1025,7 @@ void VideoSendStreamTest::TestNackRetransmission(
           rtp_packet.Ssrc() == retransmit_ssrc_ &&
           retransmit_ssrc_ != media_ssrc_) {
         // Assume correct RTX packet. Extract original sequence number.
-        ArrayView<const uint8_t> payload = rtp_packet.payload();
+        std::span<const uint8_t> payload = rtp_packet.payload();
         sequence_number = (payload[0] << 8) + payload[1];
       }
 
@@ -1183,7 +1183,7 @@ void VideoSendStreamTest::TestPacketFragmentationSize(TestVideoFormat format,
     }
 
    private:
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       size_t length = packet.size();
       RtpPacket rtp_packet;
       EXPECT_TRUE(rtp_packet.Parse(packet));
@@ -1406,7 +1406,7 @@ TEST_F(VideoSendStreamTest, NoPaddingWhenVideoIsMuted) {
           capturer_(nullptr) {}
 
    private:
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       MutexLock lock(&mutex_);
       last_packet_time_ms_ = clock_->TimeInMilliseconds();
 
@@ -1434,7 +1434,7 @@ TEST_F(VideoSendStreamTest, NoPaddingWhenVideoIsMuted) {
       return SEND_PACKET;
     }
 
-    Action OnSendRtcp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtcp(std::span<const uint8_t> packet) override {
       MutexLock lock(&mutex_);
       const int kNoPacketsThresholdMs = 2000;
       if (test_state_ == kWaitingForNoPackets &&
@@ -1501,7 +1501,7 @@ TEST_F(VideoSendStreamTest, PaddingIsPrimarilyRetransmissions) {
       call_ = sender_call;
     }
 
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       MutexLock lock(&mutex_);
 
       RtpPacket rtp_packet;
@@ -1577,7 +1577,7 @@ TEST_F(VideoSendStreamTest, MinTransmitBitrateRespectsRemb) {
           task_safety_flag_(PendingTaskSafetyFlag::CreateDetached()) {}
 
    private:
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       if (IsRtcpPacket(packet))
         return DROP_PACKET;
 
@@ -1701,7 +1701,7 @@ TEST_F(VideoSendStreamTest, ChangingNetworkRoute) {
           RtpExtension::kTransportSequenceNumberUri, kExtensionId));
     }
 
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       RTC_DCHECK_RUN_ON(&module_process_thread_);
       task_queue_->PostTask([this]() {
         RTC_DCHECK_RUN_ON(&task_queue_thread_);
@@ -1801,7 +1801,7 @@ TEST_F(VideoSendStreamTest, DISABLED_RelayToDirectRoute) {
       call_ = sender_call;
     }
 
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       RTC_DCHECK_RUN_ON(&module_process_thread_);
       task_queue_->PostTask([this]() {
         RTC_DCHECK_RUN_ON(&task_queue_thread_);
@@ -1885,7 +1885,7 @@ TEST_F(VideoSendStreamTest, ChangingTransportOverhead) {
       call_ = sender_call;
     }
 
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       EXPECT_LE(packet.size(), kMaxRtpPacketSize);
       MutexLock lock(&lock_);
       if (++packets_sent_ < 100)
@@ -1993,7 +1993,7 @@ class MaxPaddingSetTest : public test::SendTest {
   }
 
   // Called on the pacer thread.
-  Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+  Action OnSendRtp(std::span<const uint8_t> packet) override {
     RTC_DCHECK_RUN_ON(&module_process_thread_);
 
     // Check the stats on the correct thread and signal the 'complete' flag
@@ -2610,7 +2610,7 @@ TEST_F(VideoSendStreamTest, RtcpSenderReportContainsMediaBytesSent) {
           media_bytes_sent_(0) {}
 
    private:
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       MutexLock lock(&mutex_);
       RtpPacket rtp_packet;
       EXPECT_TRUE(rtp_packet.Parse(packet));
@@ -2619,7 +2619,7 @@ TEST_F(VideoSendStreamTest, RtcpSenderReportContainsMediaBytesSent) {
       return SEND_PACKET;
     }
 
-    Action OnSendRtcp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtcp(std::span<const uint8_t> packet) override {
       MutexLock lock(&mutex_);
       test::RtcpPacketParser parser;
       EXPECT_TRUE(parser.Parse(packet));
@@ -3082,12 +3082,12 @@ class Vp9HeaderObserver : public test::SendTest {
     }
   }
 
-  Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+  Action OnSendRtp(std::span<const uint8_t> packet) override {
     RtpPacket rtp_packet;
     EXPECT_TRUE(rtp_packet.Parse(packet));
 
     EXPECT_EQ(kVp9PayloadType, rtp_packet.PayloadType());
-    ArrayView<const uint8_t> rtp_payload = rtp_packet.payload();
+    std::span<const uint8_t> rtp_payload = rtp_packet.payload();
 
     bool new_packet = !last_packet_sequence_number_.has_value() ||
                       IsNewerSequenceNumber(rtp_packet.SequenceNumber(),
@@ -3794,7 +3794,7 @@ TEST_F(VideoSendStreamTest, RemoveOverheadFromBandwidth) {
       EXPECT_FALSE(send_config->rtp.extensions.empty());
     }
 
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       MutexLock lock(&mutex_);
       first_packet_sent_ = true;
       return SEND_PACKET;
@@ -3977,7 +3977,7 @@ class ContentSwitchTest : public test::SendTest {
     done_ = true;
   }
 
-  Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+  Action OnSendRtp(std::span<const uint8_t> packet) override {
     task_queue_->PostTask([this]() {
       MutexLock lock(&mutex_);
       if (done_)
@@ -4169,7 +4169,7 @@ void VideoSendStreamTest::TestTemporalLayers(
       int temporal_idx;
     };
 
-    bool ParsePayload(ArrayView<const uint8_t> packet,
+    bool ParsePayload(std::span<const uint8_t> packet,
                       ParsedPacket& parsed) const {
       RtpPacket rtp_packet;
       EXPECT_TRUE(rtp_packet.Parse(packet));
@@ -4196,7 +4196,7 @@ void VideoSendStreamTest::TestTemporalLayers(
       return true;
     }
 
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       ParsedPacket parsed;
       if (!ParsePayload(packet, parsed))
         return SEND_PACKET;
diff --git a/video/video_stream_encoder_unittest.cc b/video/video_stream_encoder_unittest.cc
index f41cfeaca6..5a3eba7de6 100644
--- a/video/video_stream_encoder_unittest.cc
+++ b/video/video_stream_encoder_unittest.cc
@@ -16,6 +16,7 @@
 #include <limits>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
@@ -25,7 +26,6 @@
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
 #include "api/adaptation/resource.h"
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/environment/environment_factory.h"
 #include "api/field_trials.h"
@@ -255,7 +255,7 @@ class FakeNV12NativeBuffer : public VideoFrameBuffer {
     return nv12_buffer_->ToI420();
   }
   scoped_refptr<VideoFrameBuffer> GetMappedFrameBuffer(
-      ArrayView<VideoFrameBuffer::Type> types) override {
+      std::span<VideoFrameBuffer::Type> types) override {
     if (absl::c_find(types, Type::kNV12) != types.end()) {
       return nv12_buffer_;
     }

From cbe3cb16172685dd7f9b38f1c91cd7ebb605f37a Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Thu, 5 Mar 2026 06:13:37 -0800
Subject: [PATCH 208/472] Roll chromium_revision 9d278b8335..44cac2a0af
 (1594505:1594634)

Change log: https://chromium.googlesource.com/chromium/src/+log/9d278b8335..44cac2a0af
Full diff: https://chromium.googlesource.com/chromium/src/+/9d278b8335..44cac2a0af

Changed dependencies
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/2b015a25fa..add05663e6
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/7d23eff99a..6a91250a4b
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/aa674c67b9..db2dce87ec
* src/third_party/androidx/cipd: 4fkujxcHswgZM_Abba388nGkoYd7fgfTF_0jCmWzodMC..1-25o844ZqTB3NxqGHhgpHC_CG6J-Yod2Jj7VsVYTDQC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/745d3ebf40..2127bd31b6
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/eea4d31f58..265496e862
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/f6077f1db9..02364664bb
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/1ddf8dac42..aca79bce63
DEPS diff: https://chromium.googlesource.com/chromium/src/+/9d278b8335..44cac2a0af/DEPS

BUG=None

Change-Id: I32e5185cb3e3e4f31f29e8cb347d933ff996153f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453880
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47061}
---
 DEPS | 18 +++++++++---------
 1 file changed, 9 insertions(+), 9 deletions(-)

diff --git a/DEPS b/DEPS
index 8ff1c56eff..62c303782d 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '9d278b833580deddd27a32636a6f56ccf5c17611',
+  'chromium_revision': '44cac2a0af0b07a8b5a0ea81b9af2f69964f443f',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@2b015a25fa7e9b762b5863e8616fad4b0b92f093',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@add05663e6258b17026b5fa488ba9d96d961c788',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@7d23eff99a735cea4f99711d0ddde40e714af6df',
+    'https://chromium.googlesource.com/chromium/src/testing@6a91250a4bfd191c04b3f42357f76f59ed78447f',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@aa674c67b9b66dc8c48d86d609f681140bbe11d7',
+    'https://chromium.googlesource.com/chromium/src/third_party@db2dce87eca65ce0b769a2440e2292cb2c57be27',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -488,7 +488,7 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@745d3ebf4064a4ea7c0a115ecbba5affa2609bcd',
+    'https://boringssl.googlesource.com/boringssl.git@2127bd31b60ecc072559f5fa4cca027eec3a3530',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@8be0e3114685fcc1589561067282edf75ea1259a',
   'src/third_party/catapult':
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'f6077f1db9bc56d81cdd1874e2310ff21438060c',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '02364664bb5dfa6c04d4b274aa818f089aa8e42e',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@1ddf8dac421f5a8acdb162781e21723e597692fd',
+    'https://chromium.googlesource.com/chromium/src/tools@aca79bce63b7bcc68f33310f7c48c6caa6f1af9a',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': '4fkujxcHswgZM_Abba388nGkoYd7fgfTF_0jCmWzodMC',
+          'version': '1-25o844ZqTB3NxqGHhgpHC_CG6J-Yod2Jj7VsVYTDQC',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -864,7 +864,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7364b490b5f78d58efe23ea76e74210fd6c3c76f',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + 'eea4d31f58572591bdaa3b654227e48efb495b96',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '265496e862d809be98b002dea70acf1a266aa1a0',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From a04d42299eb90bd50278c27b70e97624fe62f9f3 Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Wed, 4 Mar 2026 10:26:10 +0100
Subject: [PATCH 209/472] Delete deprecated variant of
 IceControllerInterface::connections

Make non-deprecated variant `GetConnections` pure virtual.

Bug: webrtc:42226052
Change-Id: I08172f9eefef3019d3584615a78e3e0e07fdbf18
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453580
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47062}
---
 p2p/base/basic_ice_controller.h     |  5 -----
 p2p/base/ice_controller_interface.h | 13 +------------
 p2p/test/mock_ice_controller.h      |  1 -
 3 files changed, 1 insertion(+), 18 deletions(-)

diff --git a/p2p/base/basic_ice_controller.h b/p2p/base/basic_ice_controller.h
index 35083c45ef..9e32c5b080 100644
--- a/p2p/base/basic_ice_controller.h
+++ b/p2p/base/basic_ice_controller.h
@@ -48,11 +48,6 @@ class BasicIceController : public IceControllerInterface {
   ArrayView<const Connection* const> GetConnections() const override {
     return connections_;
   }
-  ArrayView<const Connection*> connections() const override {
-    return ArrayView<const Connection*>(
-        const_cast<const Connection**>(connections_.data()),
-        connections_.size());
-  }
 
   bool HasPingableConnection() const override;
 
diff --git a/p2p/base/ice_controller_interface.h b/p2p/base/ice_controller_interface.h
index 53bf509b04..90e23a97fe 100644
--- a/p2p/base/ice_controller_interface.h
+++ b/p2p/base/ice_controller_interface.h
@@ -22,7 +22,6 @@
 #include "p2p/base/ice_switch_reason.h"
 #include "p2p/base/ice_transport_internal.h"
 #include "p2p/base/transport_description.h"
-#include "rtc_base/checks.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
@@ -112,17 +111,7 @@ class IceControllerInterface {
   virtual void OnConnectionDestroyed(const Connection* connection) = 0;
 
   // These are all connections that has been added and not destroyed.
-  virtual ArrayView<const Connection* const> GetConnections() const {
-    // Stub implementation to simplify downstream roll.
-    RTC_CHECK_NOTREACHED();
-    return {};
-  }
-  // TODO(bugs.webrtc.org/15702): Remove this after downstream is cleaned up.
-  virtual ArrayView<const Connection*> connections() const {
-    // Stub implementation to simplify downstream removal.
-    RTC_CHECK_NOTREACHED();
-    return {};
-  }
+  virtual ArrayView<const Connection* const> GetConnections() const = 0;
 
   // Is there a pingable connection ?
   // This function is used to boot-strap pinging, after this returns true
diff --git a/p2p/test/mock_ice_controller.h b/p2p/test/mock_ice_controller.h
index 5eb3bf5cfa..f654688247 100644
--- a/p2p/test/mock_ice_controller.h
+++ b/p2p/test/mock_ice_controller.h
@@ -40,7 +40,6 @@ class MockIceController : public IceControllerInterface {
               GetConnections,
               (),
               (const, override));
-  MOCK_METHOD(ArrayView<const Connection*>, connections, (), (const, override));
   MOCK_METHOD(bool, HasPingableConnection, (), (const, override));
   MOCK_METHOD(PingResult, GetConnectionToPing, (Timestamp), (override));
   MOCK_METHOD(bool,

From e3798d21b8ebab896d888a7097e92f3eb32d31d9 Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Mon, 2 Mar 2026 17:13:11 +0100
Subject: [PATCH 210/472] Use clock from webrtc::Environment in java
 TimestamplAligner
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Move function to query webrtc time from static function in
TimestampAligner to local in Environment

Bug: webrtc:42223992
Change-Id: I59cea50399cb873b875a5e9efafbc2ea58af35f1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452943
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Zoé Lepaul <zlep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47063}
---
 sdk/android/BUILD.gn                          |  1 +
 sdk/android/api/org/webrtc/Environment.java   | 10 ++++-
 .../api/org/webrtc/TimestampAligner.java      | 26 ++++++++----
 sdk/android/src/jni/environment.cc            |  4 ++
 sdk/android/src/jni/timestamp_aligner.cc      | 41 +++++++++++++------
 5 files changed, 61 insertions(+), 21 deletions(-)

diff --git a/sdk/android/BUILD.gn b/sdk/android/BUILD.gn
index 35d1ea4176..bcaddfae3a 100644
--- a/sdk/android/BUILD.gn
+++ b/sdk/android/BUILD.gn
@@ -272,6 +272,7 @@ if (is_android) {
 
     deps = [
       ":base_java",
+      ":environment_java",
       ":generated_video_egl_jni_java",
       ":generated_video_jni_java",
       ":video_api_java",
diff --git a/sdk/android/api/org/webrtc/Environment.java b/sdk/android/api/org/webrtc/Environment.java
index 10749ec4e2..5808776d01 100644
--- a/sdk/android/api/org/webrtc/Environment.java
+++ b/sdk/android/api/org/webrtc/Environment.java
@@ -39,6 +39,14 @@ public final class Environment implements AutoCloseable {
     return webrtcEnv;
   }
 
+  /**
+   * Returns current time as controlled by webrtcEnv. By default this is
+   * normally same as System.nanoTime(), but call this function to be safe.
+   */
+  public long getCurrentTimeNanos() {
+    return nativeCurrentTimeNanos(webrtcEnv);
+  }
+
   @Override
   public void close() {
     nativeFree(webrtcEnv);
@@ -49,6 +57,6 @@ public final class Environment implements AutoCloseable {
   }
 
   private static native long nativeCreate(@Nullable String fieldTrials);
-
+  private static native long nativeCurrentTimeNanos(long webrtcEnv);
   private static native void nativeFree(long webrtcEnv);
 }
diff --git a/sdk/android/api/org/webrtc/TimestampAligner.java b/sdk/android/api/org/webrtc/TimestampAligner.java
index 37c92b31e1..a93f07c37f 100644
--- a/sdk/android/api/org/webrtc/TimestampAligner.java
+++ b/sdk/android/api/org/webrtc/TimestampAligner.java
@@ -12,22 +12,33 @@ package org.webrtc;
 
 /**
  * The TimestampAligner class helps translating camera timestamps into the same timescale as is
- * used by webrtc::TimeNanos(). Some cameras have built in timestamping which is more accurate than
+ * used by webrtc. Some cameras have built in timestamping which is more accurate than
  * reading the system clock, but using a different epoch and unknown clock drift. Frame timestamps
- * in webrtc should use webrtc::TimeNanos (system monotonic time), and this class provides a filter
- * which lets us use the webrtc::TimeNanos timescale, and at the same time take advantage of higher
- * accuracy of the camera clock. This class is a wrapper on top of webrtc::TimestampAligner.
+ * in webrtc should use clock from the webrtc::Environment (usually the system monotonic time), and
+ * this class provides a filter which lets us use the webrtc::Environment timescale, and at the same
+ * time take advantage of higheraccuracy of the camera clock.
+ * This class is a wrapper on top of webrtc::TimestampAligner.
  */
 public class TimestampAligner {
   /**
    * Wrapper around webrtc::TimeNanos(). This is normally same as System.nanoTime(), but call this
    * function to be safe.
    */
+  @Deprecated
   public static long getRtcTimeNanos() {
-    return nativeRtcTimeNanos();
+    return Environment.builder().build().getCurrentTimeNanos();
   }
 
-  private volatile long nativeTimestampAligner = nativeCreateTimestampAligner();
+  private volatile long nativeTimestampAligner;
+
+  @Deprecated
+  public TimestampAligner() {
+    this(Environment.builder().build());
+  }
+
+  public TimestampAligner(Environment webrtcEnv) {
+    nativeTimestampAligner = nativeCreateTimestampAligner(webrtcEnv.ref());
+  }
 
   /**
    * Translates camera timestamps to the same timescale as is used by webrtc::TimeNanos().
@@ -52,8 +63,7 @@ public class TimestampAligner {
     }
   }
 
-  private static native long nativeRtcTimeNanos();
-  private static native long nativeCreateTimestampAligner();
+  private static native long nativeCreateTimestampAligner(long webrtcEnvRef);
   private static native void nativeReleaseTimestampAligner(long timestampAligner);
   private static native long nativeTranslateTimestamp(long timestampAligner, long cameraTimeNs);
 }
diff --git a/sdk/android/src/jni/environment.cc b/sdk/android/src/jni/environment.cc
index a93afe7bb0..bf751e47d1 100644
--- a/sdk/android/src/jni/environment.cc
+++ b/sdk/android/src/jni/environment.cc
@@ -27,6 +27,10 @@ void JNI_Environment_Free(JNIEnv* /*env*/, jlong webrtcEnv) {
   delete reinterpret_cast<Environment*>(webrtcEnv);
 }
 
+jlong JNI_Environment_CurrentTimeNanos(JNIEnv* env, jlong webrtcEnv) {
+  return reinterpret_cast<Environment*>(webrtcEnv)->clock().CurrentTime().ns();
+}
+
 jlong JNI_Environment_Create(JNIEnv* env,
                              const jni_zero::JavaRef<jstring>& fieldTrials) {
   std::unique_ptr<FieldTrialsView> field_trials;
diff --git a/sdk/android/src/jni/timestamp_aligner.cc b/sdk/android/src/jni/timestamp_aligner.cc
index 7226886cb0..12fbd0052f 100644
--- a/sdk/android/src/jni/timestamp_aligner.cc
+++ b/sdk/android/src/jni/timestamp_aligner.cc
@@ -12,34 +12,51 @@
 
 #include <jni.h>
 
-#include "rtc_base/time_utils.h"
+#include <cstdint>
+
+#include "api/environment/environment.h"
 #include "sdk/android/generated_video_jni/TimestampAligner_jni.h"
-#include "sdk/android/src/jni/jni_helpers.h"
+#include "sdk/android/native_api/jni/java_types.h"
 
 namespace webrtc {
 namespace jni {
+namespace {
 
-static jlong JNI_TimestampAligner_RtcTimeNanos(JNIEnv* env) {
-  return TimeNanos();
-}
+class TimestampAlignerWithClock {
+ public:
+  explicit TimestampAlignerWithClock(const Environment& env) : env_(env) {}
 
-static jlong JNI_TimestampAligner_CreateTimestampAligner(JNIEnv* env) {
-  return jlongFromPointer(new TimestampAligner());
+  int64_t TranslateTimestampNs(int64_t camera_time_ns) {
+    return aligner_.TranslateTimestamp(
+               /*capture_time_us=*/camera_time_ns / 1'000,
+               /*system_time_us=*/env_.clock().TimeInMicroseconds()) *
+           1'000;
+  }
+
+ private:
+  const Environment env_;
+  TimestampAligner aligner_;
+};
+
+}  // namespace
+
+static jlong JNI_TimestampAligner_CreateTimestampAligner(JNIEnv* env,
+                                                         jlong webrtcEnvRef) {
+  return NativeToJavaPointer(new TimestampAlignerWithClock(
+      *reinterpret_cast<const Environment*>(webrtcEnvRef)));
 }
 
 static void JNI_TimestampAligner_ReleaseTimestampAligner(
     JNIEnv* env,
     jlong timestamp_aligner) {
-  delete reinterpret_cast<TimestampAligner*>(timestamp_aligner);
+  delete reinterpret_cast<TimestampAlignerWithClock*>(timestamp_aligner);
 }
 
 static jlong JNI_TimestampAligner_TranslateTimestamp(JNIEnv* env,
                                                      jlong timestamp_aligner,
                                                      jlong camera_time_ns) {
-  return reinterpret_cast<TimestampAligner*>(timestamp_aligner)
-             ->TranslateTimestamp(camera_time_ns / kNumNanosecsPerMicrosec,
-                                  TimeMicros()) *
-         kNumNanosecsPerMicrosec;
+  return reinterpret_cast<TimestampAlignerWithClock*>(timestamp_aligner)
+      ->TranslateTimestampNs(camera_time_ns);
 }
 
 }  // namespace jni

From 59486f6d13a911b7f751f8c19f91ad85eeabb9ba Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Thu, 5 Mar 2026 11:34:59 -0800
Subject: [PATCH 211/472] Revert "Use clock from webrtc::Environment in java
 TimestamplAligner"
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This reverts commit e3798d21b8ebab896d888a7097e92f3eb32d31d9.

Reason for revert: breaks downstream

Original change's description:
> Use clock from webrtc::Environment in java TimestamplAligner
>
> Move function to query webrtc time from static function in
> TimestampAligner to local in Environment
>
> Bug: webrtc:42223992
> Change-Id: I59cea50399cb873b875a5e9efafbc2ea58af35f1
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452943
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Zoé Lepaul <zlep@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#47063}

Bug: webrtc:42223992
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I878ee497f63ee8f61ab2f9c259e36fe1950f4c6b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453940
Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47064}
---
 sdk/android/BUILD.gn                          |  1 -
 sdk/android/api/org/webrtc/Environment.java   | 10 +----
 .../api/org/webrtc/TimestampAligner.java      | 26 ++++--------
 sdk/android/src/jni/environment.cc            |  4 --
 sdk/android/src/jni/timestamp_aligner.cc      | 41 ++++++-------------
 5 files changed, 21 insertions(+), 61 deletions(-)

diff --git a/sdk/android/BUILD.gn b/sdk/android/BUILD.gn
index bcaddfae3a..35d1ea4176 100644
--- a/sdk/android/BUILD.gn
+++ b/sdk/android/BUILD.gn
@@ -272,7 +272,6 @@ if (is_android) {
 
     deps = [
       ":base_java",
-      ":environment_java",
       ":generated_video_egl_jni_java",
       ":generated_video_jni_java",
       ":video_api_java",
diff --git a/sdk/android/api/org/webrtc/Environment.java b/sdk/android/api/org/webrtc/Environment.java
index 5808776d01..10749ec4e2 100644
--- a/sdk/android/api/org/webrtc/Environment.java
+++ b/sdk/android/api/org/webrtc/Environment.java
@@ -39,14 +39,6 @@ public final class Environment implements AutoCloseable {
     return webrtcEnv;
   }
 
-  /**
-   * Returns current time as controlled by webrtcEnv. By default this is
-   * normally same as System.nanoTime(), but call this function to be safe.
-   */
-  public long getCurrentTimeNanos() {
-    return nativeCurrentTimeNanos(webrtcEnv);
-  }
-
   @Override
   public void close() {
     nativeFree(webrtcEnv);
@@ -57,6 +49,6 @@ public final class Environment implements AutoCloseable {
   }
 
   private static native long nativeCreate(@Nullable String fieldTrials);
-  private static native long nativeCurrentTimeNanos(long webrtcEnv);
+
   private static native void nativeFree(long webrtcEnv);
 }
diff --git a/sdk/android/api/org/webrtc/TimestampAligner.java b/sdk/android/api/org/webrtc/TimestampAligner.java
index a93f07c37f..37c92b31e1 100644
--- a/sdk/android/api/org/webrtc/TimestampAligner.java
+++ b/sdk/android/api/org/webrtc/TimestampAligner.java
@@ -12,33 +12,22 @@ package org.webrtc;
 
 /**
  * The TimestampAligner class helps translating camera timestamps into the same timescale as is
- * used by webrtc. Some cameras have built in timestamping which is more accurate than
+ * used by webrtc::TimeNanos(). Some cameras have built in timestamping which is more accurate than
  * reading the system clock, but using a different epoch and unknown clock drift. Frame timestamps
- * in webrtc should use clock from the webrtc::Environment (usually the system monotonic time), and
- * this class provides a filter which lets us use the webrtc::Environment timescale, and at the same
- * time take advantage of higheraccuracy of the camera clock.
- * This class is a wrapper on top of webrtc::TimestampAligner.
+ * in webrtc should use webrtc::TimeNanos (system monotonic time), and this class provides a filter
+ * which lets us use the webrtc::TimeNanos timescale, and at the same time take advantage of higher
+ * accuracy of the camera clock. This class is a wrapper on top of webrtc::TimestampAligner.
  */
 public class TimestampAligner {
   /**
    * Wrapper around webrtc::TimeNanos(). This is normally same as System.nanoTime(), but call this
    * function to be safe.
    */
-  @Deprecated
   public static long getRtcTimeNanos() {
-    return Environment.builder().build().getCurrentTimeNanos();
+    return nativeRtcTimeNanos();
   }
 
-  private volatile long nativeTimestampAligner;
-
-  @Deprecated
-  public TimestampAligner() {
-    this(Environment.builder().build());
-  }
-
-  public TimestampAligner(Environment webrtcEnv) {
-    nativeTimestampAligner = nativeCreateTimestampAligner(webrtcEnv.ref());
-  }
+  private volatile long nativeTimestampAligner = nativeCreateTimestampAligner();
 
   /**
    * Translates camera timestamps to the same timescale as is used by webrtc::TimeNanos().
@@ -63,7 +52,8 @@ public class TimestampAligner {
     }
   }
 
-  private static native long nativeCreateTimestampAligner(long webrtcEnvRef);
+  private static native long nativeRtcTimeNanos();
+  private static native long nativeCreateTimestampAligner();
   private static native void nativeReleaseTimestampAligner(long timestampAligner);
   private static native long nativeTranslateTimestamp(long timestampAligner, long cameraTimeNs);
 }
diff --git a/sdk/android/src/jni/environment.cc b/sdk/android/src/jni/environment.cc
index bf751e47d1..a93afe7bb0 100644
--- a/sdk/android/src/jni/environment.cc
+++ b/sdk/android/src/jni/environment.cc
@@ -27,10 +27,6 @@ void JNI_Environment_Free(JNIEnv* /*env*/, jlong webrtcEnv) {
   delete reinterpret_cast<Environment*>(webrtcEnv);
 }
 
-jlong JNI_Environment_CurrentTimeNanos(JNIEnv* env, jlong webrtcEnv) {
-  return reinterpret_cast<Environment*>(webrtcEnv)->clock().CurrentTime().ns();
-}
-
 jlong JNI_Environment_Create(JNIEnv* env,
                              const jni_zero::JavaRef<jstring>& fieldTrials) {
   std::unique_ptr<FieldTrialsView> field_trials;
diff --git a/sdk/android/src/jni/timestamp_aligner.cc b/sdk/android/src/jni/timestamp_aligner.cc
index 12fbd0052f..7226886cb0 100644
--- a/sdk/android/src/jni/timestamp_aligner.cc
+++ b/sdk/android/src/jni/timestamp_aligner.cc
@@ -12,51 +12,34 @@
 
 #include <jni.h>
 
-#include <cstdint>
-
-#include "api/environment/environment.h"
+#include "rtc_base/time_utils.h"
 #include "sdk/android/generated_video_jni/TimestampAligner_jni.h"
-#include "sdk/android/native_api/jni/java_types.h"
+#include "sdk/android/src/jni/jni_helpers.h"
 
 namespace webrtc {
 namespace jni {
-namespace {
 
-class TimestampAlignerWithClock {
- public:
-  explicit TimestampAlignerWithClock(const Environment& env) : env_(env) {}
+static jlong JNI_TimestampAligner_RtcTimeNanos(JNIEnv* env) {
+  return TimeNanos();
+}
 
-  int64_t TranslateTimestampNs(int64_t camera_time_ns) {
-    return aligner_.TranslateTimestamp(
-               /*capture_time_us=*/camera_time_ns / 1'000,
-               /*system_time_us=*/env_.clock().TimeInMicroseconds()) *
-           1'000;
-  }
-
- private:
-  const Environment env_;
-  TimestampAligner aligner_;
-};
-
-}  // namespace
-
-static jlong JNI_TimestampAligner_CreateTimestampAligner(JNIEnv* env,
-                                                         jlong webrtcEnvRef) {
-  return NativeToJavaPointer(new TimestampAlignerWithClock(
-      *reinterpret_cast<const Environment*>(webrtcEnvRef)));
+static jlong JNI_TimestampAligner_CreateTimestampAligner(JNIEnv* env) {
+  return jlongFromPointer(new TimestampAligner());
 }
 
 static void JNI_TimestampAligner_ReleaseTimestampAligner(
     JNIEnv* env,
     jlong timestamp_aligner) {
-  delete reinterpret_cast<TimestampAlignerWithClock*>(timestamp_aligner);
+  delete reinterpret_cast<TimestampAligner*>(timestamp_aligner);
 }
 
 static jlong JNI_TimestampAligner_TranslateTimestamp(JNIEnv* env,
                                                      jlong timestamp_aligner,
                                                      jlong camera_time_ns) {
-  return reinterpret_cast<TimestampAlignerWithClock*>(timestamp_aligner)
-      ->TranslateTimestampNs(camera_time_ns);
+  return reinterpret_cast<TimestampAligner*>(timestamp_aligner)
+             ->TranslateTimestamp(camera_time_ns / kNumNanosecsPerMicrosec,
+                                  TimeMicros()) *
+         kNumNanosecsPerMicrosec;
 }
 
 }  // namespace jni

From 2095aff3ce3b842d868bfb4afe56d562b0fc8f64 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Thu, 5 Mar 2026 12:04:17 -0800
Subject: [PATCH 212/472] Roll chromium_revision 44cac2a0af..caadb00f3d
 (1594634:1594862)

Change log: https://chromium.googlesource.com/chromium/src/+log/44cac2a0af..caadb00f3d
Full diff: https://chromium.googlesource.com/chromium/src/+/44cac2a0af..caadb00f3d

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/51fd5e3b61..92e11019ed
* src/buildtools/linux64: git_revision:a155c8c78af4030169a8da4886dc0c6b170535de..git_revision:d8c2f07d653520568da7cace755a87dad241b72d
* src/buildtools/mac: git_revision:a155c8c78af4030169a8da4886dc0c6b170535de..git_revision:d8c2f07d653520568da7cace755a87dad241b72d
* src/buildtools/win: git_revision:a155c8c78af4030169a8da4886dc0c6b170535de..git_revision:d8c2f07d653520568da7cace755a87dad241b72d
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/add05663e6..8ae867bf6b
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/6a91250a4b..d655a043c9
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/db2dce87ec..418b3c14d5
* src/third_party/androidx/cipd: 1-25o844ZqTB3NxqGHhgpHC_CG6J-Yod2Jj7VsVYTDQC..cbyKZsfMFZDYX5M7c5y-F8b2kOcy1zhCj75x1vHVtfMC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/2127bd31b6..b8b7f30825
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/a1e573faed..aceafdaa59
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/4a1e0fb594..3cae447548
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/02364664bb..f6d82460ec
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/aca79bce63..158b0470d7
DEPS diff: https://chromium.googlesource.com/chromium/src/+/44cac2a0af..caadb00f3d/DEPS

BUG=None

Change-Id: I5bf91e79d02da668b05c2e3908bc404e5504faf3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453960
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47065}
---
 DEPS | 28 ++++++++++++++--------------
 1 file changed, 14 insertions(+), 14 deletions(-)

diff --git a/DEPS b/DEPS
index 62c303782d..fb1b9d1bbf 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '44cac2a0af0b07a8b5a0ea81b9af2f69964f443f',
+  'chromium_revision': 'caadb00f3d641dd7e2deed7899d6a0af1978ec4d',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@51fd5e3b61c86e0601a4803fabe0f1baf209260b',
+    'https://chromium.googlesource.com/chromium/src/build@92e11019ed747d0d2b79c813862a7ff0c2d2f9d3',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@fef8d6947fa54202e8a00c58d5ac8fd15780bc33',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,19 +78,19 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@add05663e6258b17026b5fa488ba9d96d961c788',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@8ae867bf6b41a4bc727ef8f17160d8d6e1a5362b',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@6a91250a4bfd191c04b3f42357f76f59ed78447f',
+    'https://chromium.googlesource.com/chromium/src/testing@d655a043c946d1ff38c10d912541a3732916af29',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@db2dce87eca65ce0b769a2440e2292cb2c57be27',
+    'https://chromium.googlesource.com/chromium/src/third_party@418b3c14d55646854e8b4170fb4dfa80813a289b',
 
   'src/buildtools/linux64': {
     'packages': [
       {
         'package': 'gn/gn/linux-${{arch}}',
-        'version': 'git_revision:a155c8c78af4030169a8da4886dc0c6b170535de',
+        'version': 'git_revision:d8c2f07d653520568da7cace755a87dad241b72d',
       }
     ],
     'dep_type': 'cipd',
@@ -100,7 +100,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/mac-${{arch}}',
-        'version': 'git_revision:a155c8c78af4030169a8da4886dc0c6b170535de',
+        'version': 'git_revision:d8c2f07d653520568da7cace755a87dad241b72d',
       }
     ],
     'dep_type': 'cipd',
@@ -110,7 +110,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/windows-amd64',
-        'version': 'git_revision:a155c8c78af4030169a8da4886dc0c6b170535de',
+        'version': 'git_revision:d8c2f07d653520568da7cace755a87dad241b72d',
       }
     ],
     'dep_type': 'cipd',
@@ -488,7 +488,7 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@2127bd31b60ecc072559f5fa4cca027eec3a3530',
+    'https://boringssl.googlesource.com/boringssl.git@b8b7f308253df4a65f709a0f17d37c89dbcb9111',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@8be0e3114685fcc1589561067282edf75ea1259a',
   'src/third_party/catapult':
@@ -505,13 +505,13 @@ deps = {
   'src/third_party/crc32c/src':
     'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558',
   'src/third_party/depot_tools':
-    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@a1e573faed3fd7e783fa340ff4af4e4193574d7b',
+    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@aceafdaa59d7f59397abf1812298e065f39db958',
   'src/third_party/ffmpeg':
     'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@946d97db8d906277085e361892b7efda5152e2f1',
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@4a1e0fb594588a81e11187d0c34507a22a141e42',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@3cae4475487e1715235bd037acff6214ac16f10c',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '02364664bb5dfa6c04d4b274aa818f089aa8e42e',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'f6d82460ec55f78526aa05aca7d5e9bed3250244',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@aca79bce63b7bcc68f33310f7c48c6caa6f1af9a',
+    'https://chromium.googlesource.com/chromium/src/tools@158b0470d7cd0b4a82a6b46148c2d3cedb1d5566',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': '1-25o844ZqTB3NxqGHhgpHC_CG6J-Yod2Jj7VsVYTDQC',
+          'version': 'cbyKZsfMFZDYX5M7c5y-F8b2kOcy1zhCj75x1vHVtfMC',
       },
     ],
     'condition': 'checkout_android and non_git_source',

From b605b809e718f6e21f9a2d728e77a893df10df72 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Thu, 5 Mar 2026 14:06:56 +0100
Subject: [PATCH 213/472] Update first packet callbacks to include SSRC
 information

Update the received packet callbacks to pass the SSRC of the incoming
packet. This allows RtpReceiver and RtpTransceiver to associate the
"first packet" notification with a specific media stream, providing
better context for state changes and down the line inform the signaling
thread of when an SSRC is signaled. That in turn will facilitate the
ability of being able to read the ssrc of receivers on the signaling
thread without a thread hop.

Additionally, rename SetFirstPacketReceivedCallback and
SetFirstPacketSentCallback to include the "_n" suffix, explicitly
indicating that these operations must run on the network thread.

Bug: none
Change-Id: Ib71058f17d1f4d506a7d46da1f012269b923b6b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449460
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47066}
---
 pc/audio_rtp_receiver.cc             |  5 ++--
 pc/audio_rtp_receiver.h              |  4 +--
 pc/channel.cc                        | 14 ++++-----
 pc/channel.h                         | 15 +++++-----
 pc/channel_interface.h               |  9 +++---
 pc/rtp_receiver.h                    |  4 +--
 pc/rtp_transceiver.cc                | 45 +++++++++++++++-------------
 pc/rtp_transceiver.h                 |  6 ++--
 pc/rtp_transceiver_unittest.cc       |  8 ++---
 pc/test/mock_channel_interface.h     |  9 +++---
 pc/test/mock_rtp_receiver_internal.h |  4 +--
 pc/video_rtp_receiver.cc             |  5 ++--
 pc/video_rtp_receiver.h              |  4 +--
 13 files changed, 72 insertions(+), 60 deletions(-)

diff --git a/pc/audio_rtp_receiver.cc b/pc/audio_rtp_receiver.cc
index e1f5acb032..b90e70a863 100644
--- a/pc/audio_rtp_receiver.cc
+++ b/pc/audio_rtp_receiver.cc
@@ -376,7 +376,7 @@ void AudioRtpReceiver::SetMediaChannel(
       static_cast<VoiceMediaReceiveChannelInterface*>(media_channel);
 }
 
-void AudioRtpReceiver::NotifyFirstPacketReceived() {
+void AudioRtpReceiver::NotifyFirstPacketReceived(uint32_t ssrc) {
   RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
   if (observer_) {
     observer_->OnFirstPacketReceived(media_type());
@@ -384,7 +384,8 @@ void AudioRtpReceiver::NotifyFirstPacketReceived() {
   received_first_packet_ = true;
 }
 
-void AudioRtpReceiver::NotifyFirstPacketReceivedAfterReceptiveChange() {
+void AudioRtpReceiver::NotifyFirstPacketReceivedAfterReceptiveChange(
+    uint32_t ssrc) {
   RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
   if (observer_) {
     observer_->OnFirstPacketReceivedAfterReceptiveChange(media_type());
diff --git a/pc/audio_rtp_receiver.h b/pc/audio_rtp_receiver.h
index 70247478a4..a8b07a0186 100644
--- a/pc/audio_rtp_receiver.h
+++ b/pc/audio_rtp_receiver.h
@@ -111,8 +111,8 @@ class AudioRtpReceiver : public ObserverInterface,
   absl::AnyInvocable<void() &&> GetSetupForMediaChannel(uint32_t ssrc) override;
   absl::AnyInvocable<void() &&> GetSetupForUnsignaledMediaChannel() override;
   std::optional<uint32_t> ssrc() const override;
-  void NotifyFirstPacketReceived() override;
-  void NotifyFirstPacketReceivedAfterReceptiveChange() override;
+  void NotifyFirstPacketReceived(uint32_t ssrc) override;
+  void NotifyFirstPacketReceivedAfterReceptiveChange(uint32_t ssrc) override;
   void set_stream_ids(std::vector<std::string> stream_ids) override;
   void set_transport(
       scoped_refptr<DtlsTransportInterface> dtls_transport) override;
diff --git a/pc/channel.cc b/pc/channel.cc
index 0cea1a549e..7b31a16b57 100644
--- a/pc/channel.cc
+++ b/pc/channel.cc
@@ -428,12 +428,12 @@ void BaseChannel::OnNetworkRouteChanged(
   media_send_channel()->OnNetworkRouteChanged(transport_name(), new_route);
 }
 
-void BaseChannel::SetFirstPacketReceivedCallback(
-    absl::AnyInvocable<void() &&> callback) {
+void BaseChannel::SetFirstPacketReceivedCallback_n(
+    absl::AnyInvocable<void(const RtpPacketReceived&) &&> callback) {
   RTC_DCHECK_RUN_ON(network_thread());
   RTC_DCHECK(!on_first_packet_received_ || !callback);
 
-  // TODO(bugs.webrtc.org/11992): Rename SetFirstPacketReceivedCallback to
+  // TODO(bugs.webrtc.org/11992): Rename SetFirstPacketReceivedCallback_n to
   // something that indicates network thread initialization/uninitialization and
   // call Init_n() / Deinit_n() respectively.
   // if (!callback)
@@ -442,7 +442,7 @@ void BaseChannel::SetFirstPacketReceivedCallback(
   on_first_packet_received_ = std::move(callback);
 }
 
-void BaseChannel::SetFirstPacketSentCallback(
+void BaseChannel::SetFirstPacketSentCallback_n(
     absl::AnyInvocable<void() &&> callback) {
   RTC_DCHECK_RUN_ON(network_thread());
   RTC_DCHECK(!on_first_packet_sent_ || !callback);
@@ -451,7 +451,7 @@ void BaseChannel::SetFirstPacketSentCallback(
 }
 
 void BaseChannel::SetPacketReceivedCallback_n(
-    absl::AnyInvocable<void()> callback) {
+    absl::AnyInvocable<void(const RtpPacketReceived&)> callback) {
   RTC_DCHECK_RUN_ON(network_thread());
   RTC_DCHECK(!on_packet_received_n_ || !callback);
 
@@ -520,7 +520,7 @@ void BaseChannel::OnRtpPacket(const RtpPacketReceived& parsed_packet) {
   RTC_DCHECK(network_initialized());
 
   if (on_first_packet_received_) {
-    std::move(on_first_packet_received_)();
+    std::move(on_first_packet_received_)(parsed_packet);
     on_first_packet_received_ = nullptr;
   }
 
@@ -542,7 +542,7 @@ void BaseChannel::OnRtpPacket(const RtpPacketReceived& parsed_packet) {
     return;
   }
   if (on_packet_received_n_) {
-    on_packet_received_n_();
+    on_packet_received_n_(parsed_packet);
   }
   media_receive_channel()->OnPacketReceived(parsed_packet);
 }
diff --git a/pc/channel.h b/pc/channel.h
index 615bb57185..995a17e96b 100644
--- a/pc/channel.h
+++ b/pc/channel.h
@@ -148,12 +148,13 @@ class BaseChannel : public ChannelInterface,
   }
 
   // Used for latency measurements.
-  void SetFirstPacketReceivedCallback(
-      absl::AnyInvocable<void() &&> callback) override;
-  void SetFirstPacketSentCallback(
+  void SetFirstPacketReceivedCallback_n(
+      absl::AnyInvocable<void(const RtpPacketReceived&) &&> callback) override;
+  void SetFirstPacketSentCallback_n(
       absl::AnyInvocable<void() &&> callback) override;
 
-  void SetPacketReceivedCallback_n(absl::AnyInvocable<void()> callback) override
+  void SetPacketReceivedCallback_n(
+      absl::AnyInvocable<void(const RtpPacketReceived&)> callback) override
       RTC_RUN_ON(network_thread());
 
   // From RtpTransport - public for testing only
@@ -322,13 +323,13 @@ class BaseChannel : public ChannelInterface,
   scoped_refptr<PendingTaskSafetyFlag> alive_;
 
   // The functions are deleted after they have been called.
-  absl::AnyInvocable<void() &&> on_first_packet_received_
-      RTC_GUARDED_BY(network_thread());
+  absl::AnyInvocable<void(const RtpPacketReceived&) &&>
+      on_first_packet_received_ RTC_GUARDED_BY(network_thread());
   absl::AnyInvocable<void() &&> on_first_packet_sent_
       RTC_GUARDED_BY(network_thread());
 
   // Used to unmute.
-  absl::AnyInvocable<void()> on_packet_received_n_
+  absl::AnyInvocable<void(const RtpPacketReceived&)> on_packet_received_n_
       RTC_GUARDED_BY(network_thread());
 
   RtpTransportInternal* rtp_transport_ RTC_GUARDED_BY(network_thread()) =
diff --git a/pc/channel_interface.h b/pc/channel_interface.h
index efeea359ba..1e1c9bd32e 100644
--- a/pc/channel_interface.h
+++ b/pc/channel_interface.h
@@ -25,6 +25,7 @@
 
 namespace webrtc {
 class Call;
+class RtpPacketReceived;
 class VideoBitrateAllocatorFactory;
 class VideoChannel;
 class VoiceChannel;
@@ -75,14 +76,14 @@ class ChannelInterface {
   virtual void Enable(bool enable) = 0;
 
   // Used for latency measurements.
-  virtual void SetFirstPacketReceivedCallback(
-      absl::AnyInvocable<void() &&> callback) = 0;
-  virtual void SetFirstPacketSentCallback(
+  virtual void SetFirstPacketReceivedCallback_n(
+      absl::AnyInvocable<void(const RtpPacketReceived&) &&> callback) = 0;
+  virtual void SetFirstPacketSentCallback_n(
       absl::AnyInvocable<void() &&> callback) = 0;
 
   // Used to unmute.
   virtual void SetPacketReceivedCallback_n(
-      absl::AnyInvocable<void()> callback) = 0;
+      absl::AnyInvocable<void(const RtpPacketReceived&)> callback) = 0;
 
   // Channel control
   virtual bool SetLocalContent(const MediaContentDescription* content,
diff --git a/pc/rtp_receiver.h b/pc/rtp_receiver.h
index 3ef0274811..45399b684d 100644
--- a/pc/rtp_receiver.h
+++ b/pc/rtp_receiver.h
@@ -71,9 +71,9 @@ class RtpReceiverInternal : public RtpReceiverInterface {
 
   // Call this to notify the RtpReceiver when the first packet has been received
   // on the corresponding channel.
-  virtual void NotifyFirstPacketReceived() = 0;
+  virtual void NotifyFirstPacketReceived(uint32_t ssrc) = 0;
   // Similar to the above but done whenever the receptive flag changed.
-  virtual void NotifyFirstPacketReceivedAfterReceptiveChange() = 0;
+  virtual void NotifyFirstPacketReceivedAfterReceptiveChange(uint32_t ssrc) = 0;
 
   // Set the associated remote media streams for this receiver. The remote track
   // will be removed from any streams that are no longer present and added to
diff --git a/pc/rtp_transceiver.cc b/pc/rtp_transceiver.cc
index 0ccad43623..e92b8afb8c 100644
--- a/pc/rtp_transceiver.cc
+++ b/pc/rtp_transceiver.cc
@@ -538,20 +538,24 @@ RTCError RtpTransceiver::SetChannel(
         if (transport) {
           transport_name = transport->transport_name();
         }
-        channel->SetFirstPacketReceivedCallback([thread = thread_, flag = flag,
-                                                 this]() mutable {
-          thread->PostTask(
-              SafeTask(std::move(flag), [this]() { OnFirstPacketReceived(); }));
-        });
-        channel->SetFirstPacketSentCallback([thread = thread_, flag = flag,
-                                             this]() mutable {
-          thread->PostTask(
-              SafeTask(std::move(flag), [this]() { OnFirstPacketSent(); }));
-        });
-        channel->SetPacketReceivedCallback_n([this, flag = flag]() {
-          RTC_DCHECK_RUN_ON(context()->network_thread());
-          OnPacketReceived(flag);
-        });
+        channel->SetFirstPacketReceivedCallback_n(
+            [thread = thread_, flag = flag,
+             this](const RtpPacketReceived& packet) mutable {
+              thread->PostTask(
+                  SafeTask(std::move(flag), [this, ssrc = packet.Ssrc()]() {
+                    OnFirstPacketReceived(ssrc);
+                  }));
+            });
+        channel->SetFirstPacketSentCallback_n(
+            [thread = thread_, flag = flag, this]() mutable {
+              thread->PostTask(
+                  SafeTask(std::move(flag), [this]() { OnFirstPacketSent(); }));
+            });
+        channel->SetPacketReceivedCallback_n(
+            [this, flag = flag](const RtpPacketReceived& packet) {
+              RTC_DCHECK_RUN_ON(context()->network_thread());
+              OnPacketReceived(packet.Ssrc(), flag);
+            });
         return RTCError::OK();
       });
 
@@ -584,8 +588,8 @@ absl::AnyInvocable<void() &&> RtpTransceiver::GetClearChannelNetworkTask() {
   ChannelInterface* channel = channel_.get();
   return [channel, flag = network_thread_safety_] {
     flag->SetNotAlive();
-    channel->SetFirstPacketReceivedCallback(nullptr);
-    channel->SetFirstPacketSentCallback(nullptr);
+    channel->SetFirstPacketReceivedCallback_n(nullptr);
+    channel->SetFirstPacketSentCallback_n(nullptr);
     channel->SetPacketReceivedCallback_n(nullptr);
     channel->SetRtpTransport(nullptr);
   };
@@ -779,14 +783,15 @@ MediaEngineInterface* RtpTransceiver::media_engine() {
   return media_engine_ref_->media_engine();
 }
 
-void RtpTransceiver::OnFirstPacketReceived() {
+void RtpTransceiver::OnFirstPacketReceived(uint32_t ssrc) {
   for (const auto& receiver : receivers_) {
-    receiver->internal()->NotifyFirstPacketReceived();
+    receiver->internal()->NotifyFirstPacketReceived(ssrc);
   }
 }
 
 // RTC_RUN_ON(context()->network_thread())
 void RtpTransceiver::OnPacketReceived(
+    uint32_t ssrc,
     scoped_refptr<PendingTaskSafetyFlag> safety) {
   if (!receptive_n_) {
     return;
@@ -795,13 +800,13 @@ void RtpTransceiver::OnPacketReceived(
     return;
   }
   packet_notified_after_receptive_ = true;
-  thread_->PostTask(SafeTask(safety, [this]() {
+  thread_->PostTask(SafeTask(safety, [this, ssrc]() {
     RTC_DCHECK_RUN_ON(thread_);
     if (stopping() || stopped() || !receptive_) {
       return;
     }
     for (const auto& receiver : receivers_) {
-      receiver->internal()->NotifyFirstPacketReceivedAfterReceptiveChange();
+      receiver->internal()->NotifyFirstPacketReceivedAfterReceptiveChange(ssrc);
     }
   }));
 }
diff --git a/pc/rtp_transceiver.h b/pc/rtp_transceiver.h
index 431e4cffde..6fc217674c 100644
--- a/pc/rtp_transceiver.h
+++ b/pc/rtp_transceiver.h
@@ -13,6 +13,7 @@
 
 #include <stddef.h>
 
+#include <cstdint>
 #include <memory>
 #include <optional>
 #include <string>
@@ -404,8 +405,9 @@ class RtpTransceiver : public RtpTransceiverInterface {
   CodecVendor& codec_vendor() {
     return *codec_lookup_helper_->GetCodecVendor();
   }
-  void OnFirstPacketReceived();
-  void OnPacketReceived(scoped_refptr<PendingTaskSafetyFlag> safety)
+  void OnFirstPacketReceived(uint32_t ssrc);
+  void OnPacketReceived(uint32_t ssrc,
+                        scoped_refptr<PendingTaskSafetyFlag> safety)
       RTC_RUN_ON(context()->network_thread());
   void OnFirstPacketSent();
   // Stops the receivers synchronously and returns a task that stops the
diff --git a/pc/rtp_transceiver_unittest.cc b/pc/rtp_transceiver_unittest.cc
index 1e42603ec2..bf09c33d3c 100644
--- a/pc/rtp_transceiver_unittest.cc
+++ b/pc/rtp_transceiver_unittest.cc
@@ -126,7 +126,7 @@ TEST_F(RtpTransceiverTest, CannotSetChannelOnStoppedTransceiver) {
   auto channel1 = std::make_unique<NiceMock<MockChannelInterface>>();
   EXPECT_CALL(*channel1, media_type()).WillRepeatedly(Return(MediaType::AUDIO));
   EXPECT_CALL(*channel1, mid()).WillRepeatedly(ReturnRef(content_name));
-  EXPECT_CALL(*channel1, SetFirstPacketReceivedCallback(_));
+  EXPECT_CALL(*channel1, SetFirstPacketReceivedCallback_n(_));
   EXPECT_CALL(*channel1, SetRtpTransport(_)).WillRepeatedly(Return(true));
   auto channel1_ptr = channel1.get();
   transceiver->SetChannel(std::move(channel1), [&](const std::string& mid) {
@@ -143,7 +143,7 @@ TEST_F(RtpTransceiverTest, CannotSetChannelOnStoppedTransceiver) {
   EXPECT_CALL(*channel2, media_type()).WillRepeatedly(Return(MediaType::AUDIO));
 
   // Clear the current channel - required to allow SetChannel()
-  EXPECT_CALL(*channel1_ptr, SetFirstPacketReceivedCallback(_));
+  EXPECT_CALL(*channel1_ptr, SetFirstPacketReceivedCallback_n(_));
   transceiver->ClearChannel();
   ASSERT_FALSE(transceiver->HasChannel());
   // Channel can no longer be set, so this call should be a no-op.
@@ -161,7 +161,7 @@ TEST_F(RtpTransceiverTest, CanUnsetChannelOnStoppedTransceiver) {
   auto channel = std::make_unique<NiceMock<MockChannelInterface>>();
   EXPECT_CALL(*channel, media_type()).WillRepeatedly(Return(MediaType::VIDEO));
   EXPECT_CALL(*channel, mid()).WillRepeatedly(ReturnRef(content_name));
-  EXPECT_CALL(*channel, SetFirstPacketReceivedCallback(_))
+  EXPECT_CALL(*channel, SetFirstPacketReceivedCallback_n(_))
       .WillRepeatedly(testing::Return());
   EXPECT_CALL(*channel, SetRtpTransport(_)).WillRepeatedly(Return(true));
 
@@ -195,7 +195,7 @@ TEST_F(RtpTransceiverTest, TransportNameIsUpdated) {
   auto channel = std::make_unique<NiceMock<MockChannelInterface>>();
   EXPECT_CALL(*channel, media_type()).WillRepeatedly(Return(MediaType::AUDIO));
   EXPECT_CALL(*channel, mid()).WillRepeatedly(ReturnRef(content_name));
-  EXPECT_CALL(*channel, SetFirstPacketReceivedCallback(_))
+  EXPECT_CALL(*channel, SetFirstPacketReceivedCallback_n(_))
       .WillRepeatedly(Return());
   EXPECT_CALL(*channel, SetRtpTransport(_)).WillRepeatedly(Return(true));
 
diff --git a/pc/test/mock_channel_interface.h b/pc/test/mock_channel_interface.h
index c4f96e5fb4..1dc4ddd57e 100644
--- a/pc/test/mock_channel_interface.h
+++ b/pc/test/mock_channel_interface.h
@@ -26,6 +26,7 @@
 #include "test/gmock.h"
 
 namespace webrtc {
+class RtpPacketReceived;
 
 // Mock class for BaseChannel.
 // Use this class in unit tests to avoid dependency on a specific
@@ -60,16 +61,16 @@ class MockChannelInterface : public ChannelInterface {
   MOCK_METHOD(const std::string&, mid, (), (const, override));
   MOCK_METHOD(void, Enable, (bool), (override));
   MOCK_METHOD(void,
-              SetFirstPacketReceivedCallback,
-              (absl::AnyInvocable<void() &&>),
+              SetFirstPacketReceivedCallback_n,
+              (absl::AnyInvocable<void(const RtpPacketReceived&) &&>),
               (override));
   MOCK_METHOD(void,
-              SetFirstPacketSentCallback,
+              SetFirstPacketSentCallback_n,
               (absl::AnyInvocable<void() &&>),
               (override));
   MOCK_METHOD(void,
               SetPacketReceivedCallback_n,
-              (absl::AnyInvocable<void()>),
+              (absl::AnyInvocable<void(const RtpPacketReceived&)>),
               (override));
   MOCK_METHOD(bool,
               SetLocalContent,
diff --git a/pc/test/mock_rtp_receiver_internal.h b/pc/test/mock_rtp_receiver_internal.h
index 8d3bc701dd..29fbd4f54d 100644
--- a/pc/test/mock_rtp_receiver_internal.h
+++ b/pc/test/mock_rtp_receiver_internal.h
@@ -81,10 +81,10 @@ class MockRtpReceiverInternal : public RtpReceiverInternal {
               (),
               (override));
   MOCK_METHOD(std::optional<uint32_t>, ssrc, (), (const, override));
-  MOCK_METHOD(void, NotifyFirstPacketReceived, (), (override));
+  MOCK_METHOD(void, NotifyFirstPacketReceived, (uint32_t), (override));
   MOCK_METHOD(void,
               NotifyFirstPacketReceivedAfterReceptiveChange,
-              (),
+              (uint32_t),
               (override));
   MOCK_METHOD(void, set_stream_ids, (std::vector<std::string>), (override));
   MOCK_METHOD(void,
diff --git a/pc/video_rtp_receiver.cc b/pc/video_rtp_receiver.cc
index f3e4f9db7f..b296c818a5 100644
--- a/pc/video_rtp_receiver.cc
+++ b/pc/video_rtp_receiver.cc
@@ -334,7 +334,7 @@ void VideoRtpReceiver::SetMediaChannel_w(
     source_->ClearCallback();
 }
 
-void VideoRtpReceiver::NotifyFirstPacketReceived() {
+void VideoRtpReceiver::NotifyFirstPacketReceived(uint32_t ssrc) {
   RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
   if (observer_) {
     observer_->OnFirstPacketReceived(media_type());
@@ -342,7 +342,8 @@ void VideoRtpReceiver::NotifyFirstPacketReceived() {
   received_first_packet_ = true;
 }
 
-void VideoRtpReceiver::NotifyFirstPacketReceivedAfterReceptiveChange() {
+void VideoRtpReceiver::NotifyFirstPacketReceivedAfterReceptiveChange(
+    uint32_t ssrc) {
   RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
   if (observer_) {
     observer_->OnFirstPacketReceivedAfterReceptiveChange(media_type());
diff --git a/pc/video_rtp_receiver.h b/pc/video_rtp_receiver.h
index 6025bed1ca..ec0d8bb39b 100644
--- a/pc/video_rtp_receiver.h
+++ b/pc/video_rtp_receiver.h
@@ -91,8 +91,8 @@ class VideoRtpReceiver : public RtpReceiverInternal {
   absl::AnyInvocable<void() &&> GetSetupForMediaChannel(uint32_t ssrc) override;
   absl::AnyInvocable<void() &&> GetSetupForUnsignaledMediaChannel() override;
   std::optional<uint32_t> ssrc() const override;
-  void NotifyFirstPacketReceived() override;
-  void NotifyFirstPacketReceivedAfterReceptiveChange() override;
+  void NotifyFirstPacketReceived(uint32_t ssrc) override;
+  void NotifyFirstPacketReceivedAfterReceptiveChange(uint32_t ssrc) override;
   void set_stream_ids(std::vector<std::string> stream_ids) override;
   void set_transport(
       scoped_refptr<DtlsTransportInterface> dtls_transport) override;

From 4a795bb4aedbe3d34cc79a7c2dbb3400ef2df4c6 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Thu, 5 Mar 2026 14:10:47 +0100
Subject: [PATCH 214/472] Remove unnecessary `using` statements from
 sdp_offer_answer.cc

# Using No-Try since `webrtc_linux_chromium infra failed` several times

No-Try: true
Bug: none
Change-Id: I7e0910383fee8f34bd0f0ae0b5ac3a050c6bbe8f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453861
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47067}
---
 pc/sdp_offer_answer.cc | 14 --------------
 1 file changed, 14 deletions(-)

diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc
index aa6fe33e34..020aa11eee 100644
--- a/pc/sdp_offer_answer.cc
+++ b/pc/sdp_offer_answer.cc
@@ -109,21 +109,7 @@
 #include "rtc_base/weak_ptr.h"
 #include "system_wrappers/include/metrics.h"
 
-using ::webrtc::ContentInfo;
-using ::webrtc::ContentInfos;
-using webrtc::MediaContentDescription;
-using ::webrtc::MediaProtocolType;
-using ::webrtc::RidDescription;
-using ::webrtc::RidDirection;
-using ::webrtc::SessionDescription;
-using ::webrtc::SimulcastDescription;
-using ::webrtc::SimulcastLayer;
-using ::webrtc::SimulcastLayerList;
-using ::webrtc::StreamParams;
-using ::webrtc::TransportInfo;
-
 namespace webrtc {
-
 namespace {
 
 class ScopedOperationsBatcher {

From 68ef77be4d15e67bcb1a45276b94b40b38b291a4 Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Thu, 5 Mar 2026 14:37:43 +0100
Subject: [PATCH 215/472] Replace ArrayView with std::span in test/

This change generated using mostly search and replace:
- MakeArrayView replaced with std::span
- ArrayView replaced with std::span
- Run include cleaner
- Removed dependency on array_view target

Bug: webrtc:439801349
Change-Id: Id3b440e8f86559fcb596a7896a62c395fb40496c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453860
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47068}
---
 test/BUILD.gn                                 | 17 +---
 test/call_test.h                              |  4 +-
 test/direct_transport.cc                      | 16 ++--
 test/direct_transport.h                       | 14 +--
 test/fuzzers/BUILD.gn                         | 41 ---------
 test/fuzzers/agc_fuzzer.cc                    |  1 -
 test/fuzzers/audio_decoder_fuzzer.cc          |  4 +-
 test/fuzzers/audio_encoder_fuzzer.cc          |  1 -
 .../fuzzers/audio_processing_fuzzer_helper.cc |  4 +-
 test/fuzzers/base64_encode_fuzzer.cc          |  4 +-
 test/fuzzers/comfort_noise_decoder_fuzzer.cc  |  6 +-
 test/fuzzers/dcsctp_packet_fuzzer.cc          |  1 -
 test/fuzzers/flexfec_header_reader_fuzzer.cc  |  4 +-
 .../forward_error_correction_fuzzer.cc        |  4 +-
 test/fuzzers/neteq_rtp_fuzzer.cc              | 10 +--
 test/fuzzers/neteq_signal_fuzzer.cc           |  6 +-
 test/fuzzers/rtcp_receiver_fuzzer.cc          |  4 +-
 ..._depacketizer_av1_assemble_frame_fuzzer.cc |  4 +-
 .../rtp_dependency_descriptor_fuzzer.cc       |  4 +-
 .../rtp_video_layers_allocation_fuzzer.cc     |  6 +-
 test/fuzzers/turn_unwrap_fuzzer.cc            |  5 +-
 test/fuzzers/ulpfec_header_reader_fuzzer.cc   |  4 +-
 test/fuzzers/ulpfec_receiver_fuzzer.cc        |  1 -
 test/fuzzers/vp8_qp_parser_fuzzer.cc          |  4 +-
 test/fuzzers/vp8_replay_fuzzer.cc             |  4 +-
 test/fuzzers/vp9_encoder_references_fuzzer.cc | 11 +--
 test/fuzzers/vp9_replay_fuzzer.cc             |  4 +-
 test/fuzzers/webrtc_fuzzer_main.cc            |  4 +-
 test/layer_filtering_transport.cc             | 12 +--
 test/layer_filtering_transport.h              | 12 +--
 test/mappable_native_buffer.cc                |  6 +-
 test/mappable_native_buffer.h                 |  6 +-
 test/mock_audio_encoder.cc                    | 10 +--
 test/mock_audio_encoder.h                     | 16 ++--
 test/mock_transport.h                         |  6 +-
 test/network/BUILD.gn                         |  1 -
 test/network/network_emulation_manager.cc     |  6 +-
 test/network/network_emulation_manager.h      |  6 +-
 test/null_transport.cc                        |  6 +-
 test/null_transport.h                         |  6 +-
 test/pc/e2e/BUILD.gn                          |  5 --
 test/pc/e2e/analyzer/video/BUILD.gn           |  6 --
 .../video/default_video_quality_analyzer.cc   |  4 +-
 .../video/default_video_quality_analyzer.h    |  4 +-
 ...ideo_quality_analyzer_frames_comparator.cc |  4 +-
 ...video_quality_analyzer_frames_comparator.h |  4 +-
 .../default_video_quality_analyzer_test.cc    |  6 +-
 .../video/example_video_quality_analyzer.cc   |  4 +-
 .../video/example_video_quality_analyzer.h    |  4 +-
 .../pc/e2e/analyzer/video/names_collection.cc |  4 +-
 test/pc/e2e/analyzer/video/names_collection.h |  4 +-
 ...video_quality_analyzer_injection_helper.cc |  4 +-
 .../video_quality_analyzer_injection_helper.h |  4 +-
 test/pc/e2e/echo/echo_emulation.cc            |  6 +-
 test/pc/e2e/echo/echo_emulation.h             |  6 +-
 test/pc/e2e/peer_params_preprocessor.cc       |  6 +-
 test/pc/e2e/sdp/sdp_changer.cc                | 12 +--
 test/pc/e2e/sdp/sdp_changer.h                 | 10 +--
 ..._based_network_quality_metrics_reporter.cc |  1 -
 ...d_network_quality_metrics_reporter_test.cc |  4 +-
 test/peer_scenario/BUILD.gn                   |  1 -
 .../bwe_integration_tests/BUILD.gn            |  1 -
 .../bwe_integration_tests/l4s_test.cc         |  1 -
 .../bwe_integration_tests/scream_test.cc      | 10 +--
 .../bwe_integration_tests/stats_utilities.cc  |  4 +-
 .../bwe_integration_tests/stats_utilities.h   |  4 +-
 test/peer_scenario/scenario_connection.cc     | 10 +--
 test/peer_scenario/scenario_connection.h      |  6 +-
 test/rtcp_packet_parser.cc                    |  4 +-
 test/rtcp_packet_parser.h                     |  7 +-
 test/rtp_file_reader.cc                       |  4 +-
 test/rtp_file_reader_unittest.cc              |  4 +-
 test/rtp_rtcp_observer.h                      | 18 ++--
 test/scenario/BUILD.gn                        |  1 -
 test/scenario/call_client.cc                  |  6 +-
 test/scenario/call_client.h                   |  6 +-
 test/scenario/network_node.cc                 |  6 +-
 test/scenario/network_node.h                  |  6 +-
 test/testsupport/perf_test.cc                 |  8 +-
 test/testsupport/perf_test.h                  |  4 +-
 .../testsupport/perf_test_histogram_writer.cc |  4 +-
 test/testsupport/perf_test_result_writer.h    |  4 +-
 test/testsupport/video_frame_writer.cc        |  6 +-
 test/video_codec_tester.cc                    |  4 +-
 test/video_frame_matcher.cc                   | 89 +++++++++----------
 85 files changed, 261 insertions(+), 344 deletions(-)

diff --git a/test/BUILD.gn b/test/BUILD.gn
index 96da3e9f11..dbe5920024 100644
--- a/test/BUILD.gn
+++ b/test/BUILD.gn
@@ -234,7 +234,6 @@ rtc_library("video_test_common") {
     ":fileutils",
     ":frame_utils",
     ":test_video_capturer",
-    "../api:array_view",
     "../api:create_frame_generator",
     "../api:fec_controller_api",
     "../api:frame_generator_api",
@@ -346,7 +345,6 @@ rtc_library("rtp_test_utils") {
   ]
 
   deps = [
-    "../api:array_view",
     "../api:rtp_parameters",
     "../modules/rtp_rtcp",
     "../modules/rtp_rtcp:rtp_rtcp_format",
@@ -371,7 +369,6 @@ rtc_library("perf_test") {
     "testsupport/perf_test_result_writer.h",
   ]
   deps = [
-    "../api:array_view",
     "../api/numerics",
     "../rtc_base:checks",
     "../rtc_base:criticalsection",
@@ -505,7 +502,6 @@ rtc_library("video_frame_writer") {
   ]
 
   deps = [
-    "../api:array_view",
     "../api:scoped_refptr",
     "../api/test/video:video_frame_writer",
     "../api/video:video_frame",
@@ -724,7 +720,6 @@ if (rtc_include_tests) {
         ":video_test_support",
         ":wait_until",
         ":y4m_frame_generator",
-        "../api:array_view",
         "../api:create_frame_generator",
         "../api:create_simulcast_test_fixture_api",
         "../api:create_time_controller",
@@ -983,7 +978,6 @@ rtc_library("direct_transport") {
     "direct_transport.h",
   ]
   deps = [
-    "../api:array_view",
     "../api:rtp_headers",
     "../api:rtp_parameters",
     "../api:sequence_checker",
@@ -1059,10 +1053,7 @@ rtc_library("null_transport") {
     "null_transport.cc",
     "null_transport.h",
   ]
-  deps = [
-    "../api:array_view",
-    "../api:transport_api",
-  ]
+  deps = [ "../api:transport_api" ]
 }
 
 rtc_library("encoder_settings") {
@@ -1128,7 +1119,6 @@ rtc_library("mock_transport") {
   ]
   deps = [
     ":test_support",
-    "../api:array_view",
     "../api:transport_api",
   ]
 }
@@ -1161,7 +1151,6 @@ rtc_library("video_frame_matcher") {
   ]
   deps = [
     ":test_support",
-    "../api:array_view",
     "../api:scoped_refptr",
     "../api/video:video_frame",
     "../rtc_base:checks",
@@ -1256,7 +1245,6 @@ rtc_library("audio_codec_mocks") {
 
   deps = [
     ":test_support",
-    "../api:array_view",
     "../api:make_ref_counted",
     "../api:scoped_refptr",
     "../api/audio_codecs:audio_codecs_api",
@@ -1278,7 +1266,6 @@ if (!build_with_chromium) {
       "testsupport/copy_to_file_audio_capturer.h",
     ]
     deps = [
-      "../api:array_view",
       "../common_audio",
       "../modules/audio_device:test_audio_device_module",
       "../rtc_base:buffer",
@@ -1334,7 +1321,6 @@ if (!build_with_chromium) {
       ":video_test_common",
       ":video_test_constants",
       "..//test/network:simulated_network",
-      "../api:array_view",
       "../api:create_frame_generator",
       "../api:fec_controller_api",
       "../api:field_trials",
@@ -1485,7 +1471,6 @@ rtc_library("video_codec_tester") {
   deps = [
     ":video_frame_writer",
     ":video_test_support",
-    "../api:array_view",
     "../api:create_frame_generator",
     "../api:field_trials_view",
     "../api:frame_generator_api",
diff --git a/test/call_test.h b/test/call_test.h
index b1baabc337..86bbe39cf6 100644
--- a/test/call_test.h
+++ b/test/call_test.h
@@ -15,10 +15,10 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/audio_device.h"
 #include "api/audio/audio_processing.h"
 #include "api/audio_codecs/audio_decoder_factory.h"
@@ -83,7 +83,7 @@ class CallTest : public ::testing::Test, public RtpPacketSinkInterface {
 
   void RegisterRtpExtension(const RtpExtension& extension);
   // Returns header extensions that can be parsed by the transport.
-  ArrayView<const RtpExtension> GetRegisteredExtensions() {
+  std::span<const RtpExtension> GetRegisteredExtensions() {
     return rtp_extensions_;
   }
 
diff --git a/test/direct_transport.cc b/test/direct_transport.cc
index f2708257e6..0d3c28a0a7 100644
--- a/test/direct_transport.cc
+++ b/test/direct_transport.cc
@@ -14,9 +14,9 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 
-#include "api/array_view.h"
 #include "api/call/transport.h"
 #include "api/environment/environment.h"
 #include "api/media_types.h"
@@ -45,7 +45,7 @@ Demuxer::Demuxer(const std::map<uint8_t, MediaType>& payload_type_map)
 
 MediaType Demuxer::GetMediaType(const uint8_t* packet_data,
                                 const size_t packet_length) const {
-  if (IsRtpPacket(MakeArrayView(packet_data, packet_length))) {
+  if (IsRtpPacket(std::span(packet_data, packet_length))) {
     RTC_CHECK_GE(packet_length, 2);
     const uint8_t payload_type = packet_data[1] & 0x7f;
     std::map<uint8_t, MediaType>::const_iterator it =
@@ -62,8 +62,8 @@ DirectTransport::DirectTransport(
     std::unique_ptr<SimulatedPacketReceiverInterface> pipe,
     Call* send_call,
     const std::map<uint8_t, MediaType>& payload_type_map,
-    ArrayView<const RtpExtension> audio_extensions,
-    ArrayView<const RtpExtension> video_extensions)
+    std::span<const RtpExtension> audio_extensions,
+    std::span<const RtpExtension> video_extensions)
     : clock_(*Clock::GetRealTimeClock()),
       send_call_(send_call),
       task_queue_(task_queue),
@@ -80,8 +80,8 @@ DirectTransport::DirectTransport(
     std::unique_ptr<SimulatedPacketReceiverInterface> pipe,
     Call* send_call,
     const std::map<uint8_t, MediaType>& payload_type_map,
-    ArrayView<const RtpExtension> audio_extensions,
-    ArrayView<const RtpExtension> video_extensions)
+    std::span<const RtpExtension> audio_extensions,
+    std::span<const RtpExtension> video_extensions)
     : env_(env),
       clock_(env.clock()),
       send_call_(send_call),
@@ -101,7 +101,7 @@ void DirectTransport::SetReceiver(PacketReceiver* receiver) {
   fake_network_->SetReceiver(receiver);
 }
 
-bool DirectTransport::SendRtp(ArrayView<const uint8_t> data,
+bool DirectTransport::SendRtp(std::span<const uint8_t> data,
                               const PacketOptions& options) {
   if (send_call_) {
     SentPacketInfo sent_packet(options.packet_id, clock_.TimeInMilliseconds());
@@ -139,7 +139,7 @@ bool DirectTransport::SendRtp(ArrayView<const uint8_t> data,
   return true;
 }
 
-bool DirectTransport::SendRtcp(ArrayView<const uint8_t> data,
+bool DirectTransport::SendRtcp(std::span<const uint8_t> data,
                                const PacketOptions& /* options */) {
   fake_network_->DeliverRtcpPacket(CopyOnWriteBuffer(data));
   MutexLock lock(&process_lock_);
diff --git a/test/direct_transport.h b/test/direct_transport.h
index 9fba1ab1e6..79c0da6bb2 100644
--- a/test/direct_transport.h
+++ b/test/direct_transport.h
@@ -15,8 +15,8 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/call/transport.h"
 #include "api/environment/environment.h"
 #include "api/media_types.h"
@@ -57,16 +57,16 @@ class DirectTransport : public Transport {
                   std::unique_ptr<SimulatedPacketReceiverInterface> pipe,
                   Call* send_call,
                   const std::map<uint8_t, MediaType>& payload_type_map,
-                  ArrayView<const RtpExtension> audio_extensions,
-                  ArrayView<const RtpExtension> video_extensions);
+                  std::span<const RtpExtension> audio_extensions,
+                  std::span<const RtpExtension> video_extensions);
 
   DirectTransport(const Environment& env,
                   TaskQueueBase* task_queue,
                   std::unique_ptr<SimulatedPacketReceiverInterface> pipe,
                   Call* send_call,
                   const std::map<uint8_t, MediaType>& payload_type_map,
-                  ArrayView<const RtpExtension> audio_extensions,
-                  ArrayView<const RtpExtension> video_extensions);
+                  std::span<const RtpExtension> audio_extensions,
+                  std::span<const RtpExtension> video_extensions);
 
   ~DirectTransport() override;
 
@@ -78,9 +78,9 @@ class DirectTransport : public Transport {
   using Transport::SendRtcp;
   using Transport::SendRtp;
 
-  bool SendRtp(ArrayView<const uint8_t> data,
+  bool SendRtp(std::span<const uint8_t> data,
                const PacketOptions& options) override;
-  bool SendRtcp(ArrayView<const uint8_t> data,
+  bool SendRtcp(std::span<const uint8_t> data,
                 const PacketOptions& options) override;
 
   int GetAverageDelayMs();
diff --git a/test/fuzzers/BUILD.gn b/test/fuzzers/BUILD.gn
index 2a78afab87..e690727fd8 100644
--- a/test/fuzzers/BUILD.gn
+++ b/test/fuzzers/BUILD.gn
@@ -15,7 +15,6 @@ rtc_library("webrtc_fuzzer_main") {
   testonly = true
   deps = [
     ":fuzz_data_helper",
-    "../../api:array_view",
     "../../rtc_base:logging",
     "//testing/libfuzzer:fuzzing_engine_main",
   ]
@@ -99,7 +98,6 @@ webrtc_fuzzer_test("h264_depacketizer_fuzzer") {
 webrtc_fuzzer_test("vp8_depacketizer_fuzzer") {
   sources = [ "vp8_depacketizer_fuzzer.cc" ]
   deps = [
-    "../../api:array_view",
     "../../modules/rtp_rtcp",
     "../../modules/rtp_rtcp:rtp_video_header",
   ]
@@ -108,7 +106,6 @@ webrtc_fuzzer_test("vp8_depacketizer_fuzzer") {
 webrtc_fuzzer_test("vp9_depacketizer_fuzzer") {
   sources = [ "vp9_depacketizer_fuzzer.cc" ]
   deps = [
-    "../../api:array_view",
     "../../modules/rtp_rtcp",
     "../../modules/rtp_rtcp:rtp_video_header",
   ]
@@ -117,7 +114,6 @@ webrtc_fuzzer_test("vp9_depacketizer_fuzzer") {
 webrtc_fuzzer_test("vp8_qp_parser_fuzzer") {
   sources = [ "vp8_qp_parser_fuzzer.cc" ]
   deps = [
-    "../../api:array_view",
     "../../modules/video_coding:video_coding_utility",
     "../../modules/video_coding/",
   ]
@@ -126,7 +122,6 @@ webrtc_fuzzer_test("vp8_qp_parser_fuzzer") {
 webrtc_fuzzer_test("vp9_qp_parser_fuzzer") {
   sources = [ "vp9_qp_parser_fuzzer.cc" ]
   deps = [
-    "../../api:array_view",
     "../../modules/video_coding:video_coding_utility",
     "../../modules/video_coding/",
   ]
@@ -135,7 +130,6 @@ webrtc_fuzzer_test("vp9_qp_parser_fuzzer") {
 webrtc_fuzzer_test("h264_bitstream_parser_fuzzer") {
   sources = [ "h264_bitstream_parser_fuzzer.cc" ]
   deps = [
-    "../../api:array_view",
     "../../common_video",
     "../../modules/video_coding/",
   ]
@@ -145,7 +139,6 @@ if (rtc_use_h265) {
   webrtc_fuzzer_test("h265_bitstream_parser_fuzzer") {
     sources = [ "h265_bitstream_parser_fuzzer.cc" ]
     deps = [
-      "../../api:array_view",
       "../../common_video",
       "../../modules/video_coding/",
     ]
@@ -154,7 +147,6 @@ if (rtc_use_h265) {
   webrtc_fuzzer_test("h265_depacketizer_fuzzer") {
     sources = [ "h265_depacketizer_fuzzer.cc" ]
     deps = [
-      "../../api:array_view",
       "../../modules/rtp_rtcp",
       "../../rtc_base:copy_on_write_buffer",
     ]
@@ -164,7 +156,6 @@ if (rtc_use_h265) {
 webrtc_fuzzer_test("forward_error_correction_fuzzer") {
   sources = [ "forward_error_correction_fuzzer.cc" ]
   deps = [
-    "../../api:array_view",
     "../../api:scoped_refptr",
     "../../modules/rtp_rtcp",
     "../../modules/rtp_rtcp:rtp_rtcp_format",
@@ -175,7 +166,6 @@ webrtc_fuzzer_test("forward_error_correction_fuzzer") {
 webrtc_fuzzer_test("flexfec_header_reader_fuzzer") {
   sources = [ "flexfec_header_reader_fuzzer.cc" ]
   deps = [
-    "../../api:array_view",
     "../../api:scoped_refptr",
     "../../modules/rtp_rtcp",
     "../../modules/rtp_rtcp:rtp_rtcp_format",
@@ -198,7 +188,6 @@ webrtc_fuzzer_test("flexfec_sender_fuzzer") {
 webrtc_fuzzer_test("ulpfec_header_reader_fuzzer") {
   sources = [ "ulpfec_header_reader_fuzzer.cc" ]
   deps = [
-    "../../api:array_view",
     "../../api:scoped_refptr",
     "../../modules/rtp_rtcp",
     "../../modules/rtp_rtcp:fec_test_helper",
@@ -226,7 +215,6 @@ webrtc_fuzzer_test("ulpfec_receiver_fuzzer") {
   sources = [ "ulpfec_receiver_fuzzer.cc" ]
   deps = [
     ":fuzz_data_helper",
-    "../../api:array_view",
     "../../modules/rtp_rtcp",
     "../../modules/rtp_rtcp:rtp_rtcp_format",
     "../../system_wrappers",
@@ -252,7 +240,6 @@ webrtc_fuzzer_test("rtp_video_frame_assembler_fuzzer") {
 webrtc_fuzzer_test("rtcp_receiver_fuzzer") {
   sources = [ "rtcp_receiver_fuzzer.cc" ]
   deps = [
-    "../../api:array_view",
     "../../api:field_trials",
     "../../api/environment",
     "../../api/environment:environment_factory",
@@ -279,7 +266,6 @@ webrtc_fuzzer_test("rtp_packet_fuzzer") {
 webrtc_fuzzer_test("rtp_packetizer_av1_fuzzer") {
   sources = [ "rtp_packetizer_av1_fuzzer.cc" ]
   deps = [
-    "../../api:array_view",
     "../../api/video:video_frame_type",
     "../../modules/rtp_rtcp",
     "utils:validate_rtp_packetizer",
@@ -290,7 +276,6 @@ if (rtc_use_h265) {
   webrtc_fuzzer_test("rtp_packetizer_h265_fuzzer") {
     sources = [ "rtp_packetizer_h265_fuzzer.cc" ]
     deps = [
-      "../../api:array_view",
       "../../api/video:video_frame_type",
       "../../modules/rtp_rtcp",
       "utils:validate_rtp_packetizer",
@@ -301,7 +286,6 @@ if (rtc_use_h265) {
 webrtc_fuzzer_test("rtp_format_h264_fuzzer") {
   sources = [ "rtp_format_h264_fuzzer.cc" ]
   deps = [
-    "../../api:array_view",
     "../../modules/rtp_rtcp",
     "../../modules/video_coding:codec_globals_headers",
     "utils:validate_rtp_packetizer",
@@ -311,7 +295,6 @@ webrtc_fuzzer_test("rtp_format_h264_fuzzer") {
 webrtc_fuzzer_test("rtp_format_vp8_fuzzer") {
   sources = [ "rtp_format_vp8_fuzzer.cc" ]
   deps = [
-    "../../api:array_view",
     "../../modules/rtp_rtcp",
     "../../modules/video_coding:codec_globals_headers",
     "utils:validate_rtp_packetizer",
@@ -321,7 +304,6 @@ webrtc_fuzzer_test("rtp_format_vp8_fuzzer") {
 webrtc_fuzzer_test("rtp_format_vp9_fuzzer") {
   sources = [ "rtp_format_vp9_fuzzer.cc" ]
   deps = [
-    "../../api:array_view",
     "../../modules/rtp_rtcp",
     "../../modules/video_coding:codec_globals_headers",
     "utils:validate_rtp_packetizer",
@@ -331,7 +313,6 @@ webrtc_fuzzer_test("rtp_format_vp9_fuzzer") {
 webrtc_fuzzer_test("receive_side_congestion_controller_fuzzer") {
   sources = [ "receive_side_congestion_controller_fuzzer.cc" ]
   deps = [
-    "../../api:array_view",
     "../../api:rtp_parameters",
     "../../api/environment:environment_factory",
     "../../api/units:time_delta",
@@ -351,7 +332,6 @@ rtc_library("audio_decoder_fuzzer") {
   ]
   deps = [
     ":fuzz_data_helper",
-    "../../api:array_view",
     "../../api/audio_codecs:audio_codecs_api",
     "../../modules/rtp_rtcp:rtp_rtcp_format",
     "../../rtc_base:checks",
@@ -388,7 +368,6 @@ webrtc_fuzzer_test("audio_decoder_multiopus_fuzzer") {
   sources = [ "audio_decoder_multistream_opus_fuzzer.cc" ]
   deps = [
     ":audio_decoder_fuzzer",
-    "../../api:array_view",
     "../../api/audio_codecs/opus:audio_decoder_multiopus",
     "../../api/audio_codecs/opus:audio_decoder_opus_config",
     "../../rtc_base:checks",
@@ -420,7 +399,6 @@ rtc_library("audio_encoder_fuzzer") {
   ]
   deps = [
     ":fuzz_data_helper",
-    "../../api:array_view",
     "../../api/audio_codecs:audio_codecs_api",
     "../../rtc_base:buffer",
     "../../rtc_base:checks",
@@ -442,7 +420,6 @@ webrtc_fuzzer_test("audio_encoder_opus_fuzzer") {
 webrtc_fuzzer_test("turn_unwrap_fuzzer") {
   sources = [ "turn_unwrap_fuzzer.cc" ]
   deps = [
-    "../../api:array_view",
     "../../media",
     "../../media:turn_utils",
   ]
@@ -451,7 +428,6 @@ webrtc_fuzzer_test("turn_unwrap_fuzzer") {
 webrtc_fuzzer_test("neteq_rtp_fuzzer") {
   sources = [ "neteq_rtp_fuzzer.cc" ]
   deps = [
-    "../../api:array_view",
     "../../api:rtp_headers",
     "../../api/audio_codecs:audio_codecs_api",
     "../../api/audio_codecs:builtin_audio_decoder_factory",
@@ -468,7 +444,6 @@ webrtc_fuzzer_test("neteq_rtp_fuzzer") {
 webrtc_fuzzer_test("neteq_signal_fuzzer") {
   sources = [ "neteq_signal_fuzzer.cc" ]
   deps = [
-    "../../api:array_view",
     "../../api:rtp_headers",
     "../../api/audio_codecs:audio_codecs_api",
     "../../api/audio_codecs:builtin_audio_decoder_factory",
@@ -532,7 +507,6 @@ if (!build_with_chromium) {
 webrtc_fuzzer_test("stun_parser_fuzzer") {
   sources = [ "stun_parser_fuzzer.cc" ]
   deps = [
-    "../../api:array_view",
     "../../api/transport:stun_types",
     "../../rtc_base:byte_buffer",
   ]
@@ -558,7 +532,6 @@ rtc_library("audio_processing_fuzzer_helper") {
   ]
   deps = [
     ":fuzz_data_helper",
-    "../../api:array_view",
     "../../api:scoped_refptr",
     "../../api/audio:audio_frame_api",
     "../../api/audio:audio_processing",
@@ -572,7 +545,6 @@ webrtc_fuzzer_test("audio_processing_fuzzer") {
   sources = [ "audio_processing_configs_fuzzer.cc" ]
   deps = [
     ":audio_processing_fuzzer_helper",
-    "../../api:array_view",
     "../../api:field_trials",
     "../../api:scoped_refptr",
     "../../api/audio:aec3_factory",
@@ -594,7 +566,6 @@ webrtc_fuzzer_test("audio_processing_fuzzer") {
 webrtc_fuzzer_test("audio_processing_sample_rate_fuzzer") {
   sources = [ "audio_processing_sample_rate_fuzzer.cc" ]
   deps = [
-    "../../api:array_view",
     "../../api:scoped_refptr",
     "../../api/audio:audio_frame_api",
     "../../api/audio:audio_processing",
@@ -612,7 +583,6 @@ webrtc_fuzzer_test("agc_fuzzer") {
   sources = [ "agc_fuzzer.cc" ]
   deps = [
     ":fuzz_data_helper",
-    "../../api:array_view",
     "../../api/audio:audio_processing",
     "../../modules/audio_processing",
     "../../modules/audio_processing:audio_buffer",
@@ -645,7 +615,6 @@ webrtc_fuzzer_test("aec3_fuzzer") {
   sources = [ "aec3_fuzzer.cc" ]
   deps = [
     ":fuzz_data_helper",
-    "../../api:array_view",
     "../../api/audio:aec3_config",
     "../../api/audio:audio_processing",
     "../../api/environment:environment_factory",
@@ -657,7 +626,6 @@ webrtc_fuzzer_test("aec3_fuzzer") {
 webrtc_fuzzer_test("comfort_noise_decoder_fuzzer") {
   sources = [ "comfort_noise_decoder_fuzzer.cc" ]
   deps = [
-    "../../api:array_view",
     "../../modules/audio_coding:webrtc_cng",
     "../../rtc_base:buffer",
   ]
@@ -668,7 +636,6 @@ webrtc_fuzzer_test("rtp_depacketizer_av1_assemble_frame_fuzzer") {
   seed_corpus = "corpora/rtp-depacketizer-av1-assemble-frame-corpus"
   deps = [
     ":fuzz_data_helper",
-    "../../api:array_view",
     "../../modules/rtp_rtcp",
   ]
 }
@@ -677,7 +644,6 @@ webrtc_fuzzer_test("rtp_dependency_descriptor_fuzzer") {
   sources = [ "rtp_dependency_descriptor_fuzzer.cc" ]
   seed_corpus = "corpora/dependency_descriptor-corpus"
   deps = [
-    "../../api:array_view",
     "../../api/transport/rtp:dependency_descriptor",
     "../../common_video/generic_frame_descriptor",
     "../../modules/rtp_rtcp:rtp_rtcp_format",
@@ -689,7 +655,6 @@ webrtc_fuzzer_test("rtp_video_layers_allocation_fuzzer") {
   sources = [ "rtp_video_layers_allocation_fuzzer.cc" ]
   seed_corpus = "corpora/video_layers_allocation-corpus"
   deps = [
-    "../../api:array_view",
     "../../api/video:video_layers_allocation",
     "../../modules/rtp_rtcp:rtp_rtcp_format",
     "../../rtc_base:checks",
@@ -721,7 +686,6 @@ webrtc_fuzzer_test("frame_buffer_fuzzer") {
   sources = [ "frame_buffer_fuzzer.cc" ]
   deps = [
     ":fuzz_data_helper",
-    "../../api:array_view",
     "../../api:field_trials",
     "../../api/video:encoded_frame",
     "../../api/video:frame_buffer",
@@ -759,7 +723,6 @@ webrtc_fuzzer_test("sctp_utils_fuzzer") {
 webrtc_fuzzer_test("dcsctp_socket_fuzzer") {
   sources = [ "dcsctp_socket_fuzzer.cc" ]
   deps = [
-    "../../api:array_view",
     "../../net/dcsctp/fuzzers:dcsctp_fuzzers",
     "../../net/dcsctp/public:socket",
     "../../net/dcsctp/public:types",
@@ -782,7 +745,6 @@ webrtc_fuzzer_test("ssl_certificate_fuzzer") {
 webrtc_fuzzer_test("vp8_replay_fuzzer") {
   sources = [ "vp8_replay_fuzzer.cc" ]
   deps = [
-    "../../api:array_view",
     "../../api/video_codecs:video_codecs_api",
     "../../call:video_receive_stream_api",
     "../../modules/video_coding:codec_globals_headers",
@@ -796,7 +758,6 @@ if (rtc_build_libvpx) {
     sources = [ "vp9_encoder_references_fuzzer.cc" ]
     deps = [
       "..:test_support",
-      "../../api:array_view",
       "../../api:field_trials_view",
       "../../api/environment:environment_factory",
       "../../api/video:encoded_image",
@@ -825,7 +786,6 @@ if (rtc_build_libvpx) {
 webrtc_fuzzer_test("vp9_replay_fuzzer") {
   sources = [ "vp9_replay_fuzzer.cc" ]
   deps = [
-    "../../api:array_view",
     "../../api/video_codecs:video_codecs_api",
     "../../call:video_receive_stream_api",
     "utils:rtp_replayer",
@@ -841,7 +801,6 @@ webrtc_fuzzer_test("webrtc_base64_decode_fuzzer") {
 webrtc_fuzzer_test("webrtc_base64_encode_fuzzer") {
   sources = [ "base64_encode_fuzzer.cc" ]
   deps = [
-    "../../api:array_view",
     "../../rtc_base:base64",
     "../../rtc_base:checks",
   ]
diff --git a/test/fuzzers/agc_fuzzer.cc b/test/fuzzers/agc_fuzzer.cc
index 7b6aca1198..a8353c592a 100644
--- a/test/fuzzers/agc_fuzzer.cc
+++ b/test/fuzzers/agc_fuzzer.cc
@@ -13,7 +13,6 @@
 #include <memory>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/audio_processing.h"
 #include "modules/audio_processing/agc/gain_control.h"
 #include "modules/audio_processing/audio_buffer.h"
diff --git a/test/fuzzers/audio_decoder_fuzzer.cc b/test/fuzzers/audio_decoder_fuzzer.cc
index 84d84171db..753e4e74e5 100644
--- a/test/fuzzers/audio_decoder_fuzzer.cc
+++ b/test/fuzzers/audio_decoder_fuzzer.cc
@@ -13,8 +13,8 @@
 #include <cstddef>
 #include <cstdint>
 #include <limits>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_decoder.h"
 #include "modules/rtp_rtcp/source/byte_io.h"
 #include "test/fuzzers/fuzz_data_helper.h"
@@ -49,7 +49,7 @@ void FuzzAudioDecoder(DecoderFunctionType decode_type,
                       int sample_rate_hz,
                       size_t max_decoded_bytes,
                       int16_t* decoded) {
-  ArrayView<const uint8_t> data = fuzz_data.ReadRemaining();
+  std::span<const uint8_t> data = fuzz_data.ReadRemaining();
   const uint8_t* data_ptr = data.data();
   size_t remaining_size = data.size();
   size_t packet_len;
diff --git a/test/fuzzers/audio_encoder_fuzzer.cc b/test/fuzzers/audio_encoder_fuzzer.cc
index a0bdcfd596..54cbd9bec9 100644
--- a/test/fuzzers/audio_encoder_fuzzer.cc
+++ b/test/fuzzers/audio_encoder_fuzzer.cc
@@ -14,7 +14,6 @@
 #include <cstring>
 #include <memory>
 
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/checks.h"
diff --git a/test/fuzzers/audio_processing_fuzzer_helper.cc b/test/fuzzers/audio_processing_fuzzer_helper.cc
index dffe40d1fa..56e947f711 100644
--- a/test/fuzzers/audio_processing_fuzzer_helper.cc
+++ b/test/fuzzers/audio_processing_fuzzer_helper.cc
@@ -16,8 +16,8 @@
 #include <cstddef>
 #include <cstdint>
 #include <cstring>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/audio/audio_frame.h"
 #include "api/audio/audio_processing.h"
 #include "api/scoped_refptr.h"
@@ -42,7 +42,7 @@ void GenerateFloatFrame(FuzzDataHelper* fuzz_data,
     std::fill(float_frames[i], float_frames[i] + samples_per_input_channel, 0);
     const size_t read_bytes = sizeof(float) * samples_per_input_channel;
     if (fuzz_data->CanReadBytes(read_bytes)) {
-      ArrayView<const uint8_t> byte_array =
+      std::span<const uint8_t> byte_array =
           fuzz_data->ReadByteArray(read_bytes);
       memmove(float_frames[i], byte_array.data(), read_bytes);
     }
diff --git a/test/fuzzers/base64_encode_fuzzer.cc b/test/fuzzers/base64_encode_fuzzer.cc
index 5edaa6ede9..46693f78ab 100644
--- a/test/fuzzers/base64_encode_fuzzer.cc
+++ b/test/fuzzers/base64_encode_fuzzer.cc
@@ -11,9 +11,9 @@
 #include <cstdint>
 #include <cstring>
 #include <optional>
+#include <span>
 #include <string>
 
-#include "api/array_view.h"
 #include "rtc_base/base64.h"
 #include "rtc_base/checks.h"
 #include "test/fuzzers/fuzz_data_helper.h"
@@ -21,7 +21,7 @@
 namespace webrtc {
 
 void FuzzOneInput(FuzzDataHelper fuzz_data) {
-  ArrayView<const uint8_t> raw = fuzz_data.ReadRemaining();
+  std::span<const uint8_t> raw = fuzz_data.ReadRemaining();
   std::optional<std::string> decoded_encoded_data =
       Base64Decode(Base64Encode(raw));
   RTC_CHECK(decoded_encoded_data.has_value());
diff --git a/test/fuzzers/comfort_noise_decoder_fuzzer.cc b/test/fuzzers/comfort_noise_decoder_fuzzer.cc
index 6ee89e2eec..147a98694d 100644
--- a/test/fuzzers/comfort_noise_decoder_fuzzer.cc
+++ b/test/fuzzers/comfort_noise_decoder_fuzzer.cc
@@ -11,8 +11,8 @@
 #include <algorithm>
 #include <cstddef>
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_coding/codecs/cng/webrtc_cng.h"
 #include "rtc_base/buffer.h"
 #include "test/fuzzers/fuzz_data_helper.h"
@@ -21,7 +21,7 @@ namespace webrtc {
 namespace test {
 namespace {
 
-void FuzzOneInputTest(webrtc::ArrayView<const uint8_t> data) {
+void FuzzOneInputTest(std::span<const uint8_t> data) {
   FuzzDataHelper fuzz_data(data);
   ComfortNoiseDecoder cng_decoder;
 
@@ -44,7 +44,7 @@ void FuzzOneInputTest(webrtc::ArrayView<const uint8_t> data) {
     BufferT<int16_t> output = BufferT<int16_t>::CreateWithCapacity(output_size);
     for (size_t i = 0; i < num_generate_calls; ++i) {
       output.SetSize(0);
-      output.AppendData(output_size, [&](ArrayView<int16_t> out) {
+      output.AppendData(output_size, [&](std::span<int16_t> out) {
         cng_decoder.Generate(out, new_period);
         return output_size;
       });
diff --git a/test/fuzzers/dcsctp_packet_fuzzer.cc b/test/fuzzers/dcsctp_packet_fuzzer.cc
index a65c8f20b0..2c365e7f93 100644
--- a/test/fuzzers/dcsctp_packet_fuzzer.cc
+++ b/test/fuzzers/dcsctp_packet_fuzzer.cc
@@ -10,7 +10,6 @@
 #include <cstddef>
 #include <optional>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/chunk/chunk.h"
 #include "net/dcsctp/packet/sctp_packet.h"
 #include "test/fuzzers/fuzz_data_helper.h"
diff --git a/test/fuzzers/flexfec_header_reader_fuzzer.cc b/test/fuzzers/flexfec_header_reader_fuzzer.cc
index 1f397f29c1..9531b14d65 100644
--- a/test/fuzzers/flexfec_header_reader_fuzzer.cc
+++ b/test/fuzzers/flexfec_header_reader_fuzzer.cc
@@ -12,8 +12,8 @@
 #include <cstddef>
 #include <cstdint>
 #include <cstring>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/scoped_refptr.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/flexfec_header_reader_writer.h"
@@ -30,7 +30,7 @@ void FuzzOneInput(FuzzDataHelper fuzz_data) {
   packet.pkt = webrtc::scoped_refptr<Packet>(new Packet());
   const size_t packet_size =
       std::min(fuzz_data.size(), static_cast<size_t>(IP_PACKET_SIZE));
-  ArrayView<const uint8_t> raw = fuzz_data.ReadByteArray(packet_size);
+  std::span<const uint8_t> raw = fuzz_data.ReadByteArray(packet_size);
   packet.pkt->data.EnsureCapacity(IP_PACKET_SIZE);
   packet.pkt->data.SetData(raw.data(), raw.size());
 
diff --git a/test/fuzzers/forward_error_correction_fuzzer.cc b/test/fuzzers/forward_error_correction_fuzzer.cc
index 588fbdf139..79719f0c81 100644
--- a/test/fuzzers/forward_error_correction_fuzzer.cc
+++ b/test/fuzzers/forward_error_correction_fuzzer.cc
@@ -12,8 +12,8 @@
 #include <cstdint>
 #include <cstring>
 #include <memory>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/scoped_refptr.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/byte_io.h"
@@ -83,7 +83,7 @@ void FuzzOneInput(FuzzDataHelper fuzz_data) {
   uint8_t packet_loss;
   while (true) {
     if (!fuzz_buffer.ReadBytes(
-            webrtc::ArrayView<uint8_t>(packet_buffer, kPacketSize))) {
+            std::span<uint8_t>(packet_buffer, kPacketSize))) {
       return;
     }
     if (!fuzz_buffer.ReadUInt8(&reordering))
diff --git a/test/fuzzers/neteq_rtp_fuzzer.cc b/test/fuzzers/neteq_rtp_fuzzer.cc
index b62e580650..5d13d85d0a 100644
--- a/test/fuzzers/neteq_rtp_fuzzer.cc
+++ b/test/fuzzers/neteq_rtp_fuzzer.cc
@@ -15,10 +15,10 @@
 #include <limits>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
@@ -43,12 +43,12 @@ class SineGenerator : public EncodeNetEqInput::Generator {
   explicit SineGenerator(int sample_rate_hz)
       : sample_rate_hz_(sample_rate_hz) {}
 
-  webrtc::ArrayView<const int16_t> Generate(size_t num_samples) override {
+  std::span<const int16_t> Generate(size_t num_samples) override {
     if (samples_.size() < num_samples) {
       samples_.resize(num_samples);
     }
 
-    webrtc::ArrayView<int16_t> output(samples_.data(), num_samples);
+    std::span<int16_t> output(samples_.data(), num_samples);
     for (auto& x : output) {
       x = static_cast<int16_t>(2000.0 * std::sin(phase_));
       phase_ += 2 * kPi * kFreqHz / sample_rate_hz_;
@@ -66,7 +66,7 @@ class SineGenerator : public EncodeNetEqInput::Generator {
 
 class FuzzRtpInput : public NetEqInput {
  public:
-  explicit FuzzRtpInput(webrtc::ArrayView<const uint8_t> data) : data_(data) {
+  explicit FuzzRtpInput(std::span<const uint8_t> data) : data_(data) {
     AudioEncoderPcm16B::Config config;
     config.payload_type = kPayloadType;
     config.sample_rate_hz = 32000;
@@ -165,7 +165,7 @@ class FuzzRtpInput : public NetEqInput {
   }
 
   bool ended_ = false;
-  webrtc::ArrayView<const uint8_t> data_;
+  std::span<const uint8_t> data_;
   size_t data_ix_ = 0;
   std::unique_ptr<EncodeNetEqInput> input_;
   std::unique_ptr<RtpPacketReceived> packet_;
diff --git a/test/fuzzers/neteq_signal_fuzzer.cc b/test/fuzzers/neteq_signal_fuzzer.cc
index 477227c386..8b828de7da 100644
--- a/test/fuzzers/neteq_signal_fuzzer.cc
+++ b/test/fuzzers/neteq_signal_fuzzer.cc
@@ -16,10 +16,10 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
@@ -50,12 +50,12 @@ class SineAndNoiseGenerator : public EncodeNetEqInput::Generator {
         noise_generator_(fuzz_data_.ReadOrDefaultValueNotZero<uint64_t>(1)) {}
 
   // Generates num_samples of the sine-gaussian mixture.
-  webrtc::ArrayView<const int16_t> Generate(size_t num_samples) override {
+  std::span<const int16_t> Generate(size_t num_samples) override {
     if (samples_.size() < num_samples) {
       samples_.resize(num_samples);
     }
 
-    webrtc::ArrayView<int16_t> output(samples_.data(), num_samples);
+    std::span<int16_t> output(samples_.data(), num_samples);
     // Randomize an amplitude between 0 and 32768; use 65000/2 if we are out of
     // fuzzer data.
     const float amplitude = fuzz_data_.ReadOrDefaultValue<uint16_t>(65000) / 2;
diff --git a/test/fuzzers/rtcp_receiver_fuzzer.cc b/test/fuzzers/rtcp_receiver_fuzzer.cc
index 9eebc417ce..e856ab3a4a 100644
--- a/test/fuzzers/rtcp_receiver_fuzzer.cc
+++ b/test/fuzzers/rtcp_receiver_fuzzer.cc
@@ -10,9 +10,9 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/environment/environment_factory.h"
 #include "api/field_trials.h"
 #include "modules/rtp_rtcp/include/report_block_data.h"
@@ -37,7 +37,7 @@ class NullModuleRtpRtcp : public RTCPReceiver::ModuleRtpRtcp {
   void OnRequestSendReport() override {}
   void OnReceivedNack(const std::vector<uint16_t>&) override {}
   void OnReceivedRtcpReportBlocks(
-      webrtc::ArrayView<const ReportBlockData> report_blocks) override {}
+      std::span<const ReportBlockData> report_blocks) override {}
 };
 
 }  // namespace
diff --git a/test/fuzzers/rtp_depacketizer_av1_assemble_frame_fuzzer.cc b/test/fuzzers/rtp_depacketizer_av1_assemble_frame_fuzzer.cc
index d4049b93a0..9f2cc0c9e3 100644
--- a/test/fuzzers/rtp_depacketizer_av1_assemble_frame_fuzzer.cc
+++ b/test/fuzzers/rtp_depacketizer_av1_assemble_frame_fuzzer.cc
@@ -10,15 +10,15 @@
 #include <stddef.h>
 #include <stdint.h>
 
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/rtp_rtcp/source/video_rtp_depacketizer_av1.h"
 #include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
 void FuzzOneInput(FuzzDataHelper fuzz_data) {
-  std::vector<webrtc::ArrayView<const uint8_t>> rtp_payloads;
+  std::vector<std::span<const uint8_t>> rtp_payloads;
 
   // Convert plain array of bytes into array of array bytes.
   while (fuzz_data.CanReadBytes(sizeof(uint16_t))) {
diff --git a/test/fuzzers/rtp_dependency_descriptor_fuzzer.cc b/test/fuzzers/rtp_dependency_descriptor_fuzzer.cc
index 55d6ba8a37..f02ff5ed1d 100644
--- a/test/fuzzers/rtp_dependency_descriptor_fuzzer.cc
+++ b/test/fuzzers/rtp_dependency_descriptor_fuzzer.cc
@@ -12,9 +12,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <utility>
 
-#include "api/array_view.h"
 #include "api/transport/rtp/dependency_descriptor.h"
 #include "modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h"
 #include "rtc_base/checks.h"
@@ -57,7 +57,7 @@ void FuzzOneInput(FuzzDataHelper fuzz_data) {
     uint8_t some_memory[256];
     // That should be true because value_size <= next_size < 256
     RTC_CHECK_LT(value_size, 256);
-    webrtc::ArrayView<uint8_t> write_buffer(some_memory, value_size);
+    std::span<uint8_t> write_buffer(some_memory, value_size);
     RTC_CHECK(RtpDependencyDescriptorExtension::Write(write_buffer, structure1,
                                                       descriptor1));
 
diff --git a/test/fuzzers/rtp_video_layers_allocation_fuzzer.cc b/test/fuzzers/rtp_video_layers_allocation_fuzzer.cc
index cf5b4fe3fb..b403681a87 100644
--- a/test/fuzzers/rtp_video_layers_allocation_fuzzer.cc
+++ b/test/fuzzers/rtp_video_layers_allocation_fuzzer.cc
@@ -11,8 +11,8 @@
 #include <cstddef>
 #include <cstdint>
 #include <limits>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/video/video_layers_allocation.h"
 #include "modules/rtp_rtcp/source/rtp_video_layers_allocation_extension.h"
 #include "rtc_base/checks.h"
@@ -29,7 +29,7 @@ void FuzzOneInput(FuzzDataHelper fuzz_data) {
   if (fuzz_data.size() > kMaxSize) {
     return;
   }
-  ArrayView<const uint8_t> raw = fuzz_data.ReadRemaining();
+  std::span<const uint8_t> raw = fuzz_data.ReadRemaining();
 
   VideoLayersAllocation allocation1;
   if (!RtpVideoLayersAllocationExtension::Parse(raw, &allocation1)) {
@@ -44,7 +44,7 @@ void FuzzOneInput(FuzzDataHelper fuzz_data) {
   RTC_CHECK_LE(value_size, raw.size());
   uint8_t some_memory[kMaxSize];
   RTC_CHECK_LE(value_size, kMaxSize);
-  webrtc::ArrayView<uint8_t> write_buffer(some_memory, value_size);
+  std::span<uint8_t> write_buffer(some_memory, value_size);
   RTC_CHECK(
       RtpVideoLayersAllocationExtension::Write(write_buffer, allocation1));
 
diff --git a/test/fuzzers/turn_unwrap_fuzzer.cc b/test/fuzzers/turn_unwrap_fuzzer.cc
index 3f3670edc8..27d80cfd58 100644
--- a/test/fuzzers/turn_unwrap_fuzzer.cc
+++ b/test/fuzzers/turn_unwrap_fuzzer.cc
@@ -11,7 +11,8 @@
 #include <stddef.h>
 #include <stdint.h>
 
-#include "api/array_view.h"
+#include <span>
+
 #include "media/base/turn_utils.h"
 #include "test/fuzzers/fuzz_data_helper.h"
 
@@ -19,7 +20,7 @@ namespace webrtc {
 void FuzzOneInput(FuzzDataHelper fuzz_data) {
   size_t content_position;
   size_t content_size;
-  ArrayView<const uint8_t> raw = fuzz_data.ReadRemaining();
+  std::span<const uint8_t> raw = fuzz_data.ReadRemaining();
   webrtc::UnwrapTurnPacket(raw.data(), raw.size(), &content_position,
                            &content_size);
 }
diff --git a/test/fuzzers/ulpfec_header_reader_fuzzer.cc b/test/fuzzers/ulpfec_header_reader_fuzzer.cc
index aa7a0027a3..97abe357ed 100644
--- a/test/fuzzers/ulpfec_header_reader_fuzzer.cc
+++ b/test/fuzzers/ulpfec_header_reader_fuzzer.cc
@@ -12,8 +12,8 @@
 #include <cstddef>
 #include <cstdint>
 #include <cstring>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/scoped_refptr.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/forward_error_correction.h"
@@ -30,7 +30,7 @@ void FuzzOneInput(FuzzDataHelper fuzz_data) {
   packet.pkt = webrtc::scoped_refptr<Packet>(new Packet());
   const size_t packet_size =
       std::min(fuzz_data.size(), static_cast<size_t>(IP_PACKET_SIZE));
-  ArrayView<const uint8_t> raw = fuzz_data.ReadByteArray(packet_size);
+  std::span<const uint8_t> raw = fuzz_data.ReadByteArray(packet_size);
   packet.pkt->data.EnsureCapacity(IP_PACKET_SIZE);
   packet.pkt->data.SetData(raw.data(), raw.size());
 
diff --git a/test/fuzzers/ulpfec_receiver_fuzzer.cc b/test/fuzzers/ulpfec_receiver_fuzzer.cc
index 6e70da957b..827d8720b1 100644
--- a/test/fuzzers/ulpfec_receiver_fuzzer.cc
+++ b/test/fuzzers/ulpfec_receiver_fuzzer.cc
@@ -11,7 +11,6 @@
 #include <cstddef>
 #include <cstdint>
 
-#include "api/array_view.h"
 #include "modules/rtp_rtcp/include/recovered_packet_receiver.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
diff --git a/test/fuzzers/vp8_qp_parser_fuzzer.cc b/test/fuzzers/vp8_qp_parser_fuzzer.cc
index 7c771e905f..f60df078d1 100644
--- a/test/fuzzers/vp8_qp_parser_fuzzer.cc
+++ b/test/fuzzers/vp8_qp_parser_fuzzer.cc
@@ -9,15 +9,15 @@
  */
 #include <cstddef>
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/video_coding/utility/vp8_header_parser.h"
 #include "test/fuzzers/fuzz_data_helper.h"
 
 namespace webrtc {
 void FuzzOneInput(FuzzDataHelper fuzz_data) {
   int qp;
-  ArrayView<const uint8_t> raw = fuzz_data.ReadRemaining();
+  std::span<const uint8_t> raw = fuzz_data.ReadRemaining();
   vp8::GetQp(raw.data(), raw.size(), &qp);
 }
 }  // namespace webrtc
diff --git a/test/fuzzers/vp8_replay_fuzzer.cc b/test/fuzzers/vp8_replay_fuzzer.cc
index 8f95050b53..153c222bb8 100644
--- a/test/fuzzers/vp8_replay_fuzzer.cc
+++ b/test/fuzzers/vp8_replay_fuzzer.cc
@@ -12,10 +12,10 @@
 #include <stdint.h>
 
 #include <memory>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/video_codecs/sdp_video_format.h"
 #include "call/video_receive_stream.h"
 #include "test/fuzzers/fuzz_data_helper.h"
@@ -40,7 +40,7 @@ void FuzzOneInput(FuzzDataHelper fuzz_data) {
   std::vector<VideoReceiveStreamInterface::Config> replay_configs;
   replay_configs.push_back(std::move(vp8_config));
 
-  ArrayView<const uint8_t> raw = fuzz_data.ReadRemaining();
+  std::span<const uint8_t> raw = fuzz_data.ReadRemaining();
   test::RtpReplayer::Replay(std::move(stream_state), std::move(replay_configs),
                             raw.data(), raw.size());
 }
diff --git a/test/fuzzers/vp9_encoder_references_fuzzer.cc b/test/fuzzers/vp9_encoder_references_fuzzer.cc
index 35b58628a4..cb84d5c274 100644
--- a/test/fuzzers/vp9_encoder_references_fuzzer.cc
+++ b/test/fuzzers/vp9_encoder_references_fuzzer.cc
@@ -18,7 +18,6 @@
 #include "absl/algorithm/container.h"
 #include "absl/container/inlined_vector.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/environment/environment_factory.h"
 #include "api/field_trials_view.h"
 #include "api/video/encoded_image.h"
@@ -128,9 +127,8 @@ class FrameValidator : public EncodedImageCallback {
     }
   }
 
-  void CheckGenericReferences(
-      webrtc::ArrayView<const int64_t> frame_dependencies,
-      const GenericFrameInfo& generic_info) const {
+  void CheckGenericReferences(std::span<const int64_t> frame_dependencies,
+                              const GenericFrameInfo& generic_info) const {
     for (int64_t dependency_frame_id : frame_dependencies) {
       RTC_CHECK_GE(dependency_frame_id, 0);
       const LayerFrame& dependency = Frame(dependency_frame_id);
@@ -140,7 +138,7 @@ class FrameValidator : public EncodedImageCallback {
   }
 
   void CheckGenericAndCodecSpecificReferencesAreConsistent(
-      webrtc::ArrayView<const int64_t> frame_dependencies,
+      std::span<const int64_t> frame_dependencies,
       const CodecSpecificInfo& info,
       const LayerFrame& layer_frame) const {
     const CodecSpecificInfoVP9& vp9_info = info.codecSpecific.VP9;
@@ -148,8 +146,7 @@ class FrameValidator : public EncodedImageCallback {
 
     RTC_CHECK_EQ(generic_info.spatial_id, layer_frame.spatial_id);
     RTC_CHECK_EQ(generic_info.temporal_id, layer_frame.temporal_id);
-    auto picture_id_diffs =
-        webrtc::MakeArrayView(vp9_info.p_diff, vp9_info.num_ref_pics);
+    auto picture_id_diffs = std::span(vp9_info.p_diff, vp9_info.num_ref_pics);
     RTC_CHECK_EQ(
         frame_dependencies.size(),
         picture_id_diffs.size() + (vp9_info.inter_layer_predicted ? 1 : 0));
diff --git a/test/fuzzers/vp9_replay_fuzzer.cc b/test/fuzzers/vp9_replay_fuzzer.cc
index b996326da7..42452c95d8 100644
--- a/test/fuzzers/vp9_replay_fuzzer.cc
+++ b/test/fuzzers/vp9_replay_fuzzer.cc
@@ -12,10 +12,10 @@
 #include <stdint.h>
 
 #include <memory>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/video_codecs/sdp_video_format.h"
 #include "call/video_receive_stream.h"
 #include "test/fuzzers/fuzz_data_helper.h"
@@ -39,7 +39,7 @@ void FuzzOneInput(FuzzDataHelper fuzz_data) {
   std::vector<VideoReceiveStreamInterface::Config> replay_configs;
   replay_configs.push_back(std::move(vp9_config));
 
-  ArrayView<const uint8_t> raw = fuzz_data.ReadRemaining();
+  std::span<const uint8_t> raw = fuzz_data.ReadRemaining();
   test::RtpReplayer::Replay(std::move(stream_state), std::move(replay_configs),
                             raw.data(), raw.size());
 }
diff --git a/test/fuzzers/webrtc_fuzzer_main.cc b/test/fuzzers/webrtc_fuzzer_main.cc
index bd277c748a..bceed160d4 100644
--- a/test/fuzzers/webrtc_fuzzer_main.cc
+++ b/test/fuzzers/webrtc_fuzzer_main.cc
@@ -13,8 +13,8 @@
 // fuzzing efficiency.
 
 #include <cstddef>
+#include <span>
 
-#include "api/array_view.h"
 #include "rtc_base/logging.h"
 #include "test/fuzzers/fuzz_data_helper.h"
 
@@ -40,7 +40,7 @@ extern void FuzzOneInput(FuzzDataHelper fuzz_data);
 
 extern "C" int LLVMFuzzerTestOneInput(const unsigned char* data, size_t size) {
   InitializeWebRtcFuzzDefaults();
-  webrtc::FuzzDataHelper fuzz_data(webrtc::MakeArrayView(data, size));
+  webrtc::FuzzDataHelper fuzz_data(std::span(data, size));
   webrtc::FuzzOneInput(fuzz_data);
   return 0;
 }
diff --git a/test/layer_filtering_transport.cc b/test/layer_filtering_transport.cc
index 9a2996a335..ab5fd2c881 100644
--- a/test/layer_filtering_transport.cc
+++ b/test/layer_filtering_transport.cc
@@ -14,9 +14,9 @@
 #include <cstdint>
 #include <map>
 #include <memory>
+#include <span>
 #include <utility>
 
-#include "api/array_view.h"
 #include "api/call/transport.h"
 #include "api/environment/environment.h"
 #include "api/media_types.h"
@@ -50,8 +50,8 @@ LayerFilteringTransport::LayerFilteringTransport(
     const std::map<uint8_t, MediaType>& payload_type_map,
     uint32_t ssrc_to_filter_min,
     uint32_t ssrc_to_filter_max,
-    ArrayView<const RtpExtension> audio_extensions,
-    ArrayView<const RtpExtension> video_extensions)
+    std::span<const RtpExtension> audio_extensions,
+    std::span<const RtpExtension> video_extensions)
     : DirectTransport(env,
                       task_queue,
                       std::move(pipe),
@@ -79,8 +79,8 @@ LayerFilteringTransport::LayerFilteringTransport(
     int selected_tl,
     int selected_sl,
     const std::map<uint8_t, MediaType>& payload_type_map,
-    ArrayView<const RtpExtension> audio_extensions,
-    ArrayView<const RtpExtension> video_extensions)
+    std::span<const RtpExtension> audio_extensions,
+    std::span<const RtpExtension> video_extensions)
     : LayerFilteringTransport(env,
                               task_queue,
                               std::move(pipe),
@@ -99,7 +99,7 @@ bool LayerFilteringTransport::DiscardedLastPacket() const {
   return discarded_last_packet_;
 }
 
-bool LayerFilteringTransport::SendRtp(ArrayView<const uint8_t> packet,
+bool LayerFilteringTransport::SendRtp(std::span<const uint8_t> packet,
                                       const PacketOptions& options) {
   if (selected_tl_ == -1 && selected_sl_ == -1) {
     // Nothing to change, forward the packet immediately.
diff --git a/test/layer_filtering_transport.h b/test/layer_filtering_transport.h
index 9c8e7ead43..bca2756688 100644
--- a/test/layer_filtering_transport.h
+++ b/test/layer_filtering_transport.h
@@ -13,8 +13,8 @@
 #include <cstdint>
 #include <map>
 #include <memory>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/call/transport.h"
 #include "api/environment/environment.h"
 #include "api/media_types.h"
@@ -43,8 +43,8 @@ class LayerFilteringTransport : public test::DirectTransport {
       const std::map<uint8_t, MediaType>& payload_type_map,
       uint32_t ssrc_to_filter_min,
       uint32_t ssrc_to_filter_max,
-      ArrayView<const RtpExtension> audio_extensions,
-      ArrayView<const RtpExtension> video_extensions);
+      std::span<const RtpExtension> audio_extensions,
+      std::span<const RtpExtension> video_extensions);
   LayerFilteringTransport(
       const Environment& env,
       TaskQueueBase* task_queue,
@@ -55,10 +55,10 @@ class LayerFilteringTransport : public test::DirectTransport {
       int selected_tl,
       int selected_sl,
       const std::map<uint8_t, MediaType>& payload_type_map,
-      ArrayView<const RtpExtension> audio_extensions,
-      ArrayView<const RtpExtension> video_extensions);
+      std::span<const RtpExtension> audio_extensions,
+      std::span<const RtpExtension> video_extensions);
   bool DiscardedLastPacket() const;
-  bool SendRtp(ArrayView<const uint8_t> data,
+  bool SendRtp(std::span<const uint8_t> data,
                const PacketOptions& options) override;
 
  private:
diff --git a/test/mappable_native_buffer.cc b/test/mappable_native_buffer.cc
index 4a8f6ca8c0..e04c2f150a 100644
--- a/test/mappable_native_buffer.cc
+++ b/test/mappable_native_buffer.cc
@@ -11,11 +11,11 @@
 #include "test/mappable_native_buffer.h"
 
 #include <cstdint>
+#include <span>
 #include <utility>
 #include <vector>
 
 #include "absl/algorithm/container.h"
-#include "api/array_view.h"
 #include "api/make_ref_counted.h"
 #include "api/scoped_refptr.h"
 #include "api/video/i420_buffer.h"
@@ -96,7 +96,7 @@ MappableNativeBuffer::ScaledBuffer::ToI420() {
 
 scoped_refptr<VideoFrameBuffer>
 MappableNativeBuffer::ScaledBuffer::GetMappedFrameBuffer(
-    ArrayView<VideoFrameBuffer::Type> types) {
+    std::span<VideoFrameBuffer::Type> types) {
   if (absl::c_find(types, parent_->mappable_type_) == types.end())
     return nullptr;
   return parent_->GetOrCreateMappedBuffer(width_, height_);
@@ -128,7 +128,7 @@ scoped_refptr<I420BufferInterface> MappableNativeBuffer::ToI420() {
 }
 
 scoped_refptr<VideoFrameBuffer> MappableNativeBuffer::GetMappedFrameBuffer(
-    ArrayView<VideoFrameBuffer::Type> types) {
+    std::span<VideoFrameBuffer::Type> types) {
   return FullSizeBuffer()->GetMappedFrameBuffer(types);
 }
 
diff --git a/test/mappable_native_buffer.h b/test/mappable_native_buffer.h
index ddbc657ef4..c85cf74078 100644
--- a/test/mappable_native_buffer.h
+++ b/test/mappable_native_buffer.h
@@ -12,9 +12,9 @@
 #define TEST_MAPPABLE_NATIVE_BUFFER_H_
 
 #include <cstdint>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/scoped_refptr.h"
 #include "api/video/video_frame.h"
 #include "api/video/video_frame_buffer.h"
@@ -64,7 +64,7 @@ class MappableNativeBuffer : public VideoFrameBuffer {
 
   scoped_refptr<I420BufferInterface> ToI420() override;
   scoped_refptr<VideoFrameBuffer> GetMappedFrameBuffer(
-      ArrayView<VideoFrameBuffer::Type> types) override;
+      std::span<VideoFrameBuffer::Type> types) override;
 
   // Gets all the buffers that have been mapped so far, including mappings of
   // cropped and scaled buffers.
@@ -94,7 +94,7 @@ class MappableNativeBuffer : public VideoFrameBuffer {
 
     scoped_refptr<I420BufferInterface> ToI420() override;
     scoped_refptr<VideoFrameBuffer> GetMappedFrameBuffer(
-        ArrayView<VideoFrameBuffer::Type> types) override;
+        std::span<VideoFrameBuffer::Type> types) override;
 
    private:
     friend class RefCountedObject<ScaledBuffer>;
diff --git a/test/mock_audio_encoder.cc b/test/mock_audio_encoder.cc
index 817bcc4c96..5c2fe6013e 100644
--- a/test/mock_audio_encoder.cc
+++ b/test/mock_audio_encoder.cc
@@ -12,8 +12,8 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/checks.h"
@@ -33,7 +33,7 @@ MockAudioEncoder::FakeEncoding::FakeEncoding(size_t encoded_bytes) {
 
 AudioEncoder::EncodedInfo MockAudioEncoder::FakeEncoding::operator()(
     uint32_t timestamp,
-    ArrayView<const int16_t> audio,
+    std::span<const int16_t> audio,
     Buffer* encoded) {
   encoded->SetSize(encoded->size() + info_.encoded_bytes);
   return info_;
@@ -42,17 +42,17 @@ AudioEncoder::EncodedInfo MockAudioEncoder::FakeEncoding::operator()(
 MockAudioEncoder::CopyEncoding::~CopyEncoding() = default;
 
 MockAudioEncoder::CopyEncoding::CopyEncoding(AudioEncoder::EncodedInfo info,
-                                             ArrayView<const uint8_t> payload)
+                                             std::span<const uint8_t> payload)
     : info_(info), payload_(payload) {}
 
-MockAudioEncoder::CopyEncoding::CopyEncoding(ArrayView<const uint8_t> payload)
+MockAudioEncoder::CopyEncoding::CopyEncoding(std::span<const uint8_t> payload)
     : payload_(payload) {
   info_.encoded_bytes = payload_.size();
 }
 
 AudioEncoder::EncodedInfo MockAudioEncoder::CopyEncoding::operator()(
     uint32_t timestamp,
-    ArrayView<const int16_t> audio,
+    std::span<const int16_t> audio,
     Buffer* encoded) {
   RTC_CHECK(encoded);
   RTC_CHECK_LE(info_.encoded_bytes, payload_.size());
diff --git a/test/mock_audio_encoder.h b/test/mock_audio_encoder.h
index bb82570f6e..059dfafe7a 100644
--- a/test/mock_audio_encoder.h
+++ b/test/mock_audio_encoder.h
@@ -14,10 +14,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <utility>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/units/data_rate.h"
 #include "api/units/time_delta.h"
@@ -73,7 +73,7 @@ class MockAudioEncoder : public AudioEncoder {
   MOCK_METHOD(EncodedInfo,
               EncodeImpl,
               (uint32_t timestamp,
-               webrtc::ArrayView<const int16_t> audio,
+               std::span<const int16_t> audio,
                webrtc::Buffer*),
               (override));
 
@@ -88,7 +88,7 @@ class MockAudioEncoder : public AudioEncoder {
     explicit FakeEncoding(size_t encoded_bytes);
 
     AudioEncoder::EncodedInfo operator()(uint32_t timestamp,
-                                         ArrayView<const int16_t> audio,
+                                         std::span<const int16_t> audio,
                                          Buffer* encoded);
 
    private:
@@ -102,23 +102,23 @@ class MockAudioEncoder : public AudioEncoder {
     // Creates a functor that will return `info` and append the data in the
     // payload to the buffer given as input to it. Up to info.encoded_bytes are
     // appended - make sure the payload is big enough!  Since it uses an
-    // ArrayView, it _does not_ copy the payload. Make sure it doesn't fall out
+    // std::span, it _does not_ copy the payload. Make sure it doesn't fall out
     // of scope!
     CopyEncoding(AudioEncoder::EncodedInfo info,
-                 ArrayView<const uint8_t> payload);
+                 std::span<const uint8_t> payload);
 
     // Shorthand version of the constructor above, for when you wish to append
     // the whole payload and do not care about any EncodedInfo attribute other
     // than encoded_bytes.
-    explicit CopyEncoding(ArrayView<const uint8_t> payload);
+    explicit CopyEncoding(std::span<const uint8_t> payload);
 
     AudioEncoder::EncodedInfo operator()(uint32_t timestamp,
-                                         ArrayView<const int16_t> audio,
+                                         std::span<const int16_t> audio,
                                          Buffer* encoded);
 
    private:
     AudioEncoder::EncodedInfo info_;
-    ArrayView<const uint8_t> payload_;
+    std::span<const uint8_t> payload_;
   };
 };
 
diff --git a/test/mock_transport.h b/test/mock_transport.h
index 5598e269d0..ec1fc6944f 100644
--- a/test/mock_transport.h
+++ b/test/mock_transport.h
@@ -12,8 +12,8 @@
 #define TEST_MOCK_TRANSPORT_H_
 
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/call/transport.h"
 #include "test/gmock.h"
 
@@ -26,11 +26,11 @@ class MockTransport : public Transport {
 
   MOCK_METHOD(bool,
               SendRtp,
-              (webrtc::ArrayView<const uint8_t>, const PacketOptions&),
+              (std::span<const uint8_t>, const PacketOptions&),
               (override));
   MOCK_METHOD(bool,
               SendRtcp,
-              (webrtc::ArrayView<const uint8_t>, const PacketOptions&),
+              (std::span<const uint8_t>, const PacketOptions&),
               (override));
 };
 
diff --git a/test/network/BUILD.gn b/test/network/BUILD.gn
index 2e9818f673..033094d8f9 100644
--- a/test/network/BUILD.gn
+++ b/test/network/BUILD.gn
@@ -29,7 +29,6 @@ rtc_library("emulated_network") {
   deps = [
     ":simulated_network",
     "..:create_test_environment",
-    "../../api:array_view",
     "../../api:async_dns_resolver",
     "../../api:field_trials_view",
     "../../api:network_emulation_manager_api",
diff --git a/test/network/network_emulation_manager.cc b/test/network/network_emulation_manager.cc
index 2a531d87c3..02b36d6833 100644
--- a/test/network/network_emulation_manager.cc
+++ b/test/network/network_emulation_manager.cc
@@ -16,11 +16,11 @@
 #include <functional>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
 #include "absl/base/nullability.h"
-#include "api/array_view.h"
 #include "api/field_trials_view.h"
 #include "api/task_queue/task_queue_factory.h"
 #include "api/test/network_emulation/cross_traffic.h"
@@ -329,7 +329,7 @@ NetworkEmulationManagerImpl::CreateEmulatedNetworkManagerInterface(
 }
 
 void NetworkEmulationManagerImpl::GetStats(
-    ArrayView<EmulatedEndpoint* const> endpoints,
+    std::span<EmulatedEndpoint* const> endpoints,
     std::function<void(EmulatedNetworkStats)> stats_callback) {
   task_queue_.PostTask([endpoints, stats_callback, env = env_,
                         stats_gathering_mode = stats_gathering_mode_]() {
@@ -347,7 +347,7 @@ void NetworkEmulationManagerImpl::GetStats(
 }
 
 void NetworkEmulationManagerImpl::GetStats(
-    ArrayView<EmulatedNetworkNode* const> nodes,
+    std::span<EmulatedNetworkNode* const> nodes,
     std::function<void(EmulatedNetworkNodeStats)> stats_callback) {
   task_queue_.PostTask([nodes, stats_callback, env = env_,
                         stats_gathering_mode = stats_gathering_mode_]() {
diff --git a/test/network/network_emulation_manager.h b/test/network/network_emulation_manager.h
index 33482ae83c..297a14ce1f 100644
--- a/test/network/network_emulation_manager.h
+++ b/test/network/network_emulation_manager.h
@@ -18,11 +18,11 @@
 #include <memory>
 #include <optional>
 #include <set>
+#include <span>
 #include <utility>
 #include <vector>
 
 #include "absl/base/nullability.h"
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/test/network_emulation/cross_traffic.h"
 #include "api/test/network_emulation/network_emulation_interfaces.h"
@@ -86,11 +86,11 @@ class NetworkEmulationManagerImpl : public NetworkEmulationManager {
       const std::vector<EmulatedEndpoint*>& endpoints) override;
 
   void GetStats(
-      ArrayView<EmulatedEndpoint* const> endpoints,
+      std::span<EmulatedEndpoint* const> endpoints,
       std::function<void(EmulatedNetworkStats)> stats_callback) override;
 
   void GetStats(
-      ArrayView<EmulatedNetworkNode* const> nodes,
+      std::span<EmulatedNetworkNode* const> nodes,
       std::function<void(EmulatedNetworkNodeStats)> stats_callback) override;
 
   TimeController* time_controller() override { return time_controller_.get(); }
diff --git a/test/null_transport.cc b/test/null_transport.cc
index ae6e19ba9f..a6060d5c4c 100644
--- a/test/null_transport.cc
+++ b/test/null_transport.cc
@@ -10,19 +10,19 @@
 #include "test/null_transport.h"
 
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/call/transport.h"
 
 namespace webrtc {
 namespace test {
 
-bool NullTransport::SendRtp(ArrayView<const uint8_t> packet,
+bool NullTransport::SendRtp(std::span<const uint8_t> packet,
                             const PacketOptions& options) {
   return true;
 }
 
-bool NullTransport::SendRtcp(ArrayView<const uint8_t> packet,
+bool NullTransport::SendRtcp(std::span<const uint8_t> packet,
                              const PacketOptions& options) {
   return true;
 }
diff --git a/test/null_transport.h b/test/null_transport.h
index a65c6d77d4..43199ab196 100644
--- a/test/null_transport.h
+++ b/test/null_transport.h
@@ -11,8 +11,8 @@
 #define TEST_NULL_TRANSPORT_H_
 
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/call/transport.h"
 
 namespace webrtc {
@@ -22,9 +22,9 @@ class PacketReceiver;
 namespace test {
 class NullTransport : public Transport {
  public:
-  bool SendRtp(ArrayView<const uint8_t> packet,
+  bool SendRtp(std::span<const uint8_t> packet,
                const PacketOptions& options) override;
-  bool SendRtcp(ArrayView<const uint8_t> packet,
+  bool SendRtcp(std::span<const uint8_t> packet,
                 const PacketOptions& options) override;
 };
 }  // namespace test
diff --git a/test/pc/e2e/BUILD.gn b/test/pc/e2e/BUILD.gn
index a832b28b87..058a6941db 100644
--- a/test/pc/e2e/BUILD.gn
+++ b/test/pc/e2e/BUILD.gn
@@ -48,7 +48,6 @@ if (!build_with_chromium) {
         "echo/echo_emulation.h",
       ]
       deps = [
-        "../../../api:array_view",
         "../../../api:sequence_checker",
         "../../../api/test/pclf:media_configuration",
         "../../../modules/audio_device:test_audio_device_module",
@@ -199,7 +198,6 @@ if (!build_with_chromium) {
       ]
       deps = [
         "../..:fileutils",
-        "../../../api:array_view",
         "../../../api:rtp_parameters",
         "../../../api/test/pclf:media_configuration",
         "../../../api/test/pclf:media_quality_test_params",
@@ -404,7 +402,6 @@ if (!build_with_chromium) {
         ":peerconnection_quality_test",
         ":stats_based_network_quality_metrics_reporter",
         "../..:test_support",
-        "../../../api:array_view",
         "../../../api:create_network_emulation_manager",
         "../../../api:create_peer_connection_quality_test_frame_generator",
         "../../../api:network_emulation_manager_api",
@@ -570,7 +567,6 @@ if (!build_with_chromium) {
       ":metric_metadata_keys",
       "../..:perf_test",
       "../..:test_flags",
-      "../../../api:array_view",
       "../../../api:network_emulation_manager_api",
       "../../../api:peer_connection_quality_test_fixture_api",
       "../../../api:rtc_stats_api",
@@ -637,7 +633,6 @@ if (!build_with_chromium) {
       "sdp/sdp_changer.h",
     ]
     deps = [
-      "../../../api:array_view",
       "../../../api:jsep",
       "../../../api:rtp_parameters",
       "../../../api:rtp_transceiver_direction",
diff --git a/test/pc/e2e/analyzer/video/BUILD.gn b/test/pc/e2e/analyzer/video/BUILD.gn
index 9d0ab440a4..c830de2201 100644
--- a/test/pc/e2e/analyzer/video/BUILD.gn
+++ b/test/pc/e2e/analyzer/video/BUILD.gn
@@ -200,7 +200,6 @@ rtc_library("example_video_quality_analyzer") {
   ]
 
   deps = [
-    "../../../../../api:array_view",
     "../../../../../api:video_quality_analyzer_api",
     "../../../../../api/video:encoded_image",
     "../../../../../api/video:video_frame",
@@ -244,7 +243,6 @@ rtc_library("default_video_quality_analyzer_internal") {
     ":default_video_quality_analyzer_shared",
     ":multi_reader_queue",
     "../..:metric_metadata_keys",
-    "../../../../../api:array_view",
     "../../../../../api:scoped_refptr",
     "../../../../../api/numerics",
     "../../../../../api/units:data_size",
@@ -299,7 +297,6 @@ if (!build_with_chromium) {
       "../../../../../rtc_base:logging",
       "../../../../../rtc_base:macromagic",
       "../../../../../rtc_base/synchronization:mutex",
-      "../../../../../rtc_base/synchronization:mutex",
       "../../../../../system_wrappers",
       "//third_party/abseil-cpp/absl/flags:flag",
       "//third_party/abseil-cpp/absl/strings:string_view",
@@ -318,7 +315,6 @@ if (!build_with_chromium) {
       ":default_video_quality_analyzer_shared",
       "../..:metric_metadata_keys",
       "../../../..:test_flags",
-      "../../../../../api:array_view",
       "../../../../../api:rtc_stats_api",
       "../../../../../api:rtp_packet_info",
       "../../../../../api:scoped_refptr",
@@ -397,7 +393,6 @@ if (!build_with_chromium) {
       "../../../..:test_video_capturer",
       "../../../..:video_frame_writer",
       "../../../..:video_test_common",
-      "../../../../../api:array_view",
       "../../../../../api:rtc_stats_api",
       "../../../../../api:scoped_refptr",
       "../../../../../api:stats_observer_interface",
@@ -541,7 +536,6 @@ if (rtc_include_tests) {
         ":default_video_quality_analyzer",
         ":default_video_quality_analyzer_shared",
         "../../../..:test_support",
-        "../../../../../api:array_view",
         "../../../../../api:create_frame_generator",
         "../../../../../api:frame_generator_api",
         "../../../../../api:rtp_packet_info",
diff --git a/test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc b/test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc
index 5130dd6a3c..e75f5902d0 100644
--- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc
+++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc
@@ -17,13 +17,13 @@
 #include <memory>
 #include <optional>
 #include <set>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/flags/flag.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/numerics/samples_stats_counter.h"
 #include "api/rtp_packet_info.h"
 #include "api/test/metrics/metric.h"
@@ -150,7 +150,7 @@ DefaultVideoQualityAnalyzer::~DefaultVideoQualityAnalyzer() {
 }
 
 void DefaultVideoQualityAnalyzer::Start(std::string test_case_name,
-                                        ArrayView<const std::string> peer_names,
+                                        std::span<const std::string> peer_names,
                                         int max_threads_count) {
   test_label_ = std::move(test_case_name);
   frames_comparator_.Start(max_threads_count);
diff --git a/test/pc/e2e/analyzer/video/default_video_quality_analyzer.h b/test/pc/e2e/analyzer/video/default_video_quality_analyzer.h
index 6b056d07e1..58eceff583 100644
--- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer.h
+++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer.h
@@ -16,12 +16,12 @@
 #include <map>
 #include <memory>
 #include <set>
+#include <span>
 #include <string>
 #include <unordered_map>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/scoped_refptr.h"
 #include "api/stats/rtc_stats_report.h"
 #include "api/test/metrics/metrics_logger.h"
@@ -51,7 +51,7 @@ class DefaultVideoQualityAnalyzer : public VideoQualityAnalyzerInterface {
   ~DefaultVideoQualityAnalyzer() override;
 
   void Start(std::string test_case_name,
-             ArrayView<const std::string> peer_names,
+             std::span<const std::string> peer_names,
              int max_threads_count) override;
   uint16_t OnFrameCaptured(absl::string_view peer_name,
                            const std::string& stream_label,
diff --git a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator.cc b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator.cc
index 0918489f44..14fe6d8f7c 100644
--- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator.cc
+++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator.cc
@@ -14,11 +14,11 @@
 #include <cstddef>
 #include <map>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/numerics/samples_stats_counter.h"
 #include "api/scoped_refptr.h"
 #include "api/units/time_delta.h"
@@ -308,7 +308,7 @@ void DefaultVideoQualityAnalyzerFramesComparator::EnsureStatsForStream(
 }
 
 void DefaultVideoQualityAnalyzerFramesComparator::RegisterParticipantInCall(
-    ArrayView<std::pair<InternalStatsKey, Timestamp>> stream_started_time,
+    std::span<std::pair<InternalStatsKey, Timestamp>> stream_started_time,
     Timestamp start_time) {
   MutexLock lock(&mutex_);
   RTC_CHECK_EQ(state_, State::kActive)
diff --git a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator.h b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator.h
index 8b7bab6555..8b90fe2681 100644
--- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator.h
+++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator.h
@@ -16,10 +16,10 @@
 #include <deque>
 #include <map>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/numerics/samples_stats_counter.h"
 #include "api/units/timestamp.h"
 #include "api/video/video_frame.h"
@@ -95,7 +95,7 @@ class DefaultVideoQualityAnalyzerFramesComparator {
   //     has to be created.
   // `start_time` - call start time.
   void RegisterParticipantInCall(
-      ArrayView<std::pair<InternalStatsKey, Timestamp>> stream_started_time,
+      std::span<std::pair<InternalStatsKey, Timestamp>> stream_started_time,
       Timestamp start_time);
 
   // `captured` - video frame captured by sender to use for PSNR/SSIM
diff --git a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_test.cc b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_test.cc
index 5c32e1f22e..1a7c8636b5 100644
--- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_test.cc
+++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_test.cc
@@ -18,12 +18,12 @@
 #include <memory>
 #include <optional>
 #include <set>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/numerics/samples_stats_counter.h"
 #include "api/rtp_packet_info.h"
 #include "api/rtp_packet_infos.h"
@@ -110,7 +110,7 @@ VideoFrame DeepCopy(const VideoFrame& frame) {
 }
 
 std::vector<StatsSample> GetSortedSamples(const SamplesStatsCounter& counter) {
-  ArrayView<const StatsSample> view = counter.GetTimedSamples();
+  std::span<const StatsSample> view = counter.GetTimedSamples();
   std::vector<StatsSample> out(view.begin(), view.end());
   std::sort(out.begin(), out.end(),
             [](const StatsSample& a, const StatsSample& b) {
@@ -120,7 +120,7 @@ std::vector<StatsSample> GetSortedSamples(const SamplesStatsCounter& counter) {
 }
 
 std::vector<double> GetTimeSortedValues(const SamplesStatsCounter& counter) {
-  ArrayView<const StatsSample> view = counter.GetTimedSamples();
+  std::span<const StatsSample> view = counter.GetTimedSamples();
   std::vector<StatsSample> sorted(view.begin(), view.end());
   std::sort(sorted.begin(), sorted.end(),
             [](const StatsSample& a, const StatsSample& b) {
diff --git a/test/pc/e2e/analyzer/video/example_video_quality_analyzer.cc b/test/pc/e2e/analyzer/video/example_video_quality_analyzer.cc
index 8bd9453335..5f9195d074 100644
--- a/test/pc/e2e/analyzer/video/example_video_quality_analyzer.cc
+++ b/test/pc/e2e/analyzer/video/example_video_quality_analyzer.cc
@@ -11,10 +11,10 @@
 #include "test/pc/e2e/analyzer/video/example_video_quality_analyzer.h"
 
 #include <cstdint>
+#include <span>
 #include <string>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/video/encoded_image.h"
 #include "api/video/video_frame.h"
 #include "rtc_base/checks.h"
@@ -27,7 +27,7 @@ ExampleVideoQualityAnalyzer::ExampleVideoQualityAnalyzer() = default;
 ExampleVideoQualityAnalyzer::~ExampleVideoQualityAnalyzer() = default;
 
 void ExampleVideoQualityAnalyzer::Start(std::string test_case_name,
-                                        ArrayView<const std::string> peer_names,
+                                        std::span<const std::string> peer_names,
                                         int max_threads_count) {}
 
 uint16_t ExampleVideoQualityAnalyzer::OnFrameCaptured(
diff --git a/test/pc/e2e/analyzer/video/example_video_quality_analyzer.h b/test/pc/e2e/analyzer/video/example_video_quality_analyzer.h
index 59b056a74c..fa0d97f814 100644
--- a/test/pc/e2e/analyzer/video/example_video_quality_analyzer.h
+++ b/test/pc/e2e/analyzer/video/example_video_quality_analyzer.h
@@ -14,10 +14,10 @@
 #include <cstdint>
 #include <map>
 #include <set>
+#include <span>
 #include <string>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/test/video_quality_analyzer_interface.h"
 #include "api/video/encoded_image.h"
 #include "api/video/video_frame.h"
@@ -36,7 +36,7 @@ class ExampleVideoQualityAnalyzer : public VideoQualityAnalyzerInterface {
   ~ExampleVideoQualityAnalyzer() override;
 
   void Start(std::string test_case_name,
-             ArrayView<const std::string> peer_names,
+             std::span<const std::string> peer_names,
              int max_threads_count) override;
   uint16_t OnFrameCaptured(absl::string_view peer_name,
                            const std::string& stream_label,
diff --git a/test/pc/e2e/analyzer/video/names_collection.cc b/test/pc/e2e/analyzer/video/names_collection.cc
index 7a5548c510..b0f2bb3ae7 100644
--- a/test/pc/e2e/analyzer/video/names_collection.cc
+++ b/test/pc/e2e/analyzer/video/names_collection.cc
@@ -13,15 +13,15 @@
 #include <cstddef>
 #include <optional>
 #include <set>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 
 namespace webrtc {
 
-NamesCollection::NamesCollection(ArrayView<const std::string> names) {
+NamesCollection::NamesCollection(std::span<const std::string> names) {
   names_ = std::vector<std::string>(names.begin(), names.end());
   for (size_t i = 0; i < names_.size(); ++i) {
     index_.emplace(names_[i], i);
diff --git a/test/pc/e2e/analyzer/video/names_collection.h b/test/pc/e2e/analyzer/video/names_collection.h
index 5ba0729f1c..d5345a1f82 100644
--- a/test/pc/e2e/analyzer/video/names_collection.h
+++ b/test/pc/e2e/analyzer/video/names_collection.h
@@ -15,11 +15,11 @@
 #include <map>
 #include <optional>
 #include <set>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 
 namespace webrtc {
 
@@ -40,7 +40,7 @@ class NamesCollection {
  public:
   NamesCollection() = default;
 
-  explicit NamesCollection(ArrayView<const std::string> names);
+  explicit NamesCollection(std::span<const std::string> names);
 
   // Returns amount of currently presented names in the collection.
   size_t size() const { return size_; }
diff --git a/test/pc/e2e/analyzer/video/video_quality_analyzer_injection_helper.cc b/test/pc/e2e/analyzer/video/video_quality_analyzer_injection_helper.cc
index b311c2acfa..dbb57d01a9 100644
--- a/test/pc/e2e/analyzer/video/video_quality_analyzer_injection_helper.cc
+++ b/test/pc/e2e/analyzer/video/video_quality_analyzer_injection_helper.cc
@@ -12,13 +12,13 @@
 
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/memory/memory.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/scoped_refptr.h"
 #include "api/stats/rtc_stats_report.h"
 #include "api/test/pclf/media_configuration.h"
@@ -152,7 +152,7 @@ VideoQualityAnalyzerInjectionHelper::CreateVideoSink(
 
 void VideoQualityAnalyzerInjectionHelper::Start(
     std::string test_case_name,
-    ArrayView<const std::string> peer_names,
+    std::span<const std::string> peer_names,
     int max_threads_count) {
   analyzer_->Start(std::move(test_case_name), peer_names, max_threads_count);
   extractor_->Start(peer_names.size());
diff --git a/test/pc/e2e/analyzer/video/video_quality_analyzer_injection_helper.h b/test/pc/e2e/analyzer/video/video_quality_analyzer_injection_helper.h
index 99b2580e58..bc21921d19 100644
--- a/test/pc/e2e/analyzer/video/video_quality_analyzer_injection_helper.h
+++ b/test/pc/e2e/analyzer/video/video_quality_analyzer_injection_helper.h
@@ -12,11 +12,11 @@
 #define TEST_PC_E2E_ANALYZER_VIDEO_VIDEO_QUALITY_ANALYZER_INJECTION_HELPER_H_
 
 #include <memory>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/scoped_refptr.h"
 #include "api/stats/rtc_stats_report.h"
 #include "api/test/pclf/media_configuration.h"
@@ -77,7 +77,7 @@ class VideoQualityAnalyzerInjectionHelper : public StatsObserverInterface {
       bool report_infra_metrics);
 
   void Start(std::string test_case_name,
-             ArrayView<const std::string> peer_names,
+             std::span<const std::string> peer_names,
              int max_threads_count = 1);
 
   // Registers new call participant to the underlying video quality analyzer.
diff --git a/test/pc/e2e/echo/echo_emulation.cc b/test/pc/e2e/echo/echo_emulation.cc
index 14c5466a73..58e0cb93eb 100644
--- a/test/pc/e2e/echo/echo_emulation.cc
+++ b/test/pc/e2e/echo/echo_emulation.cc
@@ -13,9 +13,9 @@
 #include <cstdint>
 #include <limits>
 #include <memory>
+#include <span>
 #include <utility>
 
-#include "api/array_view.h"
 #include "api/sequence_checker.h"
 #include "api/test/pclf/media_configuration.h"
 #include "modules/audio_device/include/test_audio_device.h"
@@ -47,7 +47,7 @@ EchoEmulatingCapturer::EchoEmulatingCapturer(
   capturer_thread_.Detach();
 }
 
-void EchoEmulatingCapturer::OnAudioRendered(ArrayView<const int16_t> data) {
+void EchoEmulatingCapturer::OnAudioRendered(std::span<const int16_t> data) {
   RTC_DCHECK_RUN_ON(&renderer_thread_);
   if (!recording_started_) {
     // Because rendering can start before capturing in the beginning we can have
@@ -114,7 +114,7 @@ EchoEmulatingRenderer::EchoEmulatingRenderer(
   RTC_DCHECK(echo_emulating_capturer_);
 }
 
-bool EchoEmulatingRenderer::Render(ArrayView<const int16_t> data) {
+bool EchoEmulatingRenderer::Render(std::span<const int16_t> data) {
   if (!data.empty()) {
     echo_emulating_capturer_->OnAudioRendered(data);
   }
diff --git a/test/pc/e2e/echo/echo_emulation.h b/test/pc/e2e/echo/echo_emulation.h
index 9e0aaba05e..716aed17cb 100644
--- a/test/pc/e2e/echo/echo_emulation.h
+++ b/test/pc/e2e/echo/echo_emulation.h
@@ -13,9 +13,9 @@
 
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/sequence_checker.h"
 #include "api/test/pclf/media_configuration.h"
 #include "modules/audio_device/include/test_audio_device.h"
@@ -34,7 +34,7 @@ class EchoEmulatingCapturer : public TestAudioDeviceModule::Capturer {
       std::unique_ptr<TestAudioDeviceModule::Capturer> capturer,
       EchoEmulationConfig config);
 
-  void OnAudioRendered(ArrayView<const int16_t> data);
+  void OnAudioRendered(std::span<const int16_t> data);
 
   int SamplingFrequency() const override {
     return delegate_->SamplingFrequency();
@@ -69,7 +69,7 @@ class EchoEmulatingRenderer : public TestAudioDeviceModule::Renderer {
     return delegate_->SamplingFrequency();
   }
   int NumChannels() const override { return delegate_->NumChannels(); }
-  bool Render(ArrayView<const int16_t> data) override;
+  bool Render(std::span<const int16_t> data) override;
 
  private:
   std::unique_ptr<TestAudioDeviceModule::Renderer> delegate_;
diff --git a/test/pc/e2e/peer_params_preprocessor.cc b/test/pc/e2e/peer_params_preprocessor.cc
index 5cfbab3f94..259f9c3493 100644
--- a/test/pc/e2e/peer_params_preprocessor.cc
+++ b/test/pc/e2e/peer_params_preprocessor.cc
@@ -14,12 +14,12 @@
 #include <memory>
 #include <optional>
 #include <set>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtp_parameters.h"
 #include "api/test/pclf/media_configuration.h"
 #include "api/test/pclf/media_quality_test_params.h"
@@ -49,7 +49,7 @@ class PeerParamsPreprocessor::DefaultNamesProvider {
   // instance.
   explicit DefaultNamesProvider(
       absl::string_view prefix,
-      ArrayView<const absl::string_view> default_names = {})
+      std::span<const absl::string_view> default_names = {})
       : prefix_(prefix), default_names_(default_names) {}
 
   void MaybeSetName(std::optional<std::string>& name) {
@@ -77,7 +77,7 @@ class PeerParamsPreprocessor::DefaultNamesProvider {
   }
 
   const std::string prefix_;
-  const ArrayView<const absl::string_view> default_names_;
+  const std::span<const absl::string_view> default_names_;
 
   std::set<std::string> known_names_;
   size_t counter_ = 0;
diff --git a/test/pc/e2e/sdp/sdp_changer.cc b/test/pc/e2e/sdp/sdp_changer.cc
index b1fd867020..5d95f86d51 100644
--- a/test/pc/e2e/sdp/sdp_changer.cc
+++ b/test/pc/e2e/sdp/sdp_changer.cc
@@ -14,12 +14,12 @@
 #include <cstdint>
 #include <map>
 #include <memory>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/jsep.h"
 #include "api/media_types.h"
 #include "api/rtp_parameters.h"
@@ -51,7 +51,7 @@ std::string CodecRequiredParamsToString(
 }
 
 std::string SupportedCodecsToString(
-    ArrayView<const RtpCodecCapability> supported_codecs) {
+    std::span<const RtpCodecCapability> supported_codecs) {
   StringBuilder out;
   for (const auto& codec : supported_codecs) {
     out << codec.name;
@@ -70,11 +70,11 @@ std::string SupportedCodecsToString(
 }  // namespace
 
 std::vector<RtpCodecCapability> FilterVideoCodecCapabilities(
-    ArrayView<const VideoCodecConfig> video_codecs,
+    std::span<const VideoCodecConfig> video_codecs,
     bool use_rtx,
     bool use_ulpfec,
     bool use_flexfec,
-    ArrayView<const RtpCodecCapability> supported_codecs) {
+    std::span<const RtpCodecCapability> supported_codecs) {
   std::vector<RtpCodecCapability> output_codecs;
   // Find requested codecs among supported and add them to output in the order
   // they were requested.
@@ -527,7 +527,7 @@ LocalAndRemoteSdp SignalingInterceptor::PatchVp9Answer(
 
 std::vector<std::unique_ptr<IceCandidate>>
 SignalingInterceptor::PatchOffererIceCandidates(
-    ArrayView<const IceCandidate* const> candidates) {
+    std::span<const IceCandidate* const> candidates) {
   std::vector<std::unique_ptr<IceCandidate>> out;
   for (auto* candidate : candidates) {
     auto simulcast_info_it =
@@ -551,7 +551,7 @@ SignalingInterceptor::PatchOffererIceCandidates(
 
 std::vector<std::unique_ptr<IceCandidate>>
 SignalingInterceptor::PatchAnswererIceCandidates(
-    ArrayView<const IceCandidate* const> candidates) {
+    std::span<const IceCandidate* const> candidates) {
   std::vector<std::unique_ptr<IceCandidate>> out;
   for (auto* candidate : candidates) {
     auto simulcast_info_it =
diff --git a/test/pc/e2e/sdp/sdp_changer.h b/test/pc/e2e/sdp/sdp_changer.h
index 63ac48627b..84582b6dfc 100644
--- a/test/pc/e2e/sdp/sdp_changer.h
+++ b/test/pc/e2e/sdp/sdp_changer.h
@@ -13,12 +13,12 @@
 
 #include <map>
 #include <memory>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/jsep.h"
 #include "api/rtp_parameters.h"
 #include "api/test/pclf/media_configuration.h"
@@ -42,11 +42,11 @@ namespace webrtc_pc_e2e {
 // vector and they will be added in the same order, as they were in
 // `supported_codecs`.
 std::vector<RtpCodecCapability> FilterVideoCodecCapabilities(
-    ArrayView<const VideoCodecConfig> video_codecs,
+    std::span<const VideoCodecConfig> video_codecs,
     bool use_rtx,
     bool use_ulpfec,
     bool use_flexfec,
-    ArrayView<const RtpCodecCapability> supported_codecs);
+    std::span<const RtpCodecCapability> supported_codecs);
 
 struct LocalAndRemoteSdp {
   LocalAndRemoteSdp(std::unique_ptr<SessionDescriptionInterface> local_sdp,
@@ -84,9 +84,9 @@ class SignalingInterceptor {
       const VideoCodecConfig& first_codec);
 
   std::vector<std::unique_ptr<IceCandidate>> PatchOffererIceCandidates(
-      ArrayView<const IceCandidate* const> candidates);
+      std::span<const IceCandidate* const> candidates);
   std::vector<std::unique_ptr<IceCandidate>> PatchAnswererIceCandidates(
-      ArrayView<const IceCandidate* const> candidates);
+      std::span<const IceCandidate* const> candidates);
 
  private:
   // Contains information about simulcast section, that is required to perform
diff --git a/test/pc/e2e/stats_based_network_quality_metrics_reporter.cc b/test/pc/e2e/stats_based_network_quality_metrics_reporter.cc
index d798a8ebf9..9431b05e7e 100644
--- a/test/pc/e2e/stats_based_network_quality_metrics_reporter.cc
+++ b/test/pc/e2e/stats_based_network_quality_metrics_reporter.cc
@@ -20,7 +20,6 @@
 
 #include "absl/flags/flag.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/scoped_refptr.h"
 #include "api/sequence_checker.h"
 #include "api/stats/rtc_stats_report.h"
diff --git a/test/pc/e2e/stats_based_network_quality_metrics_reporter_test.cc b/test/pc/e2e/stats_based_network_quality_metrics_reporter_test.cc
index d4044f0ec0..bdd5528101 100644
--- a/test/pc/e2e/stats_based_network_quality_metrics_reporter_test.cc
+++ b/test/pc/e2e/stats_based_network_quality_metrics_reporter_test.cc
@@ -13,12 +13,12 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/test/create_network_emulation_manager.h"
 #include "api/test/metrics/metric.h"
 #include "api/test/metrics/metrics_logger.h"
@@ -68,7 +68,7 @@ void AddDefaultAudioVideoPeer(absl::string_view peer_name,
 }
 
 std::optional<Metric> FindMeetricByName(absl::string_view name,
-                                        ArrayView<const Metric> metrics) {
+                                        std::span<const Metric> metrics) {
   for (const Metric& metric : metrics) {
     if (metric.name == name) {
       return metric;
diff --git a/test/peer_scenario/BUILD.gn b/test/peer_scenario/BUILD.gn
index 01401b024d..a0b3c2adb8 100644
--- a/test/peer_scenario/BUILD.gn
+++ b/test/peer_scenario/BUILD.gn
@@ -29,7 +29,6 @@ if (rtc_include_tests) {
       "..:fileutils",
       "..:frame_generator_capturer",
       "..:test_support",
-      "../../api:array_view",
       "../../api:audio_options_api",
       "../../api:candidate",
       "../../api:create_modular_peer_connection_factory",
diff --git a/test/peer_scenario/bwe_integration_tests/BUILD.gn b/test/peer_scenario/bwe_integration_tests/BUILD.gn
index 3c65d961bd..729c2b0e5a 100644
--- a/test/peer_scenario/bwe_integration_tests/BUILD.gn
+++ b/test/peer_scenario/bwe_integration_tests/BUILD.gn
@@ -34,7 +34,6 @@ if (rtc_include_tests) {
         "..:peer_scenario",
         "../..:create_frame_generator_capturer",
         "../..:test_support",
-        "../../../api:array_view",
         "../../../api:audio_options_api",
         "../../../api:jsep",
         "../../../api:make_ref_counted",
diff --git a/test/peer_scenario/bwe_integration_tests/l4s_test.cc b/test/peer_scenario/bwe_integration_tests/l4s_test.cc
index b3c949b58f..98fdc9636d 100644
--- a/test/peer_scenario/bwe_integration_tests/l4s_test.cc
+++ b/test/peer_scenario/bwe_integration_tests/l4s_test.cc
@@ -15,7 +15,6 @@
 
 #include "absl/strings/str_cat.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio_options.h"
 #include "api/jsep.h"
 #include "api/rtc_error.h"
diff --git a/test/peer_scenario/bwe_integration_tests/scream_test.cc b/test/peer_scenario/bwe_integration_tests/scream_test.cc
index dac4e2ff6f..33b5901f24 100644
--- a/test/peer_scenario/bwe_integration_tests/scream_test.cc
+++ b/test/peer_scenario/bwe_integration_tests/scream_test.cc
@@ -11,11 +11,11 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio_options.h"
 #include "api/rtp_parameters.h"
 #include "api/scoped_refptr.h"
@@ -188,12 +188,12 @@ struct SendMediaTestResult {
   std::vector<scoped_refptr<const RTCStatsReport>> caller_stats;
   std::vector<scoped_refptr<const RTCStatsReport>> callee_stats;
 
-  ArrayView<const scoped_refptr<const RTCStatsReport>> caller() const {
-    return ArrayView<const scoped_refptr<const RTCStatsReport>>(caller_stats);
+  std::span<const scoped_refptr<const RTCStatsReport>> caller() const {
+    return std::span<const scoped_refptr<const RTCStatsReport>>(caller_stats);
   }
 
-  ArrayView<const scoped_refptr<const RTCStatsReport>> callee() const {
-    return ArrayView<const scoped_refptr<const RTCStatsReport>>(callee_stats);
+  std::span<const scoped_refptr<const RTCStatsReport>> callee() const {
+    return std::span<const scoped_refptr<const RTCStatsReport>>(callee_stats);
   }
 };
 
diff --git a/test/peer_scenario/bwe_integration_tests/stats_utilities.cc b/test/peer_scenario/bwe_integration_tests/stats_utilities.cc
index 49a6ca2fae..422414c5d1 100644
--- a/test/peer_scenario/bwe_integration_tests/stats_utilities.cc
+++ b/test/peer_scenario/bwe_integration_tests/stats_utilities.cc
@@ -11,9 +11,9 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string_view>
 
-#include "api/array_view.h"
 #include "api/make_ref_counted.h"
 #include "api/scoped_refptr.h"
 #include "api/stats/rtc_stats_report.h"
@@ -41,7 +41,7 @@ scoped_refptr<const RTCStatsReport> GetStatsAndProcess(
 
 std::optional<scoped_refptr<const RTCStatsReport>> GetFirstReportAtOrAfter(
     Timestamp time,
-    ArrayView<const scoped_refptr<const RTCStatsReport>> reports) {
+    std::span<const scoped_refptr<const RTCStatsReport>> reports) {
   if (reports.empty()) {
     return std::nullopt;
   }
diff --git a/test/peer_scenario/bwe_integration_tests/stats_utilities.h b/test/peer_scenario/bwe_integration_tests/stats_utilities.h
index ed34399114..716e39c2e0 100644
--- a/test/peer_scenario/bwe_integration_tests/stats_utilities.h
+++ b/test/peer_scenario/bwe_integration_tests/stats_utilities.h
@@ -12,9 +12,9 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string_view>
 
-#include "api/array_view.h"
 #include "api/scoped_refptr.h"
 #include "api/stats/rtc_stats_report.h"
 #include "api/units/data_rate.h"
@@ -32,7 +32,7 @@ scoped_refptr<const RTCStatsReport> GetStatsAndProcess(
 
 std::optional<scoped_refptr<const RTCStatsReport>> GetFirstReportAtOrAfter(
     Timestamp time,
-    ArrayView<const scoped_refptr<const RTCStatsReport>> reports);
+    std::span<const scoped_refptr<const RTCStatsReport>> reports);
 
 DataRate GetAvailableSendBitrate(
     const scoped_refptr<const RTCStatsReport>& report);
diff --git a/test/peer_scenario/scenario_connection.cc b/test/peer_scenario/scenario_connection.cc
index 1bb53c2732..f7e66e6197 100644
--- a/test/peer_scenario/scenario_connection.cc
+++ b/test/peer_scenario/scenario_connection.cc
@@ -11,12 +11,12 @@
 
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/candidate.h"
 #include "api/environment/environment.h"
 #include "api/jsep.h"
@@ -64,8 +64,8 @@ class ScenarioIceConnectionImpl : public ScenarioIceConnection,
                             IceConnectionObserver* observer);
   ~ScenarioIceConnectionImpl() override;
 
-  void SendRtpPacket(ArrayView<const uint8_t> packet_view) override;
-  void SendRtcpPacket(ArrayView<const uint8_t> packet_view) override;
+  void SendRtpPacket(std::span<const uint8_t> packet_view) override;
+  void SendRtcpPacket(std::span<const uint8_t> packet_view) override;
 
   void SetRemoteSdp(SdpType type, const std::string& remote_sdp) override;
   void SetLocalSdp(SdpType type, const std::string& local_sdp) override;
@@ -187,7 +187,7 @@ JsepTransportController::Config ScenarioIceConnectionImpl::CreateJsepConfig() {
 }
 
 void ScenarioIceConnectionImpl::SendRtpPacket(
-    ArrayView<const uint8_t> packet_view) {
+    std::span<const uint8_t> packet_view) {
   CopyOnWriteBuffer packet(packet_view.data(), packet_view.size(),
                            kMaxRtpPacketLen);
   network_thread_->PostTask([this, packet = std::move(packet)]() mutable {
@@ -199,7 +199,7 @@ void ScenarioIceConnectionImpl::SendRtpPacket(
 }
 
 void ScenarioIceConnectionImpl::SendRtcpPacket(
-    ArrayView<const uint8_t> packet_view) {
+    std::span<const uint8_t> packet_view) {
   CopyOnWriteBuffer packet(packet_view.data(), packet_view.size(),
                            kMaxRtpPacketLen);
   network_thread_->PostTask([this, packet = std::move(packet)]() mutable {
diff --git a/test/peer_scenario/scenario_connection.h b/test/peer_scenario/scenario_connection.h
index d4d7ec8169..7e953a5e2e 100644
--- a/test/peer_scenario/scenario_connection.h
+++ b/test/peer_scenario/scenario_connection.h
@@ -12,11 +12,11 @@
 
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/candidate.h"
 #include "api/environment/environment.h"
 #include "api/jsep.h"
@@ -51,8 +51,8 @@ class ScenarioIceConnection {
   virtual ~ScenarioIceConnection() = default;
 
   // Posts tasks to send packets to network thread.
-  virtual void SendRtpPacket(ArrayView<const uint8_t> packet_view) = 0;
-  virtual void SendRtcpPacket(ArrayView<const uint8_t> packet_view) = 0;
+  virtual void SendRtpPacket(std::span<const uint8_t> packet_view) = 0;
+  virtual void SendRtcpPacket(std::span<const uint8_t> packet_view) = 0;
 
   // Used for ICE configuration, called on signaling thread.
   virtual void SetRemoteSdp(SdpType type, const std::string& remote_sdp) = 0;
diff --git a/test/rtcp_packet_parser.cc b/test/rtcp_packet_parser.cc
index eb74ad96b9..6296ffa0c3 100644
--- a/test/rtcp_packet_parser.cc
+++ b/test/rtcp_packet_parser.cc
@@ -11,8 +11,8 @@
 #include "test/rtcp_packet_parser.h"
 
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/rtp_rtcp/source/rtcp_packet/app.h"
 #include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
 #include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
@@ -38,7 +38,7 @@ namespace test {
 RtcpPacketParser::RtcpPacketParser() = default;
 RtcpPacketParser::~RtcpPacketParser() = default;
 
-bool RtcpPacketParser::Parse(ArrayView<const uint8_t> data) {
+bool RtcpPacketParser::Parse(std::span<const uint8_t> data) {
   ++processed_rtcp_packets_;
 
   const uint8_t* const buffer = data.data();
diff --git a/test/rtcp_packet_parser.h b/test/rtcp_packet_parser.h
index 84ef5366ec..cdb2cde124 100644
--- a/test/rtcp_packet_parser.h
+++ b/test/rtcp_packet_parser.h
@@ -15,7 +15,8 @@
 #include <stddef.h>
 #include <stdint.h>
 
-#include "api/array_view.h"
+#include <span>
+
 #include "modules/rtp_rtcp/source/rtcp_packet/app.h"
 #include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
 #include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
@@ -47,7 +48,7 @@ bool ParseSinglePacket(const uint8_t* buffer, size_t size, Packet* packet) {
 }
 // Same function, but takes raw buffer as single argument instead of pair.
 template <typename Packet>
-bool ParseSinglePacket(ArrayView<const uint8_t> buffer, Packet* packet) {
+bool ParseSinglePacket(std::span<const uint8_t> buffer, Packet* packet) {
   return ParseSinglePacket(buffer.data(), buffer.size(), packet);
 }
 
@@ -79,7 +80,7 @@ class RtcpPacketParser {
   RtcpPacketParser();
   ~RtcpPacketParser();
 
-  bool Parse(ArrayView<const uint8_t> packet);
+  bool Parse(std::span<const uint8_t> packet);
 
   PacketCounter<rtcp::App>* app() { return &app_; }
   PacketCounter<rtcp::Bye>* bye() { return &bye_; }
diff --git a/test/rtp_file_reader.cc b/test/rtp_file_reader.cc
index 296e3dc0b4..6d4f58e1e8 100644
--- a/test/rtp_file_reader.cc
+++ b/test/rtp_file_reader.cc
@@ -17,11 +17,11 @@
 #include <map>
 #include <memory>
 #include <set>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "modules/rtp_rtcp/source/rtp_util.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
@@ -470,7 +470,7 @@ class PcapReader : public RtpFileReaderImpl {
 
   int ProcessPacket(RtpPacketMarker& marker,
                     const std::set<uint32_t>& ssrc_filter,
-                    ArrayView<const uint8_t> packet) {
+                    std::span<const uint8_t> packet) {
     if (IsRtcpPacket(packet)) {
       marker.payload_type = packet[1];
       packets_.push_back(marker);
diff --git a/test/rtp_file_reader_unittest.cc b/test/rtp_file_reader_unittest.cc
index 7236f6c9d3..998cd380db 100644
--- a/test/rtp_file_reader_unittest.cc
+++ b/test/rtp_file_reader_unittest.cc
@@ -13,9 +13,9 @@
 #include <cstdint>
 #include <map>
 #include <memory>
+#include <span>
 #include <string>
 
-#include "api/array_view.h"
 #include "modules/rtp_rtcp/source/rtp_util.h"
 #include "test/gtest.h"
 #include "test/testsupport/file_utils.h"
@@ -87,7 +87,7 @@ class TestPcapFileReader : public ::testing::Test {
     PacketsPerSsrc pps;
     test::RtpPacket packet;
     while (rtp_packet_source_->NextPacket(&packet)) {
-      ArrayView<const uint8_t> raw(packet.data, packet.length);
+      std::span<const uint8_t> raw(packet.data, packet.length);
       if (IsRtpPacket(raw)) {
         pps[ParseRtpSsrc(raw)]++;
       }
diff --git a/test/rtp_rtcp_observer.h b/test/rtp_rtcp_observer.h
index 3fd9327a72..94de6489c2 100644
--- a/test/rtp_rtcp_observer.h
+++ b/test/rtp_rtcp_observer.h
@@ -13,10 +13,10 @@
 #include <cstdint>
 #include <map>
 #include <memory>
+#include <span>
 #include <utility>
 
 #include "absl/flags/flag.h"
-#include "api/array_view.h"
 #include "api/call/transport.h"
 #include "api/environment/environment.h"
 #include "api/media_types.h"
@@ -54,19 +54,19 @@ class RtpRtcpObserver {
     return observation_complete_.Wait(timeout_);
   }
 
-  virtual Action OnSendRtp(ArrayView<const uint8_t> packet) {
+  virtual Action OnSendRtp(std::span<const uint8_t> packet) {
     return SEND_PACKET;
   }
 
-  virtual Action OnSendRtcp(ArrayView<const uint8_t> packet) {
+  virtual Action OnSendRtcp(std::span<const uint8_t> packet) {
     return SEND_PACKET;
   }
 
-  virtual Action OnReceiveRtp(ArrayView<const uint8_t> packet) {
+  virtual Action OnReceiveRtp(std::span<const uint8_t> packet) {
     return SEND_PACKET;
   }
 
-  virtual Action OnReceiveRtcp(ArrayView<const uint8_t> packet) {
+  virtual Action OnReceiveRtcp(std::span<const uint8_t> packet) {
     return SEND_PACKET;
   }
 
@@ -91,8 +91,8 @@ class PacketTransport : public test::DirectTransport {
                   TransportType transport_type,
                   const std::map<uint8_t, MediaType>& payload_type_map,
                   std::unique_ptr<SimulatedPacketReceiverInterface> nw_pipe,
-                  ArrayView<const RtpExtension> audio_extensions,
-                  ArrayView<const RtpExtension> video_extensions)
+                  std::span<const RtpExtension> audio_extensions,
+                  std::span<const RtpExtension> video_extensions)
       : test::DirectTransport(env,
                               task_queue,
                               std::move(nw_pipe),
@@ -104,7 +104,7 @@ class PacketTransport : public test::DirectTransport {
         transport_type_(transport_type) {}
 
  private:
-  bool SendRtp(ArrayView<const uint8_t> packet,
+  bool SendRtp(std::span<const uint8_t> packet,
                const PacketOptions& options) override {
     EXPECT_TRUE(IsRtpPacket(packet));
     RtpRtcpObserver::Action action = RtpRtcpObserver::SEND_PACKET;
@@ -126,7 +126,7 @@ class PacketTransport : public test::DirectTransport {
     return true;
   }
 
-  bool SendRtcp(ArrayView<const uint8_t> packet,
+  bool SendRtcp(std::span<const uint8_t> packet,
                 const PacketOptions& options) override {
     EXPECT_TRUE(IsRtcpPacket(packet));
     RtpRtcpObserver::Action action = RtpRtcpObserver::SEND_PACKET;
diff --git a/test/scenario/BUILD.gn b/test/scenario/BUILD.gn
index ddfba64d8d..ec99e60ba7 100644
--- a/test/scenario/BUILD.gn
+++ b/test/scenario/BUILD.gn
@@ -83,7 +83,6 @@ if (rtc_include_tests && !build_with_chromium) {
       "../:test_support",
       "../:video_test_common",
       "../..//test/network:simulated_network",
-      "../../api:array_view",
       "../../api:create_frame_generator",
       "../../api:fec_controller_api",
       "../../api:field_trials",
diff --git a/test/scenario/call_client.cc b/test/scenario/call_client.cc
index 6c25ea6a78..d6c538c3a6 100644
--- a/test/scenario/call_client.cc
+++ b/test/scenario/call_client.cc
@@ -13,10 +13,10 @@
 #include <cstdint>
 #include <functional>
 #include <memory>
+#include <span>
 #include <string>
 #include <utility>
 
-#include "api/array_view.h"
 #include "api/audio/builtin_audio_processing_builder.h"
 #include "api/environment/environment.h"
 #include "api/environment/environment_factory.h"
@@ -309,14 +309,14 @@ void CallClient::UpdateBitrateConstraints(
 }
 
 void CallClient::SetAudioReceiveRtpHeaderExtensions(
-    ArrayView<RtpExtension> extensions) {
+    std::span<RtpExtension> extensions) {
   SendTask([this, &extensions]() {
     audio_extensions_ = RtpHeaderExtensionMap(extensions);
   });
 }
 
 void CallClient::SetVideoReceiveRtpHeaderExtensions(
-    ArrayView<RtpExtension> extensions) {
+    std::span<RtpExtension> extensions) {
   SendTask([this, &extensions]() {
     video_extensions_ = RtpHeaderExtensionMap(extensions);
   });
diff --git a/test/scenario/call_client.h b/test/scenario/call_client.h
index a3f0295103..53a7700e64 100644
--- a/test/scenario/call_client.h
+++ b/test/scenario/call_client.h
@@ -14,11 +14,11 @@
 #include <functional>
 #include <map>
 #include <memory>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/audio_device.h"
 #include "api/audio/audio_processing.h"
 #include "api/environment/environment.h"
@@ -135,8 +135,8 @@ class CallClient : public EmulatedNetworkReceiverInterface {
   void UpdateBitrateConstraints(const BitrateConstraints& constraints);
   void SetRemoteBitrate(DataRate bitrate);
 
-  void SetAudioReceiveRtpHeaderExtensions(ArrayView<RtpExtension> extensions);
-  void SetVideoReceiveRtpHeaderExtensions(ArrayView<RtpExtension> extensions);
+  void SetAudioReceiveRtpHeaderExtensions(std::span<RtpExtension> extensions);
+  void SetVideoReceiveRtpHeaderExtensions(std::span<RtpExtension> extensions);
 
   // Sets the network adapter id used next time the network route changes.
   void UpdateNetworkAdapterId(int adapter_id);
diff --git a/test/scenario/network_node.cc b/test/scenario/network_node.cc
index 364f2e6da9..8a4b802718 100644
--- a/test/scenario/network_node.cc
+++ b/test/scenario/network_node.cc
@@ -12,10 +12,10 @@
 #include <cstdint>
 #include <functional>
 #include <memory>
+#include <span>
 #include <utility>
 
 #include "absl/cleanup/cleanup.h"
-#include "api/array_view.h"
 #include "api/call/transport.h"
 #include "api/sequence_checker.h"
 #include "api/test/network_emulation/network_emulation_interfaces.h"
@@ -101,7 +101,7 @@ NetworkNodeTransport::NetworkNodeTransport(Clock* sender_clock,
 
 NetworkNodeTransport::~NetworkNodeTransport() = default;
 
-bool NetworkNodeTransport::SendRtp(ArrayView<const uint8_t> packet,
+bool NetworkNodeTransport::SendRtp(std::span<const uint8_t> packet,
                                    const PacketOptions& options) {
   int64_t send_time_ms = sender_clock_->TimeInMilliseconds();
   SentPacketInfo sent_packet;
@@ -122,7 +122,7 @@ bool NetworkNodeTransport::SendRtp(ArrayView<const uint8_t> packet,
   return true;
 }
 
-bool NetworkNodeTransport::SendRtcp(ArrayView<const uint8_t> packet,
+bool NetworkNodeTransport::SendRtcp(std::span<const uint8_t> packet,
                                     const PacketOptions& options) {
   CopyOnWriteBuffer buffer(packet);
   MutexLock lock(&mutex_);
diff --git a/test/scenario/network_node.h b/test/scenario/network_node.h
index e0086fa6b4..3a8f706468 100644
--- a/test/scenario/network_node.h
+++ b/test/scenario/network_node.h
@@ -13,8 +13,8 @@
 #include <cstdint>
 #include <functional>
 #include <memory>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/call/transport.h"
 #include "api/sequence_checker.h"
 #include "api/test/network_emulation/network_emulation_interfaces.h"
@@ -60,9 +60,9 @@ class NetworkNodeTransport : public Transport {
 
   void UpdateAdapterId(int adapter_id);
 
-  bool SendRtp(ArrayView<const uint8_t> packet,
+  bool SendRtp(std::span<const uint8_t> packet,
                const PacketOptions& options) override;
-  bool SendRtcp(ArrayView<const uint8_t> packet,
+  bool SendRtcp(std::span<const uint8_t> packet,
                 const PacketOptions& options) override;
 
   void Connect(EmulatedEndpoint* endpoint,
diff --git a/test/testsupport/perf_test.cc b/test/testsupport/perf_test.cc
index 0d1bb07a8c..ae89393983 100644
--- a/test/testsupport/perf_test.cc
+++ b/test/testsupport/perf_test.cc
@@ -15,12 +15,12 @@
 #include <cstdio>
 #include <ostream>
 #include <set>
+#include <span>
 #include <sstream>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/numerics/samples_stats_counter.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/strings/string_builder.h"
@@ -49,7 +49,7 @@ std::string UnitWithDirection(absl::string_view units,
 
 std::vector<SamplesStatsCounter::StatsSample> GetSortedSamples(
     const SamplesStatsCounter& counter) {
-  ArrayView<const SamplesStatsCounter::StatsSample> view =
+  std::span<const SamplesStatsCounter::StatsSample> view =
       counter.GetTimedSamples();
   std::vector<SamplesStatsCounter::StatsSample> out(view.begin(), view.end());
   std::stable_sort(out.begin(), out.end(),
@@ -182,7 +182,7 @@ class ResultsLinePrinter {
 
   void PrintResultList(absl::string_view graph_name,
                        absl::string_view trace_name,
-                       const ArrayView<const double> values,
+                       const std::span<const double> values,
                        absl::string_view units,
                        const bool important,
                        test::ImproveDirection improve_direction) {
@@ -339,7 +339,7 @@ void PrintResultMeanAndError(absl::string_view measurement,
 void PrintResultList(absl::string_view measurement,
                      absl::string_view modifier,
                      absl::string_view trace,
-                     const ArrayView<const double> values,
+                     const std::span<const double> values,
                      absl::string_view units,
                      bool important,
                      ImproveDirection improve_direction) {
diff --git a/test/testsupport/perf_test.h b/test/testsupport/perf_test.h
index 9773b7c873..f87faf7b92 100644
--- a/test/testsupport/perf_test.h
+++ b/test/testsupport/perf_test.h
@@ -12,11 +12,11 @@
 #define TEST_TESTSUPPORT_PERF_TEST_H_
 
 #include <cstdio>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/numerics/samples_stats_counter.h"
 
 namespace webrtc {
@@ -76,7 +76,7 @@ void PrintResultList(
     absl::string_view measurement,
     absl::string_view modifier,
     absl::string_view user_story,
-    ArrayView<const double> values,
+    std::span<const double> values,
     absl::string_view units,
     bool important,
     ImproveDirection improve_direction = ImproveDirection::kNone);
diff --git a/test/testsupport/perf_test_histogram_writer.cc b/test/testsupport/perf_test_histogram_writer.cc
index b93a9a4b31..aaf95c7da1 100644
--- a/test/testsupport/perf_test_histogram_writer.cc
+++ b/test/testsupport/perf_test_histogram_writer.cc
@@ -13,11 +13,11 @@
 #include <cstdlib>
 #include <map>
 #include <memory>
+#include <span>
 #include <string>
 #include <utility>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/strings/string_builder.h"
@@ -71,7 +71,7 @@ class PerfTestHistogramWriter : public PerfTestResultWriter {
   }
   void LogResultList(absl::string_view graph_name,
                      absl::string_view trace_name,
-                     const ArrayView<const double> values,
+                     const std::span<const double> values,
                      absl::string_view units,
                      const bool important,
                      ImproveDirection improve_direction) override {
diff --git a/test/testsupport/perf_test_result_writer.h b/test/testsupport/perf_test_result_writer.h
index a4c52e9763..e5ad092745 100644
--- a/test/testsupport/perf_test_result_writer.h
+++ b/test/testsupport/perf_test_result_writer.h
@@ -11,10 +11,10 @@
 #ifndef TEST_TESTSUPPORT_PERF_TEST_RESULT_WRITER_H_
 #define TEST_TESTSUPPORT_PERF_TEST_RESULT_WRITER_H_
 
+#include <span>
 #include <string>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "test/testsupport/perf_test.h"
 
 namespace webrtc {
@@ -43,7 +43,7 @@ class PerfTestResultWriter {
   virtual void LogResultList(
       absl::string_view graph_name,
       absl::string_view trace_name,
-      ArrayView<const double> values,
+      std::span<const double> values,
       absl::string_view units,
       bool important,
       webrtc::test::ImproveDirection improve_direction) = 0;
diff --git a/test/testsupport/video_frame_writer.cc b/test/testsupport/video_frame_writer.cc
index e69c57c0be..221cd0a89c 100644
--- a/test/testsupport/video_frame_writer.cc
+++ b/test/testsupport/video_frame_writer.cc
@@ -15,10 +15,10 @@
 #include <cstdlib>
 #include <limits>
 #include <memory>
+#include <span>
 #include <string>
 #include <utility>
 
-#include "api/array_view.h"
 #include "api/scoped_refptr.h"
 #include "api/video/i420_buffer.h"
 #include "api/video/video_frame.h"
@@ -45,7 +45,7 @@ Buffer ExtractI420BufferWithSize(const VideoFrame& frame,
     size_t length =
         CalcBufferSize(VideoType::kI420, scaled->width(), scaled->height());
     Buffer buffer = Buffer::CreateWithCapacity(length);
-    buffer.AppendData(length, [&](ArrayView<uint8_t> buffer) {
+    buffer.AppendData(length, [&](std::span<uint8_t> buffer) {
       RTC_CHECK_NE(ExtractBuffer(scaled, length, buffer.data()), -1);
       return length;
     });
@@ -56,7 +56,7 @@ Buffer ExtractI420BufferWithSize(const VideoFrame& frame,
   size_t length =
       CalcBufferSize(VideoType::kI420, frame.width(), frame.height());
   Buffer buffer = Buffer::CreateWithCapacity(length);
-  buffer.AppendData(length, [&](ArrayView<uint8_t> buffer) {
+  buffer.AppendData(length, [&](std::span<uint8_t> buffer) {
     RTC_CHECK_NE(ExtractBuffer(frame, length, buffer.data()), -1);
     return length;
   });
diff --git a/test/video_codec_tester.cc b/test/video_codec_tester.cc
index 78b632f0a0..a12c688dc0 100644
--- a/test/video_codec_tester.cc
+++ b/test/video_codec_tester.cc
@@ -23,6 +23,7 @@
 #include <numeric>
 #include <optional>
 #include <set>
+#include <span>
 #include <string>
 #include <tuple>
 #include <utility>
@@ -31,7 +32,6 @@
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/match.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/field_trials_view.h"
 #include "api/make_ref_counted.h"
@@ -587,7 +587,7 @@ class VideoCodecAnalyzer : public VideoCodecTester::VideoCodecStats {
 
       Frame superframe = subframes.back();
       for (const Frame& frame :
-           ArrayView<Frame>(subframes).subspan(0, subframes.size() - 1)) {
+           std::span<Frame>(subframes).subspan(0, subframes.size() - 1)) {
         superframe.decoded |= frame.decoded;
         superframe.encoded |= frame.encoded;
         superframe.frame_size += frame.frame_size;
diff --git a/test/video_frame_matcher.cc b/test/video_frame_matcher.cc
index e2ade33480..8e63eb2975 100644
--- a/test/video_frame_matcher.cc
+++ b/test/video_frame_matcher.cc
@@ -12,9 +12,9 @@
 
 #include <cstdint>
 #include <ostream>
+#include <span>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/scoped_refptr.h"
 #include "api/video/video_frame.h"
 #include "api/video/video_frame_buffer.h"
@@ -27,7 +27,7 @@ namespace {
 using ::testing::Matcher;
 using ::testing::MatchResultListener;
 
-void PrintOneChannelToOs(ArrayView<const uint8_t> data,
+void PrintOneChannelToOs(std::span<const uint8_t> data,
                          int width,
                          int height,
                          std::ostream& os) {
@@ -39,7 +39,7 @@ void PrintOneChannelToOs(ArrayView<const uint8_t> data,
   }
 }
 
-void PrintOneChannelToListener(ArrayView<const uint8_t> data,
+void PrintOneChannelToListener(std::span<const uint8_t> data,
                                int width,
                                int height,
                                MatchResultListener& listener) {
@@ -105,30 +105,29 @@ class PixelValuesEqualMatcher {
     // done in order to give a more detailed error message.
     bool is_same = true;
     is_same &= DataMatricesAreEqual(
-        MakeArrayView(
+        std::span(
             expected_frame_buffer_->DataY(),
             expected_frame_buffer_->width() * expected_frame_buffer_->height()),
-        MakeArrayView(
-            actual_frame_buffer->DataY(),
-            actual_frame_buffer->width() * actual_frame_buffer->height()),
+        std::span(actual_frame_buffer->DataY(),
+                  actual_frame_buffer->width() * actual_frame_buffer->height()),
         width_, height_, SampleType::kY, *listener);
 
     is_same &= DataMatricesAreEqual(
-        MakeArrayView(expected_frame_buffer_->DataU(),
-                      expected_frame_buffer_->ChromaWidth() *
-                          expected_frame_buffer_->ChromaHeight()),
-        MakeArrayView(actual_frame_buffer->DataU(),
-                      actual_frame_buffer->ChromaWidth() *
-                          actual_frame_buffer->ChromaHeight()),
+        std::span(expected_frame_buffer_->DataU(),
+                  expected_frame_buffer_->ChromaWidth() *
+                      expected_frame_buffer_->ChromaHeight()),
+        std::span(actual_frame_buffer->DataU(),
+                  actual_frame_buffer->ChromaWidth() *
+                      actual_frame_buffer->ChromaHeight()),
         width_ / 2, height_ / 2, SampleType::kU, *listener);
 
     is_same &= DataMatricesAreEqual(
-        MakeArrayView(expected_frame_buffer_->DataV(),
-                      expected_frame_buffer_->ChromaWidth() *
-                          expected_frame_buffer_->ChromaHeight()),
-        MakeArrayView(actual_frame_buffer->DataV(),
-                      actual_frame_buffer->ChromaWidth() *
-                          actual_frame_buffer->ChromaHeight()),
+        std::span(expected_frame_buffer_->DataV(),
+                  expected_frame_buffer_->ChromaWidth() *
+                      expected_frame_buffer_->ChromaHeight()),
+        std::span(actual_frame_buffer->DataV(),
+                  actual_frame_buffer->ChromaWidth() *
+                      actual_frame_buffer->ChromaHeight()),
         width_ / 2, height_ / 2, SampleType::kV, *listener);
 
     return is_same;
@@ -136,50 +135,46 @@ class PixelValuesEqualMatcher {
 
   void DescribeTo(std::ostream* os) const {
     *os << "is the actual frame to have the following Y channel:\n";
-    PrintOneChannelToOs(MakeArrayView(expected_frame_buffer_->DataY(),
-                                      expected_frame_buffer_->width() *
-                                          expected_frame_buffer_->height()),
+    PrintOneChannelToOs(std::span(expected_frame_buffer_->DataY(),
+                                  expected_frame_buffer_->width() *
+                                      expected_frame_buffer_->height()),
                         width_, height_, *os);
 
     *os << "is the actual frame to have the following U channel:\n";
-    PrintOneChannelToOs(
-        MakeArrayView(expected_frame_buffer_->DataU(),
-                      expected_frame_buffer_->ChromaWidth() *
-                          expected_frame_buffer_->ChromaHeight()),
-        width_ / 2, height_ / 2, *os);
+    PrintOneChannelToOs(std::span(expected_frame_buffer_->DataU(),
+                                  expected_frame_buffer_->ChromaWidth() *
+                                      expected_frame_buffer_->ChromaHeight()),
+                        width_ / 2, height_ / 2, *os);
 
     *os << "or the actual frame to have the following V channel:\n";
-    PrintOneChannelToOs(
-        MakeArrayView(expected_frame_buffer_->DataV(),
-                      expected_frame_buffer_->ChromaWidth() *
-                          expected_frame_buffer_->ChromaHeight()),
-        width_ / 2, height_ / 2, *os);
+    PrintOneChannelToOs(std::span(expected_frame_buffer_->DataV(),
+                                  expected_frame_buffer_->ChromaWidth() *
+                                      expected_frame_buffer_->ChromaHeight()),
+                        width_ / 2, height_ / 2, *os);
   }
 
   void DescribeNegationTo(std::ostream* os) const {
     *os << "is the actual frame to not have the following Y channel:\n";
-    PrintOneChannelToOs(MakeArrayView(expected_frame_buffer_->DataY(),
-                                      expected_frame_buffer_->width() *
-                                          expected_frame_buffer_->height()),
+    PrintOneChannelToOs(std::span(expected_frame_buffer_->DataY(),
+                                  expected_frame_buffer_->width() *
+                                      expected_frame_buffer_->height()),
                         width_, height_, *os);
 
     *os << "is the actual frame to not have the following U channel:\n";
-    PrintOneChannelToOs(
-        MakeArrayView(expected_frame_buffer_->DataU(),
-                      expected_frame_buffer_->ChromaWidth() *
-                          expected_frame_buffer_->ChromaHeight()),
-        width_ / 2, height_ / 2, *os);
+    PrintOneChannelToOs(std::span(expected_frame_buffer_->DataU(),
+                                  expected_frame_buffer_->ChromaWidth() *
+                                      expected_frame_buffer_->ChromaHeight()),
+                        width_ / 2, height_ / 2, *os);
 
     *os << "and the actual frame to not have the following V channel:\n";
-    PrintOneChannelToOs(
-        MakeArrayView(expected_frame_buffer_->DataV(),
-                      expected_frame_buffer_->ChromaWidth() *
-                          expected_frame_buffer_->ChromaHeight()),
-        width_ / 2, height_ / 2, *os);
+    PrintOneChannelToOs(std::span(expected_frame_buffer_->DataV(),
+                                  expected_frame_buffer_->ChromaWidth() *
+                                      expected_frame_buffer_->ChromaHeight()),
+                        width_ / 2, height_ / 2, *os);
   }
 
-  bool DataMatricesAreEqual(ArrayView<const uint8_t> expected_data,
-                            ArrayView<const uint8_t> actual_data,
+  bool DataMatricesAreEqual(std::span<const uint8_t> expected_data,
+                            std::span<const uint8_t> actual_data,
                             int width,
                             int height,
                             SampleType sample_type,

From 2b891540c1169898293cf2d519886cebd3f4db37 Mon Sep 17 00:00:00 2001
From: Jordan <rop@google.com>
Date: Fri, 23 Jan 2026 08:25:30 +0000
Subject: [PATCH 216/472] Set Update Mechanism to Manual

This CL sets the Update Mechanism to Manual in README files.

Bug: chromium:445311061
Change-Id: I097f2006fef4fa3c4906cbe11f5795cffd878c76
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/443320
Commit-Queue: Jordan Brown <rop@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47069}
---
 README.chromium                               | 1 +
 modules/third_party/portaudio/README.chromium | 1 +
 2 files changed, 2 insertions(+)

diff --git a/README.chromium b/README.chromium
index a97e58fa5c..b4c19c0575 100644
--- a/README.chromium
+++ b/README.chromium
@@ -6,6 +6,7 @@ License: BSD-3-Clause
 License File: LICENSE
 Shipped: yes
 Security Critical: yes
+Update Mechanism: Manual
 Mitigated: CVE-2022-2294
 CVE-2022-2294: Fixed by https://crbug.com/40060120.
 
diff --git a/modules/third_party/portaudio/README.chromium b/modules/third_party/portaudio/README.chromium
index d912d912e2..ac300bb34a 100644
--- a/modules/third_party/portaudio/README.chromium
+++ b/modules/third_party/portaudio/README.chromium
@@ -6,6 +6,7 @@ Date: 2022-04-12
 License: MIT
 License File: LICENSE
 Security Critical: yes
+Update Mechanism: Manual
 Shipped: yes
 
 Description:

From 6d1746c826940b1cdabf3367e75399e4174bf91a Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Thu, 5 Mar 2026 20:07:17 -0800
Subject: [PATCH 217/472] Update WebRTC code version (2026-03-06T04:06:41).

Bug: None
Change-Id: I54dc6681b067878bcf9432c2de06596b7cb4b6d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/454141
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47070}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index 78f8f0461b..64a5a9ac3f 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-05T04:06:26";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-06T04:06:41";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From ce5c7eff7373220a4912eccb456927d1e721549d Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Thu, 5 Mar 2026 20:58:48 +0000
Subject: [PATCH 218/472] Remove unsafe_buffers entries that are obsolete

After the changes to ArrayView, it seems that a number of the
unsafe buffers exceptions have become obsolete.
Removing them from the exceptions list.

Bug: webrtc:478086887
Change-Id: I6924935969d63ca4546cd796074c701910315d8d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453920
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47071}
---
 unsafe_buffers_paths.txt | 46 ----------------------------------------
 1 file changed, 46 deletions(-)

diff --git a/unsafe_buffers_paths.txt b/unsafe_buffers_paths.txt
index ad7cbb7728..359d4a3fa5 100644
--- a/unsafe_buffers_paths.txt
+++ b/unsafe_buffers_paths.txt
@@ -27,7 +27,6 @@
 
 # File-level exceptions
 
--api/array_view.h
 -api/array_view_unittest.cc
 -api/audio/audio_processing.h
 -api/audio/audio_view.h
@@ -68,7 +67,6 @@
 -api/webrtc_sdp_unittest.cc
 -audio/audio_state_unittest.cc
 -audio/channel_receive.cc
--audio/channel_send.cc
 -audio/channel_send_unittest.cc
 -audio/nack_tracker_unittest.cc
 -audio/remix_resample_unittest.cc
@@ -83,7 +81,6 @@
 -call/rtp_demuxer_unittest.cc
 -call/rtp_payload_params.cc
 -call/rtp_payload_params_unittest.cc
--call/rtp_video_sender.cc
 -call/rtx_receive_stream_unittest.cc
 -common_audio/audio_converter.cc
 -common_audio/audio_converter_unittest.cc
@@ -98,7 +95,6 @@
 -common_audio/fir_filter_sse.cc
 -common_audio/include/audio_util.h
 -common_audio/real_fourier_ooura.cc
--common_audio/resampler/push_resampler.cc
 -common_audio/resampler/push_sinc_resampler.cc
 -common_audio/resampler/resampler_unittest.cc
 -common_audio/resampler/sinusoidal_linear_chirp_source.cc
@@ -145,9 +141,6 @@
 -common_audio/signal_processing/sqrt_of_one_minus_x_squared.c
 -common_audio/signal_processing/downsample_fast.c
 -common_audio/signal_processing/dot_product_with_scale.cc
--common_audio/third_party/ooura/fft_size_128/ooura_fft.cc
--common_audio/third_party/ooura/fft_size_128/ooura_fft_sse2.cc
--common_audio/third_party/ooura/fft_size_256/fft4g.cc
 -common_audio/vad/vad_core.c
 -common_audio/vad/vad_core_unittest.cc
 -common_audio/vad/vad_filterbank_unittest.cc
@@ -180,18 +173,15 @@
 -logging/rtc_event_log/encoder/delta_encoding_unittest.cc
 -logging/rtc_event_log/events/rtc_event_field_encoding.cc
 -logging/rtc_event_log/events/rtc_event_field_encoding_parser.cc
--logging/rtc_event_log/events/rtc_event_field_encoding_unittest.cc
 -logging/rtc_event_log/rtc_event_log_parser.cc
 -logging/rtc_event_log/rtc_event_log_parser.h
 -logging/rtc_event_log/rtc_event_log_unittest_helper.cc
--media/base/fake_network_interface.h
 -media/base/fake_rtp.cc
 -media/base/fake_video_renderer.cc
 -media/base/rtp_utils.cc
 -media/base/rtp_utils_unittest.cc
 -media/base/stream_params_unittest.cc
 -media/base/test_utils.h
--media/base/turn_utils.cc
 -media/base/video_adapter_unittest.cc
 -media/engine/simulcast_encoder_adapter.cc
 -media/engine/simulcast_encoder_adapter_unittest.cc
@@ -291,22 +281,13 @@
 -modules/audio_processing/aec3/aec3_fft.cc
 -modules/audio_processing/aec3/block_delay_buffer.cc
 -modules/audio_processing/aec3/block_delay_buffer_unittest.cc
--modules/audio_processing/aec3/block_framer.cc
--modules/audio_processing/aec3/comfort_noise_generator.cc
 -modules/audio_processing/aec3/decimator_unittest.cc
--modules/audio_processing/aec3/dominant_nearend_detector.cc
 -modules/audio_processing/aec3/echo_canceller3.cc
 -modules/audio_processing/aec3/echo_canceller3_unittest.cc
--modules/audio_processing/aec3/echo_remover.cc
 -modules/audio_processing/aec3/filter_analyzer.cc
--modules/audio_processing/aec3/frame_blocker.cc
 -modules/audio_processing/aec3/matched_filter.cc
 -modules/audio_processing/aec3/matched_filter_avx2.cc
--modules/audio_processing/aec3/reverb_decay_estimator.cc
--modules/audio_processing/aec3/reverb_frequency_response.cc
--modules/audio_processing/aec3/signal_dependent_erle_estimator.cc
 -modules/audio_processing/aec3/suppression_filter.cc
--modules/audio_processing/aec3/suppression_gain.cc
 -modules/audio_processing/aec3/transparent_mode.cc
 -modules/audio_processing/aec3/neural_residual_echo_estimator/neural_feature_extractor.cc
 -modules/audio_processing/aec3/neural_residual_echo_estimator/neural_feature_extractor_unittest.cc
@@ -329,13 +310,8 @@
 -modules/audio_processing/agc2/clipping_predictor_unittest.cc
 -modules/audio_processing/agc2/input_volume_controller.cc
 -modules/audio_processing/agc2/input_volume_controller_unittest.cc
--modules/audio_processing/agc2/rnn_vad/auto_correlation.cc
--modules/audio_processing/agc2/rnn_vad/features_extraction.cc
--modules/audio_processing/agc2/rnn_vad/features_extraction_unittest.cc
--modules/audio_processing/agc2/rnn_vad/lp_residual.cc
 -modules/audio_processing/agc2/rnn_vad/ring_buffer.h
 -modules/audio_processing/agc2/rnn_vad/sequence_buffer.h
--modules/audio_processing/agc2/rnn_vad/sequence_buffer_unittest.cc
 -modules/audio_processing/agc2/rnn_vad/spectral_features_internal_unittest.cc
 -modules/audio_processing/agc2/rnn_vad/spectral_features_unittest.cc
 -modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer.h
@@ -344,7 +320,6 @@
 -modules/audio_processing/test/aec_dump_based_simulator.cc
 -modules/audio_processing/test/audio_buffer_tools.cc
 -modules/audio_processing/test/audio_processing_simulator.h
--modules/audio_processing/test/bitexactness_tools.cc
 -modules/audio_processing/capture_levels_adjuster/audio_samples_scaler.cc
 -modules/audio_processing/capture_levels_adjuster/audio_samples_scaler_unittest.cc
 -modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster_unittest.cc
@@ -377,7 +352,6 @@
 -modules/audio_processing/vad/voice_activity_detector.cc
 -modules/audio_processing/ns/noise_suppressor.cc
 -modules/audio_processing/ns/noise_suppressor_unittest.cc
--modules/audio_processing/logging/apm_data_dumper.h
 -modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc
 -modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc
 -modules/desktop_capture/desktop_and_cursor_composer.cc
@@ -410,7 +384,6 @@
 -modules/remote_bitrate_estimator/tools/rtp_to_text.cc
 -modules/rtp_rtcp/source/byte_io.h
 -modules/rtp_rtcp/source/byte_io_unittest.cc
--modules/rtp_rtcp/source/corruption_detection_extension.cc
 -modules/rtp_rtcp/source/fec_private_tables_bursty_unittest.cc
 -modules/rtp_rtcp/source/fec_test_helper.cc
 -modules/rtp_rtcp/source/flexfec_03_header_reader_writer.cc
@@ -462,7 +435,6 @@
 -modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
 -modules/rtp_rtcp/source/rtp_format_video_generic.cc
 -modules/rtp_rtcp/source/rtp_format_vp8.cc
--modules/rtp_rtcp/source/rtp_format_vp8_test_helper.cc
 -modules/rtp_rtcp/source/rtp_format_vp9.cc
 -modules/rtp_rtcp/source/rtp_format_vp9_unittest.cc
 -modules/rtp_rtcp/source/rtp_generic_frame_descriptor.cc
@@ -501,10 +473,6 @@
 -modules/rtp_rtcp/source/rtp_packetizer_av1_unittest.cc
 -modules/rtp_rtcp/test/testFec/test_fec.cc
 -modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc
--modules/third_party/fft/fft.c
--modules/third_party/g711/g711.c
--modules/third_party/g722/g722_decode.c
--modules/third_party/g722/g722_encode.c
 -modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc
 -modules/video_coding/codecs/test/videoprocessor.cc
 -modules/video_coding/codecs/av1/av1_svc_config.cc
@@ -522,12 +490,6 @@
 -modules/video_coding/codecs/vp9/svc_config.cc
 -modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc
 -modules/video_coding/codecs/vp9/test/vp9_impl_unittest.cc
--modules/video_coding/deprecated/decoding_state.cc
--modules/video_coding/deprecated/frame_buffer.cc
--modules/video_coding/deprecated/jitter_buffer_unittest.cc
--modules/video_coding/deprecated/receiver_unittest.cc
--modules/video_coding/deprecated/session_info.cc
--modules/video_coding/deprecated/session_info_unittest.cc
 -modules/video_coding/encoded_frame.cc
 -modules/video_coding/frame_helpers.cc
 -modules/video_coding/frame_helpers_unittest.cc
@@ -561,10 +523,6 @@
 -modules/video_coding/utility/vp8_header_parser.cc
 -modules/video_coding/utility/vp9_uncompressed_header_parser.cc
 -modules/video_coding/video_codec_initializer.cc
--modules/video_capture/linux/device_info_v4l2.cc
--modules/video_capture/linux/pipewire_session.cc
--modules/video_capture/linux/video_capture_v4l2.cc
--modules/video_capture/video_capture_impl.cc
 -net/dcsctp/packet/bounded_byte_reader.h
 -net/dcsctp/packet/bounded_byte_writer.h
 -net/dcsctp/packet/tlv_trait.h
@@ -576,7 +534,6 @@
 -p2p/base/port_unittest.cc
 -p2p/base/stun_request.cc
 -p2p/base/transport_description.cc
--p2p/base/turn_port.cc
 -p2p/base/turn_port_unittest.cc
 -p2p/client/basic_port_allocator.cc
 -p2p/dtls/dtls_transport.h
@@ -586,7 +543,6 @@
 -p2p/test/nat_socket_factory.cc
 -p2p/test/nat_unittest.cc
 -p2p/test/turn_server.cc
--pc/channel_unittest.cc
 -pc/media_session_unittest.cc
 -pc/peer_connection_interface_unittest.cc
 -pc/rtc_stats_collector_unittest.cc
@@ -597,11 +553,9 @@
 -rtc_base/bitstream_reader_unittest.cc
 -rtc_base/bounded_inline_vector.h
 -rtc_base/bounded_inline_vector_impl.h
--rtc_base/bounded_inline_vector_unittest.cc
 -rtc_base/buffer_queue_unittest.cc
 -rtc_base/byte_buffer.cc
 -rtc_base/byte_buffer.h
--rtc_base/byte_order.h
 -rtc_base/checks.cc
 -rtc_base/containers/flat_tree.h
 -rtc_base/containers/flat_tree_unittest.cc

From 0626b4b12768cccfe66a7447625ff4e1238aa0e8 Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Thu, 5 Mar 2026 16:05:25 +0100
Subject: [PATCH 219/472] Replace ArrayView with std::span in sdk/

This change generated using mostly search and replace:
- MakeArrayView replaced with std::span
- ArrayView replaced with std::span
- Remove all array_view includes and dependencies
- Add <span> include in each file std::span is mentioned.
- Make conversion from objc NSData* more explicit

Bug: webrtc:439801349
Change-Id: I61c5ff0668f8612be103d76086d0f89b9e9a62d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453863
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47072}
---
 sdk/BUILD.gn                                  |  5 -----
 sdk/android/BUILD.gn                          |  4 ----
 sdk/android/native_api/jni/java_types.cc      |  6 ++---
 sdk/android/native_api/jni/java_types.h       |  6 ++---
 .../src/jni/audio_device/aaudio_player.cc     |  6 ++---
 .../src/jni/audio_device/aaudio_recorder.cc   |  6 ++---
 .../src/jni/audio_device/opensles_player.cc   |  8 +++----
 .../src/jni/audio_device/opensles_recorder.cc |  4 ++--
 sdk/android/src/jni/video_encoder_wrapper.cc  |  6 ++---
 sdk/android/src/jni/video_encoder_wrapper.h   |  4 ++--
 .../video_codec/RTCVideoDecoderH264.mm        |  9 ++++----
 .../components/video_codec/nalu_rewriter.cc   | 22 +++++++++----------
 .../components/video_codec/nalu_rewriter.h    | 18 +++++++--------
 sdk/objc/native/src/audio/audio_device_ios.mm |  6 ++---
 sdk/objc/native/src/objc_audio_device.mm      |  9 ++++----
 sdk/objc/unittests/nalu_rewriter_xctest.mm    | 13 ++++++-----
 16 files changed, 63 insertions(+), 69 deletions(-)

diff --git a/sdk/BUILD.gn b/sdk/BUILD.gn
index b669d4cbf8..065df50da8 100644
--- a/sdk/BUILD.gn
+++ b/sdk/BUILD.gn
@@ -316,7 +316,6 @@ if (is_ios || is_mac) {
           ":audio_session_observer",
           ":base_objc",
           ":core_audio_helpers_objc",
-          "../api:array_view",
           "../api:scoped_refptr",
           "../api:sequence_checker",
           "../api/audio:audio_device",
@@ -506,7 +505,6 @@ if (is_ios || is_mac) {
       deps = [
         ":audio_device_api_objc",
         ":core_audio_helpers_objc",
-        "../api:array_view",
         "../api:make_ref_counted",
         "../api:refcountedbase",
         "../api:scoped_refptr",
@@ -1221,7 +1219,6 @@ if (is_ios || is_mac) {
             ":videoframebuffer_objc",
             ":videosource_objc",
             ":videotoolbox_objc",
-            "../api:array_view",
             "../api:jsep",
             "../api:rtc_error_matchers",
             "../api:scoped_refptr",
@@ -1756,7 +1753,6 @@ if (is_ios || is_mac) {
         "objc/components/video_codec/nalu_rewriter.h",
       ]
       deps = [
-        "../api:array_view",
         "../common_video",
         "../modules/video_coding:webrtc_h264",
         "../rtc_base:buffer",
@@ -1795,7 +1791,6 @@ if (is_ios || is_mac) {
         ":video_toolbox_cc",
         ":videocodec_objc",
         ":videoframebuffer_objc",
-        "../api:array_view",
         "../api/video:video_rtp_headers",
         "../api/video_codecs:video_codecs_api",
         "../common_video",
diff --git a/sdk/android/BUILD.gn b/sdk/android/BUILD.gn
index 35d1ea4176..b593a1adc2 100644
--- a/sdk/android/BUILD.gn
+++ b/sdk/android/BUILD.gn
@@ -707,7 +707,6 @@ if (current_os == "linux" || is_android) {
       ":internal_jni",
       ":native_api_jni",
       ":videoframe_jni",
-      "../../api:array_view",
       "../../api:libjingle_peerconnection_api",
       "../../api:make_ref_counted",
       "../../api:media_stream_interface",
@@ -1045,7 +1044,6 @@ if (current_os == "linux" || is_android) {
       ":generated_external_classes_jni",
       ":generated_native_api_jni",
       ":internal_jni",
-      "../../api:array_view",
       "../../api:sequence_checker",
       "../../rtc_base:checks",
       "//third_party/abseil-cpp/absl/strings:string_view",
@@ -1346,7 +1344,6 @@ if (current_os == "linux" || is_android) {
       deps = [
         ":audio_device_module_base",
         ":base_jni",
-        "../../api:array_view",
         "../../api:sequence_checker",
         "../../api/task_queue",
         "../../modules/audio_device",
@@ -1375,7 +1372,6 @@ if (current_os == "linux" || is_android) {
     deps = [
       ":audio_device_module_base",
       ":base_jni",
-      "../../api:array_view",
       "../../api:refcountedbase",
       "../../api:scoped_refptr",
       "../../api:sequence_checker",
diff --git a/sdk/android/native_api/jni/java_types.cc b/sdk/android/native_api/jni/java_types.cc
index e257669f10..9581b2d412 100644
--- a/sdk/android/native_api/jni/java_types.cc
+++ b/sdk/android/native_api/jni/java_types.cc
@@ -17,12 +17,12 @@
 #include <cstring>
 #include <map>
 #include <optional>
+#include <span>
 #include <string>
 #include <type_traits>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/sequence_checker.h"
 #include "rtc_base/checks.h"
 #include "sdk/android/generated_external_classes_jni/ArrayList_jni.h"
@@ -248,7 +248,7 @@ ScopedJavaLocalRef<jstring> NativeToJavaString(
 
 ScopedJavaLocalRef<jbyteArray> NativeToJavaByteArray(
     JNIEnv* env,
-    ArrayView<int8_t> container) {
+    std::span<int8_t> container) {
   jni_zero::ScopedJavaLocalRef<jbyteArray> jarray =
       jni_zero::ScopedJavaLocalRef<jbyteArray>::Adopt(
           env, env->NewByteArray(container.size()));
@@ -261,7 +261,7 @@ ScopedJavaLocalRef<jbyteArray> NativeToJavaByteArray(
 
 ScopedJavaLocalRef<jintArray> NativeToJavaIntArray(
     JNIEnv* env,
-    ArrayView<int32_t> container) {
+    std::span<int32_t> container) {
   jni_zero::ScopedJavaLocalRef<jintArray> jarray =
       jni_zero::ScopedJavaLocalRef<jintArray>::Adopt(
           env, env->NewIntArray(container.size()));
diff --git a/sdk/android/native_api/jni/java_types.h b/sdk/android/native_api/jni/java_types.h
index 9f6c0174f6..c736af3d4d 100644
--- a/sdk/android/native_api/jni/java_types.h
+++ b/sdk/android/native_api/jni/java_types.h
@@ -23,11 +23,11 @@
 #include <cstdint>
 #include <map>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/sequence_checker.h"
 #include "rtc_base/checks.h"
 #include "sdk/android/native_api/jni/scoped_java_ref.h"
@@ -245,10 +245,10 @@ ScopedJavaLocalRef<jobjectArray> NativeToJavaObjectArray(
 
 ScopedJavaLocalRef<jbyteArray> NativeToJavaByteArray(
     JNIEnv* env,
-    ArrayView<int8_t> container);
+    std::span<int8_t> container);
 ScopedJavaLocalRef<jintArray> NativeToJavaIntArray(
     JNIEnv* env,
-    ArrayView<int32_t> container);
+    std::span<int32_t> container);
 
 std::vector<int8_t> JavaToNativeByteArray(
     JNIEnv* env,
diff --git a/sdk/android/src/jni/audio_device/aaudio_player.cc b/sdk/android/src/jni/audio_device/aaudio_player.cc
index 2bf9946a94..4ce33b9044 100644
--- a/sdk/android/src/jni/audio_device/aaudio_player.cc
+++ b/sdk/android/src/jni/audio_device/aaudio_player.cc
@@ -11,8 +11,8 @@
 #include "sdk/android/src/jni/audio_device/aaudio_player.h"
 
 #include <memory>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/task_queue/task_queue_base.h"
 #include "modules/audio_device/fine_audio_buffer.h"
 #include "rtc_base/checks.h"
@@ -207,8 +207,8 @@ aaudio_data_callback_result_t AAudioPlayer::OnDataCallback(void* audio_data,
     memset(audio_data, 0, num_bytes);
   } else {
     fine_audio_buffer_->GetPlayoutData(
-        webrtc::MakeArrayView(static_cast<int16_t*>(audio_data),
-                              aaudio_.samples_per_frame() * num_frames),
+        std::span(static_cast<int16_t*>(audio_data),
+                  aaudio_.samples_per_frame() * num_frames),
         static_cast<int>(latency_millis_ + 0.5));
   }
 
diff --git a/sdk/android/src/jni/audio_device/aaudio_recorder.cc b/sdk/android/src/jni/audio_device/aaudio_recorder.cc
index 8867cf8a80..5a3945b051 100644
--- a/sdk/android/src/jni/audio_device/aaudio_recorder.cc
+++ b/sdk/android/src/jni/audio_device/aaudio_recorder.cc
@@ -11,8 +11,8 @@
 #include "sdk/android/src/jni/audio_device/aaudio_recorder.h"
 
 #include <memory>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_device/fine_audio_buffer.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
@@ -190,8 +190,8 @@ aaudio_data_callback_result_t AAudioRecorder::OnDataCallback(
   // Copy recorded audio in `audio_data` to the WebRTC sink using the
   // FineAudioBuffer object.
   fine_audio_buffer_->DeliverRecordedData(
-      webrtc::MakeArrayView(static_cast<const int16_t*>(audio_data),
-                            aaudio_.samples_per_frame() * num_frames),
+      std::span(static_cast<const int16_t*>(audio_data),
+                aaudio_.samples_per_frame() * num_frames),
       static_cast<int>(latency_millis_ + 0.5));
 
   return AAUDIO_CALLBACK_RESULT_CONTINUE;
diff --git a/sdk/android/src/jni/audio_device/opensles_player.cc b/sdk/android/src/jni/audio_device/opensles_player.cc
index 1f659e9e13..f679872ad9 100644
--- a/sdk/android/src/jni/audio_device/opensles_player.cc
+++ b/sdk/android/src/jni/audio_device/opensles_player.cc
@@ -21,9 +21,9 @@
 #include <iterator>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 
-#include "api/array_view.h"
 #include "api/audio/audio_device_defines.h"
 #include "api/environment/environment.h"
 #include "api/scoped_refptr.h"
@@ -430,9 +430,9 @@ void OpenSLESPlayer::EnqueuePlayoutData(bool silence) {
     // OpenSL ES. Use hardcoded delay estimate since OpenSL ES does not support
     // delay estimation.
     fine_audio_buffer_->GetPlayoutData(
-        webrtc::ArrayView<int16_t>(audio_buffers_[buffer_index_].get(),
-                                   audio_parameters_.frames_per_buffer() *
-                                       audio_parameters_.channels()),
+        std::span<int16_t>(audio_buffers_[buffer_index_].get(),
+                           audio_parameters_.frames_per_buffer() *
+                               audio_parameters_.channels()),
         25);
   }
   // Enqueue the decoded audio buffer for playback.
diff --git a/sdk/android/src/jni/audio_device/opensles_recorder.cc b/sdk/android/src/jni/audio_device/opensles_recorder.cc
index efdbff2e79..ccce2cc7f3 100644
--- a/sdk/android/src/jni/audio_device/opensles_recorder.cc
+++ b/sdk/android/src/jni/audio_device/opensles_recorder.cc
@@ -19,9 +19,9 @@
 #include <cstdint>
 #include <iterator>
 #include <memory>
+#include <span>
 #include <utility>
 
-#include "api/array_view.h"
 #include "api/audio/audio_device_defines.h"
 #include "api/environment/environment.h"
 #include "api/scoped_refptr.h"
@@ -394,7 +394,7 @@ void OpenSLESRecorder::ReadBufferQueue() {
   // OpenSL ES anyhow. Hence, as is, the WebRTC based AEC (which would use
   // these estimates) will never be active.
   fine_audio_buffer_->DeliverRecordedData(
-      webrtc::ArrayView<const int16_t>(
+      std::span<const int16_t>(
           audio_buffers_[buffer_index_].get(),
           audio_parameters_.frames_per_buffer() * audio_parameters_.channels()),
       25);
diff --git a/sdk/android/src/jni/video_encoder_wrapper.cc b/sdk/android/src/jni/video_encoder_wrapper.cc
index 492adb2466..93570b0304 100644
--- a/sdk/android/src/jni/video_encoder_wrapper.cc
+++ b/sdk/android/src/jni/video_encoder_wrapper.cc
@@ -16,10 +16,10 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <vector>
 
 #include "absl/memory/memory.h"
-#include "api/array_view.h"
 #include "api/video/render_resolution.h"
 #include "api/video/video_bitrate_allocation.h"
 #include "api/video/video_codec_constants.h"
@@ -322,7 +322,7 @@ void VideoEncoderWrapper::OnEncodedFrame(
   // This is a bit subtle. The `frame` variable from the lambda capture is
   // const. Which implies that (i) we need to make a copy to be able to
   // write to the metadata, and (ii) we should avoid using the .data()
-  // method (including implicit conversion to ArrayView) on the non-const
+  // method (including implicit conversion to std::span) on the non-const
   // copy, since that would trigget a copy operation on the underlying
   // CopyOnWriteBuffer.
   EncodedImage frame_copy = frame;
@@ -366,7 +366,7 @@ int32_t VideoEncoderWrapper::HandleReturnCode(JNIEnv* jni,
   return WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE;
 }
 
-int VideoEncoderWrapper::ParseQp(ArrayView<const uint8_t> buffer) {
+int VideoEncoderWrapper::ParseQp(std::span<const uint8_t> buffer) {
   int qp;
   bool success;
   switch (codec_settings_.codecType) {
diff --git a/sdk/android/src/jni/video_encoder_wrapper.h b/sdk/android/src/jni/video_encoder_wrapper.h
index 2dec560783..91c9715132 100644
--- a/sdk/android/src/jni/video_encoder_wrapper.h
+++ b/sdk/android/src/jni/video_encoder_wrapper.h
@@ -18,9 +18,9 @@
 #include <deque>
 #include <memory>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/video/encoded_image.h"
 #include "api/video/video_bitrate_allocation.h"
 #include "api/video/video_frame.h"
@@ -80,7 +80,7 @@ class VideoEncoderWrapper : public VideoEncoder {
                            const JavaRef<jobject>& j_value,
                            const char* method_name);
 
-  int ParseQp(ArrayView<const uint8_t> buffer);
+  int ParseQp(std::span<const uint8_t> buffer);
 
   CodecSpecificInfo ParseCodecSpecificInfo(const EncodedImage& frame);
 
diff --git a/sdk/objc/components/video_codec/RTCVideoDecoderH264.mm b/sdk/objc/components/video_codec/RTCVideoDecoderH264.mm
index 429d69b82e..7b68164e79 100644
--- a/sdk/objc/components/video_codec/RTCVideoDecoderH264.mm
+++ b/sdk/objc/components/video_codec/RTCVideoDecoderH264.mm
@@ -13,6 +13,8 @@
 
 #import <VideoToolbox/VideoToolbox.h>
 
+#include <span>
+
 #import "base/RTCVideoFrame.h"
 #import "base/RTCVideoFrameBuffer.h"
 #import "components/video_frame_buffer/RTCCVPixelBuffer.h"
@@ -23,7 +25,6 @@
 #import "helpers/UIDevice+RTCDevice.h"
 #endif
 
-#include "api/array_view.h"
 #include "modules/video_coding/include/video_error_codes.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
@@ -116,7 +117,7 @@ void decompressionOutputCallback(void *decoderRef,
 
   webrtc::ScopedCFTypeRef<CMVideoFormatDescriptionRef> inputFormat =
       webrtc::ScopedCF(
-          webrtc::CreateVideoFormatDescription(webrtc::MakeArrayView(
+          webrtc::CreateVideoFormatDescription(std::span<const uint8_t>(
               (uint8_t *)inputImage.buffer.bytes, inputImage.buffer.length)));
   if (inputFormat) {
     // Check if the video format has changed, and reinitialize decoder if
@@ -140,8 +141,8 @@ void decompressionOutputCallback(void *decoderRef,
   }
   CMSampleBufferRef sampleBuffer = nullptr;
   if (!webrtc::H264AnnexBBufferToCMSampleBuffer(
-          webrtc::MakeArrayView((uint8_t *)inputImage.buffer.bytes,
-                                inputImage.buffer.length),
+          std::span<const uint8_t>((uint8_t *)inputImage.buffer.bytes,
+                                   inputImage.buffer.length),
           _videoFormat,
           &sampleBuffer,
           _memoryPool)) {
diff --git a/sdk/objc/components/video_codec/nalu_rewriter.cc b/sdk/objc/components/video_codec/nalu_rewriter.cc
index 5d73376687..271d8597c3 100644
--- a/sdk/objc/components/video_codec/nalu_rewriter.cc
+++ b/sdk/objc/components/video_codec/nalu_rewriter.cc
@@ -16,9 +16,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <cstring>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "common_video/h264/h264_common.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/checks.h"
@@ -137,7 +137,7 @@ bool H264CMSampleBufferToAnnexBBuffer(CMSampleBufferRef avcc_sample_buffer,
   return true;
 }
 
-bool H264AnnexBBufferToCMSampleBuffer(ArrayView<const uint8_t> annexb_buffer,
+bool H264AnnexBBufferToCMSampleBuffer(std::span<const uint8_t> annexb_buffer,
                                       CMVideoFormatDescriptionRef video_format,
                                       CMSampleBufferRef* out_sample_buffer,
                                       CMMemoryPoolRef memory_pool) {
@@ -148,7 +148,7 @@ bool H264AnnexBBufferToCMSampleBuffer(ArrayView<const uint8_t> annexb_buffer,
   AnnexBBufferReader reader(annexb_buffer);
   if (reader.SeekToNextNaluOfType(kSps)) {
     // Buffer contains an SPS NALU - skip it and the following PPS
-    ArrayView<const uint8_t> data;
+    std::span<const uint8_t> data;
     if (!reader.ReadNalu(data)) {
       RTC_LOG(LS_ERROR) << "Failed to read SPS";
       return false;
@@ -205,9 +205,9 @@ bool H264AnnexBBufferToCMSampleBuffer(ArrayView<const uint8_t> annexb_buffer,
 
   // Write Avcc NALUs into block buffer memory.
   AvccBufferWriter writer(
-      MakeArrayView(reinterpret_cast<uint8_t*>(data_ptr), block_buffer_size));
+      std::span(reinterpret_cast<uint8_t*>(data_ptr), block_buffer_size));
   while (reader.BytesRemaining() > 0) {
-    ArrayView<const uint8_t> nalu_data;
+    std::span<const uint8_t> nalu_data;
     if (reader.ReadNalu(nalu_data)) {
       writer.WriteNalu(nalu_data);
     }
@@ -227,7 +227,7 @@ bool H264AnnexBBufferToCMSampleBuffer(ArrayView<const uint8_t> annexb_buffer,
 }
 
 CMVideoFormatDescriptionRef CreateVideoFormatDescription(
-    ArrayView<const uint8_t> annexb_buffer) {
+    std::span<const uint8_t> annexb_buffer) {
   const uint8_t* param_set_ptrs[2] = {};
   size_t param_set_sizes[2] = {};
   AnnexBBufferReader reader(annexb_buffer);
@@ -235,7 +235,7 @@ CMVideoFormatDescriptionRef CreateVideoFormatDescription(
   if (!reader.SeekToNextNaluOfType(kSps)) {
     return nullptr;
   }
-  ArrayView<const uint8_t> param_set;
+  std::span<const uint8_t> param_set;
   if (reader.ReadNalu(param_set)) {
     param_set_ptrs[0] = param_set.data();
     param_set_sizes[0] = param_set.size();
@@ -262,7 +262,7 @@ CMVideoFormatDescriptionRef CreateVideoFormatDescription(
   return description;
 }
 
-AnnexBBufferReader::AnnexBBufferReader(ArrayView<const uint8_t> annexb_buffer)
+AnnexBBufferReader::AnnexBBufferReader(std::span<const uint8_t> annexb_buffer)
     : buffer_(annexb_buffer) {
   offsets_ = H264::FindNaluIndices(annexb_buffer);
   offset_ = offsets_.begin();
@@ -270,7 +270,7 @@ AnnexBBufferReader::AnnexBBufferReader(ArrayView<const uint8_t> annexb_buffer)
 
 AnnexBBufferReader::~AnnexBBufferReader() = default;
 
-bool AnnexBBufferReader::ReadNalu(ArrayView<const uint8_t>& out_nalu) {
+bool AnnexBBufferReader::ReadNalu(std::span<const uint8_t>& out_nalu) {
   if (offset_ == offsets_.end()) {
     return false;
   }
@@ -301,10 +301,10 @@ bool AnnexBBufferReader::SeekToNextNaluOfType(NaluType type) {
   return false;
 }
 
-AvccBufferWriter::AvccBufferWriter(ArrayView<uint8_t> avcc_buffer)
+AvccBufferWriter::AvccBufferWriter(std::span<uint8_t> avcc_buffer)
     : buffer_(avcc_buffer) {}
 
-bool AvccBufferWriter::WriteNalu(ArrayView<const uint8_t> data) {
+bool AvccBufferWriter::WriteNalu(std::span<const uint8_t> data) {
   // Check if we can write this length of data.
   if (data.size() + kAvccHeaderByteSize > BytesRemaining()) {
     return false;
diff --git a/sdk/objc/components/video_codec/nalu_rewriter.h b/sdk/objc/components/video_codec/nalu_rewriter.h
index 507aa67a69..c789aa2fd4 100644
--- a/sdk/objc/components/video_codec/nalu_rewriter.h
+++ b/sdk/objc/components/video_codec/nalu_rewriter.h
@@ -16,9 +16,9 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "common_video/h264/h264_common.h"
 #include "rtc_base/buffer.h"
 
@@ -39,7 +39,7 @@ bool H264CMSampleBufferToAnnexBBuffer(CMSampleBufferRef avcc_sample_buffer,
 // If `is_keyframe` is true then `video_format` is ignored since the format will
 // be read from the buffer. Otherwise `video_format` must be provided.
 // Caller is responsible for releasing the created sample buffer.
-bool H264AnnexBBufferToCMSampleBuffer(ArrayView<const uint8_t> annexb_buffer,
+bool H264AnnexBBufferToCMSampleBuffer(std::span<const uint8_t> annexb_buffer,
                                       CMVideoFormatDescriptionRef video_format,
                                       CMSampleBufferRef* out_sample_buffer,
                                       CMMemoryPoolRef memory_pool);
@@ -48,19 +48,19 @@ bool H264AnnexBBufferToCMSampleBuffer(ArrayView<const uint8_t> annexb_buffer,
 // the Annex B buffer. If there is no such information, nullptr is returned.
 // The caller is responsible for releasing the description.
 CMVideoFormatDescriptionRef CreateVideoFormatDescription(
-    ArrayView<const uint8_t> annexb_buffer);
+    std::span<const uint8_t> annexb_buffer);
 
 // Helper class for reading NALUs from an RTP Annex B buffer.
 class AnnexBBufferReader final {
  public:
-  explicit AnnexBBufferReader(ArrayView<const uint8_t> annexb_buffer);
+  explicit AnnexBBufferReader(std::span<const uint8_t> annexb_buffer);
   ~AnnexBBufferReader();
   AnnexBBufferReader(const AnnexBBufferReader& other) = delete;
   void operator=(const AnnexBBufferReader& other) = delete;
 
   // Returns a pointer to the beginning of the next NALU slice without the
   // header bytes and its length. Returns false if no more slices remain.
-  bool ReadNalu(ArrayView<const uint8_t>& out_nalu);
+  bool ReadNalu(std::span<const uint8_t>& out_nalu);
 
   // Returns the number of unread NALU bytes, including the size of the header.
   // If the buffer has no remaining NALUs this will return zero.
@@ -81,7 +81,7 @@ class AnnexBBufferReader final {
                             size_t length,
                             size_t offset) const;
 
-  const ArrayView<const uint8_t> buffer_;
+  const std::span<const uint8_t> buffer_;
   std::vector<NaluIndex> offsets_;
   std::vector<NaluIndex>::iterator offset_;
 };
@@ -89,20 +89,20 @@ class AnnexBBufferReader final {
 // Helper class for writing NALUs using avcc format into a buffer.
 class AvccBufferWriter final {
  public:
-  explicit AvccBufferWriter(ArrayView<uint8_t> avcc_buffer);
+  explicit AvccBufferWriter(std::span<uint8_t> avcc_buffer);
   ~AvccBufferWriter() {}
   AvccBufferWriter(const AvccBufferWriter& other) = delete;
   void operator=(const AvccBufferWriter& other) = delete;
 
   // Writes the data slice into the buffer. Returns false if there isn't
   // enough space left.
-  bool WriteNalu(ArrayView<const uint8_t> data);
+  bool WriteNalu(std::span<const uint8_t> data);
 
   // Returns the unused bytes in the buffer.
   size_t BytesRemaining() const;
 
  private:
-  ArrayView<uint8_t> buffer_;
+  std::span<uint8_t> buffer_;
 };
 
 }  // namespace webrtc
diff --git a/sdk/objc/native/src/audio/audio_device_ios.mm b/sdk/objc/native/src/audio/audio_device_ios.mm
index 7a95973031..c97a333642 100644
--- a/sdk/objc/native/src/audio/audio_device_ios.mm
+++ b/sdk/objc/native/src/audio/audio_device_ios.mm
@@ -15,8 +15,8 @@
 
 #include <mach/mach_time.h>
 #include <cmath>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/task_queue/pending_task_safety_flag.h"
 #include "helpers.h"
@@ -530,8 +530,8 @@ OSStatus AudioDeviceIOS::OnGetPlayoutData(AudioUnitRenderActionFlags* flags,
   // the native I/O audio unit) and copy the result to the audio buffer in the
   // `io_data` destination.
   fine_audio_buffer_->GetPlayoutData(
-      webrtc::ArrayView<int16_t>(static_cast<int16_t*>(audio_buffer->mData),
-                                 num_frames),
+      std::span<int16_t>(static_cast<int16_t*>(audio_buffer->mData),
+                         num_frames),
       playout_delay_ms);
 
   last_hw_output_latency_update_sample_count_ += num_frames;
diff --git a/sdk/objc/native/src/objc_audio_device.mm b/sdk/objc/native/src/objc_audio_device.mm
index a7400f25fd..2daa50a554 100644
--- a/sdk/objc/native/src/objc_audio_device.mm
+++ b/sdk/objc/native/src/objc_audio_device.mm
@@ -11,6 +11,7 @@
 #include "objc_audio_device.h"
 
 #include <memory>
+#include <span>
 
 #import "components/audio/RTCAudioDevice.h"
 #import "helpers/AudioTimeStamp+Nanoseconds.h"
@@ -447,8 +448,8 @@ OSStatus ObjCAudioDeviceModule::OnDeliverRecordedData(
                audio_buffer->mNumberChannels == 2);
 
     record_fine_audio_buffer_->DeliverRecordedData(
-        webrtc::ArrayView<const int16_t>(
-            static_cast<int16_t*>(audio_buffer->mData), num_frames),
+        std::span<const int16_t>(static_cast<int16_t*>(audio_buffer->mData),
+                                 num_frames),
         cached_recording_delay_ms_.load(),
         AudioTimeStampGetNanoseconds(time_stamp));
     return noErr;
@@ -527,8 +528,8 @@ OSStatus ObjCAudioDeviceModule::OnGetPlayoutData(
   // Read decoded 16-bit PCM samples from WebRTC into the
   // `io_data` destination buffer.
   playout_fine_audio_buffer_->GetPlayoutData(
-      webrtc::ArrayView<int16_t>(static_cast<int16_t*>(audio_buffer->mData),
-                                 num_frames * audio_buffer->mNumberChannels),
+      std::span<int16_t>(static_cast<int16_t*>(audio_buffer->mData),
+                         num_frames * audio_buffer->mNumberChannels),
       cached_playout_delay_ms_.load());
 
   return noErr;
diff --git a/sdk/objc/unittests/nalu_rewriter_xctest.mm b/sdk/objc/unittests/nalu_rewriter_xctest.mm
index 06cbd7aea0..82260a51e1 100644
--- a/sdk/objc/unittests/nalu_rewriter_xctest.mm
+++ b/sdk/objc/unittests/nalu_rewriter_xctest.mm
@@ -8,7 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "api/array_view.h"
+#include <span>
+
 #include "common_video/h264/h264_common.h"
 #include "components/video_codec/nalu_rewriter.h"
 #include "rtc_base/gunit.h"
@@ -74,7 +75,7 @@ static const uint8_t SPS_PPS_BUFFER[] = {
 
 - (void)testReadEmptyInput {
   webrtc::AnnexBBufferReader reader({});
-  webrtc::ArrayView<const uint8_t> nalu;
+  std::span<const uint8_t> nalu;
   XCTAssertEqual(0u, reader.BytesRemaining());
   XCTAssertFalse(reader.ReadNalu(nalu));
 }
@@ -82,7 +83,7 @@ static const uint8_t SPS_PPS_BUFFER[] = {
 - (void)testReadSingleNalu {
   const uint8_t annex_b_test_data[] = {0x00, 0x00, 0x00, 0x01, 0xAA};
   webrtc::AnnexBBufferReader reader(annex_b_test_data);
-  webrtc::ArrayView<const uint8_t> nalu;
+  std::span<const uint8_t> nalu;
   XCTAssertEqual(std::size(annex_b_test_data), reader.BytesRemaining());
   XCTAssertTrue(reader.ReadNalu(nalu));
   XCTAssertEqual(annex_b_test_data + 4, nalu.data());
@@ -94,7 +95,7 @@ static const uint8_t SPS_PPS_BUFFER[] = {
 - (void)testReadSingleNalu3ByteHeader {
   const uint8_t annex_b_test_data[] = {0x00, 0x00, 0x01, 0xAA};
   webrtc::AnnexBBufferReader reader(annex_b_test_data);
-  webrtc::ArrayView<const uint8_t> nalu;
+  std::span<const uint8_t> nalu;
   XCTAssertEqual(std::size(annex_b_test_data), reader.BytesRemaining());
   XCTAssertTrue(reader.ReadNalu(nalu));
   XCTAssertEqual(annex_b_test_data + 3, nalu.data());
@@ -110,7 +111,7 @@ static const uint8_t SPS_PPS_BUFFER[] = {
                                        0x00, 0x00, 0x00, 0xFF};
   // clang-format on
   webrtc::AnnexBBufferReader reader(annex_b_test_data);
-  webrtc::ArrayView<const uint8_t> nalu;
+  std::span<const uint8_t> nalu;
   XCTAssertEqual(0u, reader.BytesRemaining());
   XCTAssertFalse(reader.ReadNalu(nalu));
 }
@@ -124,7 +125,7 @@ static const uint8_t SPS_PPS_BUFFER[] = {
                                        0x00, 0x00, 0x01, 0xAA, 0xBB};
   // clang-format on
   webrtc::AnnexBBufferReader reader(annex_b_test_data);
-  webrtc::ArrayView<const uint8_t> nalu;
+  std::span<const uint8_t> nalu;
   XCTAssertEqual(std::size(annex_b_test_data), reader.BytesRemaining());
   XCTAssertTrue(reader.ReadNalu(nalu));
   XCTAssertEqual(annex_b_test_data + 4, nalu.data());

From 0bcd472228e69f80322808d03eb924fa27d15b63 Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Thu, 5 Mar 2026 16:12:01 +0100
Subject: [PATCH 220/472] Replace ArrayView with std::span in rtc_tools/

Bug: webrtc:439801349
Change-Id: If9fe1e47ce3f3bf93b873acf8d1784610975ef15
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453900
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47073}
---
 rtc_tools/BUILD.gn                            |  2 --
 rtc_tools/frame_analyzer/frame_analyzer.cc    |  4 +--
 .../frame_analyzer/video_color_aligner.cc     | 32 +++++++++----------
 3 files changed, 18 insertions(+), 20 deletions(-)

diff --git a/rtc_tools/BUILD.gn b/rtc_tools/BUILD.gn
index f490928b83..3b2b6c0c49 100644
--- a/rtc_tools/BUILD.gn
+++ b/rtc_tools/BUILD.gn
@@ -99,7 +99,6 @@ rtc_library("video_quality_analysis") {
   ]
   deps = [
     ":video_file_reader",
-    "../api:array_view",
     "../api:make_ref_counted",
     "../api:scoped_refptr",
     "../api/numerics",
@@ -142,7 +141,6 @@ if (!is_component_build) {
       ":video_file_reader",
       ":video_file_writer",
       ":video_quality_analysis",
-      "../api:array_view",
       "../api:make_ref_counted",
       "../api:scoped_refptr",
       "../api/test/metrics:chrome_perf_dashboard_metrics_exporter",
diff --git a/rtc_tools/frame_analyzer/frame_analyzer.cc b/rtc_tools/frame_analyzer/frame_analyzer.cc
index 71367c6640..05e1ac5d6d 100644
--- a/rtc_tools/frame_analyzer/frame_analyzer.cc
+++ b/rtc_tools/frame_analyzer/frame_analyzer.cc
@@ -14,6 +14,7 @@
 #include <cstdio>
 #include <cstdlib>
 #include <memory>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
@@ -21,7 +22,6 @@
 #include "absl/flags/flag.h"
 #include "absl/flags/parse.h"
 #include "absl/strings/match.h"
-#include "api/array_view.h"
 #include "api/scoped_refptr.h"
 #include "api/test/metrics/chrome_perf_dashboard_metrics_exporter.h"
 #include "api/test/metrics/global_metrics_logger_and_exporter.h"
@@ -138,7 +138,7 @@ class FrameAnalyzerMetricsExporter : public webrtc::test::MetricsExporter {
   FrameAnalyzerMetricsExporter& operator=(const FrameAnalyzerMetricsExporter&) =
       delete;
 
-  bool Export(webrtc::ArrayView<const webrtc::test::Metric> metrics) override {
+  bool Export(std::span<const webrtc::test::Metric> metrics) override {
     for (const webrtc::test::Metric& metric : metrics) {
       PrintMetric(metric);
     }
diff --git a/rtc_tools/frame_analyzer/video_color_aligner.cc b/rtc_tools/frame_analyzer/video_color_aligner.cc
index b4515883a0..f63a190d4a 100644
--- a/rtc_tools/frame_analyzer/video_color_aligner.cc
+++ b/rtc_tools/frame_analyzer/video_color_aligner.cc
@@ -15,9 +15,9 @@
 #include <cmath>
 #include <cstddef>
 #include <cstdint>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/make_ref_counted.h"
 #include "api/scoped_refptr.h"
 #include "api/video/i420_buffer.h"
@@ -36,11 +36,11 @@ namespace {
 // Helper function for AdjustColors(). This functions calculates a single output
 // row for y with the given color coefficients. The u/v channels are assumed to
 // be subsampled by a factor of 2, which is the case of I420.
-void CalculateYChannel(ArrayView<const uint8_t> y_data,
-                       ArrayView<const uint8_t> u_data,
-                       ArrayView<const uint8_t> v_data,
+void CalculateYChannel(std::span<const uint8_t> y_data,
+                       std::span<const uint8_t> u_data,
+                       std::span<const uint8_t> v_data,
                        const std::array<float, 4>& coeff,
-                       ArrayView<uint8_t> output) {
+                       std::span<uint8_t> output) {
   RTC_CHECK_EQ(y_data.size(), output.size());
   // Each u/v element represents two y elements. Make sure we have enough to
   // cover the Y values.
@@ -75,11 +75,11 @@ void CalculateYChannel(ArrayView<const uint8_t> y_data,
 // Helper function for AdjustColors(). This functions calculates a single output
 // row for either u or v, with the given color coefficients. Y, U, and V are
 // assumed to be the same size, i.e. no subsampling.
-void CalculateUVChannel(ArrayView<const uint8_t> y_data,
-                        ArrayView<const uint8_t> u_data,
-                        ArrayView<const uint8_t> v_data,
+void CalculateUVChannel(std::span<const uint8_t> y_data,
+                        std::span<const uint8_t> u_data,
+                        std::span<const uint8_t> v_data,
                         const std::array<float, 4>& coeff,
-                        ArrayView<uint8_t> output) {
+                        std::span<uint8_t> output) {
   RTC_CHECK_EQ(y_data.size(), u_data.size());
   RTC_CHECK_EQ(y_data.size(), v_data.size());
   RTC_CHECK_EQ(y_data.size(), output.size());
@@ -201,13 +201,13 @@ scoped_refptr<I420BufferInterface> AdjustColors(
   // Fill in the adjusted data row by row.
   for (int y = 0; y < frame->height(); ++y) {
     const int half_y = y / 2;
-    ArrayView<const uint8_t> y_row(frame->DataY() + frame->StrideY() * y,
+    std::span<const uint8_t> y_row(frame->DataY() + frame->StrideY() * y,
                                    frame->width());
-    ArrayView<const uint8_t> u_row(frame->DataU() + frame->StrideU() * half_y,
+    std::span<const uint8_t> u_row(frame->DataU() + frame->StrideU() * half_y,
                                    frame->ChromaWidth());
-    ArrayView<const uint8_t> v_row(frame->DataV() + frame->StrideV() * half_y,
+    std::span<const uint8_t> v_row(frame->DataV() + frame->StrideV() * half_y,
                                    frame->ChromaWidth());
-    ArrayView<uint8_t> output_y_row(
+    std::span<uint8_t> output_y_row(
         adjusted_frame->MutableDataY() + adjusted_frame->StrideY() * y,
         frame->width());
 
@@ -215,13 +215,13 @@ scoped_refptr<I420BufferInterface> AdjustColors(
 
     // Chroma channels only exist every second row for I420.
     if (y % 2 == 0) {
-      ArrayView<const uint8_t> downscaled_y_row(
+      std::span<const uint8_t> downscaled_y_row(
           downscaled_y_plane.data() + frame->ChromaWidth() * half_y,
           frame->ChromaWidth());
-      ArrayView<uint8_t> output_u_row(
+      std::span<uint8_t> output_u_row(
           adjusted_frame->MutableDataU() + adjusted_frame->StrideU() * half_y,
           frame->ChromaWidth());
-      ArrayView<uint8_t> output_v_row(
+      std::span<uint8_t> output_v_row(
           adjusted_frame->MutableDataV() + adjusted_frame->StrideV() * half_y,
           frame->ChromaWidth());
 

From af411f52472297833af0813b727c7b4644221f04 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Thu, 5 Mar 2026 23:12:04 +0100
Subject: [PATCH 221/472] Update raw audio sink management for unsignaled
 streams
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Ensure the default raw audio sink is correctly detached when an
unsignaled stream is removed or promoted. Previously, when a stream
was deregistered, the associated ProxySink could retain a bad
pointer to the default sink.

This change modifies the VoiceEngine to:
* Explicitly clear the raw audio sink from a stream when it is
  removed from the unsignaled stream list.
* Properly hand over the default sink to the next available
  unsignaled stream if the current primary stream is deregistered.
* Simplify the logic for detaching the sink from the previous latest
  stream when a new unsignaled stream is identified.

Bug: chromium:486421954
Fixes: chromium:486421954
Change-Id: Ied7062e95a1749a9d8f95f77ec2b0dc81ab00ad1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/454040
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47074}
---
 media/engine/webrtc_voice_engine.cc          | 26 ++++++++++++++--
 media/engine/webrtc_voice_engine_unittest.cc | 31 ++++++++++++++++++++
 2 files changed, 54 insertions(+), 3 deletions(-)

diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index 857f6dbef9..d13ce532fa 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -2698,9 +2698,14 @@ bool WebRtcVoiceReceiveChannel::MaybeCreateDefaultReceiveStream(
   // it up to the *latest* unsignaled stream we've seen, in order to support
   // the case where the SSRC of one unsignaled stream changes.
   if (default_sink_) {
-    for (uint32_t drop_ssrc : unsignaled_recv_ssrcs_) {
-      auto it = recv_streams_.find(drop_ssrc);
-      it->second->SetRawAudioSink(nullptr);
+    // The new ssrc has already been appended to `unsignaled_recv_ssrcs_`.
+    // If there are 2 or more streams, the stream at `size - 2` is the previous
+    // latest stream which currently possesses the default sink.
+    if (unsignaled_recv_ssrcs_.size() >= 2) {
+      // Detach the default sink from the previous latest stream.
+      uint32_t prev_ssrc =
+          unsignaled_recv_ssrcs_[unsignaled_recv_ssrcs_.size() - 2];
+      SetRawAudioSink(prev_ssrc, nullptr);
     }
     std::unique_ptr<AudioSinkInterface> proxy_sink(
         new ProxySink(default_sink_.get()));
@@ -2907,7 +2912,22 @@ bool WebRtcVoiceReceiveChannel::MaybeDeregisterUnsignaledRecvStream(
   RTC_DCHECK_RUN_ON(worker_thread_);
   auto it = absl::c_find(unsignaled_recv_ssrcs_, ssrc);
   if (it != unsignaled_recv_ssrcs_.end()) {
+    bool is_latest_unsignaled = (it == unsignaled_recv_ssrcs_.end() - 1);
     unsignaled_recv_ssrcs_.erase(it);
+    if (default_sink_) {
+      // Detach the default sink from the deregistered stream. This is needed
+      // to prevent the ProxySink from holding a dangling pointer to the
+      // default_sink_.
+      SetRawAudioSink(ssrc, nullptr);
+      if (is_latest_unsignaled && !unsignaled_recv_ssrcs_.empty()) {
+        // The deregistered stream was the latest unsignaled stream, so it held
+        // the default sink. Since it was removed, we must pass the default sink
+        // to the *new* latest unsignaled stream via a new ProxySink.
+        std::unique_ptr<AudioSinkInterface> proxy_sink(
+            new ProxySink(default_sink_.get()));
+        SetRawAudioSink(unsignaled_recv_ssrcs_.back(), std::move(proxy_sink));
+      }
+    }
     return true;
   }
   return false;
diff --git a/media/engine/webrtc_voice_engine_unittest.cc b/media/engine/webrtc_voice_engine_unittest.cc
index 831ca66c92..0d66c96a08 100644
--- a/media/engine/webrtc_voice_engine_unittest.cc
+++ b/media/engine/webrtc_voice_engine_unittest.cc
@@ -3708,6 +3708,37 @@ TEST_P(WebRtcVoiceEngineTestFake, OnReadyToSendSignalsNetworkState) {
   EXPECT_EQ(kNetworkUp, call_.GetNetworkState(MediaType::VIDEO));
 }
 
+// Test that when an unsignaled stream is promoted to a signaled stream,
+// its ProxySink doesn't hold a dangling raw pointer if the default sink
+// is subsequently destroyed.
+TEST_P(WebRtcVoiceEngineTestFake,
+       ProxySinkSurvivesUnsignaledToSignaledPromotion) {
+  EXPECT_TRUE(SetupChannel());
+  std::unique_ptr<FakeAudioSink> fake_sink(new FakeAudioSink());
+
+  // Set the default sink.
+  receive_channel_->SetDefaultRawAudioSink(std::move(fake_sink));
+
+  // Deliver an RTP packet to create an unsignaled stream.
+  DeliverPacket(kPcmuFrame);
+  const AudioSinkInterface* proxy_sink = GetRecvStream(kSsrc1).sink();
+  EXPECT_NE(nullptr, proxy_sink);
+
+  // Promote the unsignaled stream to a signaled stream.
+  StreamParams sp = StreamParams::CreateLegacy(kSsrc1);
+  EXPECT_TRUE(receive_channel_->AddRecvStream(sp));
+
+  // The proxy sink should be removed from the stream upon promotion.
+  EXPECT_EQ(nullptr, GetRecvStream(kSsrc1).sink());
+
+  // Destroy the original sink by passing nullptr.
+  receive_channel_->SetDefaultRawAudioSink(nullptr);
+
+  // Note: calling proxy_sink->OnData would crash here if the proxy_sink
+  // would still be attached to the stream and hold a dangling pointer to
+  // default_sink_. But we've verified that it's detached, so that won't happen.
+}
+
 // Test that playout is still started after changing parameters
 TEST_P(WebRtcVoiceEngineTestFake, PreservePlayoutWhenRecreateRecvStream) {
   SetupRecvStream();

From 7f5a8b656cf69708359a8179d226dda8d2206205 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Jes=C3=BAs=20de=20Vicente=20Pe=C3=B1a?=
 <devicentepena@webrtc.org>
Date: Thu, 5 Mar 2026 14:16:53 +0100
Subject: [PATCH 222/472] Propagate delay estimator fields to AecState.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Additionally, it removes some unused delay estimator fields such as `blocks_since_last_change` or `blocks_since_last_update`. The rest of the fields are correctly propagated from
RenderDelayController to AecState.

This change is bitexact.

Bug: webrtc:489633233
Change-Id: Ibee53a4e9b7c0fdac310ec5b4dee170ec5dc208b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453641
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47075}
---
 modules/audio_processing/aec3/aec_state.cc    |  3 +-
 .../audio_processing/aec3/delay_estimate.h    |  2 --
 .../aec3/render_delay_controller.cc           | 29 ++-----------------
 3 files changed, 3 insertions(+), 31 deletions(-)

diff --git a/modules/audio_processing/aec3/aec_state.cc b/modules/audio_processing/aec3/aec_state.cc
index abc2ccb0d6..750c89ab2f 100644
--- a/modules/audio_processing/aec3/aec_state.cc
+++ b/modules/audio_processing/aec3/aec_state.cc
@@ -380,8 +380,7 @@ void AecState::FilterDelay::Update(
     const std::optional<DelayEstimate>& external_delay,
     size_t blocks_with_proper_filter_adaptation) {
   // Update the delay based on the external delay.
-  if (external_delay &&
-      (!external_delay_ || external_delay_->delay != external_delay->delay)) {
+  if (external_delay) {
     external_delay_ = external_delay;
   }
 
diff --git a/modules/audio_processing/aec3/delay_estimate.h b/modules/audio_processing/aec3/delay_estimate.h
index 7838a0c255..2ba03fd98b 100644
--- a/modules/audio_processing/aec3/delay_estimate.h
+++ b/modules/audio_processing/aec3/delay_estimate.h
@@ -24,8 +24,6 @@ struct DelayEstimate {
 
   Quality quality;
   size_t delay;
-  size_t blocks_since_last_change = 0;
-  size_t blocks_since_last_update = 0;
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_processing/aec3/render_delay_controller.cc b/modules/audio_processing/aec3/render_delay_controller.cc
index a4046f26c1..f7243d1c7c 100644
--- a/modules/audio_processing/aec3/render_delay_controller.cc
+++ b/modules/audio_processing/aec3/render_delay_controller.cc
@@ -58,7 +58,6 @@ class RenderDelayControllerImpl final : public RenderDelayController {
   RenderDelayControllerMetrics metrics_;
   std::optional<DelayEstimate> delay_samples_;
   size_t capture_call_counter_ = 0;
-  int delay_change_counter_ = 0;
   DelayEstimate::Quality last_delay_estimate_quality_;
 };
 
@@ -102,7 +101,6 @@ void RenderDelayControllerImpl::Reset(bool reset_delay_confidence) {
   delay_ = std::nullopt;
   delay_samples_ = std::nullopt;
   delay_estimator_.Reset(reset_delay_confidence);
-  delay_change_counter_ = 0;
   if (reset_delay_confidence) {
     last_delay_estimate_quality_ = DelayEstimate::Quality::kCoarse;
   }
@@ -119,29 +117,7 @@ std::optional<DelayEstimate> RenderDelayControllerImpl::GetDelay(
   auto delay_samples = delay_estimator_.EstimateDelay(render_buffer, capture);
 
   if (delay_samples) {
-    if (!delay_samples_ || delay_samples->delay != delay_samples_->delay) {
-      delay_change_counter_ = 0;
-    }
-    if (delay_samples_) {
-      delay_samples_->blocks_since_last_change =
-          delay_samples_->delay == delay_samples->delay
-              ? delay_samples_->blocks_since_last_change + 1
-              : 0;
-      delay_samples_->blocks_since_last_update = 0;
-      delay_samples_->delay = delay_samples->delay;
-      delay_samples_->quality = delay_samples->quality;
-    } else {
-      delay_samples_ = delay_samples;
-    }
-  } else {
-    if (delay_samples_) {
-      ++delay_samples_->blocks_since_last_change;
-      ++delay_samples_->blocks_since_last_update;
-    }
-  }
-
-  if (delay_change_counter_ < 2 * kNumBlocksPerSecond) {
-    ++delay_change_counter_;
+    delay_samples_ = delay_samples;
   }
 
   if (delay_samples_) {
@@ -151,7 +127,7 @@ std::optional<DelayEstimate> RenderDelayControllerImpl::GetDelay(
         delay_samples_->quality == DelayEstimate::Quality::kRefined;
     delay_ = ComputeBufferDelay(
         delay_, use_hysteresis ? hysteresis_limit_blocks_ : 0, *delay_samples_);
-    last_delay_estimate_quality_ = delay_samples_->quality;
+    last_delay_estimate_quality_ = delay_->quality;
   }
 
   metrics_.Update(delay_samples_ ? std::optional<size_t>(delay_samples_->delay)
@@ -163,7 +139,6 @@ std::optional<DelayEstimate> RenderDelayControllerImpl::GetDelay(
                         delay_samples ? delay_samples->delay : 0);
   data_dumper_->DumpRaw("aec3_render_delay_controller_buffer_delay",
                         delay_ ? delay_->delay : 0);
-
   return delay_;
 }
 

From ecde302f3f4e4f4149ad7eda697bb9309955e57d Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Fri, 6 Mar 2026 09:37:17 +0100
Subject: [PATCH 223/472] Check maximum buffer size in
 ResamplerHelper::MaybeResample
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Verify that the target sample count does not exceed the maximum allowed
size for an AudioFrame. Previously, requesting a resample operation with
a high number of channels or a high sample rate could result in a target
data size that exceeded the internal limits of the AudioFrame class.

This change adds a validation check before starting the resampling
process. If the calculated target size—based on the desired sample rate
and channel count—surpasses kMaxDataSizeSamples, the operation now
safely aborts. In such cases, the error is logged, the audio frame is
muted to avoid undefined behavior, and the function returns false.

Bug: chromium:486349161
Fixes: chromium:486349161
Change-Id: Ia0d8abcd390f90a590c07c0606f9b6c968f663e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/454000
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47076}
---
 modules/audio_coding/BUILD.gn                 |  2 +
 modules/audio_coding/DEPS                     |  1 +
 modules/audio_coding/acm2/acm_resampler.cc    | 15 ++++
 .../acm2/acm_resampler_unittest.cc            | 79 +++++++++++++++++++
 4 files changed, 97 insertions(+)
 create mode 100644 modules/audio_coding/acm2/acm_resampler_unittest.cc

diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index 016485c4e8..132b293f79 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -47,6 +47,7 @@ rtc_library("audio_coding") {
     "../../api/neteq:default_neteq_factory",
     "../../api/neteq:neteq_api",
     "../../api/units:timestamp",
+    "../../audio/utility:audio_frame_operations",
     "../../common_audio",
     "../../common_audio:common_audio_c",
     "../../rtc_base:buffer",
@@ -1400,6 +1401,7 @@ if (rtc_include_tests) {
 
       sources = [
         "acm2/acm_remixing_unittest.cc",
+        "acm2/acm_resampler_unittest.cc",
         "acm2/audio_coding_module_unittest.cc",
         "acm2/call_statistics_unittest.cc",
         "audio_network_adaptor/audio_network_adaptor_impl_unittest.cc",
diff --git a/modules/audio_coding/DEPS b/modules/audio_coding/DEPS
index 3dc9624a4b..be1291bf2c 100644
--- a/modules/audio_coding/DEPS
+++ b/modules/audio_coding/DEPS
@@ -1,4 +1,5 @@
 include_rules = [
+  "+audio/utility",
   "+call",
   "+common_audio",
   "+logging/rtc_event_log",
diff --git a/modules/audio_coding/acm2/acm_resampler.cc b/modules/audio_coding/acm2/acm_resampler.cc
index a8c4ba8ef0..e9c2861101 100644
--- a/modules/audio_coding/acm2/acm_resampler.cc
+++ b/modules/audio_coding/acm2/acm_resampler.cc
@@ -11,12 +11,15 @@
 #include "modules/audio_coding/acm2/acm_resampler.h"
 
 #include <array>
+#include <cstddef>
 #include <cstdint>
 
 #include "absl/algorithm/container.h"
 #include "api/audio/audio_frame.h"
 #include "api/audio/audio_view.h"
+#include "audio/utility/audio_frame_operations.h"
 #include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
 
 namespace webrtc {
 namespace acm2 {
@@ -38,6 +41,18 @@ bool ResamplerHelper::MaybeResample(int desired_sample_rate_hz,
       (desired_sample_rate_hz != -1) &&
       (current_sample_rate_hz != desired_sample_rate_hz);
 
+  if (need_resampling) {
+    const size_t target_size =
+        audio_frame->num_channels_ *
+        SampleRateToDefaultChannelSize(desired_sample_rate_hz);
+    if (target_size > AudioFrame::kMaxDataSizeSamples) {
+      RTC_LOG(LS_ERROR) << "AudioFrame cannot hold resampled data.";
+      AudioFrameOperations::Mute(audio_frame);
+      audio_frame->SetSampleRateAndChannelSize(desired_sample_rate_hz);
+      return false;
+    }
+  }
+
   if (need_resampling && !resampled_last_output_frame_) {
     // Prime the resampler with the last frame.
     InterleavedView<const int16_t> src(last_audio_buffer_.data(),
diff --git a/modules/audio_coding/acm2/acm_resampler_unittest.cc b/modules/audio_coding/acm2/acm_resampler_unittest.cc
new file mode 100644
index 0000000000..f59715150b
--- /dev/null
+++ b/modules/audio_coding/acm2/acm_resampler_unittest.cc
@@ -0,0 +1,79 @@
+/*
+ *  Copyright (c) 2026 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/acm2/acm_resampler.h"
+
+#include <cstddef>
+#include <cstdint>
+#include <vector>
+
+#include "api/audio/audio_frame.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace acm2 {
+
+TEST(ResamplerHelperTest, MaybeResampleCheckForMaxSize) {
+  ResamplerHelper resampler;
+  AudioFrame audio_frame;
+
+  // Create an audio frame that requires resampling from 32kHz to 48kHz
+  // with a very high number of channels (24).
+  const int kCurrentSampleRateHz = 32000;
+  const int kDesiredSampleRateHz = 48000;
+  const size_t kChannels = 24;
+
+  // 10 ms of data at 32kHz = 320 samples per channel.
+  std::vector<int16_t> dummy_data(320 * 24, 0);
+  audio_frame.UpdateFrame(0, dummy_data.data(), 320, kCurrentSampleRateHz,
+                          AudioFrame::kNormalSpeech, AudioFrame::kVadActive,
+                          kChannels);
+
+  // The resampler prime path will attempt to allocate a buffer that is
+  // kChannels * (kDesiredSampleRateHz / 100) = 24 * 480 = 11520 samples,
+  // which exceeds AudioFrame::kMaxDataSizeSamples (7680).
+  const bool resample_success =
+      resampler.MaybeResample(kDesiredSampleRateHz, &audio_frame);
+
+  // Verify that MaybeResample correctly detects the buffer size condition and
+  // safely aborts the operation by returning false and muting the frame.
+  EXPECT_FALSE(resample_success);
+  EXPECT_TRUE(audio_frame.muted());
+  EXPECT_EQ(audio_frame.sample_rate_hz_, kDesiredSampleRateHz);
+  EXPECT_EQ(audio_frame.num_channels_, kChannels);
+}
+
+TEST(ResamplerHelperTest, MaybeResampleValidMaxSize) {
+  ResamplerHelper resampler;
+  AudioFrame audio_frame;
+
+  // Ensure that resampling within the valid buffer size does not trigger the
+  // muting behavior. We'll use a valid number of channels (e.g. 1) that will
+  // not exceed the bounds.
+  const int kCurrentSampleRateHz = 32000;
+  const int kDesiredSampleRateHz = 48000;
+  const size_t kChannels = 1;
+
+  std::vector<int16_t> dummy_data(320 * 1, 1000);
+  audio_frame.UpdateFrame(0, dummy_data.data(), 320, kCurrentSampleRateHz,
+                          AudioFrame::kNormalSpeech, AudioFrame::kVadActive,
+                          kChannels);
+
+  const bool resample_success =
+      resampler.MaybeResample(kDesiredSampleRateHz, &audio_frame);
+
+  EXPECT_TRUE(resample_success);
+  EXPECT_FALSE(audio_frame.muted());
+  EXPECT_EQ(audio_frame.sample_rate_hz_, kDesiredSampleRateHz);
+  EXPECT_EQ(audio_frame.num_channels_, kChannels);
+}
+
+}  // namespace acm2
+}  // namespace webrtc

From 924dd2ecf8f8bac0ed8e41e7829de586b3c65ab8 Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Fri, 6 Mar 2026 11:58:09 +0000
Subject: [PATCH 224/472] Reduce the number of webrtc:: prefixes in code
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This is achieved by moving most anonymous namespaces inside the
webrtc namespace and modifying as appropriate.

Bug: None
Change-Id: Iff299c45f92ae5aaa736722ed6630df9ad5525bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/454240
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47077}
---
 api/rtc_error.cc                              | 22 ++++----
 call/rtp_bitrate_configurator.cc              |  3 +-
 common_audio/window_generator.cc              | 10 ++--
 common_video/h264/sps_parser.cc               |  4 +-
 common_video/h265/h265_bitstream_parser.cc    |  4 +-
 common_video/h265/h265_pps_parser.cc          |  4 +-
 common_video/h265/h265_sps_parser.cc          | 11 ++--
 .../events/rtc_event_field_encoding.cc        |  4 +-
 .../events/rtc_event_field_encoding_parser.cc | 27 +++++-----
 media/base/video_adapter.cc                   |  4 +-
 media/engine/fake_video_codec_factory.cc      |  6 +--
 .../audio_coding/codecs/opus/test/blocker.cc  |  4 +-
 .../codecs/opus/test/blocker_unittest.cc      |  8 +--
 .../opus/test/lapped_transform_unittest.cc    | 10 ++--
 .../neteq/packet_buffer_unittest.cc           | 18 +++----
 modules/audio_coding/test/TestAllCodecs.cc    |  4 +-
 .../audio_processing/aec3/matched_filter.cc   | 29 +++++-----
 .../residual_echo_detector.cc                 |  6 +--
 .../audio_processing/test/protobuf_utils.cc   |  4 +-
 .../linux/x11/mouse_cursor_monitor_x11.cc     |  4 +-
 modules/rtp_rtcp/source/dtmf_queue.cc         |  3 +-
 .../source/fec_private_tables_bursty.cc       |  3 +-
 .../source/fec_private_tables_random.cc       |  3 +-
 .../forward_error_correction_internal.cc      |  3 +-
 p2p/base/default_ice_transport_factory.cc     | 12 ++---
 p2p/base/stun_dictionary_unittest.cc          | 16 +++---
 p2p/base/turn_port_unittest.cc                | 54 +++++++------------
 p2p/base/wrapping_active_ice_controller.cc    |  5 --
 p2p/client/basic_port_allocator_unittest.cc   | 12 ++---
 ...dtls_stun_piggyback_controller_unittest.cc | 29 +++++-----
 p2p/dtls/dtls_utils.cc                        |  4 +-
 pc/jitter_buffer_delay.cc                     |  4 +-
 pc/media_session.cc                           | 50 +++++++----------
 pc/media_session_unittest.cc                  |  1 -
 pc/rtp_sender_receiver_unittest.cc            |  6 +--
 rtc_base/bit_buffer.cc                        |  6 +--
 rtc_base/checks.cc                            |  3 +-
 rtc_base/cpu_info.cc                          |  8 +--
 rtc_base/cpu_time.cc                          |  4 +-
 rtc_base/cpu_time_unittest.cc                 | 11 ++--
 rtc_base/event_tracer_unittest.cc             | 12 ++---
 .../event_based_exponential_moving_average.cc |  4 +-
 ...sed_exponential_moving_average_unittest.cc |  4 +-
 rtc_base/openssl_session_cache_unittest.cc    |  4 +-
 rtc_base/physical_socket_server.cc            | 13 ++---
 rtc_base/thread.cc                            |  4 +-
 rtc_base/win/create_direct3d_device.cc        |  4 +-
 rtc_base/win/get_activation_factory.cc        |  4 +-
 rtc_base/win/hstring.cc                       |  4 +-
 rtc_base/win/windows_version.cc               |  4 ++
 .../peer_connection_client.cc                 | 48 ++++++++---------
 rtc_tools/video_replay.cc                     |  2 +
 sdk/android/src/jni/jni_helpers.h             |  3 --
 test/fake_vp8_encoder.cc                      |  8 +--
 test/network/emulated_turn_server.cc          | 48 ++++++++---------
 test/rtp_file_reader.cc                       |  6 +--
 test/testsupport/test_artifacts.cc            | 10 +++-
 57 files changed, 289 insertions(+), 316 deletions(-)

diff --git a/api/rtc_error.cc b/api/rtc_error.cc
index 65f220bc94..cda1fe15f3 100644
--- a/api/rtc_error.cc
+++ b/api/rtc_error.cc
@@ -16,6 +16,8 @@
 #include "absl/strings/string_view.h"
 #include "rtc_base/strings/string_builder.h"
 
+namespace webrtc {
+
 namespace {
 
 absl::string_view kRTCErrorTypeNames[] = {
@@ -32,11 +34,10 @@ absl::string_view kRTCErrorTypeNames[] = {
     "INTERNAL_ERROR",
     "OPERATION_ERROR_WITH_DATA",
 };
-static_assert(
-    static_cast<int>(webrtc::RTCErrorType::OPERATION_ERROR_WITH_DATA) ==
-        (std::size(kRTCErrorTypeNames) - 1),
-    "kRTCErrorTypeNames must have as many strings as RTCErrorType "
-    "has values.");
+static_assert(static_cast<int>(RTCErrorType::OPERATION_ERROR_WITH_DATA) ==
+                  (std::size(kRTCErrorTypeNames) - 1),
+              "kRTCErrorTypeNames must have as many strings as RTCErrorType "
+              "has values.");
 
 absl::string_view kRTCErrorDetailTypeNames[] = {
     "NONE",
@@ -48,16 +49,13 @@ absl::string_view kRTCErrorDetailTypeNames[] = {
     "HARDWARE_ENCODER_NOT_AVAILABLE",
     "HARDWARE_ENCODER_ERROR",
 };
-static_assert(
-    static_cast<int>(webrtc::RTCErrorDetailType::HARDWARE_ENCODER_ERROR) ==
-        (std::size(kRTCErrorDetailTypeNames) - 1),
-    "kRTCErrorDetailTypeNames must have as many strings as "
-    "RTCErrorDetailType has values.");
+static_assert(static_cast<int>(RTCErrorDetailType::HARDWARE_ENCODER_ERROR) ==
+                  (std::size(kRTCErrorDetailTypeNames) - 1),
+              "kRTCErrorDetailTypeNames must have as many strings as "
+              "RTCErrorDetailType has values.");
 
 }  // namespace
 
-namespace webrtc {
-
 // static
 RTCError RTCError::OK() {
   return RTCError();
diff --git a/call/rtp_bitrate_configurator.cc b/call/rtp_bitrate_configurator.cc
index 3e7912b6bc..17f0ca6c83 100644
--- a/call/rtp_bitrate_configurator.cc
+++ b/call/rtp_bitrate_configurator.cc
@@ -17,6 +17,8 @@
 #include "api/units/data_rate.h"
 #include "rtc_base/checks.h"
 
+namespace webrtc {
+
 namespace {
 
 // Returns its smallest positive argument. If neither argument is positive,
@@ -33,7 +35,6 @@ int MinPositive(int a, int b) {
 
 }  // namespace
 
-namespace webrtc {
 RtpBitrateConfigurator::RtpBitrateConfigurator(
     const BitrateConstraints& bitrate_config)
     : bitrate_config_(bitrate_config), base_bitrate_config_(bitrate_config) {
diff --git a/common_audio/window_generator.cc b/common_audio/window_generator.cc
index 5a6de1aec1..30465879c1 100644
--- a/common_audio/window_generator.cc
+++ b/common_audio/window_generator.cc
@@ -17,13 +17,13 @@
 
 #include "rtc_base/checks.h"
 
-using std::complex;
+namespace webrtc {
 
 namespace {
 
 // Modified Bessel function of order 0 for complex inputs.
-complex<float> I0(complex<float> x) {
-  complex<float> y = x / 3.75f;
+std::complex<float> I0(std::complex<float> x) {
+  std::complex<float> y = x / 3.75f;
   y *= y;
   return 1.0f + y * (3.5156229f +
                      y * (3.0899424f +
@@ -34,8 +34,6 @@ complex<float> I0(complex<float> x) {
 
 }  // namespace
 
-namespace webrtc {
-
 void WindowGenerator::Hanning(int length, float* window) {
   RTC_CHECK_GT(length, 1);
   RTC_CHECK(window != nullptr);
@@ -55,7 +53,7 @@ void WindowGenerator::KaiserBesselDerived(float alpha,
   float sum = 0.0f;
 
   for (size_t i = 0; i <= half; ++i) {
-    complex<float> r = (4.0f * i) / length - 1.0f;
+    std::complex<float> r = (4.0f * i) / length - 1.0f;
     sum += I0(std::numbers::pi_v<float> * alpha * sqrt(1.0f - r * r)).real();
     window[i] = sum;
   }
diff --git a/common_video/h264/sps_parser.cc b/common_video/h264/sps_parser.cc
index cf99700a71..2ee431c7e5 100644
--- a/common_video/h264/sps_parser.cc
+++ b/common_video/h264/sps_parser.cc
@@ -19,13 +19,13 @@
 #include "common_video/h264/h264_common.h"
 #include "rtc_base/bitstream_reader.h"
 
+namespace webrtc {
+
 namespace {
 constexpr int kScalingDeltaMin = -128;
 constexpr int kScaldingDeltaMax = 127;
 }  // namespace
 
-namespace webrtc {
-
 SpsParser::SpsState::SpsState() = default;
 SpsParser::SpsState::SpsState(const SpsState&) = default;
 SpsParser::SpsState::~SpsState() = default;
diff --git a/common_video/h265/h265_bitstream_parser.cc b/common_video/h265/h265_bitstream_parser.cc
index 23fbbf2f46..b00080dae6 100644
--- a/common_video/h265/h265_bitstream_parser.cc
+++ b/common_video/h265/h265_bitstream_parser.cc
@@ -66,6 +66,8 @@
     }                                                                    \
   } while (0)
 
+namespace webrtc {
+
 namespace {
 
 constexpr int kMaxAbsQpDeltaValue = 51;
@@ -75,8 +77,6 @@ constexpr int kMaxRefIdxActive = 15;
 
 }  // namespace
 
-namespace webrtc {
-
 H265BitstreamParser::H265BitstreamParser() = default;
 H265BitstreamParser::~H265BitstreamParser() = default;
 
diff --git a/common_video/h265/h265_pps_parser.cc b/common_video/h265/h265_pps_parser.cc
index c94e362ec9..a5f7adb7f9 100644
--- a/common_video/h265/h265_pps_parser.cc
+++ b/common_video/h265/h265_pps_parser.cc
@@ -52,14 +52,14 @@
     }                                                                    \
   } while (0)
 
+namespace webrtc {
+
 namespace {
 constexpr int kMaxNumTileColumnWidth = 19;
 constexpr int kMaxNumTileRowHeight = 21;
 constexpr int kMaxRefIdxActive = 15;
 }  // namespace
 
-namespace webrtc {
-
 // General note: this is based off the 08/2021 version of the H.265 standard.
 // You can find it on this page:
 // http://www.itu.int/rec/T-REC-H.265
diff --git a/common_video/h265/h265_sps_parser.cc b/common_video/h265/h265_sps_parser.cc
index 1461b7400c..0916de5ad3 100644
--- a/common_video/h265/h265_sps_parser.cc
+++ b/common_video/h265/h265_sps_parser.cc
@@ -53,20 +53,19 @@
     }                                                                    \
   } while (0)
 
+namespace webrtc {
+
 namespace {
-using OptionalSps = std::optional<webrtc::H265SpsParser::SpsState>;
+using OptionalSps = std::optional<H265SpsParser::SpsState>;
 using OptionalShortTermRefPicSet =
-    std::optional<webrtc::H265SpsParser::ShortTermRefPicSet>;
-using OptionalProfileTierLevel =
-    std::optional<webrtc::H265SpsParser::ProfileTierLevel>;
+    std::optional<H265SpsParser::ShortTermRefPicSet>;
+using OptionalProfileTierLevel = std::optional<H265SpsParser::ProfileTierLevel>;
 
 constexpr int kMaxNumSizeIds = 4;
 constexpr int kMaxNumMatrixIds = 6;
 constexpr int kMaxNumCoefs = 64;
 }  // namespace
 
-namespace webrtc {
-
 H265SpsParser::ShortTermRefPicSet::ShortTermRefPicSet() = default;
 
 H265SpsParser::ProfileTierLevel::ProfileTierLevel() = default;
diff --git a/logging/rtc_event_log/events/rtc_event_field_encoding.cc b/logging/rtc_event_log/events/rtc_event_field_encoding.cc
index e43526a411..2fdfe220cb 100644
--- a/logging/rtc_event_log/events/rtc_event_field_encoding.cc
+++ b/logging/rtc_event_log/events/rtc_event_field_encoding.cc
@@ -26,6 +26,8 @@
 #include "logging/rtc_event_log/events/rtc_event_field_extraction.h"
 #include "rtc_base/checks.h"
 
+namespace webrtc {
+
 using webrtc_event_logging::UnsignedDelta;
 
 namespace {
@@ -60,8 +62,6 @@ std::string SerializeLittleEndian(uint64_t value, uint8_t bytes) {
 
 }  // namespace
 
-namespace webrtc {
-
 std::string EncodeOptionalValuePositions(std::vector<bool> positions) {
   BitWriter writer((positions.size() + 7) / 8);
   for (bool position : positions) {
diff --git a/logging/rtc_event_log/events/rtc_event_field_encoding_parser.cc b/logging/rtc_event_log/events/rtc_event_field_encoding_parser.cc
index 006f5f1e65..3e04b42ca4 100644
--- a/logging/rtc_event_log/events/rtc_event_field_encoding_parser.cc
+++ b/logging/rtc_event_log/events/rtc_event_field_encoding_parser.cc
@@ -1,4 +1,3 @@
-
 /*
  *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
  *
@@ -27,27 +26,27 @@
 #include "rtc_base/bitstream_reader.h"
 #include "rtc_base/checks.h"
 
+namespace webrtc {
+
 namespace {
-std::optional<webrtc::FieldType> ConvertFieldType(uint64_t value) {
+std::optional<FieldType> ConvertFieldType(uint64_t value) {
   switch (value) {
-    case static_cast<uint64_t>(webrtc::FieldType::kFixed8):
-      return webrtc::FieldType::kFixed8;
-    case static_cast<uint64_t>(webrtc::FieldType::kFixed32):
-      return webrtc::FieldType::kFixed32;
-    case static_cast<uint64_t>(webrtc::FieldType::kFixed64):
-      return webrtc::FieldType::kFixed64;
-    case static_cast<uint64_t>(webrtc::FieldType::kVarInt):
-      return webrtc::FieldType::kVarInt;
-    case static_cast<uint64_t>(webrtc::FieldType::kString):
-      return webrtc::FieldType::kString;
+    case static_cast<uint64_t>(FieldType::kFixed8):
+      return FieldType::kFixed8;
+    case static_cast<uint64_t>(FieldType::kFixed32):
+      return FieldType::kFixed32;
+    case static_cast<uint64_t>(FieldType::kFixed64):
+      return FieldType::kFixed64;
+    case static_cast<uint64_t>(FieldType::kVarInt):
+      return FieldType::kVarInt;
+    case static_cast<uint64_t>(FieldType::kString):
+      return FieldType::kString;
     default:
       return std::nullopt;
   }
 }
 }  // namespace
 
-namespace webrtc {
-
 uint64_t EventParser::ReadLittleEndian(uint8_t bytes) {
   RTC_DCHECK_LE(bytes, sizeof(uint64_t));
   RTC_DCHECK_GE(bytes, 1);
diff --git a/media/base/video_adapter.cc b/media/base/video_adapter.cc
index 7300282bd0..533bb95cd4 100644
--- a/media/base/video_adapter.cc
+++ b/media/base/video_adapter.cc
@@ -29,6 +29,8 @@
 #include "rtc_base/synchronization/mutex.h"
 #include "rtc_base/time_utils.h"
 
+namespace webrtc {
+
 namespace {
 
 struct Fraction {
@@ -136,8 +138,6 @@ std::optional<std::pair<int, int>> Swap(
 
 }  // namespace
 
-namespace webrtc {
-
 VideoAdapter::VideoAdapter(int source_resolution_alignment)
     : frames_in_(0),
       frames_out_(0),
diff --git a/media/engine/fake_video_codec_factory.cc b/media/engine/fake_video_codec_factory.cc
index 1c62353544..c9c114b4eb 100644
--- a/media/engine/fake_video_codec_factory.cc
+++ b/media/engine/fake_video_codec_factory.cc
@@ -22,13 +22,13 @@
 #include "test/fake_decoder.h"
 #include "test/fake_encoder.h"
 
+namespace webrtc {
+
 namespace {
 
 const char kFakeCodecFactoryCodecName[] = "FakeCodec";
 
-}  // anonymous namespace
-
-namespace webrtc {
+}  // namespace
 
 std::vector<SdpVideoFormat> FakeVideoEncoderFactory::GetSupportedFormats()
     const {
diff --git a/modules/audio_coding/codecs/opus/test/blocker.cc b/modules/audio_coding/codecs/opus/test/blocker.cc
index 321a3f9770..b12943fe52 100644
--- a/modules/audio_coding/codecs/opus/test/blocker.cc
+++ b/modules/audio_coding/codecs/opus/test/blocker.cc
@@ -14,6 +14,8 @@
 
 #include "rtc_base/checks.h"
 
+namespace webrtc {
+
 namespace {
 
 // Adds `a` and `b` frame by frame into `result` (basically matrix addition).
@@ -94,8 +96,6 @@ size_t gcd(size_t a, size_t b) {
 
 }  // namespace
 
-namespace webrtc {
-
 Blocker::Blocker(size_t chunk_size,
                  size_t block_size,
                  size_t num_input_channels,
diff --git a/modules/audio_coding/codecs/opus/test/blocker_unittest.cc b/modules/audio_coding/codecs/opus/test/blocker_unittest.cc
index 47c53fbcd0..3f03795e5b 100644
--- a/modules/audio_coding/codecs/opus/test/blocker_unittest.cc
+++ b/modules/audio_coding/codecs/opus/test/blocker_unittest.cc
@@ -18,10 +18,12 @@
 #include "common_audio/channel_buffer.h"
 #include "test/gtest.h"
 
+namespace webrtc {
+
 namespace {
 
 // Callback Function to add 3 to every sample in the signal.
-class PlusThreeBlockerCallback : public webrtc::BlockerCallback {
+class PlusThreeBlockerCallback : public BlockerCallback {
  public:
   void ProcessBlock(const float* const* input,
                     size_t num_frames,
@@ -37,7 +39,7 @@ class PlusThreeBlockerCallback : public webrtc::BlockerCallback {
 };
 
 // No-op Callback Function.
-class CopyBlockerCallback : public webrtc::BlockerCallback {
+class CopyBlockerCallback : public BlockerCallback {
  public:
   void ProcessBlock(const float* const* input,
                     size_t num_frames,
@@ -54,8 +56,6 @@ class CopyBlockerCallback : public webrtc::BlockerCallback {
 
 }  // namespace
 
-namespace webrtc {
-
 // Tests blocking with a window that multiplies the signal by 2, a callback
 // that adds 3 to each sample in the signal, and different combinations of chunk
 // size, block size, and shift amount.
diff --git a/modules/audio_coding/codecs/opus/test/lapped_transform_unittest.cc b/modules/audio_coding/codecs/opus/test/lapped_transform_unittest.cc
index 4ccba02475..3b5abedb61 100644
--- a/modules/audio_coding/codecs/opus/test/lapped_transform_unittest.cc
+++ b/modules/audio_coding/codecs/opus/test/lapped_transform_unittest.cc
@@ -18,11 +18,13 @@
 #include "rtc_base/checks.h"
 #include "test/gtest.h"
 
-using std::complex;
+namespace webrtc {
 
 namespace {
 
-class NoopCallback : public webrtc::LappedTransform::Callback {
+using std::complex;
+
+class NoopCallback : public LappedTransform::Callback {
  public:
   NoopCallback() : block_num_(0) {}
 
@@ -44,7 +46,7 @@ class NoopCallback : public webrtc::LappedTransform::Callback {
   size_t block_num_;
 };
 
-class FftCheckerCallback : public webrtc::LappedTransform::Callback {
+class FftCheckerCallback : public LappedTransform::Callback {
  public:
   FftCheckerCallback() : block_num_(0) {}
 
@@ -85,8 +87,6 @@ void SetFloatArray(float value, int rows, int cols, float* const* array) {
 
 }  // namespace
 
-namespace webrtc {
-
 TEST(LappedTransformTest, Windowless) {
   const size_t kChannels = 3;
   const size_t kChunkLength = 512;
diff --git a/modules/audio_coding/neteq/packet_buffer_unittest.cc b/modules/audio_coding/neteq/packet_buffer_unittest.cc
index c3869fb428..3e34602e74 100644
--- a/modules/audio_coding/neteq/packet_buffer_unittest.cc
+++ b/modules/audio_coding/neteq/packet_buffer_unittest.cc
@@ -30,6 +30,8 @@
 #include "test/gmock.h"
 #include "test/gtest.h"
 
+namespace webrtc {
+
 using ::testing::_;
 using ::testing::InSequence;
 using ::testing::MockFunction;
@@ -37,7 +39,7 @@ using ::testing::Return;
 using ::testing::StrictMock;
 
 namespace {
-class MockEncodedAudioFrame : public webrtc::AudioDecoder::EncodedAudioFrame {
+class MockEncodedAudioFrame : public AudioDecoder::EncodedAudioFrame {
  public:
   MOCK_METHOD(size_t, Duration, (), (const, override));
 
@@ -45,7 +47,7 @@ class MockEncodedAudioFrame : public webrtc::AudioDecoder::EncodedAudioFrame {
 
   MOCK_METHOD(std::optional<DecodeResult>,
               Decode,
-              (webrtc::ArrayView<int16_t> decoded),
+              (ArrayView<int16_t> decoded),
               (const, override));
 };
 
@@ -55,9 +57,9 @@ class PacketGenerator {
   PacketGenerator(uint16_t seq_no, uint32_t ts, uint8_t pt, int frame_size);
   virtual ~PacketGenerator() {}
   void Reset(uint16_t seq_no, uint32_t ts, uint8_t pt, int frame_size);
-  webrtc::Packet NextPacket(
+  Packet NextPacket(
       int payload_size_bytes,
-      std::unique_ptr<webrtc::AudioDecoder::EncodedAudioFrame> audio_frame);
+      std::unique_ptr<AudioDecoder::EncodedAudioFrame> audio_frame);
 
   uint16_t seq_no_;
   uint32_t ts_;
@@ -82,10 +84,10 @@ void PacketGenerator::Reset(uint16_t seq_no,
   frame_size_ = frame_size;
 }
 
-webrtc::Packet PacketGenerator::NextPacket(
+Packet PacketGenerator::NextPacket(
     int payload_size_bytes,
-    std::unique_ptr<webrtc::AudioDecoder::EncodedAudioFrame> audio_frame) {
-  webrtc::Packet packet;
+    std::unique_ptr<AudioDecoder::EncodedAudioFrame> audio_frame) {
+  Packet packet;
   packet.sequence_number = seq_no_;
   packet.timestamp = ts_;
   packet.payload_type = pt_;
@@ -109,8 +111,6 @@ struct PacketsToInsert {
 
 }  // namespace
 
-namespace webrtc {
-
 // Start of test definitions.
 
 TEST(PacketBuffer, CreateAndDestroy) {
diff --git a/modules/audio_coding/test/TestAllCodecs.cc b/modules/audio_coding/test/TestAllCodecs.cc
index 0914f0acfe..e45f1fddb9 100644
--- a/modules/audio_coding/test/TestAllCodecs.cc
+++ b/modules/audio_coding/test/TestAllCodecs.cc
@@ -48,12 +48,12 @@
     EXPECT_GE(f, 0) << "Error Calling API"; \
   } while (0)
 
+namespace webrtc {
+
 namespace {
 const size_t kVariableSize = std::numeric_limits<size_t>::max();
 }  // namespace
 
-namespace webrtc {
-
 // Class for simulating packet handling.
 TestPack::TestPack()
     : neteq_(nullptr),
diff --git a/modules/audio_processing/aec3/matched_filter.cc b/modules/audio_processing/aec3/matched_filter.cc
index 866f941697..5c86e27311 100644
--- a/modules/audio_processing/aec3/matched_filter.cc
+++ b/modules/audio_processing/aec3/matched_filter.cc
@@ -33,6 +33,8 @@
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
 
+namespace webrtc {
+
 namespace {
 
 // Subsample rate used for computing the accumulated error.
@@ -41,8 +43,8 @@ namespace {
 constexpr int kAccumulatedErrorSubSampleRate = 4;
 
 void UpdateAccumulatedError(
-    const webrtc::ArrayView<const float> instantaneous_accumulated_error,
-    const webrtc::ArrayView<float> accumulated_error,
+    const ArrayView<const float> instantaneous_accumulated_error,
+    const ArrayView<float> accumulated_error,
     float one_over_error_sum_anchor) {
   static constexpr float kSmoothConstantIncreases = 0.015f;
   for (size_t k = 0; k < instantaneous_accumulated_error.size(); ++k) {
@@ -57,7 +59,7 @@ void UpdateAccumulatedError(
   }
 }
 
-size_t ComputePreEchoLag(const webrtc::ArrayView<const float> accumulated_error,
+size_t ComputePreEchoLag(const ArrayView<const float> accumulated_error,
                          size_t lag,
                          size_t alignment_shift_winner) {
   static constexpr float kPreEchoThreshold = 0.5f;
@@ -77,7 +79,6 @@ size_t ComputePreEchoLag(const webrtc::ArrayView<const float> accumulated_error,
 
 }  // namespace
 
-namespace webrtc {
 namespace aec3 {
 
 #if defined(WEBRTC_HAS_NEON)
@@ -92,13 +93,13 @@ void MatchedFilterCoreWithAccumulatedError_NEON(
     size_t x_start_index,
     float x2_sum_threshold,
     float smoothing,
-    webrtc::ArrayView<const float> x,
-    webrtc::ArrayView<const float> y,
-    webrtc::ArrayView<float> h,
+    ArrayView<const float> x,
+    ArrayView<const float> y,
+    ArrayView<float> h,
     bool* filters_updated,
     float* error_sum,
-    webrtc::ArrayView<float> accumulated_error,
-    webrtc::ArrayView<float> scratch_memory) {
+    ArrayView<float> accumulated_error,
+    ArrayView<float> scratch_memory) {
   const int h_size = static_cast<int>(h.size());
   const int x_size = static_cast<int>(x.size());
   RTC_DCHECK_EQ(0, h_size % 4);
@@ -172,14 +173,14 @@ void MatchedFilterCoreWithAccumulatedError_NEON(
 void MatchedFilterCore_NEON(size_t x_start_index,
                             float x2_sum_threshold,
                             float smoothing,
-                            webrtc::ArrayView<const float> x,
-                            webrtc::ArrayView<const float> y,
-                            webrtc::ArrayView<float> h,
+                            ArrayView<const float> x,
+                            ArrayView<const float> y,
+                            ArrayView<float> h,
                             bool* filters_updated,
                             float* error_sum,
                             bool compute_accumulated_error,
-                            webrtc::ArrayView<float> accumulated_error,
-                            webrtc::ArrayView<float> scratch_memory) {
+                            ArrayView<float> accumulated_error,
+                            ArrayView<float> scratch_memory) {
   const int h_size = static_cast<int>(h.size());
   const int x_size = static_cast<int>(x.size());
   RTC_DCHECK_EQ(0, h_size % 4);
diff --git a/modules/audio_processing/residual_echo_detector.cc b/modules/audio_processing/residual_echo_detector.cc
index 437d857c22..67e2552768 100644
--- a/modules/audio_processing/residual_echo_detector.cc
+++ b/modules/audio_processing/residual_echo_detector.cc
@@ -23,9 +23,11 @@
 #include "rtc_base/logging.h"
 #include "system_wrappers/include/metrics.h"
 
+namespace webrtc {
+
 namespace {
 
-float Power(webrtc::ArrayView<const float> input) {
+float Power(ArrayView<const float> input) {
   if (input.empty()) {
     return 0.f;
   }
@@ -42,8 +44,6 @@ constexpr size_t kAggregationBufferSize = 10 * 100;
 
 }  // namespace
 
-namespace webrtc {
-
 std::atomic<int> ResidualEchoDetector::instance_count_(0);
 
 ResidualEchoDetector::ResidualEchoDetector()
diff --git a/modules/audio_processing/test/protobuf_utils.cc b/modules/audio_processing/test/protobuf_utils.cc
index a9195f299f..fb0c08209a 100644
--- a/modules/audio_processing/test/protobuf_utils.cc
+++ b/modules/audio_processing/test/protobuf_utils.cc
@@ -19,6 +19,8 @@
 #include "rtc_base/protobuf_utils.h"
 #include "rtc_base/system/arch.h"
 
+namespace webrtc {
+
 namespace {
 // Allocates new memory in the memory owned by the unique_ptr to fit the raw
 // message and returns the number of bytes read when having a string stream as
@@ -41,8 +43,6 @@ size_t ReadMessageBytesFromString(std::stringstream* input,
 }
 }  // namespace
 
-namespace webrtc {
-
 size_t ReadMessageBytesFromFile(FILE* file, std::unique_ptr<uint8_t[]>* bytes) {
 // The "wire format" for the size is little-endian. Assume we're running on
 // a little-endian machine.
diff --git a/modules/desktop_capture/linux/x11/mouse_cursor_monitor_x11.cc b/modules/desktop_capture/linux/x11/mouse_cursor_monitor_x11.cc
index 732aa4d29d..72f2716e03 100644
--- a/modules/desktop_capture/linux/x11/mouse_cursor_monitor_x11.cc
+++ b/modules/desktop_capture/linux/x11/mouse_cursor_monitor_x11.cc
@@ -30,6 +30,8 @@
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
 
+namespace webrtc {
+
 namespace {
 
 // WindowCapturer returns window IDs of X11 windows with WM_STATE attribute.
@@ -64,8 +66,6 @@ Window GetTopLevelWindow(Display* display, Window window) {
 
 }  // namespace
 
-namespace webrtc {
-
 MouseCursorMonitorX11::MouseCursorMonitorX11(
     const DesktopCaptureOptions& options,
     Window window)
diff --git a/modules/rtp_rtcp/source/dtmf_queue.cc b/modules/rtp_rtcp/source/dtmf_queue.cc
index 3f054c7177..53f429b5dd 100644
--- a/modules/rtp_rtcp/source/dtmf_queue.cc
+++ b/modules/rtp_rtcp/source/dtmf_queue.cc
@@ -15,11 +15,12 @@
 #include "rtc_base/checks.h"
 #include "rtc_base/synchronization/mutex.h"
 
+namespace webrtc {
+
 namespace {
 constexpr size_t kDtmfOutbandMax = 20;
 }  // namespace
 
-namespace webrtc {
 DtmfQueue::DtmfQueue() {}
 
 DtmfQueue::~DtmfQueue() {}
diff --git a/modules/rtp_rtcp/source/fec_private_tables_bursty.cc b/modules/rtp_rtcp/source/fec_private_tables_bursty.cc
index d0b9da4a3f..84a4857e7b 100644
--- a/modules/rtp_rtcp/source/fec_private_tables_bursty.cc
+++ b/modules/rtp_rtcp/source/fec_private_tables_bursty.cc
@@ -12,6 +12,8 @@
 
 #include <cstdint>
 
+namespace webrtc {
+
 namespace {
 // clang-format off
 #define kMaskBursty1_1 \
@@ -639,7 +641,6 @@ namespace {
 // clang-format on
 }  // namespace
 
-namespace webrtc {
 namespace fec_private_tables {
 
 const uint8_t kPacketMaskBurstyTbl[] = {
diff --git a/modules/rtp_rtcp/source/fec_private_tables_random.cc b/modules/rtp_rtcp/source/fec_private_tables_random.cc
index 885aeb5ef2..b5cd8bccbd 100644
--- a/modules/rtp_rtcp/source/fec_private_tables_random.cc
+++ b/modules/rtp_rtcp/source/fec_private_tables_random.cc
@@ -12,6 +12,8 @@
 
 #include <cstdint>
 
+namespace webrtc {
+
 namespace {
 // clang-format off
 #define kMaskRandom1_1 \
@@ -639,7 +641,6 @@ namespace {
 // clang-format on
 }  // namespace
 
-namespace webrtc {
 namespace fec_private_tables {
 
 const uint8_t kPacketMaskRandomTbl[] = {
diff --git a/modules/rtp_rtcp/source/forward_error_correction_internal.cc b/modules/rtp_rtcp/source/forward_error_correction_internal.cc
index bc749b6297..46457794e8 100644
--- a/modules/rtp_rtcp/source/forward_error_correction_internal.cc
+++ b/modules/rtp_rtcp/source/forward_error_correction_internal.cc
@@ -20,6 +20,8 @@
 #include "modules/rtp_rtcp/source/fec_private_tables_random.h"
 #include "rtc_base/checks.h"
 
+namespace webrtc {
+
 namespace {
 // Allow for different modes of protection for packets in UEP case.
 enum ProtectionMode {
@@ -141,7 +143,6 @@ void ShiftFitSubMask(int num_mask_bytes,
 
 }  // namespace
 
-namespace webrtc {
 namespace internal {
 
 PacketMaskTable::PacketMaskTable(FecMaskType fec_mask_type,
diff --git a/p2p/base/default_ice_transport_factory.cc b/p2p/base/default_ice_transport_factory.cc
index 6c314fd827..5c2fb88339 100644
--- a/p2p/base/default_ice_transport_factory.cc
+++ b/p2p/base/default_ice_transport_factory.cc
@@ -23,20 +23,20 @@
 #include "p2p/base/ice_controller_interface.h"
 #include "p2p/base/p2p_transport_channel.h"
 
+namespace webrtc {
+
 namespace {
 
-class BasicIceControllerFactory : public webrtc::IceControllerFactoryInterface {
+class BasicIceControllerFactory : public IceControllerFactoryInterface {
  public:
-  std::unique_ptr<webrtc::IceControllerInterface> Create(
-      const webrtc::IceControllerFactoryArgs& args) override {
-    return std::make_unique<webrtc::BasicIceController>(args);
+  std::unique_ptr<IceControllerInterface> Create(
+      const IceControllerFactoryArgs& args) override {
+    return std::make_unique<BasicIceController>(args);
   }
 };
 
 }  // namespace
 
-namespace webrtc {
-
 DefaultIceTransport::DefaultIceTransport(
     std::unique_ptr<P2PTransportChannel> internal)
     : internal_(std::move(internal)) {}
diff --git a/p2p/base/stun_dictionary_unittest.cc b/p2p/base/stun_dictionary_unittest.cc
index 0e292f85ea..506443bee9 100644
--- a/p2p/base/stun_dictionary_unittest.cc
+++ b/p2p/base/stun_dictionary_unittest.cc
@@ -22,10 +22,11 @@
 #include "rtc_base/socket_address.h"
 #include "test/gtest.h"
 
+namespace webrtc {
+
 namespace {
 
-void Sync(webrtc::StunDictionaryView& dictionary,
-          webrtc::StunDictionaryWriter& writer) {
+void Sync(StunDictionaryView& dictionary, StunDictionaryWriter& writer) {
   int pending = writer.Pending();
   auto delta = writer.CreateDelta();
   if (delta == nullptr) {
@@ -43,18 +44,17 @@ void Sync(webrtc::StunDictionaryView& dictionary,
   }
 }
 
-void XorToggle(webrtc::StunByteStringAttribute& attr, size_t byte) {
+void XorToggle(StunByteStringAttribute& attr, size_t byte) {
   ASSERT_TRUE(attr.length() > byte);
   uint8_t val = attr.GetByte(byte);
   uint8_t new_val = val ^ (128 - (byte & 255));
   attr.SetByte(byte, new_val);
 }
 
-std::unique_ptr<webrtc::StunByteStringAttribute> Crop(
-    const webrtc::StunByteStringAttribute& attr,
+std::unique_ptr<StunByteStringAttribute> Crop(
+    const StunByteStringAttribute& attr,
     int new_length) {
-  auto new_attr =
-      std::make_unique<webrtc::StunByteStringAttribute>(attr.type());
+  auto new_attr = std::make_unique<StunByteStringAttribute>(attr.type());
   std::string content = std::string(attr.string_view());
   content.erase(new_length);
   new_attr->CopyBytes(content);
@@ -63,8 +63,6 @@ std::unique_ptr<webrtc::StunByteStringAttribute> Crop(
 
 }  // namespace
 
-namespace webrtc {
-
 constexpr int kKey1 = 100;
 
 TEST(StunDictionary, CreateEmptyDictionaryWriter) {
diff --git a/p2p/base/turn_port_unittest.cc b/p2p/base/turn_port_unittest.cc
index 9a7e2f1415..99efabc637 100644
--- a/p2p/base/turn_port_unittest.cc
+++ b/p2p/base/turn_port_unittest.cc
@@ -70,6 +70,8 @@
 #include <dirent.h>  // IWYU pragma: keep
 #endif
 
+namespace webrtc {
+
 namespace {
 using ::testing::_;
 using ::testing::DoAll;
@@ -79,20 +81,15 @@ using ::testing::Ne;
 using ::testing::Return;
 using ::testing::ReturnPointee;
 using ::testing::SetArgPointee;
-using ::webrtc::CreateEnvironment;
-using ::webrtc::Environment;
-using ::webrtc::IceCandidateType;
-using ::webrtc::SocketAddress;
 
 const SocketAddress kLocalAddr1("11.11.11.11", 0);
 const SocketAddress kLocalAddr2("22.22.22.22", 0);
 const SocketAddress kLocalIPv6Addr("2401:fa00:4:1000:be30:5bff:fee5:c3", 0);
 const SocketAddress kLocalIPv6Addr2("2401:fa00:4:2000:be30:5bff:fee5:d4", 0);
-const SocketAddress kTurnUdpIntAddr("99.99.99.3", webrtc::TURN_SERVER_PORT);
-const SocketAddress kTurnTcpIntAddr("99.99.99.4", webrtc::TURN_SERVER_PORT);
+const SocketAddress kTurnUdpIntAddr("99.99.99.3", TURN_SERVER_PORT);
+const SocketAddress kTurnTcpIntAddr("99.99.99.4", TURN_SERVER_PORT);
 const SocketAddress kTurnUdpExtAddr("99.99.99.5", 0);
-const SocketAddress kTurnAlternateIntAddr("99.99.99.6",
-                                          webrtc::TURN_SERVER_PORT);
+const SocketAddress kTurnAlternateIntAddr("99.99.99.6", TURN_SERVER_PORT);
 // Port for redirecting to a TCP Web server. Should not work.
 const SocketAddress kTurnDangerousAddr("99.99.99.7", 81);
 // Port 53 (the DNS port); should work.
@@ -102,11 +99,11 @@ const SocketAddress kTurnPort80Addr("99.99.99.7", 80);
 // Port 443 (the HTTPS port); should work.
 const SocketAddress kTurnPort443Addr("99.99.99.7", 443);
 // The default TURN server port.
-const SocketAddress kTurnIntAddr("99.99.99.7", webrtc::TURN_SERVER_PORT);
+const SocketAddress kTurnIntAddr("99.99.99.7", TURN_SERVER_PORT);
 const SocketAddress kTurnIPv6IntAddr("2400:4030:2:2c00:be30:abcd:efab:cdef",
-                                     webrtc::TURN_SERVER_PORT);
+                                     TURN_SERVER_PORT);
 const SocketAddress kTurnUdpIPv6IntAddr("2400:4030:1:2c00:be30:abcd:efab:cdef",
-                                        webrtc::TURN_SERVER_PORT);
+                                        TURN_SERVER_PORT);
 const SocketAddress kTurnInvalidAddr("www.google.invalid.", 3478);
 const SocketAddress kTurnValidAddr("www.google.valid.", 3478);
 
@@ -129,28 +126,18 @@ constexpr unsigned int kResolverTimeout = 10000;
 
 constexpr uint64_t kTiebreakerDefault = 44444;
 
-const webrtc::ProtocolAddress kTurnUdpProtoAddr(kTurnUdpIntAddr,
-                                                webrtc::PROTO_UDP);
-const webrtc::ProtocolAddress kTurnTcpProtoAddr(kTurnTcpIntAddr,
-                                                webrtc::PROTO_TCP);
-const webrtc::ProtocolAddress kTurnTlsProtoAddr(kTurnTcpIntAddr,
-                                                webrtc::PROTO_TLS);
-const webrtc::ProtocolAddress kTurnUdpIPv6ProtoAddr(kTurnUdpIPv6IntAddr,
-                                                    webrtc::PROTO_UDP);
-const webrtc::ProtocolAddress kTurnDangerousProtoAddr(kTurnDangerousAddr,
-                                                      webrtc::PROTO_TCP);
-const webrtc::ProtocolAddress kTurnPort53ProtoAddr(kTurnPort53Addr,
-                                                   webrtc::PROTO_TCP);
-const webrtc::ProtocolAddress kTurnPort80ProtoAddr(kTurnPort80Addr,
-                                                   webrtc::PROTO_TCP);
-const webrtc::ProtocolAddress kTurnPort443ProtoAddr(kTurnPort443Addr,
-                                                    webrtc::PROTO_TCP);
-const webrtc::ProtocolAddress kTurnPortInvalidHostnameProtoAddr(
-    kTurnInvalidAddr,
-    webrtc::PROTO_UDP);
-const webrtc::ProtocolAddress kTurnPortValidHostnameProtoAddr(
-    kTurnValidAddr,
-    webrtc::PROTO_UDP);
+const ProtocolAddress kTurnUdpProtoAddr(kTurnUdpIntAddr, PROTO_UDP);
+const ProtocolAddress kTurnTcpProtoAddr(kTurnTcpIntAddr, PROTO_TCP);
+const ProtocolAddress kTurnTlsProtoAddr(kTurnTcpIntAddr, PROTO_TLS);
+const ProtocolAddress kTurnUdpIPv6ProtoAddr(kTurnUdpIPv6IntAddr, PROTO_UDP);
+const ProtocolAddress kTurnDangerousProtoAddr(kTurnDangerousAddr, PROTO_TCP);
+const ProtocolAddress kTurnPort53ProtoAddr(kTurnPort53Addr, PROTO_TCP);
+const ProtocolAddress kTurnPort80ProtoAddr(kTurnPort80Addr, PROTO_TCP);
+const ProtocolAddress kTurnPort443ProtoAddr(kTurnPort443Addr, PROTO_TCP);
+const ProtocolAddress kTurnPortInvalidHostnameProtoAddr(kTurnInvalidAddr,
+                                                        PROTO_UDP);
+const ProtocolAddress kTurnPortValidHostnameProtoAddr(kTurnValidAddr,
+                                                      PROTO_UDP);
 
 #if defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID)
 int GetFDCount() {
@@ -169,7 +156,6 @@ int GetFDCount() {
 
 }  // unnamed namespace
 
-namespace webrtc {
 
 class TurnPortTestVirtualSocketServer : public VirtualSocketServer {
  public:
diff --git a/p2p/base/wrapping_active_ice_controller.cc b/p2p/base/wrapping_active_ice_controller.cc
index 05d9a1f706..a09d057afa 100644
--- a/p2p/base/wrapping_active_ice_controller.cc
+++ b/p2p/base/wrapping_active_ice_controller.cc
@@ -30,11 +30,6 @@
 #include "rtc_base/logging.h"
 #include "rtc_base/thread.h"
 
-namespace {
-using ::webrtc::SafeTask;
-using ::webrtc::TimeDelta;
-}  // unnamed namespace
-
 namespace webrtc {
 
 WrappingActiveIceController::WrappingActiveIceController(
diff --git a/p2p/client/basic_port_allocator_unittest.cc b/p2p/client/basic_port_allocator_unittest.cc
index fba34695a0..41a1d60002 100644
--- a/p2p/client/basic_port_allocator_unittest.cc
+++ b/p2p/client/basic_port_allocator_unittest.cc
@@ -132,26 +132,26 @@ static const char kTurnPassword[] = "test";
 // Add some margin of error for slow bots.
 static const int kStunTimeoutMs = webrtc::STUN_TOTAL_TIMEOUT;
 
+namespace webrtc {
+
 namespace {
 
 void CheckStunKeepaliveIntervalOfAllReadyPorts(
-    const webrtc::PortAllocatorSession* allocator_session,
+    const PortAllocatorSession* allocator_session,
     int expected) {
   auto ready_ports = allocator_session->ReadyPorts();
   for (const auto* port : ready_ports) {
     if (port->Type() == IceCandidateType::kSrflx ||
         (port->Type() == IceCandidateType::kHost &&
-         port->GetProtocol() == webrtc::PROTO_UDP)) {
-      EXPECT_EQ(
-          static_cast<const webrtc::UDPPort*>(port)->stun_keepalive_delay(),
-          webrtc::TimeDelta::Millis(expected));
+         port->GetProtocol() == PROTO_UDP)) {
+      EXPECT_EQ(static_cast<const UDPPort*>(port)->stun_keepalive_delay(),
+                TimeDelta::Millis(expected));
     }
   }
 }
 
 }  // namespace
 
-namespace webrtc {
 
 class BasicPortAllocatorTestBase : public ::testing::Test {
  public:
diff --git a/p2p/dtls/dtls_stun_piggyback_controller_unittest.cc b/p2p/dtls/dtls_stun_piggyback_controller_unittest.cc
index 05887a8bfe..59e5299f8f 100644
--- a/p2p/dtls/dtls_stun_piggyback_controller_unittest.cc
+++ b/p2p/dtls/dtls_stun_piggyback_controller_unittest.cc
@@ -27,6 +27,8 @@
 #include "test/gmock.h"
 #include "test/gtest.h"
 
+namespace webrtc {
+
 namespace {
 // Extracted from a stock DTLS call using Wireshark.
 // Each packet (apart from the last) is truncated to
@@ -61,8 +63,8 @@ const std::vector<uint8_t> empty = {};
 
 const std::vector<uint8_t> kPayload = {0x1, 0x2, 0x3};
 
-std::vector<uint32_t> FromAckAttribute(webrtc::ArrayView<uint8_t> attr) {
-  webrtc::ByteBufferReader ack_reader(attr);
+std::vector<uint32_t> FromAckAttribute(ArrayView<uint8_t> attr) {
+  ByteBufferReader ack_reader(attr);
   std::vector<uint32_t> values;
   uint32_t value;
   while (ack_reader.ReadUInt32(&value)) {
@@ -79,29 +81,26 @@ std::vector<uint8_t> FakeDtlsPacket(uint16_t packet_number) {
   return packet;
 }
 
-std::unique_ptr<webrtc::StunByteStringAttribute> WrapInStun(
-    webrtc::IceAttributeType type,
-    absl::string_view data) {
-  return std::make_unique<webrtc::StunByteStringAttribute>(type, data);
+std::unique_ptr<StunByteStringAttribute> WrapInStun(IceAttributeType type,
+                                                    absl::string_view data) {
+  return std::make_unique<StunByteStringAttribute>(type, data);
 }
 
-std::unique_ptr<webrtc::StunByteStringAttribute> WrapInStun(
-    webrtc::IceAttributeType type,
+std::unique_ptr<StunByteStringAttribute> WrapInStun(
+    IceAttributeType type,
     const std::vector<uint8_t>& data) {
-  return std::make_unique<webrtc::StunByteStringAttribute>(type, data.data(),
-                                                           data.size());
+  return std::make_unique<StunByteStringAttribute>(type, data.data(),
+                                                   data.size());
 }
 
-std::unique_ptr<webrtc::StunByteStringAttribute> WrapInStun(
-    webrtc::IceAttributeType type,
+std::unique_ptr<StunByteStringAttribute> WrapInStun(
+    IceAttributeType type,
     const std::vector<uint32_t>& data) {
-  return std::make_unique<webrtc::StunByteStringAttribute>(type, data);
+  return std::make_unique<StunByteStringAttribute>(type, data);
 }
 
 }  // namespace
 
-namespace webrtc {
-
 using ::testing::ElementsAreArray;
 using ::testing::MockFunction;
 using ::testing::NotNull;
diff --git a/p2p/dtls/dtls_utils.cc b/p2p/dtls/dtls_utils.cc
index 8744d6ef1e..a348fa1ad6 100644
--- a/p2p/dtls/dtls_utils.cc
+++ b/p2p/dtls/dtls_utils.cc
@@ -21,6 +21,8 @@
 #include "rtc_base/checks.h"
 #include "rtc_base/crc32.h"
 
+namespace webrtc {
+
 namespace {
 // https://datatracker.ietf.org/doc/html/rfc5246#appendix-A.1
 const uint8_t kDtlsChangeCipherSpecRecord = 20;
@@ -28,8 +30,6 @@ const uint8_t kDtlsHandshakeRecord = 22;
 
 }  // namespace
 
-namespace webrtc {
-
 bool IsDtlsPacket(ArrayView<const uint8_t> payload) {
   const uint8_t* u = payload.data();
   return (payload.size() >= kDtlsRecordHeaderLen && (u[0] > 19 && u[0] < 64));
diff --git a/pc/jitter_buffer_delay.cc b/pc/jitter_buffer_delay.cc
index 7e8c3a5593..60c38256c3 100644
--- a/pc/jitter_buffer_delay.cc
+++ b/pc/jitter_buffer_delay.cc
@@ -16,13 +16,13 @@
 #include "rtc_base/numerics/safe_conversions.h"
 #include "rtc_base/numerics/safe_minmax.h"
 
+namespace webrtc {
+
 namespace {
 constexpr int kDefaultDelay = 0;
 constexpr int kMaximumDelayMs = 10000;
 }  // namespace
 
-namespace webrtc {
-
 void JitterBufferDelay::Set(std::optional<double> delay_seconds) {
   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   cached_delay_seconds_ = delay_seconds;
diff --git a/pc/media_session.cc b/pc/media_session.cc
index 3b34e6b728..0ff4bef4c1 100644
--- a/pc/media_session.cc
+++ b/pc/media_session.cc
@@ -52,51 +52,47 @@
 #include "rtc_base/thread.h"
 #include "rtc_base/unique_id_generator.h"
 
+namespace webrtc {
+
 namespace {
 
-using webrtc::RTCError;
-using webrtc::RTCErrorType;
-using webrtc::RtpTransceiverDirection;
-using webrtc::UniqueRandomIdGenerator;
-
-webrtc::RtpExtension RtpExtensionFromCapability(
-    const webrtc::RtpHeaderExtensionCapability& capability) {
-  return webrtc::RtpExtension(capability.uri,
-                              capability.preferred_id.value_or(1),
-                              capability.preferred_encrypt);
+RtpExtension RtpExtensionFromCapability(
+    const RtpHeaderExtensionCapability& capability) {
+  return RtpExtension(capability.uri, capability.preferred_id.value_or(1),
+                      capability.preferred_encrypt);
 }
 
-webrtc::RtpHeaderExtensions RtpHeaderExtensionsFromCapabilities(
-    const std::vector<webrtc::RtpHeaderExtensionCapability>& capabilities) {
-  webrtc::RtpHeaderExtensions exts;
+RtpHeaderExtensions RtpHeaderExtensionsFromCapabilities(
+    const std::vector<RtpHeaderExtensionCapability>& capabilities) {
+  RtpHeaderExtensions exts;
   for (const auto& capability : capabilities) {
     exts.push_back(RtpExtensionFromCapability(capability));
   }
   return exts;
 }
 
-std::vector<webrtc::RtpHeaderExtensionCapability>
+std::vector<RtpHeaderExtensionCapability>
 UnstoppedRtpHeaderExtensionCapabilities(
-    std::vector<webrtc::RtpHeaderExtensionCapability> capabilities) {
+    std::vector<RtpHeaderExtensionCapability> capabilities) {
   std::erase_if(
-      capabilities, [](const webrtc::RtpHeaderExtensionCapability& capability) {
+      capabilities, [](const RtpHeaderExtensionCapability& capability) {
         return capability.direction == RtpTransceiverDirection::kStopped;
       });
   return capabilities;
 }
 
-bool IsCapabilityPresent(const webrtc::RtpHeaderExtensionCapability& capability,
-                         const webrtc::RtpHeaderExtensions& extensions) {
+bool IsCapabilityPresent(const RtpHeaderExtensionCapability& capability,
+                         const RtpHeaderExtensions& extensions) {
   return std::find_if(extensions.begin(), extensions.end(),
-                      [&capability](const webrtc::RtpExtension& extension) {
+                      [&capability](const RtpExtension& extension) {
                         return capability.uri == extension.uri;
                       }) != extensions.end();
 }
 
-webrtc::RtpHeaderExtensions UnstoppedOrPresentRtpHeaderExtensions(
-    const std::vector<webrtc::RtpHeaderExtensionCapability>& capabilities,
-    const webrtc::RtpHeaderExtensions& all_encountered_extensions) {
-  webrtc::RtpHeaderExtensions extensions;
+RtpHeaderExtensions UnstoppedOrPresentRtpHeaderExtensions(
+    const std::vector<RtpHeaderExtensionCapability>& capabilities,
+    const RtpHeaderExtensions& all_encountered_extensions) {
+  RtpHeaderExtensions extensions;
   for (const auto& capability : capabilities) {
     if (capability.direction != RtpTransceiverDirection::kStopped ||
         IsCapabilityPresent(capability, all_encountered_extensions)) {
@@ -106,12 +102,6 @@ webrtc::RtpHeaderExtensions UnstoppedOrPresentRtpHeaderExtensions(
   return extensions;
 }
 
-}  // namespace
-
-namespace webrtc {
-
-namespace {
-
 bool ContainsRtxCodec(const std::vector<Codec>& codecs) {
   return absl::c_find_if(codecs, [](const Codec& c) {
            return c.GetResiliencyType() == Codec::ResiliencyType::kRtx;
@@ -708,7 +698,7 @@ MediaSessionDescriptionFactory::filtered_rtp_header_extensions(
     RtpHeaderExtensions extensions) const {
   if (!is_unified_plan_) {
     // Remove extensions only supported with unified-plan.
-    std::erase_if(extensions, [](const webrtc::RtpExtension& extension) {
+    std::erase_if(extensions, [](const RtpExtension& extension) {
       return extension.uri == RtpExtension::kMidUri ||
              extension.uri == RtpExtension::kRidUri ||
              extension.uri == RtpExtension::kRepairedRidUri;
diff --git a/pc/media_session_unittest.cc b/pc/media_session_unittest.cc
index 3c97818fe1..858f77947e 100644
--- a/pc/media_session_unittest.cc
+++ b/pc/media_session_unittest.cc
@@ -87,7 +87,6 @@ using ::testing::SizeIs;
 using ::testing::UnorderedElementsAreArray;
 using ::testing::Values;
 using ::testing::ValuesIn;
-using ::webrtc::UniqueRandomIdGenerator;
 
 using Candidates = std::vector<Candidate>;
 
diff --git a/pc/rtp_sender_receiver_unittest.cc b/pc/rtp_sender_receiver_unittest.cc
index f900787d8c..6598701cad 100644
--- a/pc/rtp_sender_receiver_unittest.cc
+++ b/pc/rtp_sender_receiver_unittest.cc
@@ -69,6 +69,8 @@
 #include "test/run_loop.h"
 #include "test/wait_until.h"
 
+namespace webrtc {
+
 namespace {
 
 constexpr char kStreamId1[] = "local_stream_1";
@@ -81,15 +83,13 @@ constexpr uint32_t kAudioSsrc2 = 101;
 constexpr uint32_t kVideoSsrcSimulcast = 102;
 constexpr uint32_t kVideoSimulcastLayerCount = 2;
 
-class MockSetStreamsObserver
-    : public webrtc::RtpSenderBase::SetStreamsObserver {
+class MockSetStreamsObserver : public RtpSenderBase::SetStreamsObserver {
  public:
   MOCK_METHOD(void, OnSetStreams, (), (override));
 };
 
 }  // namespace
 
-namespace webrtc {
 
 using ::testing::ContainerEq;
 using RidList = std::vector<std::string>;
diff --git a/rtc_base/bit_buffer.cc b/rtc_base/bit_buffer.cc
index 0d8c75a84c..1b862e72db 100644
--- a/rtc_base/bit_buffer.cc
+++ b/rtc_base/bit_buffer.cc
@@ -1,5 +1,5 @@
 /*
- *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *  Copyright 2015 The WebRTC Project Authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
@@ -21,6 +21,8 @@
 #include "api/array_view.h"
 #include "rtc_base/checks.h"
 
+namespace webrtc {
+
 namespace {
 
 // Returns the highest byte of `val` in a uint8_t.
@@ -52,8 +54,6 @@ uint8_t WritePartialByte(uint8_t source,
 
 }  // namespace
 
-namespace webrtc {
-
 BitBufferWriter::BitBufferWriter(ArrayView<uint8_t> bytes)
     : BitBufferWriter(bytes.data(), bytes.size()) {}
 
diff --git a/rtc_base/checks.cc b/rtc_base/checks.cc
index a2e4079f23..d1e00947d9 100644
--- a/rtc_base/checks.cc
+++ b/rtc_base/checks.cc
@@ -37,6 +37,8 @@
 
 #include "rtc_base/checks.h"
 
+namespace webrtc {
+
 namespace {
 
 #if defined(__GNUC__)
@@ -59,7 +61,6 @@ void AppendFormat(std::string* s, const char* fmt, ...) {
 }
 }  // namespace
 
-namespace webrtc {
 namespace webrtc_checks_impl {
 
 #if !defined(WEBRTC_CHROMIUM_BUILD)
diff --git a/rtc_base/cpu_info.cc b/rtc_base/cpu_info.cc
index 32e2eb35b2..f79edd3bea 100644
--- a/rtc_base/cpu_info.cc
+++ b/rtc_base/cpu_info.cc
@@ -43,9 +43,11 @@
 
 // Parts of this file derived from Chromium's base/cpu.cc.
 
+namespace webrtc {
+
 namespace {
 
-uint32_t DetectNumberOfCores() {
+uint32_t DetectNumberOfCoresHelper() {
   int number_of_cores = 0;
 
 #if defined(WEBRTC_WIN)
@@ -121,8 +123,6 @@ inline void __cpuid(int cpu_info[4], int info_type) {
 
 }  // namespace
 
-namespace webrtc {
-
 namespace cpu_info {
 
 uint32_t DetectNumberOfCores() {
@@ -130,7 +130,7 @@ uint32_t DetectNumberOfCores() {
   // is running in a sandbox, we may only be able to read the value once (before
   // the sandbox is initialized) and not thereafter.
   // For more information see crbug.com/176522.
-  static const uint32_t logical_cpus = ::DetectNumberOfCores();
+  static const uint32_t logical_cpus = DetectNumberOfCoresHelper();
   return logical_cpus;
 }
 
diff --git a/rtc_base/cpu_time.cc b/rtc_base/cpu_time.cc
index 2da8c5e570..0efadd31d6 100644
--- a/rtc_base/cpu_time.cc
+++ b/rtc_base/cpu_time.cc
@@ -36,6 +36,8 @@
 #include <zircon/status.h>
 #endif
 
+namespace webrtc {
+
 #if defined(WEBRTC_WIN)
 namespace {
 // FILETIME resolution is 100 nanosecs.
@@ -43,8 +45,6 @@ const int64_t kNanosecsPerFiletime = 100;
 }  // namespace
 #endif
 
-namespace webrtc {
-
 int64_t GetProcessCpuTimeNanos() {
 #if defined(WEBRTC_FUCHSIA)
   zx_info_task_runtime_t runtime_info;
diff --git a/rtc_base/cpu_time_unittest.cc b/rtc_base/cpu_time_unittest.cc
index 8bbc48c5b2..d42abb3122 100644
--- a/rtc_base/cpu_time_unittest.cc
+++ b/rtc_base/cpu_time_unittest.cc
@@ -26,6 +26,8 @@
 #define MAYBE_TEST(test_name) test_name
 #endif
 
+namespace webrtc {
+
 namespace {
 constexpr int kAllowedErrorMillisecs = 30;
 constexpr int kProcessingTimeMillisecs = 500;
@@ -34,17 +36,14 @@ constexpr int kWorkingThreads = 2;
 // Consumes approximately kProcessingTimeMillisecs of CPU time in single thread.
 void WorkingFunction(int64_t* counter) {
   *counter = 0;
-  int64_t stop_cpu_time =
-      webrtc::GetThreadCpuTimeNanos() +
-      kProcessingTimeMillisecs * webrtc::kNumNanosecsPerMillisec;
-  while (webrtc::GetThreadCpuTimeNanos() < stop_cpu_time) {
+  int64_t stop_cpu_time = GetThreadCpuTimeNanos() +
+                          kProcessingTimeMillisecs * kNumNanosecsPerMillisec;
+  while (GetThreadCpuTimeNanos() < stop_cpu_time) {
     (*counter)++;
   }
 }
 }  // namespace
 
-namespace webrtc {
-
 // A minimal test which can be run on instrumented builds, so that they're at
 // least exercising the code to check for memory leaks/etc.
 TEST(CpuTimeTest, BasicTest) {
diff --git a/rtc_base/event_tracer_unittest.cc b/rtc_base/event_tracer_unittest.cc
index eae19a2588..dce51d89cb 100644
--- a/rtc_base/event_tracer_unittest.cc
+++ b/rtc_base/event_tracer_unittest.cc
@@ -15,22 +15,24 @@
 #include "rtc_base/trace_event.h"
 #include "test/gtest.h"
 
+namespace webrtc {
+
 namespace {
 
 class TestStatistics {
  public:
   void Reset() {
-    webrtc::MutexLock lock(&mutex_);
+    MutexLock lock(&mutex_);
     events_logged_ = 0;
   }
 
   void Increment() {
-    webrtc::MutexLock lock(&mutex_);
+    MutexLock lock(&mutex_);
     ++events_logged_;
   }
 
   int Count() const {
-    webrtc::MutexLock lock(&mutex_);
+    MutexLock lock(&mutex_);
     return events_logged_;
   }
 
@@ -41,14 +43,12 @@ class TestStatistics {
   }
 
  private:
-  mutable webrtc::Mutex mutex_;
+  mutable Mutex mutex_;
   int events_logged_ RTC_GUARDED_BY(mutex_) = 0;
 };
 
 }  // namespace
 
-namespace webrtc {
-
 TEST(EventTracerTest, EventTracerDisabled) {
   { TRACE_EVENT0("webrtc-test", "EventTracerDisabled"); }
   EXPECT_FALSE(TestStatistics::Get()->Count());
diff --git a/rtc_base/numerics/event_based_exponential_moving_average.cc b/rtc_base/numerics/event_based_exponential_moving_average.cc
index 419902dc52..072f1a1aed 100644
--- a/rtc_base/numerics/event_based_exponential_moving_average.cc
+++ b/rtc_base/numerics/event_based_exponential_moving_average.cc
@@ -17,6 +17,8 @@
 
 #include "rtc_base/checks.h"
 
+namespace webrtc {
+
 namespace {
 
 // For a normal distributed value, the 95% double sided confidence interval is
@@ -25,8 +27,6 @@ constexpr double ninetyfive_percent_confidence = 1.96;
 
 }  // namespace
 
-namespace webrtc {
-
 // `half_time` specifies how much weight will be given to old samples,
 // a sample gets exponentially less weight so that it's 50%
 // after `half_time` time units has passed.
diff --git a/rtc_base/numerics/event_based_exponential_moving_average_unittest.cc b/rtc_base/numerics/event_based_exponential_moving_average_unittest.cc
index 14ef9cb5e7..fcd7b733ea 100644
--- a/rtc_base/numerics/event_based_exponential_moving_average_unittest.cc
+++ b/rtc_base/numerics/event_based_exponential_moving_average_unittest.cc
@@ -16,6 +16,8 @@
 
 #include "test/gtest.h"
 
+namespace webrtc {
+
 namespace {
 
 constexpr int kHalfTime = 500;
@@ -23,8 +25,6 @@ constexpr double kError = 0.1;
 
 }  // namespace
 
-namespace webrtc {
-
 TEST(EventBasedExponentialMovingAverageTest, NoValue) {
   EventBasedExponentialMovingAverage average(kHalfTime);
 
diff --git a/rtc_base/openssl_session_cache_unittest.cc b/rtc_base/openssl_session_cache_unittest.cc
index 4b4da72211..148934497e 100644
--- a/rtc_base/openssl_session_cache_unittest.cc
+++ b/rtc_base/openssl_session_cache_unittest.cc
@@ -17,6 +17,8 @@
 #include "rtc_base/ssl_stream_adapter.h"
 #include "test/gtest.h"
 
+namespace webrtc {
+
 namespace {
 // Use methods that avoid X509 objects if possible.
 SSL_CTX* NewDtlsContext() {
@@ -44,8 +46,6 @@ SSL_SESSION* NewSslSession(SSL_CTX* ssl_ctx) {
 
 }  // namespace
 
-namespace webrtc {
-
 TEST(OpenSSLSessionCache, DTLSModeSetCorrectly) {
   SSL_CTX* ssl_ctx = NewDtlsContext();
 
diff --git a/rtc_base/physical_socket_server.cc b/rtc_base/physical_socket_server.cc
index e11f630271..66f649306c 100644
--- a/rtc_base/physical_socket_server.cc
+++ b/rtc_base/physical_socket_server.cc
@@ -114,6 +114,8 @@ typedef char* SockOptArg;
 #endif  // !defined(EPOLLRDHUP)
 #endif  // defined(WEBRTC_LINUX)
 
+namespace webrtc {
+
 namespace {
 
 // RFC-3168, Section 5. ECN is the two least significant bits.
@@ -121,7 +123,7 @@ constexpr uint8_t kEcnMask = 0x03;
 
 #if defined(WEBRTC_POSIX)
 
-webrtc::EcnMarking EcnFromDs(uint8_t ds) {
+EcnMarking EcnFromDs(uint8_t ds) {
   // RFC-3168, Section 5.
   constexpr uint8_t ECN_ECT1 = 0x01;
   constexpr uint8_t ECN_ECT0 = 0x02;
@@ -129,15 +131,15 @@ webrtc::EcnMarking EcnFromDs(uint8_t ds) {
   const uint8_t ecn = ds & kEcnMask;
 
   if (ecn == ECN_ECT1) {
-    return webrtc::EcnMarking::kEct1;
+    return EcnMarking::kEct1;
   }
   if (ecn == ECN_ECT0) {
-    return webrtc::EcnMarking::kEct0;
+    return EcnMarking::kEct0;
   }
   if (ecn == ECN_CE) {
-    return webrtc::EcnMarking::kCe;
+    return EcnMarking::kCe;
   }
-  return webrtc::EcnMarking::kNotEct;
+  return EcnMarking::kNotEct;
 }
 
 #endif
@@ -156,7 +158,6 @@ class ScopedSetTrue {
 
 }  // namespace
 
-namespace webrtc {
 
 PhysicalSocket::PhysicalSocket(PhysicalSocketServer* ss, SOCKET s)
     : ss_(ss),
diff --git a/rtc_base/thread.cc b/rtc_base/thread.cc
index b3c541ffa8..4bea295d6b 100644
--- a/rtc_base/thread.cc
+++ b/rtc_base/thread.cc
@@ -1,5 +1,5 @@
 /*
- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *  Copyright 2004 The WebRTC Project Authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
@@ -77,6 +77,7 @@ void* objc_autoreleasePoolPush(void);
 void objc_autoreleasePoolPop(void* pool);
 }
 
+namespace webrtc {
 namespace {
 class ScopedAutoReleasePool {
  public:
@@ -87,6 +88,7 @@ class ScopedAutoReleasePool {
   void* const pool_;
 };
 }  // namespace
+}  // namespace webrtc
 #endif
 
 namespace webrtc {
diff --git a/rtc_base/win/create_direct3d_device.cc b/rtc_base/win/create_direct3d_device.cc
index 1854fc57cb..ce9e003a16 100644
--- a/rtc_base/win/create_direct3d_device.cc
+++ b/rtc_base/win/create_direct3d_device.cc
@@ -12,6 +12,8 @@
 
 #include <libloaderapi.h>
 
+namespace webrtc {
+
 namespace {
 
 FARPROC LoadD3D11Function(const char* function_name) {
@@ -30,8 +32,6 @@ GetCreateDirect3D11DeviceFromDXGIDevice() {
 
 }  // namespace
 
-namespace webrtc {
-
 bool ResolveCoreWinRTDirect3DDelayload() {
   return GetCreateDirect3D11DeviceFromDXGIDevice();
 }
diff --git a/rtc_base/win/get_activation_factory.cc b/rtc_base/win/get_activation_factory.cc
index 0559fe7ac2..b875f7bf1f 100644
--- a/rtc_base/win/get_activation_factory.cc
+++ b/rtc_base/win/get_activation_factory.cc
@@ -15,6 +15,8 @@
 
 #include "rtc_base/win/hstring.h"
 
+namespace webrtc {
+
 namespace {
 
 FARPROC LoadComBaseFunction(const char* function_name) {
@@ -32,8 +34,6 @@ decltype(&::RoGetActivationFactory) GetRoGetActivationFactoryFunction() {
 
 }  // namespace
 
-namespace webrtc {
-
 bool ResolveCoreWinRTDelayload() {
   return GetRoGetActivationFactoryFunction() &&
          ResolveCoreWinRTStringDelayload();
diff --git a/rtc_base/win/hstring.cc b/rtc_base/win/hstring.cc
index d92112f6c6..a16050493f 100644
--- a/rtc_base/win/hstring.cc
+++ b/rtc_base/win/hstring.cc
@@ -15,6 +15,8 @@
 
 #include <cstdint>
 
+namespace webrtc {
+
 namespace {
 
 FARPROC LoadComBaseFunction(const char* function_name) {
@@ -39,8 +41,6 @@ decltype(&::WindowsDeleteString) GetWindowsDeleteString() {
 
 }  // namespace
 
-namespace webrtc {
-
 bool ResolveCoreWinRTStringDelayload() {
   return GetWindowsDeleteString() && GetWindowsCreateString();
 }
diff --git a/rtc_base/win/windows_version.cc b/rtc_base/win/windows_version.cc
index f059ea258a..5c1c5581f1 100644
--- a/rtc_base/win/windows_version.cc
+++ b/rtc_base/win/windows_version.cc
@@ -24,6 +24,8 @@
 
 #if !defined(WINUWP)
 
+namespace webrtc {
+
 namespace {
 
 typedef BOOL(WINAPI* GetProductInfoPtr)(DWORD, DWORD, DWORD, DWORD, PDWORD);
@@ -167,6 +169,8 @@ class RegKey {
 
 }  // namespace
 
+}  // namespace webrtc
+
 #endif  // !defined(WINUWP)
 
 namespace webrtc {
diff --git a/rtc_tools/data_channel_benchmark/peer_connection_client.cc b/rtc_tools/data_channel_benchmark/peer_connection_client.cc
index 7be633151b..37d6e956e5 100644
--- a/rtc_tools/data_channel_benchmark/peer_connection_client.cc
+++ b/rtc_tools/data_channel_benchmark/peer_connection_client.cc
@@ -41,18 +41,20 @@
 #include "rtc_base/thread.h"
 #include "rtc_tools/data_channel_benchmark/signaling_interface.h"
 
+namespace webrtc {
+
 namespace {
 
 constexpr char kStunServer[] = "stun:stun.l.google.com:19302";
 
 class SetLocalDescriptionObserverAdapter
-    : public webrtc::SetLocalDescriptionObserverInterface {
+    : public SetLocalDescriptionObserverInterface {
  public:
-  using Callback = std::function<void(webrtc::RTCError)>;
-  static webrtc::scoped_refptr<SetLocalDescriptionObserverAdapter> Create(
+  using Callback = std::function<void(RTCError)>;
+  static scoped_refptr<SetLocalDescriptionObserverAdapter> Create(
       Callback callback) {
-    return webrtc::scoped_refptr<SetLocalDescriptionObserverAdapter>(
-        new webrtc::RefCountedObject<SetLocalDescriptionObserverAdapter>(
+    return scoped_refptr<SetLocalDescriptionObserverAdapter>(
+        new RefCountedObject<SetLocalDescriptionObserverAdapter>(
             std::move(callback)));
   }
 
@@ -61,7 +63,7 @@ class SetLocalDescriptionObserverAdapter
   ~SetLocalDescriptionObserverAdapter() override = default;
 
  private:
-  void OnSetLocalDescriptionComplete(webrtc::RTCError error) override {
+  void OnSetLocalDescriptionComplete(RTCError error) override {
     callback_(std::move(error));
   }
 
@@ -69,13 +71,13 @@ class SetLocalDescriptionObserverAdapter
 };
 
 class SetRemoteDescriptionObserverAdapter
-    : public webrtc::SetRemoteDescriptionObserverInterface {
+    : public SetRemoteDescriptionObserverInterface {
  public:
-  using Callback = std::function<void(webrtc::RTCError)>;
-  static webrtc::scoped_refptr<SetRemoteDescriptionObserverAdapter> Create(
+  using Callback = std::function<void(RTCError)>;
+  static scoped_refptr<SetRemoteDescriptionObserverAdapter> Create(
       Callback callback) {
-    return webrtc::scoped_refptr<SetRemoteDescriptionObserverAdapter>(
-        new webrtc::RefCountedObject<SetRemoteDescriptionObserverAdapter>(
+    return scoped_refptr<SetRemoteDescriptionObserverAdapter>(
+        new RefCountedObject<SetRemoteDescriptionObserverAdapter>(
             std::move(callback)));
   }
 
@@ -84,7 +86,7 @@ class SetRemoteDescriptionObserverAdapter
   ~SetRemoteDescriptionObserverAdapter() override = default;
 
  private:
-  void OnSetRemoteDescriptionComplete(webrtc::RTCError error) override {
+  void OnSetRemoteDescriptionComplete(RTCError error) override {
     callback_(std::move(error));
   }
 
@@ -92,16 +94,16 @@ class SetRemoteDescriptionObserverAdapter
 };
 
 class CreateSessionDescriptionObserverAdapter
-    : public webrtc::CreateSessionDescriptionObserver {
+    : public CreateSessionDescriptionObserver {
  public:
-  using Success = std::function<void(webrtc::SessionDescriptionInterface*)>;
-  using Failure = std::function<void(webrtc::RTCError)>;
+  using Success = std::function<void(SessionDescriptionInterface*)>;
+  using Failure = std::function<void(RTCError)>;
 
-  static webrtc::scoped_refptr<CreateSessionDescriptionObserverAdapter> Create(
+  static scoped_refptr<CreateSessionDescriptionObserverAdapter> Create(
       Success success,
       Failure failure) {
-    return webrtc::scoped_refptr<CreateSessionDescriptionObserverAdapter>(
-        new webrtc::RefCountedObject<CreateSessionDescriptionObserverAdapter>(
+    return scoped_refptr<CreateSessionDescriptionObserverAdapter>(
+        new RefCountedObject<CreateSessionDescriptionObserverAdapter>(
             std::move(success), std::move(failure)));
   }
 
@@ -110,13 +112,9 @@ class CreateSessionDescriptionObserverAdapter
   ~CreateSessionDescriptionObserverAdapter() override = default;
 
  private:
-  void OnSuccess(webrtc::SessionDescriptionInterface* desc) override {
-    success_(desc);
-  }
+  void OnSuccess(SessionDescriptionInterface* desc) override { success_(desc); }
 
-  void OnFailure(webrtc::RTCError error) override {
-    failure_(std::move(error));
-  }
+  void OnFailure(RTCError error) override { failure_(std::move(error)); }
 
   Success success_;
   Failure failure_;
@@ -124,8 +122,6 @@ class CreateSessionDescriptionObserverAdapter
 
 }  // namespace
 
-namespace webrtc {
-
 PeerConnectionClient::PeerConnectionClient(
     PeerConnectionFactoryInterface* factory,
     SignalingInterface* signaling)
diff --git a/rtc_tools/video_replay.cc b/rtc_tools/video_replay.cc
index ceb13849df..2f7a1b541f 100644
--- a/rtc_tools/video_replay.cc
+++ b/rtc_tools/video_replay.cc
@@ -227,6 +227,7 @@ std::string ReadFileToString(const std::string& file_path) {
 }  // namespace
 
 namespace webrtc {
+
 namespace {
 
 class NullRenderer : public VideoSinkInterface<VideoFrame> {
@@ -738,6 +739,7 @@ void RtpReplay() {
 }
 
 }  // namespace
+
 }  // namespace webrtc
 
 int main(int argc, char* argv[]) {
diff --git a/sdk/android/src/jni/jni_helpers.h b/sdk/android/src/jni/jni_helpers.h
index d86c8fa4ad..950190ece0 100644
--- a/sdk/android/src/jni/jni_helpers.h
+++ b/sdk/android/src/jni/jni_helpers.h
@@ -38,9 +38,6 @@
 namespace webrtc {
 namespace jni {
 
-// TODO(sakal): Remove once clients have migrated.
-using ::webrtc::JavaToStdMapStrings;
-
 // Deprecated, use NativeToJavaPointer.
 inline long jlongFromPointer(void* ptr) {
   return NativeToJavaPointer(ptr);
diff --git a/test/fake_vp8_encoder.cc b/test/fake_vp8_encoder.cc
index f32e281719..7a1e3fe88d 100644
--- a/test/fake_vp8_encoder.cc
+++ b/test/fake_vp8_encoder.cc
@@ -29,6 +29,10 @@
 #include "rtc_base/synchronization/mutex.h"
 #include "test/fake_encoder.h"
 
+namespace webrtc {
+
+namespace test {
+
 namespace {
 
 // Write width and height to the payload the same way as the real encoder does.
@@ -49,10 +53,6 @@ void WriteFakeVp8(unsigned char* payload,
 }
 }  // namespace
 
-namespace webrtc {
-
-namespace test {
-
 FakeVp8Encoder::FakeVp8Encoder(const Environment& env) : FakeEncoder(env) {
   sequence_checker_.Detach();
 }
diff --git a/test/network/emulated_turn_server.cc b/test/network/emulated_turn_server.cc
index e6d23f16b6..ff2fc96b1a 100644
--- a/test/network/emulated_turn_server.cc
+++ b/test/network/emulated_turn_server.cc
@@ -34,68 +34,66 @@
 #include "rtc_base/task_queue_for_test.h"
 #include "rtc_base/thread.h"
 
+namespace webrtc {
+namespace test {
+
 namespace {
 
 const char kTestRealm[] = "example.org";
 const char kTestSoftware[] = "TestTurnServer";
 
 // A wrapper class for webrtc::TurnServer to allocate sockets.
-class PacketSocketFactoryWrapper : public webrtc::PacketSocketFactory {
+class PacketSocketFactoryWrapper : public PacketSocketFactory {
  public:
-  explicit PacketSocketFactoryWrapper(
-      webrtc::test::EmulatedTURNServer* turn_server)
+  explicit PacketSocketFactoryWrapper(EmulatedTURNServer* turn_server)
       : turn_server_(turn_server) {}
   ~PacketSocketFactoryWrapper() override {}
 
   // This method is called from TurnServer when making a TURN ALLOCATION.
   // It will create a socket on the `peer_` endpoint.
-  std::unique_ptr<webrtc::AsyncPacketSocket> CreateUdpSocket(
-      const webrtc::Environment& env,
-      const webrtc::SocketAddress& address,
+  std::unique_ptr<AsyncPacketSocket> CreateUdpSocket(
+      const Environment& env,
+      const SocketAddress& address,
       uint16_t min_port,
       uint16_t max_port) override {
     return turn_server_->CreatePeerSocket();
   }
 
-  std::unique_ptr<webrtc::AsyncListenSocket> CreateServerTcpSocket(
-      const webrtc::Environment& env,
-      const webrtc::SocketAddress& local_address,
+  std::unique_ptr<AsyncListenSocket> CreateServerTcpSocket(
+      const Environment& env,
+      const SocketAddress& local_address,
       uint16_t min_port,
       uint16_t max_port,
       int opts) override {
     return nullptr;
   }
-  std::unique_ptr<webrtc::AsyncPacketSocket> CreateClientTcpSocket(
-      const webrtc::Environment& env,
-      const webrtc::SocketAddress& local_address,
-      const webrtc::SocketAddress& remote_address,
-      const webrtc::PacketSocketTcpOptions& tcp_options) override {
+  std::unique_ptr<AsyncPacketSocket> CreateClientTcpSocket(
+      const Environment& env,
+      const SocketAddress& local_address,
+      const SocketAddress& remote_address,
+      const PacketSocketTcpOptions& tcp_options) override {
     return nullptr;
   }
-  std::unique_ptr<webrtc::AsyncDnsResolverInterface> CreateAsyncDnsResolver()
-      override {
+  std::unique_ptr<AsyncDnsResolverInterface> CreateAsyncDnsResolver() override {
     return nullptr;
   }
 
-  std::unique_ptr<webrtc::AsyncPacketSocket> CreateClientUdpSocket(
-      const webrtc::Environment& env,
-      const webrtc::SocketAddress& local_address,
-      const webrtc::SocketAddress& remote_address,
+  std::unique_ptr<AsyncPacketSocket> CreateClientUdpSocket(
+      const Environment& env,
+      const SocketAddress& local_address,
+      const SocketAddress& remote_address,
       uint16_t min_port,
       uint16_t max_port,
-      const webrtc::PacketSocketTcpOptions& udp_options) override {
+      const PacketSocketTcpOptions& udp_options) override {
     return nullptr;
   }
 
  private:
-  webrtc::test::EmulatedTURNServer* turn_server_;
+  EmulatedTURNServer* turn_server_;
 };
 
 }  //  namespace
 
-namespace webrtc {
-namespace test {
-
 // A wrapper class for copying data between an AsyncPacketSocket and a
 // EmulatedEndpoint. This is used by the webrtc::TurnServer when
 // sending data back into the emulated network.
diff --git a/test/rtp_file_reader.cc b/test/rtp_file_reader.cc
index 6d4f58e1e8..704b919005 100644
--- a/test/rtp_file_reader.cc
+++ b/test/rtp_file_reader.cc
@@ -27,6 +27,9 @@
 #include "rtc_base/logging.h"
 #include "rtc_base/time_utils.h"
 
+namespace webrtc {
+namespace test {
+
 namespace {
 constexpr size_t kRtpDumpFirstLineLength = 80;
 constexpr uint16_t kRtpDumpPacketHeaderSize = 8;
@@ -109,9 +112,6 @@ bool ReadUint16(uint16_t* out, FILE* file) {
 
 }  // namespace
 
-namespace webrtc {
-namespace test {
-
 class RtpFileReaderImpl : public RtpFileReader {
  public:
   virtual bool Init(FILE* file, const std::set<uint32_t>& ssrc_filter) = 0;
diff --git a/test/testsupport/test_artifacts.cc b/test/testsupport/test_artifacts.cc
index c700ddc4c0..5363c9a3a9 100644
--- a/test/testsupport/test_artifacts.cc
+++ b/test/testsupport/test_artifacts.cc
@@ -19,16 +19,22 @@
 #include "rtc_base/system/file_wrapper.h"
 #include "test/testsupport/file_utils.h"
 
+namespace webrtc {
+namespace test {
+
 namespace {
 const std::string& DefaultArtifactPath() {
-  static const std::string path = webrtc::test::OutputPathWithRandomDirectory();
+  static const std::string path = OutputPathWithRandomDirectory();
   return path;
 }
 }  // namespace
 
+}  // namespace test
+}  // namespace webrtc
+
 ABSL_FLAG(std::string,
           test_artifacts_dir,
-          DefaultArtifactPath().c_str(),
+          webrtc::test::DefaultArtifactPath().c_str(),
           "The output folder where test output should be saved.");
 
 namespace webrtc {

From fb5a86aecea031efdbfb2403a52383e7082d7366 Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Fri, 6 Mar 2026 13:32:25 +0100
Subject: [PATCH 225/472] Replace ArrayView with std::span in pc/

Bug: webrtc:439801349
Change-Id: Ia62137fc296eabf19a9b3d3a13a95f2494d3f881
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/454342
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47078}
---
 pc/BUILD.gn                        | 12 ------------
 pc/channel_unittest.cc             | 20 ++++++++------------
 pc/data_channel_controller.cc      |  4 ++--
 pc/data_channel_controller.h       |  4 ++--
 pc/datagram_connection_internal.cc |  4 ++--
 pc/datagram_connection_internal.h  |  4 ++--
 pc/datagram_connection_unittest.cc | 12 ++++++------
 pc/dtls_srtp_transport_unittest.cc |  6 +++---
 pc/jsep_transport.cc               |  4 ++--
 pc/jsep_transport_unittest.cc      |  4 ++--
 pc/media_session_unittest.cc       | 14 +++++++-------
 pc/rtc_stats_collector.cc          |  1 -
 pc/rtp_transceiver.cc              |  6 +++---
 pc/rtp_transceiver.h               | 10 +++++-----
 pc/rtp_transport.cc                |  1 -
 pc/sdp_offer_answer.cc             |  4 ++--
 pc/srtp_session.cc                 |  1 -
 pc/srtp_session_unittest.cc        | 16 ++++++++--------
 pc/srtp_transport.cc               |  1 -
 pc/srtp_transport_unittest.cc      |  6 +++---
 pc/track_media_info_map.cc         | 18 +++++++++---------
 pc/track_media_info_map.h          |  8 ++++----
 22 files changed, 70 insertions(+), 90 deletions(-)

diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 2a641da2c2..20e7f5d8e2 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -131,7 +131,6 @@ rtc_library("dtls_srtp_transport") {
   ]
   deps = [
     ":srtp_transport",
-    "../api:array_view",
     "../api:dtls_transport_interface",
     "../api:field_trials_view",
     "../p2p:dtls_transport_internal",
@@ -200,7 +199,6 @@ rtc_library("jsep_transport") {
     ":sctp_transport",
     ":session_description",
     ":transport_stats",
-    "../api:array_view",
     "../api:candidate",
     "../api:dtls_transport_interface",
     "../api:ice_transport_interface",
@@ -583,7 +581,6 @@ rtc_library("rtp_transport") {
   deps = [
     ":rtp_transport_internal",
     ":session_description",
-    "../api:array_view",
     "../api:field_trials_view",
     "../api:sequence_checker",
     "../api/task_queue",
@@ -686,7 +683,6 @@ rtc_library("srtp_session") {
     "srtp_session.h",
   ]
   deps = [
-    "../api:array_view",
     "../api:field_trials_view",
     "../api:sequence_checker",
     "../modules/rtp_rtcp:rtp_rtcp_format",
@@ -718,7 +714,6 @@ rtc_library("srtp_transport") {
   deps = [
     ":rtp_transport",
     ":srtp_session",
-    "../api:array_view",
     "../api:field_trials_view",
     "../api/units:timestamp",
     "../call:rtp_receiver",
@@ -926,7 +921,6 @@ rtc_library("data_channel_controller") {
     ":peer_connection_internal",
     ":sctp_data_channel",
     ":sctp_utils",
-    "../api:array_view",
     "../api:data_channel_event_observer_interface",
     "../api:data_channel_interface",
     "../api:priority",
@@ -1010,7 +1004,6 @@ rtc_library("rtc_stats_collector") {
     ":rtp_transceiver",
     ":track_media_info_map",
     ":transport_stats",
-    "../api:array_view",
     "../api:candidate",
     "../api:data_channel_interface",
     "../api:dtls_transport_interface",
@@ -1135,7 +1128,6 @@ rtc_library("sdp_offer_answer") {
     ":transceiver_list",
     ":usage_pattern",
     ":webrtc_session_description_factory",
-    "../api:array_view",
     "../api:audio_options_api",
     "../api:candidate",
     "../api:jsep",
@@ -1468,7 +1460,6 @@ rtc_library("track_media_info_map") {
     "track_media_info_map.h",
   ]
   deps = [
-    "../api:array_view",
     "../api:rtp_parameters",
     "../media:media_channel",
     "../rtc_base:checks",
@@ -1677,7 +1668,6 @@ rtc_library("rtp_transceiver") {
     ":rtp_transport_internal",
     ":session_description",
     ":video_rtp_receiver",
-    "../api:array_view",
     "../api:audio_options_api",
     "../api:jsep",
     "../api:make_ref_counted",
@@ -2123,7 +2113,6 @@ rtc_library("datagram_connection_internal") {
     ":dtls_srtp_transport",
     ":dtls_transport",
     ":ice_transport",
-    "../api:array_view",
     "../api:candidate",
     "../api:datagram_connection",
     "../api:dtls_transport_interface",
@@ -2233,7 +2222,6 @@ if (rtc_include_tests && !build_with_chromium) {
       ":transport_stats",
       ":used_ids",
       ":video_rtp_receiver",
-      "../api:array_view",
       "../api:audio_options_api",
       "../api:candidate",
       "../api:datagram_connection",
diff --git a/pc/channel_unittest.cc b/pc/channel_unittest.cc
index 68c35f678c..0c00c480dd 100644
--- a/pc/channel_unittest.cc
+++ b/pc/channel_unittest.cc
@@ -14,12 +14,12 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/functional/any_invocable.h"
-#include "api/array_view.h"
 #include "api/audio_options.h"
 #include "api/crypto/crypto_options.h"
 #include "api/field_trials.h"
@@ -68,7 +68,6 @@ namespace {
 using ::testing::AllOf;
 using ::testing::ElementsAre;
 using ::testing::Field;
-using ::webrtc::ArrayView;
 using ::webrtc::CreateTestFieldTrials;
 using ::webrtc::DtlsTransportInternal;
 using ::webrtc::FakeVoiceMediaReceiveChannel;
@@ -149,8 +148,8 @@ class ChannelTest : public ::testing::Test {
   };
 
   ChannelTest(bool verify_playout,
-              webrtc::ArrayView<const uint8_t> rtp_data,
-              webrtc::ArrayView<const uint8_t> rtcp_data,
+              std::span<const uint8_t> rtp_data,
+              std::span<const uint8_t> rtcp_data,
               NetworkIsWorker network_is_worker)
       : verify_playout_(verify_playout),
         rtp_packet_(rtp_data.data(), rtp_data.size()),
@@ -528,9 +527,8 @@ class ChannelTest : public ::testing::Test {
                                int pl_type) {
     webrtc::Buffer data(rtp_packet_.data(), rtp_packet_.size());
     // Set SSRC in the rtp packet copy.
-    webrtc::SetBE32(webrtc::ArrayView<uint8_t>(data).subspan(8, 4), ssrc);
-    webrtc::SetBE16(webrtc::ArrayView<uint8_t>(data).subspan(2, 2),
-                    sequence_number);
+    webrtc::SetBE32(std::span<uint8_t>(data).subspan(8, 4), ssrc);
+    webrtc::SetBE16(std::span<uint8_t>(data).subspan(2, 2), sequence_number);
     if (pl_type >= 0) {
       webrtc::Set8(data, 1, static_cast<uint8_t>(pl_type));
     }
@@ -1270,7 +1268,7 @@ class ChannelTest : public ::testing::Test {
     EXPECT_TRUE(CheckNoRtp2());
   }
 
-  void SendBundleToBundle(ArrayView<const int, 2> pl_types,
+  void SendBundleToBundle(std::span<const int, 2> pl_types,
                           bool rtcp_mux,
                           bool secure) {
     int sequence_number1_1 = 0, sequence_number2_2 = 0;
@@ -1553,9 +1551,7 @@ class ChannelTest : public ::testing::Test {
   }
 
  protected:
-  void WaitForThreads() {
-    WaitForThreads(webrtc::ArrayView<webrtc::Thread*>());
-  }
+  void WaitForThreads() { WaitForThreads(std::span<webrtc::Thread*>()); }
   static void ProcessThreadQueue(webrtc::Thread* thread) {
     RTC_DCHECK(thread->IsCurrent());
     while (!thread->empty()) {
@@ -1565,7 +1561,7 @@ class ChannelTest : public ::testing::Test {
   static void FlushCurrentThread() {
     webrtc::Thread::Current()->ProcessMessages(0);
   }
-  void WaitForThreads(webrtc::ArrayView<webrtc::Thread*> threads) {
+  void WaitForThreads(std::span<webrtc::Thread*> threads) {
     // `threads` and current thread post packets to network thread.
     for (webrtc::Thread* thread : threads) {
       SendTask(thread, [thread] { ProcessThreadQueue(thread); });
diff --git a/pc/data_channel_controller.cc b/pc/data_channel_controller.cc
index 7807ccb6ee..19c81954d4 100644
--- a/pc/data_channel_controller.cc
+++ b/pc/data_channel_controller.cc
@@ -14,6 +14,7 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
@@ -21,7 +22,6 @@
 #include "absl/algorithm/container.h"
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/data_channel_event_observer_interface.h"
 #include "api/data_channel_interface.h"
 #include "api/priority.h"
@@ -527,7 +527,7 @@ void DataChannelController::set_data_channel_transport(
 std::optional<Message> DataChannelController::BuildObserverMessage(
     StreamId sid,
     DataMessageType type,
-    ArrayView<const uint8_t> payload,
+    std::span<const uint8_t> payload,
     Message::Direction direction) const {
   RTC_DCHECK_RUN_ON(network_thread());
 
diff --git a/pc/data_channel_controller.h b/pc/data_channel_controller.h
index 2553810c0c..de9aa7b7db 100644
--- a/pc/data_channel_controller.h
+++ b/pc/data_channel_controller.h
@@ -15,10 +15,10 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/data_channel_event_observer_interface.h"
 #include "api/data_channel_interface.h"
 #include "api/priority.h"
@@ -154,7 +154,7 @@ class DataChannelController : public SctpDataChannelControllerInterface,
   BuildObserverMessage(
       StreamId sid,
       DataMessageType type,
-      ArrayView<const uint8_t> payload,
+      std::span<const uint8_t> payload,
       DataChannelEventObserverInterface::Message::Direction direction) const
       RTC_RUN_ON(network_thread());
 
diff --git a/pc/datagram_connection_internal.cc b/pc/datagram_connection_internal.cc
index afddb3aa36..be25bbadfc 100644
--- a/pc/datagram_connection_internal.cc
+++ b/pc/datagram_connection_internal.cc
@@ -14,11 +14,11 @@
 #include <functional>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/candidate.h"
 #include "api/crypto/crypto_options.h"
 #include "api/datagram_connection.h"
@@ -254,7 +254,7 @@ void DatagramConnectionInternal::SetRemoteDtlsParameters(
 }
 
 void DatagramConnectionInternal::SendPackets(
-    ArrayView<PacketSendParameters> packets) {
+    std::span<PacketSendParameters> packets) {
   RTC_DCHECK_RUN_ON(&sequence_checker_);
   for (size_t i = 0; i < packets.size(); ++i) {
     SendSinglePacket(packets[i],
diff --git a/pc/datagram_connection_internal.h b/pc/datagram_connection_internal.h
index 68705b7cba..11aa5a64cc 100644
--- a/pc/datagram_connection_internal.h
+++ b/pc/datagram_connection_internal.h
@@ -13,11 +13,11 @@
 #include <cstddef>
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <string>
 
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/candidate.h"
 #include "api/datagram_connection.h"
 #include "api/environment/environment.h"
@@ -62,7 +62,7 @@ class RTC_EXPORT DatagramConnectionInternal : public DatagramConnection,
                                const uint8_t* digest,
                                size_t digest_len,
                                SSLRole ssl_role) override;
-  void SendPackets(ArrayView<PacketSendParameters> packets) override;
+  void SendPackets(std::span<PacketSendParameters> packets) override;
 
   void Terminate(
       absl::AnyInvocable<void()> terminate_complete_callback) override;
diff --git a/pc/datagram_connection_unittest.cc b/pc/datagram_connection_unittest.cc
index 37507b5fd9..fae8c8b39d 100644
--- a/pc/datagram_connection_unittest.cc
+++ b/pc/datagram_connection_unittest.cc
@@ -13,11 +13,11 @@
 #include <cstring>
 #include <memory>
 #include <set>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/candidate.h"
 #include "api/environment/environment.h"
 #include "api/environment/environment_factory.h"
@@ -190,7 +190,7 @@ TEST_F(DatagramConnectionTest, ObserverCalledOnReceivedRtpPacket) {
   bool callback_called = false;
   EXPECT_CALL(*observer1_ptr_, OnPacketReceived(_, _))
       .WillOnce(
-          [&](ArrayView<const uint8_t> data,
+          [&](std::span<const uint8_t> data,
               const DatagramConnection::Observer::PacketMetadata& metadata) {
             EXPECT_EQ(data.size(), packet_data.size());
             EXPECT_EQ(
@@ -256,7 +256,7 @@ TEST_F(DatagramConnectionTest, RtpPacketsAreReceived) {
 
   EXPECT_CALL(*observer2_ptr_, OnPacketReceived(_, _))
       .WillOnce(
-          [&](ArrayView<const uint8_t> received_data,
+          [&](std::span<const uint8_t> received_data,
               const DatagramConnection::Observer::PacketMetadata& metadata) {
             EXPECT_EQ(received_data.size(), data.size());
             EXPECT_EQ(memcmp(received_data.data(), data.data(), data.size()),
@@ -319,7 +319,7 @@ TEST_F(DatagramConnectionTest, SendMultipleRtpPackets) {
   EXPECT_CALL(*observer2_ptr_, OnPacketReceived(_, _))
       .Times(data.size())
       .WillRepeatedly(
-          [&](ArrayView<const uint8_t> received_data,
+          [&](std::span<const uint8_t> received_data,
               const DatagramConnection::Observer::PacketMetadata& metadata) {
             received_packets.emplace_back(received_data.begin(),
                                           received_data.end());
@@ -395,7 +395,7 @@ TEST_F(DatagramConnectionTest, NonRtpPacketsInSRTPModeAreDTLSProtected) {
   bool callback_called = false;
   EXPECT_CALL(*observer2_ptr_, OnPacketReceived(_, _))
       .WillOnce(
-          [&](ArrayView<const uint8_t> received_data,
+          [&](std::span<const uint8_t> received_data,
               const DatagramConnection::Observer::PacketMetadata& metadata) {
             // Check the data is decrypted correctly.
             EXPECT_EQ(received_data.size(), non_rtp_data.size());
@@ -490,7 +490,7 @@ TEST_F(DatagramConnectionTest, DirectDtlsPacketsAreReceived) {
 
   EXPECT_CALL(*observer2_ptr_, OnPacketReceived(_, _))
       .WillOnce(
-          [&](ArrayView<const uint8_t> received_data,
+          [&](std::span<const uint8_t> received_data,
               const DatagramConnection::Observer::PacketMetadata& metadata) {
             EXPECT_EQ(received_data.size(), data.size());
             EXPECT_EQ(memcmp(received_data.data(), data.data(), data.size()),
diff --git a/pc/dtls_srtp_transport_unittest.cc b/pc/dtls_srtp_transport_unittest.cc
index 3279b3124b..e42d9b75f4 100644
--- a/pc/dtls_srtp_transport_unittest.cc
+++ b/pc/dtls_srtp_transport_unittest.cc
@@ -14,11 +14,11 @@
 #include <cstring>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/field_trials.h"
 #include "api/make_ref_counted.h"
 #include "api/transport/ecn_marking.h"
@@ -138,7 +138,7 @@ class DtlsSrtpTransportTest : public ::testing::Test {
     memcpy(rtp_packet_data, kPcmuFrame, rtp_len);
     // In order to be able to run this test function multiple times we can not
     // use the same sequence number twice. Increase the sequence number by one.
-    SetBE16(ArrayView<uint8_t>(rtp_packet_buffer).subspan(2),
+    SetBE16(std::span<uint8_t>(rtp_packet_buffer).subspan(2),
             ++sequence_number_);
     CopyOnWriteBuffer rtp_packet1to2(rtp_packet_data, rtp_len, packet_size);
     CopyOnWriteBuffer rtp_packet2to1(rtp_packet_data, rtp_len, packet_size);
@@ -211,7 +211,7 @@ class DtlsSrtpTransportTest : public ::testing::Test {
     memcpy(rtp_packet_data, kPcmuFrameWithExtensions, rtp_len);
     // In order to be able to run this test function multiple times we can not
     // use the same sequence number twice. Increase the sequence number by one.
-    SetBE16(ArrayView<uint8_t>(rtp_packet_buffer).subspan(2),
+    SetBE16(std::span<uint8_t>(rtp_packet_buffer).subspan(2),
             ++sequence_number_);
     CopyOnWriteBuffer rtp_packet1to2(rtp_packet_data, rtp_len, packet_size);
     CopyOnWriteBuffer rtp_packet2to1(rtp_packet_data, rtp_len, packet_size);
diff --git a/pc/jsep_transport.cc b/pc/jsep_transport.cc
index 82471c7181..415aa507b9 100644
--- a/pc/jsep_transport.cc
+++ b/pc/jsep_transport.cc
@@ -13,13 +13,13 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/candidate.h"
 #include "api/dtls_transport_interface.h"
 #include "api/ice_transport_interface.h"
@@ -433,7 +433,7 @@ RTCError JsepTransport::NegotiateAndSetDtlsParameters(
   } else {
     // We are not doing DTLS
     remote_fingerprint =
-        std::make_unique<SSLFingerprint>("", ArrayView<const uint8_t>());
+        std::make_unique<SSLFingerprint>("", std::span<const uint8_t>());
   }
   // Now that we have negotiated everything, push it downward.
   // Note that we cache the result so that if we have race conditions
diff --git a/pc/jsep_transport_unittest.cc b/pc/jsep_transport_unittest.cc
index a9f4189fda..3f49e7a755 100644
--- a/pc/jsep_transport_unittest.cc
+++ b/pc/jsep_transport_unittest.cc
@@ -15,12 +15,12 @@
 #include <memory>
 #include <optional>
 #include <ostream>
+#include <span>
 #include <string>
 #include <tuple>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/candidate.h"
 #include "api/field_trials.h"
 #include "api/ice_transport_interface.h"
@@ -1173,7 +1173,7 @@ class JsepTransport2HeaderExtensionTest
     memcpy(rtp_packet_data, kPcmuFrameWithExtensions, rtp_len);
     // In order to be able to run this test function multiple times we can not
     // use the same sequence number twice. Increase the sequence number by one.
-    SetBE16(ArrayView<uint8_t>(rtp_packet_buffer).subspan(2),
+    SetBE16(std::span<uint8_t>(rtp_packet_buffer).subspan(2),
             ++sequence_number_);
     CopyOnWriteBuffer rtp_packet(rtp_packet_data, rtp_len, packet_size);
 
diff --git a/pc/media_session_unittest.cc b/pc/media_session_unittest.cc
index 858f77947e..87aac0d674 100644
--- a/pc/media_session_unittest.cc
+++ b/pc/media_session_unittest.cc
@@ -15,6 +15,7 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <tuple>
 #include <vector>
@@ -22,7 +23,6 @@
 #include "absl/algorithm/container.h"
 #include "absl/strings/match.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/candidate.h"
 #include "api/environment/environment.h"
@@ -134,22 +134,22 @@ class CodecLookupHelperForTesting : public CodecLookupHelper {
     SetVideoCodecs(codecs, codecs);
   }
 
-  void SetAudioCodecs(ArrayView<const Codec> codecs) {
+  void SetAudioCodecs(std::span<const Codec> codecs) {
     SetAudioCodecs(std::vector<Codec>(codecs.begin(), codecs.end()));
   }
 
-  void SetVideoCodecs(ArrayView<const Codec> codecs) {
+  void SetVideoCodecs(std::span<const Codec> codecs) {
     SetVideoCodecs(std::vector<Codec>(codecs.begin(), codecs.end()));
   }
 
   template <typename U, size_t N>
   void SetAudioCodecs(U (&array)[N]) {
-    SetAudioCodecs(ArrayView<const Codec>(&array[0], N));
+    SetAudioCodecs(std::span<const Codec>(&array[0], N));
   }
 
   template <typename U, size_t N>
   void SetVideoCodecs(U (&array)[N]) {
-    SetVideoCodecs(ArrayView<const Codec>(&array[0], N));
+    SetVideoCodecs(std::span<const Codec>(&array[0], N));
   }
 
   void ClearAudioCodecs() { SetAudioCodecs(std::vector<Codec>()); }
@@ -260,8 +260,8 @@ const Codec kVideoCodecsH265Level6[] = {
     CreateVideoCodec(96, kH265MainProfileLevel6Sdp)};
 
 // Match two codec lists for content, but ignore the ID.
-bool CodecListsMatch(ArrayView<const Codec> list1,
-                     ArrayView<const Codec> list2) {
+bool CodecListsMatch(std::span<const Codec> list1,
+                     std::span<const Codec> list2) {
   if (list1.size() != list2.size()) {
     return false;
   }
diff --git a/pc/rtc_stats_collector.cc b/pc/rtc_stats_collector.cc
index 19582ad274..517d6f259f 100644
--- a/pc/rtc_stats_collector.cc
+++ b/pc/rtc_stats_collector.cc
@@ -24,7 +24,6 @@
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/str_cat.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio/audio_device.h"
 #include "api/audio/audio_processing_statistics.h"
 #include "api/candidate.h"
diff --git a/pc/rtp_transceiver.cc b/pc/rtp_transceiver.cc
index e92b8afb8c..520509277f 100644
--- a/pc/rtp_transceiver.cc
+++ b/pc/rtp_transceiver.cc
@@ -15,6 +15,7 @@
 #include <iterator>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
@@ -23,7 +24,6 @@
 #include "absl/base/nullability.h"
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio_options.h"
 #include "api/crypto/crypto_options.h"
 #include "api/environment/environment.h"
@@ -1020,7 +1020,7 @@ RtpTransceiver::GetStopTransceiverProcedure() {
 }
 
 RTCError RtpTransceiver::SetCodecPreferences(
-    ArrayView<RtpCodecCapability> codec_capabilities) {
+    std::span<RtpCodecCapability> codec_capabilities) {
   RTC_DCHECK(unified_plan_);
   // 3. If codecs is an empty list, set transceiver's [[PreferredCodecs]] slot
   // to codecs and abort these steps.
@@ -1182,7 +1182,7 @@ bool IsMandatoryHeaderExtension(absl::string_view uri) {
 }
 
 RTCError RtpTransceiver::SetHeaderExtensionsToNegotiate(
-    ArrayView<const RtpHeaderExtensionCapability> header_extensions) {
+    std::span<const RtpHeaderExtensionCapability> header_extensions) {
   RTC_DCHECK_RUN_ON(thread_);
   // https://w3c.github.io/webrtc-extensions/#dom-rtcrtptransceiver-setheaderextensionstonegotiate
   if (header_extensions.size() != header_extensions_to_negotiate_.size()) {
diff --git a/pc/rtp_transceiver.h b/pc/rtp_transceiver.h
index 6fc217674c..cad5276f6b 100644
--- a/pc/rtp_transceiver.h
+++ b/pc/rtp_transceiver.h
@@ -16,6 +16,7 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
@@ -23,7 +24,6 @@
 #include "absl/base/nullability.h"
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio_options.h"
 #include "api/crypto/crypto_options.h"
 #include "api/environment/environment.h"
@@ -343,7 +343,7 @@ class RtpTransceiver : public RtpTransceiverInterface {
   bool receptive() const override;
   RTCError StopStandard() override;
   void StopInternal() override;
-  RTCError SetCodecPreferences(ArrayView<RtpCodecCapability> codecs) override;
+  RTCError SetCodecPreferences(std::span<RtpCodecCapability> codecs) override;
   // TODO(https://crbug.com/webrtc/391275081): Delete codec_preferences() in
   // favor of filtered_codec_preferences() because it's not used anywhere.
   std::vector<RtpCodecCapability> codec_preferences() const override;
@@ -357,7 +357,7 @@ class RtpTransceiver : public RtpTransceiverInterface {
       const override;
 
   RTCError SetHeaderExtensionsToNegotiate(
-      ArrayView<const RtpHeaderExtensionCapability> header_extensions) override;
+      std::span<const RtpHeaderExtensionCapability> header_extensions) override;
 
   // Called on the signaling thread when the local or remote content description
   // is updated. Used to update the negotiated header extensions.
@@ -508,7 +508,7 @@ PROXY_CONSTMETHOD0(std::optional<RtpTransceiverDirection>, fired_direction)
 PROXY_CONSTMETHOD0(bool, receptive)
 PROXY_METHOD0(RTCError, StopStandard)
 PROXY_METHOD0(void, StopInternal)
-PROXY_METHOD1(RTCError, SetCodecPreferences, ArrayView<RtpCodecCapability>)
+PROXY_METHOD1(RTCError, SetCodecPreferences, std::span<RtpCodecCapability>)
 PROXY_CONSTMETHOD0(std::vector<RtpCodecCapability>, codec_preferences)
 PROXY_CONSTMETHOD0(std::vector<RtpHeaderExtensionCapability>,
                    GetHeaderExtensionsToNegotiate)
@@ -516,7 +516,7 @@ PROXY_CONSTMETHOD0(std::vector<RtpHeaderExtensionCapability>,
                    GetNegotiatedHeaderExtensions)
 PROXY_METHOD1(RTCError,
               SetHeaderExtensionsToNegotiate,
-              ArrayView<const RtpHeaderExtensionCapability>)
+              std::span<const RtpHeaderExtensionCapability>)
 END_PROXY_MAP(RtpTransceiver)
 
 }  // namespace webrtc
diff --git a/pc/rtp_transport.cc b/pc/rtp_transport.cc
index 5dcea00104..c23046f51c 100644
--- a/pc/rtp_transport.cc
+++ b/pc/rtp_transport.cc
@@ -19,7 +19,6 @@
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/sequence_checker.h"
 #include "api/task_queue/pending_task_safety_flag.h"
 #include "api/task_queue/task_queue_base.h"
diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc
index 020aa11eee..10f2f13fc1 100644
--- a/pc/sdp_offer_answer.cc
+++ b/pc/sdp_offer_answer.cc
@@ -20,6 +20,7 @@
 #include <optional>
 #include <queue>
 #include <set>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
@@ -31,7 +32,6 @@
 #include "absl/strings/str_cat.h"
 #include "absl/strings/str_split.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/candidate.h"
 #include "api/crypto/crypto_options.h"
 #include "api/environment/environment.h"
@@ -285,7 +285,7 @@ std::string GetSetDescriptionErrorMessage(ContentSource source,
   return oss.Release();
 }
 
-std::string GetStreamIdsString(ArrayView<const std::string> stream_ids) {
+std::string GetStreamIdsString(std::span<const std::string> stream_ids) {
   std::string output = "streams=[";
   const char* separator = "";
   for (const auto& stream_id : stream_ids) {
diff --git a/pc/srtp_session.cc b/pc/srtp_session.cc
index 361c924485..def6bfab9b 100644
--- a/pc/srtp_session.cc
+++ b/pc/srtp_session.cc
@@ -15,7 +15,6 @@
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/field_trials_view.h"
 #include "modules/rtp_rtcp/source/rtp_util.h"
 #include "rtc_base/buffer.h"
diff --git a/pc/srtp_session_unittest.cc b/pc/srtp_session_unittest.cc
index a9ab7af918..743fd12d18 100644
--- a/pc/srtp_session_unittest.cc
+++ b/pc/srtp_session_unittest.cc
@@ -13,9 +13,9 @@
 #include <cstdint>
 #include <cstring>
 #include <limits>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/field_trials.h"
 #include "media/base/fake_rtp.h"
 #include "pc/test/srtp_test_util.h"
@@ -215,7 +215,7 @@ TEST_F(SrtpSessionTest, TestReplay) {
 
   // Initial sequence number.
   SetBE16(
-      ArrayView<uint8_t>(rtp_packet_.MutableData<uint8_t>(), rtp_packet_.size())
+      std::span<uint8_t>(rtp_packet_.MutableData<uint8_t>(), rtp_packet_.size())
           .subspan(2, 2),
       seqnum_big);
   EXPECT_TRUE(s1_.ProtectRtp(rtp_packet_));
@@ -223,7 +223,7 @@ TEST_F(SrtpSessionTest, TestReplay) {
 
   // Replay within the 1024 window should succeed.
   SetBE16(
-      ArrayView<uint8_t>(rtp_packet_.MutableData<uint8_t>(), rtp_packet_.size())
+      std::span<uint8_t>(rtp_packet_.MutableData<uint8_t>(), rtp_packet_.size())
           .subspan(2, 2),
       seqnum_big - replay_window + 1);
   EXPECT_TRUE(s1_.ProtectRtp(rtp_packet_));
@@ -231,7 +231,7 @@ TEST_F(SrtpSessionTest, TestReplay) {
 
   // Replay out side of the 1024 window should fail.
   SetBE16(
-      ArrayView<uint8_t>(rtp_packet_.MutableData<uint8_t>(), rtp_packet_.size())
+      std::span<uint8_t>(rtp_packet_.MutableData<uint8_t>(), rtp_packet_.size())
           .subspan(2, 2),
       seqnum_big - replay_window - 1);
   EXPECT_FALSE(s1_.ProtectRtp(rtp_packet_));
@@ -239,14 +239,14 @@ TEST_F(SrtpSessionTest, TestReplay) {
 
   // Increment sequence number to a small number.
   SetBE16(
-      ArrayView<uint8_t>(rtp_packet_.MutableData<uint8_t>(), rtp_packet_.size())
+      std::span<uint8_t>(rtp_packet_.MutableData<uint8_t>(), rtp_packet_.size())
           .subspan(2, 2),
       seqnum_small);
   EXPECT_TRUE(s1_.ProtectRtp(rtp_packet_));
 
   // Replay around 0 but out side of the 1024 window should fail.
   SetBE16(
-      ArrayView<uint8_t>(rtp_packet_.MutableData<uint8_t>(), rtp_packet_.size())
+      std::span<uint8_t>(rtp_packet_.MutableData<uint8_t>(), rtp_packet_.size())
           .subspan(2, 2),
       kMaxSeqnum + seqnum_small - replay_window - 1);
   EXPECT_FALSE(s1_.ProtectRtp(rtp_packet_));
@@ -254,7 +254,7 @@ TEST_F(SrtpSessionTest, TestReplay) {
 
   // Replay around 0 but within the 1024 window should succeed.
   for (uint16_t seqnum = 65000; seqnum < 65003; ++seqnum) {
-    SetBE16(ArrayView<uint8_t>(rtp_packet_.MutableData<uint8_t>(),
+    SetBE16(std::span<uint8_t>(rtp_packet_.MutableData<uint8_t>(),
                                rtp_packet_.size())
                 .subspan(2, 2),
             seqnum);
@@ -268,7 +268,7 @@ TEST_F(SrtpSessionTest, TestReplay) {
   // number in libsrtp, eventually the new sequence number would go out side
   // of the window.
   SetBE16(
-      ArrayView<uint8_t>(rtp_packet_.MutableData<uint8_t>(), rtp_packet_.size())
+      std::span<uint8_t>(rtp_packet_.MutableData<uint8_t>(), rtp_packet_.size())
           .subspan(2, 2),
       seqnum_small + 1);
   EXPECT_TRUE(s1_.ProtectRtp(rtp_packet_));
diff --git a/pc/srtp_transport.cc b/pc/srtp_transport.cc
index 3eaf514fd3..ae843d6110 100644
--- a/pc/srtp_transport.cc
+++ b/pc/srtp_transport.cc
@@ -15,7 +15,6 @@
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/field_trials_view.h"
 #include "api/units/timestamp.h"
 #include "call/rtp_demuxer.h"
diff --git a/pc/srtp_transport_unittest.cc b/pc/srtp_transport_unittest.cc
index 3ef3545876..20a7c458fb 100644
--- a/pc/srtp_transport_unittest.cc
+++ b/pc/srtp_transport_unittest.cc
@@ -14,10 +14,10 @@
 #include <cstring>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/field_trials.h"
 #include "api/transport/ecn_marking.h"
 #include "api/units/timestamp.h"
@@ -113,7 +113,7 @@ class SrtpTransportTest : public ::testing::Test {
     memcpy(rtp_packet_data, kPcmuFrame, rtp_len);
     // In order to be able to run this test function multiple times we can not
     // use the same sequence number twice. Increase the sequence number by one.
-    SetBE16(ArrayView<uint8_t>(rtp_packet_buffer).subspan(2),
+    SetBE16(std::span<uint8_t>(rtp_packet_buffer).subspan(2),
             ++sequence_number_);
     CopyOnWriteBuffer rtp_packet1to2(rtp_packet_data, rtp_len, packet_size);
     CopyOnWriteBuffer rtp_packet2to1(rtp_packet_data, rtp_len, packet_size);
@@ -215,7 +215,7 @@ class SrtpTransportTest : public ::testing::Test {
     memcpy(rtp_packet_data, kPcmuFrameWithExtensions, rtp_len);
     // In order to be able to run this test function multiple times we can not
     // use the same sequence number twice. Increase the sequence number by one.
-    SetBE16(ArrayView<uint8_t>(rtp_packet_buffer).subspan(2),
+    SetBE16(std::span<uint8_t>(rtp_packet_buffer).subspan(2),
             ++sequence_number_);
     CopyOnWriteBuffer rtp_packet1to2(rtp_packet_data, rtp_len, packet_size);
     CopyOnWriteBuffer rtp_packet2to1(rtp_packet_data, rtp_len, packet_size);
diff --git a/pc/track_media_info_map.cc b/pc/track_media_info_map.cc
index 9cf2a9c641..7150f34b16 100644
--- a/pc/track_media_info_map.cc
+++ b/pc/track_media_info_map.cc
@@ -13,10 +13,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 
-#include "api/array_view.h"
 #include "api/media_types.h"
 #include "api/rtp_parameters.h"
 #include "media/base/media_channel.h"
@@ -40,7 +40,7 @@ const V* FindAddressOrNull(const flat_map<K, V>& map, const K& key) {
 }
 
 flat_map<uint32_t, int> GetSenderAttachmentIds(
-    ArrayView<const TrackMediaInfoMap::RtpSenderSignalInfo> senders,
+    std::span<const TrackMediaInfoMap::RtpSenderSignalInfo> senders,
     MediaType media_type) {
   flat_map<uint32_t, int> result;
   for (const auto& sender : senders) {
@@ -53,8 +53,8 @@ flat_map<uint32_t, int> GetSenderAttachmentIds(
 }
 
 flat_map<uint32_t, int> GetReceiverAttachmentIds(
-    ArrayView<const TrackMediaInfoMap::RtpReceiverSignalInfo> receivers,
-    ArrayView<const RtpParameters> receiver_parameters,
+    std::span<const TrackMediaInfoMap::RtpReceiverSignalInfo> receivers,
+    std::span<const RtpParameters> receiver_parameters,
     MediaType media_type) {
   RTC_DCHECK_EQ(receivers.size(), receiver_parameters.size());
   flat_map<uint32_t, int> result;
@@ -73,8 +73,8 @@ flat_map<uint32_t, int> GetReceiverAttachmentIds(
 }
 
 flat_map<uint32_t, std::string> GetReceiverTrackIds(
-    ArrayView<const TrackMediaInfoMap::RtpReceiverSignalInfo> receivers,
-    ArrayView<const RtpParameters> receiver_parameters,
+    std::span<const TrackMediaInfoMap::RtpReceiverSignalInfo> receivers,
+    std::span<const RtpParameters> receiver_parameters,
     MediaType media_type) {
   RTC_DCHECK_EQ(receivers.size(), receiver_parameters.size());
   flat_map<uint32_t, std::string> result;
@@ -131,9 +131,9 @@ flat_map<uint32_t, const VideoSenderInfo*> GetVideoSenderInfos(
 TrackMediaInfoMap::TrackMediaInfoMap(
     std::optional<VoiceMediaInfo> voice_media_info,
     std::optional<VideoMediaInfo> video_media_info,
-    ArrayView<const RtpSenderSignalInfo> senders,
-    ArrayView<const RtpReceiverSignalInfo> receivers,
-    ArrayView<const RtpParameters> receiver_parameters)
+    std::span<const RtpSenderSignalInfo> senders,
+    std::span<const RtpReceiverSignalInfo> receivers,
+    std::span<const RtpParameters> receiver_parameters)
     : voice_media_info_(std::move(voice_media_info)),
       video_media_info_(std::move(video_media_info)),
       audio_sender_attachment_id_by_ssrc_(
diff --git a/pc/track_media_info_map.h b/pc/track_media_info_map.h
index 64013015ac..8e63aa8143 100644
--- a/pc/track_media_info_map.h
+++ b/pc/track_media_info_map.h
@@ -13,9 +13,9 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 
-#include "api/array_view.h"
 #include "api/media_types.h"
 #include "api/rtp_parameters.h"
 #include "media/base/media_channel.h"
@@ -45,9 +45,9 @@ class TrackMediaInfoMap {
 
   TrackMediaInfoMap(std::optional<VoiceMediaInfo> voice_media_info,
                     std::optional<VideoMediaInfo> video_media_info,
-                    ArrayView<const RtpSenderSignalInfo> senders,
-                    ArrayView<const RtpReceiverSignalInfo> receivers,
-                    ArrayView<const RtpParameters> receiver_parameters);
+                    std::span<const RtpSenderSignalInfo> senders,
+                    std::span<const RtpReceiverSignalInfo> receivers,
+                    std::span<const RtpParameters> receiver_parameters);
 
   const std::optional<VoiceMediaInfo>& voice_media_info() const {
     return voice_media_info_;

From 99928704b940deb64a261c38de628ff0e528568d Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Fri, 6 Mar 2026 14:02:49 -0800
Subject: [PATCH 226/472] Roll chromium_revision caadb00f3d..c22ce439a0
 (1594862:1595685)

Change log: https://chromium.googlesource.com/chromium/src/+log/caadb00f3d..c22ce439a0
Full diff: https://chromium.googlesource.com/chromium/src/+/caadb00f3d..c22ce439a0

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/92e11019ed..4e55723cd7
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/8ae867bf6b..8da238cb7e
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/d655a043c9..cbc3c0cf69
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/418b3c14d5..0c56ed891b
* src/third_party/androidx/cipd: cbyKZsfMFZDYX5M7c5y-F8b2kOcy1zhCj75x1vHVtfMC..G93sLsFQ30pYjUz4kY55FggeIFragvakLDCkWrLtbOEC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/b8b7f30825..5bf54f4ac7
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/ab750b7590..d903007eb7
* src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/a18fb0c5f8..568f6a9253
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/aceafdaa59..4ce8ba39a3
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/265496e862..d8c8ee6fb2
* src/third_party/freetype/src: https://chromium.googlesource.com/chromium/src/third_party/freetype2.git/+log/d262bd978c..aa8ca27f9b
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/3cae447548..a18875d1df
* src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/e490aa627e..4e2524f6ae
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/f6d82460ec..7d3fdf89c3
* src/third_party/r8/cipd: CQBWQVHNMmNBTkKhqA-Rnel-94ivC4c9jRROa8bvIUkC..CoaQUbD7pDX3OuETi_O5wkv_Y198RVIUaSobSEx9zfoC
* src/third_party/rust-toolchain: Linux_x64/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-1-llvmorg-23-init-5669-g8a0be0bc.tar.xz,Mac/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-1-llvmorg-23-init-5669-g8a0be0bc.tar.xz,Mac_arm64/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-1-llvmorg-23-init-5669-g8a0be0bc.tar.xz,Win/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-1-llvmorg-23-init-5669-g8a0be0bc.tar.xz..Linux_x64/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-2-llvmorg-23-init-5669-g8a0be0bc.tar.xz,Mac/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-2-llvmorg-23-init-5669-g8a0be0bc.tar.xz,Mac_arm64/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-2-llvmorg-23-init-5669-g8a0be0bc.tar.xz,Win/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-2-llvmorg-23-init-5669-g8a0be0bc.tar.xz
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/158b0470d7..6870c91a2b
DEPS diff: https://chromium.googlesource.com/chromium/src/+/caadb00f3d..c22ce439a0/DEPS

BUG=None

Change-Id: I91d952480dc10a21367bcf049ce8c60417588257
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/454440
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47079}
---
 DEPS | 66 ++++++++++++++++++++++++++++++------------------------------
 1 file changed, 33 insertions(+), 33 deletions(-)

diff --git a/DEPS b/DEPS
index fb1b9d1bbf..db87646089 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': 'caadb00f3d641dd7e2deed7899d6a0af1978ec4d',
+  'chromium_revision': 'c22ce439a0c6903da40a102f8e338f114b105eca',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@92e11019ed747d0d2b79c813862a7ff0c2d2f9d3',
+    'https://chromium.googlesource.com/chromium/src/build@4e55723cd763c38199eae267bb0f933445d8e44a',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@fef8d6947fa54202e8a00c58d5ac8fd15780bc33',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@8ae867bf6b41a4bc727ef8f17160d8d6e1a5362b',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@8da238cb7ee4beb61d8a272d9c2c0da7e6909c79',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@d655a043c946d1ff38c10d912541a3732916af29',
+    'https://chromium.googlesource.com/chromium/src/testing@cbc3c0cf69aa7a8b59230216a2fb97f15b1b10c4',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@418b3c14d55646854e8b4170fb4dfa80813a289b',
+    'https://chromium.googlesource.com/chromium/src/third_party@0c56ed891bcd96446651f4021878ae7c9b904cd8',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -298,31 +298,31 @@ deps = {
     'bucket': 'chromium-browser-clang',
     'objects': [
       {
-        'object_name': 'Linux_x64/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-1-llvmorg-23-init-5669-g8a0be0bc.tar.xz',
-        'sha256sum': 'ced6078ebd927bc45de379e645b0f0a8ad61ef751c06eb107ac4bc083daab9b4',
-        'size_bytes': 270674956,
-        'generation': 1772218383559628,
+        'object_name': 'Linux_x64/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-2-llvmorg-23-init-5669-g8a0be0bc.tar.xz',
+        'sha256sum': '71ea8d8f981d1093ac30e884aefca789962a2eb6e0c31dcddc6fdc83a33c5bd6',
+        'size_bytes': 270756972,
+        'generation': 1772730062708394,
         'condition': 'host_os == "linux" and non_git_source',
       },
       {
-        'object_name': 'Mac/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-1-llvmorg-23-init-5669-g8a0be0bc.tar.xz',
-        'sha256sum': 'ec05e6e8d011c8f576df228b7c09d169f6eb8627b0241e17e779b0fd1ee9a869',
-        'size_bytes': 258393452,
-        'generation': 1772218385203632,
+        'object_name': 'Mac/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-2-llvmorg-23-init-5669-g8a0be0bc.tar.xz',
+        'sha256sum': '434ada87593ad29df8eb6bb266d37edcdc7380d1bafbf907f7d39c22d5739084',
+        'size_bytes': 258367196,
+        'generation': 1772730064822309,
         'condition': 'host_os == "mac" and host_cpu == "x64"',
       },
       {
-        'object_name': 'Mac_arm64/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-1-llvmorg-23-init-5669-g8a0be0bc.tar.xz',
-        'sha256sum': '6d7cde3b7c7a5286a87f9a60aa8c9f0928555a39c8b71b9d039cbf12a805fe2b',
-        'size_bytes': 241760428,
-        'generation': 1772218386869177,
+        'object_name': 'Mac_arm64/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-2-llvmorg-23-init-5669-g8a0be0bc.tar.xz',
+        'sha256sum': 'c8e89b991e325f8bfd64fc7a04cfa0eb0b7a953af01fc0a23f0a4b1c1dc34280',
+        'size_bytes': 241717196,
+        'generation': 1772730066996695,
         'condition': 'host_os == "mac" and host_cpu == "arm64"',
       },
       {
-        'object_name': 'Win/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-1-llvmorg-23-init-5669-g8a0be0bc.tar.xz',
-        'sha256sum': '8146753e3451a66e1d4dc9e73af29305b5ea2107fae88739880ddf7a7635ffb6',
-        'size_bytes': 409426524,
-        'generation': 1772218388518369,
+        'object_name': 'Win/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-2-llvmorg-23-init-5669-g8a0be0bc.tar.xz',
+        'sha256sum': '2a2463ffc005f0c98be420203bd5a0d5b54b1b6669bb2a9d491cfcab2338f851',
+        'size_bytes': 409485148,
+        'generation': 1772730069048739,
         'condition': 'host_os == "win"',
       },
     ],
@@ -331,13 +331,13 @@ deps = {
   'src/third_party/clang-format/script':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@c2725e0622e1a86d55f14514f2177a39efea4a0e',
   'src/third_party/compiler-rt/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@a18fb0c5f8c7b5fc6996dfc77b3fb3df92ff549d',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@568f6a925307495d6ba838a68d0dfe7ab375c946',
   'src/third_party/libc++/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@7ab65651aed6802d2599dcb7a73b1f82d5179d05',
   'src/third_party/libc++abi/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@8f11bb1d4438d0239d0dfc1bd9456a9f31629dda',
   'src/third_party/llvm-libc/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@e490aa627ee45c36557b933ac659a8f04a26c758',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@4e2524f6aecac8c50998cd7b673d912239d0e54e',
   'src/third_party/libunwind/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@78884e23fe39cf5cc6987ea188a9b802d65a21c9',
 
@@ -488,11 +488,11 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@b8b7f308253df4a65f709a0f17d37c89dbcb9111',
+    'https://boringssl.googlesource.com/boringssl.git@5bf54f4ac7f73e8cc625d1292a39edd508f64e2f',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@8be0e3114685fcc1589561067282edf75ea1259a',
   'src/third_party/catapult':
-    'https://chromium.googlesource.com/catapult.git@ab750b7590e913a2dcf493fae9c93d0b649f5552',
+    'https://chromium.googlesource.com/catapult.git@d903007eb7787e17c1314f1fdb4787c9a1f7c7ec',
   'src/third_party/ced/src': {
     'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5',
   },
@@ -505,13 +505,13 @@ deps = {
   'src/third_party/crc32c/src':
     'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558',
   'src/third_party/depot_tools':
-    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@aceafdaa59d7f59397abf1812298e065f39db958',
+    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@4ce8ba39a3488397a2d1494f167020f21de502f3',
   'src/third_party/ffmpeg':
     'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@946d97db8d906277085e361892b7efda5152e2f1',
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@3cae4475487e1715235bd037acff6214ac16f10c',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@a18875d1df6a3fdac2cd7339a4fdb6aff29db89e',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -519,7 +519,7 @@ deps = {
       'condition': 'checkout_linux',
   },
   'src/third_party/freetype/src':
-    'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@d262bd978c3ea303289153dba1ae8a6dc4ac747a',
+    'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@aa8ca27f9b48bbb99f02d605b7dbf1d6a016cbcb',
   'src/third_party/harfbuzz-ng/src':
     'https://chromium.googlesource.com/external/github.com/harfbuzz/harfbuzz.git@fa2908bf16d2ccd6623f4d575455fea72a1a722b',
   'src/third_party/google_benchmark/src': {
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'f6d82460ec55f78526aa05aca7d5e9bed3250244',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '7d3fdf89c361951a421fb45ff91b1f74768137ac',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -644,7 +644,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/r8',
-              'version': 'CQBWQVHNMmNBTkKhqA-Rnel-94ivC4c9jRROa8bvIUkC',
+              'version': 'CoaQUbD7pDX3OuETi_O5wkv_Y198RVIUaSobSEx9zfoC',
           },
       ],
       'condition': 'checkout_android',
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@158b0470d7cd0b4a82a6b46148c2d3cedb1d5566',
+    'https://chromium.googlesource.com/chromium/src/tools@6870c91a2b73cae054a5e00f802b3108bc959619',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'cbyKZsfMFZDYX5M7c5y-F8b2kOcy1zhCj75x1vHVtfMC',
+          'version': 'G93sLsFQ30pYjUz4kY55FggeIFragvakLDCkWrLtbOEC',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -864,7 +864,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7364b490b5f78d58efe23ea76e74210fd6c3c76f',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '265496e862d809be98b002dea70acf1a266aa1a0',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + 'd8c8ee6fb2cfe1af53b020ff10a70fbd9b47ccad',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From fda4e0da0ccc8b487eb3c8bd701c353b32353288 Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Fri, 6 Mar 2026 20:05:02 -0800
Subject: [PATCH 227/472] Update WebRTC code version (2026-03-07T04:04:41).

Bug: None
Change-Id: I4c208b2f95b098e32a20cea2460c38824369d497
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/454520
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47080}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index 64a5a9ac3f..fa5aebef55 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-06T04:06:41";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-07T04:04:41";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From 8223963924188cfb30dc05f777d51962abe75435 Mon Sep 17 00:00:00 2001
From: Philipp Hancke <philipp.hancke@googlemail.com>
Date: Mon, 2 Mar 2026 12:14:17 +0100
Subject: [PATCH 228/472] abs-send-time: clarify "close to the metal" as "after
 pacing"

in the past this was interpreted as "after encryption and in
the network process" but that did not make much difference.

Doing this after pacing is still a good idea

Bug: webrtc:487743907
Change-Id: I3b14225c29c210db0b12b54ca51814fe2836e1e8
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452960
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#47081}
---
 docs/native-code/rtp-hdrext/abs-send-time/README.md | 4 +++-
 1 file changed, 3 insertions(+), 1 deletion(-)

diff --git a/docs/native-code/rtp-hdrext/abs-send-time/README.md b/docs/native-code/rtp-hdrext/abs-send-time/README.md
index 86c3c733dc..ab34492cc8 100644
--- a/docs/native-code/rtp-hdrext/abs-send-time/README.md
+++ b/docs/native-code/rtp-hdrext/abs-send-time/README.md
@@ -26,6 +26,8 @@ Relation to NTP timestamps: abs_send_time_24 = (ntp_timestamp_64 >> 14) &
 0x00ffffff ; NTP timestamp is 32 bits for whole seconds, 32 bits fraction of
 second.
 
-Notes: Packets are time stamped when going out, preferably close to metal.
+Notes: Packets are time stamped as close as possible to the time the packet is
+sent on the wire. The current implementation allocates space before pacing
+and stamps them after pacing and before encryption.
 Intermediate RTP relays (entities possibly altering the stream) should remove
 the extension or set its own timestamp.

From 1d40b86162d96c95f9624c60633c3f3890a6d4a5 Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Fri, 6 Mar 2026 07:53:38 +0000
Subject: [PATCH 229/472] Cleanup: Remove suppress_plan_b_deprecation config

This is no longer used to suppress deprecations in tests,
and was done in an inappropriate way anyway.

Bug: webrtc:467984837
Change-Id: Ie453ec4e5e63b9dad15d222a6d1655dc8978c6e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/454200
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47082}
---
 BUILD.gn | 8 --------
 1 file changed, 8 deletions(-)

diff --git a/BUILD.gn b/BUILD.gn
index 654077f303..cd4379fb3f 100644
--- a/BUILD.gn
+++ b/BUILD.gn
@@ -132,14 +132,6 @@ config("absl_flags_configs") {
   defines = [ "ABSL_FLAGS_STRIP_NAMES=0" ]
 }
 
-config("suppress_plan_b_deprecation") {
-  if (is_clang) {
-    cflags = [ "-Wno-deprecated-declarations" ]
-  } else if (is_win && !is_clang) {
-    cflags = [ "/wd4996" ]
-  }
-}
-
 config("library_impl_config") {
   # Build targets that contain WebRTC implementation need this macro to
   # be defined in order to correctly export symbols when is_component_build

From 7962f775089e67780c29381f086485d810af1385 Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Mon, 9 Mar 2026 09:07:32 +0100
Subject: [PATCH 230/472] Use clock from webrtc::Environment in java
 TimestamplAligner
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Move function to query webrtc time from static function in
TimestampAligner to local in Environment

Reland of https://webrtc-review.googlesource.com/c/src/+/452943

Bug: webrtc:42223992
Change-Id: Ia1c63045a1162a35336979a8af60e71a52c9b309
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453981
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Zoé Lepaul <zlep@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47083}
---
 sdk/android/BUILD.gn                          | 43 +++++++++++++++----
 sdk/android/api/org/webrtc/Environment.java   | 10 ++++-
 .../api/org/webrtc/TimestampAligner.java      | 26 +++++++----
 sdk/android/src/jni/environment.cc            |  4 ++
 sdk/android/src/jni/timestamp_aligner.cc      | 41 ++++++++++++------
 5 files changed, 95 insertions(+), 29 deletions(-)

diff --git a/sdk/android/BUILD.gn b/sdk/android/BUILD.gn
index b593a1adc2..f709c40ffc 100644
--- a/sdk/android/BUILD.gn
+++ b/sdk/android/BUILD.gn
@@ -272,6 +272,7 @@ if (is_android) {
 
     deps = [
       ":base_java",
+      ":environment_java",
       ":generated_video_egl_jni_java",
       ":generated_video_jni_java",
       ":video_api_java",
@@ -701,8 +702,10 @@ if (current_os == "linux" || is_android) {
       "src/jni/yuv_helper.cc",
     ]
 
+    allow_poison = [ "environment_construction" ]
     deps = [
       ":base_jni",
+      ":environment_jni",
       ":generated_video_jni",
       ":internal_jni",
       ":native_api_jni",
@@ -932,7 +935,10 @@ if (current_os == "linux" || is_android) {
 
   rtc_library("default_video_codec_factory_jni") {
     visibility = [ "*" ]
-    allow_poison = [ "software_video_codecs" ]
+    allow_poison = [
+      "software_video_codecs",
+      "environment_construction",
+    ]
     deps = [
       ":swcodecs_jni",
       ":video_jni",
@@ -941,7 +947,10 @@ if (current_os == "linux" || is_android) {
 
   rtc_library("libvpx_vp8_jni") {
     visibility = [ "*" ]
-    allow_poison = [ "software_video_codecs" ]
+    allow_poison = [
+      "software_video_codecs",
+      "environment_construction",
+    ]
     sources = [ "src/jni/vp8_codec.cc" ]
     deps = [
       ":base_jni",
@@ -955,7 +964,10 @@ if (current_os == "linux" || is_android) {
 
   rtc_library("libvpx_vp9_jni") {
     visibility = [ "*" ]
-    allow_poison = [ "software_video_codecs" ]
+    allow_poison = [
+      "software_video_codecs",
+      "environment_construction",
+    ]
     sources = [ "src/jni/vp9_codec.cc" ]
     deps = [
       ":base_jni",
@@ -969,7 +981,10 @@ if (current_os == "linux" || is_android) {
 
   rtc_library("libaom_av1_encoder_jni") {
     visibility = [ "*" ]
-    allow_poison = [ "software_video_codecs" ]
+    allow_poison = [
+      "software_video_codecs",
+      "environment_construction",
+    ]
     sources = [ "src/jni/libaom_av1_encoder.cc" ]
     deps = [
       ":base_jni",
@@ -983,7 +998,10 @@ if (current_os == "linux" || is_android) {
 
   rtc_library("dav1d_av1_jni") {
     visibility = [ "*" ]
-    allow_poison = [ "software_video_codecs" ]
+    allow_poison = [
+      "software_video_codecs",
+      "environment_construction",
+    ]
     sources = [ "src/jni/dav1d_codec.cc" ]
     deps = [
       ":base_jni",
@@ -995,7 +1013,10 @@ if (current_os == "linux" || is_android) {
 
   rtc_library("swcodecs_jni") {
     visibility = [ "*" ]
-    allow_poison = [ "software_video_codecs" ]
+    allow_poison = [
+      "software_video_codecs",
+      "environment_construction",
+    ]
     sources = [
       "src/jni/software_video_decoder_factory.cc",
       "src/jni/software_video_encoder_factory.cc",
@@ -1102,7 +1123,10 @@ if (current_os == "linux" || is_android) {
   # objects.
   rtc_library("native_api_codecs") {
     visibility = [ "*" ]
-    allow_poison = [ "audio_codecs" ]  # TODO(bugs.webrtc.org/8396): Remove.
+    allow_poison = [
+      "audio_codecs",  # TODO(bugs.webrtc.org/8396): Remove.
+      "environment_construction",
+    ]
     sources = [
       "native_api/codecs/wrapper.cc",
       "native_api/codecs/wrapper.h",
@@ -1175,7 +1199,10 @@ if (current_os == "linux" || is_android) {
   # video interfaces from their Java equivalents.
   rtc_library("native_api_video") {
     visibility = [ "*" ]
-    allow_poison = [ "audio_codecs" ]  # TODO(bugs.webrtc.org/8396): Remove.
+    allow_poison = [
+      "audio_codecs",  # TODO(bugs.webrtc.org/8396): Remove.
+      "environment_construction",
+    ]
     sources = [
       "native_api/video/video_source.cc",
       "native_api/video/video_source.h",
diff --git a/sdk/android/api/org/webrtc/Environment.java b/sdk/android/api/org/webrtc/Environment.java
index 10749ec4e2..5808776d01 100644
--- a/sdk/android/api/org/webrtc/Environment.java
+++ b/sdk/android/api/org/webrtc/Environment.java
@@ -39,6 +39,14 @@ public final class Environment implements AutoCloseable {
     return webrtcEnv;
   }
 
+  /**
+   * Returns current time as controlled by webrtcEnv. By default this is
+   * normally same as System.nanoTime(), but call this function to be safe.
+   */
+  public long getCurrentTimeNanos() {
+    return nativeCurrentTimeNanos(webrtcEnv);
+  }
+
   @Override
   public void close() {
     nativeFree(webrtcEnv);
@@ -49,6 +57,6 @@ public final class Environment implements AutoCloseable {
   }
 
   private static native long nativeCreate(@Nullable String fieldTrials);
-
+  private static native long nativeCurrentTimeNanos(long webrtcEnv);
   private static native void nativeFree(long webrtcEnv);
 }
diff --git a/sdk/android/api/org/webrtc/TimestampAligner.java b/sdk/android/api/org/webrtc/TimestampAligner.java
index 37c92b31e1..a93f07c37f 100644
--- a/sdk/android/api/org/webrtc/TimestampAligner.java
+++ b/sdk/android/api/org/webrtc/TimestampAligner.java
@@ -12,22 +12,33 @@ package org.webrtc;
 
 /**
  * The TimestampAligner class helps translating camera timestamps into the same timescale as is
- * used by webrtc::TimeNanos(). Some cameras have built in timestamping which is more accurate than
+ * used by webrtc. Some cameras have built in timestamping which is more accurate than
  * reading the system clock, but using a different epoch and unknown clock drift. Frame timestamps
- * in webrtc should use webrtc::TimeNanos (system monotonic time), and this class provides a filter
- * which lets us use the webrtc::TimeNanos timescale, and at the same time take advantage of higher
- * accuracy of the camera clock. This class is a wrapper on top of webrtc::TimestampAligner.
+ * in webrtc should use clock from the webrtc::Environment (usually the system monotonic time), and
+ * this class provides a filter which lets us use the webrtc::Environment timescale, and at the same
+ * time take advantage of higheraccuracy of the camera clock.
+ * This class is a wrapper on top of webrtc::TimestampAligner.
  */
 public class TimestampAligner {
   /**
    * Wrapper around webrtc::TimeNanos(). This is normally same as System.nanoTime(), but call this
    * function to be safe.
    */
+  @Deprecated
   public static long getRtcTimeNanos() {
-    return nativeRtcTimeNanos();
+    return Environment.builder().build().getCurrentTimeNanos();
   }
 
-  private volatile long nativeTimestampAligner = nativeCreateTimestampAligner();
+  private volatile long nativeTimestampAligner;
+
+  @Deprecated
+  public TimestampAligner() {
+    this(Environment.builder().build());
+  }
+
+  public TimestampAligner(Environment webrtcEnv) {
+    nativeTimestampAligner = nativeCreateTimestampAligner(webrtcEnv.ref());
+  }
 
   /**
    * Translates camera timestamps to the same timescale as is used by webrtc::TimeNanos().
@@ -52,8 +63,7 @@ public class TimestampAligner {
     }
   }
 
-  private static native long nativeRtcTimeNanos();
-  private static native long nativeCreateTimestampAligner();
+  private static native long nativeCreateTimestampAligner(long webrtcEnvRef);
   private static native void nativeReleaseTimestampAligner(long timestampAligner);
   private static native long nativeTranslateTimestamp(long timestampAligner, long cameraTimeNs);
 }
diff --git a/sdk/android/src/jni/environment.cc b/sdk/android/src/jni/environment.cc
index a93afe7bb0..bf751e47d1 100644
--- a/sdk/android/src/jni/environment.cc
+++ b/sdk/android/src/jni/environment.cc
@@ -27,6 +27,10 @@ void JNI_Environment_Free(JNIEnv* /*env*/, jlong webrtcEnv) {
   delete reinterpret_cast<Environment*>(webrtcEnv);
 }
 
+jlong JNI_Environment_CurrentTimeNanos(JNIEnv* env, jlong webrtcEnv) {
+  return reinterpret_cast<Environment*>(webrtcEnv)->clock().CurrentTime().ns();
+}
+
 jlong JNI_Environment_Create(JNIEnv* env,
                              const jni_zero::JavaRef<jstring>& fieldTrials) {
   std::unique_ptr<FieldTrialsView> field_trials;
diff --git a/sdk/android/src/jni/timestamp_aligner.cc b/sdk/android/src/jni/timestamp_aligner.cc
index 7226886cb0..12fbd0052f 100644
--- a/sdk/android/src/jni/timestamp_aligner.cc
+++ b/sdk/android/src/jni/timestamp_aligner.cc
@@ -12,34 +12,51 @@
 
 #include <jni.h>
 
-#include "rtc_base/time_utils.h"
+#include <cstdint>
+
+#include "api/environment/environment.h"
 #include "sdk/android/generated_video_jni/TimestampAligner_jni.h"
-#include "sdk/android/src/jni/jni_helpers.h"
+#include "sdk/android/native_api/jni/java_types.h"
 
 namespace webrtc {
 namespace jni {
+namespace {
 
-static jlong JNI_TimestampAligner_RtcTimeNanos(JNIEnv* env) {
-  return TimeNanos();
-}
+class TimestampAlignerWithClock {
+ public:
+  explicit TimestampAlignerWithClock(const Environment& env) : env_(env) {}
 
-static jlong JNI_TimestampAligner_CreateTimestampAligner(JNIEnv* env) {
-  return jlongFromPointer(new TimestampAligner());
+  int64_t TranslateTimestampNs(int64_t camera_time_ns) {
+    return aligner_.TranslateTimestamp(
+               /*capture_time_us=*/camera_time_ns / 1'000,
+               /*system_time_us=*/env_.clock().TimeInMicroseconds()) *
+           1'000;
+  }
+
+ private:
+  const Environment env_;
+  TimestampAligner aligner_;
+};
+
+}  // namespace
+
+static jlong JNI_TimestampAligner_CreateTimestampAligner(JNIEnv* env,
+                                                         jlong webrtcEnvRef) {
+  return NativeToJavaPointer(new TimestampAlignerWithClock(
+      *reinterpret_cast<const Environment*>(webrtcEnvRef)));
 }
 
 static void JNI_TimestampAligner_ReleaseTimestampAligner(
     JNIEnv* env,
     jlong timestamp_aligner) {
-  delete reinterpret_cast<TimestampAligner*>(timestamp_aligner);
+  delete reinterpret_cast<TimestampAlignerWithClock*>(timestamp_aligner);
 }
 
 static jlong JNI_TimestampAligner_TranslateTimestamp(JNIEnv* env,
                                                      jlong timestamp_aligner,
                                                      jlong camera_time_ns) {
-  return reinterpret_cast<TimestampAligner*>(timestamp_aligner)
-             ->TranslateTimestamp(camera_time_ns / kNumNanosecsPerMicrosec,
-                                  TimeMicros()) *
-         kNumNanosecsPerMicrosec;
+  return reinterpret_cast<TimestampAlignerWithClock*>(timestamp_aligner)
+      ->TranslateTimestampNs(camera_time_ns);
 }
 
 }  // namespace jni

From 380b0a977590df0ffd1ebca8af28463ecd06e198 Mon Sep 17 00:00:00 2001
From: Philipp Hancke <philipp.hancke@googlemail.com>
Date: Thu, 5 Mar 2026 13:27:56 +0100
Subject: [PATCH 231/472] pc: add test for non-muxed RTCP
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

which requires two separate SRTP sessions on the transports due to the different DTLS handshakes and keys.

Bug: chromium:41314178
Change-Id: Idb707f8822e52baeb0f9f388804fcbe22a504fda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453821
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47084}
---
 pc/peer_connection_jsep_unittest.cc | 35 +++++++++++++++++++++++++++++
 pc/srtp_transport.h                 |  2 ++
 2 files changed, 37 insertions(+)

diff --git a/pc/peer_connection_jsep_unittest.cc b/pc/peer_connection_jsep_unittest.cc
index 72e54c0b82..104591b0a8 100644
--- a/pc/peer_connection_jsep_unittest.cc
+++ b/pc/peer_connection_jsep_unittest.cc
@@ -31,11 +31,13 @@
 #include "api/rtp_transceiver_direction.h"
 #include "api/rtp_transceiver_interface.h"
 #include "api/scoped_refptr.h"
+#include "api/stats/rtcstats_objects.h"
 #include "api/test/rtc_error_matchers.h"
 #include "media/base/stream_params.h"
 #include "p2p/base/p2p_constants.h"
 #include "p2p/base/transport_info.h"
 #include "pc/media_session.h"
+#include "pc/peer_connection.h"
 #include "pc/peer_connection_wrapper.h"
 #include "pc/session_description.h"
 #include "pc/test/fake_audio_capture_module.h"
@@ -61,6 +63,8 @@ namespace webrtc {
 using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;
 using ::testing::Combine;
 using ::testing::ElementsAre;
+using ::testing::NotNull;
+using ::testing::SizeIs;
 using ::testing::UnorderedElementsAre;
 using ::testing::Values;
 
@@ -2379,6 +2383,37 @@ TEST_F(PeerConnectionJsepTest, BundleOnlySectionDoesNotNeedRtcpMux) {
   EXPECT_TRUE(callee->SetRemoteDescription(std::move(offer)));
 }
 
+TEST_F(PeerConnectionJsepTest, RtcpMuxNotNegotiated) {
+  RTCConfiguration config;
+  // Non-standard as `require` is the only standardized value but we can not
+  // remove support for non-mux.
+  config.rtcp_mux_policy =
+      PeerConnection::RtcpMuxPolicy::kRtcpMuxPolicyNegotiate;
+  auto caller = CreatePeerConnection(config);
+  auto callee = CreatePeerConnection(config);
+  caller->AddTransceiver(MediaType::AUDIO);
+  std::unique_ptr<SessionDescriptionInterface> offer =
+      caller->CreateOfferAndSetAsLocal();
+  ASSERT_THAT(offer, NotNull());
+  ASSERT_THAT(offer->description()->contents(), SizeIs(1));
+  // Remove BUNDLE and rtcp-mux.
+  offer->description()->RemoveGroupByName(GROUP_TYPE_BUNDLE);
+  offer->description()->contents()[0].media_description()->set_rtcp_mux(false);
+
+  EXPECT_TRUE(callee->SetRemoteDescription(std::move(offer)));
+  std::unique_ptr<SessionDescriptionInterface> answer =
+      callee->CreateAnswerAndSetAsLocal();
+  EXPECT_THAT(answer, NotNull());
+  EXPECT_TRUE(caller->SetRemoteDescription(std::move(answer)));
+
+  auto report = caller->GetStats();
+  ASSERT_THAT(report, NotNull());
+
+  std::vector<const RTCTransportStats*> transports =
+      report->GetStatsOfType<RTCTransportStats>();
+  EXPECT_THAT(transports, SizeIs(2));
+}
+
 // This test is a regression test for crbug.com/410960672
 TEST_F(PeerConnectionJsepTest, OfferRollbackRemoveReoffer) {
   auto caller = CreatePeerConnection();
diff --git a/pc/srtp_transport.h b/pc/srtp_transport.h
index b157220dc7..e8b10d78c0 100644
--- a/pc/srtp_transport.h
+++ b/pc/srtp_transport.h
@@ -114,6 +114,8 @@ class SrtpTransport : public RtpTransport {
 
   std::unique_ptr<SrtpSession> send_session_;
   std::unique_ptr<SrtpSession> recv_session_;
+  // Non-muxed RTCP requires different SRTP sessions as it leads to
+  // separate DTLS handshakes.
   std::unique_ptr<SrtpSession> send_rtcp_session_;
   std::unique_ptr<SrtpSession> recv_rtcp_session_;
 

From aade3867c165bd51ac4b15b2ae620bb849e851f6 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Erik=20Spr=C3=A5ng?= <sprang@webrtc.org>
Date: Mon, 9 Mar 2026 11:06:52 +0100
Subject: [PATCH 232/472] Fix L1T3 behavior in VideoCodec tests.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

The VideoCodecTestFixtureImpl wasn't properly configuring the temporal
layer count unless either simulcast or spatial layer count was > 1.

The result was that when e.g. libvpx vp9 was configured in L1T3 mode,
the codec tried to produce temporal layers but the bitrate allocator
put all of the bitrate in TL0 - causing a single stream with the wrong
overall bitrate.

Bug: None
Change-Id: Ib5cd38f7ea59f207d2930ea4acce4655022f08e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/454701
Auto-Submit: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47085}
---
 .../test/videocodec_test_fixture_impl.cc      | 16 ++++++-------
 .../codecs/test/videocodec_test_libvpx.cc     | 24 +++++++++++++++++--
 2 files changed, 29 insertions(+), 11 deletions(-)

diff --git a/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc b/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc
index a9c079c4e5..4260669d5b 100644
--- a/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc
+++ b/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc
@@ -116,15 +116,14 @@ void ConfigureSimulcast(const FieldTrialsView& trials,
   }
 }
 
-void ConfigureSvc(VideoCodec* codec_settings) {
-  RTC_CHECK_EQ(kVideoCodecVP9, codec_settings->codecType);
-
+void ConfigureSvc(VideoCodec* codec_settings,
+                  size_t num_spatial_layers,
+                  size_t num_temporal_layers) {
   const std::vector<SpatialLayer> layers = GetSvcConfig(
       codec_settings->width, codec_settings->height, kDefaultMaxFramerateFps,
-      /*first_active_layer=*/0, codec_settings->VP9()->numberOfSpatialLayers,
-      codec_settings->VP9()->numberOfTemporalLayers,
+      /*first_active_layer=*/0, num_spatial_layers, num_temporal_layers,
       /* is_screen_sharing = */ false);
-  ASSERT_EQ(codec_settings->VP9()->numberOfSpatialLayers, layers.size())
+  ASSERT_EQ(num_spatial_layers, layers.size())
       << "GetSvcConfig returned fewer spatial layers than configured.";
 
   for (size_t i = 0; i < layers.size(); ++i) {
@@ -288,9 +287,8 @@ void VideoCodecTestFixtureImpl::Config::SetCodecSettings(
 
   if (codec_settings.numberOfSimulcastStreams > 1) {
     ConfigureSimulcast(field_trials, &codec_settings);
-  } else if (codec_settings.codecType == kVideoCodecVP9 &&
-             codec_settings.VP9()->numberOfSpatialLayers > 1) {
-    ConfigureSvc(&codec_settings);
+  } else if (num_spatial_layers > 1 || num_temporal_layers > 1) {
+    ConfigureSvc(&codec_settings, num_spatial_layers, num_temporal_layers);
   }
 }
 
diff --git a/modules/video_coding/codecs/test/videocodec_test_libvpx.cc b/modules/video_coding/codecs/test/videocodec_test_libvpx.cc
index cf67e5d3b7..3ed183cf7d 100644
--- a/modules/video_coding/codecs/test/videocodec_test_libvpx.cc
+++ b/modules/video_coding/codecs/test/videocodec_test_libvpx.cc
@@ -304,8 +304,28 @@ TEST(VideoCodecTestLibvpx, VeryLowBitrateVP9) {
   fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
 }
 
-// TODO(marpan): Add temporal layer test for VP9, once changes are in
-// vp9 wrapper for this.
+TEST(VideoCodecTestLibvpx, Vp9L1T3) {
+  auto config = CreateConfig();
+  config.SetCodecSettings(kVp9CodecName, 1, 1, /*num_temporal_layers=*/3, true,
+                          true, false, kCifWidth, kCifHeight);
+
+  auto fixture = CreateVideoCodecTestFixture(config);
+
+  std::vector<RateControlThresholds> rc_thresholds = {
+      {.max_avg_bitrate_mismatch_percent = 60,
+       .max_time_to_reach_target_bitrate_sec = 3,
+       .max_avg_framerate_mismatch_percent = 75,
+       .max_avg_buffer_level_sec = 1,
+       .max_max_key_frame_delay_sec = 0.5,
+       .max_max_delta_frame_delay_sec = 0.4,
+       .max_num_spatial_resizes = 2,
+       .max_num_key_frames = 1}};
+
+  std::vector<RateProfile> rate_profiles = {
+      {.target_kbps = 500, .input_fps = 30, .frame_num = 0}};
+
+  fixture->RunTest(rate_profiles, &rc_thresholds, nullptr, nullptr);
+}
 
 #endif  // defined(RTC_ENABLE_VP9)
 

From 5788235ac856f62f1522d1491c4a8b00dba10c82 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Mon, 9 Mar 2026 10:55:12 +0100
Subject: [PATCH 233/472] Consolidate RTP header extension validation in
 BaseChannel

Refactor WebRtcVideoEngine and WebRtcVoiceEngine to eliminate redundant
RTP header extension mapping and validation. This change shifts the
responsibility for verifying extension IDs and URI consistency from the
media engine level to BaseChannel.

Key changes:
* Remove ValidateRtpExtensions and its manual overlapping checks from
  WebRtcMediaEngine, as these are now centralized.
* Eliminate recv_rtp_extension_map_ and IdentifyExtensions calls from
  video and voice receive channels.
* Update media channels to filter extensions dynamically when
  requesting RTP parameters instead of maintaining a local filtered
  cache.
* Add CheckRtpExtensionValidity to BaseChannel to verify ID ranges,
  duplicates, and URI reassignment.
* Add historical_rtp_header_extensions_ to BaseChannel to track
  previously seen mappings and prevent illegal ID-to-URI reassignments
  during session renegotiation.

Bug: webrtc:42220456
Change-Id: I165e7f9af21b73d6e2ba66ee9367d140f00826eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453661
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47086}
---
 media/engine/webrtc_media_engine.cc          |  59 ---------
 media/engine/webrtc_media_engine.h           |   6 -
 media/engine/webrtc_media_engine_unittest.cc |  62 ----------
 media/engine/webrtc_video_engine.cc          |  73 +----------
 media/engine/webrtc_video_engine.h           |   5 -
 media/engine/webrtc_video_engine_unittest.cc |  50 --------
 media/engine/webrtc_voice_engine.cc          |  78 +-----------
 media/engine/webrtc_voice_engine.h           |   6 +-
 pc/channel.cc                                | 123 ++++++++++++++-----
 pc/channel.h                                 |  21 +++-
 pc/channel_unittest.cc                       | 108 ++++++++++++++++
 11 files changed, 227 insertions(+), 364 deletions(-)

diff --git a/media/engine/webrtc_media_engine.cc b/media/engine/webrtc_media_engine.cc
index 1700892027..96bacd8621 100644
--- a/media/engine/webrtc_media_engine.cc
+++ b/media/engine/webrtc_media_engine.cc
@@ -11,7 +11,6 @@
 #include "media/engine/webrtc_media_engine.h"
 
 #include <algorithm>
-#include <map>
 #include <string>
 #include <vector>
 
@@ -47,63 +46,6 @@ void DiscardRedundantExtensions(
 }
 }  // namespace
 
-bool ValidateRtpExtensions(ArrayView<const RtpExtension> extensions,
-                           ArrayView<const RtpExtension> old_extensions) {
-  bool id_used[1 + RtpExtension::kMaxId] = {false};
-  for (const auto& extension : extensions) {
-    if (extension.id < RtpExtension::kMinId ||
-        extension.id > RtpExtension::kMaxId) {
-      RTC_LOG(LS_ERROR) << "Bad RTP extension ID: " << extension.ToString();
-      return false;
-    }
-    if (id_used[extension.id]) {
-      RTC_LOG(LS_ERROR) << "Duplicate RTP extension ID: "
-                        << extension.ToString();
-      return false;
-    }
-    id_used[extension.id] = true;
-  }
-  // Validate the extension list against the already negotiated extensions.
-  // Re-registering is OK, re-mapping (either same URL at new ID or same
-  // ID used with new URL) is an illegal remap.
-
-  // This is required in order to avoid a crash when registering an
-  // extension. A better structure would use the registered extensions
-  // in the RTPSender. This requires spinning through:
-  //
-  // WebRtcVoiceMediaChannel::::WebRtcAudioSendStream::stream_ (pointer)
-  // AudioSendStream::rtp_rtcp_module_ (pointer)
-  // ModuleRtpRtcpImpl2::rtp_sender_ (pointer)
-  // RtpSenderContext::packet_generator (struct member)
-  // RTPSender::rtp_header_extension_map_ (class member)
-  //
-  // Getting at this seems like a hard slog.
-  if (!old_extensions.empty()) {
-    absl::string_view urimap[1 + RtpExtension::kMaxId];
-    std::map<absl::string_view, int> idmap;
-    for (const auto& old_extension : old_extensions) {
-      urimap[old_extension.id] = old_extension.uri;
-      idmap[old_extension.uri] = old_extension.id;
-    }
-    for (const auto& extension : extensions) {
-      if (!urimap[extension.id].empty() &&
-          urimap[extension.id] != extension.uri) {
-        RTC_LOG(LS_ERROR) << "Extension negotiation failure: " << extension.id
-                          << " was mapped to " << urimap[extension.id]
-                          << " but is proposed changed to " << extension.uri;
-        return false;
-      }
-      const auto& it = idmap.find(extension.uri);
-      if (it != idmap.end() && it->second != extension.id) {
-        RTC_LOG(LS_ERROR) << "Extension negotation failure: " << extension.uri
-                          << " was identified by " << it->second
-                          << " but is proposed changed to " << extension.id;
-        return false;
-      }
-    }
-  }
-  return true;
-}
 
 std::vector<RtpExtension> FilterRtpExtensions(
     const std::vector<RtpExtension>& extensions,
@@ -111,7 +53,6 @@ std::vector<RtpExtension> FilterRtpExtensions(
     bool filter_redundant_extensions,
     const FieldTrialsView& trials) {
   // Don't check against old parameters; this should have been done earlier.
-  RTC_DCHECK(ValidateRtpExtensions(extensions, {}));
   RTC_DCHECK(supported);
   std::vector<RtpExtension> result;
 
diff --git a/media/engine/webrtc_media_engine.h b/media/engine/webrtc_media_engine.h
index 488fe3f3e8..493548e645 100644
--- a/media/engine/webrtc_media_engine.h
+++ b/media/engine/webrtc_media_engine.h
@@ -14,7 +14,6 @@
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/field_trials_view.h"
 #include "api/rtp_parameters.h"
 #include "api/transport/bitrate_settings.h"
@@ -22,11 +21,6 @@
 
 namespace webrtc {
 
-// Verify that extension IDs are within 1-byte extension range and are not
-// overlapping, and that they form a legal change from previously registerd
-// extensions (if any).
-bool ValidateRtpExtensions(ArrayView<const RtpExtension> extennsions,
-                           ArrayView<const RtpExtension> old_extensions);
 
 // Discard any extensions not validated by the 'supported' predicate. Duplicate
 // extensions are removed if 'filter_redundant_extensions' is set, and also any
diff --git a/media/engine/webrtc_media_engine_unittest.cc b/media/engine/webrtc_media_engine_unittest.cc
index 69f5f1b015..5c125b2b82 100644
--- a/media/engine/webrtc_media_engine_unittest.cc
+++ b/media/engine/webrtc_media_engine_unittest.cc
@@ -11,7 +11,6 @@
 #include "media/engine/webrtc_media_engine.h"
 
 #include <string>
-#include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
@@ -66,67 +65,6 @@ bool IsSorted(const std::vector<RtpExtension>& extensions) {
 }
 }  // namespace
 
-TEST(WebRtcMediaEngineTest, ValidateRtpExtensionsEmptyList) {
-  std::vector<RtpExtension> extensions;
-  EXPECT_TRUE(ValidateRtpExtensions(extensions, {}));
-}
-
-TEST(WebRtcMediaEngineTest, ValidateRtpExtensionsAllGood) {
-  std::vector<RtpExtension> extensions = MakeUniqueExtensions();
-  EXPECT_TRUE(ValidateRtpExtensions(extensions, {}));
-}
-
-TEST(WebRtcMediaEngineTest, ValidateRtpExtensionsOutOfRangeId_Low) {
-  std::vector<RtpExtension> extensions = MakeUniqueExtensions();
-  extensions.push_back(RtpExtension("foo", 0));
-  EXPECT_FALSE(ValidateRtpExtensions(extensions, {}));
-}
-
-TEST(WebRtcMediaEngineTest, ValidateRtpExtensionsOutOfRangeIdHigh) {
-  std::vector<RtpExtension> extensions = MakeUniqueExtensions();
-  extensions.push_back(RtpExtension("foo", 256));
-  EXPECT_FALSE(ValidateRtpExtensions(extensions, {}));
-}
-
-TEST(WebRtcMediaEngineTest, ValidateRtpExtensionsOverlappingIdsStartOfSet) {
-  std::vector<RtpExtension> extensions = MakeUniqueExtensions();
-  extensions.push_back(RtpExtension("foo", 1));
-  EXPECT_FALSE(ValidateRtpExtensions(extensions, {}));
-}
-
-TEST(WebRtcMediaEngineTest, ValidateRtpExtensionsOverlappingIdsEndOfSet) {
-  std::vector<RtpExtension> extensions = MakeUniqueExtensions();
-  extensions.push_back(RtpExtension("foo", 255));
-  EXPECT_FALSE(ValidateRtpExtensions(extensions, {}));
-}
-
-TEST(WebRtcMediaEngineTest, ValidateRtpExtensionsEmptyToEmpty) {
-  std::vector<RtpExtension> extensions;
-  EXPECT_TRUE(ValidateRtpExtensions(extensions, extensions));
-}
-
-TEST(WebRtcMediaEngineTest, ValidateRtpExtensionsNoChange) {
-  std::vector<RtpExtension> extensions = MakeUniqueExtensions();
-  EXPECT_TRUE(ValidateRtpExtensions(extensions, extensions));
-}
-
-TEST(WebRtcMediaEngineTest, ValidateRtpExtensionsChangeIdNotUrl) {
-  std::vector<RtpExtension> old_extensions = MakeUniqueExtensions();
-  std::vector<RtpExtension> new_extensions = old_extensions;
-  std::swap(new_extensions[0].id, new_extensions[1].id);
-
-  EXPECT_FALSE(ValidateRtpExtensions(new_extensions, old_extensions));
-}
-
-TEST(WebRtcMediaEngineTest, ValidateRtpExtensionsChangeIdForUrl) {
-  std::vector<RtpExtension> old_extensions = MakeUniqueExtensions();
-  std::vector<RtpExtension> new_extensions = old_extensions;
-  // Change first extension to something not generated by MakeUniqueExtensions
-  new_extensions[0].id = 123;
-
-  EXPECT_FALSE(ValidateRtpExtensions(new_extensions, old_extensions));
-}
-
 TEST(WebRtcMediaEngineTest, FilterRtpExtensionsEmptyList) {
   std::vector<RtpExtension> extensions;
   FieldTrials trials = CreateTestFieldTrials();
diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc
index 7ce882c7ad..f1cd60d8f5 100644
--- a/media/engine/webrtc_video_engine.cc
+++ b/media/engine/webrtc_video_engine.cc
@@ -87,7 +87,6 @@
 #include "media/engine/webrtc_media_engine.h"
 #include "modules/rtp_rtcp/include/receive_statistics.h"
 #include "modules/rtp_rtcp/include/rtcp_statistics.h"
-#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "modules/video_coding/svc/scalability_mode_util.h"
@@ -1075,8 +1074,7 @@ std::vector<VideoCodecSettings> WebRtcVideoSendChannel::SelectSendVideoCodecs(
 bool WebRtcVideoSendChannel::GetChangedSenderParameters(
     const VideoSenderParameters& params,
     ChangedSenderParameters* changed_params) const {
-  if (!ValidateCodecFormats(params.codecs) ||
-      !ValidateRtpExtensions(params.extensions, send_rtp_extensions_)) {
+  if (!ValidateCodecFormats(params.codecs)) {
     return false;
   }
 
@@ -2852,13 +2850,16 @@ RtpParameters WebRtcVideoReceiveChannel::GetRtpReceiverParameters(
     return RtpParameters();
   }
   rtp_params = it->second->GetRtpParameters();
-  rtp_params.header_extensions = recv_rtp_extensions_;
 
   // Add codecs, which any stream is prepared to receive.
   for (const Codec& codec : recv_params_.codecs) {
     rtp_params.codecs.push_back(codec.ToCodecParameters());
   }
 
+  rtp_params.header_extensions = FilterRtpExtensions(
+      recv_params_.extensions, RtpExtension::IsSupportedForVideo, false,
+      env_.field_trials());
+
   return rtp_params;
 }
 
@@ -2885,8 +2886,7 @@ RtpParameters WebRtcVideoReceiveChannel::GetDefaultRtpReceiveParameters()
 bool WebRtcVideoReceiveChannel::GetChangedReceiverParameters(
     const VideoReceiverParameters& params,
     ChangedReceiverParameters* changed_params) const {
-  if (!ValidateCodecFormats(params.codecs) ||
-      !ValidateRtpExtensions(params.extensions, recv_rtp_extensions_)) {
+  if (!ValidateCodecFormats(params.codecs)) {
     return false;
   }
 
@@ -2922,15 +2922,6 @@ bool WebRtcVideoReceiveChannel::GetChangedReceiverParameters(
         std::optional<std::vector<VideoCodecSettings>>(mapped_codecs);
   }
 
-  // Handle RTP header extensions.
-  std::vector<RtpExtension> filtered_extensions =
-      FilterRtpExtensions(params.extensions, RtpExtension::IsSupportedForVideo,
-                          false, env_.field_trials());
-  if (filtered_extensions != recv_rtp_extensions_) {
-    changed_params->rtp_header_extensions =
-        std::optional<std::vector<RtpExtension>>(filtered_extensions);
-  }
-
   int flexfec_payload_type = mapped_codecs.front().flexfec_payload_type;
   if (flexfec_payload_type != recv_flexfec_payload_type_) {
     changed_params->flexfec_payload_type = flexfec_payload_type;
@@ -2959,16 +2950,6 @@ bool WebRtcVideoReceiveChannel::SetReceiverParameters(
                       << *changed_params.flexfec_payload_type;
     recv_flexfec_payload_type_ = *changed_params.flexfec_payload_type;
   }
-  if (changed_params.rtp_header_extensions) {
-    recv_rtp_extensions_ = *changed_params.rtp_header_extensions;
-    call_->network_thread()->PostTask(SafeTask(
-        network_thread_safety_,
-        [this, recv_rtp_extension_map =
-                   RtpHeaderExtensionMap(recv_rtp_extensions_)]() mutable {
-          RTC_DCHECK_RUN_ON(&network_thread_checker_);
-          recv_rtp_extension_map_ = std::move(recv_rtp_extension_map);
-        }));
-  }
   if (changed_params.codec_settings) {
     RTC_DLOG(LS_INFO) << "Changing recv codecs from "
                       << CodecSettingsVectorToString(recv_codecs_) << " to "
@@ -3239,48 +3220,6 @@ void WebRtcVideoReceiveChannel::OnPacketReceived(RtpPacketReceived packet) {
   // depending on configuration set at object initialization.
   RTC_DCHECK_RUN_ON(&network_thread_checker_);
 
-  // TODO(bugs.webrtc.org/7135): The mapping of extension ID (on-packet
-  // representation) to C++ class object (internal representation) is currently
-  // associated with the per-channel object (e.g. WebRtcVideoReceiveChannel).
-  // However, this mapping is more logically associated with per-media-stream
-  // objects (like ModuleRtpRtcpImpl2), which handle specific streams flowing
-  // in one direction (usually a single SSRC).
-  // This mapping should be moved out of the media engines altogether, storing
-  // it instead at the stream level initialized by the Transport. RTP packets
-  // consumed by media channels would thus have header extensions distinguished
-  // by C++ class rather than negotiated header extension ID.
-#if RTC_DCHECK_IS_ON
-  RtpPacketReceived original_packet = packet;
-#endif
-  packet.IdentifyExtensions(recv_rtp_extension_map_);
-#if RTC_DCHECK_IS_ON
-  for (int i = kRtpExtensionNone + 1; i < kRtpExtensionNumberOfExtensions;
-       ++i) {
-    RTPExtensionType type = static_cast<RTPExtensionType>(i);
-    // TODO(bugs.webrtc.org/7135): Only verify if the extension
-    // is registered in either map to avoid unnecessary checks.
-    if (recv_rtp_extension_map_.IsRegistered(type)) {
-      // We only verify equality when the IDs match. This check is
-      // intentionally weak because the transport map (from `original_packet`)
-      // contains the merged extensions for all MIDs in a BUNDLE group,
-      // while the local map only contains extensions for this specific MID.
-      // Therefore, the transport map might contain extensions not present
-      // in the local map, but if both maps contain an ID for a specific
-      // extension type, they must both parse it consistently.
-      if (original_packet.extension_manager().GetId(type) ==
-          recv_rtp_extension_map_.GetId(type)) {
-        bool transport_has = original_packet.HasExtension(type);
-        bool local_has = packet.HasExtension(type);
-        RTC_DCHECK_EQ(transport_has, local_has)
-            << "Extension mapping mismatch for type " << type << ". "
-            << "Transport map has: " << transport_has << " (ID from transport: "
-            << static_cast<int>(original_packet.extension_manager().GetId(type))
-            << "). Local map has: " << local_has << " (ID from local: "
-            << static_cast<int>(recv_rtp_extension_map_.GetId(type)) << ").";
-      }
-    }
-  }
-#endif
   packet.set_payload_type_frequency(kVideoPayloadTypeFrequency);
   if (!packet.arrival_time().IsFinite()) {
     packet.set_arrival_time(env_.clock().CurrentTime());
diff --git a/media/engine/webrtc_video_engine.h b/media/engine/webrtc_video_engine.h
index 3e2e7f2e0b..f0a3412534 100644
--- a/media/engine/webrtc_video_engine.h
+++ b/media/engine/webrtc_video_engine.h
@@ -61,7 +61,6 @@
 #include "media/base/media_config.h"
 #include "media/base/media_engine.h"
 #include "media/base/stream_params.h"
-#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/network/sent_packet.h"
@@ -765,10 +764,6 @@ class WebRtcVideoReceiveChannel : public MediaChannelUtil,
 
   VideoDecoderFactory* const decoder_factory_ RTC_GUARDED_BY(thread_checker_);
   std::vector<VideoCodecSettings> recv_codecs_ RTC_GUARDED_BY(thread_checker_);
-  RtpHeaderExtensionMap recv_rtp_extension_map_
-      RTC_GUARDED_BY(network_thread_checker_);
-  std::vector<RtpExtension> recv_rtp_extensions_
-      RTC_GUARDED_BY(thread_checker_);
   // See reason for keeping track of the FlexFEC payload type separately in
   // comment in WebRtcVideoChannel::ChangedReceiverParameters.
   int recv_flexfec_payload_type_ RTC_GUARDED_BY(thread_checker_);
diff --git a/media/engine/webrtc_video_engine_unittest.cc b/media/engine/webrtc_video_engine_unittest.cc
index 03f25b5a4f..311d7281b0 100644
--- a/media/engine/webrtc_video_engine_unittest.cc
+++ b/media/engine/webrtc_video_engine_unittest.cc
@@ -3354,57 +3354,7 @@ TEST_F(WebRtcVideoChannelTest,
                RtpExtension::kTimestampOffsetUri);
 }
 
-TEST_F(WebRtcVideoChannelTest, SetSendRtpHeaderExtensionsRejectsIncorrectIds) {
-  const int kIncorrectIds[] = {-2, -1, 0, 15, 16};
-  for (int incorrect_id : kIncorrectIds) {
-    send_parameters_.extensions.push_back(
-        RtpExtension(RtpExtension::kTimestampOffsetUri, incorrect_id));
-    EXPECT_FALSE(send_channel_->SetSenderParameters(send_parameters_))
-        << "Bad extension id '" << incorrect_id << "' accepted.";
-  }
-}
 
-TEST_F(WebRtcVideoChannelTest, SetRecvRtpHeaderExtensionsRejectsIncorrectIds) {
-  const int kIncorrectIds[] = {-2, -1, 0, 15, 16};
-  for (int incorrect_id : kIncorrectIds) {
-    recv_parameters_.extensions.push_back(
-        RtpExtension(RtpExtension::kTimestampOffsetUri, incorrect_id));
-    EXPECT_FALSE(receive_channel_->SetReceiverParameters(recv_parameters_))
-        << "Bad extension id '" << incorrect_id << "' accepted.";
-  }
-}
-
-TEST_F(WebRtcVideoChannelTest, SetSendRtpHeaderExtensionsRejectsDuplicateIds) {
-  const int id = 1;
-  send_parameters_.extensions.push_back(
-      RtpExtension(RtpExtension::kTimestampOffsetUri, id));
-  send_parameters_.extensions.push_back(
-      RtpExtension(RtpExtension::kAbsSendTimeUri, id));
-  EXPECT_FALSE(send_channel_->SetSenderParameters(send_parameters_));
-
-  // Duplicate entries are also not supported.
-  send_parameters_.extensions.clear();
-  send_parameters_.extensions.push_back(
-      RtpExtension(RtpExtension::kTimestampOffsetUri, id));
-  send_parameters_.extensions.push_back(send_parameters_.extensions.back());
-  EXPECT_FALSE(send_channel_->SetSenderParameters(send_parameters_));
-}
-
-TEST_F(WebRtcVideoChannelTest, SetRecvRtpHeaderExtensionsRejectsDuplicateIds) {
-  const int id = 1;
-  recv_parameters_.extensions.push_back(
-      RtpExtension(RtpExtension::kTimestampOffsetUri, id));
-  recv_parameters_.extensions.push_back(
-      RtpExtension(RtpExtension::kAbsSendTimeUri, id));
-  EXPECT_FALSE(receive_channel_->SetReceiverParameters(recv_parameters_));
-
-  // Duplicate entries are also not supported.
-  recv_parameters_.extensions.clear();
-  recv_parameters_.extensions.push_back(
-      RtpExtension(RtpExtension::kTimestampOffsetUri, id));
-  recv_parameters_.extensions.push_back(recv_parameters_.extensions.back());
-  EXPECT_FALSE(receive_channel_->SetReceiverParameters(recv_parameters_));
-}
 
 TEST_F(WebRtcVideoChannelTest, OnPacketReceivedIdentifiesExtensions) {
   VideoReceiverParameters parameters = recv_parameters_;
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index d13ce532fa..c8bda9884f 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -84,7 +84,6 @@
 #include "media/engine/webrtc_media_engine.h"
 #include "modules/async_audio_processing/async_audio_processing.h"
 #include "modules/audio_mixer/audio_mixer_impl.h"
-#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "rtc_base/checks.h"
@@ -291,7 +290,6 @@ AudioReceiveStreamInterface::Config BuildReceiveStreamConfig(
     bool enable_non_sender_rtt,
     RtcpMode rtcp_mode,
     const std::vector<std::string>& stream_ids,
-    const std::vector<RtpExtension>& /* extensions */,
     Transport* rtcp_send_transport,
     const scoped_refptr<AudioDecoderFactory>& decoder_factory,
     const std::map<int, SdpAudioFormat>& decoder_map,
@@ -1369,10 +1367,6 @@ bool WebRtcVoiceSendChannel::SetSenderParameters(
     return false;
   }
 
-  if (!ValidateRtpExtensions(params.extensions, send_rtp_extensions_)) {
-    return false;
-  }
-
   if (ExtmapAllowMixed() != params.extmap_allow_mixed) {
     SetExtmapAllowMixed(params.extmap_allow_mixed);
     for (auto& it : send_streams_) {
@@ -2186,26 +2180,11 @@ bool WebRtcVoiceReceiveChannel::SetReceiverParameters(
     return false;
   }
 
-  if (!ValidateRtpExtensions(params.extensions, recv_rtp_extensions_)) {
-    return false;
-  }
-  std::vector<RtpExtension> filtered_extensions =
-      FilterRtpExtensions(params.extensions, RtpExtension::IsSupportedForAudio,
-                          false, env_.field_trials());
-  if (recv_rtp_extensions_ != filtered_extensions) {
-    recv_rtp_extensions_.swap(filtered_extensions);
-    call_->network_thread()->PostTask(SafeTask(
-        network_thread_safety_,
-        [this, recv_rtp_extension_map =
-                   RtpHeaderExtensionMap(recv_rtp_extensions_)]() mutable {
-          RTC_DCHECK_RUN_ON(&network_thread_checker_);
-          recv_rtp_extension_map_ = std::move(recv_rtp_extension_map);
-        }));
-  }
   // RTCP mode, NACK, and receive-side RTT are not configured here because they
   // enable send functionality in the receive channels. This functionality is
   // instead configured using the SetReceiveRtcpMode, SetReceiveNackEnabled, and
   // SetReceiveNonSenderRttEnabled methods.
+  recv_params_ = params;
   return true;
 }
 
@@ -2223,7 +2202,10 @@ RtpParameters WebRtcVoiceReceiveChannel::GetRtpReceiverParameters(
   }
   rtp_params.encodings.emplace_back();
   rtp_params.encodings.back().ssrc = it->second->stream().remote_ssrc();
-  rtp_params.header_extensions = recv_rtp_extensions_;
+
+  rtp_params.header_extensions = FilterRtpExtensions(
+      recv_params_.extensions, RtpExtension::IsSupportedForAudio, false,
+      env_.field_trials());
 
   for (const Codec& codec : recv_codecs_) {
     rtp_params.codecs.push_back(codec.ToCodecParameters());
@@ -2452,8 +2434,7 @@ bool WebRtcVoiceReceiveChannel::AddRecvStream(const StreamParams& sp) {
   // Create a new channel for receiving audio data.
   auto config = BuildReceiveStreamConfig(
       ssrc, recv_nack_enabled_, enable_non_sender_rtt_, recv_rtcp_mode_,
-      sp.stream_ids(), recv_rtp_extensions_, transport(),
-      engine()->decoder_factory_, decoder_map_,
+      sp.stream_ids(), transport(), engine()->decoder_factory_, decoder_map_,
       engine()->audio_jitter_buffer_max_packets_,
       engine()->audio_jitter_buffer_fast_accelerate_,
       engine()->audio_jitter_buffer_min_delay_ms_, unsignaled_frame_decryptor_,
@@ -2604,53 +2585,6 @@ void WebRtcVoiceReceiveChannel::SetFrameDecryptor(
 void WebRtcVoiceReceiveChannel::OnPacketReceived(RtpPacketReceived packet) {
   RTC_DCHECK_RUN_ON(&network_thread_checker_);
 
-  // TODO(bugs.webrtc.org/11993): This code is very similar to what
-  // WebRtcVideoChannel::OnPacketReceived does. For maintainability and
-  // consistency it would be good to move the interaction with
-  // call_->Receiver() to a common implementation and provide a callback on
-  // the worker thread for the exception case (DELIVERY_UNKNOWN_SSRC) and
-  // how retry is attempted.
-
-  // TODO(bugs.webrtc.org/7135): The mapping of extension ID (on-packet
-  // representation) to C++ class object (internal representation) is currently
-  // associated with the per-channel object (e.g. WebRtcVoiceReceiveChannel).
-  // However, this mapping is more logically associated with per-media-stream
-  // objects (like ModuleRtpRtcpImpl2), which handle specific streams flowing
-  // in one direction (usually a single SSRC).
-  // This mapping should be moved out of the media engines altogether, storing
-  // it instead at the stream level initialized by the Transport. RTP packets
-  // consumed by media channels would thus have header extensions distinguished
-  // by C++ class rather than negotiated header extension ID.
-#if RTC_DCHECK_IS_ON
-  RtpPacketReceived original_packet = packet;
-#endif
-  packet.IdentifyExtensions(recv_rtp_extension_map_);
-#if RTC_DCHECK_IS_ON
-  for (int i = kRtpExtensionNone + 1; i < kRtpExtensionNumberOfExtensions;
-       ++i) {
-    RTPExtensionType type = static_cast<RTPExtensionType>(i);
-    if (recv_rtp_extension_map_.IsRegistered(type)) {
-      // We only verify equality when the IDs match. This check is
-      // intentionally weak because the transport map (from `original_packet`)
-      // contains the merged extensions for all MIDs in a BUNDLE group,
-      // while the local map only contains extensions for this specific MID.
-      // Therefore, the transport map might contain extensions not present
-      // in the local map, but if both maps agree on an ID for a specific
-      // extension type, they must both parse it consistently.
-      if (original_packet.extension_manager().GetId(type) ==
-          recv_rtp_extension_map_.GetId(type)) {
-        bool transport_has = original_packet.HasExtension(type);
-        bool local_has = packet.HasExtension(type);
-        RTC_DCHECK_EQ(transport_has, local_has)
-            << "Extension mapping mismatch for type " << type << ". "
-            << "Transport map has: " << transport_has << " (ID from transport: "
-            << static_cast<int>(original_packet.extension_manager().GetId(type))
-            << "). Local map has: " << local_has << " (ID from local: "
-            << static_cast<int>(recv_rtp_extension_map_.GetId(type)) << ").";
-      }
-    }
-  }
-#endif
   if (!packet.arrival_time().IsFinite()) {
     packet.set_arrival_time(env_.clock().CurrentTime());
   }
diff --git a/media/engine/webrtc_voice_engine.h b/media/engine/webrtc_voice_engine.h
index c4facf5890..0100fc405e 100644
--- a/media/engine/webrtc_voice_engine.h
+++ b/media/engine/webrtc_voice_engine.h
@@ -58,7 +58,6 @@
 #include "media/base/media_config.h"
 #include "media/base/media_engine.h"
 #include "media/base/stream_params.h"
-#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/network/sent_packet.h"
@@ -464,9 +463,8 @@ class WebRtcVoiceReceiveChannel final
   class WebRtcAudioReceiveStream;
   std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_
       RTC_GUARDED_BY(worker_thread_);
-  std::vector<RtpExtension> recv_rtp_extensions_ RTC_GUARDED_BY(worker_thread_);
-  RtpHeaderExtensionMap recv_rtp_extension_map_
-      RTC_GUARDED_BY(network_thread_checker_);
+
+  AudioReceiverParameters recv_params_ RTC_GUARDED_BY(worker_thread_);
 
   std::optional<AudioSendStream::Config::SendCodecSpec> send_codec_spec_
       RTC_GUARDED_BY(worker_thread_);
diff --git a/pc/channel.cc b/pc/channel.cc
index 7b31a16b57..2c39a407a8 100644
--- a/pc/channel.cc
+++ b/pc/channel.cc
@@ -11,6 +11,7 @@
 #include "pc/channel.h"
 
 #include <algorithm>
+#include <bitset>
 #include <cstdint>
 #include <memory>
 #include <optional>
@@ -549,17 +550,30 @@ void BaseChannel::OnRtpPacket(const RtpPacketReceived& parsed_packet) {
 
 bool BaseChannel::MaybeUpdateDemuxerAndRtpExtensions_w(
     bool update_demuxer,
-    std::optional<RtpHeaderExtensions> extensions,
+    const RtpHeaderExtensions& extensions,
     std::string& error_desc) {
-  if (extensions) {
-    if (rtp_header_extensions_ == extensions) {
-      extensions.reset();  // No need to update header extensions.
-    } else {
-      rtp_header_extensions_ = *extensions;
+  bool update_extensions = true;
+  if (rtp_header_extensions_ == extensions) {
+    update_extensions = false;  // No need to update header extensions.
+  } else {
+    if (!CheckRtpExtensionValidity(extensions, error_desc)) {
+      return false;
+    }
+    rtp_header_extensions_ = extensions;
+
+    for (const auto& extension : extensions) {
+      if (extension.id == 0)
+        continue;
+      if (absl::c_find_if(historical_rtp_header_extensions_,
+                          [&](const RtpExtension& ext) {
+                            return ext.id == extension.id;
+                          }) == historical_rtp_header_extensions_.end()) {
+        historical_rtp_header_extensions_.push_back(extension);
+      }
     }
   }
 
-  if (!update_demuxer && !extensions)
+  if (!update_demuxer && !update_extensions)
     return true;  // No update needed.
 
   // TODO(bugs.webrtc.org/13536): See if we can do this asynchronously.
@@ -576,8 +590,8 @@ bool BaseChannel::MaybeUpdateDemuxerAndRtpExtensions_w(
       return false;
     }
 
-    if (extensions) {
-      rtp_transport_->RegisterRtpHeaderExtensionMap(mid(), *extensions);
+    if (update_extensions) {
+      rtp_transport_->RegisterRtpHeaderExtensionMap(mid(), extensions);
     }
 
     if (!update_demuxer)
@@ -917,6 +931,48 @@ bool BaseChannel::ClearHandledPayloadTypes() {
   return !was_empty;
 }
 
+bool BaseChannel::CheckRtpExtensionValidity(
+    const RtpHeaderExtensions& extensions,
+    std::string& error_desc) const {
+  std::bitset<1 + RtpExtension::kMaxId> id_used;
+  for (const auto& extension : extensions) {
+    if (extension.id == 0)
+      continue;
+    if (extension.id < RtpExtension::kMinId ||
+        extension.id > RtpExtension::kMaxId) {
+      error_desc = StringFormat("Bad RTP extension ID: %s",
+                                extension.ToString().c_str());
+      return false;
+    }
+    if (id_used[extension.id]) {
+      error_desc = StringFormat("Duplicate RTP extension ID: %s",
+                                extension.ToString().c_str());
+      return false;
+    }
+    id_used[extension.id] = true;
+  }
+
+  for (const auto& new_extension : extensions) {
+    if (new_extension.id == 0)
+      continue;
+    auto it = absl::c_find_if(
+        historical_rtp_header_extensions_,
+        [&](const RtpExtension& ext) { return ext.id == new_extension.id; });
+    if (it != historical_rtp_header_extensions_.end() &&
+        it->uri != new_extension.uri) {
+      error_desc = StringFormat(
+          "Failed to update RTP header extensions for m-section with "
+          "mid='%s'. RTP extension ID reassignment from %s to %s for ID "
+          "%d.",
+          mid().c_str(), it->uri.c_str(), new_extension.uri.c_str(),
+          new_extension.id);
+      return false;
+    }
+  }
+
+  return true;
+}
+
 void BaseChannel::SignalSentPacket_n(const SentPacketInfo& sent_packet) {
   RTC_DCHECK_RUN_ON(network_thread());
   RTC_DCHECK(network_initialized());
@@ -975,7 +1031,11 @@ bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content,
 
   RtpHeaderExtensions header_extensions =
       GetDeduplicatedRtpHeaderExtensions(content->rtp_header_extensions());
-  bool update_header_extensions = true;
+
+  if (!CheckRtpExtensionValidity(header_extensions, error_desc)) {
+    return false;
+  }
+
   // TODO: issues.webrtc.org/383078466 - remove if pushdown on answer is enough.
   media_send_channel()->SetExtmapAllowMixed(content->extmap_allow_mixed());
 
@@ -1030,11 +1090,7 @@ bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content,
   // RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(0);
 
   bool success = MaybeUpdateDemuxerAndRtpExtensions_w(
-      criteria_modified,
-      update_header_extensions
-          ? std::optional<RtpHeaderExtensions>(recv_params.extensions)
-          : std::nullopt,
-      error_desc);
+      criteria_modified, recv_params.extensions, error_desc);
 
   // RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(1);
 
@@ -1052,6 +1108,10 @@ bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content,
                                         &send_params);
   send_params.mid = mid();
 
+  if (!CheckRtpExtensionValidity(send_params.extensions, error_desc)) {
+    return false;
+  }
+
   bool parameters_applied =
       media_send_channel()->SetSenderParameters(send_params);
   if (!parameters_applied) {
@@ -1091,17 +1151,13 @@ bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content,
     return false;
   }
 
+  bool success = true;
   if (type == SdpType::kAnswer || type == SdpType::kPrAnswer) {
-    bool success = MaybeUpdateDemuxerAndRtpExtensions_w(
-        /*update_demuxer=*/false,
-        std::optional<RtpHeaderExtensions>(last_recv_params_.extensions),
-        error_desc);
-    if (!success) {
-      return false;
-    }
+    success = MaybeUpdateDemuxerAndRtpExtensions_w(
+        /*update_demuxer=*/false, last_recv_params_.extensions, error_desc);
   }
 
-  return true;
+  return success;
 }
 
 VideoChannel::VideoChannel(
@@ -1153,12 +1209,15 @@ bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
 
   RtpHeaderExtensions header_extensions =
       GetDeduplicatedRtpHeaderExtensions(content->rtp_header_extensions());
-  bool update_header_extensions = true;
+
+  if (!CheckRtpExtensionValidity(header_extensions, error_desc)) {
+    return false;
+  }
+
   // TODO: issues.webrtc.org/383078466 - remove if pushdown on answer is enough.
   media_send_channel()->SetExtmapAllowMixed(content->extmap_allow_mixed());
 
   VideoReceiverParameters recv_params = last_recv_params_;
-
   MediaChannelParametersFromMediaDescription(
       content, header_extensions,
       RtpTransceiverDirectionHasRecv(content->direction()), &recv_params);
@@ -1219,11 +1278,7 @@ bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
   RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(0);
 
   bool success = MaybeUpdateDemuxerAndRtpExtensions_w(
-      criteria_modified,
-      update_header_extensions
-          ? std::optional<RtpHeaderExtensions>(recv_params.extensions)
-          : std::nullopt,
-      error_desc);
+      criteria_modified, recv_params.extensions, error_desc);
 
   RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(1);
 
@@ -1242,6 +1297,10 @@ bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content,
   send_params.mid = mid();
   send_params.conference_mode = content->conference_mode();
 
+  if (!CheckRtpExtensionValidity(send_params.extensions, error_desc)) {
+    return false;
+  }
+
   VideoReceiverParameters recv_params = last_recv_params_;
 
   // Ensure that there is a matching packetization for each receive codec. If we
@@ -1283,9 +1342,7 @@ bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content,
 
   if (type == SdpType::kAnswer || type == SdpType::kPrAnswer) {
     bool success = MaybeUpdateDemuxerAndRtpExtensions_w(
-        /*update_demuxer=*/false,
-        std::optional<RtpHeaderExtensions>(last_recv_params_.extensions),
-        error_desc);
+        /*update_demuxer=*/false, last_recv_params_.extensions, error_desc);
     if (!success) {
       return false;
     }
diff --git a/pc/channel.h b/pc/channel.h
index 995a17e96b..d651d066a9 100644
--- a/pc/channel.h
+++ b/pc/channel.h
@@ -290,17 +290,20 @@ class BaseChannel : public ChannelInterface,
   // clearing.
   bool ClearHandledPayloadTypes() RTC_RUN_ON(worker_thread());
 
-  // Hops to the network thread to update the transport if an update is
-  // requested. If `update_demuxer` is false and `extensions` is not set, the
-  // function simply returns. If either of these is set, the function updates
-  // the transport with either or both of the demuxer criteria and the supplied
-  // rtp header extensions.
+  // Checks that the provided RTP header extensions are valid.
+  // This verifies that all extension IDs are within the valid range,
+  // that there are no duplicate IDs, and that no existing extension ID
+  // has been reassigned to a different URI.
+  bool CheckRtpExtensionValidity(const RtpHeaderExtensions& extensions,
+                                 std::string& error_desc) const
+      RTC_RUN_ON(worker_thread());
+
   // Returns `true` if either an update wasn't needed or one was successfully
   // applied. If the return value is `false`, then updating the demuxer criteria
   // failed, which needs to be treated as an error.
   bool MaybeUpdateDemuxerAndRtpExtensions_w(
       bool update_demuxer,
-      std::optional<RtpHeaderExtensions> extensions,
+      const RtpHeaderExtensions& extensions,
       std::string& error_desc) RTC_RUN_ON(worker_thread());
 
   bool RegisterRtpDemuxerSink_w() RTC_RUN_ON(worker_thread());
@@ -365,6 +368,12 @@ class BaseChannel : public ChannelInterface,
   flat_set<uint8_t> payload_types_ RTC_GUARDED_BY(worker_thread());
   // A stored copy of the rtp header extensions as applied to the transport.
   RtpHeaderExtensions rtp_header_extensions_ RTC_GUARDED_BY(worker_thread());
+
+  // Set of all historic RTP header extensions mapped, keyed by URI,
+  // to ensure no ID-URI reassignment occurs per RFC 8285.
+  RtpHeaderExtensions historical_rtp_header_extensions_
+      RTC_GUARDED_BY(worker_thread());
+
   // TODO(bugs.webrtc.org/12239): Modified on worker thread, accessed
   // on network thread in RegisterRtpDemuxerSink_n (called from Init_w)
   RtpDemuxerCriteria demuxer_criteria_;
diff --git a/pc/channel_unittest.cc b/pc/channel_unittest.cc
index 0c00c480dd..3f588332bc 100644
--- a/pc/channel_unittest.cc
+++ b/pc/channel_unittest.cc
@@ -704,6 +704,80 @@ class ChannelTest : public ::testing::Test {
                                         RtpExtension::kVideoRotationUri))));
   }
 
+  void TestDuplicateRtpHeaderExtensionIds() {
+    typename T::Content local;
+    CreateContent(/*flags=*/0, kPcmuCodec, kH264Codec, &local);
+    local.set_rtp_header_extensions({
+        RtpExtension(RtpExtension::kTransportSequenceNumberUri, 1),
+        RtpExtension(RtpExtension::kVideoRotationUri, 1),
+    });
+
+    CreateChannels(0, 0);
+    std::string err;
+    EXPECT_FALSE(channel1_->SetLocalContent(&local, SdpType::kOffer, err));
+    EXPECT_THAT(err, ::testing::HasSubstr("Duplicate RTP extension ID"));
+  }
+
+  void TestInvalidRtpHeaderExtensionIds() {
+    typename T::Content local;
+    CreateContent(/*flags=*/0, kPcmuCodec, kH264Codec, &local);
+    local.set_rtp_header_extensions({
+        RtpExtension(RtpExtension::kTransportSequenceNumberUri, 256),
+    });
+
+    CreateChannels(0, 0);
+    std::string err;
+    EXPECT_FALSE(channel1_->SetLocalContent(&local, SdpType::kOffer, err));
+    EXPECT_THAT(err, ::testing::HasSubstr("Bad RTP extension ID"));
+  }
+
+  void TestRtpHeaderExtensionIdReassignment() {
+    typename T::Content local;
+    CreateContent(/*flags=*/0, kPcmuCodec, kH264Codec, &local);
+    local.set_rtp_header_extensions({
+        RtpExtension(RtpExtension::kTransportSequenceNumberUri, 1),
+    });
+
+    CreateChannels(0, 0);
+    std::string err;
+    ASSERT_TRUE(channel1_->SetLocalContent(&local, SdpType::kOffer, err));
+
+    typename T::Content local_updated;
+    CreateContent(/*flags=*/0, kPcmuCodec, kH264Codec, &local_updated);
+    local_updated.set_rtp_header_extensions({
+        RtpExtension(RtpExtension::kVideoRotationUri, 1),
+    });
+    EXPECT_FALSE(
+        channel1_->SetLocalContent(&local_updated, SdpType::kOffer, err));
+    EXPECT_THAT(err, ::testing::HasSubstr("RTP extension ID reassignment"));
+  }
+
+  void TestRtpHeaderExtensionIdHistoryReassignment() {
+    typename T::Content local;
+    CreateContent(/*flags=*/0, kPcmuCodec, kH264Codec, &local);
+    local.set_rtp_header_extensions({
+        RtpExtension(RtpExtension::kTransportSequenceNumberUri, 1),
+    });
+
+    CreateChannels(0, 0);
+    std::string err;
+    ASSERT_TRUE(channel1_->SetLocalContent(&local, SdpType::kOffer, err));
+
+    typename T::Content local_empty;
+    CreateContent(/*flags=*/0, kPcmuCodec, kH264Codec, &local_empty);
+    local_empty.set_rtp_header_extensions({});
+    ASSERT_TRUE(channel1_->SetLocalContent(&local_empty, SdpType::kOffer, err));
+
+    typename T::Content local_updated;
+    CreateContent(/*flags=*/0, kPcmuCodec, kH264Codec, &local_updated);
+    local_updated.set_rtp_header_extensions({
+        RtpExtension(RtpExtension::kVideoRotationUri, 1),
+    });
+    EXPECT_FALSE(
+        channel1_->SetLocalContent(&local_updated, SdpType::kOffer, err));
+    EXPECT_THAT(err, ::testing::HasSubstr("RTP extension ID reassignment"));
+  }
+
   // Test that SetLocalContent and SetRemoteContent properly configure
   // extmap-allow-mixed.
   void TestSetContentsExtmapAllowMixedCaller(bool offer, bool answer) {
@@ -1942,6 +2016,23 @@ TEST_F(VoiceChannelSingleThreadTest, RemovesExtensionNotPresentInLocalAnswer) {
   Base::TestRemovesExtensionNotPresentInLocalAnswer();
 }
 
+TEST_F(VoiceChannelSingleThreadTest, DuplicateRtpHeaderExtensionIds) {
+  Base::TestDuplicateRtpHeaderExtensionIds();
+}
+
+TEST_F(VoiceChannelSingleThreadTest, InvalidRtpHeaderExtensionIds) {
+  Base::TestInvalidRtpHeaderExtensionIds();
+}
+
+TEST_F(VoiceChannelSingleThreadTest, RtpHeaderExtensionIdReassignment) {
+  Base::TestRtpHeaderExtensionIdReassignment();
+}
+
+TEST_F(VoiceChannelSingleThreadTest,
+       TestRtpHeaderExtensionIdHistoryReassignment) {
+  Base::TestRtpHeaderExtensionIdHistoryReassignment();
+}
+
 // VoiceChannelDoubleThreadTest
 TEST_F(VoiceChannelDoubleThreadTest, TestInit) {
   Base::TestInit();
@@ -2230,6 +2321,23 @@ TEST_F(VideoChannelSingleThreadTest, RemovesExtensionNotPresentInLocalAnswer) {
   Base::TestRemovesExtensionNotPresentInLocalAnswer();
 }
 
+TEST_F(VideoChannelSingleThreadTest, DuplicateRtpHeaderExtensionIds) {
+  Base::TestDuplicateRtpHeaderExtensionIds();
+}
+
+TEST_F(VideoChannelSingleThreadTest, InvalidRtpHeaderExtensionIds) {
+  Base::TestInvalidRtpHeaderExtensionIds();
+}
+
+TEST_F(VideoChannelSingleThreadTest, RtpHeaderExtensionIdReassignment) {
+  Base::TestRtpHeaderExtensionIdReassignment();
+}
+
+TEST_F(VideoChannelSingleThreadTest,
+       TestRtpHeaderExtensionIdHistoryReassignment) {
+  Base::TestRtpHeaderExtensionIdHistoryReassignment();
+}
+
 TEST_F(VideoChannelSingleThreadTest, TestSetLocalOfferWithPacketization) {
   const webrtc::Codec kVp8Codec = webrtc::CreateVideoCodec(97, "VP8");
   webrtc::Codec vp9_codec = webrtc::CreateVideoCodec(98, "VP9");

From 06a10bde23991b52fbc9b0c6ce17a38b38a68112 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Mon, 9 Mar 2026 11:55:40 +0100
Subject: [PATCH 234/472] Replace EncoderSwitchRequestCallback with
 absl::AnyInvocable

This refactoring replaces the EncoderSwitchRequestCallback interface
with absl::AnyInvocable in VideoStreamEncoderSettings.

To ensure the callback survives stream recreation, the callback binding
has been moved to RecreateWebRtcStream, relying on a new `send_channel_`
back-pointer within WebRtcVideoSendStream.

Bug: b/478050997
Change-Id: I1f587d563553c4f2f3d8fd02c4d9ed4473829dc4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/454480
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47087}
---
 api/video/BUILD.gn                           |   1 +
 api/video/video_stream_encoder_settings.h    |  25 ++--
 call/BUILD.gn                                |   2 +
 call/call.cc                                 |  15 ++-
 call/call.h                                  |   8 +-
 media/BUILD.gn                               |   1 +
 media/engine/fake_webrtc_call.cc             |  18 ++-
 media/engine/fake_webrtc_call.h              |  12 +-
 media/engine/webrtc_video_engine.cc          | 111 +++++++++--------
 media/engine/webrtc_video_engine.h           |  14 +--
 media/engine/webrtc_video_engine_unittest.cc |  60 +++++-----
 pc/peer_connection_rtp_unittest.cc           |  20 ++--
 pc/rtp_transceiver.cc                        |   2 +-
 test/call_test.cc                            |   2 +-
 test/scenario/video_stream.cc                |   2 +-
 video/video_send_stream_impl.cc              |  15 ++-
 video/video_send_stream_impl.h               |  31 ++---
 video/video_send_stream_impl_unittest.cc     |   4 +-
 video/video_stream_encoder.cc                |  32 ++---
 video/video_stream_encoder.h                 |   7 +-
 video/video_stream_encoder_unittest.cc       | 120 +++++++++----------
 21 files changed, 274 insertions(+), 228 deletions(-)

diff --git a/api/video/BUILD.gn b/api/video/BUILD.gn
index 14cf3604b4..d6186040dc 100644
--- a/api/video/BUILD.gn
+++ b/api/video/BUILD.gn
@@ -349,6 +349,7 @@ rtc_source_set("video_stream_encoder") {
     "../adaptation:resource_adaptation_api",
     "../units:data_rate",
     "../video_codecs:video_codecs_api",
+    "//third_party/abseil-cpp/absl/functional:any_invocable",
   ]
 }
 
diff --git a/api/video/video_stream_encoder_settings.h b/api/video/video_stream_encoder_settings.h
index d02bcbe739..5936dc786c 100644
--- a/api/video/video_stream_encoder_settings.h
+++ b/api/video/video_stream_encoder_settings.h
@@ -11,6 +11,9 @@
 #ifndef API_VIDEO_VIDEO_STREAM_ENCODER_SETTINGS_H_
 #define API_VIDEO_VIDEO_STREAM_ENCODER_SETTINGS_H_
 
+#include <optional>
+
+#include "absl/functional/any_invocable.h"
 #include "api/video/video_bitrate_allocator_factory.h"
 #include "api/video_codecs/sdp_video_format.h"
 #include "api/video_codecs/video_encoder.h"
@@ -18,18 +21,13 @@
 
 namespace webrtc {
 
-class EncoderSwitchRequestCallback {
- public:
-  virtual ~EncoderSwitchRequestCallback() {}
-
-  // Requests switch to next negotiated encoder.
-  virtual void RequestEncoderFallback() = 0;
-
-  // Requests switch to a specific encoder. If the encoder is not available and
-  // `allow_default_fallback` is `true` the default fallback is invoked.
-  virtual void RequestEncoderSwitch(const SdpVideoFormat& format,
-                                    bool allow_default_fallback) = 0;
-};
+// Requests switch to a specific encoder. If `format` is nullopt, a fallback
+// to the next negotiated encoder is requested. If the requested encoder is
+// not available and `allow_default_fallback` is `true`, the default fallback
+// is invoked.
+using EncoderSwitchRequestCallback =
+    absl::AnyInvocable<void(std::optional<SdpVideoFormat> format,
+                            bool allow_default_fallback)>;
 
 struct VideoStreamEncoderSettings {
   explicit VideoStreamEncoderSettings(
@@ -43,9 +41,6 @@ struct VideoStreamEncoderSettings {
   // Ownership stays with WebrtcVideoEngine (delegated from PeerConnection).
   VideoEncoderFactory* encoder_factory = nullptr;
 
-  // Requests the WebRtcVideoChannel to perform a codec switch.
-  EncoderSwitchRequestCallback* encoder_switch_request_callback = nullptr;
-
   // Ownership stays with WebrtcVideoEngine (delegated from PeerConnection).
   VideoBitrateAllocatorFactory* bitrate_allocator_factory = nullptr;
 
diff --git a/call/BUILD.gn b/call/BUILD.gn
index dd7cbac1ac..232c66aef8 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -68,6 +68,7 @@ rtc_library("call_interfaces") {
     "../api/transport:network_control",
     "../api/units:time_delta",
     "../api/units:timestamp",
+    "../api/video:video_stream_encoder",
     "../modules/async_audio_processing",
     "../modules/congestion_controller/rtp:congestion_controller_feedback_stats",
     "../modules/rtp_rtcp",
@@ -323,6 +324,7 @@ rtc_library("call") {
     "../api/units:data_size",
     "../api/units:time_delta",
     "../api/units:timestamp",
+    "../api/video:video_stream_encoder",
     "../audio",
     "../logging:rtc_event_audio",
     "../logging:rtc_event_rtp_rtcp",
diff --git a/call/call.cc b/call/call.cc
index 052c8a76fe..40a35c978a 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -43,6 +43,7 @@
 #include "api/units/data_size.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
+#include "api/video/video_stream_encoder_settings.h"
 #include "audio/audio_receive_stream.h"
 #include "audio/audio_send_stream.h"
 #include "audio/audio_state.h"
@@ -225,10 +226,13 @@ class Call final : public webrtc::Call,
 
   webrtc::VideoSendStream* CreateVideoSendStream(
       webrtc::VideoSendStream::Config config,
-      VideoEncoderConfig encoder_config) override;
+      VideoEncoderConfig encoder_config,
+      EncoderSwitchRequestCallback encoder_switch_request_callback =
+          nullptr) override;
   webrtc::VideoSendStream* CreateVideoSendStream(
       webrtc::VideoSendStream::Config config,
       VideoEncoderConfig encoder_config,
+      EncoderSwitchRequestCallback encoder_switch_request_callback,
       std::unique_ptr<FecController> fec_controller) override;
   void DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) override;
 
@@ -499,6 +503,7 @@ std::unique_ptr<Call> Call::Create(CallConfig config) {
 VideoSendStream* Call::CreateVideoSendStream(
     VideoSendStream::Config /* config */,
     VideoEncoderConfig /* encoder_config */,
+    EncoderSwitchRequestCallback /* encoder_switch_request_callback */,
     std::unique_ptr<FecController> /* fec_controller */) {
   return nullptr;
 }
@@ -879,6 +884,7 @@ void Call::DestroyAudioReceiveStream(
 webrtc::VideoSendStream* Call::CreateVideoSendStream(
     webrtc::VideoSendStream::Config config,
     VideoEncoderConfig encoder_config,
+    EncoderSwitchRequestCallback encoder_switch_request_callback,
     std::unique_ptr<FecController> fec_controller) {
   TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream");
   RTC_DCHECK_RUN_ON(worker_thread_);
@@ -902,7 +908,8 @@ webrtc::VideoSendStream* Call::CreateVideoSendStream(
       transport_send_.get(), config_.encode_metronome, bitrate_allocator_.get(),
       video_send_delay_stats_.get(), std::move(config),
       std::move(encoder_config), suspended_video_send_ssrcs_,
-      suspended_video_payload_states_, std::move(fec_controller));
+      suspended_video_payload_states_, std::move(fec_controller),
+      std::move(encoder_switch_request_callback));
 
   for (uint32_t ssrc : ssrcs) {
     RTC_DCHECK(video_send_ssrcs_.find(ssrc) == video_send_ssrcs_.end());
@@ -923,7 +930,8 @@ webrtc::VideoSendStream* Call::CreateVideoSendStream(
 
 webrtc::VideoSendStream* Call::CreateVideoSendStream(
     webrtc::VideoSendStream::Config config,
-    VideoEncoderConfig encoder_config) {
+    VideoEncoderConfig encoder_config,
+    EncoderSwitchRequestCallback encoder_switch_request_callback) {
   RTC_DCHECK_RUN_ON(worker_thread_);
   if (config_.fec_controller_factory) {
     RTC_LOG(LS_INFO) << "External FEC Controller will be used.";
@@ -933,6 +941,7 @@ webrtc::VideoSendStream* Call::CreateVideoSendStream(
           ? config_.fec_controller_factory->CreateFecController(env_)
           : std::make_unique<FecControllerDefault>(env_);
   return CreateVideoSendStream(std::move(config), std::move(encoder_config),
+                               std::move(encoder_switch_request_callback),
                                std::move(fec_controller));
 }
 
diff --git a/call/call.h b/call/call.h
index 8f58370194..636c0f9737 100644
--- a/call/call.h
+++ b/call/call.h
@@ -24,6 +24,7 @@
 #include "api/scoped_refptr.h"
 #include "api/task_queue/task_queue_base.h"
 #include "api/transport/bitrate_settings.h"
+#include "api/video/video_stream_encoder_settings.h"
 #include "call/audio_receive_stream.h"
 #include "call/audio_send_stream.h"
 #include "call/call_config.h"
@@ -80,11 +81,14 @@ class Call {
 
   virtual VideoSendStream* CreateVideoSendStream(
       VideoSendStream::Config config,
-      VideoEncoderConfig encoder_config) = 0;
+      VideoEncoderConfig encoder_config,
+      EncoderSwitchRequestCallback encoder_switch_request_callback =
+          nullptr) = 0;
   virtual VideoSendStream* CreateVideoSendStream(
       VideoSendStream::Config config,
       VideoEncoderConfig encoder_config,
-      std::unique_ptr<FecController> fec_controller);
+      EncoderSwitchRequestCallback encoder_switch_request_callback,
+      std::unique_ptr<FecController> fec_controller) = 0;
   virtual void DestroyVideoSendStream(VideoSendStream* send_stream) = 0;
 
   virtual VideoReceiveStreamInterface* CreateVideoReceiveStream(
diff --git a/media/BUILD.gn b/media/BUILD.gn
index 47e6d7b403..fbe70c99bc 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -857,6 +857,7 @@ if (rtc_include_tests) {
       "../api/video:video_frame",
       "../api/video:video_frame_type",
       "../api/video:video_rtp_headers",
+      "../api/video:video_stream_encoder",
       "../api/video_codecs:scalability_mode",
       "../api/video_codecs:video_codecs_api",
       "../call:call_interfaces",
diff --git a/media/engine/fake_webrtc_call.cc b/media/engine/fake_webrtc_call.cc
index cc537a4667..20ed9865a8 100644
--- a/media/engine/fake_webrtc_call.cc
+++ b/media/engine/fake_webrtc_call.cc
@@ -12,6 +12,7 @@
 
 #include <cstdint>
 #include <map>
+#include <memory>
 #include <string>
 #include <utility>
 #include <vector>
@@ -24,6 +25,7 @@
 #include "api/call/audio_sink.h"
 #include "api/crypto/frame_decryptor_interface.h"
 #include "api/environment/environment.h"
+#include "api/fec_controller.h"
 #include "api/frame_transformer_interface.h"
 #include "api/make_ref_counted.h"
 #include "api/media_types.h"
@@ -35,6 +37,7 @@
 #include "api/task_queue/task_queue_base.h"
 #include "api/units/timestamp.h"
 #include "api/video/video_source_interface.h"
+#include "api/video/video_stream_encoder_settings.h"
 #include "api/video_codecs/video_codec.h"
 #include "api/video_codecs/video_encoder.h"
 #include "call/audio_receive_stream.h"
@@ -586,7 +589,20 @@ void FakeCall::DestroyAudioReceiveStream(
 
 VideoSendStream* FakeCall::CreateVideoSendStream(
     VideoSendStream::Config config,
-    VideoEncoderConfig encoder_config) {
+    VideoEncoderConfig encoder_config,
+    EncoderSwitchRequestCallback encoder_switch_request_callback) {
+  FakeVideoSendStream* fake_stream = new FakeVideoSendStream(
+      env_, std::move(config), std::move(encoder_config));
+  video_send_streams_.push_back(fake_stream);
+  ++num_created_send_streams_;
+  return fake_stream;
+}
+
+VideoSendStream* FakeCall::CreateVideoSendStream(
+    VideoSendStream::Config config,
+    VideoEncoderConfig encoder_config,
+    EncoderSwitchRequestCallback encoder_switch_request_callback,
+    std::unique_ptr<FecController> fec_controller) {
   FakeVideoSendStream* fake_stream = new FakeVideoSendStream(
       env_, std::move(config), std::move(encoder_config));
   video_send_streams_.push_back(fake_stream);
diff --git a/media/engine/fake_webrtc_call.h b/media/engine/fake_webrtc_call.h
index ac12bd53f0..e4dff9a22b 100644
--- a/media/engine/fake_webrtc_call.h
+++ b/media/engine/fake_webrtc_call.h
@@ -37,6 +37,7 @@
 #include "api/audio_codecs/audio_format.h"
 #include "api/crypto/frame_decryptor_interface.h"
 #include "api/environment/environment.h"
+#include "api/fec_controller.h"
 #include "api/frame_transformer_interface.h"
 #include "api/media_types.h"
 #include "api/rtp_headers.h"
@@ -51,6 +52,7 @@
 #include "api/video/video_frame.h"
 #include "api/video/video_sink_interface.h"
 #include "api/video/video_source_interface.h"
+#include "api/video/video_stream_encoder_settings.h"
 #include "api/video_codecs/video_codec.h"
 #include "call/audio_receive_stream.h"
 #include "call/audio_send_stream.h"
@@ -441,7 +443,15 @@ class FakeCall final : public Call, public PacketReceiver {
 
   VideoSendStream* CreateVideoSendStream(
       VideoSendStream::Config config,
-      VideoEncoderConfig encoder_config) override;
+      VideoEncoderConfig encoder_config,
+      EncoderSwitchRequestCallback encoder_switch_request_callback =
+          nullptr) override;
+
+  VideoSendStream* CreateVideoSendStream(
+      VideoSendStream::Config config,
+      VideoEncoderConfig encoder_config,
+      EncoderSwitchRequestCallback encoder_switch_request_callback,
+      std::unique_ptr<FecController> fec_controller) override;
   void DestroyVideoSendStream(VideoSendStream* send_stream) override;
 
   VideoReceiveStreamInterface* CreateVideoReceiveStream(
diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc
index f1cd60d8f5..8bfffdc7e4 100644
--- a/media/engine/webrtc_video_engine.cc
+++ b/media/engine/webrtc_video_engine.cc
@@ -1239,68 +1239,63 @@ bool WebRtcVideoSendChannel::SetSenderParameters(
   return ApplyChangedParams(changed_params);
 }
 
-void WebRtcVideoSendChannel::RequestEncoderFallback() {
-  if (!worker_thread_->IsCurrent()) {
-    worker_thread_->PostTask(
-        SafeTask(task_safety_.flag(), [this] { RequestEncoderFallback(); }));
-    return;
-  }
-
-  RTC_DCHECK_RUN_ON(&thread_checker_);
-  if (negotiated_codecs_.size() <= 1) {
-    RTC_LOG(LS_WARNING) << "Encoder failed but no fallback codec is available";
-    return;
-  }
-
-  ChangedSenderParameters params;
-  params.negotiated_codecs = negotiated_codecs_;
-  params.negotiated_codecs->erase(params.negotiated_codecs->begin());
-  params.send_codec = params.negotiated_codecs->front();
-  if (ApplyChangedParams(params) && parameters_changed_callback_) {
-    parameters_changed_callback_();
-  }
-}
-
-void WebRtcVideoSendChannel::RequestEncoderSwitch(const SdpVideoFormat& format,
-                                                  bool allow_default_fallback) {
+void WebRtcVideoSendChannel::RequestEncoderSwitch(
+    std::optional<SdpVideoFormat> format,
+    bool allow_default_fallback) {
   if (!worker_thread_->IsCurrent()) {
     worker_thread_->PostTask(
         SafeTask(task_safety_.flag(), [this, format, allow_default_fallback] {
-          RequestEncoderSwitch(format, allow_default_fallback);
+          RequestEncoderSwitch(std::move(format), allow_default_fallback);
         }));
     return;
   }
 
   RTC_DCHECK_RUN_ON(&thread_checker_);
+  RTC_DCHECK(format.has_value() || allow_default_fallback);
 
-  for (const VideoCodecSettings& codec_setting : negotiated_codecs_) {
-    if (format.IsSameCodec(
-            {codec_setting.codec.name, codec_setting.codec.params})) {
-      VideoCodecSettings new_codec_setting = codec_setting;
-      for (const auto& kv : format.parameters) {
-        new_codec_setting.codec.params[kv.first] = kv.second;
-      }
+  ChangedSenderParameters params;
+  if (!format) {
+    if (negotiated_codecs_.size() <= 1) {
+      RTC_LOG(LS_WARNING)
+          << "Encoder failed but no fallback codec is available";
+      return;
+    }
 
-      if (send_codec() == new_codec_setting) {
-        // Already using this codec, no switch required.
-        return;
-      }
+    params.negotiated_codecs = negotiated_codecs_;
+    params.negotiated_codecs->erase(params.negotiated_codecs->begin());
+    params.send_codec = params.negotiated_codecs->front();
+  } else {
+    auto it = absl::c_find_if(
+        negotiated_codecs_, [&](const VideoCodecSettings& codec_setting) {
+          return format->IsSameCodec(
+              {codec_setting.codec.name, codec_setting.codec.params});
+        });
+    if (it == negotiated_codecs_.end()) {
+      RTC_LOG(LS_WARNING) << "Failed to switch encoder to: "
+                          << format->ToString()
+                          << ". Is default fallback allowed: "
+                          << allow_default_fallback;
 
-      ChangedSenderParameters params;
-      params.send_codec = new_codec_setting;
-      if (ApplyChangedParams(params) && parameters_changed_callback_) {
-        parameters_changed_callback_();
+      if (allow_default_fallback) {
+        RequestEncoderSwitch(std::nullopt, true);
       }
       return;
     }
+
+    VideoCodecSettings new_codec_setting = *it;
+    for (const auto& kv : format->parameters) {
+      new_codec_setting.codec.params[kv.first] = kv.second;
+    }
+
+    if (send_codec() == new_codec_setting) {
+      return;
+    }
+
+    params.send_codec = new_codec_setting;
   }
 
-  RTC_LOG(LS_WARNING) << "Failed to switch encoder to: " << format.ToString()
-                      << ". Is default fallback allowed: "
-                      << allow_default_fallback;
-
-  if (allow_default_fallback) {
-    RequestEncoderFallback();
+  if (ApplyChangedParams(params) && parameters_changed_callback_) {
+    parameters_changed_callback_();
   }
 }
 
@@ -1581,7 +1576,6 @@ bool WebRtcVideoSendChannel::AddSendStream(const StreamParams& sp) {
   config.encoder_settings.encoder_factory = encoder_factory_;
   config.encoder_settings.bitrate_allocator_factory =
       bitrate_allocator_factory_;
-  config.encoder_settings.encoder_switch_request_callback = this;
 
   config.crypto_options = crypto_options_;
   config.rtp.extmap_allow_mixed = ExtmapAllowMixed();
@@ -1590,7 +1584,7 @@ bool WebRtcVideoSendChannel::AddSendStream(const StreamParams& sp) {
       video_config_.enable_send_packet_batching;
 
   WebRtcVideoSendStream* stream = new WebRtcVideoSendStream(
-      env_, call_, sp, std::move(config), default_send_options_,
+      this, env_, call_, sp, std::move(config), default_send_options_,
       video_config_.enable_cpu_adaptation, bitrate_config_.max_bitrate_bps,
       send_codec(), send_codecs_, send_rtp_extensions_, send_params_);
 
@@ -1814,6 +1808,7 @@ WebRtcVideoSendChannel::WebRtcVideoSendStream::VideoSendStreamParameters::
       codec_settings_list(codec_settings_list) {}
 
 WebRtcVideoSendChannel::WebRtcVideoSendStream::WebRtcVideoSendStream(
+    WebRtcVideoSendChannel* send_channel,
     const Environment& env,
     Call* call,
     const StreamParams& sp,
@@ -1827,7 +1822,8 @@ WebRtcVideoSendChannel::WebRtcVideoSendStream::WebRtcVideoSendStream(
     // TODO(deadbeef): Don't duplicate information between send_params,
     // rtp_extensions, options, etc.
     const VideoSenderParameters& send_params)
-    : env_(env),
+    : send_channel_(send_channel),
+      env_(env),
       worker_thread_(call->worker_thread()),
       ssrcs_(sp.ssrcs),
       ssrc_groups_(sp.ssrc_groups),
@@ -2702,6 +2698,13 @@ void WebRtcVideoSendChannel::WebRtcVideoSendStream::RecreateWebRtcStream() {
       ConfigureVideoEncoderSettings(parameters_.codec_settings->codec);
 
   VideoSendStream::Config config = parameters_.config.Copy();
+
+  auto encoder_switch_request_callback =
+      [send_channel = send_channel_](std::optional<SdpVideoFormat> format,
+                                     bool allow_default_fallback) {
+        send_channel->RequestEncoderSwitch(std::move(format),
+                                           allow_default_fallback);
+      };
   if (!config.rtp.rtx.ssrcs.empty() && config.rtp.rtx.payload_type == -1) {
     RTC_LOG(LS_WARNING) << "RTX SSRCs configured but there's no configured RTX "
                            "payload type the set codec. Ignoring RTX.";
@@ -2728,8 +2731,9 @@ void WebRtcVideoSendChannel::WebRtcVideoSendStream::RecreateWebRtcStream() {
     // GetStats and DestroyVideoSendStream.
     VideoSendStream::Stats stats = stream_->GetStats();
     call_->DestroyVideoSendStream(stream_);
-    stream_ = call_->CreateVideoSendStream(std::move(config),
-                                           parameters_.encoder_config.Copy());
+    stream_ = call_->CreateVideoSendStream(
+        std::move(config), parameters_.encoder_config.Copy(),
+        std::move(encoder_switch_request_callback));
 
     // A new stream is created without any scaling or limitations, so these
     // flags don't apply until the new stream experiences an adaptation event.
@@ -2741,8 +2745,9 @@ void WebRtcVideoSendChannel::WebRtcVideoSendStream::RecreateWebRtcStream() {
 
     stream_->SetStats(stats);
   } else {
-    stream_ = call_->CreateVideoSendStream(std::move(config),
-                                           parameters_.encoder_config.Copy());
+    stream_ = call_->CreateVideoSendStream(
+        std::move(config), parameters_.encoder_config.Copy(),
+        std::move(encoder_switch_request_callback));
   }
   if (!rtp_parameters_.encodings.empty() &&
       rtp_parameters_.encodings[0].csrcs.has_value()) {
diff --git a/media/engine/webrtc_video_engine.h b/media/engine/webrtc_video_engine.h
index f0a3412534..307768f2a3 100644
--- a/media/engine/webrtc_video_engine.h
+++ b/media/engine/webrtc_video_engine.h
@@ -47,7 +47,6 @@
 #include "api/video/video_frame.h"
 #include "api/video/video_sink_interface.h"
 #include "api/video/video_source_interface.h"
-#include "api/video/video_stream_encoder_settings.h"
 #include "api/video_codecs/sdp_video_format.h"
 #include "api/video_codecs/video_encoder_factory.h"
 #include "call/call.h"
@@ -159,8 +158,7 @@ struct VideoCodecSettings {
 };
 
 class WebRtcVideoSendChannel : public MediaChannelUtil,
-                               public VideoMediaSendChannelInterface,
-                               public EncoderSwitchRequestCallback {
+                               public VideoMediaSendChannelInterface {
  public:
   WebRtcVideoSendChannel(
       const Environment& env,
@@ -253,10 +251,10 @@ class WebRtcVideoSendChannel : public MediaChannelUtil,
     ADAPTREASON_BANDWIDTH = 2,
   };
 
-  // Implements EncoderSwitchRequestCallback.
-  void RequestEncoderFallback() override;
-  void RequestEncoderSwitch(const SdpVideoFormat& format,
-                            bool allow_default_fallback) override;
+  // Called to request an encoder switch or fallback.
+  // See also EncoderSwitchRequestCallback.
+  void RequestEncoderSwitch(std::optional<SdpVideoFormat> format,
+                            bool allow_default_fallback);
 
   void GenerateSendKeyFrame(uint32_t ssrc,
                             const std::vector<std::string>& rids) override;
@@ -318,6 +316,7 @@ class WebRtcVideoSendChannel : public MediaChannelUtil,
   class WebRtcVideoSendStream {
    public:
     WebRtcVideoSendStream(
+        WebRtcVideoSendChannel* send_channel,
         const Environment& env,
         Call* call,
         const StreamParams& sp,
@@ -402,6 +401,7 @@ class WebRtcVideoSendChannel : public MediaChannelUtil,
     DegradationPreference GetDegradationPreference() const
         RTC_EXCLUSIVE_LOCKS_REQUIRED(&thread_checker_);
 
+    WebRtcVideoSendChannel* const send_channel_;
     const Environment env_;
     RTC_NO_UNIQUE_ADDRESS SequenceChecker thread_checker_;
     TaskQueueBase* const worker_thread_;
diff --git a/media/engine/webrtc_video_engine_unittest.cc b/media/engine/webrtc_video_engine_unittest.cc
index 311d7281b0..d6f9d3872e 100644
--- a/media/engine/webrtc_video_engine_unittest.cc
+++ b/media/engine/webrtc_video_engine_unittest.cc
@@ -2598,19 +2598,19 @@ TEST_F(WebRtcVideoChannelBaseTest,
   ASSERT_TRUE(codec);
   EXPECT_EQ("VP9", codec->name);
 
-  SendImpl()->RequestEncoderFallback();
+  SendImpl()->RequestEncoderSwitch(std::nullopt, true);
   time_controller_.AdvanceTime(kFrameDuration);
   codec = send_channel_->GetSendCodec();
   ASSERT_TRUE(codec);
   EXPECT_EQ("AV1", codec->name);
 
-  SendImpl()->RequestEncoderFallback();
+  SendImpl()->RequestEncoderSwitch(std::nullopt, true);
   time_controller_.AdvanceTime(kFrameDuration);
   codec = send_channel_->GetSendCodec();
   ASSERT_TRUE(codec);
   EXPECT_EQ("VP8", codec->name);
 
-  SendImpl()->RequestEncoderFallback();
+  SendImpl()->RequestEncoderSwitch(std::nullopt, true);
   time_controller_.AdvanceTime(kFrameDuration);
 
   FrameForwarder frame_forwarder;
@@ -2620,7 +2620,7 @@ TEST_F(WebRtcVideoChannelBaseTest,
 
 #if defined(RTC_ENABLE_VP9)
 
-TEST_F(WebRtcVideoChannelBaseTest, RequestEncoderFallback) {
+TEST_F(WebRtcVideoChannelBaseTest, RequestEncoderSwitchWithNullopt) {
   VideoSenderParameters parameters;
   parameters.codecs.push_back(GetEngineCodec("VP9"));
   parameters.codecs.push_back(GetEngineCodec("VP8"));
@@ -2630,16 +2630,16 @@ TEST_F(WebRtcVideoChannelBaseTest, RequestEncoderFallback) {
   ASSERT_TRUE(codec);
   EXPECT_EQ("VP9", codec->name);
 
-  // RequestEncoderFallback will post a task to the worker thread (which is also
+  // RequestEncoderSwitch will post a task to the worker thread (which is also
   // the current thread), hence the ProcessMessages call.
-  SendImpl()->RequestEncoderFallback();
+  SendImpl()->RequestEncoderSwitch(std::nullopt, true);
   time_controller_.AdvanceTime(kFrameDuration);
   codec = send_channel_->GetSendCodec();
   ASSERT_TRUE(codec);
   EXPECT_EQ("VP8", codec->name);
 
   // No other codec to fall back to, keep using VP8.
-  SendImpl()->RequestEncoderFallback();
+  SendImpl()->RequestEncoderSwitch(std::nullopt, true);
   time_controller_.AdvanceTime(kFrameDuration);
   codec = send_channel_->GetSendCodec();
   ASSERT_TRUE(codec);
@@ -2712,9 +2712,9 @@ TEST_F(WebRtcVideoChannelBaseTest, SendCodecIsMovedToFrontInRtpParameters) {
   ASSERT_EQ(send_codecs.size(), 2u);
   EXPECT_THAT("VP9", send_codecs[0].name);
 
-  // RequestEncoderFallback will post a task to the worker thread (which is also
+  // RequestEncoderSwitch will post a task to the worker thread (which is also
   // the current thread), hence the ProcessMessages call.
-  SendImpl()->RequestEncoderFallback();
+  SendImpl()->RequestEncoderSwitch(std::nullopt, true);
   time_controller_.AdvanceTime(kFrameDuration);
 
   send_codecs = send_channel_->GetRtpSendParameters(kSsrc).codecs;
@@ -3903,8 +3903,6 @@ TEST_F(Vp9SettingsTest, MultipleSsrcsEnablesSvc) {
   FakeVideoSendStream* stream =
       AddSendStream(CreateSimStreamParams("cname", ssrcs));
 
-  VideoSendStream::Config config = stream->GetConfig().Copy();
-
   FrameForwarder frame_forwarder;
   EXPECT_TRUE(send_channel_->SetVideoSend(ssrcs[0], nullptr, &frame_forwarder));
   send_channel_->SetSend(true);
@@ -3932,7 +3930,7 @@ TEST_F(Vp9SettingsTest, SvcModeCreatesSingleRtpStream) {
   FakeVideoSendStream* stream =
       AddSendStream(CreateSimStreamParams("cname", ssrcs));
 
-  VideoSendStream::Config config = stream->GetConfig().Copy();
+  const VideoSendStream::Config& config = stream->GetConfig();
 
   // Despite 3 ssrcs provided, single layer is used.
   EXPECT_EQ(1u, config.rtp.ssrcs.size());
@@ -3994,8 +3992,6 @@ TEST_F(Vp9SettingsTest, MaxBitrateDeterminedBySvcResolutions) {
   FakeVideoSendStream* stream =
       AddSendStream(CreateSimStreamParams("cname", ssrcs));
 
-  VideoSendStream::Config config = stream->GetConfig().Copy();
-
   FrameForwarder frame_forwarder;
   EXPECT_TRUE(send_channel_->SetVideoSend(ssrcs[0], nullptr, &frame_forwarder));
   send_channel_->SetSend(true);
@@ -4037,8 +4033,6 @@ TEST_F(Vp9SettingsTest, Vp9SvcTargetBitrateCappedByMax) {
   FakeVideoSendStream* stream =
       AddSendStream(CreateSimStreamParams("cname", ssrcs));
 
-  VideoSendStream::Config config = stream->GetConfig().Copy();
-
   FrameForwarder frame_forwarder;
   EXPECT_TRUE(send_channel_->SetVideoSend(ssrcs[0], nullptr, &frame_forwarder));
   send_channel_->SetSend(true);
@@ -4386,7 +4380,7 @@ TEST_F(WebRtcVideoChannelTest, SetDefaultSendCodecs) {
   const std::vector<uint32_t> rtx_ssrcs = MAKE_VECTOR(kRtxSsrcs1);
   FakeVideoSendStream* stream =
       AddSendStream(CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs));
-  VideoSendStream::Config config = stream->GetConfig().Copy();
+  const VideoSendStream::Config& config = stream->GetConfig();
 
   // Make sure NACK and FEC are enabled on the correct payload types.
   EXPECT_EQ(1000, config.rtp.nack.rtp_history_ms);
@@ -4405,7 +4399,7 @@ TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithoutPacketization) {
   EXPECT_TRUE(send_channel_->SetSenderParameters(parameters));
 
   FakeVideoSendStream* stream = AddSendStream();
-  const VideoSendStream::Config config = stream->GetConfig().Copy();
+  const VideoSendStream::Config& config = stream->GetConfig();
   EXPECT_FALSE(config.rtp.raw_payload);
 }
 
@@ -4416,7 +4410,7 @@ TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithPacketization) {
   EXPECT_TRUE(send_channel_->SetSenderParameters(parameters));
 
   FakeVideoSendStream* stream = AddSendStream();
-  const VideoSendStream::Config config = stream->GetConfig().Copy();
+  const VideoSendStream::Config& config = stream->GetConfig();
   EXPECT_TRUE(config.rtp.raw_payload);
 }
 
@@ -4426,7 +4420,7 @@ TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithPacketization) {
 // default.
 TEST_F(WebRtcVideoChannelTest, FlexfecSendCodecWithoutSsrcNotExposedByDefault) {
   FakeVideoSendStream* stream = AddSendStream();
-  VideoSendStream::Config config = stream->GetConfig().Copy();
+  const VideoSendStream::Config& config = stream->GetConfig();
 
   EXPECT_EQ(-1, config.rtp.flexfec.payload_type);
   EXPECT_EQ(0U, config.rtp.flexfec.ssrc);
@@ -4436,7 +4430,7 @@ TEST_F(WebRtcVideoChannelTest, FlexfecSendCodecWithoutSsrcNotExposedByDefault) {
 TEST_F(WebRtcVideoChannelTest, FlexfecSendCodecWithSsrcNotExposedByDefault) {
   FakeVideoSendStream* stream = AddSendStream(
       CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc));
-  VideoSendStream::Config config = stream->GetConfig().Copy();
+  const VideoSendStream::Config& config = stream->GetConfig();
 
   EXPECT_EQ(-1, config.rtp.flexfec.payload_type);
   EXPECT_EQ(0U, config.rtp.flexfec.ssrc);
@@ -4634,7 +4628,7 @@ class WebRtcVideoChannelFlexfecSendRecvTest : public WebRtcVideoChannelTest {
 
 TEST_F(WebRtcVideoChannelFlexfecSendRecvTest, SetDefaultSendCodecsWithoutSsrc) {
   FakeVideoSendStream* stream = AddSendStream();
-  VideoSendStream::Config config = stream->GetConfig().Copy();
+  const VideoSendStream::Config& config = stream->GetConfig();
 
   EXPECT_EQ(GetEngineCodec("flexfec-03").id, config.rtp.flexfec.payload_type);
   EXPECT_EQ(0U, config.rtp.flexfec.ssrc);
@@ -4644,7 +4638,7 @@ TEST_F(WebRtcVideoChannelFlexfecSendRecvTest, SetDefaultSendCodecsWithoutSsrc) {
 TEST_F(WebRtcVideoChannelFlexfecSendRecvTest, SetDefaultSendCodecsWithSsrc) {
   FakeVideoSendStream* stream = AddSendStream(
       CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc));
-  VideoSendStream::Config config = stream->GetConfig().Copy();
+  const VideoSendStream::Config& config = stream->GetConfig();
 
   EXPECT_EQ(GetEngineCodec("flexfec-03").id, config.rtp.flexfec.payload_type);
   EXPECT_EQ(kFlexfecSsrc, config.rtp.flexfec.ssrc);
@@ -4658,7 +4652,7 @@ TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithoutFec) {
   ASSERT_TRUE(send_channel_->SetSenderParameters(parameters));
 
   FakeVideoSendStream* stream = AddSendStream();
-  VideoSendStream::Config config = stream->GetConfig().Copy();
+  const VideoSendStream::Config& config = stream->GetConfig();
 
   EXPECT_EQ(-1, config.rtp.ulpfec.ulpfec_payload_type);
   EXPECT_EQ(-1, config.rtp.ulpfec.red_payload_type);
@@ -4670,7 +4664,7 @@ TEST_F(WebRtcVideoChannelFlexfecSendRecvTest, SetSendCodecsWithoutFec) {
   ASSERT_TRUE(send_channel_->SetSenderParameters(parameters));
 
   FakeVideoSendStream* stream = AddSendStream();
-  VideoSendStream::Config config = stream->GetConfig().Copy();
+  const VideoSendStream::Config& config = stream->GetConfig();
 
   EXPECT_EQ(-1, config.rtp.flexfec.payload_type);
 }
@@ -4717,7 +4711,7 @@ TEST_F(WebRtcVideoChannelFlexfecRecvTest,
   ASSERT_TRUE(send_channel_->SetSenderParameters(parameters));
 
   FakeVideoSendStream* stream = AddSendStream();
-  VideoSendStream::Config config = stream->GetConfig().Copy();
+  const VideoSendStream::Config& config = stream->GetConfig();
 
   EXPECT_EQ(-1, config.rtp.flexfec.payload_type);
   EXPECT_EQ(0u, config.rtp.flexfec.ssrc);
@@ -4733,7 +4727,7 @@ TEST_F(WebRtcVideoChannelFlexfecRecvTest,
 
   FakeVideoSendStream* stream = AddSendStream(
       CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc));
-  VideoSendStream::Config config = stream->GetConfig().Copy();
+  const VideoSendStream::Config& config = stream->GetConfig();
 
   EXPECT_EQ(-1, config.rtp.flexfec.payload_type);
   EXPECT_EQ(0u, config.rtp.flexfec.ssrc);
@@ -4820,7 +4814,7 @@ TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithoutFecDisablesFec) {
   ASSERT_TRUE(send_channel_->SetSenderParameters(parameters));
 
   FakeVideoSendStream* stream = AddSendStream();
-  VideoSendStream::Config config = stream->GetConfig().Copy();
+  const VideoSendStream::Config& config = stream->GetConfig();
 
   EXPECT_EQ(GetEngineCodec("ulpfec").id, config.rtp.ulpfec.ulpfec_payload_type);
 
@@ -4828,8 +4822,8 @@ TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithoutFecDisablesFec) {
   ASSERT_TRUE(send_channel_->SetSenderParameters(parameters));
   stream = fake_call_->GetVideoSendStreams()[0];
   ASSERT_TRUE(stream != nullptr);
-  config = stream->GetConfig().Copy();
-  EXPECT_EQ(-1, config.rtp.ulpfec.ulpfec_payload_type)
+  const VideoSendStream::Config& config2 = stream->GetConfig();
+  EXPECT_EQ(-1, config2.rtp.ulpfec.ulpfec_payload_type)
       << "SetSendCodec without ULPFEC should disable current ULPFEC.";
 }
 
@@ -4842,7 +4836,7 @@ TEST_F(WebRtcVideoChannelFlexfecSendRecvTest,
 
   FakeVideoSendStream* stream = AddSendStream(
       CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc));
-  VideoSendStream::Config config = stream->GetConfig().Copy();
+  const VideoSendStream::Config& config = stream->GetConfig();
 
   EXPECT_EQ(GetEngineCodec("flexfec-03").id, config.rtp.flexfec.payload_type);
   EXPECT_EQ(kFlexfecSsrc, config.rtp.flexfec.ssrc);
@@ -4853,8 +4847,8 @@ TEST_F(WebRtcVideoChannelFlexfecSendRecvTest,
   ASSERT_TRUE(send_channel_->SetSenderParameters(parameters));
   stream = fake_call_->GetVideoSendStreams()[0];
   ASSERT_TRUE(stream != nullptr);
-  config = stream->GetConfig().Copy();
-  EXPECT_EQ(-1, config.rtp.flexfec.payload_type)
+  const VideoSendStream::Config& config2 = stream->GetConfig();
+  EXPECT_EQ(-1, config2.rtp.flexfec.payload_type)
       << "SetSendCodec without FlexFEC should disable current FlexFEC.";
 }
 
diff --git a/pc/peer_connection_rtp_unittest.cc b/pc/peer_connection_rtp_unittest.cc
index d0d83520a5..f2fcba417a 100644
--- a/pc/peer_connection_rtp_unittest.cc
+++ b/pc/peer_connection_rtp_unittest.cc
@@ -36,7 +36,6 @@
 #include "api/test/rtc_error_matchers.h"
 #include "api/units/data_rate.h"
 #include "api/video/render_resolution.h"
-#include "api/video/video_stream_encoder_settings.h"
 #include "api/video_codecs/sdp_video_format.h"
 #include "api/video_codecs/video_decoder_factory_template.h"
 #include "api/video_codecs/video_decoder_factory_template_dav1d_adapter.h"
@@ -2138,11 +2137,10 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan,
   worker_thread->BlockingCall([&] {
     // Simulate VideoStreamEncoder calling RequestEncoderSwitch after getting
     // the format from the EncoderSelector.
-    auto* switch_callback = static_cast<webrtc::EncoderSwitchRequestCallback*>(
-        static_cast<webrtc::WebRtcVideoSendChannel*>(
-            media_channel->AsVideoSendChannel()));
-    switch_callback->RequestEncoderSwitch(format_to_switch_to,
-                                          /*allow_default_fallback=*/false);
+    auto* video_send_channel = static_cast<webrtc::WebRtcVideoSendChannel*>(
+        media_channel->AsVideoSendChannel());
+    video_send_channel->RequestEncoderSwitch(std::move(format_to_switch_to),
+                                             /*allow_default_fallback=*/false);
   });
 
   // Allow the signaling thread to process the cache invalidation task posted
@@ -2186,15 +2184,13 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan, SendParamsCorrectWhenEncoderFallback) {
 
   worker_thread->BlockingCall([&] {
     // Simulate an encoder fallback on the worker thread.
-    auto* fallback_callback =
-        static_cast<webrtc::EncoderSwitchRequestCallback*>(
-            static_cast<webrtc::WebRtcVideoSendChannel*>(
-                media_channel->AsVideoSendChannel()));
-    fallback_callback->RequestEncoderFallback();
+    auto* video_send_channel = static_cast<webrtc::WebRtcVideoSendChannel*>(
+        media_channel->AsVideoSendChannel());
+    video_send_channel->RequestEncoderSwitch(std::nullopt, true);
   });
 
   // Allow the signaling thread to process the cache invalidation task posted
-  // by RequestEncoderFallback.
+  // by RequestEncoderSwitch(std::nullopt, true).
   run_loop_.Flush();
 
   // Call GetParameters. This triggers an internal consistency check in the
diff --git a/pc/rtp_transceiver.cc b/pc/rtp_transceiver.cc
index 520509277f..0701dca781 100644
--- a/pc/rtp_transceiver.cc
+++ b/pc/rtp_transceiver.cc
@@ -389,7 +389,7 @@ RtpTransceiver::RtpTransceiver(
   // This should be possible without a blocking call to the worker, perhaps done
   // asynchronously. At the moment this is complicated by the fact that
   // construction of the channels actually changes the settings of the engine.
-  context_->worker_thread()->BlockingCall([&]() mutable {
+  context_->worker_thread()->BlockingCall([&]() {
     RTC_DCHECK_RUN_ON(this->context()->worker_thread());
     auto channels = CreateMediaContentChannels(
         media_type_, env_, media_engine(), call, media_config, audio_options,
diff --git a/test/call_test.cc b/test/call_test.cc
index c25a6c828c..40cbc44854 100644
--- a/test/call_test.cc
+++ b/test/call_test.cc
@@ -607,7 +607,7 @@ void CallTest::CreateVideoSendStreams() {
     if (fec_controller_factory_) {
       video_send_streams_[i] = sender_call_->CreateVideoSendStream(
           video_send_configs_[i].Copy(), video_encoder_configs_[i].Copy(),
-          fec_controller_factory_->CreateFecController(send_env_));
+          nullptr, fec_controller_factory_->CreateFecController(send_env_));
     } else {
       video_send_streams_[i] = sender_call_->CreateVideoSendStream(
           video_send_configs_[i].Copy(), video_encoder_configs_[i].Copy());
diff --git a/test/scenario/video_stream.cc b/test/scenario/video_stream.cc
index 8cffa689c4..86e5111c07 100644
--- a/test/scenario/video_stream.cc
+++ b/test/scenario/video_stream.cc
@@ -428,7 +428,7 @@ SendVideoStream::SendVideoStream(CallClient* sender,
   sender_->SendTask([&] {
     if (config.stream.fec_controller_factory) {
       send_stream_ = sender_->call_->CreateVideoSendStream(
-          std::move(send_config), std::move(encoder_config),
+          std::move(send_config), std::move(encoder_config), nullptr,
           config.stream.fec_controller_factory->CreateFecController(
               sender_->env_));
     } else {
diff --git a/video/video_send_stream_impl.cc b/video/video_send_stream_impl.cc
index d6ff87ee85..ad9130bb1d 100644
--- a/video/video_send_stream_impl.cc
+++ b/video/video_send_stream_impl.cc
@@ -343,11 +343,12 @@ std::unique_ptr<VideoStreamEncoderInterface> CreateVideoStreamEncoder(
     const Environment& env,
     int num_cpu_cores,
     SendStatisticsProxy* stats_proxy,
-    const VideoStreamEncoderSettings& encoder_settings,
+    VideoStreamEncoderSettings encoder_settings,
     VideoStreamEncoder::BitrateAllocationCallbackType
         bitrate_allocation_callback_type,
     Metronome* metronome,
-    VideoEncoderFactory::EncoderSelectorInterface* encoder_selector) {
+    VideoEncoderFactory::EncoderSelectorInterface* encoder_selector,
+    EncoderSwitchRequestCallback encoder_switch_request_callback) {
   std::unique_ptr<TaskQueueBase, TaskQueueDeleter> encoder_queue =
       env.task_queue_factory().CreateTaskQueue(
           "VideoEncoderQueue",
@@ -356,13 +357,13 @@ std::unique_ptr<VideoStreamEncoderInterface> CreateVideoStreamEncoder(
               : TaskQueueFactory::Priority::kNormal);
   TaskQueueBase* encoder_queue_ptr = encoder_queue.get();
   return std::make_unique<VideoStreamEncoder>(
-      env, num_cpu_cores, stats_proxy, encoder_settings,
+      env, num_cpu_cores, stats_proxy, std::move(encoder_settings),
       std::make_unique<OveruseFrameDetector>(env, stats_proxy),
       FrameCadenceAdapterInterface::Create(
           &env.clock(), encoder_queue_ptr, metronome,
           /*worker_queue=*/TaskQueueBase::Current(), env.field_trials()),
       std::move(encoder_queue), bitrate_allocation_callback_type,
-      encoder_selector);
+      encoder_selector, std::move(encoder_switch_request_callback));
 }
 
 bool HasActiveEncodings(const VideoEncoderConfig& config) {
@@ -398,6 +399,7 @@ VideoSendStreamImpl::VideoSendStreamImpl(
     const std::map<uint32_t, RtpState>& suspended_ssrcs,
     const std::map<uint32_t, RtpPayloadState>& suspended_payload_states,
     std::unique_ptr<FecController> fec_controller,
+    EncoderSwitchRequestCallback encoder_switch_request_callback,
     std::unique_ptr<VideoStreamEncoderInterface> video_stream_encoder_for_test)
     : env_(env),
       transport_(transport),
@@ -415,11 +417,12 @@ VideoSendStreamImpl::VideoSendStreamImpl(
                     env_,
                     num_cpu_cores,
                     &stats_proxy_,
-                    config_.encoder_settings,
+                    std::move(config_.encoder_settings),
                     GetBitrateAllocationCallbackType(config_,
                                                      env_.field_trials()),
                     metronome,
-                    config_.encoder_selector)),
+                    config_.encoder_selector,
+                    std::move(encoder_switch_request_callback))),
       encoder_feedback_(
           env_,
           SupportsPerLayerPictureLossIndication(
diff --git a/video/video_send_stream_impl.h b/video/video_send_stream_impl.h
index 9ed23081fa..88845a1969 100644
--- a/video/video_send_stream_impl.h
+++ b/video/video_send_stream_impl.h
@@ -41,6 +41,7 @@
 #include "api/video/video_frame.h"
 #include "api/video/video_layers_allocation.h"
 #include "api/video/video_source_interface.h"
+#include "api/video/video_stream_encoder_settings.h"
 #include "api/video_codecs/video_encoder.h"
 #include "call/bitrate_allocator.h"
 #include "call/rtp_config.h"
@@ -88,20 +89,22 @@ class VideoSendStreamImpl : public webrtc::VideoSendStream,
   using RtpStateMap = std::map<uint32_t, RtpState>;
   using RtpPayloadStateMap = std::map<uint32_t, RtpPayloadState>;
 
-  VideoSendStreamImpl(const Environment& env,
-                      int num_cpu_cores,
-                      RtcpRttStats* call_stats,
-                      RtpTransportControllerSendInterface* transport,
-                      Metronome* metronome,
-                      BitrateAllocatorInterface* bitrate_allocator,
-                      SendDelayStats* send_delay_stats,
-                      VideoSendStream::Config config,
-                      VideoEncoderConfig encoder_config,
-                      const RtpStateMap& suspended_ssrcs,
-                      const RtpPayloadStateMap& suspended_payload_states,
-                      std::unique_ptr<FecController> fec_controller,
-                      std::unique_ptr<VideoStreamEncoderInterface>
-                          video_stream_encoder_for_test = nullptr);
+  VideoSendStreamImpl(
+      const Environment& env,
+      int num_cpu_cores,
+      RtcpRttStats* call_stats,
+      RtpTransportControllerSendInterface* transport,
+      Metronome* metronome,
+      BitrateAllocatorInterface* bitrate_allocator,
+      SendDelayStats* send_delay_stats,
+      VideoSendStream::Config config,
+      VideoEncoderConfig encoder_config,
+      const RtpStateMap& suspended_ssrcs,
+      const RtpPayloadStateMap& suspended_payload_states,
+      std::unique_ptr<FecController> fec_controller,
+      EncoderSwitchRequestCallback encoder_switch_request_callback = nullptr,
+      std::unique_ptr<VideoStreamEncoderInterface>
+          video_stream_encoder_for_test = nullptr);
   ~VideoSendStreamImpl() override;
 
   void DeliverRtcp(std::span<const uint8_t> packet);
diff --git a/video/video_send_stream_impl_unittest.cc b/video/video_send_stream_impl_unittest.cc
index 1c300242e5..4bead83d26 100644
--- a/video/video_send_stream_impl_unittest.cc
+++ b/video/video_send_stream_impl_unittest.cc
@@ -207,7 +207,9 @@ class VideoSendStreamImplTest : public ::testing::Test {
         /*metronome=*/nullptr, &bitrate_allocator_, &send_delay_stats_,
         config_.Copy(), std::move(encoder_config), suspended_ssrcs,
         suspended_payload_states,
-        /*fec_controller=*/nullptr, std::move(video_stream_encoder));
+        /*fec_controller=*/nullptr,
+        /*encoder_switch_request_callback=*/nullptr,
+        std::move(video_stream_encoder));
 
     // The call to GetStartBitrate() executes asynchronously on the tq.
     // Ensure all tasks get to run.
diff --git a/video/video_stream_encoder.cc b/video/video_stream_encoder.cc
index 3a56e77630..177ae421a6 100644
--- a/video/video_stream_encoder.cc
+++ b/video/video_stream_encoder.cc
@@ -706,23 +706,26 @@ VideoStreamEncoder::VideoStreamEncoder(
     const Environment& env,
     uint32_t number_of_cores,
     VideoStreamEncoderObserver* encoder_stats_observer,
-    const VideoStreamEncoderSettings& settings,
+    VideoStreamEncoderSettings settings,
     std::unique_ptr<OveruseFrameDetector> overuse_detector,
     std::unique_ptr<FrameCadenceAdapterInterface> frame_cadence_adapter,
     std::unique_ptr<TaskQueueBase, TaskQueueDeleter> encoder_queue,
     BitrateAllocationCallbackType allocation_cb_type,
-    VideoEncoderFactory::EncoderSelectorInterface* encoder_selector)
+    VideoEncoderFactory::EncoderSelectorInterface* encoder_selector,
+    EncoderSwitchRequestCallback encoder_switch_request_callback)
     : env_(env),
       worker_queue_(TaskQueueBase::Current()),
       number_of_cores_(number_of_cores),
-      settings_(settings),
+      settings_(std::move(settings)),
+      encoder_switch_request_callback_(
+          std::move(encoder_switch_request_callback)),
       allocation_cb_type_(allocation_cb_type),
       rate_control_settings_(env_.field_trials()),
       encoder_selector_from_constructor_(encoder_selector),
       encoder_selector_from_factory_(
           encoder_selector_from_constructor_
               ? nullptr
-              : settings.encoder_factory->GetEncoderSelector()),
+              : settings_.encoder_factory->GetEncoderSelector()),
       encoder_selector_(encoder_selector_from_constructor_
                             ? encoder_selector_from_constructor_
                             : encoder_selector_from_factory_.get()),
@@ -1568,7 +1571,7 @@ void VideoStreamEncoder::ReconfigureEncoder() {
 
 void VideoStreamEncoder::RequestEncoderSwitch() {
   bool is_encoder_switching_supported =
-      settings_.encoder_switch_request_callback != nullptr;
+      encoder_switch_request_callback_ != nullptr;
   bool is_encoder_selector_available = encoder_selector_ != nullptr;
 
   RTC_LOG(LS_INFO) << "RequestEncoderSwitch."
@@ -1591,7 +1594,8 @@ void VideoStreamEncoder::RequestEncoderSwitch() {
     if (!env_.field_trials().IsDisabled(
             kSwitchEncoderFollowCodecPreferenceOrderFieldTrial)) {
       encoder_fallback_requested_ = true;
-      settings_.encoder_switch_request_callback->RequestEncoderFallback();
+      encoder_switch_request_callback_(std::nullopt,
+                                       /*allow_default_fallback=*/false);
       return;
     } else {
       preferred_fallback_encoder =
@@ -1599,8 +1603,8 @@ void VideoStreamEncoder::RequestEncoderSwitch() {
     }
   }
 
-  settings_.encoder_switch_request_callback->RequestEncoderSwitch(
-      *preferred_fallback_encoder, /*allow_default_fallback=*/true);
+  encoder_switch_request_callback_(*preferred_fallback_encoder,
+                                   /*allow_default_fallback=*/true);
 }
 
 void VideoStreamEncoder::OnEncoderSettingsChanged() {
@@ -1936,11 +1940,11 @@ void VideoStreamEncoder::MaybeEncodeVideoFrame(const VideoFrame& video_frame,
       video_frame.is_texture() != last_frame_info_->is_texture) {
     if ((!last_frame_info_ || video_frame.width() != last_frame_info_->width ||
          video_frame.height() != last_frame_info_->height) &&
-        settings_.encoder_switch_request_callback && encoder_selector_) {
+        encoder_switch_request_callback_ && encoder_selector_) {
       if (auto encoder = encoder_selector_->OnResolutionChange(
               {video_frame.width(), video_frame.height()})) {
-        settings_.encoder_switch_request_callback->RequestEncoderSwitch(
-            *encoder, /*allow_default_fallback=*/false);
+        encoder_switch_request_callback_(*encoder,
+                                         /*allow_default_fallback=*/false);
       }
     }
 
@@ -2526,11 +2530,11 @@ void VideoStreamEncoder::OnBitrateUpdated(DataRate target_bitrate,
   const bool video_is_suspended = target_bitrate == DataRate::Zero();
   const bool video_suspension_changed = video_is_suspended != EncoderPaused();
 
-  if (!video_is_suspended && settings_.encoder_switch_request_callback &&
+  if (!video_is_suspended && encoder_switch_request_callback_ &&
       encoder_selector_) {
     if (auto encoder = encoder_selector_->OnAvailableBitrate(link_allocation)) {
-      settings_.encoder_switch_request_callback->RequestEncoderSwitch(
-          *encoder, /*allow_default_fallback=*/false);
+      encoder_switch_request_callback_(*encoder,
+                                       /*allow_default_fallback=*/false);
     }
   }
 
diff --git a/video/video_stream_encoder.h b/video/video_stream_encoder.h
index 2e28008864..f6c4ca7346 100644
--- a/video/video_stream_encoder.h
+++ b/video/video_stream_encoder.h
@@ -93,13 +93,13 @@ class VideoStreamEncoder : public VideoStreamEncoderInterface,
       const Environment& env,
       uint32_t number_of_cores,
       VideoStreamEncoderObserver* encoder_stats_observer,
-      const VideoStreamEncoderSettings& settings,
+      VideoStreamEncoderSettings settings,
       std::unique_ptr<OveruseFrameDetector> overuse_detector,
       std::unique_ptr<FrameCadenceAdapterInterface> frame_cadence_adapter,
       std::unique_ptr<TaskQueueBase, TaskQueueDeleter> encoder_queue,
       BitrateAllocationCallbackType allocation_cb_type,
-      VideoEncoderFactory::EncoderSelectorInterface* encoder_selector =
-          nullptr);
+      VideoEncoderFactory::EncoderSelectorInterface* encoder_selector = nullptr,
+      EncoderSwitchRequestCallback encoder_switch_request_callback = nullptr);
   ~VideoStreamEncoder() override;
 
   VideoStreamEncoder(const VideoStreamEncoder&) = delete;
@@ -307,6 +307,7 @@ class VideoStreamEncoder : public VideoStreamEncoderInterface,
 
   EncoderSink* sink_ = nullptr;
   const VideoStreamEncoderSettings settings_;
+  EncoderSwitchRequestCallback encoder_switch_request_callback_;
   const BitrateAllocationCallbackType allocation_cb_type_;
   const RateControlSettings rate_control_settings_;
 
diff --git a/video/video_stream_encoder_unittest.cc b/video/video_stream_encoder_unittest.cc
index 5a3eba7de6..41c91a79ff 100644
--- a/video/video_stream_encoder_unittest.cc
+++ b/video/video_stream_encoder_unittest.cc
@@ -138,6 +138,7 @@ using ::testing::Le;
 using ::testing::Lt;
 using ::testing::Matcher;
 using ::testing::Mock;
+using ::testing::MockFunction;
 using ::testing::NiceMock;
 using ::testing::Optional;
 using ::testing::Return;
@@ -440,21 +441,24 @@ class VideoStreamEncoderUnderTest : public VideoStreamEncoder {
       std::unique_ptr<FrameCadenceAdapterInterface> cadence_adapter,
       std::unique_ptr<TaskQueueBase, TaskQueueDeleter> encoder_queue,
       SendStatisticsProxy* stats_proxy,
-      const VideoStreamEncoderSettings& settings,
+      VideoStreamEncoderSettings settings,
       VideoStreamEncoder::BitrateAllocationCallbackType
           allocation_callback_type,
-      int num_cores)
+      int num_cores,
+      EncoderSwitchRequestCallback encoder_switch_request_callback = nullptr)
       : VideoStreamEncoder(
             env,
             num_cores,
             stats_proxy,
-            settings,
+            std::move(settings),
             std::unique_ptr<OveruseFrameDetector>(
                 overuse_detector_proxy_ =
                     new CpuOveruseDetectorProxy(env, stats_proxy)),
             std::move(cadence_adapter),
             std::move(encoder_queue),
-            allocation_callback_type),
+            allocation_callback_type,
+            nullptr,  // encoder_selector
+            std::move(encoder_switch_request_callback)),
         time_controller_(time_controller),
         fake_cpu_resource_(FakeResource::Create("FakeResource[CPU]")),
         fake_quality_resource_(FakeResource::Create("FakeResource[QP]")),
@@ -760,7 +764,7 @@ class SimpleVideoStreamEncoderFactory {
     auto result = std::make_unique<AdaptedVideoStreamEncoder>(
         env_,
         /*number_of_cores=*/1,
-        /*stats_proxy=*/stats_proxy_.get(), encoder_settings_,
+        /*stats_proxy=*/stats_proxy_.get(), std::move(encoder_settings_),
         std::make_unique<CpuOveruseDetectorProxy>(env_,
                                                   /*stats_proxy=*/nullptr),
         std::move(zero_hertz_adapter), std::move(encoder_queue),
@@ -902,7 +906,8 @@ class VideoStreamEncoderTest : public ::testing::Test {
             video_send_config_,
             VideoEncoderConfig::ContentType::kRealtimeVideo,
             field_trials_)),
-        sink_(&time_controller_, &fake_encoder_) {}
+        sink_(&time_controller_, &fake_encoder_),
+        encoder_switch_request_callback_(nullptr) {}
 
   void SetUp() override {
     metrics::Reset();
@@ -953,11 +958,12 @@ class VideoStreamEncoderTest : public ::testing::Test {
       env_ = factory.Create();
     }
 
+    VideoStreamEncoderSettings settings = video_send_config_.encoder_settings;
     video_stream_encoder_ = std::make_unique<VideoStreamEncoderUnderTest>(
         env_, &time_controller_, std::move(cadence_adapter),
-        std::move(encoder_queue), stats_proxy_.get(),
-        video_send_config_.encoder_settings, allocation_callback_type,
-        num_cores);
+        std::move(encoder_queue), stats_proxy_.get(), std::move(settings),
+        allocation_callback_type, num_cores,
+        std::move(encoder_switch_request_callback_));
     video_stream_encoder_->SetSink(&sink_, /*rotation_applied=*/false);
     video_stream_encoder_->SetSource(&video_source_,
                                      DegradationPreference::MAINTAIN_FRAMERATE);
@@ -1770,6 +1776,7 @@ class VideoStreamEncoderTest : public ::testing::Test {
   TestSink sink_;
   AdaptingFrameForwarder video_source_{&time_controller_};
   std::unique_ptr<VideoStreamEncoderUnderTest> video_stream_encoder_;
+  EncoderSwitchRequestCallback encoder_switch_request_callback_;
 };
 
 TEST_F(VideoStreamEncoderTest, EncodeOneFrame) {
@@ -8429,13 +8436,6 @@ TEST_F(VideoStreamEncoderTest, EncoderRatesPropagatedOnReconfigure) {
   video_stream_encoder_->Stop();
 }
 
-struct MockEncoderSwitchRequestCallback : public EncoderSwitchRequestCallback {
-  MOCK_METHOD(void, RequestEncoderFallback, (), (override));
-  MOCK_METHOD(void,
-              RequestEncoderSwitch,
-              (const SdpVideoFormat& format, bool allow_default_fallback),
-              (override));
-};
 
 TEST_F(VideoStreamEncoderTest, EncoderSelectorCurrentEncoderIsSignaled) {
   constexpr int kDontCare = 100;
@@ -8465,9 +8465,9 @@ TEST_F(VideoStreamEncoderTest, EncoderSelectorBitrateSwitch) {
   constexpr int kDontCare = 100;
 
   NiceMock<MockEncoderSelector> encoder_selector;
-  StrictMock<MockEncoderSwitchRequestCallback> switch_callback;
-  video_send_config_.encoder_settings.encoder_switch_request_callback =
-      &switch_callback;
+  StrictMock<MockFunction<void(std::optional<SdpVideoFormat>, bool)>>
+      switch_callback;
+  encoder_switch_request_callback_ = switch_callback.AsStdFunction();
   auto encoder_factory = std::make_unique<test::VideoEncoderProxyFactory>(
       &fake_encoder_, &encoder_selector);
   video_send_config_.encoder_settings.encoder_factory = encoder_factory.get();
@@ -8478,8 +8478,8 @@ TEST_F(VideoStreamEncoderTest, EncoderSelectorBitrateSwitch) {
   ON_CALL(encoder_selector, OnAvailableBitrate)
       .WillByDefault(Return(SdpVideoFormat("AV1")));
   EXPECT_CALL(switch_callback,
-              RequestEncoderSwitch(Field(&SdpVideoFormat::name, "AV1"),
-                                   /*allow_default_fallback=*/false));
+              Call(Optional(Field(&SdpVideoFormat::name, "AV1")),
+                   /*allow_default_fallback=*/false));
 
   video_stream_encoder_->OnBitrateUpdatedAndWaitForManagedResources(
       /*target_bitrate=*/DataRate::KilobitsPerSec(50),
@@ -8495,9 +8495,9 @@ TEST_F(VideoStreamEncoderTest, EncoderSelectorBitrateSwitch) {
 
 TEST_F(VideoStreamEncoderTest, EncoderSelectorResolutionSwitch) {
   NiceMock<MockEncoderSelector> encoder_selector;
-  StrictMock<MockEncoderSwitchRequestCallback> switch_callback;
-  video_send_config_.encoder_settings.encoder_switch_request_callback =
-      &switch_callback;
+  StrictMock<MockFunction<void(std::optional<SdpVideoFormat>, bool)>>
+      switch_callback;
+  encoder_switch_request_callback_ = switch_callback.AsStdFunction();
   auto encoder_factory = std::make_unique<test::VideoEncoderProxyFactory>(
       &fake_encoder_, &encoder_selector);
   video_send_config_.encoder_settings.encoder_factory = encoder_factory.get();
@@ -8510,8 +8510,8 @@ TEST_F(VideoStreamEncoderTest, EncoderSelectorResolutionSwitch) {
   EXPECT_CALL(encoder_selector, OnResolutionChange(RenderResolution(320, 240)))
       .WillOnce(Return(SdpVideoFormat("AV1")));
   EXPECT_CALL(switch_callback,
-              RequestEncoderSwitch(Field(&SdpVideoFormat::name, "AV1"),
-                                   /*allow_default_fallback=*/false));
+              Call(Optional(Field(&SdpVideoFormat::name, "AV1")),
+                   /*allow_default_fallback=*/false));
 
   video_stream_encoder_->OnBitrateUpdatedAndWaitForManagedResources(
       /*target_bitrate=*/DataRate::KilobitsPerSec(800),
@@ -8536,9 +8536,9 @@ TEST_F(VideoStreamEncoderTest, EncoderSelectorBrokenEncoderSwitch) {
 
   NiceMock<MockVideoEncoder> video_encoder;
   NiceMock<MockEncoderSelector> encoder_selector;
-  StrictMock<MockEncoderSwitchRequestCallback> switch_callback;
-  video_send_config_.encoder_settings.encoder_switch_request_callback =
-      &switch_callback;
+  StrictMock<MockFunction<void(std::optional<SdpVideoFormat>, bool)>>
+      switch_callback;
+  encoder_switch_request_callback_ = switch_callback.AsStdFunction();
   auto encoder_factory = std::make_unique<test::VideoEncoderProxyFactory>(
       &video_encoder, &encoder_selector);
   video_send_config_.encoder_settings.encoder_factory = encoder_factory.get();
@@ -8564,8 +8564,8 @@ TEST_F(VideoStreamEncoderTest, EncoderSelectorBrokenEncoderSwitch) {
 
   Event encode_attempted;
   EXPECT_CALL(switch_callback,
-              RequestEncoderSwitch(Field(&SdpVideoFormat::name, "AV2"),
-                                   /*allow_default_fallback=*/true))
+              Call(Optional(Field(&SdpVideoFormat::name, "AV2")),
+                   /*allow_default_fallback=*/true))
       .WillOnce([&encode_attempted]() { encode_attempted.Set(); });
 
   video_source_.IncomingCapturedFrame(CreateFrame(1, kDontCare, kDontCare));
@@ -8585,9 +8585,9 @@ TEST_F(VideoStreamEncoderTest, EncoderSelectorBrokenEncoderSwitch) {
 TEST_F(VideoStreamEncoderTest, SwitchEncoderOnInitFailureWithEncoderSelector) {
   NiceMock<MockVideoEncoder> video_encoder;
   NiceMock<MockEncoderSelector> encoder_selector;
-  StrictMock<MockEncoderSwitchRequestCallback> switch_callback;
-  video_send_config_.encoder_settings.encoder_switch_request_callback =
-      &switch_callback;
+  StrictMock<MockFunction<void(std::optional<SdpVideoFormat>, bool)>>
+      switch_callback;
+  encoder_switch_request_callback_ = switch_callback.AsStdFunction();
   auto encoder_factory = std::make_unique<test::VideoEncoderProxyFactory>(
       &video_encoder, &encoder_selector);
   video_send_config_.encoder_settings.encoder_factory = encoder_factory.get();
@@ -8608,8 +8608,8 @@ TEST_F(VideoStreamEncoderTest, SwitchEncoderOnInitFailureWithEncoderSelector) {
 
   Event encode_attempted;
   EXPECT_CALL(switch_callback,
-              RequestEncoderSwitch(Field(&SdpVideoFormat::name, "AV2"),
-                                   /*allow_default_fallback=*/true))
+              Call(Optional(Field(&SdpVideoFormat::name, "AV2")),
+                   /*allow_default_fallback=*/true))
       .WillOnce([&encode_attempted]() { encode_attempted.Set(); });
 
   video_source_.IncomingCapturedFrame(CreateFrame(1, nullptr));
@@ -8632,9 +8632,9 @@ TEST_F(VideoStreamEncoderTest,
       "WebRTC-SwitchEncoderFollowCodecPreferenceOrder", "Disabled");
 
   NiceMock<MockVideoEncoder> video_encoder;
-  StrictMock<MockEncoderSwitchRequestCallback> switch_callback;
-  video_send_config_.encoder_settings.encoder_switch_request_callback =
-      &switch_callback;
+  StrictMock<MockFunction<void(std::optional<SdpVideoFormat>, bool)>>
+      switch_callback;
+  encoder_switch_request_callback_ = switch_callback.AsStdFunction();
   auto encoder_factory = std::make_unique<test::VideoEncoderProxyFactory>(
       &video_encoder, /*encoder_selector=*/nullptr);
   video_send_config_.encoder_settings.encoder_factory = encoder_factory.get();
@@ -8656,8 +8656,8 @@ TEST_F(VideoStreamEncoderTest,
 
   Event encode_attempted;
   EXPECT_CALL(switch_callback,
-              RequestEncoderSwitch(Field(&SdpVideoFormat::name, "VP8"),
-                                   /*allow_default_fallback=*/true))
+              Call(Optional(Field(&SdpVideoFormat::name, "VP8")),
+                   /*allow_default_fallback=*/true))
       .WillOnce([&encode_attempted]() { encode_attempted.Set(); });
 
   video_source_.IncomingCapturedFrame(CreateFrame(1, nullptr));
@@ -8682,9 +8682,9 @@ TEST_F(VideoStreamEncoderTest, NullEncoderReturnSwitch) {
   constexpr int kDontCare = 100;
 
   NiceMock<MockEncoderSelector> encoder_selector;
-  StrictMock<MockEncoderSwitchRequestCallback> switch_callback;
-  video_send_config_.encoder_settings.encoder_switch_request_callback =
-      &switch_callback;
+  StrictMock<MockFunction<void(std::optional<SdpVideoFormat>, bool)>>
+      switch_callback;
+  encoder_switch_request_callback_ = switch_callback.AsStdFunction();
   auto encoder_factory =
       std::make_unique<test::VideoEncoderNullableProxyFactory>(
           /*encoder=*/nullptr, &encoder_selector);
@@ -8706,8 +8706,8 @@ TEST_F(VideoStreamEncoderTest, NullEncoderReturnSwitch) {
       .WillByDefault(Return(SdpVideoFormat("AV2")));
   Event encode_attempted;
   EXPECT_CALL(switch_callback,
-              RequestEncoderSwitch(Field(&SdpVideoFormat::name, "AV2"),
-                                   /*allow_default_fallback=*/_))
+              Call(Optional(Field(&SdpVideoFormat::name, "AV2")),
+                   /*allow_default_fallback=*/_))
       .WillOnce([&encode_attempted]() { encode_attempted.Set(); });
 
   video_source_.IncomingCapturedFrame(CreateFrame(1, kDontCare, kDontCare));
@@ -8730,9 +8730,9 @@ TEST_F(VideoStreamEncoderTest, NoPreferenceDefaultFallbackToVP8Disabled) {
   constexpr int kNumFrames = 8;
 
   NiceMock<MockVideoEncoder> video_encoder;
-  StrictMock<MockEncoderSwitchRequestCallback> switch_callback;
-  video_send_config_.encoder_settings.encoder_switch_request_callback =
-      &switch_callback;
+  StrictMock<MockFunction<void(std::optional<SdpVideoFormat>, bool)>>
+      switch_callback;
+  encoder_switch_request_callback_ = switch_callback.AsStdFunction();
   auto encoder_factory = std::make_unique<test::VideoEncoderProxyFactory>(
       &video_encoder, /*encoder_selector=*/nullptr);
   video_send_config_.encoder_settings.encoder_factory = encoder_factory.get();
@@ -8754,7 +8754,7 @@ TEST_F(VideoStreamEncoderTest, NoPreferenceDefaultFallbackToVP8Disabled) {
   EXPECT_CALL(video_encoder, Encode)
       .WillOnce(Return(WEBRTC_VIDEO_CODEC_ENCODER_FAILURE));
 
-  EXPECT_CALL(switch_callback, RequestEncoderFallback());
+  EXPECT_CALL(switch_callback, Call(Eq(std::nullopt), _));
 
   VideoFrame frame = CreateFrame(1, kDontCare, kDontCare);
   for (int i = 0; i < kNumFrames; ++i) {
@@ -8791,9 +8791,9 @@ TEST_F(VideoStreamEncoderTest,
       "WebRTC-SwitchEncoderFollowCodecPreferenceOrder", "Enabled");
 
   NiceMock<MockVideoEncoder> video_encoder;
-  StrictMock<MockEncoderSwitchRequestCallback> switch_callback;
-  video_send_config_.encoder_settings.encoder_switch_request_callback =
-      &switch_callback;
+  StrictMock<MockFunction<void(std::optional<SdpVideoFormat>, bool)>>
+      switch_callback;
+  encoder_switch_request_callback_ = switch_callback.AsStdFunction();
   auto encoder_factory = std::make_unique<test::VideoEncoderProxyFactory>(
       &video_encoder, /*encoder_selector=*/nullptr);
   video_send_config_.encoder_settings.encoder_factory = encoder_factory.get();
@@ -8820,7 +8820,7 @@ TEST_F(VideoStreamEncoderTest,
       .WillOnce(Return(WEBRTC_VIDEO_CODEC_ENCODER_FAILURE))
       .WillRepeatedly(Return(WEBRTC_VIDEO_CODEC_OK));
 
-  EXPECT_CALL(switch_callback, RequestEncoderFallback());
+  EXPECT_CALL(switch_callback, Call(Eq(std::nullopt), _));
 
   // Encode() will be called once and will return a failure code. All subsequent
   // frames will be dropped.
@@ -8866,9 +8866,9 @@ TEST_F(VideoStreamEncoderTest, NoPreferenceDefaultFallbackToVP8Enabled) {
       "WebRTC-SwitchEncoderFollowCodecPreferenceOrder", "Disabled");
 
   NiceMock<MockVideoEncoder> video_encoder;
-  StrictMock<MockEncoderSwitchRequestCallback> switch_callback;
-  video_send_config_.encoder_settings.encoder_switch_request_callback =
-      &switch_callback;
+  StrictMock<MockFunction<void(std::optional<SdpVideoFormat>, bool)>>
+      switch_callback;
+  encoder_switch_request_callback_ = switch_callback.AsStdFunction();
   auto encoder_factory = std::make_unique<test::VideoEncoderProxyFactory>(
       &video_encoder, /*encoder_selector=*/nullptr);
   video_send_config_.encoder_settings.encoder_factory = encoder_factory.get();
@@ -8896,8 +8896,8 @@ TEST_F(VideoStreamEncoderTest, NoPreferenceDefaultFallbackToVP8Enabled) {
 
   // Fallback request will be asking for switching to VP8.
   EXPECT_CALL(switch_callback,
-              RequestEncoderSwitch(Field(&SdpVideoFormat::name, "VP8"),
-                                   /*allow_default_fallback=*/true));
+              Call(Optional(Field(&SdpVideoFormat::name, "VP8")),
+                   /*allow_default_fallback=*/true));
 
   VideoFrame frame = CreateFrame(1, kDontCare, kDontCare);
   video_source_.IncomingCapturedFrame(frame);
@@ -10433,7 +10433,7 @@ TEST(VideoStreamEncoderSimpleTest, CreateDestroy) {
   // doing anything else. This should be fine since the posted init task will
   // simply be deleted.
   VideoStreamEncoder encoder(
-      env, 1, &stats_proxy, encoder_settings,
+      env, 1, &stats_proxy, std::move(encoder_settings),
       std::make_unique<CpuOveruseDetectorProxy>(env, &stats_proxy),
       std::move(adapter), std::move(encoder_queue),
       VideoStreamEncoder::BitrateAllocationCallbackType::

From eb887af623d901dc7e913bfa683b661117d0fad3 Mon Sep 17 00:00:00 2001
From: Ilya Nikolaevskiy <ilnik@webrtc.org>
Date: Mon, 9 Mar 2026 12:52:14 +0100
Subject: [PATCH 235/472] Remove deprecated charset handling in JniHelper
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Chromium->Webrtc roll is blocked due to compile error here.

Bug: none
Change-Id: Ibe76b7480309841ffee0efe2bc85de453c6b7ee9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/455201
Commit-Queue: Zoé Lepaul <zlep@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Zoé Lepaul <zlep@webrtc.org>
Auto-Submit: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47088}
---
 sdk/android/src/java/org/webrtc/JniHelper.java | 7 ++-----
 1 file changed, 2 insertions(+), 5 deletions(-)

diff --git a/sdk/android/src/java/org/webrtc/JniHelper.java b/sdk/android/src/java/org/webrtc/JniHelper.java
index 0d56d5d92b..9cacd5f0cc 100644
--- a/sdk/android/src/java/org/webrtc/JniHelper.java
+++ b/sdk/android/src/java/org/webrtc/JniHelper.java
@@ -11,6 +11,7 @@
 package org.webrtc;
 
 import java.io.UnsupportedEncodingException;
+import java.nio.charset.StandardCharsets;
 import java.util.Map;
 
 /**
@@ -21,11 +22,7 @@ class JniHelper {
   // TODO(bugs.webrtc.org/8632): Remove.
   @CalledByNative
   static byte[] getStringBytes(String s) {
-    try {
-      return s.getBytes("ISO-8859-1");
-    } catch (UnsupportedEncodingException e) {
-      throw new RuntimeException("ISO-8859-1 is unsupported");
-    }
+    return s.getBytes(StandardCharsets.ISO_8859_1);
   }
 
   // TODO(bugs.webrtc.org/8632): Remove.

From c45d806c09ab232372d52d962a050838bdf0ee77 Mon Sep 17 00:00:00 2001
From: Philipp Hancke <philipp.hancke@googlemail.com>
Date: Tue, 27 Jan 2026 16:10:09 +0100
Subject: [PATCH 236/472] Reland "Reland "sdk: add RTC_OBJC_TYPE to ios Metal
 renderer classes""
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This is a reland of commit 14e19b2f6ce0e284b186007124383724df0162d9

Original change's description:
> Reland "sdk: add RTC_OBJC_TYPE to ios Metal renderer classes"
>
> This is a reland of commit 89e8619bfdacfd1364bf3fc6bc5388c92629ab71
>
> Original change's description:
> > sdk: add RTC_OBJC_TYPE to ios Metal renderer classes
> >
> > Bug: webrtc:460166677
> > Change-Id: Id8883d40957e26046897632c4f6ed2d9c36bcf4e
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/427320
> > Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> > Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#46309}
>
> Bug: webrtc:460166677
> Change-Id: Ia31eec951ca8c3dbe2389b8b0244ce0d540bfcc1
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/430900
> Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#46595}

Bug: webrtc:460166677
Change-Id: I399d333f729127d8a45795b7b60b9a96511a9c17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/442282
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#47089}
---
 .../renderer/metal/RTCMTLI420Renderer.h       |  2 +-
 .../renderer/metal/RTCMTLI420Renderer.mm      |  2 +-
 .../renderer/metal/RTCMTLNSVideoView.m        |  4 +--
 .../renderer/metal/RTCMTLNV12Renderer.h       |  2 +-
 .../renderer/metal/RTCMTLNV12Renderer.mm      |  2 +-
 .../renderer/metal/RTCMTLRGBRenderer.h        |  2 +-
 .../renderer/metal/RTCMTLRGBRenderer.mm       |  2 +-
 .../renderer/metal/RTCMTLRenderer+Private.h   |  4 +--
 .../renderer/metal/RTCMTLRenderer.h           | 12 ++++---
 .../renderer/metal/RTCMTLRenderer.mm          |  2 +-
 .../renderer/metal/RTCMTLVideoView.m          | 33 ++++++++++++-------
 sdk/objc/unittests/RTCMTLVideoView_xctest.m   |  6 ++--
 12 files changed, 43 insertions(+), 30 deletions(-)

diff --git a/sdk/objc/components/renderer/metal/RTCMTLI420Renderer.h b/sdk/objc/components/renderer/metal/RTCMTLI420Renderer.h
index e5987fe22a..1c45ab90b5 100644
--- a/sdk/objc/components/renderer/metal/RTCMTLI420Renderer.h
+++ b/sdk/objc/components/renderer/metal/RTCMTLI420Renderer.h
@@ -13,5 +13,5 @@
 #import "RTCMTLRenderer.h"
 
 NS_AVAILABLE(10_11, 9_0)
-@interface RTCMTLI420Renderer : RTCMTLRenderer
+@interface RTC_OBJC_TYPE (RTCMTLI420Renderer) : RTC_OBJC_TYPE (RTCMTLRenderer)
 @end
diff --git a/sdk/objc/components/renderer/metal/RTCMTLI420Renderer.mm b/sdk/objc/components/renderer/metal/RTCMTLI420Renderer.mm
index c085b9260f..c162b4c399 100644
--- a/sdk/objc/components/renderer/metal/RTCMTLI420Renderer.mm
+++ b/sdk/objc/components/renderer/metal/RTCMTLI420Renderer.mm
@@ -70,7 +70,7 @@ static NSString *const shaderSource = MTL_STRINGIFY(
       return half4(out);
     });
 
-@implementation RTCMTLI420Renderer {
+@implementation RTC_OBJC_TYPE (RTCMTLI420Renderer) {
   // Textures.
   id<MTLTexture> _yTexture;
   id<MTLTexture> _uTexture;
diff --git a/sdk/objc/components/renderer/metal/RTCMTLNSVideoView.m b/sdk/objc/components/renderer/metal/RTCMTLNSVideoView.m
index 168045c09d..0c5a21391f 100644
--- a/sdk/objc/components/renderer/metal/RTCMTLNSVideoView.m
+++ b/sdk/objc/components/renderer/metal/RTCMTLNSVideoView.m
@@ -24,7 +24,7 @@
 @end
 
 @implementation RTC_OBJC_TYPE (RTCMTLNSVideoView) {
-  id<RTCMTLRenderer> _renderer;
+  id<RTC_OBJC_TYPE(RTCMTLRenderer)> _renderer;
 }
 
 @synthesize delegate = _delegate;
@@ -64,7 +64,7 @@
     _metalView.framebufferOnly = YES;
     _metalView.delegate = self;
 
-    _renderer = [[RTCMTLI420Renderer alloc] init];
+    _renderer = [[RTC_OBJC_TYPE(RTCMTLI420Renderer) alloc] init];
     if (![(RTCMTLI420Renderer *)_renderer addRenderingDestination:_metalView]) {
       _renderer = nil;
     };
diff --git a/sdk/objc/components/renderer/metal/RTCMTLNV12Renderer.h b/sdk/objc/components/renderer/metal/RTCMTLNV12Renderer.h
index 866b7ea17e..b6edac2e77 100644
--- a/sdk/objc/components/renderer/metal/RTCMTLNV12Renderer.h
+++ b/sdk/objc/components/renderer/metal/RTCMTLNV12Renderer.h
@@ -13,6 +13,6 @@
 #import "RTCMTLRenderer.h"
 
 NS_AVAILABLE(10_11, 9_0)
-@interface RTCMTLNV12Renderer : RTCMTLRenderer
+@interface RTC_OBJC_TYPE (RTCMTLNV12Renderer) : RTC_OBJC_TYPE (RTCMTLRenderer)
 
 @end
diff --git a/sdk/objc/components/renderer/metal/RTCMTLNV12Renderer.mm b/sdk/objc/components/renderer/metal/RTCMTLNV12Renderer.mm
index 2849e0fef0..f1f9bb85e6 100644
--- a/sdk/objc/components/renderer/metal/RTCMTLNV12Renderer.mm
+++ b/sdk/objc/components/renderer/metal/RTCMTLNV12Renderer.mm
@@ -63,7 +63,7 @@ static NSString *const shaderSource = MTL_STRINGIFY(
       return half4(out);
     });
 
-@implementation RTCMTLNV12Renderer {
+@implementation RTC_OBJC_TYPE (RTCMTLNV12Renderer) {
   // Textures.
   CVMetalTextureCacheRef _textureCache;
   id<MTLTexture> _yTexture;
diff --git a/sdk/objc/components/renderer/metal/RTCMTLRGBRenderer.h b/sdk/objc/components/renderer/metal/RTCMTLRGBRenderer.h
index 9db422cd22..9acc75af1b 100644
--- a/sdk/objc/components/renderer/metal/RTCMTLRGBRenderer.h
+++ b/sdk/objc/components/renderer/metal/RTCMTLRGBRenderer.h
@@ -17,6 +17,6 @@
  * kCVPixelFormatType_32ARGB.
  */
 NS_AVAILABLE(10_11, 9_0)
-@interface RTCMTLRGBRenderer : RTCMTLRenderer
+@interface RTC_OBJC_TYPE (RTCMTLRGBRenderer) : RTC_OBJC_TYPE (RTCMTLRenderer)
 
 @end
diff --git a/sdk/objc/components/renderer/metal/RTCMTLRGBRenderer.mm b/sdk/objc/components/renderer/metal/RTCMTLRGBRenderer.mm
index df59c6fdfd..71a8270a5c 100644
--- a/sdk/objc/components/renderer/metal/RTCMTLRGBRenderer.mm
+++ b/sdk/objc/components/renderer/metal/RTCMTLRGBRenderer.mm
@@ -57,7 +57,7 @@ static NSString *const shaderSource = MTL_STRINGIFY(
       return out;
     });
 
-@implementation RTCMTLRGBRenderer {
+@implementation RTC_OBJC_TYPE (RTCMTLRGBRenderer) {
   // Textures.
   CVMetalTextureCacheRef _textureCache;
   id<MTLTexture> _texture;
diff --git a/sdk/objc/components/renderer/metal/RTCMTLRenderer+Private.h b/sdk/objc/components/renderer/metal/RTCMTLRenderer+Private.h
index 8e99fc4614..77dab1e57d 100644
--- a/sdk/objc/components/renderer/metal/RTCMTLRenderer+Private.h
+++ b/sdk/objc/components/renderer/metal/RTCMTLRenderer+Private.h
@@ -16,8 +16,8 @@
 
 NS_ASSUME_NONNULL_BEGIN
 
-@interface RTCMTLRenderer (Private)
-- (nullable id<MTLDevice>)currentMetalDevice;
+@interface RTC_OBJC_TYPE (RTCMTLRenderer)
+(Private) - (nullable id<MTLDevice>)currentMetalDevice;
 - (NSString *)shaderSource;
 - (BOOL)setupTexturesForFrame:(nonnull RTC_OBJC_TYPE(RTCVideoFrame) *)frame;
 - (void)uploadTexturesToRenderEncoder:
diff --git a/sdk/objc/components/renderer/metal/RTCMTLRenderer.h b/sdk/objc/components/renderer/metal/RTCMTLRenderer.h
index 84aa3fe7fc..f513fccba1 100644
--- a/sdk/objc/components/renderer/metal/RTCMTLRenderer.h
+++ b/sdk/objc/components/renderer/metal/RTCMTLRenderer.h
@@ -21,19 +21,21 @@ NS_ASSUME_NONNULL_BEGIN
 /**
  * Protocol defining ability to render RTCVideoFrame in Metal enabled views.
  */
-@protocol RTCMTLRenderer <NSObject>
+@protocol RTC_OBJC_TYPE
+(RTCMTLRenderer)<NSObject>
 
 /**
- * Method to be implemented to perform actual rendering of the provided frame.
+ * Method to be implemented to perform actual rendering of the provided
+ * frame.
  *
  * @param frame The frame to be rendered.
  */
-- (void)drawFrame:(RTC_OBJC_TYPE(RTCVideoFrame) *)frame;
+- (void)drawFrame : (RTC_OBJC_TYPE(RTCVideoFrame) *)frame;
 
 /**
  * Sets the provided view as rendering destination if possible.
  *
- * If not possible method returns NO and callers of the method are responisble
+ * If not possible method returns NO and callers of the method are responsible
  * for performing cleanups.
  */
 
@@ -49,7 +51,7 @@ NS_ASSUME_NONNULL_BEGIN
  * Implementation of RTCMTLRenderer protocol.
  */
 NS_AVAILABLE(10_11, 9_0)
-@interface RTCMTLRenderer : NSObject <RTCMTLRenderer>
+@interface RTC_OBJC_TYPE (RTCMTLRenderer) : NSObject <RTC_OBJC_TYPE(RTCMTLRenderer)>
 
 /** @abstract   A wrapped RTCVideoRotation, or nil.
     @discussion When not nil, the rotation of the actual frame is ignored when
diff --git a/sdk/objc/components/renderer/metal/RTCMTLRenderer.mm b/sdk/objc/components/renderer/metal/RTCMTLRenderer.mm
index 5323633415..a8b00c9334 100644
--- a/sdk/objc/components/renderer/metal/RTCMTLRenderer.mm
+++ b/sdk/objc/components/renderer/metal/RTCMTLRenderer.mm
@@ -136,7 +136,7 @@ static inline void getCubeVertexData(int cropX,
 // In future we might use triple buffering method if it improves performance.
 static const NSInteger kMaxInflightBuffers = 1;
 
-@implementation RTCMTLRenderer {
+@implementation RTC_OBJC_TYPE (RTCMTLRenderer) {
   __kindof MTKView *_view;
 
   // Controller.
diff --git a/sdk/objc/components/renderer/metal/RTCMTLVideoView.m b/sdk/objc/components/renderer/metal/RTCMTLVideoView.m
index 7b1755dc6c..cce13a1f8d 100644
--- a/sdk/objc/components/renderer/metal/RTCMTLVideoView.m
+++ b/sdk/objc/components/renderer/metal/RTCMTLVideoView.m
@@ -25,15 +25,26 @@
 // To avoid unreconized symbol linker errors, we're taking advantage of the objc
 // runtime. Linking errors occur when compiling for architectures that don't
 // support Metal.
+#define RTC_OBJC_TYPE_PREFIX_STRING_HELPER(x) #x
+#define RTC_OBJC_TYPE_PREFIX_STRING(x) \
+  RTC_OBJC_TYPE_PREFIX_STRING_HELPER(x)
+
 #define MTKViewClass NSClassFromString(@"MTKView")
-#define RTCMTLNV12RendererClass NSClassFromString(@"RTCMTLNV12Renderer")
-#define RTCMTLI420RendererClass NSClassFromString(@"RTCMTLI420Renderer")
-#define RTCMTLRGBRendererClass NSClassFromString(@"RTCMTLRGBRenderer")
+#define RTCMTLNV12RendererClass                      \
+  NSClassFromString(@"" RTC_OBJC_TYPE_PREFIX_STRING( \
+      RTC_OBJC_TYPE_PREFIX) @"RTCMTLNV12Renderer")
+#define RTCMTLI420RendererClass                      \
+  NSClassFromString(@"" RTC_OBJC_TYPE_PREFIX_STRING( \
+      RTC_OBJC_TYPE_PREFIX) @"RTCMTLI420Renderer")
+#define RTCMTLRGBRendererClass                       \
+  NSClassFromString(@"" RTC_OBJC_TYPE_PREFIX_STRING( \
+      RTC_OBJC_TYPE_PREFIX) @"RTCMTLRGBRenderer")
 
 @interface RTC_OBJC_TYPE (RTCMTLVideoView)
-()<MTKViewDelegate> @property(nonatomic) RTCMTLI420Renderer *rendererI420;
-@property(nonatomic) RTCMTLNV12Renderer *rendererNV12;
-@property(nonatomic) RTCMTLRGBRenderer *rendererRGB;
+()<MTKViewDelegate> @property(nonatomic) RTC_OBJC_TYPE(RTCMTLI420Renderer) *
+    rendererI420;
+@property(nonatomic) RTC_OBJC_TYPE(RTCMTLNV12Renderer) * rendererNV12;
+@property(nonatomic) RTC_OBJC_TYPE(RTCMTLRGBRenderer) * rendererRGB;
 @property(nonatomic) MTKView *metalView;
 @property(atomic) RTC_OBJC_TYPE(RTCVideoFrame) * videoFrame;
 @property(nonatomic) CGSize videoFrameSize;
@@ -94,16 +105,16 @@
   return [[MTKViewClass alloc] initWithFrame:frame];
 }
 
-+ (RTCMTLNV12Renderer *)createNV12Renderer {
++ (RTC_OBJC_TYPE(RTCMTLNV12Renderer) *)createNV12Renderer {
   return [[RTCMTLNV12RendererClass alloc] init];
 }
 
-+ (RTCMTLI420Renderer *)createI420Renderer {
++ (RTC_OBJC_TYPE(RTCMTLI420Renderer) *)createI420Renderer {
   return [[RTCMTLI420RendererClass alloc] init];
 }
 
-+ (RTCMTLRGBRenderer *)createRGBRenderer {
-  return [[RTCMTLRGBRenderer alloc] init];
++ (RTC_OBJC_TYPE(RTCMTLRGBRenderer) *)createRGBRenderer {
+  return [[RTCMTLRGBRendererClass alloc] init];
 }
 
 - (void)configure {
@@ -150,7 +161,7 @@
     return;
   }
 
-  RTCMTLRenderer *renderer;
+  RTC_OBJC_TYPE(RTCMTLRenderer) * renderer;
   if ([videoFrame.buffer
           isKindOfClass:[RTC_OBJC_TYPE(RTCCVPixelBuffer) class]]) {
     RTC_OBJC_TYPE(RTCCVPixelBuffer) *buffer =
diff --git a/sdk/objc/unittests/RTCMTLVideoView_xctest.m b/sdk/objc/unittests/RTCMTLVideoView_xctest.m
index fc140a4662..437f094046 100644
--- a/sdk/objc/unittests/RTCMTLVideoView_xctest.m
+++ b/sdk/objc/unittests/RTCMTLVideoView_xctest.m
@@ -32,8 +32,8 @@ static size_t kBufferHeight = 200;
 
 + (BOOL)isMetalAvailable;
 + (UIView *)createMetalView:(CGRect)frame;
-+ (id<RTCMTLRenderer>)createNV12Renderer;
-+ (id<RTCMTLRenderer>)createI420Renderer;
++ (id<RTC_OBJC_TYPE(RTCMTLRenderer)>)createNV12Renderer;
++ (id<RTC_OBJC_TYPE(RTCMTLRenderer)>)createI420Renderer;
 - (void)drawInMTKView:(id)view;
 @end
 
@@ -95,7 +95,7 @@ static size_t kBufferHeight = 200;
 }
 
 - (id)rendererMockWithSuccessfulSetup:(BOOL)success {
-  id rendererMock = OCMClassMock([RTCMTLRenderer class]);
+  id rendererMock = OCMClassMock([RTC_OBJC_TYPE(RTCMTLRenderer) class]);
   OCMStub([rendererMock addRenderingDestination:[OCMArg any]])
       .andReturn(success);
   return rendererMock;

From a97bb5fc837c0043dc9707e4c50cdf71a30b6dc3 Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Fri, 6 Mar 2026 15:38:49 +0100
Subject: [PATCH 237/472] Replace ArrayView with std::span in p2p/

Bug: webrtc:439801349
Change-Id: I8eba2062fae69944dab44afb6459c45427deafa3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/454302
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47090}
---
 p2p/BUILD.gn                                  | 18 ------------
 p2p/base/async_stun_tcp_socket.cc             |  6 ++--
 p2p/base/async_stun_tcp_socket.h              |  4 +--
 p2p/base/async_stun_tcp_socket_unittest.cc    |  1 -
 p2p/base/basic_ice_controller.h               |  4 +--
 p2p/base/connection.cc                        |  4 +--
 p2p/base/connection_unittest.cc               |  1 -
 p2p/base/ice_agent_interface.h                |  7 +++--
 p2p/base/ice_controller_interface.h           |  4 +--
 p2p/base/ice_transport_internal.h             |  4 +--
 p2p/base/p2p_transport_channel.cc             | 10 +++----
 p2p/base/p2p_transport_channel.h              |  8 ++---
 p2p/base/p2p_transport_channel_unittest.cc    | 12 ++++----
 p2p/base/port.cc                              |  5 ++--
 p2p/base/port_unittest.cc                     |  1 -
 p2p/base/stun_request.cc                      |  4 +--
 p2p/base/stun_request.h                       |  4 +--
 p2p/base/turn_port.cc                         |  6 ++--
 p2p/base/turn_port_unittest.cc                |  6 ++--
 p2p/dtls/dtls_stun_piggyback_callbacks.h      | 10 +++----
 p2p/dtls/dtls_stun_piggyback_controller.cc    | 12 ++++----
 p2p/dtls/dtls_stun_piggyback_controller.h     | 14 ++++-----
 ...dtls_stun_piggyback_controller_unittest.cc | 16 +++++-----
 p2p/dtls/dtls_transport.cc                    | 29 +++++++++----------
 p2p/dtls/dtls_transport.h                     | 10 +++----
 p2p/dtls/dtls_transport_unittest.cc           | 22 +++++++-------
 p2p/dtls/dtls_utils.cc                        | 24 +++++++--------
 p2p/dtls/dtls_utils.h                         | 20 ++++++-------
 p2p/dtls/dtls_utils_unittest.cc               |  4 +--
 p2p/dtls/fake_dtls_transport.h                |  8 ++---
 p2p/test/fake_ice_transport.h                 |  6 ++--
 p2p/test/mock_ice_agent.h                     |  7 +++--
 p2p/test/mock_ice_controller.h                |  4 +--
 p2p/test/nat_server.cc                        |  5 ++--
 p2p/test/nat_socket_factory.cc                |  4 +--
 p2p/test/nat_socket_factory.h                 |  4 +--
 p2p/test/test_port.cc                         |  6 ++--
 p2p/test/test_port.h                          |  4 +--
 p2p/test/turn_server.cc                       | 12 ++++----
 p2p/test/turn_server.h                        |  8 ++---
 40 files changed, 158 insertions(+), 180 deletions(-)

diff --git a/p2p/BUILD.gn b/p2p/BUILD.gn
index f08c9eb7f5..0b11310ec5 100644
--- a/p2p/BUILD.gn
+++ b/p2p/BUILD.gn
@@ -47,7 +47,6 @@ rtc_library("async_stun_tcp_socket") {
     "base/async_stun_tcp_socket.h",
   ]
   deps = [
-    "../api:array_view",
     "../api/environment",
     "../api/transport:stun_types",
     "../rtc_base:async_packet_socket",
@@ -78,7 +77,6 @@ rtc_library("basic_ice_controller") {
     ":p2p_constants",
     ":p2p_transport_channel_ice_field_trials",
     ":transport_description",
-    "../api:array_view",
     "../api:candidate",
     "../api/environment",
     "../api/transport:enums",
@@ -188,7 +186,6 @@ rtc_library("connection") {
     ":port_interface",
     ":stun_request",
     ":transport_description",
-    "../api:array_view",
     "../api:candidate",
     "../api:rtc_error",
     "../api:sequence_checker",
@@ -267,7 +264,6 @@ rtc_library("dtls_transport") {
     ":dtls_utils",
     ":ice_transport_internal",
     ":packet_transport_internal",
-    "../api:array_view",
     "../api:dtls_transport_interface",
     "../api:field_trials_view",
     "../api:ice_transport_interface",
@@ -345,7 +341,6 @@ rtc_source_set("ice_agent_interface") {
     ":connection",
     ":ice_switch_reason",
     ":transport_description",
-    "../api:array_view",
     "../api/units:timestamp",
   ]
 }
@@ -360,7 +355,6 @@ rtc_library("ice_controller_interface") {
     ":ice_switch_reason",
     ":ice_transport_internal",
     ":transport_description",
-    "../api:array_view",
     "../api/units:time_delta",
     "../api/units:timestamp",
     "../rtc_base:checks",
@@ -414,7 +408,6 @@ rtc_library("ice_transport_internal") {
     ":port",
     ":stun_dictionary",
     ":transport_description",
-    "../api:array_view",
     "../api:candidate",
     "../api:peer_connection_interface",
     "../api:rtc_error",
@@ -468,7 +461,6 @@ rtc_library("p2p_transport_channel") {
     ":stun_dictionary",
     ":transport_description",
     ":wrapping_active_ice_controller",
-    "../api:array_view",
     "../api:async_dns_resolver",
     "../api:candidate",
     "../api:field_trials_view",
@@ -550,7 +542,6 @@ rtc_library("port") {
     ":port_interface",
     ":stun_request",
     ":transport_description",
-    "../api:array_view",
     "../api:candidate",
     "../api:local_network_access_permission",
     "../api:packet_socket_factory",
@@ -655,7 +646,6 @@ rtc_library("dtls_utils") {
     "dtls/dtls_utils.h",
   ]
   deps = [
-    "../api:array_view",
     "../rtc_base:buffer",
     "../rtc_base:checks",
     "../rtc_base:crc32",
@@ -671,7 +661,6 @@ rtc_library("dtls_stun_piggyback_controller") {
   ]
   deps = [
     ":dtls_utils",
-    "../api:array_view",
     "../api:sequence_checker",
     "../api/transport:stun_types",
     "../rtc_base:checks",
@@ -733,7 +722,6 @@ rtc_library("stun_request") {
     "base/stun_request.h",
   ]
   deps = [
-    "../api:array_view",
     "../api:sequence_checker",
     "../api/environment",
     "../api/task_queue",
@@ -840,7 +828,6 @@ rtc_library("turn_port") {
     ":port_interface",
     ":relay_port_factory_interface",
     ":stun_request",
-    "../api:array_view",
     "../api:async_dns_resolver",
     "../api:candidate",
     "../api:local_network_access_permission",
@@ -944,7 +931,6 @@ if (rtc_include_tests) {
       ":ice_transport_internal",
       ":port",
       ":transport_description",
-      "../api:array_view",
       "../api:candidate",
       "../api:field_trials",
       "../api:ice_transport_interface",
@@ -1047,7 +1033,6 @@ if (rtc_include_tests) {
       ":packet_transport_internal",
       ":port_interface",
       ":transport_description",
-      "../api:array_view",
       "../api:async_dns_resolver",
       "../api:candidate",
       "../api:dtls_transport_interface",
@@ -1168,7 +1153,6 @@ if (rtc_include_tests) {
       ":transport_description_factory",
       ":turn_port",
       ":wrapping_active_ice_controller",
-      "../api:array_view",
       "../api:async_dns_resolver",
       "../api:candidate",
       "../api:create_network_emulation_manager",
@@ -1325,7 +1309,6 @@ rtc_library("test_port") {
     ":p2p_constants",
     ":port",
     ":port_interface",
-    "../api:array_view",
     "../api:candidate",
     "../api/transport:stun_types",
     "../rtc_base:async_packet_socket",
@@ -1353,7 +1336,6 @@ rtc_library("p2p_server_utils") {
     ":async_stun_tcp_socket",
     ":port",
     ":port_interface",
-    "../api:array_view",
     "../api:packet_socket_factory",
     "../api:sequence_checker",
     "../api/environment",
diff --git a/p2p/base/async_stun_tcp_socket.cc b/p2p/base/async_stun_tcp_socket.cc
index 3344f5a3b1..9501b66a12 100644
--- a/p2p/base/async_stun_tcp_socket.cc
+++ b/p2p/base/async_stun_tcp_socket.cc
@@ -14,10 +14,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <utility>
 
 #include "absl/base/nullability.h"
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/transport/stun.h"
 #include "rtc_base/async_packet_socket.h"
@@ -89,7 +89,7 @@ int AsyncStunTCPSocket::Send(const void* pv,
   return static_cast<int>(cb);
 }
 
-size_t AsyncStunTCPSocket::ProcessInput(ArrayView<const uint8_t> data) {
+size_t AsyncStunTCPSocket::ProcessInput(std::span<const uint8_t> data) {
   SocketAddress remote_addr(GetRemoteAddress());
   // STUN packet - First 4 bytes. Total header size is 20 bytes.
   // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
@@ -129,7 +129,7 @@ size_t AsyncStunTCPSocket::GetExpectedLength(const void* data,
                                              size_t len,
                                              int* pad_bytes) {
   *pad_bytes = 0;
-  ArrayView<const uint8_t> view(static_cast<const uint8_t*>(data), len);
+  std::span<const uint8_t> view(static_cast<const uint8_t*>(data), len);
   PacketLength pkt_len = GetBE16(view.subspan(kPacketLenOffset, 2));
   size_t expected_pkt_len;
   uint16_t msg_type = GetBE16(view);
diff --git a/p2p/base/async_stun_tcp_socket.h b/p2p/base/async_stun_tcp_socket.h
index e3bc0b9fd9..c1e1924adb 100644
--- a/p2p/base/async_stun_tcp_socket.h
+++ b/p2p/base/async_stun_tcp_socket.h
@@ -14,9 +14,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <memory>
+#include <span>
 
 #include "absl/base/nullability.h"
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "rtc_base/async_packet_socket.h"
 #include "rtc_base/async_tcp_socket.h"
@@ -35,7 +35,7 @@ class AsyncStunTCPSocket : public AsyncTCPSocketBase {
   int Send(const void* pv,
            size_t cb,
            const AsyncSocketPacketOptions& options) override;
-  size_t ProcessInput(ArrayView<const uint8_t> data) override;
+  size_t ProcessInput(std::span<const uint8_t> data) override;
 
  private:
   // This method returns the message hdr + length written in the header.
diff --git a/p2p/base/async_stun_tcp_socket_unittest.cc b/p2p/base/async_stun_tcp_socket_unittest.cc
index 25677c3594..574561c062 100644
--- a/p2p/base/async_stun_tcp_socket_unittest.cc
+++ b/p2p/base/async_stun_tcp_socket_unittest.cc
@@ -18,7 +18,6 @@
 #include <utility>
 
 #include "absl/memory/memory.h"
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "rtc_base/async_packet_socket.h"
 #include "rtc_base/async_tcp_socket.h"
diff --git a/p2p/base/basic_ice_controller.h b/p2p/base/basic_ice_controller.h
index 9e32c5b080..d0120b4a48 100644
--- a/p2p/base/basic_ice_controller.h
+++ b/p2p/base/basic_ice_controller.h
@@ -17,9 +17,9 @@
 #include <map>
 #include <optional>
 #include <set>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
@@ -45,7 +45,7 @@ class BasicIceController : public IceControllerInterface {
   void SetSelectedConnection(const Connection* selected_connection) override;
   void AddConnection(const Connection* connection) override;
   void OnConnectionDestroyed(const Connection* connection) override;
-  ArrayView<const Connection* const> GetConnections() const override {
+  std::span<const Connection* const> GetConnections() const override {
     return connections_;
   }
 
diff --git a/p2p/base/connection.cc b/p2p/base/connection.cc
index 0e3b2a9c74..6bb2e578b3 100644
--- a/p2p/base/connection.cc
+++ b/p2p/base/connection.cc
@@ -15,6 +15,7 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
@@ -22,7 +23,6 @@
 #include "absl/algorithm/container.h"
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/candidate.h"
 #include "api/environment/environment.h"
 #include "api/rtc_error.h"
@@ -653,7 +653,7 @@ void Connection::MaybeHandleDtlsPiggybackingAttributes(
       msg->GetByteString(STUN_ATTR_META_DTLS_IN_STUN);
   const StunByteStringAttribute* dtls_piggyback_ack =
       msg->GetByteString(STUN_ATTR_META_DTLS_IN_STUN_ACK);
-  std::optional<ArrayView<uint8_t>> piggyback_data;
+  std::optional<std::span<uint8_t>> piggyback_data;
   if (dtls_piggyback_attr != nullptr) {
     piggyback_data = dtls_piggyback_attr->array_view();
   }
diff --git a/p2p/base/connection_unittest.cc b/p2p/base/connection_unittest.cc
index 1e08606c28..02ce1f736a 100644
--- a/p2p/base/connection_unittest.cc
+++ b/p2p/base/connection_unittest.cc
@@ -22,7 +22,6 @@
 
 #include "absl/memory/memory.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/environment/environment_factory.h"
 #include "api/rtc_error.h"
diff --git a/p2p/base/ice_agent_interface.h b/p2p/base/ice_agent_interface.h
index e2c3f966fd..2047be08dd 100644
--- a/p2p/base/ice_agent_interface.h
+++ b/p2p/base/ice_agent_interface.h
@@ -11,7 +11,8 @@
 #ifndef P2P_BASE_ICE_AGENT_INTERFACE_H_
 #define P2P_BASE_ICE_AGENT_INTERFACE_H_
 
-#include "api/array_view.h"
+#include <span>
+
 #include "api/units/timestamp.h"
 #include "p2p/base/connection.h"
 #include "p2p/base/ice_switch_reason.h"
@@ -62,7 +63,7 @@ class IceAgentInterface {
   //
   // SignalStateChange will not be triggered.
   virtual void ForgetLearnedStateForConnections(
-      ArrayView<const Connection* const> connections) = 0;
+      std::span<const Connection* const> connections) = 0;
 
   // Send a STUN ping request for the given connection.
   virtual void SendPingRequest(const Connection* connection) = 0;
@@ -74,7 +75,7 @@ class IceAgentInterface {
   // Prune away the given connections. Returns true if pruning is permitted and
   // successfully performed.
   virtual bool PruneConnections(
-      ArrayView<const Connection* const> connections) = 0;
+      std::span<const Connection* const> connections) = 0;
 };
 
 }  //  namespace webrtc
diff --git a/p2p/base/ice_controller_interface.h b/p2p/base/ice_controller_interface.h
index 90e23a97fe..6aa5b6024d 100644
--- a/p2p/base/ice_controller_interface.h
+++ b/p2p/base/ice_controller_interface.h
@@ -12,10 +12,10 @@
 #define P2P_BASE_ICE_CONTROLLER_INTERFACE_H_
 
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
 #include "p2p/base/connection.h"
@@ -111,7 +111,7 @@ class IceControllerInterface {
   virtual void OnConnectionDestroyed(const Connection* connection) = 0;
 
   // These are all connections that has been added and not destroyed.
-  virtual ArrayView<const Connection* const> GetConnections() const = 0;
+  virtual std::span<const Connection* const> GetConnections() const = 0;
 
   // Is there a pingable connection ?
   // This function is used to boot-strap pinging, after this returns true
diff --git a/p2p/base/ice_transport_internal.h b/p2p/base/ice_transport_internal.h
index 1e3321e178..a30db71a62 100644
--- a/p2p/base/ice_transport_internal.h
+++ b/p2p/base/ice_transport_internal.h
@@ -14,12 +14,12 @@
 #include <cstdint>
 #include <functional>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/functional/any_invocable.h"
-#include "api/array_view.h"
 #include "api/candidate.h"
 #include "api/peer_connection_interface.h"
 #include "api/rtc_error.h"
@@ -410,7 +410,7 @@ class RTC_EXPORT IceTransportInternal : public PacketTransportInternal {
 
   CallbackList<IceTransportInternal*,
                const StunDictionaryView&,
-               ArrayView<uint16_t>>
+               std::span<uint16_t>>
       dictionary_view_updated_callback_list_;
   CallbackList<IceTransportInternal*, const StunDictionaryWriter&>
       dictionary_writer_synced_callback_list_;
diff --git a/p2p/base/p2p_transport_channel.cc b/p2p/base/p2p_transport_channel.cc
index 0b5a6a55a2..76b5006e55 100644
--- a/p2p/base/p2p_transport_channel.cc
+++ b/p2p/base/p2p_transport_channel.cc
@@ -19,6 +19,7 @@
 #include <memory>
 #include <optional>
 #include <set>
+#include <span>
 #include <string>
 #include <tuple>
 #include <utility>
@@ -29,7 +30,6 @@
 #include "absl/memory/memory.h"
 #include "absl/strings/match.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/async_dns_resolver.h"
 #include "api/candidate.h"
 #include "api/environment/environment.h"
@@ -333,7 +333,7 @@ void P2PTransportChannel::AddConnection(Connection* connection) {
 }
 
 void P2PTransportChannel::ForgetLearnedStateForConnections(
-    ArrayView<const Connection* const> connections) {
+    std::span<const Connection* const> connections) {
   for (const Connection* con : connections) {
     FromIceController(con)->ForgetLearnedState();
   }
@@ -1705,9 +1705,9 @@ DiffServCodePoint P2PTransportChannel::DefaultDscpValue() const {
   return static_cast<DiffServCodePoint>(it->second);
 }
 
-ArrayView<Connection* const> P2PTransportChannel::connections() const {
+std::span<Connection* const> P2PTransportChannel::connections() const {
   RTC_DCHECK_RUN_ON(&network_thread_);
-  return ArrayView<Connection* const>(connections_.data(), connections_.size());
+  return std::span<Connection* const>(connections_.data(), connections_.size());
 }
 
 void P2PTransportChannel::RemoveConnectionForTest(Connection* connection) {
@@ -1806,7 +1806,7 @@ bool P2PTransportChannel::AllowedToPruneConnections() const {
 }
 
 bool P2PTransportChannel::PruneConnections(
-    ArrayView<const Connection* const> connections) {
+    std::span<const Connection* const> connections) {
   RTC_DCHECK_RUN_ON(&network_thread_);
   if (!AllowedToPruneConnections()) {
     RTC_LOG(LS_WARNING) << "Not allowed to prune connections";
diff --git a/p2p/base/p2p_transport_channel.h b/p2p/base/p2p_transport_channel.h
index e44b6eece4..7339920c41 100644
--- a/p2p/base/p2p_transport_channel.h
+++ b/p2p/base/p2p_transport_channel.h
@@ -26,11 +26,11 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/async_dns_resolver.h"
 #include "api/candidate.h"
 #include "api/environment/environment.h"
@@ -164,9 +164,9 @@ class RTC_EXPORT P2PTransportChannel : public IceTransportInternal,
   void SwitchSelectedConnection(const Connection* connection,
                                 IceSwitchReason reason) override;
   void ForgetLearnedStateForConnections(
-      ArrayView<const Connection* const> connections) override;
+      std::span<const Connection* const> connections) override;
   bool PruneConnections(
-      ArrayView<const Connection* const> connections) override;
+      std::span<const Connection* const> connections) override;
 
   // TODO(honghaiz): Remove this method once the reference of it in
   // Chromoting is removed.
@@ -210,7 +210,7 @@ class RTC_EXPORT P2PTransportChannel : public IceTransportInternal,
   void MarkConnectionPinged(Connection* conn);
 
   // Public for unit tests.
-  ArrayView<Connection* const> connections() const;
+  std::span<Connection* const> connections() const;
   void RemoveConnectionForTest(Connection* connection);
 
   // Public for unit tests.
diff --git a/p2p/base/p2p_transport_channel_unittest.cc b/p2p/base/p2p_transport_channel_unittest.cc
index f6eac835aa..4ec6d91ab4 100644
--- a/p2p/base/p2p_transport_channel_unittest.cc
+++ b/p2p/base/p2p_transport_channel_unittest.cc
@@ -17,6 +17,7 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <tuple>
 #include <utility>
@@ -25,7 +26,6 @@
 #include "absl/algorithm/container.h"
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/async_dns_resolver.h"
 #include "api/candidate.h"
 #include "api/environment/environment.h"
@@ -2561,7 +2561,7 @@ class P2PTransportChannelMultihomedTest : public P2PTransportChannelTest {
   }
 
   Connection* GetBestConnection(P2PTransportChannel* channel) {
-    ArrayView<Connection* const> connections = channel->connections();
+    std::span<Connection* const> connections = channel->connections();
     auto it = absl::c_find(connections, channel->selected_connection());
     if (it == connections.end()) {
       return nullptr;
@@ -2570,7 +2570,7 @@ class P2PTransportChannelMultihomedTest : public P2PTransportChannelTest {
   }
 
   Connection* GetBackupConnection(P2PTransportChannel* channel) {
-    ArrayView<Connection* const> connections = channel->connections();
+    std::span<Connection* const> connections = channel->connections();
     auto it = absl::c_find_if_not(connections, [channel](Connection* conn) {
       return conn == channel->selected_connection();
     });
@@ -2583,7 +2583,7 @@ class P2PTransportChannelMultihomedTest : public P2PTransportChannelTest {
   void DestroyAllButBestConnection(P2PTransportChannel* channel) {
     const Connection* selected_connection = channel->selected_connection();
     // Copy the list of connections since the original will be modified.
-    ArrayView<Connection* const> view = channel->connections();
+    std::span<Connection* const> view = channel->connections();
     std::vector<Connection*> connections(view.begin(), view.end());
     for (Connection* conn : connections) {
       if (conn != selected_connection)
@@ -3567,7 +3567,7 @@ TEST_F(P2PTransportChannelMultihomedTest, StunDictionaryPerformsSync) {
   CreateChannels(env);
 
   MockFunction<void(IceTransportInternal*, const StunDictionaryView&,
-                    ArrayView<uint16_t>)>
+                    std::span<uint16_t>)>
       view_updated_func;
   ep2_ch1()->AddDictionaryViewUpdatedCallback(
       "tag", view_updated_func.AsStdFunction());
@@ -7132,7 +7132,7 @@ class P2PTransportChannelTestDtlsInStun : public P2PTransportChannelTestBase {
     return make_pair(absl::string_view(pending_packet_), std::nullopt);
   }
 
-  void piggyback_data_received(std::optional<ArrayView<uint8_t>> data,
+  void piggyback_data_received(std::optional<std::span<uint8_t>> data,
                                std::optional<std::vector<uint32_t>> ack) {}
 
   ScopedFakeClock clock_;
diff --git a/p2p/base/port.cc b/p2p/base/port.cc
index 932ff54945..e7ef807bd5 100644
--- a/p2p/base/port.cc
+++ b/p2p/base/port.cc
@@ -15,6 +15,7 @@
 #include <functional>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
@@ -23,7 +24,6 @@
 #include "absl/functional/any_invocable.h"
 #include "absl/memory/memory.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/candidate.h"
 #include "api/local_network_access_permission.h"
 #include "api/sequence_checker.h"
@@ -468,8 +468,7 @@ bool Port::GetStunMessage(const char* data,
   // Parse the request message.  If the packet is not a complete and correct
   // STUN message, then ignore it.
   std::unique_ptr<IceMessage> stun_msg(new IceMessage());
-  ByteBufferReader buf(
-      MakeArrayView(reinterpret_cast<const uint8_t*>(data), size));
+  ByteBufferReader buf(std::span(reinterpret_cast<const uint8_t*>(data), size));
   if (!stun_msg->Read(&buf) || (buf.Length() > 0)) {
     return false;
   }
diff --git a/p2p/base/port_unittest.cc b/p2p/base/port_unittest.cc
index a0ca85f9f6..82b005ccf2 100644
--- a/p2p/base/port_unittest.cc
+++ b/p2p/base/port_unittest.cc
@@ -22,7 +22,6 @@
 
 #include "absl/memory/memory.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/async_dns_resolver.h"
 #include "api/candidate.h"
 #include "api/environment/environment.h"
diff --git a/p2p/base/stun_request.cc b/p2p/base/stun_request.cc
index c5867be0cd..3afabe1e42 100644
--- a/p2p/base/stun_request.cc
+++ b/p2p/base/stun_request.cc
@@ -15,12 +15,12 @@
 #include <cstdint>
 #include <functional>
 #include <memory>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/sequence_checker.h"
 #include "api/task_queue/pending_task_safety_flag.h"
@@ -191,7 +191,7 @@ bool StunRequestManager::empty() const {
   return requests_.empty();
 }
 
-bool StunRequestManager::CheckResponse(ArrayView<const uint8_t> payload) {
+bool StunRequestManager::CheckResponse(std::span<const uint8_t> payload) {
   RTC_DCHECK_RUN_ON(thread_);
   // Check the appropriate bytes of the stream to see if they match the
   // transaction ID of a response we are expecting.
diff --git a/p2p/base/stun_request.h b/p2p/base/stun_request.h
index b16720d8e2..ccd284a997 100644
--- a/p2p/base/stun_request.h
+++ b/p2p/base/stun_request.h
@@ -16,9 +16,9 @@
 #include <functional>
 #include <map>
 #include <memory>
+#include <span>
 #include <string>
 
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/task_queue/pending_task_safety_flag.h"
 #include "api/task_queue/task_queue_base.h"
@@ -70,7 +70,7 @@ class StunRequestManager {
   // Determines whether the given message is a response to one of the
   // outstanding requests, and if so, processes it appropriately.
   bool CheckResponse(StunMessage* msg);
-  bool CheckResponse(ArrayView<const uint8_t> payload);
+  bool CheckResponse(std::span<const uint8_t> payload);
 
   // Called from a StunRequest when a timeout occurs.
   void OnRequestTimedOut(StunRequest* request);
diff --git a/p2p/base/turn_port.cc b/p2p/base/turn_port.cc
index 9a22505231..1a6f3a7078 100644
--- a/p2p/base/turn_port.cc
+++ b/p2p/base/turn_port.cc
@@ -15,6 +15,7 @@
 #include <cstdint>
 #include <functional>
 #include <memory>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
@@ -23,7 +24,6 @@
 #include "absl/memory/memory.h"
 #include "absl/strings/match.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/candidate.h"
 #include "api/local_network_access_permission.h"
 #include "api/packet_socket_factory.h"
@@ -1118,7 +1118,7 @@ void TurnPort::HandleChannelData(uint16_t channel_id,
   //   +-------------------------------+
 
   // Extract header fields from the message.
-  ArrayView<const uint8_t> payload = packet.payload();
+  std::span<const uint8_t> payload = packet.payload();
   uint16_t len = GetBE16(payload.subspan(2, 2));
   if (len > payload.size() - TURN_CHANNEL_HEADER_SIZE) {
     RTC_LOG(LS_WARNING) << ToString()
@@ -1867,7 +1867,7 @@ int TurnEntry::Send(const void* data,
     buf.WriteUInt16(channel_id_);
     buf.WriteUInt16(static_cast<uint16_t>(size));
     buf.Write(
-        ArrayView<const uint8_t>(reinterpret_cast<const uint8_t*>(data), size));
+        std::span<const uint8_t>(reinterpret_cast<const uint8_t*>(data), size));
   }
   AsyncSocketPacketOptions modified_options(options);
   modified_options.info_signaled_after_sent.turn_overhead_bytes =
diff --git a/p2p/base/turn_port_unittest.cc b/p2p/base/turn_port_unittest.cc
index 99efabc637..a5d7fd7e44 100644
--- a/p2p/base/turn_port_unittest.cc
+++ b/p2p/base/turn_port_unittest.cc
@@ -14,13 +14,13 @@
 #include <list>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/candidate.h"
 #include "api/environment/environment.h"
 #include "api/environment/environment_factory.h"
@@ -1027,7 +1027,7 @@ class TurnLoggingIdValidator : public StunMessageObserver {
       }
     }
   }
-  void ReceivedChannelData(ArrayView<const uint8_t> packet) override {}
+  void ReceivedChannelData(std::span<const uint8_t> packet) override {}
 
  private:
   const char* expect_val_;
@@ -1960,7 +1960,7 @@ class MessageObserver : public StunMessageObserver {
     }
   }
 
-  void ReceivedChannelData(ArrayView<const uint8_t> payload) override {
+  void ReceivedChannelData(std::span<const uint8_t> payload) override {
     if (channel_data_counter_ != nullptr) {
       (*channel_data_counter_)++;
     }
diff --git a/p2p/dtls/dtls_stun_piggyback_callbacks.h b/p2p/dtls/dtls_stun_piggyback_callbacks.h
index 241d51e964..e3cb33afbc 100644
--- a/p2p/dtls/dtls_stun_piggyback_callbacks.h
+++ b/p2p/dtls/dtls_stun_piggyback_callbacks.h
@@ -13,12 +13,12 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/transport/stun.h"
 #include "rtc_base/checks.h"
 
@@ -38,10 +38,10 @@ class DtlsStunPiggybackCallbacks {
           /* request-type */ StunMessageType)>&& send_data,
 
       // Function invoked when receiving a STUN_BINDING { REQUEST / RESPONSE }
-      // contains the optional ArrayView of the DTLS_IN_STUN attribute and the
+      // contains the optional std::span of the DTLS_IN_STUN attribute and the
       // optional uint32_t vector DTLS_IN_STUN_ACK attribute.
       absl::AnyInvocable<void(
-          std::optional<ArrayView<uint8_t>> /* recv_data */,
+          std::optional<std::span<uint8_t>> /* recv_data */,
           std::optional<std::vector<uint32_t>> /* recv_acks */)>&& recv_data)
       : send_data_(std::move(send_data)), recv_data_(std::move(recv_data)) {
     RTC_DCHECK(
@@ -58,7 +58,7 @@ class DtlsStunPiggybackCallbacks {
     return send_data_(request_type);
   }
 
-  void recv_data(std::optional<ArrayView<uint8_t>> data,
+  void recv_data(std::optional<std::span<uint8_t>> data,
                  std::optional<std::vector<uint32_t>> acks) {
     RTC_DCHECK(recv_data_);
     return recv_data_(data, acks);
@@ -75,7 +75,7 @@ class DtlsStunPiggybackCallbacks {
                                std::optional<std::vector<uint32_t>>>(
       /* request-type */ StunMessageType)>
       send_data_;
-  absl::AnyInvocable<void(std::optional<ArrayView<uint8_t>> /* recv_data */,
+  absl::AnyInvocable<void(std::optional<std::span<uint8_t>> /* recv_data */,
                           std::optional<std::vector<uint32_t>> /* recv_acks */)>
       recv_data_;
 };
diff --git a/p2p/dtls/dtls_stun_piggyback_controller.cc b/p2p/dtls/dtls_stun_piggyback_controller.cc
index f0b84c62a8..cb7d0c4cfe 100644
--- a/p2p/dtls/dtls_stun_piggyback_controller.cc
+++ b/p2p/dtls/dtls_stun_piggyback_controller.cc
@@ -13,13 +13,13 @@
 #include <algorithm>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
 #include "absl/container/flat_hash_set.h"
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/sequence_checker.h"
 #include "api/transport/stun.h"
 #include "p2p/dtls/dtls_utils.h"
@@ -31,7 +31,7 @@
 namespace webrtc {
 
 DtlsStunPiggybackController::DtlsStunPiggybackController(
-    absl::AnyInvocable<void(ArrayView<const uint8_t>)> dtls_data_callback,
+    absl::AnyInvocable<void(std::span<const uint8_t>)> dtls_data_callback,
     // NOLINTNEXTLINE(readability/casting) - not a cast; false positive!
     absl::AnyInvocable<void(bool) &&> piggyback_complete_callback)
     : dtls_data_callback_(std::move(dtls_data_callback)),
@@ -92,7 +92,7 @@ void DtlsStunPiggybackController::SetDtlsFailed() {
   CallCompleteCallback(/*success=*/false);
 }
 
-void DtlsStunPiggybackController::CapturePacket(ArrayView<const uint8_t> data) {
+void DtlsStunPiggybackController::CapturePacket(std::span<const uint8_t> data) {
   RTC_DCHECK_RUN_ON(&sequence_checker_);
   if (!IsDtlsPacket(data)) {
     return;
@@ -159,14 +159,14 @@ DtlsStunPiggybackController::GetAckToPiggyback(
   return handshake_messages_received_;
 }
 
-std::vector<ArrayView<const uint8_t>>
+std::vector<std::span<const uint8_t>>
 DtlsStunPiggybackController::GetPending() {
   RTC_DCHECK_RUN_ON(&sequence_checker_);
   return pending_packets_.GetAll();
 }
 
 void DtlsStunPiggybackController::ReportDataPiggybacked(
-    std::optional<ArrayView<uint8_t>> data,
+    std::optional<std::span<uint8_t>> data,
     std::optional<std::vector<uint32_t>> acks) {
   RTC_DCHECK_RUN_ON(&sequence_checker_);
 
@@ -234,7 +234,7 @@ void DtlsStunPiggybackController::ReportDataPiggybacked(
 }
 
 void DtlsStunPiggybackController::ReportDtlsPacket(
-    ArrayView<const uint8_t> data) {
+    std::span<const uint8_t> data) {
   RTC_DCHECK_RUN_ON(&sequence_checker_);
 
   if (state_ == State::OFF || state_ == State::COMPLETE) {
diff --git a/p2p/dtls/dtls_stun_piggyback_controller.h b/p2p/dtls/dtls_stun_piggyback_controller.h
index 89cd27be1c..18af1bdef3 100644
--- a/p2p/dtls/dtls_stun_piggyback_controller.h
+++ b/p2p/dtls/dtls_stun_piggyback_controller.h
@@ -13,11 +13,11 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <vector>
 
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/sequence_checker.h"
 #include "api/transport/stun.h"
 #include "p2p/dtls/dtls_utils.h"
@@ -37,7 +37,7 @@ class DtlsStunPiggybackController {
   // dtls_data_callback will be called with any DTLS packets received
   // piggybacked.
   DtlsStunPiggybackController(
-      absl::AnyInvocable<void(ArrayView<const uint8_t>)> dtls_data_callback,
+      absl::AnyInvocable<void(std::span<const uint8_t>)> dtls_data_callback,
       // NOLINTNEXTLINE(readability/casting) - not a cast; false positive!
       absl::AnyInvocable<void(bool) &&> piggyback_complete_callback);
 
@@ -79,7 +79,7 @@ class DtlsStunPiggybackController {
 
   // Intercepts DTLS packets which should go into the STUN packets during the
   // handshake.
-  void CapturePacket(ArrayView<const uint8_t> data);
+  void CapturePacket(std::span<const uint8_t> data);
   void ClearCachedPacketForTesting();
 
   // Inform piggybackcontroller that a flight is complete.
@@ -91,17 +91,17 @@ class DtlsStunPiggybackController {
       StunMessageType stun_message_type);
   std::optional<const std::vector<uint32_t>> GetAckToPiggyback(
       StunMessageType stun_message_type);
-  std::vector<ArrayView<const uint8_t>> GetPending();
+  std::vector<std::span<const uint8_t>> GetPending();
 
   // Called by Connection when receiving a STUN BINDING { REQUEST / RESPONSE }.
-  void ReportDataPiggybacked(std::optional<ArrayView<uint8_t>> data,
+  void ReportDataPiggybacked(std::optional<std::span<uint8_t>> data,
                              std::optional<std::vector<uint32_t>> acks);
 
   // Called by
   // * DTLSTransport when receiving a DTLS packet (possibly after the packet
   //   was emitted by this class).
   // * This class when processing a DTLS packet.
-  void ReportDtlsPacket(ArrayView<const uint8_t> data);
+  void ReportDtlsPacket(std::span<const uint8_t> data);
 
   int GetCountOfReceivedData() const { return data_recv_count_; }
 
@@ -109,7 +109,7 @@ class DtlsStunPiggybackController {
   State state_ RTC_GUARDED_BY(sequence_checker_) = State::TENTATIVE;
   bool writing_packets_ RTC_GUARDED_BY(sequence_checker_) = false;
   PacketStash pending_packets_ RTC_GUARDED_BY(sequence_checker_);
-  absl::AnyInvocable<void(ArrayView<const uint8_t>)> dtls_data_callback_
+  absl::AnyInvocable<void(std::span<const uint8_t>)> dtls_data_callback_
       RTC_GUARDED_BY(sequence_checker_);
   // NOLINTNEXTLINE(readability/casting) - not a cast; false positive!
   absl::AnyInvocable<void(bool) &&> piggyback_complete_callback_
diff --git a/p2p/dtls/dtls_stun_piggyback_controller_unittest.cc b/p2p/dtls/dtls_stun_piggyback_controller_unittest.cc
index 59e5299f8f..7ff86af6d9 100644
--- a/p2p/dtls/dtls_stun_piggyback_controller_unittest.cc
+++ b/p2p/dtls/dtls_stun_piggyback_controller_unittest.cc
@@ -13,11 +13,11 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/transport/stun.h"
 #include "p2p/dtls/dtls_utils.h"
 #include "rtc_base/byte_buffer.h"
@@ -63,7 +63,7 @@ const std::vector<uint8_t> empty = {};
 
 const std::vector<uint8_t> kPayload = {0x1, 0x2, 0x3};
 
-std::vector<uint32_t> FromAckAttribute(ArrayView<uint8_t> attr) {
+std::vector<uint32_t> FromAckAttribute(std::span<uint8_t> attr) {
   ByteBufferReader ack_reader(attr);
   std::vector<uint32_t> values;
   uint32_t value;
@@ -110,10 +110,10 @@ class DtlsStunPiggybackControllerTest : public ::testing::Test {
  protected:
   DtlsStunPiggybackControllerTest()
       : client_(
-            [this](ArrayView<const uint8_t> data) { ClientPacketSink(data); },
+            [this](std::span<const uint8_t> data) { ClientPacketSink(data); },
             [this](bool success) { ClientCompleteCallback(success); }),
         server_(
-            [this](ArrayView<const uint8_t> data) { ServerPacketSink(data); },
+            [this](std::span<const uint8_t> data) { ServerPacketSink(data); },
             [this](bool success) { ServerCompleteCallback(success); }),
         packet_(kPayload, SocketAddress(), std::nullopt) {}
 
@@ -127,7 +127,7 @@ class DtlsStunPiggybackControllerTest : public ::testing::Test {
       client_.ClearCachedPacketForTesting();
     }
     std::unique_ptr<StunByteStringAttribute> attr_data;
-    std::optional<ArrayView<uint8_t>> view_data;
+    std::optional<std::span<uint8_t>> view_data;
     if (auto data = client_.GetDataToPiggyback(type)) {
       attr_data = WrapInStun(STUN_ATTR_META_DTLS_IN_STUN, *data);
       view_data = attr_data->array_view();
@@ -160,7 +160,7 @@ class DtlsStunPiggybackControllerTest : public ::testing::Test {
       server_.ClearCachedPacketForTesting();
     }
     std::unique_ptr<StunByteStringAttribute> attr_data;
-    std::optional<ArrayView<uint8_t>> view_data;
+    std::optional<std::span<uint8_t>> view_data;
     if (auto data = server_.GetDataToPiggyback(type)) {
       attr_data = WrapInStun(STUN_ATTR_META_DTLS_IN_STUN, *data);
       view_data = attr_data->array_view();
@@ -195,8 +195,8 @@ class DtlsStunPiggybackControllerTest : public ::testing::Test {
   DtlsStunPiggybackController client_;
   DtlsStunPiggybackController server_;
 
-  MOCK_METHOD(void, ClientPacketSink, (ArrayView<const uint8_t>));
-  MOCK_METHOD(void, ServerPacketSink, (ArrayView<const uint8_t>));
+  MOCK_METHOD(void, ClientPacketSink, (std::span<const uint8_t>));
+  MOCK_METHOD(void, ServerPacketSink, (std::span<const uint8_t>));
 
   MOCK_METHOD(void, ClientCompleteCallback, (bool));
   MOCK_METHOD(void, ServerCompleteCallback, (bool));
diff --git a/p2p/dtls/dtls_transport.cc b/p2p/dtls/dtls_transport.cc
index ae516ad704..1796ddfe77 100644
--- a/p2p/dtls/dtls_transport.cc
+++ b/p2p/dtls/dtls_transport.cc
@@ -15,13 +15,13 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/crypto/crypto_options.h"
 #include "api/dtls_transport_interface.h"
 #include "api/environment/environment.h"
@@ -128,7 +128,7 @@ constexpr int kDisabledHandshakeTimeoutMs = 3600 * 1000 * 24;
 
 constexpr uint32_t kMaxCachedClientHello = 4;
 
-static bool IsRtpPacket(ArrayView<const uint8_t> payload) {
+static bool IsRtpPacket(std::span<const uint8_t> payload) {
   const uint8_t* u = payload.data();
   return (payload.size() >= kMinRtpPacketLen && (u[0] & 0xC0) == 0x80);
 }
@@ -149,7 +149,7 @@ void StreamInterfaceChannel::SetDtlsStunPiggybackController(
   dtls_stun_piggyback_controller_ = dtls_stun_piggyback_controller;
 }
 
-StreamResult StreamInterfaceChannel::Read(ArrayView<uint8_t> buffer,
+StreamResult StreamInterfaceChannel::Read(std::span<uint8_t> buffer,
                                           size_t& read,
                                           int& /* error */) {
   RTC_DCHECK_RUN_ON(&callback_sequence_);
@@ -177,7 +177,7 @@ void StreamInterfaceChannel::ClearNextPacketOptions() {
   next_packet_options_.reset();
 }
 
-StreamResult StreamInterfaceChannel::Write(ArrayView<const uint8_t> data,
+StreamResult StreamInterfaceChannel::Write(std::span<const uint8_t> data,
                                            size_t& written,
                                            int& /* error */) {
   RTC_DCHECK_RUN_ON(&callback_sequence_);
@@ -209,7 +209,7 @@ bool StreamInterfaceChannel::Flush() {
   return false;
 }
 
-bool StreamInterfaceChannel::OnPacketReceived(ArrayView<const uint8_t> data) {
+bool StreamInterfaceChannel::OnPacketReceived(std::span<const uint8_t> data) {
   RTC_DCHECK_RUN_ON(&callback_sequence_);
   if (packets_.size() > 0) {
     RTC_LOG(LS_WARNING) << "Packet already in queue.";
@@ -260,7 +260,7 @@ DtlsTransportInternalImpl::DtlsTransportInternalImpl(
           crypto_options.ephemeral_key_exchange_cipher_groups.GetEnabled()),
       ssl_max_version_(max_version),
       dtls_stun_piggyback_controller_(
-          [this](ArrayView<const uint8_t> piggybacked_dtls_packet) {
+          [this](std::span<const uint8_t> piggybacked_dtls_packet) {
             if (piggybacked_dtls_callback_ == nullptr) {
               return;
             }
@@ -655,7 +655,7 @@ int DtlsTransportInternalImpl::SendPacket(
       if (flags & PF_SRTP_BYPASS) {
         RTC_DCHECK(!srtp_ciphers_.empty());
         if (!IsRtpPacket(
-                MakeArrayView(reinterpret_cast<const uint8_t*>(data), size))) {
+                std::span(reinterpret_cast<const uint8_t*>(data), size))) {
           return -1;
         }
 
@@ -669,8 +669,8 @@ int DtlsTransportInternalImpl::SendPacket(
         // StreamInterfaceChannel::Write function. Such change would remove the
         // need of the next_packet_options_.
         StreamResult result = dtls_->Write(
-            MakeArrayView(reinterpret_cast<const uint8_t*>(data), size),
-            written, error);
+            std::span(reinterpret_cast<const uint8_t*>(data), size), written,
+            error);
         if (result != SR_SUCCESS) {
           // Explicitly clear the next packet options, in case no packet was
           // sent.
@@ -775,7 +775,7 @@ void DtlsTransportInternalImpl::ConnectToIceTransport() {
         }
         return std::make_pair(data, ack);
       },
-      [&](std::optional<ArrayView<uint8_t>> data,
+      [&](std::optional<std::span<uint8_t>> data,
           std::optional<std::vector<uint32_t>> acks) {
         if (!dtls_in_stun_) {
           return;
@@ -1016,10 +1016,9 @@ void DtlsTransportInternalImpl::OnDtlsEvent(int sig, int err) {
         // TODO(bugs.webrtc.org/15368): It should be possible to use information
         // from the original packet here to populate socket address and
         // timestamp.
-        NotifyPacketReceived(
-            ReceivedIpPacket(MakeArrayView(buf, read), SocketAddress(),
-                             env_.clock().CurrentTime(), EcnMarking::kNotEct,
-                             ReceivedIpPacket::kDtlsDecrypted));
+        NotifyPacketReceived(ReceivedIpPacket(
+            std::span(buf, read), SocketAddress(), env_.clock().CurrentTime(),
+            EcnMarking::kNotEct, ReceivedIpPacket::kDtlsDecrypted));
       } else if (ret == SR_EOS) {
         // Remote peer shut down the association with no error.
         RTC_LOG(LS_INFO) << ToString() << ": DTLS transport closed by remote";
@@ -1109,7 +1108,7 @@ void DtlsTransportInternalImpl::MaybeStartDtls() {
 
 // Called from OnReadPacket when a DTLS packet is received.
 bool DtlsTransportInternalImpl::HandleDtlsPacket(
-    ArrayView<const uint8_t> payload) {
+    std::span<const uint8_t> payload) {
   // Pass to the StreamInterfaceChannel which ends up being passed to the DTLS
   // stack.
   return downward_->OnPacketReceived(payload);
diff --git a/p2p/dtls/dtls_transport.h b/p2p/dtls/dtls_transport.h
index 7176ccefad..9eed516996 100644
--- a/p2p/dtls/dtls_transport.h
+++ b/p2p/dtls/dtls_transport.h
@@ -16,12 +16,12 @@
 #include <functional>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/crypto/crypto_options.h"
 #include "api/dtls_transport_interface.h"
 #include "api/environment/environment.h"
@@ -76,7 +76,7 @@ class StreamInterfaceChannel : public StreamInterface {
   StreamInterfaceChannel& operator=(const StreamInterfaceChannel&) = delete;
 
   // Push in a packet; this gets pulled out from Read().
-  bool OnPacketReceived(ArrayView<const uint8_t> data);
+  bool OnPacketReceived(std::span<const uint8_t> data);
 
   // Sets the options for the next packet to be written to ice_transport,
   // corresponding to the next Write() call. Safe since BoringSSL guarantees
@@ -89,10 +89,10 @@ class StreamInterfaceChannel : public StreamInterface {
   // Implementations of StreamInterface
   StreamState GetState() const override;
   void Close() override;
-  StreamResult Read(ArrayView<uint8_t> buffer,
+  StreamResult Read(std::span<uint8_t> buffer,
                     size_t& read,
                     int& error) override;
-  StreamResult Write(ArrayView<const uint8_t> data,
+  StreamResult Write(std::span<const uint8_t> data,
                      size_t& written,
                      int& error) override;
   bool Flush() override;
@@ -293,7 +293,7 @@ class DtlsTransportInternalImpl : public DtlsTransportInternal {
   void OnNetworkRouteChanged(std::optional<NetworkRoute> network_route);
   bool SetupDtls();
   void MaybeStartDtls();
-  bool HandleDtlsPacket(ArrayView<const uint8_t> payload);
+  bool HandleDtlsPacket(std::span<const uint8_t> payload);
   void OnDtlsHandshakeError(SSLHandshakeError error);
   void ConfigureHandshakeTimeout();
   void UpdateHandshakeTimeout();
diff --git a/p2p/dtls/dtls_transport_unittest.cc b/p2p/dtls/dtls_transport_unittest.cc
index b73a0db9fd..98b0ba5eee 100644
--- a/p2p/dtls/dtls_transport_unittest.cc
+++ b/p2p/dtls/dtls_transport_unittest.cc
@@ -17,6 +17,7 @@
 #include <memory>
 #include <optional>
 #include <set>
+#include <span>
 #include <string>
 #include <tuple>
 #include <utility>
@@ -25,7 +26,6 @@
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/str_cat.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/crypto/crypto_options.h"
 #include "api/dtls_transport_interface.h"
 #include "api/field_trials.h"
@@ -278,7 +278,7 @@ class DtlsTestClient {
       memset(packet.get(), sent & 0xff, size);
       packet[0] = (srtp) ? kRtpLeadByte : 0x00;
       SetBE32(
-          ArrayView<uint8_t>(
+          std::span<uint8_t>(
               reinterpret_cast<uint8_t*>(packet.get() + kPacketNumOffset), 4),
           static_cast<uint32_t>(sent));
 
@@ -312,7 +312,7 @@ class DtlsTestClient {
   size_t NumPacketsReceived() { return received_.size(); }
 
   // Inverse of SendPackets.
-  bool VerifyPacket(ArrayView<const uint8_t> payload, uint32_t* out_num) {
+  bool VerifyPacket(std::span<const uint8_t> payload, uint32_t* out_num) {
     const uint8_t* data = payload.data();
     size_t size = payload.size();
 
@@ -320,7 +320,7 @@ class DtlsTestClient {
       return false;
     }
     uint32_t packet_num =
-        GetBE32(ArrayView<const uint8_t>(data + kPacketNumOffset, 4));
+        GetBE32(std::span<const uint8_t>(data + kPacketNumOffset, 4));
     for (size_t i = kPacketHeaderLen; i < size; ++i) {
       if (data[i] != (packet_num & 0xff)) {
         return false;
@@ -338,7 +338,7 @@ class DtlsTestClient {
       return false;
     }
     uint32_t packet_num =
-        GetBE32(ArrayView<const uint8_t>(data + kPacketNumOffset, 4));
+        GetBE32(std::span<const uint8_t>(data + kPacketNumOffset, 4));
     int num_matches = 0;
     for (size_t i = kPacketNumOffset; i < size; ++i) {
       if (data[i] == (packet_num & 0xff)) {
@@ -385,7 +385,7 @@ class DtlsTestClient {
 
   SentPacketInfo sent_packet() const { return sent_packet_; }
 
-  bool IsDtlsCiphertextPacket(ArrayView<const uint8_t> payload) {
+  bool IsDtlsCiphertextPacket(std::span<const uint8_t> payload) {
     return IsDtlsPacket(payload) &&
            (payload.data()[0] > 31 && payload.data()[0] < 64);
   }
@@ -477,7 +477,7 @@ class FakeSSLStreamAdapter : public SSLStreamAdapter {
   int StartSSL() override { return impl_->StartSSL(); }
   SSLPeerCertificateDigestError SetPeerCertificateDigest(
       absl::string_view digest_alg,
-      ArrayView<const uint8_t> digest_val) override {
+      std::span<const uint8_t> digest_val) override {
     return impl_->SetPeerCertificateDigest(digest_alg, digest_val);
   }
   std::unique_ptr<SSLCertChain> GetPeerSSLCertChain() const override {
@@ -521,12 +521,12 @@ class FakeSSLStreamAdapter : public SSLStreamAdapter {
   // StreamInterface overrides.
   StreamState GetState() const override { return impl_->GetState(); }
   void Close() override { impl_->Close(); }
-  StreamResult Read(ArrayView<uint8_t> buffer,
+  StreamResult Read(std::span<uint8_t> buffer,
                     size_t& read,
                     int& error) override {
     return impl_->Read(buffer, read, error);
   }
-  StreamResult Write(ArrayView<const uint8_t> data,
+  StreamResult Write(std::span<const uint8_t> data,
                      size_t& written,
                      int& error) override {
     if (write_error_) {
@@ -768,7 +768,7 @@ TEST_F(DtlsTransportInternalImplTest, TestSendPacketWithOptions) {
   std::unique_ptr<char[]> packet(new char[size]);
   memset(packet.get(), 0, size);
   packet[0] = 0x00;
-  SetBE32(ArrayView<uint8_t>(
+  SetBE32(std::span<uint8_t>(
               reinterpret_cast<uint8_t*>(packet.get() + kPacketNumOffset), 4),
           0);
 
@@ -800,7 +800,7 @@ TEST_F(DtlsTransportInternalImplTest, TestSendSrtpBypassPacketWithOptions) {
   std::unique_ptr<char[]> packet(new char[size]);
   memset(packet.get(), 0, size);
   packet[0] = kRtpLeadByte;  // Make it look like an SRTP packet.
-  SetBE32(ArrayView<uint8_t>(
+  SetBE32(std::span<uint8_t>(
               reinterpret_cast<uint8_t*>(packet.get() + kPacketNumOffset), 4),
           0);
 
diff --git a/p2p/dtls/dtls_utils.cc b/p2p/dtls/dtls_utils.cc
index a348fa1ad6..87d188fc1d 100644
--- a/p2p/dtls/dtls_utils.cc
+++ b/p2p/dtls/dtls_utils.cc
@@ -13,10 +13,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <vector>
 
 #include "absl/container/flat_hash_set.h"
-#include "api/array_view.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/crc32.h"
@@ -30,12 +30,12 @@ const uint8_t kDtlsHandshakeRecord = 22;
 
 }  // namespace
 
-bool IsDtlsPacket(ArrayView<const uint8_t> payload) {
+bool IsDtlsPacket(std::span<const uint8_t> payload) {
   const uint8_t* u = payload.data();
   return (payload.size() >= kDtlsRecordHeaderLen && (u[0] > 19 && u[0] < 64));
 }
 
-bool IsDtlsClientHelloPacket(ArrayView<const uint8_t> payload) {
+bool IsDtlsClientHelloPacket(std::span<const uint8_t> payload) {
   if (!IsDtlsPacket(payload)) {
     return false;
   }
@@ -43,7 +43,7 @@ bool IsDtlsClientHelloPacket(ArrayView<const uint8_t> payload) {
   return payload.size() > 17 && u[0] == kDtlsHandshakeRecord && u[13] == 1;
 }
 
-bool IsDtlsHandshakePacket(ArrayView<const uint8_t> payload) {
+bool IsDtlsHandshakePacket(std::span<const uint8_t> payload) {
   if (!IsDtlsPacket(payload)) {
     return false;
   }
@@ -55,11 +55,11 @@ bool IsDtlsHandshakePacket(ArrayView<const uint8_t> payload) {
                                  payload[0] == kDtlsChangeCipherSpecRecord);
 }
 
-uint32_t ComputeDtlsPacketHash(ArrayView<const uint8_t> dtls_packet) {
+uint32_t ComputeDtlsPacketHash(std::span<const uint8_t> dtls_packet) {
   return ComputeCrc32(dtls_packet.data(), dtls_packet.size());
 }
 
-bool PacketStash::AddIfUnique(ArrayView<const uint8_t> packet) {
+bool PacketStash::AddIfUnique(std::span<const uint8_t> packet) {
   uint32_t h = ComputeDtlsPacketHash(packet);
   for (const auto& [hash, p] : packets_) {
     if (h == hash) {
@@ -72,7 +72,7 @@ bool PacketStash::AddIfUnique(ArrayView<const uint8_t> packet) {
   return true;
 }
 
-void PacketStash::Add(ArrayView<const uint8_t> packet) {
+void PacketStash::Add(std::span<const uint8_t> packet) {
   packets_.push_back(
       {.hash = ComputeDtlsPacketHash(packet),
        .buffer = std::make_unique<Buffer>(packet.data(), packet.size())});
@@ -110,20 +110,20 @@ void PacketStash::Prune(uint32_t max_size) {
   }
 }
 
-ArrayView<const uint8_t> PacketStash::GetNext() {
+std::span<const uint8_t> PacketStash::GetNext() {
   RTC_DCHECK(!packets_.empty());
   auto pos = pos_;
   pos_ = (pos + 1) % packets_.size();
   const auto& buffer = packets_[pos].buffer;
-  return ArrayView<const uint8_t>(buffer->data(), buffer->size());
+  return std::span<const uint8_t>(buffer->data(), buffer->size());
 }
 
-std::vector<ArrayView<const uint8_t>> PacketStash::GetAll() const {
-  std::vector<ArrayView<const uint8_t>> ret;
+std::vector<std::span<const uint8_t>> PacketStash::GetAll() const {
+  std::vector<std::span<const uint8_t>> ret;
   ret.reserve(packets_.size());
   for (const auto& buffer : packets_) {
     const uint8_t* ptr = buffer.buffer->data();
-    ret.push_back(ArrayView<const uint8_t>(ptr, buffer.buffer->size()));
+    ret.push_back(std::span<const uint8_t>(ptr, buffer.buffer->size()));
   }
   return ret;
 }
diff --git a/p2p/dtls/dtls_utils.h b/p2p/dtls/dtls_utils.h
index 3a60159562..e2ed5dc33b 100644
--- a/p2p/dtls/dtls_utils.h
+++ b/p2p/dtls/dtls_utils.h
@@ -14,10 +14,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <vector>
 
 #include "absl/container/flat_hash_set.h"
-#include "api/array_view.h"
 #include "rtc_base/buffer.h"
 
 namespace webrtc {
@@ -25,23 +25,23 @@ namespace webrtc {
 const size_t kDtlsRecordHeaderLen = 13;
 const size_t kMaxDtlsPacketLen = 2048;
 
-bool IsDtlsPacket(ArrayView<const uint8_t> payload);
-bool IsDtlsClientHelloPacket(ArrayView<const uint8_t> payload);
-bool IsDtlsHandshakePacket(ArrayView<const uint8_t> payload);
+bool IsDtlsPacket(std::span<const uint8_t> payload);
+bool IsDtlsClientHelloPacket(std::span<const uint8_t> payload);
+bool IsDtlsHandshakePacket(std::span<const uint8_t> payload);
 
-uint32_t ComputeDtlsPacketHash(ArrayView<const uint8_t> dtls_packet);
+uint32_t ComputeDtlsPacketHash(std::span<const uint8_t> dtls_packet);
 
 class PacketStash {
  public:
   PacketStash() {}
 
-  void Add(ArrayView<const uint8_t> packet);
-  bool AddIfUnique(ArrayView<const uint8_t> packet);
+  void Add(std::span<const uint8_t> packet);
+  bool AddIfUnique(std::span<const uint8_t> packet);
   // Returns number of elements that were removed.
   size_t Prune(const absl::flat_hash_set<uint32_t>& packet_hashes);
   void Prune(uint32_t max_size);
-  ArrayView<const uint8_t> GetNext();
-  std::vector<ArrayView<const uint8_t>> GetAll() const;
+  std::span<const uint8_t> GetNext();
+  std::vector<std::span<const uint8_t>> GetAll() const;
 
   void clear() {
     packets_.clear();
@@ -50,7 +50,7 @@ class PacketStash {
   bool empty() const { return packets_.empty(); }
   int size() const { return packets_.size(); }
 
-  static uint32_t Hash(ArrayView<const uint8_t> packet) {
+  static uint32_t Hash(std::span<const uint8_t> packet) {
     return ComputeDtlsPacketHash(packet);
   }
 
diff --git a/p2p/dtls/dtls_utils_unittest.cc b/p2p/dtls/dtls_utils_unittest.cc
index 5cd3011fad..8c3c771315 100644
--- a/p2p/dtls/dtls_utils_unittest.cc
+++ b/p2p/dtls/dtls_utils_unittest.cc
@@ -11,15 +11,15 @@
 #include "p2p/dtls/dtls_utils.h"
 
 #include <cstdint>
+#include <span>
 #include <vector>
 
 #include "absl/container/flat_hash_set.h"
-#include "api/array_view.h"
 #include "test/gtest.h"
 
 namespace webrtc {
 
-std::vector<uint8_t> ToVector(ArrayView<const uint8_t> array) {
+std::vector<uint8_t> ToVector(std::span<const uint8_t> array) {
   return std::vector<uint8_t>(array.begin(), array.end());
 }
 
diff --git a/p2p/dtls/fake_dtls_transport.h b/p2p/dtls/fake_dtls_transport.h
index 6444bf7be9..0beb398cc3 100644
--- a/p2p/dtls/fake_dtls_transport.h
+++ b/p2p/dtls/fake_dtls_transport.h
@@ -15,12 +15,12 @@
 #include <cstring>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/dtls_transport_interface.h"
 #include "api/ice_transport_interface.h"
 #include "api/rtc_error.h"
@@ -56,7 +56,7 @@ class FakeDtlsTransport : public DtlsTransportInternal {
             ice_transport_ref_->internal())),
         transport_name_(ice_transport_->transport_name()),
         component_(ice_transport_->component()),
-        dtls_fingerprint_("", ArrayView<const uint8_t>()) {
+        dtls_fingerprint_("", std::span<const uint8_t>()) {
     RTC_DCHECK(ice_transport_);
     ice_transport_->RegisterReceivedPacketCallback(
         this, [&](PacketTransportInternal* transport,
@@ -72,7 +72,7 @@ class FakeDtlsTransport : public DtlsTransportInternal {
       : owned_ice_transport_(std::move(ice)),
         transport_name_(owned_ice_transport_->transport_name()),
         component_(owned_ice_transport_->component()),
-        dtls_fingerprint_("", ArrayView<const uint8_t>()) {
+        dtls_fingerprint_("", std::span<const uint8_t>()) {
     ice_transport_ = owned_ice_transport_.get();
     ice_transport_->RegisterReceivedPacketCallback(
         this, [&](PacketTransportInternal* transport,
@@ -187,7 +187,7 @@ class FakeDtlsTransport : public DtlsTransportInternal {
   bool SetRemoteFingerprint(absl::string_view alg,
                             const uint8_t* digest,
                             size_t digest_len) {
-    dtls_fingerprint_ = SSLFingerprint(alg, MakeArrayView(digest, digest_len));
+    dtls_fingerprint_ = SSLFingerprint(alg, std::span(digest, digest_len));
     return true;
   }
   bool SetDtlsRole(SSLRole role) override {
diff --git a/p2p/test/fake_ice_transport.h b/p2p/test/fake_ice_transport.h
index f6cd168a59..864bd40131 100644
--- a/p2p/test/fake_ice_transport.h
+++ b/p2p/test/fake_ice_transport.h
@@ -16,13 +16,13 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/candidate.h"
 #include "api/field_trials.h"
 #include "api/ice_transport_interface.h"
@@ -574,7 +574,7 @@ class FakeIceTransportInternal : public IceTransportInternal {
                         << " attr: " << (dtls_piggyback_attr != nullptr)
                         << " ack: " << (dtls_piggyback_ack != nullptr);
       if (!dtls_stun_piggyback_callbacks_.empty()) {
-        std::optional<ArrayView<uint8_t>> piggyback_attr;
+        std::optional<std::span<uint8_t>> piggyback_attr;
         if (dtls_piggyback_attr) {
           piggyback_attr = dtls_piggyback_attr->array_view();
         }
@@ -614,7 +614,7 @@ class FakeIceTransportInternal : public IceTransportInternal {
     }
 
     std::unique_ptr<IceMessage> stun_msg(new IceMessage());
-    ByteBufferReader buf(MakeArrayView(packet.data(), packet.size()));
+    ByteBufferReader buf(std::span(packet.data(), packet.size()));
     RTC_CHECK(stun_msg->Read(&buf));
     return stun_msg;
   }
diff --git a/p2p/test/mock_ice_agent.h b/p2p/test/mock_ice_agent.h
index 7613a76e83..c555bf0aee 100644
--- a/p2p/test/mock_ice_agent.h
+++ b/p2p/test/mock_ice_agent.h
@@ -11,7 +11,8 @@
 #ifndef P2P_TEST_MOCK_ICE_AGENT_H_
 #define P2P_TEST_MOCK_ICE_AGENT_H_
 
-#include "api/array_view.h"
+#include <span>
+
 #include "api/units/timestamp.h"
 #include "p2p/base/connection.h"
 #include "p2p/base/ice_agent_interface.h"
@@ -32,7 +33,7 @@ class MockIceAgent : public IceAgentInterface {
   MOCK_METHOD(void, UpdateState, (), (override));
   MOCK_METHOD(void,
               ForgetLearnedStateForConnections,
-              (ArrayView<const Connection* const>),
+              (std::span<const Connection* const>),
               (override));
   MOCK_METHOD(void, SendPingRequest, (const Connection*), (override));
   MOCK_METHOD(void,
@@ -41,7 +42,7 @@ class MockIceAgent : public IceAgentInterface {
               (override));
   MOCK_METHOD(bool,
               PruneConnections,
-              (ArrayView<const Connection* const>),
+              (std::span<const Connection* const>),
               (override));
 };
 
diff --git a/p2p/test/mock_ice_controller.h b/p2p/test/mock_ice_controller.h
index f654688247..70adf88024 100644
--- a/p2p/test/mock_ice_controller.h
+++ b/p2p/test/mock_ice_controller.h
@@ -12,9 +12,9 @@
 #define P2P_TEST_MOCK_ICE_CONTROLLER_H_
 
 #include <memory>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/units/timestamp.h"
 #include "p2p/base/connection.h"
 #include "p2p/base/ice_controller_factory_interface.h"
@@ -36,7 +36,7 @@ class MockIceController : public IceControllerInterface {
   MOCK_METHOD(void, SetSelectedConnection, (const Connection*), (override));
   MOCK_METHOD(void, AddConnection, (const Connection*), (override));
   MOCK_METHOD(void, OnConnectionDestroyed, (const Connection*), (override));
-  MOCK_METHOD(ArrayView<const Connection* const>,
+  MOCK_METHOD(std::span<const Connection* const>,
               GetConnections,
               (),
               (const, override));
diff --git a/p2p/test/nat_server.cc b/p2p/test/nat_server.cc
index 46ade7e0ea..6a897b1718 100644
--- a/p2p/test/nat_server.cc
+++ b/p2p/test/nat_server.cc
@@ -14,9 +14,9 @@
 #include <cstdint>
 #include <cstring>
 #include <memory>
+#include <span>
 #include <utility>
 
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "p2p/test/nat_socket_factory.h"
 #include "p2p/test/nat_types.h"
@@ -116,8 +116,7 @@ class NATProxyServerSocket : public AsyncProxyServerSocket {
 
     SocketAddress dest_addr;
     size_t address_length = UnpackAddressFromNAT(
-        MakeArrayView(reinterpret_cast<const uint8_t*>(data), *len),
-        &dest_addr);
+        std::span(reinterpret_cast<const uint8_t*>(data), *len), &dest_addr);
     *len -= address_length;
     if (*len > 0) {
       memmove(data, data + address_length, *len);
diff --git a/p2p/test/nat_socket_factory.cc b/p2p/test/nat_socket_factory.cc
index c0ba6d9e43..02abc96aa1 100644
--- a/p2p/test/nat_socket_factory.cc
+++ b/p2p/test/nat_socket_factory.cc
@@ -15,8 +15,8 @@
 #include <cstring>
 #include <memory>
 #include <set>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
@@ -64,7 +64,7 @@ void PackAddressForNAT(const SocketAddress& remote_addr, Buffer& buf) {
 // Decodes the remote address from a packet that has been encoded with the nat's
 // quasi-STUN format. Returns the length of the address (i.e., the offset into
 // data where the original packet starts).
-size_t UnpackAddressFromNAT(ArrayView<const uint8_t> buf,
+size_t UnpackAddressFromNAT(std::span<const uint8_t> buf,
                             SocketAddress* remote_addr) {
   RTC_CHECK(buf.size() >= 8);
   RTC_DCHECK(buf.data()[0] == 0);
diff --git a/p2p/test/nat_socket_factory.h b/p2p/test/nat_socket_factory.h
index b22d6eb64d..f1b388e989 100644
--- a/p2p/test/nat_socket_factory.h
+++ b/p2p/test/nat_socket_factory.h
@@ -16,8 +16,8 @@
 #include <map>
 #include <memory>
 #include <set>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/units/time_delta.h"
 #include "p2p/test/nat_server.h"
@@ -177,7 +177,7 @@ class NATSocketServer : public SocketServer, public NATInternalSocketFactory {
 
 // Free-standing NAT helper functions.
 void PackAddressForNAT(const SocketAddress& remote_addr, Buffer& buf);
-size_t UnpackAddressFromNAT(ArrayView<const uint8_t> buf,
+size_t UnpackAddressFromNAT(std::span<const uint8_t> buf,
                             SocketAddress* remote_addr);
 }  //  namespace webrtc
 
diff --git a/p2p/test/test_port.cc b/p2p/test/test_port.cc
index fb0c9984dc..eb4161cf1d 100644
--- a/p2p/test/test_port.cc
+++ b/p2p/test/test_port.cc
@@ -13,10 +13,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <utility>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/candidate.h"
 #include "api/transport/stun.h"
 #include "p2p/base/connection.h"
@@ -39,9 +39,9 @@ TestPort::TestPort(const PortParametersRef& args,
     : Port(args, IceCandidateType::kHost, min_port, max_port) {}
 TestPort::~TestPort() = default;
 
-ArrayView<const uint8_t> TestPort::last_stun_buf() {
+std::span<const uint8_t> TestPort::last_stun_buf() {
   if (!last_stun_buf_)
-    return ArrayView<const uint8_t>();
+    return std::span<const uint8_t>();
   return *last_stun_buf_;
 }
 IceMessage* TestPort::last_stun_msg() {
diff --git a/p2p/test/test_port.h b/p2p/test/test_port.h
index b1ddcc96f5..8d10797527 100644
--- a/p2p/test/test_port.h
+++ b/p2p/test/test_port.h
@@ -14,9 +14,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <memory>
+#include <span>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/candidate.h"
 #include "api/transport/stun.h"
 #include "p2p/base/connection.h"
@@ -41,7 +41,7 @@ class TestPort : public Port {
   using Port::GetStunMessage;
 
   // The last StunMessage that was sent on this Port.
-  ArrayView<const uint8_t> last_stun_buf();
+  std::span<const uint8_t> last_stun_buf();
   IceMessage* last_stun_msg();
   int last_stun_error_code();
 
diff --git a/p2p/test/turn_server.cc b/p2p/test/turn_server.cc
index a4fc03b819..5b59ebd83f 100644
--- a/p2p/test/turn_server.cc
+++ b/p2p/test/turn_server.cc
@@ -14,6 +14,7 @@
 #include <cstddef>
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <string>
 #include <tuple>  // for std::tie
 #include <utility>
@@ -21,7 +22,6 @@
 #include "absl/algorithm/container.h"
 #include "absl/memory/memory.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/packet_socket_factory.h"
 #include "api/sequence_checker.h"
@@ -173,7 +173,7 @@ void TurnServer::OnInternalPacket(AsyncPacketSocket* socket,
                                   const ReceivedIpPacket& packet) {
   RTC_DCHECK_RUN_ON(thread_);
   // Fail if the packet is too small to even contain a channel header.
-  ArrayView<const uint8_t> payload = packet.payload();
+  std::span<const uint8_t> payload = packet.payload();
   if (payload.size() < TURN_CHANNEL_HEADER_SIZE) {
     return;
   }
@@ -197,7 +197,7 @@ void TurnServer::OnInternalPacket(AsyncPacketSocket* socket,
 }
 
 void TurnServer::HandleStunMessage(TurnServerConnection* conn,
-                                   ArrayView<const uint8_t> payload,
+                                   std::span<const uint8_t> payload,
                                    EcnMarking ecn) {
   RTC_DCHECK_RUN_ON(thread_);
   TurnMessage msg;
@@ -398,7 +398,7 @@ bool TurnServer::ValidateNonce(absl::string_view nonce) const {
   // Decode the timestamp.
   int64_t then;
   char* p = reinterpret_cast<char*>(&then);
-  size_t len = hex_decode(ArrayView<char>(p, sizeof(then)),
+  size_t len = hex_decode(std::span<char>(p, sizeof(then)),
                           nonce.substr(0, sizeof(then) * 2));
   if (len != sizeof(then)) {
     return false;
@@ -784,7 +784,7 @@ void TurnServerAllocation::HandleChannelBindRequest(const TurnMessage* msg) {
   SendResponse(&response);
 }
 
-void TurnServerAllocation::HandleChannelData(ArrayView<const uint8_t> payload,
+void TurnServerAllocation::HandleChannelData(std::span<const uint8_t> payload,
                                              EcnMarking ecn) {
   // Extract the channel number from the data.
   uint16_t channel_id = GetBE16(payload);
@@ -809,7 +809,7 @@ void TurnServerAllocation::OnExternalPacket(AsyncPacketSocket* socket,
     ByteBufferWriter buf;
     buf.WriteUInt16(channel->id);
     buf.WriteUInt16(static_cast<uint16_t>(packet.payload().size()));
-    buf.Write(ArrayView<const uint8_t>(packet.payload()));
+    buf.Write(std::span<const uint8_t>(packet.payload()));
     server_->Send(&conn_, buf, packet.ecn());
   } else if (!server_->enable_permission_checks_ ||
              HasPermission(packet.source_address().ipaddr())) {
diff --git a/p2p/test/turn_server.h b/p2p/test/turn_server.h
index 501bafe465..1258682cd9 100644
--- a/p2p/test/turn_server.h
+++ b/p2p/test/turn_server.h
@@ -16,11 +16,11 @@
 #include <list>
 #include <map>
 #include <memory>
+#include <span>
 #include <string>
 #include <utility>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/packet_socket_factory.h"
 #include "api/sequence_checker.h"
@@ -97,7 +97,7 @@ class TurnServerAllocation final {
   std::string ToString() const;
 
   void HandleTurnMessage(const TurnMessage* msg, EcnMarking ecn);
-  void HandleChannelData(ArrayView<const uint8_t> payload, EcnMarking ecn);
+  void HandleChannelData(std::span<const uint8_t> payload, EcnMarking ecn);
 
  private:
   struct Channel {
@@ -176,7 +176,7 @@ class TurnRedirectInterface {
 class StunMessageObserver {
  public:
   virtual void ReceivedMessage(const TurnMessage* msg) = 0;
-  virtual void ReceivedChannelData(ArrayView<const uint8_t> payload) = 0;
+  virtual void ReceivedChannelData(std::span<const uint8_t> payload) = 0;
   virtual ~StunMessageObserver() {}
 };
 
@@ -290,7 +290,7 @@ class TurnServer {
   void OnInternalSocketClose(AsyncPacketSocket* socket, int err);
 
   void HandleStunMessage(TurnServerConnection* conn,
-                         ArrayView<const uint8_t> payload,
+                         std::span<const uint8_t> payload,
                          EcnMarking ecn) RTC_RUN_ON(thread_);
   void HandleBindingRequest(TurnServerConnection* conn, const StunMessage* msg)
       RTC_RUN_ON(thread_);

From 98d86b9e7dd0c6f2bcee46d750f837bce6567c87 Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Mon, 9 Mar 2026 11:32:55 +0000
Subject: [PATCH 238/472] Remove unused members from WebRtcVoiceEngine and
 channels

This cleanup removes member variables that were declared but never
used in the implementation, likely as a result of previous
refactorings or the send/receive channel split.

- WebRtcVoiceEngine: removes is_dumping_aec_
- WebRtcVoiceSendChannel: removes unsignaled_frame_transformer_
- WebRtcVoiceReceiveChannel: removes send_codec_spec_

Bug: None
Change-Id: I555ceca1b72f8d0f1d9e79b92945fda6850fc2bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/455200
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47091}
---
 media/engine/webrtc_voice_engine.h | 6 ------
 1 file changed, 6 deletions(-)

diff --git a/media/engine/webrtc_voice_engine.h b/media/engine/webrtc_voice_engine.h
index 0100fc405e..97907fd4f9 100644
--- a/media/engine/webrtc_voice_engine.h
+++ b/media/engine/webrtc_voice_engine.h
@@ -162,7 +162,6 @@ class WebRtcVoiceEngine final : public VoiceEngineInterface {
   scoped_refptr<AudioState> audio_state_ RTC_GUARDED_BY(worker_thread_checker_);
   const std::vector<Codec> legacy_send_codecs_;
   const std::vector<Codec> legacy_recv_codecs_;
-  bool is_dumping_aec_ RTC_GUARDED_BY(worker_thread_checker_) = false;
   bool initialized_ RTC_GUARDED_BY(worker_thread_checker_) = false;
 
   // Jitter buffer settings for new streams.
@@ -304,8 +303,6 @@ class WebRtcVoiceSendChannel final : public MediaChannelUtil,
   // Per peer connection crypto options that last for the lifetime of the peer
   // connection.
   const CryptoOptions crypto_options_;
-  scoped_refptr<FrameTransformerInterface> unsignaled_frame_transformer_
-      RTC_GUARDED_BY(worker_thread_);
 
   // Callback invoked whenever the list of SSRCs changes.
   absl::AnyInvocable<void(const std::set<uint32_t>&)>
@@ -466,9 +463,6 @@ class WebRtcVoiceReceiveChannel final
 
   AudioReceiverParameters recv_params_ RTC_GUARDED_BY(worker_thread_);
 
-  std::optional<AudioSendStream::Config::SendCodecSpec> send_codec_spec_
-      RTC_GUARDED_BY(worker_thread_);
-
   // Per peer connection crypto options that last for the lifetime of the peer
   // connection.
   const CryptoOptions crypto_options_;

From 00a33a40b832726c75714491da107ac445555476 Mon Sep 17 00:00:00 2001
From: Jan Grulich <grulja@gmail.com>
Date: Tue, 3 Mar 2026 13:24:54 +0100
Subject: [PATCH 239/472] Wayland screencast: fix data race in CaptureFrame()

CaptureFrame() was reading pw_stream_ without holding any lock, which
races with StopAndCleanupStream() and StartScreenCastStream() accessing
it from other threads. Remove the unguarded pw_stream_ check and rely
solely on latest_available_frame_, which is already protected by
latest_frame_lock_.

Bug: webrtc:42223634
Change-Id: I8e1df44f4ec971948411be45cdf4140b2dcfc553
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453020
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Reviewed-by: Andreas Pehrson <apehrson@mozilla.com>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#47092}
---
 .../linux/wayland/shared_screencast_stream.cc          | 10 +++++-----
 1 file changed, 5 insertions(+), 5 deletions(-)

diff --git a/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc b/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc
index 44eab8fa33..d611ca0858 100644
--- a/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc
+++ b/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc
@@ -653,14 +653,14 @@ void SharedScreenCastStreamPrivate::StopAndCleanupStream() {
     pw_stream_destroy(pw_stream_);
     pw_stream_ = nullptr;
 
-    {
-      MutexLock lock(&queue_lock_);
-      queue_.Reset();
-    }
     {
       MutexLock latest_frame_lock(&latest_frame_lock_);
       latest_available_frame_ = nullptr;
     }
+    {
+      MutexLock lock(&queue_lock_);
+      queue_.Reset();
+    }
   }
 
   if (pw_core_) {
@@ -681,7 +681,7 @@ std::unique_ptr<SharedDesktopFrame>
 SharedScreenCastStreamPrivate::CaptureFrame() {
   MutexLock latest_frame_lock(&latest_frame_lock_);
 
-  if (!pw_stream_ || !latest_available_frame_) {
+  if (!latest_available_frame_) {
     return std::unique_ptr<SharedDesktopFrame>{};
   }
 

From 5f9534d9205a870d40aff649967a38fbde7d7653 Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Thu, 5 Mar 2026 03:01:52 -0800
Subject: [PATCH 240/472] Revert^3 "Default enable
 WebRTC-HeaderExtensionNegotiateMemory"

This reverts commit 2f069781037ac1f9965d39152c6c97ea98ce6aff.

Reland "Default enable WebRTC-HeaderExtensionNegotiateMemory"

This reverts commit 2f069781037ac1f9965d39152c6c97ea98ce6aff.

Reason for revert: Abundance of caution - want more canary time.

Original change's description:
> Revert^2 "Default enable WebRTC-HeaderExtensionNegotiateMemory"
>
> This reverts commit 74d988fd57f8f1b6722d8ac6f49c686b9f2bc196.
>
> Reland "Default enable WebRTC-HeaderExtensionNegotiateMemory"
>
> This reverts commit 74d988fd57f8f1b6722d8ac6f49c686b9f2bc196.
>
> Reason for revert: Added code to skip stopped transceivers
>
> Original change's description:
> > Revert "Default enable WebRTC-HeaderExtensionNegotiateMemory"
> >
> > This reverts commit 4ecf297a4c0af68ff8af8c3a8e6b0832bddd14d5.
> >
> > Reason for revert: Impacted unexpected application usages
> >
> > Original change's description:
> > > Default enable WebRTC-HeaderExtensionNegotiateMemory
> > >
> > > Spec change: https://github.com/w3c/webrtc-extensions/pull/238
> > >
> > > Chromestatus entry for change: https://chromestatus.com/feature/5135528638939136
> > >
> > > Bug: webrtc:439514253
> > > Change-Id: I296f2f06dd69b4e30be419b44c926acfd2722e2d
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/407821
> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > > Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> > > Cr-Commit-Position: refs/heads/main@{#45993}
> >
> > Bug: webrtc:439514253, chromium:467164231
> > Change-Id: I95dbc8cbaf1b769ffaea291801684bc26e4b9ccb
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/433102
> > Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#46456}
>
> Bug: webrtc:439514253, chromium:467164231
> Change-Id: I2df85ba25016374a52abb13037cdca74fb1a395d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453240
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#47038}

No-Try: True
Bug: webrtc:439514253, chromium:467164231, webrtc:489813847
Change-Id: Id2151293ffe14d2238a1363dd7df679c8a27c42f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453820
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47093}
---
 pc/rtp_transceiver.cc          | 6 +++---
 pc/rtp_transmission_manager.cc | 3 +--
 2 files changed, 4 insertions(+), 5 deletions(-)

diff --git a/pc/rtp_transceiver.cc b/pc/rtp_transceiver.cc
index 0701dca781..f1d558210c 100644
--- a/pc/rtp_transceiver.cc
+++ b/pc/rtp_transceiver.cc
@@ -1224,19 +1224,19 @@ void RtpTransceiver::OnNegotiationUpdate(
   RTC_DCHECK(content);
   if (sdp_type == SdpType::kAnswer || sdp_type == SdpType::kPrAnswer) {
     negotiated_header_extensions_ = content->rtp_header_extensions();
-    if (!env_.field_trials().IsDisabled(
+    if (env_.field_trials().IsEnabled(
             "WebRTC-HeaderExtensionNegotiateMemory")) {
       header_extensions_to_negotiate_ = GetNegotiatedHeaderExtensions();
     }
   } else if (sdp_type == SdpType::kOffer) {
-    if (!env_.field_trials().IsDisabled(
+    if (env_.field_trials().IsEnabled(
             "WebRTC-HeaderExtensionNegotiateMemory")) {
       header_extensions_for_rollback_ = header_extensions_to_negotiate_;
       header_extensions_to_negotiate_ =
           GetOfferedAndImplementedHeaderExtensions(content);
     }
   } else if (sdp_type == SdpType::kRollback) {
-    if (!env_.field_trials().IsDisabled(
+    if (env_.field_trials().IsEnabled(
             "WebRTC-HeaderExtensionNegotiateMemory")) {
       RTC_CHECK(!header_extensions_for_rollback_.empty());
       header_extensions_to_negotiate_ = header_extensions_for_rollback_;
diff --git a/pc/rtp_transmission_manager.cc b/pc/rtp_transmission_manager.cc
index b67a3fe9e7..dc6e4bc6f4 100644
--- a/pc/rtp_transmission_manager.cc
+++ b/pc/rtp_transmission_manager.cc
@@ -284,8 +284,7 @@ RtpTransmissionManager::CreateAndAddTransceiver(
   RTC_DCHECK(!FindSenderById(sender_id));
   std::vector<RtpHeaderExtensionCapability> header_extensions =
       std::move(header_extensions_to_negotiate);
-  if (!env_.field_trials().IsDisabled(
-          "WebRTC-HeaderExtensionNegotiateMemory")) {
+  if (env_.field_trials().IsEnabled("WebRTC-HeaderExtensionNegotiateMemory")) {
     // If we have already negotiated header extensions for this type,
     // and it is not stopped,
     // reuse the negotiated state for new transceivers of the same type.

From 7efbf637b0bf4d9d2296a44550c51e50b4c44532 Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Mon, 9 Mar 2026 21:05:47 -0700
Subject: [PATCH 241/472] Update WebRTC code version (2026-03-10T04:05:46).

Bug: None
Change-Id: Ib47eaea55d9c2826008d8e9a3464e40b3cfe3b30
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/455303
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47094}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index fa5aebef55..5a346f0c72 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-07T04:04:41";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-10T04:05:46";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From 3907bf2bdbd8e1a1f93563457214f4f996349bf7 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Mon, 9 Mar 2026 22:31:21 +0100
Subject: [PATCH 242/472] Use asynchronous posting for PeerConnection signaling
 tasks

Update RunOnSignalingThread to use PostTask with thread safety flags
instead of BlockingCall. The places where RunOnSignalingThread is used,
bypass the proxy but used BlockingCall to remain functionally compatible
with the proxies. The methods are however asynchronous, so the
BlockingCall can be avoided. That was not done in the first step when
RunOnSignalingThread was introduced in order to not bring too many
changes (the main focus was on allowing non thread-hopping getters).
This change is a follow up step that transitions the initial step of the
task execution from a synchronous, blocking operation to an asynchronous
when called from a different thread.

As a btw change, SetLocalDescription and SetRemoteDescription now
consistently manage the lifetime of their arguments using
std::unique_ptr and scoped_refptr to ensure that
SessionDescriptionInterface and SetSessionDescriptionObserver objects
remain valid until the asynchronous task completes on the signaling
thread, preventing potential use-after-free or memory leak issues.

Bug: webrtc:442220720
Change-Id: I71cda2643c8ddd89622436df08bf889c0a35c56d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/454980
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47095}
---
 pc/peer_connection.cc  | 29 +++++++++++++----------
 pc/sdp_offer_answer.cc | 54 +++++++++++++-----------------------------
 pc/sdp_offer_answer.h  | 13 ++++++----
 3 files changed, 40 insertions(+), 56 deletions(-)

diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc
index 57812745ea..cd4ad8b463 100644
--- a/pc/peer_connection.cc
+++ b/pc/peer_connection.cc
@@ -1538,12 +1538,15 @@ void PeerConnection::CreateAnswer(CreateSessionDescriptionObserver* observer,
 void PeerConnection::SetLocalDescription(
     SetSessionDescriptionObserver* observer,
     SessionDescriptionInterface* desc) {
-  if (!CheckValidSetDescription(observer, desc, signaling_thread()))
+  auto desc_ptr = std::unique_ptr<SessionDescriptionInterface>(desc);
+  if (!CheckValidSetDescription(observer, desc_ptr, signaling_thread()))
     return;
 
-  RunOnSignalingThread([this, desc, observer]() mutable {
+  RunOnSignalingThread([this, desc = std::move(desc_ptr),
+                        observer = scoped_refptr<SetSessionDescriptionObserver>(
+                            observer)]() mutable {
     RTC_DCHECK_RUN_ON(signaling_thread());
-    sdp_handler_->SetLocalDescription(observer, desc);
+    sdp_handler_->SetLocalDescription(std::move(observer), std::move(desc));
   });
 }
 
@@ -1564,7 +1567,8 @@ void PeerConnection::SetLocalDescription(
 void PeerConnection::SetLocalDescription(
     SetSessionDescriptionObserver* observer) {
   RTC_DCHECK_RUN_ON(signaling_thread());
-  sdp_handler_->SetLocalDescription(observer);
+  sdp_handler_->SetLocalDescription(
+      scoped_refptr<SetSessionDescriptionObserver>(observer));
 }
 
 void PeerConnection::SetLocalDescription(
@@ -1576,12 +1580,15 @@ void PeerConnection::SetLocalDescription(
 void PeerConnection::SetRemoteDescription(
     SetSessionDescriptionObserver* observer,
     SessionDescriptionInterface* desc) {
-  if (!CheckValidSetDescription(observer, desc, signaling_thread()))
+  auto desc_ptr = std::unique_ptr<SessionDescriptionInterface>(desc);
+  if (!CheckValidSetDescription(observer, desc_ptr, signaling_thread()))
     return;
 
-  RunOnSignalingThread([this, desc, observer]() mutable {
+  RunOnSignalingThread([this, desc = std::move(desc_ptr),
+                        observer = scoped_refptr<SetSessionDescriptionObserver>(
+                            observer)]() mutable {
     RTC_DCHECK_RUN_ON(signaling_thread());
-    sdp_handler_->SetRemoteDescription(observer, desc);
+    sdp_handler_->SetRemoteDescription(std::move(observer), std::move(desc));
   });
 }
 
@@ -3196,12 +3203,8 @@ void PeerConnection::RunOnSignalingThread(absl::AnyInvocable<void() &&> task) {
   if (signaling_thread()->IsCurrent()) {
     std::move(task)();
   } else {
-    // Consider if we can use PostTask instead in some situations:
-    //   signaling_thread()->PostTask(
-    //      SafeTask(signaling_thread_safety_.flag(), std::move(task)));
-    // Currently we use BlockingCall() to be compatible with how the
-    // api proxies work by default.
-    signaling_thread()->BlockingCall([&] { std::move(task)(); });
+    signaling_thread()->PostTask(
+        SafeTask(signaling_thread_safety_.flag(), std::move(task)));
   }
 }
 
diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc
index 10f2f13fc1..590a46563f 100644
--- a/pc/sdp_offer_answer.cc
+++ b/pc/sdp_offer_answer.cc
@@ -1733,19 +1733,18 @@ void SdpOfferAnswerHandler::CreateOffer(
 }
 
 void SdpOfferAnswerHandler::SetLocalDescription(
-    SetSessionDescriptionObserver* observer,
-    SessionDescriptionInterface* desc_ptr) {
+    scoped_refptr<SetSessionDescriptionObserver> observer,
+    std::unique_ptr<SessionDescriptionInterface> desc) {
   RTC_LOG_THREAD_BLOCK_COUNT();
   RTC_DCHECK_RUN_ON(signaling_thread());
-  RTC_DCHECK(desc_ptr);
+  RTC_DCHECK(desc);
   RTC_DCHECK(observer);
   // Chain this operation. If asynchronous operations are pending on the chain,
   // this operation will be queued to be invoked, otherwise the contents of the
   // lambda will execute immediately.
   operations_chain_->ChainOperation(
       [this_weak_ptr = weak_ptr_factory_.GetWeakPtr(),
-       observer_refptr = scoped_refptr<SetSessionDescriptionObserver>(observer),
-       desc = std::unique_ptr<SessionDescriptionInterface>(desc_ptr)](
+       observer_refptr = observer, desc = std::move(desc)](
           std::function<void()> operations_chain_callback) mutable {
         // Abort early if `this_weak_ptr` is no longer valid.
         if (!this_weak_ptr) {
@@ -1801,13 +1800,13 @@ void SdpOfferAnswerHandler::SetLocalDescription(
 }
 
 void SdpOfferAnswerHandler::SetLocalDescription(
-    SetSessionDescriptionObserver* observer) {
+    scoped_refptr<SetSessionDescriptionObserver> observer) {
   RTC_LOG_THREAD_BLOCK_COUNT();
   RTC_DCHECK_RUN_ON(signaling_thread());
-  RTC_DCHECK(observer);
-  SetLocalDescription(make_ref_counted<SetSessionDescriptionObserverAdapter>(
-      weak_ptr_factory_.GetWeakPtr(),
-      scoped_refptr<SetSessionDescriptionObserver>(observer)));
+  scoped_refptr<SetLocalDescriptionObserverInterface> adapter =
+      make_ref_counted<SetSessionDescriptionObserverAdapter>(
+          weak_ptr_factory_.GetWeakPtr(), std::move(observer));
+  SetLocalDescription(std::move(adapter));
 }
 
 void SdpOfferAnswerHandler::SetLocalDescription(
@@ -2144,36 +2143,15 @@ RTCError SdpOfferAnswerHandler::ApplyLocalDescription(
 }
 
 void SdpOfferAnswerHandler::SetRemoteDescription(
-    SetSessionDescriptionObserver* observer,
-    SessionDescriptionInterface* desc_ptr) {
+    scoped_refptr<SetSessionDescriptionObserver> observer,
+    std::unique_ptr<SessionDescriptionInterface> desc) {
   RTC_DCHECK_RUN_ON(signaling_thread());
   RTC_DCHECK(observer);
-  RTC_DCHECK(desc_ptr);
-  // Chain this operation. If asynchronous operations are pending on the chain,
-  // this operation will be queued to be invoked, otherwise the contents of the
-  // lambda will execute immediately.
-  operations_chain_->ChainOperation(
-      [this_weak_ptr = weak_ptr_factory_.GetWeakPtr(),
-       observer_refptr = scoped_refptr<SetSessionDescriptionObserver>(observer),
-       desc = std::unique_ptr<SessionDescriptionInterface>(desc_ptr)](
-          std::function<void()> operations_chain_callback) mutable {
-        // Abort early if `this_weak_ptr` is no longer valid.
-        if (!this_weak_ptr) {
-          // For consistency with SetSessionDescriptionObserverAdapter whose
-          // posted messages doesn't get processed when the PC is destroyed, we
-          // do not inform `observer_refptr` that the operation failed.
-          operations_chain_callback();
-          return;
-        }
-        // SetSessionDescriptionObserverAdapter takes care of making sure the
-        // `observer_refptr` is invoked in a posted message.
-        this_weak_ptr->DoSetRemoteDescription(
-            std::make_unique<RemoteDescriptionOperation>(
-                this_weak_ptr.get(), std::move(desc),
-                make_ref_counted<SetSessionDescriptionObserverAdapter>(
-                    this_weak_ptr, observer_refptr),
-                std::move(operations_chain_callback)));
-      });
+  RTC_DCHECK(desc);
+  SetRemoteDescription(
+      std::move(desc),
+      make_ref_counted<SetSessionDescriptionObserverAdapter>(
+          weak_ptr_factory_.GetWeakPtr(), std::move(observer)));
 }
 
 void SdpOfferAnswerHandler::SetRemoteDescription(
diff --git a/pc/sdp_offer_answer.h b/pc/sdp_offer_answer.h
index 05fca5920e..7be0f6626a 100644
--- a/pc/sdp_offer_answer.h
+++ b/pc/sdp_offer_answer.h
@@ -148,15 +148,18 @@ class SdpOfferAnswerHandler : public SdpStateProvider {
       scoped_refptr<SetLocalDescriptionObserverInterface> observer);
   void SetLocalDescription(
       scoped_refptr<SetLocalDescriptionObserverInterface> observer);
-  void SetLocalDescription(SetSessionDescriptionObserver* observer,
-                           SessionDescriptionInterface* desc);
-  void SetLocalDescription(SetSessionDescriptionObserver* observer);
+  void SetLocalDescription(
+      scoped_refptr<SetSessionDescriptionObserver> observer,
+      std::unique_ptr<SessionDescriptionInterface> desc);
+  void SetLocalDescription(
+      scoped_refptr<SetSessionDescriptionObserver> observer);
 
   void SetRemoteDescription(
       std::unique_ptr<SessionDescriptionInterface> desc,
       scoped_refptr<SetRemoteDescriptionObserverInterface> observer);
-  void SetRemoteDescription(SetSessionDescriptionObserver* observer,
-                            SessionDescriptionInterface* desc);
+  void SetRemoteDescription(
+      scoped_refptr<SetSessionDescriptionObserver> observer,
+      std::unique_ptr<SessionDescriptionInterface> desc);
 
   PeerConnectionInterface::RTCConfiguration GetConfiguration();
   RTCError SetConfiguration(

From d4ef54417d8c961796008aa8766fb2b5dbf93134 Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Mon, 9 Mar 2026 15:12:30 +0000
Subject: [PATCH 243/472] Emit CCFB parameters per codec, not wildcard

This works around the inability of some parsers to handle wildcards.

Bug: webrtc:489794442
Change-Id: I4633e1d27139e13ec571bbd169ed38c324922248
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/455280
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47096}
---
 api/webrtc_sdp.cc                             | 42 ++++++++++++++-----
 pc/congestion_control_integrationtest.cc      |  9 +++-
 .../bwe_integration_tests/l4s_test.cc         | 13 ++++--
 3 files changed, 48 insertions(+), 16 deletions(-)

diff --git a/api/webrtc_sdp.cc b/api/webrtc_sdp.cc
index f404b90eb7..aab9251257 100644
--- a/api/webrtc_sdp.cc
+++ b/api/webrtc_sdp.cc
@@ -1117,12 +1117,21 @@ void AddOrReplaceCodec(MediaContentDescription* content_desc,
 
 void BuildRtpmap(const MediaContentDescription* media_desc,
                  const MediaType media_type,
+                 bool use_wildcard,
                  std::string* message) {
   RTC_DCHECK(message != nullptr);
   RTC_DCHECK(media_desc != nullptr);
   StringBuilder os;
   if (media_type == MediaType::VIDEO) {
-    for (const Codec& codec : media_desc->codecs()) {
+    for (Codec codec : media_desc->codecs()) {
+      // Include transport-wide codec parameters.
+      // TODO: issues.webrtc.org/489794442 - Remove when wildcards are
+      // OK to send.
+      if (!use_wildcard) {
+        if (media_desc->rtcp_fb_ack_ccfb()) {
+          codec.feedback_params.Add({"ack", "ccfb"});
+        }
+      }
       // RFC 4566
       // a=rtpmap:<payload type> <encoding name>/<clock rate>
       // [/<encodingparameters>]
@@ -1140,8 +1149,16 @@ void BuildRtpmap(const MediaContentDescription* media_desc,
     std::vector<int> ptimes;
     std::vector<int> maxptimes;
     int max_minptime = 0;
-    for (const Codec& codec : media_desc->codecs()) {
+    for (Codec codec : media_desc->codecs()) {
       RTC_DCHECK(!codec.name.empty());
+      // Include transport-wide codec parameters.
+      // TODO: issues.webrtc.org/489794442 - Remove when wildcards are
+      // OK to send.
+      if (!use_wildcard) {
+        if (media_desc->rtcp_fb_ack_ccfb()) {
+          codec.feedback_params.Add({"ack", "ccfb"});
+        }
+      }
       // RFC 4566
       // a=rtpmap:<payload type> <encoding name>/<clock rate>
       // [/<encodingparameters>]
@@ -1183,18 +1200,21 @@ void BuildRtpmap(const MediaContentDescription* media_desc,
       AddAttributeLine(kCodecParamPTime, ptime, message);
     }
   }
-  if (media_desc->rtcp_fb_ack_ccfb()) {
-    // RFC 8888 section 6
-    InitAttrLine(kAttributeRtcpFb, &os);
-    os << kSdpDelimiterColon;
-    os << "* ack ccfb";
-    AddLine(os.str(), message);
+  if (use_wildcard) {
+    if (media_desc->rtcp_fb_ack_ccfb()) {
+      // RFC 8888 section 6
+      InitAttrLine(kAttributeRtcpFb, &os);
+      os << kSdpDelimiterColon;
+      os << "* ack ccfb";
+      AddLine(os.str(), message);
+    }
   }
 }
 
 void BuildRtpContentAttributes(const MediaContentDescription* media_desc,
                                const MediaType media_type,
                                int msid_signaling,
+                               bool use_wildcard,
                                std::string* message) {
   SimulcastSdpSerializer serializer;
   StringBuilder os;
@@ -1289,7 +1309,7 @@ void BuildRtpContentAttributes(const MediaContentDescription* media_desc,
   // RFC 4566
   // a=rtpmap:<payload type> <encoding name>/<clock rate>
   // [/<encodingparameters>]
-  BuildRtpmap(media_desc, media_type, message);
+  BuildRtpmap(media_desc, media_type, use_wildcard, message);
 
   for (const StreamParams& track : media_desc->streams()) {
     // Build the ssrc-group lines.
@@ -1449,7 +1469,9 @@ void BuildMediaDescription(const ContentInfo* content_info,
   if (IsDtlsSctp(media_desc->protocol())) {
     BuildSctpContentAttributes(media_desc, message);
   } else if (IsRtpProtocol(media_desc->protocol())) {
-    BuildRtpContentAttributes(media_desc, media_type, msid_signaling, message);
+    // TODO: issues.webrtc.org/48979442 - Make this field trial controlled
+    BuildRtpContentAttributes(media_desc, media_type, msid_signaling,
+                              /* use_wildcard= */ false, message);
   }
 }
 
diff --git a/pc/congestion_control_integrationtest.cc b/pc/congestion_control_integrationtest.cc
index b712b6ba25..4cf7793e04 100644
--- a/pc/congestion_control_integrationtest.cc
+++ b/pc/congestion_control_integrationtest.cc
@@ -13,6 +13,7 @@
 
 #include <memory>
 #include <string>
+#include <string_view>
 #include <vector>
 
 #include "absl/strings/str_cat.h"
@@ -31,6 +32,7 @@
 
 namespace webrtc {
 
+using ::testing::ContainsRegex;
 using ::testing::Eq;
 using ::testing::Field;
 using ::testing::Gt;
@@ -45,6 +47,9 @@ class PeerConnectionCongestionControlTest
       : PeerConnectionIntegrationBaseTest(SdpSemantics::kUnifiedPlan) {}
 };
 
+// This regexp matches both wildcard and non-wildcard ccfb lines.
+constexpr std::string_view ccfb_regex = "a=rtcp-fb:[0-9*]* ack ccfb\r\n";
+
 TEST_F(PeerConnectionCongestionControlTest,
        OfferDoesNotContainCcfbEvenIfEnabled) {
   SetFieldTrials("WebRTC-RFC8888CongestionControlFeedback/Enabled/");
@@ -53,7 +58,7 @@ TEST_F(PeerConnectionCongestionControlTest,
   std::unique_ptr<SessionDescriptionInterface> offer =
       caller()->CreateOfferAndWait();
   std::string offer_str = absl::StrCat(*offer);
-  EXPECT_THAT(offer_str, Not(HasSubstr("a=rtcp-fb:* ack ccfb\r\n")));
+  EXPECT_THAT(offer_str, Not(ContainsRegex(ccfb_regex)));
 }
 
 TEST_F(PeerConnectionCongestionControlTest, OfferContainsCcfbIfFieldTrial) {
@@ -63,7 +68,7 @@ TEST_F(PeerConnectionCongestionControlTest, OfferContainsCcfbIfFieldTrial) {
   std::unique_ptr<SessionDescriptionInterface> offer =
       caller()->CreateOfferAndWait();
   std::string offer_str = absl::StrCat(*offer);
-  EXPECT_THAT(offer_str, HasSubstr("a=rtcp-fb:* ack ccfb\r\n"));
+  EXPECT_THAT(offer_str, ContainsRegex(ccfb_regex));
 }
 
 TEST_F(PeerConnectionCongestionControlTest, ReceiveOfferSetsCcfbFlag) {
diff --git a/test/peer_scenario/bwe_integration_tests/l4s_test.cc b/test/peer_scenario/bwe_integration_tests/l4s_test.cc
index 98fdc9636d..1d1f5c3f97 100644
--- a/test/peer_scenario/bwe_integration_tests/l4s_test.cc
+++ b/test/peer_scenario/bwe_integration_tests/l4s_test.cc
@@ -11,6 +11,7 @@
 #include <atomic>
 #include <map>
 #include <string>
+#include <string_view>
 #include <vector>
 
 #include "absl/strings/str_cat.h"
@@ -54,12 +55,16 @@ using test::GetPacketsSentWithEct1;
 using test::GetStatsAndProcess;
 using test::PeerScenario;
 using test::PeerScenarioClient;
+using ::testing::ContainsRegex;
 using ::testing::HasSubstr;
 using ::testing::TestWithParam;
 
 // RTC event logs can be gathered from these tests.
 // Add --peer_logs=true --peer_logs_root=/tmp/l4s/ to write logs to /tmp/l4s
 
+// This regexp matches both wildcard and non-wildcard ccfb lines.
+constexpr std::string_view ccfb_regex = "a=rtcp-fb:[0-9*]* ack ccfb\r\n";
+
 // Helper class used for counting RTCP feedback messages.
 class RtcpFeedbackCounter {
  public:
@@ -161,7 +166,7 @@ TEST(L4STest, NegotiateAndUseCcfbIfEnabled) {
         // Check that the offer contain both congestion control feedback
         // according to RFC 8888, and transport-cc and the header extension
         // http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
-        EXPECT_THAT(offer_str, HasSubstr("a=rtcp-fb:* ack ccfb\r\n"));
+        EXPECT_THAT(offer_str, ContainsRegex(ccfb_regex));
         EXPECT_THAT(offer_str, HasSubstr("transport-cc"));
         EXPECT_THAT(
             offer_str,
@@ -170,7 +175,7 @@ TEST(L4STest, NegotiateAndUseCcfbIfEnabled) {
       },
       [&](const SessionDescriptionInterface& answer) {
         std::string answer_str = absl::StrCat(answer);
-        EXPECT_THAT(answer_str, HasSubstr("a=rtcp-fb:* ack ccfb\r\n"));
+        EXPECT_THAT(answer_str, ContainsRegex(ccfb_regex));
         // Check that the answer does not contain transport-cc nor the
         // header extension
         // http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
@@ -266,7 +271,7 @@ TEST(L4STest, NoCcfbSentAfterRenegotiationAndCallerCachesLocalDescription) {
   std::string answer_str;
   caller->pc()->local_description()->ToString(&answer_str);
   ASSERT_FALSE(answer_str.empty());
-  ASSERT_THAT(answer_str, HasSubstr("a=rtcp-fb:* ack ccfb\r\n"));
+  ASSERT_THAT(answer_str, ContainsRegex(ccfb_regex));
 
   callee->CreateAndSetSdp(
       [&](SessionDescriptionInterface* /*munge_offer*/) {
@@ -274,7 +279,7 @@ TEST(L4STest, NoCcfbSentAfterRenegotiationAndCallerCachesLocalDescription) {
       },
       [&](std::string offer) {
         // Callee does not support ccfb and does not have it in the offer.
-        ASSERT_THAT(offer, Not(HasSubstr("a=rtcp-fb:* ack ccfb\r\n")));
+        ASSERT_THAT(offer, Not(ContainsRegex(ccfb_regex)));
         caller->SetRemoteDescription(
             offer, SdpType::kOffer, [&](RTCError error) {
               ASSERT_TRUE(error.ok());

From d095dd2d1896a0105ea3f657442f2eb7630bdf65 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Erik=20Spr=C3=A5ng?= <sprang@webrtc.org>
Date: Tue, 3 Mar 2026 14:53:59 +0100
Subject: [PATCH 244/472] Clean up some video-related OWNERS files.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This removes some folk who are not that active in libwebrtc anymore.

Bug: None
Change-Id: I798a5c079ab8cb9119ac8708caa59597c16dbe01
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453260
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47097}
---
 api/OWNERS                   | 1 -
 api/video/OWNERS             | 1 -
 api/video_codecs/OWNERS      | 1 -
 common_video/OWNERS          | 3 ---
 media/OWNERS                 | 2 --
 modules/video_capture/OWNERS | 1 -
 modules/video_coding/OWNERS  | 2 --
 video/OWNERS                 | 2 --
 8 files changed, 13 deletions(-)

diff --git a/api/OWNERS b/api/OWNERS
index ebaea3b1e7..5eeb7f6340 100644
--- a/api/OWNERS
+++ b/api/OWNERS
@@ -1,5 +1,4 @@
 hta@webrtc.org
-magjed@webrtc.org
 perkj@webrtc.org
 tommi@webrtc.org
 
diff --git a/api/video/OWNERS b/api/video/OWNERS
index a30dab0903..071178a41e 100644
--- a/api/video/OWNERS
+++ b/api/video/OWNERS
@@ -1,5 +1,4 @@
 brandtr@webrtc.org
-magjed@webrtc.org
 philipel@webrtc.org
 sprang@webrtc.org
 
diff --git a/api/video_codecs/OWNERS b/api/video_codecs/OWNERS
index f73b04f829..6766cdf9b9 100644
--- a/api/video_codecs/OWNERS
+++ b/api/video_codecs/OWNERS
@@ -1,4 +1,3 @@
-magjed@webrtc.org
 sprang@webrtc.org
 brandtr@webrtc.org
 philipel@webrtc.org
diff --git a/common_video/OWNERS b/common_video/OWNERS
index 1c080874f0..4b06c4e32b 100644
--- a/common_video/OWNERS
+++ b/common_video/OWNERS
@@ -1,5 +1,2 @@
-magjed@webrtc.org
-marpan@webrtc.org
 sprang@webrtc.org
 ssilkin@webrtc.org
-stefan@webrtc.org
diff --git a/media/OWNERS b/media/OWNERS
index cf40b1c25d..4003d48c49 100644
--- a/media/OWNERS
+++ b/media/OWNERS
@@ -1,8 +1,6 @@
 brandtr@webrtc.org
 ilnik@webrtc.org
 sprang@webrtc.org
-magjed@webrtc.org
-mflodman@webrtc.org
 perkj@webrtc.org
 
 # Audio-related changes:
diff --git a/modules/video_capture/OWNERS b/modules/video_capture/OWNERS
index 364d66d36f..65306a0a1c 100644
--- a/modules/video_capture/OWNERS
+++ b/modules/video_capture/OWNERS
@@ -1,4 +1,3 @@
 ilnik@webrtc.org
-mflodman@webrtc.org
 perkj@webrtc.org
 tkchin@webrtc.org
diff --git a/modules/video_coding/OWNERS b/modules/video_coding/OWNERS
index 5073079d34..f9f3e9bb09 100644
--- a/modules/video_coding/OWNERS
+++ b/modules/video_coding/OWNERS
@@ -1,8 +1,6 @@
 asapersson@webrtc.org
 brandtr@webrtc.org
 ilnik@webrtc.org
-marpan@webrtc.org
 philipel@webrtc.org
 sprang@webrtc.org
 ssilkin@webrtc.org
-stefan@webrtc.org
diff --git a/video/OWNERS b/video/OWNERS
index 3a278f3a85..bbfd8db420 100644
--- a/video/OWNERS
+++ b/video/OWNERS
@@ -1,7 +1,5 @@
 asapersson@webrtc.org
 ilnik@webrtc.org
-mflodman@webrtc.org
 philipel@webrtc.org
 sprang@webrtc.org
 ssilkin@webrtc.org
-stefan@webrtc.org

From b9f91bc93fe5922a4e0f71019baef79e82f94e39 Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Mon, 9 Mar 2026 12:45:31 +0000
Subject: [PATCH 245/472] Clean up unused members in WebRtcVideoEngine classes

Following the split of the two-way video channel into separate send and
receive channels, several members and methods became unused or redundant.

This CL:
- Removes unused private methods Construct and
  FillBandwidthEstimationStats from WebRtcVideoSendChannel.
- Removes unused member variables last_unsignalled_ssrc_creation_time_ms_
  and rtp_config_ from WebRtcVideoSendChannel.
- Removes unused member variable rtp_config_ from
  WebRtcVideoReceiveChannel.
- Removes redundant initialization of rtcp_mode in
  WebRtcVideoReceiveChannel::ConfigureReceiverRtp.

Bug: None
Change-Id: Ice3aa1788f60a2c192c271e42d503e17fc4c649c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/454702
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47098}
---
 media/engine/webrtc_video_engine.cc |  4 ----
 media/engine/webrtc_video_engine.h  | 19 -------------------
 2 files changed, 23 deletions(-)

diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc
index 8bfffdc7e4..fc04a6c2b6 100644
--- a/media/engine/webrtc_video_engine.cc
+++ b/media/engine/webrtc_video_engine.cc
@@ -3076,10 +3076,6 @@ void WebRtcVideoReceiveChannel::ConfigureReceiverRtp(
 
   config->rtp.remote_ssrc = ssrc;
 
-  // The mode and rtx time is determined by a call to the configuration
-  // function.
-  config->rtp.rtcp_mode = rtp_config_.rtcp_mode;
-
   sp.GetFidSsrc(ssrc, &config->rtp.rtx_ssrc);
 
   // TODO(brandtr): Generalize when we add support for multistream protection.
diff --git a/media/engine/webrtc_video_engine.h b/media/engine/webrtc_video_engine.h
index 307768f2a3..d79174f6b0 100644
--- a/media/engine/webrtc_video_engine.h
+++ b/media/engine/webrtc_video_engine.h
@@ -432,8 +432,6 @@ class WebRtcVideoSendChannel : public MediaChannelUtil,
     const bool disable_automatic_resize_;
   };
 
-  void Construct(Call* call, WebRtcVideoEngine* engine);
-
   // Get all codecs that are compatible with the receiver.
   std::vector<VideoCodecSettings> SelectSendVideoCodecs(
       const std::vector<VideoCodecSettings>& remote_mapped_codecs) const
@@ -441,9 +439,6 @@ class WebRtcVideoSendChannel : public MediaChannelUtil,
 
   void FillSenderStats(VideoMediaSendInfo* info, bool log_stats)
       RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
-  void FillBandwidthEstimationStats(const Call::Stats& stats,
-                                    VideoMediaInfo* info)
-      RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
   void FillSendCodecStats(VideoMediaSendInfo* video_media_info)
       RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
 
@@ -471,8 +466,6 @@ class WebRtcVideoSendChannel : public MediaChannelUtil,
   // Using primary-ssrc (first ssrc) as key.
   std::map<uint32_t, WebRtcVideoSendStream*> send_streams_
       RTC_GUARDED_BY(thread_checker_);
-  std::optional<int64_t> last_unsignalled_ssrc_creation_time_ms_
-      RTC_GUARDED_BY(thread_checker_);
   std::set<uint32_t> send_ssrcs_ RTC_GUARDED_BY(thread_checker_);
 
   std::optional<VideoCodecSettings> send_codec_ RTC_GUARDED_BY(thread_checker_);
@@ -489,8 +482,6 @@ class WebRtcVideoSendChannel : public MediaChannelUtil,
   // See reason for keeping track of the FlexFEC payload type separately in
   // comment in WebRtcVideoChannel::ChangedReceiverParameters.
   BitrateConstraints bitrate_config_ RTC_GUARDED_BY(thread_checker_);
-  // TODO(deadbeef): Don't duplicate information between
-  // send_params/recv_params, rtp_extensions, options, etc.
   VideoSenderParameters send_params_ RTC_GUARDED_BY(thread_checker_);
   VideoOptions default_send_options_ RTC_GUARDED_BY(thread_checker_);
   int64_t last_send_stats_log_ms_ RTC_GUARDED_BY(thread_checker_);
@@ -498,11 +489,6 @@ class WebRtcVideoSendChannel : public MediaChannelUtil,
   // connection.
   const CryptoOptions crypto_options_ RTC_GUARDED_BY(thread_checker_);
 
-  // RTP parameters that need to be set when creating a video receive stream.
-  // Only used in Receiver mode - in Both mode, it reads those things from the
-  // codec.
-  VideoReceiveStreamInterface::Config::Rtp rtp_config_;
-
   // Callback invoked whenever the list of SSRCs changes.
   absl::AnyInvocable<void(const std::set<uint32_t>&)>
       ssrc_list_changed_callback_;
@@ -785,11 +771,6 @@ class WebRtcVideoReceiveChannel : public MediaChannelUtil,
   scoped_refptr<FrameTransformerInterface> unsignaled_frame_transformer_
       RTC_GUARDED_BY(thread_checker_);
 
-  // RTP parameters that need to be set when creating a video receive stream.
-  // Only used in Receiver mode - in Both mode, it reads those things from the
-  // codec.
-  VideoReceiveStreamInterface::Config::Rtp rtp_config_;
-
   const int receive_buffer_size_;
 };
 

From 96625b9b48887e947a8e9737483d69efbf02e239 Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Tue, 10 Mar 2026 09:40:33 +0000
Subject: [PATCH 246/472] Remove deprecated charset handling in
 AsyncHttpUrlConnection

This was blocking chromium->webrtc roll.

Bug: None
Change-Id: Ie9833da682303b08331ab48867e5065b55f23b96
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/455282
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47099}
---
 .../src/org/appspot/apprtc/util/AsyncHttpURLConnection.java    | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

diff --git a/examples/androidapp/src/org/appspot/apprtc/util/AsyncHttpURLConnection.java b/examples/androidapp/src/org/appspot/apprtc/util/AsyncHttpURLConnection.java
index 5a8ae20961..bc655bac8b 100644
--- a/examples/androidapp/src/org/appspot/apprtc/util/AsyncHttpURLConnection.java
+++ b/examples/androidapp/src/org/appspot/apprtc/util/AsyncHttpURLConnection.java
@@ -16,6 +16,7 @@ import java.io.OutputStream;
 import java.net.HttpURLConnection;
 import java.net.SocketTimeoutException;
 import java.net.URL;
+import java.nio.charset.Charset;
 import java.util.Scanner;
 
 /**
@@ -59,7 +60,7 @@ public class AsyncHttpURLConnection {
       HttpURLConnection connection = (HttpURLConnection) new URL(url).openConnection();
       byte[] postData = new byte[0];
       if (message != null) {
-        postData = message.getBytes("UTF-8");
+        postData = message.getBytes(Charset.forName("UTF-8"));
       }
       connection.setRequestMethod(method);
       connection.setUseCaches(false);

From 3965af1ab963fc116c7b78a8dd64789a41414669 Mon Sep 17 00:00:00 2001
From: Jan Grulich <grulja@gmail.com>
Date: Tue, 10 Mar 2026 10:58:46 +0100
Subject: [PATCH 247/472] PipeWire: call pw_deinit() only when running against
 PipeWire 3.49+

pw_deinit() was not reetrant in old PipeWire versions and it only got
fixed with PipeWire 3.49. This is causing a crash for old Ubuntu 22.04
with PipeWire 3.48 when sharing screen multiple times.

This also moves all the PipeWire version parsing logic to pipewire_utils
from screencast_utils as it makes more sense and is used from there now.

Bug: chromium:490340738
Change-Id: I62ec7adfcba040b4ab99bf535b3f16b5e4b27c1a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/455180
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Andreas Pehrson <apehrson@mozilla.com>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#47100}
---
 .../linux/wayland/screencast_stream_utils.cc  | 47 ----------------
 .../linux/wayland/screencast_stream_utils.h   | 18 -------
 modules/portal/pipewire_utils.cc              | 53 ++++++++++++++++++-
 modules/portal/pipewire_utils.h               | 16 ++++++
 4 files changed, 68 insertions(+), 66 deletions(-)

diff --git a/modules/desktop_capture/linux/wayland/screencast_stream_utils.cc b/modules/desktop_capture/linux/wayland/screencast_stream_utils.cc
index 7e4ad533d6..a590cacbe6 100644
--- a/modules/desktop_capture/linux/wayland/screencast_stream_utils.cc
+++ b/modules/desktop_capture/linux/wayland/screencast_stream_utils.cc
@@ -26,10 +26,7 @@
 #include <tuple>
 #include <vector>
 
-#include "absl/strings/string_view.h"
 #include "modules/desktop_capture/linux/wayland/egl_dmabuf.h"
-#include "rtc_base/string_encode.h"
-#include "rtc_base/string_to_number.h"
 
 #if !PW_CHECK_VERSION(0, 3, 29)
 #define SPA_POD_PROP_FLAG_MANDATORY (1u << 3)
@@ -44,50 +41,6 @@ constexpr uint32_t kSupportedPixelFormats[] = {
     SPA_VIDEO_FORMAT_BGRA, SPA_VIDEO_FORMAT_RGBA, SPA_VIDEO_FORMAT_BGRx,
     SPA_VIDEO_FORMAT_RGBx};
 
-PipeWireVersion PipeWireVersion::Parse(const absl::string_view& version) {
-  std::vector<absl::string_view> parsed_version = split(version, '.');
-
-  if (parsed_version.size() != 3) {
-    return {};
-  }
-
-  std::optional<int> major = StringToNumber<int>(parsed_version.at(0));
-  std::optional<int> minor = StringToNumber<int>(parsed_version.at(1));
-  std::optional<int> micro = StringToNumber<int>(parsed_version.at(2));
-
-  // Return invalid version if we failed to parse it
-  if (!major || !minor || !micro) {
-    return {};
-  }
-
-  return {.major = major.value(),
-          .minor = minor.value(),
-          .micro = micro.value(),
-          .full_version = std::string(version)};
-}
-
-bool PipeWireVersion::operator>=(const PipeWireVersion& other) {
-  if (!major && !minor && !micro) {
-    return false;
-  }
-
-  return std::tie(major, minor, micro) >=
-         std::tie(other.major, other.minor, other.micro);
-}
-
-bool PipeWireVersion::operator<=(const PipeWireVersion& other) {
-  if (!major && !minor && !micro) {
-    return false;
-  }
-
-  return std::tie(major, minor, micro) <=
-         std::tie(other.major, other.minor, other.micro);
-}
-
-absl::string_view PipeWireVersion::ToStringView() const {
-  return full_version;
-}
-
 void BuildBaseFormatParams(spa_pod_builder* builder,
                            uint32_t format,
                            const struct spa_rectangle* resolution,
diff --git a/modules/desktop_capture/linux/wayland/screencast_stream_utils.h b/modules/desktop_capture/linux/wayland/screencast_stream_utils.h
index 27d58d22cb..4e3492e859 100644
--- a/modules/desktop_capture/linux/wayland/screencast_stream_utils.h
+++ b/modules/desktop_capture/linux/wayland/screencast_stream_utils.h
@@ -15,8 +15,6 @@
 
 #include <vector>
 
-#include "absl/strings/string_view.h"
-
 struct spa_pod;
 struct spa_pod_builder;
 struct spa_rectangle;
@@ -26,22 +24,6 @@ namespace webrtc {
 
 class EglDrmDevice;
 
-struct PipeWireVersion {
-  static PipeWireVersion Parse(const absl::string_view& version);
-
-  // Returns whether current version is newer or same as required version
-  bool operator>=(const PipeWireVersion& other);
-  // Returns whether current version is older or same as required version
-  bool operator<=(const PipeWireVersion& other);
-
-  absl::string_view ToStringView() const;
-
-  int major = 0;
-  int minor = 0;
-  int micro = 0;
-  std::string full_version;
-};
-
 // Builds base video format parameters. The format parameter consists of:
 // - SPA_FORMAT_mediaType with SPA_MEDIA_TYPE_video
 // - SPA_FORMAT_mediaSubtype with SPA_MEDIA_SUBTYPE_raw
diff --git a/modules/portal/pipewire_utils.cc b/modules/portal/pipewire_utils.cc
index 91a2ae6951..850726cc58 100644
--- a/modules/portal/pipewire_utils.cc
+++ b/modules/portal/pipewire_utils.cc
@@ -12,7 +12,15 @@
 
 #include <pipewire/pipewire.h>
 
+#include <optional>
+#include <string>
+#include <string_view>
+#include <tuple>
+#include <vector>
+
 #include "rtc_base/sanitizer.h"
+#include "rtc_base/string_encode.h"
+#include "rtc_base/string_to_number.h"
 
 #if defined(WEBRTC_DLOPEN_PIPEWIRE)
 #include "modules/portal/pipewire_stubs.h"
@@ -20,6 +28,45 @@
 
 namespace webrtc {
 
+constexpr PipeWireVersion kReentrantDeinitMinVersion = {.major = 0,
+                                                        .minor = 3,
+                                                        .micro = 49};
+
+PipeWireVersion PipeWireVersion::Parse(const std::string_view& version) {
+  std::vector<std::string_view> parsed_version = split(version, '.');
+
+  if (parsed_version.size() != 3) {
+    return {};
+  }
+
+  std::optional<int> major = StringToNumber<int>(parsed_version.at(0));
+  std::optional<int> minor = StringToNumber<int>(parsed_version.at(1));
+  std::optional<int> micro = StringToNumber<int>(parsed_version.at(2));
+
+  // Return invalid version if we failed to parse it
+  if (!major || !minor || !micro) {
+    return {};
+  }
+
+  return {.major = major.value(),
+          .minor = minor.value(),
+          .micro = micro.value(),
+          .full_version = std::string(version)};
+}
+
+bool PipeWireVersion::operator>=(const PipeWireVersion& other) {
+  if (!major && !minor && !micro) {
+    return false;
+  }
+
+  return std::tie(major, minor, micro) >=
+         std::tie(other.major, other.minor, other.micro);
+}
+
+std::string_view PipeWireVersion::ToStringView() const {
+  return full_version;
+}
+
 RTC_NO_SANITIZE("cfi-icall")
 bool InitializePipeWire() {
 #if defined(WEBRTC_DLOPEN_PIPEWIRE)
@@ -57,7 +104,11 @@ PipeWireInitializer::PipeWireInitializer() {
 
 RTC_NO_SANITIZE("cfi-icall")
 PipeWireInitializer::~PipeWireInitializer() {
-  pw_deinit();
+  PipeWireVersion pw_client_version =
+      PipeWireVersion::Parse(pw_get_library_version());
+  if (pw_client_version >= kReentrantDeinitMinVersion) {
+    pw_deinit();
+  }
 }
 
 }  // namespace webrtc
diff --git a/modules/portal/pipewire_utils.h b/modules/portal/pipewire_utils.h
index c22f296f5c..d0734c19be 100644
--- a/modules/portal/pipewire_utils.h
+++ b/modules/portal/pipewire_utils.h
@@ -17,6 +17,8 @@
 
 #include <cerrno>
 #include <cstdint>
+#include <string>
+#include <string_view>
 
 // static
 struct dma_buf_sync {
@@ -34,6 +36,20 @@ namespace webrtc {
 
 constexpr int kInvalidPipeWireFd = -1;
 
+struct PipeWireVersion {
+  static PipeWireVersion Parse(const std::string_view& version);
+
+  // Returns whether current version is newer or same as required version
+  bool operator>=(const PipeWireVersion& other);
+
+  std::string_view ToStringView() const;
+
+  int major = 0;
+  int minor = 0;
+  int micro = 0;
+  std::string full_version;
+};
+
 // Prepare PipeWire so that it is ready to be used. If it needs to be dlopen'd
 // this will do so. Note that this does not guarantee a PipeWire server is
 // running nor does it establish a connection to one.

From 4a6be3888874f6412cb0b24fcacd5f4b59af5f1a Mon Sep 17 00:00:00 2001
From: Tony Herre <herre@google.com>
Date: Fri, 6 Mar 2026 12:32:08 +0100
Subject: [PATCH 248/472] Check RtcTransport payload isn't too short before
 checking it's an RTP packet

Bug: chromium:488803429
Change-Id: Iaf6204c1f3bce1a8786c00ecfa7faabe78bb9021
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/454280
Auto-Submit: Tony Herre <herre@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47101}
---
 pc/datagram_connection_internal.cc |  4 +++-
 pc/datagram_connection_unittest.cc | 26 ++++++++++++++++++++++++++
 2 files changed, 29 insertions(+), 1 deletion(-)

diff --git a/pc/datagram_connection_internal.cc b/pc/datagram_connection_internal.cc
index be25bbadfc..38c2e43280 100644
--- a/pc/datagram_connection_internal.cc
+++ b/pc/datagram_connection_internal.cc
@@ -58,6 +58,7 @@ using PacketMetadata = DatagramConnection::Observer::PacketMetadata;
 
 const size_t kMaxRtpPacketLen = 2048;
 const size_t kIceUfragLength = 16;
+const size_t kMinPayloadLengthForRtpCheck = 2;
 
 // Helper function to create IceTransportInit
 IceTransportInit CreateIceTransportInit(const Environment& env,
@@ -291,7 +292,8 @@ void DatagramConnectionInternal::SendSinglePacket(
     return;
   }
 
-  if (IsRtpOrRtcpPacket(packet.payload[0])) {
+  if (packet.payload.size() >= kMinPayloadLengthForRtpCheck &&
+      IsRtpOrRtcpPacket(packet.payload[0])) {
     // Copy the payload into a buffer with some extra capacity to allow space
     // for the SRTP encryption tag to be added.
     CopyOnWriteBuffer buffer(packet.payload.data(), packet.payload.size(),
diff --git a/pc/datagram_connection_unittest.cc b/pc/datagram_connection_unittest.cc
index fae8c8b39d..a352816b38 100644
--- a/pc/datagram_connection_unittest.cc
+++ b/pc/datagram_connection_unittest.cc
@@ -517,5 +517,31 @@ TEST_F(DatagramConnectionTest, DirectDtlsPacketsAreReceived) {
   EXPECT_TRUE(callbacks.send_outcome);
 }
 
+TEST_F(DatagramConnectionTest, SingleBytePacketsAreSent) {
+  CreateConnections(WireProtocol::kDtls);
+  Connect();
+  ASSERT_TRUE(
+      WaitUntil([&]() { return conn1_->Writable() && conn2_->Writable(); }));
+
+  std::vector<uint8_t> data = {1};
+  bool callback_called = false;
+  EXPECT_CALL(*observer1_ptr_, OnSendOutcome(_))
+      .WillOnce([&](const SendOutcome& outcome) {
+        EXPECT_EQ(outcome.id, 1u);
+        EXPECT_EQ(outcome.status, SendOutcome::Status::kSuccess);
+        EXPECT_NE(outcome.send_time, Timestamp::MinusInfinity());
+        callback_called = true;
+        loop_.Quit();
+      });
+  std::vector<PacketSendParameters> packets = {
+      PacketSendParameters{.id = 1, .payload = data}};
+  conn1_->SendPackets(packets);
+  // For direct DTLS, the sent packet should be larger than the data due to
+  // DTLS overhead.
+  EXPECT_GT(ice1_->last_sent_packet().size(), data.size());
+  loop_.Run();
+  EXPECT_TRUE(callback_called);
+}
+
 }  // namespace
 }  // namespace webrtc

From a18009bcbbaafe4532d425fdd1c223c86f21550a Mon Sep 17 00:00:00 2001
From: Fredrik Solenberg <solenberg@google.com>
Date: Mon, 9 Mar 2026 13:43:30 +0000
Subject: [PATCH 249/472] Removed obsolete suppression
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

The referenced test no longer exists.

Bug: webrtc:42232319
Change-Id: I74ff70955f2220d776b6fa1eadd045d897d22987
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/455140
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47102}
---
 tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc | 6 ------
 1 file changed, 6 deletions(-)

diff --git a/tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc b/tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc
index a3a4bdbdd0..1725255e41 100644
--- a/tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc
+++ b/tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc
@@ -44,12 +44,6 @@ char kTSanDefaultSuppressions[] =
     // https://code.google.com/p/libyuv/issues/detail?id=508
     "race:InitCpuFlags\n"
 
-    // Test-only race due to PeerConnection::session() being virtual for
-    // testing. The stats collector may call session() before or after the
-    // destructor begins executing, which modifies the vtable.
-    "race:*RTCStatsIntegrationTest_GetsStatsWhileDestroyingPeerConnections_"
-    "Test::TestBody\n"
-
     // http://crbug.com/244856
     "race:libpulsecommon*.so\n"
 

From 86c4b5bd084919a4b97ca6b13c930385d01bcf0c Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Mon, 9 Mar 2026 15:40:14 +0100
Subject: [PATCH 250/472] Replace ArrayView with std::span in media/

Search&Replace MakeArrayView and ArrayView with std::span
Search&Replace include "api/array_view.h" with include <span>
Remove build dependencies on array_view target

In rtp_utils_unittest remove unnecessary span constants touched by rename above.

Bug: webrtc:439801349
Change-Id: I7cf2dd49622dd457455474aedc9bede3262d5df5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/455281
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47103}
---
 media/BUILD.gn                               | 12 ---------
 media/base/fake_network_interface.h          |  4 +--
 media/base/media_channel_impl.cc             |  6 ++---
 media/base/media_channel_impl.h              |  7 ++---
 media/base/media_engine.cc                   |  8 +++---
 media/base/media_engine.h                    |  8 +++---
 media/base/rtp_utils.cc                      | 17 ++++++------
 media/base/rtp_utils.h                       | 10 +++----
 media/base/rtp_utils_unittest.cc             | 15 +++--------
 media/base/stream_params.cc                  | 10 +++----
 media/base/stream_params_unittest.cc         |  4 +--
 media/base/turn_utils.cc                     |  6 ++---
 media/base/video_common.cc                   |  1 -
 media/engine/fake_webrtc_call.cc             |  8 +++---
 media/engine/fake_webrtc_call.h              |  8 +++---
 media/engine/simulcast_encoder_adapter.cc    |  6 ++---
 media/engine/webrtc_media_engine.cc          |  4 +--
 media/engine/webrtc_video_engine.cc          |  4 +--
 media/engine/webrtc_voice_engine_unittest.cc | 28 ++++++++++----------
 media/sctp/dcsctp_transport.cc               | 10 +++----
 media/sctp/dcsctp_transport.h                | 10 +++----
 21 files changed, 83 insertions(+), 103 deletions(-)

diff --git a/media/BUILD.gn b/media/BUILD.gn
index fbe70c99bc..9b026bf244 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -69,7 +69,6 @@ rtc_library("rtc_media_base") {
     ":video_broadcaster",
     ":video_common",
     ":video_source_base",
-    "../api:array_view",
     "../api:audio_options_api",
     "../api:call_api",
     "../api:field_trials_view",
@@ -218,7 +217,6 @@ rtc_library("video_common") {
     "base/video_common.h",
   ]
   deps = [
-    "../api:array_view",
     "../rtc_base:stringutils",
     "../rtc_base:timeutils",
     "../rtc_base/system:rtc_export",
@@ -238,7 +236,6 @@ rtc_library("media_engine") {
     ":rtc_media_config",
     ":stream_params",
     ":video_common",
-    "../api:array_view",
     "../api:audio_options_api",
     "../api:field_trials_view",
     "../api:rtc_error",
@@ -273,7 +270,6 @@ rtc_library("media_channel_impl") {
     ":media_channel",
     ":rtp_utils",
     ":stream_params",
-    "../api:array_view",
     "../api:audio_options_api",
     "../api:call_api",
     "../api:frame_transformer_interface",
@@ -416,7 +412,6 @@ rtc_library("rtp_utils") {
   ]
   deps = [
     ":turn_utils",
-    "../api:array_view",
     "../modules/rtp_rtcp:rtp_rtcp_format",
     "../rtc_base:async_packet_socket",
     "../rtc_base:byte_order",
@@ -434,7 +429,6 @@ rtc_library("stream_params") {
   ]
   deps = [
     ":rid_description",
-    "../api:array_view",
     "../rtc_base:checks",
     "../rtc_base:stringutils",
     "../rtc_base:unique_id_generator",
@@ -457,7 +451,6 @@ rtc_library("turn_utils") {
     "base/turn_utils.h",
   ]
   deps = [
-    "../api:array_view",
     "../api/transport:stun_types",
     "../rtc_base:byte_order",
     "../rtc_base/system:rtc_export",
@@ -486,7 +479,6 @@ rtc_library("rtc_simulcast_encoder_adapter") {
   deps = [
     ":rtc_sdp_video_format_utils",
     ":video_common",
-    "../api:array_view",
     "../api:fec_controller_api",
     "../api:field_trials_view",
     "../api:scoped_refptr",
@@ -594,7 +586,6 @@ rtc_library("rtc_audio_video") {
     ":rtc_media_config",
     ":rtp_utils",
     ":stream_params",
-    "../api:array_view",
     "../api:audio_options_api",
     "../api:call_api",
     "../api:field_trials_view",
@@ -731,7 +722,6 @@ if (rtc_build_dcsctp) {
     deps = [
       ":media_channel",
       ":rtc_data_sctp_transport_internal",
-      "../api:array_view",
       "../api:data_channel_interface",
       "../api:dtls_transport_interface",
       "../api:field_trials_view",
@@ -822,7 +812,6 @@ if (rtc_include_tests) {
       ":rtp_utils",
       ":stream_params",
       ":video_common",
-      "../api:array_view",
       "../api:audio_options_api",
       "../api:call_api",
       "../api:fec_controller_api",
@@ -966,7 +955,6 @@ if (rtc_include_tests) {
         ":stream_params",
         ":turn_utils",
         ":video_common",
-        "../api:array_view",
         "../api:audio_options_api",
         "../api:call_api",
         "../api:create_simulcast_test_fixture_api",
diff --git a/media/base/fake_network_interface.h b/media/base/fake_network_interface.h
index fa758aa4af..4573c43ca8 100644
--- a/media/base/fake_network_interface.h
+++ b/media/base/fake_network_interface.h
@@ -15,10 +15,10 @@
 #include <cstdint>
 #include <map>
 #include <set>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/task_queue/pending_task_safety_flag.h"
 #include "api/task_queue/task_queue_base.h"
@@ -192,7 +192,7 @@ class FakeNetworkInterface : public MediaChannelNetworkInterface {
   void SetRtpSsrc(uint32_t ssrc, CopyOnWriteBuffer& buffer) {
     RTC_CHECK_GE(buffer.size(), 12);
     SetBE32(
-        ArrayView<uint8_t>(buffer.MutableData(), buffer.size()).subspan(8, 4),
+        std::span<uint8_t>(buffer.MutableData(), buffer.size()).subspan(8, 4),
         ssrc);
   }
 
diff --git a/media/base/media_channel_impl.cc b/media/base/media_channel_impl.cc
index d8f3b41988..cb74a2710e 100644
--- a/media/base/media_channel_impl.cc
+++ b/media/base/media_channel_impl.cc
@@ -12,11 +12,11 @@
 
 #include <cstdint>
 #include <map>
+#include <span>
 #include <string>
 #include <utility>
 
 #include "absl/functional/any_invocable.h"
-#include "api/array_view.h"
 #include "api/audio_options.h"
 #include "api/call/transport.h"
 #include "api/media_stream_interface.h"
@@ -190,7 +190,7 @@ MediaChannelUtil::TransportForMediaChannels::TranslatePacketOptions(
 }
 
 bool MediaChannelUtil::TransportForMediaChannels::SendRtcp(
-    ArrayView<const uint8_t> packet,
+    std::span<const uint8_t> packet,
     const PacketOptions& options) {
   auto send = [this, packet = CopyOnWriteBuffer(packet, kMaxRtpPacketLen),
                options]() mutable {
@@ -206,7 +206,7 @@ bool MediaChannelUtil::TransportForMediaChannels::SendRtcp(
 }
 
 bool MediaChannelUtil::TransportForMediaChannels::SendRtp(
-    ArrayView<const uint8_t> packet,
+    std::span<const uint8_t> packet,
     const PacketOptions& options) {
   auto send = [this, packet = CopyOnWriteBuffer(packet, kMaxRtpPacketLen),
                options]() mutable {
diff --git a/media/base/media_channel_impl.h b/media/base/media_channel_impl.h
index 958018beb8..9826cdf25e 100644
--- a/media/base/media_channel_impl.h
+++ b/media/base/media_channel_impl.h
@@ -14,7 +14,8 @@
 #include <stddef.h>
 #include <stdint.h>
 
-#include "api/array_view.h"
+#include <span>
+
 #include "api/call/transport.h"
 #include "api/scoped_refptr.h"
 #include "api/sequence_checker.h"
@@ -89,9 +90,9 @@ class MediaChannelUtil {
     ~TransportForMediaChannels() override;
 
     // Implementation of Transport
-    bool SendRtp(ArrayView<const uint8_t> packet,
+    bool SendRtp(std::span<const uint8_t> packet,
                  const PacketOptions& options) override;
-    bool SendRtcp(ArrayView<const uint8_t> packet,
+    bool SendRtcp(std::span<const uint8_t> packet,
                   const PacketOptions& options) override;
 
     // Not implementation of Transport
diff --git a/media/base/media_engine.cc b/media/base/media_engine.cc
index ba0a91ef45..52da64db52 100644
--- a/media/base/media_engine.cc
+++ b/media/base/media_engine.cc
@@ -14,12 +14,12 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/algorithm/container.h"
-#include "api/array_view.h"
 #include "api/field_trials_view.h"
 #include "api/rtc_error.h"
 #include "api/rtp_headers.h"
@@ -90,7 +90,7 @@ std::vector<RtpExtension> GetDefaultEnabledRtpHeaderExtensions(
 }
 
 RTCError CheckScalabilityModeValues(const RtpParameters& rtp_parameters,
-                                    ArrayView<const Codec> send_codecs,
+                                    std::span<const Codec> send_codecs,
                                     std::optional<Codec> send_codec) {
   if (send_codecs.empty()) {
     // This is an audio sender or an extra check in the stack where the codec
@@ -161,7 +161,7 @@ RTCError CheckScalabilityModeValues(const RtpParameters& rtp_parameters,
 }
 
 RTCError CheckRtpParametersValues(const RtpParameters& rtp_parameters,
-                                  ArrayView<const Codec> send_codecs,
+                                  std::span<const Codec> send_codecs,
                                   std::optional<Codec> send_codec,
                                   const FieldTrialsView& field_trials) {
   bool has_scale_resolution_down_to = false;
@@ -280,7 +280,7 @@ RTCError CheckRtpParametersInvalidModificationAndValues(
 RTCError CheckRtpParametersInvalidModificationAndValues(
     const RtpParameters& old_rtp_parameters,
     const RtpParameters& rtp_parameters,
-    ArrayView<const Codec> send_codecs,
+    std::span<const Codec> send_codecs,
     std::optional<Codec> send_codec,
     const FieldTrialsView& field_trials) {
   if (rtp_parameters.encodings.size() != old_rtp_parameters.encodings.size()) {
diff --git a/media/base/media_engine.h b/media/base/media_engine.h
index fd1de0699e..0ac3277d50 100644
--- a/media/base/media_engine.h
+++ b/media/base/media_engine.h
@@ -14,9 +14,9 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/audio_device.h"
 #include "api/audio_codecs/audio_decoder_factory.h"
 #include "api/audio_codecs/audio_encoder_factory.h"
@@ -43,13 +43,13 @@ class Call;
 // Checks that the scalability_mode value of each encoding is supported by at
 // least one video codec of the list. If the list is empty, no check is done.
 RTCError CheckScalabilityModeValues(const RtpParameters& new_parameters,
-                                    ArrayView<const Codec> send_codecs,
+                                    std::span<const Codec> send_codecs,
                                     std::optional<Codec> send_codec);
 
 // Checks the parameters have valid and supported values, and checks parameters
 // with CheckScalabilityModeValues().
 RTCError CheckRtpParametersValues(const RtpParameters& new_parameters,
-                                  ArrayView<const Codec> send_codecs,
+                                  std::span<const Codec> send_codecs,
                                   std::optional<Codec> send_codec,
                                   const FieldTrialsView& field_trials);
 
@@ -58,7 +58,7 @@ RTCError CheckRtpParametersValues(const RtpParameters& new_parameters,
 RTCError CheckRtpParametersInvalidModificationAndValues(
     const RtpParameters& old_parameters,
     const RtpParameters& new_parameters,
-    ArrayView<const Codec> send_codecs,
+    std::span<const Codec> send_codecs,
     std::optional<Codec> send_codec,
     const FieldTrialsView& field_trials);
 
diff --git a/media/base/rtp_utils.cc b/media/base/rtp_utils.cc
index a6448d762c..3a2002eafe 100644
--- a/media/base/rtp_utils.cc
+++ b/media/base/rtp_utils.cc
@@ -12,12 +12,13 @@
 
 #include <cstdint>
 #include <cstring>
+#include <span>
 #include <vector>
 
 // PacketTimeUpdateParams is defined in asyncpacketsocket.h.
 // TODO(sergeyu): Find more appropriate place for PacketTimeUpdateParams.
+
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "media/base/turn_utils.h"
 #include "modules/rtp_rtcp/source/rtp_util.h"
 #include "rtc_base/async_packet_socket.h"
@@ -73,7 +74,7 @@ void UpdateAbsSendTimeExtensionValue(uint8_t* extension_data,
 
 // Assumes `length` is actual packet length + tag length. Updates HMAC at end of
 // the RTP packet.
-void UpdateRtpAuthTag(ArrayView<uint8_t> rtp,
+void UpdateRtpAuthTag(std::span<uint8_t> rtp,
                       const PacketTimeUpdateParams& packet_time_params) {
   // If there is no key, return.
   if (packet_time_params.srtp_auth_key.empty()) {
@@ -148,7 +149,7 @@ bool GetRtcpSsrc(const void* data, size_t len, uint32_t* value) {
   if (pl_type == kRtcpTypeSDES)
     return false;
   *value = GetBE32(
-      ArrayView<const uint8_t>(static_cast<const uint8_t*>(data) + 4, 4));
+      std::span<const uint8_t>(static_cast<const uint8_t*>(data) + 4, 4));
   return true;
 }
 
@@ -176,7 +177,7 @@ absl::string_view RtpPacketTypeToString(RtpPacketType packet_type) {
   RTC_CHECK_NOTREACHED();
 }
 
-RtpPacketType InferRtpPacketType(ArrayView<const uint8_t> packet) {
+RtpPacketType InferRtpPacketType(std::span<const uint8_t> packet) {
   if (IsRtcpPacket(packet)) {
     return RtpPacketType::kRtcp;
   }
@@ -186,7 +187,7 @@ RtpPacketType InferRtpPacketType(ArrayView<const uint8_t> packet) {
   return RtpPacketType::kUnknown;
 }
 
-bool ValidateRtpHeader(ArrayView<const uint8_t> rtp, size_t* header_length) {
+bool ValidateRtpHeader(std::span<const uint8_t> rtp, size_t* header_length) {
   size_t length = rtp.size();
   if (header_length) {
     *header_length = 0;
@@ -238,7 +239,7 @@ bool ValidateRtpHeader(ArrayView<const uint8_t> rtp, size_t* header_length) {
 
 // ValidateRtpHeader() must be called before this method to make sure, we have
 // a sane rtp packet.
-bool UpdateRtpAbsSendTimeExtension(ArrayView<uint8_t> packet,
+bool UpdateRtpAbsSendTimeExtension(std::span<uint8_t> packet,
                                    int extension_id,
                                    uint64_t time_us) {
   //  0                   1                   2                   3
@@ -265,7 +266,7 @@ bool UpdateRtpAbsSendTimeExtension(ArrayView<uint8_t> packet,
   uint8_t* rtp = packet.data();
   rtp += header_length_without_extension;
 
-  ArrayView<const uint8_t> extension_header =
+  std::span<const uint8_t> extension_header =
       packet.subspan(header_length_without_extension, kRtpExtensionHeaderLen);
 
   // Getting extension profile ID and length.
@@ -356,7 +357,7 @@ bool UpdateRtpAbsSendTimeExtension(ArrayView<uint8_t> packet,
   return found;
 }
 
-bool ApplyPacketOptions(ArrayView<uint8_t> data,
+bool ApplyPacketOptions(std::span<uint8_t> data,
                         const PacketTimeUpdateParams& packet_time_params,
                         uint64_t time_us) {
   RTC_DCHECK(!data.empty());
diff --git a/media/base/rtp_utils.h b/media/base/rtp_utils.h
index b9fae21a4a..a8f1a128b9 100644
--- a/media/base/rtp_utils.h
+++ b/media/base/rtp_utils.h
@@ -13,9 +13,9 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "rtc_base/async_packet_socket.h"
 #include "rtc_base/system/rtc_export.h"
 
@@ -45,7 +45,7 @@ bool GetRtcpType(const void* data, size_t len, int* value);
 bool GetRtcpSsrc(const void* data, size_t len, uint32_t* value);
 
 // Checks the packet header to determine if it can be an RTP or RTCP packet.
-RtpPacketType InferRtpPacketType(ArrayView<const uint8_t> packet);
+RtpPacketType InferRtpPacketType(std::span<const uint8_t> packet);
 // True if |payload type| is 0-127.
 bool IsValidRtpPayloadType(int payload_type);
 
@@ -56,18 +56,18 @@ bool IsValidRtpPacketSize(RtpPacketType packet_type, size_t size);
 absl::string_view RtpPacketTypeToString(RtpPacketType packet_type);
 
 // Verifies that a packet has a valid RTP header.
-bool RTC_EXPORT ValidateRtpHeader(ArrayView<const uint8_t> rtp,
+bool RTC_EXPORT ValidateRtpHeader(std::span<const uint8_t> rtp,
                                   size_t* header_length);
 
 // Helper method which updates the absolute send time extension if present.
-bool UpdateRtpAbsSendTimeExtension(ArrayView<uint8_t> rtp,
+bool UpdateRtpAbsSendTimeExtension(std::span<uint8_t> rtp,
                                    int extension_id,
                                    uint64_t time_us);
 
 // Applies specified `options` to the packet. It updates the absolute send time
 // extension header if it is present present then updates HMAC.
 bool RTC_EXPORT
-ApplyPacketOptions(ArrayView<uint8_t> data,
+ApplyPacketOptions(std::span<uint8_t> data,
                    const PacketTimeUpdateParams& packet_time_params,
                    uint64_t time_us);
 
diff --git a/media/base/rtp_utils_unittest.cc b/media/base/rtp_utils_unittest.cc
index af3263e7e2..269e4d6c9a 100644
--- a/media/base/rtp_utils_unittest.cc
+++ b/media/base/rtp_utils_unittest.cc
@@ -14,7 +14,6 @@
 #include <cstring>
 #include <vector>
 
-#include "api/array_view.h"
 #include "media/base/fake_rtp.h"
 #include "rtc_base/async_packet_socket.h"
 #include "test/gtest.h"
@@ -72,13 +71,6 @@ static const int kAstIndexInOneByteRtpMsg = 21;
 // and in message `kRtpMsgWithTwoByteAbsSendTimeExtension`.
 static const int kAstIndexInTwoByteRtpMsg = 21;
 
-static const ArrayView<const uint8_t> kPcmuFrameArrayView =
-    MakeArrayView(kPcmuFrame, sizeof(kPcmuFrame));
-static const ArrayView<const uint8_t> kRtcpReportArrayView =
-    MakeArrayView(kRtcpReport, sizeof(kRtcpReport));
-static const ArrayView<const uint8_t> kInvalidPacketArrayView =
-    MakeArrayView(kInvalidPacket, sizeof(kInvalidPacket));
-
 TEST(RtpUtilsTest, GetRtcp) {
   int pt;
   EXPECT_TRUE(GetRtcpType(kRtcpReport, sizeof(kRtcpReport), &pt));
@@ -280,10 +272,9 @@ TEST(RtpUtilsTest, ApplyPacketOptionsWithAuthParamsAndAbsSendTime) {
 }
 
 TEST(RtpUtilsTest, InferRtpPacketType) {
-  EXPECT_EQ(RtpPacketType::kRtp, InferRtpPacketType(kPcmuFrameArrayView));
-  EXPECT_EQ(RtpPacketType::kRtcp, InferRtpPacketType(kRtcpReportArrayView));
-  EXPECT_EQ(RtpPacketType::kUnknown,
-            InferRtpPacketType(kInvalidPacketArrayView));
+  EXPECT_EQ(RtpPacketType::kRtp, InferRtpPacketType(kPcmuFrame));
+  EXPECT_EQ(RtpPacketType::kRtcp, InferRtpPacketType(kRtcpReport));
+  EXPECT_EQ(RtpPacketType::kUnknown, InferRtpPacketType(kInvalidPacket));
 }
 
 }  // namespace webrtc
diff --git a/media/base/stream_params.cc b/media/base/stream_params.cc
index 4cae898ef7..d330d9c60f 100644
--- a/media/base/stream_params.cc
+++ b/media/base/stream_params.cc
@@ -11,12 +11,12 @@
 #include "media/base/stream_params.h"
 
 #include <cstdint>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/algorithm/container.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "media/base/rid_description.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/strings/string_builder.h"
@@ -25,7 +25,7 @@
 namespace webrtc {
 namespace {
 
-void AppendSsrcs(ArrayView<const uint32_t> ssrcs, SimpleStringBuilder* sb) {
+void AppendSsrcs(std::span<const uint32_t> ssrcs, SimpleStringBuilder* sb) {
   *sb << "ssrcs:[";
   const char* delimiter = "";
   for (uint32_t ssrc : ssrcs) {
@@ -35,7 +35,7 @@ void AppendSsrcs(ArrayView<const uint32_t> ssrcs, SimpleStringBuilder* sb) {
   *sb << "]";
 }
 
-void AppendSsrcGroups(ArrayView<const SsrcGroup> ssrc_groups,
+void AppendSsrcGroups(std::span<const SsrcGroup> ssrc_groups,
                       SimpleStringBuilder* sb) {
   *sb << "ssrc_groups:";
   const char* delimiter = "";
@@ -45,7 +45,7 @@ void AppendSsrcGroups(ArrayView<const SsrcGroup> ssrc_groups,
   }
 }
 
-void AppendStreamIds(ArrayView<const std::string> stream_ids,
+void AppendStreamIds(std::span<const std::string> stream_ids,
                      SimpleStringBuilder* sb) {
   *sb << "stream_ids:";
   const char* delimiter = "";
@@ -55,7 +55,7 @@ void AppendStreamIds(ArrayView<const std::string> stream_ids,
   }
 }
 
-void AppendRids(ArrayView<const RidDescription> rids, SimpleStringBuilder* sb) {
+void AppendRids(std::span<const RidDescription> rids, SimpleStringBuilder* sb) {
   *sb << "rids:[";
   const char* delimiter = "";
   for (const RidDescription& rid : rids) {
diff --git a/media/base/stream_params_unittest.cc b/media/base/stream_params_unittest.cc
index 49a8926c86..332022a5f8 100644
--- a/media/base/stream_params_unittest.cc
+++ b/media/base/stream_params_unittest.cc
@@ -12,10 +12,10 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "media/base/test_utils.h"
 #include "rtc_base/unique_id_generator.h"
 #include "test/gmock.h"
@@ -29,7 +29,7 @@ static const uint32_t kSsrcs2[] = {1, 2};
 
 static webrtc::StreamParams CreateStreamParamsWithSsrcGroup(
     const std::string& semantics,
-    webrtc::ArrayView<const uint32_t> ssrcs_in) {
+    std::span<const uint32_t> ssrcs_in) {
   webrtc::StreamParams stream;
   std::vector<uint32_t> ssrcs(ssrcs_in.begin(), ssrcs_in.end());
   webrtc::SsrcGroup sg(semantics, ssrcs);
diff --git a/media/base/turn_utils.cc b/media/base/turn_utils.cc
index 77f0f18ae5..b3e14bd1a1 100644
--- a/media/base/turn_utils.cc
+++ b/media/base/turn_utils.cc
@@ -12,8 +12,8 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/transport/stun.h"
 #include "rtc_base/byte_order.h"
 
@@ -32,7 +32,7 @@ bool IsTurnSendIndicationPacket(const uint8_t* data, size_t length) {
     return false;
   }
 
-  uint16_t type = GetBE16(ArrayView<const uint8_t>(data, 2));
+  uint16_t type = GetBE16(std::span<const uint8_t>(data, 2));
   return (type == TURN_SEND_INDICATION);
 }
 
@@ -42,7 +42,7 @@ bool UnwrapTurnPacket(const uint8_t* packet,
                       size_t packet_size,
                       size_t* content_position,
                       size_t* content_size) {
-  ArrayView<const uint8_t> data_view(packet, packet_size);
+  std::span<const uint8_t> data_view(packet, packet_size);
   if (IsTurnChannelData(packet, packet_size)) {
     // Turn Channel Message header format.
     //   0                   1                   2                   3
diff --git a/media/base/video_common.cc b/media/base/video_common.cc
index 053a250cd8..38b55624bc 100644
--- a/media/base/video_common.cc
+++ b/media/base/video_common.cc
@@ -13,7 +13,6 @@
 #include <cstdint>
 #include <string>
 
-#include "api/array_view.h"
 #include "rtc_base/strings/string_builder.h"
 
 namespace webrtc {
diff --git a/media/engine/fake_webrtc_call.cc b/media/engine/fake_webrtc_call.cc
index 20ed9865a8..a9c1a2cbe3 100644
--- a/media/engine/fake_webrtc_call.cc
+++ b/media/engine/fake_webrtc_call.cc
@@ -13,6 +13,7 @@
 #include <cstdint>
 #include <map>
 #include <memory>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
@@ -20,7 +21,6 @@
 #include "absl/algorithm/container.h"
 #include "absl/strings/string_view.h"
 #include "api/adaptation/resource.h"
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/call/audio_sink.h"
 #include "api/crypto/frame_decryptor_interface.h"
@@ -124,11 +124,11 @@ void FakeAudioReceiveStream::SetStats(
 }
 
 bool FakeAudioReceiveStream::VerifyLastPacket(
-    ArrayView<const uint8_t> data) const {
+    std::span<const uint8_t> data) const {
   return last_packet_ == Buffer(data.data(), data.size());
 }
 
-bool FakeAudioReceiveStream::DeliverRtp(ArrayView<const uint8_t> packet,
+bool FakeAudioReceiveStream::DeliverRtp(std::span<const uint8_t> packet,
                                         int64_t /* packet_time_us */) {
   ++received_packets_;
   last_packet_.SetData(packet);
@@ -295,7 +295,7 @@ VideoSendStream::Stats FakeVideoSendStream::GetStats() {
   return stats_;
 }
 
-void FakeVideoSendStream::SetCsrcs(ArrayView<const uint32_t> csrcs) {}
+void FakeVideoSendStream::SetCsrcs(std::span<const uint32_t> csrcs) {}
 
 void FakeVideoSendStream::ReconfigureVideoEncoder(VideoEncoderConfig config) {
   ReconfigureVideoEncoder(std::move(config), nullptr);
diff --git a/media/engine/fake_webrtc_call.h b/media/engine/fake_webrtc_call.h
index e4dff9a22b..8bd0f0a6bf 100644
--- a/media/engine/fake_webrtc_call.h
+++ b/media/engine/fake_webrtc_call.h
@@ -25,13 +25,13 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
 #include "api/adaptation/resource.h"
-#include "api/array_view.h"
 #include "api/audio/audio_frame.h"
 #include "api/audio/audio_mixer.h"
 #include "api/audio_codecs/audio_format.h"
@@ -125,10 +125,10 @@ class FakeAudioReceiveStream final : public AudioReceiveStreamInterface {
   const AudioReceiveStreamInterface::Config& GetConfig() const;
   void SetStats(const AudioReceiveStreamInterface::Stats& stats);
   int received_packets() const { return received_packets_; }
-  bool VerifyLastPacket(ArrayView<const uint8_t> data) const;
+  bool VerifyLastPacket(std::span<const uint8_t> data) const;
   const AudioSinkInterface* sink() const { return sink_; }
   float gain() const { return gain_; }
-  bool DeliverRtp(ArrayView<const uint8_t> packet, int64_t packet_time_us);
+  bool DeliverRtp(std::span<const uint8_t> packet, int64_t packet_time_us);
   bool started() const { return started_; }
   int base_mininum_playout_delay_ms() const {
     return base_mininum_playout_delay_ms_;
@@ -206,7 +206,7 @@ class FakeVideoSendStream final : public VideoSendStream,
   int GetLastHeight() const;
   int64_t GetLastTimestamp() const;
   void SetStats(const VideoSendStream::Stats& stats) override;
-  void SetCsrcs(ArrayView<const uint32_t> csrcs) override;
+  void SetCsrcs(std::span<const uint32_t> csrcs) override;
   int num_encoder_reconfigurations() const {
     return num_encoder_reconfigurations_;
   }
diff --git a/media/engine/simulcast_encoder_adapter.cc b/media/engine/simulcast_encoder_adapter.cc
index 3e10c92aec..c25d0a5394 100644
--- a/media/engine/simulcast_encoder_adapter.cc
+++ b/media/engine/simulcast_encoder_adapter.cc
@@ -19,6 +19,7 @@
 #include <memory>
 #include <numeric>
 #include <optional>
+#include <span>
 #include <string>
 #include <tuple>
 #include <utility>
@@ -26,7 +27,6 @@
 
 #include "absl/algorithm/container.h"
 #include "absl/base/nullability.h"
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/fec_controller_override.h"
 #include "api/field_trials_view.h"
@@ -131,7 +131,7 @@ bool StreamQualityCompare(const SimulcastStream& a, const SimulcastStream& b) {
 }
 
 void GetLowestAndHighestQualityStreamIndixes(
-    ArrayView<const SimulcastStream> streams,
+    std::span<const SimulcastStream> streams,
     int* lowest_quality_stream_idx,
     int* highest_quality_stream_idx) {
   const auto lowest_highest_quality_streams =
@@ -385,7 +385,7 @@ int SimulcastEncoderAdapter::InitEncode(
   int highest_quality_stream_idx = 0;
   if (!is_legacy_singlecast) {
     GetLowestAndHighestQualityStreamIndixes(
-        ArrayView<SimulcastStream>(codec_.simulcastStream,
+        std::span<SimulcastStream>(codec_.simulcastStream,
                                    total_streams_count_),
         &lowest_quality_stream_idx, &highest_quality_stream_idx);
   }
diff --git a/media/engine/webrtc_media_engine.cc b/media/engine/webrtc_media_engine.cc
index 96bacd8621..4f586d5efc 100644
--- a/media/engine/webrtc_media_engine.cc
+++ b/media/engine/webrtc_media_engine.cc
@@ -11,12 +11,12 @@
 #include "media/engine/webrtc_media_engine.h"
 
 #include <algorithm>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/algorithm/container.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/field_trials_view.h"
 #include "api/rtp_parameters.h"
 #include "api/transport/bitrate_settings.h"
@@ -30,7 +30,7 @@ namespace {
 // Remove mutually exclusive extensions with lower priority.
 void DiscardRedundantExtensions(
     std::vector<RtpExtension>* extensions,
-    ArrayView<const char* const> extensions_decreasing_prio) {
+    std::span<const char* const> extensions_decreasing_prio) {
   RTC_DCHECK(extensions);
   bool found = false;
   for (const char* uri : extensions_decreasing_prio) {
diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc
index fc04a6c2b6..bb776670c9 100644
--- a/media/engine/webrtc_video_engine.cc
+++ b/media/engine/webrtc_video_engine.cc
@@ -19,6 +19,7 @@
 #include <memory>
 #include <optional>
 #include <set>
+#include <span>
 #include <string>
 #include <unordered_set>
 #include <utility>
@@ -29,7 +30,6 @@
 #include "absl/functional/bind_front.h"
 #include "absl/strings/match.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/crypto/crypto_options.h"
 #include "api/crypto/frame_decryptor_interface.h"
 #include "api/environment/environment.h"
@@ -734,7 +734,7 @@ std::string CodecSettingsVectorToString(
 }
 
 void ExtractCodecInformation(
-    ArrayView<const VideoCodecSettings> recv_codecs,
+    std::span<const VideoCodecSettings> recv_codecs,
     std::map<int, int>& rtx_associated_payload_types,
     std::set<int>& raw_payload_types,
     std::vector<VideoReceiveStreamInterface::Decoder>& decoders) {
diff --git a/media/engine/webrtc_voice_engine_unittest.cc b/media/engine/webrtc_voice_engine_unittest.cc
index 0d66c96a08..8195403501 100644
--- a/media/engine/webrtc_voice_engine_unittest.cc
+++ b/media/engine/webrtc_voice_engine_unittest.cc
@@ -17,13 +17,13 @@
 #include <memory>
 #include <optional>
 #include <set>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/match.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio/audio_processing.h"
 #include "api/audio/builtin_audio_processing_builder.h"
 #include "api/audio_codecs/audio_format.h"
@@ -336,7 +336,7 @@ class WebRtcVoiceEngineTestFake : public ::testing::TestWithParam<bool> {
     EXPECT_FALSE(call_.GetAudioSendStream(kSsrcX));
   }
 
-  void DeliverPacket(ArrayView<const uint8_t> data) {
+  void DeliverPacket(std::span<const uint8_t> data) {
     RtpPacketReceived packet;
     packet.Parse(data);
     receive_channel_->OnPacketReceived(packet);
@@ -2779,7 +2779,7 @@ TEST_P(WebRtcVoiceEngineTestFake, RecvWithMultipleStreams) {
   uint8_t packets[4][sizeof(kPcmuFrame)];
   for (size_t i = 0; i < std::size(packets); ++i) {
     memcpy(packets[i], kPcmuFrame, sizeof(kPcmuFrame));
-    SetBE32(ArrayView<uint8_t>(packets[i] + 8, 4), static_cast<uint32_t>(i));
+    SetBE32(std::span<uint8_t>(packets[i] + 8, 4), static_cast<uint32_t>(i));
   }
 
   const FakeAudioReceiveStream& s1 = GetRecvStream(ssrc1);
@@ -2868,9 +2868,9 @@ TEST_P(WebRtcVoiceEngineTestFake,
   // Deliver a couple packets with unsignaled SSRCs.
   uint8_t packet[sizeof(kPcmuFrame)];
   memcpy(packet, kPcmuFrame, sizeof(kPcmuFrame));
-  SetBE32(ArrayView<uint8_t>(&packet[8], 4), 0x1234);
+  SetBE32(std::span<uint8_t>(&packet[8], 4), 0x1234);
   DeliverPacket(packet);
-  SetBE32(ArrayView<uint8_t>(&packet[8], 4), 0x5678);
+  SetBE32(std::span<uint8_t>(&packet[8], 4), 0x5678);
   DeliverPacket(packet);
 
   // Verify that the receive streams were created.
@@ -2892,7 +2892,7 @@ TEST_P(WebRtcVoiceEngineTestFake, RecvMultipleUnsignaled) {
 
   // Note that SSRC = 0 is not supported.
   for (uint32_t ssrc = 1; ssrc < (1 + kMaxUnsignaledRecvStreams); ++ssrc) {
-    SetBE32(ArrayView<uint8_t>(&packet[8], 4), ssrc);
+    SetBE32(std::span<uint8_t>(&packet[8], 4), ssrc);
     DeliverPacket(packet);
 
     // Verify we have one new stream for each loop iteration.
@@ -2903,7 +2903,7 @@ TEST_P(WebRtcVoiceEngineTestFake, RecvMultipleUnsignaled) {
 
   // Sending on the same SSRCs again should not create new streams.
   for (uint32_t ssrc = 1; ssrc < (1 + kMaxUnsignaledRecvStreams); ++ssrc) {
-    SetBE32(ArrayView<uint8_t>(&packet[8], 4), ssrc);
+    SetBE32(std::span<uint8_t>(&packet[8], 4), ssrc);
     DeliverPacket(packet);
 
     EXPECT_EQ(kMaxUnsignaledRecvStreams, call_.GetAudioReceiveStreams().size());
@@ -2913,7 +2913,7 @@ TEST_P(WebRtcVoiceEngineTestFake, RecvMultipleUnsignaled) {
 
   // Send on another SSRC, the oldest unsignaled stream (SSRC=1) is replaced.
   constexpr uint32_t kAnotherSsrc = 667;
-  SetBE32(ArrayView<uint8_t>(&packet[8], 4), kAnotherSsrc);
+  SetBE32(std::span<uint8_t>(&packet[8], 4), kAnotherSsrc);
   DeliverPacket(packet);
 
   const auto& streams = call_.GetAudioReceiveStreams();
@@ -2938,7 +2938,7 @@ TEST_P(WebRtcVoiceEngineTestFake, RecvUnsignaledAfterSignaled) {
 
   // Add a known stream, send packet and verify we got it.
   const uint32_t signaled_ssrc = 1;
-  SetBE32(ArrayView<uint8_t>(&packet[8], 4), signaled_ssrc);
+  SetBE32(std::span<uint8_t>(&packet[8], 4), signaled_ssrc);
   EXPECT_TRUE(AddRecvStream(signaled_ssrc));
   DeliverPacket(packet);
   EXPECT_TRUE(GetRecvStream(signaled_ssrc).VerifyLastPacket(packet));
@@ -2947,7 +2947,7 @@ TEST_P(WebRtcVoiceEngineTestFake, RecvUnsignaledAfterSignaled) {
   // Note that the first unknown SSRC cannot be 0, because we only support
   // creating receive streams for SSRC!=0.
   const uint32_t unsignaled_ssrc = 7011;
-  SetBE32(ArrayView<uint8_t>(&packet[8], 4), unsignaled_ssrc);
+  SetBE32(std::span<uint8_t>(&packet[8], 4), unsignaled_ssrc);
   DeliverPacket(packet);
   EXPECT_TRUE(GetRecvStream(unsignaled_ssrc).VerifyLastPacket(packet));
   EXPECT_EQ(2u, call_.GetAudioReceiveStreams().size());
@@ -2955,7 +2955,7 @@ TEST_P(WebRtcVoiceEngineTestFake, RecvUnsignaledAfterSignaled) {
   DeliverPacket(packet);
   EXPECT_EQ(2, GetRecvStream(unsignaled_ssrc).received_packets());
 
-  SetBE32(ArrayView<uint8_t>(&packet[8], 4), signaled_ssrc);
+  SetBE32(std::span<uint8_t>(&packet[8], 4), signaled_ssrc);
   DeliverPacket(packet);
   EXPECT_EQ(2, GetRecvStream(signaled_ssrc).received_packets());
   EXPECT_EQ(2u, call_.GetAudioReceiveStreams().size());
@@ -3420,7 +3420,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetOutputVolumeUnsignaledRecvStream) {
   // Spawn an unsignaled stream by sending a packet - gain should be 2.
   uint8_t pcmuFrame2[sizeof(kPcmuFrame)];
   memcpy(pcmuFrame2, kPcmuFrame, sizeof(kPcmuFrame));
-  SetBE32(ArrayView<uint8_t>(&pcmuFrame2[8], 4), kSsrcX);
+  SetBE32(std::span<uint8_t>(&pcmuFrame2[8], 4), kSsrcX);
   DeliverPacket(pcmuFrame2);
   EXPECT_DOUBLE_EQ(2, GetRecvStream(kSsrcX).gain());
 
@@ -3482,7 +3482,7 @@ TEST_P(WebRtcVoiceEngineTestFake,
   // Spawn an unsignaled stream by sending a packet - delay should be 100.
   uint8_t pcmuFrame2[sizeof(kPcmuFrame)];
   memcpy(pcmuFrame2, kPcmuFrame, sizeof(kPcmuFrame));
-  SetBE32(ArrayView<uint8_t>(&pcmuFrame2[8], 4), kSsrcX);
+  SetBE32(std::span<uint8_t>(&pcmuFrame2[8], 4), kSsrcX);
   DeliverPacket(pcmuFrame2);
   EXPECT_EQ(
       100, receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrcX).value_or(-1));
@@ -3659,7 +3659,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetRawAudioSinkUnsignaledRecvStream) {
   // and the previous unsignaled stream should lose it.
   uint8_t pcmuFrame2[sizeof(kPcmuFrame)];
   memcpy(pcmuFrame2, kPcmuFrame, sizeof(kPcmuFrame));
-  SetBE32(ArrayView<uint8_t>(&pcmuFrame2[8], 4), kSsrcX);
+  SetBE32(std::span<uint8_t>(&pcmuFrame2[8], 4), kSsrcX);
   DeliverPacket(pcmuFrame2);
   if (kMaxUnsignaledRecvStreams > 1) {
     EXPECT_EQ(nullptr, GetRecvStream(kSsrc1).sink());
diff --git a/media/sctp/dcsctp_transport.cc b/media/sctp/dcsctp_transport.cc
index 63f3d52eeb..03ffc69ae8 100644
--- a/media/sctp/dcsctp_transport.cc
+++ b/media/sctp/dcsctp_transport.cc
@@ -17,12 +17,12 @@
 #include <limits>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/data_channel_interface.h"
 #include "api/dtls_transport_interface.h"
 #include "api/environment/environment.h"
@@ -440,7 +440,7 @@ void DcSctpTransport::SetBufferedAmountLowThreshold(int sid, size_t bytes) {
 }
 
 SendPacketStatus DcSctpTransport::SendPacketWithStatus(
-    ArrayView<const uint8_t> data) {
+    std::span<const uint8_t> data) {
   RTC_DCHECK_RUN_ON(network_thread_);
   RTC_DCHECK(socket_);
 
@@ -591,7 +591,7 @@ void DcSctpTransport::OnConnectionRestarted() {
 }
 
 void DcSctpTransport::OnStreamsResetFailed(
-    ArrayView<const dcsctp::StreamID> outgoing_streams,
+    std::span<const dcsctp::StreamID> outgoing_streams,
     absl::string_view reason) {
   // TODO(orphis): Need a test to check for correct behavior
   for (auto& stream_id : outgoing_streams) {
@@ -603,7 +603,7 @@ void DcSctpTransport::OnStreamsResetFailed(
 }
 
 void DcSctpTransport::OnStreamsResetPerformed(
-    ArrayView<const dcsctp::StreamID> outgoing_streams) {
+    std::span<const dcsctp::StreamID> outgoing_streams) {
   RTC_DCHECK_RUN_ON(network_thread_);
   for (auto& stream_id : outgoing_streams) {
     RTC_LOG(LS_INFO) << debug_name_
@@ -631,7 +631,7 @@ void DcSctpTransport::OnStreamsResetPerformed(
 }
 
 void DcSctpTransport::OnIncomingStreamsReset(
-    ArrayView<const dcsctp::StreamID> incoming_streams) {
+    std::span<const dcsctp::StreamID> incoming_streams) {
   RTC_DCHECK_RUN_ON(network_thread_);
   for (auto& stream_id : incoming_streams) {
     RTC_LOG(LS_INFO) << debug_name_
diff --git a/media/sctp/dcsctp_transport.h b/media/sctp/dcsctp_transport.h
index 143d4b3c26..bf945c9462 100644
--- a/media/sctp/dcsctp_transport.h
+++ b/media/sctp/dcsctp_transport.h
@@ -16,11 +16,11 @@
 #include <functional>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/dtls_transport_interface.h"
 #include "api/environment/environment.h"
 #include "api/field_trials_view.h"
@@ -83,7 +83,7 @@ class DcSctpTransport : public SctpTransportInternal,
  private:
   // dcsctp::DcSctpSocketCallbacks
   dcsctp::SendPacketStatus SendPacketWithStatus(
-      ArrayView<const uint8_t> data) override;
+      std::span<const uint8_t> data) override;
   std::unique_ptr<dcsctp::Timeout> CreateTimeout(
       TaskQueueBase::DelayPrecision precision) override;
   dcsctp::TimeMs TimeMillis() override;
@@ -96,12 +96,12 @@ class DcSctpTransport : public SctpTransportInternal,
   void OnConnected() override;
   void OnClosed() override;
   void OnConnectionRestarted() override;
-  void OnStreamsResetFailed(ArrayView<const dcsctp::StreamID> outgoing_streams,
+  void OnStreamsResetFailed(std::span<const dcsctp::StreamID> outgoing_streams,
                             absl::string_view reason) override;
   void OnStreamsResetPerformed(
-      ArrayView<const dcsctp::StreamID> outgoing_streams) override;
+      std::span<const dcsctp::StreamID> outgoing_streams) override;
   void OnIncomingStreamsReset(
-      ArrayView<const dcsctp::StreamID> incoming_streams) override;
+      std::span<const dcsctp::StreamID> incoming_streams) override;
 
   // Transport callbacks
   void ConnectTransportSignals();

From 77a305351d9846d9d291d8beca01c93ce1c3880a Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Tue, 10 Mar 2026 05:04:27 -0700
Subject: [PATCH 251/472] Roll chromium_revision c22ce439a0..07798bc2eb
 (1595685:1596985)

Change log: https://chromium.googlesource.com/chromium/src/+log/c22ce439a0..07798bc2eb
Full diff: https://chromium.googlesource.com/chromium/src/+/c22ce439a0..07798bc2eb

Changed dependencies
* siso_version: git_revision:a4a219a47e8a8d1a4a2fe306ffa70d65ab4b23b8..git_revision:bea0f245b6973702ddcb457090706d03145855f8
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/4e55723cd7..98d6658ed1
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/8da238cb7e..632417adc3
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/cbc3c0cf69..07d5e4bcb0
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/0c56ed891b..cca29c0f05
* src/third_party/android_build_tools/error_prone/cipd: zUmV9Nh4JfIOMh9QU7LhIhPvodo_Aka-0IAIPIcLKVcC..Y-wIN3nTyykC3fwieDSmffv26BOT2wXRVPktZSw_oLUC
* src/third_party/android_build_tools/lint/cipd: Ow8jpaRkHsJyhElyBOWWKBWl0KKNNuqbAE-cYWSPwQAC..1N_9yErKkcZgqyxBghA0bb5L5AFEPZCt36Gv2o8_OAwC
* src/third_party/android_build_tools/manifest_merger/cipd: tu2pGdrcOsdBnUbmZgzOgxIrZh-YJhlAL7ICIZRTjH4C..xT8A6Q_b44_iYKqAMvRkWmvID-ps-I6uBDfRTQv68bcC
* src/third_party/android_deps/autorolled/cipd: gOFPZuLmUt7S4rnhusfD5p-Fhy3qllnFyNIwDbThlusC..umaYTBt4Vsa5QZYp57rLJDNzoXalsR2hPGwfYbpZYfwC
* src/third_party/androidx/cipd: G93sLsFQ30pYjUz4kY55FggeIFragvakLDCkWrLtbOEC.._YK_XyeZvkOivG1aqms80wS3mgx5K-L9yHOhZDzvlI4C
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/5bf54f4ac7..476f483518
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/d903007eb7..e0ebf38a01
* src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/568f6a9253..338a5c004c
* src/third_party/dav1d/libdav1d: https://chromium.googlesource.com/external/github.com/videolan/dav1d.git/+log/b546257f77..241a6b236a
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/d8c8ee6fb2..81550faea4
* src/third_party/freetype/src: https://chromium.googlesource.com/chromium/src/third_party/freetype2.git/+log/aa8ca27f9b..1c6e43ab90
* src/third_party/fuzztest/src: https://chromium.googlesource.com/external/github.com/google/fuzztest.git/+log/1f7726d61f..e58f8dd957
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/a18875d1df..5a3a5d5314
* src/third_party/kotlin_stdlib/cipd: 9ldUaVxyYClDj9QDPGLyxhrrLE40VxHjm65wiU_4xaoC..KABHiJiUrNNzDUFrZt5xyABtjqZCS-xG3fIdF-pe19IC
* src/third_party/libaom/source/libaom: https://aomedia.googlesource.com/aom.git/+log/39606bf4ba..98ce0d2a61
* src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/4e2524f6ae..8df8a70a83
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/7d3fdf89c3..5268fe512e
* src/third_party/siso/cipd: git_revision:a4a219a47e8a8d1a4a2fe306ffa70d65ab4b23b8..git_revision:bea0f245b6973702ddcb457090706d03145855f8
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/6870c91a2b..9e8a98181c
DEPS diff: https://chromium.googlesource.com/chromium/src/+/c22ce439a0..07798bc2eb/DEPS

BUG=None

Change-Id: Idfae4c90b0336f6cbec2d23eb5f68c9447a70725
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/455520
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47104}
---
 DEPS | 48 ++++++++++++++++++++++++------------------------
 1 file changed, 24 insertions(+), 24 deletions(-)

diff --git a/DEPS b/DEPS
index db87646089..d552d47ba5 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': 'c22ce439a0c6903da40a102f8e338f114b105eca',
+  'chromium_revision': '07798bc2eb402da8fcfaa15692a487a2312b4320',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -50,7 +50,7 @@ vars = {
   # reclient CIPD package version
   'reclient_version': 're_client_version:0.185.0.db415f21-gomaip',
   # siso CIPD package version.
-  'siso_version': 'git_revision:a4a219a47e8a8d1a4a2fe306ffa70d65ab4b23b8',
+  'siso_version': 'git_revision:bea0f245b6973702ddcb457090706d03145855f8',
 
   # ninja CIPD package.
   'ninja_package': 'infra/3pp/tools/ninja/',
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@4e55723cd763c38199eae267bb0f933445d8e44a',
+    'https://chromium.googlesource.com/chromium/src/build@98d6658ed11a1b5abc2d29feec55c51d03861692',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@fef8d6947fa54202e8a00c58d5ac8fd15780bc33',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@8da238cb7ee4beb61d8a272d9c2c0da7e6909c79',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@632417adc3fb78ab22c12a3951f15f8e55e61080',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@cbc3c0cf69aa7a8b59230216a2fb97f15b1b10c4',
+    'https://chromium.googlesource.com/chromium/src/testing@07d5e4bcb0826b49ab4318ff0100332fdc6f4f22',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@0c56ed891bcd96446651f4021878ae7c9b904cd8',
+    'https://chromium.googlesource.com/chromium/src/third_party@cca29c0f05afea0156c8f7946ed402af2573d144',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -331,13 +331,13 @@ deps = {
   'src/third_party/clang-format/script':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@c2725e0622e1a86d55f14514f2177a39efea4a0e',
   'src/third_party/compiler-rt/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@568f6a925307495d6ba838a68d0dfe7ab375c946',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@338a5c004c774a8927899b1f1c0c25a82d14510f',
   'src/third_party/libc++/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@7ab65651aed6802d2599dcb7a73b1f82d5179d05',
   'src/third_party/libc++abi/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@8f11bb1d4438d0239d0dfc1bd9456a9f31629dda',
   'src/third_party/llvm-libc/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@4e2524f6aecac8c50998cd7b673d912239d0e54e',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@8df8a70a8349a709b8996a11279eb226cae5aced',
   'src/third_party/libunwind/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@78884e23fe39cf5cc6987ea188a9b802d65a21c9',
 
@@ -435,7 +435,7 @@ deps = {
       'packages': [
           {
                'package': 'chromium/third_party/android_build_tools/error_prone',
-               'version': 'zUmV9Nh4JfIOMh9QU7LhIhPvodo_Aka-0IAIPIcLKVcC',
+               'version': 'Y-wIN3nTyykC3fwieDSmffv26BOT2wXRVPktZSw_oLUC',
           },
       ],
       'condition': 'checkout_android',
@@ -457,7 +457,7 @@ deps = {
       'packages': [
           {
                'package': 'chromium/third_party/android_build_tools/lint',
-               'version': 'Ow8jpaRkHsJyhElyBOWWKBWl0KKNNuqbAE-cYWSPwQAC',
+               'version': '1N_9yErKkcZgqyxBghA0bb5L5AFEPZCt36Gv2o8_OAwC',
           },
       ],
       'condition': 'checkout_android and non_git_source',
@@ -488,11 +488,11 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@5bf54f4ac7f73e8cc625d1292a39edd508f64e2f',
+    'https://boringssl.googlesource.com/boringssl.git@476f4835186050a44f7fe6f088abbd279b634194',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@8be0e3114685fcc1589561067282edf75ea1259a',
   'src/third_party/catapult':
-    'https://chromium.googlesource.com/catapult.git@d903007eb7787e17c1314f1fdb4787c9a1f7c7ec',
+    'https://chromium.googlesource.com/catapult.git@e0ebf38a01214aba11f31daa1c743782def031d5',
   'src/third_party/ced/src': {
     'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5',
   },
@@ -511,7 +511,7 @@ deps = {
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@a18875d1df6a3fdac2cd7339a4fdb6aff29db89e',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@5a3a5d53145b94895610825e783a8896a61a3c73',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -519,7 +519,7 @@ deps = {
       'condition': 'checkout_linux',
   },
   'src/third_party/freetype/src':
-    'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@aa8ca27f9b48bbb99f02d605b7dbf1d6a016cbcb',
+    'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@1c6e43ab90af00261334c6cd214a5a33178eacd4',
   'src/third_party/harfbuzz-ng/src':
     'https://chromium.googlesource.com/external/github.com/harfbuzz/harfbuzz.git@fa2908bf16d2ccd6623f4d575455fea72a1a722b',
   'src/third_party/google_benchmark/src': {
@@ -573,7 +573,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/kotlin_stdlib',
-              'version': '9ldUaVxyYClDj9QDPGLyxhrrLE40VxHjm65wiU_4xaoC',
+              'version': 'KABHiJiUrNNzDUFrZt5xyABtjqZCS-xG3fIdF-pe19IC',
           },
       ],
       'condition': 'checkout_android',
@@ -593,15 +593,15 @@ deps = {
   'src/third_party/libFuzzer/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt/lib/fuzzer.git@bea408a6e01f0f7e6c82a43121fe3af4506c932e',
   'src/third_party/fuzztest/src':
-    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@1f7726d61f7afa9aca1198a9395ede472ed70366',
+    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@e58f8dd9577523f636704a96b907137f36fd4b5b',
   'src/third_party/libjpeg_turbo':
     'https://chromium.googlesource.com/chromium/deps/libjpeg_turbo.git@d1f5f2393e0d51f840207342ae86e55a86443288',
   'src/third_party/libsrtp':
     'https://chromium.googlesource.com/chromium/deps/libsrtp.git@e8383771af8aa4096f5bcfe3743a5ea128f88a9a',
   'src/third_party/dav1d/libdav1d':
-    'https://chromium.googlesource.com/external/github.com/videolan/dav1d.git@b546257f770768b2c88258c533da38b91a06f737',
+    'https://chromium.googlesource.com/external/github.com/videolan/dav1d.git@241a6b236a9c84441c872119c1f31b4074c0589d',
   'src/third_party/libaom/source/libaom':
-    'https://aomedia.googlesource.com/aom.git@39606bf4bae2befef8c32bcdfab402226bb33783',
+    'https://aomedia.googlesource.com/aom.git@98ce0d2a610f1465ea14d525bce9fa46d4b05c03',
   'src/third_party/libgav1/src':
     Var('chromium_git') + '/codecs/libgav1.git' + '@' + '40f58ed32ff39071c3f2a51056dbc49a070af0dc',
   'src/third_party/libunwindstack': {
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '7d3fdf89c361951a421fb45ff91b1f74768137ac',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '5268fe512ed19082bd833cdea8e5abbba0d0599e',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@6870c91a2b73cae054a5e00f802b3108bc959619',
+    'https://chromium.googlesource.com/chromium/src/tools@9e8a98181c1b2243facb3a9953ea9b4bfcbdfc41',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'G93sLsFQ30pYjUz4kY55FggeIFragvakLDCkWrLtbOEC',
+          'version': '_YK_XyeZvkOivG1aqms80wS3mgx5K-L9yHOhZDzvlI4C',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -718,7 +718,7 @@ deps = {
       'packages': [
           {
                'package': 'chromium/third_party/android_build_tools/manifest_merger',
-               'version': 'tu2pGdrcOsdBnUbmZgzOgxIrZh-YJhlAL7ICIZRTjH4C',
+               'version': 'xT8A6Q_b44_iYKqAMvRkWmvID-ps-I6uBDfRTQv68bcC',
           },
       ],
       'condition': 'checkout_android',
@@ -842,7 +842,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/android_deps/autorolled',
-              'version': 'gOFPZuLmUt7S4rnhusfD5p-Fhy3qllnFyNIwDbThlusC',
+              'version': 'umaYTBt4Vsa5QZYp57rLJDNzoXalsR2hPGwfYbpZYfwC',
           },
       ],
       'condition': 'checkout_android and non_git_source',
@@ -864,7 +864,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7364b490b5f78d58efe23ea76e74210fd6c3c76f',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + 'd8c8ee6fb2cfe1af53b020ff10a70fbd9b47ccad',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '81550faea475b1234ff31068605ac9536c9f2f92',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From 2217b0c66cc7dae4227b33c0ddb7eefdc9645a58 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Fri, 6 Mar 2026 21:45:27 +0100
Subject: [PATCH 252/472] Remove test code related to unused variables
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

`input_aecdump` and `processed_capture_samples` are never used,
so the code related to those variables, can be removed.

Bug: none
Change-Id: I4f0b7db01f1e218f62e732ab9a247dd0b9708666
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356542
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47105}
---
 .../test/aec_dump_based_simulator.cc          | 43 +++----------------
 .../test/audio_processing_simulator.cc        |  9 +---
 .../test/audio_processing_simulator.h         |  4 --
 .../test/audioproc_float_impl.cc              | 16 ++-----
 .../audio_processing/test/protobuf_utils.cc   | 34 ---------------
 .../audio_processing/test/protobuf_utils.h    |  6 ---
 modules/audio_processing/test/test_utils.cc   | 19 --------
 modules/audio_processing/test/test_utils.h    | 19 --------
 8 files changed, 10 insertions(+), 140 deletions(-)

diff --git a/modules/audio_processing/test/aec_dump_based_simulator.cc b/modules/audio_processing/test/aec_dump_based_simulator.cc
index 7af6d414c0..f78ab46a50 100644
--- a/modules/audio_processing/test/aec_dump_based_simulator.cc
+++ b/modules/audio_processing/test/aec_dump_based_simulator.cc
@@ -19,7 +19,6 @@
 #include <iostream>
 #include <memory>
 #include <optional>
-#include <sstream>  // no-presubmit-check TODO(webrtc:8982)
 #include <string>
 #include <utility>
 
@@ -79,18 +78,6 @@ bool VerifyFloatBitExactness(const audioproc::Stream& msg,
   return true;
 }
 
-// Selectively reads the next proto-buf message from dump-file or string input.
-// Returns a bool indicating whether a new message was available.
-bool ReadNextMessage(bool use_dump_file,
-                     FILE* dump_input_file,
-                     std::stringstream& input,
-                     audioproc::Event& event_msg) {
-  if (use_dump_file) {
-    return ReadMessageFromFile(dump_input_file, &event_msg);
-  }
-  return ReadMessageFromString(&input, &event_msg);
-}
-
 }  // namespace
 
 AecDumpBasedSimulator::AecDumpBasedSimulator(
@@ -256,19 +243,12 @@ void AecDumpBasedSimulator::Process() {
         CheckedDivExact(sample_rate_hz, kChunksPerSecond), 1));
   }
 
-  const bool use_dump_file = !settings_.aec_dump_input_string.has_value();
-  std::stringstream input;
-  if (use_dump_file) {
-    dump_input_file_ =
-        OpenFile(settings_.aec_dump_input_filename->c_str(), "rb");
-  } else {
-    input << settings_.aec_dump_input_string.value();
-  }
+  dump_input_file_ = OpenFile(settings_.aec_dump_input_filename->c_str(), "rb");
 
   audioproc::Event event_msg;
   int capture_frames_since_init = 0;
   int init_index = 0;
-  while (ReadNextMessage(use_dump_file, dump_input_file_, input, event_msg)) {
+  while (ReadMessageFromFile(dump_input_file_, &event_msg)) {
     SelectivelyToggleDataDumping(init_index, capture_frames_since_init);
     HandleEvent(event_msg, capture_frames_since_init, init_index);
 
@@ -281,28 +261,19 @@ void AecDumpBasedSimulator::Process() {
               *settings_.init_to_process >= init_index);
   }
 
-  if (use_dump_file) {
-    fclose(dump_input_file_);
-  }
+  fclose(dump_input_file_);
 
   DetachAecDump();
 }
 
 void AecDumpBasedSimulator::Analyze() {
-  const bool use_dump_file = !settings_.aec_dump_input_string.has_value();
-  std::stringstream input;
-  if (use_dump_file) {
-    dump_input_file_ =
-        OpenFile(settings_.aec_dump_input_filename->c_str(), "rb");
-  } else {
-    input << settings_.aec_dump_input_string.value();
-  }
+  dump_input_file_ = OpenFile(settings_.aec_dump_input_filename->c_str(), "rb");
 
   audioproc::Event event_msg;
   int num_capture_frames = 0;
   int num_render_frames = 0;
   int init_index = 0;
-  while (ReadNextMessage(use_dump_file, dump_input_file_, input, event_msg)) {
+  while (ReadMessageFromFile(dump_input_file_, &event_msg)) {
     if (event_msg.type() == audioproc::Event::INIT) {
       ++init_index;
       constexpr float kNumFramesPerSecond = 100.f;
@@ -323,9 +294,7 @@ void AecDumpBasedSimulator::Analyze() {
     }
   }
 
-  if (use_dump_file) {
-    fclose(dump_input_file_);
-  }
+  fclose(dump_input_file_);
 }
 
 void AecDumpBasedSimulator::HandleEvent(const audioproc::Event& event_msg,
diff --git a/modules/audio_processing/test/audio_processing_simulator.cc b/modules/audio_processing/test/audio_processing_simulator.cc
index 66e271308f..aa1af1737c 100644
--- a/modules/audio_processing/test/audio_processing_simulator.cc
+++ b/modules/audio_processing/test/audio_processing_simulator.cc
@@ -222,11 +222,7 @@ void AudioProcessingSimulator::ProcessStream(bool fixed_interface) {
     applied_input_volume_ = ap_->recommended_stream_analog_level();
   }
 
-  if (buffer_memory_writer_) {
-    RTC_CHECK(!buffer_file_writer_);
-    buffer_memory_writer_->Write(*out_buf_);
-  } else if (buffer_file_writer_) {
-    RTC_CHECK(!buffer_memory_writer_);
+  if (buffer_file_writer_) {
     buffer_file_writer_->Write(*out_buf_);
   }
 
@@ -373,9 +369,6 @@ void AudioProcessingSimulator::SetupOutput() {
                       static_cast<size_t>(out_config_.num_channels()),
                       settings_.wav_output_format));
     buffer_file_writer_.reset(new ChannelBufferWavWriter(std::move(out_file)));
-  } else if (settings_.aec_dump_input_string.has_value()) {
-    buffer_memory_writer_ = std::make_unique<ChannelBufferVectorWriter>(
-        settings_.processed_capture_samples);
   }
 
   if (settings_.linear_aec_output_filename) {
diff --git a/modules/audio_processing/test/audio_processing_simulator.h b/modules/audio_processing/test/audio_processing_simulator.h
index 3ac6023905..1bf5c1b49f 100644
--- a/modules/audio_processing/test/audio_processing_simulator.h
+++ b/modules/audio_processing/test/audio_processing_simulator.h
@@ -21,7 +21,6 @@
 #include <vector>
 
 #include "absl/base/nullability.h"
-#include "absl/strings/string_view.h"
 #include "api/audio/audio_processing.h"
 #include "api/scoped_refptr.h"
 #include "common_audio/channel_buffer.h"
@@ -152,8 +151,6 @@ struct SimulationSettings {
   std::optional<std::string> call_order_input_filename;
   std::optional<std::string> call_order_output_filename;
   std::optional<std::string> aec_settings_filename;
-  std::optional<absl::string_view> aec_dump_input_string;
-  std::vector<float>* processed_capture_samples = nullptr;
   bool analysis_only = false;
   std::optional<int> dump_start_frame;
   std::optional<int> dump_end_frame;
@@ -244,7 +241,6 @@ class AudioProcessingSimulator {
   size_t num_reverse_process_stream_calls_ = 0;
   std::unique_ptr<ChannelBufferWavWriter> buffer_file_writer_;
   std::unique_ptr<ChannelBufferWavWriter> reverse_buffer_file_writer_;
-  std::unique_ptr<ChannelBufferVectorWriter> buffer_memory_writer_;
   std::unique_ptr<WavWriter> linear_aec_output_file_writer_;
   ApiCallStatistics api_call_statistics_;
   std::ofstream residual_echo_likelihood_graph_writer_;
diff --git a/modules/audio_processing/test/audioproc_float_impl.cc b/modules/audio_processing/test/audioproc_float_impl.cc
index a6b26f1808..589fa1ad97 100644
--- a/modules/audio_processing/test/audioproc_float_impl.cc
+++ b/modules/audio_processing/test/audioproc_float_impl.cc
@@ -565,11 +565,6 @@ void PerformBasicParameterSanityChecks(const SimulationSettings& settings) {
         "Error: The aec dump file cannot be specified "
         "together with input wav files!\n");
 
-    ReportConditionalErrorAndExit(
-        !!settings.aec_dump_input_string,
-        "Error: The aec dump input string cannot be specified "
-        "together with input wav files!\n");
-
     ReportConditionalErrorAndExit(!!settings.artificial_nearend_filename,
                                   "Error: The artificial nearend cannot be "
                                   "specified together with input wav files!\n");
@@ -585,13 +580,8 @@ void PerformBasicParameterSanityChecks(const SimulationSettings& settings) {
         "must be specified if the reverse output wav filename is specified!\n");
   } else {
     ReportConditionalErrorAndExit(
-        !settings.aec_dump_input_filename && !settings.aec_dump_input_string,
-        "Error: Either the aec dump input file, the wav "
-        "input file or the aec dump input string must be specified!\n");
-    ReportConditionalErrorAndExit(
-        settings.aec_dump_input_filename && settings.aec_dump_input_string,
-        "Error: The aec dump input file cannot be specified together with the "
-        "aec dump input string!\n");
+        !settings.aec_dump_input_filename,
+        "Error: The aec dump input file must be specified!\n");
   }
 
   ReportConditionalErrorAndExit(settings.use_aec && !(*settings.use_aec) &&
@@ -857,7 +847,7 @@ int RunSimulation(
   RTC_CHECK(audio_processing);
 
   std::unique_ptr<AudioProcessingSimulator> processor;
-  if (settings.aec_dump_input_filename || settings.aec_dump_input_string) {
+  if (settings.aec_dump_input_filename) {
     processor = std::make_unique<AecDumpBasedSimulator>(
         settings, std::move(audio_processing));
   } else {
diff --git a/modules/audio_processing/test/protobuf_utils.cc b/modules/audio_processing/test/protobuf_utils.cc
index fb0c08209a..227966e4b3 100644
--- a/modules/audio_processing/test/protobuf_utils.cc
+++ b/modules/audio_processing/test/protobuf_utils.cc
@@ -14,35 +14,12 @@
 #include <cstdint>
 #include <cstdio>
 #include <memory>
-#include <sstream>  // no-presubmit-check TODO(webrtc:8982)
 
 #include "rtc_base/protobuf_utils.h"
 #include "rtc_base/system/arch.h"
 
 namespace webrtc {
 
-namespace {
-// Allocates new memory in the memory owned by the unique_ptr to fit the raw
-// message and returns the number of bytes read when having a string stream as
-// input.
-size_t ReadMessageBytesFromString(std::stringstream* input,
-                                  std::unique_ptr<uint8_t[]>* bytes) {
-  int32_t size = 0;
-  input->read(reinterpret_cast<char*>(&size), sizeof(int32_t));
-  int32_t size_read = input->gcount();
-  if (size_read != sizeof(int32_t))
-    return 0;
-  if (size <= 0)
-    return 0;
-
-  *bytes = std::make_unique<uint8_t[]>(size);
-  input->read(reinterpret_cast<char*>(bytes->get()),
-              size * sizeof((*bytes)[0]));
-  size_read = input->gcount();
-  return size_read == size ? size : 0;
-}
-}  // namespace
-
 size_t ReadMessageBytesFromFile(FILE* file, std::unique_ptr<uint8_t[]>* bytes) {
 // The "wire format" for the size is little-endian. Assume we're running on
 // a little-endian machine.
@@ -70,15 +47,4 @@ bool ReadMessageFromFile(FILE* file, MessageLite* msg) {
   return msg->ParseFromArray(bytes.get(), size);
 }
 
-// Returns true on success, false on error or end of string stream.
-bool ReadMessageFromString(std::stringstream* input, MessageLite* msg) {
-  std::unique_ptr<uint8_t[]> bytes;
-  size_t size = ReadMessageBytesFromString(input, &bytes);
-  if (!size)
-    return false;
-
-  msg->Clear();
-  return msg->ParseFromArray(bytes.get(), size);
-}
-
 }  // namespace webrtc
diff --git a/modules/audio_processing/test/protobuf_utils.h b/modules/audio_processing/test/protobuf_utils.h
index 2685a4c75a..77f14913f1 100644
--- a/modules/audio_processing/test/protobuf_utils.h
+++ b/modules/audio_processing/test/protobuf_utils.h
@@ -15,7 +15,6 @@
 #include <cstdint>
 #include <cstdio>
 #include <memory>
-#include <sstream>  // no-presubmit-check TODO(webrtc:8982)
 
 #include "rtc_base/protobuf_utils.h"
 
@@ -31,11 +30,6 @@ size_t ReadMessageBytesFromFile(FILE* file, std::unique_ptr<uint8_t[]>* bytes);
 // Returns true on success, false on error or end-of-file.
 bool ReadMessageFromFile(FILE* file, MessageLite* msg);
 
-// Returns true on success, false on error or end of string stream.
-bool ReadMessageFromString(
-    std::stringstream* input,  // no-presubmit-check TODO(webrtc:8982)
-    MessageLite* msg);
-
 }  // namespace webrtc
 
 #endif  // MODULES_AUDIO_PROCESSING_TEST_PROTOBUF_UTILS_H_
diff --git a/modules/audio_processing/test/test_utils.cc b/modules/audio_processing/test/test_utils.cc
index d4b221d668..24cf176626 100644
--- a/modules/audio_processing/test/test_utils.cc
+++ b/modules/audio_processing/test/test_utils.cc
@@ -113,25 +113,6 @@ void ChannelBufferWavWriter::Write(const ChannelBuffer<float>& buffer) {
   file_->WriteSamples(&interleaved_[0], interleaved_.size());
 }
 
-ChannelBufferVectorWriter::ChannelBufferVectorWriter(std::vector<float>* output)
-    : output_(output) {
-  RTC_DCHECK(output_);
-}
-
-ChannelBufferVectorWriter::~ChannelBufferVectorWriter() = default;
-
-void ChannelBufferVectorWriter::Write(const ChannelBuffer<float>& buffer) {
-  // Account for sample rate changes throughout a simulation.
-  interleaved_buffer_.resize(buffer.size());
-  InterleavedView<float> view(&interleaved_buffer_[0], buffer.num_frames(),
-                              buffer.num_channels());
-  Interleave(buffer.channels(), buffer.num_frames(), buffer.num_channels(),
-             view);
-  size_t old_size = output_->size();
-  output_->resize(old_size + interleaved_buffer_.size());
-  FloatToFloatS16(interleaved_buffer_.data(), interleaved_buffer_.size(),
-                  output_->data() + old_size);
-}
 
 FILE* OpenFile(absl::string_view filename, absl::string_view mode) {
   std::string filename_str(filename);
diff --git a/modules/audio_processing/test/test_utils.h b/modules/audio_processing/test/test_utils.h
index e979c85978..114cd13b10 100644
--- a/modules/audio_processing/test/test_utils.h
+++ b/modules/audio_processing/test/test_utils.h
@@ -101,25 +101,6 @@ class ChannelBufferWavWriter final {
   std::vector<float> interleaved_;
 };
 
-// Takes a pointer to a vector. Allows appending the samples of channel buffers
-// to the given vector, by interleaving the samples and converting them to float
-// S16.
-class ChannelBufferVectorWriter final {
- public:
-  explicit ChannelBufferVectorWriter(std::vector<float>* output);
-  ChannelBufferVectorWriter(const ChannelBufferVectorWriter&) = delete;
-  ChannelBufferVectorWriter& operator=(const ChannelBufferVectorWriter&) =
-      delete;
-  ~ChannelBufferVectorWriter();
-
-  // Creates an interleaved copy of `buffer`, converts the samples to float S16
-  // and appends the result to output_.
-  void Write(const ChannelBuffer<float>& buffer);
-
- private:
-  std::vector<float> interleaved_buffer_;
-  std::vector<float>* output_;
-};
 
 // Exits on failure; do not use in unit tests.
 FILE* OpenFile(absl::string_view filename, absl::string_view mode);

From 7040fbcb5357f81d9b86b60ad4ba70426a25cbef Mon Sep 17 00:00:00 2001
From: Philipp Hancke <philipp.hancke@googlemail.com>
Date: Fri, 20 Feb 2026 08:41:57 +0100
Subject: [PATCH 253/472] build: change more tests to use rtc_test instead of
 rtc_exectuable

which requires clobbering on the ios bots. Also remove _temp suffix
from base64 benchmark.

Bug: webrtc:467294026
Change-Id: I4d1fe493c41438ab2f83bc1c3acbecc024b15128
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/443384
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47106}
---
 BUILD.gn                      |  2 +-
 modules/audio_coding/BUILD.gn | 12 ++++--------
 modules/rtp_rtcp/BUILD.gn     |  3 +--
 p2p/BUILD.gn                  |  3 +--
 rtc_base/BUILD.gn             |  2 +-
 tools_webrtc/get_landmines.py |  4 +++-
 6 files changed, 11 insertions(+), 15 deletions(-)

diff --git a/BUILD.gn b/BUILD.gn
index cd4379fb3f..4d154f8e19 100644
--- a/BUILD.gn
+++ b/BUILD.gn
@@ -694,7 +694,7 @@ if (rtc_include_tests && !build_with_chromium) {
     rtc_executable("benchmarks") {
       testonly = true
       deps = [
-        "rtc_base:base64_benchmark_temp",
+        "rtc_base:base64_benchmark",
         "rtc_base/synchronization:mutex_benchmark",
         "test:benchmark_main",
       ]
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index 132b293f79..d8e1e9788a 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -1280,7 +1280,7 @@ if (rtc_include_tests) {
       ]
     }
 
-    rtc_executable("neteq_opus_quality_test") {
+    rtc_test("neteq_opus_quality_test") {
       testonly = true
 
       sources = [ "neteq/test/neteq_opus_quality_test.cc" ]
@@ -1295,7 +1295,6 @@ if (rtc_include_tests) {
         "../../api/audio_codecs:audio_codecs_api",
         "../../rtc_base:buffer",
         "../../rtc_base:checks",
-        "../../test:test_main",
         "../../test:test_support",
         "//testing/gtest",
         "//third_party/abseil-cpp/absl/flags:flag",
@@ -1318,7 +1317,7 @@ if (rtc_include_tests) {
       ]
     }
 
-    rtc_executable("neteq_pcmu_quality_test") {
+    rtc_test("neteq_pcmu_quality_test") {
       testonly = true
 
       sources = [ "neteq/test/neteq_pcmu_quality_test.cc" ]
@@ -1333,14 +1332,13 @@ if (rtc_include_tests) {
         "../../rtc_base:checks",
         "../../rtc_base:safe_conversions",
         "../../test:fileutils",
-        "../../test:test_main",
         "../../test:test_support",
         "//testing/gtest",
         "//third_party/abseil-cpp/absl/flags:flag",
       ]
     }
 
-    rtc_executable("neteq_pcm16b_quality_test") {
+    rtc_test("neteq_pcm16b_quality_test") {
       testonly = true
 
       sources = [ "neteq/test/neteq_pcm16b_quality_test.cc" ]
@@ -1355,7 +1353,6 @@ if (rtc_include_tests) {
         "../../rtc_base:checks",
         "../../rtc_base:safe_conversions",
         "../../test:fileutils",
-        "../../test:test_main",
         "../../test:test_support",
         "//testing/gtest",
         "//third_party/abseil-cpp/absl/flags:flag",
@@ -1380,7 +1377,7 @@ if (rtc_include_tests) {
   }
 
   if (!build_with_chromium) {
-    rtc_executable("webrtc_opus_fec_test") {
+    rtc_test("webrtc_opus_fec_test") {
       testonly = true
 
       sources = [ "codecs/opus/opus_fec_test.cc" ]
@@ -1389,7 +1386,6 @@ if (rtc_include_tests) {
         ":webrtc_opus",
         "../../common_audio",
         "../../test:fileutils",
-        "../../test:test_main",
         "../../test:test_support",
         "//testing/gtest",
       ]
diff --git a/modules/rtp_rtcp/BUILD.gn b/modules/rtp_rtcp/BUILD.gn
index 5eda096822..3915a10786 100644
--- a/modules/rtp_rtcp/BUILD.gn
+++ b/modules/rtp_rtcp/BUILD.gn
@@ -490,7 +490,7 @@ rtc_library("rtp_packetizer_av1_test_helper") {
 
 if (rtc_include_tests) {
   if (!build_with_chromium) {
-    rtc_executable("test_packet_masks_metrics") {
+    rtc_test("test_packet_masks_metrics") {
       testonly = true
 
       sources = [
@@ -503,7 +503,6 @@ if (rtc_include_tests) {
         "..:module_fec_api",
         "../../rtc_base:checks",
         "../../test:fileutils",
-        "../../test:test_main",
         "../../test:test_support",
         "//testing/gtest",
       ]
diff --git a/p2p/BUILD.gn b/p2p/BUILD.gn
index 0b11310ec5..0c97fa00a1 100644
--- a/p2p/BUILD.gn
+++ b/p2p/BUILD.gn
@@ -1235,7 +1235,7 @@ if (rtc_include_tests) {
       "//third_party/abseil-cpp/absl/strings:string_view",
     ]
   }
-  rtc_executable("dtls_ice_integration_bench") {
+  rtc_test("dtls_ice_integration_bench") {
     testonly = true
 
     sources = [
@@ -1283,7 +1283,6 @@ if (rtc_include_tests) {
       "../rtc_base:ssl_adapter",
       "../rtc_base:threading",
       "../test:create_test_field_trials",
-      "../test:test_main",
       "../test:test_support",
       "../test:wait_until",
       "//third_party/abseil-cpp/absl/algorithm:container",
diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
index b589a35ec1..89498c5804 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -1994,7 +1994,7 @@ rtc_library("cpu_info") {
 
 if (rtc_include_tests) {
   if (rtc_enable_google_benchmarks) {
-    rtc_library("base64_benchmark_temp") {
+    rtc_library("base64_benchmark") {
       testonly = true
       sources = [ "base64_benchmark.cc" ]
       deps = [
diff --git a/tools_webrtc/get_landmines.py b/tools_webrtc/get_landmines.py
index aa4edbefb0..65c29dbb4c 100755
--- a/tools_webrtc/get_landmines.py
+++ b/tools_webrtc/get_landmines.py
@@ -53,7 +53,7 @@ def print_landmines():  # pylint: disable=invalid-name
         print('Clobber because of libc++ issue - take 5 (crbug.com/1337238)')
         print('Clobber because of libc++ issue - take 6 (crbug.com/1337238)')
         print('Clobber because b/367066321')
-    if host_os() == 'mac':
+    if host_os() == 'mac':  # includes iOS
         print('Clobber due to iOS compile errors (crbug.com/694721)')
         print('Clobber to unblock https://codereview.webrtc.org/2709573003')
         print('Clobber to fix https://codereview.webrtc.org/2709573003 after '
@@ -69,6 +69,8 @@ def print_landmines():  # pylint: disable=invalid-name
         print('Switching rtc_executable to rtc_test')
         print('Lets clobber iOS due to signing issue b/396118151')
         print('Lets clobber iOS due to signing issue b/396118151 (2nd try)')
+        print('Lets clobber iOS due to rtc_executable -> rtc_test')
+        print('Lets clobber iOS due to rtc_executable -> rtc_test')
     if host_os() == 'android':
         print('Clobber due to Android "compile confirm no-op" errors.')
 

From 565bbdd51ceb39ec8e0ee251b72452499b47853e Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Fri, 6 Mar 2026 15:44:05 +0100
Subject: [PATCH 254/472] Replace ArrayView with std::span in rtc_base/

Search&Replace MakeArrayView and ArrayView with std::span
Search&Replace include "api/array_view.h" with include <span>
Remove build dependencies on array_view target
Rename tests that were named PrefixArrayView
 from "Prefixstd::span" to "PrefixSpan"

Bug: webrtc:439801349
Change-Id: Iad951423733c8676c7356047f115eb3fcf2e885d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/454241
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47107}
---
 modules/audio_device/mac/audio_device_mac.cc |  7 +-
 modules/audio_device/mac/audio_device_mac.h  |  3 +-
 rtc_base/BUILD.gn                            | 27 +------
 rtc_base/async_tcp_socket.cc                 |  8 +-
 rtc_base/async_tcp_socket.h                  |  6 +-
 rtc_base/base64.h                            |  4 +-
 rtc_base/bit_buffer.cc                       |  4 +-
 rtc_base/bit_buffer.h                        |  5 +-
 rtc_base/bit_buffer_unittest.cc              |  4 +-
 rtc_base/bitstream_reader.h                  |  6 +-
 rtc_base/bitstream_reader_unittest.cc        | 12 +--
 rtc_base/buffer.h                            | 29 ++++---
 rtc_base/buffer_unittest.cc                  | 28 +++----
 rtc_base/byte_buffer.cc                      |  6 +-
 rtc_base/byte_buffer.h                       | 18 ++---
 rtc_base/byte_buffer_unittest.cc             | 46 +++++------
 rtc_base/byte_order.h                        | 30 ++++----
 rtc_base/byte_order_unittest.cc              |  6 +-
 rtc_base/copy_on_write_buffer.cc             |  6 +-
 rtc_base/copy_on_write_buffer.h              | 16 ++--
 rtc_base/copy_on_write_buffer_unittest.cc    |  6 +-
 rtc_base/memory/BUILD.gn                     |  2 -
 rtc_base/memory/fifo_buffer.cc               |  6 +-
 rtc_base/memory/fifo_buffer.h                |  6 +-
 rtc_base/memory/fifo_buffer_unittest.cc      | 80 ++++++++++----------
 rtc_base/memory_stream.cc                    |  6 +-
 rtc_base/memory_stream.h                     |  6 +-
 rtc_base/network.cc                          |  6 +-
 rtc_base/network.h                           |  4 +-
 rtc_base/network/BUILD.gn                    |  1 -
 rtc_base/network/received_packet.cc          |  6 +-
 rtc_base/network/received_packet.h           |  8 +-
 rtc_base/network_unittest.cc                 |  4 +-
 rtc_base/openssl_stream_adapter.cc           | 14 ++--
 rtc_base/openssl_stream_adapter.h            |  8 +-
 rtc_base/physical_socket_server.cc           |  4 +-
 rtc_base/server_socket_adapters.cc           |  6 +-
 rtc_base/socket.cc                           |  4 +-
 rtc_base/socket_adapters.cc                  |  8 +-
 rtc_base/socket_adapters.h                   |  6 +-
 rtc_base/ssl_certificate.cc                  |  4 +-
 rtc_base/ssl_fingerprint.cc                  | 10 +--
 rtc_base/ssl_fingerprint.h                   |  4 +-
 rtc_base/ssl_identity.cc                     |  4 +-
 rtc_base/ssl_stream_adapter.cc               |  4 +-
 rtc_base/ssl_stream_adapter.h                |  6 +-
 rtc_base/ssl_stream_adapter_unittest.cc      | 26 +++----
 rtc_base/stream.h                            |  6 +-
 rtc_base/string_encode.cc                    |  6 +-
 rtc_base/string_encode.h                     |  6 +-
 rtc_base/string_encode_unittest.cc           | 28 +++----
 rtc_base/strings/string_builder.cc           |  4 +-
 rtc_base/strings/string_builder.h            |  6 +-
 rtc_base/text2pcap.cc                        |  4 +-
 rtc_base/text2pcap.h                         |  5 +-
 rtc_base/unique_id_generator.cc              |  6 +-
 rtc_base/unique_id_generator.h               | 10 +--
 rtc_base/virtual_socket_unittest.cc          |  6 +-
 rtc_base/zero_memory.h                       |  5 +-
 rtc_base/zero_memory_unittest.cc             |  6 +-
 60 files changed, 293 insertions(+), 325 deletions(-)

diff --git a/modules/audio_device/mac/audio_device_mac.cc b/modules/audio_device/mac/audio_device_mac.cc
index 1ff98af0f4..e6e97bff53 100644
--- a/modules/audio_device/mac/audio_device_mac.cc
+++ b/modules/audio_device/mac/audio_device_mac.cc
@@ -15,6 +15,7 @@
 #include <sys/sysctl.h>  // sysctlbyname()
 
 #include <memory>
+#include <span>
 #include <vector>
 
 #include "modules/third_party/portaudio/pa_ringbuffer.h"
@@ -834,7 +835,7 @@ int32_t AudioDeviceMac::PlayoutDeviceName(uint16_t index,
   }
 
   return GetDeviceName(kAudioDevicePropertyScopeOutput, index,
-                       ArrayView<char>(name, kAdmMaxDeviceNameSize));
+                       std::span<char>(name, kAdmMaxDeviceNameSize));
 }
 
 int32_t AudioDeviceMac::RecordingDeviceName(uint16_t index,
@@ -853,7 +854,7 @@ int32_t AudioDeviceMac::RecordingDeviceName(uint16_t index,
   }
 
   return GetDeviceName(kAudioDevicePropertyScopeInput, index,
-                       ArrayView<char>(name, kAdmMaxDeviceNameSize));
+                       std::span<char>(name, kAdmMaxDeviceNameSize));
 }
 
 int16_t AudioDeviceMac::RecordingDevices() {
@@ -1644,7 +1645,7 @@ int32_t AudioDeviceMac::GetNumberDevices(const AudioObjectPropertyScope scope,
 
 int32_t AudioDeviceMac::GetDeviceName(const AudioObjectPropertyScope scope,
                                       const uint16_t index,
-                                      ArrayView<char> name) {
+                                      std::span<char> name) {
   OSStatus err = noErr;
   AudioDeviceID deviceIds[MaxNumberDevices];
 
diff --git a/modules/audio_device/mac/audio_device_mac.h b/modules/audio_device/mac/audio_device_mac.h
index c80de22b46..4598e1a5b1 100644
--- a/modules/audio_device/mac/audio_device_mac.h
+++ b/modules/audio_device/mac/audio_device_mac.h
@@ -17,6 +17,7 @@
 
 #include <atomic>
 #include <memory>
+#include <span>
 
 #include "absl/strings/string_view.h"
 #include "modules/audio_device/audio_device_generic.h"
@@ -178,7 +179,7 @@ class AudioDeviceMac : public AudioDeviceGeneric {
 
   int32_t GetDeviceName(AudioObjectPropertyScope scope,
                         uint16_t index,
-                        ArrayView<char> name);
+                        std::span<char> name);
 
   int32_t InitDevice(uint16_t userDeviceIndex,
                      AudioDeviceID& deviceId,
diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
index 89498c5804..59fea18a7b 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -29,7 +29,6 @@ rtc_library("bitstream_reader") {
   deps = [
     ":checks",
     ":safe_conversions",
-    "../api:array_view",
     "//third_party/abseil-cpp/absl/base:core_headers",
     "//third_party/abseil-cpp/absl/numeric:bits",
     "//third_party/abseil-cpp/absl/strings:string_view",
@@ -80,7 +79,6 @@ rtc_source_set("buffer") {
     ":checks",
     ":type_traits",
     ":zero_memory",
-    "../api:array_view",
     "//third_party/abseil-cpp/absl/algorithm:container",
     "//third_party/abseil-cpp/absl/base:core_headers",
     "//third_party/abseil-cpp/absl/strings:string_view",
@@ -92,7 +90,6 @@ rtc_source_set("byte_order") {
   sources = [ "byte_order.h" ]
   deps = [
     ":checks",
-    "../api:array_view",
     "system:arch",
     "//third_party/abseil-cpp/absl/base:core_headers",
   ]
@@ -143,7 +140,6 @@ rtc_library("bit_buffer") {
   ]
   deps = [
     ":checks",
-    "../api:array_view",
     "../api/units:data_size",
     "//third_party/abseil-cpp/absl/numeric:bits",
     "//third_party/abseil-cpp/absl/strings:string_view",
@@ -159,7 +155,6 @@ rtc_library("byte_buffer") {
   deps = [
     ":buffer",
     ":byte_order",
-    "../api:array_view",
     "//third_party/abseil-cpp/absl/base:core_headers",
     "//third_party/abseil-cpp/absl/strings:string_view",
   ]
@@ -190,7 +185,6 @@ rtc_library("copy_on_write_buffer") {
     ":checks",
     ":refcount",
     ":type_traits",
-    "../api:array_view",
     "../api:make_ref_counted",
     "../api:scoped_refptr",
     "system:rtc_export",
@@ -385,10 +379,7 @@ rtc_library("zero_memory") {
     "zero_memory.cc",
     "zero_memory.h",
   ]
-  deps = [
-    ":checks",
-    "../api:array_view",
-  ]
+  deps = [ ":checks" ]
 }
 
 rtc_library("platform_thread_types") {
@@ -643,7 +634,6 @@ rtc_library("stringutils") {
   deps = [
     ":checks",
     ":safe_minmax",
-    "../api:array_view",
     "//third_party/abseil-cpp/absl/base:core_headers",
     "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/strings:string_view",
@@ -660,7 +650,6 @@ rtc_library("text2pcap") {
     ":logging",
     ":stringutils",
     ":timeutils",
-    "../api:array_view",
     "system:rtc_export",
     "//third_party/abseil-cpp/absl/strings:string_view",
   ]
@@ -1056,7 +1045,6 @@ rtc_library("threading") {
     ":socket_address",
     ":socket_server",
     ":timeutils",
-    "../api:array_view",
     "../api:async_dns_resolver",
     "../api:function_view",
     "../api:location",
@@ -1124,7 +1112,6 @@ rtc_library("socket") {
     ":ip_address",
     ":net_helpers",
     ":socket_address",
-    "../api:array_view",
     "../api/transport:ecn_marking",
     "../api/units:timestamp",
     "system:rtc_export",
@@ -1273,7 +1260,6 @@ rtc_library("network") {
     ":socket_factory",
     ":stringutils",
     ":threading",
-    "../api:array_view",
     "../api:field_trials_view",
     "../api:scoped_refptr",
     "../api:sequence_checker",
@@ -1332,7 +1318,6 @@ rtc_library("socket_adapters") {
     ":socket_address",
     ":stringutils",
     ":zero_memory",
-    "../api:array_view",
     "//third_party/abseil-cpp/absl/functional:any_invocable",
     "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/strings:string_view",
@@ -1365,7 +1350,6 @@ rtc_library("async_tcp_socket") {
     ":socket",
     ":socket_address",
     ":timeutils",
-    "../api:array_view",
     "../api/environment",
     "../api/units:timestamp",
     "network:received_packet",
@@ -1522,7 +1506,6 @@ rtc_library("unique_id_generator") {
     ":crypto_random",
     ":macromagic",
     ":stringutils",
-    "../api:array_view",
     "../api:sequence_checker",
     "synchronization:mutex",
     "system:no_unique_address",
@@ -1551,7 +1534,6 @@ rtc_library("stream") {
     ":logging",
     ":macromagic",
     ":threading",
-    "../api:array_view",
     "../api:sequence_checker",
     "system:no_unique_address",
     "system:rtc_export",
@@ -1657,7 +1639,6 @@ rtc_library("ssl") {
     ":ssl_header",
     ":stringutils",
     ":timeutils",
-    "../api:array_view",
     "../api:refcountedbase",
     "../api:scoped_refptr",
     "system:rtc_export",
@@ -1733,7 +1714,6 @@ rtc_library("ssl_adapter") {
     ":stringutils",
     ":threading",
     ":timeutils",
-    "../api:array_view",
     "../api:field_trials_view",
     "../api:sequence_checker",
     "../api/task_queue",
@@ -1869,7 +1849,6 @@ rtc_library("rtc_base_tests_utils") {
     ":stringutils",
     ":threading",
     ":timeutils",
-    "../api:array_view",
     "../api:make_ref_counted",
     "../api:refcountedbase",
     "../api:scoped_refptr",
@@ -1942,7 +1921,6 @@ if (!rtc_rusty_base64) {
       "base64.h",
     ]
     deps = [
-      "../api:array_view",
       "//third_party/abseil-cpp/absl/algorithm:container",
       "//third_party/abseil-cpp/absl/strings",
       "//third_party/abseil-cpp/absl/strings:string_view",
@@ -1968,7 +1946,6 @@ if (!rtc_rusty_base64) {
     ]
     deps = [
       ":base64_rust_bridge",
-      "../api:array_view",
       "//build/rust:cxx_cppdeps",
       "//third_party/abseil-cpp/absl/strings:string_view",
     ]
@@ -2185,7 +2162,6 @@ if (rtc_include_tests) {
         ":timestamp_aligner",
         ":timeutils",
         ":zero_memory",
-        "../api:array_view",
         "../api:make_ref_counted",
         "../api:ref_count",
         "../api:scoped_refptr",
@@ -2359,7 +2335,6 @@ if (rtc_include_tests) {
         ":threading",
         ":timeutils",
         ":unique_id_generator",
-        "../api:array_view",
         "../api:field_trials",
         "../api:field_trials_view",
         "../api:location",
diff --git a/rtc_base/async_tcp_socket.cc b/rtc_base/async_tcp_socket.cc
index ad2867ed22..02aa98fffe 100644
--- a/rtc_base/async_tcp_socket.cc
+++ b/rtc_base/async_tcp_socket.cc
@@ -15,11 +15,11 @@
 #include <cstdint>
 #include <cstring>
 #include <memory>
+#include <span>
 #include <utility>
 
 #include "absl/base/nullability.h"
 #include "absl/memory/memory.h"
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "rtc_base/async_packet_socket.h"
 #include "rtc_base/byte_order.h"
@@ -127,7 +127,7 @@ int AsyncTCPSocketBase::SendTo(const void* pv,
 
 int AsyncTCPSocketBase::FlushOutBuffer() {
   RTC_DCHECK_GT(outbuf_.size(), 0);
-  ArrayView<uint8_t> view = outbuf_;
+  std::span<uint8_t> view = outbuf_;
   int res;
   while (!view.empty()) {
     res = socket_->Send(view.data(), view.size());
@@ -212,7 +212,7 @@ void AsyncTCPSocketBase::OnReadEvent(Socket* socket) {
   } else {
     if (bytes_remaining > 0) {
       // Move remaining bytes to beginning of buffer.
-      inbuf_.SetData(ArrayView<uint8_t>(inbuf_).subspan(processed));
+      inbuf_.SetData(std::span<uint8_t>(inbuf_).subspan(processed));
     } else {
       inbuf_.Clear();
     }
@@ -272,7 +272,7 @@ int AsyncTCPSocket::Send(const void* pv,
   return static_cast<int>(cb);
 }
 
-size_t AsyncTCPSocket::ProcessInput(ArrayView<const uint8_t> data) {
+size_t AsyncTCPSocket::ProcessInput(std::span<const uint8_t> data) {
   SocketAddress remote_addr(GetRemoteAddress());
 
   size_t processed_bytes = 0;
diff --git a/rtc_base/async_tcp_socket.h b/rtc_base/async_tcp_socket.h
index fdd5e3a9af..c937fdd08f 100644
--- a/rtc_base/async_tcp_socket.h
+++ b/rtc_base/async_tcp_socket.h
@@ -14,9 +14,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <memory>
+#include <span>
 
 #include "absl/base/nullability.h"
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "rtc_base/async_packet_socket.h"
 #include "rtc_base/buffer.h"
@@ -42,7 +42,7 @@ class AsyncTCPSocketBase : public AsyncPacketSocket {
            size_t cb,
            const AsyncSocketPacketOptions& options) override = 0;
   // Must return the number of bytes processed.
-  virtual size_t ProcessInput(ArrayView<const uint8_t> data) = 0;
+  virtual size_t ProcessInput(std::span<const uint8_t> data) = 0;
 
   SocketAddress GetLocalAddress() const override;
   SocketAddress GetRemoteAddress() const override;
@@ -92,7 +92,7 @@ class AsyncTCPSocket : public AsyncTCPSocketBase {
   int Send(const void* pv,
            size_t cb,
            const AsyncSocketPacketOptions& options) override;
-  size_t ProcessInput(ArrayView<const uint8_t>) override;
+  size_t ProcessInput(std::span<const uint8_t>) override;
 
  private:
   const Environment env_;
diff --git a/rtc_base/base64.h b/rtc_base/base64.h
index 9a83d8b07d..540fc9a3e2 100644
--- a/rtc_base/base64.h
+++ b/rtc_base/base64.h
@@ -13,16 +13,16 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 
 namespace webrtc {
 
 std::string Base64Encode(absl::string_view data);
 
-inline std::string Base64Encode(ArrayView<const uint8_t> data) {
+inline std::string Base64Encode(std::span<const uint8_t> data) {
   return Base64Encode(absl::string_view(
       reinterpret_cast<const char*>(data.data()), data.size()));
 }
diff --git a/rtc_base/bit_buffer.cc b/rtc_base/bit_buffer.cc
index 1b862e72db..ab2be82123 100644
--- a/rtc_base/bit_buffer.cc
+++ b/rtc_base/bit_buffer.cc
@@ -15,10 +15,10 @@
 #include <cstdint>
 #include <cstring>
 #include <limits>
+#include <span>
 
 #include "absl/numeric/bits.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "rtc_base/checks.h"
 
 namespace webrtc {
@@ -54,7 +54,7 @@ uint8_t WritePartialByte(uint8_t source,
 
 }  // namespace
 
-BitBufferWriter::BitBufferWriter(ArrayView<uint8_t> bytes)
+BitBufferWriter::BitBufferWriter(std::span<uint8_t> bytes)
     : BitBufferWriter(bytes.data(), bytes.size()) {}
 
 BitBufferWriter::BitBufferWriter(uint8_t* bytes, size_t byte_count)
diff --git a/rtc_base/bit_buffer.h b/rtc_base/bit_buffer.h
index 1c78662e18..575a7ef77e 100644
--- a/rtc_base/bit_buffer.h
+++ b/rtc_base/bit_buffer.h
@@ -14,8 +14,9 @@
 #include <stddef.h>  // For size_t.
 #include <stdint.h>  // For integer types.
 
+#include <span>
+
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/units/data_size.h"
 
 namespace webrtc {
@@ -29,7 +30,7 @@ class BitBufferWriter {
   static constexpr DataSize kMaxLeb128Length = DataSize::Bytes(10);
 
   // Constructs a bit buffer for the writable buffer of `bytes`.
-  explicit BitBufferWriter(ArrayView<uint8_t> bytes);
+  explicit BitBufferWriter(std::span<uint8_t> bytes);
   BitBufferWriter(uint8_t* bytes, size_t byte_count);
 
   BitBufferWriter(const BitBufferWriter&) = delete;
diff --git a/rtc_base/bit_buffer_unittest.cc b/rtc_base/bit_buffer_unittest.cc
index bca818c2b2..b607c4003c 100644
--- a/rtc_base/bit_buffer_unittest.cc
+++ b/rtc_base/bit_buffer_unittest.cc
@@ -13,8 +13,8 @@
 #include <cstddef>
 #include <cstdint>
 #include <limits>
+#include <span>
 
-#include "api/array_view.h"
 #include "rtc_base/bitstream_reader.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
@@ -166,7 +166,7 @@ TEST(BitBufferWriterTest, SymmetricReadWrite) {
   // That should be all that fits in the buffer.
   EXPECT_FALSE(buffer.WriteBits(1, 1));
 
-  BitstreamReader reader(MakeArrayView(bytes, 4));
+  BitstreamReader reader(std::span(bytes, 4));
   EXPECT_EQ(reader.ReadBits(3), 0x2u);
   EXPECT_EQ(reader.ReadBits(2), 0x1u);
   EXPECT_EQ(reader.ReadBits(7), 0x53u);
diff --git a/rtc_base/bitstream_reader.h b/rtc_base/bitstream_reader.h
index 622534ef70..5b9ee98292 100644
--- a/rtc_base/bitstream_reader.h
+++ b/rtc_base/bitstream_reader.h
@@ -13,12 +13,12 @@
 
 #include <stdint.h>
 
+#include <span>
 #include <string>
 #include <type_traits>
 
 #include "absl/base/attributes.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/numerics/safe_conversions.h"
 
@@ -33,7 +33,7 @@ namespace webrtc {
 class BitstreamReader {
  public:
   explicit BitstreamReader(
-      ArrayView<const uint8_t> bytes ABSL_ATTRIBUTE_LIFETIME_BOUND);
+      std::span<const uint8_t> bytes ABSL_ATTRIBUTE_LIFETIME_BOUND);
   explicit BitstreamReader(
       absl::string_view bytes ABSL_ATTRIBUTE_LIFETIME_BOUND);
   BitstreamReader(const BitstreamReader&) = default;
@@ -126,7 +126,7 @@ class BitstreamReader {
   mutable bool last_read_is_verified_ = true;
 };
 
-inline BitstreamReader::BitstreamReader(ArrayView<const uint8_t> bytes)
+inline BitstreamReader::BitstreamReader(std::span<const uint8_t> bytes)
     : bytes_(bytes.data()),
       remaining_bits_(checked_cast<int>(bytes.size() * 8)) {}
 
diff --git a/rtc_base/bitstream_reader_unittest.cc b/rtc_base/bitstream_reader_unittest.cc
index 4ad6624db4..f3f8f9a3c2 100644
--- a/rtc_base/bitstream_reader_unittest.cc
+++ b/rtc_base/bitstream_reader_unittest.cc
@@ -15,10 +15,10 @@
 #include <cstdint>
 #include <limits>
 #include <optional>
+#include <span>
 #include <utility>
 
 #include "absl/numeric/bits.h"
-#include "api/array_view.h"
 #include "rtc_base/checks.h"
 #include "test/gtest.h"
 
@@ -197,14 +197,14 @@ TEST(BitstreamReaderTest, ReadBits) {
 }
 
 TEST(BitstreamReaderTest, ReadZeroBits) {
-  BitstreamReader reader(ArrayView<const uint8_t>{});
+  BitstreamReader reader(std::span<const uint8_t>{});
 
   EXPECT_EQ(reader.ReadBits(0), 0u);
   EXPECT_TRUE(reader.Ok());
 }
 
 TEST(BitstreamReaderTest, ReadBitFromEmptyArray) {
-  BitstreamReader reader(ArrayView<const uint8_t>{});
+  BitstreamReader reader(std::span<const uint8_t>{});
 
   // Trying to read from the empty array shouldn't dereference the pointer,
   // i.e. shouldn't crash.
@@ -213,7 +213,7 @@ TEST(BitstreamReaderTest, ReadBitFromEmptyArray) {
 }
 
 TEST(BitstreamReaderTest, ReadBitsFromEmptyArray) {
-  BitstreamReader reader(ArrayView<const uint8_t>{});
+  BitstreamReader reader(std::span<const uint8_t>{});
 
   // Trying to read from the empty array shouldn't dereference the pointer,
   // i.e. shouldn't crash.
@@ -324,12 +324,12 @@ TEST(BitstreamReaderTest, NoGolombOverread) {
   const uint8_t bytes[] = {0x00, 0xFF, 0xFF};
   // Make sure the bit buffer correctly enforces byte length on golomb reads.
   // If it didn't, the above buffer would be valid at 3 bytes.
-  BitstreamReader reader1(MakeArrayView(bytes, 1));
+  BitstreamReader reader1(std::span(bytes, 1));
   // When parse fails, `ReadExponentialGolomb` may return any number.
   reader1.ReadExponentialGolomb();
   EXPECT_FALSE(reader1.Ok());
 
-  BitstreamReader reader2(MakeArrayView(bytes, 2));
+  BitstreamReader reader2(std::span(bytes, 2));
   reader2.ReadExponentialGolomb();
   EXPECT_FALSE(reader2.Ok());
 
diff --git a/rtc_base/buffer.h b/rtc_base/buffer.h
index f030f6b525..88b2ff76dd 100644
--- a/rtc_base/buffer.h
+++ b/rtc_base/buffer.h
@@ -22,7 +22,6 @@
 
 #include "absl/algorithm/container.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/type_traits.h"
 #include "rtc_base/zero_memory.h"
@@ -215,12 +214,12 @@ class BufferT {
 
   T& operator[](size_t index) {
     RTC_DCHECK_LT(index, size_);
-    return ArrayView<T>(*this)[index];
+    return std::span<T>(*this)[index];
   }
 
   T operator[](size_t index) const {
     RTC_DCHECK_LT(index, size_);
-    return ArrayView<const T>(*this)[index];
+    return std::span<const T>(*this)[index];
   }
 
   iterator begin() { return std::span(*this).begin(); }
@@ -263,12 +262,12 @@ class BufferT {
   // Replaces the data in the buffer with at most `max_elements` of data, using
   // the function `setter`, which should have the following signature:
   //
-  //   size_t setter(ArrayView<U> view)
+  //   size_t setter(std::span<U> view)
   //
-  // `setter` is given an appropriately typed ArrayView of length exactly
+  // `setter` is given an appropriately typed std::span of length exactly
   // `max_elements` that describes the area where it should write the data; it
   // should return the number of elements actually written. (If it doesn't fill
-  // the whole ArrayView, it should leave the unused space at the end.)
+  // the whole std::span, it should leave the unused space at the end.)
   template <typename U = T,
             typename F,
             typename std::enable_if<
@@ -298,9 +297,9 @@ class BufferT {
     const size_t new_size = size_ + size;
     EnsureCapacityWithHeadroom(new_size, true);
     static_assert(sizeof(T) == sizeof(U), "");
-    ArrayView<const U> source(data, size);
-    ArrayView<T> destination =
-        ArrayView<T>(data_.get(), capacity_).subspan(size_, size);
+    std::span<const U> source(data, size);
+    std::span<T> destination =
+        std::span<T>(data_.get(), capacity_).subspan(size_, size);
     absl::c_copy(source, destination.begin());
     size_ = new_size;
     RTC_DCHECK(IsConsistent());
@@ -331,12 +330,12 @@ class BufferT {
   // Appends at most `max_elements` to the end of the buffer, using the function
   // `setter`, which should have the following signature:
   //
-  //   size_t setter(ArrayView<U> view)
+  //   size_t setter(std::span<U> view)
   //
-  // `setter` is given an appropriately typed ArrayView of length exactly
+  // `setter` is given an appropriately typed std::span of length exactly
   // `max_elements` that describes the area where it should write the data; it
   // should return the number of elements actually written. (If it doesn't fill
-  // the whole ArrayView, it should leave the unused space at the end.)
+  // the whole std::span, it should leave the unused space at the end.)
   template <typename U = T,
             typename F,
             typename std::enable_if<
@@ -346,7 +345,7 @@ class BufferT {
     const size_t old_size = size_;
     SetSizeInternal(old_size + max_elements);
     size_t written_elements =
-        setter(ArrayView<U>(data<U>(), size()).subspan(old_size));
+        setter(std::span<U>(data<U>(), size()).subspan(old_size));
 
     RTC_CHECK_LE(written_elements, max_elements);
     size_ = old_size + written_elements;
@@ -448,7 +447,7 @@ class BufferT {
       // It would be sufficient to only zero "size_" elements, as all other
       // methods already ensure that the unused capacity contains no sensitive
       // data---but better safe than sorry.
-      ExplicitZeroMemory(ArrayView<T>(data_.get(), capacity_));
+      ExplicitZeroMemory(std::span<T>(data_.get(), capacity_));
     }
   }
 
@@ -456,7 +455,7 @@ class BufferT {
   void ZeroTrailingData(size_t count) {
     RTC_DCHECK(IsConsistent());
     RTC_DCHECK_LE(count, capacity_ - size_);
-    ExplicitZeroMemory(MakeArrayView(data(), capacity_).subspan(size_));
+    ExplicitZeroMemory(std::span(data(), capacity_).subspan(size_));
   }
 
   // Precondition for all methods except Clear, operator= and the destructor.
diff --git a/rtc_base/buffer_unittest.cc b/rtc_base/buffer_unittest.cc
index 58857f486c..4c226675db 100644
--- a/rtc_base/buffer_unittest.cc
+++ b/rtc_base/buffer_unittest.cc
@@ -14,10 +14,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <cstring>
+#include <span>
 #include <utility>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "rtc_base/checks.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
@@ -36,7 +36,7 @@ auto kTestData =
                             0xa, 0xb, 0xc, 0xd, 0xe, 0xf});
 
 uint8_t* TestDataAtIndex(int index) {
-  return ArrayView<uint8_t>(kTestData).subspan(index).data();
+  return std::span<uint8_t>(kTestData).subspan(index).data();
 }
 
 void TestBuf(const Buffer& b1, size_t size, size_t capacity) {
@@ -109,7 +109,7 @@ TEST(BufferTest, TestAppendData) {
   EXPECT_EQ(buf, Buffer(exp));
   Buffer buf2;
   buf2.AppendData(buf);
-  buf2.AppendData(ArrayView<uint8_t>(buf));
+  buf2.AppendData(std::span<uint8_t>(buf));
   const int8_t exp2[] = {0x4, 0x5, 0x6, 0xa, 0xb, 0x4, 0x5, 0x6, 0xa, 0xb};
   EXPECT_EQ(buf2, Buffer(exp2));
 }
@@ -127,7 +127,7 @@ TEST(BufferTest, TestSetAndAppendWithUnknownArg) {
   buf.AppendData(TestDataContainer());
   EXPECT_EQ(6u, buf.size());
   EXPECT_EQ(0, memcmp(buf.data(), kTestData.data(), 3));
-  EXPECT_EQ(0, memcmp(ArrayView<uint8_t>(buf).subspan(3).data(),
+  EXPECT_EQ(0, memcmp(std::span<uint8_t>(buf).subspan(3).data(),
                       kTestData.data(), 3));
   EXPECT_THAT(buf, ElementsAre(0, 1, 2, 0, 1, 2));
 }
@@ -250,7 +250,7 @@ TEST(BufferTest, TestClear) {
 }
 
 TEST(BufferTest, TestLambdaSetAppend) {
-  auto setter = [](ArrayView<uint8_t> av) {
+  auto setter = [](std::span<uint8_t> av) {
     for (int i = 0; i != 15; ++i)
       av[i] = kTestData[i];
     return 15;
@@ -270,7 +270,7 @@ TEST(BufferTest, TestLambdaSetAppend) {
 }
 
 TEST(BufferTest, TestLambdaSetAppendSigned) {
-  auto setter = [](ArrayView<int8_t> av) {
+  auto setter = [](std::span<int8_t> av) {
     for (int i = 0; i != 15; ++i)
       av[i] = kTestData[i];
     return 15;
@@ -290,7 +290,7 @@ TEST(BufferTest, TestLambdaSetAppendSigned) {
 }
 
 TEST(BufferTest, TestLambdaAppendEmpty) {
-  auto setter = [](ArrayView<uint8_t> av) {
+  auto setter = [](std::span<uint8_t> av) {
     for (int i = 0; i != 15; ++i)
       av[i] = kTestData[i];
     return 15;
@@ -308,7 +308,7 @@ TEST(BufferTest, TestLambdaAppendEmpty) {
 }
 
 TEST(BufferTest, TestLambdaAppendPartial) {
-  auto setter = [](ArrayView<uint8_t> av) {
+  auto setter = [](std::span<uint8_t> av) {
     for (int i = 0; i != 7; ++i)
       av[i] = kTestData[i];
     return 7;
@@ -324,7 +324,7 @@ TEST(BufferTest, TestLambdaAppendPartial) {
 
 TEST(BufferTest, TestMutableLambdaSetAppend) {
   uint8_t magic_number = 17;
-  auto setter = [magic_number](ArrayView<uint8_t> av) mutable {
+  auto setter = [magic_number](std::span<uint8_t> av) mutable {
     for (int i = 0; i != 15; ++i) {
       av[i] = magic_number;
       ++magic_number;
@@ -484,7 +484,7 @@ TEST(ZeroOnFreeBufferTest, TestZeroOnSetData) {
   const size_t old_capacity = buf.capacity();
   constexpr size_t offset = 1;
   // Pointer to the last five bytes of the underlying buffer.
-  auto to_be_zeroed = ArrayView<const uint8_t>(buf).subspan(2);
+  auto to_be_zeroed = std::span<const uint8_t>(buf).subspan(2);
   buf.SetData(TestDataAtIndex(offset), 2);
   // Sanity checks to make sure the underlying heap memory was not reallocated.
   EXPECT_EQ(old_data, buf.data());
@@ -498,7 +498,7 @@ TEST(ZeroOnFreeBufferTest, TestZeroOnSetData) {
 
 TEST(ZeroOnFreeBufferTest, TestZeroOnSetDataFromSetter) {
   static constexpr size_t offset = 1;
-  const auto setter = [](ArrayView<uint8_t> av) {
+  const auto setter = [](std::span<uint8_t> av) {
     for (int i = 0; i != 2; ++i)
       av[i] = kTestData[offset + i];
     return 2;
@@ -506,7 +506,7 @@ TEST(ZeroOnFreeBufferTest, TestZeroOnSetDataFromSetter) {
 
   ZeroOnFreeBuffer<uint8_t> buf(kTestData.data(), 7);
   const uint8_t* old_data = buf.data();
-  auto to_be_zeroed = ArrayView<const uint8_t>(buf).subspan(2);
+  auto to_be_zeroed = std::span<const uint8_t>(buf).subspan(2);
   const size_t old_capacity = buf.capacity();
   buf.SetData(2, setter);
   // Sanity checks to make sure the underlying heap memory was not reallocated.
@@ -521,7 +521,7 @@ TEST(ZeroOnFreeBufferTest, TestZeroOnSetDataFromSetter) {
 
 TEST(ZeroOnFreeBufferTest, TestZeroOnSetSize) {
   ZeroOnFreeBuffer<uint8_t> buf(kTestData.data(), 7);
-  auto to_be_zeroed = ArrayView<const uint8_t>(buf).subspan(2);
+  auto to_be_zeroed = std::span<const uint8_t>(buf).subspan(2);
   const uint8_t* old_data = buf.data();
   const size_t old_capacity = buf.capacity();
   buf.SetSize(2);
@@ -537,7 +537,7 @@ TEST(ZeroOnFreeBufferTest, TestZeroOnSetSize) {
 
 TEST(ZeroOnFreeBufferTest, TestZeroOnClear) {
   ZeroOnFreeBuffer<uint8_t> buf(kTestData.data(), 7);
-  auto to_be_zeroed = ArrayView<const uint8_t>(buf);
+  auto to_be_zeroed = std::span<const uint8_t>(buf);
   const uint8_t* old_data = buf.data();
   const size_t old_capacity = buf.capacity();
   buf.Clear();
diff --git a/rtc_base/byte_buffer.cc b/rtc_base/byte_buffer.cc
index 719720d777..b3ebb1ec8a 100644
--- a/rtc_base/byte_buffer.cc
+++ b/rtc_base/byte_buffer.cc
@@ -12,10 +12,10 @@
 
 #include <cstdint>
 #include <cstring>
+#include <span>
 #include <string>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "rtc_base/byte_order.h"
 
 namespace webrtc {
@@ -25,7 +25,7 @@ ByteBufferWriter::ByteBufferWriter() : ByteBufferWriterT() {}
 ByteBufferWriter::ByteBufferWriter(const uint8_t* bytes, size_t len)
     : ByteBufferWriterT(bytes, len) {}
 
-ByteBufferReader::ByteBufferReader(ArrayView<const uint8_t> bytes) {
+ByteBufferReader::ByteBufferReader(std::span<const uint8_t> bytes) {
   Construct(bytes.data(), bytes.size());
 }
 
@@ -146,7 +146,7 @@ bool ByteBufferReader::ReadStringView(absl::string_view* val, size_t len) {
   return true;
 }
 
-bool ByteBufferReader::ReadBytes(ArrayView<uint8_t> val) {
+bool ByteBufferReader::ReadBytes(std::span<uint8_t> val) {
   if (val.empty()) {
     return true;
   }
diff --git a/rtc_base/byte_buffer.h b/rtc_base/byte_buffer.h
index cc7d6487a8..64648d0a01 100644
--- a/rtc_base/byte_buffer.h
+++ b/rtc_base/byte_buffer.h
@@ -14,11 +14,11 @@
 #include <stddef.h>
 #include <stdint.h>
 
+#include <span>
 #include <string>
 
 #include "absl/base/attributes.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/byte_order.h"
 
@@ -41,8 +41,8 @@ class ByteBufferWriterT {
   const value_type* Data() const { return buffer_.data(); }
   size_t Length() const { return buffer_.size(); }
   size_t Capacity() const { return buffer_.capacity(); }
-  ArrayView<const value_type> DataView() const {
-    return MakeArrayView(Data(), Length());
+  std::span<const value_type> DataView() const {
+    return std::span(Data(), Length());
   }
   // Accessor that returns a string_view, independent of underlying type.
   // Intended to provide access for existing users that expect char*
@@ -97,12 +97,12 @@ class ByteBufferWriterT {
                        val.size());
   }
   // Write an array of bytes (uint8_t)
-  [[deprecated("issues.webrtc.org/4225170 - use Write(ArrayView)")]]
+  [[deprecated("issues.webrtc.org/4225170 - use Write(std::span)")]]
   void WriteBytes(const uint8_t* val, size_t len) {
     WriteBytesInternal(reinterpret_cast<const value_type*>(val), len);
   }
 
-  void Write(ArrayView<const value_type> data) {
+  void Write(std::span<const value_type> data) {
     WriteBytesInternal(data.data(), data.size());
   }
 
@@ -157,7 +157,7 @@ class ByteBufferWriter : public ByteBufferWriterT<BufferT<uint8_t>> {
 class ByteBufferReader {
  public:
   explicit ByteBufferReader(
-      ArrayView<const uint8_t> bytes ABSL_ATTRIBUTE_LIFETIME_BOUND);
+      std::span<const uint8_t> bytes ABSL_ATTRIBUTE_LIFETIME_BOUND);
 
   explicit ByteBufferReader(const ByteBufferWriter& buf);
 
@@ -168,8 +168,8 @@ class ByteBufferReader {
   // Returns number of unprocessed bytes.
   size_t Length() const { return end_ - start_; }
   // Returns a view of the unprocessed data. Does not move current position.
-  ArrayView<const uint8_t> DataView() const {
-    return ArrayView<const uint8_t>(bytes_ + start_, end_ - start_);
+  std::span<const uint8_t> DataView() const {
+    return std::span<const uint8_t>(bytes_ + start_, end_ - start_);
   }
 
   // Read a next value from the buffer. Return false if there isn't
@@ -181,7 +181,7 @@ class ByteBufferReader {
   bool ReadUInt64(uint64_t* val);
   bool ReadUVarint(uint64_t* val);
   // Copies the val.size() next bytes into val.data().
-  bool ReadBytes(ArrayView<uint8_t> val);
+  bool ReadBytes(std::span<uint8_t> val);
   // Appends next `len` bytes from the buffer to `val`. Returns false
   // if there is less than `len` bytes left.
   bool ReadString(std::string* val, size_t len);
diff --git a/rtc_base/byte_buffer_unittest.cc b/rtc_base/byte_buffer_unittest.cc
index 267f107ee6..cdc34bed4a 100644
--- a/rtc_base/byte_buffer_unittest.cc
+++ b/rtc_base/byte_buffer_unittest.cc
@@ -12,11 +12,11 @@
 
 #include <cstdint>
 #include <cstring>
+#include <span>
 #include <string>
 #include <utility>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/byte_order.h"
 #include "test/gmock.h"
@@ -56,11 +56,11 @@ TEST(ByteBufferTest, TestByteOrder) {
     EXPECT_EQ(n64, HostToNetwork64(n64));
 
     // GetBE converts big endian to little endian here.
-    EXPECT_EQ(n16 >> 8, GetBE16(ArrayView<const uint8_t>(
+    EXPECT_EQ(n16 >> 8, GetBE16(std::span<const uint8_t>(
                             reinterpret_cast<const uint8_t*>(&n16), 2)));
-    EXPECT_EQ(n32 >> 24, GetBE32(ArrayView<const uint8_t>(
+    EXPECT_EQ(n32 >> 24, GetBE32(std::span<const uint8_t>(
                              reinterpret_cast<const uint8_t*>(&n32), 4)));
-    EXPECT_EQ(n64 >> 56, GetBE64(ArrayView<const uint8_t>(
+    EXPECT_EQ(n64 >> 56, GetBE64(std::span<const uint8_t>(
                              reinterpret_cast<const uint8_t*>(&n64), 8)));
   } else {
     // The host is little endian.
@@ -69,22 +69,22 @@ TEST(ByteBufferTest, TestByteOrder) {
     EXPECT_NE(n64, HostToNetwork64(n64));
 
     // GetBE converts little endian to big endian here.
-    EXPECT_EQ(GetBE16(ArrayView<const uint8_t>(
+    EXPECT_EQ(GetBE16(std::span<const uint8_t>(
                   reinterpret_cast<const uint8_t*>(&n16), 2)),
               HostToNetwork16(n16));
-    EXPECT_EQ(GetBE32(ArrayView<const uint8_t>(
+    EXPECT_EQ(GetBE32(std::span<const uint8_t>(
                   reinterpret_cast<const uint8_t*>(&n32), 4)),
               HostToNetwork32(n32));
-    EXPECT_EQ(GetBE64(ArrayView<const uint8_t>(
+    EXPECT_EQ(GetBE64(std::span<const uint8_t>(
                   reinterpret_cast<const uint8_t*>(&n64), 8)),
               HostToNetwork64(n64));
 
     // GetBE converts little endian to big endian here.
-    EXPECT_EQ(n16 << 8, GetBE16(ArrayView<const uint8_t>(
+    EXPECT_EQ(n16 << 8, GetBE16(std::span<const uint8_t>(
                             reinterpret_cast<const uint8_t*>(&n16), 2)));
-    EXPECT_EQ(n32 << 24, GetBE32(ArrayView<const uint8_t>(
+    EXPECT_EQ(n32 << 24, GetBE32(std::span<const uint8_t>(
                              reinterpret_cast<const uint8_t*>(&n32), 4)));
-    EXPECT_EQ(n64 << 56, GetBE64(ArrayView<const uint8_t>(
+    EXPECT_EQ(n64 << 56, GetBE64(std::span<const uint8_t>(
                              reinterpret_cast<const uint8_t*>(&n64), 8)));
   }
 }
@@ -117,7 +117,7 @@ TEST(ByteBufferTest, TestBufferLength) {
 
 TEST(ByteBufferTest, TestReadWriteBuffer) {
   ByteBufferWriter buffer;
-  ByteBufferReader read_buf(ArrayView<const uint8_t>{});
+  ByteBufferReader read_buf(std::span<const uint8_t>{});
   uint8_t ru8;
   EXPECT_FALSE(read_buf.ReadUInt8(&ru8));
 
@@ -183,7 +183,7 @@ TEST(ByteBufferTest, TestReadWriteBuffer) {
 
   // Write and read bytes
   uint8_t write_bytes[] = {3, 2, 1};
-  buffer.Write(ArrayView<const uint8_t>(write_bytes, 3));
+  buffer.Write(std::span<const uint8_t>(write_bytes, 3));
   ByteBufferReader read_buf7(buffer);
   uint8_t read_bytes[3];
   EXPECT_TRUE(read_buf7.ReadBytes(read_bytes));
@@ -260,10 +260,10 @@ TEST(ByteBufferTest, TestWriteBuffer) {
   EXPECT_EQ(read_buf11.Length(), 0U);
 }
 
-TEST(ByteBufferTest, TestWriteArrayView) {
+TEST(ByteBufferTest, TestWriteSpan) {
   const uint8_t write_data[3] = {3, 2, 1};
-  // Write and read arrayview
-  ArrayView<const uint8_t> write_view(write_data);
+  // Write and read std::span
+  std::span<const uint8_t> write_view(write_data);
   ByteBufferWriter buffer;
   buffer.Write(write_view);
   ByteBufferReader read_buf12(buffer);
@@ -288,7 +288,7 @@ TEST(ByteBufferTest, TestReadStringView) {
   for (const auto& test : tests)
     buffer += test;
 
-  ArrayView<const uint8_t> bytes(reinterpret_cast<const uint8_t*>(&buffer[0]),
+  std::span<const uint8_t> bytes(reinterpret_cast<const uint8_t*>(&buffer[0]),
                                  buffer.size());
 
   ByteBufferReader read_buf(bytes);
@@ -359,9 +359,9 @@ TEST(ByteBufferTest, TestReadWriteUVarint) {
   EXPECT_EQ(size, read_buffer.Length());
 }
 
-TEST(ByteBufferTest, ReadFromArrayView) {
+TEST(ByteBufferTest, ReadFromSpan) {
   const uint8_t buf[] = {'a', 'b', 'c'};
-  ArrayView<const uint8_t> view(buf, 3);
+  std::span<const uint8_t> view(buf, 3);
 
   ByteBufferReader read_buffer(view);
   uint8_t val;
@@ -374,18 +374,18 @@ TEST(ByteBufferTest, ReadFromArrayView) {
   EXPECT_FALSE(read_buffer.ReadUInt8(&val));
 }
 
-TEST(ByteBufferTest, ReadToArrayView) {
+TEST(ByteBufferTest, ReadToSpan) {
   const uint8_t buf[] = {'a', 'b', 'c'};
-  ArrayView<const uint8_t> stored_view(buf, 3);
+  std::span<const uint8_t> stored_view(buf, 3);
   ByteBufferReader read_buffer(stored_view);
   uint8_t result[] = {'1', '2', '3'};
-  EXPECT_TRUE(read_buffer.ReadBytes(MakeArrayView(result, 2)));
+  EXPECT_TRUE(read_buffer.ReadBytes(std::span(result, 2)));
   EXPECT_EQ(result[0], 'a');
   EXPECT_EQ(result[1], 'b');
   EXPECT_EQ(result[2], '3');
-  EXPECT_TRUE(read_buffer.ReadBytes(MakeArrayView(&result[2], 1)));
+  EXPECT_TRUE(read_buffer.ReadBytes(std::span(&result[2], 1)));
   EXPECT_EQ(result[2], 'c');
-  EXPECT_FALSE(read_buffer.ReadBytes(MakeArrayView(result, 1)));
+  EXPECT_FALSE(read_buffer.ReadBytes(std::span(result, 1)));
 }
 
 }  // namespace webrtc
diff --git a/rtc_base/byte_order.h b/rtc_base/byte_order.h
index 8a31c6307f..bf44c450e6 100644
--- a/rtc_base/byte_order.h
+++ b/rtc_base/byte_order.h
@@ -14,8 +14,8 @@
 #include <cstdint>
 #include <cstdlib>
 #include <cstring>
+#include <span>
 
-#include "api/array_view.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/system/arch.h"  // IWYU pragma: keep
 
@@ -94,86 +94,86 @@ namespace webrtc {
 
 // Reading and writing of little and big-endian numbers from memory
 
-inline void Set8(ArrayView<uint8_t> data, size_t offset, uint8_t v) {
+inline void Set8(std::span<uint8_t> data, size_t offset, uint8_t v) {
   data[offset] = v;
 }
 
-inline uint8_t Get8(ArrayView<const uint8_t> data, size_t offset) {
+inline uint8_t Get8(std::span<const uint8_t> data, size_t offset) {
   return data[offset];
 }
 
-inline void SetBE16(ArrayView<uint8_t> data, uint16_t v) {
+inline void SetBE16(std::span<uint8_t> data, uint16_t v) {
   RTC_DCHECK_GE(data.size(), sizeof(v));
   uint16_t val = htobe16(v);
   memcpy(data.data(), &val, sizeof(val));
 }
 
-inline void SetBE32(ArrayView<uint8_t> data, uint32_t v) {
+inline void SetBE32(std::span<uint8_t> data, uint32_t v) {
   RTC_DCHECK_GE(data.size(), sizeof(v));
   uint32_t val = htobe32(v);
   memcpy(data.data(), &val, sizeof(val));
 }
 
-inline void SetBE64(ArrayView<uint8_t> data, uint64_t v) {
+inline void SetBE64(std::span<uint8_t> data, uint64_t v) {
   RTC_DCHECK_GE(data.size(), sizeof(v));
   uint64_t val = htobe64(v);
   memcpy(data.data(), &val, sizeof(val));
 }
 
-inline uint16_t GetBE16(ArrayView<const uint8_t> data) {
+inline uint16_t GetBE16(std::span<const uint8_t> data) {
   uint16_t val;
   RTC_DCHECK_GE(data.size(), sizeof(val));
   memcpy(&val, data.data(), sizeof(val));
   return be16toh(val);
 }
 
-inline uint32_t GetBE32(ArrayView<const uint8_t> data) {
+inline uint32_t GetBE32(std::span<const uint8_t> data) {
   uint32_t val;
   RTC_DCHECK_GE(data.size(), sizeof(val));
   memcpy(&val, data.data(), sizeof(val));
   return be32toh(val);
 }
 
-inline uint64_t GetBE64(ArrayView<const uint8_t> data) {
+inline uint64_t GetBE64(std::span<const uint8_t> data) {
   uint64_t val;
   RTC_DCHECK_GE(data.size(), sizeof(val));
   memcpy(&val, data.data(), sizeof(val));
   return be64toh(val);
 }
 
-inline void SetLE16(ArrayView<uint8_t> data, uint16_t v) {
+inline void SetLE16(std::span<uint8_t> data, uint16_t v) {
   RTC_DCHECK_GE(data.size(), sizeof(v));
   uint16_t val = htole16(v);
   memcpy(data.data(), &val, sizeof(val));
 }
 
-inline void SetLE32(ArrayView<uint8_t> data, uint32_t v) {
+inline void SetLE32(std::span<uint8_t> data, uint32_t v) {
   RTC_DCHECK_GE(data.size(), sizeof(v));
   uint32_t val = htole32(v);
   memcpy(data.data(), &val, sizeof(val));
 }
 
-inline void SetLE64(ArrayView<uint8_t> data, uint64_t v) {
+inline void SetLE64(std::span<uint8_t> data, uint64_t v) {
   RTC_DCHECK_GE(data.size(), sizeof(v));
   uint64_t val = htole64(v);
   memcpy(data.data(), &val, sizeof(val));
 }
 
-inline uint16_t GetLE16(ArrayView<const uint8_t> data) {
+inline uint16_t GetLE16(std::span<const uint8_t> data) {
   uint16_t val;
   RTC_DCHECK_GE(data.size(), sizeof(val));
   memcpy(&val, data.data(), sizeof(val));
   return le16toh(val);
 }
 
-inline uint32_t GetLE32(ArrayView<const uint8_t> data) {
+inline uint32_t GetLE32(std::span<const uint8_t> data) {
   uint32_t val;
   RTC_DCHECK_GE(data.size(), sizeof(val));
   memcpy(&val, data.data(), sizeof(val));
   return le32toh(val);
 }
 
-inline uint64_t GetLE64(ArrayView<const uint8_t> data) {
+inline uint64_t GetLE64(std::span<const uint8_t> data) {
   uint64_t val;
   RTC_DCHECK_GE(data.size(), sizeof(val));
   memcpy(&val, data.data(), sizeof(val));
diff --git a/rtc_base/byte_order_unittest.cc b/rtc_base/byte_order_unittest.cc
index 75a370c7b8..893c0e734b 100644
--- a/rtc_base/byte_order_unittest.cc
+++ b/rtc_base/byte_order_unittest.cc
@@ -13,8 +13,8 @@
 #include <stdint.h>
 
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "test/gtest.h"
 
 namespace webrtc {
@@ -22,7 +22,7 @@ namespace webrtc {
 // Test memory set functions put values into memory in expected order.
 TEST(ByteOrderTest, TestSet) {
   uint8_t buf[8] = {0u, 0u, 0u, 0u, 0u, 0u, 0u, 0u};
-  ArrayView<uint8_t> av(buf);
+  std::span<uint8_t> av(buf);
   Set8(av, 0, 0xfb);
   Set8(av, 1, 0x12);
   EXPECT_EQ(0xfb, buf[0]);
@@ -66,7 +66,7 @@ TEST(ByteOrderTest, TestSet) {
 // Test memory get functions get values from memory in expected order.
 TEST(ByteOrderTest, TestGet) {
   uint8_t buf[8];
-  ArrayView<const uint8_t> av(buf);
+  std::span<const uint8_t> av(buf);
   buf[0] = 0x01u;
   buf[1] = 0x23u;
   buf[2] = 0x45u;
diff --git a/rtc_base/copy_on_write_buffer.cc b/rtc_base/copy_on_write_buffer.cc
index be1fc302a6..68e98ce729 100644
--- a/rtc_base/copy_on_write_buffer.cc
+++ b/rtc_base/copy_on_write_buffer.cc
@@ -15,11 +15,11 @@
 #include <cstdint>
 #include <cstring>
 #include <memory>
+#include <span>
 #include <utility>
 
 #include "absl/algorithm/container.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/make_ref_counted.h"
 #include "api/scoped_refptr.h"
 #include "rtc_base/checks.h"
@@ -110,7 +110,7 @@ void CopyOnWriteBuffer::Clear() {
   RTC_DCHECK(IsConsistent());
 }
 
-void CopyOnWriteBuffer::Set(ArrayView<const uint8_t> data) {
+void CopyOnWriteBuffer::Set(std::span<const uint8_t> data) {
   RTC_DCHECK(IsConsistent());
   if (data.empty()) {
     offset_ = 0;
@@ -129,7 +129,7 @@ void CopyOnWriteBuffer::Set(ArrayView<const uint8_t> data) {
   RTC_DCHECK(IsConsistent());
 }
 
-void CopyOnWriteBuffer::Append(ArrayView<const uint8_t> data) {
+void CopyOnWriteBuffer::Append(std::span<const uint8_t> data) {
   RTC_DCHECK(IsConsistent());
   if (data.empty()) {
     return;
diff --git a/rtc_base/copy_on_write_buffer.h b/rtc_base/copy_on_write_buffer.h
index 14554ed2e7..cda92f1e86 100644
--- a/rtc_base/copy_on_write_buffer.h
+++ b/rtc_base/copy_on_write_buffer.h
@@ -14,11 +14,11 @@
 #include <cstddef>
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <type_traits>
 #include <utility>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/scoped_refptr.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/checks.h"
@@ -30,7 +30,7 @@ namespace webrtc {
 
 class RTC_EXPORT CopyOnWriteBuffer {
  public:
-  using const_iterator = ArrayView<const uint8_t>::iterator;
+  using const_iterator = std::span<const uint8_t>::iterator;
 
   // An empty buffer.
   CopyOnWriteBuffer();
@@ -163,7 +163,7 @@ class RTC_EXPORT CopyOnWriteBuffer {
             typename std::enable_if<
                 internal::BufferCompat<uint8_t, T>::value>::type* = nullptr>
   void SetData(const T* data, size_t size) {
-    Set(MakeArrayView(reinterpret_cast<const uint8_t*>(data), size));
+    Set(std::span(reinterpret_cast<const uint8_t*>(data), size));
   }
 
   template <typename T,
@@ -181,7 +181,7 @@ class RTC_EXPORT CopyOnWriteBuffer {
             typename std::enable_if<
                 internal::BufferCompat<uint8_t, T>::value>::type* = nullptr>
   void AppendData(const T* data, size_t size) {
-    Append(MakeArrayView(reinterpret_cast<const uint8_t*>(data), size));
+    Append(std::span(reinterpret_cast<const uint8_t*>(data), size));
   }
 
   template <typename T,
@@ -238,7 +238,7 @@ class RTC_EXPORT CopyOnWriteBuffer {
    public:
     explicit RawBuffer(size_t size);
 
-    ArrayView<uint8_t> data() { return MakeArrayView(data_.get(), size_); }
+    std::span<uint8_t> data() { return std::span(data_.get(), size_); }
     size_t capacity() const { return size_; }
 
    private:
@@ -249,7 +249,7 @@ class RTC_EXPORT CopyOnWriteBuffer {
 
   static scoped_refptr<RefCountedBuffer> CreateBuffer(size_t capacity);
 
-  ArrayView<const uint8_t> AsConstSpan() const {
+  std::span<const uint8_t> AsConstSpan() const {
     RTC_DCHECK(IsConsistent());
     if (buffer_ == nullptr) {
       return {};
@@ -257,8 +257,8 @@ class RTC_EXPORT CopyOnWriteBuffer {
     return buffer_->data().subspan(offset_, size_);
   }
 
-  void Set(ArrayView<const uint8_t> buffer);
-  void Append(ArrayView<const uint8_t> buffer);
+  void Set(std::span<const uint8_t> buffer);
+  void Append(std::span<const uint8_t> buffer);
 
   // Create a copy of the underlying data if it is referenced from other Buffer
   // objects or there is not enough capacity.
diff --git a/rtc_base/copy_on_write_buffer_unittest.cc b/rtc_base/copy_on_write_buffer_unittest.cc
index 59247e5762..e42d986dcb 100644
--- a/rtc_base/copy_on_write_buffer_unittest.cc
+++ b/rtc_base/copy_on_write_buffer_unittest.cc
@@ -13,11 +13,11 @@
 #include <cstddef>
 #include <cstdint>
 #include <cstring>
+#include <span>
 #include <type_traits>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "test/gtest.h"
 
 namespace webrtc {
@@ -374,8 +374,8 @@ TEST(CopyOnWriteBufferTest, SlicesAreIndependent) {
 TEST(CopyOnWriteBufferTest, AcceptsVectorLikeTypes) {
   std::vector<uint8_t> a = {1, 2};
   std::vector<int8_t> b = {3, 4};
-  ArrayView<uint8_t> c(a);
-  ArrayView<const int8_t> d(b);
+  std::span<uint8_t> c(a);
+  std::span<const int8_t> d(b);
 
   CopyOnWriteBuffer a_buf(a);
   CopyOnWriteBuffer b_buf(b);
diff --git a/rtc_base/memory/BUILD.gn b/rtc_base/memory/BUILD.gn
index 8162e3a665..60da01950a 100644
--- a/rtc_base/memory/BUILD.gn
+++ b/rtc_base/memory/BUILD.gn
@@ -36,7 +36,6 @@ rtc_library("fifo_buffer") {
     "..:checks",
     "..:macromagic",
     "..:stream",
-    "../../api:array_view",
     "../../api:sequence_checker",
     "../../api/task_queue",
     "../../api/task_queue:pending_task_safety_flag",
@@ -60,7 +59,6 @@ rtc_library("unittests") {
     ":fifo_buffer",
     "..:stream",
     "..:threading",
-    "../../api:array_view",
     "../../test:test_support",
   ]
 }
diff --git a/rtc_base/memory/fifo_buffer.cc b/rtc_base/memory/fifo_buffer.cc
index faebc928f3..34f0fef488 100644
--- a/rtc_base/memory/fifo_buffer.cc
+++ b/rtc_base/memory/fifo_buffer.cc
@@ -14,8 +14,8 @@
 #include <cstddef>
 #include <cstdint>
 #include <cstring>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/sequence_checker.h"
 #include "api/task_queue/task_queue_base.h"
 #include "rtc_base/checks.h"
@@ -56,7 +56,7 @@ StreamState FifoBuffer::GetState() const {
   return state_;
 }
 
-StreamResult FifoBuffer::Read(ArrayView<uint8_t> buffer,
+StreamResult FifoBuffer::Read(std::span<uint8_t> buffer,
                               size_t& bytes_read,
                               int& error) {
   RTC_DCHECK_RUN_ON(&callback_sequence_);
@@ -79,7 +79,7 @@ StreamResult FifoBuffer::Read(ArrayView<uint8_t> buffer,
   return result;
 }
 
-StreamResult FifoBuffer::Write(ArrayView<const uint8_t> buffer,
+StreamResult FifoBuffer::Write(std::span<const uint8_t> buffer,
                                size_t& bytes_written,
                                int& error) {
   RTC_DCHECK_RUN_ON(&callback_sequence_);
diff --git a/rtc_base/memory/fifo_buffer.h b/rtc_base/memory/fifo_buffer.h
index 70fee304b8..0f0de40939 100644
--- a/rtc_base/memory/fifo_buffer.h
+++ b/rtc_base/memory/fifo_buffer.h
@@ -14,8 +14,8 @@
 #include <cstddef>
 #include <cstdint>
 #include <memory>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/sequence_checker.h"
 #include "api/task_queue/pending_task_safety_flag.h"
 #include "api/task_queue/task_queue_base.h"
@@ -42,10 +42,10 @@ class FifoBuffer final : public StreamInterface {
 
   // StreamInterface methods
   StreamState GetState() const override;
-  StreamResult Read(ArrayView<uint8_t> buffer,
+  StreamResult Read(std::span<uint8_t> buffer,
                     size_t& bytes_read,
                     int& error) override;
-  StreamResult Write(ArrayView<const uint8_t> buffer,
+  StreamResult Write(std::span<const uint8_t> buffer,
                      size_t& bytes_written,
                      int& error) override;
   void Close() override;
diff --git a/rtc_base/memory/fifo_buffer_unittest.cc b/rtc_base/memory/fifo_buffer_unittest.cc
index 22f78a4461..e5f2070449 100644
--- a/rtc_base/memory/fifo_buffer_unittest.cc
+++ b/rtc_base/memory/fifo_buffer_unittest.cc
@@ -12,8 +12,8 @@
 
 #include <cstdint>
 #include <cstring>
+#include <span>
 
-#include "api/array_view.h"
 #include "rtc_base/stream.h"
 #include "rtc_base/thread.h"
 #include "test/gtest.h"
@@ -33,49 +33,49 @@ TEST(FifoBufferTest, TestAll) {
   // Test assumptions about base state
   EXPECT_EQ(SS_OPEN, buf.GetState());
   int error;
-  EXPECT_EQ(SR_BLOCK, buf.Read(MakeArrayView(out, kSize), bytes, error));
+  EXPECT_EQ(SR_BLOCK, buf.Read(std::span(out, kSize), bytes, error));
   EXPECT_TRUE(nullptr != buf.GetWriteBuffer(&bytes));
   EXPECT_EQ(kSize, bytes);
   buf.ConsumeWriteBuffer(0);
 
   // Try a full write
-  EXPECT_EQ(SR_SUCCESS, buf.Write(MakeArrayView(in, kSize), bytes, error));
+  EXPECT_EQ(SR_SUCCESS, buf.Write(std::span(in, kSize), bytes, error));
   EXPECT_EQ(kSize, bytes);
 
   // Try a write that should block
-  EXPECT_EQ(SR_BLOCK, buf.Write(MakeArrayView(in, kSize), bytes, error));
+  EXPECT_EQ(SR_BLOCK, buf.Write(std::span(in, kSize), bytes, error));
 
   // Try a full read
-  EXPECT_EQ(SR_SUCCESS, buf.Read(MakeArrayView(out, kSize), bytes, error));
+  EXPECT_EQ(SR_SUCCESS, buf.Read(std::span(out, kSize), bytes, error));
   EXPECT_EQ(kSize, bytes);
   EXPECT_EQ(0, memcmp(in, out, kSize));
 
   // Try a read that should block
-  EXPECT_EQ(SR_BLOCK, buf.Read(MakeArrayView(out, kSize), bytes, error));
+  EXPECT_EQ(SR_BLOCK, buf.Read(std::span(out, kSize), bytes, error));
 
   // Try a too-big write
-  EXPECT_EQ(SR_SUCCESS, buf.Write(MakeArrayView(in, kSize * 2), bytes, error));
+  EXPECT_EQ(SR_SUCCESS, buf.Write(std::span(in, kSize * 2), bytes, error));
   EXPECT_EQ(bytes, kSize);
 
   // Try a too-big read
-  EXPECT_EQ(SR_SUCCESS, buf.Read(MakeArrayView(out, kSize * 2), bytes, error));
+  EXPECT_EQ(SR_SUCCESS, buf.Read(std::span(out, kSize * 2), bytes, error));
   EXPECT_EQ(kSize, bytes);
   EXPECT_EQ(0, memcmp(in, out, kSize));
 
   // Try some small writes and reads
-  EXPECT_EQ(SR_SUCCESS, buf.Write(MakeArrayView(in, kSize / 2), bytes, error));
+  EXPECT_EQ(SR_SUCCESS, buf.Write(std::span(in, kSize / 2), bytes, error));
   EXPECT_EQ(kSize / 2, bytes);
-  EXPECT_EQ(SR_SUCCESS, buf.Read(MakeArrayView(out, kSize / 2), bytes, error));
+  EXPECT_EQ(SR_SUCCESS, buf.Read(std::span(out, kSize / 2), bytes, error));
   EXPECT_EQ(kSize / 2, bytes);
   EXPECT_EQ(0, memcmp(in, out, kSize / 2));
-  EXPECT_EQ(SR_SUCCESS, buf.Write(MakeArrayView(in, kSize / 2), bytes, error));
+  EXPECT_EQ(SR_SUCCESS, buf.Write(std::span(in, kSize / 2), bytes, error));
   EXPECT_EQ(kSize / 2, bytes);
-  EXPECT_EQ(SR_SUCCESS, buf.Write(MakeArrayView(in, kSize / 2), bytes, error));
+  EXPECT_EQ(SR_SUCCESS, buf.Write(std::span(in, kSize / 2), bytes, error));
   EXPECT_EQ(kSize / 2, bytes);
-  EXPECT_EQ(SR_SUCCESS, buf.Read(MakeArrayView(out, kSize / 2), bytes, error));
+  EXPECT_EQ(SR_SUCCESS, buf.Read(std::span(out, kSize / 2), bytes, error));
   EXPECT_EQ(kSize / 2, bytes);
   EXPECT_EQ(0, memcmp(in, out, kSize / 2));
-  EXPECT_EQ(SR_SUCCESS, buf.Read(MakeArrayView(out, kSize / 2), bytes, error));
+  EXPECT_EQ(SR_SUCCESS, buf.Read(std::span(out, kSize / 2), bytes, error));
   EXPECT_EQ(kSize / 2, bytes);
   EXPECT_EQ(0, memcmp(in, out, kSize / 2));
 
@@ -87,23 +87,22 @@ TEST(FifoBufferTest, TestAll) {
   // XXXXWWWWWWWWXXXX 4567012345670123
   // RRRRXXXXXXXXRRRR ....01234567....
   // ....RRRRRRRR.... ................
-  EXPECT_EQ(SR_SUCCESS,
-            buf.Write(MakeArrayView(in, kSize * 3 / 4), bytes, error));
+  EXPECT_EQ(SR_SUCCESS, buf.Write(std::span(in, kSize * 3 / 4), bytes, error));
   EXPECT_EQ(kSize * 3 / 4, bytes);
-  EXPECT_EQ(SR_SUCCESS, buf.Read(MakeArrayView(out, kSize / 2), bytes, error));
+  EXPECT_EQ(SR_SUCCESS, buf.Read(std::span(out, kSize / 2), bytes, error));
   EXPECT_EQ(kSize / 2, bytes);
   EXPECT_EQ(0, memcmp(in, out, kSize / 2));
-  EXPECT_EQ(SR_SUCCESS, buf.Write(MakeArrayView(in, kSize / 2), bytes, error));
+  EXPECT_EQ(SR_SUCCESS, buf.Write(std::span(in, kSize / 2), bytes, error));
   EXPECT_EQ(kSize / 2, bytes);
-  EXPECT_EQ(SR_SUCCESS, buf.Read(MakeArrayView(out, kSize / 4), bytes, error));
+  EXPECT_EQ(SR_SUCCESS, buf.Read(std::span(out, kSize / 4), bytes, error));
   EXPECT_EQ(kSize / 4, bytes);
   EXPECT_EQ(0, memcmp(in + kSize / 2, out, kSize / 4));
-  EXPECT_EQ(SR_SUCCESS, buf.Write(MakeArrayView(in, kSize / 2), bytes, error));
+  EXPECT_EQ(SR_SUCCESS, buf.Write(std::span(in, kSize / 2), bytes, error));
   EXPECT_EQ(kSize / 2, bytes);
-  EXPECT_EQ(SR_SUCCESS, buf.Read(MakeArrayView(out, kSize / 2), bytes, error));
+  EXPECT_EQ(SR_SUCCESS, buf.Read(std::span(out, kSize / 2), bytes, error));
   EXPECT_EQ(kSize / 2, bytes);
   EXPECT_EQ(0, memcmp(in, out, kSize / 2));
-  EXPECT_EQ(SR_SUCCESS, buf.Read(MakeArrayView(out, kSize / 2), bytes, error));
+  EXPECT_EQ(SR_SUCCESS, buf.Read(std::span(out, kSize / 2), bytes, error));
   EXPECT_EQ(kSize / 2, bytes);
   EXPECT_EQ(0, memcmp(in, out, kSize / 2));
 
@@ -112,16 +111,16 @@ TEST(FifoBufferTest, TestAll) {
   buf.ConsumeWriteBuffer(0);
 
   // Try using GetReadData to do a full read
-  EXPECT_EQ(SR_SUCCESS, buf.Write(MakeArrayView(in, kSize), bytes, error));
+  EXPECT_EQ(SR_SUCCESS, buf.Write(std::span(in, kSize), bytes, error));
   q = buf.GetReadData(&bytes);
   EXPECT_TRUE(nullptr != q);
   EXPECT_EQ(kSize, bytes);
   EXPECT_EQ(0, memcmp(q, in, kSize));
   buf.ConsumeReadData(kSize);
-  EXPECT_EQ(SR_BLOCK, buf.Read(MakeArrayView(out, kSize), bytes, error));
+  EXPECT_EQ(SR_BLOCK, buf.Read(std::span(out, kSize), bytes, error));
 
   // Try using GetReadData to do some small reads
-  EXPECT_EQ(SR_SUCCESS, buf.Write(MakeArrayView(in, kSize), bytes, error));
+  EXPECT_EQ(SR_SUCCESS, buf.Write(std::span(in, kSize), bytes, error));
   q = buf.GetReadData(&bytes);
   EXPECT_TRUE(nullptr != q);
   EXPECT_EQ(kSize, bytes);
@@ -132,7 +131,7 @@ TEST(FifoBufferTest, TestAll) {
   EXPECT_EQ(kSize / 2, bytes);
   EXPECT_EQ(0, memcmp(q, in + kSize / 2, kSize / 2));
   buf.ConsumeReadData(kSize / 2);
-  EXPECT_EQ(SR_BLOCK, buf.Read(MakeArrayView(out, kSize), bytes, error));
+  EXPECT_EQ(SR_BLOCK, buf.Read(std::span(out, kSize), bytes, error));
 
   // Try using GetReadData in a wraparound case
   // WWWWWWWWWWWWWWWW 0123456789ABCDEF
@@ -140,10 +139,9 @@ TEST(FifoBufferTest, TestAll) {
   // WWWWWWWW....XXXX 01234567....CDEF
   // ............RRRR 01234567........
   // RRRRRRRR........ ................
-  EXPECT_EQ(SR_SUCCESS, buf.Write(MakeArrayView(in, kSize), bytes, error));
-  EXPECT_EQ(SR_SUCCESS,
-            buf.Read(MakeArrayView(out, kSize * 3 / 4), bytes, error));
-  EXPECT_EQ(SR_SUCCESS, buf.Write(MakeArrayView(in, kSize / 2), bytes, error));
+  EXPECT_EQ(SR_SUCCESS, buf.Write(std::span(in, kSize), bytes, error));
+  EXPECT_EQ(SR_SUCCESS, buf.Read(std::span(out, kSize * 3 / 4), bytes, error));
+  EXPECT_EQ(SR_SUCCESS, buf.Write(std::span(in, kSize / 2), bytes, error));
   q = buf.GetReadData(&bytes);
   EXPECT_TRUE(nullptr != q);
   EXPECT_EQ(kSize / 4, bytes);
@@ -165,7 +163,7 @@ TEST(FifoBufferTest, TestAll) {
   EXPECT_EQ(kSize, bytes);
   memcpy(p, in, kSize);
   buf.ConsumeWriteBuffer(kSize);
-  EXPECT_EQ(SR_SUCCESS, buf.Read(MakeArrayView(out, kSize), bytes, error));
+  EXPECT_EQ(SR_SUCCESS, buf.Read(std::span(out, kSize), bytes, error));
   EXPECT_EQ(kSize, bytes);
   EXPECT_EQ(0, memcmp(in, out, kSize));
 
@@ -180,7 +178,7 @@ TEST(FifoBufferTest, TestAll) {
   EXPECT_EQ(kSize / 2, bytes);
   memcpy(p, in + kSize / 2, kSize / 2);
   buf.ConsumeWriteBuffer(kSize / 2);
-  EXPECT_EQ(SR_SUCCESS, buf.Read(MakeArrayView(out, kSize), bytes, error));
+  EXPECT_EQ(SR_SUCCESS, buf.Read(std::span(out, kSize), bytes, error));
   EXPECT_EQ(kSize, bytes);
   EXPECT_EQ(0, memcmp(in, out, kSize));
 
@@ -190,9 +188,8 @@ TEST(FifoBufferTest, TestAll) {
   // ........XXXXWWWW ........89AB0123
   // WWWW....XXXXXXXX 4567....89AB0123
   // RRRR....RRRRRRRR ................
-  EXPECT_EQ(SR_SUCCESS,
-            buf.Write(MakeArrayView(in, kSize * 3 / 4), bytes, error));
-  EXPECT_EQ(SR_SUCCESS, buf.Read(MakeArrayView(out, kSize / 2), bytes, error));
+  EXPECT_EQ(SR_SUCCESS, buf.Write(std::span(in, kSize * 3 / 4), bytes, error));
+  EXPECT_EQ(SR_SUCCESS, buf.Read(std::span(out, kSize / 2), bytes, error));
   p = buf.GetWriteBuffer(&bytes);
   EXPECT_TRUE(nullptr != p);
   EXPECT_EQ(kSize / 4, bytes);
@@ -203,23 +200,22 @@ TEST(FifoBufferTest, TestAll) {
   EXPECT_EQ(kSize / 2, bytes);
   memcpy(p, in + kSize / 4, kSize / 4);
   buf.ConsumeWriteBuffer(kSize / 4);
-  EXPECT_EQ(SR_SUCCESS,
-            buf.Read(MakeArrayView(out, kSize * 3 / 4), bytes, error));
+  EXPECT_EQ(SR_SUCCESS, buf.Read(std::span(out, kSize * 3 / 4), bytes, error));
   EXPECT_EQ(kSize * 3 / 4, bytes);
   EXPECT_EQ(0, memcmp(in + kSize / 2, out, kSize / 4));
   EXPECT_EQ(0, memcmp(in, out + kSize / 4, kSize / 4));
 
   // Check that the stream is now empty
-  EXPECT_EQ(SR_BLOCK, buf.Read(MakeArrayView(out, kSize), bytes, error));
+  EXPECT_EQ(SR_BLOCK, buf.Read(std::span(out, kSize), bytes, error));
 
   // Write to the stream, close it, read the remaining bytes
-  EXPECT_EQ(SR_SUCCESS, buf.Write(MakeArrayView(in, kSize / 2), bytes, error));
+  EXPECT_EQ(SR_SUCCESS, buf.Write(std::span(in, kSize / 2), bytes, error));
   buf.Close();
   EXPECT_EQ(SS_CLOSED, buf.GetState());
-  EXPECT_EQ(SR_EOS, buf.Write(MakeArrayView(in, kSize / 2), bytes, error));
-  EXPECT_EQ(SR_SUCCESS, buf.Read(MakeArrayView(out, kSize / 2), bytes, error));
+  EXPECT_EQ(SR_EOS, buf.Write(std::span(in, kSize / 2), bytes, error));
+  EXPECT_EQ(SR_SUCCESS, buf.Read(std::span(out, kSize / 2), bytes, error));
   EXPECT_EQ(0, memcmp(in, out, kSize / 2));
-  EXPECT_EQ(SR_EOS, buf.Read(MakeArrayView(out, kSize / 2), bytes, error));
+  EXPECT_EQ(SR_EOS, buf.Read(std::span(out, kSize / 2), bytes, error));
 }
 
 TEST(FifoBufferTest, FullBufferCheck) {
diff --git a/rtc_base/memory_stream.cc b/rtc_base/memory_stream.cc
index 9134d610e4..7994449e64 100644
--- a/rtc_base/memory_stream.cc
+++ b/rtc_base/memory_stream.cc
@@ -14,8 +14,8 @@
 #include <cerrno>
 #include <cstdint>
 #include <cstring>
+#include <span>
 
-#include "api/array_view.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/stream.h"
 
@@ -25,7 +25,7 @@ StreamState MemoryStream::GetState() const {
   return SS_OPEN;
 }
 
-StreamResult MemoryStream::Read(ArrayView<uint8_t> buffer,
+StreamResult MemoryStream::Read(std::span<uint8_t> buffer,
                                 size_t& bytes_read,
                                 int& error) {
   if (seek_position_ >= data_length_) {
@@ -45,7 +45,7 @@ StreamResult MemoryStream::Read(ArrayView<uint8_t> buffer,
   return SR_SUCCESS;
 }
 
-StreamResult MemoryStream::Write(ArrayView<const uint8_t> buffer,
+StreamResult MemoryStream::Write(std::span<const uint8_t> buffer,
                                  size_t& bytes_written,
                                  int& error) {
   size_t available = buffer_length_ - seek_position_;
diff --git a/rtc_base/memory_stream.h b/rtc_base/memory_stream.h
index 312bde6703..13dd396d34 100644
--- a/rtc_base/memory_stream.h
+++ b/rtc_base/memory_stream.h
@@ -14,8 +14,8 @@
 #include <stddef.h>
 
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "rtc_base/stream.h"
 
 namespace webrtc {
@@ -28,10 +28,10 @@ class MemoryStream final : public StreamInterface {
   ~MemoryStream() override;
 
   StreamState GetState() const override;
-  StreamResult Read(ArrayView<uint8_t> buffer,
+  StreamResult Read(std::span<uint8_t> buffer,
                     size_t& bytes_read,
                     int& error) override;
-  StreamResult Write(ArrayView<const uint8_t> buffer,
+  StreamResult Write(std::span<const uint8_t> buffer,
                      size_t& bytes_written,
                      int& error) override;
   void Close() override;
diff --git a/rtc_base/network.cc b/rtc_base/network.cc
index 676a11273f..fd1e3e739c 100644
--- a/rtc_base/network.cc
+++ b/rtc_base/network.cc
@@ -16,6 +16,7 @@
 #include <cstring>
 #include <map>
 #include <memory>
+#include <span>
 #include <string>
 #include <tuple>
 #include <utility>
@@ -26,7 +27,6 @@
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/match.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/field_trials_view.h"
 #include "api/sequence_checker.h"
@@ -567,7 +567,7 @@ Network* NetworkManagerBase::GetNetworkFromAddress(const IPAddress& ip) const {
   return nullptr;
 }
 
-bool NetworkManagerBase::IsVpnMacAddress(ArrayView<const uint8_t> address) {
+bool NetworkManagerBase::IsVpnMacAddress(std::span<const uint8_t> address) {
   if (address.data() == nullptr && address.empty()) {
     return false;
   }
@@ -910,7 +910,7 @@ bool BasicNetworkManager::CreateNetworks(
             adapter_type = ADAPTER_TYPE_VPN;
           }
           if (adapter_type != ADAPTER_TYPE_VPN &&
-              IsVpnMacAddress(ArrayView<const uint8_t>(
+              IsVpnMacAddress(std::span<const uint8_t>(
                   reinterpret_cast<const uint8_t*>(
                       adapter_addrs->PhysicalAddress),
                   adapter_addrs->PhysicalAddressLength))) {
diff --git a/rtc_base/network.h b/rtc_base/network.h
index e4a5ba6300..4a6bdd0a3b 100644
--- a/rtc_base/network.h
+++ b/rtc_base/network.h
@@ -15,6 +15,7 @@
 
 #include <map>
 #include <memory>
+#include <span>
 #include <string>
 #include <tuple>
 #include <utility>
@@ -23,7 +24,6 @@
 #include "absl/base/nullability.h"
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/field_trials_view.h"
 #include "api/scoped_refptr.h"
@@ -470,7 +470,7 @@ class RTC_EXPORT NetworkManagerBase : public NetworkManager {
 
   // Check if MAC address in |bytes| is one of the pre-defined
   // MAC addresses for know VPNs.
-  static bool IsVpnMacAddress(ArrayView<const uint8_t> address);
+  static bool IsVpnMacAddress(std::span<const uint8_t> address);
 
  protected:
   // Updates `networks_` with the networks listed in `list`. If
diff --git a/rtc_base/network/BUILD.gn b/rtc_base/network/BUILD.gn
index 3801b73249..d0705cbec0 100644
--- a/rtc_base/network/BUILD.gn
+++ b/rtc_base/network/BUILD.gn
@@ -26,7 +26,6 @@ rtc_library("received_packet") {
   deps = [
     "..:checks",
     "..:socket_address",
-    "../../api:array_view",
     "../../api/transport:ecn_marking",
     "../../api/units:timestamp",
     "../system:rtc_export",
diff --git a/rtc_base/network/received_packet.cc b/rtc_base/network/received_packet.cc
index 3f08e2d099..1677b6b0cf 100644
--- a/rtc_base/network/received_packet.cc
+++ b/rtc_base/network/received_packet.cc
@@ -13,9 +13,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <utility>
 
-#include "api/array_view.h"
 #include "api/transport/ecn_marking.h"
 #include "api/units/timestamp.h"
 #include "rtc_base/checks.h"
@@ -23,7 +23,7 @@
 
 namespace webrtc {
 
-ReceivedIpPacket::ReceivedIpPacket(ArrayView<const uint8_t> payload,
+ReceivedIpPacket::ReceivedIpPacket(std::span<const uint8_t> payload,
                                    const SocketAddress& source_address,
                                    std::optional<Timestamp> arrival_time,
                                    EcnMarking ecn,
@@ -48,7 +48,7 @@ ReceivedIpPacket ReceivedIpPacket::CreateFromLegacy(
     const SocketAddress& source_address) {
   RTC_DCHECK(packet_time_us == -1 || packet_time_us >= 0);
   return ReceivedIpPacket(
-      MakeArrayView(data, size), source_address,
+      std::span(data, size), source_address,
       (packet_time_us >= 0)
           ? std::optional<Timestamp>(Timestamp::Micros(packet_time_us))
           : std::nullopt);
diff --git a/rtc_base/network/received_packet.h b/rtc_base/network/received_packet.h
index 1078f807a6..2868047fa4 100644
--- a/rtc_base/network/received_packet.h
+++ b/rtc_base/network/received_packet.h
@@ -13,8 +13,8 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/transport/ecn_marking.h"
 #include "api/units/timestamp.h"
 #include "rtc_base/socket_address.h"
@@ -37,7 +37,7 @@ class RTC_EXPORT ReceivedIpPacket {
 
   // Caller must keep memory pointed to by payload and address valid for the
   // lifetime of this ReceivedPacket.
-  ReceivedIpPacket(ArrayView<const uint8_t> payload,
+  ReceivedIpPacket(std::span<const uint8_t> payload,
                    const SocketAddress& source_address,
                    std::optional<Timestamp> arrival_time = std::nullopt,
                    EcnMarking ecn = EcnMarking::kNotEct,
@@ -47,7 +47,7 @@ class RTC_EXPORT ReceivedIpPacket {
 
   // Address/port of the packet sender.
   const SocketAddress& source_address() const { return source_address_; }
-  ArrayView<const uint8_t> payload() const { return payload_; }
+  std::span<const uint8_t> payload() const { return payload_; }
 
   // Timestamp when this packet was received. Not available on all socket
   // implementations.
@@ -74,7 +74,7 @@ class RTC_EXPORT ReceivedIpPacket {
       const SocketAddress& = SocketAddress());
 
  private:
-  ArrayView<const uint8_t> payload_;
+  std::span<const uint8_t> payload_;
   std::optional<Timestamp> arrival_time_;
   const SocketAddress& source_address_;
   EcnMarking ecn_;
diff --git a/rtc_base/network_unittest.cc b/rtc_base/network_unittest.cc
index f9438b06c6..380580f2b2 100644
--- a/rtc_base/network_unittest.cc
+++ b/rtc_base/network_unittest.cc
@@ -14,6 +14,7 @@
 #include <cstdlib>
 #include <cstring>
 #include <memory>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
@@ -22,7 +23,6 @@
 #include "absl/strings/match.h"
 #include "absl/strings/str_cat.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/environment/environment_factory.h"
 #include "api/field_trials.h"
@@ -1745,7 +1745,7 @@ TEST_F(NetworkTest, HardcodedVpn) {
   EXPECT_TRUE(NetworkManagerBase::IsVpnMacAddress(global));
 
   EXPECT_FALSE(
-      NetworkManagerBase::IsVpnMacAddress(ArrayView<const uint8_t>(cisco, 5)));
+      NetworkManagerBase::IsVpnMacAddress(std::span<const uint8_t>(cisco, 5)));
   EXPECT_FALSE(NetworkManagerBase::IsVpnMacAddress(five_bytes));
   EXPECT_FALSE(NetworkManagerBase::IsVpnMacAddress(unknown));
   EXPECT_FALSE(NetworkManagerBase::IsVpnMacAddress({}));
diff --git a/rtc_base/openssl_stream_adapter.cc b/rtc_base/openssl_stream_adapter.cc
index 0a410f8e80..e3a75db8f4 100644
--- a/rtc_base/openssl_stream_adapter.cc
+++ b/rtc_base/openssl_stream_adapter.cc
@@ -21,6 +21,7 @@
 #include <cstring>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
@@ -28,7 +29,6 @@
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/str_format.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/field_trials_view.h"
 #include "api/sequence_checker.h"
 #include "api/task_queue/pending_task_safety_flag.h"
@@ -248,7 +248,7 @@ int stream_read(BIO* b, char* out, int outl) {
   size_t read;
   int error;
   StreamResult result = stream->Read(
-      MakeArrayView(reinterpret_cast<uint8_t*>(out), outl), read, error);
+      std::span(reinterpret_cast<uint8_t*>(out), outl), read, error);
   if (result == SR_SUCCESS) {
     return checked_cast<int>(read);
   } else if (result == SR_BLOCK) {
@@ -266,7 +266,7 @@ int stream_write(BIO* b, const char* in, int inl) {
   size_t written;
   int error;
   StreamResult result = stream->Write(
-      MakeArrayView(reinterpret_cast<const uint8_t*>(in), inl), written, error);
+      std::span(reinterpret_cast<const uint8_t*>(in), inl), written, error);
   if (result == SR_SUCCESS) {
     return checked_cast<int>(written);
   } else if (result == SR_BLOCK) {
@@ -366,7 +366,7 @@ void OpenSSLStreamAdapter::SetServerRole(SSLRole role) {
 
 SSLPeerCertificateDigestError OpenSSLStreamAdapter::SetPeerCertificateDigest(
     absl::string_view digest_alg,
-    ArrayView<const uint8_t> digest_val) {
+    std::span<const uint8_t> digest_val) {
   RTC_DCHECK(!peer_certificate_verified_);
   RTC_DCHECK(!HasPeerCertificateDigest());
   size_t expected_len;
@@ -521,7 +521,7 @@ bool OpenSSLStreamAdapter::AppendSrtpKeyingMaterial(
   // use_context            optional, can be null, 0 (IN). Not used by WebRTC.
   bool success = false;
   key_buffer.AppendData(
-      key_material_size, [&](ArrayView<uint8_t> keying_material) -> size_t {
+      key_material_size, [&](std::span<uint8_t> keying_material) -> size_t {
         int result = SSL_export_keying_material(
             ssl_, keying_material.data(), keying_material.size(),
             kDtlsSrtpExporterLabel.data(), kDtlsSrtpExporterLabel.size(),
@@ -664,7 +664,7 @@ void OpenSSLStreamAdapter::SetMTU(int mtu) {
 //
 // StreamInterface Implementation
 //
-StreamResult OpenSSLStreamAdapter::Write(ArrayView<const uint8_t> data,
+StreamResult OpenSSLStreamAdapter::Write(std::span<const uint8_t> data,
                                          size_t& written,
                                          int& error) {
   RTC_DLOG(LS_VERBOSE) << "OpenSSLStreamAdapter::Write(" << data.size() << ")";
@@ -722,7 +722,7 @@ StreamResult OpenSSLStreamAdapter::Write(ArrayView<const uint8_t> data,
   // not reached
 }
 
-StreamResult OpenSSLStreamAdapter::Read(ArrayView<uint8_t> data,
+StreamResult OpenSSLStreamAdapter::Read(std::span<uint8_t> data,
                                         size_t& read,
                                         int& error) {
   RTC_DLOG(LS_VERBOSE) << "OpenSSLStreamAdapter::Read(" << data.size() << ")";
diff --git a/rtc_base/openssl_stream_adapter.h b/rtc_base/openssl_stream_adapter.h
index 5307eff203..b8e531de67 100644
--- a/rtc_base/openssl_stream_adapter.h
+++ b/rtc_base/openssl_stream_adapter.h
@@ -17,12 +17,12 @@
 
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/ssl_certificate.h"
 #ifdef OPENSSL_IS_BORINGSSL
@@ -82,7 +82,7 @@ class OpenSSLStreamAdapter final : public SSLStreamAdapter {
   void SetServerRole(SSLRole role = SSL_SERVER) override;
   SSLPeerCertificateDigestError SetPeerCertificateDigest(
       absl::string_view digest_alg,
-      ArrayView<const uint8_t> digest_val) override;
+      std::span<const uint8_t> digest_val) override;
 
   std::unique_ptr<SSLCertChain> GetPeerSSLCertChain() const override;
 
@@ -95,8 +95,8 @@ class OpenSSLStreamAdapter final : public SSLStreamAdapter {
   void UpdateRetransmissionTimeout(int timeout_ms) override;
   void SetMTU(int mtu) override;
 
-  StreamResult Read(ArrayView<uint8_t> data, size_t& read, int& error) override;
-  StreamResult Write(ArrayView<const uint8_t> data,
+  StreamResult Read(std::span<uint8_t> data, size_t& read, int& error) override;
+  StreamResult Write(std::span<const uint8_t> data,
                      size_t& written,
                      int& error) override;
   void Close() override;
diff --git a/rtc_base/physical_socket_server.cc b/rtc_base/physical_socket_server.cc
index 66f649306c..fb916fb50e 100644
--- a/rtc_base/physical_socket_server.cc
+++ b/rtc_base/physical_socket_server.cc
@@ -17,9 +17,9 @@
 #include <cstdint>
 #include <cstring>
 #include <memory>
+#include <span>
 #include <utility>
 
-#include "api/array_view.h"
 #include "api/async_dns_resolver.h"
 #include "api/transport/ecn_marking.h"
 #include "api/units/time_delta.h"
@@ -493,7 +493,7 @@ int PhysicalSocket::RecvFrom(ReceiveBuffer& buffer) {
   static constexpr int BUF_SIZE = 64 * 1024;
   buffer.payload.EnsureCapacity(BUF_SIZE);
   int received;
-  buffer.payload.SetData(BUF_SIZE, [&](ArrayView<uint8_t> payload) {
+  buffer.payload.SetData(BUF_SIZE, [&](std::span<uint8_t> payload) {
     received =
         DoReadFromSocket(payload.data(), payload.size(), &buffer.source_address,
                          &timestamp, ecn_ ? &buffer.ecn : nullptr);
diff --git a/rtc_base/server_socket_adapters.cc b/rtc_base/server_socket_adapters.cc
index 339afb78b0..85047e6f9a 100644
--- a/rtc_base/server_socket_adapters.cc
+++ b/rtc_base/server_socket_adapters.cc
@@ -13,8 +13,8 @@
 #include <cstddef>
 #include <cstdint>
 #include <cstring>
+#include <span>
 
-#include "api/array_view.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/socket.h"
 #include "rtc_base/socket_adapters.h"
@@ -34,7 +34,7 @@ AsyncSSLServerSocket::AsyncSSLServerSocket(Socket* socket)
 
 void AsyncSSLServerSocket::ProcessInput(char* data, size_t* len) {
   // We only accept client hello messages.
-  const ArrayView<const uint8_t> client_hello =
+  const std::span<const uint8_t> client_hello =
       AsyncSSLSocket::SslClientHello();
   if (*len < client_hello.size()) {
     return;
@@ -51,7 +51,7 @@ void AsyncSSLServerSocket::ProcessInput(char* data, size_t* len) {
   // Clients should not send more data until the handshake is completed.
   RTC_DCHECK(*len == 0);
 
-  const ArrayView<const uint8_t> server_hello =
+  const std::span<const uint8_t> server_hello =
       AsyncSSLSocket::SslServerHello();
   // Send a server hello back to the client.
   DirectSend(server_hello.data(), server_hello.size());
diff --git a/rtc_base/socket.cc b/rtc_base/socket.cc
index dee6caa320..744311ce9a 100644
--- a/rtc_base/socket.cc
+++ b/rtc_base/socket.cc
@@ -12,8 +12,8 @@
 
 #include <algorithm>
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/units/timestamp.h"
 #include "rtc_base/buffer.h"
 
@@ -24,7 +24,7 @@ int Socket::RecvFrom(ReceiveBuffer& buffer) {
   int64_t timestamp = -1;
   int len;
   buffer.payload.EnsureCapacity(BUF_SIZE);
-  buffer.payload.SetData(BUF_SIZE, [&](ArrayView<uint8_t> payload) {
+  buffer.payload.SetData(BUF_SIZE, [&](std::span<uint8_t> payload) {
     len = RecvFrom(payload.data(), payload.size(), &buffer.source_address,
                    &timestamp);
     return std::max(len, 0);
diff --git a/rtc_base/socket_adapters.cc b/rtc_base/socket_adapters.cc
index 60d52b2bb6..56b20bdd49 100644
--- a/rtc_base/socket_adapters.cc
+++ b/rtc_base/socket_adapters.cc
@@ -15,8 +15,8 @@
 #include <cstddef>
 #include <cstdint>
 #include <cstring>
+#include <span>
 
-#include "api/array_view.h"
 #include "rtc_base/async_socket.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
@@ -135,8 +135,8 @@ static const uint8_t kSslClientHello[] = {
 };
 
 // static
-ArrayView<const uint8_t> AsyncSSLSocket::SslClientHello() {
-  // Implicit conversion directly from kSslClientHello to ArrayView fails when
+std::span<const uint8_t> AsyncSSLSocket::SslClientHello() {
+  // Implicit conversion directly from kSslClientHello to std::span fails when
   // built with gcc.
   return {kSslClientHello, sizeof(kSslClientHello)};
 }
@@ -163,7 +163,7 @@ static const uint8_t kSslServerHello[] = {
 };
 
 // static
-ArrayView<const uint8_t> AsyncSSLSocket::SslServerHello() {
+std::span<const uint8_t> AsyncSSLSocket::SslServerHello() {
   return {kSslServerHello, sizeof(kSslServerHello)};
 }
 
diff --git a/rtc_base/socket_adapters.h b/rtc_base/socket_adapters.h
index 3abd888ad3..e0954a9fe4 100644
--- a/rtc_base/socket_adapters.h
+++ b/rtc_base/socket_adapters.h
@@ -13,8 +13,8 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "rtc_base/async_socket.h"
 #include "rtc_base/socket.h"
 #include "rtc_base/socket_address.h"
@@ -59,8 +59,8 @@ class BufferedReadAdapter : public AsyncSocketAdapter {
 // fake SSL handshake. Used for "ssltcp" P2P functionality.
 class AsyncSSLSocket : public BufferedReadAdapter {
  public:
-  static ArrayView<const uint8_t> SslClientHello();
-  static ArrayView<const uint8_t> SslServerHello();
+  static std::span<const uint8_t> SslClientHello();
+  static std::span<const uint8_t> SslServerHello();
 
   explicit AsyncSSLSocket(Socket* socket);
 
diff --git a/rtc_base/ssl_certificate.cc b/rtc_base/ssl_certificate.cc
index 403638cd67..e144d39bde 100644
--- a/rtc_base/ssl_certificate.cc
+++ b/rtc_base/ssl_certificate.cc
@@ -13,13 +13,13 @@
 #include <cstddef>
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/algorithm/container.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "rtc_base/base64.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/openssl.h"
@@ -83,7 +83,7 @@ std::unique_ptr<SSLCertificateStats> SSLCertificate::GetStats() const {
 
   Buffer der_buffer;
   ToDER(&der_buffer);
-  ArrayView<const uint8_t> der_view(der_buffer);
+  std::span<const uint8_t> der_view(der_buffer);
   std::string der_base64 = Base64Encode(der_view);
 
   return std::make_unique<SSLCertificateStats>(std::move(fingerprint),
diff --git a/rtc_base/ssl_fingerprint.cc b/rtc_base/ssl_fingerprint.cc
index 52253630a7..6911d8b5e8 100644
--- a/rtc_base/ssl_fingerprint.cc
+++ b/rtc_base/ssl_fingerprint.cc
@@ -14,11 +14,11 @@
 #include <cstddef>
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <string>
 
 #include "absl/algorithm/container.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/message_digest.h"
@@ -68,13 +68,13 @@ std::unique_ptr<SSLFingerprint> SSLFingerprint::CreateUniqueFromRfc4572(
 
   char value[MessageDigest::kMaxSize];
   size_t value_len =
-      hex_decode_with_delimiter(ArrayView<char>(value), fingerprint, ':');
+      hex_decode_with_delimiter(std::span<char>(value), fingerprint, ':');
   if (!value_len)
     return nullptr;
 
   return std::make_unique<SSLFingerprint>(
       algorithm,
-      ArrayView<const uint8_t>(reinterpret_cast<uint8_t*>(value), value_len));
+      std::span<const uint8_t>(reinterpret_cast<uint8_t*>(value), value_len));
 }
 
 std::unique_ptr<SSLFingerprint> SSLFingerprint::CreateFromCertificate(
@@ -96,13 +96,13 @@ std::unique_ptr<SSLFingerprint> SSLFingerprint::CreateFromCertificate(
 }
 
 SSLFingerprint::SSLFingerprint(absl::string_view algorithm,
-                               ArrayView<const uint8_t> digest_view)
+                               std::span<const uint8_t> digest_view)
     : algorithm(algorithm), digest(digest_view.data(), digest_view.size()) {}
 
 SSLFingerprint::SSLFingerprint(absl::string_view algorithm,
                                const uint8_t* digest_in,
                                size_t digest_len)
-    : SSLFingerprint(algorithm, MakeArrayView(digest_in, digest_len)) {}
+    : SSLFingerprint(algorithm, std::span(digest_in, digest_len)) {}
 
 bool SSLFingerprint::operator==(const SSLFingerprint& other) const {
   return algorithm == other.algorithm && digest == other.digest;
diff --git a/rtc_base/ssl_fingerprint.h b/rtc_base/ssl_fingerprint.h
index 89757f4865..103e4a29d7 100644
--- a/rtc_base/ssl_fingerprint.h
+++ b/rtc_base/ssl_fingerprint.h
@@ -15,10 +15,10 @@
 #include <stdint.h>
 
 #include <memory>
+#include <span>
 #include <string>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "rtc_base/copy_on_write_buffer.h"
 #include "rtc_base/rtc_certificate.h"
 #include "rtc_base/ssl_certificate.h"
@@ -53,7 +53,7 @@ struct RTC_EXPORT SSLFingerprint {
       const RTCCertificate& cert);
 
   SSLFingerprint(absl::string_view algorithm,
-                 ArrayView<const uint8_t> digest_view);
+                 std::span<const uint8_t> digest_view);
   // TODO(steveanton): Remove once downstream projects have moved off of this.
   SSLFingerprint(absl::string_view algorithm,
                  const uint8_t* digest_in,
diff --git a/rtc_base/ssl_identity.cc b/rtc_base/ssl_identity.cc
index 2186d3a60f..f68797afa3 100644
--- a/rtc_base/ssl_identity.cc
+++ b/rtc_base/ssl_identity.cc
@@ -18,11 +18,11 @@
 #include <ctime>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "rtc_base/checks.h"
 #ifdef OPENSSL_IS_BORINGSSL
 #include "rtc_base/boringssl_identity.h"
@@ -206,7 +206,7 @@ std::string SSLIdentity::DerToPem(absl::string_view pem_type,
   StringBuilder result;
   result << "-----BEGIN " << pem_type << "-----\n";
 
-  ArrayView<const uint8_t> data_view(data, length);
+  std::span<const uint8_t> data_view(data, length);
   std::string b64_encoded = Base64Encode(data_view);
   // Divide the Base-64 encoded data into 64-character chunks, as per 4.3.2.4
   // of RFC 1421.
diff --git a/rtc_base/ssl_stream_adapter.cc b/rtc_base/ssl_stream_adapter.cc
index d02318bc68..59216f0357 100644
--- a/rtc_base/ssl_stream_adapter.cc
+++ b/rtc_base/ssl_stream_adapter.cc
@@ -15,13 +15,13 @@
 #include <memory>
 #include <optional>
 #include <set>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/field_trials_view.h"
 #include "rtc_base/openssl_stream_adapter.h"
 #include "rtc_base/ssl_identity.h"
@@ -181,7 +181,7 @@ bool SSLStreamAdapter::SetPeerCertificateDigest(
     SSLPeerCertificateDigestError* error) {
   unsigned char* nonconst_val = const_cast<unsigned char*>(digest_val);
   SSLPeerCertificateDigestError ret = SetPeerCertificateDigest(
-      digest_alg, ArrayView<uint8_t>(nonconst_val, digest_len));
+      digest_alg, std::span<uint8_t>(nonconst_val, digest_len));
   if (error)
     *error = ret;
   return ret == SSLPeerCertificateDigestError::NONE;
diff --git a/rtc_base/ssl_stream_adapter.h b/rtc_base/ssl_stream_adapter.h
index f35076c55b..75e672902b 100644
--- a/rtc_base/ssl_stream_adapter.h
+++ b/rtc_base/ssl_stream_adapter.h
@@ -17,12 +17,12 @@
 #include <memory>
 #include <optional>
 #include <set>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/field_trials_view.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/checks.h"
@@ -191,9 +191,9 @@ class SSLStreamAdapter : public StreamInterface {
   // Returns SSLPeerCertificateDigestError::NONE if successful.
   virtual SSLPeerCertificateDigestError SetPeerCertificateDigest(
       absl::string_view digest_alg,
-      ArrayView<const uint8_t> digest_val) = 0;
+      std::span<const uint8_t> digest_val) = 0;
   [[deprecated(
-      "Use SetPeerCertificateDigest with ArrayView instead")]] virtual bool
+      "Use SetPeerCertificateDigest with std::span instead")]] virtual bool
   SetPeerCertificateDigest(absl::string_view digest_alg,
                            const unsigned char* digest_val,
                            size_t digest_len,
diff --git a/rtc_base/ssl_stream_adapter_unittest.cc b/rtc_base/ssl_stream_adapter_unittest.cc
index 668a98c9da..67bfc103e3 100644
--- a/rtc_base/ssl_stream_adapter_unittest.cc
+++ b/rtc_base/ssl_stream_adapter_unittest.cc
@@ -25,6 +25,7 @@
 #include <ctime>
 #include <memory>
 #include <set>
+#include <span>
 #include <string>
 #include <tuple>
 #include <utility>
@@ -32,7 +33,6 @@
 
 #include "absl/memory/memory.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/crypto/crypto_options.h"
 #include "api/field_trials.h"
 #include "api/sequence_checker.h"
@@ -237,13 +237,13 @@ class StreamWrapper : public StreamInterface {
   }
   size_t GetFlushCountForTesting() { return flush_count_; }
 
-  StreamResult Read(ArrayView<uint8_t> buffer,
+  StreamResult Read(std::span<uint8_t> buffer,
                     size_t& read,
                     int& error) override {
     return stream_->Read(buffer, read, error);
   }
 
-  StreamResult Write(ArrayView<const uint8_t> data,
+  StreamResult Write(std::span<const uint8_t> data,
                      size_t& written,
                      int& error) override {
     return stream_->Write(data, written, error);
@@ -277,7 +277,7 @@ class SSLDummyStream final : public StreamInterface {
 
   StreamState GetState() const override { return SS_OPEN; }
 
-  StreamResult Read(ArrayView<uint8_t> buffer,
+  StreamResult Read(std::span<uint8_t> buffer,
                     size_t& read,
                     int& error) override {
     StreamResult r;
@@ -318,13 +318,13 @@ class SSLDummyStream final : public StreamInterface {
   }
 
   // Write to the outgoing FifoBuffer
-  StreamResult WriteData(ArrayView<const uint8_t> data,
+  StreamResult WriteData(std::span<const uint8_t> data,
                          size_t& written,
                          int& error) {
     return out_->Write(data, written, error);
   }
 
-  StreamResult Write(ArrayView<const uint8_t> data,
+  StreamResult Write(std::span<const uint8_t> data,
                      size_t& written,
                      int& error) override;
 
@@ -362,7 +362,7 @@ class BufferQueueStream : public StreamInterface {
   StreamState GetState() const override { return SS_OPEN; }
 
   // Reading a buffer queue stream will either succeed or block.
-  StreamResult Read(ArrayView<uint8_t> buffer,
+  StreamResult Read(std::span<uint8_t> buffer,
                     size_t& read,
                     int& error) override {
     const bool was_writable = buffer_.is_writable();
@@ -376,7 +376,7 @@ class BufferQueueStream : public StreamInterface {
   }
 
   // Writing to a buffer queue stream will either succeed or block.
-  StreamResult Write(ArrayView<const uint8_t> data,
+  StreamResult Write(std::span<const uint8_t> data,
                      size_t& written,
                      int& error) override {
     const bool was_readable = buffer_.is_readable();
@@ -762,12 +762,12 @@ class SSLStreamAdapterTestBase : public ::testing::Test {
       RTC_LOG(LS_VERBOSE) << "Damaging packet";
       memcpy(&buf[0], data, data_len);
       buf[data_len - 1]++;
-      return from->WriteData(MakeArrayView(&buf[0], data_len), written, error);
+      return from->WriteData(std::span(&buf[0], data_len), written, error);
     }
 
     return from->WriteData(
-        MakeArrayView(reinterpret_cast<const uint8_t*>(data), data_len),
-        written, error);
+        std::span(reinterpret_cast<const uint8_t*>(data), data_len), written,
+        error);
   }
 
   void SetDelay(int delay) { delay_ = delay; }
@@ -926,7 +926,7 @@ class SSLStreamAdapterTestDTLSBase : public SSLStreamAdapterTestBase {
       size_t sent;
       int error;
       StreamResult rv =
-          client_ssl_->Write(MakeArrayView(packet, packet_size_), sent, error);
+          client_ssl_->Write(std::span(packet, packet_size_), sent, error);
       if (rv == SR_SUCCESS) {
         RTC_LOG(LS_VERBOSE) << "Sent: " << sent_;
         sent_++;
@@ -1016,7 +1016,7 @@ class SSLStreamAdapterTestDTLSBase : public SSLStreamAdapterTestBase {
   std::set<int> received_;
 };
 
-webrtc::StreamResult SSLDummyStream::Write(ArrayView<const uint8_t> data,
+webrtc::StreamResult SSLDummyStream::Write(std::span<const uint8_t> data,
                                            size_t& written,
                                            int& error) {
   RTC_LOG(LS_VERBOSE) << "Writing to loopback " << data.size();
diff --git a/rtc_base/stream.h b/rtc_base/stream.h
index a8191567d5..07da2266e3 100644
--- a/rtc_base/stream.h
+++ b/rtc_base/stream.h
@@ -13,10 +13,10 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 #include <utility>
 
 #include "absl/functional/any_invocable.h"
-#include "api/array_view.h"
 #include "api/sequence_checker.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/system/no_unique_address.h"
@@ -74,10 +74,10 @@ class RTC_EXPORT StreamInterface {
   //  SR_EOS: the end-of-stream has been reached, or the stream is in the
   //    SS_CLOSED state.
 
-  virtual StreamResult Read(ArrayView<uint8_t> buffer,
+  virtual StreamResult Read(std::span<uint8_t> buffer,
                             size_t& read,
                             int& error) = 0;
-  virtual StreamResult Write(ArrayView<const uint8_t> data,
+  virtual StreamResult Write(std::span<const uint8_t> data,
                              size_t& written,
                              int& error) = 0;
 
diff --git a/rtc_base/string_encode.cc b/rtc_base/string_encode.cc
index 6b0b1e0049..717b832390 100644
--- a/rtc_base/string_encode.cc
+++ b/rtc_base/string_encode.cc
@@ -11,11 +11,11 @@
 #include "rtc_base/string_encode.h"
 
 #include <cstdio>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "rtc_base/checks.h"
 
 namespace webrtc {
@@ -91,7 +91,7 @@ std::string hex_encode_with_delimiter(absl::string_view source,
   return s;
 }
 
-size_t hex_decode_with_delimiter(ArrayView<char> cbuffer,
+size_t hex_decode_with_delimiter(std::span<char> cbuffer,
                                  absl::string_view source,
                                  char delimiter) {
   if (cbuffer.empty())
@@ -131,7 +131,7 @@ size_t hex_decode_with_delimiter(ArrayView<char> cbuffer,
   return bufpos;
 }
 
-size_t hex_decode(ArrayView<char> buffer, absl::string_view source) {
+size_t hex_decode(std::span<char> buffer, absl::string_view source) {
   return hex_decode_with_delimiter(buffer, source, 0);
 }
 
diff --git a/rtc_base/string_encode.h b/rtc_base/string_encode.h
index fb2a6e021a..187d8379a4 100644
--- a/rtc_base/string_encode.h
+++ b/rtc_base/string_encode.h
@@ -14,12 +14,12 @@
 #include <stddef.h>
 
 #include <optional>
+#include <span>
 #include <string>
 #include <type_traits>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/string_to_number.h"
 #include "rtc_base/strings/string_format.h"  // IWYU pragma: keep
@@ -34,13 +34,13 @@ std::string hex_encode(absl::string_view str);
 std::string hex_encode_with_delimiter(absl::string_view source, char delimiter);
 
 // hex_decode converts ascii hex to binary.
-size_t hex_decode(ArrayView<char> buffer, absl::string_view source);
+size_t hex_decode(std::span<char> buffer, absl::string_view source);
 
 // hex_decode, assuming that there is a delimiter between every byte
 // pair.
 // `delimiter` == 0 means no delimiter
 // If the buffer is too short or the data is invalid, we return 0.
-size_t hex_decode_with_delimiter(ArrayView<char> buffer,
+size_t hex_decode_with_delimiter(std::span<char> buffer,
                                  absl::string_view source,
                                  char delimiter);
 
diff --git a/rtc_base/string_encode_unittest.cc b/rtc_base/string_encode_unittest.cc
index d26cd6b82e..701e52d243 100644
--- a/rtc_base/string_encode_unittest.cc
+++ b/rtc_base/string_encode_unittest.cc
@@ -11,11 +11,11 @@
 #include "rtc_base/string_encode.h"
 
 #include <cstring>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "test/gtest.h"
 
 namespace webrtc {
@@ -39,7 +39,7 @@ class HexEncodeTest : public ::testing::Test {
 TEST_F(HexEncodeTest, TestWithNoDelimiter) {
   std::string encoded = hex_encode(data_view_);
   EXPECT_EQ("80818283848586878889", encoded);
-  dec_res_ = hex_decode(ArrayView<char>(decoded_), encoded);
+  dec_res_ = hex_decode(std::span<char>(decoded_), encoded);
   ASSERT_EQ(sizeof(data_), dec_res_);
   ASSERT_EQ(0, memcmp(data_, decoded_, dec_res_));
 }
@@ -48,7 +48,7 @@ TEST_F(HexEncodeTest, TestWithNoDelimiter) {
 TEST_F(HexEncodeTest, TestWithDelimiter) {
   std::string encoded = hex_encode_with_delimiter(data_view_, ':');
   EXPECT_EQ("80:81:82:83:84:85:86:87:88:89", encoded);
-  dec_res_ = hex_decode_with_delimiter(ArrayView<char>(decoded_), encoded, ':');
+  dec_res_ = hex_decode_with_delimiter(std::span<char>(decoded_), encoded, ':');
   ASSERT_EQ(sizeof(data_), dec_res_);
   ASSERT_EQ(0, memcmp(data_, decoded_, dec_res_));
 }
@@ -56,7 +56,7 @@ TEST_F(HexEncodeTest, TestWithDelimiter) {
 // Test that encoding with one delimiter and decoding with another fails.
 TEST_F(HexEncodeTest, TestWithWrongDelimiter) {
   std::string encoded = hex_encode_with_delimiter(data_view_, ':');
-  dec_res_ = hex_decode_with_delimiter(ArrayView<char>(decoded_), encoded, '/');
+  dec_res_ = hex_decode_with_delimiter(std::span<char>(decoded_), encoded, '/');
   ASSERT_EQ(0U, dec_res_);
 }
 
@@ -64,7 +64,7 @@ TEST_F(HexEncodeTest, TestWithWrongDelimiter) {
 TEST_F(HexEncodeTest, TestExpectedDelimiter) {
   std::string encoded = hex_encode(data_view_);
   EXPECT_EQ(sizeof(data_) * 2, encoded.size());
-  dec_res_ = hex_decode_with_delimiter(ArrayView<char>(decoded_), encoded, ':');
+  dec_res_ = hex_decode_with_delimiter(std::span<char>(decoded_), encoded, ':');
   ASSERT_EQ(0U, dec_res_);
 }
 
@@ -72,7 +72,7 @@ TEST_F(HexEncodeTest, TestExpectedDelimiter) {
 TEST_F(HexEncodeTest, TestExpectedNoDelimiter) {
   std::string encoded = hex_encode_with_delimiter(data_view_, ':');
   EXPECT_EQ(sizeof(data_) * 3 - 1, encoded.size());
-  dec_res_ = hex_decode(ArrayView<char>(decoded_), encoded);
+  dec_res_ = hex_decode(std::span<char>(decoded_), encoded);
   ASSERT_EQ(0U, dec_res_);
 }
 
@@ -80,7 +80,7 @@ TEST_F(HexEncodeTest, TestExpectedNoDelimiter) {
 TEST_F(HexEncodeTest, TestZeroLengthNoDelimiter) {
   std::string encoded = hex_encode("");
   EXPECT_TRUE(encoded.empty());
-  dec_res_ = hex_decode(ArrayView<char>(decoded_), encoded);
+  dec_res_ = hex_decode(std::span<char>(decoded_), encoded);
   ASSERT_EQ(0U, dec_res_);
 }
 
@@ -88,47 +88,47 @@ TEST_F(HexEncodeTest, TestZeroLengthNoDelimiter) {
 TEST_F(HexEncodeTest, TestZeroLengthWithDelimiter) {
   std::string encoded = hex_encode_with_delimiter("", ':');
   EXPECT_TRUE(encoded.empty());
-  dec_res_ = hex_decode_with_delimiter(ArrayView<char>(decoded_), encoded, ':');
+  dec_res_ = hex_decode_with_delimiter(std::span<char>(decoded_), encoded, ':');
   ASSERT_EQ(0U, dec_res_);
 }
 
 // Test that decoding into a too-small output buffer fails.
 TEST_F(HexEncodeTest, TestDecodeTooShort) {
   dec_res_ =
-      hex_decode_with_delimiter(ArrayView<char>(decoded_, 4), "0123456789", 0);
+      hex_decode_with_delimiter(std::span<char>(decoded_, 4), "0123456789", 0);
   ASSERT_EQ(0U, dec_res_);
   ASSERT_EQ(0x7f, decoded_[4]);
 }
 
 // Test that decoding non-hex data fails.
 TEST_F(HexEncodeTest, TestDecodeBogusData) {
-  dec_res_ = hex_decode_with_delimiter(ArrayView<char>(decoded_), "axyz", 0);
+  dec_res_ = hex_decode_with_delimiter(std::span<char>(decoded_), "axyz", 0);
   ASSERT_EQ(0U, dec_res_);
 }
 
 // Test that decoding an odd number of hex characters fails.
 TEST_F(HexEncodeTest, TestDecodeOddHexDigits) {
-  dec_res_ = hex_decode_with_delimiter(ArrayView<char>(decoded_), "012", 0);
+  dec_res_ = hex_decode_with_delimiter(std::span<char>(decoded_), "012", 0);
   ASSERT_EQ(0U, dec_res_);
 }
 
 // Test that decoding a string with too many delimiters fails.
 TEST_F(HexEncodeTest, TestDecodeWithDelimiterTooManyDelimiters) {
-  dec_res_ = hex_decode_with_delimiter(ArrayView<char>(decoded_, 4),
+  dec_res_ = hex_decode_with_delimiter(std::span<char>(decoded_, 4),
                                        "01::23::45::67", ':');
   ASSERT_EQ(0U, dec_res_);
 }
 
 // Test that decoding a string with a leading delimiter fails.
 TEST_F(HexEncodeTest, TestDecodeWithDelimiterLeadingDelimiter) {
-  dec_res_ = hex_decode_with_delimiter(ArrayView<char>(decoded_, 4),
+  dec_res_ = hex_decode_with_delimiter(std::span<char>(decoded_, 4),
                                        ":01:23:45:67", ':');
   ASSERT_EQ(0U, dec_res_);
 }
 
 // Test that decoding a string with a trailing delimiter fails.
 TEST_F(HexEncodeTest, TestDecodeWithDelimiterTrailingDelimiter) {
-  dec_res_ = hex_decode_with_delimiter(ArrayView<char>(decoded_, 4),
+  dec_res_ = hex_decode_with_delimiter(std::span<char>(decoded_, 4),
                                        "01:23:45:67:", ':');
   ASSERT_EQ(0U, dec_res_);
 }
diff --git a/rtc_base/strings/string_builder.cc b/rtc_base/strings/string_builder.cc
index 094bbd1939..f52445a8f4 100644
--- a/rtc_base/strings/string_builder.cc
+++ b/rtc_base/strings/string_builder.cc
@@ -13,15 +13,15 @@
 #include <cstdarg>
 #include <cstdio>
 #include <cstring>
+#include <span>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/numerics/safe_minmax.h"
 
 namespace webrtc {
 
-SimpleStringBuilder::SimpleStringBuilder(ArrayView<char> buffer)
+SimpleStringBuilder::SimpleStringBuilder(std::span<char> buffer)
     : buffer_(buffer) {
   buffer_[0] = '\0';
   RTC_DCHECK(IsConsistent());
diff --git a/rtc_base/strings/string_builder.h b/rtc_base/strings/string_builder.h
index 9f6fb33b4e..41efe9aa42 100644
--- a/rtc_base/strings/string_builder.h
+++ b/rtc_base/strings/string_builder.h
@@ -12,13 +12,13 @@
 #define RTC_BASE_STRINGS_STRING_BUILDER_H_
 
 #include <cstdio>
+#include <span>
 #include <string>
 #include <utility>
 
 #include "absl/strings/has_absl_stringify.h"
 #include "absl/strings/str_cat.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 
 namespace webrtc {
 
@@ -29,7 +29,7 @@ namespace webrtc {
 // read via `str()`.
 class SimpleStringBuilder {
  public:
-  explicit SimpleStringBuilder(ArrayView<char> buffer);
+  explicit SimpleStringBuilder(std::span<char> buffer);
   SimpleStringBuilder(const SimpleStringBuilder&) = delete;
   SimpleStringBuilder& operator=(const SimpleStringBuilder&) = delete;
 
@@ -76,7 +76,7 @@ class SimpleStringBuilder {
   // size allows the buffer to be stack allocated, which helps performance.
   // Having a fixed size is furthermore useful to avoid unnecessary resizing
   // while building it.
-  const ArrayView<char> buffer_;
+  const std::span<char> buffer_;
 
   // Represents the number of characters written to the buffer.
   // This does not include the terminating '\0'.
diff --git a/rtc_base/text2pcap.cc b/rtc_base/text2pcap.cc
index 74f8448faa..8b0e73d20e 100644
--- a/rtc_base/text2pcap.cc
+++ b/rtc_base/text2pcap.cc
@@ -10,16 +10,16 @@
 #include "rtc_base/text2pcap.h"
 
 #include <cstdint>
+#include <span>
 #include <string>
 
-#include "api/array_view.h"
 #include "rtc_base/strings/string_builder.h"
 #include "rtc_base/time_utils.h"
 
 namespace webrtc {
 
 std::string Text2Pcap::DumpPacket(bool outbound,
-                                  ArrayView<const uint8_t> payload,
+                                  std::span<const uint8_t> payload,
                                   int64_t timestamp_ms) {
   webrtc::StringBuilder s;
   s << "\n" << (outbound ? "O " : "I ");
diff --git a/rtc_base/text2pcap.h b/rtc_base/text2pcap.h
index 1f384ec7ed..5826e97ff4 100644
--- a/rtc_base/text2pcap.h
+++ b/rtc_base/text2pcap.h
@@ -11,10 +11,9 @@
 #define RTC_BASE_TEXT2PCAP_H_
 
 #include <cstdint>
+#include <span>
 #include <string>
 
-#include "api/array_view.h"
-
 namespace webrtc {
 
 class Text2Pcap {
@@ -29,7 +28,7 @@ class Text2Pcap {
   // newline and has a grep-able suffix (e.g. ` # SCTP_PACKET` or ` # RTP_DUMP`)
   // for easy extraction from logs.
   static std::string DumpPacket(bool outbound,
-                                ArrayView<const uint8_t> payload,
+                                std::span<const uint8_t> payload,
                                 int64_t timestamp_ms);
 };
 
diff --git a/rtc_base/unique_id_generator.cc b/rtc_base/unique_id_generator.cc
index d09f35bedf..a6cac6b314 100644
--- a/rtc_base/unique_id_generator.cc
+++ b/rtc_base/unique_id_generator.cc
@@ -13,11 +13,11 @@
 #include <cstdint>
 #include <limits>
 #include <optional>
+#include <span>
 #include <string>
 
 #include "absl/strings/str_cat.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/crypto_random.h"
 #include "rtc_base/string_to_number.h"
@@ -26,7 +26,7 @@
 namespace webrtc {
 
 UniqueRandomIdGenerator::UniqueRandomIdGenerator() : known_ids_() {}
-UniqueRandomIdGenerator::UniqueRandomIdGenerator(ArrayView<uint32_t> known_ids)
+UniqueRandomIdGenerator::UniqueRandomIdGenerator(std::span<uint32_t> known_ids)
     : known_ids_(known_ids.begin(), known_ids.end()) {}
 
 UniqueRandomIdGenerator::~UniqueRandomIdGenerator() = default;
@@ -49,7 +49,7 @@ bool UniqueRandomIdGenerator::AddKnownId(uint32_t value) {
 }
 
 UniqueStringGenerator::UniqueStringGenerator() : unique_number_generator_() {}
-UniqueStringGenerator::UniqueStringGenerator(ArrayView<std::string> known_ids) {
+UniqueStringGenerator::UniqueStringGenerator(std::span<std::string> known_ids) {
   for (const std::string& str : known_ids) {
     AddKnownId(str);
   }
diff --git a/rtc_base/unique_id_generator.h b/rtc_base/unique_id_generator.h
index 64491c4eba..4dd31ee261 100644
--- a/rtc_base/unique_id_generator.h
+++ b/rtc_base/unique_id_generator.h
@@ -14,11 +14,11 @@
 #include <cstdint>
 #include <limits>
 #include <set>
+#include <span>
 #include <string>
 #include <type_traits>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/sequence_checker.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/synchronization/mutex.h"
@@ -41,7 +41,7 @@ class UniqueNumberGenerator {
   typedef TIntegral value_type;
   UniqueNumberGenerator();
   // Creates a generator that will never return any value from the given list.
-  explicit UniqueNumberGenerator(ArrayView<TIntegral> known_ids);
+  explicit UniqueNumberGenerator(std::span<TIntegral> known_ids);
   ~UniqueNumberGenerator();
 
   // Generates a number that this generator has never produced before.
@@ -74,7 +74,7 @@ class UniqueRandomIdGenerator {
   typedef uint32_t value_type;
   UniqueRandomIdGenerator();
   // Create a generator that will never return any value from the given list.
-  explicit UniqueRandomIdGenerator(ArrayView<uint32_t> known_ids);
+  explicit UniqueRandomIdGenerator(std::span<uint32_t> known_ids);
   ~UniqueRandomIdGenerator();
 
   // Generates a random id that this generator has never produced before.
@@ -105,7 +105,7 @@ class UniqueStringGenerator {
  public:
   typedef std::string value_type;
   UniqueStringGenerator();
-  explicit UniqueStringGenerator(ArrayView<std::string> known_ids);
+  explicit UniqueStringGenerator(std::span<std::string> known_ids);
   ~UniqueStringGenerator();
 
   std::string GenerateString();
@@ -126,7 +126,7 @@ UniqueNumberGenerator<TIntegral>::UniqueNumberGenerator() : counter_(0) {}
 
 template <typename TIntegral>
 UniqueNumberGenerator<TIntegral>::UniqueNumberGenerator(
-    ArrayView<TIntegral> known_ids)
+    std::span<TIntegral> known_ids)
     : counter_(0), known_ids_(known_ids.begin(), known_ids.end()) {}
 
 template <typename TIntegral>
diff --git a/rtc_base/virtual_socket_unittest.cc b/rtc_base/virtual_socket_unittest.cc
index 8a7c806263..e3081b8c84 100644
--- a/rtc_base/virtual_socket_unittest.cc
+++ b/rtc_base/virtual_socket_unittest.cc
@@ -17,11 +17,11 @@
 #include <cstring>
 #include <ctime>
 #include <memory>
+#include <span>
 #include <utility>
 #include <vector>
 
 #include "absl/memory/memory.h"
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/transport/ecn_marking.h"
 #include "api/units/time_delta.h"
@@ -186,8 +186,8 @@ class VirtualSocketServerTest : public ::testing::Test {
     } else if (post_ip.family() == AF_INET6) {
       in6_addr post_ip6 = post_ip.ipv6_address();
       in6_addr pre_ip6 = pre_ip.ipv6_address();
-      ArrayView<uint8_t, 16> post_bytes = post_ip6.s6_addr;
-      ArrayView<uint8_t, 16> pre_bytes = pre_ip6.s6_addr;
+      std::span<uint8_t, 16> post_bytes = post_ip6.s6_addr;
+      std::span<uint8_t, 16> pre_bytes = pre_ip6.s6_addr;
       EXPECT_THAT(post_bytes.subspan(12, 4),
                   Pointwise(Eq(), pre_bytes.subspan(12, 4)));
     }
diff --git a/rtc_base/zero_memory.h b/rtc_base/zero_memory.h
index 6229a1e3f6..5af33ad8c8 100644
--- a/rtc_base/zero_memory.h
+++ b/rtc_base/zero_memory.h
@@ -13,10 +13,9 @@
 
 #include <stddef.h>
 
+#include <span>
 #include <type_traits>
 
-#include "api/array_view.h"
-
 namespace webrtc {
 
 // Fill memory with zeros in a way that the compiler doesn't optimize it away
@@ -26,7 +25,7 @@ void ExplicitZeroMemory(void* ptr, size_t len);
 template <typename T,
           typename std::enable_if<!std::is_const<T>::value &&
                                   std::is_trivial<T>::value>::type* = nullptr>
-void ExplicitZeroMemory(ArrayView<T> a) {
+void ExplicitZeroMemory(std::span<T> a) {
   ExplicitZeroMemory(a.data(), a.size());
 }
 
diff --git a/rtc_base/zero_memory_unittest.cc b/rtc_base/zero_memory_unittest.cc
index 5a8804c216..dde2898d6c 100644
--- a/rtc_base/zero_memory_unittest.cc
+++ b/rtc_base/zero_memory_unittest.cc
@@ -12,8 +12,8 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "test/gtest.h"
 
 namespace webrtc {
@@ -30,13 +30,13 @@ TEST(ZeroMemoryTest, TestZeroMemory) {
   }
 }
 
-TEST(ZeroMemoryTest, TestZeroArrayView) {
+TEST(ZeroMemoryTest, TestZeroSpan) {
   static const size_t kBufferSize = 32;
   uint8_t buffer[kBufferSize];
   for (size_t i = 0; i < kBufferSize; i++) {
     buffer[i] = static_cast<uint8_t>(i + 1);
   }
-  ExplicitZeroMemory(ArrayView<uint8_t>(buffer, sizeof(buffer)));
+  ExplicitZeroMemory(std::span<uint8_t>(buffer, sizeof(buffer)));
   for (size_t i = 0; i < kBufferSize; i++) {
     EXPECT_EQ(buffer[i], 0);
   }

From 2018a4f9edfa804ec15f53ad3136ab5fcc9b1412 Mon Sep 17 00:00:00 2001
From: Fredrik Solenberg <solenberg@google.com>
Date: Tue, 10 Mar 2026 14:23:12 +0000
Subject: [PATCH 255/472] Consolidate FakeAudioStream and FakeVideoStream
 classes for tests.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Bug: webrtc:42233500
Change-Id: I31fc6151e38087436227ed21f4a219d784b9f651
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/455241
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47108}
---
 pc/BUILD.gn                           |   2 +
 pc/legacy_stats_collector_unittest.cc | 110 ++++-------------
 pc/rtc_stats_collector_unittest.cc    | 167 +++++---------------------
 pc/test/fake_audio_track.h            |  79 ++++++++++++
 pc/test/fake_video_track.h            |  60 +++++++++
 pc/test/fake_video_track_source.h     |  15 +++
 6 files changed, 210 insertions(+), 223 deletions(-)
 create mode 100644 pc/test/fake_audio_track.h
 create mode 100644 pc/test/fake_video_track.h

diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 20e7f5d8e2..9863aa6da6 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -2903,12 +2903,14 @@ if (rtc_include_tests && !build_with_chromium) {
     sources = [
       "test/fake_audio_capture_module.cc",
       "test/fake_audio_capture_module.h",
+      "test/fake_audio_track.h",
       "test/fake_data_channel_controller.h",
       "test/fake_peer_connection_base.h",
       "test/fake_peer_connection_for_stats.h",
       "test/fake_periodic_video_source.h",
       "test/fake_periodic_video_track_source.h",
       "test/fake_rtc_certificate_generator.h",
+      "test/fake_video_track.h",
       "test/fake_video_track_renderer.h",
       "test/fake_video_track_source.h",
       "test/frame_generator_capturer_video_track_source.h",
diff --git a/pc/legacy_stats_collector_unittest.cc b/pc/legacy_stats_collector_unittest.cc
index 676b2eec49..63c33ec177 100644
--- a/pc/legacy_stats_collector_unittest.cc
+++ b/pc/legacy_stats_collector_unittest.cc
@@ -28,7 +28,6 @@
 #include "api/legacy_stats_types.h"
 #include "api/make_ref_counted.h"
 #include "api/media_stream_interface.h"
-#include "api/media_stream_track.h"
 #include "api/media_types.h"
 #include "api/peer_connection_interface.h"
 #include "api/rtp_sender_interface.h"
@@ -42,6 +41,7 @@
 #include "pc/media_stream.h"
 #include "pc/peer_connection_internal.h"
 #include "pc/sctp_data_channel.h"
+#include "pc/test/fake_audio_track.h"
 #include "pc/test/fake_peer_connection_for_stats.h"
 #include "pc/test/fake_video_track_source.h"
 #include "pc/test/mock_rtp_receiver_internal.h"
@@ -85,82 +85,20 @@ constexpr char kLocalTrackId[] = "local_track_id";
 constexpr char kRemoteTrackId[] = "remote_track_id";
 constexpr uint32_t kSsrcOfTrack = 1234;
 
-class FakeAudioProcessor : public AudioProcessorInterface {
- public:
-  FakeAudioProcessor() {}
-  ~FakeAudioProcessor() override {}
-
- private:
-  AudioProcessorInterface::AudioProcessorStatistics GetStats(
-      bool has_recv_streams) override {
-    AudioProcessorStatistics stats;
-    if (has_recv_streams) {
-      stats.apm_statistics.echo_return_loss = 2.0;
-      stats.apm_statistics.echo_return_loss_enhancement = 3.0;
-      stats.apm_statistics.delay_median_ms = 4;
-      stats.apm_statistics.delay_standard_deviation_ms = 5;
-    }
-    return stats;
+// Returns a FakeAudioTrack with the given ID and configuration.
+scoped_refptr<AudioTrackInterface> CreateFakeAudioTrack(const std::string& id,
+                                                        bool return_stats) {
+  auto processor = make_ref_counted<FakeAudioProcessor>();
+  processor->return_stats = return_stats;
+  if (return_stats) {
+    processor->stats.apm_statistics.echo_return_loss = 2.0;
+    processor->stats.apm_statistics.echo_return_loss_enhancement = 3.0;
+    processor->stats.apm_statistics.delay_median_ms = 4;
+    processor->stats.apm_statistics.delay_standard_deviation_ms = 5;
   }
-};
-
-class FakeAudioTrack : public MediaStreamTrack<AudioTrackInterface> {
- public:
-  explicit FakeAudioTrack(const std::string& id)
-      : MediaStreamTrack<AudioTrackInterface>(id),
-        processor_(make_ref_counted<FakeAudioProcessor>()) {}
-  std::string kind() const override { return "audio"; }
-  AudioSourceInterface* GetSource() const override { return nullptr; }
-  void AddSink(AudioTrackSinkInterface* sink) override {}
-  void RemoveSink(AudioTrackSinkInterface* sink) override {}
-  bool GetSignalLevel(int* level) override {
-    *level = 1;
-    return true;
-  }
-  scoped_refptr<AudioProcessorInterface> GetAudioProcessor() override {
-    return processor_;
-  }
-
- private:
-  scoped_refptr<FakeAudioProcessor> processor_;
-};
-
-// This fake audio processor is used to verify that the undesired initial values
-// (-1) will be filtered out.
-class FakeAudioProcessorWithInitValue : public AudioProcessorInterface {
- public:
-  FakeAudioProcessorWithInitValue() {}
-  ~FakeAudioProcessorWithInitValue() override {}
-
- private:
-  AudioProcessorInterface::AudioProcessorStatistics GetStats(
-      bool /*has_recv_streams*/) override {
-    AudioProcessorStatistics stats;
-    return stats;
-  }
-};
-
-class FakeAudioTrackWithInitValue
-    : public MediaStreamTrack<AudioTrackInterface> {
- public:
-  explicit FakeAudioTrackWithInitValue(const std::string& id)
-      : MediaStreamTrack<AudioTrackInterface>(id),
-        processor_(make_ref_counted<FakeAudioProcessorWithInitValue>()) {}
-  std::string kind() const override { return "audio"; }
-  AudioSourceInterface* GetSource() const override { return nullptr; }
-  void AddSink(AudioTrackSinkInterface* sink) override {}
-  void RemoveSink(AudioTrackSinkInterface* sink) override {}
-  bool GetSignalLevel(int* level) override {
-    *level = 1;
-    return true;
-  }
-  scoped_refptr<AudioProcessorInterface> GetAudioProcessor() override {
-    return processor_;
-  }
-
- private:
-  scoped_refptr<FakeAudioProcessorWithInitValue> processor_;
-};
+  return FakeAudioTrack::Create(id, MediaStreamTrackInterface::kLive,
+                                processor);
+}
 
 bool GetValue(const StatsReport* report,
               StatsReport::StatsValueName name,
@@ -600,7 +538,7 @@ class LegacyStatsCollectorTest : public ::testing::Test {
         pc, clock_, [this]() { return GetUtcTimeNowMs(); });
   }
 
-  void VerifyAudioTrackStats(FakeAudioTrack* audio_track,
+  void VerifyAudioTrackStats(AudioTrackInterface* audio_track,
                              LegacyStatsCollector* stats,
                              const VoiceMediaInfo& voice_info,
                              StatsReports* reports) {
@@ -841,7 +779,7 @@ class StatsCollectorTrackTest : public LegacyStatsCollectorTest,
   scoped_refptr<RtpSenderInterface> AddOutgoingAudioTrack(
       FakePeerConnectionForStats* pc,
       LegacyStatsCollector* stats) {
-    audio_track_ = make_ref_counted<FakeAudioTrack>(kLocalTrackId);
+    audio_track_ = CreateFakeAudioTrack(kLocalTrackId, true);
     if (GetParam()) {
       if (!stream_)
         stream_ = MediaStream::Create("streamid");
@@ -859,7 +797,7 @@ class StatsCollectorTrackTest : public LegacyStatsCollectorTest,
   // Adds a incoming audio track with a given SSRC into the stats.
   void AddIncomingAudioTrack(FakePeerConnectionForStats* pc,
                              LegacyStatsCollector* stats) {
-    audio_track_ = make_ref_counted<FakeAudioTrack>(kRemoteTrackId);
+    audio_track_ = CreateFakeAudioTrack(kRemoteTrackId, true);
     if (GetParam()) {
       if (stream_ == nullptr)
         stream_ = MediaStream::Create("streamid");
@@ -878,7 +816,7 @@ class StatsCollectorTrackTest : public LegacyStatsCollectorTest,
 
   scoped_refptr<MediaStream> stream_;
   scoped_refptr<VideoTrack> video_track_;
-  scoped_refptr<FakeAudioTrack> audio_track_;
+  scoped_refptr<AudioTrackInterface> audio_track_;
 };
 
 TEST(StatsCollectionTest, DetachAndMerge) {
@@ -1593,8 +1531,7 @@ TEST_P(StatsCollectorTrackTest, FilterOutNegativeInitialValues) {
 
   // Create a local stream with a local audio track and adds it to the stats.
   stream_ = MediaStream::Create("streamid");
-  auto local_track =
-      make_ref_counted<FakeAudioTrackWithInitValue>(kLocalTrackId);
+  auto local_track = CreateFakeAudioTrack(kLocalTrackId, false);
   stream_->AddTrack(scoped_refptr<AudioTrackInterface>(local_track.get()));
   RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc->AddSender(CreateMockSender(local_track, kSsrcOfTrack));
@@ -1608,7 +1545,7 @@ TEST_P(StatsCollectorTrackTest, FilterOutNegativeInitialValues) {
   scoped_refptr<MediaStream> remote_stream(
       MediaStream::Create("remotestreamid"));
   scoped_refptr<AudioTrackInterface> remote_track =
-      make_ref_counted<FakeAudioTrackWithInitValue>(kRemoteTrackId);
+      CreateFakeAudioTrack(kRemoteTrackId, false);
   remote_stream->AddTrack(remote_track);
   RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc->AddReceiver(CreateMockReceiver(remote_track, kSsrcOfTrack));
@@ -1793,7 +1730,7 @@ TEST_P(StatsCollectorTrackTest, LocalAndRemoteTracksWithSameSsrc) {
   scoped_refptr<MediaStream> remote_stream(
       MediaStream::Create("remotestreamid"));
   scoped_refptr<AudioTrackInterface> remote_track =
-      make_ref_counted<FakeAudioTrack>(kRemoteTrackId);
+      CreateFakeAudioTrack(kRemoteTrackId, true);
   RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc->AddReceiver(CreateMockReceiver(remote_track, kSsrcOfTrack));
   RTC_ALLOW_PLAN_B_DEPRECATION_END();
@@ -1893,7 +1830,7 @@ TEST_P(StatsCollectorTrackTest, TwoLocalTracksWithSameSsrc) {
 
   // Create a new audio track and adds it to the stream and stats.
   static const std::string kNewTrackId = "new_track_id";
-  auto new_audio_track = make_ref_counted<FakeAudioTrack>(kNewTrackId);
+  auto new_audio_track = CreateFakeAudioTrack(kNewTrackId, true);
   RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc->AddSender(CreateMockSender(new_audio_track, kSsrcOfTrack));
   RTC_ALLOW_PLAN_B_DEPRECATION_END();
@@ -1926,8 +1863,7 @@ TEST_P(StatsCollectorTrackTest, TwoLocalSendersWithSameTrack) {
   auto pc = CreatePeerConnection();
   auto stats = CreateStatsCollector(pc.get());
 
-  auto local_track =
-      make_ref_counted<FakeAudioTrackWithInitValue>(kLocalTrackId);
+  auto local_track = CreateFakeAudioTrack(kLocalTrackId, false);
   RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
   pc->AddSender(CreateMockSender(local_track, kFirstSsrc));
   stats->AddLocalAudioTrack(local_track.get(), kFirstSsrc);
diff --git a/pc/rtc_stats_collector_unittest.cc b/pc/rtc_stats_collector_unittest.cc
index 4c69afd636..55f19763d2 100644
--- a/pc/rtc_stats_collector_unittest.cc
+++ b/pc/rtc_stats_collector_unittest.cc
@@ -32,7 +32,6 @@
 #include "api/environment/environment.h"
 #include "api/make_ref_counted.h"
 #include "api/media_stream_interface.h"
-#include "api/media_stream_track.h"
 #include "api/media_types.h"
 #include "api/peer_connection_interface.h"
 #include "api/rtp_parameters.h"
@@ -48,11 +47,7 @@
 #include "api/units/data_rate.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
-#include "api/video/recordable_encoded_frame.h"
 #include "api/video/video_content_type.h"
-#include "api/video/video_frame.h"
-#include "api/video/video_sink_interface.h"
-#include "api/video/video_source_interface.h"
 #include "api/video/video_timing.h"
 #include "api/video_codecs/scalability_mode.h"
 #include "call/call.h"
@@ -72,8 +67,11 @@
 #include "pc/peer_connection_internal.h"
 #include "pc/sctp_data_channel.h"
 #include "pc/stream_collection.h"
+#include "pc/test/fake_audio_track.h"
 #include "pc/test/fake_data_channel_controller.h"
 #include "pc/test/fake_peer_connection_for_stats.h"
+#include "pc/test/fake_video_track.h"
+#include "pc/test/fake_video_track_source.h"
 #include "pc/test/mock_data_channel.h"
 #include "pc/test/mock_rtp_receiver_internal.h"
 #include "pc/test/mock_rtp_sender_internal.h"
@@ -199,129 +197,26 @@ class MockStatsCollectorCallback : public RTCStatsCollectorCallback {
               (override));
 };
 
-class FakeAudioProcessor : public AudioProcessorInterface {
- public:
-  FakeAudioProcessor() {}
-  ~FakeAudioProcessor() override {}
-
- private:
-  AudioProcessorInterface::AudioProcessorStatistics GetStats(
-      bool has_recv_streams) override {
-    AudioProcessorStatistics stats;
-    stats.apm_statistics.echo_return_loss = 2.0;
-    stats.apm_statistics.echo_return_loss_enhancement = 3.0;
-    return stats;
+// Helper functions to create fake tracks for stats tests.
+scoped_refptr<AudioTrackInterface> CreateFakeAudioTrackForStats(
+    const std::string& id,
+    MediaStreamTrackInterface::TrackState state,
+    bool create_fake_audio_processor) {
+  scoped_refptr<FakeAudioProcessor> processor = nullptr;
+  if (create_fake_audio_processor) {
+    processor = make_ref_counted<FakeAudioProcessor>();
+    processor->stats.apm_statistics.echo_return_loss = 2.0;
+    processor->stats.apm_statistics.echo_return_loss_enhancement = 3.0;
   }
-};
+  return FakeAudioTrack::Create(id, state, processor);
+}
 
-class FakeAudioTrackForStats : public MediaStreamTrack<AudioTrackInterface> {
- public:
-  static scoped_refptr<FakeAudioTrackForStats> Create(
-      const std::string& id,
-      MediaStreamTrackInterface::TrackState state,
-      bool create_fake_audio_processor) {
-    auto audio_track_stats = make_ref_counted<FakeAudioTrackForStats>(id);
-    audio_track_stats->set_state(state);
-    if (create_fake_audio_processor) {
-      audio_track_stats->processor_ = make_ref_counted<FakeAudioProcessor>();
-    }
-    return audio_track_stats;
-  }
-
-  explicit FakeAudioTrackForStats(const std::string& id)
-      : MediaStreamTrack<AudioTrackInterface>(id) {}
-
-  std::string kind() const override {
-    return MediaStreamTrackInterface::kAudioKind;
-  }
-  AudioSourceInterface* GetSource() const override { return nullptr; }
-  void AddSink(AudioTrackSinkInterface* sink) override {}
-  void RemoveSink(AudioTrackSinkInterface* sink) override {}
-  bool GetSignalLevel(int* level) override { return false; }
-  scoped_refptr<AudioProcessorInterface> GetAudioProcessor() override {
-    return processor_;
-  }
-
- private:
-  scoped_refptr<FakeAudioProcessor> processor_;
-};
-
-class FakeVideoTrackSourceForStats : public VideoTrackSourceInterface {
- public:
-  static scoped_refptr<FakeVideoTrackSourceForStats> Create(int input_width,
-                                                            int input_height) {
-    return make_ref_counted<FakeVideoTrackSourceForStats>(input_width,
-                                                          input_height);
-  }
-
-  FakeVideoTrackSourceForStats(int input_width, int input_height)
-      : input_width_(input_width), input_height_(input_height) {}
-  ~FakeVideoTrackSourceForStats() override {}
-
-  // VideoTrackSourceInterface
-  bool is_screencast() const override { return false; }
-  std::optional<bool> needs_denoising() const override { return false; }
-  bool GetStats(VideoTrackSourceInterface::Stats* stats) override {
-    stats->input_width = input_width_;
-    stats->input_height = input_height_;
-    return true;
-  }
-  // MediaSourceInterface (part of VideoTrackSourceInterface)
-  MediaSourceInterface::SourceState state() const override {
-    return MediaSourceInterface::SourceState::kLive;
-  }
-  bool remote() const override { return false; }
-  // NotifierInterface (part of MediaSourceInterface)
-  void RegisterObserver(ObserverInterface* observer) override {}
-  void UnregisterObserver(ObserverInterface* observer) override {}
-  // webrtc::VideoSourceInterface<VideoFrame> (part of
-  // VideoTrackSourceInterface)
-  void AddOrUpdateSink(VideoSinkInterface<VideoFrame>* sink,
-                       const VideoSinkWants& wants) override {}
-  void RemoveSink(VideoSinkInterface<VideoFrame>* sink) override {}
-  bool SupportsEncodedOutput() const override { return false; }
-  void GenerateKeyFrame() override {}
-  void AddEncodedSink(
-      VideoSinkInterface<RecordableEncodedFrame>* sink) override {}
-  void RemoveEncodedSink(
-      VideoSinkInterface<RecordableEncodedFrame>* sink) override {}
-
- private:
-  int input_width_;
-  int input_height_;
-};
-
-class FakeVideoTrackForStats : public MediaStreamTrack<VideoTrackInterface> {
- public:
-  static scoped_refptr<FakeVideoTrackForStats> Create(
-      const std::string& id,
-      MediaStreamTrackInterface::TrackState state,
-      scoped_refptr<VideoTrackSourceInterface> source) {
-    auto video_track =
-        make_ref_counted<FakeVideoTrackForStats>(id, std::move(source));
-    video_track->set_state(state);
-    return video_track;
-  }
-
-  FakeVideoTrackForStats(const std::string& id,
-                         scoped_refptr<VideoTrackSourceInterface> source)
-      : MediaStreamTrack<VideoTrackInterface>(id), source_(source) {}
-
-  std::string kind() const override {
-    return MediaStreamTrackInterface::kVideoKind;
-  }
-
-  void AddOrUpdateSink(VideoSinkInterface<VideoFrame>* sink,
-                       const VideoSinkWants& wants) override {}
-  void RemoveSink(VideoSinkInterface<VideoFrame>* sink) override {}
-
-  VideoTrackSourceInterface* GetSource() const override {
-    return source_.get();
-  }
-
- private:
-  scoped_refptr<VideoTrackSourceInterface> source_;
-};
+scoped_refptr<VideoTrackInterface> CreateFakeVideoTrackForStats(
+    const std::string& id,
+    MediaStreamTrackInterface::TrackState state,
+    scoped_refptr<VideoTrackSourceInterface> source) {
+  return FakeVideoTrack::Create(id, state, source);
+}
 
 scoped_refptr<MediaStreamTrackInterface> CreateFakeTrack(
     MediaType media_type,
@@ -329,11 +224,11 @@ scoped_refptr<MediaStreamTrackInterface> CreateFakeTrack(
     MediaStreamTrackInterface::TrackState track_state,
     bool create_fake_audio_processor = false) {
   if (media_type == MediaType::AUDIO) {
-    return FakeAudioTrackForStats::Create(track_id, track_state,
-                                          create_fake_audio_processor);
+    return CreateFakeAudioTrackForStats(track_id, track_state,
+                                        create_fake_audio_processor);
   } else {
     RTC_DCHECK_EQ(media_type, MediaType::VIDEO);
-    return FakeVideoTrackForStats::Create(track_id, track_state, nullptr);
+    return CreateFakeVideoTrackForStats(track_id, track_state, nullptr);
   }
 }
 
@@ -3261,9 +3156,9 @@ TEST_P(RTCStatsCollectorTest, RTCVideoSourceStatsCollectedForSenderWithTrack) {
   pc_->AddVideoChannel("VideoMid", "TransportName", video_media_info);
   RTC_ALLOW_PLAN_B_DEPRECATION_END();
 
-  auto video_source = FakeVideoTrackSourceForStats::Create(kVideoSourceWidth,
-                                                           kVideoSourceHeight);
-  auto video_track = FakeVideoTrackForStats::Create(
+  auto video_source = FakeVideoTrackSource::Create(false);
+  video_source->SetSize(kVideoSourceWidth, kVideoSourceHeight);
+  auto video_track = CreateFakeVideoTrackForStats(
       "LocalVideoTrackID", MediaStreamTrackInterface::kLive, video_source);
   scoped_refptr<MockRtpSenderInternal> sender =
       CreateMockSender(MediaType::VIDEO, video_track, kSsrc, kAttachmentId, {});
@@ -3313,9 +3208,9 @@ TEST_P(RTCStatsCollectorTest,
   pc_->AddVideoChannel("VideoMid", "TransportName", video_media_info);
   RTC_ALLOW_PLAN_B_DEPRECATION_END();
 
-  auto video_source = FakeVideoTrackSourceForStats::Create(kVideoSourceWidth,
-                                                           kVideoSourceHeight);
-  auto video_track = FakeVideoTrackForStats::Create(
+  auto video_source = FakeVideoTrackSource::Create(false);
+  video_source->SetSize(kVideoSourceWidth, kVideoSourceHeight);
+  auto video_track = CreateFakeVideoTrackForStats(
       "LocalVideoTrackID", MediaStreamTrackInterface::kLive, video_source);
   scoped_refptr<MockRtpSenderInternal> sender = CreateMockSender(
       MediaType::VIDEO, video_track, kNoSsrc, kAttachmentId, {});
@@ -3352,7 +3247,7 @@ TEST_P(RTCStatsCollectorTest,
   pc_->AddVideoChannel("VideoMid", "TransportName", video_media_info);
   RTC_ALLOW_PLAN_B_DEPRECATION_END();
 
-  auto video_track = FakeVideoTrackForStats::Create(
+  auto video_track = CreateFakeVideoTrackForStats(
       "LocalVideoTrackID", MediaStreamTrackInterface::kLive,
       /*source=*/nullptr);
   scoped_refptr<MockRtpSenderInternal> sender =
diff --git a/pc/test/fake_audio_track.h b/pc/test/fake_audio_track.h
new file mode 100644
index 0000000000..42b001686e
--- /dev/null
+++ b/pc/test/fake_audio_track.h
@@ -0,0 +1,79 @@
+/*
+ *  Copyright 2026 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef PC_TEST_FAKE_AUDIO_TRACK_H_
+#define PC_TEST_FAKE_AUDIO_TRACK_H_
+
+#include <string>
+
+#include "api/make_ref_counted.h"
+#include "api/media_stream_interface.h"
+#include "api/media_stream_track.h"
+#include "api/scoped_refptr.h"
+
+namespace webrtc {
+
+class FakeAudioProcessor : public AudioProcessorInterface {
+ public:
+  FakeAudioProcessor() = default;
+  ~FakeAudioProcessor() override = default;
+
+  AudioProcessorInterface::AudioProcessorStatistics GetStats(
+      bool /*has_recv_streams*/) override {
+    return return_stats ? stats
+                        : AudioProcessorInterface::AudioProcessorStatistics();
+  }
+
+  AudioProcessorInterface::AudioProcessorStatistics stats;
+  bool return_stats = true;
+};
+
+class FakeAudioTrack : public MediaStreamTrack<AudioTrackInterface> {
+ public:
+  static scoped_refptr<FakeAudioTrack> Create(
+      const std::string& id,
+      MediaStreamTrackInterface::TrackState state,
+      scoped_refptr<FakeAudioProcessor> processor = nullptr) {
+    auto audio_track = make_ref_counted<FakeAudioTrack>(id, processor);
+    audio_track->set_state(state);
+    return audio_track;
+  }
+
+  explicit FakeAudioTrack(const std::string& id,
+                          scoped_refptr<FakeAudioProcessor> processor = nullptr)
+      : MediaStreamTrack<AudioTrackInterface>(id),
+        processor_(processor ? processor
+                             : make_ref_counted<FakeAudioProcessor>()) {}
+
+  std::string kind() const override {
+    return MediaStreamTrackInterface::kAudioKind;
+  }
+  AudioSourceInterface* GetSource() const override { return nullptr; }
+  void AddSink(AudioTrackSinkInterface* sink) override {}
+  void RemoveSink(AudioTrackSinkInterface* sink) override {}
+  bool GetSignalLevel(int* level) override {
+    *level = 1;
+    return true;
+  }
+  scoped_refptr<AudioProcessorInterface> GetAudioProcessor() override {
+    return processor_;
+  }
+
+  void set_processor(scoped_refptr<AudioProcessorInterface> processor) {
+    processor_ = processor;
+  }
+
+ private:
+  scoped_refptr<AudioProcessorInterface> processor_;
+};
+
+}  // namespace webrtc
+
+#endif  // PC_TEST_FAKE_AUDIO_TRACK_H_
diff --git a/pc/test/fake_video_track.h b/pc/test/fake_video_track.h
new file mode 100644
index 0000000000..ce11982710
--- /dev/null
+++ b/pc/test/fake_video_track.h
@@ -0,0 +1,60 @@
+/*
+ *  Copyright 2026 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef PC_TEST_FAKE_VIDEO_TRACK_H_
+#define PC_TEST_FAKE_VIDEO_TRACK_H_
+
+#include <string>
+#include <utility>
+
+#include "api/make_ref_counted.h"
+#include "api/media_stream_interface.h"
+#include "api/media_stream_track.h"
+#include "api/scoped_refptr.h"
+#include "api/video/video_frame.h"
+#include "api/video/video_sink_interface.h"
+#include "api/video/video_source_interface.h"
+
+namespace webrtc {
+
+class FakeVideoTrack : public MediaStreamTrack<VideoTrackInterface> {
+ public:
+  static scoped_refptr<FakeVideoTrack> Create(
+      const std::string& id,
+      MediaStreamTrackInterface::TrackState state,
+      scoped_refptr<VideoTrackSourceInterface> source) {
+    auto video_track = make_ref_counted<FakeVideoTrack>(id, std::move(source));
+    video_track->set_state(state);
+    return video_track;
+  }
+
+  FakeVideoTrack(const std::string& id,
+                 scoped_refptr<VideoTrackSourceInterface> source)
+      : MediaStreamTrack<VideoTrackInterface>(id), source_(source) {}
+
+  std::string kind() const override {
+    return MediaStreamTrackInterface::kVideoKind;
+  }
+
+  void AddOrUpdateSink(VideoSinkInterface<VideoFrame>* sink,
+                       const VideoSinkWants& wants) override {}
+  void RemoveSink(VideoSinkInterface<VideoFrame>* sink) override {}
+
+  VideoTrackSourceInterface* GetSource() const override {
+    return source_.get();
+  }
+
+ private:
+  scoped_refptr<VideoTrackSourceInterface> source_;
+};
+
+}  // namespace webrtc
+
+#endif  // PC_TEST_FAKE_VIDEO_TRACK_H_
diff --git a/pc/test/fake_video_track_source.h b/pc/test/fake_video_track_source.h
index df56f625ca..241f73a5a5 100644
--- a/pc/test/fake_video_track_source.h
+++ b/pc/test/fake_video_track_source.h
@@ -12,6 +12,7 @@
 #define PC_TEST_FAKE_VIDEO_TRACK_SOURCE_H_
 
 #include "api/make_ref_counted.h"
+#include "api/media_stream_interface.h"
 #include "api/scoped_refptr.h"
 #include "api/video/video_frame.h"
 #include "api/video/video_source_interface.h"
@@ -31,6 +32,18 @@ class FakeVideoTrackSource : public VideoTrackSource {
   static scoped_refptr<FakeVideoTrackSource> Create() { return Create(false); }
 
   bool is_screencast() const override { return is_screencast_; }
+  bool GetStats(VideoTrackSourceInterface::Stats* stats) override {
+    if (!stats)
+      return false;
+    stats->input_width = width_;
+    stats->input_height = height_;
+    return true;
+  }
+
+  void SetSize(int width, int height) {
+    width_ = width;
+    height_ = height;
+  }
 
   void InjectFrame(const VideoFrame& frame) {
     video_broadcaster_.OnFrame(frame);
@@ -47,6 +60,8 @@ class FakeVideoTrackSource : public VideoTrackSource {
 
  private:
   const bool is_screencast_;
+  int width_ = 0;
+  int height_ = 0;
   VideoBroadcaster video_broadcaster_;
 };
 

From c34d367f3763beca8af91b74b5c62faf4c88f4eb Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Tue, 10 Mar 2026 09:04:37 -0700
Subject: [PATCH 256/472] Roll chromium_revision 07798bc2eb..5a3b6f09e6
 (1596985:1597105)

Change log: https://chromium.googlesource.com/chromium/src/+log/07798bc2eb..5a3b6f09e6
Full diff: https://chromium.googlesource.com/chromium/src/+/07798bc2eb..5a3b6f09e6

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/98d6658ed1..7d29c62736
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/632417adc3..6ef5c9aa5d
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/07d5e4bcb0..4d76c99bba
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/cca29c0f05..272f0dea53
* src/third_party/androidx/cipd: _YK_XyeZvkOivG1aqms80wS3mgx5K-L9yHOhZDzvlI4C..o4DXbfi661jCyVTmGyk5pLrrimnXDqgV-d8-dxfdZ_AC
* src/third_party/freetype/src: https://chromium.googlesource.com/chromium/src/third_party/freetype2.git/+log/1c6e43ab90..dc17b79e1a
* src/third_party/fuzztest/src: https://chromium.googlesource.com/external/github.com/google/fuzztest.git/+log/e58f8dd957..b6d8465706
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/5268fe512e..8464e2906a
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/9e8a98181c..d3a6eb533e
DEPS diff: https://chromium.googlesource.com/chromium/src/+/07798bc2eb..5a3b6f09e6/DEPS

BUG=None

Change-Id: I36885b98fdf13ffe35bc4c3bcb65ef5e6b571645
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/455580
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47109}
---
 DEPS | 20 ++++++++++----------
 1 file changed, 10 insertions(+), 10 deletions(-)

diff --git a/DEPS b/DEPS
index d552d47ba5..b331a7b8c8 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '07798bc2eb402da8fcfaa15692a487a2312b4320',
+  'chromium_revision': '5a3b6f09e6b6f23679e127a9c95db9e34c8acaf3',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@98d6658ed11a1b5abc2d29feec55c51d03861692',
+    'https://chromium.googlesource.com/chromium/src/build@7d29c62736304aeaac07df0b75546ffd2b75917a',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@fef8d6947fa54202e8a00c58d5ac8fd15780bc33',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@632417adc3fb78ab22c12a3951f15f8e55e61080',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@6ef5c9aa5d741427e0a26482175d474df91c8e5f',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@07d5e4bcb0826b49ab4318ff0100332fdc6f4f22',
+    'https://chromium.googlesource.com/chromium/src/testing@4d76c99bbae0d4b04478317d8f150ad0f4c8e871',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@cca29c0f05afea0156c8f7946ed402af2573d144',
+    'https://chromium.googlesource.com/chromium/src/third_party@272f0dea537684cb152ccc8a6af582dd6b1fa367',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -519,7 +519,7 @@ deps = {
       'condition': 'checkout_linux',
   },
   'src/third_party/freetype/src':
-    'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@1c6e43ab90af00261334c6cd214a5a33178eacd4',
+    'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@dc17b79e1acec15b1dcbfb669a4a46db08cd4656',
   'src/third_party/harfbuzz-ng/src':
     'https://chromium.googlesource.com/external/github.com/harfbuzz/harfbuzz.git@fa2908bf16d2ccd6623f4d575455fea72a1a722b',
   'src/third_party/google_benchmark/src': {
@@ -593,7 +593,7 @@ deps = {
   'src/third_party/libFuzzer/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt/lib/fuzzer.git@bea408a6e01f0f7e6c82a43121fe3af4506c932e',
   'src/third_party/fuzztest/src':
-    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@e58f8dd9577523f636704a96b907137f36fd4b5b',
+    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@b6d8465706537cdd7e4cf89c9eb5587b5e16b96e',
   'src/third_party/libjpeg_turbo':
     'https://chromium.googlesource.com/chromium/deps/libjpeg_turbo.git@d1f5f2393e0d51f840207342ae86e55a86443288',
   'src/third_party/libsrtp':
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '5268fe512ed19082bd833cdea8e5abbba0d0599e',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '8464e2906ac98eb874fe861a0b789755d9e3650d',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@9e8a98181c1b2243facb3a9953ea9b4bfcbdfc41',
+    'https://chromium.googlesource.com/chromium/src/tools@d3a6eb533ef3c871cdaef2d7967de90e233256cf',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': '_YK_XyeZvkOivG1aqms80wS3mgx5K-L9yHOhZDzvlI4C',
+          'version': 'o4DXbfi661jCyVTmGyk5pLrrimnXDqgV-d8-dxfdZ_AC',
       },
     ],
     'condition': 'checkout_android and non_git_source',

From 84ff19ec2e2c2d1c3d9dd7a33e3f81e62c30aa31 Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Tue, 10 Mar 2026 21:04:02 -0700
Subject: [PATCH 257/472] Update WebRTC code version (2026-03-11T04:03:48).

Bug: None
Change-Id: I94b81acb1a93329d3c680cb88be9e524b2261cd7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/455680
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47110}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index 5a346f0c72..2c7bb48be2 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-10T04:05:46";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-11T04:03:48";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From 8281b933c9f9909a6ddfaa9628d4b2694c1ba3d9 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Tue, 10 Mar 2026 23:02:30 -0700
Subject: [PATCH 258/472] Roll chromium_revision 5a3b6f09e6..a5382cea3d
 (1597105:1597531)

Change log: https://chromium.googlesource.com/chromium/src/+log/5a3b6f09e6..a5382cea3d
Full diff: https://chromium.googlesource.com/chromium/src/+/5a3b6f09e6..a5382cea3d

Changed dependencies
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/6ef5c9aa5d..76aa1c78ed
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/4d76c99bba..e7a1e88ecc
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/272f0dea53..256d17dbe5
* src/third_party/androidx/cipd: o4DXbfi661jCyVTmGyk5pLrrimnXDqgV-d8-dxfdZ_AC..uw8FaVpFdPaFx2Y4c1mBUMRwGB52J-PturMUyulx_hkC
* src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/338a5c004c..c8a5676846
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/81550faea4..4387e32481
* src/third_party/freetype/src: https://chromium.googlesource.com/chromium/src/third_party/freetype2.git/+log/dc17b79e1a..99b479dc34
* src/third_party/fuzztest/src: https://chromium.googlesource.com/external/github.com/google/fuzztest.git/+log/b6d8465706..54ed362692
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/5a3a5d5314..50957c5c47
* src/third_party/kotlinc/current: JEwTOoywRjDr5YoLIeG8Qzb3nT7OauqHdhfL6tIbyP0C..rZVNs6pgsRU0LupGqWhLJwl508-H7iam-fnzDW6X37MC
* src/third_party/libunwind/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git/+log/78884e23fe..db838d9185
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/8464e2906a..4cead1b0fc
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/d3a6eb533e..0510739a11
DEPS diff: https://chromium.googlesource.com/chromium/src/+/5a3b6f09e6..a5382cea3d/DEPS

BUG=None

Change-Id: I78152428d945cabaf9dd25aa00ec6791f675bb44
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/455720
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47111}
---
 DEPS | 28 ++++++++++++++--------------
 1 file changed, 14 insertions(+), 14 deletions(-)

diff --git a/DEPS b/DEPS
index b331a7b8c8..bcb8c09b6e 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '5a3b6f09e6b6f23679e127a9c95db9e34c8acaf3',
+  'chromium_revision': 'a5382cea3d06856d6689fa29b66bc31dda86ca70',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@6ef5c9aa5d741427e0a26482175d474df91c8e5f',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@76aa1c78ed7a95f9894283eaa96c0bc34ffa5be4',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@4d76c99bbae0d4b04478317d8f150ad0f4c8e871',
+    'https://chromium.googlesource.com/chromium/src/testing@e7a1e88ecc35a70f3d198674042753ec9e8e3650',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@272f0dea537684cb152ccc8a6af582dd6b1fa367',
+    'https://chromium.googlesource.com/chromium/src/third_party@256d17dbe537bf8fdd8937e1858063b9aa4f3e4f',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -331,7 +331,7 @@ deps = {
   'src/third_party/clang-format/script':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@c2725e0622e1a86d55f14514f2177a39efea4a0e',
   'src/third_party/compiler-rt/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@338a5c004c774a8927899b1f1c0c25a82d14510f',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@c8a567684659fe05ede3a220965b94fd71c4a7cb',
   'src/third_party/libc++/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@7ab65651aed6802d2599dcb7a73b1f82d5179d05',
   'src/third_party/libc++abi/src':
@@ -339,7 +339,7 @@ deps = {
   'src/third_party/llvm-libc/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@8df8a70a8349a709b8996a11279eb226cae5aced',
   'src/third_party/libunwind/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@78884e23fe39cf5cc6987ea188a9b802d65a21c9',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@db838d918570d4e381ecf9f5cc70a0098c9c2cd6',
 
   'src/third_party/test_fonts/test_fonts': {
       'dep_type': 'gcs',
@@ -511,7 +511,7 @@ deps = {
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@5a3a5d53145b94895610825e783a8896a61a3c73',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@50957c5c47de1c85637fc9dafe6d4eb4c1224f6b',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -519,7 +519,7 @@ deps = {
       'condition': 'checkout_linux',
   },
   'src/third_party/freetype/src':
-    'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@dc17b79e1acec15b1dcbfb669a4a46db08cd4656',
+    'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@99b479dc34728936b006679a31e12b8cf432fc55',
   'src/third_party/harfbuzz-ng/src':
     'https://chromium.googlesource.com/external/github.com/harfbuzz/harfbuzz.git@fa2908bf16d2ccd6623f4d575455fea72a1a722b',
   'src/third_party/google_benchmark/src': {
@@ -584,7 +584,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/kotlinc',
-              'version': 'JEwTOoywRjDr5YoLIeG8Qzb3nT7OauqHdhfL6tIbyP0C',
+              'version': 'rZVNs6pgsRU0LupGqWhLJwl508-H7iam-fnzDW6X37MC',
           },
       ],
       'condition': 'checkout_android',
@@ -593,7 +593,7 @@ deps = {
   'src/third_party/libFuzzer/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt/lib/fuzzer.git@bea408a6e01f0f7e6c82a43121fe3af4506c932e',
   'src/third_party/fuzztest/src':
-    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@b6d8465706537cdd7e4cf89c9eb5587b5e16b96e',
+    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@54ed362692177e93df0fa202cae404b57694e828',
   'src/third_party/libjpeg_turbo':
     'https://chromium.googlesource.com/chromium/deps/libjpeg_turbo.git@d1f5f2393e0d51f840207342ae86e55a86443288',
   'src/third_party/libsrtp':
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '8464e2906ac98eb874fe861a0b789755d9e3650d',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '4cead1b0fc9987f994de97c7914e1f0c94047579',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@d3a6eb533ef3c871cdaef2d7967de90e233256cf',
+    'https://chromium.googlesource.com/chromium/src/tools@0510739a115e1aed8de0dfb2abc59088219be0b2',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'o4DXbfi661jCyVTmGyk5pLrrimnXDqgV-d8-dxfdZ_AC',
+          'version': 'uw8FaVpFdPaFx2Y4c1mBUMRwGB52J-PturMUyulx_hkC',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -864,7 +864,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7364b490b5f78d58efe23ea76e74210fd6c3c76f',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '81550faea475b1234ff31068605ac9536c9f2f92',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '4387e3248109bef2c07a8b69a2b8249d7af5dcf0',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From c4e1adff96bbaf40d6584fd6b1aecb2068c730e9 Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Tue, 10 Mar 2026 12:41:10 +0100
Subject: [PATCH 259/472] Replace ArrayView with std::span in logging/

Search&Replace MakeArrayView and ArrayView with std::span
Search&Replace include "api/array_view.h" with include <span>
Remove build dependencies on array_view target

Bug: webrtc:439801349
Change-Id: I9bdaea50fd34abff83cc7571a7fadc55a940df1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/455500
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47112}
---
 logging/BUILD.gn                              | 14 -----
 .../dependency_descriptor_encoder_decoder.cc  |  9 ++--
 .../dependency_descriptor_encoder_decoder.h   |  4 +-
 .../encoder/rtc_event_log_encoder_legacy.cc   |  4 +-
 .../encoder/rtc_event_log_encoder_legacy.h    |  4 +-
 .../rtc_event_log_encoder_new_format.cc       | 52 +++++++++----------
 .../rtc_event_log_encoder_new_format.h        | 52 +++++++++----------
 .../encoder/rtc_event_log_encoder_v3.h        |  4 +-
 .../fixed_length_encoding_parameters_v3.cc    |  4 +-
 .../fixed_length_encoding_parameters_v3.h     |  4 +-
 .../events/rtc_event_alr_state.h              |  4 +-
 .../rtc_event_audio_network_adaptation.h      |  4 +-
 .../events/rtc_event_audio_playout.h          |  4 +-
 .../rtc_event_audio_receive_stream_config.h   |  4 +-
 .../rtc_event_audio_send_stream_config.h      |  4 +-
 .../events/rtc_event_begin_log.cc             |  8 +--
 .../events/rtc_event_begin_log.h              |  4 +-
 .../events/rtc_event_bwe_update_delay_based.h |  4 +-
 .../events/rtc_event_bwe_update_loss_based.h  |  4 +-
 .../events/rtc_event_bwe_update_scream.h      |  4 +-
 .../events/rtc_event_definition.h             | 18 +++----
 .../events/rtc_event_dtls_transport_state.h   |  4 +-
 .../events/rtc_event_dtls_writable_state.h    |  4 +-
 .../rtc_event_log/events/rtc_event_end_log.cc |  8 +--
 .../rtc_event_log/events/rtc_event_end_log.h  |  4 +-
 .../events/rtc_event_field_encoding.cc        | 10 ++--
 .../events/rtc_event_field_encoding.h         | 14 ++---
 .../events/rtc_event_field_encoding_parser.cc | 14 ++---
 .../events/rtc_event_field_encoding_parser.h  | 42 +++++++--------
 .../rtc_event_field_encoding_unittest.cc      | 10 ++--
 .../events/rtc_event_frame_decoded.h          |  4 +-
 .../events/rtc_event_ice_candidate_pair.h     |  4 +-
 .../rtc_event_ice_candidate_pair_config.h     |  4 +-
 .../events/rtc_event_probe_cluster_created.h  |  4 +-
 .../events/rtc_event_probe_result_failure.h   |  4 +-
 .../events/rtc_event_probe_result_success.h   |  4 +-
 .../events/rtc_event_remote_estimate.h        |  4 +-
 .../events/rtc_event_route_change.h           |  4 +-
 .../events/rtc_event_rtcp_packet_incoming.cc  |  4 +-
 .../events/rtc_event_rtcp_packet_incoming.h   |  6 +--
 .../events/rtc_event_rtcp_packet_outgoing.cc  |  4 +-
 .../events/rtc_event_rtcp_packet_outgoing.h   |  6 +--
 .../events/rtc_event_rtp_packet_incoming.h    | 10 ++--
 .../events/rtc_event_rtp_packet_outgoing.h    | 10 ++--
 .../rtc_event_video_receive_stream_config.h   |  4 +-
 .../rtc_event_video_send_stream_config.h      |  4 +-
 .../rtc_event_log/rtc_event_log2rtp_dump.cc   |  4 +-
 .../rtc_event_log_unittest_helper.cc          |  8 +--
 48 files changed, 200 insertions(+), 215 deletions(-)

diff --git a/logging/BUILD.gn b/logging/BUILD.gn
index 9f5b865ba1..40914333eb 100644
--- a/logging/BUILD.gn
+++ b/logging/BUILD.gn
@@ -58,7 +58,6 @@ rtc_library("rtc_event_field") {
   deps = [
     ":rtc_event_log_parse_status",
     ":rtc_event_number_encodings",
-    "../api:array_view",
     "../api/rtc_event_log",
     "../api/units:timestamp",
     "../rtc_base:bitstream_reader",
@@ -92,7 +91,6 @@ rtc_library("rtc_event_pacing") {
   deps = [
     ":rtc_event_field",
     ":rtc_event_log_parse_status",
-    "../api:array_view",
     "../api/rtc_event_log",
     "../api/units:timestamp",
     "//third_party/abseil-cpp/absl/memory",
@@ -119,7 +117,6 @@ rtc_library("rtc_event_audio") {
     ":rtc_event_field",
     ":rtc_event_log_parse_status",
     ":rtc_stream_config",
-    "../api:array_view",
     "../api/rtc_event_log",
     "../api/units:timestamp",
     "../modules/audio_coding:audio_network_adaptor_config",
@@ -140,7 +137,6 @@ rtc_library("rtc_event_begin_end") {
   deps = [
     ":rtc_event_field",
     ":rtc_event_log_parse_status",
-    "../api:array_view",
     "../api/rtc_event_log",
     "../api/units:timestamp",
     "//third_party/abseil-cpp/absl/strings:string_view",
@@ -168,7 +164,6 @@ rtc_library("rtc_event_bwe") {
   deps = [
     ":rtc_event_field",
     ":rtc_event_log_parse_status",
-    "../api:array_view",
     "../api/rtc_event_log",
     "../api/transport:bandwidth_usage",
     "../api/units:data_rate",
@@ -188,7 +183,6 @@ rtc_library("rtc_event_bwe_update_scream") {
 
   deps = [
     ":rtc_event_log_parse_status",
-    "../api:array_view",
     "../api/rtc_event_log",
     "../api/units:data_rate",
     "../api/units:data_size",
@@ -209,7 +203,6 @@ rtc_library("rtc_event_frame_events") {
   deps = [
     ":rtc_event_field",
     ":rtc_event_log_parse_status",
-    "../api:array_view",
     "../api/rtc_event_log",
     "../api/units:timestamp",
     "../api/video:video_frame",
@@ -236,7 +229,6 @@ rtc_library("rtc_event_rtp_rtcp") {
   deps = [
     ":rtc_event_field",
     ":rtc_event_log_parse_status",
-    "../api:array_view",
     "../api:rtp_headers",
     "../api/rtc_event_log",
     "../api/units:timestamp",
@@ -261,7 +253,6 @@ rtc_library("rtc_event_video") {
     ":rtc_event_field",
     ":rtc_event_log_parse_status",
     ":rtc_stream_config",
-    "../api:array_view",
     "../api/rtc_event_log",
     "../api/units:timestamp",
     "../rtc_base:checks",
@@ -373,7 +364,6 @@ rtc_library("rtc_event_log_impl_encoder") {
       ":rtc_event_rtp_rtcp",
       ":rtc_event_video",
       ":rtc_stream_config",
-      "../api:array_view",
       "../api:candidate",
       "../api:dtls_transport_interface",
       "../api/rtc_event_log",
@@ -407,7 +397,6 @@ if (rtc_enable_protobuf) {
       ":rtc_event_log_optional_blob_encoding",
       ":rtc_event_log_parse_status",
       ":rtc_event_log_proto",  # Why does this need to be included here?
-      "../api:array_view",
       "../rtc_base:bitstream_reader",
       "../rtc_base:checks",
       "../rtc_base:logging",
@@ -598,7 +587,6 @@ if (rtc_enable_protobuf) {
         ":rtc_event_rtp_rtcp",
         ":rtc_event_video",
         ":rtc_stream_config",
-        "../api:array_view",
         "../api:candidate",
         "../api:dtls_transport_interface",
         "../api:field_trials",
@@ -650,7 +638,6 @@ if (rtc_enable_protobuf) {
         deps = [
           ":rtc_event_log_parser",
           ":rtc_event_rtp_rtcp",
-          "../api:array_view",
           "../api:rtp_headers",
           "../api/rtc_event_log",
           "../modules/rtp_rtcp",
@@ -688,7 +675,6 @@ rtc_library("ice_log") {
   deps = [
     ":rtc_event_field",
     ":rtc_event_log_parse_status",
-    "../api:array_view",
     "../api:candidate",
     "../api:dtls_transport_interface",
     "../api:libjingle_logging_api",
diff --git a/logging/rtc_event_log/dependency_descriptor_encoder_decoder.cc b/logging/rtc_event_log/dependency_descriptor_encoder_decoder.cc
index ca947ddfb0..995524780d 100644
--- a/logging/rtc_event_log/dependency_descriptor_encoder_decoder.cc
+++ b/logging/rtc_event_log/dependency_descriptor_encoder_decoder.cc
@@ -13,11 +13,11 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "logging/rtc_event_log/encoder/delta_encoding.h"
 #include "logging/rtc_event_log/encoder/optional_blob_encoding.h"
 #include "logging/rtc_event_log/events/rtc_event_log_parse_status.h"
@@ -30,7 +30,7 @@ namespace webrtc {
 // static
 std::optional<rtclog2::DependencyDescriptorsWireInfo>
 RtcEventLogDependencyDescriptorEncoderDecoder::Encode(
-    const std::vector<ArrayView<const uint8_t>>& raw_dd_data) {
+    const std::vector<std::span<const uint8_t>>& raw_dd_data) {
   if (raw_dd_data.empty()) {
     return {};
   }
@@ -43,9 +43,8 @@ RtcEventLogDependencyDescriptorEncoderDecoder::Encode(
   }
 
   rtclog2::DependencyDescriptorsWireInfo res;
-  const ArrayView<const uint8_t>& base_dd = raw_dd_data[0];
-  auto delta_dds =
-      MakeArrayView(raw_dd_data.data(), raw_dd_data.size()).subspan(1);
+  const std::span<const uint8_t>& base_dd = raw_dd_data[0];
+  auto delta_dds = std::span(raw_dd_data.data(), raw_dd_data.size()).subspan(1);
 
   // Start and end bit.
   {
diff --git a/logging/rtc_event_log/dependency_descriptor_encoder_decoder.h b/logging/rtc_event_log/dependency_descriptor_encoder_decoder.h
index 80f35f22f6..d0867e90ae 100644
--- a/logging/rtc_event_log/dependency_descriptor_encoder_decoder.h
+++ b/logging/rtc_event_log/dependency_descriptor_encoder_decoder.h
@@ -14,9 +14,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "logging/rtc_event_log/events/rtc_event_log_parse_status.h"
 #include "logging/rtc_event_log/rtc_event_log2_proto_include.h"  // IWYU pragma: keep
 
@@ -25,7 +25,7 @@ namespace webrtc {
 class RtcEventLogDependencyDescriptorEncoderDecoder {
  public:
   static std::optional<rtclog2::DependencyDescriptorsWireInfo> Encode(
-      const std::vector<ArrayView<const uint8_t>>& raw_dd_data);
+      const std::vector<std::span<const uint8_t>>& raw_dd_data);
   static RtcEventLogParseStatusOr<std::vector<std::vector<uint8_t>>> Decode(
       const rtclog2::DependencyDescriptorsWireInfo& dd_wire_info,
       size_t num_packets);
diff --git a/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc b/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc
index 5434a4092d..c225616d74 100644
--- a/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc
+++ b/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc
@@ -15,10 +15,10 @@
 #include <deque>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/candidate.h"
 #include "api/rtc_event_log/rtc_event.h"
 #include "api/rtp_headers.h"
@@ -753,7 +753,7 @@ std::string RtcEventLogEncoderLegacy::EncodeRtcpPacket(int64_t timestamp_us,
 
 std::string RtcEventLogEncoderLegacy::EncodeRtpPacket(
     int64_t timestamp_us,
-    ArrayView<const uint8_t> header,
+    std::span<const uint8_t> header,
     size_t packet_length,
     int probe_cluster_id,
     bool is_incoming) {
diff --git a/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.h b/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.h
index c03d3dbb4a..98252cce9c 100644
--- a/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.h
+++ b/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.h
@@ -15,9 +15,9 @@
 #include <cstdint>
 #include <deque>
 #include <memory>
+#include <span>
 #include <string>
 
-#include "api/array_view.h"
 #include "api/rtc_event_log/rtc_event.h"
 #include "logging/rtc_event_log/encoder/rtc_event_log_encoder.h"
 #include "rtc_base/buffer.h"
@@ -100,7 +100,7 @@ class RtcEventLogEncoderLegacy final : public RtcEventLogEncoder {
                                const Buffer& packet,
                                bool is_incoming);
   std::string EncodeRtpPacket(int64_t timestamp_us,
-                              ArrayView<const uint8_t> header,
+                              std::span<const uint8_t> header,
                               size_t packet_length,
                               int probe_cluster_id,
                               bool is_incoming);
diff --git a/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.cc b/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.cc
index 47396b7174..5d1342e509 100644
--- a/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.cc
+++ b/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.cc
@@ -17,11 +17,11 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <type_traits>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/candidate.h"
 #include "api/dtls_transport_interface.h"
 #include "api/field_trials_view.h"
@@ -340,7 +340,7 @@ size_t RemoveNonAllowlistedRtcpBlocks(const Buffer& packet, uint8_t* buffer) {
 }
 
 template <typename EventType, typename ProtoType>
-void EncodeRtcpPacket(ArrayView<const EventType*> batch,
+void EncodeRtcpPacket(std::span<const EventType*> batch,
                       ProtoType* proto_batch) {
   if (batch.empty()) {
     return;
@@ -468,7 +468,7 @@ void RtcEventLogEncoderNewFormat::EncodeRtpPacket(const Batch& batch,
     // TODO(webrtc:14975) Remove this kill switch after DD in RTC event log has
     //                    been rolled out.
     if (encode_dependency_descriptor_) {
-      std::vector<ArrayView<const uint8_t>> raw_dds(batch.size());
+      std::vector<std::span<const uint8_t>> raw_dds(batch.size());
       bool has_dd = false;
       for (size_t i = 0; i < batch.size(); ++i) {
         raw_dds[i] =
@@ -973,7 +973,7 @@ std::string RtcEventLogEncoderNewFormat::EncodeBatch(
 }
 
 void RtcEventLogEncoderNewFormat::EncodeAlrState(
-    ArrayView<const RtcEventAlrState*> batch,
+    std::span<const RtcEventAlrState*> batch,
     rtclog2::EventStream* event_stream) {
   for (const RtcEventAlrState* base_event : batch) {
     rtclog2::AlrState* proto_batch = event_stream->add_alr_states();
@@ -984,7 +984,7 @@ void RtcEventLogEncoderNewFormat::EncodeAlrState(
 }
 
 void RtcEventLogEncoderNewFormat::EncodeAudioNetworkAdaptation(
-    ArrayView<const RtcEventAudioNetworkAdaptation*> batch,
+    std::span<const RtcEventAudioNetworkAdaptation*> batch,
     rtclog2::EventStream* event_stream) {
   if (batch.empty())
     return;
@@ -1135,7 +1135,7 @@ void RtcEventLogEncoderNewFormat::EncodeAudioNetworkAdaptation(
 }
 
 void RtcEventLogEncoderNewFormat::EncodeAudioPlayout(
-    ArrayView<const RtcEventAudioPlayout*> batch,
+    std::span<const RtcEventAudioPlayout*> batch,
     rtclog2::EventStream* event_stream) {
   if (batch.empty())
     return;
@@ -1177,7 +1177,7 @@ void RtcEventLogEncoderNewFormat::EncodeAudioPlayout(
 }
 
 void RtcEventLogEncoderNewFormat::EncodeNetEqSetMinimumDelay(
-    ArrayView<const RtcEventNetEqSetMinimumDelay*> batch,
+    std::span<const RtcEventNetEqSetMinimumDelay*> batch,
     rtclog2::EventStream* event_stream) {
   if (encode_neteq_set_minimum_delay_kill_switch_) {
     return;
@@ -1235,7 +1235,7 @@ void RtcEventLogEncoderNewFormat::EncodeNetEqSetMinimumDelay(
 }
 
 void RtcEventLogEncoderNewFormat::EncodeAudioRecvStreamConfig(
-    ArrayView<const RtcEventAudioReceiveStreamConfig*> batch,
+    std::span<const RtcEventAudioReceiveStreamConfig*> batch,
     rtclog2::EventStream* event_stream) {
   for (const RtcEventAudioReceiveStreamConfig* base_event : batch) {
     rtclog2::AudioRecvStreamConfig* proto_batch =
@@ -1254,7 +1254,7 @@ void RtcEventLogEncoderNewFormat::EncodeAudioRecvStreamConfig(
 }
 
 void RtcEventLogEncoderNewFormat::EncodeAudioSendStreamConfig(
-    ArrayView<const RtcEventAudioSendStreamConfig*> batch,
+    std::span<const RtcEventAudioSendStreamConfig*> batch,
     rtclog2::EventStream* event_stream) {
   for (const RtcEventAudioSendStreamConfig* base_event : batch) {
     rtclog2::AudioSendStreamConfig* proto_batch =
@@ -1272,7 +1272,7 @@ void RtcEventLogEncoderNewFormat::EncodeAudioSendStreamConfig(
 }
 
 void RtcEventLogEncoderNewFormat::EncodeBweUpdateDelayBased(
-    ArrayView<const RtcEventBweUpdateDelayBased*> batch,
+    std::span<const RtcEventBweUpdateDelayBased*> batch,
     rtclog2::EventStream* event_stream) {
   if (batch.empty())
     return;
@@ -1329,7 +1329,7 @@ void RtcEventLogEncoderNewFormat::EncodeBweUpdateDelayBased(
 }
 
 void RtcEventLogEncoderNewFormat::EncodeBweUpdateLossBased(
-    ArrayView<const RtcEventBweUpdateLossBased*> batch,
+    std::span<const RtcEventBweUpdateLossBased*> batch,
     rtclog2::EventStream* event_stream) {
   if (batch.empty())
     return;
@@ -1393,7 +1393,7 @@ void RtcEventLogEncoderNewFormat::EncodeBweUpdateLossBased(
 }
 
 void RtcEventLogEncoderNewFormat::EncodeBweUpdateScream(
-    ArrayView<const RtcEventBweUpdateScream*> batch,
+    std::span<const RtcEventBweUpdateScream*> batch,
     rtclog2::EventStream* event_stream) {
   if (batch.empty())
     return;
@@ -1490,7 +1490,7 @@ void RtcEventLogEncoderNewFormat::EncodeBweUpdateScream(
 }
 
 void RtcEventLogEncoderNewFormat::EncodeDtlsTransportState(
-    ArrayView<const RtcEventDtlsTransportState*> batch,
+    std::span<const RtcEventDtlsTransportState*> batch,
     rtclog2::EventStream* event_stream) {
   for (const RtcEventDtlsTransportState* base_event : batch) {
     rtclog2::DtlsTransportStateEvent* proto_batch =
@@ -1502,7 +1502,7 @@ void RtcEventLogEncoderNewFormat::EncodeDtlsTransportState(
 }
 
 void RtcEventLogEncoderNewFormat::EncodeDtlsWritableState(
-    ArrayView<const RtcEventDtlsWritableState*> batch,
+    std::span<const RtcEventDtlsWritableState*> batch,
     rtclog2::EventStream* event_stream) {
   for (const RtcEventDtlsWritableState* base_event : batch) {
     rtclog2::DtlsWritableState* proto_batch =
@@ -1513,7 +1513,7 @@ void RtcEventLogEncoderNewFormat::EncodeDtlsWritableState(
 }
 
 void RtcEventLogEncoderNewFormat::EncodeProbeClusterCreated(
-    ArrayView<const RtcEventProbeClusterCreated*> batch,
+    std::span<const RtcEventProbeClusterCreated*> batch,
     rtclog2::EventStream* event_stream) {
   for (const RtcEventProbeClusterCreated* base_event : batch) {
     rtclog2::BweProbeCluster* proto_batch = event_stream->add_probe_clusters();
@@ -1526,7 +1526,7 @@ void RtcEventLogEncoderNewFormat::EncodeProbeClusterCreated(
 }
 
 void RtcEventLogEncoderNewFormat::EncodeProbeResultFailure(
-    ArrayView<const RtcEventProbeResultFailure*> batch,
+    std::span<const RtcEventProbeResultFailure*> batch,
     rtclog2::EventStream* event_stream) {
   for (const RtcEventProbeResultFailure* base_event : batch) {
     rtclog2::BweProbeResultFailure* proto_batch =
@@ -1540,7 +1540,7 @@ void RtcEventLogEncoderNewFormat::EncodeProbeResultFailure(
 }
 
 void RtcEventLogEncoderNewFormat::EncodeProbeResultSuccess(
-    ArrayView<const RtcEventProbeResultSuccess*> batch,
+    std::span<const RtcEventProbeResultSuccess*> batch,
     rtclog2::EventStream* event_stream) {
   for (const RtcEventProbeResultSuccess* base_event : batch) {
     rtclog2::BweProbeResultSuccess* proto_batch =
@@ -1553,7 +1553,7 @@ void RtcEventLogEncoderNewFormat::EncodeProbeResultSuccess(
 }
 
 void RtcEventLogEncoderNewFormat::EncodeRouteChange(
-    ArrayView<const RtcEventRouteChange*> batch,
+    std::span<const RtcEventRouteChange*> batch,
     rtclog2::EventStream* event_stream) {
   for (const RtcEventRouteChange* base_event : batch) {
     rtclog2::RouteChange* proto_batch = event_stream->add_route_changes();
@@ -1565,7 +1565,7 @@ void RtcEventLogEncoderNewFormat::EncodeRouteChange(
 }
 
 void RtcEventLogEncoderNewFormat::EncodeRemoteEstimate(
-    ArrayView<const RtcEventRemoteEstimate*> batch,
+    std::span<const RtcEventRemoteEstimate*> batch,
     rtclog2::EventStream* event_stream) {
   if (batch.empty())
     return;
@@ -1637,7 +1637,7 @@ void RtcEventLogEncoderNewFormat::EncodeRemoteEstimate(
 }
 
 void RtcEventLogEncoderNewFormat::EncodeRtcpPacketIncoming(
-    ArrayView<const RtcEventRtcpPacketIncoming*> batch,
+    std::span<const RtcEventRtcpPacketIncoming*> batch,
     rtclog2::EventStream* event_stream) {
   if (batch.empty()) {
     return;
@@ -1646,7 +1646,7 @@ void RtcEventLogEncoderNewFormat::EncodeRtcpPacketIncoming(
 }
 
 void RtcEventLogEncoderNewFormat::EncodeRtcpPacketOutgoing(
-    ArrayView<const RtcEventRtcpPacketOutgoing*> batch,
+    std::span<const RtcEventRtcpPacketOutgoing*> batch,
     rtclog2::EventStream* event_stream) {
   if (batch.empty()) {
     return;
@@ -1665,7 +1665,7 @@ void RtcEventLogEncoderNewFormat::EncodeRtpPacketIncoming(
 }
 
 void RtcEventLogEncoderNewFormat::EncodeFramesDecoded(
-    ArrayView<const RtcEventFrameDecoded* const> batch,
+    std::span<const RtcEventFrameDecoded* const> batch,
     rtclog2::EventStream* event_stream) {
   if (batch.empty()) {
     return;
@@ -1774,7 +1774,7 @@ void RtcEventLogEncoderNewFormat::EncodeRtpPacketOutgoing(
 }
 
 void RtcEventLogEncoderNewFormat::EncodeVideoRecvStreamConfig(
-    ArrayView<const RtcEventVideoReceiveStreamConfig*> batch,
+    std::span<const RtcEventVideoReceiveStreamConfig*> batch,
     rtclog2::EventStream* event_stream) {
   for (const RtcEventVideoReceiveStreamConfig* base_event : batch) {
     rtclog2::VideoRecvStreamConfig* proto_batch =
@@ -1794,7 +1794,7 @@ void RtcEventLogEncoderNewFormat::EncodeVideoRecvStreamConfig(
 }
 
 void RtcEventLogEncoderNewFormat::EncodeVideoSendStreamConfig(
-    ArrayView<const RtcEventVideoSendStreamConfig*> batch,
+    std::span<const RtcEventVideoSendStreamConfig*> batch,
     rtclog2::EventStream* event_stream) {
   for (const RtcEventVideoSendStreamConfig* base_event : batch) {
     rtclog2::VideoSendStreamConfig* proto_batch =
@@ -1813,7 +1813,7 @@ void RtcEventLogEncoderNewFormat::EncodeVideoSendStreamConfig(
 }
 
 void RtcEventLogEncoderNewFormat::EncodeIceCandidatePairConfig(
-    ArrayView<const RtcEventIceCandidatePairConfig*> batch,
+    std::span<const RtcEventIceCandidatePairConfig*> batch,
     rtclog2::EventStream* event_stream) {
   for (const RtcEventIceCandidatePairConfig* base_event : batch) {
     rtclog2::IceCandidatePairConfig* proto_batch =
@@ -1842,7 +1842,7 @@ void RtcEventLogEncoderNewFormat::EncodeIceCandidatePairConfig(
 }
 
 void RtcEventLogEncoderNewFormat::EncodeIceCandidatePairEvent(
-    ArrayView<const RtcEventIceCandidatePair*> batch,
+    std::span<const RtcEventIceCandidatePair*> batch,
     rtclog2::EventStream* event_stream) {
   for (const RtcEventIceCandidatePair* base_event : batch) {
     rtclog2::IceCandidatePairEvent* proto_batch =
diff --git a/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.h b/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.h
index 57e6f8dea8..099c5df2b2 100644
--- a/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.h
+++ b/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.h
@@ -15,10 +15,10 @@
 #include <deque>
 #include <map>
 #include <memory>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/field_trials_view.h"
 #include "api/rtc_event_log/rtc_event.h"
 #include "logging/rtc_event_log/encoder/rtc_event_log_encoder.h"
@@ -73,66 +73,66 @@ class RtcEventLogEncoderNewFormat final : public RtcEventLogEncoder {
 
  private:
   // Encoding entry-point for the various RtcEvent subclasses.
-  void EncodeAlrState(ArrayView<const RtcEventAlrState*> batch,
+  void EncodeAlrState(std::span<const RtcEventAlrState*> batch,
                       rtclog2::EventStream* event_stream);
   void EncodeAudioNetworkAdaptation(
-      ArrayView<const RtcEventAudioNetworkAdaptation*> batch,
+      std::span<const RtcEventAudioNetworkAdaptation*> batch,
       rtclog2::EventStream* event_stream);
-  void EncodeAudioPlayout(ArrayView<const RtcEventAudioPlayout*> batch,
+  void EncodeAudioPlayout(std::span<const RtcEventAudioPlayout*> batch,
                           rtclog2::EventStream* event_stream);
   void EncodeAudioRecvStreamConfig(
-      ArrayView<const RtcEventAudioReceiveStreamConfig*> batch,
+      std::span<const RtcEventAudioReceiveStreamConfig*> batch,
       rtclog2::EventStream* event_stream);
   void EncodeAudioSendStreamConfig(
-      ArrayView<const RtcEventAudioSendStreamConfig*> batch,
+      std::span<const RtcEventAudioSendStreamConfig*> batch,
       rtclog2::EventStream* event_stream);
   void EncodeBweUpdateDelayBased(
-      ArrayView<const RtcEventBweUpdateDelayBased*> batch,
+      std::span<const RtcEventBweUpdateDelayBased*> batch,
       rtclog2::EventStream* event_stream);
   void EncodeBweUpdateLossBased(
-      ArrayView<const RtcEventBweUpdateLossBased*> batch,
+      std::span<const RtcEventBweUpdateLossBased*> batch,
       rtclog2::EventStream* event_stream);
-  void EncodeBweUpdateScream(ArrayView<const RtcEventBweUpdateScream*> batch,
+  void EncodeBweUpdateScream(std::span<const RtcEventBweUpdateScream*> batch,
                              rtclog2::EventStream* event_stream);
   void EncodeDtlsTransportState(
-      ArrayView<const RtcEventDtlsTransportState*> batch,
+      std::span<const RtcEventDtlsTransportState*> batch,
       rtclog2::EventStream* event_stream);
   void EncodeDtlsWritableState(
-      ArrayView<const RtcEventDtlsWritableState*> batch,
+      std::span<const RtcEventDtlsWritableState*> batch,
       rtclog2::EventStream* event_stream);
-  void EncodeFramesDecoded(ArrayView<const RtcEventFrameDecoded* const> batch,
+  void EncodeFramesDecoded(std::span<const RtcEventFrameDecoded* const> batch,
                            rtclog2::EventStream* event_stream);
   void EncodeIceCandidatePairConfig(
-      ArrayView<const RtcEventIceCandidatePairConfig*> batch,
+      std::span<const RtcEventIceCandidatePairConfig*> batch,
       rtclog2::EventStream* event_stream);
   void EncodeIceCandidatePairEvent(
-      ArrayView<const RtcEventIceCandidatePair*> batch,
+      std::span<const RtcEventIceCandidatePair*> batch,
       rtclog2::EventStream* event_stream);
-  void EncodeLoggingStarted(ArrayView<const RtcEventLoggingStarted*> batch,
+  void EncodeLoggingStarted(std::span<const RtcEventLoggingStarted*> batch,
                             rtclog2::EventStream* event_stream);
-  void EncodeLoggingStopped(ArrayView<const RtcEventLoggingStopped*> batch,
+  void EncodeLoggingStopped(std::span<const RtcEventLoggingStopped*> batch,
                             rtclog2::EventStream* event_stream);
   void EncodeNetEqSetMinimumDelay(
-      ArrayView<const RtcEventNetEqSetMinimumDelay*> batch,
+      std::span<const RtcEventNetEqSetMinimumDelay*> batch,
       rtclog2::EventStream* event_stream);
   void EncodeProbeClusterCreated(
-      ArrayView<const RtcEventProbeClusterCreated*> batch,
+      std::span<const RtcEventProbeClusterCreated*> batch,
       rtclog2::EventStream* event_stream);
   void EncodeProbeResultFailure(
-      ArrayView<const RtcEventProbeResultFailure*> batch,
+      std::span<const RtcEventProbeResultFailure*> batch,
       rtclog2::EventStream* event_stream);
   void EncodeProbeResultSuccess(
-      ArrayView<const RtcEventProbeResultSuccess*> batch,
+      std::span<const RtcEventProbeResultSuccess*> batch,
       rtclog2::EventStream* event_stream);
-  void EncodeRouteChange(ArrayView<const RtcEventRouteChange*> batch,
+  void EncodeRouteChange(std::span<const RtcEventRouteChange*> batch,
                          rtclog2::EventStream* event_stream);
-  void EncodeRemoteEstimate(ArrayView<const RtcEventRemoteEstimate*> batch,
+  void EncodeRemoteEstimate(std::span<const RtcEventRemoteEstimate*> batch,
                             rtclog2::EventStream* event_stream);
   void EncodeRtcpPacketIncoming(
-      ArrayView<const RtcEventRtcpPacketIncoming*> batch,
+      std::span<const RtcEventRtcpPacketIncoming*> batch,
       rtclog2::EventStream* event_stream);
   void EncodeRtcpPacketOutgoing(
-      ArrayView<const RtcEventRtcpPacketOutgoing*> batch,
+      std::span<const RtcEventRtcpPacketOutgoing*> batch,
       rtclog2::EventStream* event_stream);
   void EncodeRtpPacketIncoming(
       const std::map<uint32_t, std::vector<const RtcEventRtpPacketIncoming*>>&
@@ -143,10 +143,10 @@ class RtcEventLogEncoderNewFormat final : public RtcEventLogEncoder {
           batch,
       rtclog2::EventStream* event_stream);
   void EncodeVideoRecvStreamConfig(
-      ArrayView<const RtcEventVideoReceiveStreamConfig*> batch,
+      std::span<const RtcEventVideoReceiveStreamConfig*> batch,
       rtclog2::EventStream* event_stream);
   void EncodeVideoSendStreamConfig(
-      ArrayView<const RtcEventVideoSendStreamConfig*> batch,
+      std::span<const RtcEventVideoSendStreamConfig*> batch,
       rtclog2::EventStream* event_stream);
   template <typename Batch, typename ProtoType>
   void EncodeRtpPacket(const Batch& batch, ProtoType* proto_batch);
diff --git a/logging/rtc_event_log/encoder/rtc_event_log_encoder_v3.h b/logging/rtc_event_log/encoder/rtc_event_log_encoder_v3.h
index d5d2b6e342..c77c65a09a 100644
--- a/logging/rtc_event_log/encoder/rtc_event_log_encoder_v3.h
+++ b/logging/rtc_event_log/encoder/rtc_event_log_encoder_v3.h
@@ -16,9 +16,9 @@
 #include <functional>
 #include <map>
 #include <memory>
+#include <span>
 #include <string>
 
-#include "api/array_view.h"
 #include "api/rtc_event_log/rtc_event.h"
 #include "logging/rtc_event_log/encoder/rtc_event_log_encoder.h"
 
@@ -39,7 +39,7 @@ class RtcEventLogEncoderV3 final : public RtcEventLogEncoder {
 
  private:
   std::map<RtcEvent::Type,
-           std::function<std::string(webrtc::ArrayView<const RtcEvent*>)>>
+           std::function<std::string(std::span<const RtcEvent*>)>>
       encoders_;
 };
 
diff --git a/logging/rtc_event_log/events/fixed_length_encoding_parameters_v3.cc b/logging/rtc_event_log/events/fixed_length_encoding_parameters_v3.cc
index e2ef9b9425..33530775ce 100644
--- a/logging/rtc_event_log/events/fixed_length_encoding_parameters_v3.cc
+++ b/logging/rtc_event_log/events/fixed_length_encoding_parameters_v3.cc
@@ -13,8 +13,8 @@
 #include <algorithm>
 #include <cstdint>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "logging/rtc_event_log/events/rtc_event_field_extraction.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
@@ -29,7 +29,7 @@ namespace webrtc {
 FixedLengthEncodingParametersV3
 FixedLengthEncodingParametersV3::CalculateParameters(
     uint64_t base,
-    const ArrayView<const uint64_t> values,
+    const std::span<const uint64_t> values,
     uint64_t value_bit_width,
     bool values_optional) {
   // As a special case, if all of the elements are identical to the base
diff --git a/logging/rtc_event_log/events/fixed_length_encoding_parameters_v3.h b/logging/rtc_event_log/events/fixed_length_encoding_parameters_v3.h
index 9e4f5298f4..044e37e299 100644
--- a/logging/rtc_event_log/events/fixed_length_encoding_parameters_v3.h
+++ b/logging/rtc_event_log/events/fixed_length_encoding_parameters_v3.h
@@ -13,8 +13,8 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "logging/rtc_event_log/events/rtc_event_field_extraction.h"
 
 namespace webrtc {
@@ -34,7 +34,7 @@ class FixedLengthEncodingParametersV3 final {
 
   static FixedLengthEncodingParametersV3 CalculateParameters(
       uint64_t base,
-      ArrayView<const uint64_t> values,
+      std::span<const uint64_t> values,
       uint64_t value_bit_width,
       bool values_optional);
   static std::optional<FixedLengthEncodingParametersV3> ParseDeltaHeader(
diff --git a/logging/rtc_event_log/events/rtc_event_alr_state.h b/logging/rtc_event_log/events/rtc_event_alr_state.h
index a7321873f7..09fc129588 100644
--- a/logging/rtc_event_log/events/rtc_event_alr_state.h
+++ b/logging/rtc_event_log/events/rtc_event_alr_state.h
@@ -13,11 +13,11 @@
 
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtc_event_log/rtc_event.h"
 #include "api/units/timestamp.h"
 #include "logging/rtc_event_log/events/rtc_event_definition.h"
@@ -53,7 +53,7 @@ class RtcEventAlrState final : public RtcEvent {
 
   bool in_alr() const { return in_alr_; }
 
-  static std::string Encode(ArrayView<const RtcEvent*> batch) {
+  static std::string Encode(std::span<const RtcEvent*> batch) {
     return RtcEventAlrState::definition_.EncodeBatch(batch);
   }
 
diff --git a/logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h b/logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h
index ea78ef4b4a..b0fa86e6bf 100644
--- a/logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h
+++ b/logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h
@@ -15,11 +15,11 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtc_event_log/rtc_event.h"
 #include "api/units/timestamp.h"
 #include "logging/rtc_event_log/events/rtc_event_log_parse_status.h"
@@ -66,7 +66,7 @@ class RtcEventAudioNetworkAdaptation final : public RtcEvent {
   const std::optional<bool>& enable_dtx() const { return enable_dtx_; }
   const std::optional<size_t>& num_channels() const { return num_channels_; }
 
-  static std::string Encode(ArrayView<const RtcEvent*> /* batch */) {
+  static std::string Encode(std::span<const RtcEvent*> /* batch */) {
     // TODO(terelius): Implement
     return "";
   }
diff --git a/logging/rtc_event_log/events/rtc_event_audio_playout.h b/logging/rtc_event_log/events/rtc_event_audio_playout.h
index d9683ed449..e05830d286 100644
--- a/logging/rtc_event_log/events/rtc_event_audio_playout.h
+++ b/logging/rtc_event_log/events/rtc_event_audio_playout.h
@@ -15,11 +15,11 @@
 
 #include <map>
 #include <memory>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtc_event_log/rtc_event.h"
 #include "api/units/timestamp.h"
 #include "logging/rtc_event_log/events/rtc_event_definition.h"
@@ -55,7 +55,7 @@ class RtcEventAudioPlayout final : public RtcEvent {
 
   uint32_t ssrc() const { return ssrc_; }
 
-  static std::string Encode(ArrayView<const RtcEvent*> batch) {
+  static std::string Encode(std::span<const RtcEvent*> batch) {
     return RtcEventAudioPlayout::definition_.EncodeBatch(batch);
   }
 
diff --git a/logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.h b/logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.h
index 5e74754972..965d974f20 100644
--- a/logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.h
+++ b/logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.h
@@ -13,11 +13,11 @@
 
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtc_event_log/rtc_event.h"
 #include "api/units/timestamp.h"
 #include "logging/rtc_event_log/events/rtc_event_log_parse_status.h"
@@ -53,7 +53,7 @@ class RtcEventAudioReceiveStreamConfig final : public RtcEvent {
 
   const rtclog::StreamConfig& config() const { return *config_; }
 
-  static std::string Encode(ArrayView<const RtcEvent*> /* batch */) {
+  static std::string Encode(std::span<const RtcEvent*> /* batch */) {
     // TODO(terelius): Implement
     return "";
   }
diff --git a/logging/rtc_event_log/events/rtc_event_audio_send_stream_config.h b/logging/rtc_event_log/events/rtc_event_audio_send_stream_config.h
index 861d067b03..5978cf2d78 100644
--- a/logging/rtc_event_log/events/rtc_event_audio_send_stream_config.h
+++ b/logging/rtc_event_log/events/rtc_event_audio_send_stream_config.h
@@ -13,11 +13,11 @@
 
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtc_event_log/rtc_event.h"
 #include "api/units/timestamp.h"
 #include "logging/rtc_event_log/events/rtc_event_log_parse_status.h"
@@ -53,7 +53,7 @@ class RtcEventAudioSendStreamConfig final : public RtcEvent {
 
   const rtclog::StreamConfig& config() const { return *config_; }
 
-  static std::string Encode(ArrayView<const RtcEvent*> /* batch */) {
+  static std::string Encode(std::span<const RtcEvent*> /* batch */) {
     // TODO(terelius): Implement
     return "";
   }
diff --git a/logging/rtc_event_log/events/rtc_event_begin_log.cc b/logging/rtc_event_log/events/rtc_event_begin_log.cc
index acdb0f8bbe..1f3d290e4b 100644
--- a/logging/rtc_event_log/events/rtc_event_begin_log.cc
+++ b/logging/rtc_event_log/events/rtc_event_begin_log.cc
@@ -11,11 +11,11 @@
 #include "logging/rtc_event_log/events/rtc_event_begin_log.h"
 
 #include <cstdint>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtc_event_log/rtc_event.h"
 #include "api/units/timestamp.h"
 #include "logging/rtc_event_log/events/rtc_event_field_encoding.h"
@@ -30,7 +30,7 @@ RtcEventBeginLog::RtcEventBeginLog(Timestamp timestamp,
 
 RtcEventBeginLog::~RtcEventBeginLog() = default;
 
-std::string RtcEventBeginLog::Encode(ArrayView<const RtcEvent*> batch) {
+std::string RtcEventBeginLog::Encode(std::span<const RtcEvent*> batch) {
   EventEncoder encoder(event_params_, batch);
 
   encoder.EncodeField(
@@ -49,7 +49,7 @@ RtcEventLogParseStatus RtcEventBeginLog::Parse(
   if (!status.ok())
     return status;
 
-  ArrayView<LoggedStartEvent> output_batch =
+  std::span<LoggedStartEvent> output_batch =
       ExtendLoggedBatch(output, parser.NumEventsInBatch());
 
   constexpr FieldParameters timestamp_params{
@@ -57,7 +57,7 @@ RtcEventLogParseStatus RtcEventBeginLog::Parse(
       .field_id = FieldParameters::kTimestampField,
       .field_type = FieldType::kVarInt,
       .value_width = 64};
-  RtcEventLogParseStatusOr<ArrayView<uint64_t>> result =
+  RtcEventLogParseStatusOr<std::span<uint64_t>> result =
       parser.ParseNumericField(timestamp_params);
   if (!result.ok())
     return result.status();
diff --git a/logging/rtc_event_log/events/rtc_event_begin_log.h b/logging/rtc_event_log/events/rtc_event_begin_log.h
index 0cbad4c821..7136833d69 100644
--- a/logging/rtc_event_log/events/rtc_event_begin_log.h
+++ b/logging/rtc_event_log/events/rtc_event_begin_log.h
@@ -12,11 +12,11 @@
 #define LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_BEGIN_LOG_H_
 
 #include <cstdint>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtc_event_log/rtc_event.h"
 #include "api/units/timestamp.h"
 #include "logging/rtc_event_log/events/rtc_event_field_encoding.h"
@@ -53,7 +53,7 @@ class RtcEventBeginLog final : public RtcEvent {
   Type GetType() const override { return kType; }
   bool IsConfigEvent() const override { return false; }
 
-  static std::string Encode(ArrayView<const RtcEvent*> batch);
+  static std::string Encode(std::span<const RtcEvent*> batch);
 
   static RtcEventLogParseStatus Parse(absl::string_view encoded_bytes,
                                       bool batched,
diff --git a/logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h b/logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h
index 417d24828a..0fb1041850 100644
--- a/logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h
+++ b/logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h
@@ -15,11 +15,11 @@
 
 #include <limits>
 #include <memory>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtc_event_log/rtc_event.h"
 #include "api/transport/bandwidth_usage.h"
 #include "api/units/timestamp.h"
@@ -104,7 +104,7 @@ class RtcEventBweUpdateDelayBased final : public RtcEvent {
   int32_t bitrate_bps() const { return bitrate_bps_; }
   BandwidthUsage detector_state() const { return detector_state_; }
 
-  static std::string Encode(ArrayView<const RtcEvent*> batch) {
+  static std::string Encode(std::span<const RtcEvent*> batch) {
     return RtcEventBweUpdateDelayBased::definition_.EncodeBatch(batch);
   }
 
diff --git a/logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h b/logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h
index b8548004a6..9f10f93018 100644
--- a/logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h
+++ b/logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h
@@ -14,11 +14,11 @@
 #include <stdint.h>
 
 #include <memory>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtc_event_log/rtc_event.h"
 #include "api/units/timestamp.h"
 #include "logging/rtc_event_log/events/rtc_event_log_parse_status.h"
@@ -64,7 +64,7 @@ class RtcEventBweUpdateLossBased final : public RtcEvent {
   uint8_t fraction_loss() const { return fraction_loss_; }
   int32_t total_packets() const { return total_packets_; }
 
-  static std::string Encode(ArrayView<const RtcEvent*> /* batch */) {
+  static std::string Encode(std::span<const RtcEvent*> /* batch */) {
     // TODO(terelius): Implement
     return "";
   }
diff --git a/logging/rtc_event_log/events/rtc_event_bwe_update_scream.h b/logging/rtc_event_log/events/rtc_event_bwe_update_scream.h
index 538e7977f4..0580ffa7c2 100644
--- a/logging/rtc_event_log/events/rtc_event_bwe_update_scream.h
+++ b/logging/rtc_event_log/events/rtc_event_bwe_update_scream.h
@@ -14,11 +14,11 @@
 #include <stdint.h>
 
 #include <memory>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtc_event_log/rtc_event.h"
 #include "api/units/data_rate.h"
 #include "api/units/data_size.h"
@@ -82,7 +82,7 @@ class RtcEventBweUpdateScream final : public RtcEvent {
   uint32_t avg_queue_delay_ms() const { return avg_queue_delay_ms_; }
   uint32_t l4s_marked_permille() const { return l4s_marked_permille_; }
 
-  static std::string Encode(ArrayView<const RtcEvent*> batch) {
+  static std::string Encode(std::span<const RtcEvent*> batch) {
     // TODO(terelius): Implement
     return "";
   }
diff --git a/logging/rtc_event_log/events/rtc_event_definition.h b/logging/rtc_event_log/events/rtc_event_definition.h
index f924c7cdc7..0c221edfb3 100644
--- a/logging/rtc_event_log/events/rtc_event_definition.h
+++ b/logging/rtc_event_log/events/rtc_event_definition.h
@@ -12,11 +12,11 @@
 #define LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_DEFINITION_H_
 
 #include <cstdint>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtc_event_log/rtc_event.h"
 #include "logging/rtc_event_log/events/rtc_event_field_encoding.h"
 #include "logging/rtc_event_log/events/rtc_event_field_encoding_parser.h"
@@ -35,8 +35,8 @@ struct RtcEventFieldDefinition {
 template <typename EventType, typename LoggedType, typename... Ts>
 class RtcEventDefinitionImpl {
  public:
-  void EncodeImpl(EventEncoder&, ArrayView<const RtcEvent*>) const {}
-  RtcEventLogParseStatus ParseImpl(EventParser&, ArrayView<LoggedType>) const {
+  void EncodeImpl(EventEncoder&, std::span<const RtcEvent*>) const {}
+  RtcEventLogParseStatus ParseImpl(EventParser&, std::span<LoggedType>) const {
     return RtcEventLogParseStatus::Success();
   }
 };
@@ -51,15 +51,15 @@ class RtcEventDefinitionImpl<EventType, LoggedType, T, Ts...> {
       : field_(field), rest_(rest...) {}
 
   void EncodeImpl(EventEncoder& encoder,
-                  ArrayView<const RtcEvent*> batch) const {
+                  std::span<const RtcEvent*> batch) const {
     auto values = ExtractRtcEventMember(batch, field_.event_member);
     encoder.EncodeField(field_.params, values);
     rest_.EncodeImpl(encoder, batch);
   }
 
   RtcEventLogParseStatus ParseImpl(EventParser& parser,
-                                   ArrayView<LoggedType> output_batch) const {
-    RtcEventLogParseStatusOr<ArrayView<uint64_t>> result =
+                                   std::span<LoggedType> output_batch) const {
+    RtcEventLogParseStatusOr<std::span<uint64_t>> result =
         parser.ParseNumericField(field_.params);
     if (!result.ok())
       return result.status();
@@ -106,7 +106,7 @@ class RtcEventDefinition {
       RtcEventFieldDefinition<EventType, LoggedType, Ts>... fields)
       : params_(params), fields_(fields...) {}
 
-  std::string EncodeBatch(ArrayView<const RtcEvent*> batch) const {
+  std::string EncodeBatch(std::span<const RtcEvent*> batch) const {
     EventEncoder encoder(params_, batch);
     fields_.EncodeImpl(encoder, batch);
     return encoder.AsString();
@@ -120,7 +120,7 @@ class RtcEventDefinition {
     if (!status.ok())
       return status;
 
-    ArrayView<LoggedType> output_batch =
+    std::span<LoggedType> output_batch =
         ExtendLoggedBatch(output, parser.NumEventsInBatch());
 
     constexpr FieldParameters timestamp_params{
@@ -128,7 +128,7 @@ class RtcEventDefinition {
         .field_id = FieldParameters::kTimestampField,
         .field_type = FieldType::kVarInt,
         .value_width = 64};
-    RtcEventLogParseStatusOr<ArrayView<uint64_t>> result =
+    RtcEventLogParseStatusOr<std::span<uint64_t>> result =
         parser.ParseNumericField(timestamp_params);
     if (!result.ok())
       return result.status();
diff --git a/logging/rtc_event_log/events/rtc_event_dtls_transport_state.h b/logging/rtc_event_log/events/rtc_event_dtls_transport_state.h
index a2ffbcf62f..aa821e1934 100644
--- a/logging/rtc_event_log/events/rtc_event_dtls_transport_state.h
+++ b/logging/rtc_event_log/events/rtc_event_dtls_transport_state.h
@@ -13,11 +13,11 @@
 
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/dtls_transport_interface.h"
 #include "api/rtc_event_log/rtc_event.h"
 #include "api/units/timestamp.h"
@@ -50,7 +50,7 @@ class RtcEventDtlsTransportState : public RtcEvent {
     return dtls_transport_state_;
   }
 
-  static std::string Encode(ArrayView<const RtcEvent*> /* batch */) {
+  static std::string Encode(std::span<const RtcEvent*> /* batch */) {
     // TODO(terelius): Implement
     return "";
   }
diff --git a/logging/rtc_event_log/events/rtc_event_dtls_writable_state.h b/logging/rtc_event_log/events/rtc_event_dtls_writable_state.h
index f1da19f8db..4ebf693a9c 100644
--- a/logging/rtc_event_log/events/rtc_event_dtls_writable_state.h
+++ b/logging/rtc_event_log/events/rtc_event_dtls_writable_state.h
@@ -13,11 +13,11 @@
 
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtc_event_log/rtc_event.h"
 #include "api/units/timestamp.h"
 #include "logging/rtc_event_log/events/rtc_event_log_parse_status.h"
@@ -50,7 +50,7 @@ class RtcEventDtlsWritableState : public RtcEvent {
 
   bool writable() const { return writable_; }
 
-  static std::string Encode(ArrayView<const RtcEvent*> /* batch */) {
+  static std::string Encode(std::span<const RtcEvent*> /* batch */) {
     // TODO(terelius): Implement
     return "";
   }
diff --git a/logging/rtc_event_log/events/rtc_event_end_log.cc b/logging/rtc_event_log/events/rtc_event_end_log.cc
index e5455daff9..e912e800aa 100644
--- a/logging/rtc_event_log/events/rtc_event_end_log.cc
+++ b/logging/rtc_event_log/events/rtc_event_end_log.cc
@@ -11,11 +11,11 @@
 #include "logging/rtc_event_log/events/rtc_event_end_log.h"
 
 #include <cstdint>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtc_event_log/rtc_event.h"
 #include "api/units/timestamp.h"
 #include "logging/rtc_event_log/events/rtc_event_field_encoding.h"
@@ -29,7 +29,7 @@ RtcEventEndLog::RtcEventEndLog(Timestamp timestamp)
 
 RtcEventEndLog::~RtcEventEndLog() = default;
 
-std::string RtcEventEndLog::Encode(ArrayView<const RtcEvent*> batch) {
+std::string RtcEventEndLog::Encode(std::span<const RtcEvent*> batch) {
   EventEncoder encoder(event_params_, batch);
   return encoder.AsString();
 }
@@ -43,7 +43,7 @@ RtcEventLogParseStatus RtcEventEndLog::Parse(
   if (!status.ok())
     return status;
 
-  ArrayView<LoggedStopEvent> output_batch =
+  std::span<LoggedStopEvent> output_batch =
       ExtendLoggedBatch(output, parser.NumEventsInBatch());
 
   constexpr FieldParameters timestamp_params{
@@ -51,7 +51,7 @@ RtcEventLogParseStatus RtcEventEndLog::Parse(
       .field_id = FieldParameters::kTimestampField,
       .field_type = FieldType::kVarInt,
       .value_width = 64};
-  RtcEventLogParseStatusOr<ArrayView<uint64_t>> result =
+  RtcEventLogParseStatusOr<std::span<uint64_t>> result =
       parser.ParseNumericField(timestamp_params);
   if (!result.ok())
     return result.status();
diff --git a/logging/rtc_event_log/events/rtc_event_end_log.h b/logging/rtc_event_log/events/rtc_event_end_log.h
index bb7f7f4bb7..aba1043eb4 100644
--- a/logging/rtc_event_log/events/rtc_event_end_log.h
+++ b/logging/rtc_event_log/events/rtc_event_end_log.h
@@ -12,11 +12,11 @@
 #define LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_END_LOG_H_
 
 #include <cstdint>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtc_event_log/rtc_event.h"
 #include "api/units/timestamp.h"
 #include "logging/rtc_event_log/events/rtc_event_field_encoding.h"
@@ -46,7 +46,7 @@ class RtcEventEndLog final : public RtcEvent {
   Type GetType() const override { return kType; }
   bool IsConfigEvent() const override { return false; }
 
-  static std::string Encode(ArrayView<const RtcEvent*> batch);
+  static std::string Encode(std::span<const RtcEvent*> batch);
 
   static RtcEventLogParseStatus Parse(absl::string_view encoded_bytes,
                                       bool batched,
diff --git a/logging/rtc_event_log/events/rtc_event_field_encoding.cc b/logging/rtc_event_log/events/rtc_event_field_encoding.cc
index 2fdfe220cb..b0f6290f8d 100644
--- a/logging/rtc_event_log/events/rtc_event_field_encoding.cc
+++ b/logging/rtc_event_log/events/rtc_event_field_encoding.cc
@@ -14,11 +14,11 @@
 #include <cstdint>
 #include <limits>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtc_event_log/rtc_event.h"
 #include "logging/rtc_event_log/encoder/bit_writer.h"
 #include "logging/rtc_event_log/encoder/var_int.h"
@@ -107,7 +107,7 @@ std::optional<FieldType> ConvertFieldType(uint64_t value) {
 
 std::string EncodeDeltasV3(FixedLengthEncodingParametersV3 params,
                            uint64_t base,
-                           ArrayView<const uint64_t> values) {
+                           std::span<const uint64_t> values) {
   size_t outputbound = (values.size() * params.delta_bit_width() + 7) / 8;
   BitWriter writer(outputbound);
 
@@ -142,7 +142,7 @@ std::string EncodeDeltasV3(FixedLengthEncodingParametersV3 params,
 }
 
 EventEncoder::EventEncoder(EventParameters params,
-                           ArrayView<const RtcEvent*> batch) {
+                           std::span<const RtcEvent*> batch) {
   batch_size_ = batch.size();
   if (!batch.empty()) {
     // Encode event type.
@@ -212,9 +212,9 @@ void EventEncoder::EncodeField(const FieldParameters& params,
   const bool values_optional = values.size() != batch_size_;
 
   // Compute delta parameters
-  ArrayView<const uint64_t> all_values(values);
+  std::span<const uint64_t> all_values(values);
   uint64_t base = values[0];
-  ArrayView<const uint64_t> remaining_values = all_values.subspan(1);
+  std::span<const uint64_t> remaining_values = all_values.subspan(1);
 
   FixedLengthEncodingParametersV3 delta_params =
       FixedLengthEncodingParametersV3::CalculateParameters(
diff --git a/logging/rtc_event_log/events/rtc_event_field_encoding.h b/logging/rtc_event_log/events/rtc_event_field_encoding.h
index baa50b2e6d..ba8427aeb0 100644
--- a/logging/rtc_event_log/events/rtc_event_field_encoding.h
+++ b/logging/rtc_event_log/events/rtc_event_field_encoding.h
@@ -14,12 +14,12 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <type_traits>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtc_event_log/rtc_event.h"
 #include "logging/rtc_event_log/events/fixed_length_encoding_parameters_v3.h"
 #include "logging/rtc_event_log/events/rtc_event_field_extraction.h"
@@ -71,7 +71,7 @@ struct FieldParameters {
 // The EventEncoder is used to encode a batch of events.
 class EventEncoder {
  public:
-  EventEncoder(EventParameters params, ArrayView<const RtcEvent*> batch);
+  EventEncoder(EventParameters params, std::span<const RtcEvent*> batch);
 
   void EncodeField(const FieldParameters& params,
                    const std::vector<uint64_t>& values,
@@ -94,7 +94,7 @@ class EventEncoder {
 std::string EncodeSingleValue(uint64_t value, FieldType field_type);
 std::string EncodeDeltasV3(FixedLengthEncodingParametersV3 params,
                            uint64_t base,
-                           ArrayView<const uint64_t> values);
+                           std::span<const uint64_t> values);
 
 // Given a batch of RtcEvents and a member pointer, extract that
 // member from each event in the batch. Signed integer members are
@@ -107,7 +107,7 @@ std::string EncodeDeltasV3(FixedLengthEncodingParametersV3 params,
 template <typename T,
           typename E,
           std::enable_if_t<std::is_integral<T>::value, bool> = true>
-std::vector<uint64_t> ExtractRtcEventMember(ArrayView<const RtcEvent*> batch,
+std::vector<uint64_t> ExtractRtcEventMember(std::span<const RtcEvent*> batch,
                                             const T E::* member) {
   std::vector<uint64_t> values;
   values.reserve(batch.size());
@@ -128,7 +128,7 @@ std::vector<uint64_t> ExtractRtcEventMember(ArrayView<const RtcEvent*> batch,
 template <typename T,
           typename E,
           std::enable_if_t<std::is_integral<T>::value, bool> = true>
-ValuesWithPositions ExtractRtcEventMember(ArrayView<const RtcEvent*> batch,
+ValuesWithPositions ExtractRtcEventMember(std::span<const RtcEvent*> batch,
                                           const std::optional<T> E::* member) {
   ValuesWithPositions result;
   result.position_mask.reserve(batch.size());
@@ -149,7 +149,7 @@ ValuesWithPositions ExtractRtcEventMember(ArrayView<const RtcEvent*> batch,
 template <typename T,
           typename E,
           std::enable_if_t<std::is_enum<T>::value, bool> = true>
-std::vector<uint64_t> ExtractRtcEventMember(ArrayView<const RtcEvent*> batch,
+std::vector<uint64_t> ExtractRtcEventMember(std::span<const RtcEvent*> batch,
                                             const T E::* member) {
   std::vector<uint64_t> values;
   values.reserve(batch.size());
@@ -164,7 +164,7 @@ std::vector<uint64_t> ExtractRtcEventMember(ArrayView<const RtcEvent*> batch,
 // Extract a string field from a batch of RtcEvents.
 template <typename E>
 std::vector<absl::string_view> ExtractRtcEventMember(
-    ArrayView<const RtcEvent*> batch,
+    std::span<const RtcEvent*> batch,
     const std::string E::* member) {
   std::vector<absl::string_view> values;
   values.reserve(batch.size());
diff --git a/logging/rtc_event_log/events/rtc_event_field_encoding_parser.cc b/logging/rtc_event_log/events/rtc_event_field_encoding_parser.cc
index 3e04b42ca4..ef97c46b61 100644
--- a/logging/rtc_event_log/events/rtc_event_field_encoding_parser.cc
+++ b/logging/rtc_event_log/events/rtc_event_field_encoding_parser.cc
@@ -14,10 +14,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <tuple>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "logging/rtc_event_log/encoder/var_int.h"
 #include "logging/rtc_event_log/events/fixed_length_encoding_parameters_v3.h"
 #include "logging/rtc_event_log/events/rtc_event_field_encoding.h"
@@ -355,15 +355,15 @@ RtcEventLogParseStatus EventParser::ParseField(const FieldParameters& params) {
   return RtcEventLogParseStatus::Success();
 }
 
-RtcEventLogParseStatusOr<ArrayView<absl::string_view>>
+RtcEventLogParseStatusOr<std::span<absl::string_view>>
 EventParser::ParseStringField(const FieldParameters& params,
                               bool required_field) {
-  using StatusOr = RtcEventLogParseStatusOr<ArrayView<absl::string_view>>;
+  using StatusOr = RtcEventLogParseStatusOr<std::span<absl::string_view>>;
   RTC_DCHECK_EQ(params.field_type, FieldType::kString);
   auto status = ParseField(params);
   if (!status.ok())
     return StatusOr(status);
-  ArrayView<absl::string_view> strings = GetStrings();
+  std::span<absl::string_view> strings = GetStrings();
   if (required_field && strings.size() != NumEventsInBatch()) {
     return StatusOr::Error("Required string field not found", __FILE__,
                            __LINE__);
@@ -371,15 +371,15 @@ EventParser::ParseStringField(const FieldParameters& params,
   return StatusOr(strings);
 }
 
-RtcEventLogParseStatusOr<ArrayView<uint64_t>> EventParser::ParseNumericField(
+RtcEventLogParseStatusOr<std::span<uint64_t>> EventParser::ParseNumericField(
     const FieldParameters& params,
     bool required_field) {
-  using StatusOr = RtcEventLogParseStatusOr<ArrayView<uint64_t>>;
+  using StatusOr = RtcEventLogParseStatusOr<std::span<uint64_t>>;
   RTC_DCHECK_NE(params.field_type, FieldType::kString);
   auto status = ParseField(params);
   if (!status.ok())
     return StatusOr(status);
-  ArrayView<uint64_t> values = GetValues();
+  std::span<uint64_t> values = GetValues();
   if (required_field && values.size() != NumEventsInBatch()) {
     return StatusOr::Error("Required numerical field not found", __FILE__,
                            __LINE__);
diff --git a/logging/rtc_event_log/events/rtc_event_field_encoding_parser.h b/logging/rtc_event_log/events/rtc_event_field_encoding_parser.h
index 2de51895a3..93b68cf392 100644
--- a/logging/rtc_event_log/events/rtc_event_field_encoding_parser.h
+++ b/logging/rtc_event_log/events/rtc_event_field_encoding_parser.h
@@ -14,13 +14,13 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <type_traits>
 #include <vector>
 
 #include "absl/base/attributes.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/units/timestamp.h"
 #include "logging/rtc_event_log/events/fixed_length_encoding_parameters_v3.h"
 #include "logging/rtc_event_log/events/rtc_event_field_encoding.h"
@@ -33,8 +33,8 @@ namespace webrtc {
 class EventParser {
  public:
   struct ValueAndPostionView {
-    ArrayView<uint64_t> values;
-    ArrayView<uint8_t> positions;
+    std::span<uint64_t> values;
+    std::span<uint8_t> positions;
   };
 
   EventParser() = default;
@@ -48,10 +48,10 @@ class EventParser {
   // other fields that may occur before it. If 'required_field == true',
   // then failing to find the field is an error, otherwise the functions
   // return success, but with an empty view of values.
-  RtcEventLogParseStatusOr<ArrayView<absl::string_view>> ParseStringField(
+  RtcEventLogParseStatusOr<std::span<absl::string_view>> ParseStringField(
       const FieldParameters& params,
       bool required_field = true);
-  RtcEventLogParseStatusOr<ArrayView<uint64_t>> ParseNumericField(
+  RtcEventLogParseStatusOr<std::span<uint64_t>> ParseNumericField(
       const FieldParameters& params,
       bool required_field = true);
   RtcEventLogParseStatusOr<ValueAndPostionView> ParseOptionalNumericField(
@@ -90,9 +90,9 @@ class EventParser {
   void SetError() { error_ = true; }
   bool Ok() const { return !error_; }
 
-  ArrayView<uint64_t> GetValues() { return values_; }
-  ArrayView<uint8_t> GetPositions() { return positions_; }
-  ArrayView<absl::string_view> GetStrings() { return strings_; }
+  std::span<uint64_t> GetValues() { return values_; }
+  std::span<uint8_t> GetPositions() { return positions_; }
+  std::span<absl::string_view> GetStrings() { return strings_; }
 
   void ClearTemporaries() {
     positions_.clear();
@@ -122,9 +122,9 @@ template <typename T,
           typename E,
           std::enable_if_t<std::is_integral<T>::value, bool> = true>
 ABSL_MUST_USE_RESULT RtcEventLogParseStatus
-PopulateRtcEventMember(const ArrayView<uint64_t> values,
+PopulateRtcEventMember(const std::span<uint64_t> values,
                        T E::* member,
-                       ArrayView<E> output) {
+                       std::span<E> output) {
   size_t batch_size = values.size();
   RTC_CHECK_EQ(output.size(), batch_size);
   for (size_t i = 0; i < batch_size; ++i) {
@@ -138,10 +138,10 @@ template <typename T,
           typename E,
           std::enable_if_t<std::is_integral<T>::value, bool> = true>
 ABSL_MUST_USE_RESULT RtcEventLogParseStatus
-PopulateRtcEventMember(const ArrayView<uint8_t> positions,
-                       const ArrayView<uint64_t> values,
+PopulateRtcEventMember(const std::span<uint8_t> positions,
+                       const std::span<uint64_t> values,
                        std::optional<T> E::* member,
-                       ArrayView<E> output) {
+                       std::span<E> output) {
   size_t batch_size = positions.size();
   RTC_CHECK_EQ(output.size(), batch_size);
   RTC_CHECK_LE(values.size(), batch_size);
@@ -164,9 +164,9 @@ template <typename T,
           typename E,
           std::enable_if_t<std::is_enum<T>::value, bool> = true>
 ABSL_MUST_USE_RESULT RtcEventLogParseStatus
-PopulateRtcEventMember(const ArrayView<uint64_t> values,
+PopulateRtcEventMember(const std::span<uint64_t> values,
                        T E::* member,
-                       ArrayView<E> output) {
+                       std::span<E> output) {
   size_t batch_size = values.size();
   RTC_CHECK_EQ(output.size(), batch_size);
   for (size_t i = 0; i < batch_size; ++i) {
@@ -182,9 +182,9 @@ PopulateRtcEventMember(const ArrayView<uint64_t> values,
 // Same as above, but for string fields.
 template <typename E>
 ABSL_MUST_USE_RESULT RtcEventLogParseStatus
-PopulateRtcEventMember(const ArrayView<absl::string_view> values,
+PopulateRtcEventMember(const std::span<absl::string_view> values,
                        std::string E::* member,
-                       ArrayView<E> output) {
+                       std::span<E> output) {
   size_t batch_size = values.size();
   RTC_CHECK_EQ(output.size(), batch_size);
   for (size_t i = 0; i < batch_size; ++i) {
@@ -197,9 +197,9 @@ PopulateRtcEventMember(const ArrayView<absl::string_view> values,
 // N.B. Assumes that the encoded value uses millisecond precision.
 template <typename E>
 ABSL_MUST_USE_RESULT RtcEventLogParseStatus
-PopulateRtcEventTimestamp(const ArrayView<uint64_t>& values,
+PopulateRtcEventTimestamp(const std::span<uint64_t>& values,
                           Timestamp E::* timestamp,
-                          ArrayView<E> output) {
+                          std::span<E> output) {
   size_t batch_size = values.size();
   RTC_CHECK_EQ(batch_size, output.size());
   for (size_t i = 0; i < batch_size; ++i) {
@@ -210,10 +210,10 @@ PopulateRtcEventTimestamp(const ArrayView<uint64_t>& values,
 }
 
 template <typename E>
-ArrayView<E> ExtendLoggedBatch(std::vector<E>& output, size_t new_elements) {
+std::span<E> ExtendLoggedBatch(std::vector<E>& output, size_t new_elements) {
   size_t old_size = output.size();
   output.resize(old_size + new_elements);
-  ArrayView<E> output_batch = output;
+  std::span<E> output_batch = output;
   output_batch = output_batch.subspan(old_size);
   RTC_DCHECK_EQ(output_batch.size(), new_elements);
   return output_batch;
diff --git a/logging/rtc_event_log/events/rtc_event_field_encoding_unittest.cc b/logging/rtc_event_log/events/rtc_event_field_encoding_unittest.cc
index 218197ace4..95a7a58c5c 100644
--- a/logging/rtc_event_log/events/rtc_event_field_encoding_unittest.cc
+++ b/logging/rtc_event_log/events/rtc_event_field_encoding_unittest.cc
@@ -14,13 +14,13 @@
 #include <cstdio>
 #include <limits>
 #include <optional>
+#include <span>
 #include <string>
 #include <tuple>
 #include <type_traits>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtc_event_log/rtc_event.h"
 #include "logging/rtc_event_log/encoder/var_int.h"
 #include "logging/rtc_event_log/events/rtc_event_field_encoding_parser.h"
@@ -366,8 +366,8 @@ class RtcEventFieldTest : public ::testing::Test {
     size_t size_before = parser_.RemainingBytes();
     auto result = parser_.ParseOptionalNumericField(params);
     ASSERT_TRUE(result.ok()) << result.message().c_str();
-    ArrayView<uint64_t> values = result.value().values;
-    ArrayView<uint8_t> positions = result.value().positions;
+    std::span<uint64_t> values = result.value().values;
+    std::span<uint8_t> positions = result.value().positions;
     ASSERT_EQ(positions.size(), expected_values.size());
     auto value_it = values.begin();
     for (size_t i = 0; i < expected_values.size(); i++) {
@@ -396,8 +396,8 @@ class RtcEventFieldTest : public ::testing::Test {
     auto result =
         parser_.ParseOptionalNumericField(params, /*required_field=*/false);
     ASSERT_TRUE(result.ok()) << result.message().c_str();
-    ArrayView<uint64_t> values = result.value().values;
-    ArrayView<uint8_t> positions = result.value().positions;
+    std::span<uint64_t> values = result.value().values;
+    std::span<uint8_t> positions = result.value().positions;
     EXPECT_EQ(positions.size(), 0u);
     EXPECT_EQ(values.size(), 0u);
   }
diff --git a/logging/rtc_event_log/events/rtc_event_frame_decoded.h b/logging/rtc_event_log/events/rtc_event_frame_decoded.h
index e7265458f2..c11664b959 100644
--- a/logging/rtc_event_log/events/rtc_event_frame_decoded.h
+++ b/logging/rtc_event_log/events/rtc_event_frame_decoded.h
@@ -15,11 +15,11 @@
 
 #include <map>
 #include <memory>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtc_event_log/rtc_event.h"
 #include "api/units/timestamp.h"
 #include "api/video/video_codec_type.h"
@@ -65,7 +65,7 @@ class RtcEventFrameDecoded final : public RtcEvent {
   VideoCodecType codec() const { return codec_; }
   uint8_t qp() const { return qp_; }
 
-  static std::string Encode(ArrayView<const RtcEvent*> /* batch */) {
+  static std::string Encode(std::span<const RtcEvent*> /* batch */) {
     // TODO(terelius): Implement
     return "";
   }
diff --git a/logging/rtc_event_log/events/rtc_event_ice_candidate_pair.h b/logging/rtc_event_log/events/rtc_event_ice_candidate_pair.h
index 3029c322d1..a935a3ae67 100644
--- a/logging/rtc_event_log/events/rtc_event_ice_candidate_pair.h
+++ b/logging/rtc_event_log/events/rtc_event_ice_candidate_pair.h
@@ -14,11 +14,11 @@
 #include <stdint.h>
 
 #include <memory>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtc_event_log/rtc_event.h"
 #include "api/units/timestamp.h"
 #include "logging/rtc_event_log/events/rtc_event_log_parse_status.h"
@@ -73,7 +73,7 @@ class RtcEventIceCandidatePair final : public RtcEvent {
   uint32_t candidate_pair_id() const { return candidate_pair_id_; }
   uint32_t transaction_id() const { return transaction_id_; }
 
-  static std::string Encode(ArrayView<const RtcEvent*> /* batch */) {
+  static std::string Encode(std::span<const RtcEvent*> /* batch */) {
     // TODO(terelius): Implement
     return "";
   }
diff --git a/logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.h b/logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.h
index 76bd1f7d35..097f789a56 100644
--- a/logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.h
+++ b/logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.h
@@ -14,11 +14,11 @@
 #include <stdint.h>
 
 #include <memory>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/candidate.h"
 #include "api/rtc_event_log/rtc_event.h"
 #include "api/units/timestamp.h"
@@ -118,7 +118,7 @@ class RtcEventIceCandidatePairConfig final : public RtcEvent {
     return candidate_pair_desc_;
   }
 
-  static std::string Encode(ArrayView<const RtcEvent*> /* batch */) {
+  static std::string Encode(std::span<const RtcEvent*> /* batch */) {
     // TODO(terelius): Implement
     return "";
   }
diff --git a/logging/rtc_event_log/events/rtc_event_probe_cluster_created.h b/logging/rtc_event_log/events/rtc_event_probe_cluster_created.h
index c789257fc6..f777f30e8a 100644
--- a/logging/rtc_event_log/events/rtc_event_probe_cluster_created.h
+++ b/logging/rtc_event_log/events/rtc_event_probe_cluster_created.h
@@ -14,11 +14,11 @@
 #include <stdint.h>
 
 #include <memory>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtc_event_log/rtc_event.h"
 #include "api/units/timestamp.h"
 #include "logging/rtc_event_log/events/rtc_event_log_parse_status.h"
@@ -69,7 +69,7 @@ class RtcEventProbeClusterCreated final : public RtcEvent {
   uint32_t min_probes() const { return min_probes_; }
   uint32_t min_bytes() const { return min_bytes_; }
 
-  static std::string Encode(ArrayView<const RtcEvent*> /* batch */) {
+  static std::string Encode(std::span<const RtcEvent*> /* batch */) {
     // TODO(terelius): Implement
     return "";
   }
diff --git a/logging/rtc_event_log/events/rtc_event_probe_result_failure.h b/logging/rtc_event_log/events/rtc_event_probe_result_failure.h
index 5d90b77498..985757b917 100644
--- a/logging/rtc_event_log/events/rtc_event_probe_result_failure.h
+++ b/logging/rtc_event_log/events/rtc_event_probe_result_failure.h
@@ -14,11 +14,11 @@
 #include <stdint.h>
 
 #include <memory>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtc_event_log/rtc_event.h"
 #include "api/units/timestamp.h"
 #include "logging/rtc_event_log/events/rtc_event_log_parse_status.h"
@@ -63,7 +63,7 @@ class RtcEventProbeResultFailure final : public RtcEvent {
   int32_t id() const { return id_; }
   ProbeFailureReason failure_reason() const { return failure_reason_; }
 
-  static std::string Encode(ArrayView<const RtcEvent*> /* batch */) {
+  static std::string Encode(std::span<const RtcEvent*> /* batch */) {
     // TODO(terelius): Implement
     return "";
   }
diff --git a/logging/rtc_event_log/events/rtc_event_probe_result_success.h b/logging/rtc_event_log/events/rtc_event_probe_result_success.h
index 198c6066e1..488cabe940 100644
--- a/logging/rtc_event_log/events/rtc_event_probe_result_success.h
+++ b/logging/rtc_event_log/events/rtc_event_probe_result_success.h
@@ -14,11 +14,11 @@
 #include <stdint.h>
 
 #include <memory>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtc_event_log/rtc_event.h"
 #include "api/units/timestamp.h"
 #include "logging/rtc_event_log/events/rtc_event_log_parse_status.h"
@@ -56,7 +56,7 @@ class RtcEventProbeResultSuccess final : public RtcEvent {
   int32_t id() const { return id_; }
   int32_t bitrate_bps() const { return bitrate_bps_; }
 
-  static std::string Encode(ArrayView<const RtcEvent*> /* batch */) {
+  static std::string Encode(std::span<const RtcEvent*> /* batch */) {
     // TODO(terelius): Implement
     return "";
   }
diff --git a/logging/rtc_event_log/events/rtc_event_remote_estimate.h b/logging/rtc_event_log/events/rtc_event_remote_estimate.h
index c1de1c6fa2..fe4a49e18d 100644
--- a/logging/rtc_event_log/events/rtc_event_remote_estimate.h
+++ b/logging/rtc_event_log/events/rtc_event_remote_estimate.h
@@ -12,11 +12,11 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtc_event_log/rtc_event.h"
 #include "api/units/data_rate.h"
 #include "api/units/timestamp.h"
@@ -48,7 +48,7 @@ class RtcEventRemoteEstimate final : public RtcEvent {
   Type GetType() const override { return kType; }
   bool IsConfigEvent() const override { return false; }
 
-  static std::string Encode(ArrayView<const RtcEvent*> /* batch */) {
+  static std::string Encode(std::span<const RtcEvent*> /* batch */) {
     // TODO(terelius): Implement
     return "";
   }
diff --git a/logging/rtc_event_log/events/rtc_event_route_change.h b/logging/rtc_event_log/events/rtc_event_route_change.h
index 6eb8a9c3a6..4827ed4743 100644
--- a/logging/rtc_event_log/events/rtc_event_route_change.h
+++ b/logging/rtc_event_log/events/rtc_event_route_change.h
@@ -13,11 +13,11 @@
 
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtc_event_log/rtc_event.h"
 #include "api/units/timestamp.h"
 #include "logging/rtc_event_log/events/rtc_event_log_parse_status.h"
@@ -53,7 +53,7 @@ class RtcEventRouteChange final : public RtcEvent {
   bool connected() const { return connected_; }
   uint32_t overhead() const { return overhead_; }
 
-  static std::string Encode(ArrayView<const RtcEvent*> /* batch */) {
+  static std::string Encode(std::span<const RtcEvent*> /* batch */) {
     // TODO(terelius): Implement
     return "";
   }
diff --git a/logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.cc b/logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.cc
index e5c834a1c6..8d0426a26e 100644
--- a/logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.cc
+++ b/logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.cc
@@ -12,15 +12,15 @@
 
 #include <cstdint>
 #include <memory>
+#include <span>
 
 #include "absl/memory/memory.h"
-#include "api/array_view.h"
 #include "api/rtc_event_log/rtc_event.h"
 
 namespace webrtc {
 
 RtcEventRtcpPacketIncoming::RtcEventRtcpPacketIncoming(
-    ArrayView<const uint8_t> packet)
+    std::span<const uint8_t> packet)
     : packet_(packet.data(), packet.size()) {}
 
 RtcEventRtcpPacketIncoming::RtcEventRtcpPacketIncoming(
diff --git a/logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.h b/logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.h
index 69c2e6b852..c23ae21800 100644
--- a/logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.h
+++ b/logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.h
@@ -14,11 +14,11 @@
 #include <stdint.h>
 
 #include <memory>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtc_event_log/rtc_event.h"
 #include "logging/rtc_event_log/events/logged_rtp_rtcp.h"
 #include "logging/rtc_event_log/events/rtc_event_log_parse_status.h"
@@ -30,7 +30,7 @@ class RtcEventRtcpPacketIncoming final : public RtcEvent {
  public:
   static constexpr Type kType = Type::RtcpPacketIncoming;
 
-  explicit RtcEventRtcpPacketIncoming(ArrayView<const uint8_t> packet);
+  explicit RtcEventRtcpPacketIncoming(std::span<const uint8_t> packet);
   ~RtcEventRtcpPacketIncoming() override;
 
   Type GetType() const override { return kType; }
@@ -40,7 +40,7 @@ class RtcEventRtcpPacketIncoming final : public RtcEvent {
 
   const Buffer& packet() const { return packet_; }
 
-  static std::string Encode(ArrayView<const RtcEvent*> /* batch */) {
+  static std::string Encode(std::span<const RtcEvent*> /* batch */) {
     // TODO(terelius): Implement
     return "";
   }
diff --git a/logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.cc b/logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.cc
index 5926e5c505..1b8472797c 100644
--- a/logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.cc
+++ b/logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.cc
@@ -12,15 +12,15 @@
 
 #include <cstdint>
 #include <memory>
+#include <span>
 
 #include "absl/memory/memory.h"
-#include "api/array_view.h"
 #include "api/rtc_event_log/rtc_event.h"
 
 namespace webrtc {
 
 RtcEventRtcpPacketOutgoing::RtcEventRtcpPacketOutgoing(
-    ArrayView<const uint8_t> packet)
+    std::span<const uint8_t> packet)
     : packet_(packet.data(), packet.size()) {}
 
 RtcEventRtcpPacketOutgoing::RtcEventRtcpPacketOutgoing(
diff --git a/logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.h b/logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.h
index 9916c94211..4704fea037 100644
--- a/logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.h
+++ b/logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.h
@@ -14,11 +14,11 @@
 #include <stdint.h>
 
 #include <memory>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtc_event_log/rtc_event.h"
 #include "logging/rtc_event_log/events/logged_rtp_rtcp.h"
 #include "logging/rtc_event_log/events/rtc_event_log_parse_status.h"
@@ -30,7 +30,7 @@ class RtcEventRtcpPacketOutgoing final : public RtcEvent {
  public:
   static constexpr Type kType = Type::RtcpPacketOutgoing;
 
-  explicit RtcEventRtcpPacketOutgoing(ArrayView<const uint8_t> packet);
+  explicit RtcEventRtcpPacketOutgoing(std::span<const uint8_t> packet);
   ~RtcEventRtcpPacketOutgoing() override;
 
   Type GetType() const override { return kType; }
@@ -40,7 +40,7 @@ class RtcEventRtcpPacketOutgoing final : public RtcEvent {
 
   const Buffer& packet() const { return packet_; }
 
-  static std::string Encode(ArrayView<const RtcEvent*> /* batch */) {
+  static std::string Encode(std::span<const RtcEvent*> /* batch */) {
     // TODO(terelius): Implement
     return "";
   }
diff --git a/logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.h b/logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.h
index 03bb5ef626..23cd3ae52a 100644
--- a/logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.h
+++ b/logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.h
@@ -16,12 +16,12 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtc_event_log/rtc_event.h"
 #include "logging/rtc_event_log/events/logged_rtp_rtcp.h"
 #include "logging/rtc_event_log/events/rtc_event_log_parse_status.h"
@@ -47,8 +47,8 @@ class RtcEventRtpPacketIncoming final : public RtcEvent {
 
   size_t packet_length() const { return packet_.size(); }
 
-  ArrayView<const uint8_t> RawHeader() const {
-    return MakeArrayView(packet_.data(), header_length());
+  std::span<const uint8_t> RawHeader() const {
+    return std::span(packet_.data(), header_length());
   }
   uint32_t Ssrc() const { return packet_.Ssrc(); }
   uint32_t Timestamp() const { return packet_.Timestamp(); }
@@ -60,7 +60,7 @@ class RtcEventRtpPacketIncoming final : public RtcEvent {
     return packet_.GetExtension<ExtensionTrait>(std::forward<Args>(args)...);
   }
   template <typename ExtensionTrait>
-  ArrayView<const uint8_t> GetRawExtension() const {
+  std::span<const uint8_t> GetRawExtension() const {
     return packet_.GetRawExtension<ExtensionTrait>();
   }
   template <typename ExtensionTrait>
@@ -75,7 +75,7 @@ class RtcEventRtpPacketIncoming final : public RtcEvent {
     return rtx_original_sequence_number_;
   }
 
-  static std::string Encode(ArrayView<const RtcEvent*> /* batch */) {
+  static std::string Encode(std::span<const RtcEvent*> /* batch */) {
     // TODO(terelius): Implement
     return "";
   }
diff --git a/logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h b/logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h
index bfe55fd369..ce3931e3fa 100644
--- a/logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h
+++ b/logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h
@@ -16,12 +16,12 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtc_event_log/rtc_event.h"
 #include "logging/rtc_event_log/events/logged_rtp_rtcp.h"
 #include "logging/rtc_event_log/events/rtc_event_log_parse_status.h"
@@ -48,8 +48,8 @@ class RtcEventRtpPacketOutgoing final : public RtcEvent {
 
   size_t packet_length() const { return packet_.size(); }
 
-  ArrayView<const uint8_t> RawHeader() const {
-    return MakeArrayView(packet_.data(), header_length());
+  std::span<const uint8_t> RawHeader() const {
+    return std::span(packet_.data(), header_length());
   }
   uint32_t Ssrc() const { return packet_.Ssrc(); }
   uint32_t Timestamp() const { return packet_.Timestamp(); }
@@ -61,7 +61,7 @@ class RtcEventRtpPacketOutgoing final : public RtcEvent {
     return packet_.GetExtension<ExtensionTrait>(std::forward<Args>(args)...);
   }
   template <typename ExtensionTrait>
-  ArrayView<const uint8_t> GetRawExtension() const {
+  std::span<const uint8_t> GetRawExtension() const {
     return packet_.GetRawExtension<ExtensionTrait>();
   }
   template <typename ExtensionTrait>
@@ -77,7 +77,7 @@ class RtcEventRtpPacketOutgoing final : public RtcEvent {
     return rtx_original_sequence_number_;
   }
 
-  static std::string Encode(ArrayView<const RtcEvent*> /* batch */) {
+  static std::string Encode(std::span<const RtcEvent*> /* batch */) {
     // TODO(terelius): Implement
     return "";
   }
diff --git a/logging/rtc_event_log/events/rtc_event_video_receive_stream_config.h b/logging/rtc_event_log/events/rtc_event_video_receive_stream_config.h
index f8c222f88c..c1c5103d2b 100644
--- a/logging/rtc_event_log/events/rtc_event_video_receive_stream_config.h
+++ b/logging/rtc_event_log/events/rtc_event_video_receive_stream_config.h
@@ -13,11 +13,11 @@
 
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtc_event_log/rtc_event.h"
 #include "api/units/timestamp.h"
 #include "logging/rtc_event_log/events/rtc_event_log_parse_status.h"
@@ -53,7 +53,7 @@ class RtcEventVideoReceiveStreamConfig final : public RtcEvent {
 
   const rtclog::StreamConfig& config() const { return *config_; }
 
-  static std::string Encode(ArrayView<const RtcEvent*> /* batch */) {
+  static std::string Encode(std::span<const RtcEvent*> /* batch */) {
     // TODO(terelius): Implement
     return "";
   }
diff --git a/logging/rtc_event_log/events/rtc_event_video_send_stream_config.h b/logging/rtc_event_log/events/rtc_event_video_send_stream_config.h
index 2d5b85c56f..eeb0810069 100644
--- a/logging/rtc_event_log/events/rtc_event_video_send_stream_config.h
+++ b/logging/rtc_event_log/events/rtc_event_video_send_stream_config.h
@@ -13,11 +13,11 @@
 
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtc_event_log/rtc_event.h"
 #include "api/units/timestamp.h"
 #include "logging/rtc_event_log/events/rtc_event_log_parse_status.h"
@@ -53,7 +53,7 @@ class RtcEventVideoSendStreamConfig final : public RtcEvent {
 
   const rtclog::StreamConfig& config() const { return *config_; }
 
-  static std::string Encode(ArrayView<const RtcEvent*> /* batch */) {
+  static std::string Encode(std::span<const RtcEvent*> /* batch */) {
     // TODO(terelius): Implement
     return "";
   }
diff --git a/logging/rtc_event_log/rtc_event_log2rtp_dump.cc b/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
index 67b136e4c8..abe5a44d4c 100644
--- a/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
+++ b/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
@@ -13,6 +13,7 @@
 #include <iostream>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
@@ -20,7 +21,6 @@
 #include "absl/flags/parse.h"
 #include "absl/flags/usage.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtp_headers.h"
 #include "logging/rtc_event_log/events/logged_rtp_rtcp.h"
 #include "logging/rtc_event_log/rtc_event_log_parser.h"
@@ -108,7 +108,7 @@ void ConvertRtpPacket(
   reconstructed_packet.SetTimestamp(incoming.rtp.header.timestamp);
   reconstructed_packet.SetSsrc(incoming.rtp.header.ssrc);
   if (incoming.rtp.header.numCSRCs > 0) {
-    reconstructed_packet.SetCsrcs(webrtc::ArrayView<const uint32_t>(
+    reconstructed_packet.SetCsrcs(std::span<const uint32_t>(
         incoming.rtp.header.arrOfCSRCs, incoming.rtp.header.numCSRCs));
   }
 
diff --git a/logging/rtc_event_log/rtc_event_log_unittest_helper.cc b/logging/rtc_event_log/rtc_event_log_unittest_helper.cc
index db573e7913..b711b7212e 100644
--- a/logging/rtc_event_log/rtc_event_log_unittest_helper.cc
+++ b/logging/rtc_event_log/rtc_event_log_unittest_helper.cc
@@ -17,12 +17,12 @@
 #include <memory>
 #include <numeric>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
 #include "absl/algorithm/container.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/candidate.h"
 #include "api/dtls_transport_interface.h"
 #include "api/rtc_event_log/rtc_event_log.h"
@@ -127,7 +127,7 @@ constexpr ExtensionPair kExtensions[kMaxNumExtensions] = {
      .name = RtpExtension::kDependencyDescriptorUri}};
 
 template <typename T>
-void ShuffleInPlace(Random* prng, ArrayView<T> array) {
+void ShuffleInPlace(Random* prng, std::span<T> array) {
   RTC_DCHECK_LE(array.size(), std::numeric_limits<uint32_t>::max());
   for (uint32_t i = 0; i + 1 < array.size(); i++) {
     uint32_t other = prng->Rand(i, static_cast<uint32_t>(array.size() - 1));
@@ -657,7 +657,7 @@ RtpHeaderExtensionMap EventGenerator::NewRtpHeaderExtensionMap(
   std::vector<int> id(RtpExtension::kOneByteHeaderExtensionMaxId -
                       RtpExtension::kMinId + 1);
   std::iota(id.begin(), id.end(), RtpExtension::kMinId);
-  ShuffleInPlace(&prng_, ArrayView<int>(id));
+  ShuffleInPlace(&prng_, std::span<int>(id));
 
   auto not_excluded = [&](RTPExtensionType type) -> bool {
     return !absl::c_linear_search(excluded_extensions, type);
@@ -1031,7 +1031,7 @@ void EventVerifier::VerifyLoggedDependencyDescriptor(
     const Event& packet,
     const std::vector<uint8_t>& logged_dd) const {
   if (expect_dependency_descriptor_rtp_header_extension_is_set_) {
-    ArrayView<const uint8_t> original =
+    std::span<const uint8_t> original =
         packet.template GetRawExtension<RtpDependencyDescriptorExtension>();
     EXPECT_THAT(logged_dd, ElementsAreArray(original));
   } else {

From 9e3381393a233cac896085fc3f26bdac83caaaef Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Wed, 11 Mar 2026 09:28:58 +0000
Subject: [PATCH 260/472] Fix payload type allocation issues for Audio RED and
 MID recycling.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This CL addresses several issues identified while enabling the
WebRTC-PayloadTypesInTransport field trial:
- Enforce media type equality in codec matching to prevent Audio/Video
RED collisions.
- Relax RED matching rules to allow negotiation with unlinked RED codecs.
- Validate media type during MID recycling in CodecVendor to prevent
incorrect codec merging.
- Update documentation with the video implementation strategy.

Bug: webrtc:360058654
Change-Id: I1ecbdad2179d346b3682875d823889aa076e353e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453001
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47113}
---
 g3doc/todo/payload_type_redesign.md           | 127 ++++
 media/base/codec_comparators.cc               |  51 +-
 media/base/codec_comparators_unittest.cc      |  57 ++
 pc/BUILD.gn                                   |   1 +
 pc/codec_vendor.cc                            |   7 +-
 ...er_connection_encodings_integrationtest.cc |  58 ++
 pc/peer_connection_media_unittest.cc          |   4 +-
 ...er_connection_stability_integrationtest.cc | 549 ++++++++++++++++++
 8 files changed, 833 insertions(+), 21 deletions(-)
 create mode 100644 g3doc/todo/payload_type_redesign.md

diff --git a/g3doc/todo/payload_type_redesign.md b/g3doc/todo/payload_type_redesign.md
new file mode 100644
index 0000000000..5fa5f12bca
--- /dev/null
+++ b/g3doc/todo/payload_type_redesign.md
@@ -0,0 +1,127 @@
+# Payload type allocation redesign
+
+## Background: What a payload type is
+
+The payload type is a property of a codec that is established between a sender
+and a receiver using SDP offer/answer.
+
+Each end of the connection independently indicates the list of codecs it is
+willing to send and/or receive, and assigns a payload type to each. It is
+conventional but not required to use the same payload types in an answer as
+those suggested in an offer.
+
+For SDP sendonly media sections, it indicates a willingness to send; for
+recvonly media sections, it indicates a willingness to receive; for sendrecv
+media sections, it indicates a willingness to receive AND a willingness to send.
+Presence in a media section does not guarantee that it will be used. (RFC 3264)
+
+## Current allocation strategy
+
+The payload types are assigned by the VideoEngine and VoiceEngine according to a
+fixed list. If the assignments collide with already established payload types,
+the assignments are munged in the CodecVendor class so that there are no
+colliding payload types.
+
+The PayloadTypeRecorder class records what payload types are assigned - there is
+one case (PR-answer followed by a later Answer that has different PTs) where
+they are changed, but otherwise, once the PayloadTypeRecorder has assigned them,
+that PT is permanent for that transport and that direction.
+
+Picking a payload type is done in the PayloadTypePicker class. This is preloaded
+with some assignments, so that if a PT is free, the commonly used PT for that
+codec can be used.
+
+## Desired future strategy
+
+There should be no assignment of payload types in VideoEngine and VoiceEngine.
+Payload types should be picked only when creating an offer or answer; they are
+assigned permanently in SetLocalDescription / SetRemoteDescription.
+
+Once an assignment is made, it should never need to change (apart from the case
+noted above).
+
+## Dealing with resiliency mechanisms
+
+Resiliency mechanisms (RED, RTX) are signaled using "codecs", and are
+traditionally represented as such within the library. They associate with the
+codec they are resiliency for by using the "apt=" parameter in their a=fmtp
+lines. However, this complicates assignment, since that parameter cannot be
+filled in until the payload type for the "protected" codec is assigned.
+
+## Backwards compatibility issues
+
+Experience has shown that changing the PT allocation strategy often trips up
+applications that have depended on the result of the old allocation strategy, so
+user-visible changes should be avoided. One example is that it's preferred (but
+not strictly required) to have the RTX PT assigned the PT number of 1 more than
+the PT of the codec it refers to.
+
+## Current implementation status
+
+The new strategy is implemented for audio codecs. Several issues that caused
+test failures when enabling the `WebRTC-PayloadTypesInTransport` field trial
+have been identified and fixed:
+
+- **Audio/Video RED Collision:** RED codecs of different media types were
+  incorrectly matching, leading to payload type conflicts.
+  `MatchesWithCodecRules` now enforces media type equality.
+- **MID Recycling:** When a MID was recycled for a different media type (e.g.,
+  Audio -> Video), `CodecVendor` was incorrectly merging codecs from the old
+  description. This has been fixed by validating the media type before merging.
+- **RED Matching Logic:** Relaxed the matching rules for RED to allow
+  negotiation to proceed even when parameters (linking RED to primary codecs)
+  are not yet populated, as this linking now happens late in the `CodecVendor`.
+
+The new strategy is not yet implemented for video codecs.
+
+## Implementation Strategy for Video
+
+The goal is to transition video codec handling to the same late-assignment model
+used for audio.
+
+### 1. Unified Codec Collection
+
+- Implement `VideoCodecsFromFactory` in `pc/typed_codec_vendor.cc`. This will
+  query the `VideoEncoderFactory` and `VideoDecoderFactory` to gather supported
+  `SdpVideoFormat`s.
+- These codecs will be initialized with `kIdNotSet`, allowing
+  `SdpPayloadTypeSuggester` to assign payload types only during the creation of
+  an offer or answer.
+
+### 2. Payload Type Suggester Integration
+
+- Ensure `PayloadTypePicker` has a complete list of preferred payload types for
+  video codecs (VP8, VP9, H.264, AV1, etc.) to maintain stable and conventional
+  assignments.
+- Update `CodecVendor` to use the unassigned video codec list when the
+  `WebRTC-PayloadTypesInTransport` trial is active.
+
+### 3. Resiliency and Parameter Linking
+
+- Refine `MergeCodecs` and `AssignCodecIdsAndLinkRed` to handle video-specific
+  resiliency:
+  - **RTX:** Correctly link RTX codecs to their primary video codecs by matching
+    names and specific parameters (like `profile-level-id` for H.264).
+  - **RED/ULPFEC:** Ensure parameters for video redundancy are correctly
+    populated once the primary codec PTs are assigned.
+
+### 4. Verification and Testing
+
+- **Integration Tests:** Enable the `WebRTC-PayloadTypesInTransport` trial in
+  `peerconnection_unittests` and `rtc_unittests` to identify any video-specific
+  regressions.
+- **Stable PT Tests:** Add coverage to ensure that payload types remain stable
+  across renegotiations, even when the order of codecs in the transceiver
+  preferences changes.
+- **MID Recycling:** Verify that MID recycling for video-to-audio and vice-versa
+  works correctly without PT collisions or crashes.
+
+## Desired next steps
+
+Analyze the current situation. (Done)
+
+### Test, isolate and fix failures
+
+The identified failures in `PeerConnectionEncodingsIntegrationTest` and
+`PeerConnectionIntegrationTest` have been resolved. Next, focus on implementing
+the video strategy outlined above.
diff --git a/media/base/codec_comparators.cc b/media/base/codec_comparators.cc
index 9da9d4aca5..cb67f671a6 100644
--- a/media/base/codec_comparators.cc
+++ b/media/base/codec_comparators.cc
@@ -163,15 +163,7 @@ bool MatchesWithReferenceAttributesAndComparator(
         potential_match.params.find(kCodecParamNotInNameValueFormat);
     bool has_parameters_1 = red_parameters_1 != codec_to_match.params.end();
     bool has_parameters_2 = red_parameters_2 != potential_match.params.end();
-    // If codec_to_match has unassigned PT and no parameter,
-    // we assume that it'll be assigned later and return a match.
-    // Note - this should be deleted. It's untidy.
-    if (potential_match.id == Codec::kIdNotSet && !has_parameters_2) {
-      return true;
-    }
-    if (codec_to_match.id == Codec::kIdNotSet && !has_parameters_1) {
-      return true;
-    }
+
     if (has_parameters_1 && has_parameters_2) {
       // Different levels of redundancy between offer and answer are OK
       // since RED is considered to be declarative.
@@ -203,7 +195,16 @@ bool MatchesWithReferenceAttributesAndComparator(
       return true;
     }
     if (!has_parameters_1 && !has_parameters_2) {
-      // Both parameters are missing. Happens for video RED.
+      return true;
+    }
+    // Exactly one lacks parameters.
+    // Allow match if it is an audio RED codec and at least one of the
+    // codecs has an unassigned payload type or they have the same ID.
+    if (codec_to_match.type == Codec::Type::kAudio &&
+        codec_to_match.name == kRedCodecName &&
+        (codec_to_match.id == Codec::kIdNotSet ||
+         potential_match.id == Codec::kIdNotSet ||
+         codec_to_match.id == potential_match.id)) {
       return true;
     }
     return false;
@@ -295,6 +296,11 @@ bool MatchesWithCodecRules(const Codec& left_codec, const Codec& right_codec) {
        right_codec.id <= kLowerDynamicRangeMax) ||
       (right_codec.id >= kUpperDynamicRangeMin &&
        right_codec.id <= kUpperDynamicRangeMax);
+
+  if (left_codec.type != right_codec.type) {
+    return false;
+  }
+
   bool matches_id;
   if ((is_id_in_dynamic_range && is_codec_id_in_dynamic_range) ||
       left_codec.id == Codec::kIdNotSet || right_codec.id == Codec::kIdNotSet) {
@@ -360,12 +366,22 @@ std::optional<Codec> FindMatchingCodec(const std::vector<Codec>& codecs1,
 bool IsSameRtpCodec(const Codec& codec, const RtpCodec& rtp_codec) {
   RtpCodecParameters rtp_codec2 = codec.ToCodecParameters();
 
-  return absl::EqualsIgnoreCase(rtp_codec.name, rtp_codec2.name) &&
-         rtp_codec.kind == rtp_codec2.kind &&
-         rtp_codec.num_channels == rtp_codec2.num_channels &&
-         rtp_codec.clock_rate == rtp_codec2.clock_rate &&
-         InsertDefaultParams(rtp_codec.name, rtp_codec.parameters) ==
-             InsertDefaultParams(rtp_codec2.name, rtp_codec2.parameters);
+  if (!absl::EqualsIgnoreCase(rtp_codec.name, rtp_codec2.name) ||
+      rtp_codec.kind != rtp_codec2.kind ||
+      rtp_codec.num_channels != rtp_codec2.num_channels ||
+      rtp_codec.clock_rate != rtp_codec2.clock_rate) {
+    return false;
+  }
+
+  // audio/RED should ignore the parameters which specify payload types so
+  // can not be compared.
+  if (rtp_codec.kind == MediaType::AUDIO &&
+      absl::EqualsIgnoreCase(rtp_codec.name, kRedCodecName)) {
+    return true;
+  }
+
+  return InsertDefaultParams(rtp_codec.name, rtp_codec.parameters) ==
+         InsertDefaultParams(rtp_codec2.name, rtp_codec2.parameters);
 }
 
 bool IsSameRtpCodecIgnoringLevel(const Codec& codec,
@@ -393,7 +409,8 @@ bool IsSameRtpCodecIgnoringLevel(const Codec& codec,
   }
   // audio/RED should ignore the parameters which specify payload types so
   // can not be compared.
-  if (rtp_codec.kind == MediaType::AUDIO && rtp_codec.name == kRedCodecName) {
+  if (rtp_codec.kind == MediaType::AUDIO &&
+      absl::EqualsIgnoreCase(rtp_codec.name, kRedCodecName)) {
     return true;
   }
 
diff --git a/media/base/codec_comparators_unittest.cc b/media/base/codec_comparators_unittest.cc
index 2f78042c16..048765670e 100644
--- a/media/base/codec_comparators_unittest.cc
+++ b/media/base/codec_comparators_unittest.cc
@@ -105,6 +105,63 @@ TEST(CodecComparatorsTest, MatchesWithReferenceAttributesRed) {
   EXPECT_FALSE(MatchesWithReferenceAttributes(codec_1, codec_5));
 }
 
+TEST(CodecComparatorsTest, RedParametersMismatch) {
+  Codec with_params = CreateAudioCodec(101, kRedCodecName, 48000, 2);
+  with_params.SetParam(kCodecParamNotInNameValueFormat, "111/111");
+
+  Codec no_params = CreateAudioCodec(102, kRedCodecName, 48000, 2);
+
+  // Case 1: Exactly one lacks parameters, both have IDs -> SHOULD NOT match for
+  // audio RED.
+  EXPECT_FALSE(MatchesWithReferenceAttributes(with_params, no_params));
+  EXPECT_FALSE(MatchesWithReferenceAttributes(no_params, with_params));
+
+  // Case 2: Exactly one lacks parameters, the one WITHOUT parameters is
+  // unassigned -> SHOULD match.
+  // This is the case with late assignment.
+  Codec no_params_unassigned = no_params;
+  no_params_unassigned.id = Codec::kIdNotSet;
+  EXPECT_TRUE(
+      MatchesWithReferenceAttributes(with_params, no_params_unassigned));
+  EXPECT_TRUE(
+      MatchesWithReferenceAttributes(no_params_unassigned, with_params));
+
+  // Case 3: Exactly one lacks parameters, the one WITH parameters is
+  // unassigned -> SHOULD match for audio RED because one side is unassigned.
+  Codec with_params_unassigned = with_params;
+  with_params_unassigned.id = Codec::kIdNotSet;
+  EXPECT_TRUE(
+      MatchesWithReferenceAttributes(with_params_unassigned, no_params));
+  EXPECT_TRUE(
+      MatchesWithReferenceAttributes(no_params, with_params_unassigned));
+
+  // Case 4: Exactly one lacks parameters, both are unassigned -> SHOULD match.
+  EXPECT_TRUE(MatchesWithReferenceAttributes(with_params_unassigned,
+                                             no_params_unassigned));
+  EXPECT_TRUE(MatchesWithReferenceAttributes(no_params_unassigned,
+                                             with_params_unassigned));
+
+  // Case 5: Both lack parameters -> SHOULD match.
+  Codec no_params_2 = CreateAudioCodec(103, kRedCodecName, 48000, 2);
+  EXPECT_TRUE(MatchesWithReferenceAttributes(no_params, no_params_2));
+
+  // A video RED codec should not match any audio RED codec,
+  // independent of parameters.
+  Codec video_red_codec = CreateVideoCodec(107, kRedCodecName);
+  EXPECT_FALSE(MatchesWithReferenceAttributes(no_params, video_red_codec));
+  EXPECT_FALSE(MatchesWithReferenceAttributes(with_params, video_red_codec));
+
+  // Two video RED codecs with different IDs and one lacking parameters
+  // should NOT match.
+  Codec video_red_with_params = CreateVideoCodec(108, kRedCodecName);
+  video_red_with_params.SetParam(kCodecParamNotInNameValueFormat, "120/120");
+  Codec video_red_no_params = CreateVideoCodec(109, kRedCodecName);
+  EXPECT_FALSE(MatchesWithReferenceAttributes(video_red_with_params,
+                                              video_red_no_params));
+  EXPECT_FALSE(MatchesWithReferenceAttributes(video_red_no_params,
+                                              video_red_with_params));
+}
+
 struct TestParams {
   std::string name;
   SdpVideoFormat codec1;
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 9863aa6da6..4315b46bb2 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -407,6 +407,7 @@ rtc_library("codec_vendor") {
     ":typed_codec_vendor",
     ":used_ids",
     "../api:field_trials_view",
+    "../api:payload_type",
     "../api:rtc_error",
     "../api:rtp_parameters",
     "../api:rtp_transceiver_direction",
diff --git a/pc/codec_vendor.cc b/pc/codec_vendor.cc
index 1d865fd7de..50f3ef818f 100644
--- a/pc/codec_vendor.cc
+++ b/pc/codec_vendor.cc
@@ -23,6 +23,7 @@
 #include "absl/strings/string_view.h"
 #include "api/field_trials_view.h"
 #include "api/media_types.h"
+#include "api/payload_type.h"
 #include "api/rtc_error.h"
 #include "api/rtp_parameters.h"
 #include "api/rtp_transceiver_direction.h"
@@ -620,7 +621,8 @@ RTCErrorOr<std::vector<Codec>> CodecVendor::GetNegotiatedCodecsForOffer(
   CodecList codecs;
   std::string mid = media_description_options.mid;
   // If current content exists and is not being recycled, use its codecs.
-  if (current_content && current_content->mid() == mid) {
+  if (current_content && current_content->mid() == mid &&
+      IsMediaContentOfType(current_content, media_description_options.type)) {
     RTCErrorOr<CodecList> checked_codec_list =
         CodecList::Create(current_content->media_description()->codecs());
     if (!checked_codec_list.ok()) {
@@ -758,7 +760,8 @@ RTCErrorOr<Codecs> CodecVendor::GetNegotiatedCodecsForAnswer(
   RTC_LOG_THREAD_BLOCK_COUNT();
   CodecList codecs;
   std::string mid = media_description_options.mid;
-  if (current_content && current_content->mid() == mid) {
+  if (current_content && current_content->mid() == mid &&
+      IsMediaContentOfType(current_content, media_description_options.type)) {
     RTCErrorOr<CodecList> checked_codec_list =
         CodecList::Create(current_content->media_description()->codecs());
     if (!checked_codec_list.ok()) {
diff --git a/pc/peer_connection_encodings_integrationtest.cc b/pc/peer_connection_encodings_integrationtest.cc
index 6b84d6bf93..9b9ee258c1 100644
--- a/pc/peer_connection_encodings_integrationtest.cc
+++ b/pc/peer_connection_encodings_integrationtest.cc
@@ -2129,6 +2129,64 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
   EXPECT_EQ(parameters.codecs[0].name, opus->name);
 }
 
+TEST_F(PeerConnectionEncodingsIntegrationTest,
+       EncodingParametersRedEnabledBeforeNegotiationAudioWithFieldTrial) {
+  scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper =
+      CreatePc("WebRTC-PayloadTypesInTransport/Enabled/");
+  scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper =
+      CreatePc("WebRTC-PayloadTypesInTransport/Enabled/");
+  ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
+
+  std::vector<RtpCodecCapability> send_codecs =
+      local_pc_wrapper->pc_factory()
+          ->GetRtpSenderCapabilities(MediaType::AUDIO)
+          .codecs;
+
+  std::optional<RtpCodecCapability> opus =
+      local_pc_wrapper->FindFirstSendCodecWithName(MediaType::AUDIO, "opus");
+  ASSERT_TRUE(opus);
+
+  std::optional<RtpCodecCapability> red =
+      local_pc_wrapper->FindFirstSendCodecWithName(MediaType::AUDIO, "red");
+  ASSERT_TRUE(red);
+
+  RtpTransceiverInit init;
+  init.direction = RtpTransceiverDirection::kSendOnly;
+  RtpEncodingParameters encoding_parameters;
+  encoding_parameters.codec = opus;
+  init.send_encodings.push_back(encoding_parameters);
+
+  auto transceiver_or_error =
+      local_pc_wrapper->pc()->AddTransceiver(MediaType::AUDIO, init);
+  ASSERT_TRUE(transceiver_or_error.ok());
+  scoped_refptr<RtpTransceiverInterface> audio_transceiver =
+      transceiver_or_error.MoveValue();
+
+  // Preferring RED over Opus should enable RED with Opus encoding.
+  send_codecs[0] = red.value();
+  send_codecs[1] = opus.value();
+
+  ASSERT_TRUE(audio_transceiver->SetCodecPreferences(send_codecs).ok());
+  NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper);
+  local_pc_wrapper->WaitForConnection();
+  remote_pc_wrapper->WaitForConnection();
+
+  RtpParameters parameters = audio_transceiver->sender()->GetParameters();
+  EXPECT_EQ(parameters.encodings[0].codec, opus);
+  EXPECT_EQ(parameters.codecs[0].name, red->name);
+
+  // Check that it's possible to switch back to Opus without RED.
+  send_codecs[0] = opus.value();
+  send_codecs[1] = red.value();
+
+  ASSERT_TRUE(audio_transceiver->SetCodecPreferences(send_codecs).ok());
+  NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper);
+
+  parameters = audio_transceiver->sender()->GetParameters();
+  EXPECT_EQ(parameters.encodings[0].codec, opus);
+  EXPECT_EQ(parameters.codecs[0].name, opus->name);
+}
+
 TEST_F(PeerConnectionEncodingsIntegrationTest,
        SetParametersRejectsScalabilityModeForSelectedCodec) {
   scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
diff --git a/pc/peer_connection_media_unittest.cc b/pc/peer_connection_media_unittest.cc
index 3890be2dc3..a4671c9906 100644
--- a/pc/peer_connection_media_unittest.cc
+++ b/pc/peer_connection_media_unittest.cc
@@ -1376,10 +1376,10 @@ TEST_P(PeerConnectionMediaTest, RedFmtpPayloadTypeMustMatchBaseCodecs) {
       CreateAudioCodec(120, "foo", kDefaultAudioClockRateHz, 1));
   callee_fake_codecs.push_back(
       CreateAudioCodec(121, kRedCodecName, kDefaultAudioClockRateHz, 1));
-  callee_fake_codecs.push_back(
-      CreateAudioCodec(122, "bar", kDefaultAudioClockRateHz, 1));
   callee_fake_codecs.back().SetParam(kCodecParamNotInNameValueFormat,
                                      "122/122");
+  callee_fake_codecs.push_back(
+      CreateAudioCodec(122, "bar", kDefaultAudioClockRateHz, 1));
   auto callee_fake_engine = std::make_unique<FakeMediaEngine>();
   callee_fake_engine->SetAudioCodecs(callee_fake_codecs);
   auto callee = CreatePeerConnectionWithAudio(std::move(callee_fake_engine));
diff --git a/pc/peer_connection_stability_integrationtest.cc b/pc/peer_connection_stability_integrationtest.cc
index 8d3052215b..3acfd99619 100644
--- a/pc/peer_connection_stability_integrationtest.cc
+++ b/pc/peer_connection_stability_integrationtest.cc
@@ -949,5 +949,554 @@ TEST_F(PeerConnectionIntegrationTest, BasicOfferAnswerPayloadTypesStable) {
               ElementsAreArray(this_golden.callee_remote));
 }
 
+TEST_F(PeerConnectionIntegrationTest,
+       BasicOfferAnswerPayloadTypesStableWithFieldTrial) {
+  SetFieldTrials("WebRTC-PayloadTypesInTransport/Enabled/");
+  FactorySignature factory_signature;
+  ASSERT_THAT(factory_signature.id(),
+              Not(Eq(FactorySignature::Id::kNotRecognized)));
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignalingForSdpOnly();
+  caller()->AddAudioVideoTracks();
+  callee()->AddAudioVideoTracks();
+  // Start offer/answer exchange and wait for it to complete.
+  caller()->CreateAndSetAndSignalOffer();
+
+  ASSERT_THAT(
+      WaitUntil([&] { return SignalingStateStable(); }, ::testing::IsTrue()),
+      IsRtcOk());
+
+  // Extract PT and codec from all media sections, and check that they
+  // are stable (what was expected).
+  // Maintenance: In order to get a new golden set of strings, make the list
+  // empty and run. Gmock will output a valid C++ array initializer for you.
+
+  std::vector<ResultingCodecList> golden_answers = {
+      {.factory_id = FactorySignature::Id::kWebRtcTipOfTree,
+       .caller_local =
+           {"1 [111:audio/opus/48000/2;minptime=10;useinbandfec=1]",
+            "1 [63:audio/red/48000/2;=111/111]",
+            "1 [9:audio/G722/8000/1]",
+            "1 [0:audio/PCMU/8000/1]",
+            "1 [8:audio/PCMA/8000/1]",
+            "1 [13:audio/CN/8000/1]",
+            "1 [110:audio/telephone-event/48000/1]",
+            "1 [126:audio/telephone-event/8000/1]",
+            "2 [96:video/VP8/90000/0]",
+            "2 [97:video/rtx/90000/0;apt=96]",
+            "2 "
+            "[103:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-id="
+            "42001f]",
+            "2 [104:video/rtx/90000/0;apt=103]",
+            "2 "
+            "[107:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-id="
+            "42001f]",
+            "2 [108:video/rtx/90000/0;apt=107]",
+            "2 "
+            "[109:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-id="
+            "42e01f]",
+            "2 [114:video/rtx/90000/0;apt=109]",
+            "2 "
+            "[115:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-id="
+            "42e01f]",
+            "2 [116:video/rtx/90000/0;apt=115]",
+            "2 "
+            "[117:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-id="
+            "4d001f]",
+            "2 [118:video/rtx/90000/0;apt=117]",
+            "2 "
+            "[39:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-id="
+            "4d001f]",
+            "2 [40:video/rtx/90000/0;apt=39]",
+            "2 [45:video/AV1/90000/0;level-idx=5;profile=0;tier=0]",
+            "2 [46:video/rtx/90000/0;apt=45]",
+            "2 [98:video/VP9/90000/0;profile-id=0]",
+            "2 [99:video/rtx/90000/0;apt=98]",
+            "2 [100:video/VP9/90000/0;profile-id=2]",
+            "2 [101:video/rtx/90000/0;apt=100]",
+            "2 [119:video/red/90000/0]",
+            "2 [120:video/rtx/90000/0;apt=119]",
+            "2 [121:video/ulpfec/90000/0]"},
+       .caller_remote =
+           {"1 [111:audio/opus/48000/2;minptime=10;useinbandfec=1]",
+            "1 [63:audio/red/48000/2;=111/111]",
+            "1 [9:audio/G722/8000/1]",
+            "1 [0:audio/PCMU/8000/1]",
+            "1 [8:audio/PCMA/8000/1]",
+            "1 [13:audio/CN/8000/1]",
+            "1 [110:audio/telephone-event/48000/1]",
+            "1 [126:audio/telephone-event/8000/1]",
+            "2 [96:video/VP8/90000/0]",
+            "2 [97:video/rtx/90000/0;apt=96]",
+            "2 "
+            "[103:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-id="
+            "42001f]",
+            "2 [104:video/rtx/90000/0;apt=103]",
+            "2 "
+            "[107:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-id="
+            "42001f]",
+            "2 [108:video/rtx/90000/0;apt=107]",
+            "2 "
+            "[109:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-id="
+            "42e01f]",
+            "2 [114:video/rtx/90000/0;apt=109]",
+            "2 "
+            "[115:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-id="
+            "42e01f]",
+            "2 [116:video/rtx/90000/0;apt=115]",
+            "2 "
+            "[117:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-id="
+            "4d001f]",
+            "2 [118:video/rtx/90000/0;apt=117]",
+            "2 "
+            "[39:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-id="
+            "4d001f]",
+            "2 [40:video/rtx/90000/0;apt=39]",
+            "2 [45:video/AV1/90000/0;level-idx=5;profile=0;tier=0]",
+            "2 [46:video/rtx/90000/0;apt=45]",
+            "2 [98:video/VP9/90000/0;profile-id=0]",
+            "2 [99:video/rtx/90000/0;apt=98]",
+            "2 [100:video/VP9/90000/0;profile-id=2]",
+            "2 [101:video/rtx/90000/0;apt=100]",
+            "2 [119:video/red/90000/0]",
+            "2 [120:video/rtx/90000/0;apt=119]",
+            "2 [121:video/ulpfec/90000/0]"},
+       .callee_local =
+           {"1 [111:audio/opus/48000/2;minptime=10;useinbandfec=1]",
+            "1 [63:audio/red/48000/2;=111/111]",
+            "1 [9:audio/G722/8000/1]",
+            "1 [0:audio/PCMU/8000/1]",
+            "1 [8:audio/PCMA/8000/1]",
+            "1 [13:audio/CN/8000/1]",
+            "1 [110:audio/telephone-event/48000/1]",
+            "1 [126:audio/telephone-event/8000/1]",
+            "2 [96:video/VP8/90000/0]",
+            "2 [97:video/rtx/90000/0;apt=96]",
+            "2 "
+            "[103:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-id="
+            "42001f]",
+            "2 [104:video/rtx/90000/0;apt=103]",
+            "2 "
+            "[107:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-id="
+            "42001f]",
+            "2 [108:video/rtx/90000/0;apt=107]",
+            "2 "
+            "[109:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-id="
+            "42e01f]",
+            "2 [114:video/rtx/90000/0;apt=109]",
+            "2 "
+            "[115:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-id="
+            "42e01f]",
+            "2 [116:video/rtx/90000/0;apt=115]",
+            "2 "
+            "[117:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-id="
+            "4d001f]",
+            "2 [118:video/rtx/90000/0;apt=117]",
+            "2 "
+            "[39:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-id="
+            "4d001f]",
+            "2 [40:video/rtx/90000/0;apt=39]",
+            "2 [45:video/AV1/90000/0;level-idx=5;profile=0;tier=0]",
+            "2 [46:video/rtx/90000/0;apt=45]",
+            "2 [98:video/VP9/90000/0;profile-id=0]",
+            "2 [99:video/rtx/90000/0;apt=98]",
+            "2 [100:video/VP9/90000/0;profile-id=2]",
+            "2 [101:video/rtx/90000/0;apt=100]",
+            "2 [119:video/red/90000/0]",
+            "2 [120:video/rtx/90000/0;apt=119]",
+            "2 [121:video/ulpfec/90000/0]"},
+       .callee_remote =
+           {"1 [111:audio/opus/48000/2;minptime=10;useinbandfec=1]",
+            "1 [63:audio/red/48000/2;=111/111]",
+            "1 [9:audio/G722/8000/1]",
+            "1 [0:audio/PCMU/8000/1]",
+            "1 [8:audio/PCMA/8000/1]",
+            "1 [13:audio/CN/8000/1]",
+            "1 [110:audio/telephone-event/48000/1]",
+            "1 [126:audio/telephone-event/8000/1]",
+            "2 [96:video/VP8/90000/0]",
+            "2 [97:video/rtx/90000/0;apt=96]",
+            "2 "
+            "[103:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-id="
+            "42001f]",
+            "2 [104:video/rtx/90000/0;apt=103]",
+            "2 "
+            "[107:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-id="
+            "42001f]",
+            "2 [108:video/rtx/90000/0;apt=107]",
+            "2 "
+            "[109:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-id="
+            "42e01f]",
+            "2 [114:video/rtx/90000/0;apt=109]",
+            "2 "
+            "[115:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-id="
+            "42e01f]",
+            "2 [116:video/rtx/90000/0;apt=115]",
+            "2 "
+            "[117:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-id="
+            "4d001f]",
+            "2 [118:video/rtx/90000/0;apt=117]",
+            "2 "
+            "[39:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-id="
+            "4d001f]",
+            "2 [40:video/rtx/90000/0;apt=39]",
+            "2 [45:video/AV1/90000/0;level-idx=5;profile=0;tier=0]",
+            "2 [46:video/rtx/90000/0;apt=45]",
+            "2 [98:video/VP9/90000/0;profile-id=0]",
+            "2 [99:video/rtx/90000/0;apt=98]",
+            "2 [100:video/VP9/90000/0;profile-id=2]",
+            "2 [101:video/rtx/90000/0;apt=100]",
+            "2 [119:video/red/90000/0]",
+            "2 [120:video/rtx/90000/0;apt=119]",
+            "2 [121:video/ulpfec/90000/0]"}},
+
+      {.factory_id = FactorySignature::Id::kWebRtcMoreConfigs1,
+       .caller_local =
+           {"1 [111:audio/opus/48000/2;minptime=10;useinbandfec=1]",
+            "1 [63:audio/red/48000/2;=111/111]", "1 [9:audio/G722/8000/1]",
+            "1 [0:audio/PCMU/8000/1]", "1 [8:audio/PCMA/8000/1]",
+            "1 [13:audio/CN/8000/1]", "1 [110:audio/telephone-event/48000/1]",
+            "1 [126:audio/telephone-event/8000/1]", "2 [96:video/VP8/90000/0]",
+            "2 [97:video/rtx/90000/0;apt=96]",
+            "2 [39:video/AV1/90000/0;level-idx=5;profile=0;tier=0]",
+            "2 [40:video/rtx/90000/0;apt=39]",
+            "2 [98:video/VP9/90000/0;profile-id=0]",
+            "2 [99:video/rtx/90000/0;apt=98]",
+            "2 [100:video/VP9/90000/0;profile-id=2]",
+            "2 [101:video/rtx/90000/0;apt=100]", "2 [103:video/red/90000/0]",
+            "2 [104:video/rtx/90000/0;apt=103]",
+            "2 [107:video/ulpfec/90000/0]"},
+       .caller_remote =
+           {"1 [111:audio/opus/48000/2;minptime=10;useinbandfec=1]",
+            "1 [63:audio/red/48000/2;=111/111]", "1 [9:audio/G722/8000/1]",
+            "1 [0:audio/PCMU/8000/1]", "1 [8:audio/PCMA/8000/1]",
+            "1 [13:audio/CN/8000/1]", "1 [110:audio/telephone-event/48000/1]",
+            "1 [126:audio/telephone-event/8000/1]", "2 [96:video/VP8/90000/0]",
+            "2 [97:video/rtx/90000/0;apt=96]",
+            "2 [39:video/AV1/90000/0;level-idx=5;profile=0;tier=0]",
+            "2 [40:video/rtx/90000/0;apt=39]",
+            "2 [98:video/VP9/90000/0;profile-id=0]",
+            "2 [99:video/rtx/90000/0;apt=98]",
+            "2 [100:video/VP9/90000/0;profile-id=2]",
+            "2 [101:video/rtx/90000/0;apt=100]", "2 [103:video/red/90000/0]",
+            "2 [104:video/rtx/90000/0;apt=103]",
+            "2 [107:video/ulpfec/90000/0]"},
+       .callee_local =
+           {"1 [111:audio/opus/48000/2;minptime=10;useinbandfec=1]",
+            "1 [63:audio/red/48000/2;=111/111]", "1 [9:audio/G722/8000/1]",
+            "1 [0:audio/PCMU/8000/1]", "1 [8:audio/PCMA/8000/1]",
+            "1 [13:audio/CN/8000/1]", "1 [110:audio/telephone-event/48000/1]",
+            "1 [126:audio/telephone-event/8000/1]", "2 [96:video/VP8/90000/0]",
+            "2 [97:video/rtx/90000/0;apt=96]",
+            "2 [39:video/AV1/90000/0;level-idx=5;profile=0;tier=0]",
+            "2 [40:video/rtx/90000/0;apt=39]",
+            "2 [98:video/VP9/90000/0;profile-id=0]",
+            "2 [99:video/rtx/90000/0;apt=98]",
+            "2 [100:video/VP9/90000/0;profile-id=2]",
+            "2 [101:video/rtx/90000/0;apt=100]", "2 [103:video/red/90000/0]",
+            "2 [104:video/rtx/90000/0;apt=103]",
+            "2 [107:video/ulpfec/90000/0]"},
+       .callee_remote =
+           {"1 [111:audio/opus/48000/2;minptime=10;useinbandfec=1]",
+            "1 [63:audio/red/48000/2;=111/111]", "1 [9:audio/G722/8000/1]",
+            "1 [0:audio/PCMU/8000/1]", "1 [8:audio/PCMA/8000/1]",
+            "1 [13:audio/CN/8000/1]", "1 [110:audio/telephone-event/48000/1]",
+            "1 [126:audio/telephone-event/8000/1]", "2 [96:video/VP8/90000/0]",
+            "2 [97:video/rtx/90000/0;apt=96]",
+            "2 [39:video/AV1/90000/0;level-idx=5;profile=0;tier=0]",
+            "2 [40:video/rtx/90000/0;apt=39]",
+            "2 [98:video/VP9/90000/0;profile-id=0]",
+            "2 [99:video/rtx/90000/0;apt=98]",
+            "2 [100:video/VP9/90000/0;profile-id=2]",
+            "2 [101:video/rtx/90000/0;apt=100]", "2 [103:video/red/90000/0]",
+            "2 [104:video/rtx/90000/0;apt=103]",
+            "2 [107:video/ulpfec/90000/0]"}},
+      {.factory_id = FactorySignature::Id::kWebRtcAndroid,
+       .caller_local =
+           {"1 [111:audio/opus/48000/2;minptime=10;useinbandfec=1]",
+            "1 [63:audio/red/48000/2;=111/111]", "1 [9:audio/G722/8000/1]",
+            "1 [0:audio/PCMU/8000/1]", "1 [8:audio/PCMA/8000/1]",
+            "1 [13:audio/CN/8000/1]", "1 [110:audio/telephone-event/48000/1]",
+            "1 [126:audio/telephone-event/8000/1]", "2 [96:video/VP8/90000/0]",
+            "2 [97:video/rtx/90000/0;apt=96]",
+            "2 [39:video/AV1/90000/0;level-idx=5;profile=0;tier=0]",
+            "2 [40:video/rtx/90000/0;apt=39]",
+            "2 [98:video/VP9/90000/0;profile-id=0]",
+            "2 [99:video/rtx/90000/0;apt=98]", "2 [100:video/red/90000/0]",
+            "2 [101:video/rtx/90000/0;apt=100]",
+            "2 [103:video/ulpfec/90000/0]"},
+       .caller_remote =
+           {"1 [111:audio/opus/48000/2;minptime=10;useinbandfec=1]",
+            "1 [63:audio/red/48000/2;=111/111]", "1 [9:audio/G722/8000/1]",
+            "1 [0:audio/PCMU/8000/1]", "1 [8:audio/PCMA/8000/1]",
+            "1 [13:audio/CN/8000/1]", "1 [110:audio/telephone-event/48000/1]",
+            "1 [126:audio/telephone-event/8000/1]", "2 [96:video/VP8/90000/0]",
+            "2 [97:video/rtx/90000/0;apt=96]",
+            "2 [39:video/AV1/90000/0;level-idx=5;profile=0;tier=0]",
+            "2 [40:video/rtx/90000/0;apt=39]",
+            "2 [98:video/VP9/90000/0;profile-id=0]",
+            "2 [99:video/rtx/90000/0;apt=98]", "2 [100:video/red/90000/0]",
+            "2 [101:video/rtx/90000/0;apt=100]",
+            "2 [103:video/ulpfec/90000/0]"},
+       .callee_local =
+           {"1 [111:audio/opus/48000/2;minptime=10;useinbandfec=1]",
+            "1 [63:audio/red/48000/2;=111/111]", "1 [9:audio/G722/8000/1]",
+            "1 [0:audio/PCMU/8000/1]", "1 [8:audio/PCMA/8000/1]",
+            "1 [13:audio/CN/8000/1]", "1 [110:audio/telephone-event/48000/1]",
+            "1 [126:audio/telephone-event/8000/1]", "2 [96:video/VP8/90000/0]",
+            "2 [97:video/rtx/90000/0;apt=96]",
+            "2 [39:video/AV1/90000/0;level-idx=5;profile=0;tier=0]",
+            "2 [40:video/rtx/90000/0;apt=39]",
+            "2 [98:video/VP9/90000/0;profile-id=0]",
+            "2 [99:video/rtx/90000/0;apt=98]", "2 [100:video/red/90000/0]",
+            "2 [101:video/rtx/90000/0;apt=100]",
+            "2 [103:video/ulpfec/90000/0]"},
+       .callee_remote =
+           {"1 [111:audio/opus/48000/2;minptime=10;useinbandfec=1]",
+            "1 [63:audio/red/48000/2;=111/111]", "1 [9:audio/G722/8000/1]",
+            "1 [0:audio/PCMU/8000/1]", "1 [8:audio/PCMA/8000/1]",
+            "1 [13:audio/CN/8000/1]", "1 [110:audio/telephone-event/48000/1]",
+            "1 [126:audio/telephone-event/8000/1]", "2 [96:video/VP8/90000/0]",
+            "2 [97:video/rtx/90000/0;apt=96]",
+            "2 [39:video/AV1/90000/0;level-idx=5;profile=0;tier=0]",
+            "2 [40:video/rtx/90000/0;apt=39]",
+            "2 [98:video/VP9/90000/0;profile-id=0]",
+            "2 [99:video/rtx/90000/0;apt=98]", "2 [100:video/red/90000/0]",
+            "2 [101:video/rtx/90000/0;apt=100]",
+            "2 [103:video/ulpfec/90000/0]"}},
+      {.factory_id = FactorySignature::Id::kGoogleInternal,
+       .caller_local =
+           {"1 [111:audio/opus/48000/2;minptime=10;useinbandfec=1]",
+            "1 [63:audio/red/48000/2;=111/111]",
+            "1 [9:audio/G722/8000/1]",
+            "1 [0:audio/PCMU/8000/1]",
+            "1 [8:audio/PCMA/8000/1]",
+            "1 [13:audio/CN/8000/1]",
+            "1 [110:audio/telephone-event/48000/1]",
+            "1 [126:audio/telephone-event/8000/1]",
+            "2 [96:video/VP8/90000/0]",
+            "2 [97:video/rtx/90000/0;apt=96]",
+            "2 "
+            "[100:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-"
+            "id=42001f]",
+            "2 [101:video/rtx/90000/0;apt=100]",
+            "2 "
+            "[103:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-"
+            "id=42001f]",
+            "2 [104:video/rtx/90000/0;apt=103]",
+            "2 "
+            "[107:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-"
+            "id=42e01f]",
+            "2 [108:video/rtx/90000/0;apt=107]",
+            "2 "
+            "[109:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-"
+            "id=42e01f]",
+            "2 [114:video/rtx/90000/0;apt=109]",
+            "2 "
+            "[115:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-"
+            "id=4d001f]",
+            "2 [116:video/rtx/90000/0;apt=115]",
+            "2 "
+            "[39:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-"
+            "id=4d001f]",
+            "2 [40:video/rtx/90000/0;apt=39]",
+            "2 [98:video/VP9/90000/0;profile-id=0]",
+            "2 [99:video/rtx/90000/0;apt=98]",
+            "2 [117:video/red/90000/0]",
+            "2 [118:video/rtx/90000/0;apt=117]",
+            "2 [119:video/ulpfec/90000/0]"},
+       .caller_remote =
+           {"1 [111:audio/opus/48000/2;minptime=10;useinbandfec=1]",
+            "1 [63:audio/red/48000/2;=111/111]",
+            "1 [9:audio/G722/8000/1]",
+            "1 [0:audio/PCMU/8000/1]",
+            "1 [8:audio/PCMA/8000/1]",
+            "1 [13:audio/CN/8000/1]",
+            "1 [110:audio/telephone-event/48000/1]",
+            "1 [126:audio/telephone-event/8000/1]",
+            "2 [96:video/VP8/90000/0]",
+            "2 [97:video/rtx/90000/0;apt=96]",
+            "2 "
+            "[100:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-"
+            "id=42001f]",
+            "2 [101:video/rtx/90000/0;apt=100]",
+            "2 "
+            "[103:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-"
+            "id=42001f]",
+            "2 [104:video/rtx/90000/0;apt=103]",
+            "2 "
+            "[107:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-"
+            "id=42e01f]",
+            "2 [108:video/rtx/90000/0;apt=107]",
+            "2 "
+            "[109:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-"
+            "id=42e01f]",
+            "2 [114:video/rtx/90000/0;apt=109]",
+            "2 "
+            "[115:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-"
+            "id=4d001f]",
+            "2 [116:video/rtx/90000/0;apt=115]",
+            "2 "
+            "[39:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-"
+            "id=4d001f]",
+            "2 [40:video/rtx/90000/0;apt=39]",
+            "2 [98:video/VP9/90000/0;profile-id=0]",
+            "2 [99:video/rtx/90000/0;apt=98]",
+            "2 [117:video/red/90000/0]",
+            "2 [118:video/rtx/90000/0;apt=117]",
+            "2 [119:video/ulpfec/90000/0]"},
+       .callee_local =
+           {"1 [111:audio/opus/48000/2;minptime=10;useinbandfec=1]",
+            "1 [63:audio/red/48000/2;=111/111]",
+            "1 [9:audio/G722/8000/1]",
+            "1 [0:audio/PCMU/8000/1]",
+            "1 [8:audio/PCMA/8000/1]",
+            "1 [13:audio/CN/8000/1]",
+            "1 [110:audio/telephone-event/48000/1]",
+            "1 [126:audio/telephone-event/8000/1]",
+            "2 [96:video/VP8/90000/0]",
+            "2 [97:video/rtx/90000/0;apt=96]",
+            "2 "
+            "[100:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-"
+            "id=42001f]",
+            "2 [101:video/rtx/90000/0;apt=100]",
+            "2 "
+            "[103:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-"
+            "id=42001f]",
+            "2 [104:video/rtx/90000/0;apt=103]",
+            "2 "
+            "[107:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-"
+            "id=42e01f]",
+            "2 [108:video/rtx/90000/0;apt=107]",
+            "2 "
+            "[109:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-"
+            "id=42e01f]",
+            "2 [114:video/rtx/90000/0;apt=109]",
+            "2 "
+            "[115:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-"
+            "id=4d001f]",
+            "2 [116:video/rtx/90000/0;apt=115]",
+            "2 "
+            "[39:video/H264/90000/"
+            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-"
+            "id=4d001f]",
+            "2 [40:video/rtx/90000/0;apt=39]",
+            "2 [98:video/VP9/90000/0;profile-id=0]",
+            "2 [99:video/rtx/90000/0;apt=98]",
+            "2 [117:video/red/90000/0]",
+            "2 [118:video/rtx/90000/0;apt=117]",
+            "2 [119:video/ulpfec/90000/0]"},
+       .callee_remote = {
+           "1 [111:audio/opus/48000/2;minptime=10;useinbandfec=1]",
+           "1 [63:audio/red/48000/2;=111/111]",
+           "1 [9:audio/G722/8000/1]",
+           "1 [0:audio/PCMU/8000/1]",
+           "1 [8:audio/PCMA/8000/1]",
+           "1 [13:audio/CN/8000/1]",
+           "1 [110:audio/telephone-event/48000/1]",
+           "1 [126:audio/telephone-event/8000/1]",
+           "2 [96:video/VP8/90000/0]",
+           "2 [97:video/rtx/90000/0;apt=96]",
+           "2 "
+           "[100:video/H264/90000/"
+           "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-id="
+           "42001f]",
+           "2 [101:video/rtx/90000/0;apt=100]",
+           "2 "
+           "[103:video/H264/90000/"
+           "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-id="
+           "42001f]",
+           "2 [104:video/rtx/90000/0;apt=103]",
+           "2 "
+           "[107:video/H264/90000/"
+           "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-id="
+           "42e01f]",
+           "2 [108:video/rtx/90000/0;apt=107]",
+           "2 "
+           "[109:video/H264/90000/"
+           "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-id="
+           "42e01f]",
+           "2 [114:video/rtx/90000/0;apt=109]",
+           "2 "
+           "[115:video/H264/90000/"
+           "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-id="
+           "4d001f]",
+           "2 [116:video/rtx/90000/0;apt=115]",
+           "2 "
+           "[39:video/H264/90000/"
+           "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-id="
+           "4d001f]",
+           "2 [40:video/rtx/90000/0;apt=39]",
+           "2 [98:video/VP9/90000/0;profile-id=0]",
+           "2 [99:video/rtx/90000/0;apt=98]",
+           "2 [117:video/red/90000/0]",
+           "2 [118:video/rtx/90000/0;apt=117]",
+           "2 [119:video/ulpfec/90000/0]"}}};
+  auto this_golden_it =
+      std::find_if(golden_answers.begin(), golden_answers.end(),
+                   [&](const ResultingCodecList& candidate) {
+                     return candidate.factory_id == factory_signature.id();
+                   });
+  ASSERT_THAT(this_golden_it, Not(Eq(golden_answers.end())))
+      << "Add this result set to golden_answers:\n"
+      << DumpAsResultingCodecList(
+             factory_signature.id(),
+             CodecList(*caller()->pc()->local_description()),
+             CodecList(*caller()->pc()->remote_description()),
+             CodecList(*callee()->pc()->local_description()),
+             CodecList(*callee()->pc()->remote_description()));
+
+  const ResultingCodecList& this_golden = *this_golden_it;
+  EXPECT_THAT(CodecList(*caller()->pc()->local_description()),
+              ElementsAreArray(this_golden.caller_local));
+  EXPECT_THAT(CodecList(*caller()->pc()->remote_description()),
+              ElementsAreArray(this_golden.caller_remote));
+  EXPECT_THAT(CodecList(*callee()->pc()->local_description()),
+              ElementsAreArray(this_golden.callee_local));
+  EXPECT_THAT(CodecList(*callee()->pc()->remote_description()),
+              ElementsAreArray(this_golden.callee_remote));
+}
+
 }  // namespace
 }  // namespace webrtc

From db488784994e64c61c52b8a003b7f2d460bec8c1 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Wed, 11 Mar 2026 05:04:09 -0700
Subject: [PATCH 261/472] Roll chromium_revision a5382cea3d..b937d39273
 (1597531:1597657)

Change log: https://chromium.googlesource.com/chromium/src/+log/a5382cea3d..b937d39273
Full diff: https://chromium.googlesource.com/chromium/src/+/a5382cea3d..b937d39273

Changed dependencies
* src/buildtools/linux64: git_revision:d8c2f07d653520568da7cace755a87dad241b72d..git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9
* src/buildtools/mac: git_revision:d8c2f07d653520568da7cace755a87dad241b72d..git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9
* src/buildtools/win: git_revision:d8c2f07d653520568da7cace755a87dad241b72d..git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/76aa1c78ed..94f693fcb6
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/256d17dbe5..a133c94b51
* src/third_party/androidx/cipd: uw8FaVpFdPaFx2Y4c1mBUMRwGB52J-PturMUyulx_hkC..Hi79Lu6qQ-Fbcjwgf_f-NEYJweP4QdZqjJ0lXwc0PpcC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/476f483518..eedf10c1f4
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/50957c5c47..260c6fd37c
* src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/8df8a70a83..044963a253
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/0510739a11..9765576870
DEPS diff: https://chromium.googlesource.com/chromium/src/+/a5382cea3d..b937d39273/DEPS

BUG=None

Change-Id: Ifcf8deb472e0d2cf1d96b4446f007b2c84f36ec8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/455781
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47114}
---
 DEPS | 22 +++++++++++-----------
 1 file changed, 11 insertions(+), 11 deletions(-)

diff --git a/DEPS b/DEPS
index bcb8c09b6e..dffcecaef3 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': 'a5382cea3d06856d6689fa29b66bc31dda86ca70',
+  'chromium_revision': 'b937d39273b52034e8542dea4b040587be0a127f',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -78,19 +78,19 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@76aa1c78ed7a95f9894283eaa96c0bc34ffa5be4',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@94f693fcb6dde967f11668a9e21eec1a4ec8b49c',
     'condition': 'checkout_ios',
   },
   'src/testing':
     'https://chromium.googlesource.com/chromium/src/testing@e7a1e88ecc35a70f3d198674042753ec9e8e3650',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@256d17dbe537bf8fdd8937e1858063b9aa4f3e4f',
+    'https://chromium.googlesource.com/chromium/src/third_party@a133c94b51ac034d5f375cc1746e66dcdee57282',
 
   'src/buildtools/linux64': {
     'packages': [
       {
         'package': 'gn/gn/linux-${{arch}}',
-        'version': 'git_revision:d8c2f07d653520568da7cace755a87dad241b72d',
+        'version': 'git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9',
       }
     ],
     'dep_type': 'cipd',
@@ -100,7 +100,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/mac-${{arch}}',
-        'version': 'git_revision:d8c2f07d653520568da7cace755a87dad241b72d',
+        'version': 'git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9',
       }
     ],
     'dep_type': 'cipd',
@@ -110,7 +110,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/windows-amd64',
-        'version': 'git_revision:d8c2f07d653520568da7cace755a87dad241b72d',
+        'version': 'git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9',
       }
     ],
     'dep_type': 'cipd',
@@ -337,7 +337,7 @@ deps = {
   'src/third_party/libc++abi/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@8f11bb1d4438d0239d0dfc1bd9456a9f31629dda',
   'src/third_party/llvm-libc/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@8df8a70a8349a709b8996a11279eb226cae5aced',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@044963a253be3b8b2a63a1b2e0f3e361b01bc186',
   'src/third_party/libunwind/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@db838d918570d4e381ecf9f5cc70a0098c9c2cd6',
 
@@ -488,7 +488,7 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@476f4835186050a44f7fe6f088abbd279b634194',
+    'https://boringssl.googlesource.com/boringssl.git@eedf10c1f4ab44629e2fe3fe6f07e2fb0897c0f4',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@8be0e3114685fcc1589561067282edf75ea1259a',
   'src/third_party/catapult':
@@ -511,7 +511,7 @@ deps = {
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@50957c5c47de1c85637fc9dafe6d4eb4c1224f6b',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@260c6fd37c502e984836a446c5d90e4a1298036d',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@0510739a115e1aed8de0dfb2abc59088219be0b2',
+    'https://chromium.googlesource.com/chromium/src/tools@9765576870401b230a826e0e4f7a12851575306e',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'uw8FaVpFdPaFx2Y4c1mBUMRwGB52J-PturMUyulx_hkC',
+          'version': 'Hi79Lu6qQ-Fbcjwgf_f-NEYJweP4QdZqjJ0lXwc0PpcC',
       },
     ],
     'condition': 'checkout_android and non_git_source',

From e4f9241450a411e56dc4a56c400defe7ad19a1c0 Mon Sep 17 00:00:00 2001
From: Ilya Nikolaevskiy <ilnik@webrtc.org>
Date: Wed, 11 Mar 2026 06:24:57 -0700
Subject: [PATCH 262/472] Revert "Fix payload type allocation issues for Audio
 RED and MID recycling."
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This reverts commit 9e3381393a233cac896085fc3f26bdac83caaaef.

Reason for revert: Breaks webrtc-importer

Original change's description:
> Fix payload type allocation issues for Audio RED and MID recycling.
>
> This CL addresses several issues identified while enabling the
> WebRTC-PayloadTypesInTransport field trial:
> - Enforce media type equality in codec matching to prevent Audio/Video
> RED collisions.
> - Relax RED matching rules to allow negotiation with unlinked RED codecs.
> - Validate media type during MID recycling in CodecVendor to prevent
> incorrect codec merging.
> - Update documentation with the video implementation strategy.
>
> Bug: webrtc:360058654
> Change-Id: I1ecbdad2179d346b3682875d823889aa076e353e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453001
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#47113}

Bug: webrtc:360058654
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I5f7c642e936c1b06781b78f10293837d1262d450
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/455820
Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47115}
---
 g3doc/todo/payload_type_redesign.md           | 127 ----
 media/base/codec_comparators.cc               |  51 +-
 media/base/codec_comparators_unittest.cc      |  57 --
 pc/BUILD.gn                                   |   1 -
 pc/codec_vendor.cc                            |   7 +-
 ...er_connection_encodings_integrationtest.cc |  58 --
 pc/peer_connection_media_unittest.cc          |   4 +-
 ...er_connection_stability_integrationtest.cc | 549 ------------------
 8 files changed, 21 insertions(+), 833 deletions(-)
 delete mode 100644 g3doc/todo/payload_type_redesign.md

diff --git a/g3doc/todo/payload_type_redesign.md b/g3doc/todo/payload_type_redesign.md
deleted file mode 100644
index 5fa5f12bca..0000000000
--- a/g3doc/todo/payload_type_redesign.md
+++ /dev/null
@@ -1,127 +0,0 @@
-# Payload type allocation redesign
-
-## Background: What a payload type is
-
-The payload type is a property of a codec that is established between a sender
-and a receiver using SDP offer/answer.
-
-Each end of the connection independently indicates the list of codecs it is
-willing to send and/or receive, and assigns a payload type to each. It is
-conventional but not required to use the same payload types in an answer as
-those suggested in an offer.
-
-For SDP sendonly media sections, it indicates a willingness to send; for
-recvonly media sections, it indicates a willingness to receive; for sendrecv
-media sections, it indicates a willingness to receive AND a willingness to send.
-Presence in a media section does not guarantee that it will be used. (RFC 3264)
-
-## Current allocation strategy
-
-The payload types are assigned by the VideoEngine and VoiceEngine according to a
-fixed list. If the assignments collide with already established payload types,
-the assignments are munged in the CodecVendor class so that there are no
-colliding payload types.
-
-The PayloadTypeRecorder class records what payload types are assigned - there is
-one case (PR-answer followed by a later Answer that has different PTs) where
-they are changed, but otherwise, once the PayloadTypeRecorder has assigned them,
-that PT is permanent for that transport and that direction.
-
-Picking a payload type is done in the PayloadTypePicker class. This is preloaded
-with some assignments, so that if a PT is free, the commonly used PT for that
-codec can be used.
-
-## Desired future strategy
-
-There should be no assignment of payload types in VideoEngine and VoiceEngine.
-Payload types should be picked only when creating an offer or answer; they are
-assigned permanently in SetLocalDescription / SetRemoteDescription.
-
-Once an assignment is made, it should never need to change (apart from the case
-noted above).
-
-## Dealing with resiliency mechanisms
-
-Resiliency mechanisms (RED, RTX) are signaled using "codecs", and are
-traditionally represented as such within the library. They associate with the
-codec they are resiliency for by using the "apt=" parameter in their a=fmtp
-lines. However, this complicates assignment, since that parameter cannot be
-filled in until the payload type for the "protected" codec is assigned.
-
-## Backwards compatibility issues
-
-Experience has shown that changing the PT allocation strategy often trips up
-applications that have depended on the result of the old allocation strategy, so
-user-visible changes should be avoided. One example is that it's preferred (but
-not strictly required) to have the RTX PT assigned the PT number of 1 more than
-the PT of the codec it refers to.
-
-## Current implementation status
-
-The new strategy is implemented for audio codecs. Several issues that caused
-test failures when enabling the `WebRTC-PayloadTypesInTransport` field trial
-have been identified and fixed:
-
-- **Audio/Video RED Collision:** RED codecs of different media types were
-  incorrectly matching, leading to payload type conflicts.
-  `MatchesWithCodecRules` now enforces media type equality.
-- **MID Recycling:** When a MID was recycled for a different media type (e.g.,
-  Audio -> Video), `CodecVendor` was incorrectly merging codecs from the old
-  description. This has been fixed by validating the media type before merging.
-- **RED Matching Logic:** Relaxed the matching rules for RED to allow
-  negotiation to proceed even when parameters (linking RED to primary codecs)
-  are not yet populated, as this linking now happens late in the `CodecVendor`.
-
-The new strategy is not yet implemented for video codecs.
-
-## Implementation Strategy for Video
-
-The goal is to transition video codec handling to the same late-assignment model
-used for audio.
-
-### 1. Unified Codec Collection
-
-- Implement `VideoCodecsFromFactory` in `pc/typed_codec_vendor.cc`. This will
-  query the `VideoEncoderFactory` and `VideoDecoderFactory` to gather supported
-  `SdpVideoFormat`s.
-- These codecs will be initialized with `kIdNotSet`, allowing
-  `SdpPayloadTypeSuggester` to assign payload types only during the creation of
-  an offer or answer.
-
-### 2. Payload Type Suggester Integration
-
-- Ensure `PayloadTypePicker` has a complete list of preferred payload types for
-  video codecs (VP8, VP9, H.264, AV1, etc.) to maintain stable and conventional
-  assignments.
-- Update `CodecVendor` to use the unassigned video codec list when the
-  `WebRTC-PayloadTypesInTransport` trial is active.
-
-### 3. Resiliency and Parameter Linking
-
-- Refine `MergeCodecs` and `AssignCodecIdsAndLinkRed` to handle video-specific
-  resiliency:
-  - **RTX:** Correctly link RTX codecs to their primary video codecs by matching
-    names and specific parameters (like `profile-level-id` for H.264).
-  - **RED/ULPFEC:** Ensure parameters for video redundancy are correctly
-    populated once the primary codec PTs are assigned.
-
-### 4. Verification and Testing
-
-- **Integration Tests:** Enable the `WebRTC-PayloadTypesInTransport` trial in
-  `peerconnection_unittests` and `rtc_unittests` to identify any video-specific
-  regressions.
-- **Stable PT Tests:** Add coverage to ensure that payload types remain stable
-  across renegotiations, even when the order of codecs in the transceiver
-  preferences changes.
-- **MID Recycling:** Verify that MID recycling for video-to-audio and vice-versa
-  works correctly without PT collisions or crashes.
-
-## Desired next steps
-
-Analyze the current situation. (Done)
-
-### Test, isolate and fix failures
-
-The identified failures in `PeerConnectionEncodingsIntegrationTest` and
-`PeerConnectionIntegrationTest` have been resolved. Next, focus on implementing
-the video strategy outlined above.
diff --git a/media/base/codec_comparators.cc b/media/base/codec_comparators.cc
index cb67f671a6..9da9d4aca5 100644
--- a/media/base/codec_comparators.cc
+++ b/media/base/codec_comparators.cc
@@ -163,7 +163,15 @@ bool MatchesWithReferenceAttributesAndComparator(
         potential_match.params.find(kCodecParamNotInNameValueFormat);
     bool has_parameters_1 = red_parameters_1 != codec_to_match.params.end();
     bool has_parameters_2 = red_parameters_2 != potential_match.params.end();
-
+    // If codec_to_match has unassigned PT and no parameter,
+    // we assume that it'll be assigned later and return a match.
+    // Note - this should be deleted. It's untidy.
+    if (potential_match.id == Codec::kIdNotSet && !has_parameters_2) {
+      return true;
+    }
+    if (codec_to_match.id == Codec::kIdNotSet && !has_parameters_1) {
+      return true;
+    }
     if (has_parameters_1 && has_parameters_2) {
       // Different levels of redundancy between offer and answer are OK
       // since RED is considered to be declarative.
@@ -195,16 +203,7 @@ bool MatchesWithReferenceAttributesAndComparator(
       return true;
     }
     if (!has_parameters_1 && !has_parameters_2) {
-      return true;
-    }
-    // Exactly one lacks parameters.
-    // Allow match if it is an audio RED codec and at least one of the
-    // codecs has an unassigned payload type or they have the same ID.
-    if (codec_to_match.type == Codec::Type::kAudio &&
-        codec_to_match.name == kRedCodecName &&
-        (codec_to_match.id == Codec::kIdNotSet ||
-         potential_match.id == Codec::kIdNotSet ||
-         codec_to_match.id == potential_match.id)) {
+      // Both parameters are missing. Happens for video RED.
       return true;
     }
     return false;
@@ -296,11 +295,6 @@ bool MatchesWithCodecRules(const Codec& left_codec, const Codec& right_codec) {
        right_codec.id <= kLowerDynamicRangeMax) ||
       (right_codec.id >= kUpperDynamicRangeMin &&
        right_codec.id <= kUpperDynamicRangeMax);
-
-  if (left_codec.type != right_codec.type) {
-    return false;
-  }
-
   bool matches_id;
   if ((is_id_in_dynamic_range && is_codec_id_in_dynamic_range) ||
       left_codec.id == Codec::kIdNotSet || right_codec.id == Codec::kIdNotSet) {
@@ -366,22 +360,12 @@ std::optional<Codec> FindMatchingCodec(const std::vector<Codec>& codecs1,
 bool IsSameRtpCodec(const Codec& codec, const RtpCodec& rtp_codec) {
   RtpCodecParameters rtp_codec2 = codec.ToCodecParameters();
 
-  if (!absl::EqualsIgnoreCase(rtp_codec.name, rtp_codec2.name) ||
-      rtp_codec.kind != rtp_codec2.kind ||
-      rtp_codec.num_channels != rtp_codec2.num_channels ||
-      rtp_codec.clock_rate != rtp_codec2.clock_rate) {
-    return false;
-  }
-
-  // audio/RED should ignore the parameters which specify payload types so
-  // can not be compared.
-  if (rtp_codec.kind == MediaType::AUDIO &&
-      absl::EqualsIgnoreCase(rtp_codec.name, kRedCodecName)) {
-    return true;
-  }
-
-  return InsertDefaultParams(rtp_codec.name, rtp_codec.parameters) ==
-         InsertDefaultParams(rtp_codec2.name, rtp_codec2.parameters);
+  return absl::EqualsIgnoreCase(rtp_codec.name, rtp_codec2.name) &&
+         rtp_codec.kind == rtp_codec2.kind &&
+         rtp_codec.num_channels == rtp_codec2.num_channels &&
+         rtp_codec.clock_rate == rtp_codec2.clock_rate &&
+         InsertDefaultParams(rtp_codec.name, rtp_codec.parameters) ==
+             InsertDefaultParams(rtp_codec2.name, rtp_codec2.parameters);
 }
 
 bool IsSameRtpCodecIgnoringLevel(const Codec& codec,
@@ -409,8 +393,7 @@ bool IsSameRtpCodecIgnoringLevel(const Codec& codec,
   }
   // audio/RED should ignore the parameters which specify payload types so
   // can not be compared.
-  if (rtp_codec.kind == MediaType::AUDIO &&
-      absl::EqualsIgnoreCase(rtp_codec.name, kRedCodecName)) {
+  if (rtp_codec.kind == MediaType::AUDIO && rtp_codec.name == kRedCodecName) {
     return true;
   }
 
diff --git a/media/base/codec_comparators_unittest.cc b/media/base/codec_comparators_unittest.cc
index 048765670e..2f78042c16 100644
--- a/media/base/codec_comparators_unittest.cc
+++ b/media/base/codec_comparators_unittest.cc
@@ -105,63 +105,6 @@ TEST(CodecComparatorsTest, MatchesWithReferenceAttributesRed) {
   EXPECT_FALSE(MatchesWithReferenceAttributes(codec_1, codec_5));
 }
 
-TEST(CodecComparatorsTest, RedParametersMismatch) {
-  Codec with_params = CreateAudioCodec(101, kRedCodecName, 48000, 2);
-  with_params.SetParam(kCodecParamNotInNameValueFormat, "111/111");
-
-  Codec no_params = CreateAudioCodec(102, kRedCodecName, 48000, 2);
-
-  // Case 1: Exactly one lacks parameters, both have IDs -> SHOULD NOT match for
-  // audio RED.
-  EXPECT_FALSE(MatchesWithReferenceAttributes(with_params, no_params));
-  EXPECT_FALSE(MatchesWithReferenceAttributes(no_params, with_params));
-
-  // Case 2: Exactly one lacks parameters, the one WITHOUT parameters is
-  // unassigned -> SHOULD match.
-  // This is the case with late assignment.
-  Codec no_params_unassigned = no_params;
-  no_params_unassigned.id = Codec::kIdNotSet;
-  EXPECT_TRUE(
-      MatchesWithReferenceAttributes(with_params, no_params_unassigned));
-  EXPECT_TRUE(
-      MatchesWithReferenceAttributes(no_params_unassigned, with_params));
-
-  // Case 3: Exactly one lacks parameters, the one WITH parameters is
-  // unassigned -> SHOULD match for audio RED because one side is unassigned.
-  Codec with_params_unassigned = with_params;
-  with_params_unassigned.id = Codec::kIdNotSet;
-  EXPECT_TRUE(
-      MatchesWithReferenceAttributes(with_params_unassigned, no_params));
-  EXPECT_TRUE(
-      MatchesWithReferenceAttributes(no_params, with_params_unassigned));
-
-  // Case 4: Exactly one lacks parameters, both are unassigned -> SHOULD match.
-  EXPECT_TRUE(MatchesWithReferenceAttributes(with_params_unassigned,
-                                             no_params_unassigned));
-  EXPECT_TRUE(MatchesWithReferenceAttributes(no_params_unassigned,
-                                             with_params_unassigned));
-
-  // Case 5: Both lack parameters -> SHOULD match.
-  Codec no_params_2 = CreateAudioCodec(103, kRedCodecName, 48000, 2);
-  EXPECT_TRUE(MatchesWithReferenceAttributes(no_params, no_params_2));
-
-  // A video RED codec should not match any audio RED codec,
-  // independent of parameters.
-  Codec video_red_codec = CreateVideoCodec(107, kRedCodecName);
-  EXPECT_FALSE(MatchesWithReferenceAttributes(no_params, video_red_codec));
-  EXPECT_FALSE(MatchesWithReferenceAttributes(with_params, video_red_codec));
-
-  // Two video RED codecs with different IDs and one lacking parameters
-  // should NOT match.
-  Codec video_red_with_params = CreateVideoCodec(108, kRedCodecName);
-  video_red_with_params.SetParam(kCodecParamNotInNameValueFormat, "120/120");
-  Codec video_red_no_params = CreateVideoCodec(109, kRedCodecName);
-  EXPECT_FALSE(MatchesWithReferenceAttributes(video_red_with_params,
-                                              video_red_no_params));
-  EXPECT_FALSE(MatchesWithReferenceAttributes(video_red_no_params,
-                                              video_red_with_params));
-}
-
 struct TestParams {
   std::string name;
   SdpVideoFormat codec1;
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 4315b46bb2..9863aa6da6 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -407,7 +407,6 @@ rtc_library("codec_vendor") {
     ":typed_codec_vendor",
     ":used_ids",
     "../api:field_trials_view",
-    "../api:payload_type",
     "../api:rtc_error",
     "../api:rtp_parameters",
     "../api:rtp_transceiver_direction",
diff --git a/pc/codec_vendor.cc b/pc/codec_vendor.cc
index 50f3ef818f..1d865fd7de 100644
--- a/pc/codec_vendor.cc
+++ b/pc/codec_vendor.cc
@@ -23,7 +23,6 @@
 #include "absl/strings/string_view.h"
 #include "api/field_trials_view.h"
 #include "api/media_types.h"
-#include "api/payload_type.h"
 #include "api/rtc_error.h"
 #include "api/rtp_parameters.h"
 #include "api/rtp_transceiver_direction.h"
@@ -621,8 +620,7 @@ RTCErrorOr<std::vector<Codec>> CodecVendor::GetNegotiatedCodecsForOffer(
   CodecList codecs;
   std::string mid = media_description_options.mid;
   // If current content exists and is not being recycled, use its codecs.
-  if (current_content && current_content->mid() == mid &&
-      IsMediaContentOfType(current_content, media_description_options.type)) {
+  if (current_content && current_content->mid() == mid) {
     RTCErrorOr<CodecList> checked_codec_list =
         CodecList::Create(current_content->media_description()->codecs());
     if (!checked_codec_list.ok()) {
@@ -760,8 +758,7 @@ RTCErrorOr<Codecs> CodecVendor::GetNegotiatedCodecsForAnswer(
   RTC_LOG_THREAD_BLOCK_COUNT();
   CodecList codecs;
   std::string mid = media_description_options.mid;
-  if (current_content && current_content->mid() == mid &&
-      IsMediaContentOfType(current_content, media_description_options.type)) {
+  if (current_content && current_content->mid() == mid) {
     RTCErrorOr<CodecList> checked_codec_list =
         CodecList::Create(current_content->media_description()->codecs());
     if (!checked_codec_list.ok()) {
diff --git a/pc/peer_connection_encodings_integrationtest.cc b/pc/peer_connection_encodings_integrationtest.cc
index 9b9ee258c1..6b84d6bf93 100644
--- a/pc/peer_connection_encodings_integrationtest.cc
+++ b/pc/peer_connection_encodings_integrationtest.cc
@@ -2129,64 +2129,6 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
   EXPECT_EQ(parameters.codecs[0].name, opus->name);
 }
 
-TEST_F(PeerConnectionEncodingsIntegrationTest,
-       EncodingParametersRedEnabledBeforeNegotiationAudioWithFieldTrial) {
-  scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper =
-      CreatePc("WebRTC-PayloadTypesInTransport/Enabled/");
-  scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper =
-      CreatePc("WebRTC-PayloadTypesInTransport/Enabled/");
-  ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
-
-  std::vector<RtpCodecCapability> send_codecs =
-      local_pc_wrapper->pc_factory()
-          ->GetRtpSenderCapabilities(MediaType::AUDIO)
-          .codecs;
-
-  std::optional<RtpCodecCapability> opus =
-      local_pc_wrapper->FindFirstSendCodecWithName(MediaType::AUDIO, "opus");
-  ASSERT_TRUE(opus);
-
-  std::optional<RtpCodecCapability> red =
-      local_pc_wrapper->FindFirstSendCodecWithName(MediaType::AUDIO, "red");
-  ASSERT_TRUE(red);
-
-  RtpTransceiverInit init;
-  init.direction = RtpTransceiverDirection::kSendOnly;
-  RtpEncodingParameters encoding_parameters;
-  encoding_parameters.codec = opus;
-  init.send_encodings.push_back(encoding_parameters);
-
-  auto transceiver_or_error =
-      local_pc_wrapper->pc()->AddTransceiver(MediaType::AUDIO, init);
-  ASSERT_TRUE(transceiver_or_error.ok());
-  scoped_refptr<RtpTransceiverInterface> audio_transceiver =
-      transceiver_or_error.MoveValue();
-
-  // Preferring RED over Opus should enable RED with Opus encoding.
-  send_codecs[0] = red.value();
-  send_codecs[1] = opus.value();
-
-  ASSERT_TRUE(audio_transceiver->SetCodecPreferences(send_codecs).ok());
-  NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper);
-  local_pc_wrapper->WaitForConnection();
-  remote_pc_wrapper->WaitForConnection();
-
-  RtpParameters parameters = audio_transceiver->sender()->GetParameters();
-  EXPECT_EQ(parameters.encodings[0].codec, opus);
-  EXPECT_EQ(parameters.codecs[0].name, red->name);
-
-  // Check that it's possible to switch back to Opus without RED.
-  send_codecs[0] = opus.value();
-  send_codecs[1] = red.value();
-
-  ASSERT_TRUE(audio_transceiver->SetCodecPreferences(send_codecs).ok());
-  NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper);
-
-  parameters = audio_transceiver->sender()->GetParameters();
-  EXPECT_EQ(parameters.encodings[0].codec, opus);
-  EXPECT_EQ(parameters.codecs[0].name, opus->name);
-}
-
 TEST_F(PeerConnectionEncodingsIntegrationTest,
        SetParametersRejectsScalabilityModeForSelectedCodec) {
   scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
diff --git a/pc/peer_connection_media_unittest.cc b/pc/peer_connection_media_unittest.cc
index a4671c9906..3890be2dc3 100644
--- a/pc/peer_connection_media_unittest.cc
+++ b/pc/peer_connection_media_unittest.cc
@@ -1376,10 +1376,10 @@ TEST_P(PeerConnectionMediaTest, RedFmtpPayloadTypeMustMatchBaseCodecs) {
       CreateAudioCodec(120, "foo", kDefaultAudioClockRateHz, 1));
   callee_fake_codecs.push_back(
       CreateAudioCodec(121, kRedCodecName, kDefaultAudioClockRateHz, 1));
-  callee_fake_codecs.back().SetParam(kCodecParamNotInNameValueFormat,
-                                     "122/122");
   callee_fake_codecs.push_back(
       CreateAudioCodec(122, "bar", kDefaultAudioClockRateHz, 1));
+  callee_fake_codecs.back().SetParam(kCodecParamNotInNameValueFormat,
+                                     "122/122");
   auto callee_fake_engine = std::make_unique<FakeMediaEngine>();
   callee_fake_engine->SetAudioCodecs(callee_fake_codecs);
   auto callee = CreatePeerConnectionWithAudio(std::move(callee_fake_engine));
diff --git a/pc/peer_connection_stability_integrationtest.cc b/pc/peer_connection_stability_integrationtest.cc
index 3acfd99619..8d3052215b 100644
--- a/pc/peer_connection_stability_integrationtest.cc
+++ b/pc/peer_connection_stability_integrationtest.cc
@@ -949,554 +949,5 @@ TEST_F(PeerConnectionIntegrationTest, BasicOfferAnswerPayloadTypesStable) {
               ElementsAreArray(this_golden.callee_remote));
 }
 
-TEST_F(PeerConnectionIntegrationTest,
-       BasicOfferAnswerPayloadTypesStableWithFieldTrial) {
-  SetFieldTrials("WebRTC-PayloadTypesInTransport/Enabled/");
-  FactorySignature factory_signature;
-  ASSERT_THAT(factory_signature.id(),
-              Not(Eq(FactorySignature::Id::kNotRecognized)));
-  ASSERT_TRUE(CreatePeerConnectionWrappers());
-  ConnectFakeSignalingForSdpOnly();
-  caller()->AddAudioVideoTracks();
-  callee()->AddAudioVideoTracks();
-  // Start offer/answer exchange and wait for it to complete.
-  caller()->CreateAndSetAndSignalOffer();
-
-  ASSERT_THAT(
-      WaitUntil([&] { return SignalingStateStable(); }, ::testing::IsTrue()),
-      IsRtcOk());
-
-  // Extract PT and codec from all media sections, and check that they
-  // are stable (what was expected).
-  // Maintenance: In order to get a new golden set of strings, make the list
-  // empty and run. Gmock will output a valid C++ array initializer for you.
-
-  std::vector<ResultingCodecList> golden_answers = {
-      {.factory_id = FactorySignature::Id::kWebRtcTipOfTree,
-       .caller_local =
-           {"1 [111:audio/opus/48000/2;minptime=10;useinbandfec=1]",
-            "1 [63:audio/red/48000/2;=111/111]",
-            "1 [9:audio/G722/8000/1]",
-            "1 [0:audio/PCMU/8000/1]",
-            "1 [8:audio/PCMA/8000/1]",
-            "1 [13:audio/CN/8000/1]",
-            "1 [110:audio/telephone-event/48000/1]",
-            "1 [126:audio/telephone-event/8000/1]",
-            "2 [96:video/VP8/90000/0]",
-            "2 [97:video/rtx/90000/0;apt=96]",
-            "2 "
-            "[103:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-id="
-            "42001f]",
-            "2 [104:video/rtx/90000/0;apt=103]",
-            "2 "
-            "[107:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-id="
-            "42001f]",
-            "2 [108:video/rtx/90000/0;apt=107]",
-            "2 "
-            "[109:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-id="
-            "42e01f]",
-            "2 [114:video/rtx/90000/0;apt=109]",
-            "2 "
-            "[115:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-id="
-            "42e01f]",
-            "2 [116:video/rtx/90000/0;apt=115]",
-            "2 "
-            "[117:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-id="
-            "4d001f]",
-            "2 [118:video/rtx/90000/0;apt=117]",
-            "2 "
-            "[39:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-id="
-            "4d001f]",
-            "2 [40:video/rtx/90000/0;apt=39]",
-            "2 [45:video/AV1/90000/0;level-idx=5;profile=0;tier=0]",
-            "2 [46:video/rtx/90000/0;apt=45]",
-            "2 [98:video/VP9/90000/0;profile-id=0]",
-            "2 [99:video/rtx/90000/0;apt=98]",
-            "2 [100:video/VP9/90000/0;profile-id=2]",
-            "2 [101:video/rtx/90000/0;apt=100]",
-            "2 [119:video/red/90000/0]",
-            "2 [120:video/rtx/90000/0;apt=119]",
-            "2 [121:video/ulpfec/90000/0]"},
-       .caller_remote =
-           {"1 [111:audio/opus/48000/2;minptime=10;useinbandfec=1]",
-            "1 [63:audio/red/48000/2;=111/111]",
-            "1 [9:audio/G722/8000/1]",
-            "1 [0:audio/PCMU/8000/1]",
-            "1 [8:audio/PCMA/8000/1]",
-            "1 [13:audio/CN/8000/1]",
-            "1 [110:audio/telephone-event/48000/1]",
-            "1 [126:audio/telephone-event/8000/1]",
-            "2 [96:video/VP8/90000/0]",
-            "2 [97:video/rtx/90000/0;apt=96]",
-            "2 "
-            "[103:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-id="
-            "42001f]",
-            "2 [104:video/rtx/90000/0;apt=103]",
-            "2 "
-            "[107:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-id="
-            "42001f]",
-            "2 [108:video/rtx/90000/0;apt=107]",
-            "2 "
-            "[109:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-id="
-            "42e01f]",
-            "2 [114:video/rtx/90000/0;apt=109]",
-            "2 "
-            "[115:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-id="
-            "42e01f]",
-            "2 [116:video/rtx/90000/0;apt=115]",
-            "2 "
-            "[117:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-id="
-            "4d001f]",
-            "2 [118:video/rtx/90000/0;apt=117]",
-            "2 "
-            "[39:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-id="
-            "4d001f]",
-            "2 [40:video/rtx/90000/0;apt=39]",
-            "2 [45:video/AV1/90000/0;level-idx=5;profile=0;tier=0]",
-            "2 [46:video/rtx/90000/0;apt=45]",
-            "2 [98:video/VP9/90000/0;profile-id=0]",
-            "2 [99:video/rtx/90000/0;apt=98]",
-            "2 [100:video/VP9/90000/0;profile-id=2]",
-            "2 [101:video/rtx/90000/0;apt=100]",
-            "2 [119:video/red/90000/0]",
-            "2 [120:video/rtx/90000/0;apt=119]",
-            "2 [121:video/ulpfec/90000/0]"},
-       .callee_local =
-           {"1 [111:audio/opus/48000/2;minptime=10;useinbandfec=1]",
-            "1 [63:audio/red/48000/2;=111/111]",
-            "1 [9:audio/G722/8000/1]",
-            "1 [0:audio/PCMU/8000/1]",
-            "1 [8:audio/PCMA/8000/1]",
-            "1 [13:audio/CN/8000/1]",
-            "1 [110:audio/telephone-event/48000/1]",
-            "1 [126:audio/telephone-event/8000/1]",
-            "2 [96:video/VP8/90000/0]",
-            "2 [97:video/rtx/90000/0;apt=96]",
-            "2 "
-            "[103:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-id="
-            "42001f]",
-            "2 [104:video/rtx/90000/0;apt=103]",
-            "2 "
-            "[107:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-id="
-            "42001f]",
-            "2 [108:video/rtx/90000/0;apt=107]",
-            "2 "
-            "[109:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-id="
-            "42e01f]",
-            "2 [114:video/rtx/90000/0;apt=109]",
-            "2 "
-            "[115:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-id="
-            "42e01f]",
-            "2 [116:video/rtx/90000/0;apt=115]",
-            "2 "
-            "[117:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-id="
-            "4d001f]",
-            "2 [118:video/rtx/90000/0;apt=117]",
-            "2 "
-            "[39:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-id="
-            "4d001f]",
-            "2 [40:video/rtx/90000/0;apt=39]",
-            "2 [45:video/AV1/90000/0;level-idx=5;profile=0;tier=0]",
-            "2 [46:video/rtx/90000/0;apt=45]",
-            "2 [98:video/VP9/90000/0;profile-id=0]",
-            "2 [99:video/rtx/90000/0;apt=98]",
-            "2 [100:video/VP9/90000/0;profile-id=2]",
-            "2 [101:video/rtx/90000/0;apt=100]",
-            "2 [119:video/red/90000/0]",
-            "2 [120:video/rtx/90000/0;apt=119]",
-            "2 [121:video/ulpfec/90000/0]"},
-       .callee_remote =
-           {"1 [111:audio/opus/48000/2;minptime=10;useinbandfec=1]",
-            "1 [63:audio/red/48000/2;=111/111]",
-            "1 [9:audio/G722/8000/1]",
-            "1 [0:audio/PCMU/8000/1]",
-            "1 [8:audio/PCMA/8000/1]",
-            "1 [13:audio/CN/8000/1]",
-            "1 [110:audio/telephone-event/48000/1]",
-            "1 [126:audio/telephone-event/8000/1]",
-            "2 [96:video/VP8/90000/0]",
-            "2 [97:video/rtx/90000/0;apt=96]",
-            "2 "
-            "[103:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-id="
-            "42001f]",
-            "2 [104:video/rtx/90000/0;apt=103]",
-            "2 "
-            "[107:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-id="
-            "42001f]",
-            "2 [108:video/rtx/90000/0;apt=107]",
-            "2 "
-            "[109:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-id="
-            "42e01f]",
-            "2 [114:video/rtx/90000/0;apt=109]",
-            "2 "
-            "[115:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-id="
-            "42e01f]",
-            "2 [116:video/rtx/90000/0;apt=115]",
-            "2 "
-            "[117:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-id="
-            "4d001f]",
-            "2 [118:video/rtx/90000/0;apt=117]",
-            "2 "
-            "[39:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-id="
-            "4d001f]",
-            "2 [40:video/rtx/90000/0;apt=39]",
-            "2 [45:video/AV1/90000/0;level-idx=5;profile=0;tier=0]",
-            "2 [46:video/rtx/90000/0;apt=45]",
-            "2 [98:video/VP9/90000/0;profile-id=0]",
-            "2 [99:video/rtx/90000/0;apt=98]",
-            "2 [100:video/VP9/90000/0;profile-id=2]",
-            "2 [101:video/rtx/90000/0;apt=100]",
-            "2 [119:video/red/90000/0]",
-            "2 [120:video/rtx/90000/0;apt=119]",
-            "2 [121:video/ulpfec/90000/0]"}},
-
-      {.factory_id = FactorySignature::Id::kWebRtcMoreConfigs1,
-       .caller_local =
-           {"1 [111:audio/opus/48000/2;minptime=10;useinbandfec=1]",
-            "1 [63:audio/red/48000/2;=111/111]", "1 [9:audio/G722/8000/1]",
-            "1 [0:audio/PCMU/8000/1]", "1 [8:audio/PCMA/8000/1]",
-            "1 [13:audio/CN/8000/1]", "1 [110:audio/telephone-event/48000/1]",
-            "1 [126:audio/telephone-event/8000/1]", "2 [96:video/VP8/90000/0]",
-            "2 [97:video/rtx/90000/0;apt=96]",
-            "2 [39:video/AV1/90000/0;level-idx=5;profile=0;tier=0]",
-            "2 [40:video/rtx/90000/0;apt=39]",
-            "2 [98:video/VP9/90000/0;profile-id=0]",
-            "2 [99:video/rtx/90000/0;apt=98]",
-            "2 [100:video/VP9/90000/0;profile-id=2]",
-            "2 [101:video/rtx/90000/0;apt=100]", "2 [103:video/red/90000/0]",
-            "2 [104:video/rtx/90000/0;apt=103]",
-            "2 [107:video/ulpfec/90000/0]"},
-       .caller_remote =
-           {"1 [111:audio/opus/48000/2;minptime=10;useinbandfec=1]",
-            "1 [63:audio/red/48000/2;=111/111]", "1 [9:audio/G722/8000/1]",
-            "1 [0:audio/PCMU/8000/1]", "1 [8:audio/PCMA/8000/1]",
-            "1 [13:audio/CN/8000/1]", "1 [110:audio/telephone-event/48000/1]",
-            "1 [126:audio/telephone-event/8000/1]", "2 [96:video/VP8/90000/0]",
-            "2 [97:video/rtx/90000/0;apt=96]",
-            "2 [39:video/AV1/90000/0;level-idx=5;profile=0;tier=0]",
-            "2 [40:video/rtx/90000/0;apt=39]",
-            "2 [98:video/VP9/90000/0;profile-id=0]",
-            "2 [99:video/rtx/90000/0;apt=98]",
-            "2 [100:video/VP9/90000/0;profile-id=2]",
-            "2 [101:video/rtx/90000/0;apt=100]", "2 [103:video/red/90000/0]",
-            "2 [104:video/rtx/90000/0;apt=103]",
-            "2 [107:video/ulpfec/90000/0]"},
-       .callee_local =
-           {"1 [111:audio/opus/48000/2;minptime=10;useinbandfec=1]",
-            "1 [63:audio/red/48000/2;=111/111]", "1 [9:audio/G722/8000/1]",
-            "1 [0:audio/PCMU/8000/1]", "1 [8:audio/PCMA/8000/1]",
-            "1 [13:audio/CN/8000/1]", "1 [110:audio/telephone-event/48000/1]",
-            "1 [126:audio/telephone-event/8000/1]", "2 [96:video/VP8/90000/0]",
-            "2 [97:video/rtx/90000/0;apt=96]",
-            "2 [39:video/AV1/90000/0;level-idx=5;profile=0;tier=0]",
-            "2 [40:video/rtx/90000/0;apt=39]",
-            "2 [98:video/VP9/90000/0;profile-id=0]",
-            "2 [99:video/rtx/90000/0;apt=98]",
-            "2 [100:video/VP9/90000/0;profile-id=2]",
-            "2 [101:video/rtx/90000/0;apt=100]", "2 [103:video/red/90000/0]",
-            "2 [104:video/rtx/90000/0;apt=103]",
-            "2 [107:video/ulpfec/90000/0]"},
-       .callee_remote =
-           {"1 [111:audio/opus/48000/2;minptime=10;useinbandfec=1]",
-            "1 [63:audio/red/48000/2;=111/111]", "1 [9:audio/G722/8000/1]",
-            "1 [0:audio/PCMU/8000/1]", "1 [8:audio/PCMA/8000/1]",
-            "1 [13:audio/CN/8000/1]", "1 [110:audio/telephone-event/48000/1]",
-            "1 [126:audio/telephone-event/8000/1]", "2 [96:video/VP8/90000/0]",
-            "2 [97:video/rtx/90000/0;apt=96]",
-            "2 [39:video/AV1/90000/0;level-idx=5;profile=0;tier=0]",
-            "2 [40:video/rtx/90000/0;apt=39]",
-            "2 [98:video/VP9/90000/0;profile-id=0]",
-            "2 [99:video/rtx/90000/0;apt=98]",
-            "2 [100:video/VP9/90000/0;profile-id=2]",
-            "2 [101:video/rtx/90000/0;apt=100]", "2 [103:video/red/90000/0]",
-            "2 [104:video/rtx/90000/0;apt=103]",
-            "2 [107:video/ulpfec/90000/0]"}},
-      {.factory_id = FactorySignature::Id::kWebRtcAndroid,
-       .caller_local =
-           {"1 [111:audio/opus/48000/2;minptime=10;useinbandfec=1]",
-            "1 [63:audio/red/48000/2;=111/111]", "1 [9:audio/G722/8000/1]",
-            "1 [0:audio/PCMU/8000/1]", "1 [8:audio/PCMA/8000/1]",
-            "1 [13:audio/CN/8000/1]", "1 [110:audio/telephone-event/48000/1]",
-            "1 [126:audio/telephone-event/8000/1]", "2 [96:video/VP8/90000/0]",
-            "2 [97:video/rtx/90000/0;apt=96]",
-            "2 [39:video/AV1/90000/0;level-idx=5;profile=0;tier=0]",
-            "2 [40:video/rtx/90000/0;apt=39]",
-            "2 [98:video/VP9/90000/0;profile-id=0]",
-            "2 [99:video/rtx/90000/0;apt=98]", "2 [100:video/red/90000/0]",
-            "2 [101:video/rtx/90000/0;apt=100]",
-            "2 [103:video/ulpfec/90000/0]"},
-       .caller_remote =
-           {"1 [111:audio/opus/48000/2;minptime=10;useinbandfec=1]",
-            "1 [63:audio/red/48000/2;=111/111]", "1 [9:audio/G722/8000/1]",
-            "1 [0:audio/PCMU/8000/1]", "1 [8:audio/PCMA/8000/1]",
-            "1 [13:audio/CN/8000/1]", "1 [110:audio/telephone-event/48000/1]",
-            "1 [126:audio/telephone-event/8000/1]", "2 [96:video/VP8/90000/0]",
-            "2 [97:video/rtx/90000/0;apt=96]",
-            "2 [39:video/AV1/90000/0;level-idx=5;profile=0;tier=0]",
-            "2 [40:video/rtx/90000/0;apt=39]",
-            "2 [98:video/VP9/90000/0;profile-id=0]",
-            "2 [99:video/rtx/90000/0;apt=98]", "2 [100:video/red/90000/0]",
-            "2 [101:video/rtx/90000/0;apt=100]",
-            "2 [103:video/ulpfec/90000/0]"},
-       .callee_local =
-           {"1 [111:audio/opus/48000/2;minptime=10;useinbandfec=1]",
-            "1 [63:audio/red/48000/2;=111/111]", "1 [9:audio/G722/8000/1]",
-            "1 [0:audio/PCMU/8000/1]", "1 [8:audio/PCMA/8000/1]",
-            "1 [13:audio/CN/8000/1]", "1 [110:audio/telephone-event/48000/1]",
-            "1 [126:audio/telephone-event/8000/1]", "2 [96:video/VP8/90000/0]",
-            "2 [97:video/rtx/90000/0;apt=96]",
-            "2 [39:video/AV1/90000/0;level-idx=5;profile=0;tier=0]",
-            "2 [40:video/rtx/90000/0;apt=39]",
-            "2 [98:video/VP9/90000/0;profile-id=0]",
-            "2 [99:video/rtx/90000/0;apt=98]", "2 [100:video/red/90000/0]",
-            "2 [101:video/rtx/90000/0;apt=100]",
-            "2 [103:video/ulpfec/90000/0]"},
-       .callee_remote =
-           {"1 [111:audio/opus/48000/2;minptime=10;useinbandfec=1]",
-            "1 [63:audio/red/48000/2;=111/111]", "1 [9:audio/G722/8000/1]",
-            "1 [0:audio/PCMU/8000/1]", "1 [8:audio/PCMA/8000/1]",
-            "1 [13:audio/CN/8000/1]", "1 [110:audio/telephone-event/48000/1]",
-            "1 [126:audio/telephone-event/8000/1]", "2 [96:video/VP8/90000/0]",
-            "2 [97:video/rtx/90000/0;apt=96]",
-            "2 [39:video/AV1/90000/0;level-idx=5;profile=0;tier=0]",
-            "2 [40:video/rtx/90000/0;apt=39]",
-            "2 [98:video/VP9/90000/0;profile-id=0]",
-            "2 [99:video/rtx/90000/0;apt=98]", "2 [100:video/red/90000/0]",
-            "2 [101:video/rtx/90000/0;apt=100]",
-            "2 [103:video/ulpfec/90000/0]"}},
-      {.factory_id = FactorySignature::Id::kGoogleInternal,
-       .caller_local =
-           {"1 [111:audio/opus/48000/2;minptime=10;useinbandfec=1]",
-            "1 [63:audio/red/48000/2;=111/111]",
-            "1 [9:audio/G722/8000/1]",
-            "1 [0:audio/PCMU/8000/1]",
-            "1 [8:audio/PCMA/8000/1]",
-            "1 [13:audio/CN/8000/1]",
-            "1 [110:audio/telephone-event/48000/1]",
-            "1 [126:audio/telephone-event/8000/1]",
-            "2 [96:video/VP8/90000/0]",
-            "2 [97:video/rtx/90000/0;apt=96]",
-            "2 "
-            "[100:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-"
-            "id=42001f]",
-            "2 [101:video/rtx/90000/0;apt=100]",
-            "2 "
-            "[103:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-"
-            "id=42001f]",
-            "2 [104:video/rtx/90000/0;apt=103]",
-            "2 "
-            "[107:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-"
-            "id=42e01f]",
-            "2 [108:video/rtx/90000/0;apt=107]",
-            "2 "
-            "[109:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-"
-            "id=42e01f]",
-            "2 [114:video/rtx/90000/0;apt=109]",
-            "2 "
-            "[115:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-"
-            "id=4d001f]",
-            "2 [116:video/rtx/90000/0;apt=115]",
-            "2 "
-            "[39:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-"
-            "id=4d001f]",
-            "2 [40:video/rtx/90000/0;apt=39]",
-            "2 [98:video/VP9/90000/0;profile-id=0]",
-            "2 [99:video/rtx/90000/0;apt=98]",
-            "2 [117:video/red/90000/0]",
-            "2 [118:video/rtx/90000/0;apt=117]",
-            "2 [119:video/ulpfec/90000/0]"},
-       .caller_remote =
-           {"1 [111:audio/opus/48000/2;minptime=10;useinbandfec=1]",
-            "1 [63:audio/red/48000/2;=111/111]",
-            "1 [9:audio/G722/8000/1]",
-            "1 [0:audio/PCMU/8000/1]",
-            "1 [8:audio/PCMA/8000/1]",
-            "1 [13:audio/CN/8000/1]",
-            "1 [110:audio/telephone-event/48000/1]",
-            "1 [126:audio/telephone-event/8000/1]",
-            "2 [96:video/VP8/90000/0]",
-            "2 [97:video/rtx/90000/0;apt=96]",
-            "2 "
-            "[100:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-"
-            "id=42001f]",
-            "2 [101:video/rtx/90000/0;apt=100]",
-            "2 "
-            "[103:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-"
-            "id=42001f]",
-            "2 [104:video/rtx/90000/0;apt=103]",
-            "2 "
-            "[107:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-"
-            "id=42e01f]",
-            "2 [108:video/rtx/90000/0;apt=107]",
-            "2 "
-            "[109:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-"
-            "id=42e01f]",
-            "2 [114:video/rtx/90000/0;apt=109]",
-            "2 "
-            "[115:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-"
-            "id=4d001f]",
-            "2 [116:video/rtx/90000/0;apt=115]",
-            "2 "
-            "[39:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-"
-            "id=4d001f]",
-            "2 [40:video/rtx/90000/0;apt=39]",
-            "2 [98:video/VP9/90000/0;profile-id=0]",
-            "2 [99:video/rtx/90000/0;apt=98]",
-            "2 [117:video/red/90000/0]",
-            "2 [118:video/rtx/90000/0;apt=117]",
-            "2 [119:video/ulpfec/90000/0]"},
-       .callee_local =
-           {"1 [111:audio/opus/48000/2;minptime=10;useinbandfec=1]",
-            "1 [63:audio/red/48000/2;=111/111]",
-            "1 [9:audio/G722/8000/1]",
-            "1 [0:audio/PCMU/8000/1]",
-            "1 [8:audio/PCMA/8000/1]",
-            "1 [13:audio/CN/8000/1]",
-            "1 [110:audio/telephone-event/48000/1]",
-            "1 [126:audio/telephone-event/8000/1]",
-            "2 [96:video/VP8/90000/0]",
-            "2 [97:video/rtx/90000/0;apt=96]",
-            "2 "
-            "[100:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-"
-            "id=42001f]",
-            "2 [101:video/rtx/90000/0;apt=100]",
-            "2 "
-            "[103:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-"
-            "id=42001f]",
-            "2 [104:video/rtx/90000/0;apt=103]",
-            "2 "
-            "[107:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-"
-            "id=42e01f]",
-            "2 [108:video/rtx/90000/0;apt=107]",
-            "2 "
-            "[109:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-"
-            "id=42e01f]",
-            "2 [114:video/rtx/90000/0;apt=109]",
-            "2 "
-            "[115:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-"
-            "id=4d001f]",
-            "2 [116:video/rtx/90000/0;apt=115]",
-            "2 "
-            "[39:video/H264/90000/"
-            "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-"
-            "id=4d001f]",
-            "2 [40:video/rtx/90000/0;apt=39]",
-            "2 [98:video/VP9/90000/0;profile-id=0]",
-            "2 [99:video/rtx/90000/0;apt=98]",
-            "2 [117:video/red/90000/0]",
-            "2 [118:video/rtx/90000/0;apt=117]",
-            "2 [119:video/ulpfec/90000/0]"},
-       .callee_remote = {
-           "1 [111:audio/opus/48000/2;minptime=10;useinbandfec=1]",
-           "1 [63:audio/red/48000/2;=111/111]",
-           "1 [9:audio/G722/8000/1]",
-           "1 [0:audio/PCMU/8000/1]",
-           "1 [8:audio/PCMA/8000/1]",
-           "1 [13:audio/CN/8000/1]",
-           "1 [110:audio/telephone-event/48000/1]",
-           "1 [126:audio/telephone-event/8000/1]",
-           "2 [96:video/VP8/90000/0]",
-           "2 [97:video/rtx/90000/0;apt=96]",
-           "2 "
-           "[100:video/H264/90000/"
-           "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-id="
-           "42001f]",
-           "2 [101:video/rtx/90000/0;apt=100]",
-           "2 "
-           "[103:video/H264/90000/"
-           "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-id="
-           "42001f]",
-           "2 [104:video/rtx/90000/0;apt=103]",
-           "2 "
-           "[107:video/H264/90000/"
-           "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-id="
-           "42e01f]",
-           "2 [108:video/rtx/90000/0;apt=107]",
-           "2 "
-           "[109:video/H264/90000/"
-           "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-id="
-           "42e01f]",
-           "2 [114:video/rtx/90000/0;apt=109]",
-           "2 "
-           "[115:video/H264/90000/"
-           "0;level-asymmetry-allowed=1;packetization-mode=1;profile-level-id="
-           "4d001f]",
-           "2 [116:video/rtx/90000/0;apt=115]",
-           "2 "
-           "[39:video/H264/90000/"
-           "0;level-asymmetry-allowed=1;packetization-mode=0;profile-level-id="
-           "4d001f]",
-           "2 [40:video/rtx/90000/0;apt=39]",
-           "2 [98:video/VP9/90000/0;profile-id=0]",
-           "2 [99:video/rtx/90000/0;apt=98]",
-           "2 [117:video/red/90000/0]",
-           "2 [118:video/rtx/90000/0;apt=117]",
-           "2 [119:video/ulpfec/90000/0]"}}};
-  auto this_golden_it =
-      std::find_if(golden_answers.begin(), golden_answers.end(),
-                   [&](const ResultingCodecList& candidate) {
-                     return candidate.factory_id == factory_signature.id();
-                   });
-  ASSERT_THAT(this_golden_it, Not(Eq(golden_answers.end())))
-      << "Add this result set to golden_answers:\n"
-      << DumpAsResultingCodecList(
-             factory_signature.id(),
-             CodecList(*caller()->pc()->local_description()),
-             CodecList(*caller()->pc()->remote_description()),
-             CodecList(*callee()->pc()->local_description()),
-             CodecList(*callee()->pc()->remote_description()));
-
-  const ResultingCodecList& this_golden = *this_golden_it;
-  EXPECT_THAT(CodecList(*caller()->pc()->local_description()),
-              ElementsAreArray(this_golden.caller_local));
-  EXPECT_THAT(CodecList(*caller()->pc()->remote_description()),
-              ElementsAreArray(this_golden.caller_remote));
-  EXPECT_THAT(CodecList(*callee()->pc()->local_description()),
-              ElementsAreArray(this_golden.callee_local));
-  EXPECT_THAT(CodecList(*callee()->pc()->remote_description()),
-              ElementsAreArray(this_golden.callee_remote));
-}
-
 }  // namespace
 }  // namespace webrtc

From 510e4f2f3f31065c7c998f65cb1ef440cd387547 Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Sat, 7 Mar 2026 11:01:55 +0100
Subject: [PATCH 263/472] Replace ArrayView with std::span in net/dcsctp/

Search&Replace MakeArrayView and ArrayView with std::span
Search&Replace include "api/array_view.h" with include <span>
Remove build dependencies on array_view target

Bug: webrtc:439801349
Change-Id: Ib1f7d5f042ed1b1d331cb3171259340e391870e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/454303
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47116}
---
 net/dcsctp/fuzzers/BUILD.gn                   |  2 --
 net/dcsctp/fuzzers/dcsctp_fuzzers.cc          |  8 ++---
 net/dcsctp/fuzzers/dcsctp_fuzzers.h           | 15 +++++-----
 net/dcsctp/fuzzers/dcsctp_fuzzers_test.cc     |  1 -
 net/dcsctp/packet/BUILD.gn                    | 12 +-------
 net/dcsctp/packet/bounded_byte_reader.h       | 15 +++++-----
 net/dcsctp/packet/bounded_byte_reader_test.cc |  1 -
 net/dcsctp/packet/bounded_byte_writer.h       | 10 +++----
 net/dcsctp/packet/bounded_byte_writer_test.cc |  1 -
 net/dcsctp/packet/chunk/abort_chunk.cc        |  7 ++---
 net/dcsctp/packet/chunk/abort_chunk.h         |  4 +--
 net/dcsctp/packet/chunk/chunk.cc              |  6 ++--
 net/dcsctp/packet/chunk/chunk.h               |  6 ++--
 net/dcsctp/packet/chunk/cookie_ack_chunk.cc   |  5 ++--
 net/dcsctp/packet/chunk/cookie_ack_chunk.h    |  5 ++--
 net/dcsctp/packet/chunk/cookie_echo_chunk.cc  |  4 +--
 net/dcsctp/packet/chunk/cookie_echo_chunk.h   |  9 +++---
 .../packet/chunk/cookie_echo_chunk_test.cc    |  1 -
 net/dcsctp/packet/chunk/data_chunk.cc         |  5 ++--
 net/dcsctp/packet/chunk/data_chunk.h          |  4 +--
 net/dcsctp/packet/chunk/data_chunk_test.cc    |  1 -
 net/dcsctp/packet/chunk/data_common.h         |  4 +--
 net/dcsctp/packet/chunk/error_chunk.cc        |  7 ++---
 net/dcsctp/packet/chunk/error_chunk.h         |  4 +--
 net/dcsctp/packet/chunk/error_chunk_test.cc   |  1 -
 net/dcsctp/packet/chunk/forward_tsn_chunk.cc  |  6 ++--
 net/dcsctp/packet/chunk/forward_tsn_chunk.h   |  5 ++--
 .../packet/chunk/forward_tsn_chunk_test.cc    |  2 +-
 net/dcsctp/packet/chunk/forward_tsn_common.h  |  4 +--
 .../packet/chunk/heartbeat_ack_chunk.cc       |  6 ++--
 net/dcsctp/packet/chunk/heartbeat_ack_chunk.h |  5 ++--
 .../packet/chunk/heartbeat_ack_chunk_test.cc  |  1 -
 .../packet/chunk/heartbeat_request_chunk.cc   |  6 ++--
 .../packet/chunk/heartbeat_request_chunk.h    |  4 +--
 .../chunk/heartbeat_request_chunk_test.cc     |  1 -
 net/dcsctp/packet/chunk/idata_chunk.cc        |  5 ++--
 net/dcsctp/packet/chunk/idata_chunk.h         |  4 +--
 net/dcsctp/packet/chunk/idata_chunk_test.cc   |  1 -
 net/dcsctp/packet/chunk/iforward_tsn_chunk.cc |  6 ++--
 net/dcsctp/packet/chunk/iforward_tsn_chunk.h  |  5 ++--
 .../packet/chunk/iforward_tsn_chunk_test.cc   |  1 -
 net/dcsctp/packet/chunk/init_ack_chunk.cc     |  7 ++---
 net/dcsctp/packet/chunk/init_ack_chunk.h      |  5 ++--
 .../packet/chunk/init_ack_chunk_test.cc       |  1 -
 net/dcsctp/packet/chunk/init_chunk.cc         |  7 ++---
 net/dcsctp/packet/chunk/init_chunk.h          |  4 +--
 net/dcsctp/packet/chunk/reconfig_chunk.cc     |  6 ++--
 net/dcsctp/packet/chunk/reconfig_chunk.h      |  5 ++--
 .../packet/chunk/reconfig_chunk_test.cc       |  1 -
 net/dcsctp/packet/chunk/sack_chunk.cc         |  5 ++--
 net/dcsctp/packet/chunk/sack_chunk.h          |  6 ++--
 net/dcsctp/packet/chunk/sack_chunk_test.cc    |  1 -
 net/dcsctp/packet/chunk/shutdown_ack_chunk.cc |  5 ++--
 net/dcsctp/packet/chunk/shutdown_ack_chunk.h  |  5 ++--
 net/dcsctp/packet/chunk/shutdown_chunk.cc     |  4 +--
 net/dcsctp/packet/chunk/shutdown_chunk.h      |  5 ++--
 .../packet/chunk/shutdown_complete_chunk.cc   |  4 +--
 .../packet/chunk/shutdown_complete_chunk.h    |  4 +--
 net/dcsctp/packet/crc32c.cc                   |  4 +--
 net/dcsctp/packet/crc32c.h                    |  5 ++--
 ...okie_received_while_shutting_down_cause.cc |  6 ++--
 ...ookie_received_while_shutting_down_cause.h |  4 +--
 .../invalid_mandatory_parameter_cause.cc      |  5 ++--
 .../invalid_mandatory_parameter_cause.h       |  4 +--
 .../invalid_stream_identifier_cause.cc        |  4 +--
 .../invalid_stream_identifier_cause.h         |  4 +--
 .../missing_mandatory_parameter_cause.cc      |  4 +--
 .../missing_mandatory_parameter_cause.h       |  8 ++---
 .../missing_mandatory_parameter_cause_test.cc |  1 -
 .../packet/error_cause/no_user_data_cause.cc  |  4 +--
 .../packet/error_cause/no_user_data_cause.h   |  5 ++--
 .../out_of_resource_error_cause.cc            |  5 ++--
 .../error_cause/out_of_resource_error_cause.h |  4 +--
 .../error_cause/protocol_violation_cause.cc   |  6 ++--
 .../error_cause/protocol_violation_cause.h    |  4 +--
 .../protocol_violation_cause_test.cc          |  1 -
 ...f_an_association_with_new_address_cause.cc |  4 +--
 ...of_an_association_with_new_address_cause.h |  8 ++---
 ...association_with_new_address_cause_test.cc |  1 -
 .../error_cause/stale_cookie_error_cause.cc   |  4 +--
 .../error_cause/stale_cookie_error_cause.h    |  4 +--
 .../unrecognized_chunk_type_cause.cc          |  4 +--
 .../unrecognized_chunk_type_cause.h           |  6 ++--
 .../unrecognized_chunk_type_cause_test.cc     |  1 -
 .../unrecognized_parameter_cause.cc           |  4 +--
 .../unrecognized_parameter_cause.h            |  8 ++---
 .../unrecognized_parameter_cause_test.cc      |  1 -
 .../error_cause/unresolvable_address_cause.cc |  4 +--
 .../error_cause/unresolvable_address_cause.h  |  8 ++---
 .../unresolvable_address_cause_test.cc        |  1 -
 .../error_cause/user_initiated_abort_cause.cc |  6 ++--
 .../error_cause/user_initiated_abort_cause.h  |  4 +--
 .../user_initiated_abort_cause_test.cc        |  1 -
 .../add_incoming_streams_request_parameter.cc |  5 ++--
 .../add_incoming_streams_request_parameter.h  |  4 +--
 .../add_outgoing_streams_request_parameter.cc |  5 ++--
 .../add_outgoing_streams_request_parameter.h  |  4 +--
 .../forward_tsn_supported_parameter.cc        |  5 ++--
 .../forward_tsn_supported_parameter.h         |  4 +--
 .../parameter/heartbeat_info_parameter.cc     |  4 +--
 .../parameter/heartbeat_info_parameter.h      |  8 ++---
 .../incoming_ssn_reset_request_parameter.cc   |  4 +--
 .../incoming_ssn_reset_request_parameter.h    |  6 ++--
 ...coming_ssn_reset_request_parameter_test.cc |  1 -
 .../outgoing_ssn_reset_request_parameter.cc   |  4 +--
 .../outgoing_ssn_reset_request_parameter.h    |  6 ++--
 ...tgoing_ssn_reset_request_parameter_test.cc |  1 -
 net/dcsctp/packet/parameter/parameter.cc      |  9 +++---
 net/dcsctp/packet/parameter/parameter.h       | 12 ++++----
 net/dcsctp/packet/parameter/parameter_test.cc |  4 +--
 .../reconfiguration_response_parameter.cc     |  4 +--
 .../reconfiguration_response_parameter.h      |  4 +--
 .../ssn_tsn_reset_request_parameter.cc        |  4 +--
 .../ssn_tsn_reset_request_parameter.h         |  4 +--
 .../parameter/state_cookie_parameter.cc       |  4 +--
 .../packet/parameter/state_cookie_parameter.h |  8 ++---
 .../parameter/state_cookie_parameter_test.cc  |  1 -
 .../supported_extensions_parameter.cc         |  4 +--
 .../supported_extensions_parameter.h          |  6 ++--
 .../supported_extensions_parameter_test.cc    |  1 -
 ...ero_checksum_acceptable_chunk_parameter.cc |  5 ++--
 ...zero_checksum_acceptable_chunk_parameter.h |  4 +--
 net/dcsctp/packet/sctp_packet.cc              | 11 ++++---
 net/dcsctp/packet/sctp_packet.h               | 14 ++++-----
 net/dcsctp/packet/sctp_packet_test.cc         |  1 -
 net/dcsctp/packet/tlv_trait.h                 |  8 ++---
 net/dcsctp/packet/tlv_trait_test.cc           | 10 +++----
 net/dcsctp/public/BUILD.gn                    |  4 ---
 net/dcsctp/public/dcsctp_message.h            |  4 +--
 net/dcsctp/public/dcsctp_socket.h             | 30 ++++++++-----------
 net/dcsctp/public/mock_dcsctp_socket.h        | 14 ++++-----
 net/dcsctp/public/packet_observer.h           |  7 ++---
 .../public/text_pcap_packet_observer.cc       |  9 +++---
 net/dcsctp/public/text_pcap_packet_observer.h |  8 ++---
 net/dcsctp/rx/BUILD.gn                        |  5 ----
 .../rx/interleaved_reassembly_streams.cc      |  7 ++---
 .../rx/interleaved_reassembly_streams.h       | 11 ++++---
 net/dcsctp/rx/reassembly_queue.cc             | 20 +++++--------
 net/dcsctp/rx/reassembly_queue.h              | 12 ++++----
 net/dcsctp/rx/reassembly_queue_test.cc        |  8 ++---
 net/dcsctp/rx/reassembly_streams.h            |  9 +++---
 .../rx/traditional_reassembly_streams.cc      |  7 ++---
 .../rx/traditional_reassembly_streams.h       | 11 ++++---
 net/dcsctp/socket/BUILD.gn                    |  6 ----
 net/dcsctp/socket/callback_deferrer.cc        | 10 +++----
 net/dcsctp/socket/callback_deferrer.h         | 11 ++++---
 net/dcsctp/socket/dcsctp_socket.cc            | 19 ++++++------
 net/dcsctp/socket/dcsctp_socket.h             | 18 +++++------
 .../socket/dcsctp_socket_network_test.cc      | 15 +++++-----
 net/dcsctp/socket/dcsctp_socket_test.cc       |  1 -
 net/dcsctp/socket/heartbeat_handler.cc        |  4 +--
 .../socket/mock_dcsctp_socket_callbacks.h     | 12 ++++----
 net/dcsctp/socket/packet_sender.cc            |  4 +--
 net/dcsctp/socket/packet_sender.h             |  8 ++---
 net/dcsctp/socket/packet_sender_test.cc       |  5 ++--
 net/dcsctp/socket/state_cookie.cc             |  4 +--
 net/dcsctp/socket/state_cookie.h              |  4 +--
 net/dcsctp/socket/stream_reset_handler.cc     |  4 +--
 net/dcsctp/socket/stream_reset_handler.h      |  4 +--
 .../socket/stream_reset_handler_test.cc       |  1 -
 .../socket/transmission_control_block_test.cc |  5 ++--
 net/dcsctp/tx/BUILD.gn                        |  3 --
 net/dcsctp/tx/outstanding_data.cc             |  8 ++---
 net/dcsctp/tx/outstanding_data.h              | 18 +++++------
 net/dcsctp/tx/retransmission_queue.cc         |  1 -
 net/dcsctp/tx/rr_send_queue.cc                |  6 ++--
 net/dcsctp/tx/rr_send_queue.h                 |  1 -
 167 files changed, 405 insertions(+), 534 deletions(-)

diff --git a/net/dcsctp/fuzzers/BUILD.gn b/net/dcsctp/fuzzers/BUILD.gn
index cd5df5697e..8daabb81e7 100644
--- a/net/dcsctp/fuzzers/BUILD.gn
+++ b/net/dcsctp/fuzzers/BUILD.gn
@@ -11,7 +11,6 @@ import("../../../webrtc.gni")
 rtc_library("dcsctp_fuzzers") {
   testonly = true
   deps = [
-    "../../../api:array_view",
     "../../../api/task_queue",
     "../../../api/units:timestamp",
     "../../../rtc_base:checks",
@@ -38,7 +37,6 @@ if (rtc_include_tests) {
 
     deps = [
       ":dcsctp_fuzzers",
-      "../../../api:array_view",
       "../../../test:test_support",
       "../public:types",
       "../socket:dcsctp_socket",
diff --git a/net/dcsctp/fuzzers/dcsctp_fuzzers.cc b/net/dcsctp/fuzzers/dcsctp_fuzzers.cc
index 3f8c4d1f7b..9a732db440 100644
--- a/net/dcsctp/fuzzers/dcsctp_fuzzers.cc
+++ b/net/dcsctp/fuzzers/dcsctp_fuzzers.cc
@@ -12,11 +12,11 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/chunk/abort_chunk.h"
 #include "net/dcsctp/packet/chunk/chunk.h"
@@ -82,7 +82,7 @@ enum class StartingState : int {
 // State about the current fuzzing iteration
 class FuzzState {
  public:
-  explicit FuzzState(webrtc::ArrayView<const uint8_t> data) : data_(data) {}
+  explicit FuzzState(std::span<const uint8_t> data) : data_(data) {}
 
   uint8_t GetByte() {
     uint8_t value = 0;
@@ -101,7 +101,7 @@ class FuzzState {
  private:
   uint32_t tsn_ = kRandomValue;
   uint32_t mid_ = 0;
-  webrtc::ArrayView<const uint8_t> data_;
+  std::span<const uint8_t> data_;
   size_t offset_ = 0;
 };
 
@@ -419,7 +419,7 @@ std::vector<uint8_t> GeneratePacket(FuzzState& state) {
 
 void FuzzSocket(DcSctpSocketInterface& socket,
                 FuzzerCallbacks& cb,
-                webrtc::ArrayView<const uint8_t> data) {
+                std::span<const uint8_t> data) {
   if (data.size() < kMinInputLength || data.size() > kMaxInputLength) {
     return;
   }
diff --git a/net/dcsctp/fuzzers/dcsctp_fuzzers.h b/net/dcsctp/fuzzers/dcsctp_fuzzers.h
index 62f7b7f0b4..bc4afc682f 100644
--- a/net/dcsctp/fuzzers/dcsctp_fuzzers.h
+++ b/net/dcsctp/fuzzers/dcsctp_fuzzers.h
@@ -17,10 +17,10 @@
 #include <memory>
 #include <optional>
 #include <set>
+#include <span>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/task_queue/task_queue_base.h"
 #include "api/units/timestamp.h"
 #include "net/dcsctp/public/dcsctp_message.h"
@@ -66,7 +66,7 @@ class FuzzerTimeout : public Timeout {
 class FuzzerCallbacks : public DcSctpSocketCallbacks {
  public:
   static constexpr int kRandomValue = 42;
-  void SendPacket(webrtc::ArrayView<const uint8_t> data) override {
+  void SendPacket(std::span<const uint8_t> data) override {
     sent_packets_.emplace_back(std::vector<uint8_t>(data.begin(), data.end()));
   }
   std::unique_ptr<Timeout> CreateTimeout(
@@ -86,13 +86,12 @@ class FuzzerCallbacks : public DcSctpSocketCallbacks {
   void OnConnected() override {}
   void OnClosed() override {}
   void OnConnectionRestarted() override {}
-  void OnStreamsResetFailed(
-      webrtc::ArrayView<const StreamID> /* outgoing_streams */,
-      absl::string_view /* reason */) override {}
+  void OnStreamsResetFailed(std::span<const StreamID> /* outgoing_streams */,
+                            absl::string_view /* reason */) override {}
   void OnStreamsResetPerformed(
-      webrtc::ArrayView<const StreamID> outgoing_streams) override {}
+      std::span<const StreamID> outgoing_streams) override {}
   void OnIncomingStreamsReset(
-      webrtc::ArrayView<const StreamID> incoming_streams) override {}
+      std::span<const StreamID> incoming_streams) override {}
 
   std::vector<uint8_t> ConsumeSentPacket() {
     if (sent_packets_.empty()) {
@@ -125,7 +124,7 @@ class FuzzerCallbacks : public DcSctpSocketCallbacks {
 // API methods.
 void FuzzSocket(DcSctpSocketInterface& socket,
                 FuzzerCallbacks& cb,
-                webrtc::ArrayView<const uint8_t> data);
+                std::span<const uint8_t> data);
 
 }  // namespace dcsctp_fuzzers
 }  // namespace dcsctp
diff --git a/net/dcsctp/fuzzers/dcsctp_fuzzers_test.cc b/net/dcsctp/fuzzers/dcsctp_fuzzers_test.cc
index bbb00d72ba..b1b0d50a6b 100644
--- a/net/dcsctp/fuzzers/dcsctp_fuzzers_test.cc
+++ b/net/dcsctp/fuzzers/dcsctp_fuzzers_test.cc
@@ -11,7 +11,6 @@
 
 #include <cstdint>
 
-#include "api/array_view.h"
 #include "net/dcsctp/public/dcsctp_options.h"
 #include "net/dcsctp/socket/dcsctp_socket.h"
 #include "test/gtest.h"
diff --git a/net/dcsctp/packet/BUILD.gn b/net/dcsctp/packet/BUILD.gn
index 852788b3f5..7b8d0876ac 100644
--- a/net/dcsctp/packet/BUILD.gn
+++ b/net/dcsctp/packet/BUILD.gn
@@ -13,10 +13,7 @@ group("packet") {
 }
 
 rtc_source_set("bounded_io") {
-  deps = [
-    "../../../api:array_view",
-    "../../../rtc_base:checks",
-  ]
+  deps = [ "../../../rtc_base:checks" ]
   sources = [
     "bounded_byte_reader.h",
     "bounded_byte_writer.h",
@@ -26,7 +23,6 @@ rtc_source_set("bounded_io") {
 rtc_library("tlv_trait") {
   deps = [
     ":bounded_io",
-    "../../../api:array_view",
     "../../../rtc_base:logging",
   ]
   sources = [
@@ -46,7 +42,6 @@ rtc_source_set("data") {
 
 rtc_library("crc32c") {
   deps = [
-    "../../../api:array_view",
     "//third_party/abseil-cpp/absl/crc:crc32c",
     "//third_party/abseil-cpp/absl/numeric:bits",
     "//third_party/abseil-cpp/absl/strings:string_view",
@@ -61,7 +56,6 @@ rtc_library("parameter") {
   deps = [
     ":bounded_io",
     ":tlv_trait",
-    "../../../api:array_view",
     "../../../rtc_base:logging",
     "../../../rtc_base:stringutils",
     "../common:internal_types",
@@ -104,7 +98,6 @@ rtc_library("error_cause") {
     ":bounded_io",
     ":parameter",
     ":tlv_trait",
-    "../../../api:array_view",
     "../../../rtc_base:logging",
     "../../../rtc_base:stringutils",
     "../common:internal_types",
@@ -150,7 +143,6 @@ rtc_library("chunk") {
     ":data",
     ":parameter",
     ":tlv_trait",
-    "../../../api:array_view",
     "../../../rtc_base:checks",
     "../../../rtc_base:logging",
     "../../../rtc_base:stringutils",
@@ -219,7 +211,6 @@ rtc_library("sctp_packet") {
     ":bounded_io",
     ":chunk",
     ":crc32c",
-    "../../../api:array_view",
     "../../../rtc_base:checks",
     "../../../rtc_base:logging",
     "../../../rtc_base:stringutils",
@@ -248,7 +239,6 @@ if (rtc_include_tests) {
       ":parameter",
       ":sctp_packet",
       ":tlv_trait",
-      "../../../api:array_view",
       "../../../test:test_support",
       "../common:internal_types",
       "../common:math",
diff --git a/net/dcsctp/packet/bounded_byte_reader.h b/net/dcsctp/packet/bounded_byte_reader.h
index deeb090c9a..4b25bd71ed 100644
--- a/net/dcsctp/packet/bounded_byte_reader.h
+++ b/net/dcsctp/packet/bounded_byte_reader.h
@@ -13,8 +13,8 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "rtc_base/checks.h"
 
 namespace dcsctp {
@@ -39,7 +39,7 @@ inline uint32_t LoadBigEndian32(const uint8_t* data) {
 }
 }  // namespace internal
 
-// BoundedByteReader wraps an ArrayView and divides it into two parts; A fixed
+// BoundedByteReader wraps an std::span and divides it into two parts; A fixed
 // size - which is the template parameter - and a variable size, which is what
 // remains in `data` after the `FixedSize`.
 //
@@ -49,12 +49,11 @@ inline uint32_t LoadBigEndian32(const uint8_t* data) {
 //
 // The variable sized portion can either be used to create sub-readers, which
 // themselves would provide compile-time bounds-checking, or the entire variable
-// sized portion can be retrieved as an ArrayView.
+// sized portion can be retrieved as an std::span.
 template <int FixedSize>
 class BoundedByteReader {
  public:
-  explicit BoundedByteReader(webrtc::ArrayView<const uint8_t> data)
-      : data_(data) {
+  explicit BoundedByteReader(std::span<const uint8_t> data) : data_(data) {
     RTC_CHECK(data.size() >= FixedSize);
   }
 
@@ -82,19 +81,19 @@ class BoundedByteReader {
   BoundedByteReader<SubSize> sub_reader(size_t variable_offset) const {
     RTC_CHECK(FixedSize + variable_offset + SubSize <= data_.size());
 
-    webrtc::ArrayView<const uint8_t> sub_span =
+    std::span<const uint8_t> sub_span =
         data_.subspan(FixedSize + variable_offset, SubSize);
     return BoundedByteReader<SubSize>(sub_span);
   }
 
   size_t variable_data_size() const { return data_.size() - FixedSize; }
 
-  webrtc::ArrayView<const uint8_t> variable_data() const {
+  std::span<const uint8_t> variable_data() const {
     return data_.subspan(FixedSize);
   }
 
  private:
-  const webrtc::ArrayView<const uint8_t> data_;
+  const std::span<const uint8_t> data_;
 };
 
 }  // namespace dcsctp
diff --git a/net/dcsctp/packet/bounded_byte_reader_test.cc b/net/dcsctp/packet/bounded_byte_reader_test.cc
index 6aaeb01a26..9ec798aea0 100644
--- a/net/dcsctp/packet/bounded_byte_reader_test.cc
+++ b/net/dcsctp/packet/bounded_byte_reader_test.cc
@@ -12,7 +12,6 @@
 
 #include <cstdint>
 
-#include "api/array_view.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
 
diff --git a/net/dcsctp/packet/bounded_byte_writer.h b/net/dcsctp/packet/bounded_byte_writer.h
index 8f1ad1e0c6..e124f3300a 100644
--- a/net/dcsctp/packet/bounded_byte_writer.h
+++ b/net/dcsctp/packet/bounded_byte_writer.h
@@ -15,8 +15,8 @@
 #include <cstddef>
 #include <cstdint>
 #include <cstring>
+#include <span>
 
-#include "api/array_view.h"
 #include "rtc_base/checks.h"
 
 namespace dcsctp {
@@ -45,7 +45,7 @@ inline void StoreBigEndian32(uint8_t* data, uint32_t val) {
 }
 }  // namespace internal
 
-// BoundedByteWriter wraps an ArrayView and divides it into two parts; A fixed
+// BoundedByteWriter wraps an std::span and divides it into two parts; A fixed
 // size - which is the template parameter - and a variable size, which is what
 // remains in `data` after the `FixedSize`.
 //
@@ -59,7 +59,7 @@ inline void StoreBigEndian32(uint8_t* data, uint32_t val) {
 template <int FixedSize>
 class BoundedByteWriter {
  public:
-  explicit BoundedByteWriter(webrtc::ArrayView<uint8_t> data) : data_(data) {
+  explicit BoundedByteWriter(std::span<uint8_t> data) : data_(data) {
     RTC_CHECK(data.size() >= FixedSize);
   }
 
@@ -91,7 +91,7 @@ class BoundedByteWriter {
         data_.subspan(FixedSize + variable_offset, SubSize));
   }
 
-  void CopyToVariableData(webrtc::ArrayView<const uint8_t> source) {
+  void CopyToVariableData(std::span<const uint8_t> source) {
     size_t copy_size = std::min(source.size(), data_.size() - FixedSize);
     if (source.data() == nullptr || copy_size == 0) {
       return;
@@ -100,7 +100,7 @@ class BoundedByteWriter {
   }
 
  private:
-  webrtc::ArrayView<uint8_t> data_;
+  std::span<uint8_t> data_;
 };
 }  // namespace dcsctp
 
diff --git a/net/dcsctp/packet/bounded_byte_writer_test.cc b/net/dcsctp/packet/bounded_byte_writer_test.cc
index e1591897bf..8128f768dc 100644
--- a/net/dcsctp/packet/bounded_byte_writer_test.cc
+++ b/net/dcsctp/packet/bounded_byte_writer_test.cc
@@ -13,7 +13,6 @@
 #include <cstdint>
 #include <vector>
 
-#include "api/array_view.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
 
diff --git a/net/dcsctp/packet/chunk/abort_chunk.cc b/net/dcsctp/packet/chunk/abort_chunk.cc
index 47daa56cf2..d15a6ec7c4 100644
--- a/net/dcsctp/packet/chunk/abort_chunk.cc
+++ b/net/dcsctp/packet/chunk/abort_chunk.cc
@@ -11,11 +11,11 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "net/dcsctp/packet/bounded_byte_writer.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
@@ -34,8 +34,7 @@ namespace dcsctp {
 //  \                                                               \
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
-std::optional<AbortChunk> AbortChunk::Parse(
-    webrtc::ArrayView<const uint8_t> data) {
+std::optional<AbortChunk> AbortChunk::Parse(std::span<const uint8_t> data) {
   std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
   if (!reader.has_value()) {
     return std::nullopt;
@@ -51,7 +50,7 @@ std::optional<AbortChunk> AbortChunk::Parse(
 }
 
 void AbortChunk::SerializeTo(std::vector<uint8_t>& out) const {
-  webrtc::ArrayView<const uint8_t> error_causes = error_causes_.data();
+  std::span<const uint8_t> error_causes = error_causes_.data();
   BoundedByteWriter<kHeaderSize> writer = AllocateTLV(out, error_causes.size());
   writer.Store8<1>(filled_in_verification_tag_ ? 0 : (1 << kFlagsBitT));
   writer.CopyToVariableData(error_causes);
diff --git a/net/dcsctp/packet/chunk/abort_chunk.h b/net/dcsctp/packet/chunk/abort_chunk.h
index 772c52087a..c7356d2663 100644
--- a/net/dcsctp/packet/chunk/abort_chunk.h
+++ b/net/dcsctp/packet/chunk/abort_chunk.h
@@ -13,11 +13,11 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/chunk/chunk.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
 #include "net/dcsctp/packet/tlv_trait.h"
@@ -42,7 +42,7 @@ class AbortChunk : public Chunk, public TLVTrait<AbortChunkConfig> {
   AbortChunk(AbortChunk&& other) = default;
   AbortChunk& operator=(AbortChunk&& other) = default;
 
-  static std::optional<AbortChunk> Parse(webrtc::ArrayView<const uint8_t> data);
+  static std::optional<AbortChunk> Parse(std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
diff --git a/net/dcsctp/packet/chunk/chunk.cc b/net/dcsctp/packet/chunk/chunk.cc
index 9332f37c5b..5ef0091af8 100644
--- a/net/dcsctp/packet/chunk/chunk.cc
+++ b/net/dcsctp/packet/chunk/chunk.cc
@@ -11,9 +11,9 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/chunk/abort_chunk.h"
 #include "net/dcsctp/packet/chunk/cookie_ack_chunk.h"
 #include "net/dcsctp/packet/chunk/cookie_echo_chunk.h"
@@ -37,7 +37,7 @@ namespace dcsctp {
 
 template <class Chunk>
 bool ParseAndPrint(uint8_t chunk_type,
-                   webrtc::ArrayView<const uint8_t> data,
+                   std::span<const uint8_t> data,
                    webrtc::StringBuilder& sb) {
   if (chunk_type == Chunk::kType) {
     std::optional<Chunk> c = Chunk::Parse(data);
@@ -51,7 +51,7 @@ bool ParseAndPrint(uint8_t chunk_type,
   return false;
 }
 
-std::string DebugConvertChunkToString(webrtc::ArrayView<const uint8_t> data) {
+std::string DebugConvertChunkToString(std::span<const uint8_t> data) {
   webrtc::StringBuilder sb;
 
   if (data.empty()) {
diff --git a/net/dcsctp/packet/chunk/chunk.h b/net/dcsctp/packet/chunk/chunk.h
index 3eac270ac9..8d5b3c787a 100644
--- a/net/dcsctp/packet/chunk/chunk.h
+++ b/net/dcsctp/packet/chunk/chunk.h
@@ -10,13 +10,11 @@
 #ifndef NET_DCSCTP_PACKET_CHUNK_CHUNK_H_
 #define NET_DCSCTP_PACKET_CHUNK_CHUNK_H_
 
-
 #include <cstdint>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
-
 namespace dcsctp {
 
 // Base class for all SCTP chunks
@@ -40,7 +38,7 @@ class Chunk {
 
 // Introspects the chunk in `data` and returns a human readable textual
 // representation of it, to be used in debugging.
-std::string DebugConvertChunkToString(webrtc::ArrayView<const uint8_t> data);
+std::string DebugConvertChunkToString(std::span<const uint8_t> data);
 
 struct ChunkConfig {
   static constexpr int kTypeSizeInBytes = 1;
diff --git a/net/dcsctp/packet/chunk/cookie_ack_chunk.cc b/net/dcsctp/packet/chunk/cookie_ack_chunk.cc
index b01ded9acc..781bcf293c 100644
--- a/net/dcsctp/packet/chunk/cookie_ack_chunk.cc
+++ b/net/dcsctp/packet/chunk/cookie_ack_chunk.cc
@@ -11,11 +11,10 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
-
 namespace dcsctp {
 
 // https://tools.ietf.org/html/rfc4960#section-3.3.12
@@ -27,7 +26,7 @@ namespace dcsctp {
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
 std::optional<CookieAckChunk> CookieAckChunk::Parse(
-    webrtc::ArrayView<const uint8_t> data) {
+    std::span<const uint8_t> data) {
   if (!ParseTLV(data).has_value()) {
     return std::nullopt;
   }
diff --git a/net/dcsctp/packet/chunk/cookie_ack_chunk.h b/net/dcsctp/packet/chunk/cookie_ack_chunk.h
index 872757c73b..d670af9d21 100644
--- a/net/dcsctp/packet/chunk/cookie_ack_chunk.h
+++ b/net/dcsctp/packet/chunk/cookie_ack_chunk.h
@@ -13,10 +13,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/chunk/chunk.h"
 #include "net/dcsctp/packet/tlv_trait.h"
 
@@ -35,8 +35,7 @@ class CookieAckChunk : public Chunk, public TLVTrait<CookieAckChunkConfig> {
 
   CookieAckChunk() {}
 
-  static std::optional<CookieAckChunk> Parse(
-      webrtc::ArrayView<const uint8_t> data);
+  static std::optional<CookieAckChunk> Parse(std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
diff --git a/net/dcsctp/packet/chunk/cookie_echo_chunk.cc b/net/dcsctp/packet/chunk/cookie_echo_chunk.cc
index 4e5a80d389..4cf309f288 100644
--- a/net/dcsctp/packet/chunk/cookie_echo_chunk.cc
+++ b/net/dcsctp/packet/chunk/cookie_echo_chunk.cc
@@ -11,10 +11,10 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "net/dcsctp/packet/bounded_byte_writer.h"
 
@@ -32,7 +32,7 @@ namespace dcsctp {
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
 std::optional<CookieEchoChunk> CookieEchoChunk::Parse(
-    webrtc::ArrayView<const uint8_t> data) {
+    std::span<const uint8_t> data) {
   std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
   if (!reader.has_value()) {
     return std::nullopt;
diff --git a/net/dcsctp/packet/chunk/cookie_echo_chunk.h b/net/dcsctp/packet/chunk/cookie_echo_chunk.h
index 94cb10435c..67cee7a5fe 100644
--- a/net/dcsctp/packet/chunk/cookie_echo_chunk.h
+++ b/net/dcsctp/packet/chunk/cookie_echo_chunk.h
@@ -13,10 +13,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/chunk/chunk.h"
 #include "net/dcsctp/packet/tlv_trait.h"
 
@@ -33,16 +33,15 @@ class CookieEchoChunk : public Chunk, public TLVTrait<CookieEchoChunkConfig> {
  public:
   static constexpr int kType = CookieEchoChunkConfig::kType;
 
-  explicit CookieEchoChunk(webrtc::ArrayView<const uint8_t> cookie)
+  explicit CookieEchoChunk(std::span<const uint8_t> cookie)
       : cookie_(cookie.begin(), cookie.end()) {}
 
-  static std::optional<CookieEchoChunk> Parse(
-      webrtc::ArrayView<const uint8_t> data);
+  static std::optional<CookieEchoChunk> Parse(std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
 
-  webrtc::ArrayView<const uint8_t> cookie() const { return cookie_; }
+  std::span<const uint8_t> cookie() const { return cookie_; }
 
  private:
   std::vector<uint8_t> cookie_;
diff --git a/net/dcsctp/packet/chunk/cookie_echo_chunk_test.cc b/net/dcsctp/packet/chunk/cookie_echo_chunk_test.cc
index 9f3f24cdf1..2a120c5ce8 100644
--- a/net/dcsctp/packet/chunk/cookie_echo_chunk_test.cc
+++ b/net/dcsctp/packet/chunk/cookie_echo_chunk_test.cc
@@ -12,7 +12,6 @@
 #include <cstdint>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/testing/testing_macros.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
diff --git a/net/dcsctp/packet/chunk/data_chunk.cc b/net/dcsctp/packet/chunk/data_chunk.cc
index 80da4eba79..60ffd85f8b 100644
--- a/net/dcsctp/packet/chunk/data_chunk.cc
+++ b/net/dcsctp/packet/chunk/data_chunk.cc
@@ -11,10 +11,10 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -43,8 +43,7 @@ namespace dcsctp {
 //  \                                                               \
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
-std::optional<DataChunk> DataChunk::Parse(
-    webrtc::ArrayView<const uint8_t> data) {
+std::optional<DataChunk> DataChunk::Parse(std::span<const uint8_t> data) {
   std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
   if (!reader.has_value()) {
     return std::nullopt;
diff --git a/net/dcsctp/packet/chunk/data_chunk.h b/net/dcsctp/packet/chunk/data_chunk.h
index 54bcdfb9aa..cfd5acab55 100644
--- a/net/dcsctp/packet/chunk/data_chunk.h
+++ b/net/dcsctp/packet/chunk/data_chunk.h
@@ -13,11 +13,11 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/chunk/chunk.h"
 #include "net/dcsctp/packet/chunk/data_common.h"
@@ -60,7 +60,7 @@ class DataChunk : public AnyDataChunk, public TLVTrait<DataChunkConfig> {
   DataChunk(TSN tsn, Data&& data, bool immediate_ack)
       : AnyDataChunk(tsn, std::move(data), immediate_ack) {}
 
-  static std::optional<DataChunk> Parse(webrtc::ArrayView<const uint8_t> data);
+  static std::optional<DataChunk> Parse(std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
diff --git a/net/dcsctp/packet/chunk/data_chunk_test.cc b/net/dcsctp/packet/chunk/data_chunk_test.cc
index e8791f74fa..4f3d37f55f 100644
--- a/net/dcsctp/packet/chunk/data_chunk_test.cc
+++ b/net/dcsctp/packet/chunk/data_chunk_test.cc
@@ -12,7 +12,6 @@
 #include <cstdint>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/public/types.h"
 #include "net/dcsctp/testing/testing_macros.h"
diff --git a/net/dcsctp/packet/chunk/data_common.h b/net/dcsctp/packet/chunk/data_common.h
index 5ded1e8879..361745131f 100644
--- a/net/dcsctp/packet/chunk/data_common.h
+++ b/net/dcsctp/packet/chunk/data_common.h
@@ -11,10 +11,10 @@
 #define NET_DCSCTP_PACKET_CHUNK_DATA_COMMON_H_
 
 #include <cstdint>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/chunk/chunk.h"
 #include "net/dcsctp/packet/data.h"
@@ -54,7 +54,7 @@ class AnyDataChunk : public Chunk {
   MID mid() const { return data_.mid; }
   FSN fsn() const { return data_.fsn; }
   PPID ppid() const { return data_.ppid; }
-  webrtc::ArrayView<const uint8_t> payload() const { return data_.payload; }
+  std::span<const uint8_t> payload() const { return data_.payload; }
 
   // Extracts the Data from the chunk, as a destructive action.
   Data extract() && { return std::move(data_); }
diff --git a/net/dcsctp/packet/chunk/error_chunk.cc b/net/dcsctp/packet/chunk/error_chunk.cc
index 90d1dc0259..ea48248c7e 100644
--- a/net/dcsctp/packet/chunk/error_chunk.cc
+++ b/net/dcsctp/packet/chunk/error_chunk.cc
@@ -11,11 +11,11 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "net/dcsctp/packet/bounded_byte_writer.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
@@ -34,8 +34,7 @@ namespace dcsctp {
 //  \                                                               \
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
-std::optional<ErrorChunk> ErrorChunk::Parse(
-    webrtc::ArrayView<const uint8_t> data) {
+std::optional<ErrorChunk> ErrorChunk::Parse(std::span<const uint8_t> data) {
   std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
   if (!reader.has_value()) {
     return std::nullopt;
@@ -49,7 +48,7 @@ std::optional<ErrorChunk> ErrorChunk::Parse(
 }
 
 void ErrorChunk::SerializeTo(std::vector<uint8_t>& out) const {
-  webrtc::ArrayView<const uint8_t> error_causes = error_causes_.data();
+  std::span<const uint8_t> error_causes = error_causes_.data();
   BoundedByteWriter<kHeaderSize> writer = AllocateTLV(out, error_causes.size());
   writer.CopyToVariableData(error_causes);
 }
diff --git a/net/dcsctp/packet/chunk/error_chunk.h b/net/dcsctp/packet/chunk/error_chunk.h
index b395b6b1c2..ed0d4c1087 100644
--- a/net/dcsctp/packet/chunk/error_chunk.h
+++ b/net/dcsctp/packet/chunk/error_chunk.h
@@ -13,11 +13,11 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/chunk/chunk.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
 #include "net/dcsctp/packet/tlv_trait.h"
@@ -41,7 +41,7 @@ class ErrorChunk : public Chunk, public TLVTrait<ErrorChunkConfig> {
   ErrorChunk(ErrorChunk&& other) = default;
   ErrorChunk& operator=(ErrorChunk&& other) = default;
 
-  static std::optional<ErrorChunk> Parse(webrtc::ArrayView<const uint8_t> data);
+  static std::optional<ErrorChunk> Parse(std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
diff --git a/net/dcsctp/packet/chunk/error_chunk_test.cc b/net/dcsctp/packet/chunk/error_chunk_test.cc
index 0d034da2b2..25b9434005 100644
--- a/net/dcsctp/packet/chunk/error_chunk_test.cc
+++ b/net/dcsctp/packet/chunk/error_chunk_test.cc
@@ -12,7 +12,6 @@
 #include <cstdint>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/error_cause/unrecognized_chunk_type_cause.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
 #include "net/dcsctp/testing/testing_macros.h"
diff --git a/net/dcsctp/packet/chunk/forward_tsn_chunk.cc b/net/dcsctp/packet/chunk/forward_tsn_chunk.cc
index dbbfa60ccc..dcf3f04475 100644
--- a/net/dcsctp/packet/chunk/forward_tsn_chunk.cc
+++ b/net/dcsctp/packet/chunk/forward_tsn_chunk.cc
@@ -12,11 +12,11 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -44,7 +44,7 @@ namespace dcsctp {
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
 std::optional<ForwardTsnChunk> ForwardTsnChunk::Parse(
-    webrtc::ArrayView<const uint8_t> data) {
+    std::span<const uint8_t> data) {
   std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
   if (!reader.has_value()) {
     return std::nullopt;
@@ -69,7 +69,7 @@ std::optional<ForwardTsnChunk> ForwardTsnChunk::Parse(
 }
 
 void ForwardTsnChunk::SerializeTo(std::vector<uint8_t>& out) const {
-  webrtc::ArrayView<const SkippedStream> skipped = skipped_streams();
+  std::span<const SkippedStream> skipped = skipped_streams();
   size_t variable_size = skipped.size() * kSkippedStreamBufferSize;
   BoundedByteWriter<kHeaderSize> writer = AllocateTLV(out, variable_size);
 
diff --git a/net/dcsctp/packet/chunk/forward_tsn_chunk.h b/net/dcsctp/packet/chunk/forward_tsn_chunk.h
index b65d2ff124..6260456cad 100644
--- a/net/dcsctp/packet/chunk/forward_tsn_chunk.h
+++ b/net/dcsctp/packet/chunk/forward_tsn_chunk.h
@@ -13,11 +13,11 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/chunk/chunk.h"
 #include "net/dcsctp/packet/chunk/forward_tsn_common.h"
@@ -41,8 +41,7 @@ class ForwardTsnChunk : public AnyForwardTsnChunk,
                   std::vector<SkippedStream> skipped_streams)
       : AnyForwardTsnChunk(new_cumulative_tsn, std::move(skipped_streams)) {}
 
-  static std::optional<ForwardTsnChunk> Parse(
-      webrtc::ArrayView<const uint8_t> data);
+  static std::optional<ForwardTsnChunk> Parse(std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
diff --git a/net/dcsctp/packet/chunk/forward_tsn_chunk_test.cc b/net/dcsctp/packet/chunk/forward_tsn_chunk_test.cc
index 982e7ce229..1f8ed5a216 100644
--- a/net/dcsctp/packet/chunk/forward_tsn_chunk_test.cc
+++ b/net/dcsctp/packet/chunk/forward_tsn_chunk_test.cc
@@ -10,9 +10,9 @@
 #include "net/dcsctp/packet/chunk/forward_tsn_chunk.h"
 
 #include <cstdint>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/chunk/forward_tsn_common.h"
 #include "net/dcsctp/public/types.h"
diff --git a/net/dcsctp/packet/chunk/forward_tsn_common.h b/net/dcsctp/packet/chunk/forward_tsn_common.h
index b2e9168835..082d2e0ed0 100644
--- a/net/dcsctp/packet/chunk/forward_tsn_common.h
+++ b/net/dcsctp/packet/chunk/forward_tsn_common.h
@@ -10,10 +10,10 @@
 #ifndef NET_DCSCTP_PACKET_CHUNK_FORWARD_TSN_COMMON_H_
 #define NET_DCSCTP_PACKET_CHUNK_FORWARD_TSN_COMMON_H_
 
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/chunk/chunk.h"
 #include "net/dcsctp/public/types.h"
@@ -51,7 +51,7 @@ class AnyForwardTsnChunk : public Chunk {
 
   TSN new_cumulative_tsn() const { return new_cumulative_tsn_; }
 
-  webrtc::ArrayView<const SkippedStream> skipped_streams() const {
+  std::span<const SkippedStream> skipped_streams() const {
     return skipped_streams_;
   }
 
diff --git a/net/dcsctp/packet/chunk/heartbeat_ack_chunk.cc b/net/dcsctp/packet/chunk/heartbeat_ack_chunk.cc
index 0de304016b..842447f8cb 100644
--- a/net/dcsctp/packet/chunk/heartbeat_ack_chunk.cc
+++ b/net/dcsctp/packet/chunk/heartbeat_ack_chunk.cc
@@ -11,11 +11,11 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "net/dcsctp/packet/bounded_byte_writer.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
@@ -35,7 +35,7 @@ namespace dcsctp {
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
 std::optional<HeartbeatAckChunk> HeartbeatAckChunk::Parse(
-    webrtc::ArrayView<const uint8_t> data) {
+    std::span<const uint8_t> data) {
   std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
   if (!reader.has_value()) {
     return std::nullopt;
@@ -50,7 +50,7 @@ std::optional<HeartbeatAckChunk> HeartbeatAckChunk::Parse(
 }
 
 void HeartbeatAckChunk::SerializeTo(std::vector<uint8_t>& out) const {
-  webrtc::ArrayView<const uint8_t> parameters = parameters_.data();
+  std::span<const uint8_t> parameters = parameters_.data();
   BoundedByteWriter<kHeaderSize> writer = AllocateTLV(out, parameters.size());
   writer.CopyToVariableData(parameters);
 }
diff --git a/net/dcsctp/packet/chunk/heartbeat_ack_chunk.h b/net/dcsctp/packet/chunk/heartbeat_ack_chunk.h
index 3e6ecd4e0f..319a12f70f 100644
--- a/net/dcsctp/packet/chunk/heartbeat_ack_chunk.h
+++ b/net/dcsctp/packet/chunk/heartbeat_ack_chunk.h
@@ -13,11 +13,11 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/chunk/chunk.h"
 #include "net/dcsctp/packet/parameter/heartbeat_info_parameter.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
@@ -43,8 +43,7 @@ class HeartbeatAckChunk : public Chunk,
   HeartbeatAckChunk(HeartbeatAckChunk&& other) = default;
   HeartbeatAckChunk& operator=(HeartbeatAckChunk&& other) = default;
 
-  static std::optional<HeartbeatAckChunk> Parse(
-      webrtc::ArrayView<const uint8_t> data);
+  static std::optional<HeartbeatAckChunk> Parse(std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
diff --git a/net/dcsctp/packet/chunk/heartbeat_ack_chunk_test.cc b/net/dcsctp/packet/chunk/heartbeat_ack_chunk_test.cc
index 6e41e7b272..9d5ae63846 100644
--- a/net/dcsctp/packet/chunk/heartbeat_ack_chunk_test.cc
+++ b/net/dcsctp/packet/chunk/heartbeat_ack_chunk_test.cc
@@ -13,7 +13,6 @@
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/parameter/heartbeat_info_parameter.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
 #include "net/dcsctp/testing/testing_macros.h"
diff --git a/net/dcsctp/packet/chunk/heartbeat_request_chunk.cc b/net/dcsctp/packet/chunk/heartbeat_request_chunk.cc
index 4b76fff7e8..eca9bac6d3 100644
--- a/net/dcsctp/packet/chunk/heartbeat_request_chunk.cc
+++ b/net/dcsctp/packet/chunk/heartbeat_request_chunk.cc
@@ -11,11 +11,11 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "net/dcsctp/packet/bounded_byte_writer.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
@@ -35,7 +35,7 @@ namespace dcsctp {
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
 std::optional<HeartbeatRequestChunk> HeartbeatRequestChunk::Parse(
-    webrtc::ArrayView<const uint8_t> data) {
+    std::span<const uint8_t> data) {
   std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
   if (!reader.has_value()) {
     return std::nullopt;
@@ -50,7 +50,7 @@ std::optional<HeartbeatRequestChunk> HeartbeatRequestChunk::Parse(
 }
 
 void HeartbeatRequestChunk::SerializeTo(std::vector<uint8_t>& out) const {
-  webrtc::ArrayView<const uint8_t> parameters = parameters_.data();
+  std::span<const uint8_t> parameters = parameters_.data();
   BoundedByteWriter<kHeaderSize> writer = AllocateTLV(out, parameters.size());
   writer.CopyToVariableData(parameters);
 }
diff --git a/net/dcsctp/packet/chunk/heartbeat_request_chunk.h b/net/dcsctp/packet/chunk/heartbeat_request_chunk.h
index 31eefbcb88..04798fa1de 100644
--- a/net/dcsctp/packet/chunk/heartbeat_request_chunk.h
+++ b/net/dcsctp/packet/chunk/heartbeat_request_chunk.h
@@ -13,11 +13,11 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/chunk/chunk.h"
 #include "net/dcsctp/packet/parameter/heartbeat_info_parameter.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
@@ -43,7 +43,7 @@ class HeartbeatRequestChunk : public Chunk,
   HeartbeatRequestChunk& operator=(HeartbeatRequestChunk&& other) = default;
 
   static std::optional<HeartbeatRequestChunk> Parse(
-      webrtc::ArrayView<const uint8_t> data);
+      std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
diff --git a/net/dcsctp/packet/chunk/heartbeat_request_chunk_test.cc b/net/dcsctp/packet/chunk/heartbeat_request_chunk_test.cc
index ab50025550..5701349c38 100644
--- a/net/dcsctp/packet/chunk/heartbeat_request_chunk_test.cc
+++ b/net/dcsctp/packet/chunk/heartbeat_request_chunk_test.cc
@@ -13,7 +13,6 @@
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/parameter/heartbeat_info_parameter.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
 #include "net/dcsctp/testing/testing_macros.h"
diff --git a/net/dcsctp/packet/chunk/idata_chunk.cc b/net/dcsctp/packet/chunk/idata_chunk.cc
index 7f0fce7a33..815c058af6 100644
--- a/net/dcsctp/packet/chunk/idata_chunk.cc
+++ b/net/dcsctp/packet/chunk/idata_chunk.cc
@@ -11,10 +11,10 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -45,8 +45,7 @@ namespace dcsctp {
 //  \                                                               \
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
-std::optional<IDataChunk> IDataChunk::Parse(
-    webrtc::ArrayView<const uint8_t> data) {
+std::optional<IDataChunk> IDataChunk::Parse(std::span<const uint8_t> data) {
   std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
   if (!reader.has_value()) {
     return std::nullopt;
diff --git a/net/dcsctp/packet/chunk/idata_chunk.h b/net/dcsctp/packet/chunk/idata_chunk.h
index 23b38ef645..d99dcd9ded 100644
--- a/net/dcsctp/packet/chunk/idata_chunk.h
+++ b/net/dcsctp/packet/chunk/idata_chunk.h
@@ -13,11 +13,11 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/chunk/chunk.h"
 #include "net/dcsctp/packet/chunk/data_common.h"
@@ -60,7 +60,7 @@ class IDataChunk : public AnyDataChunk, public TLVTrait<IDataChunkConfig> {
   explicit IDataChunk(TSN tsn, Data&& data, bool immediate_ack)
       : AnyDataChunk(tsn, std::move(data), immediate_ack) {}
 
-  static std::optional<IDataChunk> Parse(webrtc::ArrayView<const uint8_t> data);
+  static std::optional<IDataChunk> Parse(std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
diff --git a/net/dcsctp/packet/chunk/idata_chunk_test.cc b/net/dcsctp/packet/chunk/idata_chunk_test.cc
index d65b31b925..73c9480884 100644
--- a/net/dcsctp/packet/chunk/idata_chunk_test.cc
+++ b/net/dcsctp/packet/chunk/idata_chunk_test.cc
@@ -12,7 +12,6 @@
 #include <cstdint>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/data.h"
 #include "net/dcsctp/public/types.h"
diff --git a/net/dcsctp/packet/chunk/iforward_tsn_chunk.cc b/net/dcsctp/packet/chunk/iforward_tsn_chunk.cc
index 47c0304a11..1d8fdab936 100644
--- a/net/dcsctp/packet/chunk/iforward_tsn_chunk.cc
+++ b/net/dcsctp/packet/chunk/iforward_tsn_chunk.cc
@@ -12,11 +12,11 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -49,7 +49,7 @@ namespace dcsctp {
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
 std::optional<IForwardTsnChunk> IForwardTsnChunk::Parse(
-    webrtc::ArrayView<const uint8_t> data) {
+    std::span<const uint8_t> data) {
   std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
   if (!reader.has_value()) {
     return std::nullopt;
@@ -77,7 +77,7 @@ std::optional<IForwardTsnChunk> IForwardTsnChunk::Parse(
 }
 
 void IForwardTsnChunk::SerializeTo(std::vector<uint8_t>& out) const {
-  webrtc::ArrayView<const SkippedStream> skipped = skipped_streams();
+  std::span<const SkippedStream> skipped = skipped_streams();
   size_t variable_size = skipped.size() * kSkippedStreamBufferSize;
   BoundedByteWriter<kHeaderSize> writer = AllocateTLV(out, variable_size);
 
diff --git a/net/dcsctp/packet/chunk/iforward_tsn_chunk.h b/net/dcsctp/packet/chunk/iforward_tsn_chunk.h
index ae60fd71eb..b6e114c2c3 100644
--- a/net/dcsctp/packet/chunk/iforward_tsn_chunk.h
+++ b/net/dcsctp/packet/chunk/iforward_tsn_chunk.h
@@ -13,11 +13,11 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/chunk/chunk.h"
 #include "net/dcsctp/packet/chunk/forward_tsn_common.h"
@@ -41,8 +41,7 @@ class IForwardTsnChunk : public AnyForwardTsnChunk,
                    std::vector<SkippedStream> skipped_streams)
       : AnyForwardTsnChunk(new_cumulative_tsn, std::move(skipped_streams)) {}
 
-  static std::optional<IForwardTsnChunk> Parse(
-      webrtc::ArrayView<const uint8_t> data);
+  static std::optional<IForwardTsnChunk> Parse(std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
diff --git a/net/dcsctp/packet/chunk/iforward_tsn_chunk_test.cc b/net/dcsctp/packet/chunk/iforward_tsn_chunk_test.cc
index d7f3479242..909304a3d0 100644
--- a/net/dcsctp/packet/chunk/iforward_tsn_chunk_test.cc
+++ b/net/dcsctp/packet/chunk/iforward_tsn_chunk_test.cc
@@ -12,7 +12,6 @@
 #include <cstdint>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/chunk/forward_tsn_common.h"
 #include "net/dcsctp/public/types.h"
diff --git a/net/dcsctp/packet/chunk/init_ack_chunk.cc b/net/dcsctp/packet/chunk/init_ack_chunk.cc
index ce837f7908..2c6f4f7736 100644
--- a/net/dcsctp/packet/chunk/init_ack_chunk.cc
+++ b/net/dcsctp/packet/chunk/init_ack_chunk.cc
@@ -11,11 +11,11 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -44,8 +44,7 @@ namespace dcsctp {
 //  \                                                               \
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
-std::optional<InitAckChunk> InitAckChunk::Parse(
-    webrtc::ArrayView<const uint8_t> data) {
+std::optional<InitAckChunk> InitAckChunk::Parse(std::span<const uint8_t> data) {
   std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
   if (!reader.has_value()) {
     return std::nullopt;
@@ -66,7 +65,7 @@ std::optional<InitAckChunk> InitAckChunk::Parse(
 }
 
 void InitAckChunk::SerializeTo(std::vector<uint8_t>& out) const {
-  webrtc::ArrayView<const uint8_t> parameters = parameters_.data();
+  std::span<const uint8_t> parameters = parameters_.data();
   BoundedByteWriter<kHeaderSize> writer = AllocateTLV(out, parameters.size());
 
   writer.Store32<4>(*initiate_tag_);
diff --git a/net/dcsctp/packet/chunk/init_ack_chunk.h b/net/dcsctp/packet/chunk/init_ack_chunk.h
index 19baea6776..4220a80df9 100644
--- a/net/dcsctp/packet/chunk/init_ack_chunk.h
+++ b/net/dcsctp/packet/chunk/init_ack_chunk.h
@@ -13,11 +13,11 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/chunk/chunk.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
@@ -52,8 +52,7 @@ class InitAckChunk : public Chunk, public TLVTrait<InitAckChunkConfig> {
   InitAckChunk(InitAckChunk&& other) = default;
   InitAckChunk& operator=(InitAckChunk&& other) = default;
 
-  static std::optional<InitAckChunk> Parse(
-      webrtc::ArrayView<const uint8_t> data);
+  static std::optional<InitAckChunk> Parse(std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
diff --git a/net/dcsctp/packet/chunk/init_ack_chunk_test.cc b/net/dcsctp/packet/chunk/init_ack_chunk_test.cc
index 2741d6757b..11bfa7cc07 100644
--- a/net/dcsctp/packet/chunk/init_ack_chunk_test.cc
+++ b/net/dcsctp/packet/chunk/init_ack_chunk_test.cc
@@ -13,7 +13,6 @@
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/parameter/forward_tsn_supported_parameter.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
diff --git a/net/dcsctp/packet/chunk/init_chunk.cc b/net/dcsctp/packet/chunk/init_chunk.cc
index f08e35b331..9e29df276f 100644
--- a/net/dcsctp/packet/chunk/init_chunk.cc
+++ b/net/dcsctp/packet/chunk/init_chunk.cc
@@ -11,11 +11,11 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -44,8 +44,7 @@ namespace dcsctp {
 //  \                                                               \
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
-std::optional<InitChunk> InitChunk::Parse(
-    webrtc::ArrayView<const uint8_t> data) {
+std::optional<InitChunk> InitChunk::Parse(std::span<const uint8_t> data) {
   std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
   if (!reader.has_value()) {
     return std::nullopt;
@@ -67,7 +66,7 @@ std::optional<InitChunk> InitChunk::Parse(
 }
 
 void InitChunk::SerializeTo(std::vector<uint8_t>& out) const {
-  webrtc::ArrayView<const uint8_t> parameters = parameters_.data();
+  std::span<const uint8_t> parameters = parameters_.data();
   BoundedByteWriter<kHeaderSize> writer = AllocateTLV(out, parameters.size());
 
   writer.Store32<4>(*initiate_tag_);
diff --git a/net/dcsctp/packet/chunk/init_chunk.h b/net/dcsctp/packet/chunk/init_chunk.h
index d86ca4b534..2b40691bf2 100644
--- a/net/dcsctp/packet/chunk/init_chunk.h
+++ b/net/dcsctp/packet/chunk/init_chunk.h
@@ -13,11 +13,11 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/chunk/chunk.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
@@ -52,7 +52,7 @@ class InitChunk : public Chunk, public TLVTrait<InitChunkConfig> {
   InitChunk(InitChunk&& other) = default;
   InitChunk& operator=(InitChunk&& other) = default;
 
-  static std::optional<InitChunk> Parse(webrtc::ArrayView<const uint8_t> data);
+  static std::optional<InitChunk> Parse(std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
diff --git a/net/dcsctp/packet/chunk/reconfig_chunk.cc b/net/dcsctp/packet/chunk/reconfig_chunk.cc
index ce5d630666..5440016d77 100644
--- a/net/dcsctp/packet/chunk/reconfig_chunk.cc
+++ b/net/dcsctp/packet/chunk/reconfig_chunk.cc
@@ -11,11 +11,11 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "net/dcsctp/packet/bounded_byte_writer.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
@@ -39,7 +39,7 @@ namespace dcsctp {
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
 std::optional<ReConfigChunk> ReConfigChunk::Parse(
-    webrtc::ArrayView<const uint8_t> data) {
+    std::span<const uint8_t> data) {
   std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
   if (!reader.has_value()) {
     return std::nullopt;
@@ -55,7 +55,7 @@ std::optional<ReConfigChunk> ReConfigChunk::Parse(
 }
 
 void ReConfigChunk::SerializeTo(std::vector<uint8_t>& out) const {
-  webrtc::ArrayView<const uint8_t> parameters = parameters_.data();
+  std::span<const uint8_t> parameters = parameters_.data();
   BoundedByteWriter<kHeaderSize> writer = AllocateTLV(out, parameters.size());
   writer.CopyToVariableData(parameters);
 }
diff --git a/net/dcsctp/packet/chunk/reconfig_chunk.h b/net/dcsctp/packet/chunk/reconfig_chunk.h
index 227a1d2991..bf8cbcdc43 100644
--- a/net/dcsctp/packet/chunk/reconfig_chunk.h
+++ b/net/dcsctp/packet/chunk/reconfig_chunk.h
@@ -13,11 +13,11 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/chunk/chunk.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
 #include "net/dcsctp/packet/tlv_trait.h"
@@ -38,8 +38,7 @@ class ReConfigChunk : public Chunk, public TLVTrait<ReConfigChunkConfig> {
   explicit ReConfigChunk(Parameters parameters)
       : parameters_(std::move(parameters)) {}
 
-  static std::optional<ReConfigChunk> Parse(
-      webrtc::ArrayView<const uint8_t> data);
+  static std::optional<ReConfigChunk> Parse(std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
diff --git a/net/dcsctp/packet/chunk/reconfig_chunk_test.cc b/net/dcsctp/packet/chunk/reconfig_chunk_test.cc
index 296f7ec5fe..261eb6306d 100644
--- a/net/dcsctp/packet/chunk/reconfig_chunk_test.cc
+++ b/net/dcsctp/packet/chunk/reconfig_chunk_test.cc
@@ -12,7 +12,6 @@
 #include <cstdint>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
diff --git a/net/dcsctp/packet/chunk/sack_chunk.cc b/net/dcsctp/packet/chunk/sack_chunk.cc
index ef77331bbd..906ebdb507 100644
--- a/net/dcsctp/packet/chunk/sack_chunk.cc
+++ b/net/dcsctp/packet/chunk/sack_chunk.cc
@@ -13,10 +13,10 @@
 #include <cstdint>
 #include <optional>
 #include <set>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -57,8 +57,7 @@ namespace dcsctp {
 //  |                       Duplicate TSN X                         |
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
-std::optional<SackChunk> SackChunk::Parse(
-    webrtc::ArrayView<const uint8_t> data) {
+std::optional<SackChunk> SackChunk::Parse(std::span<const uint8_t> data) {
   std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
   if (!reader.has_value()) {
     return std::nullopt;
diff --git a/net/dcsctp/packet/chunk/sack_chunk.h b/net/dcsctp/packet/chunk/sack_chunk.h
index 026d52d8c1..7410e85ac0 100644
--- a/net/dcsctp/packet/chunk/sack_chunk.h
+++ b/net/dcsctp/packet/chunk/sack_chunk.h
@@ -14,11 +14,11 @@
 #include <cstdint>
 #include <optional>
 #include <set>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/chunk/chunk.h"
 #include "net/dcsctp/packet/tlv_trait.h"
@@ -55,14 +55,14 @@ class SackChunk : public Chunk, public TLVTrait<SackChunkConfig> {
         a_rwnd_(a_rwnd),
         gap_ack_blocks_(std::move(gap_ack_blocks)),
         duplicate_tsns_(std::move(duplicate_tsns)) {}
-  static std::optional<SackChunk> Parse(webrtc::ArrayView<const uint8_t> data);
+  static std::optional<SackChunk> Parse(std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
 
   TSN cumulative_tsn_ack() const { return cumulative_tsn_ack_; }
   uint32_t a_rwnd() const { return a_rwnd_; }
-  webrtc::ArrayView<const GapAckBlock> gap_ack_blocks() const {
+  std::span<const GapAckBlock> gap_ack_blocks() const {
     return gap_ack_blocks_;
   }
   const std::set<TSN>& duplicate_tsns() const { return duplicate_tsns_; }
diff --git a/net/dcsctp/packet/chunk/sack_chunk_test.cc b/net/dcsctp/packet/chunk/sack_chunk_test.cc
index 2f5c0d0931..70fd96b6a8 100644
--- a/net/dcsctp/packet/chunk/sack_chunk_test.cc
+++ b/net/dcsctp/packet/chunk/sack_chunk_test.cc
@@ -12,7 +12,6 @@
 #include <cstdint>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/testing/testing_macros.h"
 #include "test/gmock.h"
diff --git a/net/dcsctp/packet/chunk/shutdown_ack_chunk.cc b/net/dcsctp/packet/chunk/shutdown_ack_chunk.cc
index f78276d079..2bedc71c73 100644
--- a/net/dcsctp/packet/chunk/shutdown_ack_chunk.cc
+++ b/net/dcsctp/packet/chunk/shutdown_ack_chunk.cc
@@ -11,11 +11,10 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
-
 namespace dcsctp {
 
 // https://tools.ietf.org/html/rfc4960#section-3.3.9
@@ -27,7 +26,7 @@ namespace dcsctp {
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
 std::optional<ShutdownAckChunk> ShutdownAckChunk::Parse(
-    webrtc::ArrayView<const uint8_t> data) {
+    std::span<const uint8_t> data) {
   if (!ParseTLV(data).has_value()) {
     return std::nullopt;
   }
diff --git a/net/dcsctp/packet/chunk/shutdown_ack_chunk.h b/net/dcsctp/packet/chunk/shutdown_ack_chunk.h
index 18550f85a9..07da0a4602 100644
--- a/net/dcsctp/packet/chunk/shutdown_ack_chunk.h
+++ b/net/dcsctp/packet/chunk/shutdown_ack_chunk.h
@@ -13,10 +13,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/chunk/chunk.h"
 #include "net/dcsctp/packet/tlv_trait.h"
 
@@ -35,8 +35,7 @@ class ShutdownAckChunk : public Chunk, public TLVTrait<ShutdownAckChunkConfig> {
 
   ShutdownAckChunk() {}
 
-  static std::optional<ShutdownAckChunk> Parse(
-      webrtc::ArrayView<const uint8_t> data);
+  static std::optional<ShutdownAckChunk> Parse(std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
diff --git a/net/dcsctp/packet/chunk/shutdown_chunk.cc b/net/dcsctp/packet/chunk/shutdown_chunk.cc
index a205f0d97e..26ac869a8a 100644
--- a/net/dcsctp/packet/chunk/shutdown_chunk.cc
+++ b/net/dcsctp/packet/chunk/shutdown_chunk.cc
@@ -11,10 +11,10 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -32,7 +32,7 @@ namespace dcsctp {
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
 std::optional<ShutdownChunk> ShutdownChunk::Parse(
-    webrtc::ArrayView<const uint8_t> data) {
+    std::span<const uint8_t> data) {
   std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
   if (!reader.has_value()) {
     return std::nullopt;
diff --git a/net/dcsctp/packet/chunk/shutdown_chunk.h b/net/dcsctp/packet/chunk/shutdown_chunk.h
index 2dec9c172c..d591553d55 100644
--- a/net/dcsctp/packet/chunk/shutdown_chunk.h
+++ b/net/dcsctp/packet/chunk/shutdown_chunk.h
@@ -13,10 +13,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/chunk/chunk.h"
 #include "net/dcsctp/packet/tlv_trait.h"
@@ -37,8 +37,7 @@ class ShutdownChunk : public Chunk, public TLVTrait<ShutdownChunkConfig> {
   explicit ShutdownChunk(TSN cumulative_tsn_ack)
       : cumulative_tsn_ack_(cumulative_tsn_ack) {}
 
-  static std::optional<ShutdownChunk> Parse(
-      webrtc::ArrayView<const uint8_t> data);
+  static std::optional<ShutdownChunk> Parse(std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
diff --git a/net/dcsctp/packet/chunk/shutdown_complete_chunk.cc b/net/dcsctp/packet/chunk/shutdown_complete_chunk.cc
index 4a596d910a..82edde6b06 100644
--- a/net/dcsctp/packet/chunk/shutdown_complete_chunk.cc
+++ b/net/dcsctp/packet/chunk/shutdown_complete_chunk.cc
@@ -11,10 +11,10 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "net/dcsctp/packet/bounded_byte_writer.h"
 
@@ -29,7 +29,7 @@ namespace dcsctp {
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
 std::optional<ShutdownCompleteChunk> ShutdownCompleteChunk::Parse(
-    webrtc::ArrayView<const uint8_t> data) {
+    std::span<const uint8_t> data) {
   std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
   if (!reader.has_value()) {
     return std::nullopt;
diff --git a/net/dcsctp/packet/chunk/shutdown_complete_chunk.h b/net/dcsctp/packet/chunk/shutdown_complete_chunk.h
index e8cc570d5c..2f85602b74 100644
--- a/net/dcsctp/packet/chunk/shutdown_complete_chunk.h
+++ b/net/dcsctp/packet/chunk/shutdown_complete_chunk.h
@@ -13,10 +13,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/chunk/chunk.h"
 #include "net/dcsctp/packet/tlv_trait.h"
 
@@ -38,7 +38,7 @@ class ShutdownCompleteChunk : public Chunk,
       : tag_reflected_(tag_reflected) {}
 
   static std::optional<ShutdownCompleteChunk> Parse(
-      webrtc::ArrayView<const uint8_t> data);
+      std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
diff --git a/net/dcsctp/packet/crc32c.cc b/net/dcsctp/packet/crc32c.cc
index a55749b2bd..c32e28bcc9 100644
--- a/net/dcsctp/packet/crc32c.cc
+++ b/net/dcsctp/packet/crc32c.cc
@@ -10,15 +10,15 @@
 #include "net/dcsctp/packet/crc32c.h"
 
 #include <cstdint>
+#include <span>
 
 #include "absl/crc/crc32c.h"
 #include "absl/numeric/bits.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 
 namespace dcsctp {
 
-uint32_t GenerateCrc32C(webrtc::ArrayView<const uint8_t> data) {
+uint32_t GenerateCrc32C(std::span<const uint8_t> data) {
   absl::crc32c_t crc32 = absl::ComputeCrc32c(absl::string_view(
       reinterpret_cast<const char*>(data.data()), data.size()));
   return absl::byteswap(static_cast<uint32_t>(crc32));
diff --git a/net/dcsctp/packet/crc32c.h b/net/dcsctp/packet/crc32c.h
index 71ef16ceeb..a8e8900be4 100644
--- a/net/dcsctp/packet/crc32c.h
+++ b/net/dcsctp/packet/crc32c.h
@@ -11,13 +11,12 @@
 #define NET_DCSCTP_PACKET_CRC32C_H_
 
 #include <cstdint>
-
-#include "api/array_view.h"
+#include <span>
 
 namespace dcsctp {
 
 // Generates the CRC32C checksum of `data`.
-uint32_t GenerateCrc32C(webrtc::ArrayView<const uint8_t> data);
+uint32_t GenerateCrc32C(std::span<const uint8_t> data);
 
 }  // namespace dcsctp
 
diff --git a/net/dcsctp/packet/error_cause/cookie_received_while_shutting_down_cause.cc b/net/dcsctp/packet/error_cause/cookie_received_while_shutting_down_cause.cc
index 629ff2e30c..d6194120c2 100644
--- a/net/dcsctp/packet/error_cause/cookie_received_while_shutting_down_cause.cc
+++ b/net/dcsctp/packet/error_cause/cookie_received_while_shutting_down_cause.cc
@@ -11,11 +11,10 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
-
 namespace dcsctp {
 
 // https://tools.ietf.org/html/rfc4960#section-3.3.10.10
@@ -25,8 +24,7 @@ namespace dcsctp {
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
 std::optional<CookieReceivedWhileShuttingDownCause>
-CookieReceivedWhileShuttingDownCause::Parse(
-    webrtc::ArrayView<const uint8_t> data) {
+CookieReceivedWhileShuttingDownCause::Parse(std::span<const uint8_t> data) {
   if (!ParseTLV(data).has_value()) {
     return std::nullopt;
   }
diff --git a/net/dcsctp/packet/error_cause/cookie_received_while_shutting_down_cause.h b/net/dcsctp/packet/error_cause/cookie_received_while_shutting_down_cause.h
index 360a31664d..8d61069127 100644
--- a/net/dcsctp/packet/error_cause/cookie_received_while_shutting_down_cause.h
+++ b/net/dcsctp/packet/error_cause/cookie_received_while_shutting_down_cause.h
@@ -13,10 +13,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
 #include "net/dcsctp/packet/tlv_trait.h"
 
@@ -39,7 +39,7 @@ class CookieReceivedWhileShuttingDownCause
   CookieReceivedWhileShuttingDownCause() {}
 
   static std::optional<CookieReceivedWhileShuttingDownCause> Parse(
-      webrtc::ArrayView<const uint8_t> data);
+      std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
diff --git a/net/dcsctp/packet/error_cause/invalid_mandatory_parameter_cause.cc b/net/dcsctp/packet/error_cause/invalid_mandatory_parameter_cause.cc
index eb361e3596..5b008faf80 100644
--- a/net/dcsctp/packet/error_cause/invalid_mandatory_parameter_cause.cc
+++ b/net/dcsctp/packet/error_cause/invalid_mandatory_parameter_cause.cc
@@ -11,11 +11,10 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
-
 namespace dcsctp {
 
 // https://tools.ietf.org/html/rfc4960#section-3.3.10.7
@@ -25,7 +24,7 @@ namespace dcsctp {
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
 std::optional<InvalidMandatoryParameterCause>
-InvalidMandatoryParameterCause::Parse(webrtc::ArrayView<const uint8_t> data) {
+InvalidMandatoryParameterCause::Parse(std::span<const uint8_t> data) {
   if (!ParseTLV(data).has_value()) {
     return std::nullopt;
   }
diff --git a/net/dcsctp/packet/error_cause/invalid_mandatory_parameter_cause.h b/net/dcsctp/packet/error_cause/invalid_mandatory_parameter_cause.h
index 734b72979d..59361c63ea 100644
--- a/net/dcsctp/packet/error_cause/invalid_mandatory_parameter_cause.h
+++ b/net/dcsctp/packet/error_cause/invalid_mandatory_parameter_cause.h
@@ -13,10 +13,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
 #include "net/dcsctp/packet/tlv_trait.h"
 
@@ -38,7 +38,7 @@ class InvalidMandatoryParameterCause
   InvalidMandatoryParameterCause() {}
 
   static std::optional<InvalidMandatoryParameterCause> Parse(
-      webrtc::ArrayView<const uint8_t> data);
+      std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
diff --git a/net/dcsctp/packet/error_cause/invalid_stream_identifier_cause.cc b/net/dcsctp/packet/error_cause/invalid_stream_identifier_cause.cc
index 7cba7fb7b7..fb94f243cb 100644
--- a/net/dcsctp/packet/error_cause/invalid_stream_identifier_cause.cc
+++ b/net/dcsctp/packet/error_cause/invalid_stream_identifier_cause.cc
@@ -11,10 +11,10 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "net/dcsctp/packet/bounded_byte_writer.h"
 #include "net/dcsctp/public/types.h"
@@ -31,7 +31,7 @@ namespace dcsctp {
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
 std::optional<InvalidStreamIdentifierCause> InvalidStreamIdentifierCause::Parse(
-    webrtc::ArrayView<const uint8_t> data) {
+    std::span<const uint8_t> data) {
   std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
   if (!reader.has_value()) {
     return std::nullopt;
diff --git a/net/dcsctp/packet/error_cause/invalid_stream_identifier_cause.h b/net/dcsctp/packet/error_cause/invalid_stream_identifier_cause.h
index b58809e51a..2cb977d212 100644
--- a/net/dcsctp/packet/error_cause/invalid_stream_identifier_cause.h
+++ b/net/dcsctp/packet/error_cause/invalid_stream_identifier_cause.h
@@ -13,10 +13,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
 #include "net/dcsctp/packet/tlv_trait.h"
 #include "net/dcsctp/public/types.h"
@@ -40,7 +40,7 @@ class InvalidStreamIdentifierCause
       : stream_id_(stream_id) {}
 
   static std::optional<InvalidStreamIdentifierCause> Parse(
-      webrtc::ArrayView<const uint8_t> data);
+      std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
diff --git a/net/dcsctp/packet/error_cause/missing_mandatory_parameter_cause.cc b/net/dcsctp/packet/error_cause/missing_mandatory_parameter_cause.cc
index db769f9c12..2cd90d90ea 100644
--- a/net/dcsctp/packet/error_cause/missing_mandatory_parameter_cause.cc
+++ b/net/dcsctp/packet/error_cause/missing_mandatory_parameter_cause.cc
@@ -12,10 +12,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "net/dcsctp/packet/bounded_byte_writer.h"
 #include "rtc_base/logging.h"
@@ -37,7 +37,7 @@ namespace dcsctp {
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
 std::optional<MissingMandatoryParameterCause>
-MissingMandatoryParameterCause::Parse(webrtc::ArrayView<const uint8_t> data) {
+MissingMandatoryParameterCause::Parse(std::span<const uint8_t> data) {
   std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
   if (!reader.has_value()) {
     return std::nullopt;
diff --git a/net/dcsctp/packet/error_cause/missing_mandatory_parameter_cause.h b/net/dcsctp/packet/error_cause/missing_mandatory_parameter_cause.h
index 04b33d8236..8ca8cdc2f1 100644
--- a/net/dcsctp/packet/error_cause/missing_mandatory_parameter_cause.h
+++ b/net/dcsctp/packet/error_cause/missing_mandatory_parameter_cause.h
@@ -13,10 +13,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
 #include "net/dcsctp/packet/tlv_trait.h"
 
@@ -36,17 +36,17 @@ class MissingMandatoryParameterCause
   static constexpr int kType = MissingMandatoryParameterCauseConfig::kType;
 
   explicit MissingMandatoryParameterCause(
-      webrtc::ArrayView<const uint16_t> missing_parameter_types)
+      std::span<const uint16_t> missing_parameter_types)
       : missing_parameter_types_(missing_parameter_types.begin(),
                                  missing_parameter_types.end()) {}
 
   static std::optional<MissingMandatoryParameterCause> Parse(
-      webrtc::ArrayView<const uint8_t> data);
+      std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
 
-  webrtc::ArrayView<const uint16_t> missing_parameter_types() const {
+  std::span<const uint16_t> missing_parameter_types() const {
     return missing_parameter_types_;
   }
 
diff --git a/net/dcsctp/packet/error_cause/missing_mandatory_parameter_cause_test.cc b/net/dcsctp/packet/error_cause/missing_mandatory_parameter_cause_test.cc
index ade3667484..9a83ff3ce9 100644
--- a/net/dcsctp/packet/error_cause/missing_mandatory_parameter_cause_test.cc
+++ b/net/dcsctp/packet/error_cause/missing_mandatory_parameter_cause_test.cc
@@ -12,7 +12,6 @@
 #include <cstdint>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/testing/testing_macros.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
diff --git a/net/dcsctp/packet/error_cause/no_user_data_cause.cc b/net/dcsctp/packet/error_cause/no_user_data_cause.cc
index f1f3acc40e..379d1ba4c4 100644
--- a/net/dcsctp/packet/error_cause/no_user_data_cause.cc
+++ b/net/dcsctp/packet/error_cause/no_user_data_cause.cc
@@ -11,10 +11,10 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -32,7 +32,7 @@ namespace dcsctp {
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
 std::optional<NoUserDataCause> NoUserDataCause::Parse(
-    webrtc::ArrayView<const uint8_t> data) {
+    std::span<const uint8_t> data) {
   std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
   if (!reader.has_value()) {
     return std::nullopt;
diff --git a/net/dcsctp/packet/error_cause/no_user_data_cause.h b/net/dcsctp/packet/error_cause/no_user_data_cause.h
index ad67b33356..cef1ee7f81 100644
--- a/net/dcsctp/packet/error_cause/no_user_data_cause.h
+++ b/net/dcsctp/packet/error_cause/no_user_data_cause.h
@@ -13,10 +13,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
 #include "net/dcsctp/packet/tlv_trait.h"
@@ -36,8 +36,7 @@ class NoUserDataCause : public Parameter,
 
   explicit NoUserDataCause(TSN tsn) : tsn_(tsn) {}
 
-  static std::optional<NoUserDataCause> Parse(
-      webrtc::ArrayView<const uint8_t> data);
+  static std::optional<NoUserDataCause> Parse(std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
diff --git a/net/dcsctp/packet/error_cause/out_of_resource_error_cause.cc b/net/dcsctp/packet/error_cause/out_of_resource_error_cause.cc
index 91313335c0..b27d8a2e93 100644
--- a/net/dcsctp/packet/error_cause/out_of_resource_error_cause.cc
+++ b/net/dcsctp/packet/error_cause/out_of_resource_error_cause.cc
@@ -11,11 +11,10 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
-
 namespace dcsctp {
 
 // https://tools.ietf.org/html/rfc4960#section-3.3.10.4
@@ -25,7 +24,7 @@ namespace dcsctp {
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
 std::optional<OutOfResourceErrorCause> OutOfResourceErrorCause::Parse(
-    webrtc::ArrayView<const uint8_t> data) {
+    std::span<const uint8_t> data) {
   if (!ParseTLV(data).has_value()) {
     return std::nullopt;
   }
diff --git a/net/dcsctp/packet/error_cause/out_of_resource_error_cause.h b/net/dcsctp/packet/error_cause/out_of_resource_error_cause.h
index 6e4e9eef24..63e3dd2b30 100644
--- a/net/dcsctp/packet/error_cause/out_of_resource_error_cause.h
+++ b/net/dcsctp/packet/error_cause/out_of_resource_error_cause.h
@@ -13,10 +13,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
 #include "net/dcsctp/packet/tlv_trait.h"
 
@@ -37,7 +37,7 @@ class OutOfResourceErrorCause : public Parameter,
   OutOfResourceErrorCause() {}
 
   static std::optional<OutOfResourceErrorCause> Parse(
-      webrtc::ArrayView<const uint8_t> data);
+      std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
diff --git a/net/dcsctp/packet/error_cause/protocol_violation_cause.cc b/net/dcsctp/packet/error_cause/protocol_violation_cause.cc
index 6196472aaf..bc1a4cc5a6 100644
--- a/net/dcsctp/packet/error_cause/protocol_violation_cause.cc
+++ b/net/dcsctp/packet/error_cause/protocol_violation_cause.cc
@@ -11,10 +11,10 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "net/dcsctp/packet/bounded_byte_writer.h"
 #include "rtc_base/strings/string_builder.h"
@@ -33,7 +33,7 @@ namespace dcsctp {
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
 std::optional<ProtocolViolationCause> ProtocolViolationCause::Parse(
-    webrtc::ArrayView<const uint8_t> data) {
+    std::span<const uint8_t> data) {
   std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
   if (!reader.has_value()) {
     return std::nullopt;
@@ -46,7 +46,7 @@ std::optional<ProtocolViolationCause> ProtocolViolationCause::Parse(
 void ProtocolViolationCause::SerializeTo(std::vector<uint8_t>& out) const {
   BoundedByteWriter<kHeaderSize> writer =
       AllocateTLV(out, additional_information_.size());
-  writer.CopyToVariableData(webrtc::MakeArrayView(
+  writer.CopyToVariableData(std::span(
       reinterpret_cast<const uint8_t*>(additional_information_.data()),
       additional_information_.size()));
 }
diff --git a/net/dcsctp/packet/error_cause/protocol_violation_cause.h b/net/dcsctp/packet/error_cause/protocol_violation_cause.h
index 9e265e9391..85e8176c2e 100644
--- a/net/dcsctp/packet/error_cause/protocol_violation_cause.h
+++ b/net/dcsctp/packet/error_cause/protocol_violation_cause.h
@@ -13,11 +13,11 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
 #include "net/dcsctp/packet/tlv_trait.h"
 
@@ -39,7 +39,7 @@ class ProtocolViolationCause : public Parameter,
       : additional_information_(additional_information) {}
 
   static std::optional<ProtocolViolationCause> Parse(
-      webrtc::ArrayView<const uint8_t> data);
+      std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
diff --git a/net/dcsctp/packet/error_cause/protocol_violation_cause_test.cc b/net/dcsctp/packet/error_cause/protocol_violation_cause_test.cc
index de3ef22086..b0f9878214 100644
--- a/net/dcsctp/packet/error_cause/protocol_violation_cause_test.cc
+++ b/net/dcsctp/packet/error_cause/protocol_violation_cause_test.cc
@@ -11,7 +11,6 @@
 
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
 #include "net/dcsctp/testing/testing_macros.h"
 #include "test/gmock.h"
diff --git a/net/dcsctp/packet/error_cause/restart_of_an_association_with_new_address_cause.cc b/net/dcsctp/packet/error_cause/restart_of_an_association_with_new_address_cause.cc
index 8694ff40e0..f15eacd880 100644
--- a/net/dcsctp/packet/error_cause/restart_of_an_association_with_new_address_cause.cc
+++ b/net/dcsctp/packet/error_cause/restart_of_an_association_with_new_address_cause.cc
@@ -11,10 +11,10 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "net/dcsctp/packet/bounded_byte_writer.h"
 
@@ -33,7 +33,7 @@ namespace dcsctp {
 
 std::optional<RestartOfAnAssociationWithNewAddressesCause>
 RestartOfAnAssociationWithNewAddressesCause::Parse(
-    webrtc::ArrayView<const uint8_t> data) {
+    std::span<const uint8_t> data) {
   std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
   if (!reader.has_value()) {
     return std::nullopt;
diff --git a/net/dcsctp/packet/error_cause/restart_of_an_association_with_new_address_cause.h b/net/dcsctp/packet/error_cause/restart_of_an_association_with_new_address_cause.h
index 14b6c4ec3c..6cfe28adb0 100644
--- a/net/dcsctp/packet/error_cause/restart_of_an_association_with_new_address_cause.h
+++ b/net/dcsctp/packet/error_cause/restart_of_an_association_with_new_address_cause.h
@@ -13,10 +13,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
 #include "net/dcsctp/packet/tlv_trait.h"
 
@@ -38,16 +38,16 @@ class RestartOfAnAssociationWithNewAddressesCause
       RestartOfAnAssociationWithNewAddressesCauseConfig::kType;
 
   explicit RestartOfAnAssociationWithNewAddressesCause(
-      webrtc::ArrayView<const uint8_t> new_address_tlvs)
+      std::span<const uint8_t> new_address_tlvs)
       : new_address_tlvs_(new_address_tlvs.begin(), new_address_tlvs.end()) {}
 
   static std::optional<RestartOfAnAssociationWithNewAddressesCause> Parse(
-      webrtc::ArrayView<const uint8_t> data);
+      std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
 
-  webrtc::ArrayView<const uint8_t> new_address_tlvs() const {
+  std::span<const uint8_t> new_address_tlvs() const {
     return new_address_tlvs_;
   }
 
diff --git a/net/dcsctp/packet/error_cause/restart_of_an_association_with_new_address_cause_test.cc b/net/dcsctp/packet/error_cause/restart_of_an_association_with_new_address_cause_test.cc
index 0c3b3fcb30..c32ada22eb 100644
--- a/net/dcsctp/packet/error_cause/restart_of_an_association_with_new_address_cause_test.cc
+++ b/net/dcsctp/packet/error_cause/restart_of_an_association_with_new_address_cause_test.cc
@@ -12,7 +12,6 @@
 #include <cstdint>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/testing/testing_macros.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
diff --git a/net/dcsctp/packet/error_cause/stale_cookie_error_cause.cc b/net/dcsctp/packet/error_cause/stale_cookie_error_cause.cc
index 996ca071ec..00c9ab49a8 100644
--- a/net/dcsctp/packet/error_cause/stale_cookie_error_cause.cc
+++ b/net/dcsctp/packet/error_cause/stale_cookie_error_cause.cc
@@ -11,10 +11,10 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "net/dcsctp/packet/bounded_byte_writer.h"
 #include "rtc_base/strings/string_builder.h"
@@ -30,7 +30,7 @@ namespace dcsctp {
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
 std::optional<StaleCookieErrorCause> StaleCookieErrorCause::Parse(
-    webrtc::ArrayView<const uint8_t> data) {
+    std::span<const uint8_t> data) {
   std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
   if (!reader.has_value()) {
     return std::nullopt;
diff --git a/net/dcsctp/packet/error_cause/stale_cookie_error_cause.h b/net/dcsctp/packet/error_cause/stale_cookie_error_cause.h
index cb5bace37c..82bf61e7ee 100644
--- a/net/dcsctp/packet/error_cause/stale_cookie_error_cause.h
+++ b/net/dcsctp/packet/error_cause/stale_cookie_error_cause.h
@@ -13,10 +13,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
 #include "net/dcsctp/packet/tlv_trait.h"
 
@@ -38,7 +38,7 @@ class StaleCookieErrorCause : public Parameter,
       : staleness_us_(staleness_us) {}
 
   static std::optional<StaleCookieErrorCause> Parse(
-      webrtc::ArrayView<const uint8_t> data);
+      std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
diff --git a/net/dcsctp/packet/error_cause/unrecognized_chunk_type_cause.cc b/net/dcsctp/packet/error_cause/unrecognized_chunk_type_cause.cc
index 6a12fff64d..59d015ce97 100644
--- a/net/dcsctp/packet/error_cause/unrecognized_chunk_type_cause.cc
+++ b/net/dcsctp/packet/error_cause/unrecognized_chunk_type_cause.cc
@@ -11,11 +11,11 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "net/dcsctp/packet/bounded_byte_writer.h"
 #include "rtc_base/strings/string_builder.h"
@@ -32,7 +32,7 @@ namespace dcsctp {
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
 std::optional<UnrecognizedChunkTypeCause> UnrecognizedChunkTypeCause::Parse(
-    webrtc::ArrayView<const uint8_t> data) {
+    std::span<const uint8_t> data) {
   std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
   if (!reader.has_value()) {
     return std::nullopt;
diff --git a/net/dcsctp/packet/error_cause/unrecognized_chunk_type_cause.h b/net/dcsctp/packet/error_cause/unrecognized_chunk_type_cause.h
index 48c9bc9f36..6b8c506155 100644
--- a/net/dcsctp/packet/error_cause/unrecognized_chunk_type_cause.h
+++ b/net/dcsctp/packet/error_cause/unrecognized_chunk_type_cause.h
@@ -13,11 +13,11 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
 #include "net/dcsctp/packet/tlv_trait.h"
 
@@ -40,12 +40,12 @@ class UnrecognizedChunkTypeCause
       : unrecognized_chunk_(std::move(unrecognized_chunk)) {}
 
   static std::optional<UnrecognizedChunkTypeCause> Parse(
-      webrtc::ArrayView<const uint8_t> data);
+      std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
 
-  webrtc::ArrayView<const uint8_t> unrecognized_chunk() const {
+  std::span<const uint8_t> unrecognized_chunk() const {
     return unrecognized_chunk_;
   }
 
diff --git a/net/dcsctp/packet/error_cause/unrecognized_chunk_type_cause_test.cc b/net/dcsctp/packet/error_cause/unrecognized_chunk_type_cause_test.cc
index e058f5f827..7a5f78c017 100644
--- a/net/dcsctp/packet/error_cause/unrecognized_chunk_type_cause_test.cc
+++ b/net/dcsctp/packet/error_cause/unrecognized_chunk_type_cause_test.cc
@@ -12,7 +12,6 @@
 #include <cstdint>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/testing/testing_macros.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
diff --git a/net/dcsctp/packet/error_cause/unrecognized_parameter_cause.cc b/net/dcsctp/packet/error_cause/unrecognized_parameter_cause.cc
index 73df61197a..3dc884e1c9 100644
--- a/net/dcsctp/packet/error_cause/unrecognized_parameter_cause.cc
+++ b/net/dcsctp/packet/error_cause/unrecognized_parameter_cause.cc
@@ -11,10 +11,10 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "net/dcsctp/packet/bounded_byte_writer.h"
 
@@ -30,7 +30,7 @@ namespace dcsctp {
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
 std::optional<UnrecognizedParametersCause> UnrecognizedParametersCause::Parse(
-    webrtc::ArrayView<const uint8_t> data) {
+    std::span<const uint8_t> data) {
   std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
   if (!reader.has_value()) {
     return std::nullopt;
diff --git a/net/dcsctp/packet/error_cause/unrecognized_parameter_cause.h b/net/dcsctp/packet/error_cause/unrecognized_parameter_cause.h
index 3088705258..08cb3cce9f 100644
--- a/net/dcsctp/packet/error_cause/unrecognized_parameter_cause.h
+++ b/net/dcsctp/packet/error_cause/unrecognized_parameter_cause.h
@@ -13,10 +13,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
 #include "net/dcsctp/packet/tlv_trait.h"
 
@@ -36,17 +36,17 @@ class UnrecognizedParametersCause
   static constexpr int kType = UnrecognizedParametersCauseConfig::kType;
 
   explicit UnrecognizedParametersCause(
-      webrtc::ArrayView<const uint8_t> unrecognized_parameters)
+      std::span<const uint8_t> unrecognized_parameters)
       : unrecognized_parameters_(unrecognized_parameters.begin(),
                                  unrecognized_parameters.end()) {}
 
   static std::optional<UnrecognizedParametersCause> Parse(
-      webrtc::ArrayView<const uint8_t> data);
+      std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
 
-  webrtc::ArrayView<const uint8_t> unrecognized_parameters() const {
+  std::span<const uint8_t> unrecognized_parameters() const {
     return unrecognized_parameters_;
   }
 
diff --git a/net/dcsctp/packet/error_cause/unrecognized_parameter_cause_test.cc b/net/dcsctp/packet/error_cause/unrecognized_parameter_cause_test.cc
index 189205c2a1..5a39034358 100644
--- a/net/dcsctp/packet/error_cause/unrecognized_parameter_cause_test.cc
+++ b/net/dcsctp/packet/error_cause/unrecognized_parameter_cause_test.cc
@@ -12,7 +12,6 @@
 #include <cstdint>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/testing/testing_macros.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
diff --git a/net/dcsctp/packet/error_cause/unresolvable_address_cause.cc b/net/dcsctp/packet/error_cause/unresolvable_address_cause.cc
index 77cb7ae3ba..af3885ded4 100644
--- a/net/dcsctp/packet/error_cause/unresolvable_address_cause.cc
+++ b/net/dcsctp/packet/error_cause/unresolvable_address_cause.cc
@@ -11,10 +11,10 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "net/dcsctp/packet/bounded_byte_writer.h"
 
@@ -30,7 +30,7 @@ namespace dcsctp {
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
 std::optional<UnresolvableAddressCause> UnresolvableAddressCause::Parse(
-    webrtc::ArrayView<const uint8_t> data) {
+    std::span<const uint8_t> data) {
   std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
   if (!reader.has_value()) {
     return std::nullopt;
diff --git a/net/dcsctp/packet/error_cause/unresolvable_address_cause.h b/net/dcsctp/packet/error_cause/unresolvable_address_cause.h
index 3ab94eb674..c6914533e6 100644
--- a/net/dcsctp/packet/error_cause/unresolvable_address_cause.h
+++ b/net/dcsctp/packet/error_cause/unresolvable_address_cause.h
@@ -13,10 +13,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
 #include "net/dcsctp/packet/tlv_trait.h"
 
@@ -36,17 +36,17 @@ class UnresolvableAddressCause
   static constexpr int kType = UnresolvableAddressCauseConfig::kType;
 
   explicit UnresolvableAddressCause(
-      webrtc::ArrayView<const uint8_t> unresolvable_address)
+      std::span<const uint8_t> unresolvable_address)
       : unresolvable_address_(unresolvable_address.begin(),
                               unresolvable_address.end()) {}
 
   static std::optional<UnresolvableAddressCause> Parse(
-      webrtc::ArrayView<const uint8_t> data);
+      std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
 
-  webrtc::ArrayView<const uint8_t> unresolvable_address() const {
+  std::span<const uint8_t> unresolvable_address() const {
     return unresolvable_address_;
   }
 
diff --git a/net/dcsctp/packet/error_cause/unresolvable_address_cause_test.cc b/net/dcsctp/packet/error_cause/unresolvable_address_cause_test.cc
index 4c76683ed3..d4c60cf637 100644
--- a/net/dcsctp/packet/error_cause/unresolvable_address_cause_test.cc
+++ b/net/dcsctp/packet/error_cause/unresolvable_address_cause_test.cc
@@ -12,7 +12,6 @@
 #include <cstdint>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/testing/testing_macros.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
diff --git a/net/dcsctp/packet/error_cause/user_initiated_abort_cause.cc b/net/dcsctp/packet/error_cause/user_initiated_abort_cause.cc
index 1b67aaa52f..f2ad53fca8 100644
--- a/net/dcsctp/packet/error_cause/user_initiated_abort_cause.cc
+++ b/net/dcsctp/packet/error_cause/user_initiated_abort_cause.cc
@@ -11,10 +11,10 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "net/dcsctp/packet/bounded_byte_writer.h"
 #include "rtc_base/strings/string_builder.h"
@@ -33,7 +33,7 @@ namespace dcsctp {
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
 std::optional<UserInitiatedAbortCause> UserInitiatedAbortCause::Parse(
-    webrtc::ArrayView<const uint8_t> data) {
+    std::span<const uint8_t> data) {
   std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
   if (!reader.has_value()) {
     return std::nullopt;
@@ -49,7 +49,7 @@ std::optional<UserInitiatedAbortCause> UserInitiatedAbortCause::Parse(
 void UserInitiatedAbortCause::SerializeTo(std::vector<uint8_t>& out) const {
   BoundedByteWriter<kHeaderSize> writer =
       AllocateTLV(out, upper_layer_abort_reason_.size());
-  writer.CopyToVariableData(webrtc::MakeArrayView(
+  writer.CopyToVariableData(std::span(
       reinterpret_cast<const uint8_t*>(upper_layer_abort_reason_.data()),
       upper_layer_abort_reason_.size()));
 }
diff --git a/net/dcsctp/packet/error_cause/user_initiated_abort_cause.h b/net/dcsctp/packet/error_cause/user_initiated_abort_cause.h
index 8cea2bcf70..2b9d21b752 100644
--- a/net/dcsctp/packet/error_cause/user_initiated_abort_cause.h
+++ b/net/dcsctp/packet/error_cause/user_initiated_abort_cause.h
@@ -13,11 +13,11 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
 #include "net/dcsctp/packet/tlv_trait.h"
 
@@ -39,7 +39,7 @@ class UserInitiatedAbortCause : public Parameter,
       : upper_layer_abort_reason_(upper_layer_abort_reason) {}
 
   static std::optional<UserInitiatedAbortCause> Parse(
-      webrtc::ArrayView<const uint8_t> data);
+      std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
diff --git a/net/dcsctp/packet/error_cause/user_initiated_abort_cause_test.cc b/net/dcsctp/packet/error_cause/user_initiated_abort_cause_test.cc
index 683900778f..3a6437887d 100644
--- a/net/dcsctp/packet/error_cause/user_initiated_abort_cause_test.cc
+++ b/net/dcsctp/packet/error_cause/user_initiated_abort_cause_test.cc
@@ -11,7 +11,6 @@
 
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
 #include "net/dcsctp/testing/testing_macros.h"
 #include "test/gmock.h"
diff --git a/net/dcsctp/packet/parameter/add_incoming_streams_request_parameter.cc b/net/dcsctp/packet/parameter/add_incoming_streams_request_parameter.cc
index b0fab6a521..9e8a68e0c2 100644
--- a/net/dcsctp/packet/parameter/add_incoming_streams_request_parameter.cc
+++ b/net/dcsctp/packet/parameter/add_incoming_streams_request_parameter.cc
@@ -11,10 +11,10 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -35,8 +35,7 @@ namespace dcsctp {
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
 std::optional<AddIncomingStreamsRequestParameter>
-AddIncomingStreamsRequestParameter::Parse(
-    webrtc::ArrayView<const uint8_t> data) {
+AddIncomingStreamsRequestParameter::Parse(std::span<const uint8_t> data) {
   std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
   if (!reader.has_value()) {
     return std::nullopt;
diff --git a/net/dcsctp/packet/parameter/add_incoming_streams_request_parameter.h b/net/dcsctp/packet/parameter/add_incoming_streams_request_parameter.h
index fe3b306af5..298c845fb5 100644
--- a/net/dcsctp/packet/parameter/add_incoming_streams_request_parameter.h
+++ b/net/dcsctp/packet/parameter/add_incoming_streams_request_parameter.h
@@ -13,10 +13,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
 #include "net/dcsctp/packet/tlv_trait.h"
@@ -43,7 +43,7 @@ class AddIncomingStreamsRequestParameter
         nbr_of_new_streams_(nbr_of_new_streams) {}
 
   static std::optional<AddIncomingStreamsRequestParameter> Parse(
-      webrtc::ArrayView<const uint8_t> data);
+      std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
diff --git a/net/dcsctp/packet/parameter/add_outgoing_streams_request_parameter.cc b/net/dcsctp/packet/parameter/add_outgoing_streams_request_parameter.cc
index 7310474feb..b4d0906848 100644
--- a/net/dcsctp/packet/parameter/add_outgoing_streams_request_parameter.cc
+++ b/net/dcsctp/packet/parameter/add_outgoing_streams_request_parameter.cc
@@ -11,10 +11,10 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -35,8 +35,7 @@ namespace dcsctp {
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
 std::optional<AddOutgoingStreamsRequestParameter>
-AddOutgoingStreamsRequestParameter::Parse(
-    webrtc::ArrayView<const uint8_t> data) {
+AddOutgoingStreamsRequestParameter::Parse(std::span<const uint8_t> data) {
   std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
   if (!reader.has_value()) {
     return std::nullopt;
diff --git a/net/dcsctp/packet/parameter/add_outgoing_streams_request_parameter.h b/net/dcsctp/packet/parameter/add_outgoing_streams_request_parameter.h
index ac1735f95f..44085ed481 100644
--- a/net/dcsctp/packet/parameter/add_outgoing_streams_request_parameter.h
+++ b/net/dcsctp/packet/parameter/add_outgoing_streams_request_parameter.h
@@ -13,10 +13,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
 #include "net/dcsctp/packet/tlv_trait.h"
@@ -43,7 +43,7 @@ class AddOutgoingStreamsRequestParameter
         nbr_of_new_streams_(nbr_of_new_streams) {}
 
   static std::optional<AddOutgoingStreamsRequestParameter> Parse(
-      webrtc::ArrayView<const uint8_t> data);
+      std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
diff --git a/net/dcsctp/packet/parameter/forward_tsn_supported_parameter.cc b/net/dcsctp/packet/parameter/forward_tsn_supported_parameter.cc
index c095dddbbb..0b548444ab 100644
--- a/net/dcsctp/packet/parameter/forward_tsn_supported_parameter.cc
+++ b/net/dcsctp/packet/parameter/forward_tsn_supported_parameter.cc
@@ -11,11 +11,10 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
-
 namespace dcsctp {
 
 // https://tools.ietf.org/html/rfc3758#section-3.1
@@ -26,7 +25,7 @@ namespace dcsctp {
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
 std::optional<ForwardTsnSupportedParameter> ForwardTsnSupportedParameter::Parse(
-    webrtc::ArrayView<const uint8_t> data) {
+    std::span<const uint8_t> data) {
   if (!ParseTLV(data).has_value()) {
     return std::nullopt;
   }
diff --git a/net/dcsctp/packet/parameter/forward_tsn_supported_parameter.h b/net/dcsctp/packet/parameter/forward_tsn_supported_parameter.h
index 03e492f8c7..eee26d09d8 100644
--- a/net/dcsctp/packet/parameter/forward_tsn_supported_parameter.h
+++ b/net/dcsctp/packet/parameter/forward_tsn_supported_parameter.h
@@ -13,10 +13,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
 #include "net/dcsctp/packet/tlv_trait.h"
 
@@ -38,7 +38,7 @@ class ForwardTsnSupportedParameter
   ForwardTsnSupportedParameter() {}
 
   static std::optional<ForwardTsnSupportedParameter> Parse(
-      webrtc::ArrayView<const uint8_t> data);
+      std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
diff --git a/net/dcsctp/packet/parameter/heartbeat_info_parameter.cc b/net/dcsctp/packet/parameter/heartbeat_info_parameter.cc
index e6120612a7..3b6f534aa6 100644
--- a/net/dcsctp/packet/parameter/heartbeat_info_parameter.cc
+++ b/net/dcsctp/packet/parameter/heartbeat_info_parameter.cc
@@ -11,10 +11,10 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "net/dcsctp/packet/bounded_byte_writer.h"
 #include "rtc_base/strings/string_builder.h"
@@ -34,7 +34,7 @@ namespace dcsctp {
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
 std::optional<HeartbeatInfoParameter> HeartbeatInfoParameter::Parse(
-    webrtc::ArrayView<const uint8_t> data) {
+    std::span<const uint8_t> data) {
   std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
   if (!reader.has_value()) {
     return std::nullopt;
diff --git a/net/dcsctp/packet/parameter/heartbeat_info_parameter.h b/net/dcsctp/packet/parameter/heartbeat_info_parameter.h
index 82c18dbe31..38b78f4eee 100644
--- a/net/dcsctp/packet/parameter/heartbeat_info_parameter.h
+++ b/net/dcsctp/packet/parameter/heartbeat_info_parameter.h
@@ -13,10 +13,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
 #include "net/dcsctp/packet/tlv_trait.h"
 
@@ -34,16 +34,16 @@ class HeartbeatInfoParameter : public Parameter,
  public:
   static constexpr int kType = HeartbeatInfoParameterConfig::kType;
 
-  explicit HeartbeatInfoParameter(webrtc::ArrayView<const uint8_t> info)
+  explicit HeartbeatInfoParameter(std::span<const uint8_t> info)
       : info_(info.begin(), info.end()) {}
 
   static std::optional<HeartbeatInfoParameter> Parse(
-      webrtc::ArrayView<const uint8_t> data);
+      std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
 
-  webrtc::ArrayView<const uint8_t> info() const { return info_; }
+  std::span<const uint8_t> info() const { return info_; }
 
  private:
   std::vector<uint8_t> info_;
diff --git a/net/dcsctp/packet/parameter/incoming_ssn_reset_request_parameter.cc b/net/dcsctp/packet/parameter/incoming_ssn_reset_request_parameter.cc
index fe43dae0a9..05d1b0464b 100644
--- a/net/dcsctp/packet/parameter/incoming_ssn_reset_request_parameter.cc
+++ b/net/dcsctp/packet/parameter/incoming_ssn_reset_request_parameter.cc
@@ -12,11 +12,11 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -42,7 +42,7 @@ namespace dcsctp {
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
 std::optional<IncomingSSNResetRequestParameter>
-IncomingSSNResetRequestParameter::Parse(webrtc::ArrayView<const uint8_t> data) {
+IncomingSSNResetRequestParameter::Parse(std::span<const uint8_t> data) {
   std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
   if (!reader.has_value()) {
     return std::nullopt;
diff --git a/net/dcsctp/packet/parameter/incoming_ssn_reset_request_parameter.h b/net/dcsctp/packet/parameter/incoming_ssn_reset_request_parameter.h
index 77e453a2e9..2569ef072a 100644
--- a/net/dcsctp/packet/parameter/incoming_ssn_reset_request_parameter.h
+++ b/net/dcsctp/packet/parameter/incoming_ssn_reset_request_parameter.h
@@ -13,11 +13,11 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
 #include "net/dcsctp/packet/tlv_trait.h"
@@ -45,7 +45,7 @@ class IncomingSSNResetRequestParameter
         stream_ids_(std::move(stream_ids)) {}
 
   static std::optional<IncomingSSNResetRequestParameter> Parse(
-      webrtc::ArrayView<const uint8_t> data);
+      std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
@@ -53,7 +53,7 @@ class IncomingSSNResetRequestParameter
   ReconfigRequestSN request_sequence_number() const {
     return request_sequence_number_;
   }
-  webrtc::ArrayView<const StreamID> stream_ids() const { return stream_ids_; }
+  std::span<const StreamID> stream_ids() const { return stream_ids_; }
 
  private:
   static constexpr size_t kStreamIdSize = sizeof(uint16_t);
diff --git a/net/dcsctp/packet/parameter/incoming_ssn_reset_request_parameter_test.cc b/net/dcsctp/packet/parameter/incoming_ssn_reset_request_parameter_test.cc
index 5b10fb371f..ede45c1577 100644
--- a/net/dcsctp/packet/parameter/incoming_ssn_reset_request_parameter_test.cc
+++ b/net/dcsctp/packet/parameter/incoming_ssn_reset_request_parameter_test.cc
@@ -12,7 +12,6 @@
 #include <cstdint>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/public/types.h"
 #include "net/dcsctp/testing/testing_macros.h"
diff --git a/net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.cc b/net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.cc
index 39f45fd00c..bc3ae9ae6f 100644
--- a/net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.cc
+++ b/net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.cc
@@ -12,11 +12,11 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -46,7 +46,7 @@ namespace dcsctp {
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
 std::optional<OutgoingSSNResetRequestParameter>
-OutgoingSSNResetRequestParameter::Parse(webrtc::ArrayView<const uint8_t> data) {
+OutgoingSSNResetRequestParameter::Parse(std::span<const uint8_t> data) {
   std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
   if (!reader.has_value()) {
     return std::nullopt;
diff --git a/net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.h b/net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.h
index 71399afb6a..d3c2b5e4fb 100644
--- a/net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.h
+++ b/net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.h
@@ -13,11 +13,11 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
 #include "net/dcsctp/packet/tlv_trait.h"
@@ -49,7 +49,7 @@ class OutgoingSSNResetRequestParameter
         stream_ids_(std::move(stream_ids)) {}
 
   static std::optional<OutgoingSSNResetRequestParameter> Parse(
-      webrtc::ArrayView<const uint8_t> data);
+      std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
@@ -61,7 +61,7 @@ class OutgoingSSNResetRequestParameter
     return response_sequence_number_;
   }
   TSN sender_last_assigned_tsn() const { return sender_last_assigned_tsn_; }
-  webrtc::ArrayView<const StreamID> stream_ids() const { return stream_ids_; }
+  std::span<const StreamID> stream_ids() const { return stream_ids_; }
 
  private:
   static constexpr size_t kStreamIdSize = sizeof(uint16_t);
diff --git a/net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter_test.cc b/net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter_test.cc
index f61b8a0cec..0ab234d028 100644
--- a/net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter_test.cc
+++ b/net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter_test.cc
@@ -12,7 +12,6 @@
 #include <cstdint>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/public/types.h"
 #include "net/dcsctp/testing/testing_macros.h"
diff --git a/net/dcsctp/packet/parameter/parameter.cc b/net/dcsctp/packet/parameter/parameter.cc
index ad787f958d..3cece5effe 100644
--- a/net/dcsctp/packet/parameter/parameter.cc
+++ b/net/dcsctp/packet/parameter/parameter.cc
@@ -12,9 +12,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/math.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "rtc_base/logging.h"
@@ -37,7 +37,7 @@ Parameters::Builder& Parameters::Builder::Add(const Parameter& p) {
 }
 
 std::vector<ParameterDescriptor> Parameters::descriptors() const {
-  webrtc::ArrayView<const uint8_t> span(data_);
+  std::span<const uint8_t> span(data_);
   std::vector<ParameterDescriptor> result;
   while (!span.empty()) {
     BoundedByteReader<kParameterHeaderSize> header(span);
@@ -53,10 +53,9 @@ std::vector<ParameterDescriptor> Parameters::descriptors() const {
   return result;
 }
 
-std::optional<Parameters> Parameters::Parse(
-    webrtc::ArrayView<const uint8_t> data) {
+std::optional<Parameters> Parameters::Parse(std::span<const uint8_t> data) {
   // Validate the parameter descriptors
-  webrtc::ArrayView<const uint8_t> span(data);
+  std::span<const uint8_t> span(data);
   while (!span.empty()) {
     if (span.size() < kParameterHeaderSize) {
       RTC_DLOG(LS_WARNING) << "Insufficient parameter length";
diff --git a/net/dcsctp/packet/parameter/parameter.h b/net/dcsctp/packet/parameter/parameter.h
index e64de97170..30a2feb4b8 100644
--- a/net/dcsctp/packet/parameter/parameter.h
+++ b/net/dcsctp/packet/parameter/parameter.h
@@ -10,16 +10,14 @@
 #ifndef NET_DCSCTP_PACKET_PARAMETER_PARAMETER_H_
 #define NET_DCSCTP_PACKET_PARAMETER_PARAMETER_H_
 
-
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <type_traits>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
-
 namespace dcsctp {
 
 class Parameter {
@@ -35,10 +33,10 @@ class Parameter {
 };
 
 struct ParameterDescriptor {
-  ParameterDescriptor(uint16_t type, webrtc::ArrayView<const uint8_t> data)
+  ParameterDescriptor(uint16_t type, std::span<const uint8_t> data)
       : type(type), data(data) {}
   uint16_t type;
-  webrtc::ArrayView<const uint8_t> data;
+  std::span<const uint8_t> data;
 };
 
 class Parameters {
@@ -53,13 +51,13 @@ class Parameters {
     std::vector<uint8_t> data_;
   };
 
-  static std::optional<Parameters> Parse(webrtc::ArrayView<const uint8_t> data);
+  static std::optional<Parameters> Parse(std::span<const uint8_t> data);
 
   Parameters() {}
   Parameters(Parameters&& other) = default;
   Parameters& operator=(Parameters&& other) = default;
 
-  webrtc::ArrayView<const uint8_t> data() const { return data_; }
+  std::span<const uint8_t> data() const { return data_; }
   std::vector<ParameterDescriptor> descriptors() const;
 
   template <typename P>
diff --git a/net/dcsctp/packet/parameter/parameter_test.cc b/net/dcsctp/packet/parameter/parameter_test.cc
index 366a740dba..97046cd1de 100644
--- a/net/dcsctp/packet/parameter/parameter_test.cc
+++ b/net/dcsctp/packet/parameter/parameter_test.cc
@@ -10,9 +10,9 @@
 #include "net/dcsctp/packet/parameter/parameter.h"
 
 #include <cstdint>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.h"
 #include "net/dcsctp/public/types.h"
@@ -33,7 +33,7 @@ TEST(ParameterTest, SerializeDeserializeParameter) {
                                                 TSN(789), {StreamID(42)}))
           .Build();
 
-  webrtc::ArrayView<const uint8_t> serialized = parameters.data();
+  std::span<const uint8_t> serialized = parameters.data();
 
   ASSERT_HAS_VALUE_AND_ASSIGN(Parameters parsed, Parameters::Parse(serialized));
   auto descriptors = parsed.descriptors();
diff --git a/net/dcsctp/packet/parameter/reconfiguration_response_parameter.cc b/net/dcsctp/packet/parameter/reconfiguration_response_parameter.cc
index d4460f7582..ffdef5cfae 100644
--- a/net/dcsctp/packet/parameter/reconfiguration_response_parameter.cc
+++ b/net/dcsctp/packet/parameter/reconfiguration_response_parameter.cc
@@ -12,12 +12,12 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <type_traits>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -63,7 +63,7 @@ absl::string_view ToString(ReconfigurationResponseParameter::Result result) {
 }
 
 std::optional<ReconfigurationResponseParameter>
-ReconfigurationResponseParameter::Parse(webrtc::ArrayView<const uint8_t> data) {
+ReconfigurationResponseParameter::Parse(std::span<const uint8_t> data) {
   std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
   if (!reader.has_value()) {
     return std::nullopt;
diff --git a/net/dcsctp/packet/parameter/reconfiguration_response_parameter.h b/net/dcsctp/packet/parameter/reconfiguration_response_parameter.h
index 6fb4b357d8..f719dee53f 100644
--- a/net/dcsctp/packet/parameter/reconfiguration_response_parameter.h
+++ b/net/dcsctp/packet/parameter/reconfiguration_response_parameter.h
@@ -13,11 +13,11 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
 #include "net/dcsctp/packet/tlv_trait.h"
@@ -65,7 +65,7 @@ class ReconfigurationResponseParameter
         receiver_next_tsn_(receiver_next_tsn) {}
 
   static std::optional<ReconfigurationResponseParameter> Parse(
-      webrtc::ArrayView<const uint8_t> data);
+      std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
diff --git a/net/dcsctp/packet/parameter/ssn_tsn_reset_request_parameter.cc b/net/dcsctp/packet/parameter/ssn_tsn_reset_request_parameter.cc
index ee0b7e3a74..e29bb6fc30 100644
--- a/net/dcsctp/packet/parameter/ssn_tsn_reset_request_parameter.cc
+++ b/net/dcsctp/packet/parameter/ssn_tsn_reset_request_parameter.cc
@@ -11,10 +11,10 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -33,7 +33,7 @@ namespace dcsctp {
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
 std::optional<SSNTSNResetRequestParameter> SSNTSNResetRequestParameter::Parse(
-    webrtc::ArrayView<const uint8_t> data) {
+    std::span<const uint8_t> data) {
   std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
   if (!reader.has_value()) {
     return std::nullopt;
diff --git a/net/dcsctp/packet/parameter/ssn_tsn_reset_request_parameter.h b/net/dcsctp/packet/parameter/ssn_tsn_reset_request_parameter.h
index 1beb0b867a..468c18c42d 100644
--- a/net/dcsctp/packet/parameter/ssn_tsn_reset_request_parameter.h
+++ b/net/dcsctp/packet/parameter/ssn_tsn_reset_request_parameter.h
@@ -13,10 +13,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
 #include "net/dcsctp/packet/tlv_trait.h"
@@ -41,7 +41,7 @@ class SSNTSNResetRequestParameter
       : request_sequence_number_(request_sequence_number) {}
 
   static std::optional<SSNTSNResetRequestParameter> Parse(
-      webrtc::ArrayView<const uint8_t> data);
+      std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
diff --git a/net/dcsctp/packet/parameter/state_cookie_parameter.cc b/net/dcsctp/packet/parameter/state_cookie_parameter.cc
index cc1c59758b..e1a59245e3 100644
--- a/net/dcsctp/packet/parameter/state_cookie_parameter.cc
+++ b/net/dcsctp/packet/parameter/state_cookie_parameter.cc
@@ -11,10 +11,10 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "net/dcsctp/packet/bounded_byte_writer.h"
 #include "rtc_base/strings/string_builder.h"
@@ -24,7 +24,7 @@ namespace dcsctp {
 // https://tools.ietf.org/html/rfc4960#section-3.3.3.1
 
 std::optional<StateCookieParameter> StateCookieParameter::Parse(
-    webrtc::ArrayView<const uint8_t> data) {
+    std::span<const uint8_t> data) {
   std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
   if (!reader.has_value()) {
     return std::nullopt;
diff --git a/net/dcsctp/packet/parameter/state_cookie_parameter.h b/net/dcsctp/packet/parameter/state_cookie_parameter.h
index 30de131402..fe723f114b 100644
--- a/net/dcsctp/packet/parameter/state_cookie_parameter.h
+++ b/net/dcsctp/packet/parameter/state_cookie_parameter.h
@@ -13,10 +13,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
 #include "net/dcsctp/packet/tlv_trait.h"
 
@@ -34,16 +34,16 @@ class StateCookieParameter : public Parameter,
  public:
   static constexpr int kType = StateCookieParameterConfig::kType;
 
-  explicit StateCookieParameter(webrtc::ArrayView<const uint8_t> data)
+  explicit StateCookieParameter(std::span<const uint8_t> data)
       : data_(data.begin(), data.end()) {}
 
   static std::optional<StateCookieParameter> Parse(
-      webrtc::ArrayView<const uint8_t> data);
+      std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
 
-  webrtc::ArrayView<const uint8_t> data() const { return data_; }
+  std::span<const uint8_t> data() const { return data_; }
 
  private:
   std::vector<uint8_t> data_;
diff --git a/net/dcsctp/packet/parameter/state_cookie_parameter_test.cc b/net/dcsctp/packet/parameter/state_cookie_parameter_test.cc
index 4093764768..6ce1f92a35 100644
--- a/net/dcsctp/packet/parameter/state_cookie_parameter_test.cc
+++ b/net/dcsctp/packet/parameter/state_cookie_parameter_test.cc
@@ -12,7 +12,6 @@
 #include <cstdint>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/testing/testing_macros.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
diff --git a/net/dcsctp/packet/parameter/supported_extensions_parameter.cc b/net/dcsctp/packet/parameter/supported_extensions_parameter.cc
index edbcc98bc3..4da5f2a77a 100644
--- a/net/dcsctp/packet/parameter/supported_extensions_parameter.cc
+++ b/net/dcsctp/packet/parameter/supported_extensions_parameter.cc
@@ -11,11 +11,11 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "net/dcsctp/packet/bounded_byte_writer.h"
 #include "rtc_base/strings/str_join.h"
@@ -38,7 +38,7 @@ namespace dcsctp {
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
 std::optional<SupportedExtensionsParameter> SupportedExtensionsParameter::Parse(
-    webrtc::ArrayView<const uint8_t> data) {
+    std::span<const uint8_t> data) {
   std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
   if (!reader.has_value()) {
     return std::nullopt;
diff --git a/net/dcsctp/packet/parameter/supported_extensions_parameter.h b/net/dcsctp/packet/parameter/supported_extensions_parameter.h
index 1953391a61..81b3da4a53 100644
--- a/net/dcsctp/packet/parameter/supported_extensions_parameter.h
+++ b/net/dcsctp/packet/parameter/supported_extensions_parameter.h
@@ -14,11 +14,11 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
 #include "net/dcsctp/packet/tlv_trait.h"
 
@@ -41,7 +41,7 @@ class SupportedExtensionsParameter
       : chunk_types_(std::move(chunk_types)) {}
 
   static std::optional<SupportedExtensionsParameter> Parse(
-      webrtc::ArrayView<const uint8_t> data);
+      std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
@@ -51,7 +51,7 @@ class SupportedExtensionsParameter
            chunk_types_.end();
   }
 
-  webrtc::ArrayView<const uint8_t> chunk_types() const { return chunk_types_; }
+  std::span<const uint8_t> chunk_types() const { return chunk_types_; }
 
  private:
   std::vector<uint8_t> chunk_types_;
diff --git a/net/dcsctp/packet/parameter/supported_extensions_parameter_test.cc b/net/dcsctp/packet/parameter/supported_extensions_parameter_test.cc
index e76d3b0fe2..76f6376cb5 100644
--- a/net/dcsctp/packet/parameter/supported_extensions_parameter_test.cc
+++ b/net/dcsctp/packet/parameter/supported_extensions_parameter_test.cc
@@ -12,7 +12,6 @@
 #include <cstdint>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/testing/testing_macros.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
diff --git a/net/dcsctp/packet/parameter/zero_checksum_acceptable_chunk_parameter.cc b/net/dcsctp/packet/parameter/zero_checksum_acceptable_chunk_parameter.cc
index adad39377a..23f1b39541 100644
--- a/net/dcsctp/packet/parameter/zero_checksum_acceptable_chunk_parameter.cc
+++ b/net/dcsctp/packet/parameter/zero_checksum_acceptable_chunk_parameter.cc
@@ -11,10 +11,10 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "net/dcsctp/packet/bounded_byte_writer.h"
 #include "net/dcsctp/public/types.h"
@@ -32,8 +32,7 @@ namespace dcsctp {
 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
 std::optional<ZeroChecksumAcceptableChunkParameter>
-ZeroChecksumAcceptableChunkParameter::Parse(
-    webrtc::ArrayView<const uint8_t> data) {
+ZeroChecksumAcceptableChunkParameter::Parse(std::span<const uint8_t> data) {
   std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
   if (!reader.has_value()) {
     return std::nullopt;
diff --git a/net/dcsctp/packet/parameter/zero_checksum_acceptable_chunk_parameter.h b/net/dcsctp/packet/parameter/zero_checksum_acceptable_chunk_parameter.h
index 139b6f612f..a9084a8aab 100644
--- a/net/dcsctp/packet/parameter/zero_checksum_acceptable_chunk_parameter.h
+++ b/net/dcsctp/packet/parameter/zero_checksum_acceptable_chunk_parameter.h
@@ -13,10 +13,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/parameter/parameter.h"
 #include "net/dcsctp/packet/tlv_trait.h"
 #include "net/dcsctp/public/types.h"
@@ -42,7 +42,7 @@ class ZeroChecksumAcceptableChunkParameter
       : error_detection_method_(error_detection_method) {}
 
   static std::optional<ZeroChecksumAcceptableChunkParameter> Parse(
-      webrtc::ArrayView<const uint8_t> data);
+      std::span<const uint8_t> data);
 
   void SerializeTo(std::vector<uint8_t>& out) const override;
   std::string ToString() const override;
diff --git a/net/dcsctp/packet/sctp_packet.cc b/net/dcsctp/packet/sctp_packet.cc
index 6a6802b5a6..1b5893a579 100644
--- a/net/dcsctp/packet/sctp_packet.cc
+++ b/net/dcsctp/packet/sctp_packet.cc
@@ -12,10 +12,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/common/math.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
@@ -105,9 +105,8 @@ std::vector<uint8_t> SctpPacket::Builder::Build(bool write_checksum) {
   return out;
 }
 
-std::optional<SctpPacket> SctpPacket::Parse(
-    webrtc::ArrayView<const uint8_t> data,
-    const DcSctpOptions& options) {
+std::optional<SctpPacket> SctpPacket::Parse(std::span<const uint8_t> data,
+                                            const DcSctpOptions& options) {
   if (data.size() < kHeaderSize + kChunkTlvHeaderSize ||
       data.size() > kMaxUdpPacketSize) {
     RTC_DLOG(LS_WARNING) << "Invalid packet size";
@@ -162,8 +161,8 @@ std::optional<SctpPacket> SctpPacket::Parse(
 
   std::vector<ChunkDescriptor> descriptors;
   descriptors.reserve(kExpectedDescriptorCount);
-  webrtc::ArrayView<const uint8_t> descriptor_data =
-      webrtc::ArrayView<const uint8_t>(data_copy).subspan(kHeaderSize);
+  std::span<const uint8_t> descriptor_data =
+      std::span<const uint8_t>(data_copy).subspan(kHeaderSize);
   while (!descriptor_data.empty()) {
     if (descriptor_data.size() < kChunkTlvHeaderSize) {
       RTC_DLOG(LS_WARNING) << "Too small chunk";
diff --git a/net/dcsctp/packet/sctp_packet.h b/net/dcsctp/packet/sctp_packet.h
index bd602eddcc..2c4db51b6c 100644
--- a/net/dcsctp/packet/sctp_packet.h
+++ b/net/dcsctp/packet/sctp_packet.h
@@ -13,10 +13,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/chunk/chunk.h"
 #include "net/dcsctp/public/dcsctp_options.h"
@@ -38,13 +38,11 @@ class SctpPacket {
   static constexpr size_t kHeaderSize = 12;
 
   struct ChunkDescriptor {
-    ChunkDescriptor(uint8_t type,
-                    uint8_t flags,
-                    webrtc::ArrayView<const uint8_t> data)
+    ChunkDescriptor(uint8_t type, uint8_t flags, std::span<const uint8_t> data)
         : type(type), flags(flags), data(data) {}
     uint8_t type;
     uint8_t flags;
-    webrtc::ArrayView<const uint8_t> data;
+    std::span<const uint8_t> data;
   };
 
   SctpPacket(SctpPacket&& other) = default;
@@ -87,16 +85,14 @@ class SctpPacket {
   };
 
   // Parses `data` as an SCTP packet and returns it if it validates.
-  static std::optional<SctpPacket> Parse(webrtc::ArrayView<const uint8_t> data,
+  static std::optional<SctpPacket> Parse(std::span<const uint8_t> data,
                                          const DcSctpOptions& options);
 
   // Returns the SCTP common header.
   const CommonHeader& common_header() const { return common_header_; }
 
   // Returns the chunks (types and offsets) within the packet.
-  webrtc::ArrayView<const ChunkDescriptor> descriptors() const {
-    return descriptors_;
-  }
+  std::span<const ChunkDescriptor> descriptors() const { return descriptors_; }
 
  private:
   SctpPacket(const CommonHeader& common_header,
diff --git a/net/dcsctp/packet/sctp_packet_test.cc b/net/dcsctp/packet/sctp_packet_test.cc
index 2209f21bea..2ccaa0ab6d 100644
--- a/net/dcsctp/packet/sctp_packet_test.cc
+++ b/net/dcsctp/packet/sctp_packet_test.cc
@@ -13,7 +13,6 @@
 #include <cstdint>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/common/math.h"
 #include "net/dcsctp/packet/chunk/abort_chunk.h"
diff --git a/net/dcsctp/packet/tlv_trait.h b/net/dcsctp/packet/tlv_trait.h
index e032c8e434..ef18e6be8c 100644
--- a/net/dcsctp/packet/tlv_trait.h
+++ b/net/dcsctp/packet/tlv_trait.h
@@ -14,9 +14,9 @@
 #include <cstdint>
 #include <cstring>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "net/dcsctp/packet/bounded_byte_writer.h"
 
@@ -77,7 +77,7 @@ class TLVTrait {
   // Validates the data with regards to size, alignment and type.
   // If valid, returns a bounded buffer.
   static std::optional<BoundedByteReader<Config::kHeaderSize>> ParseTLV(
-      webrtc::ArrayView<const uint8_t> data) {
+      std::span<const uint8_t> data) {
     if (data.size() < Config::kHeaderSize) {
       tlv_trait_impl::ReportInvalidSize(data.size(), Config::kHeaderSize);
       return std::nullopt;
@@ -133,7 +133,7 @@ class TLVTrait {
     out.resize(offset + size);
 
     BoundedByteWriter<kTlvHeaderSize> tlv_header(
-        webrtc::ArrayView<uint8_t>(out.data() + offset, kTlvHeaderSize));
+        std::span<uint8_t>(out.data() + offset, kTlvHeaderSize));
     if (Config::kTypeSizeInBytes == 1) {
       tlv_header.template Store8<0>(static_cast<uint8_t>(Config::kType));
     } else {
@@ -142,7 +142,7 @@ class TLVTrait {
     tlv_header.template Store16<2>(size);
 
     return BoundedByteWriter<Config::kHeaderSize>(
-        webrtc::ArrayView<uint8_t>(out.data() + offset, size));
+        std::span<uint8_t>(out.data() + offset, size));
   }
 
  private:
diff --git a/net/dcsctp/packet/tlv_trait_test.cc b/net/dcsctp/packet/tlv_trait_test.cc
index 1c43ee65b2..fd992d33e3 100644
--- a/net/dcsctp/packet/tlv_trait_test.cc
+++ b/net/dcsctp/packet/tlv_trait_test.cc
@@ -12,9 +12,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "net/dcsctp/packet/bounded_byte_writer.h"
 #include "test/gmock.h"
@@ -44,11 +44,11 @@ class OneByteChunk : public TLVTrait<OneByteTypeConfig> {
     writer.Store16<10>(0x0708);
 
     uint8_t variable_data[kVariableSize] = {0xDE, 0xAD, 0xBE, 0xEF};
-    writer.CopyToVariableData(webrtc::ArrayView<const uint8_t>(variable_data));
+    writer.CopyToVariableData(std::span<const uint8_t>(variable_data));
   }
 
   static std::optional<BoundedByteReader<OneByteTypeConfig::kHeaderSize>> Parse(
-      webrtc::ArrayView<const uint8_t> data) {
+      std::span<const uint8_t> data) {
     return ParseTLV(data);
   }
 };
@@ -93,11 +93,11 @@ class TwoByteChunk : public TLVTrait<TwoByteTypeConfig> {
     writer.Store32<4>(0x01020304U);
 
     uint8_t variable_data[] = {0x05, 0x06, 0x07, 0x08, 0xDE, 0xAD, 0xBE, 0xEF};
-    writer.CopyToVariableData(webrtc::ArrayView<const uint8_t>(variable_data));
+    writer.CopyToVariableData(std::span<const uint8_t>(variable_data));
   }
 
   static std::optional<BoundedByteReader<TwoByteTypeConfig::kHeaderSize>> Parse(
-      webrtc::ArrayView<const uint8_t> data) {
+      std::span<const uint8_t> data) {
     return ParseTLV(data);
   }
 };
diff --git a/net/dcsctp/public/BUILD.gn b/net/dcsctp/public/BUILD.gn
index a6eb3df4f1..af542e7792 100644
--- a/net/dcsctp/public/BUILD.gn
+++ b/net/dcsctp/public/BUILD.gn
@@ -10,7 +10,6 @@ import("../../../webrtc.gni")
 
 rtc_source_set("types") {
   deps = [
-    "../../../api:array_view",
     "../../../api/units:time_delta",
     "../../../rtc_base:strong_alias",
   ]
@@ -24,7 +23,6 @@ rtc_source_set("types") {
 rtc_library("socket") {
   deps = [
     ":types",
-    "../../../api:array_view",
     "../../../api/task_queue",
     "../../../api/units:timestamp",
     "../../../rtc_base:strong_alias",
@@ -63,7 +61,6 @@ rtc_source_set("mocks") {
     ":factory",
     ":socket",
     ":types",
-    "../../../api:array_view",
     "../../../test:test_support",
     "//third_party/abseil-cpp/absl/strings:string_view",
   ]
@@ -73,7 +70,6 @@ rtc_library("utils") {
   deps = [
     ":socket",
     ":types",
-    "../../../api:array_view",
     "../../../rtc_base:logging",
     "../../../rtc_base:stringutils",
     "../../../rtc_base:text2pcap",
diff --git a/net/dcsctp/public/dcsctp_message.h b/net/dcsctp/public/dcsctp_message.h
index bcf4486f2d..6d002f0291 100644
--- a/net/dcsctp/public/dcsctp_message.h
+++ b/net/dcsctp/public/dcsctp_message.h
@@ -11,10 +11,10 @@
 #define NET_DCSCTP_PUBLIC_DCSCTP_MESSAGE_H_
 
 #include <cstdint>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/public/types.h"
 
 namespace dcsctp {
@@ -39,7 +39,7 @@ class DcSctpMessage {
   PPID ppid() const { return ppid_; }
 
   // The payload of the message.
-  webrtc::ArrayView<const uint8_t> payload() const { return payload_; }
+  std::span<const uint8_t> payload() const { return payload_; }
 
   // When destructing the message, extracts the payload.
   std::vector<uint8_t> ReleasePayload() && { return std::move(payload_); }
diff --git a/net/dcsctp/public/dcsctp_socket.h b/net/dcsctp/public/dcsctp_socket.h
index 24a50ea0cd..d399197259 100644
--- a/net/dcsctp/public/dcsctp_socket.h
+++ b/net/dcsctp/public/dcsctp_socket.h
@@ -14,11 +14,11 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <vector>
 
 #include "absl/base/attributes.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/task_queue/task_queue_base.h"
 #include "api/units/timestamp.h"
 #include "net/dcsctp/public/dcsctp_handover_state.h"
@@ -287,14 +287,13 @@ class DcSctpSocketCallbacks {
   //
   // Note that it's NOT ALLOWED to call into this library from within this
   // callback.
-  virtual void SendPacket(webrtc::ArrayView<const uint8_t> /* data */) {}
+  virtual void SendPacket(std::span<const uint8_t> /* data */) {}
 
   // Called when the library wants the packet serialized as `data` to be sent.
   //
   // Note that it's NOT ALLOWED to call into this library from within this
   // callback.
-  virtual SendPacketStatus SendPacketWithStatus(
-      webrtc::ArrayView<const uint8_t> data) {
+  virtual SendPacketStatus SendPacketWithStatus(std::span<const uint8_t> data) {
     SendPacket(data);
     return SendPacketStatus::kSuccess;
   }
@@ -410,22 +409,21 @@ class DcSctpSocketCallbacks {
   // Indicates that a stream reset request has failed.
   //
   // It is allowed to call into this library from within this callback.
-  virtual void OnStreamsResetFailed(
-      webrtc::ArrayView<const StreamID> outgoing_streams,
-      absl::string_view reason) = 0;
+  virtual void OnStreamsResetFailed(std::span<const StreamID> outgoing_streams,
+                                    absl::string_view reason) = 0;
 
   // Indicates that a stream reset request has been performed.
   //
   // It is allowed to call into this library from within this callback.
   virtual void OnStreamsResetPerformed(
-      webrtc::ArrayView<const StreamID> outgoing_streams) = 0;
+      std::span<const StreamID> outgoing_streams) = 0;
 
   // When a peer has reset some of its outgoing streams, this will be called. An
   // empty list indicates that all streams have been reset.
   //
   // It is allowed to call into this library from within this callback.
   virtual void OnIncomingStreamsReset(
-      webrtc::ArrayView<const StreamID> incoming_streams) = 0;
+      std::span<const StreamID> incoming_streams) = 0;
 
   // Will be called when the amount of data buffered to be sent falls to or
   // below the threshold set when calling `SetBufferedAmountLowThreshold`.
@@ -531,7 +529,7 @@ class DcSctpSocketInterface {
   virtual ~DcSctpSocketInterface() = default;
 
   // To be called when an incoming SCTP packet is to be processed.
-  virtual void ReceivePacket(webrtc::ArrayView<const uint8_t> data) = 0;
+  virtual void ReceivePacket(std::span<const uint8_t> data) = 0;
 
   // Returns the number of received messages that can be retrieved by calling
   // calling `::GetNextMessage`.
@@ -558,9 +556,8 @@ class DcSctpSocketInterface {
   // Finishes the out-of-bands connection sequence and returns `true` if this
   // was successful. This will also trigger
   // `DcSctpSocketCallbacks::OnConnected`.
-  virtual bool ConnectWithConnectionToken(
-      webrtc::ArrayView<const uint8_t> my_data,
-      webrtc::ArrayView<const uint8_t> peer_data) {
+  virtual bool ConnectWithConnectionToken(std::span<const uint8_t> my_data,
+                                          std::span<const uint8_t> peer_data) {
     return false;
   }
 
@@ -615,9 +612,8 @@ class DcSctpSocketInterface {
   //
   // This has identical semantics to Send, except that it may coalesce many
   // messages into a single SCTP packet if they would fit.
-  virtual std::vector<SendStatus> SendMany(
-      webrtc::ArrayView<DcSctpMessage> messages,
-      const SendOptions& send_options) = 0;
+  virtual std::vector<SendStatus> SendMany(std::span<DcSctpMessage> messages,
+                                           const SendOptions& send_options) = 0;
 
   // Resetting streams is an asynchronous operation and the results will
   // be notified using `DcSctpSocketCallbacks::OnStreamsResetDone()` on success
@@ -635,7 +631,7 @@ class DcSctpSocketInterface {
   // supports stream resetting. Calling this method on e.g. a closed association
   // or streams that don't support resetting will not perform any operation.
   virtual ResetStreamsStatus ResetStreams(
-      webrtc::ArrayView<const StreamID> outgoing_streams) = 0;
+      std::span<const StreamID> outgoing_streams) = 0;
 
   // Returns the number of bytes of data currently queued to be sent on a given
   // stream.
diff --git a/net/dcsctp/public/mock_dcsctp_socket.h b/net/dcsctp/public/mock_dcsctp_socket.h
index ce8de2ac32..8a49a94d74 100644
--- a/net/dcsctp/public/mock_dcsctp_socket.h
+++ b/net/dcsctp/public/mock_dcsctp_socket.h
@@ -13,9 +13,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/public/dcsctp_handover_state.h"
 #include "net/dcsctp/public/dcsctp_message.h"
 #include "net/dcsctp/public/dcsctp_options.h"
@@ -27,10 +27,7 @@ namespace dcsctp {
 
 class MockDcSctpSocket : public DcSctpSocketInterface {
  public:
-  MOCK_METHOD(void,
-              ReceivePacket,
-              (webrtc::ArrayView<const uint8_t> data),
-              (override));
+  MOCK_METHOD(void, ReceivePacket, (std::span<const uint8_t> data), (override));
 
   MOCK_METHOD(size_t, MessagesReady, (), (const, override));
   MOCK_METHOD(std::optional<DcSctpMessage>, GetNextMessage, (), (override));
@@ -40,8 +37,7 @@ class MockDcSctpSocket : public DcSctpSocketInterface {
   MOCK_METHOD(void, Connect, (), (override));
   MOCK_METHOD(bool,
               ConnectWithConnectionToken,
-              (webrtc::ArrayView<const uint8_t>,
-               webrtc::ArrayView<const uint8_t>),
+              (std::span<const uint8_t>, std::span<const uint8_t>),
               (override));
 
   MOCK_METHOD(void,
@@ -76,13 +72,13 @@ class MockDcSctpSocket : public DcSctpSocketInterface {
 
   MOCK_METHOD(std::vector<SendStatus>,
               SendMany,
-              (webrtc::ArrayView<DcSctpMessage> messages,
+              (std::span<DcSctpMessage> messages,
                const SendOptions& send_options),
               (override));
 
   MOCK_METHOD(ResetStreamsStatus,
               ResetStreams,
-              (webrtc::ArrayView<const StreamID> outgoing_streams),
+              (std::span<const StreamID> outgoing_streams),
               (override));
 
   MOCK_METHOD(size_t, buffered_amount, (StreamID stream_id), (const, override));
diff --git a/net/dcsctp/public/packet_observer.h b/net/dcsctp/public/packet_observer.h
index c0fa6b937d..68affb3f38 100644
--- a/net/dcsctp/public/packet_observer.h
+++ b/net/dcsctp/public/packet_observer.h
@@ -11,8 +11,8 @@
 #define NET_DCSCTP_PUBLIC_PACKET_OBSERVER_H_
 
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "net/dcsctp/public/types.h"
 
 namespace dcsctp {
@@ -24,13 +24,12 @@ class PacketObserver {
   virtual ~PacketObserver() = default;
   // Called when a packet is sent, with the current time (in milliseconds) as
   // `now`, and the packet payload as `payload`.
-  virtual void OnSentPacket(TimeMs now,
-                            webrtc::ArrayView<const uint8_t> payload) = 0;
+  virtual void OnSentPacket(TimeMs now, std::span<const uint8_t> payload) = 0;
 
   // Called when a packet is received, with the current time (in milliseconds)
   // as `now`, and the packet payload as `payload`.
   virtual void OnReceivedPacket(TimeMs now,
-                                webrtc::ArrayView<const uint8_t> payload) = 0;
+                                std::span<const uint8_t> payload) = 0;
 };
 }  // namespace dcsctp
 
diff --git a/net/dcsctp/public/text_pcap_packet_observer.cc b/net/dcsctp/public/text_pcap_packet_observer.cc
index 7fed9717c6..93092bd298 100644
--- a/net/dcsctp/public/text_pcap_packet_observer.cc
+++ b/net/dcsctp/public/text_pcap_packet_observer.cc
@@ -10,17 +10,16 @@
 #include "net/dcsctp/public/text_pcap_packet_observer.h"
 
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "net/dcsctp/public/types.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/text2pcap.h"
 
 namespace dcsctp {
 
-void TextPcapPacketObserver::OnSentPacket(
-    dcsctp::TimeMs now,
-    webrtc::ArrayView<const uint8_t> payload) {
+void TextPcapPacketObserver::OnSentPacket(dcsctp::TimeMs now,
+                                          std::span<const uint8_t> payload) {
   RTC_LOG(LS_VERBOSE) << "\n"
                       << webrtc::Text2Pcap::DumpPacket(/*outbound=*/true,
                                                        payload, *now)
@@ -29,7 +28,7 @@ void TextPcapPacketObserver::OnSentPacket(
 
 void TextPcapPacketObserver::OnReceivedPacket(
     dcsctp::TimeMs now,
-    webrtc::ArrayView<const uint8_t> payload) {
+    std::span<const uint8_t> payload) {
   RTC_LOG(LS_VERBOSE) << "\n"
                       << webrtc::Text2Pcap::DumpPacket(/*outbound=*/false,
                                                        payload, *now)
diff --git a/net/dcsctp/public/text_pcap_packet_observer.h b/net/dcsctp/public/text_pcap_packet_observer.h
index c368b3da74..8c8d07c967 100644
--- a/net/dcsctp/public/text_pcap_packet_observer.h
+++ b/net/dcsctp/public/text_pcap_packet_observer.h
@@ -11,10 +11,10 @@
 #define NET_DCSCTP_PUBLIC_TEXT_PCAP_PACKET_OBSERVER_H_
 
 #include <cstdint>
+#include <span>
 #include <string>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "net/dcsctp/public/packet_observer.h"
 #include "net/dcsctp/public/types.h"
 
@@ -27,17 +27,17 @@ class TextPcapPacketObserver : public dcsctp::PacketObserver {
 
   // Implementation of `dcsctp::PacketObserver`.
   void OnSentPacket(dcsctp::TimeMs now,
-                    webrtc::ArrayView<const uint8_t> payload) override;
+                    std::span<const uint8_t> payload) override;
 
   void OnReceivedPacket(dcsctp::TimeMs now,
-                        webrtc::ArrayView<const uint8_t> payload) override;
+                        std::span<const uint8_t> payload) override;
 
   // Prints a packet to the log. Exposed to allow it to be used in compatibility
   // tests suites that don't use PacketObserver.
   static void PrintPacket(absl::string_view prefix,
                           absl::string_view socket_name,
                           dcsctp::TimeMs now,
-                          webrtc::ArrayView<const uint8_t> payload);
+                          std::span<const uint8_t> payload);
 
  private:
   const std::string name_;
diff --git a/net/dcsctp/rx/BUILD.gn b/net/dcsctp/rx/BUILD.gn
index 3a3e91e3bb..6545ac71ea 100644
--- a/net/dcsctp/rx/BUILD.gn
+++ b/net/dcsctp/rx/BUILD.gn
@@ -28,7 +28,6 @@ rtc_library("data_tracker") {
 
 rtc_source_set("reassembly_streams") {
   deps = [
-    "../../../api:array_view",
     "../common:sequence_numbers",
     "../packet:chunk",
     "../packet:data",
@@ -42,7 +41,6 @@ rtc_source_set("reassembly_streams") {
 rtc_library("interleaved_reassembly_streams") {
   deps = [
     ":reassembly_streams",
-    "../../../api:array_view",
     "../../../rtc_base:checks",
     "../../../rtc_base:logging",
     "../common:internal_types",
@@ -62,7 +60,6 @@ rtc_library("interleaved_reassembly_streams") {
 rtc_library("traditional_reassembly_streams") {
   deps = [
     ":reassembly_streams",
-    "../../../api:array_view",
     "../../../rtc_base:checks",
     "../../../rtc_base:logging",
     "../common:internal_types",
@@ -85,7 +82,6 @@ rtc_library("reassembly_queue") {
     ":interleaved_reassembly_streams",
     ":reassembly_streams",
     ":traditional_reassembly_streams",
-    "../../../api:array_view",
     "../../../rtc_base:checks",
     "../../../rtc_base:logging",
     "../../../rtc_base:stringutils",
@@ -115,7 +111,6 @@ if (rtc_include_tests) {
       ":reassembly_queue",
       ":reassembly_streams",
       ":traditional_reassembly_streams",
-      "../../../api:array_view",
       "../../../api/task_queue",
       "../../../api/units:time_delta",
       "../../../api/units:timestamp",
diff --git a/net/dcsctp/rx/interleaved_reassembly_streams.cc b/net/dcsctp/rx/interleaved_reassembly_streams.cc
index 0e4dbe21fe..6950082501 100644
--- a/net/dcsctp/rx/interleaved_reassembly_streams.cc
+++ b/net/dcsctp/rx/interleaved_reassembly_streams.cc
@@ -13,13 +13,13 @@
 #include <cstdint>
 #include <iterator>
 #include <map>
+#include <span>
 #include <tuple>
 #include <utility>
 #include <vector>
 
 #include "absl/algorithm/container.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/common/sequence_numbers.h"
 #include "net/dcsctp/packet/chunk/forward_tsn_common.h"
@@ -224,8 +224,7 @@ int InterleavedReassemblyStreams::Add(UnwrappedTSN tsn, Data data) {
 
 size_t InterleavedReassemblyStreams::HandleForwardTsn(
     UnwrappedTSN /* new_cumulative_ack_tsn */,
-    webrtc::ArrayView<const AnyForwardTsnChunk::SkippedStream>
-        skipped_streams) {
+    std::span<const AnyForwardTsnChunk::SkippedStream> skipped_streams) {
   size_t removed_bytes = 0;
   for (const auto& skipped : skipped_streams) {
     removed_bytes +=
@@ -236,7 +235,7 @@ size_t InterleavedReassemblyStreams::HandleForwardTsn(
 }
 
 void InterleavedReassemblyStreams::ResetStreams(
-    webrtc::ArrayView<const StreamID> stream_ids) {
+    std::span<const StreamID> stream_ids) {
   if (stream_ids.empty()) {
     for (auto& entry : streams_) {
       entry.second.Reset();
diff --git a/net/dcsctp/rx/interleaved_reassembly_streams.h b/net/dcsctp/rx/interleaved_reassembly_streams.h
index 45d661ed73..c956b1b9e3 100644
--- a/net/dcsctp/rx/interleaved_reassembly_streams.h
+++ b/net/dcsctp/rx/interleaved_reassembly_streams.h
@@ -12,11 +12,11 @@
 
 #include <cstddef>
 #include <map>
+#include <span>
 #include <tuple>
 #include <utility>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/common/sequence_numbers.h"
 #include "net/dcsctp/packet/chunk/forward_tsn_common.h"
@@ -36,12 +36,11 @@ class InterleavedReassemblyStreams : public ReassemblyStreams {
 
   int Add(UnwrappedTSN tsn, Data data) override;
 
-  size_t HandleForwardTsn(
-      UnwrappedTSN new_cumulative_ack_tsn,
-      webrtc::ArrayView<const AnyForwardTsnChunk::SkippedStream>
-          skipped_streams) override;
+  size_t HandleForwardTsn(UnwrappedTSN new_cumulative_ack_tsn,
+                          std::span<const AnyForwardTsnChunk::SkippedStream>
+                              skipped_streams) override;
 
-  void ResetStreams(webrtc::ArrayView<const StreamID> stream_ids) override;
+  void ResetStreams(std::span<const StreamID> stream_ids) override;
 
   HandoverReadinessStatus GetHandoverReadiness() const override;
   void AddHandoverState(DcSctpSocketHandoverState& state) override;
diff --git a/net/dcsctp/rx/reassembly_queue.cc b/net/dcsctp/rx/reassembly_queue.cc
index 5607221ca4..1d28f686a0 100644
--- a/net/dcsctp/rx/reassembly_queue.cc
+++ b/net/dcsctp/rx/reassembly_queue.cc
@@ -12,11 +12,11 @@
 #include <cstddef>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/common/sequence_numbers.h"
 #include "net/dcsctp/packet/chunk/forward_tsn_common.h"
@@ -56,8 +56,7 @@ ReassemblyQueue::ReassemblyQueue(absl::string_view log_prefix,
       watermark_bytes_(max_size_bytes * kHighWatermarkLimit),
       streams_(CreateStreams(
           log_prefix_,
-          [this](webrtc::ArrayView<const UnwrappedTSN> tsns,
-                 DcSctpMessage message) {
+          [this](std::span<const UnwrappedTSN> tsns, DcSctpMessage message) {
             AddReassembledMessage(tsns, std::move(message));
           },
           use_message_interleaving)) {}
@@ -110,7 +109,7 @@ void ReassemblyQueue::Add(TSN tsn, Data data) {
 }
 
 void ReassemblyQueue::ResetStreamsAndLeaveDeferredReset(
-    webrtc::ArrayView<const StreamID> stream_ids) {
+    std::span<const StreamID> stream_ids) {
   RTC_DLOG(LS_VERBOSE) << log_prefix_ << "Resetting streams: ["
                        << webrtc::StrJoin(stream_ids, ",",
                                           [](webrtc::StringBuilder& sb,
@@ -141,9 +140,8 @@ void ReassemblyQueue::ResetStreamsAndLeaveDeferredReset(
   RTC_DCHECK(IsConsistent());
 }
 
-void ReassemblyQueue::EnterDeferredReset(
-    TSN sender_last_assigned_tsn,
-    webrtc::ArrayView<const StreamID> streams) {
+void ReassemblyQueue::EnterDeferredReset(TSN sender_last_assigned_tsn,
+                                         std::span<const StreamID> streams) {
   if (!deferred_reset_streams_.has_value()) {
     RTC_DLOG(LS_VERBOSE) << log_prefix_
                          << "Entering deferred reset; sender_last_assigned_tsn="
@@ -165,9 +163,8 @@ std::optional<DcSctpMessage> ReassemblyQueue::GetNextMessage() {
   return ret;
 }
 
-void ReassemblyQueue::AddReassembledMessage(
-    webrtc::ArrayView<const UnwrappedTSN> tsns,
-    DcSctpMessage message) {
+void ReassemblyQueue::AddReassembledMessage(std::span<const UnwrappedTSN> tsns,
+                                            DcSctpMessage message) {
   RTC_DLOG(LS_VERBOSE) << log_prefix_ << "Assembled message from TSN=["
                        << webrtc::StrJoin(
                               tsns, ",",
@@ -184,8 +181,7 @@ void ReassemblyQueue::AddReassembledMessage(
 
 void ReassemblyQueue::HandleForwardTsn(
     TSN new_cumulative_tsn,
-    webrtc::ArrayView<const AnyForwardTsnChunk::SkippedStream>
-        skipped_streams) {
+    std::span<const AnyForwardTsnChunk::SkippedStream> skipped_streams) {
   UnwrappedTSN tsn = tsn_unwrapper_.Unwrap(new_cumulative_tsn);
 
   if (deferred_reset_streams_.has_value() &&
diff --git a/net/dcsctp/rx/reassembly_queue.h b/net/dcsctp/rx/reassembly_queue.h
index 504584243c..f35ab565f9 100644
--- a/net/dcsctp/rx/reassembly_queue.h
+++ b/net/dcsctp/rx/reassembly_queue.h
@@ -14,12 +14,12 @@
 #include <deque>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/common/sequence_numbers.h"
 #include "net/dcsctp/packet/chunk/forward_tsn_common.h"
@@ -93,17 +93,15 @@ class ReassemblyQueue {
   // partial reliability.
   void HandleForwardTsn(
       TSN new_cumulative_tsn,
-      webrtc::ArrayView<const AnyForwardTsnChunk::SkippedStream>
-          skipped_streams);
+      std::span<const AnyForwardTsnChunk::SkippedStream> skipped_streams);
 
   // Resets the provided streams and leaves deferred reset processing, if
   // enabled.
-  void ResetStreamsAndLeaveDeferredReset(
-      webrtc::ArrayView<const StreamID> stream_ids);
+  void ResetStreamsAndLeaveDeferredReset(std::span<const StreamID> stream_ids);
 
   // Enters deferred reset processing.
   void EnterDeferredReset(TSN sender_last_assigned_tsn,
-                          webrtc::ArrayView<const StreamID> streams);
+                          std::span<const StreamID> streams);
 
   // The number of payload bytes that have been queued. Note that the actual
   // memory usage is higher due to additional overhead of tracking received
@@ -142,7 +140,7 @@ class ReassemblyQueue {
   };
 
   bool IsConsistent() const;
-  void AddReassembledMessage(webrtc::ArrayView<const UnwrappedTSN> tsns,
+  void AddReassembledMessage(std::span<const UnwrappedTSN> tsns,
                              DcSctpMessage message);
 
   const absl::string_view log_prefix_;
diff --git a/net/dcsctp/rx/reassembly_queue_test.cc b/net/dcsctp/rx/reassembly_queue_test.cc
index 4d28b5155d..5191054679 100644
--- a/net/dcsctp/rx/reassembly_queue_test.cc
+++ b/net/dcsctp/rx/reassembly_queue_test.cc
@@ -14,9 +14,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <iterator>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/handover_testing.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/chunk/forward_tsn_common.h"
@@ -103,7 +103,7 @@ TEST_F(ReassemblyQueueTest, SingleUnorderedChunkMessage) {
 
 TEST_F(ReassemblyQueueTest, LargeUnorderedChunkAllPermutations) {
   std::vector<uint32_t> tsns = {10, 11, 12, 13};
-  webrtc::ArrayView<const uint8_t> payload(kLongPayload);
+  std::span<const uint8_t> payload(kLongPayload);
   do {
     ReassemblyQueue reasm("log: ", kBufferSize);
 
@@ -138,7 +138,7 @@ TEST_F(ReassemblyQueueTest, SingleOrderedChunkMessage) {
 
 TEST_F(ReassemblyQueueTest, ManySmallOrderedMessages) {
   std::vector<uint32_t> tsns = {10, 11, 12, 13};
-  webrtc::ArrayView<const uint8_t> payload(kLongPayload);
+  std::span<const uint8_t> payload(kLongPayload);
   do {
     ReassemblyQueue reasm("log: ", kBufferSize);
     for (size_t i = 0; i < tsns.size(); i++) {
@@ -348,7 +348,7 @@ TEST_F(ReassemblyQueueTest, UnorderedInterleavedMessagesAllPermutations) {
   StreamID stream_ids[] = {StreamID(1), StreamID(2), StreamID(1),
                            StreamID(1), StreamID(2), StreamID(2)};
   FSN fsns[] = {FSN(0), FSN(0), FSN(1), FSN(2), FSN(1), FSN(2)};
-  webrtc::ArrayView<const uint8_t> payload(kSixBytePayload);
+  std::span<const uint8_t> payload(kSixBytePayload);
   do {
     ReassemblyQueue reasm("log: ", kBufferSize,
                           /*use_message_interleaving=*/true);
diff --git a/net/dcsctp/rx/reassembly_streams.h b/net/dcsctp/rx/reassembly_streams.h
index d801ef729e..5d10a08026 100644
--- a/net/dcsctp/rx/reassembly_streams.h
+++ b/net/dcsctp/rx/reassembly_streams.h
@@ -12,8 +12,8 @@
 
 #include <cstddef>
 #include <functional>
+#include <span>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/sequence_numbers.h"
 #include "net/dcsctp/packet/chunk/forward_tsn_common.h"
 #include "net/dcsctp/packet/data.h"
@@ -43,7 +43,7 @@ class ReassemblyStreams {
   // message has been assembled as well as indicating from which TSNs this
   // message was assembled from.
   using OnAssembledMessage =
-      std::function<void(webrtc::ArrayView<const UnwrappedTSN> tsns,
+      std::function<void(std::span<const UnwrappedTSN> tsns,
                          DcSctpMessage message)>;
 
   virtual ~ReassemblyStreams() = default;
@@ -68,13 +68,12 @@ class ReassemblyStreams {
   // this operation.
   virtual size_t HandleForwardTsn(
       UnwrappedTSN new_cumulative_ack_tsn,
-      webrtc::ArrayView<const AnyForwardTsnChunk::SkippedStream>
-          skipped_streams) = 0;
+      std::span<const AnyForwardTsnChunk::SkippedStream> skipped_streams) = 0;
 
   // Called for incoming (possibly deferred) RE_CONFIG chunks asking for
   // either a few streams, or all streams (when the list is empty) to be
   // reset - to have their next SSN or Message ID to be zero.
-  virtual void ResetStreams(webrtc::ArrayView<const StreamID> stream_ids) = 0;
+  virtual void ResetStreams(std::span<const StreamID> stream_ids) = 0;
 
   virtual HandoverReadinessStatus GetHandoverReadiness() const = 0;
   virtual void AddHandoverState(DcSctpSocketHandoverState& state) = 0;
diff --git a/net/dcsctp/rx/traditional_reassembly_streams.cc b/net/dcsctp/rx/traditional_reassembly_streams.cc
index cae1d7dc61..53932503de 100644
--- a/net/dcsctp/rx/traditional_reassembly_streams.cc
+++ b/net/dcsctp/rx/traditional_reassembly_streams.cc
@@ -15,13 +15,13 @@
 #include <map>
 #include <numeric>
 #include <optional>
+#include <span>
 #include <tuple>
 #include <utility>
 #include <vector>
 
 #include "absl/algorithm/container.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/common/sequence_numbers.h"
 #include "net/dcsctp/packet/chunk/forward_tsn_common.h"
@@ -288,8 +288,7 @@ int TraditionalReassemblyStreams::Add(UnwrappedTSN tsn, Data data) {
 
 size_t TraditionalReassemblyStreams::HandleForwardTsn(
     UnwrappedTSN new_cumulative_ack_tsn,
-    webrtc::ArrayView<const AnyForwardTsnChunk::SkippedStream>
-        skipped_streams) {
+    std::span<const AnyForwardTsnChunk::SkippedStream> skipped_streams) {
   size_t bytes_removed = 0;
   // The `skipped_streams` only cover ordered messages - need to
   // iterate all unordered streams manually to remove those chunks.
@@ -307,7 +306,7 @@ size_t TraditionalReassemblyStreams::HandleForwardTsn(
 }
 
 void TraditionalReassemblyStreams::ResetStreams(
-    webrtc::ArrayView<const StreamID> stream_ids) {
+    std::span<const StreamID> stream_ids) {
   if (stream_ids.empty()) {
     for (auto& [stream_id, stream] : ordered_streams_) {
       RTC_DLOG(LS_VERBOSE) << log_prefix_
diff --git a/net/dcsctp/rx/traditional_reassembly_streams.h b/net/dcsctp/rx/traditional_reassembly_streams.h
index dca34fbed5..0338291fcf 100644
--- a/net/dcsctp/rx/traditional_reassembly_streams.h
+++ b/net/dcsctp/rx/traditional_reassembly_streams.h
@@ -12,9 +12,9 @@
 
 #include <cstddef>
 #include <map>
+#include <span>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/common/sequence_numbers.h"
 #include "net/dcsctp/packet/chunk/forward_tsn_common.h"
@@ -35,12 +35,11 @@ class TraditionalReassemblyStreams : public ReassemblyStreams {
 
   int Add(UnwrappedTSN tsn, Data data) override;
 
-  size_t HandleForwardTsn(
-      UnwrappedTSN new_cumulative_ack_tsn,
-      webrtc::ArrayView<const AnyForwardTsnChunk::SkippedStream>
-          skipped_streams) override;
+  size_t HandleForwardTsn(UnwrappedTSN new_cumulative_ack_tsn,
+                          std::span<const AnyForwardTsnChunk::SkippedStream>
+                              skipped_streams) override;
 
-  void ResetStreams(webrtc::ArrayView<const StreamID> stream_ids) override;
+  void ResetStreams(std::span<const StreamID> stream_ids) override;
 
   HandoverReadinessStatus GetHandoverReadiness() const override;
   void AddHandoverState(DcSctpSocketHandoverState& state) override;
diff --git a/net/dcsctp/socket/BUILD.gn b/net/dcsctp/socket/BUILD.gn
index 951bfffabe..fdb255a8f8 100644
--- a/net/dcsctp/socket/BUILD.gn
+++ b/net/dcsctp/socket/BUILD.gn
@@ -22,7 +22,6 @@ rtc_source_set("context") {
 rtc_library("heartbeat_handler") {
   deps = [
     ":context",
-    "../../../api:array_view",
     "../../../api/units:time_delta",
     "../../../api/units:timestamp",
     "../../../rtc_base:checks",
@@ -46,7 +45,6 @@ rtc_library("heartbeat_handler") {
 rtc_library("stream_reset_handler") {
   deps = [
     ":context",
-    "../../../api:array_view",
     "../../../api/units:time_delta",
     "../../../rtc_base:checks",
     "../../../rtc_base:logging",
@@ -73,7 +71,6 @@ rtc_library("stream_reset_handler") {
 
 rtc_library("packet_sender") {
   deps = [
-    "../../../api:array_view",
     "../packet:sctp_packet",
     "../public:socket",
   ]
@@ -123,7 +120,6 @@ rtc_library("dcsctp_socket") {
     ":packet_sender",
     ":stream_reset_handler",
     ":transmission_control_block",
-    "../../../api:array_view",
     "../../../api/task_queue",
     "../../../api/units:time_delta",
     "../../../api/units:timestamp",
@@ -164,7 +160,6 @@ if (rtc_include_tests) {
     testonly = true
     sources = [ "mock_dcsctp_socket_callbacks.h" ]
     deps = [
-      "../../../api:array_view",
       "../../../api/task_queue",
       "../../../api/units:time_delta",
       "../../../api/units:timestamp",
@@ -205,7 +200,6 @@ if (rtc_include_tests) {
       ":packet_sender",
       ":stream_reset_handler",
       ":transmission_control_block",
-      "../../../api:array_view",
       "../../../api:create_network_emulation_manager",
       "../../../api:network_emulation_manager_api",
       "../../../api:simulated_network_api",
diff --git a/net/dcsctp/socket/callback_deferrer.cc b/net/dcsctp/socket/callback_deferrer.cc
index 595732a21e..5ed033acc3 100644
--- a/net/dcsctp/socket/callback_deferrer.cc
+++ b/net/dcsctp/socket/callback_deferrer.cc
@@ -11,13 +11,13 @@
 
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <string>
 #include <utility>
 #include <variant>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/task_queue/task_queue_base.h"
 #include "net/dcsctp/public/dcsctp_message.h"
 #include "net/dcsctp/public/dcsctp_socket.h"
@@ -51,7 +51,7 @@ void CallbackDeferrer::TriggerDeferred() {
 }
 
 SendPacketStatus CallbackDeferrer::SendPacketWithStatus(
-    webrtc::ArrayView<const uint8_t> data) {
+    std::span<const uint8_t> data) {
   // Will not be deferred - call directly.
   return underlying_.SendPacketWithStatus(data);
 }
@@ -140,7 +140,7 @@ void CallbackDeferrer::OnConnectionRestarted() {
 }
 
 void CallbackDeferrer::OnStreamsResetFailed(
-    webrtc::ArrayView<const StreamID> outgoing_streams,
+    std::span<const StreamID> outgoing_streams,
     absl::string_view reason) {
   RTC_DCHECK(prepared_);
   deferred_.emplace_back(
@@ -154,7 +154,7 @@ void CallbackDeferrer::OnStreamsResetFailed(
 }
 
 void CallbackDeferrer::OnStreamsResetPerformed(
-    webrtc::ArrayView<const StreamID> outgoing_streams) {
+    std::span<const StreamID> outgoing_streams) {
   RTC_DCHECK(prepared_);
   deferred_.emplace_back(
       +[](CallbackData data, DcSctpSocketCallbacks& cb) {
@@ -166,7 +166,7 @@ void CallbackDeferrer::OnStreamsResetPerformed(
 }
 
 void CallbackDeferrer::OnIncomingStreamsReset(
-    webrtc::ArrayView<const StreamID> incoming_streams) {
+    std::span<const StreamID> incoming_streams) {
   RTC_DCHECK(prepared_);
   deferred_.emplace_back(
       +[](CallbackData data, DcSctpSocketCallbacks& cb) {
diff --git a/net/dcsctp/socket/callback_deferrer.h b/net/dcsctp/socket/callback_deferrer.h
index 83e41528a8..c0ea134bbc 100644
--- a/net/dcsctp/socket/callback_deferrer.h
+++ b/net/dcsctp/socket/callback_deferrer.h
@@ -12,13 +12,13 @@
 
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <string>
 #include <utility>
 #include <variant>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/task_queue/task_queue_base.h"
 #include "api/units/timestamp.h"
 #include "net/dcsctp/public/dcsctp_message.h"
@@ -61,8 +61,7 @@ class CallbackDeferrer : public DcSctpSocketCallbacks {
       : underlying_(underlying) {}
 
   // Implementation of DcSctpSocketCallbacks
-  SendPacketStatus SendPacketWithStatus(
-      webrtc::ArrayView<const uint8_t> data) override;
+  SendPacketStatus SendPacketWithStatus(std::span<const uint8_t> data) override;
   std::unique_ptr<Timeout> CreateTimeout(
       webrtc::TaskQueueBase::DelayPrecision precision) override;
   TimeMs TimeMillis() override;
@@ -75,12 +74,12 @@ class CallbackDeferrer : public DcSctpSocketCallbacks {
   void OnConnected() override;
   void OnClosed() override;
   void OnConnectionRestarted() override;
-  void OnStreamsResetFailed(webrtc::ArrayView<const StreamID> outgoing_streams,
+  void OnStreamsResetFailed(std::span<const StreamID> outgoing_streams,
                             absl::string_view reason) override;
   void OnStreamsResetPerformed(
-      webrtc::ArrayView<const StreamID> outgoing_streams) override;
+      std::span<const StreamID> outgoing_streams) override;
   void OnIncomingStreamsReset(
-      webrtc::ArrayView<const StreamID> incoming_streams) override;
+      std::span<const StreamID> incoming_streams) override;
   void OnBufferedAmountLow(StreamID stream_id) override;
   void OnTotalBufferedAmountLow() override;
 
diff --git a/net/dcsctp/socket/dcsctp_socket.cc b/net/dcsctp/socket/dcsctp_socket.cc
index dd54f75a8b..0bcdbd5dd6 100644
--- a/net/dcsctp/socket/dcsctp_socket.cc
+++ b/net/dcsctp/socket/dcsctp_socket.cc
@@ -16,13 +16,13 @@
 #include <limits>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/functional/bind_front.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/task_queue/task_queue_base.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
@@ -172,7 +172,7 @@ TieTag MakeTieTag(DcSctpSocketCallbacks& cb) {
 }
 
 SctpImplementation DeterminePeerImplementation(
-    webrtc::ArrayView<const uint8_t> cookie) {
+    std::span<const uint8_t> cookie) {
   if (cookie.size() > 8) {
     absl::string_view magic(reinterpret_cast<const char*>(cookie.data()), 8);
     if (magic == "dcSCTP00") {
@@ -365,8 +365,8 @@ std::vector<uint8_t> DcSctpSocket::GenerateConnectionToken(
 }
 
 bool DcSctpSocket::ConnectWithConnectionToken(
-    webrtc::ArrayView<const uint8_t> my_data,
-    webrtc::ArrayView<const uint8_t> peer_data) {
+    std::span<const uint8_t> my_data,
+    std::span<const uint8_t> peer_data) {
   CallbackDeferrer::ScopedDeferrer deferrer(callbacks_);
 
   std::optional<InitChunk> my_init = InitChunk::Parse(my_data);
@@ -552,7 +552,7 @@ SendStatus DcSctpSocket::Send(DcSctpMessage message,
 }
 
 std::vector<SendStatus> DcSctpSocket::SendMany(
-    webrtc::ArrayView<DcSctpMessage> messages,
+    std::span<DcSctpMessage> messages,
     const SendOptions& send_options) {
   CallbackDeferrer::ScopedDeferrer deferrer(callbacks_);
   Timestamp now = callbacks_.Now();
@@ -618,7 +618,7 @@ SendStatus DcSctpSocket::InternalSend(const DcSctpMessage& message,
 }
 
 ResetStreamsStatus DcSctpSocket::ResetStreams(
-    webrtc::ArrayView<const StreamID> outgoing_streams) {
+    std::span<const StreamID> outgoing_streams) {
   CallbackDeferrer::ScopedDeferrer deferrer(callbacks_);
 
   if (tcb_ == nullptr) {
@@ -837,7 +837,7 @@ void DcSctpSocket::HandleTimeout(TimeoutID timeout_id) {
   RTC_DCHECK(IsConsistent());
 }
 
-void DcSctpSocket::ReceivePacket(webrtc::ArrayView<const uint8_t> data) {
+void DcSctpSocket::ReceivePacket(std::span<const uint8_t> data) {
   CallbackDeferrer::ScopedDeferrer deferrer(callbacks_);
 
   ++metrics_.rx_packets_count;
@@ -887,8 +887,7 @@ void DcSctpSocket::ReceivePacket(webrtc::ArrayView<const uint8_t> data) {
   RTC_DCHECK(IsConsistent());
 }
 
-void DcSctpSocket::DebugPrintOutgoing(
-    webrtc::ArrayView<const uint8_t> payload) {
+void DcSctpSocket::DebugPrintOutgoing(std::span<const uint8_t> payload) {
   auto packet = SctpPacket::Parse(payload, options_);
   RTC_DCHECK(packet.has_value());
 
@@ -1067,7 +1066,7 @@ TimeDelta DcSctpSocket::OnShutdownTimerExpiry() {
   return tcb_->current_rto();
 }
 
-void DcSctpSocket::OnSentPacket(webrtc::ArrayView<const uint8_t> packet,
+void DcSctpSocket::OnSentPacket(std::span<const uint8_t> packet,
                                 SendPacketStatus status) {
   // The packet observer is invoked even if the packet was failed to be sent, to
   // indicate an attempt was made.
diff --git a/net/dcsctp/socket/dcsctp_socket.h b/net/dcsctp/socket/dcsctp_socket.h
index a3e3cb3f5f..5f78957b05 100644
--- a/net/dcsctp/socket/dcsctp_socket.h
+++ b/net/dcsctp/socket/dcsctp_socket.h
@@ -15,11 +15,11 @@
 #include <functional>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/units/time_delta.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/chunk/data_common.h"
@@ -66,7 +66,7 @@ class DcSctpSocket : public DcSctpSocketInterface {
   DcSctpSocket& operator=(const DcSctpSocket&) = delete;
 
   // Implementation of `DcSctpSocketInterface`.
-  void ReceivePacket(webrtc::ArrayView<const uint8_t> data) override;
+  void ReceivePacket(std::span<const uint8_t> data) override;
   size_t MessagesReady() const override;
   std::optional<DcSctpMessage> GetNextMessage() override;
   void HandleTimeout(TimeoutID timeout_id) override;
@@ -74,19 +74,18 @@ class DcSctpSocket : public DcSctpSocketInterface {
   static std::vector<uint8_t> GenerateConnectionToken(
       const DcSctpOptions& options,
       std::function<uint32_t(uint32_t low, uint32_t high)> get_random_uint32);
-  bool ConnectWithConnectionToken(
-      webrtc::ArrayView<const uint8_t> my_data,
-      webrtc::ArrayView<const uint8_t> peer_data) override;
+  bool ConnectWithConnectionToken(std::span<const uint8_t> my_data,
+                                  std::span<const uint8_t> peer_data) override;
 
   void RestoreFromState(const DcSctpSocketHandoverState& state) override;
   void Shutdown() override;
   void Close() override;
   SendStatus Send(DcSctpMessage message,
                   const SendOptions& send_options) override;
-  std::vector<SendStatus> SendMany(webrtc::ArrayView<DcSctpMessage> messages,
+  std::vector<SendStatus> SendMany(std::span<DcSctpMessage> messages,
                                    const SendOptions& send_options) override;
   ResetStreamsStatus ResetStreams(
-      webrtc::ArrayView<const StreamID> outgoing_streams) override;
+      std::span<const StreamID> outgoing_streams) override;
   SocketState state() const override;
   const DcSctpOptions& options() const override { return options_; }
   void SetMaxMessageSize(size_t max_message_size) override;
@@ -151,8 +150,7 @@ class DcSctpSocket : public DcSctpSocketInterface {
   webrtc::TimeDelta OnInitTimerExpiry();
   webrtc::TimeDelta OnCookieTimerExpiry();
   webrtc::TimeDelta OnShutdownTimerExpiry();
-  void OnSentPacket(webrtc::ArrayView<const uint8_t> packet,
-                    SendPacketStatus status);
+  void OnSentPacket(std::span<const uint8_t> packet, SendPacketStatus status);
   // Sends SHUTDOWN or SHUTDOWN-ACK if the socket is shutting down and if all
   // outstanding data has been acknowledged.
   void MaybeSendShutdownOrAck();
@@ -174,7 +172,7 @@ class DcSctpSocket : public DcSctpSocketInterface {
   bool ValidatePacket(const SctpPacket& packet);
   // Parses `payload`, which is a serialized packet that is just going to be
   // sent and prints all chunks.
-  void DebugPrintOutgoing(webrtc::ArrayView<const uint8_t> payload);
+  void DebugPrintOutgoing(std::span<const uint8_t> payload);
   // Called whenever data has been received, or the cumulative acknowledgment
   // TSN has moved, that may result in delivering messages.
   void MaybeDeliverMessages();
diff --git a/net/dcsctp/socket/dcsctp_socket_network_test.cc b/net/dcsctp/socket/dcsctp_socket_network_test.cc
index 41b71ba475..3aa1efca20 100644
--- a/net/dcsctp/socket/dcsctp_socket_network_test.cc
+++ b/net/dcsctp/socket/dcsctp_socket_network_test.cc
@@ -12,13 +12,13 @@
 #include <functional>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/base/nullability.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/task_queue/pending_task_safety_flag.h"
 #include "api/task_queue/task_queue_base.h"
 #include "api/test/create_network_emulation_manager.h"
@@ -126,7 +126,7 @@ class BoundSocket : public webrtc::EmulatedNetworkReceiverInterface {
     receiver_ = std::move(receiver);
   }
 
-  void SendPacket(webrtc::ArrayView<const uint8_t> data) {
+  void SendPacket(std::span<const uint8_t> data) {
     endpoint_->SendPacket(source_address_, dest_address_,
                           webrtc::CopyOnWriteBuffer(data.data(), data.size()));
   }
@@ -191,7 +191,7 @@ class SctpActor : public DcSctpSocketCallbacks {
     }
   }
 
-  void SendPacket(webrtc::ArrayView<const uint8_t> data) override {
+  void SendPacket(std::span<const uint8_t> data) override {
     emulated_socket_.SendPacket(data);
   }
 
@@ -227,15 +227,14 @@ class SctpActor : public DcSctpSocketCallbacks {
 
   void OnConnectionRestarted() override {}
 
-  void OnStreamsResetFailed(
-      webrtc::ArrayView<const StreamID> /* outgoing_streams */,
-      absl::string_view /* reason */) override {}
+  void OnStreamsResetFailed(std::span<const StreamID> /* outgoing_streams */,
+                            absl::string_view /* reason */) override {}
 
   void OnStreamsResetPerformed(
-      webrtc::ArrayView<const StreamID> /* outgoing_streams */) override {}
+      std::span<const StreamID> /* outgoing_streams */) override {}
 
   void OnIncomingStreamsReset(
-      webrtc::ArrayView<const StreamID> /* incoming_streams */) override {}
+      std::span<const StreamID> /* incoming_streams */) override {}
 
   void NotifyOutgoingMessageBufferEmpty() override {}
 
diff --git a/net/dcsctp/socket/dcsctp_socket_test.cc b/net/dcsctp/socket/dcsctp_socket_test.cc
index 88aef3dbbf..fcab2677ec 100644
--- a/net/dcsctp/socket/dcsctp_socket_test.cc
+++ b/net/dcsctp/socket/dcsctp_socket_test.cc
@@ -20,7 +20,6 @@
 
 #include "absl/flags/flag.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
 #include "net/dcsctp/common/handover_testing.h"
diff --git a/net/dcsctp/socket/heartbeat_handler.cc b/net/dcsctp/socket/heartbeat_handler.cc
index 1b52900d13..3162662b24 100644
--- a/net/dcsctp/socket/heartbeat_handler.cc
+++ b/net/dcsctp/socket/heartbeat_handler.cc
@@ -13,12 +13,12 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
 #include "absl/functional/bind_front.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
@@ -66,7 +66,7 @@ class HeartbeatInfo {
   }
 
   static std::optional<HeartbeatInfo> Deserialize(
-      webrtc::ArrayView<const uint8_t> data) {
+      std::span<const uint8_t> data) {
     if (data.size() != kBufferSize) {
       RTC_LOG(LS_WARNING) << "Invalid heartbeat info: " << data.size()
                           << " bytes";
diff --git a/net/dcsctp/socket/mock_dcsctp_socket_callbacks.h b/net/dcsctp/socket/mock_dcsctp_socket_callbacks.h
index 7e405e5379..c80f37331a 100644
--- a/net/dcsctp/socket/mock_dcsctp_socket_callbacks.h
+++ b/net/dcsctp/socket/mock_dcsctp_socket_callbacks.h
@@ -14,12 +14,12 @@
 #include <deque>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/task_queue/task_queue_base.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
@@ -60,7 +60,7 @@ class MockDcSctpSocketCallbacks : public DcSctpSocketCallbacks {
         random_(internal::GetUniqueSeed()),
         timeout_manager_([this]() { return now_; }) {
     ON_CALL(*this, SendPacketWithStatus)
-        .WillByDefault([this](webrtc::ArrayView<const uint8_t> data) {
+        .WillByDefault([this](std::span<const uint8_t> data) {
           sent_packets_.emplace_back(
               std::vector<uint8_t>(data.begin(), data.end()));
           return SendPacketStatus::kSuccess;
@@ -87,7 +87,7 @@ class MockDcSctpSocketCallbacks : public DcSctpSocketCallbacks {
 
   MOCK_METHOD(SendPacketStatus,
               SendPacketWithStatus,
-              (webrtc::ArrayView<const uint8_t> data),
+              (std::span<const uint8_t> data),
               (override));
 
   std::unique_ptr<Timeout> CreateTimeout(
@@ -116,16 +116,16 @@ class MockDcSctpSocketCallbacks : public DcSctpSocketCallbacks {
   MOCK_METHOD(void, OnConnectionRestarted, (), (override));
   MOCK_METHOD(void,
               OnStreamsResetFailed,
-              (webrtc::ArrayView<const StreamID> outgoing_streams,
+              (std::span<const StreamID> outgoing_streams,
                absl::string_view reason),
               (override));
   MOCK_METHOD(void,
               OnStreamsResetPerformed,
-              (webrtc::ArrayView<const StreamID> outgoing_streams),
+              (std::span<const StreamID> outgoing_streams),
               (override));
   MOCK_METHOD(void,
               OnIncomingStreamsReset,
-              (webrtc::ArrayView<const StreamID> incoming_streams),
+              (std::span<const StreamID> incoming_streams),
               (override));
   MOCK_METHOD(void, OnBufferedAmountLow, (StreamID stream_id), (override));
   MOCK_METHOD(void, OnTotalBufferedAmountLow, (), (override));
diff --git a/net/dcsctp/socket/packet_sender.cc b/net/dcsctp/socket/packet_sender.cc
index 9abe1b48d7..3f4de4d906 100644
--- a/net/dcsctp/socket/packet_sender.cc
+++ b/net/dcsctp/socket/packet_sender.cc
@@ -11,17 +11,17 @@
 
 #include <cstdint>
 #include <functional>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/sctp_packet.h"
 #include "net/dcsctp/public/dcsctp_socket.h"
 
 namespace dcsctp {
 
 PacketSender::PacketSender(DcSctpSocketCallbacks& callbacks,
-                           std::function<void(webrtc::ArrayView<const uint8_t>,
+                           std::function<void(std::span<const uint8_t>,
                                               SendPacketStatus)> on_sent_packet)
     : callbacks_(callbacks), on_sent_packet_(std::move(on_sent_packet)) {}
 
diff --git a/net/dcsctp/socket/packet_sender.h b/net/dcsctp/socket/packet_sender.h
index 69a74f9ff8..d0039d0d21 100644
--- a/net/dcsctp/socket/packet_sender.h
+++ b/net/dcsctp/socket/packet_sender.h
@@ -12,8 +12,8 @@
 
 #include <cstdint>
 #include <functional>
+#include <span>
 
-#include "api/array_view.h"
 #include "net/dcsctp/packet/sctp_packet.h"
 #include "net/dcsctp/public/dcsctp_socket.h"
 
@@ -25,8 +25,8 @@ namespace dcsctp {
 class PacketSender {
  public:
   PacketSender(DcSctpSocketCallbacks& callbacks,
-               std::function<void(webrtc::ArrayView<const uint8_t>,
-                                  SendPacketStatus)> on_sent_packet);
+               std::function<void(std::span<const uint8_t>, SendPacketStatus)>
+                   on_sent_packet);
 
   // Sends the packet, and returns true if it was sent successfully.
   bool Send(SctpPacket::Builder& builder, bool write_checksum = true);
@@ -36,7 +36,7 @@ class PacketSender {
 
   // Callback that will be triggered for every send attempt, indicating the
   // status of the operation.
-  std::function<void(webrtc::ArrayView<const uint8_t>, SendPacketStatus)>
+  std::function<void(std::span<const uint8_t>, SendPacketStatus)>
       on_sent_packet_;
 };
 }  // namespace dcsctp
diff --git a/net/dcsctp/socket/packet_sender_test.cc b/net/dcsctp/socket/packet_sender_test.cc
index be54580908..620072a57c 100644
--- a/net/dcsctp/socket/packet_sender_test.cc
+++ b/net/dcsctp/socket/packet_sender_test.cc
@@ -10,8 +10,8 @@
 #include "net/dcsctp/socket/packet_sender.h"
 
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/chunk/cookie_ack_chunk.h"
 #include "net/dcsctp/packet/sctp_packet.h"
@@ -37,8 +37,7 @@ class PacketSenderTest : public testing::Test {
 
   DcSctpOptions options_;
   testing::NiceMock<MockDcSctpSocketCallbacks> callbacks_;
-  testing::MockFunction<void(webrtc::ArrayView<const uint8_t>,
-                             SendPacketStatus)>
+  testing::MockFunction<void(std::span<const uint8_t>, SendPacketStatus)>
       on_send_fn_;
   PacketSender sender_;
 };
diff --git a/net/dcsctp/socket/state_cookie.cc b/net/dcsctp/socket/state_cookie.cc
index 68a237b7ae..d5b05b5408 100644
--- a/net/dcsctp/socket/state_cookie.cc
+++ b/net/dcsctp/socket/state_cookie.cc
@@ -12,9 +12,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/bounded_byte_reader.h"
 #include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -50,7 +50,7 @@ std::vector<uint8_t> StateCookie::Serialize() {
 }
 
 std::optional<StateCookie> StateCookie::Deserialize(
-    webrtc::ArrayView<const uint8_t> cookie) {
+    std::span<const uint8_t> cookie) {
   if (cookie.size() != kCookieSize) {
     RTC_DLOG(LS_WARNING) << "Invalid state cookie: " << cookie.size()
                          << " bytes";
diff --git a/net/dcsctp/socket/state_cookie.h b/net/dcsctp/socket/state_cookie.h
index 90073586ab..752f05157b 100644
--- a/net/dcsctp/socket/state_cookie.h
+++ b/net/dcsctp/socket/state_cookie.h
@@ -13,9 +13,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/socket/capabilities.h"
 
@@ -50,7 +50,7 @@ class StateCookie {
 
   // Deserializes the cookie, and returns std::nullopt if that failed.
   static std::optional<StateCookie> Deserialize(
-      webrtc::ArrayView<const uint8_t> cookie);
+      std::span<const uint8_t> cookie);
 
   VerificationTag peer_tag() const { return peer_tag_; }
   VerificationTag my_tag() const { return my_tag_; }
diff --git a/net/dcsctp/socket/stream_reset_handler.cc b/net/dcsctp/socket/stream_reset_handler.cc
index 950142c847..c4573e7a43 100644
--- a/net/dcsctp/socket/stream_reset_handler.cc
+++ b/net/dcsctp/socket/stream_reset_handler.cc
@@ -12,9 +12,9 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/units/time_delta.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/packet/chunk/reconfig_chunk.h"
@@ -355,7 +355,7 @@ ReConfigChunk StreamResetHandler::MakeReconfigChunk() {
 }
 
 void StreamResetHandler::ResetStreams(
-    webrtc::ArrayView<const StreamID> outgoing_streams) {
+    std::span<const StreamID> outgoing_streams) {
   for (StreamID stream_id : outgoing_streams) {
     retransmission_queue_->PrepareResetStream(stream_id);
   }
diff --git a/net/dcsctp/socket/stream_reset_handler.h b/net/dcsctp/socket/stream_reset_handler.h
index e54c946d10..06c1379f64 100644
--- a/net/dcsctp/socket/stream_reset_handler.h
+++ b/net/dcsctp/socket/stream_reset_handler.h
@@ -12,12 +12,12 @@
 
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
 #include "absl/functional/bind_front.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/units/time_delta.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/common/sequence_numbers.h"
@@ -97,7 +97,7 @@ class StreamResetHandler {
   // time and also multiple times. It will enqueue requests that can't be
   // directly fulfilled, and will asynchronously process them when any ongoing
   // request has completed.
-  void ResetStreams(webrtc::ArrayView<const StreamID> outgoing_streams);
+  void ResetStreams(std::span<const StreamID> outgoing_streams);
 
   // Creates a Reset Streams request that must be sent if returned. Will start
   // the reconfig timer. Will return std::nullopt if there is no need to
diff --git a/net/dcsctp/socket/stream_reset_handler_test.cc b/net/dcsctp/socket/stream_reset_handler_test.cc
index cb2454aedf..b609bb2bf4 100644
--- a/net/dcsctp/socket/stream_reset_handler_test.cc
+++ b/net/dcsctp/socket/stream_reset_handler_test.cc
@@ -16,7 +16,6 @@
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/task_queue/task_queue_base.h"
 #include "api/units/time_delta.h"
 #include "net/dcsctp/common/handover_testing.h"
diff --git a/net/dcsctp/socket/transmission_control_block_test.cc b/net/dcsctp/socket/transmission_control_block_test.cc
index 005df89d09..070c38ed93 100644
--- a/net/dcsctp/socket/transmission_control_block_test.cc
+++ b/net/dcsctp/socket/transmission_control_block_test.cc
@@ -11,8 +11,8 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/task_queue/task_queue_base.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/public/dcsctp_options.h"
@@ -49,8 +49,7 @@ class TransmissionControlBlockTest : public testing::Test {
   Capabilities capabilities_;
   StrictMock<MockDcSctpSocketCallbacks> callbacks_;
   StrictMock<MockSendQueue> send_queue_;
-  testing::MockFunction<void(webrtc::ArrayView<const uint8_t>,
-                             SendPacketStatus)>
+  testing::MockFunction<void(std::span<const uint8_t>, SendPacketStatus)>
       on_send_fn_;
   testing::MockFunction<bool()> on_connection_established;
   PacketSender sender_;
diff --git a/net/dcsctp/tx/BUILD.gn b/net/dcsctp/tx/BUILD.gn
index 2a1645eae8..41eaca3a5a 100644
--- a/net/dcsctp/tx/BUILD.gn
+++ b/net/dcsctp/tx/BUILD.gn
@@ -22,7 +22,6 @@ rtc_library("rr_send_queue") {
   deps = [
     ":send_queue",
     ":stream_scheduler",
-    "../../../api:array_view",
     "../../../api/units:time_delta",
     "../../../api/units:timestamp",
     "../../../rtc_base:checks",
@@ -89,7 +88,6 @@ rtc_library("retransmission_timeout") {
 
 rtc_library("outstanding_data") {
   deps = [
-    "../../../api:array_view",
     "../../../api/units:time_delta",
     "../../../api/units:timestamp",
     "../../../rtc_base:checks",
@@ -112,7 +110,6 @@ rtc_library("retransmission_queue") {
   deps = [
     ":outstanding_data",
     ":send_queue",
-    "../../../api:array_view",
     "../../../api/units:time_delta",
     "../../../api/units:timestamp",
     "../../../rtc_base:checks",
diff --git a/net/dcsctp/tx/outstanding_data.cc b/net/dcsctp/tx/outstanding_data.cc
index 37ba07b822..89f1ea2a77 100644
--- a/net/dcsctp/tx/outstanding_data.cc
+++ b/net/dcsctp/tx/outstanding_data.cc
@@ -15,10 +15,10 @@
 #include <map>
 #include <optional>
 #include <set>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
 #include "net/dcsctp/common/internal_types.h"
@@ -141,7 +141,7 @@ void OutstandingData::AckChunk(AckInfo& ack_info,
 
 OutstandingData::AckInfo OutstandingData::HandleSack(
     UnwrappedTSN cumulative_tsn_ack,
-    webrtc::ArrayView<const SackChunk::GapAckBlock> gap_ack_blocks,
+    std::span<const SackChunk::GapAckBlock> gap_ack_blocks,
     bool is_in_fast_recovery) {
   bool cumulative_tsn_ack_advanced =
       cumulative_tsn_ack > last_cumulative_tsn_ack_;
@@ -205,7 +205,7 @@ void OutstandingData::RemoveAcked(UnwrappedTSN cumulative_tsn_ack,
 
 void OutstandingData::AckGapBlocks(
     UnwrappedTSN cumulative_tsn_ack,
-    webrtc::ArrayView<const SackChunk::GapAckBlock> gap_ack_blocks,
+    std::span<const SackChunk::GapAckBlock> gap_ack_blocks,
     AckInfo& ack_info) {
   // Mark all non-gaps as ACKED (but they can't be removed) as (from RFC)
   // "SCTP considers the information carried in the Gap Ack Blocks in the
@@ -226,7 +226,7 @@ void OutstandingData::AckGapBlocks(
 
 void OutstandingData::NackBetweenAckBlocks(
     UnwrappedTSN cumulative_tsn_ack,
-    webrtc::ArrayView<const SackChunk::GapAckBlock> gap_ack_blocks,
+    std::span<const SackChunk::GapAckBlock> gap_ack_blocks,
     bool is_in_fast_recovery,
     bool cumulative_tsn_acked_advanced,
     OutstandingData::AckInfo& ack_info) {
diff --git a/net/dcsctp/tx/outstanding_data.h b/net/dcsctp/tx/outstanding_data.h
index 81fb4c27af..5303764151 100644
--- a/net/dcsctp/tx/outstanding_data.h
+++ b/net/dcsctp/tx/outstanding_data.h
@@ -16,10 +16,10 @@
 #include <functional>
 #include <optional>
 #include <set>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
 #include "net/dcsctp/common/internal_types.h"
@@ -89,10 +89,9 @@ class OutstandingData {
         last_cumulative_tsn_ack_(last_cumulative_tsn_ack),
         discard_from_send_queue_(std::move(discard_from_send_queue)) {}
 
-  AckInfo HandleSack(
-      UnwrappedTSN cumulative_tsn_ack,
-      webrtc::ArrayView<const SackChunk::GapAckBlock> gap_ack_blocks,
-      bool is_in_fast_recovery);
+  AckInfo HandleSack(UnwrappedTSN cumulative_tsn_ack,
+                     std::span<const SackChunk::GapAckBlock> gap_ack_blocks,
+                     bool is_in_fast_recovery);
 
   // Returns as many of the chunks that are eligible for fast retransmissions
   // and that would fit in a single packet of `max_size`. The eligible chunks
@@ -314,10 +313,9 @@ class OutstandingData {
 
   // Will mark the chunks covered by the `gap_ack_blocks` from an incoming SACK
   // as "acked" and update `ack_info` by adding new TSNs to `added_tsns`.
-  void AckGapBlocks(
-      UnwrappedTSN cumulative_tsn_ack,
-      webrtc::ArrayView<const SackChunk::GapAckBlock> gap_ack_blocks,
-      AckInfo& ack_info);
+  void AckGapBlocks(UnwrappedTSN cumulative_tsn_ack,
+                    std::span<const SackChunk::GapAckBlock> gap_ack_blocks,
+                    AckInfo& ack_info);
 
   // Mark chunks reported as "missing", as "nacked" or "to be retransmitted"
   // depending how many times this has happened. Only packets up until
@@ -325,7 +323,7 @@ class OutstandingData {
   // nacked/retransmitted. The method will set `ack_info.has_packet_loss`.
   void NackBetweenAckBlocks(
       UnwrappedTSN cumulative_tsn_ack,
-      webrtc::ArrayView<const SackChunk::GapAckBlock> gap_ack_blocks,
+      std::span<const SackChunk::GapAckBlock> gap_ack_blocks,
       bool is_in_fast_recovery,
       bool cumulative_tsn_acked_advanced,
       OutstandingData::AckInfo& ack_info);
diff --git a/net/dcsctp/tx/retransmission_queue.cc b/net/dcsctp/tx/retransmission_queue.cc
index 9557a9be22..3dc2bc7134 100644
--- a/net/dcsctp/tx/retransmission_queue.cc
+++ b/net/dcsctp/tx/retransmission_queue.cc
@@ -19,7 +19,6 @@
 
 #include "absl/algorithm/container.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
 #include "net/dcsctp/common/internal_types.h"
diff --git a/net/dcsctp/tx/rr_send_queue.cc b/net/dcsctp/tx/rr_send_queue.cc
index d27094d68d..2b0883ae99 100644
--- a/net/dcsctp/tx/rr_send_queue.cc
+++ b/net/dcsctp/tx/rr_send_queue.cc
@@ -16,13 +16,13 @@
 #include <map>
 #include <optional>
 #include <set>
+#include <span>
 #include <tuple>
 #include <utility>
 #include <vector>
 
 #include "absl/algorithm/container.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
 #include "net/dcsctp/common/internal_types.h"
@@ -175,8 +175,8 @@ std::optional<SendQueue::DataToSend> RRSendQueue::OutgoingStream::Produce(
     }
 
     // Grab the next `max_size` fragment from this message and calculate flags.
-    webrtc::ArrayView<const uint8_t> message_payload = message.payload();
-    webrtc::ArrayView<const uint8_t> chunk_payload = message_payload.subspan(
+    std::span<const uint8_t> message_payload = message.payload();
+    std::span<const uint8_t> chunk_payload = message_payload.subspan(
         item.remaining_offset,
         std::min(message_payload.size() - item.remaining_offset, max_size));
     Data::IsBeginning is_beginning(chunk_payload.data() ==
diff --git a/net/dcsctp/tx/rr_send_queue.h b/net/dcsctp/tx/rr_send_queue.h
index 07148cccc9..ae24c49ba8 100644
--- a/net/dcsctp/tx/rr_send_queue.h
+++ b/net/dcsctp/tx/rr_send_queue.h
@@ -20,7 +20,6 @@
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/units/timestamp.h"
 #include "net/dcsctp/common/internal_types.h"
 #include "net/dcsctp/public/dcsctp_handover_state.h"

From 201168a998a4368ebc4349fc12485c2299f731f9 Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Wed, 11 Mar 2026 12:59:53 +0100
Subject: [PATCH 264/472] Replace ArrayView with std::span in
 modules/audio_processing/aec3

Search&Replace MakeArrayView and ArrayView with std::span
Search&Replace include "api/array_view.h" with include <span>
Remove <span> include where std::span is not mentioned in the file
Remove build dependencies on array_view target

No-Iwyu: suggested missing includes are questionable, and such question is out of scope of this change.
Bug: webrtc:439801349
Change-Id: Iaa233c972c1420563862f43530ed354cddbcaf8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/455800
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47117}
---
 modules/audio_processing/aec3/BUILD.gn        | 15 +----
 .../aec3/adaptive_fir_filter.cc               | 18 +++---
 .../aec3/adaptive_fir_filter.h                |  1 -
 .../aec3/adaptive_fir_filter_avx2.cc          |  6 +-
 .../aec3/adaptive_fir_filter_erl.cc           | 10 ++--
 .../aec3/adaptive_fir_filter_erl.h            | 12 ++--
 .../aec3/adaptive_fir_filter_erl_avx2.cc      |  4 +-
 .../aec3/adaptive_fir_filter_unittest.cc      |  6 +-
 modules/audio_processing/aec3/aec3_fft.cc     |  8 +--
 modules/audio_processing/aec3/aec3_fft.h      | 12 ++--
 modules/audio_processing/aec3/aec_state.cc    | 30 +++++-----
 modules/audio_processing/aec3/aec_state.h     | 26 ++++----
 .../audio_processing/aec3/alignment_mixer.cc  |  8 +--
 .../audio_processing/aec3/alignment_mixer.h   |  6 +-
 .../aec3/alignment_mixer_unittest.cc          |  1 -
 modules/audio_processing/aec3/block.h         | 12 ++--
 .../aec3/block_delay_buffer.cc                |  4 +-
 modules/audio_processing/aec3/block_framer.cc |  4 +-
 modules/audio_processing/aec3/block_framer.h  |  4 +-
 .../aec3/block_framer_unittest.cc             | 24 ++++----
 .../aec3/block_processor_unittest.cc          |  4 +-
 .../aec3/comfort_noise_generator.cc           | 16 ++---
 .../aec3/comfort_noise_generator.h            | 10 ++--
 modules/audio_processing/aec3/decimator.cc    | 12 ++--
 modules/audio_processing/aec3/decimator.h     |  4 +-
 .../aec3/decimator_unittest.cc                |  8 +--
 .../aec3/dominant_nearend_detector.cc         | 10 ++--
 .../aec3/dominant_nearend_detector.h          |  8 +--
 .../audio_processing/aec3/echo_audibility.cc  |  8 +--
 .../audio_processing/aec3/echo_audibility.h   |  8 +--
 .../audio_processing/aec3/echo_canceller3.cc  | 36 +++++------
 .../audio_processing/aec3/echo_canceller3.h   |  8 +--
 .../aec3/echo_canceller3_unittest.cc          | 26 ++++----
 .../aec3/echo_path_delay_estimator.cc         |  4 +-
 modules/audio_processing/aec3/echo_remover.cc | 58 +++++++++---------
 .../audio_processing/aec3/erl_estimator.cc    |  8 +--
 modules/audio_processing/aec3/erl_estimator.h |  6 +-
 .../audio_processing/aec3/erle_estimator.cc   | 10 ++--
 .../audio_processing/aec3/erle_estimator.h    | 18 +++---
 .../aec3/erle_estimator_unittest.cc           | 20 +++----
 modules/audio_processing/aec3/fft_data.h      |  6 +-
 .../audio_processing/aec3/fft_data_avx2.cc    |  4 +-
 .../audio_processing/aec3/filter_analyzer.cc  | 16 ++---
 .../audio_processing/aec3/filter_analyzer.h   | 16 ++---
 .../audio_processing/aec3/frame_blocker.cc    |  4 +-
 modules/audio_processing/aec3/frame_blocker.h |  4 +-
 .../aec3/frame_blocker_unittest.cc            | 34 +++++------
 .../aec3/fullband_erle_estimator.cc           |  8 +--
 .../aec3/fullband_erle_estimator.h            | 10 ++--
 .../audio_processing/aec3/matched_filter.cc   | 60 +++++++++----------
 .../audio_processing/aec3/matched_filter.h    | 44 +++++++-------
 .../aec3/matched_filter_avx2.cc               | 22 +++----
 .../aec3/matched_filter_lag_aggregator.cc     |  4 +-
 .../aec3/matched_filter_lag_aggregator.h      |  4 +-
 .../aec3/matched_filter_unittest.cc           | 10 ++--
 .../audio_processing/aec3/moving_average.cc   |  6 +-
 .../audio_processing/aec3/moving_average.h    |  5 +-
 .../audio_processing/aec3/nearend_detector.h  |  8 +--
 .../neural_residual_echo_estimator/BUILD.gn   |  2 -
 .../neural_feature_extractor.cc               | 18 +++---
 .../neural_feature_extractor.h                | 18 +++---
 .../neural_feature_extractor_unittest.cc      | 24 ++++----
 .../neural_residual_echo_estimator_impl.cc    | 42 ++++++-------
 .../neural_residual_echo_estimator_impl.h     | 30 +++++-----
 ...l_residual_echo_estimator_impl_unittest.cc | 24 ++++----
 .../aec3/refined_filter_update_gain.cc        |  4 +-
 .../aec3/refined_filter_update_gain.h         |  4 +-
 modules/audio_processing/aec3/render_buffer.h |  6 +-
 .../aec3/render_delay_buffer.cc               |  8 +--
 .../aec3/render_signal_analyzer.cc            |  6 +-
 .../aec3/render_signal_analyzer_unittest.cc   |  1 -
 .../aec3/residual_echo_estimator.cc           | 40 ++++++-------
 .../aec3/residual_echo_estimator.h            | 18 +++---
 .../aec3/residual_echo_estimator_unittest.cc  |  4 +-
 .../aec3/reverb_decay_estimator.cc            | 14 ++---
 .../aec3/reverb_decay_estimator.h             |  9 ++-
 .../aec3/reverb_frequency_response.cc         | 10 ++--
 .../aec3/reverb_frequency_response.h          |  4 +-
 modules/audio_processing/aec3/reverb_model.cc |  9 ++-
 modules/audio_processing/aec3/reverb_model.h  | 10 ++--
 .../aec3/reverb_model_estimator.cc            | 10 ++--
 .../aec3/reverb_model_estimator.h             | 12 ++--
 .../aec3/reverb_model_estimator_unittest.cc   |  4 +-
 .../aec3/signal_dependent_erle_estimator.cc   | 28 ++++-----
 .../aec3/signal_dependent_erle_estimator.h    | 26 ++++----
 ...ignal_dependent_erle_estimator_unittest.cc | 10 ++--
 .../aec3/stationarity_estimator.cc            | 12 ++--
 .../aec3/stationarity_estimator.h             | 12 ++--
 .../aec3/subband_erle_estimator.cc            | 14 ++---
 .../aec3/subband_erle_estimator.h             | 20 +++----
 .../aec3/subband_nearend_detector.cc          |  8 +--
 .../aec3/subband_nearend_detector.h           |  8 +--
 modules/audio_processing/aec3/subtractor.cc   | 14 ++---
 modules/audio_processing/aec3/subtractor.h    |  8 +--
 .../aec3/subtractor_output.cc                 |  5 +-
 .../audio_processing/aec3/subtractor_output.h |  4 +-
 .../aec3/subtractor_output_analyzer.cc        |  4 +-
 .../aec3/subtractor_output_analyzer.h         |  4 +-
 .../aec3/subtractor_unittest.cc               |  8 +--
 .../aec3/suppression_filter.cc                |  8 +--
 .../aec3/suppression_filter.h                 |  8 +--
 .../audio_processing/aec3/suppression_gain.cc | 40 ++++++-------
 .../audio_processing/aec3/suppression_gain.h  | 32 +++++-----
 modules/audio_processing/aec3/vector_math.h   | 22 +++----
 .../audio_processing/aec3/vector_math_avx2.cc | 12 ++--
 105 files changed, 686 insertions(+), 710 deletions(-)

diff --git a/modules/audio_processing/aec3/BUILD.gn b/modules/audio_processing/aec3/BUILD.gn
index 1f5a2717c9..10b055f4e2 100644
--- a/modules/audio_processing/aec3/BUILD.gn
+++ b/modules/audio_processing/aec3/BUILD.gn
@@ -143,7 +143,6 @@ rtc_library("aec3") {
     "..:apm_logging",
     "..:audio_buffer",
     "..:high_pass_filter",
-    "../../../api:array_view",
     "../../../api:field_trials_view",
     "../../../api/audio:aec3_config",
     "../../../api/audio:echo_control",
@@ -176,10 +175,7 @@ rtc_source_set("aec3_common") {
 
 rtc_source_set("aec3_block") {
   sources = [ "block.h" ]
-  deps = [
-    ":aec3_common",
-    "../../../api:array_view",
-  ]
+  deps = [ ":aec3_common" ]
 }
 
 rtc_source_set("aec3_fft") {
@@ -187,7 +183,6 @@ rtc_source_set("aec3_fft") {
   deps = [
     ":aec3_common",
     ":fft_data",
-    "../../../api:array_view",
     "../../../common_audio/third_party/ooura:fft_size_128",
     "../../../rtc_base:checks",
     "../../../rtc_base/system:arch",
@@ -205,7 +200,6 @@ rtc_source_set("render_buffer") {
     ":aec3_block",
     ":aec3_common",
     ":fft_data",
-    "../../../api:array_view",
     "../../../rtc_base:checks",
     "../../../rtc_base/system:arch",
   ]
@@ -219,7 +213,6 @@ rtc_source_set("adaptive_fir_filter") {
     ":fft_data",
     ":render_buffer",
     "..:apm_logging",
-    "../../../api:array_view",
     "../../../rtc_base/system:arch",
     "//third_party/abseil-cpp/absl/strings:string_view",
   ]
@@ -229,7 +222,6 @@ rtc_source_set("adaptive_fir_filter_erl") {
   sources = [ "adaptive_fir_filter_erl.h" ]
   deps = [
     ":aec3_common",
-    "../../../api:array_view",
     "../../../rtc_base/system:arch",
   ]
 }
@@ -238,7 +230,6 @@ rtc_source_set("matched_filter") {
   sources = [ "matched_filter.h" ]
   deps = [
     ":aec3_common",
-    "../../../api:array_view",
     "../../../rtc_base:gtest_prod",
     "../../../rtc_base/system:arch",
   ]
@@ -248,7 +239,6 @@ rtc_source_set("vector_math") {
   sources = [ "vector_math.h" ]
   deps = [
     ":aec3_common",
-    "../../../api:array_view",
     "../../../rtc_base:checks",
     "../../../rtc_base/system:arch",
   ]
@@ -258,7 +248,6 @@ rtc_source_set("fft_data") {
   sources = [ "fft_data.h" ]
   deps = [
     ":aec3_common",
-    "../../../api:array_view",
     "../../../rtc_base:checks",
     "../../../rtc_base/system:arch",
   ]
@@ -292,7 +281,6 @@ if (current_cpu == "x86" || current_cpu == "x64") {
       ":matched_filter",
       ":render_buffer",
       ":vector_math",
-      "../../../api:array_view",
       "../../../rtc_base:checks",
     ]
   }
@@ -329,7 +317,6 @@ if (rtc_include_tests) {
       "..:audio_buffer",
       "..:audio_processing",
       "..:high_pass_filter",
-      "../../../api:array_view",
       "../../../api:field_trials",
       "../../../api/audio:aec3_config",
       "../../../api/audio:echo_control",
diff --git a/modules/audio_processing/aec3/adaptive_fir_filter.cc b/modules/audio_processing/aec3/adaptive_fir_filter.cc
index 6a2e3ba712..52718c0526 100644
--- a/modules/audio_processing/aec3/adaptive_fir_filter.cc
+++ b/modules/audio_processing/aec3/adaptive_fir_filter.cc
@@ -14,9 +14,9 @@
 #include <array>
 #include <cmath>
 #include <cstddef>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/fft_data.h"
 #include "modules/audio_processing/aec3/render_buffer.h"
@@ -134,7 +134,7 @@ void AdaptPartitions(const RenderBuffer& render_buffer,
                      const FftData& G,
                      size_t num_partitions,
                      std::vector<std::vector<FftData>>* H) {
-  ArrayView<const std::vector<FftData>> render_buffer_data =
+  std::span<const std::vector<FftData>> render_buffer_data =
       render_buffer.GetFftBuffer();
   size_t index = render_buffer.Position();
   const size_t num_render_channels = render_buffer_data[index].size();
@@ -157,7 +157,7 @@ void AdaptPartitions_Neon(const RenderBuffer& render_buffer,
                           const FftData& G,
                           size_t num_partitions,
                           std::vector<std::vector<FftData>>* H) {
-  webrtc::ArrayView<const std::vector<FftData>> render_buffer_data =
+  std::span<const std::vector<FftData>> render_buffer_data =
       render_buffer.GetFftBuffer();
   const size_t num_render_channels = render_buffer_data[0].size();
   const size_t lim1 = std::min(
@@ -223,7 +223,7 @@ void AdaptPartitions_Sse2(const RenderBuffer& render_buffer,
                           const FftData& G,
                           size_t num_partitions,
                           std::vector<std::vector<FftData>>* H) {
-  ArrayView<const std::vector<FftData>> render_buffer_data =
+  std::span<const std::vector<FftData>> render_buffer_data =
       render_buffer.GetFftBuffer();
   const size_t num_render_channels = render_buffer_data[0].size();
   const size_t lim1 = std::min(
@@ -294,7 +294,7 @@ void ApplyFilter(const RenderBuffer& render_buffer,
   S->re.fill(0.f);
   S->im.fill(0.f);
 
-  ArrayView<const std::vector<FftData>> render_buffer_data =
+  std::span<const std::vector<FftData>> render_buffer_data =
       render_buffer.GetFftBuffer();
   size_t index = render_buffer.Position();
   const size_t num_render_channels = render_buffer_data[index].size();
@@ -319,12 +319,12 @@ void ApplyFilter_Neon(const RenderBuffer& render_buffer,
                       const std::vector<std::vector<FftData>>& H,
                       FftData* S) {
   // const RenderBuffer& render_buffer,
-  //                     webrtc::ArrayView<const FftData> H,
+  //                     std::span<const FftData> H,
   //                     FftData* S) {
   RTC_DCHECK_GE(H.size(), H.size() - 1);
   S->Clear();
 
-  webrtc::ArrayView<const std::vector<FftData>> render_buffer_data =
+  std::span<const std::vector<FftData>> render_buffer_data =
       render_buffer.GetFftBuffer();
   const size_t num_render_channels = render_buffer_data[0].size();
   const size_t lim1 = std::min(
@@ -389,13 +389,13 @@ void ApplyFilter_Sse2(const RenderBuffer& render_buffer,
                       const std::vector<std::vector<FftData>>& H,
                       FftData* S) {
   // const RenderBuffer& render_buffer,
-  //                     webrtc::ArrayView<const FftData> H,
+  //                     std::span<const FftData> H,
   //                     FftData* S) {
   RTC_DCHECK_GE(H.size(), H.size() - 1);
   S->re.fill(0.f);
   S->im.fill(0.f);
 
-  ArrayView<const std::vector<FftData>> render_buffer_data =
+  std::span<const std::vector<FftData>> render_buffer_data =
       render_buffer.GetFftBuffer();
   const size_t num_render_channels = render_buffer_data[0].size();
   const size_t lim1 = std::min(
diff --git a/modules/audio_processing/aec3/adaptive_fir_filter.h b/modules/audio_processing/aec3/adaptive_fir_filter.h
index 34c06f4367..681cd23d56 100644
--- a/modules/audio_processing/aec3/adaptive_fir_filter.h
+++ b/modules/audio_processing/aec3/adaptive_fir_filter.h
@@ -17,7 +17,6 @@
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/aec3_fft.h"
 #include "modules/audio_processing/aec3/fft_data.h"
diff --git a/modules/audio_processing/aec3/adaptive_fir_filter_avx2.cc b/modules/audio_processing/aec3/adaptive_fir_filter_avx2.cc
index f7ba1fda86..f250f84806 100644
--- a/modules/audio_processing/aec3/adaptive_fir_filter_avx2.cc
+++ b/modules/audio_processing/aec3/adaptive_fir_filter_avx2.cc
@@ -13,9 +13,9 @@
 #include <algorithm>
 #include <array>
 #include <cstddef>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/adaptive_fir_filter.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/fft_data.h"
@@ -63,7 +63,7 @@ void AdaptPartitions_Avx2(const RenderBuffer& render_buffer,
                           const FftData& G,
                           size_t num_partitions,
                           std::vector<std::vector<FftData>>* H) {
-  ArrayView<const std::vector<FftData>> render_buffer_data =
+  std::span<const std::vector<FftData>> render_buffer_data =
       render_buffer.GetFftBuffer();
   const size_t num_render_channels = render_buffer_data[0].size();
   const size_t lim1 = std::min(
@@ -134,7 +134,7 @@ void ApplyFilter_Avx2(const RenderBuffer& render_buffer,
   S->re.fill(0.f);
   S->im.fill(0.f);
 
-  ArrayView<const std::vector<FftData>> render_buffer_data =
+  std::span<const std::vector<FftData>> render_buffer_data =
       render_buffer.GetFftBuffer();
   const size_t num_render_channels = render_buffer_data[0].size();
   const size_t lim1 = std::min(
diff --git a/modules/audio_processing/aec3/adaptive_fir_filter_erl.cc b/modules/audio_processing/aec3/adaptive_fir_filter_erl.cc
index 11f83af7fe..77d1487352 100644
--- a/modules/audio_processing/aec3/adaptive_fir_filter_erl.cc
+++ b/modules/audio_processing/aec3/adaptive_fir_filter_erl.cc
@@ -14,9 +14,9 @@
 #include <array>
 #include <cstddef>
 #include <functional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "rtc_base/checks.h"
 
@@ -36,7 +36,7 @@ namespace aec3 {
 // Computes and stores the echo return loss estimate of the filter, which is the
 // sum of the partition frequency responses.
 void ErlComputer(const std::vector<std::array<float, kFftLengthBy2Plus1>>& H2,
-                 ArrayView<float> erl) {
+                 std::span<float> erl) {
   std::fill(erl.begin(), erl.end(), 0.f);
   for (auto& H2_j : H2) {
     std::transform(H2_j.begin(), H2_j.end(), erl.begin(), erl.begin(),
@@ -49,7 +49,7 @@ void ErlComputer(const std::vector<std::array<float, kFftLengthBy2Plus1>>& H2,
 // sum of the partition frequency responses.
 void ErlComputer_NEON(
     const std::vector<std::array<float, kFftLengthBy2Plus1>>& H2,
-    webrtc::ArrayView<float> erl) {
+    std::span<float> erl) {
   std::fill(erl.begin(), erl.end(), 0.f);
   for (auto& H2_j : H2) {
     for (size_t k = 0; k < kFftLengthBy2; k += 4) {
@@ -68,7 +68,7 @@ void ErlComputer_NEON(
 // sum of the partition frequency responses.
 void ErlComputer_SSE2(
     const std::vector<std::array<float, kFftLengthBy2Plus1>>& H2,
-    ArrayView<float> erl) {
+    std::span<float> erl) {
   std::fill(erl.begin(), erl.end(), 0.f);
   for (auto& H2_j : H2) {
     for (size_t k = 0; k < kFftLengthBy2; k += 4) {
@@ -86,7 +86,7 @@ void ErlComputer_SSE2(
 
 void ComputeErl(const Aec3Optimization& optimization,
                 const std::vector<std::array<float, kFftLengthBy2Plus1>>& H2,
-                ArrayView<float> erl) {
+                std::span<float> erl) {
   RTC_DCHECK_EQ(kFftLengthBy2Plus1, erl.size());
   // Update the frequency response and echo return loss for the filter.
   switch (optimization) {
diff --git a/modules/audio_processing/aec3/adaptive_fir_filter_erl.h b/modules/audio_processing/aec3/adaptive_fir_filter_erl.h
index 68da2d2d66..d504d270ec 100644
--- a/modules/audio_processing/aec3/adaptive_fir_filter_erl.h
+++ b/modules/audio_processing/aec3/adaptive_fir_filter_erl.h
@@ -14,9 +14,9 @@
 #include <stddef.h>
 
 #include <array>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "rtc_base/system/arch.h"
 
@@ -26,20 +26,20 @@ namespace aec3 {
 // Computes and stores the echo return loss estimate of the filter, which is the
 // sum of the partition frequency responses.
 void ErlComputer(const std::vector<std::array<float, kFftLengthBy2Plus1>>& H2,
-                 ArrayView<float> erl);
+                 std::span<float> erl);
 #if defined(WEBRTC_HAS_NEON)
 void ErlComputer_NEON(
     const std::vector<std::array<float, kFftLengthBy2Plus1>>& H2,
-    webrtc::ArrayView<float> erl);
+    std::span<float> erl);
 #endif
 #if defined(WEBRTC_ARCH_X86_FAMILY)
 void ErlComputer_SSE2(
     const std::vector<std::array<float, kFftLengthBy2Plus1>>& H2,
-    ArrayView<float> erl);
+    std::span<float> erl);
 
 void ErlComputer_AVX2(
     const std::vector<std::array<float, kFftLengthBy2Plus1>>& H2,
-    ArrayView<float> erl);
+    std::span<float> erl);
 #endif
 
 }  // namespace aec3
@@ -47,7 +47,7 @@ void ErlComputer_AVX2(
 // Computes the echo return loss based on a frequency response.
 void ComputeErl(const Aec3Optimization& optimization,
                 const std::vector<std::array<float, kFftLengthBy2Plus1>>& H2,
-                ArrayView<float> erl);
+                std::span<float> erl);
 
 }  // namespace webrtc
 
diff --git a/modules/audio_processing/aec3/adaptive_fir_filter_erl_avx2.cc b/modules/audio_processing/aec3/adaptive_fir_filter_erl_avx2.cc
index 6abbcaec5a..c3606aa10b 100644
--- a/modules/audio_processing/aec3/adaptive_fir_filter_erl_avx2.cc
+++ b/modules/audio_processing/aec3/adaptive_fir_filter_erl_avx2.cc
@@ -13,9 +13,9 @@
 #include <algorithm>
 #include <array>
 #include <cstddef>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/adaptive_fir_filter_erl.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 
@@ -27,7 +27,7 @@ namespace aec3 {
 // sum of the partition frequency responses.
 void ErlComputer_AVX2(
     const std::vector<std::array<float, kFftLengthBy2Plus1>>& H2,
-    ArrayView<float> erl) {
+    std::span<float> erl) {
   std::fill(erl.begin(), erl.end(), 0.f);
   for (auto& H2_j : H2) {
     for (size_t k = 0; k < kFftLengthBy2; k += 8) {
diff --git a/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc b/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc
index 2e600be46d..60e9628657 100644
--- a/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc
+++ b/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc
@@ -16,11 +16,11 @@
 #include <memory>
 #include <numeric>
 #include <optional>
+#include <span>
 #include <string>
 #include <tuple>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "api/environment/environment_factory.h"
 #include "modules/audio_processing/aec3/adaptive_fir_filter_erl.h"
@@ -546,11 +546,11 @@ TEST_P(AdaptiveFirFilterMultiChannel, FilterAndAdapt) {
         num_render_channels);
     for (size_t ch = 0; ch < num_render_channels; ++ch) {
       x_hp_filter[ch] = std::make_unique<CascadedBiQuadFilter>(
-          ArrayView<const CascadedBiQuadFilter::BiQuadCoefficients>(
+          std::span<const CascadedBiQuadFilter::BiQuadCoefficients>(
               kHighPassFilterCoefficients));
     }
     CascadedBiQuadFilter y_hp_filter(
-        (ArrayView<const CascadedBiQuadFilter::BiQuadCoefficients>(
+        (std::span<const CascadedBiQuadFilter::BiQuadCoefficients>(
             kHighPassFilterCoefficients)));
 
     SCOPED_TRACE(ProduceDebugText(num_render_channels, delay_samples));
diff --git a/modules/audio_processing/aec3/aec3_fft.cc b/modules/audio_processing/aec3/aec3_fft.cc
index 7652e306a0..360590e358 100644
--- a/modules/audio_processing/aec3/aec3_fft.cc
+++ b/modules/audio_processing/aec3/aec3_fft.cc
@@ -14,8 +14,8 @@
 #include <array>
 #include <functional>
 #include <iterator>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/fft_data.h"
 #include "rtc_base/checks.h"
@@ -88,7 +88,7 @@ bool IsSse2Available() {
 Aec3Fft::Aec3Fft() : ooura_fft_(IsSse2Available()) {}
 
 // TODO(peah): Change x to be std::array once the rest of the code allows this.
-void Aec3Fft::ZeroPaddedFft(ArrayView<const float> x,
+void Aec3Fft::ZeroPaddedFft(std::span<const float> x,
                             Window window,
                             FftData* X) const {
   RTC_DCHECK(X);
@@ -114,8 +114,8 @@ void Aec3Fft::ZeroPaddedFft(ArrayView<const float> x,
   Fft(&fft, X);
 }
 
-void Aec3Fft::PaddedFft(ArrayView<const float> x,
-                        ArrayView<const float> x_old,
+void Aec3Fft::PaddedFft(std::span<const float> x,
+                        std::span<const float> x_old,
                         Window window,
                         FftData* X) const {
   RTC_DCHECK(X);
diff --git a/modules/audio_processing/aec3/aec3_fft.h b/modules/audio_processing/aec3/aec3_fft.h
index 83d2a2e919..2362a51df0 100644
--- a/modules/audio_processing/aec3/aec3_fft.h
+++ b/modules/audio_processing/aec3/aec3_fft.h
@@ -12,8 +12,8 @@
 #define MODULES_AUDIO_PROCESSING_AEC3_AEC3_FFT_H_
 
 #include <array>
+#include <span>
 
-#include "api/array_view.h"
 #include "common_audio/third_party/ooura/fft_size_128/ooura_fft.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/fft_data.h"
@@ -48,19 +48,19 @@ class Aec3Fft {
 
   // Windows the input using a Hanning window, and then adds padding of
   // kFftLengthBy2 initial zeros before computing the Fft.
-  void ZeroPaddedFft(ArrayView<const float> x, Window window, FftData* X) const;
+  void ZeroPaddedFft(std::span<const float> x, Window window, FftData* X) const;
 
   // Concatenates the kFftLengthBy2 values long x and x_old before computing the
   // Fft. After that, x is copied to x_old.
-  void PaddedFft(ArrayView<const float> x,
-                 ArrayView<const float> x_old,
+  void PaddedFft(std::span<const float> x,
+                 std::span<const float> x_old,
                  FftData* X) const {
     PaddedFft(x, x_old, Window::kRectangular, X);
   }
 
   // Padded Fft using a time-domain window.
-  void PaddedFft(ArrayView<const float> x,
-                 ArrayView<const float> x_old,
+  void PaddedFft(std::span<const float> x,
+                 std::span<const float> x_old,
                  Window window,
                  FftData* X) const;
 
diff --git a/modules/audio_processing/aec3/aec_state.cc b/modules/audio_processing/aec3/aec_state.cc
index 750c89ab2f..d97fda3b9c 100644
--- a/modules/audio_processing/aec3/aec_state.cc
+++ b/modules/audio_processing/aec3/aec_state.cc
@@ -17,9 +17,9 @@
 #include <cstddef>
 #include <numeric>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "api/environment/environment.h"
 #include "api/field_trials_view.h"
@@ -59,7 +59,7 @@ void ComputeAvgRenderReverb(
     int delay_blocks,
     float reverb_decay,
     ReverbModel* reverb_model,
-    ArrayView<float, kFftLengthBy2Plus1> reverb_power_spectrum) {
+    std::span<float, kFftLengthBy2Plus1> reverb_power_spectrum) {
   RTC_DCHECK(reverb_model);
   const size_t num_render_channels = spectrum_buffer.buffer[0].size();
   int idx_at_delay =
@@ -67,13 +67,13 @@ void ComputeAvgRenderReverb(
   int idx_past = spectrum_buffer.IncIndex(idx_at_delay);
 
   std::array<float, kFftLengthBy2Plus1> X2_data;
-  ArrayView<const float> X2;
+  std::span<const float> X2;
   if (num_render_channels > 1) {
     auto average_channels =
         [](size_t num_render_channels,
-           ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+           std::span<const std::array<float, kFftLengthBy2Plus1>>
                spectrum_band_0,
-           ArrayView<float, kFftLengthBy2Plus1> render_power) {
+           std::span<float, kFftLengthBy2Plus1> render_power) {
           std::fill(render_power.begin(), render_power.end(), 0.f);
           for (size_t ch = 0; ch < num_render_channels; ++ch) {
             for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) {
@@ -101,7 +101,7 @@ void ComputeAvgRenderReverb(
     X2 = spectrum_buffer.buffer[idx_at_delay][/*channel=*/0];
   }
 
-  ArrayView<const float, kFftLengthBy2Plus1> reverb_power =
+  std::span<const float, kFftLengthBy2Plus1> reverb_power =
       reverb_model->reverb();
   for (size_t k = 0; k < X2.size(); ++k) {
     reverb_power_spectrum[k] = X2[k] + reverb_power[k];
@@ -112,7 +112,7 @@ void ComputeAvgRenderReverb(
 
 std::atomic<int> AecState::instance_count_(0);
 
-void AecState::GetResidualEchoScaling(ArrayView<float> residual_scaling) const {
+void AecState::GetResidualEchoScaling(std::span<float> residual_scaling) const {
   bool filter_has_had_time_to_converge;
   if (config_.filter.conservative_initial_phase) {
     filter_has_had_time_to_converge =
@@ -189,13 +189,13 @@ void AecState::HandleEchoPathChange(
 
 void AecState::Update(
     const std::optional<DelayEstimate>& external_delay,
-    ArrayView<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
+    std::span<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
         adaptive_filter_frequency_responses,
-    ArrayView<const std::vector<float>> adaptive_filter_impulse_responses,
+    std::span<const std::vector<float>> adaptive_filter_impulse_responses,
     const RenderBuffer& render_buffer,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> E2_refined,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> Y2,
-    ArrayView<const SubtractorOutput> subtractor_output) {
+    std::span<const std::array<float, kFftLengthBy2Plus1>> E2_refined,
+    std::span<const std::array<float, kFftLengthBy2Plus1>> Y2,
+    std::span<const SubtractorOutput> subtractor_output) {
   RTC_DCHECK_EQ(num_capture_channels_, Y2.size());
   RTC_DCHECK_EQ(num_capture_channels_, subtractor_output.size());
   RTC_DCHECK_EQ(num_capture_channels_,
@@ -376,7 +376,7 @@ AecState::FilterDelay::FilterDelay(const EchoCanceller3Config& config,
       min_filter_delay_(delay_headroom_blocks_) {}
 
 void AecState::FilterDelay::Update(
-    ArrayView<const int> analyzer_filter_delay_estimates_blocks,
+    std::span<const int> analyzer_filter_delay_estimates_blocks,
     const std::optional<DelayEstimate>& external_delay,
     size_t blocks_with_proper_filter_adaptation) {
   // Update the delay based on the external delay.
@@ -464,7 +464,7 @@ void AecState::SaturationDetector::Update(
     const Block& x,
     bool saturated_capture,
     bool usable_linear_estimate,
-    ArrayView<const SubtractorOutput> subtractor_output,
+    std::span<const SubtractorOutput> subtractor_output,
     float echo_path_gain) {
   saturated_echo_ = false;
   if (!saturated_capture) {
@@ -482,7 +482,7 @@ void AecState::SaturationDetector::Update(
   } else {
     float max_sample = 0.f;
     for (int ch = 0; ch < x.NumChannels(); ++ch) {
-      ArrayView<const float, kBlockSize> x_ch = x.View(/*band=*/0, ch);
+      std::span<const float, kBlockSize> x_ch = x.View(/*band=*/0, ch);
       for (float sample : x_ch) {
         max_sample = std::max(max_sample, fabsf(sample));
       }
diff --git a/modules/audio_processing/aec3/aec_state.h b/modules/audio_processing/aec3/aec_state.h
index 9c5adc7d3c..df5bc65b9b 100644
--- a/modules/audio_processing/aec3/aec_state.h
+++ b/modules/audio_processing/aec3/aec_state.h
@@ -17,9 +17,9 @@
 #include <atomic>
 #include <memory>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "api/environment/environment.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
@@ -67,7 +67,7 @@ class AecState {
 
   // Returns the appropriate scaling of the residual echo to match the
   // audibility.
-  void GetResidualEchoScaling(ArrayView<float> residual_scaling) const;
+  void GetResidualEchoScaling(std::span<float> residual_scaling) const;
 
   // Returns whether the stationary properties of the signals are used in the
   // aec.
@@ -76,13 +76,13 @@ class AecState {
   }
 
   // Returns the ERLE.
-  ArrayView<const std::array<float, kFftLengthBy2Plus1>> Erle(
+  std::span<const std::array<float, kFftLengthBy2Plus1>> Erle(
       bool onset_compensated) const {
     return erle_estimator_.Erle(onset_compensated);
   }
 
   // Returns the non-capped ERLE.
-  ArrayView<const std::array<float, kFftLengthBy2Plus1>> ErleUnbounded() const {
+  std::span<const std::array<float, kFftLengthBy2Plus1>> ErleUnbounded() const {
     return erle_estimator_.ErleUnbounded();
   }
 
@@ -129,7 +129,7 @@ class AecState {
   }
 
   // Return the frequency response of the reverberant echo.
-  ArrayView<const float> GetReverbFrequencyResponse() const {
+  std::span<const float> GetReverbFrequencyResponse() const {
     return reverb_model_estimator_.GetReverbFrequencyResponse();
   }
 
@@ -143,13 +143,13 @@ class AecState {
   // TODO(bugs.webrtc.org/10913): Compute multi-channel ERL.
   void Update(
       const std::optional<DelayEstimate>& external_delay,
-      ArrayView<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
+      std::span<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
           adaptive_filter_frequency_responses,
-      ArrayView<const std::vector<float>> adaptive_filter_impulse_responses,
+      std::span<const std::vector<float>> adaptive_filter_impulse_responses,
       const RenderBuffer& render_buffer,
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>> E2_refined,
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>> Y2,
-      ArrayView<const SubtractorOutput> subtractor_output);
+      std::span<const std::array<float, kFftLengthBy2Plus1>> E2_refined,
+      std::span<const std::array<float, kFftLengthBy2Plus1>> Y2,
+      std::span<const SubtractorOutput> subtractor_output);
 
   // Returns filter length in blocks.
   int FilterLengthBlocks() const {
@@ -215,7 +215,7 @@ class AecState {
 
     // Returns the delay in blocks relative to the beginning of the filter that
     // corresponds to the direct path of the echo.
-    ArrayView<const int> DirectPathFilterDelays() const {
+    std::span<const int> DirectPathFilterDelays() const {
       return filter_delays_blocks_;
     }
 
@@ -224,7 +224,7 @@ class AecState {
     int MinDirectPathFilterDelay() const { return min_filter_delay_; }
 
     // Updates the delay estimates based on new data.
-    void Update(ArrayView<const int> analyzer_filter_delay_estimates_blocks,
+    void Update(std::span<const int> analyzer_filter_delay_estimates_blocks,
                 const std::optional<DelayEstimate>& external_delay,
                 size_t blocks_with_proper_filter_adaptation);
 
@@ -287,7 +287,7 @@ class AecState {
     void Update(const Block& x,
                 bool saturated_capture,
                 bool usable_linear_estimate,
-                ArrayView<const SubtractorOutput> subtractor_output,
+                std::span<const SubtractorOutput> subtractor_output,
                 float echo_path_gain);
 
    private:
diff --git a/modules/audio_processing/aec3/alignment_mixer.cc b/modules/audio_processing/aec3/alignment_mixer.cc
index 0370aebcaf..c3e36b22b7 100644
--- a/modules/audio_processing/aec3/alignment_mixer.cc
+++ b/modules/audio_processing/aec3/alignment_mixer.cc
@@ -12,8 +12,8 @@
 #include <algorithm>
 #include <cstddef>
 #include <cstring>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/block.h"
@@ -70,7 +70,7 @@ AlignmentMixer::AlignmentMixer(size_t num_channels,
 }
 
 void AlignmentMixer::ProduceOutput(const Block& x,
-                                   ArrayView<float, kBlockSize> y) {
+                                   std::span<float, kBlockSize> y) {
   RTC_DCHECK_EQ(x.NumChannels(), num_channels_);
 
   if (selection_variant_ == MixingVariant::kDownmix) {
@@ -85,7 +85,7 @@ void AlignmentMixer::ProduceOutput(const Block& x,
 }
 
 void AlignmentMixer::Downmix(const Block& x,
-                             ArrayView<float, kBlockSize> y) const {
+                             std::span<float, kBlockSize> y) const {
   RTC_DCHECK_EQ(x.NumChannels(), num_channels_);
   RTC_DCHECK_GE(num_channels_, 2);
   std::memcpy(&y[0], x.View(/*band=*/0, /*channel=*/0).data(),
@@ -122,7 +122,7 @@ int AlignmentMixer::SelectChannel(const Block& x) {
 
   for (int ch = 0; ch < num_ch_to_analyze; ++ch) {
     float x2_sum = 0.f;
-    ArrayView<const float, kBlockSize> x_ch = x.View(/*band=*/0, ch);
+    std::span<const float, kBlockSize> x_ch = x.View(/*band=*/0, ch);
     for (size_t i = 0; i < kBlockSize; ++i) {
       x2_sum += x_ch[i] * x_ch[i];
     }
diff --git a/modules/audio_processing/aec3/alignment_mixer.h b/modules/audio_processing/aec3/alignment_mixer.h
index 29bf14b675..ecc015d787 100644
--- a/modules/audio_processing/aec3/alignment_mixer.h
+++ b/modules/audio_processing/aec3/alignment_mixer.h
@@ -13,9 +13,9 @@
 
 #include <array>
 #include <cstddef>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/block.h"
@@ -36,7 +36,7 @@ class AlignmentMixer {
                  float excitation_limit,
                  bool prefer_first_two_channels);
 
-  void ProduceOutput(const Block& x, ArrayView<float, kBlockSize> y);
+  void ProduceOutput(const Block& x, std::span<float, kBlockSize> y);
 
   enum class MixingVariant { kDownmix, kAdaptive, kFixed };
 
@@ -51,7 +51,7 @@ class AlignmentMixer {
   int selected_channel_ = 0;
   size_t block_counter_ = 0;
 
-  void Downmix(const Block& x, ArrayView<float, kBlockSize> y) const;
+  void Downmix(const Block& x, std::span<float, kBlockSize> y) const;
   int SelectChannel(const Block& x);
 };
 }  // namespace webrtc
diff --git a/modules/audio_processing/aec3/alignment_mixer_unittest.cc b/modules/audio_processing/aec3/alignment_mixer_unittest.cc
index 72c2649a48..e21e2118e8 100644
--- a/modules/audio_processing/aec3/alignment_mixer_unittest.cc
+++ b/modules/audio_processing/aec3/alignment_mixer_unittest.cc
@@ -15,7 +15,6 @@
 #include <cstddef>
 #include <string>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/block.h"
 #include "rtc_base/checks.h"
diff --git a/modules/audio_processing/aec3/block.h b/modules/audio_processing/aec3/block.h
index 75f032d0f2..164ba1824a 100644
--- a/modules/audio_processing/aec3/block.h
+++ b/modules/audio_processing/aec3/block.h
@@ -12,10 +12,10 @@
 #define MODULES_AUDIO_PROCESSING_AEC3_BLOCK_H_
 
 #include <algorithm>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 
 namespace webrtc {
@@ -58,14 +58,14 @@ class Block {
     return begin(band, channel) + kBlockSize;
   }
 
-  // Access data via ArrayView.
-  ArrayView<float, kBlockSize> View(int band, int channel) {
-    return ArrayView<float, kBlockSize>(&data_[GetIndex(band, channel)],
+  // Access data via std::span.
+  std::span<float, kBlockSize> View(int band, int channel) {
+    return std::span<float, kBlockSize>(&data_[GetIndex(band, channel)],
                                         kBlockSize);
   }
 
-  ArrayView<const float, kBlockSize> View(int band, int channel) const {
-    return ArrayView<const float, kBlockSize>(&data_[GetIndex(band, channel)],
+  std::span<const float, kBlockSize> View(int band, int channel) const {
+    return std::span<const float, kBlockSize>(&data_[GetIndex(band, channel)],
                                               kBlockSize);
   }
 
diff --git a/modules/audio_processing/aec3/block_delay_buffer.cc b/modules/audio_processing/aec3/block_delay_buffer.cc
index 3d67aae6cb..84df4e50c3 100644
--- a/modules/audio_processing/aec3/block_delay_buffer.cc
+++ b/modules/audio_processing/aec3/block_delay_buffer.cc
@@ -10,9 +10,9 @@
 #include "modules/audio_processing/aec3/block_delay_buffer.h"
 
 #include <cstddef>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/audio_buffer.h"
 #include "rtc_base/checks.h"
 
@@ -44,7 +44,7 @@ void BlockDelayBuffer::DelaySignal(AudioBuffer* frame) {
   for (size_t ch = 0; ch < num_channels; ++ch) {
     RTC_DCHECK_EQ(buf_[ch].size(), frame->num_bands());
     RTC_DCHECK_EQ(buf_[ch].size(), num_bands);
-    ArrayView<float* const> frame_ch(frame->split_bands(ch), num_bands);
+    std::span<float* const> frame_ch(frame->split_bands(ch), num_bands);
     const size_t delay = delay_;
 
     for (size_t band = 0; band < num_bands; ++band) {
diff --git a/modules/audio_processing/aec3/block_framer.cc b/modules/audio_processing/aec3/block_framer.cc
index 58d1effba7..f9cadb9a8a 100644
--- a/modules/audio_processing/aec3/block_framer.cc
+++ b/modules/audio_processing/aec3/block_framer.cc
@@ -12,9 +12,9 @@
 
 #include <algorithm>
 #include <cstddef>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/block.h"
 #include "rtc_base/checks.h"
@@ -54,7 +54,7 @@ void BlockFramer::InsertBlock(const Block& block) {
 
 void BlockFramer::InsertBlockAndExtractSubFrame(
     const Block& block,
-    std::vector<std::vector<ArrayView<float>>>* sub_frame) {
+    std::vector<std::vector<std::span<float>>>* sub_frame) {
   RTC_DCHECK(sub_frame);
   RTC_DCHECK_EQ(num_bands_, block.NumBands());
   RTC_DCHECK_EQ(num_channels_, block.NumChannels());
diff --git a/modules/audio_processing/aec3/block_framer.h b/modules/audio_processing/aec3/block_framer.h
index c7ddc4987e..896bc6b5dd 100644
--- a/modules/audio_processing/aec3/block_framer.h
+++ b/modules/audio_processing/aec3/block_framer.h
@@ -12,9 +12,9 @@
 #define MODULES_AUDIO_PROCESSING_AEC3_BLOCK_FRAMER_H_
 
 #include <cstddef>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/block.h"
 
 namespace webrtc {
@@ -37,7 +37,7 @@ class BlockFramer {
   // Adds a 64 sample block and extracts an 80 sample subframe.
   void InsertBlockAndExtractSubFrame(
       const Block& block,
-      std::vector<std::vector<ArrayView<float>>>* sub_frame);
+      std::vector<std::vector<std::span<float>>>* sub_frame);
 
  private:
   const size_t num_bands_;
diff --git a/modules/audio_processing/aec3/block_framer_unittest.cc b/modules/audio_processing/aec3/block_framer_unittest.cc
index b898f32db6..fa9aaff15f 100644
--- a/modules/audio_processing/aec3/block_framer_unittest.cc
+++ b/modules/audio_processing/aec3/block_framer_unittest.cc
@@ -11,10 +11,10 @@
 #include "modules/audio_processing/aec3/block_framer.h"
 
 #include <cstddef>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/block.h"
 #include "rtc_base/checks.h"
@@ -26,12 +26,12 @@ namespace {
 
 void SetupSubFrameView(
     std::vector<std::vector<std::vector<float>>>* sub_frame,
-    std::vector<std::vector<ArrayView<float>>>* sub_frame_view) {
+    std::vector<std::vector<std::span<float>>>* sub_frame_view) {
   for (size_t band = 0; band < sub_frame_view->size(); ++band) {
     for (size_t channel = 0; channel < (*sub_frame_view)[band].size();
          ++channel) {
       (*sub_frame_view)[band][channel] =
-          ArrayView<float>((*sub_frame)[band][channel].data(),
+          std::span<float>((*sub_frame)[band][channel].data(),
                            (*sub_frame)[band][channel].size());
     }
   }
@@ -52,7 +52,7 @@ float ComputeSampleValue(size_t chunk_counter,
 bool VerifySubFrame(
     size_t sub_frame_counter,
     int offset,
-    const std::vector<std::vector<ArrayView<float>>>& sub_frame_view) {
+    const std::vector<std::vector<std::span<float>>>& sub_frame_view) {
   for (size_t band = 0; band < sub_frame_view.size(); ++band) {
     for (size_t channel = 0; channel < sub_frame_view[band].size(); ++channel) {
       for (size_t sample = 0; sample < sub_frame_view[band][channel].size();
@@ -89,8 +89,8 @@ void RunFramerTest(int sample_rate_hz, size_t num_channels) {
   std::vector<std::vector<std::vector<float>>> output_sub_frame(
       num_bands, std::vector<std::vector<float>>(
                      num_channels, std::vector<float>(kSubFrameLength, 0.f)));
-  std::vector<std::vector<ArrayView<float>>> output_sub_frame_view(
-      num_bands, std::vector<ArrayView<float>>(num_channels));
+  std::vector<std::vector<std::span<float>>> output_sub_frame_view(
+      num_bands, std::vector<std::span<float>>(num_channels));
   SetupSubFrameView(&output_sub_frame, &output_sub_frame_view);
   BlockFramer framer(num_bands, num_channels);
 
@@ -128,9 +128,9 @@ void RunWronglySizedInsertAndExtractParametersTest(
       num_sub_frame_bands,
       std::vector<std::vector<float>>(
           num_sub_frame_channels, std::vector<float>(sub_frame_length, 0.f)));
-  std::vector<std::vector<ArrayView<float>>> output_sub_frame_view(
+  std::vector<std::vector<std::span<float>>> output_sub_frame_view(
       output_sub_frame.size(),
-      std::vector<ArrayView<float>>(num_sub_frame_channels));
+      std::vector<std::span<float>>(num_sub_frame_channels));
   SetupSubFrameView(&output_sub_frame, &output_sub_frame_view);
   BlockFramer framer(correct_num_bands, correct_num_channels);
   EXPECT_DEATH(
@@ -151,9 +151,9 @@ void RunWronglySizedInsertParameterTest(int sample_rate_hz,
       correct_num_bands,
       std::vector<std::vector<float>>(
           correct_num_channels, std::vector<float>(kSubFrameLength, 0.f)));
-  std::vector<std::vector<ArrayView<float>>> output_sub_frame_view(
+  std::vector<std::vector<std::span<float>>> output_sub_frame_view(
       output_sub_frame.size(),
-      std::vector<ArrayView<float>>(correct_num_channels));
+      std::vector<std::span<float>>(correct_num_channels));
   SetupSubFrameView(&output_sub_frame, &output_sub_frame_view);
   BlockFramer framer(correct_num_bands, correct_num_channels);
   framer.InsertBlockAndExtractSubFrame(correct_block, &output_sub_frame_view);
@@ -178,8 +178,8 @@ void RunWronglyInsertOrderTest(int sample_rate_hz,
       correct_num_bands,
       std::vector<std::vector<float>>(
           num_channels, std::vector<float>(kSubFrameLength, 0.f)));
-  std::vector<std::vector<ArrayView<float>>> output_sub_frame_view(
-      output_sub_frame.size(), std::vector<ArrayView<float>>(num_channels));
+  std::vector<std::vector<std::span<float>>> output_sub_frame_view(
+      output_sub_frame.size(), std::vector<std::span<float>>(num_channels));
   SetupSubFrameView(&output_sub_frame, &output_sub_frame_view);
   BlockFramer framer(correct_num_bands, num_channels);
   for (size_t k = 0; k < num_preceeding_api_calls; ++k) {
diff --git a/modules/audio_processing/aec3/block_processor_unittest.cc b/modules/audio_processing/aec3/block_processor_unittest.cc
index fe9596d4c4..bc085841d2 100644
--- a/modules/audio_processing/aec3/block_processor_unittest.cc
+++ b/modules/audio_processing/aec3/block_processor_unittest.cc
@@ -13,10 +13,10 @@
 #include <cstddef>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "api/environment/environment.h"
 #include "api/environment/environment_factory.h"
@@ -118,7 +118,7 @@ std::string ProduceDebugText(int sample_rate_hz) {
   return ss.Release();
 }
 
-void FillSampleVector(int call_counter, int delay, ArrayView<float> samples) {
+void FillSampleVector(int call_counter, int delay, std::span<float> samples) {
   for (size_t i = 0; i < samples.size(); ++i) {
     samples[i] = (call_counter - delay) * 10000.0f + i;
   }
diff --git a/modules/audio_processing/aec3/comfort_noise_generator.cc b/modules/audio_processing/aec3/comfort_noise_generator.cc
index 8df81e5cec..b7f30de870 100644
--- a/modules/audio_processing/aec3/comfort_noise_generator.cc
+++ b/modules/audio_processing/aec3/comfort_noise_generator.cc
@@ -18,9 +18,9 @@
 #include <memory>
 #include <numbers>
 #include <numeric>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/fft_data.h"
@@ -60,8 +60,8 @@ constexpr std::array<float,32> kSqrt2Sin = {
 
 void GenerateRandomSinTableIndices(
     uint32_t& seed,
-    ArrayView<int16_t, kFftLengthBy2 - 1> re_sin_table_indices,
-    ArrayView<int16_t, kFftLengthBy2 - 1> im_sin_table_indices) {
+    std::span<int16_t, kFftLengthBy2 - 1> re_sin_table_indices,
+    std::span<int16_t, kFftLengthBy2 - 1> im_sin_table_indices) {
   RTC_DCHECK_EQ(re_sin_table_indices.size(), im_sin_table_indices.size());
 
   constexpr int32_t kSeedMask = 0x80000000 - 1;
@@ -84,8 +84,8 @@ void GenerateRandomSinTableIndices(
 void GenerateComfortNoise(
     Aec3Optimization optimization,
     const std::array<float, kFftLengthBy2Plus1>& N2,
-    ArrayView<int16_t, kFftLengthBy2 - 1> re_sin_table_indices,
-    ArrayView<int16_t, kFftLengthBy2 - 1> im_sin_table_indices,
+    std::span<int16_t, kFftLengthBy2 - 1> re_sin_table_indices,
+    std::span<int16_t, kFftLengthBy2 - 1> im_sin_table_indices,
     FftData* lower_band_noise,
     FftData* upper_band_noise) {
   FftData* N_low = lower_band_noise;
@@ -151,9 +151,9 @@ ComfortNoiseGenerator::~ComfortNoiseGenerator() = default;
 
 void ComfortNoiseGenerator::Compute(
     bool saturated_capture,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> capture_spectrum,
-    ArrayView<FftData> lower_band_noise,
-    ArrayView<FftData> upper_band_noise) {
+    std::span<const std::array<float, kFftLengthBy2Plus1>> capture_spectrum,
+    std::span<FftData> lower_band_noise,
+    std::span<FftData> upper_band_noise) {
   const auto& Y2 = capture_spectrum;
 
   if (!saturated_capture) {
diff --git a/modules/audio_processing/aec3/comfort_noise_generator.h b/modules/audio_processing/aec3/comfort_noise_generator.h
index 3672185f99..c28e269ade 100644
--- a/modules/audio_processing/aec3/comfort_noise_generator.h
+++ b/modules/audio_processing/aec3/comfort_noise_generator.h
@@ -16,9 +16,9 @@
 #include <array>
 #include <cstddef>
 #include <memory>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/fft_data.h"
@@ -53,12 +53,12 @@ class ComfortNoiseGenerator {
   // Computes the comfort noise.
   void Compute(
       bool saturated_capture,
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>> capture_spectrum,
-      ArrayView<FftData> lower_band_noise,
-      ArrayView<FftData> upper_band_noise);
+      std::span<const std::array<float, kFftLengthBy2Plus1>> capture_spectrum,
+      std::span<FftData> lower_band_noise,
+      std::span<FftData> upper_band_noise);
 
   // Returns the estimate of the background noise spectrum.
-  ArrayView<const std::array<float, kFftLengthBy2Plus1>> NoiseSpectrum() const {
+  std::span<const std::array<float, kFftLengthBy2Plus1>> NoiseSpectrum() const {
     return N2_;
   }
 
diff --git a/modules/audio_processing/aec3/decimator.cc b/modules/audio_processing/aec3/decimator.cc
index d0f0e7c82c..473273d2c1 100644
--- a/modules/audio_processing/aec3/decimator.cc
+++ b/modules/audio_processing/aec3/decimator.cc
@@ -11,8 +11,8 @@
 
 #include <array>
 #include <cstddef>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/utility/cascaded_biquad_filter.h"
 #include "rtc_base/checks.h"
@@ -59,20 +59,20 @@ Decimator::Decimator(size_t down_sampling_factor)
     : down_sampling_factor_(down_sampling_factor),
       anti_aliasing_filter_(
           down_sampling_factor_ == 4
-              ? ArrayView<const CascadedBiQuadFilter::BiQuadCoefficients>(
+              ? std::span<const CascadedBiQuadFilter::BiQuadCoefficients>(
                     kLowPassFilterDs4)
-              : ArrayView<const CascadedBiQuadFilter::BiQuadCoefficients>(
+              : std::span<const CascadedBiQuadFilter::BiQuadCoefficients>(
                     kBandPassFilterDs8)),
       noise_reduction_filter_(
           down_sampling_factor_ == 8
-              ? (ArrayView<const CascadedBiQuadFilter::BiQuadCoefficients>(
+              ? (std::span<const CascadedBiQuadFilter::BiQuadCoefficients>(
                     kPassThroughFilter))
-              : (ArrayView<const CascadedBiQuadFilter::BiQuadCoefficients>(
+              : (std::span<const CascadedBiQuadFilter::BiQuadCoefficients>(
                     kHighPassFilter))) {
   RTC_DCHECK(down_sampling_factor_ == 4 || down_sampling_factor_ == 8);
 }
 
-void Decimator::Decimate(ArrayView<const float> in, ArrayView<float> out) {
+void Decimator::Decimate(std::span<const float> in, std::span<float> out) {
   RTC_DCHECK_EQ(kBlockSize, in.size());
   RTC_DCHECK_EQ(kBlockSize / down_sampling_factor_, out.size());
   std::array<float, kBlockSize> x;
diff --git a/modules/audio_processing/aec3/decimator.h b/modules/audio_processing/aec3/decimator.h
index 805616948f..51d6489515 100644
--- a/modules/audio_processing/aec3/decimator.h
+++ b/modules/audio_processing/aec3/decimator.h
@@ -12,8 +12,8 @@
 #define MODULES_AUDIO_PROCESSING_AEC3_DECIMATOR_H_
 
 #include <cstddef>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/utility/cascaded_biquad_filter.h"
 
 namespace webrtc {
@@ -27,7 +27,7 @@ class Decimator {
   Decimator& operator=(const Decimator&) = delete;
 
   // Downsamples the signal.
-  void Decimate(ArrayView<const float> in, ArrayView<float> out);
+  void Decimate(std::span<const float> in, std::span<float> out);
 
  private:
   const size_t down_sampling_factor_;
diff --git a/modules/audio_processing/aec3/decimator_unittest.cc b/modules/audio_processing/aec3/decimator_unittest.cc
index 8424c36cc7..d0eb010b68 100644
--- a/modules/audio_processing/aec3/decimator_unittest.cc
+++ b/modules/audio_processing/aec3/decimator_unittest.cc
@@ -16,10 +16,10 @@
 #include <cstring>
 #include <numbers>
 #include <numeric>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/strings/string_builder.h"
@@ -60,17 +60,17 @@ void ProduceDecimatedSinusoidalOutputPower(int sample_rate_hz,
   for (size_t k = 0; k < kNumBlocks; ++k) {
     std::vector<float> sub_block(sub_block_size);
     decimator.Decimate(
-        ArrayView<const float>(&input[k * kBlockSize], kBlockSize), sub_block);
+        std::span<const float>(&input[k * kBlockSize], kBlockSize), sub_block);
 
     std::copy(sub_block.begin(), sub_block.end(),
               output.begin() + k * sub_block_size);
   }
 
   ASSERT_GT(kNumBlocks, kNumStartupBlocks);
-  ArrayView<const float> input_to_evaluate(
+  std::span<const float> input_to_evaluate(
       &input[kNumStartupBlocks * kBlockSize],
       (kNumBlocks - kNumStartupBlocks) * kBlockSize);
-  ArrayView<const float> output_to_evaluate(
+  std::span<const float> output_to_evaluate(
       &output[kNumStartupBlocks * sub_block_size],
       (kNumBlocks - kNumStartupBlocks) * sub_block_size);
   *input_power =
diff --git a/modules/audio_processing/aec3/dominant_nearend_detector.cc b/modules/audio_processing/aec3/dominant_nearend_detector.cc
index c37fa9a4ba..35aff9341b 100644
--- a/modules/audio_processing/aec3/dominant_nearend_detector.cc
+++ b/modules/audio_processing/aec3/dominant_nearend_detector.cc
@@ -14,8 +14,8 @@
 #include <array>
 #include <cstddef>
 #include <numeric>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "rtc_base/checks.h"
@@ -35,15 +35,15 @@ DominantNearendDetector::DominantNearendDetector(
       hold_counters_(num_capture_channels_) {}
 
 void DominantNearendDetector::Update(
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> nearend_spectrum,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+    std::span<const std::array<float, kFftLengthBy2Plus1>> nearend_spectrum,
+    std::span<const std::array<float, kFftLengthBy2Plus1>>
         residual_echo_spectrum,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+    std::span<const std::array<float, kFftLengthBy2Plus1>>
         comfort_noise_spectrum,
     bool initial_state) {
   nearend_state_ = false;
 
-  auto low_frequency_energy = [](ArrayView<const float> spectrum) {
+  auto low_frequency_energy = [](std::span<const float> spectrum) {
     RTC_DCHECK_LE(16, spectrum.size());
     return std::accumulate(spectrum.begin() + 1, spectrum.begin() + 16, 0.f);
   };
diff --git a/modules/audio_processing/aec3/dominant_nearend_detector.h b/modules/audio_processing/aec3/dominant_nearend_detector.h
index afd1516042..cbab36e3a4 100644
--- a/modules/audio_processing/aec3/dominant_nearend_detector.h
+++ b/modules/audio_processing/aec3/dominant_nearend_detector.h
@@ -13,9 +13,9 @@
 
 #include <array>
 #include <cstddef>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/nearend_detector.h"
@@ -33,10 +33,10 @@ class DominantNearendDetector : public NearendDetector {
 
   // Updates the state selection based on latest spectral estimates.
   void Update(
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>> nearend_spectrum,
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+      std::span<const std::array<float, kFftLengthBy2Plus1>> nearend_spectrum,
+      std::span<const std::array<float, kFftLengthBy2Plus1>>
           residual_echo_spectrum,
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+      std::span<const std::array<float, kFftLengthBy2Plus1>>
           comfort_noise_spectrum,
       bool initial_state) override;
 
diff --git a/modules/audio_processing/aec3/echo_audibility.cc b/modules/audio_processing/aec3/echo_audibility.cc
index b7a9430290..55191ef4ad 100644
--- a/modules/audio_processing/aec3/echo_audibility.cc
+++ b/modules/audio_processing/aec3/echo_audibility.cc
@@ -13,10 +13,10 @@
 #include <algorithm>
 #include <cmath>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/block_buffer.h"
 #include "modules/audio_processing/aec3/render_buffer.h"
@@ -33,7 +33,7 @@ EchoAudibility::EchoAudibility(bool use_render_stationarity_at_init)
 EchoAudibility::~EchoAudibility() = default;
 
 void EchoAudibility::Update(const RenderBuffer& render_buffer,
-                            ArrayView<const float> average_reverb,
+                            std::span<const float> average_reverb,
                             int delay_blocks,
                             bool external_delay_seen) {
   UpdateRenderNoiseEstimator(render_buffer.GetSpectrumBuffer(),
@@ -53,7 +53,7 @@ void EchoAudibility::Reset() {
 
 void EchoAudibility::UpdateRenderStationarityFlags(
     const RenderBuffer& render_buffer,
-    ArrayView<const float> average_reverb,
+    std::span<const float> average_reverb,
     int min_channel_delay_blocks) {
   const SpectrumBuffer& spectrum_buffer = render_buffer.GetSpectrumBuffer();
   int idx_at_delay = spectrum_buffer.OffsetIndex(spectrum_buffer.read,
@@ -101,7 +101,7 @@ bool EchoAudibility::IsRenderTooLow(const BlockBuffer& block_buffer) {
          idx = block_buffer.IncIndex(idx)) {
       float max_abs_over_channels = 0.f;
       for (int ch = 0; ch < num_render_channels; ++ch) {
-        ArrayView<const float, kBlockSize> block =
+        std::span<const float, kBlockSize> block =
             block_buffer.buffer[idx].View(/*band=*/0, /*channel=*/ch);
         auto r = std::minmax_element(block.begin(), block.end());
         float max_abs_channel =
diff --git a/modules/audio_processing/aec3/echo_audibility.h b/modules/audio_processing/aec3/echo_audibility.h
index 44df9266c1..35edb5d592 100644
--- a/modules/audio_processing/aec3/echo_audibility.h
+++ b/modules/audio_processing/aec3/echo_audibility.h
@@ -14,8 +14,8 @@
 #include <stddef.h>
 
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/block_buffer.h"
 #include "modules/audio_processing/aec3/render_buffer.h"
 #include "modules/audio_processing/aec3/spectrum_buffer.h"
@@ -33,12 +33,12 @@ class EchoAudibility {
 
   // Feed new render data to the echo audibility estimator.
   void Update(const RenderBuffer& render_buffer,
-              ArrayView<const float> average_reverb,
+              std::span<const float> average_reverb,
               int min_channel_delay_blocks,
               bool external_delay_seen);
   // Get the residual echo scaling.
   void GetResidualEchoScaling(bool filter_has_had_time_to_converge,
-                              ArrayView<float> residual_scaling) const {
+                              std::span<float> residual_scaling) const {
     for (size_t band = 0; band < residual_scaling.size(); ++band) {
       if (render_stationarity_.IsBandStationary(band) &&
           (filter_has_had_time_to_converge ||
@@ -61,7 +61,7 @@ class EchoAudibility {
 
   // Updates the render stationarity flags for the current frame.
   void UpdateRenderStationarityFlags(const RenderBuffer& render_buffer,
-                                     ArrayView<const float> average_reverb,
+                                     std::span<const float> average_reverb,
                                      int delay_blocks);
 
   // Updates the noise estimator with the new render data since the previous
diff --git a/modules/audio_processing/aec3/echo_canceller3.cc b/modules/audio_processing/aec3/echo_canceller3.cc
index 7bf2aa3856..5e99a5b3bc 100644
--- a/modules/audio_processing/aec3/echo_canceller3.cc
+++ b/modules/audio_processing/aec3/echo_canceller3.cc
@@ -14,12 +14,12 @@
 #include <cstddef>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "api/audio/echo_control.h"
 #include "api/audio/neural_residual_echo_estimator.h"
@@ -45,7 +45,7 @@ namespace {
 
 enum class EchoCanceller3ApiCall { kCapture, kRender };
 
-bool DetectSaturation(ArrayView<const float> y) {
+bool DetectSaturation(std::span<const float> y) {
   for (size_t k = 0; k < y.size(); ++k) {
     if (y[k] >= 32700.0f || y[k] <= -32700.0f) {
       return true;
@@ -102,14 +102,14 @@ void RetrieveFieldTrialValue(const FieldTrialsView& field_trials,
 void FillSubFrameView(
     AudioBuffer* frame,
     size_t sub_frame_index,
-    std::vector<std::vector<ArrayView<float>>>* sub_frame_view) {
+    std::vector<std::vector<std::span<float>>>* sub_frame_view) {
   RTC_DCHECK_GE(1, sub_frame_index);
   RTC_DCHECK_LE(0, sub_frame_index);
   RTC_DCHECK_EQ(frame->num_bands(), sub_frame_view->size());
   RTC_DCHECK_EQ(frame->num_channels(), (*sub_frame_view)[0].size());
   for (size_t band = 0; band < sub_frame_view->size(); ++band) {
     for (size_t channel = 0; channel < (*sub_frame_view)[0].size(); ++channel) {
-      (*sub_frame_view)[band][channel] = ArrayView<float>(
+      (*sub_frame_view)[band][channel] = std::span<float>(
           &frame->split_bands(channel)[band][sub_frame_index * kSubFrameLength],
           kSubFrameLength);
     }
@@ -120,7 +120,7 @@ void FillSubFrameView(
     bool proper_downmix_needed,
     std::vector<std::vector<std::vector<float>>>* frame,
     size_t sub_frame_index,
-    std::vector<std::vector<ArrayView<float>>>* sub_frame_view) {
+    std::vector<std::vector<std::span<float>>>* sub_frame_view) {
   RTC_DCHECK_GE(1, sub_frame_index);
   RTC_DCHECK_EQ(frame->size(), sub_frame_view->size());
   const size_t frame_num_channels = (*frame)[0].size();
@@ -148,7 +148,7 @@ void FillSubFrameView(
       }
     }
     for (size_t band = 0; band < frame->size(); ++band) {
-      (*sub_frame_view)[band][/*channel=*/0] = ArrayView<float>(
+      (*sub_frame_view)[band][/*channel=*/0] = std::span<float>(
           &(*frame)[band][/*channel=*/0][sub_frame_index * kSubFrameLength],
           kSubFrameLength);
     }
@@ -156,7 +156,7 @@ void FillSubFrameView(
     RTC_DCHECK_EQ(frame_num_channels, sub_frame_num_channels);
     for (size_t band = 0; band < frame->size(); ++band) {
       for (size_t channel = 0; channel < (*frame)[band].size(); ++channel) {
-        (*sub_frame_view)[band][channel] = ArrayView<float>(
+        (*sub_frame_view)[band][channel] = std::span<float>(
             &(*frame)[band][channel][sub_frame_index * kSubFrameLength],
             kSubFrameLength);
       }
@@ -176,9 +176,9 @@ void ProcessCaptureFrameContent(
     BlockFramer* output_framer,
     BlockProcessor* block_processor,
     Block* linear_output_block,
-    std::vector<std::vector<ArrayView<float>>>* linear_output_sub_frame_view,
+    std::vector<std::vector<std::span<float>>>* linear_output_sub_frame_view,
     Block* capture_block,
-    std::vector<std::vector<ArrayView<float>>>* capture_sub_frame_view) {
+    std::vector<std::vector<std::span<float>>>* capture_sub_frame_view) {
   FillSubFrameView(capture, sub_frame_index, capture_sub_frame_view);
 
   if (linear_output) {
@@ -238,7 +238,7 @@ void BufferRenderFrameContent(
     FrameBlocker* render_blocker,
     BlockProcessor* block_processor,
     Block* block,
-    std::vector<std::vector<ArrayView<float>>>* sub_frame_view) {
+    std::vector<std::vector<std::span<float>>>* sub_frame_view) {
   FillSubFrameView(proper_downmix_needed, render_frame, sub_frame_index,
                    sub_frame_view);
   render_blocker->InsertSubFrameAndExtractBlock(*sub_frame_view, block);
@@ -264,7 +264,7 @@ void CopyBufferIntoFrame(const AudioBuffer& buffer,
   RTC_DCHECK_EQ(AudioBuffer::kSplitBandSize, (*frame)[0][0].size());
   for (size_t band = 0; band < num_bands; ++band) {
     for (size_t channel = 0; channel < num_channels; ++channel) {
-      ArrayView<const float> buffer_view(
+      std::span<const float> buffer_view(
           &buffer.split_bands_const(channel)[band][0],
           AudioBuffer::kSplitBandSize);
       std::copy(buffer_view.begin(), buffer_view.end(),
@@ -790,7 +790,7 @@ EchoCanceller3::EchoCanceller3(
       capture_block_(num_bands_, num_capture_channels_),
       capture_sub_frame_view_(
           num_bands_,
-          std::vector<ArrayView<float>>(num_capture_channels_)) {
+          std::vector<std::span<float>>(num_capture_channels_)) {
   RTC_DCHECK(ValidFullBandRate(sample_rate_hz_));
 
   if (config_selector_.active_config().delay.fixed_capture_delay_samples > 0) {
@@ -811,8 +811,8 @@ EchoCanceller3::EchoCanceller3(
         new BlockFramer(/*num_bands=*/1, num_capture_channels_));
     linear_output_block_ =
         std::make_unique<Block>(/*num_bands=*/1, num_capture_channels_);
-    linear_output_sub_frame_view_ = std::vector<std::vector<ArrayView<float>>>(
-        1, std::vector<ArrayView<float>>(num_capture_channels_));
+    linear_output_sub_frame_view_ = std::vector<std::vector<std::span<float>>>(
+        1, std::vector<std::span<float>>(num_capture_channels_));
   }
 
   Initialize();
@@ -845,8 +845,8 @@ void EchoCanceller3::Initialize() {
       num_render_channels_to_aec_, num_capture_channels_,
       neural_residual_echo_estimator_);
 
-  render_sub_frame_view_ = std::vector<std::vector<ArrayView<float>>>(
-      num_bands_, std::vector<ArrayView<float>>(num_render_channels_to_aec_));
+  render_sub_frame_view_ = std::vector<std::vector<std::span<float>>>(
+      num_bands_, std::vector<std::span<float>>(num_render_channels_to_aec_));
 }
 
 void EchoCanceller3::AnalyzeRender(const AudioBuffer& render) {
@@ -865,7 +865,7 @@ void EchoCanceller3::AnalyzeCapture(const AudioBuffer& capture) {
                         capture.channels_const()[0], sample_rate_hz_, 1);
   saturated_microphone_signal_ = false;
   for (size_t channel = 0; channel < capture.num_channels(); ++channel) {
-    saturated_microphone_signal_ |= DetectSaturation(ArrayView<const float>(
+    saturated_microphone_signal_ |= DetectSaturation(std::span<const float>(
         capture.channels_const()[channel], capture.num_frames()));
     if (saturated_microphone_signal_) {
       break;
@@ -904,7 +904,7 @@ void EchoCanceller3::ProcessCapture(AudioBuffer* capture,
     block_delay_buffer_->DelaySignal(capture);
   }
 
-  ArrayView<float> capture_lower_band = ArrayView<float>(
+  std::span<float> capture_lower_band = std::span<float>(
       &capture->split_bands(0)[0][0], AudioBuffer::kSplitBandSize);
 
   data_dumper_->DumpWav("aec3_capture_input", capture_lower_band, 16000, 1);
diff --git a/modules/audio_processing/aec3/echo_canceller3.h b/modules/audio_processing/aec3/echo_canceller3.h
index 9005cc00b3..e12ee0abe9 100644
--- a/modules/audio_processing/aec3/echo_canceller3.h
+++ b/modules/audio_processing/aec3/echo_canceller3.h
@@ -16,9 +16,9 @@
 #include <atomic>
 #include <memory>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "api/audio/echo_control.h"
 #include "api/audio/neural_residual_echo_estimator.h"
@@ -220,11 +220,11 @@ class EchoCanceller3 : public EchoControl {
   std::unique_ptr<Block> linear_output_block_
       RTC_GUARDED_BY(capture_race_checker_);
   Block capture_block_ RTC_GUARDED_BY(capture_race_checker_);
-  std::vector<std::vector<ArrayView<float>>> render_sub_frame_view_
+  std::vector<std::vector<std::span<float>>> render_sub_frame_view_
       RTC_GUARDED_BY(capture_race_checker_);
-  std::vector<std::vector<ArrayView<float>>> linear_output_sub_frame_view_
+  std::vector<std::vector<std::span<float>>> linear_output_sub_frame_view_
       RTC_GUARDED_BY(capture_race_checker_);
-  std::vector<std::vector<ArrayView<float>>> capture_sub_frame_view_
+  std::vector<std::vector<std::span<float>>> capture_sub_frame_view_
       RTC_GUARDED_BY(capture_race_checker_);
   std::unique_ptr<BlockDelayBuffer> block_delay_buffer_
       RTC_GUARDED_BY(capture_race_checker_);
diff --git a/modules/audio_processing/aec3/echo_canceller3_unittest.cc b/modules/audio_processing/aec3/echo_canceller3_unittest.cc
index 01f0a6ea49..69d71ad810 100644
--- a/modules/audio_processing/aec3/echo_canceller3_unittest.cc
+++ b/modules/audio_processing/aec3/echo_canceller3_unittest.cc
@@ -16,11 +16,11 @@
 #include <deque>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "api/audio/echo_control.h"
 #include "api/audio/neural_residual_echo_estimator.h"
@@ -100,8 +100,8 @@ bool VerifyOutputFrameBitexactness(size_t frame_length,
   return true;
 }
 
-bool VerifyOutputFrameBitexactness(ArrayView<const float> reference,
-                                   ArrayView<const float> frame,
+bool VerifyOutputFrameBitexactness(std::span<const float> reference,
+                                   std::span<const float> frame,
                                    int offset) {
   for (size_t k = 0; k < frame.size(); ++k) {
     int reference_index = static_cast<int>(k) + offset;
@@ -200,10 +200,10 @@ void RunAecInStereo(AudioBuffer& buffer,
                     EchoCanceller3& aec3,
                     float channel_0_value,
                     float channel_1_value) {
-  ArrayView<float> data_channel_0(&buffer.channels()[0][0],
+  std::span<float> data_channel_0(&buffer.channels()[0][0],
                                   buffer.num_frames());
   std::fill(data_channel_0.begin(), data_channel_0.end(), channel_0_value);
-  ArrayView<float> data_channel_1(&buffer.channels()[1][0],
+  std::span<float> data_channel_1(&buffer.channels()[1][0],
                                   buffer.num_frames());
   std::fill(data_channel_1.begin(), data_channel_1.end(), channel_1_value);
   aec3.AnalyzeRender(&buffer);
@@ -214,7 +214,7 @@ void RunAecInStereo(AudioBuffer& buffer,
 void RunAecInSMono(AudioBuffer& buffer,
                    EchoCanceller3& aec3,
                    float channel_0_value) {
-  ArrayView<float> data_channel_0(&buffer.channels()[0][0],
+  std::span<float> data_channel_0(&buffer.channels()[0][0],
                                   buffer.num_frames());
   std::fill(data_channel_0.begin(), data_channel_0.end(), channel_0_value);
   aec3.AnalyzeRender(&buffer);
@@ -1171,14 +1171,14 @@ TEST(EchoCanceller3, InjectedNeuralResidualEchoEstimatorIsUsed) {
     NeuralResidualEchoEstimatorMock() {}
 
     void Estimate(const Block& render,
-                  ArrayView<const std::array<float, 64>> capture,
-                  ArrayView<const std::array<float, 64>> linear_aec_output,
-                  ArrayView<const std::array<float, 65>> S2_linear,
-                  ArrayView<const std::array<float, 65>> Y2,
-                  ArrayView<const std::array<float, 65>> E2,
+                  std::span<const std::array<float, 64>> capture,
+                  std::span<const std::array<float, 64>> linear_aec_output,
+                  std::span<const std::array<float, 65>> S2_linear,
+                  std::span<const std::array<float, 65>> Y2,
+                  std::span<const std::array<float, 65>> E2,
                   bool dominant_nearend,
-                  ArrayView<std::array<float, 65>> R2,
-                  ArrayView<std::array<float, 65>> R2_unbounded) override {
+                  std::span<std::array<float, 65>> R2,
+                  std::span<std::array<float, 65>> R2_unbounded) override {
       residual_echo_estimate_requested_ = true;
       for (auto& R2_ch : R2) {
         R2_ch.fill(0.0f);
diff --git a/modules/audio_processing/aec3/echo_path_delay_estimator.cc b/modules/audio_processing/aec3/echo_path_delay_estimator.cc
index 30b66896b3..db238d82be 100644
--- a/modules/audio_processing/aec3/echo_path_delay_estimator.cc
+++ b/modules/audio_processing/aec3/echo_path_delay_estimator.cc
@@ -12,8 +12,8 @@
 #include <array>
 #include <cstddef>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/block.h"
@@ -67,7 +67,7 @@ std::optional<DelayEstimate> EchoPathDelayEstimator::EstimateDelay(
     const DownsampledRenderBuffer& render_buffer,
     const Block& capture) {
   std::array<float, kBlockSize> downsampled_capture_data;
-  ArrayView<float> downsampled_capture(downsampled_capture_data.data(),
+  std::span<float> downsampled_capture(downsampled_capture_data.data(),
                                        sub_block_size_);
 
   std::array<float, kBlockSize> downmixed_capture;
diff --git a/modules/audio_processing/aec3/echo_remover.cc b/modules/audio_processing/aec3/echo_remover.cc
index b6f3a7b24f..0d3e1d1c7a 100644
--- a/modules/audio_processing/aec3/echo_remover.cc
+++ b/modules/audio_processing/aec3/echo_remover.cc
@@ -16,9 +16,9 @@
 #include <cstddef>
 #include <memory>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "api/audio/echo_control.h"
 #include "api/audio/neural_residual_echo_estimator.h"
@@ -74,9 +74,9 @@ void LinearEchoPower(const FftData& E,
 }
 
 // Fades between two input signals using a fix-sized transition.
-void SignalTransition(ArrayView<const float> from,
-                      ArrayView<const float> to,
-                      ArrayView<float> out) {
+void SignalTransition(std::span<const float> from,
+                      std::span<const float> to,
+                      std::span<float> out) {
   RTC_DCHECK_EQ(from.size(), to.size());
   RTC_DCHECK_EQ(from.size(), out.size());
   if (from.data() == to.data()) {
@@ -100,8 +100,8 @@ void SignalTransition(ArrayView<const float> from,
 // Computes a windowed (square root Hanning) padded FFT and updates the related
 // memory.
 void WindowedPaddedFft(const Aec3Fft& fft,
-                       ArrayView<const float> v,
-                       ArrayView<float> v_old,
+                       std::span<const float> v,
+                       std::span<float> v_old,
                        FftData* V) {
   fft.PaddedFft(v, v_old, Aec3Fft::Window::kSqrtHanning, V);
   std::copy(v.begin(), v.end(), v_old.begin());
@@ -134,7 +134,7 @@ bool UseRefinedOutput(const SubtractorOutput& subtractor_output) {
 void FormLinearFilterOutput(bool refined_filter_output_last_selected,
                             bool use_refined_output,
                             const SubtractorOutput& subtractor_output,
-                            ArrayView<float> output) {
+                            std::span<float> output) {
   RTC_DCHECK_EQ(subtractor_output.e_refined.size(), output.size());
   RTC_DCHECK_EQ(subtractor_output.e_coarse.size(), output.size());
 
@@ -318,48 +318,48 @@ void EchoRemoverImpl::ProcessCapture(
   std::array<FftData, kMaxNumChannelsOnStack> high_band_comfort_noise_stack;
   std::array<SubtractorOutput, kMaxNumChannelsOnStack> subtractor_output_stack;
 
-  ArrayView<std::array<float, kFftLengthBy2>> e(e_stack.data(),
+  std::span<std::array<float, kFftLengthBy2>> e(e_stack.data(),
                                                 num_capture_channels_);
-  ArrayView<std::array<float, kFftLengthBy2Plus1>> Y2(Y2_stack.data(),
+  std::span<std::array<float, kFftLengthBy2Plus1>> Y2(Y2_stack.data(),
                                                       num_capture_channels_);
-  ArrayView<std::array<float, kFftLengthBy2Plus1>> E2(E2_stack.data(),
+  std::span<std::array<float, kFftLengthBy2Plus1>> E2(E2_stack.data(),
                                                       num_capture_channels_);
-  ArrayView<std::array<float, kFftLengthBy2Plus1>> R2(R2_stack.data(),
+  std::span<std::array<float, kFftLengthBy2Plus1>> R2(R2_stack.data(),
                                                       num_capture_channels_);
-  ArrayView<std::array<float, kFftLengthBy2Plus1>> R2_unbounded(
+  std::span<std::array<float, kFftLengthBy2Plus1>> R2_unbounded(
       R2_unbounded_stack.data(), num_capture_channels_);
-  ArrayView<std::array<float, kFftLengthBy2Plus1>> S2_linear(
+  std::span<std::array<float, kFftLengthBy2Plus1>> S2_linear(
       S2_linear_stack.data(), num_capture_channels_);
-  ArrayView<FftData> Y(Y_stack.data(), num_capture_channels_);
-  ArrayView<FftData> E(E_stack.data(), num_capture_channels_);
-  ArrayView<FftData> comfort_noise(comfort_noise_stack.data(),
+  std::span<FftData> Y(Y_stack.data(), num_capture_channels_);
+  std::span<FftData> E(E_stack.data(), num_capture_channels_);
+  std::span<FftData> comfort_noise(comfort_noise_stack.data(),
                                    num_capture_channels_);
-  ArrayView<FftData> high_band_comfort_noise(
+  std::span<FftData> high_band_comfort_noise(
       high_band_comfort_noise_stack.data(), num_capture_channels_);
-  ArrayView<SubtractorOutput> subtractor_output(subtractor_output_stack.data(),
+  std::span<SubtractorOutput> subtractor_output(subtractor_output_stack.data(),
                                                 num_capture_channels_);
   if (NumChannelsOnHeap(num_capture_channels_) > 0) {
     // If the stack-allocated space is too small, use the heap for storing the
     // microphone data.
-    e = ArrayView<std::array<float, kFftLengthBy2>>(e_heap_.data(),
+    e = std::span<std::array<float, kFftLengthBy2>>(e_heap_.data(),
                                                     num_capture_channels_);
-    Y2 = ArrayView<std::array<float, kFftLengthBy2Plus1>>(
+    Y2 = std::span<std::array<float, kFftLengthBy2Plus1>>(
         Y2_heap_.data(), num_capture_channels_);
-    E2 = ArrayView<std::array<float, kFftLengthBy2Plus1>>(
+    E2 = std::span<std::array<float, kFftLengthBy2Plus1>>(
         E2_heap_.data(), num_capture_channels_);
-    R2 = ArrayView<std::array<float, kFftLengthBy2Plus1>>(
+    R2 = std::span<std::array<float, kFftLengthBy2Plus1>>(
         R2_heap_.data(), num_capture_channels_);
-    R2_unbounded = ArrayView<std::array<float, kFftLengthBy2Plus1>>(
+    R2_unbounded = std::span<std::array<float, kFftLengthBy2Plus1>>(
         R2_unbounded_heap_.data(), num_capture_channels_);
-    S2_linear = ArrayView<std::array<float, kFftLengthBy2Plus1>>(
+    S2_linear = std::span<std::array<float, kFftLengthBy2Plus1>>(
         S2_linear_heap_.data(), num_capture_channels_);
-    Y = ArrayView<FftData>(Y_heap_.data(), num_capture_channels_);
-    E = ArrayView<FftData>(E_heap_.data(), num_capture_channels_);
+    Y = std::span<FftData>(Y_heap_.data(), num_capture_channels_);
+    E = std::span<FftData>(E_heap_.data(), num_capture_channels_);
     comfort_noise =
-        ArrayView<FftData>(comfort_noise_heap_.data(), num_capture_channels_);
-    high_band_comfort_noise = ArrayView<FftData>(
+        std::span<FftData>(comfort_noise_heap_.data(), num_capture_channels_);
+    high_band_comfort_noise = std::span<FftData>(
         high_band_comfort_noise_heap_.data(), num_capture_channels_);
-    subtractor_output = ArrayView<SubtractorOutput>(
+    subtractor_output = std::span<SubtractorOutput>(
         subtractor_output_heap_.data(), num_capture_channels_);
   }
 
diff --git a/modules/audio_processing/aec3/erl_estimator.cc b/modules/audio_processing/aec3/erl_estimator.cc
index 2bf6ee9e72..d5894cf0a6 100644
--- a/modules/audio_processing/aec3/erl_estimator.cc
+++ b/modules/audio_processing/aec3/erl_estimator.cc
@@ -15,9 +15,9 @@
 #include <cstddef>
 #include <iterator>
 #include <numeric>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "rtc_base/checks.h"
 
@@ -46,8 +46,8 @@ void ErlEstimator::Reset() {
 
 void ErlEstimator::Update(
     const std::vector<bool>& converged_filters,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> render_spectra,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> capture_spectra) {
+    std::span<const std::array<float, kFftLengthBy2Plus1>> render_spectra,
+    std::span<const std::array<float, kFftLengthBy2Plus1>> capture_spectra) {
   const size_t num_capture_channels = converged_filters.size();
   RTC_DCHECK_EQ(capture_spectra.size(), num_capture_channels);
 
@@ -90,7 +90,7 @@ void ErlEstimator::Update(
 
   const size_t num_render_channels = render_spectra.size();
   std::array<float, kFftLengthBy2Plus1> max_render_spectrum_data;
-  ArrayView<const float, kFftLengthBy2Plus1> max_render_spectrum =
+  std::span<const float, kFftLengthBy2Plus1> max_render_spectrum =
       render_spectra[/*channel=*/0];
   if (num_render_channels > 1) {
     std::copy(render_spectra[0].begin(), render_spectra[0].end(),
diff --git a/modules/audio_processing/aec3/erl_estimator.h b/modules/audio_processing/aec3/erl_estimator.h
index b793ddec78..2b2ab4342e 100644
--- a/modules/audio_processing/aec3/erl_estimator.h
+++ b/modules/audio_processing/aec3/erl_estimator.h
@@ -14,9 +14,9 @@
 #include <stddef.h>
 
 #include <array>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 
 namespace webrtc {
@@ -36,8 +36,8 @@ class ErlEstimator {
   // Updates the ERL estimate.
   void Update(
       const std::vector<bool>& converged_filters,
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>> render_spectra,
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>> capture_spectra);
+      std::span<const std::array<float, kFftLengthBy2Plus1>> render_spectra,
+      std::span<const std::array<float, kFftLengthBy2Plus1>> capture_spectra);
 
   // Returns the most recent ERL estimate.
   const std::array<float, kFftLengthBy2Plus1>& Erl() const { return erl_; }
diff --git a/modules/audio_processing/aec3/erle_estimator.cc b/modules/audio_processing/aec3/erle_estimator.cc
index 15de377594..206103e42b 100644
--- a/modules/audio_processing/aec3/erle_estimator.cc
+++ b/modules/audio_processing/aec3/erle_estimator.cc
@@ -13,9 +13,9 @@
 #include <array>
 #include <cstddef>
 #include <memory>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "api/environment/environment.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
@@ -56,11 +56,11 @@ void ErleEstimator::Reset(bool delay_change) {
 
 void ErleEstimator::Update(
     const RenderBuffer& render_buffer,
-    ArrayView<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
+    std::span<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
         filter_frequency_responses,
-    ArrayView<const float, kFftLengthBy2Plus1> avg_render_spectrum_with_reverb,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> capture_spectra,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> subtractor_spectra,
+    std::span<const float, kFftLengthBy2Plus1> avg_render_spectrum_with_reverb,
+    std::span<const std::array<float, kFftLengthBy2Plus1>> capture_spectra,
+    std::span<const std::array<float, kFftLengthBy2Plus1>> subtractor_spectra,
     const std::vector<bool>& converged_filters) {
   RTC_DCHECK_EQ(subband_erle_estimator_.Erle(/*onset_compensated=*/true).size(),
                 capture_spectra.size());
diff --git a/modules/audio_processing/aec3/erle_estimator.h b/modules/audio_processing/aec3/erle_estimator.h
index fe6e71a355..fb275107ff 100644
--- a/modules/audio_processing/aec3/erle_estimator.h
+++ b/modules/audio_processing/aec3/erle_estimator.h
@@ -16,9 +16,9 @@
 #include <array>
 #include <memory>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "api/environment/environment.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
@@ -46,16 +46,16 @@ class ErleEstimator {
   // Updates the ERLE estimates.
   void Update(
       const RenderBuffer& render_buffer,
-      ArrayView<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
+      std::span<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
           filter_frequency_responses,
-      ArrayView<const float, kFftLengthBy2Plus1>
+      std::span<const float, kFftLengthBy2Plus1>
           avg_render_spectrum_with_reverb,
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>> capture_spectra,
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>> subtractor_spectra,
+      std::span<const std::array<float, kFftLengthBy2Plus1>> capture_spectra,
+      std::span<const std::array<float, kFftLengthBy2Plus1>> subtractor_spectra,
       const std::vector<bool>& converged_filters);
 
   // Returns the most recent subband ERLE estimates.
-  ArrayView<const std::array<float, kFftLengthBy2Plus1>> Erle(
+  std::span<const std::array<float, kFftLengthBy2Plus1>> Erle(
       bool onset_compensated) const {
     return signal_dependent_erle_estimator_
                ? signal_dependent_erle_estimator_->Erle(onset_compensated)
@@ -63,7 +63,7 @@ class ErleEstimator {
   }
 
   // Returns the non-capped subband ERLE.
-  ArrayView<const std::array<float, kFftLengthBy2Plus1>> ErleUnbounded() const {
+  std::span<const std::array<float, kFftLengthBy2Plus1>> ErleUnbounded() const {
     // Unbounded ERLE is only used with the subband erle estimator where the
     // ERLE is often capped at low values. When the signal dependent ERLE
     // estimator is used the capped ERLE is returned.
@@ -75,7 +75,7 @@ class ErleEstimator {
 
   // Returns the subband ERLE that are estimated during onsets (only used for
   // testing).
-  ArrayView<const std::array<float, kFftLengthBy2Plus1>> ErleDuringOnsets()
+  std::span<const std::array<float, kFftLengthBy2Plus1>> ErleDuringOnsets()
       const {
     return subband_erle_estimator_.ErleDuringOnsets();
   }
@@ -90,7 +90,7 @@ class ErleEstimator {
   // vector with content between 0 and 1 where 1 indicates that, at this current
   // time instant, the linear filter is reaching its maximum subtraction
   // performance.
-  ArrayView<const std::optional<float>> GetInstLinearQualityEstimates() const {
+  std::span<const std::optional<float>> GetInstLinearQualityEstimates() const {
     return fullband_erle_estimator_.GetInstLinearQualityEstimates();
   }
 
diff --git a/modules/audio_processing/aec3/erle_estimator_unittest.cc b/modules/audio_processing/aec3/erle_estimator_unittest.cc
index ce75cfcafe..a6497386e9 100644
--- a/modules/audio_processing/aec3/erle_estimator_unittest.cc
+++ b/modules/audio_processing/aec3/erle_estimator_unittest.cc
@@ -15,10 +15,10 @@
 #include <cmath>
 #include <cstddef>
 #include <memory>
+#include <span>
 #include <tuple>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "api/environment/environment_factory.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
@@ -38,7 +38,7 @@ constexpr float kTrueErleOnsets = 1.0f;
 constexpr float kEchoPathGain = 3.f;
 
 void VerifyErleBands(
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> erle,
+    std::span<const std::array<float, kFftLengthBy2Plus1>> erle,
     float reference_lf,
     float reference_hf) {
   for (size_t ch = 0; ch < erle.size(); ++ch) {
@@ -51,7 +51,7 @@ void VerifyErleBands(
   }
 }
 
-void VerifyErle(ArrayView<const std::array<float, kFftLengthBy2Plus1>> erle,
+void VerifyErle(std::span<const std::array<float, kFftLengthBy2Plus1>> erle,
                 float erle_time_domain,
                 float reference_lf,
                 float reference_hf) {
@@ -60,8 +60,8 @@ void VerifyErle(ArrayView<const std::array<float, kFftLengthBy2Plus1>> erle,
 }
 
 void VerifyErleGreaterOrEqual(
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> erle1,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> erle2) {
+    std::span<const std::array<float, kFftLengthBy2Plus1>> erle1,
+    std::span<const std::array<float, kFftLengthBy2Plus1>> erle2) {
   for (size_t ch = 0; ch < erle1.size(); ++ch) {
     for (size_t i = 0; i < kFftLengthBy2Plus1; ++i) {
       EXPECT_GE(erle1[ch][i], erle2[ch][i]);
@@ -90,8 +90,8 @@ void FormFarendTimeFrame(Block* x) {
 void FormFarendFrame(const RenderBuffer& render_buffer,
                      float erle,
                      std::array<float, kFftLengthBy2Plus1>* X2,
-                     ArrayView<std::array<float, kFftLengthBy2Plus1>> E2,
-                     ArrayView<std::array<float, kFftLengthBy2Plus1>> Y2) {
+                     std::span<std::array<float, kFftLengthBy2Plus1>> E2,
+                     std::span<std::array<float, kFftLengthBy2Plus1>> Y2) {
   const auto& spectrum_buffer = render_buffer.GetSpectrumBuffer();
   const int num_render_channels = spectrum_buffer.buffer[0].size();
   const int num_capture_channels = Y2.size();
@@ -114,8 +114,8 @@ void FormFarendFrame(const RenderBuffer& render_buffer,
 
 void FormNearendFrame(Block* x,
                       std::array<float, kFftLengthBy2Plus1>* X2,
-                      ArrayView<std::array<float, kFftLengthBy2Plus1>> E2,
-                      ArrayView<std::array<float, kFftLengthBy2Plus1>> Y2) {
+                      std::span<std::array<float, kFftLengthBy2Plus1>> E2,
+                      std::span<std::array<float, kFftLengthBy2Plus1>> Y2) {
   for (int band = 0; band < x->NumBands(); ++band) {
     for (int ch = 0; ch < x->NumChannels(); ++ch) {
       std::fill(x->begin(band, ch), x->end(band, ch), 0.f);
@@ -130,7 +130,7 @@ void FormNearendFrame(Block* x,
 }
 
 void GetFilterFreq(size_t delay_headroom_samples,
-                   ArrayView<std::vector<std::array<float, kFftLengthBy2Plus1>>>
+                   std::span<std::vector<std::array<float, kFftLengthBy2Plus1>>>
                        filter_frequency_response) {
   const size_t delay_headroom_blocks = delay_headroom_samples / kBlockSize;
   for (size_t ch = 0; ch < filter_frequency_response[0].size(); ++ch) {
diff --git a/modules/audio_processing/aec3/fft_data.h b/modules/audio_processing/aec3/fft_data.h
index b227496388..6bab8c9c1a 100644
--- a/modules/audio_processing/aec3/fft_data.h
+++ b/modules/audio_processing/aec3/fft_data.h
@@ -14,8 +14,8 @@
 #include <algorithm>
 #include <array>
 #include <cstddef>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "rtc_base/checks.h"
 
@@ -43,11 +43,11 @@ struct FftData {
   }
 
   // Computes the power spectrum of the data.
-  void SpectrumAVX2(ArrayView<float> power_spectrum) const;
+  void SpectrumAVX2(std::span<float> power_spectrum) const;
 
   // Computes the power spectrum of the data.
   void Spectrum(Aec3Optimization optimization,
-                ArrayView<float> power_spectrum) const {
+                std::span<float> power_spectrum) const {
     RTC_DCHECK_EQ(kFftLengthBy2Plus1, power_spectrum.size());
     switch (optimization) {
 #if defined(WEBRTC_ARCH_X86_FAMILY)
diff --git a/modules/audio_processing/aec3/fft_data_avx2.cc b/modules/audio_processing/aec3/fft_data_avx2.cc
index eaa93c0db1..97c34dfc31 100644
--- a/modules/audio_processing/aec3/fft_data_avx2.cc
+++ b/modules/audio_processing/aec3/fft_data_avx2.cc
@@ -11,8 +11,8 @@
 #include <immintrin.h>
 
 #include <cstddef>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/fft_data.h"
 #include "rtc_base/checks.h"
@@ -20,7 +20,7 @@
 namespace webrtc {
 
 // Computes the power spectrum of the data.
-void FftData::SpectrumAVX2(ArrayView<float> power_spectrum) const {
+void FftData::SpectrumAVX2(std::span<float> power_spectrum) const {
   RTC_DCHECK_EQ(kFftLengthBy2Plus1, power_spectrum.size());
   for (size_t k = 0; k < kFftLengthBy2; k += 8) {
     __m256 r = _mm256_loadu_ps(&re[k]);
diff --git a/modules/audio_processing/aec3/filter_analyzer.cc b/modules/audio_processing/aec3/filter_analyzer.cc
index cf64737109..e9445cf664 100644
--- a/modules/audio_processing/aec3/filter_analyzer.cc
+++ b/modules/audio_processing/aec3/filter_analyzer.cc
@@ -16,9 +16,9 @@
 #include <cmath>
 #include <cstddef>
 #include <numeric>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/block.h"
@@ -29,7 +29,7 @@
 namespace webrtc {
 namespace {
 
-size_t FindPeakIndex(ArrayView<const float> filter_time_domain,
+size_t FindPeakIndex(std::span<const float> filter_time_domain,
                      size_t peak_index_in,
                      size_t start_sample,
                      size_t end_sample) {
@@ -78,7 +78,7 @@ void FilterAnalyzer::Reset() {
 }
 
 void FilterAnalyzer::Update(
-    ArrayView<const std::vector<float>> filters_time_domain,
+    std::span<const std::vector<float>> filters_time_domain,
     const RenderBuffer& render_buffer,
     bool* any_filter_consistent,
     float* max_echo_path_gain) {
@@ -107,7 +107,7 @@ void FilterAnalyzer::Update(
 }
 
 void FilterAnalyzer::AnalyzeRegion(
-    ArrayView<const std::vector<float>> filters_time_domain,
+    std::span<const std::vector<float>> filters_time_domain,
     const RenderBuffer& render_buffer) {
   // Preprocess the filter to avoid issues with low-frequency components in the
   // filter.
@@ -139,7 +139,7 @@ void FilterAnalyzer::AnalyzeRegion(
   }
 }
 
-void FilterAnalyzer::UpdateFilterGain(ArrayView<const float> filter_time_domain,
+void FilterAnalyzer::UpdateFilterGain(std::span<const float> filter_time_domain,
                                       FilterAnalysisState* st) {
   bool sufficient_time_to_converge =
       blocks_since_reset_ > 5 * kNumBlocksPerSecond;
@@ -159,7 +159,7 @@ void FilterAnalyzer::UpdateFilterGain(ArrayView<const float> filter_time_domain,
 }
 
 void FilterAnalyzer::PreProcessFilters(
-    ArrayView<const std::vector<float>> filters_time_domain) {
+    std::span<const std::vector<float>> filters_time_domain) {
   for (size_t ch = 0; ch < filters_time_domain.size(); ++ch) {
     RTC_DCHECK_LT(region_.start_sample_, filters_time_domain[ch].size());
     RTC_DCHECK_LT(region_.end_sample_, filters_time_domain[ch].size());
@@ -224,7 +224,7 @@ void FilterAnalyzer::ConsistentFilterDetector::Reset() {
 }
 
 bool FilterAnalyzer::ConsistentFilterDetector::Detect(
-    ArrayView<const float> filter_to_analyze,
+    std::span<const float> filter_to_analyze,
     const FilterRegion& region,
     const Block& x_block,
     size_t peak_index,
@@ -268,7 +268,7 @@ bool FilterAnalyzer::ConsistentFilterDetector::Detect(
   if (significant_peak_) {
     bool active_render_block = false;
     for (int ch = 0; ch < x_block.NumChannels(); ++ch) {
-      ArrayView<const float, kBlockSize> x_channel =
+      std::span<const float, kBlockSize> x_channel =
           x_block.View(/*band=*/0, ch);
       const float x_energy = std::inner_product(
           x_channel.begin(), x_channel.end(), x_channel.begin(), 0.f);
diff --git a/modules/audio_processing/aec3/filter_analyzer.h b/modules/audio_processing/aec3/filter_analyzer.h
index 83c48650c5..28c7912f9d 100644
--- a/modules/audio_processing/aec3/filter_analyzer.h
+++ b/modules/audio_processing/aec3/filter_analyzer.h
@@ -15,9 +15,9 @@
 
 #include <atomic>
 #include <memory>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "modules/audio_processing/aec3/block.h"
 
@@ -40,13 +40,13 @@ class FilterAnalyzer {
   void Reset();
 
   // Updates the estimates with new input data.
-  void Update(ArrayView<const std::vector<float>> filters_time_domain,
+  void Update(std::span<const std::vector<float>> filters_time_domain,
               const RenderBuffer& render_buffer,
               bool* any_filter_consistent,
               float* max_echo_path_gain);
 
   // Returns the delay in blocks for each filter.
-  ArrayView<const int> FilterDelaysBlocks() const {
+  std::span<const int> FilterDelaysBlocks() const {
     return filter_delays_blocks_;
   }
 
@@ -59,7 +59,7 @@ class FilterAnalyzer {
   }
 
   // Returns the preprocessed filter.
-  ArrayView<const std::vector<float>> GetAdjustedFilters() const {
+  std::span<const std::vector<float>> GetAdjustedFilters() const {
     return h_highpass_;
   }
 
@@ -69,13 +69,13 @@ class FilterAnalyzer {
  private:
   struct FilterAnalysisState;
 
-  void AnalyzeRegion(ArrayView<const std::vector<float>> filters_time_domain,
+  void AnalyzeRegion(std::span<const std::vector<float>> filters_time_domain,
                      const RenderBuffer& render_buffer);
 
-  void UpdateFilterGain(ArrayView<const float> filters_time_domain,
+  void UpdateFilterGain(std::span<const float> filters_time_domain,
                         FilterAnalysisState* st);
   void PreProcessFilters(
-      ArrayView<const std::vector<float>> filters_time_domain);
+      std::span<const std::vector<float>> filters_time_domain);
 
   void ResetRegion();
 
@@ -90,7 +90,7 @@ class FilterAnalyzer {
    public:
     explicit ConsistentFilterDetector(const EchoCanceller3Config& config);
     void Reset();
-    bool Detect(ArrayView<const float> filter_to_analyze,
+    bool Detect(std::span<const float> filter_to_analyze,
                 const FilterRegion& region,
                 const Block& x_block,
                 size_t peak_index,
diff --git a/modules/audio_processing/aec3/frame_blocker.cc b/modules/audio_processing/aec3/frame_blocker.cc
index 97246e14ca..2b93e7a65b 100644
--- a/modules/audio_processing/aec3/frame_blocker.cc
+++ b/modules/audio_processing/aec3/frame_blocker.cc
@@ -12,9 +12,9 @@
 
 #include <algorithm>
 #include <cstddef>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/block.h"
 #include "rtc_base/checks.h"
@@ -38,7 +38,7 @@ FrameBlocker::FrameBlocker(size_t num_bands, size_t num_channels)
 FrameBlocker::~FrameBlocker() = default;
 
 void FrameBlocker::InsertSubFrameAndExtractBlock(
-    const std::vector<std::vector<ArrayView<float>>>& sub_frame,
+    const std::vector<std::vector<std::span<float>>>& sub_frame,
     Block* block) {
   RTC_DCHECK(block);
   RTC_DCHECK_EQ(num_bands_, block->NumBands());
diff --git a/modules/audio_processing/aec3/frame_blocker.h b/modules/audio_processing/aec3/frame_blocker.h
index f9dd88b76a..f035fed33f 100644
--- a/modules/audio_processing/aec3/frame_blocker.h
+++ b/modules/audio_processing/aec3/frame_blocker.h
@@ -13,9 +13,9 @@
 
 #include <stddef.h>
 
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/block.h"
 
 namespace webrtc {
@@ -32,7 +32,7 @@ class FrameBlocker {
   // Inserts one 80 sample multiband subframe from the multiband frame and
   // extracts one 64 sample multiband block.
   void InsertSubFrameAndExtractBlock(
-      const std::vector<std::vector<ArrayView<float>>>& sub_frame,
+      const std::vector<std::vector<std::span<float>>>& sub_frame,
       Block* block);
   // Reports whether a multiband block of 64 samples is available for
   // extraction.
diff --git a/modules/audio_processing/aec3/frame_blocker_unittest.cc b/modules/audio_processing/aec3/frame_blocker_unittest.cc
index d32cd80b29..eadfe71dfc 100644
--- a/modules/audio_processing/aec3/frame_blocker_unittest.cc
+++ b/modules/audio_processing/aec3/frame_blocker_unittest.cc
@@ -11,10 +11,10 @@
 #include "modules/audio_processing/aec3/frame_blocker.h"
 
 #include <cstddef>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/block.h"
 #include "modules/audio_processing/aec3/block_framer.h"
@@ -55,12 +55,12 @@ void FillSubFrameView(
     size_t sub_frame_counter,
     int offset,
     std::vector<std::vector<std::vector<float>>>* sub_frame,
-    std::vector<std::vector<ArrayView<float>>>* sub_frame_view) {
+    std::vector<std::vector<std::span<float>>>* sub_frame_view) {
   FillSubFrame(sub_frame_counter, offset, sub_frame);
   for (size_t band = 0; band < sub_frame_view->size(); ++band) {
     for (size_t channel = 0; channel < (*sub_frame_view)[band].size();
          ++channel) {
-      (*sub_frame_view)[band][channel] = ArrayView<float>(
+      (*sub_frame_view)[band][channel] = std::span<float>(
           &(*sub_frame)[band][channel][0], (*sub_frame)[band][channel].size());
     }
   }
@@ -69,7 +69,7 @@ void FillSubFrameView(
 bool VerifySubFrame(
     size_t sub_frame_counter,
     int offset,
-    const std::vector<std::vector<ArrayView<float>>>& sub_frame_view) {
+    const std::vector<std::vector<std::span<float>>>& sub_frame_view) {
   std::vector<std::vector<std::vector<float>>> reference_sub_frame(
       sub_frame_view.size(),
       std::vector<std::vector<float>>(
@@ -115,8 +115,8 @@ void RunBlockerTest(int sample_rate_hz, size_t num_channels) {
   std::vector<std::vector<std::vector<float>>> input_sub_frame(
       num_bands, std::vector<std::vector<float>>(
                      num_channels, std::vector<float>(kSubFrameLength, 0.f)));
-  std::vector<std::vector<ArrayView<float>>> input_sub_frame_view(
-      num_bands, std::vector<ArrayView<float>>(num_channels));
+  std::vector<std::vector<std::span<float>>> input_sub_frame_view(
+      num_bands, std::vector<std::span<float>>(num_channels));
   FrameBlocker blocker(num_bands, num_channels);
 
   size_t block_counter = 0;
@@ -153,10 +153,10 @@ void RunBlockerAndFramerTest(int sample_rate_hz, size_t num_channels) {
   std::vector<std::vector<std::vector<float>>> output_sub_frame(
       num_bands, std::vector<std::vector<float>>(
                      num_channels, std::vector<float>(kSubFrameLength, 0.f)));
-  std::vector<std::vector<ArrayView<float>>> output_sub_frame_view(
-      num_bands, std::vector<ArrayView<float>>(num_channels));
-  std::vector<std::vector<ArrayView<float>>> input_sub_frame_view(
-      num_bands, std::vector<ArrayView<float>>(num_channels));
+  std::vector<std::vector<std::span<float>>> output_sub_frame_view(
+      num_bands, std::vector<std::span<float>>(num_channels));
+  std::vector<std::vector<std::span<float>>> input_sub_frame_view(
+      num_bands, std::vector<std::span<float>>(num_channels));
   FrameBlocker blocker(num_bands, num_channels);
   BlockFramer framer(num_bands, num_channels);
 
@@ -203,9 +203,9 @@ void RunWronglySizedInsertAndExtractParametersTest(
       num_sub_frame_bands,
       std::vector<std::vector<float>>(
           num_sub_frame_channels, std::vector<float>(sub_frame_length, 0.f)));
-  std::vector<std::vector<ArrayView<float>>> input_sub_frame_view(
+  std::vector<std::vector<std::span<float>>> input_sub_frame_view(
       input_sub_frame.size(),
-      std::vector<ArrayView<float>>(num_sub_frame_channels));
+      std::vector<std::span<float>>(num_sub_frame_channels));
   FillSubFrameView(0, 0, &input_sub_frame, &input_sub_frame_view);
   FrameBlocker blocker(correct_num_bands, correct_num_channels);
   EXPECT_DEATH(
@@ -226,9 +226,9 @@ void RunWronglySizedExtractParameterTest(int sample_rate_hz,
       correct_num_bands,
       std::vector<std::vector<float>>(
           correct_num_channels, std::vector<float>(kSubFrameLength, 0.f)));
-  std::vector<std::vector<ArrayView<float>>> input_sub_frame_view(
+  std::vector<std::vector<std::span<float>>> input_sub_frame_view(
       input_sub_frame.size(),
-      std::vector<ArrayView<float>>(correct_num_channels));
+      std::vector<std::span<float>>(correct_num_channels));
   FillSubFrameView(0, 0, &input_sub_frame, &input_sub_frame_view);
   FrameBlocker blocker(correct_num_bands, correct_num_channels);
   blocker.InsertSubFrameAndExtractBlock(input_sub_frame_view, &correct_block);
@@ -251,8 +251,8 @@ void RunWrongExtractOrderTest(int sample_rate_hz,
   std::vector<std::vector<std::vector<float>>> input_sub_frame(
       num_bands, std::vector<std::vector<float>>(
                      num_channels, std::vector<float>(kSubFrameLength, 0.f)));
-  std::vector<std::vector<ArrayView<float>>> input_sub_frame_view(
-      input_sub_frame.size(), std::vector<ArrayView<float>>(num_channels));
+  std::vector<std::vector<std::span<float>>> input_sub_frame_view(
+      input_sub_frame.size(), std::vector<std::span<float>>(num_channels));
   FillSubFrameView(0, 0, &input_sub_frame, &input_sub_frame_view);
   FrameBlocker blocker(num_bands, num_channels);
   for (size_t k = 0; k < num_preceeding_api_calls; ++k) {
@@ -398,7 +398,7 @@ TEST(FrameBlockerDeathTest, NullBlockParameter) {
   std::vector<std::vector<std::vector<float>>> sub_frame(
       1, std::vector<std::vector<float>>(
              1, std::vector<float>(kSubFrameLength, 0.f)));
-  std::vector<std::vector<ArrayView<float>>> sub_frame_view(sub_frame.size());
+  std::vector<std::vector<std::span<float>>> sub_frame_view(sub_frame.size());
   FillSubFrameView(0, 0, &sub_frame, &sub_frame_view);
   EXPECT_DEATH(
       FrameBlocker(1, 1).InsertSubFrameAndExtractBlock(sub_frame_view, nullptr),
diff --git a/modules/audio_processing/aec3/fullband_erle_estimator.cc b/modules/audio_processing/aec3/fullband_erle_estimator.cc
index 07acfa98a3..e31ec29331 100644
--- a/modules/audio_processing/aec3/fullband_erle_estimator.cc
+++ b/modules/audio_processing/aec3/fullband_erle_estimator.cc
@@ -16,9 +16,9 @@
 #include <memory>
 #include <numeric>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/logging/apm_data_dumper.h"
@@ -60,9 +60,9 @@ void FullBandErleEstimator::Reset() {
 }
 
 void FullBandErleEstimator::Update(
-    ArrayView<const float> X2,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> Y2,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> E2,
+    std::span<const float> X2,
+    std::span<const std::array<float, kFftLengthBy2Plus1>> Y2,
+    std::span<const std::array<float, kFftLengthBy2Plus1>> E2,
     const std::vector<bool>& converged_filters) {
   for (size_t ch = 0; ch < Y2.size(); ++ch) {
     if (converged_filters[ch]) {
diff --git a/modules/audio_processing/aec3/fullband_erle_estimator.h b/modules/audio_processing/aec3/fullband_erle_estimator.h
index bb710c96d0..17af958750 100644
--- a/modules/audio_processing/aec3/fullband_erle_estimator.h
+++ b/modules/audio_processing/aec3/fullband_erle_estimator.h
@@ -16,9 +16,9 @@
 #include <cstddef>
 #include <memory>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/logging/apm_data_dumper.h"
@@ -36,9 +36,9 @@ class FullBandErleEstimator {
   void Reset();
 
   // Updates the ERLE estimator.
-  void Update(ArrayView<const float> X2,
-              ArrayView<const std::array<float, kFftLengthBy2Plus1>> Y2,
-              ArrayView<const std::array<float, kFftLengthBy2Plus1>> E2,
+  void Update(std::span<const float> X2,
+              std::span<const std::array<float, kFftLengthBy2Plus1>> Y2,
+              std::span<const std::array<float, kFftLengthBy2Plus1>> E2,
               const std::vector<bool>& converged_filters);
 
   // Returns the fullband ERLE estimates in log2 units.
@@ -52,7 +52,7 @@ class FullBandErleEstimator {
 
   // Returns an estimation of the current linear filter quality. It returns a
   // float number between 0 and 1 mapping 1 to the highest possible quality.
-  ArrayView<const std::optional<float>> GetInstLinearQualityEstimates() const {
+  std::span<const std::optional<float>> GetInstLinearQualityEstimates() const {
     return linear_filters_qualities_;
   }
 
diff --git a/modules/audio_processing/aec3/matched_filter.cc b/modules/audio_processing/aec3/matched_filter.cc
index 5c86e27311..961d8a88db 100644
--- a/modules/audio_processing/aec3/matched_filter.cc
+++ b/modules/audio_processing/aec3/matched_filter.cc
@@ -25,8 +25,8 @@
 #include <cstddef>
 #include <initializer_list>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/downsampled_render_buffer.h"
 #include "modules/audio_processing/logging/apm_data_dumper.h"
@@ -43,8 +43,8 @@ namespace {
 constexpr int kAccumulatedErrorSubSampleRate = 4;
 
 void UpdateAccumulatedError(
-    const ArrayView<const float> instantaneous_accumulated_error,
-    const ArrayView<float> accumulated_error,
+    const std::span<const float> instantaneous_accumulated_error,
+    const std::span<float> accumulated_error,
     float one_over_error_sum_anchor) {
   static constexpr float kSmoothConstantIncreases = 0.015f;
   for (size_t k = 0; k < instantaneous_accumulated_error.size(); ++k) {
@@ -59,7 +59,7 @@ void UpdateAccumulatedError(
   }
 }
 
-size_t ComputePreEchoLag(const ArrayView<const float> accumulated_error,
+size_t ComputePreEchoLag(const std::span<const float> accumulated_error,
                          size_t lag,
                          size_t alignment_shift_winner) {
   static constexpr float kPreEchoThreshold = 0.5f;
@@ -93,13 +93,13 @@ void MatchedFilterCoreWithAccumulatedError_NEON(
     size_t x_start_index,
     float x2_sum_threshold,
     float smoothing,
-    ArrayView<const float> x,
-    ArrayView<const float> y,
-    ArrayView<float> h,
+    std::span<const float> x,
+    std::span<const float> y,
+    std::span<float> h,
     bool* filters_updated,
     float* error_sum,
-    ArrayView<float> accumulated_error,
-    ArrayView<float> scratch_memory) {
+    std::span<float> accumulated_error,
+    std::span<float> scratch_memory) {
   const int h_size = static_cast<int>(h.size());
   const int x_size = static_cast<int>(x.size());
   RTC_DCHECK_EQ(0, h_size % 4);
@@ -173,14 +173,14 @@ void MatchedFilterCoreWithAccumulatedError_NEON(
 void MatchedFilterCore_NEON(size_t x_start_index,
                             float x2_sum_threshold,
                             float smoothing,
-                            ArrayView<const float> x,
-                            ArrayView<const float> y,
-                            ArrayView<float> h,
+                            std::span<const float> x,
+                            std::span<const float> y,
+                            std::span<float> h,
                             bool* filters_updated,
                             float* error_sum,
                             bool compute_accumulated_error,
-                            ArrayView<float> accumulated_error,
-                            ArrayView<float> scratch_memory) {
+                            std::span<float> accumulated_error,
+                            std::span<float> scratch_memory) {
   const int h_size = static_cast<int>(h.size());
   const int x_size = static_cast<int>(x.size());
   RTC_DCHECK_EQ(0, h_size % 4);
@@ -290,13 +290,13 @@ void MatchedFilterCore_NEON(size_t x_start_index,
 void MatchedFilterCore_AccumulatedError_SSE2(size_t x_start_index,
                                              float x2_sum_threshold,
                                              float smoothing,
-                                             ArrayView<const float> x,
-                                             ArrayView<const float> y,
-                                             ArrayView<float> h,
+                                             std::span<const float> x,
+                                             std::span<const float> y,
+                                             std::span<float> h,
                                              bool* filters_updated,
                                              float* error_sum,
-                                             ArrayView<float> accumulated_error,
-                                             ArrayView<float> scratch_memory) {
+                                             std::span<float> accumulated_error,
+                                             std::span<float> scratch_memory) {
   const int h_size = static_cast<int>(h.size());
   const int x_size = static_cast<int>(x.size());
   RTC_DCHECK_EQ(0, h_size % 8);
@@ -386,14 +386,14 @@ void MatchedFilterCore_AccumulatedError_SSE2(size_t x_start_index,
 void MatchedFilterCore_SSE2(size_t x_start_index,
                             float x2_sum_threshold,
                             float smoothing,
-                            ArrayView<const float> x,
-                            ArrayView<const float> y,
-                            ArrayView<float> h,
+                            std::span<const float> x,
+                            std::span<const float> y,
+                            std::span<float> h,
                             bool* filters_updated,
                             float* error_sum,
                             bool compute_accumulated_error,
-                            ArrayView<float> accumulated_error,
-                            ArrayView<float> scratch_memory) {
+                            std::span<float> accumulated_error,
+                            std::span<float> scratch_memory) {
   if (compute_accumulated_error) {
     return MatchedFilterCore_AccumulatedError_SSE2(
         x_start_index, x2_sum_threshold, smoothing, x, y, h, filters_updated,
@@ -498,13 +498,13 @@ void MatchedFilterCore_SSE2(size_t x_start_index,
 void MatchedFilterCore(size_t x_start_index,
                        float x2_sum_threshold,
                        float smoothing,
-                       ArrayView<const float> x,
-                       ArrayView<const float> y,
-                       ArrayView<float> h,
+                       std::span<const float> x,
+                       std::span<const float> y,
+                       std::span<float> h,
                        bool* filters_updated,
                        float* error_sum,
                        bool compute_accumulated_error,
-                       ArrayView<float> accumulated_error) {
+                       std::span<float> accumulated_error) {
   if (compute_accumulated_error) {
     std::fill(accumulated_error.begin(), accumulated_error.end(), 0.0f);
   }
@@ -556,7 +556,7 @@ void MatchedFilterCore(size_t x_start_index,
   }
 }
 
-size_t MaxSquarePeakIndex(ArrayView<const float> h) {
+size_t MaxSquarePeakIndex(std::span<const float> h) {
   if (h.size() < 2) {
     return 0;
   }
@@ -656,7 +656,7 @@ void MatchedFilter::Reset(bool full_reset) {
 }
 
 void MatchedFilter::Update(const DownsampledRenderBuffer& render_buffer,
-                           ArrayView<const float> capture,
+                           std::span<const float> capture,
                            bool use_slow_smoothing) {
   RTC_DCHECK_EQ(sub_block_size_, capture.size());
   auto& y = capture;
diff --git a/modules/audio_processing/aec3/matched_filter.h b/modules/audio_processing/aec3/matched_filter.h
index 2a910b7eb2..993670eafc 100644
--- a/modules/audio_processing/aec3/matched_filter.h
+++ b/modules/audio_processing/aec3/matched_filter.h
@@ -14,9 +14,9 @@
 #include <stddef.h>
 
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "rtc_base/system/arch.h"
 
@@ -33,14 +33,14 @@ namespace aec3 {
 void MatchedFilterCore_NEON(size_t x_start_index,
                             float x2_sum_threshold,
                             float smoothing,
-                            webrtc::ArrayView<const float> x,
-                            webrtc::ArrayView<const float> y,
-                            webrtc::ArrayView<float> h,
+                            std::span<const float> x,
+                            std::span<const float> y,
+                            std::span<float> h,
                             bool* filters_updated,
                             float* error_sum,
                             bool compute_accumulation_error,
-                            webrtc::ArrayView<float> accumulated_error,
-                            webrtc::ArrayView<float> scratch_memory);
+                            std::span<float> accumulated_error,
+                            std::span<float> scratch_memory);
 
 #endif
 
@@ -50,27 +50,27 @@ void MatchedFilterCore_NEON(size_t x_start_index,
 void MatchedFilterCore_SSE2(size_t x_start_index,
                             float x2_sum_threshold,
                             float smoothing,
-                            ArrayView<const float> x,
-                            ArrayView<const float> y,
-                            ArrayView<float> h,
+                            std::span<const float> x,
+                            std::span<const float> y,
+                            std::span<float> h,
                             bool* filters_updated,
                             float* error_sum,
                             bool compute_accumulated_error,
-                            ArrayView<float> accumulated_error,
-                            ArrayView<float> scratch_memory);
+                            std::span<float> accumulated_error,
+                            std::span<float> scratch_memory);
 
 // Filter core for the matched filter that is optimized for AVX2.
 void MatchedFilterCore_AVX2(size_t x_start_index,
                             float x2_sum_threshold,
                             float smoothing,
-                            ArrayView<const float> x,
-                            ArrayView<const float> y,
-                            ArrayView<float> h,
+                            std::span<const float> x,
+                            std::span<const float> y,
+                            std::span<float> h,
                             bool* filters_updated,
                             float* error_sum,
                             bool compute_accumulated_error,
-                            ArrayView<float> accumulated_error,
-                            ArrayView<float> scratch_memory);
+                            std::span<float> accumulated_error,
+                            std::span<float> scratch_memory);
 
 #endif
 
@@ -78,16 +78,16 @@ void MatchedFilterCore_AVX2(size_t x_start_index,
 void MatchedFilterCore(size_t x_start_index,
                        float x2_sum_threshold,
                        float smoothing,
-                       ArrayView<const float> x,
-                       ArrayView<const float> y,
-                       ArrayView<float> h,
+                       std::span<const float> x,
+                       std::span<const float> y,
+                       std::span<float> h,
                        bool* filters_updated,
                        float* error_sum,
                        bool compute_accumulation_error,
-                       ArrayView<float> accumulated_error);
+                       std::span<float> accumulated_error);
 
 // Find largest peak of squared values in array.
-size_t MaxSquarePeakIndex(ArrayView<const float> h);
+size_t MaxSquarePeakIndex(std::span<const float> h);
 
 }  // namespace aec3
 
@@ -125,7 +125,7 @@ class MatchedFilter {
 
   // Updates the correlation with the values in the capture buffer.
   void Update(const DownsampledRenderBuffer& render_buffer,
-              ArrayView<const float> capture,
+              std::span<const float> capture,
               bool use_slow_smoothing);
 
   // Resets the matched filter.
diff --git a/modules/audio_processing/aec3/matched_filter_avx2.cc b/modules/audio_processing/aec3/matched_filter_avx2.cc
index 1312c83db6..0b9360c210 100644
--- a/modules/audio_processing/aec3/matched_filter_avx2.cc
+++ b/modules/audio_processing/aec3/matched_filter_avx2.cc
@@ -12,8 +12,8 @@
 
 #include <algorithm>
 #include <cstddef>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/matched_filter.h"
 #include "rtc_base/checks.h"
 
@@ -35,13 +35,13 @@ inline __m128 hsum_ab(__m256 a, __m256 b) {
 void MatchedFilterCore_AccumulatedError_AVX2(size_t x_start_index,
                                              float x2_sum_threshold,
                                              float smoothing,
-                                             ArrayView<const float> x,
-                                             ArrayView<const float> y,
-                                             ArrayView<float> h,
+                                             std::span<const float> x,
+                                             std::span<const float> y,
+                                             std::span<float> h,
                                              bool* filters_updated,
                                              float* error_sum,
-                                             ArrayView<float> accumulated_error,
-                                             ArrayView<float> scratch_memory) {
+                                             std::span<float> accumulated_error,
+                                             std::span<float> scratch_memory) {
   const int h_size = static_cast<int>(h.size());
   const int x_size = static_cast<int>(x.size());
   RTC_DCHECK_EQ(0, h_size % 16);
@@ -141,14 +141,14 @@ void MatchedFilterCore_AccumulatedError_AVX2(size_t x_start_index,
 void MatchedFilterCore_AVX2(size_t x_start_index,
                             float x2_sum_threshold,
                             float smoothing,
-                            ArrayView<const float> x,
-                            ArrayView<const float> y,
-                            ArrayView<float> h,
+                            std::span<const float> x,
+                            std::span<const float> y,
+                            std::span<float> h,
                             bool* filters_updated,
                             float* error_sum,
                             bool compute_accumulated_error,
-                            ArrayView<float> accumulated_error,
-                            ArrayView<float> scratch_memory) {
+                            std::span<float> accumulated_error,
+                            std::span<float> scratch_memory) {
   if (compute_accumulated_error) {
     return MatchedFilterCore_AccumulatedError_AVX2(
         x_start_index, x2_sum_threshold, smoothing, x, y, h, filters_updated,
diff --git a/modules/audio_processing/aec3/matched_filter_lag_aggregator.cc b/modules/audio_processing/aec3/matched_filter_lag_aggregator.cc
index 4909182157..51e5db84b5 100644
--- a/modules/audio_processing/aec3/matched_filter_lag_aggregator.cc
+++ b/modules/audio_processing/aec3/matched_filter_lag_aggregator.cc
@@ -14,8 +14,8 @@
 #include <iterator>
 #include <memory>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/delay_estimate.h"
@@ -81,7 +81,7 @@ std::optional<DelayEstimate> MatchedFilterLagAggregator::Aggregate(
   if (lag_estimate) {
     highest_peak_aggregator_.Aggregate(
         std::max(0, static_cast<int>(lag_estimate->lag) - headroom_));
-    ArrayView<const int> histogram = highest_peak_aggregator_.histogram();
+    std::span<const int> histogram = highest_peak_aggregator_.histogram();
     int candidate = highest_peak_aggregator_.candidate();
     significant_candidate_found_ = significant_candidate_found_ ||
                                    histogram[candidate] > thresholds_.converged;
diff --git a/modules/audio_processing/aec3/matched_filter_lag_aggregator.h b/modules/audio_processing/aec3/matched_filter_lag_aggregator.h
index 26cafbca15..d53464fb42 100644
--- a/modules/audio_processing/aec3/matched_filter_lag_aggregator.h
+++ b/modules/audio_processing/aec3/matched_filter_lag_aggregator.h
@@ -15,9 +15,9 @@
 #include <cstddef>
 #include <memory>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "modules/audio_processing/aec3/delay_estimate.h"
 #include "modules/audio_processing/aec3/matched_filter.h"
@@ -81,7 +81,7 @@ class MatchedFilterLagAggregator {
     void Reset();
     void Aggregate(int lag);
     int candidate() const { return candidate_; }
-    ArrayView<const int> histogram() const { return histogram_; }
+    std::span<const int> histogram() const { return histogram_; }
 
    private:
     std::vector<int> histogram_;
diff --git a/modules/audio_processing/aec3/matched_filter_unittest.cc b/modules/audio_processing/aec3/matched_filter_unittest.cc
index dc75bcccce..7369ec2a57 100644
--- a/modules/audio_processing/aec3/matched_filter_unittest.cc
+++ b/modules/audio_processing/aec3/matched_filter_unittest.cc
@@ -16,10 +16,10 @@
 #include <cstdlib>
 #include <iterator>
 #include <memory>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/block.h"
@@ -305,7 +305,7 @@ TEST_P(MatchedFilterTest, LagEstimation) {
 
         render_delay_buffer->PrepareCaptureProcessing();
         std::array<float, kBlockSize> downsampled_capture_data;
-        ArrayView<float> downsampled_capture(downsampled_capture_data.data(),
+        std::span<float> downsampled_capture(downsampled_capture_data.data(),
                                              sub_block_size);
         capture_decimator.Decimate(capture[0], downsampled_capture);
         filter.Update(render_delay_buffer->GetDownsampledRenderBuffer(),
@@ -380,7 +380,7 @@ TEST_P(MatchedFilterTest, PreEchoEstimation) {
       }
       render_delay_buffer->PrepareCaptureProcessing();
       std::array<float, kBlockSize> downsampled_capture_data;
-      ArrayView<float> downsampled_capture(downsampled_capture_data.data(),
+      std::span<float> downsampled_capture(downsampled_capture_data.data(),
                                            sub_block_size);
       capture_decimator.Decimate(capture[0], downsampled_capture);
       filter.Update(render_delay_buffer->GetDownsampledRenderBuffer(),
@@ -421,7 +421,7 @@ TEST_P(MatchedFilterTest, LagNotReliableForUncorrelatedRenderAndCapture) {
 
     Block render(kNumBands, kNumChannels);
     std::array<float, kBlockSize> capture_data;
-    ArrayView<float> capture(capture_data.data(), sub_block_size);
+    std::span<float> capture(capture_data.data(), sub_block_size);
     std::fill(capture.begin(), capture.end(), 0.f);
     ApmDataDumper data_dumper(0);
     std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
@@ -486,7 +486,7 @@ TEST_P(MatchedFilterTest, LagNotUpdatedForLowLevelRender) {
       }
       std::copy(render.begin(0, 0), render.end(0, 0), capture[0].begin());
       std::array<float, kBlockSize> downsampled_capture_data;
-      ArrayView<float> downsampled_capture(downsampled_capture_data.data(),
+      std::span<float> downsampled_capture(downsampled_capture_data.data(),
                                            sub_block_size);
       capture_decimator.Decimate(capture[0], downsampled_capture);
       filter.Update(render_delay_buffer->GetDownsampledRenderBuffer(),
diff --git a/modules/audio_processing/aec3/moving_average.cc b/modules/audio_processing/aec3/moving_average.cc
index e1e5589ff3..503b86bf9b 100644
--- a/modules/audio_processing/aec3/moving_average.cc
+++ b/modules/audio_processing/aec3/moving_average.cc
@@ -14,8 +14,8 @@
 #include <algorithm>
 #include <cstddef>
 #include <functional>
+#include <span>
 
-#include "api/array_view.h"
 #include "rtc_base/checks.h"
 
 namespace webrtc {
@@ -33,8 +33,8 @@ MovingAverage::MovingAverage(size_t num_elem, size_t mem_len)
 
 MovingAverage::~MovingAverage() = default;
 
-void MovingAverage::Average(ArrayView<const float> input,
-                            ArrayView<float> output) {
+void MovingAverage::Average(std::span<const float> input,
+                            std::span<float> output) {
   RTC_DCHECK(input.size() == num_elem_);
   RTC_DCHECK(output.size() == num_elem_);
 
diff --git a/modules/audio_processing/aec3/moving_average.h b/modules/audio_processing/aec3/moving_average.h
index 68f2fd0fcc..77e2a5907d 100644
--- a/modules/audio_processing/aec3/moving_average.h
+++ b/modules/audio_processing/aec3/moving_average.h
@@ -13,10 +13,9 @@
 
 #include <stddef.h>
 
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
-
 namespace webrtc {
 namespace aec3 {
 
@@ -29,7 +28,7 @@ class MovingAverage {
 
   // Computes the average of input and mem_len-1 previous inputs and stores the
   // result in output.
-  void Average(ArrayView<const float> input, ArrayView<float> output);
+  void Average(std::span<const float> input, std::span<float> output);
 
  private:
   const size_t num_elem_;
diff --git a/modules/audio_processing/aec3/nearend_detector.h b/modules/audio_processing/aec3/nearend_detector.h
index a1f30a977c..c95c19edc1 100644
--- a/modules/audio_processing/aec3/nearend_detector.h
+++ b/modules/audio_processing/aec3/nearend_detector.h
@@ -12,8 +12,8 @@
 #define MODULES_AUDIO_PROCESSING_AEC3_NEAREND_DETECTOR_H_
 
 #include <array>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 
 namespace webrtc {
@@ -27,10 +27,10 @@ class NearendDetector {
 
   // Updates the state selection based on latest spectral estimates.
   virtual void Update(
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>> nearend_spectrum,
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+      std::span<const std::array<float, kFftLengthBy2Plus1>> nearend_spectrum,
+      std::span<const std::array<float, kFftLengthBy2Plus1>>
           residual_echo_spectrum,
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+      std::span<const std::array<float, kFftLengthBy2Plus1>>
           comfort_noise_spectrum,
       bool initial_state) = 0;
 };
diff --git a/modules/audio_processing/aec3/neural_residual_echo_estimator/BUILD.gn b/modules/audio_processing/aec3/neural_residual_echo_estimator/BUILD.gn
index 120d46a4f6..029a0114f6 100644
--- a/modules/audio_processing/aec3/neural_residual_echo_estimator/BUILD.gn
+++ b/modules/audio_processing/aec3/neural_residual_echo_estimator/BUILD.gn
@@ -34,7 +34,6 @@ if (rtc_enable_protobuf) {
       "..:aec3_block",
       "..:aec3_common",
       "../..:apm_logging",
-      "../../../../api:array_view",
       "../../../../api/audio:aec3_config",
       "../../../../api/audio:neural_residual_echo_estimator_api",
       "../../../../common_audio",
@@ -70,7 +69,6 @@ if (rtc_include_tests) {
         ":neural_residual_echo_estimator_proto",
         "..:aec3_block",
         "..:aec3_common",
-        "../../../../api:array_view",
         "../../../../rtc_base:checks",
         "../../../../rtc_base:random",
         "../../../../test:fileutils",
diff --git a/modules/audio_processing/aec3/neural_residual_echo_estimator/neural_feature_extractor.cc b/modules/audio_processing/aec3/neural_residual_echo_estimator/neural_feature_extractor.cc
index 45c08e1c70..63615c9eb1 100644
--- a/modules/audio_processing/aec3/neural_residual_echo_estimator/neural_feature_extractor.cc
+++ b/modules/audio_processing/aec3/neural_residual_echo_estimator/neural_feature_extractor.cc
@@ -15,9 +15,9 @@
 #include <cmath>
 #include <cstring>
 #include <memory>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "common_audio/window_generator.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "rtc_base/checks.h"
@@ -46,7 +46,7 @@ std::vector<float> GetSqrtHanningWindow(int frame_size, float scale) {
 }
 
 std::array<float, kBlockSize> AverageAllChannels(
-    ArrayView<const ArrayView<const float, kBlockSize>> all_channels) {
+    std::span<const std::span<const float, kBlockSize>> all_channels) {
   std::array<float, kBlockSize> summed_block;
   summed_block.fill(0.0f);
   const float scale = kScale * 1.0f / all_channels.size();
@@ -93,7 +93,7 @@ bool TimeDomainFeatureExtractor::ReadyForInference() const {
 }
 
 void TimeDomainFeatureExtractor::UpdateBuffers(
-    ArrayView<const ArrayView<const float, kBlockSize>> all_channels,
+    std::span<const std::span<const float, kBlockSize>> all_channels,
     ModelInputEnum input_type) {
   if (!RequiredInput(input_type)) {
     return;
@@ -105,7 +105,7 @@ void TimeDomainFeatureExtractor::UpdateBuffers(
                       summed_block.end());
 }
 
-void TimeDomainFeatureExtractor::PrepareModelInput(ArrayView<float> model_input,
+void TimeDomainFeatureExtractor::PrepareModelInput(std::span<float> model_input,
                                                    ModelInputEnum input_type) {
   if (!RequiredInput(input_type)) {
     return;
@@ -174,10 +174,10 @@ bool FrequencyDomainFeatureExtractor::ReadyForInference() const {
 }
 
 void FrequencyDomainFeatureExtractor::ComputeAndAddPowerSpectra(
-    ArrayView<const float> frame,
+    std::span<const float> frame,
     std::unique_ptr<PffftState>& pffft_state,
     int number_channels,
-    ArrayView<float> power_spectra) {
+    std::span<float> power_spectra) {
   const float kAverageScale = 1.0f / number_channels;
   if (pffft_state == nullptr) {
     pffft_state = std::make_unique<PffftState>(frame_size_);
@@ -202,7 +202,7 @@ void FrequencyDomainFeatureExtractor::ComputeAndAddPowerSpectra(
 }
 
 void FrequencyDomainFeatureExtractor::UpdateBuffers(
-    ArrayView<const ArrayView<const float, kBlockSize>> all_channels,
+    std::span<const std::span<const float, kBlockSize>> all_channels,
     ModelInputEnum input_type) {
   if (!RequiredInput(input_type)) {
     return;
@@ -211,14 +211,14 @@ void FrequencyDomainFeatureExtractor::UpdateBuffers(
       input_buffer_[static_cast<size_t>(input_type)];
   input_buffer.resize(all_channels.size());
   for (size_t ch = 0; ch < all_channels.size(); ++ch) {
-    const ArrayView<const float, kBlockSize>& frame_in = all_channels[ch];
+    const std::span<const float, kBlockSize>& frame_in = all_channels[ch];
     std::vector<float>& input_buffer_ch = input_buffer[ch];
     input_buffer_ch.insert(input_buffer_ch.end(), frame_in.begin(),
                            frame_in.end());
   }
 }
 void FrequencyDomainFeatureExtractor::PrepareModelInput(
-    ArrayView<float> model_input,
+    std::span<float> model_input,
     ModelInputEnum input_type) {
   if (!RequiredInput(input_type)) {
     return;
diff --git a/modules/audio_processing/aec3/neural_residual_echo_estimator/neural_feature_extractor.h b/modules/audio_processing/aec3/neural_residual_echo_estimator/neural_feature_extractor.h
index 5dba121d32..2071fb944a 100644
--- a/modules/audio_processing/aec3/neural_residual_echo_estimator/neural_feature_extractor.h
+++ b/modules/audio_processing/aec3/neural_residual_echo_estimator/neural_feature_extractor.h
@@ -13,9 +13,9 @@
 
 #include <cstring>
 #include <memory>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "third_party/pffft/src/pffft.h"
 
@@ -45,11 +45,11 @@ class FeatureExtractor {
 
   // Buffers the frames for matching the expecting inference step size.
   virtual void UpdateBuffers(
-      ArrayView<const ArrayView<const float, kBlockSize>> all_channels,
+      std::span<const std::span<const float, kBlockSize>> all_channels,
       ModelInputEnum input_type) = 0;
 
   // Uses the internal buffer data for producing the model input tensors.
-  virtual void PrepareModelInput(ArrayView<float> model_input,
+  virtual void PrepareModelInput(std::span<float> model_input,
                                  ModelInputEnum input_type) = 0;
 
   // Resets the internal state of the feature extractor.
@@ -66,10 +66,10 @@ class TimeDomainFeatureExtractor : public FeatureExtractor {
   bool ReadyForInference() const override;
 
   void UpdateBuffers(
-      ArrayView<const ArrayView<const float, kBlockSize>> all_channels,
+      std::span<const std::span<const float, kBlockSize>> all_channels,
       ModelInputEnum input_type) override;
 
-  void PrepareModelInput(ArrayView<float> model_input,
+  void PrepareModelInput(std::span<float> model_input,
                          ModelInputEnum input_type) override;
 
  private:
@@ -87,10 +87,10 @@ class FrequencyDomainFeatureExtractor : public FeatureExtractor {
   bool ReadyForInference() const override;
 
   void UpdateBuffers(
-      ArrayView<const ArrayView<const float, kBlockSize>> all_channels,
+      std::span<const std::span<const float, kBlockSize>> all_channels,
       ModelInputEnum input_type) override;
 
-  void PrepareModelInput(ArrayView<float> model_input,
+  void PrepareModelInput(std::span<float> model_input,
                          ModelInputEnum input_type) override;
 
  private:
@@ -108,10 +108,10 @@ class FrequencyDomainFeatureExtractor : public FeatureExtractor {
     float* const data_;
   };
 
-  void ComputeAndAddPowerSpectra(ArrayView<const float> frame,
+  void ComputeAndAddPowerSpectra(std::span<const float> frame,
                                  std::unique_ptr<PffftState>& pffft_state,
                                  int number_channels,
-                                 ArrayView<float> power_spectra);
+                                 std::span<float> power_spectra);
 
   const size_t step_size_;
   const int frame_size_;
diff --git a/modules/audio_processing/aec3/neural_residual_echo_estimator/neural_feature_extractor_unittest.cc b/modules/audio_processing/aec3/neural_residual_echo_estimator/neural_feature_extractor_unittest.cc
index 66ccf6e958..757af1bc7a 100644
--- a/modules/audio_processing/aec3/neural_residual_echo_estimator/neural_feature_extractor_unittest.cc
+++ b/modules/audio_processing/aec3/neural_residual_echo_estimator/neural_feature_extractor_unittest.cc
@@ -53,9 +53,9 @@ print(format_as_cpp_array(expected_output2, "expected_output2"))
 
 #include <array>
 #include <cstddef>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
@@ -229,9 +229,9 @@ class FrequencyDomainFeatureExtractorTest
     : public ::testing::TestWithParam<int> {
  protected:
   void UpdateBlock(FrequencyDomainFeatureExtractor& extractor,
-                   ArrayView<const float, kBlockSize> block,
+                   std::span<const float, kBlockSize> block,
                    int num_channels) {
-    std::vector<ArrayView<const float, kBlockSize>> all_blocks;
+    std::vector<std::span<const float, kBlockSize>> all_blocks;
     all_blocks.reserve(num_channels);
     for (int i = 0; i < num_channels; ++i) {
       all_blocks.push_back(block);
@@ -248,7 +248,7 @@ TEST_P(FrequencyDomainFeatureExtractorTest, BasicTest) {
   std::vector<float> output(kStepSize + 1);
   // First frame.
   for (int i = 0; i < kStepSize; i += kBlockSize) {
-    ArrayView<const float, kBlockSize> block(&noise1_scaled[i], kBlockSize);
+    std::span<const float, kBlockSize> block(&noise1_scaled[i], kBlockSize);
     this->UpdateBlock(extractor, block, num_channels);
   }
   EXPECT_TRUE(extractor.ReadyForInference());
@@ -258,7 +258,7 @@ TEST_P(FrequencyDomainFeatureExtractorTest, BasicTest) {
   EXPECT_FALSE(extractor.ReadyForInference());
 
   for (int i = kStepSize; i < 2 * kStepSize; i += kBlockSize) {
-    ArrayView<const float, kBlockSize> block(&noise1_scaled[i], kBlockSize);
+    std::span<const float, kBlockSize> block(&noise1_scaled[i], kBlockSize);
     this->UpdateBlock(extractor, block, num_channels);
   }
   EXPECT_TRUE(extractor.ReadyForInference());
@@ -270,7 +270,7 @@ TEST_P(FrequencyDomainFeatureExtractorTest, BasicTest) {
   EXPECT_FALSE(extractor.ReadyForInference());
   // Second frame.
   for (int i = 0; i < kStepSize; i += kBlockSize) {
-    ArrayView<const float, kBlockSize> block(&noise2_scaled[i], kBlockSize);
+    std::span<const float, kBlockSize> block(&noise2_scaled[i], kBlockSize);
     this->UpdateBlock(extractor, block, num_channels);
   }
   EXPECT_TRUE(extractor.ReadyForInference());
@@ -279,7 +279,7 @@ TEST_P(FrequencyDomainFeatureExtractorTest, BasicTest) {
   }
 
   for (int i = kStepSize; i < 2 * kStepSize; i += kBlockSize) {
-    ArrayView<const float, kBlockSize> block(&noise2_scaled[i], kBlockSize);
+    std::span<const float, kBlockSize> block(&noise2_scaled[i], kBlockSize);
     this->UpdateBlock(extractor, block, num_channels);
   }
   EXPECT_TRUE(extractor.ReadyForInference());
@@ -318,11 +318,11 @@ TEST(TimeDomainFeatureExtractorTest, BasicTest) {
     block1[i] = i;
     block2[i] = i + kBlockSize;
   }
-  ArrayView<const float, kBlockSize> block1_view(block1);
-  ArrayView<const ArrayView<const float, kBlockSize>> all_blocks1(&block1_view,
+  std::span<const float, kBlockSize> block1_view(block1);
+  std::span<const std::span<const float, kBlockSize>> all_blocks1(&block1_view,
                                                                   1);
-  ArrayView<const float, kBlockSize> block2_view(block2);
-  ArrayView<const ArrayView<const float, kBlockSize>> all_blocks2(&block2_view,
+  std::span<const float, kBlockSize> block2_view(block2);
+  std::span<const std::span<const float, kBlockSize>> all_blocks2(&block2_view,
                                                                   1);
 
   for (auto input_type : kExpectedInputs) {
@@ -346,7 +346,7 @@ TEST(TimeDomainFeatureExtractorTest, BasicTest) {
 TEST(TimeDomainFeatureExtractorTest, ResetsState) {
   TimeDomainFeatureExtractor extractor(kStepSize);
   const std::array<float, kBlockSize> block{};
-  const std::array<const ArrayView<const float, kBlockSize>, 1> all_blocks = {
+  const std::array<const std::span<const float, kBlockSize>, 1> all_blocks = {
       block};
   EXPECT_FALSE(extractor.ReadyForInference());
   for (size_t i = 0; i < kStepSize / kBlockSize; ++i) {
diff --git a/modules/audio_processing/aec3/neural_residual_echo_estimator/neural_residual_echo_estimator_impl.cc b/modules/audio_processing/aec3/neural_residual_echo_estimator/neural_residual_echo_estimator_impl.cc
index 1cec3a3367..0c80ef18ce 100644
--- a/modules/audio_processing/aec3/neural_residual_echo_estimator/neural_residual_echo_estimator_impl.cc
+++ b/modules/audio_processing/aec3/neural_residual_echo_estimator/neural_residual_echo_estimator_impl.cc
@@ -17,12 +17,12 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/base/nullability.h"
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "api/audio/neural_residual_echo_estimator.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
@@ -82,8 +82,8 @@ std::optional<audioproc::ReeModelMetadata> ReadModelMetadata(
 
 // Downsamples the model output mask to the AEC3 frequency resolution and
 // transforms it from a nearend prediction to an echo power mask.
-void DownsampleAndTransformMask(ArrayView<const float> mask,
-                                ArrayView<float> downsampled_mask) {
+void DownsampleAndTransformMask(std::span<const float> mask,
+                                std::span<float> downsampled_mask) {
   const int kDownsampleFactor =
       static_cast<int>((mask.size() - 1) / kFftLengthBy2);
   downsampled_mask[0] = mask[0];
@@ -293,7 +293,7 @@ class TfLiteModelRunner : public NeuralResidualEchoEstimatorImpl::ModelRunner {
     for (const auto input_enum :
          {ModelInputEnum::kMic, ModelInputEnum::kLinearAecOutput,
           ModelInputEnum::kAecRef}) {
-      ArrayView<float> input_tensor = GetInput(input_enum);
+      std::span<float> input_tensor = GetInput(input_enum);
       std::fill(input_tensor.begin(), input_tensor.end(), 0.0f);
     }
   }
@@ -305,34 +305,34 @@ class TfLiteModelRunner : public NeuralResidualEchoEstimatorImpl::ModelRunner {
     for (const auto input_enum :
          {ModelInputEnum::kMic, ModelInputEnum::kLinearAecOutput,
           ModelInputEnum::kAecRef}) {
-      ArrayView<float> input_tensor = GetInput(input_enum);
+      std::span<float> input_tensor = GetInput(input_enum);
       std::fill(input_tensor.begin(), input_tensor.end(), 0.0f);
     }
   }
 
   int StepSize() const override { return step_size_; }
 
-  ArrayView<float> GetInput(
+  std::span<float> GetInput(
       FeatureExtractor::ModelInputEnum input_enum) override {
     size_t index = input_tensor_indexes_[static_cast<size_t>(input_enum)];
     TfLiteTensor* input_tensor = tflite_interpreter_->tensor(index);
     float* input_typed_tensor =
         reinterpret_cast<float*>(input_tensor->data.data);
-    return ArrayView<float>(input_typed_tensor,
+    return std::span<float>(input_typed_tensor,
                             tflite::NumElements(input_tensor));
   }
 
-  ArrayView<const float> GetOutput(
+  std::span<const float> GetOutput(
       FeatureExtractor::ModelOutputEnum output_enum) override {
     if (!use_unbounded_mask_ &&
         output_enum == ModelOutputEnum::kUnboundedEchoMask) {
-      return ArrayView<const float>();
+      return std::span<const float>();
     }
     size_t index = output_tensor_indexes_[static_cast<size_t>(output_enum)];
     const TfLiteTensor* output_tensor = tflite_interpreter_->tensor(index);
     const float* output_typed_tensor =
         reinterpret_cast<const float*>(output_tensor->data.data);
-    return ArrayView<const float>(output_typed_tensor,
+    return std::span<const float>(output_typed_tensor,
                                   tflite::NumElements(output_tensor));
   }
 
@@ -488,14 +488,14 @@ void NeuralResidualEchoEstimatorImpl::Reset() {
 
 void NeuralResidualEchoEstimatorImpl::Estimate(
     const Block& render,
-    ArrayView<const std::array<float, kBlockSize>> y,
-    ArrayView<const std::array<float, kBlockSize>> e,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> S2,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> Y2,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> E2,
+    std::span<const std::array<float, kBlockSize>> y,
+    std::span<const std::array<float, kBlockSize>> e,
+    std::span<const std::array<float, kFftLengthBy2Plus1>> S2,
+    std::span<const std::array<float, kFftLengthBy2Plus1>> Y2,
+    std::span<const std::array<float, kFftLengthBy2Plus1>> E2,
     bool dominant_nearend,
-    ArrayView<std::array<float, kFftLengthBy2Plus1>> R2,
-    ArrayView<std::array<float, kFftLengthBy2Plus1>> R2_unbounded) {
+    std::span<std::array<float, kFftLengthBy2Plus1>> R2,
+    std::span<std::array<float, kFftLengthBy2Plus1>> R2_unbounded) {
   DumpInputs(render, y, e);
   render_channels_.clear();
   for (int i = 0; i < render.NumChannels(); ++i) {
@@ -525,11 +525,11 @@ void NeuralResidualEchoEstimatorImpl::Estimate(
         ModelInputEnum::kAecRef);
     if (model_runner_->Invoke()) {
       // Downsample output mask to match the AEC3 frequency resolution.
-      ArrayView<const float> output_mask =
+      std::span<const float> output_mask =
           model_runner_->GetOutput(ModelOutputEnum::kEchoMask);
       DownsampleAndTransformMask(output_mask, output_mask_);
       if (use_unbounded_mask_) {
-        ArrayView<const float> output_mask_unbounded =
+        std::span<const float> output_mask_unbounded =
             model_runner_->GetOutput(ModelOutputEnum::kUnboundedEchoMask);
         DownsampleAndTransformMask(output_mask_unbounded,
                                    output_mask_unbounded_);
@@ -589,8 +589,8 @@ EchoCanceller3Config NeuralResidualEchoEstimatorImpl::GetConfiguration(
 
 void NeuralResidualEchoEstimatorImpl::DumpInputs(
     const Block& render,
-    ArrayView<const std::array<float, kBlockSize>> y,
-    ArrayView<const std::array<float, kBlockSize>> e) {
+    std::span<const std::array<float, kBlockSize>> y,
+    std::span<const std::array<float, kBlockSize>> e) {
   data_dumper_->DumpWav("ml_ree_mic_input", y[0], 16000, 1);
   data_dumper_->DumpWav("ml_ree_linear_aec_output", e[0], 16000, 1);
   data_dumper_->DumpWav("ml_ree_aec_ref", render.View(0, 0), 16000, 1);
diff --git a/modules/audio_processing/aec3/neural_residual_echo_estimator/neural_residual_echo_estimator_impl.h b/modules/audio_processing/aec3/neural_residual_echo_estimator/neural_residual_echo_estimator_impl.h
index 4ee8634091..2c2f091089 100644
--- a/modules/audio_processing/aec3/neural_residual_echo_estimator/neural_residual_echo_estimator_impl.h
+++ b/modules/audio_processing/aec3/neural_residual_echo_estimator/neural_residual_echo_estimator_impl.h
@@ -13,10 +13,10 @@
 
 #include <array>
 #include <memory>
+#include <span>
 #include <vector>
 
 #include "absl/base/nullability.h"
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "api/audio/neural_residual_echo_estimator.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
@@ -45,9 +45,9 @@ class NeuralResidualEchoEstimatorImpl : public NeuralResidualEchoEstimator {
     virtual ~ModelRunner() = default;
 
     virtual int StepSize() const = 0;
-    virtual ArrayView<float> GetInput(
+    virtual std::span<float> GetInput(
         FeatureExtractor::ModelInputEnum input_enum) = 0;
-    virtual ArrayView<const float> GetOutput(
+    virtual std::span<const float> GetOutput(
         FeatureExtractor::ModelOutputEnum output_enum) = 0;
     virtual const audioproc::ReeModelMetadata& GetMetadata() const = 0;
     virtual bool Invoke() = 0;
@@ -71,14 +71,14 @@ class NeuralResidualEchoEstimatorImpl : public NeuralResidualEchoEstimator {
 
   void Estimate(
       const Block& render,
-      ArrayView<const std::array<float, kBlockSize>> y,
-      ArrayView<const std::array<float, kBlockSize>> e,
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>> S2,
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>> Y2,
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>> E2,
+      std::span<const std::array<float, kBlockSize>> y,
+      std::span<const std::array<float, kBlockSize>> e,
+      std::span<const std::array<float, kFftLengthBy2Plus1>> S2,
+      std::span<const std::array<float, kFftLengthBy2Plus1>> Y2,
+      std::span<const std::array<float, kFftLengthBy2Plus1>> E2,
       bool dominant_nearend,
-      ArrayView<std::array<float, kFftLengthBy2Plus1>> R2,
-      ArrayView<std::array<float, kFftLengthBy2Plus1>> R2_unbounded) override;
+      std::span<std::array<float, kFftLengthBy2Plus1>> R2,
+      std::span<std::array<float, kFftLengthBy2Plus1>> R2_unbounded) override;
 
   EchoCanceller3Config GetConfiguration(bool multi_channel) const override;
 
@@ -86,8 +86,8 @@ class NeuralResidualEchoEstimatorImpl : public NeuralResidualEchoEstimator {
 
  private:
   void DumpInputs(const Block& render,
-                  ArrayView<const std::array<float, kBlockSize>> y,
-                  ArrayView<const std::array<float, kBlockSize>> e);
+                  std::span<const std::array<float, kBlockSize>> y,
+                  std::span<const std::array<float, kBlockSize>> e);
 
   // Encapsulates all ML model invocation work.
   const std::unique_ptr<ModelRunner> model_runner_;
@@ -100,9 +100,9 @@ class NeuralResidualEchoEstimatorImpl : public NeuralResidualEchoEstimator {
   std::array<float, kFftLengthBy2Plus1> output_mask_;
   std::array<float, kFftLengthBy2Plus1> output_mask_unbounded_;
 
-  std::vector<ArrayView<const float, kBlockSize>> render_channels_;
-  std::vector<ArrayView<const float, kBlockSize>> y_channels_;
-  std::vector<ArrayView<const float, kBlockSize>> e_channels_;
+  std::vector<std::span<const float, kBlockSize>> render_channels_;
+  std::vector<std::span<const float, kBlockSize>> y_channels_;
+  std::vector<std::span<const float, kBlockSize>> e_channels_;
 
   static int instance_count_;
   // Pointer to a data dumper that is used for debugging purposes.
diff --git a/modules/audio_processing/aec3/neural_residual_echo_estimator/neural_residual_echo_estimator_impl_unittest.cc b/modules/audio_processing/aec3/neural_residual_echo_estimator/neural_residual_echo_estimator_impl_unittest.cc
index 3da7b77715..a8681fb1e2 100644
--- a/modules/audio_processing/aec3/neural_residual_echo_estimator/neural_residual_echo_estimator_impl_unittest.cc
+++ b/modules/audio_processing/aec3/neural_residual_echo_estimator/neural_residual_echo_estimator_impl_unittest.cc
@@ -16,12 +16,12 @@
 #include <cstdint>
 #include <cstring>
 #include <memory>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/block.h"
 #include "modules/audio_processing/aec3/neural_residual_echo_estimator/neural_feature_extractor.h"
@@ -78,36 +78,34 @@ class MockModelRunner : public NeuralResidualEchoEstimatorImpl::ModelRunner {
 
   int StepSize() const override { return constants_.step_size; }
 
-  ArrayView<float> GetInput(ModelInputEnum input_enum) override {
+  std::span<float> GetInput(ModelInputEnum input_enum) override {
     switch (input_enum) {
       case ModelInputEnum::kMic:
-        return webrtc::ArrayView<float>(input_mic_.data(),
-                                        constants_.frame_size);
+        return std::span<float>(input_mic_.data(), constants_.frame_size);
       case ModelInputEnum::kLinearAecOutput:
-        return webrtc::ArrayView<float>(input_linear_aec_output_.data(),
-                                        constants_.frame_size);
+        return std::span<float>(input_linear_aec_output_.data(),
+                                constants_.frame_size);
       case ModelInputEnum::kAecRef:
-        return webrtc::ArrayView<float>(input_aec_ref_.data(),
-                                        constants_.frame_size);
+        return std::span<float>(input_aec_ref_.data(), constants_.frame_size);
       case ModelInputEnum::kModelState:
       case ModelInputEnum::kNumInputs:
         RTC_CHECK(false);
-        return webrtc::ArrayView<float>();
+        return std::span<float>();
     }
   }
 
-  ArrayView<const float> GetOutput(ModelOutputEnum output_enum) override {
+  std::span<const float> GetOutput(ModelOutputEnum output_enum) override {
     switch (output_enum) {
       case ModelOutputEnum::kEchoMask:
       case ModelOutputEnum::kUnboundedEchoMask:
-        return ArrayView<const float>(output_echo_mask_.data(),
+        return std::span<const float>(output_echo_mask_.data(),
                                       constants_.frame_size_by_2_plus_1);
       case ModelOutputEnum::kModelState:
         // Mock model state output if needed, for now return empty
-        return ArrayView<const float>();
+        return std::span<const float>();
       default:
         RTC_CHECK(false);
-        return ArrayView<float>();
+        return std::span<float>();
     }
   }
 
diff --git a/modules/audio_processing/aec3/refined_filter_update_gain.cc b/modules/audio_processing/aec3/refined_filter_update_gain.cc
index 80a55c21b3..3686903970 100644
--- a/modules/audio_processing/aec3/refined_filter_update_gain.cc
+++ b/modules/audio_processing/aec3/refined_filter_update_gain.cc
@@ -14,8 +14,8 @@
 #include <array>
 #include <atomic>
 #include <cstddef>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/echo_path_variability.h"
@@ -71,7 +71,7 @@ void RefinedFilterUpdateGain::Compute(
     const std::array<float, kFftLengthBy2Plus1>& render_power,
     const RenderSignalAnalyzer& render_signal_analyzer,
     const SubtractorOutput& subtractor_output,
-    ArrayView<const float> erl,
+    std::span<const float> erl,
     size_t size_partitions,
     bool saturated_capture_signal,
     bool disallow_leakage_diverged,
diff --git a/modules/audio_processing/aec3/refined_filter_update_gain.h b/modules/audio_processing/aec3/refined_filter_update_gain.h
index 660a0137b5..937ba8f974 100644
--- a/modules/audio_processing/aec3/refined_filter_update_gain.h
+++ b/modules/audio_processing/aec3/refined_filter_update_gain.h
@@ -16,8 +16,8 @@
 #include <array>
 #include <atomic>
 #include <memory>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 
@@ -49,7 +49,7 @@ class RefinedFilterUpdateGain {
   void Compute(const std::array<float, kFftLengthBy2Plus1>& render_power,
                const RenderSignalAnalyzer& render_signal_analyzer,
                const SubtractorOutput& subtractor_output,
-               ArrayView<const float> erl,
+               std::span<const float> erl,
                size_t size_partitions,
                bool saturated_capture_signal,
                bool disallow_leakage_diverged,
diff --git a/modules/audio_processing/aec3/render_buffer.h b/modules/audio_processing/aec3/render_buffer.h
index 8f76bd04af..f25416f828 100644
--- a/modules/audio_processing/aec3/render_buffer.h
+++ b/modules/audio_processing/aec3/render_buffer.h
@@ -14,9 +14,9 @@
 #include <stddef.h>
 
 #include <array>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/block.h"
 #include "modules/audio_processing/aec3/block_buffer.h"
@@ -48,7 +48,7 @@ class RenderBuffer {
   }
 
   // Get the spectrum from one of the FFTs in the buffer.
-  ArrayView<const std::array<float, kFftLengthBy2Plus1>> Spectrum(
+  std::span<const std::array<float, kFftLengthBy2Plus1>> Spectrum(
       int buffer_offset_ffts) const {
     int position = spectrum_buffer_->OffsetIndex(spectrum_buffer_->read,
                                                  buffer_offset_ffts);
@@ -56,7 +56,7 @@ class RenderBuffer {
   }
 
   // Returns the circular fft buffer.
-  ArrayView<const std::vector<FftData>> GetFftBuffer() const {
+  std::span<const std::vector<FftData>> GetFftBuffer() const {
     return fft_buffer_->buffer;
   }
 
diff --git a/modules/audio_processing/aec3/render_delay_buffer.cc b/modules/audio_processing/aec3/render_delay_buffer.cc
index b595eefc9e..ecc7431fa4 100644
--- a/modules/audio_processing/aec3/render_delay_buffer.cc
+++ b/modules/audio_processing/aec3/render_delay_buffer.cc
@@ -19,9 +19,9 @@
 #include <memory>
 #include <numeric>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/aec3_fft.h"
@@ -104,7 +104,7 @@ class RenderDelayBufferImpl final : public RenderDelayBuffer {
   int ComputeDelay() const;
   void ApplyTotalDelay(int delay);
   void InsertBlock(const Block& block, int previous_write);
-  bool DetectActiveRender(ArrayView<const float> x) const;
+  bool DetectActiveRender(std::span<const float> x) const;
   bool DetectExcessRenderBlocks();
   void IncrementWriteIndices();
   void IncrementLowRateReadIndices();
@@ -405,7 +405,7 @@ void RenderDelayBufferImpl::InsertBlock(const Block& block,
   if (render_linear_amplitude_gain_ != 1.f) {
     for (size_t band = 0; band < num_bands; ++band) {
       for (size_t ch = 0; ch < num_render_channels; ++ch) {
-        ArrayView<float, kBlockSize> b_view = b.buffer[b.write].View(band, ch);
+        std::span<float, kBlockSize> b_view = b.buffer[b.write].View(band, ch);
         for (float& sample : b_view) {
           sample *= render_linear_amplitude_gain_;
         }
@@ -428,7 +428,7 @@ void RenderDelayBufferImpl::InsertBlock(const Block& block,
   }
 }
 
-bool RenderDelayBufferImpl::DetectActiveRender(ArrayView<const float> x) const {
+bool RenderDelayBufferImpl::DetectActiveRender(std::span<const float> x) const {
   const float x_energy = std::inner_product(x.begin(), x.end(), x.begin(), 0.f);
   return x_energy > (config_.render_levels.active_render_limit *
                      config_.render_levels.active_render_limit) *
diff --git a/modules/audio_processing/aec3/render_signal_analyzer.cc b/modules/audio_processing/aec3/render_signal_analyzer.cc
index ae013e7189..9efb4bc2a4 100644
--- a/modules/audio_processing/aec3/render_signal_analyzer.cc
+++ b/modules/audio_processing/aec3/render_signal_analyzer.cc
@@ -15,9 +15,9 @@
 #include <cmath>
 #include <cstddef>
 #include <optional>
+#include <span>
 #include <utility>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/block.h"
@@ -43,7 +43,7 @@ void IdentifySmallNarrowBandRegions(
 
   std::array<size_t, kFftLengthBy2 - 1> channel_counters;
   channel_counters.fill(0);
-  ArrayView<const std::array<float, kFftLengthBy2Plus1>> X2 =
+  std::span<const std::array<float, kFftLengthBy2Plus1>> X2 =
       render_buffer.Spectrum(*delay_partitions);
   for (size_t ch = 0; ch < X2.size(); ++ch) {
     for (size_t k = 1; k < kFftLengthBy2; ++k) {
@@ -74,7 +74,7 @@ void IdentifyStrongNarrowBandComponent(const RenderBuffer& render_buffer,
   const Block& x_latest = render_buffer.GetBlock(0);
   float max_peak_level = 0.f;
   for (int channel = 0; channel < x_latest.NumChannels(); ++channel) {
-    ArrayView<const float, kFftLengthBy2Plus1> X2_latest =
+    std::span<const float, kFftLengthBy2Plus1> X2_latest =
         render_buffer.Spectrum(0)[channel];
 
     // Identify the spectral peak.
diff --git a/modules/audio_processing/aec3/render_signal_analyzer_unittest.cc b/modules/audio_processing/aec3/render_signal_analyzer_unittest.cc
index 9cb8ebe005..7a14adb75a 100644
--- a/modules/audio_processing/aec3/render_signal_analyzer_unittest.cc
+++ b/modules/audio_processing/aec3/render_signal_analyzer_unittest.cc
@@ -19,7 +19,6 @@
 #include <optional>
 #include <string>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/aec3_fft.h"
diff --git a/modules/audio_processing/aec3/residual_echo_estimator.cc b/modules/audio_processing/aec3/residual_echo_estimator.cc
index 1c4c3a0137..65793d46c1 100644
--- a/modules/audio_processing/aec3/residual_echo_estimator.cc
+++ b/modules/audio_processing/aec3/residual_echo_estimator.cc
@@ -14,9 +14,9 @@
 #include <array>
 #include <cstddef>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "api/audio/neural_residual_echo_estimator.h"
 #include "api/environment/environment.h"
@@ -89,9 +89,9 @@ void GetRenderIndexesToAnalyze(
 // Estimates the residual echo power based on the echo return loss enhancement
 // (ERLE) and the linear power estimate.
 void LinearEstimate(
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> S2_linear,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> erle,
-    ArrayView<std::array<float, kFftLengthBy2Plus1>> R2) {
+    std::span<const std::array<float, kFftLengthBy2Plus1>> S2_linear,
+    std::span<const std::array<float, kFftLengthBy2Plus1>> erle,
+    std::span<std::array<float, kFftLengthBy2Plus1>> R2) {
   RTC_DCHECK_EQ(S2_linear.size(), erle.size());
   RTC_DCHECK_EQ(S2_linear.size(), R2.size());
 
@@ -108,7 +108,7 @@ void LinearEstimate(
 // gain.
 void NonLinearEstimate(float echo_path_gain,
                        const std::array<float, kFftLengthBy2Plus1>& X2,
-                       ArrayView<std::array<float, kFftLengthBy2Plus1>> R2) {
+                       std::span<std::array<float, kFftLengthBy2Plus1>> R2) {
   const size_t num_capture_channels = R2.size();
   for (size_t ch = 0; ch < num_capture_channels; ++ch) {
     for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) {
@@ -119,7 +119,7 @@ void NonLinearEstimate(float echo_path_gain,
 
 // Applies a soft noise gate to the echo generating power.
 void ApplyNoiseGate(const EchoCanceller3Config::EchoModel& config,
-                    ArrayView<float, kFftLengthBy2Plus1> X2) {
+                    std::span<float, kFftLengthBy2Plus1> X2) {
   for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) {
     if (config.noise_gate_power > X2[k]) {
       X2[k] = std::max(0.f, X2[k] - config.noise_gate_slope *
@@ -134,7 +134,7 @@ void EchoGeneratingPower(size_t num_render_channels,
                          const SpectrumBuffer& spectrum_buffer,
                          const EchoCanceller3Config::EchoModel& echo_model,
                          int filter_delay_blocks,
-                         ArrayView<float, kFftLengthBy2Plus1> X2) {
+                         std::span<float, kFftLengthBy2Plus1> X2) {
   int idx_stop;
   int idx_start;
   GetRenderIndexesToAnalyze(spectrum_buffer, echo_model, filter_delay_blocks,
@@ -193,14 +193,14 @@ ResidualEchoEstimator::~ResidualEchoEstimator() = default;
 void ResidualEchoEstimator::Estimate(
     const AecState& aec_state,
     const RenderBuffer& render_buffer,
-    ArrayView<const std::array<float, kFftLengthBy2>> capture,
-    ArrayView<const std::array<float, kFftLengthBy2>> linear_aec_output,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> S2_linear,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> Y2,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> E2,
+    std::span<const std::array<float, kFftLengthBy2>> capture,
+    std::span<const std::array<float, kFftLengthBy2>> linear_aec_output,
+    std::span<const std::array<float, kFftLengthBy2Plus1>> S2_linear,
+    std::span<const std::array<float, kFftLengthBy2Plus1>> Y2,
+    std::span<const std::array<float, kFftLengthBy2Plus1>> E2,
     bool dominant_nearend,
-    ArrayView<std::array<float, kFftLengthBy2Plus1>> R2,
-    ArrayView<std::array<float, kFftLengthBy2Plus1>> R2_unbounded) {
+    std::span<std::array<float, kFftLengthBy2Plus1>> R2,
+    std::span<std::array<float, kFftLengthBy2Plus1>> R2_unbounded) {
   RTC_DCHECK_EQ(R2.size(), Y2.size());
   RTC_DCHECK_EQ(R2.size(), S2_linear.size());
 
@@ -323,9 +323,9 @@ void ResidualEchoEstimator::Reset() {
 void ResidualEchoEstimator::UpdateRenderNoisePower(
     const RenderBuffer& render_buffer) {
   std::array<float, kFftLengthBy2Plus1> render_power_data;
-  ArrayView<const std::array<float, kFftLengthBy2Plus1>> X2 =
+  std::span<const std::array<float, kFftLengthBy2Plus1>> X2 =
       render_buffer.Spectrum(0);
-  ArrayView<const float, kFftLengthBy2Plus1> render_power = X2[/*channel=*/0];
+  std::span<const float, kFftLengthBy2Plus1> render_power = X2[/*channel=*/0];
   if (num_render_channels_ > 1) {
     render_power_data.fill(0.f);
     for (size_t ch = 0; ch < num_render_channels_; ++ch) {
@@ -369,9 +369,9 @@ void ResidualEchoEstimator::UpdateReverb(ReverbType reverb_type,
 
   // Compute render power for the reverb.
   std::array<float, kFftLengthBy2Plus1> render_power_data;
-  ArrayView<const std::array<float, kFftLengthBy2Plus1>> X2 =
+  std::span<const std::array<float, kFftLengthBy2Plus1>> X2 =
       render_buffer.Spectrum(first_reverb_partition);
-  ArrayView<const float, kFftLengthBy2Plus1> render_power = X2[/*channel=*/0];
+  std::span<const float, kFftLengthBy2Plus1> render_power = X2[/*channel=*/0];
   if (num_render_channels_ > 1) {
     render_power_data.fill(0.f);
     for (size_t ch = 0; ch < num_render_channels_; ++ch) {
@@ -397,11 +397,11 @@ void ResidualEchoEstimator::UpdateReverb(ReverbType reverb_type,
 }
 // Adds the estimated power of the reverb to the residual echo power.
 void ResidualEchoEstimator::AddReverb(
-    ArrayView<std::array<float, kFftLengthBy2Plus1>> R2) const {
+    std::span<std::array<float, kFftLengthBy2Plus1>> R2) const {
   const size_t num_capture_channels = R2.size();
 
   // Add the reverb power.
-  ArrayView<const float, kFftLengthBy2Plus1> reverb_power =
+  std::span<const float, kFftLengthBy2Plus1> reverb_power =
       echo_reverb_.reverb();
   for (size_t ch = 0; ch < num_capture_channels; ++ch) {
     for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) {
diff --git a/modules/audio_processing/aec3/residual_echo_estimator.h b/modules/audio_processing/aec3/residual_echo_estimator.h
index 22358bca2a..fe1a4f868d 100644
--- a/modules/audio_processing/aec3/residual_echo_estimator.h
+++ b/modules/audio_processing/aec3/residual_echo_estimator.h
@@ -13,8 +13,8 @@
 
 #include <array>
 #include <cstddef>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "api/audio/neural_residual_echo_estimator.h"
 #include "api/environment/environment.h"
@@ -40,14 +40,14 @@ class ResidualEchoEstimator {
   void Estimate(
       const AecState& aec_state,
       const RenderBuffer& render_buffer,
-      ArrayView<const std::array<float, kFftLengthBy2>> capture,
-      ArrayView<const std::array<float, kFftLengthBy2>> linear_aec_output,
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>> S2_linear,
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>> Y2,
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>> E2,
+      std::span<const std::array<float, kFftLengthBy2>> capture,
+      std::span<const std::array<float, kFftLengthBy2>> linear_aec_output,
+      std::span<const std::array<float, kFftLengthBy2Plus1>> S2_linear,
+      std::span<const std::array<float, kFftLengthBy2Plus1>> Y2,
+      std::span<const std::array<float, kFftLengthBy2Plus1>> E2,
       bool dominant_nearend,
-      ArrayView<std::array<float, kFftLengthBy2Plus1>> R2,
-      ArrayView<std::array<float, kFftLengthBy2Plus1>> R2_unbounded);
+      std::span<std::array<float, kFftLengthBy2Plus1>> R2,
+      std::span<std::array<float, kFftLengthBy2Plus1>> R2_unbounded);
 
  private:
   enum class ReverbType { kLinear, kNonLinear };
@@ -67,7 +67,7 @@ class ResidualEchoEstimator {
 
   // Adds the estimated unmodelled echo power to the residual echo power
   // estimate.
-  void AddReverb(ArrayView<std::array<float, kFftLengthBy2Plus1>> R2) const;
+  void AddReverb(std::span<std::array<float, kFftLengthBy2Plus1>> R2) const;
 
   // Gets the echo path gain to apply.
   float GetEchoPathGain(const AecState& aec_state,
diff --git a/modules/audio_processing/aec3/residual_echo_estimator_unittest.cc b/modules/audio_processing/aec3/residual_echo_estimator_unittest.cc
index 05c8c58b3a..a3351c831e 100644
--- a/modules/audio_processing/aec3/residual_echo_estimator_unittest.cc
+++ b/modules/audio_processing/aec3/residual_echo_estimator_unittest.cc
@@ -16,10 +16,10 @@
 #include <memory>
 #include <numeric>
 #include <optional>
+#include <span>
 #include <tuple>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "api/environment/environment.h"
 #include "api/environment/environment_factory.h"
@@ -122,7 +122,7 @@ class ResidualEchoEstimatorTest {
                         R2_, R2_unbounded_);
   }
 
-  ArrayView<const std::array<float, kFftLengthBy2Plus1>> R2() const {
+  std::span<const std::array<float, kFftLengthBy2Plus1>> R2() const {
     return R2_;
   }
 
diff --git a/modules/audio_processing/aec3/reverb_decay_estimator.cc b/modules/audio_processing/aec3/reverb_decay_estimator.cc
index dbedfc79e0..b010629ece 100644
--- a/modules/audio_processing/aec3/reverb_decay_estimator.cc
+++ b/modules/audio_processing/aec3/reverb_decay_estimator.cc
@@ -16,8 +16,8 @@
 #include <cstddef>
 #include <numeric>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/logging/apm_data_dumper.h"
@@ -34,7 +34,7 @@ constexpr float kEarlyReverbFirstPointAtLinearRegressors =
     -0.5f * kBlocksPerSection * kFftLengthBy2 + 0.5f;
 
 // Averages the values in a block of size kFftLengthBy2;
-float BlockAverage(ArrayView<const float> v, size_t block_index) {
+float BlockAverage(std::span<const float> v, size_t block_index) {
   constexpr float kOneByFftLengthBy2 = 1.f / kFftLengthBy2;
   const int i = block_index * kFftLengthBy2;
   RTC_DCHECK_GE(v.size(), i + kFftLengthBy2);
@@ -62,7 +62,7 @@ constexpr float SymmetricArithmetricSum(int N) {
 }
 
 // Returns the peak energy of an impulse response.
-float BlockEnergyPeak(ArrayView<const float> h, int peak_block) {
+float BlockEnergyPeak(std::span<const float> h, int peak_block) {
   RTC_DCHECK_LE((peak_block + 1) * kFftLengthBy2, h.size());
   RTC_DCHECK_GE(peak_block, 0);
   float peak_value =
@@ -73,7 +73,7 @@ float BlockEnergyPeak(ArrayView<const float> h, int peak_block) {
 }
 
 // Returns the average energy of an impulse response block.
-float BlockEnergyAverage(ArrayView<const float> h, int block_index) {
+float BlockEnergyAverage(std::span<const float> h, int block_index) {
   RTC_DCHECK_LE((block_index + 1) * kFftLengthBy2, h.size());
   RTC_DCHECK_GE(block_index, 0);
   constexpr float kOneByFftLengthBy2 = 1.f / kFftLengthBy2;
@@ -103,7 +103,7 @@ ReverbDecayEstimator::ReverbDecayEstimator(const EchoCanceller3Config& config)
 
 ReverbDecayEstimator::~ReverbDecayEstimator() = default;
 
-void ReverbDecayEstimator::Update(ArrayView<const float> filter,
+void ReverbDecayEstimator::Update(std::span<const float> filter,
                                   const std::optional<float>& filter_quality,
                                   int filter_delay_blocks,
                                   bool usable_linear_filter,
@@ -159,7 +159,7 @@ void ReverbDecayEstimator::ResetDecayEstimation() {
   late_reverb_end_ = 0;
 }
 
-void ReverbDecayEstimator::EstimateDecay(ArrayView<const float> filter,
+void ReverbDecayEstimator::EstimateDecay(std::span<const float> filter,
                                          int peak_block) {
   auto& h = filter;
   RTC_DCHECK_EQ(0, h.size() % kFftLengthBy2);
@@ -223,7 +223,7 @@ void ReverbDecayEstimator::EstimateDecay(ArrayView<const float> filter,
   early_reverb_estimator_.Reset();
 }
 
-void ReverbDecayEstimator::AnalyzeFilter(ArrayView<const float> filter) {
+void ReverbDecayEstimator::AnalyzeFilter(std::span<const float> filter) {
   auto h = filter.subspan(block_to_analyze_ * kFftLengthBy2, kFftLengthBy2);
 
   // Compute squared filter coeffiecients for the block to analyze_;
diff --git a/modules/audio_processing/aec3/reverb_decay_estimator.h b/modules/audio_processing/aec3/reverb_decay_estimator.h
index 721cd2bbec..08fd6ca900 100644
--- a/modules/audio_processing/aec3/reverb_decay_estimator.h
+++ b/modules/audio_processing/aec3/reverb_decay_estimator.h
@@ -12,10 +12,9 @@
 #define MODULES_AUDIO_PROCESSING_AEC3_REVERB_DECAY_ESTIMATOR_H_
 
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
-
 namespace webrtc {
 
 class ApmDataDumper;
@@ -27,7 +26,7 @@ class ReverbDecayEstimator {
   explicit ReverbDecayEstimator(const EchoCanceller3Config& config);
   ~ReverbDecayEstimator();
   // Updates the decay estimate.
-  void Update(ArrayView<const float> filter,
+  void Update(std::span<const float> filter,
               const std::optional<float>& filter_quality,
               int filter_delay_blocks,
               bool usable_linear_filter,
@@ -45,8 +44,8 @@ class ReverbDecayEstimator {
   void Dump(ApmDataDumper* data_dumper) const;
 
  private:
-  void EstimateDecay(ArrayView<const float> filter, int peak_block);
-  void AnalyzeFilter(ArrayView<const float> filter);
+  void EstimateDecay(std::span<const float> filter, int peak_block);
+  void AnalyzeFilter(std::span<const float> filter);
 
   void ResetDecayEstimation();
 
diff --git a/modules/audio_processing/aec3/reverb_frequency_response.cc b/modules/audio_processing/aec3/reverb_frequency_response.cc
index c375ad1820..574eefc4c9 100644
--- a/modules/audio_processing/aec3/reverb_frequency_response.cc
+++ b/modules/audio_processing/aec3/reverb_frequency_response.cc
@@ -15,9 +15,9 @@
 #include <cstddef>
 #include <numeric>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "rtc_base/checks.h"
 
@@ -28,8 +28,8 @@ namespace {
 // Computes the ratio of the energies between the direct path and the tail. The
 // energy is computed in the power spectrum domain discarding the DC
 // contributions.
-float AverageDecayWithinFilter(ArrayView<const float> freq_resp_direct_path,
-                               ArrayView<const float> freq_resp_tail) {
+float AverageDecayWithinFilter(std::span<const float> freq_resp_direct_path,
+                               std::span<const float> freq_resp_tail) {
   // Skipping the DC for the ratio computation
   constexpr size_t kSkipBins = 1;
   RTC_CHECK_EQ(freq_resp_direct_path.size(), freq_resp_tail.size());
@@ -76,10 +76,10 @@ void ReverbFrequencyResponse::Update(
         frequency_response,
     int filter_delay_blocks,
     float linear_filter_quality) {
-  ArrayView<const float> freq_resp_tail(
+  std::span<const float> freq_resp_tail(
       frequency_response[frequency_response.size() - 1]);
 
-  ArrayView<const float> freq_resp_direct_path(
+  std::span<const float> freq_resp_direct_path(
       frequency_response[filter_delay_blocks]);
 
   float average_decay =
diff --git a/modules/audio_processing/aec3/reverb_frequency_response.h b/modules/audio_processing/aec3/reverb_frequency_response.h
index 3fecba7396..ee36bce0e0 100644
--- a/modules/audio_processing/aec3/reverb_frequency_response.h
+++ b/modules/audio_processing/aec3/reverb_frequency_response.h
@@ -13,9 +13,9 @@
 
 #include <array>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 
 namespace webrtc {
@@ -35,7 +35,7 @@ class ReverbFrequencyResponse {
               bool stationary_block);
 
   // Returns the estimated frequency response for the reverb.
-  ArrayView<const float> FrequencyResponse() const { return tail_response_; }
+  std::span<const float> FrequencyResponse() const { return tail_response_; }
 
  private:
   void Update(const std::vector<std::array<float, kFftLengthBy2Plus1>>&
diff --git a/modules/audio_processing/aec3/reverb_model.cc b/modules/audio_processing/aec3/reverb_model.cc
index 7bd719c76d..63c30cbaa6 100644
--- a/modules/audio_processing/aec3/reverb_model.cc
+++ b/modules/audio_processing/aec3/reverb_model.cc
@@ -11,8 +11,7 @@
 #include "modules/audio_processing/aec3/reverb_model.h"
 
 #include <cstddef>
-
-#include "api/array_view.h"
+#include <span>
 
 namespace webrtc {
 
@@ -27,7 +26,7 @@ void ReverbModel::Reset() {
 }
 
 void ReverbModel::UpdateReverbNoFreqShaping(
-    ArrayView<const float> power_spectrum,
+    std::span<const float> power_spectrum,
     float power_spectrum_scaling,
     float reverb_decay) {
   if (reverb_decay > 0) {
@@ -39,8 +38,8 @@ void ReverbModel::UpdateReverbNoFreqShaping(
   }
 }
 
-void ReverbModel::UpdateReverb(ArrayView<const float> power_spectrum,
-                               ArrayView<const float> power_spectrum_scaling,
+void ReverbModel::UpdateReverb(std::span<const float> power_spectrum,
+                               std::span<const float> power_spectrum_scaling,
                                float reverb_decay) {
   if (reverb_decay > 0) {
     // Update the estimate of the reverberant power.
diff --git a/modules/audio_processing/aec3/reverb_model.h b/modules/audio_processing/aec3/reverb_model.h
index 15e24895e6..a05f38016f 100644
--- a/modules/audio_processing/aec3/reverb_model.h
+++ b/modules/audio_processing/aec3/reverb_model.h
@@ -12,8 +12,8 @@
 #define MODULES_AUDIO_PROCESSING_AEC3_REVERB_MODEL_H_
 
 #include <array>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 
 namespace webrtc {
@@ -29,7 +29,7 @@ class ReverbModel {
   void Reset();
 
   // Returns the reverb.
-  ArrayView<const float, kFftLengthBy2Plus1> reverb() const { return reverb_; }
+  std::span<const float, kFftLengthBy2Plus1> reverb() const { return reverb_; }
 
   // The methods UpdateReverbNoFreqShaping and UpdateReverb update the
   // estimate of the reverberation contribution to an input/output power
@@ -37,13 +37,13 @@ class ReverbModel {
   // power spectrum is pre-scaled. Use the method UpdateReverb when a different
   // scaling should be applied per frequency and UpdateReverb_no_freq_shape if
   // the same scaling should be used for all the frequencies.
-  void UpdateReverbNoFreqShaping(ArrayView<const float> power_spectrum,
+  void UpdateReverbNoFreqShaping(std::span<const float> power_spectrum,
                                  float power_spectrum_scaling,
                                  float reverb_decay);
 
   // Update the reverb based on new data.
-  void UpdateReverb(ArrayView<const float> power_spectrum,
-                    ArrayView<const float> power_spectrum_scaling,
+  void UpdateReverb(std::span<const float> power_spectrum,
+                    std::span<const float> power_spectrum_scaling,
                     float reverb_decay);
 
  private:
diff --git a/modules/audio_processing/aec3/reverb_model_estimator.cc b/modules/audio_processing/aec3/reverb_model_estimator.cc
index 6465299909..afbd110e6d 100644
--- a/modules/audio_processing/aec3/reverb_model_estimator.cc
+++ b/modules/audio_processing/aec3/reverb_model_estimator.cc
@@ -14,9 +14,9 @@
 #include <cstddef>
 #include <memory>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/reverb_decay_estimator.h"
@@ -41,11 +41,11 @@ ReverbModelEstimator::ReverbModelEstimator(const EchoCanceller3Config& config,
 ReverbModelEstimator::~ReverbModelEstimator() = default;
 
 void ReverbModelEstimator::Update(
-    ArrayView<const std::vector<float>> impulse_responses,
-    ArrayView<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
+    std::span<const std::vector<float>> impulse_responses,
+    std::span<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
         frequency_responses,
-    ArrayView<const std::optional<float>> linear_filter_qualities,
-    ArrayView<const int> filter_delays_blocks,
+    std::span<const std::optional<float>> linear_filter_qualities,
+    std::span<const int> filter_delays_blocks,
     const std::vector<bool>& usable_linear_estimates,
     bool stationary_block) {
   const size_t num_capture_channels = reverb_decay_estimators_.size();
diff --git a/modules/audio_processing/aec3/reverb_model_estimator.h b/modules/audio_processing/aec3/reverb_model_estimator.h
index 08bbc86447..50bd97e066 100644
--- a/modules/audio_processing/aec3/reverb_model_estimator.h
+++ b/modules/audio_processing/aec3/reverb_model_estimator.h
@@ -15,9 +15,9 @@
 #include <cstddef>
 #include <memory>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "modules/audio_processing/aec3/aec3_common.h"  // kFftLengthBy2Plus1
 #include "modules/audio_processing/aec3/reverb_decay_estimator.h"
@@ -36,11 +36,11 @@ class ReverbModelEstimator {
 
   // Updates the estimates based on new data.
   void Update(
-      ArrayView<const std::vector<float>> impulse_responses,
-      ArrayView<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
+      std::span<const std::vector<float>> impulse_responses,
+      std::span<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
           frequency_responses,
-      ArrayView<const std::optional<float>> linear_filter_qualities,
-      ArrayView<const int> filter_delays_blocks,
+      std::span<const std::optional<float>> linear_filter_qualities,
+      std::span<const int> filter_delays_blocks,
       const std::vector<bool>& usable_linear_estimates,
       bool stationary_block);
 
@@ -54,7 +54,7 @@ class ReverbModelEstimator {
 
   // Return the frequency response of the reverberant echo.
   // TODO(peah): Correct to properly support multiple channels.
-  ArrayView<const float> GetReverbFrequencyResponse() const {
+  std::span<const float> GetReverbFrequencyResponse() const {
     return reverb_frequency_responses_[0].FrequencyResponse();
   }
 
diff --git a/modules/audio_processing/aec3/reverb_model_estimator_unittest.cc b/modules/audio_processing/aec3/reverb_model_estimator_unittest.cc
index e3cde9e4e1..8f5be5a74b 100644
--- a/modules/audio_processing/aec3/reverb_model_estimator_unittest.cc
+++ b/modules/audio_processing/aec3/reverb_model_estimator_unittest.cc
@@ -16,9 +16,9 @@
 #include <cstddef>
 #include <numeric>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/aec3_fft.h"
@@ -110,7 +110,7 @@ void ReverbModelEstimatorTest::CreateImpulseResponseWithDecay() {
       H_j.Spectrum(Aec3Optimization::kNone, H2_[ch][j]);
     }
   }
-  ArrayView<float> H2_tail(H2_[0][H2_[0].size() - 1]);
+  std::span<float> H2_tail(H2_[0][H2_[0].size() - 1]);
   true_power_tail_ = std::accumulate(H2_tail.begin(), H2_tail.end(), 0.f);
 }
 void ReverbModelEstimatorTest::RunEstimator() {
diff --git a/modules/audio_processing/aec3/signal_dependent_erle_estimator.cc b/modules/audio_processing/aec3/signal_dependent_erle_estimator.cc
index d0a854c57b..571b4823de 100644
--- a/modules/audio_processing/aec3/signal_dependent_erle_estimator.cc
+++ b/modules/audio_processing/aec3/signal_dependent_erle_estimator.cc
@@ -16,9 +16,9 @@
 #include <functional>
 #include <memory>
 #include <numeric>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/render_buffer.h"
@@ -187,13 +187,13 @@ void SignalDependentErleEstimator::Reset() {
 // correction factor to the erle that is given as an input to this method.
 void SignalDependentErleEstimator::Update(
     const RenderBuffer& render_buffer,
-    ArrayView<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
+    std::span<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
         filter_frequency_responses,
-    ArrayView<const float, kFftLengthBy2Plus1> X2,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> Y2,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> E2,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> average_erle,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+    std::span<const float, kFftLengthBy2Plus1> X2,
+    std::span<const std::array<float, kFftLengthBy2Plus1>> Y2,
+    std::span<const std::array<float, kFftLengthBy2Plus1>> E2,
+    std::span<const std::array<float, kFftLengthBy2Plus1>> average_erle,
+    std::span<const std::array<float, kFftLengthBy2Plus1>>
         average_erle_onset_compensated,
     const std::vector<bool>& converged_filters) {
   RTC_DCHECK_GT(num_sections_, 1);
@@ -241,7 +241,7 @@ void SignalDependentErleEstimator::Dump(
 // together constitute 90% of the estimated echo energy.
 void SignalDependentErleEstimator::ComputeNumberOfActiveFilterSections(
     const RenderBuffer& render_buffer,
-    ArrayView<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
+    std::span<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
         filter_frequency_responses) {
   RTC_DCHECK_GT(num_sections_, 1);
   // Computes an approximation of the power spectrum if the filter would have
@@ -254,17 +254,17 @@ void SignalDependentErleEstimator::ComputeNumberOfActiveFilterSections(
 }
 
 void SignalDependentErleEstimator::UpdateCorrectionFactors(
-    ArrayView<const float, kFftLengthBy2Plus1> X2,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> Y2,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> E2,
+    std::span<const float, kFftLengthBy2Plus1> X2,
+    std::span<const std::array<float, kFftLengthBy2Plus1>> Y2,
+    std::span<const std::array<float, kFftLengthBy2Plus1>> E2,
     const std::vector<bool>& converged_filters) {
   for (size_t ch = 0; ch < converged_filters.size(); ++ch) {
     if (converged_filters[ch]) {
       constexpr float kX2BandEnergyThreshold = 44015068.0f;
       constexpr float kSmthConstantDecreases = 0.1f;
       constexpr float kSmthConstantIncreases = kSmthConstantDecreases / 2.f;
-      auto subband_powers = [](ArrayView<const float> power_spectrum,
-                               ArrayView<float> power_spectrum_subbands) {
+      auto subband_powers = [](std::span<const float> power_spectrum,
+                               std::span<float> power_spectrum_subbands) {
         for (size_t subband = 0; subband < kSubbands; ++subband) {
           RTC_DCHECK_LE(kBandBoundaries[subband + 1], power_spectrum.size());
           power_spectrum_subbands[subband] = std::accumulate(
@@ -354,7 +354,7 @@ void SignalDependentErleEstimator::UpdateCorrectionFactors(
 
 void SignalDependentErleEstimator::ComputeEchoEstimatePerFilterSection(
     const RenderBuffer& render_buffer,
-    ArrayView<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
+    std::span<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
         filter_frequency_responses) {
   const SpectrumBuffer& spectrum_render_buffer =
       render_buffer.GetSpectrumBuffer();
diff --git a/modules/audio_processing/aec3/signal_dependent_erle_estimator.h b/modules/audio_processing/aec3/signal_dependent_erle_estimator.h
index 4c58bd4f4f..992057d78f 100644
--- a/modules/audio_processing/aec3/signal_dependent_erle_estimator.h
+++ b/modules/audio_processing/aec3/signal_dependent_erle_estimator.h
@@ -14,9 +14,9 @@
 #include <array>
 #include <cstddef>
 #include <memory>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/render_buffer.h"
@@ -39,7 +39,7 @@ class SignalDependentErleEstimator {
   void Reset();
 
   // Returns the Erle per frequency subband.
-  ArrayView<const std::array<float, kFftLengthBy2Plus1>> Erle(
+  std::span<const std::array<float, kFftLengthBy2Plus1>> Erle(
       bool onset_compensated) const {
     return onset_compensated && use_onset_detection_ ? erle_onset_compensated_
                                                      : erle_;
@@ -49,13 +49,13 @@ class SignalDependentErleEstimator {
   // to be an estimation of the average Erle achieved by the linear filter.
   void Update(
       const RenderBuffer& render_buffer,
-      ArrayView<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
+      std::span<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
           filter_frequency_response,
-      ArrayView<const float, kFftLengthBy2Plus1> X2,
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>> Y2,
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>> E2,
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>> average_erle,
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+      std::span<const float, kFftLengthBy2Plus1> X2,
+      std::span<const std::array<float, kFftLengthBy2Plus1>> Y2,
+      std::span<const std::array<float, kFftLengthBy2Plus1>> E2,
+      std::span<const std::array<float, kFftLengthBy2Plus1>> average_erle,
+      std::span<const std::array<float, kFftLengthBy2Plus1>>
           average_erle_onset_compensated,
       const std::vector<bool>& converged_filters);
 
@@ -66,18 +66,18 @@ class SignalDependentErleEstimator {
  private:
   void ComputeNumberOfActiveFilterSections(
       const RenderBuffer& render_buffer,
-      ArrayView<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
+      std::span<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
           filter_frequency_responses);
 
   void UpdateCorrectionFactors(
-      ArrayView<const float, kFftLengthBy2Plus1> X2,
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>> Y2,
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>> E2,
+      std::span<const float, kFftLengthBy2Plus1> X2,
+      std::span<const std::array<float, kFftLengthBy2Plus1>> Y2,
+      std::span<const std::array<float, kFftLengthBy2Plus1>> E2,
       const std::vector<bool>& converged_filters);
 
   void ComputeEchoEstimatePerFilterSection(
       const RenderBuffer& render_buffer,
-      ArrayView<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
+      std::span<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
           filter_frequency_responses);
 
   void ComputeActiveFilterSections();
diff --git a/modules/audio_processing/aec3/signal_dependent_erle_estimator_unittest.cc b/modules/audio_processing/aec3/signal_dependent_erle_estimator_unittest.cc
index c834b39e94..611b25ca07 100644
--- a/modules/audio_processing/aec3/signal_dependent_erle_estimator_unittest.cc
+++ b/modules/audio_processing/aec3/signal_dependent_erle_estimator_unittest.cc
@@ -14,10 +14,10 @@
 #include <array>
 #include <cstddef>
 #include <memory>
+#include <span>
 #include <tuple>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/block.h"
@@ -56,14 +56,14 @@ class TestInputs {
              size_t num_capture_channels);
   ~TestInputs();
   const RenderBuffer& GetRenderBuffer() { return *render_buffer_; }
-  ArrayView<const float, kFftLengthBy2Plus1> GetX2() { return X2_; }
-  ArrayView<const std::array<float, kFftLengthBy2Plus1>> GetY2() const {
+  std::span<const float, kFftLengthBy2Plus1> GetX2() { return X2_; }
+  std::span<const std::array<float, kFftLengthBy2Plus1>> GetY2() const {
     return Y2_;
   }
-  ArrayView<const std::array<float, kFftLengthBy2Plus1>> GetE2() const {
+  std::span<const std::array<float, kFftLengthBy2Plus1>> GetE2() const {
     return E2_;
   }
-  ArrayView<const std::vector<std::array<float, kFftLengthBy2Plus1>>> GetH2()
+  std::span<const std::vector<std::array<float, kFftLengthBy2Plus1>>> GetH2()
       const {
     return H2_;
   }
diff --git a/modules/audio_processing/aec3/stationarity_estimator.cc b/modules/audio_processing/aec3/stationarity_estimator.cc
index cba546bdd1..7b3f8f8c09 100644
--- a/modules/audio_processing/aec3/stationarity_estimator.cc
+++ b/modules/audio_processing/aec3/stationarity_estimator.cc
@@ -14,8 +14,8 @@
 #include <array>
 #include <atomic>
 #include <cstddef>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/spectrum_buffer.h"
 #include "modules/audio_processing/logging/apm_data_dumper.h"
@@ -45,7 +45,7 @@ void StationarityEstimator::Reset() {
 
 // Update just the noise estimator. Usefull until the delay is known
 void StationarityEstimator::UpdateNoiseEstimator(
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> spectrum) {
+    std::span<const std::array<float, kFftLengthBy2Plus1>> spectrum) {
   noise_.Update(spectrum);
   data_dumper_->DumpRaw("aec3_stationarity_noise_spectrum", noise_.Spectrum());
   data_dumper_->DumpRaw("aec3_stationarity_is_block_stationary",
@@ -54,7 +54,7 @@ void StationarityEstimator::UpdateNoiseEstimator(
 
 void StationarityEstimator::UpdateStationarityFlags(
     const SpectrumBuffer& spectrum_buffer,
-    ArrayView<const float> render_reverb_contribution_spectrum,
+    std::span<const float> render_reverb_contribution_spectrum,
     int idx_current,
     int num_lookahead) {
   std::array<int, kWindowLength> indexes;
@@ -99,7 +99,7 @@ bool StationarityEstimator::IsBlockStationary() const {
 
 bool StationarityEstimator::EstimateBandStationarity(
     const SpectrumBuffer& spectrum_buffer,
-    ArrayView<const float> average_reverb,
+    std::span<const float> average_reverb,
     const std::array<int, kWindowLength>& indexes,
     size_t band) const {
   constexpr float kThrStationarity = 10.f;
@@ -168,12 +168,12 @@ void StationarityEstimator::NoiseSpectrum::Reset() {
 }
 
 void StationarityEstimator::NoiseSpectrum::Update(
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> spectrum) {
+    std::span<const std::array<float, kFftLengthBy2Plus1>> spectrum) {
   RTC_DCHECK_LE(1, spectrum[0].size());
   const int num_render_channels = static_cast<int>(spectrum.size());
 
   std::array<float, kFftLengthBy2Plus1> avg_spectrum_data;
-  ArrayView<const float> avg_spectrum;
+  std::span<const float> avg_spectrum;
   if (num_render_channels == 1) {
     avg_spectrum = spectrum[0];
   } else {
diff --git a/modules/audio_processing/aec3/stationarity_estimator.h b/modules/audio_processing/aec3/stationarity_estimator.h
index 0c10a099b0..e0a66cac0b 100644
--- a/modules/audio_processing/aec3/stationarity_estimator.h
+++ b/modules/audio_processing/aec3/stationarity_estimator.h
@@ -15,8 +15,8 @@
 #include <atomic>
 #include <cstddef>
 #include <memory>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/aec3_common.h"  // kFftLengthBy2Plus1...
 #include "rtc_base/checks.h"
 
@@ -35,13 +35,13 @@ class StationarityEstimator {
 
   // Update just the noise estimator. Usefull until the delay is known
   void UpdateNoiseEstimator(
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>> spectrum);
+      std::span<const std::array<float, kFftLengthBy2Plus1>> spectrum);
 
   // Update the flag indicating whether this current frame is stationary. For
   // getting a more robust estimation, it looks at future and/or past frames.
   void UpdateStationarityFlags(
       const SpectrumBuffer& spectrum_buffer,
-      ArrayView<const float> render_reverb_contribution_spectrum,
+      std::span<const float> render_reverb_contribution_spectrum,
       int idx_current,
       int num_lookahead);
 
@@ -61,7 +61,7 @@ class StationarityEstimator {
   // Get an estimation of the stationarity for the current band by looking
   // at the past/present/future available data.
   bool EstimateBandStationarity(const SpectrumBuffer& spectrum_buffer,
-                                ArrayView<const float> average_reverb,
+                                std::span<const float> average_reverb,
                                 const std::array<int, kWindowLength>& indexes,
                                 size_t band) const;
 
@@ -86,10 +86,10 @@ class StationarityEstimator {
 
     // Update the noise power spectrum with a new frame.
     void Update(
-        ArrayView<const std::array<float, kFftLengthBy2Plus1>> spectrum);
+        std::span<const std::array<float, kFftLengthBy2Plus1>> spectrum);
 
     // Get the noise estimation power spectrum.
-    ArrayView<const float> Spectrum() const { return noise_spectrum_; }
+    std::span<const float> Spectrum() const { return noise_spectrum_; }
 
     // Get the noise power spectrum at a certain band.
     float Power(size_t band) const {
diff --git a/modules/audio_processing/aec3/subband_erle_estimator.cc b/modules/audio_processing/aec3/subband_erle_estimator.cc
index 8db40a2579..c71d5e8d06 100644
--- a/modules/audio_processing/aec3/subband_erle_estimator.cc
+++ b/modules/audio_processing/aec3/subband_erle_estimator.cc
@@ -15,9 +15,9 @@
 #include <cstddef>
 #include <functional>
 #include <memory>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "api/environment/environment.h"
 #include "api/field_trials_view.h"
@@ -83,9 +83,9 @@ void SubbandErleEstimator::Reset() {
 }
 
 void SubbandErleEstimator::Update(
-    ArrayView<const float, kFftLengthBy2Plus1> X2,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> Y2,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> E2,
+    std::span<const float, kFftLengthBy2Plus1> X2,
+    std::span<const std::array<float, kFftLengthBy2Plus1>> Y2,
+    std::span<const std::array<float, kFftLengthBy2Plus1>> E2,
     const std::vector<bool>& converged_filters) {
   UpdateAccumulatedSpectra(X2, Y2, E2, converged_filters);
   UpdateBands(converged_filters);
@@ -221,9 +221,9 @@ void SubbandErleEstimator::ResetAccumulatedSpectra() {
 }
 
 void SubbandErleEstimator::UpdateAccumulatedSpectra(
-    ArrayView<const float, kFftLengthBy2Plus1> X2,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> Y2,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> E2,
+    std::span<const float, kFftLengthBy2Plus1> X2,
+    std::span<const std::array<float, kFftLengthBy2Plus1>> Y2,
+    std::span<const std::array<float, kFftLengthBy2Plus1>> E2,
     const std::vector<bool>& converged_filters) {
   auto& st = accum_spectra_;
   RTC_DCHECK_EQ(st.E2.size(), E2.size());
diff --git a/modules/audio_processing/aec3/subband_erle_estimator.h b/modules/audio_processing/aec3/subband_erle_estimator.h
index fdfaad8872..5e639fe2e4 100644
--- a/modules/audio_processing/aec3/subband_erle_estimator.h
+++ b/modules/audio_processing/aec3/subband_erle_estimator.h
@@ -15,9 +15,9 @@
 
 #include <array>
 #include <memory>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "api/environment/environment.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
@@ -37,25 +37,25 @@ class SubbandErleEstimator {
   void Reset();
 
   // Updates the ERLE estimate.
-  void Update(ArrayView<const float, kFftLengthBy2Plus1> X2,
-              ArrayView<const std::array<float, kFftLengthBy2Plus1>> Y2,
-              ArrayView<const std::array<float, kFftLengthBy2Plus1>> E2,
+  void Update(std::span<const float, kFftLengthBy2Plus1> X2,
+              std::span<const std::array<float, kFftLengthBy2Plus1>> Y2,
+              std::span<const std::array<float, kFftLengthBy2Plus1>> E2,
               const std::vector<bool>& converged_filters);
 
   // Returns the ERLE estimate.
-  ArrayView<const std::array<float, kFftLengthBy2Plus1>> Erle(
+  std::span<const std::array<float, kFftLengthBy2Plus1>> Erle(
       bool onset_compensated) const {
     return onset_compensated && use_onset_detection_ ? erle_onset_compensated_
                                                      : erle_;
   }
 
   // Returns the non-capped ERLE estimate.
-  ArrayView<const std::array<float, kFftLengthBy2Plus1>> ErleUnbounded() const {
+  std::span<const std::array<float, kFftLengthBy2Plus1>> ErleUnbounded() const {
     return erle_unbounded_;
   }
 
   // Returns the ERLE estimate at onsets (only used for testing).
-  ArrayView<const std::array<float, kFftLengthBy2Plus1>> ErleDuringOnsets()
+  std::span<const std::array<float, kFftLengthBy2Plus1>> ErleDuringOnsets()
       const {
     return erle_during_onsets_;
   }
@@ -76,9 +76,9 @@ class SubbandErleEstimator {
   };
 
   void UpdateAccumulatedSpectra(
-      ArrayView<const float, kFftLengthBy2Plus1> X2,
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>> Y2,
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>> E2,
+      std::span<const float, kFftLengthBy2Plus1> X2,
+      std::span<const std::array<float, kFftLengthBy2Plus1>> Y2,
+      std::span<const std::array<float, kFftLengthBy2Plus1>> E2,
       const std::vector<bool>& converged_filters);
 
   void ResetAccumulatedSpectra();
diff --git a/modules/audio_processing/aec3/subband_nearend_detector.cc b/modules/audio_processing/aec3/subband_nearend_detector.cc
index 37b356367b..e569a3f131 100644
--- a/modules/audio_processing/aec3/subband_nearend_detector.cc
+++ b/modules/audio_processing/aec3/subband_nearend_detector.cc
@@ -13,8 +13,8 @@
 #include <array>
 #include <cstddef>
 #include <numeric>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/moving_average.h"
@@ -34,10 +34,10 @@ SubbandNearendDetector::SubbandNearendDetector(
           1.f / (config_.subband2.high - config_.subband2.low + 1)) {}
 
 void SubbandNearendDetector::Update(
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> nearend_spectrum,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+    std::span<const std::array<float, kFftLengthBy2Plus1>> nearend_spectrum,
+    std::span<const std::array<float, kFftLengthBy2Plus1>>
     /* residual_echo_spectrum */,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+    std::span<const std::array<float, kFftLengthBy2Plus1>>
         comfort_noise_spectrum,
     bool /* initial_state */) {
   nearend_state_ = false;
diff --git a/modules/audio_processing/aec3/subband_nearend_detector.h b/modules/audio_processing/aec3/subband_nearend_detector.h
index ab733b9acb..635e063b24 100644
--- a/modules/audio_processing/aec3/subband_nearend_detector.h
+++ b/modules/audio_processing/aec3/subband_nearend_detector.h
@@ -13,9 +13,9 @@
 
 #include <array>
 #include <cstddef>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/moving_average.h"
@@ -34,10 +34,10 @@ class SubbandNearendDetector : public NearendDetector {
 
   // Updates the state selection based on latest spectral estimates.
   void Update(
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>> nearend_spectrum,
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+      std::span<const std::array<float, kFftLengthBy2Plus1>> nearend_spectrum,
+      std::span<const std::array<float, kFftLengthBy2Plus1>>
           residual_echo_spectrum,
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+      std::span<const std::array<float, kFftLengthBy2Plus1>>
           comfort_noise_spectrum,
       bool initial_state) override;
 
diff --git a/modules/audio_processing/aec3/subtractor.cc b/modules/audio_processing/aec3/subtractor.cc
index bfce13f42d..8f9d7752e9 100644
--- a/modules/audio_processing/aec3/subtractor.cc
+++ b/modules/audio_processing/aec3/subtractor.cc
@@ -14,9 +14,9 @@
 #include <array>
 #include <cstddef>
 #include <memory>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "api/environment/environment.h"
 #include "api/field_trials_view.h"
@@ -48,7 +48,7 @@ bool UseCoarseFilterResetHangover(const FieldTrialsView& field_trials) {
 
 void PredictionError(const Aec3Fft& fft,
                      const FftData& S,
-                     ArrayView<const float> y,
+                     std::span<const float> y,
                      std::array<float, kBlockSize>* e,
                      std::array<float, kBlockSize>* s) {
   std::array<float, kFftLength> tmp;
@@ -64,10 +64,10 @@ void PredictionError(const Aec3Fft& fft,
   }
 }
 
-void ScaleFilterOutput(ArrayView<const float> y,
+void ScaleFilterOutput(std::span<const float> y,
                        float factor,
-                       ArrayView<float> e,
-                       ArrayView<float> s) {
+                       std::span<float> e,
+                       std::span<float> s) {
   RTC_DCHECK_EQ(y.size(), e.size());
   RTC_DCHECK_EQ(y.size(), s.size());
   for (size_t k = 0; k < y.size(); ++k) {
@@ -197,7 +197,7 @@ void Subtractor::Process(const RenderBuffer& render_buffer,
                          const Block& capture,
                          const RenderSignalAnalyzer& render_signal_analyzer,
                          const AecState& aec_state,
-                         ArrayView<SubtractorOutput> outputs) {
+                         std::span<SubtractorOutput> outputs) {
   RTC_DCHECK_EQ(num_capture_channels_, capture.NumChannels());
 
   // Compute the render powers.
@@ -223,7 +223,7 @@ void Subtractor::Process(const RenderBuffer& render_buffer,
   // Process all capture channels
   for (size_t ch = 0; ch < num_capture_channels_; ++ch) {
     SubtractorOutput& output = outputs[ch];
-    ArrayView<const float> y = capture.View(/*band=*/0, ch);
+    std::span<const float> y = capture.View(/*band=*/0, ch);
     FftData& E_refined = output.E_refined;
     FftData E_coarse;
     std::array<float, kBlockSize>& e_refined = output.e_refined;
diff --git a/modules/audio_processing/aec3/subtractor.h b/modules/audio_processing/aec3/subtractor.h
index 148bdf2c1f..e0402900eb 100644
--- a/modules/audio_processing/aec3/subtractor.h
+++ b/modules/audio_processing/aec3/subtractor.h
@@ -15,9 +15,9 @@
 #include <cmath>
 #include <cstddef>
 #include <memory>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "api/environment/environment.h"
 #include "modules/audio_processing/aec3/adaptive_fir_filter.h"
@@ -54,7 +54,7 @@ class Subtractor {
                const Block& capture,
                const RenderSignalAnalyzer& render_signal_analyzer,
                const AecState& aec_state,
-               ArrayView<SubtractorOutput> outputs);
+               std::span<SubtractorOutput> outputs);
 
   void HandleEchoPathChange(const EchoPathVariability& echo_path_variability);
 
@@ -77,7 +77,7 @@ class Subtractor {
   void DumpFilters() {
     data_dumper_->DumpRaw(
         "aec3_subtractor_h_refined",
-        ArrayView<const float>(
+        std::span<const float>(
             refined_impulse_responses_[0].data(),
             GetTimeDomainLength(
                 refined_filters_[0]->max_filter_size_partitions())));
@@ -85,7 +85,7 @@ class Subtractor {
       RTC_DCHECK_GT(coarse_impulse_responses_.size(), 0);
       data_dumper_->DumpRaw(
           "aec3_subtractor_h_coarse",
-          ArrayView<const float>(
+          std::span<const float>(
               coarse_impulse_responses_[0].data(),
               GetTimeDomainLength(
                   coarse_filter_[0]->max_filter_size_partitions())));
diff --git a/modules/audio_processing/aec3/subtractor_output.cc b/modules/audio_processing/aec3/subtractor_output.cc
index 7a5bd17ef9..811014c90c 100644
--- a/modules/audio_processing/aec3/subtractor_output.cc
+++ b/modules/audio_processing/aec3/subtractor_output.cc
@@ -12,8 +12,7 @@
 
 #include <algorithm>
 #include <numeric>
-
-#include "api/array_view.h"
+#include <span>
 
 namespace webrtc {
 
@@ -36,7 +35,7 @@ void SubtractorOutput::Reset() {
   y2 = 0.f;
 }
 
-void SubtractorOutput::ComputeMetrics(ArrayView<const float> y) {
+void SubtractorOutput::ComputeMetrics(std::span<const float> y) {
   const auto sum_of_squares = [](float a, float b) { return a + b * b; };
   y2 = std::accumulate(y.begin(), y.end(), 0.f, sum_of_squares);
   e2_refined =
diff --git a/modules/audio_processing/aec3/subtractor_output.h b/modules/audio_processing/aec3/subtractor_output.h
index 10349f5937..6fb1d0d8be 100644
--- a/modules/audio_processing/aec3/subtractor_output.h
+++ b/modules/audio_processing/aec3/subtractor_output.h
@@ -12,8 +12,8 @@
 #define MODULES_AUDIO_PROCESSING_AEC3_SUBTRACTOR_OUTPUT_H_
 
 #include <array>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/fft_data.h"
 
@@ -44,7 +44,7 @@ struct SubtractorOutput {
   void Reset();
 
   // Updates the powers of the signals.
-  void ComputeMetrics(ArrayView<const float> y);
+  void ComputeMetrics(std::span<const float> y);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_processing/aec3/subtractor_output_analyzer.cc b/modules/audio_processing/aec3/subtractor_output_analyzer.cc
index f0cb67278f..52d806c837 100644
--- a/modules/audio_processing/aec3/subtractor_output_analyzer.cc
+++ b/modules/audio_processing/aec3/subtractor_output_analyzer.cc
@@ -12,8 +12,8 @@
 
 #include <algorithm>
 #include <cstddef>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/subtractor_output.h"
 #include "rtc_base/checks.h"
@@ -24,7 +24,7 @@ SubtractorOutputAnalyzer::SubtractorOutputAnalyzer(size_t num_capture_channels)
     : filters_converged_(num_capture_channels, false) {}
 
 void SubtractorOutputAnalyzer::Update(
-    ArrayView<const SubtractorOutput> subtractor_output,
+    std::span<const SubtractorOutput> subtractor_output,
     bool* any_filter_converged,
     bool* any_coarse_filter_converged,
     bool* all_filters_diverged) {
diff --git a/modules/audio_processing/aec3/subtractor_output_analyzer.h b/modules/audio_processing/aec3/subtractor_output_analyzer.h
index 760ffe63b0..9559dab272 100644
--- a/modules/audio_processing/aec3/subtractor_output_analyzer.h
+++ b/modules/audio_processing/aec3/subtractor_output_analyzer.h
@@ -12,9 +12,9 @@
 #define MODULES_AUDIO_PROCESSING_AEC3_SUBTRACTOR_OUTPUT_ANALYZER_H_
 
 #include <cstddef>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/subtractor_output.h"
 
 namespace webrtc {
@@ -26,7 +26,7 @@ class SubtractorOutputAnalyzer {
   ~SubtractorOutputAnalyzer() = default;
 
   // Analyses the subtractor output.
-  void Update(ArrayView<const SubtractorOutput> subtractor_output,
+  void Update(std::span<const SubtractorOutput> subtractor_output,
               bool* any_filter_converged,
               bool* any_coarse_filter_converged,
               bool* all_filters_diverged);
diff --git a/modules/audio_processing/aec3/subtractor_unittest.cc b/modules/audio_processing/aec3/subtractor_unittest.cc
index 9d2be96e52..5d30146e94 100644
--- a/modules/audio_processing/aec3/subtractor_unittest.cc
+++ b/modules/audio_processing/aec3/subtractor_unittest.cc
@@ -16,11 +16,11 @@
 #include <memory>
 #include <numeric>
 #include <optional>
+#include <span>
 #include <string>
 #include <tuple>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "api/environment/environment.h"
 #include "api/environment/environment_factory.h"
@@ -106,14 +106,14 @@ std::vector<float> RunSubtractorTest(
       num_render_channels);
   for (size_t ch = 0; ch < num_render_channels; ++ch) {
     x_hp_filter[ch] = std::make_unique<CascadedBiQuadFilter>(
-        ArrayView<const CascadedBiQuadFilter::BiQuadCoefficients>(
+        std::span<const CascadedBiQuadFilter::BiQuadCoefficients>(
             kHighPassFilterCoefficients));
   }
   std::vector<std::unique_ptr<CascadedBiQuadFilter>> y_hp_filter(
       num_capture_channels);
   for (size_t ch = 0; ch < num_capture_channels; ++ch) {
     y_hp_filter[ch] = std::make_unique<CascadedBiQuadFilter>(
-        ArrayView<const CascadedBiQuadFilter::BiQuadCoefficients>(
+        std::span<const CascadedBiQuadFilter::BiQuadCoefficients>(
             kHighPassFilterCoefficients));
   }
 
@@ -130,7 +130,7 @@ std::vector<float> RunSubtractorTest(
     } else {
       for (size_t capture_ch = 0; capture_ch < num_capture_channels;
            ++capture_ch) {
-        ArrayView<float> y_view = y.View(/*band=*/0, capture_ch);
+        std::span<float> y_view = y.View(/*band=*/0, capture_ch);
         for (size_t render_ch = 0; render_ch < num_render_channels;
              ++render_ch) {
           std::array<float, kBlockSize> y_channel;
diff --git a/modules/audio_processing/aec3/suppression_filter.cc b/modules/audio_processing/aec3/suppression_filter.cc
index fa44591b13..b5a6f79fb2 100644
--- a/modules/audio_processing/aec3/suppression_filter.cc
+++ b/modules/audio_processing/aec3/suppression_filter.cc
@@ -15,9 +15,9 @@
 #include <cmath>
 #include <cstring>
 #include <iterator>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/block.h"
 #include "modules/audio_processing/aec3/fft_data.h"
@@ -86,11 +86,11 @@ SuppressionFilter::SuppressionFilter(Aec3Optimization optimization,
 SuppressionFilter::~SuppressionFilter() = default;
 
 void SuppressionFilter::ApplyGain(
-    ArrayView<const FftData> comfort_noise,
-    ArrayView<const FftData> comfort_noise_high_band,
+    std::span<const FftData> comfort_noise,
+    std::span<const FftData> comfort_noise_high_band,
     const std::array<float, kFftLengthBy2Plus1>& suppression_gain,
     float high_bands_gain,
-    ArrayView<const FftData> E_lowest_band,
+    std::span<const FftData> E_lowest_band,
     Block* e) {
   RTC_DCHECK(e);
   RTC_DCHECK_EQ(e->NumBands(), NumBandsForRate(sample_rate_hz_));
diff --git a/modules/audio_processing/aec3/suppression_filter.h b/modules/audio_processing/aec3/suppression_filter.h
index 4a6de28eba..a37d783551 100644
--- a/modules/audio_processing/aec3/suppression_filter.h
+++ b/modules/audio_processing/aec3/suppression_filter.h
@@ -13,9 +13,9 @@
 
 #include <array>
 #include <cstddef>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/aec3_fft.h"
 #include "modules/audio_processing/aec3/block.h"
@@ -33,11 +33,11 @@ class SuppressionFilter {
   SuppressionFilter(const SuppressionFilter&) = delete;
   SuppressionFilter& operator=(const SuppressionFilter&) = delete;
 
-  void ApplyGain(ArrayView<const FftData> comfort_noise,
-                 ArrayView<const FftData> comfort_noise_high_bands,
+  void ApplyGain(std::span<const FftData> comfort_noise,
+                 std::span<const FftData> comfort_noise_high_bands,
                  const std::array<float, kFftLengthBy2Plus1>& suppression_gain,
                  float high_bands_gain,
-                 ArrayView<const FftData> E_lowest_band,
+                 std::span<const FftData> E_lowest_band,
                  Block* e);
 
  private:
diff --git a/modules/audio_processing/aec3/suppression_gain.cc b/modules/audio_processing/aec3/suppression_gain.cc
index 0ba04aec9f..bd92aff2e8 100644
--- a/modules/audio_processing/aec3/suppression_gain.cc
+++ b/modules/audio_processing/aec3/suppression_gain.cc
@@ -18,8 +18,8 @@
 #include <memory>
 #include <numeric>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/aec_state.h"
@@ -84,13 +84,13 @@ void LimitHighFrequencyGains(const EchoCanceller3Config::Suppressor& config,
 
 // Scales the echo according to assessed audibility at the other end.
 void WeightEchoForAudibility(const EchoCanceller3Config& config,
-                             ArrayView<const float> echo,
-                             ArrayView<float> weighted_echo) {
+                             std::span<const float> echo,
+                             std::span<float> weighted_echo) {
   RTC_DCHECK_EQ(kFftLengthBy2Plus1, echo.size());
   RTC_DCHECK_EQ(kFftLengthBy2Plus1, weighted_echo.size());
 
   auto weigh = [](float threshold, float normalizer, size_t begin, size_t end,
-                  ArrayView<const float> echo, ArrayView<float> weighted_echo) {
+                  std::span<const float> echo, std::span<float> weighted_echo) {
     for (size_t k = begin; k < end; ++k) {
       if (echo[k] < threshold) {
         float tmp = (threshold - echo[k]) * normalizer;
@@ -122,8 +122,8 @@ void WeightEchoForAudibility(const EchoCanceller3Config& config,
 std::atomic<int> SuppressionGain::instance_count_(0);
 
 float SuppressionGain::UpperBandsGain(
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> echo_spectrum,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+    std::span<const std::array<float, kFftLengthBy2Plus1>> echo_spectrum,
+    std::span<const std::array<float, kFftLengthBy2Plus1>>
         comfort_noise_spectrum,
     const std::optional<int>& narrow_peak_band,
     bool saturated_echo,
@@ -189,7 +189,7 @@ float SuppressionGain::UpperBandsGain(
   if (!dominant_nearend_detector_->IsNearendState()) {
     // Bound the upper gain during significant echo activity.
     const auto& cfg = config_.suppressor.high_bands_suppression;
-    auto low_frequency_energy = [](ArrayView<const float> spectrum) {
+    auto low_frequency_energy = [](std::span<const float> spectrum) {
       RTC_DCHECK_LE(16, spectrum.size());
       return std::accumulate(spectrum.begin() + 1, spectrum.begin() + 16, 0.f);
     };
@@ -230,12 +230,12 @@ void SuppressionGain::GainToNoAudibleEcho(
 
 // Compute the minimum gain as the attenuating gain to put the signal just
 // above the zero sample values.
-void SuppressionGain::GetMinGain(ArrayView<const float> weighted_residual_echo,
-                                 ArrayView<const float> last_nearend,
-                                 ArrayView<const float> last_echo,
+void SuppressionGain::GetMinGain(std::span<const float> weighted_residual_echo,
+                                 std::span<const float> last_nearend,
+                                 std::span<const float> last_echo,
                                  bool low_noise_render,
                                  bool saturated_echo,
-                                 ArrayView<float> min_gain) const {
+                                 std::span<float> min_gain) const {
   if (!saturated_echo) {
     const float min_echo_power =
         low_noise_render ? config_.echo_audibility.low_render_limit
@@ -271,7 +271,7 @@ void SuppressionGain::GetMinGain(ArrayView<const float> weighted_residual_echo,
 
 // Compute the maximum gain by limiting the gain increase from the previous
 // gain.
-void SuppressionGain::GetMaxGain(ArrayView<float> max_gain) const {
+void SuppressionGain::GetMaxGain(std::span<float> max_gain) const {
   const auto& inc = dominant_nearend_detector_->IsNearendState()
                         ? nearend_params_.max_inc_factor
                         : normal_params_.max_inc_factor;
@@ -284,9 +284,9 @@ void SuppressionGain::GetMaxGain(ArrayView<float> max_gain) const {
 void SuppressionGain::LowerBandGain(
     bool low_noise_render,
     const AecState& aec_state,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> suppressor_input,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> residual_echo,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> comfort_noise,
+    std::span<const std::array<float, kFftLengthBy2Plus1>> suppressor_input,
+    std::span<const std::array<float, kFftLengthBy2Plus1>> residual_echo,
+    std::span<const std::array<float, kFftLengthBy2Plus1>> comfort_noise,
     bool clock_drift,
     std::array<float, kFftLengthBy2Plus1>* gain) {
   gain->fill(1.f);
@@ -375,13 +375,13 @@ SuppressionGain::SuppressionGain(const EchoCanceller3Config& config,
 SuppressionGain::~SuppressionGain() = default;
 
 void SuppressionGain::GetGain(
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> nearend_spectrum,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>> echo_spectrum,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+    std::span<const std::array<float, kFftLengthBy2Plus1>> nearend_spectrum,
+    std::span<const std::array<float, kFftLengthBy2Plus1>> echo_spectrum,
+    std::span<const std::array<float, kFftLengthBy2Plus1>>
         residual_echo_spectrum,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+    std::span<const std::array<float, kFftLengthBy2Plus1>>
         residual_echo_spectrum_unbounded,
-    ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+    std::span<const std::array<float, kFftLengthBy2Plus1>>
         comfort_noise_spectrum,
     const RenderSignalAnalyzer& render_signal_analyzer,
     const AecState& aec_state,
diff --git a/modules/audio_processing/aec3/suppression_gain.h b/modules/audio_processing/aec3/suppression_gain.h
index b0a56d04ad..550b533cbf 100644
--- a/modules/audio_processing/aec3/suppression_gain.h
+++ b/modules/audio_processing/aec3/suppression_gain.h
@@ -16,9 +16,9 @@
 #include <cstddef>
 #include <memory>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/aec_state.h"
@@ -42,13 +42,13 @@ class SuppressionGain {
   SuppressionGain& operator=(const SuppressionGain&) = delete;
 
   void GetGain(
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>> nearend_spectrum,
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>> echo_spectrum,
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+      std::span<const std::array<float, kFftLengthBy2Plus1>> nearend_spectrum,
+      std::span<const std::array<float, kFftLengthBy2Plus1>> echo_spectrum,
+      std::span<const std::array<float, kFftLengthBy2Plus1>>
           residual_echo_spectrum,
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+      std::span<const std::array<float, kFftLengthBy2Plus1>>
           residual_echo_spectrum_unbounded,
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+      std::span<const std::array<float, kFftLengthBy2Plus1>>
           comfort_noise_spectrum,
       const RenderSignalAnalyzer& render_signal_analyzer,
       const AecState& aec_state,
@@ -67,8 +67,8 @@ class SuppressionGain {
  private:
   // Computes the gain to apply for the bands beyond the first band.
   float UpperBandsGain(
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>> echo_spectrum,
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+      std::span<const std::array<float, kFftLengthBy2Plus1>> echo_spectrum,
+      std::span<const std::array<float, kFftLengthBy2Plus1>>
           comfort_noise_spectrum,
       const std::optional<int>& narrow_peak_band,
       bool saturated_echo,
@@ -83,20 +83,20 @@ class SuppressionGain {
   void LowerBandGain(
       bool stationary_with_low_power,
       const AecState& aec_state,
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>> suppressor_input,
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>> residual_echo,
-      ArrayView<const std::array<float, kFftLengthBy2Plus1>> comfort_noise,
+      std::span<const std::array<float, kFftLengthBy2Plus1>> suppressor_input,
+      std::span<const std::array<float, kFftLengthBy2Plus1>> residual_echo,
+      std::span<const std::array<float, kFftLengthBy2Plus1>> comfort_noise,
       bool clock_drift,
       std::array<float, kFftLengthBy2Plus1>* gain);
 
-  void GetMinGain(ArrayView<const float> weighted_residual_echo,
-                  ArrayView<const float> last_nearend,
-                  ArrayView<const float> last_echo,
+  void GetMinGain(std::span<const float> weighted_residual_echo,
+                  std::span<const float> last_nearend,
+                  std::span<const float> last_echo,
                   bool low_noise_render,
                   bool saturated_echo,
-                  ArrayView<float> min_gain) const;
+                  std::span<float> min_gain) const;
 
-  void GetMaxGain(ArrayView<float> max_gain) const;
+  void GetMaxGain(std::span<float> max_gain) const;
 
   class LowNoiseRenderDetector {
    public:
diff --git a/modules/audio_processing/aec3/vector_math.h b/modules/audio_processing/aec3/vector_math.h
index 9fd7792d3f..d8b5ecf3d1 100644
--- a/modules/audio_processing/aec3/vector_math.h
+++ b/modules/audio_processing/aec3/vector_math.h
@@ -14,8 +14,8 @@
 #include <algorithm>
 #include <cmath>
 #include <functional>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "rtc_base/checks.h"
 
@@ -38,8 +38,8 @@ class VectorMath {
       : optimization_(optimization) {}
 
   // Elementwise square root.
-  void SqrtAVX2(ArrayView<float> x);
-  void Sqrt(ArrayView<float> x) {
+  void SqrtAVX2(std::span<float> x);
+  void Sqrt(std::span<float> x) {
     switch (optimization_) {
 #if defined(WEBRTC_ARCH_X86_FAMILY)
       case Aec3Optimization::kSse2: {
@@ -112,12 +112,12 @@ class VectorMath {
   }
 
   // Elementwise vector multiplication z = x * y.
-  void MultiplyAVX2(ArrayView<const float> x,
-                    ArrayView<const float> y,
-                    ArrayView<float> z);
-  void Multiply(ArrayView<const float> x,
-                ArrayView<const float> y,
-                ArrayView<float> z) {
+  void MultiplyAVX2(std::span<const float> x,
+                    std::span<const float> y,
+                    std::span<float> z);
+  void Multiply(std::span<const float> x,
+                std::span<const float> y,
+                std::span<float> z) {
     RTC_DCHECK_EQ(z.size(), x.size());
     RTC_DCHECK_EQ(z.size(), y.size());
     switch (optimization_) {
@@ -167,8 +167,8 @@ class VectorMath {
   }
 
   // Elementwise vector accumulation z += x.
-  void AccumulateAVX2(ArrayView<const float> x, ArrayView<float> z);
-  void Accumulate(ArrayView<const float> x, ArrayView<float> z) {
+  void AccumulateAVX2(std::span<const float> x, std::span<float> z);
+  void Accumulate(std::span<const float> x, std::span<float> z) {
     RTC_DCHECK_EQ(z.size(), x.size());
     switch (optimization_) {
 #if defined(WEBRTC_ARCH_X86_FAMILY)
diff --git a/modules/audio_processing/aec3/vector_math_avx2.cc b/modules/audio_processing/aec3/vector_math_avx2.cc
index da4dd6e515..2288420f40 100644
--- a/modules/audio_processing/aec3/vector_math_avx2.cc
+++ b/modules/audio_processing/aec3/vector_math_avx2.cc
@@ -11,8 +11,8 @@
 #include <immintrin.h>
 
 #include <cmath>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/aec3/vector_math.h"
 #include "rtc_base/checks.h"
 
@@ -20,7 +20,7 @@ namespace webrtc {
 namespace aec3 {
 
 // Elementwise square root.
-void VectorMath::SqrtAVX2(ArrayView<float> x) {
+void VectorMath::SqrtAVX2(std::span<float> x) {
   const int x_size = static_cast<int>(x.size());
   const int vector_limit = x_size >> 3;
 
@@ -37,9 +37,9 @@ void VectorMath::SqrtAVX2(ArrayView<float> x) {
 }
 
 // Elementwise vector multiplication z = x * y.
-void VectorMath::MultiplyAVX2(ArrayView<const float> x,
-                              ArrayView<const float> y,
-                              ArrayView<float> z) {
+void VectorMath::MultiplyAVX2(std::span<const float> x,
+                              std::span<const float> y,
+                              std::span<float> z) {
   RTC_DCHECK_EQ(z.size(), x.size());
   RTC_DCHECK_EQ(z.size(), y.size());
   const int x_size = static_cast<int>(x.size());
@@ -59,7 +59,7 @@ void VectorMath::MultiplyAVX2(ArrayView<const float> x,
 }
 
 // Elementwise vector accumulation z += x.
-void VectorMath::AccumulateAVX2(ArrayView<const float> x, ArrayView<float> z) {
+void VectorMath::AccumulateAVX2(std::span<const float> x, std::span<float> z) {
   RTC_DCHECK_EQ(z.size(), x.size());
   const int x_size = static_cast<int>(x.size());
   const int vector_limit = x_size >> 3;

From 17a2fb58ea7db7b17bdbcf82b400bd311558a027 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Wed, 11 Mar 2026 11:03:49 -0700
Subject: [PATCH 265/472] Roll chromium_revision b937d39273..8fdfcb5d76
 (1597657:1597876)

Change log: https://chromium.googlesource.com/chromium/src/+log/b937d39273..8fdfcb5d76
Full diff: https://chromium.googlesource.com/chromium/src/+/b937d39273..8fdfcb5d76

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/7d29c62736..c9202dd8ca
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/94f693fcb6..00efde9573
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/e7a1e88ecc..bacbfe72d8
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/a133c94b51..22a76fe23b
* src/third_party/androidx/cipd: Hi79Lu6qQ-Fbcjwgf_f-NEYJweP4QdZqjJ0lXwc0PpcC..A6oLcxGdFmfrPoaAcVYoVYHlkB3InvLKiIKnIgmGJVcC
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/e0ebf38a01..a2f6285fa5
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/4ce8ba39a3..a4e677522d
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/4387e32481..c4eb3c4f4c
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/260c6fd37c..3ca09e4818
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/4cead1b0fc..c0a3840504
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/9765576870..82543c0c7f
DEPS diff: https://chromium.googlesource.com/chromium/src/+/b937d39273..8fdfcb5d76/DEPS

BUG=None

Change-Id: Ica2e37fd2735bac28e9c7b69c9539b487e5666d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/455900
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47118}
---
 DEPS | 24 ++++++++++++------------
 1 file changed, 12 insertions(+), 12 deletions(-)

diff --git a/DEPS b/DEPS
index dffcecaef3..211c40da09 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': 'b937d39273b52034e8542dea4b040587be0a127f',
+  'chromium_revision': '8fdfcb5d762df924ac3e77514c178343e2aeead3',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@7d29c62736304aeaac07df0b75546ffd2b75917a',
+    'https://chromium.googlesource.com/chromium/src/build@c9202dd8ca1d4255bd9e505988b945746ddb378a',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@fef8d6947fa54202e8a00c58d5ac8fd15780bc33',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@94f693fcb6dde967f11668a9e21eec1a4ec8b49c',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@00efde9573f5ecad0307619539c633c4eb805905',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@e7a1e88ecc35a70f3d198674042753ec9e8e3650',
+    'https://chromium.googlesource.com/chromium/src/testing@bacbfe72d8a48eac176f063aefc517cd84786b91',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@a133c94b51ac034d5f375cc1746e66dcdee57282',
+    'https://chromium.googlesource.com/chromium/src/third_party@22a76fe23bae393322386681d4bfa508b0da44ea',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -492,7 +492,7 @@ deps = {
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@8be0e3114685fcc1589561067282edf75ea1259a',
   'src/third_party/catapult':
-    'https://chromium.googlesource.com/catapult.git@e0ebf38a01214aba11f31daa1c743782def031d5',
+    'https://chromium.googlesource.com/catapult.git@a2f6285fa55a02491af892e15ac308fd8315d754',
   'src/third_party/ced/src': {
     'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5',
   },
@@ -505,13 +505,13 @@ deps = {
   'src/third_party/crc32c/src':
     'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558',
   'src/third_party/depot_tools':
-    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@4ce8ba39a3488397a2d1494f167020f21de502f3',
+    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@a4e677522da5c326b656dc9584b3a4f6cde9fd05',
   'src/third_party/ffmpeg':
     'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@946d97db8d906277085e361892b7efda5152e2f1',
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@260c6fd37c502e984836a446c5d90e4a1298036d',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@3ca09e4818f4362e75c9c224f1d4c2065db75a0a',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '4cead1b0fc9987f994de97c7914e1f0c94047579',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'c0a3840504ad618393057eb1fc121810c210dfd2',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@9765576870401b230a826e0e4f7a12851575306e',
+    'https://chromium.googlesource.com/chromium/src/tools@82543c0c7fa9a6436b46102e26e57e09aefa192d',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'Hi79Lu6qQ-Fbcjwgf_f-NEYJweP4QdZqjJ0lXwc0PpcC',
+          'version': 'A6oLcxGdFmfrPoaAcVYoVYHlkB3InvLKiIKnIgmGJVcC',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -864,7 +864,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7364b490b5f78d58efe23ea76e74210fd6c3c76f',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '4387e3248109bef2c07a8b69a2b8249d7af5dcf0',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + 'c4eb3c4f4c91511eb5c76d608e07dde2a1cc8b23',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From 4a39f9487a8acc3c3b32ec0edd32185246317e59 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Wed, 11 Mar 2026 16:28:52 +0100
Subject: [PATCH 266/472] Move Call::receive_time_calculator_ to network thread

This CL moves the usage of receive_time_calculator_ in Call from the
worker thread to the network thread. It extracts the worker thread logic
of DeliverRtpPacket into DeliverRtpPacket_w to allow clearer thread
annotations.

Also includes new g3doc for a simple overview of the RTP receive flow.

Bug: webrtc:11993
Change-Id: I143a548888474b4ebd14a9041d5cb649738e0775
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/454304
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47119}
---
 call/call.cc              |  42 ++++++++++----
 g3doc/rtp_receive_flow.md | 116 ++++++++++++++++++++++++++++++++++++++
 pc/peer_connection.cc     |  18 +++---
 3 files changed, 154 insertions(+), 22 deletions(-)
 create mode 100644 g3doc/rtp_receive_flow.md

diff --git a/call/call.cc b/call/call.cc
index 40a35c978a..a7fd3e3142 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -347,6 +347,11 @@ class Call final : public webrtc::Call,
   void DeliverRtcp(MediaType media_type, CopyOnWriteBuffer packet)
       RTC_RUN_ON(network_thread_);
 
+  void DeliverRtpPacket_w(MediaType media_type,
+                          RtpPacketReceived packet,
+                          OnUndemuxablePacketHandler undemuxable_packet_handler)
+      RTC_RUN_ON(worker_thread_);
+
   AudioReceiveStreamImpl* FindAudioStreamForSyncGroup(
       absl::string_view sync_group) RTC_RUN_ON(worker_thread_);
   void ConfigureSync(absl::string_view sync_group) RTC_RUN_ON(worker_thread_);
@@ -443,7 +448,8 @@ class Call final : public webrtc::Call,
   RepeatingTaskHandle receive_side_cc_periodic_task_;
   RepeatingTaskHandle elastic_bandwidth_allocation_task_;
 
-  const std::unique_ptr<ReceiveTimeCalculator> receive_time_calculator_;
+  const std::unique_ptr<ReceiveTimeCalculator> receive_time_calculator_
+      RTC_GUARDED_BY(network_thread_);
 
   const std::unique_ptr<SendDelayStats> video_send_delay_stats_;
   const Timestamp start_of_call_;
@@ -1367,17 +1373,7 @@ void Call::DeliverRtpPacket(
     MediaType media_type,
     RtpPacketReceived packet,
     OnUndemuxablePacketHandler undemuxable_packet_handler) {
-  if (!worker_thread_->IsCurrent()) {
-    worker_thread_->PostTask(SafeTask(
-        task_safety_.flag(),
-        [this, media_type, packet = std::move(packet),
-         handler = std::move(undemuxable_packet_handler)]() mutable {
-          DeliverRtpPacket(media_type, std::move(packet), std::move(handler));
-        }));
-    return;
-  }
-  RTC_DCHECK_RUN_ON(worker_thread_);
-  RTC_DCHECK(packet.arrival_time().IsFinite());
+  RTC_DCHECK_RUN_ON(network_thread_);
 
   if (receive_time_calculator_) {
     int64_t packet_time_us = packet.arrival_time().us();
@@ -1388,6 +1384,28 @@ void Call::DeliverRtpPacket(
     packet.set_arrival_time(Timestamp::Micros(packet_time_us));
   }
 
+  if (worker_thread_->IsCurrent()) {
+    RTC_DCHECK_RUN_ON(worker_thread_);
+    DeliverRtpPacket_w(media_type, std::move(packet),
+                       std::move(undemuxable_packet_handler));
+  } else {
+    worker_thread_->PostTask(SafeTask(
+        task_safety_.flag(),
+        [this, media_type, packet = std::move(packet),
+         handler = std::move(undemuxable_packet_handler)]() mutable {
+          RTC_DCHECK_RUN_ON(worker_thread_);
+          DeliverRtpPacket_w(media_type, std::move(packet), std::move(handler));
+        }));
+  }
+}
+
+void Call::DeliverRtpPacket_w(
+    MediaType media_type,
+    RtpPacketReceived packet,
+    OnUndemuxablePacketHandler undemuxable_packet_handler) {
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  RTC_DCHECK(packet.arrival_time().IsFinite());
+
   NotifyBweOfReceivedPacket(packet, media_type);
 
   // Packets that are successfully demuxed are logged by their respective
diff --git a/g3doc/rtp_receive_flow.md b/g3doc/rtp_receive_flow.md
new file mode 100644
index 0000000000..24ce1c9e0e
--- /dev/null
+++ b/g3doc/rtp_receive_flow.md
@@ -0,0 +1,116 @@
+# WebRTC incoming RTP packet flow
+
+This document details the flow of incoming (receive) RTP packets across the WebRTC codebase, tracing the path from the network socket to the video receive stream.
+
+## Threading Model
+*   **Network Thread**: The majority of the initial receive and demuxing flow occurs synchronously on the network thread to minimize delays. 
+*   **Worker Thread**: Packet processing, dispatching encoded frames/buffers to jitter buffers. Actual decoding tends to happen on different threads however.
+
+---
+
+## Call Stack and Component Flow
+
+### 1. Network / Socket Layer 🌐 *(Network Thread)*
+Packets arrive from the network via UDP or TCP sockets. Platform-specific socket implementations trigger an event when bytes are ready to be read.
+*   **`AsyncUDPSocket` / `PhysicalSocket`**
+    *   `OnReadEvent` 
+
+### 2. Transport Routing 🛤️ *(Network Thread)*
+The bytes move up through the ICE/STUN/TURN connection channels and DTLS layer. 
+*   **`P2PTransportChannel`** -> `OnReadPacket`
+    *   Routes via **`Connection::OnReadPacket`**
+*   **`DtlsTransport`** -> `OnReadPacket`
+    *   Decrypts SRTP packets into cleartext bytes if necessary.
+
+### 3. RTP Transport & Parsing 📦 *(Network Thread)*
+The raw bytes reach the RTP core structures where they are parsed into proper C++ WebRTC objects.
+*   **`RtpTransport::OnRtpPacketReceived(const ReceivedIpPacket&)`**
+    *   Converts the byte buffer string into a `RtpPacketReceived`.
+    *   Extracts header extensions based on the active `RtpHeaderExtensionMap`.
+*   **`RtpTransport::DemuxPacket`**
+    *   Passes the parsed object down to the demuxer tree.
+
+### 4. RTP Demuxer (Level 1) 🔀 *(Network Thread)*
+*   **`RtpDemuxer::OnRtpPacket(const RtpPacketReceived&)`**
+    *   **Demuxing Happens Here**: This demuxer maps the packet using rules like **MID**, **RSID**, or **SSRC**.
+    *   Routes the packet to the matched `RtpPacketSinkInterface` (usually a `BaseChannel`).
+
+### 5. BaseChannel & Media Layer 📺 *(Network Thread)*
+*   **`BaseChannel::OnRtpPacket(const RtpPacketReceived&)`**
+    *   Implements `RtpPacketSinkInterface`. Serves as the base conduit for specific Media Channels (audio or video).
+*   **`WebRtcVideoReceiveChannel::OnPacketReceived(RtpPacketReceived)`**
+    *   Delegates the received packet directly up to the `Call` interface: `call_->Receiver()->DeliverRtpPacket(...)`
+
+### 6. Call Interface & Demuxer (Level 2) 📞 *(Network Thread -> Worker Thread)*
+*   **`Call::DeliverRtpPacket`**
+    *   Evaluates `receive_time_calculator_` directly on the Network Thread to avoid jitter/delay.
+    *   **Thread Hop**: Dispatches a task to the **Worker Thread** and invokes `Call::DeliverRtpPacket_w`.
+*   **`Call::DeliverRtpPacket_w`**
+    *   Executes on the Worker Thread.
+    *   Employs a secondary, internal demuxer logic to find the specific receive stream that matches the SSRC.
+    *   Calls the appropriate video or audio receive stream.
+
+### 7. Stream Receiver 📥 *(Worker Thread)*
+*   **`RtpVideoStreamReceiver2::OnRtpPacket(const RtpPacketReceived&)`**
+    *   Executes on the Worker Thread.
+    *   Extracts metadata, processes NACKs/RTCP feedback based on the incoming sequence numbers.
+    *   Inserts the payload into the `PacketBuffer` (Jitter Buffer) to wait for frame completion.
+    *   Once a complete video frame is assembled (or for deeper stream operations), the `VideoFrame` is decoded by `VCMGenericDecoder`.
+
+---
+
+## RTP Receive Packet Flow Diagram
+
+```text
+       [Network]
+           |
+           v
++-----------------------+
+|   AsyncUDPSocket /    |  (Network Thread)
+|   PhysicalSocket      |
++-----------------------+
+           |
+           v
++-----------------------+
+|  P2PTransportChannel  |  (Network Thread)
+|  / DtlsTransport      |
++-----------------------+
+           |
+           v
++-----------------------+
+|     RtpTransport      |  (Network Thread)
++-----------------------+
+           |
+           v
++-----------------------+
+|      RtpDemuxer       |  (Network Thread)
++-----------------------+
+           |
+           v
++-----------------------+
+|     BaseChannel       |  (Network Thread)
+|  & MediaReceiveChannel|
++-----------------------+
+           |
+           v
++-----------------------+
+| Call::DeliverRtpPacket|  (Network Thread)
++-----------------------+
+           |
+           | Thread Hop
+           v
++-----------------------+
+| Call::                |  (Worker Thread)
+| DeliverRtpPacket_w    |
++-----------------------+
+           |
+           v
++-----------------------+
+| Stream Receiver       |  (Worker Thread)
+| (e.g., RtpVideo-      |
+|  StreamReceiver2)     |
++-----------------------+
+           |
+           v
+      [Decoder]
+```
diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc
index cd4ad8b463..e2a4adedc9 100644
--- a/pc/peer_connection.cc
+++ b/pc/peer_connection.cc
@@ -3181,16 +3181,14 @@ PeerConnection::InitializeRtcpCallback() {
 absl::AnyInvocable<void(const RtpPacketReceived& parsed_packet) const>
 PeerConnection::InitializeUnDemuxablePacketHandler() {
   return [this](const RtpPacketReceived& parsed_packet) {
-    worker_thread()->PostTask(
-        SafeTask(worker_thread_safety_, [this, parsed_packet]() {
-          // Deliver the packet anyway to Call to allow Call to do BWE.
-          // Even if there is no media receiver, the packet has still
-          // been received on the network and has been correcly parsed.
-          call_ptr_->Receiver()->DeliverRtpPacket(
-              MediaType::ANY, parsed_packet,
-              /*undemuxable_packet_handler=*/
-              [](const RtpPacketReceived& packet) { return false; });
-        }));
+    RTC_DCHECK_RUN_ON(network_thread());
+    // Deliver the packet anyway to Call to allow Call to do BWE.
+    // Even if there is no media receiver, the packet has still
+    // been received on the network and has been correctly parsed.
+    call_ptr_->Receiver()->DeliverRtpPacket(
+        MediaType::ANY, parsed_packet,
+        /*undemuxable_packet_handler=*/
+        [](const RtpPacketReceived& packet) { return false; });
   };
 }
 

From ca88c7da2099ed61922476d915880c2361a7b4c7 Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Mon, 9 Mar 2026 13:18:12 +0000
Subject: [PATCH 267/472] Remove some unused calls from Video Engine classes

SendCodecsHaLntf, SendCodecRtxTime and SendCodecRtcpMode
were introduced during the send/receive split.
Refactoring has removed the need for them. They can be deleted.

Bug: None
Change-Id: I328a8f8dfcc89743270f87c307a558276bb730a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/455240
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47120}
---
 media/base/fake_media_engine.h     |  3 ---
 media/base/media_channel.h         |  4 ----
 media/engine/webrtc_video_engine.h | 20 --------------------
 3 files changed, 27 deletions(-)

diff --git a/media/base/fake_media_engine.h b/media/base/fake_media_engine.h
index 75e204d361..6cd6e66641 100644
--- a/media/base/fake_media_engine.h
+++ b/media/base/fake_media_engine.h
@@ -749,14 +749,11 @@ class FakeVideoMediaSendChannel
 
   void GenerateSendKeyFrame(uint32_t ssrc,
                             const std::vector<std::string>& rids) override;
-  RtcpMode SendCodecRtcpMode() const override { return RtcpMode::kCompound; }
   void SetSsrcListChangedCallback(
       absl::AnyInvocable<void(const std::set<uint32_t>&)> /* callback */)
       override {}
 
-  bool SendCodecHasLntf() const override { return false; }
   bool SendCodecHasNack() const override { return false; }
-  std::optional<int> SendCodecRtxTime() const override { return std::nullopt; }
   bool GetStats(VideoMediaSendInfo* info) override;
   bool SetOptions(const VideoOptions& options) override;
 
diff --git a/media/base/media_channel.h b/media/base/media_channel.h
index 3cc8f7ba2f..93693ec740 100644
--- a/media/base/media_channel.h
+++ b/media/base/media_channel.h
@@ -982,10 +982,6 @@ class VideoMediaSendChannelInterface : public MediaSendChannelInterface {
   // so that it's getting the send stream stats separately by calling
   // GetStats(), and merges with BandwidthEstimationInfo by itself.
   virtual void FillBitrateInfo(BandwidthEstimationInfo* bwe_info) = 0;
-  // Information queries to support SetReceiverFeedbackParameters
-  virtual RtcpMode SendCodecRtcpMode() const = 0;
-  virtual bool SendCodecHasLntf() const = 0;
-  virtual std::optional<int> SendCodecRtxTime() const = 0;
 };
 
 class VideoMediaReceiveChannelInterface : public MediaReceiveChannelInterface {
diff --git a/media/engine/webrtc_video_engine.h b/media/engine/webrtc_video_engine.h
index d79174f6b0..e6b4b23bdb 100644
--- a/media/engine/webrtc_video_engine.h
+++ b/media/engine/webrtc_video_engine.h
@@ -263,19 +263,6 @@ class WebRtcVideoSendChannel : public MediaChannelUtil,
       uint32_t ssrc,
       scoped_refptr<FrameTransformerInterface> frame_transformer) override;
   // Information queries to support SetReceiverFeedbackParameters
-  RtcpMode SendCodecRtcpMode() const override {
-    RTC_DCHECK_RUN_ON(&thread_checker_);
-    return send_params_.rtcp.reduced_size ? RtcpMode::kReducedSize
-                                          : RtcpMode::kCompound;
-  }
-
-  bool SendCodecHasLntf() const override {
-    RTC_DCHECK_RUN_ON(&thread_checker_);
-    if (!send_codec()) {
-      return false;
-    }
-    return HasLntf(send_codec()->codec);
-  }
   bool SendCodecHasNack() const override {
     RTC_DCHECK_RUN_ON(&thread_checker_);
     if (!send_codec()) {
@@ -283,13 +270,6 @@ class WebRtcVideoSendChannel : public MediaChannelUtil,
     }
     return HasNack(send_codec()->codec);
   }
-  std::optional<int> SendCodecRtxTime() const override {
-    RTC_DCHECK_RUN_ON(&thread_checker_);
-    if (!send_codec()) {
-      return std::nullopt;
-    }
-    return send_codec()->rtx_time;
-  }
 
  private:
   struct ChangedSenderParameters {

From da6f7ffbf033849966c809d8e4a6ea67898f081f Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Wed, 11 Mar 2026 15:14:01 -0700
Subject: [PATCH 268/472] Roll chromium_revision 8fdfcb5d76..94ef474520
 (1597876:1598026)

Change log: https://chromium.googlesource.com/chromium/src/+log/8fdfcb5d76..94ef474520
Full diff: https://chromium.googlesource.com/chromium/src/+/8fdfcb5d76..94ef474520

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/c9202dd8ca..08d96365cc
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/00efde9573..c82693e275
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/bacbfe72d8..ff008b8125
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/22a76fe23b..6fd21700ca
* src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/c8a5676846..e1e57b8776
* src/third_party/fuzztest/src: https://chromium.googlesource.com/external/github.com/google/fuzztest.git/+log/54ed362692..2145ac8a75
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/3ca09e4818..a2bf3f10d5
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/c0a3840504..08b509bdad
* src/third_party/r8/cipd: CoaQUbD7pDX3OuETi_O5wkv_Y198RVIUaSobSEx9zfoC..6BLixpU9eIdUBP0egFPaRGM00adKPdziwUcg5ThtXFYC
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/82543c0c7f..cf631c2a37
DEPS diff: https://chromium.googlesource.com/chromium/src/+/8fdfcb5d76..94ef474520/DEPS

BUG=None

Change-Id: Ib35a4e1539cdbe95349933a6993aeac9dcfb9311
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/455960
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47121}
---
 DEPS | 22 +++++++++++-----------
 1 file changed, 11 insertions(+), 11 deletions(-)

diff --git a/DEPS b/DEPS
index 211c40da09..04bd103bbc 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '8fdfcb5d762df924ac3e77514c178343e2aeead3',
+  'chromium_revision': '94ef474520ff0ce405215917e0915fa57414c48f',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@c9202dd8ca1d4255bd9e505988b945746ddb378a',
+    'https://chromium.googlesource.com/chromium/src/build@08d96365cc4f5c0548c11d2aff921a352c095352',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@fef8d6947fa54202e8a00c58d5ac8fd15780bc33',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@00efde9573f5ecad0307619539c633c4eb805905',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@c82693e275c2c184e2848f462c26e29c9dc588f3',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@bacbfe72d8a48eac176f063aefc517cd84786b91',
+    'https://chromium.googlesource.com/chromium/src/testing@ff008b8125f1295a0d0f3b90bb0c5647a827fe37',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@22a76fe23bae393322386681d4bfa508b0da44ea',
+    'https://chromium.googlesource.com/chromium/src/third_party@6fd21700ca093f45029ddba35e1f38763cfa90e8',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -331,7 +331,7 @@ deps = {
   'src/third_party/clang-format/script':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@c2725e0622e1a86d55f14514f2177a39efea4a0e',
   'src/third_party/compiler-rt/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@c8a567684659fe05ede3a220965b94fd71c4a7cb',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@e1e57b877616abd748cdd63bb78e913e26f5bdfa',
   'src/third_party/libc++/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@7ab65651aed6802d2599dcb7a73b1f82d5179d05',
   'src/third_party/libc++abi/src':
@@ -511,7 +511,7 @@ deps = {
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@3ca09e4818f4362e75c9c224f1d4c2065db75a0a',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@a2bf3f10d5e58d00ebee93d0f312e0fd6882380e',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -593,7 +593,7 @@ deps = {
   'src/third_party/libFuzzer/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt/lib/fuzzer.git@bea408a6e01f0f7e6c82a43121fe3af4506c932e',
   'src/third_party/fuzztest/src':
-    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@54ed362692177e93df0fa202cae404b57694e828',
+    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@2145ac8a757695d4ad68112943641dcce9221e07',
   'src/third_party/libjpeg_turbo':
     'https://chromium.googlesource.com/chromium/deps/libjpeg_turbo.git@d1f5f2393e0d51f840207342ae86e55a86443288',
   'src/third_party/libsrtp':
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'c0a3840504ad618393057eb1fc121810c210dfd2',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '08b509bdad26c91f5f34d9ff97c9362a111f3a72',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -644,7 +644,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/r8',
-              'version': 'CoaQUbD7pDX3OuETi_O5wkv_Y198RVIUaSobSEx9zfoC',
+              'version': '6BLixpU9eIdUBP0egFPaRGM00adKPdziwUcg5ThtXFYC',
           },
       ],
       'condition': 'checkout_android',
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@82543c0c7fa9a6436b46102e26e57e09aefa192d',
+    'https://chromium.googlesource.com/chromium/src/tools@cf631c2a373a1f10bef7dae3d8cf4efd8dc29aba',
 
   'src/third_party/espresso': {
       'packages': [

From 5f1f4d5d8e2e6e486c390137081e65bb41f2848d Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Wed, 11 Mar 2026 19:03:31 -0700
Subject: [PATCH 269/472] Roll chromium_revision 94ef474520..68ba4ea02a
 (1598026:1598147)

Change log: https://chromium.googlesource.com/chromium/src/+log/94ef474520..68ba4ea02a
Full diff: https://chromium.googlesource.com/chromium/src/+/94ef474520..68ba4ea02a

Changed dependencies
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/c82693e275..73655355dd
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/ff008b8125..2c01d5acaa
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/6fd21700ca..1df9fda10d
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/a2bf3f10d5..ee2aee8b5e
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/08b509bdad..3a764e6f63
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/cf631c2a37..a9a47e61f4
DEPS diff: https://chromium.googlesource.com/chromium/src/+/94ef474520..68ba4ea02a/DEPS

BUG=None

Change-Id: Ica7f73869e92fb151d248c8544a6624f3288a61f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456000
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47122}
---
 DEPS | 14 +++++++-------
 1 file changed, 7 insertions(+), 7 deletions(-)

diff --git a/DEPS b/DEPS
index 04bd103bbc..10973337aa 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '94ef474520ff0ce405215917e0915fa57414c48f',
+  'chromium_revision': '68ba4ea02ab80d730e9538c74a9605b99c4a35df',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@c82693e275c2c184e2848f462c26e29c9dc588f3',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@73655355dd4e6dd8b1b989049ba32740d2caf398',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@ff008b8125f1295a0d0f3b90bb0c5647a827fe37',
+    'https://chromium.googlesource.com/chromium/src/testing@2c01d5acaa65b2cc65a60d6b6b21cfbc247aba32',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@6fd21700ca093f45029ddba35e1f38763cfa90e8',
+    'https://chromium.googlesource.com/chromium/src/third_party@1df9fda10dcbc12b9810a0a3b481b5696236fee8',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -511,7 +511,7 @@ deps = {
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@a2bf3f10d5e58d00ebee93d0f312e0fd6882380e',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@ee2aee8b5e104d83a973b7d9be07fdb3681626a7',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '08b509bdad26c91f5f34d9ff97c9362a111f3a72',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '3a764e6f6339ae06c19540682dad4a1a9021f072',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@cf631c2a373a1f10bef7dae3d8cf4efd8dc29aba',
+    'https://chromium.googlesource.com/chromium/src/tools@a9a47e61f490f9242807eac79bb4d46c808652a9',
 
   'src/third_party/espresso': {
       'packages': [

From 815295ac223f5f5a90e5269258c09245ada471d8 Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Wed, 11 Mar 2026 21:12:30 -0700
Subject: [PATCH 270/472] Update WebRTC code version (2026-03-12T04:11:50).

Bug: None
Change-Id: Ib5a517a68d06a7b7f9c468b5e058ef4c8cda3092
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456021
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47123}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index 2c7bb48be2..b2c4e5e121 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-11T04:03:48";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-12T04:11:50";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From 4420b583ec6fa525bad04e2b5c7189c7ecfb6b14 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Thu, 12 Mar 2026 01:04:24 -0700
Subject: [PATCH 271/472] Roll chromium_revision 68ba4ea02a..42535d5df1
 (1598147:1598261)

Change log: https://chromium.googlesource.com/chromium/src/+log/68ba4ea02a..42535d5df1
Full diff: https://chromium.googlesource.com/chromium/src/+/68ba4ea02a..42535d5df1

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/08d96365cc..59b2d010a9
* src/buildtools/linux64: git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9..git_revision:83c605e5dcc3db2945291da1760ca9abd6343338
* src/buildtools/mac: git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9..git_revision:83c605e5dcc3db2945291da1760ca9abd6343338
* src/buildtools/win: git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9..git_revision:83c605e5dcc3db2945291da1760ca9abd6343338
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/73655355dd..93aa68ef77
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/1df9fda10d..3aceea1e90
* src/third_party/androidx/cipd: A6oLcxGdFmfrPoaAcVYoVYHlkB3InvLKiIKnIgmGJVcC..Mn6UxK4fOcnps2oJ1ikX7T1Nf93dtMBkX68uqIyoQ0sC
* src/third_party/cpuinfo/src: https://chromium.googlesource.com/external/github.com/pytorch/cpuinfo.git/+log/7364b490b5..7607ca5004
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/c4eb3c4f4c..2a2456c873
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/ee2aee8b5e..4d052614c1
* src/third_party/r8/cipd: 6BLixpU9eIdUBP0egFPaRGM00adKPdziwUcg5ThtXFYC..AVViRP05v8_iDI-43CYldqOEnoNz2SUTr_nBxP_z-NwC
* src/third_party/tflite/src: https://chromium.googlesource.com/external/github.com/tensorflow/tensorflow.git/+log/f920077085..d20369f4f2
* src/third_party/xnnpack/src: https://chromium.googlesource.com/external/github.com/google/XNNPACK.git/+log/26feacc3eb..15e04ff8b6
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/a9a47e61f4..a06f75e8e5
DEPS diff: https://chromium.googlesource.com/chromium/src/+/68ba4ea02a..42535d5df1/DEPS

BUG=None

Change-Id: Ia5cba7a6e2796ccdfa654b372fe6c348be7c42f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456080
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47124}
---
 DEPS | 30 +++++++++++++++---------------
 1 file changed, 15 insertions(+), 15 deletions(-)

diff --git a/DEPS b/DEPS
index 10973337aa..6c49e57ee5 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '68ba4ea02ab80d730e9538c74a9605b99c4a35df',
+  'chromium_revision': '42535d5df198ffb06207d77688ee4e1b98fe1372',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@08d96365cc4f5c0548c11d2aff921a352c095352',
+    'https://chromium.googlesource.com/chromium/src/build@59b2d010a999e350f13f7009402f9006c17ef618',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@fef8d6947fa54202e8a00c58d5ac8fd15780bc33',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,19 +78,19 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@73655355dd4e6dd8b1b989049ba32740d2caf398',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@93aa68ef77a3e3ab9c018e69a20fa1874364f4fd',
     'condition': 'checkout_ios',
   },
   'src/testing':
     'https://chromium.googlesource.com/chromium/src/testing@2c01d5acaa65b2cc65a60d6b6b21cfbc247aba32',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@1df9fda10dcbc12b9810a0a3b481b5696236fee8',
+    'https://chromium.googlesource.com/chromium/src/third_party@3aceea1e90f83044f80492efc7a1376a6c4850fa',
 
   'src/buildtools/linux64': {
     'packages': [
       {
         'package': 'gn/gn/linux-${{arch}}',
-        'version': 'git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9',
+        'version': 'git_revision:83c605e5dcc3db2945291da1760ca9abd6343338',
       }
     ],
     'dep_type': 'cipd',
@@ -100,7 +100,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/mac-${{arch}}',
-        'version': 'git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9',
+        'version': 'git_revision:83c605e5dcc3db2945291da1760ca9abd6343338',
       }
     ],
     'dep_type': 'cipd',
@@ -110,7 +110,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/windows-amd64',
-        'version': 'git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9',
+        'version': 'git_revision:83c605e5dcc3db2945291da1760ca9abd6343338',
       }
     ],
     'dep_type': 'cipd',
@@ -511,7 +511,7 @@ deps = {
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@ee2aee8b5e104d83a973b7d9be07fdb3681626a7',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@4d052614c1fdd60815db6ce6dc7d63ec391c0988',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -644,7 +644,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/r8',
-              'version': '6BLixpU9eIdUBP0egFPaRGM00adKPdziwUcg5ThtXFYC',
+              'version': 'AVViRP05v8_iDI-43CYldqOEnoNz2SUTr_nBxP_z-NwC',
           },
       ],
       'condition': 'checkout_android',
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@a9a47e61f490f9242807eac79bb4d46c808652a9',
+    'https://chromium.googlesource.com/chromium/src/tools@a06f75e8e5504a8535903a2a941b2cc361cd54ea',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'A6oLcxGdFmfrPoaAcVYoVYHlkB3InvLKiIKnIgmGJVcC',
+          'version': 'Mn6UxK4fOcnps2oJ1ikX7T1Nf93dtMBkX68uqIyoQ0sC',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -786,7 +786,7 @@ deps = {
   },
 
   'src/third_party/tflite/src':
-    Var('chromium_git') + '/external/github.com/tensorflow/tensorflow.git' + '@' + 'f9200770858f5721b7fa95db7aa3e4cd712fe9df',
+    Var('chromium_git') + '/external/github.com/tensorflow/tensorflow.git' + '@' + 'd20369f4f226598800e0d406db567d2d1c17fa95',
 
   'src/third_party/turbine/cipd': {
       'packages': [
@@ -852,7 +852,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/google/pthreadpool.git' + '@' + '9003ee6c137cea3b94161bd5c614fb43be523ee1',
 
   'src/third_party/xnnpack/src':
-    Var('chromium_git') + '/external/github.com/google/XNNPACK.git' + '@' + '26feacc3ebe30d1f3a82e1ff3938ef9859d74640',
+    Var('chromium_git') + '/external/github.com/google/XNNPACK.git' + '@' + '15e04ff8b61e0bc2118bad268447be5b2075aa66',
 
   'src/third_party/farmhash/src':
     Var('chromium_git') + '/external/github.com/google/farmhash.git' + '@' + '816a4ae622e964763ca0862d9dbd19324a1eaf45',
@@ -861,10 +861,10 @@ deps = {
     Var('chromium_git') + '/external/github.com/google/ruy.git' + '@' + '2af88863614a8298689cc52b1a47b3fcad7be835',
 
   'src/third_party/cpuinfo/src':
-    Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7364b490b5f78d58efe23ea76e74210fd6c3c76f',
+    Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7607ca500436b37ad23fb8d18614bec7796b68a7',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + 'c4eb3c4f4c91511eb5c76d608e07dde2a1cc8b23',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '2a2456c873d6f4bf43a9adf919f13ad917d57c2d',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From 77cd0a7759c94e1ad457c8e4daa2e6ff53c12e5e Mon Sep 17 00:00:00 2001
From: Jeremy Leconte <jleconte@webrtc.org>
Date: Thu, 12 Mar 2026 08:12:12 +0100
Subject: [PATCH 272/472] Make the targets leaking through
 'libjingle_peerconnection_api' public

Change-Id: Id8fbe6d5476c1c7559a72afa79c658d514932813
Bug: webrtc:467294026
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456061
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#47125}
---
 api/BUILD.gn | 9 +++++++++
 1 file changed, 9 insertions(+)

diff --git a/api/BUILD.gn b/api/BUILD.gn
index 4759de57c8..876043a072 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -366,6 +366,7 @@ rtc_library("jsep") {
 }
 
 rtc_library("data_channel_interface") {
+  visibility = [ "*" ]
   sources = [
     "data_channel_interface.cc",
     "data_channel_interface.h",
@@ -382,6 +383,7 @@ rtc_library("data_channel_interface") {
 }
 
 rtc_library("legacy_stats_types") {
+  visibility = [ "*" ]
   sources = [
     "legacy_stats_types.cc",
     "legacy_stats_types.h",
@@ -400,6 +402,7 @@ rtc_library("legacy_stats_types") {
 }
 
 rtc_library("peer_connection_interface") {
+  visibility = [ "*" ]
   sources = [
     "peer_connection_interface.cc",
     "peer_connection_interface.h",
@@ -474,6 +477,7 @@ rtc_library("peer_connection_interface") {
 }
 
 rtc_library("rtp_receiver_interface") {
+  visibility = [ "*" ]
   sources = [
     "rtp_receiver_interface.cc",
     "rtp_receiver_interface.h",
@@ -492,6 +496,7 @@ rtc_library("rtp_receiver_interface") {
 }
 
 rtc_library("rtp_transceiver_interface") {
+  visibility = [ "*" ]
   sources = [
     "rtp_transceiver_interface.cc",
     "rtp_transceiver_interface.h",
@@ -512,6 +517,7 @@ rtc_library("rtp_transceiver_interface") {
 }
 
 rtc_library("set_local_description_observer_interface") {
+  visibility = [ "*" ]
   sources = [ "set_local_description_observer_interface.h" ]
   deps = [
     ":ref_count",
@@ -520,6 +526,7 @@ rtc_library("set_local_description_observer_interface") {
 }
 
 rtc_library("set_remote_description_observer_interface") {
+  visibility = [ "*" ]
   sources = [ "set_remote_description_observer_interface.h" ]
   deps = [
     ":ref_count",
@@ -528,6 +535,7 @@ rtc_library("set_remote_description_observer_interface") {
 }
 
 rtc_library("uma_metrics") {
+  visibility = [ "*" ]
   sources = [ "uma_metrics.h" ]
   deps = [
     ":ref_count",
@@ -536,6 +544,7 @@ rtc_library("uma_metrics") {
 }
 
 rtc_library("video_track_source_proxy_factory") {
+  visibility = [ "*" ]
   sources = [ "video_track_source_proxy_factory.h" ]
   deps = [
     ":media_stream_interface",

From ed82d23c962839237b0db0a8128e12424762f63a Mon Sep 17 00:00:00 2001
From: Philipp Hancke <philipp.hancke@googlemail.com>
Date: Wed, 11 Mar 2026 22:14:18 +0100
Subject: [PATCH 273/472] build: clean up api/BUILD.gn

using
  xargs tools_webrtc/gn_check_autofix.py -r api/BUILD.gn
followed by
  tools_webrtc/gn_check_autofix -C out/Default

The removal of the "call" dep from the "enable_media" target
and "stats/rtcstats" from the "create_modular_peer_connection_factory"
targets were reverted due to linker failures.

Changes to the libjingle_peerconnection legacy target were reverted.

Bug: webrtc:467294026
Change-Id: Iac9c6b0bdac23cfd5441bd0d9968202c15f70b1e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/455821
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47126}
---
 api/BUILD.gn | 137 +++++++--------------------------------------------
 1 file changed, 19 insertions(+), 118 deletions(-)

diff --git a/api/BUILD.gn b/api/BUILD.gn
index 876043a072..fea2f3ec62 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -126,7 +126,6 @@ rtc_library("rtp_headers") {
     "rtp_headers.h",
   ]
   deps = [
-    ":array_view",
     "../rtc_base:checks",
     "../rtc_base/system:rtc_export",
     "units:timestamp",
@@ -142,7 +141,6 @@ rtc_library("rtp_packet_info") {
     "rtp_packet_infos.h",
   ]
   deps = [
-    ":array_view",
     ":make_ref_counted",
     ":refcountedbase",
     ":rtp_headers",
@@ -168,7 +166,6 @@ rtc_library("media_stream_interface") {
   ]
   deps = [
     ":audio_options_api",
-    ":make_ref_counted",
     ":ref_count",
     ":rtp_parameters",
     ":scoped_refptr",
@@ -199,7 +196,6 @@ rtc_library("candidate") {
     "../rtc_base:crc32",
     "../rtc_base:crypto_random",
     "../rtc_base:ip_address",
-    "../rtc_base:logging",
     "../rtc_base:net_helper",
     "../rtc_base:network_constants",
     "../rtc_base:socket_address",
@@ -227,9 +223,7 @@ rtc_source_set("ice_transport_interface") {
   deps = [
     ":async_dns_resolver",
     ":local_network_access_permission",
-    ":packet_socket_factory",
     ":ref_count",
-    ":rtc_error",
     ":scoped_refptr",
     "environment",
     "rtc_event_log",
@@ -265,10 +259,7 @@ rtc_library("dtmf_sender_interface") {
   visibility = [ "*" ]
 
   sources = [ "dtmf_sender_interface.h" ]
-  deps = [
-    ":media_stream_interface",
-    ":ref_count",
-  ]
+  deps = [ ":ref_count" ]
 }
 
 rtc_library("rtp_sender_interface") {
@@ -537,10 +528,7 @@ rtc_library("set_remote_description_observer_interface") {
 rtc_library("uma_metrics") {
   visibility = [ "*" ]
   sources = [ "uma_metrics.h" ]
-  deps = [
-    ":ref_count",
-    ":rtc_error",
-  ]
+  deps = []
 }
 
 rtc_library("video_track_source_proxy_factory") {
@@ -548,8 +536,6 @@ rtc_library("video_track_source_proxy_factory") {
   sources = [ "video_track_source_proxy_factory.h" ]
   deps = [
     ":media_stream_interface",
-    ":ref_count",
-    ":rtc_error",
     ":scoped_refptr",
     "../rtc_base:threading",
     "../rtc_base/system:rtc_export",
@@ -690,16 +676,12 @@ rtc_library("frame_transformer_interface") {
   ]
   deps = [
     ":array_view",
-    ":make_ref_counted",
-    ":payload_type",
     ":ref_count",
     ":scoped_refptr",
     "../rtc_base:checks",
-    "../rtc_base:refcount",
     "../rtc_base/system:rtc_export",
     "units:time_delta",
     "units:timestamp",
-    "video:encoded_frame",
     "video:video_frame_metadata",
   ]
 }
@@ -750,7 +732,6 @@ rtc_source_set("async_dns_resolver") {
   visibility = [ "*" ]
   sources = [ "async_dns_resolver.h" ]
   deps = [
-    "../rtc_base:checks",
     "../rtc_base:socket_address",
     "../rtc_base/system:rtc_export",
     "//third_party/abseil-cpp/absl/functional:any_invocable",
@@ -761,9 +742,7 @@ rtc_source_set("local_network_access_permission") {
   visibility = [ "*" ]
   sources = [ "local_network_access_permission.h" ]
   deps = [
-    "../rtc_base:checks",
     "../rtc_base:socket_address",
-    "../rtc_base/system:rtc_export",
     "//third_party/abseil-cpp/absl/functional:any_invocable",
   ]
 }
@@ -803,7 +782,6 @@ rtc_source_set("video_quality_analyzer_api") {
     "../rtc_base:checks",
     "video:encoded_image",
     "video:video_frame",
-    "video:video_rtp_headers",
     "video_codecs:video_codecs_api",
     "//third_party/abseil-cpp/absl/strings:string_view",
   ]
@@ -921,39 +899,11 @@ rtc_source_set("peer_connection_quality_test_fixture_api") {
   sources = [ "test/peerconnection_quality_test_fixture.h" ]
 
   deps = [
-    ":array_view",
-    ":audio_quality_analyzer_api",
-    ":fec_controller_api",
-    ":frame_generator_api",
-    ":function_view",
-    ":media_stream_interface",
-    ":network_state_predictor_api",
-    ":packet_socket_factory",
-    ":rtp_parameters",
-    ":simulated_network_api",
     ":stats_observer_interface",
     ":track_id_stream_info_map",
-    ":video_quality_analyzer_api",
-    "../media:media_constants",
-    "../rtc_base:checks",
-    "../rtc_base:network",
-    "../rtc_base:rtc_certificate_generator",
-    "../rtc_base:ssl",
-    "../rtc_base:stringutils",
-    "../rtc_base:threading",
-    "../test:fileutils",
-    "audio:audio_mixer_api",
-    "audio:audio_processing",
-    "rtc_event_log",
-    "task_queue",
-    "test/pclf:media_configuration",
     "test/pclf:media_quality_test_params",
     "test/pclf:peer_configurer",
-    "test/video:video_frame_writer",
-    "transport:network_control",
     "units:time_delta",
-    "video:video_frame",
-    "video_codecs:video_codecs_api",
     "//third_party/abseil-cpp/absl/base:core_headers",
     "//third_party/abseil-cpp/absl/memory",
     "//third_party/abseil-cpp/absl/strings:string_view",
@@ -1001,10 +951,6 @@ if (rtc_include_tests) {
         ":network_state_predictor_api",
         ":rtp_parameters",
         ":simulated_network_api",
-        "../call:fake_network",
-        "../call:rtp_interfaces",
-        "../test:test_common",
-        "../test:video_test_common",
         "../video/config:encoder_config",
         "transport:bitrate_settings",
         "transport:network_control",
@@ -1059,7 +1005,6 @@ rtc_library("create_frame_generator") {
     "../system_wrappers",
     "../test:frame_generator_impl",
     "environment",
-    "environment:environment_factory",
     "//third_party/abseil-cpp/absl/base:nullability",
     "//third_party/abseil-cpp/absl/strings:string_view",
   ]
@@ -1127,12 +1072,10 @@ rtc_source_set("rtc_stats_api") {
   ]
 
   deps = [
-    ":make_ref_counted",
     ":ref_count",
+    ":refcountedbase",
     ":scoped_refptr",
-    "../api:refcountedbase",
     "../rtc_base:checks",
-    "../rtc_base:refcount",
     "../rtc_base/system:rtc_export",
     "units:timestamp",
   ]
@@ -1158,11 +1101,7 @@ rtc_library("transport_api") {
     "call/transport.cc",
     "call/transport.h",
   ]
-  deps = [
-    ":array_view",
-    ":refcountedbase",
-    ":scoped_refptr",
-  ]
+  deps = [ ":array_view" ]
 }
 
 rtc_source_set("bitrate_allocation") {
@@ -1179,7 +1118,6 @@ rtc_source_set("simulated_network_api") {
   visibility = [ "*" ]
   sources = [ "test/simulated_network.h" ]
   deps = [
-    "../rtc_base:random",
     "transport:ecn_marking",
     "units:data_rate",
     "units:data_size",
@@ -1198,13 +1136,11 @@ rtc_library("network_emulation_manager_api") {
   deps = [
     ":array_view",
     ":field_trials_view",
-    ":packet_socket_factory",
     ":peer_network_dependencies",
     ":simulated_network_api",
     ":time_controller",
     "../rtc_base:checks",
     "../rtc_base:ip_address",
-    "../rtc_base:network",
     "../rtc_base:network_constants",
     "../rtc_base:socket_address",
     "../test/network:simulated_network",
@@ -1227,11 +1163,9 @@ rtc_library("time_controller") {
     ":function_view",
     "../rtc_base:socket_server",
     "../rtc_base:threading",
-    "../rtc_base/synchronization:yield_policy",
     "../system_wrappers",
     "task_queue",
     "units:time_delta",
-    "units:timestamp",
     "//third_party/abseil-cpp/absl/strings:string_view",
   ]
 }
@@ -1279,18 +1213,13 @@ rtc_library("ice_transport_factory") {
   deps = [
     ":ice_transport_interface",
     ":make_ref_counted",
-    ":packet_socket_factory",
     ":scoped_refptr",
     ":sequence_checker",
-    "../p2p:connection",
     "../p2p:ice_transport_internal",
     "../p2p:p2p_constants",
     "../p2p:p2p_transport_channel",
-    "../p2p:port_allocator",
     "../rtc_base:macromagic",
-    "../rtc_base:threading",
     "../rtc_base/system:rtc_export",
-    "rtc_event_log",
   ]
 }
 
@@ -1328,9 +1257,7 @@ rtc_library("datagram_connection") {
     ":candidate",
     ":ref_count",
     "../p2p:transport_description",
-    "../rtc_base:macromagic",
     "../rtc_base/system:rtc_export",
-    "environment",
     "units:timestamp",
     "//third_party/abseil-cpp/absl/functional:any_invocable",
     "//third_party/abseil-cpp/absl/strings:string_view",
@@ -1346,11 +1273,9 @@ rtc_library("datagram_connection_factory") {
   deps = [
     ":datagram_connection",
     ":make_ref_counted",
-    ":ref_count",
     ":scoped_refptr",
     "../p2p:port_allocator",
     "../pc:datagram_connection_internal",
-    "../rtc_base:macromagic",
     "../rtc_base:ssl",
     "../rtc_base/system:rtc_export",
     "environment",
@@ -1369,8 +1294,6 @@ if (rtc_include_tests) {
       ]
 
       deps = [
-        ":scoped_refptr",
-        "../modules/audio_processing",
         "../modules/audio_processing:audioproc_f_impl",
         "audio:audio_processing",
         "audio:builtin_audio_processing_builder",
@@ -1388,7 +1311,6 @@ if (rtc_include_tests) {
       deps = [
         ":neteq_simulator_api",
         "../modules/audio_coding:neteq_test_factory",
-        "../rtc_base:checks",
         "neteq:neteq_api",
         "//third_party/abseil-cpp/absl/flags:flag",
         "//third_party/abseil-cpp/absl/flags:parse",
@@ -1425,9 +1347,9 @@ if (rtc_include_tests) {
       "test/videocodec_test_stats.h",
     ]
     deps = [
-      "../api/units:data_rate",
-      "../api/units:frequency",
       "../rtc_base:stringutils",
+      "units:data_rate",
+      "units:frequency",
       "video:video_frame_type",
     ]
   }
@@ -1440,7 +1362,6 @@ if (rtc_include_tests) {
       ":field_trials",
       ":videocodec_test_stats_api",
       "../modules/video_coding:codec_globals_headers",
-      "../modules/video_coding:video_codec_interface",
       "video:encoded_image",
       "video:video_frame",
       "video_codecs:video_codecs_api",
@@ -1456,7 +1377,6 @@ if (rtc_include_tests) {
     ]
     deps = [
       ":videocodec_test_fixture_api",
-      "../modules/video_coding:video_codecs_test_framework",
       "../modules/video_coding:videocodec_test_impl",
       "video_codecs:video_codecs_api",
     ]
@@ -1480,7 +1400,7 @@ if (rtc_include_tests) {
     sources = [ "test/mock_audio_sink.h" ]
 
     deps = [
-      "../api:media_stream_interface",
+      ":media_stream_interface",
       "../test:test_support",
     ]
   }
@@ -1532,8 +1452,6 @@ if (rtc_include_tests) {
     deps = [
       ":array_view",
       ":rtp_parameters",
-
-      # For api/crypto/frame_encryptor_interface.h
       "../test:test_support",
       "crypto:frame_encryptor_interface",
     ]
@@ -1567,10 +1485,10 @@ if (rtc_include_tests) {
     testonly = true
     sources = [ "test/mock_encoder_selector.h" ]
     deps = [
-      "../api/video_codecs:video_codecs_api",
       "../test:test_support",
       "units:data_rate",
       "video:render_resolution",
+      "video_codecs:video_codecs_api",
     ]
   }
 
@@ -1583,8 +1501,6 @@ if (rtc_include_tests) {
     ]
     deps = [
       ":array_view",
-      ":make_ref_counted",
-      ":ref_count",
       ":rtp_parameters",
       "../rtc_base:checks",
       "../rtc_base:refcount",
@@ -1601,7 +1517,6 @@ if (rtc_include_tests) {
     ]
     deps = [
       ":array_view",
-      ":make_ref_counted",
       ":rtp_parameters",
       "../rtc_base:checks",
       "crypto:frame_decryptor_interface",
@@ -1643,7 +1558,6 @@ if (rtc_include_tests) {
     sources = [ "test/mock_peerconnectioninterface.h" ]
 
     deps = [
-      ":candidate",
       ":data_channel_event_observer_interface",
       ":data_channel_interface",
       ":dtls_transport_interface",
@@ -1652,24 +1566,21 @@ if (rtc_include_tests) {
       ":make_ref_counted",
       ":media_stream_interface",
       ":peer_connection_interface",
-      ":ref_count",
       ":rtc_error",
       ":rtc_stats_api",
       ":rtp_parameters",
       ":rtp_receiver_interface",
       ":rtp_sender_interface",
       ":rtp_transceiver_interface",
+      ":scoped_refptr",
       ":sctp_transport_interface",
-      ":set_local_description_observer_interface",
       ":set_remote_description_observer_interface",
-      "../api:scoped_refptr",
       "../rtc_base:refcount",
       "../rtc_base:threading",
       "../test:test_support",
       "adaptation:resource_adaptation_api",
       "transport:bandwidth_estimation_settings",
       "transport:bitrate_settings",
-      "transport:network_control",
     ]
   }
 
@@ -1698,7 +1609,6 @@ if (rtc_include_tests) {
     deps = [
       ":array_view",
       ":frame_transformer_interface",
-      ":payload_type",
       "../test:test_support",
       "units:time_delta",
       "units:timestamp",
@@ -1756,9 +1666,9 @@ if (rtc_include_tests) {
       ":rtp_transceiver_direction",
       ":rtp_transceiver_interface",
       ":scoped_refptr",
-      "../api/crypto:frame_decryptor_interface",
       "../rtc_base:refcount",
       "../test:test_support",
+      "crypto:frame_decryptor_interface",
       "crypto:frame_encryptor_interface",
       "transport/rtp:rtp_source",
       "video_codecs:video_codecs_api",
@@ -1773,10 +1683,9 @@ if (rtc_include_tests) {
     deps = [
       ":array_view",
       ":frame_transformer_interface",
-      ":payload_type",
-      "../api/units:timestamp",
       "../test:test_support",
       "units:time_delta",
+      "units:timestamp",
     ]
   }
 
@@ -1788,7 +1697,6 @@ if (rtc_include_tests) {
     deps = [
       ":array_view",
       ":frame_transformer_interface",
-      ":payload_type",
       "../test:test_support",
       "units:time_delta",
       "units:timestamp",
@@ -1802,9 +1710,9 @@ if (rtc_include_tests) {
     sources = [ "test/mock_video_bitrate_allocator.h" ]
 
     deps = [
-      "../api/video:video_bitrate_allocator",
       "../test:test_support",
       "video:video_bitrate_allocation",
+      "video:video_bitrate_allocator",
     ]
   }
 
@@ -1814,10 +1722,10 @@ if (rtc_include_tests) {
     sources = [ "test/mock_video_bitrate_allocator_factory.h" ]
 
     deps = [
-      "../api/video:video_bitrate_allocator_factory",
       "../test:test_support",
       "environment",
       "video:video_bitrate_allocator",
+      "video:video_bitrate_allocator_factory",
       "video_codecs:video_codecs_api",
     ]
   }
@@ -1831,9 +1739,9 @@ if (rtc_include_tests) {
     ]
 
     deps = [
-      "../api/video_codecs:video_codecs_api",
       "../test:test_support",
       "environment",
+      "video_codecs:video_codecs_api",
     ]
   }
 
@@ -1843,10 +1751,10 @@ if (rtc_include_tests) {
     sources = [ "test/mock_video_decoder.h" ]
 
     deps = [
-      "../api/video_codecs:video_codecs_api",
       "../test:test_support",
       "video:encoded_image",
       "video:video_frame",
+      "video_codecs:video_codecs_api",
     ]
   }
 
@@ -1857,11 +1765,11 @@ if (rtc_include_tests) {
 
     deps = [
       ":fec_controller_api",
-      "../api/video_codecs:video_codecs_api",
       "../test:test_support",
       "video:encoded_image",
       "video:video_frame",
       "video:video_frame_type",
+      "video_codecs:video_codecs_api",
     ]
   }
 
@@ -1871,9 +1779,8 @@ if (rtc_include_tests) {
     sources = [ "test/mock_video_track.h" ]
 
     deps = [
-      ":ref_count",
-      "../api:media_stream_interface",
-      "../api:scoped_refptr",
+      ":media_stream_interface",
+      ":scoped_refptr",
       "../rtc_base:refcount",
       "../test:test_support",
       "video:video_frame",
@@ -2037,11 +1944,9 @@ if (rtc_include_tests) {
       ":mock_frame_decryptor",
       ":mock_frame_encryptor",
       ":mock_media_stream_interface",
-      ":mock_packet_socket_factory",
       ":mock_peer_connection_factory_interface",
       ":mock_peerconnectioninterface",
       ":mock_rtp",
-      ":mock_transformable_audio_frame",
       ":mock_transformable_frame",
       ":mock_transformable_video_frame",
       ":mock_video_bitrate_allocator",
@@ -2064,7 +1969,7 @@ rtc_library("field_trials_registry") {
   ]
   deps = [
     ":field_trials_view",
-    "../experiments:registered_field_trials",
+    "../experiments:registered_field_trials",  # keep
     "../rtc_base:checks",
     "../rtc_base:logging",
     "../rtc_base/containers:flat_set",
@@ -2115,14 +2020,10 @@ rtc_library("frame_transformer_factory") {
   ]
   deps = [
     ":frame_transformer_interface",
-    ":ref_count",
-    ":scoped_refptr",
     "../audio",
     "../modules/rtp_rtcp",
     "../rtc_base:checks",
     "../rtc_base/system:rtc_export",
-    "video:encoded_frame",
-    "video:video_frame_metadata",
   ]
 }
 

From ba64784ea6d67c58275b0e732d0cad4027684b49 Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Thu, 12 Mar 2026 08:20:08 +0000
Subject: [PATCH 274/472] pc: Enhance stability integration test
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

with AbslStringify and prioritized checks

- Added AbslStringify to FactorySignature::Id to support logging its
   name.
- Updated BasicOfferAnswerPayloadTypesStable to check codec content
  (unordered) before checking sequence, providing better failure
  signals.
- Enhanced failure messages to include the Factory ID.
- Added a unit test for DumpAsResultingCodecList to verify its output.

Bug: webrtc:397895867
Change-Id: Ia6fe1ca3dc6f5154b597c132969fef3574c4b59d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456062
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47127}
---
 ...er_connection_stability_integrationtest.cc | 77 +++++++++++++++++--
 1 file changed, 69 insertions(+), 8 deletions(-)

diff --git a/pc/peer_connection_stability_integrationtest.cc b/pc/peer_connection_stability_integrationtest.cc
index 8d3052215b..eb8948b5df 100644
--- a/pc/peer_connection_stability_integrationtest.cc
+++ b/pc/peer_connection_stability_integrationtest.cc
@@ -48,6 +48,7 @@ namespace {
 using ::testing::ElementsAreArray;
 using ::testing::Eq;
 using ::testing::Not;
+using ::testing::UnorderedElementsAreArray;
 
 class FactorySignature {
  public:
@@ -61,6 +62,28 @@ class FactorySignature {
     kWebRtcAndroid,
     kGoogleInternal,
   };
+
+  template <typename Sink>
+  friend void AbslStringify(Sink& sink, Id id) {
+    switch (id) {
+      case Id::kNotRecognized:
+        sink.Append("kNotRecognized");
+        break;
+      case Id::kWebRtcTipOfTree:
+        sink.Append("kWebRtcTipOfTree");
+        break;
+      case Id::kWebRtcMoreConfigs1:
+        sink.Append("kWebRtcMoreConfigs1");
+        break;
+      case Id::kWebRtcAndroid:
+        sink.Append("kWebRtcAndroid");
+        break;
+      case Id::kGoogleInternal:
+        sink.Append("kGoogleInternal");
+        break;
+    }
+  }
+
   Id id() { return id_; }
   FactorySignature() {
     ExtractSignatureStrings();
@@ -377,10 +400,7 @@ class PeerConnectionIntegrationTest : public PeerConnectionIntegrationBaseTest {
                                        std::vector<std::string> callee_local,
                                        std::vector<std::string> callee_remote) {
     StringBuilder sb;
-    // TODO: issues.webrtc.org/397895867 - change kChangeThis to the name of
-    // the value. Requires adding an AbslStringifier to the enum.
-    sb << "\n{" << ".factory_id = FactorySignature::Id::kChangeThis"
-       << static_cast<int>(id) << ",\n"
+    sb << "\n{" << ".factory_id = FactorySignature::Id::" << id << ",\n"
        << ".caller_local = {";
     for (const std::string& str : caller_local) {
       sb << "\"" << str << "\",\n";
@@ -940,13 +960,54 @@ TEST_F(PeerConnectionIntegrationTest, BasicOfferAnswerPayloadTypesStable) {
 
   const ResultingCodecList& this_golden = *this_golden_it;
   EXPECT_THAT(CodecList(*caller()->pc()->local_description()),
-              ElementsAreArray(this_golden.caller_local));
+              UnorderedElementsAreArray(this_golden.caller_local))
+      << "Factory ID: " << factory_signature.id();
   EXPECT_THAT(CodecList(*caller()->pc()->remote_description()),
-              ElementsAreArray(this_golden.caller_remote));
+              UnorderedElementsAreArray(this_golden.caller_remote))
+      << "Factory ID: " << factory_signature.id();
   EXPECT_THAT(CodecList(*callee()->pc()->local_description()),
-              ElementsAreArray(this_golden.callee_local));
+              UnorderedElementsAreArray(this_golden.callee_local))
+      << "Factory ID: " << factory_signature.id();
   EXPECT_THAT(CodecList(*callee()->pc()->remote_description()),
-              ElementsAreArray(this_golden.callee_remote));
+              UnorderedElementsAreArray(this_golden.callee_remote))
+      << "Factory ID: " << factory_signature.id();
+
+  if (HasFailure()) {
+    return;
+  }
+
+  EXPECT_THAT(CodecList(*caller()->pc()->local_description()),
+              ElementsAreArray(this_golden.caller_local))
+      << "Factory ID: " << factory_signature.id();
+  EXPECT_THAT(CodecList(*caller()->pc()->remote_description()),
+              ElementsAreArray(this_golden.caller_remote))
+      << "Factory ID: " << factory_signature.id();
+  EXPECT_THAT(CodecList(*callee()->pc()->local_description()),
+              ElementsAreArray(this_golden.callee_local))
+      << "Factory ID: " << factory_signature.id();
+  EXPECT_THAT(CodecList(*callee()->pc()->remote_description()),
+              ElementsAreArray(this_golden.callee_remote))
+      << "Factory ID: " << factory_signature.id();
+}
+
+TEST_F(PeerConnectionIntegrationTest,
+       DumpAsResultingCodecListProducesExpectedOutput) {
+  std::vector<std::string> caller_local = {"cl1"};
+  std::vector<std::string> caller_remote = {"cr1"};
+  std::vector<std::string> callee_local = {"ce_l1"};
+  std::vector<std::string> callee_remote = {"ce_r1"};
+
+  std::string output = DumpAsResultingCodecList(
+      FactorySignature::Id::kWebRtcTipOfTree, caller_local, caller_remote,
+      callee_local, callee_remote);
+
+  EXPECT_THAT(output,
+              Eq("\n{.factory_id = FactorySignature::Id::kWebRtcTipOfTree,\n"
+                 ".caller_local = {\"cl1\",\n},\n"
+                 " .caller_remote = {\"cr1\",\n},\n"
+                 " .callee_local = {\"ce_l1\",\n},\n"
+                 " .callee_remote = {\"ce_r1\",\n"
+                 "}}\n"));
 }
 
 }  // namespace

From 9c1612bf5bb8e72364e6c27a10763a83c59733c4 Mon Sep 17 00:00:00 2001
From: Anton Podavalov <tonypo@google.com>
Date: Wed, 11 Mar 2026 16:20:56 -0700
Subject: [PATCH 275/472] Encapsulate and make RTP sequence number available in
 RtpPacketInfo.

Bug: b/462514208
Change-Id: I63f9235253f229a8059f752d6186409fc1eb24f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452722
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47128}
---
 api/BUILD.gn                    |   2 +
 api/rtp_packet_info.cc          |  39 +++++++---
 api/rtp_packet_info.h           |  20 ++++--
 api/rtp_packet_info_unittest.cc | 121 ++++++++++++++++++++++++++------
 4 files changed, 145 insertions(+), 37 deletions(-)

diff --git a/api/BUILD.gn b/api/BUILD.gn
index fea2f3ec62..8c62fe2a0d 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -145,6 +145,7 @@ rtc_library("rtp_packet_info") {
     ":refcountedbase",
     ":rtp_headers",
     ":scoped_refptr",
+    "../modules/rtp_rtcp:rtp_rtcp_format",
     "../rtc_base/system:rtc_export",
     "units:time_delta",
     "units:timestamp",
@@ -1871,6 +1872,7 @@ if (rtc_include_tests) {
       "../media:media_constants",
       "../media:rid_description",
       "../media:stream_params",
+      "../modules/rtp_rtcp:rtp_rtcp_format",
       "../p2p:p2p_constants",
       "../p2p:transport_description",
       "../p2p:transport_info",
diff --git a/api/rtp_packet_info.cc b/api/rtp_packet_info.cc
index 557247b523..6bbb135a5d 100644
--- a/api/rtp_packet_info.cc
+++ b/api/rtp_packet_info.cc
@@ -18,24 +18,48 @@
 
 #include "api/rtp_headers.h"
 #include "api/units/timestamp.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
 
 namespace webrtc {
 
 RtpPacketInfo::RtpPacketInfo()
-    : ssrc_(0), rtp_timestamp_(0), receive_time_(Timestamp::MinusInfinity()) {}
+    : sequence_number_(0),
+      ssrc_(0),
+      rtp_timestamp_(0),
+      receive_time_(Timestamp::MinusInfinity()) {}
+
+RtpPacketInfo::RtpPacketInfo(const RtpPacketReceived& rtp_packet)
+    : sequence_number_(rtp_packet.SequenceNumber()),
+      ssrc_(rtp_packet.Ssrc()),
+      csrcs_(rtp_packet.Csrcs()),
+      rtp_timestamp_(rtp_packet.Timestamp()),
+      receive_time_(rtp_packet.arrival_time()) {
+  AudioLevel audio_level;
+  if (rtp_packet.GetExtension<AudioLevelExtension>(&audio_level)) {
+    audio_level_ = audio_level.level();
+  }
+
+  AbsoluteCaptureTime capture_time;
+  if (rtp_packet.GetExtension<AbsoluteCaptureTimeExtension>(&capture_time)) {
+    absolute_capture_time_ = std::move(capture_time);
+  }
+}
 
 RtpPacketInfo::RtpPacketInfo(uint32_t ssrc,
                              std::vector<uint32_t> csrcs,
                              uint32_t rtp_timestamp,
                              Timestamp receive_time)
-    : ssrc_(ssrc),
+    : sequence_number_(0),
+      ssrc_(ssrc),
       csrcs_(std::move(csrcs)),
       rtp_timestamp_(rtp_timestamp),
       receive_time_(receive_time) {}
 
 RtpPacketInfo::RtpPacketInfo(const RTPHeader& rtp_header,
                              Timestamp receive_time)
-    : ssrc_(rtp_header.ssrc),
+    : sequence_number_(rtp_header.sequenceNumber),
+      ssrc_(rtp_header.ssrc),
       rtp_timestamp_(rtp_header.timestamp),
       receive_time_(receive_time) {
   const auto& extension = rtp_header.extension;
@@ -50,13 +74,6 @@ RtpPacketInfo::RtpPacketInfo(const RTPHeader& rtp_header,
   absolute_capture_time_ = extension.absolute_capture_time;
 }
 
-bool operator==(const RtpPacketInfo& lhs, const RtpPacketInfo& rhs) {
-  return (lhs.ssrc() == rhs.ssrc()) && (lhs.csrcs() == rhs.csrcs()) &&
-         (lhs.rtp_timestamp() == rhs.rtp_timestamp()) &&
-         (lhs.receive_time() == rhs.receive_time()) &&
-         (lhs.audio_level() == rhs.audio_level()) &&
-         (lhs.absolute_capture_time() == rhs.absolute_capture_time()) &&
-         (lhs.local_capture_clock_offset() == rhs.local_capture_clock_offset());
-}
+bool operator==(const RtpPacketInfo& lhs, const RtpPacketInfo& rhs) = default;
 
 }  // namespace webrtc
diff --git a/api/rtp_packet_info.h b/api/rtp_packet_info.h
index 4059108c83..943975b319 100644
--- a/api/rtp_packet_info.h
+++ b/api/rtp_packet_info.h
@@ -23,6 +23,8 @@
 
 namespace webrtc {
 
+class RtpPacketReceived;
+
 //
 // Structure to hold information about a received `RtpPacket`. It is primarily
 // used to carry per-packet information from when a packet is received until
@@ -32,11 +34,17 @@ class RTC_EXPORT RtpPacketInfo {
  public:
   RtpPacketInfo();
 
+  explicit RtpPacketInfo(const RtpPacketReceived& rtp_packet);
+
+  // TODO: bugs.webrtc.org/42225366 - The constructor will be deprecated. Use
+  // RtpPacketInfo(const RtpPacketReceived& rtp_packet).
   RtpPacketInfo(uint32_t ssrc,
                 std::vector<uint32_t> csrcs,
                 uint32_t rtp_timestamp,
                 Timestamp receive_time);
 
+  // TODO: bugs.webrtc.org/42225366 - The constructor will be deprecated. Use
+  // RtpPacketInfo(const RtpPacketReceived& rtp_packet).
   RtpPacketInfo(const RTPHeader& rtp_header, Timestamp receive_time);
 
   RtpPacketInfo(const RtpPacketInfo& other) = default;
@@ -47,6 +55,9 @@ class RTC_EXPORT RtpPacketInfo {
   uint32_t ssrc() const { return ssrc_; }
   void set_ssrc(uint32_t value) { ssrc_ = value; }
 
+  uint16_t sequence_number() const { return sequence_number_; }
+  void set_sequence_number(uint16_t value) { sequence_number_ = value; }
+
   const std::vector<uint32_t>& csrcs() const { return csrcs_; }
   void set_csrcs(std::vector<uint32_t> value) { csrcs_ = std::move(value); }
 
@@ -80,9 +91,12 @@ class RTC_EXPORT RtpPacketInfo {
     return *this;
   }
 
+  friend bool operator==(const RtpPacketInfo& lhs, const RtpPacketInfo& rhs);
+
  private:
   // Fields from the RTP header:
   // https://tools.ietf.org/html/rfc3550#section-5.1
+  uint16_t sequence_number_;
   uint32_t ssrc_;
   std::vector<uint32_t> csrcs_;
   uint32_t rtp_timestamp_;
@@ -106,12 +120,6 @@ class RTC_EXPORT RtpPacketInfo {
   std::optional<TimeDelta> local_capture_clock_offset_;
 };
 
-bool operator==(const RtpPacketInfo& lhs, const RtpPacketInfo& rhs);
-
-inline bool operator!=(const RtpPacketInfo& lhs, const RtpPacketInfo& rhs) {
-  return !(lhs == rhs);
-}
-
 }  // namespace webrtc
 
 #endif  // API_RTP_PACKET_INFO_H_
diff --git a/api/rtp_packet_info_unittest.cc b/api/rtp_packet_info_unittest.cc
index 9aea11d24c..c61d1bdd0d 100644
--- a/api/rtp_packet_info_unittest.cc
+++ b/api/rtp_packet_info_unittest.cc
@@ -17,9 +17,29 @@
 #include "api/rtp_headers.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "rtc_base/checks.h"
+#include "test/gmock.h"
 #include "test/gtest.h"
 
 namespace webrtc {
+namespace {
+
+using ::testing::ElementsAreArray;
+using ::testing::Optional;
+
+template <typename ExtensionType,
+          typename ExtensionValue = ExtensionType::value_type>
+RtpPacketReceived CreateRtpPacketReceivedWithExtension(ExtensionValue value) {
+  RtpHeaderExtensionMap extensions;
+  extensions.Register<ExtensionType>(5);
+  RtpPacketReceived packet(&extensions);
+  RTC_CHECK(packet.SetExtension<ExtensionType>(value))
+      << "Unable to set extension.";
+  return packet;
+}
 
 TEST(RtpPacketInfoTest, Ssrc) {
   constexpr uint32_t kValue = 4038189233;
@@ -44,13 +64,12 @@ TEST(RtpPacketInfoTest, Ssrc) {
   rhs = RtpPacketInfo();
   EXPECT_NE(rhs.ssrc(), kValue);
 
-  rhs = RtpPacketInfo(/*ssrc=*/kValue, /*csrcs=*/{}, /*rtp_timestamp=*/{},
-                      /*receive_time=*/Timestamp::Zero());
+  rhs.set_ssrc(kValue);
   EXPECT_EQ(rhs.ssrc(), kValue);
 }
 
-TEST(RtpPacketInfoTest, Csrcs) {
-  const std::vector<uint32_t> value = {4038189233, 3016333617, 1207992985};
+TEST(RtpPacketInfoTest, SequenceNumber) {
+  constexpr uint16_t kValue = 34653;
 
   RtpPacketInfo lhs;
   RtpPacketInfo rhs;
@@ -58,8 +77,8 @@ TEST(RtpPacketInfoTest, Csrcs) {
   EXPECT_TRUE(lhs == rhs);
   EXPECT_FALSE(lhs != rhs);
 
-  rhs.set_csrcs(value);
-  EXPECT_EQ(rhs.csrcs(), value);
+  rhs.set_sequence_number(kValue);
+  EXPECT_EQ(rhs.sequence_number(), kValue);
 
   EXPECT_FALSE(lhs == rhs);
   EXPECT_TRUE(lhs != rhs);
@@ -70,11 +89,37 @@ TEST(RtpPacketInfoTest, Csrcs) {
   EXPECT_FALSE(lhs != rhs);
 
   rhs = RtpPacketInfo();
-  EXPECT_NE(rhs.csrcs(), value);
+  EXPECT_NE(rhs.sequence_number(), kValue);
 
-  rhs = RtpPacketInfo(/*ssrc=*/{}, /*csrcs=*/value, /*rtp_timestamp=*/{},
-                      /*receive_time=*/Timestamp::Zero());
-  EXPECT_EQ(rhs.csrcs(), value);
+  rhs.set_sequence_number(kValue);
+  EXPECT_EQ(rhs.sequence_number(), kValue);
+}
+
+TEST(RtpPacketInfoTest, Csrcs) {
+  const std::vector<uint32_t> kValue = {4038189233, 3016333617, 1207992985};
+
+  RtpPacketInfo lhs;
+  RtpPacketInfo rhs;
+
+  EXPECT_TRUE(lhs == rhs);
+  EXPECT_FALSE(lhs != rhs);
+
+  rhs.set_csrcs(kValue);
+  EXPECT_EQ(rhs.csrcs(), kValue);
+
+  EXPECT_FALSE(lhs == rhs);
+  EXPECT_TRUE(lhs != rhs);
+
+  lhs = rhs;
+
+  EXPECT_TRUE(lhs == rhs);
+  EXPECT_FALSE(lhs != rhs);
+
+  rhs = RtpPacketInfo();
+  EXPECT_NE(rhs.csrcs(), kValue);
+
+  rhs.set_csrcs(kValue);
+  EXPECT_EQ(rhs.csrcs(), kValue);
 }
 
 TEST(RtpPacketInfoTest, RtpTimestamp) {
@@ -100,8 +145,7 @@ TEST(RtpPacketInfoTest, RtpTimestamp) {
   rhs = RtpPacketInfo();
   EXPECT_NE(rhs.rtp_timestamp(), kValue);
 
-  rhs = RtpPacketInfo(/*ssrc=*/{}, /*csrcs=*/{}, /*rtp_timestamp=*/kValue,
-                      /*receive_time=*/Timestamp::Zero());
+  rhs.set_rtp_timestamp(kValue);
   EXPECT_EQ(rhs.rtp_timestamp(), kValue);
 }
 
@@ -128,8 +172,7 @@ TEST(RtpPacketInfoTest, ReceiveTimeMs) {
   rhs = RtpPacketInfo();
   EXPECT_NE(rhs.receive_time(), kValue);
 
-  rhs = RtpPacketInfo(/*ssrc=*/{}, /*csrcs=*/{}, /*rtp_timestamp=*/{},
-                      /*receive_time=*/kValue);
+  rhs.set_receive_time(kValue);
   EXPECT_EQ(rhs.receive_time(), kValue);
 }
 
@@ -156,8 +199,6 @@ TEST(RtpPacketInfoTest, AudioLevel) {
   rhs = RtpPacketInfo();
   EXPECT_NE(rhs.audio_level(), kValue);
 
-  rhs = RtpPacketInfo(/*ssrc=*/{}, /*csrcs=*/{}, /*rtp_timestamp=*/{},
-                      /*receive_time=*/Timestamp::Zero());
   rhs.set_audio_level(kValue);
   EXPECT_EQ(rhs.audio_level(), kValue);
 }
@@ -186,8 +227,6 @@ TEST(RtpPacketInfoTest, AbsoluteCaptureTime) {
   rhs = RtpPacketInfo();
   EXPECT_NE(rhs.absolute_capture_time(), kValue);
 
-  rhs = RtpPacketInfo(/*ssrc=*/{}, /*csrcs=*/{}, /*rtp_timestamp=*/{},
-                      /*receive_time=*/Timestamp::Zero());
   rhs.set_absolute_capture_time(kValue);
   EXPECT_EQ(rhs.absolute_capture_time(), kValue);
 }
@@ -215,10 +254,52 @@ TEST(RtpPacketInfoTest, LocalCaptureClockOffset) {
   rhs = RtpPacketInfo();
   EXPECT_EQ(rhs.local_capture_clock_offset(), std::nullopt);
 
-  rhs = RtpPacketInfo(/*ssrc=*/{}, /*csrcs=*/{}, /*rtp_timestamp=*/{},
-                      /*receive_time=*/Timestamp::Zero());
   rhs.set_local_capture_clock_offset(kValue);
   EXPECT_EQ(rhs.local_capture_clock_offset(), kValue);
 }
 
+TEST(RtpPacketInfoTest, RtpPacketReceivedNoExtensions) {
+  constexpr Timestamp kArrivalTime = Timestamp::Micros(8868963877546349045LL);
+  constexpr uint32_t kSsrc = 4038189233;
+  constexpr uint16_t kSequenceNumber = 1234;
+  constexpr uint32_t kRtpTimestamp = 5684353;
+  const std::vector<uint32_t> kCsrcs = {15, 60};
+  RtpPacketReceived packet;
+  packet.set_arrival_time(kArrivalTime);
+  packet.SetSsrc(kSsrc);
+  packet.SetCsrcs(kCsrcs);
+  packet.SetSequenceNumber(kSequenceNumber);
+  packet.SetTimestamp(kRtpTimestamp);
+
+  RtpPacketInfo rtp_packet_info(packet);
+
+  EXPECT_EQ(rtp_packet_info.ssrc(), kSsrc);
+  EXPECT_EQ(rtp_packet_info.sequence_number(), kSequenceNumber);
+  EXPECT_THAT(rtp_packet_info.csrcs(), ElementsAreArray(kCsrcs));
+  EXPECT_EQ(rtp_packet_info.rtp_timestamp(), kRtpTimestamp);
+  EXPECT_EQ(rtp_packet_info.receive_time(), kArrivalTime);
+}
+
+TEST(RtpPacketInfoTest, RtpPacketReceivedAudioLevel) {
+  constexpr uint8_t kValue = 14;
+  RtpPacketReceived packet =
+      CreateRtpPacketReceivedWithExtension<AudioLevelExtension>(
+          AudioLevel(/*voice_activity=*/true, /*level=*/kValue));
+
+  EXPECT_THAT(RtpPacketInfo(packet).audio_level(), Optional(kValue));
+}
+
+TEST(RtpPacketInfoTest, RtpPacketReceivedAbsoluteCaptureTime) {
+  const AbsoluteCaptureTime kAbsoluteCaptureTime = {
+      .absolute_capture_timestamp = 1493489345934859334,
+      .estimated_capture_clock_offset = 3453534534};
+  RtpPacketReceived packet =
+      CreateRtpPacketReceivedWithExtension<AbsoluteCaptureTimeExtension>(
+          kAbsoluteCaptureTime);
+
+  EXPECT_THAT(RtpPacketInfo(packet).absolute_capture_time(),
+              Optional(kAbsoluteCaptureTime));
+}
+
+}  // namespace
 }  // namespace webrtc

From 7fddd93e81a836e11dbfe0f2577dd8273a7e47e2 Mon Sep 17 00:00:00 2001
From: Per K <perkj@webrtc.org>
Date: Wed, 11 Mar 2026 17:06:07 +0000
Subject: [PATCH 276/472] Make scream  more resilient to delay spikes
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Use the minimum queue_delay from the last two feedback when calculating average queue_delay. This decrease the chance of reducing reference window due to increased delay that is only seen in one feedback.

Also replace usage of queue_delay_dev_norm with DelayBasedCongestionControl::ref_window_scale_factor_due_to_avg_min_delay and DelayBasedCongestionControl::ref_window_scale_factor_due_to_latency_difference
Both are used similarly to the old queue_delay_dev_norm, but are calculated based on the minimum average min queue delay and the average
latency difference per feeedback.

This replace previous:
if (l4s_alpha < 0.0001)
   increment_t *= max(0.1, 1.0 - qdelay_avg / (qdelay_target / 4))
end

Bug: webrtc:447037083
Change-Id: Ic713258b2d0a50b486dc93a89f7c947156d86907
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453300
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47129}
---
 .../scream/delay_based_congestion_control.cc  |  71 ++++++++---
 .../scream/delay_based_congestion_control.h   |  40 +++----
 ...delay_based_congestion_control_unittest.cc | 112 ++++++++++++++++++
 .../scream_network_controller_unittest.cc     |   2 +-
 .../congestion_controller/scream/scream_v2.cc |  25 ++--
 .../scream/scream_v2_parameters.cc            |  11 +-
 .../scream/scream_v2_parameters.h             |  12 +-
 .../rtc_event_log_visualizer/analyze_bwe.cc   |  55 +++++++--
 .../rtc_event_log_visualizer/analyze_bwe.h    |   5 +-
 .../rtc_event_log_visualizer/analyzer.cc      |   7 ++
 rtc_tools/rtc_event_log_visualizer/analyzer.h |   1 +
 .../log_scream_simulation.cc                  |  21 ++--
 .../log_scream_simulation.h                   |   9 +-
 rtc_tools/rtc_event_log_visualizer/main.cc    |   2 +-
 .../bwe_integration_tests/scream_test.cc      |  60 ++++++----
 15 files changed, 324 insertions(+), 109 deletions(-)

diff --git a/modules/congestion_controller/scream/delay_based_congestion_control.cc b/modules/congestion_controller/scream/delay_based_congestion_control.cc
index 4cc5e0d31b..94be767cb1 100644
--- a/modules/congestion_controller/scream/delay_based_congestion_control.cc
+++ b/modules/congestion_controller/scream/delay_based_congestion_control.cc
@@ -39,15 +39,16 @@ void DelayBasedCongestionControl::OnTransportPacketsFeedback(
   if (received_packets.empty()) {
     return;
   }
-  TimeDelta one_way_delay_sum;
+
   TimeDelta min_one_way_delay = TimeDelta::PlusInfinity();
+  TimeDelta max_one_way_delay = TimeDelta::Zero();
 
   for (const PacketResult& packet : received_packets) {
     TimeDelta one_way_delay =
         packet.receive_time - packet.sent_packet.send_time;
     next_base_delay_ = std::min(next_base_delay_, one_way_delay);
-    one_way_delay_sum += one_way_delay;
     min_one_way_delay = std::min(min_one_way_delay, one_way_delay);
+    max_one_way_delay = std::max(max_one_way_delay, one_way_delay);
   }
   // `arrival_time_offset` is null if TWCC is used. We assume feedback was sent
   // when the last sent packet was received.
@@ -66,8 +67,12 @@ void DelayBasedCongestionControl::OnTransportPacketsFeedback(
   } else {
     min_queue_delay_above_threshold_start_ = Timestamp::MinusInfinity();
   }
-  UpdateQueueDelayAverage(one_way_delay_sum /
-                          static_cast<int>(received_packets.size()));
+  UpdateQueueDelayAverage(std::min(min_queue_delay, last_queue_delay_sample_));
+  last_queue_delay_sample_ = min_queue_delay;
+  UpdateQueueDelayMinAverage(min_queue_delay);
+  UpdateLatencyDifferenceAverage(min_one_way_delay.IsFinite()
+                                     ? max_one_way_delay - min_one_way_delay
+                                     : TimeDelta::Zero());
 
   if (msg.feedback_time - last_base_delay_update_ >=
       params_.base_delay_history_update_interval.Get()) {
@@ -78,26 +83,37 @@ void DelayBasedCongestionControl::OnTransportPacketsFeedback(
 }
 
 void DelayBasedCongestionControl::UpdateQueueDelayAverage(
-    TimeDelta one_way_delay) {
-  TimeDelta current_qdelay = one_way_delay - min_base_delay();
-
+    TimeDelta min_qdelay_in_feedback) {
   // `queue_delay_avg_` is updated with a slow attack,fast decay EWMA
   // filter.
-  if (current_qdelay < queue_delay_avg_) {
-    queue_delay_avg_ = current_qdelay;
+  if (min_qdelay_in_feedback < queue_delay_avg_) {
+    queue_delay_avg_ = min_qdelay_in_feedback;
   } else {
     queue_delay_avg_ =
-        params_.queue_delay_avg_g.Get() * current_qdelay +
+        params_.queue_delay_avg_g.Get() * min_qdelay_in_feedback +
         (1.0 - params_.queue_delay_avg_g.Get()) * queue_delay_avg_;
   }
+}
 
-  queue_delay_dev_norm_ =
-      (1.0 - params_.queue_delay_dev_avg_g.Get()) * queue_delay_dev_norm_ +
-      params_.queue_delay_dev_avg_g.Get() *
-          std::clamp((current_qdelay -
-                      params_.queue_delay_dev_normalization.Get() / 4) /
-                         params_.queue_delay_dev_normalization.Get(),
-                     0.0, 0.2);
+void DelayBasedCongestionControl::UpdateQueueDelayMinAverage(
+    TimeDelta packet_qdelay) {
+  RTC_DCHECK(packet_qdelay >= TimeDelta::Zero());
+  queue_delay_min_avg_ =
+      (1.0 - params_.delay_min_and_latency_diff_avg_g.Get()) *
+          queue_delay_min_avg_ +
+      params_.delay_min_and_latency_diff_avg_g.Get() *
+          std::min(packet_qdelay, 2 * params_.queue_delay_min_threshold.Get());
+}
+
+void DelayBasedCongestionControl::UpdateLatencyDifferenceAverage(
+    TimeDelta packet_latency_diff) {
+  RTC_DCHECK(packet_latency_diff >= TimeDelta::Zero());
+  latency_difference_avg_ =
+      (1.0 - params_.delay_min_and_latency_diff_avg_g.Get()) *
+          latency_difference_avg_ +
+      params_.delay_min_and_latency_diff_avg_g.Get() *
+          std::min(packet_latency_diff,
+                   2 * params_.latency_diff_threshold.Get());
 }
 
 void DelayBasedCongestionControl::UpdateSmoothedRtt(TimeDelta rtt_sample) {
@@ -123,7 +139,26 @@ void DelayBasedCongestionControl::ResetQueueDelay() {
   min_queue_delay_above_threshold_start_ = Timestamp::MinusInfinity();
   last_update_qdelay_avg_time_ = Timestamp::MinusInfinity();
   queue_delay_avg_ = TimeDelta::PlusInfinity();
-  queue_delay_dev_norm_ = 0.0;
+  queue_delay_min_avg_ = TimeDelta::Zero();
+  latency_difference_avg_ = TimeDelta::Zero();
+}
+
+double
+DelayBasedCongestionControl::ref_window_scale_factor_due_to_avg_min_delay(
+    bool allow_zero) const {
+  TimeDelta norm = params_.queue_delay_min_threshold.Get();
+  // Reaches 0.1 at norm, and 1.0 at norm / 4
+  return std::clamp(0.1 + 1.2 * (norm - queue_delay_min_avg_) / norm,
+                    allow_zero ? 0.0 : 0.1, 1.0);
+}
+
+double
+DelayBasedCongestionControl::ref_window_scale_factor_due_to_latency_difference()
+    const {
+  TimeDelta norm = params_.latency_diff_threshold.Get();
+  // Reaches 0.1 at norm, and 1.0 at norm / 4
+  return std::clamp(0.1 + 1.2 * (norm - latency_difference_avg_) / norm, 0.1,
+                    1.0);
 }
 
 DataSize DelayBasedCongestionControl::UpdateReferenceWindow(
diff --git a/modules/congestion_controller/scream/delay_based_congestion_control.h b/modules/congestion_controller/scream/delay_based_congestion_control.h
index d8041b4fb6..91d14831d5 100644
--- a/modules/congestion_controller/scream/delay_based_congestion_control.h
+++ b/modules/congestion_controller/scream/delay_based_congestion_control.h
@@ -63,29 +63,21 @@ class DelayBasedCongestionControl {
 
   TimeDelta queue_delay() const { return queue_delay_avg_; }
 
-  double queue_delay_dev_norm() const { return queue_delay_dev_norm_; }
+  TimeDelta queue_delay_min_avg() const { return queue_delay_min_avg_; }
+  TimeDelta latency_difference_avg() const { return latency_difference_avg_; }
 
-  // Scales the reference window increase/decrease due to increased delay if L4S
-  // does not appear to be used. See 4.2.2.1.
-  double ref_window_scale_factor_due_to_increased_delay() const {
-    return std::clamp(
-        1 - queue_delay_avg_ / (params_.queue_delay_target.Get() / 4), 0.1,
-        1.0);
-  }
+  // Scale factor used for scaling reference window increase/decrease due to
+  // minimum average queue delay per feedback. The scale factor is 0.1
+  // if the delay is higher than the threshold, and increases linearly to 1.0 if
+  // the delay is lower than the threshold / 4.
+  double ref_window_scale_factor_due_to_avg_min_delay(
+      bool allow_zero = false) const;
 
-  // Scales the reference window increase/decrease due to delay variation.
-  // See 4.2.2.1 and 4.2.2.2. The scale factor is 0.0 if the delay variation
-  // is higher than the threshold, and decreases linearly to 1.0 if the delay
-  // variation is zero.
-  double ref_window_scale_factor_due_to_delay_variation(
-      double ref_window_mss_ratio) const {
-    // 4.2.1.4.
-    double queue_delay_dev_norm_threshold =
-        std::max(0.05, 0.1 * (1.0 - ref_window_mss_ratio / 0.1));
-    return std::clamp((queue_delay_dev_norm_threshold - queue_delay_dev_norm_) /
-                          queue_delay_dev_norm_threshold,
-                      0.0, 1.0);
-  }
+  // Scale factor used for scaling reference window increase if
+  // latency differences per feedback is larger than the threshold. If the send
+  // rate is close to link capacity, the difference between min and max latency
+  // per feedback increases and the scale factor decreases.
+  double ref_window_scale_factor_due_to_latency_difference() const;
 
   TimeDelta rtt() const { return last_smoothed_rtt_; }
 
@@ -97,6 +89,8 @@ class DelayBasedCongestionControl {
   }
   void UpdateSmoothedRtt(TimeDelta rtt_sample);
   void UpdateQueueDelayAverage(TimeDelta one_way_delay);
+  void UpdateQueueDelayMinAverage(TimeDelta packet_qdelay);
+  void UpdateLatencyDifferenceAverage(TimeDelta packet_latency_diff);
 
   const ScreamV2Parameters params_;
 
@@ -111,8 +105,10 @@ class DelayBasedCongestionControl {
   Timestamp min_queue_delay_above_threshold_start_ = Timestamp::MinusInfinity();
   TimeDelta last_smoothed_rtt_ = TimeDelta::Zero();
   Timestamp last_update_qdelay_avg_time_ = Timestamp::MinusInfinity();
+  TimeDelta last_queue_delay_sample_ = TimeDelta::PlusInfinity();
   TimeDelta queue_delay_avg_ = TimeDelta::PlusInfinity();
-  double queue_delay_dev_norm_ = 0.0;
+  TimeDelta queue_delay_min_avg_ = TimeDelta::Zero();
+  TimeDelta latency_difference_avg_ = TimeDelta::Zero();
 };
 
 }  // namespace webrtc
diff --git a/modules/congestion_controller/scream/delay_based_congestion_control_unittest.cc b/modules/congestion_controller/scream/delay_based_congestion_control_unittest.cc
index a2940ae417..eea60d3c7d 100644
--- a/modules/congestion_controller/scream/delay_based_congestion_control_unittest.cc
+++ b/modules/congestion_controller/scream/delay_based_congestion_control_unittest.cc
@@ -229,6 +229,118 @@ TEST(DelayBasedCongestionControlTest,
   EXPECT_FALSE(delay_controller.IsQueueDrainedInTime(clock.CurrentTime()));
 }
 
+TEST(DelayBasedCongestionControlTest,
+     RefWindowScaleFactorDueToMinAverageQueueDelay) {
+  SimulatedClock clock(Timestamp::Seconds(1234));
+  Environment env = CreateTestEnvironment({.time = &clock});
+  ScreamV2Parameters params(env.field_trials());
+  DelayBasedCongestionControl delay_controller(params);
+
+  const TimeDelta kQueueDelyMinThreshold =
+      params.queue_delay_min_threshold.Get();
+
+  // Helper to establish queue delay.
+  // Assumes base delay is 100ms.
+  auto feed_packets = [&](TimeDelta qdelay, int count) {
+    for (int i = 0; i < count; ++i) {
+      clock.AdvanceTime(TimeDelta::Millis(10));
+      TransportPacketsFeedback msg;
+      msg.feedback_time = clock.CurrentTime();
+      PacketResult packet;
+      packet.receive_time = clock.CurrentTime();
+      packet.sent_packet.send_time =
+          clock.CurrentTime() - TimeDelta::Millis(100) - qdelay;
+      packet.sent_packet.sequence_number = i;
+      msg.packet_feedbacks.push_back(packet);
+      delay_controller.OnTransportPacketsFeedback(msg);
+    }
+  };
+
+  // Establish base delay (100ms) with qdelay=0.
+  feed_packets(TimeDelta::Zero(), 100);
+  EXPECT_DOUBLE_EQ(
+      delay_controller.ref_window_scale_factor_due_to_avg_min_delay(), 1.0);
+
+  // Drive queue delay to kQueueDelyMinThreshold.
+  feed_packets(kQueueDelyMinThreshold + TimeDelta::Millis(10), 250);
+  EXPECT_NEAR(delay_controller.ref_window_scale_factor_due_to_avg_min_delay(),
+              0.1, 0.01);
+
+  // Drive queue delay to `0.5 * kQueueDelyMinThreshold`.
+  // Expected: 0.1 + 1.2 * 0.5 = 0.7
+  feed_packets(kQueueDelyMinThreshold / 2, 250);
+  EXPECT_NEAR(delay_controller.ref_window_scale_factor_due_to_avg_min_delay(),
+              0.7, 0.01);
+
+  // Drive queue delay to `0.25 * kQueueDelyMinThreshold`.
+  // Expected: 0.1 + 1.2 * 0.75 = 1.0.
+  feed_packets(kQueueDelyMinThreshold / 4, 250);
+  EXPECT_NEAR(delay_controller.ref_window_scale_factor_due_to_avg_min_delay(),
+              1.0, 0.01);
+}
+
+TEST(DelayBasedCongestionControlTest,
+     RefWindowScaleFactorDueToLatencyDifference) {
+  SimulatedClock clock(Timestamp::Seconds(1234));
+  Environment env = CreateTestEnvironment({.time = &clock});
+  ScreamV2Parameters params(env.field_trials());
+  DelayBasedCongestionControl delay_controller(params);
+
+  const TimeDelta kLatencyThreshold = params.latency_diff_threshold.Get();
+
+  // Helper to establish latency differences within feedbacks.
+  // Assumes a base one-way delay of 100ms.
+  auto feed_packets = [&](TimeDelta latency_diff, int count) {
+    for (int i = 0; i < count; ++i) {
+      clock.AdvanceTime(TimeDelta::Millis(10));
+      TransportPacketsFeedback msg;
+      msg.feedback_time = clock.CurrentTime();
+
+      PacketResult packet1;
+      packet1.receive_time = clock.CurrentTime();
+      packet1.sent_packet.send_time =
+          clock.CurrentTime() - TimeDelta::Millis(100);
+      packet1.sent_packet.sequence_number = i * 2;
+
+      PacketResult packet2;
+      packet2.receive_time = clock.CurrentTime();
+      packet2.sent_packet.send_time =
+          clock.CurrentTime() - TimeDelta::Millis(100) - latency_diff;
+      packet2.sent_packet.sequence_number = i * 2 + 1;
+
+      msg.packet_feedbacks.push_back(packet1);
+      msg.packet_feedbacks.push_back(packet2);
+      delay_controller.OnTransportPacketsFeedback(msg);
+    }
+  };
+
+  // Establish 0 latency difference.
+  feed_packets(TimeDelta::Zero(), 100);
+  EXPECT_DOUBLE_EQ(
+      delay_controller.ref_window_scale_factor_due_to_latency_difference(),
+      1.0);
+
+  // Drive to `kLatencyThreshold`.
+  feed_packets(kLatencyThreshold + TimeDelta::Millis(10), 250);
+  EXPECT_NEAR(
+      delay_controller.ref_window_scale_factor_due_to_latency_difference(), 0.1,
+      0.01);
+
+  // Drive to `0.5 * kLatencyThreshold`.
+  // Expected: 0.1 + 1.2 * 0.5 = 0.7
+  feed_packets(kLatencyThreshold / 2, 250);
+  EXPECT_NEAR(
+      delay_controller.ref_window_scale_factor_due_to_latency_difference(), 0.7,
+      0.01);
+
+  // Drive to `0.25 * kLatencyThreshold`.
+  // Expected: 0.1 + 1.2 * 0.75 = 1.0.
+  feed_packets(kLatencyThreshold / 4, 250);
+  EXPECT_NEAR(
+      delay_controller.ref_window_scale_factor_due_to_latency_difference(), 1.0,
+      0.01);
+}
+
 // TODO: bugs.webrtc.org/447037083 - add tests for clock drift in feedback NTP
 // time.
 
diff --git a/modules/congestion_controller/scream/scream_network_controller_unittest.cc b/modules/congestion_controller/scream/scream_network_controller_unittest.cc
index b3a0d90419..745f70d79d 100644
--- a/modules/congestion_controller/scream/scream_network_controller_unittest.cc
+++ b/modules/congestion_controller/scream/scream_network_controller_unittest.cc
@@ -420,7 +420,7 @@ TEST(ScreamControllerTest, PaddingStopIfNetworkCongested) {
   PaddingTestResult result = ProcessUntilPaddingStartAndStop(
       clock, scream_controller, feedback_generator);
 
-  EXPECT_LT(result.target_rate, DataRate::KilobitsPerSec(700));
+  EXPECT_LT(result.target_rate, DataRate::KilobitsPerSec(720));
   // Padding should stop when congestion is detected.
   EXPECT_LT(result.padding_stop - result.padding_start, TimeDelta::Seconds(1));
 }
diff --git a/modules/congestion_controller/scream/scream_v2.cc b/modules/congestion_controller/scream/scream_v2.cc
index 3836c939ca..fc5d533d19 100644
--- a/modules/congestion_controller/scream/scream_v2.cc
+++ b/modules/congestion_controller/scream/scream_v2.cc
@@ -180,8 +180,7 @@ void ScreamV2::UpdateRefWindow(const TransportPacketsFeedback& msg) {
         // of competing TCP Prague flows.
         backoff *=
             std::max(0.1, delay_based_congestion_control_
-                              .ref_window_scale_factor_due_to_delay_variation(
-                                  ref_window_mss_ratio()));
+                              .ref_window_scale_factor_due_to_avg_min_delay());
       }
 
       if (msg.feedback_time - last_reaction_to_congestion_time_ >
@@ -233,23 +232,17 @@ void ScreamV2::UpdateRefWindow(const TransportPacketsFeedback& msg) {
         increase_scale_factor *
         std::max(0.25, ref_window_scale_factor_close_to_ref_window_i());
 
-    // Limit increase if L4S not enabled and queue delay is increased.
-    if (l4s_alpha_ < 0.0001) {
-      increase_scale_factor =
-          increase_scale_factor *
-          delay_based_congestion_control_
-              .ref_window_scale_factor_due_to_increased_delay();
-    }
-
     // Put a additional restriction on reference window growth if rtt varies a
     // lot.
     // Better to enforce a slow increase in reference window and get
     // a more stable bitrate.
+    increase_scale_factor = increase_scale_factor *
+                            delay_based_congestion_control_
+                                .ref_window_scale_factor_due_to_avg_min_delay();
     increase_scale_factor =
         increase_scale_factor *
-        std::max(0.1, delay_based_congestion_control_
-                          .ref_window_scale_factor_due_to_delay_variation(
-                              ref_window_mss_ratio()));
+        delay_based_congestion_control_
+            .ref_window_scale_factor_due_to_latency_difference();
 
     // Use lower multiplicative scale factor if congestion was detected
     // recently.
@@ -299,8 +292,6 @@ void ScreamV2::UpdateRefWindow(const TransportPacketsFeedback& msg) {
       << " is_virtual_ce=" << is_virtual_ce << " is_loss=" << is_loss
       << " smoothed_rtt=" << delay_based_congestion_control_.rtt().ms()
       << ", queue_delay=" << delay_based_congestion_control_.queue_delay().ms()
-      << ", queue_delay_dev_norm="
-      << delay_based_congestion_control_.queue_delay_dev_norm()
       << " feedback_hold" << feedback_hold_time_.ms()
       << ", target_rate =" << target_rate_.kbps();
 }
@@ -312,8 +303,8 @@ DataSize ScreamV2::max_data_in_flight() const {
       (params_.ref_window_overhead_max.Get() -
        params_.ref_window_overhead_min.Get()) *
           delay_based_congestion_control_
-              .ref_window_scale_factor_due_to_delay_variation(
-                  ref_window_mss_ratio());
+              .ref_window_scale_factor_due_to_avg_min_delay(
+                  /*allow_zero=*/true);
 
   return ref_window_ * ref_window_overhead;
 }
diff --git a/modules/congestion_controller/scream/scream_v2_parameters.cc b/modules/congestion_controller/scream/scream_v2_parameters.cc
index 16c91b2d4d..63c01c450a 100644
--- a/modules/congestion_controller/scream/scream_v2_parameters.cc
+++ b/modules/congestion_controller/scream/scream_v2_parameters.cc
@@ -41,8 +41,10 @@ ScreamV2Parameters::ScreamV2Parameters(const FieldTrialsView& trials)
       ref_window_overhead_min("RefWinMin", 1.5),
       ref_window_overhead_max("RefWinMax", 3.0),
       queue_delay_avg_g("QDelayAvgG", 1.0 / 4.0),
-      queue_delay_dev_avg_g("QDelayDevAvgG", 1.0 / 64.0),
-      queue_delay_dev_normalization("QDelayDevNorm", TimeDelta::Millis(25)),
+      delay_min_and_latency_diff_avg_g("DelayMinAndLatencyDiffAvgG",
+                                       1.0 / 32.0),
+      queue_delay_min_threshold("QDelayMinThreshold", TimeDelta::Millis(10)),
+      latency_diff_threshold("LatencyDiffThreshold", TimeDelta::Millis(10)),
       base_delay_window_length("BaseDelayWindowLength", 10),
       base_delay_history_update_interval("BaseDelayHistoryUpdateInterval",
                                          TimeDelta::Minutes(1)),
@@ -76,8 +78,9 @@ ScreamV2Parameters::ScreamV2Parameters(const FieldTrialsView& trials)
                    &ref_window_overhead_min,
                    &ref_window_overhead_max,
                    &queue_delay_avg_g,
-                   &queue_delay_dev_avg_g,
-                   &queue_delay_dev_normalization,
+                   &delay_min_and_latency_diff_avg_g,
+                   &queue_delay_min_threshold,
+                   &latency_diff_threshold,
                    &base_delay_window_length,
                    &base_delay_history_update_interval,
                    &queue_delay_target,
diff --git a/modules/congestion_controller/scream/scream_v2_parameters.h b/modules/congestion_controller/scream/scream_v2_parameters.h
index 53c8296039..9dafdba008 100644
--- a/modules/congestion_controller/scream/scream_v2_parameters.h
+++ b/modules/congestion_controller/scream/scream_v2_parameters.h
@@ -74,10 +74,16 @@ struct ScreamV2Parameters {
   // Exponentially Weighted Moving Average (EWMA) factor for updating queue
   // delay.
   FieldTrialParameter<double> queue_delay_avg_g;
-  FieldTrialParameter<double> queue_delay_dev_avg_g;
+  // Exponentially Weighted Moving Average (EWMA) factor for updating average
+  // min queue delay and average latency difference.
+  FieldTrialParameter<double> delay_min_and_latency_diff_avg_g;
 
-  // Normalization factor for queue delay variation.
-  FieldTrialParameter<TimeDelta> queue_delay_dev_normalization;
+  // Thresholds for average min queue delay. Used for limiting ref_window
+  // increase.
+  FieldTrialParameter<TimeDelta> queue_delay_min_threshold;
+  // Thresholds for average latency difference. Used for limiting ref_window
+  // increase.
+  FieldTrialParameter<TimeDelta> latency_diff_threshold;
 
   // Determines the length of the base delay history when estimating one way
   // delay (owd)
diff --git a/rtc_tools/rtc_event_log_visualizer/analyze_bwe.cc b/rtc_tools/rtc_event_log_visualizer/analyze_bwe.cc
index d0cf941983..30682e4fdd 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyze_bwe.cc
+++ b/rtc_tools/rtc_event_log_visualizer/analyze_bwe.cc
@@ -644,6 +644,41 @@ void CreateGoogCcSimulationGraph(const ParsedRtcEventLog& parsed_log,
   plot->SetTitle("Simulated BWE behavior");
 }
 
+void CreateScreamSimulationDelayGraph(const ParsedRtcEventLog& parsed_log,
+                                      const AnalyzerConfig& config,
+                                      Plot* plot) {
+  TimeSeries smoothed_rtt_series("Smoothed RTT", LineStyle::kStep);
+  TimeSeries queue_delay_series("Queue delay", LineStyle::kStep);
+  TimeSeries queue_delay_min_avg_series("Queue delay min avg",
+                                        LineStyle::kStep);
+  TimeSeries latency_difference_avg_series("Latency difference avg",
+                                           LineStyle::kStep);
+
+  LogScreamSimulation simulation({.rate_window = config.window_duration_},
+                                 config.env_);
+  simulation.ProcessEventsInLog(parsed_log);
+
+  for (const LogScreamSimulation::State& state : simulation.updates()) {
+    smoothed_rtt_series.points.emplace_back(config.GetCallTimeSec(state.time),
+                                            state.smoothed_rtt.ms());
+    queue_delay_series.points.emplace_back(config.GetCallTimeSec(state.time),
+                                           state.queue_delay.ms());
+    queue_delay_min_avg_series.points.emplace_back(
+        config.GetCallTimeSec(state.time), state.queue_delay_min_avg.ms());
+    latency_difference_avg_series.points.emplace_back(
+        config.GetCallTimeSec(state.time), state.latency_difference_avg.ms());
+  }
+  plot->AppendTimeSeries(std::move(smoothed_rtt_series));
+  plot->AppendTimeSeries(std::move(queue_delay_series));
+  plot->AppendTimeSeries(std::move(queue_delay_min_avg_series));
+  plot->AppendTimeSeries(std::move(latency_difference_avg_series));
+
+  plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)",
+                 kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 50, "Delay (ms)", kBottomMargin, kTopMargin);
+  plot->SetTitle("Simulated Scream delays");
+}
+
 void CreateScreamSimulationBitrateGraph(const ParsedRtcEventLog& parsed_log,
                                         const AnalyzerConfig& config,
                                         Plot* plot) {
@@ -757,10 +792,10 @@ void CreateScreamSimulationRatiosGraph(const ParsedRtcEventLog& parsed_log,
   TimeSeries queue_delay_dev_norm_series("QueueDelayDevNorm", LineStyle::kStep);
   TimeSeries l4s_alpha_series("L4sAlpha", LineStyle::kStep);
   TimeSeries l4s_alpha_v_series("L4sAlphaV", LineStyle::kStep);
-  TimeSeries ref_window_scale_factor_due_to_increased_delay(
-      "RefWindowScaleFactorDueToIncreasedDelay", LineStyle::kStep);
-  TimeSeries ref_window_scale_factor_due_to_delay_variation(
-      "RefWindowScaleFactorDueToDelayVariation", LineStyle::kStep);
+  TimeSeries ref_window_scale_factor_due_to_min_delay_variation(
+      "RefWindowScaleFactorDueToAvgMinQueueDelay", LineStyle::kStep);
+  TimeSeries ref_window_scale_factor_due_to_latency_difference(
+      "RefWindowScaleFactorDueToAvgLatencyDifference", LineStyle::kStep);
   TimeSeries ref_window_scale_factor_close_to_ref_window_i(
       "RefWindowScaleFactorCloseToRefWindowI", LineStyle::kStep);
   TimeSeries ref_window_combined_increase_scale_factor(
@@ -777,12 +812,12 @@ void CreateScreamSimulationRatiosGraph(const ParsedRtcEventLog& parsed_log,
                                          state.l4s_alpha);
     l4s_alpha_v_series.points.emplace_back(config.GetCallTimeSec(state.time),
                                            state.l4s_alpha_v);
-    ref_window_scale_factor_due_to_increased_delay.points.emplace_back(
+    ref_window_scale_factor_due_to_min_delay_variation.points.emplace_back(
         config.GetCallTimeSec(state.time),
-        state.ref_window_scale_factor_due_to_increased_delay);
-    ref_window_scale_factor_due_to_delay_variation.points.emplace_back(
+        state.ref_window_scale_factor_due_to_avg_min_delay);
+    ref_window_scale_factor_due_to_latency_difference.points.emplace_back(
         config.GetCallTimeSec(state.time),
-        state.ref_window_scale_factor_due_to_delay_variation);
+        state.ref_window_scale_factor_due_to_latency_difference);
     ref_window_scale_factor_close_to_ref_window_i.points.emplace_back(
         config.GetCallTimeSec(state.time),
         state.ref_window_scale_factor_close_to_ref_window_i);
@@ -794,9 +829,9 @@ void CreateScreamSimulationRatiosGraph(const ParsedRtcEventLog& parsed_log,
   plot->AppendTimeSeries(std::move(l4s_alpha_series));
   plot->AppendTimeSeries(std::move(l4s_alpha_v_series));
   plot->AppendTimeSeries(
-      std::move(ref_window_scale_factor_due_to_increased_delay));
+      std::move(ref_window_scale_factor_due_to_min_delay_variation));
   plot->AppendTimeSeries(
-      std::move(ref_window_scale_factor_due_to_delay_variation));
+      std::move(ref_window_scale_factor_due_to_latency_difference));
   plot->AppendTimeSeries(
       std::move(ref_window_scale_factor_close_to_ref_window_i));
   plot->AppendTimeSeries(std::move(ref_window_combined_increase_scale_factor));
diff --git a/rtc_tools/rtc_event_log_visualizer/analyze_bwe.h b/rtc_tools/rtc_event_log_visualizer/analyze_bwe.h
index fd4b041c78..5de8598abd 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyze_bwe.h
+++ b/rtc_tools/rtc_event_log_visualizer/analyze_bwe.h
@@ -40,10 +40,13 @@ void CreateGoogCcSimulationGraph(const ParsedRtcEventLog& parsed_log,
                                  const AnalyzerConfig& config,
                                  Plot* plot);
 
+void CreateScreamSimulationDelayGraph(const ParsedRtcEventLog& parsed_log,
+                                      const AnalyzerConfig& config,
+                                      Plot* plot);
+
 void CreateScreamSimulationBitrateGraph(const ParsedRtcEventLog& parsed_log,
                                         const AnalyzerConfig& config,
                                         Plot* plot);
-
 void CreateScreamSimulationRefWindowGraph(const ParsedRtcEventLog& parsed_log,
                                           const AnalyzerConfig& config,
                                           Plot* plot);
diff --git a/rtc_tools/rtc_event_log_visualizer/analyzer.cc b/rtc_tools/rtc_event_log_visualizer/analyzer.cc
index 72ef110236..a44d6832c7 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyzer.cc
+++ b/rtc_tools/rtc_event_log_visualizer/analyzer.cc
@@ -172,6 +172,9 @@ void EventLogAnalyzer::InitializeMapOfNamedGraphs(bool show_detector_state,
   plots_.RegisterPlot("simulated_goog_cc", [this](Plot* plot) {
     this->CreateGoogCcSimulationGraph(plot);
   });
+  plots_.RegisterPlot("simulated_scream_delay", [this](Plot* plot) {
+    this->CreateScreamSimulationDelayGraph(plot);
+  });
   plots_.RegisterPlot("simulated_scream_bitrates", [this](Plot* plot) {
     this->CreateScreamSimulationBitrateGraph(plot);
   });
@@ -403,6 +406,10 @@ void EventLogAnalyzer::CreateGoogCcSimulationGraph(Plot* plot) const {
   webrtc::CreateGoogCcSimulationGraph(parsed_log_, config_, plot);
 }
 
+void EventLogAnalyzer::CreateScreamSimulationDelayGraph(Plot* plot) const {
+  webrtc::CreateScreamSimulationDelayGraph(parsed_log_, config_, plot);
+}
+
 void EventLogAnalyzer::CreateScreamSimulationBitrateGraph(Plot* plot) const {
   webrtc::CreateScreamSimulationBitrateGraph(parsed_log_, config_, plot);
 }
diff --git a/rtc_tools/rtc_event_log_visualizer/analyzer.h b/rtc_tools/rtc_event_log_visualizer/analyzer.h
index 511d002036..60d2244e0e 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyzer.h
+++ b/rtc_tools/rtc_event_log_visualizer/analyzer.h
@@ -131,6 +131,7 @@ class EventLogAnalyzer {
   void CreateScreamRefWindowGraph(Plot* plot) const;
   void CreateScreamDelayEstimateGraph(Plot* plot) const;
   void CreateGoogCcSimulationGraph(Plot* plot) const;
+  void CreateScreamSimulationDelayGraph(Plot* plot) const;
   void CreateScreamSimulationBitrateGraph(Plot* plot) const;
   void CreateScreamSimulationRefWindowGraph(Plot* plot) const;
   void CreateScreamSimulationRatiosGraph(Plot* plot) const;
diff --git a/rtc_tools/rtc_event_log_visualizer/log_scream_simulation.cc b/rtc_tools/rtc_event_log_visualizer/log_scream_simulation.cc
index 5e97896aa5..e52e131395 100644
--- a/rtc_tools/rtc_event_log_visualizer/log_scream_simulation.cc
+++ b/rtc_tools/rtc_event_log_visualizer/log_scream_simulation.cc
@@ -172,17 +172,18 @@ void LogScreamSimulation::LogState(const TransportPacketsFeedback& msg) {
       .max_data_in_flight = scream_->max_data_in_flight(),
       .data_in_flight = data_in_flight_,
       .send_window_usage = send_window_usage_,
-      .queue_delay_dev_norm =
-          scream_->delay_based_congestion_control().queue_delay_dev_norm(),
-      .ref_window_scale_factor_due_to_increased_delay =
-          scream_->delay_based_congestion_control().IsQueueDelayDetected()
-              ? 0.0
-              : scream_->delay_based_congestion_control()
-                    .ref_window_scale_factor_due_to_increased_delay(),
-      .ref_window_scale_factor_due_to_delay_variation =
+      .smoothed_rtt = scream_->delay_based_congestion_control().rtt(),
+      .queue_delay = scream_->delay_based_congestion_control().queue_delay(),
+      .queue_delay_min_avg =
+          scream_->delay_based_congestion_control().queue_delay_min_avg(),
+      .latency_difference_avg =
+          scream_->delay_based_congestion_control().latency_difference_avg(),
+      .ref_window_scale_factor_due_to_avg_min_delay =
           scream_->delay_based_congestion_control()
-              .ref_window_scale_factor_due_to_delay_variation(
-                  scream_->ref_window_mss_ratio()),
+              .ref_window_scale_factor_due_to_avg_min_delay(),
+      .ref_window_scale_factor_due_to_latency_difference =
+          scream_->delay_based_congestion_control()
+              .ref_window_scale_factor_due_to_latency_difference(),
       .ref_window_scale_factor_close_to_ref_window_i =
           scream_->ref_window_scale_factor_close_to_ref_window_i(),
       .ref_window_combined_increase_scale_factor =
diff --git a/rtc_tools/rtc_event_log_visualizer/log_scream_simulation.h b/rtc_tools/rtc_event_log_visualizer/log_scream_simulation.h
index 9641f77eb7..252880a83f 100644
--- a/rtc_tools/rtc_event_log_visualizer/log_scream_simulation.h
+++ b/rtc_tools/rtc_event_log_visualizer/log_scream_simulation.h
@@ -55,9 +55,14 @@ class LogScreamSimulation {
     // last packet was sent before the state was captured.
     SendWindowUsage send_window_usage = kBelowRefWindow;
 
+    TimeDelta smoothed_rtt = TimeDelta::Zero();
+    TimeDelta queue_delay = TimeDelta::Zero();
+    TimeDelta queue_delay_min_avg = TimeDelta::Zero();
+    TimeDelta latency_difference_avg = TimeDelta::Zero();
+
     double queue_delay_dev_norm = 0;
-    double ref_window_scale_factor_due_to_increased_delay = 0.0;
-    double ref_window_scale_factor_due_to_delay_variation = 0.0;
+    double ref_window_scale_factor_due_to_avg_min_delay = 0.0;
+    double ref_window_scale_factor_due_to_latency_difference = 0.0;
     double ref_window_scale_factor_close_to_ref_window_i = 0.0;
     double ref_window_combined_increase_scale_factor = 0.0;
     double l4s_alpha = 0.0;
diff --git a/rtc_tools/rtc_event_log_visualizer/main.cc b/rtc_tools/rtc_event_log_visualizer/main.cc
index 67ac267d80..f299d9a318 100644
--- a/rtc_tools/rtc_event_log_visualizer/main.cc
+++ b/rtc_tools/rtc_event_log_visualizer/main.cc
@@ -251,7 +251,7 @@ int main(int argc, char* argv[]) {
        {"incoming_bitrate", "outgoing_bitrate", "incoming_ecn_feedback",
         "outgoing_ecn_feedback"}},
       {"scream",
-       {"scream_delay_estimates", "scream_ref_window",
+       {"scream_ref_window", "simulated_scream_delay",
         "simulated_scream_bitrates", "simulated_scream_ref_window",
         "simulated_scream_ratios", "network_delay_feedback", "pacer_delay"}}};
 
diff --git a/test/peer_scenario/bwe_integration_tests/scream_test.cc b/test/peer_scenario/bwe_integration_tests/scream_test.cc
index 33b5901f24..8ed6a436b0 100644
--- a/test/peer_scenario/bwe_integration_tests/scream_test.cc
+++ b/test/peer_scenario/bwe_integration_tests/scream_test.cc
@@ -337,7 +337,7 @@ TEST(
   EXPECT_THAT(
       result.caller().subspan(1, 3),
       Each(AvailableSendBitrateIsBetween(DataRate::KilobitsPerSec(10),
-                                         DataRate::KilobitsPerSec(100))));
+                                         DataRate::KilobitsPerSec(120))));
   EXPECT_THAT(result.caller()[3],
               CurrentRoundTripTimeIsBetween(TimeDelta::Millis(40),
                                             TimeDelta::Millis(200)));
@@ -347,7 +347,7 @@ TEST(
   // is disabled.
   EXPECT_THAT(result.caller_stats.back(),
               AvailableSendBitrateIsBetween(DataRate::KilobitsPerSec(100),
-                                            DataRate::KilobitsPerSec(5000)));
+                                            DataRate::KilobitsPerSec(6500)));
 }
 
 TEST(ScreamTest, MaybeTest(LinkCapacity600KbpsRtt20msNoEcn)) {
@@ -363,8 +363,8 @@ TEST(ScreamTest, MaybeTest(LinkCapacity600KbpsRtt20msNoEcn)) {
 
   SendMediaTestResult result = SendMediaInOneDirection(std::move(params), s);
   EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
-                                              DataRate::KilobitsPerSec(300),
-                                              DataRate::KilobitsPerSec(700))));
+                                              DataRate::KilobitsPerSec(200),
+                                              DataRate::KilobitsPerSec(800))));
 }
 
 TEST(ScreamTest, MaybeTest(LinkCapacity600KbpsRtt100msEcn)) {
@@ -467,7 +467,7 @@ TEST(ScreamTest, MaybeTest(LinkCapacity1500KbpsRtt30msNoEcn)) {
   SendMediaTestResult result = SendMediaInOneDirection(std::move(params), s);
   EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
                                               DataRate::KilobitsPerSec(800),
-                                              DataRate::KilobitsPerSec(1900))));
+                                              DataRate::KilobitsPerSec(2000))));
 }
 
 TEST(ScreamTest, MaybeTest(LinkCapacity2MbpsRtt50msNoEcn)) {
@@ -484,7 +484,7 @@ TEST(ScreamTest, MaybeTest(LinkCapacity2MbpsRtt50msNoEcn)) {
   SendMediaTestResult result = SendMediaInOneDirection(std::move(params), s);
   EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
                                               DataRate::KilobitsPerSec(1300),
-                                              DataRate::KilobitsPerSec(2300))));
+                                              DataRate::KilobitsPerSec(2400))));
 }
 
 TEST(ScreamTest, MaybeTest(LinkCapacity2MbpsRtt50msEcn)) {
@@ -525,6 +525,26 @@ TEST(ScreamTest, MaybeTest(LinkCapacity2MbpsRtt50msNoEcnWithGoogCC)) {
                                             DataRate::KilobitsPerSec(2600)));
 }
 
+TEST(ScreamTest, MaybeTest(LinkCapacity4MbpsRtt50ms10MsDelayStdDev)) {
+  PeerScenario s(*testing::UnitTest::GetInstance()->current_test_info());
+  NetworkEmulationManager::SimulatedNetworkNode::Builder network_builder(
+      s.net());
+  network_builder.delay_ms(25);
+  network_builder.delay_standard_deviation_ms(10);
+  network_builder.capacity_Mbps(4);
+  SendMediaTestParams params;
+  params.test_duration = TimeDelta::Seconds(30);
+  params.callee_to_caller_path =
+      CreateNetworkPath(network_builder, /*use_dual_pi= */ false);
+  params.caller_to_callee_path =
+      CreateNetworkPath(network_builder, /*use_dual_pi= */ false);
+
+  SendMediaTestResult result = SendMediaInOneDirection(std::move(params), s);
+  EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
+                                              DataRate::KilobitsPerSec(700),
+                                              DataRate::KilobitsPerSec(4100))));
+}
+
 TEST(ScreamTest, MaybeTest(LinkCapacity2MbpsRtt50msNoEcnWithTwcc)) {
   PeerScenario s(*testing::UnitTest::GetInstance()->current_test_info());
   SendMediaTestParams params;
@@ -545,7 +565,7 @@ TEST(ScreamTest, MaybeTest(LinkCapacity2MbpsRtt50msNoEcnWithTwcc)) {
   // approximately.
   EXPECT_THAT(result.caller().subspan(5), Each(AvailableSendBitrateIsBetween(
                                               DataRate::KilobitsPerSec(1200),
-                                              DataRate::KilobitsPerSec(2300))));
+                                              DataRate::KilobitsPerSec(2400))));
 }
 
 TEST(ScreamTest, MaybeTest(CallerPauseSendingVideoIfFeedbackNotReceived)) {
@@ -627,7 +647,7 @@ TEST(ScreamTest, MaybeTest(ScreencastSlideChange2Mbit50msRttNoEcn)) {
   // Ignore estimate during rampup.
   EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
                                               DataRate::KilobitsPerSec(1200),
-                                              DataRate::KilobitsPerSec(2600))));
+                                              DataRate::KilobitsPerSec(2800))));
 }
 
 TEST(ScreamTest, MaybeTest(ScreencastSlideChangeRepeatedDelaySpikes)) {
@@ -645,12 +665,9 @@ TEST(ScreamTest, MaybeTest(ScreencastSlideChangeRepeatedDelaySpikes)) {
                             .change_interval = TimeDelta::Seconds(5)}}};
   SendMediaTestResult result = SendMediaInOneDirection(std::move(params), s);
 
-  // Ignore estimate during rampup.
-  // TODO: bugs.webrtc.org/447037083 - consider improving resilience by limiting
-  // queue delay from one feedback.
-  EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
-                                              DataRate::KilobitsPerSec(100),
-                                              DataRate::KilobitsPerSec(4000))));
+  EXPECT_THAT(result.caller(), Each(AvailableSendBitrateIsBetween(
+                                   DataRate::KilobitsPerSec(500),
+                                   DataRate::KilobitsPerSec(4300))));
 }
 
 TEST(ScreamTest, MaybeTest(LinkCapacity5MbitRepeatedDelaySpikesNoEcn)) {
@@ -664,16 +681,14 @@ TEST(ScreamTest, MaybeTest(LinkCapacity5MbitRepeatedDelaySpikesNoEcn)) {
       TimeDelta::Millis(200));
   SendMediaTestResult result = SendMediaInOneDirection(std::move(params), s);
 
-  // TODO: bugs.webrtc.org/447037083 - consider improving resilience by limiting
-  // queue delay from one feedback.
   EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
-                                              DataRate::KilobitsPerSec(500),
+                                              DataRate::KilobitsPerSec(1200),
                                               DataRate::KilobitsPerSec(5000))));
 }
 
 TEST(ScreamTest, MaybeTest(RampupFastOnLinkCapacity50Mbit20MsRttNoEcn)) {
   PeerScenario s(*testing::UnitTest::GetInstance()->current_test_info());
-  SendMediaTestParams params{.test_duration = TimeDelta::Millis(300),
+  SendMediaTestParams params{.test_duration = TimeDelta::Seconds(1),
                              .stats_interval = TimeDelta::Millis(100)};
   params.caller_to_callee_path =
       CreateNetworkPath(s, /*use_dual_pi= */ false,
@@ -685,9 +700,14 @@ TEST(ScreamTest, MaybeTest(RampupFastOnLinkCapacity50Mbit20MsRttNoEcn)) {
 
   // TODO: bugs.webrtc.org/447037083 - On a good network, we should be able to
   // let the bitrate increase by 50% every RTT.
-  EXPECT_THAT(result.caller_stats.back(),
+  // After 300ms.
+  EXPECT_THAT(result.caller_stats[2],
               AvailableSendBitrateIsBetween(DataRate::KilobitsPerSec(500),
                                             DataRate::KilobitsPerSec(5000)));
+  // After 1s.
+  EXPECT_THAT(result.caller_stats.back(),
+              AvailableSendBitrateIsBetween(DataRate::KilobitsPerSec(1500),
+                                            DataRate::KilobitsPerSec(5000)));
 }
 
 TEST(ScreamTest, MaybeTest(LinkCapacity100Kbit50msRttNoEcn)) {
@@ -767,7 +787,7 @@ TEST(ScreamTest, MaybeTest(ReturnLinkWithBurstLoss)) {
             GetPacketsSent(result.caller_stats[5]));
   EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
                                               DataRate::KilobitsPerSec(300),
-                                              DataRate::KilobitsPerSec(1100))));
+                                              DataRate::KilobitsPerSec(1300))));
 }
 
 TEST(ScreamTest, MaybeTest(SendVideoOnlyReturnLinkWithBurstLoss)) {

From 561305fc4ff8f318735a088a77859f19e310de45 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Per=20=C3=85hgren?= <peah@webrtc.org>
Date: Thu, 12 Mar 2026 12:16:38 +0100
Subject: [PATCH 277/472] Removed the AEC3 namespace
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This CL removes the AEC3 namespace to reduce the number of nested
namespaces.

As part of this removal, one of the functions (ComputeFrequencyResponse(..), and one class (MovingAverage), had to be renamed to avoid naming clashes.

Bug: webrtc:42233815
Change-Id: I0f4dfac095f557cf1d6ee7d170c2f405eb0f7a9e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/454360
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47130}
---
 modules/audio_processing/aec3/BUILD.gn        |  6 ++--
 .../aec3/adaptive_fir_filter.cc               | 33 +++++++----------
 .../aec3/adaptive_fir_filter.h                |  5 +--
 .../aec3/adaptive_fir_filter_avx2.cc          |  3 --
 .../aec3/adaptive_fir_filter_erl.cc           | 12 +++----
 .../aec3/adaptive_fir_filter_erl.h            |  3 --
 .../aec3/adaptive_fir_filter_erl_avx2.cc      |  3 --
 .../aec3/adaptive_fir_filter_erl_unittest.cc  |  2 --
 .../aec3/adaptive_fir_filter_unittest.cc      |  8 ++---
 .../aec3/comfort_noise_generator.cc           |  2 +-
 .../aec3/comfort_noise_generator.h            |  3 --
 .../aec3/comfort_noise_generator_unittest.cc  |  2 --
 .../aec3/echo_remover_metrics.cc              | 28 +++++++--------
 .../aec3/echo_remover_metrics.h               |  4 ---
 .../aec3/echo_remover_metrics_unittest.cc     | 26 +++++++-------
 .../audio_processing/aec3/matched_filter.cc   | 22 +++++-------
 .../audio_processing/aec3/matched_filter.h    |  4 ---
 .../aec3/matched_filter_avx2.cc               |  2 --
 .../aec3/matched_filter_unittest.cc           |  2 --
 ..._average.cc => moving_average_spectrum.cc} | 12 +++----
 ...ng_average.h => moving_average_spectrum.h} | 18 +++++-----
 ...cc => moving_average_spectrum_unittest.cc} | 10 +++---
 .../aec3/subband_nearend_detector.cc          |  6 ++--
 .../aec3/subband_nearend_detector.h           |  4 +--
 .../aec3/suppression_filter.cc                |  2 +-
 .../audio_processing/aec3/suppression_gain.cc |  8 ++---
 .../audio_processing/aec3/suppression_gain.h  |  4 +--
 .../aec3/suppression_gain_unittest.cc         |  2 --
 modules/audio_processing/aec3/vector_math.h   |  3 --
 .../audio_processing/aec3/vector_math_avx2.cc |  2 --
 .../aec3/vector_math_unittest.cc              | 36 +++++++++----------
 31 files changed, 107 insertions(+), 170 deletions(-)
 rename modules/audio_processing/aec3/{moving_average.cc => moving_average_spectrum.cc} (81%)
 rename modules/audio_processing/aec3/{moving_average.h => moving_average_spectrum.h} (65%)
 rename modules/audio_processing/aec3/{moving_average_unittest.cc => moving_average_spectrum_unittest.cc} (92%)

diff --git a/modules/audio_processing/aec3/BUILD.gn b/modules/audio_processing/aec3/BUILD.gn
index 10b055f4e2..6b5279b56a 100644
--- a/modules/audio_processing/aec3/BUILD.gn
+++ b/modules/audio_processing/aec3/BUILD.gn
@@ -75,8 +75,8 @@ rtc_library("aec3") {
     "matched_filter.cc",
     "matched_filter_lag_aggregator.cc",
     "matched_filter_lag_aggregator.h",
-    "moving_average.cc",
-    "moving_average.h",
+    "moving_average_spectrum.cc",
+    "moving_average_spectrum.h",
     "multi_channel_content_detector.cc",
     "multi_channel_content_detector.h",
     "nearend_detector.h",
@@ -369,7 +369,7 @@ if (rtc_include_tests) {
         "frame_blocker_unittest.cc",
         "matched_filter_lag_aggregator_unittest.cc",
         "matched_filter_unittest.cc",
-        "moving_average_unittest.cc",
+        "moving_average_spectrum_unittest.cc",
         "multi_channel_content_detector_unittest.cc",
         "refined_filter_update_gain_unittest.cc",
         "render_buffer_unittest.cc",
diff --git a/modules/audio_processing/aec3/adaptive_fir_filter.cc b/modules/audio_processing/aec3/adaptive_fir_filter.cc
index 52718c0526..142f786f67 100644
--- a/modules/audio_processing/aec3/adaptive_fir_filter.cc
+++ b/modules/audio_processing/aec3/adaptive_fir_filter.cc
@@ -34,10 +34,8 @@
 
 namespace webrtc {
 
-namespace aec3 {
-
 // Computes and stores the frequency response of the filter.
-void ComputeFrequencyResponse(
+void ComputeFrequencyResponse_C(
     size_t num_partitions,
     const std::vector<std::vector<FftData>>& H,
     std::vector<std::array<float, kFftLengthBy2Plus1>>* H2) {
@@ -455,8 +453,6 @@ void ApplyFilter_Sse2(const RenderBuffer& render_buffer,
 }
 #endif
 
-}  // namespace aec3
-
 namespace {
 
 // Ensures that the newly added filter partitions after a size increase are set
@@ -563,19 +559,19 @@ void AdaptiveFirFilter::Filter(const RenderBuffer& render_buffer,
   switch (optimization_) {
 #if defined(WEBRTC_ARCH_X86_FAMILY)
     case Aec3Optimization::kSse2:
-      aec3::ApplyFilter_Sse2(render_buffer, current_size_partitions_, H_, S);
+      ApplyFilter_Sse2(render_buffer, current_size_partitions_, H_, S);
       break;
     case Aec3Optimization::kAvx2:
-      aec3::ApplyFilter_Avx2(render_buffer, current_size_partitions_, H_, S);
+      ApplyFilter_Avx2(render_buffer, current_size_partitions_, H_, S);
       break;
 #endif
 #if defined(WEBRTC_HAS_NEON)
     case Aec3Optimization::kNeon:
-      aec3::ApplyFilter_Neon(render_buffer, current_size_partitions_, H_, S);
+      ApplyFilter_Neon(render_buffer, current_size_partitions_, H_, S);
       break;
 #endif
     default:
-      aec3::ApplyFilter(render_buffer, current_size_partitions_, H_, S);
+      ApplyFilter(render_buffer, current_size_partitions_, H_, S);
   }
 }
 
@@ -607,19 +603,19 @@ void AdaptiveFirFilter::ComputeFrequencyResponse(
   switch (optimization_) {
 #if defined(WEBRTC_ARCH_X86_FAMILY)
     case Aec3Optimization::kSse2:
-      aec3::ComputeFrequencyResponse_Sse2(current_size_partitions_, H_, H2);
+      ComputeFrequencyResponse_Sse2(current_size_partitions_, H_, H2);
       break;
     case Aec3Optimization::kAvx2:
-      aec3::ComputeFrequencyResponse_Avx2(current_size_partitions_, H_, H2);
+      ComputeFrequencyResponse_Avx2(current_size_partitions_, H_, H2);
       break;
 #endif
 #if defined(WEBRTC_HAS_NEON)
     case Aec3Optimization::kNeon:
-      aec3::ComputeFrequencyResponse_Neon(current_size_partitions_, H_, H2);
+      ComputeFrequencyResponse_Neon(current_size_partitions_, H_, H2);
       break;
 #endif
     default:
-      aec3::ComputeFrequencyResponse(current_size_partitions_, H_, H2);
+      ComputeFrequencyResponse_C(current_size_partitions_, H_, H2);
   }
 }
 
@@ -632,22 +628,19 @@ void AdaptiveFirFilter::AdaptAndUpdateSize(const RenderBuffer& render_buffer,
   switch (optimization_) {
 #if defined(WEBRTC_ARCH_X86_FAMILY)
     case Aec3Optimization::kSse2:
-      aec3::AdaptPartitions_Sse2(render_buffer, G, current_size_partitions_,
-                                 &H_);
+      AdaptPartitions_Sse2(render_buffer, G, current_size_partitions_, &H_);
       break;
     case Aec3Optimization::kAvx2:
-      aec3::AdaptPartitions_Avx2(render_buffer, G, current_size_partitions_,
-                                 &H_);
+      AdaptPartitions_Avx2(render_buffer, G, current_size_partitions_, &H_);
       break;
 #endif
 #if defined(WEBRTC_HAS_NEON)
     case Aec3Optimization::kNeon:
-      aec3::AdaptPartitions_Neon(render_buffer, G, current_size_partitions_,
-                                 &H_);
+      AdaptPartitions_Neon(render_buffer, G, current_size_partitions_, &H_);
       break;
 #endif
     default:
-      aec3::AdaptPartitions(render_buffer, G, current_size_partitions_, &H_);
+      AdaptPartitions(render_buffer, G, current_size_partitions_, &H_);
   }
 }
 
diff --git a/modules/audio_processing/aec3/adaptive_fir_filter.h b/modules/audio_processing/aec3/adaptive_fir_filter.h
index 681cd23d56..d960a2aa03 100644
--- a/modules/audio_processing/aec3/adaptive_fir_filter.h
+++ b/modules/audio_processing/aec3/adaptive_fir_filter.h
@@ -25,9 +25,8 @@
 #include "rtc_base/system/arch.h"
 
 namespace webrtc {
-namespace aec3 {
 // Computes and stores the frequency response of the filter.
-void ComputeFrequencyResponse(
+void ComputeFrequencyResponse_C(
     size_t num_partitions,
     const std::vector<std::vector<FftData>>& H,
     std::vector<std::array<float, kFftLengthBy2Plus1>>* H2);
@@ -95,8 +94,6 @@ void ApplyFilter_Avx2(const RenderBuffer& render_buffer,
                       FftData* S);
 #endif
 
-}  // namespace aec3
-
 // Provides a frequency domain adaptive filter functionality.
 class AdaptiveFirFilter {
  public:
diff --git a/modules/audio_processing/aec3/adaptive_fir_filter_avx2.cc b/modules/audio_processing/aec3/adaptive_fir_filter_avx2.cc
index f250f84806..327f0fb16d 100644
--- a/modules/audio_processing/aec3/adaptive_fir_filter_avx2.cc
+++ b/modules/audio_processing/aec3/adaptive_fir_filter_avx2.cc
@@ -24,8 +24,6 @@
 
 namespace webrtc {
 
-namespace aec3 {
-
 // Computes and stores the frequency response of the filter.
 void ComputeFrequencyResponse_Avx2(
     size_t num_partitions,
@@ -193,5 +191,4 @@ void ApplyFilter_Avx2(const RenderBuffer& render_buffer,
   } while (p < lim2);
 }
 
-}  // namespace aec3
 }  // namespace webrtc
diff --git a/modules/audio_processing/aec3/adaptive_fir_filter_erl.cc b/modules/audio_processing/aec3/adaptive_fir_filter_erl.cc
index 77d1487352..19ffe86ef8 100644
--- a/modules/audio_processing/aec3/adaptive_fir_filter_erl.cc
+++ b/modules/audio_processing/aec3/adaptive_fir_filter_erl.cc
@@ -31,8 +31,6 @@
 
 namespace webrtc {
 
-namespace aec3 {
-
 // Computes and stores the echo return loss estimate of the filter, which is the
 // sum of the partition frequency responses.
 void ErlComputer(const std::vector<std::array<float, kFftLengthBy2Plus1>>& H2,
@@ -82,8 +80,6 @@ void ErlComputer_SSE2(
 }
 #endif
 
-}  // namespace aec3
-
 void ComputeErl(const Aec3Optimization& optimization,
                 const std::vector<std::array<float, kFftLengthBy2Plus1>>& H2,
                 std::span<float> erl) {
@@ -92,19 +88,19 @@ void ComputeErl(const Aec3Optimization& optimization,
   switch (optimization) {
 #if defined(WEBRTC_ARCH_X86_FAMILY)
     case Aec3Optimization::kSse2:
-      aec3::ErlComputer_SSE2(H2, erl);
+      ErlComputer_SSE2(H2, erl);
       break;
     case Aec3Optimization::kAvx2:
-      aec3::ErlComputer_AVX2(H2, erl);
+      ErlComputer_AVX2(H2, erl);
       break;
 #endif
 #if defined(WEBRTC_HAS_NEON)
     case Aec3Optimization::kNeon:
-      aec3::ErlComputer_NEON(H2, erl);
+      ErlComputer_NEON(H2, erl);
       break;
 #endif
     default:
-      aec3::ErlComputer(H2, erl);
+      ErlComputer(H2, erl);
   }
 }
 
diff --git a/modules/audio_processing/aec3/adaptive_fir_filter_erl.h b/modules/audio_processing/aec3/adaptive_fir_filter_erl.h
index d504d270ec..60a5c1e678 100644
--- a/modules/audio_processing/aec3/adaptive_fir_filter_erl.h
+++ b/modules/audio_processing/aec3/adaptive_fir_filter_erl.h
@@ -21,7 +21,6 @@
 #include "rtc_base/system/arch.h"
 
 namespace webrtc {
-namespace aec3 {
 
 // Computes and stores the echo return loss estimate of the filter, which is the
 // sum of the partition frequency responses.
@@ -42,8 +41,6 @@ void ErlComputer_AVX2(
     std::span<float> erl);
 #endif
 
-}  // namespace aec3
-
 // Computes the echo return loss based on a frequency response.
 void ComputeErl(const Aec3Optimization& optimization,
                 const std::vector<std::array<float, kFftLengthBy2Plus1>>& H2,
diff --git a/modules/audio_processing/aec3/adaptive_fir_filter_erl_avx2.cc b/modules/audio_processing/aec3/adaptive_fir_filter_erl_avx2.cc
index c3606aa10b..3b67f02247 100644
--- a/modules/audio_processing/aec3/adaptive_fir_filter_erl_avx2.cc
+++ b/modules/audio_processing/aec3/adaptive_fir_filter_erl_avx2.cc
@@ -21,8 +21,6 @@
 
 namespace webrtc {
 
-namespace aec3 {
-
 // Computes and stores the echo return loss estimate of the filter, which is the
 // sum of the partition frequency responses.
 void ErlComputer_AVX2(
@@ -40,5 +38,4 @@ void ErlComputer_AVX2(
   }
 }
 
-}  // namespace aec3
 }  // namespace webrtc
diff --git a/modules/audio_processing/aec3/adaptive_fir_filter_erl_unittest.cc b/modules/audio_processing/aec3/adaptive_fir_filter_erl_unittest.cc
index d95aad1a24..64d4f04b0b 100644
--- a/modules/audio_processing/aec3/adaptive_fir_filter_erl_unittest.cc
+++ b/modules/audio_processing/aec3/adaptive_fir_filter_erl_unittest.cc
@@ -25,7 +25,6 @@
 #endif
 
 namespace webrtc {
-namespace aec3 {
 
 #if defined(WEBRTC_HAS_NEON)
 // Verifies that the optimized method for echo return loss computation is
@@ -105,5 +104,4 @@ TEST(AdaptiveFirFilter, UpdateErlAvx2Optimization) {
 
 #endif
 
-}  // namespace aec3
 }  // namespace webrtc
diff --git a/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc b/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc
index 60e9628657..e4c5481204 100644
--- a/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc
+++ b/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc
@@ -52,7 +52,6 @@
 #endif
 
 namespace webrtc {
-namespace aec3 {
 namespace {
 
 std::string ProduceDebugText(size_t num_render_channels, size_t delay) {
@@ -168,7 +167,7 @@ TEST_P(AdaptiveFirFilterOneTwoFourEightRenderChannels,
       }
     }
 
-    ComputeFrequencyResponse(num_partitions, H, &H2);
+    ComputeFrequencyResponse_C(num_partitions, H, &H2);
     ComputeFrequencyResponse_Neon(num_partitions, H, &H2_Neon);
 
     for (size_t p = 0; p < num_partitions; ++p) {
@@ -348,7 +347,7 @@ TEST_P(AdaptiveFirFilterOneTwoFourEightRenderChannels,
         }
       }
 
-      ComputeFrequencyResponse(num_partitions, H, &H2);
+      ComputeFrequencyResponse_C(num_partitions, H, &H2);
       ComputeFrequencyResponse_Sse2(num_partitions, H, &H2_Sse2);
 
       for (size_t p = 0; p < num_partitions; ++p) {
@@ -383,7 +382,7 @@ TEST_P(AdaptiveFirFilterOneTwoFourEightRenderChannels,
         }
       }
 
-      ComputeFrequencyResponse(num_partitions, H, &H2);
+      ComputeFrequencyResponse_C(num_partitions, H, &H2);
       ComputeFrequencyResponse_Avx2(num_partitions, H, &H2_Avx2);
 
       for (size_t p = 0; p < num_partitions; ++p) {
@@ -620,5 +619,4 @@ TEST_P(AdaptiveFirFilterMultiChannel, FilterAndAdapt) {
   }
 }
 
-}  // namespace aec3
 }  // namespace webrtc
diff --git a/modules/audio_processing/aec3/comfort_noise_generator.cc b/modules/audio_processing/aec3/comfort_noise_generator.cc
index b7f30de870..58b2323d25 100644
--- a/modules/audio_processing/aec3/comfort_noise_generator.cc
+++ b/modules/audio_processing/aec3/comfort_noise_generator.cc
@@ -94,7 +94,7 @@ void GenerateComfortNoise(
   // Compute square root spectrum.
   std::array<float, kFftLengthBy2Plus1> N;
   std::copy(N2.begin(), N2.end(), N.begin());
-  aec3::VectorMath(optimization).Sqrt(N);
+  VectorMath(optimization).Sqrt(N);
 
   // Compute the noise level for the upper bands.
   constexpr float kOneByNumBands = 1.f / (kFftLengthBy2Plus1 / 2 + 1);
diff --git a/modules/audio_processing/aec3/comfort_noise_generator.h b/modules/audio_processing/aec3/comfort_noise_generator.h
index c28e269ade..05cd43e79a 100644
--- a/modules/audio_processing/aec3/comfort_noise_generator.h
+++ b/modules/audio_processing/aec3/comfort_noise_generator.h
@@ -25,7 +25,6 @@
 #include "rtc_base/system/arch.h"
 
 namespace webrtc {
-namespace aec3 {
 #if defined(WEBRTC_ARCH_X86_FAMILY)
 
 void EstimateComfortNoise_SSE2(const std::array<float, kFftLengthBy2Plus1>& N2,
@@ -38,8 +37,6 @@ void EstimateComfortNoise(const std::array<float, kFftLengthBy2Plus1>& N2,
                           FftData* lower_band_noise,
                           FftData* upper_band_noise);
 
-}  // namespace aec3
-
 // Generates the comfort noise.
 class ComfortNoiseGenerator {
  public:
diff --git a/modules/audio_processing/aec3/comfort_noise_generator_unittest.cc b/modules/audio_processing/aec3/comfort_noise_generator_unittest.cc
index 17ce0477d7..e985023109 100644
--- a/modules/audio_processing/aec3/comfort_noise_generator_unittest.cc
+++ b/modules/audio_processing/aec3/comfort_noise_generator_unittest.cc
@@ -23,7 +23,6 @@
 #include "test/gtest.h"
 
 namespace webrtc {
-namespace aec3 {
 namespace {
 
 float Power(const FftData& N) {
@@ -70,5 +69,4 @@ TEST(ComfortNoiseGenerator, CorrectLevel) {
   }
 }
 
-}  // namespace aec3
 }  // namespace webrtc
diff --git a/modules/audio_processing/aec3/echo_remover_metrics.cc b/modules/audio_processing/aec3/echo_remover_metrics.cc
index fce3964da5..a3f31675b0 100644
--- a/modules/audio_processing/aec3/echo_remover_metrics.cc
+++ b/modules/audio_processing/aec3/echo_remover_metrics.cc
@@ -78,35 +78,35 @@ void EchoRemoverMetrics::Update(
       case kMetricsCollectionBlocks + 2:
         RTC_HISTOGRAM_COUNTS_LINEAR(
             "WebRTC.Audio.EchoCanceller.Erl.Value",
-            aec3::TransformDbMetricForReporting(true, 0.f, 59.f, 30.f, 1.f,
-                                                erl_time_domain_.sum_value),
+            TransformDbMetricForReporting(true, 0.f, 59.f, 30.f, 1.f,
+                                          erl_time_domain_.sum_value),
             0, 59, 30);
         RTC_HISTOGRAM_COUNTS_LINEAR(
             "WebRTC.Audio.EchoCanceller.Erl.Max",
-            aec3::TransformDbMetricForReporting(true, 0.f, 59.f, 30.f, 1.f,
-                                                erl_time_domain_.ceil_value),
+            TransformDbMetricForReporting(true, 0.f, 59.f, 30.f, 1.f,
+                                          erl_time_domain_.ceil_value),
             0, 59, 30);
         RTC_HISTOGRAM_COUNTS_LINEAR(
             "WebRTC.Audio.EchoCanceller.Erl.Min",
-            aec3::TransformDbMetricForReporting(true, 0.f, 59.f, 30.f, 1.f,
-                                                erl_time_domain_.floor_value),
+            TransformDbMetricForReporting(true, 0.f, 59.f, 30.f, 1.f,
+                                          erl_time_domain_.floor_value),
             0, 59, 30);
         break;
       case kMetricsCollectionBlocks + 3:
         RTC_HISTOGRAM_COUNTS_LINEAR(
             "WebRTC.Audio.EchoCanceller.Erle.Value",
-            aec3::TransformDbMetricForReporting(false, 0.f, 19.f, 0.f, 1.f,
-                                                erle_time_domain_.sum_value),
+            TransformDbMetricForReporting(false, 0.f, 19.f, 0.f, 1.f,
+                                          erle_time_domain_.sum_value),
             0, 19, 20);
         RTC_HISTOGRAM_COUNTS_LINEAR(
             "WebRTC.Audio.EchoCanceller.Erle.Max",
-            aec3::TransformDbMetricForReporting(false, 0.f, 19.f, 0.f, 1.f,
-                                                erle_time_domain_.ceil_value),
+            TransformDbMetricForReporting(false, 0.f, 19.f, 0.f, 1.f,
+                                          erle_time_domain_.ceil_value),
             0, 19, 20);
         RTC_HISTOGRAM_COUNTS_LINEAR(
             "WebRTC.Audio.EchoCanceller.Erle.Min",
-            aec3::TransformDbMetricForReporting(false, 0.f, 19.f, 0.f, 1.f,
-                                                erle_time_domain_.floor_value),
+            TransformDbMetricForReporting(false, 0.f, 19.f, 0.f, 1.f,
+                                          erle_time_domain_.floor_value),
             0, 19, 20);
         metrics_reported_ = true;
         RTC_DCHECK_EQ(kMetricsReportingIntervalBlocks, block_counter_);
@@ -120,8 +120,6 @@ void EchoRemoverMetrics::Update(
   }
 }
 
-namespace aec3 {
-
 void UpdateDbMetric(const std::array<float, kFftLengthBy2Plus1>& value,
                     std::array<EchoRemoverMetrics::DbMetric, 2>* statistic) {
   RTC_DCHECK(statistic);
@@ -153,6 +151,4 @@ int TransformDbMetricForReporting(bool negate,
   return static_cast<int>(SafeClamp(new_value, min_value, max_value));
 }
 
-}  // namespace aec3
-
 }  // namespace webrtc
diff --git a/modules/audio_processing/aec3/echo_remover_metrics.h b/modules/audio_processing/aec3/echo_remover_metrics.h
index aec8084d78..ebcd27772f 100644
--- a/modules/audio_processing/aec3/echo_remover_metrics.h
+++ b/modules/audio_processing/aec3/echo_remover_metrics.h
@@ -56,8 +56,6 @@ class EchoRemoverMetrics {
   bool metrics_reported_ = false;
 };
 
-namespace aec3 {
-
 // Updates a banded metric of type DbMetric with the values in the supplied
 // array.
 void UpdateDbMetric(const std::array<float, kFftLengthBy2Plus1>& value,
@@ -71,8 +69,6 @@ int TransformDbMetricForReporting(bool negate,
                                   float scaling,
                                   float value);
 
-}  // namespace aec3
-
 }  // namespace webrtc
 
 #endif  // MODULES_AUDIO_PROCESSING_AEC3_ECHO_REMOVER_METRICS_H_
diff --git a/modules/audio_processing/aec3/echo_remover_metrics_unittest.cc b/modules/audio_processing/aec3/echo_remover_metrics_unittest.cc
index 85b04b77e0..b087ad152f 100644
--- a/modules/audio_processing/aec3/echo_remover_metrics_unittest.cc
+++ b/modules/audio_processing/aec3/echo_remover_metrics_unittest.cc
@@ -31,7 +31,7 @@ namespace webrtc {
 TEST(UpdateDbMetricDeathTest, NullValue) {
   std::array<float, kFftLengthBy2Plus1> value;
   value.fill(0.f);
-  EXPECT_DEATH(aec3::UpdateDbMetric(value, nullptr), "");
+  EXPECT_DEATH(UpdateDbMetric(value, nullptr), "");
 }
 
 #endif
@@ -46,7 +46,7 @@ TEST(UpdateDbMetric, Updating) {
   std::fill(value.begin(), value.begin() + 32, kValue0);
   std::fill(value.begin() + 32, value.begin() + 64, kValue1);
 
-  aec3::UpdateDbMetric(value, &statistic);
+  UpdateDbMetric(value, &statistic);
   EXPECT_FLOAT_EQ(kValue0, statistic[0].sum_value);
   EXPECT_FLOAT_EQ(kValue0, statistic[0].ceil_value);
   EXPECT_FLOAT_EQ(kValue0, statistic[0].floor_value);
@@ -54,7 +54,7 @@ TEST(UpdateDbMetric, Updating) {
   EXPECT_FLOAT_EQ(kValue1, statistic[1].ceil_value);
   EXPECT_FLOAT_EQ(kValue1, statistic[1].floor_value);
 
-  aec3::UpdateDbMetric(value, &statistic);
+  UpdateDbMetric(value, &statistic);
   EXPECT_FLOAT_EQ(2.f * kValue0, statistic[0].sum_value);
   EXPECT_FLOAT_EQ(kValue0, statistic[0].ceil_value);
   EXPECT_FLOAT_EQ(kValue0, statistic[0].floor_value);
@@ -78,26 +78,26 @@ TEST(TransformDbMetricForReporting, DbFsScaling) {
   EXPECT_NEAR(offset, -90.3f, 0.1f);
   EXPECT_EQ(
       static_cast<int>(30.3f),
-      aec3::TransformDbMetricForReporting(
-          true, 0.f, 90.f, offset, 1.f / (kBlockSize * kBlockSize), X2[0]));
+      TransformDbMetricForReporting(true, 0.f, 90.f, offset,
+                                    1.f / (kBlockSize * kBlockSize), X2[0]));
 }
 
 // Verifies that the TransformDbMetricForReporting method is able to properly
 // limit the output.
 TEST(TransformDbMetricForReporting, Limits) {
-  EXPECT_EQ(0, aec3::TransformDbMetricForReporting(false, 0.f, 10.f, 0.f, 1.f,
-                                                   0.001f));
-  EXPECT_EQ(10, aec3::TransformDbMetricForReporting(false, 0.f, 10.f, 0.f, 1.f,
-                                                    100.f));
+  EXPECT_EQ(0,
+            TransformDbMetricForReporting(false, 0.f, 10.f, 0.f, 1.f, 0.001f));
+  EXPECT_EQ(10,
+            TransformDbMetricForReporting(false, 0.f, 10.f, 0.f, 1.f, 100.f));
 }
 
 // Verifies that the TransformDbMetricForReporting method is able to properly
 // negate output.
 TEST(TransformDbMetricForReporting, Negate) {
-  EXPECT_EQ(10, aec3::TransformDbMetricForReporting(true, -20.f, 20.f, 0.f, 1.f,
-                                                    0.1f));
-  EXPECT_EQ(-10, aec3::TransformDbMetricForReporting(true, -20.f, 20.f, 0.f,
-                                                     1.f, 10.f));
+  EXPECT_EQ(10,
+            TransformDbMetricForReporting(true, -20.f, 20.f, 0.f, 1.f, 0.1f));
+  EXPECT_EQ(-10,
+            TransformDbMetricForReporting(true, -20.f, 20.f, 0.f, 1.f, 10.f));
 }
 
 // Verify the Update functionality of DbMetric.
diff --git a/modules/audio_processing/aec3/matched_filter.cc b/modules/audio_processing/aec3/matched_filter.cc
index 961d8a88db..69962e9f1a 100644
--- a/modules/audio_processing/aec3/matched_filter.cc
+++ b/modules/audio_processing/aec3/matched_filter.cc
@@ -79,8 +79,6 @@ size_t ComputePreEchoLag(const std::span<const float> accumulated_error,
 
 }  // namespace
 
-namespace aec3 {
-
 #if defined(WEBRTC_HAS_NEON)
 
 inline float SumAllElements(float32x4_t elements) {
@@ -591,8 +589,6 @@ size_t MaxSquarePeakIndex(std::span<const float> h) {
   return lag_estimate1;
 }
 
-}  // namespace aec3
-
 MatchedFilter::MatchedFilter(ApmDataDumper* data_dumper,
                              Aec3Optimization optimization,
                              size_t sub_block_size,
@@ -694,13 +690,13 @@ void MatchedFilter::Update(const DownsampledRenderBuffer& render_buffer,
     switch (optimization_) {
 #if defined(WEBRTC_ARCH_X86_FAMILY)
       case Aec3Optimization::kSse2:
-        aec3::MatchedFilterCore_SSE2(
+        MatchedFilterCore_SSE2(
             x_start_index, x2_sum_threshold, smoothing, render_buffer.buffer, y,
             filters_[n], &filters_updated, &error_sum, compute_pre_echo,
             instantaneous_accumulated_error_, scratch_memory_);
         break;
       case Aec3Optimization::kAvx2:
-        aec3::MatchedFilterCore_AVX2(
+        MatchedFilterCore_AVX2(
             x_start_index, x2_sum_threshold, smoothing, render_buffer.buffer, y,
             filters_[n], &filters_updated, &error_sum, compute_pre_echo,
             instantaneous_accumulated_error_, scratch_memory_);
@@ -708,23 +704,23 @@ void MatchedFilter::Update(const DownsampledRenderBuffer& render_buffer,
 #endif
 #if defined(WEBRTC_HAS_NEON)
       case Aec3Optimization::kNeon:
-        aec3::MatchedFilterCore_NEON(
+        MatchedFilterCore_NEON(
             x_start_index, x2_sum_threshold, smoothing, render_buffer.buffer, y,
             filters_[n], &filters_updated, &error_sum, compute_pre_echo,
             instantaneous_accumulated_error_, scratch_memory_);
         break;
 #endif
       default:
-        aec3::MatchedFilterCore(x_start_index, x2_sum_threshold, smoothing,
-                                render_buffer.buffer, y, filters_[n],
-                                &filters_updated, &error_sum, compute_pre_echo,
-                                instantaneous_accumulated_error_);
+        MatchedFilterCore(x_start_index, x2_sum_threshold, smoothing,
+                          render_buffer.buffer, y, filters_[n],
+                          &filters_updated, &error_sum, compute_pre_echo,
+                          instantaneous_accumulated_error_);
     }
 
     // Estimate the lag in the matched filter as the distance to the portion in
     // the filter that contributes the most to the matched filter output. This
     // is detected as the peak of the matched filter.
-    const size_t lag_estimate = aec3::MaxSquarePeakIndex(filters_[n]);
+    const size_t lag_estimate = MaxSquarePeakIndex(filters_[n]);
     const bool reliable =
         lag_estimate > 2 && lag_estimate < (filters_[n].size() - 10) &&
         error_sum < matching_filter_threshold_ * error_sum_anchor;
@@ -798,7 +794,7 @@ void MatchedFilter::LogFilterProperties(int /* sample_rate_hz */,
 
 void MatchedFilter::Dump() {
   for (size_t n = 0; n < filters_.size(); ++n) {
-    const size_t lag_estimate = aec3::MaxSquarePeakIndex(filters_[n]);
+    const size_t lag_estimate = MaxSquarePeakIndex(filters_[n]);
     std::string dumper_filter = "aec3_correlator_" + std::to_string(n) + "_h";
     data_dumper_->DumpRaw(dumper_filter.c_str(), filters_[n]);
     std::string dumper_lag = "aec3_correlator_lag_" + std::to_string(n);
diff --git a/modules/audio_processing/aec3/matched_filter.h b/modules/audio_processing/aec3/matched_filter.h
index 993670eafc..b0b7a0dff1 100644
--- a/modules/audio_processing/aec3/matched_filter.h
+++ b/modules/audio_processing/aec3/matched_filter.h
@@ -25,8 +25,6 @@ namespace webrtc {
 class ApmDataDumper;
 struct DownsampledRenderBuffer;
 
-namespace aec3 {
-
 #if defined(WEBRTC_HAS_NEON)
 
 // Filter core for the matched filter that is optimized for NEON.
@@ -89,8 +87,6 @@ void MatchedFilterCore(size_t x_start_index,
 // Find largest peak of squared values in array.
 size_t MaxSquarePeakIndex(std::span<const float> h);
 
-}  // namespace aec3
-
 // Produces recursively updated cross-correlation estimates for several signal
 // shifts where the intra-shift spacing is uniform.
 class MatchedFilter {
diff --git a/modules/audio_processing/aec3/matched_filter_avx2.cc b/modules/audio_processing/aec3/matched_filter_avx2.cc
index 0b9360c210..eacd6486b1 100644
--- a/modules/audio_processing/aec3/matched_filter_avx2.cc
+++ b/modules/audio_processing/aec3/matched_filter_avx2.cc
@@ -18,7 +18,6 @@
 #include "rtc_base/checks.h"
 
 namespace webrtc {
-namespace aec3 {
 
 // Let ha denote the horizontal of a, and hb the horizontal sum of b
 // returns [ha, hb, ha, hb]
@@ -259,5 +258,4 @@ void MatchedFilterCore_AVX2(size_t x_start_index,
   }
 }
 
-}  // namespace aec3
 }  // namespace webrtc
diff --git a/modules/audio_processing/aec3/matched_filter_unittest.cc b/modules/audio_processing/aec3/matched_filter_unittest.cc
index 7369ec2a57..d7f6f5e0bd 100644
--- a/modules/audio_processing/aec3/matched_filter_unittest.cc
+++ b/modules/audio_processing/aec3/matched_filter_unittest.cc
@@ -41,7 +41,6 @@
 #endif
 
 namespace webrtc {
-namespace aec3 {
 namespace {
 
 std::string ProduceDebugText(size_t delay, size_t down_sampling_factor) {
@@ -565,5 +564,4 @@ INSTANTIATE_TEST_SUITE_P(_,
 
 #endif
 
-}  // namespace aec3
 }  // namespace webrtc
diff --git a/modules/audio_processing/aec3/moving_average.cc b/modules/audio_processing/aec3/moving_average_spectrum.cc
similarity index 81%
rename from modules/audio_processing/aec3/moving_average.cc
rename to modules/audio_processing/aec3/moving_average_spectrum.cc
index 503b86bf9b..3884c4a57a 100644
--- a/modules/audio_processing/aec3/moving_average.cc
+++ b/modules/audio_processing/aec3/moving_average_spectrum.cc
@@ -9,7 +9,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "modules/audio_processing/aec3/moving_average.h"
+#include "modules/audio_processing/aec3/moving_average_spectrum.h"
 
 #include <algorithm>
 #include <cstddef>
@@ -19,9 +19,8 @@
 #include "rtc_base/checks.h"
 
 namespace webrtc {
-namespace aec3 {
 
-MovingAverage::MovingAverage(size_t num_elem, size_t mem_len)
+MovingAverageSpectrum::MovingAverageSpectrum(size_t num_elem, size_t mem_len)
     : num_elem_(num_elem),
       mem_len_(mem_len - 1),
       scaling_(1.0f / static_cast<float>(mem_len)),
@@ -31,10 +30,10 @@ MovingAverage::MovingAverage(size_t num_elem, size_t mem_len)
   RTC_DCHECK(mem_len > 0);
 }
 
-MovingAverage::~MovingAverage() = default;
+MovingAverageSpectrum::~MovingAverageSpectrum() = default;
 
-void MovingAverage::Average(std::span<const float> input,
-                            std::span<float> output) {
+void MovingAverageSpectrum::Average(std::span<const float> input,
+                                    std::span<float> output) {
   RTC_DCHECK(input.size() == num_elem_);
   RTC_DCHECK(output.size() == num_elem_);
 
@@ -58,5 +57,4 @@ void MovingAverage::Average(std::span<const float> input,
   }
 }
 
-}  // namespace aec3
 }  // namespace webrtc
diff --git a/modules/audio_processing/aec3/moving_average.h b/modules/audio_processing/aec3/moving_average_spectrum.h
similarity index 65%
rename from modules/audio_processing/aec3/moving_average.h
rename to modules/audio_processing/aec3/moving_average_spectrum.h
index 77e2a5907d..b4a3763311 100644
--- a/modules/audio_processing/aec3/moving_average.h
+++ b/modules/audio_processing/aec3/moving_average_spectrum.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef MODULES_AUDIO_PROCESSING_AEC3_MOVING_AVERAGE_H_
-#define MODULES_AUDIO_PROCESSING_AEC3_MOVING_AVERAGE_H_
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_MOVING_AVERAGE_SPECTRUM_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_MOVING_AVERAGE_SPECTRUM_H_
 
 #include <stddef.h>
 
@@ -17,14 +17,13 @@
 #include <vector>
 
 namespace webrtc {
-namespace aec3 {
 
-class MovingAverage {
+class MovingAverageSpectrum {
  public:
-  // Creates an instance of MovingAverage that accepts inputs of length num_elem
-  // and averages over mem_len inputs.
-  MovingAverage(size_t num_elem, size_t mem_len);
-  ~MovingAverage();
+  // Creates an instance of MovingAverageSpectrum that accepts inputs of length
+  // num_elem and averages over mem_len inputs.
+  MovingAverageSpectrum(size_t num_elem, size_t mem_len);
+  ~MovingAverageSpectrum();
 
   // Computes the average of input and mem_len-1 previous inputs and stores the
   // result in output.
@@ -38,7 +37,6 @@ class MovingAverage {
   size_t mem_index_;
 };
 
-}  // namespace aec3
 }  // namespace webrtc
 
-#endif  // MODULES_AUDIO_PROCESSING_AEC3_MOVING_AVERAGE_H_
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_MOVING_AVERAGE_SPECTRUM_H_
diff --git a/modules/audio_processing/aec3/moving_average_unittest.cc b/modules/audio_processing/aec3/moving_average_spectrum_unittest.cc
similarity index 92%
rename from modules/audio_processing/aec3/moving_average_unittest.cc
rename to modules/audio_processing/aec3/moving_average_spectrum_unittest.cc
index ca985176cd..718069b611 100644
--- a/modules/audio_processing/aec3/moving_average_unittest.cc
+++ b/modules/audio_processing/aec3/moving_average_spectrum_unittest.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "modules/audio_processing/aec3/moving_average.h"
+#include "modules/audio_processing/aec3/moving_average_spectrum.h"
 
 #include <array>
 #include <cstddef>
@@ -17,11 +17,11 @@
 
 namespace webrtc {
 
-TEST(MovingAverage, Average) {
+TEST(MovingAverageSpectrum, Average) {
   constexpr size_t num_elem = 4;
   constexpr size_t mem_len = 3;
   constexpr float e = 1e-6f;
-  aec3::MovingAverage ma(num_elem, mem_len);
+  MovingAverageSpectrum ma(num_elem, mem_len);
   std::array<float, num_elem> data1 = {1, 2, 3, 4};
   std::array<float, num_elem> data2 = {5, 1, 9, 7};
   std::array<float, num_elem> data3 = {3, 3, 5, 6};
@@ -53,11 +53,11 @@ TEST(MovingAverage, Average) {
   EXPECT_NEAR(output[3], (data2[3] + data3[3] + data4[3]) / 3.0f, e);
 }
 
-TEST(MovingAverage, PassThrough) {
+TEST(MovingAverageSpectrum, PassThrough) {
   constexpr size_t num_elem = 4;
   constexpr size_t mem_len = 1;
   constexpr float e = 1e-6f;
-  aec3::MovingAverage ma(num_elem, mem_len);
+  MovingAverageSpectrum ma(num_elem, mem_len);
   std::array<float, num_elem> data1 = {1, 2, 3, 4};
   std::array<float, num_elem> data2 = {5, 1, 9, 7};
   std::array<float, num_elem> data3 = {3, 3, 5, 6};
diff --git a/modules/audio_processing/aec3/subband_nearend_detector.cc b/modules/audio_processing/aec3/subband_nearend_detector.cc
index e569a3f131..39908acf59 100644
--- a/modules/audio_processing/aec3/subband_nearend_detector.cc
+++ b/modules/audio_processing/aec3/subband_nearend_detector.cc
@@ -17,7 +17,7 @@
 
 #include "api/audio/echo_canceller3_config.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
-#include "modules/audio_processing/aec3/moving_average.h"
+#include "modules/audio_processing/aec3/moving_average_spectrum.h"
 
 namespace webrtc {
 SubbandNearendDetector::SubbandNearendDetector(
@@ -26,8 +26,8 @@ SubbandNearendDetector::SubbandNearendDetector(
     : config_(config),
       num_capture_channels_(num_capture_channels),
       nearend_smoothers_(num_capture_channels_,
-                         aec3::MovingAverage(kFftLengthBy2Plus1,
-                                             config_.nearend_average_blocks)),
+                         MovingAverageSpectrum(kFftLengthBy2Plus1,
+                                               config_.nearend_average_blocks)),
       one_over_subband_length1_(
           1.f / (config_.subband1.high - config_.subband1.low + 1)),
       one_over_subband_length2_(
diff --git a/modules/audio_processing/aec3/subband_nearend_detector.h b/modules/audio_processing/aec3/subband_nearend_detector.h
index 635e063b24..5b4333199d 100644
--- a/modules/audio_processing/aec3/subband_nearend_detector.h
+++ b/modules/audio_processing/aec3/subband_nearend_detector.h
@@ -18,7 +18,7 @@
 
 #include "api/audio/echo_canceller3_config.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
-#include "modules/audio_processing/aec3/moving_average.h"
+#include "modules/audio_processing/aec3/moving_average_spectrum.h"
 #include "modules/audio_processing/aec3/nearend_detector.h"
 
 namespace webrtc {
@@ -44,7 +44,7 @@ class SubbandNearendDetector : public NearendDetector {
  private:
   const EchoCanceller3Config::Suppressor::SubbandNearendDetection config_;
   const size_t num_capture_channels_;
-  std::vector<aec3::MovingAverage> nearend_smoothers_;
+  std::vector<MovingAverageSpectrum> nearend_smoothers_;
   const float one_over_subband_length1_;
   const float one_over_subband_length2_;
   bool nearend_state_ = false;
diff --git a/modules/audio_processing/aec3/suppression_filter.cc b/modules/audio_processing/aec3/suppression_filter.cc
index b5a6f79fb2..bd895e8c5a 100644
--- a/modules/audio_processing/aec3/suppression_filter.cc
+++ b/modules/audio_processing/aec3/suppression_filter.cc
@@ -100,7 +100,7 @@ void SuppressionFilter::ApplyGain(
   for (size_t i = 0; i < kFftLengthBy2Plus1; ++i) {
     noise_gain[i] = 1.f - suppression_gain[i] * suppression_gain[i];
   }
-  aec3::VectorMath(optimization_).Sqrt(noise_gain);
+  VectorMath(optimization_).Sqrt(noise_gain);
 
   const float high_bands_noise_scaling =
       0.4f * std::sqrt(1.f - high_bands_gain * high_bands_gain);
diff --git a/modules/audio_processing/aec3/suppression_gain.cc b/modules/audio_processing/aec3/suppression_gain.cc
index bd92aff2e8..aab8a325a3 100644
--- a/modules/audio_processing/aec3/suppression_gain.cc
+++ b/modules/audio_processing/aec3/suppression_gain.cc
@@ -25,7 +25,7 @@
 #include "modules/audio_processing/aec3/aec_state.h"
 #include "modules/audio_processing/aec3/block.h"
 #include "modules/audio_processing/aec3/dominant_nearend_detector.h"
-#include "modules/audio_processing/aec3/moving_average.h"
+#include "modules/audio_processing/aec3/moving_average_spectrum.h"
 #include "modules/audio_processing/aec3/render_signal_analyzer.h"
 #include "modules/audio_processing/aec3/subband_nearend_detector.h"
 #include "modules/audio_processing/aec3/vector_math.h"
@@ -333,7 +333,7 @@ void SuppressionGain::LowerBandGain(
   std::copy(gain->begin(), gain->end(), last_gain_.begin());
 
   // Transform gains to amplitude domain.
-  aec3::VectorMath(optimization_).Sqrt(*gain);
+  VectorMath(optimization_).Sqrt(*gain);
 }
 
 SuppressionGain::SuppressionGain(const EchoCanceller3Config& config,
@@ -350,8 +350,8 @@ SuppressionGain::SuppressionGain(const EchoCanceller3Config& config,
       last_echo_(num_capture_channels_, {0}),
       nearend_smoothers_(
           num_capture_channels_,
-          aec3::MovingAverage(kFftLengthBy2Plus1,
-                              config.suppressor.nearend_average_blocks)),
+          MovingAverageSpectrum(kFftLengthBy2Plus1,
+                                config.suppressor.nearend_average_blocks)),
       nearend_params_(config_.suppressor.last_lf_band,
                       config_.suppressor.first_hf_band,
                       config_.suppressor.nearend_tuning),
diff --git a/modules/audio_processing/aec3/suppression_gain.h b/modules/audio_processing/aec3/suppression_gain.h
index 550b533cbf..4bcd0d23c3 100644
--- a/modules/audio_processing/aec3/suppression_gain.h
+++ b/modules/audio_processing/aec3/suppression_gain.h
@@ -23,7 +23,7 @@
 #include "modules/audio_processing/aec3/aec3_common.h"
 #include "modules/audio_processing/aec3/aec_state.h"
 #include "modules/audio_processing/aec3/block.h"
-#include "modules/audio_processing/aec3/moving_average.h"
+#include "modules/audio_processing/aec3/moving_average_spectrum.h"
 #include "modules/audio_processing/aec3/nearend_detector.h"
 #include "modules/audio_processing/aec3/render_signal_analyzer.h"
 #include "modules/audio_processing/logging/apm_data_dumper.h"
@@ -130,7 +130,7 @@ class SuppressionGain {
   LowNoiseRenderDetector low_render_detector_;
   bool initial_state_ = true;
   int initial_state_change_counter_ = 0;
-  std::vector<aec3::MovingAverage> nearend_smoothers_;
+  std::vector<MovingAverageSpectrum> nearend_smoothers_;
   const GainParameters nearend_params_;
   const GainParameters normal_params_;
   // Determines if the dominant nearend detector uses the unbounded residual
diff --git a/modules/audio_processing/aec3/suppression_gain_unittest.cc b/modules/audio_processing/aec3/suppression_gain_unittest.cc
index 2da9589261..60d290d78b 100644
--- a/modules/audio_processing/aec3/suppression_gain_unittest.cc
+++ b/modules/audio_processing/aec3/suppression_gain_unittest.cc
@@ -34,7 +34,6 @@
 #include "test/gtest.h"
 
 namespace webrtc {
-namespace aec3 {
 
 #if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
 
@@ -160,5 +159,4 @@ TEST(SuppressionGain, BasicGainComputation) {
                 [](float a) { EXPECT_NEAR(0.0f, a, 0.001f); });
 }
 
-}  // namespace aec3
 }  // namespace webrtc
diff --git a/modules/audio_processing/aec3/vector_math.h b/modules/audio_processing/aec3/vector_math.h
index d8b5ecf3d1..c733aa0228 100644
--- a/modules/audio_processing/aec3/vector_math.h
+++ b/modules/audio_processing/aec3/vector_math.h
@@ -29,7 +29,6 @@
 #endif
 
 namespace webrtc {
-namespace aec3 {
 
 // Provides optimizations for mathematical operations based on vectors.
 class VectorMath {
@@ -220,8 +219,6 @@ class VectorMath {
   Aec3Optimization optimization_;
 };
 
-}  // namespace aec3
-
 }  // namespace webrtc
 
 #endif  // MODULES_AUDIO_PROCESSING_AEC3_VECTOR_MATH_H_
diff --git a/modules/audio_processing/aec3/vector_math_avx2.cc b/modules/audio_processing/aec3/vector_math_avx2.cc
index 2288420f40..ec25104873 100644
--- a/modules/audio_processing/aec3/vector_math_avx2.cc
+++ b/modules/audio_processing/aec3/vector_math_avx2.cc
@@ -17,7 +17,6 @@
 #include "rtc_base/checks.h"
 
 namespace webrtc {
-namespace aec3 {
 
 // Elementwise square root.
 void VectorMath::SqrtAVX2(std::span<float> x) {
@@ -77,5 +76,4 @@ void VectorMath::AccumulateAVX2(std::span<const float> x, std::span<float> z) {
   }
 }
 
-}  // namespace aec3
 }  // namespace webrtc
diff --git a/modules/audio_processing/aec3/vector_math_unittest.cc b/modules/audio_processing/aec3/vector_math_unittest.cc
index a47263b305..1f3525b6ac 100644
--- a/modules/audio_processing/aec3/vector_math_unittest.cc
+++ b/modules/audio_processing/aec3/vector_math_unittest.cc
@@ -34,9 +34,9 @@ TEST(VectorMath, Sqrt) {
   }
 
   std::copy(x.begin(), x.end(), z.begin());
-  aec3::VectorMath(Aec3Optimization::kNone).Sqrt(z);
+  VectorMath(Aec3Optimization::kNone).Sqrt(z);
   std::copy(x.begin(), x.end(), z_neon.begin());
-  aec3::VectorMath(Aec3Optimization::kNeon).Sqrt(z_neon);
+  VectorMath(Aec3Optimization::kNeon).Sqrt(z_neon);
   for (size_t k = 0; k < z.size(); ++k) {
     EXPECT_NEAR(z[k], z_neon[k], 0.0001f);
     EXPECT_NEAR(sqrtf(x[k]), z_neon[k], 0.0001f);
@@ -54,8 +54,8 @@ TEST(VectorMath, Multiply) {
     y[k] = (2.f / 3.f) * k;
   }
 
-  aec3::VectorMath(Aec3Optimization::kNone).Multiply(x, y, z);
-  aec3::VectorMath(Aec3Optimization::kNeon).Multiply(x, y, z_neon);
+  VectorMath(Aec3Optimization::kNone).Multiply(x, y, z);
+  VectorMath(Aec3Optimization::kNeon).Multiply(x, y, z_neon);
   for (size_t k = 0; k < z.size(); ++k) {
     EXPECT_FLOAT_EQ(z[k], z_neon[k]);
     EXPECT_FLOAT_EQ(x[k] * y[k], z_neon[k]);
@@ -72,8 +72,8 @@ TEST(VectorMath, Accumulate) {
     z[k] = z_neon[k] = 2.f * k;
   }
 
-  aec3::VectorMath(Aec3Optimization::kNone).Accumulate(x, z);
-  aec3::VectorMath(Aec3Optimization::kNeon).Accumulate(x, z_neon);
+  VectorMath(Aec3Optimization::kNone).Accumulate(x, z);
+  VectorMath(Aec3Optimization::kNeon).Accumulate(x, z_neon);
   for (size_t k = 0; k < z.size(); ++k) {
     EXPECT_FLOAT_EQ(z[k], z_neon[k]);
     EXPECT_FLOAT_EQ(x[k] + 2.f * x[k], z_neon[k]);
@@ -94,9 +94,9 @@ TEST(VectorMath, Sse2Sqrt) {
     }
 
     std::copy(x.begin(), x.end(), z.begin());
-    aec3::VectorMath(Aec3Optimization::kNone).Sqrt(z);
+    VectorMath(Aec3Optimization::kNone).Sqrt(z);
     std::copy(x.begin(), x.end(), z_sse2.begin());
-    aec3::VectorMath(Aec3Optimization::kSse2).Sqrt(z_sse2);
+    VectorMath(Aec3Optimization::kSse2).Sqrt(z_sse2);
     EXPECT_EQ(z, z_sse2);
     for (size_t k = 0; k < z.size(); ++k) {
       EXPECT_FLOAT_EQ(z[k], z_sse2[k]);
@@ -116,9 +116,9 @@ TEST(VectorMath, Avx2Sqrt) {
     }
 
     std::copy(x.begin(), x.end(), z.begin());
-    aec3::VectorMath(Aec3Optimization::kNone).Sqrt(z);
+    VectorMath(Aec3Optimization::kNone).Sqrt(z);
     std::copy(x.begin(), x.end(), z_avx2.begin());
-    aec3::VectorMath(Aec3Optimization::kAvx2).Sqrt(z_avx2);
+    VectorMath(Aec3Optimization::kAvx2).Sqrt(z_avx2);
     EXPECT_EQ(z, z_avx2);
     for (size_t k = 0; k < z.size(); ++k) {
       EXPECT_FLOAT_EQ(z[k], z_avx2[k]);
@@ -139,8 +139,8 @@ TEST(VectorMath, Sse2Multiply) {
       y[k] = (2.f / 3.f) * k;
     }
 
-    aec3::VectorMath(Aec3Optimization::kNone).Multiply(x, y, z);
-    aec3::VectorMath(Aec3Optimization::kSse2).Multiply(x, y, z_sse2);
+    VectorMath(Aec3Optimization::kNone).Multiply(x, y, z);
+    VectorMath(Aec3Optimization::kSse2).Multiply(x, y, z_sse2);
     for (size_t k = 0; k < z.size(); ++k) {
       EXPECT_FLOAT_EQ(z[k], z_sse2[k]);
       EXPECT_FLOAT_EQ(x[k] * y[k], z_sse2[k]);
@@ -160,8 +160,8 @@ TEST(VectorMath, Avx2Multiply) {
       y[k] = (2.f / 3.f) * k;
     }
 
-    aec3::VectorMath(Aec3Optimization::kNone).Multiply(x, y, z);
-    aec3::VectorMath(Aec3Optimization::kAvx2).Multiply(x, y, z_avx2);
+    VectorMath(Aec3Optimization::kNone).Multiply(x, y, z);
+    VectorMath(Aec3Optimization::kAvx2).Multiply(x, y, z_avx2);
     for (size_t k = 0; k < z.size(); ++k) {
       EXPECT_FLOAT_EQ(z[k], z_avx2[k]);
       EXPECT_FLOAT_EQ(x[k] * y[k], z_avx2[k]);
@@ -180,8 +180,8 @@ TEST(VectorMath, Sse2Accumulate) {
       z[k] = z_sse2[k] = 2.f * k;
     }
 
-    aec3::VectorMath(Aec3Optimization::kNone).Accumulate(x, z);
-    aec3::VectorMath(Aec3Optimization::kSse2).Accumulate(x, z_sse2);
+    VectorMath(Aec3Optimization::kNone).Accumulate(x, z);
+    VectorMath(Aec3Optimization::kSse2).Accumulate(x, z_sse2);
     for (size_t k = 0; k < z.size(); ++k) {
       EXPECT_FLOAT_EQ(z[k], z_sse2[k]);
       EXPECT_FLOAT_EQ(x[k] + 2.f * x[k], z_sse2[k]);
@@ -200,8 +200,8 @@ TEST(VectorMath, Avx2Accumulate) {
       z[k] = z_avx2[k] = 2.f * k;
     }
 
-    aec3::VectorMath(Aec3Optimization::kNone).Accumulate(x, z);
-    aec3::VectorMath(Aec3Optimization::kAvx2).Accumulate(x, z_avx2);
+    VectorMath(Aec3Optimization::kNone).Accumulate(x, z);
+    VectorMath(Aec3Optimization::kAvx2).Accumulate(x, z_avx2);
     for (size_t k = 0; k < z.size(); ++k) {
       EXPECT_FLOAT_EQ(z[k], z_avx2[k]);
       EXPECT_FLOAT_EQ(x[k] + 2.f * x[k], z_avx2[k]);

From 7c36f795d925c3f08fb89611166cc6b2e2de02a6 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Erik=20Spr=C3=A5ng?= <sprang@webrtc.org>
Date: Thu, 12 Mar 2026 11:55:13 +0100
Subject: [PATCH 278/472] Add workaround for OpenH264 related to U/V strides.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This CL adds a workaround for a bug in OpenH264, where using an input
frame where the U and V strides are different may result in a crash.

This fix should be removed once the root cause has been addressed.

Bug: chromium:491655161
Change-Id: I2e8bccefab3ffcd08bf1e086763407ceb461bba7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456121
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47131}
---
 .../codecs/h264/h264_encoder_impl.cc          |  5 ++++
 .../codecs/h264/h264_encoder_impl_unittest.cc | 26 +++++++++++++++++++
 2 files changed, 31 insertions(+)

diff --git a/modules/video_coding/codecs/h264/h264_encoder_impl.cc b/modules/video_coding/codecs/h264/h264_encoder_impl.cc
index 02602029b8..b682013e62 100644
--- a/modules/video_coding/codecs/h264/h264_encoder_impl.cc
+++ b/modules/video_coding/codecs/h264/h264_encoder_impl.cc
@@ -450,6 +450,11 @@ int32_t H264EncoderImpl::Encode(
                       << " image to I420. Can't encode frame.";
     return WEBRTC_VIDEO_CODEC_ENCODER_FAILURE;
   }
+  if (frame_buffer->StrideU() != frame_buffer->StrideV()) {
+    // TODO: crbug.com/chromium:491655161 - Remove once the root cause is fixed.
+    RTC_LOG(LS_ERROR) << "OpenH264 requires the U and V strides to be equal.";
+    return WEBRTC_VIDEO_CODEC_ENCODER_FAILURE;
+  }
   RTC_CHECK(frame_buffer->type() == VideoFrameBuffer::Type::kI420 ||
             frame_buffer->type() == VideoFrameBuffer::Type::kI420A);
 
diff --git a/modules/video_coding/codecs/h264/h264_encoder_impl_unittest.cc b/modules/video_coding/codecs/h264/h264_encoder_impl_unittest.cc
index 0149a8cd86..e5aade5188 100644
--- a/modules/video_coding/codecs/h264/h264_encoder_impl_unittest.cc
+++ b/modules/video_coding/codecs/h264/h264_encoder_impl_unittest.cc
@@ -18,6 +18,7 @@
 #include "api/test/create_frame_generator.h"
 #include "api/test/frame_generator_interface.h"
 #include "api/test/mock_video_encoder.h"
+#include "api/video/i420_buffer.h"
 #include "api/video/video_codec_type.h"
 #include "api/video/video_frame.h"
 #include "api/video_codecs/video_codec.h"
@@ -137,6 +138,31 @@ TEST(H264EncoderImplTest, OnFrameDropped) {
   }
 }
 
+TEST(H264EncoderImplTest, RejectsFramesWithUnequalChromaStrides) {
+  H264EncoderImpl encoder(CreateEnvironment(), {});
+  VideoCodec codec_settings;
+  SetDefaultSettings(&codec_settings);
+  MockEncodedImageCallback callback;
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder.InitEncode(&codec_settings, kSettings));
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder.RegisterEncodeCompleteCallback(&callback));
+  // Create a VideoFrame where the U and V strides are different.
+  auto buffer = I420Buffer::Create(
+      /*width=*/codec_settings.width,
+      /*height=*/codec_settings.height,
+      /*stride_y=*/codec_settings.width,
+      /*stride_u=*/(codec_settings.width + 1) / 2,
+      /*stride_v=*/(codec_settings.width + 1) / 2 + 1);
+
+  VideoFrame frame = VideoFrame::Builder()
+                         .set_video_frame_buffer(buffer)
+                         .set_rtp_timestamp(0)
+                         .build();
+
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_ENCODER_FAILURE, encoder.Encode(frame, nullptr));
+}
+
 }  // anonymous namespace
 
 }  // namespace webrtc

From 0b3311e536ba488d36a731d66aa5d2fc917e0b67 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Bj=C3=B6rn=20Terelius?= <terelius@webrtc.org>
Date: Thu, 12 Mar 2026 12:54:17 +0100
Subject: [PATCH 279/472] Run iwyu on current CL if no files are given on
 command line
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Bug: None
Change-Id: I9e14e0f37bf50570dbb021543db9131a3638fb71
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456120
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47132}
---
 tools_webrtc/iwyu/apply_include_cleaner.py | 25 +++++++++++++++++++---
 1 file changed, 22 insertions(+), 3 deletions(-)

diff --git a/tools_webrtc/iwyu/apply_include_cleaner.py b/tools_webrtc/iwyu/apply_include_cleaner.py
index ad89a526a0..5f2470a6d3 100755
--- a/tools_webrtc/iwyu/apply_include_cleaner.py
+++ b/tools_webrtc/iwyu/apply_include_cleaner.py
@@ -43,7 +43,7 @@ import pathlib
 import os
 import subprocess
 import sys
-from typing import Tuple
+from typing import List, Tuple
 
 _CLEANER_BINARY_PATH = pathlib.Path(
     "third_party/llvm-build/Release+Asserts/bin/clang-include-cleaner")
@@ -111,7 +111,7 @@ def _parse_args() -> argparse.Namespace:
         formatter_class=argparse.ArgumentDefaultsHelpFormatter,
     )
     parser.add_argument("files",
-                        nargs="+",
+                        nargs="*",
                         type=_valid_file,
                         help="List of files to process")
     parser.add_argument(
@@ -227,6 +227,20 @@ def _modified_content(content: str) -> str:
     return modified_content
 
 
+def _fetch_modified_files(revision: str) -> List[pathlib.Path]:
+    print(f"Trying to find modified files relative to {revision}")
+    result = subprocess.run(
+        ["git", "diff", "--name-only", "--diff-filter=d", revision],
+        capture_output=True,
+        check=False)
+    if result.returncode != 0:
+        print(f"Failed to run git diff on {revision}, ",
+              f"{result.stderr.decode().strip()}")
+        return []
+    files = result.stdout.decode().split()
+    print("Found files:", '\n'.join(files))
+    return [_valid_file(file.strip()) for file in files]
+
 # Transitioning the cmd type to tuple to prevent modification of
 # the original command from the callsite in main...
 def apply_include_cleaner_to_file(file_path: pathlib.Path, should_modify: bool,
@@ -293,12 +307,17 @@ def main() -> None:
         should_modify = True
 
     changes_generated = False
+    files = args.files
+    if not files:
+        files = _fetch_modified_files("@{upstream}")
+    if not files:
+        files = _fetch_modified_files("main")
     # TODO(dorhen@meta): Ideally don't iterate on the files
     # and execute cleaner on each, but instead execute the
     # cleaner binary once - passing in all files.
     # e.g instead of `cleaner foo.cc && cleaner bar.cc`
     # do `cleaner foo.cc bar.cc`
-    for file in args.files:
+    for file in files:
         if not file.suffix in _SUFFICES:
             continue
         if not _is_built(file, args.work_dir):

From 447a59dcd83ca3fdc134a2f443187a949860445c Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Thu, 12 Mar 2026 11:43:00 +0000
Subject: [PATCH 280/472] Reland "Fix payload type allocation issues for Audio
 RED and MID recycling."
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This reverts commit e4f9241450a411e56dc4a56c400defe7ad19a1c0.

Reason for revert: Removed wrong tests for stability-with-flag.

Original change's description:
> Revert "Fix payload type allocation issues for Audio RED and MID recycling."
>
> This reverts commit 9e3381393a233cac896085fc3f26bdac83caaaef.
>
> Reason for revert: Breaks webrtc-importer
>
> Original change's description:
> > Fix payload type allocation issues for Audio RED and MID recycling.
> >
> > This CL addresses several issues identified while enabling the
> > WebRTC-PayloadTypesInTransport field trial:
> > - Enforce media type equality in codec matching to prevent Audio/Video
> > RED collisions.
> > - Relax RED matching rules to allow negotiation with unlinked RED codecs.
> > - Validate media type during MID recycling in CodecVendor to prevent
> > incorrect codec merging.
> > - Update documentation with the video implementation strategy.
> >
> > Bug: webrtc:360058654
> > Change-Id: I1ecbdad2179d346b3682875d823889aa076e353e
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453001
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#47113}
>
> Bug: webrtc:360058654
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Change-Id: I5f7c642e936c1b06781b78f10293837d1262d450
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/455820
> Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
> Owners-Override: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#47115}

Bug: webrtc:360058654
Change-Id: I4000cf06bd87a23f042a137a835f827580f7d0cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456100
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47133}
---
 g3doc/todo/payload_type_redesign.md           | 127 ++++++++++++++++++
 media/base/codec_comparators.cc               |  51 ++++---
 media/base/codec_comparators_unittest.cc      |  57 ++++++++
 pc/BUILD.gn                                   |   1 +
 pc/codec_vendor.cc                            |   7 +-
 ...er_connection_encodings_integrationtest.cc |  58 ++++++++
 pc/peer_connection_media_unittest.cc          |   4 +-
 7 files changed, 284 insertions(+), 21 deletions(-)
 create mode 100644 g3doc/todo/payload_type_redesign.md

diff --git a/g3doc/todo/payload_type_redesign.md b/g3doc/todo/payload_type_redesign.md
new file mode 100644
index 0000000000..5fa5f12bca
--- /dev/null
+++ b/g3doc/todo/payload_type_redesign.md
@@ -0,0 +1,127 @@
+# Payload type allocation redesign
+
+## Background: What a payload type is
+
+The payload type is a property of a codec that is established between a sender
+and a receiver using SDP offer/answer.
+
+Each end of the connection independently indicates the list of codecs it is
+willing to send and/or receive, and assigns a payload type to each. It is
+conventional but not required to use the same payload types in an answer as
+those suggested in an offer.
+
+For SDP sendonly media sections, it indicates a willingness to send; for
+recvonly media sections, it indicates a willingness to receive; for sendrecv
+media sections, it indicates a willingness to receive AND a willingness to send.
+Presence in a media section does not guarantee that it will be used. (RFC 3264)
+
+## Current allocation strategy
+
+The payload types are assigned by the VideoEngine and VoiceEngine according to a
+fixed list. If the assignments collide with already established payload types,
+the assignments are munged in the CodecVendor class so that there are no
+colliding payload types.
+
+The PayloadTypeRecorder class records what payload types are assigned - there is
+one case (PR-answer followed by a later Answer that has different PTs) where
+they are changed, but otherwise, once the PayloadTypeRecorder has assigned them,
+that PT is permanent for that transport and that direction.
+
+Picking a payload type is done in the PayloadTypePicker class. This is preloaded
+with some assignments, so that if a PT is free, the commonly used PT for that
+codec can be used.
+
+## Desired future strategy
+
+There should be no assignment of payload types in VideoEngine and VoiceEngine.
+Payload types should be picked only when creating an offer or answer; they are
+assigned permanently in SetLocalDescription / SetRemoteDescription.
+
+Once an assignment is made, it should never need to change (apart from the case
+noted above).
+
+## Dealing with resiliency mechanisms
+
+Resiliency mechanisms (RED, RTX) are signaled using "codecs", and are
+traditionally represented as such within the library. They associate with the
+codec they are resiliency for by using the "apt=" parameter in their a=fmtp
+lines. However, this complicates assignment, since that parameter cannot be
+filled in until the payload type for the "protected" codec is assigned.
+
+## Backwards compatibility issues
+
+Experience has shown that changing the PT allocation strategy often trips up
+applications that have depended on the result of the old allocation strategy, so
+user-visible changes should be avoided. One example is that it's preferred (but
+not strictly required) to have the RTX PT assigned the PT number of 1 more than
+the PT of the codec it refers to.
+
+## Current implementation status
+
+The new strategy is implemented for audio codecs. Several issues that caused
+test failures when enabling the `WebRTC-PayloadTypesInTransport` field trial
+have been identified and fixed:
+
+- **Audio/Video RED Collision:** RED codecs of different media types were
+  incorrectly matching, leading to payload type conflicts.
+  `MatchesWithCodecRules` now enforces media type equality.
+- **MID Recycling:** When a MID was recycled for a different media type (e.g.,
+  Audio -> Video), `CodecVendor` was incorrectly merging codecs from the old
+  description. This has been fixed by validating the media type before merging.
+- **RED Matching Logic:** Relaxed the matching rules for RED to allow
+  negotiation to proceed even when parameters (linking RED to primary codecs)
+  are not yet populated, as this linking now happens late in the `CodecVendor`.
+
+The new strategy is not yet implemented for video codecs.
+
+## Implementation Strategy for Video
+
+The goal is to transition video codec handling to the same late-assignment model
+used for audio.
+
+### 1. Unified Codec Collection
+
+- Implement `VideoCodecsFromFactory` in `pc/typed_codec_vendor.cc`. This will
+  query the `VideoEncoderFactory` and `VideoDecoderFactory` to gather supported
+  `SdpVideoFormat`s.
+- These codecs will be initialized with `kIdNotSet`, allowing
+  `SdpPayloadTypeSuggester` to assign payload types only during the creation of
+  an offer or answer.
+
+### 2. Payload Type Suggester Integration
+
+- Ensure `PayloadTypePicker` has a complete list of preferred payload types for
+  video codecs (VP8, VP9, H.264, AV1, etc.) to maintain stable and conventional
+  assignments.
+- Update `CodecVendor` to use the unassigned video codec list when the
+  `WebRTC-PayloadTypesInTransport` trial is active.
+
+### 3. Resiliency and Parameter Linking
+
+- Refine `MergeCodecs` and `AssignCodecIdsAndLinkRed` to handle video-specific
+  resiliency:
+  - **RTX:** Correctly link RTX codecs to their primary video codecs by matching
+    names and specific parameters (like `profile-level-id` for H.264).
+  - **RED/ULPFEC:** Ensure parameters for video redundancy are correctly
+    populated once the primary codec PTs are assigned.
+
+### 4. Verification and Testing
+
+- **Integration Tests:** Enable the `WebRTC-PayloadTypesInTransport` trial in
+  `peerconnection_unittests` and `rtc_unittests` to identify any video-specific
+  regressions.
+- **Stable PT Tests:** Add coverage to ensure that payload types remain stable
+  across renegotiations, even when the order of codecs in the transceiver
+  preferences changes.
+- **MID Recycling:** Verify that MID recycling for video-to-audio and vice-versa
+  works correctly without PT collisions or crashes.
+
+## Desired next steps
+
+Analyze the current situation. (Done)
+
+### Test, isolate and fix failures
+
+The identified failures in `PeerConnectionEncodingsIntegrationTest` and
+`PeerConnectionIntegrationTest` have been resolved. Next, focus on implementing
+the video strategy outlined above.
diff --git a/media/base/codec_comparators.cc b/media/base/codec_comparators.cc
index 9da9d4aca5..cb67f671a6 100644
--- a/media/base/codec_comparators.cc
+++ b/media/base/codec_comparators.cc
@@ -163,15 +163,7 @@ bool MatchesWithReferenceAttributesAndComparator(
         potential_match.params.find(kCodecParamNotInNameValueFormat);
     bool has_parameters_1 = red_parameters_1 != codec_to_match.params.end();
     bool has_parameters_2 = red_parameters_2 != potential_match.params.end();
-    // If codec_to_match has unassigned PT and no parameter,
-    // we assume that it'll be assigned later and return a match.
-    // Note - this should be deleted. It's untidy.
-    if (potential_match.id == Codec::kIdNotSet && !has_parameters_2) {
-      return true;
-    }
-    if (codec_to_match.id == Codec::kIdNotSet && !has_parameters_1) {
-      return true;
-    }
+
     if (has_parameters_1 && has_parameters_2) {
       // Different levels of redundancy between offer and answer are OK
       // since RED is considered to be declarative.
@@ -203,7 +195,16 @@ bool MatchesWithReferenceAttributesAndComparator(
       return true;
     }
     if (!has_parameters_1 && !has_parameters_2) {
-      // Both parameters are missing. Happens for video RED.
+      return true;
+    }
+    // Exactly one lacks parameters.
+    // Allow match if it is an audio RED codec and at least one of the
+    // codecs has an unassigned payload type or they have the same ID.
+    if (codec_to_match.type == Codec::Type::kAudio &&
+        codec_to_match.name == kRedCodecName &&
+        (codec_to_match.id == Codec::kIdNotSet ||
+         potential_match.id == Codec::kIdNotSet ||
+         codec_to_match.id == potential_match.id)) {
       return true;
     }
     return false;
@@ -295,6 +296,11 @@ bool MatchesWithCodecRules(const Codec& left_codec, const Codec& right_codec) {
        right_codec.id <= kLowerDynamicRangeMax) ||
       (right_codec.id >= kUpperDynamicRangeMin &&
        right_codec.id <= kUpperDynamicRangeMax);
+
+  if (left_codec.type != right_codec.type) {
+    return false;
+  }
+
   bool matches_id;
   if ((is_id_in_dynamic_range && is_codec_id_in_dynamic_range) ||
       left_codec.id == Codec::kIdNotSet || right_codec.id == Codec::kIdNotSet) {
@@ -360,12 +366,22 @@ std::optional<Codec> FindMatchingCodec(const std::vector<Codec>& codecs1,
 bool IsSameRtpCodec(const Codec& codec, const RtpCodec& rtp_codec) {
   RtpCodecParameters rtp_codec2 = codec.ToCodecParameters();
 
-  return absl::EqualsIgnoreCase(rtp_codec.name, rtp_codec2.name) &&
-         rtp_codec.kind == rtp_codec2.kind &&
-         rtp_codec.num_channels == rtp_codec2.num_channels &&
-         rtp_codec.clock_rate == rtp_codec2.clock_rate &&
-         InsertDefaultParams(rtp_codec.name, rtp_codec.parameters) ==
-             InsertDefaultParams(rtp_codec2.name, rtp_codec2.parameters);
+  if (!absl::EqualsIgnoreCase(rtp_codec.name, rtp_codec2.name) ||
+      rtp_codec.kind != rtp_codec2.kind ||
+      rtp_codec.num_channels != rtp_codec2.num_channels ||
+      rtp_codec.clock_rate != rtp_codec2.clock_rate) {
+    return false;
+  }
+
+  // audio/RED should ignore the parameters which specify payload types so
+  // can not be compared.
+  if (rtp_codec.kind == MediaType::AUDIO &&
+      absl::EqualsIgnoreCase(rtp_codec.name, kRedCodecName)) {
+    return true;
+  }
+
+  return InsertDefaultParams(rtp_codec.name, rtp_codec.parameters) ==
+         InsertDefaultParams(rtp_codec2.name, rtp_codec2.parameters);
 }
 
 bool IsSameRtpCodecIgnoringLevel(const Codec& codec,
@@ -393,7 +409,8 @@ bool IsSameRtpCodecIgnoringLevel(const Codec& codec,
   }
   // audio/RED should ignore the parameters which specify payload types so
   // can not be compared.
-  if (rtp_codec.kind == MediaType::AUDIO && rtp_codec.name == kRedCodecName) {
+  if (rtp_codec.kind == MediaType::AUDIO &&
+      absl::EqualsIgnoreCase(rtp_codec.name, kRedCodecName)) {
     return true;
   }
 
diff --git a/media/base/codec_comparators_unittest.cc b/media/base/codec_comparators_unittest.cc
index 2f78042c16..048765670e 100644
--- a/media/base/codec_comparators_unittest.cc
+++ b/media/base/codec_comparators_unittest.cc
@@ -105,6 +105,63 @@ TEST(CodecComparatorsTest, MatchesWithReferenceAttributesRed) {
   EXPECT_FALSE(MatchesWithReferenceAttributes(codec_1, codec_5));
 }
 
+TEST(CodecComparatorsTest, RedParametersMismatch) {
+  Codec with_params = CreateAudioCodec(101, kRedCodecName, 48000, 2);
+  with_params.SetParam(kCodecParamNotInNameValueFormat, "111/111");
+
+  Codec no_params = CreateAudioCodec(102, kRedCodecName, 48000, 2);
+
+  // Case 1: Exactly one lacks parameters, both have IDs -> SHOULD NOT match for
+  // audio RED.
+  EXPECT_FALSE(MatchesWithReferenceAttributes(with_params, no_params));
+  EXPECT_FALSE(MatchesWithReferenceAttributes(no_params, with_params));
+
+  // Case 2: Exactly one lacks parameters, the one WITHOUT parameters is
+  // unassigned -> SHOULD match.
+  // This is the case with late assignment.
+  Codec no_params_unassigned = no_params;
+  no_params_unassigned.id = Codec::kIdNotSet;
+  EXPECT_TRUE(
+      MatchesWithReferenceAttributes(with_params, no_params_unassigned));
+  EXPECT_TRUE(
+      MatchesWithReferenceAttributes(no_params_unassigned, with_params));
+
+  // Case 3: Exactly one lacks parameters, the one WITH parameters is
+  // unassigned -> SHOULD match for audio RED because one side is unassigned.
+  Codec with_params_unassigned = with_params;
+  with_params_unassigned.id = Codec::kIdNotSet;
+  EXPECT_TRUE(
+      MatchesWithReferenceAttributes(with_params_unassigned, no_params));
+  EXPECT_TRUE(
+      MatchesWithReferenceAttributes(no_params, with_params_unassigned));
+
+  // Case 4: Exactly one lacks parameters, both are unassigned -> SHOULD match.
+  EXPECT_TRUE(MatchesWithReferenceAttributes(with_params_unassigned,
+                                             no_params_unassigned));
+  EXPECT_TRUE(MatchesWithReferenceAttributes(no_params_unassigned,
+                                             with_params_unassigned));
+
+  // Case 5: Both lack parameters -> SHOULD match.
+  Codec no_params_2 = CreateAudioCodec(103, kRedCodecName, 48000, 2);
+  EXPECT_TRUE(MatchesWithReferenceAttributes(no_params, no_params_2));
+
+  // A video RED codec should not match any audio RED codec,
+  // independent of parameters.
+  Codec video_red_codec = CreateVideoCodec(107, kRedCodecName);
+  EXPECT_FALSE(MatchesWithReferenceAttributes(no_params, video_red_codec));
+  EXPECT_FALSE(MatchesWithReferenceAttributes(with_params, video_red_codec));
+
+  // Two video RED codecs with different IDs and one lacking parameters
+  // should NOT match.
+  Codec video_red_with_params = CreateVideoCodec(108, kRedCodecName);
+  video_red_with_params.SetParam(kCodecParamNotInNameValueFormat, "120/120");
+  Codec video_red_no_params = CreateVideoCodec(109, kRedCodecName);
+  EXPECT_FALSE(MatchesWithReferenceAttributes(video_red_with_params,
+                                              video_red_no_params));
+  EXPECT_FALSE(MatchesWithReferenceAttributes(video_red_no_params,
+                                              video_red_with_params));
+}
+
 struct TestParams {
   std::string name;
   SdpVideoFormat codec1;
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 9863aa6da6..4315b46bb2 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -407,6 +407,7 @@ rtc_library("codec_vendor") {
     ":typed_codec_vendor",
     ":used_ids",
     "../api:field_trials_view",
+    "../api:payload_type",
     "../api:rtc_error",
     "../api:rtp_parameters",
     "../api:rtp_transceiver_direction",
diff --git a/pc/codec_vendor.cc b/pc/codec_vendor.cc
index 1d865fd7de..50f3ef818f 100644
--- a/pc/codec_vendor.cc
+++ b/pc/codec_vendor.cc
@@ -23,6 +23,7 @@
 #include "absl/strings/string_view.h"
 #include "api/field_trials_view.h"
 #include "api/media_types.h"
+#include "api/payload_type.h"
 #include "api/rtc_error.h"
 #include "api/rtp_parameters.h"
 #include "api/rtp_transceiver_direction.h"
@@ -620,7 +621,8 @@ RTCErrorOr<std::vector<Codec>> CodecVendor::GetNegotiatedCodecsForOffer(
   CodecList codecs;
   std::string mid = media_description_options.mid;
   // If current content exists and is not being recycled, use its codecs.
-  if (current_content && current_content->mid() == mid) {
+  if (current_content && current_content->mid() == mid &&
+      IsMediaContentOfType(current_content, media_description_options.type)) {
     RTCErrorOr<CodecList> checked_codec_list =
         CodecList::Create(current_content->media_description()->codecs());
     if (!checked_codec_list.ok()) {
@@ -758,7 +760,8 @@ RTCErrorOr<Codecs> CodecVendor::GetNegotiatedCodecsForAnswer(
   RTC_LOG_THREAD_BLOCK_COUNT();
   CodecList codecs;
   std::string mid = media_description_options.mid;
-  if (current_content && current_content->mid() == mid) {
+  if (current_content && current_content->mid() == mid &&
+      IsMediaContentOfType(current_content, media_description_options.type)) {
     RTCErrorOr<CodecList> checked_codec_list =
         CodecList::Create(current_content->media_description()->codecs());
     if (!checked_codec_list.ok()) {
diff --git a/pc/peer_connection_encodings_integrationtest.cc b/pc/peer_connection_encodings_integrationtest.cc
index 6b84d6bf93..9b9ee258c1 100644
--- a/pc/peer_connection_encodings_integrationtest.cc
+++ b/pc/peer_connection_encodings_integrationtest.cc
@@ -2129,6 +2129,64 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
   EXPECT_EQ(parameters.codecs[0].name, opus->name);
 }
 
+TEST_F(PeerConnectionEncodingsIntegrationTest,
+       EncodingParametersRedEnabledBeforeNegotiationAudioWithFieldTrial) {
+  scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper =
+      CreatePc("WebRTC-PayloadTypesInTransport/Enabled/");
+  scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper =
+      CreatePc("WebRTC-PayloadTypesInTransport/Enabled/");
+  ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
+
+  std::vector<RtpCodecCapability> send_codecs =
+      local_pc_wrapper->pc_factory()
+          ->GetRtpSenderCapabilities(MediaType::AUDIO)
+          .codecs;
+
+  std::optional<RtpCodecCapability> opus =
+      local_pc_wrapper->FindFirstSendCodecWithName(MediaType::AUDIO, "opus");
+  ASSERT_TRUE(opus);
+
+  std::optional<RtpCodecCapability> red =
+      local_pc_wrapper->FindFirstSendCodecWithName(MediaType::AUDIO, "red");
+  ASSERT_TRUE(red);
+
+  RtpTransceiverInit init;
+  init.direction = RtpTransceiverDirection::kSendOnly;
+  RtpEncodingParameters encoding_parameters;
+  encoding_parameters.codec = opus;
+  init.send_encodings.push_back(encoding_parameters);
+
+  auto transceiver_or_error =
+      local_pc_wrapper->pc()->AddTransceiver(MediaType::AUDIO, init);
+  ASSERT_TRUE(transceiver_or_error.ok());
+  scoped_refptr<RtpTransceiverInterface> audio_transceiver =
+      transceiver_or_error.MoveValue();
+
+  // Preferring RED over Opus should enable RED with Opus encoding.
+  send_codecs[0] = red.value();
+  send_codecs[1] = opus.value();
+
+  ASSERT_TRUE(audio_transceiver->SetCodecPreferences(send_codecs).ok());
+  NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper);
+  local_pc_wrapper->WaitForConnection();
+  remote_pc_wrapper->WaitForConnection();
+
+  RtpParameters parameters = audio_transceiver->sender()->GetParameters();
+  EXPECT_EQ(parameters.encodings[0].codec, opus);
+  EXPECT_EQ(parameters.codecs[0].name, red->name);
+
+  // Check that it's possible to switch back to Opus without RED.
+  send_codecs[0] = opus.value();
+  send_codecs[1] = red.value();
+
+  ASSERT_TRUE(audio_transceiver->SetCodecPreferences(send_codecs).ok());
+  NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper);
+
+  parameters = audio_transceiver->sender()->GetParameters();
+  EXPECT_EQ(parameters.encodings[0].codec, opus);
+  EXPECT_EQ(parameters.codecs[0].name, opus->name);
+}
+
 TEST_F(PeerConnectionEncodingsIntegrationTest,
        SetParametersRejectsScalabilityModeForSelectedCodec) {
   scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
diff --git a/pc/peer_connection_media_unittest.cc b/pc/peer_connection_media_unittest.cc
index 3890be2dc3..a4671c9906 100644
--- a/pc/peer_connection_media_unittest.cc
+++ b/pc/peer_connection_media_unittest.cc
@@ -1376,10 +1376,10 @@ TEST_P(PeerConnectionMediaTest, RedFmtpPayloadTypeMustMatchBaseCodecs) {
       CreateAudioCodec(120, "foo", kDefaultAudioClockRateHz, 1));
   callee_fake_codecs.push_back(
       CreateAudioCodec(121, kRedCodecName, kDefaultAudioClockRateHz, 1));
-  callee_fake_codecs.push_back(
-      CreateAudioCodec(122, "bar", kDefaultAudioClockRateHz, 1));
   callee_fake_codecs.back().SetParam(kCodecParamNotInNameValueFormat,
                                      "122/122");
+  callee_fake_codecs.push_back(
+      CreateAudioCodec(122, "bar", kDefaultAudioClockRateHz, 1));
   auto callee_fake_engine = std::make_unique<FakeMediaEngine>();
   callee_fake_engine->SetAudioCodecs(callee_fake_codecs);
   auto callee = CreatePeerConnectionWithAudio(std::move(callee_fake_engine));

From 63cadb1c4ff5a5b10836ffa694e9269548072321 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Bj=C3=B6rn=20Terelius?= <terelius@webrtc.org>
Date: Thu, 12 Mar 2026 13:13:27 +0100
Subject: [PATCH 281/472] Remove unsupported python event log analyzer
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

The tool doesn't support current event log format. Use
rtc_tools/rtc_event_log_visualizer or rtc_tools/rtc_event_log_to_text
instead.

Bug: None
Change-Id: Ieca3ba581a102f7a28a5480b45c53bad2622d606
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456180
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#47134}
---
 PRESUBMIT.py                                  |   5 +-
 rtc_tools/BUILD.gn                            |  18 +-
 rtc_tools/py_event_log_analyzer/README.md     |  26 --
 rtc_tools/py_event_log_analyzer/misc.py       |  85 -----
 rtc_tools/py_event_log_analyzer/misc_test.py  |  74 ----
 rtc_tools/py_event_log_analyzer/pb_parse.py   |  51 ---
 .../py_event_log_analyzer/rtp_analyzer.py     | 340 ------------------
 .../py_event_log_analyzer/rtp_analyzer.sh     |  10 -
 .../rtp_analyzer_test.py                      |  62 ----
 9 files changed, 2 insertions(+), 669 deletions(-)
 delete mode 100644 rtc_tools/py_event_log_analyzer/README.md
 delete mode 100644 rtc_tools/py_event_log_analyzer/misc.py
 delete mode 100755 rtc_tools/py_event_log_analyzer/misc_test.py
 delete mode 100644 rtc_tools/py_event_log_analyzer/pb_parse.py
 delete mode 100644 rtc_tools/py_event_log_analyzer/rtp_analyzer.py
 delete mode 100755 rtc_tools/py_event_log_analyzer/rtp_analyzer.sh
 delete mode 100755 rtc_tools/py_event_log_analyzer/rtp_analyzer_test.py

diff --git a/PRESUBMIT.py b/PRESUBMIT.py
index cfb9220c3d..60a78a5e4b 100755
--- a/PRESUBMIT.py
+++ b/PRESUBMIT.py
@@ -832,10 +832,7 @@ def RunPythonTests(input_api, output_api):
         'process_perf_results_test.py',
     ]
 
-    test_directories = [
-        input_api.PresubmitLocalPath(),
-        Join('rtc_tools', 'py_event_log_analyzer'),
-    ] + [
+    test_directories = [input_api.PresubmitLocalPath()] + [
         root for root, _, files in os.walk(Join('tools_webrtc')) if any(
             f.endswith('_test.py') and f not in excluded_files for f in files)
     ]
diff --git a/rtc_tools/BUILD.gn b/rtc_tools/BUILD.gn
index 3b2b6c0c49..b964e43859 100644
--- a/rtc_tools/BUILD.gn
+++ b/rtc_tools/BUILD.gn
@@ -29,10 +29,7 @@ group("rtc_tools") {
     deps += [ ":tools_unittests" ]
   }
   if (rtc_include_tests && rtc_enable_protobuf) {
-    deps += [
-      ":rtp_analyzer",
-      "network_tester",
-    ]
+    deps += [ "network_tester" ]
   }
   if (rtc_include_tests && rtc_enable_protobuf && !build_with_chromium) {
     deps += [
@@ -623,17 +620,4 @@ if (rtc_include_tests) {
       }  # unpack_aecdump
     }
   }
-
-  if (rtc_enable_protobuf) {
-    copy("rtp_analyzer") {
-      sources = [
-        "py_event_log_analyzer/misc.py",
-        "py_event_log_analyzer/pb_parse.py",
-        "py_event_log_analyzer/rtp_analyzer.py",
-        "py_event_log_analyzer/rtp_analyzer.sh",
-      ]
-      outputs = [ "$root_build_dir/{{source_file_part}}" ]
-      deps = [ "../logging:rtc_event_log_proto" ]
-    }  # rtp_analyzer
-  }
 }
diff --git a/rtc_tools/py_event_log_analyzer/README.md b/rtc_tools/py_event_log_analyzer/README.md
deleted file mode 100644
index 12f436c3d7..0000000000
--- a/rtc_tools/py_event_log_analyzer/README.md
+++ /dev/null
@@ -1,26 +0,0 @@
-# RTP log analyzer
-This file describes how to set up and use the RTP log analyzer.
-
-## Build
-
-```shell
-$ autoninja -C out/my_build webrtc:rtp_analyzer
-```
-
-## Usage
-
-```shell
-./out/my_build/rtp_analyzer.sh [options] /path/to/rtc_event.log
-```
-
-where `/path/to/rtc_event.log` is a recorded RTC event log, which is stored in
-protobuf format. Such logs are generated in multiple ways, e.g. by Chrome
-through the chrome://webrtc-internals page.
-
-Use `--help` for the options.
-
-The script requires Python (2.7 or 3+) and it has the following dependencies:
-Dependencies (available on pip):
-- matplotlib (http://matplotlib.org/)
-- numpy (http://www.numpy.org/)
-- protobuf (https://pypi.org/project/protobuf/)
diff --git a/rtc_tools/py_event_log_analyzer/misc.py b/rtc_tools/py_event_log_analyzer/misc.py
deleted file mode 100644
index c21f0c466b..0000000000
--- a/rtc_tools/py_event_log_analyzer/misc.py
+++ /dev/null
@@ -1,85 +0,0 @@
-#  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
-#
-#  Use of this source code is governed by a BSD-style license
-#  that can be found in the LICENSE file in the root of the source
-#  tree. An additional intellectual property rights grant can be found
-#  in the file PATENTS.  All contributing project authors may
-#  be found in the AUTHORS file in the root of the source tree.
-"""Utility functions for calculating statistics.
-"""
-
-from __future__ import division
-import collections
-import sys
-
-
-def CountReordered(sequence_numbers):
-    """Returns number of reordered indices.
-
-  A reordered index is an index `i` for which sequence_numbers[i] >=
-  sequence_numbers[i + 1]
-  """
-    return sum(1 for (s1, s2) in zip(sequence_numbers, sequence_numbers[1:])
-               if s1 >= s2)
-
-
-def SsrcNormalizedSizeTable(data_points):
-    """Counts proportion of data for every SSRC.
-
-  Args:
-     data_points: list of pb_parse.DataPoint
-
-  Returns:
-     A dictionary mapping from every SSRC in the data points. The
-     value of an SSRC `s` is the proportion of sizes of packets with
-     SSRC `s` to the total size of all packets.
-
-  """
-    mapping = collections.defaultdict(int)
-    for point in data_points:
-        mapping[point.ssrc] += point.size
-    return NormalizeCounter(mapping)
-
-
-def NormalizeCounter(counter):
-    """Returns a normalized version of the dictionary `counter`.
-
-  Does not modify `counter`.
-
-  Returns:
-    A new dictionary, in which every value in `counter`
-    has been divided by the total to sum up to 1.
-  """
-    total = sum(counter.values())
-    return {key: counter[key] / total for key in counter}
-
-
-def Unwrap(data, mod):
-    """Returns `data` unwrapped modulo `mod`. Does not modify data.
-
-  Adds integer multiples of mod to all elements of data except the
-  first, such that all pairs of consecutive elements (a, b) satisfy
-  -mod / 2 <= b - a < mod / 2.
-
-  E.g. Unwrap([0, 1, 2, 0, 1, 2, 7, 8], 3) -> [0, 1, 2, 3,
-  4, 5, 4, 5]
-  """
-    lst = data[:]
-    for i in range(1, len(data)):
-        lst[i] = lst[i - 1] + (lst[i] - lst[i - 1] + mod // 2) % mod - (mod //
-                                                                        2)
-    return lst
-
-
-def SsrcDirections(data_points):
-    ssrc_is_incoming = {}
-    for point in data_points:
-        ssrc_is_incoming[point.ssrc] = point.incoming
-    return ssrc_is_incoming
-
-
-# Python 2/3-compatible input function
-if sys.version_info[0] <= 2:
-    get_input = raw_input  # pylint: disable=invalid-name
-else:
-    get_input = input  # pylint: disable=invalid-name
diff --git a/rtc_tools/py_event_log_analyzer/misc_test.py b/rtc_tools/py_event_log_analyzer/misc_test.py
deleted file mode 100755
index 1ee17419a2..0000000000
--- a/rtc_tools/py_event_log_analyzer/misc_test.py
+++ /dev/null
@@ -1,74 +0,0 @@
-#!/usr/bin/env python3
-#  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
-#
-#  Use of this source code is governed by a BSD-style license
-#  that can be found in the LICENSE file in the root of the source
-#  tree. An additional intellectual property rights grant can be found
-#  in the file PATENTS.  All contributing project authors may
-#  be found in the AUTHORS file in the root of the source tree.
-"""Run the tests with
-
-      python misc_test.py
-or
-      python3 misc_test.py
-"""
-
-from __future__ import division
-from __future__ import absolute_import
-import random
-import unittest
-from six.moves import range
-from six.moves import zip
-
-import misc
-
-class TestMisc(unittest.TestCase):
-  def testUnwrapMod3(self):
-    data = [0, 1, 2, 0, -1, -2, -3, -4]
-    unwrapped_3 = misc.Unwrap(data, 3)
-    self.assertEqual([0, 1, 2, 3, 2, 1, 0, -1], unwrapped_3)
-
-  def testUnwrapMod4(self):
-    data = [0, 1, 2, 0, -1, -2, -3, -4]
-    unwrapped_4 = misc.Unwrap(data, 4)
-    self.assertEqual([0, 1, 2, 0, -1, -2, -3, -4], unwrapped_4)
-
-  def testDataShouldNotChangeAfterUnwrap(self):
-    data = [0, 1, 2, 0, -1, -2, -3, -4]
-    _ = misc.Unwrap(data, 4)
-
-    self.assertEqual([0, 1, 2, 0, -1, -2, -3, -4], data)
-
-  def testRandomlyMultiplesOfModAdded(self):
-    # `unwrap` definition says only multiples of mod are added.
-    random_data = [random.randint(0, 9) for _ in range(100)]
-
-    for mod in range(1, 100):
-      random_data_unwrapped_mod = misc.Unwrap(random_data, mod)
-
-      for (old_a, a) in zip(random_data, random_data_unwrapped_mod):
-        self.assertEqual((old_a - a) % mod, 0)
-
-  def testRandomlyAgainstInequalityDefinition(self):
-    # Data has to satisfy -mod/2 <= difference < mod/2 for every
-    # difference between consecutive values after unwrap.
-    random_data = [random.randint(0, 9) for _ in range(100)]
-
-    for mod in range(1, 100):
-      random_data_unwrapped_mod = misc.Unwrap(random_data, mod)
-
-      for (a, b) in zip(random_data_unwrapped_mod,
-                        random_data_unwrapped_mod[1:]):
-        self.assertTrue(-mod / 2 <= b - a < mod / 2)
-
-  def testRandomlyDataShouldNotChangeAfterUnwrap(self):
-    random_data = [random.randint(0, 9) for _ in range(100)]
-    random_data_copy = random_data[:]
-    for mod in range(1, 100):
-      _ = misc.Unwrap(random_data, mod)
-
-      self.assertEqual(random_data, random_data_copy)
-
-
-if __name__ == "__main__":
-  unittest.main()
diff --git a/rtc_tools/py_event_log_analyzer/pb_parse.py b/rtc_tools/py_event_log_analyzer/pb_parse.py
deleted file mode 100644
index 23e6ae4487..0000000000
--- a/rtc_tools/py_event_log_analyzer/pb_parse.py
+++ /dev/null
@@ -1,51 +0,0 @@
-#  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
-#
-#  Use of this source code is governed by a BSD-style license
-#  that can be found in the LICENSE file in the root of the source
-#  tree. An additional intellectual property rights grant can be found
-#  in the file PATENTS.  All contributing project authors may
-#  be found in the AUTHORS file in the root of the source tree.
-"""Parses protobuf RTC dumps."""
-
-from __future__ import division
-import struct
-import pyproto.logging.rtc_event_log.rtc_event_log_pb2 as rtc_pb
-
-
-class DataPoint(object):
-    """Simple container class for RTP events."""
-
-    def __init__(self, rtp_header_str, packet_size, arrival_timestamp_us,
-                 incoming):
-        """Builds a data point by parsing an RTP header, size and arrival time.
-
-    RTP header structure is defined in RFC 3550 section 5.1.
-    """
-        self.size = packet_size
-        self.arrival_timestamp_ms = arrival_timestamp_us / 1000
-        self.incoming = incoming
-        header = struct.unpack_from("!HHII", rtp_header_str, 0)
-        (first2header_bytes, self.sequence_number, self.timestamp,
-         self.ssrc) = header
-        self.payload_type = first2header_bytes & 0b01111111
-        self.marker_bit = (first2header_bytes & 0b10000000) >> 7
-
-
-def ParseProtobuf(file_path):
-    """Parses RTC event log from protobuf file.
-
-  Args:
-       file_path: path to protobuf file of RTC event stream
-
-  Returns:
-    all RTP packet events from the event stream as a list of DataPoints
-  """
-    event_stream = rtc_pb.EventStream()
-    with open(file_path, "rb") as f:
-        event_stream.ParseFromString(f.read())
-
-    return [
-        DataPoint(event.rtp_packet.header, event.rtp_packet.packet_length,
-                  event.timestamp_us, event.rtp_packet.incoming)
-        for event in event_stream.stream if event.HasField("rtp_packet")
-    ]
diff --git a/rtc_tools/py_event_log_analyzer/rtp_analyzer.py b/rtc_tools/py_event_log_analyzer/rtp_analyzer.py
deleted file mode 100644
index 5844997877..0000000000
--- a/rtc_tools/py_event_log_analyzer/rtp_analyzer.py
+++ /dev/null
@@ -1,340 +0,0 @@
-#!/usr/bin/env python3
-#  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
-#
-#  Use of this source code is governed by a BSD-style license
-#  that can be found in the LICENSE file in the root of the source
-#  tree. An additional intellectual property rights grant can be found
-#  in the file PATENTS.  All contributing project authors may
-#  be found in the AUTHORS file in the root of the source tree.
-"""Displays statistics and plots graphs from RTC protobuf dump."""
-
-from __future__ import division
-from __future__ import print_function
-
-from __future__ import absolute_import
-import collections
-import optparse
-import os
-import sys
-from six.moves import range
-from six.moves import zip
-
-import matplotlib.pyplot as plt
-import numpy
-
-import misc
-import pb_parse
-
-
-class RTPStatistics:
-  """Has methods for calculating and plotting RTP stream statistics."""
-
-  BANDWIDTH_SMOOTHING_WINDOW_SIZE = 10
-  PLOT_RESOLUTION_MS = 50
-
-  def __init__(self, data_points):
-    """Initializes object with data_points and computes simple statistics.
-
-    Computes percentages of number of packets and packet sizes by
-    SSRC.
-
-    Args:
-        data_points: list of pb_parse.DataPoints on which statistics are
-            calculated.
-
-    """
-
-    self.data_points = data_points
-    self.ssrc_frequencies = misc.NormalizeCounter(
-        collections.Counter([pt.ssrc for pt in self.data_points]))
-    self.ssrc_size_table = misc.SsrcNormalizedSizeTable(self.data_points)
-    self.bandwidth_kbps = None
-    self.smooth_bw_kbps = None
-
-  def PrintHeaderStatistics(self):
-    print("{:>6}{:>14}{:>14}{:>6}{:>6}{:>3}{:>11}".format(
-        "SeqNo", "TimeStamp", "SendTime", "Size", "PT", "M", "SSRC"))
-    for point in self.data_points:
-      print("{:>6}{:>14}{:>14}{:>6}{:>6}{:>3}{:>11}".format(
-          point.sequence_number, point.timestamp,
-          int(point.arrival_timestamp_ms), point.size, point.payload_type,
-          point.marker_bit, "0x{:x}".format(point.ssrc)))
-
-  def PrintSsrcInfo(self, ssrc_id, ssrc):
-    """Prints packet and size statistics for a given SSRC.
-
-    Args:
-        ssrc_id: textual identifier of SSRC printed beside statistics for it.
-        ssrc: SSRC by which to filter data and display statistics
-    """
-    filtered_ssrc = [point for point in self.data_points if point.ssrc == ssrc]
-    payloads = misc.NormalizeCounter(
-        collections.Counter([point.payload_type for point in filtered_ssrc]))
-
-    payload_info = "payload type(s): {}".format(", ".join(
-        str(payload) for payload in payloads))
-    print("{} 0x{:x} {}, {:.2f}% packets, {:.2f}% data".format(
-        ssrc_id, ssrc, payload_info, self.ssrc_frequencies[ssrc] * 100,
-        self.ssrc_size_table[ssrc] * 100))
-    print("  packet sizes:")
-    (bin_counts,
-     bin_bounds) = numpy.histogram([point.size for point in filtered_ssrc],
-                                   bins=5,
-                                   density=False)
-    bin_proportions = bin_counts / sum(bin_counts)
-    print("\n".join([
-        " {:.1f} - {:.1f}: {:.2f}%".format(bin_bounds[i], bin_bounds[i + 1],
-                                           bin_proportions[i] * 100)
-        for i in range(len(bin_proportions))
-    ]))
-
-  def ChooseSsrc(self):
-    """Queries user for SSRC."""
-
-    if len(self.ssrc_frequencies) == 1:
-      chosen_ssrc = list(self.ssrc_frequencies.keys())[0]
-      self.PrintSsrcInfo("", chosen_ssrc)
-      return chosen_ssrc
-
-    ssrc_is_incoming = misc.SsrcDirections(self.data_points)
-    incoming = [ssrc for ssrc in ssrc_is_incoming if ssrc_is_incoming[ssrc]]
-    outgoing = [ssrc for ssrc in ssrc_is_incoming if not ssrc_is_incoming[ssrc]]
-
-    print("\nIncoming:\n")
-    for (i, ssrc) in enumerate(incoming):
-      self.PrintSsrcInfo(i, ssrc)
-
-    print("\nOutgoing:\n")
-    for (i, ssrc) in enumerate(outgoing):
-      self.PrintSsrcInfo(i + len(incoming), ssrc)
-
-    while True:
-      chosen_index = int(misc.get_input("choose one> "))
-      if 0 <= chosen_index < len(self.ssrc_frequencies):
-        return (incoming + outgoing)[chosen_index]
-      print("Invalid index!")
-
-  def FilterSsrc(self, chosen_ssrc):
-    """Filters and wraps data points.
-
-    Removes data points with `ssrc != chosen_ssrc`. Unwraps sequence
-    numbers and timestamps for the chosen selection.
-    """
-    self.data_points = [
-        point for point in self.data_points if point.ssrc == chosen_ssrc
-    ]
-    unwrapped_sequence_numbers = misc.Unwrap(
-        [point.sequence_number for point in self.data_points], 2**16 - 1)
-    for (data_point, sequence_number) in zip(self.data_points,
-                                             unwrapped_sequence_numbers):
-      data_point.sequence_number = sequence_number
-
-    unwrapped_timestamps = misc.Unwrap(
-        [point.timestamp for point in self.data_points], 2**32 - 1)
-
-    for (data_point, timestamp) in zip(self.data_points, unwrapped_timestamps):
-      data_point.timestamp = timestamp
-
-  def PrintSequenceNumberStatistics(self):
-    seq_no_set = set(point.sequence_number for point in self.data_points)
-    missing_sequence_numbers = max(seq_no_set) - min(seq_no_set) + (
-        1 - len(seq_no_set))
-    print("Missing sequence numbers: {} out of {}  ({:.2f}%)".format(
-        missing_sequence_numbers, len(seq_no_set),
-        100 * missing_sequence_numbers / len(seq_no_set)))
-    print("Duplicated packets: {}".format(
-        len(self.data_points) - len(seq_no_set)))
-    print("Reordered packets: {}".format(
-        misc.CountReordered(
-            [point.sequence_number for point in self.data_points])))
-
-  def EstimateFrequency(self, always_query_sample_rate):
-    """Estimates frequency and updates data.
-
-    Guesses the most probable frequency by looking at changes in
-    timestamps (RFC 3550 section 5.1), calculates clock drifts and
-    sending time of packets. Updates `self.data_points` with changes
-    in delay and send time.
-    """
-    delta_timestamp = (self.data_points[-1].timestamp -
-                       self.data_points[0].timestamp)
-    delta_arr_timestamp = float((self.data_points[-1].arrival_timestamp_ms -
-                                 self.data_points[0].arrival_timestamp_ms))
-    freq_est = delta_timestamp / delta_arr_timestamp
-
-    freq_vec = [8, 16, 32, 48, 90]
-    freq = None
-    for f in freq_vec:
-      if abs((freq_est - f) / f) < 0.05:
-        freq = f
-
-    print("Estimated frequency: {:.3f}kHz".format(freq_est))
-    if freq is None or always_query_sample_rate:
-      if not always_query_sample_rate:
-        print("Frequency could not be guessed.", end=" ")
-      freq = int(misc.get_input("Input frequency (in kHz)> "))
-    else:
-      print("Guessed frequency: {}kHz".format(freq))
-
-    for point in self.data_points:
-      point.real_send_time_ms = (point.timestamp -
-                                 self.data_points[0].timestamp) / freq
-      point.delay = point.arrival_timestamp_ms - point.real_send_time_ms
-
-  def PrintDurationStatistics(self):
-    """Prints delay, clock drift and bitrate statistics."""
-
-    min_delay = min(point.delay for point in self.data_points)
-
-    for point in self.data_points:
-      point.absdelay = point.delay - min_delay
-
-    stream_duration_sender = self.data_points[-1].real_send_time_ms / 1000
-    print("Stream duration at sender: {:.1f} seconds".format(
-        stream_duration_sender))
-
-    arrival_timestamps_ms = [
-        point.arrival_timestamp_ms for point in self.data_points
-    ]
-    stream_duration_receiver = (max(arrival_timestamps_ms) -
-                                min(arrival_timestamps_ms)) / 1000
-    print("Stream duration at receiver: {:.1f} seconds".format(
-        stream_duration_receiver))
-
-    print("Clock drift: {:.2f}%".format(
-        100 * (stream_duration_receiver / stream_duration_sender - 1)))
-
-    total_size = sum(point.size for point in self.data_points) * 8 / 1000
-    print("Send average bitrate: {:.2f} kbps".format(total_size /
-                                                     stream_duration_sender))
-
-    print("Receive average bitrate: {:.2f} kbps".format(
-        total_size / stream_duration_receiver))
-
-  def RemoveReordered(self):
-    last = self.data_points[0]
-    data_points_ordered = [last]
-    for point in self.data_points[1:]:
-      if point.sequence_number > last.sequence_number and (
-          point.real_send_time_ms > last.real_send_time_ms):
-        data_points_ordered.append(point)
-        last = point
-    self.data_points = data_points_ordered
-
-  def ComputeBandwidth(self):
-    """Computes bandwidth averaged over several consecutive packets.
-
-    The number of consecutive packets used in the average is
-    BANDWIDTH_SMOOTHING_WINDOW_SIZE. Averaging is done with
-    numpy.correlate.
-    """
-    start_ms = self.data_points[0].real_send_time_ms
-    stop_ms = self.data_points[-1].real_send_time_ms
-    (self.bandwidth_kbps, _) = numpy.histogram(
-        [point.real_send_time_ms for point in self.data_points],
-        bins=numpy.arange(start_ms, stop_ms, RTPStatistics.PLOT_RESOLUTION_MS),
-        weights=[
-            point.size * 8 / RTPStatistics.PLOT_RESOLUTION_MS
-            for point in self.data_points
-        ])
-    correlate_filter = (
-        numpy.ones(RTPStatistics.BANDWIDTH_SMOOTHING_WINDOW_SIZE) /
-        RTPStatistics.BANDWIDTH_SMOOTHING_WINDOW_SIZE)
-    self.smooth_bw_kbps = numpy.correlate(self.bandwidth_kbps, correlate_filter)
-
-  def PlotStatistics(self):
-    """Plots changes in delay and average bandwidth."""
-
-    start_ms = self.data_points[0].real_send_time_ms
-    stop_ms = self.data_points[-1].real_send_time_ms
-    time_axis = numpy.arange(start_ms / 1000, stop_ms / 1000,
-                             RTPStatistics.PLOT_RESOLUTION_MS / 1000)
-
-    delay = CalculateDelay(start_ms, stop_ms, RTPStatistics.PLOT_RESOLUTION_MS,
-                           self.data_points)
-
-    plt.figure(1)
-    plt.plot(time_axis, delay[:len(time_axis)])
-    plt.xlabel("Send time [s]")
-    plt.ylabel("Relative transport delay [ms]")
-
-    plt.figure(2)
-    plt.plot(time_axis[:len(self.smooth_bw_kbps)], self.smooth_bw_kbps)
-    plt.xlabel("Send time [s]")
-    plt.ylabel("Bandwidth [kbps]")
-
-    plt.show()
-
-
-def CalculateDelay(start, stop, step, points):
-  """Quantizes the time coordinates for the delay.
-
-  Quantizes points by rounding the timestamps downwards to the nearest
-  point in the time sequence start, start+step, start+2*step... Takes
-  the average of the delays of points rounded to the same. Returns
-  masked array, in which time points with no value are masked.
-
-  """
-  grouped_delays = [[] for _ in numpy.arange(start, stop + step, step)]
-  rounded_value_index = lambda x: int((x - start) / step)
-  for point in points:
-    grouped_delays[rounded_value_index(point.real_send_time_ms)].append(
-        point.absdelay)
-  regularized_delays = [
-      numpy.average(arr) if arr else -1 for arr in grouped_delays
-  ]
-  return numpy.ma.masked_values(regularized_delays, -1)
-
-
-def main():
-  usage = "Usage: %prog [options] <filename of rtc event log>"
-  parser = optparse.OptionParser(usage=usage)
-  parser.add_option("--dump_header_to_stdout",
-                    default=False,
-                    action="store_true",
-                    help="print header info to stdout; similar to rtp_analyze")
-  parser.add_option("--query_sample_rate",
-                    default=False,
-                    action="store_true",
-                    help="always query user for real sample rate")
-
-  parser.add_option("--working_directory",
-                    default=None,
-                    action="store",
-                    help="directory in which to search for relative paths")
-
-  (options, args) = parser.parse_args()
-
-  if len(args) < 1:
-    parser.print_help()
-    sys.exit(0)
-
-  input_file = args[0]
-
-  if options.working_directory and not os.path.isabs(input_file):
-    input_file = os.path.join(options.working_directory, input_file)
-
-  data_points = pb_parse.ParseProtobuf(input_file)
-  rtp_stats = RTPStatistics(data_points)
-
-  if options.dump_header_to_stdout:
-    print("Printing header info to stdout.", file=sys.stderr)
-    rtp_stats.PrintHeaderStatistics()
-    sys.exit(0)
-
-  chosen_ssrc = rtp_stats.ChooseSsrc()
-  print("Chosen SSRC: 0X{:X}".format(chosen_ssrc))
-
-  rtp_stats.FilterSsrc(chosen_ssrc)
-
-  print("Statistics:")
-  rtp_stats.PrintSequenceNumberStatistics()
-  rtp_stats.EstimateFrequency(options.query_sample_rate)
-  rtp_stats.PrintDurationStatistics()
-  rtp_stats.RemoveReordered()
-  rtp_stats.ComputeBandwidth()
-  rtp_stats.PlotStatistics()
-
-
-if __name__ == "__main__":
-  main()
diff --git a/rtc_tools/py_event_log_analyzer/rtp_analyzer.sh b/rtc_tools/py_event_log_analyzer/rtp_analyzer.sh
deleted file mode 100755
index 7467e493d2..0000000000
--- a/rtc_tools/py_event_log_analyzer/rtp_analyzer.sh
+++ /dev/null
@@ -1,10 +0,0 @@
-#!/bin/bash
-#  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
-#
-#  Use of this source code is governed by a BSD-style license
-#  that can be found in the LICENSE file in the root of the source
-#  tree. An additional intellectual property rights grant can be found
-#  in the file PATENTS.  All contributing project authors may
-#  be found in the AUTHORS file in the root of the source tree.
-BASE_DIR=`dirname $0`
-python "${BASE_DIR}/rtp_analyzer.py" $@ --working_dir $BASE_DIR
diff --git a/rtc_tools/py_event_log_analyzer/rtp_analyzer_test.py b/rtc_tools/py_event_log_analyzer/rtp_analyzer_test.py
deleted file mode 100755
index a078c3098f..0000000000
--- a/rtc_tools/py_event_log_analyzer/rtp_analyzer_test.py
+++ /dev/null
@@ -1,62 +0,0 @@
-#!/usr/bin/env python3
-#  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
-#
-#  Use of this source code is governed by a BSD-style license
-#  that can be found in the LICENSE file in the root of the source
-#  tree. An additional intellectual property rights grant can be found
-#  in the file PATENTS.  All contributing project authors may
-#  be found in the AUTHORS file in the root of the source tree.
-"""Run the tests with
-
-      python rtp_analyzer_test.py
-or
-      python3 rtp_analyzer_test.py
-"""
-
-from __future__ import absolute_import
-from __future__ import print_function
-import collections
-import unittest
-
-MISSING_NUMPY = False  # pylint: disable=invalid-name
-try:
-  import numpy
-  import rtp_analyzer
-except ImportError:
-  MISSING_NUMPY = True
-
-FakePoint = collections.namedtuple("FakePoint",
-                                   ["real_send_time_ms", "absdelay"])
-
-
-class TestDelay(unittest.TestCase):
-  def AssertMaskEqual(self, masked_array, data, mask):
-    self.assertEqual(list(masked_array.data), data)
-
-    if isinstance(masked_array.mask, numpy.bool_):
-      array_mask = masked_array.mask
-    else:
-      array_mask = list(masked_array.mask)
-    self.assertEqual(array_mask, mask)
-
-  def testCalculateDelaySimple(self):
-    points = [FakePoint(0, 0), FakePoint(1, 0)]
-    mask = rtp_analyzer.CalculateDelay(0, 1, 1, points)
-    self.AssertMaskEqual(mask, [0, 0], False)
-
-  def testCalculateDelayMissing(self):
-    points = [FakePoint(0, 0), FakePoint(2, 0)]
-    mask = rtp_analyzer.CalculateDelay(0, 2, 1, points)
-    self.AssertMaskEqual(mask, [0, -1, 0], [False, True, False])
-
-  def testCalculateDelayBorders(self):
-    points = [FakePoint(0, 0), FakePoint(2, 0)]
-    mask = rtp_analyzer.CalculateDelay(0, 3, 2, points)
-    self.AssertMaskEqual(mask, [0, 0, -1], [False, False, True])
-
-
-if __name__ == "__main__":
-  if MISSING_NUMPY:
-    print("Missing numpy, skipping test.")
-  else:
-    unittest.main()

From 71cf81a4f3fe8f34b7fbcd9eaa37d212e6e16fca Mon Sep 17 00:00:00 2001
From: Mirko Bonadei <mbonadei@webrtc.org>
Date: Thu, 12 Mar 2026 07:24:43 -0700
Subject: [PATCH 282/472] Revert "Roll chromium_revision 68ba4ea02a..42535d5df1
 (1598147:1598261)"

This reverts commit 4420b583ec6fa525bad04e2b5c7189c7ecfb6b14.

Reason for revert: Rolling back Chromium rolls that broke CI but not CQ android device tests.

Original change's description:
> Roll chromium_revision 68ba4ea02a..42535d5df1 (1598147:1598261)
>
> Change log: https://chromium.googlesource.com/chromium/src/+log/68ba4ea02a..42535d5df1
> Full diff: https://chromium.googlesource.com/chromium/src/+/68ba4ea02a..42535d5df1
>
> Changed dependencies
> * src/build: https://chromium.googlesource.com/chromium/src/build/+log/08d96365cc..59b2d010a9
> * src/buildtools/linux64: git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9..git_revision:83c605e5dcc3db2945291da1760ca9abd6343338
> * src/buildtools/mac: git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9..git_revision:83c605e5dcc3db2945291da1760ca9abd6343338
> * src/buildtools/win: git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9..git_revision:83c605e5dcc3db2945291da1760ca9abd6343338
> * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/73655355dd..93aa68ef77
> * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/1df9fda10d..3aceea1e90
> * src/third_party/androidx/cipd: A6oLcxGdFmfrPoaAcVYoVYHlkB3InvLKiIKnIgmGJVcC..Mn6UxK4fOcnps2oJ1ikX7T1Nf93dtMBkX68uqIyoQ0sC
> * src/third_party/cpuinfo/src: https://chromium.googlesource.com/external/github.com/pytorch/cpuinfo.git/+log/7364b490b5..7607ca5004
> * src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/c4eb3c4f4c..2a2456c873
> * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/ee2aee8b5e..4d052614c1
> * src/third_party/r8/cipd: 6BLixpU9eIdUBP0egFPaRGM00adKPdziwUcg5ThtXFYC..AVViRP05v8_iDI-43CYldqOEnoNz2SUTr_nBxP_z-NwC
> * src/third_party/tflite/src: https://chromium.googlesource.com/external/github.com/tensorflow/tensorflow.git/+log/f920077085..d20369f4f2
> * src/third_party/xnnpack/src: https://chromium.googlesource.com/external/github.com/google/XNNPACK.git/+log/26feacc3eb..15e04ff8b6
> * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/a9a47e61f4..a06f75e8e5
> DEPS diff: https://chromium.googlesource.com/chromium/src/+/68ba4ea02a..42535d5df1/DEPS
>
> BUG=None
>
> Change-Id: Ia5cba7a6e2796ccdfa654b372fe6c348be7c42f0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456080
> Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
> Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
> Cr-Commit-Position: refs/heads/main@{#47124}

Bug: b/492115880
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I1a80e9dc4b46c49c604ad73fb0254db76fcfb471
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456260
Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47135}
---
 DEPS | 30 +++++++++++++++---------------
 1 file changed, 15 insertions(+), 15 deletions(-)

diff --git a/DEPS b/DEPS
index 6c49e57ee5..10973337aa 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '42535d5df198ffb06207d77688ee4e1b98fe1372',
+  'chromium_revision': '68ba4ea02ab80d730e9538c74a9605b99c4a35df',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@59b2d010a999e350f13f7009402f9006c17ef618',
+    'https://chromium.googlesource.com/chromium/src/build@08d96365cc4f5c0548c11d2aff921a352c095352',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@fef8d6947fa54202e8a00c58d5ac8fd15780bc33',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,19 +78,19 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@93aa68ef77a3e3ab9c018e69a20fa1874364f4fd',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@73655355dd4e6dd8b1b989049ba32740d2caf398',
     'condition': 'checkout_ios',
   },
   'src/testing':
     'https://chromium.googlesource.com/chromium/src/testing@2c01d5acaa65b2cc65a60d6b6b21cfbc247aba32',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@3aceea1e90f83044f80492efc7a1376a6c4850fa',
+    'https://chromium.googlesource.com/chromium/src/third_party@1df9fda10dcbc12b9810a0a3b481b5696236fee8',
 
   'src/buildtools/linux64': {
     'packages': [
       {
         'package': 'gn/gn/linux-${{arch}}',
-        'version': 'git_revision:83c605e5dcc3db2945291da1760ca9abd6343338',
+        'version': 'git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9',
       }
     ],
     'dep_type': 'cipd',
@@ -100,7 +100,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/mac-${{arch}}',
-        'version': 'git_revision:83c605e5dcc3db2945291da1760ca9abd6343338',
+        'version': 'git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9',
       }
     ],
     'dep_type': 'cipd',
@@ -110,7 +110,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/windows-amd64',
-        'version': 'git_revision:83c605e5dcc3db2945291da1760ca9abd6343338',
+        'version': 'git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9',
       }
     ],
     'dep_type': 'cipd',
@@ -511,7 +511,7 @@ deps = {
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@4d052614c1fdd60815db6ce6dc7d63ec391c0988',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@ee2aee8b5e104d83a973b7d9be07fdb3681626a7',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -644,7 +644,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/r8',
-              'version': 'AVViRP05v8_iDI-43CYldqOEnoNz2SUTr_nBxP_z-NwC',
+              'version': '6BLixpU9eIdUBP0egFPaRGM00adKPdziwUcg5ThtXFYC',
           },
       ],
       'condition': 'checkout_android',
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@a06f75e8e5504a8535903a2a941b2cc361cd54ea',
+    'https://chromium.googlesource.com/chromium/src/tools@a9a47e61f490f9242807eac79bb4d46c808652a9',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'Mn6UxK4fOcnps2oJ1ikX7T1Nf93dtMBkX68uqIyoQ0sC',
+          'version': 'A6oLcxGdFmfrPoaAcVYoVYHlkB3InvLKiIKnIgmGJVcC',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -786,7 +786,7 @@ deps = {
   },
 
   'src/third_party/tflite/src':
-    Var('chromium_git') + '/external/github.com/tensorflow/tensorflow.git' + '@' + 'd20369f4f226598800e0d406db567d2d1c17fa95',
+    Var('chromium_git') + '/external/github.com/tensorflow/tensorflow.git' + '@' + 'f9200770858f5721b7fa95db7aa3e4cd712fe9df',
 
   'src/third_party/turbine/cipd': {
       'packages': [
@@ -852,7 +852,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/google/pthreadpool.git' + '@' + '9003ee6c137cea3b94161bd5c614fb43be523ee1',
 
   'src/third_party/xnnpack/src':
-    Var('chromium_git') + '/external/github.com/google/XNNPACK.git' + '@' + '15e04ff8b61e0bc2118bad268447be5b2075aa66',
+    Var('chromium_git') + '/external/github.com/google/XNNPACK.git' + '@' + '26feacc3ebe30d1f3a82e1ff3938ef9859d74640',
 
   'src/third_party/farmhash/src':
     Var('chromium_git') + '/external/github.com/google/farmhash.git' + '@' + '816a4ae622e964763ca0862d9dbd19324a1eaf45',
@@ -861,10 +861,10 @@ deps = {
     Var('chromium_git') + '/external/github.com/google/ruy.git' + '@' + '2af88863614a8298689cc52b1a47b3fcad7be835',
 
   'src/third_party/cpuinfo/src':
-    Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7607ca500436b37ad23fb8d18614bec7796b68a7',
+    Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7364b490b5f78d58efe23ea76e74210fd6c3c76f',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '2a2456c873d6f4bf43a9adf919f13ad917d57c2d',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + 'c4eb3c4f4c91511eb5c76d608e07dde2a1cc8b23',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From 6d9c409eea2351a2f52ba74d380a262ebf9c4767 Mon Sep 17 00:00:00 2001
From: Mirko Bonadei <mbonadei@webrtc.org>
Date: Thu, 12 Mar 2026 07:26:13 -0700
Subject: [PATCH 283/472] Revert "Roll chromium_revision 94ef474520..68ba4ea02a
 (1598026:1598147)"

This reverts commit 5f1f4d5d8e2e6e486c390137081e65bb41f2848d.

Reason for revert: Rolling back Chromium rolls that broke CI but not CQ android device tests.

Original change's description:
> Roll chromium_revision 94ef474520..68ba4ea02a (1598026:1598147)
>
> Change log: https://chromium.googlesource.com/chromium/src/+log/94ef474520..68ba4ea02a
> Full diff: https://chromium.googlesource.com/chromium/src/+/94ef474520..68ba4ea02a
>
> Changed dependencies
> * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/c82693e275..73655355dd
> * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/ff008b8125..2c01d5acaa
> * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/6fd21700ca..1df9fda10d
> * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/a2bf3f10d5..ee2aee8b5e
> * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/08b509bdad..3a764e6f63
> * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/cf631c2a37..a9a47e61f4
> DEPS diff: https://chromium.googlesource.com/chromium/src/+/94ef474520..68ba4ea02a/DEPS
>
> BUG=None
>
> Change-Id: Ica7f73869e92fb151d248c8544a6624f3288a61f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456000
> Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
> Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
> Cr-Commit-Position: refs/heads/main@{#47122}

Bug: b/492115880
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I62b59ab7498ae199986ea1eb61885aded5d44d05
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456243
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47136}
---
 DEPS | 14 +++++++-------
 1 file changed, 7 insertions(+), 7 deletions(-)

diff --git a/DEPS b/DEPS
index 10973337aa..04bd103bbc 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '68ba4ea02ab80d730e9538c74a9605b99c4a35df',
+  'chromium_revision': '94ef474520ff0ce405215917e0915fa57414c48f',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@73655355dd4e6dd8b1b989049ba32740d2caf398',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@c82693e275c2c184e2848f462c26e29c9dc588f3',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@2c01d5acaa65b2cc65a60d6b6b21cfbc247aba32',
+    'https://chromium.googlesource.com/chromium/src/testing@ff008b8125f1295a0d0f3b90bb0c5647a827fe37',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@1df9fda10dcbc12b9810a0a3b481b5696236fee8',
+    'https://chromium.googlesource.com/chromium/src/third_party@6fd21700ca093f45029ddba35e1f38763cfa90e8',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -511,7 +511,7 @@ deps = {
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@ee2aee8b5e104d83a973b7d9be07fdb3681626a7',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@a2bf3f10d5e58d00ebee93d0f312e0fd6882380e',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '3a764e6f6339ae06c19540682dad4a1a9021f072',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '08b509bdad26c91f5f34d9ff97c9362a111f3a72',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@a9a47e61f490f9242807eac79bb4d46c808652a9',
+    'https://chromium.googlesource.com/chromium/src/tools@cf631c2a373a1f10bef7dae3d8cf4efd8dc29aba',
 
   'src/third_party/espresso': {
       'packages': [

From b32c246801a2c9a5b61619d23828e9cae6e283d7 Mon Sep 17 00:00:00 2001
From: Mirko Bonadei <mbonadei@webrtc.org>
Date: Thu, 12 Mar 2026 07:28:09 -0700
Subject: [PATCH 284/472] Revert "Roll chromium_revision 8fdfcb5d76..94ef474520
 (1597876:1598026)"

This reverts commit da6f7ffbf033849966c809d8e4a6ea67898f081f.

Reason for revert: Rolling back Chromium rolls that broke CI but not CQ android device tests.

Original change's description:
> Roll chromium_revision 8fdfcb5d76..94ef474520 (1597876:1598026)
>
> Change log: https://chromium.googlesource.com/chromium/src/+log/8fdfcb5d76..94ef474520
> Full diff: https://chromium.googlesource.com/chromium/src/+/8fdfcb5d76..94ef474520
>
> Changed dependencies
> * src/build: https://chromium.googlesource.com/chromium/src/build/+log/c9202dd8ca..08d96365cc
> * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/00efde9573..c82693e275
> * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/bacbfe72d8..ff008b8125
> * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/22a76fe23b..6fd21700ca
> * src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/c8a5676846..e1e57b8776
> * src/third_party/fuzztest/src: https://chromium.googlesource.com/external/github.com/google/fuzztest.git/+log/54ed362692..2145ac8a75
> * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/3ca09e4818..a2bf3f10d5
> * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/c0a3840504..08b509bdad
> * src/third_party/r8/cipd: CoaQUbD7pDX3OuETi_O5wkv_Y198RVIUaSobSEx9zfoC..6BLixpU9eIdUBP0egFPaRGM00adKPdziwUcg5ThtXFYC
> * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/82543c0c7f..cf631c2a37
> DEPS diff: https://chromium.googlesource.com/chromium/src/+/8fdfcb5d76..94ef474520/DEPS
>
> BUG=None
>
> Change-Id: Ib35a4e1539cdbe95349933a6993aeac9dcfb9311
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/455960
> Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
> Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
> Cr-Commit-Position: refs/heads/main@{#47121}

Bug: b/492115880
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: Id804c2333c0e3780af9aaa2ed27ba2073d7a3b5c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456244
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47137}
---
 DEPS | 22 +++++++++++-----------
 1 file changed, 11 insertions(+), 11 deletions(-)

diff --git a/DEPS b/DEPS
index 04bd103bbc..211c40da09 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '94ef474520ff0ce405215917e0915fa57414c48f',
+  'chromium_revision': '8fdfcb5d762df924ac3e77514c178343e2aeead3',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@08d96365cc4f5c0548c11d2aff921a352c095352',
+    'https://chromium.googlesource.com/chromium/src/build@c9202dd8ca1d4255bd9e505988b945746ddb378a',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@fef8d6947fa54202e8a00c58d5ac8fd15780bc33',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@c82693e275c2c184e2848f462c26e29c9dc588f3',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@00efde9573f5ecad0307619539c633c4eb805905',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@ff008b8125f1295a0d0f3b90bb0c5647a827fe37',
+    'https://chromium.googlesource.com/chromium/src/testing@bacbfe72d8a48eac176f063aefc517cd84786b91',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@6fd21700ca093f45029ddba35e1f38763cfa90e8',
+    'https://chromium.googlesource.com/chromium/src/third_party@22a76fe23bae393322386681d4bfa508b0da44ea',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -331,7 +331,7 @@ deps = {
   'src/third_party/clang-format/script':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@c2725e0622e1a86d55f14514f2177a39efea4a0e',
   'src/third_party/compiler-rt/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@e1e57b877616abd748cdd63bb78e913e26f5bdfa',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@c8a567684659fe05ede3a220965b94fd71c4a7cb',
   'src/third_party/libc++/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@7ab65651aed6802d2599dcb7a73b1f82d5179d05',
   'src/third_party/libc++abi/src':
@@ -511,7 +511,7 @@ deps = {
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@a2bf3f10d5e58d00ebee93d0f312e0fd6882380e',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@3ca09e4818f4362e75c9c224f1d4c2065db75a0a',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -593,7 +593,7 @@ deps = {
   'src/third_party/libFuzzer/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt/lib/fuzzer.git@bea408a6e01f0f7e6c82a43121fe3af4506c932e',
   'src/third_party/fuzztest/src':
-    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@2145ac8a757695d4ad68112943641dcce9221e07',
+    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@54ed362692177e93df0fa202cae404b57694e828',
   'src/third_party/libjpeg_turbo':
     'https://chromium.googlesource.com/chromium/deps/libjpeg_turbo.git@d1f5f2393e0d51f840207342ae86e55a86443288',
   'src/third_party/libsrtp':
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '08b509bdad26c91f5f34d9ff97c9362a111f3a72',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'c0a3840504ad618393057eb1fc121810c210dfd2',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -644,7 +644,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/r8',
-              'version': '6BLixpU9eIdUBP0egFPaRGM00adKPdziwUcg5ThtXFYC',
+              'version': 'CoaQUbD7pDX3OuETi_O5wkv_Y198RVIUaSobSEx9zfoC',
           },
       ],
       'condition': 'checkout_android',
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@cf631c2a373a1f10bef7dae3d8cf4efd8dc29aba',
+    'https://chromium.googlesource.com/chromium/src/tools@82543c0c7fa9a6436b46102e26e57e09aefa192d',
 
   'src/third_party/espresso': {
       'packages': [

From f784e442c742306ee8c4327081b9173f5441dfde Mon Sep 17 00:00:00 2001
From: Mirko Bonadei <mbonadei@webrtc.org>
Date: Thu, 12 Mar 2026 07:29:21 -0700
Subject: [PATCH 285/472] Revert "Roll chromium_revision b937d39273..8fdfcb5d76
 (1597657:1597876)"

This reverts commit 17a2fb58ea7db7b17bdbcf82b400bd311558a027.

Reason for revert: Rolling back Chromium rolls that broke CI but not CQ android device tests.

Original change's description:
> Roll chromium_revision b937d39273..8fdfcb5d76 (1597657:1597876)
>
> Change log: https://chromium.googlesource.com/chromium/src/+log/b937d39273..8fdfcb5d76
> Full diff: https://chromium.googlesource.com/chromium/src/+/b937d39273..8fdfcb5d76
>
> Changed dependencies
> * src/build: https://chromium.googlesource.com/chromium/src/build/+log/7d29c62736..c9202dd8ca
> * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/94f693fcb6..00efde9573
> * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/e7a1e88ecc..bacbfe72d8
> * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/a133c94b51..22a76fe23b
> * src/third_party/androidx/cipd: Hi79Lu6qQ-Fbcjwgf_f-NEYJweP4QdZqjJ0lXwc0PpcC..A6oLcxGdFmfrPoaAcVYoVYHlkB3InvLKiIKnIgmGJVcC
> * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/e0ebf38a01..a2f6285fa5
> * src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/4ce8ba39a3..a4e677522d
> * src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/4387e32481..c4eb3c4f4c
> * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/260c6fd37c..3ca09e4818
> * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/4cead1b0fc..c0a3840504
> * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/9765576870..82543c0c7f
> DEPS diff: https://chromium.googlesource.com/chromium/src/+/b937d39273..8fdfcb5d76/DEPS
>
> BUG=None
>
> Change-Id: Ica2e37fd2735bac28e9c7b69c9539b487e5666d2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/455900
> Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
> Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
> Cr-Commit-Position: refs/heads/main@{#47118}

Bug: b/492115880
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I57d717d9faaefe85efb54beb1f5387d8a4581a2f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456280
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47138}
---
 DEPS | 24 ++++++++++++------------
 1 file changed, 12 insertions(+), 12 deletions(-)

diff --git a/DEPS b/DEPS
index 211c40da09..dffcecaef3 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '8fdfcb5d762df924ac3e77514c178343e2aeead3',
+  'chromium_revision': 'b937d39273b52034e8542dea4b040587be0a127f',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@c9202dd8ca1d4255bd9e505988b945746ddb378a',
+    'https://chromium.googlesource.com/chromium/src/build@7d29c62736304aeaac07df0b75546ffd2b75917a',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@fef8d6947fa54202e8a00c58d5ac8fd15780bc33',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@00efde9573f5ecad0307619539c633c4eb805905',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@94f693fcb6dde967f11668a9e21eec1a4ec8b49c',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@bacbfe72d8a48eac176f063aefc517cd84786b91',
+    'https://chromium.googlesource.com/chromium/src/testing@e7a1e88ecc35a70f3d198674042753ec9e8e3650',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@22a76fe23bae393322386681d4bfa508b0da44ea',
+    'https://chromium.googlesource.com/chromium/src/third_party@a133c94b51ac034d5f375cc1746e66dcdee57282',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -492,7 +492,7 @@ deps = {
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@8be0e3114685fcc1589561067282edf75ea1259a',
   'src/third_party/catapult':
-    'https://chromium.googlesource.com/catapult.git@a2f6285fa55a02491af892e15ac308fd8315d754',
+    'https://chromium.googlesource.com/catapult.git@e0ebf38a01214aba11f31daa1c743782def031d5',
   'src/third_party/ced/src': {
     'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5',
   },
@@ -505,13 +505,13 @@ deps = {
   'src/third_party/crc32c/src':
     'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558',
   'src/third_party/depot_tools':
-    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@a4e677522da5c326b656dc9584b3a4f6cde9fd05',
+    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@4ce8ba39a3488397a2d1494f167020f21de502f3',
   'src/third_party/ffmpeg':
     'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@946d97db8d906277085e361892b7efda5152e2f1',
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@3ca09e4818f4362e75c9c224f1d4c2065db75a0a',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@260c6fd37c502e984836a446c5d90e4a1298036d',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'c0a3840504ad618393057eb1fc121810c210dfd2',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '4cead1b0fc9987f994de97c7914e1f0c94047579',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@82543c0c7fa9a6436b46102e26e57e09aefa192d',
+    'https://chromium.googlesource.com/chromium/src/tools@9765576870401b230a826e0e4f7a12851575306e',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'A6oLcxGdFmfrPoaAcVYoVYHlkB3InvLKiIKnIgmGJVcC',
+          'version': 'Hi79Lu6qQ-Fbcjwgf_f-NEYJweP4QdZqjJ0lXwc0PpcC',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -864,7 +864,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7364b490b5f78d58efe23ea76e74210fd6c3c76f',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + 'c4eb3c4f4c91511eb5c76d608e07dde2a1cc8b23',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '4387e3248109bef2c07a8b69a2b8249d7af5dcf0',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From 3e0a723522b893a4498264f5d5dd7b8a1283bb2d Mon Sep 17 00:00:00 2001
From: Philipp Hancke <philipp.hancke@googlemail.com>
Date: Thu, 12 Mar 2026 14:46:16 +0100
Subject: [PATCH 286/472] build: gn_check_autofix should retain blank lines,
 comments and # keep tagged lines

Bug: webrtc:467294026
Change-Id: I613ef701e678563a3f9e41e7a008b42540137916
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456141
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#47139}
---
 api/BUILD.gn                     |  8 ++++----
 tools_webrtc/gn_check_autofix.py | 25 +++++++++++++++----------
 2 files changed, 19 insertions(+), 14 deletions(-)

diff --git a/api/BUILD.gn b/api/BUILD.gn
index 8c62fe2a0d..9df13cc56d 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -662,10 +662,10 @@ rtc_library("libjingle_peerconnection_api") {
     # Basically, don't add stuff here. You might break sensitive downstream
     # targets like pnacl. API should not depend on anything outside of this
     # file, really. All these should arguably go away in time.
-    "../media:rtc_media_config",
-    "../rtc_base:checks",
-    "../rtc_base:threading",
-    "../rtc_base/system:rtc_export",
+    "../media:rtc_media_config",  # keep
+    "../rtc_base:checks",  # keep
+    "../rtc_base:threading",  # keep
+    "../rtc_base/system:rtc_export",  # keep
   ]
 }
 
diff --git a/tools_webrtc/gn_check_autofix.py b/tools_webrtc/gn_check_autofix.py
index 99d57dfe70..0732f4fe61 100755
--- a/tools_webrtc/gn_check_autofix.py
+++ b/tools_webrtc/gn_check_autofix.py
@@ -166,20 +166,25 @@ def rebase(base_path, dependency_path, dependency):
 
 def _update_deps(new_lines, current_deps_lines_content, block_type):
     """Helper to finalize and append a processed deps block to new_lines."""
-    deps_to_keep = []
-    for dep_line in current_deps_lines_content:
-        # Keep absolute dependencies.
-        deps_to_keep.extend(re.findall(r'"(//[^"]*)"', dep_line))
-        # Keep test targets.
-        deps_to_keep.extend(re.findall(r'"([^"]*_unittests)"', dep_line))
-
     if block_type == 'deps_set':
         new_lines.append('deps = [\n')
     elif block_type == 'deps_add':
         new_lines.append('deps += [\n')
-
-    for dep in deps_to_keep:
-        new_lines.append('"' + dep + '",\n')
+    for dep_line in current_deps_lines_content:
+        line = dep_line.strip()
+        # Keep blank lines (inserted for formatting), comments and
+        # lines tagged with "  # keep".
+        if not line or line.startswith('#') or line.endswith(' # keep'):
+            new_lines.append(dep_line)
+            continue
+        deps_to_keep = []
+        # Keep absolute dependencies.
+        deps_to_keep.extend(re.findall(r'"(//[^"]*)"', dep_line))
+        # Keep _unittest and _unittests target.
+        deps_to_keep.extend(
+            re.findall(r'"([^"]*[_:](?:unit)?tests?)"', dep_line))
+        for dep in deps_to_keep:
+            new_lines.append('"' + dep + '",\n')
     new_lines.append(']\n')
 
 

From aafe141c9ac3e016b6e6dda4b1d60a97a99b55d9 Mon Sep 17 00:00:00 2001
From: Jan Grulich <grulja@gmail.com>
Date: Thu, 12 Mar 2026 15:11:27 +0100
Subject: [PATCH 287/472] PipeWire capture: add CFI suppression for
 StopCapture()

We added recently spa_hook_remove() call that gets inlined and there is
an indirect call through a C function pointer, which CFI cannot verify.

Bug: chromium:491979284
Change-Id: Ide5ba4d2cb2c3c3ad63e107cb6df1cd42c6578fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456240
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47140}
---
 modules/video_capture/linux/video_capture_pipewire.cc | 1 +
 1 file changed, 1 insertion(+)

diff --git a/modules/video_capture/linux/video_capture_pipewire.cc b/modules/video_capture/linux/video_capture_pipewire.cc
index 58021f9f69..ca35498016 100644
--- a/modules/video_capture/linux/video_capture_pipewire.cc
+++ b/modules/video_capture/linux/video_capture_pipewire.cc
@@ -250,6 +250,7 @@ int32_t VideoCaptureModulePipeWire::StartCapture(
   return 0;
 }
 
+RTC_NO_SANITIZE("cfi-icall")
 int32_t VideoCaptureModulePipeWire::StopCapture() {
   RTC_DCHECK_RUN_ON(&api_checker_);
 

From 4f037dde4a198fddfb51bb7d4d335bfcc3adb659 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Thu, 12 Mar 2026 13:04:12 -0700
Subject: [PATCH 288/472] Roll chromium_revision b937d39273..32ae6a1187
 (1597657:1598618)

Change log: https://chromium.googlesource.com/chromium/src/+log/b937d39273..32ae6a1187
Full diff: https://chromium.googlesource.com/chromium/src/+/b937d39273..32ae6a1187

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/7d29c62736..5261982778
* src/buildtools: https://chromium.googlesource.com/chromium/src/buildtools/+log/fef8d6947f..95ed44cf5f
* src/buildtools/linux64: git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9..git_revision:2941d2a26fa9cef3103f10e1dee5da7cd404008a
* src/buildtools/mac: git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9..git_revision:2941d2a26fa9cef3103f10e1dee5da7cd404008a
* src/buildtools/win: git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9..git_revision:2941d2a26fa9cef3103f10e1dee5da7cd404008a
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/94f693fcb6..6d06b9d888
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/e7a1e88ecc..14711b1d7a
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/a133c94b51..d46a522070
* src/third_party/android_build_tools/aapt2/cipd: vUP1cs7krVsNHbUJpwSDR7rvWuYLGzINVa9xopPpt74C..gsaUgZUqoyD0XG1E9-xesSWknHZINEuS00iSEncvlE0C
* src/third_party/android_build_tools/lint/cipd: 1N_9yErKkcZgqyxBghA0bb5L5AFEPZCt36Gv2o8_OAwC..UdJPLkDOu5joGtPdwsjc1Ltg5hcadLKPynhbd6NacsoC
* src/third_party/android_build_tools/manifest_merger/cipd: xT8A6Q_b44_iYKqAMvRkWmvID-ps-I6uBDfRTQv68bcC..KYQSw0Ej8zT_IRNL02DxEmk0nCvkykopkQhsGk_HE0EC
* src/third_party/android_deps/autorolled/cipd: umaYTBt4Vsa5QZYp57rLJDNzoXalsR2hPGwfYbpZYfwC..3VYcqPssBf1YI1iK0N2UUMsT2Ik5BlMrvm-wOVr1fMQC
* src/third_party/androidx/cipd: Hi79Lu6qQ-Fbcjwgf_f-NEYJweP4QdZqjJ0lXwc0PpcC..Mn6UxK4fOcnps2oJ1ikX7T1Nf93dtMBkX68uqIyoQ0sC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/eedf10c1f4..5e289a7952
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/e0ebf38a01..b9ddc08533
* src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/c8a5676846..e1e57b8776
* src/third_party/cpuinfo/src: https://chromium.googlesource.com/external/github.com/pytorch/cpuinfo.git/+log/7364b490b5..7607ca5004
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/4ce8ba39a3..3de3b74fc6
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/4387e32481..2a2456c873
* src/third_party/fuzztest/src: https://chromium.googlesource.com/external/github.com/google/fuzztest.git/+log/54ed362692..41e160679e
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/260c6fd37c..c68e2a4d5d
* src/third_party/harfbuzz-ng/src: https://chromium.googlesource.com/external/github.com/harfbuzz/harfbuzz.git/+log/fa2908bf16..5d4e96ad8d
* src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/044963a253..76c8a4be56
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/4cead1b0fc..d798e01b99
* src/third_party/r8/cipd: CoaQUbD7pDX3OuETi_O5wkv_Y198RVIUaSobSEx9zfoC..UMLn7pE7DklTsk-SIiMAwch46m8rSF1vG2HQAbMuZvMC
* src/third_party/tflite/src: https://chromium.googlesource.com/external/github.com/tensorflow/tensorflow.git/+log/f920077085..d20369f4f2
* src/third_party/xnnpack/src: https://chromium.googlesource.com/external/github.com/google/XNNPACK.git/+log/26feacc3eb..15e04ff8b6
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/9765576870..7544e32bf3
DEPS diff: https://chromium.googlesource.com/chromium/src/+/b937d39273..32ae6a1187/DEPS

BUG=None

Change-Id: I0143315f704c805e18aac95a47fbaa8fb69978b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456320
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47141}
---
 DEPS | 58 +++++++++++++++++++++++++++++-----------------------------
 1 file changed, 29 insertions(+), 29 deletions(-)

diff --git a/DEPS b/DEPS
index dffcecaef3..3e8b583607 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': 'b937d39273b52034e8542dea4b040587be0a127f',
+  'chromium_revision': '32ae6a1187736ad8a28d172c899db2860d984628',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,28 +69,28 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@7d29c62736304aeaac07df0b75546ffd2b75917a',
+    'https://chromium.googlesource.com/chromium/src/build@5261982778f9c1edc28c5eb9422d71ed638f09dd',
   'src/buildtools':
-    'https://chromium.googlesource.com/chromium/src/buildtools@fef8d6947fa54202e8a00c58d5ac8fd15780bc33',
+    'https://chromium.googlesource.com/chromium/src/buildtools@95ed44cf5f06dbb5861030b91c9db9ccb4316762',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
   'src/examples/androidtests/third_party/gradle': {
     'url': 'https://chromium.googlesource.com/external/github.com/gradle/gradle.git@f2d1fb54a951d8b11d25748e4711bec8d128d7e3',
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@94f693fcb6dde967f11668a9e21eec1a4ec8b49c',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@6d06b9d888596e1b97bcc4bd1a68c0ad84eefe16',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@e7a1e88ecc35a70f3d198674042753ec9e8e3650',
+    'https://chromium.googlesource.com/chromium/src/testing@14711b1d7ae2fc3bdb460b816b6921c012aeb1d3',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@a133c94b51ac034d5f375cc1746e66dcdee57282',
+    'https://chromium.googlesource.com/chromium/src/third_party@d46a52207098d4e7bc21a01a97d2d133c2375e34',
 
   'src/buildtools/linux64': {
     'packages': [
       {
         'package': 'gn/gn/linux-${{arch}}',
-        'version': 'git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9',
+        'version': 'git_revision:2941d2a26fa9cef3103f10e1dee5da7cd404008a',
       }
     ],
     'dep_type': 'cipd',
@@ -100,7 +100,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/mac-${{arch}}',
-        'version': 'git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9',
+        'version': 'git_revision:2941d2a26fa9cef3103f10e1dee5da7cd404008a',
       }
     ],
     'dep_type': 'cipd',
@@ -110,7 +110,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/windows-amd64',
-        'version': 'git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9',
+        'version': 'git_revision:2941d2a26fa9cef3103f10e1dee5da7cd404008a',
       }
     ],
     'dep_type': 'cipd',
@@ -331,13 +331,13 @@ deps = {
   'src/third_party/clang-format/script':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@c2725e0622e1a86d55f14514f2177a39efea4a0e',
   'src/third_party/compiler-rt/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@c8a567684659fe05ede3a220965b94fd71c4a7cb',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@e1e57b877616abd748cdd63bb78e913e26f5bdfa',
   'src/third_party/libc++/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@7ab65651aed6802d2599dcb7a73b1f82d5179d05',
   'src/third_party/libc++abi/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@8f11bb1d4438d0239d0dfc1bd9456a9f31629dda',
   'src/third_party/llvm-libc/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@044963a253be3b8b2a63a1b2e0f3e361b01bc186',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@76c8a4be56babbe6104f5cd9eda25a95dd277b7a',
   'src/third_party/libunwind/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@db838d918570d4e381ecf9f5cc70a0098c9c2cd6',
 
@@ -402,7 +402,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/android_build_tools/aapt2',
-              'version': 'vUP1cs7krVsNHbUJpwSDR7rvWuYLGzINVa9xopPpt74C',
+              'version': 'gsaUgZUqoyD0XG1E9-xesSWknHZINEuS00iSEncvlE0C',
           },
       ],
       'condition': 'checkout_android',
@@ -457,7 +457,7 @@ deps = {
       'packages': [
           {
                'package': 'chromium/third_party/android_build_tools/lint',
-               'version': '1N_9yErKkcZgqyxBghA0bb5L5AFEPZCt36Gv2o8_OAwC',
+               'version': 'UdJPLkDOu5joGtPdwsjc1Ltg5hcadLKPynhbd6NacsoC',
           },
       ],
       'condition': 'checkout_android and non_git_source',
@@ -488,11 +488,11 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@eedf10c1f4ab44629e2fe3fe6f07e2fb0897c0f4',
+    'https://boringssl.googlesource.com/boringssl.git@5e289a79529ff730ee6473772813e2b2f8c8cda5',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@8be0e3114685fcc1589561067282edf75ea1259a',
   'src/third_party/catapult':
-    'https://chromium.googlesource.com/catapult.git@e0ebf38a01214aba11f31daa1c743782def031d5',
+    'https://chromium.googlesource.com/catapult.git@b9ddc08533aca2b2878bc05f2cce8e3051616313',
   'src/third_party/ced/src': {
     'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5',
   },
@@ -505,13 +505,13 @@ deps = {
   'src/third_party/crc32c/src':
     'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558',
   'src/third_party/depot_tools':
-    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@4ce8ba39a3488397a2d1494f167020f21de502f3',
+    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@3de3b74fc6d503ef2d9ad408804abb222715070b',
   'src/third_party/ffmpeg':
     'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@946d97db8d906277085e361892b7efda5152e2f1',
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@260c6fd37c502e984836a446c5d90e4a1298036d',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@c68e2a4d5d6dd29e0a25dc8de956ef06ad622a4c',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -521,7 +521,7 @@ deps = {
   'src/third_party/freetype/src':
     'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@99b479dc34728936b006679a31e12b8cf432fc55',
   'src/third_party/harfbuzz-ng/src':
-    'https://chromium.googlesource.com/external/github.com/harfbuzz/harfbuzz.git@fa2908bf16d2ccd6623f4d575455fea72a1a722b',
+    'https://chromium.googlesource.com/external/github.com/harfbuzz/harfbuzz.git@5d4e96ad8d00fc871ffa17707b2ca08fa850e7d6',
   'src/third_party/google_benchmark/src': {
     'url': 'https://chromium.googlesource.com/external/github.com/google/benchmark.git@188e8278990a9069ffc84441cb5a024fd0bede37',
   },
@@ -593,7 +593,7 @@ deps = {
   'src/third_party/libFuzzer/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt/lib/fuzzer.git@bea408a6e01f0f7e6c82a43121fe3af4506c932e',
   'src/third_party/fuzztest/src':
-    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@54ed362692177e93df0fa202cae404b57694e828',
+    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@41e160679e3c010a860e674e7997e2b285c6e0a2',
   'src/third_party/libjpeg_turbo':
     'https://chromium.googlesource.com/chromium/deps/libjpeg_turbo.git@d1f5f2393e0d51f840207342ae86e55a86443288',
   'src/third_party/libsrtp':
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '4cead1b0fc9987f994de97c7914e1f0c94047579',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'd798e01b99a063720c65992ad79903390fb373c4',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -644,7 +644,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/r8',
-              'version': 'CoaQUbD7pDX3OuETi_O5wkv_Y198RVIUaSobSEx9zfoC',
+              'version': 'UMLn7pE7DklTsk-SIiMAwch46m8rSF1vG2HQAbMuZvMC',
           },
       ],
       'condition': 'checkout_android',
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@9765576870401b230a826e0e4f7a12851575306e',
+    'https://chromium.googlesource.com/chromium/src/tools@7544e32bf37a06ac5a8fc02814fa7ae2ca251c55',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'Hi79Lu6qQ-Fbcjwgf_f-NEYJweP4QdZqjJ0lXwc0PpcC',
+          'version': 'Mn6UxK4fOcnps2oJ1ikX7T1Nf93dtMBkX68uqIyoQ0sC',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -718,7 +718,7 @@ deps = {
       'packages': [
           {
                'package': 'chromium/third_party/android_build_tools/manifest_merger',
-               'version': 'xT8A6Q_b44_iYKqAMvRkWmvID-ps-I6uBDfRTQv68bcC',
+               'version': 'KYQSw0Ej8zT_IRNL02DxEmk0nCvkykopkQhsGk_HE0EC',
           },
       ],
       'condition': 'checkout_android',
@@ -786,7 +786,7 @@ deps = {
   },
 
   'src/third_party/tflite/src':
-    Var('chromium_git') + '/external/github.com/tensorflow/tensorflow.git' + '@' + 'f9200770858f5721b7fa95db7aa3e4cd712fe9df',
+    Var('chromium_git') + '/external/github.com/tensorflow/tensorflow.git' + '@' + 'd20369f4f226598800e0d406db567d2d1c17fa95',
 
   'src/third_party/turbine/cipd': {
       'packages': [
@@ -842,7 +842,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/android_deps/autorolled',
-              'version': 'umaYTBt4Vsa5QZYp57rLJDNzoXalsR2hPGwfYbpZYfwC',
+              'version': '3VYcqPssBf1YI1iK0N2UUMsT2Ik5BlMrvm-wOVr1fMQC',
           },
       ],
       'condition': 'checkout_android and non_git_source',
@@ -852,7 +852,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/google/pthreadpool.git' + '@' + '9003ee6c137cea3b94161bd5c614fb43be523ee1',
 
   'src/third_party/xnnpack/src':
-    Var('chromium_git') + '/external/github.com/google/XNNPACK.git' + '@' + '26feacc3ebe30d1f3a82e1ff3938ef9859d74640',
+    Var('chromium_git') + '/external/github.com/google/XNNPACK.git' + '@' + '15e04ff8b61e0bc2118bad268447be5b2075aa66',
 
   'src/third_party/farmhash/src':
     Var('chromium_git') + '/external/github.com/google/farmhash.git' + '@' + '816a4ae622e964763ca0862d9dbd19324a1eaf45',
@@ -861,10 +861,10 @@ deps = {
     Var('chromium_git') + '/external/github.com/google/ruy.git' + '@' + '2af88863614a8298689cc52b1a47b3fcad7be835',
 
   'src/third_party/cpuinfo/src':
-    Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7364b490b5f78d58efe23ea76e74210fd6c3c76f',
+    Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7607ca500436b37ad23fb8d18614bec7796b68a7',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '4387e3248109bef2c07a8b69a2b8249d7af5dcf0',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '2a2456c873d6f4bf43a9adf919f13ad917d57c2d',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From 9fec1823163729910222ffd52a6c3440683f740f Mon Sep 17 00:00:00 2001
From: Jeremy Leconte <jleconte@google.com>
Date: Thu, 12 Mar 2026 14:57:00 -0700
Subject: [PATCH 289/472] Revert "Roll chromium_revision b937d39273..32ae6a1187
 (1597657:1598618)"

This reverts commit 4f037dde4a198fddfb51bb7d4d335bfcc3adb659.

Reason for revert: breaks CI

Original change's description:
> Roll chromium_revision b937d39273..32ae6a1187 (1597657:1598618)
>
> Change log: https://chromium.googlesource.com/chromium/src/+log/b937d39273..32ae6a1187
> Full diff: https://chromium.googlesource.com/chromium/src/+/b937d39273..32ae6a1187
>
> Changed dependencies
> * src/build: https://chromium.googlesource.com/chromium/src/build/+log/7d29c62736..5261982778
> * src/buildtools: https://chromium.googlesource.com/chromium/src/buildtools/+log/fef8d6947f..95ed44cf5f
> * src/buildtools/linux64: git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9..git_revision:2941d2a26fa9cef3103f10e1dee5da7cd404008a
> * src/buildtools/mac: git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9..git_revision:2941d2a26fa9cef3103f10e1dee5da7cd404008a
> * src/buildtools/win: git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9..git_revision:2941d2a26fa9cef3103f10e1dee5da7cd404008a
> * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/94f693fcb6..6d06b9d888
> * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/e7a1e88ecc..14711b1d7a
> * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/a133c94b51..d46a522070
> * src/third_party/android_build_tools/aapt2/cipd: vUP1cs7krVsNHbUJpwSDR7rvWuYLGzINVa9xopPpt74C..gsaUgZUqoyD0XG1E9-xesSWknHZINEuS00iSEncvlE0C
> * src/third_party/android_build_tools/lint/cipd: 1N_9yErKkcZgqyxBghA0bb5L5AFEPZCt36Gv2o8_OAwC..UdJPLkDOu5joGtPdwsjc1Ltg5hcadLKPynhbd6NacsoC
> * src/third_party/android_build_tools/manifest_merger/cipd: xT8A6Q_b44_iYKqAMvRkWmvID-ps-I6uBDfRTQv68bcC..KYQSw0Ej8zT_IRNL02DxEmk0nCvkykopkQhsGk_HE0EC
> * src/third_party/android_deps/autorolled/cipd: umaYTBt4Vsa5QZYp57rLJDNzoXalsR2hPGwfYbpZYfwC..3VYcqPssBf1YI1iK0N2UUMsT2Ik5BlMrvm-wOVr1fMQC
> * src/third_party/androidx/cipd: Hi79Lu6qQ-Fbcjwgf_f-NEYJweP4QdZqjJ0lXwc0PpcC..Mn6UxK4fOcnps2oJ1ikX7T1Nf93dtMBkX68uqIyoQ0sC
> * src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/eedf10c1f4..5e289a7952
> * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/e0ebf38a01..b9ddc08533
> * src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/c8a5676846..e1e57b8776
> * src/third_party/cpuinfo/src: https://chromium.googlesource.com/external/github.com/pytorch/cpuinfo.git/+log/7364b490b5..7607ca5004
> * src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/4ce8ba39a3..3de3b74fc6
> * src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/4387e32481..2a2456c873
> * src/third_party/fuzztest/src: https://chromium.googlesource.com/external/github.com/google/fuzztest.git/+log/54ed362692..41e160679e
> * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/260c6fd37c..c68e2a4d5d
> * src/third_party/harfbuzz-ng/src: https://chromium.googlesource.com/external/github.com/harfbuzz/harfbuzz.git/+log/fa2908bf16..5d4e96ad8d
> * src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/044963a253..76c8a4be56
> * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/4cead1b0fc..d798e01b99
> * src/third_party/r8/cipd: CoaQUbD7pDX3OuETi_O5wkv_Y198RVIUaSobSEx9zfoC..UMLn7pE7DklTsk-SIiMAwch46m8rSF1vG2HQAbMuZvMC
> * src/third_party/tflite/src: https://chromium.googlesource.com/external/github.com/tensorflow/tensorflow.git/+log/f920077085..d20369f4f2
> * src/third_party/xnnpack/src: https://chromium.googlesource.com/external/github.com/google/XNNPACK.git/+log/26feacc3eb..15e04ff8b6
> * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/9765576870..7544e32bf3
> DEPS diff: https://chromium.googlesource.com/chromium/src/+/b937d39273..32ae6a1187/DEPS
>
> BUG=None
>
> Change-Id: I0143315f704c805e18aac95a47fbaa8fb69978b8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456320
> Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
> Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
> Cr-Commit-Position: refs/heads/main@{#47141}

Bug: None
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: Ifaf3e08f7111f364a15377d97c85f87f7481c877
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456122
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#47142}
---
 DEPS | 58 +++++++++++++++++++++++++++++-----------------------------
 1 file changed, 29 insertions(+), 29 deletions(-)

diff --git a/DEPS b/DEPS
index 3e8b583607..dffcecaef3 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '32ae6a1187736ad8a28d172c899db2860d984628',
+  'chromium_revision': 'b937d39273b52034e8542dea4b040587be0a127f',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,28 +69,28 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@5261982778f9c1edc28c5eb9422d71ed638f09dd',
+    'https://chromium.googlesource.com/chromium/src/build@7d29c62736304aeaac07df0b75546ffd2b75917a',
   'src/buildtools':
-    'https://chromium.googlesource.com/chromium/src/buildtools@95ed44cf5f06dbb5861030b91c9db9ccb4316762',
+    'https://chromium.googlesource.com/chromium/src/buildtools@fef8d6947fa54202e8a00c58d5ac8fd15780bc33',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
   'src/examples/androidtests/third_party/gradle': {
     'url': 'https://chromium.googlesource.com/external/github.com/gradle/gradle.git@f2d1fb54a951d8b11d25748e4711bec8d128d7e3',
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@6d06b9d888596e1b97bcc4bd1a68c0ad84eefe16',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@94f693fcb6dde967f11668a9e21eec1a4ec8b49c',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@14711b1d7ae2fc3bdb460b816b6921c012aeb1d3',
+    'https://chromium.googlesource.com/chromium/src/testing@e7a1e88ecc35a70f3d198674042753ec9e8e3650',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@d46a52207098d4e7bc21a01a97d2d133c2375e34',
+    'https://chromium.googlesource.com/chromium/src/third_party@a133c94b51ac034d5f375cc1746e66dcdee57282',
 
   'src/buildtools/linux64': {
     'packages': [
       {
         'package': 'gn/gn/linux-${{arch}}',
-        'version': 'git_revision:2941d2a26fa9cef3103f10e1dee5da7cd404008a',
+        'version': 'git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9',
       }
     ],
     'dep_type': 'cipd',
@@ -100,7 +100,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/mac-${{arch}}',
-        'version': 'git_revision:2941d2a26fa9cef3103f10e1dee5da7cd404008a',
+        'version': 'git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9',
       }
     ],
     'dep_type': 'cipd',
@@ -110,7 +110,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/windows-amd64',
-        'version': 'git_revision:2941d2a26fa9cef3103f10e1dee5da7cd404008a',
+        'version': 'git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9',
       }
     ],
     'dep_type': 'cipd',
@@ -331,13 +331,13 @@ deps = {
   'src/third_party/clang-format/script':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@c2725e0622e1a86d55f14514f2177a39efea4a0e',
   'src/third_party/compiler-rt/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@e1e57b877616abd748cdd63bb78e913e26f5bdfa',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@c8a567684659fe05ede3a220965b94fd71c4a7cb',
   'src/third_party/libc++/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@7ab65651aed6802d2599dcb7a73b1f82d5179d05',
   'src/third_party/libc++abi/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@8f11bb1d4438d0239d0dfc1bd9456a9f31629dda',
   'src/third_party/llvm-libc/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@76c8a4be56babbe6104f5cd9eda25a95dd277b7a',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@044963a253be3b8b2a63a1b2e0f3e361b01bc186',
   'src/third_party/libunwind/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@db838d918570d4e381ecf9f5cc70a0098c9c2cd6',
 
@@ -402,7 +402,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/android_build_tools/aapt2',
-              'version': 'gsaUgZUqoyD0XG1E9-xesSWknHZINEuS00iSEncvlE0C',
+              'version': 'vUP1cs7krVsNHbUJpwSDR7rvWuYLGzINVa9xopPpt74C',
           },
       ],
       'condition': 'checkout_android',
@@ -457,7 +457,7 @@ deps = {
       'packages': [
           {
                'package': 'chromium/third_party/android_build_tools/lint',
-               'version': 'UdJPLkDOu5joGtPdwsjc1Ltg5hcadLKPynhbd6NacsoC',
+               'version': '1N_9yErKkcZgqyxBghA0bb5L5AFEPZCt36Gv2o8_OAwC',
           },
       ],
       'condition': 'checkout_android and non_git_source',
@@ -488,11 +488,11 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@5e289a79529ff730ee6473772813e2b2f8c8cda5',
+    'https://boringssl.googlesource.com/boringssl.git@eedf10c1f4ab44629e2fe3fe6f07e2fb0897c0f4',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@8be0e3114685fcc1589561067282edf75ea1259a',
   'src/third_party/catapult':
-    'https://chromium.googlesource.com/catapult.git@b9ddc08533aca2b2878bc05f2cce8e3051616313',
+    'https://chromium.googlesource.com/catapult.git@e0ebf38a01214aba11f31daa1c743782def031d5',
   'src/third_party/ced/src': {
     'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5',
   },
@@ -505,13 +505,13 @@ deps = {
   'src/third_party/crc32c/src':
     'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558',
   'src/third_party/depot_tools':
-    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@3de3b74fc6d503ef2d9ad408804abb222715070b',
+    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@4ce8ba39a3488397a2d1494f167020f21de502f3',
   'src/third_party/ffmpeg':
     'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@946d97db8d906277085e361892b7efda5152e2f1',
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@c68e2a4d5d6dd29e0a25dc8de956ef06ad622a4c',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@260c6fd37c502e984836a446c5d90e4a1298036d',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -521,7 +521,7 @@ deps = {
   'src/third_party/freetype/src':
     'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@99b479dc34728936b006679a31e12b8cf432fc55',
   'src/third_party/harfbuzz-ng/src':
-    'https://chromium.googlesource.com/external/github.com/harfbuzz/harfbuzz.git@5d4e96ad8d00fc871ffa17707b2ca08fa850e7d6',
+    'https://chromium.googlesource.com/external/github.com/harfbuzz/harfbuzz.git@fa2908bf16d2ccd6623f4d575455fea72a1a722b',
   'src/third_party/google_benchmark/src': {
     'url': 'https://chromium.googlesource.com/external/github.com/google/benchmark.git@188e8278990a9069ffc84441cb5a024fd0bede37',
   },
@@ -593,7 +593,7 @@ deps = {
   'src/third_party/libFuzzer/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt/lib/fuzzer.git@bea408a6e01f0f7e6c82a43121fe3af4506c932e',
   'src/third_party/fuzztest/src':
-    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@41e160679e3c010a860e674e7997e2b285c6e0a2',
+    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@54ed362692177e93df0fa202cae404b57694e828',
   'src/third_party/libjpeg_turbo':
     'https://chromium.googlesource.com/chromium/deps/libjpeg_turbo.git@d1f5f2393e0d51f840207342ae86e55a86443288',
   'src/third_party/libsrtp':
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'd798e01b99a063720c65992ad79903390fb373c4',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '4cead1b0fc9987f994de97c7914e1f0c94047579',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -644,7 +644,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/r8',
-              'version': 'UMLn7pE7DklTsk-SIiMAwch46m8rSF1vG2HQAbMuZvMC',
+              'version': 'CoaQUbD7pDX3OuETi_O5wkv_Y198RVIUaSobSEx9zfoC',
           },
       ],
       'condition': 'checkout_android',
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@7544e32bf37a06ac5a8fc02814fa7ae2ca251c55',
+    'https://chromium.googlesource.com/chromium/src/tools@9765576870401b230a826e0e4f7a12851575306e',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'Mn6UxK4fOcnps2oJ1ikX7T1Nf93dtMBkX68uqIyoQ0sC',
+          'version': 'Hi79Lu6qQ-Fbcjwgf_f-NEYJweP4QdZqjJ0lXwc0PpcC',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -718,7 +718,7 @@ deps = {
       'packages': [
           {
                'package': 'chromium/third_party/android_build_tools/manifest_merger',
-               'version': 'KYQSw0Ej8zT_IRNL02DxEmk0nCvkykopkQhsGk_HE0EC',
+               'version': 'xT8A6Q_b44_iYKqAMvRkWmvID-ps-I6uBDfRTQv68bcC',
           },
       ],
       'condition': 'checkout_android',
@@ -786,7 +786,7 @@ deps = {
   },
 
   'src/third_party/tflite/src':
-    Var('chromium_git') + '/external/github.com/tensorflow/tensorflow.git' + '@' + 'd20369f4f226598800e0d406db567d2d1c17fa95',
+    Var('chromium_git') + '/external/github.com/tensorflow/tensorflow.git' + '@' + 'f9200770858f5721b7fa95db7aa3e4cd712fe9df',
 
   'src/third_party/turbine/cipd': {
       'packages': [
@@ -842,7 +842,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/android_deps/autorolled',
-              'version': '3VYcqPssBf1YI1iK0N2UUMsT2Ik5BlMrvm-wOVr1fMQC',
+              'version': 'umaYTBt4Vsa5QZYp57rLJDNzoXalsR2hPGwfYbpZYfwC',
           },
       ],
       'condition': 'checkout_android and non_git_source',
@@ -852,7 +852,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/google/pthreadpool.git' + '@' + '9003ee6c137cea3b94161bd5c614fb43be523ee1',
 
   'src/third_party/xnnpack/src':
-    Var('chromium_git') + '/external/github.com/google/XNNPACK.git' + '@' + '15e04ff8b61e0bc2118bad268447be5b2075aa66',
+    Var('chromium_git') + '/external/github.com/google/XNNPACK.git' + '@' + '26feacc3ebe30d1f3a82e1ff3938ef9859d74640',
 
   'src/third_party/farmhash/src':
     Var('chromium_git') + '/external/github.com/google/farmhash.git' + '@' + '816a4ae622e964763ca0862d9dbd19324a1eaf45',
@@ -861,10 +861,10 @@ deps = {
     Var('chromium_git') + '/external/github.com/google/ruy.git' + '@' + '2af88863614a8298689cc52b1a47b3fcad7be835',
 
   'src/third_party/cpuinfo/src':
-    Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7607ca500436b37ad23fb8d18614bec7796b68a7',
+    Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7364b490b5f78d58efe23ea76e74210fd6c3c76f',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '2a2456c873d6f4bf43a9adf919f13ad917d57c2d',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '4387e3248109bef2c07a8b69a2b8249d7af5dcf0',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From 04bdac9d47d522c4b58ef87bc7f4823bd8de79f8 Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Thu, 12 Mar 2026 07:12:37 -0700
Subject: [PATCH 290/472] Revert^4 "Default enable
 WebRTC-HeaderExtensionNegotiateMemory"

This reverts commit 5f9534d9205a870d40aff649967a38fbde7d7653.

Reland "Default enable WebRTC-HeaderExtensionNegotiateMemory"

This reverts commit 5f9534d9205a870d40aff649967a38fbde7d7653.

Reason for revert: M147 is now cut, landing for M148.

Original change's description:
> Revert^3 "Default enable WebRTC-HeaderExtensionNegotiateMemory"
>
> This reverts commit 2f069781037ac1f9965d39152c6c97ea98ce6aff.
>
> Reland "Default enable WebRTC-HeaderExtensionNegotiateMemory"
>
> This reverts commit 2f069781037ac1f9965d39152c6c97ea98ce6aff.
>
> Reason for revert: Abundance of caution - want more canary time.
>
> Original change's description:
> > Revert^2 "Default enable WebRTC-HeaderExtensionNegotiateMemory"
> >
> > This reverts commit 74d988fd57f8f1b6722d8ac6f49c686b9f2bc196.
> >
> > Reland "Default enable WebRTC-HeaderExtensionNegotiateMemory"
> >
> > This reverts commit 74d988fd57f8f1b6722d8ac6f49c686b9f2bc196.
> >
> > Reason for revert: Added code to skip stopped transceivers
> >
> > Original change's description:
> > > Revert "Default enable WebRTC-HeaderExtensionNegotiateMemory"
> > >
> > > This reverts commit 4ecf297a4c0af68ff8af8c3a8e6b0832bddd14d5.
> > >
> > > Reason for revert: Impacted unexpected application usages
> > >
> > > Original change's description:
> > > > Default enable WebRTC-HeaderExtensionNegotiateMemory
> > > >
> > > > Spec change: https://github.com/w3c/webrtc-extensions/pull/238
> > > >
> > > > Chromestatus entry for change: https://chromestatus.com/feature/5135528638939136
> > > >
> > > > Bug: webrtc:439514253
> > > > Change-Id: I296f2f06dd69b4e30be419b44c926acfd2722e2d
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/407821
> > > > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > > > Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/main@{#45993}
> > >
> > > Bug: webrtc:439514253, chromium:467164231
> > > Change-Id: I95dbc8cbaf1b769ffaea291801684bc26e4b9ccb
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/433102
> > > Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > > Cr-Commit-Position: refs/heads/main@{#46456}
> >
> > Bug: webrtc:439514253, chromium:467164231
> > Change-Id: I2df85ba25016374a52abb13037cdca74fb1a395d
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453240
> > Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> > Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#47038}
>
> No-Try: True
> Bug: webrtc:439514253, chromium:467164231, webrtc:489813847
> Change-Id: Id2151293ffe14d2238a1363dd7df679c8a27c42f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453820
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
> Cr-Commit-Position: refs/heads/main@{#47093}

Bug: webrtc:439514253, chromium:467164231, webrtc:489813847
Change-Id: I5249afdfaef3bc4d72570448c33d04ff71ddec11
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456241
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47143}
---
 pc/rtp_transceiver.cc          | 6 +++---
 pc/rtp_transmission_manager.cc | 3 ++-
 2 files changed, 5 insertions(+), 4 deletions(-)

diff --git a/pc/rtp_transceiver.cc b/pc/rtp_transceiver.cc
index f1d558210c..0701dca781 100644
--- a/pc/rtp_transceiver.cc
+++ b/pc/rtp_transceiver.cc
@@ -1224,19 +1224,19 @@ void RtpTransceiver::OnNegotiationUpdate(
   RTC_DCHECK(content);
   if (sdp_type == SdpType::kAnswer || sdp_type == SdpType::kPrAnswer) {
     negotiated_header_extensions_ = content->rtp_header_extensions();
-    if (env_.field_trials().IsEnabled(
+    if (!env_.field_trials().IsDisabled(
             "WebRTC-HeaderExtensionNegotiateMemory")) {
       header_extensions_to_negotiate_ = GetNegotiatedHeaderExtensions();
     }
   } else if (sdp_type == SdpType::kOffer) {
-    if (env_.field_trials().IsEnabled(
+    if (!env_.field_trials().IsDisabled(
             "WebRTC-HeaderExtensionNegotiateMemory")) {
       header_extensions_for_rollback_ = header_extensions_to_negotiate_;
       header_extensions_to_negotiate_ =
           GetOfferedAndImplementedHeaderExtensions(content);
     }
   } else if (sdp_type == SdpType::kRollback) {
-    if (env_.field_trials().IsEnabled(
+    if (!env_.field_trials().IsDisabled(
             "WebRTC-HeaderExtensionNegotiateMemory")) {
       RTC_CHECK(!header_extensions_for_rollback_.empty());
       header_extensions_to_negotiate_ = header_extensions_for_rollback_;
diff --git a/pc/rtp_transmission_manager.cc b/pc/rtp_transmission_manager.cc
index dc6e4bc6f4..b67a3fe9e7 100644
--- a/pc/rtp_transmission_manager.cc
+++ b/pc/rtp_transmission_manager.cc
@@ -284,7 +284,8 @@ RtpTransmissionManager::CreateAndAddTransceiver(
   RTC_DCHECK(!FindSenderById(sender_id));
   std::vector<RtpHeaderExtensionCapability> header_extensions =
       std::move(header_extensions_to_negotiate);
-  if (env_.field_trials().IsEnabled("WebRTC-HeaderExtensionNegotiateMemory")) {
+  if (!env_.field_trials().IsDisabled(
+          "WebRTC-HeaderExtensionNegotiateMemory")) {
     // If we have already negotiated header extensions for this type,
     // and it is not stopped,
     // reuse the negotiated state for new transceivers of the same type.

From c5945824e8ec8e95c3b92b6bbdfb25073dfa6c31 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Thu, 12 Mar 2026 11:16:18 +0100
Subject: [PATCH 291/472] Replace RtpDemuxerCriteria member with discrete
 variables
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Instead of keeping an RtpDemuxerCriteria object as a member, replace it
with distinct member variables (mid_, payload_types_, ssrcs_) inside
BaseChannel. The RtpDemuxerCriteria object is now constructed on the
stack, on the network thread, when needed (a step that clarifies the
intended threading slightly).

This disentangles the mid, which is const and must be readable from the
worker and network threads, from other demuxer properties. The channel
implementations also only use the ssrcs and payload_types properties of
the RtpDemuxerCriteria, don't need the full object and now there is not
a duplicate copy of payload_types_.

Those changes unlock the ability to further move parts of the channel
implementation over to the network thread and get rid of some of the
thread hopping.

Some more work needs to be done to fully move the demuxer state to the
network thread however.

Bug: webrtc:42222117
Change-Id: I42f8861c0ed04a8d19656edba7f0053142f07b06
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456060
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47144}
---
 pc/channel.cc | 98 ++++++++++++++++++++++++---------------------------
 pc/channel.h  | 15 +++++---
 2 files changed, 57 insertions(+), 56 deletions(-)

diff --git a/pc/channel.cc b/pc/channel.cc
index 2c39a407a8..a3f4f40f95 100644
--- a/pc/channel.cc
+++ b/pc/channel.cc
@@ -31,6 +31,7 @@
 #include "api/sequence_checker.h"
 #include "api/task_queue/pending_task_safety_flag.h"
 #include "api/task_queue/task_queue_base.h"
+#include "call/rtp_demuxer.h"
 #include "media/base/codec.h"
 #include "media/base/media_channel.h"
 #include "media/base/rid_description.h"
@@ -192,7 +193,7 @@ BaseChannel::BaseChannel(
           crypto_options.srtp.enable_encrypted_rtp_header_extensions
               ? RtpExtension::kPreferEncryptedExtension
               : RtpExtension::kDiscardEncryptedExtension),
-      demuxer_criteria_(mid),
+      mid_(std::string(mid)),
       ssrc_generator_(ssrc_generator) {
   RTC_DCHECK_RUN_ON(worker_thread_);
   RTC_DCHECK(media_send_channel_);
@@ -225,7 +226,8 @@ bool BaseChannel::ConnectToRtpTransport_n(RtpTransportInternal* rtp_transport) {
 
   // We don't need to call OnDemuxerCriteriaUpdatePending/Complete because
   // there's no previous criteria to worry about.
-  if (!rtp_transport->RegisterRtpDemuxerSink(demuxer_criteria_, this)) {
+  RtpDemuxerCriteria criteria = demuxer_criteria();
+  if (!rtp_transport->RegisterRtpDemuxerSink(criteria, this)) {
     return false;
   }
   rtp_transport_ = rtp_transport;
@@ -578,10 +580,11 @@ bool BaseChannel::MaybeUpdateDemuxerAndRtpExtensions_w(
 
   // TODO(bugs.webrtc.org/13536): See if we can do this asynchronously.
 
-  if (update_demuxer)
+  if (update_demuxer) {
     media_receive_channel()->OnDemuxerCriteriaUpdatePending();
+  }
 
-  bool success = network_thread()->BlockingCall([&]() mutable {
+  bool success = network_thread()->BlockingCall([&] {
     RTC_DCHECK_RUN_ON(network_thread());
     if (!rtp_transport_) {
       // To repro this situation, run the
@@ -597,10 +600,11 @@ bool BaseChannel::MaybeUpdateDemuxerAndRtpExtensions_w(
     if (!update_demuxer)
       return true;
 
-    if (!rtp_transport_->RegisterRtpDemuxerSink(demuxer_criteria_, this)) {
+    RtpDemuxerCriteria criteria = demuxer_criteria();
+    if (!rtp_transport_->RegisterRtpDemuxerSink(criteria, this)) {
       error_desc =
           StringFormat("Failed to apply demuxer criteria for '%s': '%s'.",
-                       mid().c_str(), demuxer_criteria_.ToString().c_str());
+                       mid().c_str(), criteria.ToString().c_str());
       return false;
     }
     return true;
@@ -614,28 +618,36 @@ bool BaseChannel::MaybeUpdateDemuxerAndRtpExtensions_w(
 
 bool BaseChannel::RegisterRtpDemuxerSink_w() {
   media_receive_channel()->OnDemuxerCriteriaUpdatePending();
-  // Copy demuxer criteria, since they're a worker-thread variable
-  // and we want to pass them to the network thread
-  bool ret = network_thread_->BlockingCall(
-      [this, demuxer_criteria = demuxer_criteria_] {
-        RTC_DCHECK_RUN_ON(network_thread());
-        if (!rtp_transport_) {
-          // Transport was disconnected before attempting to update the
-          // criteria. This can happen while setting the remote description.
-          // See chromium:1295469 for an example.
-          return false;
-        }
-        // Note that RegisterRtpDemuxerSink first unregisters the sink if
-        // already registered. So this will change the state of the class
-        // whether the call succeeds or not.
-        return rtp_transport_->RegisterRtpDemuxerSink(demuxer_criteria, this);
-      });
+  bool ret = network_thread_->BlockingCall([&] {
+    RTC_DCHECK_RUN_ON(network_thread());
+    if (!rtp_transport_) {
+      // Transport was disconnected before attempting to update the
+      // criteria. This can happen while setting the remote description.
+      // See chromium:1295469 for an example.
+      return false;
+    }
+    // Note that RegisterRtpDemuxerSink first unregisters the sink if
+    // already registered. So this will change the state of the class
+    // whether the call succeeds or not.
+    RtpDemuxerCriteria criteria = demuxer_criteria();
+    return rtp_transport_->RegisterRtpDemuxerSink(criteria, this);
+  });
 
   media_receive_channel()->OnDemuxerCriteriaUpdateComplete();
 
   return ret;
 }
 
+RtpDemuxerCriteria BaseChannel::demuxer_criteria() const {
+  RTC_DCHECK_RUN_ON(network_thread());
+  RtpDemuxerCriteria criteria(mid_);
+  if (payload_type_demuxing_enabled_) {
+    criteria.payload_types() = payload_types_;
+  }
+  criteria.ssrcs() = ssrcs_;
+  return criteria;
+}
+
 void BaseChannel::EnableMedia_w() {
   if (enabled_)
     return;
@@ -702,8 +714,6 @@ bool BaseChannel::SetPayloadTypeDemuxingEnabled_w(bool enabled) {
 
   payload_type_demuxing_enabled_ = enabled;
 
-  bool config_changed = false;
-
   if (!enabled) {
     // TODO(crbug.com/11477): This will remove *all* unsignaled streams (those
     // without an explicitly signaled SSRC), which may include streams that
@@ -711,27 +721,16 @@ bool BaseChannel::SetPayloadTypeDemuxingEnabled_w(bool enabled) {
     // streams that were matched based on payload type alone, but currently
     // there is no straightforward way to identify those streams.
     media_receive_channel()->ResetUnsignaledRecvStream();
-    if (!demuxer_criteria_.payload_types().empty()) {
-      config_changed = true;
-      demuxer_criteria_.payload_types().clear();
+    if (!payload_types_.empty()) {
+      return RegisterRtpDemuxerSink_w();
     }
   } else if (!payload_types_.empty()) {
-    for (const auto& type : payload_types_) {
-      if (demuxer_criteria_.payload_types().insert(type).second) {
-        config_changed = true;
-      }
-    }
-  } else {
-    RTC_DCHECK(demuxer_criteria_.payload_types().empty());
+    return RegisterRtpDemuxerSink_w();
   }
 
   RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(0);
 
-  if (!config_changed)
-    return true;
-
-  // Note: This synchronously hops to the network thread.
-  return RegisterRtpDemuxerSink_w();
+  return true;
 }
 
 bool BaseChannel::UpdateLocalStreams_w(const std::vector<StreamParams>& streams,
@@ -881,8 +880,8 @@ bool BaseChannel::UpdateRemoteStreams_w(const MediaContentDescription* content,
     ssrcs.insert(new_stream.ssrcs.begin(), new_stream.ssrcs.end());
   }
 
-  if (demuxer_criteria_.ssrcs() != ssrcs) {
-    demuxer_criteria_.ssrcs() = std::move(ssrcs);
+  if (ssrcs_ != ssrcs) {
+    ssrcs_ = std::move(ssrcs);
     needs_re_registration = true;
   }
 
@@ -912,23 +911,17 @@ RtpHeaderExtensions BaseChannel::GetDeduplicatedRtpHeaderExtensions(
 }
 
 bool BaseChannel::MaybeAddHandledPayloadType(int payload_type) {
-  bool demuxer_criteria_modified = false;
-  if (payload_type_demuxing_enabled_) {
-    demuxer_criteria_modified = demuxer_criteria_.payload_types()
-                                    .insert(static_cast<uint8_t>(payload_type))
-                                    .second;
-  }
   // Even if payload type demuxing is currently disabled, we need to remember
   // the payload types in case it's re-enabled later.
-  payload_types_.insert(static_cast<uint8_t>(payload_type));
-  return demuxer_criteria_modified;
+  bool inserted =
+      payload_types_.insert(static_cast<uint8_t>(payload_type)).second;
+  return inserted && payload_type_demuxing_enabled_;
 }
 
 bool BaseChannel::ClearHandledPayloadTypes() {
-  const bool was_empty = demuxer_criteria_.payload_types().empty();
-  demuxer_criteria_.payload_types().clear();
+  const bool was_empty = payload_types_.empty();
   payload_types_.clear();
-  return !was_empty;
+  return !was_empty && payload_type_demuxing_enabled_;
 }
 
 bool BaseChannel::CheckRtpExtensionValidity(
@@ -1066,6 +1059,7 @@ bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content,
 
   if (type == SdpType::kAnswer || type == SdpType::kPrAnswer) {
     AudioSenderParameter send_params = last_send_params_;
+    RTC_DCHECK(!send_params.mid.empty());
     send_params.extensions = header_extensions;
     send_params.extmap_allow_mixed = content->extmap_allow_mixed();
     if (!media_send_channel()->SetSenderParameters(send_params)) {
diff --git a/pc/channel.h b/pc/channel.h
index d651d066a9..8a30c46490 100644
--- a/pc/channel.h
+++ b/pc/channel.h
@@ -95,7 +95,7 @@ class BaseChannel : public ChannelInterface,
 
   TaskQueueBase* worker_thread() const { return worker_thread_; }
   Thread* network_thread() const { return network_thread_; }
-  const std::string& mid() const override { return demuxer_criteria_.mid(); }
+  const std::string& mid() const override { return mid_; }
   // TODO(deadbeef): This is redundant; remove this.
   absl::string_view transport_name() const override {
     RTC_DCHECK_RUN_ON(network_thread());
@@ -306,6 +306,9 @@ class BaseChannel : public ChannelInterface,
       const RtpHeaderExtensions& extensions,
       std::string& error_desc) RTC_RUN_ON(worker_thread());
 
+  // Registers a demuxer criteria with the transport, on the network thread.
+  // This function will fail if there's no transport of if a sink is already
+  // registered for this channel's demuxer_critera().
   bool RegisterRtpDemuxerSink_w() RTC_RUN_ON(worker_thread());
 
   // Return description of media channel to facilitate logging
@@ -319,6 +322,11 @@ class BaseChannel : public ChannelInterface,
       RTC_RUN_ON(network_thread());
   void DisconnectFromRtpTransport_n() RTC_RUN_ON(network_thread());
   void SignalSentPacket_n(const SentPacketInfo& sent_packet);
+  // Only called on the network thread, so should be marked as
+  // `RTC_RUN_ON(network_thread());` but currently accesses variables
+  // that are associated with the worker thread. Those operations are
+  // done in a synchronous way, so they're safe, but not ideal.
+  RtpDemuxerCriteria demuxer_criteria() const RTC_NO_THREAD_SAFETY_ANALYSIS;
 
   TaskQueueBase* const worker_thread_;
   Thread* const network_thread_;
@@ -374,9 +382,8 @@ class BaseChannel : public ChannelInterface,
   RtpHeaderExtensions historical_rtp_header_extensions_
       RTC_GUARDED_BY(worker_thread());
 
-  // TODO(bugs.webrtc.org/12239): Modified on worker thread, accessed
-  // on network thread in RegisterRtpDemuxerSink_n (called from Init_w)
-  RtpDemuxerCriteria demuxer_criteria_;
+  const std::string mid_;
+  flat_set<uint32_t> ssrcs_ RTC_GUARDED_BY(worker_thread());
   // This generator is used to generate SSRCs for local streams.
   // This is needed in cases where SSRCs are not negotiated or set explicitly
   // like in Simulcast.

From 2d4de9fc657f4450923805f7ed0acca77d32995b Mon Sep 17 00:00:00 2001
From: Philipp Hancke <philipp.hancke@googlemail.com>
Date: Thu, 12 Mar 2026 13:34:34 +0100
Subject: [PATCH 292/472] build: clean up api:enable_media deps
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

which requires fixing a transitive dependency

Bug: webrtc:467294026
Change-Id: Ia227d593de825efe84ac2a93522d1b6977112df9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456142
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#47145}
---
 api/BUILD.gn | 2 --
 pc/BUILD.gn  | 1 +
 2 files changed, 1 insertion(+), 2 deletions(-)

diff --git a/api/BUILD.gn b/api/BUILD.gn
index 9df13cc56d..d7587e6fef 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -26,7 +26,6 @@ rtc_library("enable_media") {
   deps = [
     ":peer_connection_interface",
     ":scoped_refptr",
-    "../call",
     "../call:call_interfaces",
     "../media:media_engine",
     "../media:rtc_audio_video",
@@ -76,7 +75,6 @@ rtc_library("create_modular_peer_connection_factory") {
     "../pc:peer_connection_factory_proxy",
     "../rtc_base:threading",
     "../rtc_base/system:rtc_export",
-    "../stats:rtc_stats",
   ]
 }
 
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 4315b46bb2..03fabe9df5 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -1065,6 +1065,7 @@ rtc_library("rtc_stats_traversal") {
     "../api:rtc_stats_api",
     "../api:scoped_refptr",
     "../rtc_base:checks",
+    "../stats:rtc_stats",
   ]
 }
 

From c6d1a824c269da34beba528e57f9d17f1222dfcc Mon Sep 17 00:00:00 2001
From: Per K <perkj@webrtc.org>
Date: Thu, 12 Mar 2026 20:33:30 +0000
Subject: [PATCH 293/472] Increase MSS to 1280bytes to increase BWE ramp up
 speed
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This increase how much reference window can grow per update.
It also matches the WebRTC MTU size.


Bug: webrtc:447037083
Change-Id: I7a32915213a2c26e18dbc387e10aa0f9ee6693a4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456246
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47146}
---
 .../scream/scream_network_controller.cc          |  3 +++
 .../scream/scream_network_controller_unittest.cc |  4 ++--
 .../scream/scream_v2_parameters.cc               |  2 +-
 .../scream/scream_v2_unittest.cc                 |  4 ++--
 .../bwe_integration_tests/scream_test.cc         | 16 +++++-----------
 5 files changed, 13 insertions(+), 16 deletions(-)

diff --git a/modules/congestion_controller/scream/scream_network_controller.cc b/modules/congestion_controller/scream/scream_network_controller.cc
index d50ccc4d5d..396c2be21f 100644
--- a/modules/congestion_controller/scream/scream_network_controller.cc
+++ b/modules/congestion_controller/scream/scream_network_controller.cc
@@ -44,6 +44,9 @@ ScreamNetworkController::ScreamNetworkController(NetworkControllerConfig config)
       starting_rate_(
           config.constraints.starting_rate.value_or(kDefaultStartRate)),
       streams_config_(config.stream_based_config),
+      max_seen_total_allocated_bitrate_(
+          config.stream_based_config.max_total_allocated_bitrate.value_or(
+              DataRate::Zero())),
       last_padding_interval_started_(Timestamp::Zero()) {
   UpdateScreamTargetBitrateConstraints();
 }
diff --git a/modules/congestion_controller/scream/scream_network_controller_unittest.cc b/modules/congestion_controller/scream/scream_network_controller_unittest.cc
index 745f70d79d..1a9dcabf15 100644
--- a/modules/congestion_controller/scream/scream_network_controller_unittest.cc
+++ b/modules/congestion_controller/scream/scream_network_controller_unittest.cc
@@ -449,10 +449,10 @@ TEST(ScreamControllerTest, PeriodicallyAllowPadding) {
   TimeDelta padding_duration = result_1.padding_stop - result_1.padding_start;
   TimeDelta time_between_padding =
       result_2.padding_start - result_1.padding_stop;
-  EXPECT_GT(padding_duration, TimeDelta::Millis(2800));
+  EXPECT_GT(padding_duration, TimeDelta::Millis(2500));
   EXPECT_LT(padding_duration, TimeDelta::Millis(3100));
 
-  EXPECT_GT(time_between_padding, TimeDelta::Millis(2900));
+  EXPECT_GT(time_between_padding, TimeDelta::Millis(2500));
   EXPECT_LT(time_between_padding, TimeDelta::Millis(3300));
 }
 
diff --git a/modules/congestion_controller/scream/scream_v2_parameters.cc b/modules/congestion_controller/scream/scream_v2_parameters.cc
index 63c01c450a..7c70de148d 100644
--- a/modules/congestion_controller/scream/scream_v2_parameters.cc
+++ b/modules/congestion_controller/scream/scream_v2_parameters.cc
@@ -23,7 +23,7 @@ ScreamV2Parameters::ScreamV2Parameters(const FieldTrialsView& trials)
       l4s_avg_g_down("L4sAvgGDown", 1.0 / 128.0),
       smoothed_rtt_avg_g_up("SmoothedRttAvgGUp", 1.0 / 8.0),
       smoothed_rtt_avg_g_down("SmoothedRttAvgGDown", 1.0 / 8.0),
-      max_segment_size("MaxSegmentSize", DataSize::Bytes(1000)),
+      max_segment_size("MaxSegmentSize", DataSize::Bytes(1280)),
       bytes_in_flight_head_room("BytesInFlightHeadRoom", 1.1),
       beta_loss("BetaLoss", 0.7),
       post_congestion_delay_rtts("PostCongestionDelayRtts", 100),
diff --git a/modules/congestion_controller/scream/scream_v2_unittest.cc b/modules/congestion_controller/scream/scream_v2_unittest.cc
index df15e48886..84ccf60607 100644
--- a/modules/congestion_controller/scream/scream_v2_unittest.cc
+++ b/modules/congestion_controller/scream/scream_v2_unittest.cc
@@ -180,9 +180,9 @@ TEST(ScreamV2Test, ReferenceWindowIncreaseToDataInflight) {
     clock.AdvanceTime(feedback_interval);
   }
   // Target rate can increase up to 1.1 * data_in_flight + Max Segment Size(
-  // default 1000 bytes) when no max target rate has been set.
+  // default 1280 bytes) when no max target rate has been set.
   EXPECT_EQ(scream.ref_window(),
-            1.1 * feedback.data_in_flight + DataSize::Bytes(1000));
+            1.1 * feedback.data_in_flight + DataSize::Bytes(1280));
 }
 
 TEST(ScreamV2Test, CalculatesL4sAlpha) {
diff --git a/test/peer_scenario/bwe_integration_tests/scream_test.cc b/test/peer_scenario/bwe_integration_tests/scream_test.cc
index 8ed6a436b0..0cf1d13812 100644
--- a/test/peer_scenario/bwe_integration_tests/scream_test.cc
+++ b/test/peer_scenario/bwe_integration_tests/scream_test.cc
@@ -337,7 +337,7 @@ TEST(
   EXPECT_THAT(
       result.caller().subspan(1, 3),
       Each(AvailableSendBitrateIsBetween(DataRate::KilobitsPerSec(10),
-                                         DataRate::KilobitsPerSec(120))));
+                                         DataRate::KilobitsPerSec(140))));
   EXPECT_THAT(result.caller()[3],
               CurrentRoundTripTimeIsBetween(TimeDelta::Millis(40),
                                             TimeDelta::Millis(200)));
@@ -698,16 +698,10 @@ TEST(ScreamTest, MaybeTest(RampupFastOnLinkCapacity50Mbit20MsRttNoEcn)) {
                         DataRate::KilobitsPerSec(50000), TimeDelta::Millis(10));
   SendMediaTestResult result = SendMediaInOneDirection(std::move(params), s);
 
-  // TODO: bugs.webrtc.org/447037083 - On a good network, we should be able to
-  // let the bitrate increase by 50% every RTT.
-  // After 300ms.
-  EXPECT_THAT(result.caller_stats[2],
-              AvailableSendBitrateIsBetween(DataRate::KilobitsPerSec(500),
-                                            DataRate::KilobitsPerSec(5000)));
-  // After 1s.
-  EXPECT_THAT(result.caller_stats.back(),
-              AvailableSendBitrateIsBetween(DataRate::KilobitsPerSec(1500),
-                                            DataRate::KilobitsPerSec(5000)));
+  // After 400ms.
+  EXPECT_THAT(result.caller_stats[3],
+              AvailableSendBitrateIsBetween(DataRate::KilobitsPerSec(1200),
+                                            DataRate::KilobitsPerSec(2500)));
 }
 
 TEST(ScreamTest, MaybeTest(LinkCapacity100Kbit50msRttNoEcn)) {

From ea1fce97e610d7c0323a01faf271506af96b6c01 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Fri, 13 Mar 2026 01:58:40 -0700
Subject: [PATCH 294/472] Roll chromium_revision b937d39273..8f11ead6b3
 (1597657:1598931)

Change log: https://chromium.googlesource.com/chromium/src/+log/b937d39273..8f11ead6b3
Full diff: https://chromium.googlesource.com/chromium/src/+/b937d39273..8f11ead6b3

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/7d29c62736..618025ac18
* src/buildtools: https://chromium.googlesource.com/chromium/src/buildtools/+log/fef8d6947f..95ed44cf5f
* src/buildtools/linux64: git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9..git_revision:2941d2a26fa9cef3103f10e1dee5da7cd404008a
* src/buildtools/mac: git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9..git_revision:2941d2a26fa9cef3103f10e1dee5da7cd404008a
* src/buildtools/win: git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9..git_revision:2941d2a26fa9cef3103f10e1dee5da7cd404008a
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/94f693fcb6..72d8d02177
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/e7a1e88ecc..92de5f9238
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/a133c94b51..4b35a84f2e
* src/third_party/android_build_tools/aapt2/cipd: vUP1cs7krVsNHbUJpwSDR7rvWuYLGzINVa9xopPpt74C..gsaUgZUqoyD0XG1E9-xesSWknHZINEuS00iSEncvlE0C
* src/third_party/android_build_tools/lint/cipd: 1N_9yErKkcZgqyxBghA0bb5L5AFEPZCt36Gv2o8_OAwC..UdJPLkDOu5joGtPdwsjc1Ltg5hcadLKPynhbd6NacsoC
* src/third_party/android_build_tools/manifest_merger/cipd: xT8A6Q_b44_iYKqAMvRkWmvID-ps-I6uBDfRTQv68bcC..KYQSw0Ej8zT_IRNL02DxEmk0nCvkykopkQhsGk_HE0EC
* src/third_party/android_deps/autorolled/cipd: umaYTBt4Vsa5QZYp57rLJDNzoXalsR2hPGwfYbpZYfwC..3VYcqPssBf1YI1iK0N2UUMsT2Ik5BlMrvm-wOVr1fMQC
* src/third_party/androidx/cipd: Hi79Lu6qQ-Fbcjwgf_f-NEYJweP4QdZqjJ0lXwc0PpcC..Mn6UxK4fOcnps2oJ1ikX7T1Nf93dtMBkX68uqIyoQ0sC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/eedf10c1f4..de8c7af5e7
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/e0ebf38a01..b9ddc08533
* src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/c8a5676846..c8ee656688
* src/third_party/cpuinfo/src: https://chromium.googlesource.com/external/github.com/pytorch/cpuinfo.git/+log/7364b490b5..7607ca5004
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/4ce8ba39a3..6b62257bd9
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/4387e32481..2a2456c873
* src/third_party/fuzztest/src: https://chromium.googlesource.com/external/github.com/google/fuzztest.git/+log/54ed362692..41e160679e
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/260c6fd37c..9751f44f3f
* src/third_party/harfbuzz-ng/src: https://chromium.googlesource.com/external/github.com/harfbuzz/harfbuzz.git/+log/fa2908bf16..5d4e96ad8d
* src/third_party/libaom/source/libaom: https://aomedia.googlesource.com/aom.git/+log/98ce0d2a61..0c15af06af
* src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/044963a253..76c8a4be56
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/4cead1b0fc..15e97d25a9
* src/third_party/r8/cipd: CoaQUbD7pDX3OuETi_O5wkv_Y198RVIUaSobSEx9zfoC..UMLn7pE7DklTsk-SIiMAwch46m8rSF1vG2HQAbMuZvMC
* src/third_party/tflite/src: https://chromium.googlesource.com/external/github.com/tensorflow/tensorflow.git/+log/f920077085..d20369f4f2
* src/third_party/xnnpack/src: https://chromium.googlesource.com/external/github.com/google/XNNPACK.git/+log/26feacc3eb..15e04ff8b6
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/9765576870..815b983a1c
* src/tools/luci-go: git_revision:eefaa865c43930859313a63d426740cdd0e22673..git_revision:09ad27bb7a42580550b9063ef8d0a39a5f01a338
* src/tools/luci-go: git_revision:eefaa865c43930859313a63d426740cdd0e22673..git_revision:09ad27bb7a42580550b9063ef8d0a39a5f01a338
* src/tools/luci-go: git_revision:eefaa865c43930859313a63d426740cdd0e22673..git_revision:09ad27bb7a42580550b9063ef8d0a39a5f01a338
DEPS diff: https://chromium.googlesource.com/chromium/src/+/b937d39273..8f11ead6b3/DEPS

BUG=None

Change-Id: I2f18b9bc0a8067e6c283cf3805b04a140ac923bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456440
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47147}
---
 DEPS | 66 ++++++++++++++++++++++++++++++------------------------------
 1 file changed, 33 insertions(+), 33 deletions(-)

diff --git a/DEPS b/DEPS
index dffcecaef3..12bd7209dd 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': 'b937d39273b52034e8542dea4b040587be0a127f',
+  'chromium_revision': '8f11ead6b30adfcb8eb5f504d92c596adbe1a1e5',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,28 +69,28 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@7d29c62736304aeaac07df0b75546ffd2b75917a',
+    'https://chromium.googlesource.com/chromium/src/build@618025ac18566ba090c4b86328ea77f83a6c4cae',
   'src/buildtools':
-    'https://chromium.googlesource.com/chromium/src/buildtools@fef8d6947fa54202e8a00c58d5ac8fd15780bc33',
+    'https://chromium.googlesource.com/chromium/src/buildtools@95ed44cf5f06dbb5861030b91c9db9ccb4316762',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
   'src/examples/androidtests/third_party/gradle': {
     'url': 'https://chromium.googlesource.com/external/github.com/gradle/gradle.git@f2d1fb54a951d8b11d25748e4711bec8d128d7e3',
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@94f693fcb6dde967f11668a9e21eec1a4ec8b49c',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@72d8d02177f725c05690fd05c52f5d9779e7bf71',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@e7a1e88ecc35a70f3d198674042753ec9e8e3650',
+    'https://chromium.googlesource.com/chromium/src/testing@92de5f92381b699b0b3139e2a25b6bd7beb85253',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@a133c94b51ac034d5f375cc1746e66dcdee57282',
+    'https://chromium.googlesource.com/chromium/src/third_party@4b35a84f2e22d8fc094fcb615c12f2288706adc6',
 
   'src/buildtools/linux64': {
     'packages': [
       {
         'package': 'gn/gn/linux-${{arch}}',
-        'version': 'git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9',
+        'version': 'git_revision:2941d2a26fa9cef3103f10e1dee5da7cd404008a',
       }
     ],
     'dep_type': 'cipd',
@@ -100,7 +100,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/mac-${{arch}}',
-        'version': 'git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9',
+        'version': 'git_revision:2941d2a26fa9cef3103f10e1dee5da7cd404008a',
       }
     ],
     'dep_type': 'cipd',
@@ -110,7 +110,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/windows-amd64',
-        'version': 'git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9',
+        'version': 'git_revision:2941d2a26fa9cef3103f10e1dee5da7cd404008a',
       }
     ],
     'dep_type': 'cipd',
@@ -331,13 +331,13 @@ deps = {
   'src/third_party/clang-format/script':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@c2725e0622e1a86d55f14514f2177a39efea4a0e',
   'src/third_party/compiler-rt/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@c8a567684659fe05ede3a220965b94fd71c4a7cb',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@c8ee6566881410a159d66e99081779d28124fd3f',
   'src/third_party/libc++/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@7ab65651aed6802d2599dcb7a73b1f82d5179d05',
   'src/third_party/libc++abi/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@8f11bb1d4438d0239d0dfc1bd9456a9f31629dda',
   'src/third_party/llvm-libc/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@044963a253be3b8b2a63a1b2e0f3e361b01bc186',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@76c8a4be56babbe6104f5cd9eda25a95dd277b7a',
   'src/third_party/libunwind/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@db838d918570d4e381ecf9f5cc70a0098c9c2cd6',
 
@@ -402,7 +402,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/android_build_tools/aapt2',
-              'version': 'vUP1cs7krVsNHbUJpwSDR7rvWuYLGzINVa9xopPpt74C',
+              'version': 'gsaUgZUqoyD0XG1E9-xesSWknHZINEuS00iSEncvlE0C',
           },
       ],
       'condition': 'checkout_android',
@@ -457,7 +457,7 @@ deps = {
       'packages': [
           {
                'package': 'chromium/third_party/android_build_tools/lint',
-               'version': '1N_9yErKkcZgqyxBghA0bb5L5AFEPZCt36Gv2o8_OAwC',
+               'version': 'UdJPLkDOu5joGtPdwsjc1Ltg5hcadLKPynhbd6NacsoC',
           },
       ],
       'condition': 'checkout_android and non_git_source',
@@ -488,11 +488,11 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@eedf10c1f4ab44629e2fe3fe6f07e2fb0897c0f4',
+    'https://boringssl.googlesource.com/boringssl.git@de8c7af5e7c40153eb6cc0a79ff61a43f4bb03f9',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@8be0e3114685fcc1589561067282edf75ea1259a',
   'src/third_party/catapult':
-    'https://chromium.googlesource.com/catapult.git@e0ebf38a01214aba11f31daa1c743782def031d5',
+    'https://chromium.googlesource.com/catapult.git@b9ddc08533aca2b2878bc05f2cce8e3051616313',
   'src/third_party/ced/src': {
     'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5',
   },
@@ -505,13 +505,13 @@ deps = {
   'src/third_party/crc32c/src':
     'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558',
   'src/third_party/depot_tools':
-    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@4ce8ba39a3488397a2d1494f167020f21de502f3',
+    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@6b62257bd9fc9ee0e00d10712a0f3db63132ba2f',
   'src/third_party/ffmpeg':
     'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@946d97db8d906277085e361892b7efda5152e2f1',
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@260c6fd37c502e984836a446c5d90e4a1298036d',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@9751f44f3f6640142d242ddac084f7a80dce4aac',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -521,7 +521,7 @@ deps = {
   'src/third_party/freetype/src':
     'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@99b479dc34728936b006679a31e12b8cf432fc55',
   'src/third_party/harfbuzz-ng/src':
-    'https://chromium.googlesource.com/external/github.com/harfbuzz/harfbuzz.git@fa2908bf16d2ccd6623f4d575455fea72a1a722b',
+    'https://chromium.googlesource.com/external/github.com/harfbuzz/harfbuzz.git@5d4e96ad8d00fc871ffa17707b2ca08fa850e7d6',
   'src/third_party/google_benchmark/src': {
     'url': 'https://chromium.googlesource.com/external/github.com/google/benchmark.git@188e8278990a9069ffc84441cb5a024fd0bede37',
   },
@@ -593,7 +593,7 @@ deps = {
   'src/third_party/libFuzzer/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt/lib/fuzzer.git@bea408a6e01f0f7e6c82a43121fe3af4506c932e',
   'src/third_party/fuzztest/src':
-    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@54ed362692177e93df0fa202cae404b57694e828',
+    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@41e160679e3c010a860e674e7997e2b285c6e0a2',
   'src/third_party/libjpeg_turbo':
     'https://chromium.googlesource.com/chromium/deps/libjpeg_turbo.git@d1f5f2393e0d51f840207342ae86e55a86443288',
   'src/third_party/libsrtp':
@@ -601,7 +601,7 @@ deps = {
   'src/third_party/dav1d/libdav1d':
     'https://chromium.googlesource.com/external/github.com/videolan/dav1d.git@241a6b236a9c84441c872119c1f31b4074c0589d',
   'src/third_party/libaom/source/libaom':
-    'https://aomedia.googlesource.com/aom.git@98ce0d2a610f1465ea14d525bce9fa46d4b05c03',
+    'https://aomedia.googlesource.com/aom.git@0c15af06af108af03cbe237e4bffa4b700b76d81',
   'src/third_party/libgav1/src':
     Var('chromium_git') + '/codecs/libgav1.git' + '@' + '40f58ed32ff39071c3f2a51056dbc49a070af0dc',
   'src/third_party/libunwindstack': {
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '4cead1b0fc9987f994de97c7914e1f0c94047579',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '15e97d25a95aed5a96c92553e254b7e491bf7369',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -644,7 +644,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/r8',
-              'version': 'CoaQUbD7pDX3OuETi_O5wkv_Y198RVIUaSobSEx9zfoC',
+              'version': 'UMLn7pE7DklTsk-SIiMAwch46m8rSF1vG2HQAbMuZvMC',
           },
       ],
       'condition': 'checkout_android',
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@9765576870401b230a826e0e4f7a12851575306e',
+    'https://chromium.googlesource.com/chromium/src/tools@815b983a1c95803681d04b28e82b22a358f078b4',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'Hi79Lu6qQ-Fbcjwgf_f-NEYJweP4QdZqjJ0lXwc0PpcC',
+          'version': 'Mn6UxK4fOcnps2oJ1ikX7T1Nf93dtMBkX68uqIyoQ0sC',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -718,7 +718,7 @@ deps = {
       'packages': [
           {
                'package': 'chromium/third_party/android_build_tools/manifest_merger',
-               'version': 'xT8A6Q_b44_iYKqAMvRkWmvID-ps-I6uBDfRTQv68bcC',
+               'version': 'KYQSw0Ej8zT_IRNL02DxEmk0nCvkykopkQhsGk_HE0EC',
           },
       ],
       'condition': 'checkout_android',
@@ -786,7 +786,7 @@ deps = {
   },
 
   'src/third_party/tflite/src':
-    Var('chromium_git') + '/external/github.com/tensorflow/tensorflow.git' + '@' + 'f9200770858f5721b7fa95db7aa3e4cd712fe9df',
+    Var('chromium_git') + '/external/github.com/tensorflow/tensorflow.git' + '@' + 'd20369f4f226598800e0d406db567d2d1c17fa95',
 
   'src/third_party/turbine/cipd': {
       'packages': [
@@ -808,15 +808,15 @@ deps = {
       'packages': [
         {
           'package': 'infra/tools/luci/cas/${{platform}}',
-          'version': 'git_revision:eefaa865c43930859313a63d426740cdd0e22673',
+          'version': 'git_revision:09ad27bb7a42580550b9063ef8d0a39a5f01a338',
         },
         {
           'package': 'infra/tools/luci/isolate/${{platform}}',
-          'version': 'git_revision:eefaa865c43930859313a63d426740cdd0e22673',
+          'version': 'git_revision:09ad27bb7a42580550b9063ef8d0a39a5f01a338',
         },
         {
           'package': 'infra/tools/luci/swarming/${{platform}}',
-          'version': 'git_revision:eefaa865c43930859313a63d426740cdd0e22673',
+          'version': 'git_revision:09ad27bb7a42580550b9063ef8d0a39a5f01a338',
         }
       ],
       'dep_type': 'cipd',
@@ -842,7 +842,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/android_deps/autorolled',
-              'version': 'umaYTBt4Vsa5QZYp57rLJDNzoXalsR2hPGwfYbpZYfwC',
+              'version': '3VYcqPssBf1YI1iK0N2UUMsT2Ik5BlMrvm-wOVr1fMQC',
           },
       ],
       'condition': 'checkout_android and non_git_source',
@@ -852,7 +852,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/google/pthreadpool.git' + '@' + '9003ee6c137cea3b94161bd5c614fb43be523ee1',
 
   'src/third_party/xnnpack/src':
-    Var('chromium_git') + '/external/github.com/google/XNNPACK.git' + '@' + '26feacc3ebe30d1f3a82e1ff3938ef9859d74640',
+    Var('chromium_git') + '/external/github.com/google/XNNPACK.git' + '@' + '15e04ff8b61e0bc2118bad268447be5b2075aa66',
 
   'src/third_party/farmhash/src':
     Var('chromium_git') + '/external/github.com/google/farmhash.git' + '@' + '816a4ae622e964763ca0862d9dbd19324a1eaf45',
@@ -861,10 +861,10 @@ deps = {
     Var('chromium_git') + '/external/github.com/google/ruy.git' + '@' + '2af88863614a8298689cc52b1a47b3fcad7be835',
 
   'src/third_party/cpuinfo/src':
-    Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7364b490b5f78d58efe23ea76e74210fd6c3c76f',
+    Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7607ca500436b37ad23fb8d18614bec7796b68a7',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '4387e3248109bef2c07a8b69a2b8249d7af5dcf0',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '2a2456c873d6f4bf43a9adf919f13ad917d57c2d',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From e429887d22fd5f9c5910f0a50391e6046371a5eb Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Thu, 12 Mar 2026 12:00:51 +0100
Subject: [PATCH 295/472] Add encoder switch request callback to
 VideoMediaSendChannel

Introduce a callback mechanism to intercept and handle encoder switch
requests within the video media engine. This makes it so that codec
switch/fallback requests first go through the the RtpTransceiver on the
signaling thread. That is consistent with other parameter related
changes and allows the transceiver to update the sender object(s) before
the codec change is applied.

Specifically, the changes include:
* Define EncoderSwitchRequestCallback and EncoderSwitchRequestAction
  in VideoMediaSendChannelInterface.
* Updating WebRtcVideoSendChannel to optionally use the callback
  instead of posting directly to the worker thread when an encoder
  switch is requested.
* Implementing GetEncoderSwitchRequestCallback in RtpTransceiver
  to ensure that when a switch occurs, cached RTP parameters on the
  signaling thread are cleared before the switch action is executed
  on the worker thread.
* Updating RtpSender to support clearing cached parameters via
  std::optional.
* Adding RTC_DCHECK to SafeTask and SafeInvocable to catch if
  the provided safety flag is invalid, early.

Bug: b/478050997
Change-Id: Id81a219a0f3dab6a167248c21513723504fa6e79
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/454700
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sameer Vijaykar <samvi@google.com>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47148}
---
 api/task_queue/pending_task_safety_flag.h    |  3 +
 media/base/fake_media_engine.cc              |  4 +-
 media/base/fake_media_engine.h               |  4 +-
 media/base/media_channel.h                   |  4 ++
 media/base/media_engine.cc                   |  2 +-
 media/base/media_engine.h                    |  8 ++-
 media/engine/webrtc_video_engine.cc          | 36 +++++++----
 media/engine/webrtc_video_engine.h           | 16 ++++-
 media/engine/webrtc_video_engine_unittest.cc | 42 ++++++++++++
 pc/rtp_sender.cc                             | 16 +++--
 pc/rtp_sender.h                              |  4 +-
 pc/rtp_transceiver.cc                        | 68 +++++++++++++++++---
 pc/rtp_transceiver.h                         |  6 +-
 pc/test/mock_rtp_sender_internal.h           |  6 +-
 14 files changed, 179 insertions(+), 40 deletions(-)

diff --git a/api/task_queue/pending_task_safety_flag.h b/api/task_queue/pending_task_safety_flag.h
index 875976eede..a3c3cbd02e 100644
--- a/api/task_queue/pending_task_safety_flag.h
+++ b/api/task_queue/pending_task_safety_flag.h
@@ -19,6 +19,7 @@
 #include "api/scoped_refptr.h"
 #include "api/sequence_checker.h"
 #include "api/task_queue/task_queue_base.h"
+#include "rtc_base/checks.h"
 #include "rtc_base/system/no_unique_address.h"
 #include "rtc_base/system/rtc_export.h"
 
@@ -165,6 +166,7 @@ class RTC_EXPORT ScopedTaskSafetyDetached final {
 inline absl::AnyInvocable<void() &&> SafeTask(
     scoped_refptr<PendingTaskSafetyFlag> flag,
     absl::AnyInvocable<void() &&> task) {
+  RTC_DCHECK(flag);
   return [flag = std::move(flag), task = std::move(task)]() mutable {
     if (flag->alive()) {
       std::move(task)();
@@ -176,6 +178,7 @@ inline absl::AnyInvocable<void() &&> SafeTask(
 inline absl::AnyInvocable<void()> SafeInvocable(
     scoped_refptr<PendingTaskSafetyFlag> flag,
     absl::AnyInvocable<void()> task) {
+  RTC_DCHECK(flag);
   return [flag = std::move(flag), task = std::move(task)]() mutable {
     if (flag->alive()) {
       task();
diff --git a/media/base/fake_media_engine.cc b/media/base/fake_media_engine.cc
index a406b2a36c..cad3c587ed 100644
--- a/media/base/fake_media_engine.cc
+++ b/media/base/fake_media_engine.cc
@@ -654,7 +654,9 @@ FakeVideoEngine::CreateSendChannel(
     const MediaConfig& /* config */,
     const VideoOptions& options,
     const CryptoOptions& /* crypto_options */,
-    VideoBitrateAllocatorFactory* /* video_bitrate_allocator_factory */) {
+    VideoBitrateAllocatorFactory* /* video_bitrate_allocator_factory */,
+    VideoMediaSendChannelInterface::EncoderSwitchRequestCallback
+    /* video_encoder_switch_request_callback */) {
   std::unique_ptr<FakeVideoMediaSendChannel> ch =
       std::make_unique<FakeVideoMediaSendChannel>(options,
                                                   call->network_thread());
diff --git a/media/base/fake_media_engine.h b/media/base/fake_media_engine.h
index 6cd6e66641..4e5e43a5e3 100644
--- a/media/base/fake_media_engine.h
+++ b/media/base/fake_media_engine.h
@@ -886,7 +886,9 @@ class FakeVideoEngine : public VideoEngineInterface {
       const MediaConfig& config,
       const VideoOptions& options,
       const CryptoOptions& crypto_options,
-      VideoBitrateAllocatorFactory* video_bitrate_allocator_factory) override;
+      VideoBitrateAllocatorFactory* video_bitrate_allocator_factory,
+      VideoMediaSendChannelInterface::EncoderSwitchRequestCallback
+          video_encoder_switch_request_callback = nullptr) override;
   std::unique_ptr<VideoMediaReceiveChannelInterface> CreateReceiveChannel(
       const Environment& env,
       Call* call,
diff --git a/media/base/media_channel.h b/media/base/media_channel.h
index 93693ec740..cdc19280cc 100644
--- a/media/base/media_channel.h
+++ b/media/base/media_channel.h
@@ -960,6 +960,10 @@ struct VideoReceiverParameters : MediaChannelParameters {};
 
 class VideoMediaSendChannelInterface : public MediaSendChannelInterface {
  public:
+  using EncoderSwitchRequestAction = absl::AnyInvocable<void() &&>;
+  using EncoderSwitchRequestCallback =
+      absl::AnyInvocable<void(EncoderSwitchRequestAction)>;
+
   virtual bool SetSenderParameters(const VideoSenderParameters& params) = 0;
   // Starts or stops transmission (and potentially capture) of local video.
   virtual bool SetSend(bool send) = 0;
diff --git a/media/base/media_engine.cc b/media/base/media_engine.cc
index 52da64db52..78bb62af9b 100644
--- a/media/base/media_engine.cc
+++ b/media/base/media_engine.cc
@@ -79,7 +79,7 @@ RtpParameters CreateRtpParametersWithEncodings(StreamParams sp) {
 
 std::vector<RtpExtension> GetDefaultEnabledRtpHeaderExtensions(
     const RtpHeaderExtensionQueryInterface& query_interface,
-    const webrtc::FieldTrialsView* field_trials) {
+    const FieldTrialsView* field_trials) {
   std::vector<RtpExtension> extensions;
   for (const auto& entry :
        query_interface.GetRtpHeaderExtensions(field_trials)) {
diff --git a/media/base/media_engine.h b/media/base/media_engine.h
index 0ac3277d50..c85db96653 100644
--- a/media/base/media_engine.h
+++ b/media/base/media_engine.h
@@ -77,7 +77,7 @@ class RtpHeaderExtensionQueryInterface {
   // Returns a vector of RtpHeaderExtensionCapability, whose direction is
   // kStopped if the extension is stopped (not used) by default.
   virtual std::vector<RtpHeaderExtensionCapability> GetRtpHeaderExtensions(
-      const webrtc::FieldTrialsView* field_trials) const = 0;
+      const FieldTrialsView* field_trials) const = 0;
 };
 
 class VoiceEngineInterface : public RtpHeaderExtensionQueryInterface {
@@ -153,7 +153,9 @@ class VideoEngineInterface : public RtpHeaderExtensionQueryInterface {
       const MediaConfig& config,
       const VideoOptions& options,
       const CryptoOptions& crypto_options,
-      VideoBitrateAllocatorFactory* video_bitrate_allocator_factory) = 0;
+      VideoBitrateAllocatorFactory* video_bitrate_allocator_factory,
+      VideoMediaSendChannelInterface::EncoderSwitchRequestCallback
+          video_encoder_switch_request_callback = nullptr) = 0;
 
   virtual std::unique_ptr<VideoMediaReceiveChannelInterface>
   CreateReceiveChannel(const Environment& env,
@@ -240,7 +242,7 @@ RtpParameters CreateRtpParametersWithEncodings(StreamParams sp);
 // GetRtpHeaderExtensions() that are not kStopped.
 std::vector<RtpExtension> GetDefaultEnabledRtpHeaderExtensions(
     const RtpHeaderExtensionQueryInterface& query_interface,
-    const webrtc::FieldTrialsView* field_trials);
+    const FieldTrialsView* field_trials);
 
 }  //  namespace webrtc
 
diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc
index bb776670c9..91f2c9b333 100644
--- a/media/engine/webrtc_video_engine.cc
+++ b/media/engine/webrtc_video_engine.cc
@@ -829,10 +829,13 @@ WebRtcVideoEngine::CreateSendChannel(
     const MediaConfig& config,
     const VideoOptions& options,
     const CryptoOptions& crypto_options,
-    VideoBitrateAllocatorFactory* video_bitrate_allocator_factory) {
+    VideoBitrateAllocatorFactory* video_bitrate_allocator_factory,
+    VideoMediaSendChannelInterface::EncoderSwitchRequestCallback
+        video_encoder_switch_request_callback) {
   return std::make_unique<WebRtcVideoSendChannel>(
       env, call, config, options, crypto_options, encoder_factory_.get(),
-      video_bitrate_allocator_factory);
+      video_bitrate_allocator_factory,
+      std::move(video_encoder_switch_request_callback));
 }
 std::unique_ptr<VideoMediaReceiveChannelInterface>
 WebRtcVideoEngine::CreateReceiveChannel(const Environment& env,
@@ -920,7 +923,9 @@ WebRtcVideoSendChannel::WebRtcVideoSendChannel(
     const VideoOptions& options,
     const CryptoOptions& crypto_options,
     VideoEncoderFactory* encoder_factory,
-    VideoBitrateAllocatorFactory* bitrate_allocator_factory)
+    VideoBitrateAllocatorFactory* bitrate_allocator_factory,
+    VideoMediaSendChannelInterface::EncoderSwitchRequestCallback
+        video_encoder_switch_request_callback)
     : MediaChannelUtil(call->network_thread(), config.enable_dscp),
       env_(env),
       worker_thread_(call->worker_thread()),
@@ -931,7 +936,9 @@ WebRtcVideoSendChannel::WebRtcVideoSendChannel(
       bitrate_allocator_factory_(bitrate_allocator_factory),
       default_send_options_(options),
       last_send_stats_log_ms_(-1),
-      crypto_options_(crypto_options) {
+      crypto_options_(crypto_options),
+      encoder_switch_request_callback_(
+          std::move(video_encoder_switch_request_callback)) {
   RTC_DCHECK_RUN_ON(&thread_checker_);
 }
 
@@ -1242,14 +1249,21 @@ bool WebRtcVideoSendChannel::SetSenderParameters(
 void WebRtcVideoSendChannel::RequestEncoderSwitch(
     std::optional<SdpVideoFormat> format,
     bool allow_default_fallback) {
-  if (!worker_thread_->IsCurrent()) {
-    worker_thread_->PostTask(
-        SafeTask(task_safety_.flag(), [this, format, allow_default_fallback] {
-          RequestEncoderSwitch(std::move(format), allow_default_fallback);
-        }));
-    return;
+  auto task =
+      SafeTask(task_safety_.flag(), [this, format, allow_default_fallback] {
+        RTC_DCHECK_RUN_ON(&thread_checker_);
+        ApplyEncoderSwitch(std::move(format), allow_default_fallback);
+      });
+  if (encoder_switch_request_callback_) {
+    encoder_switch_request_callback_(std::move(task));
+  } else {
+    worker_thread_->PostTask(std::move(task));
   }
+}
 
+void WebRtcVideoSendChannel::ApplyEncoderSwitch(
+    std::optional<SdpVideoFormat> format,
+    bool allow_default_fallback) {
   RTC_DCHECK_RUN_ON(&thread_checker_);
   RTC_DCHECK(format.has_value() || allow_default_fallback);
 
@@ -1277,7 +1291,7 @@ void WebRtcVideoSendChannel::RequestEncoderSwitch(
                           << allow_default_fallback;
 
       if (allow_default_fallback) {
-        RequestEncoderSwitch(std::nullopt, true);
+        ApplyEncoderSwitch(std::nullopt, true);
       }
       return;
     }
diff --git a/media/engine/webrtc_video_engine.h b/media/engine/webrtc_video_engine.h
index e6b4b23bdb..b28211a41f 100644
--- a/media/engine/webrtc_video_engine.h
+++ b/media/engine/webrtc_video_engine.h
@@ -106,7 +106,9 @@ class WebRtcVideoEngine : public VideoEngineInterface {
       const MediaConfig& config,
       const VideoOptions& options,
       const CryptoOptions& crypto_options,
-      VideoBitrateAllocatorFactory* video_bitrate_allocator_factory) override;
+      VideoBitrateAllocatorFactory* video_bitrate_allocator_factory,
+      VideoMediaSendChannelInterface::EncoderSwitchRequestCallback
+          video_encoder_switch_request_callback = nullptr) override;
   std::unique_ptr<VideoMediaReceiveChannelInterface> CreateReceiveChannel(
       const Environment& env,
       Call* call,
@@ -167,7 +169,9 @@ class WebRtcVideoSendChannel : public MediaChannelUtil,
       const VideoOptions& options,
       const CryptoOptions& crypto_options,
       VideoEncoderFactory* encoder_factory,
-      VideoBitrateAllocatorFactory* bitrate_allocator_factory);
+      VideoBitrateAllocatorFactory* bitrate_allocator_factory,
+      VideoMediaSendChannelInterface::EncoderSwitchRequestCallback
+          video_encoder_switch_request_callback);
   ~WebRtcVideoSendChannel() override;
 
   MediaType media_type() const override { return MediaType::VIDEO; }
@@ -291,6 +295,10 @@ class WebRtcVideoSendChannel : public MediaChannelUtil,
   bool ApplyChangedParams(const ChangedSenderParameters& changed_params);
   bool ValidateSendSsrcAvailability(const StreamParams& sp) const
       RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
+  // Executes the actual encoder switch operation.
+  void ApplyEncoderSwitch(std::optional<SdpVideoFormat> format,
+                          bool allow_default_fallback)
+      RTC_RUN_ON(thread_checker_);
 
   // Wrapper for the sender part.
   class WebRtcVideoSendStream {
@@ -476,6 +484,10 @@ class WebRtcVideoSendChannel : public MediaChannelUtil,
   // Callback invoked whenever the sender parameters change.
   absl::AnyInvocable<void()> parameters_changed_callback_
       RTC_GUARDED_BY(&thread_checker_);
+
+  // Callback to intercept the encoder switch request.
+  VideoMediaSendChannelInterface::EncoderSwitchRequestCallback
+      encoder_switch_request_callback_;
 };
 
 class WebRtcVideoReceiveChannel : public MediaChannelUtil,
diff --git a/media/engine/webrtc_video_engine_unittest.cc b/media/engine/webrtc_video_engine_unittest.cc
index d6f9d3872e..fdd105de71 100644
--- a/media/engine/webrtc_video_engine_unittest.cc
+++ b/media/engine/webrtc_video_engine_unittest.cc
@@ -115,6 +115,7 @@
 #include "rtc_base/experiments/min_video_bitrate_experiment.h"
 #include "rtc_base/numerics/safe_conversions.h"
 #include "rtc_base/socket.h"
+#include "rtc_base/thread.h"
 #include "test/create_test_environment.h"
 #include "test/create_test_field_trials.h"
 #include "test/fake_decoder.h"
@@ -835,6 +836,47 @@ TEST_F(WebRtcVideoEngineTest, CanConstructDecoderForVp9EncoderFactory) {
 }
 #endif  // defined(RTC_ENABLE_VP9)
 
+TEST_F(WebRtcVideoEngineTest, EncoderSwitchRequestCallback) {
+  AddSupportedVideoCodecType("VP8");
+
+  bool callback_called = false;
+  auto callback =
+      [&callback_called, call = call_.get(), main_thread = Thread::Current()](
+          VideoMediaSendChannelInterface::EncoderSwitchRequestAction action) {
+        call->worker_thread()->PostTask([action = std::move(action),
+                                         main_thread,
+                                         &callback_called]() mutable {
+          std::move(action)();
+          main_thread->PostTask([&callback_called] { callback_called = true; });
+        });
+      };
+
+  std::unique_ptr<VideoMediaSendChannelInterface> send_channel =
+      engine_->CreateSendChannel(
+          env_, call_.get(), GetMediaConfig(), VideoOptions(), CryptoOptions(),
+          video_bitrate_allocator_factory_.get(), std::move(callback));
+
+  VideoSenderParameters parameters;
+  parameters.codecs.push_back(GetEngineCodec("VP8"));
+  EXPECT_TRUE(send_channel->SetSenderParameters(parameters));
+
+  send_channel->OnReadyToSend(true);
+  EXPECT_TRUE(send_channel->AddSendStream(StreamParams::CreateLegacy(kSsrc)));
+  EXPECT_TRUE(send_channel->SetSend(true));
+
+  auto* web_rtc_send_channel =
+      static_cast<WebRtcVideoSendChannel*>(send_channel.get());
+
+  std::unique_ptr<Thread> bg_thread = Thread::Create();
+  bg_thread->Start();
+  bg_thread->BlockingCall(
+      [&] { web_rtc_send_channel->RequestEncoderSwitch(std::nullopt, true); });
+
+  EXPECT_TRUE(time_controller_.Wait([&] { return callback_called; }));
+
+  EXPECT_TRUE(send_channel->RemoveSendStream(kSsrc));
+}
+
 TEST_F(WebRtcVideoEngineTest, PropagatesInputFrameTimestamp) {
   AddSupportedVideoCodecType("VP8");
   FakeCall* fake_call = new FakeCall(env_);
diff --git a/pc/rtp_sender.cc b/pc/rtp_sender.cc
index ca3b131fb3..3a4e6b778f 100644
--- a/pc/rtp_sender.cc
+++ b/pc/rtp_sender.cc
@@ -261,9 +261,15 @@ void RtpSenderBase::SetEncoderSelectorOnChannel() {
   });
 }
 
-void RtpSenderBase::SetCachedParameters(RtpParameters parameters) {
+void RtpSenderBase::SetCachedParameters(
+    std::optional<RtpParameters> parameters) {
   RTC_DCHECK_RUN_ON(signaling_thread_);
-  cached_parameters_ = std::move(parameters);
+  if (parameters.has_value()) {
+    cached_parameters_ = std::move(*parameters);
+  } else {
+    cached_parameters_.reset();
+    last_transaction_id_.reset();
+  }
 }
 
 void RtpSenderBase::SetMediaChannel(MediaSendChannelInterface* media_channel) {
@@ -383,10 +389,8 @@ RtpParameters RtpSenderBase::GetParameters() const {
           << "Cached: " << cached_filtered << "\n"
           << "Result: " << result;
     }
-    // TODO(b/478050997): Re-enable this check once the downstream issue is
-    // resolved.
-    // RTC_DCHECK(cached_filtered == result)
-    //    << "The cached value should have been equal (filtered)";
+    RTC_DCHECK(cached_filtered == result)
+        << "The cached value should have been equal (filtered)";
   }
 #endif
   return result;
diff --git a/pc/rtp_sender.h b/pc/rtp_sender.h
index dc9591b0e2..848db55281 100644
--- a/pc/rtp_sender.h
+++ b/pc/rtp_sender.h
@@ -90,7 +90,7 @@ class RtpSenderInternal : public RtpSenderInterface {
   virtual RTCError SetParametersInternal(const RtpParameters& parameters,
                                          SetParametersCallback,
                                          bool blocking) = 0;
-  virtual void SetCachedParameters(RtpParameters parameters) = 0;
+  virtual void SetCachedParameters(std::optional<RtpParameters> parameters) = 0;
 
   // GetParameters and SetParameters will remove deactivated simulcast layers
   // and restore them on SetParameters. This is probably a Bad Idea, but we
@@ -155,7 +155,7 @@ class RtpSenderBase : public RtpSenderInternal, public ObserverInterface {
   RTCError SetParametersInternal(const RtpParameters& parameters,
                                  SetParametersCallback callback = nullptr,
                                  bool blocking = true) override;
-  void SetCachedParameters(RtpParameters parameters) override;
+  void SetCachedParameters(std::optional<RtpParameters> parameters) override;
   RTCError CheckSetParameters(const RtpParameters& parameters);
   RtpParameters GetParametersInternalWithAllLayers() const override;
   RTCError SetParametersInternalWithAllLayers(
diff --git a/pc/rtp_transceiver.cc b/pc/rtp_transceiver.cc
index 0701dca781..a36eab46cf 100644
--- a/pc/rtp_transceiver.cc
+++ b/pc/rtp_transceiver.cc
@@ -231,7 +231,9 @@ CreateMediaContentChannels(
     const AudioOptions& audio_options,
     const VideoOptions& video_options,
     const CryptoOptions& crypto_options,
-    VideoBitrateAllocatorFactory* video_bitrate_allocator_factory) {
+    VideoBitrateAllocatorFactory* video_bitrate_allocator_factory,
+    VideoMediaSendChannelInterface::EncoderSwitchRequestCallback
+        video_encoder_switch_request_callback = nullptr) {
   if (media_type == MediaType::AUDIO) {
     return {media_engine->voice().CreateSendChannel(
                 env, call, media_config, audio_options, crypto_options),
@@ -240,7 +242,8 @@ CreateMediaContentChannels(
   }
   return {media_engine->video().CreateSendChannel(
               env, call, media_config, video_options, crypto_options,
-              video_bitrate_allocator_factory),
+              video_bitrate_allocator_factory,
+              std::move(video_encoder_switch_request_callback)),
           media_engine->video().CreateReceiveChannel(
               env, call, media_config, video_options, crypto_options)};
 }
@@ -286,8 +289,11 @@ RtpTransceiver::RtpTransceiver(const Environment& env,
       thread_(context->signaling_thread()),
       unified_plan_(false),
       media_type_(media_type),
+      signaling_thread_safety_(PendingTaskSafetyFlag::CreateAttachedToTaskQueue(
+          /*alive=*/false,
+          context->signaling_thread())),
       network_thread_safety_(PendingTaskSafetyFlag::CreateAttachedToTaskQueue(
-          true,
+          /*alive=*/true,
           context->network_thread())),
       context_(context),
       codec_lookup_helper_(codec_lookup_helper),
@@ -310,8 +316,11 @@ RtpTransceiver::RtpTransceiver(
       thread_(context->signaling_thread()),
       unified_plan_(true),
       media_type_(sender->media_type()),
+      signaling_thread_safety_(PendingTaskSafetyFlag::CreateAttachedToTaskQueue(
+          /*alive=*/false,
+          context->signaling_thread())),
       network_thread_safety_(PendingTaskSafetyFlag::CreateAttachedToTaskQueue(
-          true,
+          /*alive=*/true,
           context->network_thread())),
       context_(context),
       codec_lookup_helper_(codec_lookup_helper),
@@ -365,8 +374,11 @@ RtpTransceiver::RtpTransceiver(
       thread_(context->signaling_thread()),
       unified_plan_(true),
       media_type_(media_type),
+      signaling_thread_safety_(PendingTaskSafetyFlag::CreateAttachedToTaskQueue(
+          /*alive=*/false,
+          context->signaling_thread())),
       network_thread_safety_(PendingTaskSafetyFlag::CreateAttachedToTaskQueue(
-          true,
+          /*alive=*/true,
           context->network_thread())),
       media_engine_ref_(nullptr),
       context_(context),
@@ -393,7 +405,8 @@ RtpTransceiver::RtpTransceiver(
     RTC_DCHECK_RUN_ON(this->context()->worker_thread());
     auto channels = CreateMediaContentChannels(
         media_type_, env_, media_engine(), call, media_config, audio_options,
-        video_options, crypto_options, video_bitrate_allocator_factory);
+        video_options, crypto_options, video_bitrate_allocator_factory,
+        GetEncoderSwitchRequestCallback());
     owned_send_channel_ = std::move(channels.first);
     owned_receive_channel_ = std::move(channels.second);
     senders_.push_back(CreateSender(media_type_, env_, context_, legacy_stats_,
@@ -444,6 +457,15 @@ RTCError RtpTransceiver::CreateChannel(
 
   mid_ = mid;
 
+  if (!signaling_thread_safety_) {
+    // This code path is hit during rollback.
+    signaling_thread_safety_ = PendingTaskSafetyFlag::Create();
+  } else {
+    // Newly constructed.
+    RTC_DCHECK(!signaling_thread_safety_->alive());
+    signaling_thread_safety_->SetAlive();
+  }
+
   std::unique_ptr<ChannelInterface> new_channel;
   // TODO(bugs.webrtc.org/11992): CreateVideoChannel internally switches to
   // the worker thread. We shouldn't be using the `call_ptr_` hack here but
@@ -471,7 +493,7 @@ RTCError RtpTransceiver::CreateChannel(
       auto channels = CreateMediaContentChannels(
           media_type(), env_, media_engine(), call_ptr, media_config,
           audio_options, video_options, crypto_options,
-          video_bitrate_allocator_factory);
+          video_bitrate_allocator_factory, GetEncoderSwitchRequestCallback());
       media_send_channel = std::move(channels.first);
       media_receive_channel = std::move(channels.second);
       SetMediaChannels(media_send_channel.get(), media_receive_channel.get());
@@ -513,7 +535,6 @@ RTCError RtpTransceiver::SetChannel(
 
   RTC_DCHECK_EQ(media_type(), channel->media_type());
   RTC_DCHECK(mid_ || channel->mid().empty());
-  signaling_thread_safety_ = PendingTaskSafetyFlag::Create();
   channel_ = std::move(channel);
   transport_name_ = std::nullopt;
 
@@ -578,7 +599,7 @@ absl::AnyInvocable<void() &&> RtpTransceiver::GetClearChannelNetworkTask() {
   // combine these into one function to avoid an ordering mistake?
 
   if (!channel_) {
-    RTC_DCHECK(!signaling_thread_safety_);
+    RTC_DCHECK(!signaling_thread_safety_ || !signaling_thread_safety_->alive());
     return nullptr;
   }
 
@@ -598,7 +619,7 @@ absl::AnyInvocable<void() &&> RtpTransceiver::GetClearChannelNetworkTask() {
 absl::AnyInvocable<void() &&> RtpTransceiver::GetDeleteChannelWorkerTask(
     bool stop_senders) {
   RTC_DCHECK_RUN_ON(thread_);
-  RTC_DCHECK(signaling_thread_safety_ == nullptr)
+  RTC_DCHECK(!signaling_thread_safety_ || !signaling_thread_safety_->alive())
       << "GetClearChannelNetworkTask() must be called first";
 
   if (!channel_) {
@@ -668,6 +689,33 @@ void RtpTransceiver::SetMediaChannels(MediaSendChannelInterface* send,
     receiver->internal()->SetMediaChannel(receive);
   }
 }
+VideoMediaSendChannelInterface::EncoderSwitchRequestCallback
+RtpTransceiver::GetEncoderSwitchRequestCallback() {
+  if (media_type() != MediaType::VIDEO) {
+    return nullptr;
+  }
+  RTC_DCHECK(signaling_thread_safety_);
+  // Return a task that first clears the sender parameter cache on the signaling
+  // thread and then posts a task to apply the codec switch changes to the
+  // parameters on the worker.
+  return
+      [this, signaling_thread = context_->signaling_thread(),
+       worker_thread = context_->worker_thread(),
+       signaling_safety = signaling_thread_safety_](
+          VideoMediaSendChannelInterface::EncoderSwitchRequestAction action) {
+        // Called on the encoder task queue.
+        signaling_thread->PostTask(SafeTask(
+            signaling_safety,
+            [this, worker_thread, action = std::move(action)]() mutable {
+              for (const auto& sender : senders_) {
+                sender->internal()->SetCachedParameters(std::nullopt);
+              }
+              worker_thread->PostTask([action = std::move(action)]() mutable {
+                std::move(action)();
+              });
+            }));
+      };
+}
 
 // RTC_RUN_ON(context()->worker_thread());
 void RtpTransceiver::ClearMediaChannelReferences() {
diff --git a/pc/rtp_transceiver.h b/pc/rtp_transceiver.h
index cad5276f6b..81e1855c12 100644
--- a/pc/rtp_transceiver.h
+++ b/pc/rtp_transceiver.h
@@ -423,6 +423,9 @@ class RtpTransceiver : public RtpTransceiverInterface {
       RTC_RUN_ON(context()->worker_thread());
   void ClearMediaChannelReferences() RTC_RUN_ON(context()->worker_thread());
 
+  VideoMediaSendChannelInterface::EncoderSwitchRequestCallback
+  GetEncoderSwitchRequestCallback();
+
   RTCError UpdateCodecPreferencesCaches(
       const std::vector<RtpCodecCapability>& codecs);
   // Helper function for handling extensions during O/A
@@ -438,8 +441,7 @@ class RtpTransceiver : public RtpTransceiverInterface {
   TaskQueueBase* const thread_;
   const bool unified_plan_;
   const MediaType media_type_;
-  scoped_refptr<PendingTaskSafetyFlag> signaling_thread_safety_
-      RTC_GUARDED_BY(thread_);
+  scoped_refptr<PendingTaskSafetyFlag> signaling_thread_safety_;
   scoped_refptr<PendingTaskSafetyFlag> network_thread_safety_;
   std::vector<scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>
       senders_;
diff --git a/pc/test/mock_rtp_sender_internal.h b/pc/test/mock_rtp_sender_internal.h
index 6d14a12cef..eeba6fac6f 100644
--- a/pc/test/mock_rtp_sender_internal.h
+++ b/pc/test/mock_rtp_sender_internal.h
@@ -13,6 +13,7 @@
 
 #include <cstdint>
 #include <memory>
+#include <optional>
 #include <string>
 #include <vector>
 
@@ -60,7 +61,10 @@ class MockRtpSenderInternal : public RtpSenderInternal {
               set_transport,
               (webrtc::scoped_refptr<DtlsTransportInterface>),
               (override));
-  MOCK_METHOD(void, SetCachedParameters, (RtpParameters), (override));
+  MOCK_METHOD(void,
+              SetCachedParameters,
+              (std::optional<RtpParameters>),
+              (override));
   MOCK_METHOD(RtpParameters, GetParameters, (), (const, override));
   MOCK_METHOD(RtpParameters,
               GetParametersInternal,

From 1ab988ec607e2ace22bbdd86b6d3841ee3549152 Mon Sep 17 00:00:00 2001
From: Per K <perkj@webrtc.org>
Date: Fri, 13 Mar 2026 11:24:17 +0000
Subject: [PATCH 296/472] Fix flakiness in
 ScreamTest.SendVideoOnlyReturnLinkWithBurstLoss

After increasing MSS, BWE rampup is more aggressive.

Bug: webrtc:447037083
Change-Id: Ie1890ec8cfef255a2871b0c71ba32919da7d6a0c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456480
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47149}
---
 test/peer_scenario/bwe_integration_tests/scream_test.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/test/peer_scenario/bwe_integration_tests/scream_test.cc b/test/peer_scenario/bwe_integration_tests/scream_test.cc
index 0cf1d13812..b8dfec15cc 100644
--- a/test/peer_scenario/bwe_integration_tests/scream_test.cc
+++ b/test/peer_scenario/bwe_integration_tests/scream_test.cc
@@ -806,7 +806,7 @@ TEST(ScreamTest, MaybeTest(SendVideoOnlyReturnLinkWithBurstLoss)) {
             GetPacketsSent(result.caller_stats[5]));
   EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
                                               DataRate::KilobitsPerSec(10),
-                                              DataRate::KilobitsPerSec(1100))));
+                                              DataRate::KilobitsPerSec(1200))));
 }
 
 // Test that Scream adapt to a link with traffic policing on the network path

From a0e526cca6524d60bc831d85e7b93bc3c7e7eefd Mon Sep 17 00:00:00 2001
From: Jeremy Leconte <jleconte@google.com>
Date: Fri, 13 Mar 2026 05:58:33 -0700
Subject: [PATCH 297/472] Revert "Roll chromium_revision b937d39273..8f11ead6b3
 (1597657:1598931)"

This reverts commit ea1fce97e610d7c0323a01faf271506af96b6c01.

Reason for revert: breaks android CI bot

Original change's description:
> Roll chromium_revision b937d39273..8f11ead6b3 (1597657:1598931)
>
> Change log: https://chromium.googlesource.com/chromium/src/+log/b937d39273..8f11ead6b3
> Full diff: https://chromium.googlesource.com/chromium/src/+/b937d39273..8f11ead6b3
>
> Changed dependencies
> * src/build: https://chromium.googlesource.com/chromium/src/build/+log/7d29c62736..618025ac18
> * src/buildtools: https://chromium.googlesource.com/chromium/src/buildtools/+log/fef8d6947f..95ed44cf5f
> * src/buildtools/linux64: git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9..git_revision:2941d2a26fa9cef3103f10e1dee5da7cd404008a
> * src/buildtools/mac: git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9..git_revision:2941d2a26fa9cef3103f10e1dee5da7cd404008a
> * src/buildtools/win: git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9..git_revision:2941d2a26fa9cef3103f10e1dee5da7cd404008a
> * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/94f693fcb6..72d8d02177
> * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/e7a1e88ecc..92de5f9238
> * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/a133c94b51..4b35a84f2e
> * src/third_party/android_build_tools/aapt2/cipd: vUP1cs7krVsNHbUJpwSDR7rvWuYLGzINVa9xopPpt74C..gsaUgZUqoyD0XG1E9-xesSWknHZINEuS00iSEncvlE0C
> * src/third_party/android_build_tools/lint/cipd: 1N_9yErKkcZgqyxBghA0bb5L5AFEPZCt36Gv2o8_OAwC..UdJPLkDOu5joGtPdwsjc1Ltg5hcadLKPynhbd6NacsoC
> * src/third_party/android_build_tools/manifest_merger/cipd: xT8A6Q_b44_iYKqAMvRkWmvID-ps-I6uBDfRTQv68bcC..KYQSw0Ej8zT_IRNL02DxEmk0nCvkykopkQhsGk_HE0EC
> * src/third_party/android_deps/autorolled/cipd: umaYTBt4Vsa5QZYp57rLJDNzoXalsR2hPGwfYbpZYfwC..3VYcqPssBf1YI1iK0N2UUMsT2Ik5BlMrvm-wOVr1fMQC
> * src/third_party/androidx/cipd: Hi79Lu6qQ-Fbcjwgf_f-NEYJweP4QdZqjJ0lXwc0PpcC..Mn6UxK4fOcnps2oJ1ikX7T1Nf93dtMBkX68uqIyoQ0sC
> * src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/eedf10c1f4..de8c7af5e7
> * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/e0ebf38a01..b9ddc08533
> * src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/c8a5676846..c8ee656688
> * src/third_party/cpuinfo/src: https://chromium.googlesource.com/external/github.com/pytorch/cpuinfo.git/+log/7364b490b5..7607ca5004
> * src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/4ce8ba39a3..6b62257bd9
> * src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/4387e32481..2a2456c873
> * src/third_party/fuzztest/src: https://chromium.googlesource.com/external/github.com/google/fuzztest.git/+log/54ed362692..41e160679e
> * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/260c6fd37c..9751f44f3f
> * src/third_party/harfbuzz-ng/src: https://chromium.googlesource.com/external/github.com/harfbuzz/harfbuzz.git/+log/fa2908bf16..5d4e96ad8d
> * src/third_party/libaom/source/libaom: https://aomedia.googlesource.com/aom.git/+log/98ce0d2a61..0c15af06af
> * src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/044963a253..76c8a4be56
> * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/4cead1b0fc..15e97d25a9
> * src/third_party/r8/cipd: CoaQUbD7pDX3OuETi_O5wkv_Y198RVIUaSobSEx9zfoC..UMLn7pE7DklTsk-SIiMAwch46m8rSF1vG2HQAbMuZvMC
> * src/third_party/tflite/src: https://chromium.googlesource.com/external/github.com/tensorflow/tensorflow.git/+log/f920077085..d20369f4f2
> * src/third_party/xnnpack/src: https://chromium.googlesource.com/external/github.com/google/XNNPACK.git/+log/26feacc3eb..15e04ff8b6
> * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/9765576870..815b983a1c
> * src/tools/luci-go: git_revision:eefaa865c43930859313a63d426740cdd0e22673..git_revision:09ad27bb7a42580550b9063ef8d0a39a5f01a338
> * src/tools/luci-go: git_revision:eefaa865c43930859313a63d426740cdd0e22673..git_revision:09ad27bb7a42580550b9063ef8d0a39a5f01a338
> * src/tools/luci-go: git_revision:eefaa865c43930859313a63d426740cdd0e22673..git_revision:09ad27bb7a42580550b9063ef8d0a39a5f01a338
> DEPS diff: https://chromium.googlesource.com/chromium/src/+/b937d39273..8f11ead6b3/DEPS
>
> BUG=None
>
> Change-Id: I2f18b9bc0a8067e6c283cf3805b04a140ac923bb
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456440
> Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
> Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
> Cr-Commit-Position: refs/heads/main@{#47147}

Bug: None
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: Idb4693e6fe3a5f3297b8db510f472159884d46d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456461
Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#47150}
---
 DEPS | 66 ++++++++++++++++++++++++++++++------------------------------
 1 file changed, 33 insertions(+), 33 deletions(-)

diff --git a/DEPS b/DEPS
index 12bd7209dd..dffcecaef3 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '8f11ead6b30adfcb8eb5f504d92c596adbe1a1e5',
+  'chromium_revision': 'b937d39273b52034e8542dea4b040587be0a127f',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,28 +69,28 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@618025ac18566ba090c4b86328ea77f83a6c4cae',
+    'https://chromium.googlesource.com/chromium/src/build@7d29c62736304aeaac07df0b75546ffd2b75917a',
   'src/buildtools':
-    'https://chromium.googlesource.com/chromium/src/buildtools@95ed44cf5f06dbb5861030b91c9db9ccb4316762',
+    'https://chromium.googlesource.com/chromium/src/buildtools@fef8d6947fa54202e8a00c58d5ac8fd15780bc33',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
   'src/examples/androidtests/third_party/gradle': {
     'url': 'https://chromium.googlesource.com/external/github.com/gradle/gradle.git@f2d1fb54a951d8b11d25748e4711bec8d128d7e3',
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@72d8d02177f725c05690fd05c52f5d9779e7bf71',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@94f693fcb6dde967f11668a9e21eec1a4ec8b49c',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@92de5f92381b699b0b3139e2a25b6bd7beb85253',
+    'https://chromium.googlesource.com/chromium/src/testing@e7a1e88ecc35a70f3d198674042753ec9e8e3650',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@4b35a84f2e22d8fc094fcb615c12f2288706adc6',
+    'https://chromium.googlesource.com/chromium/src/third_party@a133c94b51ac034d5f375cc1746e66dcdee57282',
 
   'src/buildtools/linux64': {
     'packages': [
       {
         'package': 'gn/gn/linux-${{arch}}',
-        'version': 'git_revision:2941d2a26fa9cef3103f10e1dee5da7cd404008a',
+        'version': 'git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9',
       }
     ],
     'dep_type': 'cipd',
@@ -100,7 +100,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/mac-${{arch}}',
-        'version': 'git_revision:2941d2a26fa9cef3103f10e1dee5da7cd404008a',
+        'version': 'git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9',
       }
     ],
     'dep_type': 'cipd',
@@ -110,7 +110,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/windows-amd64',
-        'version': 'git_revision:2941d2a26fa9cef3103f10e1dee5da7cd404008a',
+        'version': 'git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9',
       }
     ],
     'dep_type': 'cipd',
@@ -331,13 +331,13 @@ deps = {
   'src/third_party/clang-format/script':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@c2725e0622e1a86d55f14514f2177a39efea4a0e',
   'src/third_party/compiler-rt/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@c8ee6566881410a159d66e99081779d28124fd3f',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@c8a567684659fe05ede3a220965b94fd71c4a7cb',
   'src/third_party/libc++/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@7ab65651aed6802d2599dcb7a73b1f82d5179d05',
   'src/third_party/libc++abi/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@8f11bb1d4438d0239d0dfc1bd9456a9f31629dda',
   'src/third_party/llvm-libc/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@76c8a4be56babbe6104f5cd9eda25a95dd277b7a',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@044963a253be3b8b2a63a1b2e0f3e361b01bc186',
   'src/third_party/libunwind/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@db838d918570d4e381ecf9f5cc70a0098c9c2cd6',
 
@@ -402,7 +402,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/android_build_tools/aapt2',
-              'version': 'gsaUgZUqoyD0XG1E9-xesSWknHZINEuS00iSEncvlE0C',
+              'version': 'vUP1cs7krVsNHbUJpwSDR7rvWuYLGzINVa9xopPpt74C',
           },
       ],
       'condition': 'checkout_android',
@@ -457,7 +457,7 @@ deps = {
       'packages': [
           {
                'package': 'chromium/third_party/android_build_tools/lint',
-               'version': 'UdJPLkDOu5joGtPdwsjc1Ltg5hcadLKPynhbd6NacsoC',
+               'version': '1N_9yErKkcZgqyxBghA0bb5L5AFEPZCt36Gv2o8_OAwC',
           },
       ],
       'condition': 'checkout_android and non_git_source',
@@ -488,11 +488,11 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@de8c7af5e7c40153eb6cc0a79ff61a43f4bb03f9',
+    'https://boringssl.googlesource.com/boringssl.git@eedf10c1f4ab44629e2fe3fe6f07e2fb0897c0f4',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@8be0e3114685fcc1589561067282edf75ea1259a',
   'src/third_party/catapult':
-    'https://chromium.googlesource.com/catapult.git@b9ddc08533aca2b2878bc05f2cce8e3051616313',
+    'https://chromium.googlesource.com/catapult.git@e0ebf38a01214aba11f31daa1c743782def031d5',
   'src/third_party/ced/src': {
     'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5',
   },
@@ -505,13 +505,13 @@ deps = {
   'src/third_party/crc32c/src':
     'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558',
   'src/third_party/depot_tools':
-    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@6b62257bd9fc9ee0e00d10712a0f3db63132ba2f',
+    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@4ce8ba39a3488397a2d1494f167020f21de502f3',
   'src/third_party/ffmpeg':
     'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@946d97db8d906277085e361892b7efda5152e2f1',
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@9751f44f3f6640142d242ddac084f7a80dce4aac',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@260c6fd37c502e984836a446c5d90e4a1298036d',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -521,7 +521,7 @@ deps = {
   'src/third_party/freetype/src':
     'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@99b479dc34728936b006679a31e12b8cf432fc55',
   'src/third_party/harfbuzz-ng/src':
-    'https://chromium.googlesource.com/external/github.com/harfbuzz/harfbuzz.git@5d4e96ad8d00fc871ffa17707b2ca08fa850e7d6',
+    'https://chromium.googlesource.com/external/github.com/harfbuzz/harfbuzz.git@fa2908bf16d2ccd6623f4d575455fea72a1a722b',
   'src/third_party/google_benchmark/src': {
     'url': 'https://chromium.googlesource.com/external/github.com/google/benchmark.git@188e8278990a9069ffc84441cb5a024fd0bede37',
   },
@@ -593,7 +593,7 @@ deps = {
   'src/third_party/libFuzzer/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt/lib/fuzzer.git@bea408a6e01f0f7e6c82a43121fe3af4506c932e',
   'src/third_party/fuzztest/src':
-    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@41e160679e3c010a860e674e7997e2b285c6e0a2',
+    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@54ed362692177e93df0fa202cae404b57694e828',
   'src/third_party/libjpeg_turbo':
     'https://chromium.googlesource.com/chromium/deps/libjpeg_turbo.git@d1f5f2393e0d51f840207342ae86e55a86443288',
   'src/third_party/libsrtp':
@@ -601,7 +601,7 @@ deps = {
   'src/third_party/dav1d/libdav1d':
     'https://chromium.googlesource.com/external/github.com/videolan/dav1d.git@241a6b236a9c84441c872119c1f31b4074c0589d',
   'src/third_party/libaom/source/libaom':
-    'https://aomedia.googlesource.com/aom.git@0c15af06af108af03cbe237e4bffa4b700b76d81',
+    'https://aomedia.googlesource.com/aom.git@98ce0d2a610f1465ea14d525bce9fa46d4b05c03',
   'src/third_party/libgav1/src':
     Var('chromium_git') + '/codecs/libgav1.git' + '@' + '40f58ed32ff39071c3f2a51056dbc49a070af0dc',
   'src/third_party/libunwindstack': {
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '15e97d25a95aed5a96c92553e254b7e491bf7369',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '4cead1b0fc9987f994de97c7914e1f0c94047579',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -644,7 +644,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/r8',
-              'version': 'UMLn7pE7DklTsk-SIiMAwch46m8rSF1vG2HQAbMuZvMC',
+              'version': 'CoaQUbD7pDX3OuETi_O5wkv_Y198RVIUaSobSEx9zfoC',
           },
       ],
       'condition': 'checkout_android',
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@815b983a1c95803681d04b28e82b22a358f078b4',
+    'https://chromium.googlesource.com/chromium/src/tools@9765576870401b230a826e0e4f7a12851575306e',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'Mn6UxK4fOcnps2oJ1ikX7T1Nf93dtMBkX68uqIyoQ0sC',
+          'version': 'Hi79Lu6qQ-Fbcjwgf_f-NEYJweP4QdZqjJ0lXwc0PpcC',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -718,7 +718,7 @@ deps = {
       'packages': [
           {
                'package': 'chromium/third_party/android_build_tools/manifest_merger',
-               'version': 'KYQSw0Ej8zT_IRNL02DxEmk0nCvkykopkQhsGk_HE0EC',
+               'version': 'xT8A6Q_b44_iYKqAMvRkWmvID-ps-I6uBDfRTQv68bcC',
           },
       ],
       'condition': 'checkout_android',
@@ -786,7 +786,7 @@ deps = {
   },
 
   'src/third_party/tflite/src':
-    Var('chromium_git') + '/external/github.com/tensorflow/tensorflow.git' + '@' + 'd20369f4f226598800e0d406db567d2d1c17fa95',
+    Var('chromium_git') + '/external/github.com/tensorflow/tensorflow.git' + '@' + 'f9200770858f5721b7fa95db7aa3e4cd712fe9df',
 
   'src/third_party/turbine/cipd': {
       'packages': [
@@ -808,15 +808,15 @@ deps = {
       'packages': [
         {
           'package': 'infra/tools/luci/cas/${{platform}}',
-          'version': 'git_revision:09ad27bb7a42580550b9063ef8d0a39a5f01a338',
+          'version': 'git_revision:eefaa865c43930859313a63d426740cdd0e22673',
         },
         {
           'package': 'infra/tools/luci/isolate/${{platform}}',
-          'version': 'git_revision:09ad27bb7a42580550b9063ef8d0a39a5f01a338',
+          'version': 'git_revision:eefaa865c43930859313a63d426740cdd0e22673',
         },
         {
           'package': 'infra/tools/luci/swarming/${{platform}}',
-          'version': 'git_revision:09ad27bb7a42580550b9063ef8d0a39a5f01a338',
+          'version': 'git_revision:eefaa865c43930859313a63d426740cdd0e22673',
         }
       ],
       'dep_type': 'cipd',
@@ -842,7 +842,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/android_deps/autorolled',
-              'version': '3VYcqPssBf1YI1iK0N2UUMsT2Ik5BlMrvm-wOVr1fMQC',
+              'version': 'umaYTBt4Vsa5QZYp57rLJDNzoXalsR2hPGwfYbpZYfwC',
           },
       ],
       'condition': 'checkout_android and non_git_source',
@@ -852,7 +852,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/google/pthreadpool.git' + '@' + '9003ee6c137cea3b94161bd5c614fb43be523ee1',
 
   'src/third_party/xnnpack/src':
-    Var('chromium_git') + '/external/github.com/google/XNNPACK.git' + '@' + '15e04ff8b61e0bc2118bad268447be5b2075aa66',
+    Var('chromium_git') + '/external/github.com/google/XNNPACK.git' + '@' + '26feacc3ebe30d1f3a82e1ff3938ef9859d74640',
 
   'src/third_party/farmhash/src':
     Var('chromium_git') + '/external/github.com/google/farmhash.git' + '@' + '816a4ae622e964763ca0862d9dbd19324a1eaf45',
@@ -861,10 +861,10 @@ deps = {
     Var('chromium_git') + '/external/github.com/google/ruy.git' + '@' + '2af88863614a8298689cc52b1a47b3fcad7be835',
 
   'src/third_party/cpuinfo/src':
-    Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7607ca500436b37ad23fb8d18614bec7796b68a7',
+    Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7364b490b5f78d58efe23ea76e74210fd6c3c76f',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '2a2456c873d6f4bf43a9adf919f13ad917d57c2d',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '4387e3248109bef2c07a8b69a2b8249d7af5dcf0',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From ff07f34d6bf0b445d7cad3703bb9643d06856425 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Fri, 13 Mar 2026 12:00:59 +0100
Subject: [PATCH 298/472] Move payload_type_demuxing_enabled_ to the network
 thread
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Bug: webrtc:42222117
Change-Id: I1414d6dc78e928029de5835f584b22dc5cfd85e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456245
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47151}
---
 pc/channel.cc          | 82 +++++++++++++++++++++---------------------
 pc/channel.h           | 12 ++-----
 pc/rtp_transceiver.cc  |  2 +-
 pc/sdp_offer_answer.cc | 10 +++---
 4 files changed, 49 insertions(+), 57 deletions(-)

diff --git a/pc/channel.cc b/pc/channel.cc
index a3f4f40f95..3f70792fe3 100644
--- a/pc/channel.cc
+++ b/pc/channel.cc
@@ -52,6 +52,7 @@
 #include "rtc_base/socket.h"
 #include "rtc_base/strings/string_format.h"
 #include "rtc_base/thread.h"
+#include "rtc_base/thread_annotations.h"
 #include "rtc_base/trace_event.h"
 #include "rtc_base/unique_id_generator.h"
 
@@ -361,11 +362,44 @@ bool BaseChannel::SetPayloadTypeDemuxingEnabled(bool enabled) {
   // network thread. At the moment there's a workaround for inconsistent state
   // between the worker and network thread because of this (see
   // OnDemuxerCriteriaUpdatePending elsewhere in this file) and
-  // SetPayloadTypeDemuxingEnabled_w has a BlockingCall over to the network
+  // UpdateRemoteStreams_w has a BlockingCall over to the network
   // thread to apply state updates.
-  RTC_DCHECK_RUN_ON(worker_thread());
+  RTC_DCHECK_RUN_ON(network_thread());
   TRACE_EVENT0("webrtc", "BaseChannel::SetPayloadTypeDemuxingEnabled");
-  return SetPayloadTypeDemuxingEnabled_w(enabled);
+
+  if (enabled == payload_type_demuxing_enabled_) {
+    return true;
+  }
+
+  payload_type_demuxing_enabled_ = enabled;
+
+  if (!enabled) {
+    worker_thread_->PostTask(SafeTask(alive_, [this] {
+      // TODO(crbug.com/11477): This will remove *all* unsignaled streams (those
+      // without an explicitly signaled SSRC), which may include streams that
+      // were matched to this channel by MID or RID. Ideally we'd remove only
+      // the streams that were matched based on payload type alone, but
+      // currently there is no straightforward way to identify those streams.
+      media_receive_channel()->ResetUnsignaledRecvStream();
+    }));
+  }
+
+  // TODO: bugs.webrtc.org/42222117 - payload_types_ is guarded by the worker
+  // thread, but we read it here on the network thread. This is safe because
+  // this method is called synchronously during SDP signaling, during which no
+  // tasks that modify payload_types_ can be running on the worker thread.
+  auto has_payload_types = [this]() RTC_NO_THREAD_SAFETY_ANALYSIS {
+    return !payload_types_.empty();
+  };
+
+  if (has_payload_types()) {
+    if (!rtp_transport_) {
+      return false;
+    }
+    return rtp_transport_->RegisterRtpDemuxerSink(demuxer_criteria(), this);
+  }
+
+  return true;
 }
 
 bool BaseChannel::IsReadyToSendMedia_w() const {
@@ -705,33 +739,6 @@ void BaseChannel::ChannelNotWritable_n() {
   RTC_LOG(LS_INFO) << "Channel not writable (" << ToString() << ")";
 }
 
-bool BaseChannel::SetPayloadTypeDemuxingEnabled_w(bool enabled) {
-  RTC_LOG_THREAD_BLOCK_COUNT();
-
-  if (enabled == payload_type_demuxing_enabled_) {
-    return true;
-  }
-
-  payload_type_demuxing_enabled_ = enabled;
-
-  if (!enabled) {
-    // TODO(crbug.com/11477): This will remove *all* unsignaled streams (those
-    // without an explicitly signaled SSRC), which may include streams that
-    // were matched to this channel by MID or RID. Ideally we'd remove only the
-    // streams that were matched based on payload type alone, but currently
-    // there is no straightforward way to identify those streams.
-    media_receive_channel()->ResetUnsignaledRecvStream();
-    if (!payload_types_.empty()) {
-      return RegisterRtpDemuxerSink_w();
-    }
-  } else if (!payload_types_.empty()) {
-    return RegisterRtpDemuxerSink_w();
-  }
-
-  RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(0);
-
-  return true;
-}
 
 bool BaseChannel::UpdateLocalStreams_w(const std::vector<StreamParams>& streams,
                                        SdpType type,
@@ -820,8 +827,9 @@ bool BaseChannel::UpdateRemoteStreams_w(const MediaContentDescription* content,
     RTC_DLOG(LS_VERBOSE) << "UpdateRemoteStreams_w: remote side will not send "
                             "- disable payload type demuxing for "
                          << ToString();
-    if (ClearHandledPayloadTypes()) {
-      needs_re_registration = payload_type_demuxing_enabled_;
+    if (!payload_types_.empty()) {
+      payload_types_.clear();
+      needs_re_registration = true;
     }
   }
 
@@ -913,15 +921,7 @@ RtpHeaderExtensions BaseChannel::GetDeduplicatedRtpHeaderExtensions(
 bool BaseChannel::MaybeAddHandledPayloadType(int payload_type) {
   // Even if payload type demuxing is currently disabled, we need to remember
   // the payload types in case it's re-enabled later.
-  bool inserted =
-      payload_types_.insert(static_cast<uint8_t>(payload_type)).second;
-  return inserted && payload_type_demuxing_enabled_;
-}
-
-bool BaseChannel::ClearHandledPayloadTypes() {
-  const bool was_empty = payload_types_.empty();
-  payload_types_.clear();
-  return !was_empty && payload_type_demuxing_enabled_;
+  return payload_types_.insert(static_cast<uint8_t>(payload_type)).second;
 }
 
 bool BaseChannel::CheckRtpExtensionValidity(
diff --git a/pc/channel.h b/pc/channel.h
index 8a30c46490..c53c1b44f5 100644
--- a/pc/channel.h
+++ b/pc/channel.h
@@ -136,7 +136,8 @@ class BaseChannel : public ChannelInterface,
   // This method will also remove any existing streams that were bound to this
   // channel on the basis of payload type, since one of these streams might
   // actually belong to a new channel. See: crbug.com/webrtc/11477
-  bool SetPayloadTypeDemuxingEnabled(bool enabled) override;
+  bool SetPayloadTypeDemuxingEnabled(bool enabled) override
+      RTC_RUN_ON(network_thread());
 
   void Enable(bool enable) override;
 
@@ -249,9 +250,6 @@ class BaseChannel : public ChannelInterface,
   void ChannelWritable_n() RTC_RUN_ON(network_thread());
   void ChannelNotWritable_n() RTC_RUN_ON(network_thread());
 
-  bool SetPayloadTypeDemuxingEnabled_w(bool enabled)
-      RTC_RUN_ON(worker_thread());
-
   // Should be called whenever the conditions for
   // IsReadyToReceiveMedia/IsReadyToSendMedia are satisfied (or unsatisfied).
   // Updates the send/recv state of the media channel.
@@ -286,10 +284,6 @@ class BaseChannel : public ChannelInterface,
   // is needed.
   bool MaybeAddHandledPayloadType(int payload_type) RTC_RUN_ON(worker_thread());
 
-  // Returns true if the demuxer payload type criteria was non-empty before
-  // clearing.
-  bool ClearHandledPayloadTypes() RTC_RUN_ON(worker_thread());
-
   // Checks that the provided RTP header extensions are valid.
   // This verifies that all extension IDs are within the valid range,
   // that there are no duplicate IDs, and that no existing extension ID
@@ -364,7 +358,7 @@ class BaseChannel : public ChannelInterface,
   // call to the worker thread, so it should be safe.
   bool enabled_ RTC_GUARDED_BY(worker_thread()) = false;
   bool enabled_s_ RTC_GUARDED_BY(signaling_thread()) = false;
-  bool payload_type_demuxing_enabled_ RTC_GUARDED_BY(worker_thread()) = true;
+  bool payload_type_demuxing_enabled_ RTC_GUARDED_BY(network_thread()) = true;
   std::vector<StreamParams> local_streams_ RTC_GUARDED_BY(worker_thread());
   std::vector<StreamParams> remote_streams_ RTC_GUARDED_BY(worker_thread());
   RtpTransceiverDirection local_content_direction_
diff --git a/pc/rtp_transceiver.cc b/pc/rtp_transceiver.cc
index a36eab46cf..63576448eb 100644
--- a/pc/rtp_transceiver.cc
+++ b/pc/rtp_transceiver.cc
@@ -1366,7 +1366,7 @@ bool RtpTransceiver::SetChannelContent(
 }
 
 bool RtpTransceiver::SetChannelPayloadTypeDemuxingEnabled(bool enabled) {
-  RTC_DCHECK_RUN_ON(context()->worker_thread());
+  RTC_DCHECK_RUN_ON(context()->network_thread());
   RTC_DCHECK(channel_);
   return channel_->SetPayloadTypeDemuxingEnabled(enabled);
 }
diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc
index 590a46563f..6048d98657 100644
--- a/pc/sdp_offer_answer.cc
+++ b/pc/sdp_offer_answer.cc
@@ -5962,12 +5962,10 @@ bool SdpOfferAnswerHandler::UpdatePayloadTypeDemuxingState(
     return true;
   }
 
-  // TODO(bugs.webrtc.org/11993): This BlockingCall() will also block on the
-  // network thread for every demuxer sink that needs to be updated. The demuxer
-  // state needs to be fully (and only) managed on the network thread and once
-  // that's the case, there's no need to stop by on the worker. Ideally we could
-  // also do this without blocking.
-  return context_->worker_thread()->BlockingCall([&channels_to_update]() {
+  // TODO(bugs.webrtc.org/11993): The demuxer state is now fully managed on
+  // the network thread. So we do a single blocking call here to the network
+  // thread. Ideally we could also do this without blocking.
+  return context_->network_thread()->BlockingCall([&channels_to_update]() {
     for (const auto& it : channels_to_update) {
       if (!it.second->SetChannelPayloadTypeDemuxingEnabled(it.first)) {
         // Note that the state has already been irrevocably changed at this

From 5460d74631c9df16c9f0d160fb872c9834f8bb59 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Erik=20Spr=C3=A5ng?= <sprang@webrtc.org>
Date: Fri, 13 Mar 2026 10:56:43 +0100
Subject: [PATCH 299/472] Reject frames with inconsistent U/V strides.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

All of openh264, libvpx and libaom seems to have issues with this. For
now, rejecting frames with this property. This should be seen as a
temporary workaround to be removed when the implementations have been
updated to support it.

Bug: chromium:492213293
Change-Id: I44e784ed5f70dc514afbecbcd1aa7957c372a238
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456203
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47152}
---
 modules/video_coding/BUILD.gn                 |  1 +
 modules/video_coding/codecs/av1/BUILD.gn      |  2 +
 .../codecs/av1/libaom_av1_encoder.cc          |  7 ++
 .../codecs/av1/libaom_av1_unittest.cc         | 79 +++++++++++++++++++
 .../codecs/h264/h264_encoder_impl_unittest.cc | 42 +++++++++-
 .../codecs/vp8/libvpx_vp8_encoder.cc          | 22 ++++++
 .../codecs/vp8/test/vp8_impl_unittest.cc      | 51 ++++++++++++
 .../codecs/vp9/libvpx_vp9_encoder.cc          | 14 ++++
 .../codecs/vp9/test/vp9_impl_unittest.cc      | 51 ++++++++++++
 9 files changed, 268 insertions(+), 1 deletion(-)

diff --git a/modules/video_coding/BUILD.gn b/modules/video_coding/BUILD.gn
index e381e862fe..34ff20e3f5 100644
--- a/modules/video_coding/BUILD.gn
+++ b/modules/video_coding/BUILD.gn
@@ -1087,6 +1087,7 @@ if (rtc_include_tests) {
       "../../api:create_videocodec_test_fixture_api",
       "../../api:field_trials",
       "../../api:frame_generator_api",
+      "../../api:make_ref_counted",
       "../../api:mock_video_codec_factory",
       "../../api:mock_video_decoder",
       "../../api:mock_video_encoder",
diff --git a/modules/video_coding/codecs/av1/BUILD.gn b/modules/video_coding/codecs/av1/BUILD.gn
index 1a03a7e365..1e94e36c81 100644
--- a/modules/video_coding/codecs/av1/BUILD.gn
+++ b/modules/video_coding/codecs/av1/BUILD.gn
@@ -130,6 +130,8 @@ if (rtc_include_tests) {
       "../..:video_codec_interface",
       "../../../../api:array_view",
       "../../../../api:field_trials",
+      "../../../../api:make_ref_counted",
+      "../../../../api:scoped_refptr",
       "../../../../api/environment",
       "../../../../api/environment:environment_factory",
       "../../../../api/transport/rtp:dependency_descriptor",
diff --git a/modules/video_coding/codecs/av1/libaom_av1_encoder.cc b/modules/video_coding/codecs/av1/libaom_av1_encoder.cc
index 4aa7199715..1079cb8d0a 100644
--- a/modules/video_coding/codecs/av1/libaom_av1_encoder.cc
+++ b/modules/video_coding/codecs/av1/libaom_av1_encoder.cc
@@ -878,6 +878,13 @@ int32_t LibaomAv1Encoder::Encode(
                                mapped_buffer->height());
       auto i420_buffer = mapped_buffer->GetI420();
       RTC_DCHECK(i420_buffer);
+
+      // TODO: crbug.com/492213293 - Remove once the root cause is fixed.
+      if (i420_buffer->StrideU() != i420_buffer->StrideV()) {
+        RTC_LOG(LS_ERROR) << "Libaom requires the U and V strides to be equal.";
+        return WEBRTC_VIDEO_CODEC_ENCODER_FAILURE;
+      }
+
       RTC_CHECK_EQ(i420_buffer->width(), frame_for_encode_->d_w);
       RTC_CHECK_EQ(i420_buffer->height(), frame_for_encode_->d_h);
       frame_for_encode_->planes[AOM_PLANE_Y] =
diff --git a/modules/video_coding/codecs/av1/libaom_av1_unittest.cc b/modules/video_coding/codecs/av1/libaom_av1_unittest.cc
index 58c15f801a..df6f57216e 100644
--- a/modules/video_coding/codecs/av1/libaom_av1_unittest.cc
+++ b/modules/video_coding/codecs/av1/libaom_av1_unittest.cc
@@ -22,13 +22,18 @@
 #include "api/environment/environment.h"
 #include "api/environment/environment_factory.h"
 #include "api/field_trials.h"
+#include "api/make_ref_counted.h"
+#include "api/scoped_refptr.h"
+#include "api/test/mock_video_encoder.h"
 #include "api/transport/rtp/dependency_descriptor.h"
 #include "api/units/data_rate.h"
 #include "api/units/data_size.h"
 #include "api/units/time_delta.h"
 #include "api/video/encoded_image.h"
+#include "api/video/i420_buffer.h"
 #include "api/video/video_bitrate_allocation.h"
 #include "api/video/video_frame.h"
+#include "api/video/video_frame_buffer.h"
 #include "api/video_codecs/scalability_mode.h"
 #include "api/video_codecs/video_codec.h"
 #include "api/video_codecs/video_decoder.h"
@@ -234,6 +239,80 @@ TEST(LibaomAv1Test, InitReleaseRepeatedly) {
   }
 }
 
+TEST(LibaomAv1Test, RejectsNativeFramesWithUnequalChromaStrides) {
+  const Environment env = CreateEnvironment();
+  std::unique_ptr<VideoEncoder> encoder = CreateLibaomAv1Encoder(env);
+  VideoCodec codec_settings = DefaultCodecSettings();
+  ASSERT_EQ(encoder->InitEncode(&codec_settings, DefaultEncoderSettings()),
+            WEBRTC_VIDEO_CODEC_OK);
+
+  VideoBitrateAllocation allocation;
+  allocation.SetBitrate(0, 0, 300000);
+  encoder->SetRates(VideoEncoder::RateControlParameters(
+      allocation, codec_settings.maxFramerate));
+
+  MockEncodedImageCallback callback;
+  encoder->RegisterEncodeCompleteCallback(&callback);
+
+  class FakeNativeBuffer : public VideoFrameBuffer {
+   public:
+    FakeNativeBuffer(int width, int height) : width_(width), height_(height) {}
+    Type type() const override { return Type::kNative; }
+    int width() const override { return width_; }
+    int height() const override { return height_; }
+    scoped_refptr<I420BufferInterface> ToI420() override {
+      return I420Buffer::Create(width_, height_, width_, (width_ + 1) / 2,
+                                (width_ + 1) / 2 + 1);
+    }
+
+   private:
+    int width_;
+    int height_;
+  };
+
+  auto buffer = make_ref_counted<FakeNativeBuffer>(codec_settings.width,
+                                                   codec_settings.height);
+
+  VideoFrame frame = VideoFrame::Builder()
+                         .set_video_frame_buffer(buffer)
+                         .set_rtp_timestamp(0)
+                         .build();
+
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_ENCODER_FAILURE,
+            encoder->Encode(frame, nullptr));
+}
+
+TEST(LibaomAv1Test, RejectsI420FramesWithUnequalChromaStrides) {
+  const Environment env = CreateEnvironment();
+  std::unique_ptr<VideoEncoder> encoder = CreateLibaomAv1Encoder(env);
+  VideoCodec codec_settings = DefaultCodecSettings();
+  ASSERT_EQ(encoder->InitEncode(&codec_settings, DefaultEncoderSettings()),
+            WEBRTC_VIDEO_CODEC_OK);
+
+  VideoBitrateAllocation allocation;
+  allocation.SetBitrate(0, 0, 300000);
+  encoder->SetRates(VideoEncoder::RateControlParameters(
+      allocation, codec_settings.maxFramerate));
+
+  MockEncodedImageCallback callback;
+  encoder->RegisterEncodeCompleteCallback(&callback);
+
+  auto buffer = I420Buffer::Create(
+      /*width=*/codec_settings.width,
+      /*height=*/codec_settings.height,
+      /*stride_y=*/codec_settings.width,
+      /*stride_u=*/(codec_settings.width + 1) / 2,
+      /*stride_v=*/(codec_settings.width + 1) / 2 + 1);
+
+  VideoFrame frame = VideoFrame::Builder()
+                         .set_video_frame_buffer(buffer)
+                         .set_rtp_timestamp(0)
+                         .build();
+
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_ENCODER_FAILURE,
+            encoder->Encode(frame, nullptr));
+}
+
 struct LayerId {
   friend bool operator==(const LayerId& lhs, const LayerId& rhs) {
     return std::tie(lhs.spatial_id, lhs.temporal_id) ==
diff --git a/modules/video_coding/codecs/h264/h264_encoder_impl_unittest.cc b/modules/video_coding/codecs/h264/h264_encoder_impl_unittest.cc
index e5aade5188..c4f0dad975 100644
--- a/modules/video_coding/codecs/h264/h264_encoder_impl_unittest.cc
+++ b/modules/video_coding/codecs/h264/h264_encoder_impl_unittest.cc
@@ -15,12 +15,15 @@
 #include <optional>
 
 #include "api/environment/environment_factory.h"
+#include "api/make_ref_counted.h"
+#include "api/scoped_refptr.h"
 #include "api/test/create_frame_generator.h"
 #include "api/test/frame_generator_interface.h"
 #include "api/test/mock_video_encoder.h"
 #include "api/video/i420_buffer.h"
 #include "api/video/video_codec_type.h"
 #include "api/video/video_frame.h"
+#include "api/video/video_frame_buffer.h"
 #include "api/video_codecs/video_codec.h"
 #include "api/video_codecs/video_encoder.h"
 #include "modules/video_coding/codecs/h264/include/h264_globals.h"
@@ -138,7 +141,7 @@ TEST(H264EncoderImplTest, OnFrameDropped) {
   }
 }
 
-TEST(H264EncoderImplTest, RejectsFramesWithUnequalChromaStrides) {
+TEST(H264EncoderImplTest, RejectsI420FramesWithUnequalChromaStrides) {
   H264EncoderImpl encoder(CreateEnvironment(), {});
   VideoCodec codec_settings;
   SetDefaultSettings(&codec_settings);
@@ -163,6 +166,43 @@ TEST(H264EncoderImplTest, RejectsFramesWithUnequalChromaStrides) {
   EXPECT_EQ(WEBRTC_VIDEO_CODEC_ENCODER_FAILURE, encoder.Encode(frame, nullptr));
 }
 
+TEST(H264EncoderImplTest, RejectsNativeFramesWithUnequalChromaStrides) {
+  H264EncoderImpl encoder(CreateEnvironment(), {});
+  VideoCodec codec_settings;
+  SetDefaultSettings(&codec_settings);
+  MockEncodedImageCallback callback;
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder.InitEncode(&codec_settings, kSettings));
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder.RegisterEncodeCompleteCallback(&callback));
+
+  class FakeNativeBuffer : public VideoFrameBuffer {
+   public:
+    FakeNativeBuffer(int width, int height) : width_(width), height_(height) {}
+    Type type() const override { return Type::kNative; }
+    int width() const override { return width_; }
+    int height() const override { return height_; }
+    scoped_refptr<I420BufferInterface> ToI420() override {
+      return I420Buffer::Create(width_, height_, width_, (width_ + 1) / 2,
+                                (width_ + 1) / 2 + 1);
+    }
+
+   private:
+    int width_;
+    int height_;
+  };
+
+  auto buffer = make_ref_counted<FakeNativeBuffer>(codec_settings.width,
+                                                   codec_settings.height);
+
+  VideoFrame frame = VideoFrame::Builder()
+                         .set_video_frame_buffer(buffer)
+                         .set_rtp_timestamp(0)
+                         .build();
+
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_ENCODER_FAILURE, encoder.Encode(frame, nullptr));
+}
+
 }  // anonymous namespace
 
 }  // namespace webrtc
diff --git a/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc b/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc
index 3659d4f561..8fb8c972f2 100644
--- a/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc
+++ b/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc
@@ -1493,6 +1493,13 @@ std::vector<scoped_refptr<VideoFrameBuffer>> LibvpxVp8Encoder::PrepareBuffers(
                         << " image to I420. Can't encode frame.";
       return {};
     }
+
+    // TODO: crbug.com/492213293 - Remove once the root cause is fixed.
+    if (converted_buffer->StrideU() != converted_buffer->StrideV()) {
+      RTC_LOG(LS_ERROR) << "Libvpx requires the U and V strides to be equal.";
+      return {};
+    }
+
     RTC_CHECK(converted_buffer->type() == VideoFrameBuffer::Type::kI420 ||
               converted_buffer->type() == VideoFrameBuffer::Type::kI420A);
 
@@ -1564,6 +1571,21 @@ std::vector<scoped_refptr<VideoFrameBuffer>> LibvpxVp8Encoder::PrepareBuffers(
     SetRawImagePlanes(&raw_images_[i], scaled_buffer.get());
     prepared_buffers.push_back(scaled_buffer);
   }
+
+  // TODO: crbug.com/492213293 - Remove once the root cause is fixed.
+  for (const scoped_refptr<VideoFrameBuffer>& prepared_buffer :
+       prepared_buffers) {
+    if (prepared_buffer->type() == VideoFrameBuffer::Type::kI420 ||
+        prepared_buffer->type() == VideoFrameBuffer::Type::kI420A) {
+      auto i420_buffer = prepared_buffer->GetI420();
+      RTC_DCHECK(i420_buffer);
+      if (i420_buffer->StrideU() != i420_buffer->StrideV()) {
+        RTC_LOG(LS_ERROR) << "Libvpx requires the U and V strides to be equal.";
+        return {};
+      }
+    }
+  }
+
   return prepared_buffers;
 }
 
diff --git a/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc b/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
index 9e0ceb3152..7033b4ecb4 100644
--- a/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
+++ b/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
@@ -22,6 +22,7 @@
 #include "absl/memory/memory.h"
 #include "api/environment/environment_factory.h"
 #include "api/field_trials.h"
+#include "api/make_ref_counted.h"
 #include "api/scoped_refptr.h"
 #include "api/test/create_frame_generator.h"
 #include "api/test/frame_generator_interface.h"
@@ -30,6 +31,7 @@
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
 #include "api/video/encoded_image.h"
+#include "api/video/i420_buffer.h"
 #include "api/video/render_resolution.h"
 #include "api/video/video_bitrate_allocation.h"
 #include "api/video/video_codec_constants.h"
@@ -229,6 +231,55 @@ TEST_F(TestVp8Impl, EncodeFrameAndRelease) {
             encoder_->Encode(NextInputFrame(), nullptr));
 }
 
+TEST_F(TestVp8Impl, RejectsI420FramesWithUnequalChromaStrides) {
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+
+  auto buffer = I420Buffer::Create(
+      /*width=*/kWidth,
+      /*height=*/kHeight,
+      /*stride_y=*/kWidth,
+      /*stride_u=*/(kWidth + 1) / 2,
+      /*stride_v=*/(kWidth + 1) / 2 + 1);
+
+  VideoFrame frame = VideoFrame::Builder()
+                         .set_video_frame_buffer(buffer)
+                         .set_rtp_timestamp(0)
+                         .build();
+
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_ERROR, encoder_->Encode(frame, nullptr));
+}
+
+TEST_F(TestVp8Impl, RejectsNativeFramesWithUnequalChromaStrides) {
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+
+  class FakeNativeBuffer : public VideoFrameBuffer {
+   public:
+    FakeNativeBuffer(int width, int height) : width_(width), height_(height) {}
+    Type type() const override { return Type::kNative; }
+    int width() const override { return width_; }
+    int height() const override { return height_; }
+    scoped_refptr<I420BufferInterface> ToI420() override {
+      return I420Buffer::Create(width_, height_, width_, (width_ + 1) / 2,
+                                (width_ + 1) / 2 + 1);
+    }
+
+   private:
+    int width_;
+    int height_;
+  };
+
+  auto buffer = make_ref_counted<FakeNativeBuffer>(kWidth, kHeight);
+
+  VideoFrame frame = VideoFrame::Builder()
+                         .set_video_frame_buffer(buffer)
+                         .set_rtp_timestamp(0)
+                         .build();
+
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_ERROR, encoder_->Encode(frame, nullptr));
+}
+
 TEST_F(TestVp8Impl, EncodeNv12FrameSimulcast) {
   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
diff --git a/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc b/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc
index 0d440cda2a..105a6e8215 100644
--- a/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc
+++ b/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc
@@ -1215,6 +1215,13 @@ int LibvpxVp9Encoder::Encode(const VideoFrame& input_image,
           i010_buffer = i010_copy.get();
         }
       }
+
+      // TODO: crbug.com/492213293 - Remove once the root cause is fixed.
+      if (i010_buffer->StrideU() != i010_buffer->StrideV()) {
+        RTC_LOG(LS_ERROR) << "Libvpx requires the U and V strides to be equal.";
+        return WEBRTC_VIDEO_CODEC_ERROR;
+      }
+
       MaybeRewrapRawWithFormat(VPX_IMG_FMT_I42016, i010_buffer->width(),
                                i010_buffer->height());
       raw_->planes[VPX_PLANE_Y] = const_cast<uint8_t*>(
@@ -2159,6 +2166,13 @@ scoped_refptr<VideoFrameBuffer> LibvpxVp9Encoder::PrepareBufferForProfile0(
                                mapped_buffer->height());
       const I420BufferInterface* i420_buffer = mapped_buffer->GetI420();
       RTC_DCHECK(i420_buffer);
+
+      // TODO: crbug.com/492213293 - Remove once the root cause is fixed.
+      if (i420_buffer->StrideU() != i420_buffer->StrideV()) {
+        RTC_LOG(LS_ERROR) << "Libvpx requires the U and V strides to be equal.";
+        return {};
+      }
+
       raw_->planes[VPX_PLANE_Y] = const_cast<uint8_t*>(i420_buffer->DataY());
       raw_->planes[VPX_PLANE_U] = const_cast<uint8_t*>(i420_buffer->DataU());
       raw_->planes[VPX_PLANE_V] = const_cast<uint8_t*>(i420_buffer->DataV());
diff --git a/modules/video_coding/codecs/vp9/test/vp9_impl_unittest.cc b/modules/video_coding/codecs/vp9/test/vp9_impl_unittest.cc
index 9a5273e88b..c6ffdb941e 100644
--- a/modules/video_coding/codecs/vp9/test/vp9_impl_unittest.cc
+++ b/modules/video_coding/codecs/vp9/test/vp9_impl_unittest.cc
@@ -23,6 +23,7 @@
 #include "api/array_view.h"
 #include "api/environment/environment_factory.h"
 #include "api/field_trials.h"
+#include "api/make_ref_counted.h"
 #include "api/scoped_refptr.h"
 #include "api/test/create_frame_generator.h"
 #include "api/test/frame_generator_interface.h"
@@ -31,6 +32,7 @@
 #include "api/units/timestamp.h"
 #include "api/video/color_space.h"
 #include "api/video/encoded_image.h"
+#include "api/video/i420_buffer.h"
 #include "api/video/video_bitrate_allocation.h"
 #include "api/video/video_codec_constants.h"
 #include "api/video/video_codec_type.h"
@@ -269,6 +271,55 @@ TEST_P(TestVp9ImplForPixelFormat, CheckPresentationTimestamp) {
             encoded_frame.PresentationTimestamp()->us());
 }
 
+TEST_F(TestVp9Impl, RejectsI420FramesWithUnequalChromaStrides) {
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+
+  auto buffer = I420Buffer::Create(
+      /*width=*/kWidth,
+      /*height=*/kHeight,
+      /*stride_y=*/kWidth,
+      /*stride_u=*/(kWidth + 1) / 2,
+      /*stride_v=*/(kWidth + 1) / 2 + 1);
+
+  VideoFrame frame = VideoFrame::Builder()
+                         .set_video_frame_buffer(buffer)
+                         .set_rtp_timestamp(0)
+                         .build();
+
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_ERROR, encoder_->Encode(frame, nullptr));
+}
+
+TEST_F(TestVp9Impl, RejectsNativeFramesWithUnequalChromaStrides) {
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+
+  class FakeNativeBuffer : public VideoFrameBuffer {
+   public:
+    FakeNativeBuffer(int width, int height) : width_(width), height_(height) {}
+    Type type() const override { return Type::kNative; }
+    int width() const override { return width_; }
+    int height() const override { return height_; }
+    scoped_refptr<I420BufferInterface> ToI420() override {
+      return I420Buffer::Create(width_, height_, width_, (width_ + 1) / 2,
+                                (width_ + 1) / 2 + 1);
+    }
+
+   private:
+    int width_;
+    int height_;
+  };
+
+  auto buffer = make_ref_counted<FakeNativeBuffer>(kWidth, kHeight);
+
+  VideoFrame frame = VideoFrame::Builder()
+                         .set_video_frame_buffer(buffer)
+                         .set_rtp_timestamp(0)
+                         .build();
+
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_ERROR, encoder_->Encode(frame, nullptr));
+}
+
 TEST_F(TestVp9Impl, SwitchInputPixelFormatsWithoutReconfigure) {
   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Encode(NextInputFrame(), nullptr));
   EncodedImage encoded_frame;

From d460e60e1955be72f3479a944fabde401fe848a2 Mon Sep 17 00:00:00 2001
From: Sam Zackrisson <saza@webrtc.org>
Date: Fri, 13 Mar 2026 14:10:34 +0000
Subject: [PATCH 300/472] AEC3: Fix render buffer headroom calculation on
 underruns

The read and write pointers can meet. Currently, this causes an overflow in the headroom calculation (headroom == buffer size) which offsets the buffers needlessly. This causes a loss of transparency when using the neural residual echo estimation module.

Tested:
  1. audioproc_f is bitexact on a large set of aecdumps, without ML-REE
  2. I see improved transparency on a dump with underruns, with ML-REE

Bug: b:487988676, webrtc:442444736
Change-Id: I902a20f28c603f85a1c10ae5b525788e3a54c285
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456301
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47153}
---
 modules/audio_processing/aec3/render_buffer.h | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/modules/audio_processing/aec3/render_buffer.h b/modules/audio_processing/aec3/render_buffer.h
index f25416f828..6fc0317c6b 100644
--- a/modules/audio_processing/aec3/render_buffer.h
+++ b/modules/audio_processing/aec3/render_buffer.h
@@ -88,12 +88,12 @@ class RenderBuffer {
   int Headroom() const {
     // The write and read indices are decreased over time.
     int headroom =
-        fft_buffer_->write < fft_buffer_->read
+        fft_buffer_->write <= fft_buffer_->read
             ? fft_buffer_->read - fft_buffer_->write
             : fft_buffer_->size - fft_buffer_->write + fft_buffer_->read;
 
     RTC_DCHECK_LE(0, headroom);
-    RTC_DCHECK_GE(fft_buffer_->size, headroom);
+    RTC_DCHECK_GT(fft_buffer_->size, headroom);
 
     return headroom;
   }

From cdc36b7fbf44ad1953624cc9f289f199a8aaeb5a Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Fri, 13 Mar 2026 09:03:40 -0700
Subject: [PATCH 301/472] Roll chromium_revision b937d39273..4ab63722a5
 (1597657:1599101)

Change log: https://chromium.googlesource.com/chromium/src/+log/b937d39273..4ab63722a5
Full diff: https://chromium.googlesource.com/chromium/src/+/b937d39273..4ab63722a5

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/7d29c62736..618025ac18
* src/buildtools: https://chromium.googlesource.com/chromium/src/buildtools/+log/fef8d6947f..95ed44cf5f
* src/buildtools/linux64: git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9..git_revision:2941d2a26fa9cef3103f10e1dee5da7cd404008a
* src/buildtools/mac: git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9..git_revision:2941d2a26fa9cef3103f10e1dee5da7cd404008a
* src/buildtools/win: git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9..git_revision:2941d2a26fa9cef3103f10e1dee5da7cd404008a
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/94f693fcb6..d5e7161333
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/e7a1e88ecc..9a122b7537
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/a133c94b51..b8b47ad376
* src/third_party/android_build_tools/aapt2/cipd: vUP1cs7krVsNHbUJpwSDR7rvWuYLGzINVa9xopPpt74C..gsaUgZUqoyD0XG1E9-xesSWknHZINEuS00iSEncvlE0C
* src/third_party/android_build_tools/lint/cipd: 1N_9yErKkcZgqyxBghA0bb5L5AFEPZCt36Gv2o8_OAwC..UdJPLkDOu5joGtPdwsjc1Ltg5hcadLKPynhbd6NacsoC
* src/third_party/android_build_tools/manifest_merger/cipd: xT8A6Q_b44_iYKqAMvRkWmvID-ps-I6uBDfRTQv68bcC..KYQSw0Ej8zT_IRNL02DxEmk0nCvkykopkQhsGk_HE0EC
* src/third_party/android_deps/autorolled/cipd: umaYTBt4Vsa5QZYp57rLJDNzoXalsR2hPGwfYbpZYfwC..3VYcqPssBf1YI1iK0N2UUMsT2Ik5BlMrvm-wOVr1fMQC
* src/third_party/androidx/cipd: Hi79Lu6qQ-Fbcjwgf_f-NEYJweP4QdZqjJ0lXwc0PpcC..Mn6UxK4fOcnps2oJ1ikX7T1Nf93dtMBkX68uqIyoQ0sC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/eedf10c1f4..de8c7af5e7
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/e0ebf38a01..ac70fe6a2c
* src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/c8a5676846..c8ee656688
* src/third_party/cpuinfo/src: https://chromium.googlesource.com/external/github.com/pytorch/cpuinfo.git/+log/7364b490b5..7607ca5004
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/4ce8ba39a3..6b62257bd9
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/4387e32481..2a2456c873
* src/third_party/fuzztest/src: https://chromium.googlesource.com/external/github.com/google/fuzztest.git/+log/54ed362692..41e160679e
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/260c6fd37c..9751f44f3f
* src/third_party/harfbuzz-ng/src: https://chromium.googlesource.com/external/github.com/harfbuzz/harfbuzz.git/+log/fa2908bf16..5d4e96ad8d
* src/third_party/libaom/source/libaom: https://aomedia.googlesource.com/aom.git/+log/98ce0d2a61..0c15af06af
* src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/044963a253..76c8a4be56
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/4cead1b0fc..39802b97ff
* src/third_party/r8/cipd: CoaQUbD7pDX3OuETi_O5wkv_Y198RVIUaSobSEx9zfoC..UMLn7pE7DklTsk-SIiMAwch46m8rSF1vG2HQAbMuZvMC
* src/third_party/tflite/src: https://chromium.googlesource.com/external/github.com/tensorflow/tensorflow.git/+log/f920077085..d20369f4f2
* src/third_party/xnnpack/src: https://chromium.googlesource.com/external/github.com/google/XNNPACK.git/+log/26feacc3eb..15e04ff8b6
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/9765576870..f90bc5461d
* src/tools/luci-go: git_revision:eefaa865c43930859313a63d426740cdd0e22673..git_revision:09ad27bb7a42580550b9063ef8d0a39a5f01a338
* src/tools/luci-go: git_revision:eefaa865c43930859313a63d426740cdd0e22673..git_revision:09ad27bb7a42580550b9063ef8d0a39a5f01a338
* src/tools/luci-go: git_revision:eefaa865c43930859313a63d426740cdd0e22673..git_revision:09ad27bb7a42580550b9063ef8d0a39a5f01a338
DEPS diff: https://chromium.googlesource.com/chromium/src/+/b937d39273..4ab63722a5/DEPS

BUG=None

Change-Id: I1e1f000695f41909dd13f296ad4f7f4f70ea58c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456540
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47154}
---
 DEPS | 66 ++++++++++++++++++++++++++++++------------------------------
 1 file changed, 33 insertions(+), 33 deletions(-)

diff --git a/DEPS b/DEPS
index dffcecaef3..acad6f72ff 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': 'b937d39273b52034e8542dea4b040587be0a127f',
+  'chromium_revision': '4ab63722a5f5006a5d6fc9374d20da8e78d684d8',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,28 +69,28 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@7d29c62736304aeaac07df0b75546ffd2b75917a',
+    'https://chromium.googlesource.com/chromium/src/build@618025ac18566ba090c4b86328ea77f83a6c4cae',
   'src/buildtools':
-    'https://chromium.googlesource.com/chromium/src/buildtools@fef8d6947fa54202e8a00c58d5ac8fd15780bc33',
+    'https://chromium.googlesource.com/chromium/src/buildtools@95ed44cf5f06dbb5861030b91c9db9ccb4316762',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
   'src/examples/androidtests/third_party/gradle': {
     'url': 'https://chromium.googlesource.com/external/github.com/gradle/gradle.git@f2d1fb54a951d8b11d25748e4711bec8d128d7e3',
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@94f693fcb6dde967f11668a9e21eec1a4ec8b49c',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@d5e71613336debf95fff22f7ce2986f01eb781f9',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@e7a1e88ecc35a70f3d198674042753ec9e8e3650',
+    'https://chromium.googlesource.com/chromium/src/testing@9a122b7537744517c0128fbf55eb749bce0558fa',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@a133c94b51ac034d5f375cc1746e66dcdee57282',
+    'https://chromium.googlesource.com/chromium/src/third_party@b8b47ad3763aae32c8c8f56446e59b3bf51e32af',
 
   'src/buildtools/linux64': {
     'packages': [
       {
         'package': 'gn/gn/linux-${{arch}}',
-        'version': 'git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9',
+        'version': 'git_revision:2941d2a26fa9cef3103f10e1dee5da7cd404008a',
       }
     ],
     'dep_type': 'cipd',
@@ -100,7 +100,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/mac-${{arch}}',
-        'version': 'git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9',
+        'version': 'git_revision:2941d2a26fa9cef3103f10e1dee5da7cd404008a',
       }
     ],
     'dep_type': 'cipd',
@@ -110,7 +110,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/windows-amd64',
-        'version': 'git_revision:129ce6b9af1a93143acd8f8dea6fb8c6cb65abc9',
+        'version': 'git_revision:2941d2a26fa9cef3103f10e1dee5da7cd404008a',
       }
     ],
     'dep_type': 'cipd',
@@ -331,13 +331,13 @@ deps = {
   'src/third_party/clang-format/script':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@c2725e0622e1a86d55f14514f2177a39efea4a0e',
   'src/third_party/compiler-rt/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@c8a567684659fe05ede3a220965b94fd71c4a7cb',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@c8ee6566881410a159d66e99081779d28124fd3f',
   'src/third_party/libc++/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@7ab65651aed6802d2599dcb7a73b1f82d5179d05',
   'src/third_party/libc++abi/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@8f11bb1d4438d0239d0dfc1bd9456a9f31629dda',
   'src/third_party/llvm-libc/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@044963a253be3b8b2a63a1b2e0f3e361b01bc186',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@76c8a4be56babbe6104f5cd9eda25a95dd277b7a',
   'src/third_party/libunwind/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@db838d918570d4e381ecf9f5cc70a0098c9c2cd6',
 
@@ -402,7 +402,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/android_build_tools/aapt2',
-              'version': 'vUP1cs7krVsNHbUJpwSDR7rvWuYLGzINVa9xopPpt74C',
+              'version': 'gsaUgZUqoyD0XG1E9-xesSWknHZINEuS00iSEncvlE0C',
           },
       ],
       'condition': 'checkout_android',
@@ -457,7 +457,7 @@ deps = {
       'packages': [
           {
                'package': 'chromium/third_party/android_build_tools/lint',
-               'version': '1N_9yErKkcZgqyxBghA0bb5L5AFEPZCt36Gv2o8_OAwC',
+               'version': 'UdJPLkDOu5joGtPdwsjc1Ltg5hcadLKPynhbd6NacsoC',
           },
       ],
       'condition': 'checkout_android and non_git_source',
@@ -488,11 +488,11 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@eedf10c1f4ab44629e2fe3fe6f07e2fb0897c0f4',
+    'https://boringssl.googlesource.com/boringssl.git@de8c7af5e7c40153eb6cc0a79ff61a43f4bb03f9',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@8be0e3114685fcc1589561067282edf75ea1259a',
   'src/third_party/catapult':
-    'https://chromium.googlesource.com/catapult.git@e0ebf38a01214aba11f31daa1c743782def031d5',
+    'https://chromium.googlesource.com/catapult.git@ac70fe6a2cd8d142ff08e4692fa3230fb954a211',
   'src/third_party/ced/src': {
     'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5',
   },
@@ -505,13 +505,13 @@ deps = {
   'src/third_party/crc32c/src':
     'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558',
   'src/third_party/depot_tools':
-    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@4ce8ba39a3488397a2d1494f167020f21de502f3',
+    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@6b62257bd9fc9ee0e00d10712a0f3db63132ba2f',
   'src/third_party/ffmpeg':
     'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@946d97db8d906277085e361892b7efda5152e2f1',
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@260c6fd37c502e984836a446c5d90e4a1298036d',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@9751f44f3f6640142d242ddac084f7a80dce4aac',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -521,7 +521,7 @@ deps = {
   'src/third_party/freetype/src':
     'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@99b479dc34728936b006679a31e12b8cf432fc55',
   'src/third_party/harfbuzz-ng/src':
-    'https://chromium.googlesource.com/external/github.com/harfbuzz/harfbuzz.git@fa2908bf16d2ccd6623f4d575455fea72a1a722b',
+    'https://chromium.googlesource.com/external/github.com/harfbuzz/harfbuzz.git@5d4e96ad8d00fc871ffa17707b2ca08fa850e7d6',
   'src/third_party/google_benchmark/src': {
     'url': 'https://chromium.googlesource.com/external/github.com/google/benchmark.git@188e8278990a9069ffc84441cb5a024fd0bede37',
   },
@@ -593,7 +593,7 @@ deps = {
   'src/third_party/libFuzzer/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt/lib/fuzzer.git@bea408a6e01f0f7e6c82a43121fe3af4506c932e',
   'src/third_party/fuzztest/src':
-    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@54ed362692177e93df0fa202cae404b57694e828',
+    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@41e160679e3c010a860e674e7997e2b285c6e0a2',
   'src/third_party/libjpeg_turbo':
     'https://chromium.googlesource.com/chromium/deps/libjpeg_turbo.git@d1f5f2393e0d51f840207342ae86e55a86443288',
   'src/third_party/libsrtp':
@@ -601,7 +601,7 @@ deps = {
   'src/third_party/dav1d/libdav1d':
     'https://chromium.googlesource.com/external/github.com/videolan/dav1d.git@241a6b236a9c84441c872119c1f31b4074c0589d',
   'src/third_party/libaom/source/libaom':
-    'https://aomedia.googlesource.com/aom.git@98ce0d2a610f1465ea14d525bce9fa46d4b05c03',
+    'https://aomedia.googlesource.com/aom.git@0c15af06af108af03cbe237e4bffa4b700b76d81',
   'src/third_party/libgav1/src':
     Var('chromium_git') + '/codecs/libgav1.git' + '@' + '40f58ed32ff39071c3f2a51056dbc49a070af0dc',
   'src/third_party/libunwindstack': {
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '4cead1b0fc9987f994de97c7914e1f0c94047579',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '39802b97ff447996f184cfa2f749a89aff4a1e49',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -644,7 +644,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/r8',
-              'version': 'CoaQUbD7pDX3OuETi_O5wkv_Y198RVIUaSobSEx9zfoC',
+              'version': 'UMLn7pE7DklTsk-SIiMAwch46m8rSF1vG2HQAbMuZvMC',
           },
       ],
       'condition': 'checkout_android',
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@9765576870401b230a826e0e4f7a12851575306e',
+    'https://chromium.googlesource.com/chromium/src/tools@f90bc5461dc97886f567f9106341ee9d95bd9042',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'Hi79Lu6qQ-Fbcjwgf_f-NEYJweP4QdZqjJ0lXwc0PpcC',
+          'version': 'Mn6UxK4fOcnps2oJ1ikX7T1Nf93dtMBkX68uqIyoQ0sC',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -718,7 +718,7 @@ deps = {
       'packages': [
           {
                'package': 'chromium/third_party/android_build_tools/manifest_merger',
-               'version': 'xT8A6Q_b44_iYKqAMvRkWmvID-ps-I6uBDfRTQv68bcC',
+               'version': 'KYQSw0Ej8zT_IRNL02DxEmk0nCvkykopkQhsGk_HE0EC',
           },
       ],
       'condition': 'checkout_android',
@@ -786,7 +786,7 @@ deps = {
   },
 
   'src/third_party/tflite/src':
-    Var('chromium_git') + '/external/github.com/tensorflow/tensorflow.git' + '@' + 'f9200770858f5721b7fa95db7aa3e4cd712fe9df',
+    Var('chromium_git') + '/external/github.com/tensorflow/tensorflow.git' + '@' + 'd20369f4f226598800e0d406db567d2d1c17fa95',
 
   'src/third_party/turbine/cipd': {
       'packages': [
@@ -808,15 +808,15 @@ deps = {
       'packages': [
         {
           'package': 'infra/tools/luci/cas/${{platform}}',
-          'version': 'git_revision:eefaa865c43930859313a63d426740cdd0e22673',
+          'version': 'git_revision:09ad27bb7a42580550b9063ef8d0a39a5f01a338',
         },
         {
           'package': 'infra/tools/luci/isolate/${{platform}}',
-          'version': 'git_revision:eefaa865c43930859313a63d426740cdd0e22673',
+          'version': 'git_revision:09ad27bb7a42580550b9063ef8d0a39a5f01a338',
         },
         {
           'package': 'infra/tools/luci/swarming/${{platform}}',
-          'version': 'git_revision:eefaa865c43930859313a63d426740cdd0e22673',
+          'version': 'git_revision:09ad27bb7a42580550b9063ef8d0a39a5f01a338',
         }
       ],
       'dep_type': 'cipd',
@@ -842,7 +842,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/android_deps/autorolled',
-              'version': 'umaYTBt4Vsa5QZYp57rLJDNzoXalsR2hPGwfYbpZYfwC',
+              'version': '3VYcqPssBf1YI1iK0N2UUMsT2Ik5BlMrvm-wOVr1fMQC',
           },
       ],
       'condition': 'checkout_android and non_git_source',
@@ -852,7 +852,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/google/pthreadpool.git' + '@' + '9003ee6c137cea3b94161bd5c614fb43be523ee1',
 
   'src/third_party/xnnpack/src':
-    Var('chromium_git') + '/external/github.com/google/XNNPACK.git' + '@' + '26feacc3ebe30d1f3a82e1ff3938ef9859d74640',
+    Var('chromium_git') + '/external/github.com/google/XNNPACK.git' + '@' + '15e04ff8b61e0bc2118bad268447be5b2075aa66',
 
   'src/third_party/farmhash/src':
     Var('chromium_git') + '/external/github.com/google/farmhash.git' + '@' + '816a4ae622e964763ca0862d9dbd19324a1eaf45',
@@ -861,10 +861,10 @@ deps = {
     Var('chromium_git') + '/external/github.com/google/ruy.git' + '@' + '2af88863614a8298689cc52b1a47b3fcad7be835',
 
   'src/third_party/cpuinfo/src':
-    Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7364b490b5f78d58efe23ea76e74210fd6c3c76f',
+    Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7607ca500436b37ad23fb8d18614bec7796b68a7',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '4387e3248109bef2c07a8b69a2b8249d7af5dcf0',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '2a2456c873d6f4bf43a9adf919f13ad917d57c2d',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From 2ee2982cb31b5b0d8a5faf84d3f9cf6d9250395d Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Fri, 13 Mar 2026 13:06:28 -0700
Subject: [PATCH 302/472] Roll chromium_revision 4ab63722a5..fb6250728d
 (1599101:1599271)

Change log: https://chromium.googlesource.com/chromium/src/+log/4ab63722a5..fb6250728d
Full diff: https://chromium.googlesource.com/chromium/src/+/4ab63722a5..fb6250728d

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/618025ac18..f6edad717a
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/d5e7161333..b8d65e8ffc
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/9a122b7537..d6a35dd284
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/b8b47ad376..c0f75ebc2b
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/ac70fe6a2c..6ea42cfd29
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/9751f44f3f..5917a15d52
* src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/76c8a4be56..5336596e25
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/39802b97ff..1e165a6097
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/f90bc5461d..b351fad746
DEPS diff: https://chromium.googlesource.com/chromium/src/+/4ab63722a5..fb6250728d/DEPS

BUG=None

Change-Id: Ic8a7d9d0686520d85ecbc59581f653974a7b3690
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456560
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47155}
---
 DEPS | 20 ++++++++++----------
 1 file changed, 10 insertions(+), 10 deletions(-)

diff --git a/DEPS b/DEPS
index acad6f72ff..dd26c4617d 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '4ab63722a5f5006a5d6fc9374d20da8e78d684d8',
+  'chromium_revision': 'fb6250728d2950b9d5c81b59951fc5a543533020',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@618025ac18566ba090c4b86328ea77f83a6c4cae',
+    'https://chromium.googlesource.com/chromium/src/build@f6edad717a93248a82e594b9aac9a977e2b99194',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@95ed44cf5f06dbb5861030b91c9db9ccb4316762',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@d5e71613336debf95fff22f7ce2986f01eb781f9',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@b8d65e8ffc0cc998b17b20474a9d795cd0720a34',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@9a122b7537744517c0128fbf55eb749bce0558fa',
+    'https://chromium.googlesource.com/chromium/src/testing@d6a35dd28410cf457e4ef5bf27df2bb2dac13693',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@b8b47ad3763aae32c8c8f56446e59b3bf51e32af',
+    'https://chromium.googlesource.com/chromium/src/third_party@c0f75ebc2b2dffadb9e9d0278be337dbfc69551c',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -337,7 +337,7 @@ deps = {
   'src/third_party/libc++abi/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@8f11bb1d4438d0239d0dfc1bd9456a9f31629dda',
   'src/third_party/llvm-libc/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@76c8a4be56babbe6104f5cd9eda25a95dd277b7a',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@5336596e25df74e6f0f748ac65d8fe784c39b79b',
   'src/third_party/libunwind/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@db838d918570d4e381ecf9f5cc70a0098c9c2cd6',
 
@@ -492,7 +492,7 @@ deps = {
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@8be0e3114685fcc1589561067282edf75ea1259a',
   'src/third_party/catapult':
-    'https://chromium.googlesource.com/catapult.git@ac70fe6a2cd8d142ff08e4692fa3230fb954a211',
+    'https://chromium.googlesource.com/catapult.git@6ea42cfd29b4409cfdb830d1058d112359eaa9f3',
   'src/third_party/ced/src': {
     'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5',
   },
@@ -511,7 +511,7 @@ deps = {
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@9751f44f3f6640142d242ddac084f7a80dce4aac',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@5917a15d52c42c4bcae0a5706fdc34e5627c718d',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '39802b97ff447996f184cfa2f749a89aff4a1e49',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '1e165a6097073b6e1758a382f5e902d52962d78d',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@f90bc5461dc97886f567f9106341ee9d95bd9042',
+    'https://chromium.googlesource.com/chromium/src/tools@b351fad746e8056058166d7eb03b0f2f58cf65d7',
 
   'src/third_party/espresso': {
       'packages': [

From 2d45ac97748f5a5b2ab6cd12390d838e006fcf1e Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Fri, 13 Mar 2026 17:03:17 -0700
Subject: [PATCH 303/472] Roll chromium_revision fb6250728d..6fb43a733b
 (1599271:1599381)

Change log: https://chromium.googlesource.com/chromium/src/+log/fb6250728d..6fb43a733b
Full diff: https://chromium.googlesource.com/chromium/src/+/fb6250728d..6fb43a733b

Changed dependencies
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/b8d65e8ffc..c0f94dfabe
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/d6a35dd284..1e307ded23
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/c0f75ebc2b..c04ff6bae4
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/5917a15d52..d7bb3583ac
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/1e165a6097..9657154a81
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/b351fad746..9ae257371f
DEPS diff: https://chromium.googlesource.com/chromium/src/+/fb6250728d..6fb43a733b/DEPS

BUG=None

Change-Id: I341b296c3cfceb60a3b34bd3a41c512898424c16
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456601
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47156}
---
 DEPS | 14 +++++++-------
 1 file changed, 7 insertions(+), 7 deletions(-)

diff --git a/DEPS b/DEPS
index dd26c4617d..abe8ee3e91 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': 'fb6250728d2950b9d5c81b59951fc5a543533020',
+  'chromium_revision': '6fb43a733bb14ed79eea2d465ca78a22f5f1c891',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@b8d65e8ffc0cc998b17b20474a9d795cd0720a34',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@c0f94dfabea2ed699e7064ef062967a63ae2bfa5',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@d6a35dd28410cf457e4ef5bf27df2bb2dac13693',
+    'https://chromium.googlesource.com/chromium/src/testing@1e307ded23fd26c78a6594e3e922d3fbf8a63472',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@c0f75ebc2b2dffadb9e9d0278be337dbfc69551c',
+    'https://chromium.googlesource.com/chromium/src/third_party@c04ff6bae43cfea66cb165caca1f4c74f4d471ce',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -511,7 +511,7 @@ deps = {
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@5917a15d52c42c4bcae0a5706fdc34e5627c718d',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@d7bb3583ac34040c41954ae0f287c06e0227712f',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '1e165a6097073b6e1758a382f5e902d52962d78d',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '9657154a815d2581f774da1298ba386074c7c0ae',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@b351fad746e8056058166d7eb03b0f2f58cf65d7',
+    'https://chromium.googlesource.com/chromium/src/tools@9ae257371f6397a5259aa7528aee3b4d11a53149',
 
   'src/third_party/espresso': {
       'packages': [

From bee8d812628e997937e1b428659d10c5d1821045 Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Fri, 13 Mar 2026 21:12:08 -0700
Subject: [PATCH 304/472] Update WebRTC code version (2026-03-14T04:12:07).

Bug: None
Change-Id: I88c2189e4d05f80b7ae0cb1ee3ebe780194590ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456640
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47157}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index b2c4e5e121..aaeae422c9 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-12T04:11:50";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-14T04:12:07";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From d92a2b71352d0a6d7e0f29c801ddfe4afac7b1d5 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Sat, 14 Mar 2026 01:02:31 -0700
Subject: [PATCH 305/472] Roll chromium_revision 6fb43a733b..7c89d33808
 (1599381:1599481)

Change log: https://chromium.googlesource.com/chromium/src/+log/6fb43a733b..7c89d33808
Full diff: https://chromium.googlesource.com/chromium/src/+/6fb43a733b..7c89d33808

Changed dependencies
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/c0f94dfabe..c6927afeaf
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/c04ff6bae4..49e270aca3
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/de8c7af5e7..d61cb8f3df
* src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/c8ee656688..ea94b34eeb
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/2a2456c873..9ae0e0f195
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/d7bb3583ac..ff564156ae
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/9ae257371f..3da14e582d
DEPS diff: https://chromium.googlesource.com/chromium/src/+/6fb43a733b..7c89d33808/DEPS

BUG=None

Change-Id: I9d7cb6a10564d5187c9afdcae254be69f1798aa9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456661
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47158}
---
 DEPS | 16 ++++++++--------
 1 file changed, 8 insertions(+), 8 deletions(-)

diff --git a/DEPS b/DEPS
index abe8ee3e91..d26c730e83 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '6fb43a733bb14ed79eea2d465ca78a22f5f1c891',
+  'chromium_revision': '7c89d33808e551aed6122c1f324864784011c158',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@c0f94dfabea2ed699e7064ef062967a63ae2bfa5',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@c6927afeafc35502c4c7e121a8a289869936e50d',
     'condition': 'checkout_ios',
   },
   'src/testing':
     'https://chromium.googlesource.com/chromium/src/testing@1e307ded23fd26c78a6594e3e922d3fbf8a63472',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@c04ff6bae43cfea66cb165caca1f4c74f4d471ce',
+    'https://chromium.googlesource.com/chromium/src/third_party@49e270aca33f4bc476be355ee9c6b1b38e95ec6c',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -331,7 +331,7 @@ deps = {
   'src/third_party/clang-format/script':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@c2725e0622e1a86d55f14514f2177a39efea4a0e',
   'src/third_party/compiler-rt/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@c8ee6566881410a159d66e99081779d28124fd3f',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@ea94b34eeb6351bda6a3185cc5393263a7c5a1d4',
   'src/third_party/libc++/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@7ab65651aed6802d2599dcb7a73b1f82d5179d05',
   'src/third_party/libc++abi/src':
@@ -488,7 +488,7 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@de8c7af5e7c40153eb6cc0a79ff61a43f4bb03f9',
+    'https://boringssl.googlesource.com/boringssl.git@d61cb8f3df753b28098c13213afbeba42c475252',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@8be0e3114685fcc1589561067282edf75ea1259a',
   'src/third_party/catapult':
@@ -511,7 +511,7 @@ deps = {
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@d7bb3583ac34040c41954ae0f287c06e0227712f',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@ff564156ae9399be9cb1efd5084d910fb3dc6d97',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@9ae257371f6397a5259aa7528aee3b4d11a53149',
+    'https://chromium.googlesource.com/chromium/src/tools@3da14e582d503895a3dd224246c3742489180aa6',
 
   'src/third_party/espresso': {
       'packages': [
@@ -864,7 +864,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7607ca500436b37ad23fb8d18614bec7796b68a7',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '2a2456c873d6f4bf43a9adf919f13ad917d57c2d',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '9ae0e0f195e0a3b83e86d11b2ff6f4f8be254803',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From b7e65ec7f0a767ee6b4bfcd257656cf77efc0503 Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Sat, 14 Mar 2026 21:08:21 -0700
Subject: [PATCH 306/472] Update WebRTC code version (2026-03-15T04:07:45).

Bug: None
Change-Id: Ia0311829b7882117268623e1c4e110d8f6c57f08
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456880
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47159}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index aaeae422c9..fc656e2b72 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-14T04:12:07";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-15T04:07:45";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From b3456342cecd2e1c584ae45ef64ccf9f860d4f9c Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Sun, 15 Mar 2026 03:03:28 -0700
Subject: [PATCH 307/472] Roll chromium_revision 7c89d33808..ae756e4ad3
 (1599481:1599588)

Change log: https://chromium.googlesource.com/chromium/src/+log/7c89d33808..ae756e4ad3
Full diff: https://chromium.googlesource.com/chromium/src/+/7c89d33808..ae756e4ad3

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/f6edad717a..f05f529c87
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/c6927afeaf..4e7f4b668a
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/49e270aca3..549c1a8885
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/d61cb8f3df..75a1350717
* src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/ea94b34eeb..b93edfeca7
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/9ae0e0f195..929785924c
* src/third_party/libvpx/source/libvpx: https://chromium.googlesource.com/webm/libvpx.git/+log/9a2d3d1f46..e1dd14963a
* src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/5336596e25..fd9bad0830
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/9657154a81..ba9ab62963
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/3da14e582d..098576b942
DEPS diff: https://chromium.googlesource.com/chromium/src/+/7c89d33808..ae756e4ad3/DEPS

BUG=None

Change-Id: I3668172dc46d3ea00ef3c38ee1d54623a3f67c93
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456940
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47160}
---
 DEPS | 22 +++++++++++-----------
 1 file changed, 11 insertions(+), 11 deletions(-)

diff --git a/DEPS b/DEPS
index d26c730e83..d01602f91d 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '7c89d33808e551aed6122c1f324864784011c158',
+  'chromium_revision': 'ae756e4ad3ed3cfe56861f95a42ab8b4ea1786f1',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@f6edad717a93248a82e594b9aac9a977e2b99194',
+    'https://chromium.googlesource.com/chromium/src/build@f05f529c87391f7526e8d85305873c0b2aae9ffb',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@95ed44cf5f06dbb5861030b91c9db9ccb4316762',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@c6927afeafc35502c4c7e121a8a289869936e50d',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@4e7f4b668a84fdacc646a478b5e6e5c48fc2ba34',
     'condition': 'checkout_ios',
   },
   'src/testing':
     'https://chromium.googlesource.com/chromium/src/testing@1e307ded23fd26c78a6594e3e922d3fbf8a63472',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@49e270aca33f4bc476be355ee9c6b1b38e95ec6c',
+    'https://chromium.googlesource.com/chromium/src/third_party@549c1a8885dbd6dbbc1b0e3278d082114287d753',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -331,13 +331,13 @@ deps = {
   'src/third_party/clang-format/script':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@c2725e0622e1a86d55f14514f2177a39efea4a0e',
   'src/third_party/compiler-rt/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@ea94b34eeb6351bda6a3185cc5393263a7c5a1d4',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@b93edfeca73cf0d8537cf1c84ede88a9768c7bab',
   'src/third_party/libc++/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@7ab65651aed6802d2599dcb7a73b1f82d5179d05',
   'src/third_party/libc++abi/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@8f11bb1d4438d0239d0dfc1bd9456a9f31629dda',
   'src/third_party/llvm-libc/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@5336596e25df74e6f0f748ac65d8fe784c39b79b',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@fd9bad0830907c4729426180b276e67e6a05b4e0',
   'src/third_party/libunwind/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@db838d918570d4e381ecf9f5cc70a0098c9c2cd6',
 
@@ -488,7 +488,7 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@d61cb8f3df753b28098c13213afbeba42c475252',
+    'https://boringssl.googlesource.com/boringssl.git@75a1350717c122d5f1edf8938a9c17f1ce79f888',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@8be0e3114685fcc1589561067282edf75ea1259a',
   'src/third_party/catapult':
@@ -609,11 +609,11 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '9657154a815d2581f774da1298ba386074c7c0ae',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'ba9ab629630aa6a013247075f53493279c5b4897',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
-    'https://chromium.googlesource.com/webm/libvpx.git@9a2d3d1f46afbdfa9b9820a9fd3aacb084e65e2f',
+    'https://chromium.googlesource.com/webm/libvpx.git@e1dd14963a005523d0eb3a72954921934b3a8be7',
   'src/third_party/libyuv':
     'https://chromium.googlesource.com/libyuv/libyuv.git@30809ff64a9ca5e45f86439c0d474c2d3eef3d05',
   'src/third_party/lss': {
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@3da14e582d503895a3dd224246c3742489180aa6',
+    'https://chromium.googlesource.com/chromium/src/tools@098576b9425d9330c6757b5cb35b8e2c2c3c19e5',
 
   'src/third_party/espresso': {
       'packages': [
@@ -864,7 +864,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7607ca500436b37ad23fb8d18614bec7796b68a7',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '9ae0e0f195e0a3b83e86d11b2ff6f4f8be254803',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '929785924cf4c3d68946e9a24beebbd92b1d2b50',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From 1a2db2d3c3a64fcc1d2fbcaf4dceb3dac43198af Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Sun, 15 Mar 2026 21:14:32 -0700
Subject: [PATCH 308/472] Roll chromium_revision ae756e4ad3..88388785e2
 (1599588:1599692)

Change log: https://chromium.googlesource.com/chromium/src/+log/ae756e4ad3..88388785e2
Full diff: https://chromium.googlesource.com/chromium/src/+/ae756e4ad3..88388785e2

Changed dependencies
* siso_version: git_revision:bea0f245b6973702ddcb457090706d03145855f8..git_revision:c7dcf419b19a4a7377fa8e3e3f5fad5dffe10cf2
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/4e7f4b668a..abae91664f
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/1e307ded23..e887719b51
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/549c1a8885..2c52a3200f
* src/third_party/android_build_tools/error_prone/cipd: Y-wIN3nTyykC3fwieDSmffv26BOT2wXRVPktZSw_oLUC..ax2FOQ16-lz2R1o1P-xDhd2KOAFoY1wjZ-lQgZkygUYC
* src/third_party/android_build_tools/lint/cipd: UdJPLkDOu5joGtPdwsjc1Ltg5hcadLKPynhbd6NacsoC..lBgjWB8NdI2Mhnsy0SHkCyCZ3pbO1Qe4Zk0JZHs3yAIC
* src/third_party/android_build_tools/manifest_merger/cipd: KYQSw0Ej8zT_IRNL02DxEmk0nCvkykopkQhsGk_HE0EC..pdCSlKLBAxy6TfAEU6FCxCqA8rgvQ70Vhsodl6VKdKQC
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/ff564156ae..e94fcd7163
* src/third_party/kotlin_stdlib/cipd: KABHiJiUrNNzDUFrZt5xyABtjqZCS-xG3fIdF-pe19IC..mLhm2Qx7LKeQaTHlt5dOXbInxRZJzfhO2BhmUPklxBUC
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/ba9ab62963..9887f1b3cb
* src/third_party/siso/cipd: git_revision:bea0f245b6973702ddcb457090706d03145855f8..git_revision:c7dcf419b19a4a7377fa8e3e3f5fad5dffe10cf2
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/098576b942..f2abc599fc
DEPS diff: https://chromium.googlesource.com/chromium/src/+/ae756e4ad3..88388785e2/DEPS

BUG=None

Change-Id: Ie40eebc98232af8c272d03d7b88b1fe9faba8d30
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/457101
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47161}
---
 DEPS | 24 ++++++++++++------------
 1 file changed, 12 insertions(+), 12 deletions(-)

diff --git a/DEPS b/DEPS
index d01602f91d..145cd2bcec 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': 'ae756e4ad3ed3cfe56861f95a42ab8b4ea1786f1',
+  'chromium_revision': '88388785e2a97340d8c9cc1e74f24d4e71c092ab',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -50,7 +50,7 @@ vars = {
   # reclient CIPD package version
   'reclient_version': 're_client_version:0.185.0.db415f21-gomaip',
   # siso CIPD package version.
-  'siso_version': 'git_revision:bea0f245b6973702ddcb457090706d03145855f8',
+  'siso_version': 'git_revision:c7dcf419b19a4a7377fa8e3e3f5fad5dffe10cf2',
 
   # ninja CIPD package.
   'ninja_package': 'infra/3pp/tools/ninja/',
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@4e7f4b668a84fdacc646a478b5e6e5c48fc2ba34',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@abae91664fe2284bda82ed4be85a36e0ba2f8d56',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@1e307ded23fd26c78a6594e3e922d3fbf8a63472',
+    'https://chromium.googlesource.com/chromium/src/testing@e887719b51ac536e7af5e1a5dd71e974520e86db',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@549c1a8885dbd6dbbc1b0e3278d082114287d753',
+    'https://chromium.googlesource.com/chromium/src/third_party@2c52a3200fb53172df565b17089f0749a4f3ad27',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -435,7 +435,7 @@ deps = {
       'packages': [
           {
                'package': 'chromium/third_party/android_build_tools/error_prone',
-               'version': 'Y-wIN3nTyykC3fwieDSmffv26BOT2wXRVPktZSw_oLUC',
+               'version': 'ax2FOQ16-lz2R1o1P-xDhd2KOAFoY1wjZ-lQgZkygUYC',
           },
       ],
       'condition': 'checkout_android',
@@ -457,7 +457,7 @@ deps = {
       'packages': [
           {
                'package': 'chromium/third_party/android_build_tools/lint',
-               'version': 'UdJPLkDOu5joGtPdwsjc1Ltg5hcadLKPynhbd6NacsoC',
+               'version': 'lBgjWB8NdI2Mhnsy0SHkCyCZ3pbO1Qe4Zk0JZHs3yAIC',
           },
       ],
       'condition': 'checkout_android and non_git_source',
@@ -511,7 +511,7 @@ deps = {
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@ff564156ae9399be9cb1efd5084d910fb3dc6d97',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@e94fcd71631c896295fc7d6874165505e3673e2d',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -573,7 +573,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/kotlin_stdlib',
-              'version': 'KABHiJiUrNNzDUFrZt5xyABtjqZCS-xG3fIdF-pe19IC',
+              'version': 'mLhm2Qx7LKeQaTHlt5dOXbInxRZJzfhO2BhmUPklxBUC',
           },
       ],
       'condition': 'checkout_android',
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'ba9ab629630aa6a013247075f53493279c5b4897',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '9887f1b3cb6c1c9c8c72a515a882d5eeea0d13c6',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@098576b9425d9330c6757b5cb35b8e2c2c3c19e5',
+    'https://chromium.googlesource.com/chromium/src/tools@f2abc599fc2693206c7367c4ce17772156b813ec',
 
   'src/third_party/espresso': {
       'packages': [
@@ -718,7 +718,7 @@ deps = {
       'packages': [
           {
                'package': 'chromium/third_party/android_build_tools/manifest_merger',
-               'version': 'KYQSw0Ej8zT_IRNL02DxEmk0nCvkykopkQhsGk_HE0EC',
+               'version': 'pdCSlKLBAxy6TfAEU6FCxCqA8rgvQ70Vhsodl6VKdKQC',
           },
       ],
       'condition': 'checkout_android',

From d11adf6359e4fe872e4a6f836805e68c7f1a58d3 Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Sun, 15 Mar 2026 21:05:04 -0700
Subject: [PATCH 309/472] Update WebRTC code version (2026-03-16T04:04:41).

Bug: None
Change-Id: I00b2e9604e09ce9035dbc912f3249b280896e147
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/457100
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47162}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index fc656e2b72..5f3ef9fc12 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-15T04:07:45";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-16T04:04:41";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From d9b92fe1bc81f8172efdf737e97f2413ec99f850 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Lionel=20Koenig=20G=C3=A9las?= <lionelk@webrtc.org>
Date: Mon, 16 Mar 2026 07:55:05 +0100
Subject: [PATCH 310/472] Make max target input level for input controller
 -12dB

Launching the experiment "Increased maximum target" for
InputVolumeController: UMA on chrome showed that input level of APM is
roughly 1dBfs while output level remain unchanged and number of analog
gain change is reduced.

Bug: webrtc:457791164
Change-Id: Ic2a5befededfdfd309c7b5e817cdd18b1a1c9d9c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/457140
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47163}
---
 .../agc2/input_volume_controller.cc               | 15 +--------------
 .../agc2/input_volume_controller.h                |  3 +--
 2 files changed, 2 insertions(+), 16 deletions(-)

diff --git a/modules/audio_processing/agc2/input_volume_controller.cc b/modules/audio_processing/agc2/input_volume_controller.cc
index 34acde78d3..ea78a67a8a 100644
--- a/modules/audio_processing/agc2/input_volume_controller.cc
+++ b/modules/audio_processing/agc2/input_volume_controller.cc
@@ -133,19 +133,6 @@ int GetSpeechLevelRmsErrorDb(float speech_level_dbfs,
 
   return rms_error_db;
 }
-
-int GetTargetRangeMaxDbfs(const InputVolumeController::Config& config,
-                          const FieldTrialsView& field_trials) {
-  if (field_trials.IsEnabled("WebRTC-Agc2MaxSpeechLevelExperimental")) {
-    RTC_LOG(LS_INFO) << "AGC2 input volume controller using experimental "
-                        "maximum speech level";
-    return config.target_range_experimental_max_dbfs;
-  } else {
-    RTC_LOG(LS_INFO)
-        << "AGC2 input volume controller using default maximum speech level";
-    return config.target_range_max_dbfs;
-  }
-}
 }  // namespace
 
 MonoInputVolumeController::MonoInputVolumeController(
@@ -388,7 +375,7 @@ InputVolumeController::InputVolumeController(
       frames_since_clipped_(config.clipped_wait_frames),
       clipping_rate_log_counter_(0),
       clipping_rate_log_(0.0f),
-      target_range_max_dbfs_(GetTargetRangeMaxDbfs(config, field_trials)),
+      target_range_max_dbfs_(config.target_range_max_dbfs),
       target_range_min_dbfs_(config.target_range_min_dbfs),
       channel_controllers_(num_capture_channels) {
   RTC_LOG(LS_INFO)
diff --git a/modules/audio_processing/agc2/input_volume_controller.h b/modules/audio_processing/agc2/input_volume_controller.h
index dbb9f7f773..ab7f125d10 100644
--- a/modules/audio_processing/agc2/input_volume_controller.h
+++ b/modules/audio_processing/agc2/input_volume_controller.h
@@ -56,8 +56,7 @@ class InputVolumeController final {
     // adjustments are done based on the speech level. For speech levels below
     // and above the range, the targets `target_range_min_dbfs` and
     // `target_range_max_dbfs` are used, respectively.
-    int target_range_max_dbfs = -30;
-    int target_range_experimental_max_dbfs = -12;
+    int target_range_max_dbfs = -12;
     int target_range_min_dbfs = -50;
     // Number of wait frames between the recommended input volume updates.
     int update_input_volume_wait_frames = 100;

From 0e036e13f489856783ada4c761a8c8640d307db5 Mon Sep 17 00:00:00 2001
From: Fredrik Solenberg <solenberg@google.com>
Date: Fri, 13 Mar 2026 10:27:07 +0000
Subject: [PATCH 311/472] Fix integer overflow in WebRtc_CreateBuffer
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Added a check to ensure element_count * element_size does not overflow
size_t. Subsequent writes to the buffer could result in out-of-bounds
heap writes if the multiplication wraps around to a small number.

Bug: None
Change-Id: I0a37dfdf19fcee8b19765e8aec316d0886876d09
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456460
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47164}
---
 common_audio/ring_buffer.c           | 5 +++++
 common_audio/ring_buffer_unittest.cc | 3 +++
 2 files changed, 8 insertions(+)

diff --git a/common_audio/ring_buffer.c b/common_audio/ring_buffer.c
index a3fabd058f..c7ff782507 100644
--- a/common_audio/ring_buffer.c
+++ b/common_audio/ring_buffer.c
@@ -14,6 +14,7 @@
 #include "common_audio/ring_buffer.h"
 
 #include <stddef.h>  // size_t
+#include <stdint.h>
 #include <stdlib.h>
 #include <string.h>
 
@@ -56,6 +57,10 @@ RingBuffer* WebRtc_CreateBuffer(size_t element_count, size_t element_size) {
     return NULL;
   }
 
+  if (element_count > SIZE_MAX / element_size) {
+    return NULL;
+  }
+
   self = malloc(sizeof(RingBuffer));
   if (!self) {
     return NULL;
diff --git a/common_audio/ring_buffer_unittest.cc b/common_audio/ring_buffer_unittest.cc
index f93f042d9c..a87de9d0d0 100644
--- a/common_audio/ring_buffer_unittest.cc
+++ b/common_audio/ring_buffer_unittest.cc
@@ -11,6 +11,7 @@
 #include "common_audio/ring_buffer.h"
 
 #include <algorithm>
+#include <cstdint>
 #include <cstdio>
 #include <cstdlib>
 #include <ctime>
@@ -142,6 +143,8 @@ TEST(RingBufferTest, PassingNulltoReadBufferForcesMemcpy) {
 TEST(RingBufferTest, CreateHandlesErrors) {
   EXPECT_TRUE(WebRtc_CreateBuffer(0, 1) == nullptr);
   EXPECT_TRUE(WebRtc_CreateBuffer(1, 0) == nullptr);
+  EXPECT_TRUE(WebRtc_CreateBuffer(SIZE_MAX, 2) == nullptr);
+  EXPECT_TRUE(WebRtc_CreateBuffer(2, SIZE_MAX) == nullptr);
   RingBuffer* buffer = WebRtc_CreateBuffer(1, 1);
   EXPECT_TRUE(buffer != nullptr);
   WebRtc_FreeBuffer(buffer);

From a7f267f8b01ad6a2a11eced7267e61eff932b2ab Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Fri, 13 Mar 2026 19:35:01 +0100
Subject: [PATCH 312/472] Fix missing mediaSourceId in simulcast outbound-rtp
 stats
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

When simulcast is enabled, only the primary SSRC was mapped to the
sender's attachment ID in TrackMediaInfoMap. This resulted in missing
mediaSourceId properties for secondary simulcast outgoing stats. Updated
TrackMediaInfoMap to map all valid SSRCs provided in the sender's
RtpParameters.

Bug: webrtc:492108787
Fixes: webrtc:492108787
Change-Id: I688c169628675d7be3114e75b978a4ad4343dcc0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456520
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47165}
---
 pc/peer_connection.cc               | 10 ++++++++--
 pc/rtc_stats_collector.cc           | 12 ++++++++++--
 pc/rtc_stats_collector.h            |  2 ++
 pc/rtc_stats_collector_unittest.cc  | 29 ++++++++++++++++++++++++++---
 pc/track_media_info_map.cc          | 22 +++++++++++++++++-----
 pc/track_media_info_map.h           |  1 +
 pc/track_media_info_map_unittest.cc |  4 +++-
 7 files changed, 67 insertions(+), 13 deletions(-)

diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc
index e2a4adedc9..78a914253c 100644
--- a/pc/peer_connection.cc
+++ b/pc/peer_connection.cc
@@ -1375,7 +1375,10 @@ void PeerConnection::GetStats(
   TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
   RTC_DCHECK_RUN_ON(signaling_thread());
   RTC_DCHECK(callback);
-  RTC_DCHECK_DISALLOW_THREAD_BLOCKING_CALLS();
+  // TODO: bugs.webrtc.org/492108787 - Re-enable this when fixing the
+  // call to `sender->internal()->GetParametersInternal()` which can
+  // block when cached parameters aren't available.
+  // RTC_DCHECK_DISALLOW_THREAD_BLOCKING_CALLS();
   scoped_refptr<RtpSenderInternal> internal_sender;
   if (selector) {
     for (const auto& proxy_transceiver :
@@ -1407,7 +1410,10 @@ void PeerConnection::GetStats(
   TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
   RTC_DCHECK_RUN_ON(signaling_thread());
   RTC_DCHECK(callback);
-  RTC_DCHECK_DISALLOW_THREAD_BLOCKING_CALLS();
+  // TODO: bugs.webrtc.org/492108787 - Re-enable this when fixing the
+  // call to `sender->internal()->GetParametersInternal()` which can
+  // block when cached parameters aren't available.
+  // RTC_DCHECK_DISALLOW_THREAD_BLOCKING_CALLS();
   scoped_refptr<RtpReceiverInternal> internal_receiver;
   if (selector) {
     for (const auto& proxy_transceiver :
diff --git a/pc/rtc_stats_collector.cc b/pc/rtc_stats_collector.cc
index 517d6f259f..ea0ae61435 100644
--- a/pc/rtc_stats_collector.cc
+++ b/pc/rtc_stats_collector.cc
@@ -2269,6 +2269,7 @@ RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w() {
 
   std::vector<RtpTransceiverStatsInfo> transceiver_stats_infos;
   std::vector<TransceiverReferences> transceiver_references;
+  std::vector<std::vector<RtpParameters>> all_sender_parameters;
   // These are used to invoke GetStats for all the media channels together in
   // one worker thread hop.
   std::map<VoiceMediaSendChannelInterface*, VoiceMediaSendInfo>
@@ -2296,12 +2297,16 @@ RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w() {
                                    scoped_refptr<RtpTransceiver>(transceiver)};
 
     RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
+    std::vector<RtpParameters> sender_parameters;
     for (const auto& sender : transceiver->senders()) {
       stats.sender_infos.push_back(
           {.ssrc = sender->ssrc(),
            .attachment_id = sender->internal()->AttachmentId(),
            .media_type = sender->media_type()});
+      sender_parameters.push_back(sender->internal()->GetParametersInternal(
+          /*may_use_cache=*/true, /*with_all_layers=*/true));
     }
+    all_sender_parameters.push_back(std::move(sender_parameters));
     for (const auto& receiver : transceiver->receivers()) {
       stats.receiver_infos.push_back(
           {.track_id = receiver->track() ? receiver->track()->id() : "",
@@ -2339,6 +2344,7 @@ RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w() {
   // which also needs info from the worker thread.
   return [this, transceiver_stats_infos = std::move(transceiver_stats_infos),
           transceiver_references = std::move(transceiver_references),
+          all_sender_parameters = std::move(all_sender_parameters),
           voice_send_stats = std::move(voice_send_stats),
           voice_receive_stats = std::move(voice_receive_stats),
           video_send_stats = std::move(video_send_stats),
@@ -2348,6 +2354,7 @@ RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w() {
     WorkerThreadResult worker_result;
     worker_result.results.transceiver_stats_infos =
         std::move(transceiver_stats_infos);
+    worker_result.results.sender_parameters = std::move(all_sender_parameters);
     worker_result.transceiver_references = std::move(transceiver_references);
 
     for (auto& pair : voice_send_stats) {
@@ -2415,8 +2422,9 @@ RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w() {
 
       stats.track_media_info_map = std::make_unique<TrackMediaInfoMap>(
           std::move(voice_media_info), std::move(video_media_info),
-          std::move(stats.sender_infos), std::move(stats.receiver_infos),
-          std::move(receiver_parameters));
+          std::move(stats.sender_infos),
+          std::move(worker_result.results.sender_parameters[i]),
+          std::move(stats.receiver_infos), std::move(receiver_parameters));
       if (stats.media_type == MediaType::AUDIO) {
         has_audio_receiver |= stats.has_receivers;
       }
diff --git a/pc/rtc_stats_collector.h b/pc/rtc_stats_collector.h
index 5e9bbfe463..5103c6e3cb 100644
--- a/pc/rtc_stats_collector.h
+++ b/pc/rtc_stats_collector.h
@@ -26,6 +26,7 @@
 #include "api/data_channel_interface.h"
 #include "api/environment/environment.h"
 #include "api/media_types.h"
+#include "api/rtp_parameters.h"
 #include "api/rtp_transceiver_direction.h"
 #include "api/scoped_refptr.h"
 #include "api/stats/rtc_stats_collector_callback.h"
@@ -155,6 +156,7 @@ class RTCStatsCollector {
  private:
   struct StatsGatheringResults {
     std::vector<RtpTransceiverStatsInfo> transceiver_stats_infos;
+    std::vector<std::vector<RtpParameters>> sender_parameters;
     Call::Stats call_stats;
     std::optional<AudioDeviceModule::Stats> audio_device_stats;
   };
diff --git a/pc/rtc_stats_collector_unittest.cc b/pc/rtc_stats_collector_unittest.cc
index 55f19763d2..e3ffa34aac 100644
--- a/pc/rtc_stats_collector_unittest.cc
+++ b/pc/rtc_stats_collector_unittest.cc
@@ -2690,9 +2690,17 @@ TEST_P(RTCStatsCollectorTest, CollectRTCOutboundRtpStreamStats_Video) {
   auto video_media_channels =
       pc_->AddVideoChannel("VideoMid", "TransportName", video_media_info);
   RTC_ALLOW_PLAN_B_DEPRECATION_END();
-  stats_->SetupLocalTrackAndSender(MediaType::VIDEO, "LocalVideoTrackID", 1,
-                                   true,
-                                   /*attachment_id=*/50);
+  auto video_sender = stats_->SetupLocalTrackAndSender(
+      MediaType::VIDEO, "LocalVideoTrackID", 1, true,
+      /*attachment_id=*/50);
+  EXPECT_CALL(*video_sender, GetParametersInternal(_, _)).WillRepeatedly([] {
+    RtpParameters params;
+    params.encodings.push_back(RtpEncodingParameters());
+    params.encodings[0].ssrc = 1;
+    params.encodings.push_back(RtpEncodingParameters());
+    params.encodings[1].ssrc = 2;
+    return params;
+  });
 
   scoped_refptr<const RTCStatsReport> report =
       stats_->GetStatsReport(main_thread_);
@@ -2757,6 +2765,21 @@ TEST_P(RTCStatsCollectorTest, CollectRTCOutboundRtpStreamStats_Video) {
       report->Get(expected_video.id())->cast_to<RTCOutboundRtpStreamStats>(),
       expected_video);
 
+  // Find the sub-stats for the second simulcast outbound RTP stream and verify
+  // that its `media_source_id` is set.
+  std::string id_of_second_ssrc;
+  for (const auto* outbound_rtp : stats_of_my_type) {
+    if (outbound_rtp->ssrc.value_or(0) == 2) {
+      id_of_second_ssrc = outbound_rtp->id();
+      break;
+    }
+  }
+  ASSERT_TRUE(report->Get(id_of_second_ssrc));
+  auto& second_video =
+      report->Get(id_of_second_ssrc)->cast_to<RTCOutboundRtpStreamStats>();
+  EXPECT_TRUE(second_video.media_source_id.has_value());
+  EXPECT_EQ(*second_video.media_source_id, "SV50");
+
   // Set previously undefined values and "GetStats" again.
   video_media_info.senders[0].qp_sum = 9;
   expected_video.qp_sum = 9;
diff --git a/pc/track_media_info_map.cc b/pc/track_media_info_map.cc
index 7150f34b16..91d12c509b 100644
--- a/pc/track_media_info_map.cc
+++ b/pc/track_media_info_map.cc
@@ -41,13 +41,24 @@ const V* FindAddressOrNull(const flat_map<K, V>& map, const K& key) {
 
 flat_map<uint32_t, int> GetSenderAttachmentIds(
     std::span<const TrackMediaInfoMap::RtpSenderSignalInfo> senders,
+    std::span<const RtpParameters> sender_parameters,
     MediaType media_type) {
+  RTC_DCHECK_EQ(senders.size(), sender_parameters.size());
   flat_map<uint32_t, int> result;
-  for (const auto& sender : senders) {
-    if (sender.media_type != media_type || sender.ssrc == 0) {
+  for (size_t i = 0; i < senders.size(); ++i) {
+    const auto& sender = senders[i];
+    if (sender.media_type != media_type) {
       continue;
     }
-    result[sender.ssrc] = sender.attachment_id;
+    const auto& parameters = sender_parameters[i];
+    if (sender.ssrc != 0) {
+      result[sender.ssrc] = sender.attachment_id;
+    }
+    for (const auto& encoding : parameters.encodings) {
+      if (encoding.ssrc) {
+        result[*encoding.ssrc] = sender.attachment_id;
+      }
+    }
   }
   return result;
 }
@@ -132,14 +143,15 @@ TrackMediaInfoMap::TrackMediaInfoMap(
     std::optional<VoiceMediaInfo> voice_media_info,
     std::optional<VideoMediaInfo> video_media_info,
     std::span<const RtpSenderSignalInfo> senders,
+    std::span<const RtpParameters> sender_parameters,
     std::span<const RtpReceiverSignalInfo> receivers,
     std::span<const RtpParameters> receiver_parameters)
     : voice_media_info_(std::move(voice_media_info)),
       video_media_info_(std::move(video_media_info)),
       audio_sender_attachment_id_by_ssrc_(
-          GetSenderAttachmentIds(senders, MediaType::AUDIO)),
+          GetSenderAttachmentIds(senders, sender_parameters, MediaType::AUDIO)),
       video_sender_attachment_id_by_ssrc_(
-          GetSenderAttachmentIds(senders, MediaType::VIDEO)),
+          GetSenderAttachmentIds(senders, sender_parameters, MediaType::VIDEO)),
       audio_receiver_attachment_id_by_ssrc_(
           GetReceiverAttachmentIds(receivers,
                                    receiver_parameters,
diff --git a/pc/track_media_info_map.h b/pc/track_media_info_map.h
index 8e63aa8143..f68ec7504c 100644
--- a/pc/track_media_info_map.h
+++ b/pc/track_media_info_map.h
@@ -46,6 +46,7 @@ class TrackMediaInfoMap {
   TrackMediaInfoMap(std::optional<VoiceMediaInfo> voice_media_info,
                     std::optional<VideoMediaInfo> video_media_info,
                     std::span<const RtpSenderSignalInfo> senders,
+                    std::span<const RtpParameters> sender_parameters,
                     std::span<const RtpReceiverSignalInfo> receivers,
                     std::span<const RtpParameters> receiver_parameters);
 
diff --git a/pc/track_media_info_map_unittest.cc b/pc/track_media_info_map_unittest.cc
index 01b8cedd2c..fd46ed5759 100644
--- a/pc/track_media_info_map_unittest.cc
+++ b/pc/track_media_info_map_unittest.cc
@@ -175,12 +175,14 @@ class TrackMediaInfoMapTest : public ::testing::Test {
   // into the map.
   TrackMediaInfoMap InitializeMap() {
     std::vector<TrackMediaInfoMap::RtpSenderSignalInfo> sender_infos;
+    std::vector<RtpParameters> sender_params;
     for (const auto& sender : rtp_senders_) {
       sender_infos.push_back({
           .ssrc = sender->ssrc(),
           .attachment_id = sender->AttachmentId(),
           .media_type = sender->media_type(),
       });
+      sender_params.push_back(sender->GetParameters());
     }
     std::vector<TrackMediaInfoMap::RtpReceiverSignalInfo> receiver_infos;
     std::vector<RtpParameters> receiver_params;
@@ -204,7 +206,7 @@ class TrackMediaInfoMapTest : public ::testing::Test {
     }
     return TrackMediaInfoMap(std::move(voice_media_info),
                              std::move(video_media_info), sender_infos,
-                             receiver_infos, receiver_params);
+                             sender_params, receiver_infos, receiver_params);
   }
 
  private:

From 254fefeb3a756fd2e829f714e020d19d4bad1410 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Fri, 13 Mar 2026 23:22:26 +0100
Subject: [PATCH 313/472] Fetch sender parameters asynchronously during
 GetStats
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Resolve an issue where calling GetStats could block while fetching send
parameters. Previously, fetching sender parameters relied on
`GetParametersInternal`, which blocks when cached parameters are
unavailable.

This moves the retrieval of sender RTP parameters to the worker thread.
Parameters are now fetched directly from the media send channel during
the asynchronous stats collection phase.

This change avoids the thread blocking on the signaling thread and
re-enables the `RTC_DCHECK_DISALLOW_THREAD_BLOCKING_CALLS` checks.

Bug: webrtc:492108787
Change-Id: I941bed643ff46c4e52b96ae9fbc08a64d7cff7c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456580
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47166}
---
 pc/peer_connection.cc              | 10 +----
 pc/rtc_stats_collector.cc          | 32 ++++++++++-----
 pc/rtc_stats_collector.h           |  2 +-
 pc/rtc_stats_collector_unittest.cc | 65 ++++++++++++++++++++++++++----
 4 files changed, 83 insertions(+), 26 deletions(-)

diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc
index 78a914253c..e2a4adedc9 100644
--- a/pc/peer_connection.cc
+++ b/pc/peer_connection.cc
@@ -1375,10 +1375,7 @@ void PeerConnection::GetStats(
   TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
   RTC_DCHECK_RUN_ON(signaling_thread());
   RTC_DCHECK(callback);
-  // TODO: bugs.webrtc.org/492108787 - Re-enable this when fixing the
-  // call to `sender->internal()->GetParametersInternal()` which can
-  // block when cached parameters aren't available.
-  // RTC_DCHECK_DISALLOW_THREAD_BLOCKING_CALLS();
+  RTC_DCHECK_DISALLOW_THREAD_BLOCKING_CALLS();
   scoped_refptr<RtpSenderInternal> internal_sender;
   if (selector) {
     for (const auto& proxy_transceiver :
@@ -1410,10 +1407,7 @@ void PeerConnection::GetStats(
   TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
   RTC_DCHECK_RUN_ON(signaling_thread());
   RTC_DCHECK(callback);
-  // TODO: bugs.webrtc.org/492108787 - Re-enable this when fixing the
-  // call to `sender->internal()->GetParametersInternal()` which can
-  // block when cached parameters aren't available.
-  // RTC_DCHECK_DISALLOW_THREAD_BLOCKING_CALLS();
+  RTC_DCHECK_DISALLOW_THREAD_BLOCKING_CALLS();
   scoped_refptr<RtpReceiverInternal> internal_receiver;
   if (selector) {
     for (const auto& proxy_transceiver :
diff --git a/pc/rtc_stats_collector.cc b/pc/rtc_stats_collector.cc
index ea0ae61435..6a2325b6dd 100644
--- a/pc/rtc_stats_collector.cc
+++ b/pc/rtc_stats_collector.cc
@@ -2266,10 +2266,10 @@ RTCStatsCollector::PrepareTransportCertificateStats_n(
 absl::AnyInvocable<RTCStatsCollector::WorkerThreadResult()>
 RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w() {
   RTC_DCHECK_RUN_ON(signaling_thread_);
+  RTC_DCHECK_DISALLOW_THREAD_BLOCKING_CALLS();
 
   std::vector<RtpTransceiverStatsInfo> transceiver_stats_infos;
   std::vector<TransceiverReferences> transceiver_references;
-  std::vector<std::vector<RtpParameters>> all_sender_parameters;
   // These are used to invoke GetStats for all the media channels together in
   // one worker thread hop.
   std::map<VoiceMediaSendChannelInterface*, VoiceMediaSendInfo>
@@ -2297,16 +2297,12 @@ RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w() {
                                    scoped_refptr<RtpTransceiver>(transceiver)};
 
     RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
-    std::vector<RtpParameters> sender_parameters;
     for (const auto& sender : transceiver->senders()) {
       stats.sender_infos.push_back(
           {.ssrc = sender->ssrc(),
            .attachment_id = sender->internal()->AttachmentId(),
            .media_type = sender->media_type()});
-      sender_parameters.push_back(sender->internal()->GetParametersInternal(
-          /*may_use_cache=*/true, /*with_all_layers=*/true));
     }
-    all_sender_parameters.push_back(std::move(sender_parameters));
     for (const auto& receiver : transceiver->receivers()) {
       stats.receiver_infos.push_back(
           {.track_id = receiver->track() ? receiver->track()->id() : "",
@@ -2344,7 +2340,6 @@ RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w() {
   // which also needs info from the worker thread.
   return [this, transceiver_stats_infos = std::move(transceiver_stats_infos),
           transceiver_references = std::move(transceiver_references),
-          all_sender_parameters = std::move(all_sender_parameters),
           voice_send_stats = std::move(voice_send_stats),
           voice_receive_stats = std::move(voice_receive_stats),
           video_send_stats = std::move(video_send_stats),
@@ -2354,7 +2349,6 @@ RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w() {
     WorkerThreadResult worker_result;
     worker_result.results.transceiver_stats_infos =
         std::move(transceiver_stats_infos);
-    worker_result.results.sender_parameters = std::move(all_sender_parameters);
     worker_result.transceiver_references = std::move(transceiver_references);
 
     for (auto& pair : voice_send_stats) {
@@ -2394,6 +2388,27 @@ RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w() {
         continue;
       }
 
+      std::vector<RtpParameters> sender_parameters;
+      for (const auto& sender : stats.sender_infos) {
+        RtpParameters params;
+        if (sender.ssrc != 0 && stats.has_channel) {
+          auto& transceiver = refs.transceiver;
+          MediaSendChannelInterface* media_channel =
+              (stats.media_type == MediaType::AUDIO)
+                  ? static_cast<MediaSendChannelInterface*>(
+                        transceiver->voice_media_send_channel())
+                  : static_cast<MediaSendChannelInterface*>(
+                        transceiver->video_media_send_channel());
+          if (media_channel) {
+            params = media_channel->GetRtpSendParameters(sender.ssrc);
+          }
+        }
+        // `sender_parameters` must be the same size as `stats.sender_infos` to
+        // allow 1:1 mapping when TrackMediaInfoMap is constructed. Empty params
+        // are safely ignored since they have no encodings/SSRCs.
+        sender_parameters.push_back(std::move(params));
+      }
+
       std::vector<RtpParameters> receiver_parameters;
       for (const auto& receiver : refs.receivers) {
         receiver_parameters.push_back(receiver->GetParameters());
@@ -2422,8 +2437,7 @@ RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w() {
 
       stats.track_media_info_map = std::make_unique<TrackMediaInfoMap>(
           std::move(voice_media_info), std::move(video_media_info),
-          std::move(stats.sender_infos),
-          std::move(worker_result.results.sender_parameters[i]),
+          std::move(stats.sender_infos), std::move(sender_parameters),
           std::move(stats.receiver_infos), std::move(receiver_parameters));
       if (stats.media_type == MediaType::AUDIO) {
         has_audio_receiver |= stats.has_receivers;
diff --git a/pc/rtc_stats_collector.h b/pc/rtc_stats_collector.h
index 5103c6e3cb..c873006066 100644
--- a/pc/rtc_stats_collector.h
+++ b/pc/rtc_stats_collector.h
@@ -156,13 +156,13 @@ class RTCStatsCollector {
  private:
   struct StatsGatheringResults {
     std::vector<RtpTransceiverStatsInfo> transceiver_stats_infos;
-    std::vector<std::vector<RtpParameters>> sender_parameters;
     Call::Stats call_stats;
     std::optional<AudioDeviceModule::Stats> audio_device_stats;
   };
 
   struct WorkerThreadResult {
     StatsGatheringResults results;
+    std::vector<std::vector<RtpParameters>> sender_parameters;
     std::vector<TransceiverReferences> transceiver_references;
   };
   struct CollectionContext;
diff --git a/pc/rtc_stats_collector_unittest.cc b/pc/rtc_stats_collector_unittest.cc
index e3ffa34aac..c044f87f0c 100644
--- a/pc/rtc_stats_collector_unittest.cc
+++ b/pc/rtc_stats_collector_unittest.cc
@@ -26,6 +26,7 @@
 #include "absl/strings/str_replace.h"
 #include "api/audio/audio_device.h"
 #include "api/audio/audio_processing_statistics.h"
+#include "api/audio_options.h"
 #include "api/candidate.h"
 #include "api/data_channel_interface.h"
 #include "api/dtls_transport_interface.h"
@@ -54,6 +55,7 @@
 #include "common_video/include/quality_limitation_reason.h"
 #include "json/reader.h"
 #include "json/value.h"
+#include "media/base/fake_media_engine.h"
 #include "media/base/media_channel.h"
 #include "media/base/stream_params.h"
 #include "modules/rtp_rtcp/include/report_block_data.h"
@@ -65,6 +67,7 @@
 #include "p2p/base/transport_description.h"
 #include "pc/media_stream.h"
 #include "pc/peer_connection_internal.h"
+#include "pc/rtp_sender.h"
 #include "pc/sctp_data_channel.h"
 #include "pc/stream_collection.h"
 #include "pc/test/fake_audio_track.h"
@@ -2693,14 +2696,14 @@ TEST_P(RTCStatsCollectorTest, CollectRTCOutboundRtpStreamStats_Video) {
   auto video_sender = stats_->SetupLocalTrackAndSender(
       MediaType::VIDEO, "LocalVideoTrackID", 1, true,
       /*attachment_id=*/50);
-  EXPECT_CALL(*video_sender, GetParametersInternal(_, _)).WillRepeatedly([] {
-    RtpParameters params;
-    params.encodings.push_back(RtpEncodingParameters());
-    params.encodings[0].ssrc = 1;
-    params.encodings.push_back(RtpEncodingParameters());
-    params.encodings[1].ssrc = 2;
-    return params;
-  });
+
+  // Set up the fake video channel to return the expected RtpParameters
+  // containing both simulcast encodings for the primary SSRC.
+  StreamParams sp;
+  sp.add_ssrc(1);
+  sp.add_ssrc(2);
+  sp.ssrc_groups.push_back(SsrcGroup(kSimSsrcGroupSemantics, {1, 2}));
+  video_media_channels.first->AddSendStream(sp);
 
   scoped_refptr<const RTCStatsReport> report =
       stats_->GetStatsReport(main_thread_);
@@ -3823,6 +3826,52 @@ TEST_P(RTCStatsCollectorTest, GetStatsWithReceiverSelector) {
   EXPECT_FALSE(receiver_report->Get(graph.media_source_id));
 }
 
+TEST_P(RTCStatsCollectorTest,
+       GetStatsDoesNotBlockWhenFetchingSenderParameters) {
+  // We must create a new FakePeerConnectionForStats with a separate worker
+  // thread because the test base uses the current thread for all three, which
+  // bypasses the thread hop and avoids the
+  // RTC_DCHECK_DISALLOW_THREAD_BLOCKING_CALLS check.
+  auto worker_thread = Thread::Create();
+  worker_thread->Start();
+  pc_ = make_ref_counted<FakePeerConnectionForStats>(env_, worker_thread.get());
+  SetStatsTimestampWithEnvironmentClock(GetParam());
+  // Create a track and a real sender to ensure we test the actual blocking
+  // behavior in RtpSenderBase::GetParametersInternal.
+  scoped_refptr<MediaStreamTrackInterface> track = CreateFakeTrack(
+      MediaType::AUDIO, "audioTrack", MediaStreamTrackInterface::kLive);
+
+  auto fake_media_channel = std::make_unique<FakeVoiceMediaSendChannel>(
+      webrtc::AudioOptions(), pc_->network_thread());
+
+  scoped_refptr<AudioRtpSender> sender = AudioRtpSender::Create(
+      env_, pc_->signaling_thread(), pc_->worker_thread(), "sender_id",
+      /*stats=*/nullptr, /*set_streams_observer=*/nullptr,
+      /*media_channel=*/nullptr);
+
+  worker_thread->BlockingCall([&] {
+    fake_media_channel->AddSendStream(StreamParams::CreateLegacy(1234));
+    sender->SetMediaChannel(fake_media_channel.get());
+  });
+
+  sender->SetSsrc(1234);
+  sender->SetTrack(track.get());
+
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
+  pc_->AddSender(sender);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
+
+  scoped_refptr<const RTCStatsReport> report;
+  stats_->stats_collector().GetStatsReport(
+      RTCStatsObtainer::Create(&report, [this] { main_thread_.Quit(); }));
+  main_thread_.Run();
+
+  sender->Stop();
+  sender = nullptr;
+  stats_.reset();
+  pc_ = nullptr;
+}
+
 TEST_P(RTCStatsCollectorTest, GetStatsWithNullSenderSelector) {
   ExampleStatsGraph graph = SetupExampleStatsGraphForSelectorTests();
   scoped_refptr<const RTCStatsReport> empty_report;

From c69418852fda1517ff8f4d0dfe41427eb0d0100b Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Jes=C3=BAs=20de=20Vicente=20Pe=C3=B1a?=
 <devicentepena@webrtc.org>
Date: Fri, 13 Mar 2026 08:51:32 +0100
Subject: [PATCH 314/472] Re-adding apm_data_dumper.h to the list of unsafe
 buffer exceptions.

This is breaking the compilation of
[audioproc_f](https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/rtc_tools/BUILD.gn;drc=63cadb1c4ff5a5b10836ffa694e9269548072321;l=591)
together with apm_debug_dump = true.

Bug: webrtc:478086887
Change-Id: I09fffbb6694f36036ed4bae773571968619bb799
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456420
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47167}
---
 unsafe_buffers_paths.txt | 1 +
 1 file changed, 1 insertion(+)

diff --git a/unsafe_buffers_paths.txt b/unsafe_buffers_paths.txt
index 359d4a3fa5..a1522f446e 100644
--- a/unsafe_buffers_paths.txt
+++ b/unsafe_buffers_paths.txt
@@ -324,6 +324,7 @@
 -modules/audio_processing/capture_levels_adjuster/audio_samples_scaler_unittest.cc
 -modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster_unittest.cc
 -modules/audio_processing/high_pass_filter.cc
+-modules/audio_processing/logging/apm_data_dumper.h
 -modules/audio_processing/post_filter.cc
 -modules/audio_processing/splitting_filter.cc
 -modules/audio_processing/splitting_filter_unittest.cc

From 629212ab1d91c54952e04297a111a07dfcf82e48 Mon Sep 17 00:00:00 2001
From: Evan Shrubsole <eshr@webrtc.org>
Date: Fri, 6 Mar 2026 11:48:52 +0000
Subject: [PATCH 315/472] Refactor StunRequestTest to use
 GlobalSimulatedTimeController

- Replace ScopedFakeClock and AutoThread with GlobalSimulatedTimeController.
- Use CreateTestEnvironment to initialize the Environment.
- Use simulated time AdvanceTime instead of global fake clocks.
- Cleanup unused headers and legacy code.

Bug: webrtc:42223992
Bug: webrtc:469327588
Bug: webrtc:381524905
Change-Id: I10f28bfc50d3205252e0c99dc41d3de06a6a6964
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/454300
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47168}
---
 p2p/base/stun_request_unittest.cc | 49 +++++++++++++++----------------
 1 file changed, 23 insertions(+), 26 deletions(-)

diff --git a/p2p/base/stun_request_unittest.cc b/p2p/base/stun_request_unittest.cc
index d819d939cb..a0998a3239 100644
--- a/p2p/base/stun_request_unittest.cc
+++ b/p2p/base/stun_request_unittest.cc
@@ -10,24 +10,22 @@
 
 #include "p2p/base/stun_request.h"
 
+#include <array>
 #include <cstddef>
-#include <cstdint>
 #include <memory>
 #include <string>
 #include <utility>
-#include <vector>
 
 #include "api/environment/environment.h"
 #include "api/test/rtc_error_matchers.h"
 #include "api/transport/stun.h"
 #include "api/units/time_delta.h"
-#include "rtc_base/fake_clock.h"
-#include "rtc_base/gunit.h"
+#include "api/units/timestamp.h"
 #include "rtc_base/logging.h"
-#include "rtc_base/thread.h"
 #include "test/create_test_environment.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/time_controller/simulated_time_controller.h"
 #include "test/wait_until.h"
 
 namespace webrtc {
@@ -43,20 +41,14 @@ std::unique_ptr<StunMessage> CreateStunMessage(
   return msg;
 }
 
-int TotalDelay(int sends) {
-  std::vector<int> delays = {0,    250,   750,   1750,  3750,
-                             7750, 15750, 23750, 31750, 39750};
-  return delays[sends];
-}
-}  // namespace
-
 class StunRequestThunker;
 
 class StunRequestTest : public ::testing::Test {
  public:
   StunRequestTest()
-      : env_(CreateTestEnvironment()),
-        manager_(Thread::Current(),
+      : time_controller_(Timestamp::Seconds(12345)),
+        env_(CreateTestEnvironment({.time = &time_controller_})),
+        manager_(time_controller_.GetMainThread(),
                  [this](const void* data, size_t size, StunRequest* request) {
                    OnSendPacket(data, size, request);
                  }),
@@ -83,7 +75,7 @@ class StunRequestTest : public ::testing::Test {
   virtual void OnTimeout() { timeout_ = true; }
 
  protected:
-  AutoThread main_thread_;
+  GlobalSimulatedTimeController time_controller_;
   const Environment env_;
   StunRequestManager manager_;
   int request_count_;
@@ -164,25 +156,30 @@ TEST_F(StunRequestTest, TestUnexpected) {
   EXPECT_FALSE(timeout_);
 }
 
-// Test that requests are sent at the right times.
 TEST_F(StunRequestTest, TestBackoff) {
-  ScopedFakeClock fake_clock;
   std::unique_ptr<StunRequestThunker> request = CreateStunRequest();
   std::unique_ptr<StunMessage> res =
       request->CreateResponseMessage(STUN_BINDING_RESPONSE);
+  constexpr auto kTotalDelays = std::to_array<TimeDelta>(
+      {TimeDelta::Zero(), TimeDelta::Millis(250), TimeDelta::Millis(750),
+       TimeDelta::Millis(1750), TimeDelta::Millis(3750),
+       TimeDelta::Millis(7750), TimeDelta::Millis(15750),
+       TimeDelta::Millis(23750), TimeDelta::Millis(31750),
+       TimeDelta::Millis(39750)});
 
-  int64_t start = env_.clock().TimeInMilliseconds();
   manager_.Send(std::move(request));
-  for (int i = 0; i < 9; ++i) {
+  Timestamp start = env_.clock().CurrentTime();
+  for (size_t i = 0; i < kTotalDelays.size() - 1; ++i) {
     EXPECT_THAT(WaitUntil([&] { return request_count_; }, Ne(i),
                           {.timeout = TimeDelta::Millis(STUN_TOTAL_TIMEOUT),
-                           .clock = &fake_clock}),
+                           .clock = &time_controller_}),
                 IsRtcOk());
-    int64_t elapsed = env_.clock().TimeInMilliseconds() - start;
-    RTC_DLOG(LS_INFO) << "STUN request #" << (i + 1) << " sent at " << elapsed
-                      << " ms";
-    EXPECT_EQ(TotalDelay(i), elapsed);
+    TimeDelta elapsed = env_.clock().CurrentTime() - start;
+    RTC_DLOG(LS_INFO) << "STUN request #" << (i + 1) << " sent at " << elapsed;
+    EXPECT_EQ(kTotalDelays[i], elapsed);
   }
+  ASSERT_EQ(request_count_, 9);
+
   EXPECT_TRUE(manager_.CheckResponse(res.get()));
 
   EXPECT_TRUE(response_ == res.get());
@@ -193,13 +190,12 @@ TEST_F(StunRequestTest, TestBackoff) {
 
 // Test that we timeout properly if no response is received.
 TEST_F(StunRequestTest, TestTimeout) {
-  ScopedFakeClock fake_clock;
   std::unique_ptr<StunRequestThunker> request = CreateStunRequest();
   std::unique_ptr<StunMessage> res =
       request->CreateResponseMessage(STUN_BINDING_RESPONSE);
 
   manager_.Send(std::move(request));
-  SIMULATED_WAIT(false, STUN_TOTAL_TIMEOUT, fake_clock);
+  time_controller_.AdvanceTime(TimeDelta::Millis(STUN_TOTAL_TIMEOUT));
 
   EXPECT_FALSE(manager_.CheckResponse(res.get()));
   EXPECT_TRUE(response_ == nullptr);
@@ -284,4 +280,5 @@ TEST_F(StunRequestReentranceTest, TestError) {
   EXPECT_FALSE(timeout_);
 }
 
+}  // namespace
 }  // namespace webrtc

From 457f8ae7e2e29c23a972adb7aaf5b004af67cfd5 Mon Sep 17 00:00:00 2001
From: Evan Shrubsole <eshr@webrtc.org>
Date: Mon, 16 Mar 2026 10:34:13 +0000
Subject: [PATCH 316/472] Use GlobalSimulatedTimeController in
 local_network_access_port_unittest

This removes uses of ScopedFakeClock and AutoThread.

Bug: webrtc:469327588,webrtc:42223992
Change-Id: Iaefd14a9c09eee15147d1b562a0296f16a6a6964
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/457242
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47169}
---
 .../local_network_access_port_unittest.cc     | 31 ++++++++++---------
 1 file changed, 16 insertions(+), 15 deletions(-)

diff --git a/p2p/base/local_network_access_port_unittest.cc b/p2p/base/local_network_access_port_unittest.cc
index 01292a01c3..9f27cb7766 100644
--- a/p2p/base/local_network_access_port_unittest.cc
+++ b/p2p/base/local_network_access_port_unittest.cc
@@ -23,6 +23,7 @@
 #include "api/test/mock_async_dns_resolver.h"
 #include "api/test/mock_local_network_access_permission.h"
 #include "api/test/rtc_error_matchers.h"
+#include "api/units/timestamp.h"
 #include "p2p/base/port.h"
 #include "p2p/base/port_allocator.h"
 #include "p2p/base/stun_port.h"
@@ -32,17 +33,16 @@
 #include "p2p/test/test_stun_server.h"
 #include "p2p/test/test_turn_server.h"
 #include "p2p/test/turn_server.h"
-#include "rtc_base/fake_clock.h"
 #include "rtc_base/net_helper.h"
 #include "rtc_base/net_helpers.h"
 #include "rtc_base/network.h"
 #include "rtc_base/socket.h"
 #include "rtc_base/socket_address.h"
-#include "rtc_base/thread.h"
 #include "rtc_base/virtual_socket_server.h"
 #include "test/create_test_environment.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/time_controller/simulated_time_controller.h"
 #include "test/wait_until.h"
 
 namespace webrtc {
@@ -78,8 +78,9 @@ class LocalNetworkAccessPortTest
 
     switch (server_type()) {
       case kStun:
-        stun_server_ = TestStunServer::Create(env_, {server_address(), 5000},
-                                              ss_, thread_);
+        stun_server_ =
+            TestStunServer::Create(env_, {server_address(), 5000}, ss_,
+                                   *time_controller_.GetMainThread());
         break;
       case kTurn:
         turn_server_.AddInternalSocket({server_address(), 5000}, PROTO_UDP);
@@ -118,7 +119,7 @@ class LocalNetworkAccessPortTest
     ProtocolAddress proto_server_address({server_address, 5000}, PROTO_UDP);
     CreateRelayPortArgs args = {
         .env = env_,
-        .network_thread = &thread_,
+        .network_thread = time_controller_.GetMainThread(),
         .socket_factory = &socket_factory_,
         .network = &network_,
         .server_address = &proto_server_address,
@@ -145,7 +146,7 @@ class LocalNetworkAccessPortTest
       LocalNetworkAccessPermissionFactoryInterface& lna_permission_factory) {
     Port::PortParametersRef params = {
         .env = env_,
-        .network_thread = &thread_,
+        .network_thread = time_controller_.GetMainThread(),
         .socket_factory = &socket_factory_,
         .network = &network_,
         .ice_username_fragment = kIceUfrag,
@@ -187,10 +188,10 @@ class LocalNetworkAccessPortTest
   bool port_ready_ = false;
   bool port_error_ = false;
 
-  ScopedFakeClock fake_clock_;
-  const Environment env_ = CreateTestEnvironment();
   VirtualSocketServer ss_;
-  AutoSocketServerThread thread_{&ss_};
+  GlobalSimulatedTimeController time_controller_{Timestamp::Micros(1234567),
+                                                 &ss_};
+  const Environment env_ = CreateTestEnvironment({.time = &time_controller_});
   MockDnsResolvingPacketSocketFactory socket_factory_{&ss_};
 
   const bool is_using_ipv6_{SocketAddress(server_address(), 5000).family() ==
@@ -199,8 +200,8 @@ class LocalNetworkAccessPortTest
                                                     : kLocalAddr};
   Network network_{"unittest", "unittest", local_address_.ipaddr(), 32};
 
-  TestTurnServer turn_server_{env_, &thread_, &ss_, kTurnUdpIntAddr,
-                              kTurnUdpExtAddr};
+  TestTurnServer turn_server_{env_, time_controller_.GetMainThread(), &ss_,
+                              kTurnUdpIntAddr, kTurnUdpExtAddr};
   TestStunServer::StunServerPtr stun_server_;
 };
 
@@ -265,13 +266,13 @@ TEST_P(LocalNetworkAccessPortTest, ResolvedAddress) {
   if (lna_fake_result() == LnaFakeResult::kPermissionNotNeeded ||
       lna_fake_result() == LnaFakeResult::kPermissionGranted) {
     EXPECT_THAT(WaitUntil([&] { return port_ready_; }, IsTrue(),
-                          {.clock = &fake_clock_}),
+                          {.clock = &time_controller_}),
                 IsRtcOk());
     EXPECT_EQ(1u, port->Candidates().size());
     EXPECT_NE(SOCKET_ERROR, port->GetError());
   } else {
     EXPECT_THAT(WaitUntil([&] { return port_error_; }, IsTrue(),
-                          {.clock = &fake_clock_}),
+                          {.clock = &time_controller_}),
                 IsRtcOk());
     EXPECT_EQ(0u, port->Candidates().size());
     EXPECT_NE(SOCKET_ERROR, port->GetError());
@@ -288,13 +289,13 @@ TEST_P(LocalNetworkAccessPortTest, UnresolvedAddress) {
   if (lna_fake_result() == LnaFakeResult::kPermissionNotNeeded ||
       lna_fake_result() == LnaFakeResult::kPermissionGranted) {
     EXPECT_THAT(WaitUntil([&] { return port_ready_; }, IsTrue(),
-                          {.clock = &fake_clock_}),
+                          {.clock = &time_controller_}),
                 IsRtcOk());
     EXPECT_EQ(1u, port->Candidates().size());
     EXPECT_NE(SOCKET_ERROR, port->GetError());
   } else {
     EXPECT_THAT(WaitUntil([&] { return port_error_; }, IsTrue(),
-                          {.clock = &fake_clock_}),
+                          {.clock = &time_controller_}),
                 IsRtcOk());
     EXPECT_EQ(0u, port->Candidates().size());
     EXPECT_NE(SOCKET_ERROR, port->GetError());

From 9056a05a34cfcc276fcb63be34032e157a8737fd Mon Sep 17 00:00:00 2001
From: Evan Shrubsole <eshr@webrtc.org>
Date: Mon, 16 Mar 2026 10:44:11 +0000
Subject: [PATCH 317/472] Use injected clock in ssl_stream_adapter_unittest

Removes use of AutoThread and ScopedFakeClock to boot.

Bug: webrtc:469327588,webrtc:42223992
Change-Id: I1b06f1666ec68af67fa1896b364981f26a6a6964
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/455941
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47170}
---
 rtc_base/BUILD.gn                       |  1 +
 rtc_base/ssl_stream_adapter_unittest.cc | 42 ++++++++++++-------------
 2 files changed, 22 insertions(+), 21 deletions(-)

diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
index 59fea18a7b..b98dcd6862 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -2358,6 +2358,7 @@ if (rtc_include_tests) {
         "../test:run_loop",
         "../test:test_support",
         "../test:wait_until",
+        "../test/time_controller",
         "memory:fifo_buffer",
         "network:received_packet",
         "synchronization:mutex",
diff --git a/rtc_base/ssl_stream_adapter_unittest.cc b/rtc_base/ssl_stream_adapter_unittest.cc
index 67bfc103e3..8bd26c3873 100644
--- a/rtc_base/ssl_stream_adapter_unittest.cc
+++ b/rtc_base/ssl_stream_adapter_unittest.cc
@@ -39,22 +39,22 @@
 #include "api/task_queue/pending_task_safety_flag.h"
 #include "api/test/rtc_error_matchers.h"
 #include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/buffer_queue.h"
 #include "rtc_base/callback_list.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/crypto_random.h"
-#include "rtc_base/fake_clock.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/message_digest.h"
 #include "rtc_base/ssl_certificate.h"
 #include "rtc_base/ssl_identity.h"
 #include "rtc_base/stream.h"
 #include "rtc_base/thread.h"
-#include "rtc_base/time_utils.h"
 #include "test/create_test_field_trials.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/time_controller/simulated_time_controller.h"
 #include "test/wait_until.h"
 
 namespace webrtc {
@@ -591,20 +591,21 @@ class SSLStreamAdapterTestBase : public ::testing::Test {
                                (server_ssl_->GetState() == SS_OPEN);
                       },
                       ::testing::IsTrue(),
-                      {.timeout = handshake_wait_, .clock = &clock_}),
+                      {.timeout = handshake_wait_, .clock = &time_controller_}),
                   IsRtcOk());
     } else {
-      EXPECT_THAT(WaitUntil([&] { return client_ssl_->GetState(); },
-                            ::testing::Eq(SS_CLOSED),
-                            {.timeout = handshake_wait_, .clock = &clock_}),
-                  IsRtcOk());
+      EXPECT_THAT(
+          WaitUntil([&] { return client_ssl_->GetState(); },
+                    ::testing::Eq(SS_CLOSED),
+                    {.timeout = handshake_wait_, .clock = &time_controller_}),
+          IsRtcOk());
     }
   }
 
   // This tests that we give up after one hour.
   // Only works for BoringSSL which allows advancing the fake clock.
   void TestHandshakeTimeout() {
-    int64_t time_start = clock_.TimeNanos();
+    Timestamp time_start = time_controller_.GetClock()->CurrentTime();
     TimeDelta time_increment = TimeDelta::Millis(1000);
 
     if (!dtls_) {
@@ -632,16 +633,16 @@ class SSLStreamAdapterTestBase : public ::testing::Test {
     // Now wait for the handshake to timeout (or fail after an hour of simulated
     // time).
     while (client_ssl_->GetState() == SS_OPENING &&
-           (TimeDiff(clock_.TimeNanos(), time_start) <
-            3600 * kNumNanosecsPerSec)) {
+           (time_controller_.GetClock()->CurrentTime() - time_start <
+            TimeDelta::Minutes(60))) {
       EXPECT_THAT(WaitUntil(
                       [&] {
                         return !((client_ssl_->GetState() == SS_OPEN) &&
                                  (server_ssl_->GetState() == SS_OPEN));
                       },
-                      ::testing::IsTrue(), {.clock = &clock_}),
+                      ::testing::IsTrue(), {.clock = &time_controller_}),
                   IsRtcOk());
-      clock_.AdvanceTime(time_increment);
+      time_controller_.AdvanceTime(time_increment);
     }
     EXPECT_EQ(client_ssl_->GetState(), SS_CLOSED);
   }
@@ -671,7 +672,7 @@ class SSLStreamAdapterTestBase : public ::testing::Test {
                              server_ssl_->IsTlsConnected();
                     },
                     ::testing::IsTrue(),
-                    {.timeout = handshake_wait_, .clock = &clock_}),
+                    {.timeout = handshake_wait_, .clock = &time_controller_}),
                 IsRtcOk());
 
     // Until the identity has been verified, the state should still be
@@ -863,8 +864,7 @@ class SSLStreamAdapterTestBase : public ::testing::Test {
     return server_ssl_->GetIdentityForTesting();
   }
 
-  AutoThread main_thread_;
-  ScopedFakeClock clock_;
+  GlobalSimulatedTimeController time_controller_{Timestamp::Micros(1234567)};
   std::string client_cert_pem_;
   std::string client_private_key_pem_;
   KeyParams client_key_type_;
@@ -983,19 +983,19 @@ class SSLStreamAdapterTestDTLSBase : public SSLStreamAdapterTestBase {
 
     WriteData();
 
-    EXPECT_THAT(
-        WaitUntil([&] { return sent_; }, ::testing::Eq(count_),
-                  {.timeout = TimeDelta::Millis(10000), .clock = &clock_}),
-        IsRtcOk());
+    EXPECT_THAT(WaitUntil([&] { return sent_; }, ::testing::Eq(count_),
+                          {.timeout = TimeDelta::Millis(10000),
+                           .clock = &time_controller_}),
+                IsRtcOk());
     RTC_LOG(LS_INFO) << "sent_ == " << sent_;
 
     if (damage_) {
-      clock_.AdvanceTime(TimeDelta::Millis(2000));
+      time_controller_.AdvanceTime(TimeDelta::Millis(2000));
       EXPECT_EQ(0U, received_.size());
     } else if (loss_ == 0) {
       EXPECT_THAT(WaitUntil([&] { return received_.size(); },
                             ::testing::Eq(static_cast<size_t>(sent_)),
-                            {.clock = &clock_}),
+                            {.clock = &time_controller_}),
                   IsRtcOk());
     } else {
       RTC_LOG(LS_INFO) << "Sent " << sent_ << " packets; received "

From 68f867ee5be771f7a6a55a2304122fd3af40c531 Mon Sep 17 00:00:00 2001
From: Fredrik Solenberg <solenberg@google.com>
Date: Mon, 16 Mar 2026 09:00:07 +0000
Subject: [PATCH 318/472] Remove WebRTC-RtcEventLogEncodeDependencyDescriptor
 field trial

Bug: webrtc:42225280
Change-Id: Ieafe836ff7c69cefa5d860abb4abb617d5bf4dd6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456500
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47171}
---
 experiments/field_trials.py                   |  3 --
 .../rtc_event_log_encoder_new_format.cc       | 33 ++++++++-----------
 .../rtc_event_log_encoder_new_format.h        |  1 -
 .../encoder/rtc_event_log_encoder_unittest.cc | 18 ----------
 4 files changed, 13 insertions(+), 42 deletions(-)

diff --git a/experiments/field_trials.py b/experiments/field_trials.py
index 9faa63a5b5..679fa3c4d6 100755
--- a/experiments/field_trials.py
+++ b/experiments/field_trials.py
@@ -200,9 +200,6 @@ ACTIVE_FIELD_TRIALS: FrozenSet[FieldTrial] = frozenset([
     FieldTrial('WebRTC-ReceiveBufferSize',
                42225927,
                date(2024, 4, 1)),
-    FieldTrial('WebRTC-RtcEventLogEncodeDependencyDescriptor',
-               42225280,
-               date(2024, 4, 1)),
     FieldTrial('WebRTC-RtcEventLogEncodeNetEqSetMinimumDelayKillSwitch',
                42225058,
                date(2024, 4, 1)),
diff --git a/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.cc b/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.cc
index 5d1342e509..67667fcf29 100644
--- a/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.cc
+++ b/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.cc
@@ -465,23 +465,18 @@ void RtcEventLogEncoderNewFormat::EncodeRtpPacket(const Batch& batch,
   }
 
   {
-    // TODO(webrtc:14975) Remove this kill switch after DD in RTC event log has
-    //                    been rolled out.
-    if (encode_dependency_descriptor_) {
-      std::vector<std::span<const uint8_t>> raw_dds(batch.size());
-      bool has_dd = false;
-      for (size_t i = 0; i < batch.size(); ++i) {
-        raw_dds[i] =
-            batch[i]
-                ->template GetRawExtension<RtpDependencyDescriptorExtension>();
-        has_dd |= !raw_dds[i].empty();
-      }
-      if (has_dd) {
-        if (auto dd_encoded =
-                RtcEventLogDependencyDescriptorEncoderDecoder::Encode(
-                    raw_dds)) {
-          *proto_batch->mutable_dependency_descriptor() = *dd_encoded;
-        }
+    std::vector<std::span<const uint8_t>> raw_dds(batch.size());
+    bool has_dd = false;
+    for (size_t i = 0; i < batch.size(); ++i) {
+      raw_dds[i] =
+          batch[i]
+              ->template GetRawExtension<RtpDependencyDescriptorExtension>();
+      has_dd |= !raw_dds[i].empty();
+    }
+    if (has_dd) {
+      if (auto dd_encoded =
+              RtcEventLogDependencyDescriptorEncoderDecoder::Encode(raw_dds)) {
+        *proto_batch->mutable_dependency_descriptor() = *dd_encoded;
       }
     }
   }
@@ -704,9 +699,7 @@ void RtcEventLogEncoderNewFormat::EncodeRtpPacket(const Batch& batch,
 RtcEventLogEncoderNewFormat::RtcEventLogEncoderNewFormat(
     const FieldTrialsView& field_trials)
     : encode_neteq_set_minimum_delay_kill_switch_(field_trials.IsEnabled(
-          "WebRTC-RtcEventLogEncodeNetEqSetMinimumDelayKillSwitch")),
-      encode_dependency_descriptor_(!field_trials.IsDisabled(
-          "WebRTC-RtcEventLogEncodeDependencyDescriptor")) {}
+          "WebRTC-RtcEventLogEncodeNetEqSetMinimumDelayKillSwitch")) {}
 
 std::string RtcEventLogEncoderNewFormat::EncodeLogStart(int64_t timestamp_us,
                                                         int64_t utc_time_us) {
diff --git a/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.h b/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.h
index 099c5df2b2..85a7515fb8 100644
--- a/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.h
+++ b/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.h
@@ -152,7 +152,6 @@ class RtcEventLogEncoderNewFormat final : public RtcEventLogEncoder {
   void EncodeRtpPacket(const Batch& batch, ProtoType* proto_batch);
 
   const bool encode_neteq_set_minimum_delay_kill_switch_;
-  const bool encode_dependency_descriptor_;
 };
 
 }  // namespace webrtc
diff --git a/logging/rtc_event_log/encoder/rtc_event_log_encoder_unittest.cc b/logging/rtc_event_log/encoder/rtc_event_log_encoder_unittest.cc
index 9e604f03da..409dfc0efb 100644
--- a/logging/rtc_event_log/encoder/rtc_event_log_encoder_unittest.cc
+++ b/logging/rtc_event_log/encoder/rtc_event_log_encoder_unittest.cc
@@ -23,7 +23,6 @@
 #include <utility>
 #include <vector>
 
-#include "api/field_trials.h"
 #include "api/field_trials_view.h"
 #include "api/rtc_event_log/rtc_event.h"
 #include "api/rtc_event_log/rtc_event_log.h"
@@ -1308,23 +1307,6 @@ TEST_P(RtcEventLogEncoderTest, RtcEventRtpPacketOutgoing) {
   TestRtpPackets<RtcEventRtpPacketOutgoing, LoggedRtpPacketOutgoing>(*encoder);
 }
 
-TEST_P(RtcEventLogEncoderTest,
-       RtcEventRtpPacketIncomingNoDependencyDescriptor) {
-  FieldTrials no_dd = CreateTestFieldTrials(
-      "WebRTC-RtcEventLogEncodeDependencyDescriptor/Disabled/");
-  std::unique_ptr<RtcEventLogEncoder> encoder = CreateEncoder(no_dd);
-  verifier_.ExpectDependencyDescriptorExtensionIsSet(false);
-  TestRtpPackets<RtcEventRtpPacketIncoming, LoggedRtpPacketIncoming>(*encoder);
-}
-
-TEST_P(RtcEventLogEncoderTest,
-       RtcEventRtpPacketOutgoingNoDependencyDescriptor) {
-  FieldTrials no_dd = CreateTestFieldTrials(
-      "WebRTC-RtcEventLogEncodeDependencyDescriptor/Disabled/");
-  std::unique_ptr<RtcEventLogEncoder> encoder = CreateEncoder(no_dd);
-  verifier_.ExpectDependencyDescriptorExtensionIsSet(false);
-  TestRtpPackets<RtcEventRtpPacketOutgoing, LoggedRtpPacketOutgoing>(*encoder);
-}
 
 // TODO(eladalon/terelius): Test with multiple events in the batch.
 TEST_P(RtcEventLogEncoderTest, RtcEventVideoReceiveStreamConfig) {

From f78e89d49d8ca973449689b7e739a06ea40f8c82 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Mon, 16 Mar 2026 05:04:14 -0700
Subject: [PATCH 319/472] Roll chromium_revision 88388785e2..54dac6b1cc
 (1599692:1599815)

Change log: https://chromium.googlesource.com/chromium/src/+log/88388785e2..54dac6b1cc
Full diff: https://chromium.googlesource.com/chromium/src/+/88388785e2..54dac6b1cc

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/f05f529c87..f123ee3617
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/abae91664f..8c2c252980
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/e887719b51..6bfb3976ea
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/2c52a3200f..a6d109b540
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/929785924c..ea13a98dec
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/e94fcd7163..fc9a77f811
* src/third_party/kotlinc/current: rZVNs6pgsRU0LupGqWhLJwl508-H7iam-fnzDW6X37MC..RcyJsRii1TkItQ8HjsvzQnXGvIZ0FJvpF4bxYeFr7qAC
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/9887f1b3cb..0950d1381a
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/f2abc599fc..9613832c59
DEPS diff: https://chromium.googlesource.com/chromium/src/+/88388785e2..54dac6b1cc/DEPS

BUG=None

Change-Id: I4ea4444c272e01f1e5a7bbf2de95d362a9720df3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/457280
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47172}
---
 DEPS | 20 ++++++++++----------
 1 file changed, 10 insertions(+), 10 deletions(-)

diff --git a/DEPS b/DEPS
index 145cd2bcec..22d92a18ec 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '88388785e2a97340d8c9cc1e74f24d4e71c092ab',
+  'chromium_revision': '54dac6b1cc5193c4e3bf18c3ddd72ae240915616',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@f05f529c87391f7526e8d85305873c0b2aae9ffb',
+    'https://chromium.googlesource.com/chromium/src/build@f123ee3617656ae843bd7f68f173c651fe2ec4bf',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@95ed44cf5f06dbb5861030b91c9db9ccb4316762',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@abae91664fe2284bda82ed4be85a36e0ba2f8d56',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@8c2c2529805c9898c19a00deec2220fda3cff216',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@e887719b51ac536e7af5e1a5dd71e974520e86db',
+    'https://chromium.googlesource.com/chromium/src/testing@6bfb3976eaef2083da2bbf6999b57a48750604b2',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@2c52a3200fb53172df565b17089f0749a4f3ad27',
+    'https://chromium.googlesource.com/chromium/src/third_party@a6d109b540a22bb212b1a476abda9b9926d45700',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -511,7 +511,7 @@ deps = {
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@e94fcd71631c896295fc7d6874165505e3673e2d',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@fc9a77f8116cec6594e1f317c51baa23ae949e0c',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -584,7 +584,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/kotlinc',
-              'version': 'rZVNs6pgsRU0LupGqWhLJwl508-H7iam-fnzDW6X37MC',
+              'version': 'RcyJsRii1TkItQ8HjsvzQnXGvIZ0FJvpF4bxYeFr7qAC',
           },
       ],
       'condition': 'checkout_android',
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '9887f1b3cb6c1c9c8c72a515a882d5eeea0d13c6',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '0950d1381ab9020cbf2686d74ea905ea64db2d2c',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@f2abc599fc2693206c7367c4ce17772156b813ec',
+    'https://chromium.googlesource.com/chromium/src/tools@9613832c59fd7f88115281a3c9e7be9e91043cac',
 
   'src/third_party/espresso': {
       'packages': [
@@ -864,7 +864,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7607ca500436b37ad23fb8d18614bec7796b68a7',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '929785924cf4c3d68946e9a24beebbd92b1d2b50',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + 'ea13a98decd497a8c5588fb5de71b57bcf10d864',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From 91067be02b399223e64696ea197e1198c46a42fb Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Lionel=20Koenig=20G=C3=A9las?= <lionelk@webrtc.org>
Date: Mon, 16 Mar 2026 14:14:03 +0100
Subject: [PATCH 320/472] VAD SplitFilter may overflow integerr.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

VAD Split filter is design such that, in some specific cases, signed
integer overflow can occurs. Those are fine and will only affect the
output audio making some clicks and/or digital noise.

Bug: webrtc:490331112
Change-Id: Ib4a2eb01a84f6b11e7be0d3df55a892379018178
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/457301
Commit-Queue: Lionel Koenig <lionelk@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47173}
---
 common_audio/vad/vad_filterbank.c | 4 +++-
 1 file changed, 3 insertions(+), 1 deletion(-)

diff --git a/common_audio/vad/vad_filterbank.c b/common_audio/vad/vad_filterbank.c
index 32830fa06f..04cfba908b 100644
--- a/common_audio/vad/vad_filterbank.c
+++ b/common_audio/vad/vad_filterbank.c
@@ -12,6 +12,7 @@
 
 #include "common_audio/signal_processing/include/signal_processing_library.h"
 #include "rtc_base/checks.h"
+#include "rtc_base/sanitizer.h"
 
 // Constants used in LogOfEnergy().
 static const int16_t kLogConst = 24660;          // 160*log10(2) in Q9.
@@ -79,7 +80,8 @@ static void HighPassFilter(const int16_t* data_in,
 // - filter_coefficient [i]   : Given in Q15.
 // - filter_state       [i/o] : State of the filter given in Q(-1).
 // - data_out           [o]   : Output audio signal given in Q(-1).
-static void AllPassFilter(const int16_t* data_in,
+static void RTC_NO_SANITIZE("signed-integer-overflow") // https://bugs.webrtc.org/490331112
+    AllPassFilter(const int16_t* data_in,
                           size_t data_length,
                           int16_t filter_coefficient,
                           int16_t* filter_state,

From aa217206b9ce8b929dc56d112d670a5931ef8cc1 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Mon, 16 Mar 2026 09:04:26 -0700
Subject: [PATCH 321/472] Roll chromium_revision 54dac6b1cc..432c30e3e1
 (1599815:1599933)

Change log: https://chromium.googlesource.com/chromium/src/+log/54dac6b1cc..432c30e3e1
Full diff: https://chromium.googlesource.com/chromium/src/+/54dac6b1cc..432c30e3e1

Changed dependencies
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/8c2c252980..6c591c18ef
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/6bfb3976ea..93c3f9205a
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/a6d109b540..7048751cc7
* src/third_party/kotlin_stdlib/cipd: mLhm2Qx7LKeQaTHlt5dOXbInxRZJzfhO2BhmUPklxBUC..uq9bdsIxS9Is_mAZr9OWBymcLtxheKkgzeUSLuZKhJUC
* src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/fd9bad0830..851188a132
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/0950d1381a..09f2475998
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/9613832c59..358b58f130
DEPS diff: https://chromium.googlesource.com/chromium/src/+/54dac6b1cc..432c30e3e1/DEPS

BUG=None

Change-Id: Id708ed043570b977911ad37cc58ab5f26dec172e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/457340
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47174}
---
 DEPS | 16 ++++++++--------
 1 file changed, 8 insertions(+), 8 deletions(-)

diff --git a/DEPS b/DEPS
index 22d92a18ec..a26410e477 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '54dac6b1cc5193c4e3bf18c3ddd72ae240915616',
+  'chromium_revision': '432c30e3e1dcd03cecb869a1e23162d8ea1b9af2',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@8c2c2529805c9898c19a00deec2220fda3cff216',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@6c591c18ef88a9816d7b77c38f52f9b7a80bfcc0',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@6bfb3976eaef2083da2bbf6999b57a48750604b2',
+    'https://chromium.googlesource.com/chromium/src/testing@93c3f9205a28e0191af7b4f091855d7d2b162faa',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@a6d109b540a22bb212b1a476abda9b9926d45700',
+    'https://chromium.googlesource.com/chromium/src/third_party@7048751cc7450f6e937418379d2a95551973625f',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -337,7 +337,7 @@ deps = {
   'src/third_party/libc++abi/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@8f11bb1d4438d0239d0dfc1bd9456a9f31629dda',
   'src/third_party/llvm-libc/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@fd9bad0830907c4729426180b276e67e6a05b4e0',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@851188a1329844b28215a7eb9f503b8e0f4c1ea0',
   'src/third_party/libunwind/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@db838d918570d4e381ecf9f5cc70a0098c9c2cd6',
 
@@ -573,7 +573,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/kotlin_stdlib',
-              'version': 'mLhm2Qx7LKeQaTHlt5dOXbInxRZJzfhO2BhmUPklxBUC',
+              'version': 'uq9bdsIxS9Is_mAZr9OWBymcLtxheKkgzeUSLuZKhJUC',
           },
       ],
       'condition': 'checkout_android',
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '0950d1381ab9020cbf2686d74ea905ea64db2d2c',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '09f2475998ce66a7710f26640ddd144ce82613b6',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@9613832c59fd7f88115281a3c9e7be9e91043cac',
+    'https://chromium.googlesource.com/chromium/src/tools@358b58f13032d19f1652d6da63f6ea45c34b8008',
 
   'src/third_party/espresso': {
       'packages': [

From db6adf2e9f9a213f9045e1c388713fe727245727 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Per=20=C3=85hgren?= <peah@webrtc.org>
Date: Mon, 16 Mar 2026 20:38:36 +0100
Subject: [PATCH 322/472] Removed the configuration options for multi_channel
 processing
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This CL removes the explicit configuration options for whether
the audio processing should be performed in a multi- or
single-channel manner.
This change means that the choice of whether to do multi- or
single-channel audio processing will be made based on the
content of the audio sent to APM for processing, and the
audio output requested by APM.

Bug: chromium:464314991
Change-Id: I55a53f2565c81a7d7ad2e378f944ffcf07c71f4f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453000
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47175}
---
 api/audio/audio_processing.cc                 |  2 --
 api/audio/audio_processing.h                  |  5 ---
 .../audio_processing/audio_processing_impl.cc | 35 ++-----------------
 .../audio_processing/audio_processing_impl.h  | 10 ++----
 .../audio_processing_unittest.cc              |  2 --
 .../test/audio_processing_simulator.cc        |  8 -----
 .../test/audio_processing_simulator.h         |  2 --
 .../test/audioproc_float_impl.cc              | 14 --------
 .../audio_processing_configs_fuzzer.cc        |  2 --
 .../audio_processing_sample_rate_fuzzer.cc    |  2 --
 10 files changed, 4 insertions(+), 78 deletions(-)

diff --git a/api/audio/audio_processing.cc b/api/audio/audio_processing.cc
index bf0af030fe..12ea953c85 100644
--- a/api/audio/audio_processing.cc
+++ b/api/audio/audio_processing.cc
@@ -131,8 +131,6 @@ std::string AudioProcessing::Config::ToString() const {
              "pipeline: { "
              "maximum_internal_processing_rate: "
           << pipeline.maximum_internal_processing_rate
-          << ", multi_channel_render: " << pipeline.multi_channel_render
-          << ", multi_channel_capture: " << pipeline.multi_channel_capture
           << " }, pre_amplifier: { enabled: " << pre_amplifier.enabled
           << ", fixed_gain_factor: " << pre_amplifier.fixed_gain_factor
           << " },capture_level_adjustment: { enabled: "
diff --git a/api/audio/audio_processing.h b/api/audio/audio_processing.h
index 1e0ea0beee..4c3408d4c2 100644
--- a/api/audio/audio_processing.h
+++ b/api/audio/audio_processing.h
@@ -146,11 +146,6 @@ class RTC_EXPORT AudioProcessing : public RefCountInterface {
       // Maximum allowed processing rate used internally. May only be set to
       // 32000 or 48000 and any differing values will be treated as 48000.
       int maximum_internal_processing_rate = 32000;
-      // Allow multi-channel processing of render audio.
-      bool multi_channel_render = true;
-      // Allow multi-channel processing of capture audio when AEC3 is active
-      // or a custom AEC is injected.
-      bool multi_channel_capture = true;
       // Indicates how to downmix multi-channel capture audio to mono (when
       // needed).
       DownmixMethod capture_downmix_method = DownmixMethod::kAverageChannels;
diff --git a/modules/audio_processing/audio_processing_impl.cc b/modules/audio_processing/audio_processing_impl.cc
index e86cc3cf69..2956078359 100644
--- a/modules/audio_processing/audio_processing_impl.cc
+++ b/modules/audio_processing/audio_processing_impl.cc
@@ -468,10 +468,6 @@ AudioProcessingImpl::AudioProcessingImpl(
                   std::move(capture_analyzer),
                   std::move(neural_residual_echo_estimator)),
       constants_(
-          !env.field_trials().IsEnabled(
-              "WebRTC-ApmExperimentalMultiChannelRenderKillSwitch"),
-          !env.field_trials().IsEnabled(
-              "WebRTC-ApmExperimentalMultiChannelCaptureKillSwitch"),
           EnforceSplitBandHpf(env.field_trials()),
           MinimizeProcessingForUnusedOutput(env.field_trials()),
           env.field_trials().IsEnabled("WebRTC-ApmEnforce48kHzProcessingRate")),
@@ -653,16 +649,9 @@ void AudioProcessingImpl::InitializeLocked(const ProcessingConfig& config) {
              render_processing_rate == 48000);
 
   if (submodule_states_.RenderMultiBandSubModulesActive()) {
-    // By default, downmix the render stream to mono for analysis. This has been
-    // demonstrated to work well for AEC in most practical scenarios.
-    const bool multi_channel_render = config_.pipeline.multi_channel_render &&
-                                      constants_.multi_channel_render_support;
-    int render_processing_num_channels =
-        multi_channel_render
-            ? formats_.api_format.reverse_input_stream().num_channels()
-            : 1;
     formats_.render_processing_format =
-        StreamConfig(render_processing_rate, render_processing_num_channels);
+        StreamConfig(render_processing_rate,
+                     formats_.api_format.reverse_input_stream().num_channels());
   } else {
     formats_.render_processing_format = StreamConfig(
         formats_.api_format.reverse_input_stream().sample_rate_hz(),
@@ -680,10 +669,6 @@ void AudioProcessingImpl::ApplyConfig(const AudioProcessing::Config& config) {
   RTC_LOG(LS_INFO) << "AudioProcessing::ApplyConfig: " << config.ToString();
 
   const bool pipeline_config_changed =
-      config_.pipeline.multi_channel_render !=
-          config.pipeline.multi_channel_render ||
-      config_.pipeline.multi_channel_capture !=
-          config.pipeline.multi_channel_capture ||
       config_.pipeline.maximum_internal_processing_rate !=
           config.pipeline.maximum_internal_processing_rate ||
       config_.pipeline.capture_downmix_method !=
@@ -778,12 +763,6 @@ size_t AudioProcessingImpl::num_input_channels() const {
 }
 
 size_t AudioProcessingImpl::num_proc_channels() const {
-  // Used as callback from submodules, hence locking is not allowed.
-  const bool multi_channel_capture = config_.pipeline.multi_channel_capture &&
-                                     constants_.multi_channel_capture_support;
-  if (submodule_states_.EchoControllerEnabled() && !multi_channel_capture) {
-    return 1;
-  }
   return num_output_channels();
 }
 
@@ -1312,16 +1291,6 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
     capture_buffer->SplitIntoFrequencyBands();
   }
 
-  const bool multi_channel_capture = config_.pipeline.multi_channel_capture &&
-                                     constants_.multi_channel_capture_support;
-  if (submodules_.echo_controller && !multi_channel_capture) {
-    // Force down-mixing of the number of channels after the detection of
-    // capture signal saturation.
-    // TODO(peah): Look into ensuring that this kind of tampering with the
-    // AudioBuffer functionality should not be needed.
-    capture_buffer->set_num_channels(1);
-  }
-
   if (submodules_.high_pass_filter &&
       (!config_.high_pass_filter.apply_in_full_band ||
        constants_.enforce_split_band_hpf)) {
diff --git a/modules/audio_processing/audio_processing_impl.h b/modules/audio_processing/audio_processing_impl.h
index c058961472..302cf43bb2 100644
--- a/modules/audio_processing/audio_processing_impl.h
+++ b/modules/audio_processing/audio_processing_impl.h
@@ -420,20 +420,14 @@ class AudioProcessingImpl : public AudioProcessing {
 
   // APM constants.
   const struct ApmConstants {
-    ApmConstants(bool multi_channel_render_support,
-                 bool multi_channel_capture_support,
-                 bool enforce_split_band_hpf,
+    ApmConstants(bool enforce_split_band_hpf,
                  bool minimize_processing_for_unused_output,
                  bool enforce_48_khz_max_internal_processing_rate)
-        : multi_channel_render_support(multi_channel_render_support),
-          multi_channel_capture_support(multi_channel_capture_support),
-          enforce_split_band_hpf(enforce_split_band_hpf),
+        : enforce_split_band_hpf(enforce_split_band_hpf),
           minimize_processing_for_unused_output(
               minimize_processing_for_unused_output),
           enforce_48_khz_max_internal_processing_rate(
               enforce_48_khz_max_internal_processing_rate) {}
-    bool multi_channel_render_support;
-    bool multi_channel_capture_support;
     bool enforce_split_band_hpf;
     bool minimize_processing_for_unused_output;
     bool enforce_48_khz_max_internal_processing_rate;
diff --git a/modules/audio_processing/audio_processing_unittest.cc b/modules/audio_processing/audio_processing_unittest.cc
index 7078370c64..03472bd568 100644
--- a/modules/audio_processing/audio_processing_unittest.cc
+++ b/modules/audio_processing/audio_processing_unittest.cc
@@ -2056,8 +2056,6 @@ void RunApmRateAndChannelTest(ArrayView<const int> sample_rates_hz,
                               ArrayView<const int> capture_channel_counts,
                               ArrayView<const bool> mono_capture_output) {
   AudioProcessing::Config apm_config;
-  apm_config.pipeline.multi_channel_render = true;
-  apm_config.pipeline.multi_channel_capture = true;
   apm_config.echo_canceller.enabled = true;
   scoped_refptr<AudioProcessing> apm =
       BuiltinAudioProcessingBuilder(apm_config).Build(CreateEnvironment());
diff --git a/modules/audio_processing/test/audio_processing_simulator.cc b/modules/audio_processing/test/audio_processing_simulator.cc
index aa1af1737c..be0a1c64b8 100644
--- a/modules/audio_processing/test/audio_processing_simulator.cc
+++ b/modules/audio_processing/test/audio_processing_simulator.cc
@@ -418,14 +418,6 @@ void AudioProcessingSimulator::ConfigureAudioProcessor() {
   if (settings_.use_ts) {
     apm_config.transient_suppression.enabled = *settings_.use_ts != 0;
   }
-  if (settings_.multi_channel_render) {
-    apm_config.pipeline.multi_channel_render = *settings_.multi_channel_render;
-  }
-
-  if (settings_.multi_channel_capture) {
-    apm_config.pipeline.multi_channel_capture =
-        *settings_.multi_channel_capture;
-  }
 
   if (settings_.use_adaptive_stereo_downmixing_for_aec) {
     if (*settings_.use_adaptive_stereo_downmixing_for_aec) {
diff --git a/modules/audio_processing/test/audio_processing_simulator.h b/modules/audio_processing/test/audio_processing_simulator.h
index 1bf5c1b49f..511981ccc5 100644
--- a/modules/audio_processing/test/audio_processing_simulator.h
+++ b/modules/audio_processing/test/audio_processing_simulator.h
@@ -128,8 +128,6 @@ struct SimulationSettings {
   std::optional<int> maximum_internal_processing_rate;
   int initial_mic_level;
   bool simulate_mic_gain = false;
-  std::optional<bool> multi_channel_render;
-  std::optional<bool> multi_channel_capture;
   std::optional<int> simulated_mic_kind;
   std::optional<int> frame_for_sending_capture_output_used_false;
   std::optional<int> frame_for_sending_capture_output_used_true;
diff --git a/modules/audio_processing/test/audioproc_float_impl.cc b/modules/audio_processing/test/audioproc_float_impl.cc
index 589fa1ad97..d41aaed0b6 100644
--- a/modules/audio_processing/test/audioproc_float_impl.cc
+++ b/modules/audio_processing/test/audioproc_float_impl.cc
@@ -223,16 +223,6 @@ ABSL_FLAG(int,
           0,
           "Activate (1) or deactivate(0) the analog mic gain simulation in the "
           "test code");
-ABSL_FLAG(int,
-          multi_channel_render,
-          kParameterNotSpecifiedValue,
-          "Activate (1) or deactivate (0) multi-channel render processing in "
-          "APM pipeline");
-ABSL_FLAG(int,
-          multi_channel_capture,
-          kParameterNotSpecifiedValue,
-          "Activate (1) or deactivate (0) multi-channel capture processing in "
-          "APM pipeline");
 ABSL_FLAG(int,
           simulated_mic_kind,
           kParameterNotSpecifiedValue,
@@ -482,10 +472,6 @@ SimulationSettings CreateSettings() {
   SetSettingIfSpecified(absl::GetFlag(FLAGS_aec_settings),
                         &settings.aec_settings_filename);
   settings.initial_mic_level = absl::GetFlag(FLAGS_initial_mic_level);
-  SetSettingIfFlagSet(absl::GetFlag(FLAGS_multi_channel_render),
-                      &settings.multi_channel_render);
-  SetSettingIfFlagSet(absl::GetFlag(FLAGS_multi_channel_capture),
-                      &settings.multi_channel_capture);
   settings.simulate_mic_gain = absl::GetFlag(FLAGS_simulate_mic_gain);
   SetSettingIfSpecified(absl::GetFlag(FLAGS_simulated_mic_kind),
                         &settings.simulated_mic_kind);
diff --git a/test/fuzzers/audio_processing_configs_fuzzer.cc b/test/fuzzers/audio_processing_configs_fuzzer.cc
index 614cc66b37..03ee5f3836 100644
--- a/test/fuzzers/audio_processing_configs_fuzzer.cc
+++ b/test/fuzzers/audio_processing_configs_fuzzer.cc
@@ -105,8 +105,6 @@ webrtc::scoped_refptr<AudioProcessing> CreateApm(FuzzDataHelper* fuzz_data,
   }
 
   webrtc::AudioProcessing::Config apm_config;
-  apm_config.pipeline.multi_channel_render = true;
-  apm_config.pipeline.multi_channel_capture = true;
   apm_config.echo_canceller.enabled = use_aec || use_aecm;
   apm_config.high_pass_filter.enabled = use_hpf;
   apm_config.gain_controller1.enabled = use_agc;
diff --git a/test/fuzzers/audio_processing_sample_rate_fuzzer.cc b/test/fuzzers/audio_processing_sample_rate_fuzzer.cc
index 3f7bec7e4c..e893f021ea 100644
--- a/test/fuzzers/audio_processing_sample_rate_fuzzer.cc
+++ b/test/fuzzers/audio_processing_sample_rate_fuzzer.cc
@@ -94,8 +94,6 @@ void FuzzOneInput(FuzzDataHelper fuzz_data) {
           : nullptr;
   scoped_refptr<AudioProcessing> apm =
       BuiltinAudioProcessingBuilder()
-          .SetConfig({.pipeline = {.multi_channel_render = true,
-                                   .multi_channel_capture = true}})
           .SetCapturePostProcessing(std::move(capture_processor))
           .SetRenderPreProcessing(std::move(render_processor))
           .Build(CreateEnvironment());

From 2789ba637fa2e1d4e2915013bb05bb1ee7c5c296 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Mon, 16 Mar 2026 17:02:44 -0700
Subject: [PATCH 323/472] Roll chromium_revision 432c30e3e1..3233143d85
 (1599933:1600202)

Change log: https://chromium.googlesource.com/chromium/src/+log/432c30e3e1..3233143d85
Full diff: https://chromium.googlesource.com/chromium/src/+/432c30e3e1..3233143d85

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/f123ee3617..7813ec8df3
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/6c591c18ef..128bca2b90
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/93c3f9205a..42240910e8
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/7048751cc7..e1a6414b85
* src/third_party/android_deps/autorolled/cipd: 3VYcqPssBf1YI1iK0N2UUMsT2Ik5BlMrvm-wOVr1fMQC..Iv8_79hT6eno7q3HWp8otQ560vHfs_HhOBqWLLyLCscC
* src/third_party/androidx/cipd: Mn6UxK4fOcnps2oJ1ikX7T1Nf93dtMBkX68uqIyoQ0sC..ZmqTPlStkea9SodRTzUZVUTdZXl4vnQOGDPtfys86o0C
* src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/b93edfeca7..573ea4ab86
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/6b62257bd9..4e58caf8aa
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/fc9a77f811..47d41c464c
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/09f2475998..f2a3fbdcf9
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/358b58f130..a3649439ca
DEPS diff: https://chromium.googlesource.com/chromium/src/+/432c30e3e1..3233143d85/DEPS

BUG=None

Change-Id: I74a9e531e5ecf98ca63961c55e9840a6a2c805b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/457420
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47176}
---
 DEPS | 24 ++++++++++++------------
 1 file changed, 12 insertions(+), 12 deletions(-)

diff --git a/DEPS b/DEPS
index a26410e477..eecde967a4 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '432c30e3e1dcd03cecb869a1e23162d8ea1b9af2',
+  'chromium_revision': '3233143d858e3ec2fe72af75a0f46a9153b327d9',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@f123ee3617656ae843bd7f68f173c651fe2ec4bf',
+    'https://chromium.googlesource.com/chromium/src/build@7813ec8df3267cb270df610f31f655ae744068cf',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@95ed44cf5f06dbb5861030b91c9db9ccb4316762',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@6c591c18ef88a9816d7b77c38f52f9b7a80bfcc0',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@128bca2b90af0378d90a101a1d6dc0040a0805c7',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@93c3f9205a28e0191af7b4f091855d7d2b162faa',
+    'https://chromium.googlesource.com/chromium/src/testing@42240910e8b5a902abee45db1cd952f321011faf',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@7048751cc7450f6e937418379d2a95551973625f',
+    'https://chromium.googlesource.com/chromium/src/third_party@e1a6414b8541fc81b89286f0089e1ee6faf0885a',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -331,7 +331,7 @@ deps = {
   'src/third_party/clang-format/script':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@c2725e0622e1a86d55f14514f2177a39efea4a0e',
   'src/third_party/compiler-rt/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@b93edfeca73cf0d8537cf1c84ede88a9768c7bab',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@573ea4ab862e838d5c8512f654799d14deca07e7',
   'src/third_party/libc++/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@7ab65651aed6802d2599dcb7a73b1f82d5179d05',
   'src/third_party/libc++abi/src':
@@ -505,13 +505,13 @@ deps = {
   'src/third_party/crc32c/src':
     'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558',
   'src/third_party/depot_tools':
-    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@6b62257bd9fc9ee0e00d10712a0f3db63132ba2f',
+    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@4e58caf8aa08befe78cf40d08ba33da072b15cf9',
   'src/third_party/ffmpeg':
     'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@946d97db8d906277085e361892b7efda5152e2f1',
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@fc9a77f8116cec6594e1f317c51baa23ae949e0c',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@47d41c464c54ff2f37aa78338f96d0d32af35443',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '09f2475998ce66a7710f26640ddd144ce82613b6',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'f2a3fbdcf9f40c072276e39ad86790c2add615f1',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@358b58f13032d19f1652d6da63f6ea45c34b8008',
+    'https://chromium.googlesource.com/chromium/src/tools@a3649439caa5bf0ee925b0c8df890c3f3914e8e6',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'Mn6UxK4fOcnps2oJ1ikX7T1Nf93dtMBkX68uqIyoQ0sC',
+          'version': 'ZmqTPlStkea9SodRTzUZVUTdZXl4vnQOGDPtfys86o0C',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -842,7 +842,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/android_deps/autorolled',
-              'version': '3VYcqPssBf1YI1iK0N2UUMsT2Ik5BlMrvm-wOVr1fMQC',
+              'version': 'Iv8_79hT6eno7q3HWp8otQ560vHfs_HhOBqWLLyLCscC',
           },
       ],
       'condition': 'checkout_android and non_git_source',

From 1d7d15dbe39e825c8aebecb30ce0b8cef9ab4d29 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Mon, 16 Mar 2026 21:13:00 -0700
Subject: [PATCH 324/472] Roll chromium_revision 3233143d85..8ab1ce0d36
 (1600202:1600317)

Change log: https://chromium.googlesource.com/chromium/src/+log/3233143d85..8ab1ce0d36
Full diff: https://chromium.googlesource.com/chromium/src/+/3233143d85..8ab1ce0d36

Changed dependencies
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/128bca2b90..7a44f96317
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/42240910e8..72d6c0bbf5
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/e1a6414b85..bebaff3d43
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/f2a3fbdcf9..e7ca9ec9a0
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/a3649439ca..72781fe59b
DEPS diff: https://chromium.googlesource.com/chromium/src/+/3233143d85..8ab1ce0d36/DEPS

BUG=None

Change-Id: Ie5a7cfcb28d9ad8a99413156ad58d22d0f9b7072
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/457460
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47177}
---
 DEPS | 12 ++++++------
 1 file changed, 6 insertions(+), 6 deletions(-)

diff --git a/DEPS b/DEPS
index eecde967a4..cc4ff821a8 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '3233143d858e3ec2fe72af75a0f46a9153b327d9',
+  'chromium_revision': '8ab1ce0d367dc9ed4f2467374678b65792f47b56',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@128bca2b90af0378d90a101a1d6dc0040a0805c7',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@7a44f96317315121533cee03e2672d8288963480',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@42240910e8b5a902abee45db1cd952f321011faf',
+    'https://chromium.googlesource.com/chromium/src/testing@72d6c0bbf548ba2cdf7dcb1c735617485767374d',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@e1a6414b8541fc81b89286f0089e1ee6faf0885a',
+    'https://chromium.googlesource.com/chromium/src/third_party@bebaff3d4325cd319f96dc82c125313f85ccb8cc',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'f2a3fbdcf9f40c072276e39ad86790c2add615f1',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'e7ca9ec9a0d663b279b8936aa9548d9e2c4ffd46',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@a3649439caa5bf0ee925b0c8df890c3f3914e8e6',
+    'https://chromium.googlesource.com/chromium/src/tools@72781fe59b2bbea2abaa35f3f1e37dae04e77a5a',
 
   'src/third_party/espresso': {
       'packages': [

From 6a75a33b29e7eaea1e9d21d30d6be254783a60a3 Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Mon, 16 Mar 2026 21:10:43 -0700
Subject: [PATCH 325/472] Update WebRTC code version (2026-03-17T04:10:08).

Bug: None
Change-Id: I6c7587f74bbf8924d82fc08c21baa1b682df1b05
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/457440
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47178}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index 5f3ef9fc12..e1facb4943 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-16T04:04:41";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-17T04:10:08";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From 4b7e544b67e773b26ce7e0c318e3e47fdf7d63ba Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Mon, 16 Mar 2026 17:49:05 +0100
Subject: [PATCH 326/472] Move ssrcs_ and payload_types_ to the network thread
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Moves the remaining demuxer criteria related variables in the channel
classes, ssrcs_ and payload_types_, to the network thread. State updates
are now passed as optional parameters and inlined into
`MaybeUpdateDemuxerAndRtpExtensions_w` and `RegisterRtpDemuxerSink_w`.
These methods apply the modifications atomically within a blocking call
to the network thread.

Specific changes include:
* Updating annotations to `RTC_GUARDED_BY(network_thread())`.
* Removing `MaybeAddHandledPayloadType` to construct payloads prior
  to the thread hop.
* Restricting `demuxer_criteria()` to run only on the network thread.

Bug: webrtc:42222117
Change-Id: If5a14bd5626536118e3caa7c87237a94d7ba2230
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456300
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47179}
---
 pc/channel.cc | 104 ++++++++++++++++++++++++++++++--------------------
 pc/channel.h  |  23 +++++------
 2 files changed, 71 insertions(+), 56 deletions(-)

diff --git a/pc/channel.cc b/pc/channel.cc
index 3f70792fe3..f1705378f6 100644
--- a/pc/channel.cc
+++ b/pc/channel.cc
@@ -52,7 +52,6 @@
 #include "rtc_base/socket.h"
 #include "rtc_base/strings/string_format.h"
 #include "rtc_base/thread.h"
-#include "rtc_base/thread_annotations.h"
 #include "rtc_base/trace_event.h"
 #include "rtc_base/unique_id_generator.h"
 
@@ -384,15 +383,7 @@ bool BaseChannel::SetPayloadTypeDemuxingEnabled(bool enabled) {
     }));
   }
 
-  // TODO: bugs.webrtc.org/42222117 - payload_types_ is guarded by the worker
-  // thread, but we read it here on the network thread. This is safe because
-  // this method is called synchronously during SDP signaling, during which no
-  // tasks that modify payload_types_ can be running on the worker thread.
-  auto has_payload_types = [this]() RTC_NO_THREAD_SAFETY_ANALYSIS {
-    return !payload_types_.empty();
-  };
-
-  if (has_payload_types()) {
+  if (!payload_types_.empty()) {
     if (!rtp_transport_) {
       return false;
     }
@@ -586,7 +577,9 @@ void BaseChannel::OnRtpPacket(const RtpPacketReceived& parsed_packet) {
 
 bool BaseChannel::MaybeUpdateDemuxerAndRtpExtensions_w(
     bool update_demuxer,
+    std::optional<flat_set<uint8_t>> payload_types,
     const RtpHeaderExtensions& extensions,
+    std::optional<flat_set<uint32_t>> ssrcs,
     std::string& error_desc) {
   bool update_extensions = true;
   if (rtp_header_extensions_ == extensions) {
@@ -609,12 +602,14 @@ bool BaseChannel::MaybeUpdateDemuxerAndRtpExtensions_w(
     }
   }
 
-  if (!update_demuxer && !update_extensions)
+  if (!update_demuxer && !update_extensions && !payload_types.has_value() &&
+      !ssrcs.has_value()) {
     return true;  // No update needed.
+  }
 
   // TODO(bugs.webrtc.org/13536): See if we can do this asynchronously.
 
-  if (update_demuxer) {
+  if (update_demuxer || payload_types.has_value() || ssrcs.has_value()) {
     media_receive_channel()->OnDemuxerCriteriaUpdatePending();
   }
 
@@ -631,6 +626,20 @@ bool BaseChannel::MaybeUpdateDemuxerAndRtpExtensions_w(
       rtp_transport_->RegisterRtpHeaderExtensionMap(mid(), extensions);
     }
 
+    if (payload_types) {
+      if (payload_types_ != *payload_types) {
+        payload_types_ = std::move(*payload_types);
+        update_demuxer = true;
+      }
+    }
+
+    if (ssrcs) {
+      if (ssrcs_ != *ssrcs) {
+        ssrcs_ = std::move(*ssrcs);
+        update_demuxer = true;
+      }
+    }
+
     if (!update_demuxer)
       return true;
 
@@ -644,13 +653,15 @@ bool BaseChannel::MaybeUpdateDemuxerAndRtpExtensions_w(
     return true;
   });
 
-  if (update_demuxer)
+  if (update_demuxer || payload_types.has_value() || ssrcs.has_value())
     media_receive_channel()->OnDemuxerCriteriaUpdateComplete();
 
   return success;
 }
 
-bool BaseChannel::RegisterRtpDemuxerSink_w() {
+bool BaseChannel::RegisterRtpDemuxerSink_w(
+    bool clear_payload_types,
+    std::optional<flat_set<uint32_t>> ssrcs) {
   media_receive_channel()->OnDemuxerCriteriaUpdatePending();
   bool ret = network_thread_->BlockingCall([&] {
     RTC_DCHECK_RUN_ON(network_thread());
@@ -660,6 +671,25 @@ bool BaseChannel::RegisterRtpDemuxerSink_w() {
       // See chromium:1295469 for an example.
       return false;
     }
+
+    bool needs_re_registration = false;
+
+    if (clear_payload_types && !payload_types_.empty()) {
+      payload_types_.clear();
+      needs_re_registration = true;
+    }
+
+    if (ssrcs) {
+      if (ssrcs_ != *ssrcs) {
+        ssrcs_ = std::move(*ssrcs);
+        needs_re_registration = true;
+      }
+    }
+
+    if (!needs_re_registration) {
+      return true;
+    }
+
     // Note that RegisterRtpDemuxerSink first unregisters the sink if
     // already registered. So this will change the state of the class
     // whether the call succeeds or not.
@@ -822,15 +852,12 @@ bool BaseChannel::UpdateRemoteStreams_w(const MediaContentDescription* content,
                                         SdpType type,
                                         std::string& error_desc) {
   RTC_LOG_THREAD_BLOCK_COUNT();
-  bool needs_re_registration = false;
+  bool clear_payload_types = false;
   if (!RtpTransceiverDirectionHasSend(content->direction())) {
     RTC_DLOG(LS_VERBOSE) << "UpdateRemoteStreams_w: remote side will not send "
                             "- disable payload type demuxing for "
                          << ToString();
-    if (!payload_types_.empty()) {
-      payload_types_.clear();
-      needs_re_registration = true;
-    }
+    clear_payload_types = true;
   }
 
   const std::vector<StreamParams>& streams = content->streams();
@@ -888,15 +915,10 @@ bool BaseChannel::UpdateRemoteStreams_w(const MediaContentDescription* content,
     ssrcs.insert(new_stream.ssrcs.begin(), new_stream.ssrcs.end());
   }
 
-  if (ssrcs_ != ssrcs) {
-    ssrcs_ = std::move(ssrcs);
-    needs_re_registration = true;
-  }
-
   RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(0);
 
   // Re-register the sink to update after changing the demuxer criteria.
-  if (needs_re_registration && !RegisterRtpDemuxerSink_w()) {
+  if (!RegisterRtpDemuxerSink_w(clear_payload_types, std::move(ssrcs))) {
     error_desc = StringFormat("Failed to set up audio demuxing for mid='%s'.",
                               mid().c_str());
     return false;
@@ -918,12 +940,6 @@ RtpHeaderExtensions BaseChannel::GetDeduplicatedRtpHeaderExtensions(
                                                    extensions_filter_);
 }
 
-bool BaseChannel::MaybeAddHandledPayloadType(int payload_type) {
-  // Even if payload type demuxing is currently disabled, we need to remember
-  // the payload types in case it's re-enabled later.
-  return payload_types_.insert(static_cast<uint8_t>(payload_type)).second;
-}
-
 bool BaseChannel::CheckRtpExtensionValidity(
     const RtpHeaderExtensions& extensions,
     std::string& error_desc) const {
@@ -1046,12 +1062,11 @@ bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content,
     return false;
   }
 
-  bool criteria_modified = false;
+  std::optional<flat_set<uint8_t>> payload_types;
   if (RtpTransceiverDirectionHasRecv(content->direction())) {
+    payload_types.emplace();
     for (const Codec& codec : content->codecs()) {
-      if (MaybeAddHandledPayloadType(codec.id)) {
-        criteria_modified = true;
-      }
+      payload_types->insert(codec.id);
     }
   }
 
@@ -1084,7 +1099,9 @@ bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content,
   // RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(0);
 
   bool success = MaybeUpdateDemuxerAndRtpExtensions_w(
-      criteria_modified, recv_params.extensions, error_desc);
+      /*update_demuxer=*/false, std::move(payload_types),
+      recv_params.extensions,
+      /*ssrcs=*/std::nullopt, error_desc);
 
   // RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(1);
 
@@ -1148,7 +1165,8 @@ bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content,
   bool success = true;
   if (type == SdpType::kAnswer || type == SdpType::kPrAnswer) {
     success = MaybeUpdateDemuxerAndRtpExtensions_w(
-        /*update_demuxer=*/false, last_recv_params_.extensions, error_desc);
+        /*update_demuxer=*/false, /*payload_types=*/std::nullopt,
+        last_recv_params_.extensions, /*ssrcs=*/std::nullopt, error_desc);
   }
 
   return success;
@@ -1241,11 +1259,11 @@ bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
     return false;
   }
 
-  bool criteria_modified = false;
+  std::optional<flat_set<uint8_t>> payload_types;
   if (RtpTransceiverDirectionHasRecv(content->direction())) {
+    payload_types.emplace();
     for (const Codec& codec : content->codecs()) {
-      if (MaybeAddHandledPayloadType(codec.id))
-        criteria_modified = true;
+      payload_types->insert(codec.id);
     }
   }
 
@@ -1272,7 +1290,8 @@ bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
   RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(0);
 
   bool success = MaybeUpdateDemuxerAndRtpExtensions_w(
-      criteria_modified, recv_params.extensions, error_desc);
+      /*update_demuxer=*/false, std::move(payload_types),
+      recv_params.extensions, /*ssrcs=*/std::nullopt, error_desc);
 
   RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(1);
 
@@ -1336,7 +1355,8 @@ bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content,
 
   if (type == SdpType::kAnswer || type == SdpType::kPrAnswer) {
     bool success = MaybeUpdateDemuxerAndRtpExtensions_w(
-        /*update_demuxer=*/false, last_recv_params_.extensions, error_desc);
+        /*update_demuxer=*/false, /*payload_types=*/std::nullopt,
+        last_recv_params_.extensions, /*ssrcs=*/std::nullopt, error_desc);
     if (!success) {
       return false;
     }
diff --git a/pc/channel.h b/pc/channel.h
index c53c1b44f5..f4488cd612 100644
--- a/pc/channel.h
+++ b/pc/channel.h
@@ -278,12 +278,6 @@ class BaseChannel : public ChannelInterface,
   RtpHeaderExtensions GetDeduplicatedRtpHeaderExtensions(
       const RtpHeaderExtensions& extensions);
 
-  // Add `payload_type` to `demuxer_criteria_` if payload type demuxing is
-  // enabled.
-  // Returns true if the demuxer payload type changed and a re-registration
-  // is needed.
-  bool MaybeAddHandledPayloadType(int payload_type) RTC_RUN_ON(worker_thread());
-
   // Checks that the provided RTP header extensions are valid.
   // This verifies that all extension IDs are within the valid range,
   // that there are no duplicate IDs, and that no existing extension ID
@@ -297,13 +291,17 @@ class BaseChannel : public ChannelInterface,
   // failed, which needs to be treated as an error.
   bool MaybeUpdateDemuxerAndRtpExtensions_w(
       bool update_demuxer,
+      std::optional<flat_set<uint8_t>> payload_types,
       const RtpHeaderExtensions& extensions,
+      std::optional<flat_set<uint32_t>> ssrcs,
       std::string& error_desc) RTC_RUN_ON(worker_thread());
 
   // Registers a demuxer criteria with the transport, on the network thread.
   // This function will fail if there's no transport of if a sink is already
   // registered for this channel's demuxer_critera().
-  bool RegisterRtpDemuxerSink_w() RTC_RUN_ON(worker_thread());
+  bool RegisterRtpDemuxerSink_w(bool clear_payload_types,
+                                std::optional<flat_set<uint32_t>> ssrcs)
+      RTC_RUN_ON(worker_thread());
 
   // Return description of media channel to facilitate logging
   std::string ToString() const;
@@ -316,11 +314,8 @@ class BaseChannel : public ChannelInterface,
       RTC_RUN_ON(network_thread());
   void DisconnectFromRtpTransport_n() RTC_RUN_ON(network_thread());
   void SignalSentPacket_n(const SentPacketInfo& sent_packet);
-  // Only called on the network thread, so should be marked as
-  // `RTC_RUN_ON(network_thread());` but currently accesses variables
-  // that are associated with the worker thread. Those operations are
-  // done in a synchronous way, so they're safe, but not ideal.
-  RtpDemuxerCriteria demuxer_criteria() const RTC_NO_THREAD_SAFETY_ANALYSIS;
+  // Only called on the network thread.
+  RtpDemuxerCriteria demuxer_criteria() const RTC_RUN_ON(network_thread());
 
   TaskQueueBase* const worker_thread_;
   Thread* const network_thread_;
@@ -367,7 +362,7 @@ class BaseChannel : public ChannelInterface,
       RTC_GUARDED_BY(worker_thread()) = RtpTransceiverDirection::kInactive;
 
   // Cached list of payload types, used if payload type demuxing is re-enabled.
-  flat_set<uint8_t> payload_types_ RTC_GUARDED_BY(worker_thread());
+  flat_set<uint8_t> payload_types_ RTC_GUARDED_BY(network_thread());
   // A stored copy of the rtp header extensions as applied to the transport.
   RtpHeaderExtensions rtp_header_extensions_ RTC_GUARDED_BY(worker_thread());
 
@@ -377,7 +372,7 @@ class BaseChannel : public ChannelInterface,
       RTC_GUARDED_BY(worker_thread());
 
   const std::string mid_;
-  flat_set<uint32_t> ssrcs_ RTC_GUARDED_BY(worker_thread());
+  flat_set<uint32_t> ssrcs_ RTC_GUARDED_BY(network_thread());
   // This generator is used to generate SSRCs for local streams.
   // This is needed in cases where SSRCs are not negotiated or set explicitly
   // like in Simulcast.

From ae5414e8464c440b7f6fca85705ad3351156a95f Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Tue, 17 Mar 2026 05:02:24 -0700
Subject: [PATCH 327/472] Roll chromium_revision 8ab1ce0d36..eb32defcec
 (1600317:1600457)

Change log: https://chromium.googlesource.com/chromium/src/+log/8ab1ce0d36..eb32defcec
Full diff: https://chromium.googlesource.com/chromium/src/+/8ab1ce0d36..eb32defcec

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/7813ec8df3..00b379c45d
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/7a44f96317..17402a0a52
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/72d6c0bbf5..5b9eb17956
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/bebaff3d43..17adda09b6
* src/third_party/androidx/cipd: ZmqTPlStkea9SodRTzUZVUTdZXl4vnQOGDPtfys86o0C..wxjbyb5KcZ7SzBmiQuEU3Ou1KQf6VobxrK6ImmKDdvcC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/75a1350717..c9adcbb559
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/47d41c464c..2ab1a62145
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/72781fe59b..d4d3f92cac
DEPS diff: https://chromium.googlesource.com/chromium/src/+/8ab1ce0d36..eb32defcec/DEPS

BUG=None

Change-Id: If6375379925c450d0f8989dc791f48a595c81d51
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/457600
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47180}
---
 DEPS | 18 +++++++++---------
 1 file changed, 9 insertions(+), 9 deletions(-)

diff --git a/DEPS b/DEPS
index cc4ff821a8..91f3e4f7af 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '8ab1ce0d367dc9ed4f2467374678b65792f47b56',
+  'chromium_revision': 'eb32defcecc7ddcb69ea920b630f20cd6150106b',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@7813ec8df3267cb270df610f31f655ae744068cf',
+    'https://chromium.googlesource.com/chromium/src/build@00b379c45d62148b8b0f273cb295356d6039739d',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@95ed44cf5f06dbb5861030b91c9db9ccb4316762',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@7a44f96317315121533cee03e2672d8288963480',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@17402a0a523e51c56598f39bdf68fd5dd9c81b25',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@72d6c0bbf548ba2cdf7dcb1c735617485767374d',
+    'https://chromium.googlesource.com/chromium/src/testing@5b9eb179561e84eb1fea1d331dd9a473a85ab7e0',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@bebaff3d4325cd319f96dc82c125313f85ccb8cc',
+    'https://chromium.googlesource.com/chromium/src/third_party@17adda09b6a68ca16fc6156a06db61aff6dbc20d',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -488,7 +488,7 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@75a1350717c122d5f1edf8938a9c17f1ce79f888',
+    'https://boringssl.googlesource.com/boringssl.git@c9adcbb55901715856a6aed27a2f15b5f91680b9',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@8be0e3114685fcc1589561067282edf75ea1259a',
   'src/third_party/catapult':
@@ -511,7 +511,7 @@ deps = {
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@47d41c464c54ff2f37aa78338f96d0d32af35443',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@2ab1a6214568e5a0ddf85a9d47b055e80492d389',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@72781fe59b2bbea2abaa35f3f1e37dae04e77a5a',
+    'https://chromium.googlesource.com/chromium/src/tools@d4d3f92cacfdf6d3ef316dcad8273d8d98f3393f',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'ZmqTPlStkea9SodRTzUZVUTdZXl4vnQOGDPtfys86o0C',
+          'version': 'wxjbyb5KcZ7SzBmiQuEU3Ou1KQf6VobxrK6ImmKDdvcC',
       },
     ],
     'condition': 'checkout_android and non_git_source',

From 60f506109ec4e3b97a255acd2453dadb3daeff46 Mon Sep 17 00:00:00 2001
From: Evan Shrubsole <eshr@webrtc.org>
Date: Mon, 16 Mar 2026 10:15:28 +0000
Subject: [PATCH 328/472] Remove use of ScopedFakeClock in
 video_encoder_software_fallback_wrapper_unittest
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

The fake clock was not used so was completely removed.

Bug: webrtc:469327588,webrtc:42223992
Change-Id: I5cd06b94801fc0f3d5fc91734719ce636a6a6964
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/457241
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47181}
---
 api/video_codecs/test/BUILD.gn                |  3 +-
 ...oder_software_fallback_wrapper_unittest.cc | 50 ++++++++-----------
 2 files changed, 23 insertions(+), 30 deletions(-)

diff --git a/api/video_codecs/test/BUILD.gn b/api/video_codecs/test/BUILD.gn
index d905633eee..f1861c7f99 100644
--- a/api/video_codecs/test/BUILD.gn
+++ b/api/video_codecs/test/BUILD.gn
@@ -41,13 +41,13 @@ if (rtc_include_tests) {
       "../../../modules/video_coding:webrtc_vp8",
       "../../../rtc_base:checks",
       "../../../rtc_base:rtc_base_tests_utils",
+      "../../../test:create_test_environment",
       "../../../test:create_test_field_trials",
       "../../../test:fake_video_codecs",
       "../../../test:test_support",
       "../../../test:video_test_common",
       "../../environment",
       "../../environment:environment_factory",
-      "../../units:timestamp",
       "../../video:encoded_image",
       "../../video:resolution",
       "../../video:video_bitrate_allocation",
@@ -56,6 +56,7 @@ if (rtc_include_tests) {
       "../../video:video_frame_type",
       "../../video:video_rtp_headers",
       "//testing/gtest",
+      "//third_party/abseil-cpp/absl/strings:string_view",
     ]
   }
 
diff --git a/api/video_codecs/test/video_encoder_software_fallback_wrapper_unittest.cc b/api/video_codecs/test/video_encoder_software_fallback_wrapper_unittest.cc
index 542b194407..e69c9c1793 100644
--- a/api/video_codecs/test/video_encoder_software_fallback_wrapper_unittest.cc
+++ b/api/video_codecs/test/video_encoder_software_fallback_wrapper_unittest.cc
@@ -18,12 +18,11 @@
 #include <utility>
 #include <vector>
 
+#include "absl/strings/string_view.h"
 #include "api/environment/environment.h"
-#include "api/environment/environment_factory.h"
 #include "api/fec_controller_override.h"
 #include "api/scoped_refptr.h"
 #include "api/test/mock_video_encoder.h"
-#include "api/units/timestamp.h"
 #include "api/video/encoded_image.h"
 #include "api/video/i420_buffer.h"
 #include "api/video/video_bitrate_allocator.h"
@@ -38,8 +37,7 @@
 #include "modules/video_coding/include/video_codec_interface.h"
 #include "modules/video_coding/include/video_error_codes.h"
 #include "modules/video_coding/utility/simulcast_rate_allocator.h"
-#include "rtc_base/fake_clock.h"
-#include "test/create_test_field_trials.h"
+#include "test/create_test_environment.h"
 #include "test/fake_encoder.h"
 #include "test/fake_texture_frame.h"
 #include "test/gmock.h"
@@ -205,7 +203,7 @@ class VideoEncoderSoftwareFallbackWrapperTest
   explicit VideoEncoderSoftwareFallbackWrapperTest(
       CountingFakeEncoder* fake_sw_encoder)
       : VideoEncoderSoftwareFallbackWrapperTestBase(
-            CreateEnvironment(),
+            CreateTestEnvironment(),
             std::unique_ptr<VideoEncoder>(fake_sw_encoder)),
         fake_sw_encoder_(fake_sw_encoder) {
     fake_sw_encoder_->implementation_name_ = "fake_sw_encoder";
@@ -495,17 +493,15 @@ TEST_F(VideoEncoderSoftwareFallbackWrapperTest,
 
 class ForcedFallbackTest : public VideoEncoderSoftwareFallbackWrapperTestBase {
  public:
-  explicit ForcedFallbackTest(const Environment& env)
-      : VideoEncoderSoftwareFallbackWrapperTestBase(env,
-                                                    CreateVp8Encoder(env)) {}
+  explicit ForcedFallbackTest(absl::string_view field_trials)
+      : VideoEncoderSoftwareFallbackWrapperTestBase(
+            CreateTestEnvironment({.field_trials = field_trials}),
+            CreateVp8Encoder(
+                CreateTestEnvironment({.field_trials = field_trials}))) {}
 
   ~ForcedFallbackTest() override {}
 
- protected:
-  void SetUp() override {
-    clock_.SetTime(Timestamp::Micros(1234));
-    ConfigureVp8Codec();
-  }
+  void SetUp() override { ConfigureVp8Codec(); }
 
   void TearDown() override {
     if (wrapper_initialized_) {
@@ -553,24 +549,20 @@ class ForcedFallbackTest : public VideoEncoderSoftwareFallbackWrapperTestBase {
     EncodeFrame(expected_ret);
     CheckLastEncoderName(expected_name);
   }
-
-  ScopedFakeClock clock_;
 };
 
 class ForcedFallbackTestEnabled : public ForcedFallbackTest {
  public:
   ForcedFallbackTestEnabled()
-      : ForcedFallbackTest(CreateEnvironment(CreateTestFieldTrialsPtr(
-            std::string(kFieldTrial) + "/Enabled-" +
-            std::to_string(kMinPixelsPerFrame) + "," +
-            std::to_string(kWidth * kHeight) + ",30000/"))) {}
+      : ForcedFallbackTest(std::string(kFieldTrial) + "/Enabled-" +
+                           std::to_string(kMinPixelsPerFrame) + "," +
+                           std::to_string(kWidth * kHeight) + ",30000/") {}
 };
 
 class ForcedFallbackTestDisabled : public ForcedFallbackTest {
  public:
   ForcedFallbackTestDisabled()
-      : ForcedFallbackTest(CreateEnvironment(CreateTestFieldTrialsPtr(
-            std::string(kFieldTrial) + "/Disabled/"))) {}
+      : ForcedFallbackTest(std::string(kFieldTrial) + "/Disabled/") {}
 };
 
 TEST_F(ForcedFallbackTestDisabled, NoFallbackWithoutFieldTrial) {
@@ -716,7 +708,7 @@ TEST(SoftwareFallbackEncoderTest, BothRateControllersNotTrusted) {
 
   std::unique_ptr<VideoEncoder> wrapper =
       CreateVideoEncoderSoftwareFallbackWrapper(
-          CreateEnvironment(), std::unique_ptr<VideoEncoder>(sw_encoder),
+          CreateTestEnvironment(), std::unique_ptr<VideoEncoder>(sw_encoder),
           std::unique_ptr<VideoEncoder>(hw_encoder),
           /*prefer_temporal_support=*/false);
   EXPECT_FALSE(wrapper->GetEncoderInfo().has_trusted_rate_controller);
@@ -732,7 +724,7 @@ TEST(SoftwareFallbackEncoderTest, SwRateControllerTrusted) {
 
   std::unique_ptr<VideoEncoder> wrapper =
       CreateVideoEncoderSoftwareFallbackWrapper(
-          CreateEnvironment(), std::unique_ptr<VideoEncoder>(sw_encoder),
+          CreateTestEnvironment(), std::unique_ptr<VideoEncoder>(sw_encoder),
           std::unique_ptr<VideoEncoder>(hw_encoder),
           /*prefer_temporal_support=*/false);
   EXPECT_FALSE(wrapper->GetEncoderInfo().has_trusted_rate_controller);
@@ -748,7 +740,7 @@ TEST(SoftwareFallbackEncoderTest, HwRateControllerTrusted) {
 
   std::unique_ptr<VideoEncoder> wrapper =
       CreateVideoEncoderSoftwareFallbackWrapper(
-          CreateEnvironment(), std::unique_ptr<VideoEncoder>(sw_encoder),
+          CreateTestEnvironment(), std::unique_ptr<VideoEncoder>(sw_encoder),
           std::unique_ptr<VideoEncoder>(hw_encoder),
           /*prefer_temporal_support=*/false);
   EXPECT_TRUE(wrapper->GetEncoderInfo().has_trusted_rate_controller);
@@ -779,7 +771,7 @@ TEST(SoftwareFallbackEncoderTest, BothRateControllersTrusted) {
 
   std::unique_ptr<VideoEncoder> wrapper =
       CreateVideoEncoderSoftwareFallbackWrapper(
-          CreateEnvironment(), std::unique_ptr<VideoEncoder>(sw_encoder),
+          CreateTestEnvironment(), std::unique_ptr<VideoEncoder>(sw_encoder),
           std::unique_ptr<VideoEncoder>(hw_encoder),
           /*prefer_temporal_support=*/false);
   EXPECT_TRUE(wrapper->GetEncoderInfo().has_trusted_rate_controller);
@@ -795,7 +787,7 @@ TEST(SoftwareFallbackEncoderTest, ReportsHardwareAccelerated) {
 
   std::unique_ptr<VideoEncoder> wrapper =
       CreateVideoEncoderSoftwareFallbackWrapper(
-          CreateEnvironment(), std::unique_ptr<VideoEncoder>(sw_encoder),
+          CreateTestEnvironment(), std::unique_ptr<VideoEncoder>(sw_encoder),
           std::unique_ptr<VideoEncoder>(hw_encoder),
           /*prefer_temporal_support=*/false);
   EXPECT_TRUE(wrapper->GetEncoderInfo().is_hardware_accelerated);
@@ -825,7 +817,7 @@ TEST(SoftwareFallbackEncoderTest, ConfigureHardwareOnSecondAttempt) {
 
   std::unique_ptr<VideoEncoder> wrapper =
       CreateVideoEncoderSoftwareFallbackWrapper(
-          CreateEnvironment(), std::unique_ptr<VideoEncoder>(sw_encoder),
+          CreateTestEnvironment(), std::unique_ptr<VideoEncoder>(sw_encoder),
           std::unique_ptr<VideoEncoder>(hw_encoder),
           /*prefer_temporal_support=*/false);
   EXPECT_TRUE(wrapper->GetEncoderInfo().is_hardware_accelerated);
@@ -868,7 +860,7 @@ class PreferTemporalLayersFallbackTest : public ::testing::Test {
         .WillRepeatedly(Return(WEBRTC_VIDEO_CODEC_OK));
 
     wrapper_ = CreateVideoEncoderSoftwareFallbackWrapper(
-        CreateEnvironment(), std::unique_ptr<VideoEncoder>(sw_),
+        CreateTestEnvironment(), std::unique_ptr<VideoEncoder>(sw_),
         std::unique_ptr<VideoEncoder>(hw_),
         /*prefer_temporal_support=*/true);
 
@@ -1143,7 +1135,7 @@ INSTANTIATE_TEST_SUITE_P(
 TEST_P(ResolutionBasedFallbackTest, VerifyForcedEncoderFallback) {
   const ResolutionBasedFallbackTestParams& params = GetParam();
   const Environment env =
-      CreateEnvironment(CreateTestFieldTrialsPtr(params.field_trials));
+      CreateTestEnvironment({.field_trials = params.field_trials});
   auto primary = std::make_unique<test::FakeEncoder>(env);
   primary->SetImplementationName("primary");
   auto fallback = std::make_unique<test::FakeEncoder>(env);

From 210a0991ffdbcad3fb4660ea209291e5ad9211e3 Mon Sep 17 00:00:00 2001
From: Evan Shrubsole <eshr@webrtc.org>
Date: Tue, 17 Mar 2026 06:42:45 -0700
Subject: [PATCH 329/472] Revert "Removed the configuration options for
 multi_channel processing"
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This reverts commit db6adf2e9f9a213f9045e1c388713fe727245727.

Reason for revert: Breaks webrtc=>chrome roll https://crrev.com/c/7672366

Original change's description:
> Removed the configuration options for multi_channel processing
>
> This CL removes the explicit configuration options for whether
> the audio processing should be performed in a multi- or
> single-channel manner.
> This change means that the choice of whether to do multi- or
> single-channel audio processing will be made based on the
> content of the audio sent to APM for processing, and the
> audio output requested by APM.
>
> Bug: chromium:464314991
> Change-Id: I55a53f2565c81a7d7ad2e378f944ffcf07c71f4f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453000
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#47175}

Bug: chromium:464314991
Change-Id: If2888c379fe3a5bfdac6c12225a12705a1377c30
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/457640
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47182}
---
 api/audio/audio_processing.cc                 |  2 ++
 api/audio/audio_processing.h                  |  5 +++
 .../audio_processing/audio_processing_impl.cc | 35 +++++++++++++++++--
 .../audio_processing/audio_processing_impl.h  | 10 ++++--
 .../audio_processing_unittest.cc              |  2 ++
 .../test/audio_processing_simulator.cc        |  8 +++++
 .../test/audio_processing_simulator.h         |  2 ++
 .../test/audioproc_float_impl.cc              | 14 ++++++++
 .../audio_processing_configs_fuzzer.cc        |  2 ++
 .../audio_processing_sample_rate_fuzzer.cc    |  2 ++
 10 files changed, 78 insertions(+), 4 deletions(-)

diff --git a/api/audio/audio_processing.cc b/api/audio/audio_processing.cc
index 12ea953c85..bf0af030fe 100644
--- a/api/audio/audio_processing.cc
+++ b/api/audio/audio_processing.cc
@@ -131,6 +131,8 @@ std::string AudioProcessing::Config::ToString() const {
              "pipeline: { "
              "maximum_internal_processing_rate: "
           << pipeline.maximum_internal_processing_rate
+          << ", multi_channel_render: " << pipeline.multi_channel_render
+          << ", multi_channel_capture: " << pipeline.multi_channel_capture
           << " }, pre_amplifier: { enabled: " << pre_amplifier.enabled
           << ", fixed_gain_factor: " << pre_amplifier.fixed_gain_factor
           << " },capture_level_adjustment: { enabled: "
diff --git a/api/audio/audio_processing.h b/api/audio/audio_processing.h
index 4c3408d4c2..1e0ea0beee 100644
--- a/api/audio/audio_processing.h
+++ b/api/audio/audio_processing.h
@@ -146,6 +146,11 @@ class RTC_EXPORT AudioProcessing : public RefCountInterface {
       // Maximum allowed processing rate used internally. May only be set to
       // 32000 or 48000 and any differing values will be treated as 48000.
       int maximum_internal_processing_rate = 32000;
+      // Allow multi-channel processing of render audio.
+      bool multi_channel_render = true;
+      // Allow multi-channel processing of capture audio when AEC3 is active
+      // or a custom AEC is injected.
+      bool multi_channel_capture = true;
       // Indicates how to downmix multi-channel capture audio to mono (when
       // needed).
       DownmixMethod capture_downmix_method = DownmixMethod::kAverageChannels;
diff --git a/modules/audio_processing/audio_processing_impl.cc b/modules/audio_processing/audio_processing_impl.cc
index 2956078359..e86cc3cf69 100644
--- a/modules/audio_processing/audio_processing_impl.cc
+++ b/modules/audio_processing/audio_processing_impl.cc
@@ -468,6 +468,10 @@ AudioProcessingImpl::AudioProcessingImpl(
                   std::move(capture_analyzer),
                   std::move(neural_residual_echo_estimator)),
       constants_(
+          !env.field_trials().IsEnabled(
+              "WebRTC-ApmExperimentalMultiChannelRenderKillSwitch"),
+          !env.field_trials().IsEnabled(
+              "WebRTC-ApmExperimentalMultiChannelCaptureKillSwitch"),
           EnforceSplitBandHpf(env.field_trials()),
           MinimizeProcessingForUnusedOutput(env.field_trials()),
           env.field_trials().IsEnabled("WebRTC-ApmEnforce48kHzProcessingRate")),
@@ -649,9 +653,16 @@ void AudioProcessingImpl::InitializeLocked(const ProcessingConfig& config) {
              render_processing_rate == 48000);
 
   if (submodule_states_.RenderMultiBandSubModulesActive()) {
+    // By default, downmix the render stream to mono for analysis. This has been
+    // demonstrated to work well for AEC in most practical scenarios.
+    const bool multi_channel_render = config_.pipeline.multi_channel_render &&
+                                      constants_.multi_channel_render_support;
+    int render_processing_num_channels =
+        multi_channel_render
+            ? formats_.api_format.reverse_input_stream().num_channels()
+            : 1;
     formats_.render_processing_format =
-        StreamConfig(render_processing_rate,
-                     formats_.api_format.reverse_input_stream().num_channels());
+        StreamConfig(render_processing_rate, render_processing_num_channels);
   } else {
     formats_.render_processing_format = StreamConfig(
         formats_.api_format.reverse_input_stream().sample_rate_hz(),
@@ -669,6 +680,10 @@ void AudioProcessingImpl::ApplyConfig(const AudioProcessing::Config& config) {
   RTC_LOG(LS_INFO) << "AudioProcessing::ApplyConfig: " << config.ToString();
 
   const bool pipeline_config_changed =
+      config_.pipeline.multi_channel_render !=
+          config.pipeline.multi_channel_render ||
+      config_.pipeline.multi_channel_capture !=
+          config.pipeline.multi_channel_capture ||
       config_.pipeline.maximum_internal_processing_rate !=
           config.pipeline.maximum_internal_processing_rate ||
       config_.pipeline.capture_downmix_method !=
@@ -763,6 +778,12 @@ size_t AudioProcessingImpl::num_input_channels() const {
 }
 
 size_t AudioProcessingImpl::num_proc_channels() const {
+  // Used as callback from submodules, hence locking is not allowed.
+  const bool multi_channel_capture = config_.pipeline.multi_channel_capture &&
+                                     constants_.multi_channel_capture_support;
+  if (submodule_states_.EchoControllerEnabled() && !multi_channel_capture) {
+    return 1;
+  }
   return num_output_channels();
 }
 
@@ -1291,6 +1312,16 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
     capture_buffer->SplitIntoFrequencyBands();
   }
 
+  const bool multi_channel_capture = config_.pipeline.multi_channel_capture &&
+                                     constants_.multi_channel_capture_support;
+  if (submodules_.echo_controller && !multi_channel_capture) {
+    // Force down-mixing of the number of channels after the detection of
+    // capture signal saturation.
+    // TODO(peah): Look into ensuring that this kind of tampering with the
+    // AudioBuffer functionality should not be needed.
+    capture_buffer->set_num_channels(1);
+  }
+
   if (submodules_.high_pass_filter &&
       (!config_.high_pass_filter.apply_in_full_band ||
        constants_.enforce_split_band_hpf)) {
diff --git a/modules/audio_processing/audio_processing_impl.h b/modules/audio_processing/audio_processing_impl.h
index 302cf43bb2..c058961472 100644
--- a/modules/audio_processing/audio_processing_impl.h
+++ b/modules/audio_processing/audio_processing_impl.h
@@ -420,14 +420,20 @@ class AudioProcessingImpl : public AudioProcessing {
 
   // APM constants.
   const struct ApmConstants {
-    ApmConstants(bool enforce_split_band_hpf,
+    ApmConstants(bool multi_channel_render_support,
+                 bool multi_channel_capture_support,
+                 bool enforce_split_band_hpf,
                  bool minimize_processing_for_unused_output,
                  bool enforce_48_khz_max_internal_processing_rate)
-        : enforce_split_band_hpf(enforce_split_band_hpf),
+        : multi_channel_render_support(multi_channel_render_support),
+          multi_channel_capture_support(multi_channel_capture_support),
+          enforce_split_band_hpf(enforce_split_band_hpf),
           minimize_processing_for_unused_output(
               minimize_processing_for_unused_output),
           enforce_48_khz_max_internal_processing_rate(
               enforce_48_khz_max_internal_processing_rate) {}
+    bool multi_channel_render_support;
+    bool multi_channel_capture_support;
     bool enforce_split_band_hpf;
     bool minimize_processing_for_unused_output;
     bool enforce_48_khz_max_internal_processing_rate;
diff --git a/modules/audio_processing/audio_processing_unittest.cc b/modules/audio_processing/audio_processing_unittest.cc
index 03472bd568..7078370c64 100644
--- a/modules/audio_processing/audio_processing_unittest.cc
+++ b/modules/audio_processing/audio_processing_unittest.cc
@@ -2056,6 +2056,8 @@ void RunApmRateAndChannelTest(ArrayView<const int> sample_rates_hz,
                               ArrayView<const int> capture_channel_counts,
                               ArrayView<const bool> mono_capture_output) {
   AudioProcessing::Config apm_config;
+  apm_config.pipeline.multi_channel_render = true;
+  apm_config.pipeline.multi_channel_capture = true;
   apm_config.echo_canceller.enabled = true;
   scoped_refptr<AudioProcessing> apm =
       BuiltinAudioProcessingBuilder(apm_config).Build(CreateEnvironment());
diff --git a/modules/audio_processing/test/audio_processing_simulator.cc b/modules/audio_processing/test/audio_processing_simulator.cc
index be0a1c64b8..aa1af1737c 100644
--- a/modules/audio_processing/test/audio_processing_simulator.cc
+++ b/modules/audio_processing/test/audio_processing_simulator.cc
@@ -418,6 +418,14 @@ void AudioProcessingSimulator::ConfigureAudioProcessor() {
   if (settings_.use_ts) {
     apm_config.transient_suppression.enabled = *settings_.use_ts != 0;
   }
+  if (settings_.multi_channel_render) {
+    apm_config.pipeline.multi_channel_render = *settings_.multi_channel_render;
+  }
+
+  if (settings_.multi_channel_capture) {
+    apm_config.pipeline.multi_channel_capture =
+        *settings_.multi_channel_capture;
+  }
 
   if (settings_.use_adaptive_stereo_downmixing_for_aec) {
     if (*settings_.use_adaptive_stereo_downmixing_for_aec) {
diff --git a/modules/audio_processing/test/audio_processing_simulator.h b/modules/audio_processing/test/audio_processing_simulator.h
index 511981ccc5..1bf5c1b49f 100644
--- a/modules/audio_processing/test/audio_processing_simulator.h
+++ b/modules/audio_processing/test/audio_processing_simulator.h
@@ -128,6 +128,8 @@ struct SimulationSettings {
   std::optional<int> maximum_internal_processing_rate;
   int initial_mic_level;
   bool simulate_mic_gain = false;
+  std::optional<bool> multi_channel_render;
+  std::optional<bool> multi_channel_capture;
   std::optional<int> simulated_mic_kind;
   std::optional<int> frame_for_sending_capture_output_used_false;
   std::optional<int> frame_for_sending_capture_output_used_true;
diff --git a/modules/audio_processing/test/audioproc_float_impl.cc b/modules/audio_processing/test/audioproc_float_impl.cc
index d41aaed0b6..589fa1ad97 100644
--- a/modules/audio_processing/test/audioproc_float_impl.cc
+++ b/modules/audio_processing/test/audioproc_float_impl.cc
@@ -223,6 +223,16 @@ ABSL_FLAG(int,
           0,
           "Activate (1) or deactivate(0) the analog mic gain simulation in the "
           "test code");
+ABSL_FLAG(int,
+          multi_channel_render,
+          kParameterNotSpecifiedValue,
+          "Activate (1) or deactivate (0) multi-channel render processing in "
+          "APM pipeline");
+ABSL_FLAG(int,
+          multi_channel_capture,
+          kParameterNotSpecifiedValue,
+          "Activate (1) or deactivate (0) multi-channel capture processing in "
+          "APM pipeline");
 ABSL_FLAG(int,
           simulated_mic_kind,
           kParameterNotSpecifiedValue,
@@ -472,6 +482,10 @@ SimulationSettings CreateSettings() {
   SetSettingIfSpecified(absl::GetFlag(FLAGS_aec_settings),
                         &settings.aec_settings_filename);
   settings.initial_mic_level = absl::GetFlag(FLAGS_initial_mic_level);
+  SetSettingIfFlagSet(absl::GetFlag(FLAGS_multi_channel_render),
+                      &settings.multi_channel_render);
+  SetSettingIfFlagSet(absl::GetFlag(FLAGS_multi_channel_capture),
+                      &settings.multi_channel_capture);
   settings.simulate_mic_gain = absl::GetFlag(FLAGS_simulate_mic_gain);
   SetSettingIfSpecified(absl::GetFlag(FLAGS_simulated_mic_kind),
                         &settings.simulated_mic_kind);
diff --git a/test/fuzzers/audio_processing_configs_fuzzer.cc b/test/fuzzers/audio_processing_configs_fuzzer.cc
index 03ee5f3836..614cc66b37 100644
--- a/test/fuzzers/audio_processing_configs_fuzzer.cc
+++ b/test/fuzzers/audio_processing_configs_fuzzer.cc
@@ -105,6 +105,8 @@ webrtc::scoped_refptr<AudioProcessing> CreateApm(FuzzDataHelper* fuzz_data,
   }
 
   webrtc::AudioProcessing::Config apm_config;
+  apm_config.pipeline.multi_channel_render = true;
+  apm_config.pipeline.multi_channel_capture = true;
   apm_config.echo_canceller.enabled = use_aec || use_aecm;
   apm_config.high_pass_filter.enabled = use_hpf;
   apm_config.gain_controller1.enabled = use_agc;
diff --git a/test/fuzzers/audio_processing_sample_rate_fuzzer.cc b/test/fuzzers/audio_processing_sample_rate_fuzzer.cc
index e893f021ea..3f7bec7e4c 100644
--- a/test/fuzzers/audio_processing_sample_rate_fuzzer.cc
+++ b/test/fuzzers/audio_processing_sample_rate_fuzzer.cc
@@ -94,6 +94,8 @@ void FuzzOneInput(FuzzDataHelper fuzz_data) {
           : nullptr;
   scoped_refptr<AudioProcessing> apm =
       BuiltinAudioProcessingBuilder()
+          .SetConfig({.pipeline = {.multi_channel_render = true,
+                                   .multi_channel_capture = true}})
           .SetCapturePostProcessing(std::move(capture_processor))
           .SetRenderPreProcessing(std::move(render_processor))
           .Build(CreateEnvironment());

From 438e870e152ce588fd990202574a221a26e06eb7 Mon Sep 17 00:00:00 2001
From: Henrik Lundin <henrik.lundin@webrtc.org>
Date: Tue, 17 Mar 2026 15:18:28 +0000
Subject: [PATCH 330/472] Update kMaxPacketBufferSize to support 50ms @ 48kHz
 in RtpFileReader

Increased kMaxPacketBufferSize from 3500 to 5100 bytes to accommodate
50 ms packets at 48 kHz sample rate (4800 bytes for PCM16) plus the
original 300 bytes of overhead.

Bug: none
Change-Id: Ic89f5bf3883b10c7c97c4584bbc452e0b2b90ee8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/457621
Auto-Submit: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47183}
---
 test/rtp_file_reader.h | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/test/rtp_file_reader.h b/test/rtp_file_reader.h
index c7224af108..f889fbe509 100644
--- a/test/rtp_file_reader.h
+++ b/test/rtp_file_reader.h
@@ -20,9 +20,9 @@ namespace webrtc {
 namespace test {
 
 struct RtpPacket {
-  // Accommodate for 50 ms packets of 32 kHz PCM16 samples (3200 bytes) plus
+  // Accommodate for 50 ms packets of 48 kHz PCM16 samples (4800 bytes) plus
   // some overhead.
-  static const size_t kMaxPacketBufferSize = 3500;
+  static const size_t kMaxPacketBufferSize = 5100;
   uint8_t data[kMaxPacketBufferSize];
   size_t length;
   // The length the packet had on wire. Will be different from `length` when

From 4637b33722cc5b68ed5d12a7f588fd36ac4a01d9 Mon Sep 17 00:00:00 2001
From: Per K <perkj@webrtc.org>
Date: Tue, 17 Mar 2026 15:20:01 +0000
Subject: [PATCH 331/472] Ensure Scream only treat reported lost packets once
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Also add test to TransportFeedbackAdapterTest
- to verify that lost feedback packets are not treated as lost sent RTP packets.
- to verify that packets reported as implicitly and explicitly lost has the flag reported_lost_for_the_first_time set only the first time it is reported lost.

Bug: webrtc:483936079
Change-Id: Id820783fe8f7c774451d50b7a8dac4a07881dc16
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/457641
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47184}
---
 .../transport_feedback_adapter_unittest.cc    | 108 ++++++++++++++++++
 .../congestion_controller/scream/scream_v2.cc |   2 +-
 .../scream/scream_v2_unittest.cc              |  37 ++++++
 .../scream/test/cc_feedback_generator.cc      |   1 +
 4 files changed, 147 insertions(+), 1 deletion(-)

diff --git a/modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc b/modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc
index 1abf22f5f4..bdb63b7aad 100644
--- a/modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc
+++ b/modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc
@@ -606,6 +606,59 @@ TEST_P(TransportFeedbackAdapterTest, TransportPacketFeedbackHasDataInFlight) {
   EXPECT_EQ(adapted_feedback_2->data_in_flight, DataSize::Zero());
 }
 
+TEST_P(TransportFeedbackAdapterTest,
+       PacketsInLostFeedbackAreNotReportedAsLost) {
+  TransportFeedbackAdapter adapter;
+
+  std::vector<PacketTemplate> packets = {
+      {.transport_sequence_number = 1,
+       .rtp_sequence_number = 101,
+       .send_timestamp = Timestamp::Millis(100),
+       .receive_timestamp = Timestamp::Millis(120)},
+      {.transport_sequence_number = 2,
+       .rtp_sequence_number = 102,
+       .send_timestamp = Timestamp::Millis(110),
+       .receive_timestamp = Timestamp::Millis(130)},
+      {.transport_sequence_number = 3,
+       .rtp_sequence_number = 103,
+       .send_timestamp = Timestamp::Millis(120),
+       .receive_timestamp = Timestamp::Millis(140)},
+      {.transport_sequence_number = 4,
+       .rtp_sequence_number = 104,
+       .send_timestamp = Timestamp::Millis(130),
+       .receive_timestamp = Timestamp::Millis(150)}};
+
+  for (const PacketTemplate& packet : packets) {
+    adapter.AddPacket(CreatePacketToSend(packet), packet.pacing_info,
+                      /*overhead=*/0u, TimeNow());
+    adapter.ProcessSentPacket(SentPacketInfo(packet.transport_sequence_number,
+                                             packet.send_timestamp.ms()));
+  }
+
+  // Feedback 1: Covers packet 1
+  std::vector<PacketTemplate> feedback1_packets = {packets[0]};
+  CreateAndProcessFeedback(feedback1_packets, adapter);
+
+  // Feedback 2 (Lost network packet, never processed): would cover packet 2.
+
+  // Feedback 3: Covers packet 3
+  std::vector<PacketTemplate> feedback3_packets = {packets[2]};
+  std::optional<TransportPacketsFeedback> adapted_feedback3 =
+      CreateAndProcessFeedback(feedback3_packets, adapter);
+
+  // Packet 2 is not reported as lost.
+  EXPECT_FALSE(FindFeedback(adapted_feedback3, 2).has_value());
+
+  // Feedback 4: Covers packet 4
+  std::vector<PacketTemplate> feedback4_packets = {packets[3]};
+  std::optional<TransportPacketsFeedback> adapted_feedback4 =
+      CreateAndProcessFeedback(feedback4_packets, adapter);
+
+  // Both CCFB and TWCC never infer packet 2 as lost unless explicitly skipped
+  // within the sequence number range of a single received feedback packet.
+  EXPECT_FALSE(FindFeedback(adapted_feedback4, 2).has_value());
+}
+
 TEST(TransportFeedbackAdapterCongestionFeedbackTest,
      CongestionControlFeedbackResultHasEcn) {
   TransportFeedbackAdapter adapter;
@@ -812,6 +865,61 @@ TEST(TransportFeedbackAdapterCongestionFeedbackTest,
   EXPECT_FALSE(packet_feedback->reported_lost_for_the_first_time);
 }
 
+TEST(TransportFeedbackAdapterCongestionFeedbackTest,
+     CongestionControlFeedbackResultReportsImplicitlyLostPacketOnce) {
+  TransportFeedbackAdapter adapter;
+
+  PacketTemplate packets[] = {{.ssrc = 1,
+                               .transport_sequence_number = 1,
+                               .rtp_sequence_number = 101,
+                               .send_timestamp = Timestamp::Millis(100)},
+                              {.ssrc = 2,
+                               .transport_sequence_number = 2,
+                               .rtp_sequence_number = 201,
+                               .send_timestamp = Timestamp::Millis(110)},
+                              {.ssrc = 1,
+                               .transport_sequence_number = 3,
+                               .rtp_sequence_number = 102,
+                               .send_timestamp = Timestamp::Millis(120)},
+                              {.ssrc = 2,
+                               .transport_sequence_number = 4,
+                               .rtp_sequence_number = 202,
+                               .send_timestamp = Timestamp::Millis(120)}};
+
+  for (const PacketTemplate& packet : packets) {
+    adapter.AddPacket(CreatePacketToSend(packet), packet.pacing_info,
+                      /*overhead=*/0u, TimeNow());
+
+    adapter.ProcessSentPacket(SentPacketInfo(packet.transport_sequence_number,
+                                             packet.send_timestamp.ms()));
+  }
+
+  // Produce feedback where 2nd packet is lost.
+  packets[0].receive_timestamp = Timestamp::Millis(200);
+  packets[2].receive_timestamp = Timestamp::Millis(220);
+  std::vector<PacketTemplate> feedback_1 = {packets[0], packets[2]};
+  std::optional<PacketResult> packet_feedback = FindFeedback(
+      adapter.ProcessCongestionControlFeedback(
+          BuildRtcpCongestionControlFeedbackPacket(feedback_1), TimeNow()),
+      /*transport_sequence_number=*/2);
+  ASSERT_TRUE(packet_feedback.has_value());
+  EXPECT_FALSE(packet_feedback->IsReceived());
+
+  // Produce feedback where 2nd packet is reported lost and 4th packet is
+  // received.
+  packets[3].receive_timestamp = Timestamp::Millis(240);
+  std::vector<PacketTemplate> feedback_2 = {packets[1], packets[3]};
+  rtcp::CongestionControlFeedback rtcp_feedback =
+      BuildRtcpCongestionControlFeedbackPacket(feedback_2);
+  // 2nd packet is still lost.
+  packet_feedback = FindFeedback(
+      adapter.ProcessCongestionControlFeedback(rtcp_feedback, TimeNow()),
+      /*transport_sequence_number=*/2);
+  ASSERT_TRUE(packet_feedback.has_value());
+  EXPECT_FALSE(packet_feedback->IsReceived());
+  EXPECT_FALSE(packet_feedback->reported_lost_for_the_first_time);
+}
+
 TEST(TransportFeedbackAdapterCongestionFeedbackTest,
      CongestionControlFeedbackResultReportsRecoveredPacketOnce) {
   TransportFeedbackAdapter adapter;
diff --git a/modules/congestion_controller/scream/scream_v2.cc b/modules/congestion_controller/scream/scream_v2.cc
index fc5d533d19..ea458762cf 100644
--- a/modules/congestion_controller/scream/scream_v2.cc
+++ b/modules/congestion_controller/scream/scream_v2.cc
@@ -45,7 +45,7 @@ DataSize DataUnitsAckedAndNotMarked(const TransportPacketsFeedback& msg) {
 
 bool HasLostPackets(const TransportPacketsFeedback& msg) {
   for (const auto& packet : msg.PacketsWithFeedback()) {
-    if (!packet.IsReceived()) {
+    if (!packet.IsReceived() && packet.reported_lost_for_the_first_time) {
       return true;
     }
   }
diff --git a/modules/congestion_controller/scream/scream_v2_unittest.cc b/modules/congestion_controller/scream/scream_v2_unittest.cc
index 84ccf60607..39d89363b7 100644
--- a/modules/congestion_controller/scream/scream_v2_unittest.cc
+++ b/modules/congestion_controller/scream/scream_v2_unittest.cc
@@ -161,6 +161,43 @@ TEST(ScreamV2Test, ReferenceWindowIncreaseLessPerStepIfCeDetected) {
   EXPECT_GT(scream_1.ref_window(), scream_2.ref_window());
 }
 
+TEST(ScreamV2Test, ReferenceWindowDecreaseIfPacketsAreLostForTheFirstTime) {
+  SimulatedClock clock(Timestamp::Seconds(1'234));
+  Environment env = CreateTestEnvironment({.time = &clock});
+  ScreamV2 scream(env);
+
+  TransportPacketsFeedback feedback =
+      CreateFeedback(clock.CurrentTime(), /*rtt=*/TimeDelta::Millis(10),
+                     /*number_of_ect1_packets=*/20,
+                     /*number_of_packets_in_flight=*/20);
+
+  scream.OnTransportPacketsFeedback(feedback);
+  DataSize ref_window = scream.ref_window();
+  clock.AdvanceTime(TimeDelta::Millis(25));
+
+  TransportPacketsFeedback loss_feedback =
+      CreateFeedback(clock.CurrentTime(), /*rtt=*/TimeDelta::Millis(10),
+                     /*number_of_ect1_packets=*/5,
+                     /*number_of_packets_in_flight=*/5);
+  loss_feedback.packet_feedbacks[3].receive_time = Timestamp::PlusInfinity();
+  loss_feedback.packet_feedbacks[3].reported_lost_for_the_first_time = true;
+
+  scream.OnTransportPacketsFeedback(loss_feedback);
+  EXPECT_LT(scream.ref_window(), ref_window);
+  ref_window = scream.ref_window();
+
+  clock.AdvanceTime(TimeDelta::Millis(25));
+  TransportPacketsFeedback loss_feedback2 =
+      CreateFeedback(clock.CurrentTime(), /*rtt=*/TimeDelta::Millis(10),
+                     /*number_of_ect1_packets=*/5,
+                     /*number_of_packets_in_flight=*/5);
+  loss_feedback2.packet_feedbacks[0].receive_time = Timestamp::PlusInfinity();
+  loss_feedback2.packet_feedbacks[0].reported_lost_for_the_first_time = false;
+
+  scream.OnTransportPacketsFeedback(loss_feedback2);
+  EXPECT_GE(scream.ref_window(), ref_window);
+}
+
 TEST(ScreamV2Test, ReferenceWindowIncreaseToDataInflight) {
   SimulatedClock clock(Timestamp::Seconds(1'234));
   Environment env = CreateTestEnvironment({.time = &clock});
diff --git a/modules/congestion_controller/scream/test/cc_feedback_generator.cc b/modules/congestion_controller/scream/test/cc_feedback_generator.cc
index b754a2045b..1da5c34623 100644
--- a/modules/congestion_controller/scream/test/cc_feedback_generator.cc
+++ b/modules/congestion_controller/scream/test/cc_feedback_generator.cc
@@ -146,6 +146,7 @@ CcFeedbackGenerator::MaybeSendFeedback(Timestamp time) {
       PacketResult packet_result;
       packet_result.sent_packet.send_time =
           packets_in_flight_.front().send_time();
+      packet_result.reported_lost_for_the_first_time = true;
       packet_result.sent_packet.size = packets_in_flight_.front().packet_size();
       packets_in_flight_.pop_front();
       feedback.packet_feedbacks.push_back(packet_result);

From 6f76a57334e3d1f4629fdbf7f9d9bc3c38feadc9 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Tue, 17 Mar 2026 09:03:41 -0700
Subject: [PATCH 332/472] Roll chromium_revision eb32defcec..323d808a85
 (1600457:1600582)

Change log: https://chromium.googlesource.com/chromium/src/+log/eb32defcec..323d808a85
Full diff: https://chromium.googlesource.com/chromium/src/+/eb32defcec..323d808a85

Changed dependencies
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/17402a0a52..9879d6fdbb
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/5b9eb17956..31e1172ce8
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/17adda09b6..00e926ee2a
* src/third_party/androidx/cipd: wxjbyb5KcZ7SzBmiQuEU3Ou1KQf6VobxrK6ImmKDdvcC..cOeVS8IOLGMV68S9vCCdb9hdKvQOTDppiUZ5rIYtIZoC
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/6ea42cfd29..9bc4a2aee9
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/2ab1a62145..dc0931970a
* src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/851188a132..ea38df85bb
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/e7ca9ec9a0..4e4843ae07
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/d4d3f92cac..8a6bac0f6f
DEPS diff: https://chromium.googlesource.com/chromium/src/+/eb32defcec..323d808a85/DEPS

BUG=None

Change-Id: I851f7b3cb3307bce4222f7d640260e57d6fec2e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/457680
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47185}
---
 DEPS | 20 ++++++++++----------
 1 file changed, 10 insertions(+), 10 deletions(-)

diff --git a/DEPS b/DEPS
index 91f3e4f7af..d8fa77e459 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': 'eb32defcecc7ddcb69ea920b630f20cd6150106b',
+  'chromium_revision': '323d808a85b70c9defaf74279cf8de312aaf70d1',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@17402a0a523e51c56598f39bdf68fd5dd9c81b25',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@9879d6fdbb5cf31c1e4545cd528c9c22b720edab',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@5b9eb179561e84eb1fea1d331dd9a473a85ab7e0',
+    'https://chromium.googlesource.com/chromium/src/testing@31e1172ce87cfcea0bf34b71bb0a89085e184f36',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@17adda09b6a68ca16fc6156a06db61aff6dbc20d',
+    'https://chromium.googlesource.com/chromium/src/third_party@00e926ee2a354b1e2cab3e805cfdc8b04357574e',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -337,7 +337,7 @@ deps = {
   'src/third_party/libc++abi/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@8f11bb1d4438d0239d0dfc1bd9456a9f31629dda',
   'src/third_party/llvm-libc/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@851188a1329844b28215a7eb9f503b8e0f4c1ea0',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@ea38df85bbd4203028153aad85fdb148a9dfc5c3',
   'src/third_party/libunwind/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@db838d918570d4e381ecf9f5cc70a0098c9c2cd6',
 
@@ -492,7 +492,7 @@ deps = {
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@8be0e3114685fcc1589561067282edf75ea1259a',
   'src/third_party/catapult':
-    'https://chromium.googlesource.com/catapult.git@6ea42cfd29b4409cfdb830d1058d112359eaa9f3',
+    'https://chromium.googlesource.com/catapult.git@9bc4a2aee9361b6d2018c2f230c27c64a9e99b36',
   'src/third_party/ced/src': {
     'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5',
   },
@@ -511,7 +511,7 @@ deps = {
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@2ab1a6214568e5a0ddf85a9d47b055e80492d389',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@dc0931970a247ce29c9ea2d5944d5734651e55e6',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'e7ca9ec9a0d663b279b8936aa9548d9e2c4ffd46',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '4e4843ae073504d2f58540031a8f41471fa86a76',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@d4d3f92cacfdf6d3ef316dcad8273d8d98f3393f',
+    'https://chromium.googlesource.com/chromium/src/tools@8a6bac0f6f4bcce2df08f3817e59a7bbe07afd89',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'wxjbyb5KcZ7SzBmiQuEU3Ou1KQf6VobxrK6ImmKDdvcC',
+          'version': 'cOeVS8IOLGMV68S9vCCdb9hdKvQOTDppiUZ5rIYtIZoC',
       },
     ],
     'condition': 'checkout_android and non_git_source',

From 16be9d989debc23bfd2d1bca93728833eb93625d Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Erik=20Spr=C3=A5ng?= <sprang@webrtc.org>
Date: Tue, 17 Mar 2026 15:51:18 +0100
Subject: [PATCH 333/472] Fix racy pending frame status in
 SimulcastEncoderAdapter.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

While each sub-encoder (in a StreamContext) can be assumed to process
frames in order, there is the chance for a data race since the
`OnEncodedImage` callback can happen asynchronously from another thread.
This CL makes sure to protect the `pending_rtp_timestamps_` list using a
mutex to prevent issue in such cases.

A pretty complex unit test is also added, that is consistely able to
reproduce this problem when executed under TSAN.

Bug: webrtc:467444018
Change-Id: I595dd403558549b338ac6a67ebc87f6cb610a481
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/457620
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47186}
---
 media/BUILD.gn                                |   6 +
 media/engine/simulcast_encoder_adapter.cc     |  37 +++-
 media/engine/simulcast_encoder_adapter.h      |   3 +-
 .../simulcast_encoder_adapter_unittest.cc     | 194 ++++++++++++++++--
 4 files changed, 209 insertions(+), 31 deletions(-)

diff --git a/media/BUILD.gn b/media/BUILD.gn
index 9b026bf244..d02dea6768 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -967,6 +967,7 @@ if (rtc_include_tests) {
         "../api:mock_video_bitrate_allocator_factory",
         "../api:mock_video_codec_factory",
         "../api:mock_video_decoder",
+        "../api:mock_video_encoder",
         "../api:priority",
         "../api:ref_count",
         "../api:rtc_error",
@@ -986,6 +987,7 @@ if (rtc_include_tests) {
         "../api/crypto:options",
         "../api/environment",
         "../api/environment:environment_factory",
+        "../api/task_queue",
         "../api/test/video:function_video_factory",
         "../api/transport:bitrate_settings",
         "../api/transport:datagram_transport_interface",
@@ -1037,13 +1039,17 @@ if (rtc_include_tests) {
         "../rtc_base:byte_order",
         "../rtc_base:checks",
         "../rtc_base:copy_on_write_buffer",
+        "../rtc_base:cpu_info",
         "../rtc_base:dscp",
+        "../rtc_base:macromagic",
+        "../rtc_base:rtc_event",
         "../rtc_base:safe_conversions",
         "../rtc_base:socket",
         "../rtc_base:threading",
         "../rtc_base:timeutils",
         "../rtc_base:unique_id_generator",
         "../rtc_base/experiments:min_video_bitrate_experiment",
+        "../rtc_base/synchronization:mutex",
         "../rtc_base/system:file_wrapper",
         "../system_wrappers",
         "../test:audio_codec_mocks",
diff --git a/media/engine/simulcast_encoder_adapter.cc b/media/engine/simulcast_encoder_adapter.cc
index c25d0a5394..fbed6ef915 100644
--- a/media/engine/simulcast_encoder_adapter.cc
+++ b/media/engine/simulcast_encoder_adapter.cc
@@ -196,9 +196,9 @@ SimulcastEncoderAdapter::StreamContext::StreamContext(
       is_keyframe_needed_(false),
       is_paused_(is_paused) {
   if (parent_) {
+    // If we have a parent, this encoder is not running in bypass mode and we
+    // should funnnel the callbacks back via this stream context.
     encoder_context_->encoder().RegisterEncodeCompleteCallback(this);
-  } else {
-    RTC_LOG(LS_ERROR) << "[SEA] StreamContext ctor parent is null";
   }
 }
 
@@ -212,6 +212,8 @@ SimulcastEncoderAdapter::StreamContext::StreamContext(StreamContext&& rhs)
       is_keyframe_needed_(rhs.is_keyframe_needed_),
       is_paused_(rhs.is_paused_) {
   if (parent_) {
+    // If we have a parent, this encoder is not running in bypass mode and we
+    // should funnel the callbacks back via this stream context.
     encoder_context_->encoder().RegisterEncodeCompleteCallback(this);
   }
 }
@@ -273,9 +275,13 @@ void SimulcastEncoderAdapter::StreamContext::OnFrameDropped(
 int SimulcastEncoderAdapter::StreamContext::Encode(
     const VideoFrame& frame,
     const std::vector<VideoFrameType>* frame_types) {
-  pending_rtp_timestamps_.push_back(frame.rtp_timestamp());
+  {
+    MutexLock lock(&queue_mutex_);
+    pending_rtp_timestamps_.push_back(frame.rtp_timestamp());
+  }
   int ret = encoder().Encode(frame, frame_types);
   if (ret != WEBRTC_VIDEO_CODEC_OK) {
+    MutexLock lock(&queue_mutex_);
     RTC_DCHECK_EQ(pending_rtp_timestamps_.back(), frame.rtp_timestamp());
     pending_rtp_timestamps_.pop_back();
   }
@@ -284,15 +290,24 @@ int SimulcastEncoderAdapter::StreamContext::Encode(
 
 void SimulcastEncoderAdapter::StreamContext::MaybeCullOldTimestamps(
     uint32_t new_rtp_timestamp) {
-  while (!pending_rtp_timestamps_.empty() &&
-         pending_rtp_timestamps_.front() != new_rtp_timestamp &&
-         IsNewerTimestamp(new_rtp_timestamp, pending_rtp_timestamps_.front())) {
-    parent_->OnFrameDropped(pending_rtp_timestamps_.front(), stream_idx_);
-    pending_rtp_timestamps_.pop_front();
+  std::vector<uint32_t> dropped_timestamps;
+  {
+    MutexLock lock(&queue_mutex_);
+    while (
+        !pending_rtp_timestamps_.empty() &&
+        pending_rtp_timestamps_.front() != new_rtp_timestamp &&
+        IsNewerTimestamp(new_rtp_timestamp, pending_rtp_timestamps_.front())) {
+      dropped_timestamps.push_back(pending_rtp_timestamps_.front());
+      pending_rtp_timestamps_.pop_front();
+    }
+    if (!pending_rtp_timestamps_.empty() &&
+        pending_rtp_timestamps_.front() == new_rtp_timestamp) {
+      pending_rtp_timestamps_.pop_front();
+    }
   }
-  if (!pending_rtp_timestamps_.empty() &&
-      pending_rtp_timestamps_.front() == new_rtp_timestamp) {
-    pending_rtp_timestamps_.pop_front();
+  // Make callback outside lock to avoid potential deadlocks.
+  for (uint32_t timestamp : dropped_timestamps) {
+    parent_->OnFrameDropped(timestamp, stream_idx_);
   }
 }
 
diff --git a/media/engine/simulcast_encoder_adapter.h b/media/engine/simulcast_encoder_adapter.h
index 0bc6c40eb1..99ca19d431 100644
--- a/media/engine/simulcast_encoder_adapter.h
+++ b/media/engine/simulcast_encoder_adapter.h
@@ -159,7 +159,8 @@ class RTC_EXPORT SimulcastEncoderAdapter : public VideoEncoder {
     SimulcastEncoderAdapter* const parent_;
     std::unique_ptr<EncoderContext> encoder_context_;
     std::unique_ptr<FramerateController> framerate_controller_;
-    std::deque<uint32_t> pending_rtp_timestamps_;
+    mutable Mutex queue_mutex_;
+    std::deque<uint32_t> pending_rtp_timestamps_ RTC_GUARDED_BY(queue_mutex_);
     const int stream_idx_;
     const uint16_t width_;
     const uint16_t height_;
diff --git a/media/engine/simulcast_encoder_adapter_unittest.cc b/media/engine/simulcast_encoder_adapter_unittest.cc
index ca0bb2bded..df2894c339 100644
--- a/media/engine/simulcast_encoder_adapter_unittest.cc
+++ b/media/engine/simulcast_encoder_adapter_unittest.cc
@@ -10,9 +10,11 @@
 
 #include "media/engine/simulcast_encoder_adapter.h"
 
+#include <stddef.h>
+#include <stdint.h>
+
 #include <array>
-#include <cstddef>
-#include <cstdint>
+#include <atomic>
 #include <memory>
 #include <optional>
 #include <string>
@@ -21,6 +23,7 @@
 #include <vector>
 
 #include "absl/container/inlined_vector.h"
+#include "absl/functional/any_invocable.h"
 #include "api/environment/environment.h"
 #include "api/environment/environment_factory.h"
 #include "api/fec_controller_override.h"
@@ -28,13 +31,17 @@
 #include "api/make_ref_counted.h"
 #include "api/rtp_parameters.h"
 #include "api/scoped_refptr.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/task_queue/task_queue_factory.h"
 #include "api/test/create_simulcast_test_fixture.h"
 #include "api/test/mock_video_bitrate_allocator.h"
 #include "api/test/mock_video_decoder.h"
+#include "api/test/mock_video_encoder.h"
 #include "api/test/simulcast_test_fixture.h"
 #include "api/test/video/function_video_decoder_factory.h"
 #include "api/test/video/function_video_encoder_factory.h"
 #include "api/units/data_rate.h"
+#include "api/units/timestamp.h"
 #include "api/video/encoded_image.h"
 #include "api/video/i420_buffer.h"
 #include "api/video/video_bitrate_allocation.h"
@@ -58,6 +65,10 @@
 #include "modules/video_coding/utility/simulcast_rate_allocator.h"
 #include "modules/video_coding/utility/simulcast_test_fixture_impl.h"
 #include "rtc_base/checks.h"
+#include "rtc_base/cpu_info.h"
+#include "rtc_base/event.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
 #include "test/create_test_field_trials.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
@@ -198,6 +209,7 @@ class MockVideoEncoder;
 
 class MockVideoEncoderFactory : public VideoEncoderFactory {
  public:
+  explicit MockVideoEncoderFactory(const Environment& env) : env_(env) {}
   std::vector<SdpVideoFormat> GetSupportedFormats() const override;
 
   std::unique_ptr<VideoEncoder> Create(const Environment& env,
@@ -223,10 +235,18 @@ class MockVideoEncoderFactory : public VideoEncoderFactory {
   void set_fallback_from_simulcast(std::optional<int32_t> return_value) {
     fallback_from_simulcast_ = return_value;
   }
+  TaskQueueBase* GetAsyncEncoderQueue() {
+    if (!async_encoder_queue_) {
+      async_encoder_queue_ = env_.task_queue_factory().CreateTaskQueue(
+          "AsyncEncoderQueue", TaskQueueFactory::Priority::NORMAL);
+    }
+    return async_encoder_queue_.get();
+  }
 
   void DestroyVideoEncoder(VideoEncoder* encoder);
 
  private:
+  const Environment& env_;
   bool create_video_encoder_return_nullptr_ = false;
   int32_t init_encode_return_value_ = 0;
   std::optional<int32_t> fallback_from_simulcast_;
@@ -236,6 +256,7 @@ class MockVideoEncoderFactory : public VideoEncoderFactory {
   std::vector<uint32_t> requested_resolution_alignments_ = {1, 1, 1};
   bool supports_simulcast_ = false;
   std::vector<VideoEncoder::ResolutionBitrateLimits> resolution_bitrate_limits_;
+  std::unique_ptr<TaskQueueBase, TaskQueueDeleter> async_encoder_queue_;
 };
 
 class MockVideoEncoder : public VideoEncoder {
@@ -272,7 +293,11 @@ class MockVideoEncoder : public VideoEncoder {
     return 0;
   }
 
-  MOCK_METHOD(int32_t, Release, (), (override));
+  MOCK_METHOD(int32_t, ReleaseMock, ());
+  int32_t Release() override {
+    RunUntilIdle();
+    return ReleaseMock();
+  }
 
   void SetRates(const RateControlParameters& parameters) override {
     last_set_rates_ = parameters;
@@ -319,6 +344,23 @@ class MockVideoEncoder : public VideoEncoder {
     callback_->OnEncodedImage(image, &codec_specific_info);
   }
 
+  void PostTask(absl::AnyInvocable<void() &&> task) {
+    if (is_async_) {
+      factory_->GetAsyncEncoderQueue()->PostTask(std::move(task));
+    } else {
+      std::move(task)();
+    }
+  }
+
+  void set_is_async(bool is_async) {
+    if (is_async_ && !is_async) {
+      // Turning from asynchronous to synchronous. Wait for any pending tasks to
+      // complete.
+      RunUntilIdle();
+    }
+    is_async_ = is_async;
+  }
+
   void set_supports_native_handle(bool enabled) {
     supports_native_handle_ = enabled;
   }
@@ -388,6 +430,13 @@ class MockVideoEncoder : public VideoEncoder {
   SdpVideoFormat video_format() const { return video_format_; }
 
  private:
+  void RunUntilIdle() {
+    // This is dangerous, only do it here for testing.
+    Event event;
+    factory_->GetAsyncEncoderQueue()->PostTask([&event] { event.Set(); });
+    event.Wait(Event::kForever);
+  }
+
   MockVideoEncoderFactory* const factory_;
   bool supports_native_handle_ = false;
   std::string implementation_name_ = "unknown";
@@ -405,6 +454,7 @@ class MockVideoEncoder : public VideoEncoder {
   std::optional<bool> is_qp_trusted_;
   SdpVideoFormat video_format_;
   std::vector<VideoEncoder::ResolutionBitrateLimits> resolution_bitrate_limits;
+  bool is_async_ = false;
 
   VideoCodec codec_;
   EncodedImageCallback* callback_;
@@ -467,8 +517,9 @@ class TestSimulcastEncoderAdapterFakeHelper {
       bool use_fallback_factory,
       const SdpVideoFormat& video_format)
       : env_(env),
+        primary_factory_(env_),
         fallback_factory_(use_fallback_factory
-                              ? std::make_unique<MockVideoEncoderFactory>()
+                              ? std::make_unique<MockVideoEncoderFactory>(env_)
                               : nullptr),
         video_format_(video_format) {}
 
@@ -508,6 +559,7 @@ class TestSimulcastEncoderAdapterFake : public ::testing::Test,
         env_, use_fallback_factory_,
         SdpVideoFormat("VP8", sdp_video_parameters_));
     adapter_ = helper_->CreateMockEncoderAdapter();
+    MutexLock lock(&mutex_);
     last_encoded_image_width_ = std::nullopt;
     last_encoded_image_height_ = std::nullopt;
     last_encoded_image_simulcast_index_ = std::nullopt;
@@ -526,6 +578,7 @@ class TestSimulcastEncoderAdapterFake : public ::testing::Test,
   Result OnEncodedImage(
       const EncodedImage& encoded_image,
       const CodecSpecificInfo* /* codec_specific_info */) override {
+    MutexLock lock(&mutex_);
     last_encoded_image_width_ = encoded_image._encodedWidth;
     last_encoded_image_height_ = encoded_image._encodedHeight;
     last_encoded_image_simulcast_index_ = encoded_image.SimulcastIndex();
@@ -547,6 +600,7 @@ class TestSimulcastEncoderAdapterFake : public ::testing::Test,
   bool GetLastEncodedImageInfo(std::optional<int>* out_width,
                                std::optional<int>* out_height,
                                std::optional<int>* out_simulcast_index) {
+    MutexLock lock(&mutex_);
     if (!last_encoded_image_width_.has_value()) {
       return false;
     }
@@ -700,13 +754,14 @@ class TestSimulcastEncoderAdapterFake : public ::testing::Test,
 
  protected:
   FieldTrials field_trials_ = CreateTestFieldTrials();
-  Environment env_ = CreateEnvironment(field_trials_.CreateCopy());
+  Environment env_ = EnvironmentFactory().Create();
   std::unique_ptr<TestSimulcastEncoderAdapterFakeHelper> helper_;
   std::unique_ptr<VideoEncoder> adapter_;
   VideoCodec codec_;
-  std::optional<int> last_encoded_image_width_;
-  std::optional<int> last_encoded_image_height_;
-  std::optional<int> last_encoded_image_simulcast_index_;
+  std::optional<int> last_encoded_image_width_ RTC_GUARDED_BY(mutex_);
+  std::optional<int> last_encoded_image_height_ RTC_GUARDED_BY(mutex_);
+  std::optional<int> last_encoded_image_simulcast_index_ RTC_GUARDED_BY(mutex_);
+  Mutex mutex_;
   std::unique_ptr<VideoBitrateAllocator> rate_allocator_;
   bool use_fallback_factory_;
   CodecParameterMap sdp_video_parameters_;
@@ -937,11 +992,11 @@ TEST_F(TestSimulcastEncoderAdapterFake, ReusesEncodersInOrder) {
       .WillOnce(Return(WEBRTC_VIDEO_CODEC_OK));
   frame_types.resize(3, VideoFrameType::kVideoFrameKey);
   EXPECT_EQ(0, adapter_->Encode(input_frame, &frame_types));
-  EXPECT_CALL(*original_encoders[0], Release())
+  EXPECT_CALL(*original_encoders[0], ReleaseMock())
       .WillOnce(Return(WEBRTC_VIDEO_CODEC_OK));
-  EXPECT_CALL(*original_encoders[1], Release())
+  EXPECT_CALL(*original_encoders[1], ReleaseMock())
       .WillOnce(Return(WEBRTC_VIDEO_CODEC_OK));
-  EXPECT_CALL(*original_encoders[2], Release())
+  EXPECT_CALL(*original_encoders[2], ReleaseMock())
       .WillOnce(Return(WEBRTC_VIDEO_CODEC_OK));
   EXPECT_EQ(0, adapter_->Release());
 
@@ -964,9 +1019,9 @@ TEST_F(TestSimulcastEncoderAdapterFake, ReusesEncodersInOrder) {
       .WillOnce(Return(WEBRTC_VIDEO_CODEC_OK));
   frame_types.resize(2, VideoFrameType::kVideoFrameKey);
   EXPECT_EQ(0, adapter_->Encode(input_frame, &frame_types));
-  EXPECT_CALL(*original_encoders[0], Release())
+  EXPECT_CALL(*original_encoders[0], ReleaseMock())
       .WillOnce(Return(WEBRTC_VIDEO_CODEC_OK));
-  EXPECT_CALL(*original_encoders[1], Release())
+  EXPECT_CALL(*original_encoders[1], ReleaseMock())
       .WillOnce(Return(WEBRTC_VIDEO_CODEC_OK));
   EXPECT_EQ(0, adapter_->Release());
 
@@ -986,7 +1041,7 @@ TEST_F(TestSimulcastEncoderAdapterFake, ReusesEncodersInOrder) {
       .WillOnce(Return(WEBRTC_VIDEO_CODEC_OK));
   frame_types.resize(1, VideoFrameType::kVideoFrameKey);
   EXPECT_EQ(0, adapter_->Encode(input_frame, &frame_types));
-  EXPECT_CALL(*original_encoders[0], Release())
+  EXPECT_CALL(*original_encoders[0], ReleaseMock())
       .WillOnce(Return(WEBRTC_VIDEO_CODEC_OK));
   EXPECT_EQ(0, adapter_->Release());
 
@@ -1012,11 +1067,11 @@ TEST_F(TestSimulcastEncoderAdapterFake, ReusesEncodersInOrder) {
       .WillOnce(Return(WEBRTC_VIDEO_CODEC_OK));
   frame_types.resize(3, VideoFrameType::kVideoFrameKey);
   EXPECT_EQ(0, adapter_->Encode(input_frame, &frame_types));
-  EXPECT_CALL(*original_encoders[0], Release())
+  EXPECT_CALL(*original_encoders[0], ReleaseMock())
       .WillOnce(Return(WEBRTC_VIDEO_CODEC_OK));
-  EXPECT_CALL(*new_encoders[1], Release())
+  EXPECT_CALL(*new_encoders[1], ReleaseMock())
       .WillOnce(Return(WEBRTC_VIDEO_CODEC_OK));
-  EXPECT_CALL(*new_encoders[2], Release())
+  EXPECT_CALL(*new_encoders[2], ReleaseMock())
       .WillOnce(Return(WEBRTC_VIDEO_CODEC_OK));
   EXPECT_EQ(0, adapter_->Release());
 }
@@ -2577,14 +2632,14 @@ TEST_F(TestSimulcastEncoderAdapterFake,
   EXPECT_EQ(helper_->factory()->encoders()[0], prev_encoder);
 
   // Singlecast, an upper stream is active. Encoder should be recreated.
-  EXPECT_CALL(*prev_encoder, Release()).Times(1);
+  EXPECT_CALL(*prev_encoder, ReleaseMock()).Times(1);
   SetupCodec(/*active_streams=*/{false, true});
   ASSERT_EQ(1u, helper_->factory()->encoders().size());
   EXPECT_NE(helper_->factory()->encoders()[0], prev_encoder);
 
   // Singlecast, the lowest stream is active. Encoder should be recreated.
   prev_encoder = helper_->factory()->encoders()[0];
-  EXPECT_CALL(*prev_encoder, Release()).Times(1);
+  EXPECT_CALL(*prev_encoder, ReleaseMock()).Times(1);
   SetupCodec(/*active_streams=*/{true, false});
   ASSERT_EQ(1u, helper_->factory()->encoders().size());
   EXPECT_NE(helper_->factory()->encoders()[0], prev_encoder);
@@ -2785,5 +2840,106 @@ TEST_F(TestSimulcastEncoderAdapterFake,
             adapter_->Encode(input_frame, &frame_types));
 }
 
+TEST_F(TestSimulcastEncoderAdapterFake, ConcurrentEncodeAndOnEncodedImage) {
+  // The test setup here is quite complex, but that is needed in order to make
+  // sure tools like TSAN is able to reliable detect issue in the wrapper
+  // encoder instances used for each stream. The code below makes sure that
+  // there is concurrent usage of both an `Encode()` to request encoding of
+  // a new frame at the same time that an encoder is calling `OnEncodedImage()`
+  // on a separate thread.
+
+  if (cpu_info::DetectNumberOfCores() <= 2) {
+    GTEST_SKIP()
+        << "Skipping test on low core count machine due to risk of flakiness.";
+  }
+
+  SetUp();
+  helper_->factory()->set_supports_simulcast(false);
+  SetupCodec({true, true});
+
+  std::vector<MockVideoEncoder*> encoders = helper_->factory()->encoders();
+  ASSERT_EQ(2u, encoders.size());
+  encoders[0]->set_is_async(true);
+  encoders[1]->set_is_async(true);
+
+  std::atomic<bool> wait_for_second_frame{false};
+
+  EXPECT_CALL(*encoders[0], ReleaseMock()).WillRepeatedly(testing::Return(0));
+  EXPECT_CALL(*encoders[1], ReleaseMock()).WillRepeatedly(testing::Return(0));
+
+  MockEncodedImageCallback mock_encoded_image_callback;
+  EXPECT_CALL(mock_encoded_image_callback, OnEncodedImage)
+      .WillRepeatedly(Return(
+          EncodedImageCallback::Result(EncodedImageCallback::Result::OK)));
+  adapter_->RegisterEncodeCompleteCallback(&mock_encoded_image_callback);
+  EXPECT_CALL(*encoders[0], Encode)
+      .WillRepeatedly([&wait_for_second_frame, encoders](
+                          const VideoFrame& frame,
+                          const std::vector<VideoFrameType>* types) {
+        if (frame.rtp_timestamp() == 100) {
+          encoders[0]->PostTask([width = frame.width(), height = frame.height(),
+                                 rtp_timestamp = frame.rtp_timestamp(),
+                                 &wait_for_second_frame,
+                                 encoder = encoders[0]]() {
+            // Spin until the main thread is just about to encode the second
+            // frame. Using relaxed memory order prevents TSAN from seeing a
+            // happens-before edge but abort after 1000000 iterations.
+            constexpr int kMaxLoops = 1000000;
+            int loops = 0;
+            while (!wait_for_second_frame.load(std::memory_order_relaxed) &&
+                   ++loops <= kMaxLoops) {
+            }
+            EXPECT_LT(loops, kMaxLoops);
+            encoder->SendEncodedImage(width, height, rtp_timestamp,
+                                      /*simulcast_index=*/1);
+          });
+        }
+        return WEBRTC_VIDEO_CODEC_OK;
+      });
+
+  EXPECT_CALL(*encoders[1], Encode)
+      .WillRepeatedly([&](const VideoFrame& frame,
+                          const std::vector<VideoFrameType>* types) {
+        encoders[1]->SendEncodedImage(frame.width() / 2, frame.height() / 2,
+                                      frame.rtp_timestamp(),
+                                      /*simulcast_index=*/0);
+        return WEBRTC_VIDEO_CODEC_OK;
+      });
+
+  scoped_refptr<VideoFrameBuffer> buffer(I420Buffer::Create(1280, 720));
+
+  std::vector<VideoFrameType> frame_types0{VideoFrameType::kVideoFrameKey,
+                                           VideoFrameType::kVideoFrameKey};
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            adapter_->Encode(VideoFrame::Builder()
+                                 .set_video_frame_buffer(buffer)
+                                 .set_rtp_timestamp(100)
+                                 .set_timestamp_ms(0)
+                                 .build(),
+                             &frame_types0));
+
+  std::vector<VideoFrameType> frame_types1{VideoFrameType::kVideoFrameDelta,
+                                           VideoFrameType::kVideoFrameDelta};
+
+  // Signal the async thread to proceed. To maximize the chance of a collision,
+  // we use a relaxed atomic flag which TSAN won't treat as a happens-before
+  // edge.
+  wait_for_second_frame.store(true, std::memory_order_relaxed);
+
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            adapter_->Encode(VideoFrame::Builder()
+                                 .set_video_frame_buffer(buffer)
+                                 .set_rtp_timestamp(90100)
+                                 .set_timestamp_ms(1000)
+                                 .build(),
+                             &frame_types1));
+
+  // The callbacks should have executed via the task queue, but we don't have
+  // strict ordering guarantees on the main thread until we call adapter
+  // Release, which will indirectly call Release on the encoders, flushing
+  // their queues.
+  EXPECT_EQ(0, adapter_->Release());
+}
+
 }  // namespace test
 }  // namespace webrtc

From 262568c8c375391e2df6a12691d2a3f1b59bc3c5 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Tue, 17 Mar 2026 13:03:00 -0700
Subject: [PATCH 334/472] Roll chromium_revision 323d808a85..19b05ed41b
 (1600582:1600751)

Change log: https://chromium.googlesource.com/chromium/src/+log/323d808a85..19b05ed41b
Full diff: https://chromium.googlesource.com/chromium/src/+/323d808a85..19b05ed41b

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/00b379c45d..5c31389b03
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/9879d6fdbb..c1c27ffc20
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/31e1172ce8..7953480e67
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/00e926ee2a..0bc2df1a46
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/dc0931970a..e14fa05100
* src/third_party/r8/cipd: UMLn7pE7DklTsk-SIiMAwch46m8rSF1vG2HQAbMuZvMC..o1h6_6tQfYYay5tJcyLeEi-HODtdK5nFoPd6fls0oTkC
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/8a6bac0f6f..fa4e6f5160
DEPS diff: https://chromium.googlesource.com/chromium/src/+/323d808a85..19b05ed41b/DEPS

BUG=None

Change-Id: Ie7dc17af0335f65eb27907ee07dff32bb06a2477
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/457720
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47187}
---
 DEPS | 16 ++++++++--------
 1 file changed, 8 insertions(+), 8 deletions(-)

diff --git a/DEPS b/DEPS
index d8fa77e459..adbf56a926 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '323d808a85b70c9defaf74279cf8de312aaf70d1',
+  'chromium_revision': '19b05ed41bafecd62c8aeb02854faba3fb5c3375',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@00b379c45d62148b8b0f273cb295356d6039739d',
+    'https://chromium.googlesource.com/chromium/src/build@5c31389b03e59818d2ee155be40cbaf1798cbd20',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@95ed44cf5f06dbb5861030b91c9db9ccb4316762',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@9879d6fdbb5cf31c1e4545cd528c9c22b720edab',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@c1c27ffc20d75442d2334a308fdaa797ae5d816b',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@31e1172ce87cfcea0bf34b71bb0a89085e184f36',
+    'https://chromium.googlesource.com/chromium/src/testing@7953480e671e06cb123edc3e067675f07ea07974',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@00e926ee2a354b1e2cab3e805cfdc8b04357574e',
+    'https://chromium.googlesource.com/chromium/src/third_party@0bc2df1a46cbf66035d205ef1642dff693a7e7f1',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -511,7 +511,7 @@ deps = {
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@dc0931970a247ce29c9ea2d5944d5734651e55e6',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@e14fa05100494f45aab3838f09764c59e628b427',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -644,7 +644,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/r8',
-              'version': 'UMLn7pE7DklTsk-SIiMAwch46m8rSF1vG2HQAbMuZvMC',
+              'version': 'o1h6_6tQfYYay5tJcyLeEi-HODtdK5nFoPd6fls0oTkC',
           },
       ],
       'condition': 'checkout_android',
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@8a6bac0f6f4bcce2df08f3817e59a7bbe07afd89',
+    'https://chromium.googlesource.com/chromium/src/tools@fa4e6f516009593a81741f98cbc417ae251a5a07',
 
   'src/third_party/espresso': {
       'packages': [

From 29ae1267d1a3c8a087164a35e23784ada337fdf9 Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Tue, 17 Mar 2026 21:02:50 -0700
Subject: [PATCH 335/472] Update WebRTC code version (2026-03-18T04:02:27).

Bug: None
Change-Id: I23e06da791205a291a6d07294d434e5cc990b34e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/457840
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47188}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index e1facb4943..e361931b2e 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-17T04:10:08";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-18T04:02:27";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From deeeb70e3a3bf003da1c58d217fea4d0e7d4b595 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Wed, 18 Mar 2026 01:03:28 -0700
Subject: [PATCH 336/472] Roll chromium_revision 19b05ed41b..b30d84fa40
 (1600751:1601092)

Change log: https://chromium.googlesource.com/chromium/src/+log/19b05ed41b..b30d84fa40
Full diff: https://chromium.googlesource.com/chromium/src/+/19b05ed41b..b30d84fa40

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/5c31389b03..7344e69ded
* src/buildtools/linux64: git_revision:2941d2a26fa9cef3103f10e1dee5da7cd404008a..git_revision:d22fa2d126d8e4db585e9db0f3fb2fb4e0bf601b
* src/buildtools/mac: git_revision:2941d2a26fa9cef3103f10e1dee5da7cd404008a..git_revision:d22fa2d126d8e4db585e9db0f3fb2fb4e0bf601b
* src/buildtools/win: git_revision:2941d2a26fa9cef3103f10e1dee5da7cd404008a..git_revision:d22fa2d126d8e4db585e9db0f3fb2fb4e0bf601b
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/c1c27ffc20..904cd4f067
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/7953480e67..8527fd9f72
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/0bc2df1a46..793a1bc367
* src/third_party/android_deps/autorolled/cipd: Iv8_79hT6eno7q3HWp8otQ560vHfs_HhOBqWLLyLCscC..bSNUpmo1wIHCZI5w3gxpUEAdVSqFPcTlstxTARVxcuwC
* src/third_party/androidx/cipd: cOeVS8IOLGMV68S9vCCdb9hdKvQOTDppiUZ5rIYtIZoC..IxP1FmK52UK5we5ZE7jEkvED0_fBvkCslzXKDVIuV9QC
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/9bc4a2aee9..f7a1fb2d8f
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/4e58caf8aa..95d1432b5c
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/ea13a98dec..1726a92900
* src/third_party/fuzztest/src: https://chromium.googlesource.com/external/github.com/google/fuzztest.git/+log/41e160679e..3e42dc3c30
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/e14fa05100..c0728bdae7
* src/third_party/libaom/source/libaom: https://aomedia.googlesource.com/aom.git/+log/0c15af06af..34f25197c6
* src/third_party/libunwindstack: https://chromium.googlesource.com/chromium/src/third_party/libunwindstack.git/+log/0928ad0d25..333fcafb91
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/4e4843ae07..9b59f48e11
* src/third_party/tflite/src: https://chromium.googlesource.com/external/github.com/tensorflow/tensorflow.git/+log/d20369f4f2..24d66fbe6c
* src/third_party/xnnpack/src: https://chromium.googlesource.com/external/github.com/google/XNNPACK.git/+log/15e04ff8b6..ee91cc745b
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/fa4e6f5160..4dc3ed6779
DEPS diff: https://chromium.googlesource.com/chromium/src/+/19b05ed41b..b30d84fa40/DEPS

BUG=None

Change-Id: Ie35110a41ed0dd642a10adf8cfee7deed0c74edd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/457781
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47189}
---
 DEPS | 42 +++++++++++++++++++++---------------------
 1 file changed, 21 insertions(+), 21 deletions(-)

diff --git a/DEPS b/DEPS
index adbf56a926..bc998a8512 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '19b05ed41bafecd62c8aeb02854faba3fb5c3375',
+  'chromium_revision': 'b30d84fa40626ba910ed277984b3cc7737f49c2e',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@5c31389b03e59818d2ee155be40cbaf1798cbd20',
+    'https://chromium.googlesource.com/chromium/src/build@7344e69ded7ed29ef7134b9d4f54763245a43466',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@95ed44cf5f06dbb5861030b91c9db9ccb4316762',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,19 +78,19 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@c1c27ffc20d75442d2334a308fdaa797ae5d816b',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@904cd4f06715973dbd0f092eee0aa8e2c5d7667e',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@7953480e671e06cb123edc3e067675f07ea07974',
+    'https://chromium.googlesource.com/chromium/src/testing@8527fd9f72b14778901d9cac901736c97213344f',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@0bc2df1a46cbf66035d205ef1642dff693a7e7f1',
+    'https://chromium.googlesource.com/chromium/src/third_party@793a1bc3679b136c77dfffe8c74b09a31786b9c0',
 
   'src/buildtools/linux64': {
     'packages': [
       {
         'package': 'gn/gn/linux-${{arch}}',
-        'version': 'git_revision:2941d2a26fa9cef3103f10e1dee5da7cd404008a',
+        'version': 'git_revision:d22fa2d126d8e4db585e9db0f3fb2fb4e0bf601b',
       }
     ],
     'dep_type': 'cipd',
@@ -100,7 +100,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/mac-${{arch}}',
-        'version': 'git_revision:2941d2a26fa9cef3103f10e1dee5da7cd404008a',
+        'version': 'git_revision:d22fa2d126d8e4db585e9db0f3fb2fb4e0bf601b',
       }
     ],
     'dep_type': 'cipd',
@@ -110,7 +110,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/windows-amd64',
-        'version': 'git_revision:2941d2a26fa9cef3103f10e1dee5da7cd404008a',
+        'version': 'git_revision:d22fa2d126d8e4db585e9db0f3fb2fb4e0bf601b',
       }
     ],
     'dep_type': 'cipd',
@@ -492,7 +492,7 @@ deps = {
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@8be0e3114685fcc1589561067282edf75ea1259a',
   'src/third_party/catapult':
-    'https://chromium.googlesource.com/catapult.git@9bc4a2aee9361b6d2018c2f230c27c64a9e99b36',
+    'https://chromium.googlesource.com/catapult.git@f7a1fb2d8f3d155f6d52e95fa486bf91a9ebca81',
   'src/third_party/ced/src': {
     'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5',
   },
@@ -505,13 +505,13 @@ deps = {
   'src/third_party/crc32c/src':
     'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558',
   'src/third_party/depot_tools':
-    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@4e58caf8aa08befe78cf40d08ba33da072b15cf9',
+    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@95d1432b5c506cc6b500b34c23f06c4708139fe9',
   'src/third_party/ffmpeg':
     'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@946d97db8d906277085e361892b7efda5152e2f1',
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@e14fa05100494f45aab3838f09764c59e628b427',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@c0728bdae796f214a9fc56a617b21075c2ed3b40',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -593,7 +593,7 @@ deps = {
   'src/third_party/libFuzzer/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt/lib/fuzzer.git@bea408a6e01f0f7e6c82a43121fe3af4506c932e',
   'src/third_party/fuzztest/src':
-    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@41e160679e3c010a860e674e7997e2b285c6e0a2',
+    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@3e42dc3c305dd22a22e7d2996c7e24177f3a43ad',
   'src/third_party/libjpeg_turbo':
     'https://chromium.googlesource.com/chromium/deps/libjpeg_turbo.git@d1f5f2393e0d51f840207342ae86e55a86443288',
   'src/third_party/libsrtp':
@@ -601,15 +601,15 @@ deps = {
   'src/third_party/dav1d/libdav1d':
     'https://chromium.googlesource.com/external/github.com/videolan/dav1d.git@241a6b236a9c84441c872119c1f31b4074c0589d',
   'src/third_party/libaom/source/libaom':
-    'https://aomedia.googlesource.com/aom.git@0c15af06af108af03cbe237e4bffa4b700b76d81',
+    'https://aomedia.googlesource.com/aom.git@34f25197c6e9efef129992bb763aefa6036119fd',
   'src/third_party/libgav1/src':
     Var('chromium_git') + '/codecs/libgav1.git' + '@' + '40f58ed32ff39071c3f2a51056dbc49a070af0dc',
   'src/third_party/libunwindstack': {
-      'url': 'https://chromium.googlesource.com/chromium/src/third_party/libunwindstack.git@0928ad0d25e4af07c8be5ab06d0ca584f9f4ceb5',
+      'url': 'https://chromium.googlesource.com/chromium/src/third_party/libunwindstack.git@333fcafb91bd3830c5ef814c071ff73df9cdc976',
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '4e4843ae073504d2f58540031a8f41471fa86a76',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '9b59f48e11a56a8d2dddd60299e7e25b68b3072c',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@fa4e6f516009593a81741f98cbc417ae251a5a07',
+    'https://chromium.googlesource.com/chromium/src/tools@4dc3ed6779e386922affe81b6b933cc534f2fc32',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'cOeVS8IOLGMV68S9vCCdb9hdKvQOTDppiUZ5rIYtIZoC',
+          'version': 'IxP1FmK52UK5we5ZE7jEkvED0_fBvkCslzXKDVIuV9QC',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -786,7 +786,7 @@ deps = {
   },
 
   'src/third_party/tflite/src':
-    Var('chromium_git') + '/external/github.com/tensorflow/tensorflow.git' + '@' + 'd20369f4f226598800e0d406db567d2d1c17fa95',
+    Var('chromium_git') + '/external/github.com/tensorflow/tensorflow.git' + '@' + '24d66fbe6c5e87d291207494e4e83d39de3f7d90',
 
   'src/third_party/turbine/cipd': {
       'packages': [
@@ -842,7 +842,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/android_deps/autorolled',
-              'version': 'Iv8_79hT6eno7q3HWp8otQ560vHfs_HhOBqWLLyLCscC',
+              'version': 'bSNUpmo1wIHCZI5w3gxpUEAdVSqFPcTlstxTARVxcuwC',
           },
       ],
       'condition': 'checkout_android and non_git_source',
@@ -852,7 +852,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/google/pthreadpool.git' + '@' + '9003ee6c137cea3b94161bd5c614fb43be523ee1',
 
   'src/third_party/xnnpack/src':
-    Var('chromium_git') + '/external/github.com/google/XNNPACK.git' + '@' + '15e04ff8b61e0bc2118bad268447be5b2075aa66',
+    Var('chromium_git') + '/external/github.com/google/XNNPACK.git' + '@' + 'ee91cc745bc715bfa38c5e8241aeb435ca59f433',
 
   'src/third_party/farmhash/src':
     Var('chromium_git') + '/external/github.com/google/farmhash.git' + '@' + '816a4ae622e964763ca0862d9dbd19324a1eaf45',
@@ -864,7 +864,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7607ca500436b37ad23fb8d18614bec7796b68a7',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + 'ea13a98decd497a8c5588fb5de71b57bcf10d864',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '1726a929009be60a5ed049d23052cbb7ce6f2f96',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From 4c8479b3025805a32eaba43a99364da7eccc7a1b Mon Sep 17 00:00:00 2001
From: Philipp Hancke <philipp.hancke@googlemail.com>
Date: Tue, 17 Mar 2026 12:54:10 +0100
Subject: [PATCH 337/472] IWYU: regenerate compile_commands.json on every run

which seems to avoid the
  unable to find plugin 'unsafe-buffers'
error.

Bug: webrtc:485150285
Change-Id: I51c563072e05a6b8d694de9fe96cb81279e738f1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/457582
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Christoffer Dewerin <jansson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47190}
---
 tools_webrtc/apply_clang_tidy.py           |  2 +-
 tools_webrtc/iwyu/apply_include_cleaner.py | 15 +++++++--------
 2 files changed, 8 insertions(+), 9 deletions(-)

diff --git a/tools_webrtc/apply_clang_tidy.py b/tools_webrtc/apply_clang_tidy.py
index c61b6e75ca..b7a4065a35 100755
--- a/tools_webrtc/apply_clang_tidy.py
+++ b/tools_webrtc/apply_clang_tidy.py
@@ -92,7 +92,7 @@ def _generate_compile_commands(work_dir: pathlib.Path) -> None:
     by the include cleaner binary.
 
     Args:
-        work_dir: gn out dir where the compile_commands json file exists
+        work_dir: gn out dir
     """
     compile_commands_path = work_dir / "compile_commands.json"
     print("Generating compile commands file...")
diff --git a/tools_webrtc/iwyu/apply_include_cleaner.py b/tools_webrtc/iwyu/apply_include_cleaner.py
index 5f2470a6d3..488e2cb6f7 100755
--- a/tools_webrtc/iwyu/apply_include_cleaner.py
+++ b/tools_webrtc/iwyu/apply_include_cleaner.py
@@ -183,16 +183,15 @@ def _generate_compile_commands(work_dir: pathlib.Path) -> None:
     by the include cleaner binary.
 
     Args:
-        work_dir: gn out dir where the compile_commands json file exists
+        work_dir: gn out dir
     """
     compile_commands_path = work_dir / "compile_commands.json"
-    if not compile_commands_path.is_file():
-        print("Generating compile commands file...")
-        subprocess.run(
-            ["tools/clang/scripts/generate_compdb.py", "-p", work_dir],
-            stdout=compile_commands_path.open(mode="w+"),
-            check=True,
-        )
+    print("Generating compile commands file...")
+    subprocess.run(
+        ["tools/clang/scripts/generate_compdb.py", "-p", work_dir],
+        stdout=compile_commands_path.open(mode="w+"),
+        check=True,
+    )
 
 
 def _modified_output(output: str, content: str) -> str:

From 32d49244ef411093e53664338075d65a2c376f34 Mon Sep 17 00:00:00 2001
From: Evan Shrubsole <eshr@webrtc.org>
Date: Tue, 17 Mar 2026 20:05:02 +0000
Subject: [PATCH 338/472] Use GlobalSimulatedTimecontroller in
 turn_port_unittest

Removes uses of SIMULATED_WAIT and AutoThread.

Bug: webrtc:469327588,webrtc:42223992,webrtc:381524905
Change-Id: I68166aeb65758ddbb92c405113cf8dac6a6a6964
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/457740
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47191}
---
 p2p/BUILD.gn                   |   1 +
 p2p/base/turn_port_unittest.cc | 531 ++++++++++++++++-----------------
 2 files changed, 251 insertions(+), 281 deletions(-)

diff --git a/p2p/BUILD.gn b/p2p/BUILD.gn
index 0c97fa00a1..d59a176b96 100644
--- a/p2p/BUILD.gn
+++ b/p2p/BUILD.gn
@@ -1172,6 +1172,7 @@ if (rtc_include_tests) {
       "../api:scoped_refptr",
       "../api:sequence_checker",
       "../api:simulated_network_api",
+      "../api:time_controller",
       "../api/crypto:options",
       "../api/environment",
       "../api/environment:environment_factory",
diff --git a/p2p/base/turn_port_unittest.cc b/p2p/base/turn_port_unittest.cc
index a5d7fd7e44..6823fbd1e2 100644
--- a/p2p/base/turn_port_unittest.cc
+++ b/p2p/base/turn_port_unittest.cc
@@ -23,7 +23,6 @@
 #include "absl/strings/string_view.h"
 #include "api/candidate.h"
 #include "api/environment/environment.h"
-#include "api/environment/environment_factory.h"
 #include "api/packet_socket_factory.h"
 #include "api/test/mock_async_dns_resolver.h"
 #include "api/test/rtc_error_matchers.h"
@@ -50,8 +49,6 @@
 #include "rtc_base/buffer.h"
 #include "rtc_base/byte_buffer.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/fake_clock.h"
-#include "rtc_base/gunit.h"
 #include "rtc_base/ip_address.h"
 #include "rtc_base/net_helper.h"
 #include "rtc_base/net_helpers.h"
@@ -62,8 +59,10 @@
 #include "rtc_base/thread.h"
 #include "rtc_base/virtual_socket_server.h"
 #include "system_wrappers/include/metrics.h"
+#include "test/create_test_environment.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/time_controller/simulated_time_controller.h"
 #include "test/wait_until.h"
 
 #if defined(WEBRTC_POSIX)
@@ -116,13 +115,13 @@ constexpr char kTurnUsername[] = "test";
 constexpr char kTurnPassword[] = "test";
 // This test configures the virtual socket server to simulate delay so that we
 // can verify operations take no more than the expected number of round trips.
-constexpr unsigned int kSimulatedRtt = 50;
+constexpr TimeDelta kSimulatedRtt = TimeDelta::Millis(50);
 // Connection destruction may happen asynchronously, but it should only
 // take one simulated clock tick.
-constexpr unsigned int kConnectionDestructionDelay = 1;
+constexpr TimeDelta kConnectionDestructionDelay = TimeDelta::Millis(1);
 // This used to be 1 second, but that's not always enough for getaddrinfo().
 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5191
-constexpr unsigned int kResolverTimeout = 10000;
+constexpr TimeDelta kResolverTimeout = TimeDelta::Millis(10000);
 
 constexpr uint64_t kTiebreakerDefault = 44444;
 
@@ -156,13 +155,12 @@ int GetFDCount() {
 
 }  // unnamed namespace
 
-
 class TurnPortTestVirtualSocketServer : public VirtualSocketServer {
  public:
   TurnPortTestVirtualSocketServer() {
     // This configures the virtual socket server to always add a simulated
     // delay of exactly half of kSimulatedRtt.
-    set_delay_mean(kSimulatedRtt / 2);
+    set_delay_mean((kSimulatedRtt / 2).ms());
     UpdateDelayDistribution();
   }
 
@@ -200,14 +198,11 @@ class TurnPortTest : public ::testing::Test, public TurnPort::CallbacksForTest {
  public:
   TurnPortTest()
       : ss_(new TurnPortTestVirtualSocketServer()),
-        main_(ss_.get()),
-        turn_server_(env_, &main_, ss_.get(), kTurnUdpIntAddr, kTurnUdpExtAddr),
-        socket_factory_(ss_.get()) {
-    // Some code uses "last received time == 0" to represent "nothing received
-    // so far", so we need to start the fake clock at a nonzero time...
-    // TODO(deadbeef): Fix this.
-    fake_clock_.AdvanceTime(TimeDelta::Seconds(1));
-  }
+        time_controller_(Timestamp::Seconds(1), ss_.get()),
+        env_(CreateTestEnvironment({.time = &time_controller_})),
+        main_(time_controller_.GetMainThread()),
+        turn_server_(env_, main_, ss_.get(), kTurnUdpIntAddr, kTurnUdpExtAddr),
+        socket_factory_(ss_.get()) {}
 
   void OnTurnPortComplete(Port* port) { turn_ready_ = true; }
   void OnTurnPortError(Port* port) { turn_error_ = true; }
@@ -287,7 +282,7 @@ class TurnPortTest : public ::testing::Test, public TurnPort::CallbacksForTest {
     RelayServerConfig config;
     config.credentials = RelayCredentials(username, password);
     CreateRelayPortArgs args = {.env = env_};
-    args.network_thread = &main_;
+    args.network_thread = main_;
     args.socket_factory = socket_factory();
     args.network = network;
     args.username = kIceUfrag1;
@@ -334,7 +329,7 @@ class TurnPortTest : public ::testing::Test, public TurnPort::CallbacksForTest {
     RelayServerConfig config;
     config.credentials = RelayCredentials(username, password);
     CreateRelayPortArgs args = {.env = env_};
-    args.network_thread = &main_;
+    args.network_thread = main_;
     args.socket_factory = socket_factory();
     args.network = MakeNetwork(kLocalAddr1);
     args.username = kIceUfrag1;
@@ -373,7 +368,7 @@ class TurnPortTest : public ::testing::Test, public TurnPort::CallbacksForTest {
 
   void CreateUdpPort(const SocketAddress& address) {
     udp_port_ = UDPPort::Create({.env = env_,
-                                 .network_thread = &main_,
+                                 .network_thread = main_,
                                  .socket_factory = socket_factory(),
                                  .network = MakeNetwork(address),
                                  .ice_username_fragment = kIceUfrag2,
@@ -392,22 +387,21 @@ class TurnPortTest : public ::testing::Test, public TurnPort::CallbacksForTest {
     ASSERT_TRUE(turn_port_ != nullptr);
     turn_port_->PrepareAddress();
     ASSERT_THAT(WaitUntil([&] { return turn_ready_; }, IsTrue(),
-                          {.timeout = TimeDelta::Millis(
-                               TimeToGetTurnCandidate(protocol_type)),
-                           .clock = &fake_clock_}),
+                          {.timeout = TimeToGetTurnCandidate(protocol_type),
+                           .clock = &time_controller_}),
                 IsRtcOk());
 
     CreateUdpPort();
     udp_port_->PrepareAddress();
-    ASSERT_THAT(WaitUntil([&] { return udp_ready_; }, IsTrue(),
-                          {.timeout = TimeDelta::Millis(kSimulatedRtt),
-                           .clock = &fake_clock_}),
-                IsRtcOk());
+    ASSERT_THAT(
+        WaitUntil([&] { return udp_ready_; }, IsTrue(),
+                  {.timeout = kSimulatedRtt, .clock = &time_controller_}),
+        IsRtcOk());
   }
 
   // Returns the fake clock time to establish a connection over the given
   // protocol.
-  int TimeToConnect(ProtocolType protocol_type) {
+  TimeDelta TimeToConnect(ProtocolType protocol_type) {
     switch (protocol_type) {
       case PROTO_TCP:
         // The virtual socket server will delay by a fixed half a round trip
@@ -420,13 +414,13 @@ class TurnPortTest : public ::testing::Test, public TurnPort::CallbacksForTest {
       case PROTO_UDP:
       default:
         // UDP requires no round trips to set up the connection.
-        return 0;
+        return TimeDelta::Zero();
     }
   }
 
   // Returns the total fake clock time to establish a connection with a TURN
   // server over the given protocol and to allocate a TURN candidate.
-  int TimeToGetTurnCandidate(ProtocolType protocol_type) {
+  TimeDelta TimeToGetTurnCandidate(ProtocolType protocol_type) {
     // For a simple allocation, the first Allocate message will return with an
     // error asking for credentials and will succeed after the second Allocate
     // message.
@@ -439,7 +433,7 @@ class TurnPortTest : public ::testing::Test, public TurnPort::CallbacksForTest {
   // 3. Connect to alternate TURN server
   // 4. Send Allocate and receive a request for credentials
   // 5. Send Allocate with credentials and receive allocation
-  int TimeToGetAlternateTurnCandidate(ProtocolType protocol_type) {
+  TimeDelta TimeToGetAlternateTurnCandidate(ProtocolType protocol_type) {
     return 3 * kSimulatedRtt + 2 * TimeToConnect(protocol_type);
   }
 
@@ -463,12 +457,11 @@ class TurnPortTest : public ::testing::Test, public TurnPort::CallbacksForTest {
     return true;
   }
 
-  void TestTurnAllocateSucceeds(unsigned int timeout) {
+  void TestTurnAllocateSucceeds(TimeDelta timeout) {
     ASSERT_TRUE(turn_port_);
     turn_port_->PrepareAddress();
     EXPECT_THAT(WaitUntil([&] { return turn_ready_; }, IsTrue(),
-                          {.timeout = TimeDelta::Millis(timeout),
-                           .clock = &fake_clock_}),
+                          {.timeout = timeout, .clock = &time_controller_}),
                 IsRtcOk());
     ASSERT_EQ(1U, turn_port_->Candidates().size());
     EXPECT_EQ(kTurnUdpExtAddr.ipaddr(),
@@ -481,9 +474,8 @@ class TurnPortTest : public ::testing::Test, public TurnPort::CallbacksForTest {
     ASSERT_TRUE(turn_port_);
     turn_port_->PrepareAddress();
     ASSERT_THAT(WaitUntil([&] { return turn_ready_; }, IsTrue(),
-                          {.timeout = TimeDelta::Millis(
-                               TimeToGetTurnCandidate(protocol_type)),
-                           .clock = &fake_clock_}),
+                          {.timeout = TimeToGetTurnCandidate(protocol_type),
+                           .clock = &time_controller_}),
                 IsRtcOk());
     ASSERT_EQ(1U, turn_port_->Candidates().size());
     EXPECT_EQ(turn_port_->Candidates()[0].url(), expected_url);
@@ -505,11 +497,11 @@ class TurnPortTest : public ::testing::Test, public TurnPort::CallbacksForTest {
     const SocketAddress old_addr = turn_port_->server_address().address;
 
     turn_port_->PrepareAddress();
-    EXPECT_THAT(WaitUntil([&] { return turn_ready_; }, IsTrue(),
-                          {.timeout = TimeDelta::Millis(
-                               TimeToGetAlternateTurnCandidate(protocol_type)),
-                           .clock = &fake_clock_}),
-                IsRtcOk());
+    EXPECT_THAT(
+        WaitUntil([&] { return turn_ready_; }, IsTrue(),
+                  {.timeout = TimeToGetAlternateTurnCandidate(protocol_type),
+                   .clock = &time_controller_}),
+        IsRtcOk());
     // Retrieve the address again, the turn port's address should be
     // changed.
     const SocketAddress new_addr = turn_port_->server_address().address;
@@ -532,11 +524,11 @@ class TurnPortTest : public ::testing::Test, public TurnPort::CallbacksForTest {
     turn_port_->PrepareAddress();
     // Need time to connect to TURN server, send Allocate request and receive
     // redirect notice.
-    EXPECT_THAT(WaitUntil([&] { return turn_error_; }, IsTrue(),
-                          {.timeout = TimeDelta::Millis(
-                               kSimulatedRtt + TimeToConnect(protocol_type)),
-                           .clock = &fake_clock_}),
-                IsRtcOk());
+    EXPECT_THAT(
+        WaitUntil([&] { return turn_error_; }, IsTrue(),
+                  {.timeout = kSimulatedRtt + TimeToConnect(protocol_type),
+                   .clock = &time_controller_}),
+        IsRtcOk());
   }
 
   void TestTurnAlternateServerPingPong(ProtocolType protocol_type) {
@@ -553,11 +545,11 @@ class TurnPortTest : public ::testing::Test, public TurnPort::CallbacksForTest {
                    ProtocolAddress(kTurnIntAddr, protocol_type));
 
     turn_port_->PrepareAddress();
-    EXPECT_THAT(WaitUntil([&] { return turn_error_; }, IsTrue(),
-                          {.timeout = TimeDelta::Millis(
-                               TimeToGetAlternateTurnCandidate(protocol_type)),
-                           .clock = &fake_clock_}),
-                IsRtcOk());
+    EXPECT_THAT(
+        WaitUntil([&] { return turn_error_; }, IsTrue(),
+                  {.timeout = TimeToGetAlternateTurnCandidate(protocol_type),
+                   .clock = &time_controller_}),
+        IsRtcOk());
     ASSERT_EQ(0U, turn_port_->Candidates().size());
     SocketAddress address;
     // Verify that we have exhausted all alternate servers instead of
@@ -579,11 +571,11 @@ class TurnPortTest : public ::testing::Test, public TurnPort::CallbacksForTest {
                    ProtocolAddress(kTurnIntAddr, protocol_type));
 
     turn_port_->PrepareAddress();
-    EXPECT_THAT(WaitUntil([&] { return turn_error_; }, IsTrue(),
-                          {.timeout = TimeDelta::Millis(
-                               TimeToGetAlternateTurnCandidate(protocol_type)),
-                           .clock = &fake_clock_}),
-                IsRtcOk());
+    EXPECT_THAT(
+        WaitUntil([&] { return turn_error_; }, IsTrue(),
+                  {.timeout = TimeToGetAlternateTurnCandidate(protocol_type),
+                   .clock = &time_controller_}),
+        IsRtcOk());
     ASSERT_EQ(0U, turn_port_->Candidates().size());
   }
 
@@ -622,15 +614,14 @@ class TurnPortTest : public ::testing::Test, public TurnPort::CallbacksForTest {
 
     turn_port_->PrepareAddress();
     EXPECT_THAT(WaitUntil([&] { return turn_error_; }, IsTrue(),
-                          {.timeout = TimeDelta::Millis(
-                               TimeToGetTurnCandidate(protocol_type)),
-                           .clock = &fake_clock_}),
+                          {.timeout = TimeToGetTurnCandidate(protocol_type),
+                           .clock = &time_controller_}),
                 IsRtcOk());
 
     // Wait for some extra time, and make sure no packets were received on the
     // loopback port we created (or in the case of TCP, no connection attempt
     // occurred).
-    SIMULATED_WAIT(false, kSimulatedRtt, fake_clock_);
+    time_controller_.AdvanceTime(kSimulatedRtt);
     if (protocol_type == PROTO_UDP) {
       char buf[1];
       EXPECT_EQ(-1, loopback_socket->Recv(&buf, 1, nullptr));
@@ -650,7 +641,9 @@ class TurnPortTest : public ::testing::Test, public TurnPort::CallbacksForTest {
                                                     Port::ORIGIN_MESSAGE);
     ASSERT_TRUE(conn1 != nullptr);
     conn1->Ping();
-    SIMULATED_WAIT(!turn_unknown_address_, kSimulatedRtt * 2, fake_clock_);
+    EXPECT_TRUE(
+        WaitUntil([&] { return !turn_unknown_address_; },
+                  {.timeout = kSimulatedRtt * 2, .clock = &time_controller_}));
     EXPECT_FALSE(turn_unknown_address_);
     EXPECT_FALSE(conn1->receiving());
     EXPECT_EQ(Connection::STATE_WRITE_INIT, conn1->write_state());
@@ -661,28 +654,27 @@ class TurnPortTest : public ::testing::Test, public TurnPort::CallbacksForTest {
     ASSERT_TRUE(conn2 != nullptr);
     ASSERT_THAT(
         WaitUntil([&] { return turn_create_permission_success_; }, IsTrue(),
-                  {.timeout = TimeDelta::Millis(kSimulatedRtt),
-                   .clock = &fake_clock_}),
+                  {.timeout = kSimulatedRtt, .clock = &time_controller_}),
         IsRtcOk());
     conn2->Ping();
 
     // Two hops from TURN port to UDP port through TURN server, thus two RTTs.
-    EXPECT_THAT(WaitUntil([&] { return conn2->write_state(); },
-                          Eq(Connection::STATE_WRITABLE),
-                          {.timeout = TimeDelta::Millis(kSimulatedRtt * 2),
-                           .clock = &fake_clock_}),
-                IsRtcOk());
+    EXPECT_THAT(
+        WaitUntil([&] { return conn2->write_state(); },
+                  Eq(Connection::STATE_WRITABLE),
+                  {.timeout = kSimulatedRtt * 2, .clock = &time_controller_}),
+        IsRtcOk());
     EXPECT_TRUE(conn1->receiving());
     EXPECT_TRUE(conn2->receiving());
     EXPECT_EQ(Connection::STATE_WRITE_INIT, conn1->write_state());
 
     // Send another ping from UDP to TURN.
     conn1->Ping();
-    EXPECT_THAT(WaitUntil([&] { return conn1->write_state(); },
-                          Eq(Connection::STATE_WRITABLE),
-                          {.timeout = TimeDelta::Millis(kSimulatedRtt * 2),
-                           .clock = &fake_clock_}),
-                IsRtcOk());
+    EXPECT_THAT(
+        WaitUntil([&] { return conn1->write_state(); },
+                  Eq(Connection::STATE_WRITABLE),
+                  {.timeout = kSimulatedRtt * 2, .clock = &time_controller_}),
+        IsRtcOk());
     EXPECT_TRUE(conn2->receiving());
   }
 
@@ -702,16 +694,15 @@ class TurnPortTest : public ::testing::Test, public TurnPort::CallbacksForTest {
     ASSERT_TRUE(conn2 != nullptr);
     ASSERT_THAT(
         WaitUntil([&] { return turn_create_permission_success_; }, IsTrue(),
-                  {.timeout = TimeDelta::Millis(kSimulatedRtt),
-                   .clock = &fake_clock_}),
+                  {.timeout = kSimulatedRtt, .clock = &time_controller_}),
         IsRtcOk());
     // Make sure turn connection can receive.
     conn1->Ping();
-    EXPECT_THAT(WaitUntil([&] { return conn1->write_state(); },
-                          Eq(Connection::STATE_WRITABLE),
-                          {.timeout = TimeDelta::Millis(kSimulatedRtt * 2),
-                           .clock = &fake_clock_}),
-                IsRtcOk());
+    EXPECT_THAT(
+        WaitUntil([&] { return conn1->write_state(); },
+                  Eq(Connection::STATE_WRITABLE),
+                  {.timeout = kSimulatedRtt * 2, .clock = &time_controller_}),
+        IsRtcOk());
     EXPECT_FALSE(turn_unknown_address_);
 
     // Destroy the connection on the TURN port. The TurnEntry still exists, so
@@ -719,16 +710,16 @@ class TurnPortTest : public ::testing::Test, public TurnPort::CallbacksForTest {
     turn_port_->DestroyConnection(conn2);
 
     conn1->Ping();
-    EXPECT_THAT(WaitUntil([&] { return turn_unknown_address_; }, IsTrue(),
-                          {.timeout = TimeDelta::Millis(kSimulatedRtt),
-                           .clock = &fake_clock_}),
-                IsRtcOk());
+    EXPECT_THAT(
+        WaitUntil([&] { return turn_unknown_address_; }, IsTrue(),
+                  {.timeout = kSimulatedRtt, .clock = &time_controller_}),
+        IsRtcOk());
 
     // Wait for TurnEntry to expire. Timeout is 5 minutes.
     // Expect that it still processes an incoming ping and signals the
     // unknown address.
     turn_unknown_address_ = false;
-    fake_clock_.AdvanceTime(TimeDelta::Seconds(5 * 60));
+    time_controller_.AdvanceTime(TimeDelta::Seconds(5 * 60));
 
     // TODO(chromium:1395625): When `TurnPort` doesn't find connection objects
     // for incoming packets, it forwards calls to the parent class, `Port`. This
@@ -754,19 +745,19 @@ class TurnPortTest : public ::testing::Test, public TurnPort::CallbacksForTest {
     conn1->set_remote_password_for_test(pwd);
 
     conn1->Ping();
-    EXPECT_THAT(WaitUntil([&] { return turn_unknown_address_; }, IsTrue(),
-                          {.timeout = TimeDelta::Millis(kSimulatedRtt),
-                           .clock = &fake_clock_}),
-                IsRtcOk());
+    EXPECT_THAT(
+        WaitUntil([&] { return turn_unknown_address_; }, IsTrue(),
+                  {.timeout = kSimulatedRtt, .clock = &time_controller_}),
+        IsRtcOk());
 
     // If the connection is created again, it will start to receive pings.
     conn2 = turn_port_->CreateConnection(udp_port_->Candidates()[0],
                                          Port::ORIGIN_MESSAGE);
     conn1->Ping();
-    EXPECT_THAT(WaitUntil([&] { return conn2->receiving(); }, IsTrue(),
-                          {.timeout = TimeDelta::Millis(kSimulatedRtt),
-                           .clock = &fake_clock_}),
-                IsRtcOk());
+    EXPECT_THAT(
+        WaitUntil([&] { return conn2->receiving(); }, IsTrue(),
+                  {.timeout = kSimulatedRtt, .clock = &time_controller_}),
+        IsRtcOk());
   }
 
   void TestTurnSendData(ProtocolType protocol_type, bool expect_ecn_propagate) {
@@ -795,17 +786,17 @@ class TurnPortTest : public ::testing::Test, public TurnPort::CallbacksForTest {
       OnConnectionSignalDestroyed(connection);
     });
     conn1->Ping();
-    EXPECT_THAT(WaitUntil([&] { return conn1->write_state(); },
-                          Eq(Connection::STATE_WRITABLE),
-                          {.timeout = TimeDelta::Millis(kSimulatedRtt * 2),
-                           .clock = &fake_clock_}),
-                IsRtcOk());
+    EXPECT_THAT(
+        WaitUntil([&] { return conn1->write_state(); },
+                  Eq(Connection::STATE_WRITABLE),
+                  {.timeout = kSimulatedRtt * 2, .clock = &time_controller_}),
+        IsRtcOk());
     conn2->Ping();
-    EXPECT_THAT(WaitUntil([&] { return conn2->write_state(); },
-                          Eq(Connection::STATE_WRITABLE),
-                          {.timeout = TimeDelta::Millis(kSimulatedRtt * 2),
-                           .clock = &fake_clock_}),
-                IsRtcOk());
+    EXPECT_THAT(
+        WaitUntil([&] { return conn2->write_state(); },
+                  Eq(Connection::STATE_WRITABLE),
+                  {.timeout = kSimulatedRtt * 2, .clock = &time_controller_}),
+        IsRtcOk());
 
     // Send some data.
     size_t num_packets = 256;
@@ -817,7 +808,7 @@ class TurnPortTest : public ::testing::Test, public TurnPort::CallbacksForTest {
       options.ect_1 = (i % 2 == 0);
       conn1->Send(buf, i + 1, options);
       conn2->Send(buf, i + 1, options);
-      SIMULATED_WAIT(false, kSimulatedRtt, fake_clock_);
+      time_controller_.AdvanceTime(kSimulatedRtt);
     }
 
     // Check the data.
@@ -845,8 +836,7 @@ class TurnPortTest : public ::testing::Test, public TurnPort::CallbacksForTest {
     EXPECT_THAT(
         WaitUntil([&] { return turn_server_.server()->allocations().size(); },
                   Eq(0U),
-                  {.timeout = TimeDelta::Millis(kSimulatedRtt),
-                   .clock = &fake_clock_}),
+                  {.timeout = kSimulatedRtt, .clock = &time_controller_}),
         IsRtcOk());
   }
 
@@ -875,17 +865,17 @@ class TurnPortTest : public ::testing::Test, public TurnPort::CallbacksForTest {
     });
 
     conn1->Ping();
-    EXPECT_THAT(WaitUntil([&] { return conn1->write_state(); },
-                          Eq(Connection::STATE_WRITABLE),
-                          {.timeout = TimeDelta::Millis(kSimulatedRtt * 2),
-                           .clock = &fake_clock_}),
-                IsRtcOk());
+    EXPECT_THAT(
+        WaitUntil([&] { return conn1->write_state(); },
+                  Eq(Connection::STATE_WRITABLE),
+                  {.timeout = kSimulatedRtt * 2, .clock = &time_controller_}),
+        IsRtcOk());
     conn2->Ping();
-    EXPECT_THAT(WaitUntil([&] { return conn2->write_state(); },
-                          Eq(Connection::STATE_WRITABLE),
-                          {.timeout = TimeDelta::Millis(kSimulatedRtt * 2),
-                           .clock = &fake_clock_}),
-                IsRtcOk());
+    EXPECT_THAT(
+        WaitUntil([&] { return conn2->write_state(); },
+                  Eq(Connection::STATE_WRITABLE),
+                  {.timeout = kSimulatedRtt * 2, .clock = &time_controller_}),
+        IsRtcOk());
 
     // Send some data from Udp to TurnPort.
     unsigned char buf[256] = {0};
@@ -896,10 +886,10 @@ class TurnPortTest : public ::testing::Test, public TurnPort::CallbacksForTest {
     turn_port_->Release();
 
     // Wait for the TurnPort to signal closed.
-    ASSERT_THAT(WaitUntil([&] { return turn_port_closed_; }, IsTrue(),
-                          {.timeout = TimeDelta::Millis(kSimulatedRtt),
-                           .clock = &fake_clock_}),
-                IsRtcOk());
+    ASSERT_THAT(
+        WaitUntil([&] { return turn_port_closed_; }, IsTrue(),
+                  {.timeout = kSimulatedRtt, .clock = &time_controller_}),
+        IsRtcOk());
 
     // But the data should have arrived first.
     ASSERT_EQ(1ul, turn_packets_.size());
@@ -923,14 +913,11 @@ class TurnPortTest : public ::testing::Test, public TurnPort::CallbacksForTest {
 
   virtual PacketSocketFactory* socket_factory() { return &socket_factory_; }
 
-  ScopedFakeClock fake_clock_;
-  const Environment env_ = CreateEnvironment();
-  // When a "create port" helper method is called with an IP, we create a
-  // Network with that IP and add it to this list. Using a list instead of a
-  // vector so that when it grows, pointers aren't invalidated.
   std::list<Network> networks_;
   std::unique_ptr<TurnPortTestVirtualSocketServer> ss_;
-  AutoSocketServerThread main_;
+  GlobalSimulatedTimeController time_controller_;
+  const Environment env_;
+  webrtc::Thread* main_;
   std::unique_ptr<AsyncPacketSocket> socket_;
   TestTurnServer turn_server_;
   std::unique_ptr<TurnPort> turn_port_;
@@ -996,7 +983,7 @@ TEST_F(TurnPortTest, TestReconstructedServerUrlForHostname) {
   // the error will be set and contain the url.
   turn_port_->PrepareAddress();
   EXPECT_THAT(WaitUntil([&] { return turn_error_; }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(kResolverTimeout)}),
+                        {.timeout = kResolverTimeout}),
               IsRtcOk());
   std::string server_url =
       "turn:" + kTurnInvalidAddr.ToString() + "?transport=udp";
@@ -1052,16 +1039,16 @@ TEST_F(TurnPortTest, TestTurnAllocateWithoutLoggingId) {
 TEST_F(TurnPortTest, TestTurnBadCredentials) {
   CreateTurnPort(kTurnUsername, "bad", kTurnUdpProtoAddr);
   turn_port_->PrepareAddress();
-  EXPECT_THAT(WaitUntil([&] { return turn_error_; }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(kSimulatedRtt * 3),
-                         .clock = &fake_clock_}),
-              IsRtcOk());
+  EXPECT_THAT(
+      WaitUntil([&] { return turn_error_; }, IsTrue(),
+                {.timeout = kSimulatedRtt * 3, .clock = &time_controller_}),
+      IsRtcOk());
   ASSERT_EQ(0U, turn_port_->Candidates().size());
-  EXPECT_THAT(WaitUntil([&] { return error_event_.error_code; },
-                        Eq(STUN_ERROR_UNAUTHORIZED),
-                        {.timeout = TimeDelta::Millis(kSimulatedRtt * 3),
-                         .clock = &fake_clock_}),
-              IsRtcOk());
+  EXPECT_THAT(
+      WaitUntil([&] { return error_event_.error_code; },
+                Eq(STUN_ERROR_UNAUTHORIZED),
+                {.timeout = kSimulatedRtt * 3, .clock = &time_controller_}),
+      IsRtcOk());
   EXPECT_EQ(error_event_.error_text, "Unauthorized.");
 }
 
@@ -1070,10 +1057,10 @@ TEST_F(TurnPortTest, TestTurnBadCredentials) {
 TEST_F(TurnPortTest, TestServerAddressFamilyMismatch) {
   CreateTurnPort(kTurnUsername, kTurnPassword, kTurnUdpIPv6ProtoAddr);
   turn_port_->PrepareAddress();
-  EXPECT_THAT(WaitUntil([&] { return turn_error_; }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(kSimulatedRtt * 3),
-                         .clock = &fake_clock_}),
-              IsRtcOk());
+  EXPECT_THAT(
+      WaitUntil([&] { return turn_error_; }, IsTrue(),
+                {.timeout = kSimulatedRtt * 3, .clock = &time_controller_}),
+      IsRtcOk());
   ASSERT_EQ(0U, turn_port_->Candidates().size());
   EXPECT_EQ(0, error_event_.error_code);
 }
@@ -1084,10 +1071,10 @@ TEST_F(TurnPortTest, TestServerAddressFamilyMismatch6) {
   CreateTurnPort(kLocalIPv6Addr, kTurnUsername, kTurnPassword,
                  kTurnUdpProtoAddr);
   turn_port_->PrepareAddress();
-  EXPECT_THAT(WaitUntil([&] { return turn_error_; }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(kSimulatedRtt * 3),
-                         .clock = &fake_clock_}),
-              IsRtcOk());
+  EXPECT_THAT(
+      WaitUntil([&] { return turn_error_; }, IsTrue(),
+                {.timeout = kSimulatedRtt * 3, .clock = &time_controller_}),
+      IsRtcOk());
   ASSERT_EQ(0U, turn_port_->Candidates().size());
   EXPECT_EQ(0, error_event_.error_code);
 }
@@ -1141,13 +1128,11 @@ TEST_F(TurnPortTest,
   // Shouldn't take more than 1 RTT to realize the bound address isn't the one
   // expected.
   EXPECT_THAT(WaitUntil([&] { return turn_error_; }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(kSimulatedRtt),
-                         .clock = &fake_clock_}),
+                        {.timeout = kSimulatedRtt, .clock = &time_controller_}),
               IsRtcOk());
   EXPECT_THAT(WaitUntil([&] { return error_event_.error_code; },
                         Eq(STUN_ERROR_SERVER_NOT_REACHABLE),
-                        {.timeout = TimeDelta::Millis(kSimulatedRtt),
-                         .clock = &fake_clock_}),
+                        {.timeout = kSimulatedRtt, .clock = &time_controller_}),
               IsRtcOk());
   EXPECT_NE(error_event_.error_text.find('.'), std::string::npos);
   EXPECT_NE(error_event_.address.find(kLocalAddr2.HostAsSensitiveURIString()),
@@ -1180,10 +1165,10 @@ TEST_F(TurnPortTest, TurnTcpAllocationNotDiscardedIfNotBoundToBestIP) {
   turn_port_->PrepareAddress();
 
   // Candidate should be gathered as normally.
-  EXPECT_THAT(WaitUntil([&] { return turn_ready_; }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(kSimulatedRtt * 3),
-                         .clock = &fake_clock_}),
-              IsRtcOk());
+  EXPECT_THAT(
+      WaitUntil([&] { return turn_ready_; }, IsTrue(),
+                {.timeout = kSimulatedRtt * 3, .clock = &time_controller_}),
+      IsRtcOk());
   ASSERT_EQ(1U, turn_port_->Candidates().size());
 
   // Verify that the socket actually used the alternate address, otherwise this
@@ -1209,10 +1194,10 @@ TEST_F(TurnPortTest, TCPPortNotDiscardedIfBoundToTemporaryIP) {
   turn_port_->PrepareAddress();
 
   // Candidate should be gathered as normally.
-  EXPECT_THAT(WaitUntil([&] { return turn_ready_; }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(kSimulatedRtt * 3),
-                         .clock = &fake_clock_}),
-              IsRtcOk());
+  EXPECT_THAT(
+      WaitUntil([&] { return turn_ready_; }, IsTrue(),
+                {.timeout = kSimulatedRtt * 3, .clock = &time_controller_}),
+      IsRtcOk());
   ASSERT_EQ(1U, turn_port_->Candidates().size());
 }
 
@@ -1224,7 +1209,7 @@ TEST_F(TurnPortTest, TestTurnTcpOnAddressResolveFailure) {
                  ProtocolAddress(kTurnInvalidAddr, PROTO_TCP));
   turn_port_->PrepareAddress();
   EXPECT_THAT(WaitUntil([&] { return turn_error_; }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(kResolverTimeout)}),
+                        {.timeout = kResolverTimeout}),
               IsRtcOk());
   // As VSS doesn't provide DNS resolution, name resolve will fail. TurnPort
   // will proceed in creating a TCP socket which will fail as there is no
@@ -1232,8 +1217,7 @@ TEST_F(TurnPortTest, TestTurnTcpOnAddressResolveFailure) {
   EXPECT_EQ(SOCKET_ERROR, turn_port_->error());
   EXPECT_THAT(WaitUntil([&] { return error_event_.error_code; },
                         Eq(STUN_ERROR_SERVER_NOT_REACHABLE),
-                        {.timeout = TimeDelta::Millis(kSimulatedRtt),
-                         .clock = &fake_clock_}),
+                        {.timeout = kSimulatedRtt, .clock = &time_controller_}),
               IsRtcOk());
   std::string server_url =
       "turn:" + kTurnInvalidAddr.ToString() + "?transport=tcp";
@@ -1248,7 +1232,7 @@ TEST_F(TurnPortTest, TestTurnTlsOnAddressResolveFailure) {
                  ProtocolAddress(kTurnInvalidAddr, PROTO_TLS));
   turn_port_->PrepareAddress();
   EXPECT_THAT(WaitUntil([&] { return turn_error_; }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(kResolverTimeout)}),
+                        {.timeout = kResolverTimeout}),
               IsRtcOk());
   EXPECT_EQ(SOCKET_ERROR, turn_port_->error());
 }
@@ -1260,7 +1244,7 @@ TEST_F(TurnPortTest, TestTurnUdpOnAddressResolveFailure) {
                  ProtocolAddress(kTurnInvalidAddr, PROTO_UDP));
   turn_port_->PrepareAddress();
   EXPECT_THAT(WaitUntil([&] { return turn_error_; }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(kResolverTimeout)}),
+                        {.timeout = kResolverTimeout}),
               IsRtcOk());
   // Error from turn port will not be socket error.
   EXPECT_NE(SOCKET_ERROR, turn_port_->error());
@@ -1270,10 +1254,10 @@ TEST_F(TurnPortTest, TestTurnUdpOnAddressResolveFailure) {
 TEST_F(TurnPortTest, TestTurnAllocateBadPassword) {
   CreateTurnPort(kTurnUsername, "bad", kTurnUdpProtoAddr);
   turn_port_->PrepareAddress();
-  EXPECT_THAT(WaitUntil([&] { return turn_error_; }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(kSimulatedRtt * 2),
-                         .clock = &fake_clock_}),
-              IsRtcOk());
+  EXPECT_THAT(
+      WaitUntil([&] { return turn_error_; }, IsTrue(),
+                {.timeout = kSimulatedRtt * 2, .clock = &time_controller_}),
+      IsRtcOk());
   ASSERT_EQ(0U, turn_port_->Candidates().size());
 }
 
@@ -1283,16 +1267,16 @@ TEST_F(TurnPortTest, TestTurnAllocateNonceResetAfterAllocateMismatch) {
   // Do a normal allocation first.
   CreateTurnPort(kTurnUsername, kTurnPassword, kTurnUdpProtoAddr);
   turn_port_->PrepareAddress();
-  EXPECT_THAT(WaitUntil([&] { return turn_ready_; }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(kSimulatedRtt * 2),
-                         .clock = &fake_clock_}),
-              IsRtcOk());
+  EXPECT_THAT(
+      WaitUntil([&] { return turn_ready_; }, IsTrue(),
+                {.timeout = kSimulatedRtt * 2, .clock = &time_controller_}),
+      IsRtcOk());
   SocketAddress first_addr(turn_port_->socket()->GetLocalAddress());
   // Destroy the turnport while keeping the drop probability to 1 to
   // suppress the release of the allocation at the server.
   ss_->set_drop_probability(1.0);
   turn_port_.reset();
-  SIMULATED_WAIT(false, kSimulatedRtt, fake_clock_);
+  time_controller_.AdvanceTime(kSimulatedRtt);
   ss_->set_drop_probability(0.0);
 
   // Force the socket server to assign the same port.
@@ -1312,10 +1296,10 @@ TEST_F(TurnPortTest, TestTurnAllocateNonceResetAfterAllocateMismatch) {
   // Four round trips; first we'll get "stale nonce", then
   // "allocate mismatch", then "stale nonce" again, then finally it will
   // succeed.
-  EXPECT_THAT(WaitUntil([&] { return turn_ready_; }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(kSimulatedRtt * 4),
-                         .clock = &fake_clock_}),
-              IsRtcOk());
+  EXPECT_THAT(
+      WaitUntil([&] { return turn_ready_; }, IsTrue(),
+                {.timeout = kSimulatedRtt * 4, .clock = &time_controller_}),
+      IsRtcOk());
   EXPECT_NE(first_nonce, turn_port_->nonce());
 }
 
@@ -1325,10 +1309,10 @@ TEST_F(TurnPortTest, TestTurnAllocateMismatch) {
   // Do a normal allocation first.
   CreateTurnPort(kTurnUsername, kTurnPassword, kTurnUdpProtoAddr);
   turn_port_->PrepareAddress();
-  EXPECT_THAT(WaitUntil([&] { return turn_ready_; }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(kSimulatedRtt * 2),
-                         .clock = &fake_clock_}),
-              IsRtcOk());
+  EXPECT_THAT(
+      WaitUntil([&] { return turn_ready_; }, IsTrue(),
+                {.timeout = kSimulatedRtt * 2, .clock = &time_controller_}),
+      IsRtcOk());
   SocketAddress first_addr(turn_port_->socket()->GetLocalAddress());
 
   // Clear connected_ flag on turnport to suppress the release of
@@ -1348,10 +1332,10 @@ TEST_F(TurnPortTest, TestTurnAllocateMismatch) {
   // Four round trips; first we'll get "stale nonce", then
   // "allocate mismatch", then "stale nonce" again, then finally it will
   // succeed.
-  EXPECT_THAT(WaitUntil([&] { return turn_ready_; }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(kSimulatedRtt * 4),
-                         .clock = &fake_clock_}),
-              IsRtcOk());
+  EXPECT_THAT(
+      WaitUntil([&] { return turn_ready_; }, IsTrue(),
+                {.timeout = kSimulatedRtt * 4, .clock = &time_controller_}),
+      IsRtcOk());
 
   // Verifies that the new port has a different address now.
   EXPECT_NE(first_addr, turn_port_->socket()->GetLocalAddress());
@@ -1371,10 +1355,10 @@ TEST_F(TurnPortTest, TestSharedSocketAllocateMismatch) {
   // Do a normal allocation first.
   CreateSharedTurnPort(kTurnUsername, kTurnPassword, kTurnUdpProtoAddr);
   turn_port_->PrepareAddress();
-  EXPECT_THAT(WaitUntil([&] { return turn_ready_; }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(kSimulatedRtt * 2),
-                         .clock = &fake_clock_}),
-              IsRtcOk());
+  EXPECT_THAT(
+      WaitUntil([&] { return turn_ready_; }, IsTrue(),
+                {.timeout = kSimulatedRtt * 2, .clock = &time_controller_}),
+      IsRtcOk());
   SocketAddress first_addr(turn_port_->socket()->GetLocalAddress());
 
   // Clear connected_ flag on turnport to suppress the release of
@@ -1390,10 +1374,10 @@ TEST_F(TurnPortTest, TestSharedSocketAllocateMismatch) {
 
   turn_port_->PrepareAddress();
   // Extra 2 round trips due to allocate mismatch.
-  EXPECT_THAT(WaitUntil([&] { return turn_ready_; }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(kSimulatedRtt * 4),
-                         .clock = &fake_clock_}),
-              IsRtcOk());
+  EXPECT_THAT(
+      WaitUntil([&] { return turn_ready_; }, IsTrue(),
+                {.timeout = kSimulatedRtt * 4, .clock = &time_controller_}),
+      IsRtcOk());
 
   // Verifies that the new port has a different address now.
   EXPECT_NE(first_addr, turn_port_->socket()->GetLocalAddress());
@@ -1406,10 +1390,10 @@ TEST_F(TurnPortTest, TestTurnTcpAllocateMismatch) {
 
   // Do a normal allocation first.
   turn_port_->PrepareAddress();
-  EXPECT_THAT(WaitUntil([&] { return turn_ready_; }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(kSimulatedRtt * 3),
-                         .clock = &fake_clock_}),
-              IsRtcOk());
+  EXPECT_THAT(
+      WaitUntil([&] { return turn_ready_; }, IsTrue(),
+                {.timeout = kSimulatedRtt * 3, .clock = &time_controller_}),
+      IsRtcOk());
   SocketAddress first_addr(turn_port_->socket()->GetLocalAddress());
 
   // Clear connected_ flag on turnport to suppress the release of
@@ -1427,10 +1411,10 @@ TEST_F(TurnPortTest, TestTurnTcpAllocateMismatch) {
   EXPECT_EQ(first_addr, turn_port_->socket()->GetLocalAddress());
 
   // Extra 2 round trips due to allocate mismatch.
-  EXPECT_THAT(WaitUntil([&] { return turn_ready_; }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(kSimulatedRtt * 5),
-                         .clock = &fake_clock_}),
-              IsRtcOk());
+  EXPECT_THAT(
+      WaitUntil([&] { return turn_ready_; }, IsTrue(),
+                {.timeout = kSimulatedRtt * 5, .clock = &time_controller_}),
+      IsRtcOk());
 
   // Verifies that the new port has a different address now.
   EXPECT_NE(first_addr, turn_port_->socket()->GetLocalAddress());
@@ -1450,14 +1434,12 @@ TEST_F(TurnPortTest, TestRefreshRequestGetsErrorResponse) {
   // credential.
   turn_port_->request_manager().FlushForTest(TURN_REFRESH_REQUEST);
   EXPECT_THAT(WaitUntil([&] { return turn_refresh_success_; }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(kSimulatedRtt),
-                         .clock = &fake_clock_}),
+                        {.timeout = kSimulatedRtt, .clock = &time_controller_}),
               IsRtcOk());
   // Flush it again, it will receive a bad response.
   turn_port_->request_manager().FlushForTest(TURN_REFRESH_REQUEST);
   EXPECT_THAT(WaitUntil([&] { return !turn_refresh_success_; }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(kSimulatedRtt),
-                         .clock = &fake_clock_}),
+                        {.timeout = kSimulatedRtt, .clock = &time_controller_}),
               IsRtcOk());
   EXPECT_FALSE(turn_port_->connected());
   EXPECT_TRUE(CheckAllConnectionsFailedAndPruned());
@@ -1477,17 +1459,17 @@ TEST_F(TurnPortTest, TestStopProcessingPacketsAfterClosed) {
   ASSERT_TRUE(conn2 != nullptr);
   // Make sure conn2 is writable.
   conn2->Ping();
-  EXPECT_THAT(WaitUntil([&] { return conn2->write_state(); },
-                        Eq(Connection::STATE_WRITABLE),
-                        {.timeout = TimeDelta::Millis(kSimulatedRtt * 2),
-                         .clock = &fake_clock_}),
-              IsRtcOk());
+  EXPECT_THAT(
+      WaitUntil([&] { return conn2->write_state(); },
+                Eq(Connection::STATE_WRITABLE),
+                {.timeout = kSimulatedRtt * 2, .clock = &time_controller_}),
+      IsRtcOk());
 
   turn_port_->CloseForTest();
-  SIMULATED_WAIT(false, kSimulatedRtt, fake_clock_);
+  time_controller_.AdvanceTime(kSimulatedRtt);
   turn_unknown_address_ = false;
   conn2->Ping();
-  SIMULATED_WAIT(false, kSimulatedRtt, fake_clock_);
+  time_controller_.AdvanceTime(kSimulatedRtt);
   // Since the turn port does not handle packets any more, it should not
   // SignalUnknownAddress.
   EXPECT_FALSE(turn_unknown_address_);
@@ -1522,9 +1504,9 @@ TEST_F(TurnPortTest, TestSocketCloseWillDestroyConnection) {
   EXPECT_TRUE(!turn_port_->connections().empty());
   turn_port_->socket()->NotifyClosedForTest(1);
   EXPECT_THAT(
-      WaitUntil([&] { return turn_port_->connections().empty(); }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kConnectionDestructionDelay),
-                 .clock = &fake_clock_}),
+      WaitUntil(
+          [&] { return turn_port_->connections().empty(); }, IsTrue(),
+          {.timeout = kConnectionDestructionDelay, .clock = &time_controller_}),
       IsRtcOk());
 }
 
@@ -1669,9 +1651,8 @@ TEST_F(TurnPortTest, TestRefreshCreatePermissionRequest) {
                                                   Port::ORIGIN_MESSAGE);
   ASSERT_TRUE(conn != nullptr);
   EXPECT_THAT(
-      WaitUntil(
-          [&] { return turn_create_permission_success_; }, IsTrue(),
-          {.timeout = TimeDelta::Millis(kSimulatedRtt), .clock = &fake_clock_}),
+      WaitUntil([&] { return turn_create_permission_success_; }, IsTrue(),
+                {.timeout = kSimulatedRtt, .clock = &time_controller_}),
       IsRtcOk());
   turn_create_permission_success_ = false;
   // A create-permission-request should be pending.
@@ -1681,16 +1662,14 @@ TEST_F(TurnPortTest, TestRefreshCreatePermissionRequest) {
   turn_port_->set_credentials(bad_credentials);
   turn_port_->request_manager().FlushForTest(kAllRequestsForTest);
   EXPECT_THAT(
-      WaitUntil(
-          [&] { return turn_create_permission_success_; }, IsTrue(),
-          {.timeout = TimeDelta::Millis(kSimulatedRtt), .clock = &fake_clock_}),
+      WaitUntil([&] { return turn_create_permission_success_; }, IsTrue(),
+                {.timeout = kSimulatedRtt, .clock = &time_controller_}),
       IsRtcOk());
   // Flush the requests again; the create-permission-request will fail.
   turn_port_->request_manager().FlushForTest(kAllRequestsForTest);
   EXPECT_THAT(
-      WaitUntil(
-          [&] { return !turn_create_permission_success_; }, IsTrue(),
-          {.timeout = TimeDelta::Millis(kSimulatedRtt), .clock = &fake_clock_}),
+      WaitUntil([&] { return !turn_create_permission_success_; }, IsTrue(),
+                {.timeout = kSimulatedRtt, .clock = &time_controller_}),
       IsRtcOk());
   EXPECT_TRUE(CheckConnectionFailedAndPruned(conn));
 }
@@ -1706,10 +1685,10 @@ TEST_F(TurnPortTest, TestChannelBindGetErrorResponse) {
 
   ASSERT_TRUE(conn2 != nullptr);
   conn1->Ping();
-  EXPECT_THAT(WaitUntil([&] { return conn1->writable(); }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(kSimulatedRtt * 2),
-                         .clock = &fake_clock_}),
-              IsRtcOk());
+  EXPECT_THAT(
+      WaitUntil([&] { return conn1->writable(); }, IsTrue(),
+                {.timeout = kSimulatedRtt * 2, .clock = &time_controller_}),
+      IsRtcOk());
 
   // Tell the TURN server to reject all bind requests from now on.
   turn_server_.server()->set_reject_bind_requests(true);
@@ -1718,9 +1697,8 @@ TEST_F(TurnPortTest, TestChannelBindGetErrorResponse) {
   conn1->Send(data.data(), data.length(), options);
 
   EXPECT_THAT(
-      WaitUntil(
-          [&] { return CheckConnectionFailedAndPruned(conn1); }, IsTrue(),
-          {.timeout = TimeDelta::Millis(kSimulatedRtt), .clock = &fake_clock_}),
+      WaitUntil([&] { return CheckConnectionFailedAndPruned(conn1); }, IsTrue(),
+                {.timeout = kSimulatedRtt, .clock = &time_controller_}),
       IsRtcOk());
   // Verify that packets are allowed to be sent after a bind request error.
   // They'll just use a send indication instead.
@@ -1732,8 +1710,7 @@ TEST_F(TurnPortTest, TestChannelBindGetErrorResponse) {
       });
   conn1->Send(data.data(), data.length(), options);
   EXPECT_THAT(WaitUntil([&] { return !udp_packets_.empty(); }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(kSimulatedRtt),
-                         .clock = &fake_clock_}),
+                        {.timeout = kSimulatedRtt, .clock = &time_controller_}),
               IsRtcOk());
   conn2->DeregisterReceivedPacketCallback();
 }
@@ -1771,8 +1748,7 @@ TEST_F(TurnPortTest, TestTurnLocalIPv6AddressServerIPv4) {
                  kTurnUdpProtoAddr);
   turn_port_->PrepareAddress();
   ASSERT_THAT(WaitUntil([&] { return turn_error_; }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(kSimulatedRtt),
-                         .clock = &fake_clock_}),
+                        {.timeout = kSimulatedRtt, .clock = &time_controller_}),
               IsRtcOk());
   EXPECT_TRUE(turn_port_->Candidates().empty());
 }
@@ -1797,10 +1773,10 @@ TEST_F(TurnPortTest, TestCandidateAddressFamilyMatch) {
   CreateTurnPort(kLocalIPv6Addr, kTurnUsername, kTurnPassword,
                  kTurnUdpIPv6ProtoAddr);
   turn_port_->PrepareAddress();
-  EXPECT_THAT(WaitUntil([&] { return turn_ready_; }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(kSimulatedRtt * 2),
-                         .clock = &fake_clock_}),
-              IsRtcOk());
+  EXPECT_THAT(
+      WaitUntil([&] { return turn_ready_; }, IsTrue(),
+                {.timeout = kSimulatedRtt * 2, .clock = &time_controller_}),
+      IsRtcOk());
   ASSERT_EQ(1U, turn_port_->Candidates().size());
 
   // Create an IPv4 candidate. It will match the TURN candidate.
@@ -1826,16 +1802,15 @@ TEST_F(TurnPortTest, TestConnectionFailedAndPrunedOnCreatePermissionFailure) {
   turn_server_.server()->set_reject_private_addresses(true);
   CreateTurnPort(kTurnUsername, kTurnPassword, kTurnTcpProtoAddr);
   turn_port_->PrepareAddress();
-  EXPECT_THAT(WaitUntil([&] { return turn_ready_; }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(kSimulatedRtt * 3),
-                         .clock = &fake_clock_}),
-              IsRtcOk());
+  EXPECT_THAT(
+      WaitUntil([&] { return turn_ready_; }, IsTrue(),
+                {.timeout = kSimulatedRtt * 3, .clock = &time_controller_}),
+      IsRtcOk());
 
   CreateUdpPort(SocketAddress("10.0.0.10", 0));
   udp_port_->PrepareAddress();
   EXPECT_THAT(WaitUntil([&] { return udp_ready_; }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(kSimulatedRtt),
-                         .clock = &fake_clock_}),
+                        {.timeout = kSimulatedRtt, .clock = &time_controller_}),
               IsRtcOk());
   // Create a connection.
   TestConnectionWrapper conn(turn_port_->CreateConnection(
@@ -1847,17 +1822,14 @@ TEST_F(TurnPortTest, TestConnectionFailedAndPrunedOnCreatePermissionFailure) {
   EXPECT_THAT(
       WaitUntil(
           [&] { return CheckConnectionFailedAndPruned(conn.connection()); },
-          IsTrue(),
-          {.timeout = TimeDelta::Millis(kSimulatedRtt), .clock = &fake_clock_}),
+          IsTrue(), {.timeout = kSimulatedRtt, .clock = &time_controller_}),
       IsRtcOk());
   EXPECT_THAT(
-      WaitUntil(
-          [&] { return !turn_create_permission_success_; }, IsTrue(),
-          {.timeout = TimeDelta::Millis(kSimulatedRtt), .clock = &fake_clock_}),
+      WaitUntil([&] { return !turn_create_permission_success_; }, IsTrue(),
+                {.timeout = kSimulatedRtt, .clock = &time_controller_}),
       IsRtcOk());
   // Check that the connection is not deleted asynchronously.
-  SIMULATED_WAIT(conn.connection() == nullptr, kConnectionDestructionDelay,
-                 fake_clock_);
+  time_controller_.AdvanceTime(kConnectionDestructionDelay);
   EXPECT_NE(nullptr, conn.connection());
 }
 
@@ -1923,14 +1895,14 @@ TEST_F(TurnPortTest, TestResolverShutdown) {
                  ProtocolAddress(kTurnInvalidAddr, PROTO_UDP));
   turn_port_->PrepareAddress();
   ASSERT_THAT(WaitUntil([&] { return turn_error_; }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(kResolverTimeout)}),
+                        {.timeout = kResolverTimeout}),
               IsRtcOk());
   EXPECT_TRUE(turn_port_->Candidates().empty());
   turn_port_.reset();
   Thread::Current()->PostTask([this] { test_finish_ = true; });
   // Waiting for above message to be processed.
   ASSERT_THAT(WaitUntil([&] { return test_finish_; }, IsTrue(),
-                        {.clock = &fake_clock_}),
+                        {.clock = &time_controller_}),
               IsRtcOk());
   EXPECT_EQ(last_fd_count, GetFDCount());
 }
@@ -2123,11 +2095,11 @@ TEST_F(TurnPortTest, TestTurnDangerousAlternateServer) {
 
   turn_port_->PrepareAddress();
   // This should result in an error event.
-  EXPECT_THAT(WaitUntil([&] { return error_event_.error_code; }, Ne(0),
-                        {.timeout = TimeDelta::Millis(
-                             TimeToGetAlternateTurnCandidate(protocol_type)),
-                         .clock = &fake_clock_}),
-              IsRtcOk());
+  EXPECT_THAT(
+      WaitUntil([&] { return error_event_.error_code; }, Ne(0),
+                {.timeout = TimeToGetAlternateTurnCandidate(protocol_type),
+                 .clock = &time_controller_}),
+      IsRtcOk());
   // but should NOT result in the port turning ready, and no candidates
   // should be gathered.
   EXPECT_FALSE(turn_ready_);
@@ -2153,18 +2125,17 @@ class TurnPortWithMockDnsResolverTest : public TurnPortTest {
 // Test an allocation from a TURN server specified by a hostname.
 TEST_F(TurnPortWithMockDnsResolverTest, TestHostnameResolved) {
   CreateTurnPort(kTurnUsername, kTurnPassword, kTurnPortValidHostnameProtoAddr);
-  SetDnsResolverExpectations(
-      [](MockAsyncDnsResolver* resolver,
-         MockAsyncDnsResolverResult* resolver_result) {
-        EXPECT_CALL(*resolver, Start(kTurnValidAddr, /*family=*/AF_INET, _))
-            .WillOnce([](const SocketAddress& addr, int family,
-                         absl::AnyInvocable<void()> callback) { callback(); });
-        EXPECT_CALL(*resolver, result)
-            .WillRepeatedly(ReturnPointee(resolver_result));
-        EXPECT_CALL(*resolver_result, GetError).WillRepeatedly(Return(0));
-        EXPECT_CALL(*resolver_result, GetResolvedAddress(AF_INET, _))
-            .WillOnce(DoAll(SetArgPointee<1>(kTurnUdpIntAddr), Return(true)));
-      });
+  SetDnsResolverExpectations([](MockAsyncDnsResolver* resolver,
+                                MockAsyncDnsResolverResult* resolver_result) {
+    EXPECT_CALL(*resolver, Start(kTurnValidAddr, /*family=*/AF_INET, _))
+        .WillOnce([](const SocketAddress& addr, int family,
+                     absl::AnyInvocable<void()> callback) { callback(); });
+    EXPECT_CALL(*resolver, result)
+        .WillRepeatedly(ReturnPointee(resolver_result));
+    EXPECT_CALL(*resolver_result, GetError).WillRepeatedly(Return(0));
+    EXPECT_CALL(*resolver_result, GetResolvedAddress(AF_INET, _))
+        .WillOnce(DoAll(SetArgPointee<1>(kTurnUdpIntAddr), Return(true)));
+  });
   TestTurnAllocateSucceeds(kSimulatedRtt * 2);
 }
 
@@ -2174,19 +2145,17 @@ TEST_F(TurnPortWithMockDnsResolverTest, TestHostnameResolvedIPv6Network) {
   turn_server_.AddInternalSocket(kTurnUdpIPv6IntAddr, PROTO_UDP);
   CreateTurnPort(kLocalIPv6Addr, kTurnUsername, kTurnPassword,
                  kTurnPortValidHostnameProtoAddr);
-  SetDnsResolverExpectations(
-      [](MockAsyncDnsResolver* resolver,
-         MockAsyncDnsResolverResult* resolver_result) {
-        EXPECT_CALL(*resolver, Start(kTurnValidAddr, /*family=*/AF_INET6, _))
-            .WillOnce([](const SocketAddress& addr, int family,
-                         absl::AnyInvocable<void()> callback) { callback(); });
-        EXPECT_CALL(*resolver, result)
-            .WillRepeatedly(ReturnPointee(resolver_result));
-        EXPECT_CALL(*resolver_result, GetError).WillRepeatedly(Return(0));
-        EXPECT_CALL(*resolver_result, GetResolvedAddress(AF_INET6, _))
-            .WillOnce(
-                DoAll(SetArgPointee<1>(kTurnUdpIPv6IntAddr), Return(true)));
-      });
+  SetDnsResolverExpectations([](MockAsyncDnsResolver* resolver,
+                                MockAsyncDnsResolverResult* resolver_result) {
+    EXPECT_CALL(*resolver, Start(kTurnValidAddr, /*family=*/AF_INET6, _))
+        .WillOnce([](const SocketAddress& addr, int family,
+                     absl::AnyInvocable<void()> callback) { callback(); });
+    EXPECT_CALL(*resolver, result)
+        .WillRepeatedly(ReturnPointee(resolver_result));
+    EXPECT_CALL(*resolver_result, GetError).WillRepeatedly(Return(0));
+    EXPECT_CALL(*resolver_result, GetResolvedAddress(AF_INET6, _))
+        .WillOnce(DoAll(SetArgPointee<1>(kTurnUdpIPv6IntAddr), Return(true)));
+  });
   TestTurnAllocateSucceeds(kSimulatedRtt * 2);
 }
 

From 133ca9e54b66a47ccf5b0a3d366de0b902d47be3 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Wed, 18 Mar 2026 07:15:05 -0700
Subject: [PATCH 339/472] Roll chromium_revision b30d84fa40..72553e8b4b
 (1601092:1601266)

Change log: https://chromium.googlesource.com/chromium/src/+log/b30d84fa40..72553e8b4b
Full diff: https://chromium.googlesource.com/chromium/src/+/b30d84fa40..72553e8b4b

Changed dependencies
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/904cd4f067..471b43a4ef
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/8527fd9f72..260e1270b0
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/793a1bc367..9c2ce92f1a
* src/third_party/android_deps/autorolled/cipd: bSNUpmo1wIHCZI5w3gxpUEAdVSqFPcTlstxTARVxcuwC..PGgvEOiRpg1Z5fBKSrpvnnX1CFBZU6KVNUBVkyukBKoC
* src/third_party/androidx/cipd: IxP1FmK52UK5we5ZE7jEkvED0_fBvkCslzXKDVIuV9QC..rz90VPpK87UYOL1GQGLlBARBJdK5-L_MAuzuM8bszasC
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/95d1432b5c..d819b798b7
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/1726a92900..c148dc8fad
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/9b59f48e11..08919e34b6
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/4dc3ed6779..9d8b817ce4
DEPS diff: https://chromium.googlesource.com/chromium/src/+/b30d84fa40..72553e8b4b/DEPS

BUG=None

Change-Id: Ib23618976aed0cd3c22b2f9ebb9522c2388f2ceb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/458020
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47192}
---
 DEPS | 20 ++++++++++----------
 1 file changed, 10 insertions(+), 10 deletions(-)

diff --git a/DEPS b/DEPS
index bc998a8512..8d8eefeeaa 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': 'b30d84fa40626ba910ed277984b3cc7737f49c2e',
+  'chromium_revision': '72553e8b4bdc25825c847a7093078979e87c838b',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@904cd4f06715973dbd0f092eee0aa8e2c5d7667e',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@471b43a4ef2c531b50b1109e05604bf2f7e34a98',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@8527fd9f72b14778901d9cac901736c97213344f',
+    'https://chromium.googlesource.com/chromium/src/testing@260e1270b0bd9281950ded5f58da8d8b32f43b51',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@793a1bc3679b136c77dfffe8c74b09a31786b9c0',
+    'https://chromium.googlesource.com/chromium/src/third_party@9c2ce92f1a07cd93501f76189f9dfd058994bb38',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -505,7 +505,7 @@ deps = {
   'src/third_party/crc32c/src':
     'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558',
   'src/third_party/depot_tools':
-    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@95d1432b5c506cc6b500b34c23f06c4708139fe9',
+    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@d819b798b712d4064ad12b7bd9257acf78e34cb0',
   'src/third_party/ffmpeg':
     'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@946d97db8d906277085e361892b7efda5152e2f1',
   'src/third_party/flatbuffers/src':
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '9b59f48e11a56a8d2dddd60299e7e25b68b3072c',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '08919e34b61cbb1950dd20a87f4748d5795fa2a9',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@4dc3ed6779e386922affe81b6b933cc534f2fc32',
+    'https://chromium.googlesource.com/chromium/src/tools@9d8b817ce41993df74f7d6e71a741a42ada1a8ea',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'IxP1FmK52UK5we5ZE7jEkvED0_fBvkCslzXKDVIuV9QC',
+          'version': 'rz90VPpK87UYOL1GQGLlBARBJdK5-L_MAuzuM8bszasC',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -842,7 +842,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/android_deps/autorolled',
-              'version': 'bSNUpmo1wIHCZI5w3gxpUEAdVSqFPcTlstxTARVxcuwC',
+              'version': 'PGgvEOiRpg1Z5fBKSrpvnnX1CFBZU6KVNUBVkyukBKoC',
           },
       ],
       'condition': 'checkout_android and non_git_source',
@@ -864,7 +864,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7607ca500436b37ad23fb8d18614bec7796b68a7',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '1726a929009be60a5ed049d23052cbb7ce6f2f96',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + 'c148dc8fadf4ae427fd57d4edb92b855e3d8939c',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From 76018a2ec3a7f982c5bb2288487231cd9aa4510a Mon Sep 17 00:00:00 2001
From: Palak Agarwal <agpalak@google.com>
Date: Wed, 18 Mar 2026 16:24:45 +0100
Subject: [PATCH 340/472] Change UMA logging to only log once for same kind of
 error

`FullScreenPowerPointHandler::FindEditorWindow()` was previously
recording `FullScreenFindEditorResult` UMA metrics on every frame. This
causes log spam, especially when the heuristic fails due to multiple
editors having the same title.

This CL caches the previous result in `previous_find_editor_result_` and
only records the metric when the result differs from the previous call.

Bug: chromium:409473386
Change-Id: I44500864c89dd7415327c91778086ecd667cdd15
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/458041
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Palak Agarwal <agpalak@google.com>
Cr-Commit-Position: refs/heads/main@{#47193}
---
 .../full_screen_win_application_handler.cc    | 22 +++++++++++++------
 .../win/full_screen_win_application_handler.h |  4 ++++
 2 files changed, 19 insertions(+), 7 deletions(-)

diff --git a/modules/desktop_capture/win/full_screen_win_application_handler.cc b/modules/desktop_capture/win/full_screen_win_application_handler.cc
index c7548b0007..7a245c96f8 100644
--- a/modules/desktop_capture/win/full_screen_win_application_handler.cc
+++ b/modules/desktop_capture/win/full_screen_win_application_handler.cc
@@ -36,10 +36,16 @@ void RecordFullScreenDetectorResult(FullScreenDetectorResult result) {
       static_cast<int>(FullScreenDetectorResult::kMaxValue));
 }
 
-void RecordFullScreenFindEditorResult(FullScreenFindEditorResult result) {
-  RTC_HISTOGRAM_ENUMERATION(
-      "WebRTC.Screenshare.FullScreenFindEditorResult", static_cast<int>(result),
-      static_cast<int>(FullScreenFindEditorResult::kMaxValue));
+void MaybeRecordFullScreenFindEditorResult(
+    FullScreenFindEditorResult result,
+    std::optional<FullScreenFindEditorResult>& previous_result) {
+  if (previous_result != result) {
+    RTC_HISTOGRAM_ENUMERATION(
+        "WebRTC.Screenshare.FullScreenFindEditorResult",
+        static_cast<int>(result),
+        static_cast<int>(FullScreenFindEditorResult::kMaxValue));
+    previous_result = result;
+  }
 }
 
 namespace webrtc {
@@ -246,15 +252,17 @@ DesktopCapturer::SourceId FullScreenPowerPointHandler::FindEditorWindow(
   }
 
   if (editor_ids.size() != 1) {
-    RecordFullScreenFindEditorResult(
-        FullScreenFindEditorResult::kFailureDueToSameTitleWindows);
+    MaybeRecordFullScreenFindEditorResult(
+        FullScreenFindEditorResult::kFailureDueToSameTitleWindows,
+        previous_find_editor_result_);
     // If `editor_ids` has more than one id, then there are multiple open
     // editors with the same title as the full screen slide show and then
     // there's no way of knowing which editor has opened the slide show.
     return 0;
   }
 
-  RecordFullScreenFindEditorResult(FullScreenFindEditorResult::kSuccess);
+  MaybeRecordFullScreenFindEditorResult(FullScreenFindEditorResult::kSuccess,
+                                        previous_find_editor_result_);
   return *editor_ids.begin();
 }
 
diff --git a/modules/desktop_capture/win/full_screen_win_application_handler.h b/modules/desktop_capture/win/full_screen_win_application_handler.h
index 0accd62e9e..3a51cea7e0 100644
--- a/modules/desktop_capture/win/full_screen_win_application_handler.h
+++ b/modules/desktop_capture/win/full_screen_win_application_handler.h
@@ -15,6 +15,7 @@
 
 #include <cstdint>
 #include <memory>
+#include <optional>
 #include <string>
 
 #include "modules/desktop_capture/desktop_capturer.h"
@@ -82,6 +83,9 @@ class FullScreenPowerPointHandler : public FullScreenApplicationHandler {
   mutable bool was_slide_show_created_after_capture_started_;
 
   mutable FullScreenDetectorResult full_screen_detector_result_;
+
+  mutable std::optional<FullScreenFindEditorResult>
+      previous_find_editor_result_;
 };
 
 std::unique_ptr<FullScreenApplicationHandler>

From c0854b6ad5fc50968f235d7e3d8b5c63f42dc479 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Wed, 18 Mar 2026 11:02:59 -0700
Subject: [PATCH 341/472] Roll chromium_revision 72553e8b4b..d609a75cbb
 (1601266:1601402)

Change log: https://chromium.googlesource.com/chromium/src/+log/72553e8b4b..d609a75cbb
Full diff: https://chromium.googlesource.com/chromium/src/+/72553e8b4b..d609a75cbb

Changed dependencies
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/471b43a4ef..a25f412b4d
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/260e1270b0..37f86393a4
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/9c2ce92f1a..2df5dc0d09
* src/third_party/androidx/cipd: rz90VPpK87UYOL1GQGLlBARBJdK5-L_MAuzuM8bszasC..6c2939gs3hJpDeyKGdMRo_u5z0rsDjmulzNMIQzYLyAC
* src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/573ea4ab86..c09cee4577
* src/third_party/fuzztest/src: https://chromium.googlesource.com/external/github.com/google/fuzztest.git/+log/3e42dc3c30..3874258b07
* src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/ea38df85bb..390ab7f3cf
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/9d8b817ce4..d986bada7a
DEPS diff: https://chromium.googlesource.com/chromium/src/+/72553e8b4b..d609a75cbb/DEPS

BUG=None

Change-Id: I5ed48b717b7a98024f0e454cfe81eeebf6ab57ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/457961
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47194}
---
 DEPS | 18 +++++++++---------
 1 file changed, 9 insertions(+), 9 deletions(-)

diff --git a/DEPS b/DEPS
index 8d8eefeeaa..7d830eb598 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '72553e8b4bdc25825c847a7093078979e87c838b',
+  'chromium_revision': 'd609a75cbbeb4105f5a3e473cf67810377004636',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@471b43a4ef2c531b50b1109e05604bf2f7e34a98',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@a25f412b4de1760055290cd397bc6effafda13c3',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@260e1270b0bd9281950ded5f58da8d8b32f43b51',
+    'https://chromium.googlesource.com/chromium/src/testing@37f86393a473d8574d07bcce0e7e3c336f7ce044',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@9c2ce92f1a07cd93501f76189f9dfd058994bb38',
+    'https://chromium.googlesource.com/chromium/src/third_party@2df5dc0d09038d01c5f305bf0e3b4ba5b79db384',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -331,13 +331,13 @@ deps = {
   'src/third_party/clang-format/script':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@c2725e0622e1a86d55f14514f2177a39efea4a0e',
   'src/third_party/compiler-rt/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@573ea4ab862e838d5c8512f654799d14deca07e7',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@c09cee4577254cc33ee4a7699bbe3d7ae3ea730a',
   'src/third_party/libc++/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@7ab65651aed6802d2599dcb7a73b1f82d5179d05',
   'src/third_party/libc++abi/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@8f11bb1d4438d0239d0dfc1bd9456a9f31629dda',
   'src/third_party/llvm-libc/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@ea38df85bbd4203028153aad85fdb148a9dfc5c3',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@390ab7f3cffaa613e4a0a9c5aecf37052d5a8923',
   'src/third_party/libunwind/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@db838d918570d4e381ecf9f5cc70a0098c9c2cd6',
 
@@ -593,7 +593,7 @@ deps = {
   'src/third_party/libFuzzer/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt/lib/fuzzer.git@bea408a6e01f0f7e6c82a43121fe3af4506c932e',
   'src/third_party/fuzztest/src':
-    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@3e42dc3c305dd22a22e7d2996c7e24177f3a43ad',
+    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@3874258b07ad554917c0bac6ea9ee07469af6ac9',
   'src/third_party/libjpeg_turbo':
     'https://chromium.googlesource.com/chromium/deps/libjpeg_turbo.git@d1f5f2393e0d51f840207342ae86e55a86443288',
   'src/third_party/libsrtp':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@9d8b817ce41993df74f7d6e71a741a42ada1a8ea',
+    'https://chromium.googlesource.com/chromium/src/tools@d986bada7a7d66309944e41e2082100e75eddd92',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'rz90VPpK87UYOL1GQGLlBARBJdK5-L_MAuzuM8bszasC',
+          'version': '6c2939gs3hJpDeyKGdMRo_u5z0rsDjmulzNMIQzYLyAC',
       },
     ],
     'condition': 'checkout_android and non_git_source',

From dfceedf0ba3b36c2a5605bbf6347b62069aa1d58 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Mon, 16 Mar 2026 13:14:11 +0100
Subject: [PATCH 342/472] Enhance ScopedOperationsBatcher to support tasks
 returning tasks.

A common pattern for negotiation operations that involve the signaling
thread as well as one or both of the worker/network threads, is to a)
apply partial state on the signaling thread, b) perform work on the
worker/network thread(s), c) run completion work on the signaling thread
that depends on the outcome of the previous step.

This adds support for that pattern to the ScopedOperationsBatcher so
that this pattern can be applied to batched operations.

Bug: webrtc:42222804
Change-Id: I776786216208991c1716a1ab2808c9b37f30ead8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449362
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47195}
---
 pc/sdp_offer_answer.cc | 45 ++++++++++++++++++++++++++++++++++++------
 1 file changed, 39 insertions(+), 6 deletions(-)

diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc
index 6048d98657..24abb0daf7 100644
--- a/pc/sdp_offer_answer.cc
+++ b/pc/sdp_offer_answer.cc
@@ -112,6 +112,15 @@
 namespace webrtc {
 namespace {
 
+// Batches operations to be executed synchronously on the worker thread.
+//
+// When the batcher goes out of scope (or `Run()` is explicitly called), it
+// executes all queued tasks in a single `BlockingCall` to the worker thread.
+//
+// Tasks can either have a `void` return type, or return a new task.
+// Any tasks returned by the executed worker thread tasks are collected
+// and subsequently executed on the calling thread (typically the signaling
+// thread) after the worker thread operations have completed.
 class ScopedOperationsBatcher {
  public:
   explicit ScopedOperationsBatcher(Thread* worker_thread)
@@ -122,13 +131,26 @@ class ScopedOperationsBatcher {
   ~ScopedOperationsBatcher() { Run(); }
 
   void Run() {
+    std::vector<absl::AnyInvocable<void() &&>> signaling_tasks;
     if (!tasks_.empty()) {
-      worker_thread_->BlockingCall([tasks = std::move(tasks_)]() mutable {
-        for (auto& task : tasks) {
-          std::move(task)();
+      worker_thread_->BlockingCall([&] {
+        for (auto& task : tasks_) {
+          if (task.void_task) {
+            std::move(task.void_task)();
+          } else {
+            RTC_DCHECK(task.returning_task);
+            auto ret = std::move(task.returning_task)();
+            if (ret) {
+              signaling_tasks.push_back(std::move(ret));
+            }
+          }
         }
       });
-      RTC_DCHECK(tasks_.empty());
+      tasks_.clear();
+    }
+
+    for (auto& task : signaling_tasks) {
+      std::move(task)();
     }
   }
 
@@ -136,13 +158,24 @@ class ScopedOperationsBatcher {
   // ScopedOperationsBatcher goes out of scope.
   void push_back(absl::AnyInvocable<void() &&> task) {
     if (task) {
-      tasks_.push_back(std::move(task));
+      tasks_.push_back({.void_task = std::move(task)});
+    }
+  }
+
+  void push_back(absl::AnyInvocable<absl::AnyInvocable<void() &&>() &&> task) {
+    if (task) {
+      tasks_.push_back({.returning_task = std::move(task)});
     }
   }
 
  private:
+  struct BatchedTask {
+    absl::AnyInvocable<void() &&> void_task;
+    absl::AnyInvocable<absl::AnyInvocable<void() &&>() &&> returning_task;
+  };
+
   Thread* const worker_thread_;
-  std::vector<absl::AnyInvocable<void() &&>> tasks_;
+  std::vector<BatchedTask> tasks_;
 };
 
 struct DtlsTransportAndName {

From ff23a3d012fa7db9a3e2c8ac7ee2d18a64f3aa82 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Wed, 18 Mar 2026 15:02:39 -0700
Subject: [PATCH 343/472] Roll chromium_revision d609a75cbb..ff9c037b0d
 (1601402:1601561)

Change log: https://chromium.googlesource.com/chromium/src/+log/d609a75cbb..ff9c037b0d
Full diff: https://chromium.googlesource.com/chromium/src/+/d609a75cbb..ff9c037b0d

Changed dependencies
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/a25f412b4d..5375498adc
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/37f86393a4..1200ca9f0c
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/2df5dc0d09..36d58e6b00
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/f7a1fb2d8f..df4f860924
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/d819b798b7..8e07900125
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/c148dc8fad..0fd8002b11
* src/third_party/r8/cipd: o1h6_6tQfYYay5tJcyLeEi-HODtdK5nFoPd6fls0oTkC..8ZRb6CCpZTU5dSpQyDlbusalGCjWV0sVSGTq_0Js3mcC
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/d986bada7a..1e4b145466
DEPS diff: https://chromium.googlesource.com/chromium/src/+/d609a75cbb..ff9c037b0d/DEPS

BUG=None

Change-Id: I23a9537ce1f54da75a788d3929e1209e566f0d8e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/458100
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47196}
---
 DEPS | 18 +++++++++---------
 1 file changed, 9 insertions(+), 9 deletions(-)

diff --git a/DEPS b/DEPS
index 7d830eb598..d87df856b2 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': 'd609a75cbbeb4105f5a3e473cf67810377004636',
+  'chromium_revision': 'ff9c037b0d3ce2a74c0ce7fa6a5fa42167010897',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@a25f412b4de1760055290cd397bc6effafda13c3',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@5375498adc780c22db50c5992e30cce280762eb6',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@37f86393a473d8574d07bcce0e7e3c336f7ce044',
+    'https://chromium.googlesource.com/chromium/src/testing@1200ca9f0c8da0b4aa6e5e508ed1b3ced0651d4e',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@2df5dc0d09038d01c5f305bf0e3b4ba5b79db384',
+    'https://chromium.googlesource.com/chromium/src/third_party@36d58e6b0091a1ebbd1ee6ac4643fc285d601193',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -492,7 +492,7 @@ deps = {
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@8be0e3114685fcc1589561067282edf75ea1259a',
   'src/third_party/catapult':
-    'https://chromium.googlesource.com/catapult.git@f7a1fb2d8f3d155f6d52e95fa486bf91a9ebca81',
+    'https://chromium.googlesource.com/catapult.git@df4f8609244fd2378c322760c5172cb3d4fd8725',
   'src/third_party/ced/src': {
     'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5',
   },
@@ -505,7 +505,7 @@ deps = {
   'src/third_party/crc32c/src':
     'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558',
   'src/third_party/depot_tools':
-    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@d819b798b712d4064ad12b7bd9257acf78e34cb0',
+    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@8e07900125ba28f056008bded80053b297837aab',
   'src/third_party/ffmpeg':
     'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@946d97db8d906277085e361892b7efda5152e2f1',
   'src/third_party/flatbuffers/src':
@@ -644,7 +644,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/r8',
-              'version': 'o1h6_6tQfYYay5tJcyLeEi-HODtdK5nFoPd6fls0oTkC',
+              'version': '8ZRb6CCpZTU5dSpQyDlbusalGCjWV0sVSGTq_0Js3mcC',
           },
       ],
       'condition': 'checkout_android',
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@d986bada7a7d66309944e41e2082100e75eddd92',
+    'https://chromium.googlesource.com/chromium/src/tools@1e4b1454662363af593e8350b596372bc9352f8f',
 
   'src/third_party/espresso': {
       'packages': [
@@ -864,7 +864,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7607ca500436b37ad23fb8d18614bec7796b68a7',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + 'c148dc8fadf4ae427fd57d4edb92b855e3d8939c',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '0fd8002b11852969d5a8625dd151c085bf44af57',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From 6144ca0b10eebea84ac90798163b1d67fc3d2067 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Wed, 18 Mar 2026 19:03:31 -0700
Subject: [PATCH 344/472] Roll chromium_revision ff9c037b0d..c60f7fb998
 (1601561:1601697)

Change log: https://chromium.googlesource.com/chromium/src/+log/ff9c037b0d..c60f7fb998
Full diff: https://chromium.googlesource.com/chromium/src/+/ff9c037b0d..c60f7fb998

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/7344e69ded..166f3230f6
* src/buildtools/linux64: git_revision:d22fa2d126d8e4db585e9db0f3fb2fb4e0bf601b..git_revision:335219e2893b5a3b30ec8ce27d48ec762c7de1d5
* src/buildtools/mac: git_revision:d22fa2d126d8e4db585e9db0f3fb2fb4e0bf601b..git_revision:335219e2893b5a3b30ec8ce27d48ec762c7de1d5
* src/buildtools/win: git_revision:d22fa2d126d8e4db585e9db0f3fb2fb4e0bf601b..git_revision:335219e2893b5a3b30ec8ce27d48ec762c7de1d5
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/5375498adc..74494157ca
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/1200ca9f0c..7a7d159e99
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/36d58e6b00..549deb2670
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/df4f860924..a41fdb3aa5
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/0fd8002b11..05295a818b
* src/third_party/r8/d8/cipd: CQBWQVHNMmNBTkKhqA-Rnel-94ivC4c9jRROa8bvIUkC..8ZRb6CCpZTU5dSpQyDlbusalGCjWV0sVSGTq_0Js3mcC
* src/third_party/rust-toolchain: Linux_x64/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-2-llvmorg-23-init-5669-g8a0be0bc.tar.xz,Mac/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-2-llvmorg-23-init-5669-g8a0be0bc.tar.xz,Mac_arm64/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-2-llvmorg-23-init-5669-g8a0be0bc.tar.xz,Win/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-2-llvmorg-23-init-5669-g8a0be0bc.tar.xz..Linux_x64/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-7-llvmorg-23-init-5669-g8a0be0bc.tar.xz,Mac/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-7-llvmorg-23-init-5669-g8a0be0bc.tar.xz,Mac_arm64/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-7-llvmorg-23-init-5669-g8a0be0bc.tar.xz,Win/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-7-llvmorg-23-init-5669-g8a0be0bc.tar.xz
* src/third_party/tflite/src: https://chromium.googlesource.com/external/github.com/tensorflow/tensorflow.git/+log/24d66fbe6c..4e546f6967
* src/third_party/xnnpack/src: https://chromium.googlesource.com/external/github.com/google/XNNPACK.git/+log/ee91cc745b..f1a5f31a23
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/1e4b145466..a6b6df6c66
DEPS diff: https://chromium.googlesource.com/chromium/src/+/ff9c037b0d..c60f7fb998/DEPS

BUG=None

Change-Id: Ic1faa03286e72cde436078982de0e489107c84d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/458140
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47197}
---
 DEPS | 60 ++++++++++++++++++++++++++++++------------------------------
 1 file changed, 30 insertions(+), 30 deletions(-)

diff --git a/DEPS b/DEPS
index d87df856b2..eca2389c46 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': 'ff9c037b0d3ce2a74c0ce7fa6a5fa42167010897',
+  'chromium_revision': 'c60f7fb9989b3ebd09409d614f219c3c8694d663',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@7344e69ded7ed29ef7134b9d4f54763245a43466',
+    'https://chromium.googlesource.com/chromium/src/build@166f3230f69fdb95cf3da587e6ee143c2c03c971',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@95ed44cf5f06dbb5861030b91c9db9ccb4316762',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,19 +78,19 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@5375498adc780c22db50c5992e30cce280762eb6',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@74494157cad1e80e4ebae6f3ef431f2de02444c9',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@1200ca9f0c8da0b4aa6e5e508ed1b3ced0651d4e',
+    'https://chromium.googlesource.com/chromium/src/testing@7a7d159e99818292ce6b17466c89952cc79e2cc3',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@36d58e6b0091a1ebbd1ee6ac4643fc285d601193',
+    'https://chromium.googlesource.com/chromium/src/third_party@549deb2670a0662c14c4ac47a4c5fb8927b4b6e6',
 
   'src/buildtools/linux64': {
     'packages': [
       {
         'package': 'gn/gn/linux-${{arch}}',
-        'version': 'git_revision:d22fa2d126d8e4db585e9db0f3fb2fb4e0bf601b',
+        'version': 'git_revision:335219e2893b5a3b30ec8ce27d48ec762c7de1d5',
       }
     ],
     'dep_type': 'cipd',
@@ -100,7 +100,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/mac-${{arch}}',
-        'version': 'git_revision:d22fa2d126d8e4db585e9db0f3fb2fb4e0bf601b',
+        'version': 'git_revision:335219e2893b5a3b30ec8ce27d48ec762c7de1d5',
       }
     ],
     'dep_type': 'cipd',
@@ -110,7 +110,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/windows-amd64',
-        'version': 'git_revision:d22fa2d126d8e4db585e9db0f3fb2fb4e0bf601b',
+        'version': 'git_revision:335219e2893b5a3b30ec8ce27d48ec762c7de1d5',
       }
     ],
     'dep_type': 'cipd',
@@ -298,31 +298,31 @@ deps = {
     'bucket': 'chromium-browser-clang',
     'objects': [
       {
-        'object_name': 'Linux_x64/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-2-llvmorg-23-init-5669-g8a0be0bc.tar.xz',
-        'sha256sum': '71ea8d8f981d1093ac30e884aefca789962a2eb6e0c31dcddc6fdc83a33c5bd6',
-        'size_bytes': 270756972,
-        'generation': 1772730062708394,
+        'object_name': 'Linux_x64/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-7-llvmorg-23-init-5669-g8a0be0bc.tar.xz',
+        'sha256sum': 'afbb00d27b8f9f65e6a754fb21e80dff084993285cf7f3c0020dece59c5bd67a',
+        'size_bytes': 271641712,
+        'generation': 1773769777991797,
         'condition': 'host_os == "linux" and non_git_source',
       },
       {
-        'object_name': 'Mac/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-2-llvmorg-23-init-5669-g8a0be0bc.tar.xz',
-        'sha256sum': '434ada87593ad29df8eb6bb266d37edcdc7380d1bafbf907f7d39c22d5739084',
-        'size_bytes': 258367196,
-        'generation': 1772730064822309,
+        'object_name': 'Mac/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-7-llvmorg-23-init-5669-g8a0be0bc.tar.xz',
+        'sha256sum': '70b86e82f1cb55777d40b5828ddcb80afea49510085290424b61251d22e9f959',
+        'size_bytes': 259443552,
+        'generation': 1773769780408342,
         'condition': 'host_os == "mac" and host_cpu == "x64"',
       },
       {
-        'object_name': 'Mac_arm64/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-2-llvmorg-23-init-5669-g8a0be0bc.tar.xz',
-        'sha256sum': 'c8e89b991e325f8bfd64fc7a04cfa0eb0b7a953af01fc0a23f0a4b1c1dc34280',
-        'size_bytes': 241717196,
-        'generation': 1772730066996695,
+        'object_name': 'Mac_arm64/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-7-llvmorg-23-init-5669-g8a0be0bc.tar.xz',
+        'sha256sum': 'e2e19684f31b653ce9238f6303aec22576085528c294757a7157d4ab5e1926dc',
+        'size_bytes': 242768940,
+        'generation': 1773769782590875,
         'condition': 'host_os == "mac" and host_cpu == "arm64"',
       },
       {
-        'object_name': 'Win/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-2-llvmorg-23-init-5669-g8a0be0bc.tar.xz',
-        'sha256sum': '2a2463ffc005f0c98be420203bd5a0d5b54b1b6669bb2a9d491cfcab2338f851',
-        'size_bytes': 409485148,
-        'generation': 1772730069048739,
+        'object_name': 'Win/rust-toolchain-6f54d591c3116ee7f8ce9321ddeca286810cc142-7-llvmorg-23-init-5669-g8a0be0bc.tar.xz',
+        'sha256sum': '37dd250549fed5a9765c3a88e3487409189e0c9c63b691fc77daa0b5f214bced',
+        'size_bytes': 409536908,
+        'generation': 1773769784773096,
         'condition': 'host_os == "win"',
       },
     ],
@@ -492,7 +492,7 @@ deps = {
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@8be0e3114685fcc1589561067282edf75ea1259a',
   'src/third_party/catapult':
-    'https://chromium.googlesource.com/catapult.git@df4f8609244fd2378c322760c5172cb3d4fd8725',
+    'https://chromium.googlesource.com/catapult.git@a41fdb3aa561b3ae158e35ecbf44f4a14ff1e934',
   'src/third_party/ced/src': {
     'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5',
   },
@@ -657,7 +657,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/r8',
-              'version': 'CQBWQVHNMmNBTkKhqA-Rnel-94ivC4c9jRROa8bvIUkC',
+              'version': '8ZRb6CCpZTU5dSpQyDlbusalGCjWV0sVSGTq_0Js3mcC',
           },
       ],
       'condition': 'checkout_android',
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@1e4b1454662363af593e8350b596372bc9352f8f',
+    'https://chromium.googlesource.com/chromium/src/tools@a6b6df6c66bde4514eabb56f7f10c5eeb13b3ce1',
 
   'src/third_party/espresso': {
       'packages': [
@@ -786,7 +786,7 @@ deps = {
   },
 
   'src/third_party/tflite/src':
-    Var('chromium_git') + '/external/github.com/tensorflow/tensorflow.git' + '@' + '24d66fbe6c5e87d291207494e4e83d39de3f7d90',
+    Var('chromium_git') + '/external/github.com/tensorflow/tensorflow.git' + '@' + '4e546f69670b48c230e2a2f79ccccd52e5920a01',
 
   'src/third_party/turbine/cipd': {
       'packages': [
@@ -852,7 +852,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/google/pthreadpool.git' + '@' + '9003ee6c137cea3b94161bd5c614fb43be523ee1',
 
   'src/third_party/xnnpack/src':
-    Var('chromium_git') + '/external/github.com/google/XNNPACK.git' + '@' + 'ee91cc745bc715bfa38c5e8241aeb435ca59f433',
+    Var('chromium_git') + '/external/github.com/google/XNNPACK.git' + '@' + 'f1a5f31a23b9a0f5ccf027852731b11d1d1115d0',
 
   'src/third_party/farmhash/src':
     Var('chromium_git') + '/external/github.com/google/farmhash.git' + '@' + '816a4ae622e964763ca0862d9dbd19324a1eaf45',
@@ -864,7 +864,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7607ca500436b37ad23fb8d18614bec7796b68a7',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '0fd8002b11852969d5a8625dd151c085bf44af57',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '05295a818b7e217817ef6cec7e0170870670ca14',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From af098aadc540d8c42230969e78f0cacd5e295635 Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Wed, 18 Mar 2026 21:08:49 -0700
Subject: [PATCH 345/472] Update WebRTC code version (2026-03-19T04:08:14).

Bug: None
Change-Id: I36125b41aca06aa11880463e6cbbb4f3dcd5abb0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/458200
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47198}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index e361931b2e..747c58667b 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-18T04:02:27";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-19T04:08:14";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From da87ea8bbd4bcc66bd24b81dd1bb45267e04e5bd Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Wed, 18 Mar 2026 22:51:31 +0100
Subject: [PATCH 346/472] Move ScopedOperationsBatcher to its own source files

Extract the ScopedOperationsBatcher class from sdp_offer_answer.cc into
a new, dedicated library module. This change improves code
organization by isolating the batching logic, which handles
synchronous task execution on the worker thread and optional
callback tasks on the calling thread.

As part of this refactoring:
* Created scoped_operations_batcher.h and .cc.
* Defined a new rtc_library target in pc/BUILD.gn.
* Added unit tests in scoped_operations_batcher_unittest.cc to
  verify task execution on both worker and signaling threads.
* Updated sdp_offer_answer.cc to include the new header.

Bug: webrtc:42222804
Change-Id: I090c1614917e83a3e3d9543991f1fa2dc4cf22f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/454305
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47199}
---
 pc/BUILD.gn                              | 16 +++++
 pc/scoped_operations_batcher.cc          | 68 +++++++++++++++++++++
 pc/scoped_operations_batcher.h           | 54 ++++++++++++++++
 pc/scoped_operations_batcher_unittest.cc | 78 ++++++++++++++++++++++++
 pc/sdp_offer_answer.cc                   | 66 +-------------------
 5 files changed, 217 insertions(+), 65 deletions(-)
 create mode 100644 pc/scoped_operations_batcher.cc
 create mode 100644 pc/scoped_operations_batcher.h
 create mode 100644 pc/scoped_operations_batcher_unittest.cc

diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 03fabe9df5..2755d73724 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -1069,6 +1069,19 @@ rtc_library("rtc_stats_traversal") {
   ]
 }
 
+rtc_library("scoped_operations_batcher") {
+  visibility = [ ":*" ]
+  sources = [
+    "scoped_operations_batcher.cc",
+    "scoped_operations_batcher.h",
+  ]
+  deps = [
+    "../rtc_base:checks",
+    "../rtc_base:threading",
+    "//third_party/abseil-cpp/absl/functional:any_invocable",
+  ]
+}
+
 rtc_library("sdp_munging_detector") {
   visibility = [ ":*" ]
   sources = [
@@ -1121,6 +1134,7 @@ rtc_library("sdp_offer_answer") {
     ":rtp_sender_proxy",
     ":rtp_transceiver",
     ":rtp_transmission_manager",
+    ":scoped_operations_batcher",
     ":sdp_munging_detector",
     ":sdp_payload_type_suggester",
     ":sdp_state_provider",
@@ -2502,6 +2516,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "rtp_parameters_conversion_unittest.cc",
       "rtp_sender_receiver_unittest.cc",
       "rtp_transceiver_unittest.cc",
+      "scoped_operations_batcher_unittest.cc",
       "sctp_utils_unittest.cc",
       "sdp_munging_detector_unittest.cc",
       "sdp_offer_answer_unittest.cc",
@@ -2552,6 +2567,7 @@ if (rtc_include_tests && !build_with_chromium) {
       ":rtp_transceiver",
       ":rtp_transport",
       ":rtp_transport_internal",
+      ":scoped_operations_batcher",
       ":sctp_data_channel",
       ":sctp_transport",
       ":sctp_utils",
diff --git a/pc/scoped_operations_batcher.cc b/pc/scoped_operations_batcher.cc
new file mode 100644
index 0000000000..050d0c37a3
--- /dev/null
+++ b/pc/scoped_operations_batcher.cc
@@ -0,0 +1,68 @@
+/*
+ *  Copyright 2026 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "pc/scoped_operations_batcher.h"
+
+#include <utility>
+#include <vector>
+
+#include "absl/functional/any_invocable.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/thread.h"
+
+namespace webrtc {
+
+ScopedOperationsBatcher::ScopedOperationsBatcher(Thread* worker_thread)
+    : worker_thread_(worker_thread) {
+  RTC_DCHECK(worker_thread_);
+}
+
+ScopedOperationsBatcher::~ScopedOperationsBatcher() {
+  Run();
+}
+
+void ScopedOperationsBatcher::Run() {
+  std::vector<absl::AnyInvocable<void() &&>> signaling_tasks;
+  if (!tasks_.empty()) {
+    worker_thread_->BlockingCall([&] {
+      for (auto& task : tasks_) {
+        if (task.void_task) {
+          std::move(task.void_task)();
+        } else {
+          RTC_DCHECK(task.returning_task);
+          auto ret = std::move(task.returning_task)();
+          if (ret) {
+            signaling_tasks.push_back(std::move(ret));
+          }
+        }
+      }
+    });
+    tasks_.clear();
+  }
+
+  for (auto& task : signaling_tasks) {
+    std::move(task)();
+  }
+}
+
+void ScopedOperationsBatcher::push_back(absl::AnyInvocable<void() &&> task) {
+  if (task) {
+    tasks_.push_back({.void_task = std::move(task)});
+  }
+}
+
+void ScopedOperationsBatcher::push_back(
+    absl::AnyInvocable<absl::AnyInvocable<void() &&>() &&> task) {
+  if (task) {
+    tasks_.push_back({.returning_task = std::move(task)});
+  }
+}
+
+}  // namespace webrtc
diff --git a/pc/scoped_operations_batcher.h b/pc/scoped_operations_batcher.h
new file mode 100644
index 0000000000..d9a4c23157
--- /dev/null
+++ b/pc/scoped_operations_batcher.h
@@ -0,0 +1,54 @@
+/*
+ *  Copyright 2026 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef PC_SCOPED_OPERATIONS_BATCHER_H_
+#define PC_SCOPED_OPERATIONS_BATCHER_H_
+
+#include <vector>
+
+#include "absl/functional/any_invocable.h"
+#include "rtc_base/thread.h"
+
+namespace webrtc {
+
+// Batches operations to be executed synchronously on the worker thread.
+//
+// When the batcher goes out of scope (or `Run()` is explicitly called), it
+// executes all queued tasks in a single `BlockingCall` to the worker thread.
+//
+// Tasks can either have a `void` return type, or return a new task.
+// Any tasks returned by the executed worker thread tasks are collected
+// and subsequently executed on the calling thread (typically the signaling
+// thread) after the worker thread operations have completed.
+class ScopedOperationsBatcher {
+ public:
+  explicit ScopedOperationsBatcher(Thread* worker_thread);
+  ~ScopedOperationsBatcher();
+
+  void Run();
+
+  // Queues non-nullptr tasks to be executed on the worker when the
+  // ScopedOperationsBatcher goes out of scope.
+  void push_back(absl::AnyInvocable<void() &&> task);
+  void push_back(absl::AnyInvocable<absl::AnyInvocable<void() &&>() &&> task);
+
+ private:
+  struct BatchedTask {
+    absl::AnyInvocable<void() &&> void_task;
+    absl::AnyInvocable<absl::AnyInvocable<void() &&>() &&> returning_task;
+  };
+
+  Thread* const worker_thread_;
+  std::vector<BatchedTask> tasks_;
+};
+
+}  // namespace webrtc
+
+#endif  // PC_SCOPED_OPERATIONS_BATCHER_H_
diff --git a/pc/scoped_operations_batcher_unittest.cc b/pc/scoped_operations_batcher_unittest.cc
new file mode 100644
index 0000000000..7f15ad300f
--- /dev/null
+++ b/pc/scoped_operations_batcher_unittest.cc
@@ -0,0 +1,78 @@
+/*
+ *  Copyright 2026 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "pc/scoped_operations_batcher.h"
+
+#include <memory>
+#include <utility>
+
+#include "absl/functional/any_invocable.h"
+#include "rtc_base/thread.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+TEST(ScopedOperationsBatcherTest, ExecutesTasksOnWorkerThread) {
+  auto worker_thread = Thread::Create();
+  worker_thread->Start();
+
+  bool task_executed = false;
+  bool worker_checked = false;
+
+  {
+    ScopedOperationsBatcher batcher(worker_thread.get());
+    batcher.push_back([&] {
+      task_executed = true;
+      worker_checked = worker_thread->IsCurrent();
+    });
+  }
+
+  EXPECT_TRUE(task_executed);
+  EXPECT_TRUE(worker_checked);
+}
+
+TEST(ScopedOperationsBatcherTest, ExecutesReturnedTasksOnCallingThread) {
+  // Use current thread as the calling (signaling) thread. This test runs
+  // without an explicit RunLoop, because the returned tasks are executed
+  // synchronously within the caller's context of `Run()` or
+  // `~ScopedOperationsBatcher()`.
+  auto signaling_thread = Thread::Current();
+
+  auto worker_thread = Thread::Create();
+  worker_thread->Start();
+
+  bool return_task_executed = false;
+  Thread* return_task_thread = nullptr;
+  bool task_executed = false;
+  Thread* task_thread = nullptr;
+
+  {
+    ScopedOperationsBatcher batcher(worker_thread.get());
+    absl::AnyInvocable<absl::AnyInvocable<void() &&>() &&> task =
+        [&]() -> absl::AnyInvocable<void() &&> {
+      task_executed = true;
+      task_thread = Thread::Current();
+      return [&]() {
+        return_task_executed = true;
+        return_task_thread = Thread::Current();
+      };
+    };
+    batcher.push_back(std::move(task));
+  }
+
+  EXPECT_TRUE(task_executed);
+  EXPECT_EQ(task_thread, worker_thread.get());
+  EXPECT_TRUE(return_task_executed);
+  EXPECT_EQ(return_task_thread, signaling_thread);
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc
index 24abb0daf7..3f5c83fb8d 100644
--- a/pc/sdp_offer_answer.cc
+++ b/pc/sdp_offer_answer.cc
@@ -86,6 +86,7 @@
 #include "pc/rtp_sender_proxy.h"
 #include "pc/rtp_transceiver.h"
 #include "pc/rtp_transmission_manager.h"
+#include "pc/scoped_operations_batcher.h"
 #include "pc/sdp_munging_detector.h"
 #include "pc/session_description.h"
 #include "pc/simulcast_description.h"
@@ -112,71 +113,6 @@
 namespace webrtc {
 namespace {
 
-// Batches operations to be executed synchronously on the worker thread.
-//
-// When the batcher goes out of scope (or `Run()` is explicitly called), it
-// executes all queued tasks in a single `BlockingCall` to the worker thread.
-//
-// Tasks can either have a `void` return type, or return a new task.
-// Any tasks returned by the executed worker thread tasks are collected
-// and subsequently executed on the calling thread (typically the signaling
-// thread) after the worker thread operations have completed.
-class ScopedOperationsBatcher {
- public:
-  explicit ScopedOperationsBatcher(Thread* worker_thread)
-      : worker_thread_(worker_thread) {
-    RTC_DCHECK(worker_thread_);
-  }
-
-  ~ScopedOperationsBatcher() { Run(); }
-
-  void Run() {
-    std::vector<absl::AnyInvocable<void() &&>> signaling_tasks;
-    if (!tasks_.empty()) {
-      worker_thread_->BlockingCall([&] {
-        for (auto& task : tasks_) {
-          if (task.void_task) {
-            std::move(task.void_task)();
-          } else {
-            RTC_DCHECK(task.returning_task);
-            auto ret = std::move(task.returning_task)();
-            if (ret) {
-              signaling_tasks.push_back(std::move(ret));
-            }
-          }
-        }
-      });
-      tasks_.clear();
-    }
-
-    for (auto& task : signaling_tasks) {
-      std::move(task)();
-    }
-  }
-
-  // Queues non-nullptr tasks to be executed on the worker when the
-  // ScopedOperationsBatcher goes out of scope.
-  void push_back(absl::AnyInvocable<void() &&> task) {
-    if (task) {
-      tasks_.push_back({.void_task = std::move(task)});
-    }
-  }
-
-  void push_back(absl::AnyInvocable<absl::AnyInvocable<void() &&>() &&> task) {
-    if (task) {
-      tasks_.push_back({.returning_task = std::move(task)});
-    }
-  }
-
- private:
-  struct BatchedTask {
-    absl::AnyInvocable<void() &&> void_task;
-    absl::AnyInvocable<absl::AnyInvocable<void() &&>() &&> returning_task;
-  };
-
-  Thread* const worker_thread_;
-  std::vector<BatchedTask> tasks_;
-};
 
 struct DtlsTransportAndName {
   scoped_refptr<DtlsTransport> transport;

From 5d1521d9fc2345718f5287e99c3ca08696e6544d Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Thu, 19 Mar 2026 03:04:02 -0700
Subject: [PATCH 347/472] Roll chromium_revision c60f7fb998..ae3e9c7e15
 (1601697:1601850)

Change log: https://chromium.googlesource.com/chromium/src/+log/c60f7fb998..ae3e9c7e15
Full diff: https://chromium.googlesource.com/chromium/src/+/c60f7fb998..ae3e9c7e15

Changed dependencies
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/74494157ca..288a1898c9
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/549deb2670..1c4b46273d
* src/third_party/androidx/cipd: 6c2939gs3hJpDeyKGdMRo_u5z0rsDjmulzNMIQzYLyAC..rEXI3HS2jun77b9O-v3z5Znvw5VGqcohCuE6DTm5hgIC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/c9adcbb559..4dd559a6b0
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/8e07900125..983ff6c520
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/05295a818b..36ca36d0de
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/a6b6df6c66..626cf2a6ca
DEPS diff: https://chromium.googlesource.com/chromium/src/+/c60f7fb998..ae3e9c7e15/DEPS

BUG=None

Change-Id: I8a56d665a3d95a6ea21774c3692dca6d2ca24634
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/458220
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47200}
---
 DEPS | 16 ++++++++--------
 1 file changed, 8 insertions(+), 8 deletions(-)

diff --git a/DEPS b/DEPS
index eca2389c46..5e94e4e201 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': 'c60f7fb9989b3ebd09409d614f219c3c8694d663',
+  'chromium_revision': 'ae3e9c7e15a5aa1fbb98d40e06fae45eec238004',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@74494157cad1e80e4ebae6f3ef431f2de02444c9',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@288a1898c9839a28449fae8f105c506015a5a0cf',
     'condition': 'checkout_ios',
   },
   'src/testing':
     'https://chromium.googlesource.com/chromium/src/testing@7a7d159e99818292ce6b17466c89952cc79e2cc3',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@549deb2670a0662c14c4ac47a4c5fb8927b4b6e6',
+    'https://chromium.googlesource.com/chromium/src/third_party@1c4b46273d2c386dffffc86e579b10fb6e85cd38',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -488,7 +488,7 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@c9adcbb55901715856a6aed27a2f15b5f91680b9',
+    'https://boringssl.googlesource.com/boringssl.git@4dd559a6b088c9e8704fcd9b227599be8a228f4d',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@8be0e3114685fcc1589561067282edf75ea1259a',
   'src/third_party/catapult':
@@ -505,7 +505,7 @@ deps = {
   'src/third_party/crc32c/src':
     'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558',
   'src/third_party/depot_tools':
-    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@8e07900125ba28f056008bded80053b297837aab',
+    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@983ff6c520f9703262d0c75d8038f2e6b9b0c701',
   'src/third_party/ffmpeg':
     'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@946d97db8d906277085e361892b7efda5152e2f1',
   'src/third_party/flatbuffers/src':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@a6b6df6c66bde4514eabb56f7f10c5eeb13b3ce1',
+    'https://chromium.googlesource.com/chromium/src/tools@626cf2a6ca75fff49c342ae34d7ef6a319032abd',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': '6c2939gs3hJpDeyKGdMRo_u5z0rsDjmulzNMIQzYLyAC',
+          'version': 'rEXI3HS2jun77b9O-v3z5Znvw5VGqcohCuE6DTm5hgIC',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -864,7 +864,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7607ca500436b37ad23fb8d18614bec7796b68a7',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '05295a818b7e217817ef6cec7e0170870670ca14',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '36ca36d0def87e3cf10fb0b3d75c6be87b3d3698',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From fa178bed0e9cfa955f0ea7e036a9ce257f6a8721 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Thu, 19 Mar 2026 11:55:06 +0100
Subject: [PATCH 348/472] Add yielding support to ScopedOperationsBatcher

Also including a change for the internal BatchedTask struct and use
std::variant instead as suggested in a previous CL.

Bug: webrtc:42222804
Change-Id: Ia8d10cb9ee6c0a735420232aeca82fb6d85233ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/454940
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47201}
---
 pc/BUILD.gn                              |  2 +
 pc/scoped_operations_batcher.cc          | 52 ++++++++++++++++--------
 pc/scoped_operations_batcher.h           | 26 ++++++++----
 pc/scoped_operations_batcher_unittest.cc | 47 ++++++++++++++++-----
 4 files changed, 91 insertions(+), 36 deletions(-)

diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 2755d73724..88287d5ea9 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -1076,8 +1076,10 @@ rtc_library("scoped_operations_batcher") {
     "scoped_operations_batcher.h",
   ]
   deps = [
+    "../api:sequence_checker",
     "../rtc_base:checks",
     "../rtc_base:threading",
+    "../rtc_base/system:no_unique_address",
     "//third_party/abseil-cpp/absl/functional:any_invocable",
   ]
 }
diff --git a/pc/scoped_operations_batcher.cc b/pc/scoped_operations_batcher.cc
index 050d0c37a3..00c19cfe85 100644
--- a/pc/scoped_operations_batcher.cc
+++ b/pc/scoped_operations_batcher.cc
@@ -10,18 +10,21 @@
 
 #include "pc/scoped_operations_batcher.h"
 
+#include <cstddef>
 #include <utility>
+#include <variant>
 #include <vector>
 
 #include "absl/functional/any_invocable.h"
+#include "api/sequence_checker.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/thread.h"
 
 namespace webrtc {
 
-ScopedOperationsBatcher::ScopedOperationsBatcher(Thread* worker_thread)
-    : worker_thread_(worker_thread) {
-  RTC_DCHECK(worker_thread_);
+ScopedOperationsBatcher::ScopedOperationsBatcher(Thread* target_thread)
+    : target_thread_(target_thread) {
+  RTC_DCHECK(target_thread_);
 }
 
 ScopedOperationsBatcher::~ScopedOperationsBatcher() {
@@ -29,39 +32,56 @@ ScopedOperationsBatcher::~ScopedOperationsBatcher() {
 }
 
 void ScopedOperationsBatcher::Run() {
-  std::vector<absl::AnyInvocable<void() &&>> signaling_tasks;
-  if (!tasks_.empty()) {
-    worker_thread_->BlockingCall([&] {
-      for (auto& task : tasks_) {
-        if (task.void_task) {
-          std::move(task.void_task)();
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  std::vector<absl::AnyInvocable<void() &&>> return_tasks;
+
+  size_t task_idx = 0;
+  bool target_thread_is_current = target_thread_->IsCurrent();
+
+  while (task_idx < tasks_.size()) {
+    target_thread_->BlockingCall([&] {
+      while (task_idx < tasks_.size()) {
+        if (auto* void_task =
+                std::get_if<absl::AnyInvocable<void() &&>>(&tasks_[task_idx])) {
+          std::move (*void_task)();
         } else {
-          RTC_DCHECK(task.returning_task);
-          auto ret = std::move(task.returning_task)();
+          auto* returning_task = std::get_if<
+              absl::AnyInvocable<absl::AnyInvocable<void() &&>() &&>>(
+              &tasks_[task_idx]);
+          RTC_DCHECK(returning_task);
+          auto ret = std::move(*returning_task)();
           if (ret) {
-            signaling_tasks.push_back(std::move(ret));
+            return_tasks.push_back(std::move(ret));
           }
         }
+        ++task_idx;
+        if (!target_thread_is_current && target_thread_->IsYieldRequested()) {
+          return;
+        }
       }
     });
-    tasks_.clear();
   }
 
-  for (auto& task : signaling_tasks) {
+  RTC_DCHECK_EQ(task_idx, tasks_.size());
+  tasks_.clear();
+
+  for (auto& task : return_tasks) {
     std::move(task)();
   }
 }
 
 void ScopedOperationsBatcher::push_back(absl::AnyInvocable<void() &&> task) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
   if (task) {
-    tasks_.push_back({.void_task = std::move(task)});
+    tasks_.emplace_back(std::move(task));
   }
 }
 
 void ScopedOperationsBatcher::push_back(
     absl::AnyInvocable<absl::AnyInvocable<void() &&>() &&> task) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
   if (task) {
-    tasks_.push_back({.returning_task = std::move(task)});
+    tasks_.emplace_back(std::move(task));
   }
 }
 
diff --git a/pc/scoped_operations_batcher.h b/pc/scoped_operations_batcher.h
index d9a4c23157..f5c32c1a74 100644
--- a/pc/scoped_operations_batcher.h
+++ b/pc/scoped_operations_batcher.h
@@ -11,17 +11,25 @@
 #ifndef PC_SCOPED_OPERATIONS_BATCHER_H_
 #define PC_SCOPED_OPERATIONS_BATCHER_H_
 
+#include <variant>
 #include <vector>
 
 #include "absl/functional/any_invocable.h"
+#include "api/sequence_checker.h"
+#include "rtc_base/system/no_unique_address.h"
 #include "rtc_base/thread.h"
 
 namespace webrtc {
 
-// Batches operations to be executed synchronously on the worker thread.
+// Batches operations to be executed synchronously on a target thread.
+//
+// ScopedOperationsBatcher must only be created on the signaling thread.
 //
 // When the batcher goes out of scope (or `Run()` is explicitly called), it
-// executes all queued tasks in a single `BlockingCall` to the worker thread.
+// executes all queued tasks in one or more `BlockingCall`s to the target
+// thread. If the target thread requests a yield during batch execution, the
+// batcher will cooperatively yield the thread and resume execution in a
+// subsequent `BlockingCall`.
 //
 // Tasks can either have a `void` return type, or return a new task.
 // Any tasks returned by the executed worker thread tasks are collected
@@ -29,23 +37,23 @@ namespace webrtc {
 // thread) after the worker thread operations have completed.
 class ScopedOperationsBatcher {
  public:
-  explicit ScopedOperationsBatcher(Thread* worker_thread);
+  explicit ScopedOperationsBatcher(Thread* target_thread);
   ~ScopedOperationsBatcher();
 
   void Run();
 
-  // Queues non-nullptr tasks to be executed on the worker when the
+  // Queues non-nullptr tasks to be executed on the target thread when the
   // ScopedOperationsBatcher goes out of scope.
   void push_back(absl::AnyInvocable<void() &&> task);
   void push_back(absl::AnyInvocable<absl::AnyInvocable<void() &&>() &&> task);
 
  private:
-  struct BatchedTask {
-    absl::AnyInvocable<void() &&> void_task;
-    absl::AnyInvocable<absl::AnyInvocable<void() &&>() &&> returning_task;
-  };
+  using BatchedTask =
+      std::variant<absl::AnyInvocable<void() &&>,
+                   absl::AnyInvocable<absl::AnyInvocable<void() &&>() &&>>;
 
-  Thread* const worker_thread_;
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker sequence_checker_;
+  Thread* const target_thread_;
   std::vector<BatchedTask> tasks_;
 };
 
diff --git a/pc/scoped_operations_batcher_unittest.cc b/pc/scoped_operations_batcher_unittest.cc
index 7f15ad300f..0f188f7c11 100644
--- a/pc/scoped_operations_batcher_unittest.cc
+++ b/pc/scoped_operations_batcher_unittest.cc
@@ -12,6 +12,7 @@
 
 #include <memory>
 #include <utility>
+#include <vector>
 
 #include "absl/functional/any_invocable.h"
 #include "rtc_base/thread.h"
@@ -20,23 +21,23 @@
 namespace webrtc {
 namespace {
 
-TEST(ScopedOperationsBatcherTest, ExecutesTasksOnWorkerThread) {
-  auto worker_thread = Thread::Create();
-  worker_thread->Start();
+TEST(ScopedOperationsBatcherTest, ExecutesTasksOnTargetThread) {
+  auto target_thread = Thread::Create();
+  target_thread->Start();
 
   bool task_executed = false;
-  bool worker_checked = false;
+  bool target_checked = false;
 
   {
-    ScopedOperationsBatcher batcher(worker_thread.get());
+    ScopedOperationsBatcher batcher(target_thread.get());
     batcher.push_back([&] {
       task_executed = true;
-      worker_checked = worker_thread->IsCurrent();
+      target_checked = target_thread->IsCurrent();
     });
   }
 
   EXPECT_TRUE(task_executed);
-  EXPECT_TRUE(worker_checked);
+  EXPECT_TRUE(target_checked);
 }
 
 TEST(ScopedOperationsBatcherTest, ExecutesReturnedTasksOnCallingThread) {
@@ -46,8 +47,8 @@ TEST(ScopedOperationsBatcherTest, ExecutesReturnedTasksOnCallingThread) {
   // `~ScopedOperationsBatcher()`.
   auto signaling_thread = Thread::Current();
 
-  auto worker_thread = Thread::Create();
-  worker_thread->Start();
+  auto target_thread = Thread::Create();
+  target_thread->Start();
 
   bool return_task_executed = false;
   Thread* return_task_thread = nullptr;
@@ -55,7 +56,7 @@ TEST(ScopedOperationsBatcherTest, ExecutesReturnedTasksOnCallingThread) {
   Thread* task_thread = nullptr;
 
   {
-    ScopedOperationsBatcher batcher(worker_thread.get());
+    ScopedOperationsBatcher batcher(target_thread.get());
     absl::AnyInvocable<absl::AnyInvocable<void() &&>() &&> task =
         [&]() -> absl::AnyInvocable<void() &&> {
       task_executed = true;
@@ -69,10 +70,34 @@ TEST(ScopedOperationsBatcherTest, ExecutesReturnedTasksOnCallingThread) {
   }
 
   EXPECT_TRUE(task_executed);
-  EXPECT_EQ(task_thread, worker_thread.get());
+  EXPECT_EQ(task_thread, target_thread.get());
   EXPECT_TRUE(return_task_executed);
   EXPECT_EQ(return_task_thread, signaling_thread);
 }
 
+TEST(ScopedOperationsBatcherTest, YieldsToHighPriorityTasks) {
+  auto target_thread = Thread::Create();
+  target_thread->Start();
+
+  std::vector<int> execution_order;
+
+  {
+    ScopedOperationsBatcher batcher(target_thread.get());
+    batcher.push_back([&] { execution_order.push_back(1); });
+    batcher.push_back([&] {
+      execution_order.push_back(2);
+      // Post a high-priority task that should interrupt the batch.
+      target_thread->PostHighPriorityTask(
+          [&] { execution_order.push_back(3); });
+    });
+    batcher.push_back([&] { execution_order.push_back(4); });
+    batcher.push_back([&] { execution_order.push_back(5); });
+  }
+
+  // Expect the high priority task (3) to execute immediately after the task
+  // that posted it (2). The regular tasks (4, 5) should yield.
+  EXPECT_EQ(execution_order, std::vector<int>({1, 2, 3, 4, 5}));
+}
+
 }  // namespace
 }  // namespace webrtc

From 704c29c12c3e8db3c1538254d15a12838f73599a Mon Sep 17 00:00:00 2001
From: Per K <perkj@webrtc.org>
Date: Thu, 19 Mar 2026 11:52:03 +0000
Subject: [PATCH 349/472] Attempt to deflake
 ScreamTest.LinkCapacity2MbpsRtt50msNoEcn
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

See https://ci.chromium.org/ui/p/webrtc/builders/ci/Linux32%20Debug

Bug: webrtc:483936079
Change-Id: I3839da5e118f1848ac6cf7a38dcd84062ceae797
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/458002
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47202}
---
 test/peer_scenario/bwe_integration_tests/scream_test.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/test/peer_scenario/bwe_integration_tests/scream_test.cc b/test/peer_scenario/bwe_integration_tests/scream_test.cc
index b8dfec15cc..de61e23150 100644
--- a/test/peer_scenario/bwe_integration_tests/scream_test.cc
+++ b/test/peer_scenario/bwe_integration_tests/scream_test.cc
@@ -484,7 +484,7 @@ TEST(ScreamTest, MaybeTest(LinkCapacity2MbpsRtt50msNoEcn)) {
   SendMediaTestResult result = SendMediaInOneDirection(std::move(params), s);
   EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
                                               DataRate::KilobitsPerSec(1300),
-                                              DataRate::KilobitsPerSec(2400))));
+                                              DataRate::KilobitsPerSec(2600))));
 }
 
 TEST(ScreamTest, MaybeTest(LinkCapacity2MbpsRtt50msEcn)) {

From c055986891558041817c2c7fffc516f5dca68973 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Henrik=20Bostr=C3=B6m?= <hbos@webrtc.org>
Date: Thu, 19 Mar 2026 13:23:00 +0100
Subject: [PATCH 350/472] Make Evan co-owner of WebRTC stats directories.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Bug: None
Change-Id: Ibdd2e7fe0463ddc8626a87a5fd63e656f920b4e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/458260
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47203}
---
 api/stats/OWNERS | 1 +
 stats/OWNERS     | 1 +
 2 files changed, 2 insertions(+)

diff --git a/api/stats/OWNERS b/api/stats/OWNERS
index 1556231b31..b35697f14b 100644
--- a/api/stats/OWNERS
+++ b/api/stats/OWNERS
@@ -1,3 +1,4 @@
 set noparent
 hbos@webrtc.org
+eshr@webrtc.org
 hta@webrtc.org
diff --git a/stats/OWNERS b/stats/OWNERS
index 7e98070d5d..f1ab0b433a 100644
--- a/stats/OWNERS
+++ b/stats/OWNERS
@@ -1,2 +1,3 @@
 hbos@webrtc.org
+eshr@webrtc.org
 hta@webrtc.org

From dcb0c0d4b977d38c68d5c3a8bcdd25039b777917 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Henrik=20Bostr=C3=B6m?= <hbos@webrtc.org>
Date: Thu, 19 Mar 2026 13:27:45 +0100
Subject: [PATCH 351/472] Make Evan and Guido peer_connection/webrtc_sdp
 co-owners.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Bug: None
Change-Id: If284d998572e31c82439d975cbe8b090c24feb4a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/458280
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47204}
---
 api/OWNERS | 3 +++
 1 file changed, 3 insertions(+)

diff --git a/api/OWNERS b/api/OWNERS
index 5eeb7f6340..0ef7a47c7b 100644
--- a/api/OWNERS
+++ b/api/OWNERS
@@ -6,6 +6,8 @@ tommi@webrtc.org
 deadbeef@webrtc.org
 
 per-file peer_connection*=hbos@webrtc.org
+per-file peer_connection*=eshr@webrtc.org
+per-file peer_connection*=guidou@webrtc.org
 
 per-file DEPS=mbonadei@webrtc.org
 
@@ -15,6 +17,7 @@ per-file uma_metrics.h=kron@webrtc.org
 per-file webrtc_sdp.cc = set noparent
 per-file webrtc_sdp.cc = hta@webrtc.org
 per-file webrtc_sdp.cc = hbos@webrtc.org
+per-file webrtc_sdp.cc = eshr@webrtc.org
 # If none of the above are present, may also try one of these
 per-file webrtc_sdp.cc = tommi@webrtc.org
 per-file webrtc_sdp.cc = guidou@webrtc.org

From 9fd0fd9c44c933392d13cf844b0ee2cb4d780df8 Mon Sep 17 00:00:00 2001
From: Evan Shrubsole <eshr@webrtc.org>
Date: Mon, 16 Mar 2026 11:32:22 +0000
Subject: [PATCH 352/472] Remove unused ScopedFakeClock from
 packet_router_unittest
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This was unused for these tests and passes without it.

Bug: webrtc:42223992
Change-Id: Id19b1d7886a0527d49c2f4c30b947b486a6a6964
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/457581
Reviewed-by: Erik Språng <sprang@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47205}
---
 modules/pacing/packet_router_unittest.cc | 12 ------------
 1 file changed, 12 deletions(-)

diff --git a/modules/pacing/packet_router_unittest.cc b/modules/pacing/packet_router_unittest.cc
index ee853294ff..d0143c23ac 100644
--- a/modules/pacing/packet_router_unittest.cc
+++ b/modules/pacing/packet_router_unittest.cc
@@ -20,7 +20,6 @@
 #include "api/rtp_headers.h"
 #include "api/transport/network_types.h"
 #include "api/units/data_size.h"
-#include "api/units/time_delta.h"
 #include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
@@ -29,7 +28,6 @@
 #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/fake_clock.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
 
@@ -676,7 +674,6 @@ TEST_F(PacketRouterTest, DuplicateRemovalOfSendModuleIgnored) {
 }
 
 TEST(PacketRouterRembTest, ChangeSendRtpModuleChangeRembSender) {
-  ScopedFakeClock clock;
   NiceMock<MockRtpRtcpInterface> rtp_send;
   NiceMock<MockRtpRtcpInterface> rtp_recv;
   PacketRouter packet_router;
@@ -701,7 +698,6 @@ TEST(PacketRouterRembTest, ChangeSendRtpModuleChangeRembSender) {
 // Only register receiving modules and make sure we fallback to trigger a REMB
 // packet on this one.
 TEST(PacketRouterRembTest, NoSendingRtpModule) {
-  ScopedFakeClock clock;
   NiceMock<MockRtpRtcpInterface> rtp;
   PacketRouter packet_router;
 
@@ -722,7 +718,6 @@ TEST(PacketRouterRembTest, NoSendingRtpModule) {
 }
 
 TEST(PacketRouterRembTest, NonCandidateSendRtpModuleNotUsedForRemb) {
-  ScopedFakeClock clock;
   PacketRouter packet_router;
   NiceMock<MockRtpRtcpInterface> module;
 
@@ -740,7 +735,6 @@ TEST(PacketRouterRembTest, NonCandidateSendRtpModuleNotUsedForRemb) {
 }
 
 TEST(PacketRouterRembTest, CandidateSendRtpModuleUsedForRemb) {
-  ScopedFakeClock clock;
   PacketRouter packet_router;
   NiceMock<MockRtpRtcpInterface> module;
 
@@ -758,7 +752,6 @@ TEST(PacketRouterRembTest, CandidateSendRtpModuleUsedForRemb) {
 }
 
 TEST(PacketRouterRembTest, NonCandidateReceiveRtpModuleNotUsedForRemb) {
-  ScopedFakeClock clock;
   PacketRouter packet_router;
   NiceMock<MockRtpRtcpInterface> module;
 
@@ -776,7 +769,6 @@ TEST(PacketRouterRembTest, NonCandidateReceiveRtpModuleNotUsedForRemb) {
 }
 
 TEST(PacketRouterRembTest, CandidateReceiveRtpModuleUsedForRemb) {
-  ScopedFakeClock clock;
   PacketRouter packet_router;
   NiceMock<MockRtpRtcpInterface> module;
 
@@ -795,7 +787,6 @@ TEST(PacketRouterRembTest, CandidateReceiveRtpModuleUsedForRemb) {
 
 TEST(PacketRouterRembTest,
      SendCandidatePreferredOverReceiveCandidate_SendModuleAddedFirst) {
-  ScopedFakeClock clock;
   PacketRouter packet_router;
   NiceMock<MockRtpRtcpInterface> send_module;
   NiceMock<MockRtpRtcpInterface> receive_module;
@@ -822,7 +813,6 @@ TEST(PacketRouterRembTest,
 
 TEST(PacketRouterRembTest,
      SendCandidatePreferredOverReceiveCandidate_ReceiveModuleAddedFirst) {
-  ScopedFakeClock clock;
   PacketRouter packet_router;
   NiceMock<MockRtpRtcpInterface> send_module;
   NiceMock<MockRtpRtcpInterface> receive_module;
@@ -840,7 +830,6 @@ TEST(PacketRouterRembTest,
   EXPECT_CALL(send_module, SetRemb(bitrate_estimate, ssrcs));
   EXPECT_CALL(receive_module, SetRemb(_, _)).Times(0);
 
-  clock.AdvanceTime(TimeDelta::Millis(1000));
   packet_router.SendRemb(bitrate_estimate, ssrcs);
 
   // Test tear-down
@@ -849,7 +838,6 @@ TEST(PacketRouterRembTest,
 }
 
 TEST(PacketRouterRembTest, ReceiveModuleTakesOverWhenLastSendModuleRemoved) {
-  ScopedFakeClock clock;
   PacketRouter packet_router;
   NiceMock<MockRtpRtcpInterface> send_module;
   NiceMock<MockRtpRtcpInterface> receive_module;

From 8609b574457cf5d4f3d0e6a69ac11e6a520d7a52 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Thu, 19 Mar 2026 09:04:07 -0700
Subject: [PATCH 353/472] Roll chromium_revision ae3e9c7e15..19412fd4e2
 (1601850:1602015)

Change log: https://chromium.googlesource.com/chromium/src/+log/ae3e9c7e15..19412fd4e2
Full diff: https://chromium.googlesource.com/chromium/src/+/ae3e9c7e15..19412fd4e2

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/166f3230f6..602e66a04a
* src/buildtools/linux64: git_revision:335219e2893b5a3b30ec8ce27d48ec762c7de1d5..git_revision:3007f57168dbe495a6a0a206c4ac54aca9a0d56f
* src/buildtools/mac: git_revision:335219e2893b5a3b30ec8ce27d48ec762c7de1d5..git_revision:3007f57168dbe495a6a0a206c4ac54aca9a0d56f
* src/buildtools/win: git_revision:335219e2893b5a3b30ec8ce27d48ec762c7de1d5..git_revision:3007f57168dbe495a6a0a206c4ac54aca9a0d56f
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/288a1898c9..f412678617
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/7a7d159e99..89b906b2f0
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/1c4b46273d..89ed82528e
* src/third_party/androidx/cipd: rEXI3HS2jun77b9O-v3z5Znvw5VGqcohCuE6DTm5hgIC..gCm_xSL7tlrL-NzS4-1zwSoJIS6S9wc_1cwdWYRTfx4C
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/4dd559a6b0..4d969991f6
* src/third_party/icu: https://chromium.googlesource.com/chromium/deps/icu.git/+log/ee5f27adc2..b4aae6832c
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/08919e34b6..cc7b51d2e0
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/626cf2a6ca..4376836440
DEPS diff: https://chromium.googlesource.com/chromium/src/+/ae3e9c7e15..19412fd4e2/DEPS

BUG=None

Change-Id: I2c3e14b0c1dc4000c60b0201963c0c49a2fb3e78
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/458321
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47206}
---
 DEPS | 26 +++++++++++++-------------
 1 file changed, 13 insertions(+), 13 deletions(-)

diff --git a/DEPS b/DEPS
index 5e94e4e201..f100e28a67 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': 'ae3e9c7e15a5aa1fbb98d40e06fae45eec238004',
+  'chromium_revision': '19412fd4e28c68d592569c759909a6e90c618f64',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@166f3230f69fdb95cf3da587e6ee143c2c03c971',
+    'https://chromium.googlesource.com/chromium/src/build@602e66a04a728d193d7e120c151b4044da354bb2',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@95ed44cf5f06dbb5861030b91c9db9ccb4316762',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,19 +78,19 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@288a1898c9839a28449fae8f105c506015a5a0cf',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@f4126786174cfbef5d6f6556ba885b0cd3846bed',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@7a7d159e99818292ce6b17466c89952cc79e2cc3',
+    'https://chromium.googlesource.com/chromium/src/testing@89b906b2f0a0a0c6d80fd2a18d3a28e9f39fd2ae',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@1c4b46273d2c386dffffc86e579b10fb6e85cd38',
+    'https://chromium.googlesource.com/chromium/src/third_party@89ed82528ecce2fd9d1ba84214a21adebb0ca7e7',
 
   'src/buildtools/linux64': {
     'packages': [
       {
         'package': 'gn/gn/linux-${{arch}}',
-        'version': 'git_revision:335219e2893b5a3b30ec8ce27d48ec762c7de1d5',
+        'version': 'git_revision:3007f57168dbe495a6a0a206c4ac54aca9a0d56f',
       }
     ],
     'dep_type': 'cipd',
@@ -100,7 +100,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/mac-${{arch}}',
-        'version': 'git_revision:335219e2893b5a3b30ec8ce27d48ec762c7de1d5',
+        'version': 'git_revision:3007f57168dbe495a6a0a206c4ac54aca9a0d56f',
       }
     ],
     'dep_type': 'cipd',
@@ -110,7 +110,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/windows-amd64',
-        'version': 'git_revision:335219e2893b5a3b30ec8ce27d48ec762c7de1d5',
+        'version': 'git_revision:3007f57168dbe495a6a0a206c4ac54aca9a0d56f',
       }
     ],
     'dep_type': 'cipd',
@@ -488,7 +488,7 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@4dd559a6b088c9e8704fcd9b227599be8a228f4d',
+    'https://boringssl.googlesource.com/boringssl.git@4d969991f61733b4cab1ae871b136fd4dd6c6005',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@8be0e3114685fcc1589561067282edf75ea1259a',
   'src/third_party/catapult':
@@ -537,7 +537,7 @@ deps = {
   'src/third_party/googletest/src':
     'https://chromium.googlesource.com/external/github.com/google/googletest.git@4fe3307fb2d9f86d19777c7eb0e4809e9694dde7',
   'src/third_party/icu': {
-    'url': 'https://chromium.googlesource.com/chromium/deps/icu.git@ee5f27adc28bd3f15b2c293f726d14d2e336cbd5',
+    'url': 'https://chromium.googlesource.com/chromium/deps/icu.git@b4aae6832c06df9d538d41b249403cf0678f16b4',
   },
   'src/third_party/jdk/current': {
       'packages': [
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '08919e34b61cbb1950dd20a87f4748d5795fa2a9',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'cc7b51d2e03c05961e2649f0db396f3179dd6def',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@626cf2a6ca75fff49c342ae34d7ef6a319032abd',
+    'https://chromium.googlesource.com/chromium/src/tools@4376836440f3add8199d9848e92e1900ae358867',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'rEXI3HS2jun77b9O-v3z5Znvw5VGqcohCuE6DTm5hgIC',
+          'version': 'gCm_xSL7tlrL-NzS4-1zwSoJIS6S9wc_1cwdWYRTfx4C',
       },
     ],
     'condition': 'checkout_android and non_git_source',

From 1c562d18c74ad3f8cbe0801ed79cb627aba52d30 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Thu, 19 Mar 2026 13:02:30 -0700
Subject: [PATCH 354/472] Roll chromium_revision 19412fd4e2..4ea3a3a5a8
 (1602015:1602184)

Change log: https://chromium.googlesource.com/chromium/src/+log/19412fd4e2..4ea3a3a5a8
Full diff: https://chromium.googlesource.com/chromium/src/+/19412fd4e2..4ea3a3a5a8

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/602e66a04a..66254c179e
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/f412678617..a5750002d8
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/89b906b2f0..e525a52cbd
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/89ed82528e..5093883fdb
* src/third_party/android_build_tools/manifest_merger/cipd: pdCSlKLBAxy6TfAEU6FCxCqA8rgvQ70Vhsodl6VKdKQC..5JSVccMXNkpeH9lpydgxJ3QCoNpBC5yvil7NvdsqUasC
* src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/c09cee4577..59066a6625
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/983ff6c520..9922fe92e1
* src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/390ab7f3cf..a96728b9a9
* src/third_party/tflite/src: https://chromium.googlesource.com/external/github.com/tensorflow/tensorflow.git/+log/4e546f6967..67d816626b
* src/third_party/xnnpack/src: https://chromium.googlesource.com/external/github.com/google/XNNPACK.git/+log/f1a5f31a23..811d0bd388
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/4376836440..446c8572de
DEPS diff: https://chromium.googlesource.com/chromium/src/+/19412fd4e2..4ea3a3a5a8/DEPS

BUG=None

Change-Id: I8f979392d3ba136e31a335f7ff9fd0f9eca05500
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/458420
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47207}
---
 DEPS | 24 ++++++++++++------------
 1 file changed, 12 insertions(+), 12 deletions(-)

diff --git a/DEPS b/DEPS
index f100e28a67..894faa5be2 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '19412fd4e28c68d592569c759909a6e90c618f64',
+  'chromium_revision': '4ea3a3a5a825bc995dca95d3a6b7a3135ccb7134',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@602e66a04a728d193d7e120c151b4044da354bb2',
+    'https://chromium.googlesource.com/chromium/src/build@66254c179efcd5d3f1be45e3b98486b7cf9ada50',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@95ed44cf5f06dbb5861030b91c9db9ccb4316762',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@f4126786174cfbef5d6f6556ba885b0cd3846bed',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@a5750002d8699aac0984f9292a5b2dc5a2cbcfad',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@89b906b2f0a0a0c6d80fd2a18d3a28e9f39fd2ae',
+    'https://chromium.googlesource.com/chromium/src/testing@e525a52cbd91f4c4be059d704541562b285379b3',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@89ed82528ecce2fd9d1ba84214a21adebb0ca7e7',
+    'https://chromium.googlesource.com/chromium/src/third_party@5093883fdb83da3255f9590b9b7d67ea9f08f47a',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -331,13 +331,13 @@ deps = {
   'src/third_party/clang-format/script':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@c2725e0622e1a86d55f14514f2177a39efea4a0e',
   'src/third_party/compiler-rt/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@c09cee4577254cc33ee4a7699bbe3d7ae3ea730a',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@59066a662511e1f9959b07a32c88906d7212b6a7',
   'src/third_party/libc++/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@7ab65651aed6802d2599dcb7a73b1f82d5179d05',
   'src/third_party/libc++abi/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@8f11bb1d4438d0239d0dfc1bd9456a9f31629dda',
   'src/third_party/llvm-libc/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@390ab7f3cffaa613e4a0a9c5aecf37052d5a8923',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@a96728b9a94a732ec71650710949851ddf114845',
   'src/third_party/libunwind/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@db838d918570d4e381ecf9f5cc70a0098c9c2cd6',
 
@@ -505,7 +505,7 @@ deps = {
   'src/third_party/crc32c/src':
     'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558',
   'src/third_party/depot_tools':
-    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@983ff6c520f9703262d0c75d8038f2e6b9b0c701',
+    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@9922fe92e1c5ea0033d04a3eda0dc1fa2f5464e6',
   'src/third_party/ffmpeg':
     'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@946d97db8d906277085e361892b7efda5152e2f1',
   'src/third_party/flatbuffers/src':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@4376836440f3add8199d9848e92e1900ae358867',
+    'https://chromium.googlesource.com/chromium/src/tools@446c8572de969f49fe9a7c32ba5a08fc780bb825',
 
   'src/third_party/espresso': {
       'packages': [
@@ -718,7 +718,7 @@ deps = {
       'packages': [
           {
                'package': 'chromium/third_party/android_build_tools/manifest_merger',
-               'version': 'pdCSlKLBAxy6TfAEU6FCxCqA8rgvQ70Vhsodl6VKdKQC',
+               'version': '5JSVccMXNkpeH9lpydgxJ3QCoNpBC5yvil7NvdsqUasC',
           },
       ],
       'condition': 'checkout_android',
@@ -786,7 +786,7 @@ deps = {
   },
 
   'src/third_party/tflite/src':
-    Var('chromium_git') + '/external/github.com/tensorflow/tensorflow.git' + '@' + '4e546f69670b48c230e2a2f79ccccd52e5920a01',
+    Var('chromium_git') + '/external/github.com/tensorflow/tensorflow.git' + '@' + '67d816626bf264e3b142187768601a200cbe810d',
 
   'src/third_party/turbine/cipd': {
       'packages': [
@@ -852,7 +852,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/google/pthreadpool.git' + '@' + '9003ee6c137cea3b94161bd5c614fb43be523ee1',
 
   'src/third_party/xnnpack/src':
-    Var('chromium_git') + '/external/github.com/google/XNNPACK.git' + '@' + 'f1a5f31a23b9a0f5ccf027852731b11d1d1115d0',
+    Var('chromium_git') + '/external/github.com/google/XNNPACK.git' + '@' + '811d0bd388ba8f4853c5560b6e16a5af3e8b895a',
 
   'src/third_party/farmhash/src':
     Var('chromium_git') + '/external/github.com/google/farmhash.git' + '@' + '816a4ae622e964763ca0862d9dbd19324a1eaf45',

From c0228a1b709eee29e07e2dc09502a6a6b0aeb6a1 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Thu, 19 Mar 2026 18:56:24 +0100
Subject: [PATCH 355/472] Remove deprecated StartSctpTransport method

Bug: none
Change-Id: I55627d991cab4713e13662ea4afd1e637cf48262
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/458264
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47208}
---
 pc/peer_connection_internal.h | 8 --------
 1 file changed, 8 deletions(-)

diff --git a/pc/peer_connection_internal.h b/pc/peer_connection_internal.h
index 4bf70dcf35..8314f7b23e 100644
--- a/pc/peer_connection_internal.h
+++ b/pc/peer_connection_internal.h
@@ -130,14 +130,6 @@ class PeerConnectionSdpMethods {
   // Asynchronously calls SctpTransport::Start() on the network thread for
   // `sctp_mid()` if set. Called as part of setting the local description.
   virtual RTCError StartSctpTransport(const SctpOptions& options) = 0;
-  [[deprecated("Call with SctpOptions")]]
-  virtual void StartSctpTransport(int local_port,
-                                  int remote_port,
-                                  int max_message_size) {
-    StartSctpTransport({.local_port = local_port,
-                        .remote_port = remote_port,
-                        .max_message_size = max_message_size});
-  }
 
   // Asynchronously adds a remote candidate on the network thread.
   virtual void AddRemoteCandidate(absl::string_view mid,

From c8cb3a6c32c9652118668972b1c3e50161e1a229 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Thu, 19 Mar 2026 17:02:56 -0700
Subject: [PATCH 356/472] Roll chromium_revision 4ea3a3a5a8..0939042bb0
 (1602184:1602336)

Change log: https://chromium.googlesource.com/chromium/src/+log/4ea3a3a5a8..0939042bb0
Full diff: https://chromium.googlesource.com/chromium/src/+/4ea3a3a5a8..0939042bb0

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/66254c179e..013cd9d1bc
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/a5750002d8..3d8eb6269a
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/e525a52cbd..53cc79e2fc
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/5093883fdb..ac21dad613
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/4d969991f6..28cf3dd50d
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/9922fe92e1..40b296d214
* src/third_party/fuzztest/src: https://chromium.googlesource.com/external/github.com/google/fuzztest.git/+log/3874258b07..9600119eaa
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/cc7b51d2e0..f9422e2fbe
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/446c8572de..46520ac5d1
DEPS diff: https://chromium.googlesource.com/chromium/src/+/4ea3a3a5a8..0939042bb0/DEPS

BUG=None

Change-Id: I9d12e1fc015523960e14d7887f3b6272e7c1f570
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/458421
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47209}
---
 DEPS | 20 ++++++++++----------
 1 file changed, 10 insertions(+), 10 deletions(-)

diff --git a/DEPS b/DEPS
index 894faa5be2..2eb2484088 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '4ea3a3a5a825bc995dca95d3a6b7a3135ccb7134',
+  'chromium_revision': '0939042bb095b3703f58845b4b510f20a5e674be',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@66254c179efcd5d3f1be45e3b98486b7cf9ada50',
+    'https://chromium.googlesource.com/chromium/src/build@013cd9d1bc97edce5e115f5fbe48a771ef1d89d2',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@95ed44cf5f06dbb5861030b91c9db9ccb4316762',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@a5750002d8699aac0984f9292a5b2dc5a2cbcfad',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@3d8eb6269a2842ee3d652047770ffc2e4d84e9b9',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@e525a52cbd91f4c4be059d704541562b285379b3',
+    'https://chromium.googlesource.com/chromium/src/testing@53cc79e2fca64e4075719222582ca77ff1276bc4',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@5093883fdb83da3255f9590b9b7d67ea9f08f47a',
+    'https://chromium.googlesource.com/chromium/src/third_party@ac21dad613211188b270f27bec07efd00983f2b3',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -488,7 +488,7 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@4d969991f61733b4cab1ae871b136fd4dd6c6005',
+    'https://boringssl.googlesource.com/boringssl.git@28cf3dd50d8a9c8bc810c8fa84138af8da711e97',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@8be0e3114685fcc1589561067282edf75ea1259a',
   'src/third_party/catapult':
@@ -505,7 +505,7 @@ deps = {
   'src/third_party/crc32c/src':
     'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558',
   'src/third_party/depot_tools':
-    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@9922fe92e1c5ea0033d04a3eda0dc1fa2f5464e6',
+    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@40b296d214b2fa92c4605a0244f26c2d824f3996',
   'src/third_party/ffmpeg':
     'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@946d97db8d906277085e361892b7efda5152e2f1',
   'src/third_party/flatbuffers/src':
@@ -593,7 +593,7 @@ deps = {
   'src/third_party/libFuzzer/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt/lib/fuzzer.git@bea408a6e01f0f7e6c82a43121fe3af4506c932e',
   'src/third_party/fuzztest/src':
-    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@3874258b07ad554917c0bac6ea9ee07469af6ac9',
+    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@9600119eaa4684586fef4d0144c0446470f64d87',
   'src/third_party/libjpeg_turbo':
     'https://chromium.googlesource.com/chromium/deps/libjpeg_turbo.git@d1f5f2393e0d51f840207342ae86e55a86443288',
   'src/third_party/libsrtp':
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'cc7b51d2e03c05961e2649f0db396f3179dd6def',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'f9422e2fbe0f815beb2c986d8359a4275385138f',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@446c8572de969f49fe9a7c32ba5a08fc780bb825',
+    'https://chromium.googlesource.com/chromium/src/tools@46520ac5d116e5f79bdb4826ad62cf24bff65d1d',
 
   'src/third_party/espresso': {
       'packages': [

From cb044053134262c356df437c66610fff199487ce Mon Sep 17 00:00:00 2001
From: Jordan <rop@google.com>
Date: Fri, 14 Nov 2025 07:57:59 +0000
Subject: [PATCH 357/472] Fix metadata for Portaudio library for mac

R=mflodman@webrtc.org

Bug: chromium:460543401
Change-Id: I4bac6260a2671942b31acff9933b4cdab988e60a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/425600
Auto-Submit: Jordan Brown <rop@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Jordan Brown <rop@google.com>
Cr-Commit-Position: refs/heads/main@{#47210}
---
 modules/third_party/portaudio/README.chromium | 5 +++--
 1 file changed, 3 insertions(+), 2 deletions(-)

diff --git a/modules/third_party/portaudio/README.chromium b/modules/third_party/portaudio/README.chromium
index ac300bb34a..d7d899c05d 100644
--- a/modules/third_party/portaudio/README.chromium
+++ b/modules/third_party/portaudio/README.chromium
@@ -1,7 +1,8 @@
 Name: Portaudio library for mac
 Short Name: portaudio
-URL: https://github.com/PortAudio/portaudio/tree/master/src/common
-Version: 9d8563100d841300f1689b186d131347ad43a0f6
+URL: https://github.com/PortAudio/portaudio
+Version: N/A
+Revision: 9d8563100d841300f1689b186d131347ad43a0f6
 Date: 2022-04-12
 License: MIT
 License File: LICENSE

From 16e300299573d849124573d0b7894061822c1790 Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Thu, 19 Mar 2026 21:02:24 -0700
Subject: [PATCH 358/472] Update WebRTC code version (2026-03-20T04:01:59).

Bug: None
Change-Id: Ica3acdf2d7cba4f75a6d5d3691374a6b6e3c81db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/458460
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47211}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index 747c58667b..6d4be36a08 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-19T04:08:14";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-20T04:01:59";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From 2fc65d315041e5a99a1833d96aa2128ff89e2cdb Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Thu, 19 Mar 2026 23:06:25 -0700
Subject: [PATCH 359/472] Roll chromium_revision 0939042bb0..d1fc7b8926
 (1602336:1602437)

Change log: https://chromium.googlesource.com/chromium/src/+log/0939042bb0..d1fc7b8926
Full diff: https://chromium.googlesource.com/chromium/src/+/0939042bb0..d1fc7b8926

Changed dependencies
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/3d8eb6269a..20f52cf52a
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/ac21dad613..e61263adf8
* src/third_party/androidx/cipd: gCm_xSL7tlrL-NzS4-1zwSoJIS6S9wc_1cwdWYRTfx4C..HrOnciB8BQGl5MNBfsfgysiPtdI3nERmtEhel7WeC4oC
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/36ca36d0de..3578883bb3
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/c0728bdae7..2c62d7b7a9
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/f9422e2fbe..c9aa1481b5
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/46520ac5d1..50d5e17692
DEPS diff: https://chromium.googlesource.com/chromium/src/+/0939042bb0..d1fc7b8926/DEPS

BUG=None

Change-Id: I8308f97d5cf9f5ea42892aea2fef23c8d5165ff7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/458480
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47212}
---
 DEPS | 16 ++++++++--------
 1 file changed, 8 insertions(+), 8 deletions(-)

diff --git a/DEPS b/DEPS
index 2eb2484088..3a84d03239 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '0939042bb095b3703f58845b4b510f20a5e674be',
+  'chromium_revision': 'd1fc7b8926ecd425bcb2a42b5f690cc60f9854b8',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@3d8eb6269a2842ee3d652047770ffc2e4d84e9b9',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@20f52cf52a7b171aa220240f103b15ff4c3fef45',
     'condition': 'checkout_ios',
   },
   'src/testing':
     'https://chromium.googlesource.com/chromium/src/testing@53cc79e2fca64e4075719222582ca77ff1276bc4',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@ac21dad613211188b270f27bec07efd00983f2b3',
+    'https://chromium.googlesource.com/chromium/src/third_party@e61263adf8c3a8a77d023d89e6d6ce8e9b49ece3',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -511,7 +511,7 @@ deps = {
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@c0728bdae796f214a9fc56a617b21075c2ed3b40',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@2c62d7b7a927db2572b50277ba56eb4a754c43b7',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'f9422e2fbe0f815beb2c986d8359a4275385138f',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'c9aa1481b548e78ce3d6f79eabb6eca0be044e34',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@46520ac5d116e5f79bdb4826ad62cf24bff65d1d',
+    'https://chromium.googlesource.com/chromium/src/tools@50d5e17692834d647a6b47f138480f833d7c392e',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'gCm_xSL7tlrL-NzS4-1zwSoJIS6S9wc_1cwdWYRTfx4C',
+          'version': 'HrOnciB8BQGl5MNBfsfgysiPtdI3nERmtEhel7WeC4oC',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -864,7 +864,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7607ca500436b37ad23fb8d18614bec7796b68a7',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '36ca36d0def87e3cf10fb0b3d75c6be87b3d3698',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '3578883bb3616208b8f9bb6fa03dc1fda0a16a78',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From c63366c187d96353ed69d16a6b3cc9423cbe5abe Mon Sep 17 00:00:00 2001
From: Per K <perkj@webrtc.org>
Date: Fri, 20 Mar 2026 06:53:21 +0000
Subject: [PATCH 360/472] Allow initial BWE probing with Scream without
 configured media
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Similar to Goog CC, allow padding packets to be sent the first 6s seconds, up to the configured max target bitrate, even if no max allocated bitrate per streams is set.
Probing without media must be allowed.

These "probes" are sent every 3s during the first 6s. Ie, if the initial attempt is aborted due to congestion, a second attempt will be made 3s later.


Bug: webrtc:447037083
Change-Id: I8d32e1ac2eebe5cc84142995d2332ffdc954ecb4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/458300
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47213}
---
 .../scream/scream_network_controller.cc       | 70 ++++++++++++-------
 .../scream/scream_network_controller.h        |  6 +-
 .../scream_network_controller_unittest.cc     | 43 ++++++++++--
 .../scream/scream_v2_parameters.cc            |  6 +-
 .../scream/scream_v2_parameters.h             |  8 ++-
 .../bwe_integration_tests/scream_test.cc      |  8 +--
 6 files changed, 97 insertions(+), 44 deletions(-)

diff --git a/modules/congestion_controller/scream/scream_network_controller.cc b/modules/congestion_controller/scream/scream_network_controller.cc
index 396c2be21f..e20e5898e5 100644
--- a/modules/congestion_controller/scream/scream_network_controller.cc
+++ b/modules/congestion_controller/scream/scream_network_controller.cc
@@ -46,8 +46,7 @@ ScreamNetworkController::ScreamNetworkController(NetworkControllerConfig config)
       streams_config_(config.stream_based_config),
       max_seen_total_allocated_bitrate_(
           config.stream_based_config.max_total_allocated_bitrate.value_or(
-              DataRate::Zero())),
-      last_padding_interval_started_(Timestamp::Zero()) {
+              DataRate::Zero())) {
   UpdateScreamTargetBitrateConstraints();
 }
 
@@ -65,6 +64,9 @@ NetworkControlUpdate ScreamNetworkController::CreateFirstUpdate(Timestamp now) {
   RTC_DCHECK(network_available_);
   RTC_DCHECK(!first_update_created_);
   first_update_created_ = true;
+  if (allow_initial_bwe_before_media_) {
+    initial_bwe_probe_end_time_ = now + params_.initial_probing_duration.Get();
+  }
   NetworkControlUpdate update = CreateUpdate(now);
 
   if (allow_initial_bwe_before_media_) {
@@ -87,8 +89,7 @@ NetworkControlUpdate ScreamNetworkController::CreateFirstUpdate(Timestamp now) {
 NetworkControlUpdate ScreamNetworkController::OnNetworkAvailability(
     NetworkAvailability msg) {
   network_available_ = msg.network_available;
-  if (!first_update_created_ && network_available_ &&
-      streams_config_.max_total_allocated_bitrate > DataRate::Zero()) {
+  if (!first_update_created_ && network_available_) {
     return CreateFirstUpdate(msg.at_time);
   }
   return NetworkControlUpdate();
@@ -112,8 +113,9 @@ NetworkControlUpdate ScreamNetworkController::OnNetworkRouteChange(
 
 NetworkControlUpdate ScreamNetworkController::OnProcessInterval(
     ProcessInterval msg) {
-  // Scream currently has no need for periodic processing.
-  return NetworkControlUpdate();
+  NetworkControlUpdate update;
+  update.pacer_config = MaybeCreatePacerConfig(msg.at_time);
+  return update;
 }
 
 NetworkControlUpdate ScreamNetworkController::OnRemoteBitrateReport(
@@ -204,12 +206,13 @@ NetworkControlUpdate ScreamNetworkController::CreateUpdate(Timestamp now) {
     target_rate_msg.network_estimate.bwe_period = TimeDelta::Millis(25);
     update.target_rate = target_rate_msg;
   }
-  update.pacer_config = MaybeCreatePacerConfig();
+  update.pacer_config = MaybeCreatePacerConfig(now);
   update.congestion_window = scream_->max_data_in_flight();
   return update;
 }
 
-std::optional<PacerConfig> ScreamNetworkController::MaybeCreatePacerConfig() {
+std::optional<PacerConfig> ScreamNetworkController::MaybeCreatePacerConfig(
+    Timestamp now) {
   // Allow sending packets in larger bursts if some time has passed since last
   // congestion event.
   TimeDelta pacing_window =
@@ -218,29 +221,42 @@ std::optional<PacerConfig> ScreamNetworkController::MaybeCreatePacerConfig() {
        params_.allow_large_pacing_bursts_after_congestion_time.Get())
           ? default_pacing_window_
           : TimeDelta::Millis(10);
-  DataRate target_rate = scream_->target_rate();
-  Timestamp now = env_.clock().CurrentTime();
-  DataRate padding_rate = DataRate::Zero();
-  // Allow padding if needed. Note that current max needed by streams may be
-  // lower than what the user intended since it depends on video resolution
-  // that may be scaled down due to low quality.
-  DataRate max_padding_rate =
-      std::min({max_target_rate_, max_seen_total_allocated_bitrate_,
-                2 * streams_config_.max_total_allocated_bitrate.value_or(
-                        DataRate::Zero()),
-                remote_bitrate_report_.value_or(DataRate::PlusInfinity())});
-  if (target_rate < max_padding_rate &&
+
+  bool allow_padding = false;
+  if (params_.time_between_periodic_padding.Get().IsFinite() &&
       now - scream_->last_reference_window_decrease_time() >
           params_.allow_padding_after_last_congestion_time) {
-    if (params_.periodic_padding_interval->IsFinite() &&
-        (now - last_padding_interval_started_ >
-         params_.periodic_padding_interval.Get())) {
-      last_padding_interval_started_ = now;
+    if (now < padding_interval_end_time_) {
+      // We are currently inside an active padding duration.
+      allow_padding = true;
+    } else if (now - padding_interval_end_time_ >=
+               params_.time_between_periodic_padding.Get()) {
+      // Enough time has passed since the end of the last padding interval;
+      // start a new one.
+      padding_interval_end_time_ =
+          now + params_.periodic_padding_duration.Get();
+      allow_padding = true;
     }
-    if (now - last_padding_interval_started_ <
-        params_.periodic_padding_duration.Get()) {
-      padding_rate = target_rate;
+  } else {
+    // Stop padding immediately if a congestion event occurred recently.
+    padding_interval_end_time_ = Timestamp::MinusInfinity();
+  }
+
+  DataRate padding_rate = DataRate::Zero();
+  if (allow_padding) {
+    // Padding is allowed.
+    DataRate max_padding_rate =
+        std::min({max_target_rate_, max_seen_total_allocated_bitrate_,
+                  2 * streams_config_.max_total_allocated_bitrate.value_or(
+                          DataRate::Zero()),
+                  remote_bitrate_report_.value_or(DataRate::PlusInfinity())});
+    if (max_padding_rate.IsZero() && now < initial_bwe_probe_end_time_) {
+      // If initial BWE probing is allowed, probe up to the max target rate.
+      max_padding_rate = max_target_rate_;
     }
+    DataRate target_rate = scream_->target_rate();
+    padding_rate =
+        target_rate < max_padding_rate ? target_rate : DataRate::Zero();
   }
 
   DataRate pacing_rate = scream_->pacing_rate();
diff --git a/modules/congestion_controller/scream/scream_network_controller.h b/modules/congestion_controller/scream/scream_network_controller.h
index 4dfebff552..2a03ce6741 100644
--- a/modules/congestion_controller/scream/scream_network_controller.h
+++ b/modules/congestion_controller/scream/scream_network_controller.h
@@ -50,7 +50,7 @@ class ScreamNetworkController : public NetworkControllerInterface {
   void UpdateScreamTargetBitrateConstraints();
   NetworkControlUpdate CreateFirstUpdate(Timestamp now);
   NetworkControlUpdate CreateUpdate(Timestamp now);
-  std::optional<PacerConfig> MaybeCreatePacerConfig();
+  std::optional<PacerConfig> MaybeCreatePacerConfig(Timestamp now);
 
   Environment env_;
   const ScreamV2Parameters params_;
@@ -66,8 +66,8 @@ class ScreamNetworkController : public NetworkControllerInterface {
   std::optional<DataRate> remote_bitrate_report_;
   StreamsConfig streams_config_;
   DataRate max_seen_total_allocated_bitrate_ = DataRate::Zero();
-
-  Timestamp last_padding_interval_started_;
+  Timestamp initial_bwe_probe_end_time_ = Timestamp::MinusInfinity();
+  Timestamp padding_interval_end_time_ = Timestamp::MinusInfinity();
 
   // Values last reported in a NetworkControlUpdate. Used for finding out if an
   // update needs to be reported.
diff --git a/modules/congestion_controller/scream/scream_network_controller_unittest.cc b/modules/congestion_controller/scream/scream_network_controller_unittest.cc
index 1a9dcabf15..9dc4280d4b 100644
--- a/modules/congestion_controller/scream/scream_network_controller_unittest.cc
+++ b/modules/congestion_controller/scream/scream_network_controller_unittest.cc
@@ -48,8 +48,8 @@ TEST(ScreamControllerTest, OnNetworkAvailabilityUpdatesTargetRateAndPacerRate) {
       DataRate::KilobitsPerSec(456);
   ScreamNetworkController scream_controller(config);
 
-  NetworkControlUpdate update =
-      scream_controller.OnNetworkAvailability({.network_available = true});
+  NetworkControlUpdate update = scream_controller.OnNetworkAvailability(
+      {.at_time = clock.CurrentTime(), .network_available = true});
   ASSERT_TRUE(update.has_updates());
   ASSERT_TRUE(update.target_rate.has_value());
   EXPECT_EQ(update.target_rate->target_rate, config.constraints.starting_rate);
@@ -92,7 +92,8 @@ TEST(ScreamControllerTest,
   config.stream_based_config.max_total_allocated_bitrate =
       DataRate::KilobitsPerSec(1000);
   ScreamNetworkController scream_controller(config);
-  scream_controller.OnNetworkAvailability({.network_available = true});
+  scream_controller.OnNetworkAvailability(
+      {.at_time = clock.CurrentTime(), .network_available = true});
 
   CcFeedbackGenerator feedback_generator({});
   DataRate send_rate = DataRate::KilobitsPerSec(100);
@@ -312,6 +313,8 @@ TEST(ScreamControllerTest, InitiallyPaddingIsAllowedToReachNeededRate) {
   StreamsConfig streams_config;
   streams_config.max_total_allocated_bitrate = DataRate::KilobitsPerSec(1000);
   scream_controller.OnStreamsConfig(streams_config);
+  NetworkControlUpdate update = scream_controller.OnNetworkAvailability(
+      {.at_time = clock.CurrentTime(), .network_available = true});
 
   DataRate send_rate = DataRate::KilobitsPerSec(50);
   DataRate target_rate = DataRate::Zero();
@@ -323,8 +326,7 @@ TEST(ScreamControllerTest, InitiallyPaddingIsAllowedToReachNeededRate) {
         feedback_generator.ProcessUntilNextFeedback(
             send_rate, clock,
             [&](SentPacket packet) { scream_controller.OnSentPacket(packet); });
-    NetworkControlUpdate update =
-        scream_controller.OnTransportPacketsFeedback(feedback);
+    update = scream_controller.OnTransportPacketsFeedback(feedback);
     if (update.pacer_config.has_value()) {
       if (update.pacer_config->pad_rate() != DataRate::Zero()) {
         padding_set = true;
@@ -355,6 +357,35 @@ TEST(ScreamControllerTest, InitiallyPaddingIsAllowedToReachNeededRate) {
   EXPECT_LT(padding_stop - start_time, TimeDelta::Seconds(1));
 }
 
+TEST(ScreamControllerTest, InitialProbingWithoutMaxTotalAllocatedBitrate) {
+  SimulatedClock clock(Timestamp::Seconds(1'234));
+  Environment env = CreateTestEnvironment({.time = &clock});
+  NetworkControllerConfig config(env);
+  config.stream_based_config.enable_repeated_initial_probing = true;
+  config.constraints.starting_rate = DataRate::KilobitsPerSec(100);
+  config.constraints.max_data_rate = DataRate::KilobitsPerSec(1000);
+  // Do not set config.stream_based_config.max_total_allocated_bitrate
+
+  ScreamNetworkController scream_controller(config);
+  NetworkControlUpdate update = scream_controller.OnNetworkAvailability(
+      {.at_time = clock.CurrentTime(), .network_available = true});
+
+  ASSERT_TRUE(update.pacer_config.has_value());
+  // Padding is allowed during the first 6 seconds even if
+  // max_total_allocated_bitrate is zero.
+  EXPECT_EQ(update.pacer_config->pad_rate(), config.constraints.starting_rate);
+
+  // Advance clock past the 6s initial BWE probe window.
+  clock.AdvanceTime(TimeDelta::Seconds(7));
+  update =
+      scream_controller.OnProcessInterval({.at_time = clock.CurrentTime()});
+  // Since max_total_allocated_bitrate is not set, padding should stop after the
+  // initial probe window.
+  if (update.pacer_config.has_value()) {
+    EXPECT_EQ(update.pacer_config->pad_rate(), DataRate::Zero());
+  }
+}
+
 struct PaddingTestResult {
   DataRate target_rate;
   Timestamp padding_start;
@@ -450,7 +481,7 @@ TEST(ScreamControllerTest, PeriodicallyAllowPadding) {
   TimeDelta time_between_padding =
       result_2.padding_start - result_1.padding_stop;
   EXPECT_GT(padding_duration, TimeDelta::Millis(2500));
-  EXPECT_LT(padding_duration, TimeDelta::Millis(3100));
+  EXPECT_LT(padding_duration, TimeDelta::Millis(3300));
 
   EXPECT_GT(time_between_padding, TimeDelta::Millis(2500));
   EXPECT_LT(time_between_padding, TimeDelta::Millis(3300));
diff --git a/modules/congestion_controller/scream/scream_v2_parameters.cc b/modules/congestion_controller/scream/scream_v2_parameters.cc
index 7c70de148d..68f6fe393d 100644
--- a/modules/congestion_controller/scream/scream_v2_parameters.cc
+++ b/modules/congestion_controller/scream/scream_v2_parameters.cc
@@ -52,11 +52,12 @@ ScreamV2Parameters::ScreamV2Parameters(const FieldTrialsView& trials)
       queue_delay_drain_threshold("QDelayDrainThreshold", TimeDelta::Millis(5)),
       queue_delay_drain_period("QDelayDrainPeriod", TimeDelta::Seconds(20)),
       queue_delay_drain_rtts("QDelayDrainRtts", 5),
-      periodic_padding_interval("PeriodicPadding", TimeDelta::Seconds(6)),
+      time_between_periodic_padding("PeriodicPadding", TimeDelta::Seconds(3)),
       periodic_padding_duration("PaddingDuration", TimeDelta::Seconds(3)),
       allow_padding_after_last_congestion_time(
           "AllowPaddingAfterLastCongestionTimeout",
           TimeDelta::Seconds(1)),
+      initial_probing_duration("InitialProbingDuration", TimeDelta::Seconds(6)),
       pacing_factor("PacingFactor", 1.1),
       feedback_hold_time_avg_g("FeedbackHoldTimeAvgG", 1.0 / 8.0),
       allow_large_pacing_bursts_after_congestion_time(
@@ -87,9 +88,10 @@ ScreamV2Parameters::ScreamV2Parameters(const FieldTrialsView& trials)
                    &queue_delay_drain_threshold,
                    &queue_delay_drain_period,
                    &queue_delay_drain_rtts,
-                   &periodic_padding_interval,
+                   &time_between_periodic_padding,
                    &periodic_padding_duration,
                    &allow_padding_after_last_congestion_time,
+                   &initial_probing_duration,
                    &pacing_factor,
                    &feedback_hold_time_avg_g,
                    &allow_large_pacing_bursts_after_congestion_time},
diff --git a/modules/congestion_controller/scream/scream_v2_parameters.h b/modules/congestion_controller/scream/scream_v2_parameters.h
index 9dafdba008..bd4c129652 100644
--- a/modules/congestion_controller/scream/scream_v2_parameters.h
+++ b/modules/congestion_controller/scream/scream_v2_parameters.h
@@ -108,8 +108,9 @@ struct ScreamV2Parameters {
   FieldTrialParameter<int> queue_delay_drain_rtts;
 
   // Padding is periodically used in order to increase target rate even if a
-  // stream does not produce a high enough rate.
-  FieldTrialParameter<TimeDelta> periodic_padding_interval;
+  // stream does not produce a high enough rate. Time between periodic padding
+  // is at least this long.
+  FieldTrialParameter<TimeDelta> time_between_periodic_padding;
   // Max duration padding is used when periodic padding start.
   // Padding is stopped if congestion occur.
   FieldTrialParameter<TimeDelta> periodic_padding_duration;
@@ -117,6 +118,9 @@ struct ScreamV2Parameters {
   // time reference window was reduced but at least `periodic_padding_interval`
   // must have passed since last time padding was used.
   FieldTrialParameter<TimeDelta> allow_padding_after_last_congestion_time;
+  // If initial BWE probing is allowed, allow periodic probing for this duration
+  // even of no streams have been configured.
+  FieldTrialParameter<TimeDelta> initial_probing_duration;
 
   // Factor multiplied by the current target rate to decide the pacing rate.
   FieldTrialParameter<double> pacing_factor;
diff --git a/test/peer_scenario/bwe_integration_tests/scream_test.cc b/test/peer_scenario/bwe_integration_tests/scream_test.cc
index de61e23150..ef5f84c1de 100644
--- a/test/peer_scenario/bwe_integration_tests/scream_test.cc
+++ b/test/peer_scenario/bwe_integration_tests/scream_test.cc
@@ -364,7 +364,7 @@ TEST(ScreamTest, MaybeTest(LinkCapacity600KbpsRtt20msNoEcn)) {
   SendMediaTestResult result = SendMediaInOneDirection(std::move(params), s);
   EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
                                               DataRate::KilobitsPerSec(200),
-                                              DataRate::KilobitsPerSec(800))));
+                                              DataRate::KilobitsPerSec(900))));
 }
 
 TEST(ScreamTest, MaybeTest(LinkCapacity600KbpsRtt100msEcn)) {
@@ -467,7 +467,7 @@ TEST(ScreamTest, MaybeTest(LinkCapacity1500KbpsRtt30msNoEcn)) {
   SendMediaTestResult result = SendMediaInOneDirection(std::move(params), s);
   EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
                                               DataRate::KilobitsPerSec(800),
-                                              DataRate::KilobitsPerSec(2000))));
+                                              DataRate::KilobitsPerSec(2200))));
 }
 
 TEST(ScreamTest, MaybeTest(LinkCapacity2MbpsRtt50msNoEcn)) {
@@ -564,8 +564,8 @@ TEST(ScreamTest, MaybeTest(LinkCapacity2MbpsRtt50msNoEcnWithTwcc)) {
   // BWE rampup is quite slow since feedback is only sent every 90ms
   // approximately.
   EXPECT_THAT(result.caller().subspan(5), Each(AvailableSendBitrateIsBetween(
-                                              DataRate::KilobitsPerSec(1200),
-                                              DataRate::KilobitsPerSec(2400))));
+                                              DataRate::KilobitsPerSec(700),
+                                              DataRate::KilobitsPerSec(2600))));
 }
 
 TEST(ScreamTest, MaybeTest(CallerPauseSendingVideoIfFeedbackNotReceived)) {

From 304ad08c63e22fb96e8ad98512e28494cbb9d6a9 Mon Sep 17 00:00:00 2001
From: Philipp Hancke <philipp.hancke@googlemail.com>
Date: Fri, 20 Mar 2026 10:30:03 +0100
Subject: [PATCH 361/472] gn_check_autofix: increase error limit for gn check

which after usage of -r to remove all target as of recently runs into
  Too many errors/warnings. Suppressing further messages.
  You can change the limit by passing --error-limit=<number>.
which leads to not fixing errors. Passing a sufficiently high number
as error limit avoids that.

Doing
  tools_webrtc/gn_check_autofix.py -r api/BUILD.gn
followed by
  tools_webrtc/gn_check_autofix -C out/Default
should again require no(t much) more than a single pass.

Bug: webrtc:467294026
Change-Id: I4912bdc6555256f7fbb55a0768296f0ab72f21a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/458520
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Christoffer Dewerin <jansson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47214}
---
 tools_webrtc/gn_check_autofix.py | 9 ++++++++-
 1 file changed, 8 insertions(+), 1 deletion(-)

diff --git a/tools_webrtc/gn_check_autofix.py b/tools_webrtc/gn_check_autofix.py
index 0732f4fe61..2400d88c01 100755
--- a/tools_webrtc/gn_check_autofix.py
+++ b/tools_webrtc/gn_check_autofix.py
@@ -272,7 +272,14 @@ useful to check for different configurations."""
     errors_by_file = defaultdict(lambda: defaultdict(set))
 
     if flags.local_build_dir:
-        mb_output = Run(["gn", "gen", "--check", flags.local_build_dir])
+        mb_output = Run([
+            "gn",
+            "gen",
+            "--check",
+            # High limit to expose all errors to the tool.
+            "--error-limit=20000",
+            flags.local_build_dir
+        ])
     else:
         with TemporaryDirectory() as tmp_dir:
             mb_script_path = os.path.join(SCRIPT_DIR, 'mb', 'mb.py')

From eabb52c792e758637957d4ac04ad6bf2600b5286 Mon Sep 17 00:00:00 2001
From: Philipp Hancke <philipp.hancke@googlemail.com>
Date: Wed, 11 Mar 2026 15:11:59 +0100
Subject: [PATCH 362/472] build: clean up modules/congestion_controller targets

using
  find modules/congestion_controller/ -name BUILD.gn | xargs tools_webrtc/gn_check_autofix.py -r
followed by
  tools_webrtc/gn_check_autofix -C out/Default

Bug: webrtc:467294026
Change-Id: I81961cace6bf661e142db8258cd03e8a398e4038
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/455860
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#47215}
---
 modules/congestion_controller/goog_cc/BUILD.gn     | 14 --------------
 .../goog_cc_scream_network_controller/BUILD.gn     |  5 -----
 modules/congestion_controller/pcc/BUILD.gn         |  2 --
 modules/congestion_controller/rtp/BUILD.gn         | 13 -------------
 modules/congestion_controller/scream/BUILD.gn      |  7 -------
 5 files changed, 41 deletions(-)

diff --git a/modules/congestion_controller/goog_cc/BUILD.gn b/modules/congestion_controller/goog_cc/BUILD.gn
index 5d65bc2c6d..837b1085f4 100644
--- a/modules/congestion_controller/goog_cc/BUILD.gn
+++ b/modules/congestion_controller/goog_cc/BUILD.gn
@@ -59,9 +59,7 @@ rtc_library("pushback_controller") {
   ]
   deps = [
     "../../../api:field_trials_view",
-    "../../../api/transport:network_control",
     "../../../api/units:data_size",
-    "../../../rtc_base:checks",
     "../../../rtc_base/experiments:rate_control_settings",
   ]
 }
@@ -191,7 +189,6 @@ rtc_library("delay_based_bwe") {
     "../../../rtc_base:race_checker",
     "../../../rtc_base/experiments:field_trial_parser",
     "../../../system_wrappers:metrics",
-    "../../pacing",
     "../../remote_bitrate_estimator",
   ]
 }
@@ -211,11 +208,8 @@ rtc_library("probe_controller") {
     "../../../api/units:time_delta",
     "../../../api/units:timestamp",
     "../../../logging:rtc_event_bwe",
-    "../../../logging:rtc_event_pacing",
     "../../../rtc_base:checks",
     "../../../rtc_base:logging",
-    "../../../rtc_base:macromagic",
-    "../../../rtc_base:safe_conversions",
     "../../../rtc_base/experiments:field_trial_parser",
     "../../../system_wrappers:metrics",
     "//third_party/abseil-cpp/absl/base:core_headers",
@@ -236,7 +230,6 @@ if (rtc_include_tests) {
       ":estimators",
       ":goog_cc",
       "../../../api:libjingle_logging_api",
-      "../../../api/rtc_event_log",
       "../../../api/transport:goog_cc",
       "../../../api/transport:network_control",
       "../../../api/units:data_rate",
@@ -272,14 +265,11 @@ if (rtc_include_tests) {
         ":alr_detector",
         ":delay_based_bwe",
         ":estimators",
-        ":goog_cc",
         ":loss_based_bwe_v2",
         ":probe_controller",
         ":pushback_controller",
         ":send_side_bwe",
         "../../../api:field_trials",
-        "../../../api:field_trials_view",
-        "../../../api:network_state_predictor_api",
         "../../../api/environment",
         "../../../api/environment:environment_factory",
         "../../../api/rtc_event_log",
@@ -296,9 +286,6 @@ if (rtc_include_tests) {
         "../../../logging:mocks",
         "../../../logging:rtc_event_bwe",
         "../../../rtc_base:checks",
-        "../../../rtc_base:logging",
-        "../../../rtc_base:random",
-        "../../../rtc_base:rtc_base_tests_utils",
         "../../../rtc_base:stringutils",
         "../../../rtc_base/experiments:alr_experiment",
         "../../../system_wrappers",
@@ -307,7 +294,6 @@ if (rtc_include_tests) {
         "../../../test/network:emulated_network",
         "../../../test/scenario",
         "../../../test/scenario:column_printer",
-        "../../pacing",
         "//testing/gmock",
         "//third_party/abseil-cpp/absl/base:nullability",
         "//third_party/abseil-cpp/absl/strings:string_view",
diff --git a/modules/congestion_controller/goog_cc_scream_network_controller/BUILD.gn b/modules/congestion_controller/goog_cc_scream_network_controller/BUILD.gn
index a064c4434d..3a63e1c042 100644
--- a/modules/congestion_controller/goog_cc_scream_network_controller/BUILD.gn
+++ b/modules/congestion_controller/goog_cc_scream_network_controller/BUILD.gn
@@ -14,11 +14,9 @@ rtc_library("goog_cc_scream_network_controller") {
   ]
 
   deps = [
-    "../../../api:field_trials_view",
     "../../../api/environment",
     "../../../api/transport:network_control",
     "../../../rtc_base:logging",
-    "../../../rtc_base/experiments:field_trial_parser",
     "../goog_cc",
     "../scream:scream_network_controller",
     "//third_party/abseil-cpp/absl/functional:any_invocable",
@@ -30,17 +28,14 @@ if (rtc_include_tests) {
     testonly = true
     sources = [ "goog_cc_scream_network_controller_unittest.cc" ]
     deps = [
-      ":goog_cc_scream_network_controller",
       "../../../api/environment",
       "../../../api/transport:ecn_marking",
       "../../../api/transport:goog_cc",
       "../../../api/transport:network_control",
       "../../../api/units:data_rate",
       "../../../api/units:data_size",
-      "../../../api/units:time_delta",
       "../../../test:create_test_environment",
       "../../../test:test_support",
-      "../goog_cc",
     ]
   }
 }
diff --git a/modules/congestion_controller/pcc/BUILD.gn b/modules/congestion_controller/pcc/BUILD.gn
index 5fce915259..c971116d65 100644
--- a/modules/congestion_controller/pcc/BUILD.gn
+++ b/modules/congestion_controller/pcc/BUILD.gn
@@ -73,7 +73,6 @@ rtc_library("utility_function") {
   ]
   deps = [
     ":monitor_interval",
-    "../../../api/transport:network_control",
     "../../../api/units:data_rate",
     "../../../rtc_base:checks",
   ]
@@ -87,7 +86,6 @@ rtc_library("bitrate_controller") {
   deps = [
     ":monitor_interval",
     ":utility_function",
-    "../../../api/transport:network_control",
     "../../../api/units:data_rate",
   ]
 }
diff --git a/modules/congestion_controller/rtp/BUILD.gn b/modules/congestion_controller/rtp/BUILD.gn
index 42c789c158..27c108a4f0 100644
--- a/modules/congestion_controller/rtp/BUILD.gn
+++ b/modules/congestion_controller/rtp/BUILD.gn
@@ -19,12 +19,9 @@ rtc_library("control_handler") {
     "../../../api:sequence_checker",
     "../../../api/transport:network_control",
     "../../../api/units:data_rate",
-    "../../../api/units:data_size",
     "../../../api/units:time_delta",
     "../../../rtc_base:checks",
     "../../../rtc_base:logging",
-    "../../../rtc_base:safe_conversions",
-    "../../../rtc_base:safe_minmax",
     "../../../rtc_base/system:no_unique_address",
     "../../pacing",
   ]
@@ -39,7 +36,6 @@ rtc_library("transport_feedback") {
   ]
 
   deps = [
-    "../..:module_api_public",
     "../../../api:sequence_checker",
     "../../../api/transport:ecn_marking",
     "../../../api/transport:network_control",
@@ -52,9 +48,7 @@ rtc_library("transport_feedback") {
     "../../../rtc_base:network_route",
     "../../../rtc_base:rtc_numerics",
     "../../../rtc_base/network:sent_packet",
-    "../../../rtc_base/synchronization:mutex",
     "../../../rtc_base/system:no_unique_address",
-    "../../../system_wrappers",
     "../../rtp_rtcp:ntp_time_util",
     "../../rtp_rtcp:rtp_rtcp_format",
     "//third_party/abseil-cpp/absl/algorithm:container",
@@ -75,23 +69,16 @@ if (rtc_include_tests) {
     ]
     deps = [
       ":transport_feedback",
-      "../:congestion_controller",
       "../../../api:array_view",
       "../../../api/transport:ecn_marking",
       "../../../api/transport:network_control",
       "../../../api/units:data_size",
       "../../../api/units:time_delta",
       "../../../api/units:timestamp",
-      "../../../logging:mocks",
       "../../../rtc_base:buffer",
-      "../../../rtc_base:checks",
-      "../../../rtc_base:logging",
-      "../../../rtc_base:safe_conversions",
       "../../../rtc_base/network:sent_packet",
       "../../../system_wrappers",
       "../../../test:test_support",
-      "../../pacing",
-      "../../remote_bitrate_estimator",
       "../../rtp_rtcp:ntp_time_util",
       "../../rtp_rtcp:rtp_rtcp_format",
       "//testing/gmock",
diff --git a/modules/congestion_controller/scream/BUILD.gn b/modules/congestion_controller/scream/BUILD.gn
index 1683c6f44f..d249b02892 100644
--- a/modules/congestion_controller/scream/BUILD.gn
+++ b/modules/congestion_controller/scream/BUILD.gn
@@ -21,7 +21,6 @@ rtc_library("delay_based_congestion_control") {
     "../../../api/units:time_delta",
     "../../../api/units:timestamp",
     "../../../rtc_base:checks",
-    "../../../rtc_base:logging",
     "../../../rtc_base:windowed_min_filter",
   ]
 }
@@ -47,7 +46,6 @@ rtc_library("scream_v2") {
   deps = [
     ":delay_based_congestion_control",
     ":scream_v2_parameters",
-    "../../../api:field_trials_view",
     "../../../api/environment",
     "../../../api/transport:ecn_marking",
     "../../../api/transport:network_control",
@@ -70,10 +68,7 @@ rtc_library("scream_network_controller") {
   deps = [
     ":scream_v2",
     ":scream_v2_parameters",
-    "../../../api:field_trials_view",
     "../../../api/environment",
-    "../../../api/transport:bandwidth_usage",
-    "../../../api/transport:ecn_marking",
     "../../../api/transport:network_control",
     "../../../api/units:data_rate",
     "../../../api/units:data_size",
@@ -82,7 +77,6 @@ rtc_library("scream_network_controller") {
     "../../../logging:rtc_event_bwe",
     "../../../rtc_base:checks",
     "../../../rtc_base:logging",
-    "../../../rtc_base/experiments:field_trial_parser",
   ]
 }
 
@@ -100,7 +94,6 @@ if (rtc_include_tests) {
       ":scream_v2",
       ":scream_v2_parameters",
       "../../../api/environment",
-      "../../../api/environment:environment_factory",
       "../../../api/transport:ecn_marking",
       "../../../api/transport:network_control",
       "../../../api/units:data_rate",

From 2877e294271db8509764d8d291346c80e9edfd9d Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Erik=20Spr=C3=A5ng?= <sprang@webrtc.org>
Date: Fri, 20 Mar 2026 11:36:34 +0100
Subject: [PATCH 363/472] FrameEncodeMetadataWriter: Don't make callbacks while
 holding lock.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

The FrameEncodeMetadataWriter class holds state about pending frames.
Based on callbacks, it may determine the some frame have been dropped,
and will on those cases call the EncodedImageCallback::OnFrameDropped()
method.

This CL makes sure it doesn't hold the internal mutex while making such
callbacks, as that poses a risk for deadlocks.

Bug: webrtc:467444018
Change-Id: Ie31ec6d725cda95b44ec25889609c2df84e58730
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/458580
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47216}
---
 video/frame_encode_metadata_writer.cc | 276 ++++++++++++++------------
 video/frame_encode_metadata_writer.h  |  14 +-
 2 files changed, 166 insertions(+), 124 deletions(-)

diff --git a/video/frame_encode_metadata_writer.cc b/video/frame_encode_metadata_writer.cc
index 828ae790a8..47c6080aac 100644
--- a/video/frame_encode_metadata_writer.cc
+++ b/video/frame_encode_metadata_writer.cc
@@ -16,6 +16,7 @@
 #include <memory>
 #include <optional>
 #include <utility>
+#include <vector>
 
 #include "api/environment/environment.h"
 #include "api/video/encoded_image.h"
@@ -103,145 +104,173 @@ void FrameEncodeMetadataWriter::OnSetRates(
 }
 
 void FrameEncodeMetadataWriter::OnEncodeStarted(const VideoFrame& frame) {
-  MutexLock lock(&lock_);
+  std::vector<DropNotification> drop_notifications;
+  {
+    MutexLock lock(&lock_);
 
-  timing_frames_info_.resize(num_spatial_layers_);
-  FrameMetadata metadata;
-  metadata.rtp_timestamp = frame.rtp_timestamp();
-  metadata.encode_start_time_ms = env_.clock().TimeInMilliseconds();
-  metadata.ntp_time_ms = frame.ntp_time_ms();
-  metadata.timestamp_us = frame.timestamp_us();
-  metadata.rotation = frame.rotation();
-  metadata.color_space = frame.color_space();
-  metadata.is_steady_state_refresh_frame = frame.update_rect().IsEmpty();
-  metadata.packet_infos = frame.packet_infos();
-  for (size_t si = 0; si < num_spatial_layers_; ++si) {
-    RTC_DCHECK(timing_frames_info_[si].frames.empty() ||
-               TimeDiff(frame.render_time_ms(),
-                        timing_frames_info_[si].frames.back().timestamp_us /
-                            1000) >= 0);
-    // If stream is disabled due to low bandwidth OnEncodeStarted still will be
-    // called and have to be ignored.
-    if (timing_frames_info_[si].target_bitrate_bytes_per_sec == 0)
-      continue;
-    if (timing_frames_info_[si].frames.size() == kMaxEncodeStartTimeListSize) {
-      ++stalled_encoder_logged_messages_;
-      if (stalled_encoder_logged_messages_ <= kMessagesThrottlingThreshold ||
-          stalled_encoder_logged_messages_ % kThrottleRatio == 0) {
-        RTC_LOG(LS_WARNING) << "Too many frames in the encode_start_list."
-                               " Did encoder stall?";
-        if (stalled_encoder_logged_messages_ == kMessagesThrottlingThreshold) {
-          RTC_LOG(LS_WARNING)
-              << "Too many log messages. Further stalled encoder"
-                 "warnings will be throttled.";
+    timing_frames_info_.resize(num_spatial_layers_);
+    FrameMetadata metadata;
+    metadata.rtp_timestamp = frame.rtp_timestamp();
+    metadata.encode_start_time_ms = env_.clock().TimeInMilliseconds();
+    metadata.ntp_time_ms = frame.ntp_time_ms();
+    metadata.timestamp_us = frame.timestamp_us();
+    metadata.rotation = frame.rotation();
+    metadata.color_space = frame.color_space();
+    metadata.is_steady_state_refresh_frame = frame.update_rect().IsEmpty();
+    metadata.packet_infos = frame.packet_infos();
+    for (size_t si = 0; si < num_spatial_layers_; ++si) {
+      RTC_DCHECK(timing_frames_info_[si].frames.empty() ||
+                 TimeDiff(frame.render_time_ms(),
+                          timing_frames_info_[si].frames.back().timestamp_us /
+                              1000) >= 0);
+      // If stream is disabled due to low bandwidth OnEncodeStarted still will
+      // be called and have to be ignored.
+      if (timing_frames_info_[si].target_bitrate_bytes_per_sec == 0)
+        continue;
+      if (timing_frames_info_[si].frames.size() ==
+          kMaxEncodeStartTimeListSize) {
+        ++stalled_encoder_logged_messages_;
+        if (stalled_encoder_logged_messages_ <= kMessagesThrottlingThreshold ||
+            stalled_encoder_logged_messages_ % kThrottleRatio == 0) {
+          RTC_LOG(LS_WARNING) << "Too many frames in the encode_start_list."
+                                 " Did encoder stall?";
+          if (stalled_encoder_logged_messages_ ==
+              kMessagesThrottlingThreshold) {
+            RTC_LOG(LS_WARNING)
+                << "Too many log messages. Further stalled encoder"
+                   "warnings will be throttled.";
+          }
         }
-      }
 
-      uint32_t dropped_rtp_timestamp =
-          timing_frames_info_[si].frames.front().rtp_timestamp;
-      timing_frames_info_[si].frames.pop_front();
-      frame_drop_callback_->OnFrameDropped(
-          dropped_rtp_timestamp, si,
-          !IsFrameWithRtpTimestampPending(dropped_rtp_timestamp));
+        uint32_t dropped_rtp_timestamp =
+            timing_frames_info_[si].frames.front().rtp_timestamp;
+        timing_frames_info_[si].frames.pop_front();
+        drop_notifications.push_back(
+            {dropped_rtp_timestamp, static_cast<int>(si),
+             !IsFrameWithRtpTimestampPending(dropped_rtp_timestamp)});
+      }
+      timing_frames_info_[si].frames.emplace_back(metadata);
     }
-    timing_frames_info_[si].frames.emplace_back(metadata);
+  }
+
+  for (const auto& drop : drop_notifications) {
+    frame_drop_callback_->OnFrameDropped(drop.rtp_timestamp, drop.spatial_id,
+                                         drop.is_end_of_temporal_unit);
   }
 }
 
 void FrameEncodeMetadataWriter::FillMetadataAndTimingInfo(
     size_t simulcast_svc_idx,
     EncodedImage* encoded_image) {
-  MutexLock lock(&lock_);
-  std::optional<size_t> outlier_frame_size;
-  std::optional<int64_t> encode_start_ms;
-  uint8_t timing_flags = VideoSendTiming::kNotTriggered;
+  std::vector<DropNotification> drop_notifications;
+  {
+    MutexLock lock(&lock_);
+    std::optional<size_t> outlier_frame_size;
+    std::optional<int64_t> encode_start_ms;
+    uint8_t timing_flags = VideoSendTiming::kNotTriggered;
 
-  int64_t encode_done_ms = env_.clock().TimeInMilliseconds();
+    int64_t encode_done_ms = env_.clock().TimeInMilliseconds();
 
-  encode_start_ms =
-      ExtractEncodeStartTimeAndFillMetadata(simulcast_svc_idx, encoded_image);
+    encode_start_ms = ExtractEncodeStartTimeAndFillMetadata(
+        simulcast_svc_idx, encoded_image, drop_notifications);
 
-  if (timing_frames_info_.size() > simulcast_svc_idx) {
-    size_t target_bitrate =
-        timing_frames_info_[simulcast_svc_idx].target_bitrate_bytes_per_sec;
-    if (framerate_fps_ > 0 && target_bitrate > 0) {
-      // framerate and target bitrate were reported by encoder.
-      size_t average_frame_size = target_bitrate / framerate_fps_;
-      outlier_frame_size.emplace(
-          average_frame_size *
-          codec_settings_.timing_frame_thresholds.outlier_ratio_percent / 100);
+    if (timing_frames_info_.size() > simulcast_svc_idx) {
+      size_t target_bitrate =
+          timing_frames_info_[simulcast_svc_idx].target_bitrate_bytes_per_sec;
+      if (framerate_fps_ > 0 && target_bitrate > 0) {
+        // framerate and target bitrate were reported by encoder.
+        size_t average_frame_size = target_bitrate / framerate_fps_;
+        outlier_frame_size.emplace(
+            average_frame_size *
+            codec_settings_.timing_frame_thresholds.outlier_ratio_percent /
+            100);
+      }
+    }
+
+    // Outliers trigger timing frames, but do not affect scheduled timing
+    // frames.
+    if (outlier_frame_size && encoded_image->size() >= *outlier_frame_size) {
+      timing_flags |= VideoSendTiming::kTriggeredBySize;
+    }
+
+    // Check if it's time to send a timing frame.
+    int64_t timing_frame_delay_ms =
+        encoded_image->capture_time_ms_ - last_timing_frame_time_ms_;
+    // Trigger threshold if it's a first frame, too long passed since the last
+    // timing frame, or we already sent timing frame on a different simulcast
+    // stream with the same capture time.
+    if (last_timing_frame_time_ms_ == -1 ||
+        timing_frame_delay_ms >=
+            codec_settings_.timing_frame_thresholds.delay_ms ||
+        timing_frame_delay_ms == 0) {
+      timing_flags |= VideoSendTiming::kTriggeredByTimer;
+      last_timing_frame_time_ms_ = encoded_image->capture_time_ms_;
+    }
+
+    // If encode start is not available that means that encoder uses internal
+    // source. In that case capture timestamp may be from a different clock with
+    // a drift relative to `env_.clock()`. We can't use it for Timing frames,
+    // because to being sent in the network capture time required to be less
+    // than all the other timestamps.
+    if (encode_start_ms) {
+      encoded_image->SetEncodeTime(*encode_start_ms, encode_done_ms);
+      encoded_image->timing_.flags = timing_flags;
+    } else {
+      encoded_image->timing_.flags = VideoSendTiming::kInvalid;
+    }
+
+    // Cancel timing frames for older layers.
+    if (encoded_image->is_end_of_temporal_unit().value_or(false)) {
+      DropOldOrEqualFrames(encoded_image->RtpTimestamp(), simulcast_svc_idx,
+                           drop_notifications);
     }
   }
 
-  // Outliers trigger timing frames, but do not affect scheduled timing
-  // frames.
-  if (outlier_frame_size && encoded_image->size() >= *outlier_frame_size) {
-    timing_flags |= VideoSendTiming::kTriggeredBySize;
-  }
-
-  // Check if it's time to send a timing frame.
-  int64_t timing_frame_delay_ms =
-      encoded_image->capture_time_ms_ - last_timing_frame_time_ms_;
-  // Trigger threshold if it's a first frame, too long passed since the last
-  // timing frame, or we already sent timing frame on a different simulcast
-  // stream with the same capture time.
-  if (last_timing_frame_time_ms_ == -1 ||
-      timing_frame_delay_ms >=
-          codec_settings_.timing_frame_thresholds.delay_ms ||
-      timing_frame_delay_ms == 0) {
-    timing_flags |= VideoSendTiming::kTriggeredByTimer;
-    last_timing_frame_time_ms_ = encoded_image->capture_time_ms_;
-  }
-
-  // If encode start is not available that means that encoder uses internal
-  // source. In that case capture timestamp may be from a different clock with a
-  // drift relative to `env_.clock()`. We can't use it for Timing frames,
-  // because to being sent in the network capture time required to be less than
-  // all the other timestamps.
-  if (encode_start_ms) {
-    encoded_image->SetEncodeTime(*encode_start_ms, encode_done_ms);
-    encoded_image->timing_.flags = timing_flags;
-  } else {
-    encoded_image->timing_.flags = VideoSendTiming::kInvalid;
-  }
-
-  // Cancel timing frames for older layers.
-  if (encoded_image->is_end_of_temporal_unit().value_or(false)) {
-    DropOldOrEqualFrames(encoded_image->RtpTimestamp(), simulcast_svc_idx);
+  for (const auto& drop : drop_notifications) {
+    frame_drop_callback_->OnFrameDropped(drop.rtp_timestamp, drop.spatial_id,
+                                         drop.is_end_of_temporal_unit);
   }
 }
 
 void FrameEncodeMetadataWriter::OnFrameDropped(uint32_t rtp_timestamp,
                                                int spatial_id,
                                                bool is_end_of_temporal_unit) {
-  MutexLock lock(&lock_);
-  RTC_DCHECK_LT(spatial_id, static_cast<int>(timing_frames_info_.size()));
-  bool found_timestamp = false;
-  while (!timing_frames_info_[spatial_id].frames.empty()) {
-    if (timing_frames_info_[spatial_id].frames.front().rtp_timestamp ==
-        rtp_timestamp) {
-      found_timestamp = true;
+  std::vector<DropNotification> drop_notifications;
+  {
+    MutexLock lock(&lock_);
+    RTC_DCHECK_LT(spatial_id, static_cast<int>(timing_frames_info_.size()));
+    bool found_timestamp = false;
+    while (!timing_frames_info_[spatial_id].frames.empty()) {
+      if (timing_frames_info_[spatial_id].frames.front().rtp_timestamp ==
+          rtp_timestamp) {
+        found_timestamp = true;
+        break;
+      } else if (IsNewerTimestamp(rtp_timestamp, timing_frames_info_[spatial_id]
+                                                     .frames.front()
+                                                     .rtp_timestamp)) {
+        uint32_t dropped_timestamp =
+            timing_frames_info_[spatial_id].frames.front().rtp_timestamp;
+        timing_frames_info_[spatial_id].frames.pop_front();
+        drop_notifications.push_back(
+            {dropped_timestamp, spatial_id,
+             !IsFrameWithRtpTimestampPending(dropped_timestamp)});
+        continue;
+      }
       break;
-    } else if (IsNewerTimestamp(rtp_timestamp, timing_frames_info_[spatial_id]
-                                                   .frames.front()
-                                                   .rtp_timestamp)) {
-      uint32_t dropped_timestamp =
-          timing_frames_info_[spatial_id].frames.front().rtp_timestamp;
+    }
+    if (found_timestamp) {
+      if (is_end_of_temporal_unit) {
+        DropOldOrEqualFrames(rtp_timestamp, spatial_id, drop_notifications);
+      }
       timing_frames_info_[spatial_id].frames.pop_front();
-      frame_drop_callback_->OnFrameDropped(
-          dropped_timestamp, spatial_id,
-          !IsFrameWithRtpTimestampPending(dropped_timestamp));
-      continue;
+      drop_notifications.push_back(
+          {rtp_timestamp, spatial_id, is_end_of_temporal_unit});
     }
-    break;
   }
-  if (found_timestamp) {
-    if (is_end_of_temporal_unit) {
-      DropOldOrEqualFrames(rtp_timestamp, spatial_id);
-    }
-    timing_frames_info_[spatial_id].frames.pop_front();
-    frame_drop_callback_->OnFrameDropped(rtp_timestamp, spatial_id,
-                                         is_end_of_temporal_unit);
+
+  for (const auto& drop : drop_notifications) {
+    frame_drop_callback_->OnFrameDropped(drop.rtp_timestamp, drop.spatial_id,
+                                         drop.is_end_of_temporal_unit);
   }
 }
 
@@ -276,7 +305,8 @@ void FrameEncodeMetadataWriter::Reset() {
 std::optional<int64_t>
 FrameEncodeMetadataWriter::ExtractEncodeStartTimeAndFillMetadata(
     size_t simulcast_svc_idx,
-    EncodedImage* encoded_image) {
+    EncodedImage* encoded_image,
+    std::vector<DropNotification>& drop_notifications) {
   std::optional<int64_t> result;
   size_t num_simulcast_svc_streams = timing_frames_info_.size();
   if (simulcast_svc_idx < num_simulcast_svc_streams) {
@@ -290,9 +320,9 @@ FrameEncodeMetadataWriter::ExtractEncodeStartTimeAndFillMetadata(
                             metadata_list->front().rtp_timestamp)) {
       uint32_t dropped_timestamp = metadata_list->front().rtp_timestamp;
       metadata_list->pop_front();
-      frame_drop_callback_->OnFrameDropped(
-          dropped_timestamp, simulcast_svc_idx,
-          !IsFrameWithRtpTimestampPending(dropped_timestamp));
+      drop_notifications.push_back(
+          {dropped_timestamp, static_cast<int>(simulcast_svc_idx),
+           !IsFrameWithRtpTimestampPending(dropped_timestamp)});
     }
 
     encoded_image->content_type_ =
@@ -331,8 +361,10 @@ FrameEncodeMetadataWriter::ExtractEncodeStartTimeAndFillMetadata(
   return result;
 }
 
-void FrameEncodeMetadataWriter::DropOldOrEqualFrames(uint32_t rtp_timestamp,
-                                                     int spatial_id_to_skip) {
+void FrameEncodeMetadataWriter::DropOldOrEqualFrames(
+    uint32_t rtp_timestamp,
+    int spatial_id_to_skip,
+    std::vector<DropNotification>& drop_notifications) {
   for (size_t si = 0; si < num_spatial_layers_; ++si) {
     if (spatial_id_to_skip == static_cast<int>(si)) {
       continue;
@@ -344,9 +376,9 @@ void FrameEncodeMetadataWriter::DropOldOrEqualFrames(uint32_t rtp_timestamp,
                              metadata_list->front().rtp_timestamp))) {
       uint32_t dropped_timestamp = metadata_list->front().rtp_timestamp;
       metadata_list->pop_front();
-      frame_drop_callback_->OnFrameDropped(
-          dropped_timestamp, si,
-          !IsFrameWithRtpTimestampPending(dropped_timestamp));
+      drop_notifications.push_back(
+          {dropped_timestamp, static_cast<int>(si),
+           !IsFrameWithRtpTimestampPending(dropped_timestamp)});
     }
   }
 }
diff --git a/video/frame_encode_metadata_writer.h b/video/frame_encode_metadata_writer.h
index aa400c4741..906006f145 100644
--- a/video/frame_encode_metadata_writer.h
+++ b/video/frame_encode_metadata_writer.h
@@ -57,12 +57,22 @@ class FrameEncodeMetadataWriter {
   void Reset();
 
  private:
+  struct DropNotification {
+    uint32_t rtp_timestamp;
+    int spatial_id;
+    bool is_end_of_temporal_unit;
+  };
+
   // For non-internal-source encoders, returns encode started time and fixes
   // capture timestamp for the frame, if corrupted by the encoder.
   std::optional<int64_t> ExtractEncodeStartTimeAndFillMetadata(
       size_t simulcast_svc_idx,
-      EncodedImage* encoded_image) RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
-  void DropOldOrEqualFrames(uint32_t rtp_timestamp, int spatial_id_to_skip)
+      EncodedImage* encoded_image,
+      std::vector<DropNotification>& drop_notifications)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
+  void DropOldOrEqualFrames(uint32_t rtp_timestamp,
+                            int spatial_id_to_skip,
+                            std::vector<DropNotification>& drop_notifications)
       RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
   // Returns true if any spatial layer contains a pending frame with the given
   // RTP timestamp.

From 97935343047476a081353dfcede500c24044147b Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?=C3=85sa=20Persson?= <asapersson@webrtc.org>
Date: Fri, 20 Mar 2026 12:01:24 +0000
Subject: [PATCH 364/472] Remove unused method VCMTiming::UpdateCurrentDelay.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Bug: b/493549134
Change-Id: Iff72142e90d617c3e94a72716c2b39d17baee9c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/458265
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47217}
---
 modules/video_coding/timing/timing.cc         |  43 ---
 modules/video_coding/timing/timing.h          |   7 -
 .../video_coding/timing/timing_unittest.cc    | 295 ++++++++++++------
 3 files changed, 195 insertions(+), 150 deletions(-)

diff --git a/modules/video_coding/timing/timing.cc b/modules/video_coding/timing/timing.cc
index 9485dedebb..a947b38554 100644
--- a/modules/video_coding/timing/timing.cc
+++ b/modules/video_coding/timing/timing.cc
@@ -64,7 +64,6 @@ VCMTiming::VCMTiming(Clock* clock, const FieldTrialsView& field_trials)
       max_playout_delay_(TimeDelta::Seconds(10)),
       jitter_delay_(TimeDelta::Zero()),
       current_delay_(TimeDelta::Zero()),
-      prev_frame_timestamp_(0),
       num_decoded_frames_(0),
       zero_playout_delay_min_pacing_("min_pacing",
                                      kZeroPlayoutDelayDefaultMinPacing),
@@ -81,7 +80,6 @@ void VCMTiming::Reset() {
   min_playout_delay_ = TimeDelta::Zero();
   jitter_delay_ = TimeDelta::Zero();
   current_delay_ = TimeDelta::Zero();
-  prev_frame_timestamp_ = 0;
 }
 
 void VCMTiming::set_render_delay(TimeDelta render_delay) {
@@ -121,47 +119,6 @@ void VCMTiming::SetJitterDelay(TimeDelta jitter_delay) {
   }
 }
 
-void VCMTiming::UpdateCurrentDelay(uint32_t frame_timestamp) {
-  MutexLock lock(&mutex_);
-  TimeDelta target_delay = TargetDelayInternal();
-
-  if (current_delay_.IsZero()) {
-    // Not initialized, set current delay to target.
-    current_delay_ = target_delay;
-  } else if (target_delay != current_delay_) {
-    TimeDelta delay_diff = target_delay - current_delay_;
-    // Never change the delay with more than 100 ms every second. If we're
-    // changing the delay in too large steps we will get noticeable freezes. By
-    // limiting the change we can increase the delay in smaller steps, which
-    // will be experienced as the video is played in slow motion. When lowering
-    // the delay the video will be played at a faster pace.
-    TimeDelta max_change = TimeDelta::Zero();
-    if (frame_timestamp < 0x0000ffff && prev_frame_timestamp_ > 0xffff0000) {
-      // wrap
-      max_change =
-          TimeDelta::Millis(kDelayMaxChangeMsPerS *
-                            (frame_timestamp + (static_cast<int64_t>(1) << 32) -
-                             prev_frame_timestamp_) /
-                            90000);
-    } else {
-      max_change =
-          TimeDelta::Millis(kDelayMaxChangeMsPerS *
-                            (frame_timestamp - prev_frame_timestamp_) / 90000);
-    }
-
-    if (max_change <= TimeDelta::Zero()) {
-      // Any changes less than 1 ms are truncated and will be postponed.
-      // Negative change will be due to reordering and should be ignored.
-      return;
-    }
-    delay_diff = std::max(delay_diff, -max_change);
-    delay_diff = std::min(delay_diff, max_change);
-
-    current_delay_ = current_delay_ + delay_diff;
-  }
-  prev_frame_timestamp_ = frame_timestamp;
-}
-
 void VCMTiming::UpdateCurrentDelay(Timestamp render_time,
                                    Timestamp actual_decode_time) {
   MutexLock lock(&mutex_);
diff --git a/modules/video_coding/timing/timing.h b/modules/video_coding/timing/timing.h
index fe35e8aaf7..2ccf1544ba 100644
--- a/modules/video_coding/timing/timing.h
+++ b/modules/video_coding/timing/timing.h
@@ -57,7 +57,6 @@ class VCMTiming {
   };
 
   static constexpr TimeDelta kDefaultRenderDelay = TimeDelta::Millis(10);
-  static constexpr int kDelayMaxChangeMsPerS = 100;
 
   VCMTiming(Clock* clock, const FieldTrialsView& field_trials);
   virtual ~VCMTiming() = default;
@@ -79,11 +78,6 @@ class VCMTiming {
   // Set the minimum and maximum playout delay from capture to render.
   void set_playout_delay(const VideoPlayoutDelay& playout_delay);
 
-  // Increases or decreases the current delay to get closer to the target delay.
-  // Calculates how long it has been since the previous call to this function,
-  // and increases/decreases the delay in proportion to the time difference.
-  void UpdateCurrentDelay(uint32_t frame_timestamp);
-
   // Increases or decreases the current delay to get closer to the target delay.
   // Given the actual decode time in ms and the render time in ms for a frame,
   // this function calculates how late the frame is and increases the delay
@@ -158,7 +152,6 @@ class VCMTiming {
   TimeDelta max_playout_delay_ RTC_GUARDED_BY(mutex_);
   TimeDelta jitter_delay_ RTC_GUARDED_BY(mutex_);
   TimeDelta current_delay_ RTC_GUARDED_BY(mutex_);
-  uint32_t prev_frame_timestamp_ RTC_GUARDED_BY(mutex_);
   std::optional<TimingFrameInfo> timing_frame_info_ RTC_GUARDED_BY(mutex_);
   size_t num_decoded_frames_ RTC_GUARDED_BY(mutex_);
   std::optional<int> max_composition_delay_in_frames_ RTC_GUARDED_BY(mutex_);
diff --git a/modules/video_coding/timing/timing_unittest.cc b/modules/video_coding/timing/timing_unittest.cc
index 17e1d3d677..45b26ab71f 100644
--- a/modules/video_coding/timing/timing_unittest.cc
+++ b/modules/video_coding/timing/timing_unittest.cc
@@ -26,6 +26,9 @@ namespace {
 
 constexpr Frequency k25Fps = Frequency::Hertz(25);
 constexpr Frequency k90kHz = Frequency::KiloHertz(90);
+constexpr TimeDelta kJitterDelay = TimeDelta::Millis(100);
+constexpr TimeDelta kDecodeTime = TimeDelta::Millis(20);
+constexpr TimeDelta kRenderDelay = TimeDelta::Millis(15);
 
 MATCHER(HasConsistentVideoDelayTimings, "") {
   // Delays should be non-negative.
@@ -79,101 +82,14 @@ MATCHER(HasConsistentVideoDelayTimings, "") {
   return p && m;
 }
 
-}  // namespace
-
-TEST(VCMTimingTest, JitterDelay) {
-  FieldTrials field_trials = CreateTestFieldTrials();
-  SimulatedClock clock(0);
-  VCMTiming timing(&clock, field_trials);
-  timing.Reset();
-
-  uint32_t timestamp = 0;
-  timing.UpdateCurrentDelay(timestamp);
-
-  timing.Reset();
-
-  timing.IncomingTimestamp(timestamp, clock.CurrentTime());
-  TimeDelta jitter_delay = TimeDelta::Millis(20);
-  timing.SetJitterDelay(jitter_delay);
-  timing.UpdateCurrentDelay(timestamp);
-  timing.set_render_delay(TimeDelta::Zero());
-  auto wait_time = timing.MaxWaitingTime(
-      timing.RenderTime(timestamp, clock.CurrentTime()), clock.CurrentTime(),
-      /*too_many_frames_queued=*/false);
-  // First update initializes the render time. Since we have no decode delay
-  // we get wait_time = renderTime - now - renderDelay = jitter.
-  EXPECT_EQ(jitter_delay, wait_time);
-
-  jitter_delay += TimeDelta::Millis(VCMTiming::kDelayMaxChangeMsPerS + 10);
-  timestamp += 90000;
-  clock.AdvanceTimeMilliseconds(1000);
-  timing.SetJitterDelay(jitter_delay);
-  timing.UpdateCurrentDelay(timestamp);
-  wait_time = timing.MaxWaitingTime(
-      timing.RenderTime(timestamp, clock.CurrentTime()), clock.CurrentTime(),
-      /*too_many_frames_queued=*/false);
-  // Since we gradually increase the delay we only get 100 ms every second.
-  EXPECT_EQ(jitter_delay - TimeDelta::Millis(10), wait_time);
-
-  timestamp += 90000;
-  clock.AdvanceTimeMilliseconds(1000);
-  timing.UpdateCurrentDelay(timestamp);
-  wait_time = timing.MaxWaitingTime(
-      timing.RenderTime(timestamp, clock.CurrentTime()), clock.CurrentTime(),
-      /*too_many_frames_queued=*/false);
-  EXPECT_EQ(jitter_delay, wait_time);
-
-  // Insert frames without jitter, verify that this gives the exact wait time.
-  const int kNumFrames = 300;
-  for (int i = 0; i < kNumFrames; i++) {
-    clock.AdvanceTime(1 / k25Fps);
-    timestamp += k90kHz / k25Fps;
-    timing.IncomingTimestamp(timestamp, clock.CurrentTime());
+void UpdateDecodeTimer(VCMTiming& timing,
+                       SimulatedClock& clock,
+                       TimeDelta decode_time) {
+  for (int i = 0; i < k25Fps.hertz(); ++i) {
+    clock.AdvanceTime(decode_time);
+    timing.StopDecodeTimer(decode_time, clock.CurrentTime());
+    clock.AdvanceTime(1 / k25Fps - decode_time);
   }
-  timing.UpdateCurrentDelay(timestamp);
-  wait_time = timing.MaxWaitingTime(
-      timing.RenderTime(timestamp, clock.CurrentTime()), clock.CurrentTime(),
-      /*too_many_frames_queued=*/false);
-  EXPECT_EQ(jitter_delay, wait_time);
-
-  // Add decode time estimates for 1 second.
-  const TimeDelta kDecodeTime = TimeDelta::Millis(10);
-  for (int i = 0; i < k25Fps.hertz(); i++) {
-    clock.AdvanceTime(kDecodeTime);
-    timing.StopDecodeTimer(kDecodeTime, clock.CurrentTime());
-    timestamp += k90kHz / k25Fps;
-    clock.AdvanceTime(1 / k25Fps - kDecodeTime);
-    timing.IncomingTimestamp(timestamp, clock.CurrentTime());
-  }
-  timing.UpdateCurrentDelay(timestamp);
-  wait_time = timing.MaxWaitingTime(
-      timing.RenderTime(timestamp, clock.CurrentTime()), clock.CurrentTime(),
-      /*too_many_frames_queued=*/false);
-  EXPECT_EQ(jitter_delay, wait_time);
-
-  const TimeDelta kMinTotalDelay = TimeDelta::Millis(200);
-  timing.set_min_playout_delay(kMinTotalDelay);
-  clock.AdvanceTimeMilliseconds(5000);
-  timestamp += 5 * 90000;
-  timing.UpdateCurrentDelay(timestamp);
-  const TimeDelta kRenderDelay = TimeDelta::Millis(10);
-  timing.set_render_delay(kRenderDelay);
-  wait_time = timing.MaxWaitingTime(
-      timing.RenderTime(timestamp, clock.CurrentTime()), clock.CurrentTime(),
-      /*too_many_frames_queued=*/false);
-  // We should at least have kMinTotalDelayMs - decodeTime (10) - renderTime
-  // (10) to wait.
-  EXPECT_EQ(kMinTotalDelay - kDecodeTime - kRenderDelay, wait_time);
-  // The total video delay should be equal to the min total delay.
-  EXPECT_EQ(kMinTotalDelay, timing.TargetVideoDelay());
-
-  // Reset playout delay.
-  timing.set_min_playout_delay(TimeDelta::Zero());
-  clock.AdvanceTimeMilliseconds(5000);
-  timestamp += 5 * 90000;
-  timing.UpdateCurrentDelay(timestamp);
-
-  EXPECT_THAT(timing.GetTimings(), HasConsistentVideoDelayTimings());
 }
 
 TEST(VCMTimingTest, TimestampWrapAround) {
@@ -386,6 +302,34 @@ TEST(VCMTimingTest, MaxWaitingTimeReturnsZeroIfTooManyFramesQueuedIsTrue) {
   EXPECT_THAT(timing.GetTimings(), HasConsistentVideoDelayTimings());
 }
 
+TEST(VCMTimingTest, MaxWaitingTime) {
+  FieldTrials field_trials = CreateTestFieldTrials();
+  SimulatedClock clock(0);
+  VCMTiming timing(&clock, field_trials);
+  timing.set_render_delay(kRenderDelay);
+  UpdateDecodeTimer(timing, clock, kDecodeTime);
+
+  Timestamp on_time = clock.CurrentTime() + kDecodeTime + kRenderDelay;
+
+  // Early frame.
+  Timestamp render_time = on_time + TimeDelta::Millis(1);
+  EXPECT_EQ(timing.MaxWaitingTime(render_time, clock.CurrentTime(),
+                                  /*too_many_frames_queued=*/false),
+            TimeDelta::Millis(1));
+
+  // Exactly on time.
+  render_time = on_time;
+  EXPECT_EQ(timing.MaxWaitingTime(render_time, clock.CurrentTime(),
+                                  /*too_many_frames_queued=*/false),
+            TimeDelta::Zero());
+
+  // Late frame.
+  render_time = on_time - TimeDelta::Millis(1);
+  EXPECT_EQ(timing.MaxWaitingTime(render_time, clock.CurrentTime(),
+                                  /*too_many_frames_queued=*/false),
+            TimeDelta::Millis(-1));
+}
+
 TEST(VCMTimingTest, UpdateCurrentDelayCapsWhenOffByMicroseconds) {
   FieldTrials field_trials = CreateTestFieldTrials();
   SimulatedClock clock(0);
@@ -413,7 +357,6 @@ TEST(VCMTimingTest, GetTimings) {
   FieldTrials field_trials = CreateTestFieldTrials();
   SimulatedClock clock(33);
   VCMTiming timing(&clock, field_trials);
-  timing.Reset();
 
   // Setup.
   TimeDelta render_delay = TimeDelta::Millis(11);
@@ -435,14 +378,12 @@ TEST(VCMTimingTest, GetTimings) {
   clock.AdvanceTimeMilliseconds(100);
 
   // On decoded.
-  TimeDelta decode_time = TimeDelta::Millis(4);
-  timing.StopDecodeTimer(decode_time, clock.CurrentTime());
+  UpdateDecodeTimer(timing, clock, kDecodeTime);
 
   VCMTiming::VideoDelayTimings timings = timing.GetTimings();
-  EXPECT_EQ(timings.num_decoded_frames, 1u);
+  EXPECT_GT(timings.num_decoded_frames, 0u);
   EXPECT_EQ(timings.minimum_delay, minimum_delay);
-  // A single decoded frame is not enough to calculate p95.
-  EXPECT_EQ(timings.estimated_max_decode_time, TimeDelta::Zero());
+  EXPECT_EQ(timings.estimated_max_decode_time, kDecodeTime);
   EXPECT_EQ(timings.render_delay, render_delay);
   EXPECT_EQ(timings.min_playout_delay, min_playout_delay);
   EXPECT_EQ(timings.max_playout_delay, max_playout_delay);
@@ -451,6 +392,41 @@ TEST(VCMTimingTest, GetTimings) {
   EXPECT_THAT(timings, HasConsistentVideoDelayTimings());
 }
 
+TEST(VCMTimingTest, Reset) {
+  FieldTrials field_trials = CreateTestFieldTrials();
+  SimulatedClock clock(Timestamp::Millis(33));
+  VCMTiming timing(&clock, field_trials);
+
+  timing.set_render_delay(TimeDelta::Millis(11));
+  TimeDelta min_playout_delay = TimeDelta::Millis(50);
+  TimeDelta max_playout_delay = TimeDelta::Millis(500);
+  timing.set_playout_delay({min_playout_delay, max_playout_delay});
+
+  // On complete.
+  timing.IncomingTimestamp(3000, clock.CurrentTime());
+
+  // On decodable.
+  Timestamp render_time = timing.RenderTime(3000, clock.CurrentTime());
+  timing.SetJitterDelay(TimeDelta::Millis(123));
+  timing.UpdateCurrentDelay(render_time, clock.CurrentTime());
+
+  // On decoded.
+  UpdateDecodeTimer(timing, clock, kDecodeTime);
+
+  timing.Reset();
+
+  VCMTiming::VideoDelayTimings timings = timing.GetTimings();
+  EXPECT_GT(timings.num_decoded_frames, 0u);
+  EXPECT_EQ(timings.minimum_delay, TimeDelta::Zero());
+  EXPECT_EQ(timings.estimated_max_decode_time, TimeDelta::Zero());
+  EXPECT_EQ(timings.render_delay, VCMTiming::kDefaultRenderDelay);
+  EXPECT_EQ(timings.min_playout_delay, TimeDelta::Zero());
+  EXPECT_EQ(timings.max_playout_delay, max_playout_delay);
+  EXPECT_EQ(timings.target_delay, TimeDelta::Zero());
+  EXPECT_EQ(timings.current_delay, TimeDelta::Zero());
+  EXPECT_THAT(timings, HasConsistentVideoDelayTimings());
+}
+
 TEST(VCMTimingTest, GetTimingsBeforeAndAfterValidRtpTimestamp) {
   SimulatedClock clock(33);
   FieldTrials field_trials = CreateTestFieldTrials();
@@ -492,4 +468,123 @@ TEST(VCMTimingTest, GetTimingsBeforeAndAfterValidRtpTimestamp) {
                              min_playout_delay);
 }
 
+TEST(VCMTimingTest, IncreasesCurrentDelayWhenFrameIsLate) {
+  FieldTrials field_trials = CreateTestFieldTrials();
+  SimulatedClock clock(0);
+  VCMTiming timing(&clock, field_trials);
+  timing.SetJitterDelay(kJitterDelay);
+  timing.set_render_delay(kRenderDelay);
+
+  // Current delay is initialized to jitter delay.
+  EXPECT_EQ(timing.GetTimings().current_delay, kJitterDelay);
+  EXPECT_EQ(timing.TargetVideoDelay(), kJitterDelay + kRenderDelay);
+
+  const TimeDelta kFrameDelay = TimeDelta::Millis(4);
+  // Current delay should be increased to get closer to target delay.
+  Timestamp render_time = clock.CurrentTime() + kRenderDelay;
+  Timestamp actual_decode_time = clock.CurrentTime() + kFrameDelay;
+  timing.UpdateCurrentDelay(render_time, actual_decode_time);
+
+  EXPECT_EQ(timing.GetTimings().current_delay, kJitterDelay + kFrameDelay);
+}
+
+TEST(VCMTimingTest, CapsCurrentDelayIncreaseToTarget) {
+  FieldTrials field_trials = CreateTestFieldTrials();
+  SimulatedClock clock(0);
+  VCMTiming timing(&clock, field_trials);
+  timing.SetJitterDelay(kJitterDelay);
+  timing.set_render_delay(kRenderDelay);
+
+  // Current delay is initialized to jitter delay.
+  EXPECT_EQ(timing.GetTimings().current_delay, kJitterDelay);
+  EXPECT_EQ(timing.TargetVideoDelay(), kJitterDelay + kRenderDelay);
+
+  const TimeDelta kFrameDelay = TimeDelta::Millis(588);
+  // Current delay should be increased but not exceed target delay.
+  Timestamp render_time = clock.CurrentTime() + kRenderDelay;
+  Timestamp actual_decode_time = clock.CurrentTime() + kFrameDelay;
+  timing.UpdateCurrentDelay(render_time, actual_decode_time);
+
+  EXPECT_EQ(timing.GetTimings().current_delay, kJitterDelay + kRenderDelay);
+}
+
+TEST(VCMTimingTest, KeepsCurrentDelayWhenFrameIsEarly) {
+  FieldTrials field_trials = CreateTestFieldTrials();
+  SimulatedClock clock(0);
+  VCMTiming timing(&clock, field_trials);
+  timing.SetJitterDelay(kJitterDelay);
+  timing.set_render_delay(kRenderDelay);
+
+  // Current delay is initialized to jitter delay.
+  EXPECT_EQ(timing.GetTimings().current_delay, kJitterDelay);
+  EXPECT_EQ(timing.TargetVideoDelay(), kJitterDelay + kRenderDelay);
+
+  // Frame is early.
+  // Delay should remain unchanged.
+  Timestamp render_time = clock.CurrentTime() + kRenderDelay * 2;
+  Timestamp actual_decode_time = clock.CurrentTime();
+  timing.UpdateCurrentDelay(render_time, actual_decode_time);
+
+  EXPECT_EQ(timing.GetTimings().current_delay, kJitterDelay);
+}
+
+TEST(VCMTimingTest, IncreasesCurrentDelayWhenFrameIsLateWithDecodeTime) {
+  FieldTrials field_trials = CreateTestFieldTrials();
+  SimulatedClock clock(0);
+  VCMTiming timing(&clock, field_trials);
+  timing.SetJitterDelay(kJitterDelay);
+  timing.set_render_delay(kRenderDelay);
+  UpdateDecodeTimer(timing, clock, kDecodeTime);
+
+  // Current delay is initialized to jitter delay.
+  EXPECT_EQ(timing.GetTimings().current_delay, kJitterDelay);
+  EXPECT_EQ(timing.TargetVideoDelay(),
+            kJitterDelay + kDecodeTime + kRenderDelay);
+
+  const TimeDelta kFrameDelay = TimeDelta::Millis(4);
+  // Current delay should be increased to get closer to target delay.
+  Timestamp render_time = clock.CurrentTime() + kDecodeTime + kRenderDelay;
+  Timestamp actual_decode_time = clock.CurrentTime() + kFrameDelay;
+  timing.UpdateCurrentDelay(render_time, actual_decode_time);
+
+  EXPECT_EQ(timing.GetTimings().current_delay, kJitterDelay + kFrameDelay);
+}
+
+TEST(VCMTimingTest, DecreasesCurrentDelayToTarget) {
+  FieldTrials field_trials = CreateTestFieldTrials();
+  SimulatedClock clock(0);
+  VCMTiming timing(&clock, field_trials);
+  timing.SetJitterDelay(kJitterDelay);
+  timing.set_render_delay(kRenderDelay);
+
+  // Current delay should be increased to target for late frame.
+  timing.UpdateCurrentDelay(clock.CurrentTime(),
+                            clock.CurrentTime() + TimeDelta::Millis(588));
+  EXPECT_EQ(timing.GetTimings().current_delay, timing.TargetVideoDelay());
+
+  // Reduce jitter delay.
+  timing.SetJitterDelay(kJitterDelay / 2);
+  EXPECT_EQ(timing.TargetVideoDelay(), kJitterDelay / 2 + kRenderDelay);
+
+  // Current delay should be decreased to new target for frame on-time.
+  timing.UpdateCurrentDelay(clock.CurrentTime() + kRenderDelay,
+                            clock.CurrentTime());
+  EXPECT_EQ(timing.GetTimings().current_delay, kJitterDelay / 2 + kRenderDelay);
+}
+
+TEST(VCMTimingTest, MinPlayoutDelayUpdatesTargetDelay) {
+  FieldTrials field_trials = CreateTestFieldTrials();
+  SimulatedClock clock(0);
+  VCMTiming timing(&clock, field_trials);
+  timing.SetJitterDelay(kJitterDelay);
+  timing.set_render_delay(kRenderDelay);
+
+  const TimeDelta kMinPlayout =
+      kJitterDelay + kRenderDelay + TimeDelta::Millis(50);
+  timing.set_min_playout_delay(kMinPlayout);
+
+  EXPECT_EQ(timing.TargetVideoDelay(), kMinPlayout);
+}
+
+}  // namespace
 }  // namespace webrtc

From 26b5241f6b15ea54e07b99e777af5296a99d5874 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Fri, 20 Mar 2026 07:07:51 -0700
Subject: [PATCH 365/472] Roll chromium_revision d1fc7b8926..31f6cbd695
 (1602437:1602558)

Change log: https://chromium.googlesource.com/chromium/src/+log/d1fc7b8926..31f6cbd695
Full diff: https://chromium.googlesource.com/chromium/src/+/d1fc7b8926..31f6cbd695

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/013cd9d1bc..7e958c162b
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/20f52cf52a..7858fc4146
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/e61263adf8..1ec8287478
* src/third_party/android_deps/autorolled/cipd: PGgvEOiRpg1Z5fBKSrpvnnX1CFBZU6KVNUBVkyukBKoC..pLc9v3YoCPpc5bZ-yPGyIg0hLqZeXaIpzEHHWOI09UsC
* src/third_party/androidx/cipd: HrOnciB8BQGl5MNBfsfgysiPtdI3nERmtEhel7WeC4oC..wzjBTMka_ZRUOzMrirBfHngiFx_liKeSqVeKtVlO7FwC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/28cf3dd50d..6453bd80a9
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/40b296d214..315fe92361
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/c9aa1481b5..d2aaedff4b
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/50d5e17692..1582fc6219
DEPS diff: https://chromium.googlesource.com/chromium/src/+/d1fc7b8926..31f6cbd695/DEPS

BUG=None

Change-Id: I7ca4d014a14ada71c68754692fe227220bb8ccc1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/458601
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47218}
---
 DEPS | 20 ++++++++++----------
 1 file changed, 10 insertions(+), 10 deletions(-)

diff --git a/DEPS b/DEPS
index 3a84d03239..d868b6d374 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': 'd1fc7b8926ecd425bcb2a42b5f690cc60f9854b8',
+  'chromium_revision': '31f6cbd695178167c3430e897e07cba52e316006',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@013cd9d1bc97edce5e115f5fbe48a771ef1d89d2',
+    'https://chromium.googlesource.com/chromium/src/build@7e958c162b6043bbb4b84e9d1418f8ee8f68ba34',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@95ed44cf5f06dbb5861030b91c9db9ccb4316762',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@20f52cf52a7b171aa220240f103b15ff4c3fef45',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@7858fc4146da04e5e0f5583afe3aa04337ef29ba',
     'condition': 'checkout_ios',
   },
   'src/testing':
     'https://chromium.googlesource.com/chromium/src/testing@53cc79e2fca64e4075719222582ca77ff1276bc4',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@e61263adf8c3a8a77d023d89e6d6ce8e9b49ece3',
+    'https://chromium.googlesource.com/chromium/src/third_party@1ec8287478b97a74971870a689dec6b1c9573f9d',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -488,7 +488,7 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@28cf3dd50d8a9c8bc810c8fa84138af8da711e97',
+    'https://boringssl.googlesource.com/boringssl.git@6453bd80a9fe293c0adb8ae67d5d9c233100a7b9',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@8be0e3114685fcc1589561067282edf75ea1259a',
   'src/third_party/catapult':
@@ -505,7 +505,7 @@ deps = {
   'src/third_party/crc32c/src':
     'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558',
   'src/third_party/depot_tools':
-    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@40b296d214b2fa92c4605a0244f26c2d824f3996',
+    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@315fe9236154fbe4124aeed270c9f5b96a060666',
   'src/third_party/ffmpeg':
     'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@946d97db8d906277085e361892b7efda5152e2f1',
   'src/third_party/flatbuffers/src':
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'c9aa1481b548e78ce3d6f79eabb6eca0be044e34',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'd2aaedff4b9e6fdb90edfdb57f02d79c87d23a73',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@50d5e17692834d647a6b47f138480f833d7c392e',
+    'https://chromium.googlesource.com/chromium/src/tools@1582fc6219ee1aba33145c937c50a2f2df665f8d',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'HrOnciB8BQGl5MNBfsfgysiPtdI3nERmtEhel7WeC4oC',
+          'version': 'wzjBTMka_ZRUOzMrirBfHngiFx_liKeSqVeKtVlO7FwC',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -842,7 +842,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/android_deps/autorolled',
-              'version': 'PGgvEOiRpg1Z5fBKSrpvnnX1CFBZU6KVNUBVkyukBKoC',
+              'version': 'pLc9v3YoCPpc5bZ-yPGyIg0hLqZeXaIpzEHHWOI09UsC',
           },
       ],
       'condition': 'checkout_android and non_git_source',

From 60c526de29ae7b1c203697885453c5f82c968a9d Mon Sep 17 00:00:00 2001
From: Evan Shrubsole <eshr@webrtc.org>
Date: Mon, 16 Mar 2026 13:35:34 +0000
Subject: [PATCH 366/472] Use GlobalSimulatedTimeController in
 dtmf_sender_unittest

Bug: webrtc:42223992
Change-Id: I7359fa261b641db0db6fb32e2c36ec376a6a6964
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/457580
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47219}
---
 pc/dtmf_sender_unittest.cc | 38 ++++++++++++++++++++------------------
 1 file changed, 20 insertions(+), 18 deletions(-)

diff --git a/pc/dtmf_sender_unittest.cc b/pc/dtmf_sender_unittest.cc
index 9c62cc151e..7e4063cad0 100644
--- a/pc/dtmf_sender_unittest.cc
+++ b/pc/dtmf_sender_unittest.cc
@@ -20,11 +20,11 @@
 #include "api/scoped_refptr.h"
 #include "api/test/rtc_error_matchers.h"
 #include "api/units/time_delta.h"
-#include "rtc_base/fake_clock.h"
-#include "rtc_base/thread.h"
+#include "api/units/timestamp.h"
 #include "rtc_base/time_utils.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/time_controller/simulated_time_controller.h"
 #include "test/wait_until.h"
 
 using webrtc::DtmfProviderInterface;
@@ -116,9 +116,12 @@ class FakeDtmfProvider : public DtmfProviderInterface {
 class DtmfSenderTest : public ::testing::Test {
  protected:
   DtmfSenderTest()
-      : observer_(new FakeDtmfObserver()), provider_(new FakeDtmfProvider()) {
+      : time_controller_(webrtc::Timestamp::Seconds(1)),
+        observer_(new FakeDtmfObserver()),
+        provider_(new FakeDtmfProvider()) {
     provider_->SetCanInsertDtmf(true);
-    dtmf_ = DtmfSender::Create(webrtc::Thread::Current(), provider_.get());
+    dtmf_ =
+        DtmfSender::Create(time_controller_.GetMainThread(), provider_.get());
     dtmf_->RegisterObserver(observer_.get());
   }
 
@@ -211,11 +214,10 @@ class DtmfSenderTest : public ::testing::Test {
     }
   }
 
-  webrtc::AutoThread main_thread_;
+  webrtc::GlobalSimulatedTimeController time_controller_;
   std::unique_ptr<FakeDtmfObserver> observer_;
   std::unique_ptr<FakeDtmfProvider> provider_;
   webrtc::scoped_refptr<DtmfSender> dtmf_;
-  webrtc::ScopedFakeClock fake_clock_;
 };
 
 TEST_F(DtmfSenderTest, CanInsertDtmf) {
@@ -232,7 +234,7 @@ TEST_F(DtmfSenderTest, InsertDtmf) {
   EXPECT_THAT(webrtc::WaitUntil(
                   [&] { return observer_->completed(); }, ::testing::IsTrue(),
                   {.timeout = webrtc::TimeDelta::Millis(kMaxWaitMs),
-                   .clock = &fake_clock_}),
+                   .clock = &time_controller_}),
               webrtc::IsRtcOk());
 
   // The unrecognized characters should be ignored.
@@ -252,7 +254,7 @@ TEST_F(DtmfSenderTest, InsertDtmfTwice) {
   EXPECT_THAT(webrtc::WaitUntil(
                   [&] { return observer_->tones().size(); }, ::testing::Eq(1),
                   {.timeout = webrtc::TimeDelta::Millis(kMaxWaitMs),
-                   .clock = &fake_clock_}),
+                   .clock = &time_controller_}),
               webrtc::IsRtcOk());
   VerifyExpectedState("2", duration, inter_tone_gap);
   // Insert with another tone buffer.
@@ -262,7 +264,7 @@ TEST_F(DtmfSenderTest, InsertDtmfTwice) {
   EXPECT_THAT(webrtc::WaitUntil(
                   [&] { return observer_->completed(); }, ::testing::IsTrue(),
                   {.timeout = webrtc::TimeDelta::Millis(kMaxWaitMs),
-                   .clock = &fake_clock_}),
+                   .clock = &time_controller_}),
               webrtc::IsRtcOk());
 
   std::vector<FakeDtmfProvider::DtmfInfo> dtmf_queue_ref;
@@ -281,13 +283,13 @@ TEST_F(DtmfSenderTest, InsertDtmfWhileProviderIsDeleted) {
   EXPECT_THAT(webrtc::WaitUntil(
                   [&] { return observer_->tones().size(); }, ::testing::Eq(1),
                   {.timeout = webrtc::TimeDelta::Millis(kMaxWaitMs),
-                   .clock = &fake_clock_}),
+                   .clock = &time_controller_}),
               webrtc::IsRtcOk());
   // Delete provider.
   dtmf_->OnDtmfProviderDestroyed();
   provider_.reset();
   // The queue should be discontinued so no more tone callbacks.
-  fake_clock_.AdvanceTime(webrtc::TimeDelta::Millis(200));
+  time_controller_.AdvanceTime(webrtc::TimeDelta::Millis(200));
   EXPECT_EQ(1U, observer_->tones().size());
 }
 
@@ -300,12 +302,12 @@ TEST_F(DtmfSenderTest, InsertDtmfWhileSenderIsDeleted) {
   EXPECT_THAT(webrtc::WaitUntil(
                   [&] { return observer_->tones().size(); }, ::testing::Eq(1),
                   {.timeout = webrtc::TimeDelta::Millis(kMaxWaitMs),
-                   .clock = &fake_clock_}),
+                   .clock = &time_controller_}),
               webrtc::IsRtcOk());
   // Delete the sender.
   dtmf_ = nullptr;
   // The queue should be discontinued so no more tone callbacks.
-  fake_clock_.AdvanceTime(webrtc::TimeDelta::Millis(200));
+  time_controller_.AdvanceTime(webrtc::TimeDelta::Millis(200));
   EXPECT_EQ(1U, observer_->tones().size());
 }
 
@@ -319,7 +321,7 @@ TEST_F(DtmfSenderTest, InsertEmptyTonesToCancelPreviousTask) {
   EXPECT_THAT(webrtc::WaitUntil(
                   [&] { return observer_->tones().size(); }, ::testing::Eq(1),
                   {.timeout = webrtc::TimeDelta::Millis(kMaxWaitMs),
-                   .clock = &fake_clock_}),
+                   .clock = &time_controller_}),
               webrtc::IsRtcOk());
   // Insert with another tone buffer.
   EXPECT_TRUE(dtmf_->InsertDtmf(tones2, duration, inter_tone_gap));
@@ -327,7 +329,7 @@ TEST_F(DtmfSenderTest, InsertEmptyTonesToCancelPreviousTask) {
   EXPECT_THAT(webrtc::WaitUntil(
                   [&] { return observer_->completed(); }, ::testing::IsTrue(),
                   {.timeout = webrtc::TimeDelta::Millis(kMaxWaitMs),
-                   .clock = &fake_clock_}),
+                   .clock = &time_controller_}),
               webrtc::IsRtcOk());
 
   std::vector<FakeDtmfProvider::DtmfInfo> dtmf_queue_ref;
@@ -346,7 +348,7 @@ TEST_F(DtmfSenderTest, InsertDtmfWithDefaultCommaDelay) {
   EXPECT_THAT(webrtc::WaitUntil(
                   [&] { return observer_->completed(); }, ::testing::IsTrue(),
                   {.timeout = webrtc::TimeDelta::Millis(kMaxWaitMs),
-                   .clock = &fake_clock_}),
+                   .clock = &time_controller_}),
               webrtc::IsRtcOk());
 
   VerifyOnProvider(tones, duration, inter_tone_gap);
@@ -365,7 +367,7 @@ TEST_F(DtmfSenderTest, InsertDtmfWithNonDefaultCommaDelay) {
   EXPECT_THAT(webrtc::WaitUntil(
                   [&] { return observer_->completed(); }, ::testing::IsTrue(),
                   {.timeout = webrtc::TimeDelta::Millis(kMaxWaitMs),
-                   .clock = &fake_clock_}),
+                   .clock = &time_controller_}),
               webrtc::IsRtcOk());
 
   VerifyOnProvider(tones, duration, inter_tone_gap, comma_delay);
@@ -404,7 +406,7 @@ TEST_F(DtmfSenderTest, InsertDtmfSendsAfterWait) {
   EXPECT_THAT(webrtc::WaitUntil(
                   [&] { return observer_->tones().size(); }, ::testing::Eq(1),
                   {.timeout = webrtc::TimeDelta::Millis(kMaxWaitMs),
-                   .clock = &fake_clock_}),
+                   .clock = &time_controller_}),
               webrtc::IsRtcOk());
   VerifyExpectedState("BC", duration, inter_tone_gap);
 }

From e844bd819c4663de5422be9c2cdb25204b03b41b Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Erik=20Spr=C3=A5ng?= <sprang@webrtc.org>
Date: Fri, 20 Mar 2026 18:00:56 +0100
Subject: [PATCH 367/472] Allow unspecified max allocatable bitrate in
 VideoSendStreamImpl
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

If we have no video the can be enabled, based on the VideoEncoderConfig
(e.g. the max bitrate is <= 0 or all simulcast streams are
`active=false`) - let the max bitrate be reported as 0 for the sake of
bandwidth allocation (MediaStreamAllocationConfig).

Previous to this CL, the above conditions would result in an arbitrary
value of 10Mbps to use as the max allocation limit. When creating a peer
connection with audio and video configured but the video disabled/muted,
that would result in the bandwidth estimator trying to send packet
trains to probe up to 10Mbps, potentially impacting the network and
disturbing audio even though we have no need for such as high BWE.

Bug: webrtc:494350649
Change-Id: Id6353fd93c6170610a790c084d75f3b3c5c3ee97
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449100
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47220}
---
 call/call_perf_tests.cc                       |   8 +-
 media/engine/webrtc_video_engine_unittest.cc  |  50 +++----
 .../bwe_integration_tests/bwe_ramp_up_test.cc |  28 +++-
 test/scenario/video_stream_unittest.cc        |   2 +-
 video/video_send_stream_impl.cc               | 134 ++++++++++--------
 video/video_send_stream_impl.h                |   6 +-
 video/video_send_stream_impl_unittest.cc      |  16 ++-
 7 files changed, 144 insertions(+), 100 deletions(-)

diff --git a/call/call_perf_tests.cc b/call/call_perf_tests.cc
index 9cd29d8d1c..14b9de618f 100644
--- a/call/call_perf_tests.cc
+++ b/call/call_perf_tests.cc
@@ -654,7 +654,6 @@ TEST_F(CallPerfTest, Bitrate_Kbps_NoPadWithoutMinTransmitBitrate) {
 #endif
 TEST_F(CallPerfTest, MAYBE_KeepsHighBitrateWhenReconfiguringSender) {
   static const uint32_t kInitialBitrateKbps = 400;
-  static const uint32_t kInitialBitrateOverheadKpbs = 6;
   static const uint32_t kReconfigureThresholdKbps = 600;
 
   class VideoStreamFactory
@@ -697,10 +696,9 @@ TEST_F(CallPerfTest, MAYBE_KeepsHighBitrateWhenReconfiguringSender) {
         // First time initialization. Frame size is known.
         // `expected_bitrate` is affected by bandwidth estimation before the
         // first frame arrives to the encoder.
-        uint32_t expected_bitrate =
-            last_set_bitrate_kbps_ > 0
-                ? last_set_bitrate_kbps_
-                : kInitialBitrateKbps - kInitialBitrateOverheadKpbs;
+        uint32_t expected_bitrate = last_set_bitrate_kbps_ > 0
+                                        ? last_set_bitrate_kbps_
+                                        : kInitialBitrateKbps;
         EXPECT_EQ(expected_bitrate, config->startBitrate)
             << "Encoder not initialized at expected bitrate.";
         EXPECT_EQ(test::VideoTestConstants::kDefaultWidth, config->width);
diff --git a/media/engine/webrtc_video_engine_unittest.cc b/media/engine/webrtc_video_engine_unittest.cc
index fdd105de71..368554440c 100644
--- a/media/engine/webrtc_video_engine_unittest.cc
+++ b/media/engine/webrtc_video_engine_unittest.cc
@@ -10020,6 +10020,11 @@ TEST_F(WebRtcVideoChannelBaseTest, EncoderSelectorSwitchCodec) {
   EXPECT_TRUE(send_channel_->SetSenderParameters(parameters));
   send_channel_->SetSend(true);
 
+  RtpParameters rtp_parameters = send_channel_->GetRtpSendParameters(kSsrc);
+  ASSERT_EQ(1u, rtp_parameters.encodings.size());
+  rtp_parameters.encodings[0].max_bitrate_bps = 1000000;
+  EXPECT_TRUE(send_channel_->SetRtpSendParameters(kSsrc, rtp_parameters).ok());
+
   std::optional<Codec> codec = send_channel_->GetSendCodec();
   ASSERT_TRUE(codec);
   EXPECT_EQ("VP8", codec->name);
@@ -10041,23 +10046,20 @@ TEST_F(WebRtcVideoChannelBaseTest, EncoderSelectorSwitchCodec) {
 
 TEST_F(WebRtcVideoChannelTest, ScaleResolutionDownToSinglecast) {
   VideoSenderParameters parameters;
-  parameters.codecs.push_back(GetEngineCodec("VP8"));
-  ASSERT_TRUE(send_channel_->SetSenderParameters(parameters));
-
   FakeVideoSendStream* stream = AddSendStream();
+  const uint32_t ssrc = stream->GetConfig().rtp.ssrcs[0];
   FrameForwarder frame_forwarder;
   FakeFrameSource frame_source(1280, 720, TimeDelta::Seconds(1) / 30,
                                env_.clock().CurrentTime());
-  EXPECT_TRUE(
-      send_channel_->SetVideoSend(last_ssrc_, nullptr, &frame_forwarder));
+  EXPECT_TRUE(send_channel_->SetVideoSend(ssrc, nullptr, &frame_forwarder));
+  frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
 
   {  // TEST scale_resolution_down_to < frame size
-    RtpParameters rtp_parameters =
-        send_channel_->GetRtpSendParameters(last_ssrc_);
+    RtpParameters rtp_parameters = send_channel_->GetRtpSendParameters(ssrc);
     EXPECT_EQ(1UL, rtp_parameters.encodings.size());
     rtp_parameters.encodings[0].scale_resolution_down_to = {.width = 640,
                                                             .height = 360};
-    send_channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
+    send_channel_->SetRtpSendParameters(ssrc, rtp_parameters);
 
     frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
 
@@ -10068,11 +10070,11 @@ TEST_F(WebRtcVideoChannelTest, ScaleResolutionDownToSinglecast) {
   }
 
   {  // TEST scale_resolution_down_to == frame size
-    auto rtp_parameters = send_channel_->GetRtpSendParameters(last_ssrc_);
+    auto rtp_parameters = send_channel_->GetRtpSendParameters(ssrc);
     EXPECT_EQ(1UL, rtp_parameters.encodings.size());
     rtp_parameters.encodings[0].scale_resolution_down_to = {.width = 1280,
                                                             .height = 720};
-    send_channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
+    send_channel_->SetRtpSendParameters(ssrc, rtp_parameters);
 
     frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
     auto streams = stream->GetVideoStreams();
@@ -10082,11 +10084,11 @@ TEST_F(WebRtcVideoChannelTest, ScaleResolutionDownToSinglecast) {
   }
 
   {  // TEST scale_resolution_down_to > frame size
-    auto rtp_parameters = send_channel_->GetRtpSendParameters(last_ssrc_);
+    auto rtp_parameters = send_channel_->GetRtpSendParameters(ssrc);
     EXPECT_EQ(1UL, rtp_parameters.encodings.size());
     rtp_parameters.encodings[0].scale_resolution_down_to = {.width = 2 * 1280,
                                                             .height = 2 * 720};
-    send_channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
+    send_channel_->SetRtpSendParameters(ssrc, rtp_parameters);
 
     frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
     auto streams = stream->GetVideoStreams();
@@ -10095,7 +10097,7 @@ TEST_F(WebRtcVideoChannelTest, ScaleResolutionDownToSinglecast) {
     EXPECT_EQ(checked_cast<size_t>(720), streams[0].height);
   }
 
-  EXPECT_TRUE(send_channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
+  EXPECT_TRUE(send_channel_->SetVideoSend(ssrc, nullptr, nullptr));
 }
 
 TEST_F(WebRtcVideoChannelTest, ScaleResolutionDownToSinglecastScaling) {
@@ -10104,18 +10106,18 @@ TEST_F(WebRtcVideoChannelTest, ScaleResolutionDownToSinglecastScaling) {
   ASSERT_TRUE(send_channel_->SetSenderParameters(parameters));
 
   FakeVideoSendStream* stream = AddSendStream();
+  const uint32_t ssrc = stream->GetConfig().rtp.ssrcs[0];
   FrameForwarder frame_forwarder;
   FakeFrameSource frame_source(1280, 720, TimeDelta::Seconds(1) / 30,
                                env_.clock().CurrentTime());
-  EXPECT_TRUE(
-      send_channel_->SetVideoSend(last_ssrc_, nullptr, &frame_forwarder));
+  EXPECT_TRUE(send_channel_->SetVideoSend(ssrc, nullptr, &frame_forwarder));
 
   {
-    auto rtp_parameters = send_channel_->GetRtpSendParameters(last_ssrc_);
+    auto rtp_parameters = send_channel_->GetRtpSendParameters(ssrc);
     EXPECT_EQ(1UL, rtp_parameters.encodings.size());
     rtp_parameters.encodings[0].scale_resolution_down_to = {.width = 720,
                                                             .height = 720};
-    send_channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
+    send_channel_->SetRtpSendParameters(ssrc, rtp_parameters);
 
     frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
 
@@ -10128,11 +10130,11 @@ TEST_F(WebRtcVideoChannelTest, ScaleResolutionDownToSinglecastScaling) {
   }
 
   {
-    auto rtp_parameters = send_channel_->GetRtpSendParameters(last_ssrc_);
+    auto rtp_parameters = send_channel_->GetRtpSendParameters(ssrc);
     EXPECT_EQ(1UL, rtp_parameters.encodings.size());
     rtp_parameters.encodings[0].scale_resolution_down_to = {.width = 1280,
                                                             .height = 1280};
-    send_channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
+    send_channel_->SetRtpSendParameters(ssrc, rtp_parameters);
 
     frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
 
@@ -10144,11 +10146,11 @@ TEST_F(WebRtcVideoChannelTest, ScaleResolutionDownToSinglecastScaling) {
   }
 
   {
-    auto rtp_parameters = send_channel_->GetRtpSendParameters(last_ssrc_);
+    auto rtp_parameters = send_channel_->GetRtpSendParameters(ssrc);
     EXPECT_EQ(1UL, rtp_parameters.encodings.size());
     rtp_parameters.encodings[0].scale_resolution_down_to = {.width = 650,
                                                             .height = 650};
-    send_channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
+    send_channel_->SetRtpSendParameters(ssrc, rtp_parameters);
 
     auto streams = stream->GetVideoStreams();
     ASSERT_EQ(streams.size(), 1u);
@@ -10159,11 +10161,11 @@ TEST_F(WebRtcVideoChannelTest, ScaleResolutionDownToSinglecastScaling) {
   }
 
   {
-    auto rtp_parameters = send_channel_->GetRtpSendParameters(last_ssrc_);
+    auto rtp_parameters = send_channel_->GetRtpSendParameters(ssrc);
     EXPECT_EQ(1UL, rtp_parameters.encodings.size());
     rtp_parameters.encodings[0].scale_resolution_down_to = {.width = 2560,
                                                             .height = 1440};
-    send_channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
+    send_channel_->SetRtpSendParameters(ssrc, rtp_parameters);
 
     auto streams = stream->GetVideoStreams();
     ASSERT_EQ(streams.size(), 1u);
@@ -10172,7 +10174,7 @@ TEST_F(WebRtcVideoChannelTest, ScaleResolutionDownToSinglecastScaling) {
     EXPECT_EQ(checked_cast<size_t>(720), streams[0].height);
   }
 
-  EXPECT_TRUE(send_channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
+  EXPECT_TRUE(send_channel_->SetVideoSend(ssrc, nullptr, nullptr));
 }
 
 TEST_F(WebRtcVideoChannelTest, ScaleResolutionDownToSimulcast) {
diff --git a/test/peer_scenario/bwe_integration_tests/bwe_ramp_up_test.cc b/test/peer_scenario/bwe_integration_tests/bwe_ramp_up_test.cc
index 8a0bbd37a3..4c5637a36b 100644
--- a/test/peer_scenario/bwe_integration_tests/bwe_ramp_up_test.cc
+++ b/test/peer_scenario/bwe_integration_tests/bwe_ramp_up_test.cc
@@ -16,6 +16,7 @@
 
 #include "api/jsep.h"
 #include "api/media_types.h"
+#include "api/rtp_parameters.h"
 #include "api/rtp_sender_interface.h"
 #include "api/rtp_transceiver_direction.h"
 #include "api/scoped_refptr.h"
@@ -34,6 +35,7 @@
 #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "modules/rtp_rtcp/source/rtp_util.h"
 #include "pc/media_session.h"
+#include "rtc_base/logging.h"
 #include "test/create_frame_generator_capturer.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
@@ -206,7 +208,7 @@ INSTANTIATE_TEST_SUITE_P(
              .l4s_network = true,
              .network_capacity = DataRate::KilobitsPerSec(3000),
              .expected_bwe_min = DataRate::KilobitsPerSec(1500),
-             .max_bwe = DataRate::KilobitsPerSec(3000),
+             .max_bwe = DataRate::KilobitsPerSec(3600),
          },
          {
              .test_name = "L4s500Kbit",
@@ -239,12 +241,18 @@ TEST_P(BweRampupWithInitialProbeTest, BweRampUpBothDirectionsWithoutMedia) {
   PeerScenarioClient* caller = s.CreateClient(config);
   PeerScenarioClient* callee = s.CreateClient(config);
 
-  auto transceiver = caller->pc()->AddTransceiver(MediaType::VIDEO);
-  ASSERT_TRUE(transceiver.error().ok());
+  auto transceiver_or = caller->pc()->AddTransceiver(MediaType::VIDEO);
+  ASSERT_TRUE(transceiver_or.error().ok());
+  auto transceiver = transceiver_or.value();
+
+  auto parameters = transceiver->sender()->GetParameters();
+  ASSERT_THAT(parameters.encodings, testing::SizeIs(1));
+  parameters.encodings[0].max_bitrate_bps = 3'000'000;
+  ASSERT_TRUE(transceiver->sender()->SetParameters(parameters).ok());
 
   MockRtpSenderObserver observer;
   EXPECT_CALL(observer, OnFirstPacketSent).Times(0);
-  transceiver.value()->sender()->SetObserver(&observer);
+  transceiver->sender()->SetObserver(&observer);
 
   caller->pc()->ReconfigureBandwidthEstimation(
       {.allow_probe_without_media = true});
@@ -290,6 +298,18 @@ TEST_P(BweRampupWithInitialProbeTest, BweRampUpBothDirectionsWithoutMedia) {
   // Wait for SDP negotiation.
   s.WaitAndProcess(&offer_exchange_done);
 
+  // Set max encoding rate for Callee as well.
+  ASSERT_THAT(callee->pc()->GetTransceivers(), SizeIs(1));
+  auto callee_transceiver = callee->pc()->GetTransceivers()[0];
+  auto callee_params = callee_transceiver->sender()->GetParameters();
+  if (callee_params.encodings.size() > 0) {
+    callee_params.encodings[0].max_bitrate_bps = 10'000'000;
+    ASSERT_TRUE(
+        callee_transceiver->sender()->SetParameters(callee_params).ok());
+  } else {
+    RTC_LOG(LS_WARNING) << "Callee encodings is empty after negotiation!";
+  }
+
   // Test that 1s after offer/answer exchange finish, we have a BWE estimate,
   // even though no video frames have been sent.
   s.ProcessMessages(TimeDelta::Seconds(2));
diff --git a/test/scenario/video_stream_unittest.cc b/test/scenario/video_stream_unittest.cc
index c9791bf8fc..c1fcd4da1c 100644
--- a/test/scenario/video_stream_unittest.cc
+++ b/test/scenario/video_stream_unittest.cc
@@ -281,7 +281,7 @@ TEST(VideoStreamTest, SuspendsBelowMinBitrate) {
     // Min transmit rate needs to be lower than kMinVideoBitrate for this test
     // to make sense.
     c->transport.rates.min_rate = kMinVideoBitrate / 2;
-    c->transport.rates.start_rate = kMinVideoBitrate;
+    c->transport.rates.start_rate = kMinVideoBitrate * 2;
     c->transport.rates.max_rate = kMinVideoBitrate * 2;
   });
   auto send_net = s.CreateMutableSimulationNode(
diff --git a/video/video_send_stream_impl.cc b/video/video_send_stream_impl.cc
index ad9130bb1d..301939f4eb 100644
--- a/video/video_send_stream_impl.cc
+++ b/video/video_send_stream_impl.cc
@@ -236,32 +236,50 @@ int GetEncoderPriorityBitrate(std::string codec_name,
   return priority_bitrate;
 }
 
-uint32_t GetInitialEncoderMaxBitrate(int initial_encoder_max_bitrate) {
-  if (initial_encoder_max_bitrate > 0)
-    return dchecked_cast<uint32_t>(initial_encoder_max_bitrate);
-
-  // TODO(srte): Make sure max bitrate is not set to negative values. We don't
-  // have any way to handle unset values in downstream code, such as the
-  // bitrate allocator. Previously -1 was implicitly casted to UINT32_MAX, a
-  // behaviour that is not safe. Converting to 10 Mbps should be safe for
-  // reasonable use cases as it allows adding the max of multiple streams
-  // without wrappping around.
-  const int kFallbackMaxBitrateBps = 10000000;
-  // Don't log an error for -1 since this is the default value that is used to
-  // signal that the max bitrate is unset.
-  if (initial_encoder_max_bitrate != -1) {
-    RTC_DLOG(LS_ERROR) << "ERROR: Initial encoder max bitrate = "
-                       << initial_encoder_max_bitrate << " which is <= 0!";
+DataRate GetDefaultMinVideoBitrate(VideoCodecType codec_type) {
+  if (codec_type == VideoCodecType::kVideoCodecAV1) {
+    return DataRate::BitsPerSec(kMinDefaultAv1BitrateBps);
   }
-  RTC_DLOG(LS_INFO) << "Using default encoder max bitrate = 10 Mbps";
-  return kFallbackMaxBitrateBps;
+  return DataRate::BitsPerSec(kDefaultMinVideoBitrateBps);
 }
 
-int GetDefaultMinVideoBitrateBps(VideoCodecType codec_type) {
-  if (codec_type == VideoCodecType::kVideoCodecAV1) {
-    return kMinDefaultAv1BitrateBps;
+std::optional<DataRate> GetConfiguredMaxVideoBitrate(
+    const VideoEncoderConfig& config) {
+  // If spatial layers are configured, the sum of their bitrate takes precedence
+  // over both simulcast and the global max_bitrate_bps.
+  if (!config.spatial_layers.empty()) {
+    std::optional<DataRate> spatial_layer_max_bitrate_sum;
+    for (const auto& layer : config.spatial_layers) {
+      if (layer.active && layer.maxBitrate > 0) {
+        spatial_layer_max_bitrate_sum =
+            spatial_layer_max_bitrate_sum.value_or(DataRate::Zero()) +
+            DataRate::KilobitsPerSec(layer.maxBitrate);
+      }
+    }
+    return spatial_layer_max_bitrate_sum;
   }
-  return kDefaultMinVideoBitrateBps;
+
+  // If simulcast layers are configured, the sum of their bitrate takes
+  // precedence over the global max_bitrate_bps.
+  if (!config.simulcast_layers.empty()) {
+    std::optional<DataRate> simulcast_layer_max_bitrate_sum;
+    for (const auto& stream : config.simulcast_layers) {
+      if (stream.active && stream.max_bitrate_bps > 0) {
+        simulcast_layer_max_bitrate_sum =
+            simulcast_layer_max_bitrate_sum.value_or(DataRate::Zero()) +
+            DataRate::BitsPerSec(stream.max_bitrate_bps);
+      }
+    }
+    return simulcast_layer_max_bitrate_sum;
+  }
+
+  // If single stream, use the global max_bitrate_bps.
+  if (config.max_bitrate_bps > 0) {
+    return DataRate::BitsPerSec(config.max_bitrate_bps);
+  }
+
+  // No max bitrate specified.
+  return std::nullopt;
 }
 
 size_t CalculateMaxHeaderSize(const RtpConfig& config) {
@@ -455,10 +473,9 @@ VideoSendStreamImpl::VideoSendStreamImpl(
       has_active_encodings_(HasActiveEncodings(encoder_config)),
       disable_padding_(true),
       max_padding_bitrate_(0),
-      encoder_min_bitrate_bps_(0),
-      encoder_max_bitrate_bps_(
-          GetInitialEncoderMaxBitrate(encoder_config.max_bitrate_bps)),
-      encoder_target_rate_bps_(0),
+      encoder_min_bitrate_(DataRate::Zero()),
+      encoder_max_bitrate_(GetConfiguredMaxVideoBitrate(encoder_config)),
+      encoder_target_rate_(DataRate::Zero()),
       encoder_bitrate_priority_(encoder_config.bitrate_priority),
       encoder_av1_priority_bitrate_override_bps_(
           GetEncoderPriorityBitrate(config_.rtp.payload_name,
@@ -472,7 +489,6 @@ VideoSendStreamImpl::VideoSendStreamImpl(
   RTC_DCHECK_LE(config_.rtp.payload_type, 127);
   RTC_DCHECK(!config_.rtp.ssrcs.empty());
   RTC_DCHECK(transport_);
-  RTC_DCHECK_NE(encoder_max_bitrate_bps_, 0);
   RTC_LOG(LS_INFO) << "VideoSendStreamImpl: " << config_.ToString();
 
   RTC_CHECK(
@@ -719,7 +735,7 @@ void VideoSendStreamImpl::SignalEncoderTimedOut() {
   // If the encoder has not produced anything the last kEncoderTimeOut and it
   // is supposed to, deregister as BitrateAllocatorObserver. This can happen
   // if a camera stops producing frames.
-  if (encoder_target_rate_bps_ > 0) {
+  if (!encoder_target_rate_.IsZero()) {
     RTC_LOG(LS_INFO) << "SignalEncoderTimedOut, Encoder timed out.";
     bitrate_allocator_->RemoveObserver(this);
   }
@@ -731,7 +747,7 @@ void VideoSendStreamImpl::OnBitrateAllocationUpdated(
   // the worker_queue_.
   auto task = [this, allocation] {
     RTC_DCHECK_RUN_ON(&thread_checker_);
-    if (encoder_target_rate_bps_ == 0) {
+    if (encoder_target_rate_.IsZero()) {
       return;
     }
     int64_t now_ms = env_.clock().TimeInMilliseconds();
@@ -791,8 +807,9 @@ void VideoSendStreamImpl::SignalEncoderActive() {
 
 MediaStreamAllocationConfig VideoSendStreamImpl::GetAllocationConfig() const {
   return MediaStreamAllocationConfig{
-      .min_bitrate_bps = static_cast<uint32_t>(encoder_min_bitrate_bps_),
-      .max_bitrate_bps = encoder_max_bitrate_bps_,
+      .min_bitrate_bps = encoder_min_bitrate_.bps<uint32_t>(),
+      .max_bitrate_bps =
+          encoder_max_bitrate_.value_or(DataRate::Zero()).bps<uint32_t>(),
       .pad_up_bitrate_bps =
           static_cast<uint32_t>(disable_padding_ ? 0 : max_padding_bitrate_),
       .priority_bitrate_bps = encoder_av1_priority_bitrate_override_bps_,
@@ -822,17 +839,23 @@ void VideoSendStreamImpl::OnEncoderConfigurationChanged(
 
     const std::optional<DataRate> experimental_min_bitrate =
         GetExperimentalMinVideoBitrate(env_.field_trials(), codec_type);
-    encoder_min_bitrate_bps_ =
-        experimental_min_bitrate
-            ? experimental_min_bitrate->bps()
-            : std::max(streams[0].min_bitrate_bps,
-                       GetDefaultMinVideoBitrateBps(codec_type));
+    if (experimental_min_bitrate.has_value()) {
+      encoder_min_bitrate_ = *experimental_min_bitrate;
+    } else {
+      DataRate min_bitrate =
+          streams[0].min_bitrate_bps > 0
+              ? DataRate::BitsPerSec(streams[0].min_bitrate_bps)
+              : DataRate::Zero();
+      encoder_min_bitrate_ =
+          std::max(min_bitrate, GetDefaultMinVideoBitrate(codec_type));
+    }
     double stream_bitrate_priority_sum = 0;
-    uint32_t encoder_max_bitrate_bps = 0;
+    std::optional<DataRate> max_bitrate_sum;
     for (const auto& stream : streams) {
       // We don't want to allocate more bitrate than needed to inactive streams.
-      if (stream.active) {
-        encoder_max_bitrate_bps += stream.max_bitrate_bps;
+      if (stream.active && stream.max_bitrate_bps > 0) {
+        max_bitrate_sum = max_bitrate_sum.value_or(DataRate::Zero()) +
+                          DataRate::BitsPerSec(stream.max_bitrate_bps);
       }
       if (stream.bitrate_priority) {
         RTC_DCHECK_GT(*stream.bitrate_priority, 0);
@@ -841,10 +864,8 @@ void VideoSendStreamImpl::OnEncoderConfigurationChanged(
     }
     RTC_DCHECK_GT(stream_bitrate_priority_sum, 0);
     encoder_bitrate_priority_ = stream_bitrate_priority_sum;
-    if (encoder_max_bitrate_bps > 0) {
-      encoder_max_bitrate_bps_ =
-          std::max(static_cast<uint32_t>(encoder_min_bitrate_bps_),
-                   encoder_max_bitrate_bps);
+    if (max_bitrate_sum.has_value()) {
+      encoder_max_bitrate_ = std::max(encoder_min_bitrate_, *max_bitrate_sum);
     }
 
     // TODO(bugs.webrtc.org/10266): Query the VideoBitrateAllocator instead.
@@ -931,24 +952,25 @@ uint32_t VideoSendStreamImpl::OnBitrateUpdated(BitrateAllocationUpdate update) {
       << "VideoSendStream::Start has not been called.";
 
   rtp_video_sender_->OnBitrateUpdated(update, stats_proxy_.GetSendFrameRate());
-  encoder_target_rate_bps_ = rtp_video_sender_->GetPayloadBitrateBps();
-  const uint32_t protection_bitrate_bps =
-      rtp_video_sender_->GetProtectionBitrateBps();
+  encoder_target_rate_ =
+      DataRate::BitsPerSec(rtp_video_sender_->GetPayloadBitrateBps());
+  const DataRate protection_bitrate =
+      DataRate::BitsPerSec(rtp_video_sender_->GetProtectionBitrateBps());
   DataRate link_allocation = DataRate::Zero();
-  if (encoder_target_rate_bps_ > protection_bitrate_bps) {
-    link_allocation =
-        DataRate::BitsPerSec(encoder_target_rate_bps_ - protection_bitrate_bps);
+  if (encoder_target_rate_ > protection_bitrate) {
+    link_allocation = encoder_target_rate_ - protection_bitrate;
   }
-  encoder_target_rate_bps_ =
-      std::min(encoder_max_bitrate_bps_, encoder_target_rate_bps_);
-  DataRate encoder_target_rate = DataRate::BitsPerSec(encoder_target_rate_bps_);
-  link_allocation = std::max(encoder_target_rate, link_allocation);
+  if (encoder_max_bitrate_.has_value()) {
+    encoder_target_rate_ =
+        std::min(*encoder_max_bitrate_, encoder_target_rate_);
+  }
+  link_allocation = std::max(encoder_target_rate_, link_allocation);
   video_stream_encoder_->OnBitrateUpdated(
-      encoder_target_rate, link_allocation,
+      encoder_target_rate_, link_allocation,
       dchecked_cast<uint8_t>(update.packet_loss_ratio * 256),
       update.round_trip_time.ms(), update.cwnd_reduce_ratio);
-  stats_proxy_.OnSetEncoderTargetRate(encoder_target_rate_bps_);
-  return protection_bitrate_bps;
+  stats_proxy_.OnSetEncoderTargetRate(encoder_target_rate_.bps());
+  return protection_bitrate.bps();
 }
 
 std::optional<DataRate> VideoSendStreamImpl::GetUsedRate() const {
diff --git a/video/video_send_stream_impl.h b/video/video_send_stream_impl.h
index 88845a1969..3f2f8e02bd 100644
--- a/video/video_send_stream_impl.h
+++ b/video/video_send_stream_impl.h
@@ -239,9 +239,9 @@ class VideoSendStreamImpl : public webrtc::VideoSendStream,
   bool has_active_encodings_ RTC_GUARDED_BY(thread_checker_);
   bool disable_padding_ RTC_GUARDED_BY(thread_checker_);
   int max_padding_bitrate_ RTC_GUARDED_BY(thread_checker_);
-  int encoder_min_bitrate_bps_ RTC_GUARDED_BY(thread_checker_);
-  uint32_t encoder_max_bitrate_bps_ RTC_GUARDED_BY(thread_checker_);
-  uint32_t encoder_target_rate_bps_ RTC_GUARDED_BY(thread_checker_);
+  DataRate encoder_min_bitrate_ RTC_GUARDED_BY(thread_checker_);
+  std::optional<DataRate> encoder_max_bitrate_ RTC_GUARDED_BY(thread_checker_);
+  DataRate encoder_target_rate_ RTC_GUARDED_BY(thread_checker_);
   double encoder_bitrate_priority_ RTC_GUARDED_BY(thread_checker_);
   const int encoder_av1_priority_bitrate_override_bps_
       RTC_GUARDED_BY(thread_checker_);
diff --git a/video/video_send_stream_impl_unittest.cc b/video/video_send_stream_impl_unittest.cc
index 4bead83d26..8a8e35e4bc 100644
--- a/video/video_send_stream_impl_unittest.cc
+++ b/video/video_send_stream_impl_unittest.cc
@@ -183,6 +183,8 @@ class VideoSendStreamImplTest : public ::testing::Test {
     encoder_config.simulcast_layers.push_back(VideoStream());
     encoder_config.simulcast_layers.back().active = true;
     encoder_config.simulcast_layers.back().bitrate_priority = 1.0;
+    encoder_config.simulcast_layers.back().width = 640;
+    encoder_config.simulcast_layers.back().height = 360;
     return encoder_config;
   }
 
@@ -291,7 +293,7 @@ TEST_F(VideoSendStreamImplTest,
        MaxBitrateCorrectIfActiveEncodingUpdatedAfterCreation) {
   VideoEncoderConfig one_active_encoding = TestVideoEncoderConfig();
   ASSERT_THAT(one_active_encoding.simulcast_layers, SizeIs(1));
-  one_active_encoding.max_bitrate_bps = 10'000'000;
+  one_active_encoding.max_bitrate_bps = 2'500'000;
   one_active_encoding.simulcast_layers[0].max_bitrate_bps = 2'000'000;
   VideoEncoderConfig no_active_encodings = one_active_encoding.Copy();
   no_active_encodings.simulcast_layers[0].active = false;
@@ -307,12 +309,12 @@ TEST_F(VideoSendStreamImplTest,
   time_controller_.AdvanceTime(TimeDelta::Zero());
 
   Sequence s;
-  // Expect codec max bitrate as max needed bitrate before the encoder has
-  // notifed about the actual send streams.
-  EXPECT_CALL(bitrate_allocator_,
-              AddObserver(vss_impl.get(),
-                          Field(&MediaStreamAllocationConfig::max_bitrate_bps,
-                                Eq(one_active_encoding.max_bitrate_bps))))
+  // Expect zero max bitrate before the encoder has notified about the actual
+  // send streams.
+  EXPECT_CALL(
+      bitrate_allocator_,
+      AddObserver(vss_impl.get(),
+                  Field(&MediaStreamAllocationConfig::max_bitrate_bps, Eq(0))))
       .InSequence(s);
 
   // Expect the sum of active encodings as max needed bitrate after

From 0b6045b38b4f1f226873af976c62103e32d4dc70 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Fri, 20 Mar 2026 18:48:08 +0100
Subject: [PATCH 368/472] Prevent RTCStatsCollector crash for stopped
 transceivers

Prevent a crash in RTCStatsCollector that occurs when attempting to
process transceivers with a stopped direction.

When gathering statistics, the collector iterates through all available
transceivers. If a transceiver's current direction is set to kStopped,
the internal mapping for media information is not created during the
preparation phase and there was a missing check for that.

Bug: b/494115682
Change-Id: I0748c9d627d886aaa12118411fa94e2ab8575269
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/458680
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47221}
---
 pc/rtc_stats_collector.cc          |  4 ++++
 pc/rtc_stats_collector_unittest.cc | 32 ++++++++++++++++++++++++++++++
 2 files changed, 36 insertions(+)

diff --git a/pc/rtc_stats_collector.cc b/pc/rtc_stats_collector.cc
index 6a2325b6dd..8030dd4cae 100644
--- a/pc/rtc_stats_collector.cc
+++ b/pc/rtc_stats_collector.cc
@@ -1862,6 +1862,10 @@ void RTCStatsCollector::ProduceRTPStreamStats_s(
       continue;
     }
 
+    if (stats.current_direction == RtpTransceiverDirection::kStopped) {
+      continue;
+    }
+
     if (stats.media_type == MediaType::AUDIO) {
       ProduceAudioRTPStreamStats_s(timestamp, stats, call_stats,
                                    audio_device_stats, report);
diff --git a/pc/rtc_stats_collector_unittest.cc b/pc/rtc_stats_collector_unittest.cc
index c044f87f0c..2b5e4b072f 100644
--- a/pc/rtc_stats_collector_unittest.cc
+++ b/pc/rtc_stats_collector_unittest.cc
@@ -68,6 +68,7 @@
 #include "pc/media_stream.h"
 #include "pc/peer_connection_internal.h"
 #include "pc/rtp_sender.h"
+#include "pc/rtp_transceiver.h"
 #include "pc/sctp_data_channel.h"
 #include "pc/stream_collection.h"
 #include "pc/test/fake_audio_track.h"
@@ -102,6 +103,9 @@
 #include "test/time_controller/simulated_time_controller.h"
 
 using ::testing::_;
+using ::testing::IsEmpty;
+using ::testing::Not;
+using ::testing::NotNull;
 using ::testing::Return;
 
 namespace webrtc {
@@ -900,6 +904,34 @@ TEST_P(RTCStatsCollectorTest, ToJsonProducesParseableJson) {
   EXPECT_EQ(report->size(), json_value.size());
 }
 
+TEST_P(RTCStatsCollectorTest, ProduceRTPStreamStatsWithStoppedTransceiver) {
+  const char kTransportName[] = "transport";
+  const char kMid[] = "VideoMid";
+
+  VideoMediaInfo video_media_info;
+  video_media_info.senders.emplace_back();
+  video_media_info.senders[0].add_ssrc(1);
+
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
+  pc_->AddVideoChannel(kMid, kTransportName, video_media_info);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
+
+  // Get the transceiver and set it to stopped.
+  std::vector<scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
+      transceivers = pc_->GetTransceiversInternal();
+  ASSERT_THAT(transceivers, Not(IsEmpty()));
+  scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>> transceiver =
+      transceivers[0];
+  ASSERT_TRUE(transceiver);
+  transceiver->internal()->set_current_direction(
+      RtpTransceiverDirection::kStopped);
+
+  // This should not crash.
+  scoped_refptr<const RTCStatsReport> report =
+      stats_->GetStatsReport(main_thread_);
+  EXPECT_THAT(report, NotNull());
+}
+
 TEST_P(RTCStatsCollectorTest, CollectRTCCertificateStatsSingle) {
   const char kTransportName[] = "transport";
 

From 22e6f6cec4c8eb37f2d6196332f99f9f23de0e90 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Fri, 20 Mar 2026 13:03:12 -0700
Subject: [PATCH 369/472] Roll chromium_revision 31f6cbd695..d2fdfe540d
 (1602558:1602801)

Change log: https://chromium.googlesource.com/chromium/src/+log/31f6cbd695..d2fdfe540d
Full diff: https://chromium.googlesource.com/chromium/src/+/31f6cbd695..d2fdfe540d

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/7e958c162b..376dfbb702
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/7858fc4146..88f8168044
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/53cc79e2fc..104338fa9d
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/1ec8287478..fad81ef2da
* src/third_party/androidx/cipd: wzjBTMka_ZRUOzMrirBfHngiFx_liKeSqVeKtVlO7FwC..2cwc39P1Nuw60SuLSgzP-sHbXOgoHhlhS10ck26jV3QC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/6453bd80a9..ba3de4d9b4
* src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/59066a6625..84bfdcd752
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/3578883bb3..821ab7d3e6
* src/third_party/harfbuzz-ng/src: https://chromium.googlesource.com/external/github.com/harfbuzz/harfbuzz.git/+log/5d4e96ad8d..6f4c5cec30
* src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/a96728b9a9..9b08b4a339
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/d2aaedff4b..b041e45176
* src/third_party/tflite/src: https://chromium.googlesource.com/external/github.com/tensorflow/tensorflow.git/+log/67d816626b..8cb40fbe19
* src/third_party/xnnpack/src: https://chromium.googlesource.com/external/github.com/google/XNNPACK.git/+log/811d0bd388..4334fa7cbd
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/1582fc6219..cab35c1daf
DEPS diff: https://chromium.googlesource.com/chromium/src/+/31f6cbd695..d2fdfe540d/DEPS

BUG=None

Change-Id: Icaf66b9dd1c53e24c6fba86df560624ff95f5d71
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/458702
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47222}
---
 DEPS | 30 +++++++++++++++---------------
 1 file changed, 15 insertions(+), 15 deletions(-)

diff --git a/DEPS b/DEPS
index d868b6d374..b60f0c973a 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '31f6cbd695178167c3430e897e07cba52e316006',
+  'chromium_revision': 'd2fdfe540d234357ea185e14b9721739339ea9a7',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@7e958c162b6043bbb4b84e9d1418f8ee8f68ba34',
+    'https://chromium.googlesource.com/chromium/src/build@376dfbb70294202ba3f248d459be95c26c766423',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@95ed44cf5f06dbb5861030b91c9db9ccb4316762',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@7858fc4146da04e5e0f5583afe3aa04337ef29ba',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@88f8168044098901642bae31fb008a955a79ea2c',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@53cc79e2fca64e4075719222582ca77ff1276bc4',
+    'https://chromium.googlesource.com/chromium/src/testing@104338fa9d2b5ee90064a595b537b7403c29c74f',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@1ec8287478b97a74971870a689dec6b1c9573f9d',
+    'https://chromium.googlesource.com/chromium/src/third_party@fad81ef2daf5a05d454f80d2db57981f720471ae',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -331,13 +331,13 @@ deps = {
   'src/third_party/clang-format/script':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@c2725e0622e1a86d55f14514f2177a39efea4a0e',
   'src/third_party/compiler-rt/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@59066a662511e1f9959b07a32c88906d7212b6a7',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@84bfdcd752cb00ccc8087aeb584d3c08e9da96ba',
   'src/third_party/libc++/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@7ab65651aed6802d2599dcb7a73b1f82d5179d05',
   'src/third_party/libc++abi/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@8f11bb1d4438d0239d0dfc1bd9456a9f31629dda',
   'src/third_party/llvm-libc/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@a96728b9a94a732ec71650710949851ddf114845',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@9b08b4a339fce4aaa79c30ec303eb8edeb198290',
   'src/third_party/libunwind/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@db838d918570d4e381ecf9f5cc70a0098c9c2cd6',
 
@@ -488,7 +488,7 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@6453bd80a9fe293c0adb8ae67d5d9c233100a7b9',
+    'https://boringssl.googlesource.com/boringssl.git@ba3de4d9b41906eb7e50d69edca140ef3cec053a',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@8be0e3114685fcc1589561067282edf75ea1259a',
   'src/third_party/catapult':
@@ -521,7 +521,7 @@ deps = {
   'src/third_party/freetype/src':
     'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@99b479dc34728936b006679a31e12b8cf432fc55',
   'src/third_party/harfbuzz-ng/src':
-    'https://chromium.googlesource.com/external/github.com/harfbuzz/harfbuzz.git@5d4e96ad8d00fc871ffa17707b2ca08fa850e7d6',
+    'https://chromium.googlesource.com/external/github.com/harfbuzz/harfbuzz.git@6f4c5cec306d31e6822303f5ba248a14293d588e',
   'src/third_party/google_benchmark/src': {
     'url': 'https://chromium.googlesource.com/external/github.com/google/benchmark.git@188e8278990a9069ffc84441cb5a024fd0bede37',
   },
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'd2aaedff4b9e6fdb90edfdb57f02d79c87d23a73',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'b041e451767026d64f688b3767938c0222652b31',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@1582fc6219ee1aba33145c937c50a2f2df665f8d',
+    'https://chromium.googlesource.com/chromium/src/tools@cab35c1daff5c995890360f759455b6725af5856',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'wzjBTMka_ZRUOzMrirBfHngiFx_liKeSqVeKtVlO7FwC',
+          'version': '2cwc39P1Nuw60SuLSgzP-sHbXOgoHhlhS10ck26jV3QC',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -786,7 +786,7 @@ deps = {
   },
 
   'src/third_party/tflite/src':
-    Var('chromium_git') + '/external/github.com/tensorflow/tensorflow.git' + '@' + '67d816626bf264e3b142187768601a200cbe810d',
+    Var('chromium_git') + '/external/github.com/tensorflow/tensorflow.git' + '@' + '8cb40fbe19ed5eb65da26078b8667e31e52211e9',
 
   'src/third_party/turbine/cipd': {
       'packages': [
@@ -852,7 +852,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/google/pthreadpool.git' + '@' + '9003ee6c137cea3b94161bd5c614fb43be523ee1',
 
   'src/third_party/xnnpack/src':
-    Var('chromium_git') + '/external/github.com/google/XNNPACK.git' + '@' + '811d0bd388ba8f4853c5560b6e16a5af3e8b895a',
+    Var('chromium_git') + '/external/github.com/google/XNNPACK.git' + '@' + '4334fa7cbdf8e276db976792cd21d8614910d569',
 
   'src/third_party/farmhash/src':
     Var('chromium_git') + '/external/github.com/google/farmhash.git' + '@' + '816a4ae622e964763ca0862d9dbd19324a1eaf45',
@@ -864,7 +864,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7607ca500436b37ad23fb8d18614bec7796b68a7',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '3578883bb3616208b8f9bb6fa03dc1fda0a16a78',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '821ab7d3e656e6928d342e289efaf208017d4274',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From b00f47926a3c5c0e1ee6358b0f1a61742deee8ca Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Sat, 21 Mar 2026 21:07:47 -0700
Subject: [PATCH 370/472] Update WebRTC code version (2026-03-22T04:07:46).

Bug: None
Change-Id: Id9b95f270d54d14ac15bc8ca755339c9b5433008
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/459021
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47223}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index 6d4be36a08..cca624b266 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-20T04:01:59";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-22T04:07:46";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From 1aacbbfc7a576e2d365de6f974e93fe311709ed4 Mon Sep 17 00:00:00 2001
From: kwasniow <kwasniow@cisco.com>
Date: Tue, 17 Mar 2026 10:28:09 +0100
Subject: [PATCH 371/472] Sframe API support for sender and receiver

This CL introduces the public API surface for Sframe (Secure Frame)
encryption and decryption on RTP senders and receivers. The actual
pipeline wiring is deferred to a follow-up CL; all factory methods
currently return nullptr.

New API headers (api/sframe/):

- sframe_types.h: shared enums SframeMode (per-frame / per-packet)
  and SframeCipherSuite (AES-CTR-HMAC variants, AES-GCM-128/256).
- sframe_encrypter_interface.h: SframeEncrypterInit config struct
  and SframeEncrypterInterface for sender-side key management
  (SetEncryptionKey).
- sframe_decrypter_interface.h: SframeDecrypterInit config struct
  and SframeDecrypterInterface for receiver-side key management
  (AddDecryptionKey, RemoveDecryptionKey).

Interface changes:

- RtpSenderInterface gains CreateSframeEncrypter(SframeEncrypterInit)
- RtpReceiverInterface gains CreateSframeDecrypter(SframeDecrypterInit)
- Both proxy files updated with corresponding PROXY_METHOD entries.

New RtpReceiverBase class introduced inheriting RtpReceiverInternal,
providing CreateSframeDecrypter. This mirrors the existing RtpSenderBase
pattern. AudioRtpReceiver and VideoRtpReceiver now inherit
RtpReceiverBase instead of RtpReceiverInternal, eliminating future code
duplication.

Bug: webrtc:479862368
Change-Id: I2d62b61f2d10325d8dca9f15f08a0eb0509eed01
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452260
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47224}
---
 api/BUILD.gn                            | 33 ++++++++++++++++++++
 api/rtp_receiver_interface.h            | 15 ++++++++++
 api/rtp_sender_interface.h              | 13 ++++++++
 api/sframe/sframe_decrypter_interface.h | 36 ++++++++++++++++++++++
 api/sframe/sframe_encrypter_interface.h | 40 +++++++++++++++++++++++++
 api/sframe/sframe_types.h               | 31 +++++++++++++++++++
 pc/BUILD.gn                             | 12 ++++++--
 pc/audio_rtp_receiver.cc                |  3 +-
 pc/audio_rtp_receiver.h                 |  5 +---
 pc/rtp_receiver.cc                      | 16 ++++++++++
 pc/rtp_receiver.h                       | 18 +++++++++++
 pc/rtp_receiver_proxy.h                 |  6 ++++
 pc/rtp_sender.cc                        | 10 +++++++
 pc/rtp_sender.h                         |  4 +++
 pc/rtp_sender_proxy.h                   |  4 +++
 pc/video_rtp_receiver.cc                |  3 +-
 pc/video_rtp_receiver.h                 |  6 +---
 17 files changed, 242 insertions(+), 13 deletions(-)
 create mode 100644 api/sframe/sframe_decrypter_interface.h
 create mode 100644 api/sframe/sframe_encrypter_interface.h
 create mode 100644 api/sframe/sframe_types.h

diff --git a/api/BUILD.gn b/api/BUILD.gn
index d7587e6fef..d7f763f96c 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -261,6 +261,31 @@ rtc_library("dtmf_sender_interface") {
   deps = [ ":ref_count" ]
 }
 
+rtc_library("sframe_types") {
+  visibility = [ "*" ]
+  sources = [ "sframe/sframe_types.h" ]
+}
+
+rtc_library("sframe_encrypter_interface") {
+  visibility = [ "*" ]
+  sources = [ "sframe/sframe_encrypter_interface.h" ]
+  deps = [
+    ":ref_count",
+    ":rtc_error",
+    ":sframe_types",
+  ]
+}
+
+rtc_library("sframe_decrypter_interface") {
+  visibility = [ "*" ]
+  sources = [ "sframe/sframe_decrypter_interface.h" ]
+  deps = [
+    ":ref_count",
+    ":rtc_error",
+    ":sframe_types",
+  ]
+}
+
 rtc_library("rtp_sender_interface") {
   visibility = [ "*" ]
 
@@ -277,6 +302,7 @@ rtc_library("rtp_sender_interface") {
     ":rtc_error",
     ":rtp_parameters",
     ":scoped_refptr",
+    ":sframe_encrypter_interface",
     "../rtc_base:checks",
     "../rtc_base/system:rtc_export",
     "crypto:frame_encryptor_interface",
@@ -477,8 +503,12 @@ rtc_library("rtp_receiver_interface") {
     ":frame_transformer_interface",
     ":media_stream_interface",
     ":ref_count",
+    ":rtc_error",
     ":rtp_parameters",
     ":scoped_refptr",
+    ":sframe_decrypter_interface",
+    ":sframe_types",
+    "../rtc_base:checks",
     "../rtc_base/system:rtc_export",
     "crypto:frame_decryptor_interface",
     "transport/rtp:rtp_source",
@@ -599,6 +629,9 @@ rtc_library("libjingle_peerconnection_api") {
     ":sctp_transport_interface",
     ":sequence_checker",
     ":set_local_description_observer_interface",
+    ":sframe_decrypter_interface",
+    ":sframe_encrypter_interface",
+    ":sframe_types",
     ":turn_customizer",
     "../call:rtp_interfaces",
     "../media:media_engine",
diff --git a/api/rtp_receiver_interface.h b/api/rtp_receiver_interface.h
index 9ac347b45f..7dae849c48 100644
--- a/api/rtp_receiver_interface.h
+++ b/api/rtp_receiver_interface.h
@@ -25,9 +25,13 @@
 #include "api/media_stream_interface.h"
 #include "api/media_types.h"
 #include "api/ref_count.h"
+#include "api/rtc_error.h"
 #include "api/rtp_parameters.h"
 #include "api/scoped_refptr.h"
+#include "api/sframe/sframe_decrypter_interface.h"
+#include "api/sframe/sframe_types.h"
 #include "api/transport/rtp/rtp_source.h"
+#include "rtc_base/checks.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
@@ -131,6 +135,17 @@ class RTC_EXPORT RtpReceiverInterface : public RefCountInterface,
   void SetFrameTransformer(
       scoped_refptr<FrameTransformerInterface> frame_transformer) override;
 
+  // Creates an internal Sframe decrypter and returns a handle for key
+  // management.
+  // Default implementation of CreateSframeDecrypterOrError.
+  // TODO: issues.webrtc.org/479862368 - make pure virtual when all
+  // implementations are updated
+  virtual RTCErrorOr<scoped_refptr<SframeDecrypterInterface>>
+  CreateSframeDecrypterOrError(SframeCipherSuite cipher_suite) {
+    RTC_DCHECK_NOTREACHED();
+    return RTCError();
+  }
+
  protected:
   ~RtpReceiverInterface() override = default;
 };
diff --git a/api/rtp_sender_interface.h b/api/rtp_sender_interface.h
index be839285bf..5ea6efee18 100644
--- a/api/rtp_sender_interface.h
+++ b/api/rtp_sender_interface.h
@@ -31,7 +31,9 @@
 #include "api/rtc_error.h"
 #include "api/rtp_parameters.h"
 #include "api/scoped_refptr.h"
+#include "api/sframe/sframe_encrypter_interface.h"
 #include "api/video_codecs/video_encoder_factory.h"
+#include "rtc_base/checks.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
@@ -135,6 +137,17 @@ class RTC_EXPORT RtpSenderInterface : public RefCountInterface,
   void SetFrameTransformer(scoped_refptr<FrameTransformerInterface>
                            /* frame_transformer */) override {}
 
+  // Creates an internal Sframe encrypter and returns a handle for key
+  // management.
+  // Default implementation of CreateSframeEncrypterOrError.
+  // TODO: bugs.webrtc.org/479862368 - remove when all implementations are
+  // updated
+  virtual RTCErrorOr<scoped_refptr<SframeEncrypterInterface>>
+  CreateSframeEncrypterOrError(const SframeEncrypterInit& options) {
+    RTC_DCHECK_NOTREACHED();
+    return RTCError();
+  }
+
   // TODO(crbug.com/1354101): make pure virtual again after Chrome roll.
   virtual RTCError GenerateKeyFrame(const std::vector<std::string>& rids) {
     return RTCError::OK();
diff --git a/api/sframe/sframe_decrypter_interface.h b/api/sframe/sframe_decrypter_interface.h
new file mode 100644
index 0000000000..7ad85bd1ed
--- /dev/null
+++ b/api/sframe/sframe_decrypter_interface.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright 2026 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_SFRAME_SFRAME_DECRYPTER_INTERFACE_H_
+#define API_SFRAME_SFRAME_DECRYPTER_INTERFACE_H_
+
+#include <cstdint>
+#include <span>
+
+#include "api/ref_count.h"
+#include "api/rtc_error.h"
+
+namespace webrtc {
+
+// Key management handle for Sframe receiver decryption.
+class SframeDecrypterInterface : public RefCountInterface {
+ public:
+  virtual RTCError AddDecryptionKey(uint64_t key_id,
+                                    std::span<const uint8_t> key_material) = 0;
+
+  virtual RTCError RemoveDecryptionKey(uint64_t key_id) = 0;
+
+ protected:
+  ~SframeDecrypterInterface() override = default;
+};
+
+}  // namespace webrtc
+
+#endif  // API_SFRAME_SFRAME_DECRYPTER_INTERFACE_H_
diff --git a/api/sframe/sframe_encrypter_interface.h b/api/sframe/sframe_encrypter_interface.h
new file mode 100644
index 0000000000..60babc3389
--- /dev/null
+++ b/api/sframe/sframe_encrypter_interface.h
@@ -0,0 +1,40 @@
+/*
+ *  Copyright 2026 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_SFRAME_SFRAME_ENCRYPTER_INTERFACE_H_
+#define API_SFRAME_SFRAME_ENCRYPTER_INTERFACE_H_
+
+#include <cstdint>
+#include <span>
+
+#include "api/ref_count.h"
+#include "api/rtc_error.h"
+#include "api/sframe/sframe_types.h"
+
+namespace webrtc {
+
+struct SframeEncrypterInit {
+  SframeMode mode;
+  SframeCipherSuite cipher_suite;
+};
+
+// Key management handle for Sframe sender encryption.
+class SframeEncrypterInterface : public RefCountInterface {
+ public:
+  virtual RTCError SetEncryptionKey(uint64_t key_id,
+                                    std::span<const uint8_t> key_material) = 0;
+
+ protected:
+  ~SframeEncrypterInterface() override = default;
+};
+
+}  // namespace webrtc
+
+#endif  // API_SFRAME_SFRAME_ENCRYPTER_INTERFACE_H_
diff --git a/api/sframe/sframe_types.h b/api/sframe/sframe_types.h
new file mode 100644
index 0000000000..2414bf3d9f
--- /dev/null
+++ b/api/sframe/sframe_types.h
@@ -0,0 +1,31 @@
+/*
+ *  Copyright 2026 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_SFRAME_SFRAME_TYPES_H_
+#define API_SFRAME_SFRAME_TYPES_H_
+
+namespace webrtc {
+
+enum class SframeMode {
+  kPerFrame,
+  kPerPacket,
+};
+
+enum class SframeCipherSuite {
+  kAes128CtrHmacSha256_80,
+  kAes128CtrHmacSha256_64,
+  kAes128CtrHmacSha256_32,
+  kAes128GcmSha256_128,
+  kAes256GcmSha512_128,
+};
+
+}  // namespace webrtc
+
+#endif  // API_SFRAME_SFRAME_TYPES_H_
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 88287d5ea9..8a576969ae 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -547,9 +547,12 @@ rtc_source_set("rtp_receiver_proxy") {
     "../api:dtls_transport_interface",
     "../api:frame_transformer_interface",
     "../api:media_stream_interface",
+    "../api:rtc_error",
     "../api:rtp_parameters",
     "../api:rtp_receiver_interface",
     "../api:scoped_refptr",
+    "../api:sframe_decrypter_interface",
+    "../api:sframe_types",
     "../api/crypto:frame_decryptor_interface",
     "../api/transport/rtp:rtp_source",
   ]
@@ -568,6 +571,7 @@ rtc_source_set("rtp_sender_proxy") {
     "../api:rtp_parameters",
     "../api:rtp_sender_interface",
     "../api:scoped_refptr",
+    "../api:sframe_encrypter_interface",
     "../api/crypto:frame_encryptor_interface",
     "../api/video_codecs:video_codecs_api",
   ]
@@ -1806,10 +1810,15 @@ rtc_library("rtp_receiver") {
     ":media_stream_proxy",
     "../api:dtls_transport_interface",
     "../api:media_stream_interface",
+    "../api:rtc_error",
     "../api:rtp_receiver_interface",
     "../api:scoped_refptr",
+    "../api:sequence_checker",
+    "../api:sframe_decrypter_interface",
+    "../api:sframe_types",
     "../media:media_channel",
     "../rtc_base:threading",
+    "../rtc_base/system:no_unique_address",
     "//third_party/abseil-cpp/absl/functional:any_invocable",
   ]
 }
@@ -1841,7 +1850,6 @@ rtc_library("audio_rtp_receiver") {
     "../rtc_base:checks",
     "../rtc_base:macromagic",
     "../rtc_base:threading",
-    "../rtc_base/system:no_unique_address",
     "//third_party/abseil-cpp/absl/functional:any_invocable",
     "//third_party/abseil-cpp/absl/strings:string_view",
   ]
@@ -1876,7 +1884,6 @@ rtc_library("video_rtp_receiver") {
     "../rtc_base:logging",
     "../rtc_base:macromagic",
     "../rtc_base:threading",
-    "../rtc_base/system:no_unique_address",
     "//third_party/abseil-cpp/absl/functional:any_invocable",
     "//third_party/abseil-cpp/absl/strings:string_view",
   ]
@@ -2006,6 +2013,7 @@ rtc_library("rtp_sender") {
     "../api:rtp_sender_interface",
     "../api:scoped_refptr",
     "../api:sequence_checker",
+    "../api:sframe_encrypter_interface",
     "../api/crypto:frame_encryptor_interface",
     "../api/environment",
     "../api/task_queue",
diff --git a/pc/audio_rtp_receiver.cc b/pc/audio_rtp_receiver.cc
index b90e70a863..ab3fbada46 100644
--- a/pc/audio_rtp_receiver.cc
+++ b/pc/audio_rtp_receiver.cc
@@ -34,6 +34,7 @@
 #include "pc/audio_track.h"
 #include "pc/media_stream_track_proxy.h"
 #include "pc/remote_audio_source.h"
+#include "pc/rtp_receiver.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/thread.h"
 
@@ -81,7 +82,7 @@ AudioRtpReceiver::AudioRtpReceiver(
     const std::vector<scoped_refptr<MediaStreamInterface>>& streams,
     VoiceMediaReceiveChannelInterface* voice_channel,
     RemoteAudioSource::OnAudioChannelGoneAction source_gone_action)
-    : worker_thread_(worker_thread),
+    : RtpReceiverBase(worker_thread),
       id_(receiver_id),
       source_(make_ref_counted<RemoteAudioSource>(worker_thread,
                                                   source_gone_action)),
diff --git a/pc/audio_rtp_receiver.h b/pc/audio_rtp_receiver.h
index a8b07a0186..f89ab65976 100644
--- a/pc/audio_rtp_receiver.h
+++ b/pc/audio_rtp_receiver.h
@@ -36,7 +36,6 @@
 #include "pc/media_stream_track_proxy.h"
 #include "pc/remote_audio_source.h"
 #include "pc/rtp_receiver.h"
-#include "rtc_base/system/no_unique_address.h"
 #include "rtc_base/thread.h"
 #include "rtc_base/thread_annotations.h"
 
@@ -44,7 +43,7 @@ namespace webrtc {
 
 class AudioRtpReceiver : public ObserverInterface,
                          public AudioSourceInterface::AudioObserver,
-                         public RtpReceiverInternal {
+                         public RtpReceiverBase {
  public:
   // The constructor supports optionally passing the voice channel to the
   // instance at construction time without having to call `SetMediaChannel()`
@@ -147,8 +146,6 @@ class AudioRtpReceiver : public ObserverInterface,
   void Reconfigure(bool track_enabled) RTC_RUN_ON(worker_thread_);
   void SetOutputVolume_w(double volume) RTC_RUN_ON(worker_thread_);
 
-  RTC_NO_UNIQUE_ADDRESS SequenceChecker signaling_thread_checker_;
-  Thread* const worker_thread_;
   const std::string id_;
   const scoped_refptr<RemoteAudioSource> source_;
   const scoped_refptr<AudioTrackProxyWithInternal<AudioTrack>> track_;
diff --git a/pc/rtp_receiver.cc b/pc/rtp_receiver.cc
index 31a20df17a..33dddb872d 100644
--- a/pc/rtp_receiver.cc
+++ b/pc/rtp_receiver.cc
@@ -17,7 +17,11 @@
 #include <vector>
 
 #include "api/media_stream_interface.h"
+#include "api/rtc_error.h"
 #include "api/scoped_refptr.h"
+#include "api/sequence_checker.h"
+#include "api/sframe/sframe_decrypter_interface.h"
+#include "api/sframe/sframe_types.h"
 #include "pc/media_stream.h"
 #include "pc/media_stream_proxy.h"
 #include "rtc_base/thread.h"
@@ -43,4 +47,16 @@ RtpReceiverInternal::CreateStreamsFromIds(std::vector<std::string> stream_ids) {
   return streams;
 }
 
+RtpReceiverBase::RtpReceiverBase(Thread* worker_thread)
+    : worker_thread_(worker_thread) {}
+
+RTCErrorOr<scoped_refptr<SframeDecrypterInterface>>
+RtpReceiverBase::CreateSframeDecrypterOrError(SframeCipherSuite cipher_suite) {
+  RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
+  // TODO(bugs.webrtc.org/479862368): Create the internal Sframe decryption
+  // pipeline and return a key management handle.
+  return RTCError(RTCErrorType::UNSUPPORTED_OPERATION,
+                  "Sframe decrypter not yet implemented");
+}
+
 }  // namespace webrtc
diff --git a/pc/rtp_receiver.h b/pc/rtp_receiver.h
index 45399b684d..6a2b21dc33 100644
--- a/pc/rtp_receiver.h
+++ b/pc/rtp_receiver.h
@@ -24,9 +24,15 @@
 #include "absl/functional/any_invocable.h"
 #include "api/dtls_transport_interface.h"
 #include "api/media_stream_interface.h"
+#include "api/rtc_error.h"
 #include "api/rtp_receiver_interface.h"
 #include "api/scoped_refptr.h"
+#include "api/sequence_checker.h"
+#include "api/sframe/sframe_decrypter_interface.h"
+#include "api/sframe/sframe_types.h"
 #include "media/base/media_channel.h"
+#include "rtc_base/system/no_unique_address.h"
+#include "rtc_base/thread.h"
 
 namespace webrtc {
 
@@ -97,6 +103,18 @@ class RtpReceiverInternal : public RtpReceiverInterface {
       std::vector<std::string> stream_ids);
 };
 
+class RtpReceiverBase : public RtpReceiverInternal {
+ public:
+  RTCErrorOr<scoped_refptr<SframeDecrypterInterface>>
+  CreateSframeDecrypterOrError(SframeCipherSuite cipher_suite) override;
+
+ protected:
+  explicit RtpReceiverBase(Thread* worker_thread);
+
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker signaling_thread_checker_;
+  Thread* const worker_thread_;
+};
+
 }  // namespace webrtc
 
 #endif  // PC_RTP_RECEIVER_H_
diff --git a/pc/rtp_receiver_proxy.h b/pc/rtp_receiver_proxy.h
index 223b307030..2022cbf079 100644
--- a/pc/rtp_receiver_proxy.h
+++ b/pc/rtp_receiver_proxy.h
@@ -20,9 +20,12 @@
 #include "api/frame_transformer_interface.h"
 #include "api/media_stream_interface.h"
 #include "api/media_types.h"
+#include "api/rtc_error.h"
 #include "api/rtp_parameters.h"
 #include "api/rtp_receiver_interface.h"
 #include "api/scoped_refptr.h"
+#include "api/sframe/sframe_decrypter_interface.h"
+#include "api/sframe/sframe_types.h"
 #include "api/transport/rtp/rtp_source.h"
 #include "pc/proxy.h"
 
@@ -55,6 +58,9 @@ PROXY_SECONDARY_CONSTMETHOD0(scoped_refptr<FrameDecryptorInterface>,
 PROXY_SECONDARY_METHOD1(void,
                         SetFrameTransformer,
                         scoped_refptr<FrameTransformerInterface>)
+PROXY_METHOD1(RTCErrorOr<scoped_refptr<SframeDecrypterInterface>>,
+              CreateSframeDecrypterOrError,
+              SframeCipherSuite)
 END_PROXY_MAP(RtpReceiver)
 
 }  // namespace webrtc
diff --git a/pc/rtp_sender.cc b/pc/rtp_sender.cc
index 3a4e6b778f..7af007cfa8 100644
--- a/pc/rtp_sender.cc
+++ b/pc/rtp_sender.cc
@@ -38,6 +38,7 @@
 #include "api/rtp_sender_interface.h"
 #include "api/scoped_refptr.h"
 #include "api/sequence_checker.h"
+#include "api/sframe/sframe_encrypter_interface.h"
 #include "api/task_queue/pending_task_safety_flag.h"
 #include "api/task_queue/task_queue_base.h"
 #include "api/video_codecs/video_encoder_factory.h"
@@ -916,6 +917,15 @@ void RtpSenderBase::SetFrameTransformer(
   }
 }
 
+RTCErrorOr<scoped_refptr<SframeEncrypterInterface>>
+RtpSenderBase::CreateSframeEncrypterOrError(
+    const SframeEncrypterInit& options) {
+  RTC_DCHECK_RUN_ON(signaling_thread_);
+  // TODO(bugs.webrtc.org/479862368): Implement Sframe encrypter creation.
+  return RTCError(RTCErrorType::UNSUPPORTED_OPERATION,
+                  "Sframe encrypter not yet implemented");
+}
+
 LocalAudioSinkAdapter::LocalAudioSinkAdapter() : sink_(nullptr) {}
 
 LocalAudioSinkAdapter::~LocalAudioSinkAdapter() {
diff --git a/pc/rtp_sender.h b/pc/rtp_sender.h
index 848db55281..eeec9ae1c8 100644
--- a/pc/rtp_sender.h
+++ b/pc/rtp_sender.h
@@ -37,6 +37,7 @@
 #include "api/rtp_sender_interface.h"
 #include "api/scoped_refptr.h"
 #include "api/sequence_checker.h"
+#include "api/sframe/sframe_encrypter_interface.h"
 #include "api/task_queue/pending_task_safety_flag.h"
 #include "api/task_queue/task_queue_base.h"
 #include "api/video_codecs/video_encoder_factory.h"
@@ -224,6 +225,9 @@ class RtpSenderBase : public RtpSenderInternal, public ObserverInterface {
   void SetFrameTransformer(
       scoped_refptr<FrameTransformerInterface> frame_transformer) override;
 
+  RTCErrorOr<scoped_refptr<SframeEncrypterInterface>>
+  CreateSframeEncrypterOrError(const SframeEncrypterInit& options) override;
+
   void SetEncoderSelector(
       std::unique_ptr<VideoEncoderFactory::EncoderSelectorInterface>
           encoder_selector) override;
diff --git a/pc/rtp_sender_proxy.h b/pc/rtp_sender_proxy.h
index 8e609c5e43..74d28b55ed 100644
--- a/pc/rtp_sender_proxy.h
+++ b/pc/rtp_sender_proxy.h
@@ -26,6 +26,7 @@
 #include "api/rtp_parameters.h"
 #include "api/rtp_sender_interface.h"
 #include "api/scoped_refptr.h"
+#include "api/sframe/sframe_encrypter_interface.h"
 #include "api/video_codecs/video_encoder_factory.h"
 #include "pc/proxy.h"
 
@@ -58,6 +59,9 @@ PROXY_METHOD1(void, SetStreams, const std::vector<std::string>&)
 PROXY_METHOD1(void,
               SetFrameTransformer,
               scoped_refptr<FrameTransformerInterface>)
+PROXY_METHOD1(RTCErrorOr<scoped_refptr<SframeEncrypterInterface>>,
+              CreateSframeEncrypterOrError,
+              const SframeEncrypterInit&)
 PROXY_METHOD1(void,
               SetEncoderSelector,
               std::unique_ptr<VideoEncoderFactory::EncoderSelectorInterface>)
diff --git a/pc/video_rtp_receiver.cc b/pc/video_rtp_receiver.cc
index b296c818a5..250caa90d7 100644
--- a/pc/video_rtp_receiver.cc
+++ b/pc/video_rtp_receiver.cc
@@ -34,6 +34,7 @@
 #include "api/video/video_sink_interface.h"
 #include "media/base/media_channel.h"
 #include "pc/media_stream_track_proxy.h"
+#include "pc/rtp_receiver.h"
 #include "pc/video_rtp_track_source.h"
 #include "pc/video_track.h"
 #include "rtc_base/checks.h"
@@ -57,7 +58,7 @@ VideoRtpReceiver::VideoRtpReceiver(
     absl::string_view receiver_id,
     const std::vector<scoped_refptr<MediaStreamInterface>>& streams,
     VideoMediaReceiveChannelInterface* media_channel)
-    : worker_thread_(worker_thread),
+    : RtpReceiverBase(worker_thread),
       id_(receiver_id),
       media_channel_(media_channel),
       source_(make_ref_counted<VideoRtpTrackSource>(&source_callback_)),
diff --git a/pc/video_rtp_receiver.h b/pc/video_rtp_receiver.h
index ec0d8bb39b..552539ded8 100644
--- a/pc/video_rtp_receiver.h
+++ b/pc/video_rtp_receiver.h
@@ -37,13 +37,12 @@
 #include "pc/rtp_receiver.h"
 #include "pc/video_rtp_track_source.h"
 #include "pc/video_track.h"
-#include "rtc_base/system/no_unique_address.h"
 #include "rtc_base/thread.h"
 #include "rtc_base/thread_annotations.h"
 
 namespace webrtc {
 
-class VideoRtpReceiver : public RtpReceiverInternal {
+class VideoRtpReceiver : public RtpReceiverBase {
  public:
   // An SSRC of 0 will create a receiver that will match the first SSRC it
   // sees. Must be called on signaling thread.
@@ -146,9 +145,6 @@ class VideoRtpReceiver : public RtpReceiverInternal {
     VideoRtpReceiver* const receiver_;
   } source_callback_{this};
 
-  RTC_NO_UNIQUE_ADDRESS SequenceChecker signaling_thread_checker_;
-  Thread* const worker_thread_;
-
   const std::string id_;
   VideoMediaReceiveChannelInterface* media_channel_
       RTC_GUARDED_BY(worker_thread_) = nullptr;

From 47fddff466253adf07115b778b0f132c72d9e034 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Erik=20Spr=C3=A5ng?= <sprang@webrtc.org>
Date: Fri, 20 Mar 2026 17:01:32 +0100
Subject: [PATCH 372/472] Run simulcast adapter test
 ConcurrentEncodeAndOnEncodedImage on TSAN.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

The test case
TestSimulcastEncoderAdapterFake.ConcurrentEncodeAndOnEncodedImage was
added to repro a tricky race condition. But the test doesn't make much
sense unless TSAN thread sanitizer is enabled - otherwise it mostly
wastes resources. This CL disables it unless TSAN is on.

Bug: webrtc:467444018
Change-Id: I7350483892dfa2a9b3003d09a3013368d83ced2f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/458681
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47225}
---
 media/engine/simulcast_encoder_adapter_unittest.cc | 13 ++++++++++++-
 1 file changed, 12 insertions(+), 1 deletion(-)

diff --git a/media/engine/simulcast_encoder_adapter_unittest.cc b/media/engine/simulcast_encoder_adapter_unittest.cc
index df2894c339..5c10aee2dc 100644
--- a/media/engine/simulcast_encoder_adapter_unittest.cc
+++ b/media/engine/simulcast_encoder_adapter_unittest.cc
@@ -2840,7 +2840,18 @@ TEST_F(TestSimulcastEncoderAdapterFake,
             adapter_->Encode(input_frame, &frame_types));
 }
 
-TEST_F(TestSimulcastEncoderAdapterFake, ConcurrentEncodeAndOnEncodedImage) {
+// Only run this test under TSAN, otherwise it doesn't make much sense and
+// mostly wastes CPU resources.
+#if defined(THREAD_SANITIZER)
+#define MAYBE_ConcurrentEncodeAndOnEncodedImage \
+  ConcurrentEncodeAndOnEncodedImage
+#else
+#define MAYBE_ConcurrentEncodeAndOnEncodedImage \
+  DISABLED_ConcurrentEncodeAndOnEncodedImage
+#endif
+
+TEST_F(TestSimulcastEncoderAdapterFake,
+       MAYBE_ConcurrentEncodeAndOnEncodedImage) {
   // The test setup here is quite complex, but that is needed in order to make
   // sure tools like TSAN is able to reliable detect issue in the wrapper
   // encoder instances used for each stream. The code below makes sure that

From 4f39d459159ac9a4e1a1d238f5181c1bbd1041bd Mon Sep 17 00:00:00 2001
From: Philipp Hancke <philipp.hancke@googlemail.com>
Date: Fri, 20 Mar 2026 13:59:49 +0100
Subject: [PATCH 373/472] build: remove unnecessary targets in video_coding

using
  xargs tools_webrtc/gn_check_autofix.py -r modules/video_coding/codecs/av1/BUILD.gn modules/video_coding/svc/BUILD.gn modules/video_coding/timing/BUILD.gn
followed by
  tools_webrtc/gn_check_autofix -C out/Default
with some manual reverts because of the conditional libaom dependency.

Bug: webrtc:467294026
Change-Id: I7269d0dd4139ca8ef70457e440d6fb0840710a55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/458620
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Christoffer Dewerin <jansson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47226}
---
 modules/video_coding/codecs/av1/BUILD.gn | 11 ++---------
 modules/video_coding/svc/BUILD.gn        |  7 ++-----
 modules/video_coding/timing/BUILD.gn     | 11 +----------
 3 files changed, 5 insertions(+), 24 deletions(-)

diff --git a/modules/video_coding/codecs/av1/BUILD.gn b/modules/video_coding/codecs/av1/BUILD.gn
index 1e94e36c81..5d6156f3a7 100644
--- a/modules/video_coding/codecs/av1/BUILD.gn
+++ b/modules/video_coding/codecs/av1/BUILD.gn
@@ -61,9 +61,8 @@ rtc_library("libaom_av1_encoder") {
   ]
   deps = [
     ":libaom_speed_config_factory",
+    "../..:frame_sampler",
     "../..:video_codec_interface",
-    "../..:video_coding_utility",
-    "../../:frame_sampler",
     "../../../../api:field_trials_view",
     "../../../../api:scoped_refptr",
     "../../../../api/environment",
@@ -75,19 +74,16 @@ rtc_library("libaom_av1_encoder") {
     "../../../../api/video:video_frame",
     "../../../../api/video:video_frame_type",
     "../../../../api/video:video_rtp_headers",
-    "../../../../api/video_codecs:encoder_speed_controller_factory",
     "../../../../api/video_codecs:scalability_mode",
     "../../../../api/video_codecs:video_codecs_api",
-    "../../../../common_video",
     "../../../../common_video/generic_frame_descriptor",
-    "../../../../modules/rtp_rtcp:rtp_rtcp_format",
     "../../../../rtc_base:checks",
     "../../../../rtc_base:logging",
-    "../../../../rtc_base:rtc_numerics",
     "../../../../rtc_base/experiments:encoder_info_settings",
     "../../../../rtc_base/experiments:encoder_speed_experiment",
     "../../../../rtc_base/experiments:psnr_experiment",
     "../../../../system_wrappers",
+    "../../../rtp_rtcp:rtp_rtcp_format",
     "../../svc:scalability_structures",
     "../../svc:scalable_video_controller",
     "//third_party/abseil-cpp/absl/algorithm:container",
@@ -105,12 +101,9 @@ rtc_library("libaom_speed_config_factory") {
     "libaom_speed_config_factory.h",
   ]
   deps = [
-    "../..:video_codec_interface",
-    "../..:video_coding_utility",
     "../../../../api:field_trials_view",
     "../../../../api/units:time_delta",
     "../../../../api/video_codecs:video_codecs_api",
-    "../../../../rtc_base:logging",
     "../../../../rtc_base/experiments:psnr_experiment",
   ]
 }
diff --git a/modules/video_coding/svc/BUILD.gn b/modules/video_coding/svc/BUILD.gn
index 094f44e249..7a7864b645 100644
--- a/modules/video_coding/svc/BUILD.gn
+++ b/modules/video_coding/svc/BUILD.gn
@@ -98,12 +98,12 @@ rtc_library("simulcast_to_svc_converter") {
     ":scalability_structures",
     ":scalable_video_controller",
     "..:codec_globals_headers",
+    "..:video_codec_interface",
+    "..:video_coding_utility",
     "../../../api/video:encoded_image",
     "../../../api/video:video_bitrate_allocation",
     "../../../api/video_codecs:scalability_mode",
     "../../../api/video_codecs:video_codecs_api",
-    "../../../modules/video_coding:video_codec_interface",
-    "../../../modules/video_coding:video_coding_utility",
     "../../../rtc_base:checks",
     "../../../rtc_base/system:rtc_export",
   ]
@@ -125,14 +125,12 @@ if (rtc_include_tests) {
       ":scalability_mode_util",
       ":scalability_structures",
       ":scalable_video_controller",
-      ":simulcast_to_svc_converter",
       "..:chain_diff_calculator",
       "..:frame_dependencies_calculator",
       "../../../api:array_view",
       "../../../api/transport/rtp:dependency_descriptor",
       "../../../api/video:video_bitrate_allocation",
       "../../../api/video:video_frame",
-      "../../../api/video:video_frame_type",
       "../../../api/video_codecs:scalability_mode",
       "../../../api/video_codecs:video_codecs_api",
       "../../../common_video/generic_frame_descriptor",
@@ -178,7 +176,6 @@ if (rtc_include_tests) {
       "../../../api/video:video_frame",
       "../../../api/video_codecs:scalability_mode",
       "../../../api/video_codecs:video_codecs_api",
-      "../../../rtc_base:checks",
       "../../../test:test_support",
     ]
   }
diff --git a/modules/video_coding/timing/BUILD.gn b/modules/video_coding/timing/BUILD.gn
index ea1862f98f..75de611cce 100644
--- a/modules/video_coding/timing/BUILD.gn
+++ b/modules/video_coding/timing/BUILD.gn
@@ -35,11 +35,7 @@ rtc_library("frame_delay_variation_kalman_filter") {
     "frame_delay_variation_kalman_filter.cc",
     "frame_delay_variation_kalman_filter.h",
   ]
-  deps = [
-    "../../../api/units:data_size",
-    "../../../api/units:time_delta",
-    "../../../rtc_base:checks",
-  ]
+  deps = [ "../../../rtc_base:checks" ]
   visibility = [
     ":jitter_estimator",
     ":timing_unittests",
@@ -63,7 +59,6 @@ rtc_library("jitter_estimator") {
     "../../../rtc_base:logging",
     "../../../rtc_base:rolling_accumulator",
     "../../../rtc_base:rtc_numerics",
-    "../../../rtc_base:safe_conversions",
     "../../../rtc_base/experiments:field_trial_parser",
     "../../../system_wrappers",
     "//third_party/abseil-cpp/absl/strings:string_view",
@@ -92,7 +87,6 @@ rtc_library("timestamp_extrapolator") {
     "../../../api:field_trials_view",
     "../../../api/units:time_delta",
     "../../../api/units:timestamp",
-    "../../../modules:module_api_public",
     "../../../rtc_base:logging",
     "../../../rtc_base:rtc_numerics",
     "../../../rtc_base/experiments:field_trial_parser",
@@ -117,7 +111,6 @@ rtc_library("timing_module") {
     "../../../rtc_base:checks",
     "../../../rtc_base:logging",
     "../../../rtc_base:macromagic",
-    "../../../rtc_base:rtc_numerics",
     "../../../rtc_base/experiments:field_trial_parser",
     "../../../rtc_base/synchronization:mutex",
     "../../../system_wrappers",
@@ -141,14 +134,12 @@ rtc_library("timing_unittests") {
     ":rtt_filter",
     ":timestamp_extrapolator",
     ":timing_module",
-    "../../../api:array_view",
     "../../../api:field_trials",
     "../../../api/units:data_size",
     "../../../api/units:frequency",
     "../../../api/units:time_delta",
     "../../../api/units:timestamp",
     "../../../rtc_base:histogram_percentile_counter",
-    "../../../rtc_base:timeutils",
     "../../../system_wrappers",
     "../../../system_wrappers:metrics",
     "../../../test:create_test_field_trials",

From 5adc7ad292d8b633da8c2870754f6277b5e035e4 Mon Sep 17 00:00:00 2001
From: Sam Zackrisson <saza@webrtc.org>
Date: Fri, 20 Mar 2026 14:18:57 +0000
Subject: [PATCH 374/472] Fix unsafe buffer usage error in apm_data_dumper.h

...by making the ArrayView-based Dump* methods write data, instead of the "unsafe" C-style dumpers.

Fixes compile failures for audioproc_f with apm_debug_dump = true.

Bug: None
Tested: audioproc_f data dumps bitexact with/without patch
Change-Id: I28e559f937eb0c36806e0d027728dcf9f6285a51
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/458640
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47227}
---
 .../logging/apm_data_dumper.h                 | 98 +++++++------------
 1 file changed, 33 insertions(+), 65 deletions(-)

diff --git a/modules/audio_processing/logging/apm_data_dumper.h b/modules/audio_processing/logging/apm_data_dumper.h
index 24e5ad913a..e49845f95f 100644
--- a/modules/audio_processing/logging/apm_data_dumper.h
+++ b/modules/audio_processing/logging/apm_data_dumper.h
@@ -123,13 +123,7 @@ class ApmDataDumper {
                [[maybe_unused]] const double* v,
                [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
-    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
-      return;
-
-    if (recording_activated_) {
-      FILE* file = GetRawFile(name);
-      fwrite(v, sizeof(v[0]), v_length, file);
-    }
+    DumpRaw(name, ArrayView<const double>(v, v_length), dump_set);
 #endif
   }
 
@@ -141,7 +135,8 @@ class ApmDataDumper {
       return;
 
     if (recording_activated_) {
-      DumpRaw(name, v.size(), v.data());
+      FILE* file = GetRawFile(name);
+      fwrite(v.data(), sizeof(v[0]), v.size(), file);
     }
 #endif
   }
@@ -165,13 +160,7 @@ class ApmDataDumper {
                [[maybe_unused]] const float* v,
                [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
-    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
-      return;
-
-    if (recording_activated_) {
-      FILE* file = GetRawFile(name);
-      fwrite(v, sizeof(v[0]), v_length, file);
-    }
+    DumpRaw(name, ArrayView<const float>(v, v_length), dump_set);
 #endif
   }
 
@@ -183,7 +172,8 @@ class ApmDataDumper {
       return;
 
     if (recording_activated_) {
-      DumpRaw(name, v.size(), v.data());
+      FILE* file = GetRawFile(name);
+      fwrite(v.data(), sizeof(v[0]), v.size(), file);
     }
 #endif
   }
@@ -206,16 +196,7 @@ class ApmDataDumper {
                [[maybe_unused]] const bool* v,
                [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
-    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
-      return;
-
-    if (recording_activated_) {
-      FILE* file = GetRawFile(name);
-      for (size_t k = 0; k < v_length; ++k) {
-        int16_t value = static_cast<int16_t>(v[k]);
-        fwrite(&value, sizeof(value), 1, file);
-      }
-    }
+    DumpRaw(name, ArrayView<const bool>(v, v_length), dump_set);
 #endif
   }
 
@@ -227,7 +208,11 @@ class ApmDataDumper {
       return;
 
     if (recording_activated_) {
-      DumpRaw(name, v.size(), v.data());
+      FILE* file = GetRawFile(name);
+      for (bool val : v) {
+        int16_t value = static_cast<int16_t>(val);
+        fwrite(&value, sizeof(value), 1, file);
+      }
     }
 #endif
   }
@@ -251,13 +236,7 @@ class ApmDataDumper {
                [[maybe_unused]] const int16_t* v,
                [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
-    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
-      return;
-
-    if (recording_activated_) {
-      FILE* file = GetRawFile(name);
-      fwrite(v, sizeof(v[0]), v_length, file);
-    }
+    DumpRaw(name, ArrayView<const int16_t>(v, v_length), dump_set);
 #endif
   }
 
@@ -269,7 +248,8 @@ class ApmDataDumper {
       return;
 
     if (recording_activated_) {
-      DumpRaw(name, v.size(), v.data());
+      FILE* file = GetRawFile(name);
+      fwrite(v.data(), sizeof(v[0]), v.size(), file);
     }
 #endif
   }
@@ -292,13 +272,21 @@ class ApmDataDumper {
                [[maybe_unused]] size_t v_length,
                [[maybe_unused]] const int32_t* v,
                [[maybe_unused]] int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    DumpRaw(name, ArrayView<const int32_t>(v, v_length), dump_set);
+#endif
+  }
+
+  void DumpRaw([[maybe_unused]] absl::string_view name,
+               [[maybe_unused]] ArrayView<const int32_t> v,
+               [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
     if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
       return;
 
     if (recording_activated_) {
       FILE* file = GetRawFile(name);
-      fwrite(v, sizeof(v[0]), v_length, file);
+      fwrite(v.data(), sizeof(v[0]), v.size(), file);
     }
 #endif
   }
@@ -322,40 +310,24 @@ class ApmDataDumper {
                [[maybe_unused]] const size_t* v,
                [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
-    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
-      return;
-
-    if (recording_activated_) {
-      FILE* file = GetRawFile(name);
-      fwrite(v, sizeof(v[0]), v_length, file);
-    }
+    DumpRaw(name, ArrayView<const size_t>(v, v_length), dump_set);
 #endif
   }
 
   void DumpRaw([[maybe_unused]] absl::string_view name,
-               [[maybe_unused]] ArrayView<const int32_t> v,
+               [[maybe_unused]] ArrayView<const size_t> v,
                [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
     if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
       return;
 
     if (recording_activated_) {
-      DumpRaw(name, v.size(), v.data());
+      FILE* file = GetRawFile(name);
+      fwrite(v.data(), sizeof(v[0]), v.size(), file);
     }
 #endif
   }
 
-  void DumpRaw(absl::string_view name,
-               ArrayView<const size_t> v,
-               int dump_set = kDefaultDumpSet) {
-#if WEBRTC_APM_DEBUG_DUMP == 1
-    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
-      return;
-
-    DumpRaw(name, v.size(), v.data());
-#endif
-  }
-
   void DumpWav([[maybe_unused]] absl::string_view name,
                [[maybe_unused]] size_t v_length,
                [[maybe_unused]] const float* v,
@@ -363,14 +335,8 @@ class ApmDataDumper {
                [[maybe_unused]] int num_channels,
                [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
-    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
-      return;
-
-    if (recording_activated_) {
-      WavWriter* file = GetWavFile(name, sample_rate_hz, num_channels,
-                                   WavFile::SampleFormat::kFloat);
-      file->WriteSamples(v, v_length);
-    }
+    DumpWav(name, ArrayView<const float>(v, v_length), sample_rate_hz,
+            num_channels, dump_set);
 #endif
   }
 
@@ -384,7 +350,9 @@ class ApmDataDumper {
       return;
 
     if (recording_activated_) {
-      DumpWav(name, v.size(), v.data(), sample_rate_hz, num_channels);
+      WavWriter* file = GetWavFile(name, sample_rate_hz, num_channels,
+                                   WavFile::SampleFormat::kFloat);
+      file->WriteSamples(v.data(), v.size());
     }
 #endif
   }

From bc9cf92f3d874017f1db79f64d5434b7444c2a6b Mon Sep 17 00:00:00 2001
From: Per Kjellander <perkj@webrtc.org>
Date: Mon, 23 Mar 2026 02:07:58 -0700
Subject: [PATCH 375/472] Revert "Allow unspecified max allocatable bitrate in
 VideoSendStreamImpl"
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This reverts commit e844bd819c4663de5422be9c2cdb25204b03b41b.

Reason for revert: Maybe breaks downstream projects

Original change's description:
> Allow unspecified max allocatable bitrate in VideoSendStreamImpl
>
> If we have no video the can be enabled, based on the VideoEncoderConfig
> (e.g. the max bitrate is <= 0 or all simulcast streams are
> `active=false`) - let the max bitrate be reported as 0 for the sake of
> bandwidth allocation (MediaStreamAllocationConfig).
>
> Previous to this CL, the above conditions would result in an arbitrary
> value of 10Mbps to use as the max allocation limit. When creating a peer
> connection with audio and video configured but the video disabled/muted,
> that would result in the bandwidth estimator trying to send packet
> trains to probe up to 10Mbps, potentially impacting the network and
> disturbing audio even though we have no need for such as high BWE.
>
> Bug: webrtc:494350649
> Change-Id: Id6353fd93c6170610a790c084d75f3b3c5c3ee97
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449100
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#47220}

Bug: webrtc:494350649
Change-Id: Id3accec77f8bde51c43c3eeb31db68d4f03f7754
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/458722
Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47228}
---
 call/call_perf_tests.cc                       |   8 +-
 media/engine/webrtc_video_engine_unittest.cc  |  50 ++++---
 .../bwe_integration_tests/bwe_ramp_up_test.cc |  28 +---
 test/scenario/video_stream_unittest.cc        |   2 +-
 video/video_send_stream_impl.cc               | 134 ++++++++----------
 video/video_send_stream_impl.h                |   6 +-
 video/video_send_stream_impl_unittest.cc      |  16 +--
 7 files changed, 100 insertions(+), 144 deletions(-)

diff --git a/call/call_perf_tests.cc b/call/call_perf_tests.cc
index 14b9de618f..9cd29d8d1c 100644
--- a/call/call_perf_tests.cc
+++ b/call/call_perf_tests.cc
@@ -654,6 +654,7 @@ TEST_F(CallPerfTest, Bitrate_Kbps_NoPadWithoutMinTransmitBitrate) {
 #endif
 TEST_F(CallPerfTest, MAYBE_KeepsHighBitrateWhenReconfiguringSender) {
   static const uint32_t kInitialBitrateKbps = 400;
+  static const uint32_t kInitialBitrateOverheadKpbs = 6;
   static const uint32_t kReconfigureThresholdKbps = 600;
 
   class VideoStreamFactory
@@ -696,9 +697,10 @@ TEST_F(CallPerfTest, MAYBE_KeepsHighBitrateWhenReconfiguringSender) {
         // First time initialization. Frame size is known.
         // `expected_bitrate` is affected by bandwidth estimation before the
         // first frame arrives to the encoder.
-        uint32_t expected_bitrate = last_set_bitrate_kbps_ > 0
-                                        ? last_set_bitrate_kbps_
-                                        : kInitialBitrateKbps;
+        uint32_t expected_bitrate =
+            last_set_bitrate_kbps_ > 0
+                ? last_set_bitrate_kbps_
+                : kInitialBitrateKbps - kInitialBitrateOverheadKpbs;
         EXPECT_EQ(expected_bitrate, config->startBitrate)
             << "Encoder not initialized at expected bitrate.";
         EXPECT_EQ(test::VideoTestConstants::kDefaultWidth, config->width);
diff --git a/media/engine/webrtc_video_engine_unittest.cc b/media/engine/webrtc_video_engine_unittest.cc
index 368554440c..fdd105de71 100644
--- a/media/engine/webrtc_video_engine_unittest.cc
+++ b/media/engine/webrtc_video_engine_unittest.cc
@@ -10020,11 +10020,6 @@ TEST_F(WebRtcVideoChannelBaseTest, EncoderSelectorSwitchCodec) {
   EXPECT_TRUE(send_channel_->SetSenderParameters(parameters));
   send_channel_->SetSend(true);
 
-  RtpParameters rtp_parameters = send_channel_->GetRtpSendParameters(kSsrc);
-  ASSERT_EQ(1u, rtp_parameters.encodings.size());
-  rtp_parameters.encodings[0].max_bitrate_bps = 1000000;
-  EXPECT_TRUE(send_channel_->SetRtpSendParameters(kSsrc, rtp_parameters).ok());
-
   std::optional<Codec> codec = send_channel_->GetSendCodec();
   ASSERT_TRUE(codec);
   EXPECT_EQ("VP8", codec->name);
@@ -10046,20 +10041,23 @@ TEST_F(WebRtcVideoChannelBaseTest, EncoderSelectorSwitchCodec) {
 
 TEST_F(WebRtcVideoChannelTest, ScaleResolutionDownToSinglecast) {
   VideoSenderParameters parameters;
+  parameters.codecs.push_back(GetEngineCodec("VP8"));
+  ASSERT_TRUE(send_channel_->SetSenderParameters(parameters));
+
   FakeVideoSendStream* stream = AddSendStream();
-  const uint32_t ssrc = stream->GetConfig().rtp.ssrcs[0];
   FrameForwarder frame_forwarder;
   FakeFrameSource frame_source(1280, 720, TimeDelta::Seconds(1) / 30,
                                env_.clock().CurrentTime());
-  EXPECT_TRUE(send_channel_->SetVideoSend(ssrc, nullptr, &frame_forwarder));
-  frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
+  EXPECT_TRUE(
+      send_channel_->SetVideoSend(last_ssrc_, nullptr, &frame_forwarder));
 
   {  // TEST scale_resolution_down_to < frame size
-    RtpParameters rtp_parameters = send_channel_->GetRtpSendParameters(ssrc);
+    RtpParameters rtp_parameters =
+        send_channel_->GetRtpSendParameters(last_ssrc_);
     EXPECT_EQ(1UL, rtp_parameters.encodings.size());
     rtp_parameters.encodings[0].scale_resolution_down_to = {.width = 640,
                                                             .height = 360};
-    send_channel_->SetRtpSendParameters(ssrc, rtp_parameters);
+    send_channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
 
     frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
 
@@ -10070,11 +10068,11 @@ TEST_F(WebRtcVideoChannelTest, ScaleResolutionDownToSinglecast) {
   }
 
   {  // TEST scale_resolution_down_to == frame size
-    auto rtp_parameters = send_channel_->GetRtpSendParameters(ssrc);
+    auto rtp_parameters = send_channel_->GetRtpSendParameters(last_ssrc_);
     EXPECT_EQ(1UL, rtp_parameters.encodings.size());
     rtp_parameters.encodings[0].scale_resolution_down_to = {.width = 1280,
                                                             .height = 720};
-    send_channel_->SetRtpSendParameters(ssrc, rtp_parameters);
+    send_channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
 
     frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
     auto streams = stream->GetVideoStreams();
@@ -10084,11 +10082,11 @@ TEST_F(WebRtcVideoChannelTest, ScaleResolutionDownToSinglecast) {
   }
 
   {  // TEST scale_resolution_down_to > frame size
-    auto rtp_parameters = send_channel_->GetRtpSendParameters(ssrc);
+    auto rtp_parameters = send_channel_->GetRtpSendParameters(last_ssrc_);
     EXPECT_EQ(1UL, rtp_parameters.encodings.size());
     rtp_parameters.encodings[0].scale_resolution_down_to = {.width = 2 * 1280,
                                                             .height = 2 * 720};
-    send_channel_->SetRtpSendParameters(ssrc, rtp_parameters);
+    send_channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
 
     frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
     auto streams = stream->GetVideoStreams();
@@ -10097,7 +10095,7 @@ TEST_F(WebRtcVideoChannelTest, ScaleResolutionDownToSinglecast) {
     EXPECT_EQ(checked_cast<size_t>(720), streams[0].height);
   }
 
-  EXPECT_TRUE(send_channel_->SetVideoSend(ssrc, nullptr, nullptr));
+  EXPECT_TRUE(send_channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
 }
 
 TEST_F(WebRtcVideoChannelTest, ScaleResolutionDownToSinglecastScaling) {
@@ -10106,18 +10104,18 @@ TEST_F(WebRtcVideoChannelTest, ScaleResolutionDownToSinglecastScaling) {
   ASSERT_TRUE(send_channel_->SetSenderParameters(parameters));
 
   FakeVideoSendStream* stream = AddSendStream();
-  const uint32_t ssrc = stream->GetConfig().rtp.ssrcs[0];
   FrameForwarder frame_forwarder;
   FakeFrameSource frame_source(1280, 720, TimeDelta::Seconds(1) / 30,
                                env_.clock().CurrentTime());
-  EXPECT_TRUE(send_channel_->SetVideoSend(ssrc, nullptr, &frame_forwarder));
+  EXPECT_TRUE(
+      send_channel_->SetVideoSend(last_ssrc_, nullptr, &frame_forwarder));
 
   {
-    auto rtp_parameters = send_channel_->GetRtpSendParameters(ssrc);
+    auto rtp_parameters = send_channel_->GetRtpSendParameters(last_ssrc_);
     EXPECT_EQ(1UL, rtp_parameters.encodings.size());
     rtp_parameters.encodings[0].scale_resolution_down_to = {.width = 720,
                                                             .height = 720};
-    send_channel_->SetRtpSendParameters(ssrc, rtp_parameters);
+    send_channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
 
     frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
 
@@ -10130,11 +10128,11 @@ TEST_F(WebRtcVideoChannelTest, ScaleResolutionDownToSinglecastScaling) {
   }
 
   {
-    auto rtp_parameters = send_channel_->GetRtpSendParameters(ssrc);
+    auto rtp_parameters = send_channel_->GetRtpSendParameters(last_ssrc_);
     EXPECT_EQ(1UL, rtp_parameters.encodings.size());
     rtp_parameters.encodings[0].scale_resolution_down_to = {.width = 1280,
                                                             .height = 1280};
-    send_channel_->SetRtpSendParameters(ssrc, rtp_parameters);
+    send_channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
 
     frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
 
@@ -10146,11 +10144,11 @@ TEST_F(WebRtcVideoChannelTest, ScaleResolutionDownToSinglecastScaling) {
   }
 
   {
-    auto rtp_parameters = send_channel_->GetRtpSendParameters(ssrc);
+    auto rtp_parameters = send_channel_->GetRtpSendParameters(last_ssrc_);
     EXPECT_EQ(1UL, rtp_parameters.encodings.size());
     rtp_parameters.encodings[0].scale_resolution_down_to = {.width = 650,
                                                             .height = 650};
-    send_channel_->SetRtpSendParameters(ssrc, rtp_parameters);
+    send_channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
 
     auto streams = stream->GetVideoStreams();
     ASSERT_EQ(streams.size(), 1u);
@@ -10161,11 +10159,11 @@ TEST_F(WebRtcVideoChannelTest, ScaleResolutionDownToSinglecastScaling) {
   }
 
   {
-    auto rtp_parameters = send_channel_->GetRtpSendParameters(ssrc);
+    auto rtp_parameters = send_channel_->GetRtpSendParameters(last_ssrc_);
     EXPECT_EQ(1UL, rtp_parameters.encodings.size());
     rtp_parameters.encodings[0].scale_resolution_down_to = {.width = 2560,
                                                             .height = 1440};
-    send_channel_->SetRtpSendParameters(ssrc, rtp_parameters);
+    send_channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
 
     auto streams = stream->GetVideoStreams();
     ASSERT_EQ(streams.size(), 1u);
@@ -10174,7 +10172,7 @@ TEST_F(WebRtcVideoChannelTest, ScaleResolutionDownToSinglecastScaling) {
     EXPECT_EQ(checked_cast<size_t>(720), streams[0].height);
   }
 
-  EXPECT_TRUE(send_channel_->SetVideoSend(ssrc, nullptr, nullptr));
+  EXPECT_TRUE(send_channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
 }
 
 TEST_F(WebRtcVideoChannelTest, ScaleResolutionDownToSimulcast) {
diff --git a/test/peer_scenario/bwe_integration_tests/bwe_ramp_up_test.cc b/test/peer_scenario/bwe_integration_tests/bwe_ramp_up_test.cc
index 4c5637a36b..8a0bbd37a3 100644
--- a/test/peer_scenario/bwe_integration_tests/bwe_ramp_up_test.cc
+++ b/test/peer_scenario/bwe_integration_tests/bwe_ramp_up_test.cc
@@ -16,7 +16,6 @@
 
 #include "api/jsep.h"
 #include "api/media_types.h"
-#include "api/rtp_parameters.h"
 #include "api/rtp_sender_interface.h"
 #include "api/rtp_transceiver_direction.h"
 #include "api/scoped_refptr.h"
@@ -35,7 +34,6 @@
 #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "modules/rtp_rtcp/source/rtp_util.h"
 #include "pc/media_session.h"
-#include "rtc_base/logging.h"
 #include "test/create_frame_generator_capturer.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
@@ -208,7 +206,7 @@ INSTANTIATE_TEST_SUITE_P(
              .l4s_network = true,
              .network_capacity = DataRate::KilobitsPerSec(3000),
              .expected_bwe_min = DataRate::KilobitsPerSec(1500),
-             .max_bwe = DataRate::KilobitsPerSec(3600),
+             .max_bwe = DataRate::KilobitsPerSec(3000),
          },
          {
              .test_name = "L4s500Kbit",
@@ -241,18 +239,12 @@ TEST_P(BweRampupWithInitialProbeTest, BweRampUpBothDirectionsWithoutMedia) {
   PeerScenarioClient* caller = s.CreateClient(config);
   PeerScenarioClient* callee = s.CreateClient(config);
 
-  auto transceiver_or = caller->pc()->AddTransceiver(MediaType::VIDEO);
-  ASSERT_TRUE(transceiver_or.error().ok());
-  auto transceiver = transceiver_or.value();
-
-  auto parameters = transceiver->sender()->GetParameters();
-  ASSERT_THAT(parameters.encodings, testing::SizeIs(1));
-  parameters.encodings[0].max_bitrate_bps = 3'000'000;
-  ASSERT_TRUE(transceiver->sender()->SetParameters(parameters).ok());
+  auto transceiver = caller->pc()->AddTransceiver(MediaType::VIDEO);
+  ASSERT_TRUE(transceiver.error().ok());
 
   MockRtpSenderObserver observer;
   EXPECT_CALL(observer, OnFirstPacketSent).Times(0);
-  transceiver->sender()->SetObserver(&observer);
+  transceiver.value()->sender()->SetObserver(&observer);
 
   caller->pc()->ReconfigureBandwidthEstimation(
       {.allow_probe_without_media = true});
@@ -298,18 +290,6 @@ TEST_P(BweRampupWithInitialProbeTest, BweRampUpBothDirectionsWithoutMedia) {
   // Wait for SDP negotiation.
   s.WaitAndProcess(&offer_exchange_done);
 
-  // Set max encoding rate for Callee as well.
-  ASSERT_THAT(callee->pc()->GetTransceivers(), SizeIs(1));
-  auto callee_transceiver = callee->pc()->GetTransceivers()[0];
-  auto callee_params = callee_transceiver->sender()->GetParameters();
-  if (callee_params.encodings.size() > 0) {
-    callee_params.encodings[0].max_bitrate_bps = 10'000'000;
-    ASSERT_TRUE(
-        callee_transceiver->sender()->SetParameters(callee_params).ok());
-  } else {
-    RTC_LOG(LS_WARNING) << "Callee encodings is empty after negotiation!";
-  }
-
   // Test that 1s after offer/answer exchange finish, we have a BWE estimate,
   // even though no video frames have been sent.
   s.ProcessMessages(TimeDelta::Seconds(2));
diff --git a/test/scenario/video_stream_unittest.cc b/test/scenario/video_stream_unittest.cc
index c1fcd4da1c..c9791bf8fc 100644
--- a/test/scenario/video_stream_unittest.cc
+++ b/test/scenario/video_stream_unittest.cc
@@ -281,7 +281,7 @@ TEST(VideoStreamTest, SuspendsBelowMinBitrate) {
     // Min transmit rate needs to be lower than kMinVideoBitrate for this test
     // to make sense.
     c->transport.rates.min_rate = kMinVideoBitrate / 2;
-    c->transport.rates.start_rate = kMinVideoBitrate * 2;
+    c->transport.rates.start_rate = kMinVideoBitrate;
     c->transport.rates.max_rate = kMinVideoBitrate * 2;
   });
   auto send_net = s.CreateMutableSimulationNode(
diff --git a/video/video_send_stream_impl.cc b/video/video_send_stream_impl.cc
index 301939f4eb..ad9130bb1d 100644
--- a/video/video_send_stream_impl.cc
+++ b/video/video_send_stream_impl.cc
@@ -236,50 +236,32 @@ int GetEncoderPriorityBitrate(std::string codec_name,
   return priority_bitrate;
 }
 
-DataRate GetDefaultMinVideoBitrate(VideoCodecType codec_type) {
-  if (codec_type == VideoCodecType::kVideoCodecAV1) {
-    return DataRate::BitsPerSec(kMinDefaultAv1BitrateBps);
+uint32_t GetInitialEncoderMaxBitrate(int initial_encoder_max_bitrate) {
+  if (initial_encoder_max_bitrate > 0)
+    return dchecked_cast<uint32_t>(initial_encoder_max_bitrate);
+
+  // TODO(srte): Make sure max bitrate is not set to negative values. We don't
+  // have any way to handle unset values in downstream code, such as the
+  // bitrate allocator. Previously -1 was implicitly casted to UINT32_MAX, a
+  // behaviour that is not safe. Converting to 10 Mbps should be safe for
+  // reasonable use cases as it allows adding the max of multiple streams
+  // without wrappping around.
+  const int kFallbackMaxBitrateBps = 10000000;
+  // Don't log an error for -1 since this is the default value that is used to
+  // signal that the max bitrate is unset.
+  if (initial_encoder_max_bitrate != -1) {
+    RTC_DLOG(LS_ERROR) << "ERROR: Initial encoder max bitrate = "
+                       << initial_encoder_max_bitrate << " which is <= 0!";
   }
-  return DataRate::BitsPerSec(kDefaultMinVideoBitrateBps);
+  RTC_DLOG(LS_INFO) << "Using default encoder max bitrate = 10 Mbps";
+  return kFallbackMaxBitrateBps;
 }
 
-std::optional<DataRate> GetConfiguredMaxVideoBitrate(
-    const VideoEncoderConfig& config) {
-  // If spatial layers are configured, the sum of their bitrate takes precedence
-  // over both simulcast and the global max_bitrate_bps.
-  if (!config.spatial_layers.empty()) {
-    std::optional<DataRate> spatial_layer_max_bitrate_sum;
-    for (const auto& layer : config.spatial_layers) {
-      if (layer.active && layer.maxBitrate > 0) {
-        spatial_layer_max_bitrate_sum =
-            spatial_layer_max_bitrate_sum.value_or(DataRate::Zero()) +
-            DataRate::KilobitsPerSec(layer.maxBitrate);
-      }
-    }
-    return spatial_layer_max_bitrate_sum;
+int GetDefaultMinVideoBitrateBps(VideoCodecType codec_type) {
+  if (codec_type == VideoCodecType::kVideoCodecAV1) {
+    return kMinDefaultAv1BitrateBps;
   }
-
-  // If simulcast layers are configured, the sum of their bitrate takes
-  // precedence over the global max_bitrate_bps.
-  if (!config.simulcast_layers.empty()) {
-    std::optional<DataRate> simulcast_layer_max_bitrate_sum;
-    for (const auto& stream : config.simulcast_layers) {
-      if (stream.active && stream.max_bitrate_bps > 0) {
-        simulcast_layer_max_bitrate_sum =
-            simulcast_layer_max_bitrate_sum.value_or(DataRate::Zero()) +
-            DataRate::BitsPerSec(stream.max_bitrate_bps);
-      }
-    }
-    return simulcast_layer_max_bitrate_sum;
-  }
-
-  // If single stream, use the global max_bitrate_bps.
-  if (config.max_bitrate_bps > 0) {
-    return DataRate::BitsPerSec(config.max_bitrate_bps);
-  }
-
-  // No max bitrate specified.
-  return std::nullopt;
+  return kDefaultMinVideoBitrateBps;
 }
 
 size_t CalculateMaxHeaderSize(const RtpConfig& config) {
@@ -473,9 +455,10 @@ VideoSendStreamImpl::VideoSendStreamImpl(
       has_active_encodings_(HasActiveEncodings(encoder_config)),
       disable_padding_(true),
       max_padding_bitrate_(0),
-      encoder_min_bitrate_(DataRate::Zero()),
-      encoder_max_bitrate_(GetConfiguredMaxVideoBitrate(encoder_config)),
-      encoder_target_rate_(DataRate::Zero()),
+      encoder_min_bitrate_bps_(0),
+      encoder_max_bitrate_bps_(
+          GetInitialEncoderMaxBitrate(encoder_config.max_bitrate_bps)),
+      encoder_target_rate_bps_(0),
       encoder_bitrate_priority_(encoder_config.bitrate_priority),
       encoder_av1_priority_bitrate_override_bps_(
           GetEncoderPriorityBitrate(config_.rtp.payload_name,
@@ -489,6 +472,7 @@ VideoSendStreamImpl::VideoSendStreamImpl(
   RTC_DCHECK_LE(config_.rtp.payload_type, 127);
   RTC_DCHECK(!config_.rtp.ssrcs.empty());
   RTC_DCHECK(transport_);
+  RTC_DCHECK_NE(encoder_max_bitrate_bps_, 0);
   RTC_LOG(LS_INFO) << "VideoSendStreamImpl: " << config_.ToString();
 
   RTC_CHECK(
@@ -735,7 +719,7 @@ void VideoSendStreamImpl::SignalEncoderTimedOut() {
   // If the encoder has not produced anything the last kEncoderTimeOut and it
   // is supposed to, deregister as BitrateAllocatorObserver. This can happen
   // if a camera stops producing frames.
-  if (!encoder_target_rate_.IsZero()) {
+  if (encoder_target_rate_bps_ > 0) {
     RTC_LOG(LS_INFO) << "SignalEncoderTimedOut, Encoder timed out.";
     bitrate_allocator_->RemoveObserver(this);
   }
@@ -747,7 +731,7 @@ void VideoSendStreamImpl::OnBitrateAllocationUpdated(
   // the worker_queue_.
   auto task = [this, allocation] {
     RTC_DCHECK_RUN_ON(&thread_checker_);
-    if (encoder_target_rate_.IsZero()) {
+    if (encoder_target_rate_bps_ == 0) {
       return;
     }
     int64_t now_ms = env_.clock().TimeInMilliseconds();
@@ -807,9 +791,8 @@ void VideoSendStreamImpl::SignalEncoderActive() {
 
 MediaStreamAllocationConfig VideoSendStreamImpl::GetAllocationConfig() const {
   return MediaStreamAllocationConfig{
-      .min_bitrate_bps = encoder_min_bitrate_.bps<uint32_t>(),
-      .max_bitrate_bps =
-          encoder_max_bitrate_.value_or(DataRate::Zero()).bps<uint32_t>(),
+      .min_bitrate_bps = static_cast<uint32_t>(encoder_min_bitrate_bps_),
+      .max_bitrate_bps = encoder_max_bitrate_bps_,
       .pad_up_bitrate_bps =
           static_cast<uint32_t>(disable_padding_ ? 0 : max_padding_bitrate_),
       .priority_bitrate_bps = encoder_av1_priority_bitrate_override_bps_,
@@ -839,23 +822,17 @@ void VideoSendStreamImpl::OnEncoderConfigurationChanged(
 
     const std::optional<DataRate> experimental_min_bitrate =
         GetExperimentalMinVideoBitrate(env_.field_trials(), codec_type);
-    if (experimental_min_bitrate.has_value()) {
-      encoder_min_bitrate_ = *experimental_min_bitrate;
-    } else {
-      DataRate min_bitrate =
-          streams[0].min_bitrate_bps > 0
-              ? DataRate::BitsPerSec(streams[0].min_bitrate_bps)
-              : DataRate::Zero();
-      encoder_min_bitrate_ =
-          std::max(min_bitrate, GetDefaultMinVideoBitrate(codec_type));
-    }
+    encoder_min_bitrate_bps_ =
+        experimental_min_bitrate
+            ? experimental_min_bitrate->bps()
+            : std::max(streams[0].min_bitrate_bps,
+                       GetDefaultMinVideoBitrateBps(codec_type));
     double stream_bitrate_priority_sum = 0;
-    std::optional<DataRate> max_bitrate_sum;
+    uint32_t encoder_max_bitrate_bps = 0;
     for (const auto& stream : streams) {
       // We don't want to allocate more bitrate than needed to inactive streams.
-      if (stream.active && stream.max_bitrate_bps > 0) {
-        max_bitrate_sum = max_bitrate_sum.value_or(DataRate::Zero()) +
-                          DataRate::BitsPerSec(stream.max_bitrate_bps);
+      if (stream.active) {
+        encoder_max_bitrate_bps += stream.max_bitrate_bps;
       }
       if (stream.bitrate_priority) {
         RTC_DCHECK_GT(*stream.bitrate_priority, 0);
@@ -864,8 +841,10 @@ void VideoSendStreamImpl::OnEncoderConfigurationChanged(
     }
     RTC_DCHECK_GT(stream_bitrate_priority_sum, 0);
     encoder_bitrate_priority_ = stream_bitrate_priority_sum;
-    if (max_bitrate_sum.has_value()) {
-      encoder_max_bitrate_ = std::max(encoder_min_bitrate_, *max_bitrate_sum);
+    if (encoder_max_bitrate_bps > 0) {
+      encoder_max_bitrate_bps_ =
+          std::max(static_cast<uint32_t>(encoder_min_bitrate_bps_),
+                   encoder_max_bitrate_bps);
     }
 
     // TODO(bugs.webrtc.org/10266): Query the VideoBitrateAllocator instead.
@@ -952,25 +931,24 @@ uint32_t VideoSendStreamImpl::OnBitrateUpdated(BitrateAllocationUpdate update) {
       << "VideoSendStream::Start has not been called.";
 
   rtp_video_sender_->OnBitrateUpdated(update, stats_proxy_.GetSendFrameRate());
-  encoder_target_rate_ =
-      DataRate::BitsPerSec(rtp_video_sender_->GetPayloadBitrateBps());
-  const DataRate protection_bitrate =
-      DataRate::BitsPerSec(rtp_video_sender_->GetProtectionBitrateBps());
+  encoder_target_rate_bps_ = rtp_video_sender_->GetPayloadBitrateBps();
+  const uint32_t protection_bitrate_bps =
+      rtp_video_sender_->GetProtectionBitrateBps();
   DataRate link_allocation = DataRate::Zero();
-  if (encoder_target_rate_ > protection_bitrate) {
-    link_allocation = encoder_target_rate_ - protection_bitrate;
+  if (encoder_target_rate_bps_ > protection_bitrate_bps) {
+    link_allocation =
+        DataRate::BitsPerSec(encoder_target_rate_bps_ - protection_bitrate_bps);
   }
-  if (encoder_max_bitrate_.has_value()) {
-    encoder_target_rate_ =
-        std::min(*encoder_max_bitrate_, encoder_target_rate_);
-  }
-  link_allocation = std::max(encoder_target_rate_, link_allocation);
+  encoder_target_rate_bps_ =
+      std::min(encoder_max_bitrate_bps_, encoder_target_rate_bps_);
+  DataRate encoder_target_rate = DataRate::BitsPerSec(encoder_target_rate_bps_);
+  link_allocation = std::max(encoder_target_rate, link_allocation);
   video_stream_encoder_->OnBitrateUpdated(
-      encoder_target_rate_, link_allocation,
+      encoder_target_rate, link_allocation,
       dchecked_cast<uint8_t>(update.packet_loss_ratio * 256),
       update.round_trip_time.ms(), update.cwnd_reduce_ratio);
-  stats_proxy_.OnSetEncoderTargetRate(encoder_target_rate_.bps());
-  return protection_bitrate.bps();
+  stats_proxy_.OnSetEncoderTargetRate(encoder_target_rate_bps_);
+  return protection_bitrate_bps;
 }
 
 std::optional<DataRate> VideoSendStreamImpl::GetUsedRate() const {
diff --git a/video/video_send_stream_impl.h b/video/video_send_stream_impl.h
index 3f2f8e02bd..88845a1969 100644
--- a/video/video_send_stream_impl.h
+++ b/video/video_send_stream_impl.h
@@ -239,9 +239,9 @@ class VideoSendStreamImpl : public webrtc::VideoSendStream,
   bool has_active_encodings_ RTC_GUARDED_BY(thread_checker_);
   bool disable_padding_ RTC_GUARDED_BY(thread_checker_);
   int max_padding_bitrate_ RTC_GUARDED_BY(thread_checker_);
-  DataRate encoder_min_bitrate_ RTC_GUARDED_BY(thread_checker_);
-  std::optional<DataRate> encoder_max_bitrate_ RTC_GUARDED_BY(thread_checker_);
-  DataRate encoder_target_rate_ RTC_GUARDED_BY(thread_checker_);
+  int encoder_min_bitrate_bps_ RTC_GUARDED_BY(thread_checker_);
+  uint32_t encoder_max_bitrate_bps_ RTC_GUARDED_BY(thread_checker_);
+  uint32_t encoder_target_rate_bps_ RTC_GUARDED_BY(thread_checker_);
   double encoder_bitrate_priority_ RTC_GUARDED_BY(thread_checker_);
   const int encoder_av1_priority_bitrate_override_bps_
       RTC_GUARDED_BY(thread_checker_);
diff --git a/video/video_send_stream_impl_unittest.cc b/video/video_send_stream_impl_unittest.cc
index 8a8e35e4bc..4bead83d26 100644
--- a/video/video_send_stream_impl_unittest.cc
+++ b/video/video_send_stream_impl_unittest.cc
@@ -183,8 +183,6 @@ class VideoSendStreamImplTest : public ::testing::Test {
     encoder_config.simulcast_layers.push_back(VideoStream());
     encoder_config.simulcast_layers.back().active = true;
     encoder_config.simulcast_layers.back().bitrate_priority = 1.0;
-    encoder_config.simulcast_layers.back().width = 640;
-    encoder_config.simulcast_layers.back().height = 360;
     return encoder_config;
   }
 
@@ -293,7 +291,7 @@ TEST_F(VideoSendStreamImplTest,
        MaxBitrateCorrectIfActiveEncodingUpdatedAfterCreation) {
   VideoEncoderConfig one_active_encoding = TestVideoEncoderConfig();
   ASSERT_THAT(one_active_encoding.simulcast_layers, SizeIs(1));
-  one_active_encoding.max_bitrate_bps = 2'500'000;
+  one_active_encoding.max_bitrate_bps = 10'000'000;
   one_active_encoding.simulcast_layers[0].max_bitrate_bps = 2'000'000;
   VideoEncoderConfig no_active_encodings = one_active_encoding.Copy();
   no_active_encodings.simulcast_layers[0].active = false;
@@ -309,12 +307,12 @@ TEST_F(VideoSendStreamImplTest,
   time_controller_.AdvanceTime(TimeDelta::Zero());
 
   Sequence s;
-  // Expect zero max bitrate before the encoder has notified about the actual
-  // send streams.
-  EXPECT_CALL(
-      bitrate_allocator_,
-      AddObserver(vss_impl.get(),
-                  Field(&MediaStreamAllocationConfig::max_bitrate_bps, Eq(0))))
+  // Expect codec max bitrate as max needed bitrate before the encoder has
+  // notifed about the actual send streams.
+  EXPECT_CALL(bitrate_allocator_,
+              AddObserver(vss_impl.get(),
+                          Field(&MediaStreamAllocationConfig::max_bitrate_bps,
+                                Eq(one_active_encoding.max_bitrate_bps))))
       .InSequence(s);
 
   // Expect the sum of active encodings as max needed bitrate after

From 0f29d06a3389d66864cf0d33df32d113383ebafd Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?=C3=85sa=20Persson?= <asapersson@webrtc.org>
Date: Mon, 23 Mar 2026 09:37:36 +0000
Subject: [PATCH 376/472] Move VCMTiming::kDefaultRenderDelay to anonymous
 namespace in .cc file.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

It is used in the implementation only.

Bug: b/493549134
Change-Id: Ib309296b0b5ecc32f84da4cf8cfc3e642d915c94
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/458721
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47229}
---
 modules/video_coding/timing/timing.cc          |  1 +
 modules/video_coding/timing/timing.h           | 12 +++++-------
 modules/video_coding/timing/timing_unittest.cc |  5 +++--
 3 files changed, 9 insertions(+), 9 deletions(-)

diff --git a/modules/video_coding/timing/timing.cc b/modules/video_coding/timing/timing.cc
index a947b38554..63a7befaf0 100644
--- a/modules/video_coding/timing/timing.cc
+++ b/modules/video_coding/timing/timing.cc
@@ -31,6 +31,7 @@
 namespace webrtc {
 namespace {
 
+constexpr TimeDelta kDefaultRenderDelay = TimeDelta::Millis(10);
 // Default pacing that is used for the low-latency renderer path.
 constexpr TimeDelta kZeroPlayoutDelayDefaultMinPacing = TimeDelta::Millis(8);
 constexpr TimeDelta kLowLatencyStreamMaxPlayoutDelayThreshold =
diff --git a/modules/video_coding/timing/timing.h b/modules/video_coding/timing/timing.h
index 2ccf1544ba..d0b863864e 100644
--- a/modules/video_coding/timing/timing.h
+++ b/modules/video_coding/timing/timing.h
@@ -56,26 +56,24 @@ class VCMTiming {
     TimeDelta current_delay;
   };
 
-  static constexpr TimeDelta kDefaultRenderDelay = TimeDelta::Millis(10);
-
   VCMTiming(Clock* clock, const FieldTrialsView& field_trials);
   virtual ~VCMTiming() = default;
 
   // Resets the timing to the initial state.
   void Reset();
 
-  // Set the amount of time needed to render an image. Defaults to 10 ms.
+  // Sets the amount of time needed to render an image. Defaults to 10 ms.
   void set_render_delay(TimeDelta render_delay);
 
-  // Set the minimum time the video must be delayed on the receiver to
+  // Sets the minimum time the video must be delayed on the receiver to
   // get the desired jitter buffer level.
   void SetJitterDelay(TimeDelta required_delay);
 
-  // Set/get the minimum playout delay from capture to render.
+  // Sets/gets the minimum playout delay from capture to render.
   TimeDelta min_playout_delay() const;
   void set_min_playout_delay(TimeDelta min_playout_delay);
 
-  // Set the minimum and maximum playout delay from capture to render.
+  // Sets the minimum and maximum playout delay from capture to render.
   void set_playout_delay(const VideoPlayoutDelay& playout_delay);
 
   // Increases or decreases the current delay to get closer to the target delay.
@@ -113,7 +111,7 @@ class VCMTiming {
   // render delay.
   TimeDelta TargetVideoDelay() const;
 
-  // Return current timing information.
+  // Returns current timing information.
   VideoDelayTimings GetTimings() const;
 
   void SetTimingFrameInfo(const TimingFrameInfo& info);
diff --git a/modules/video_coding/timing/timing_unittest.cc b/modules/video_coding/timing/timing_unittest.cc
index 45b26ab71f..afe11d0217 100644
--- a/modules/video_coding/timing/timing_unittest.cc
+++ b/modules/video_coding/timing/timing_unittest.cc
@@ -397,7 +397,8 @@ TEST(VCMTimingTest, Reset) {
   SimulatedClock clock(Timestamp::Millis(33));
   VCMTiming timing(&clock, field_trials);
 
-  timing.set_render_delay(TimeDelta::Millis(11));
+  const TimeDelta default_render_delay = timing.GetTimings().render_delay;
+  timing.set_render_delay(default_render_delay + TimeDelta::Millis(1));
   TimeDelta min_playout_delay = TimeDelta::Millis(50);
   TimeDelta max_playout_delay = TimeDelta::Millis(500);
   timing.set_playout_delay({min_playout_delay, max_playout_delay});
@@ -419,7 +420,7 @@ TEST(VCMTimingTest, Reset) {
   EXPECT_GT(timings.num_decoded_frames, 0u);
   EXPECT_EQ(timings.minimum_delay, TimeDelta::Zero());
   EXPECT_EQ(timings.estimated_max_decode_time, TimeDelta::Zero());
-  EXPECT_EQ(timings.render_delay, VCMTiming::kDefaultRenderDelay);
+  EXPECT_EQ(timings.render_delay, default_render_delay);
   EXPECT_EQ(timings.min_playout_delay, TimeDelta::Zero());
   EXPECT_EQ(timings.max_playout_delay, max_playout_delay);
   EXPECT_EQ(timings.target_delay, TimeDelta::Zero());

From 7e9559f8b08e61b4cf065ad891f132bccd580fc5 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Fri, 20 Mar 2026 18:42:40 +0100
Subject: [PATCH 377/472] Fix out-of-bounds write in SetupCodec due to
 excessive RIDs

In GetSendEncodingsFromRemoteDescription, limit the number of processed
simulcast layers to kMaxSimulcastStreams.

Added bounds checking in SetupCodec to avoid writing past the end of the
simulcastStream array.

Bug: chromium:486536241
Change-Id: Ie5bbe2bd3362cb4b319ec5be67c6288da8ad953a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/458720
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Philipp Hancke <philipp.hancke@googlemail.com>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47230}
---
 .../video_coding/video_codec_initializer.cc   |  8 ++--
 pc/sdp_offer_answer.cc                        |  6 +++
 pc/sdp_offer_answer_unittest.cc               | 40 +++++++++++++++++++
 3 files changed, 51 insertions(+), 3 deletions(-)

diff --git a/modules/video_coding/video_codec_initializer.cc b/modules/video_coding/video_codec_initializer.cc
index 6d329a1bce..7b3579df85 100644
--- a/modules/video_coding/video_codec_initializer.cc
+++ b/modules/video_coding/video_codec_initializer.cc
@@ -80,8 +80,8 @@ VideoCodec VideoCodecInitializer::SetupCodec(
       config.legacy_conference_mode;
 
   video_codec.SetFrameDropEnabled(config.frame_drop_enabled);
-  video_codec.numberOfSimulcastStreams =
-      static_cast<unsigned char>(streams.size());
+  video_codec.numberOfSimulcastStreams = static_cast<unsigned char>(
+      std::min(streams.size(), static_cast<size_t>(kMaxSimulcastStreams)));
   video_codec.minBitrate = streams[0].min_bitrate_bps / 1000;
   bool codec_active = false;
   // Active configuration might not be fully copied to `streams` for SVC yet.
@@ -104,7 +104,9 @@ VideoCodec VideoCodecInitializer::SetupCodec(
   int max_framerate = 0;
 
   std::optional<ScalabilityMode> scalability_mode = streams[0].scalability_mode;
-  for (size_t i = 0; i < streams.size(); ++i) {
+  const size_t num_streams =
+      std::min(streams.size(), static_cast<size_t>(kMaxSimulcastStreams));
+  for (size_t i = 0; i < num_streams; ++i) {
     SimulcastStream* sim_stream = &video_codec.simulcastStream[i];
     RTC_DCHECK_GT(streams[i].width, 0);
     RTC_DCHECK_GT(streams[i].height, 0);
diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc
index 3f5c83fb8d..cb2ed5c124 100644
--- a/pc/sdp_offer_answer.cc
+++ b/pc/sdp_offer_answer.cc
@@ -706,6 +706,12 @@ std::vector<RtpEncodingParameters> GetSendEncodingsFromRemoteDescription(
   // This is a remote description, the parameters we are after should appear
   // as receive streams.
   for (const auto& alternatives : simulcast.receive_layers()) {
+    if (result.size() >= kMaxSimulcastStreams) {
+      RTC_LOG(LS_WARNING)
+          << "Excessive simulcast layers in remote description. Clamping to "
+          << kMaxSimulcastStreams;
+      break;
+    }
     RTC_DCHECK(!alternatives.empty());
     // There is currently no way to specify or choose from alternatives.
     // We will always use the first alternative, which is the most preferred.
diff --git a/pc/sdp_offer_answer_unittest.cc b/pc/sdp_offer_answer_unittest.cc
index 14d81a170e..c216c8980a 100644
--- a/pc/sdp_offer_answer_unittest.cc
+++ b/pc/sdp_offer_answer_unittest.cc
@@ -823,6 +823,46 @@ TEST_F(SdpOfferAnswerTest, SimulcastAnswerWithPayloadType) {
   EXPECT_TRUE(pc->SetLocalDescription(std::move(answer)));
 }
 
+TEST_F(SdpOfferAnswerTest, SimulcastOfferWithExcessiveRidsClamped) {
+  auto pc = CreatePeerConnection();
+
+  std::string sdp =
+      "v=0\r\n"
+      "o=- 4131505339648218884 3 IN IP4 127.0.0.1\r\n"
+      "s=-\r\n"
+      "t=0 0\r\n"
+      "a=ice-ufrag:zGWFZ+fVXDeN6UoI/136\r\n"
+      "a=ice-pwd:9AUNgUqRNI5LSIrC1qFD2iTR\r\n"
+      "a=fingerprint:sha-256 "
+      "AD:52:52:E0:B1:37:34:21:0E:15:8E:B7:56:56:7B:B4:39:0E:6D:1C:F5:84:A7:EE:"
+      "B5:27:3E:30:B1:7D:69:42\r\n"
+      "a=setup:passive\r\n"
+      "m=video 9 UDP/TLS/RTP/SAVPF 96\r\n"
+      "c=IN IP4 0.0.0.0\r\n"
+      "a=rtcp:9 IN IP4 0.0.0.0\r\n"
+      "a=mid:0\r\n"
+      "a=extmap:9 urn:ietf:params:rtp-hdrext:sdes:mid\r\n"
+      "a=extmap:10 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id\r\n"
+      "a=recvonly\r\n"
+      "a=rtcp-mux\r\n"
+      "a=rtcp-rsize\r\n"
+      "a=rtpmap:96 VP8/90000\r\n";
+
+  for (int i = 1; i <= 9; ++i) {
+    sdp += "a=rid:" + std::to_string(i) + " recv\r\n";
+  }
+  sdp += "a=simulcast:recv 1;2;3;4;5;6;7;8;9\r\n";
+
+  std::unique_ptr<SessionDescriptionInterface> offer =
+      CreateSessionDescription(SdpType::kOffer, sdp);
+  EXPECT_TRUE(pc->SetRemoteDescription(std::move(offer)));
+
+  auto transceiver = pc->pc()->GetTransceivers()[0];
+  // Verify that the number of send encodings is clamped to 3
+  // (kMaxSimulcastStreams)
+  EXPECT_THAT(transceiver->sender()->GetParameters().encodings, SizeIs(3));
+}
+
 TEST_F(SdpOfferAnswerTest, ExpectAllSsrcsSpecifiedInSsrcGroupFid) {
   auto pc = CreatePeerConnection();
   std::string sdp =

From d0c7fd32b615a07bd0aa4edf131603bdff3e02ae Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Mon, 23 Mar 2026 14:35:26 +0100
Subject: [PATCH 378/472] Refactor BaseChannel+ChannelInterface to return
 RTCError instead of bool

Update the interface and internal signaling methods in BaseChannel,
VoiceChannel, and VideoChannel to return RTCError. This replaces the
previous pattern of returning a boolean and using a separate string
reference output parameter for the error description.

Bug: none
Change-Id: I198af89fad90a77d3c3125cc4c2d31d1a0c0efcd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/458641
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47231}
---
 pc/BUILD.gn                      |   1 +
 pc/channel.cc                    | 327 +++++++++++++-----------------
 pc/channel.h                     |  65 +++---
 pc/channel_interface.h           |  11 +-
 pc/channel_unittest.cc           | 334 ++++++++++++++-----------------
 pc/rtp_transceiver.cc            |  29 ++-
 pc/rtp_transceiver.h             |  12 +-
 pc/sdp_offer_answer.cc           |  11 +-
 pc/test/mock_channel_interface.h |   9 +-
 9 files changed, 350 insertions(+), 449 deletions(-)

diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 8a576969ae..c07cbd3016 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -115,6 +115,7 @@ rtc_source_set("channel_interface") {
     ":rtp_transport_internal",
     ":session_description",
     "../api:jsep",
+    "../api:rtc_error",
     "../api:rtp_parameters",
     "../media:media_channel",
     "../media:stream_params",
diff --git a/pc/channel.cc b/pc/channel.cc
index f1705378f6..8925882e17 100644
--- a/pc/channel.cc
+++ b/pc/channel.cc
@@ -340,20 +340,18 @@ void BaseChannel::Enable(bool enable) {
   }));
 }
 
-bool BaseChannel::SetLocalContent(const MediaContentDescription* content,
-                                  SdpType type,
-                                  std::string& error_desc) {
+RTCError BaseChannel::SetLocalContent(const MediaContentDescription* content,
+                                      SdpType type) {
   RTC_DCHECK_RUN_ON(worker_thread());
   TRACE_EVENT0("webrtc", "BaseChannel::SetLocalContent");
-  return SetLocalContent_w(content, type, error_desc);
+  return SetLocalContent_w(content, type);
 }
 
-bool BaseChannel::SetRemoteContent(const MediaContentDescription* content,
-                                   SdpType type,
-                                   std::string& error_desc) {
+RTCError BaseChannel::SetRemoteContent(const MediaContentDescription* content,
+                                       SdpType type) {
   RTC_DCHECK_RUN_ON(worker_thread());
   TRACE_EVENT0("webrtc", "BaseChannel::SetRemoteContent");
-  return SetRemoteContent_w(content, type, error_desc);
+  return SetRemoteContent_w(content, type);
 }
 
 bool BaseChannel::SetPayloadTypeDemuxingEnabled(bool enabled) {
@@ -575,18 +573,18 @@ void BaseChannel::OnRtpPacket(const RtpPacketReceived& parsed_packet) {
   media_receive_channel()->OnPacketReceived(parsed_packet);
 }
 
-bool BaseChannel::MaybeUpdateDemuxerAndRtpExtensions_w(
+RTCError BaseChannel::MaybeUpdateDemuxerAndRtpExtensions_w(
     bool update_demuxer,
     std::optional<flat_set<uint8_t>> payload_types,
     const RtpHeaderExtensions& extensions,
-    std::optional<flat_set<uint32_t>> ssrcs,
-    std::string& error_desc) {
+    std::optional<flat_set<uint32_t>> ssrcs) {
   bool update_extensions = true;
   if (rtp_header_extensions_ == extensions) {
     update_extensions = false;  // No need to update header extensions.
   } else {
-    if (!CheckRtpExtensionValidity(extensions, error_desc)) {
-      return false;
+    RTCError error = CheckRtpExtensionValidity(extensions);
+    if (!error.ok()) {
+      return error;
     }
     rtp_header_extensions_ = extensions;
 
@@ -604,7 +602,7 @@ bool BaseChannel::MaybeUpdateDemuxerAndRtpExtensions_w(
 
   if (!update_demuxer && !update_extensions && !payload_types.has_value() &&
       !ssrcs.has_value()) {
-    return true;  // No update needed.
+    return RTCError::OK();  // No update needed.
   }
 
   // TODO(bugs.webrtc.org/13536): See if we can do this asynchronously.
@@ -613,13 +611,13 @@ bool BaseChannel::MaybeUpdateDemuxerAndRtpExtensions_w(
     media_receive_channel()->OnDemuxerCriteriaUpdatePending();
   }
 
-  bool success = network_thread()->BlockingCall([&] {
+  RTCError error = network_thread()->BlockingCall([&]() -> RTCError {
     RTC_DCHECK_RUN_ON(network_thread());
     if (!rtp_transport_) {
       // To repro this situation, run the
       // `ApplyDescriptionWithSameSsrcsBundledFails` test.
-      RTC_LOG(LS_ERROR) << "No transport assigned for mid=" << mid();
-      return false;
+      return LOG_ERROR(RTCError::InvalidState()
+                       << "No transport assigned for mid=" << mid());
     }
 
     if (update_extensions) {
@@ -641,22 +639,21 @@ bool BaseChannel::MaybeUpdateDemuxerAndRtpExtensions_w(
     }
 
     if (!update_demuxer)
-      return true;
+      return RTCError::OK();
 
     RtpDemuxerCriteria criteria = demuxer_criteria();
     if (!rtp_transport_->RegisterRtpDemuxerSink(criteria, this)) {
-      error_desc =
-          StringFormat("Failed to apply demuxer criteria for '%s': '%s'.",
-                       mid().c_str(), criteria.ToString().c_str());
-      return false;
+      return RTCError::InvalidParameter()
+             << "Failed to apply demuxer criteria for '" << mid() << "': '"
+             << criteria.ToString() << "'.";
     }
-    return true;
+    return RTCError::OK();
   });
 
   if (update_demuxer || payload_types.has_value() || ssrcs.has_value())
     media_receive_channel()->OnDemuxerCriteriaUpdateComplete();
 
-  return success;
+  return error;
 }
 
 bool BaseChannel::RegisterRtpDemuxerSink_w(
@@ -769,10 +766,9 @@ void BaseChannel::ChannelNotWritable_n() {
   RTC_LOG(LS_INFO) << "Channel not writable (" << ToString() << ")";
 }
 
-
-bool BaseChannel::UpdateLocalStreams_w(const std::vector<StreamParams>& streams,
-                                       SdpType type,
-                                       std::string& error_desc) {
+RTCError BaseChannel::UpdateLocalStreams_w(
+    const std::vector<StreamParams>& streams,
+    SdpType type) {
   // In the case of RIDs (where SSRCs are not negotiated), this method will
   // generate an SSRC for each layer in StreamParams. That representation will
   // be stored internally in `local_streams_`.
@@ -785,18 +781,16 @@ bool BaseChannel::UpdateLocalStreams_w(const std::vector<StreamParams>& streams,
   // in which niether SSRCs or RIDs are negotiated.
 
   // Check for streams that have been removed.
-  bool ret = true;
   for (const StreamParams& old_stream : local_streams_) {
     if (!old_stream.has_ssrcs() ||
         GetStream(streams, StreamFinder(&old_stream))) {
       continue;
     }
     if (!media_send_channel()->RemoveSendStream(old_stream.first_ssrc())) {
-      error_desc = StringFormat(
-          "Failed to remove send stream with ssrc %u from m-section with "
-          "mid='%s'.",
-          old_stream.first_ssrc(), mid().c_str());
-      ret = false;
+      return RTCError::InvalidParameter()
+             << "Failed to remove send stream with ssrc "
+             << old_stream.first_ssrc() << " from m-section with mid='" << mid()
+             << "'.";
     }
   }
   // Check for new streams.
@@ -818,12 +812,10 @@ bool BaseChannel::UpdateLocalStreams_w(const std::vector<StreamParams>& streams,
 
     RTC_DCHECK(new_stream.has_ssrcs() || new_stream.has_rids());
     if (new_stream.has_ssrcs() && new_stream.has_rids()) {
-      error_desc = StringFormat(
-          "Failed to add send stream: %u into m-section with mid='%s'. Stream "
-          "has both SSRCs and RIDs.",
-          new_stream.first_ssrc(), mid().c_str());
-      ret = false;
-      continue;
+      return RTCError::InvalidParameter()
+             << "Failed to add send stream: " << new_stream.first_ssrc()
+             << " into m-section with mid='" << mid()
+             << "'. Stream has both SSRCs and RIDs.";
     }
 
     // At this point we use the legacy simulcast group in StreamParams to
@@ -838,19 +830,18 @@ bool BaseChannel::UpdateLocalStreams_w(const std::vector<StreamParams>& streams,
       RTC_LOG(LS_INFO) << "Add send stream ssrc: " << new_stream.ssrcs[0]
                        << " into " << ToString();
     } else {
-      error_desc = StringFormat(
-          "Failed to add send stream ssrc: %u into m-section with mid='%s'",
-          new_stream.first_ssrc(), mid().c_str());
-      ret = false;
+      return RTCError::InvalidParameter()
+             << "Failed to add send stream ssrc: " << new_stream.first_ssrc()
+             << " into m-section with mid='" << mid() << "'";
     }
   }
   local_streams_ = all_streams;
-  return ret;
+  return RTCError::OK();
 }
 
-bool BaseChannel::UpdateRemoteStreams_w(const MediaContentDescription* content,
-                                        SdpType type,
-                                        std::string& error_desc) {
+RTCError BaseChannel::UpdateRemoteStreams_w(
+    const MediaContentDescription* content,
+    SdpType type) {
   RTC_LOG_THREAD_BLOCK_COUNT();
   bool clear_payload_types = false;
   if (!RtpTransceiverDirectionHasSend(content->direction())) {
@@ -878,11 +869,10 @@ bool BaseChannel::UpdateRemoteStreams_w(const MediaContentDescription* content,
         RTC_LOG(LS_INFO) << "Remove remote ssrc: " << old_stream.first_ssrc()
                          << " from " << ToString() << ".";
       } else {
-        error_desc = StringFormat(
-            "Failed to remove remote stream with ssrc %u from m-section with "
-            "mid='%s'.",
-            old_stream.first_ssrc(), mid().c_str());
-        return false;
+        return RTCError::InvalidParameter()
+               << "Failed to remove remote stream with ssrc "
+               << old_stream.first_ssrc() << " from m-section with mid='"
+               << mid() << "'.";
       }
     }
   }
@@ -902,13 +892,12 @@ bool BaseChannel::UpdateRemoteStreams_w(const MediaContentDescription* content,
                                  : "unsignaled")
                          << " to " << ToString();
       } else {
-        error_desc =
-            StringFormat("Failed to add remote stream ssrc: %s to %s",
-                         new_stream.has_ssrcs()
-                             ? std::to_string(new_stream.first_ssrc()).c_str()
-                             : "unsignaled",
-                         ToString().c_str());
-        return false;
+        return RTCError::InvalidParameter()
+               << "Failed to add remote stream ssrc: "
+               << (new_stream.has_ssrcs()
+                       ? std::to_string(new_stream.first_ssrc())
+                       : "unsignaled")
+               << " to " << ToString();
       }
     }
     // Update the receiving SSRCs.
@@ -919,9 +908,8 @@ bool BaseChannel::UpdateRemoteStreams_w(const MediaContentDescription* content,
 
   // Re-register the sink to update after changing the demuxer criteria.
   if (!RegisterRtpDemuxerSink_w(clear_payload_types, std::move(ssrcs))) {
-    error_desc = StringFormat("Failed to set up audio demuxing for mid='%s'.",
-                              mid().c_str());
-    return false;
+    return RTCError::InvalidParameter()
+           << "Failed to set up audio demuxing for mid='" << mid() << "'.";
   }
 
   remote_streams_ = streams;
@@ -931,7 +919,7 @@ bool BaseChannel::UpdateRemoteStreams_w(const MediaContentDescription* content,
 
   RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(1);
 
-  return true;
+  return RTCError::OK();
 }
 
 RtpHeaderExtensions BaseChannel::GetDeduplicatedRtpHeaderExtensions(
@@ -940,23 +928,20 @@ RtpHeaderExtensions BaseChannel::GetDeduplicatedRtpHeaderExtensions(
                                                    extensions_filter_);
 }
 
-bool BaseChannel::CheckRtpExtensionValidity(
-    const RtpHeaderExtensions& extensions,
-    std::string& error_desc) const {
+RTCError BaseChannel::CheckRtpExtensionValidity(
+    const RtpHeaderExtensions& extensions) const {
   std::bitset<1 + RtpExtension::kMaxId> id_used;
   for (const auto& extension : extensions) {
     if (extension.id == 0)
       continue;
     if (extension.id < RtpExtension::kMinId ||
         extension.id > RtpExtension::kMaxId) {
-      error_desc = StringFormat("Bad RTP extension ID: %s",
-                                extension.ToString().c_str());
-      return false;
+      return RTCError::InvalidParameter()
+             << "Bad RTP extension ID: " << extension.ToString();
     }
     if (id_used[extension.id]) {
-      error_desc = StringFormat("Duplicate RTP extension ID: %s",
-                                extension.ToString().c_str());
-      return false;
+      return RTCError::InvalidParameter()
+             << "Duplicate RTP extension ID: " << extension.ToString();
     }
     id_used[extension.id] = true;
   }
@@ -969,17 +954,15 @@ bool BaseChannel::CheckRtpExtensionValidity(
         [&](const RtpExtension& ext) { return ext.id == new_extension.id; });
     if (it != historical_rtp_header_extensions_.end() &&
         it->uri != new_extension.uri) {
-      error_desc = StringFormat(
-          "Failed to update RTP header extensions for m-section with "
-          "mid='%s'. RTP extension ID reassignment from %s to %s for ID "
-          "%d.",
-          mid().c_str(), it->uri.c_str(), new_extension.uri.c_str(),
-          new_extension.id);
-      return false;
+      return RTCError::InvalidParameter()
+             << "Failed to update RTP header extensions for m-section with "
+             << "mid='" << mid() << "'. RTP extension ID reassignment from "
+             << it->uri << " to " << new_extension.uri << " for ID "
+             << new_extension.id << ".";
     }
   }
 
-  return true;
+  return RTCError::OK();
 }
 
 void BaseChannel::SignalSentPacket_n(const SentPacketInfo& sent_packet) {
@@ -1030,9 +1013,8 @@ void VoiceChannel::UpdateMediaSendRecvState_w() {
                    << " send=" << send << " for " << ToString();
 }
 
-bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content,
-                                     SdpType type,
-                                     std::string& error_desc) {
+RTCError VoiceChannel::SetLocalContent_w(const MediaContentDescription* content,
+                                         SdpType type) {
   TRACE_EVENT0("webrtc", "VoiceChannel::SetLocalContent_w");
   RTC_DLOG(LS_INFO) << "Setting local voice description for " << ToString();
 
@@ -1041,8 +1023,9 @@ bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content,
   RtpHeaderExtensions header_extensions =
       GetDeduplicatedRtpHeaderExtensions(content->rtp_header_extensions());
 
-  if (!CheckRtpExtensionValidity(header_extensions, error_desc)) {
-    return false;
+  RTCError error = CheckRtpExtensionValidity(header_extensions);
+  if (!error.ok()) {
+    return error;
   }
 
   // TODO: issues.webrtc.org/383078466 - remove if pushdown on answer is enough.
@@ -1055,11 +1038,10 @@ bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content,
   recv_params.mid = mid();
 
   if (!media_receive_channel()->SetReceiverParameters(recv_params)) {
-    error_desc = StringFormat(
-        "Failed to set local audio description recv parameters for m-section "
-        "with mid='%s'.",
-        mid().c_str());
-    return false;
+    return RTCError::InvalidParameter()
+           << "Failed to set local audio description recv parameters for "
+              "m-section with mid='"
+           << mid() << "'.";
   }
 
   std::optional<flat_set<uint8_t>> payload_types;
@@ -1078,17 +1060,16 @@ bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content,
     send_params.extensions = header_extensions;
     send_params.extmap_allow_mixed = content->extmap_allow_mixed();
     if (!media_send_channel()->SetSenderParameters(send_params)) {
-      error_desc = StringFormat(
-          "Failed to set send parameters for m-section with mid='%s'.",
-          mid().c_str());
-      return false;
+      return RTCError::InvalidParameter()
+             << "Failed to set send parameters for m-section with mid='"
+             << mid() << "'.";
     }
     last_send_params_ = send_params;
   }
 
-  if (!UpdateLocalStreams_w(content->streams(), type, error_desc)) {
-    RTC_DCHECK(!error_desc.empty());
-    return false;
+  error = UpdateLocalStreams_w(content->streams(), type);
+  if (!error.ok()) {
+    return error;
   }
 
   set_local_content_direction(content->direction());
@@ -1098,19 +1079,15 @@ bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content,
   // TODO: https://issues.webrtc.org/360058654 - reenable after cleanup
   // RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(0);
 
-  bool success = MaybeUpdateDemuxerAndRtpExtensions_w(
+  return MaybeUpdateDemuxerAndRtpExtensions_w(
       /*update_demuxer=*/false, std::move(payload_types),
       recv_params.extensions,
-      /*ssrcs=*/std::nullopt, error_desc);
-
-  // RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(1);
-
-  return success;
+      /*ssrcs=*/std::nullopt);
 }
 
-bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content,
-                                      SdpType type,
-                                      std::string& error_desc) {
+RTCError VoiceChannel::SetRemoteContent_w(
+    const MediaContentDescription* content,
+    SdpType type) {
   TRACE_EVENT0("webrtc", "VoiceChannel::SetRemoteContent_w");
   RTC_LOG(LS_INFO) << "Setting remote voice description for " << ToString();
 
@@ -1119,28 +1096,25 @@ bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content,
                                         &send_params);
   send_params.mid = mid();
 
-  if (!CheckRtpExtensionValidity(send_params.extensions, error_desc)) {
-    return false;
+  RTCError error = CheckRtpExtensionValidity(send_params.extensions);
+  if (!error.ok()) {
+    return error;
   }
 
-  bool parameters_applied =
-      media_send_channel()->SetSenderParameters(send_params);
-  if (!parameters_applied) {
-    error_desc = StringFormat(
-        "Failed to set remote audio description send parameters for m-section "
-        "with mid='%s'.",
-        mid().c_str());
-    return false;
+  if (!media_send_channel()->SetSenderParameters(send_params)) {
+    return RTCError::InvalidParameter()
+           << "Failed to set remote audio description send parameters for "
+              "m-section with mid='"
+           << mid() << "'.";
   }
 
   if (type == SdpType::kAnswer || type == SdpType::kPrAnswer) {
     AudioReceiverParameters recv_params = last_recv_params_;
     recv_params.extensions = send_params.extensions;
     if (!media_receive_channel()->SetReceiverParameters(recv_params)) {
-      error_desc = StringFormat(
-          "Failed to set recv parameters for m-section with mid='%s'.",
-          mid().c_str());
-      return false;
+      return RTCError::InvalidParameter()
+             << "Failed to set recv parameters for m-section with mid='"
+             << mid() << "'.";
     }
     last_recv_params_ = recv_params;
   }
@@ -1158,18 +1132,14 @@ bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content,
       media_send_channel()->SenderNonSenderRttEnabled());
   last_send_params_ = send_params;
 
-  if (!UpdateRemoteStreams_w(content, type, error_desc)) {
-    return false;
-  }
-
-  bool success = true;
-  if (type == SdpType::kAnswer || type == SdpType::kPrAnswer) {
-    success = MaybeUpdateDemuxerAndRtpExtensions_w(
+  error = UpdateRemoteStreams_w(content, type);
+  if (error.ok() && (type == SdpType::kAnswer || type == SdpType::kPrAnswer)) {
+    error = MaybeUpdateDemuxerAndRtpExtensions_w(
         /*update_demuxer=*/false, /*payload_types=*/std::nullopt,
-        last_recv_params_.extensions, /*ssrcs=*/std::nullopt, error_desc);
+        last_recv_params_.extensions, /*ssrcs=*/std::nullopt);
   }
 
-  return success;
+  return error;
 }
 
 VideoChannel::VideoChannel(
@@ -1212,9 +1182,8 @@ void VideoChannel::UpdateMediaSendRecvState_w() {
                    << " send=" << send << " for " << ToString();
 }
 
-bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
-                                     SdpType type,
-                                     std::string& error_desc) {
+RTCError VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
+                                         SdpType type) {
   TRACE_EVENT0("webrtc", "VideoChannel::SetLocalContent_w");
 
   RTC_LOG_THREAD_BLOCK_COUNT();
@@ -1222,8 +1191,9 @@ bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
   RtpHeaderExtensions header_extensions =
       GetDeduplicatedRtpHeaderExtensions(content->rtp_header_extensions());
 
-  if (!CheckRtpExtensionValidity(header_extensions, error_desc)) {
-    return false;
+  RTCError error = CheckRtpExtensionValidity(header_extensions);
+  if (!error.ok()) {
+    return error;
   }
 
   // TODO: issues.webrtc.org/383078466 - remove if pushdown on answer is enough.
@@ -1244,19 +1214,17 @@ bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
   // offer by ignoring the packetiztion and fall back to standard packetization
   // instead.
   if (type == SdpType::kAnswer || type == SdpType::kPrAnswer) {
-    RTCError status = MaybeIgnorePacketization(recv_params, send_params);
-    if (!status.ok()) {
-      error_desc = status.message();
-      return false;
+    error = MaybeIgnorePacketization(recv_params, send_params);
+    if (!error.ok()) {
+      return error;
     }
   }
 
   if (!media_receive_channel()->SetReceiverParameters(recv_params)) {
-    error_desc = StringFormat(
-        "Failed to set local video description recv parameters for m-section "
-        "with mid='%s'.",
-        mid().c_str());
-    return false;
+    return RTCError::InvalidParameter()
+           << "Failed to set local video description recv parameters for "
+              "m-section with mid='"
+           << mid() << "'.";
   }
 
   std::optional<flat_set<uint8_t>> payload_types;
@@ -1271,17 +1239,16 @@ bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
 
   if (type == SdpType::kAnswer || type == SdpType::kPrAnswer) {
     if (!media_send_channel()->SetSenderParameters(send_params)) {
-      error_desc = StringFormat(
-          "Failed to set send parameters for m-section with mid='%s'.",
-          mid().c_str());
-      return false;
+      return RTCError::InvalidParameter()
+             << "Failed to set send parameters for m-section with mid='"
+             << mid() << "'.";
     }
     last_send_params_ = send_params;
   }
 
-  if (!UpdateLocalStreams_w(content->streams(), type, error_desc)) {
-    RTC_DCHECK(!error_desc.empty());
-    return false;
+  error = UpdateLocalStreams_w(content->streams(), type);
+  if (!error.ok()) {
+    return error;
   }
 
   set_local_content_direction(content->direction());
@@ -1289,18 +1256,14 @@ bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
 
   RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(0);
 
-  bool success = MaybeUpdateDemuxerAndRtpExtensions_w(
+  return MaybeUpdateDemuxerAndRtpExtensions_w(
       /*update_demuxer=*/false, std::move(payload_types),
-      recv_params.extensions, /*ssrcs=*/std::nullopt, error_desc);
-
-  RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(1);
-
-  return success;
+      recv_params.extensions, /*ssrcs=*/std::nullopt);
 }
 
-bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content,
-                                      SdpType type,
-                                      std::string& error_desc) {
+RTCError VideoChannel::SetRemoteContent_w(
+    const MediaContentDescription* content,
+    SdpType type) {
   TRACE_EVENT0("webrtc", "VideoChannel::SetRemoteContent_w");
   RTC_LOG(LS_INFO) << "Setting remote video description for " << ToString();
 
@@ -1310,8 +1273,9 @@ bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content,
   send_params.mid = mid();
   send_params.conference_mode = content->conference_mode();
 
-  if (!CheckRtpExtensionValidity(send_params.extensions, error_desc)) {
-    return false;
+  RTCError error = CheckRtpExtensionValidity(send_params.extensions);
+  if (!error.ok()) {
+    return error;
   }
 
   VideoReceiverParameters recv_params = last_recv_params_;
@@ -1322,47 +1286,38 @@ bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content,
   // offer by ignoring the packetiztion and fall back to standard packetization
   // instead.
   if (type == SdpType::kAnswer || type == SdpType::kPrAnswer) {
-    RTCError status = MaybeIgnorePacketization(send_params, recv_params);
-    if (!status.ok()) {
-      error_desc = status.message();
-      return false;
+    error = MaybeIgnorePacketization(send_params, recv_params);
+    if (!error.ok()) {
+      return error;
     }
   }
 
   if (!media_send_channel()->SetSenderParameters(send_params)) {
-    error_desc = StringFormat(
-        "Failed to set remote video description send parameters for m-section "
-        "with mid='%s'.",
-        mid().c_str());
-    return false;
+    return RTCError::InvalidParameter()
+           << "Failed to set remote video description send parameters for "
+              "m-section with mid='"
+           << mid() << "'.";
   }
   if (type == SdpType::kAnswer || type == SdpType::kPrAnswer) {
     recv_params.extensions = send_params.extensions;
     recv_params.rtcp.reduced_size = send_params.rtcp.reduced_size;
     if (!media_receive_channel()->SetReceiverParameters(recv_params)) {
-      error_desc = StringFormat(
-          "Failed to set recv parameters for m-section with mid='%s'.",
-          mid().c_str());
-      return false;
+      return RTCError::InvalidParameter()
+             << "Failed to set recv parameters for m-section with mid='"
+             << mid() << "'.";
     }
     last_recv_params_ = recv_params;
   }
   last_send_params_ = send_params;
 
-  if (!UpdateRemoteStreams_w(content, type, error_desc)) {
-    return false;
-  }
-
-  if (type == SdpType::kAnswer || type == SdpType::kPrAnswer) {
-    bool success = MaybeUpdateDemuxerAndRtpExtensions_w(
+  error = UpdateRemoteStreams_w(content, type);
+  if (error.ok() && (type == SdpType::kAnswer || type == SdpType::kPrAnswer)) {
+    error = MaybeUpdateDemuxerAndRtpExtensions_w(
         /*update_demuxer=*/false, /*payload_types=*/std::nullopt,
-        last_recv_params_.extensions, /*ssrcs=*/std::nullopt, error_desc);
-    if (!success) {
-      return false;
-    }
+        last_recv_params_.extensions, /*ssrcs=*/std::nullopt);
   }
 
-  return true;
+  return error;
 }
 
 }  // namespace webrtc
diff --git a/pc/channel.h b/pc/channel.h
index f4488cd612..6b4438ab19 100644
--- a/pc/channel.h
+++ b/pc/channel.h
@@ -24,6 +24,7 @@
 #include "api/crypto/crypto_options.h"
 #include "api/jsep.h"
 #include "api/media_types.h"
+#include "api/rtc_error.h"
 #include "api/rtp_parameters.h"
 #include "api/rtp_transceiver_direction.h"
 #include "api/scoped_refptr.h"
@@ -122,12 +123,10 @@ class BaseChannel : public ChannelInterface,
   }
 
   // Channel control
-  bool SetLocalContent(const MediaContentDescription* content,
-                       SdpType type,
-                       std::string& error_desc) override;
-  bool SetRemoteContent(const MediaContentDescription* content,
-                        SdpType type,
-                        std::string& error_desc) override;
+  RTCError SetLocalContent(const MediaContentDescription* content,
+                           SdpType type) override;
+  RTCError SetRemoteContent(const MediaContentDescription* content,
+                            SdpType type) override;
   // Controls whether this channel will receive packets on the basis of
   // matching payload type alone. This is needed for legacy endpoints that
   // don't signal SSRCs or use MID/RID, but doesn't make sense if there is
@@ -255,21 +254,15 @@ class BaseChannel : public ChannelInterface,
   // Updates the send/recv state of the media channel.
   virtual void UpdateMediaSendRecvState_w() RTC_RUN_ON(worker_thread()) = 0;
 
-  bool UpdateLocalStreams_w(const std::vector<StreamParams>& streams,
-                            SdpType type,
-                            std::string& error_desc)
-      RTC_RUN_ON(worker_thread());
-  bool UpdateRemoteStreams_w(const MediaContentDescription* content,
-                             SdpType type,
-                             std::string& error_desc)
-      RTC_RUN_ON(worker_thread());
-  virtual bool SetLocalContent_w(const MediaContentDescription* content,
-                                 SdpType type,
-                                 std::string& error_desc)
+  RTCError UpdateLocalStreams_w(const std::vector<StreamParams>& streams,
+                                SdpType type) RTC_RUN_ON(worker_thread());
+  RTCError UpdateRemoteStreams_w(const MediaContentDescription* content,
+                                 SdpType type) RTC_RUN_ON(worker_thread());
+  virtual RTCError SetLocalContent_w(const MediaContentDescription* content,
+                                     SdpType type)
       RTC_RUN_ON(worker_thread()) = 0;
-  virtual bool SetRemoteContent_w(const MediaContentDescription* content,
-                                  SdpType type,
-                                  std::string& error_desc)
+  virtual RTCError SetRemoteContent_w(const MediaContentDescription* content,
+                                      SdpType type)
       RTC_RUN_ON(worker_thread()) = 0;
 
   // Returns a list of RTP header extensions where any extension URI is unique.
@@ -282,19 +275,17 @@ class BaseChannel : public ChannelInterface,
   // This verifies that all extension IDs are within the valid range,
   // that there are no duplicate IDs, and that no existing extension ID
   // has been reassigned to a different URI.
-  bool CheckRtpExtensionValidity(const RtpHeaderExtensions& extensions,
-                                 std::string& error_desc) const
-      RTC_RUN_ON(worker_thread());
+  RTCError CheckRtpExtensionValidity(
+      const RtpHeaderExtensions& extensions) const RTC_RUN_ON(worker_thread());
 
   // Returns `true` if either an update wasn't needed or one was successfully
   // applied. If the return value is `false`, then updating the demuxer criteria
   // failed, which needs to be treated as an error.
-  bool MaybeUpdateDemuxerAndRtpExtensions_w(
+  RTCError MaybeUpdateDemuxerAndRtpExtensions_w(
       bool update_demuxer,
       std::optional<flat_set<uint8_t>> payload_types,
       const RtpHeaderExtensions& extensions,
-      std::optional<flat_set<uint32_t>> ssrcs,
-      std::string& error_desc) RTC_RUN_ON(worker_thread());
+      std::optional<flat_set<uint32_t>> ssrcs) RTC_RUN_ON(worker_thread());
 
   // Registers a demuxer criteria with the transport, on the network thread.
   // This function will fail if there's no transport of if a sink is already
@@ -432,13 +423,10 @@ class VoiceChannel : public BaseChannel {
  private:
   // overrides from BaseChannel
   void UpdateMediaSendRecvState_w() RTC_RUN_ON(worker_thread()) override;
-  bool SetLocalContent_w(const MediaContentDescription* content,
-                         SdpType type,
-                         std::string& error_desc)
-      RTC_RUN_ON(worker_thread()) override;
-  bool SetRemoteContent_w(const MediaContentDescription* content,
-                          SdpType type,
-                          std::string& error_desc)
+  RTCError SetLocalContent_w(const MediaContentDescription* content,
+                             SdpType type) RTC_RUN_ON(worker_thread()) override;
+  RTCError SetRemoteContent_w(const MediaContentDescription* content,
+                              SdpType type)
       RTC_RUN_ON(worker_thread()) override;
 
   // Last AudioSenderParameter sent down to the media_channel() via
@@ -499,13 +487,10 @@ class VideoChannel : public BaseChannel {
  private:
   // overrides from BaseChannel
   void UpdateMediaSendRecvState_w() RTC_RUN_ON(worker_thread()) override;
-  bool SetLocalContent_w(const MediaContentDescription* content,
-                         SdpType type,
-                         std::string& error_desc)
-      RTC_RUN_ON(worker_thread()) override;
-  bool SetRemoteContent_w(const MediaContentDescription* content,
-                          SdpType type,
-                          std::string& error_desc)
+  RTCError SetLocalContent_w(const MediaContentDescription* content,
+                             SdpType type) RTC_RUN_ON(worker_thread()) override;
+  RTCError SetRemoteContent_w(const MediaContentDescription* content,
+                              SdpType type)
       RTC_RUN_ON(worker_thread()) override;
 
   // Last VideoSenderParameters sent down to the media_channel() via
diff --git a/pc/channel_interface.h b/pc/channel_interface.h
index 1e1c9bd32e..97fed01e8f 100644
--- a/pc/channel_interface.h
+++ b/pc/channel_interface.h
@@ -18,6 +18,7 @@
 #include "absl/strings/string_view.h"
 #include "api/jsep.h"
 #include "api/media_types.h"
+#include "api/rtc_error.h"
 #include "media/base/media_channel.h"
 #include "media/base/stream_params.h"
 #include "pc/rtp_transport_internal.h"
@@ -86,12 +87,10 @@ class ChannelInterface {
       absl::AnyInvocable<void(const RtpPacketReceived&)> callback) = 0;
 
   // Channel control
-  virtual bool SetLocalContent(const MediaContentDescription* content,
-                               SdpType type,
-                               std::string& error_desc) = 0;
-  virtual bool SetRemoteContent(const MediaContentDescription* content,
-                                SdpType type,
-                                std::string& error_desc) = 0;
+  virtual RTCError SetLocalContent(const MediaContentDescription* content,
+                                   SdpType type) = 0;
+  virtual RTCError SetRemoteContent(const MediaContentDescription* content,
+                                    SdpType type) = 0;
   virtual bool SetPayloadTypeDemuxingEnabled(bool enabled) = 0;
 
   // Access to the local and remote streams that were set on the channel.
diff --git a/pc/channel_unittest.cc b/pc/channel_unittest.cc
index 3f588332bc..aae219ffd5 100644
--- a/pc/channel_unittest.cc
+++ b/pc/channel_unittest.cc
@@ -24,6 +24,7 @@
 #include "api/crypto/crypto_options.h"
 #include "api/field_trials.h"
 #include "api/jsep.h"
+#include "api/rtc_error.h"
 #include "api/rtp_headers.h"
 #include "api/rtp_parameters.h"
 #include "api/rtp_transceiver_direction.h"
@@ -75,6 +76,7 @@ using ::webrtc::FakeVoiceMediaSendChannel;
 using ::webrtc::FieldTrials;
 using ::webrtc::RidDescription;
 using ::webrtc::RidDirection;
+using ::webrtc::RTCError;
 using ::webrtc::RtpExtension;
 using ::webrtc::RtpTransceiverDirection;
 using ::webrtc::SdpType;
@@ -197,6 +199,7 @@ class ChannelTest : public ::testing::Test {
                        typename T::Options(), network_thread_),
                    flags1, flags2);
   }
+
   void CreateChannels(std::unique_ptr<typename T::MediaSendChannel> ch1s,
                       std::unique_ptr<typename T::MediaReceiveChannel> ch1r,
                       std::unique_ptr<typename T::MediaSendChannel> ch2s,
@@ -305,6 +308,7 @@ class ChannelTest : public ::testing::Test {
       AddLegacyStreamInContent(kSsrc2, flags2, &remote_media_content2_);
     }
   }
+
   std::unique_ptr<typename T::Channel> CreateChannel(
       webrtc::Thread* worker_thread,
       webrtc::Thread* network_thread,
@@ -411,65 +415,61 @@ class ChannelTest : public ::testing::Test {
   }
 
   bool SendInitiate() {
-    std::string err;
-    bool result = channel1_->SetLocalContent(&local_media_content1_,
-                                             SdpType::kOffer, err);
-    if (result) {
+    RTCError result =
+        channel1_->SetLocalContent(&local_media_content1_, SdpType::kOffer);
+    if (result.ok()) {
       channel1_->Enable(true);
       FlushCurrentThread();
-      result = channel2_->SetRemoteContent(&remote_media_content1_,
-                                           SdpType::kOffer, err);
-      if (result) {
+      result =
+          channel2_->SetRemoteContent(&remote_media_content1_, SdpType::kOffer);
+      if (result.ok()) {
         ConnectFakeTransports();
         result = channel2_->SetLocalContent(&local_media_content2_,
-                                            SdpType::kAnswer, err);
+                                            SdpType::kAnswer);
       }
     }
-    return result;
+    return result.ok();
   }
 
   bool SendAccept() {
     channel2_->Enable(true);
     FlushCurrentThread();
-    std::string err;
-    return channel1_->SetRemoteContent(&remote_media_content2_,
-                                       SdpType::kAnswer, err);
+    return channel1_
+        ->SetRemoteContent(&remote_media_content2_, SdpType::kAnswer)
+        .ok();
   }
 
   bool SendOffer() {
-    std::string err;
-    bool result = channel1_->SetLocalContent(&local_media_content1_,
-                                             SdpType::kOffer, err);
-    if (result) {
+    RTCError result =
+        channel1_->SetLocalContent(&local_media_content1_, SdpType::kOffer);
+    if (result.ok()) {
       channel1_->Enable(true);
-      result = channel2_->SetRemoteContent(&remote_media_content1_,
-                                           SdpType::kOffer, err);
+      result =
+          channel2_->SetRemoteContent(&remote_media_content1_, SdpType::kOffer);
     }
-    return result;
+    return result.ok();
   }
 
   bool SendProvisionalAnswer() {
-    std::string err;
-    bool result = channel2_->SetLocalContent(&local_media_content2_,
-                                             SdpType::kPrAnswer, err);
-    if (result) {
+    RTCError result =
+        channel2_->SetLocalContent(&local_media_content2_, SdpType::kPrAnswer);
+    if (result.ok()) {
       channel2_->Enable(true);
       result = channel1_->SetRemoteContent(&remote_media_content2_,
-                                           SdpType::kPrAnswer, err);
+                                           SdpType::kPrAnswer);
       ConnectFakeTransports();
     }
-    return result;
+    return result.ok();
   }
 
   bool SendFinalAnswer() {
-    std::string err;
-    bool result = channel2_->SetLocalContent(&local_media_content2_,
-                                             SdpType::kAnswer, err);
-    if (result) {
+    RTCError result =
+        channel2_->SetLocalContent(&local_media_content2_, SdpType::kAnswer);
+    if (result.ok()) {
       result = channel1_->SetRemoteContent(&remote_media_content2_,
-                                           SdpType::kAnswer, err);
+                                           SdpType::kAnswer);
     }
-    return result;
+    return result.ok();
   }
 
   void SendRtp(typename T::MediaSendChannel* media_channel,
@@ -636,10 +636,9 @@ class ChannelTest : public ::testing::Test {
     CreateChannels(0, 0);
     typename T::Content content;
     CreateContent(0, kPcmuCodec, kH264Codec, &content);
-    std::string err;
-    EXPECT_TRUE(channel1_->SetLocalContent(&content, SdpType::kOffer, err));
+    EXPECT_TRUE(channel1_->SetLocalContent(&content, SdpType::kOffer).ok());
     EXPECT_EQ(0U, media_send_channel1_impl()->send_codecs().size());
-    EXPECT_TRUE(channel1_->SetRemoteContent(&content, SdpType::kAnswer, err));
+    EXPECT_TRUE(channel1_->SetRemoteContent(&content, SdpType::kAnswer).ok());
     ASSERT_EQ(1U, media_send_channel1_impl()->send_codecs().size());
     EXPECT_EQ(content.codecs()[0],
               media_send_channel1_impl()->send_codecs()[0]);
@@ -659,11 +658,8 @@ class ChannelTest : public ::testing::Test {
     });
 
     CreateChannels(0, 0);
-    std::string err;
-    ASSERT_TRUE(channel1_->SetLocalContent(&local, SdpType::kOffer, err))
-        << err;
-    ASSERT_TRUE(channel1_->SetRemoteContent(&remote, SdpType::kAnswer, err))
-        << err;
+    ASSERT_TRUE(channel1_->SetLocalContent(&local, SdpType::kOffer).ok());
+    ASSERT_TRUE(channel1_->SetRemoteContent(&remote, SdpType::kAnswer).ok());
 
     EXPECT_THAT(media_receive_channel1_impl()->recv_extensions(),
                 ElementsAre(AllOf(Field("id", &RtpExtension::id, 1),
@@ -688,11 +684,8 @@ class ChannelTest : public ::testing::Test {
     });
 
     CreateChannels(0, 0);
-    std::string err;
-    ASSERT_TRUE(channel1_->SetRemoteContent(&remote, SdpType::kOffer, err))
-        << err;
-    ASSERT_TRUE(channel1_->SetLocalContent(&local, SdpType::kAnswer, err))
-        << err;
+    ASSERT_TRUE(channel1_->SetRemoteContent(&remote, SdpType::kOffer).ok());
+    ASSERT_TRUE(channel1_->SetLocalContent(&local, SdpType::kAnswer).ok());
 
     EXPECT_THAT(media_receive_channel1_impl()->recv_extensions(),
                 ElementsAre(AllOf(Field("id", &RtpExtension::id, 1),
@@ -713,9 +706,10 @@ class ChannelTest : public ::testing::Test {
     });
 
     CreateChannels(0, 0);
-    std::string err;
-    EXPECT_FALSE(channel1_->SetLocalContent(&local, SdpType::kOffer, err));
-    EXPECT_THAT(err, ::testing::HasSubstr("Duplicate RTP extension ID"));
+    RTCError error = channel1_->SetLocalContent(&local, SdpType::kOffer);
+    EXPECT_FALSE(error.ok());
+    EXPECT_THAT(error.message(),
+                ::testing::HasSubstr("Duplicate RTP extension ID"));
   }
 
   void TestInvalidRtpHeaderExtensionIds() {
@@ -726,9 +720,9 @@ class ChannelTest : public ::testing::Test {
     });
 
     CreateChannels(0, 0);
-    std::string err;
-    EXPECT_FALSE(channel1_->SetLocalContent(&local, SdpType::kOffer, err));
-    EXPECT_THAT(err, ::testing::HasSubstr("Bad RTP extension ID"));
+    RTCError error = channel1_->SetLocalContent(&local, SdpType::kOffer);
+    EXPECT_FALSE(error.ok());
+    EXPECT_THAT(error.message(), ::testing::HasSubstr("Bad RTP extension ID"));
   }
 
   void TestRtpHeaderExtensionIdReassignment() {
@@ -739,17 +733,18 @@ class ChannelTest : public ::testing::Test {
     });
 
     CreateChannels(0, 0);
-    std::string err;
-    ASSERT_TRUE(channel1_->SetLocalContent(&local, SdpType::kOffer, err));
+    ASSERT_TRUE(channel1_->SetLocalContent(&local, SdpType::kOffer).ok());
 
     typename T::Content local_updated;
     CreateContent(/*flags=*/0, kPcmuCodec, kH264Codec, &local_updated);
     local_updated.set_rtp_header_extensions({
         RtpExtension(RtpExtension::kVideoRotationUri, 1),
     });
-    EXPECT_FALSE(
-        channel1_->SetLocalContent(&local_updated, SdpType::kOffer, err));
-    EXPECT_THAT(err, ::testing::HasSubstr("RTP extension ID reassignment"));
+    RTCError error =
+        channel1_->SetLocalContent(&local_updated, SdpType::kOffer);
+    EXPECT_FALSE(error.ok());
+    EXPECT_THAT(error.message(),
+                ::testing::HasSubstr("RTP extension ID reassignment"));
   }
 
   void TestRtpHeaderExtensionIdHistoryReassignment() {
@@ -760,22 +755,23 @@ class ChannelTest : public ::testing::Test {
     });
 
     CreateChannels(0, 0);
-    std::string err;
-    ASSERT_TRUE(channel1_->SetLocalContent(&local, SdpType::kOffer, err));
+    ASSERT_TRUE(channel1_->SetLocalContent(&local, SdpType::kOffer).ok());
 
     typename T::Content local_empty;
     CreateContent(/*flags=*/0, kPcmuCodec, kH264Codec, &local_empty);
     local_empty.set_rtp_header_extensions({});
-    ASSERT_TRUE(channel1_->SetLocalContent(&local_empty, SdpType::kOffer, err));
+    ASSERT_TRUE(channel1_->SetLocalContent(&local_empty, SdpType::kOffer).ok());
 
     typename T::Content local_updated;
     CreateContent(/*flags=*/0, kPcmuCodec, kH264Codec, &local_updated);
     local_updated.set_rtp_header_extensions({
         RtpExtension(RtpExtension::kVideoRotationUri, 1),
     });
-    EXPECT_FALSE(
-        channel1_->SetLocalContent(&local_updated, SdpType::kOffer, err));
-    EXPECT_THAT(err, ::testing::HasSubstr("RTP extension ID reassignment"));
+    RTCError error =
+        channel1_->SetLocalContent(&local_updated, SdpType::kOffer);
+    EXPECT_FALSE(error.ok());
+    EXPECT_THAT(error.message(),
+                ::testing::HasSubstr("RTP extension ID reassignment"));
   }
 
   // Test that SetLocalContent and SetRemoteContent properly configure
@@ -789,12 +785,12 @@ class ChannelTest : public ::testing::Test {
     auto offer_enum = offer ? (T::Content::kSession) : (T::Content::kNo);
     auto answer_enum = answer ? (T::Content::kSession) : (T::Content::kNo);
     content.set_extmap_allow_mixed_enum(offer_enum);
-    std::string err;
-    EXPECT_TRUE(channel1_->SetLocalContent(&content, SdpType::kOffer, err));
+    EXPECT_TRUE(channel1_->SetLocalContent(&content, SdpType::kOffer).ok());
     content.set_extmap_allow_mixed_enum(answer_enum);
-    EXPECT_TRUE(channel1_->SetRemoteContent(&content, SdpType::kAnswer, err));
+    EXPECT_TRUE(channel1_->SetRemoteContent(&content, SdpType::kAnswer).ok());
     EXPECT_EQ(answer, media_send_channel1_impl()->ExtmapAllowMixed());
   }
+
   void TestSetContentsExtmapAllowMixedCallee(bool offer, bool answer) {
     // For a callee, SetRemoteContent() is called first with an offer and next
     // SetLocalContent() is called with the answer.
@@ -804,10 +800,9 @@ class ChannelTest : public ::testing::Test {
     auto offer_enum = offer ? (T::Content::kSession) : (T::Content::kNo);
     auto answer_enum = answer ? (T::Content::kSession) : (T::Content::kNo);
     content.set_extmap_allow_mixed_enum(offer_enum);
-    std::string err;
-    EXPECT_TRUE(channel1_->SetRemoteContent(&content, SdpType::kOffer, err));
+    EXPECT_TRUE(channel1_->SetRemoteContent(&content, SdpType::kOffer).ok());
     content.set_extmap_allow_mixed_enum(answer_enum);
-    EXPECT_TRUE(channel1_->SetLocalContent(&content, SdpType::kAnswer, err));
+    EXPECT_TRUE(channel1_->SetLocalContent(&content, SdpType::kAnswer).ok());
     EXPECT_EQ(answer, media_send_channel1()->ExtmapAllowMixed());
   }
 
@@ -816,11 +811,10 @@ class ChannelTest : public ::testing::Test {
   void TestSetContentsNullOffer() {
     CreateChannels(0, 0);
     typename T::Content content;
-    std::string err;
-    EXPECT_TRUE(channel1_->SetLocalContent(&content, SdpType::kOffer, err));
+    EXPECT_TRUE(channel1_->SetLocalContent(&content, SdpType::kOffer).ok());
     CreateContent(0, kPcmuCodec, kH264Codec, &content);
     EXPECT_EQ(0U, media_send_channel1_impl()->send_codecs().size());
-    EXPECT_TRUE(channel1_->SetRemoteContent(&content, SdpType::kAnswer, err));
+    EXPECT_TRUE(channel1_->SetRemoteContent(&content, SdpType::kAnswer).ok());
     ASSERT_EQ(1U, media_send_channel1_impl()->send_codecs().size());
     EXPECT_EQ(content.codecs()[0],
               media_send_channel1_impl()->send_codecs()[0]);
@@ -834,13 +828,12 @@ class ChannelTest : public ::testing::Test {
     CreateContent(0, kPcmuCodec, kH264Codec, &content);
     // Both sides agree on mux. Should no longer be a separate RTCP channel.
     content.set_rtcp_mux(true);
-    std::string err;
-    EXPECT_TRUE(channel1_->SetLocalContent(&content, SdpType::kOffer, err));
-    EXPECT_TRUE(channel1_->SetRemoteContent(&content, SdpType::kAnswer, err));
+    EXPECT_TRUE(channel1_->SetLocalContent(&content, SdpType::kOffer).ok());
+    EXPECT_TRUE(channel1_->SetRemoteContent(&content, SdpType::kAnswer).ok());
     // Only initiator supports mux. Should still have a separate RTCP channel.
-    EXPECT_TRUE(channel2_->SetLocalContent(&content, SdpType::kOffer, err));
+    EXPECT_TRUE(channel2_->SetLocalContent(&content, SdpType::kOffer).ok());
     content.set_rtcp_mux(false);
-    EXPECT_TRUE(channel2_->SetRemoteContent(&content, SdpType::kAnswer, err));
+    EXPECT_TRUE(channel2_->SetRemoteContent(&content, SdpType::kAnswer).ok());
   }
 
   // Test that SetLocalContent and SetRemoteContent properly set RTCP
@@ -851,25 +844,24 @@ class ChannelTest : public ::testing::Test {
     CreateContent(0, kPcmuCodec, kH264Codec, &content);
     // Both sides agree on reduced size.
     content.set_rtcp_reduced_size(true);
-    std::string err;
     // The RTCP mode is a send property and should be configured based on
     // the remote content and not the local content.
-    EXPECT_TRUE(channel1_->SetLocalContent(&content, SdpType::kOffer, err));
+    EXPECT_TRUE(channel1_->SetLocalContent(&content, SdpType::kOffer).ok());
     EXPECT_EQ(media_receive_channel1_impl()->RtcpMode(),
               webrtc::RtcpMode::kCompound);
-    EXPECT_TRUE(channel1_->SetRemoteContent(&content, SdpType::kAnswer, err));
+    EXPECT_TRUE(channel1_->SetRemoteContent(&content, SdpType::kAnswer).ok());
     EXPECT_EQ(media_receive_channel1_impl()->RtcpMode(),
               webrtc::RtcpMode::kReducedSize);
     // Only initiator supports reduced size.
-    EXPECT_TRUE(channel2_->SetLocalContent(&content, SdpType::kOffer, err));
+    EXPECT_TRUE(channel2_->SetLocalContent(&content, SdpType::kOffer).ok());
     EXPECT_EQ(media_receive_channel2_impl()->RtcpMode(),
               webrtc::RtcpMode::kCompound);
     content.set_rtcp_reduced_size(false);
-    EXPECT_TRUE(channel2_->SetRemoteContent(&content, SdpType::kAnswer, err));
+    EXPECT_TRUE(channel2_->SetRemoteContent(&content, SdpType::kAnswer).ok());
     EXPECT_EQ(media_receive_channel2_impl()->RtcpMode(),
               webrtc::RtcpMode::kCompound);
     // Peer renegotiates without reduced size.
-    EXPECT_TRUE(channel1_->SetRemoteContent(&content, SdpType::kAnswer, err));
+    EXPECT_TRUE(channel1_->SetRemoteContent(&content, SdpType::kAnswer).ok());
     EXPECT_EQ(media_receive_channel1_impl()->RtcpMode(),
               webrtc::RtcpMode::kCompound);
   }
@@ -893,21 +885,19 @@ class ChannelTest : public ::testing::Test {
     typename T::Content content1;
     CreateContent(0, kPcmuCodec, kH264Codec, &content1);
     content1.AddStream(stream1);
-    std::string err;
-    EXPECT_TRUE(channel1_->SetLocalContent(&content1, SdpType::kOffer, err));
+    EXPECT_TRUE(channel1_->SetLocalContent(&content1, SdpType::kOffer).ok());
     channel1_->Enable(true);
     EXPECT_EQ(1u, media_send_channel1_impl()->send_streams().size());
-
-    EXPECT_TRUE(channel2_->SetRemoteContent(&content1, SdpType::kOffer, err));
+    EXPECT_TRUE(channel2_->SetRemoteContent(&content1, SdpType::kOffer).ok());
     EXPECT_EQ(1u, media_receive_channel2_impl()->recv_streams().size());
     ConnectFakeTransports();
 
     // Channel 2 do not send anything.
     typename T::Content content2;
     CreateContent(0, kPcmuCodec, kH264Codec, &content2);
-    EXPECT_TRUE(channel1_->SetRemoteContent(&content2, SdpType::kAnswer, err));
+    EXPECT_TRUE(channel1_->SetRemoteContent(&content2, SdpType::kAnswer).ok());
     EXPECT_EQ(0u, media_receive_channel1_impl()->recv_streams().size());
-    EXPECT_TRUE(channel2_->SetLocalContent(&content2, SdpType::kAnswer, err));
+    EXPECT_TRUE(channel2_->SetLocalContent(&content2, SdpType::kAnswer).ok());
     channel2_->Enable(true);
     EXPECT_EQ(0u, media_send_channel2_impl()->send_streams().size());
 
@@ -919,21 +909,21 @@ class ChannelTest : public ::testing::Test {
     typename T::Content content3;
     CreateContent(0, kPcmuCodec, kH264Codec, &content3);
     content3.AddStream(stream2);
-    EXPECT_TRUE(channel2_->SetLocalContent(&content3, SdpType::kOffer, err));
+    EXPECT_TRUE(channel2_->SetLocalContent(&content3, SdpType::kOffer).ok());
     ASSERT_EQ(1u, media_send_channel2_impl()->send_streams().size());
     EXPECT_EQ(stream2, media_send_channel2_impl()->send_streams()[0]);
 
-    EXPECT_TRUE(channel1_->SetRemoteContent(&content3, SdpType::kOffer, err));
+    EXPECT_TRUE(channel1_->SetRemoteContent(&content3, SdpType::kOffer).ok());
     ASSERT_EQ(1u, media_receive_channel1_impl()->recv_streams().size());
     EXPECT_EQ(stream2, media_receive_channel1_impl()->recv_streams()[0]);
 
     // Channel 1 replies but stop sending stream1.
     typename T::Content content4;
     CreateContent(0, kPcmuCodec, kH264Codec, &content4);
-    EXPECT_TRUE(channel1_->SetLocalContent(&content4, SdpType::kAnswer, err));
+    EXPECT_TRUE(channel1_->SetLocalContent(&content4, SdpType::kAnswer).ok());
     EXPECT_EQ(0u, media_send_channel1_impl()->send_streams().size());
 
-    EXPECT_TRUE(channel2_->SetRemoteContent(&content4, SdpType::kAnswer, err));
+    EXPECT_TRUE(channel2_->SetRemoteContent(&content4, SdpType::kAnswer).ok());
     EXPECT_EQ(0u, media_receive_channel2_impl()->recv_streams().size());
 
     SendCustomRtp2(kSsrc2, 0);
@@ -958,21 +948,23 @@ class ChannelTest : public ::testing::Test {
       EXPECT_FALSE(media_receive_channel1_impl()->playout());
     }
     EXPECT_FALSE(media_send_channel1_impl()->sending());
-    std::string err;
-    EXPECT_TRUE(channel1_->SetLocalContent(&local_media_content1_,
-                                           SdpType::kOffer, err));
+    EXPECT_TRUE(
+        channel1_->SetLocalContent(&local_media_content1_, SdpType::kOffer)
+            .ok());
     if (verify_playout_) {
       EXPECT_TRUE(media_receive_channel1_impl()->playout());
     }
     EXPECT_FALSE(media_send_channel1_impl()->sending());
-    EXPECT_TRUE(channel2_->SetRemoteContent(&local_media_content1_,
-                                            SdpType::kOffer, err));
+    EXPECT_TRUE(
+        channel2_->SetRemoteContent(&local_media_content1_, SdpType::kOffer)
+            .ok());
     if (verify_playout_) {
       EXPECT_FALSE(media_receive_channel2_impl()->playout());
     }
     EXPECT_FALSE(media_send_channel2_impl()->sending());
-    EXPECT_TRUE(channel2_->SetLocalContent(&local_media_content2_,
-                                           SdpType::kAnswer, err));
+    EXPECT_TRUE(
+        channel2_->SetLocalContent(&local_media_content2_, SdpType::kAnswer)
+            .ok());
     if (verify_playout_) {
       EXPECT_FALSE(media_receive_channel2_impl()->playout());
     }
@@ -992,8 +984,9 @@ class ChannelTest : public ::testing::Test {
       EXPECT_TRUE(media_receive_channel2_impl()->playout());
     }
     EXPECT_TRUE(media_send_channel2_impl()->sending());
-    EXPECT_TRUE(channel1_->SetRemoteContent(&local_media_content2_,
-                                            SdpType::kAnswer, err));
+    EXPECT_TRUE(
+        channel1_->SetRemoteContent(&local_media_content2_, SdpType::kAnswer)
+            .ok());
     if (verify_playout_) {
       EXPECT_TRUE(media_receive_channel1_impl()->playout());
     }
@@ -1023,12 +1016,11 @@ class ChannelTest : public ::testing::Test {
     }
     EXPECT_FALSE(media_send_channel2_impl()->sending());
 
-    std::string err;
-    EXPECT_TRUE(channel1_->SetLocalContent(&content1, SdpType::kOffer, err));
-    EXPECT_TRUE(channel2_->SetRemoteContent(&content1, SdpType::kOffer, err));
-    EXPECT_TRUE(channel2_->SetLocalContent(&content2, SdpType::kPrAnswer, err));
+    EXPECT_TRUE(channel1_->SetLocalContent(&content1, SdpType::kOffer).ok());
+    EXPECT_TRUE(channel2_->SetRemoteContent(&content1, SdpType::kOffer).ok());
+    EXPECT_TRUE(channel2_->SetLocalContent(&content2, SdpType::kPrAnswer).ok());
     EXPECT_TRUE(
-        channel1_->SetRemoteContent(&content2, SdpType::kPrAnswer, err));
+        channel1_->SetRemoteContent(&content2, SdpType::kPrAnswer).ok());
     ConnectFakeTransports();
 
     if (verify_playout_) {
@@ -1042,9 +1034,9 @@ class ChannelTest : public ::testing::Test {
 
     // Update `content2` to be RecvOnly.
     content2.set_direction(RtpTransceiverDirection::kRecvOnly);
-    EXPECT_TRUE(channel2_->SetLocalContent(&content2, SdpType::kPrAnswer, err));
+    EXPECT_TRUE(channel2_->SetLocalContent(&content2, SdpType::kPrAnswer).ok());
     EXPECT_TRUE(
-        channel1_->SetRemoteContent(&content2, SdpType::kPrAnswer, err));
+        channel1_->SetRemoteContent(&content2, SdpType::kPrAnswer).ok());
 
     if (verify_playout_) {
       EXPECT_TRUE(media_receive_channel1_impl()->playout());
@@ -1057,8 +1049,8 @@ class ChannelTest : public ::testing::Test {
 
     // Update `content2` to be SendRecv.
     content2.set_direction(RtpTransceiverDirection::kSendRecv);
-    EXPECT_TRUE(channel2_->SetLocalContent(&content2, SdpType::kAnswer, err));
-    EXPECT_TRUE(channel1_->SetRemoteContent(&content2, SdpType::kAnswer, err));
+    EXPECT_TRUE(channel2_->SetLocalContent(&content2, SdpType::kAnswer).ok());
+    EXPECT_TRUE(channel1_->SetRemoteContent(&content2, SdpType::kAnswer).ok());
 
     if (verify_playout_) {
       EXPECT_TRUE(media_receive_channel1_impl()->playout());
@@ -1072,11 +1064,11 @@ class ChannelTest : public ::testing::Test {
     // Update `content2` to be inactive on the receiver while sending at the
     // sender.
     content2.set_direction(RtpTransceiverDirection::kInactive);
-    EXPECT_TRUE(channel1_->SetLocalContent(&content1, SdpType::kOffer, err));
-    EXPECT_TRUE(channel2_->SetRemoteContent(&content1, SdpType::kOffer, err));
-    EXPECT_TRUE(channel2_->SetLocalContent(&content2, SdpType::kAnswer, err));
+    EXPECT_TRUE(channel1_->SetLocalContent(&content1, SdpType::kOffer).ok());
+    EXPECT_TRUE(channel2_->SetRemoteContent(&content1, SdpType::kOffer).ok());
+    EXPECT_TRUE(channel2_->SetLocalContent(&content2, SdpType::kAnswer).ok());
     content2.set_direction(RtpTransceiverDirection::kRecvOnly);
-    EXPECT_TRUE(channel1_->SetRemoteContent(&content2, SdpType::kAnswer, err));
+    EXPECT_TRUE(channel1_->SetRemoteContent(&content2, SdpType::kAnswer).ok());
     if (verify_playout_) {
       EXPECT_FALSE(media_receive_channel2_impl()->playout());
     }
@@ -1084,10 +1076,10 @@ class ChannelTest : public ::testing::Test {
 
     // Re-enable `content2`.
     content2.set_direction(RtpTransceiverDirection::kSendRecv);
-    EXPECT_TRUE(channel1_->SetLocalContent(&content1, SdpType::kOffer, err));
-    EXPECT_TRUE(channel2_->SetRemoteContent(&content1, SdpType::kOffer, err));
-    EXPECT_TRUE(channel2_->SetLocalContent(&content2, SdpType::kAnswer, err));
-    EXPECT_TRUE(channel1_->SetRemoteContent(&content2, SdpType::kAnswer, err));
+    EXPECT_TRUE(channel1_->SetLocalContent(&content1, SdpType::kOffer).ok());
+    EXPECT_TRUE(channel2_->SetRemoteContent(&content1, SdpType::kOffer).ok());
+    EXPECT_TRUE(channel2_->SetLocalContent(&content2, SdpType::kAnswer).ok());
+    EXPECT_TRUE(channel1_->SetRemoteContent(&content2, SdpType::kAnswer).ok());
     if (verify_playout_) {
       EXPECT_TRUE(media_receive_channel2_impl()->playout());
     }
@@ -1379,39 +1371,37 @@ class ChannelTest : public ::testing::Test {
   void TestSetContentFailure() {
     CreateChannels(0, 0);
 
-    std::string err;
     std::unique_ptr<typename T::Content> content(
         CreateMediaContentWithStream(1));
 
     media_receive_channel1_impl()->set_fail_set_recv_codecs(true);
     EXPECT_FALSE(
-        channel1_->SetLocalContent(content.get(), SdpType::kOffer, err));
+        channel1_->SetLocalContent(content.get(), SdpType::kOffer).ok());
     EXPECT_FALSE(
-        channel1_->SetLocalContent(content.get(), SdpType::kAnswer, err));
+        channel1_->SetLocalContent(content.get(), SdpType::kAnswer).ok());
 
     media_send_channel1_impl()->set_fail_set_send_codecs(true);
     EXPECT_FALSE(
-        channel1_->SetRemoteContent(content.get(), SdpType::kOffer, err));
+        channel1_->SetRemoteContent(content.get(), SdpType::kOffer).ok());
 
     media_send_channel1_impl()->set_fail_set_send_codecs(true);
     EXPECT_FALSE(
-        channel1_->SetRemoteContent(content.get(), SdpType::kAnswer, err));
+        channel1_->SetRemoteContent(content.get(), SdpType::kAnswer).ok());
   }
 
   void TestSendTwoOffers() {
     CreateChannels(0, 0);
 
-    std::string err;
     std::unique_ptr<typename T::Content> content1(
         CreateMediaContentWithStream(1));
     EXPECT_TRUE(
-        channel1_->SetLocalContent(content1.get(), SdpType::kOffer, err));
+        channel1_->SetLocalContent(content1.get(), SdpType::kOffer).ok());
     EXPECT_TRUE(media_send_channel1_impl()->HasSendStream(1));
 
     std::unique_ptr<typename T::Content> content2(
         CreateMediaContentWithStream(2));
     EXPECT_TRUE(
-        channel1_->SetLocalContent(content2.get(), SdpType::kOffer, err));
+        channel1_->SetLocalContent(content2.get(), SdpType::kOffer).ok());
     EXPECT_FALSE(media_send_channel1_impl()->HasSendStream(1));
     EXPECT_TRUE(media_send_channel1_impl()->HasSendStream(2));
   }
@@ -1419,17 +1409,16 @@ class ChannelTest : public ::testing::Test {
   void TestReceiveTwoOffers() {
     CreateChannels(0, 0);
 
-    std::string err;
     std::unique_ptr<typename T::Content> content1(
         CreateMediaContentWithStream(1));
     EXPECT_TRUE(
-        channel1_->SetRemoteContent(content1.get(), SdpType::kOffer, err));
+        channel1_->SetRemoteContent(content1.get(), SdpType::kOffer).ok());
     EXPECT_TRUE(media_receive_channel1_impl()->HasRecvStream(1));
 
     std::unique_ptr<typename T::Content> content2(
         CreateMediaContentWithStream(2));
     EXPECT_TRUE(
-        channel1_->SetRemoteContent(content2.get(), SdpType::kOffer, err));
+        channel1_->SetRemoteContent(content2.get(), SdpType::kOffer).ok());
     EXPECT_FALSE(media_receive_channel1_impl()->HasRecvStream(1));
     EXPECT_TRUE(media_receive_channel1_impl()->HasRecvStream(2));
   }
@@ -1437,19 +1426,18 @@ class ChannelTest : public ::testing::Test {
   void TestSendPrAnswer() {
     CreateChannels(0, 0);
 
-    std::string err;
     // Receive offer
     std::unique_ptr<typename T::Content> content1(
         CreateMediaContentWithStream(1));
     EXPECT_TRUE(
-        channel1_->SetRemoteContent(content1.get(), SdpType::kOffer, err));
+        channel1_->SetRemoteContent(content1.get(), SdpType::kOffer).ok());
     EXPECT_TRUE(media_receive_channel1_impl()->HasRecvStream(1));
 
     // Send PR answer
     std::unique_ptr<typename T::Content> content2(
         CreateMediaContentWithStream(2));
     EXPECT_TRUE(
-        channel1_->SetLocalContent(content2.get(), SdpType::kPrAnswer, err));
+        channel1_->SetLocalContent(content2.get(), SdpType::kPrAnswer).ok());
     EXPECT_TRUE(media_receive_channel1_impl()->HasRecvStream(1));
     EXPECT_TRUE(media_send_channel1_impl()->HasSendStream(2));
 
@@ -1457,7 +1445,7 @@ class ChannelTest : public ::testing::Test {
     std::unique_ptr<typename T::Content> content3(
         CreateMediaContentWithStream(3));
     EXPECT_TRUE(
-        channel1_->SetLocalContent(content3.get(), SdpType::kAnswer, err));
+        channel1_->SetLocalContent(content3.get(), SdpType::kAnswer).ok());
     EXPECT_TRUE(media_receive_channel1_impl()->HasRecvStream(1));
     EXPECT_FALSE(media_send_channel1_impl()->HasSendStream(2));
     EXPECT_TRUE(media_send_channel1_impl()->HasSendStream(3));
@@ -1466,19 +1454,18 @@ class ChannelTest : public ::testing::Test {
   void TestReceivePrAnswer() {
     CreateChannels(0, 0);
 
-    std::string err;
     // Send offer
     std::unique_ptr<typename T::Content> content1(
         CreateMediaContentWithStream(1));
     EXPECT_TRUE(
-        channel1_->SetLocalContent(content1.get(), SdpType::kOffer, err));
+        channel1_->SetLocalContent(content1.get(), SdpType::kOffer).ok());
     EXPECT_TRUE(media_send_channel1_impl()->HasSendStream(1));
 
     // Receive PR answer
     std::unique_ptr<typename T::Content> content2(
         CreateMediaContentWithStream(2));
     EXPECT_TRUE(
-        channel1_->SetRemoteContent(content2.get(), SdpType::kPrAnswer, err));
+        channel1_->SetRemoteContent(content2.get(), SdpType::kPrAnswer).ok());
     EXPECT_TRUE(media_send_channel1_impl()->HasSendStream(1));
     EXPECT_TRUE(media_receive_channel1_impl()->HasRecvStream(2));
 
@@ -1486,7 +1473,7 @@ class ChannelTest : public ::testing::Test {
     std::unique_ptr<typename T::Content> content3(
         CreateMediaContentWithStream(3));
     EXPECT_TRUE(
-        channel1_->SetRemoteContent(content3.get(), SdpType::kAnswer, err));
+        channel1_->SetRemoteContent(content3.get(), SdpType::kAnswer).ok());
     EXPECT_TRUE(media_send_channel1_impl()->HasSendStream(1));
     EXPECT_FALSE(media_receive_channel1_impl()->HasRecvStream(2));
     EXPECT_TRUE(media_receive_channel1_impl()->HasRecvStream(3));
@@ -1530,9 +1517,9 @@ class ChannelTest : public ::testing::Test {
 
   void DefaultMaxBitrateIsUnlimited() {
     CreateChannels(0, 0);
-    std::string err;
-    EXPECT_TRUE(channel1_->SetLocalContent(&local_media_content1_,
-                                           SdpType::kOffer, err));
+    EXPECT_TRUE(
+        channel1_->SetLocalContent(&local_media_content1_, SdpType::kOffer)
+            .ok());
     EXPECT_EQ(media_send_channel1_impl()->max_bps(), -1);
     VerifyMaxBitrate(media_send_channel1()->GetRtpSendParameters(kSsrc1),
                      std::nullopt);
@@ -1605,14 +1592,13 @@ class ChannelTest : public ::testing::Test {
     CreateChannels(0, 0);
     typename T::Content content1, content2, content3;
     CreateSimulcastContent({"f", "h", "q"}, &content1);
-    std::string err;
-    EXPECT_TRUE(channel1_->SetLocalContent(&content1, SdpType::kOffer, err));
+    EXPECT_TRUE(channel1_->SetLocalContent(&content1, SdpType::kOffer).ok());
     VerifySimulcastStreamParams(content1.streams()[0], channel1_.get());
     StreamParams stream1 = channel1_->local_streams()[0];
 
     // Create a similar offer. SetLocalContent should not remove and add.
     CreateSimulcastContent({"f", "h", "q"}, &content2);
-    EXPECT_TRUE(channel1_->SetLocalContent(&content2, SdpType::kOffer, err));
+    EXPECT_TRUE(channel1_->SetLocalContent(&content2, SdpType::kOffer).ok());
     VerifySimulcastStreamParams(content2.streams()[0], channel1_.get());
     StreamParams stream2 = channel1_->local_streams()[0];
     // Check that the streams are identical (SSRCs didn't change).
@@ -1620,7 +1606,7 @@ class ChannelTest : public ::testing::Test {
 
     // Create third offer that has same RIDs in different order.
     CreateSimulcastContent({"f", "q", "h"}, &content3);
-    EXPECT_TRUE(channel1_->SetLocalContent(&content3, SdpType::kOffer, err));
+    EXPECT_TRUE(channel1_->SetLocalContent(&content3, SdpType::kOffer).ok());
     VerifySimulcastStreamParams(content3.streams()[0], channel1_.get());
   }
 
@@ -2347,8 +2333,7 @@ TEST_F(VideoChannelSingleThreadTest, TestSetLocalOfferWithPacketization) {
 
   CreateChannels(0, 0);
 
-  std::string err;
-  EXPECT_TRUE(channel1_->SetLocalContent(&video, SdpType::kOffer, err));
+  EXPECT_TRUE(channel1_->SetLocalContent(&video, SdpType::kOffer).ok());
   EXPECT_THAT(media_send_channel1_impl()->send_codecs(), testing::IsEmpty());
   ASSERT_THAT(media_receive_channel1_impl()->recv_codecs(), testing::SizeIs(2));
   EXPECT_TRUE(
@@ -2370,9 +2355,7 @@ TEST_F(VideoChannelSingleThreadTest, TestSetRemoteOfferWithPacketization) {
 
   CreateChannels(0, 0);
 
-  std::string err;
-  EXPECT_TRUE(channel1_->SetRemoteContent(&video, SdpType::kOffer, err));
-  EXPECT_TRUE(err.empty());
+  EXPECT_TRUE(channel1_->SetRemoteContent(&video, SdpType::kOffer).ok());
   EXPECT_THAT(media_receive_channel1_impl()->recv_codecs(), testing::IsEmpty());
   ASSERT_THAT(media_send_channel1_impl()->send_codecs(), testing::SizeIs(2));
   EXPECT_TRUE(media_send_channel1_impl()->send_codecs()[0].Matches(kVp8Codec));
@@ -2392,11 +2375,8 @@ TEST_F(VideoChannelSingleThreadTest, TestSetAnswerWithPacketization) {
 
   CreateChannels(0, 0);
 
-  std::string err;
-  EXPECT_TRUE(channel1_->SetLocalContent(&video, SdpType::kOffer, err));
-  EXPECT_TRUE(err.empty());
-  EXPECT_TRUE(channel1_->SetRemoteContent(&video, SdpType::kAnswer, err));
-  EXPECT_TRUE(err.empty());
+  EXPECT_TRUE(channel1_->SetLocalContent(&video, SdpType::kOffer).ok());
+  EXPECT_TRUE(channel1_->SetRemoteContent(&video, SdpType::kAnswer).ok());
   ASSERT_THAT(media_receive_channel1_impl()->recv_codecs(), testing::SizeIs(2));
   EXPECT_TRUE(
       media_receive_channel1_impl()->recv_codecs()[0].Matches(kVp8Codec));
@@ -2426,9 +2406,8 @@ TEST_F(VideoChannelSingleThreadTest, TestSetLocalAnswerWithoutPacketization) {
 
   CreateChannels(0, 0);
 
-  std::string err;
-  EXPECT_TRUE(channel1_->SetRemoteContent(&remote_video, SdpType::kOffer, err));
-  EXPECT_TRUE(channel1_->SetLocalContent(&local_video, SdpType::kAnswer, err));
+  EXPECT_TRUE(channel1_->SetRemoteContent(&remote_video, SdpType::kOffer).ok());
+  EXPECT_TRUE(channel1_->SetLocalContent(&local_video, SdpType::kAnswer).ok());
   ASSERT_THAT(media_receive_channel1_impl()->recv_codecs(), testing::SizeIs(1));
   EXPECT_EQ(media_receive_channel1_impl()->recv_codecs()[0].packetization,
             std::nullopt);
@@ -2448,10 +2427,9 @@ TEST_F(VideoChannelSingleThreadTest, TestSetRemoteAnswerWithoutPacketization) {
 
   CreateChannels(0, 0);
 
-  std::string err;
-  EXPECT_TRUE(channel1_->SetLocalContent(&local_video, SdpType::kOffer, err));
+  EXPECT_TRUE(channel1_->SetLocalContent(&local_video, SdpType::kOffer).ok());
   EXPECT_TRUE(
-      channel1_->SetRemoteContent(&remote_video, SdpType::kAnswer, err));
+      channel1_->SetRemoteContent(&remote_video, SdpType::kAnswer).ok());
   ASSERT_THAT(media_receive_channel1_impl()->recv_codecs(), testing::SizeIs(1));
   EXPECT_EQ(media_receive_channel1_impl()->recv_codecs()[0].packetization,
             std::nullopt);
@@ -2473,12 +2451,9 @@ TEST_F(VideoChannelSingleThreadTest,
 
   CreateChannels(0, 0);
 
-  std::string err;
-  EXPECT_TRUE(channel1_->SetLocalContent(&local_video, SdpType::kOffer, err));
-  EXPECT_TRUE(err.empty());
+  EXPECT_TRUE(channel1_->SetLocalContent(&local_video, SdpType::kOffer).ok());
   EXPECT_FALSE(
-      channel1_->SetRemoteContent(&remote_video, SdpType::kAnswer, err));
-  EXPECT_FALSE(err.empty());
+      channel1_->SetRemoteContent(&remote_video, SdpType::kAnswer).ok());
   ASSERT_THAT(media_receive_channel1_impl()->recv_codecs(), testing::SizeIs(1));
   EXPECT_EQ(media_receive_channel1_impl()->recv_codecs()[0].packetization,
             webrtc::kPacketizationParamRaw);
@@ -2497,11 +2472,8 @@ TEST_F(VideoChannelSingleThreadTest,
 
   CreateChannels(0, 0);
 
-  std::string err;
-  EXPECT_TRUE(channel1_->SetRemoteContent(&remote_video, SdpType::kOffer, err));
-  EXPECT_TRUE(err.empty());
-  EXPECT_FALSE(channel1_->SetLocalContent(&local_video, SdpType::kAnswer, err));
-  EXPECT_FALSE(err.empty());
+  EXPECT_TRUE(channel1_->SetRemoteContent(&remote_video, SdpType::kOffer).ok());
+  EXPECT_FALSE(channel1_->SetLocalContent(&local_video, SdpType::kAnswer).ok());
   EXPECT_THAT(media_receive_channel1_impl()->recv_codecs(), testing::IsEmpty());
   ASSERT_THAT(media_send_channel1_impl()->send_codecs(), testing::SizeIs(1));
   EXPECT_EQ(media_send_channel1_impl()->send_codecs()[0].packetization,
@@ -2523,10 +2495,8 @@ TEST_F(VideoChannelSingleThreadTest,
   remote.set_codecs({vp8_foo, vp9});
 
   CreateChannels(0, 0);
-  std::string err;
-  ASSERT_TRUE(channel1_->SetLocalContent(&local, SdpType::kOffer, err)) << err;
-  ASSERT_TRUE(channel1_->SetRemoteContent(&remote, SdpType::kAnswer, err))
-      << err;
+  ASSERT_TRUE(channel1_->SetLocalContent(&local, SdpType::kOffer).ok());
+  ASSERT_TRUE(channel1_->SetRemoteContent(&remote, SdpType::kAnswer).ok());
 
   EXPECT_THAT(
       media_receive_channel1_impl()->recv_codecs(),
@@ -2559,10 +2529,8 @@ TEST_F(VideoChannelSingleThreadTest,
   remote.set_codecs({vp8_foo, vp8_bar, vp9_foo});
 
   CreateChannels(0, 0);
-  std::string err;
-  ASSERT_TRUE(channel1_->SetRemoteContent(&remote, SdpType::kOffer, err))
-      << err;
-  ASSERT_TRUE(channel1_->SetLocalContent(&local, SdpType::kAnswer, err)) << err;
+  ASSERT_TRUE(channel1_->SetRemoteContent(&remote, SdpType::kOffer).ok());
+  ASSERT_TRUE(channel1_->SetLocalContent(&local, SdpType::kAnswer).ok());
 
   EXPECT_THAT(
       media_receive_channel1_impl()->recv_codecs(),
@@ -2595,10 +2563,8 @@ TEST_F(VideoChannelSingleThreadTest,
   remote.set_codecs({vp8_raw, vp8});
 
   CreateChannels(0, 0);
-  std::string err;
-  ASSERT_TRUE(channel1_->SetLocalContent(&local, SdpType::kOffer, err)) << err;
-  ASSERT_TRUE(channel1_->SetRemoteContent(&remote, SdpType::kAnswer, err))
-      << err;
+  ASSERT_TRUE(channel1_->SetLocalContent(&local, SdpType::kOffer).ok());
+  ASSERT_TRUE(channel1_->SetRemoteContent(&remote, SdpType::kAnswer).ok());
 
   EXPECT_THAT(media_receive_channel1_impl()->recv_codecs(),
               UnorderedElementsAre(
@@ -2627,10 +2593,8 @@ TEST_F(VideoChannelSingleThreadTest,
   remote.set_codecs({vp8, vp8_raw});
 
   CreateChannels(0, 0);
-  std::string err;
-  ASSERT_TRUE(channel1_->SetRemoteContent(&remote, SdpType::kOffer, err))
-      << err;
-  ASSERT_TRUE(channel1_->SetLocalContent(&local, SdpType::kAnswer, err)) << err;
+  ASSERT_TRUE(channel1_->SetRemoteContent(&remote, SdpType::kOffer).ok());
+  ASSERT_TRUE(channel1_->SetLocalContent(&local, SdpType::kAnswer).ok());
 
   EXPECT_THAT(
       media_receive_channel1_impl()->recv_codecs(),
diff --git a/pc/rtp_transceiver.cc b/pc/rtp_transceiver.cc
index 63576448eb..010574895d 100644
--- a/pc/rtp_transceiver.cc
+++ b/pc/rtp_transceiver.cc
@@ -1299,33 +1299,31 @@ bool RtpTransceiver::SetChannelRtpTransport(
   return channel_->SetRtpTransport(rtp_transport);
 }
 
-bool RtpTransceiver::SetChannelLocalContent(
+RTCError RtpTransceiver::SetChannelLocalContent(
     const MediaContentDescription* content,
-    SdpType type,
-    std::string& error_desc) {
+    SdpType type) {
   RTC_DCHECK_RUN_ON(context()->signaling_thread());
   return SetChannelContent([&]() {
     RTC_DCHECK_RUN_ON(context()->worker_thread());
-    return channel_->SetLocalContent(content, type, error_desc);
+    return channel_->SetLocalContent(content, type);
   });
 }
 
-bool RtpTransceiver::SetChannelRemoteContent(
+RTCError RtpTransceiver::SetChannelRemoteContent(
     const MediaContentDescription* content,
-    SdpType type,
-    std::string& error_desc) {
+    SdpType type) {
   RTC_DCHECK_RUN_ON(context()->signaling_thread());
   return SetChannelContent([&]() {
     RTC_DCHECK_RUN_ON(context()->worker_thread());
-    return channel_->SetRemoteContent(content, type, error_desc);
+    return channel_->SetRemoteContent(content, type);
   });
 }
 
-bool RtpTransceiver::SetChannelContent(
-    absl::AnyInvocable<bool() &&> set_content) {
+RTCError RtpTransceiver::SetChannelContent(
+    absl::AnyInvocable<RTCError() &&> set_content) {
   RTC_DCHECK_RUN_ON(context()->signaling_thread());
   if (!channel_) {
-    return false;
+    return RTCError::InvalidState() << "No channel";
   }
 
   struct SenderParameters {
@@ -1343,9 +1341,10 @@ bool RtpTransceiver::SetChannelContent(
 
   // Calls the callback on the worker thread, fetches and returns the
   // RtpParameters for the senders.
-  bool result = context()->worker_thread()->BlockingCall([&]() {
-    if (!std::move(set_content)()) {
-      return false;
+  RTCError result = context()->worker_thread()->BlockingCall([&]() {
+    RTCError error = std::move(set_content)();
+    if (!error.ok()) {
+      return error;
     }
     for (auto& entry : sender_parameters) {
       if (entry.ssrc != 0) {
@@ -1353,7 +1352,7 @@ bool RtpTransceiver::SetChannelContent(
             channel_->media_send_channel()->GetRtpSendParameters(entry.ssrc);
       }
     }
-    return true;
+    return RTCError::OK();
   });
 
   for (auto& entry : sender_parameters) {
diff --git a/pc/rtp_transceiver.h b/pc/rtp_transceiver.h
index 81e1855c12..730978b9d2 100644
--- a/pc/rtp_transceiver.h
+++ b/pc/rtp_transceiver.h
@@ -377,12 +377,10 @@ class RtpTransceiver : public RtpTransceiverInterface {
   }
 
   bool SetChannelRtpTransport(RtpTransportInternal* rtp_transport);
-  bool SetChannelLocalContent(const MediaContentDescription* content,
-                              SdpType type,
-                              std::string& error_desc);
-  bool SetChannelRemoteContent(const MediaContentDescription* content,
-                               SdpType type,
-                               std::string& error_desc);
+  RTCError SetChannelLocalContent(const MediaContentDescription* content,
+                                  SdpType type);
+  RTCError SetChannelRemoteContent(const MediaContentDescription* content,
+                                   SdpType type);
   bool SetChannelPayloadTypeDemuxingEnabled(bool enabled);
   void EnableChannel(bool enable);
   const std::vector<StreamParams>& channel_local_streams() const;
@@ -433,7 +431,7 @@ class RtpTransceiver : public RtpTransceiverInterface {
   GetOfferedAndImplementedHeaderExtensions(
       const MediaContentDescription* content) const;
 
-  bool SetChannelContent(absl::AnyInvocable<bool() &&> set_content);
+  RTCError SetChannelContent(absl::AnyInvocable<RTCError() &&> set_content);
 
   const Environment env_;
   // Enforce that this object is created, used and destroyed on one thread.
diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc
index cb2ed5c124..a182232dfd 100644
--- a/pc/sdp_offer_answer.cc
+++ b/pc/sdp_offer_answer.cc
@@ -5233,13 +5233,12 @@ RTCError SdpOfferAnswerHandler::PushdownMediaDescription(
     // - crbug.com/1157227
     // - crbug.com/1187289
     for (const auto& [transceiver, content] : channels) {
-      std::string error;
-      bool success =
+      RTCError error =
           (source == CS_LOCAL)
-              ? transceiver->SetChannelLocalContent(content, type, error)
-              : transceiver->SetChannelRemoteContent(content, type, error);
-      if (!success) {
-        return LOG_ERROR(RTCError::InvalidParameter() << error);
+              ? transceiver->SetChannelLocalContent(content, type)
+              : transceiver->SetChannelRemoteContent(content, type);
+      if (!error.ok()) {
+        return error;
       }
     }
     // If local and remote are both set, we assume that it's safe to trigger
diff --git a/pc/test/mock_channel_interface.h b/pc/test/mock_channel_interface.h
index 1dc4ddd57e..b87ab9dc6a 100644
--- a/pc/test/mock_channel_interface.h
+++ b/pc/test/mock_channel_interface.h
@@ -18,6 +18,7 @@
 #include "absl/strings/string_view.h"
 #include "api/jsep.h"
 #include "api/media_types.h"
+#include "api/rtc_error.h"
 #include "media/base/media_channel.h"
 #include "media/base/stream_params.h"
 #include "pc/channel_interface.h"
@@ -72,13 +73,13 @@ class MockChannelInterface : public ChannelInterface {
               SetPacketReceivedCallback_n,
               (absl::AnyInvocable<void(const RtpPacketReceived&)>),
               (override));
-  MOCK_METHOD(bool,
+  MOCK_METHOD(RTCError,
               SetLocalContent,
-              (const webrtc::MediaContentDescription*, SdpType, std::string&),
+              (const webrtc::MediaContentDescription*, SdpType),
               (override));
-  MOCK_METHOD(bool,
+  MOCK_METHOD(RTCError,
               SetRemoteContent,
-              (const webrtc::MediaContentDescription*, SdpType, std::string&),
+              (const webrtc::MediaContentDescription*, SdpType),
               (override));
   MOCK_METHOD(bool, SetPayloadTypeDemuxingEnabled, (bool), (override));
   MOCK_METHOD(const std::vector<StreamParams>&,

From bed5b7ecd86ae5434dc8d66475229fa8c98ca999 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Mon, 23 Mar 2026 09:04:33 -0700
Subject: [PATCH 379/472] Roll chromium_revision d2fdfe540d..edf3774d66
 (1602801:1603529)

Change log: https://chromium.googlesource.com/chromium/src/+log/d2fdfe540d..edf3774d66
Full diff: https://chromium.googlesource.com/chromium/src/+/d2fdfe540d..edf3774d66

Changed dependencies
* siso_version: git_revision:c7dcf419b19a4a7377fa8e3e3f5fad5dffe10cf2..git_revision:87bad442ede1c60700dfabef5862c4a584621734
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/376dfbb702..613f1bc9c3
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/88f8168044..4f65f5693c
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/104338fa9d..67f72857b9
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/fad81ef2da..47f8a64d10
* src/third_party/android_deps/autorolled/cipd: pLc9v3YoCPpc5bZ-yPGyIg0hLqZeXaIpzEHHWOI09UsC..2mhz9b8511ym0Z01dgIFbJPgHyotC0CmmWp1piMvNcoC
* src/third_party/androidx/cipd: 2cwc39P1Nuw60SuLSgzP-sHbXOgoHhlhS10ck26jV3QC..ShH1Nbh7Gg9-IGjDdLKIPxd1x9hLFemKhXA03nWzTQYC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/ba3de4d9b4..8f1f636048
* src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/84bfdcd752..0a638b1a5e
* src/third_party/dav1d/libdav1d: https://chromium.googlesource.com/external/github.com/videolan/dav1d.git/+log/241a6b236a..d69235dd80
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/315fe92361..145eee0201
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/821ab7d3e6..f574cb9b18
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/2c62d7b7a9..90309e4a3a
* src/third_party/libaom/source/libaom: https://aomedia.googlesource.com/aom.git/+log/34f25197c6..f3dddebddd
* src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/9b08b4a339..a20a87a7eb
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/b041e45176..ac6282fcb2
* src/third_party/siso/cipd: git_revision:c7dcf419b19a4a7377fa8e3e3f5fad5dffe10cf2..git_revision:87bad442ede1c60700dfabef5862c4a584621734
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/cab35c1daf..c87604d604
DEPS diff: https://chromium.googlesource.com/chromium/src/+/d2fdfe540d..edf3774d66/DEPS

BUG=None

Change-Id: Icb66a07868c076527e006ea2c04bf8582cba57d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/459420
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47232}
---
 DEPS | 36 ++++++++++++++++++------------------
 1 file changed, 18 insertions(+), 18 deletions(-)

diff --git a/DEPS b/DEPS
index b60f0c973a..79438af869 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': 'd2fdfe540d234357ea185e14b9721739339ea9a7',
+  'chromium_revision': 'edf3774d66e0faa0c2b633c91b828c3f5d08c5e5',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -50,7 +50,7 @@ vars = {
   # reclient CIPD package version
   'reclient_version': 're_client_version:0.185.0.db415f21-gomaip',
   # siso CIPD package version.
-  'siso_version': 'git_revision:c7dcf419b19a4a7377fa8e3e3f5fad5dffe10cf2',
+  'siso_version': 'git_revision:87bad442ede1c60700dfabef5862c4a584621734',
 
   # ninja CIPD package.
   'ninja_package': 'infra/3pp/tools/ninja/',
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@376dfbb70294202ba3f248d459be95c26c766423',
+    'https://chromium.googlesource.com/chromium/src/build@613f1bc9c3886611ca9354bd3e409082277e8919',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@95ed44cf5f06dbb5861030b91c9db9ccb4316762',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@88f8168044098901642bae31fb008a955a79ea2c',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@4f65f5693caf885db847197e4e7668c83c01c0ed',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@104338fa9d2b5ee90064a595b537b7403c29c74f',
+    'https://chromium.googlesource.com/chromium/src/testing@67f72857b9ed52fc0fa5b5141475bf1d117f4813',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@fad81ef2daf5a05d454f80d2db57981f720471ae',
+    'https://chromium.googlesource.com/chromium/src/third_party@47f8a64d10b73ed3dd6b9e3e5682ae4e463a1e20',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -331,13 +331,13 @@ deps = {
   'src/third_party/clang-format/script':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@c2725e0622e1a86d55f14514f2177a39efea4a0e',
   'src/third_party/compiler-rt/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@84bfdcd752cb00ccc8087aeb584d3c08e9da96ba',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@0a638b1a5e41834d0b438b925e53c70d3108f34a',
   'src/third_party/libc++/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@7ab65651aed6802d2599dcb7a73b1f82d5179d05',
   'src/third_party/libc++abi/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@8f11bb1d4438d0239d0dfc1bd9456a9f31629dda',
   'src/third_party/llvm-libc/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@9b08b4a339fce4aaa79c30ec303eb8edeb198290',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@a20a87a7eb92e2698ac1b0b7bede4c13ccea90a3',
   'src/third_party/libunwind/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@db838d918570d4e381ecf9f5cc70a0098c9c2cd6',
 
@@ -488,7 +488,7 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@ba3de4d9b41906eb7e50d69edca140ef3cec053a',
+    'https://boringssl.googlesource.com/boringssl.git@8f1f6360480481a6950bdd702061e68ecd7d0a79',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@8be0e3114685fcc1589561067282edf75ea1259a',
   'src/third_party/catapult':
@@ -505,13 +505,13 @@ deps = {
   'src/third_party/crc32c/src':
     'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558',
   'src/third_party/depot_tools':
-    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@315fe9236154fbe4124aeed270c9f5b96a060666',
+    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@145eee02011c02fc25d2ec055b55451661f3a611',
   'src/third_party/ffmpeg':
     'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@946d97db8d906277085e361892b7efda5152e2f1',
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@2c62d7b7a927db2572b50277ba56eb4a754c43b7',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@90309e4a3a8250eaf505ba3194bbdbf6be5ea12c',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -599,9 +599,9 @@ deps = {
   'src/third_party/libsrtp':
     'https://chromium.googlesource.com/chromium/deps/libsrtp.git@e8383771af8aa4096f5bcfe3743a5ea128f88a9a',
   'src/third_party/dav1d/libdav1d':
-    'https://chromium.googlesource.com/external/github.com/videolan/dav1d.git@241a6b236a9c84441c872119c1f31b4074c0589d',
+    'https://chromium.googlesource.com/external/github.com/videolan/dav1d.git@d69235dd804b24c04ed05639cffcc912cd6cfd75',
   'src/third_party/libaom/source/libaom':
-    'https://aomedia.googlesource.com/aom.git@34f25197c6e9efef129992bb763aefa6036119fd',
+    'https://aomedia.googlesource.com/aom.git@f3dddebddd0dba76fbfb97b96b6336bcf1d3a30c',
   'src/third_party/libgav1/src':
     Var('chromium_git') + '/codecs/libgav1.git' + '@' + '40f58ed32ff39071c3f2a51056dbc49a070af0dc',
   'src/third_party/libunwindstack': {
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'b041e451767026d64f688b3767938c0222652b31',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'ac6282fcb29d43dfdaeeb808f2c7746ed15a8b9b',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@cab35c1daff5c995890360f759455b6725af5856',
+    'https://chromium.googlesource.com/chromium/src/tools@c87604d604f347955ad7c07f7f21315a5c64f49e',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': '2cwc39P1Nuw60SuLSgzP-sHbXOgoHhlhS10ck26jV3QC',
+          'version': 'ShH1Nbh7Gg9-IGjDdLKIPxd1x9hLFemKhXA03nWzTQYC',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -842,7 +842,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/android_deps/autorolled',
-              'version': 'pLc9v3YoCPpc5bZ-yPGyIg0hLqZeXaIpzEHHWOI09UsC',
+              'version': '2mhz9b8511ym0Z01dgIFbJPgHyotC0CmmWp1piMvNcoC',
           },
       ],
       'condition': 'checkout_android and non_git_source',
@@ -864,7 +864,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7607ca500436b37ad23fb8d18614bec7796b68a7',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '821ab7d3e656e6928d342e289efaf208017d4274',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + 'f574cb9b18a3e9ddf09ec80a3afb0270649d61a5',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From 82882c3084a454f3b340366d3b49e1da7c55f01c Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Mon, 23 Mar 2026 13:03:41 -0700
Subject: [PATCH 380/472] Roll chromium_revision edf3774d66..c5265f5aea
 (1603529:1603661)

Change log: https://chromium.googlesource.com/chromium/src/+log/edf3774d66..c5265f5aea
Full diff: https://chromium.googlesource.com/chromium/src/+/edf3774d66..c5265f5aea

Changed dependencies
* src/buildtools/linux64: git_revision:3007f57168dbe495a6a0a206c4ac54aca9a0d56f..git_revision:c91f7e86727683118b6096757717135463fdcdd7
* src/buildtools/mac: git_revision:3007f57168dbe495a6a0a206c4ac54aca9a0d56f..git_revision:c91f7e86727683118b6096757717135463fdcdd7
* src/buildtools/win: git_revision:3007f57168dbe495a6a0a206c4ac54aca9a0d56f..git_revision:c91f7e86727683118b6096757717135463fdcdd7
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/4f65f5693c..6ad000d2a1
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/67f72857b9..4baf362be5
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/47f8a64d10..9b87b60a47
* src/third_party/androidx/cipd: ShH1Nbh7Gg9-IGjDdLKIPxd1x9hLFemKhXA03nWzTQYC..auHen2c3yuyBQK158jQrEvU0JD_AtKDaBUqflOWWZcAC
* src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/0a638b1a5e..28c233d2e1
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/ac6282fcb2..bba888dab9
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/c87604d604..01e46c13a0
DEPS diff: https://chromium.googlesource.com/chromium/src/+/edf3774d66..c5265f5aea/DEPS

BUG=None

Change-Id: I27c6bb6076f93e58e19c5e8bb0316d6345708466
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/459500
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47233}
---
 DEPS | 22 +++++++++++-----------
 1 file changed, 11 insertions(+), 11 deletions(-)

diff --git a/DEPS b/DEPS
index 79438af869..328db20274 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': 'edf3774d66e0faa0c2b633c91b828c3f5d08c5e5',
+  'chromium_revision': 'c5265f5aea98860f032daae3336f857818358611',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -78,19 +78,19 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@4f65f5693caf885db847197e4e7668c83c01c0ed',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@6ad000d2a105276d29e79d92480c0ba9f13a6cea',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@67f72857b9ed52fc0fa5b5141475bf1d117f4813',
+    'https://chromium.googlesource.com/chromium/src/testing@4baf362be5c8cc881fb1ff3aa1cc09c5d18a9634',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@47f8a64d10b73ed3dd6b9e3e5682ae4e463a1e20',
+    'https://chromium.googlesource.com/chromium/src/third_party@9b87b60a47de2b4a53b858a1a5332a6b6033da5b',
 
   'src/buildtools/linux64': {
     'packages': [
       {
         'package': 'gn/gn/linux-${{arch}}',
-        'version': 'git_revision:3007f57168dbe495a6a0a206c4ac54aca9a0d56f',
+        'version': 'git_revision:c91f7e86727683118b6096757717135463fdcdd7',
       }
     ],
     'dep_type': 'cipd',
@@ -100,7 +100,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/mac-${{arch}}',
-        'version': 'git_revision:3007f57168dbe495a6a0a206c4ac54aca9a0d56f',
+        'version': 'git_revision:c91f7e86727683118b6096757717135463fdcdd7',
       }
     ],
     'dep_type': 'cipd',
@@ -110,7 +110,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/windows-amd64',
-        'version': 'git_revision:3007f57168dbe495a6a0a206c4ac54aca9a0d56f',
+        'version': 'git_revision:c91f7e86727683118b6096757717135463fdcdd7',
       }
     ],
     'dep_type': 'cipd',
@@ -331,7 +331,7 @@ deps = {
   'src/third_party/clang-format/script':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@c2725e0622e1a86d55f14514f2177a39efea4a0e',
   'src/third_party/compiler-rt/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@0a638b1a5e41834d0b438b925e53c70d3108f34a',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@28c233d2e119c0a260083be9247c85ccd981b18d',
   'src/third_party/libc++/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@7ab65651aed6802d2599dcb7a73b1f82d5179d05',
   'src/third_party/libc++abi/src':
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'ac6282fcb29d43dfdaeeb808f2c7746ed15a8b9b',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'bba888dab987c31f0440974c4972a3cebb87a244',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@c87604d604f347955ad7c07f7f21315a5c64f49e',
+    'https://chromium.googlesource.com/chromium/src/tools@01e46c13a01b9336d5b64ea5ff75b658d8da2d81',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'ShH1Nbh7Gg9-IGjDdLKIPxd1x9hLFemKhXA03nWzTQYC',
+          'version': 'auHen2c3yuyBQK158jQrEvU0JD_AtKDaBUqflOWWZcAC',
       },
     ],
     'condition': 'checkout_android and non_git_source',

From 712758587561f5711bbf4bb14e3032706771b360 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Mon, 23 Mar 2026 17:02:31 -0700
Subject: [PATCH 381/472] Roll chromium_revision c5265f5aea..db815f58ca
 (1603661:1603797)

Change log: https://chromium.googlesource.com/chromium/src/+log/c5265f5aea..db815f58ca
Full diff: https://chromium.googlesource.com/chromium/src/+/c5265f5aea..db815f58ca

Changed dependencies
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/6ad000d2a1..a60191e77d
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/4baf362be5..8ab6de03be
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/9b87b60a47..c2eae4f8d4
* src/third_party/fuzztest/src: https://chromium.googlesource.com/external/github.com/google/fuzztest.git/+log/9600119eaa..3f730151ee
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/90309e4a3a..2fb50818a9
* src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/a20a87a7eb..65d4df5c8e
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/bba888dab9..5d32182175
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/01e46c13a0..3366ed5009
DEPS diff: https://chromium.googlesource.com/chromium/src/+/c5265f5aea..db815f58ca/DEPS

BUG=None

Change-Id: I27746416cee725b7cc2fadd2a44d5091d0071084
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/459301
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47234}
---
 DEPS | 18 +++++++++---------
 1 file changed, 9 insertions(+), 9 deletions(-)

diff --git a/DEPS b/DEPS
index 328db20274..bded363163 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': 'c5265f5aea98860f032daae3336f857818358611',
+  'chromium_revision': 'db815f58caa934760fe8f3b900172f25df7fb0d4',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@6ad000d2a105276d29e79d92480c0ba9f13a6cea',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@a60191e77d7a91464114403f5d854262b134c314',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@4baf362be5c8cc881fb1ff3aa1cc09c5d18a9634',
+    'https://chromium.googlesource.com/chromium/src/testing@8ab6de03bef68504b2738c028c4671ecb33cc5ef',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@9b87b60a47de2b4a53b858a1a5332a6b6033da5b',
+    'https://chromium.googlesource.com/chromium/src/third_party@c2eae4f8d41e653b099718f6a146a97578069c76',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -337,7 +337,7 @@ deps = {
   'src/third_party/libc++abi/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@8f11bb1d4438d0239d0dfc1bd9456a9f31629dda',
   'src/third_party/llvm-libc/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@a20a87a7eb92e2698ac1b0b7bede4c13ccea90a3',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@65d4df5c8e930138f416611ae540b5fd9fe3a6c6',
   'src/third_party/libunwind/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@db838d918570d4e381ecf9f5cc70a0098c9c2cd6',
 
@@ -511,7 +511,7 @@ deps = {
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@90309e4a3a8250eaf505ba3194bbdbf6be5ea12c',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@2fb50818a9b4742fda5ae1b894dca8683dd3f422',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -593,7 +593,7 @@ deps = {
   'src/third_party/libFuzzer/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt/lib/fuzzer.git@bea408a6e01f0f7e6c82a43121fe3af4506c932e',
   'src/third_party/fuzztest/src':
-    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@9600119eaa4684586fef4d0144c0446470f64d87',
+    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@3f730151eede388a825d60bcdd0bc1540bcc2b90',
   'src/third_party/libjpeg_turbo':
     'https://chromium.googlesource.com/chromium/deps/libjpeg_turbo.git@d1f5f2393e0d51f840207342ae86e55a86443288',
   'src/third_party/libsrtp':
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'bba888dab987c31f0440974c4972a3cebb87a244',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '5d32182175cc1063c9c2e5af25a94e7f3ea3f22c',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@01e46c13a01b9336d5b64ea5ff75b658d8da2d81',
+    'https://chromium.googlesource.com/chromium/src/tools@3366ed5009a56d79bb7af74577acbf6c42e6c7ae',
 
   'src/third_party/espresso': {
       'packages': [

From 0046d720f71776ec05a06ce5b018b7329353b1cf Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Mon, 23 Mar 2026 21:06:54 -0700
Subject: [PATCH 382/472] Update WebRTC code version (2026-03-24T04:06:53).

Bug: None
Change-Id: I39c0ed78d6bc5731ed0cbbdbccea829b7a8e1b94
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/459560
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47235}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index cca624b266..c911c6f69e 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-22T04:07:46";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-24T04:06:53";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From c834307dfd27d63425738753f67a1e45237319c8 Mon Sep 17 00:00:00 2001
From: Per K <perkj@webrtc.org>
Date: Mon, 23 Mar 2026 16:10:36 +0000
Subject: [PATCH 383/472] Ignore BWE expectations in ScreamTest

ScreamTests are despite running in simulated time not deterministic enough on all platforms and have caused flakiness on try bots.
Therefore, ignore BWE expectations for now unless the flag  --enable-scream-bwe-expectations is set.



Bug: webrtc:447037083
Change-Id: I5a240806603e37952f08f7e900877baa6a6ff3e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/459440
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47236}
---
 .../bwe_integration_tests/BUILD.gn            |  1 +
 .../bwe_integration_tests/scream_test.cc      | 56 +++++++++++++------
 2 files changed, 41 insertions(+), 16 deletions(-)

diff --git a/test/peer_scenario/bwe_integration_tests/BUILD.gn b/test/peer_scenario/bwe_integration_tests/BUILD.gn
index 729c2b0e5a..763487ae26 100644
--- a/test/peer_scenario/bwe_integration_tests/BUILD.gn
+++ b/test/peer_scenario/bwe_integration_tests/BUILD.gn
@@ -70,6 +70,7 @@ if (rtc_include_tests) {
         "../../../rtc_base:network_constants",
         "../../network:emulated_network",
         "../../scenario",
+        "//third_party/abseil-cpp/absl/flags:flag",
         "//third_party/abseil-cpp/absl/strings",
       ]
     }
diff --git a/test/peer_scenario/bwe_integration_tests/scream_test.cc b/test/peer_scenario/bwe_integration_tests/scream_test.cc
index ef5f84c1de..538b91a181 100644
--- a/test/peer_scenario/bwe_integration_tests/scream_test.cc
+++ b/test/peer_scenario/bwe_integration_tests/scream_test.cc
@@ -16,6 +16,7 @@
 #include <utility>
 #include <vector>
 
+#include "absl/flags/flag.h"
 #include "api/audio_options.h"
 #include "api/rtp_parameters.h"
 #include "api/scoped_refptr.h"
@@ -31,6 +32,7 @@
 #include "api/units/data_size.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
+#include "rtc_base/logging.h"
 #include "test/create_frame_generator_capturer.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
@@ -39,6 +41,21 @@
 #include "test/peer_scenario/peer_scenario.h"
 #include "test/peer_scenario/peer_scenario_client.h"
 
+// Tests for SCReAM.
+// Per default, the tests currently does not set expectations on BWE. The
+// reason is due to that the tests are not deterministic enough on all
+// platforms.
+// TODO: bugs.webrtc.org/447037083 - Attempt to make tests stable enough for
+// the none deterministic behaviour to not matter.
+// To enable BWE expectations, set the flag --enable-scream-bwe-expectations.
+// To write RTC eventlogs to file, set the flag --peer_logs=true and
+// --peer_logs_root=<path>.
+
+ABSL_FLAG(bool,
+          enable_scream_bwe_expectations,
+          false,
+          "Enable BWE expectations in Scream tests.");
+
 namespace webrtc {
 namespace {
 
@@ -76,8 +93,15 @@ MATCHER_P2(AvailableSendBitrateIsBetween, low, high, "") {
   *result_listener << "the available send bitrate is " << available_bwe.kbps()
                    << "kbps, which is not between " << low.kbps() << "kbps and "
                    << high.kbps() << " kbps";
-
-  return false;
+  if (absl::GetFlag(FLAGS_enable_scream_bwe_expectations)) {
+    // TODO: bugs.webrtc.org/447037083 - Attempt to make tests stable enough for
+    // the none deterministic behaviour to not matter.
+    return false;
+  }
+  RTC_LOG(LS_ERROR) << "!!! the available send bitrate is "
+                    << available_bwe.kbps() << "kbps, which is not between "
+                    << low.kbps() << "kbps and " << high.kbps() << " kbps";
+  return true;
 }
 
 MATCHER_P2(CurrentRoundTripTimeIsBetween, low, high, "") {
@@ -398,9 +422,6 @@ TEST(ScreamTest, MaybeTest(LinkCapacity600KbpsRtt100msEcnAfterCe)) {
       {"WebRTC-Bwe-ScreamV2", "mode:only_after_ce"}};
 
   SendMediaTestResult result = SendMediaInOneDirection(std::move(params), s);
-  EXPECT_THAT(result.caller_stats.back(),
-              AvailableSendBitrateIsBetween(DataRate::KilobitsPerSec(350),
-                                            DataRate::KilobitsPerSec(660)));
 
   // All packets are sent as ECT1.
   EXPECT_EQ(GetPacketsSent(result.caller_stats.back()),
@@ -408,6 +429,9 @@ TEST(ScreamTest, MaybeTest(LinkCapacity600KbpsRtt100msEcnAfterCe)) {
   // Not all packets has been received yet.
   EXPECT_GE(GetPacketsSentWithEct1(result.caller_stats.back()),
             0.9 * (GetPacketsReceived(result.callee_stats.back())));
+  EXPECT_THAT(result.caller_stats.back(),
+              AvailableSendBitrateIsBetween(DataRate::KilobitsPerSec(350),
+                                            DataRate::KilobitsPerSec(660)));
 }
 
 // Test that we can switch from Goog CC sending ECT1 to send ECT 0 and adapt.
@@ -499,7 +523,6 @@ TEST(ScreamTest, MaybeTest(LinkCapacity2MbpsRtt50msEcn)) {
                         DataRate::KilobitsPerSec(2000), TimeDelta::Millis(25));
 
   SendMediaTestResult result = SendMediaInOneDirection(std::move(params), s);
-
   EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
                                               DataRate::KilobitsPerSec(1300),
                                               DataRate::KilobitsPerSec(2300))));
@@ -680,7 +703,6 @@ TEST(ScreamTest, MaybeTest(LinkCapacity5MbitRepeatedDelaySpikesNoEcn)) {
       s, DataRate::KilobitsPerSec(5000), TimeDelta::Millis(100),
       TimeDelta::Millis(200));
   SendMediaTestResult result = SendMediaInOneDirection(std::move(params), s);
-
   EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
                                               DataRate::KilobitsPerSec(1200),
                                               DataRate::KilobitsPerSec(5000))));
@@ -697,7 +719,6 @@ TEST(ScreamTest, MaybeTest(RampupFastOnLinkCapacity50Mbit20MsRttNoEcn)) {
       CreateNetworkPath(s, /*use_dual_pi= */ false,
                         DataRate::KilobitsPerSec(50000), TimeDelta::Millis(10));
   SendMediaTestResult result = SendMediaInOneDirection(std::move(params), s);
-
   // After 400ms.
   EXPECT_THAT(result.caller_stats[3],
               AvailableSendBitrateIsBetween(DataRate::KilobitsPerSec(1200),
@@ -730,14 +751,14 @@ TEST(ScreamTest, MaybeTest(LinkCapacity1MbitRtt50msWithShortQueuesNoEcn)) {
       network_builder.packet_queue_length(3), /*use_dual_pi= */ false);
 
   SendMediaTestResult result = SendMediaInOneDirection(std::move(params), s);
-
-  EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
-                                              DataRate::KilobitsPerSec(200),
-                                              DataRate::KilobitsPerSec(1100))));
   EXPECT_THAT(
       GetPacketsLost(result.callee_stats.back()) /
           static_cast<double>(GetPacketsSent(result.caller_stats.back())),
       Lt(0.05));
+
+  EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
+                                              DataRate::KilobitsPerSec(200),
+                                              DataRate::KilobitsPerSec(1100))));
 }
 
 TEST(ScreamTest,
@@ -755,6 +776,7 @@ TEST(ScreamTest,
 
   ASSERT_GE(GetPacketsLost(result.callee_stats.back()),
             0.05 * GetPacketsSent(result.caller_stats.back()));
+
   // Ignore estimate during rampup.
   EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
                                               DataRate::KilobitsPerSec(100),
@@ -779,6 +801,7 @@ TEST(ScreamTest, MaybeTest(ReturnLinkWithBurstLoss)) {
   // feedback is eventually received even if feedback packets are lost.
   EXPECT_GT(GetPacketsSent(result.caller_stats.back()),
             GetPacketsSent(result.caller_stats[5]));
+
   EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
                                               DataRate::KilobitsPerSec(300),
                                               DataRate::KilobitsPerSec(1300))));
@@ -804,6 +827,7 @@ TEST(ScreamTest, MaybeTest(SendVideoOnlyReturnLinkWithBurstLoss)) {
   // packets to fast if queued too long, BWE drop to a very low value.
   EXPECT_GT(GetPacketsSent(result.caller_stats.back()),
             GetPacketsSent(result.caller_stats[5]));
+
   EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
                                               DataRate::KilobitsPerSec(10),
                                               DataRate::KilobitsPerSec(1200))));
@@ -826,14 +850,14 @@ TEST(ScreamTest, MaybeTest(LinkCapacity5MbitPolicedTo256Kbit)) {
       CreateNetworkPath(network_builder, /*use_dual_pi= */ false);
 
   SendMediaTestResult result = SendMediaInOneDirection(std::move(params), s);
-
-  EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
-                                              DataRate::KilobitsPerSec(150),
-                                              DataRate::KilobitsPerSec(900))));
   EXPECT_THAT(
       GetPacketsLost(result.callee_stats.back()) /
           static_cast<double>(GetPacketsSent(result.caller_stats.back())),
       AllOf(Lt(0.08), Gt(0.01)));
+
+  EXPECT_THAT(result.caller().subspan(1), Each(AvailableSendBitrateIsBetween(
+                                              DataRate::KilobitsPerSec(150),
+                                              DataRate::KilobitsPerSec(900))));
 }
 
 TEST(ScreamTest, MaybeTest(LinkCapacity5MbitWithCrossTrafficNoEcn)) {

From 336f5fba8bc1e2d99988552d225e767b7cac5d5d Mon Sep 17 00:00:00 2001
From: Philipp Hancke <philipp.hancke@googlemail.com>
Date: Mon, 23 Mar 2026 08:04:12 +0100
Subject: [PATCH 384/472] snap: assert behavior when sctp-init changes due to
 pranswer

Also changes the SctpTransport::Start to a blocking call which allows
SetRemoteDescription to react to errors such as changing the sctp-init or sctp-port.

Bug: webrtc:426480601
Change-Id: Ifa8b97a4bbc055130c15293cda59a33281ad4f80
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/457901
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#47237}
---
 pc/data_channel_integrationtest.cc |  3 +-
 pc/peer_connection.cc              | 18 +++++++-----
 pc/peer_connection.h               |  5 ++--
 pc/peer_connection_internal.h      |  5 ++--
 pc/sctp_transport.cc               |  4 ++-
 pc/sctp_transport.h                |  4 +--
 pc/sdp_offer_answer_unittest.cc    | 47 +++++++++++++++++++++++-------
 7 files changed, 58 insertions(+), 28 deletions(-)

diff --git a/pc/data_channel_integrationtest.cc b/pc/data_channel_integrationtest.cc
index 3a4b280166..cf7a5b7459 100644
--- a/pc/data_channel_integrationtest.cc
+++ b/pc/data_channel_integrationtest.cc
@@ -1479,8 +1479,7 @@ TEST_P(DataChannelIntegrationTest, ChangingSctpPortIsNotAllowed) {
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_THAT(answer, NotNull());
 
-  // Currently SRD succeeds.
-  EXPECT_TRUE(caller()->SetRemoteDescription(std::move(answer)));
+  EXPECT_FALSE(caller()->SetRemoteDescription(std::move(answer)));
   // Check the state of the SCTP transport.
   VerifySctpState(caller(), SctpTransportState::kClosed);
 }
diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc
index e2a4adedc9..62704136ef 100644
--- a/pc/peer_connection.cc
+++ b/pc/peer_connection.cc
@@ -3116,14 +3116,16 @@ RTCError PeerConnection::StartSctpTransport(const SctpOptions& options) {
   RTC_DCHECK_RUN_ON(signaling_thread());
   RTC_DCHECK(sctp_mid_s_);
 
-  network_thread()->PostTask(
-      SafeTask(network_thread_safety_, [this, mid = *sctp_mid_s_, options] {
-        scoped_refptr<SctpTransport> sctp_transport =
-            transport_controller_n()->GetSctpTransport(mid);
-        if (sctp_transport)
-          sctp_transport->Start(options);
-      }));
-  return RTCError::OK();
+  return network_thread()->BlockingCall([this, mid = *sctp_mid_s_, options] {
+    scoped_refptr<SctpTransport> sctp_transport =
+        transport_controller_n()->GetSctpTransport(mid);
+    RTC_DCHECK(sctp_transport);
+    if (!sctp_transport || !sctp_transport->Start(options)) {
+      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
+                           "Failed to start SCTP transport.");
+    }
+    return RTCError::OK();
+  });
 }
 
 CryptoOptions PeerConnection::GetCryptoOptions() {
diff --git a/pc/peer_connection.h b/pc/peer_connection.h
index 42387f8937..15743e26dc 100644
--- a/pc/peer_connection.h
+++ b/pc/peer_connection.h
@@ -417,8 +417,9 @@ class PeerConnection : public PeerConnectionInternal,
   bool CreateDataChannelTransport(absl::string_view mid) override;
   void DestroyDataChannelTransport(RTCError error) override;
 
-  // Asynchronously calls SctpTransport::Start() on the network thread for
-  // `sctp_mid()` if set. Called as part of setting the local description.
+  // Synchronously calls SctpTransport::Start() on the network thread for
+  // `sctp_mid()` if set. Called as part of pushing down the media descriptions
+  // after a complete offer/answer.
   RTCError StartSctpTransport(const SctpOptions& options) override;
 
   // Returns the CryptoOptions set as RTCConfiguration.crypto_options for this
diff --git a/pc/peer_connection_internal.h b/pc/peer_connection_internal.h
index 8314f7b23e..f31f9992d0 100644
--- a/pc/peer_connection_internal.h
+++ b/pc/peer_connection_internal.h
@@ -127,8 +127,9 @@ class PeerConnectionSdpMethods {
       scoped_refptr<MediaStreamTrackInterface> track,
       const RtpTransceiverInit& init,
       bool fire_callback = true) = 0;
-  // Asynchronously calls SctpTransport::Start() on the network thread for
-  // `sctp_mid()` if set. Called as part of setting the local description.
+  // Synchronously calls SctpTransport::Start() on the network thread for
+  // `sctp_mid()` if set. Called as part of pushing down the media descriptions
+  // after a complete offer/answer.
   virtual RTCError StartSctpTransport(const SctpOptions& options) = 0;
 
   // Asynchronously adds a remote candidate on the network thread.
diff --git a/pc/sctp_transport.cc b/pc/sctp_transport.cc
index 063c855f35..abca8a199c 100644
--- a/pc/sctp_transport.cc
+++ b/pc/sctp_transport.cc
@@ -176,7 +176,7 @@ void SctpTransport::Clear() {
   UpdateInformation(SctpTransportState::kClosed);
 }
 
-void SctpTransport::Start(const SctpOptions& options) {
+bool SctpTransport::Start(const SctpOptions& options) {
   RTC_DCHECK_RUN_ON(owner_thread_);
   info_ =
       SctpTransportInformation(info_.state(), info_.dtls_transport(),
@@ -185,7 +185,9 @@ void SctpTransport::Start(const SctpOptions& options) {
   if (!internal()->Start(options)) {
     RTC_LOG(LS_ERROR) << "Failed to push down SCTP parameters, closing.";
     UpdateInformation(SctpTransportState::kClosed);
+    return false;
   }
+  return true;
 }
 
 void SctpTransport::UpdateInformation(SctpTransportState state) {
diff --git a/pc/sctp_transport.h b/pc/sctp_transport.h
index 32014aa0f8..c6908a9014 100644
--- a/pc/sctp_transport.h
+++ b/pc/sctp_transport.h
@@ -65,9 +65,9 @@ class SctpTransport : public SctpTransportInterface,
 
   // Internal functions
   void Clear();
-  // Initialize the webrtc::SctpTransport. This can be called from
+  // Initialize the webrtc::SctpTransport. This must be called from
   // the signaling thread.
-  void Start(const SctpOptions& options);
+  bool Start(const SctpOptions& options);
 
   // TODO(https://bugs.webrtc.org/10629): Move functions that need
   // internal() to be functions on the SctpTransport interface,
diff --git a/pc/sdp_offer_answer_unittest.cc b/pc/sdp_offer_answer_unittest.cc
index c216c8980a..a4a9c62843 100644
--- a/pc/sdp_offer_answer_unittest.cc
+++ b/pc/sdp_offer_answer_unittest.cc
@@ -2011,30 +2011,30 @@ TEST_F(SdpOfferAnswerTest, SctpInitDisabled) {
   auto pc2 = CreatePeerConnection("WebRTC-Sctp-Snap/Disabled/");
   EXPECT_TRUE(pc1->pc()->CreateDataChannelOrError("dc", nullptr).ok());
   auto offer = pc1->CreateOfferAndSetAsLocal();
-  ASSERT_NE(offer, nullptr);
+  ASSERT_THAT(offer, NotNull());
 
   {
     auto& contents = offer->description()->contents();
-    ASSERT_EQ(contents.size(), 1u);
+    ASSERT_THAT(contents, SizeIs(1));
     auto* media_description = contents[0].media_description();
-    ASSERT_TRUE(media_description);
+    ASSERT_THAT(media_description, NotNull());
     auto* sctp_description = media_description->as_sctp();
-    ASSERT_TRUE(sctp_description);
+    ASSERT_THAT(sctp_description, NotNull());
     EXPECT_FALSE(sctp_description->sctp_init());
   }
 
   RTCError error;
   EXPECT_TRUE(pc2->SetRemoteDescription(std::move(offer)));
   auto answer = pc2->CreateAnswerAndSetAsLocal();
-  ASSERT_NE(answer, nullptr);
+  ASSERT_THAT(answer, NotNull());
 
   {
     auto& contents = answer->description()->contents();
-    ASSERT_EQ(contents.size(), 1u);
+    ASSERT_THAT(contents, SizeIs(1));
     auto* media_description = contents[0].media_description();
-    ASSERT_TRUE(media_description);
+    ASSERT_THAT(media_description, NotNull());
     auto* sctp_description = media_description->as_sctp();
-    ASSERT_TRUE(sctp_description);
+    ASSERT_THAT(sctp_description, NotNull());
     EXPECT_FALSE(sctp_description->sctp_init());
   }
 
@@ -2046,7 +2046,7 @@ TEST_F(SdpOfferAnswerTest, SctpInitWithTrial) {
   auto pc2 = CreatePeerConnection("WebRTC-Sctp-Snap/Enabled/");
   EXPECT_TRUE(pc1->pc()->CreateDataChannelOrError("dc", nullptr).ok());
   auto offer = pc1->CreateOfferAndSetAsLocal();
-  ASSERT_NE(offer, nullptr);
+  ASSERT_THAT(offer, NotNull());
 
   {
     auto& contents = offer->description()->contents();
@@ -2061,7 +2061,7 @@ TEST_F(SdpOfferAnswerTest, SctpInitWithTrial) {
   RTCError error;
   EXPECT_TRUE(pc2->SetRemoteDescription(std::move(offer)));
   auto answer = pc2->CreateAnswerAndSetAsLocal();
-  ASSERT_NE(answer, nullptr);
+  ASSERT_THAT(answer, NotNull());
 
   {
     auto& contents = answer->description()->contents();
@@ -2101,7 +2101,7 @@ TEST_F(SdpOfferAnswerTest, AnswerNoSctpInitInOffer) {
 
   EXPECT_TRUE(pc->SetRemoteDescription(std::move(desc)));
   auto answer = pc->CreateAnswerAndSetAsLocal();
-  ASSERT_NE(answer, nullptr);
+  ASSERT_THAT(answer, NotNull());
   EXPECT_TRUE(answer->ToString(&sdp));
 
   auto& contents = answer->description()->contents();
@@ -2134,6 +2134,31 @@ TEST_F(SdpOfferAnswerTest, AnswerNonBase64SctpInit) {
   auto desc = CreateSessionDescription(SdpType::kOffer, sdp);
   EXPECT_EQ(desc, nullptr);
 }
+
+TEST_F(SdpOfferAnswerTest,
+       AnswerFromNewPeerAfterProvisionalAnswerFailsSnapSctpInit) {
+  auto pc1 = CreatePeerConnection("WebRTC-Sctp-Snap/Enabled/");
+  auto pc2 = CreatePeerConnection("WebRTC-Sctp-Snap/Enabled/");
+  auto pc3 = CreatePeerConnection("WebRTC-Sctp-Snap/Enabled/");
+  EXPECT_TRUE(pc1->pc()->CreateDataChannelOrError("dc", nullptr).ok());
+  auto offer = pc1->CreateOfferAndSetAsLocal();
+  ASSERT_THAT(offer, NotNull());
+
+  EXPECT_TRUE(pc2->SetRemoteDescription(offer->Clone()));
+  EXPECT_TRUE(pc3->SetRemoteDescription(std::move(offer)));
+
+  auto answer2 = pc2->CreateAnswerAndSetAsLocal();
+  ASSERT_THAT(answer2, NotNull());
+  std::string sdp;
+  answer2->ToString(&sdp);
+  auto pranswer2 = CreateSessionDescription(SdpType::kPrAnswer, sdp);
+  EXPECT_TRUE(pc1->SetRemoteDescription(std::move(pranswer2)));
+
+  // Changing the sctp-init is not supported currently.
+  auto answer3 = pc3->CreateAnswerAndSetAsLocal();
+  ASSERT_THAT(answer3, NotNull());
+  EXPECT_FALSE(pc1->SetRemoteDescription(std::move(answer3)));
+}
 #endif  // WEBRTC_HAVE_SCTP
 
 class SdpOfferAnswerDirectionTest

From b8967184f9a63308114ab63044e86e6e48379d97 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Tue, 24 Mar 2026 01:02:47 -0700
Subject: [PATCH 385/472] Roll chromium_revision db815f58ca..c9a20fd775
 (1603797:1603923)

Change log: https://chromium.googlesource.com/chromium/src/+log/db815f58ca..c9a20fd775
Full diff: https://chromium.googlesource.com/chromium/src/+/db815f58ca..c9a20fd775

Changed dependencies
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/a60191e77d..74c2aacf1a
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/8ab6de03be..1aee011cb5
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/c2eae4f8d4..b4c442492a
* src/third_party/androidx/cipd: auHen2c3yuyBQK158jQrEvU0JD_AtKDaBUqflOWWZcAC..SF_J8m-KyUb2Qlq0kZtcHq_x2kYglLBXY1_5u-og9SsC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/8f1f636048..aa0acca1ac
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/a41fdb3aa5..d7bc9113b8
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/145eee0201..d5c8d9fb01
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/2fb50818a9..173b262da7
* src/third_party/nasm: https://chromium.googlesource.com/chromium/deps/nasm.git/+log/af5eeeb054..4525285872
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/5d32182175..f6800908d7
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/3366ed5009..2ca854011d
* src/tools/luci-go: git_revision:09ad27bb7a42580550b9063ef8d0a39a5f01a338..git_revision:8cb5bd940d5f726f8a538212b2287987fcadf837
* src/tools/luci-go: git_revision:09ad27bb7a42580550b9063ef8d0a39a5f01a338..git_revision:8cb5bd940d5f726f8a538212b2287987fcadf837
* src/tools/luci-go: git_revision:09ad27bb7a42580550b9063ef8d0a39a5f01a338..git_revision:8cb5bd940d5f726f8a538212b2287987fcadf837
DEPS diff: https://chromium.googlesource.com/chromium/src/+/db815f58ca..c9a20fd775/DEPS

BUG=None

Change-Id: I6058adfcd203c7ae5811b1f8857257c39e6ff71f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/459640
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47238}
---
 DEPS | 30 +++++++++++++++---------------
 1 file changed, 15 insertions(+), 15 deletions(-)

diff --git a/DEPS b/DEPS
index bded363163..d145ddf62d 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': 'db815f58caa934760fe8f3b900172f25df7fb0d4',
+  'chromium_revision': 'c9a20fd7758f5caf1efbe4e3d4d3bb9bba47533f',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@a60191e77d7a91464114403f5d854262b134c314',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@74c2aacf1af6fbe9a88303e16afaadc195ee9eee',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@8ab6de03bef68504b2738c028c4671ecb33cc5ef',
+    'https://chromium.googlesource.com/chromium/src/testing@1aee011cb5e9ebe3e0821db60d361ea7e9d7513e',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@c2eae4f8d41e653b099718f6a146a97578069c76',
+    'https://chromium.googlesource.com/chromium/src/third_party@b4c442492a7e1eef1234ef946fb35bc5f94059d5',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -488,11 +488,11 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@8f1f6360480481a6950bdd702061e68ecd7d0a79',
+    'https://boringssl.googlesource.com/boringssl.git@aa0acca1acf36b7fb16aa8a7b60b97d5cb01404d',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@8be0e3114685fcc1589561067282edf75ea1259a',
   'src/third_party/catapult':
-    'https://chromium.googlesource.com/catapult.git@a41fdb3aa561b3ae158e35ecbf44f4a14ff1e934',
+    'https://chromium.googlesource.com/catapult.git@d7bc9113b86bdc56239ff7a7d5d1f773d37ca567',
   'src/third_party/ced/src': {
     'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5',
   },
@@ -505,13 +505,13 @@ deps = {
   'src/third_party/crc32c/src':
     'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558',
   'src/third_party/depot_tools':
-    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@145eee02011c02fc25d2ec055b55451661f3a611',
+    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@d5c8d9fb01eff58a78c69beffa1c6fc5d2f5b770',
   'src/third_party/ffmpeg':
     'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@946d97db8d906277085e361892b7efda5152e2f1',
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@2fb50818a9b4742fda5ae1b894dca8683dd3f422',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@173b262da788de397ea3f7d788156c59272f3a89',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '5d32182175cc1063c9c2e5af25a94e7f3ea3f22c',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'f6800908d7bebd44732059ed0310f6e5965d42a6',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -631,7 +631,7 @@ deps = {
 
   # Used by boringssl.
   'src/third_party/nasm': {
-      'url': 'https://chromium.googlesource.com/chromium/deps/nasm.git@af5eeeb054bebadfbb79c7bcd100a95e2ad4525f'
+      'url': 'https://chromium.googlesource.com/chromium/deps/nasm.git@45252858722aad12e545819b2d0f370eb865431b'
   },
 
   'src/third_party/openh264/src':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@3366ed5009a56d79bb7af74577acbf6c42e6c7ae',
+    'https://chromium.googlesource.com/chromium/src/tools@2ca854011dffebab4eecc00110783373a5d2ebb9',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'auHen2c3yuyBQK158jQrEvU0JD_AtKDaBUqflOWWZcAC',
+          'version': 'SF_J8m-KyUb2Qlq0kZtcHq_x2kYglLBXY1_5u-og9SsC',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -808,15 +808,15 @@ deps = {
       'packages': [
         {
           'package': 'infra/tools/luci/cas/${{platform}}',
-          'version': 'git_revision:09ad27bb7a42580550b9063ef8d0a39a5f01a338',
+          'version': 'git_revision:8cb5bd940d5f726f8a538212b2287987fcadf837',
         },
         {
           'package': 'infra/tools/luci/isolate/${{platform}}',
-          'version': 'git_revision:09ad27bb7a42580550b9063ef8d0a39a5f01a338',
+          'version': 'git_revision:8cb5bd940d5f726f8a538212b2287987fcadf837',
         },
         {
           'package': 'infra/tools/luci/swarming/${{platform}}',
-          'version': 'git_revision:09ad27bb7a42580550b9063ef8d0a39a5f01a338',
+          'version': 'git_revision:8cb5bd940d5f726f8a538212b2287987fcadf837',
         }
       ],
       'dep_type': 'cipd',

From 2c28168e10eccc3b709bcccf36426a6644af1621 Mon Sep 17 00:00:00 2001
From: Evan Shrubsole <eshr@webrtc.org>
Date: Wed, 18 Mar 2026 08:56:16 +0000
Subject: [PATCH 386/472] Use GlobalSimulatedTimeController in
 BasicPortAllocatorTest

This removes uses of SIMULATED_WAIT and AutoThread to boot. Drive by
cleanups of fixing units and removing the realtime test fixing a todo.

Adds a Waiter utility to store WaitUntil settings to reduce code
repetition.

Bug: webrtc:469327588,webrtc:381524905,webrtc:42223992
Change-Id: Idc7cb7d572140aa136fe774715c4c91d6a6a6964
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/457940
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47239}
---
 p2p/client/basic_port_allocator_unittest.cc | 778 +++++++-------------
 test/wait_until.h                           |  21 +
 test/wait_until_unittest.cc                 |  26 +
 3 files changed, 312 insertions(+), 513 deletions(-)

diff --git a/p2p/client/basic_port_allocator_unittest.cc b/p2p/client/basic_port_allocator_unittest.cc
index 41a1d60002..6f508eaa92 100644
--- a/p2p/client/basic_port_allocator_unittest.cc
+++ b/p2p/client/basic_port_allocator_unittest.cc
@@ -16,17 +16,20 @@
 #include <memory>
 #include <optional>
 #include <string>
+#include <utility>
 #include <vector>
 
 #include "absl/algorithm/container.h"
+#include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
+#include "api/async_dns_resolver.h"
 #include "api/candidate.h"
 #include "api/environment/environment.h"
-#include "api/environment/environment_factory.h"
 #include "api/field_trials.h"
 #include "api/test/rtc_error_matchers.h"
 #include "api/transport/enums.h"
 #include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
 #include "p2p/base/basic_packet_socket_factory.h"
 #include "p2p/base/p2p_constants.h"
 #include "p2p/base/port.h"
@@ -40,101 +43,85 @@
 #include "p2p/test/stun_server.h"
 #include "p2p/test/test_stun_server.h"
 #include "p2p/test/test_turn_server.h"
-#include "rtc_base/fake_clock.h"
+#include "rtc_base/checks.h"
 #include "rtc_base/fake_mdns_responder.h"
 #include "rtc_base/fake_network.h"
 #include "rtc_base/firewall_socket_server.h"
-#include "rtc_base/gunit.h"
 #include "rtc_base/ip_address.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/net_helper.h"
+#include "rtc_base/net_helpers.h"
 #include "rtc_base/net_test_helpers.h"
 #include "rtc_base/network.h"
 #include "rtc_base/network_constants.h"
 #include "rtc_base/socket.h"
 #include "rtc_base/socket_address.h"
+#include "rtc_base/socket_factory.h"
 #include "rtc_base/thread.h"
 #include "rtc_base/virtual_socket_server.h"
 #include "system_wrappers/include/metrics.h"
+#include "test/create_test_environment.h"
 #include "test/create_test_field_trials.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/time_controller/simulated_time_controller.h"
 #include "test/wait_until.h"
 
-using ::testing::Contains;
-using ::testing::Eq;
-using ::testing::IsTrue;
-using ::testing::Not;
-using ::webrtc::CreateEnvironment;
-using ::webrtc::Environment;
-using ::webrtc::IceCandidateType;
-using ::webrtc::IPAddress;
-using ::webrtc::SocketAddress;
-
 #define MAYBE_SKIP_IPV4                        \
   if (!::webrtc::HasIPv4Enabled()) {           \
     RTC_LOG(LS_INFO) << "No IPv4... skipping"; \
     return;                                    \
   }
 
-static const SocketAddress kAnyAddr("0.0.0.0", 0);
-static const SocketAddress kClientAddr("11.11.11.11", 0);
-static const SocketAddress kClientAddr2("22.22.22.22", 0);
-static const SocketAddress kLoopbackAddr("127.0.0.1", 0);
-static const SocketAddress kPrivateAddr("192.168.1.11", 0);
-static const SocketAddress kPrivateAddr2("192.168.1.12", 0);
-static const SocketAddress kClientIPv6Addr("2401:fa00:4:1000:be30:5bff:fee5:c3",
-                                           0);
-static const SocketAddress kClientIPv6Addr2(
-    "2401:fa00:4:2000:be30:5bff:fee5:c3",
-    0);
-static const SocketAddress kClientIPv6Addr3(
-    "2401:fa00:4:3000:be30:5bff:fee5:c3",
-    0);
-static const SocketAddress kClientIPv6Addr4(
-    "2401:fa00:4:4000:be30:5bff:fee5:c3",
-    0);
-static const SocketAddress kClientIPv6Addr5(
-    "2401:fa00:4:5000:be30:5bff:fee5:c3",
-    0);
-static const SocketAddress kNatUdpAddr("77.77.77.77",
-                                       webrtc::NAT_SERVER_UDP_PORT);
-static const SocketAddress kNatTcpAddr("77.77.77.77",
-                                       webrtc::NAT_SERVER_TCP_PORT);
-static const SocketAddress kRemoteClientAddr("22.22.22.22", 0);
-static const SocketAddress kStunAddr("99.99.99.1", webrtc::STUN_SERVER_PORT);
-static const SocketAddress kTurnUdpIntAddr("99.99.99.4", 3478);
-static const SocketAddress kTurnUdpIntIPv6Addr(
-    "2402:fb00:4:1000:be30:5bff:fee5:c3",
-    3479);
-static const SocketAddress kTurnTcpIntAddr("99.99.99.5", 3478);
-static const SocketAddress kTurnTcpIntIPv6Addr(
-    "2402:fb00:4:2000:be30:5bff:fee5:c3",
-    3479);
-static const SocketAddress kTurnUdpExtAddr("99.99.99.6", 0);
+namespace webrtc {
+namespace {
+
+using ::testing::Contains;
+using ::testing::Eq;
+using ::testing::IsTrue;
+using ::testing::Not;
+
+const SocketAddress kAnyAddr("0.0.0.0", 0);
+const SocketAddress kClientAddr("11.11.11.11", 0);
+const SocketAddress kClientAddr2("22.22.22.22", 0);
+const SocketAddress kLoopbackAddr("127.0.0.1", 0);
+const SocketAddress kPrivateAddr("192.168.1.11", 0);
+const SocketAddress kPrivateAddr2("192.168.1.12", 0);
+const SocketAddress kClientIPv6Addr("2401:fa00:4:1000:be30:5bff:fee5:c3", 0);
+const SocketAddress kClientIPv6Addr2("2401:fa00:4:2000:be30:5bff:fee5:c3", 0);
+const SocketAddress kClientIPv6Addr3("2401:fa00:4:3000:be30:5bff:fee5:c3", 0);
+const SocketAddress kClientIPv6Addr4("2401:fa00:4:4000:be30:5bff:fee5:c3", 0);
+const SocketAddress kClientIPv6Addr5("2401:fa00:4:5000:be30:5bff:fee5:c3", 0);
+const SocketAddress kNatUdpAddr("77.77.77.77", webrtc::NAT_SERVER_UDP_PORT);
+const SocketAddress kNatTcpAddr("77.77.77.77", webrtc::NAT_SERVER_TCP_PORT);
+const SocketAddress kRemoteClientAddr("22.22.22.22", 0);
+const SocketAddress kStunAddr("99.99.99.1", webrtc::STUN_SERVER_PORT);
+const SocketAddress kTurnUdpIntAddr("99.99.99.4", 3478);
+const SocketAddress kTurnUdpIntIPv6Addr("2402:fb00:4:1000:be30:5bff:fee5:c3",
+                                        3479);
+const SocketAddress kTurnTcpIntAddr("99.99.99.5", 3478);
+const SocketAddress kTurnTcpIntIPv6Addr("2402:fb00:4:2000:be30:5bff:fee5:c3",
+                                        3479);
+const SocketAddress kTurnUdpExtAddr("99.99.99.6", 0);
 
 // Minimum and maximum port for port range tests.
-static const int kMinPort = 10000;
-static const int kMaxPort = 10099;
+const int kMinPort = 10000;
+const int kMaxPort = 10099;
 
 // Based on ICE_UFRAG_LENGTH
-static const char kIceUfrag0[] = "UF00";
+const char kIceUfrag0[] = "UF00";
 // Based on ICE_PWD_LENGTH
-static const char kIcePwd0[] = "TESTICEPWD00000000000000";
+const char kIcePwd0[] = "TESTICEPWD00000000000000";
 
-static const char kContentName[] = "test content";
+const char kContentName[] = "test content";
 
-static const int kDefaultAllocationTimeout = 3000;
-static const char kTurnUsername[] = "test";
-static const char kTurnPassword[] = "test";
+constexpr TimeDelta kDefaultAllocationTimeout = TimeDelta::Millis(3000);
+const char kTurnUsername[] = "test";
+const char kTurnPassword[] = "test";
 
 // STUN timeout (with all retries) is webrtc::STUN_TOTAL_TIMEOUT.
 // Add some margin of error for slow bots.
-static const int kStunTimeoutMs = webrtc::STUN_TOTAL_TIMEOUT;
-
-namespace webrtc {
-
-namespace {
+constexpr TimeDelta kStunTimeout = TimeDelta::Millis(STUN_TOTAL_TIMEOUT);
 
 void CheckStunKeepaliveIntervalOfAllReadyPorts(
     const PortAllocatorSession* allocator_session,
@@ -150,28 +137,100 @@ void CheckStunKeepaliveIntervalOfAllReadyPorts(
   }
 }
 
-}  // namespace
-
-
-class BasicPortAllocatorTestBase : public ::testing::Test {
+class FakeAsyncDnsResolverResult : public AsyncDnsResolverResult {
  public:
-  BasicPortAllocatorTestBase()
+  FakeAsyncDnsResolverResult(const SocketAddress& addr, int error)
+      : addr_(addr), error_(error) {}
+
+  bool GetResolvedAddress(int family, SocketAddress* addr) const override {
+    if (error_ != 0)
+      return false;
+    if (family != AF_UNSPEC && addr_.family() != family)
+      return false;
+    *addr = addr_;
+    return true;
+  }
+
+  int GetError() const override { return error_; }
+
+ private:
+  SocketAddress addr_;
+  int error_;
+};
+
+class FakeAsyncDnsResolver : public AsyncDnsResolverInterface {
+ public:
+  void Start(const SocketAddress& addr,
+             absl::AnyInvocable<void()> callback) override {
+    StartInternal(addr, AF_UNSPEC, std::move(callback));
+  }
+
+  void Start(const SocketAddress& addr,
+             int family,
+             absl::AnyInvocable<void()> callback) override {
+    StartInternal(addr, family, std::move(callback));
+  }
+
+  const AsyncDnsResolverResult& result() const override {
+    RTC_DCHECK(result_);
+    return *result_;
+  }
+
+ private:
+  void StartInternal(const SocketAddress& addr,
+                     int family,
+                     absl::AnyInvocable<void()> callback) {
+    SocketAddress resolved = addr;
+    int error = 0;
+    if (addr.hostname() == "localhost") {
+      if (family == AF_INET6) {
+        resolved.SetIP("::1");
+      } else {
+        resolved.SetIP("127.0.0.1");
+      }
+    } else {
+      error = -1;
+    }
+    result_ = std::make_unique<FakeAsyncDnsResolverResult>(resolved, error);
+    Thread::Current()->PostTask(std::move(callback));
+  }
+
+  std::unique_ptr<FakeAsyncDnsResolverResult> result_;
+};
+
+class FakePacketSocketFactory : public BasicPacketSocketFactory {
+ public:
+  explicit FakePacketSocketFactory(SocketFactory* socket_factory)
+      : BasicPacketSocketFactory(socket_factory) {}
+
+  std::unique_ptr<AsyncDnsResolverInterface> CreateAsyncDnsResolver() override {
+    return std::make_unique<FakeAsyncDnsResolver>();
+  }
+};
+
+class BasicPortAllocatorTest : public ::testing::Test {
+ public:
+  BasicPortAllocatorTest()
       : vss_(new VirtualSocketServer()),
         fss_(new FirewallSocketServer(vss_.get())),
+        time_controller_(Timestamp::Zero(), fss_.get()),
+        env_(CreateTestEnvironment({.time = &time_controller_})),
         socket_factory_(fss_.get()),
-        thread_(fss_.get()),
+        thread_(time_controller_.GetMainThread()),
         // Note that the NAT is not used by default. ResetWithStunServerAndNat
         // must be called.
         nat_factory_(vss_.get(), kNatUdpAddr, kNatTcpAddr),
         nat_socket_factory_(new BasicPacketSocketFactory(&nat_factory_)),
-        stun_server_(TestStunServer::Create(env_, kStunAddr, *fss_, thread_)),
+        stun_server_(TestStunServer::Create(env_, kStunAddr, *fss_, *thread_)),
         turn_server_(env_,
-                     Thread::Current(),
+                     thread_,
                      fss_.get(),
                      kTurnUdpIntAddr,
                      kTurnUdpExtAddr),
-        network_manager_(&thread_),
-        candidate_allocation_done_(false) {
+        network_manager_(thread_),
+        candidate_allocation_done_(false),
+        waiter_({.timeout = kDefaultAllocationTimeout,
+                 .clock = &time_controller_}) {
     allocator_.emplace(env_, &network_manager_, &socket_factory_);
     allocator_->SetConfiguration({kStunAddr}, {}, 0, NO_PRUNE, nullptr);
 
@@ -384,12 +443,11 @@ class BasicPortAllocatorTestBase : public ::testing::Test {
 
   static bool HasNetwork(const std::vector<const Network*>& networks,
                          const Network& to_be_found) {
-    auto it =
-        absl::c_find_if(networks, [&to_be_found](const Network* network) {
-          return network->description() == to_be_found.description() &&
-                 network->name() == to_be_found.name() &&
-                 network->prefix() == to_be_found.prefix();
-        });
+    auto it = absl::c_find_if(networks, [&to_be_found](const Network* network) {
+      return network->description() == to_be_found.description() &&
+             network->name() == to_be_found.name() &&
+             network->prefix() == to_be_found.prefix();
+    });
     return it != networks.end();
   }
 
@@ -486,8 +544,8 @@ class BasicPortAllocatorTestBase : public ::testing::Test {
   void ResetWithStunServer(const SocketAddress& stun_server, bool with_nat) {
     if (with_nat) {
       nat_server_ = std::make_unique<NATServer>(
-          env_, NAT_OPEN_CONE, thread_, vss_.get(), kNatUdpAddr, kNatTcpAddr,
-          thread_, vss_.get(), SocketAddress(kNatUdpAddr.ipaddr(), 0));
+          env_, NAT_OPEN_CONE, *thread_, vss_.get(), kNatUdpAddr, kNatTcpAddr,
+          *thread_, vss_.get(), SocketAddress(kNatUdpAddr.ipaddr(), 0));
     } else {
       nat_socket_factory_ =
           std::make_unique<BasicPacketSocketFactory>(fss_.get());
@@ -504,11 +562,12 @@ class BasicPortAllocatorTestBase : public ::testing::Test {
     allocator_->set_step_delay(kMinimumStepDelay);
   }
 
-  Environment env_ = CreateEnvironment();
   std::unique_ptr<VirtualSocketServer> vss_;
   std::unique_ptr<FirewallSocketServer> fss_;
+  GlobalSimulatedTimeController time_controller_;
+  const Environment env_;
   BasicPacketSocketFactory socket_factory_;
-  AutoSocketServerThread thread_;
+  webrtc::Thread* const thread_;
   std::unique_ptr<NATServer> nat_server_;
   NATSocketFactory nat_factory_;
   std::unique_ptr<BasicPacketSocketFactory> nat_socket_factory_;
@@ -520,18 +579,8 @@ class BasicPortAllocatorTestBase : public ::testing::Test {
   std::vector<PortInterface*> ports_;
   std::vector<Candidate> candidates_;
   bool candidate_allocation_done_;
-};
+  Waiter waiter_;
 
-class BasicPortAllocatorTestWithRealClock : public BasicPortAllocatorTestBase {
-};
-
-class FakeClockBase {
- public:
-  ScopedFakeClock fake_clock;
-};
-
-class BasicPortAllocatorTest : public FakeClockBase,
-                               public BasicPortAllocatorTestBase {
  public:
   // This function starts the port/address gathering and check the existence of
   // candidates as specified. When `expect_stun_candidate` is true,
@@ -554,11 +603,7 @@ class BasicPortAllocatorTest : public FakeClockBase,
                         PORTALLOCATOR_ENABLE_SHARED_SOCKET);
     allocator().set_allow_tcp_listen(false);
     session_->StartGettingPorts();
-    EXPECT_THAT(
-        WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                  {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                   .clock = &fake_clock}),
-        IsRtcOk());
+    EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
 
     uint32_t total_candidates = 0;
     if (!host_candidate_addr.IsNil()) {
@@ -615,11 +660,7 @@ class BasicPortAllocatorTest : public FakeClockBase,
 
     ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
     session_->StartGettingPorts();
-    EXPECT_THAT(
-        WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                  {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                   .clock = &fake_clock}),
-        IsRtcOk());
+    EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
     // Three ports (one IPv4 STUN, one IPv6 STUN and one TURN) will be ready.
     EXPECT_EQ(3U, session_->ReadyPorts().size());
     EXPECT_EQ(3U, ports_.size());
@@ -660,11 +701,7 @@ class BasicPortAllocatorTest : public FakeClockBase,
 
     ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
     session_->StartGettingPorts();
-    EXPECT_THAT(
-        WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                  {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                   .clock = &fake_clock}),
-        IsRtcOk());
+    EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
     // Only 2 ports (one STUN and one TURN) are actually being used.
     EXPECT_EQ(2U, session_->ReadyPorts().size());
     // We have verified that each port, when it is added to `ports_`, it is
@@ -720,11 +757,7 @@ class BasicPortAllocatorTest : public FakeClockBase,
         PORTALLOCATOR_ENABLE_IPV6 | PORTALLOCATOR_ENABLE_IPV6_ON_WIFI);
     ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
     session_->StartGettingPorts();
-    EXPECT_THAT(
-        WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                  {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                   .clock = &fake_clock}),
-        IsRtcOk());
+    EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
     // 10 ports (4 STUN and 1 TURN ports on each interface) will be ready to
     // use.
     EXPECT_EQ(10U, session_->ReadyPorts().size());
@@ -803,11 +836,7 @@ TEST_F(BasicPortAllocatorTest, TestIgnoreOnlyLoopbackNetworkByDefault) {
   session_->set_flags(PORTALLOCATOR_DISABLE_STUN | PORTALLOCATOR_DISABLE_RELAY |
                       PORTALLOCATOR_DISABLE_TCP);
   session_->StartGettingPorts();
-  EXPECT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   EXPECT_EQ(4U, candidates_.size());
   for (const Candidate& candidate : candidates_) {
     EXPECT_LT(candidate.address().ip(), 0x12345604U);
@@ -827,11 +856,7 @@ TEST_F(BasicPortAllocatorTest, TestIgnoreNetworksAccordingToIgnoreMask) {
   session_->set_flags(PORTALLOCATOR_DISABLE_STUN | PORTALLOCATOR_DISABLE_RELAY |
                       PORTALLOCATOR_DISABLE_TCP);
   session_->StartGettingPorts();
-  EXPECT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   EXPECT_EQ(1U, candidates_.size());
   EXPECT_EQ(0x12345602U, candidates_[0].address().ip());
 }
@@ -850,11 +875,7 @@ TEST_F(BasicPortAllocatorTest,
       PORTALLOCATOR_DISABLE_TCP | PORTALLOCATOR_DISABLE_COSTLY_NETWORKS);
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  EXPECT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   // Should only get one Wi-Fi candidate.
   EXPECT_EQ(1U, candidates_.size());
   EXPECT_TRUE(HasCandidate(candidates_, IceCandidateType::kHost, "udp", wifi));
@@ -878,11 +899,7 @@ TEST_F(BasicPortAllocatorTest,
       PORTALLOCATOR_DISABLE_TCP | PORTALLOCATOR_DISABLE_COSTLY_NETWORKS);
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  EXPECT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   // Should only get two candidates, none of which is cell.
   EXPECT_EQ(2U, candidates_.size());
   EXPECT_TRUE(
@@ -910,11 +927,7 @@ TEST_F(BasicPortAllocatorTest,
       PORTALLOCATOR_DISABLE_TCP | PORTALLOCATOR_DISABLE_COSTLY_NETWORKS);
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  EXPECT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   // Should only get one Wi-Fi candidate.
   EXPECT_EQ(1U, candidates_.size());
   EXPECT_TRUE(HasCandidate(candidates_, IceCandidateType::kHost, "udp", wifi));
@@ -933,11 +946,7 @@ TEST_F(BasicPortAllocatorTest,
       PORTALLOCATOR_DISABLE_TCP | PORTALLOCATOR_DISABLE_COSTLY_NETWORKS);
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  EXPECT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   // Make sure we got the cell candidate.
   EXPECT_EQ(1U, candidates_.size());
   EXPECT_TRUE(
@@ -959,11 +968,7 @@ TEST_F(BasicPortAllocatorTest,
       PORTALLOCATOR_DISABLE_TCP | PORTALLOCATOR_DISABLE_COSTLY_NETWORKS);
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  EXPECT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   // Make sure we got both wifi and cell candidates.
   EXPECT_EQ(2U, candidates_.size());
   EXPECT_TRUE(HasCandidate(candidates_, IceCandidateType::kHost, "udp",
@@ -990,11 +995,7 @@ TEST_F(BasicPortAllocatorTest,
       PORTALLOCATOR_DISABLE_TCP | PORTALLOCATOR_DISABLE_COSTLY_NETWORKS);
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  EXPECT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   // Make sure we got only wifi candidates.
   EXPECT_EQ(2U, candidates_.size());
   EXPECT_TRUE(HasCandidate(candidates_, IceCandidateType::kHost, "udp", wifi));
@@ -1015,11 +1016,7 @@ TEST_F(BasicPortAllocatorTest,
                         PORTALLOCATOR_DISABLE_TCP);
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  ASSERT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  ASSERT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   // The VPN tap0 network should be filtered out as a costly network, and we
   // should have a UDP port and a STUN port from the Ethernet eth0.
   ASSERT_EQ(2U, ports_.size());
@@ -1043,11 +1040,7 @@ TEST_F(BasicPortAllocatorTest, MaxIpv6NetworksLimitEnforced) {
 
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  EXPECT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   EXPECT_EQ(2U, candidates_.size());
   // Ensure the expected two interfaces (eth0 and eth1) were used.
   EXPECT_TRUE(HasCandidate(candidates_, IceCandidateType::kHost, "udp",
@@ -1073,11 +1066,7 @@ TEST_F(BasicPortAllocatorTest, MaxIpv6NetworksLimitDoesNotImpactIpv4Networks) {
 
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  EXPECT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   EXPECT_EQ(3U, candidates_.size());
   // Ensure that only one IPv6 interface was used, but both IPv4 interfaces
   // were used.
@@ -1097,11 +1086,7 @@ TEST_F(BasicPortAllocatorTest, TestLoopbackNetworkInterface) {
   session_->set_flags(PORTALLOCATOR_DISABLE_STUN | PORTALLOCATOR_DISABLE_RELAY |
                       PORTALLOCATOR_DISABLE_TCP);
   session_->StartGettingPorts();
-  EXPECT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   EXPECT_EQ(1U, candidates_.size());
 }
 
@@ -1110,11 +1095,7 @@ TEST_F(BasicPortAllocatorTest, TestGetAllPortsWithMinimumStepDelay) {
   AddInterface(kClientAddr);
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  ASSERT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  ASSERT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   EXPECT_EQ(3U, candidates_.size());
   EXPECT_EQ(3U, ports_.size());
   EXPECT_TRUE(
@@ -1133,11 +1114,7 @@ TEST_F(BasicPortAllocatorTest, TestSameNetworkDownAndUpWhenSessionNotStopped) {
   AddInterface(kClientAddr, if_name);
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  ASSERT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  ASSERT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   EXPECT_EQ(3U, candidates_.size());
   EXPECT_EQ(3U, ports_.size());
   candidate_allocation_done_ = false;
@@ -1150,7 +1127,7 @@ TEST_F(BasicPortAllocatorTest, TestSameNetworkDownAndUpWhenSessionNotStopped) {
   fss_->set_tcp_sockets_enabled(false);
   fss_->set_udp_sockets_enabled(false);
   RemoveInterface(kClientAddr);
-  SIMULATED_WAIT(false, 1000, fake_clock);
+  time_controller_.AdvanceTime(TimeDelta::Millis(1000));
   EXPECT_EQ(0U, candidates_.size());
   ports_.clear();
   candidate_allocation_done_ = false;
@@ -1160,11 +1137,7 @@ TEST_F(BasicPortAllocatorTest, TestSameNetworkDownAndUpWhenSessionNotStopped) {
   fss_->set_tcp_sockets_enabled(true);
   fss_->set_udp_sockets_enabled(true);
   AddInterface(kClientAddr, if_name);
-  ASSERT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  ASSERT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   EXPECT_EQ(3U, candidates_.size());
   EXPECT_EQ(3U, ports_.size());
 }
@@ -1177,11 +1150,7 @@ TEST_F(BasicPortAllocatorTest, TestSameNetworkDownAndUpWhenSessionStopped) {
   AddInterface(kClientAddr, if_name);
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  ASSERT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  ASSERT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   EXPECT_EQ(3U, candidates_.size());
   EXPECT_EQ(3U, ports_.size());
   session_->StopGettingPorts();
@@ -1191,14 +1160,14 @@ TEST_F(BasicPortAllocatorTest, TestSameNetworkDownAndUpWhenSessionStopped) {
   RemoveInterface(kClientAddr);
   // Wait one (simulated) second and then verify no new candidates have
   // appeared.
-  SIMULATED_WAIT(false, 1000, fake_clock);
+  time_controller_.AdvanceTime(TimeDelta::Millis(1000));
   EXPECT_EQ(0U, candidates_.size());
   EXPECT_EQ(0U, ports_.size());
 
   // When the same interfaces are added again, new candidates/ports should not
   // be generated because the session has stopped.
   AddInterface(kClientAddr, if_name);
-  SIMULATED_WAIT(false, 1000, fake_clock);
+  time_controller_.AdvanceTime(TimeDelta::Millis(1000));
   EXPECT_EQ(0U, candidates_.size());
   EXPECT_EQ(0U, ports_.size());
 }
@@ -1218,11 +1187,7 @@ TEST_F(BasicPortAllocatorTest, CandidatesRegatheredAfterBindingFails) {
   fss_->set_udp_sockets_enabled(false);
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  ASSERT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  ASSERT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   // Make sure we actually prevented candidates from being gathered (other than
   // a single TCP active candidate, since that doesn't require creating a
   // socket).
@@ -1236,11 +1201,7 @@ TEST_F(BasicPortAllocatorTest, CandidatesRegatheredAfterBindingFails) {
   fss_->set_tcp_sockets_enabled(true);
   fss_->set_udp_sockets_enabled(true);
   AddInterface(kClientAddr, if_name);
-  ASSERT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  ASSERT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   // Should get UDP and TCP candidate.
   ASSERT_EQ(2U, candidates_.size());
   EXPECT_TRUE(
@@ -1258,17 +1219,14 @@ TEST_F(BasicPortAllocatorTest, TestGetAllPortsWithOneSecondStepDelay) {
   allocator_->set_step_delay(kDefaultStepDelay);
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  ASSERT_THAT(WaitUntil([&] { return candidates_.size(); }, Eq(2U),
-                        {.clock = &fake_clock}),
+  ASSERT_THAT(waiter_.Until([&] { return candidates_.size(); }, Eq(2U)),
               IsRtcOk());
   EXPECT_EQ(2U, ports_.size());
-  ASSERT_THAT(WaitUntil([&] { return candidates_.size(); }, Eq(3U),
-                        {.clock = &fake_clock}),
+  ASSERT_THAT(waiter_.Until([&] { return candidates_.size(); }, Eq(3U)),
               IsRtcOk());
   EXPECT_EQ(3U, ports_.size());
 
-  ASSERT_THAT(WaitUntil([&] { return candidates_.size(); }, Eq(3U),
-                        {.clock = &fake_clock}),
+  ASSERT_THAT(waiter_.Until([&] { return candidates_.size(); }, Eq(3U)),
               IsRtcOk());
   EXPECT_TRUE(
       HasCandidate(candidates_, IceCandidateType::kHost, "tcp", kClientAddr));
@@ -1282,11 +1240,7 @@ TEST_F(BasicPortAllocatorTest, TestSetupVideoRtpPortsWithNormalSendBuffers) {
   AddInterface(kClientAddr);
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP, "video"));
   session_->StartGettingPorts();
-  ASSERT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  ASSERT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   EXPECT_EQ(3U, candidates_.size());
   // If we Stop gathering now, we shouldn't get a second "done" callback.
   session_->StopGettingPorts();
@@ -1301,18 +1255,11 @@ TEST_F(BasicPortAllocatorTest, TestStopGetAllPorts) {
   AddInterface(kClientAddr);
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  ASSERT_THAT(
-      WaitUntil([&] { return candidates_.size(); }, Eq(2U),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  ASSERT_THAT(waiter_.Until([&] { return candidates_.size(); }, Eq(2U)),
+              IsRtcOk());
   EXPECT_EQ(2U, ports_.size());
   session_->StopGettingPorts();
-  EXPECT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
 }
 
 // Test that we restrict client ports appropriately when a port range is set.
@@ -1328,11 +1275,7 @@ TEST_F(BasicPortAllocatorTest, TestGetAllPortsPortRange) {
   EXPECT_TRUE(SetPortRange(kMinPort, kMaxPort));
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  ASSERT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  ASSERT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   EXPECT_EQ(3U, candidates_.size());
   EXPECT_EQ(3U, ports_.size());
 
@@ -1358,11 +1301,7 @@ TEST_F(BasicPortAllocatorTest, TestGetAllPortsNoAdapters) {
   AddTurnServers(kTurnUdpIntIPv6Addr, kTurnTcpIntIPv6Addr);
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  EXPECT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   EXPECT_EQ(4U, ports_.size());
   EXPECT_EQ(1,
             CountPorts(ports_, IceCandidateType::kSrflx, PROTO_UDP, kAnyAddr));
@@ -1504,11 +1443,7 @@ TEST_F(BasicPortAllocatorTest, TestDisableUdpTurn) {
                       PORTALLOCATOR_ENABLE_SHARED_SOCKET);
 
   session_->StartGettingPorts();
-  EXPECT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
 
   // Expect to see 2 ports and 2 candidates - TURN/TCP and TCP ports, TCP and
   // TURN/TCP candidates.
@@ -1531,9 +1466,7 @@ TEST_F(BasicPortAllocatorTest, TestDisableAllPorts) {
   session_->set_flags(PORTALLOCATOR_DISABLE_UDP | PORTALLOCATOR_DISABLE_STUN |
                       PORTALLOCATOR_DISABLE_RELAY | PORTALLOCATOR_DISABLE_TCP);
   session_->StartGettingPorts();
-  EXPECT_THAT(WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                        {.clock = &fake_clock}),
-              IsRtcOk());
+  EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   EXPECT_EQ(0U, candidates_.size());
 }
 
@@ -1543,11 +1476,7 @@ TEST_F(BasicPortAllocatorTest, TestGetAllPortsNoUdpSockets) {
   fss_->set_udp_sockets_enabled(false);
   ASSERT_TRUE(CreateSession(1));
   session_->StartGettingPorts();
-  ASSERT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  ASSERT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   EXPECT_EQ(1U, candidates_.size());
   EXPECT_EQ(1U, ports_.size());
   EXPECT_TRUE(
@@ -1563,11 +1492,7 @@ TEST_F(BasicPortAllocatorTest, TestGetAllPortsNoUdpSocketsNoTcpListen) {
   fss_->set_tcp_listen_enabled(false);
   ASSERT_TRUE(CreateSession(1));
   session_->StartGettingPorts();
-  ASSERT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  ASSERT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   EXPECT_EQ(1U, candidates_.size());
   EXPECT_EQ(1U, ports_.size());
   EXPECT_TRUE(
@@ -1582,7 +1507,9 @@ TEST_F(BasicPortAllocatorTest, TestGetAllPortsNoSockets) {
   fss_->set_udp_sockets_enabled(false);
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  SIMULATED_WAIT(!candidates_.empty(), 2000, fake_clock);
+  EXPECT_TRUE(
+      Waiter({.timeout = TimeDelta::Millis(2000), .clock = &time_controller_})
+          .Until([&] { return !candidates_.empty(); }));
   // TODO(deadbeef): Check candidate_allocation_done signal.
   // In case of Relay, ports creation will succeed but sockets will fail.
   // There is no error reporting from RelayEntry to handle this failure.
@@ -1594,11 +1521,8 @@ TEST_F(BasicPortAllocatorTest, TestGetAllPortsNoUdpAllowed) {
   AddInterface(kClientAddr);
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  EXPECT_THAT(
-      WaitUntil([&] { return candidates_.size(); }, Eq(2U),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_THAT(waiter_.Until([&] { return candidates_.size(); }, Eq(2U)),
+              IsRtcOk());
   EXPECT_EQ(2U, ports_.size());
   EXPECT_TRUE(
       HasCandidate(candidates_, IceCandidateType::kHost, "udp", kClientAddr));
@@ -1606,10 +1530,9 @@ TEST_F(BasicPortAllocatorTest, TestGetAllPortsNoUdpAllowed) {
       HasCandidate(candidates_, IceCandidateType::kHost, "tcp", kClientAddr));
   // We wait at least for a full STUN timeout, which
   // STUN_TOTAL_TIMEOUT seconds.
-  EXPECT_THAT(WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(STUN_TOTAL_TIMEOUT),
-                         .clock = &fake_clock}),
-              IsRtcOk());
+  EXPECT_TRUE(Waiter({.timeout = TimeDelta::Millis(STUN_TOTAL_TIMEOUT),
+                      .clock = &time_controller_})
+                  .Until([&] { return candidate_allocation_done_; }));
   // No additional (STUN) candidates.
   EXPECT_EQ(2U, candidates_.size());
 }
@@ -1623,11 +1546,7 @@ TEST_F(BasicPortAllocatorTest, TestCandidatePriorityOfMultipleInterfaces) {
                         PORTALLOCATOR_DISABLE_RELAY);
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  EXPECT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   ASSERT_EQ(2U, candidates_.size());
   EXPECT_EQ(2U, ports_.size());
   // Candidates priorities should be different.
@@ -1639,11 +1558,7 @@ TEST_F(BasicPortAllocatorTest, TestGetAllPortsRestarts) {
   AddInterface(kClientAddr);
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  EXPECT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   EXPECT_EQ(3U, candidates_.size());
   EXPECT_EQ(3U, ports_.size());
   // TODO(deadbeef): Extend this to verify ICE restart.
@@ -1661,11 +1576,7 @@ TEST_F(BasicPortAllocatorTest, TestSessionUsesOwnCandidateFilter) {
   session_->StartGettingPorts();
   // 7 candidates and 4 ports is what we would normally get (see the
   // TestGetAllPorts* tests).
-  EXPECT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   EXPECT_EQ(3U, candidates_.size());
   EXPECT_EQ(3U, ports_.size());
 }
@@ -1682,11 +1593,7 @@ TEST_F(BasicPortAllocatorTest, TestCandidateFilterWithRelayOnly) {
   allocator().SetCandidateFilter(CF_RELAY);
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  EXPECT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   EXPECT_TRUE(HasCandidate(candidates_, IceCandidateType::kRelay, "udp",
                            SocketAddress(kTurnUdpExtAddr.ipaddr(), 0)));
 
@@ -1703,11 +1610,7 @@ TEST_F(BasicPortAllocatorTest, TestCandidateFilterWithHostOnly) {
   allocator().SetCandidateFilter(CF_HOST);
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  EXPECT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   EXPECT_EQ(2U, candidates_.size());  // Host UDP/TCP candidates only.
   EXPECT_EQ(2U, ports_.size());       // UDP/TCP ports only.
   for (const Candidate& candidate : candidates_) {
@@ -1724,11 +1627,7 @@ TEST_F(BasicPortAllocatorTest, TestCandidateFilterWithReflexiveOnly) {
   allocator().SetCandidateFilter(CF_REFLEXIVE);
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  EXPECT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   // Host is behind NAT, no private address will be exposed. Hence only UDP
   // port with STUN candidate will be sent outside.
   EXPECT_EQ(1U, candidates_.size());  // Only STUN candidate.
@@ -1745,11 +1644,7 @@ TEST_F(BasicPortAllocatorTest, TestCandidateFilterWithReflexiveOnlyAndNoNAT) {
   allocator().SetCandidateFilter(CF_REFLEXIVE);
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  EXPECT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   // Host has a public address, both UDP and TCP candidates will be exposed.
   EXPECT_EQ(2U, candidates_.size());  // Local UDP + TCP candidate.
   EXPECT_EQ(2U, ports_.size());  //  UDP and TCP ports will be in ready state.
@@ -1763,11 +1658,7 @@ TEST_F(BasicPortAllocatorTest, TestEnableSharedUfrag) {
   AddInterface(kClientAddr);
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  ASSERT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  ASSERT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   EXPECT_EQ(3U, candidates_.size());
   EXPECT_TRUE(
       HasCandidate(candidates_, IceCandidateType::kHost, "udp", kClientAddr));
@@ -1792,19 +1683,12 @@ TEST_F(BasicPortAllocatorTest, TestSharedSocketWithoutNat) {
                         PORTALLOCATOR_ENABLE_SHARED_SOCKET);
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  ASSERT_THAT(
-      WaitUntil([&] { return candidates_.size(); }, Eq(2U),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  ASSERT_THAT(waiter_.Until([&] { return candidates_.size(); }, Eq(2U)),
+              IsRtcOk());
   EXPECT_EQ(2U, ports_.size());
   EXPECT_TRUE(
       HasCandidate(candidates_, IceCandidateType::kHost, "udp", kClientAddr));
-  EXPECT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
 }
 
 // Test that when PORTALLOCATOR_ENABLE_SHARED_SOCKET is enabled only one port
@@ -1818,21 +1702,14 @@ TEST_F(BasicPortAllocatorTest, TestSharedSocketWithNat) {
                         PORTALLOCATOR_ENABLE_SHARED_SOCKET);
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  ASSERT_THAT(
-      WaitUntil([&] { return candidates_.size(); }, Eq(3U),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  ASSERT_THAT(waiter_.Until([&] { return candidates_.size(); }, Eq(3U)),
+              IsRtcOk());
   ASSERT_EQ(2U, ports_.size());
   EXPECT_TRUE(
       HasCandidate(candidates_, IceCandidateType::kHost, "udp", kClientAddr));
   EXPECT_TRUE(HasCandidate(candidates_, IceCandidateType::kSrflx, "udp",
                            SocketAddress(kNatUdpAddr.ipaddr(), 0)));
-  EXPECT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   EXPECT_EQ(3U, candidates_.size());
 }
 
@@ -1853,11 +1730,7 @@ TEST_F(BasicPortAllocatorTest, TestSharedSocketWithoutNatUsingTurn) {
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
 
-  ASSERT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  ASSERT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   ASSERT_EQ(3U, candidates_.size());
   ASSERT_EQ(3U, ports_.size());
   EXPECT_TRUE(
@@ -1967,17 +1840,16 @@ TEST_F(BasicPortAllocatorTest,
 
 // Testing DNS resolve for the TURN server, this will test AllocationSequence
 // handling the unresolved address signal from TurnPort.
-// TODO(pthatcher): Make this test work with SIMULATED_WAIT. It
-// appears that it doesn't currently because of the DNS look up not
-// using the fake clock.
-TEST_F(BasicPortAllocatorTestWithRealClock,
-       TestSharedSocketWithServerAddressResolve) {
+TEST_F(BasicPortAllocatorTest, TestSharedSocketWithServerAddressResolve) {
   // This test relies on a real query for "localhost", so it won't work on an
   // IPv6-only machine.
   MAYBE_SKIP_IPV4;
   turn_server_.AddInternalSocket(SocketAddress("127.0.0.1", 3478), PROTO_UDP);
   AddInterface(kClientAddr);
-  allocator_.emplace(env_, &network_manager_, &socket_factory_);
+  // Use a fake socket factory to avoid real DNS lookups which may not used a
+  // fake clock.
+  FakePacketSocketFactory fake_socket_factory(fss_.get());
+  allocator_.emplace(env_, &network_manager_, &fake_socket_factory);
   allocator_->Initialize();
   RelayServerConfig turn_server;
   RelayCredentials credentials(kTurnUsername, kTurnPassword);
@@ -1994,10 +1866,9 @@ TEST_F(BasicPortAllocatorTestWithRealClock,
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
 
-  EXPECT_THAT(
-      WaitUntil([&] { return ports_.size(); }, Eq(2U),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout)}),
-      IsRtcOk());
+  EXPECT_THAT(waiter_.Until([&] { return ports_.size(); }, Eq(2U)), IsRtcOk());
+  session_.reset();
+  allocator_.reset();
 }
 
 // Test that when PORTALLOCATOR_ENABLE_SHARED_SOCKET is enabled only one port
@@ -2016,11 +1887,7 @@ TEST_F(BasicPortAllocatorTest, TestSharedSocketWithNatUsingTurn) {
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
 
-  ASSERT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  ASSERT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   EXPECT_EQ(3U, candidates_.size());
   ASSERT_EQ(2U, ports_.size());
   EXPECT_TRUE(
@@ -2029,11 +1896,7 @@ TEST_F(BasicPortAllocatorTest, TestSharedSocketWithNatUsingTurn) {
                            SocketAddress(kNatUdpAddr.ipaddr(), 0)));
   EXPECT_TRUE(HasCandidate(candidates_, IceCandidateType::kRelay, "udp",
                            SocketAddress(kTurnUdpExtAddr.ipaddr(), 0)));
-  EXPECT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   // Local port will be created first and then TURN port.
   // TODO(deadbeef): This isn't something the BasicPortAllocator API contract
   // guarantees...
@@ -2062,11 +1925,7 @@ TEST_F(BasicPortAllocatorTest, TestSharedSocketWithNatUsingTurnAsStun) {
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
 
-  ASSERT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  ASSERT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   EXPECT_EQ(3U, candidates_.size());
   EXPECT_TRUE(
       HasCandidate(candidates_, IceCandidateType::kHost, "udp", kClientAddr));
@@ -2101,11 +1960,7 @@ TEST_F(BasicPortAllocatorTest, TestSharedSocketWithNatUsingTurnTcpOnly) {
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
 
-  ASSERT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  ASSERT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   EXPECT_EQ(2U, candidates_.size());
   ASSERT_EQ(2U, ports_.size());
   EXPECT_TRUE(
@@ -2132,11 +1987,7 @@ TEST_F(BasicPortAllocatorTest, TestNonSharedSocketWithNatUsingTurnAsStun) {
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
 
-  ASSERT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  ASSERT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   EXPECT_EQ(3U, candidates_.size());
   ASSERT_EQ(3U, ports_.size());
   EXPECT_TRUE(
@@ -2175,11 +2026,8 @@ TEST_F(BasicPortAllocatorTest, TestSharedSocketWithNatUsingTurnAndStun) {
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
 
-  ASSERT_THAT(
-      WaitUntil([&] { return candidates_.size(); }, Eq(3U),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  ASSERT_THAT(waiter_.Until([&] { return candidates_.size(); }, Eq(3U)),
+              IsRtcOk());
   EXPECT_TRUE(
       HasCandidate(candidates_, IceCandidateType::kHost, "udp", kClientAddr));
   Candidate stun_candidate;
@@ -2205,19 +2053,15 @@ TEST_F(BasicPortAllocatorTest, TestSharedSocketNoUdpAllowed) {
   AddInterface(kClientAddr);
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  ASSERT_THAT(
-      WaitUntil([&] { return ports_.size(); }, Eq(1U),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  ASSERT_THAT(waiter_.Until([&] { return ports_.size(); }, Eq(1U)), IsRtcOk());
   EXPECT_EQ(1U, candidates_.size());
   EXPECT_TRUE(
       HasCandidate(candidates_, IceCandidateType::kHost, "udp", kClientAddr));
   // STUN timeout is 9.5sec. We need to wait to get candidate done signal.
-  EXPECT_THAT(WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(kStunTimeoutMs),
-                         .clock = &fake_clock}),
-              IsRtcOk());
+  EXPECT_TRUE(
+      Waiter({.timeout = kStunTimeout, .clock = &time_controller_}).Until([&] {
+        return candidate_allocation_done_;
+      }));
   EXPECT_EQ(1U, candidates_.size());
 }
 
@@ -2237,11 +2081,7 @@ TEST_F(BasicPortAllocatorTest, TestNetworkPermissionBlocked) {
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   EXPECT_EQ(0U, session_->flags() & PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION);
   session_->StartGettingPorts();
-  EXPECT_THAT(
-      WaitUntil([&] { return ports_.size(); }, Eq(1U),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_THAT(waiter_.Until([&] { return ports_.size(); }, Eq(1U)), IsRtcOk());
   EXPECT_EQ(1U, candidates_.size());
   EXPECT_TRUE(
       HasCandidate(candidates_, IceCandidateType::kHost, "udp", kPrivateAddr));
@@ -2258,11 +2098,7 @@ TEST_F(BasicPortAllocatorTest, TestEnableIPv6Addresses) {
   allocator_->set_step_delay(kMinimumStepDelay);
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  ASSERT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  ASSERT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   EXPECT_EQ(4U, ports_.size());
   EXPECT_EQ(4U, candidates_.size());
   EXPECT_TRUE(HasCandidate(candidates_, IceCandidateType::kHost, "udp",
@@ -2280,14 +2116,11 @@ TEST_F(BasicPortAllocatorTest, TestStopGettingPorts) {
   allocator_->set_step_delay(kDefaultStepDelay);
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  ASSERT_THAT(WaitUntil([&] { return candidates_.size(); }, Eq(2U),
-                        {.clock = &fake_clock}),
+  ASSERT_THAT(waiter_.Until([&] { return candidates_.size(); }, Eq(2U)),
               IsRtcOk());
   EXPECT_EQ(2U, ports_.size());
   session_->StopGettingPorts();
-  EXPECT_THAT(WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                        {.clock = &fake_clock}),
-              IsRtcOk());
+  EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
 
   // After stopping getting ports, adding a new interface will not start
   // getting ports again.
@@ -2296,7 +2129,7 @@ TEST_F(BasicPortAllocatorTest, TestStopGettingPorts) {
   ports_.clear();
   candidate_allocation_done_ = false;
   network_manager_.AddInterface(kClientAddr2);
-  SIMULATED_WAIT(false, 1000, fake_clock);
+  time_controller_.AdvanceTime(TimeDelta::Millis(1000));
   EXPECT_EQ(0U, candidates_.size());
   EXPECT_EQ(0U, ports_.size());
 }
@@ -2306,14 +2139,11 @@ TEST_F(BasicPortAllocatorTest, TestClearGettingPorts) {
   allocator_->set_step_delay(kDefaultStepDelay);
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  ASSERT_THAT(WaitUntil([&] { return candidates_.size(); }, Eq(2U),
-                        {.clock = &fake_clock}),
+  ASSERT_THAT(waiter_.Until([&] { return candidates_.size(); }, Eq(2U)),
               IsRtcOk());
   EXPECT_EQ(2U, ports_.size());
   session_->ClearGettingPorts();
-  EXPECT_THAT(WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                        {.clock = &fake_clock}),
-              IsRtcOk());
+  EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
 
   // After clearing getting ports, adding a new interface will start getting
   // ports again.
@@ -2322,15 +2152,10 @@ TEST_F(BasicPortAllocatorTest, TestClearGettingPorts) {
   ports_.clear();
   candidate_allocation_done_ = false;
   network_manager_.AddInterface(kClientAddr2);
-  ASSERT_THAT(WaitUntil([&] { return candidates_.size(); }, Eq(2U),
-                        {.clock = &fake_clock}),
+  ASSERT_THAT(waiter_.Until([&] { return candidates_.size(); }, Eq(2U)),
               IsRtcOk());
   EXPECT_EQ(2U, ports_.size());
-  EXPECT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
 }
 
 // Test that the ports and candidates are updated with new ufrag/pwd/etc. when
@@ -2342,12 +2167,8 @@ TEST_F(BasicPortAllocatorTest, TestTransportInformationUpdated) {
                                allocator_->turn_servers(), pool_size, NO_PRUNE);
   const PortAllocatorSession* peeked_session = allocator_->GetPooledSession();
   ASSERT_NE(nullptr, peeked_session);
-  EXPECT_THAT(
-      WaitUntil([&] { return peeked_session->CandidatesAllocationDone(); },
-                IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_TRUE(waiter_.Until(
+      [&] { return peeked_session->CandidatesAllocationDone(); }));
   // Expect that when TakePooledSession is called,
   // UpdateTransportInformationInternal will be called and the
   // BasicPortAllocatorSession will update the ufrag/pwd of ports and
@@ -2382,12 +2203,8 @@ TEST_F(BasicPortAllocatorTest, TestSetCandidateFilterAfterCandidatesGathered) {
                                allocator_->turn_servers(), pool_size, NO_PRUNE);
   const PortAllocatorSession* peeked_session = allocator_->GetPooledSession();
   ASSERT_NE(nullptr, peeked_session);
-  EXPECT_THAT(
-      WaitUntil([&] { return peeked_session->CandidatesAllocationDone(); },
-                IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_TRUE(waiter_.Until(
+      [&] { return peeked_session->CandidatesAllocationDone(); }));
   size_t initial_candidates_size = peeked_session->ReadyCandidates().size();
   size_t initial_ports_size = peeked_session->ReadyPorts().size();
   allocator_->SetCandidateFilter(CF_RELAY);
@@ -2434,41 +2251,28 @@ TEST_F(BasicPortAllocatorTest,
   allocator_->SetCandidateFilter(CF_NONE);
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  EXPECT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   EXPECT_TRUE(candidates_.empty());
   EXPECT_TRUE(ports_.empty());
 
   // Surface the relay candidate previously gathered but not signaled.
   session_->SetCandidateFilter(CF_RELAY);
-  ASSERT_THAT(
-      WaitUntil([&] { return candidates_.size(); }, Eq(1u),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  ASSERT_THAT(waiter_.Until([&] { return candidates_.size(); }, Eq(1u)),
+              IsRtcOk());
   EXPECT_TRUE(candidates_.back().is_relay());
   EXPECT_EQ(1u, ports_.size());
 
   // Surface the srflx candidate previously gathered but not signaled.
   session_->SetCandidateFilter(CF_RELAY | CF_REFLEXIVE);
-  ASSERT_THAT(
-      WaitUntil([&] { return candidates_.size(); }, Eq(2u),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  ASSERT_THAT(waiter_.Until([&] { return candidates_.size(); }, Eq(2u)),
+              IsRtcOk());
   EXPECT_TRUE(candidates_.back().is_stun());
   EXPECT_EQ(2u, ports_.size());
 
   // Surface the srflx candidate previously gathered but not signaled.
   session_->SetCandidateFilter(CF_ALL);
-  ASSERT_THAT(
-      WaitUntil([&] { return candidates_.size(); }, Eq(3u),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  ASSERT_THAT(waiter_.Until([&] { return candidates_.size(); }, Eq(3u)),
+              IsRtcOk());
   EXPECT_TRUE(candidates_.back().is_local());
   EXPECT_EQ(2u, ports_.size());
 }
@@ -2496,41 +2300,28 @@ TEST_F(
   allocator_->SetCandidateFilter(CF_NONE);
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  EXPECT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   EXPECT_TRUE(candidates_.empty());
   EXPECT_TRUE(ports_.empty());
 
   // Surface the relay candidate previously gathered but not signaled.
   session_->SetCandidateFilter(CF_RELAY);
-  EXPECT_THAT(
-      WaitUntil([&] { return candidates_.size(); }, Eq(1u),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_THAT(waiter_.Until([&] { return candidates_.size(); }, Eq(1u)),
+              IsRtcOk());
   EXPECT_TRUE(candidates_.back().is_relay());
   EXPECT_EQ(1u, ports_.size());
 
   // Surface the srflx candidate previously gathered but not signaled.
   session_->SetCandidateFilter(CF_REFLEXIVE);
-  EXPECT_THAT(
-      WaitUntil([&] { return candidates_.size(); }, Eq(2u),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_THAT(waiter_.Until([&] { return candidates_.size(); }, Eq(2u)),
+              IsRtcOk());
   EXPECT_TRUE(candidates_.back().is_stun());
   EXPECT_EQ(2u, ports_.size());
 
   // Surface the host candidate previously gathered but not signaled.
   session_->SetCandidateFilter(CF_HOST);
-  EXPECT_THAT(
-      WaitUntil([&] { return candidates_.size(); }, Eq(3u),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_THAT(waiter_.Until([&] { return candidates_.size(); }, Eq(3u)),
+              IsRtcOk());
   EXPECT_TRUE(candidates_.back().is_local());
   // We use a shared socket and UDPPort handles the srflx candidate.
   EXPECT_EQ(2u, ports_.size());
@@ -2553,11 +2344,7 @@ TEST_F(BasicPortAllocatorTest,
   allocator_->SetCandidateFilter(CF_NONE);
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  EXPECT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   auto test_invariants = [this]() {
     EXPECT_TRUE(candidates_.empty());
     EXPECT_TRUE(ports_.empty());
@@ -2568,15 +2355,15 @@ TEST_F(BasicPortAllocatorTest,
   session_->StopGettingPorts();
 
   session_->SetCandidateFilter(CF_RELAY);
-  SIMULATED_WAIT(false, kDefaultAllocationTimeout, fake_clock);
+  time_controller_.AdvanceTime(kDefaultAllocationTimeout);
   test_invariants();
 
   session_->SetCandidateFilter(CF_RELAY | CF_REFLEXIVE);
-  SIMULATED_WAIT(false, kDefaultAllocationTimeout, fake_clock);
+  time_controller_.AdvanceTime(kDefaultAllocationTimeout);
   test_invariants();
 
   session_->SetCandidateFilter(CF_ALL);
-  SIMULATED_WAIT(false, kDefaultAllocationTimeout, fake_clock);
+  time_controller_.AdvanceTime(kDefaultAllocationTimeout);
   test_invariants();
 }
 
@@ -2589,12 +2376,8 @@ TEST_F(BasicPortAllocatorTest, SetStunKeepaliveIntervalForPorts) {
                                nullptr, expected_stun_keepalive_interval);
   auto* pooled_session = allocator_->GetPooledSession();
   ASSERT_NE(nullptr, pooled_session);
-  EXPECT_THAT(
-      WaitUntil([&] { return pooled_session->CandidatesAllocationDone(); },
-                IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_TRUE(waiter_.Until(
+      [&] { return pooled_session->CandidatesAllocationDone(); }));
   CheckStunKeepaliveIntervalOfAllReadyPorts(pooled_session,
                                             expected_stun_keepalive_interval);
 }
@@ -2608,12 +2391,8 @@ TEST_F(BasicPortAllocatorTest,
                                nullptr, 123 /* stun keepalive interval */);
   auto* pooled_session = allocator_->GetPooledSession();
   ASSERT_NE(nullptr, pooled_session);
-  EXPECT_THAT(
-      WaitUntil([&] { return pooled_session->CandidatesAllocationDone(); },
-                IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_TRUE(waiter_.Until(
+      [&] { return pooled_session->CandidatesAllocationDone(); }));
   const int expected_stun_keepalive_interval = 321;
   allocator_->SetConfiguration(allocator_->stun_servers(),
                                allocator_->turn_servers(), pool_size, NO_PRUNE,
@@ -2634,11 +2413,7 @@ TEST_F(BasicPortAllocatorTest,
                                nullptr, expected_stun_keepalive_interval);
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  EXPECT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   CheckStunKeepaliveIntervalOfAllReadyPorts(session_.get(),
                                             expected_stun_keepalive_interval);
 }
@@ -2655,11 +2430,7 @@ TEST_F(BasicPortAllocatorTest,
                                nullptr, expected_stun_keepalive_interval);
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  EXPECT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   CheckStunKeepaliveIntervalOfAllReadyPorts(session_.get(),
                                             expected_stun_keepalive_interval);
 }
@@ -2681,11 +2452,7 @@ TEST_F(BasicPortAllocatorTest, HostCandidateAddressIsReplacedByHostname) {
   AddInterface(kClientAddr);
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  ASSERT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  ASSERT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   EXPECT_EQ(5u, candidates_.size());
   int num_host_udp_candidates = 0;
   int num_host_tcp_candidates = 0;
@@ -2762,11 +2529,7 @@ TEST_F(BasicPortAllocatorTest, AssignsUniqueLocalPreferencetoRelayCandidates) {
   AddInterface(kClientAddr);
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  ASSERT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  ASSERT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
   EXPECT_EQ(3u, candidates_.size());
   EXPECT_GT((candidates_[0].priority() >> 8) & 0xFFFF,
             (candidates_[1].priority() >> 8) & 0xFFFF);
@@ -2913,11 +2676,7 @@ TEST_F(BasicPortAllocatorTest, Select2DifferentIntefaces) {
 
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  EXPECT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
 
   EXPECT_EQ(2U, candidates_.size());
   // ethe1 and wifi1 were selected.
@@ -2943,11 +2702,7 @@ TEST_F(BasicPortAllocatorTest, Select3DifferentIntefaces) {
 
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  EXPECT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
 
   EXPECT_EQ(3U, candidates_.size());
   // ethe1, wifi1, and cell1 were selected.
@@ -2975,11 +2730,7 @@ TEST_F(BasicPortAllocatorTest, Select4DifferentIntefaces) {
 
   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  EXPECT_THAT(
-      WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(),
-                {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout),
-                 .clock = &fake_clock}),
-      IsRtcOk());
+  EXPECT_TRUE(waiter_.Until([&] { return candidate_allocation_done_; }));
 
   EXPECT_EQ(4U, candidates_.size());
   // ethe1, ethe2, wifi1, and cell1 were selected.
@@ -2993,4 +2744,5 @@ TEST_F(BasicPortAllocatorTest, Select4DifferentIntefaces) {
                            kClientIPv6Addr5));
 }
 
+}  // namespace
 }  // namespace webrtc
diff --git a/test/wait_until.h b/test/wait_until.h
index c5e4f8f3ba..a4c5421b65 100644
--- a/test/wait_until.h
+++ b/test/wait_until.h
@@ -99,6 +99,27 @@ template <typename Fn>
   return RTCError(RTCErrorType::INTERNAL_ERROR, listener.str());
 }
 
+// A utility class that binds WaitUntilSettings to WaitUntil methods.
+class Waiter {
+ public:
+  explicit Waiter(WaitUntilSettings settings)
+      : settings_(std::move(settings)) {}
+
+  [[nodiscard]] bool Until(FunctionView<bool()> fn) const {
+    return WaitUntil(std::move(fn), settings_);
+  }
+
+  template <typename Fn>
+  [[nodiscard]] RTCErrorOr<std::invoke_result_t<Fn>> Until(
+      const Fn& fn,
+      ::testing::Matcher<std::invoke_result_t<Fn>> matcher) const {
+    return WaitUntil(fn, std::move(matcher), settings_);
+  }
+
+ private:
+  WaitUntilSettings settings_;
+};
+
 }  // namespace webrtc
 
 #endif  // TEST_WAIT_UNTIL_H_
diff --git a/test/wait_until_unittest.cc b/test/wait_until_unittest.cc
index 181d6c1b08..a78a377010 100644
--- a/test/wait_until_unittest.cc
+++ b/test/wait_until_unittest.cc
@@ -194,5 +194,31 @@ TEST(WaitUntilTest,
   EXPECT_EQ(time_controller->GetClock()->CurrentTime(), start);
 }
 
+TEST(WaiterTest, ReturnsTrueWhenConditionIsMet) {
+  ScopedFakeClock clock;
+  Waiter waiter({.timeout = TimeDelta::Seconds(1), .clock = &clock});
+
+  int counter = 0;
+  EXPECT_TRUE(waiter.Until([&] { return ++counter == 3; }));
+  EXPECT_EQ(counter, 3);
+}
+
+TEST(WaiterTest, ReturnsResultWhenMatcherIsMet) {
+  ScopedFakeClock clock;
+  Waiter waiter({.timeout = TimeDelta::Seconds(1), .clock = &clock});
+
+  int counter = 0;
+  auto result = waiter.Until([&] { return ++counter; }, Eq(3));
+  EXPECT_THAT(result, IsRtcOkAndHolds(3));
+}
+
+TEST(WaiterTest, ReturnsFalseWhenTimeoutIsReached) {
+  SimulatedClock clock(Timestamp::Millis(1000));
+  Waiter waiter({.timeout = TimeDelta::Millis(100), .clock = &clock});
+
+  EXPECT_FALSE(waiter.Until([&] { return false; }));
+  EXPECT_EQ(clock.CurrentTime(), Timestamp::Millis(1100));
+}
+
 }  // namespace
 }  // namespace webrtc

From e68d24dcebce1c13e4f7f79f94ef0e56b3275460 Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Tue, 24 Mar 2026 08:47:05 +0000
Subject: [PATCH 387/472] Replace LOG_AND_RETURN_ERROR with "return LOG_ERROR"

Deletes the LOG_AND_RETURN_ERROR and LOG_AND_RETURN_ERROR_EX macros.
This makes for code that is easier to read.

Bug: None
Change-Id: I9ce0de3c2f6337def0ae8a3d626fd3374d4bebe0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/459600
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47240}
---
 api/rtc_error.h                 |  15 ---
 api/rtc_error_unittest.cc       |   2 +-
 media/base/codec_list.cc        |  14 +--
 media/base/media_engine.cc      | 117 ++++++++++---------
 pc/codec_vendor.cc              |  16 +--
 pc/ice_server_parsing.cc        |  80 ++++++-------
 pc/jsep_transport_controller.cc |  19 ++-
 pc/media_session.cc             |  49 ++++----
 pc/peer_connection.cc           | 149 ++++++++++++------------
 pc/sdp_offer_answer.cc          | 199 ++++++++++++++++----------------
 10 files changed, 320 insertions(+), 340 deletions(-)

diff --git a/api/rtc_error.h b/api/rtc_error.h
index ffc2b17a20..a0a19ae5ba 100644
--- a/api/rtc_error.h
+++ b/api/rtc_error.h
@@ -225,20 +225,6 @@ class RTC_EXPORT RTCError {
   std::optional<uint16_t> sctp_cause_code_;
 };
 
-// Helper macro that can be used by implementations to create an error with a
-// message and log it. `message` should be a string literal or movable
-// std::string.
-#define LOG_AND_RETURN_ERROR_EX(type, message, severity)                     \
-  {                                                                          \
-    RTC_DCHECK(type != RTCErrorType::NONE);                                  \
-    RTC_LOG(severity) << message << " (" << ::webrtc::ToString(type) << ")"; \
-    return ::webrtc::RTCError(type, message);                                \
-  }
-
-// LOG_AND_RETURN_ERROR is a s simpler variant of `LOG_AND_RETURN_ERROR_EX`.
-#define LOG_AND_RETURN_ERROR(type, message) \
-  LOG_AND_RETURN_ERROR_EX(type, message, LS_ERROR)
-
 inline RTCError LogErrorImpl(RTCError error,
                              LoggingSeverity severity,
                              const char* file,
@@ -255,7 +241,6 @@ inline RTCError LogErrorImpl(RTCError error,
   return error;
 }
 
-// A slightly more C++ looking alternative to the LOG_AND_RETURN_ERROR() macro.
 // This approach does not hide the return statement and also allows for
 // constructing/formatting the error string inline.
 //
diff --git a/api/rtc_error_unittest.cc b/api/rtc_error_unittest.cc
index 596bc6413f..7527ae6261 100644
--- a/api/rtc_error_unittest.cc
+++ b/api/rtc_error_unittest.cc
@@ -259,7 +259,7 @@ TEST(RTCErrorOrTest, BuildString) {
   EXPECT_STREQ(error.message(), "StringyBuilder #2");
 }
 
-// Tests a non macro based alternative to `LOG_AND_RETURN_ERROR`.
+// Tests `LOG_ERROR`.
 TEST(RTCErrorOrTest, BuildStringLog) {
   class LogSinkImpl : public LogSink {
    public:
diff --git a/media/base/codec_list.cc b/media/base/codec_list.cc
index c82b7209ef..4be60c4776 100644
--- a/media/base/codec_list.cc
+++ b/media/base/codec_list.cc
@@ -38,8 +38,8 @@ RTCError CheckInputConsistency(const std::vector<Codec>& codecs) {
         RTC_LOG(LS_ERROR) << "Duplicate payload type in codec list, " << codec
                           << " and " << codecs[it->second]
                           << " have the same ID";
-        LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
-                             "Duplicate payload type in codec list");
+        return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                         << "Duplicate payload type in codec list");
       }
     }
   }
@@ -74,8 +74,8 @@ RTCError CheckInputConsistency(const std::vector<Codec>& codecs) {
         if (!(FromString(apt_it->second, &associated_pt))) {
           RTC_LOG(LS_ERROR) << "Non-numeric argument to rtx apt: " << codec
                             << " apt=" << apt_it->second;
-          LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
-                               "Non-numeric argument to rtx apt parameter");
+          return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                           << "Non-numeric argument to rtx apt parameter");
         }
         if (codec.id != Codec::kIdNotSet &&
             pt_to_index.count(associated_pt) != 1) {
@@ -83,9 +83,9 @@ RTCError CheckInputConsistency(const std::vector<Codec>& codecs) {
               << "Surprising condition: RTX codec APT not found: " << codec
               << " points to a PT that occurs "
               << pt_to_index.count(associated_pt) << " times";
-          LOG_AND_RETURN_ERROR(
-              RTCErrorType::INVALID_PARAMETER,
-              "PT pointed to by rtx apt parameter does not exist");
+          return LOG_ERROR(
+              RTCError(RTCErrorType::INVALID_PARAMETER)
+              << "PT pointed to by rtx apt parameter does not exist");
         }
         // const Codec& referred_codec = codecs[pt_to_index[associated_pt]];
         // Not true:
diff --git a/media/base/media_engine.cc b/media/base/media_engine.cc
index 78bb62af9b..73a87101c0 100644
--- a/media/base/media_engine.cc
+++ b/media/base/media_engine.cc
@@ -111,10 +111,9 @@ RTCError CheckScalabilityModeValues(const RtpParameters& rtp_parameters,
         }
       }
       if (!codecFound) {
-        LOG_AND_RETURN_ERROR(
-            RTCErrorType::INVALID_MODIFICATION,
-            "Attempted to use an unsupported codec for layer " +
-                std::to_string(i));
+        return LOG_ERROR(RTCError(RTCErrorType::INVALID_MODIFICATION)
+                         << "Attempted to use an unsupported codec for layer "
+                         << i);
       }
     }
     if (rtp_parameters.encodings[i].scalability_mode) {
@@ -133,10 +132,10 @@ RTCError CheckScalabilityModeValues(const RtpParameters& rtp_parameters,
         }
 
         if (!scalabilityModeFound) {
-          LOG_AND_RETURN_ERROR(
-              RTCErrorType::INVALID_MODIFICATION,
-              "Attempted to set RtpParameters scalabilityMode "
-              "to an unsupported value for the current codecs.");
+          return LOG_ERROR(
+              RTCError(RTCErrorType::INVALID_MODIFICATION)
+              << "Attempted to set RtpParameters scalabilityMode "
+                 "to an unsupported value for the current codecs.");
         }
       } else {
         bool scalabilityModeFound = false;
@@ -148,10 +147,10 @@ RTCError CheckScalabilityModeValues(const RtpParameters& rtp_parameters,
           }
         }
         if (!scalabilityModeFound) {
-          LOG_AND_RETURN_ERROR(
-              RTCErrorType::INVALID_MODIFICATION,
-              "Attempted to set RtpParameters scalabilityMode "
-              "to an unsupported value for the current codecs.");
+          return LOG_ERROR(
+              RTCError(RTCErrorType::INVALID_MODIFICATION)
+              << "Attempted to set RtpParameters scalabilityMode "
+                 "to an unsupported value for the current codecs.");
         }
       }
     }
@@ -167,39 +166,39 @@ RTCError CheckRtpParametersValues(const RtpParameters& rtp_parameters,
   bool has_scale_resolution_down_to = false;
   for (size_t i = 0; i < rtp_parameters.encodings.size(); ++i) {
     if (rtp_parameters.encodings[i].bitrate_priority <= 0) {
-      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_RANGE,
-                           "Attempted to set RtpParameters bitrate_priority to "
-                           "an invalid number. bitrate_priority must be > 0.");
+      return LOG_ERROR(RTCError(RTCErrorType::INVALID_RANGE)
+                       << "Attempted to set RtpParameters bitrate_priority to "
+                          "an invalid number. bitrate_priority must be > 0.");
     }
     if (rtp_parameters.encodings[i].scale_resolution_down_by &&
         *rtp_parameters.encodings[i].scale_resolution_down_by < 1.0) {
-      LOG_AND_RETURN_ERROR(
-          RTCErrorType::INVALID_RANGE,
-          "Attempted to set RtpParameters scale_resolution_down_by to an "
-          "invalid value. scale_resolution_down_by must be >= 1.0");
+      return LOG_ERROR(
+          RTCError(RTCErrorType::INVALID_RANGE)
+          << "Attempted to set RtpParameters scale_resolution_down_by to an "
+             "invalid value. scale_resolution_down_by must be >= 1.0");
     }
     if (rtp_parameters.encodings[i].max_framerate &&
         *rtp_parameters.encodings[i].max_framerate < 0.0) {
-      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_RANGE,
-                           "Attempted to set RtpParameters max_framerate to an "
-                           "invalid value. max_framerate must be >= 0.0");
+      return LOG_ERROR(RTCError(RTCErrorType::INVALID_RANGE)
+                       << "Attempted to set RtpParameters max_framerate to an "
+                          "invalid value. max_framerate must be >= 0.0");
     }
     if (rtp_parameters.encodings[i].min_bitrate_bps &&
         rtp_parameters.encodings[i].max_bitrate_bps) {
       if (*rtp_parameters.encodings[i].max_bitrate_bps <
           *rtp_parameters.encodings[i].min_bitrate_bps) {
-        LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_RANGE,
-                             "Attempted to set RtpParameters min bitrate "
-                             "larger than max bitrate.");
+        return LOG_ERROR(RTCError(RTCErrorType::INVALID_RANGE)
+                         << "Attempted to set RtpParameters min bitrate "
+                            "larger than max bitrate.");
       }
     }
     if (rtp_parameters.encodings[i].num_temporal_layers) {
       if (*rtp_parameters.encodings[i].num_temporal_layers < 1 ||
           *rtp_parameters.encodings[i].num_temporal_layers >
               kMaxTemporalStreams) {
-        LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_RANGE,
-                             "Attempted to set RtpParameters "
-                             "num_temporal_layers to an invalid number.");
+        return LOG_ERROR(RTCError(RTCErrorType::INVALID_RANGE)
+                         << "Attempted to set RtpParameters "
+                            "num_temporal_layers to an invalid number.");
       }
     }
 
@@ -207,32 +206,32 @@ RTCError CheckRtpParametersValues(const RtpParameters& rtp_parameters,
       has_scale_resolution_down_to = true;
       if (rtp_parameters.encodings[i].scale_resolution_down_to->width <= 0 ||
           rtp_parameters.encodings[i].scale_resolution_down_to->height <= 0) {
-        LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
-                             "The resolution dimensions must be positive.");
+        return LOG_ERROR(RTCError(RTCErrorType::INVALID_MODIFICATION)
+                         << "The resolution dimensions must be positive.");
       }
     }
 
     if (!field_trials.IsEnabled("WebRTC-MixedCodecSimulcast")) {
       if (i > 0 && rtp_parameters.encodings[i - 1].codec !=
                        rtp_parameters.encodings[i].codec) {
-        LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
-                             "Attempted to use different codec values for "
-                             "different encodings.");
+        return LOG_ERROR(RTCError(RTCErrorType::UNSUPPORTED_OPERATION)
+                         << "Attempted to use different codec values for "
+                            "different encodings.");
       }
     }
 
     if (rtp_parameters.encodings[i].csrcs.has_value() &&
         rtp_parameters.encodings[i].csrcs.value().size() > kRtpCsrcSize) {
-      LOG_AND_RETURN_ERROR(
-          RTCErrorType::INVALID_RANGE,
-          "Attempted to set more than the maximum allowed number of CSRCs.")
+      return LOG_ERROR(
+          RTCError(RTCErrorType::INVALID_RANGE)
+          << "Attempted to set more than the maximum allowed number of CSRCs.");
     }
 
     if (i > 0 && rtp_parameters.encodings[i - 1].csrcs !=
                      rtp_parameters.encodings[i].csrcs) {
-      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
-                           "Attempted to set different CSRCs for different "
-                           "encodings.");
+      return LOG_ERROR(RTCError(RTCErrorType::INVALID_MODIFICATION)
+                       << "Attempted to set different CSRCs for different "
+                          "encodings.");
     }
   }
 
@@ -242,9 +241,9 @@ RTCError CheckRtpParametersValues(const RtpParameters& rtp_parameters,
                        return encoding.active &&
                               !encoding.scale_resolution_down_to.has_value();
                      })) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
-                         "If a resolution is specified on any encoding then "
-                         "it must be specified on all encodings.");
+    return LOG_ERROR(RTCError(RTCErrorType::INVALID_MODIFICATION)
+                     << "If a resolution is specified on any encoding then "
+                        "it must be specified on all encodings.");
   }
 
   // In a mixed codec scenario, we only support scalability modes without
@@ -258,10 +257,10 @@ RTCError CheckRtpParametersValues(const RtpParameters& rtp_parameters,
       auto num_spatial_layers =
           ScalabilityModeStringToNumSpatialLayers(*scalability_mode);
       if (num_spatial_layers && *num_spatial_layers > 1) {
-        LOG_AND_RETURN_ERROR(
-            RTCErrorType::UNSUPPORTED_OPERATION,
-            "Attempted to use a scalabilityMode with spatial layers in "
-            "a mixed codec scenario.");
+        return LOG_ERROR(
+            RTCError(RTCErrorType::UNSUPPORTED_OPERATION)
+            << "Attempted to use a scalabilityMode with spatial layers in "
+               "a mixed codec scenario.");
       }
     }
   }
@@ -284,36 +283,36 @@ RTCError CheckRtpParametersInvalidModificationAndValues(
     std::optional<Codec> send_codec,
     const FieldTrialsView& field_trials) {
   if (rtp_parameters.encodings.size() != old_rtp_parameters.encodings.size()) {
-    LOG_AND_RETURN_ERROR(
-        RTCErrorType::INVALID_MODIFICATION,
-        "Attempted to set RtpParameters with different encoding count");
+    return LOG_ERROR(
+        RTCError(RTCErrorType::INVALID_MODIFICATION)
+        << "Attempted to set RtpParameters with different encoding count");
   }
   if (rtp_parameters.rtcp != old_rtp_parameters.rtcp) {
-    LOG_AND_RETURN_ERROR(
-        RTCErrorType::INVALID_MODIFICATION,
-        "Attempted to set RtpParameters with modified RTCP parameters");
+    return LOG_ERROR(
+        RTCError(RTCErrorType::INVALID_MODIFICATION)
+        << "Attempted to set RtpParameters with modified RTCP parameters");
   }
   if (rtp_parameters.header_extensions !=
       old_rtp_parameters.header_extensions) {
-    LOG_AND_RETURN_ERROR(
-        RTCErrorType::INVALID_MODIFICATION,
-        "Attempted to set RtpParameters with modified header extensions");
+    return LOG_ERROR(
+        RTCError(RTCErrorType::INVALID_MODIFICATION)
+        << "Attempted to set RtpParameters with modified header extensions");
   }
   if (!absl::c_equal(old_rtp_parameters.encodings, rtp_parameters.encodings,
                      [](const RtpEncodingParameters& encoding1,
                         const RtpEncodingParameters& encoding2) {
                        return encoding1.rid == encoding2.rid;
                      })) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
-                         "Attempted to change RID values in the encodings.");
+    return LOG_ERROR(RTCError(RTCErrorType::INVALID_MODIFICATION)
+                     << "Attempted to change RID values in the encodings.");
   }
   if (!absl::c_equal(old_rtp_parameters.encodings, rtp_parameters.encodings,
                      [](const RtpEncodingParameters& encoding1,
                         const RtpEncodingParameters& encoding2) {
                        return encoding1.ssrc == encoding2.ssrc;
                      })) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
-                         "Attempted to set RtpParameters with modified SSRC");
+    return LOG_ERROR(RTCError(RTCErrorType::INVALID_MODIFICATION)
+                     << "Attempted to set RtpParameters with modified SSRC");
   }
 
   return CheckRtpParametersValues(rtp_parameters, send_codecs, send_codec,
diff --git a/pc/codec_vendor.cc b/pc/codec_vendor.cc
index 50f3ef818f..2adc3a62e0 100644
--- a/pc/codec_vendor.cc
+++ b/pc/codec_vendor.cc
@@ -660,10 +660,10 @@ RTCErrorOr<std::vector<Codec>> CodecVendor::GetNegotiatedCodecsForOffer(
         if (!IsMediaContentOfType(current_content,
                                   media_description_options.type)) {
           // Can happen if the remote side re-uses a MID while recycling.
-          LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
-                               "Media type for content with mid='" +
-                                   current_content->mid() +
-                                   "' does not match previous type.");
+          return LOG_ERROR(RTCError(RTCErrorType::INTERNAL_ERROR)
+                           << "Media type for content with mid='"
+                           << current_content->mid()
+                           << "' does not match previous type.");
         }
         const MediaContentDescription* mcd =
             current_content->media_description();
@@ -796,10 +796,10 @@ RTCErrorOr<Codecs> CodecVendor::GetNegotiatedCodecsForAnswer(
         if (!IsMediaContentOfType(current_content,
                                   media_description_options.type)) {
           // Can happen if the remote side re-uses a MID while recycling.
-          LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
-                               "Media type for content with mid='" +
-                                   current_content->mid() +
-                                   "' does not match previous type.");
+          return LOG_ERROR(RTCError(RTCErrorType::INTERNAL_ERROR)
+                           << "Media type for content with mid='"
+                           << current_content->mid()
+                           << "' does not match previous type.");
         }
         const MediaContentDescription* mcd =
             current_content->media_description();
diff --git a/pc/ice_server_parsing.cc b/pc/ice_server_parsing.cc
index b83cc86783..e2b87268fc 100644
--- a/pc/ice_server_parsing.cc
+++ b/pc/ice_server_parsing.cc
@@ -194,22 +194,21 @@ RTCError ParseIceServerUrl(const PeerConnectionInterface::IceServer& server,
   if (tokens.size() == kTurnTransportTokensNum) {  // ?transport= is present.
     std::vector<absl::string_view> transport_tokens = split(tokens[1], '=');
     if (transport_tokens[0] != kTransport) {
-      LOG_AND_RETURN_ERROR(
-          RTCErrorType::SYNTAX_ERROR,
-          "ICE server parsing failed: Invalid transport parameter key.");
+      return LOG_ERROR(
+          RTCError(RTCErrorType::SYNTAX_ERROR)
+          << "ICE server parsing failed: Invalid transport parameter key.");
     }
     if (transport_tokens.size() < 2) {
-      LOG_AND_RETURN_ERROR(
-          RTCErrorType::SYNTAX_ERROR,
-          "ICE server parsing failed: Transport parameter missing value.");
+      return LOG_ERROR(
+          RTCError(RTCErrorType::SYNTAX_ERROR)
+          << "ICE server parsing failed: Transport parameter missing value.");
     }
 
     std::optional<ProtocolType> proto = StringToProto(transport_tokens[1]);
     if (!proto || (*proto != PROTO_UDP && *proto != PROTO_TCP)) {
-      LOG_AND_RETURN_ERROR(
-          RTCErrorType::SYNTAX_ERROR,
-          "ICE server parsing failed: Transport parameter should "
-          "always be udp or tcp.");
+      return LOG_ERROR(RTCError(RTCErrorType::SYNTAX_ERROR)
+                       << "ICE server parsing failed: Transport parameter "
+                          "should always be udp or tcp.");
     }
     turn_transport_type = *proto;
     transport_given_explicitly = true;
@@ -219,9 +218,9 @@ RTCError ParseIceServerUrl(const PeerConnectionInterface::IceServer& server,
       GetServiceTypeAndHostnameFromUri(uri_without_transport);
   if (service_type == ServiceType::INVALID) {
     RTC_LOG(LS_ERROR) << "Invalid transport parameter in ICE URI: " << url;
-    LOG_AND_RETURN_ERROR(
-        RTCErrorType::SYNTAX_ERROR,
-        "ICE server parsing failed: Invalid transport parameter in ICE URI");
+    return LOG_ERROR(
+        RTCError(RTCErrorType::SYNTAX_ERROR)
+        << "ICE server parsing failed: Invalid transport parameter in ICE URI");
   }
 
   // GetServiceTypeAndHostnameFromUri should never give an empty hoststring
@@ -231,9 +230,9 @@ RTCError ParseIceServerUrl(const PeerConnectionInterface::IceServer& server,
   if ((service_type == ServiceType::STUN ||
        service_type == ServiceType::STUNS) &&
       tokens.size() > 1) {
-    LOG_AND_RETURN_ERROR(
-        RTCErrorType::SYNTAX_ERROR,
-        "ICE server parsing failed: Invalid stun url with query parameters");
+    return LOG_ERROR(
+        RTCError(RTCErrorType::SYNTAX_ERROR)
+        << "ICE server parsing failed: Invalid stun url with query parameters");
   }
 
   int default_port = kDefaultStunPort;
@@ -252,23 +251,23 @@ RTCError ParseIceServerUrl(const PeerConnectionInterface::IceServer& server,
   if (hoststring.find('@') != absl::string_view::npos) {
     RTC_LOG(LS_ERROR) << "Invalid url with long deprecated user@host syntax: "
                       << uri_without_transport;
-    LOG_AND_RETURN_ERROR(RTCErrorType::SYNTAX_ERROR,
-                         "ICE server parsing failed: Invalid url with long "
-                         "deprecated user@host syntax");
+    return LOG_ERROR(RTCError(RTCErrorType::SYNTAX_ERROR)
+                     << "ICE server parsing failed: Invalid url with long "
+                        "deprecated user@host syntax");
   }
 
   auto [success, address, port] =
       ParseHostnameAndPortFromString(hoststring, default_port);
   if (!success) {
     RTC_LOG(LS_ERROR) << "Invalid hostname format: " << uri_without_transport;
-    LOG_AND_RETURN_ERROR(RTCErrorType::SYNTAX_ERROR,
-                         "ICE server parsing failed: Invalid hostname format");
+    return LOG_ERROR(RTCError(RTCErrorType::SYNTAX_ERROR)
+                     << "ICE server parsing failed: Invalid hostname format");
   }
 
   if (port <= 0 || port > 0xffff) {
     RTC_LOG(LS_ERROR) << "Invalid port: " << port;
-    LOG_AND_RETURN_ERROR(RTCErrorType::SYNTAX_ERROR,
-                         "ICE server parsing failed: Invalid port");
+    return LOG_ERROR(RTCError(RTCErrorType::SYNTAX_ERROR)
+                     << "ICE server parsing failed: Invalid port");
   }
 
   switch (service_type) {
@@ -281,16 +280,15 @@ RTCError ParseIceServerUrl(const PeerConnectionInterface::IceServer& server,
       if (server.username.empty() || server.password.empty()) {
         // The WebRTC spec requires throwing an InvalidAccessError when username
         // or credential are ommitted; this is the native equivalent.
-        LOG_AND_RETURN_ERROR(
-            RTCErrorType::INVALID_PARAMETER,
-            "ICE server parsing failed: TURN server with empty "
-            "username or password");
+        return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                         << "ICE server parsing failed: TURN server with empty "
+                            "username or password");
       }
       // RFC 8489 limits the size of the STUN username field to 509 characters.
       if (server.username.size() > kMaxTurnUsernameLength) {
-        LOG_AND_RETURN_ERROR(
-            RTCErrorType::INVALID_PARAMETER,
-            "ICE server parsing failed: TURN server username is too long");
+        return LOG_ERROR(
+            RTCError(RTCErrorType::INVALID_PARAMETER)
+            << "ICE server parsing failed: TURN server username is too long");
       }
       // If the hostname field is not empty, then the server address must be
       // the resolved IP for that host, the hostname is needed later for TLS
@@ -303,11 +301,10 @@ RTCError ParseIceServerUrl(const PeerConnectionInterface::IceServer& server,
         if (!IPFromString(address, &ip)) {
           // When hostname is set, the server address must be a
           // resolved ip address.
-          LOG_AND_RETURN_ERROR(
-              RTCErrorType::INVALID_PARAMETER,
-              "ICE server parsing failed: "
-              "IceServer has hostname field set, but URI does not "
-              "contain an IP address.");
+          return LOG_ERROR(
+              RTCError(RTCErrorType::INVALID_PARAMETER)
+              << "ICE server parsing failed: IceServer has hostname field "
+                 "set, but URI does not contain an IP address.");
         }
         socket_address.SetResolvedIP(ip);
       }
@@ -328,9 +325,8 @@ RTCError ParseIceServerUrl(const PeerConnectionInterface::IceServer& server,
     default:
       // We shouldn't get to this point with an invalid service_type, we should
       // have returned an error already.
-      LOG_AND_RETURN_ERROR(
-          RTCErrorType::INTERNAL_ERROR,
-          "ICE server parsing failed: Unexpected service type");
+      return LOG_ERROR(RTCError(RTCErrorType::INTERNAL_ERROR)
+                       << "ICE server parsing failed: Unexpected service type");
   }
   return RTCError::OK();
 }
@@ -345,8 +341,8 @@ RTCError ParseIceServersOrError(
     if (!server.urls.empty()) {
       for (const std::string& url : server.urls) {
         if (url.empty()) {
-          LOG_AND_RETURN_ERROR(RTCErrorType::SYNTAX_ERROR,
-                               "ICE server parsing failed: Empty uri.");
+          return LOG_ERROR(RTCError(RTCErrorType::SYNTAX_ERROR)
+                           << "ICE server parsing failed: Empty uri.");
         }
         RTCError err =
             ParseIceServerUrl(server, url, stun_servers, turn_servers);
@@ -363,8 +359,8 @@ RTCError ParseIceServersOrError(
         return err;
       }
     } else {
-      LOG_AND_RETURN_ERROR(RTCErrorType::SYNTAX_ERROR,
-                           "ICE server parsing failed: Empty uri.");
+      return LOG_ERROR(RTCError(RTCErrorType::SYNTAX_ERROR)
+                       << "ICE server parsing failed: Empty uri.");
     }
   }
   return RTCError::OK();
diff --git a/pc/jsep_transport_controller.cc b/pc/jsep_transport_controller.cc
index 2dd7f4f857..4d2126ebd3 100644
--- a/pc/jsep_transport_controller.cc
+++ b/pc/jsep_transport_controller.cc
@@ -421,8 +421,8 @@ RTCError JsepTransportController::RollbackTransports() {
 RTCError JsepTransportController::RollbackTransports_n() {
   bundles_.Rollback();
   if (!transports_.RollbackTransports()) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
-                         "Failed to roll back transport state.");
+    return LOG_ERROR(RTCError(RTCErrorType::INTERNAL_ERROR)
+                     << "Failed to roll back transport state.");
   }
   return RTCError::OK();
 }
@@ -701,10 +701,9 @@ RTCError JsepTransportController::ApplyDescription_n(
 
     JsepTransport* transport = GetJsepTransportForMid(content_info.mid());
     if (!transport) {
-      LOG_AND_RETURN_ERROR(
-          RTCErrorType::INVALID_PARAMETER,
-          "Could not find transport for m= section with mid='" +
-              content_info.mid() + "'");
+      return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                       << "Could not find transport for m= section with mid='"
+                       << content_info.mid() << "'");
     }
 
     if (established_bundle_group &&
@@ -744,10 +743,10 @@ RTCError JsepTransportController::ApplyDescription_n(
     }
 
     if (!error.ok()) {
-      LOG_AND_RETURN_ERROR(
-          RTCErrorType::INVALID_PARAMETER,
-          "Failed to apply the description for m= section with mid='" +
-              content_info.mid() + "': " + error.message());
+      return LOG_ERROR(
+          RTCError(RTCErrorType::INVALID_PARAMETER)
+          << "Failed to apply the description for m= section with mid='"
+          << content_info.mid() << "': " << error.message());
     }
   }
   if (type == SdpType::kAnswer) {
diff --git a/pc/media_session.cc b/pc/media_session.cc
index 0ff4bef4c1..79bc2f4c5a 100644
--- a/pc/media_session.cc
+++ b/pc/media_session.cc
@@ -400,8 +400,8 @@ RTCError CreateMediaContentOffer(
   if (!AddStreamParams(media_description_options.sender_options,
                        session_options.rtcp_cname, ssrc_generator,
                        current_streams, offer, field_trials)) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
-                         "Failed to add stream parameters");
+    return LOG_ERROR(RTCError(RTCErrorType::INTERNAL_ERROR)
+                     << "Failed to add stream parameters");
   }
 
   return CreateContentOffer(media_description_options, session_options,
@@ -797,9 +797,9 @@ MediaSessionDescriptionFactory::CreateOfferOrError(
     if (!offer_bundle.content_names().empty()) {
       offer->AddGroup(offer_bundle);
       if (!UpdateTransportInfoForBundle(offer_bundle, offer.get())) {
-        LOG_AND_RETURN_ERROR(
-            RTCErrorType::INTERNAL_ERROR,
-            "CreateOffer failed to UpdateTransportInfoForBundle");
+        return LOG_ERROR(
+            RTCError(RTCErrorType::INTERNAL_ERROR)
+            << "CreateOffer failed to UpdateTransportInfoForBundle");
       }
     }
   }
@@ -830,7 +830,8 @@ MediaSessionDescriptionFactory::CreateAnswerOrError(
     const MediaSessionOptions& session_options,
     const SessionDescription* current_description) const {
   if (!offer) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR, "Called without offer.");
+    return LOG_ERROR(RTCError(RTCErrorType::INTERNAL_ERROR)
+                     << "Called without offer.");
   }
 
   // Must have options for exactly as many sections as in the offer.
@@ -989,9 +990,9 @@ MediaSessionDescriptionFactory::CreateAnswerOrError(
         // Share the same ICE credentials and crypto params across all contents,
         // as BUNDLE requires.
         if (!UpdateTransportInfoForBundle(answer_bundle, answer.get())) {
-          LOG_AND_RETURN_ERROR(
-              RTCErrorType::INTERNAL_ERROR,
-              "CreateAnswer failed to UpdateTransportInfoForBundle.");
+          return LOG_ERROR(
+              RTCError(RTCErrorType::INTERNAL_ERROR)
+              << "CreateAnswer failed to UpdateTransportInfoForBundle.");
         }
       }
     }
@@ -1332,9 +1333,9 @@ RTCError MediaSessionDescriptionFactory::AddRtpContentForAnswer(
       media_description_options.transport_options, current_description,
       !offer_content->rejected && bundle_transport == nullptr, ice_credentials);
   if (!transport) {
-    LOG_AND_RETURN_ERROR(
-        RTCErrorType::INTERNAL_ERROR,
-        "Failed to create transport answer, transport is missing");
+    return LOG_ERROR(
+        RTCError(RTCErrorType::INTERNAL_ERROR)
+        << "Failed to create transport answer, transport is missing");
   }
 
   // Pick codecs based on the requested communications direction in the offer
@@ -1387,16 +1388,16 @@ RTCError MediaSessionDescriptionFactory::AddRtpContentForAnswer(
                          ssrc_generator(), current_streams,
                          answer_content.get(),
                          transport_desc_factory_->trials())) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
-                         "Failed to set codecs in answer");
+    return LOG_ERROR(RTCError(RTCErrorType::INTERNAL_ERROR)
+                     << "Failed to set codecs in answer");
   }
   if (!CreateMediaContentAnswer(
           offer_content_description, media_description_options, session_options,
           filtered_rtp_header_extensions(header_extensions), ssrc_generator(),
           enable_encrypted_rtp_header_extensions_, current_streams,
           bundle_enabled, answer_content.get())) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
-                         "Failed to create answer");
+    return LOG_ERROR(RTCError(RTCErrorType::INTERNAL_ERROR)
+                     << "Failed to create answer");
   }
 
   bool secure = bundle_transport ? bundle_transport->description.secure()
@@ -1437,9 +1438,9 @@ RTCError MediaSessionDescriptionFactory::AddDataContentForAnswer(
       media_description_options.transport_options, current_description,
       !offer_content->rejected && bundle_transport == nullptr, ice_credentials);
   if (!data_transport) {
-    LOG_AND_RETURN_ERROR(
-        RTCErrorType::INTERNAL_ERROR,
-        "Failed to create transport answer, data transport is missing");
+    return LOG_ERROR(
+        RTCError(RTCErrorType::INTERNAL_ERROR)
+        << "Failed to create transport answer, data transport is missing");
   }
 
   bool bundle_enabled = offer_description->HasGroup(GROUP_TYPE_BUNDLE) &&
@@ -1469,8 +1470,8 @@ RTCError MediaSessionDescriptionFactory::AddDataContentForAnswer(
             RtpHeaderExtensions(), ssrc_generator(),
             enable_encrypted_rtp_header_extensions_, current_streams,
             bundle_enabled, data_answer.get())) {
-      LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
-                           "Failed to create answer");
+      return LOG_ERROR(RTCError(RTCErrorType::INTERNAL_ERROR)
+                       << "Failed to create answer");
     }
     // Respond with sctpmap if the offer uses sctpmap.
     bool offer_uses_sctpmap = offer_data_description->use_sctpmap();
@@ -1530,9 +1531,9 @@ RTCError MediaSessionDescriptionFactory::AddUnsupportedContentForAnswer(
           !offer_content->rejected && bundle_transport == nullptr,
           ice_credentials);
   if (!unsupported_transport) {
-    LOG_AND_RETURN_ERROR(
-        RTCErrorType::INTERNAL_ERROR,
-        "Failed to create transport answer, unsupported transport is missing");
+    return LOG_ERROR(RTCError(RTCErrorType::INTERNAL_ERROR)
+                     << "Failed to create transport answer, unsupported "
+                        "transport is missing");
   }
   RTC_CHECK(IsMediaContentOfType(offer_content, MediaType::UNSUPPORTED));
 
diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc
index 62704136ef..75b0b0ee38 100644
--- a/pc/peer_connection.cc
+++ b/pc/peer_connection.cc
@@ -276,9 +276,9 @@ RTCError ValidateIceCandidatePoolSize(
   // in new candidates being gathered.
   if (previous_ice_candidate_pool_size.has_value() &&
       ice_candidate_pool_size != previous_ice_candidate_pool_size.value()) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
-                         "Can't change candidate pool size after calling "
-                         "SetLocalDescription.");
+    return LOG_ERROR(RTCError(RTCErrorType::INVALID_MODIFICATION)
+                     << "Can't change candidate pool size after calling "
+                        "SetLocalDescription.");
   }
 
   return RTCError::OK();
@@ -331,8 +331,8 @@ RTCErrorOr<PeerConnectionInterface::RTCConfiguration> ApplyConfiguration(
       configuration.stable_writable_connection_ping_interval_ms;
 
   if (configuration != modified_config) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
-                         "Modifying the configuration in an unsupported way.");
+    return LOG_ERROR(RTCError(RTCErrorType::INVALID_MODIFICATION)
+                     << "Modifying the configuration in an unsupported way.");
   }
 
   RTCError err = IceConfig(modified_config).IsValid();
@@ -956,25 +956,26 @@ RTCErrorOr<scoped_refptr<RtpSenderInterface>> PeerConnection::AddTrack(
   RTC_DCHECK_RUN_ON(signaling_thread());
   TRACE_EVENT0("webrtc", "PeerConnection::AddTrack");
   if (!ConfiguredForMedia()) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
-                         "Not configured for media");
+    return LOG_ERROR(RTCError(RTCErrorType::UNSUPPORTED_OPERATION)
+                     << "Not configured for media");
   }
   if (!track) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, "Track is null.");
+    return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                     << "Track is null.");
   }
   if (!(track->kind() == MediaStreamTrackInterface::kAudioKind ||
         track->kind() == MediaStreamTrackInterface::kVideoKind)) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
-                         "Track has invalid kind: " + track->kind());
+    return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                     << "Track has invalid kind: " << track->kind());
   }
   if (IsClosed()) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
-                         "PeerConnection is closed.");
+    return LOG_ERROR(RTCError(RTCErrorType::INVALID_STATE)
+                     << "PeerConnection is closed.");
   }
   if (rtp_manager()->FindSenderForTrack(track.get())) {
-    LOG_AND_RETURN_ERROR(
-        RTCErrorType::INVALID_PARAMETER,
-        "Sender already exists for track " + track->id() + ".");
+    return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                     << "Sender already exists for track " << track->id()
+                     << ".");
   }
   RTCErrorOr<scoped_refptr<RtpSenderInterface>> sender_or_error;
   if (IsUnifiedPlan()) {
@@ -1000,15 +1001,16 @@ RTCError PeerConnection::RemoveTrackOrError(
     scoped_refptr<RtpSenderInterface> sender) {
   RTC_DCHECK_RUN_ON(signaling_thread());
   if (!ConfiguredForMedia()) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
-                         "Not configured for media");
+    return LOG_ERROR(RTCError(RTCErrorType::UNSUPPORTED_OPERATION)
+                     << "Not configured for media");
   }
   if (!sender) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, "Sender is null.");
+    return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                     << "Sender is null.");
   }
   if (IsClosed()) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
-                         "PeerConnection is closed.");
+    return LOG_ERROR(RTCError(RTCErrorType::INVALID_STATE)
+                     << "PeerConnection is closed.");
   }
   if (IsUnifiedPlan()) {
     auto transceiver = FindTransceiverBySender(sender);
@@ -1038,9 +1040,9 @@ RTCError PeerConnection::RemoveTrackOrError(
     }
     RTC_ALLOW_PLAN_B_DEPRECATION_END();
     if (!removed) {
-      LOG_AND_RETURN_ERROR(
-          RTCErrorType::INVALID_PARAMETER,
-          "Couldn't find sender " + sender->id() + " to remove.");
+      return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                       << "Couldn't find sender " << sender->id()
+                       << " to remove.");
     }
   }
   sdp_handler_->UpdateNegotiationNeeded();
@@ -1056,8 +1058,8 @@ PeerConnection::FindTransceiverBySender(
 RTCErrorOr<scoped_refptr<RtpTransceiverInterface>>
 PeerConnection::AddTransceiver(scoped_refptr<MediaStreamTrackInterface> track) {
   if (!ConfiguredForMedia()) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
-                         "Not configured for media");
+    return LOG_ERROR(RTCError(RTCErrorType::UNSUPPORTED_OPERATION)
+                     << "Not configured for media");
   }
 
   return AddTransceiver(track, RtpTransceiverInit());
@@ -1068,13 +1070,14 @@ PeerConnection::AddTransceiver(scoped_refptr<MediaStreamTrackInterface> track,
                                const RtpTransceiverInit& init) {
   RTC_DCHECK_RUN_ON(signaling_thread());
   if (!ConfiguredForMedia()) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
-                         "Not configured for media");
+    return LOG_ERROR(RTCError(RTCErrorType::UNSUPPORTED_OPERATION)
+                     << "Not configured for media");
   }
   RTC_CHECK(IsUnifiedPlan())
       << "AddTransceiver is only available with Unified Plan SdpSemantics";
   if (!track) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, "track is null");
+    return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                     << "track is null");
   }
   MediaType media_type;
   if (track->kind() == MediaStreamTrackInterface::kAudioKind) {
@@ -1082,8 +1085,8 @@ PeerConnection::AddTransceiver(scoped_refptr<MediaStreamTrackInterface> track,
   } else if (track->kind() == MediaStreamTrackInterface::kVideoKind) {
     media_type = MediaType::VIDEO;
   } else {
-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
-                         "Track kind is not audio or video");
+    return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                     << "Track kind is not audio or video");
   }
   return AddTransceiver(media_type, track, init);
 }
@@ -1098,14 +1101,14 @@ PeerConnection::AddTransceiver(MediaType media_type,
                                const RtpTransceiverInit& init) {
   RTC_DCHECK_RUN_ON(signaling_thread());
   if (!ConfiguredForMedia()) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
-                         "Not configured for media");
+    return LOG_ERROR(RTCError(RTCErrorType::UNSUPPORTED_OPERATION)
+                     << "Not configured for media");
   }
   RTC_CHECK(IsUnifiedPlan())
       << "AddTransceiver is only available with Unified Plan SdpSemantics";
   if (!(media_type == MediaType::AUDIO || media_type == MediaType::VIDEO)) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
-                         "media type is not audio or video");
+    return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                     << "media type is not audio or video");
   }
   return AddTransceiver(media_type, nullptr, init);
 }
@@ -1117,8 +1120,8 @@ PeerConnection::AddTransceiver(MediaType media_type,
                                bool update_negotiation_needed) {
   RTC_DCHECK_RUN_ON(signaling_thread());
   if (!ConfiguredForMedia()) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
-                         "Not configured for media");
+    return LOG_ERROR(RTCError(RTCErrorType::UNSUPPORTED_OPERATION)
+                     << "Not configured for media");
   }
   RTC_DCHECK(
       (media_type == MediaType::AUDIO || media_type == MediaType::VIDEO));
@@ -1134,26 +1137,26 @@ PeerConnection::AddTransceiver(MediaType media_type,
                                        return !encoding.rid.empty();
                                      });
   if (num_rids > 0 && num_rids != init.send_encodings.size()) {
-    LOG_AND_RETURN_ERROR(
-        RTCErrorType::INVALID_PARAMETER,
-        "RIDs must be provided for either all or none of the send encodings.");
+    return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                     << "RIDs must be provided for either all or none of the "
+                        "send encodings.");
   }
 
   if (num_rids > 0 && absl::c_any_of(init.send_encodings,
                                      [](const RtpEncodingParameters& encoding) {
                                        return !IsLegalRsidName(encoding.rid);
                                      })) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
-                         "Invalid RID value provided.");
+    return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                     << "Invalid RID value provided.");
   }
 
   if (absl::c_any_of(init.send_encodings,
                      [](const RtpEncodingParameters& encoding) {
                        return encoding.ssrc.has_value();
                      })) {
-    LOG_AND_RETURN_ERROR(
-        RTCErrorType::UNSUPPORTED_PARAMETER,
-        "Attempted to set an unimplemented parameter of RtpParameters.");
+    return LOG_ERROR(
+        RTCError(RTCErrorType::UNSUPPORTED_PARAMETER)
+        << "Attempted to set an unimplemented parameter of RtpParameters.");
   }
 
   RtpParameters parameters;
@@ -1189,9 +1192,9 @@ PeerConnection::AddTransceiver(MediaType media_type,
   }
 
   if (UnimplementedRtpParameterHasValue(parameters)) {
-    LOG_AND_RETURN_ERROR(
-        RTCErrorType::UNSUPPORTED_PARAMETER,
-        "Attempted to set an unimplemented parameter of RtpParameters.");
+    return LOG_ERROR(
+        RTCError(RTCErrorType::UNSUPPORTED_PARAMETER)
+        << "Attempted to set an unimplemented parameter of RtpParameters.");
   }
 
   std::vector<Codec> codecs;
@@ -1210,7 +1213,7 @@ PeerConnection::AddTransceiver(MediaType media_type,
     if (result.type() == RTCErrorType::INVALID_MODIFICATION) {
       result.set_type(RTCErrorType::UNSUPPORTED_OPERATION);
     }
-    LOG_AND_RETURN_ERROR(result.type(), result.message());
+    return LOG_ERROR(RTCError(result.type()) << result.message());
   }
 
   RTC_LOG(LS_INFO) << "Adding " << MediaTypeToString(media_type)
@@ -1486,8 +1489,8 @@ PeerConnection::CreateDataChannelOrError(const std::string& label,
   TRACE_EVENT0("webrtc", "PeerConnection::CreateDataChannel");
 
   if (IsClosed()) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
-                         "CreateDataChannelOrError: PeerConnection is closed.");
+    return LOG_ERROR(RTCError(RTCErrorType::INVALID_STATE)
+                     << "CreateDataChannelOrError: PeerConnection is closed.");
   }
 
   bool first_datachannel = !data_channel_controller_.HasUsedDataChannels();
@@ -1615,8 +1618,8 @@ RTCError PeerConnection::SetConfiguration(
   RTC_DCHECK_RUN_ON(signaling_thread());
   TRACE_EVENT0("webrtc", "PeerConnection::SetConfiguration");
   if (IsClosed()) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
-                         "SetConfiguration: PeerConnection is closed.");
+    return LOG_ERROR(RTCError(RTCErrorType::INVALID_STATE)
+                     << "SetConfiguration: PeerConnection is closed.");
   }
 
   const bool has_local_description = local_description() != nullptr;
@@ -1632,9 +1635,9 @@ RTCError PeerConnection::SetConfiguration(
 
   if (has_local_description &&
       configuration.crypto_options != configuration_.crypto_options) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
-                         "Can't change crypto_options after calling "
-                         "SetLocalDescription.");
+    return LOG_ERROR(RTCError(RTCErrorType::INVALID_MODIFICATION)
+                     << "Can't change crypto_options after calling "
+                        "SetLocalDescription.");
   }
 
   // Create a new, configuration object whose peerconnection config
@@ -1688,8 +1691,8 @@ RTCError PeerConnection::SetConfiguration(
                 modified_config.stun_candidate_keepalive_interval,
                 has_local_description);
           })) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
-                         "Failed to apply configuration to PortAllocator.");
+    return LOG_ERROR(RTCError(RTCErrorType::INTERNAL_ERROR)
+                     << "Failed to apply configuration to PortAllocator.");
   }
 
   configuration_ = modified_config;
@@ -1725,36 +1728,36 @@ RTCError PeerConnection::SetBitrate(const BitrateSettings& bitrate) {
   RTC_DCHECK_RUN_ON(worker_thread());
 
   if (!call_) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
-                         "PeerConnection is closed.");
+    return LOG_ERROR(RTCError(RTCErrorType::INVALID_STATE)
+                     << "PeerConnection is closed.");
   }
 
   const bool has_min = bitrate.min_bitrate_bps.has_value();
   const bool has_start = bitrate.start_bitrate_bps.has_value();
   const bool has_max = bitrate.max_bitrate_bps.has_value();
   if (has_min && *bitrate.min_bitrate_bps < 0) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
-                         "min_bitrate_bps <= 0");
+    return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                     << "min_bitrate_bps <= 0");
   }
   if (has_start) {
     if (has_min && *bitrate.start_bitrate_bps < *bitrate.min_bitrate_bps) {
-      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
-                           "start_bitrate_bps < min_bitrate_bps");
+      return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                       << "start_bitrate_bps < min_bitrate_bps");
     } else if (*bitrate.start_bitrate_bps < 0) {
-      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
-                           "curent_bitrate_bps < 0");
+      return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                       << "curent_bitrate_bps < 0");
     }
   }
   if (has_max) {
     if (has_start && *bitrate.max_bitrate_bps < *bitrate.start_bitrate_bps) {
-      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
-                           "max_bitrate_bps < start_bitrate_bps");
+      return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                       << "max_bitrate_bps < start_bitrate_bps");
     } else if (has_min && *bitrate.max_bitrate_bps < *bitrate.min_bitrate_bps) {
-      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
-                           "max_bitrate_bps < min_bitrate_bps");
+      return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                       << "max_bitrate_bps < min_bitrate_bps");
     } else if (*bitrate.max_bitrate_bps < 0) {
-      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
-                           "max_bitrate_bps < 0");
+      return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                       << "max_bitrate_bps < 0");
     }
   }
 
@@ -3121,8 +3124,8 @@ RTCError PeerConnection::StartSctpTransport(const SctpOptions& options) {
         transport_controller_n()->GetSctpTransport(mid);
     RTC_DCHECK(sctp_transport);
     if (!sctp_transport || !sctp_transport->Start(options)) {
-      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
-                           "Failed to start SCTP transport.");
+      return LOG_ERROR(RTCError(RTCErrorType::INVALID_MODIFICATION)
+                       << "Failed to start SCTP transport.");
     }
     return RTCError::OK();
   });
diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc
index a182232dfd..e4e996fa48 100644
--- a/pc/sdp_offer_answer.cc
+++ b/pc/sdp_offer_answer.cc
@@ -370,13 +370,13 @@ RTCError VerifyDirectionsInAnswer(const SessionDescription* local_offer,
 
     if (!RtpTransceiverDirectionHasRecv(local_direction) &&
         RtpTransceiverDirectionHasSend(remote_direction)) {
-      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
-                           "Incompatible receive direction");
+      return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                       << "Incompatible receive direction");
     }
     if (!RtpTransceiverDirectionHasSend(local_direction) &&
         RtpTransceiverDirectionHasRecv(remote_direction)) {
-      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
-                           "Incompatible send direction");
+      return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                       << "Incompatible send direction");
     }
   }
   return RTCError::OK();
@@ -413,15 +413,17 @@ RTCError VerifyCrypto(
     const TransportInfo* tinfo = desc->GetTransportInfoByName(mid);
     if (!media || !tinfo) {
       // Something is not right.
-      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, kInvalidSdp);
+      return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                       << kInvalidSdp);
     }
     if (dtls_enabled) {
       if (!tinfo->description.identity_fingerprint) {
-        LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
-                             kSdpWithoutDtlsFingerprint);
+        return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                         << kSdpWithoutDtlsFingerprint);
       }
     } else {
-      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, kSdpWithoutCrypto);
+      return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                       << kSdpWithoutCrypto);
     }
   }
   return RTCError::OK();
@@ -469,17 +471,17 @@ RTCError ValidateMids(const SessionDescription& description) {
   flat_set<absl::string_view> mids;
   for (const ContentInfo& content : description.contents()) {
     if (content.mid().empty()) {
-      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
-                           "A media section is missing a MID attribute.");
+      return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                       << "A media section is missing a MID attribute.");
     }
     if (content.mid().size() > kMidMaxSize) {
-      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
-                           "The MID attribute exceeds the maximum supported "
-                           "length of 16 characters.");
+      return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                       << "The MID attribute exceeds the maximum supported "
+                          "length of 16 characters.");
     }
     if (!mids.insert(content.mid()).second) {
-      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
-                           "Duplicate a=mid value '" + content.mid() + "'.");
+      return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                       << "Duplicate a=mid value '" << content.mid() << "'.");
     }
   }
   return RTCError::OK();
@@ -499,7 +501,7 @@ RTCError FindDuplicateCodecParameters(
        << ". All codecs must share the same type, "
           "encoding name, clock rate and parameters.";
 
-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, sb.Release());
+    return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER) << sb.Release());
   }
   payload_to_codec_parameters.insert(
       std::make_pair(codec_parameters.payload_type, codec_parameters));
@@ -520,9 +522,9 @@ RTCError ValidateBundledPayloadTypes(const SessionDescription& description) {
       const ContentInfo* content_description =
           description.GetContentByName(content_name);
       if (!content_description) {
-        LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
-                             "A BUNDLE group contains a MID='" + content_name +
-                                 "' matching no m= section.");
+        return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                         << "A BUNDLE group contains a MID='" << content_name
+                         << "' matching no m= section.");
       }
       const MediaContentDescription* media_description =
           content_description->media_description();
@@ -553,12 +555,12 @@ RTCError FindDuplicateHeaderExtensionIds(
   if (existing_extension != id_to_extension.end() &&
       !(extension.uri == existing_extension->second.uri &&
         extension.encrypt == existing_extension->second.encrypt)) {
-    LOG_AND_RETURN_ERROR(
-        RTCErrorType::INVALID_PARAMETER,
-        "A BUNDLE group contains a codec collision for "
-        "header extension id=" +
-            absl::StrCat(extension.id) +
-            ". The id must be the same across all bundled media descriptions");
+    return LOG_ERROR(
+        RTCError(RTCErrorType::INVALID_PARAMETER)
+        << "A BUNDLE group contains a codec collision for "
+           "header extension id="
+        << extension.id
+        << ". The id must be the same across all bundled media descriptions");
   }
   id_to_extension.insert(std::make_pair(extension.id, extension));
   return RTCError::OK();
@@ -577,9 +579,9 @@ RTCError ValidateBundledRtpHeaderExtensions(
       const ContentInfo* content_description =
           description.GetContentByName(content_name);
       if (!content_description) {
-        LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
-                             "A BUNDLE group contains a MID='" + content_name +
-                                 "' matching no m= section.");
+        return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                         << "A BUNDLE group contains a MID='" << content_name
+                         << "' matching no m= section.");
       }
       const MediaContentDescription* media_description =
           content_description->media_description();
@@ -616,10 +618,10 @@ RTCError ValidateRtpHeaderExtensionsForSpecSimulcast(
       return ext.uri == RtpExtension::kRidUri;
     });
     if (it == extensions.end()) {
-      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
-                           "The media section with MID='" + content.mid() +
-                               "' negotiates simulcast but does not negotiate "
-                               "the RID RTP header extension.");
+      return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                       << "The media section with MID='" << content.mid()
+                       << "' negotiates simulcast but does not negotiate "
+                          "the RID RTP header extension.");
     }
   }
   return RTCError::OK();
@@ -640,11 +642,11 @@ RTCError ValidateSsrcGroups(const SessionDescription& description) {
              group.ssrcs.size() != 2) ||
             (group.semantics == kSimSsrcGroupSemantics &&
              group.ssrcs.size() > kMaxSimulcastStreams)) {
-          LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
-                               "The media section with MID='" + content.mid() +
-                                   "' has a ssrc-group with semantics " +
-                                   group.semantics +
-                                   " and an unexpected number of SSRCs.");
+          return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                           << "The media section with MID='" << content.mid()
+                           << "' has a ssrc-group with semantics "
+                           << group.semantics
+                           << " and an unexpected number of SSRCs.");
         }
       }
     }
@@ -667,12 +669,12 @@ RTCError ValidatePayloadTypes(const SessionDescription& description) {
     if (type == MediaType::AUDIO || type == MediaType::VIDEO) {
       for (const auto& codec : media_description->codecs()) {
         if (!PayloadType::IsValid(codec.id, media_description->rtcp_mux())) {
-          LOG_AND_RETURN_ERROR(
-              RTCErrorType::INVALID_PARAMETER,
-              "The media section with MID='" + content.mid() +
-                  "' used an invalid payload type " + absl::StrCat(codec.id) +
-                  " for codec '" + codec.name + ", rtcp-mux:" +
-                  (media_description->rtcp_mux() ? "enabled" : "disabled"));
+          return LOG_ERROR(
+              RTCError(RTCErrorType::INVALID_PARAMETER)
+              << "The media section with MID='" << content.mid()
+              << "' used an invalid payload type " << codec.id << " for codec '"
+              << codec.name << ", rtcp-mux:"
+              << (media_description->rtcp_mux() ? "enabled" : "disabled"));
         }
       }
     }
@@ -2009,7 +2011,8 @@ RTCError SdpOfferAnswerHandler::ApplyLocalDescription(
 
   pending_ice_restarts_.clear();
   if (session_error() != SessionError::kNone) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR, GetSessionErrorMsg());
+    return LOG_ERROR(RTCError(RTCErrorType::INTERNAL_ERROR)
+                     << GetSessionErrorMsg());
   }
 
   // Validate SSRCs, we do not allow duplicates.
@@ -2020,9 +2023,8 @@ RTCError SdpOfferAnswerHandler::ApplyLocalDescription(
         for (uint32_t ssrc : stream.ssrcs) {
           auto result = used_ssrcs.insert(ssrc);
           if (!result.second) {
-            LOG_AND_RETURN_ERROR(
-                RTCErrorType::INVALID_PARAMETER,
-                "Duplicate ssrc " + absl::StrCat(ssrc) + " is not allowed");
+            return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                             << "Duplicate ssrc " << ssrc << " is not allowed");
           }
         }
       }
@@ -3409,11 +3411,9 @@ RTCError SdpOfferAnswerHandler::Rollback(SdpType desc_type) {
   auto state = signaling_state();
   if (state != PeerConnectionInterface::kHaveLocalOffer &&
       state != PeerConnectionInterface::kHaveRemoteOffer) {
-    LOG_AND_RETURN_ERROR(
-        RTCErrorType::INVALID_STATE,
-        (StringBuilder("Called in wrong signalingState: ")
-         << (PeerConnectionInterface::AsString(signaling_state())))
-            .Release());
+    return LOG_ERROR(RTCError(RTCErrorType::INVALID_STATE)
+                     << "Called in wrong signalingState: "
+                     << PeerConnectionInterface::AsString(signaling_state()));
   }
   RTC_DCHECK_RUN_ON(signaling_thread());
   RTC_DCHECK(IsUnifiedPlan());
@@ -3865,17 +3865,15 @@ RTCError SdpOfferAnswerHandler::ValidateSessionDescription(
   RTC_DCHECK_EQ(SessionError::kNone, session_error());
 
   if (!sdesc || !sdesc->description()) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, kInvalidSdp);
+    return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER) << kInvalidSdp);
   }
 
   SdpType type = sdesc->GetType();
   if ((source == CS_LOCAL && !ExpectSetLocalDescription(type)) ||
       (source == CS_REMOTE && !ExpectSetRemoteDescription(type))) {
-    LOG_AND_RETURN_ERROR(
-        RTCErrorType::INVALID_STATE,
-        (StringBuilder("Called in wrong state: ")
-         << PeerConnectionInterface::AsString(signaling_state()))
-            .Release());
+    return LOG_ERROR(RTCError(RTCErrorType::INVALID_STATE)
+                     << "Called in wrong state: "
+                     << PeerConnectionInterface::AsString(signaling_state()));
   }
 
   RTCError error = ValidateMids(*sdesc->description());
@@ -3894,8 +3892,8 @@ RTCError SdpOfferAnswerHandler::ValidateSessionDescription(
 
   // Verify ice-ufrag and ice-pwd.
   if (!VerifyIceUfragPwdPresent(sdesc->description(), bundle_groups_by_mid)) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
-                         kSdpWithoutIceUfragPwd);
+    return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                     << kSdpWithoutIceUfragPwd);
   }
 
   // Validate that there are no collisions of bundled payload types.
@@ -3918,8 +3916,8 @@ RTCError SdpOfferAnswerHandler::ValidateSessionDescription(
 
   if (!pc_->ValidateBundleSettings(sdesc->description(),
                                    bundle_groups_by_mid)) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
-                         kBundleWithoutRtcpMux);
+    return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                     << kBundleWithoutRtcpMux);
   }
 
   error = ValidatePayloadTypes(*sdesc->description());
@@ -3940,8 +3938,8 @@ RTCError SdpOfferAnswerHandler::ValidateSessionDescription(
     if (!MediaSectionsHaveSameCount(*offer_desc, *sdesc->description()) ||
         !MediaSectionsInSameOrder(*offer_desc, nullptr, *sdesc->description(),
                                   type)) {
-      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
-                           kMlineMismatchInAnswer);
+      return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                       << kMlineMismatchInAnswer);
     }
   } else {
     // The re-offers should respect the order of m= sections in current
@@ -3965,8 +3963,8 @@ RTCError SdpOfferAnswerHandler::ValidateSessionDescription(
     if (current_desc &&
         !MediaSectionsInSameOrder(*current_desc, secondary_current_desc,
                                   *sdesc->description(), type)) {
-      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
-                           kMlineMismatchInSubsequentOffer);
+      return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                       << kMlineMismatchInSubsequentOffer);
     }
   }
 
@@ -3981,10 +3979,10 @@ RTCError SdpOfferAnswerHandler::ValidateSessionDescription(
       if ((desc.type() == MediaType::AUDIO ||
            desc.type() == MediaType::VIDEO) &&
           desc.streams().size() > 1u) {
-        LOG_AND_RETURN_ERROR(
-            RTCErrorType::INVALID_PARAMETER,
-            "Media section has more than one track specified with a=ssrc lines "
-            "which is not supported with Unified Plan.");
+        return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                         << "Media section has more than one track specified "
+                            "with a=ssrc lines which is not supported with "
+                            "Unified Plan.");
       }
     }
     // Validate spec-simulcast which only works if the remote end negotiated the
@@ -4028,9 +4026,9 @@ RTCError SdpOfferAnswerHandler::UpdateTransceiversAndDataChannels(
     if (pc_->configuration()->bundle_policy ==
             PeerConnectionInterface::kBundlePolicyMaxBundle &&
         bundle_groups_by_mid.empty()) {
-      LOG_AND_RETURN_ERROR(
-          RTCErrorType::INVALID_PARAMETER,
-          "max-bundle configured but session description has no BUNDLE group");
+      return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                       << "max-bundle configured but session description has "
+                          "no BUNDLE group");
     }
   }
 
@@ -4121,8 +4119,8 @@ RTCError SdpOfferAnswerHandler::UpdateTransceiversAndDataChannels(
     } else if (media_type == MediaType::UNSUPPORTED) {
       RTC_LOG(LS_INFO) << "Ignoring unsupported media type";
     } else {
-      LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
-                           "Unknown section type.");
+      return LOG_ERROR(RTCError(RTCErrorType::INTERNAL_ERROR)
+                       << "Unknown section type.");
     }
   }
 
@@ -4166,8 +4164,8 @@ SdpOfferAnswerHandler::AssociateTransceiver(
     }
     if (!transceiver) {
       // This may happen normally when media sections are rejected.
-      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
-                           "Transceiver not found based on m-line index");
+      return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                       << "Transceiver not found based on m-line index");
     }
   } else {
     RTC_DCHECK_EQ(source, CS_REMOTE);
@@ -4225,9 +4223,9 @@ SdpOfferAnswerHandler::AssociateTransceiver(
   }
 
   if (transceiver->media_type() != media_desc->type()) {
-    LOG_AND_RETURN_ERROR(
-        RTCErrorType::INVALID_PARAMETER,
-        "Transceiver type does not match media description type.");
+    return LOG_ERROR(
+        RTCError(RTCErrorType::INVALID_PARAMETER)
+        << "Transceiver type does not match media description type.");
   }
 
   if (media_desc->HasSimulcast()) {
@@ -4306,8 +4304,8 @@ RTCError SdpOfferAnswerHandler::UpdateDataChannelTransport(
     error.set_error_detail(RTCErrorDetailType::DATA_CHANNEL_FAILURE);
     pc_->DestroyDataChannelTransport(error);
   } else if (!pc_->CreateDataChannelTransport(content.mid())) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
-                         "Failed to create data channel.");
+    return LOG_ERROR(RTCError(RTCErrorType::INTERNAL_ERROR)
+                     << "Failed to create data channel.");
   }
   return RTCError::OK();
 }
@@ -4895,8 +4893,8 @@ RTCError SdpOfferAnswerHandler::HandleLegacyOfferOptions(
   } else if (options.offer_to_receive_audio == 1) {
     AddUpToOneReceivingTransceiverOfType(MediaType::AUDIO);
   } else if (options.offer_to_receive_audio > 1) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_PARAMETER,
-                         "offer_to_receive_audio > 1 is not supported.");
+    return LOG_ERROR(RTCError(RTCErrorType::UNSUPPORTED_PARAMETER)
+                     << "offer_to_receive_audio > 1 is not supported.");
   }
 
   if (options.offer_to_receive_video == 0) {
@@ -4904,8 +4902,8 @@ RTCError SdpOfferAnswerHandler::HandleLegacyOfferOptions(
   } else if (options.offer_to_receive_video == 1) {
     AddUpToOneReceivingTransceiverOfType(MediaType::VIDEO);
   } else if (options.offer_to_receive_video > 1) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_PARAMETER,
-                         "offer_to_receive_video > 1 is not supported.");
+    return LOG_ERROR(RTCError(RTCErrorType::UNSUPPORTED_PARAMETER)
+                     << "offer_to_receive_video > 1 is not supported.");
   }
 
   return RTCError::OK();
@@ -5173,8 +5171,8 @@ RTCError SdpOfferAnswerHandler::PushdownMediaDescription(
       // Note that this is never expected to fail, since RtpDemuxer doesn't
       // return an error when changing payload type demux criteria, which is all
       // this does.
-      LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
-                           "Failed to update payload type demuxing state.");
+      return LOG_ERROR(RTCError(RTCErrorType::INTERNAL_ERROR)
+                       << "Failed to update payload type demuxing state.");
     }
 
     // Push down the new SDP media section for each audio/video transceiver.
@@ -5592,10 +5590,10 @@ RTCErrorOr<const ContentInfo*> SdpOfferAnswerHandler::FindContentInfo(
           return content_info.mid() == candidate->sdp_mid();
         });
     if (it == contents.end()) {
-      LOG_AND_RETURN_ERROR(
-          RTCErrorType::INVALID_PARAMETER,
-          "Mid " + candidate->sdp_mid() +
-              " specified but no media section with that mid found.");
+      return LOG_ERROR(
+          RTCError(RTCErrorType::INVALID_PARAMETER)
+          << "Mid " << candidate->sdp_mid()
+          << " specified but no media section with that mid found.");
     } else {
       return &*it;
     }
@@ -5606,16 +5604,15 @@ RTCErrorOr<const ContentInfo*> SdpOfferAnswerHandler::FindContentInfo(
     if (mediacontent_index < content_size) {
       return &description->description()->contents()[mediacontent_index];
     } else {
-      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_RANGE,
-                           "Media line index (" +
-                               absl::StrCat(candidate->sdp_mline_index()) +
-                               ") out of range (number of mlines: " +
-                               absl::StrCat(content_size) + ").");
+      return LOG_ERROR(RTCError(RTCErrorType::INVALID_RANGE)
+                       << "Media line index (" << candidate->sdp_mline_index()
+                       << ") out of range (number of mlines: " << content_size
+                       << ").");
     }
   }
 
-  LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
-                       "Neither sdp_mline_index nor sdp_mid specified.");
+  return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
+                   << "Neither sdp_mline_index nor sdp_mid specified.");
 }
 
 RTCError SdpOfferAnswerHandler::CreateChannels(const SessionDescription& desc) {
@@ -5663,8 +5660,8 @@ RTCError SdpOfferAnswerHandler::CreateChannels(const SessionDescription& desc) {
   const ContentInfo* data = GetFirstDataContent(&desc);
   if (data && !data->rejected &&
       !pc_->CreateDataChannelTransport(data->mid())) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
-                         "Failed to create data channel.");
+    return LOG_ERROR(RTCError(RTCErrorType::INTERNAL_ERROR)
+                     << "Failed to create data channel.");
   }
 
   return RTCError::OK();

From 49a0591324bf382663a1e0bc8da0d5008469cacc Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Tue, 24 Mar 2026 05:10:18 -0700
Subject: [PATCH 388/472] Roll chromium_revision c9a20fd775..ba4fdb3ab0
 (1603923:1604035)

Change log: https://chromium.googlesource.com/chromium/src/+log/c9a20fd775..ba4fdb3ab0
Full diff: https://chromium.googlesource.com/chromium/src/+/c9a20fd775..ba4fdb3ab0

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/613f1bc9c3..97f7ea8925
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/74c2aacf1a..084148e261
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/1aee011cb5..ae6eb7c16b
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/b4c442492a..72dbf4d474
* src/third_party/android_deps/autorolled/cipd: 2mhz9b8511ym0Z01dgIFbJPgHyotC0CmmWp1piMvNcoC..NDkHoZFBi462KHX9MpNy-3MfJnllAi8MytUuzRobeeQC
* src/third_party/androidx/cipd: SF_J8m-KyUb2Qlq0kZtcHq_x2kYglLBXY1_5u-og9SsC..GIgdzzHTYcjiT8XrtVjd6rjspxYt0Hs0drpRjWcNwYUC
* src/third_party/eigen3/src: https://chromium.googlesource.com/external/gitlab.com/libeigen/eigen.git/+log/f574cb9b18..002229ce47
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/173b262da7..4f623df331
* src/third_party/zstd/src: https://chromium.googlesource.com/external/github.com/facebook/zstd.git/+log/1168da0e56..3ae099b48d
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/2ca854011d..bfd723ef78
DEPS diff: https://chromium.googlesource.com/chromium/src/+/c9a20fd775..ba4fdb3ab0/DEPS

BUG=None

Change-Id: Idad529b664b8466d6f90c047ffd9f5b75035b709
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/459622
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47241}
---
 DEPS | 22 +++++++++++-----------
 1 file changed, 11 insertions(+), 11 deletions(-)

diff --git a/DEPS b/DEPS
index d145ddf62d..eed36c5cb9 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': 'c9a20fd7758f5caf1efbe4e3d4d3bb9bba47533f',
+  'chromium_revision': 'ba4fdb3ab05879f5fcf2a0ae228be827e390fe83',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@613f1bc9c3886611ca9354bd3e409082277e8919',
+    'https://chromium.googlesource.com/chromium/src/build@97f7ea8925aca5014399715eb8b1edd2473081a3',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@95ed44cf5f06dbb5861030b91c9db9ccb4316762',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@74c2aacf1af6fbe9a88303e16afaadc195ee9eee',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@084148e2616290286edef31b03fe729c75385037',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@1aee011cb5e9ebe3e0821db60d361ea7e9d7513e',
+    'https://chromium.googlesource.com/chromium/src/testing@ae6eb7c16bec1d5247a7569f4977f428803328d0',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@b4c442492a7e1eef1234ef946fb35bc5f94059d5',
+    'https://chromium.googlesource.com/chromium/src/third_party@72dbf4d474cbe174e77a3c95e89a29ac276bdb20',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -511,7 +511,7 @@ deps = {
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@173b262da788de397ea3f7d788156c59272f3a89',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@4f623df3319df7e39a89fa2f4a198d43744917b7',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@2ca854011dffebab4eecc00110783373a5d2ebb9',
+    'https://chromium.googlesource.com/chromium/src/tools@bfd723ef7807f05437772eac610625def887f4ef',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'SF_J8m-KyUb2Qlq0kZtcHq_x2kYglLBXY1_5u-og9SsC',
+          'version': 'GIgdzzHTYcjiT8XrtVjd6rjspxYt0Hs0drpRjWcNwYUC',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -800,7 +800,7 @@ deps = {
   },
 
   'src/third_party/zstd/src': {
-    'url': Var('chromium_git') + '/external/github.com/facebook/zstd.git' + '@' + '1168da0e567960d50cba1b58c9b0ba047ece4733',
+    'url': Var('chromium_git') + '/external/github.com/facebook/zstd.git' + '@' + '3ae099b48dfcfe02b1b3ba81ab85457f8a922e9f',
     'condition': 'checkout_android',
   },
 
@@ -842,7 +842,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/android_deps/autorolled',
-              'version': '2mhz9b8511ym0Z01dgIFbJPgHyotC0CmmWp1piMvNcoC',
+              'version': 'NDkHoZFBi462KHX9MpNy-3MfJnllAi8MytUuzRobeeQC',
           },
       ],
       'condition': 'checkout_android and non_git_source',
@@ -864,7 +864,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/pytorch/cpuinfo.git' + '@' + '7607ca500436b37ad23fb8d18614bec7796b68a7',
 
   'src/third_party/eigen3/src':
-    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + 'f574cb9b18a3e9ddf09ec80a3afb0270649d61a5',
+    Var('chromium_git') + '/external/gitlab.com/libeigen/eigen.git' + '@' + '002229ce470065878afb7c2f6f96d22c9a9b7ba0',
 
   'src/third_party/fp16/src':
     Var('chromium_git') + '/external/github.com/Maratyszcza/FP16.git' + '@' + '3d2de1816307bac63c16a297e8c4dc501b4076df',

From 731795bab2d89be63e20485407a850173d5d3665 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Mon, 23 Mar 2026 17:29:21 +0100
Subject: [PATCH 389/472] Refactor AddCertificateReports to prevent crash
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

When processing certificate stats with duplicate fingerprints,
ReplaceOrAddNew would delete the existing stats report that was
previously added and pointed to by first_report or prev_report.

By adding tracking for duplicate fingerprints we can find the existing
report and avoid replacing it.

Bug: chromium:486495143
Change-Id: Iabc41ae064476c1e5853cdff1dbbcab449f8df27
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/459320
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47242}
---
 pc/BUILD.gn                           |  1 +
 pc/legacy_stats_collector.cc          |  6 ++++++
 pc/legacy_stats_collector.h           |  5 +++++
 pc/legacy_stats_collector_unittest.cc | 16 ++++++++++++++++
 4 files changed, 28 insertions(+)

diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index c07cbd3016..f2045cb32a 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -1461,6 +1461,7 @@ rtc_library("legacy_stats_collector") {
     "../rtc_base/system:plan_b_only",
     "../system_wrappers",
     "//third_party/abseil-cpp/absl/base:nullability",
+    "//third_party/abseil-cpp/absl/container:flat_hash_set",
     "//third_party/abseil-cpp/absl/functional:any_invocable",
     "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/strings:string_view",
diff --git a/pc/legacy_stats_collector.cc b/pc/legacy_stats_collector.cc
index f41de1cded..6978f1b06b 100644
--- a/pc/legacy_stats_collector.cc
+++ b/pc/legacy_stats_collector.cc
@@ -21,6 +21,7 @@
 #include <vector>
 
 #include "absl/base/nullability.h"
+#include "absl/container/flat_hash_set.h"
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/str_cat.h"
 #include "absl/strings/string_view.h"
@@ -792,8 +793,13 @@ StatsReport* LegacyStatsCollector::AddCertificateReports(
 
   StatsReport* first_report = nullptr;
   StatsReport* prev_report = nullptr;
+  absl::flat_hash_set<std::string> visited_fingerprints;
   for (SSLCertificateStats* stats = cert_stats.get(); stats;
        stats = stats->issuer.get()) {
+    if (!visited_fingerprints.insert(stats->fingerprint).second) {
+      break;
+    }
+
     StatsReport::Id id(StatsReport::NewTypedId(
         StatsReport::kStatsReportTypeCertificate, stats->fingerprint));
 
diff --git a/pc/legacy_stats_collector.h b/pc/legacy_stats_collector.h
index 25e61edfdd..f36e509664 100644
--- a/pc/legacy_stats_collector.h
+++ b/pc/legacy_stats_collector.h
@@ -114,6 +114,11 @@ class LegacyStatsCollector : public LegacyStatsCollectorInterface {
 
   bool UseStandardBytesStats() const { return use_standard_bytes_stats_; }
 
+  StatsReport* AddCertificateReportsForTest(
+      std::unique_ptr<SSLCertificateStats> cert_stats) {
+    return AddCertificateReports(std::move(cert_stats));
+  }
+
  private:
   // Struct that's populated on the network thread and carries the values to
   // the signaling thread where the stats are added to the stats reports.
diff --git a/pc/legacy_stats_collector_unittest.cc b/pc/legacy_stats_collector_unittest.cc
index 63c33ec177..f03911589b 100644
--- a/pc/legacy_stats_collector_unittest.cc
+++ b/pc/legacy_stats_collector_unittest.cc
@@ -57,6 +57,7 @@
 #include "rtc_base/null_socket_server.h"
 #include "rtc_base/rtc_certificate.h"
 #include "rtc_base/socket_address.h"
+#include "rtc_base/ssl_certificate.h"
 #include "rtc_base/ssl_identity.h"
 #include "rtc_base/ssl_stream_adapter.h"
 #include "rtc_base/system/plan_b_only.h"
@@ -1446,6 +1447,21 @@ TEST_F(LegacyStatsCollectorTest, ChainedCertificateReportsCreated) {
                          remote_ders);
 }
 
+TEST_F(LegacyStatsCollectorTest,
+       AddCertificateReports_DuplicateFingerprintDoesNotCrash) {
+  auto pc = CreatePeerConnection();
+  auto stats = CreateStatsCollector(pc.get());
+
+  // Create duplicate fingerprints chain
+  auto issuer_stats = std::make_unique<SSLCertificateStats>(
+      "same_fingerprint", "sha-1", "cert_issuer", nullptr);
+  auto cert_stats = std::make_unique<SSLCertificateStats>(
+      "same_fingerprint", "sha-1", "cert_leaf", std::move(issuer_stats));
+
+  // This should not crash (Use-After-Free)
+  stats->AddCertificateReportsForTest(std::move(cert_stats));
+}
+
 // This test verifies that all certificates without chains are correctly
 // reported.
 TEST_F(LegacyStatsCollectorTest, ChainlessCertificateReportsCreated) {

From c397ee98ed8f3acd829466c0b8ff5e97073c3ff6 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Tue, 24 Mar 2026 11:47:14 +0100
Subject: [PATCH 390/472] Follow-up auto -> explicit type

Follow-up updates from a cherrypick review. Update use of auto in
rtc_stats_collector_unittest.cc and track_media_info_map.cc to use an
explicit type.

Bug: none
Change-Id: I06761f6e241602c7002551f2554ea878a5697cb6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/459661
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47243}
---
 pc/rtc_stats_collector_unittest.cc | 4 ++--
 pc/track_media_info_map.cc         | 6 +++---
 2 files changed, 5 insertions(+), 5 deletions(-)

diff --git a/pc/rtc_stats_collector_unittest.cc b/pc/rtc_stats_collector_unittest.cc
index 2b5e4b072f..35905ed3f1 100644
--- a/pc/rtc_stats_collector_unittest.cc
+++ b/pc/rtc_stats_collector_unittest.cc
@@ -2743,7 +2743,7 @@ TEST_P(RTCStatsCollectorTest, CollectRTCOutboundRtpStreamStats_Video) {
   auto stats_of_my_type = report->GetStatsOfType<RTCOutboundRtpStreamStats>();
   ASSERT_EQ(2U, stats_of_my_type.size());
   std::string id_of_first_ssrc;
-  for (const auto* outbound_rtp : stats_of_my_type) {
+  for (const RTCOutboundRtpStreamStats* outbound_rtp : stats_of_my_type) {
     if (outbound_rtp->ssrc.value_or(0) == 1) {
       id_of_first_ssrc = outbound_rtp->id();
       break;
@@ -2803,7 +2803,7 @@ TEST_P(RTCStatsCollectorTest, CollectRTCOutboundRtpStreamStats_Video) {
   // Find the sub-stats for the second simulcast outbound RTP stream and verify
   // that its `media_source_id` is set.
   std::string id_of_second_ssrc;
-  for (const auto* outbound_rtp : stats_of_my_type) {
+  for (const RTCOutboundRtpStreamStats* outbound_rtp : stats_of_my_type) {
     if (outbound_rtp->ssrc.value_or(0) == 2) {
       id_of_second_ssrc = outbound_rtp->id();
       break;
diff --git a/pc/track_media_info_map.cc b/pc/track_media_info_map.cc
index 91d12c509b..afe6a36c14 100644
--- a/pc/track_media_info_map.cc
+++ b/pc/track_media_info_map.cc
@@ -46,15 +46,15 @@ flat_map<uint32_t, int> GetSenderAttachmentIds(
   RTC_DCHECK_EQ(senders.size(), sender_parameters.size());
   flat_map<uint32_t, int> result;
   for (size_t i = 0; i < senders.size(); ++i) {
-    const auto& sender = senders[i];
+    const TrackMediaInfoMap::RtpSenderSignalInfo& sender = senders[i];
     if (sender.media_type != media_type) {
       continue;
     }
-    const auto& parameters = sender_parameters[i];
+    const RtpParameters& parameters = sender_parameters[i];
     if (sender.ssrc != 0) {
       result[sender.ssrc] = sender.attachment_id;
     }
-    for (const auto& encoding : parameters.encodings) {
+    for (const RtpEncodingParameters& encoding : parameters.encodings) {
       if (encoding.ssrc) {
         result[*encoding.ssrc] = sender.attachment_id;
       }

From f4bfe0b65413d57918aa1a731b1241d6093f79ea Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Tue, 24 Mar 2026 09:05:42 -0700
Subject: [PATCH 391/472] Roll chromium_revision ba4fdb3ab0..4acdf7133f
 (1604035:1604168)

Change log: https://chromium.googlesource.com/chromium/src/+log/ba4fdb3ab0..4acdf7133f
Full diff: https://chromium.googlesource.com/chromium/src/+/ba4fdb3ab0..4acdf7133f

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/97f7ea8925..d76036c311
* src/buildtools/linux64: git_revision:c91f7e86727683118b6096757717135463fdcdd7..git_revision:b2ac0e7a9089039e62b84d246eca83f84c540f76
* src/buildtools/mac: git_revision:c91f7e86727683118b6096757717135463fdcdd7..git_revision:b2ac0e7a9089039e62b84d246eca83f84c540f76
* src/buildtools/win: git_revision:c91f7e86727683118b6096757717135463fdcdd7..git_revision:b2ac0e7a9089039e62b84d246eca83f84c540f76
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/084148e261..998e238962
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/ae6eb7c16b..0491a27dd1
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/72dbf4d474..db50e5c540
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/d5c8d9fb01..2e5b239d79
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/f6800908d7..3c997950a0
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/bfd723ef78..465340f84f
DEPS diff: https://chromium.googlesource.com/chromium/src/+/ba4fdb3ab0..4acdf7133f/DEPS

BUG=None

Change-Id: Icf37015caf0e0c7916777f25bb654bf61684a73f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/459740
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47244}
---
 DEPS | 22 +++++++++++-----------
 1 file changed, 11 insertions(+), 11 deletions(-)

diff --git a/DEPS b/DEPS
index eed36c5cb9..dc5f55b06f 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': 'ba4fdb3ab05879f5fcf2a0ae228be827e390fe83',
+  'chromium_revision': '4acdf7133f643cd2a965fe471f646599359c7d82',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@97f7ea8925aca5014399715eb8b1edd2473081a3',
+    'https://chromium.googlesource.com/chromium/src/build@d76036c3115ecdb183ae57c8f9e922760ea8e47c',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@95ed44cf5f06dbb5861030b91c9db9ccb4316762',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,19 +78,19 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@084148e2616290286edef31b03fe729c75385037',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@998e2389625373f61d37fbd6e60bea45ea30536c',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@ae6eb7c16bec1d5247a7569f4977f428803328d0',
+    'https://chromium.googlesource.com/chromium/src/testing@0491a27dd1af96bf568bd0911c6cae53d9e4b2dc',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@72dbf4d474cbe174e77a3c95e89a29ac276bdb20',
+    'https://chromium.googlesource.com/chromium/src/third_party@db50e5c54009efb3b2fbd3aae771080dc398c5f9',
 
   'src/buildtools/linux64': {
     'packages': [
       {
         'package': 'gn/gn/linux-${{arch}}',
-        'version': 'git_revision:c91f7e86727683118b6096757717135463fdcdd7',
+        'version': 'git_revision:b2ac0e7a9089039e62b84d246eca83f84c540f76',
       }
     ],
     'dep_type': 'cipd',
@@ -100,7 +100,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/mac-${{arch}}',
-        'version': 'git_revision:c91f7e86727683118b6096757717135463fdcdd7',
+        'version': 'git_revision:b2ac0e7a9089039e62b84d246eca83f84c540f76',
       }
     ],
     'dep_type': 'cipd',
@@ -110,7 +110,7 @@ deps = {
     'packages': [
       {
         'package': 'gn/gn/windows-amd64',
-        'version': 'git_revision:c91f7e86727683118b6096757717135463fdcdd7',
+        'version': 'git_revision:b2ac0e7a9089039e62b84d246eca83f84c540f76',
       }
     ],
     'dep_type': 'cipd',
@@ -505,7 +505,7 @@ deps = {
   'src/third_party/crc32c/src':
     'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558',
   'src/third_party/depot_tools':
-    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@d5c8d9fb01eff58a78c69beffa1c6fc5d2f5b770',
+    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@2e5b239d792f082ee7c008ffea310da64d5527eb',
   'src/third_party/ffmpeg':
     'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@946d97db8d906277085e361892b7efda5152e2f1',
   'src/third_party/flatbuffers/src':
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'f6800908d7bebd44732059ed0310f6e5965d42a6',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '3c997950a05e9d0da33d95e99ad952ba275b21e4',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@bfd723ef7807f05437772eac610625def887f4ef',
+    'https://chromium.googlesource.com/chromium/src/tools@465340f84f9f81116a8e50542e7b34293d1a2567',
 
   'src/third_party/espresso': {
       'packages': [

From 0bc96c80f368842ab7eca754a8606d714fd3194b Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Erik=20Spr=C3=A5ng?= <sprang@webrtc.org>
Date: Tue, 24 Mar 2026 15:44:44 +0100
Subject: [PATCH 392/472] Reland "Allow unspecified max allocatable bitrate in
 VideoSendStreamImpl"
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This is a reland of commit e844bd819c4663de5422be9c2cdb25204b03b41b

The changes after Patchset 1 are intended to fix a corner case where a
video send stream config is used where layers (simulcast or svc) are
configured and active but have undefined max bitrate. In those cases the
max bitrate will be inherited from the global max bitrate.

Original change's description:
> Allow unspecified max allocatable bitrate in VideoSendStreamImpl
>
> If we have no video the can be enabled, based on the VideoEncoderConfig
> (e.g. the max bitrate is <= 0 or all simulcast streams are
> `active=false`) - let the max bitrate be reported as 0 for the sake of
> bandwidth allocation (MediaStreamAllocationConfig).
>
> Previous to this CL, the above conditions would result in an arbitrary
> value of 10Mbps to use as the max allocation limit. When creating a peer
> connection with audio and video configured but the video disabled/muted,
> that would result in the bandwidth estimator trying to send packet
> trains to probe up to 10Mbps, potentially impacting the network and
> disturbing audio even though we have no need for such as high BWE.
>
> Bug: webrtc:494350649
> Change-Id: Id6353fd93c6170610a790c084d75f3b3c5c3ee97
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449100
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#47220}

Bug: webrtc:494350649
Change-Id: I59e415bbd121b8661dc5b53e67a2788a335539a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/459680
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47245}
---
 media/engine/webrtc_video_engine_unittest.cc  |  50 ++---
 .../bwe_integration_tests/bwe_ramp_up_test.cc |  15 +-
 test/scenario/video_stream_unittest.cc        |   2 +-
 video/video_send_stream_impl.cc               | 183 +++++++++++------
 video/video_send_stream_impl.h                |  10 +-
 video/video_send_stream_impl_unittest.cc      | 194 +++++++++++++++++-
 6 files changed, 357 insertions(+), 97 deletions(-)

diff --git a/media/engine/webrtc_video_engine_unittest.cc b/media/engine/webrtc_video_engine_unittest.cc
index fdd105de71..368554440c 100644
--- a/media/engine/webrtc_video_engine_unittest.cc
+++ b/media/engine/webrtc_video_engine_unittest.cc
@@ -10020,6 +10020,11 @@ TEST_F(WebRtcVideoChannelBaseTest, EncoderSelectorSwitchCodec) {
   EXPECT_TRUE(send_channel_->SetSenderParameters(parameters));
   send_channel_->SetSend(true);
 
+  RtpParameters rtp_parameters = send_channel_->GetRtpSendParameters(kSsrc);
+  ASSERT_EQ(1u, rtp_parameters.encodings.size());
+  rtp_parameters.encodings[0].max_bitrate_bps = 1000000;
+  EXPECT_TRUE(send_channel_->SetRtpSendParameters(kSsrc, rtp_parameters).ok());
+
   std::optional<Codec> codec = send_channel_->GetSendCodec();
   ASSERT_TRUE(codec);
   EXPECT_EQ("VP8", codec->name);
@@ -10041,23 +10046,20 @@ TEST_F(WebRtcVideoChannelBaseTest, EncoderSelectorSwitchCodec) {
 
 TEST_F(WebRtcVideoChannelTest, ScaleResolutionDownToSinglecast) {
   VideoSenderParameters parameters;
-  parameters.codecs.push_back(GetEngineCodec("VP8"));
-  ASSERT_TRUE(send_channel_->SetSenderParameters(parameters));
-
   FakeVideoSendStream* stream = AddSendStream();
+  const uint32_t ssrc = stream->GetConfig().rtp.ssrcs[0];
   FrameForwarder frame_forwarder;
   FakeFrameSource frame_source(1280, 720, TimeDelta::Seconds(1) / 30,
                                env_.clock().CurrentTime());
-  EXPECT_TRUE(
-      send_channel_->SetVideoSend(last_ssrc_, nullptr, &frame_forwarder));
+  EXPECT_TRUE(send_channel_->SetVideoSend(ssrc, nullptr, &frame_forwarder));
+  frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
 
   {  // TEST scale_resolution_down_to < frame size
-    RtpParameters rtp_parameters =
-        send_channel_->GetRtpSendParameters(last_ssrc_);
+    RtpParameters rtp_parameters = send_channel_->GetRtpSendParameters(ssrc);
     EXPECT_EQ(1UL, rtp_parameters.encodings.size());
     rtp_parameters.encodings[0].scale_resolution_down_to = {.width = 640,
                                                             .height = 360};
-    send_channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
+    send_channel_->SetRtpSendParameters(ssrc, rtp_parameters);
 
     frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
 
@@ -10068,11 +10070,11 @@ TEST_F(WebRtcVideoChannelTest, ScaleResolutionDownToSinglecast) {
   }
 
   {  // TEST scale_resolution_down_to == frame size
-    auto rtp_parameters = send_channel_->GetRtpSendParameters(last_ssrc_);
+    auto rtp_parameters = send_channel_->GetRtpSendParameters(ssrc);
     EXPECT_EQ(1UL, rtp_parameters.encodings.size());
     rtp_parameters.encodings[0].scale_resolution_down_to = {.width = 1280,
                                                             .height = 720};
-    send_channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
+    send_channel_->SetRtpSendParameters(ssrc, rtp_parameters);
 
     frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
     auto streams = stream->GetVideoStreams();
@@ -10082,11 +10084,11 @@ TEST_F(WebRtcVideoChannelTest, ScaleResolutionDownToSinglecast) {
   }
 
   {  // TEST scale_resolution_down_to > frame size
-    auto rtp_parameters = send_channel_->GetRtpSendParameters(last_ssrc_);
+    auto rtp_parameters = send_channel_->GetRtpSendParameters(ssrc);
     EXPECT_EQ(1UL, rtp_parameters.encodings.size());
     rtp_parameters.encodings[0].scale_resolution_down_to = {.width = 2 * 1280,
                                                             .height = 2 * 720};
-    send_channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
+    send_channel_->SetRtpSendParameters(ssrc, rtp_parameters);
 
     frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
     auto streams = stream->GetVideoStreams();
@@ -10095,7 +10097,7 @@ TEST_F(WebRtcVideoChannelTest, ScaleResolutionDownToSinglecast) {
     EXPECT_EQ(checked_cast<size_t>(720), streams[0].height);
   }
 
-  EXPECT_TRUE(send_channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
+  EXPECT_TRUE(send_channel_->SetVideoSend(ssrc, nullptr, nullptr));
 }
 
 TEST_F(WebRtcVideoChannelTest, ScaleResolutionDownToSinglecastScaling) {
@@ -10104,18 +10106,18 @@ TEST_F(WebRtcVideoChannelTest, ScaleResolutionDownToSinglecastScaling) {
   ASSERT_TRUE(send_channel_->SetSenderParameters(parameters));
 
   FakeVideoSendStream* stream = AddSendStream();
+  const uint32_t ssrc = stream->GetConfig().rtp.ssrcs[0];
   FrameForwarder frame_forwarder;
   FakeFrameSource frame_source(1280, 720, TimeDelta::Seconds(1) / 30,
                                env_.clock().CurrentTime());
-  EXPECT_TRUE(
-      send_channel_->SetVideoSend(last_ssrc_, nullptr, &frame_forwarder));
+  EXPECT_TRUE(send_channel_->SetVideoSend(ssrc, nullptr, &frame_forwarder));
 
   {
-    auto rtp_parameters = send_channel_->GetRtpSendParameters(last_ssrc_);
+    auto rtp_parameters = send_channel_->GetRtpSendParameters(ssrc);
     EXPECT_EQ(1UL, rtp_parameters.encodings.size());
     rtp_parameters.encodings[0].scale_resolution_down_to = {.width = 720,
                                                             .height = 720};
-    send_channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
+    send_channel_->SetRtpSendParameters(ssrc, rtp_parameters);
 
     frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
 
@@ -10128,11 +10130,11 @@ TEST_F(WebRtcVideoChannelTest, ScaleResolutionDownToSinglecastScaling) {
   }
 
   {
-    auto rtp_parameters = send_channel_->GetRtpSendParameters(last_ssrc_);
+    auto rtp_parameters = send_channel_->GetRtpSendParameters(ssrc);
     EXPECT_EQ(1UL, rtp_parameters.encodings.size());
     rtp_parameters.encodings[0].scale_resolution_down_to = {.width = 1280,
                                                             .height = 1280};
-    send_channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
+    send_channel_->SetRtpSendParameters(ssrc, rtp_parameters);
 
     frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
 
@@ -10144,11 +10146,11 @@ TEST_F(WebRtcVideoChannelTest, ScaleResolutionDownToSinglecastScaling) {
   }
 
   {
-    auto rtp_parameters = send_channel_->GetRtpSendParameters(last_ssrc_);
+    auto rtp_parameters = send_channel_->GetRtpSendParameters(ssrc);
     EXPECT_EQ(1UL, rtp_parameters.encodings.size());
     rtp_parameters.encodings[0].scale_resolution_down_to = {.width = 650,
                                                             .height = 650};
-    send_channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
+    send_channel_->SetRtpSendParameters(ssrc, rtp_parameters);
 
     auto streams = stream->GetVideoStreams();
     ASSERT_EQ(streams.size(), 1u);
@@ -10159,11 +10161,11 @@ TEST_F(WebRtcVideoChannelTest, ScaleResolutionDownToSinglecastScaling) {
   }
 
   {
-    auto rtp_parameters = send_channel_->GetRtpSendParameters(last_ssrc_);
+    auto rtp_parameters = send_channel_->GetRtpSendParameters(ssrc);
     EXPECT_EQ(1UL, rtp_parameters.encodings.size());
     rtp_parameters.encodings[0].scale_resolution_down_to = {.width = 2560,
                                                             .height = 1440};
-    send_channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
+    send_channel_->SetRtpSendParameters(ssrc, rtp_parameters);
 
     auto streams = stream->GetVideoStreams();
     ASSERT_EQ(streams.size(), 1u);
@@ -10172,7 +10174,7 @@ TEST_F(WebRtcVideoChannelTest, ScaleResolutionDownToSinglecastScaling) {
     EXPECT_EQ(checked_cast<size_t>(720), streams[0].height);
   }
 
-  EXPECT_TRUE(send_channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
+  EXPECT_TRUE(send_channel_->SetVideoSend(ssrc, nullptr, nullptr));
 }
 
 TEST_F(WebRtcVideoChannelTest, ScaleResolutionDownToSimulcast) {
diff --git a/test/peer_scenario/bwe_integration_tests/bwe_ramp_up_test.cc b/test/peer_scenario/bwe_integration_tests/bwe_ramp_up_test.cc
index 8a0bbd37a3..2aac0dc9f7 100644
--- a/test/peer_scenario/bwe_integration_tests/bwe_ramp_up_test.cc
+++ b/test/peer_scenario/bwe_integration_tests/bwe_ramp_up_test.cc
@@ -16,6 +16,7 @@
 
 #include "api/jsep.h"
 #include "api/media_types.h"
+#include "api/rtp_parameters.h"
 #include "api/rtp_sender_interface.h"
 #include "api/rtp_transceiver_direction.h"
 #include "api/scoped_refptr.h"
@@ -206,7 +207,7 @@ INSTANTIATE_TEST_SUITE_P(
              .l4s_network = true,
              .network_capacity = DataRate::KilobitsPerSec(3000),
              .expected_bwe_min = DataRate::KilobitsPerSec(1500),
-             .max_bwe = DataRate::KilobitsPerSec(3000),
+             .max_bwe = DataRate::KilobitsPerSec(3600),
          },
          {
              .test_name = "L4s500Kbit",
@@ -239,12 +240,18 @@ TEST_P(BweRampupWithInitialProbeTest, BweRampUpBothDirectionsWithoutMedia) {
   PeerScenarioClient* caller = s.CreateClient(config);
   PeerScenarioClient* callee = s.CreateClient(config);
 
-  auto transceiver = caller->pc()->AddTransceiver(MediaType::VIDEO);
-  ASSERT_TRUE(transceiver.error().ok());
+  auto transceiver_or = caller->pc()->AddTransceiver(MediaType::VIDEO);
+  ASSERT_TRUE(transceiver_or.error().ok());
+  auto transceiver = transceiver_or.value();
+
+  auto parameters = transceiver->sender()->GetParameters();
+  ASSERT_THAT(parameters.encodings, testing::SizeIs(1));
+  parameters.encodings[0].max_bitrate_bps = 3'000'000;
+  ASSERT_TRUE(transceiver->sender()->SetParameters(parameters).ok());
 
   MockRtpSenderObserver observer;
   EXPECT_CALL(observer, OnFirstPacketSent).Times(0);
-  transceiver.value()->sender()->SetObserver(&observer);
+  transceiver->sender()->SetObserver(&observer);
 
   caller->pc()->ReconfigureBandwidthEstimation(
       {.allow_probe_without_media = true});
diff --git a/test/scenario/video_stream_unittest.cc b/test/scenario/video_stream_unittest.cc
index c9791bf8fc..c1fcd4da1c 100644
--- a/test/scenario/video_stream_unittest.cc
+++ b/test/scenario/video_stream_unittest.cc
@@ -281,7 +281,7 @@ TEST(VideoStreamTest, SuspendsBelowMinBitrate) {
     // Min transmit rate needs to be lower than kMinVideoBitrate for this test
     // to make sense.
     c->transport.rates.min_rate = kMinVideoBitrate / 2;
-    c->transport.rates.start_rate = kMinVideoBitrate;
+    c->transport.rates.start_rate = kMinVideoBitrate * 2;
     c->transport.rates.max_rate = kMinVideoBitrate * 2;
   });
   auto send_net = s.CreateMutableSimulationNode(
diff --git a/video/video_send_stream_impl.cc b/video/video_send_stream_impl.cc
index ad9130bb1d..f4d5d27c92 100644
--- a/video/video_send_stream_impl.cc
+++ b/video/video_send_stream_impl.cc
@@ -46,6 +46,7 @@
 #include "api/video/video_layers_allocation.h"
 #include "api/video/video_source_interface.h"
 #include "api/video/video_stream_encoder_settings.h"
+#include "api/video_codecs/spatial_layer.h"
 #include "api/video_codecs/video_codec.h"
 #include "api/video_codecs/video_encoder.h"
 #include "api/video_codecs/video_encoder_factory.h"
@@ -204,6 +205,36 @@ bool SameStreamsEnabled(const VideoBitrateAllocation& lhs,
   return true;
 }
 
+// Calculates the maximum bitrate across a set of layers (spatial or simulcast).
+// If there are active layers but they lack individual bitrate caps, falls back
+// to the provided global maximum bitrate. Returns std::nullopt if there are no
+// active layers, indicating that no bitrate should be allocated.
+template <typename T, typename ActiveGetter, typename MaxBitrateGetter>
+std::optional<DataRate> CalculateMaxBitrate(const std::vector<T>& layers,
+                                            ActiveGetter active_getter,
+                                            MaxBitrateGetter max_bitrate_getter,
+                                            DataRate global_max_bitrate) {
+  bool has_active_layer = false;
+  std::optional<DataRate> max_bitrate_sum;
+  for (const auto& layer : layers) {
+    if (active_getter(layer)) {
+      has_active_layer = true;
+      DataRate layer_max = max_bitrate_getter(layer);
+      if (layer_max > DataRate::Zero()) {
+        max_bitrate_sum =
+            max_bitrate_sum.value_or(DataRate::Zero()) + layer_max;
+      }
+    }
+  }
+  if (has_active_layer) {
+    if (!max_bitrate_sum.has_value() && !global_max_bitrate.IsZero()) {
+      return global_max_bitrate;
+    }
+    return max_bitrate_sum;
+  }
+  return std::nullopt;
+}
+
 // Returns an optional that has value iff TransportSeqNumExtensionConfigured
 // is `true` for the given video send stream config.
 std::optional<float> GetConfiguredPacingFactor(
@@ -236,32 +267,45 @@ int GetEncoderPriorityBitrate(std::string codec_name,
   return priority_bitrate;
 }
 
-uint32_t GetInitialEncoderMaxBitrate(int initial_encoder_max_bitrate) {
-  if (initial_encoder_max_bitrate > 0)
-    return dchecked_cast<uint32_t>(initial_encoder_max_bitrate);
-
-  // TODO(srte): Make sure max bitrate is not set to negative values. We don't
-  // have any way to handle unset values in downstream code, such as the
-  // bitrate allocator. Previously -1 was implicitly casted to UINT32_MAX, a
-  // behaviour that is not safe. Converting to 10 Mbps should be safe for
-  // reasonable use cases as it allows adding the max of multiple streams
-  // without wrappping around.
-  const int kFallbackMaxBitrateBps = 10000000;
-  // Don't log an error for -1 since this is the default value that is used to
-  // signal that the max bitrate is unset.
-  if (initial_encoder_max_bitrate != -1) {
-    RTC_DLOG(LS_ERROR) << "ERROR: Initial encoder max bitrate = "
-                       << initial_encoder_max_bitrate << " which is <= 0!";
+DataRate GetDefaultMinVideoBitrate(VideoCodecType codec_type) {
+  if (codec_type == VideoCodecType::kVideoCodecAV1) {
+    return DataRate::BitsPerSec(kMinDefaultAv1BitrateBps);
   }
-  RTC_DLOG(LS_INFO) << "Using default encoder max bitrate = 10 Mbps";
-  return kFallbackMaxBitrateBps;
+  return DataRate::BitsPerSec(kDefaultMinVideoBitrateBps);
 }
 
-int GetDefaultMinVideoBitrateBps(VideoCodecType codec_type) {
-  if (codec_type == VideoCodecType::kVideoCodecAV1) {
-    return kMinDefaultAv1BitrateBps;
+std::optional<DataRate> GetConfiguredMaxVideoBitrate(
+    const VideoEncoderConfig& config) {
+  // If spatial layers are configured, the sum of their bitrate takes precedence
+  // over both simulcast and the global max_bitrate_bps.
+  if (!config.spatial_layers.empty()) {
+    return CalculateMaxBitrate(
+        config.spatial_layers, [](const SpatialLayer& l) { return l.active; },
+        [](const SpatialLayer& l) {
+          return l.maxBitrate > 0 ? DataRate::KilobitsPerSec(l.maxBitrate)
+                                  : DataRate::Zero();
+        },
+        DataRate::BitsPerSec(std::max(config.max_bitrate_bps, 0)));
   }
-  return kDefaultMinVideoBitrateBps;
+
+  // If simulcast layers are configured, the sum of their bitrate takes
+  // precedence over the global max_bitrate_bps.
+  if (!config.simulcast_layers.empty()) {
+    return CalculateMaxBitrate(
+        config.simulcast_layers, [](const VideoStream& s) { return s.active; },
+        [](const VideoStream& s) {
+          return s.max_bitrate_bps > 0 ? DataRate::BitsPerSec(s.max_bitrate_bps)
+                                       : DataRate::Zero();
+        },
+        DataRate::BitsPerSec(std::max(config.max_bitrate_bps, 0)));
+  }
+
+  // If single stream, use the global max_bitrate_bps.
+  if (config.max_bitrate_bps > 0) {
+    return DataRate::BitsPerSec(config.max_bitrate_bps);
+  }
+
+  return std::nullopt;
 }
 
 size_t CalculateMaxHeaderSize(const RtpConfig& config) {
@@ -367,6 +411,14 @@ std::unique_ptr<VideoStreamEncoderInterface> CreateVideoStreamEncoder(
 }
 
 bool HasActiveEncodings(const VideoEncoderConfig& config) {
+  if (!config.spatial_layers.empty()) {
+    for (const auto& layer : config.spatial_layers) {
+      if (layer.active) {
+        return true;
+      }
+    }
+    return false;
+  }
   for (const VideoStream& stream : config.simulcast_layers) {
     if (stream.active) {
       return true;
@@ -455,10 +507,11 @@ VideoSendStreamImpl::VideoSendStreamImpl(
       has_active_encodings_(HasActiveEncodings(encoder_config)),
       disable_padding_(true),
       max_padding_bitrate_(0),
-      encoder_min_bitrate_bps_(0),
-      encoder_max_bitrate_bps_(
-          GetInitialEncoderMaxBitrate(encoder_config.max_bitrate_bps)),
-      encoder_target_rate_bps_(0),
+      encoder_min_bitrate_(DataRate::Zero()),
+      encoder_max_bitrate_(GetConfiguredMaxVideoBitrate(encoder_config)),
+      configured_max_bitrate_(
+          DataRate::BitsPerSec(std::max(0, encoder_config.max_bitrate_bps))),
+      encoder_target_rate_(DataRate::Zero()),
       encoder_bitrate_priority_(encoder_config.bitrate_priority),
       encoder_av1_priority_bitrate_override_bps_(
           GetEncoderPriorityBitrate(config_.rtp.payload_name,
@@ -472,7 +525,6 @@ VideoSendStreamImpl::VideoSendStreamImpl(
   RTC_DCHECK_LE(config_.rtp.payload_type, 127);
   RTC_DCHECK(!config_.rtp.ssrcs.empty());
   RTC_DCHECK(transport_);
-  RTC_DCHECK_NE(encoder_max_bitrate_bps_, 0);
   RTC_LOG(LS_INFO) << "VideoSendStreamImpl: " << config_.ToString();
 
   RTC_CHECK(
@@ -565,6 +617,8 @@ void VideoSendStreamImpl::ReconfigureVideoEncoder(
                    << " VideoSendStream config: " << config_.ToString();
 
   has_active_encodings_ = HasActiveEncodings(config);
+  configured_max_bitrate_ =
+      DataRate::BitsPerSec(std::max(0, config.max_bitrate_bps));
   if (has_active_encodings_ && rtp_video_sender_->IsActive() && !IsRunning()) {
     StartupVideoSendStream();
   } else if (!has_active_encodings_ && IsRunning()) {
@@ -719,7 +773,7 @@ void VideoSendStreamImpl::SignalEncoderTimedOut() {
   // If the encoder has not produced anything the last kEncoderTimeOut and it
   // is supposed to, deregister as BitrateAllocatorObserver. This can happen
   // if a camera stops producing frames.
-  if (encoder_target_rate_bps_ > 0) {
+  if (!encoder_target_rate_.IsZero()) {
     RTC_LOG(LS_INFO) << "SignalEncoderTimedOut, Encoder timed out.";
     bitrate_allocator_->RemoveObserver(this);
   }
@@ -731,7 +785,7 @@ void VideoSendStreamImpl::OnBitrateAllocationUpdated(
   // the worker_queue_.
   auto task = [this, allocation] {
     RTC_DCHECK_RUN_ON(&thread_checker_);
-    if (encoder_target_rate_bps_ == 0) {
+    if (encoder_target_rate_.IsZero()) {
       return;
     }
     int64_t now_ms = env_.clock().TimeInMilliseconds();
@@ -791,8 +845,9 @@ void VideoSendStreamImpl::SignalEncoderActive() {
 
 MediaStreamAllocationConfig VideoSendStreamImpl::GetAllocationConfig() const {
   return MediaStreamAllocationConfig{
-      .min_bitrate_bps = static_cast<uint32_t>(encoder_min_bitrate_bps_),
-      .max_bitrate_bps = encoder_max_bitrate_bps_,
+      .min_bitrate_bps = encoder_min_bitrate_.bps<uint32_t>(),
+      .max_bitrate_bps =
+          encoder_max_bitrate_.value_or(DataRate::Zero()).bps<uint32_t>(),
       .pad_up_bitrate_bps =
           static_cast<uint32_t>(disable_padding_ ? 0 : max_padding_bitrate_),
       .priority_bitrate_bps = encoder_av1_priority_bitrate_override_bps_,
@@ -822,18 +877,18 @@ void VideoSendStreamImpl::OnEncoderConfigurationChanged(
 
     const std::optional<DataRate> experimental_min_bitrate =
         GetExperimentalMinVideoBitrate(env_.field_trials(), codec_type);
-    encoder_min_bitrate_bps_ =
-        experimental_min_bitrate
-            ? experimental_min_bitrate->bps()
-            : std::max(streams[0].min_bitrate_bps,
-                       GetDefaultMinVideoBitrateBps(codec_type));
+    if (experimental_min_bitrate.has_value()) {
+      encoder_min_bitrate_ = *experimental_min_bitrate;
+    } else {
+      DataRate min_bitrate =
+          streams[0].min_bitrate_bps > 0
+              ? DataRate::BitsPerSec(streams[0].min_bitrate_bps)
+              : DataRate::Zero();
+      encoder_min_bitrate_ =
+          std::max(min_bitrate, GetDefaultMinVideoBitrate(codec_type));
+    }
     double stream_bitrate_priority_sum = 0;
-    uint32_t encoder_max_bitrate_bps = 0;
     for (const auto& stream : streams) {
-      // We don't want to allocate more bitrate than needed to inactive streams.
-      if (stream.active) {
-        encoder_max_bitrate_bps += stream.max_bitrate_bps;
-      }
       if (stream.bitrate_priority) {
         RTC_DCHECK_GT(*stream.bitrate_priority, 0);
         stream_bitrate_priority_sum += *stream.bitrate_priority;
@@ -841,10 +896,21 @@ void VideoSendStreamImpl::OnEncoderConfigurationChanged(
     }
     RTC_DCHECK_GT(stream_bitrate_priority_sum, 0);
     encoder_bitrate_priority_ = stream_bitrate_priority_sum;
-    if (encoder_max_bitrate_bps > 0) {
-      encoder_max_bitrate_bps_ =
-          std::max(static_cast<uint32_t>(encoder_min_bitrate_bps_),
-                   encoder_max_bitrate_bps);
+
+    // Aggregate maximum bitrate from all active streams. Active streams missing
+    // individual caps will fall back to the global maximum configured bitrate
+    // cap.
+    encoder_max_bitrate_ = CalculateMaxBitrate(
+        streams, [](const VideoStream& s) { return s.active; },
+        [](const VideoStream& s) {
+          return s.max_bitrate_bps > 0 ? DataRate::BitsPerSec(s.max_bitrate_bps)
+                                       : DataRate::Zero();
+        },
+        configured_max_bitrate_);
+
+    if (encoder_max_bitrate_.has_value()) {
+      encoder_max_bitrate_ =
+          std::max(encoder_min_bitrate_, *encoder_max_bitrate_);
     }
 
     // TODO(bugs.webrtc.org/10266): Query the VideoBitrateAllocator instead.
@@ -931,24 +997,25 @@ uint32_t VideoSendStreamImpl::OnBitrateUpdated(BitrateAllocationUpdate update) {
       << "VideoSendStream::Start has not been called.";
 
   rtp_video_sender_->OnBitrateUpdated(update, stats_proxy_.GetSendFrameRate());
-  encoder_target_rate_bps_ = rtp_video_sender_->GetPayloadBitrateBps();
-  const uint32_t protection_bitrate_bps =
-      rtp_video_sender_->GetProtectionBitrateBps();
+  encoder_target_rate_ =
+      DataRate::BitsPerSec(rtp_video_sender_->GetPayloadBitrateBps());
+  const DataRate protection_bitrate =
+      DataRate::BitsPerSec(rtp_video_sender_->GetProtectionBitrateBps());
   DataRate link_allocation = DataRate::Zero();
-  if (encoder_target_rate_bps_ > protection_bitrate_bps) {
-    link_allocation =
-        DataRate::BitsPerSec(encoder_target_rate_bps_ - protection_bitrate_bps);
+  if (encoder_target_rate_ > protection_bitrate) {
+    link_allocation = encoder_target_rate_ - protection_bitrate;
   }
-  encoder_target_rate_bps_ =
-      std::min(encoder_max_bitrate_bps_, encoder_target_rate_bps_);
-  DataRate encoder_target_rate = DataRate::BitsPerSec(encoder_target_rate_bps_);
-  link_allocation = std::max(encoder_target_rate, link_allocation);
+  if (encoder_max_bitrate_.has_value()) {
+    encoder_target_rate_ =
+        std::min(*encoder_max_bitrate_, encoder_target_rate_);
+  }
+  link_allocation = std::max(encoder_target_rate_, link_allocation);
   video_stream_encoder_->OnBitrateUpdated(
-      encoder_target_rate, link_allocation,
+      encoder_target_rate_, link_allocation,
       dchecked_cast<uint8_t>(update.packet_loss_ratio * 256),
       update.round_trip_time.ms(), update.cwnd_reduce_ratio);
-  stats_proxy_.OnSetEncoderTargetRate(encoder_target_rate_bps_);
-  return protection_bitrate_bps;
+  stats_proxy_.OnSetEncoderTargetRate(encoder_target_rate_.bps());
+  return protection_bitrate.bps();
 }
 
 std::optional<DataRate> VideoSendStreamImpl::GetUsedRate() const {
diff --git a/video/video_send_stream_impl.h b/video/video_send_stream_impl.h
index 88845a1969..dd2d2dbb80 100644
--- a/video/video_send_stream_impl.h
+++ b/video/video_send_stream_impl.h
@@ -239,9 +239,13 @@ class VideoSendStreamImpl : public webrtc::VideoSendStream,
   bool has_active_encodings_ RTC_GUARDED_BY(thread_checker_);
   bool disable_padding_ RTC_GUARDED_BY(thread_checker_);
   int max_padding_bitrate_ RTC_GUARDED_BY(thread_checker_);
-  int encoder_min_bitrate_bps_ RTC_GUARDED_BY(thread_checker_);
-  uint32_t encoder_max_bitrate_bps_ RTC_GUARDED_BY(thread_checker_);
-  uint32_t encoder_target_rate_bps_ RTC_GUARDED_BY(thread_checker_);
+  DataRate encoder_min_bitrate_ RTC_GUARDED_BY(thread_checker_);
+  // The current maximum total bitrate across all active layers, if any.
+  std::optional<DataRate> encoder_max_bitrate_ RTC_GUARDED_BY(thread_checker_);
+  // The currently configured global max bitrate (from
+  // VideoEncoderConfig::max_bitrate_bps).
+  DataRate configured_max_bitrate_ RTC_GUARDED_BY(thread_checker_);
+  DataRate encoder_target_rate_ RTC_GUARDED_BY(thread_checker_);
   double encoder_bitrate_priority_ RTC_GUARDED_BY(thread_checker_);
   const int encoder_av1_priority_bitrate_override_bps_
       RTC_GUARDED_BY(thread_checker_);
diff --git a/video/video_send_stream_impl_unittest.cc b/video/video_send_stream_impl_unittest.cc
index 4bead83d26..ebef8fa688 100644
--- a/video/video_send_stream_impl_unittest.cc
+++ b/video/video_send_stream_impl_unittest.cc
@@ -36,6 +36,7 @@
 #include "api/video/encoded_image.h"
 #include "api/video/video_bitrate_allocation.h"
 #include "api/video/video_layers_allocation.h"
+#include "api/video_codecs/spatial_layer.h"
 #include "api/video_codecs/video_encoder.h"
 #include "call/bitrate_allocator.h"
 #include "call/rtp_config.h"
@@ -183,6 +184,8 @@ class VideoSendStreamImplTest : public ::testing::Test {
     encoder_config.simulcast_layers.push_back(VideoStream());
     encoder_config.simulcast_layers.back().active = true;
     encoder_config.simulcast_layers.back().bitrate_priority = 1.0;
+    encoder_config.simulcast_layers.back().width = 640;
+    encoder_config.simulcast_layers.back().height = 360;
     return encoder_config;
   }
 
@@ -291,7 +294,7 @@ TEST_F(VideoSendStreamImplTest,
        MaxBitrateCorrectIfActiveEncodingUpdatedAfterCreation) {
   VideoEncoderConfig one_active_encoding = TestVideoEncoderConfig();
   ASSERT_THAT(one_active_encoding.simulcast_layers, SizeIs(1));
-  one_active_encoding.max_bitrate_bps = 10'000'000;
+  one_active_encoding.max_bitrate_bps = 2'500'000;
   one_active_encoding.simulcast_layers[0].max_bitrate_bps = 2'000'000;
   VideoEncoderConfig no_active_encodings = one_active_encoding.Copy();
   no_active_encodings.simulcast_layers[0].active = false;
@@ -307,12 +310,12 @@ TEST_F(VideoSendStreamImplTest,
   time_controller_.AdvanceTime(TimeDelta::Zero());
 
   Sequence s;
-  // Expect codec max bitrate as max needed bitrate before the encoder has
-  // notifed about the actual send streams.
-  EXPECT_CALL(bitrate_allocator_,
-              AddObserver(vss_impl.get(),
-                          Field(&MediaStreamAllocationConfig::max_bitrate_bps,
-                                Eq(one_active_encoding.max_bitrate_bps))))
+  // Expect zero max bitrate before the encoder has notified about the actual
+  // send streams.
+  EXPECT_CALL(
+      bitrate_allocator_,
+      AddObserver(vss_impl.get(),
+                  Field(&MediaStreamAllocationConfig::max_bitrate_bps, Eq(0))))
       .InSequence(s);
 
   // Expect the sum of active encodings as max needed bitrate after
@@ -340,6 +343,183 @@ TEST_F(VideoSendStreamImplTest,
   vss_impl->Stop();
 }
 
+TEST_F(VideoSendStreamImplTest,
+       MaxBitrateFromGlobalConfigIfSimulcastLayersMissingBitrate) {
+  VideoEncoderConfig encoder_config = TestVideoEncoderConfig();
+  ASSERT_THAT(encoder_config.simulcast_layers, SizeIs(1));
+  encoder_config.max_bitrate_bps = 2500000;
+  encoder_config.simulcast_layers[0].max_bitrate_bps = 0;
+
+  auto vss_impl = CreateVideoSendStreamImpl(encoder_config.Copy());
+
+  Sequence s;
+  // Expect global max bitrate before the encoder has notified about the stream.
+  EXPECT_CALL(bitrate_allocator_,
+              AddObserver(vss_impl.get(),
+                          Field(&MediaStreamAllocationConfig::max_bitrate_bps,
+                                Eq(encoder_config.max_bitrate_bps))))
+      .InSequence(s);
+
+  vss_impl->Start();
+  time_controller_.AdvanceTime(TimeDelta::Zero());
+
+  // Trigger configuration change with stream still having 0 max bitrate.
+  std::vector<VideoStream> streams = encoder_config.simulcast_layers;
+  streams[0].max_bitrate_bps = 0;
+
+  // Expect that the global max bitrate is still used because the stream
+  // doesn't provide a max bitrate.
+  EXPECT_CALL(bitrate_allocator_,
+              AddObserver(vss_impl.get(),
+                          Field(&MediaStreamAllocationConfig::max_bitrate_bps,
+                                Eq(encoder_config.max_bitrate_bps))))
+      .InSequence(s);
+
+  encoder_queue_->PostTask([&] {
+    static_cast<VideoStreamEncoderInterface::EncoderSink*>(vss_impl.get())
+        ->OnEncoderConfigurationChanged(
+            streams, false, VideoEncoderConfig::ContentType::kRealtimeVideo,
+            /*min_transmit_bitrate_bps*/ 30000);
+  });
+  time_controller_.AdvanceTime(TimeDelta::Zero());
+
+  EXPECT_CALL(bitrate_allocator_, RemoveObserver(vss_impl.get())).InSequence(s);
+  vss_impl->Stop();
+}
+
+TEST_F(VideoSendStreamImplTest, MaxBitrateZeroIfNoMaxSetAnywhere) {
+  VideoEncoderConfig encoder_config = TestVideoEncoderConfig();
+  ASSERT_THAT(encoder_config.simulcast_layers, SizeIs(1));
+  encoder_config.max_bitrate_bps = 0;
+  encoder_config.simulcast_layers[0].max_bitrate_bps = 0;
+
+  auto vss_impl = CreateVideoSendStreamImpl(encoder_config.Copy());
+
+  Sequence s;
+  EXPECT_CALL(
+      bitrate_allocator_,
+      AddObserver(vss_impl.get(),
+                  Field(&MediaStreamAllocationConfig::max_bitrate_bps, Eq(0))))
+      .InSequence(s);
+
+  vss_impl->Start();
+  time_controller_.AdvanceTime(TimeDelta::Zero());
+
+  std::vector<VideoStream> streams = encoder_config.simulcast_layers;
+  streams[0].max_bitrate_bps = 0;
+
+  EXPECT_CALL(
+      bitrate_allocator_,
+      AddObserver(vss_impl.get(),
+                  Field(&MediaStreamAllocationConfig::max_bitrate_bps, Eq(0))))
+      .InSequence(s);
+
+  encoder_queue_->PostTask([&] {
+    static_cast<VideoStreamEncoderInterface::EncoderSink*>(vss_impl.get())
+        ->OnEncoderConfigurationChanged(
+            streams, false, VideoEncoderConfig::ContentType::kRealtimeVideo,
+            /*min_transmit_bitrate_bps*/ 30000);
+  });
+  time_controller_.AdvanceTime(TimeDelta::Zero());
+
+  EXPECT_CALL(bitrate_allocator_, RemoveObserver(vss_impl.get())).InSequence(s);
+  vss_impl->Stop();
+}
+
+TEST_F(VideoSendStreamImplTest,
+       MaxBitrateFromGlobalConfigIfSpatialLayersMissingBitrate) {
+  VideoEncoderConfig encoder_config = TestVideoEncoderConfig();
+  encoder_config.max_bitrate_bps = 2500000;
+
+  encoder_config.simulcast_layers.clear();
+  SpatialLayer layer;
+  layer.active = true;
+  layer.maxBitrate = 0;
+  encoder_config.spatial_layers.push_back(layer);
+  encoder_config.number_of_streams = 1;
+
+  auto vss_impl = CreateVideoSendStreamImpl(encoder_config.Copy());
+
+  Sequence s;
+  EXPECT_CALL(bitrate_allocator_,
+              AddObserver(vss_impl.get(),
+                          Field(&MediaStreamAllocationConfig::max_bitrate_bps,
+                                Eq(encoder_config.max_bitrate_bps))))
+      .InSequence(s);
+
+  vss_impl->Start();
+  time_controller_.AdvanceTime(TimeDelta::Zero());
+
+  std::vector<VideoStream> streams;
+  VideoStream stream;
+  stream.active = true;
+  stream.max_bitrate_bps = 0;
+  stream.bitrate_priority = 1.0;
+  streams.push_back(stream);
+
+  EXPECT_CALL(bitrate_allocator_,
+              AddObserver(vss_impl.get(),
+                          Field(&MediaStreamAllocationConfig::max_bitrate_bps,
+                                Eq(encoder_config.max_bitrate_bps))))
+      .InSequence(s);
+
+  encoder_queue_->PostTask([&] {
+    static_cast<VideoStreamEncoderInterface::EncoderSink*>(vss_impl.get())
+        ->OnEncoderConfigurationChanged(
+            streams, false, VideoEncoderConfig::ContentType::kRealtimeVideo,
+            /*min_transmit_bitrate_bps*/ 30000);
+  });
+  time_controller_.AdvanceTime(TimeDelta::Zero());
+
+  EXPECT_CALL(bitrate_allocator_, RemoveObserver(vss_impl.get())).InSequence(s);
+  vss_impl->Stop();
+}
+
+TEST_F(VideoSendStreamImplTest,
+       MaxBitrateUpdatedOnReconfigurationIfSimulcastLayersMissingBitrate) {
+  VideoEncoderConfig encoder_config = TestVideoEncoderConfig();
+  ASSERT_THAT(encoder_config.simulcast_layers, SizeIs(1));
+  encoder_config.max_bitrate_bps = 2500000;
+  encoder_config.simulcast_layers[0].max_bitrate_bps = 0;
+
+  auto vss_impl = CreateVideoSendStreamImpl(encoder_config.Copy());
+
+  Sequence s;
+  EXPECT_CALL(bitrate_allocator_,
+              AddObserver(vss_impl.get(),
+                          Field(&MediaStreamAllocationConfig::max_bitrate_bps,
+                                Eq(encoder_config.max_bitrate_bps))))
+      .InSequence(s);
+
+  vss_impl->Start();
+  time_controller_.AdvanceTime(TimeDelta::Zero());
+
+  VideoEncoderConfig new_config = encoder_config.Copy();
+  new_config.max_bitrate_bps = 3500000;
+
+  vss_impl->ReconfigureVideoEncoder(new_config.Copy());
+
+  std::vector<VideoStream> streams = new_config.simulcast_layers;
+  streams[0].max_bitrate_bps = 0;
+
+  EXPECT_CALL(bitrate_allocator_,
+              AddObserver(vss_impl.get(),
+                          Field(&MediaStreamAllocationConfig::max_bitrate_bps,
+                                Eq(new_config.max_bitrate_bps))))
+      .InSequence(s);
+
+  encoder_queue_->PostTask([&] {
+    static_cast<VideoStreamEncoderInterface::EncoderSink*>(vss_impl.get())
+        ->OnEncoderConfigurationChanged(
+            streams, false, VideoEncoderConfig::ContentType::kRealtimeVideo,
+            /*min_transmit_bitrate_bps*/ 30000);
+  });
+  time_controller_.AdvanceTime(TimeDelta::Zero());
+
+  EXPECT_CALL(bitrate_allocator_, RemoveObserver(vss_impl.get())).InSequence(s);
+  vss_impl->Stop();
+}
+
 TEST_F(VideoSendStreamImplTest,
        DoNotRegistersAsBitrateObserverOnStrayEncodedImage) {
   auto vss_impl = CreateVideoSendStreamImpl(TestVideoEncoderConfig());

From 13e7ebafcd8c3c02a55fd70903b34d234cfd32eb Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Tue, 24 Mar 2026 13:06:52 -0700
Subject: [PATCH 393/472] Roll chromium_revision 4acdf7133f..8327d89a47
 (1604168:1604334)

Change log: https://chromium.googlesource.com/chromium/src/+log/4acdf7133f..8327d89a47
Full diff: https://chromium.googlesource.com/chromium/src/+/4acdf7133f..8327d89a47

Changed dependencies
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/998e238962..6ca5eede75
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/0491a27dd1..3fef7fabf5
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/db50e5c540..4aeca26d44
* src/third_party/androidx/cipd: GIgdzzHTYcjiT8XrtVjd6rjspxYt0Hs0drpRjWcNwYUC..lH4R_xNZoL7YwnNSJYRSgU6ve0ypQAaGXUhHSMV8238C
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/2e5b239d79..416ff6088d
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/4f623df331..488b14fd17
* src/third_party/libvpx/source/libvpx: https://chromium.googlesource.com/webm/libvpx.git/+log/e1dd14963a..3fce57ecc9
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/3c997950a0..357268a3f6
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/465340f84f..2fc91c62a5
DEPS diff: https://chromium.googlesource.com/chromium/src/+/4acdf7133f..8327d89a47/DEPS

BUG=None

Change-Id: Ie75ae1268f3e1330a8078e2f87dfc549ece88fe0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/459800
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47246}
---
 DEPS | 20 ++++++++++----------
 1 file changed, 10 insertions(+), 10 deletions(-)

diff --git a/DEPS b/DEPS
index dc5f55b06f..af4b6fe028 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '4acdf7133f643cd2a965fe471f646599359c7d82',
+  'chromium_revision': '8327d89a4740050d03a5249ed4d37293128fe124',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@998e2389625373f61d37fbd6e60bea45ea30536c',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@6ca5eede75d2546ea22a3f555c272f6dfbd1dbd7',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@0491a27dd1af96bf568bd0911c6cae53d9e4b2dc',
+    'https://chromium.googlesource.com/chromium/src/testing@3fef7fabf503f7917199692b28a5a601e418aa1b',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@db50e5c54009efb3b2fbd3aae771080dc398c5f9',
+    'https://chromium.googlesource.com/chromium/src/third_party@4aeca26d44dde61760914d708098d96da28dce80',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -505,13 +505,13 @@ deps = {
   'src/third_party/crc32c/src':
     'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558',
   'src/third_party/depot_tools':
-    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@2e5b239d792f082ee7c008ffea310da64d5527eb',
+    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@416ff6088da5735e8321c5a297b9aef8b291bc5b',
   'src/third_party/ffmpeg':
     'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@946d97db8d906277085e361892b7efda5152e2f1',
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@4f623df3319df7e39a89fa2f4a198d43744917b7',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@488b14fd17934e8bbb469b6b383e10544ea0824d',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -609,11 +609,11 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '3c997950a05e9d0da33d95e99ad952ba275b21e4',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '357268a3f619b8bcc9b9b28b373dcf32f3df74b7',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
-    'https://chromium.googlesource.com/webm/libvpx.git@e1dd14963a005523d0eb3a72954921934b3a8be7',
+    'https://chromium.googlesource.com/webm/libvpx.git@3fce57ecc905d95a4619f33d09851d68c5a88663',
   'src/third_party/libyuv':
     'https://chromium.googlesource.com/libyuv/libyuv.git@30809ff64a9ca5e45f86439c0d474c2d3eef3d05',
   'src/third_party/lss': {
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@465340f84f9f81116a8e50542e7b34293d1a2567',
+    'https://chromium.googlesource.com/chromium/src/tools@2fc91c62a51235748aadfeca64d08266e2a72896',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'GIgdzzHTYcjiT8XrtVjd6rjspxYt0Hs0drpRjWcNwYUC',
+          'version': 'lH4R_xNZoL7YwnNSJYRSgU6ve0ypQAaGXUhHSMV8238C',
       },
     ],
     'condition': 'checkout_android and non_git_source',

From f85d783f006ecd9dc5453d7e9bbe99548f3fc26b Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Tue, 24 Mar 2026 17:03:37 -0700
Subject: [PATCH 394/472] Roll chromium_revision 8327d89a47..60f1c8a981
 (1604334:1604486)

Change log: https://chromium.googlesource.com/chromium/src/+log/8327d89a47..60f1c8a981
Full diff: https://chromium.googlesource.com/chromium/src/+/8327d89a47..60f1c8a981

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/d76036c311..4f3f3e34dd
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/6ca5eede75..0a0815e462
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/3fef7fabf5..ce12ef21dd
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/4aeca26d44..8016f4df49
* src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/28c233d2e1..bb7645f5e1
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/416ff6088d..e9512dd3ca
* src/third_party/ffmpeg: https://chromium.googlesource.com/chromium/third_party/ffmpeg.git/+log/946d97db8d..b5e18fb9da
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/357268a3f6..282dfad33b
* src/third_party/tflite/src: https://chromium.googlesource.com/external/github.com/tensorflow/tensorflow.git/+log/8cb40fbe19..8fd5278490
* src/third_party/xnnpack/src: https://chromium.googlesource.com/external/github.com/google/XNNPACK.git/+log/4334fa7cbd..97f3177fd8
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/2fc91c62a5..4e3155807c
DEPS diff: https://chromium.googlesource.com/chromium/src/+/8327d89a47..60f1c8a981/DEPS

BUG=None

Change-Id: I2efe4dd9e027e7df03defc0cfa6f26edb94126e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/459840
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47247}
---
 DEPS | 24 ++++++++++++------------
 1 file changed, 12 insertions(+), 12 deletions(-)

diff --git a/DEPS b/DEPS
index af4b6fe028..49192cfb42 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '8327d89a4740050d03a5249ed4d37293128fe124',
+  'chromium_revision': '60f1c8a981ff162b3fa73e347fa9aff4a15ff54f',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@d76036c3115ecdb183ae57c8f9e922760ea8e47c',
+    'https://chromium.googlesource.com/chromium/src/build@4f3f3e34ddfc6f4df4fa398cbe6996fcddc9bf3d',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@95ed44cf5f06dbb5861030b91c9db9ccb4316762',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@6ca5eede75d2546ea22a3f555c272f6dfbd1dbd7',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@0a0815e462da0c14df3023e37ccbcec3f29eeb93',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@3fef7fabf503f7917199692b28a5a601e418aa1b',
+    'https://chromium.googlesource.com/chromium/src/testing@ce12ef21dd2649e5642c1dac03e3a0a29a45763b',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@4aeca26d44dde61760914d708098d96da28dce80',
+    'https://chromium.googlesource.com/chromium/src/third_party@8016f4df4990fb16d34ea6aafce475bd20e82d45',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -331,7 +331,7 @@ deps = {
   'src/third_party/clang-format/script':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@c2725e0622e1a86d55f14514f2177a39efea4a0e',
   'src/third_party/compiler-rt/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@28c233d2e119c0a260083be9247c85ccd981b18d',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@bb7645f5e11c9c1d719a890fcb09ccfaaa14580f',
   'src/third_party/libc++/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@7ab65651aed6802d2599dcb7a73b1f82d5179d05',
   'src/third_party/libc++abi/src':
@@ -505,9 +505,9 @@ deps = {
   'src/third_party/crc32c/src':
     'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558',
   'src/third_party/depot_tools':
-    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@416ff6088da5735e8321c5a297b9aef8b291bc5b',
+    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@e9512dd3cacc96ff3f71def8a1a3019ab5ea1b17',
   'src/third_party/ffmpeg':
-    'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@946d97db8d906277085e361892b7efda5152e2f1',
+    'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@b5e18fb9da84e26ceef30d4e4886696bf59337c0',
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '357268a3f619b8bcc9b9b28b373dcf32f3df74b7',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '282dfad33b9390cf6d4c5eccd6d9c3c104c0c450',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@2fc91c62a51235748aadfeca64d08266e2a72896',
+    'https://chromium.googlesource.com/chromium/src/tools@4e3155807c68b64914f85bc24b4eae2165e757db',
 
   'src/third_party/espresso': {
       'packages': [
@@ -786,7 +786,7 @@ deps = {
   },
 
   'src/third_party/tflite/src':
-    Var('chromium_git') + '/external/github.com/tensorflow/tensorflow.git' + '@' + '8cb40fbe19ed5eb65da26078b8667e31e52211e9',
+    Var('chromium_git') + '/external/github.com/tensorflow/tensorflow.git' + '@' + '8fd527849069a358ad6c2980b9a9b34a53c53717',
 
   'src/third_party/turbine/cipd': {
       'packages': [
@@ -852,7 +852,7 @@ deps = {
     Var('chromium_git') + '/external/github.com/google/pthreadpool.git' + '@' + '9003ee6c137cea3b94161bd5c614fb43be523ee1',
 
   'src/third_party/xnnpack/src':
-    Var('chromium_git') + '/external/github.com/google/XNNPACK.git' + '@' + '4334fa7cbdf8e276db976792cd21d8614910d569',
+    Var('chromium_git') + '/external/github.com/google/XNNPACK.git' + '@' + '97f3177fd836fff03b48a886bb130591866ad7ca',
 
   'src/third_party/farmhash/src':
     Var('chromium_git') + '/external/github.com/google/farmhash.git' + '@' + '816a4ae622e964763ca0862d9dbd19324a1eaf45',

From a08f6ce124897b26e462687774d08a0f837d9ef3 Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Tue, 24 Mar 2026 21:05:05 -0700
Subject: [PATCH 395/472] Update WebRTC code version (2026-03-25T04:04:33).

Bug: None
Change-Id: I5fad6ddf6e4ca4ffd2f5a2445595ce3d3f3902d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/459880
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47248}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index c911c6f69e..8f1b8fd42b 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-24T04:06:53";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-25T04:04:33";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From 9c36e112fb1d962c11edc22ce356eb75b045c4a7 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Tue, 24 Mar 2026 15:56:52 +0100
Subject: [PATCH 396/472] Simplify thread yielding by removing support for
 high-priority tasks

Remove the `PostHighPriorityTask` method and the internal
`high_priority_messages_` queue from the `Thread` class.

Instead, change `IsYieldRequested()` to return true whenever there are
any pending tasks in the standard message queue, rather than only
checking for high-priority tasks.

This means that the sequence of batched operations is always executed
when the thread is idle, rather than blocking the thread, ensuring that
media-related calls to the same thread get processed as fast as
possible.

Bug: webrtc:42222804
Change-Id: I758c33dc2c33d79fc5e6beea1926529f78e49255
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456760
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47249}
---
 pc/scoped_operations_batcher.cc          |   2 +-
 pc/scoped_operations_batcher_unittest.cc |  13 ++-
 rtc_base/thread.cc                       |  29 +----
 rtc_base/thread.h                        |  15 +--
 rtc_base/thread_unittest.cc              | 139 ++++-------------------
 5 files changed, 34 insertions(+), 164 deletions(-)

diff --git a/pc/scoped_operations_batcher.cc b/pc/scoped_operations_batcher.cc
index 00c19cfe85..b5b277fea9 100644
--- a/pc/scoped_operations_batcher.cc
+++ b/pc/scoped_operations_batcher.cc
@@ -55,7 +55,7 @@ void ScopedOperationsBatcher::Run() {
           }
         }
         ++task_idx;
-        if (!target_thread_is_current && target_thread_->IsYieldRequested()) {
+        if (!target_thread_is_current && target_thread_->HasPendingTasks()) {
           return;
         }
       }
diff --git a/pc/scoped_operations_batcher_unittest.cc b/pc/scoped_operations_batcher_unittest.cc
index 0f188f7c11..03aa42d89d 100644
--- a/pc/scoped_operations_batcher_unittest.cc
+++ b/pc/scoped_operations_batcher_unittest.cc
@@ -75,7 +75,7 @@ TEST(ScopedOperationsBatcherTest, ExecutesReturnedTasksOnCallingThread) {
   EXPECT_EQ(return_task_thread, signaling_thread);
 }
 
-TEST(ScopedOperationsBatcherTest, YieldsToHighPriorityTasks) {
+TEST(ScopedOperationsBatcherTest, YieldsToOtherTasks) {
   auto target_thread = Thread::Create();
   target_thread->Start();
 
@@ -86,16 +86,17 @@ TEST(ScopedOperationsBatcherTest, YieldsToHighPriorityTasks) {
     batcher.push_back([&] { execution_order.push_back(1); });
     batcher.push_back([&] {
       execution_order.push_back(2);
-      // Post a high-priority task that should interrupt the batch.
-      target_thread->PostHighPriorityTask(
-          [&] { execution_order.push_back(3); });
+      // Post a task that should interrupt the batch since we now yield to any
+      // pending task.
+      target_thread->PostTask([&] { execution_order.push_back(3); });
     });
     batcher.push_back([&] { execution_order.push_back(4); });
     batcher.push_back([&] { execution_order.push_back(5); });
   }
 
-  // Expect the high priority task (3) to execute immediately after the task
-  // that posted it (2). The regular tasks (4, 5) should yield.
+  // Expect the task (3) to execute immediately after the task
+  // that posted it (2). The operations remaining in the batch (4, 5)
+  // should resume after the thread has processed the new task.
   EXPECT_EQ(execution_order, std::vector<int>({1, 2, 3, 4, 5}));
 }
 
diff --git a/rtc_base/thread.cc b/rtc_base/thread.cc
index 4bea295d6b..81a7b6b601 100644
--- a/rtc_base/thread.cc
+++ b/rtc_base/thread.cc
@@ -391,7 +391,6 @@ void Thread::DoDestroy() {
   ThreadManager::Remove(this);
   // Clear.
   CurrentTaskQueueSetter set_current(this);
-  high_priority_messages_.clear();
   messages_.clear();
   delayed_messages_ = {};
 }
@@ -442,13 +441,6 @@ absl::AnyInvocable<void() &&> Thread::Get(int cmsWait) {
         messages_.push_back(std::move(delayed_messages_.top().functor));
         delayed_messages_.pop();
       }
-      // Pull a high priority message off the queue, if available.
-      if (!high_priority_messages_.empty()) {
-        absl::AnyInvocable<void() &&> task =
-            std::move(high_priority_messages_.front());
-        high_priority_messages_.pop_front();
-        return task;
-      }
       // Pull a message off the message queue, if available.
       if (!messages_.empty()) {
         absl::AnyInvocable<void() &&> task = std::move(messages_.front());
@@ -494,15 +486,6 @@ absl::AnyInvocable<void() &&> Thread::Get(int cmsWait) {
 void Thread::PostTaskImpl(absl::AnyInvocable<void() &&> task,
                           const PostTaskTraits& /* traits */,
                           const Location& /* location */) {
-  PostTaskInternal(std::move(task), /*high_priority=*/false);
-}
-
-void Thread::PostHighPriorityTask(absl::AnyInvocable<void() &&> task) {
-  PostTaskInternal(std::move(task), /*high_priority=*/true);
-}
-
-void Thread::PostTaskInternal(absl::AnyInvocable<void() &&> task,
-                              bool high_priority) {
   if (IsQuitting()) {
     return;
   }
@@ -513,11 +496,7 @@ void Thread::PostTaskInternal(absl::AnyInvocable<void() &&> task,
 
   {
     MutexLock lock(&mutex_);
-    if (high_priority) {
-      high_priority_messages_.push_back(std::move(task));
-    } else {
-      messages_.push_back(std::move(task));
-    }
+    messages_.push_back(std::move(task));
     WakeUpSocketServer();
   }
 }
@@ -554,7 +533,7 @@ void Thread::PostDelayedTaskImpl(absl::AnyInvocable<void() &&> task,
 int Thread::GetDelay() {
   MutexLock lock(&mutex_);
 
-  if (!high_priority_messages_.empty() || !messages_.empty())
+  if (!messages_.empty())
     return 0;
 
   if (!delayed_messages_.empty()) {
@@ -990,7 +969,7 @@ AutoSocketServerThread::~AutoSocketServerThread() {
   }
 }
 
-bool Thread::IsYieldRequested() const {
+bool Thread::HasPendingTasks() const {
   RTC_DCHECK_RUN_ON(this);
 #if RTC_DCHECK_IS_ON
   // If you've hit this, then there's a cooperative task running from inside a
@@ -998,7 +977,7 @@ bool Thread::IsYieldRequested() const {
   RTC_DCHECK_EQ(running_synchronous_blocking_call_count_, 0);
 #endif
   MutexLock lock(&mutex_);
-  return !high_priority_messages_.empty();
+  return !messages_.empty();
 }
 
 }  // namespace webrtc
diff --git a/rtc_base/thread.h b/rtc_base/thread.h
index 0629461366..24372722b3 100644
--- a/rtc_base/thread.h
+++ b/rtc_base/thread.h
@@ -317,11 +317,12 @@ class RTC_LOCKABLE RTC_EXPORT Thread : public TaskQueueBase {
   // ProcessMessages occasionally.
   virtual void Run();
 
-  // Called from cooperative tasks to know if they should yield.
+  // Returns true if there are pending tasks in the message queue.
+  // Cooperative tasks can use this to know if they should yield.
   // If a task yields, it is up to the task itself how or if to
   // continue the ongoing operation. Typically this can be handled
   // by using PostTask() to queue up a continuation task.
-  bool IsYieldRequested() const;
+  bool HasPendingTasks() const;
 
   // Convenience method to invoke a functor on another thread.
   // Blocks the current thread until execution is complete.
@@ -345,12 +346,6 @@ class RTC_LOCKABLE RTC_EXPORT Thread : public TaskQueueBase {
     return result;
   }
 
-  // Posts a task that jumps the queue (gets added as the next task to execute,
-  // ahead of other pending tasks). An internal yielding request flag is also
-  // raised that cooperative aware tasks can check by calling
-  // `IsYieldRequested()` and immediately exit.
-  void PostHighPriorityTask(absl::AnyInvocable<void() &&> task);
-
   // Allows BlockingCall to specified `thread`. Thread never will be
   // dereferenced and will be used only for reference-based comparison, so
   // instance can be safely deleted. If NDEBUG is defined and RTC_DCHECK_IS_ON
@@ -476,8 +471,6 @@ class RTC_LOCKABLE RTC_EXPORT Thread : public TaskQueueBase {
  private:
   static const int kSlowDispatchLoggingThreshold = 50;  // 50 ms
 
-  void PostTaskInternal(absl::AnyInvocable<void() &&> task, bool high_priority);
-
   // Get() will process I/O until:
   //  1) A task is available (returns it)
   //  2) cmsWait seconds have elapsed (returns empty task)
@@ -513,8 +506,6 @@ class RTC_LOCKABLE RTC_EXPORT Thread : public TaskQueueBase {
   void ClearCurrentTaskQueue();
 
   std::deque<absl::AnyInvocable<void() &&>> messages_ RTC_GUARDED_BY(mutex_);
-  std::deque<absl::AnyInvocable<void() &&>> high_priority_messages_
-      RTC_GUARDED_BY(mutex_);
   std::priority_queue<DelayedMessage> delayed_messages_ RTC_GUARDED_BY(mutex_);
   uint32_t delayed_next_num_ RTC_GUARDED_BY(mutex_);
 #if RTC_DCHECK_IS_ON
diff --git a/rtc_base/thread_unittest.cc b/rtc_base/thread_unittest.cc
index 97ed808817..ce011e7a29 100644
--- a/rtc_base/thread_unittest.cc
+++ b/rtc_base/thread_unittest.cc
@@ -873,155 +873,54 @@ TEST(ThreadPostDelayedTaskTest, IsCurrentTaskQueue) {
   EXPECT_EQ(TaskQueueBase::Current(), current_tq);
 }
 
-// Verifies that posting a high-priority task sets the "yield requested" flag
-// on the target thread, allowing a cooperative task to observe it.
-TEST(ThreadCooperativeTest, HighPriorityTaskTriggersYieldRequest) {
+// Uses `HasPendingTasks()` to detect when to yield to another posted task.
+TEST(ThreadCooperativeTest, TaskTriggersHasPendingTasks) {
   test::RunLoop loop;
   auto thread = Thread::Create();
   thread->Start();
 
-  bool yield_requested = false;
+  bool was_interrupted = false;
   Event task_started;
 
-  // Post a long running task that checks for yield
+  // Post a long running task that checks for pending tasks.
   thread->PostTask(
-      [&yield_requested, &loop, &task_started, thread = thread.get()] {
+      [&was_interrupted, &loop, &task_started, thread = thread.get()] {
         task_started.Set();
-        while (!thread->IsYieldRequested()) {
+        while (!thread->HasPendingTasks()) {
           // Busy loop/simulated work
         }
-        loop.PostTask([&yield_requested, &loop] {
-          yield_requested = true;
+        loop.PostTask([&was_interrupted, &loop] {
+          was_interrupted = true;
           loop.Quit();
         });
       });
 
-  // Wait for the task to start to ensure that the high priority task doesn't
+  // Wait for the task to start to ensure that the task doesn't
   // run first.
   task_started.Wait(Event::kForever);
 
-  // Post a high priority task that interrupts the busy loop.
-  thread->PostHighPriorityTask([] {});
+  // Post a task that interrupts the busy loop.
+  thread->PostTask([] {});
 
   loop.Run();
-  EXPECT_TRUE(yield_requested);
+  EXPECT_TRUE(was_interrupted);
 }
 
-// Verifies that a high-priority task is executed before a regular priority task
-// that was posted earlier, if the thread was busy.
-TEST(ThreadCooperativeTest, HighPriorityTaskRunsFirst) {
-  test::RunLoop loop;
-  auto thread = Thread::Create();
-  thread->Start();
-
-  std::vector<std::string> execution_order;
-  Event task1_started;
-  Event continue_execution;
-
-  // Task 1: Blocks the thread until signaled.
-  thread->PostTask([&] {
-    task1_started.Set();
-    continue_execution.Wait(Event::kForever);
-    loop.PostTask([&] { execution_order.push_back("Task1"); });
-  });
-
-  task1_started.Wait(Event::kForever);
-
-  // Task 2: Regular priority.
-  thread->PostTask(
-      [&] { loop.PostTask([&] { execution_order.push_back("Task2"); }); });
-
-  // Task 3: High priority. High priority tasks should run before "Task2".
-  thread->PostHighPriorityTask(
-      [&] { loop.PostTask([&] { execution_order.push_back("Task3"); }); });
-
-  // Signal Task 1 to continue.
-  continue_execution.Set();
-
-  // Now post a task to quit the loop.
-  thread->PostTask([&] { loop.PostTask([&] { loop.Quit(); }); });
-
-  loop.Run();
-
-  // Expect Task1 (resuming), Task3 (High Prio), Task2 (Low Prio - pushed to
-  // back).
-  EXPECT_THAT(execution_order, ElementsAre("Task1", "Task3", "Task2"));
-}
-
-// Verifies that multiple high-priority tasks execute in the sequence they were
-// posted (FIFO order).
-TEST(ThreadCooperativeTest, HighPriorityTasksExecuteInSequence) {
-  test::RunLoop loop;
-  auto thread = Thread::Create();
-  thread->Start();
-
-  std::vector<std::string> execution_order;
-  Event task1_started;
-  Event continue_execution;
-
-  // Block the thread.
-  thread->PostTask([&] {
-    task1_started.Set();
-    continue_execution.Wait(Event::kForever);
-  });
-
-  task1_started.Wait(Event::kForever);
-
-  // Post a normal priority task first to ensure high priority tasks run before
-  // it.
-  thread->PostTask(
-      [&] { loop.PostTask([&] { execution_order.push_back("Normal1"); }); });
-
-  // Post multiple high priority tasks while the thread is blocked.
-  thread->PostHighPriorityTask(
-      [&] { loop.PostTask([&] { execution_order.push_back("HP1"); }); });
-  thread->PostHighPriorityTask(
-      [&] { loop.PostTask([&] { execution_order.push_back("HP2"); }); });
-  thread->PostHighPriorityTask(
-      [&] { loop.PostTask([&] { execution_order.push_back("HP3"); }); });
-
-  // Unblock the thread.
-  continue_execution.Set();
-
-  // Quit after all tasks.
-  thread->PostTask([&] { loop.PostTask([&] { loop.Quit(); }); });
-
-  loop.Run();
-
-  EXPECT_THAT(execution_order, ElementsAre("HP1", "HP2", "HP3", "Normal1"));
-}
-
-TEST(ThreadCooperativeTest, YieldRequestedClearedAfterHighPriorityTask) {
+TEST(ThreadCooperativeTest, HasPendingTasksClearedAfterTask) {
   std::unique_ptr<Thread> thread(Thread::Create());
   thread->Start();
 
   // Initially false.
-  thread->BlockingCall([&] { EXPECT_FALSE(thread->IsYieldRequested()); });
+  thread->BlockingCall([&] { EXPECT_FALSE(thread->HasPendingTasks()); });
 
-  // Post high priority task.
-  thread->PostHighPriorityTask([&] {});
+  // Post task.
+  thread->PostTask([&] {});
 
-  // Post normal task.
-  thread->BlockingCall([&] { EXPECT_FALSE(thread->IsYieldRequested()); });
+  // Use `BlockingCall` to post normal task which implicitly blocks and waits
+  // for its functor to run, at which point the queue will be empty again.
+  thread->BlockingCall([&] { EXPECT_FALSE(thread->HasPendingTasks()); });
 }
 
-#if GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID) && RTC_DCHECK_IS_ON
-TEST(ThreadCooperativeDeathTest, YieldInSynchronousBlockingCallTriggersDeath) {
-  std::unique_ptr<Thread> background_thread = Thread::Create();
-  background_thread->Start();
-  background_thread->BlockingCall([&] {
-    // Yielding from a top-level synchronous blocking call is allowed.
-    background_thread->IsYieldRequested();
-
-    // Call a nested BlockingCall.
-    background_thread->BlockingCall([&] {
-      // This violates the "no yielding in blocking call" rule.
-      EXPECT_DEATH(background_thread->IsYieldRequested(), "");
-    });
-  });
-}
-#endif
-
 class ThreadFactory : public TaskQueueFactory {
  public:
   std::unique_ptr<TaskQueueBase, TaskQueueDeleter> CreateTaskQueue(

From a875e8cb67a106a699fe0aef0bb008669bcacbb7 Mon Sep 17 00:00:00 2001
From: Jeremy Leconte <jleconte@webrtc.org>
Date: Wed, 25 Mar 2026 07:31:06 +0100
Subject: [PATCH 397/472] build: clean up api/BUILD.gn
 libjingle_peerconnection_api target

this is a follow up for https://webrtc-review.git.corp.google.com/c/src/+/455821

Bug: webrtc:467294026
Change-Id: I3d7c5129e58b9f5fd6d2ccb958458823de89ac00
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/457900
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#47250}
---
 api/BUILD.gn | 24 ------------------------
 1 file changed, 24 deletions(-)

diff --git a/api/BUILD.gn b/api/BUILD.gn
index d7f763f96c..5e0e0f4fe0 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -598,51 +598,34 @@ rtc_library("libjingle_peerconnection_api") {
     ":audio_options_api",
     ":candidate",
     ":data_channel_event_observer_interface",
-    ":data_channel_interface",
     ":dtls_transport_interface",
     ":dtmf_sender_interface",
     ":fec_controller_api",
     ":field_trials_view",
     ":frame_transformer_interface",
     ":ice_transport_interface",
-    ":jsep",
-    ":legacy_stats_types",
     ":libjingle_logging_api",
     ":local_network_access_permission",
-    ":make_ref_counted",
     ":media_stream_interface",
     ":network_state_predictor_api",
     ":packet_socket_factory",
-    ":peer_connection_interface",
     ":priority",
     ":ref_count",
     ":rtc_error",
     ":rtc_stats_api",
-    ":rtp_packet_info",
     ":rtp_parameters",
-    ":rtp_receiver_interface",
-    ":rtp_sender_interface",
     ":rtp_transceiver_direction",
-    ":rtp_transceiver_interface",
     ":rtp_transport_factory",
     ":scoped_refptr",
-    ":sctp_transport_interface",
     ":sequence_checker",
-    ":set_local_description_observer_interface",
     ":sframe_decrypter_interface",
     ":sframe_encrypter_interface",
     ":sframe_types",
     ":turn_customizer",
-    "../call:rtp_interfaces",
-    "../media:media_engine",
-    "../p2p:connection",
     "../p2p:dtls_transport_factory",
     "../p2p:port",
     "../p2p:port_allocator",
-    "../pc:media_factory",
-    "../pc:session_description",
     "../rtc_base:copy_on_write_buffer",
-    "../rtc_base:logging",
     "../rtc_base:macromagic",
     "../rtc_base:network",
     "../rtc_base:network_constants",
@@ -650,7 +633,6 @@ rtc_library("libjingle_peerconnection_api") {
     "../rtc_base:socket_factory",
     "../rtc_base:ssl",
     "../rtc_base:ssl_adapter",
-    "../rtc_base:stringutils",
     "../rtc_base/system:no_unique_address",
     "../rtc_base/system:plan_b_only",
     "adaptation:resource_adaptation_api",
@@ -666,20 +648,14 @@ rtc_library("libjingle_peerconnection_api") {
     "metronome",
     "neteq:neteq_api",
     "rtc_event_log:rtc_event_log_factory_interface",
-    "task_queue",
     "transport:bandwidth_estimation_settings",
     "transport:bitrate_settings",
     "transport:enums",
     "transport:network_control",
     "transport:sctp_transport_factory_interface",
     "transport/rtp:rtp_source",
-    "units:data_rate",
     "units:time_delta",
-    "units:timestamp",
-    "video:encoded_image",
     "video:video_bitrate_allocator_factory",
-    "video:video_frame",
-    "video:video_rtp_headers",
     "video_codecs:video_codecs_api",
     "//third_party/abseil-cpp/absl/algorithm:container",
     "//third_party/abseil-cpp/absl/base:core_headers",

From 5e177d217195edbcdf1f54f928a942e1696197d8 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Tue, 24 Mar 2026 15:22:38 +0100
Subject: [PATCH 398/472] Harden SimpleStringBuilder safety checks

Replace RTC_DCHECK with RTC_CHECK in SimpleStringBuilder to ensure
safety constraints are enforced in all build configurations, including
release builds. This change prevents undefined behavior or silent
truncation by crashing when a buffer overflow is detected, rather than
only performing these checks in debug modes.

The modifications include:
* Updating SimpleStringBuilder to use RTC_CHECK for consistency and
  bounds verification.
* Simplifying RtpExtension::ToString and VideoReceiveStream2 by
  migrating from SimpleStringBuilder to StringBuilder.
* Simplify+clarify the StringBuilder::Release implementation.
* Update unit tests to reflect that overflow now consistently results
  in a crash.

Bug: chromium:486536241
Change-Id: Ia067508e2b9409154ad4afdbe4c0971a1dd32e1e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/459441
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47251}
---
 api/rtp_parameters.cc                       |  8 ++---
 rtc_base/BUILD.gn                           |  1 +
 rtc_base/strings/string_builder.cc          | 10 +++---
 rtc_base/strings/string_builder.h           |  7 ++--
 rtc_base/strings/string_builder_unittest.cc | 38 +--------------------
 video/video_receive_stream2.cc              |  3 +-
 6 files changed, 13 insertions(+), 54 deletions(-)

diff --git a/api/rtp_parameters.cc b/api/rtp_parameters.cc
index 3a36bfe00a..b6b2f89168 100644
--- a/api/rtp_parameters.cc
+++ b/api/rtp_parameters.cc
@@ -18,7 +18,6 @@
 
 #include "absl/strings/ascii.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/rtc_error.h"
 #include "api/rtp_transceiver_direction.h"
 #include "media/base/media_constants.h"
@@ -206,15 +205,14 @@ RtpParameters::RtpParameters(const RtpParameters& rhs) = default;
 RtpParameters::~RtpParameters() = default;
 
 std::string RtpExtension::ToString() const {
-  char buf[256];
-  SimpleStringBuilder sb(buf);
+  StringBuilder sb;
   sb << "{uri: " << uri;
   sb << ", id: " << id;
   if (encrypt) {
     sb << ", encrypt";
   }
-  sb << '}';
-  return sb.str();
+  sb << "}";
+  return sb.Release();
 }
 
 bool RtpExtension::IsSupportedForAudio(absl::string_view uri) {
diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
index b98dcd6862..29888f043b 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -2184,6 +2184,7 @@ if (rtc_include_tests) {
         "network:received_packet",
         "synchronization:mutex",
         "task_utils:repeating_task",
+        "//testing/gtest",
         "//third_party/abseil-cpp/absl/algorithm:container",
         "//third_party/abseil-cpp/absl/base:core_headers",
         "//third_party/abseil-cpp/absl/memory",
diff --git a/rtc_base/strings/string_builder.cc b/rtc_base/strings/string_builder.cc
index f52445a8f4..af8e7e85c2 100644
--- a/rtc_base/strings/string_builder.cc
+++ b/rtc_base/strings/string_builder.cc
@@ -24,7 +24,7 @@ namespace webrtc {
 SimpleStringBuilder::SimpleStringBuilder(std::span<char> buffer)
     : buffer_(buffer) {
   buffer_[0] = '\0';
-  RTC_DCHECK(IsConsistent());
+  RTC_CHECK(IsConsistent());
 }
 
 SimpleStringBuilder& SimpleStringBuilder::operator<<(char ch) {
@@ -32,13 +32,13 @@ SimpleStringBuilder& SimpleStringBuilder::operator<<(char ch) {
 }
 
 SimpleStringBuilder& SimpleStringBuilder::operator<<(absl::string_view str) {
-  RTC_DCHECK_LT(size_ + str.length(), buffer_.size())
+  RTC_CHECK_LT(size_ + str.length(), buffer_.size())
       << "Buffer size was insufficient";
   const size_t chars_added = SafeMin(str.length(), buffer_.size() - size_ - 1);
   memcpy(&buffer_[size_], str.data(), chars_added);
   size_ += chars_added;
   buffer_[size_] = '\0';
-  RTC_DCHECK(IsConsistent());
+  RTC_CHECK(IsConsistent());
   return *this;
 }
 
@@ -98,7 +98,7 @@ SimpleStringBuilder& SimpleStringBuilder::AppendFormat(const char* fmt, ...) {
   if (len >= 0) {
     const size_t chars_added = SafeMin(len, buffer_.size() - 1 - size_);
     size_ += chars_added;
-    RTC_DCHECK_EQ(len, chars_added) << "Buffer size was insufficient";
+    RTC_CHECK_EQ(len, chars_added) << "Buffer size was insufficient";
   } else {
     // This should never happen, but we're paranoid, so re-write the
     // terminator in case vsnprintf() overwrote it.
@@ -106,7 +106,7 @@ SimpleStringBuilder& SimpleStringBuilder::AppendFormat(const char* fmt, ...) {
     buffer_[size_] = '\0';
   }
   va_end(args);
-  RTC_DCHECK(IsConsistent());
+  RTC_CHECK(IsConsistent());
   return *this;
 }
 
diff --git a/rtc_base/strings/string_builder.h b/rtc_base/strings/string_builder.h
index 41efe9aa42..4a3df25ba3 100644
--- a/rtc_base/strings/string_builder.h
+++ b/rtc_base/strings/string_builder.h
@@ -155,11 +155,8 @@ class StringBuilder {
 
   size_t size() const { return str_.size(); }
 
-  std::string Release() {
-    std::string ret = std::move(str_);
-    str_.clear();
-    return ret;
-  }
+  // Moves out the internal std::string.
+  std::string Release() { return std::move(str_); }
 
   // Allows appending a printf style formatted string.
   StringBuilder& AppendFormat(const char* fmt, ...)
diff --git a/rtc_base/strings/string_builder_unittest.cc b/rtc_base/strings/string_builder_unittest.cc
index cb45313062..4764ac4cea 100644
--- a/rtc_base/strings/string_builder_unittest.cc
+++ b/rtc_base/strings/string_builder_unittest.cc
@@ -14,7 +14,6 @@
 #include <string>
 
 #include "absl/strings/string_view.h"
-#include "rtc_base/checks.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
 
@@ -81,18 +80,13 @@ TEST(SimpleStringBuilder, CanUseAbslStringForCustomTypes) {
 
 // These tests are safe to run if we have death test support or if DCHECKs are
 // off.
-#if (GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)) || !RTC_DCHECK_IS_ON
+#if GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
 
 TEST(SimpleStringBuilderDeathTest, BufferOverrunConstCharP) {
   char sb_buf[4];
   SimpleStringBuilder sb(sb_buf);
   const char* const msg = "This is just too much";
-#if RTC_DCHECK_IS_ON
   EXPECT_DEATH(sb << msg, "");
-#else
-  sb << msg;
-  EXPECT_THAT(sb.str(), ::testing::StrEq("Thi"));
-#endif
 }
 
 TEST(SimpleStringBuilderDeathTest, BufferOverrunStdString) {
@@ -100,41 +94,21 @@ TEST(SimpleStringBuilderDeathTest, BufferOverrunStdString) {
   SimpleStringBuilder sb(sb_buf);
   sb << 12;
   const std::string msg = "Aw, come on!";
-#if RTC_DCHECK_IS_ON
   EXPECT_DEATH(sb << msg, "");
-#else
-  sb << msg;
-  EXPECT_THAT(sb.str(), ::testing::StrEq("12A"));
-#endif
 }
 
 TEST(SimpleStringBuilderDeathTest, BufferOverrunInt) {
   char sb_buf[4];
   SimpleStringBuilder sb(sb_buf);
   constexpr int num = -12345;
-#if RTC_DCHECK_IS_ON
   EXPECT_DEATH(sb << num, "");
-#else
-  sb << num;
-  // If we run into the end of the buffer, resonable results are either that
-  // the append has no effect or that it's truncated at the point where the
-  // buffer ends.
-  EXPECT_THAT(sb.str(),
-              ::testing::AnyOf(::testing::StrEq(""), ::testing::StrEq("-12")));
-#endif
 }
 
 TEST(SimpleStringBuilderDeathTest, BufferOverrunDouble) {
   char sb_buf[5];
   SimpleStringBuilder sb(sb_buf);
   constexpr double num = 123.456;
-#if RTC_DCHECK_IS_ON
   EXPECT_DEATH(sb << num, "");
-#else
-  sb << num;
-  EXPECT_THAT(sb.str(),
-              ::testing::AnyOf(::testing::StrEq(""), ::testing::StrEq("123.")));
-#endif
 }
 
 TEST(SimpleStringBuilderDeathTest, BufferOverrunConstCharPAlreadyFull) {
@@ -142,12 +116,7 @@ TEST(SimpleStringBuilderDeathTest, BufferOverrunConstCharPAlreadyFull) {
   SimpleStringBuilder sb(sb_buf);
   sb << 123;
   const char* const msg = "This is just too much";
-#if RTC_DCHECK_IS_ON
   EXPECT_DEATH(sb << msg, "");
-#else
-  sb << msg;
-  EXPECT_THAT(sb.str(), ::testing::StrEq("123"));
-#endif
 }
 
 TEST(SimpleStringBuilderDeathTest, BufferOverrunIntAlreadyFull) {
@@ -155,12 +124,7 @@ TEST(SimpleStringBuilderDeathTest, BufferOverrunIntAlreadyFull) {
   SimpleStringBuilder sb(sb_buf);
   sb << "xyz";
   constexpr int num = -12345;
-#if RTC_DCHECK_IS_ON
   EXPECT_DEATH(sb << num, "");
-#else
-  sb << num;
-  EXPECT_THAT(sb.str(), ::testing::StrEq("xyz"));
-#endif
 }
 
 #endif
diff --git a/video/video_receive_stream2.cc b/video/video_receive_stream2.cc
index 93ceb84ef3..8e3f6a8132 100644
--- a/video/video_receive_stream2.cc
+++ b/video/video_receive_stream2.cc
@@ -573,8 +573,7 @@ void VideoReceiveStream2::CreateAndRegisterExternalDecoder(
   // dumped video, since it's developers-only feature for debugging.
   absl::c_replace(decoded_output_file, ';', '/');
   if (!decoded_output_file.empty()) {
-    char filename_buffer[256];
-    SimpleStringBuilder ssb(filename_buffer);
+    StringBuilder ssb;
     ssb << decoded_output_file << "/webrtc_receive_stream_" << remote_ssrc()
         << "-" << env_.clock().TimeInMicroseconds() << ".ivf";
     video_decoder =

From 277d8b2db8c3d7047ad763070ff7dea452c956ba Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Wed, 25 Mar 2026 01:03:24 -0700
Subject: [PATCH 399/472] Roll chromium_revision 60f1c8a981..d3079a981f
 (1604486:1604639)

Change log: https://chromium.googlesource.com/chromium/src/+log/60f1c8a981..d3079a981f
Full diff: https://chromium.googlesource.com/chromium/src/+/60f1c8a981..d3079a981f

Changed dependencies
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/0a0815e462..087d2d6a3f
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/ce12ef21dd..7274f92de1
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/8016f4df49..7ad8972446
* src/third_party/android_deps/autorolled/cipd: NDkHoZFBi462KHX9MpNy-3MfJnllAi8MytUuzRobeeQC..ioujkn6x8xwFvfYkiQGrRb9lX4Khn3Thhv3nsoN6lrQC
* src/third_party/androidx/cipd: lH4R_xNZoL7YwnNSJYRSgU6ve0ypQAaGXUhHSMV8238C..TUGHGHq_2U0C6NzCTxxwEIYyf1o0qWVn7Ta_T8xndiAC
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/e9512dd3ca..ce1ebad2c3
* src/third_party/fuzztest/src: https://chromium.googlesource.com/external/github.com/google/fuzztest.git/+log/3f730151ee..36b8640bfd
* src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/488b14fd17..5e9fb9cbfb
* src/third_party/libpfm4/src: https://chromium.googlesource.com/external/git.code.sf.net/p/perfmon2/libpfm4.git/+log/964baf9d35..41878eab48
* src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/65d4df5c8e..adccc44307
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/282dfad33b..54c1a44a6e
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/4e3155807c..ff09f3d9ee
DEPS diff: https://chromium.googlesource.com/chromium/src/+/60f1c8a981..d3079a981f/DEPS

BUG=None

Change-Id: Ie2fcb0aca3ed6e8990a27deab0c12acb8000d790
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/459921
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47252}
---
 DEPS | 26 +++++++++++++-------------
 1 file changed, 13 insertions(+), 13 deletions(-)

diff --git a/DEPS b/DEPS
index 49192cfb42..b7df5b3b06 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '60f1c8a981ff162b3fa73e347fa9aff4a15ff54f',
+  'chromium_revision': 'd3079a981fd07f619d0c1fc18d30b9208749b94b',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@0a0815e462da0c14df3023e37ccbcec3f29eeb93',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@087d2d6a3fcdf817624bc6a547becb6a50892a0e',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@ce12ef21dd2649e5642c1dac03e3a0a29a45763b',
+    'https://chromium.googlesource.com/chromium/src/testing@7274f92de1882295073078afb648515ced0185ea',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@8016f4df4990fb16d34ea6aafce475bd20e82d45',
+    'https://chromium.googlesource.com/chromium/src/third_party@7ad8972446f6cc626c193804fdb71d5413b0cea7',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -337,7 +337,7 @@ deps = {
   'src/third_party/libc++abi/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@8f11bb1d4438d0239d0dfc1bd9456a9f31629dda',
   'src/third_party/llvm-libc/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@65d4df5c8e930138f416611ae540b5fd9fe3a6c6',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@adccc443070c58badd6414fd9a4380ca8c78e7c4',
   'src/third_party/libunwind/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@db838d918570d4e381ecf9f5cc70a0098c9c2cd6',
 
@@ -505,13 +505,13 @@ deps = {
   'src/third_party/crc32c/src':
     'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558',
   'src/third_party/depot_tools':
-    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@e9512dd3cacc96ff3f71def8a1a3019ab5ea1b17',
+    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@ce1ebad2c35c9387186f01d77edeea28a254c955',
   'src/third_party/ffmpeg':
     'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@b5e18fb9da84e26ceef30d4e4886696bf59337c0',
   'src/third_party/flatbuffers/src':
     'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@a86afae9399bbe631d1ea0783f8816e780e236cc',
   'src/third_party/grpc/src': {
-    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@488b14fd17934e8bbb469b6b383e10544ea0824d',
+    'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@5e9fb9cbfb12a10ff9c16fbc360328d224b838d6',
   },
   # Used for embedded builds. CrOS & Linux use the system version.
   'src/third_party/fontconfig/src': {
@@ -526,7 +526,7 @@ deps = {
     'url': 'https://chromium.googlesource.com/external/github.com/google/benchmark.git@188e8278990a9069ffc84441cb5a024fd0bede37',
   },
   'src/third_party/libpfm4/src':
-    Var('chromium_git') + '/external/git.code.sf.net/p/perfmon2/libpfm4.git' + '@' + '964baf9d35d5f88d8422f96d8a82c672042e7064',
+    Var('chromium_git') + '/external/git.code.sf.net/p/perfmon2/libpfm4.git' + '@' + '41878eab48c50bb9ec5f741a013e971bb5a9dff2',
   # WebRTC-only dependency (not present in Chromium).
   'src/third_party/gtest-parallel':
     'https://chromium.googlesource.com/external/github.com/google/gtest-parallel@cd488bdedc1d2cffb98201a17afc1b298b0b90f1',
@@ -593,7 +593,7 @@ deps = {
   'src/third_party/libFuzzer/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt/lib/fuzzer.git@bea408a6e01f0f7e6c82a43121fe3af4506c932e',
   'src/third_party/fuzztest/src':
-    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@3f730151eede388a825d60bcdd0bc1540bcc2b90',
+    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@36b8640bfd7cd2cf868cd3989cf1e8e30b304be4',
   'src/third_party/libjpeg_turbo':
     'https://chromium.googlesource.com/chromium/deps/libjpeg_turbo.git@d1f5f2393e0d51f840207342ae86e55a86443288',
   'src/third_party/libsrtp':
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '282dfad33b9390cf6d4c5eccd6d9c3c104c0c450',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '54c1a44a6e524789175c80da61ccae18987dc903',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@4e3155807c68b64914f85bc24b4eae2165e757db',
+    'https://chromium.googlesource.com/chromium/src/tools@ff09f3d9ee8b2f215be6ecd7592757c98010403d',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'lH4R_xNZoL7YwnNSJYRSgU6ve0ypQAaGXUhHSMV8238C',
+          'version': 'TUGHGHq_2U0C6NzCTxxwEIYyf1o0qWVn7Ta_T8xndiAC',
       },
     ],
     'condition': 'checkout_android and non_git_source',
@@ -842,7 +842,7 @@ deps = {
       'packages': [
           {
               'package': 'chromium/third_party/android_deps/autorolled',
-              'version': 'NDkHoZFBi462KHX9MpNy-3MfJnllAi8MytUuzRobeeQC',
+              'version': 'ioujkn6x8xwFvfYkiQGrRb9lX4Khn3Thhv3nsoN6lrQC',
           },
       ],
       'condition': 'checkout_android and non_git_source',

From 9f4a5163fa396da283236c06325e61d3ea895278 Mon Sep 17 00:00:00 2001
From: Shaofan Qi <vshaqi@gmail.com>
Date: Wed, 25 Mar 2026 00:44:05 +0800
Subject: [PATCH 400/472] Use subset profile matching for H.264 decoder support

Bug: webrtc:347724928
Change-Id: Ia71b907b7f5af1e3acb9ed4959bd52378e80ccb9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/432780
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47253}
---
 api/video_codecs/h264_profile_level_id.cc     | 42 +++++++++++++++
 api/video_codecs/h264_profile_level_id.h      |  5 ++
 .../test/h264_profile_level_id_unittest.cc    | 53 +++++++++++++++++++
 media/base/codec_comparators.cc               | 14 ++---
 media/base/codec_comparators.h                |  5 ++
 media/engine/internal_decoder_factory.cc      | 20 ++++++-
 .../internal_decoder_factory_unittest.cc      | 41 ++++++++++++++
 7 files changed, 171 insertions(+), 9 deletions(-)

diff --git a/api/video_codecs/h264_profile_level_id.cc b/api/video_codecs/h264_profile_level_id.cc
index fc8555a0f2..ad30598853 100644
--- a/api/video_codecs/h264_profile_level_id.cc
+++ b/api/video_codecs/h264_profile_level_id.cc
@@ -320,4 +320,46 @@ bool H264IsSameProfileAndLevel(const CodecParameterMap& params1,
          profile_level_id->level == other_profile_level_id->level;
 }
 
+bool H264IsProfileSubsetOf(const CodecParameterMap& subset,
+                           const CodecParameterMap& superset) {
+  const std::optional<H264ProfileLevelId> subset_profile_level_id =
+      ParseSdpForH264ProfileLevelId(subset);
+  const std::optional<H264ProfileLevelId> superset_profile_level_id =
+      ParseSdpForH264ProfileLevelId(superset);
+
+  if (!subset_profile_level_id || !superset_profile_level_id) {
+    return false;
+  }
+
+  const H264Profile subset_profile = subset_profile_level_id->profile;
+  const H264Profile superset_profile = superset_profile_level_id->profile;
+
+  if (subset_profile == superset_profile) {
+    return true;
+  }
+
+  // Evaluates whether the `subset` profile is a valid subset of the `superset`
+  // profile according to ITU-T H.264 Annex A.2.
+  switch (superset_profile) {
+    case H264Profile::kProfileConstrainedBaseline:
+      return false;
+    case H264Profile::kProfileBaseline:
+    case H264Profile::kProfileMain:
+      return subset_profile == H264Profile::kProfileConstrainedBaseline;
+    case H264Profile::kProfileConstrainedHigh:
+      return subset_profile == H264Profile::kProfileConstrainedBaseline;
+    case H264Profile::kProfileHigh:
+      return subset_profile == H264Profile::kProfileConstrainedBaseline ||
+             subset_profile == H264Profile::kProfileMain ||
+             subset_profile == H264Profile::kProfileConstrainedHigh;
+    case H264Profile::kProfilePredictiveHigh444:
+      return subset_profile == H264Profile::kProfileConstrainedBaseline ||
+             subset_profile == H264Profile::kProfileMain ||
+             subset_profile == H264Profile::kProfileConstrainedHigh ||
+             subset_profile == H264Profile::kProfileHigh;
+    default:
+      return false;
+  }
+}
+
 }  // namespace webrtc
diff --git a/api/video_codecs/h264_profile_level_id.h b/api/video_codecs/h264_profile_level_id.h
index 1b21f18834..8e77c55dee 100644
--- a/api/video_codecs/h264_profile_level_id.h
+++ b/api/video_codecs/h264_profile_level_id.h
@@ -91,6 +91,11 @@ RTC_EXPORT bool H264IsSameProfile(const CodecParameterMap& params1,
 RTC_EXPORT bool H264IsSameProfileAndLevel(const CodecParameterMap& params1,
                                           const CodecParameterMap& params2);
 
+// Returns true if the profile defined in `subset` is a valid subset of the
+// profile defined in `superset` according to ITU-T H.264 Annex A.2.
+RTC_EXPORT bool H264IsProfileSubsetOf(const CodecParameterMap& subset,
+                                      const CodecParameterMap& superset);
+
 }  // namespace webrtc
 
 #endif  // API_VIDEO_CODECS_H264_PROFILE_LEVEL_ID_H_
diff --git a/api/video_codecs/test/h264_profile_level_id_unittest.cc b/api/video_codecs/test/h264_profile_level_id_unittest.cc
index 044765b44e..68768f45e5 100644
--- a/api/video_codecs/test/h264_profile_level_id_unittest.cc
+++ b/api/video_codecs/test/h264_profile_level_id_unittest.cc
@@ -168,4 +168,57 @@ TEST(H264ProfileLevelId, TestParseSdpProfileLevelIdInvalid) {
   EXPECT_FALSE(ParseSdpForH264ProfileLevelId(params));
 }
 
+TEST(H264ProfileLevelId, TestIsProfileSubsetOf) {
+  CodecParameterMap params_cb;
+  params_cb["profile-level-id"] = "42e01f";
+  CodecParameterMap params_baseline;
+  params_baseline["profile-level-id"] = "42001f";
+  CodecParameterMap params_main;
+  params_main["profile-level-id"] = "4d001f";
+  CodecParameterMap params_constrained_high;
+  params_constrained_high["profile-level-id"] = "640c1f";
+  CodecParameterMap params_high;
+  params_high["profile-level-id"] = "64001f";
+  CodecParameterMap params_high444;
+  params_high444["profile-level-id"] = "f4001f";
+
+  // Same profile.
+  EXPECT_TRUE(H264IsProfileSubsetOf(params_cb, params_cb));
+  EXPECT_TRUE(H264IsProfileSubsetOf(params_baseline, params_baseline));
+  EXPECT_TRUE(H264IsProfileSubsetOf(params_main, params_main));
+
+  // Constrained Baseline is a subset of almost everything.
+  EXPECT_TRUE(H264IsProfileSubsetOf(params_cb, params_baseline));
+  EXPECT_TRUE(H264IsProfileSubsetOf(params_cb, params_main));
+  EXPECT_TRUE(H264IsProfileSubsetOf(params_cb, params_constrained_high));
+  EXPECT_TRUE(H264IsProfileSubsetOf(params_cb, params_high));
+  EXPECT_TRUE(H264IsProfileSubsetOf(params_cb, params_high444));
+
+  // Main is a subset of High and High 4:4:4 profiles.
+  EXPECT_TRUE(H264IsProfileSubsetOf(params_main, params_high));
+  EXPECT_TRUE(H264IsProfileSubsetOf(params_main, params_high444));
+
+  // Constrained High is a subset of High and High 4:4:4.
+  EXPECT_TRUE(H264IsProfileSubsetOf(params_constrained_high, params_high));
+  EXPECT_TRUE(H264IsProfileSubsetOf(params_constrained_high, params_high444));
+
+  // High is a subset of High 4:4:4.
+  EXPECT_TRUE(H264IsProfileSubsetOf(params_high, params_high444));
+
+  // Standard Baseline is not a subset of Main/High.
+  EXPECT_FALSE(H264IsProfileSubsetOf(params_baseline, params_main));
+  EXPECT_FALSE(H264IsProfileSubsetOf(params_baseline, params_constrained_high));
+  EXPECT_FALSE(H264IsProfileSubsetOf(params_baseline, params_high));
+  EXPECT_FALSE(H264IsProfileSubsetOf(params_baseline, params_high444));
+
+  // Main allows interlaced video, Constrained High does not.
+  EXPECT_FALSE(H264IsProfileSubsetOf(params_main, params_constrained_high));
+
+  // Higher profiles are not subsets of lower profiles.
+  EXPECT_FALSE(H264IsProfileSubsetOf(params_baseline, params_cb));
+  EXPECT_FALSE(H264IsProfileSubsetOf(params_main, params_cb));
+  EXPECT_FALSE(H264IsProfileSubsetOf(params_high, params_main));
+  EXPECT_FALSE(H264IsProfileSubsetOf(params_high444, params_high));
+}
+
 }  // namespace webrtc
diff --git a/media/base/codec_comparators.cc b/media/base/codec_comparators.cc
index cb67f671a6..942e3a2973 100644
--- a/media/base/codec_comparators.cc
+++ b/media/base/codec_comparators.cc
@@ -59,12 +59,6 @@ std::string H264GetPacketizationModeOrDefault(const CodecParameterMap& params) {
   return GetFmtpParameterOrDefault(params, kH264FmtpPacketizationMode, "0");
 }
 
-bool H264IsSamePacketizationMode(const CodecParameterMap& left,
-                                 const CodecParameterMap& right) {
-  return H264GetPacketizationModeOrDefault(left) ==
-         H264GetPacketizationModeOrDefault(right);
-}
-
 std::string AV1GetTierOrDefault(const CodecParameterMap& params) {
   // If the parameter is not present, the tier MUST be inferred to be 0.
   // https://aomediacodec.github.io/av1-rtp-spec/#72-sdp-parameters
@@ -104,7 +98,7 @@ bool IsSameCodecSpecific(const std::string& name1,
   };
   if (either_name_matches(kH264CodecName))
     return H264IsSameProfile(params1, params2) &&
-           H264IsSamePacketizationMode(params1, params2);
+           IsSameH264PacketizationMode(params1, params2);
   if (either_name_matches(kVp9CodecName))
     return VP9IsSameProfile(params1, params2);
   // https://aomediacodec.github.io/av1-rtp-spec/#723-usage-with-the-sdp-offeranswer-model
@@ -417,4 +411,10 @@ bool IsSameRtpCodecIgnoringLevel(const Codec& codec,
   return params1 == params2;
 }
 
+bool IsSameH264PacketizationMode(const CodecParameterMap& left,
+                                 const CodecParameterMap& right) {
+  return H264GetPacketizationModeOrDefault(left) ==
+         H264GetPacketizationModeOrDefault(right);
+}
+
 }  // namespace webrtc
diff --git a/media/base/codec_comparators.h b/media/base/codec_comparators.h
index 1d242b119d..bf143596e3 100644
--- a/media/base/codec_comparators.h
+++ b/media/base/codec_comparators.h
@@ -44,6 +44,11 @@ bool IsSameRtpCodec(const Codec& codec, const RtpCodec& rtp_codec);
 
 // Similar to `IsSameRtpCodec` but ignoring the level related parameter.
 bool IsSameRtpCodecIgnoringLevel(const Codec& codec, const RtpCodec& rtp_codec);
+
+// Returns true if the two codecs have the same H.264 packetization-mode.
+bool IsSameH264PacketizationMode(const CodecParameterMap& left,
+                                 const CodecParameterMap& right);
+
 }  // namespace webrtc
 
 #endif  // MEDIA_BASE_CODEC_COMPARATORS_H_
diff --git a/media/engine/internal_decoder_factory.cc b/media/engine/internal_decoder_factory.cc
index f81c0f57bb..ed05e71572 100644
--- a/media/engine/internal_decoder_factory.cc
+++ b/media/engine/internal_decoder_factory.cc
@@ -16,10 +16,12 @@
 #include "absl/strings/match.h"
 #include "api/environment/environment.h"
 #include "api/video/video_codec_type.h"
+#include "api/video_codecs/h264_profile_level_id.h"
 #include "api/video_codecs/sdp_video_format.h"
 #include "api/video_codecs/video_codec.h"
 #include "api/video_codecs/video_decoder.h"
 #include "api/video_codecs/video_decoder_factory.h"
+#include "media/base/codec_comparators.h"
 #include "media/base/media_constants.h"
 #include "modules/video_coding/codecs/h264/include/h264.h"
 #include "modules/video_coding/codecs/vp8/include/vp8.h"
@@ -75,14 +77,28 @@ VideoDecoderFactory::CodecSupport InternalDecoderFactory::QueryCodecSupport(
   }
 
   CodecSupport codec_support;
-  codec_support.is_supported = format.IsCodecInList(GetSupportedFormats());
+  const std::vector<SdpVideoFormat>& supported_formats = GetSupportedFormats();
+  // For H.264, stream profile can be subset of supported profile.
+  if (absl::EqualsIgnoreCase(format.name, kH264CodecName)) {
+    for (const SdpVideoFormat& supported : supported_formats) {
+      if (absl::EqualsIgnoreCase(supported.name, kH264CodecName) &&
+          IsSameH264PacketizationMode(format.parameters,
+                                      supported.parameters) &&
+          H264IsProfileSubsetOf(format.parameters, supported.parameters)) {
+        codec_support.is_supported = true;
+        break;
+      }
+    }
+  } else {
+    codec_support.is_supported = format.IsCodecInList(supported_formats);
+  }
   return codec_support;
 }
 
 std::unique_ptr<VideoDecoder> InternalDecoderFactory::Create(
     const Environment& env,
     const SdpVideoFormat& format) {
-  if (!format.IsCodecInList(GetSupportedFormats())) {
+  if (!QueryCodecSupport(format, false).is_supported) {
     RTC_LOG(LS_WARNING) << "Trying to create decoder for unsupported format. "
                         << format.ToString();
     return nullptr;
diff --git a/media/engine/internal_decoder_factory_unittest.cc b/media/engine/internal_decoder_factory_unittest.cc
index f72110cf28..b10cc47357 100644
--- a/media/engine/internal_decoder_factory_unittest.cc
+++ b/media/engine/internal_decoder_factory_unittest.cc
@@ -149,6 +149,47 @@ TEST(InternalDecoderFactoryTest, QueryCodecSupportNoReferenceScaling) {
 #endif
 }
 
+TEST(InternalDecoderFactoryTest, QueryCodecSupportH264Profiles) {
+  InternalDecoderFactory factory;
+  auto h264_support = kH264Enabled ? kSupported : kUnsupported;
+
+  // The internal decoder factory explicitly lists Baseline, Constrained
+  // Baseline, Main, and High Predictive 4:4:4. Through profile subset matching,
+  // it should also support High and Constrained High profiles because they are
+  // subsets of High Predictive 4:4:4.
+  EXPECT_THAT(
+      factory.QueryCodecSupport(
+          SdpVideoFormat("H264", {{"profile-level-id", "42e01f"}}), false),
+      Support(h264_support));
+  EXPECT_THAT(
+      factory.QueryCodecSupport(
+          SdpVideoFormat("H264", {{"profile-level-id", "42001f"}}), false),
+      Support(h264_support));
+  EXPECT_THAT(
+      factory.QueryCodecSupport(
+          SdpVideoFormat("H264", {{"profile-level-id", "4d001f"}}), false),
+      Support(h264_support));
+  EXPECT_THAT(
+      factory.QueryCodecSupport(
+          SdpVideoFormat("H264", {{"profile-level-id", "640c1f"}}), false),
+      Support(h264_support));
+  EXPECT_THAT(
+      factory.QueryCodecSupport(
+          SdpVideoFormat("H264", {{"profile-level-id", "64001f"}}), false),
+      Support(h264_support));
+
+  EXPECT_THAT(
+      factory.QueryCodecSupport(
+          SdpVideoFormat("H264", {{"profile-level-id", "ff0000"}}), false),
+      Support(kUnsupported));
+
+  EXPECT_THAT(factory.QueryCodecSupport(
+                  SdpVideoFormat("H264", {{"profile-level-id", "42e01f"},
+                                          {"packetization-mode", "2"}}),
+                  false),
+              Support(kUnsupported));
+}
+
 TEST(InternalDecoderFactoryTest, QueryCodecSupportReferenceScaling) {
   InternalDecoderFactory factory;
   // VP9 and AV1 support for spatial layers.

From 1d5a6e111f7dc0d600c625a9ead6dd0a1b7219a9 Mon Sep 17 00:00:00 2001
From: Zhibo Wang <zhibo1.wang@intel.com>
Date: Tue, 24 Mar 2026 14:06:37 +0800
Subject: [PATCH 401/472] Use GPU texture in desktop capture

This change adds a FrameTexture member in the DesktopFrame class, so
that WGC desktop capture can pass DXGIResource handles to supported
frame consumers to save CPU-GPU copies.

Bug: chromium:40929600
Change-Id: Ide143596272e4b72c099afa2d366f58e5ef835c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/446781
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Wang, Zhibo1 <zhibo1.wang@intel.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#47254}
---
 modules/desktop_capture/BUILD.gn              |   6 +
 .../desktop_capture/desktop_capture_options.h |   8 +
 modules/desktop_capture/desktop_frame.cc      |   7 +-
 modules/desktop_capture/desktop_frame.h       |  15 +-
 .../desktop_frame_texture_unittest.cc         | 146 ++++++++++
 modules/desktop_capture/frame_texture.cc      |  18 ++
 modules/desktop_capture/frame_texture.h       |  48 ++++
 .../desktop_capture/shared_desktop_frame.cc   |   3 +-
 .../desktop_capture/win/dxgi_desktop_frame.cc |  89 ++++++
 .../desktop_capture/win/dxgi_desktop_frame.h  |  51 ++++
 .../win/wgc_capture_session.cc                | 109 ++++++-
 .../desktop_capture/win/wgc_capture_session.h |  29 ++
 .../desktop_capture/win/wgc_capturer_win.cc   |   2 +-
 .../win/wgc_capturer_win_unittest.cc          | 270 ++++++++++++++++++
 14 files changed, 788 insertions(+), 13 deletions(-)
 create mode 100644 modules/desktop_capture/desktop_frame_texture_unittest.cc
 create mode 100644 modules/desktop_capture/frame_texture.cc
 create mode 100644 modules/desktop_capture/frame_texture.h
 create mode 100644 modules/desktop_capture/win/dxgi_desktop_frame.cc
 create mode 100644 modules/desktop_capture/win/dxgi_desktop_frame.h

diff --git a/modules/desktop_capture/BUILD.gn b/modules/desktop_capture/BUILD.gn
index 78d6fc4c50..3da5304447 100644
--- a/modules/desktop_capture/BUILD.gn
+++ b/modules/desktop_capture/BUILD.gn
@@ -28,6 +28,8 @@ rtc_library("primitives") {
     "desktop_geometry.h",
     "desktop_region.cc",
     "desktop_region.h",
+    "frame_texture.cc",
+    "frame_texture.h",
     "shared_desktop_frame.cc",
     "shared_desktop_frame.h",
     "shared_memory.cc",
@@ -41,6 +43,7 @@ rtc_library("primitives") {
     "../../rtc_base:refcount",
     "../../rtc_base:stringutils",
     "../../rtc_base/system:rtc_export",
+    "//third_party/abseil-cpp/absl/base:nullability",
     "//third_party/libyuv",
   ]
 }
@@ -140,6 +143,7 @@ if (rtc_include_tests) {
       "desktop_and_cursor_composer_unittest.cc",
       "desktop_capturer_differ_wrapper_unittest.cc",
       "desktop_frame_rotation_unittest.cc",
+      "desktop_frame_texture_unittest.cc",
       "desktop_frame_unittest.cc",
       "desktop_geometry_unittest.cc",
       "desktop_region_unittest.cc",
@@ -591,6 +595,8 @@ rtc_library("desktop_capture") {
 
   if (rtc_enable_win_wgc) {
     sources += [
+      "win/dxgi_desktop_frame.cc",
+      "win/dxgi_desktop_frame.h",
       "win/wgc_capture_session.cc",
       "win/wgc_capture_session.h",
       "win/wgc_capture_source.cc",
diff --git a/modules/desktop_capture/desktop_capture_options.h b/modules/desktop_capture/desktop_capture_options.h
index bc1342a648..3ce4678f14 100644
--- a/modules/desktop_capture/desktop_capture_options.h
+++ b/modules/desktop_capture/desktop_capture_options.h
@@ -227,6 +227,13 @@ class RTC_EXPORT DesktopCaptureOptions {
   void set_wgc_include_secondary_windows(bool include) {
     wgc_include_secondary_windows_ = include;
   }
+
+  // This flag enables native texture of frame with the WGC capturer.
+  // The flag has no effect if the allow_wgc_capturer flag is false.
+  bool allow_wgc_using_texture() const { return allow_wgc_using_texture_; }
+  void set_allow_wgc_using_texture(bool allow) {
+    allow_wgc_using_texture_ = allow;
+  }
 #endif  // defined(RTC_ENABLE_WIN_WGC)
 #endif  // defined(WEBRTC_WIN)
 
@@ -285,6 +292,7 @@ class RTC_EXPORT DesktopCaptureOptions {
   bool allow_wgc_zero_hertz_ = false;
   bool wgc_require_border_ = false;
   bool wgc_include_secondary_windows_ = false;
+  bool allow_wgc_using_texture_ = false;
 #endif
 #endif
 #if defined(WEBRTC_USE_X11)
diff --git a/modules/desktop_capture/desktop_frame.cc b/modules/desktop_capture/desktop_frame.cc
index 1f0f6047b2..bfba612c82 100644
--- a/modules/desktop_capture/desktop_frame.cc
+++ b/modules/desktop_capture/desktop_frame.cc
@@ -29,9 +29,11 @@ DesktopFrame::DesktopFrame(DesktopSize size,
                            int stride,
                            FourCC pixel_format,
                            uint8_t* data,
-                           SharedMemory* shared_memory)
+                           SharedMemory* shared_memory,
+                           FrameTexture* texture)
     : data_(data),
       shared_memory_(shared_memory),
+      texture_(texture),
       size_(size),
       stride_(stride),
       pixel_format_(pixel_format),
@@ -134,6 +136,7 @@ float DesktopFrame::scale_factor() const {
 }
 
 uint8_t* DesktopFrame::GetFrameDataAtPos(const DesktopVector& pos) const {
+  RTC_DCHECK(data());
   return data() + stride() * pos.y() + DesktopFrame::kBytesPerPixel * pos.x();
 }
 
@@ -160,6 +163,7 @@ void DesktopFrame::MoveFrameInfoFrom(DesktopFrame* other) {
 }
 
 bool DesktopFrame::FrameDataIsBlack() const {
+  RTC_DCHECK(data());
   if (size().is_empty())
     return false;
 
@@ -172,6 +176,7 @@ bool DesktopFrame::FrameDataIsBlack() const {
 }
 
 void DesktopFrame::SetFrameDataToBlack() {
+  RTC_DCHECK(data());
   const uint8_t kBlackPixelValue = 0x00;
   memset(data(), kBlackPixelValue, stride() * size().height());
 }
diff --git a/modules/desktop_capture/desktop_frame.h b/modules/desktop_capture/desktop_frame.h
index 104fe63d05..bda6b7f1f9 100644
--- a/modules/desktop_capture/desktop_frame.h
+++ b/modules/desktop_capture/desktop_frame.h
@@ -17,9 +17,11 @@
 #include <optional>
 #include <vector>
 
+#include "absl/base/nullability.h"
 #include "media/base/video_common.h"
 #include "modules/desktop_capture/desktop_geometry.h"
 #include "modules/desktop_capture/desktop_region.h"
+#include "modules/desktop_capture/frame_texture.h"
 #include "modules/desktop_capture/shared_memory.h"
 #include "rtc_base/system/rtc_export.h"
 
@@ -28,6 +30,8 @@ namespace webrtc {
 const float kStandardDPI = 96.0f;
 
 // DesktopFrame represents a video frame captured from the screen.
+// TODO(crbug.com/40929600): Make data-dependent methods virtual so
+// texture-backed frames can provide their own implementations.
 class RTC_EXPORT DesktopFrame {
  public:
   // DesktopFrame objects always hold BGRA data.
@@ -62,8 +66,11 @@ class RTC_EXPORT DesktopFrame {
   // The pixel format the `DesktopFrame` is stored in.
   FourCC pixel_format() const { return pixel_format_; }
 
-  // Data buffer used for the frame.
-  uint8_t* data() const { return data_; }
+  // Data buffer used for the frame. May be nullptr for texture-backed frames.
+  uint8_t* absl_nullable data() const { return data_; }
+
+  // Texture used for the frame.
+  FrameTexture* texture() const { return texture_; }
 
   // SharedMemory used for the buffer or NULL if memory is allocated on the
   // heap. The result is guaranteed to be deleted only after the frame is
@@ -172,13 +179,15 @@ class RTC_EXPORT DesktopFrame {
                int stride,
                FourCC pixel_format,
                uint8_t* data,
-               SharedMemory* shared_memory);
+               SharedMemory* shared_memory,
+               FrameTexture* texture = nullptr);
 
   // Ownership of the buffers is defined by the classes that inherit from this
   // class. They must guarantee that the buffer is not deleted before the frame
   // is deleted.
   uint8_t* const data_;
   SharedMemory* const shared_memory_;
+  FrameTexture* texture_;
 
  private:
   const DesktopSize size_;
diff --git a/modules/desktop_capture/desktop_frame_texture_unittest.cc b/modules/desktop_capture/desktop_frame_texture_unittest.cc
new file mode 100644
index 0000000000..6e0f790079
--- /dev/null
+++ b/modules/desktop_capture/desktop_frame_texture_unittest.cc
@@ -0,0 +1,146 @@
+/*
+ *  Copyright (c) 2026 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+#include <utility>
+
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "modules/desktop_capture/frame_texture.h"
+#include "modules/desktop_capture/shared_desktop_frame.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+// Returns a non-null, non-invalid sentinel FrameTexture::Handle for testing.
+// On Windows Handle is void*, on other platforms it is int.
+FrameTexture::Handle MakeHandle() {
+  constexpr uintptr_t kValue = 0x1234;
+#if defined(WEBRTC_WIN)
+  return reinterpret_cast<FrameTexture::Handle>(kValue);
+#else
+  return static_cast<FrameTexture::Handle>(kValue);
+#endif
+}
+
+// A concrete FrameTexture subclass for testing purposes. Unlike the
+// platform-specific DXGIFrameTexture, this doesn't own any real GPU resources.
+class FakeFrameTexture : public FrameTexture {
+ public:
+  explicit FakeFrameTexture(Handle handle) : FrameTexture(handle) {}
+  ~FakeFrameTexture() override = default;
+};
+
+// A texture-backed DesktopFrame with nullptr data, mimicking DXGIDesktopFrame.
+class TextureDesktopFrame : public DesktopFrame {
+ public:
+  TextureDesktopFrame(DesktopSize size,
+                      std::unique_ptr<FakeFrameTexture> texture)
+      : DesktopFrame(size,
+                     /*stride=*/0,
+                     FOURCC_ARGB,
+                     /*data=*/nullptr,
+                     /*shared_memory=*/nullptr,
+                     texture.get()),
+        owned_texture_(std::move(texture)) {}
+
+  ~TextureDesktopFrame() override = default;
+
+ private:
+  std::unique_ptr<FakeFrameTexture> owned_texture_;
+};
+
+}  // namespace
+
+TEST(FrameTextureTest, StoresHandle) {
+  const FrameTexture::Handle handle = MakeHandle();
+  FakeFrameTexture texture(handle);
+  EXPECT_EQ(texture.handle(), handle);
+}
+
+TEST(DesktopFrameTest, DefaultTextureIsNull) {
+  auto frame = std::make_unique<BasicDesktopFrame>(DesktopSize(10, 10));
+  EXPECT_EQ(frame->texture(), nullptr);
+}
+
+TEST(DesktopFrameTest, TextureFrameHasValidTexture) {
+  const FrameTexture::Handle handle = MakeHandle();
+  auto texture = std::make_unique<FakeFrameTexture>(handle);
+  FrameTexture* raw_texture = texture.get();
+
+  auto frame = std::make_unique<TextureDesktopFrame>(DesktopSize(10, 10),
+                                                     std::move(texture));
+  EXPECT_EQ(frame->data(), nullptr);
+  EXPECT_NE(frame->texture(), nullptr);
+  EXPECT_EQ(frame->texture(), raw_texture);
+  EXPECT_EQ(frame->texture()->handle(), handle);
+}
+
+TEST(DesktopFrameTest, TextureFrameHasNullData) {
+  // Verify that a texture-backed frame with nullptr data() still works for
+  // metadata access (size, texture handle, etc.).
+  const FrameTexture::Handle handle = MakeHandle();
+  auto texture = std::make_unique<FakeFrameTexture>(handle);
+
+  auto frame = std::make_unique<TextureDesktopFrame>(DesktopSize(10, 10),
+                                                     std::move(texture));
+  EXPECT_EQ(frame->data(), nullptr);
+  EXPECT_NE(frame->texture(), nullptr);
+  EXPECT_EQ(frame->texture()->handle(), handle);
+  EXPECT_EQ(frame->size().width(), 10);
+  EXPECT_EQ(frame->size().height(), 10);
+
+  // Metadata operations should work on texture-only frames.
+  frame->set_dpi(DesktopVector(96, 96));
+  EXPECT_EQ(frame->dpi().x(), 96);
+  frame->set_capture_time_ms(123);
+  EXPECT_EQ(frame->capture_time_ms(), 123);
+  frame->mutable_updated_region()->SetRect(DesktopRect::MakeWH(10, 10));
+  EXPECT_FALSE(frame->updated_region().is_empty());
+}
+
+TEST(SharedDesktopFrameTest, TexturePropagatedToSharedFrame) {
+  const FrameTexture::Handle handle = MakeHandle();
+  auto texture = std::make_unique<FakeFrameTexture>(handle);
+  FrameTexture* raw_texture = texture.get();
+
+  auto frame = std::make_unique<TextureDesktopFrame>(DesktopSize(10, 10),
+                                                     std::move(texture));
+  auto shared_frame = SharedDesktopFrame::Wrap(std::move(frame));
+  ASSERT_NE(shared_frame, nullptr);
+  EXPECT_EQ(shared_frame->texture(), raw_texture);
+  EXPECT_EQ(shared_frame->texture()->handle(), handle);
+}
+
+TEST(SharedDesktopFrameTest, TexturePropagatedToClone) {
+  const FrameTexture::Handle handle = MakeHandle();
+  auto texture = std::make_unique<FakeFrameTexture>(handle);
+  FrameTexture* raw_texture = texture.get();
+
+  auto frame = std::make_unique<TextureDesktopFrame>(DesktopSize(10, 10),
+                                                     std::move(texture));
+  auto shared_frame = SharedDesktopFrame::Wrap(std::move(frame));
+  auto clone = shared_frame->Share();
+  ASSERT_NE(clone, nullptr);
+  EXPECT_EQ(clone->texture(), raw_texture);
+  EXPECT_EQ(clone->texture()->handle(), handle);
+  EXPECT_TRUE(shared_frame->ShareFrameWith(*clone));
+}
+
+TEST(SharedDesktopFrameTest, NullTextureForNormalFrame) {
+  auto frame = std::make_unique<BasicDesktopFrame>(DesktopSize(10, 10));
+  auto shared_frame = SharedDesktopFrame::Wrap(std::move(frame));
+  ASSERT_NE(shared_frame, nullptr);
+  EXPECT_EQ(shared_frame->texture(), nullptr);
+}
+
+}  // namespace webrtc
diff --git a/modules/desktop_capture/frame_texture.cc b/modules/desktop_capture/frame_texture.cc
new file mode 100644
index 0000000000..c9934c7a70
--- /dev/null
+++ b/modules/desktop_capture/frame_texture.cc
@@ -0,0 +1,18 @@
+/*
+ *  Copyright (c) 2026 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/frame_texture.h"
+
+namespace webrtc {
+
+FrameTexture::FrameTexture(Handle handle) : handle_(handle) {}
+FrameTexture::~FrameTexture() = default;
+
+}  // namespace webrtc
diff --git a/modules/desktop_capture/frame_texture.h b/modules/desktop_capture/frame_texture.h
new file mode 100644
index 0000000000..4051b319e1
--- /dev/null
+++ b/modules/desktop_capture/frame_texture.h
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2026 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_FRAME_TEXTURE_H_
+#define MODULES_DESKTOP_CAPTURE_FRAME_TEXTURE_H_
+
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// FrameTexture is a base class for platform-specific texture handles. It stores
+// the texture handle.
+class RTC_EXPORT FrameTexture {
+ public:
+  // Platform-specific handle type for the GPU texture resource.
+  // On Windows, this is a HANDLE (e.g., DXGI shared resource handle).
+  // On other platforms, this is an integer file descriptor.
+#if defined(WEBRTC_WIN)
+  typedef void* Handle;
+  static constexpr Handle kInvalidHandle = nullptr;
+#else
+  typedef int Handle;
+  static constexpr Handle kInvalidHandle = -1;
+#endif
+  // Platform-specific handle of the texture.
+  Handle handle() const { return handle_; }
+
+  virtual ~FrameTexture();
+
+  FrameTexture(const FrameTexture&) = delete;
+  FrameTexture& operator=(const FrameTexture&) = delete;
+
+ protected:
+  explicit FrameTexture(Handle handle);
+
+  Handle handle_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_FRAME_TEXTURE_H_
diff --git a/modules/desktop_capture/shared_desktop_frame.cc b/modules/desktop_capture/shared_desktop_frame.cc
index 57a98d7d03..d1472640fa 100644
--- a/modules/desktop_capture/shared_desktop_frame.cc
+++ b/modules/desktop_capture/shared_desktop_frame.cc
@@ -54,7 +54,8 @@ SharedDesktopFrame::SharedDesktopFrame(scoped_refptr<Core> core)
                    (*core)->stride(),
                    (*core)->pixel_format(),
                    (*core)->data(),
-                   (*core)->shared_memory()),
+                   (*core)->shared_memory(),
+                   (*core)->texture()),
       core_(core) {
   CopyFrameInfoFrom(*(core_->get()));
 }
diff --git a/modules/desktop_capture/win/dxgi_desktop_frame.cc b/modules/desktop_capture/win/dxgi_desktop_frame.cc
new file mode 100644
index 0000000000..2a929c373d
--- /dev/null
+++ b/modules/desktop_capture/win/dxgi_desktop_frame.cc
@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2026 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/win/dxgi_desktop_frame.h"
+
+#include <comdef.h>
+#include <d3d11_4.h>
+#include <dxgi1_2.h>
+
+#include <utility>
+
+#include "modules/desktop_capture/win/desktop_capture_utils.h"
+#include "rtc_base/logging.h"
+
+using Microsoft::WRL::ComPtr;
+
+namespace webrtc {
+
+namespace {
+
+class DXGIFrameTexture : public FrameTexture {
+ public:
+  DXGIFrameTexture(Handle handle, ComPtr<IUnknown> prevent_release)
+      : FrameTexture(handle), prevent_release_(std::move(prevent_release)) {}
+
+  ~DXGIFrameTexture() override {
+    if (handle_ != kInvalidHandle) {
+      CloseHandle(handle_);
+    }
+  }
+
+ private:
+  // Prevents the WGC frame pool from recycling the captured texture. Typically
+  // holds a reference to the IDirect3D11CaptureFrame. When this pointer (and
+  // hence this DXGIFrameTexture) is destroyed, the capture frame is released
+  // and the frame pool may reuse the underlying texture.
+  ComPtr<IUnknown> prevent_release_;
+};
+
+}  // namespace
+
+DXGIDesktopFrame::DXGIDesktopFrame(DesktopSize size,
+                                   int stride,
+                                   uint8_t* data,
+                                   std::unique_ptr<FrameTexture> frame_texture)
+    : DesktopFrame(size,
+                   stride,
+                   FOURCC_ARGB,
+                   data,
+                   /*shared_memory=*/nullptr,
+                   frame_texture.get()),
+      owned_frame_texture_(std::move(frame_texture)) {}
+
+// static
+std::unique_ptr<DXGIDesktopFrame> DXGIDesktopFrame::Create(
+    DesktopSize size,
+    ComPtr<ID3D11Texture2D> texture,
+    ComPtr<IUnknown> prevent_release) {
+  ComPtr<IDXGIResource1> dxgi_resource;
+  HRESULT hr = texture.As(&dxgi_resource);
+  if (FAILED(hr)) {
+    RTC_LOG(LS_ERROR) << "Failed to query IDXGIResource1: "
+                      << desktop_capture::utils::ComErrorToString(hr);
+    return nullptr;
+  }
+
+  HANDLE texture_handle;
+  hr = dxgi_resource->CreateSharedHandle(nullptr, DXGI_SHARED_RESOURCE_READ,
+                                         nullptr, &texture_handle);
+  if (FAILED(hr)) {
+    RTC_LOG(LS_ERROR) << "Failed to create shared handle for texture: "
+                      << desktop_capture::utils::ComErrorToString(hr);
+    return nullptr;
+  }
+
+  auto frame_texture = std::make_unique<DXGIFrameTexture>(
+      texture_handle, std::move(prevent_release));
+  return std::unique_ptr<DXGIDesktopFrame>(
+      new DXGIDesktopFrame(size, 0, nullptr, std::move(frame_texture)));
+}
+
+}  // namespace webrtc
diff --git a/modules/desktop_capture/win/dxgi_desktop_frame.h b/modules/desktop_capture/win/dxgi_desktop_frame.h
new file mode 100644
index 0000000000..6ab7b71660
--- /dev/null
+++ b/modules/desktop_capture/win/dxgi_desktop_frame.h
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2026 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_WIN_DXGI_DESKTOP_FRAME_H_
+#define MODULES_DESKTOP_CAPTURE_WIN_DXGI_DESKTOP_FRAME_H_
+
+#include <d3d11.h>
+#include <wrl/client.h>
+
+#include <memory>
+
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/frame_texture.h"
+
+namespace webrtc {
+
+// DesktopFrame implementation used by DXGI captures on Windows.
+// Frame texture is stored in the handle.
+class DXGIDesktopFrame : public DesktopFrame {
+ public:
+  DXGIDesktopFrame(const DXGIDesktopFrame&) = delete;
+  DXGIDesktopFrame& operator=(const DXGIDesktopFrame&) = delete;
+
+  // Creates a DXGIDesktopFrame. If `prevent_release` is provided (typically
+  // an IDirect3D11CaptureFrame), it prevents the WGC frame pool from recycling
+  // the underlying texture while downstream consumers still reference this
+  // frame.
+  static std::unique_ptr<DXGIDesktopFrame> Create(
+      DesktopSize size,
+      Microsoft::WRL::ComPtr<ID3D11Texture2D> texture,
+      Microsoft::WRL::ComPtr<IUnknown> prevent_release);
+
+ private:
+  DXGIDesktopFrame(DesktopSize size,
+                   int stride,
+                   uint8_t* data,
+                   std::unique_ptr<FrameTexture> frame_texture);
+
+  const std::unique_ptr<FrameTexture> owned_frame_texture_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_WIN_DXGI_DESKTOP_FRAME_H_
diff --git a/modules/desktop_capture/win/wgc_capture_session.cc b/modules/desktop_capture/win/wgc_capture_session.cc
index 7d02d41ce1..c7eaf549b7 100644
--- a/modules/desktop_capture/win/wgc_capture_session.cc
+++ b/modules/desktop_capture/win/wgc_capture_session.cc
@@ -8,6 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "modules/desktop_capture/win/wgc_capture_session.h"
+
 #include <DispatcherQueue.h>
 #include <windows.graphics.capture.interop.h>
 #include <windows.graphics.directX.direct3d11.interop.h>
@@ -26,8 +28,8 @@
 #include "modules/desktop_capture/desktop_frame.h"
 #include "modules/desktop_capture/desktop_geometry.h"
 #include "modules/desktop_capture/shared_desktop_frame.h"
+#include "modules/desktop_capture/win/dxgi_desktop_frame.h"
 #include "modules/desktop_capture/win/screen_capture_utils.h"
-#include "modules/desktop_capture/win/wgc_capture_session.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/thread.h"
@@ -181,6 +183,24 @@ WgcCaptureSession::~WgcCaptureSession() {
   }
 }
 
+bool WgcCaptureSession::MayContainCursor() const {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  if (!session_) {
+    return false;
+  }
+  // IsCursorCaptureEnabled was introduced in IGraphicsCaptureSession2.
+  // Default to true (cursor captured) if the interface is not available.
+  ComPtr<ABI::Windows::Graphics::Capture::IGraphicsCaptureSession2> session2;
+  HRESULT hr = session_->QueryInterface(
+      ABI::Windows::Graphics::Capture::IID_IGraphicsCaptureSession2, &session2);
+  if (FAILED(hr)) {
+    return true;
+  }
+  boolean cursor_enabled = true;
+  session2->get_IsCursorCaptureEnabled(&cursor_enabled);
+  return cursor_enabled;
+}
+
 HRESULT WgcCaptureSession::StartCapture(const DesktopCaptureOptions& options) {
   RTC_DCHECK_RUN_ON(&sequence_checker_);
   RTC_DCHECK(!is_capture_started_);
@@ -249,8 +269,17 @@ HRESULT WgcCaptureSession::StartCapture(const DesktopCaptureOptions& options) {
     return hr;
   }
 
+  allow_zero_hertz_ = options.allow_wgc_zero_hertz();
+  // Texture mode requires Windows 11 24H2+ where WGC natively skips static
+  // frames, since we cannot perform CPU-side frame comparison on GPU textures.
+  allow_using_texture_ =
+      options.allow_wgc_using_texture() && DoesWgcSkipStaticFrames();
+  RTC_LOG(LS_INFO) << "WGC capture session is outputing "
+                   << (allow_using_texture_ ? "GPU texture" : "CPU memory")
+                   << ".";
+
   hr = frame_pool_statics2->CreateFreeThreaded(
-      direct3d_device_.Get(), kPixelFormat, kNumBuffers, size_, &frame_pool_);
+      direct3d_device_.Get(), kPixelFormat, num_buffers(), size_, &frame_pool_);
   if (FAILED(hr)) {
     RecordStartCaptureResult(StartCaptureResult::kCreateFramePoolFailed);
     return hr;
@@ -262,7 +291,11 @@ HRESULT WgcCaptureSession::StartCapture(const DesktopCaptureOptions& options) {
     return hr;
   }
 
-  if (!options.prefer_cursor_embedded()) {
+  // We cannot do cursor composition on texture for now, so we need to enable
+  // cursor embedding when texture is allowed.
+  const bool embed_cursor =
+      options.prefer_cursor_embedded() || allow_using_texture_;
+  if (!embed_cursor) {
     ComPtr<ABI::Windows::Graphics::Capture::IGraphicsCaptureSession2> session2;
     if (SUCCEEDED(session_->QueryInterface(
             ABI::Windows::Graphics::Capture::IID_IGraphicsCaptureSession2,
@@ -295,8 +328,6 @@ HRESULT WgcCaptureSession::StartCapture(const DesktopCaptureOptions& options) {
         options.wgc_include_secondary_windows());
   }
 
-  allow_zero_hertz_ = options.allow_wgc_zero_hertz();
-
   hr = session_->StartCapture();
   if (FAILED(hr)) {
     RTC_LOG(LS_ERROR) << "Failed to start CaptureSession: " << hr;
@@ -316,7 +347,7 @@ bool WgcCaptureSession::WaitForFirstFrame() {
   ComPtr<WGC::IDirect3D11CaptureFrame> capture_frame = nullptr;
   // Flush the `frame_pool_` buffers so that we can receive the most recent
   // frames.
-  for (int i = 0; i < kNumBuffers; ++i) {
+  for (int i = 0; i < num_buffers(); ++i) {
     HRESULT hr = frame_pool_->TryGetNextFrame(&capture_frame);
     if (FAILED(hr)) {
       RTC_LOG(LS_ERROR) << "TryGetNextFrame failed: " << hr;
@@ -456,6 +487,9 @@ HRESULT WgcCaptureSession::CreateMappedTexture(
     UINT width,
     UINT height) {
   RTC_DCHECK_RUN_ON(&sequence_checker_);
+  if (allow_using_texture_) {
+    return S_OK;
+  }
 
   D3D11_TEXTURE2D_DESC src_desc;
   src_texture->GetDesc(&src_desc);
@@ -562,13 +596,19 @@ HRESULT WgcCaptureSession::ProcessFrame() {
     }
 
     hr = frame_pool_->Recreate(direct3d_device_.Get(), kPixelFormat,
-                               kNumBuffers, new_size);
+                               num_buffers(), new_size);
     if (FAILED(hr)) {
       RecordGetFrameResult(GetFrameResult::kRecreateFramePoolFailed);
       return hr;
     }
   }
 
+  if (allow_using_texture_) {
+    ComPtr<IUnknown> prevent_release;
+    capture_frame.As(&prevent_release);
+    return ProcessTexture(texture_2D, std::move(prevent_release));
+  }
+
   // If the size has changed since the last capture, we must be sure to use
   // the smaller dimensions. Otherwise we might overrun our buffer, or
   // read stale data from the last frame.
@@ -742,6 +782,61 @@ HRESULT WgcCaptureSession::ProcessFrame() {
   return hr;
 }
 
+HRESULT WgcCaptureSession::ProcessTexture(ComPtr<ID3D11Texture2D> texture,
+                                          ComPtr<IUnknown> capture_frame) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+
+  D3D11_TEXTURE2D_DESC desc;
+  texture->GetDesc(&desc);
+  DesktopSize size(desc.Width, desc.Height);
+
+  std::unique_ptr<DesktopFrame> texture_frame = DXGIDesktopFrame::Create(
+      size, std::move(texture), std::move(capture_frame));
+  if (!texture_frame) {
+    return E_FAIL;
+  }
+
+  if (queue_.current_frame() && queue_.current_frame()->IsShared()) {
+    RTC_LOG(LS_VERBOSE) << "Overwriting texture frame that is still shared.";
+  }
+
+  // Update the monitor and scale factor information for the new frame.
+  if (is_window_source_) {
+    monitor_ = ::MonitorFromWindow(reinterpret_cast<HWND>(source_id_),
+                                   MONITOR_DEFAULTTONEAREST);
+  } else if (!monitor_.has_value()) {
+    HMONITOR monitor;
+    if (GetHmonitorFromDeviceIndex(source_id_, &monitor)) {
+      monitor_ = monitor;
+    }
+  }
+
+  if (monitor_.has_value()) {
+    DEVICE_SCALE_FACTOR device_scale_factor = DEVICE_SCALE_FACTOR_INVALID;
+    HRESULT hr =
+        GetScaleFactorForMonitor(monitor_.value(), &device_scale_factor);
+    if (SUCCEEDED(hr) && device_scale_factor != DEVICE_SCALE_FACTOR_INVALID) {
+      texture_frame->set_device_scale_factor(
+          static_cast<float>(device_scale_factor) / 100.0f);
+    }
+  }
+
+  queue_.ReplaceCurrentFrame(
+      SharedDesktopFrame::Wrap(std::move(texture_frame)));
+
+  // We use ProcessTexture() only when WGC supports skipping static frames
+  // (Windows 11 24H2+), so we can directly mark the whole frame as damaged
+  // without doing any content comparison.
+  damage_region_.SetRect(DesktopRect::MakeSize(size));
+
+  // Sync the session size with the captured texture size.
+  size_.Width = size.width();
+  size_.Height = size.height();
+
+  RecordGetFrameResult(GetFrameResult::kSuccess);
+  return S_OK;
+}
+
 HRESULT WgcCaptureSession::OnItemClosed(WGC::IGraphicsCaptureItem* sender,
                                         IInspectable* event_args) {
   RTC_DCHECK_RUN_ON(&sequence_checker_);
diff --git a/modules/desktop_capture/win/wgc_capture_session.h b/modules/desktop_capture/win/wgc_capture_session.h
index 0e5fbd3f98..74ef4c07ba 100644
--- a/modules/desktop_capture/win/wgc_capture_session.h
+++ b/modules/desktop_capture/win/wgc_capture_session.h
@@ -63,6 +63,10 @@ class WgcCaptureSession final {
     return is_capture_started_;
   }
 
+  // Returns true if the captured frame may contain the cursor. Reads the
+  // actual IsCursorCaptureEnabled state from the WGC capture session.
+  bool MayContainCursor() const;
+
   // We keep 2 buffers in the frame pool since it results in a good compromise
   // between latency/capture-rate and the rate at which
   // Direct3D11CaptureFramePool.TryGetNextFrame returns NULL and we have to fall
@@ -70,6 +74,14 @@ class WgcCaptureSession final {
   // We make this public for tests.
   static constexpr int kNumBuffers = 2;
 
+  // When using the texture path (allow_using_texture_), each in-flight frame
+  // holds a reference to the WGC capture frame (via prevent_release) to prevent
+  // the frame pool from recycling the texture. We need more buffers so that the
+  // frame pool has free slots for new captures while downstream consumers still
+  // hold older frames. The queue holds 2 frames, and the downstream consumer
+  // (encoder) may hold 1-2 more, so we use 5 to provide sufficient headroom.
+  static constexpr int kNumBuffersForTexture = 5;
+
  private:
   class RefCountedEvent : public RefCountedNonVirtual<RefCountedEvent>,
                           public Event {
@@ -135,6 +147,13 @@ class WgcCaptureSession final {
   // Process the captured frame and copy it to the `queue_`.
   HRESULT ProcessFrame();
 
+  // Process texture and copy frame with texture to the `queue_`.
+  // `capture_frame` is stored in the DXGIDesktopFrame's FrameTexture to keep
+  // the WGC frame alive, preventing the frame pool from recycling the texture
+  // while downstream consumers still reference this frame.
+  HRESULT ProcessTexture(Microsoft::WRL::ComPtr<ID3D11Texture2D> texture,
+                         Microsoft::WRL::ComPtr<IUnknown> capture_frame);
+
   void RemoveEventHandlers();
   void RemoveItemClosedEventHandler();
   void RemoveFrameArrivedEventHandler();
@@ -144,6 +163,10 @@ class WgcCaptureSession final {
 
   bool allow_zero_hertz() const { return allow_zero_hertz_; }
 
+  int num_buffers() const {
+    return allow_using_texture_ ? kNumBuffersForTexture : kNumBuffers;
+  }
+
   std::unique_ptr<EventRegistrationToken> item_closed_token_;
   std::unique_ptr<EventRegistrationToken> frame_arrived_token_;
 
@@ -197,6 +220,12 @@ class WgcCaptureSession final {
   // adds complexity since memcmp() is performed on two successive frames.
   bool allow_zero_hertz_ = false;
 
+  // Caches the value of DesktopCaptureOptions.allow_wgc_using_texture() in
+  // StartCapture(). Store texture handle of captured frame in DesktopFrame
+  // instead of mapping texture data which can reduce copy if clients process
+  // textures directly.
+  bool allow_using_texture_ = false;
+
   // Tracks damage region updates that were reported since the last time a frame
   // was captured. Currently only supports either the complete rect being
   // captured or an empty region. Will always be empty if `allow_zero_hertz_` is
diff --git a/modules/desktop_capture/win/wgc_capturer_win.cc b/modules/desktop_capture/win/wgc_capturer_win.cc
index 2f9abc07c1..cc9ea52c1f 100644
--- a/modules/desktop_capture/win/wgc_capturer_win.cc
+++ b/modules/desktop_capture/win/wgc_capturer_win.cc
@@ -431,7 +431,7 @@ void WgcCapturerWin::CaptureFrame() {
                             capture_time_ms);
   frame->set_capture_time_ms(capture_time_ms);
   frame->set_capturer_id(DesktopCapturerId::kWgcCapturerWin);
-  frame->set_may_contain_cursor(options_.prefer_cursor_embedded());
+  frame->set_may_contain_cursor(capture_session->MayContainCursor());
   frame->set_top_left(capture_source_->GetTopLeft());
   RecordWgcCapturerResult(WgcCapturerResult::kSuccess);
   callback_->OnCaptureResult(DesktopCapturer::Result::SUCCESS,
diff --git a/modules/desktop_capture/win/wgc_capturer_win_unittest.cc b/modules/desktop_capture/win/wgc_capturer_win_unittest.cc
index 46e8e1432b..8119c704ae 100644
--- a/modules/desktop_capture/win/wgc_capturer_win_unittest.cc
+++ b/modules/desktop_capture/win/wgc_capturer_win_unittest.cc
@@ -21,6 +21,7 @@
 #include "modules/desktop_capture/desktop_capture_options.h"
 #include "modules/desktop_capture/desktop_capture_types.h"
 #include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/frame_texture.h"
 #include "modules/desktop_capture/full_screen_window_detector.h"
 #include "modules/desktop_capture/win/full_screen_win_application_handler.h"
 #include "modules/desktop_capture/win/screen_capture_utils.h"
@@ -789,4 +790,273 @@ TEST_F(WgcCapturerFullScreenDetectorTest,
             0);
 }
 
+// Tests for the WGC texture mode (allow_wgc_using_texture).
+class WgcCapturerTextureTest : public ::testing::TestWithParam<CaptureType>,
+                               public DesktopCapturer::Callback {
+ public:
+  void SetUp() override {
+    com_initializer_ =
+        std::make_unique<ScopedCOMInitializer>(ScopedCOMInitializer::kMTA);
+    ASSERT_TRUE(com_initializer_->Succeeded());
+
+    if (!IsWgcSupported(GetParam())) {
+      RTC_LOG(LS_INFO)
+          << "Skipping WgcCapturerTextureTests on unsupported platforms.";
+      GTEST_SKIP();
+    }
+  }
+
+  void SetUpForWindowCapture(int window_width = kMediumWindowWidth,
+                             int window_height = kMediumWindowHeight) {
+    auto options = DesktopCaptureOptions::CreateDefault();
+    options.set_allow_wgc_using_texture(true);
+    capturer_ = WgcCapturerWin::CreateRawWindowCapturer(options);
+    CreateWindowOnSeparateThread(window_width, window_height);
+    StartWindowThreadMessageLoop();
+    source_id_ = GetTestWindowIdFromSourceList();
+  }
+
+  void SetUpForScreenCapture() {
+    auto options = DesktopCaptureOptions::CreateDefault();
+    options.set_allow_wgc_using_texture(true);
+    capturer_ = WgcCapturerWin::CreateRawScreenCapturer(options);
+    source_id_ = GetScreenIdFromSourceList();
+  }
+
+  void TearDown() override {
+    if (window_open_) {
+      CloseTestWindow();
+    }
+  }
+
+  void CreateWindowOnSeparateThread(int window_width, int window_height) {
+    window_thread_ = Thread::Create();
+    window_thread_->SetName(kWindowThreadName, nullptr);
+    window_thread_->Start();
+    SendTask(window_thread_.get(), [this, window_width, window_height]() {
+      window_thread_id_ = GetCurrentThreadId();
+      window_info_ =
+          CreateTestWindow(kWindowTitle, window_height, window_width);
+      window_open_ = true;
+
+      while (!IsWindowResponding(window_info_.hwnd)) {
+        RTC_LOG(LS_INFO) << "Waiting for test window to become responsive in "
+                            "WgcCapturerTextureTest.";
+      }
+
+      while (!IsWindowValidAndVisible(window_info_.hwnd)) {
+        RTC_LOG(LS_INFO) << "Waiting for test window to be visible in "
+                            "WgcCapturerTextureTest.";
+      }
+    });
+
+    ASSERT_TRUE(window_thread_->RunningForTest());
+    ASSERT_FALSE(window_thread_->IsCurrent());
+  }
+
+  void StartWindowThreadMessageLoop() {
+    window_thread_->PostTask([this]() {
+      MSG msg;
+      BOOL gm;
+      while ((gm = ::GetMessage(&msg, NULL, 0, 0)) != 0 && gm != -1) {
+        ::DispatchMessage(&msg);
+        if (msg.message == kDestroyWindow) {
+          DestroyTestWindow(window_info_);
+        }
+        if (msg.message == kQuitRunning) {
+          PostQuitMessage(0);
+        }
+      }
+    });
+  }
+
+  void CloseTestWindow() {
+    ::PostThreadMessage(window_thread_id_, kDestroyWindow, 0, 0);
+    ::PostThreadMessage(window_thread_id_, kQuitRunning, 0, 0);
+    window_thread_->Stop();
+    window_open_ = false;
+  }
+
+  DesktopCapturer::SourceId GetTestWindowIdFromSourceList() {
+    intptr_t src_id = 0;
+    do {
+      DesktopCapturer::SourceList sources;
+      EXPECT_TRUE(capturer_->GetSourceList(&sources));
+      auto it = std::find_if(
+          sources.begin(), sources.end(),
+          [&](const DesktopCapturer::Source& src) {
+            return src.id == reinterpret_cast<intptr_t>(window_info_.hwnd);
+          });
+      if (it != sources.end())
+        src_id = it->id;
+    } while (src_id != reinterpret_cast<intptr_t>(window_info_.hwnd));
+    return src_id;
+  }
+
+  DesktopCapturer::SourceId GetScreenIdFromSourceList() {
+    DesktopCapturer::SourceList sources;
+    EXPECT_TRUE(capturer_->GetSourceList(&sources));
+    EXPECT_GT(sources.size(), 0ULL);
+    return sources[0].id;
+  }
+
+  void DoCapture(int num_captures = 1) {
+    const int max_tries = num_captures == 1 ? 1 : kMaxTries;
+    int success_count = 0;
+    for (int i = 0; success_count < num_captures && i < max_tries; i++) {
+      capturer_->CaptureFrame();
+      if (result_ == DesktopCapturer::Result::ERROR_PERMANENT)
+        break;
+      if (result_ == DesktopCapturer::Result::SUCCESS)
+        success_count++;
+    }
+
+    total_successful_captures_ += success_count;
+    EXPECT_EQ(success_count, num_captures);
+    EXPECT_EQ(result_, DesktopCapturer::Result::SUCCESS);
+    EXPECT_TRUE(frame_);
+  }
+
+  // DesktopCapturer::Callback interface
+  void OnCaptureResult(DesktopCapturer::Result result,
+                       std::unique_ptr<DesktopFrame> frame) override {
+    result_ = result;
+    frame_ = std::move(frame);
+  }
+
+ protected:
+  std::unique_ptr<ScopedCOMInitializer> com_initializer_;
+  DWORD window_thread_id_;
+  std::unique_ptr<Thread> window_thread_;
+  WindowInfo window_info_;
+  intptr_t source_id_;
+  bool window_open_ = false;
+  DesktopCapturer::Result result_;
+  int total_successful_captures_ = 0;
+  std::unique_ptr<DesktopFrame> frame_;
+  std::unique_ptr<DesktopCapturer> capturer_;
+};
+
+// Verifies that the captured frame in texture mode has a non-null texture with
+// a valid handle, and that the CPU data pointer is null (texture-only frame).
+TEST_P(WgcCapturerTextureTest, CapturedFrameHasTexture) {
+  if (GetParam() == CaptureType::kWindow) {
+    SetUpForWindowCapture();
+  } else {
+    SetUpForScreenCapture();
+  }
+
+  EXPECT_TRUE(capturer_->SelectSource(source_id_));
+  capturer_->Start(this);
+  DoCapture();
+
+  ASSERT_NE(frame_, nullptr);
+  EXPECT_GT(frame_->size().width(), 0);
+  EXPECT_GT(frame_->size().height(), 0);
+  ASSERT_NE(frame_->texture(), nullptr);
+  EXPECT_NE(frame_->texture()->handle(), FrameTexture::kInvalidHandle);
+  // In texture mode, frame data should be null (no CPU copy).
+  EXPECT_EQ(frame_->data(), nullptr);
+}
+
+// Verifies that consecutive captures in texture mode each produce frames with
+// valid texture handles.
+TEST_P(WgcCapturerTextureTest, MultipleCapturesProduceValidTextures) {
+  if (GetParam() == CaptureType::kWindow) {
+    SetUpForWindowCapture();
+  } else {
+    SetUpForScreenCapture();
+  }
+
+  EXPECT_TRUE(capturer_->SelectSource(source_id_));
+  capturer_->Start(this);
+
+  for (int i = 0; i < 3; ++i) {
+    DoCapture();
+    ASSERT_NE(frame_, nullptr);
+    ASSERT_NE(frame_->texture(), nullptr);
+    EXPECT_NE(frame_->texture()->handle(), FrameTexture::kInvalidHandle);
+  }
+}
+
+// Verifies that the frame pool uses `kNumBuffersForTexture` buffers when
+// texture mode is enabled (verified indirectly by successfully capturing more
+// frames than kNumBuffers without starvation).
+TEST_P(WgcCapturerTextureTest, CaptureDoesNotStarveWithTextureBuffers) {
+  if (GetParam() == CaptureType::kWindow) {
+    SetUpForWindowCapture();
+  } else {
+    SetUpForScreenCapture();
+  }
+
+  EXPECT_TRUE(capturer_->SelectSource(source_id_));
+  capturer_->Start(this);
+
+  // Capture more frames than kNumBuffers to ensure the texture buffer count
+  // (kNumBuffersForTexture) is sufficient.
+  constexpr int kCaptures = WgcCaptureSession::kNumBuffersForTexture + 1;
+  for (int i = 0; i < kCaptures; ++i) {
+    DoCapture();
+    ASSERT_NE(frame_, nullptr);
+    ASSERT_NE(frame_->texture(), nullptr);
+  }
+}
+
+INSTANTIATE_TEST_SUITE_P(SourceAgnosticTexture,
+                         WgcCapturerTextureTest,
+                         ::testing::Values(CaptureType::kWindow,
+                                           CaptureType::kScreen));
+
+// Window-specific texture mode tests.
+class WgcCapturerTextureWindowTest : public WgcCapturerTextureTest {
+ public:
+  void SetUp() override {
+    com_initializer_ =
+        std::make_unique<ScopedCOMInitializer>(ScopedCOMInitializer::kMTA);
+    ASSERT_TRUE(com_initializer_->Succeeded());
+
+    if (!IsWgcSupported(CaptureType::kWindow)) {
+      RTC_LOG(LS_INFO) << "Skipping WgcCapturerTextureWindowTest on "
+                          "unsupported platforms.";
+      GTEST_SKIP();
+    }
+  }
+};
+
+// Verifies that the cursor is embedded in the frame when texture mode is
+// enabled (since we cannot do cursor composition on texture).
+TEST_F(WgcCapturerTextureWindowTest, CursorEmbeddedInTextureMode) {
+  SetUpForWindowCapture();
+  EXPECT_TRUE(capturer_->SelectSource(source_id_));
+  capturer_->Start(this);
+  DoCapture();
+  ASSERT_NE(frame_, nullptr);
+  // When texture mode is enabled, cursor embedding should be forced on,
+  // so the frame should indicate it may contain the cursor.
+  EXPECT_TRUE(frame_->may_contain_cursor());
+}
+
+// Verifies capture continues to work after a window resize in texture mode.
+TEST_F(WgcCapturerTextureWindowTest, ResizeWindowMidTextureCapture) {
+  SetUpForWindowCapture(kSmallWindowWidth, kSmallWindowHeight);
+  EXPECT_TRUE(capturer_->SelectSource(source_id_));
+  capturer_->Start(this);
+  DoCapture();
+  ASSERT_NE(frame_, nullptr);
+  ASSERT_NE(frame_->texture(), nullptr);
+
+  // Resize the window.
+  ResizeTestWindow(window_info_.hwnd, kLargeWindowWidth, kLargeWindowHeight);
+
+  // We need more captures to flush the buffers and pick up the new size.
+  constexpr int kNumCapturesToFlushTextureBuffers =
+      WgcCaptureSession::kNumBuffersForTexture * 2 + 1;
+  DoCapture(kNumCapturesToFlushTextureBuffers);
+  ASSERT_NE(frame_, nullptr);
+  ASSERT_NE(frame_->texture(), nullptr);
+  EXPECT_NE(frame_->texture()->handle(), FrameTexture::kInvalidHandle);
+  EXPECT_GT(frame_->size().width(), 0);
+  EXPECT_GT(frame_->size().height(), 0);
+}
+
 }  // namespace webrtc

From 746e3f956305b64195183b707a244fce55231304 Mon Sep 17 00:00:00 2001
From: Evan Shrubsole <eshr@webrtc.org>
Date: Fri, 20 Mar 2026 07:52:22 +0000
Subject: [PATCH 402/472] Use GlobalSimulatedTimeController in
 P2PTransportChannelTest

Removes uses of AutoThread and SIMULATED_WAIT.

Bug: webrtc:469327588,webrtc:42223992,webrtc:381524905
Change-Id: Ied84bf98231e1c552fcdee2b3ba238a46a6a6964
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/458660
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47255}
---
 p2p/base/p2p_transport_channel_unittest.cc | 3080 +++++++++-----------
 1 file changed, 1397 insertions(+), 1683 deletions(-)

diff --git a/p2p/base/p2p_transport_channel_unittest.cc b/p2p/base/p2p_transport_channel_unittest.cc
index 4ec6d91ab4..97a9feb949 100644
--- a/p2p/base/p2p_transport_channel_unittest.cc
+++ b/p2p/base/p2p_transport_channel_unittest.cc
@@ -29,7 +29,6 @@
 #include "api/async_dns_resolver.h"
 #include "api/candidate.h"
 #include "api/environment/environment.h"
-#include "api/environment/environment_factory.h"
 #include "api/ice_transport_interface.h"
 #include "api/packet_socket_factory.h"
 #include "api/scoped_refptr.h"
@@ -72,11 +71,9 @@
 #include "rtc_base/byte_buffer.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/dscp.h"
-#include "rtc_base/fake_clock.h"
 #include "rtc_base/fake_mdns_responder.h"
 #include "rtc_base/fake_network.h"
 #include "rtc_base/firewall_socket_server.h"
-#include "rtc_base/gunit.h"
 #include "rtc_base/internal/default_socket_server.h"
 #include "rtc_base/ip_address.h"
 #include "rtc_base/logging.h"
@@ -95,9 +92,11 @@
 #include "rtc_base/time_utils.h"
 #include "rtc_base/virtual_socket_server.h"
 #include "system_wrappers/include/metrics.h"
+#include "test/create_test_environment.h"
 #include "test/create_test_field_trials.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/time_controller/simulated_time_controller.h"
 #include "test/wait_until.h"
 
 namespace webrtc {
@@ -202,7 +201,7 @@ Candidate CreateUdpCandidate(IceCandidateType type,
 }
 
 std::unique_ptr<BasicPortAllocator> CreateBasicPortAllocator(
-    const Environment& env,
+    const Environment& env_,
     NetworkManager* network_manager,
     PacketSocketFactory* socket_factory,
     const ServerAddresses& stun_servers,
@@ -218,7 +217,7 @@ std::unique_ptr<BasicPortAllocator> CreateBasicPortAllocator(
   }
   std::vector<RelayServerConfig> turn_servers(1, turn_server);
 
-  auto allocator = std::make_unique<BasicPortAllocator>(env, network_manager,
+  auto allocator = std::make_unique<BasicPortAllocator>(env_, network_manager,
                                                         socket_factory);
   allocator->Initialize();
   allocator->SetConfiguration(stun_servers, turn_servers, 0, NO_PRUNE);
@@ -306,26 +305,30 @@ class P2PTransportChannelTestBase : public ::testing::Test {
       : vss_(new VirtualSocketServer()),
         nss_(new NATSocketServer(vss_.get())),
         ss_(new FirewallSocketServer(nss_.get())),
+        time_controller_(Timestamp::Millis(1000), ss_.get()),
+        env_(CreateTestEnvironment({.time = &time_controller_})),
         socket_factory_(new BasicPacketSocketFactory(ss_.get())),
-        main_(ss_.get()),
+        main_(time_controller_.GetMainThread()),
+        ep1_(main_),
+        ep2_(main_),
         force_relay_(false) {
-    ep1_.role_ = ICEROLE_CONTROLLING;
-    ep2_.role_ = ICEROLE_CONTROLLED;
+    ep1_.SetIceRole(ICEROLE_CONTROLLING);
+    ep2_.SetIceRole(ICEROLE_CONTROLLED);
 
     metrics::Reset();
   }
 
-  void CreatePortAllocators(const Environment& env) {
-    stun_server_ = TestStunServer::Create(env, kStunAddr, *ss_, main_);
-    turn_server_.emplace(env, &main_, ss_.get(), kTurnUdpIntAddr,
+  void CreatePortAllocators() {
+    stun_server_ = TestStunServer::Create(env_, kStunAddr, *ss_, *main_);
+    turn_server_.emplace(env_, main_, ss_.get(), kTurnUdpIntAddr,
                          kTurnUdpExtAddr);
     ServerAddresses stun_servers = {kStunAddr};
-    ep1_.allocator_ = CreateBasicPortAllocator(
-        env, &ep1_.network_manager_, socket_factory_.get(), stun_servers,
-        kTurnUdpIntAddr, SocketAddress());
-    ep2_.allocator_ = CreateBasicPortAllocator(
-        env, &ep2_.network_manager_, socket_factory_.get(), stun_servers,
-        kTurnUdpIntAddr, SocketAddress());
+    ep1_.set_allocator(CreateBasicPortAllocator(
+        env_, &ep1_.network_manager(), socket_factory_.get(), stun_servers,
+        kTurnUdpIntAddr, SocketAddress()));
+    ep2_.set_allocator(CreateBasicPortAllocator(
+        env_, &ep2_.network_manager(), socket_factory_.get(), stun_servers,
+        kTurnUdpIntAddr, SocketAddress()));
   }
 
  protected:
@@ -365,7 +368,8 @@ class P2PTransportChannelTestBase : public ::testing::Test {
     TimeDelta connect_wait;
   };
 
-  struct ChannelData {
+  class ChannelData {
+   public:
     bool CheckData(const char* data, int len) {
       bool ret = false;
       if (!ch_packets_.empty()) {
@@ -376,6 +380,14 @@ class P2PTransportChannelTestBase : public ::testing::Test {
       return ret;
     }
 
+    void set_ch(std::unique_ptr<P2PTransportChannel> ch) {
+      ch_ = std::move(ch);
+    }
+    void reset_ch() { ch_.reset(); }
+    P2PTransportChannel* ch() const { return ch_.get(); }
+    std::list<std::string>& ch_packets() { return ch_packets_; }
+
+   private:
     std::string name_;  // TODO(?) - Currently not used.
     std::list<std::string> ch_packets_;
     std::unique_ptr<P2PTransportChannel> ch_;
@@ -386,27 +398,29 @@ class P2PTransportChannelTestBase : public ::testing::Test {
     Candidate candidate;
   };
 
-  struct Endpoint {
-    Endpoint()
-        : role_(ICEROLE_UNKNOWN),
+  class Endpoint {
+   public:
+    explicit Endpoint(Thread* thread)
+        : network_manager_(thread),
+          role_(ICEROLE_UNKNOWN),
           role_conflict_(false),
           save_candidates_(false) {}
-    bool HasTransport(const PacketTransportInternal* transport) {
-      return (transport == cd1_.ch_.get() || transport == cd2_.ch_.get());
+    bool HasTransport(const PacketTransportInternal* transport) const {
+      return (transport == cd1_.ch() || transport == cd2_.ch());
     }
     ChannelData* GetChannelData(PacketTransportInternal* transport) {
       if (!HasTransport(transport))
         return nullptr;
-      if (cd1_.ch_.get() == transport)
+      if (cd1_.ch() == transport)
         return &cd1_;
       else
         return &cd2_;
     }
 
     void SetIceRole(IceRole role) { role_ = role; }
-    IceRole ice_role() { return role_; }
+    IceRole ice_role() const { return role_; }
     void OnRoleConflict(bool role_conflict) { role_conflict_ = role_conflict; }
-    bool role_conflict() { return role_conflict_; }
+    bool role_conflict() const { return role_conflict_; }
     void SetAllocationStepDelay(uint32_t delay) {
       allocator_->set_step_delay(delay);
     }
@@ -418,11 +432,33 @@ class P2PTransportChannelTestBase : public ::testing::Test {
       ++ice_regathering_counter_[reason];
     }
 
-    int GetIceRegatheringCountForReason(IceRegatheringReason reason) {
-      return ice_regathering_counter_[reason];
+    int GetIceRegatheringCountForReason(IceRegatheringReason reason) const {
+      auto it = ice_regathering_counter_.find(reason);
+      return it == ice_regathering_counter_.end() ? 0 : it->second;
     }
 
-    FakeNetworkManager network_manager_{Thread::Current()};
+    FakeNetworkManager& network_manager() { return network_manager_; }
+    void set_allocator(std::unique_ptr<BasicPortAllocator> allocator) {
+      allocator_ = std::move(allocator);
+    }
+    BasicPortAllocator* allocator() const { return allocator_.get(); }
+    void set_async_dns_resolver_factory(
+        AsyncDnsResolverFactoryInterface* async_dns_resolver_factory) {
+      async_dns_resolver_factory_ = async_dns_resolver_factory;
+    }
+    AsyncDnsResolverFactoryInterface* async_dns_resolver_factory() const {
+      return async_dns_resolver_factory_;
+    }
+    ChannelData& cd1() { return cd1_; }
+    ChannelData& cd2() { return cd2_; }
+    void set_save_candidates(bool save) { save_candidates_ = save; }
+    bool save_candidates() const { return save_candidates_; }
+    std::vector<CandidateData>& saved_candidates() { return saved_candidates_; }
+    void set_ready_to_send(bool ready) { ready_to_send_ = ready; }
+    bool ready_to_send() const { return ready_to_send_; }
+
+   private:
+    FakeNetworkManager network_manager_;
     std::unique_ptr<BasicPortAllocator> allocator_;
     AsyncDnsResolverFactoryInterface* async_dns_resolver_factory_ = nullptr;
     ChannelData cd1_;
@@ -449,49 +485,60 @@ class P2PTransportChannelTestBase : public ::testing::Test {
     return new_ice;
   }
 
-  void CreateChannels(const Environment& env,
-                      const IceConfig& ep1_config,
+  Waiter DefaultWait() {
+    return Waiter({.timeout = kDefaultTimeout, .clock = &time_controller_});
+  }
+  Waiter MediumWait() {
+    return Waiter({.timeout = kMediumTimeout, .clock = &time_controller_});
+  }
+  Waiter ShortWait() {
+    return Waiter({.timeout = kShortTimeout, .clock = &time_controller_});
+  }
+  Waiter Wait(TimeDelta timeout) {
+    return Waiter({.timeout = timeout, .clock = &time_controller_});
+  }
+
+  void CreateChannels(const IceConfig& ep1_config,
                       const IceConfig& ep2_config,
                       bool renomination = false) {
     IceParameters ice_ep1_cd1_ch =
         IceParamsWithRenomination(kIceParams[0], renomination);
     IceParameters ice_ep2_cd1_ch =
         IceParamsWithRenomination(kIceParams[1], renomination);
-    ep1_.cd1_.ch_ = CreateChannel(env, 0, ICE_CANDIDATE_COMPONENT_DEFAULT,
-                                  ice_ep1_cd1_ch, ice_ep2_cd1_ch);
-    ep2_.cd1_.ch_ = CreateChannel(env, 1, ICE_CANDIDATE_COMPONENT_DEFAULT,
-                                  ice_ep2_cd1_ch, ice_ep1_cd1_ch);
-    ep1_.cd1_.ch_->SetIceConfig(ep1_config);
-    ep2_.cd1_.ch_->SetIceConfig(ep2_config);
-    ep1_.cd1_.ch_->MaybeStartGathering();
-    ep2_.cd1_.ch_->MaybeStartGathering();
-    ep1_.cd1_.ch_->allocator_session()->SubscribeIceRegathering(
+    ep1_.cd1().set_ch(CreateChannel(0, ICE_CANDIDATE_COMPONENT_DEFAULT,
+                                    ice_ep1_cd1_ch, ice_ep2_cd1_ch));
+    ep2_.cd1().set_ch(CreateChannel(1, ICE_CANDIDATE_COMPONENT_DEFAULT,
+                                    ice_ep2_cd1_ch, ice_ep1_cd1_ch));
+    ep1_.cd1().ch()->SetIceConfig(ep1_config);
+    ep2_.cd1().ch()->SetIceConfig(ep2_config);
+    ep1_.cd1().ch()->MaybeStartGathering();
+    ep2_.cd1().ch()->MaybeStartGathering();
+    ep1_.cd1().ch()->allocator_session()->SubscribeIceRegathering(
         this, [this](PortAllocatorSession* allocator_session,
                      IceRegatheringReason reason) {
           ep1_.OnIceRegathering(allocator_session, reason);
         });
-    ep2_.cd1_.ch_->allocator_session()->SubscribeIceRegathering(
+    ep2_.cd1().ch()->allocator_session()->SubscribeIceRegathering(
         this, [this](PortAllocatorSession* allocator_session,
                      IceRegatheringReason reason) {
           ep2_.OnIceRegathering(allocator_session, reason);
         });
   }
 
-  void CreateChannels(const Environment& env) {
+  void CreateChannels() {
     IceConfig default_config;
-    CreateChannels(env, default_config, default_config, false);
+    CreateChannels(default_config, default_config, false);
   }
 
   std::unique_ptr<P2PTransportChannel> CreateChannel(
-      const Environment& env,
       int endpoint,
       int component,
       const IceParameters& local_ice,
       const IceParameters& remote_ice) {
-    IceTransportInit init(env);
+    IceTransportInit init(env_);
     init.set_port_allocator(GetAllocator(endpoint));
     init.set_async_dns_resolver_factory(
-        GetEndpoint(endpoint)->async_dns_resolver_factory_);
+        GetEndpoint(endpoint)->async_dns_resolver_factory());
     auto channel = P2PTransportChannel::Create("test content name", component,
                                                std::move(init));
     channel->SubscribeReadyToSend(this,
@@ -534,18 +581,18 @@ class P2PTransportChannelTestBase : public ::testing::Test {
 
   void DestroyChannels() {
     safety_->SetNotAlive();
-    ep1_.cd1_.ch_.reset();
-    ep2_.cd1_.ch_.reset();
-    ep1_.cd2_.ch_.reset();
-    ep2_.cd2_.ch_.reset();
+    ep1_.cd1().reset_ch();
+    ep2_.cd1().reset_ch();
+    ep1_.cd2().reset_ch();
+    ep2_.cd2().reset_ch();
     // Process pending tasks that need to run for cleanup purposes such as
     // pending deletion of Connection objects (see Connection::Destroy).
     Thread::Current()->ProcessMessages(0);
   }
-  P2PTransportChannel* ep1_ch1() { return ep1_.cd1_.ch_.get(); }
-  P2PTransportChannel* ep1_ch2() { return ep1_.cd2_.ch_.get(); }
-  P2PTransportChannel* ep2_ch1() { return ep2_.cd1_.ch_.get(); }
-  P2PTransportChannel* ep2_ch2() { return ep2_.cd2_.ch_.get(); }
+  P2PTransportChannel* ep1_ch1() { return ep1_.cd1().ch(); }
+  P2PTransportChannel* ep1_ch2() { return ep1_.cd2().ch(); }
+  P2PTransportChannel* ep2_ch1() { return ep2_.cd1().ch(); }
+  P2PTransportChannel* ep2_ch2() { return ep2_.cd2().ch(); }
 
   TestTurnServer* test_turn_server() {
     EXPECT_TRUE(turn_server_.has_value());
@@ -582,10 +629,10 @@ class P2PTransportChannelTestBase : public ::testing::Test {
     }
   }
   BasicPortAllocator* GetAllocator(int endpoint) {
-    return GetEndpoint(endpoint)->allocator_.get();
+    return GetEndpoint(endpoint)->allocator();
   }
   void AddAddress(int endpoint, const SocketAddress& addr) {
-    GetEndpoint(endpoint)->network_manager_.AddInterface(addr);
+    GetEndpoint(endpoint)->network_manager().AddInterface(addr);
   }
   void AddAddress(
       int endpoint,
@@ -593,11 +640,11 @@ class P2PTransportChannelTestBase : public ::testing::Test {
       absl::string_view ifname,
       AdapterType adapter_type,
       std::optional<AdapterType> underlying_vpn_adapter_type = std::nullopt) {
-    GetEndpoint(endpoint)->network_manager_.AddInterface(
+    GetEndpoint(endpoint)->network_manager().AddInterface(
         addr, ifname, adapter_type, underlying_vpn_adapter_type);
   }
   void RemoveAddress(int endpoint, const SocketAddress& addr) {
-    GetEndpoint(endpoint)->network_manager_.RemoveInterface(addr);
+    GetEndpoint(endpoint)->network_manager().RemoveInterface(addr);
     fw()->AddRule(false, FP_ANY, FD_ANY, addr);
   }
   void SetAllocatorFlags(int endpoint, int flags) {
@@ -713,17 +760,15 @@ class P2PTransportChannelTestBase : public ::testing::Test {
     return CheckConnected(ch1, ch2) && CheckCandidatePair(ch1, ch2, from, to);
   }
 
-  void Test(const Environment& env, const Result& expected) {
-    ScopedFakeClock clock;
-    int64_t connect_start = env.clock().TimeInMilliseconds();
+  void Test(const Result& expected) {
+    webrtc::Timestamp connect_start = env_.clock().CurrentTime();
     int64_t connect_time;
 
     // Create the channels and wait for them to connect.
-    CreateChannels(env);
-    EXPECT_TRUE(WaitUntil(
-        [&] { return CheckConnected(ep1_ch1(), ep2_ch1()); },
-        {.timeout = expected.connect_wait + kShortTimeout, .clock = &clock}));
-    connect_time = env.clock().TimeInMilliseconds() - connect_start;
+    CreateChannels();
+    EXPECT_TRUE(MediumWait().Until(
+        [&] { return CheckConnected(ep1_ch1(), ep2_ch1()); }));
+    connect_time = (env_.clock().CurrentTime() - connect_start).ms();
     if (connect_time < expected.connect_wait.ms()) {
       RTC_LOG(LS_INFO) << "Connect time: " << connect_time << " ms";
     } else {
@@ -734,22 +779,20 @@ class P2PTransportChannelTestBase : public ::testing::Test {
     // Allow a few turns of the crank for the selected connections to emerge.
     // This may take up to 2 seconds.
     if (ep1_ch1()->selected_connection() && ep2_ch1()->selected_connection()) {
-      int64_t converge_start = env.clock().TimeInMilliseconds();
+      webrtc::Timestamp converge_start = env_.clock().CurrentTime();
       int64_t converge_time;
       // Verifying local and remote channel selected connection information.
       // This is done only for the RFC 5245 as controlled agent will use
       // USE-CANDIDATE from controlling (ep1) agent. We can easily predict from
       // EP1 result matrix.
-      EXPECT_TRUE(WaitUntil(
-          [&] {
-            return CheckCandidate1(expected) && CheckCandidate2(expected);
-          },
-          {.timeout = kDefaultTimeout, .clock = &clock}));
+      EXPECT_TRUE(ShortWait().Until([&] {
+        return CheckCandidate1(expected) && CheckCandidate2(expected);
+      }));
       // Also do EXPECT_EQ on each part so that failures are more verbose.
       ExpectCandidate1(expected);
       ExpectCandidate2(expected);
 
-      converge_time = env.clock().TimeInMilliseconds() - converge_start;
+      converge_time = (env_.clock().CurrentTime() - converge_start).ms();
       int64_t converge_wait = 2000;
       if (converge_time < converge_wait) {
         RTC_LOG(LS_INFO) << "Converge time: " << converge_time << " ms";
@@ -759,31 +802,27 @@ class P2PTransportChannelTestBase : public ::testing::Test {
       }
     }
     // Try sending some data to other end.
-    TestSendRecv(&clock);
+    TestSendRecv();
 
     // Destroy the channels, and wait for them to be fully cleaned up.
     DestroyChannels();
   }
 
-  void TestSendRecv(ThreadProcessingFakeClock* clock) {
+  void TestSendRecv() {
     for (int i = 0; i < 10; ++i) {
       const char* data = "ABCDEFGHIJKLMNOPQRSTUVWXYZ1234567890";
       int len = static_cast<int>(strlen(data));
       // local_channel1 <==> remote_channel1
-      EXPECT_THAT(
-          WaitUntil([&] { return SendData(ep1_ch1(), data, len); }, Eq(len),
-                    {.timeout = kMediumTimeout, .clock = clock}),
-          IsRtcOk());
-      EXPECT_TRUE(
-          WaitUntil([&] { return CheckDataOnChannel(ep2_ch1(), data, len); },
-                    {.timeout = kMediumTimeout, .clock = clock}));
-      EXPECT_THAT(
-          WaitUntil([&] { return SendData(ep2_ch1(), data, len); }, Eq(len),
-                    {.timeout = kMediumTimeout, .clock = clock}),
-          IsRtcOk());
-      EXPECT_TRUE(
-          WaitUntil([&] { return CheckDataOnChannel(ep1_ch1(), data, len); },
-                    {.timeout = kMediumTimeout, .clock = clock}));
+      EXPECT_THAT(MediumWait().Until(
+                      [&] { return SendData(ep1_ch1(), data, len); }, Eq(len)),
+                  IsRtcOk());
+      EXPECT_TRUE(MediumWait().Until(
+          [&] { return CheckDataOnChannel(ep2_ch1(), data, len); }));
+      EXPECT_THAT(MediumWait().Until(
+                      [&] { return SendData(ep2_ch1(), data, len); }, Eq(len)),
+                  IsRtcOk());
+      EXPECT_TRUE(MediumWait().Until(
+          [&] { return CheckDataOnChannel(ep1_ch1(), data, len); }));
     }
   }
 
@@ -793,11 +832,10 @@ class P2PTransportChannelTestBase : public ::testing::Test {
   // new connection using the newly generated ice candidates.
   // Before calling this function the end points must be configured.
   void TestHandleIceUfragPasswordChanged() {
-    ScopedFakeClock clock;
     ep1_ch1()->SetRemoteIceParameters(kIceParams[1]);
     ep2_ch1()->SetRemoteIceParameters(kIceParams[0]);
-    EXPECT_TRUE(WaitUntil([&] { return CheckConnected(ep1_ch1(), ep2_ch1()); },
-                          {.timeout = kMediumTimeout, .clock = &clock}));
+    EXPECT_TRUE(MediumWait().Until(
+        [&] { return CheckConnected(ep1_ch1(), ep2_ch1()); }));
 
     const Candidate* old_local_candidate1 = LocalCandidate(ep1_ch1());
     const Candidate* old_local_candidate2 = LocalCandidate(ep2_ch1());
@@ -812,26 +850,22 @@ class P2PTransportChannelTestBase : public ::testing::Test {
     ep2_ch1()->SetRemoteIceParameters(kIceParams[2]);
     ep2_ch1()->MaybeStartGathering();
 
-    EXPECT_THAT(
-        WaitUntil([&] { return LocalCandidate(ep1_ch1())->generation(); },
-                  Ne(old_local_candidate1->generation()),
-                  {.timeout = kMediumTimeout, .clock = &clock}),
-        IsRtcOk());
-    EXPECT_THAT(
-        WaitUntil([&] { return LocalCandidate(ep2_ch1())->generation(); },
-                  Ne(old_local_candidate2->generation()),
-                  {.timeout = kMediumTimeout, .clock = &clock}),
-        IsRtcOk());
-    EXPECT_THAT(
-        WaitUntil([&] { return RemoteCandidate(ep1_ch1())->generation(); },
-                  Ne(old_remote_candidate1->generation()),
-                  {.timeout = kMediumTimeout, .clock = &clock}),
-        IsRtcOk());
-    EXPECT_THAT(
-        WaitUntil([&] { return RemoteCandidate(ep2_ch1())->generation(); },
-                  Ne(old_remote_candidate2->generation()),
-                  {.timeout = kMediumTimeout, .clock = &clock}),
-        IsRtcOk());
+    EXPECT_THAT(MediumWait().Until(
+                    [&] { return LocalCandidate(ep1_ch1())->generation(); },
+                    Ne(old_local_candidate1->generation())),
+                IsRtcOk());
+    EXPECT_THAT(MediumWait().Until(
+                    [&] { return LocalCandidate(ep2_ch1())->generation(); },
+                    Ne(old_local_candidate2->generation())),
+                IsRtcOk());
+    EXPECT_THAT(MediumWait().Until(
+                    [&] { return RemoteCandidate(ep1_ch1())->generation(); },
+                    Ne(old_remote_candidate1->generation())),
+                IsRtcOk());
+    EXPECT_THAT(MediumWait().Until(
+                    [&] { return RemoteCandidate(ep2_ch1())->generation(); },
+                    Ne(old_remote_candidate2->generation())),
+                IsRtcOk());
     EXPECT_EQ(1u, RemoteCandidate(ep2_ch1())->generation());
     EXPECT_EQ(1u, RemoteCandidate(ep1_ch1())->generation());
   }
@@ -843,7 +877,7 @@ class P2PTransportChannelTestBase : public ::testing::Test {
   }
 
   void OnReadyToSend(PacketTransportInternal* transport) {
-    GetEndpoint(transport)->ready_to_send_ = true;
+    GetEndpoint(transport)->set_ready_to_send(true);
   }
 
   // We pass the candidates directly to the other side.
@@ -851,11 +885,11 @@ class P2PTransportChannelTestBase : public ::testing::Test {
     if (force_relay_ && !c.is_relay())
       return;
 
-    if (GetEndpoint(ch)->save_candidates_) {
-      GetEndpoint(ch)->saved_candidates_.push_back(
+    if (GetEndpoint(ch)->save_candidates()) {
+      GetEndpoint(ch)->saved_candidates().push_back(
           {.channel = ch, .candidate = c});
     } else {
-      main_.PostTask(SafeTask(
+      main_->PostTask(SafeTask(
           safety_, [this, ch, c = c]() mutable { AddCandidate(ch, c); }));
     }
   }
@@ -876,13 +910,13 @@ class P2PTransportChannelTestBase : public ::testing::Test {
   }
 
   void PauseCandidates(int endpoint) {
-    GetEndpoint(endpoint)->save_candidates_ = true;
+    GetEndpoint(endpoint)->set_save_candidates(true);
   }
 
   void OnCandidatesRemoved(IceTransportInternal* ch,
                            const std::vector<Candidate>& candidates) {
     // Candidate removals are not paused.
-    main_.PostTask(SafeTask(safety_, [this, ch, candidates]() mutable {
+    main_->PostTask(SafeTask(safety_, [this, ch, candidates]() mutable {
       P2PTransportChannel* rch = GetRemoteChannel(ch);
       if (rch == nullptr) {
         return;
@@ -896,7 +930,7 @@ class P2PTransportChannelTestBase : public ::testing::Test {
 
   // Tcp candidate verification has to be done when they are generated.
   void VerifySavedTcpCandidates(int endpoint, absl::string_view tcptype) {
-    for (auto& data : GetEndpoint(endpoint)->saved_candidates_) {
+    for (auto& data : GetEndpoint(endpoint)->saved_candidates()) {
       EXPECT_EQ(data.candidate.protocol(), TCP_PROTOCOL_NAME);
       EXPECT_EQ(data.candidate.tcptype(), tcptype);
       if (data.candidate.tcptype() == TCPTYPE_ACTIVE_STR) {
@@ -911,17 +945,17 @@ class P2PTransportChannelTestBase : public ::testing::Test {
 
   void ResumeCandidates(int endpoint) {
     Endpoint* ed = GetEndpoint(endpoint);
-    std::vector<CandidateData> candidates = std::move(ed->saved_candidates_);
+    std::vector<CandidateData> candidates = std::move(ed->saved_candidates());
     if (!candidates.empty()) {
-      main_.PostTask(SafeTask(
+      main_->PostTask(SafeTask(
           safety_, [this, candidates = std::move(candidates)]() mutable {
             for (CandidateData& data : candidates) {
               AddCandidate(data.channel, data.candidate);
             }
           }));
     }
-    ed->saved_candidates_.clear();
-    ed->save_candidates_ = false;
+    ed->saved_candidates().clear();
+    ed->set_save_candidates(false);
   }
 
   void AddCandidate(IceTransportInternal* channel, Candidate& candidate) {
@@ -1000,7 +1034,7 @@ class P2PTransportChannelTestBase : public ::testing::Test {
       return nullptr;
   }
   std::list<std::string>& GetPacketList(PacketTransportInternal* transport) {
-    return GetChannelData(transport)->ch_packets_;
+    return GetChannelData(transport)->ch_packets();
   }
 
   enum RemoteIceParameterSource { FROM_CANDIDATE, FROM_SETICEPARAMETERS };
@@ -1022,13 +1056,15 @@ class P2PTransportChannelTestBase : public ::testing::Test {
   void OnNominated(Connection* conn) { nominated_ = true; }
   bool nominated() { return nominated_; }
 
- private:
+ protected:
   std::unique_ptr<VirtualSocketServer> vss_;
   std::unique_ptr<NATSocketServer> nss_;
   std::unique_ptr<FirewallSocketServer> ss_;
+  GlobalSimulatedTimeController time_controller_;
+  Environment env_;
   std::unique_ptr<BasicPacketSocketFactory> socket_factory_;
 
-  AutoSocketServerThread main_;
+  Thread* main_;
   scoped_refptr<PendingTaskSafetyFlag> safety_ =
       PendingTaskSafetyFlag::Create();
   TestStunServer::StunServerPtr stun_server_;
@@ -1132,22 +1168,21 @@ const P2PTransportChannelTestBase::Result
 // Just test every combination of the configs in the Config enum.
 class P2PTransportChannelTest : public P2PTransportChannelTestBase {
  protected:
-  void ConfigureEndpoints(const Environment& env,
-                          Config config1,
+  void ConfigureEndpoints(Config config1,
                           Config config2,
                           int allocator_flags1,
                           int allocator_flags2) {
-    CreatePortAllocators(env);
-    ConfigureEndpoint(env, 0, config1);
+    CreatePortAllocators();
+    ConfigureEndpoint(0, config1);
     SetAllocatorFlags(0, allocator_flags1);
     SetAllocationStepDelay(0, kMinimumStepDelay);
-    ConfigureEndpoint(env, 1, config2);
+    ConfigureEndpoint(1, config2);
     SetAllocatorFlags(1, allocator_flags2);
     SetAllocationStepDelay(1, kMinimumStepDelay);
 
     set_remote_ice_parameter_source(FROM_SETICEPARAMETERS);
   }
-  void ConfigureEndpoint(const Environment& env, int endpoint, Config config) {
+  void ConfigureEndpoint(int endpoint, Config config) {
     switch (config) {
       case OPEN:
         AddAddress(endpoint, kPublicAddrs[endpoint]);
@@ -1159,7 +1194,7 @@ class P2PTransportChannelTest : public P2PTransportChannelTestBase {
         AddAddress(endpoint, kPrivateAddrs[endpoint]);
         // Add a single NAT of the desired type
         nat()
-            ->AddTranslator(env, kPublicAddrs[endpoint], kNatAddrs[endpoint],
+            ->AddTranslator(env_, kPublicAddrs[endpoint], kNatAddrs[endpoint],
                             static_cast<NATType>(config - NAT_FULL_CONE))
             ->AddClient(kPrivateAddrs[endpoint]);
         break;
@@ -1168,11 +1203,11 @@ class P2PTransportChannelTest : public P2PTransportChannelTestBase {
         AddAddress(endpoint, kCascadedPrivateAddrs[endpoint]);
         // Add a two cascaded NATs of the desired types
         nat()
-            ->AddTranslator(env, kPublicAddrs[endpoint], kNatAddrs[endpoint],
+            ->AddTranslator(env_, kPublicAddrs[endpoint], kNatAddrs[endpoint],
                             (config == Config::NAT_DOUBLE_CONE)
                                 ? NATType::NAT_OPEN_CONE
                                 : NATType::NAT_SYMMETRIC)
-            ->AddTranslator(env, kPrivateAddrs[endpoint],
+            ->AddTranslator(env_, kPrivateAddrs[endpoint],
                             kCascadedNatAddrs[endpoint], NAT_OPEN_CONE)
             ->AddClient(kCascadedPrivateAddrs[endpoint]);
         break;
@@ -1259,16 +1294,14 @@ const P2PTransportChannelMatrixTest::Result*
 
 // The actual tests that exercise all the various configurations.
 // Test names are of the form P2PTransportChannelTest_TestOPENToNAT_FULL_CONE
-#define P2P_TEST_DECLARATION(x, y, z)                                 \
-  TEST_P(P2PTransportChannelMatrixTest, z##Test##x##To##y) {          \
-    const Environment env =                                           \
-        CreateEnvironment(CreateTestFieldTrialsPtr(GetParam()));      \
-    ConfigureEndpoints(env, x, y, PORTALLOCATOR_ENABLE_SHARED_SOCKET, \
-                       PORTALLOCATOR_ENABLE_SHARED_SOCKET);           \
-    if (kMatrix[x][y] != NULL)                                        \
-      Test(env, *kMatrix[x][y]);                                      \
-    else                                                              \
-      RTC_LOG(LS_WARNING) << "Not yet implemented";                   \
+#define P2P_TEST_DECLARATION(x, y, z)                            \
+  TEST_P(P2PTransportChannelMatrixTest, z##Test##x##To##y) {     \
+    ConfigureEndpoints(x, y, PORTALLOCATOR_ENABLE_SHARED_SOCKET, \
+                       PORTALLOCATOR_ENABLE_SHARED_SOCKET);      \
+    if (kMatrix[x][y] != NULL)                                   \
+      Test(*kMatrix[x][y]);                                      \
+    else                                                         \
+      RTC_LOG(LS_WARNING) << "Not yet implemented";              \
   }
 
 #define P2P_TEST(x, y) P2P_TEST_DECLARATION(x, y, /* empty argument */)
@@ -1305,10 +1338,9 @@ INSTANTIATE_TEST_SUITE_P(
 // Test that we restart candidate allocation when local ufrag&pwd changed.
 // Standard Ice protocol is used.
 TEST_F(P2PTransportChannelTest, HandleUfragPwdChange) {
-  const Environment env = CreateEnvironment();
-  ConfigureEndpoints(env, OPEN, OPEN, kDefaultPortAllocatorFlags,
+  ConfigureEndpoints(OPEN, OPEN, kDefaultPortAllocatorFlags,
                      kDefaultPortAllocatorFlags);
-  CreateChannels(env);
+  CreateChannels();
   TestHandleIceUfragPasswordChanged();
   DestroyChannels();
 }
@@ -1316,29 +1348,24 @@ TEST_F(P2PTransportChannelTest, HandleUfragPwdChange) {
 // Same as above test, but with a symmetric NAT.
 // We should end up with relay<->prflx candidate pairs, with generation "1".
 TEST_F(P2PTransportChannelTest, HandleUfragPwdChangeSymmetricNat) {
-  const Environment env = CreateEnvironment();
-  ConfigureEndpoints(env, NAT_SYMMETRIC, NAT_SYMMETRIC,
-                     kDefaultPortAllocatorFlags, kDefaultPortAllocatorFlags);
-  CreateChannels(env);
+  ConfigureEndpoints(NAT_SYMMETRIC, NAT_SYMMETRIC, kDefaultPortAllocatorFlags,
+                     kDefaultPortAllocatorFlags);
+  CreateChannels();
   TestHandleIceUfragPasswordChanged();
   DestroyChannels();
 }
 
 // Test the operation of GetStats.
 TEST_F(P2PTransportChannelTest, GetStats) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-  ConfigureEndpoints(env, OPEN, OPEN, kDefaultPortAllocatorFlags,
+  ConfigureEndpoints(OPEN, OPEN, kDefaultPortAllocatorFlags,
                      kDefaultPortAllocatorFlags);
-  CreateChannels(env);
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return ep1_ch1()->receiving() && ep1_ch1()->writable() &&
-               ep2_ch1()->receiving() && ep2_ch1()->writable();
-      },
-      {.timeout = kMediumTimeout, .clock = &clock}));
+  CreateChannels();
+  EXPECT_TRUE(MediumWait().Until([&] {
+    return ep1_ch1()->receiving() && ep1_ch1()->writable() &&
+           ep2_ch1()->receiving() && ep2_ch1()->writable();
+  }));
   // Sends and receives 10 packets.
-  TestSendRecv(&clock);
+  TestSendRecv();
 
   // Try sending a packet which is discarded due to the socket being blocked.
   virtual_socket_server()->SetSendingBlocked(true);
@@ -1378,25 +1405,21 @@ TEST_F(P2PTransportChannelTest, GetStats) {
 }
 
 TEST_F(P2PTransportChannelTest, GetStatsSwitchConnection) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
   IceConfig continual_gathering_config =
       CreateIceConfig(TimeDelta::Seconds(1), GATHER_CONTINUALLY);
 
-  ConfigureEndpoints(env, OPEN, OPEN, kDefaultPortAllocatorFlags,
+  ConfigureEndpoints(OPEN, OPEN, kDefaultPortAllocatorFlags,
                      kDefaultPortAllocatorFlags);
 
   AddAddress(0, kAlternateAddrs[1], "rmnet0", ADAPTER_TYPE_CELLULAR);
 
-  CreateChannels(env, continual_gathering_config, continual_gathering_config);
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return ep1_ch1()->receiving() && ep1_ch1()->writable() &&
-               ep2_ch1()->receiving() && ep2_ch1()->writable();
-      },
-      {.timeout = kMediumTimeout, .clock = &clock}));
+  CreateChannels(continual_gathering_config, continual_gathering_config);
+  EXPECT_TRUE(MediumWait().Until([&] {
+    return ep1_ch1()->receiving() && ep1_ch1()->writable() &&
+           ep2_ch1()->receiving() && ep2_ch1()->writable();
+  }));
   // Sends and receives 10 packets.
-  TestSendRecv(&clock);
+  TestSendRecv();
 
   IceTransportStats ice_transport_stats;
   ASSERT_TRUE(ep1_ch1()->GetStats(&ice_transport_stats));
@@ -1426,12 +1449,12 @@ TEST_F(P2PTransportChannelTest, GetStatsSwitchConnection) {
       const_cast<Connection*>(old_selected_connection));
 
   EXPECT_THAT(
-      WaitUntil([&] { return ep1_ch1()->selected_connection(); }, Ne(nullptr),
-                {.timeout = kMediumTimeout, .clock = &clock}),
+      MediumWait().Until([&] { return ep1_ch1()->selected_connection(); },
+                         Ne(nullptr)),
       IsRtcOk());
 
   // Sends and receives 10 packets.
-  TestSendRecv(&clock);
+  TestSendRecv();
 
   IceTransportStats ice_transport_stats2;
   ASSERT_TRUE(ep1_ch1()->GetStats(&ice_transport_stats2));
@@ -1456,36 +1479,34 @@ TEST_F(P2PTransportChannelTest, GetStatsSwitchConnection) {
 // change if and only if continual gathering is enabled.
 TEST_F(P2PTransportChannelTest,
        TestIceRegatheringReasonContinualGatheringByNetworkChange) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-  ConfigureEndpoints(env, OPEN, OPEN, kOnlyLocalPorts, kOnlyLocalPorts);
+  ConfigureEndpoints(OPEN, OPEN, kOnlyLocalPorts, kOnlyLocalPorts);
 
   // ep1 gathers continually but ep2 does not.
   IceConfig continual_gathering_config =
       CreateIceConfig(TimeDelta::Seconds(1), GATHER_CONTINUALLY);
   IceConfig default_config;
-  CreateChannels(env, continual_gathering_config, default_config);
+  CreateChannels(continual_gathering_config, default_config);
 
-  EXPECT_TRUE(WaitUntil([&] { return CheckConnected(ep1_ch1(), ep2_ch1()); },
-                        {.timeout = kDefaultTimeout, .clock = &clock}));
+  EXPECT_TRUE(
+      MediumWait().Until([&] { return CheckConnected(ep1_ch1(), ep2_ch1()); }));
 
   // Adding address in ep1 will trigger continual gathering.
   AddAddress(0, kAlternateAddrs[0]);
-  EXPECT_THAT(
-      WaitUntil(
-          [&] {
-            return GetEndpoint(0)->GetIceRegatheringCountForReason(
-                IceRegatheringReason::NETWORK_CHANGE);
-          },
-          Eq(1), {.timeout = kDefaultTimeout, .clock = &clock}),
-      IsRtcOk());
+  EXPECT_THAT(Wait(TimeDelta::Millis(1900))
+                  .Until(
+                      [&] {
+                        return GetEndpoint(0)->GetIceRegatheringCountForReason(
+                            IceRegatheringReason::NETWORK_CHANGE);
+                      },
+                      Eq(1)),
+              IsRtcOk());
 
   ep2_ch1()->SetIceParameters(kIceParams[3]);
   ep2_ch1()->SetRemoteIceParameters(kIceParams[2]);
   ep2_ch1()->MaybeStartGathering();
 
   AddAddress(1, kAlternateAddrs[1]);
-  SIMULATED_WAIT(false, kDefaultTimeout.ms(), clock);
+  time_controller_.AdvanceTime(kDefaultTimeout);
   // ep2 has not enabled continual gathering.
   EXPECT_EQ(0, GetEndpoint(1)->GetIceRegatheringCountForReason(
                    IceRegatheringReason::NETWORK_CHANGE));
@@ -1497,9 +1518,7 @@ TEST_F(P2PTransportChannelTest,
 // failure if and only if continual gathering is enabled.
 TEST_F(P2PTransportChannelTest,
        TestIceRegatheringReasonContinualGatheringByNetworkFailure) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-  ConfigureEndpoints(env, OPEN, OPEN, kOnlyLocalPorts, kOnlyLocalPorts);
+  ConfigureEndpoints(OPEN, OPEN, kOnlyLocalPorts, kOnlyLocalPorts);
 
   // ep1 gathers continually but ep2 does not.
   IceConfig config1 =
@@ -1507,15 +1526,15 @@ TEST_F(P2PTransportChannelTest,
   config1.regather_on_failed_networks_interval = TimeDelta::Seconds(2);
   IceConfig config2;
   config2.regather_on_failed_networks_interval = TimeDelta::Seconds(2);
-  CreateChannels(env, config1, config2);
+  CreateChannels(config1, config2);
 
-  EXPECT_TRUE(WaitUntil([&] { return CheckConnected(ep1_ch1(), ep2_ch1()); },
-                        {.timeout = kDefaultTimeout, .clock = &clock}));
+  EXPECT_TRUE(DefaultWait().Until(
+      [&] { return CheckConnected(ep1_ch1(), ep2_ch1()); }));
 
   fw()->AddRule(false, FP_ANY, FD_ANY, kPublicAddrs[0]);
   // Timeout value such that all connections are deleted.
   const int kNetworkFailureTimeout = 35000;
-  SIMULATED_WAIT(false, kNetworkFailureTimeout, clock);
+  time_controller_.AdvanceTime(TimeDelta::Millis(kNetworkFailureTimeout));
   EXPECT_LE(1, GetEndpoint(0)->GetIceRegatheringCountForReason(
                    IceRegatheringReason::NETWORK_FAILURE));
   EXPECT_EQ(0, GetEndpoint(1)->GetIceRegatheringCountForReason(
@@ -1527,12 +1546,11 @@ TEST_F(P2PTransportChannelTest,
 // Test that we properly create a connection on a STUN ping from unknown address
 // when the signaling is slow.
 TEST_F(P2PTransportChannelTest, PeerReflexiveCandidateBeforeSignaling) {
-  const Environment env = CreateEnvironment();
-  ConfigureEndpoints(env, OPEN, OPEN, kDefaultPortAllocatorFlags,
+  ConfigureEndpoints(OPEN, OPEN, kDefaultPortAllocatorFlags,
                      kDefaultPortAllocatorFlags);
   // Emulate no remote parameters coming in.
   set_remote_ice_parameter_source(FROM_CANDIDATE);
-  CreateChannels(env);
+  CreateChannels();
   // Only have remote parameters come in for ep2, not ep1.
   ep2_ch1()->SetRemoteIceParameters(kIceParams[0]);
 
@@ -1542,9 +1560,10 @@ TEST_F(P2PTransportChannelTest, PeerReflexiveCandidateBeforeSignaling) {
 
   // Wait until the callee becomes writable to make sure that a ping request is
   // received by the caller before their remote ICE credentials are set.
-  ASSERT_THAT(WaitUntil([&] { return ep2_ch1()->selected_connection(); },
-                        Ne(nullptr), {.timeout = kMediumTimeout}),
-              IsRtcOk());
+  ASSERT_THAT(
+      Wait(TimeDelta::Seconds(11))
+          .Until([&] { return ep2_ch1()->selected_connection(); }, Ne(nullptr)),
+      IsRtcOk());
   // Add two sets of remote ICE credentials, so that the ones used by the
   // candidate will be generation 1 instead of 0.
   ep1_ch1()->SetRemoteIceParameters(kIceParams[3]);
@@ -1552,12 +1571,12 @@ TEST_F(P2PTransportChannelTest, PeerReflexiveCandidateBeforeSignaling) {
   // The caller should have the selected connection connected to the peer
   // reflexive candidate.
   const Connection* selected_connection = nullptr;
-  ASSERT_THAT(WaitUntil(
+  ASSERT_THAT(MediumWait().Until(
                   [&] {
                     return selected_connection =
                                ep1_ch1()->selected_connection();
                   },
-                  Ne(nullptr), {.timeout = kMediumTimeout}),
+                  Ne(nullptr)),
               IsRtcOk());
   EXPECT_TRUE(selected_connection->remote_candidate().is_prflx());
   EXPECT_EQ(kIceUfrag[1], selected_connection->remote_candidate().username());
@@ -1566,11 +1585,9 @@ TEST_F(P2PTransportChannelTest, PeerReflexiveCandidateBeforeSignaling) {
 
   ResumeCandidates(1);
   // Verify ep1's selected connection is updated to use the 'local' candidate.
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return ep1_ch1()->selected_connection()->remote_candidate().is_local();
-      },
-      {.timeout = kMediumTimeout}));
+  EXPECT_TRUE(MediumWait().Until([&] {
+    return ep1_ch1()->selected_connection()->remote_candidate().is_local();
+  }));
   EXPECT_EQ(selected_connection, ep1_ch1()->selected_connection());
   DestroyChannels();
 }
@@ -1580,11 +1597,10 @@ TEST_F(P2PTransportChannelTest, PeerReflexiveCandidateBeforeSignaling) {
 // 2. the candidate pair stats
 // until we learn the same address from signaling.
 TEST_F(P2PTransportChannelTest, PeerReflexiveRemoteCandidateIsSanitized) {
-  const Environment env = CreateEnvironment();
-  ConfigureEndpoints(env, OPEN, OPEN, kOnlyLocalPorts, kOnlyLocalPorts);
+  ConfigureEndpoints(OPEN, OPEN, kOnlyLocalPorts, kOnlyLocalPorts);
   // Emulate no remote parameters coming in.
   set_remote_ice_parameter_source(FROM_CANDIDATE);
-  CreateChannels(env);
+  CreateChannels();
   // Only have remote parameters come in for ep2, not ep1.
   ep2_ch1()->SetRemoteIceParameters(kIceParams[0]);
 
@@ -1592,13 +1608,15 @@ TEST_F(P2PTransportChannelTest, PeerReflexiveRemoteCandidateIsSanitized) {
   // candidate.
   PauseCandidates(1);
 
-  ASSERT_THAT(WaitUntil([&] { return ep2_ch1()->selected_connection(); },
-                        Ne(nullptr), {.timeout = kMediumTimeout}),
-              IsRtcOk());
+  ASSERT_THAT(
+      MediumWait().Until([&] { return ep2_ch1()->selected_connection(); },
+                         Ne(nullptr)),
+      IsRtcOk());
   ep1_ch1()->SetRemoteIceParameters(kIceParams[1]);
-  ASSERT_THAT(WaitUntil([&] { return ep1_ch1()->selected_connection(); },
-                        Ne(nullptr), {.timeout = kMediumTimeout}),
-              IsRtcOk());
+  ASSERT_THAT(
+      MediumWait().Until([&] { return ep1_ch1()->selected_connection(); },
+                         Ne(nullptr)),
+      IsRtcOk());
 
   // Check the selected candidate pair.
   auto pair_ep1 = ep1_ch1()->GetSelectedCandidatePair();
@@ -1619,12 +1637,10 @@ TEST_F(P2PTransportChannelTest, PeerReflexiveRemoteCandidateIsSanitized) {
 
   // Let ep1 receive the remote candidate to update its type from prflx to host.
   ResumeCandidates(1);
-  ASSERT_TRUE(WaitUntil(
-      [&] {
-        return ep1_ch1()->selected_connection() != nullptr &&
-               ep1_ch1()->selected_connection()->remote_candidate().is_local();
-      },
-      {.timeout = kMediumTimeout}));
+  ASSERT_TRUE(MediumWait().Until([&] {
+    return ep1_ch1()->selected_connection() != nullptr &&
+           ep1_ch1()->selected_connection()->remote_candidate().is_local();
+  }));
 
   // We should be able to reveal the address after it is learnt via
   // AddIceCandidate.
@@ -1650,12 +1666,11 @@ TEST_F(P2PTransportChannelTest, PeerReflexiveRemoteCandidateIsSanitized) {
 // Test that we properly create a connection on a STUN ping from unknown address
 // when the signaling is slow and the end points are behind NAT.
 TEST_F(P2PTransportChannelTest, PeerReflexiveCandidateBeforeSignalingWithNAT) {
-  const Environment env = CreateEnvironment();
-  ConfigureEndpoints(env, OPEN, NAT_SYMMETRIC, kDefaultPortAllocatorFlags,
+  ConfigureEndpoints(OPEN, NAT_SYMMETRIC, kDefaultPortAllocatorFlags,
                      kDefaultPortAllocatorFlags);
   // Emulate no remote parameters coming in.
   set_remote_ice_parameter_source(FROM_CANDIDATE);
-  CreateChannels(env);
+  CreateChannels();
   // Only have remote parameters come in for ep2, not ep1.
   ep2_ch1()->SetRemoteIceParameters(kIceParams[0]);
   // Pause sending ep2's candidates to ep1 until ep1 receives the peer reflexive
@@ -1664,9 +1679,10 @@ TEST_F(P2PTransportChannelTest, PeerReflexiveCandidateBeforeSignalingWithNAT) {
 
   // Wait until the callee becomes writable to make sure that a ping request is
   // received by the caller before their remote ICE credentials are set.
-  ASSERT_THAT(WaitUntil([&] { return ep2_ch1()->selected_connection(); },
-                        Ne(nullptr), {.timeout = kMediumTimeout}),
-              IsRtcOk());
+  ASSERT_THAT(
+      MediumWait().Until([&] { return ep2_ch1()->selected_connection(); },
+                         Ne(nullptr)),
+      IsRtcOk());
   // Add two sets of remote ICE credentials, so that the ones used by the
   // candidate will be generation 1 instead of 0.
   ep1_ch1()->SetRemoteIceParameters(kIceParams[3]);
@@ -1675,12 +1691,12 @@ TEST_F(P2PTransportChannelTest, PeerReflexiveCandidateBeforeSignalingWithNAT) {
   // The caller's selected connection should be connected to the peer reflexive
   // candidate.
   const Connection* selected_connection = nullptr;
-  ASSERT_THAT(WaitUntil(
+  ASSERT_THAT(MediumWait().Until(
                   [&] {
                     return selected_connection =
                                ep1_ch1()->selected_connection();
                   },
-                  Ne(nullptr), {.timeout = kMediumTimeout}),
+                  Ne(nullptr)),
               IsRtcOk());
   EXPECT_TRUE(selected_connection->remote_candidate().is_prflx());
   EXPECT_EQ(kIceUfrag[1], selected_connection->remote_candidate().username());
@@ -1689,11 +1705,9 @@ TEST_F(P2PTransportChannelTest, PeerReflexiveCandidateBeforeSignalingWithNAT) {
 
   ResumeCandidates(1);
 
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return ep1_ch1()->selected_connection()->remote_candidate().is_prflx();
-      },
-      {.timeout = kMediumTimeout}));
+  EXPECT_TRUE(MediumWait().Until([&] {
+    return ep1_ch1()->selected_connection()->remote_candidate().is_prflx();
+  }));
   EXPECT_EQ(selected_connection, ep1_ch1()->selected_connection());
   DestroyChannels();
 }
@@ -1709,21 +1723,20 @@ TEST_F(P2PTransportChannelTest, PeerReflexiveCandidateBeforeSignalingWithNAT) {
 // prioritized above new-generation candidate pairs.
 TEST_F(P2PTransportChannelTest,
        PeerReflexiveCandidateBeforeSignalingWithIceRestart) {
-  const Environment env = CreateEnvironment();
-  ConfigureEndpoints(env, OPEN, OPEN, kDefaultPortAllocatorFlags,
+  ConfigureEndpoints(OPEN, OPEN, kDefaultPortAllocatorFlags,
                      kDefaultPortAllocatorFlags);
   // Only gather relay candidates, so that when the prflx candidate arrives
   // it's prioritized above the current candidate pair.
-  GetEndpoint(0)->allocator_->SetCandidateFilter(CF_RELAY);
-  GetEndpoint(1)->allocator_->SetCandidateFilter(CF_RELAY);
+  GetEndpoint(0)->allocator()->SetCandidateFilter(CF_RELAY);
+  GetEndpoint(1)->allocator()->SetCandidateFilter(CF_RELAY);
   // Setting this allows us to control when SetRemoteIceParameters is called.
   set_remote_ice_parameter_source(FROM_CANDIDATE);
-  CreateChannels(env);
+  CreateChannels();
   // Wait for the initial connection to be made.
   ep1_ch1()->SetRemoteIceParameters(kIceParams[1]);
   ep2_ch1()->SetRemoteIceParameters(kIceParams[0]);
-  EXPECT_TRUE(WaitUntil([&] { return CheckConnected(ep1_ch1(), ep2_ch1()); },
-                        {.timeout = kDefaultTimeout}));
+  EXPECT_TRUE(
+      MediumWait().Until([&] { return CheckConnected(ep1_ch1(), ep2_ch1()); }));
 
   // Simulate an ICE restart on ep2, but don't signal the candidate or new
   // ICE parameters until after a prflx connection has been made.
@@ -1735,11 +1748,9 @@ TEST_F(P2PTransportChannelTest,
 
   // The caller should have the selected connection connected to the peer
   // reflexive candidate.
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return ep1_ch1()->selected_connection()->remote_candidate().is_prflx();
-      },
-      {.timeout = kDefaultTimeout}));
+  EXPECT_TRUE(Wait(TimeDelta::Seconds(8)).Until([&] {
+    return ep1_ch1()->selected_connection()->remote_candidate().is_prflx();
+  }));
   const Connection* prflx_selected_connection =
       ep1_ch1()->selected_connection();
 
@@ -1753,35 +1764,31 @@ TEST_F(P2PTransportChannelTest,
   // their information to update the peer reflexive candidate.
   ResumeCandidates(1);
 
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return ep1_ch1()->selected_connection()->remote_candidate().is_relay();
-      },
-      {.timeout = kDefaultTimeout}));
+  EXPECT_TRUE(DefaultWait().Until([&] {
+    return ep1_ch1()->selected_connection()->remote_candidate().is_relay();
+  }));
   EXPECT_EQ(prflx_selected_connection, ep1_ch1()->selected_connection());
   DestroyChannels();
 }
 
 // Test that if remote candidates don't have ufrag and pwd, we still work.
 TEST_F(P2PTransportChannelTest, RemoteCandidatesWithoutUfragPwd) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
   set_remote_ice_parameter_source(FROM_SETICEPARAMETERS);
-  ConfigureEndpoints(env, OPEN, OPEN, kDefaultPortAllocatorFlags,
+  ConfigureEndpoints(OPEN, OPEN, kDefaultPortAllocatorFlags,
                      kDefaultPortAllocatorFlags);
-  CreateChannels(env);
+  CreateChannels();
   const Connection* selected_connection = nullptr;
   // Wait until the callee's connections are created.
-  EXPECT_THAT(WaitUntil(
+  EXPECT_THAT(DefaultWait().Until(
                   [&] {
                     return selected_connection =
                                ep2_ch1()->selected_connection();
                   },
-                  NotNull(), {.timeout = kMediumTimeout, .clock = &clock}),
+                  NotNull()),
               IsRtcOk());
   // Wait to make sure the selected connection is not changed.
-  SIMULATED_WAIT(ep2_ch1()->selected_connection() != selected_connection,
-                 kShortTimeout.ms(), clock);
+  (void)MediumWait().Until(
+      [&] { return ep2_ch1()->selected_connection() != selected_connection; });
   EXPECT_TRUE(ep2_ch1()->selected_connection() == selected_connection);
   DestroyChannels();
 }
@@ -1789,17 +1796,15 @@ TEST_F(P2PTransportChannelTest, RemoteCandidatesWithoutUfragPwd) {
 // Test that a host behind NAT cannot be reached when incoming_only
 // is set to true.
 TEST_F(P2PTransportChannelTest, IncomingOnlyBlocked) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-  ConfigureEndpoints(env, NAT_FULL_CONE, OPEN, kDefaultPortAllocatorFlags,
+  ConfigureEndpoints(NAT_FULL_CONE, OPEN, kDefaultPortAllocatorFlags,
                      kDefaultPortAllocatorFlags);
 
   SetAllocatorFlags(0, kOnlyLocalPorts);
-  CreateChannels(env);
+  CreateChannels();
   ep1_ch1()->set_incoming_only(true);
 
   // Pump for 1 second and verify that the channels are not connected.
-  SIMULATED_WAIT(false, kShortTimeout.ms(), clock);
+  time_controller_.AdvanceTime(kShortTimeout);
 
   EXPECT_FALSE(ep1_ch1()->receiving());
   EXPECT_FALSE(ep1_ch1()->writable());
@@ -1812,17 +1817,15 @@ TEST_F(P2PTransportChannelTest, IncomingOnlyBlocked) {
 // Test that a peer behind NAT can connect to a peer that has
 // incoming_only flag set.
 TEST_F(P2PTransportChannelTest, IncomingOnlyOpen) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-  ConfigureEndpoints(env, OPEN, NAT_FULL_CONE, kDefaultPortAllocatorFlags,
+  ConfigureEndpoints(OPEN, NAT_FULL_CONE, kDefaultPortAllocatorFlags,
                      kDefaultPortAllocatorFlags);
 
   SetAllocatorFlags(0, kOnlyLocalPorts);
-  CreateChannels(env);
+  CreateChannels();
   ep1_ch1()->set_incoming_only(true);
 
-  EXPECT_TRUE(WaitUntil([&] { return CheckConnected(ep1_ch1(), ep2_ch1()); },
-                        {.timeout = kMediumTimeout, .clock = &clock}));
+  EXPECT_TRUE(
+      ShortWait().Until([&] { return CheckConnected(ep1_ch1(), ep2_ch1()); }));
 
   DestroyChannels();
 }
@@ -1831,7 +1834,6 @@ TEST_F(P2PTransportChannelTest, IncomingOnlyOpen) {
 // connections. This has been observed in some scenarios involving
 // VPNs/firewalls.
 TEST_F(P2PTransportChannelTest, CanOnlyMakeOutgoingTcpConnections) {
-  const Environment env = CreateEnvironment();
   // The PORTALLOCATOR_ENABLE_ANY_ADDRESS_PORTS flag is required if the
   // application needs this use case to work, since the application must accept
   // the tradeoff that more candidates need to be allocated.
@@ -1839,7 +1841,7 @@ TEST_F(P2PTransportChannelTest, CanOnlyMakeOutgoingTcpConnections) {
   // TODO(deadbeef): Later, make this flag the default, and do more elegant
   // things to ensure extra candidates don't waste resources?
   ConfigureEndpoints(
-      env, OPEN, OPEN,
+      OPEN, OPEN,
       kDefaultPortAllocatorFlags | PORTALLOCATOR_ENABLE_ANY_ADDRESS_PORTS,
       kDefaultPortAllocatorFlags);
   // In order to simulate nothing working but outgoing TCP connections, prevent
@@ -1847,17 +1849,15 @@ TEST_F(P2PTransportChannelTest, CanOnlyMakeOutgoingTcpConnections) {
   // "any" addresses. It can then only make a connection by using "Connect()".
   fw()->SetUnbindableIps(
       {GetAnyIP(AF_INET), GetAnyIP(AF_INET6), kPublicAddrs[0].ipaddr()});
-  CreateChannels(env);
+  CreateChannels();
   // Expect a IceCandidateType::kPrflx candidate on the side that can only make
   // outgoing connections, endpoint 0.
-  Test(env, kPrflxTcpToLocalTcp);
+  Test(kPrflxTcpToLocalTcp);
   DestroyChannels();
 }
 
 TEST_F(P2PTransportChannelTest, TestTcpConnectionsFromActiveToPassive) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-  CreatePortAllocators(env);
+  CreatePortAllocators();
   AddAddress(0, kPublicAddrs[0]);
   AddAddress(1, kPublicAddrs[1]);
 
@@ -1882,7 +1882,7 @@ TEST_F(P2PTransportChannelTest, TestTcpConnectionsFromActiveToPassive) {
   // Pause candidate so we could verify the candidate properties.
   PauseCandidates(0);
   PauseCandidates(1);
-  CreateChannels(env);
+  CreateChannels();
 
   // Verify tcp candidates.
   VerifySavedTcpCandidates(0, TCPTYPE_PASSIVE_STR);
@@ -1892,32 +1892,28 @@ TEST_F(P2PTransportChannelTest, TestTcpConnectionsFromActiveToPassive) {
   ResumeCandidates(0);
   ResumeCandidates(1);
 
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return CheckCandidatePairAndConnected(ep1_ch1(), ep2_ch1(),
-                                              kPublicAddrs[0], kPublicAddrs[1]);
-      },
-      {.timeout = kShortTimeout, .clock = &clock}));
+  EXPECT_TRUE(MediumWait().Until([&] {
+    return CheckCandidatePairAndConnected(ep1_ch1(), ep2_ch1(), kPublicAddrs[0],
+                                          kPublicAddrs[1]);
+  }));
 
-  TestSendRecv(&clock);
+  TestSendRecv();
   DestroyChannels();
 }
 
 // Test that tcptype is set on all candidates for a connection running over TCP.
 TEST_F(P2PTransportChannelTest, TestTcpConnectionTcptypeSet) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-  ConfigureEndpoints(env, BLOCK_UDP_AND_INCOMING_TCP, OPEN,
+  ConfigureEndpoints(BLOCK_UDP_AND_INCOMING_TCP, OPEN,
                      PORTALLOCATOR_ENABLE_SHARED_SOCKET,
                      PORTALLOCATOR_ENABLE_SHARED_SOCKET);
 
   SetAllowTcpListen(0, false);  // active.
   SetAllowTcpListen(1, true);   // actpass.
-  CreateChannels(env);
+  CreateChannels();
 
-  EXPECT_TRUE(WaitUntil([&] { return CheckConnected(ep1_ch1(), ep2_ch1()); },
-                        {.timeout = kMediumTimeout, .clock = &clock}));
-  SIMULATED_WAIT(false, kDefaultTimeout.ms(), clock);
+  EXPECT_TRUE(
+      ShortWait().Until([&] { return CheckConnected(ep1_ch1(), ep2_ch1()); }));
+  time_controller_.AdvanceTime(kDefaultTimeout);
 
   EXPECT_EQ(RemoteCandidate(ep1_ch1())->tcptype(), "passive");
   EXPECT_EQ(LocalCandidate(ep1_ch1())->tcptype(), "active");
@@ -1928,9 +1924,7 @@ TEST_F(P2PTransportChannelTest, TestTcpConnectionTcptypeSet) {
 }
 
 TEST_F(P2PTransportChannelTest, TestIceRoleConflict) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-  CreatePortAllocators(env);
+  CreatePortAllocators();
   AddAddress(0, kPublicAddrs[0]);
   AddAddress(1, kPublicAddrs[1]);
 
@@ -1938,52 +1932,47 @@ TEST_F(P2PTransportChannelTest, TestIceRoleConflict) {
   SetIceRole(0, ICEROLE_CONTROLLING);
   SetIceRole(1, ICEROLE_CONTROLLING);
 
-  CreateChannels(env);
+  CreateChannels();
   bool first_endpoint_has_lower_tiebreaker =
-      GetEndpoint(0)->allocator_->ice_tiebreaker() <
-      GetEndpoint(1)->allocator_->ice_tiebreaker();
+      GetEndpoint(0)->allocator()->ice_tiebreaker() <
+      GetEndpoint(1)->allocator()->ice_tiebreaker();
   // Since both the channels initiated with controlling state, the channel with
   // the lower tiebreaker should receive SignalRoleConflict.
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return GetRoleConflict(first_endpoint_has_lower_tiebreaker ? 0 : 1);
-      },
-      {.timeout = kShortTimeout, .clock = &clock}));
+  EXPECT_TRUE(MediumWait().Until([&] {
+    return GetRoleConflict(first_endpoint_has_lower_tiebreaker ? 0 : 1);
+  }));
   EXPECT_FALSE(GetRoleConflict(first_endpoint_has_lower_tiebreaker ? 1 : 0));
 
-  EXPECT_TRUE(WaitUntil([&] { return CheckConnected(ep1_ch1(), ep2_ch1()); },
-                        {.timeout = kShortTimeout, .clock = &clock}));
+  EXPECT_TRUE(
+      ShortWait().Until([&] { return CheckConnected(ep1_ch1(), ep2_ch1()); }));
 
   EXPECT_TRUE(ep1_ch1()->selected_connection() &&
               ep2_ch1()->selected_connection());
 
-  TestSendRecv(&clock);
+  TestSendRecv();
   DestroyChannels();
 }
 
 // Tests that the ice configs (protocol and role) can be passed down to ports.
 TEST_F(P2PTransportChannelTest, TestIceConfigWillPassDownToPort) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-  CreatePortAllocators(env);
+  CreatePortAllocators();
   AddAddress(0, kPublicAddrs[0]);
   AddAddress(1, kPublicAddrs[1]);
 
   SetIceRole(0, ICEROLE_CONTROLLING);
   SetIceRole(1, ICEROLE_CONTROLLING);
 
-  CreateChannels(env);
+  CreateChannels();
 
   // Pick channel with the higher tiebreaker so its role won't change unless we
   // tell it to.
   P2PTransportChannel* channel =
-      GetEndpoint(0)->allocator_->ice_tiebreaker() >
-              GetEndpoint(1)->allocator_->ice_tiebreaker()
+      GetEndpoint(0)->allocator()->ice_tiebreaker() >
+              GetEndpoint(1)->allocator()->ice_tiebreaker()
           ? ep1_ch1()
           : ep2_ch1();
 
-  EXPECT_THAT(WaitUntil([&] { return channel->ports(); }, SizeIs(2),
-                        {.timeout = kShortTimeout, .clock = &clock}),
+  EXPECT_THAT(ShortWait().Until([&] { return channel->ports(); }, SizeIs(2)),
               IsRtcOk());
 
   EXPECT_THAT(channel->ports(), Each(Property(&PortInterface::GetIceRole,
@@ -1994,42 +1983,39 @@ TEST_F(P2PTransportChannelTest, TestIceConfigWillPassDownToPort) {
   EXPECT_THAT(channel->ports(), Each(Property(&PortInterface::GetIceRole,
                                               Eq(ICEROLE_CONTROLLED))));
 
-  EXPECT_TRUE(WaitUntil([&] { return CheckConnected(ep1_ch1(), ep2_ch1()); },
-                        {.timeout = kShortTimeout, .clock = &clock}));
+  EXPECT_TRUE(
+      ShortWait().Until([&] { return CheckConnected(ep1_ch1(), ep2_ch1()); }));
 
   EXPECT_TRUE(ep1_ch1()->selected_connection() &&
               ep2_ch1()->selected_connection());
 
-  TestSendRecv(&clock);
+  TestSendRecv();
   DestroyChannels();
 }
 
 // Verify that we can set DSCP value and retrieve properly from P2PTC.
 TEST_F(P2PTransportChannelTest, TestDefaultDscpValue) {
-  const Environment env = CreateEnvironment();
-  CreatePortAllocators(env);
+  CreatePortAllocators();
   AddAddress(0, kPublicAddrs[0]);
   AddAddress(1, kPublicAddrs[1]);
 
-  CreateChannels(env);
-  EXPECT_EQ(DSCP_NO_CHANGE, GetEndpoint(0)->cd1_.ch_->DefaultDscpValue());
-  EXPECT_EQ(DSCP_NO_CHANGE, GetEndpoint(1)->cd1_.ch_->DefaultDscpValue());
-  GetEndpoint(0)->cd1_.ch_->SetOption(Socket::OPT_DSCP, DSCP_CS6);
-  GetEndpoint(1)->cd1_.ch_->SetOption(Socket::OPT_DSCP, DSCP_CS6);
-  EXPECT_EQ(DSCP_CS6, GetEndpoint(0)->cd1_.ch_->DefaultDscpValue());
-  EXPECT_EQ(DSCP_CS6, GetEndpoint(1)->cd1_.ch_->DefaultDscpValue());
-  GetEndpoint(0)->cd1_.ch_->SetOption(Socket::OPT_DSCP, DSCP_AF41);
-  GetEndpoint(1)->cd1_.ch_->SetOption(Socket::OPT_DSCP, DSCP_AF41);
-  EXPECT_EQ(DSCP_AF41, GetEndpoint(0)->cd1_.ch_->DefaultDscpValue());
-  EXPECT_EQ(DSCP_AF41, GetEndpoint(1)->cd1_.ch_->DefaultDscpValue());
+  CreateChannels();
+  EXPECT_EQ(DSCP_NO_CHANGE, GetEndpoint(0)->cd1().ch()->DefaultDscpValue());
+  EXPECT_EQ(DSCP_NO_CHANGE, GetEndpoint(1)->cd1().ch()->DefaultDscpValue());
+  GetEndpoint(0)->cd1().ch()->SetOption(Socket::OPT_DSCP, DSCP_CS6);
+  GetEndpoint(1)->cd1().ch()->SetOption(Socket::OPT_DSCP, DSCP_CS6);
+  EXPECT_EQ(DSCP_CS6, GetEndpoint(0)->cd1().ch()->DefaultDscpValue());
+  EXPECT_EQ(DSCP_CS6, GetEndpoint(1)->cd1().ch()->DefaultDscpValue());
+  GetEndpoint(0)->cd1().ch()->SetOption(Socket::OPT_DSCP, DSCP_AF41);
+  GetEndpoint(1)->cd1().ch()->SetOption(Socket::OPT_DSCP, DSCP_AF41);
+  EXPECT_EQ(DSCP_AF41, GetEndpoint(0)->cd1().ch()->DefaultDscpValue());
+  EXPECT_EQ(DSCP_AF41, GetEndpoint(1)->cd1().ch()->DefaultDscpValue());
   DestroyChannels();
 }
 
 // Verify IPv6 connection is preferred over IPv4.
 TEST_F(P2PTransportChannelTest, TestIPv6Connections) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-  CreatePortAllocators(env);
+  CreatePortAllocators();
   AddAddress(0, kIPv6PublicAddrs[0]);
   AddAddress(0, kPublicAddrs[0]);
   AddAddress(1, kIPv6PublicAddrs[1]);
@@ -2044,25 +2030,21 @@ TEST_F(P2PTransportChannelTest, TestIPv6Connections) {
   SetAllocatorFlags(
       1, PORTALLOCATOR_ENABLE_IPV6 | PORTALLOCATOR_ENABLE_IPV6_ON_WIFI);
 
-  CreateChannels(env);
+  CreateChannels();
 
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return CheckCandidatePairAndConnected(
-            ep1_ch1(), ep2_ch1(), kIPv6PublicAddrs[0], kIPv6PublicAddrs[1]);
-      },
-      {.timeout = kShortTimeout, .clock = &clock}));
+  EXPECT_TRUE(ShortWait().Until([&] {
+    return CheckCandidatePairAndConnected(
+        ep1_ch1(), ep2_ch1(), kIPv6PublicAddrs[0], kIPv6PublicAddrs[1]);
+  }));
 
-  TestSendRecv(&clock);
+  TestSendRecv();
   DestroyChannels();
 }
 
 // Testing forceful TURN connections.
 TEST_F(P2PTransportChannelTest, TestForceTurn) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
   ConfigureEndpoints(
-      env, NAT_PORT_RESTRICTED, NAT_SYMMETRIC,
+      NAT_PORT_RESTRICTED, NAT_SYMMETRIC,
       kDefaultPortAllocatorFlags | PORTALLOCATOR_ENABLE_SHARED_SOCKET,
       kDefaultPortAllocatorFlags | PORTALLOCATOR_ENABLE_SHARED_SOCKET);
   set_force_relay(true);
@@ -2070,10 +2052,10 @@ TEST_F(P2PTransportChannelTest, TestForceTurn) {
   SetAllocationStepDelay(0, kMinimumStepDelay);
   SetAllocationStepDelay(1, kMinimumStepDelay);
 
-  CreateChannels(env);
+  CreateChannels();
 
-  EXPECT_TRUE(WaitUntil([&] { return CheckConnected(ep1_ch1(), ep2_ch1()); },
-                        {.timeout = kMediumTimeout, .clock = &clock}));
+  EXPECT_TRUE(
+      ShortWait().Until([&] { return CheckConnected(ep1_ch1(), ep2_ch1()); }));
 
   EXPECT_TRUE(ep1_ch1()->selected_connection() &&
               ep2_ch1()->selected_connection());
@@ -2083,16 +2065,14 @@ TEST_F(P2PTransportChannelTest, TestForceTurn) {
   EXPECT_TRUE(RemoteCandidate(ep2_ch1())->is_relay());
   EXPECT_TRUE(LocalCandidate(ep2_ch1())->is_relay());
 
-  TestSendRecv(&clock);
+  TestSendRecv();
   DestroyChannels();
 }
 
 // Test that if continual gathering is set to true, ICE gathering state will
 // not change to "Complete", and vice versa.
 TEST_F(P2PTransportChannelTest, TestContinualGathering) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-  ConfigureEndpoints(env, OPEN, OPEN, kDefaultPortAllocatorFlags,
+  ConfigureEndpoints(OPEN, OPEN, kDefaultPortAllocatorFlags,
                      kDefaultPortAllocatorFlags);
   SetAllocationStepDelay(0, kDefaultStepDelay);
   SetAllocationStepDelay(1, kDefaultStepDelay);
@@ -2100,13 +2080,14 @@ TEST_F(P2PTransportChannelTest, TestContinualGathering) {
       CreateIceConfig(TimeDelta::Seconds(1), GATHER_CONTINUALLY);
   // By default, ep2 does not gather continually.
   IceConfig default_config;
-  CreateChannels(env, continual_gathering_config, default_config);
+  CreateChannels(continual_gathering_config, default_config);
 
-  EXPECT_TRUE(WaitUntil([&] { return CheckConnected(ep1_ch1(), ep2_ch1()); },
-                        {.timeout = kMediumTimeout, .clock = &clock}));
-  SIMULATED_WAIT(
-      IceGatheringState::kIceGatheringComplete == ep1_ch1()->gathering_state(),
-      kShortTimeout.ms(), clock);
+  EXPECT_TRUE(
+      MediumWait().Until([&] { return CheckConnected(ep1_ch1(), ep2_ch1()); }));
+  (void)MediumWait().Until([&] {
+    return IceGatheringState::kIceGatheringComplete ==
+           ep1_ch1()->gathering_state();
+  });
   EXPECT_EQ(IceGatheringState::kIceGatheringGathering,
             ep1_ch1()->gathering_state());
   // By now, ep2 should have completed gathering.
@@ -2119,13 +2100,11 @@ TEST_F(P2PTransportChannelTest, TestContinualGathering) {
 // Test that a connection succeeds when the P2PTransportChannel uses a pooled
 // PortAllocatorSession that has not yet finished gathering candidates.
 TEST_F(P2PTransportChannelTest, TestUsingPooledSessionBeforeDoneGathering) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-  ConfigureEndpoints(env, OPEN, OPEN, kDefaultPortAllocatorFlags,
+  ConfigureEndpoints(OPEN, OPEN, kDefaultPortAllocatorFlags,
                      kDefaultPortAllocatorFlags);
   // First create a pooled session for each endpoint.
-  auto& allocator_1 = GetEndpoint(0)->allocator_;
-  auto& allocator_2 = GetEndpoint(1)->allocator_;
+  auto* allocator_1 = GetEndpoint(0)->allocator();
+  auto* allocator_2 = GetEndpoint(1)->allocator();
   int pool_size = 1;
   allocator_1->SetConfiguration(allocator_1->stun_servers(),
                                 allocator_1->turn_servers(), pool_size,
@@ -2145,10 +2124,10 @@ TEST_F(P2PTransportChannelTest, TestUsingPooledSessionBeforeDoneGathering) {
   EXPECT_TRUE(pooled_session_2->ReadyPorts().empty());
   EXPECT_TRUE(pooled_session_2->ReadyCandidates().empty());
   // Now let the endpoints connect and try exchanging some data.
-  CreateChannels(env);
-  EXPECT_TRUE(WaitUntil([&] { return CheckConnected(ep1_ch1(), ep2_ch1()); },
-                        {.timeout = kMediumTimeout, .clock = &clock}));
-  TestSendRecv(&clock);
+  CreateChannels();
+  EXPECT_TRUE(
+      ShortWait().Until([&] { return CheckConnected(ep1_ch1(), ep2_ch1()); }));
+  TestSendRecv();
   // Make sure the P2PTransportChannels are actually using ports from the
   // pooled sessions.
   auto pooled_ports_1 = pooled_session_1->ReadyPorts();
@@ -2163,13 +2142,11 @@ TEST_F(P2PTransportChannelTest, TestUsingPooledSessionBeforeDoneGathering) {
 // Test that a connection succeeds when the P2PTransportChannel uses a pooled
 // PortAllocatorSession that already finished gathering candidates.
 TEST_F(P2PTransportChannelTest, TestUsingPooledSessionAfterDoneGathering) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-  ConfigureEndpoints(env, OPEN, OPEN, kDefaultPortAllocatorFlags,
+  ConfigureEndpoints(OPEN, OPEN, kDefaultPortAllocatorFlags,
                      kDefaultPortAllocatorFlags);
   // First create a pooled session for each endpoint.
-  auto& allocator_1 = GetEndpoint(0)->allocator_;
-  auto& allocator_2 = GetEndpoint(1)->allocator_;
+  auto* allocator_1 = GetEndpoint(0)->allocator();
+  auto* allocator_2 = GetEndpoint(1)->allocator();
   int pool_size = 1;
   allocator_1->SetConfiguration(allocator_1->stun_servers(),
                                 allocator_1->turn_servers(), pool_size,
@@ -2185,17 +2162,15 @@ TEST_F(P2PTransportChannelTest, TestUsingPooledSessionAfterDoneGathering) {
   ASSERT_NE(nullptr, pooled_session_2);
   // Wait for the pooled sessions to finish gathering before the
   // P2PTransportChannels try to use them.
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return pooled_session_1->CandidatesAllocationDone() &&
-               pooled_session_2->CandidatesAllocationDone();
-      },
-      {.timeout = kDefaultTimeout, .clock = &clock}));
+  EXPECT_TRUE(MediumWait().Until([&] {
+    return pooled_session_1->CandidatesAllocationDone() &&
+           pooled_session_2->CandidatesAllocationDone();
+  }));
   // Now let the endpoints connect and try exchanging some data.
-  CreateChannels(env);
-  EXPECT_TRUE(WaitUntil([&] { return CheckConnected(ep1_ch1(), ep2_ch1()); },
-                        {.timeout = kMediumTimeout, .clock = &clock}));
-  TestSendRecv(&clock);
+  CreateChannels();
+  EXPECT_TRUE(DefaultWait().Until(
+      [&] { return CheckConnected(ep1_ch1(), ep2_ch1()); }));
+  TestSendRecv();
   // Make sure the P2PTransportChannels are actually using ports from the
   // pooled sessions.
   auto pooled_ports_1 = pooled_session_1->ReadyPorts();
@@ -2214,20 +2189,18 @@ TEST_F(P2PTransportChannelTest, TestUsingPooledSessionAfterDoneGathering) {
 // class that operates on a single P2PTransportChannel, once an appropriate one
 // (which supports TURN servers and TURN candidate gathering) is available.
 TEST_F(P2PTransportChannelTest, TurnToTurnPresumedWritable) {
-  const Environment env = CreateEnvironment();
-  ConfigureEndpoints(env, OPEN, OPEN, kDefaultPortAllocatorFlags,
+  ConfigureEndpoints(OPEN, OPEN, kDefaultPortAllocatorFlags,
                      kDefaultPortAllocatorFlags);
   // Only configure one channel so we can control when the remote candidate
   // is added.
-  GetEndpoint(0)->cd1_.ch_ = CreateChannel(
-      env, 0, ICE_CANDIDATE_COMPONENT_DEFAULT, kIceParams[0], kIceParams[1]);
+  GetEndpoint(0)->cd1().set_ch(CreateChannel(0, ICE_CANDIDATE_COMPONENT_DEFAULT,
+                                             kIceParams[0], kIceParams[1]));
   IceConfig config;
   config.presume_writable_when_fully_relayed = true;
   ep1_ch1()->SetIceConfig(config);
   ep1_ch1()->MaybeStartGathering();
-  EXPECT_THAT(WaitUntil([&] { return ep1_ch1()->gathering_state(); },
-                        Eq(IceGatheringState::kIceGatheringComplete),
-                        {.timeout = kDefaultTimeout}),
+  EXPECT_THAT(MediumWait().Until([&] { return ep1_ch1()->gathering_state(); },
+                                 Eq(IceGatheringState::kIceGatheringComplete)),
               IsRtcOk());
   // Add two remote candidates; a host candidate (with higher priority)
   // and TURN candidate.
@@ -2237,15 +2210,16 @@ TEST_F(P2PTransportChannelTest, TurnToTurnPresumedWritable) {
       CreateUdpCandidate(IceCandidateType::kRelay, "2.2.2.2", 2, 0));
   // Expect that the TURN-TURN candidate pair will be prioritized since it's
   // "probably writable".
-  EXPECT_THAT(WaitUntil([&] { return ep1_ch1()->selected_connection(); },
-                        Ne(nullptr), {.timeout = kShortTimeout}),
-              IsRtcOk());
+  EXPECT_THAT(
+      DefaultWait().Until([&] { return ep1_ch1()->selected_connection(); },
+                          Ne(nullptr)),
+      IsRtcOk());
   EXPECT_TRUE(LocalCandidate(ep1_ch1())->is_relay());
   EXPECT_TRUE(RemoteCandidate(ep1_ch1())->is_relay());
   // Also expect that the channel instantly indicates that it's writable since
   // it has a TURN-TURN pair.
   EXPECT_TRUE(ep1_ch1()->writable());
-  EXPECT_TRUE(GetEndpoint(0)->ready_to_send_);
+  EXPECT_TRUE(GetEndpoint(0)->ready_to_send());
   // Also make sure we can immediately send packets.
   const char* data = "test";
   int len = static_cast<int>(strlen(data));
@@ -2256,27 +2230,24 @@ TEST_F(P2PTransportChannelTest, TurnToTurnPresumedWritable) {
 
 // Test that a TURN/peer reflexive candidate pair is also presumed writable.
 TEST_F(P2PTransportChannelTest, TurnToPrflxPresumedWritable) {
-  ScopedFakeClock fake_clock;
-  const Environment env = CreateEnvironment();
-
   // We need to add artificial network delay to verify that the connection
   // is presumed writable before it's actually writable. Without this delay
   // it would become writable instantly.
   virtual_socket_server()->set_delay_mean(50);
   virtual_socket_server()->UpdateDelayDistribution();
 
-  ConfigureEndpoints(env, NAT_SYMMETRIC, NAT_SYMMETRIC,
-                     kDefaultPortAllocatorFlags, kDefaultPortAllocatorFlags);
+  ConfigureEndpoints(NAT_SYMMETRIC, NAT_SYMMETRIC, kDefaultPortAllocatorFlags,
+                     kDefaultPortAllocatorFlags);
   // We want the remote TURN candidate to show up as prflx. To do this we need
   // to configure the server to accept packets from an address we haven't
   // explicitly installed permission for.
   test_turn_server()->set_enable_permission_checks(false);
   IceConfig config;
   config.presume_writable_when_fully_relayed = true;
-  GetEndpoint(0)->cd1_.ch_ = CreateChannel(
-      env, 0, ICE_CANDIDATE_COMPONENT_DEFAULT, kIceParams[0], kIceParams[1]);
-  GetEndpoint(1)->cd1_.ch_ = CreateChannel(
-      env, 1, ICE_CANDIDATE_COMPONENT_DEFAULT, kIceParams[1], kIceParams[0]);
+  GetEndpoint(0)->cd1().set_ch(CreateChannel(0, ICE_CANDIDATE_COMPONENT_DEFAULT,
+                                             kIceParams[0], kIceParams[1]));
+  GetEndpoint(1)->cd1().set_ch(CreateChannel(1, ICE_CANDIDATE_COMPONENT_DEFAULT,
+                                             kIceParams[1], kIceParams[0]));
   ep1_ch1()->SetIceConfig(config);
   ep2_ch1()->SetIceConfig(config);
   // Don't signal candidates from channel 2, so that channel 1 sees the TURN
@@ -2286,9 +2257,8 @@ TEST_F(P2PTransportChannelTest, TurnToPrflxPresumedWritable) {
   ep2_ch1()->MaybeStartGathering();
 
   // Wait for the TURN<->prflx connection.
-  EXPECT_TRUE(
-      WaitUntil([&] { return ep1_ch1()->receiving() && ep1_ch1()->writable(); },
-                {.timeout = kShortTimeout, .clock = &fake_clock}));
+  EXPECT_TRUE(ShortWait().Until(
+      [&] { return ep1_ch1()->receiving() && ep1_ch1()->writable(); }));
   ASSERT_NE(nullptr, ep1_ch1()->selected_connection());
   EXPECT_TRUE(LocalCandidate(ep1_ch1())->is_relay());
   EXPECT_TRUE(RemoteCandidate(ep1_ch1())->is_prflx());
@@ -2297,9 +2267,8 @@ TEST_F(P2PTransportChannelTest, TurnToPrflxPresumedWritable) {
   EXPECT_FALSE(ep1_ch1()->selected_connection()->writable());
 
   // Now wait for it to actually become writable.
-  EXPECT_TRUE(
-      WaitUntil([&] { return ep1_ch1()->selected_connection()->writable(); },
-                {.timeout = kShortTimeout, .clock = &fake_clock}));
+  EXPECT_TRUE(ShortWait().Until(
+      [&] { return ep1_ch1()->selected_connection()->writable(); }));
 
   // Explitly destroy channels, before fake clock is destroyed.
   DestroyChannels();
@@ -2308,30 +2277,26 @@ TEST_F(P2PTransportChannelTest, TurnToPrflxPresumedWritable) {
 // Test that a presumed-writable TURN<->TURN connection is preferred above an
 // unreliable connection (one that has failed to be pinged for some time).
 TEST_F(P2PTransportChannelTest, PresumedWritablePreferredOverUnreliable) {
-  ScopedFakeClock fake_clock;
-  const Environment env = CreateEnvironment();
-
-  ConfigureEndpoints(env, NAT_SYMMETRIC, NAT_SYMMETRIC,
-                     kDefaultPortAllocatorFlags, kDefaultPortAllocatorFlags);
+  ConfigureEndpoints(NAT_SYMMETRIC, NAT_SYMMETRIC, kDefaultPortAllocatorFlags,
+                     kDefaultPortAllocatorFlags);
   IceConfig config;
   config.presume_writable_when_fully_relayed = true;
-  GetEndpoint(0)->cd1_.ch_ = CreateChannel(
-      env, 0, ICE_CANDIDATE_COMPONENT_DEFAULT, kIceParams[0], kIceParams[1]);
-  GetEndpoint(1)->cd1_.ch_ = CreateChannel(
-      env, 1, ICE_CANDIDATE_COMPONENT_DEFAULT, kIceParams[1], kIceParams[0]);
+  GetEndpoint(0)->cd1().set_ch(CreateChannel(0, ICE_CANDIDATE_COMPONENT_DEFAULT,
+                                             kIceParams[0], kIceParams[1]));
+  GetEndpoint(1)->cd1().set_ch(CreateChannel(1, ICE_CANDIDATE_COMPONENT_DEFAULT,
+                                             kIceParams[1], kIceParams[0]));
   ep1_ch1()->SetIceConfig(config);
   ep2_ch1()->SetIceConfig(config);
   ep1_ch1()->MaybeStartGathering();
   ep2_ch1()->MaybeStartGathering();
   // Wait for initial connection as usual.
-  EXPECT_TRUE(WaitUntil([&] { return CheckConnected(ep1_ch1(), ep2_ch1()); },
-                        {.timeout = kShortTimeout, .clock = &fake_clock}));
+  EXPECT_TRUE(DefaultWait().Until(
+      [&] { return CheckConnected(ep1_ch1(), ep2_ch1()); }));
   const Connection* old_selected_connection = ep1_ch1()->selected_connection();
   // Destroy the second channel and wait for the current connection on the
   // first channel to become "unreliable", making it no longer writable.
-  GetEndpoint(1)->cd1_.ch_.reset();
-  EXPECT_TRUE(WaitUntil([&] { return !ep1_ch1()->writable(); },
-                        {.timeout = kDefaultTimeout, .clock = &fake_clock}));
+  GetEndpoint(1)->cd1().reset_ch();
+  EXPECT_TRUE(DefaultWait().Until([&] { return !ep1_ch1()->writable(); }));
   EXPECT_NE(nullptr, ep1_ch1()->selected_connection());
   // Add a remote TURN candidate. The first channel should still have a TURN
   // port available to make a TURN<->TURN pair that's presumed writable.
@@ -2340,7 +2305,7 @@ TEST_F(P2PTransportChannelTest, PresumedWritablePreferredOverUnreliable) {
   EXPECT_TRUE(LocalCandidate(ep1_ch1())->is_relay());
   EXPECT_TRUE(RemoteCandidate(ep1_ch1())->is_relay());
   EXPECT_TRUE(ep1_ch1()->writable());
-  EXPECT_TRUE(GetEndpoint(0)->ready_to_send_);
+  EXPECT_TRUE(GetEndpoint(0)->ready_to_send());
   EXPECT_NE(old_selected_connection, ep1_ch1()->selected_connection());
   // Explitly destroy channels, before fake clock is destroyed.
   DestroyChannels();
@@ -2349,27 +2314,26 @@ TEST_F(P2PTransportChannelTest, PresumedWritablePreferredOverUnreliable) {
 // Ensure that "SignalReadyToSend" is fired as expected with a "presumed
 // writable" connection. Previously this did not work.
 TEST_F(P2PTransportChannelTest, SignalReadyToSendWithPresumedWritable) {
-  const Environment env = CreateEnvironment();
-  ConfigureEndpoints(env, OPEN, OPEN, kDefaultPortAllocatorFlags,
+  ConfigureEndpoints(OPEN, OPEN, kDefaultPortAllocatorFlags,
                      kDefaultPortAllocatorFlags);
   // Only test one endpoint, so we can ensure the connection doesn't receive a
   // binding response and advance beyond being "presumed" writable.
-  GetEndpoint(0)->cd1_.ch_ = CreateChannel(
-      env, 0, ICE_CANDIDATE_COMPONENT_DEFAULT, kIceParams[0], kIceParams[1]);
+  GetEndpoint(0)->cd1().set_ch(CreateChannel(0, ICE_CANDIDATE_COMPONENT_DEFAULT,
+                                             kIceParams[0], kIceParams[1]));
   IceConfig config;
   config.presume_writable_when_fully_relayed = true;
   ep1_ch1()->SetIceConfig(config);
   ep1_ch1()->MaybeStartGathering();
-  EXPECT_THAT(WaitUntil([&] { return ep1_ch1()->gathering_state(); },
-                        Eq(IceGatheringState::kIceGatheringComplete),
-                        {.timeout = kDefaultTimeout}),
+  EXPECT_THAT(DefaultWait().Until([&] { return ep1_ch1()->gathering_state(); },
+                                  Eq(IceGatheringState::kIceGatheringComplete)),
               IsRtcOk());
   ep1_ch1()->AddRemoteCandidate(
       CreateUdpCandidate(IceCandidateType::kRelay, "1.1.1.1", 1, 0));
   // Sanity checking the type of the connection.
-  EXPECT_THAT(WaitUntil([&] { return ep1_ch1()->selected_connection(); },
-                        Ne(nullptr), {.timeout = kShortTimeout}),
-              IsRtcOk());
+  EXPECT_THAT(
+      DefaultWait().Until([&] { return ep1_ch1()->selected_connection(); },
+                          Ne(nullptr)),
+      IsRtcOk());
   EXPECT_TRUE(LocalCandidate(ep1_ch1())->is_relay());
   EXPECT_TRUE(RemoteCandidate(ep1_ch1())->is_relay());
 
@@ -2381,9 +2345,9 @@ TEST_F(P2PTransportChannelTest, SignalReadyToSendWithPresumedWritable) {
 
   // Reset `ready_to_send_` flag, which is set to true if the event fires as it
   // should.
-  GetEndpoint(0)->ready_to_send_ = false;
+  GetEndpoint(0)->set_ready_to_send(false);
   virtual_socket_server()->SetSendingBlocked(false);
-  EXPECT_TRUE(GetEndpoint(0)->ready_to_send_);
+  EXPECT_TRUE(GetEndpoint(0)->ready_to_send());
   EXPECT_EQ(len, SendData(ep1_ch1(), data, len));
   DestroyChannels();
 }
@@ -2393,10 +2357,8 @@ TEST_F(P2PTransportChannelTest, SignalReadyToSendWithPresumedWritable) {
 // crbug.com/webrtc/9034.
 TEST_F(P2PTransportChannelTest,
        TurnToPrflxSelectedAfterResolvingIceControllingRoleConflict) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
   // Gather only relay candidates.
-  ConfigureEndpoints(env, NAT_SYMMETRIC, NAT_SYMMETRIC,
+  ConfigureEndpoints(NAT_SYMMETRIC, NAT_SYMMETRIC,
                      kDefaultPortAllocatorFlags | PORTALLOCATOR_DISABLE_UDP |
                          PORTALLOCATOR_DISABLE_STUN | PORTALLOCATOR_DISABLE_TCP,
                      kDefaultPortAllocatorFlags | PORTALLOCATOR_DISABLE_UDP |
@@ -2409,19 +2371,18 @@ TEST_F(P2PTransportChannelTest,
   // to configure the server to accept packets from an address we haven't
   // explicitly installed permission for.
   test_turn_server()->set_enable_permission_checks(false);
-  GetEndpoint(0)->cd1_.ch_ = CreateChannel(
-      env, 0, ICE_CANDIDATE_COMPONENT_DEFAULT, kIceParams[0], kIceParams[1]);
-  GetEndpoint(1)->cd1_.ch_ = CreateChannel(
-      env, 1, ICE_CANDIDATE_COMPONENT_DEFAULT, kIceParams[1], kIceParams[0]);
+  GetEndpoint(0)->cd1().set_ch(CreateChannel(0, ICE_CANDIDATE_COMPONENT_DEFAULT,
+                                             kIceParams[0], kIceParams[1]));
+  GetEndpoint(1)->cd1().set_ch(CreateChannel(1, ICE_CANDIDATE_COMPONENT_DEFAULT,
+                                             kIceParams[1], kIceParams[0]));
   // Don't signal candidates from channel 2, so that channel 1 sees the TURN
   // candidate as peer reflexive.
   PauseCandidates(1);
   ep1_ch1()->MaybeStartGathering();
   ep2_ch1()->MaybeStartGathering();
 
-  EXPECT_TRUE(
-      WaitUntil([&] { return ep1_ch1()->receiving() && ep1_ch1()->writable(); },
-                {.timeout = kMediumTimeout, .clock = &clock}));
+  EXPECT_TRUE(ShortWait().Until(
+      [&] { return ep1_ch1()->receiving() && ep1_ch1()->writable(); }));
 
   ASSERT_NE(nullptr, ep1_ch1()->selected_connection());
 
@@ -2435,10 +2396,11 @@ TEST_F(P2PTransportChannelTest,
 // acknowledgement in the connectivity check from the remote peer.
 TEST_F(P2PTransportChannelTest,
        CanConnectWithPiggybackCheckAcknowledgementWhenCheckResponseBlocked) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment(CreateTestFieldTrialsPtr(
-      "WebRTC-PiggybackIceCheckAcknowledgement/Enabled/"));
-  ConfigureEndpoints(env, OPEN, OPEN, kOnlyLocalPorts, kOnlyLocalPorts);
+  env_ = CreateTestEnvironment(
+      {.field_trials = CreateTestFieldTrialsPtr(
+           "WebRTC-PiggybackIceCheckAcknowledgement/Enabled/"),
+       .time = &time_controller_});
+  ConfigureEndpoints(OPEN, OPEN, kOnlyLocalPorts, kOnlyLocalPorts);
   IceConfig ep1_config;
   IceConfig ep2_config =
       CreateIceConfig(TimeDelta::Seconds(1), GATHER_CONTINUALLY);
@@ -2450,20 +2412,19 @@ TEST_F(P2PTransportChannelTest,
   ep2_config.ice_unwritable_min_checks = 30;
   ep2_config.ice_inactive_timeout = TimeDelta::Seconds(60);
 
-  CreateChannels(env, ep1_config, ep2_config);
+  CreateChannels(ep1_config, ep2_config);
 
   // Wait until both sides become writable for the first time.
-  EXPECT_TRUE(WaitUntil([&] { return CheckConnected(ep1_ch1(), ep2_ch1()); },
-                        {.timeout = kDefaultTimeout, .clock = &clock}));
+  EXPECT_TRUE(
+      MediumWait().Until([&] { return CheckConnected(ep1_ch1(), ep2_ch1()); }));
   // Block the ingress traffic to ep1 so that there is no check response from
   // ep2.
   ASSERT_NE(nullptr, LocalCandidate(ep1_ch1()));
   fw()->AddRule(false, FP_ANY, FD_IN, LocalCandidate(ep1_ch1())->address());
   // Wait until ep1 becomes unwritable. At the same time ep2 should be still
   // fine so that it will keep sending pings.
-  EXPECT_TRUE(
-      WaitUntil([&] { return ep1_ch1() != nullptr && !ep1_ch1()->writable(); },
-                {.timeout = kDefaultTimeout, .clock = &clock}));
+  EXPECT_TRUE(DefaultWait().Until(
+      [&] { return ep1_ch1() != nullptr && !ep1_ch1()->writable(); }));
   EXPECT_TRUE(ep2_ch1() != nullptr && ep2_ch1()->writable());
   // Now let the pings from ep2 to flow but block any pings from ep1, so that
   // ep1 can only become writable again after receiving an incoming ping from
@@ -2472,9 +2433,8 @@ TEST_F(P2PTransportChannelTest,
   // in the current design.
   fw()->ClearRules();
   fw()->AddRule(false, FP_ANY, FD_OUT, LocalCandidate(ep1_ch1())->address());
-  EXPECT_TRUE(
-      WaitUntil([&] { return ep1_ch1() != nullptr && ep1_ch1()->writable(); },
-                {.timeout = kDefaultTimeout, .clock = &clock}));
+  EXPECT_TRUE(DefaultWait().Until(
+      [&] { return ep1_ch1() != nullptr && ep1_ch1()->writable(); }));
   DestroyChannels();
 }
 
@@ -2483,22 +2443,18 @@ TEST_F(P2PTransportChannelTest,
 // address of the outermost NAT.
 class P2PTransportChannelSameNatTest : public P2PTransportChannelTestBase {
  protected:
-  void ConfigureEndpoints(const Environment& env,
-                          Config nat_type,
-                          Config config1,
-                          Config config2) {
+  void ConfigureEndpoints(Config nat_type, Config config1, Config config2) {
     RTC_CHECK_GE(nat_type, NAT_FULL_CONE);
     RTC_CHECK_LE(nat_type, NAT_SYMMETRIC);
-    CreatePortAllocators(env);
+    CreatePortAllocators();
     NATSocketServer::Translator* outer_nat =
-        nat()->AddTranslator(env, kPublicAddrs[0], kNatAddrs[0],
+        nat()->AddTranslator(env_, kPublicAddrs[0], kNatAddrs[0],
                              static_cast<NATType>(nat_type - NAT_FULL_CONE));
-    ConfigureEndpoint(env, outer_nat, 0, config1);
-    ConfigureEndpoint(env, outer_nat, 1, config2);
+    ConfigureEndpoint(outer_nat, 0, config1);
+    ConfigureEndpoint(outer_nat, 1, config2);
     set_remote_ice_parameter_source(FROM_SETICEPARAMETERS);
   }
-  void ConfigureEndpoint(const Environment& env,
-                         NATSocketServer::Translator* nat,
+  void ConfigureEndpoint(NATSocketServer::Translator* nat,
                          int endpoint,
                          Config config) {
     RTC_CHECK(config <= NAT_SYMMETRIC);
@@ -2507,7 +2463,7 @@ class P2PTransportChannelSameNatTest : public P2PTransportChannelTestBase {
       nat->AddClient(kPrivateAddrs[endpoint]);
     } else {
       AddAddress(endpoint, kCascadedPrivateAddrs[endpoint]);
-      nat->AddTranslator(env, kPrivateAddrs[endpoint],
+      nat->AddTranslator(env_, kPrivateAddrs[endpoint],
                          kCascadedNatAddrs[endpoint],
                          static_cast<NATType>(config - NAT_FULL_CONE))
           ->AddClient(kCascadedPrivateAddrs[endpoint]);
@@ -2516,11 +2472,10 @@ class P2PTransportChannelSameNatTest : public P2PTransportChannelTestBase {
 };
 
 TEST_F(P2PTransportChannelSameNatTest, TestConesBehindSameCone) {
-  const Environment env = CreateEnvironment();
-  ConfigureEndpoints(env, NAT_FULL_CONE, NAT_FULL_CONE, NAT_FULL_CONE);
-  Test(env, P2PTransportChannelTestBase::Result(IceCandidateType::kPrflx, "udp",
-                                                IceCandidateType::kSrflx, "udp",
-                                                TimeDelta::Seconds(1)));
+  ConfigureEndpoints(NAT_FULL_CONE, NAT_FULL_CONE, NAT_FULL_CONE);
+  Test(P2PTransportChannelTestBase::Result(IceCandidateType::kPrflx, "udp",
+                                           IceCandidateType::kSrflx, "udp",
+                                           TimeDelta::Seconds(1)));
 }
 
 // Test what happens when we have multiple available pathways.
@@ -2594,21 +2549,18 @@ class P2PTransportChannelMultihomedTest : public P2PTransportChannelTest {
 
 // Test that we can establish connectivity when both peers are multihomed.
 TEST_F(P2PTransportChannelMultihomedTest, TestBasic) {
-  const Environment env = CreateEnvironment();
-  CreatePortAllocators(env);
+  CreatePortAllocators();
   AddAddress(0, kPublicAddrs[0]);
   AddAddress(0, kAlternateAddrs[0]);
   AddAddress(1, kPublicAddrs[1]);
   AddAddress(1, kAlternateAddrs[1]);
-  Test(env, kLocalUdpToLocalUdp);
+  Test(kLocalUdpToLocalUdp);
 }
 
 // Test that we can quickly switch links if an interface goes down.
 // The controlled side has two interfaces and one will die.
 TEST_F(P2PTransportChannelMultihomedTest, TestFailoverControlledSide) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-  CreatePortAllocators(env);
+  CreatePortAllocators();
   AddAddress(0, kPublicAddrs[0]);
   // Simulate failing over from Wi-Fi to cell interface.
   AddAddress(1, kPublicAddrs[1], "eth0", ADAPTER_TYPE_WIFI);
@@ -2621,14 +2573,12 @@ TEST_F(P2PTransportChannelMultihomedTest, TestFailoverControlledSide) {
   // Make the receiving timeout shorter for testing.
   IceConfig config = CreateIceConfig(TimeDelta::Seconds(1), GATHER_ONCE);
   // Create channels and let them go writable, as usual.
-  CreateChannels(env, config, config);
+  CreateChannels(config, config);
 
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return CheckCandidatePairAndConnected(ep1_ch1(), ep2_ch1(),
-                                              kPublicAddrs[0], kPublicAddrs[1]);
-      },
-      {.timeout = kMediumTimeout, .clock = &clock}));
+  EXPECT_TRUE(DefaultWait().Until([&] {
+    return CheckCandidatePairAndConnected(ep1_ch1(), ep2_ch1(), kPublicAddrs[0],
+                                          kPublicAddrs[1]);
+  }));
 
   // Blackhole any traffic to or from the public addrs.
   RTC_LOG(LS_INFO) << "Failing over...";
@@ -2636,17 +2586,15 @@ TEST_F(P2PTransportChannelMultihomedTest, TestFailoverControlledSide) {
   // The selected connections may switch, so keep references to them.
   const Connection* selected_connection1 = ep1_ch1()->selected_connection();
   // We should detect loss of receiving within 1 second or so.
-  EXPECT_TRUE(WaitUntil([&] { return !selected_connection1->receiving(); },
-                        {.timeout = kMediumTimeout, .clock = &clock}));
+  EXPECT_TRUE(
+      MediumWait().Until([&] { return !selected_connection1->receiving(); }));
 
   // We should switch over to use the alternate addr on both sides
   // when we are not receiving.
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return ep1_ch1()->selected_connection()->receiving() &&
-               ep2_ch1()->selected_connection()->receiving();
-      },
-      {.timeout = kMediumTimeout, .clock = &clock}));
+  EXPECT_TRUE(MediumWait().Until([&] {
+    return ep1_ch1()->selected_connection()->receiving() &&
+           ep2_ch1()->selected_connection()->receiving();
+  }));
   EXPECT_TRUE(LocalCandidate(ep1_ch1())->address().EqualIPs(kPublicAddrs[0]));
   EXPECT_TRUE(
       RemoteCandidate(ep1_ch1())->address().EqualIPs(kAlternateAddrs[1]));
@@ -2659,9 +2607,7 @@ TEST_F(P2PTransportChannelMultihomedTest, TestFailoverControlledSide) {
 // Test that we can quickly switch links if an interface goes down.
 // The controlling side has two interfaces and one will die.
 TEST_F(P2PTransportChannelMultihomedTest, TestFailoverControllingSide) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-  CreatePortAllocators(env);
+  CreatePortAllocators();
   // Simulate failing over from Wi-Fi to cell interface.
   AddAddress(0, kPublicAddrs[0], "eth0", ADAPTER_TYPE_WIFI);
   AddAddress(0, kAlternateAddrs[0], "wlan0", ADAPTER_TYPE_CELLULAR);
@@ -2674,13 +2620,11 @@ TEST_F(P2PTransportChannelMultihomedTest, TestFailoverControllingSide) {
   // Make the receiving timeout shorter for testing.
   IceConfig config = CreateIceConfig(TimeDelta::Seconds(1), GATHER_ONCE);
   // Create channels and let them go writable, as usual.
-  CreateChannels(env, config, config);
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return CheckCandidatePairAndConnected(ep1_ch1(), ep2_ch1(),
-                                              kPublicAddrs[0], kPublicAddrs[1]);
-      },
-      {.timeout = kMediumTimeout, .clock = &clock}));
+  CreateChannels(config, config);
+  EXPECT_TRUE(MediumWait().Until([&] {
+    return CheckCandidatePairAndConnected(ep1_ch1(), ep2_ch1(), kPublicAddrs[0],
+                                          kPublicAddrs[1]);
+  }));
 
   // Blackhole any traffic to or from the public addrs.
   RTC_LOG(LS_INFO) << "Failing over...";
@@ -2689,12 +2633,10 @@ TEST_F(P2PTransportChannelMultihomedTest, TestFailoverControllingSide) {
   // We should detect loss of receiving within 1 second or so.
   // We should switch over to use the alternate addr on both sides
   // when we are not receiving.
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return CheckCandidatePairAndConnected(
-            ep1_ch1(), ep2_ch1(), kAlternateAddrs[0], kPublicAddrs[1]);
-      },
-      {.timeout = kMediumTimeout, .clock = &clock}));
+  EXPECT_TRUE(MediumWait().Until([&] {
+    return CheckCandidatePairAndConnected(ep1_ch1(), ep2_ch1(),
+                                          kAlternateAddrs[0], kPublicAddrs[1]);
+  }));
 
   DestroyChannels();
 }
@@ -2702,9 +2644,7 @@ TEST_F(P2PTransportChannelMultihomedTest, TestFailoverControllingSide) {
 // Tests that we can quickly switch links if an interface goes down when
 // there are many connections.
 TEST_F(P2PTransportChannelMultihomedTest, TestFailoverWithManyConnections) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-  CreatePortAllocators(env);
+  CreatePortAllocators();
   test_turn_server()->AddInternalSocket(kTurnTcpIntAddr, PROTO_TCP);
   RelayServerConfig turn_server;
   turn_server.credentials = kRelayCredentials;
@@ -2742,13 +2682,11 @@ TEST_F(P2PTransportChannelMultihomedTest, TestFailoverWithManyConnections) {
   // Make the receiving timeout shorter for testing.
   IceConfig config = CreateIceConfig(TimeDelta::Seconds(1), GATHER_CONTINUALLY);
   // Create channels and let them go writable, as usual.
-  CreateChannels(env, config, config, true /* ice_renomination */);
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return CheckCandidatePairAndConnected(ep1_ch1(), ep2_ch1(), wifiIpv6[0],
-                                              wifiIpv6[1]);
-      },
-      {.timeout = kMediumTimeout, .clock = &clock}));
+  CreateChannels(config, config, true /* ice_renomination */);
+  EXPECT_TRUE(MediumWait().Until([&] {
+    return CheckCandidatePairAndConnected(ep1_ch1(), ep2_ch1(), wifiIpv6[0],
+                                          wifiIpv6[1]);
+  }));
 
   // Blackhole any traffic to or from the wifi on endpoint 1.
   RTC_LOG(LS_INFO) << "Failing over...";
@@ -2758,12 +2696,10 @@ TEST_F(P2PTransportChannelMultihomedTest, TestFailoverWithManyConnections) {
   // The selected connections may switch, so keep references to them.
   const Connection* selected_connection1 = ep1_ch1()->selected_connection();
   const Connection* selected_connection2 = ep2_ch1()->selected_connection();
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return !selected_connection1->receiving() &&
-               !selected_connection2->receiving();
-      },
-      {.timeout = kMediumTimeout, .clock = &clock}));
+  EXPECT_TRUE(MediumWait().Until([&] {
+    return !selected_connection1->receiving() &&
+           !selected_connection2->receiving();
+  }));
 
   // Per-network best connections will be pinged at relatively higher rate when
   // the selected connection becomes not receiving.
@@ -2773,9 +2709,9 @@ TEST_F(P2PTransportChannelMultihomedTest, TestFailoverWithManyConnections) {
   Timestamp last_ping_sent1 = per_network_best_connection1->LastPingSent();
   int num_pings_sent1 = per_network_best_connection1->num_pings_sent();
   EXPECT_THAT(
-      WaitUntil([&] { return per_network_best_connection1->num_pings_sent(); },
-                Gt(num_pings_sent1),
-                {.timeout = kMediumTimeout, .clock = &clock}),
+      MediumWait().Until(
+          [&] { return per_network_best_connection1->num_pings_sent(); },
+          Gt(num_pings_sent1)),
       IsRtcOk());
   ASSERT_GT(per_network_best_connection1->num_pings_sent() - num_pings_sent1,
             0);
@@ -2788,12 +2724,10 @@ TEST_F(P2PTransportChannelMultihomedTest, TestFailoverWithManyConnections) {
 
   // It should switch over to use the cellular IPv6 addr on endpoint 1 before
   // it timed out on writing.
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return CheckCandidatePairAndConnected(ep1_ch1(), ep2_ch1(),
-                                              cellularIpv6[0], wifiIpv6[1]);
-      },
-      {.timeout = kMediumTimeout, .clock = &clock}));
+  EXPECT_TRUE(MediumWait().Until([&] {
+    return CheckCandidatePairAndConnected(ep1_ch1(), ep2_ch1(), cellularIpv6[0],
+                                          wifiIpv6[1]);
+  }));
 
   DestroyChannels();
 }
@@ -2802,9 +2736,7 @@ TEST_F(P2PTransportChannelMultihomedTest, TestFailoverWithManyConnections) {
 // the nomination of the selected connection on the controlled side will
 // increase.
 TEST_F(P2PTransportChannelMultihomedTest, TestIceRenomination) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-  CreatePortAllocators(env);
+  CreatePortAllocators();
   // Simulate failing over from Wi-Fi to cell interface.
   AddAddress(0, kPublicAddrs[0], "eth0", ADAPTER_TYPE_WIFI);
   AddAddress(0, kAlternateAddrs[0], "wlan0", ADAPTER_TYPE_CELLULAR);
@@ -2819,15 +2751,13 @@ TEST_F(P2PTransportChannelMultihomedTest, TestIceRenomination) {
   // Make the receiving timeout shorter for testing.
   IceConfig config = CreateIceConfig(TimeDelta::Seconds(1), GATHER_ONCE);
   // Create channels with ICE renomination and let them go writable as usual.
-  CreateChannels(env, config, config, true);
-  ASSERT_TRUE(WaitUntil([&] { return CheckConnected(ep1_ch1(), ep2_ch1()); },
-                        {.timeout = kMediumTimeout, .clock = &clock}));
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return ep2_ch1()->selected_connection()->remote_nomination() > 0 &&
-               ep1_ch1()->selected_connection()->acked_nomination() > 0;
-      },
-      {.timeout = kDefaultTimeout, .clock = &clock}));
+  CreateChannels(config, config, true);
+  ASSERT_TRUE(
+      MediumWait().Until([&] { return CheckConnected(ep1_ch1(), ep2_ch1()); }));
+  EXPECT_TRUE(MediumWait().Until([&] {
+    return ep2_ch1()->selected_connection()->remote_nomination() > 0 &&
+           ep1_ch1()->selected_connection()->acked_nomination() > 0;
+  }));
   const Connection* selected_connection1 = ep1_ch1()->selected_connection();
   Connection* selected_connection2 =
       const_cast<Connection*>(ep2_ch1()->selected_connection());
@@ -2835,7 +2765,7 @@ TEST_F(P2PTransportChannelMultihomedTest, TestIceRenomination) {
   // `selected_connection2` should not be nominated any more since the previous
   // nomination has been acknowledged.
   ConnectSignalNominated(selected_connection2);
-  SIMULATED_WAIT(nominated(), kMediumTimeout.ms(), clock);
+  EXPECT_FALSE(DefaultWait().Until([&] { return nominated(); }));
   EXPECT_FALSE(nominated());
 
   // Blackhole any traffic to or from the public addrs.
@@ -2843,17 +2773,16 @@ TEST_F(P2PTransportChannelMultihomedTest, TestIceRenomination) {
   fw()->AddRule(false, FP_ANY, FD_ANY, kPublicAddrs[0]);
 
   // The selected connection on the controlling side should switch.
-  EXPECT_THAT(WaitUntil([&] { return ep1_ch1()->selected_connection(); },
-                        Ne(selected_connection1),
-                        {.timeout = kMediumTimeout, .clock = &clock}),
-              IsRtcOk());
+  EXPECT_THAT(
+      MediumWait().Until([&] { return ep1_ch1()->selected_connection(); },
+                         Ne(selected_connection1)),
+      IsRtcOk());
   // The connection on the controlled side should be nominated again
   // and have an increased nomination.
   EXPECT_THAT(
-      WaitUntil(
+      MediumWait().Until(
           [&] { return ep2_ch1()->selected_connection()->remote_nomination(); },
-          Gt(remote_nomination2),
-          {.timeout = kDefaultTimeout, .clock = &clock}),
+          Gt(remote_nomination2)),
       IsRtcOk());
 
   DestroyChannels();
@@ -2866,9 +2795,7 @@ TEST_F(P2PTransportChannelMultihomedTest, TestIceRenomination) {
 // TestFailoverControlledSide and TestFailoverControllingSide.
 TEST_F(P2PTransportChannelMultihomedTest,
        TestConnectionSwitchDampeningControlledSide) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-  CreatePortAllocators(env);
+  CreatePortAllocators();
   AddAddress(0, kPublicAddrs[0]);
   // Simulate failing over from Wi-Fi to cell interface.
   AddAddress(1, kPublicAddrs[1], "eth0", ADAPTER_TYPE_WIFI);
@@ -2879,14 +2806,12 @@ TEST_F(P2PTransportChannelMultihomedTest,
   SetAllocatorFlags(1, kOnlyLocalPorts);
 
   // Create channels and let them go writable, as usual.
-  CreateChannels(env);
+  CreateChannels();
 
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return CheckCandidatePairAndConnected(ep1_ch1(), ep2_ch1(),
-                                              kPublicAddrs[0], kPublicAddrs[1]);
-      },
-      {.timeout = kMediumTimeout, .clock = &clock}));
+  EXPECT_TRUE(DefaultWait().Until([&] {
+    return CheckCandidatePairAndConnected(ep1_ch1(), ep2_ch1(), kPublicAddrs[0],
+                                          kPublicAddrs[1]);
+  }));
 
   // Make the receiving timeout shorter for testing.
   IceConfig config = CreateIceConfig(TimeDelta::Seconds(1), GATHER_ONCE);
@@ -2901,20 +2826,18 @@ TEST_F(P2PTransportChannelMultihomedTest,
   // The selected connections may switch, so keep references to them.
   const Connection* selected_connection1 = ep1_ch1()->selected_connection();
   // We should detect loss of receiving within 1 second or so.
-  EXPECT_TRUE(WaitUntil([&] { return !selected_connection1->receiving(); },
-                        {.timeout = kMediumTimeout, .clock = &clock}));
+  EXPECT_TRUE(
+      MediumWait().Until([&] { return !selected_connection1->receiving(); }));
   // After a short while, the link recovers itself.
-  SIMULATED_WAIT(false, 10, clock);
+  time_controller_.AdvanceTime(TimeDelta::Millis(10));
   fw()->ClearRules();
 
   // We should remain on the public address on both sides and no connection
   // switches should have happened.
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return ep1_ch1()->selected_connection()->receiving() &&
-               ep2_ch1()->selected_connection()->receiving();
-      },
-      {.timeout = kMediumTimeout, .clock = &clock}));
+  EXPECT_TRUE(MediumWait().Until([&] {
+    return ep1_ch1()->selected_connection()->receiving() &&
+           ep2_ch1()->selected_connection()->receiving();
+  }));
   EXPECT_TRUE(RemoteCandidate(ep1_ch1())->address().EqualIPs(kPublicAddrs[1]));
   EXPECT_TRUE(LocalCandidate(ep2_ch1())->address().EqualIPs(kPublicAddrs[1]));
   EXPECT_EQ(0, reset_selected_candidate_pair_switches());
@@ -2926,9 +2849,7 @@ TEST_F(P2PTransportChannelMultihomedTest,
 // the selected connection will not switch.
 TEST_F(P2PTransportChannelMultihomedTest,
        TestConnectionSwitchDampeningControllingSide) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-  CreatePortAllocators(env);
+  CreatePortAllocators();
   // Simulate failing over from Wi-Fi to cell interface.
   AddAddress(0, kPublicAddrs[0], "eth0", ADAPTER_TYPE_WIFI);
   AddAddress(0, kAlternateAddrs[0], "wlan0", ADAPTER_TYPE_CELLULAR);
@@ -2939,13 +2860,11 @@ TEST_F(P2PTransportChannelMultihomedTest,
   SetAllocatorFlags(1, kOnlyLocalPorts);
 
   // Create channels and let them go writable, as usual.
-  CreateChannels(env);
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return CheckCandidatePairAndConnected(ep1_ch1(), ep2_ch1(),
-                                              kPublicAddrs[0], kPublicAddrs[1]);
-      },
-      {.timeout = kMediumTimeout, .clock = &clock}));
+  CreateChannels();
+  EXPECT_TRUE(MediumWait().Until([&] {
+    return CheckCandidatePairAndConnected(ep1_ch1(), ep2_ch1(), kPublicAddrs[0],
+                                          kPublicAddrs[1]);
+  }));
 
   // Make the receiving timeout shorter for testing.
   IceConfig config = CreateIceConfig(TimeDelta::Seconds(1), GATHER_ONCE);
@@ -2959,20 +2878,18 @@ TEST_F(P2PTransportChannelMultihomedTest,
   // The selected connections may switch, so keep references to them.
   const Connection* selected_connection1 = ep1_ch1()->selected_connection();
   // We should detect loss of receiving within 1 second or so.
-  EXPECT_TRUE(WaitUntil([&] { return !selected_connection1->receiving(); },
-                        {.timeout = kMediumTimeout, .clock = &clock}));
+  EXPECT_TRUE(
+      MediumWait().Until([&] { return !selected_connection1->receiving(); }));
   // The link recovers after a short while.
-  SIMULATED_WAIT(false, 10, clock);
+  time_controller_.AdvanceTime(TimeDelta::Millis(10));
   fw()->ClearRules();
 
   // We should not switch to the alternate addr on both sides because of the
   // dampening.
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return CheckCandidatePairAndConnected(ep1_ch1(), ep2_ch1(),
-                                              kPublicAddrs[0], kPublicAddrs[1]);
-      },
-      {.timeout = kMediumTimeout, .clock = &clock}));
+  EXPECT_TRUE(MediumWait().Until([&] {
+    return CheckCandidatePairAndConnected(ep1_ch1(), ep2_ch1(), kPublicAddrs[0],
+                                          kPublicAddrs[1]);
+  }));
   EXPECT_EQ(0, reset_selected_candidate_pair_switches());
   DestroyChannels();
 }
@@ -2980,9 +2897,7 @@ TEST_F(P2PTransportChannelMultihomedTest,
 // Tests that if the remote side's network failed, it won't cause the local
 // side to switch connections and networks.
 TEST_F(P2PTransportChannelMultihomedTest, TestRemoteFailover) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-  CreatePortAllocators(env);
+  CreatePortAllocators();
   // The interface names are chosen so that `cellular` would have higher
   // candidate priority and higher cost.
   auto& wifi = kPublicAddrs;
@@ -2995,7 +2910,7 @@ TEST_F(P2PTransportChannelMultihomedTest, TestRemoteFailover) {
   SetAllocatorFlags(0, kOnlyLocalPorts);
   SetAllocatorFlags(1, kOnlyLocalPorts);
   // Create channels and let them go writable, as usual.
-  CreateChannels(env);
+  CreateChannels();
   // Make the receiving timeout shorter for testing.
   // Set the backup connection ping interval to 25s.
   IceConfig config = CreateIceConfig(TimeDelta::Seconds(1), GATHER_ONCE,
@@ -3005,36 +2920,33 @@ TEST_F(P2PTransportChannelMultihomedTest, TestRemoteFailover) {
   ep1_ch1()->SetIceConfig(config);
   ep2_ch1()->SetIceConfig(config);
   // Need to wait to make sure the connections on both networks are writable.
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return CheckCandidatePairAndConnected(ep1_ch1(), ep2_ch1(), wifi[0],
-                                              wifi[1]);
-      },
-      {.timeout = kDefaultTimeout, .clock = &clock}));
+  EXPECT_TRUE(MediumWait().Until([&] {
+    return CheckCandidatePairAndConnected(ep1_ch1(), ep2_ch1(), wifi[0],
+                                          wifi[1]);
+  }));
   Connection* backup_conn =
       GetConnectionWithLocalAddress(ep1_ch1(), cellular[0]);
   ASSERT_NE(nullptr, backup_conn);
   // After a short while, the backup connection will be writable but not
   // receiving because backup connection is pinged at a slower rate.
-  EXPECT_TRUE(WaitUntil(
-      [&] { return backup_conn->writable() && !backup_conn->receiving(); },
-      {.timeout = kDefaultTimeout, .clock = &clock}));
+  EXPECT_TRUE(DefaultWait().Until(
+      [&] { return backup_conn->writable() && !backup_conn->receiving(); }));
   reset_selected_candidate_pair_switches();
   // Blackhole any traffic to or from the remote WiFi networks.
   RTC_LOG(LS_INFO) << "Failing over...";
   fw()->AddRule(false, FP_ANY, FD_ANY, wifi[1]);
 
   int num_switches = 0;
-  SIMULATED_WAIT((num_switches = reset_selected_candidate_pair_switches()) > 0,
-                 20000, clock);
+  (void)DefaultWait().Until([&] {
+    return (num_switches = reset_selected_candidate_pair_switches()) > 0;
+  });
   EXPECT_EQ(0, num_switches);
   DestroyChannels();
 }
 
 // Tests that a Wifi-Wifi connection has the highest precedence.
 TEST_F(P2PTransportChannelMultihomedTest, TestPreferWifiToWifiConnection) {
-  const Environment env = CreateEnvironment();
-  CreatePortAllocators(env);
+  CreatePortAllocators();
   // The interface names are chosen so that `cellular` would have higher
   // candidate priority if it is not for the network type.
   auto& wifi = kAlternateAddrs;
@@ -3049,12 +2961,12 @@ TEST_F(P2PTransportChannelMultihomedTest, TestPreferWifiToWifiConnection) {
   SetAllocatorFlags(1, kOnlyLocalPorts);
 
   // Create channels and let them go writable, as usual.
-  CreateChannels(env);
+  CreateChannels();
 
-  EXPECT_TRUE(
-      WaitUntil([&]() { return CheckConnected(ep1_ch1(), ep2_ch1()); }));
+  EXPECT_TRUE(DefaultWait().Until(
+      [&]() { return CheckConnected(ep1_ch1(), ep2_ch1()); }));
   // Need to wait to make sure the connections on both networks are writable.
-  EXPECT_TRUE(WaitUntil([&] {
+  EXPECT_TRUE(DefaultWait().Until([&] {
     return CheckCandidatePairAndConnected(ep1_ch1(), ep2_ch1(), wifi[0],
                                           wifi[1]);
   }));
@@ -3064,8 +2976,7 @@ TEST_F(P2PTransportChannelMultihomedTest, TestPreferWifiToWifiConnection) {
 // Tests that a Wifi-Cellular connection has higher precedence than
 // a Cellular-Cellular connection.
 TEST_F(P2PTransportChannelMultihomedTest, TestPreferWifiOverCellularNetwork) {
-  const Environment env = CreateEnvironment();
-  CreatePortAllocators(env);
+  CreatePortAllocators();
   // The interface names are chosen so that `cellular` would have higher
   // candidate priority if it is not for the network type.
   auto& wifi = kAlternateAddrs;
@@ -3079,9 +2990,9 @@ TEST_F(P2PTransportChannelMultihomedTest, TestPreferWifiOverCellularNetwork) {
   SetAllocatorFlags(1, kOnlyLocalPorts);
 
   // Create channels and let them go writable, as usual.
-  CreateChannels(env);
+  CreateChannels();
 
-  EXPECT_TRUE(WaitUntil([&]() {
+  EXPECT_TRUE(DefaultWait().Until([&]() {
     return CheckCandidatePairAndConnected(ep1_ch1(), ep2_ch1(), cellular[0],
                                           wifi[1]);
   }));
@@ -3091,8 +3002,7 @@ TEST_F(P2PTransportChannelMultihomedTest, TestPreferWifiOverCellularNetwork) {
 // Test that the backup connection is pinged at a rate no faster than
 // what was configured.
 TEST_F(P2PTransportChannelMultihomedTest, TestPingBackupConnectionRate) {
-  const Environment env = CreateEnvironment();
-  CreatePortAllocators(env);
+  CreatePortAllocators();
   AddAddress(0, kPublicAddrs[0]);
   // Adding alternate address will make sure `kPublicAddrs` has the higher
   // priority than others. This is due to FakeNetwork::AddInterface method.
@@ -3104,25 +3014,25 @@ TEST_F(P2PTransportChannelMultihomedTest, TestPingBackupConnectionRate) {
   SetAllocatorFlags(1, kOnlyLocalPorts);
 
   // Create channels and let them go writable, as usual.
-  CreateChannels(env);
-  EXPECT_TRUE(WaitUntil([&] { return CheckConnected(ep1_ch1(), ep2_ch1()); }));
+  CreateChannels();
+  EXPECT_TRUE(DefaultWait().Until(
+      [&] { return CheckConnected(ep1_ch1(), ep2_ch1()); }));
   TimeDelta backup_ping_interval = TimeDelta::Seconds(2);
   ep2_ch1()->SetIceConfig(CreateIceConfig(TimeDelta::Seconds(2), GATHER_ONCE,
                                           backup_ping_interval));
   // After the state becomes COMPLETED, the backup connection will be pinged
   // once every `backup_ping_interval` milliseconds.
-  ASSERT_THAT(WaitUntil([&] { return ep2_ch1()->GetState(); },
-                        Eq(IceTransportStateInternal::STATE_COMPLETED)),
-              IsRtcOk());
+  ASSERT_TRUE(DefaultWait().Until([&] {
+    return ep2_ch1()->GetState() == IceTransportStateInternal::STATE_COMPLETED;
+  }));
   auto connections = ep2_ch1()->connections();
   ASSERT_EQ(2U, connections.size());
   Connection* backup_conn = GetBackupConnection(ep2_ch1());
-  EXPECT_TRUE(WaitUntil([&] { return backup_conn->writable(); },
-                        {.timeout = kMediumTimeout}));
+  EXPECT_TRUE(DefaultWait().Until([&] { return backup_conn->writable(); }));
   Timestamp last_ping_response = backup_conn->LastPingResponseReceived();
-  EXPECT_THAT(WaitUntil([&] { return backup_conn->LastPingResponseReceived(); },
-                        Gt(last_ping_response), {.timeout = kDefaultTimeout}),
-              IsRtcOk());
+  EXPECT_TRUE(MediumWait().Until([&] {
+    return backup_conn->LastPingResponseReceived() > last_ping_response;
+  }));
   TimeDelta time_elapsed =
       backup_conn->LastPingResponseReceived() - last_ping_response;
   RTC_LOG(LS_INFO) << "Time elapsed: " << time_elapsed;
@@ -3134,8 +3044,7 @@ TEST_F(P2PTransportChannelMultihomedTest, TestPingBackupConnectionRate) {
 // Test that the connection is pinged at a rate no faster than
 // what was configured when stable and writable.
 TEST_F(P2PTransportChannelMultihomedTest, TestStableWritableRate) {
-  const Environment env = CreateEnvironment();
-  CreatePortAllocators(env);
+  CreatePortAllocators();
   AddAddress(0, kPublicAddrs[0]);
   // Adding alternate address will make sure `kPublicAddrs` has the higher
   // priority than others. This is due to FakeNetwork::AddInterface method.
@@ -3147,8 +3056,9 @@ TEST_F(P2PTransportChannelMultihomedTest, TestStableWritableRate) {
   SetAllocatorFlags(1, kOnlyLocalPorts);
 
   // Create channels and let them go writable, as usual.
-  CreateChannels(env);
-  EXPECT_TRUE(WaitUntil([&] { return CheckConnected(ep1_ch1(), ep2_ch1()); }));
+  CreateChannels();
+  EXPECT_TRUE(DefaultWait().Until(
+      [&] { return CheckConnected(ep1_ch1(), ep2_ch1()); }));
   // Set a value larger than the default value of 2500 ms
   TimeDelta ping_interval = TimeDelta::Millis(3'456);
   IceConfig config = CreateIceConfig(2 * ping_interval, GATHER_ONCE);
@@ -3156,22 +3066,20 @@ TEST_F(P2PTransportChannelMultihomedTest, TestStableWritableRate) {
   ep2_ch1()->SetIceConfig(config);
   // After the state becomes COMPLETED and is stable and writable, the
   // connection will be pinged once every `ping_interval`.
-  ASSERT_THAT(WaitUntil([&] { return ep2_ch1()->GetState(); },
-                        Eq(IceTransportStateInternal::STATE_COMPLETED)),
-              IsRtcOk());
+  ASSERT_TRUE(DefaultWait().Until([&] {
+    return ep2_ch1()->GetState() == IceTransportStateInternal::STATE_COMPLETED;
+  }));
   auto connections = ep2_ch1()->connections();
   ASSERT_EQ(2U, connections.size());
   Connection* conn = GetBestConnection(ep2_ch1());
-  EXPECT_TRUE(
-      WaitUntil([&] { return conn->writable(); }, {.timeout = kMediumTimeout}));
+  EXPECT_TRUE(DefaultWait().Until([&] { return conn->writable(); }));
 
   Timestamp last_ping_response = Timestamp::Zero();
   // Burn through some pings so the connection is stable.
   for (int i = 0; i < 5; i++) {
     last_ping_response = conn->LastPingResponseReceived();
-    EXPECT_THAT(WaitUntil([&] { return conn->LastPingResponseReceived(); },
-                          Gt(last_ping_response), {.timeout = kDefaultTimeout}),
-                IsRtcOk());
+    EXPECT_TRUE(DefaultWait().Until(
+        [&] { return conn->LastPingResponseReceived() > last_ping_response; }));
   }
   EXPECT_TRUE(conn->stable(last_ping_response)) << "Connection not stable";
   TimeDelta time_elapsed =
@@ -3183,23 +3091,20 @@ TEST_F(P2PTransportChannelMultihomedTest, TestStableWritableRate) {
 }
 
 TEST_F(P2PTransportChannelMultihomedTest, TestGetState) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-  CreatePortAllocators(env);
+  CreatePortAllocators();
   AddAddress(0, kAlternateAddrs[0]);
   AddAddress(0, kPublicAddrs[0]);
   AddAddress(1, kPublicAddrs[1]);
   // Create channels and let them go writable, as usual.
-  CreateChannels(env);
+  CreateChannels();
 
   // Both transport channels will reach STATE_COMPLETED quickly.
-  EXPECT_THAT(WaitUntil([&] { return ep1_ch1()->GetState(); },
-                        Eq(IceTransportStateInternal::STATE_COMPLETED),
-                        {.timeout = kShortTimeout, .clock = &clock}),
-              IsRtcOk());
-  EXPECT_THAT(WaitUntil([&] { return ep2_ch1()->GetState(); },
-                        Eq(IceTransportStateInternal::STATE_COMPLETED),
-                        {.timeout = kShortTimeout, .clock = &clock}),
+  EXPECT_THAT(
+      DefaultWait().Until([&] { return ep1_ch1()->GetState(); },
+                          Eq(IceTransportStateInternal::STATE_COMPLETED)),
+      IsRtcOk());
+  EXPECT_THAT(ShortWait().Until([&] { return ep2_ch1()->GetState(); },
+                                Eq(IceTransportStateInternal::STATE_COMPLETED)),
               IsRtcOk());
   DestroyChannels();
 }
@@ -3209,21 +3114,19 @@ TEST_F(P2PTransportChannelMultihomedTest, TestGetState) {
 // will be removed from the port list of the channel, and the respective
 // remote candidates on the other participant will be removed eventually.
 TEST_F(P2PTransportChannelMultihomedTest, TestNetworkBecomesInactive) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-  CreatePortAllocators(env);
+  CreatePortAllocators();
   AddAddress(0, kPublicAddrs[0]);
   AddAddress(1, kPublicAddrs[1]);
   // Create channels and let them go writable, as usual.
   IceConfig ep1_config =
       CreateIceConfig(TimeDelta::Seconds(2), GATHER_CONTINUALLY);
   IceConfig ep2_config = CreateIceConfig(TimeDelta::Seconds(2), GATHER_ONCE);
-  CreateChannels(env, ep1_config, ep2_config);
+  CreateChannels(ep1_config, ep2_config);
 
   SetAllocatorFlags(0, kOnlyLocalPorts);
   SetAllocatorFlags(1, kOnlyLocalPorts);
-  ASSERT_TRUE(WaitUntil([&] { return CheckConnected(ep1_ch1(), ep2_ch1()); },
-                        {.timeout = kDefaultTimeout, .clock = &clock}));
+  ASSERT_TRUE(
+      ShortWait().Until([&] { return CheckConnected(ep1_ch1(), ep2_ch1()); }));
   // More than one port has been created.
   EXPECT_LE(1U, ep1_ch1()->ports().size());
   // Endpoint 1 enabled continual gathering; the port will be removed
@@ -3231,8 +3134,8 @@ TEST_F(P2PTransportChannelMultihomedTest, TestNetworkBecomesInactive) {
   RemoveAddress(0, kPublicAddrs[0]);
   EXPECT_TRUE(ep1_ch1()->ports().empty());
   // The remote candidates will be removed eventually.
-  EXPECT_TRUE(WaitUntil([&] { return ep2_ch1()->remote_candidates().empty(); },
-                        {.clock = &clock}));
+  EXPECT_TRUE(DefaultWait().Until(
+      [&] { return ep2_ch1()->remote_candidates().empty(); }));
 
   size_t num_ports = ep2_ch1()->ports().size();
   EXPECT_LE(1U, num_ports);
@@ -3242,10 +3145,11 @@ TEST_F(P2PTransportChannelMultihomedTest, TestNetworkBecomesInactive) {
   // other participant will not be removed.
   RemoveAddress(1, kPublicAddrs[1]);
 
-  EXPECT_THAT(WaitUntil([&] { return ep2_ch1()->ports().size(); }, Eq(0U),
-                        {.timeout = kDefaultTimeout, .clock = &clock}),
-              IsRtcOk());
-  SIMULATED_WAIT(ep1_ch1()->remote_candidates().empty(), 500, clock);
+  EXPECT_THAT(
+      DefaultWait().Until([&] { return ep2_ch1()->ports().size(); }, Eq(0U)),
+      IsRtcOk());
+  (void)DefaultWait().Until(
+      [&] { return ep1_ch1()->remote_candidates().empty(); });
   EXPECT_EQ(num_remote_candidates, ep1_ch1()->remote_candidates().size());
 
   DestroyChannels();
@@ -3255,8 +3159,7 @@ TEST_F(P2PTransportChannelMultihomedTest, TestNetworkBecomesInactive) {
 // interface is added.
 TEST_F(P2PTransportChannelMultihomedTest,
        TestContinualGatheringOnNewInterface) {
-  const Environment env = CreateEnvironment();
-  CreatePortAllocators(env);
+  CreatePortAllocators();
   auto& wifi = kAlternateAddrs;
   auto& cellular = kPublicAddrs;
   AddAddress(0, wifi[0], "test_wifi0", ADAPTER_TYPE_WIFI);
@@ -3264,58 +3167,50 @@ TEST_F(P2PTransportChannelMultihomedTest,
   // Set continual gathering policy.
   IceConfig continual_gathering_config =
       CreateIceConfig(TimeDelta::Seconds(1), GATHER_CONTINUALLY);
-  CreateChannels(env, continual_gathering_config, continual_gathering_config);
+  CreateChannels(continual_gathering_config, continual_gathering_config);
   SetAllocatorFlags(0, kOnlyLocalPorts);
   SetAllocatorFlags(1, kOnlyLocalPorts);
-  EXPECT_TRUE(WaitUntil([&] { return CheckConnected(ep1_ch1(), ep2_ch1()); },
-                        {.timeout = kDefaultTimeout}));
+  EXPECT_TRUE(DefaultWait().Until(
+      [&] { return CheckConnected(ep1_ch1(), ep2_ch1()); }));
 
   // Add a new wifi interface on end point 2. We should expect a new connection
   // to be created and the new one will be the best connection.
   AddAddress(1, wifi[1], "test_wifi1", ADAPTER_TYPE_WIFI);
   const Connection* conn;
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return (conn = ep1_ch1()->selected_connection()) != nullptr &&
-               HasRemoteAddress(conn, wifi[1]);
-      },
-      {.timeout = kDefaultTimeout}));
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return (conn = ep2_ch1()->selected_connection()) != nullptr &&
-               HasLocalAddress(conn, wifi[1]);
-      },
-      {.timeout = kDefaultTimeout}));
+  EXPECT_TRUE(DefaultWait().Until([&] {
+    return (conn = ep1_ch1()->selected_connection()) != nullptr &&
+           HasRemoteAddress(conn, wifi[1]);
+  }));
+  EXPECT_TRUE(DefaultWait().Until([&] {
+    return (conn = ep2_ch1()->selected_connection()) != nullptr &&
+           HasLocalAddress(conn, wifi[1]);
+  }));
 
   // Add a new cellular interface on end point 1, we should expect a new
   // backup connection created using this new interface.
   AddAddress(0, cellular[0], "test_cellular0", ADAPTER_TYPE_CELLULAR);
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return ep1_ch1()->GetState() ==
-                   IceTransportStateInternal::STATE_COMPLETED &&
-               absl::c_any_of(ep1_ch1()->connections(),
-                              [channel = ep1_ch1(),
-                               address = cellular[0]](const Connection* conn) {
-                                return HasLocalAddress(conn, address) &&
-                                       conn != channel->selected_connection() &&
-                                       conn->writable();
-                              });
-      },
-      {.timeout = kDefaultTimeout}));
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return ep2_ch1()->GetState() ==
-                   IceTransportStateInternal::STATE_COMPLETED &&
-               absl::c_any_of(ep2_ch1()->connections(),
-                              [channel = ep2_ch1(),
-                               address = cellular[0]](const Connection* conn) {
-                                return HasRemoteAddress(conn, address) &&
-                                       conn != channel->selected_connection() &&
-                                       conn->receiving();
-                              });
-      },
-      {.timeout = kDefaultTimeout}));
+  EXPECT_TRUE(DefaultWait().Until([&] {
+    return ep1_ch1()->GetState() ==
+               IceTransportStateInternal::STATE_COMPLETED &&
+           absl::c_any_of(ep1_ch1()->connections(),
+                          [channel = ep1_ch1(),
+                           address = cellular[0]](const Connection* conn) {
+                            return HasLocalAddress(conn, address) &&
+                                   conn != channel->selected_connection() &&
+                                   conn->writable();
+                          });
+  }));
+  EXPECT_TRUE(DefaultWait().Until([&] {
+    return ep2_ch1()->GetState() ==
+               IceTransportStateInternal::STATE_COMPLETED &&
+           absl::c_any_of(ep2_ch1()->connections(),
+                          [channel = ep2_ch1(),
+                           address = cellular[0]](const Connection* conn) {
+                            return HasRemoteAddress(conn, address) &&
+                                   conn != channel->selected_connection() &&
+                                   conn->receiving();
+                          });
+  }));
 
   DestroyChannels();
 }
@@ -3323,9 +3218,7 @@ TEST_F(P2PTransportChannelMultihomedTest,
 // Tests that we can switch links via continual gathering.
 TEST_F(P2PTransportChannelMultihomedTest,
        TestSwitchLinksViaContinualGathering) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-  CreatePortAllocators(env);
+  CreatePortAllocators();
   AddAddress(0, kPublicAddrs[0]);
   AddAddress(1, kPublicAddrs[1]);
   // Use only local ports for simplicity.
@@ -3336,45 +3229,37 @@ TEST_F(P2PTransportChannelMultihomedTest,
   IceConfig continual_gathering_config =
       CreateIceConfig(TimeDelta::Seconds(1), GATHER_CONTINUALLY);
   // Create channels and let them go writable, as usual.
-  CreateChannels(env, continual_gathering_config, continual_gathering_config);
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return CheckCandidatePairAndConnected(ep1_ch1(), ep2_ch1(),
-                                              kPublicAddrs[0], kPublicAddrs[1]);
-      },
-      {.timeout = kMediumTimeout, .clock = &clock}));
+  CreateChannels(continual_gathering_config, continual_gathering_config);
+  EXPECT_TRUE(DefaultWait().Until([&] {
+    return CheckCandidatePairAndConnected(ep1_ch1(), ep2_ch1(), kPublicAddrs[0],
+                                          kPublicAddrs[1]);
+  }));
 
   // Add the new address first and then remove the other one.
   RTC_LOG(LS_INFO) << "Draining...";
   AddAddress(1, kAlternateAddrs[1]);
   RemoveAddress(1, kPublicAddrs[1]);
   // We should switch to use the alternate address after an exchange of pings.
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return CheckCandidatePairAndConnected(
-            ep1_ch1(), ep2_ch1(), kPublicAddrs[0], kAlternateAddrs[1]);
-      },
-      {.timeout = kMediumTimeout, .clock = &clock}));
+  EXPECT_TRUE(MediumWait().Until([&] {
+    return CheckCandidatePairAndConnected(ep1_ch1(), ep2_ch1(), kPublicAddrs[0],
+                                          kAlternateAddrs[1]);
+  }));
 
   // Remove one address first and then add another address.
   RTC_LOG(LS_INFO) << "Draining again...";
   RemoveAddress(1, kAlternateAddrs[1]);
   AddAddress(1, kAlternateAddrs[0]);
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return CheckCandidatePairAndConnected(
-            ep1_ch1(), ep2_ch1(), kPublicAddrs[0], kAlternateAddrs[0]);
-      },
-      {.timeout = kMediumTimeout, .clock = &clock}));
+  EXPECT_TRUE(MediumWait().Until([&] {
+    return CheckCandidatePairAndConnected(ep1_ch1(), ep2_ch1(), kPublicAddrs[0],
+                                          kAlternateAddrs[0]);
+  }));
 
   DestroyChannels();
 }
 
 // Tests that the backup connection will be restored after it is destroyed.
 TEST_F(P2PTransportChannelMultihomedTest, TestRestoreBackupConnection) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-  CreatePortAllocators(env);
+  CreatePortAllocators();
   auto& wifi = kAlternateAddrs;
   auto& cellular = kPublicAddrs;
   AddAddress(0, wifi[0], "test_wifi0", ADAPTER_TYPE_WIFI);
@@ -3388,57 +3273,46 @@ TEST_F(P2PTransportChannelMultihomedTest, TestRestoreBackupConnection) {
   // Create channels and let them go writable, as usual.
   IceConfig config = CreateIceConfig(TimeDelta::Seconds(1), GATHER_CONTINUALLY);
   config.regather_on_failed_networks_interval = TimeDelta::Seconds(2);
-  CreateChannels(env, config, config);
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return CheckCandidatePairAndConnected(ep1_ch1(), ep2_ch1(), wifi[0],
-                                              wifi[1]);
-      },
-      {.timeout = kMediumTimeout, .clock = &clock}));
+  CreateChannels(config, config);
+  EXPECT_TRUE(MediumWait().Until([&] {
+    return CheckCandidatePairAndConnected(ep1_ch1(), ep2_ch1(), wifi[0],
+                                          wifi[1]);
+  }));
 
   // Destroy all backup connections.
   DestroyAllButBestConnection(ep1_ch1());
   // Ensure the backup connection is removed first.
   EXPECT_THAT(
-      WaitUntil(
+      MediumWait().Until(
           [&] { return GetConnectionWithLocalAddress(ep1_ch1(), cellular[0]); },
-          Eq(nullptr), {.timeout = kDefaultTimeout, .clock = &clock}),
+          Eq(nullptr)),
       IsRtcOk());
   const Connection* conn;
-  EXPECT_TRUE(
-      WaitUntil(
-          [&] {
-            return (conn = GetConnectionWithLocalAddress(
-                        ep1_ch1(), cellular[0])) != nullptr &&
-                   conn != ep1_ch1()->selected_connection() && conn->writable();
-          },
-          {.timeout = kDefaultTimeout, .clock = &clock}));
+  EXPECT_TRUE(DefaultWait().Until([&] {
+    return (conn = GetConnectionWithLocalAddress(ep1_ch1(), cellular[0])) !=
+               nullptr &&
+           conn != ep1_ch1()->selected_connection() && conn->writable();
+  }));
 
   DestroyChannels();
 }
 
 TEST_F(P2PTransportChannelMultihomedTest, TestVpnDefault) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-  CreatePortAllocators(env);
+  CreatePortAllocators();
   AddAddress(0, kPublicAddrs[0], "eth0", ADAPTER_TYPE_ETHERNET);
   AddAddress(0, kAlternateAddrs[0], "vpn0", ADAPTER_TYPE_VPN);
   AddAddress(1, kPublicAddrs[1]);
 
   IceConfig config;
-  CreateChannels(env, config, config, false);
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return CheckConnected(ep1_ch1(), ep2_ch1()) &&
-               !ep1_ch1()->selected_connection()->network()->IsVpn();
-      },
-      {.timeout = kDefaultTimeout, .clock = &clock}));
+  CreateChannels(config, config, false);
+  EXPECT_TRUE(DefaultWait().Until([&] {
+    return CheckConnected(ep1_ch1(), ep2_ch1()) &&
+           !ep1_ch1()->selected_connection()->network()->IsVpn();
+  }));
 }
 
 TEST_F(P2PTransportChannelMultihomedTest, TestVpnPreferVpn) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-  CreatePortAllocators(env);
+  CreatePortAllocators();
   AddAddress(0, kPublicAddrs[0], "eth0", ADAPTER_TYPE_ETHERNET);
   AddAddress(0, kAlternateAddrs[0], "vpn0", ADAPTER_TYPE_VPN,
              ADAPTER_TYPE_CELLULAR);
@@ -3447,30 +3321,24 @@ TEST_F(P2PTransportChannelMultihomedTest, TestVpnPreferVpn) {
   IceConfig config;
   config.vpn_preference = VpnPreference::kPreferVpn;
   RTC_LOG(LS_INFO) << "KESO: config.vpn_preference: " << config.vpn_preference;
-  CreateChannels(env, config, config, false);
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return CheckConnected(ep1_ch1(), ep2_ch1()) &&
-               ep1_ch1()->selected_connection()->network()->IsVpn();
-      },
-      {.timeout = kDefaultTimeout, .clock = &clock}));
+  CreateChannels(config, config, false);
+  EXPECT_TRUE(DefaultWait().Until([&] {
+    return CheckConnected(ep1_ch1(), ep2_ch1()) &&
+           ep1_ch1()->selected_connection()->network()->IsVpn();
+  }));
 
   // Block VPN.
   fw()->AddRule(false, FP_ANY, FD_ANY, kAlternateAddrs[0]);
 
   // Check that it switches to non-VPN
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return CheckConnected(ep1_ch1(), ep2_ch1()) &&
-               !ep1_ch1()->selected_connection()->network()->IsVpn();
-      },
-      {.timeout = kDefaultTimeout, .clock = &clock}));
+  EXPECT_TRUE(DefaultWait().Until([&] {
+    return CheckConnected(ep1_ch1(), ep2_ch1()) &&
+           !ep1_ch1()->selected_connection()->network()->IsVpn();
+  }));
 }
 
 TEST_F(P2PTransportChannelMultihomedTest, TestVpnAvoidVpn) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-  CreatePortAllocators(env);
+  CreatePortAllocators();
   AddAddress(0, kPublicAddrs[0], "eth0", ADAPTER_TYPE_CELLULAR);
   AddAddress(0, kAlternateAddrs[0], "vpn0", ADAPTER_TYPE_VPN,
              ADAPTER_TYPE_ETHERNET);
@@ -3478,30 +3346,24 @@ TEST_F(P2PTransportChannelMultihomedTest, TestVpnAvoidVpn) {
 
   IceConfig config;
   config.vpn_preference = VpnPreference::kAvoidVpn;
-  CreateChannels(env, config, config, false);
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return CheckConnected(ep1_ch1(), ep2_ch1()) &&
-               !ep1_ch1()->selected_connection()->network()->IsVpn();
-      },
-      {.timeout = kDefaultTimeout, .clock = &clock}));
+  CreateChannels(config, config, false);
+  EXPECT_TRUE(DefaultWait().Until([&] {
+    return CheckConnected(ep1_ch1(), ep2_ch1()) &&
+           !ep1_ch1()->selected_connection()->network()->IsVpn();
+  }));
 
   // Block non-VPN.
   fw()->AddRule(false, FP_ANY, FD_ANY, kPublicAddrs[0]);
 
   // Check that it switches to VPN
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return CheckConnected(ep1_ch1(), ep2_ch1()) &&
-               ep1_ch1()->selected_connection()->network()->IsVpn();
-      },
-      {.timeout = kDefaultTimeout, .clock = &clock}));
+  EXPECT_TRUE(DefaultWait().Until([&] {
+    return CheckConnected(ep1_ch1(), ep2_ch1()) &&
+           ep1_ch1()->selected_connection()->network()->IsVpn();
+  }));
 }
 
 TEST_F(P2PTransportChannelMultihomedTest, TestVpnNeverVpn) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-  CreatePortAllocators(env);
+  CreatePortAllocators();
   AddAddress(0, kPublicAddrs[0], "eth0", ADAPTER_TYPE_CELLULAR);
   AddAddress(0, kAlternateAddrs[0], "vpn0", ADAPTER_TYPE_VPN,
              ADAPTER_TYPE_ETHERNET);
@@ -3509,27 +3371,23 @@ TEST_F(P2PTransportChannelMultihomedTest, TestVpnNeverVpn) {
 
   IceConfig config;
   config.vpn_preference = VpnPreference::kNeverUseVpn;
-  CreateChannels(env, config, config, false);
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return CheckConnected(ep1_ch1(), ep2_ch1()) &&
-               !ep1_ch1()->selected_connection()->network()->IsVpn();
-      },
-      {.timeout = kDefaultTimeout, .clock = &clock}));
+  CreateChannels(config, config, false);
+  EXPECT_TRUE(DefaultWait().Until([&] {
+    return CheckConnected(ep1_ch1(), ep2_ch1()) &&
+           !ep1_ch1()->selected_connection()->network()->IsVpn();
+  }));
 
   // Block non-VPN.
   fw()->AddRule(false, FP_ANY, FD_ANY, kPublicAddrs[0]);
 
   // Check that it does not switches to VPN
-  clock.AdvanceTime(kDefaultTimeout);
-  EXPECT_TRUE(WaitUntil([&] { return !CheckConnected(ep1_ch1(), ep2_ch1()); },
-                        {.timeout = kDefaultTimeout, .clock = &clock}));
+  time_controller_.AdvanceTime(kDefaultTimeout);
+  EXPECT_TRUE(DefaultWait().Until(
+      [&] { return !CheckConnected(ep1_ch1(), ep2_ch1()); }));
 }
 
 TEST_F(P2PTransportChannelMultihomedTest, TestVpnOnlyVpn) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-  CreatePortAllocators(env);
+  CreatePortAllocators();
   AddAddress(0, kPublicAddrs[0], "eth0", ADAPTER_TYPE_CELLULAR);
   AddAddress(0, kAlternateAddrs[0], "vpn0", ADAPTER_TYPE_VPN,
              ADAPTER_TYPE_ETHERNET);
@@ -3537,34 +3395,30 @@ TEST_F(P2PTransportChannelMultihomedTest, TestVpnOnlyVpn) {
 
   IceConfig config;
   config.vpn_preference = VpnPreference::kOnlyUseVpn;
-  CreateChannels(env, config, config, false);
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return CheckConnected(ep1_ch1(), ep2_ch1()) &&
-               ep1_ch1()->selected_connection()->network()->IsVpn();
-      },
-      {.timeout = kDefaultTimeout, .clock = &clock}));
+  CreateChannels(config, config, false);
+  EXPECT_TRUE(DefaultWait().Until([&] {
+    return CheckConnected(ep1_ch1(), ep2_ch1()) &&
+           ep1_ch1()->selected_connection()->network()->IsVpn();
+  }));
 
   // Block VPN.
   fw()->AddRule(false, FP_ANY, FD_ANY, kAlternateAddrs[0]);
 
   // Check that it does not switch to non-VPN
-  clock.AdvanceTime(kDefaultTimeout);
-  EXPECT_TRUE(WaitUntil([&] { return !CheckConnected(ep1_ch1(), ep2_ch1()); },
-                        {.timeout = kDefaultTimeout, .clock = &clock}));
+  time_controller_.AdvanceTime(kDefaultTimeout);
+  EXPECT_TRUE(DefaultWait().Until(
+      [&] { return !CheckConnected(ep1_ch1(), ep2_ch1()); }));
 }
 
 TEST_F(P2PTransportChannelMultihomedTest, StunDictionaryPerformsSync) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-  CreatePortAllocators(env);
+  CreatePortAllocators();
   AddAddress(0, kPublicAddrs[0], "eth0", ADAPTER_TYPE_CELLULAR);
   AddAddress(0, kAlternateAddrs[0], "vpn0", ADAPTER_TYPE_VPN,
              ADAPTER_TYPE_ETHERNET);
   AddAddress(1, kPublicAddrs[1]);
 
   // Create channels and let them go writable, as usual.
-  CreateChannels(env);
+  CreateChannels();
 
   MockFunction<void(IceTransportInternal*, const StunDictionaryView&,
                     std::span<uint16_t>)>
@@ -3584,8 +3438,8 @@ TEST_F(P2PTransportChannelMultihomedTest, StunDictionaryPerformsSync) {
         EXPECT_EQ(view.GetByteString(12)->string_view(), "keso");
       });
   EXPECT_CALL(writer_synced_func, Call).Times(1);
-  EXPECT_TRUE(WaitUntil([&] { return CheckConnected(ep1_ch1(), ep2_ch1()); },
-                        {.timeout = kMediumTimeout, .clock = &clock}));
+  EXPECT_TRUE(DefaultWait().Until(
+      [&] { return CheckConnected(ep1_ch1(), ep2_ch1()); }));
 }
 
 // A collection of tests which tests a single P2PTransportChannel by sending
@@ -3594,11 +3448,26 @@ class P2PTransportChannelPingTest : public ::testing::Test {
  public:
   P2PTransportChannelPingTest()
       : vss_(std::make_unique<VirtualSocketServer>()),
+        time_controller_(Timestamp::Millis(1000), vss_.get()),
+        env_(CreateTestEnvironment({.time = &time_controller_})),
         packet_socket_factory_(
             std::make_unique<BasicPacketSocketFactory>(vss_.get())),
-        thread_(vss_.get()) {}
+        thread_(time_controller_.GetMainThread()) {}
 
  protected:
+  Waiter DefaultWait() {
+    return Waiter({.timeout = kDefaultTimeout, .clock = &time_controller_});
+  }
+  Waiter MediumWait() {
+    return Waiter({.timeout = kMediumTimeout, .clock = &time_controller_});
+  }
+  Waiter ShortWait() {
+    return Waiter({.timeout = kShortTimeout, .clock = &time_controller_});
+  }
+  Waiter Wait(TimeDelta timeout) {
+    return Waiter({.timeout = timeout, .clock = &time_controller_});
+  }
+
   void PrepareChannel(P2PTransportChannel* ch) {
     ch->SetIceRole(ICEROLE_CONTROLLING);
     ch->SetIceParameters(kIceParams[0]);
@@ -3622,18 +3491,11 @@ class P2PTransportChannelPingTest : public ::testing::Test {
 
   Connection* WaitForConnectionTo(P2PTransportChannel* ch,
                                   absl::string_view ip,
-                                  int port_num,
-                                  ThreadProcessingFakeClock* clock = nullptr) {
-    if (clock == nullptr) {
-      EXPECT_THAT(WaitUntil([&] { return GetConnectionTo(ch, ip, port_num); },
-                            Ne(nullptr), {.timeout = kMediumTimeout}),
-                  IsRtcOk());
-    } else {
-      EXPECT_THAT(
-          WaitUntil([&] { return GetConnectionTo(ch, ip, port_num); },
-                    Ne(nullptr), {.timeout = kMediumTimeout, .clock = clock}),
-          IsRtcOk());
-    }
+                                  int port_num) {
+    EXPECT_THAT(
+        MediumWait().Until([&] { return GetConnectionTo(ch, ip, port_num); },
+                           Ne(nullptr)),
+        IsRtcOk());
     return GetConnectionTo(ch, ip, port_num);
   }
 
@@ -3679,17 +3541,16 @@ class P2PTransportChannelPingTest : public ::testing::Test {
   }
 
   Connection* CreateConnectionWithCandidate(P2PTransportChannel* channel,
-                                            ScopedFakeClock* clock,
                                             absl::string_view ip_addr,
                                             int port,
                                             int priority,
                                             bool writable) {
     channel->AddRemoteCandidate(
         CreateUdpCandidate(IceCandidateType::kHost, ip_addr, port, priority));
-    EXPECT_THAT(
-        WaitUntil([&] { return GetConnectionTo(channel, ip_addr, port); },
-                  Ne(nullptr), {.timeout = kMediumTimeout, .clock = clock}),
-        IsRtcOk());
+    EXPECT_THAT(MediumWait().Until(
+                    [&] { return GetConnectionTo(channel, ip_addr, port); },
+                    Ne(nullptr)),
+                IsRtcOk());
     Connection* conn = GetConnectionTo(channel, ip_addr, port);
 
     if (conn && writable) {
@@ -3811,10 +3672,12 @@ class P2PTransportChannelPingTest : public ::testing::Test {
     return packet_socket_factory_.get();
   }
 
- private:
+ protected:
   std::unique_ptr<VirtualSocketServer> vss_;
+  GlobalSimulatedTimeController time_controller_;
+  Environment env_;
   std::unique_ptr<PacketSocketFactory> packet_socket_factory_;
-  AutoSocketServerThread thread_;
+  Thread* thread_;
   int selected_candidate_pair_switches_ = 0;
   int last_sent_packet_id_ = -1;
   bool channel_ready_to_send_ = false;
@@ -3825,9 +3688,8 @@ class P2PTransportChannelPingTest : public ::testing::Test {
 };
 
 TEST_F(P2PTransportChannelPingTest, TestTriggeredChecks) {
-  const Environment env = CreateEnvironment();
-  FakePortAllocator pa(env, ss());
-  P2PTransportChannel ch(env, "trigger checks", 1, &pa);
+  FakePortAllocator pa(env_, ss());
+  P2PTransportChannel ch(env_, "trigger checks", 1, &pa);
   PrepareChannel(&ch);
   ch.MaybeStartGathering();
   ch.AddRemoteCandidate(
@@ -3852,9 +3714,8 @@ TEST_F(P2PTransportChannelPingTest, TestTriggeredChecks) {
 }
 
 TEST_F(P2PTransportChannelPingTest, TestAllConnectionsPingedSufficiently) {
-  const Environment env = CreateEnvironment();
-  FakePortAllocator pa(env, ss());
-  P2PTransportChannel ch(env, "ping sufficiently", 1, &pa);
+  FakePortAllocator pa(env_, ss());
+  P2PTransportChannel ch(env_, "ping sufficiently", 1, &pa);
   PrepareChannel(&ch);
   ch.MaybeStartGathering();
   ch.AddRemoteCandidate(
@@ -3870,24 +3731,20 @@ TEST_F(P2PTransportChannelPingTest, TestAllConnectionsPingedSufficiently) {
   // Low-priority connection becomes writable so that the other connection
   // is not pruned.
   conn1->ReceivedPingResponse(kLowRtt, "id");
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return conn1->num_pings_sent() >= kMinPingsAtWeakPingInterval &&
-               conn2->num_pings_sent() >= kMinPingsAtWeakPingInterval;
-      },
-      {.timeout = kDefaultTimeout}));
+  EXPECT_TRUE(MediumWait().Until([&] {
+    return conn1->num_pings_sent() >= kMinPingsAtWeakPingInterval &&
+           conn2->num_pings_sent() >= kMinPingsAtWeakPingInterval;
+  }));
 }
 
 // Verify that the connections are pinged at the right time.
 TEST_F(P2PTransportChannelPingTest, TestStunPingIntervals) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
   int RTT_RATIO = 4;
   constexpr TimeDelta kSchedulingRange = TimeDelta::Millis(200);
   constexpr TimeDelta kRttRange = TimeDelta::Millis(10);
 
-  FakePortAllocator pa(env, ss());
-  P2PTransportChannel ch(env, "TestChannel", 1, &pa);
+  FakePortAllocator pa(env_, ss());
+  P2PTransportChannel ch(env_, "TestChannel", 1, &pa);
   PrepareChannel(&ch);
   ch.MaybeStartGathering();
   ch.AddRemoteCandidate(
@@ -3895,27 +3752,27 @@ TEST_F(P2PTransportChannelPingTest, TestStunPingIntervals) {
   Connection* conn = WaitForConnectionTo(&ch, "1.1.1.1", 1);
 
   ASSERT_TRUE(conn != nullptr);
-  SIMULATED_WAIT(conn->num_pings_sent() == 1, kDefaultTimeout.ms(), clock);
+  EXPECT_TRUE(DefaultWait().Until([&] { return conn->num_pings_sent() == 1; }));
 
   // Initializing.
 
-  Timestamp start = env.clock().CurrentTime();
-  SIMULATED_WAIT(conn->num_pings_sent() >= kMinPingsAtWeakPingInterval,
-                 kDefaultTimeout.ms(), clock);
+  Timestamp start = env_.clock().CurrentTime();
+  EXPECT_TRUE(DefaultWait().Until(
+      [&] { return conn->num_pings_sent() >= kMinPingsAtWeakPingInterval; }));
   TimeDelta ping_interval =
-      (env.clock().CurrentTime() - start) / (kMinPingsAtWeakPingInterval - 1);
+      (env_.clock().CurrentTime() - start) / (kMinPingsAtWeakPingInterval - 1);
   EXPECT_EQ(ping_interval, kWeakPingInterval);
 
   // Stabilizing.
 
   conn->ReceivedPingResponse(kLowRtt, "id");
   int ping_sent_before = conn->num_pings_sent();
-  start = env.clock().CurrentTime();
+  start = env_.clock().CurrentTime();
   // The connection becomes strong but not stable because we haven't been able
   // to converge the RTT.
-  SIMULATED_WAIT(conn->num_pings_sent() == ping_sent_before + 1,
-                 kMediumTimeout.ms(), clock);
-  ping_interval = env.clock().CurrentTime() - start;
+  EXPECT_TRUE(DefaultWait().Until(
+      [&] { return conn->num_pings_sent() == ping_sent_before + 1; }));
+  ping_interval = env_.clock().CurrentTime() - start;
   EXPECT_GE(ping_interval, kWeakOrStabilizingWritableConnectionPingInterval);
   EXPECT_LE(ping_interval, kWeakOrStabilizingWritableConnectionPingInterval +
                                kSchedulingRange);
@@ -3928,10 +3785,10 @@ TEST_F(P2PTransportChannelPingTest, TestStunPingIntervals) {
     conn->ReceivedPingResponse(kLowRtt, "id");
   }
   ping_sent_before = conn->num_pings_sent();
-  start = env.clock().CurrentTime();
-  SIMULATED_WAIT(conn->num_pings_sent() == ping_sent_before + 1,
-                 kMediumTimeout.ms(), clock);
-  ping_interval = env.clock().CurrentTime() - start;
+  start = env_.clock().CurrentTime();
+  EXPECT_TRUE(MediumWait().Until(
+      [&] { return conn->num_pings_sent() == ping_sent_before + 1; }));
+  ping_interval = env_.clock().CurrentTime() - start;
   EXPECT_GE(ping_interval, kStrongAndStableWritableConnectionPingInterval);
   EXPECT_LE(ping_interval,
             kStrongAndStableWritableConnectionPingInterval + kSchedulingRange);
@@ -3941,24 +3798,24 @@ TEST_F(P2PTransportChannelPingTest, TestStunPingIntervals) {
   conn->ReceivedPingResponse(kLowRtt, "id");
   // Create a in-flight ping.
   conn->Ping();
-  start = env.clock().CurrentTime();
+  start = env_.clock().CurrentTime();
   // In-flight ping timeout and the connection will be unstable.
-  SIMULATED_WAIT(!conn->stable(env.clock().CurrentTime()), kMediumTimeout.ms(),
-                 clock);
-  TimeDelta duration = env.clock().CurrentTime() - start;
+  EXPECT_TRUE(MediumWait().Until(
+      [&] { return !conn->stable(env_.clock().CurrentTime()); }));
+  TimeDelta duration = env_.clock().CurrentTime() - start;
   EXPECT_GE(duration, 2 * conn->Rtt() - kRttRange);
   EXPECT_LE(duration, 2 * conn->Rtt() + kRttRange);
   // The connection become unstable due to not receiving ping responses.
   ping_sent_before = conn->num_pings_sent();
-  SIMULATED_WAIT(conn->num_pings_sent() == ping_sent_before + 1,
-                 kMediumTimeout.ms(), clock);
+  EXPECT_TRUE(MediumWait().Until(
+      [&] { return conn->num_pings_sent() == ping_sent_before + 1; }));
   // The interval is expected to be
   // kWeakOrStabilizingWritableConnectionPingInterval.
-  start = env.clock().CurrentTime();
+  start = env_.clock().CurrentTime();
   ping_sent_before = conn->num_pings_sent();
-  SIMULATED_WAIT(conn->num_pings_sent() == ping_sent_before + 1,
-                 kMediumTimeout.ms(), clock);
-  ping_interval = env.clock().CurrentTime() - start;
+  EXPECT_TRUE(MediumWait().Until(
+      [&] { return conn->num_pings_sent() == ping_sent_before + 1; }));
+  ping_interval = env_.clock().CurrentTime() - start;
   EXPECT_GE(ping_interval, kWeakOrStabilizingWritableConnectionPingInterval);
   EXPECT_LE(ping_interval, kWeakOrStabilizingWritableConnectionPingInterval +
                                kSchedulingRange);
@@ -3967,18 +3824,14 @@ TEST_F(P2PTransportChannelPingTest, TestStunPingIntervals) {
 // Test that we start pinging as soon as we have a connection and remote ICE
 // parameters.
 TEST_F(P2PTransportChannelPingTest, PingingStartedAsSoonAsPossible) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-
-  FakePortAllocator pa(env, ss());
-  P2PTransportChannel ch(env, "TestChannel", 1, &pa);
+  FakePortAllocator pa(env_, ss());
+  P2PTransportChannel ch(env_, "TestChannel", 1, &pa);
   ch.SetIceRole(ICEROLE_CONTROLLING);
   ch.SetIceParameters(kIceParams[0]);
   ch.MaybeStartGathering();
-  EXPECT_THAT(WaitUntil([&] { return ch.gathering_state(); },
-                        Eq(IceGatheringState::kIceGatheringComplete),
-                        {.timeout = kDefaultTimeout}),
-              IsRtcOk());
+  EXPECT_TRUE(MediumWait().Until([&] {
+    return ch.gathering_state() == IceGatheringState::kIceGatheringComplete;
+  }));
 
   // Simulate a binding request being received, creating a peer reflexive
   // candidate pair while we still don't have remote ICE parameters.
@@ -3997,22 +3850,19 @@ TEST_F(P2PTransportChannelPingTest, PingingStartedAsSoonAsPossible) {
 
   // Simulate waiting for a second (and change) and verify that no pings were
   // sent, since we don't yet have remote ICE parameters.
-  SIMULATED_WAIT(conn->num_pings_sent() > 0, 1025, clock);
+  time_controller_.AdvanceTime(TimeDelta::Millis(1025));
   EXPECT_EQ(0, conn->num_pings_sent());
 
   // Set remote ICE parameters. Now we should be able to ping. Ensure that
   // the first ping is sent as soon as possible, within one simulated clock
   // tick.
   ch.SetRemoteIceParameters(kIceParams[1]);
-  EXPECT_THAT(WaitUntil([&] { return conn->num_pings_sent(); }, Gt(0),
-                        {.clock = &clock}),
-              IsRtcOk());
+  EXPECT_TRUE(DefaultWait().Until([&] { return conn->num_pings_sent() > 0; }));
 }
 
 TEST_F(P2PTransportChannelPingTest, TestNoTriggeredChecksWhenWritable) {
-  const Environment env = CreateEnvironment();
-  FakePortAllocator pa(env, ss());
-  P2PTransportChannel ch(env, "trigger checks", 1, &pa);
+  FakePortAllocator pa(env_, ss());
+  P2PTransportChannel ch(env_, "trigger checks", 1, &pa);
   PrepareChannel(&ch);
   ch.MaybeStartGathering();
   ch.AddRemoteCandidate(
@@ -4038,9 +3888,8 @@ TEST_F(P2PTransportChannelPingTest, TestNoTriggeredChecksWhenWritable) {
 }
 
 TEST_F(P2PTransportChannelPingTest, TestFailedConnectionNotPingable) {
-  const Environment env = CreateEnvironment();
-  FakePortAllocator pa(env, ss());
-  P2PTransportChannel ch(env, "Do not ping failed connections", 1, &pa);
+  FakePortAllocator pa(env_, ss());
+  P2PTransportChannel ch(env_, "Do not ping failed connections", 1, &pa);
   PrepareChannel(&ch);
   ch.MaybeStartGathering();
   ch.AddRemoteCandidate(
@@ -4057,9 +3906,8 @@ TEST_F(P2PTransportChannelPingTest, TestFailedConnectionNotPingable) {
 }
 
 TEST_F(P2PTransportChannelPingTest, TestSignalStateChanged) {
-  const Environment env = CreateEnvironment();
-  FakePortAllocator pa(env, ss());
-  P2PTransportChannel ch(env, "state change", 1, &pa);
+  FakePortAllocator pa(env_, ss());
+  P2PTransportChannel ch(env_, "state change", 1, &pa);
   PrepareChannel(&ch);
   ch.MaybeStartGathering();
   ch.AddRemoteCandidate(
@@ -4069,9 +3917,8 @@ TEST_F(P2PTransportChannelPingTest, TestSignalStateChanged) {
   // Pruning the connection reduces the set of active connections and changes
   // the channel state.
   conn1->Prune();
-  EXPECT_THAT(WaitUntil([&] { return channel_state(); },
-                        Eq(IceTransportStateInternal::STATE_FAILED),
-                        {.timeout = kDefaultTimeout}),
+  EXPECT_THAT(DefaultWait().Until([&] { return channel_state(); },
+                                  Eq(IceTransportStateInternal::STATE_FAILED)),
               IsRtcOk());
 }
 
@@ -4082,9 +3929,8 @@ TEST_F(P2PTransportChannelPingTest, TestSignalStateChanged) {
 // parameters arrive. If a remote candidate is added with the current ICE
 // ufrag, its pwd and generation will be set properly.
 TEST_F(P2PTransportChannelPingTest, TestAddRemoteCandidateWithVariousUfrags) {
-  const Environment env = CreateEnvironment();
-  FakePortAllocator pa(env, ss());
-  P2PTransportChannel ch(env, "add candidate", 1, &pa);
+  FakePortAllocator pa(env_, ss());
+  P2PTransportChannel ch(env_, "add candidate", 1, &pa);
   PrepareChannel(&ch);
   ch.MaybeStartGathering();
   // Add a candidate with a future ufrag.
@@ -4108,7 +3954,7 @@ TEST_F(P2PTransportChannelPingTest, TestAddRemoteCandidateWithVariousUfrags) {
   // Add a candidate with an old ufrag. No connection will be created.
   ch.AddRemoteCandidate(CreateUdpCandidate(IceCandidateType::kHost, "2.2.2.2",
                                            2, 2, kIceUfrag[1]));
-  Thread::Current()->ProcessMessages(500);
+  time_controller_.AdvanceTime(TimeDelta::Millis(500));
   EXPECT_TRUE(GetConnectionTo(&ch, "2.2.2.2", 2) == nullptr);
 
   // Add a candidate with the current ufrag, its pwd and generation will be
@@ -4116,10 +3962,8 @@ TEST_F(P2PTransportChannelPingTest, TestAddRemoteCandidateWithVariousUfrags) {
   ch.AddRemoteCandidate(CreateUdpCandidate(IceCandidateType::kHost, "3.3.3.3",
                                            3, 0, kIceUfrag[2]));
   Connection* conn3 = nullptr;
-  ASSERT_THAT(
-      WaitUntil([&] { return conn3 = GetConnectionTo(&ch, "3.3.3.3", 3); },
-                Ne(nullptr), {.timeout = kMediumTimeout}),
-      IsRtcOk());
+  EXPECT_TRUE(DefaultWait().Until(
+      [&] { return (conn3 = GetConnectionTo(&ch, "3.3.3.3", 3)) != nullptr; }));
   const Candidate& new_candidate = conn3->remote_candidate();
   EXPECT_EQ(kIcePwd[2], new_candidate.password());
   EXPECT_EQ(1U, new_candidate.generation());
@@ -4137,9 +3981,8 @@ TEST_F(P2PTransportChannelPingTest, TestAddRemoteCandidateWithVariousUfrags) {
 }
 
 TEST_F(P2PTransportChannelPingTest, ConnectionResurrection) {
-  const Environment env = CreateEnvironment();
-  FakePortAllocator pa(env, ss());
-  P2PTransportChannel ch(env, "connection resurrection", 1, &pa);
+  FakePortAllocator pa(env_, ss());
+  P2PTransportChannel ch(env_, "connection resurrection", 1, &pa);
   PrepareChannel(&ch);
   ch.MaybeStartGathering();
 
@@ -4160,14 +4003,15 @@ TEST_F(P2PTransportChannelPingTest, ConnectionResurrection) {
   conn2->ReceivedPingResponse(kLowRtt, "id");
 
   // Wait for conn2 to be selected.
-  EXPECT_THAT(WaitUntil([&] { return ch.selected_connection(); }, Eq(conn2),
-                        {.timeout = kMediumTimeout}),
-              IsRtcOk());
+  EXPECT_THAT(
+      MediumWait().Until([&] { return ch.selected_connection(); }, Eq(conn2)),
+      IsRtcOk());
   // Destroy the connection to test SignalUnknownAddress.
   ch.RemoveConnectionForTest(conn1);
-  EXPECT_THAT(WaitUntil([&] { return GetConnectionTo(&ch, "1.1.1.1", 1); },
-                        Eq(nullptr), {.timeout = kMediumTimeout}),
-              IsRtcOk());
+  EXPECT_THAT(
+      MediumWait().Until([&] { return GetConnectionTo(&ch, "1.1.1.1", 1); },
+                         Eq(nullptr)),
+      IsRtcOk());
 
   // Create a minimal STUN message with prflx priority.
   IceMessage request(STUN_BINDING_REQUEST);
@@ -4195,10 +4039,8 @@ TEST_F(P2PTransportChannelPingTest, ConnectionResurrection) {
 }
 
 TEST_F(P2PTransportChannelPingTest, TestReceivingStateChange) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-  FakePortAllocator pa(env, ss());
-  P2PTransportChannel ch(env, "receiving state change", 1, &pa);
+  FakePortAllocator pa(env_, ss());
+  P2PTransportChannel ch(env_, "receiving state change", 1, &pa);
   PrepareChannel(&ch);
   // Default receiving timeout and checking receiving interval should not be too
   // small.
@@ -4210,18 +4052,16 @@ TEST_F(P2PTransportChannelPingTest, TestReceivingStateChange) {
   ch.MaybeStartGathering();
   ch.AddRemoteCandidate(
       CreateUdpCandidate(IceCandidateType::kHost, "1.1.1.1", 1, 1));
-  Connection* conn1 = WaitForConnectionTo(&ch, "1.1.1.1", 1, &clock);
+  Connection* conn1 = WaitForConnectionTo(&ch, "1.1.1.1", 1);
   ASSERT_TRUE(conn1 != nullptr);
 
-  clock.AdvanceTime(TimeDelta::Seconds(1));
+  time_controller_.AdvanceTime(TimeDelta::Seconds(1));
   conn1->ReceivedPing();
   conn1->OnReadPacket(ReceivedIpPacket::CreateFromLegacy(
-      "ABC", 3, env.clock().TimeInMicroseconds()));
+      "ABC", 3, env_.clock().TimeInMicroseconds()));
 
-  EXPECT_TRUE(WaitUntil([&] { return ch.receiving(); },
-                        {.timeout = kShortTimeout, .clock = &clock}));
-  EXPECT_TRUE(WaitUntil([&] { return !ch.receiving(); },
-                        {.timeout = kShortTimeout, .clock = &clock}));
+  EXPECT_TRUE(MediumWait().Until([&] { return ch.receiving(); }));
+  EXPECT_TRUE(ShortWait().Until([&] { return !ch.receiving(); }));
 }
 
 // The controlled side will select a connection as the "selected connection"
@@ -4231,9 +4071,8 @@ TEST_F(P2PTransportChannelPingTest, TestReceivingStateChange) {
 // selected connection changes and SignalReadyToSend will be fired if the new
 // selected connection is writable.
 TEST_F(P2PTransportChannelPingTest, TestSelectConnectionBeforeNomination) {
-  const Environment env = CreateEnvironment();
-  FakePortAllocator pa(env, ss());
-  P2PTransportChannel ch(env, "receiving state change", 1, &pa);
+  FakePortAllocator pa(env_, ss());
+  P2PTransportChannel ch(env_, "receiving state change", 1, &pa);
   PrepareChannel(&ch);
   ch.SetIceRole(ICEROLE_CONTROLLED);
   ch.MaybeStartGathering();
@@ -4251,9 +4090,9 @@ TEST_F(P2PTransportChannelPingTest, TestSelectConnectionBeforeNomination) {
 
   // A connection needs to be writable before it is selected for transmission.
   conn1->ReceivedPingResponse(kLowRtt, "id");
-  EXPECT_THAT(WaitUntil([&] { return ch.selected_connection(); }, Eq(conn1),
-                        {.timeout = kDefaultTimeout}),
-              IsRtcOk());
+  EXPECT_THAT(
+      ShortWait().Until([&] { return ch.selected_connection(); }, Eq(conn1)),
+      IsRtcOk());
   EXPECT_TRUE(CandidatePairMatchesNetworkRoute(conn1));
   EXPECT_TRUE(ConnectionMatchesChangeEvent(
       conn1, "remote candidate generation maybe changed"));
@@ -4266,9 +4105,9 @@ TEST_F(P2PTransportChannelPingTest, TestSelectConnectionBeforeNomination) {
   Connection* conn2 = WaitForConnectionTo(&ch, "2.2.2.2", 2);
   ASSERT_TRUE(conn2 != nullptr);
   conn2->ReceivedPingResponse(kLowRtt, "id");
-  EXPECT_THAT(WaitUntil([&] { return ch.selected_connection(); }, Eq(conn2),
-                        {.timeout = kDefaultTimeout}),
-              IsRtcOk());
+  EXPECT_THAT(
+      DefaultWait().Until([&] { return ch.selected_connection(); }, Eq(conn2)),
+      IsRtcOk());
   EXPECT_TRUE(CandidatePairMatchesNetworkRoute(conn2));
   EXPECT_TRUE(
       ConnectionMatchesChangeEvent(conn2, "candidate pair state changed"));
@@ -4314,9 +4153,9 @@ TEST_F(P2PTransportChannelPingTest, TestSelectConnectionBeforeNomination) {
   reset_channel_ready_to_send();
   // The selected connection switches after conn4 becomes writable.
   conn4->ReceivedPingResponse(kLowRtt, "id");
-  EXPECT_THAT(WaitUntil([&] { return ch.selected_connection(); }, Eq(conn4),
-                        {.timeout = kDefaultTimeout}),
-              IsRtcOk());
+  EXPECT_THAT(
+      DefaultWait().Until([&] { return ch.selected_connection(); }, Eq(conn4)),
+      IsRtcOk());
   EXPECT_TRUE(CandidatePairMatchesNetworkRoute(conn4));
   EXPECT_TRUE(
       ConnectionMatchesChangeEvent(conn4, "candidate pair state changed"));
@@ -4329,10 +4168,13 @@ TEST_F(P2PTransportChannelPingTest, TestSelectConnectionBeforeNomination) {
 // that sends a ping directly when a connection has been nominated
 // i.e on the ICE_CONTROLLED-side.
 TEST_F(P2PTransportChannelPingTest, TestPingOnNomination) {
-  const Environment env = CreateEnvironment(CreateTestFieldTrialsPtr(
-      "WebRTC-IceFieldTrials/send_ping_on_nomination_ice_controlled:true/"));
-  FakePortAllocator pa(env, ss());
-  P2PTransportChannel ch(env, "receiving state change", 1, &pa);
+  env_ = CreateTestEnvironment(
+      {.field_trials = CreateTestFieldTrialsPtr(
+           "WebRTC-IceFieldTrials/"
+           "send_ping_on_nomination_ice_controlled:true/"),
+       .time = &time_controller_});
+  FakePortAllocator pa(env_, ss());
+  P2PTransportChannel ch(env_, "receiving state change", 1, &pa);
   PrepareChannel(&ch);
   ch.SetIceConfig(ch.config());
   ch.SetIceRole(ICEROLE_CONTROLLED);
@@ -4344,9 +4186,9 @@ TEST_F(P2PTransportChannelPingTest, TestPingOnNomination) {
 
   // A connection needs to be writable before it is selected for transmission.
   conn1->ReceivedPingResponse(kLowRtt, "id");
-  EXPECT_THAT(WaitUntil([&] { return ch.selected_connection(); }, Eq(conn1),
-                        {.timeout = kDefaultTimeout}),
-              IsRtcOk());
+  EXPECT_THAT(
+      DefaultWait().Until([&] { return ch.selected_connection(); }, Eq(conn1)),
+      IsRtcOk());
   EXPECT_TRUE(CandidatePairMatchesNetworkRoute(conn1));
 
   // When a higher priority candidate comes in, the new connection is chosen
@@ -4356,9 +4198,9 @@ TEST_F(P2PTransportChannelPingTest, TestPingOnNomination) {
   Connection* conn2 = WaitForConnectionTo(&ch, "2.2.2.2", 2);
   ASSERT_TRUE(conn2 != nullptr);
   conn2->ReceivedPingResponse(kLowRtt, "id");
-  EXPECT_THAT(WaitUntil([&] { return ch.selected_connection(); }, Eq(conn2),
-                        {.timeout = kDefaultTimeout}),
-              IsRtcOk());
+  EXPECT_THAT(
+      DefaultWait().Until([&] { return ch.selected_connection(); }, Eq(conn2)),
+      IsRtcOk());
   EXPECT_TRUE(CandidatePairMatchesNetworkRoute(conn2));
 
   // Now nominate conn1 (low prio), it shall be choosen.
@@ -4375,10 +4217,12 @@ TEST_F(P2PTransportChannelPingTest, TestPingOnNomination) {
 // that sends a ping directly when switching to a new connection
 // on the ICE_CONTROLLING-side.
 TEST_F(P2PTransportChannelPingTest, TestPingOnSwitch) {
-  const Environment env = CreateEnvironment(CreateTestFieldTrialsPtr(
-      "WebRTC-IceFieldTrials/send_ping_on_switch_ice_controlling:true/"));
-  FakePortAllocator pa(env, ss());
-  P2PTransportChannel ch(env, "receiving state change", 1, &pa);
+  env_ = CreateTestEnvironment(
+      {.field_trials = CreateTestFieldTrialsPtr(
+           "WebRTC-IceFieldTrials/send_ping_on_switch_ice_controlling:true/"),
+       .time = &time_controller_});
+  FakePortAllocator pa(env_, ss());
+  P2PTransportChannel ch(env_, "receiving state change", 1, &pa);
   PrepareChannel(&ch);
   ch.SetIceConfig(ch.config());
   ch.SetIceRole(ICEROLE_CONTROLLING);
@@ -4390,9 +4234,9 @@ TEST_F(P2PTransportChannelPingTest, TestPingOnSwitch) {
 
   // A connection needs to be writable before it is selected for transmission.
   conn1->ReceivedPingResponse(kLowRtt, "id");
-  EXPECT_THAT(WaitUntil([&] { return ch.selected_connection(); }, Eq(conn1),
-                        {.timeout = kDefaultTimeout}),
-              IsRtcOk());
+  EXPECT_THAT(
+      DefaultWait().Until([&] { return ch.selected_connection(); }, Eq(conn1)),
+      IsRtcOk());
   EXPECT_TRUE(CandidatePairMatchesNetworkRoute(conn1));
 
   // When a higher priority candidate comes in, the new connection is chosen
@@ -4405,9 +4249,9 @@ TEST_F(P2PTransportChannelPingTest, TestPingOnSwitch) {
   const int before = conn2->num_pings_sent();
 
   conn2->ReceivedPingResponse(kLowRtt, "id");
-  EXPECT_THAT(WaitUntil([&] { return ch.selected_connection(); }, Eq(conn2),
-                        {.timeout = kDefaultTimeout}),
-              IsRtcOk());
+  EXPECT_THAT(
+      DefaultWait().Until([&] { return ch.selected_connection(); }, Eq(conn2)),
+      IsRtcOk());
   EXPECT_TRUE(CandidatePairMatchesNetworkRoute(conn2));
 
   // And the additional ping should have been sent directly.
@@ -4418,10 +4262,12 @@ TEST_F(P2PTransportChannelPingTest, TestPingOnSwitch) {
 // that sends a ping directly when selecteing a new connection
 // on the ICE_CONTROLLING-side (i.e also initial selection).
 TEST_F(P2PTransportChannelPingTest, TestPingOnSelected) {
-  const Environment env = CreateEnvironment(CreateTestFieldTrialsPtr(
-      "WebRTC-IceFieldTrials/send_ping_on_selected_ice_controlling:true/"));
-  FakePortAllocator pa(env, ss());
-  P2PTransportChannel ch(env, "receiving state change", 1, &pa);
+  env_ = CreateTestEnvironment(
+      {.field_trials = CreateTestFieldTrialsPtr(
+           "WebRTC-IceFieldTrials/send_ping_on_selected_ice_controlling:true/"),
+       .time = &time_controller_});
+  FakePortAllocator pa(env_, ss());
+  P2PTransportChannel ch(env_, "receiving state change", 1, &pa);
   PrepareChannel(&ch);
   ch.SetIceConfig(ch.config());
   ch.SetIceRole(ICEROLE_CONTROLLING);
@@ -4435,9 +4281,9 @@ TEST_F(P2PTransportChannelPingTest, TestPingOnSelected) {
 
   // A connection needs to be writable before it is selected for transmission.
   conn1->ReceivedPingResponse(kLowRtt, "id");
-  EXPECT_THAT(WaitUntil([&] { return ch.selected_connection(); }, Eq(conn1),
-                        {.timeout = kDefaultTimeout}),
-              IsRtcOk());
+  EXPECT_THAT(
+      DefaultWait().Until([&] { return ch.selected_connection(); }, Eq(conn1)),
+      IsRtcOk());
   EXPECT_TRUE(CandidatePairMatchesNetworkRoute(conn1));
 
   // And the additional ping should have been sent directly.
@@ -4451,9 +4297,8 @@ TEST_F(P2PTransportChannelPingTest, TestPingOnSelected) {
 // also sends back a ping response and set the ICE pwd in the remote candidate
 // appropriately.
 TEST_F(P2PTransportChannelPingTest, TestSelectConnectionFromUnknownAddress) {
-  const Environment env = CreateEnvironment();
-  FakePortAllocator pa(env, ss());
-  P2PTransportChannel ch(env, "receiving state change", 1, &pa);
+  FakePortAllocator pa(env_, ss());
+  P2PTransportChannel ch(env_, "receiving state change", 1, &pa);
   PrepareChannel(&ch);
   ch.SetIceRole(ICEROLE_CONTROLLED);
   ch.MaybeStartGathering();
@@ -4472,9 +4317,9 @@ TEST_F(P2PTransportChannelPingTest, TestSelectConnectionFromUnknownAddress) {
   EXPECT_EQ(conn1->stats().sent_ping_responses, 1u);
   EXPECT_NE(conn1, ch.selected_connection());
   conn1->ReceivedPingResponse(kLowRtt, "id");
-  EXPECT_THAT(WaitUntil([&] { return ch.selected_connection(); }, Eq(conn1),
-                        {.timeout = kDefaultTimeout}),
-              IsRtcOk());
+  EXPECT_THAT(
+      DefaultWait().Until([&] { return ch.selected_connection(); }, Eq(conn1)),
+      IsRtcOk());
 
   // Another connection is nominated via use_candidate.
   ch.AddRemoteCandidate(
@@ -4512,9 +4357,9 @@ TEST_F(P2PTransportChannelPingTest, TestSelectConnectionFromUnknownAddress) {
   // conn4 is not the selected connection yet because it is not writable.
   EXPECT_EQ(conn2, ch.selected_connection());
   conn4->ReceivedPingResponse(kLowRtt, "id");  // Become writable.
-  EXPECT_THAT(WaitUntil([&] { return ch.selected_connection(); }, Eq(conn4),
-                        {.timeout = kDefaultTimeout}),
-              IsRtcOk());
+  EXPECT_THAT(
+      DefaultWait().Until([&] { return ch.selected_connection(); }, Eq(conn4)),
+      IsRtcOk());
 
   // Test that the request from an unknown address contains a ufrag from an old
   // generation.
@@ -4534,9 +4379,8 @@ TEST_F(P2PTransportChannelPingTest, TestSelectConnectionFromUnknownAddress) {
 // at which point the controlled side will select that connection as
 // the "selected connection".
 TEST_F(P2PTransportChannelPingTest, TestSelectConnectionBasedOnMediaReceived) {
-  const Environment env = CreateEnvironment();
-  FakePortAllocator pa(env, ss());
-  P2PTransportChannel ch(env, "receiving state change", 1, &pa);
+  FakePortAllocator pa(env_, ss());
+  P2PTransportChannel ch(env_, "receiving state change", 1, &pa);
   PrepareChannel(&ch);
   ch.SetIceRole(ICEROLE_CONTROLLED);
   ch.MaybeStartGathering();
@@ -4545,9 +4389,9 @@ TEST_F(P2PTransportChannelPingTest, TestSelectConnectionBasedOnMediaReceived) {
   Connection* conn1 = WaitForConnectionTo(&ch, "1.1.1.1", 1);
   ASSERT_TRUE(conn1 != nullptr);
   conn1->ReceivedPingResponse(kLowRtt, "id");
-  EXPECT_THAT(WaitUntil([&] { return ch.selected_connection(); }, Eq(conn1),
-                        {.timeout = kDefaultTimeout}),
-              IsRtcOk());
+  EXPECT_THAT(
+      DefaultWait().Until([&] { return ch.selected_connection(); }, Eq(conn1)),
+      IsRtcOk());
 
   // If a data packet is received on conn2, the selected connection should
   // switch to conn2 because the controlled side must mirror the media path
@@ -4558,7 +4402,7 @@ TEST_F(P2PTransportChannelPingTest, TestSelectConnectionBasedOnMediaReceived) {
   ASSERT_TRUE(conn2 != nullptr);
   conn2->ReceivedPingResponse(kLowRtt, "id");  // Become writable and receiving.
   conn2->OnReadPacket(ReceivedIpPacket::CreateFromLegacy(
-      "ABC", 3, env.clock().TimeInMicroseconds()));
+      "ABC", 3, env_.clock().TimeInMicroseconds()));
   EXPECT_EQ(conn2, ch.selected_connection());
   conn2->ReceivedPingResponse(kLowRtt, "id");  // Become writable.
 
@@ -4579,37 +4423,34 @@ TEST_F(P2PTransportChannelPingTest, TestSelectConnectionBasedOnMediaReceived) {
   ASSERT_TRUE(conn3 != nullptr);
   EXPECT_NE(conn3, ch.selected_connection());  // Not writable yet.
   conn3->ReceivedPingResponse(kLowRtt, "id");  // Become writable.
-  EXPECT_THAT(WaitUntil([&] { return ch.selected_connection(); }, Eq(conn3),
-                        {.timeout = kDefaultTimeout}),
-              IsRtcOk());
+  EXPECT_THAT(
+      DefaultWait().Until([&] { return ch.selected_connection(); }, Eq(conn3)),
+      IsRtcOk());
 
   // Now another data packet will not switch the selected connection because the
   // selected connection was nominated by the controlling side.
   conn2->ReceivedPing();
   conn2->ReceivedPingResponse(kLowRtt, "id");
   conn2->OnReadPacket(ReceivedIpPacket::CreateFromLegacy(
-      "XYZ", 3, env.clock().TimeInMicroseconds()));
-  EXPECT_THAT(WaitUntil([&] { return ch.selected_connection(); }, Eq(conn3),
-                        {.timeout = kDefaultTimeout}),
-              IsRtcOk());
+      "XYZ", 3, env_.clock().TimeInMicroseconds()));
+  EXPECT_THAT(
+      DefaultWait().Until([&] { return ch.selected_connection(); }, Eq(conn3)),
+      IsRtcOk());
 }
 
 TEST_F(P2PTransportChannelPingTest,
        TestControlledAgentDataReceivingTakesHigherPrecedenceThanPriority) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-  clock.AdvanceTime(TimeDelta::Seconds(1));
-  FakePortAllocator pa(env, ss());
-  P2PTransportChannel ch(env, "SwitchSelectedConnection", 1, &pa);
+  time_controller_.AdvanceTime(TimeDelta::Seconds(1));
+  FakePortAllocator pa(env_, ss());
+  P2PTransportChannel ch(env_, "SwitchSelectedConnection", 1, &pa);
   PrepareChannel(&ch);
   ch.SetIceRole(ICEROLE_CONTROLLED);
   ch.MaybeStartGathering();
   // The connections have decreasing priority.
   Connection* conn1 =
-      CreateConnectionWithCandidate(&ch, &clock, "1.1.1.1", 1, 10, true);
+      CreateConnectionWithCandidate(&ch, "1.1.1.1", 1, 10, true);
   ASSERT_TRUE(conn1 != nullptr);
-  Connection* conn2 =
-      CreateConnectionWithCandidate(&ch, &clock, "2.2.2.2", 2, 9, true);
+  Connection* conn2 = CreateConnectionWithCandidate(&ch, "2.2.2.2", 2, 9, true);
   ASSERT_TRUE(conn2 != nullptr);
 
   // Initially, connections are selected based on priority.
@@ -4619,59 +4460,56 @@ TEST_F(P2PTransportChannelPingTest,
   // conn2 receives data; it becomes selected.
   // Advance the clock by 1ms so that the last data receiving timestamp of
   // conn2 is larger.
-  SIMULATED_WAIT(false, 1, clock);
+  time_controller_.AdvanceTime(TimeDelta::Millis(1));
 
   conn2->OnReadPacket(ReceivedIpPacket::CreateFromLegacy(
-      "XYZ", 3, env.clock().TimeInMicroseconds()));
+      "XYZ", 3, env_.clock().TimeInMicroseconds()));
   EXPECT_EQ(1, reset_selected_candidate_pair_switches());
   EXPECT_TRUE(CandidatePairMatchesNetworkRoute(conn2));
 
   // conn1 also receives data; it becomes selected due to priority again.
   conn1->OnReadPacket(ReceivedIpPacket::CreateFromLegacy(
-      "ABC", 3, env.clock().TimeInMicroseconds()));
+      "ABC", 3, env_.clock().TimeInMicroseconds()));
   EXPECT_EQ(1, reset_selected_candidate_pair_switches());
   EXPECT_TRUE(CandidatePairMatchesNetworkRoute(conn2));
 
   // conn2 received data more recently; it is selected now because it
   // received data more recently.
-  SIMULATED_WAIT(false, 1, clock);
+  time_controller_.AdvanceTime(TimeDelta::Millis(1));
   // Need to become writable again because it was pruned.
   conn2->ReceivedPingResponse(kLowRtt, "id");
   conn2->OnReadPacket(ReceivedIpPacket::CreateFromLegacy(
-      "ABC", 3, env.clock().TimeInMicroseconds()));
+      "ABC", 3, env_.clock().TimeInMicroseconds()));
   EXPECT_EQ(1, reset_selected_candidate_pair_switches());
   EXPECT_TRUE(CandidatePairMatchesNetworkRoute(conn2));
 
   // Make sure sorting won't reselect candidate pair.
-  SIMULATED_WAIT(false, 10, clock);
+  time_controller_.AdvanceTime(TimeDelta::Millis(10));
   EXPECT_EQ(0, reset_selected_candidate_pair_switches());
 }
 
 TEST_F(P2PTransportChannelPingTest,
        TestControlledAgentNominationTakesHigherPrecedenceThanDataReceiving) {
-  ScopedFakeClock clock;
-  clock.AdvanceTime(TimeDelta::Seconds(1));
-  const Environment env = CreateEnvironment();
+  time_controller_.AdvanceTime(TimeDelta::Seconds(1));
 
-  FakePortAllocator pa(env, ss());
-  P2PTransportChannel ch(env, "SwitchSelectedConnection", 1, &pa);
+  FakePortAllocator pa(env_, ss());
+  P2PTransportChannel ch(env_, "SwitchSelectedConnection", 1, &pa);
   PrepareChannel(&ch);
   ch.SetIceRole(ICEROLE_CONTROLLED);
   ch.MaybeStartGathering();
   // The connections have decreasing priority.
   Connection* conn1 =
-      CreateConnectionWithCandidate(&ch, &clock, "1.1.1.1", 1, 10, true);
+      CreateConnectionWithCandidate(&ch, "1.1.1.1", 1, 10, true);
   ASSERT_TRUE(conn1 != nullptr);
-  Connection* conn2 =
-      CreateConnectionWithCandidate(&ch, &clock, "2.2.2.2", 2, 9, true);
+  Connection* conn2 = CreateConnectionWithCandidate(&ch, "2.2.2.2", 2, 9, true);
   ASSERT_TRUE(conn2 != nullptr);
 
   // conn1 received data; it is the selected connection.
   // Advance the clock to have a non-zero last-data-receiving time.
-  SIMULATED_WAIT(false, 1, clock);
+  time_controller_.AdvanceTime(TimeDelta::Millis(1));
 
   conn1->OnReadPacket(ReceivedIpPacket::CreateFromLegacy(
-      "XYZ", 3, env.clock().TimeInMicroseconds()));
+      "XYZ", 3, env_.clock().TimeInMicroseconds()));
   EXPECT_EQ(1, reset_selected_candidate_pair_switches());
   EXPECT_TRUE(CandidatePairMatchesNetworkRoute(conn1));
 
@@ -4686,33 +4524,30 @@ TEST_F(P2PTransportChannelPingTest,
   EXPECT_TRUE(CandidatePairMatchesNetworkRoute(conn2));
 
   // Make sure sorting won't reselect candidate pair.
-  SIMULATED_WAIT(false, 10, clock);
+  time_controller_.AdvanceTime(TimeDelta::Millis(10));
   EXPECT_EQ(0, reset_selected_candidate_pair_switches());
 }
 
 TEST_F(P2PTransportChannelPingTest,
        TestControlledAgentSelectsConnectionWithHigherNomination) {
-  ScopedFakeClock clock;
-  clock.AdvanceTime(TimeDelta::Seconds(1));
-  const Environment env = CreateEnvironment();
+  time_controller_.AdvanceTime(TimeDelta::Seconds(1));
 
-  FakePortAllocator pa(env, ss());
-  P2PTransportChannel ch(env, "test", 1, &pa);
+  FakePortAllocator pa(env_, ss());
+  P2PTransportChannel ch(env_, "test", 1, &pa);
   PrepareChannel(&ch);
   ch.SetIceRole(ICEROLE_CONTROLLED);
   ch.MaybeStartGathering();
   // The connections have decreasing priority.
   Connection* conn1 =
-      CreateConnectionWithCandidate(&ch, &clock, "1.1.1.1", 1, 10, true);
+      CreateConnectionWithCandidate(&ch, "1.1.1.1", 1, 10, true);
   ASSERT_TRUE(conn1 != nullptr);
-  Connection* conn2 =
-      CreateConnectionWithCandidate(&ch, &clock, "2.2.2.2", 2, 9, true);
+  Connection* conn2 = CreateConnectionWithCandidate(&ch, "2.2.2.2", 2, 9, true);
   ASSERT_TRUE(conn2 != nullptr);
 
   // conn1 is the selected connection because it has a higher priority,
-  EXPECT_THAT(WaitUntil([&] { return ch.selected_connection(); }, Eq(conn1),
-                        {.timeout = kDefaultTimeout, .clock = &clock}),
-              IsRtcOk());
+  EXPECT_THAT(
+      DefaultWait().Until([&] { return ch.selected_connection(); }, Eq(conn1)),
+      IsRtcOk());
   EXPECT_TRUE(CandidatePairMatchesNetworkRoute(conn1));
   reset_selected_candidate_pair_switches();
 
@@ -4735,34 +4570,32 @@ TEST_F(P2PTransportChannelPingTest,
   EXPECT_TRUE(CandidatePairMatchesNetworkRoute(conn2));
 
   // Make sure sorting won't reselect candidate pair.
-  SIMULATED_WAIT(false, 100, clock);
+  time_controller_.AdvanceTime(TimeDelta::Millis(100));
   EXPECT_EQ(0, reset_selected_candidate_pair_switches());
 }
 
 TEST_F(P2PTransportChannelPingTest, TestEstimatedDisconnectedTime) {
-  ScopedFakeClock clock;
-  clock.AdvanceTime(TimeDelta::Seconds(1));
-  const Environment env = CreateEnvironment();
+  time_controller_.AdvanceTime(TimeDelta::Seconds(1));
 
-  FakePortAllocator pa(env, ss());
-  P2PTransportChannel ch(env, "test", 1, &pa);
+  FakePortAllocator pa(env_, ss());
+  P2PTransportChannel ch(env_, "test", 1, &pa);
   PrepareChannel(&ch);
   ch.SetIceRole(ICEROLE_CONTROLLED);
   ch.MaybeStartGathering();
   // The connections have decreasing priority.
   Connection* conn1 =
-      CreateConnectionWithCandidate(&ch, &clock, "1.1.1.1", /* port= */ 1,
+      CreateConnectionWithCandidate(&ch, "1.1.1.1", /* port= */ 1,
                                     /* priority= */ 10, /* writable= */ true);
   ASSERT_TRUE(conn1 != nullptr);
   Connection* conn2 =
-      CreateConnectionWithCandidate(&ch, &clock, "2.2.2.2", /* port= */ 2,
+      CreateConnectionWithCandidate(&ch, "2.2.2.2", /* port= */ 2,
                                     /* priority= */ 9, /* writable= */ true);
   ASSERT_TRUE(conn2 != nullptr);
 
   // conn1 is the selected connection because it has a higher priority,
-  EXPECT_THAT(WaitUntil([&] { return ch.selected_connection(); }, Eq(conn1),
-                        {.timeout = kDefaultTimeout, .clock = &clock}),
-              IsRtcOk());
+  EXPECT_THAT(
+      DefaultWait().Until([&] { return ch.selected_connection(); }, Eq(conn1)),
+      IsRtcOk());
   EXPECT_TRUE(CandidatePairMatchesNetworkRoute(conn1));
   // No estimateded disconnect time at first connect <=> value is 0.
   EXPECT_EQ(LastEstimatedDisconnectedTimeMs(), 0);
@@ -4771,11 +4604,16 @@ TEST_F(P2PTransportChannelPingTest, TestEstimatedDisconnectedTime) {
   int nomination = 1;
 
   {
-    clock.AdvanceTime(TimeDelta::Seconds(1));
+    time_controller_.AdvanceTime(TimeDelta::Seconds(1));
     // This will not parse as STUN, and is considered data
     conn1->OnReadPacket(ReceivedIpPacket::CreateFromLegacy(
-        "XYZ", 3, env.clock().TimeInMicroseconds()));
-    clock.AdvanceTime(TimeDelta::Seconds(2));
+        "XYZ", 3, env_.clock().TimeInMicroseconds()));
+    time_controller_.AdvanceTime(TimeDelta::Seconds(2));
+
+    // To be eligible for selection upon nomination, conn2 must be receiving.
+    // The real time controller fires timers that cause conn2 to lose receiving
+    // status during the 2s wait.
+    conn2->ReceivedPingResponse(kLowRtt, "id");
 
     // conn2 is nominated; it becomes selected.
     NominateConnection(conn2, nomination++);
@@ -4785,13 +4623,16 @@ TEST_F(P2PTransportChannelPingTest, TestEstimatedDisconnectedTime) {
   }
 
   {
-    clock.AdvanceTime(TimeDelta::Seconds(1));
+    time_controller_.AdvanceTime(TimeDelta::Seconds(1));
     conn2->OnReadPacket(ReceivedIpPacket::CreateFromLegacy(
-        "XYZ", 3, env.clock().TimeInMicroseconds()));
-    clock.AdvanceTime(TimeDelta::Seconds(2));
+        "XYZ", 3, env_.clock().TimeInMicroseconds()));
+    time_controller_.AdvanceTime(TimeDelta::Seconds(2));
     ReceivePingOnConnection(conn2, kIceUfrag[1], 1, nomination++);
 
-    clock.AdvanceTime(TimeDelta::Millis(500));
+    // To be eligible for selection upon nomination, conn1 must be receiving.
+    conn1->ReceivedPingResponse(kLowRtt, "id");
+
+    time_controller_.AdvanceTime(TimeDelta::Millis(500));
 
     ReceivePingOnConnection(conn1, kIceUfrag[1], 1, nomination++);
     EXPECT_EQ(conn1, ch.selected_connection());
@@ -4802,17 +4643,15 @@ TEST_F(P2PTransportChannelPingTest, TestEstimatedDisconnectedTime) {
 
 TEST_F(P2PTransportChannelPingTest,
        TestControlledAgentIgnoresSmallerNomination) {
-  ScopedFakeClock clock;
-  clock.AdvanceTime(TimeDelta::Seconds(1));
-  const Environment env = CreateEnvironment();
+  time_controller_.AdvanceTime(TimeDelta::Seconds(1));
 
-  FakePortAllocator pa(env, ss());
-  P2PTransportChannel ch(env, "test", 1, &pa);
+  FakePortAllocator pa(env_, ss());
+  P2PTransportChannel ch(env_, "test", 1, &pa);
   PrepareChannel(&ch);
   ch.SetIceRole(ICEROLE_CONTROLLED);
   ch.MaybeStartGathering();
   Connection* conn =
-      CreateConnectionWithCandidate(&ch, &clock, "1.1.1.1", 1, 10, false);
+      CreateConnectionWithCandidate(&ch, "1.1.1.1", 1, 10, false);
   ReceivePingOnConnection(conn, kIceUfrag[1], 1, 2U);
   EXPECT_EQ(2U, conn->remote_nomination());
   // Smaller nomination is ignored.
@@ -4822,20 +4661,17 @@ TEST_F(P2PTransportChannelPingTest,
 
 TEST_F(P2PTransportChannelPingTest,
        TestControlledAgentWriteStateTakesHigherPrecedenceThanNomination) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-
-  FakePortAllocator pa(env, ss());
-  P2PTransportChannel ch(env, "SwitchSelectedConnection", 1, &pa);
+  FakePortAllocator pa(env_, ss());
+  P2PTransportChannel ch(env_, "SwitchSelectedConnection", 1, &pa);
   PrepareChannel(&ch);
   ch.SetIceRole(ICEROLE_CONTROLLED);
   ch.MaybeStartGathering();
   // The connections have decreasing priority.
   Connection* conn1 =
-      CreateConnectionWithCandidate(&ch, &clock, "1.1.1.1", 1, 10, false);
+      CreateConnectionWithCandidate(&ch, "1.1.1.1", 1, 10, false);
   ASSERT_TRUE(conn1 != nullptr);
   Connection* conn2 =
-      CreateConnectionWithCandidate(&ch, &clock, "2.2.2.2", 2, 9, false);
+      CreateConnectionWithCandidate(&ch, "2.2.2.2", 2, 9, false);
   ASSERT_TRUE(conn2 != nullptr);
 
   NominateConnection(conn1);
@@ -4845,36 +4681,35 @@ TEST_F(P2PTransportChannelPingTest,
   // conn2 becomes writable; it is selected even though it is not nominated.
   conn2->ReceivedPingResponse(kLowRtt, "id");
   EXPECT_THAT(
-      WaitUntil([&] { return reset_selected_candidate_pair_switches(); }, Eq(1),
-                {.timeout = kDefaultTimeout, .clock = &clock}),
+      DefaultWait().Until(
+          [&] { return reset_selected_candidate_pair_switches(); }, Eq(1)),
+      IsRtcOk());
+  EXPECT_THAT(
+      DefaultWait().Until([&] { return ch.selected_connection(); }, Eq(conn2)),
       IsRtcOk());
-  EXPECT_THAT(WaitUntil([&] { return ch.selected_connection(); }, Eq(conn2),
-                        {.timeout = kDefaultTimeout, .clock = &clock}),
-              IsRtcOk());
   EXPECT_TRUE(CandidatePairMatchesNetworkRoute(conn2));
 
   // If conn1 is also writable, it will become selected.
   conn1->ReceivedPingResponse(kLowRtt, "id");
   EXPECT_THAT(
-      WaitUntil([&] { return reset_selected_candidate_pair_switches(); }, Eq(1),
-                {.timeout = kDefaultTimeout, .clock = &clock}),
+      DefaultWait().Until(
+          [&] { return reset_selected_candidate_pair_switches(); }, Eq(1)),
+      IsRtcOk());
+  EXPECT_THAT(
+      DefaultWait().Until([&] { return ch.selected_connection(); }, Eq(conn1)),
       IsRtcOk());
-  EXPECT_THAT(WaitUntil([&] { return ch.selected_connection(); }, Eq(conn1),
-                        {.timeout = kDefaultTimeout, .clock = &clock}),
-              IsRtcOk());
   EXPECT_TRUE(CandidatePairMatchesNetworkRoute(conn1));
 
   // Make sure sorting won't reselect candidate pair.
-  SIMULATED_WAIT(false, 10, clock);
+  time_controller_.AdvanceTime(TimeDelta::Millis(10));
   EXPECT_EQ(0, reset_selected_candidate_pair_switches());
 }
 
 // Test that if a new remote candidate has the same address and port with
 // an old one, it will be used to create a new connection.
 TEST_F(P2PTransportChannelPingTest, TestAddRemoteCandidateWithAddressReuse) {
-  const Environment env = CreateEnvironment();
-  FakePortAllocator pa(env, ss());
-  P2PTransportChannel ch(env, "candidate reuse", 1, &pa);
+  FakePortAllocator pa(env_, ss());
+  P2PTransportChannel ch(env_, "candidate reuse", 1, &pa);
   PrepareChannel(&ch);
   ch.MaybeStartGathering();
   const std::string host_address = "1.1.1.1";
@@ -4911,11 +4746,9 @@ TEST_F(P2PTransportChannelPingTest, TestAddRemoteCandidateWithAddressReuse) {
 // When the current selected connection is strong, lower-priority connections
 // will be pruned. Otherwise, lower-priority connections are kept.
 TEST_F(P2PTransportChannelPingTest, TestDontPruneWhenWeak) {
-  ScopedFakeClock clock;
-  clock.AdvanceTime(TimeDelta::Seconds(1));
-  const Environment env = CreateEnvironment();
-  FakePortAllocator pa(env, ss());
-  P2PTransportChannel ch(env, "test channel", 1, &pa);
+  time_controller_.AdvanceTime(TimeDelta::Seconds(1));
+  FakePortAllocator pa(env_, ss());
+  P2PTransportChannel ch(env_, "test channel", 1, &pa);
   PrepareChannel(&ch);
   ch.SetIceRole(ICEROLE_CONTROLLED);
   ch.MaybeStartGathering();
@@ -4930,60 +4763,54 @@ TEST_F(P2PTransportChannelPingTest, TestDontPruneWhenWeak) {
   // lower-priority are pruned.
   ch.AddRemoteCandidate(
       CreateUdpCandidate(IceCandidateType::kHost, "2.2.2.2", 2, 10));
-  Connection* conn2 = WaitForConnectionTo(&ch, "2.2.2.2", 2, &clock);
+  Connection* conn2 = WaitForConnectionTo(&ch, "2.2.2.2", 2);
   ASSERT_TRUE(conn2 != nullptr);
   conn2->ReceivedPingResponse(kLowRtt, "id");  // Becomes writable and receiving
   NominateConnection(conn2);
-  EXPECT_TRUE(WaitUntil([&] { return conn1->pruned(); },
-                        {.timeout = kMediumTimeout, .clock = &clock}));
+  EXPECT_TRUE(DefaultWait().Until([&] { return conn1->pruned(); }));
 
   ch.SetIceConfig(CreateIceConfig(TimeDelta::Millis(500), GATHER_ONCE));
   // Wait until conn2 becomes not receiving.
-  EXPECT_TRUE(WaitUntil([&] { return !conn2->receiving(); },
-                        {.timeout = kMediumTimeout, .clock = &clock}));
+  EXPECT_TRUE(MediumWait().Until([&] { return !conn2->receiving(); }));
 
   ch.AddRemoteCandidate(
       CreateUdpCandidate(IceCandidateType::kHost, "3.3.3.3", 3, 1));
-  Connection* conn3 = WaitForConnectionTo(&ch, "3.3.3.3", 3, &clock);
+  Connection* conn3 = WaitForConnectionTo(&ch, "3.3.3.3", 3);
   ASSERT_TRUE(conn3 != nullptr);
   // The selected connection should still be conn2. Even through conn3 has lower
   // priority and is not receiving/writable, it is not pruned because the
   // selected connection is not receiving.
-  SIMULATED_WAIT(conn3->pruned(), kShortTimeout.ms(), clock);
+  EXPECT_FALSE(MediumWait().Until([&] { return conn3->pruned(); }));
   EXPECT_FALSE(conn3->pruned());
 }
 
 TEST_F(P2PTransportChannelPingTest, TestDontPruneHighPriorityConnections) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-  FakePortAllocator pa(env, ss());
-  P2PTransportChannel ch(env, "test channel", 1, &pa);
+  FakePortAllocator pa(env_, ss());
+  P2PTransportChannel ch(env_, "test channel", 1, &pa);
   PrepareChannel(&ch);
   ch.SetIceRole(ICEROLE_CONTROLLED);
   ch.MaybeStartGathering();
   Connection* conn1 =
-      CreateConnectionWithCandidate(&ch, &clock, "1.1.1.1", 1, 100, true);
+      CreateConnectionWithCandidate(&ch, "1.1.1.1", 1, 100, true);
   ASSERT_TRUE(conn1 != nullptr);
   Connection* conn2 =
-      CreateConnectionWithCandidate(&ch, &clock, "2.2.2.2", 2, 200, false);
+      CreateConnectionWithCandidate(&ch, "2.2.2.2", 2, 200, false);
   ASSERT_TRUE(conn2 != nullptr);
   // Even if conn1 is writable, nominated, receiving data, it should not prune
   // conn2.
   NominateConnection(conn1);
-  SIMULATED_WAIT(false, 1, clock);
+  time_controller_.AdvanceTime(TimeDelta::Millis(1));
   conn1->OnReadPacket(ReceivedIpPacket::CreateFromLegacy(
-      "XYZ", 3, env.clock().TimeInMicroseconds()));
-  SIMULATED_WAIT(conn2->pruned(), 100, clock);
+      "XYZ", 3, env_.clock().TimeInMicroseconds()));
+  (void)ShortWait().Until([&] { return conn2->pruned(); });
   EXPECT_FALSE(conn2->pruned());
 }
 
 // Test that GetState returns the state correctly.
 TEST_F(P2PTransportChannelPingTest, TestGetState) {
-  ScopedFakeClock clock;
-  clock.AdvanceTime(TimeDelta::Seconds(1));
-  const Environment env = CreateEnvironment();
-  FakePortAllocator pa(env, ss());
-  P2PTransportChannel ch(env, "test channel", 1, &pa);
+  time_controller_.AdvanceTime(TimeDelta::Seconds(1));
+  FakePortAllocator pa(env_, ss());
+  P2PTransportChannel ch(env_, "test channel", 1, &pa);
   EXPECT_EQ(IceTransportState::kNew, ch.GetIceTransportState());
   PrepareChannel(&ch);
   ch.MaybeStartGathering();
@@ -4998,8 +4825,8 @@ TEST_F(P2PTransportChannelPingTest, TestGetState) {
   // Checking candidates that have been added with gathered candidates.
   ASSERT_GT(ch.connections().size(), 0u);
   EXPECT_EQ(IceTransportState::kChecking, ch.GetIceTransportState());
-  Connection* conn1 = WaitForConnectionTo(&ch, "1.1.1.1", 1, &clock);
-  Connection* conn2 = WaitForConnectionTo(&ch, "2.2.2.2", 2, &clock);
+  Connection* conn1 = WaitForConnectionTo(&ch, "1.1.1.1", 1);
+  Connection* conn2 = WaitForConnectionTo(&ch, "2.2.2.2", 2);
   ASSERT_TRUE(conn1 != nullptr);
   ASSERT_TRUE(conn2 != nullptr);
   // Now there are two connections, so the transport channel is connecting.
@@ -5009,15 +4836,13 @@ TEST_F(P2PTransportChannelPingTest, TestGetState) {
   EXPECT_EQ(IceTransportState::kChecking, ch.GetIceTransportState());
   // `conn1` becomes writable and receiving; it then should prune `conn2`.
   conn1->ReceivedPingResponse(kLowRtt, "id");
-  EXPECT_TRUE(WaitUntil([&] { return conn2->pruned(); },
-                        {.timeout = kShortTimeout, .clock = &clock}));
+  EXPECT_TRUE(DefaultWait().Until([&] { return conn2->pruned(); }));
   EXPECT_EQ(IceTransportStateInternal::STATE_COMPLETED, ch.GetState());
   EXPECT_EQ(IceTransportState::kConnected, ch.GetIceTransportState());
   conn1->Prune();  // All connections are pruned.
   // Need to wait until the channel state is updated.
-  EXPECT_THAT(WaitUntil([&] { return ch.GetState(); },
-                        Eq(IceTransportStateInternal::STATE_FAILED),
-                        {.timeout = kShortTimeout, .clock = &clock}),
+  EXPECT_THAT(ShortWait().Until([&] { return ch.GetState(); },
+                                Eq(IceTransportStateInternal::STATE_FAILED)),
               IsRtcOk());
   EXPECT_EQ(IceTransportState::kFailed, ch.GetIceTransportState());
 }
@@ -5025,12 +4850,10 @@ TEST_F(P2PTransportChannelPingTest, TestGetState) {
 // Test that when a low-priority connection is pruned, it is not deleted
 // right away, and it can become active and be pruned again.
 TEST_F(P2PTransportChannelPingTest, TestConnectionPrunedAgain) {
-  ScopedFakeClock clock;
-  clock.AdvanceTime(TimeDelta::Seconds(1));
-  const Environment env = CreateEnvironment();
+  time_controller_.AdvanceTime(TimeDelta::Seconds(1));
 
-  FakePortAllocator pa(env, ss());
-  P2PTransportChannel ch(env, "test channel", 1, &pa);
+  FakePortAllocator pa(env_, ss());
+  P2PTransportChannel ch(env_, "test channel", 1, &pa);
   PrepareChannel(&ch);
   IceConfig config = CreateIceConfig(TimeDelta::Seconds(1), GATHER_ONCE);
   config.receiving_switching_delay = TimeDelta::Millis(800);
@@ -5038,13 +4861,13 @@ TEST_F(P2PTransportChannelPingTest, TestConnectionPrunedAgain) {
   ch.MaybeStartGathering();
   ch.AddRemoteCandidate(
       CreateUdpCandidate(IceCandidateType::kHost, "1.1.1.1", 1, 100));
-  Connection* conn1 = WaitForConnectionTo(&ch, "1.1.1.1", 1, &clock);
+  Connection* conn1 = WaitForConnectionTo(&ch, "1.1.1.1", 1);
   ASSERT_TRUE(conn1 != nullptr);
   EXPECT_EQ(nullptr, ch.selected_connection());
   conn1->ReceivedPingResponse(kLowRtt, "id");  // Becomes writable and receiving
-  EXPECT_THAT(WaitUntil([&] { return ch.selected_connection(); }, Eq(conn1),
-                        {.timeout = kDefaultTimeout, .clock = &clock}),
-              IsRtcOk());
+  EXPECT_THAT(
+      ShortWait().Until([&] { return ch.selected_connection(); }, Eq(conn1)),
+      IsRtcOk());
 
   // Add a low-priority connection `conn2`, which will be pruned, but it will
   // not be deleted right away. Once the current selected connection becomes not
@@ -5052,74 +4875,66 @@ TEST_F(P2PTransportChannelPingTest, TestConnectionPrunedAgain) {
   // it will become the selected connection.
   ch.AddRemoteCandidate(
       CreateUdpCandidate(IceCandidateType::kHost, "2.2.2.2", 2, 1));
-  Connection* conn2 = WaitForConnectionTo(&ch, "2.2.2.2", 2, &clock);
+  Connection* conn2 = WaitForConnectionTo(&ch, "2.2.2.2", 2);
   ASSERT_TRUE(conn2 != nullptr);
-  EXPECT_TRUE(WaitUntil([&] { return !conn2->active(); },
-                        {.timeout = kDefaultTimeout, .clock = &clock}));
+  EXPECT_TRUE(DefaultWait().Until([&] { return !conn2->active(); }));
   // `conn2` should not send a ping yet.
   EXPECT_EQ(IceCandidatePairState::WAITING, conn2->state());
   EXPECT_EQ(IceTransportStateInternal::STATE_COMPLETED, ch.GetState());
   // Wait for `conn1` becoming not receiving.
-  EXPECT_TRUE(WaitUntil([&] { return !conn1->receiving(); },
-                        {.timeout = kMediumTimeout, .clock = &clock}));
+  EXPECT_TRUE(DefaultWait().Until([&] { return !conn1->receiving(); }));
   // Make sure conn2 is not deleted.
-  conn2 = WaitForConnectionTo(&ch, "2.2.2.2", 2, &clock);
+  conn2 = WaitForConnectionTo(&ch, "2.2.2.2", 2);
   ASSERT_TRUE(conn2 != nullptr);
-  EXPECT_THAT(WaitUntil([&] { return conn2->state(); },
-                        Eq(IceCandidatePairState::IN_PROGRESS),
-                        {.timeout = kDefaultTimeout, .clock = &clock}),
+  EXPECT_THAT(MediumWait().Until([&] { return conn2->state(); },
+                                 Eq(IceCandidatePairState::IN_PROGRESS)),
               IsRtcOk());
   conn2->ReceivedPingResponse(kLowRtt, "id");
-  EXPECT_THAT(WaitUntil([&] { return ch.selected_connection(); }, Eq(conn2),
-                        {.timeout = kDefaultTimeout, .clock = &clock}),
-              IsRtcOk());
+  EXPECT_THAT(
+      DefaultWait().Until([&] { return ch.selected_connection(); }, Eq(conn2)),
+      IsRtcOk());
   EXPECT_EQ(IceTransportStateInternal::STATE_CONNECTING, ch.GetState());
 
   // When `conn1` comes back again, `conn2` will be pruned again.
   conn1->ReceivedPingResponse(kLowRtt, "id");
-  EXPECT_THAT(WaitUntil([&] { return ch.selected_connection(); }, Eq(conn1),
-                        {.timeout = kDefaultTimeout, .clock = &clock}),
-              IsRtcOk());
-  EXPECT_TRUE(WaitUntil([&] { return !conn2->active(); },
-                        {.timeout = kDefaultTimeout, .clock = &clock}));
+  EXPECT_THAT(
+      DefaultWait().Until([&] { return ch.selected_connection(); }, Eq(conn1)),
+      IsRtcOk());
+  EXPECT_TRUE(DefaultWait().Until([&] { return !conn2->active(); }));
   EXPECT_EQ(IceTransportStateInternal::STATE_COMPLETED, ch.GetState());
 }
 
 // Test that if all connections in a channel has timed out on writing, they
 // will all be deleted. We use Prune to simulate write_time_out.
 TEST_F(P2PTransportChannelPingTest, TestDeleteConnectionsIfAllWriteTimedout) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-  FakePortAllocator pa(env, ss());
-  P2PTransportChannel ch(env, "test channel", 1, &pa);
+  FakePortAllocator pa(env_, ss());
+  P2PTransportChannel ch(env_, "test channel", 1, &pa);
   PrepareChannel(&ch);
   ch.MaybeStartGathering();
   // Have one connection only but later becomes write-time-out.
   ch.AddRemoteCandidate(
       CreateUdpCandidate(IceCandidateType::kHost, "1.1.1.1", 1, 100));
-  Connection* conn1 = WaitForConnectionTo(&ch, "1.1.1.1", 1, &clock);
+  Connection* conn1 = WaitForConnectionTo(&ch, "1.1.1.1", 1);
   ASSERT_TRUE(conn1 != nullptr);
   conn1->ReceivedPing();  // Becomes receiving
   conn1->Prune();
-  EXPECT_TRUE(WaitUntil([&] { return ch.connections().empty(); },
-                        {.timeout = kShortTimeout, .clock = &clock}));
+  EXPECT_TRUE(DefaultWait().Until([&] { return ch.connections().empty(); }));
 
   // Have two connections but both become write-time-out later.
   ch.AddRemoteCandidate(
       CreateUdpCandidate(IceCandidateType::kHost, "2.2.2.2", 2, 1));
-  Connection* conn2 = WaitForConnectionTo(&ch, "2.2.2.2", 2, &clock);
+  Connection* conn2 = WaitForConnectionTo(&ch, "2.2.2.2", 2);
   ASSERT_TRUE(conn2 != nullptr);
   conn2->ReceivedPing();  // Becomes receiving
   ch.AddRemoteCandidate(
       CreateUdpCandidate(IceCandidateType::kHost, "3.3.3.3", 3, 2));
-  Connection* conn3 = WaitForConnectionTo(&ch, "3.3.3.3", 3, &clock);
+  Connection* conn3 = WaitForConnectionTo(&ch, "3.3.3.3", 3);
   ASSERT_TRUE(conn3 != nullptr);
   conn3->ReceivedPing();  // Becomes receiving
   // Now prune both conn2 and conn3; they will be deleted soon.
   conn2->Prune();
   conn3->Prune();
-  EXPECT_TRUE(WaitUntil([&] { return ch.connections().empty(); },
-                        {.timeout = kShortTimeout, .clock = &clock}));
+  EXPECT_TRUE(ShortWait().Until([&] { return ch.connections().empty(); }));
 }
 
 // Tests that after a port allocator session is started, it will be stopped
@@ -5127,9 +4942,8 @@ TEST_F(P2PTransportChannelPingTest, TestDeleteConnectionsIfAllWriteTimedout) {
 // connection belonging to an old session becomes writable, it won't stop
 // the current port allocator session.
 TEST_F(P2PTransportChannelPingTest, TestStopPortAllocatorSessions) {
-  const Environment env = CreateEnvironment();
-  FakePortAllocator pa(env, ss());
-  P2PTransportChannel ch(env, "test channel", 1, &pa);
+  FakePortAllocator pa(env_, ss());
+  P2PTransportChannel ch(env_, "test channel", 1, &pa);
   PrepareChannel(&ch);
   ch.SetIceConfig(CreateIceConfig(TimeDelta::Seconds(2), GATHER_ONCE));
   ch.MaybeStartGathering();
@@ -5163,9 +4977,8 @@ TEST_F(P2PTransportChannelPingTest, TestStopPortAllocatorSessions) {
 // These ports may still have connections that need a correct role, in case that
 // the connections on it may still receive stun pings.
 TEST_F(P2PTransportChannelPingTest, TestIceRoleUpdatedOnRemovedPort) {
-  const Environment env = CreateEnvironment();
-  FakePortAllocator pa(env, ss());
-  P2PTransportChannel ch(env, "test channel", ICE_CANDIDATE_COMPONENT_DEFAULT,
+  FakePortAllocator pa(env_, ss());
+  P2PTransportChannel ch(env_, "test channel", ICE_CANDIDATE_COMPONENT_DEFAULT,
                          &pa);
   // Starts with ICEROLE_CONTROLLING.
   PrepareChannel(&ch);
@@ -5191,9 +5004,8 @@ TEST_F(P2PTransportChannelPingTest, TestIceRoleUpdatedOnRemovedPort) {
 // pings sent by those connections until they're replaced by newer-generation
 // connections.
 TEST_F(P2PTransportChannelPingTest, TestIceRoleUpdatedOnPortAfterIceRestart) {
-  const Environment env = CreateEnvironment();
-  FakePortAllocator pa(env, ss());
-  P2PTransportChannel ch(env, "test channel", ICE_CANDIDATE_COMPONENT_DEFAULT,
+  FakePortAllocator pa(env_, ss());
+  P2PTransportChannel ch(env_, "test channel", ICE_CANDIDATE_COMPONENT_DEFAULT,
                          &pa);
   // Starts with ICEROLE_CONTROLLING.
   PrepareChannel(&ch);
@@ -5216,11 +5028,8 @@ TEST_F(P2PTransportChannelPingTest, TestIceRoleUpdatedOnPortAfterIceRestart) {
 // will be destroyed. The port will only be destroyed after it is marked as
 // "pruned."
 TEST_F(P2PTransportChannelPingTest, TestPortDestroyedAfterTimeoutAndPruned) {
-  ScopedFakeClock fake_clock;
-  const Environment env = CreateEnvironment();
-
-  FakePortAllocator pa(env, ss());
-  P2PTransportChannel ch(env, "test channel", ICE_CANDIDATE_COMPONENT_DEFAULT,
+  FakePortAllocator pa(env_, ss());
+  P2PTransportChannel ch(env_, "test channel", ICE_CANDIDATE_COMPONENT_DEFAULT,
                          &pa);
   PrepareChannel(&ch);
   ch.SetIceRole(ICEROLE_CONTROLLED);
@@ -5234,7 +5043,7 @@ TEST_F(P2PTransportChannelPingTest, TestPortDestroyedAfterTimeoutAndPruned) {
   // Simulate 2 minutes going by. This should be enough time for the port to
   // time out.
   for (int second = 0; second < 120; ++second) {
-    fake_clock.AdvanceTime(TimeDelta::Seconds(1));
+    time_controller_.AdvanceTime(TimeDelta::Seconds(1));
   }
   EXPECT_EQ(nullptr, GetConnectionTo(&ch, "1.1.1.1", 1));
   // Port will not be removed because it is not pruned yet.
@@ -5243,19 +5052,20 @@ TEST_F(P2PTransportChannelPingTest, TestPortDestroyedAfterTimeoutAndPruned) {
 
   // If the session prunes all ports, the port will be destroyed.
   ch.allocator_session()->PruneAllPorts();
-  EXPECT_THAT(WaitUntil([&] { return GetPort(&ch); }, Eq(nullptr),
-                        {.clock = &fake_clock}),
-              IsRtcOk());
-  EXPECT_THAT(WaitUntil([&] { return GetPrunedPort(&ch); }, Eq(nullptr),
-                        {.clock = &fake_clock}),
+  EXPECT_THAT(DefaultWait().Until([&] { return GetPort(&ch); }, Eq(nullptr)),
               IsRtcOk());
+  EXPECT_THAT(
+      DefaultWait().Until([&] { return GetPrunedPort(&ch); }, Eq(nullptr)),
+      IsRtcOk());
 }
 
 TEST_F(P2PTransportChannelPingTest, TestMaxOutstandingPingsFieldTrial) {
-  const Environment env = CreateEnvironment(CreateTestFieldTrialsPtr(
-      "WebRTC-IceFieldTrials/max_outstanding_pings:3/"));
-  FakePortAllocator pa(env, ss());
-  P2PTransportChannel ch(env, "max", 1, &pa);
+  env_ = CreateTestEnvironment(
+      {.field_trials = CreateTestFieldTrialsPtr(
+           "WebRTC-IceFieldTrials/max_outstanding_pings:3/"),
+       .time = &time_controller_});
+  FakePortAllocator pa(env_, ss());
+  P2PTransportChannel ch(env_, "max", 1, &pa);
   ch.SetIceConfig(ch.config());
   PrepareChannel(&ch);
   ch.MaybeStartGathering();
@@ -5269,11 +5079,9 @@ TEST_F(P2PTransportChannelPingTest, TestMaxOutstandingPingsFieldTrial) {
   ASSERT_TRUE(conn1 != nullptr);
   ASSERT_TRUE(conn2 != nullptr);
 
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return conn1->num_pings_sent() == 3 && conn2->num_pings_sent() == 3;
-      },
-      {.timeout = kDefaultTimeout}));
+  EXPECT_TRUE(DefaultWait().Until([&] {
+    return conn1->num_pings_sent() == 3 && conn2->num_pings_sent() == 3;
+  }));
 
   // Check that these connections don't send any more pings.
   EXPECT_EQ(nullptr, ch.FindNextPingableConnection());
@@ -5286,11 +5094,11 @@ class P2PTransportChannelMostLikelyToWorkFirstTest
     network_manager_.AddInterface(kPublicAddrs[0]);
   }
 
-  BasicPortAllocator& CreatePortAllocator(const Environment& env) {
-    turn_server_.emplace(env, Thread::Current(), ss(), kTurnUdpIntAddr,
+  BasicPortAllocator& CreatePortAllocator(const Environment& env_) {
+    turn_server_.emplace(env_, Thread::Current(), ss(), kTurnUdpIntAddr,
                          kTurnUdpExtAddr);
     port_allocator_ = CreateBasicPortAllocator(
-        env, &network_manager_, packet_socket_factory(), ServerAddresses(),
+        env_, &network_manager_, packet_socket_factory(), ServerAddresses(),
         kTurnUdpIntAddr, SocketAddress());
     port_allocator_->set_flags(port_allocator_->flags() |
                                PORTALLOCATOR_DISABLE_STUN |
@@ -5300,10 +5108,10 @@ class P2PTransportChannelMostLikelyToWorkFirstTest
   }
 
   P2PTransportChannel& StartTransportChannel(
-      const Environment& env,
+      const Environment& env_,
       bool prioritize_most_likely_to_work,
       TimeDelta stable_writable_connection_ping_interval) {
-    channel_ = std::make_unique<P2PTransportChannel>(env, "checks", 1,
+    channel_ = std::make_unique<P2PTransportChannel>(env_, "checks", 1,
                                                      port_allocator_.get());
     IceConfig config = channel_->config();
     config.prioritize_most_likely_candidate_pairs =
@@ -5337,6 +5145,22 @@ class P2PTransportChannelMostLikelyToWorkFirstTest
     EXPECT_EQ(conn->remote_candidate().type(), remote_candidate_type);
   }
 
+ protected:
+  Waiter DefaultWait() {
+    return Waiter({.timeout = kDefaultTimeout, .clock = &time_controller_});
+  }
+  Waiter MediumWait() {
+    return Waiter({.timeout = kMediumTimeout, .clock = &time_controller_});
+  }
+  Waiter ShortWait() {
+    return Waiter({.timeout = kShortTimeout, .clock = &time_controller_});
+  }
+  Waiter Wait(TimeDelta timeout) {
+    return Waiter({.timeout = timeout, .clock = &time_controller_});
+  }
+
+  FakeNetworkManager& network_manager() { return network_manager_; }
+
  private:
   std::unique_ptr<BasicPortAllocator> port_allocator_;
   FakeNetworkManager network_manager_{Thread::Current()};
@@ -5349,13 +5173,11 @@ class P2PTransportChannelMostLikelyToWorkFirstTest
 // we have a selected connection.
 TEST_F(P2PTransportChannelMostLikelyToWorkFirstTest,
        TestRelayRelayFirstWhenNothingPingedYet) {
-  const Environment env = CreateEnvironment();
   const TimeDelta max_strong_interval = TimeDelta::Millis(500);
-  CreatePortAllocator(env);
+  CreatePortAllocator(env_);
   P2PTransportChannel& ch =
-      StartTransportChannel(env, true, max_strong_interval);
-  EXPECT_THAT(WaitUntil([&] { return ch.ports().size(); }, Eq(2),
-                        {.timeout = kDefaultTimeout}),
+      StartTransportChannel(env_, true, max_strong_interval);
+  EXPECT_THAT(DefaultWait().Until([&] { return ch.ports().size(); }, Eq(2)),
               IsRtcOk());
   EXPECT_EQ(ch.ports()[0]->Type(), IceCandidateType::kHost);
   EXPECT_EQ(ch.ports()[1]->Type(), IceCandidateType::kRelay);
@@ -5365,9 +5187,9 @@ TEST_F(P2PTransportChannelMostLikelyToWorkFirstTest,
   ch.AddRemoteCandidate(
       CreateUdpCandidate(IceCandidateType::kHost, "2.2.2.2", 2, 2));
 
-  EXPECT_THAT(WaitUntil([&] { return ch.connections().size(); }, Eq(4),
-                        {.timeout = kDefaultTimeout}),
-              IsRtcOk());
+  EXPECT_THAT(
+      DefaultWait().Until([&] { return ch.connections().size(); }, Eq(4)),
+      IsRtcOk());
 
   // Relay/Relay should be the first pingable connection.
   Connection* conn = FindNextPingableConnectionAndPingIt(&ch);
@@ -5418,21 +5240,19 @@ TEST_F(P2PTransportChannelMostLikelyToWorkFirstTest,
 // in the first round.
 TEST_F(P2PTransportChannelMostLikelyToWorkFirstTest,
        TestRelayRelayFirstWhenEverythingPinged) {
-  const Environment env = CreateEnvironment();
-  CreatePortAllocator(env);
+  CreatePortAllocator(env_);
   P2PTransportChannel& ch =
-      StartTransportChannel(env, true, TimeDelta::Millis(500));
-  EXPECT_THAT(WaitUntil([&] { return ch.ports().size(); }, Eq(2),
-                        {.timeout = kDefaultTimeout}),
+      StartTransportChannel(env_, true, TimeDelta::Millis(500));
+  EXPECT_THAT(DefaultWait().Until([&] { return ch.ports().size(); }, Eq(2)),
               IsRtcOk());
   EXPECT_EQ(ch.ports()[0]->Type(), IceCandidateType::kHost);
   EXPECT_EQ(ch.ports()[1]->Type(), IceCandidateType::kRelay);
 
   ch.AddRemoteCandidate(
       CreateUdpCandidate(IceCandidateType::kHost, "1.1.1.1", 1, 1));
-  EXPECT_THAT(WaitUntil([&] { return ch.connections().size(); }, Eq(2),
-                        {.timeout = kDefaultTimeout}),
-              IsRtcOk());
+  EXPECT_THAT(
+      DefaultWait().Until([&] { return ch.connections().size(); }, Eq(2)),
+      IsRtcOk());
 
   // Initially, only have Local/Local and Local/Relay.
   VerifyNextPingableConnection(IceCandidateType::kHost,
@@ -5443,9 +5263,9 @@ TEST_F(P2PTransportChannelMostLikelyToWorkFirstTest,
   // Remote Relay candidate arrives.
   ch.AddRemoteCandidate(
       CreateUdpCandidate(IceCandidateType::kRelay, "2.2.2.2", 2, 2));
-  EXPECT_THAT(WaitUntil([&] { return ch.connections().size(); }, Eq(4),
-                        {.timeout = kDefaultTimeout}),
-              IsRtcOk());
+  EXPECT_THAT(
+      DefaultWait().Until([&] { return ch.connections().size(); }, Eq(4)),
+      IsRtcOk());
 
   // Relay/Relay should be the first since it hasn't been pinged before.
   VerifyNextPingableConnection(IceCandidateType::kRelay,
@@ -5465,21 +5285,19 @@ TEST_F(P2PTransportChannelMostLikelyToWorkFirstTest,
 // before we re-ping Relay/Relay connections again.
 TEST_F(P2PTransportChannelMostLikelyToWorkFirstTest,
        TestNoStarvationOnNonRelayConnection) {
-  const Environment env = CreateEnvironment();
-  CreatePortAllocator(env);
+  CreatePortAllocator(env_);
   P2PTransportChannel& ch =
-      StartTransportChannel(env, true, TimeDelta::Millis(500));
-  EXPECT_THAT(WaitUntil([&] { return ch.ports().size(); }, Eq(2),
-                        {.timeout = kDefaultTimeout}),
+      StartTransportChannel(env_, true, TimeDelta::Millis(500));
+  EXPECT_THAT(DefaultWait().Until([&] { return ch.ports().size(); }, Eq(2)),
               IsRtcOk());
   EXPECT_EQ(ch.ports()[0]->Type(), IceCandidateType::kHost);
   EXPECT_EQ(ch.ports()[1]->Type(), IceCandidateType::kRelay);
 
   ch.AddRemoteCandidate(
       CreateUdpCandidate(IceCandidateType::kRelay, "1.1.1.1", 1, 1));
-  EXPECT_THAT(WaitUntil([&] { return ch.connections().size(); }, Eq(2),
-                        {.timeout = kDefaultTimeout}),
-              IsRtcOk());
+  EXPECT_THAT(
+      DefaultWait().Until([&] { return ch.connections().size(); }, Eq(2)),
+      IsRtcOk());
 
   // Initially, only have Relay/Relay and Local/Relay. Ping Relay/Relay first.
   VerifyNextPingableConnection(IceCandidateType::kRelay,
@@ -5492,9 +5310,9 @@ TEST_F(P2PTransportChannelMostLikelyToWorkFirstTest,
   // Remote Local candidate arrives.
   ch.AddRemoteCandidate(
       CreateUdpCandidate(IceCandidateType::kHost, "2.2.2.2", 2, 2));
-  EXPECT_THAT(WaitUntil([&] { return ch.connections().size(); }, Eq(4),
-                        {.timeout = kDefaultTimeout}),
-              IsRtcOk());
+  EXPECT_THAT(
+      DefaultWait().Until([&] { return ch.connections().size(); }, Eq(4)),
+      IsRtcOk());
 
   // Local/Local should be the first since it hasn't been pinged before.
   VerifyNextPingableConnection(IceCandidateType::kHost,
@@ -5514,13 +5332,14 @@ TEST_F(P2PTransportChannelMostLikelyToWorkFirstTest,
 // I.e that we never create connection between relay and non-relay.
 TEST_F(P2PTransportChannelMostLikelyToWorkFirstTest,
        TestSkipRelayToNonRelayConnectionsFieldTrial) {
-  const Environment env = CreateEnvironment(CreateTestFieldTrialsPtr(
-      "WebRTC-IceFieldTrials/skip_relay_to_non_relay_connections:true/"));
-  CreatePortAllocator(env);
+  env_ = CreateTestEnvironment(
+      {.field_trials = CreateTestFieldTrialsPtr(
+           "WebRTC-IceFieldTrials/skip_relay_to_non_relay_connections:true/"),
+       .time = &time_controller_});
+  CreatePortAllocator(env_);
   P2PTransportChannel& ch =
-      StartTransportChannel(env, true, TimeDelta::Millis(500));
-  EXPECT_THAT(WaitUntil([&] { return ch.ports().size(); }, Eq(2),
-                        {.timeout = kDefaultTimeout}),
+      StartTransportChannel(env_, true, TimeDelta::Millis(500));
+  EXPECT_THAT(DefaultWait().Until([&] { return ch.ports().size(); }, Eq(2)),
               IsRtcOk());
   EXPECT_EQ(ch.ports()[0]->Type(), IceCandidateType::kHost);
   EXPECT_EQ(ch.ports()[1]->Type(), IceCandidateType::kRelay);
@@ -5528,33 +5347,31 @@ TEST_F(P2PTransportChannelMostLikelyToWorkFirstTest,
   // Remote Relay candidate arrives.
   ch.AddRemoteCandidate(
       CreateUdpCandidate(IceCandidateType::kRelay, "1.1.1.1", 1, 1));
-  EXPECT_THAT(WaitUntil([&] { return ch.connections().size(); }, Eq(1),
-                        {.timeout = kDefaultTimeout}),
-              IsRtcOk());
+  EXPECT_THAT(
+      DefaultWait().Until([&] { return ch.connections().size(); }, Eq(1)),
+      IsRtcOk());
 
   // Remote Local candidate arrives.
   ch.AddRemoteCandidate(
       CreateUdpCandidate(IceCandidateType::kHost, "2.2.2.2", 2, 2));
-  EXPECT_THAT(WaitUntil([&] { return ch.connections().size(); }, Eq(2),
-                        {.timeout = kDefaultTimeout}),
-              IsRtcOk());
+  EXPECT_THAT(
+      DefaultWait().Until([&] { return ch.connections().size(); }, Eq(2)),
+      IsRtcOk());
 }
 
 // Test the ping sequence is UDP Relay/Relay followed by TCP Relay/Relay,
 // followed by the rest.
 TEST_F(P2PTransportChannelMostLikelyToWorkFirstTest, TestTcpTurn) {
-  const Environment env = CreateEnvironment();
   RelayServerConfig config;
   config.credentials = kRelayCredentials;
   config.ports.push_back(ProtocolAddress(kTurnTcpIntAddr, PROTO_TCP));
-  CreatePortAllocator(env).AddTurnServerForTesting(config);
+  CreatePortAllocator(env_).AddTurnServerForTesting(config);
   // Add a Tcp Turn server.
   turn_server().AddInternalSocket(kTurnTcpIntAddr, PROTO_TCP);
 
   P2PTransportChannel& ch =
-      StartTransportChannel(env, true, TimeDelta::Millis(500));
-  EXPECT_THAT(WaitUntil([&] { return ch.ports().size(); }, Eq(3),
-                        {.timeout = kDefaultTimeout}),
+      StartTransportChannel(env_, true, TimeDelta::Millis(500));
+  EXPECT_THAT(DefaultWait().Until([&] { return ch.ports().size(); }, Eq(3)),
               IsRtcOk());
   EXPECT_EQ(ch.ports()[0]->Type(), IceCandidateType::kHost);
   EXPECT_EQ(ch.ports()[1]->Type(), IceCandidateType::kRelay);
@@ -5563,9 +5380,9 @@ TEST_F(P2PTransportChannelMostLikelyToWorkFirstTest, TestTcpTurn) {
   // Remote Relay candidate arrives.
   ch.AddRemoteCandidate(
       CreateUdpCandidate(IceCandidateType::kRelay, "1.1.1.1", 1, 1));
-  EXPECT_THAT(WaitUntil([&] { return ch.connections().size(); }, Eq(3),
-                        {.timeout = kDefaultTimeout}),
-              IsRtcOk());
+  EXPECT_THAT(
+      DefaultWait().Until([&] { return ch.connections().size(); }, Eq(3)),
+      IsRtcOk());
 
   // UDP Relay/Relay should be pinged first.
   VerifyNextPingableConnection(IceCandidateType::kRelay,
@@ -5584,7 +5401,7 @@ TEST_F(P2PTransportChannelMostLikelyToWorkFirstTest, TestTcpTurn) {
 // when the address is a hostname. The destruction should happen even
 // if the channel is not destroyed.
 TEST(P2PTransportChannelResolverTest, HostnameCandidateIsResolved) {
-  const Environment env = CreateEnvironment();
+  const Environment env = CreateTestEnvironment();
   ResolverFactoryFixture resolver_fixture;
   std::unique_ptr<SocketServer> socket_server = CreateDefaultSocketServer();
   AutoSocketServerThread main_thread(socket_server.get());
@@ -5598,9 +5415,8 @@ TEST(P2PTransportChannelResolverTest, HostnameCandidateIsResolved) {
   hostname_candidate.set_address(hostname_address);
   channel->AddRemoteCandidate(hostname_candidate);
 
-  ASSERT_THAT(WaitUntil([&] { return channel->remote_candidates().size(); },
-                        Eq(1u), {.timeout = kDefaultTimeout}),
-              IsRtcOk());
+  ASSERT_TRUE(
+      WaitUntil([&] { return channel->remote_candidates().size() == 1; }));
   const RemoteCandidate& candidate = channel->remote_candidates()[0];
   EXPECT_FALSE(candidate.address().IsUnresolvedIP());
 }
@@ -5611,27 +5427,26 @@ TEST(P2PTransportChannelResolverTest, HostnameCandidateIsResolved) {
 // done.
 TEST_F(P2PTransportChannelTest,
        PeerReflexiveCandidateBeforeSignalingWithMdnsName) {
-  const Environment env = CreateEnvironment();
   // ep1 and ep2 will only gather host candidates with addresses
   // kPublicAddrs[0] and kPublicAddrs[1], respectively.
-  ConfigureEndpoints(env, OPEN, OPEN, kOnlyLocalPorts, kOnlyLocalPorts);
+  ConfigureEndpoints(OPEN, OPEN, kOnlyLocalPorts, kOnlyLocalPorts);
   // ICE parameter will be set up when creating the channels.
   set_remote_ice_parameter_source(FROM_SETICEPARAMETERS);
-  GetEndpoint(0)->network_manager_.set_mdns_responder(
+  GetEndpoint(0)->network_manager().set_mdns_responder(
       std::make_unique<FakeMdnsResponder>(Thread::Current()));
 
   ResolverFactoryFixture resolver_fixture;
-  GetEndpoint(1)->async_dns_resolver_factory_ = &resolver_fixture;
-  CreateChannels(env);
+  GetEndpoint(1)->set_async_dns_resolver_factory(&resolver_fixture);
+  CreateChannels();
   // Pause sending candidates from both endpoints until we find out what port
   // number is assgined to ep1's host candidate.
   PauseCandidates(0);
   PauseCandidates(1);
   ASSERT_THAT(
-      WaitUntil([&] { return GetEndpoint(0)->saved_candidates_.size(); },
-                Eq(1u), {.timeout = kMediumTimeout}),
+      DefaultWait().Until(
+          [&] { return GetEndpoint(0)->saved_candidates().size(); }, Eq(1u)),
       IsRtcOk());
-  const auto& local_candidate = GetEndpoint(0)->saved_candidates_[0].candidate;
+  const auto& local_candidate = GetEndpoint(0)->saved_candidates()[0].candidate;
   // The IP address of ep1's host candidate should be obfuscated.
   EXPECT_TRUE(local_candidate.address().IsUnresolvedIP());
   // This is the underlying private IP address of the same candidate at ep1.
@@ -5642,18 +5457,19 @@ TEST_F(P2PTransportChannelTest,
   // pair and start to ping. After receiving the ping, ep2 discovers a prflx
   // remote candidate and form a candidate pair as well.
   ResumeCandidates(1);
-  ASSERT_THAT(WaitUntil([&] { return ep1_ch1()->selected_connection(); },
-                        Ne(nullptr), {.timeout = kMediumTimeout}),
-              IsRtcOk());
+  ASSERT_THAT(
+      MediumWait().Until([&] { return ep1_ch1()->selected_connection(); },
+                         Ne(nullptr)),
+      IsRtcOk());
   // ep2 should have the selected connection connected to the prflx remote
   // candidate.
   const Connection* selected_connection = nullptr;
-  ASSERT_THAT(WaitUntil(
+  ASSERT_THAT(MediumWait().Until(
                   [&] {
                     return selected_connection =
                                ep2_ch1()->selected_connection();
                   },
-                  Ne(nullptr), {.timeout = kMediumTimeout}),
+                  Ne(nullptr)),
               IsRtcOk());
   EXPECT_TRUE(selected_connection->remote_candidate().is_prflx());
   EXPECT_EQ(kIceUfrag[0], selected_connection->remote_candidate().username());
@@ -5662,11 +5478,9 @@ TEST_F(P2PTransportChannelTest,
   resolver_fixture.SetAddressToReturn(local_address);
   ResumeCandidates(0);
   // Verify ep2's selected connection is updated to use the 'local' candidate.
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return ep2_ch1()->selected_connection()->remote_candidate().is_local();
-      },
-      {.timeout = kMediumTimeout}));
+  EXPECT_TRUE(MediumWait().Until([&] {
+    return ep2_ch1()->selected_connection()->remote_candidate().is_local();
+  }));
   EXPECT_EQ(selected_connection, ep2_ch1()->selected_connection());
 
   DestroyChannels();
@@ -5678,30 +5492,29 @@ TEST_F(P2PTransportChannelTest,
 // address after the resolution completes.
 TEST_F(P2PTransportChannelTest,
        PeerReflexiveCandidateDuringResolvingHostCandidateWithMdnsName) {
-  const Environment env = CreateEnvironment();
   ResolverFactoryFixture resolver_fixture;
   // Prevent resolution until triggered by FireDelayedResolution.
   resolver_fixture.DelayResolution();
 
   // ep1 and ep2 will only gather host candidates with addresses
   // kPublicAddrs[0] and kPublicAddrs[1], respectively.
-  ConfigureEndpoints(env, OPEN, OPEN, kOnlyLocalPorts, kOnlyLocalPorts);
+  ConfigureEndpoints(OPEN, OPEN, kOnlyLocalPorts, kOnlyLocalPorts);
   // ICE parameter will be set up when creating the channels.
   set_remote_ice_parameter_source(FROM_SETICEPARAMETERS);
-  GetEndpoint(0)->network_manager_.set_mdns_responder(
+  GetEndpoint(0)->network_manager().set_mdns_responder(
       std::make_unique<FakeMdnsResponder>(Thread::Current()));
-  GetEndpoint(1)->async_dns_resolver_factory_ = &resolver_fixture;
-  CreateChannels(env);
+  GetEndpoint(1)->set_async_dns_resolver_factory(&resolver_fixture);
+  CreateChannels();
   // Pause sending candidates from both endpoints until we find out what port
   // number is assgined to ep1's host candidate.
   PauseCandidates(0);
   PauseCandidates(1);
 
   ASSERT_THAT(
-      WaitUntil([&] { return GetEndpoint(0)->saved_candidates_.size(); },
-                Eq(1u), {.timeout = kMediumTimeout}),
+      MediumWait().Until(
+          [&] { return GetEndpoint(0)->saved_candidates().size(); }, Eq(1u)),
       IsRtcOk());
-  const auto& local_candidate = GetEndpoint(0)->saved_candidates_[0].candidate;
+  const auto& local_candidate = GetEndpoint(0)->saved_candidates()[0].candidate;
   // The IP address of ep1's host candidate should be obfuscated.
   ASSERT_TRUE(local_candidate.address().IsUnresolvedIP());
   // This is the underlying private IP address of the same candidate at ep1.
@@ -5713,9 +5526,10 @@ TEST_F(P2PTransportChannelTest,
   // by ep1. Let ep2 signal its host candidate with an IP address to ep1, so
   // that ep1 can form a candidate pair, select it and start to ping ep2.
   ResumeCandidates(1);
-  ASSERT_THAT(WaitUntil([&] { return ep1_ch1()->selected_connection(); },
-                        Ne(nullptr), {.timeout = kMediumTimeout}),
-              IsRtcOk());
+  ASSERT_THAT(
+      MediumWait().Until([&] { return ep1_ch1()->selected_connection(); },
+                         Ne(nullptr)),
+      IsRtcOk());
   // Let the mock resolver of ep2 receives the correct resolution.
   resolver_fixture.SetAddressToReturn(local_address);
 
@@ -5724,20 +5538,19 @@ TEST_F(P2PTransportChannelTest,
   //
   // There is a caveat in our implementation associated with this expectation.
   // See the big comment in P2PTransportChannel::OnUnknownAddress.
-  ASSERT_THAT(WaitUntil([&] { return ep2_ch1()->selected_connection(); },
-                        Ne(nullptr), {.timeout = kMediumTimeout}),
-              IsRtcOk());
+  ASSERT_THAT(
+      MediumWait().Until([&] { return ep2_ch1()->selected_connection(); },
+                         Ne(nullptr)),
+      IsRtcOk());
   EXPECT_TRUE(ep2_ch1()->selected_connection()->remote_candidate().is_prflx());
   // ep2 should also be able resolve the hostname candidate. The resolved remote
   // host candidate should be merged with the prflx remote candidate.
 
   resolver_fixture.FireDelayedResolution();
 
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return ep2_ch1()->selected_connection()->remote_candidate().is_local();
-      },
-      {.timeout = kMediumTimeout}));
+  EXPECT_TRUE(MediumWait().Until([&] {
+    return ep2_ch1()->selected_connection()->remote_candidate().is_local();
+  }));
   EXPECT_EQ(1u, ep2_ch1()->remote_candidates().size());
 
   DestroyChannels();
@@ -5747,28 +5560,27 @@ TEST_F(P2PTransportChannelTest,
 // which is obfuscated by an mDNS name, and if the peer can complete the name
 // resolution with the correct IP address, we can have a p2p connection.
 TEST_F(P2PTransportChannelTest, CanConnectWithHostCandidateWithMdnsName) {
-  const Environment env = CreateEnvironment();
   ResolverFactoryFixture resolver_fixture;
 
   // ep1 and ep2 will only gather host candidates with addresses
   // kPublicAddrs[0] and kPublicAddrs[1], respectively.
-  ConfigureEndpoints(env, OPEN, OPEN, kOnlyLocalPorts, kOnlyLocalPorts);
+  ConfigureEndpoints(OPEN, OPEN, kOnlyLocalPorts, kOnlyLocalPorts);
   // ICE parameter will be set up when creating the channels.
   set_remote_ice_parameter_source(FROM_SETICEPARAMETERS);
-  GetEndpoint(0)->network_manager_.set_mdns_responder(
+  GetEndpoint(0)->network_manager().set_mdns_responder(
       std::make_unique<FakeMdnsResponder>(Thread::Current()));
-  GetEndpoint(1)->async_dns_resolver_factory_ = &resolver_fixture;
-  CreateChannels(env);
+  GetEndpoint(1)->set_async_dns_resolver_factory(&resolver_fixture);
+  CreateChannels();
   // Pause sending candidates from both endpoints until we find out what port
   // number is assgined to ep1's host candidate.
   PauseCandidates(0);
   PauseCandidates(1);
   ASSERT_THAT(
-      WaitUntil([&] { return GetEndpoint(0)->saved_candidates_.size(); },
-                Eq(1u), {.timeout = kMediumTimeout}),
+      MediumWait().Until(
+          [&] { return GetEndpoint(0)->saved_candidates().size(); }, Eq(1u)),
       IsRtcOk());
   const auto& local_candidate_ep1 =
-      GetEndpoint(0)->saved_candidates_[0].candidate;
+      GetEndpoint(0)->saved_candidates()[0].candidate;
   // The IP address of ep1's host candidate should be obfuscated.
   EXPECT_TRUE(local_candidate_ep1.address().IsUnresolvedIP());
   // This is the underlying private IP address of the same candidate at ep1,
@@ -5782,9 +5594,10 @@ TEST_F(P2PTransportChannelTest, CanConnectWithHostCandidateWithMdnsName) {
 
   // We should be able to receive a ping from ep2 and establish a connection
   // with a peer reflexive candidate from ep2.
-  ASSERT_THAT(WaitUntil([&] { return ep1_ch1()->selected_connection(); },
-                        Ne(nullptr), {.timeout = kMediumTimeout}),
-              IsRtcOk());
+  ASSERT_THAT(
+      MediumWait().Until([&] { return ep1_ch1()->selected_connection(); },
+                         Ne(nullptr)),
+      IsRtcOk());
   EXPECT_TRUE(ep1_ch1()->selected_connection()->local_candidate().is_local());
   EXPECT_TRUE(ep1_ch1()->selected_connection()->remote_candidate().is_prflx());
 
@@ -5799,36 +5612,35 @@ TEST_F(P2PTransportChannelTest, CanConnectWithHostCandidateWithMdnsName) {
 // this remote host candidate in stats.
 TEST_F(P2PTransportChannelTest,
        CandidatesSanitizedInStatsWhenMdnsObfuscationEnabled) {
-  const Environment env = CreateEnvironment();
   ResolverFactoryFixture resolver_fixture;
 
   // ep1 and ep2 will gather host candidates with addresses
   // kPublicAddrs[0] and kPublicAddrs[1], respectively. ep1 also gathers a srflx
   // and a relay candidates.
-  ConfigureEndpoints(env, OPEN, OPEN,
+  ConfigureEndpoints(OPEN, OPEN,
                      kDefaultPortAllocatorFlags | PORTALLOCATOR_DISABLE_TCP,
                      kOnlyLocalPorts);
   // ICE parameter will be set up when creating the channels.
   set_remote_ice_parameter_source(FROM_SETICEPARAMETERS);
-  GetEndpoint(0)->network_manager_.set_mdns_responder(
+  GetEndpoint(0)->network_manager().set_mdns_responder(
       std::make_unique<FakeMdnsResponder>(Thread::Current()));
-  GetEndpoint(1)->async_dns_resolver_factory_ = &resolver_fixture;
-  CreateChannels(env);
+  GetEndpoint(1)->set_async_dns_resolver_factory(&resolver_fixture);
+  CreateChannels();
   // Pause sending candidates from both endpoints until we find out what port
   // number is assigned to ep1's host candidate.
   PauseCandidates(0);
   PauseCandidates(1);
   // Ep1 has a UDP host, a srflx and a relay candidates.
   ASSERT_THAT(
-      WaitUntil([&] { return GetEndpoint(0)->saved_candidates_.size(); },
-                Eq(3u), {.timeout = kMediumTimeout}),
+      MediumWait().Until(
+          [&] { return GetEndpoint(0)->saved_candidates().size(); }, Eq(3u)),
       IsRtcOk());
   ASSERT_THAT(
-      WaitUntil([&] { return GetEndpoint(1)->saved_candidates_.size(); },
-                Eq(1u), {.timeout = kMediumTimeout}),
+      MediumWait().Until(
+          [&] { return GetEndpoint(1)->saved_candidates().size(); }, Eq(1u)),
       IsRtcOk());
 
-  for (const auto& candidates_data : GetEndpoint(0)->saved_candidates_) {
+  for (const auto& candidates_data : GetEndpoint(0)->saved_candidates()) {
     const auto& local_candidate_ep1 = candidates_data.candidate;
     if (local_candidate_ep1.is_local()) {
       // This is the underlying private IP address of the same candidate at ep1,
@@ -5842,16 +5654,16 @@ TEST_F(P2PTransportChannelTest,
   ResumeCandidates(0);
   ResumeCandidates(1);
 
-  ASSERT_THAT(WaitUntil([&] { return ep1_ch1()->gathering_state(); },
-                        Eq(kIceGatheringComplete), {.timeout = kMediumTimeout}),
+  ASSERT_THAT(MediumWait().Until([&] { return ep1_ch1()->gathering_state(); },
+                                 Eq(kIceGatheringComplete)),
               IsRtcOk());
   // We should have the following candidate pairs on both endpoints:
   // ep1_host <-> ep2_host, ep1_srflx <-> ep2_host, ep1_relay <-> ep2_host
-  ASSERT_THAT(WaitUntil([&] { return ep1_ch1()->connections().size(); }, Eq(3u),
-                        {.timeout = kMediumTimeout}),
+  ASSERT_THAT(MediumWait().Until(
+                  [&] { return ep1_ch1()->connections().size(); }, Eq(3u)),
               IsRtcOk());
-  ASSERT_THAT(WaitUntil([&] { return ep2_ch1()->connections().size(); }, Eq(3u),
-                        {.timeout = kMediumTimeout}),
+  ASSERT_THAT(MediumWait().Until(
+                  [&] { return ep2_ch1()->connections().size(); }, Eq(3u)),
               IsRtcOk());
 
   IceTransportStats ice_transport_stats1;
@@ -5897,11 +5709,9 @@ TEST_F(P2PTransportChannelTest,
 
 TEST_F(P2PTransportChannelTest,
        ConnectingIncreasesSelectedCandidatePairChanges) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-  ConfigureEndpoints(env, OPEN, OPEN, kDefaultPortAllocatorFlags,
+  ConfigureEndpoints(OPEN, OPEN, kDefaultPortAllocatorFlags,
                      kDefaultPortAllocatorFlags);
-  CreateChannels(env);
+  CreateChannels();
 
   IceTransportStats ice_transport_stats;
   ASSERT_TRUE(ep1_ch1()->GetStats(&ice_transport_stats));
@@ -5909,8 +5719,8 @@ TEST_F(P2PTransportChannelTest,
 
   // Let the channels connect.
   EXPECT_THAT(
-      WaitUntil([&] { return ep1_ch1()->selected_connection(); }, Ne(nullptr),
-                {.timeout = kMediumTimeout, .clock = &clock}),
+      MediumWait().Until([&] { return ep1_ch1()->selected_connection(); },
+                         Ne(nullptr)),
       IsRtcOk());
 
   ASSERT_TRUE(ep1_ch1()->GetStats(&ice_transport_stats));
@@ -5921,11 +5731,9 @@ TEST_F(P2PTransportChannelTest,
 
 TEST_F(P2PTransportChannelTest,
        DisconnectedIncreasesSelectedCandidatePairChanges) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-  ConfigureEndpoints(env, OPEN, OPEN, kDefaultPortAllocatorFlags,
+  ConfigureEndpoints(OPEN, OPEN, kDefaultPortAllocatorFlags,
                      kDefaultPortAllocatorFlags);
-  CreateChannels(env);
+  CreateChannels();
 
   IceTransportStats ice_transport_stats;
   ASSERT_TRUE(ep1_ch1()->GetStats(&ice_transport_stats));
@@ -5933,8 +5741,8 @@ TEST_F(P2PTransportChannelTest,
 
   // Let the channels connect.
   EXPECT_THAT(
-      WaitUntil([&] { return ep1_ch1()->selected_connection(); }, Ne(nullptr),
-                {.timeout = kMediumTimeout, .clock = &clock}),
+      MediumWait().Until([&] { return ep1_ch1()->selected_connection(); },
+                         Ne(nullptr)),
       IsRtcOk());
 
   ASSERT_TRUE(ep1_ch1()->GetStats(&ice_transport_stats));
@@ -5945,8 +5753,8 @@ TEST_F(P2PTransportChannelTest,
     con->Prune();
   }
   EXPECT_THAT(
-      WaitUntil([&] { return ep1_ch1()->selected_connection(); }, Eq(nullptr),
-                {.timeout = kMediumTimeout, .clock = &clock}),
+      MediumWait().Until([&] { return ep1_ch1()->selected_connection(); },
+                         Eq(nullptr)),
       IsRtcOk());
 
   ASSERT_TRUE(ep1_ch1()->GetStats(&ice_transport_stats));
@@ -5957,11 +5765,9 @@ TEST_F(P2PTransportChannelTest,
 
 TEST_F(P2PTransportChannelTest,
        NewSelectionIncreasesSelectedCandidatePairChanges) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-  ConfigureEndpoints(env, OPEN, OPEN, kDefaultPortAllocatorFlags,
+  ConfigureEndpoints(OPEN, OPEN, kDefaultPortAllocatorFlags,
                      kDefaultPortAllocatorFlags);
-  CreateChannels(env);
+  CreateChannels();
 
   IceTransportStats ice_transport_stats;
   ASSERT_TRUE(ep1_ch1()->GetStats(&ice_transport_stats));
@@ -5969,8 +5775,8 @@ TEST_F(P2PTransportChannelTest,
 
   // Let the channels connect.
   EXPECT_THAT(
-      WaitUntil([&] { return ep1_ch1()->selected_connection(); }, Ne(nullptr),
-                {.timeout = kMediumTimeout, .clock = &clock}),
+      MediumWait().Until([&] { return ep1_ch1()->selected_connection(); },
+                         Ne(nullptr)),
       IsRtcOk());
 
   ASSERT_TRUE(ep1_ch1()->GetStats(&ice_transport_stats));
@@ -5984,14 +5790,11 @@ TEST_F(P2PTransportChannelTest,
       con->Prune();
     }
   }
-  EXPECT_TRUE(
-      WaitUntil(
-          [&] {
-            return ep1_ch1()->selected_connection() != nullptr &&
-                   (ep1_ch1()->GetStats(&ice_transport_stats),
-                    ice_transport_stats.selected_candidate_pair_changes >= 2u);
-          },
-          {.timeout = kMediumTimeout, .clock = &clock}));
+  EXPECT_TRUE(MediumWait().Until([&] {
+    return ep1_ch1()->selected_connection() != nullptr &&
+           (ep1_ch1()->GetStats(&ice_transport_stats),
+            ice_transport_stats.selected_candidate_pair_changes >= 2u);
+  }));
 
   ASSERT_TRUE(ep1_ch1()->GetStats(&ice_transport_stats));
   EXPECT_GE(ice_transport_stats.selected_candidate_pair_changes, 2u);
@@ -6003,27 +5806,26 @@ TEST_F(P2PTransportChannelTest,
 // when it is queried via GetSelectedCandidatePair.
 TEST_F(P2PTransportChannelTest,
        SelectedCandidatePairSanitizedWhenMdnsObfuscationEnabled) {
-  const Environment env = CreateEnvironment();
   ResolverFactoryFixture resolver_fixture;
 
   // ep1 and ep2 will gather host candidates with addresses
   // kPublicAddrs[0] and kPublicAddrs[1], respectively.
-  ConfigureEndpoints(env, OPEN, OPEN, kOnlyLocalPorts, kOnlyLocalPorts);
+  ConfigureEndpoints(OPEN, OPEN, kOnlyLocalPorts, kOnlyLocalPorts);
   // ICE parameter will be set up when creating the channels.
   set_remote_ice_parameter_source(FROM_SETICEPARAMETERS);
-  GetEndpoint(0)->network_manager_.set_mdns_responder(
+  GetEndpoint(0)->network_manager().set_mdns_responder(
       std::make_unique<FakeMdnsResponder>(Thread::Current()));
-  GetEndpoint(1)->async_dns_resolver_factory_ = &resolver_fixture;
-  CreateChannels(env);
+  GetEndpoint(1)->set_async_dns_resolver_factory(&resolver_fixture);
+  CreateChannels();
   // Pause sending candidates from both endpoints until we find out what port
   // number is assigned to ep1's host candidate.
   PauseCandidates(0);
   PauseCandidates(1);
   ASSERT_THAT(
-      WaitUntil([&] { return GetEndpoint(0)->saved_candidates_.size(); },
-                Eq(1u), {.timeout = kMediumTimeout}),
+      MediumWait().Until(
+          [&] { return GetEndpoint(0)->saved_candidates().size(); }, Eq(1u)),
       IsRtcOk());
-  const auto& candidates_data = GetEndpoint(0)->saved_candidates_[0];
+  const auto& candidates_data = GetEndpoint(0)->saved_candidates()[0];
   const auto& local_candidate_ep1 = candidates_data.candidate;
   ASSERT_TRUE(local_candidate_ep1.is_local());
   // This is the underlying private IP address of the same candidate at ep1,
@@ -6035,12 +5837,10 @@ TEST_F(P2PTransportChannelTest,
   ResumeCandidates(0);
   ResumeCandidates(1);
 
-  ASSERT_TRUE(WaitUntil(
-      [&] {
-        return ep1_ch1()->selected_connection() != nullptr &&
-               ep2_ch1()->selected_connection() != nullptr;
-      },
-      {.timeout = kMediumTimeout}));
+  ASSERT_TRUE(MediumWait().Until([&] {
+    return ep1_ch1()->selected_connection() != nullptr &&
+           ep2_ch1()->selected_connection() != nullptr;
+  }));
 
   const auto pair_ep1 = ep1_ch1()->GetSelectedCandidatePair();
   ASSERT_TRUE(pair_ep1.has_value());
@@ -6057,23 +5857,20 @@ TEST_F(P2PTransportChannelTest,
 
 TEST_F(P2PTransportChannelTest,
        NoPairOfLocalRelayCandidateWithRemoteMdnsCandidate) {
-  const Environment env = CreateEnvironment();
   const int kOnlyRelayPorts = PORTALLOCATOR_DISABLE_UDP |
                               PORTALLOCATOR_DISABLE_STUN |
                               PORTALLOCATOR_DISABLE_TCP;
   // We use one endpoint to test the behavior of adding remote candidates, and
   // this endpoint only gathers relay candidates.
-  ConfigureEndpoints(env, OPEN, OPEN, kOnlyRelayPorts,
-                     kDefaultPortAllocatorFlags);
-  GetEndpoint(0)->cd1_.ch_ = CreateChannel(
-      env, 0, ICE_CANDIDATE_COMPONENT_DEFAULT, kIceParams[0], kIceParams[1]);
+  ConfigureEndpoints(OPEN, OPEN, kOnlyRelayPorts, kDefaultPortAllocatorFlags);
+  GetEndpoint(0)->cd1().set_ch(CreateChannel(0, ICE_CANDIDATE_COMPONENT_DEFAULT,
+                                             kIceParams[0], kIceParams[1]));
   IceConfig config;
   // Start gathering and we should have only a single relay port.
   ep1_ch1()->SetIceConfig(config);
   ep1_ch1()->MaybeStartGathering();
-  EXPECT_THAT(WaitUntil([&] { return ep1_ch1()->gathering_state(); },
-                        Eq(IceGatheringState::kIceGatheringComplete),
-                        {.timeout = kDefaultTimeout}),
+  EXPECT_THAT(MediumWait().Until([&] { return ep1_ch1()->gathering_state(); },
+                                 Eq(IceGatheringState::kIceGatheringComplete)),
               IsRtcOk());
   EXPECT_EQ(1u, ep1_ch1()->ports().size());
   // Add a plain remote host candidate and three remote mDNS candidates with the
@@ -6130,7 +5927,6 @@ class MockMdnsResponder : public MdnsResponderInterface {
 
 TEST_F(P2PTransportChannelTest,
        SrflxCandidateCanBeGatheredBeforeMdnsCandidateToCreateConnection) {
-  const Environment env = CreateEnvironment();
   // ep1 and ep2 will only gather host and srflx candidates with base addresses
   // kPublicAddrs[0] and kPublicAddrs[1], respectively, and we use a shared
   // socket in gathering.
@@ -6138,7 +5934,7 @@ TEST_F(P2PTransportChannelTest,
                                       PORTALLOCATOR_DISABLE_TCP |
                                       PORTALLOCATOR_ENABLE_SHARED_SOCKET;
   // ep1 is configured with a NAT so that we do gather a srflx candidate.
-  ConfigureEndpoints(env, NAT_FULL_CONE, OPEN, kOnlyLocalAndStunPorts,
+  ConfigureEndpoints(NAT_FULL_CONE, OPEN, kOnlyLocalAndStunPorts,
                      kOnlyLocalAndStunPorts);
   // ICE parameter will be set up when creating the channels.
   set_remote_ice_parameter_source(FROM_SETICEPARAMETERS);
@@ -6148,15 +5944,16 @@ TEST_F(P2PTransportChannelTest,
   EXPECT_CALL(*mock_mdns_responder, CreateNameForAddress(_, _))
       .Times(1)
       .WillOnce(Return());
-  GetEndpoint(0)->network_manager_.set_mdns_responder(
+  GetEndpoint(0)->network_manager().set_mdns_responder(
       std::move(mock_mdns_responder));
 
-  CreateChannels(env);
+  CreateChannels();
 
   // We should be able to form a srflx-host connection to ep2.
-  ASSERT_THAT(WaitUntil([&] { return ep1_ch1()->selected_connection(); },
-                        Ne(nullptr), {.timeout = kMediumTimeout}),
-              IsRtcOk());
+  ASSERT_THAT(
+      DefaultWait().Until([&] { return ep1_ch1()->selected_connection(); },
+                          Ne(nullptr)),
+      IsRtcOk());
   EXPECT_TRUE(ep1_ch1()->selected_connection()->local_candidate().is_stun());
   EXPECT_TRUE(ep1_ch1()->selected_connection()->remote_candidate().is_local());
 
@@ -6170,52 +5967,45 @@ TEST_F(P2PTransportChannelTest,
 // removed and are still usable for necessary route switching.
 TEST_F(P2PTransportChannelTest,
        SurfaceHostCandidateOnCandidateFilterChangeFromRelayToAll) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-
   ConfigureEndpoints(
-      env, OPEN, OPEN,
+      OPEN, OPEN,
       kDefaultPortAllocatorFlags | PORTALLOCATOR_ENABLE_SHARED_SOCKET,
       kDefaultPortAllocatorFlags | PORTALLOCATOR_ENABLE_SHARED_SOCKET);
   auto* ep1 = GetEndpoint(0);
   auto* ep2 = GetEndpoint(1);
-  ep1->allocator_->SetCandidateFilter(CF_RELAY);
-  ep2->allocator_->SetCandidateFilter(CF_RELAY);
+  ep1->allocator()->SetCandidateFilter(CF_RELAY);
+  ep2->allocator()->SetCandidateFilter(CF_RELAY);
   // Enable continual gathering and also resurfacing gathered candidates upon
   // the candidate filter changed in the ICE configuration.
   IceConfig ice_config =
       CreateIceConfig(TimeDelta::Seconds(1), GATHER_CONTINUALLY);
   ice_config.surface_ice_candidates_on_ice_transport_type_changed = true;
-  CreateChannels(env, ice_config, ice_config);
+  CreateChannels(ice_config, ice_config);
   ASSERT_THAT(
-      WaitUntil([&] { return ep1_ch1()->selected_connection(); }, Ne(nullptr),
-                {.timeout = kDefaultTimeout, .clock = &clock}),
+      MediumWait().Until([&] { return ep1_ch1()->selected_connection(); },
+                         Ne(nullptr)),
       IsRtcOk());
   ASSERT_THAT(
-      WaitUntil([&] { return ep2_ch1()->selected_connection(); }, Ne(nullptr),
-                {.timeout = kDefaultTimeout, .clock = &clock}),
+      DefaultWait().Until([&] { return ep2_ch1()->selected_connection(); },
+                          Ne(nullptr)),
       IsRtcOk());
   EXPECT_TRUE(ep1_ch1()->selected_connection()->local_candidate().is_relay());
   EXPECT_TRUE(ep2_ch1()->selected_connection()->local_candidate().is_relay());
 
   // Loosen the candidate filter at ep1.
-  ep1->allocator_->SetCandidateFilter(CF_ALL);
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return ep1_ch1()->selected_connection() != nullptr &&
-               ep1_ch1()->selected_connection()->local_candidate().is_local();
-      },
-      {.timeout = kDefaultTimeout, .clock = &clock}));
+  ep1->allocator()->SetCandidateFilter(CF_ALL);
+  EXPECT_TRUE(DefaultWait().Until([&] {
+    return ep1_ch1()->selected_connection() != nullptr &&
+           ep1_ch1()->selected_connection()->local_candidate().is_local();
+  }));
   EXPECT_TRUE(ep1_ch1()->selected_connection()->remote_candidate().is_relay());
 
   // Loosen the candidate filter at ep2.
-  ep2->allocator_->SetCandidateFilter(CF_ALL);
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return ep2_ch1()->selected_connection() != nullptr &&
-               ep2_ch1()->selected_connection()->local_candidate().is_local();
-      },
-      {.timeout = kDefaultTimeout, .clock = &clock}));
+  ep2->allocator()->SetCandidateFilter(CF_ALL);
+  EXPECT_TRUE(DefaultWait().Until([&] {
+    return ep2_ch1()->selected_connection() != nullptr &&
+           ep2_ch1()->selected_connection()->local_candidate().is_local();
+  }));
   // We have migrated to a host-host candidate pair.
   EXPECT_TRUE(ep2_ch1()->selected_connection()->remote_candidate().is_local());
 
@@ -6226,11 +6016,9 @@ TEST_F(P2PTransportChannelTest,
   fw()->AddRule(false, FP_ANY, kPublicAddrs[1], kTurnUdpExtAddr);
 
   // We should be able to reuse the previously gathered relay candidates.
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return ep1_ch1()->selected_connection()->local_candidate().is_relay();
-      },
-      {.timeout = kDefaultTimeout, .clock = &clock}));
+  EXPECT_TRUE(DefaultWait().Until([&] {
+    return ep1_ch1()->selected_connection()->local_candidate().is_relay();
+  }));
   EXPECT_TRUE(ep1_ch1()->selected_connection()->remote_candidate().is_relay());
   DestroyChannels();
 }
@@ -6240,8 +6028,6 @@ TEST_F(P2PTransportChannelTest,
 // changing the candidate filter.
 TEST_F(P2PTransportChannelTest,
        SurfaceSrflxCandidateOnCandidateFilterChangeFromRelayToNoHost) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
   // We need an actual NAT so that the host candidate is not equivalent to the
   // srflx candidate; otherwise, the host candidate would still surface even
   // though we disable it via the candidate filter below. This is a result of
@@ -6250,46 +6036,42 @@ TEST_F(P2PTransportChannelTest,
   //  2. We keep the host candidate in this case in CheckCandidateFilter even
   //     though we intend to filter them.
   ConfigureEndpoints(
-      env, NAT_FULL_CONE, NAT_FULL_CONE,
+      NAT_FULL_CONE, NAT_FULL_CONE,
       kDefaultPortAllocatorFlags | PORTALLOCATOR_ENABLE_SHARED_SOCKET,
       kDefaultPortAllocatorFlags | PORTALLOCATOR_ENABLE_SHARED_SOCKET);
   auto* ep1 = GetEndpoint(0);
   auto* ep2 = GetEndpoint(1);
-  ep1->allocator_->SetCandidateFilter(CF_RELAY);
-  ep2->allocator_->SetCandidateFilter(CF_RELAY);
+  ep1->allocator()->SetCandidateFilter(CF_RELAY);
+  ep2->allocator()->SetCandidateFilter(CF_RELAY);
   // Enable continual gathering and also resurfacing gathered candidates upon
   // the candidate filter changed in the ICE configuration.
   IceConfig ice_config =
       CreateIceConfig(TimeDelta::Seconds(1), GATHER_CONTINUALLY);
   ice_config.surface_ice_candidates_on_ice_transport_type_changed = true;
-  CreateChannels(env, ice_config, ice_config);
+  CreateChannels(ice_config, ice_config);
   ASSERT_THAT(
-      WaitUntil([&] { return ep1_ch1()->selected_connection(); }, Ne(nullptr),
-                {.timeout = kDefaultTimeout, .clock = &clock}),
+      DefaultWait().Until([&] { return ep1_ch1()->selected_connection(); },
+                          Ne(nullptr)),
       IsRtcOk());
   ASSERT_THAT(
-      WaitUntil([&] { return ep2_ch1()->selected_connection(); }, Ne(nullptr),
-                {.timeout = kDefaultTimeout, .clock = &clock}),
+      DefaultWait().Until([&] { return ep2_ch1()->selected_connection(); },
+                          Ne(nullptr)),
       IsRtcOk());
   const uint32_t kCandidateFilterNoHost = CF_ALL & ~CF_HOST;
   // Loosen the candidate filter at ep1.
-  ep1->allocator_->SetCandidateFilter(kCandidateFilterNoHost);
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return ep1_ch1()->selected_connection() != nullptr &&
-               ep1_ch1()->selected_connection()->local_candidate().is_stun();
-      },
-      {.timeout = kDefaultTimeout, .clock = &clock}));
+  ep1->allocator()->SetCandidateFilter(kCandidateFilterNoHost);
+  EXPECT_TRUE(DefaultWait().Until([&] {
+    return ep1_ch1()->selected_connection() != nullptr &&
+           ep1_ch1()->selected_connection()->local_candidate().is_stun();
+  }));
   EXPECT_TRUE(ep1_ch1()->selected_connection()->remote_candidate().is_relay());
 
   // Loosen the candidate filter at ep2.
-  ep2->allocator_->SetCandidateFilter(kCandidateFilterNoHost);
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return ep2_ch1()->selected_connection() != nullptr &&
-               ep2_ch1()->selected_connection()->local_candidate().is_stun();
-      },
-      {.timeout = kDefaultTimeout, .clock = &clock}));
+  ep2->allocator()->SetCandidateFilter(kCandidateFilterNoHost);
+  EXPECT_TRUE(DefaultWait().Until([&] {
+    return ep2_ch1()->selected_connection() != nullptr &&
+           ep2_ch1()->selected_connection()->local_candidate().is_stun();
+  }));
   // We have migrated to a srflx-srflx candidate pair.
   EXPECT_TRUE(ep2_ch1()->selected_connection()->remote_candidate().is_stun());
 
@@ -6299,11 +6081,9 @@ TEST_F(P2PTransportChannelTest,
   fw()->AddRule(false, FP_ANY, kPrivateAddrs[0], kTurnUdpExtAddr);
   fw()->AddRule(false, FP_ANY, kPrivateAddrs[1], kTurnUdpExtAddr);
   // We should be able to reuse the previously gathered relay candidates.
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return ep1_ch1()->selected_connection()->local_candidate().is_relay();
-      },
-      {.timeout = kDefaultTimeout, .clock = &clock}));
+  EXPECT_TRUE(DefaultWait().Until([&] {
+    return ep1_ch1()->selected_connection()->local_candidate().is_relay();
+  }));
   EXPECT_TRUE(ep1_ch1()->selected_connection()->remote_candidate().is_relay());
   DestroyChannels();
 }
@@ -6315,37 +6095,34 @@ TEST_F(P2PTransportChannelTest,
 // gathering stopped.
 TEST_F(P2PTransportChannelTest,
        CannotSurfaceTheNewlyAllowedOnFilterChangeIfNotGatheringContinually) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-
   ConfigureEndpoints(
-      env, OPEN, OPEN,
+      OPEN, OPEN,
       kDefaultPortAllocatorFlags | PORTALLOCATOR_ENABLE_SHARED_SOCKET,
       kDefaultPortAllocatorFlags | PORTALLOCATOR_ENABLE_SHARED_SOCKET);
   auto* ep1 = GetEndpoint(0);
   auto* ep2 = GetEndpoint(1);
-  ep1->allocator_->SetCandidateFilter(CF_RELAY);
-  ep2->allocator_->SetCandidateFilter(CF_RELAY);
+  ep1->allocator()->SetCandidateFilter(CF_RELAY);
+  ep2->allocator()->SetCandidateFilter(CF_RELAY);
   // Only gather once.
   IceConfig ice_config = CreateIceConfig(TimeDelta::Seconds(1), GATHER_ONCE);
   ice_config.surface_ice_candidates_on_ice_transport_type_changed = true;
-  CreateChannels(env, ice_config, ice_config);
+  CreateChannels(ice_config, ice_config);
   ASSERT_THAT(
-      WaitUntil([&] { return ep1_ch1()->selected_connection(); }, Ne(nullptr),
-                {.timeout = kDefaultTimeout, .clock = &clock}),
+      DefaultWait().Until([&] { return ep1_ch1()->selected_connection(); },
+                          Ne(nullptr)),
       IsRtcOk());
   ASSERT_THAT(
-      WaitUntil([&] { return ep2_ch1()->selected_connection(); }, Ne(nullptr),
-                {.timeout = kDefaultTimeout, .clock = &clock}),
+      DefaultWait().Until([&] { return ep2_ch1()->selected_connection(); },
+                          Ne(nullptr)),
       IsRtcOk());
   // Loosen the candidate filter at ep1.
-  ep1->allocator_->SetCandidateFilter(CF_ALL);
+  ep1->allocator()->SetCandidateFilter(CF_ALL);
   // Wait for a period for any potential surfacing of new candidates.
-  SIMULATED_WAIT(false, kDefaultTimeout.ms(), clock);
+  time_controller_.AdvanceTime(kDefaultTimeout);
   EXPECT_TRUE(ep1_ch1()->selected_connection()->local_candidate().is_relay());
 
   // Loosen the candidate filter at ep2.
-  ep2->allocator_->SetCandidateFilter(CF_ALL);
+  ep2->allocator()->SetCandidateFilter(CF_ALL);
   EXPECT_TRUE(ep2_ch1()->selected_connection()->local_candidate().is_relay());
   DestroyChannels();
 }
@@ -6355,17 +6132,14 @@ TEST_F(P2PTransportChannelTest,
 // match the filter, are not removed.
 TEST_F(P2PTransportChannelTest,
        RestrictingCandidateFilterDoesNotRemoveRegatheredCandidates) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
-
   ConfigureEndpoints(
-      env, OPEN, OPEN,
+      OPEN, OPEN,
       kDefaultPortAllocatorFlags | PORTALLOCATOR_ENABLE_SHARED_SOCKET,
       kDefaultPortAllocatorFlags | PORTALLOCATOR_ENABLE_SHARED_SOCKET);
   auto* ep1 = GetEndpoint(0);
   auto* ep2 = GetEndpoint(1);
-  ep1->allocator_->SetCandidateFilter(CF_ALL);
-  ep2->allocator_->SetCandidateFilter(CF_ALL);
+  ep1->allocator()->SetCandidateFilter(CF_ALL);
+  ep2->allocator()->SetCandidateFilter(CF_ALL);
   // Enable continual gathering and also resurfacing gathered candidates upon
   // the candidate filter changed in the ICE configuration.
   IceConfig ice_config =
@@ -6376,32 +6150,23 @@ TEST_F(P2PTransportChannelTest,
   // gathering when we have a strongly connected candidate pair.
   PauseCandidates(0);
   PauseCandidates(1);
-  CreateChannels(env, ice_config, ice_config);
+  CreateChannels(ice_config, ice_config);
 
   // We have gathered host, srflx and relay candidates.
-  EXPECT_THAT(WaitUntil([&] { return ep1->saved_candidates_.size(); }, Eq(3u),
-                        {.timeout = kDefaultTimeout, .clock = &clock}),
+  EXPECT_THAT(DefaultWait().Until(
+                  [&] { return ep1->saved_candidates().size(); }, Eq(3u)),
               IsRtcOk());
   ResumeCandidates(0);
   ResumeCandidates(1);
-  ASSERT_TRUE(
-      WaitUntil(
-          [&] {
-            return ep1_ch1()->selected_connection() != nullptr &&
-                   ep1_ch1()
-                       ->selected_connection()
-                       ->local_candidate()
-                       .is_local() &&
-                   ep2_ch1()->selected_connection() != nullptr &&
-                   ep1_ch1()
-                       ->selected_connection()
-                       ->remote_candidate()
-                       .is_local();
-          },
-          {.timeout = kDefaultTimeout, .clock = &clock}));
+  ASSERT_TRUE(DefaultWait().Until([&] {
+    return ep1_ch1()->selected_connection() != nullptr &&
+           ep1_ch1()->selected_connection()->local_candidate().is_local() &&
+           ep2_ch1()->selected_connection() != nullptr &&
+           ep1_ch1()->selected_connection()->remote_candidate().is_local();
+  }));
   ASSERT_THAT(
-      WaitUntil([&] { return ep2_ch1()->selected_connection(); }, Ne(nullptr),
-                {.timeout = kDefaultTimeout, .clock = &clock}),
+      DefaultWait().Until([&] { return ep2_ch1()->selected_connection(); },
+                          Ne(nullptr)),
       IsRtcOk());
   // Test that we have a host-host candidate pair selected and the number of
   // candidates signaled to the remote peer stays the same.
@@ -6415,16 +6180,16 @@ TEST_F(P2PTransportChannelTest,
   test_invariants();
 
   // Set a more restrictive candidate filter at ep1.
-  ep1->allocator_->SetCandidateFilter(CF_HOST | CF_REFLEXIVE);
-  SIMULATED_WAIT(false, kDefaultTimeout.ms(), clock);
+  ep1->allocator()->SetCandidateFilter(CF_HOST | CF_REFLEXIVE);
+  time_controller_.AdvanceTime(kDefaultTimeout);
   test_invariants();
 
-  ep1->allocator_->SetCandidateFilter(CF_HOST);
-  SIMULATED_WAIT(false, kDefaultTimeout.ms(), clock);
+  ep1->allocator()->SetCandidateFilter(CF_HOST);
+  time_controller_.AdvanceTime(kDefaultTimeout);
   test_invariants();
 
-  ep1->allocator_->SetCandidateFilter(CF_NONE);
-  SIMULATED_WAIT(false, kDefaultTimeout.ms(), clock);
+  ep1->allocator()->SetCandidateFilter(CF_NONE);
+  time_controller_.AdvanceTime(kDefaultTimeout);
   test_invariants();
   DestroyChannels();
 }
@@ -6436,18 +6201,19 @@ TEST_F(P2PTransportChannelTest,
 // i.e surface_ice_candidates_on_ice_transport_type_changed requires
 // coordination outside of webrtc to function properly.
 TEST_F(P2PTransportChannelTest, SurfaceRequiresCoordination) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment(CreateTestFieldTrialsPtr(
-      "WebRTC-IceFieldTrials/skip_relay_to_non_relay_connections:true/"));
+  env_ = CreateTestEnvironment(
+      {.field_trials = CreateTestFieldTrialsPtr(
+           "WebRTC-IceFieldTrials/skip_relay_to_non_relay_connections:true/"),
+       .time = &time_controller_});
 
   ConfigureEndpoints(
-      env, OPEN, OPEN,
+      OPEN, OPEN,
       kDefaultPortAllocatorFlags | PORTALLOCATOR_ENABLE_SHARED_SOCKET,
       kDefaultPortAllocatorFlags | PORTALLOCATOR_ENABLE_SHARED_SOCKET);
   auto* ep1 = GetEndpoint(0);
   auto* ep2 = GetEndpoint(1);
-  ep1->allocator_->SetCandidateFilter(CF_RELAY);
-  ep2->allocator_->SetCandidateFilter(CF_ALL);
+  ep1->allocator()->SetCandidateFilter(CF_RELAY);
+  ep2->allocator()->SetCandidateFilter(CF_ALL);
   // Enable continual gathering and also resurfacing gathered candidates upon
   // the candidate filter changed in the ICE configuration.
   IceConfig ice_config =
@@ -6458,47 +6224,38 @@ TEST_F(P2PTransportChannelTest, SurfaceRequiresCoordination) {
   // gathering when we have a strongly connected candidate pair.
   PauseCandidates(0);
   PauseCandidates(1);
-  CreateChannels(env, ice_config, ice_config);
+  CreateChannels(ice_config, ice_config);
 
   // On the caller we only have relay,
   // on the callee we have host, srflx and relay.
-  EXPECT_THAT(WaitUntil([&] { return ep1->saved_candidates_.size(); }, Eq(1u),
-                        {.timeout = kDefaultTimeout, .clock = &clock}),
+  EXPECT_THAT(DefaultWait().Until(
+                  [&] { return ep1->saved_candidates().size(); }, Eq(1u)),
               IsRtcOk());
-  EXPECT_THAT(WaitUntil([&] { return ep2->saved_candidates_.size(); }, Eq(3u),
-                        {.timeout = kDefaultTimeout, .clock = &clock}),
+  EXPECT_THAT(DefaultWait().Until(
+                  [&] { return ep2->saved_candidates().size(); }, Eq(3u)),
               IsRtcOk());
 
   ResumeCandidates(0);
   ResumeCandidates(1);
-  ASSERT_TRUE(
-      WaitUntil(
-          [&] {
-            return ep1_ch1()->selected_connection() != nullptr &&
-                   ep1_ch1()
-                       ->selected_connection()
-                       ->local_candidate()
-                       .is_relay() &&
-                   ep2_ch1()->selected_connection() != nullptr &&
-                   ep1_ch1()
-                       ->selected_connection()
-                       ->remote_candidate()
-                       .is_relay();
-          },
-          {.timeout = kDefaultTimeout, .clock = &clock}));
+  ASSERT_TRUE(DefaultWait().Until([&] {
+    return ep1_ch1()->selected_connection() != nullptr &&
+           ep1_ch1()->selected_connection()->local_candidate().is_relay() &&
+           ep2_ch1()->selected_connection() != nullptr &&
+           ep1_ch1()->selected_connection()->remote_candidate().is_relay();
+  }));
   ASSERT_THAT(
-      WaitUntil([&] { return ep2_ch1()->selected_connection(); }, Ne(nullptr),
-                {.timeout = kDefaultTimeout, .clock = &clock}),
+      DefaultWait().Until([&] { return ep2_ch1()->selected_connection(); },
+                          Ne(nullptr)),
       IsRtcOk());
 
   // Wait until the callee discards it's candidates
   // since they don't manage to connect.
-  SIMULATED_WAIT(false, 300000, clock);
+  time_controller_.AdvanceTime(TimeDelta::Millis(300000));
 
   // And then loosen caller candidate filter.
-  ep1->allocator_->SetCandidateFilter(CF_ALL);
+  ep1->allocator()->SetCandidateFilter(CF_ALL);
 
-  SIMULATED_WAIT(false, kDefaultTimeout.ms(), clock);
+  time_controller_.AdvanceTime(kDefaultTimeout);
 
   // No p2p connection will be made, it will remain on relay.
   EXPECT_TRUE(ep1_ch1()->selected_connection() != nullptr &&
@@ -6510,120 +6267,116 @@ TEST_F(P2PTransportChannelTest, SurfaceRequiresCoordination) {
 }
 
 TEST_F(P2PTransportChannelPingTest, TestInitialSelectDampening0) {
-  constexpr int kMargin = 10;
-  ScopedFakeClock clock;
-  clock.AdvanceTime(TimeDelta::Seconds(1));
-  const Environment env = CreateEnvironment(CreateTestFieldTrialsPtr(
-      "WebRTC-IceFieldTrials/initial_select_dampening:0/"));
+  time_controller_.AdvanceTime(TimeDelta::Seconds(1));
+  env_ = CreateTestEnvironment(
+      {.field_trials = CreateTestFieldTrialsPtr(
+           "WebRTC-IceFieldTrials/initial_select_dampening:0/"),
+       .time = &time_controller_});
 
-  FakePortAllocator pa(env, ss());
-  P2PTransportChannel ch(env, "test channel", 1, &pa);
+  FakePortAllocator pa(env_, ss());
+  P2PTransportChannel ch(env_, "test channel", 1, &pa);
   PrepareChannel(&ch);
   ch.SetIceConfig(ch.config());
   ch.MaybeStartGathering();
 
   ch.AddRemoteCandidate(
       CreateUdpCandidate(IceCandidateType::kHost, "1.1.1.1", 1, 100));
-  Connection* conn1 = WaitForConnectionTo(&ch, "1.1.1.1", 1, &clock);
+  Connection* conn1 = WaitForConnectionTo(&ch, "1.1.1.1", 1);
   ASSERT_TRUE(conn1 != nullptr);
   EXPECT_EQ(nullptr, ch.selected_connection());
   conn1->ReceivedPingResponse(kLowRtt, "id");  // Becomes writable and receiving
   // It shall not be selected until 0ms has passed....i.e it should be connected
   // directly.
   EXPECT_THAT(
-      WaitUntil([&] { return ch.selected_connection(); }, Eq(conn1),
-                {.timeout = TimeDelta::Millis(kMargin), .clock = &clock}),
+      DefaultWait().Until([&] { return ch.selected_connection(); }, Eq(conn1)),
       IsRtcOk());
 }
 
 TEST_F(P2PTransportChannelPingTest, TestInitialSelectDampening) {
-  constexpr int kMargin = 10;
-  ScopedFakeClock clock;
-  clock.AdvanceTime(TimeDelta::Seconds(1));
-  const Environment env = CreateEnvironment(CreateTestFieldTrialsPtr(
-      "WebRTC-IceFieldTrials/initial_select_dampening:100/"));
+  time_controller_.AdvanceTime(TimeDelta::Seconds(1));
+  env_ = CreateTestEnvironment(
+      {.field_trials = CreateTestFieldTrialsPtr(
+           "WebRTC-IceFieldTrials/initial_select_dampening:100/"),
+       .time = &time_controller_});
 
-  FakePortAllocator pa(env, ss());
-  P2PTransportChannel ch(env, "test channel", 1, &pa);
+  FakePortAllocator pa(env_, ss());
+  P2PTransportChannel ch(env_, "test channel", 1, &pa);
   PrepareChannel(&ch);
   ch.SetIceConfig(ch.config());
   ch.MaybeStartGathering();
 
   ch.AddRemoteCandidate(
       CreateUdpCandidate(IceCandidateType::kHost, "1.1.1.1", 1, 100));
-  Connection* conn1 = WaitForConnectionTo(&ch, "1.1.1.1", 1, &clock);
+  Connection* conn1 = WaitForConnectionTo(&ch, "1.1.1.1", 1);
   ASSERT_TRUE(conn1 != nullptr);
   EXPECT_EQ(nullptr, ch.selected_connection());
   conn1->ReceivedPingResponse(kLowRtt, "id");  // Becomes writable and receiving
   // It shall not be selected until 100ms has passed.
-  SIMULATED_WAIT(conn1 == ch.selected_connection(), 100 - kMargin, clock);
+  (void)DefaultWait().Until([&] { return conn1 == ch.selected_connection(); });
   EXPECT_THAT(
-      WaitUntil([&] { return ch.selected_connection(); }, Eq(conn1),
-                {.timeout = TimeDelta::Millis(2 * kMargin), .clock = &clock}),
+      DefaultWait().Until([&] { return ch.selected_connection(); }, Eq(conn1)),
       IsRtcOk());
 }
 
 TEST_F(P2PTransportChannelPingTest, TestInitialSelectDampeningPingReceived) {
-  constexpr int kMargin = 10;
-  ScopedFakeClock clock;
-  clock.AdvanceTime(TimeDelta::Seconds(1));
-  const Environment env = CreateEnvironment(CreateTestFieldTrialsPtr(
-      "WebRTC-IceFieldTrials/initial_select_dampening_ping_received:100/"));
+  time_controller_.AdvanceTime(TimeDelta::Seconds(1));
+  env_ = CreateTestEnvironment(
+      {.field_trials = CreateTestFieldTrialsPtr(
+           "WebRTC-IceFieldTrials/initial_select_dampening_ping_received:100/"),
+       .time = &time_controller_});
 
-  FakePortAllocator pa(env, ss());
-  P2PTransportChannel ch(env, "test channel", 1, &pa);
+  FakePortAllocator pa(env_, ss());
+  P2PTransportChannel ch(env_, "test channel", 1, &pa);
   PrepareChannel(&ch);
   ch.SetIceConfig(ch.config());
   ch.MaybeStartGathering();
 
   ch.AddRemoteCandidate(
       CreateUdpCandidate(IceCandidateType::kHost, "1.1.1.1", 1, 100));
-  Connection* conn1 = WaitForConnectionTo(&ch, "1.1.1.1", 1, &clock);
+  Connection* conn1 = WaitForConnectionTo(&ch, "1.1.1.1", 1);
   ASSERT_TRUE(conn1 != nullptr);
   EXPECT_EQ(nullptr, ch.selected_connection());
   conn1->ReceivedPingResponse(kLowRtt, "id");  // Becomes writable and receiving
   conn1->ReceivedPing("id1");                  //
   // It shall not be selected until 100ms has passed.
-  SIMULATED_WAIT(conn1 == ch.selected_connection(), 100 - kMargin, clock);
+  (void)DefaultWait().Until([&] { return conn1 == ch.selected_connection(); });
   EXPECT_THAT(
-      WaitUntil([&] { return ch.selected_connection(); }, Eq(conn1),
-                {.timeout = TimeDelta::Millis(2 * kMargin), .clock = &clock}),
+      DefaultWait().Until([&] { return ch.selected_connection(); }, Eq(conn1)),
       IsRtcOk());
 }
 
 TEST_F(P2PTransportChannelPingTest, TestInitialSelectDampeningBoth) {
-  constexpr int kMargin = 10;
-  ScopedFakeClock clock;
-  clock.AdvanceTime(TimeDelta::Seconds(1));
-  const Environment env = CreateEnvironment(CreateTestFieldTrialsPtr(
-      "WebRTC-IceFieldTrials/"
-      "initial_select_dampening:100,initial_select_dampening_ping_received:"
-      "50/"));
+  time_controller_.AdvanceTime(TimeDelta::Seconds(1));
+  env_ = CreateTestEnvironment({.field_trials = CreateTestFieldTrialsPtr(
+                                    "WebRTC-IceFieldTrials/"
+                                    "initial_select_dampening:100,initial_"
+                                    "select_dampening_ping_received:"
+                                    "50/"),
+                                .time = &time_controller_});
 
-  FakePortAllocator pa(env, ss());
-  P2PTransportChannel ch(env, "test channel", 1, &pa);
+  FakePortAllocator pa(env_, ss());
+  P2PTransportChannel ch(env_, "test channel", 1, &pa);
   PrepareChannel(&ch);
   ch.SetIceConfig(ch.config());
   ch.MaybeStartGathering();
 
   ch.AddRemoteCandidate(
       CreateUdpCandidate(IceCandidateType::kHost, "1.1.1.1", 1, 100));
-  Connection* conn1 = WaitForConnectionTo(&ch, "1.1.1.1", 1, &clock);
+  Connection* conn1 = WaitForConnectionTo(&ch, "1.1.1.1", 1);
   ASSERT_TRUE(conn1 != nullptr);
   EXPECT_EQ(nullptr, ch.selected_connection());
   conn1->ReceivedPingResponse(kLowRtt, "id");  // Becomes writable and receiving
   // It shall not be selected until 100ms has passed....but only wait ~50 now.
-  SIMULATED_WAIT(conn1 == ch.selected_connection(), 50 - kMargin, clock);
+  (void)DefaultWait().Until([&] { return conn1 == ch.selected_connection(); });
   // Now receiving ping and new timeout should kick in.
   conn1->ReceivedPing("id1");  //
   EXPECT_THAT(
-      WaitUntil([&] { return ch.selected_connection(); }, Eq(conn1),
-                {.timeout = TimeDelta::Millis(2 * kMargin), .clock = &clock}),
+      DefaultWait().Until([&] { return ch.selected_connection(); }, Eq(conn1)),
       IsRtcOk());
 }
 
 TEST(P2PTransportChannelIceControllerTest, InjectIceController) {
-  const Environment env = CreateEnvironment();
+  const Environment env = CreateTestEnvironment();
   std::unique_ptr<SocketServer> socket_server = CreateDefaultSocketServer();
   AutoSocketServerThread main_thread(socket_server.get());
   MockIceControllerFactory factory;
@@ -6638,7 +6391,7 @@ TEST(P2PTransportChannelIceControllerTest, InjectIceController) {
 }
 
 TEST(P2PTransportChannel, InjectActiveIceController) {
-  const Environment env = CreateEnvironment();
+  const Environment env = CreateTestEnvironment();
   std::unique_ptr<SocketServer> socket_server = CreateDefaultSocketServer();
   AutoSocketServerThread main_thread(socket_server.get());
   MockActiveIceControllerFactory factory;
@@ -6693,10 +6446,9 @@ class ForgetLearnedStateControllerFactory
 };
 
 TEST_F(P2PTransportChannelPingTest, TestForgetLearnedState) {
-  const Environment env = CreateEnvironment();
   ForgetLearnedStateControllerFactory factory;
-  FakePortAllocator pa(env, ss());
-  IceTransportInit init(env);
+  FakePortAllocator pa(env_, ss());
+  IceTransportInit init(env_);
   init.set_port_allocator(&pa);
   init.set_ice_controller_factory(&factory);
   auto ch =
@@ -6716,9 +6468,9 @@ TEST_F(P2PTransportChannelPingTest, TestForgetLearnedState) {
 
   // Wait for conn1 to be selected.
   conn1->ReceivedPingResponse(kLowRtt, "id");
-  EXPECT_THAT(WaitUntil([&] { return ch->selected_connection(); }, Eq(conn1),
-                        {.timeout = kMediumTimeout}),
-              IsRtcOk());
+  EXPECT_THAT(
+      DefaultWait().Until([&] { return ch->selected_connection(); }, Eq(conn1)),
+      IsRtcOk());
 
   conn2->ReceivedPingResponse(kLowRtt, "id");
   EXPECT_TRUE(conn2->writable());
@@ -6730,16 +6482,14 @@ TEST_F(P2PTransportChannelPingTest, TestForgetLearnedState) {
 
   // We don't have a mock Connection, so verify this by checking that it
   // is no longer writable.
-  EXPECT_TRUE(WaitUntil([&] { return !conn2->writable(); },
-                        {.timeout = kMediumTimeout}));
+  EXPECT_TRUE(MediumWait().Until([&] { return !conn2->writable(); }));
 }
 
 TEST_F(P2PTransportChannelTest, DisableDnsLookupsWithTransportPolicyRelay) {
-  const Environment env = CreateEnvironment();
-  ConfigureEndpoints(env, OPEN, OPEN, kDefaultPortAllocatorFlags,
+  ConfigureEndpoints(OPEN, OPEN, kDefaultPortAllocatorFlags,
                      kDefaultPortAllocatorFlags);
   auto* ep1 = GetEndpoint(0);
-  ep1->allocator_->SetCandidateFilter(CF_RELAY);
+  ep1->allocator()->SetCandidateFilter(CF_RELAY);
 
   std::unique_ptr<MockAsyncDnsResolver> mock_async_resolver =
       std::make_unique<MockAsyncDnsResolver>();
@@ -6751,9 +6501,9 @@ TEST_F(P2PTransportChannelTest, DisableDnsLookupsWithTransportPolicyRelay) {
       .WillByDefault(
           [&mock_async_resolver]() { return std::move(mock_async_resolver); });
 
-  ep1->async_dns_resolver_factory_ = &mock_async_resolver_factory;
+  ep1->set_async_dns_resolver_factory(&mock_async_resolver_factory);
 
-  CreateChannels(env);
+  CreateChannels();
 
   ep1_ch1()->AddRemoteCandidate(
       CreateUdpCandidate(IceCandidateType::kHost, "hostname.test", 1, 100));
@@ -6762,11 +6512,10 @@ TEST_F(P2PTransportChannelTest, DisableDnsLookupsWithTransportPolicyRelay) {
 }
 
 TEST_F(P2PTransportChannelTest, DisableDnsLookupsWithTransportPolicyNone) {
-  const Environment env = CreateEnvironment();
-  ConfigureEndpoints(env, OPEN, OPEN, kDefaultPortAllocatorFlags,
+  ConfigureEndpoints(OPEN, OPEN, kDefaultPortAllocatorFlags,
                      kDefaultPortAllocatorFlags);
   auto* ep1 = GetEndpoint(0);
-  ep1->allocator_->SetCandidateFilter(CF_NONE);
+  ep1->allocator()->SetCandidateFilter(CF_NONE);
 
   std::unique_ptr<MockAsyncDnsResolver> mock_async_resolver =
       std::make_unique<MockAsyncDnsResolver>();
@@ -6778,9 +6527,9 @@ TEST_F(P2PTransportChannelTest, DisableDnsLookupsWithTransportPolicyNone) {
       .WillByDefault(
           [&mock_async_resolver]() { return std::move(mock_async_resolver); });
 
-  ep1->async_dns_resolver_factory_ = &mock_async_resolver_factory;
+  ep1->set_async_dns_resolver_factory(&mock_async_resolver_factory);
 
-  CreateChannels(env);
+  CreateChannels();
 
   ep1_ch1()->AddRemoteCandidate(
       CreateUdpCandidate(IceCandidateType::kHost, "hostname.test", 1, 100));
@@ -6789,11 +6538,10 @@ TEST_F(P2PTransportChannelTest, DisableDnsLookupsWithTransportPolicyNone) {
 }
 
 TEST_F(P2PTransportChannelTest, EnableDnsLookupsWithTransportPolicyNoHost) {
-  const Environment env = CreateEnvironment();
-  ConfigureEndpoints(env, OPEN, OPEN, kDefaultPortAllocatorFlags,
+  ConfigureEndpoints(OPEN, OPEN, kDefaultPortAllocatorFlags,
                      kDefaultPortAllocatorFlags);
   auto* ep1 = GetEndpoint(0);
-  ep1->allocator_->SetCandidateFilter(CF_ALL & ~CF_HOST);
+  ep1->allocator()->SetCandidateFilter(CF_ALL & ~CF_HOST);
 
   std::unique_ptr<MockAsyncDnsResolver> mock_async_resolver =
       std::make_unique<MockAsyncDnsResolver>();
@@ -6806,9 +6554,9 @@ TEST_F(P2PTransportChannelTest, EnableDnsLookupsWithTransportPolicyNoHost) {
       .WillOnce(
           [&mock_async_resolver]() { return std::move(mock_async_resolver); });
 
-  ep1->async_dns_resolver_factory_ = &mock_async_resolver_factory;
+  ep1->set_async_dns_resolver_factory(&mock_async_resolver_factory);
 
-  CreateChannels(env);
+  CreateChannels();
 
   ep1_ch1()->AddRemoteCandidate(
       CreateUdpCandidate(IceCandidateType::kHost, "hostname.test", 1, 100));
@@ -6834,13 +6582,11 @@ class LocalNetworkAccessPermissionTest
 
 TEST_P(LocalNetworkAccessPermissionTest, LiteralAddresses) {
   const auto [address, lna_fake_result] = GetParam();
-
-  const Environment env = CreateEnvironment();
-  FakePortAllocator pa(env, ss());
+  FakePortAllocator pa(env_, ss());
   FakeLocalNetworkAccessPermissionFactory lna_permission_factory(
       lna_fake_result);
 
-  IceTransportInit init(env);
+  IceTransportInit init(env_);
   init.set_port_allocator(&pa);
   init.set_lna_permission_factory(&lna_permission_factory);
 
@@ -6852,8 +6598,8 @@ TEST_P(LocalNetworkAccessPermissionTest, LiteralAddresses) {
       CreateUdpCandidate(IceCandidateType::kHost, address, 5000, 1));
 
   ASSERT_THAT(
-      WaitUntil([&] { return ch->PermissionQueriesOutstandingForTesting(); },
-                Eq(0)),
+      DefaultWait().Until(
+          [&] { return ch->PermissionQueriesOutstandingForTesting(); }, Eq(0)),
       IsRtcOk());
   if (lna_fake_result == LnaFakeResult::kPermissionNotNeeded ||
       lna_fake_result == LnaFakeResult::kPermissionGranted) {
@@ -6865,16 +6611,14 @@ TEST_P(LocalNetworkAccessPermissionTest, LiteralAddresses) {
 
 TEST_P(LocalNetworkAccessPermissionTest, UnresolvedAddresses) {
   const auto [address, lna_fake_result] = GetParam();
-
-  const Environment env = CreateEnvironment();
-  FakePortAllocator pa(env, ss());
+  FakePortAllocator pa(env_, ss());
   FakeLocalNetworkAccessPermissionFactory lna_permission_factory(
       lna_fake_result);
 
   ResolverFactoryFixture resolver_fixture;
   resolver_fixture.SetAddressToReturn({address, 5000});
 
-  IceTransportInit init(env);
+  IceTransportInit init(env_);
   init.set_port_allocator(&pa);
   init.set_lna_permission_factory(&lna_permission_factory);
   init.set_async_dns_resolver_factory(&resolver_fixture);
@@ -6887,8 +6631,8 @@ TEST_P(LocalNetworkAccessPermissionTest, UnresolvedAddresses) {
       CreateUdpCandidate(IceCandidateType::kHost, "fake.test", 5000, 1));
 
   ASSERT_THAT(
-      WaitUntil([&] { return ch->PermissionQueriesOutstandingForTesting(); },
-                Eq(0)),
+      DefaultWait().Until(
+          [&] { return ch->PermissionQueriesOutstandingForTesting(); }, Eq(0)),
       IsRtcOk());
   if (lna_fake_result == LnaFakeResult::kPermissionNotNeeded ||
       lna_fake_result == LnaFakeResult::kPermissionGranted) {
@@ -6917,18 +6661,18 @@ TEST_P(GatherAfterConnectedTest, GatherAfterConnected) {
       std::string("WebRTC-IceFieldTrials/stop_gather_on_strongly_connected:") +
       (stop_gather_on_strongly_connected ? "true/" : "false/");
 
-  ScopedFakeClock clock;
-  const Environment env =
-      CreateEnvironment(CreateTestFieldTrialsPtr(field_trial));
+  env_ = CreateTestEnvironment(
+      {.field_trials = CreateTestFieldTrialsPtr(field_trial),
+       .time = &time_controller_});
   // Use local + relay
   constexpr uint32_t flags =
       kDefaultPortAllocatorFlags | PORTALLOCATOR_ENABLE_SHARED_SOCKET |
       PORTALLOCATOR_DISABLE_STUN | PORTALLOCATOR_DISABLE_TCP;
-  ConfigureEndpoints(env, OPEN, OPEN, flags, flags);
+  ConfigureEndpoints(OPEN, OPEN, flags, flags);
   auto* ep1 = GetEndpoint(0);
   auto* ep2 = GetEndpoint(1);
-  ep1->allocator_->SetCandidateFilter(CF_ALL);
-  ep2->allocator_->SetCandidateFilter(CF_ALL);
+  ep1->allocator()->SetCandidateFilter(CF_ALL);
+  ep2->allocator()->SetCandidateFilter(CF_ALL);
 
   // Use step delay 3s which is long enough for
   // connection to be established before managing to gather relay candidates.
@@ -6937,18 +6681,16 @@ TEST_P(GatherAfterConnectedTest, GatherAfterConnected) {
   SetAllocationStepDelay(1, delay);
   IceConfig ice_config =
       CreateIceConfig(TimeDelta::Seconds(1), GATHER_CONTINUALLY);
-  CreateChannels(env, ice_config, ice_config);
+  CreateChannels(ice_config, ice_config);
 
   PauseCandidates(0);
   PauseCandidates(1);
 
   // We have gathered host candidates but not relay.
-  ASSERT_TRUE(WaitUntil(
-      [&] {
-        return ep1->saved_candidates_.size() == 1u &&
-               ep2->saved_candidates_.size() == 1u;
-      },
-      {.timeout = kDefaultTimeout, .clock = &clock}));
+  ASSERT_TRUE(DefaultWait().Until([&] {
+    return ep1->saved_candidates().size() == 1u &&
+           ep2->saved_candidates().size() == 1u;
+  }));
 
   ResumeCandidates(0);
   ResumeCandidates(1);
@@ -6956,30 +6698,25 @@ TEST_P(GatherAfterConnectedTest, GatherAfterConnected) {
   PauseCandidates(0);
   PauseCandidates(1);
 
-  ASSERT_TRUE(WaitUntil(
-      [&] {
-        return ep1_ch1()->remote_candidates().size() == 1 &&
-               ep2_ch1()->remote_candidates().size() == 1;
-      },
-      {.timeout = kDefaultTimeout, .clock = &clock}));
+  ASSERT_TRUE(DefaultWait().Until([&] {
+    return ep1_ch1()->remote_candidates().size() == 1 &&
+           ep2_ch1()->remote_candidates().size() == 1;
+  }));
 
-  ASSERT_TRUE(WaitUntil(
-      [&] {
-        return ep1_ch1()->selected_connection() &&
-               ep2_ch1()->selected_connection();
-      },
-      {.timeout = kDefaultTimeout, .clock = &clock}));
+  ASSERT_TRUE(DefaultWait().Until([&] {
+    return ep1_ch1()->selected_connection() && ep2_ch1()->selected_connection();
+  }));
 
-  clock.AdvanceTime(TimeDelta::Millis(10 * delay));
+  time_controller_.AdvanceTime(TimeDelta::Millis(10 * delay));
 
   if (stop_gather_on_strongly_connected) {
     // The relay candidates gathered has not been propagated to channel.
-    EXPECT_EQ(ep1->saved_candidates_.size(), 0u);
-    EXPECT_EQ(ep2->saved_candidates_.size(), 0u);
+    EXPECT_EQ(ep1->saved_candidates().size(), 0u);
+    EXPECT_EQ(ep2->saved_candidates().size(), 0u);
   } else {
     // The relay candidates gathered has been propagated to channel.
-    EXPECT_EQ(ep1->saved_candidates_.size(), 1u);
-    EXPECT_EQ(ep2->saved_candidates_.size(), 1u);
+    EXPECT_EQ(ep1->saved_candidates().size(), 1u);
+    EXPECT_EQ(ep2->saved_candidates().size(), 1u);
   }
 }
 
@@ -6989,20 +6726,20 @@ TEST_P(GatherAfterConnectedTest, GatherAfterConnectedMultiHomed) {
       std::string("WebRTC-IceFieldTrials/stop_gather_on_strongly_connected:") +
       (stop_gather_on_strongly_connected ? "true/" : "false/");
 
-  ScopedFakeClock clock;
-  const Environment env =
-      CreateEnvironment(CreateTestFieldTrialsPtr(field_trial));
+  env_ = CreateTestEnvironment(
+      {.field_trials = CreateTestFieldTrialsPtr(field_trial),
+       .time = &time_controller_});
 
   // Use local + relay
   constexpr uint32_t flags =
       kDefaultPortAllocatorFlags | PORTALLOCATOR_ENABLE_SHARED_SOCKET |
       PORTALLOCATOR_DISABLE_STUN | PORTALLOCATOR_DISABLE_TCP;
   AddAddress(0, kAlternateAddrs[0]);
-  ConfigureEndpoints(env, OPEN, OPEN, flags, flags);
+  ConfigureEndpoints(OPEN, OPEN, flags, flags);
   auto* ep1 = GetEndpoint(0);
   auto* ep2 = GetEndpoint(1);
-  ep1->allocator_->SetCandidateFilter(CF_ALL);
-  ep2->allocator_->SetCandidateFilter(CF_ALL);
+  ep1->allocator()->SetCandidateFilter(CF_ALL);
+  ep2->allocator()->SetCandidateFilter(CF_ALL);
 
   // Use step delay 3s which is long enough for
   // connection to be established before managing to gather relay candidates.
@@ -7011,18 +6748,16 @@ TEST_P(GatherAfterConnectedTest, GatherAfterConnectedMultiHomed) {
   SetAllocationStepDelay(1, delay);
   IceConfig ice_config =
       CreateIceConfig(TimeDelta::Seconds(1), GATHER_CONTINUALLY);
-  CreateChannels(env, ice_config, ice_config);
+  CreateChannels(ice_config, ice_config);
 
   PauseCandidates(0);
   PauseCandidates(1);
 
   // We have gathered host candidates but not relay.
-  ASSERT_TRUE(WaitUntil(
-      [&] {
-        return ep1->saved_candidates_.size() == 2u &&
-               ep2->saved_candidates_.size() == 1u;
-      },
-      {.timeout = kDefaultTimeout, .clock = &clock}));
+  ASSERT_TRUE(DefaultWait().Until([&] {
+    return ep1->saved_candidates().size() == 2u &&
+           ep2->saved_candidates().size() == 1u;
+  }));
 
   ResumeCandidates(0);
   ResumeCandidates(1);
@@ -7030,48 +6765,41 @@ TEST_P(GatherAfterConnectedTest, GatherAfterConnectedMultiHomed) {
   PauseCandidates(0);
   PauseCandidates(1);
 
-  ASSERT_TRUE(WaitUntil(
-      [&] {
-        return ep1_ch1()->remote_candidates().size() == 1 &&
-               ep2_ch1()->remote_candidates().size() == 2;
-      },
-      {.timeout = kDefaultTimeout, .clock = &clock}));
+  ASSERT_TRUE(DefaultWait().Until([&] {
+    return ep1_ch1()->remote_candidates().size() == 1 &&
+           ep2_ch1()->remote_candidates().size() == 2;
+  }));
 
-  ASSERT_TRUE(WaitUntil(
-      [&] {
-        return ep1_ch1()->selected_connection() &&
-               ep2_ch1()->selected_connection();
-      },
-      {.timeout = kDefaultTimeout, .clock = &clock}));
+  ASSERT_TRUE(DefaultWait().Until([&] {
+    return ep1_ch1()->selected_connection() && ep2_ch1()->selected_connection();
+  }));
 
-  clock.AdvanceTime(TimeDelta::Millis(10 * delay));
+  time_controller_.AdvanceTime(TimeDelta::Millis(10 * delay));
 
   if (stop_gather_on_strongly_connected) {
     // The relay candidates gathered has not been propagated to channel.
-    EXPECT_EQ(ep1->saved_candidates_.size(), 0u);
-    EXPECT_EQ(ep2->saved_candidates_.size(), 0u);
+    EXPECT_EQ(ep1->saved_candidates().size(), 0u);
+    EXPECT_EQ(ep2->saved_candidates().size(), 0u);
   } else {
     // The relay candidates gathered has been propagated.
-    EXPECT_EQ(ep1->saved_candidates_.size(), 2u);
-    EXPECT_EQ(ep2->saved_candidates_.size(), 1u);
+    EXPECT_EQ(ep1->saved_candidates().size(), 2u);
+    EXPECT_EQ(ep2->saved_candidates().size(), 1u);
   }
 }
 
 // Tests no candidates are generated with old ice ufrag/passwd after an ice
 // restart even if continual gathering is enabled.
 TEST_F(P2PTransportChannelTest, TestIceNoOldCandidatesAfterIceRestart) {
-  ScopedFakeClock clock;
-  const Environment env = CreateEnvironment();
   AddAddress(0, kAlternateAddrs[0]);
-  ConfigureEndpoints(env, OPEN, OPEN, kDefaultPortAllocatorFlags,
+  ConfigureEndpoints(OPEN, OPEN, kDefaultPortAllocatorFlags,
                      kDefaultPortAllocatorFlags);
 
   // gathers continually.
   IceConfig config = CreateIceConfig(TimeDelta::Seconds(1), GATHER_CONTINUALLY);
-  CreateChannels(env, config, config);
+  CreateChannels(config, config);
 
-  EXPECT_TRUE(WaitUntil([&] { return CheckConnected(ep1_ch1(), ep2_ch1()); },
-                        {.timeout = kDefaultTimeout, .clock = &clock}));
+  EXPECT_TRUE(DefaultWait().Until(
+      [&] { return CheckConnected(ep1_ch1(), ep2_ch1()); }));
 
   PauseCandidates(0);
 
@@ -7079,11 +6807,11 @@ TEST_F(P2PTransportChannelTest, TestIceNoOldCandidatesAfterIceRestart) {
   ep1_ch1()->MaybeStartGathering();
 
   EXPECT_THAT(
-      WaitUntil([&] { return GetEndpoint(0)->saved_candidates_.size(); }, Gt(0),
-                {.timeout = kDefaultTimeout, .clock = &clock}),
+      DefaultWait().Until(
+          [&] { return GetEndpoint(0)->saved_candidates().size(); }, Gt(0)),
       IsRtcOk());
 
-  for (const auto& cd : GetEndpoint(0)->saved_candidates_) {
+  for (const auto& cd : GetEndpoint(0)->saved_candidates()) {
     EXPECT_EQ(cd.candidate.username(), kIceUfrag[3]);
   }
 
@@ -7104,13 +6832,12 @@ class P2PTransportChannelTestDtlsInStun : public P2PTransportChannelTestBase {
 
  protected:
   void Run(bool ep1_support, bool ep2_support) {
-    const Environment env = CreateEnvironment();
-    CreatePortAllocators(env);
+    CreatePortAllocators();
     IceConfig ep1_config;
     ep1_config.dtls_handshake_in_stun = ep1_support;
     IceConfig ep2_config;
     ep2_config.dtls_handshake_in_stun = ep2_support;
-    CreateChannels(env, ep1_config, ep2_config);
+    CreateChannels(ep1_config, ep2_config);
     if (ep1_support) {
       ep1_ch1()->SetDtlsStunPiggybackCallbacks(DtlsStunPiggybackCallbacks(
           [&](auto type) { return data_to_piggyback_func(type); },
@@ -7121,8 +6848,8 @@ class P2PTransportChannelTestDtlsInStun : public P2PTransportChannelTestBase {
           [&](auto type) { return data_to_piggyback_func(type); },
           [&](auto data, auto ack) { piggyback_data_received(data, ack); }));
     }
-    EXPECT_TRUE(WaitUntil([&] { return CheckConnected(ep1_ch1(), ep2_ch1()); },
-                          {.timeout = kDefaultTimeout, .clock = &clock_}));
+    EXPECT_TRUE(DefaultWait().Until(
+        [&] { return CheckConnected(ep1_ch1(), ep2_ch1()); }));
     DestroyChannels();
   }
 
@@ -7135,7 +6862,6 @@ class P2PTransportChannelTestDtlsInStun : public P2PTransportChannelTestBase {
   void piggyback_data_received(std::optional<std::span<uint8_t>> data,
                                std::optional<std::vector<uint32_t>> ack) {}
 
-  ScopedFakeClock clock_;
   Buffer pending_packet_;
 };
 
@@ -7159,7 +6885,8 @@ class P2PTransportChannelRegatheringTest : public P2PTransportChannelPingTest {
  public:
   void SetupChannel(IceConfig config) {
     // Use a fixed environment for the allocator and channel.
-    env_ = std::make_unique<Environment>(CreateEnvironment());
+    env_ = std::make_unique<Environment>(
+        CreateTestEnvironment({.time = time_controller_.GetClock()}));
     allocator_ = std::make_unique<FakePortAllocator>(*env_, ss());
     channel_ = std::make_unique<P2PTransportChannel>(*env_, "regather", 1,
                                                      allocator_.get());
@@ -7196,7 +6923,6 @@ class P2PTransportChannelRegatheringTest : public P2PTransportChannelPingTest {
 
 TEST_F(P2PTransportChannelRegatheringTest,
        IceRegatheringDoesNotOccurIfSessionNotCleared) {
-  ScopedFakeClock clock;
   IceConfig config;
   config.regather_on_failed_networks_interval = TimeDelta::Millis(2000);
   SetupChannel(config);
@@ -7205,13 +6931,11 @@ TEST_F(P2PTransportChannelRegatheringTest,
   // Expect no regathering in the last 10s.
   // We use WaitUntil with a condition that is always false to wait for timeout,
   // but we expect it to fail (return false).
-  EXPECT_FALSE(WaitUntil([&] { return false; },
-                         {.timeout = TimeDelta::Seconds(10), .clock = &clock}));
+  EXPECT_FALSE(DefaultWait().Until([&] { return false; }));
   EXPECT_EQ(0, GetRegatheringCount(IceRegatheringReason::NETWORK_FAILURE));
 }
 
 TEST_F(P2PTransportChannelRegatheringTest, IceRegatheringRepeatsAsScheduled) {
-  ScopedFakeClock clock;
   IceConfig config;
   config.regather_on_failed_networks_interval = TimeDelta::Millis(2000);
   SetupChannel(config);
@@ -7219,30 +6943,23 @@ TEST_F(P2PTransportChannelRegatheringTest, IceRegatheringRepeatsAsScheduled) {
   channel_->allocator_session()->ClearGettingPorts();
 
   // Expect no regathering immediately (wait < 2000ms).
-  EXPECT_FALSE(
-      WaitUntil([&] { return false; },
-                {.timeout = TimeDelta::Millis(1900), .clock = &clock}));
+  EXPECT_FALSE(Wait(TimeDelta::Millis(1900)).Until([&] { return false; }));
   EXPECT_EQ(0, GetRegatheringCount(IceRegatheringReason::NETWORK_FAILURE));
 
   // Expect regathering to happen once after 2s.
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return GetRegatheringCount(IceRegatheringReason::NETWORK_FAILURE) == 1;
-      },
-      {.timeout = TimeDelta::Millis(500), .clock = &clock}));
+  EXPECT_TRUE(DefaultWait().Until([&] {
+    return GetRegatheringCount(IceRegatheringReason::NETWORK_FAILURE) == 1;
+  }));
 
   // Expect regathering to happen for another 5 times in 11s with 2s interval.
   // Total 6.
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return GetRegatheringCount(IceRegatheringReason::NETWORK_FAILURE) == 6;
-      },
-      {.timeout = TimeDelta::Seconds(11), .clock = &clock}));
+  EXPECT_TRUE(Wait(TimeDelta::Seconds(11)).Until([&] {
+    return GetRegatheringCount(IceRegatheringReason::NETWORK_FAILURE) == 6;
+  }));
 }
 
 TEST_F(P2PTransportChannelRegatheringTest,
        ScheduleOfIceRegatheringOnFailedNetworksCanBeReplaced) {
-  ScopedFakeClock clock;
   IceConfig config;
   config.regather_on_failed_networks_interval = TimeDelta::Millis(2000);
   SetupChannel(config);
@@ -7254,17 +6971,14 @@ TEST_F(P2PTransportChannelRegatheringTest,
 
   // Expect no regathering from the previous schedule (Wait 3s, previous was
   // 2s). Since we reset the schedule, it should restart counting 5s from now.
-  EXPECT_FALSE(WaitUntil([&] { return false; },
-                         {.timeout = TimeDelta::Seconds(3), .clock = &clock}));
+  EXPECT_FALSE(MediumWait().Until([&] { return false; }));
   EXPECT_EQ(0, GetRegatheringCount(IceRegatheringReason::NETWORK_FAILURE));
 
   // Expect regathering to happen twice in the last 11s (total 14s) with 5s
   // interval. First at 5s, second at 10s.
-  EXPECT_TRUE(WaitUntil(
-      [&] {
-        return GetRegatheringCount(IceRegatheringReason::NETWORK_FAILURE) == 2;
-      },
-      {.timeout = TimeDelta::Seconds(8), .clock = &clock}));
+  EXPECT_TRUE(Wait(TimeDelta::Seconds(8)).Until([&] {
+    return GetRegatheringCount(IceRegatheringReason::NETWORK_FAILURE) == 2;
+  }));
 }
 
 }  // namespace

From 3d3d455fd56b75d66bf7ad9b3d2c1cfdc3d9af8b Mon Sep 17 00:00:00 2001
From: Per K <perkj@webrtc.org>
Date: Wed, 25 Mar 2026 09:45:46 +0000
Subject: [PATCH 403/472] Correct setting Scream padding end time
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This fix a bug where padding start after 1s after a congestion event instead of ensuring that the correct time period after last padding attempt have passed.

Bug: webrtc:447037083
Change-Id: I6e0915a793abfa14648c01bde09ffc41e4f64e86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/459960
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47256}
---
 .../scream/scream_network_controller.cc       |  5 ++-
 .../scream_network_controller_unittest.cc     | 38 ++++++++++++++++++-
 2 files changed, 40 insertions(+), 3 deletions(-)

diff --git a/modules/congestion_controller/scream/scream_network_controller.cc b/modules/congestion_controller/scream/scream_network_controller.cc
index e20e5898e5..c496bc7154 100644
--- a/modules/congestion_controller/scream/scream_network_controller.cc
+++ b/modules/congestion_controller/scream/scream_network_controller.cc
@@ -64,6 +64,7 @@ NetworkControlUpdate ScreamNetworkController::CreateFirstUpdate(Timestamp now) {
   RTC_DCHECK(network_available_);
   RTC_DCHECK(!first_update_created_);
   first_update_created_ = true;
+  padding_interval_end_time_ = Timestamp::MinusInfinity();
   if (allow_initial_bwe_before_media_) {
     initial_bwe_probe_end_time_ = now + params_.initial_probing_duration.Get();
   }
@@ -237,9 +238,9 @@ std::optional<PacerConfig> ScreamNetworkController::MaybeCreatePacerConfig(
           now + params_.periodic_padding_duration.Get();
       allow_padding = true;
     }
-  } else {
+  } else if (now < padding_interval_end_time_) {
     // Stop padding immediately if a congestion event occurred recently.
-    padding_interval_end_time_ = Timestamp::MinusInfinity();
+    padding_interval_end_time_ = now;
   }
 
   DataRate padding_rate = DataRate::Zero();
diff --git a/modules/congestion_controller/scream/scream_network_controller_unittest.cc b/modules/congestion_controller/scream/scream_network_controller_unittest.cc
index 9dc4280d4b..6331b8dfc0 100644
--- a/modules/congestion_controller/scream/scream_network_controller_unittest.cc
+++ b/modules/congestion_controller/scream/scream_network_controller_unittest.cc
@@ -412,7 +412,9 @@ PaddingTestResult ProcessUntilPaddingStartAndStop(
         scream_controller.OnTransportPacketsFeedback(feedback);
     if (update.pacer_config.has_value()) {
       if (update.pacer_config->pad_rate() != DataRate::Zero()) {
-        padding_start = clock.CurrentTime();
+        if (padding_start.IsZero()) {
+          padding_start = clock.CurrentTime();
+        }
         if (increase_send_rate) {
           // Set the send rate equal to the padding rate.
           send_rate = update.pacer_config->pad_rate();
@@ -487,6 +489,40 @@ TEST(ScreamControllerTest, PeriodicallyAllowPadding) {
   EXPECT_LT(time_between_padding, TimeDelta::Millis(3300));
 }
 
+TEST(ScreamControllerTest, DelayPaddingAfterCongestion) {
+  SimulatedClock clock(Timestamp::Seconds(1'234));
+  Environment env = CreateTestEnvironment({.time = &clock});
+  NetworkControllerConfig config(env);
+  ScreamNetworkController scream_controller(config);
+  CcFeedbackGenerator feedback_generator(
+      {.network_config = {.queue_delay_ms = 10,
+                          .link_capacity = DataRate::KilobitsPerSec(500)},
+       .send_as_ect1 = true});
+  StreamsConfig streams_config;
+  streams_config.max_total_allocated_bitrate = DataRate::KilobitsPerSec(1000);
+  scream_controller.OnStreamsConfig(streams_config);
+
+  // Padding should start, but then stop since pushing to max allocated causes
+  // congestion.
+  PaddingTestResult result_1 = ProcessUntilPaddingStartAndStop(
+      clock, scream_controller, feedback_generator,
+      /*increase_send_rate=*/true);
+
+  // Ensure it was stopped quickly when congestion kicked in.
+  EXPECT_LT(result_1.padding_stop - result_1.padding_start,
+            TimeDelta::Seconds(1));
+
+  // Network recovers because increase_send_rate=false keeps sending rate at
+  // 50kbps. Wait until next padding starts.
+  PaddingTestResult result_2 = ProcessUntilPaddingStartAndStop(
+      clock, scream_controller, feedback_generator,
+      /*increase_send_rate=*/false);
+
+  TimeDelta time_until_padding = result_2.padding_start - result_1.padding_stop;
+  EXPECT_GT(time_until_padding, TimeDelta::Millis(2500));
+  EXPECT_LT(time_until_padding, TimeDelta::Millis(3500));
+}
+
 TEST(ScreamControllerTest, PadsToMinOf2xCurrentMaxAndEverSeenMax) {
   SimulatedClock clock(Timestamp::Seconds(1'234));
   Environment env = CreateTestEnvironment({.time = &clock});

From 228f04dcf40b739cdab909c2afcd7ad2d40719e6 Mon Sep 17 00:00:00 2001
From: Philipp Hancke <philipp.hancke@googlemail.com>
Date: Wed, 25 Mar 2026 09:00:18 +0100
Subject: [PATCH 404/472] pc: rename data_channel_unittest

to match the name of the file it is testing

Bug: None
Change-Id: I040f5459b9d88e022bf3a165e3c2b08c9c956711
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/459941
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#47257}
---
 pc/BUILD.gn                                                    | 2 +-
 pc/{data_channel_unittest.cc => sctp_data_channel_unittest.cc} | 3 ++-
 2 files changed, 3 insertions(+), 2 deletions(-)
 rename pc/{data_channel_unittest.cc => sctp_data_channel_unittest.cc} (99%)

diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index f2045cb32a..ca8dcd86b9 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -2490,7 +2490,6 @@ if (rtc_include_tests && !build_with_chromium) {
     sources = [
       "congestion_control_integrationtest.cc",
       "data_channel_integrationtest.cc",
-      "data_channel_unittest.cc",
       "dtmf_sender_unittest.cc",
       "ice_server_parsing_unittest.cc",
       "jitter_buffer_delay_unittest.cc",
@@ -2529,6 +2528,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "rtp_sender_receiver_unittest.cc",
       "rtp_transceiver_unittest.cc",
       "scoped_operations_batcher_unittest.cc",
+      "sctp_data_channel_unittest.cc",
       "sctp_utils_unittest.cc",
       "sdp_munging_detector_unittest.cc",
       "sdp_offer_answer_unittest.cc",
diff --git a/pc/data_channel_unittest.cc b/pc/sctp_data_channel_unittest.cc
similarity index 99%
rename from pc/data_channel_unittest.cc
rename to pc/sctp_data_channel_unittest.cc
index 82f4d852cd..06b4a8851f 100644
--- a/pc/data_channel_unittest.cc
+++ b/pc/sctp_data_channel_unittest.cc
@@ -8,6 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "pc/sctp_data_channel.h"
+
 #include <cstdint>
 #include <cstring>
 #include <memory>
@@ -25,7 +27,6 @@
 #include "api/task_queue/pending_task_safety_flag.h"
 #include "api/test/rtc_error_matchers.h"
 #include "api/transport/data_channel_transport_interface.h"
-#include "pc/sctp_data_channel.h"
 #include "pc/sctp_utils.h"
 #include "pc/test/fake_data_channel_controller.h"
 #include "rtc_base/checks.h"

From da927d7fd81d1702ca5968009120c968bb561cde Mon Sep 17 00:00:00 2001
From: Evan Shrubsole <eshr@webrtc.org>
Date: Fri, 6 Mar 2026 14:49:45 +0000
Subject: [PATCH 405/472] Refactor stun_port_unittest to use
 GlobalSimulatedTimeController

Bug: webrtc:381524905
Bug: webrtc:42223992
Bug: webrtc:469327588
Change-Id: Ib2db3c582aaf3f7b6e8e0dee39e87e666a6a6964
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/454380
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47258}
---
 p2p/BUILD.gn                   |   1 +
 p2p/base/stun_port_unittest.cc | 670 ++++++++++++++-------------------
 2 files changed, 285 insertions(+), 386 deletions(-)

diff --git a/p2p/BUILD.gn b/p2p/BUILD.gn
index d59a176b96..a18edbb3aa 100644
--- a/p2p/BUILD.gn
+++ b/p2p/BUILD.gn
@@ -1223,6 +1223,7 @@ if (rtc_include_tests) {
       "../system_wrappers:metrics",
       "../test:create_test_environment",
       "../test:create_test_field_trials",
+      "../test:run_loop",
       "../test:test_support",
       "../test:wait_until",
       "../test/time_controller",
diff --git a/p2p/base/stun_port_unittest.cc b/p2p/base/stun_port_unittest.cc
index fbdb534845..71d577976f 100644
--- a/p2p/base/stun_port_unittest.cc
+++ b/p2p/base/stun_port_unittest.cc
@@ -28,9 +28,9 @@
 #include "api/field_trials_view.h"
 #include "api/packet_socket_factory.h"
 #include "api/test/mock_async_dns_resolver.h"
-#include "api/test/rtc_error_matchers.h"
 #include "api/transport/stun.h"
 #include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
 #include "p2p/base/basic_packet_socket_factory.h"
 #include "p2p/base/port.h"
 #include "p2p/base/stun_request.h"
@@ -43,8 +43,6 @@
 #include "rtc_base/checks.h"
 #include "rtc_base/crypto_random.h"
 #include "rtc_base/dscp.h"
-#include "rtc_base/fake_clock.h"
-#include "rtc_base/gunit.h"
 #include "rtc_base/ip_address.h"
 #include "rtc_base/mdns_responder_interface.h"
 #include "rtc_base/net_helpers.h"
@@ -62,6 +60,7 @@
 #include "test/create_test_field_trials.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/time_controller/simulated_time_controller.h"
 #include "test/wait_until.h"
 
 namespace webrtc {
@@ -79,23 +78,22 @@ using ::testing::SetArgPointee;
 const SocketAddress kPrivateIP("192.168.1.12", 0);
 const SocketAddress kMsdnAddress("unittest-mdns-host-name.local", 0);
 const SocketAddress kPublicIP("212.116.91.133", 0);
-const SocketAddress kNatAddr(kPublicIP.ipaddr(), webrtc::NAT_SERVER_UDP_PORT);
+const SocketAddress kNatAddr(kPublicIP.ipaddr(), NAT_SERVER_UDP_PORT);
 const SocketAddress kStunServerAddr1("34.38.54.120", 5000);
 const SocketAddress kStunServerAddr2("34.38.54.120", 4000);
 
 const SocketAddress kPrivateIPv6("2001:4860:4860::8844", 0);
 const SocketAddress kPublicIPv6("2002:4860:4860::8844", 5000);
-const SocketAddress kNatAddrIPv6(kPublicIPv6.ipaddr(),
-                                 webrtc::NAT_SERVER_UDP_PORT);
+const SocketAddress kNatAddrIPv6(kPublicIPv6.ipaddr(), NAT_SERVER_UDP_PORT);
 const SocketAddress kStunServerAddrIPv6Addr("2003:4860:4860::8844", 5000);
 
 const SocketAddress kBadAddr("0.0.0.1", 5000);
 const SocketAddress kIPv6BadAddr("::ffff:0:1", 5000);
 const SocketAddress kValidHostnameAddr("valid-hostname", 5000);
 const SocketAddress kBadHostnameAddr("not-a-real-hostname", 5000);
-// STUN timeout (with all retries) is webrtc::STUN_TOTAL_TIMEOUT.
+// STUN timeout (with all retries) is STUN_TOTAL_TIMEOUT.
 // Add some margin of error for slow bots.
-constexpr int kTimeoutMs = webrtc::STUN_TOTAL_TIMEOUT;
+constexpr TimeDelta kTimeout = TimeDelta::Millis(STUN_TOTAL_TIMEOUT + 5000);
 // stun prio = 100 (srflx) << 24 | 30 (IPv4) << 8 | 256 - 1 (component)
 constexpr uint32_t kStunCandidatePriority = (100 << 24) | (30 << 8) | (256 - 1);
 // stun prio = 100 (srflx) << 24 | 40 (IPv6) << 8 | 256 - 1 (component)
@@ -105,81 +103,76 @@ constexpr uint64_t kTiebreakerDefault = 44444;
 
 struct IPAddressTypeTestConfig {
   absl::string_view address;
-  webrtc::IPAddressType address_type;
+  IPAddressType address_type;
 };
 
-// Used by the test framework to print the param value for parameterized tests.
-std::string PrintToString(const IPAddressTypeTestConfig& param) {
-  return std::string(param.address);
-}
-
-class FakeMdnsResponder : public webrtc::MdnsResponderInterface {
+class FakeMdnsResponder : public MdnsResponderInterface {
  public:
-  void CreateNameForAddress(const webrtc::IPAddress& addr,
+  void CreateNameForAddress(const IPAddress& addr,
                             NameCreatedCallback callback) override {
     callback(addr, kMsdnAddress.HostAsSensitiveURIString());
   }
 
-  void RemoveNameForAddress(const webrtc::IPAddress& addr,
+  void RemoveNameForAddress(const IPAddress& addr,
                             NameRemovedCallback callback) override {}
 };
 
-class FakeMdnsResponderProvider : public webrtc::MdnsResponderProvider {
+class FakeMdnsResponderProvider : public MdnsResponderProvider {
  public:
   FakeMdnsResponderProvider() : mdns_responder_(new FakeMdnsResponder()) {}
 
-  webrtc::MdnsResponderInterface* GetMdnsResponder() const override {
+  MdnsResponderInterface* GetMdnsResponder() const override {
     return mdns_responder_.get();
   }
 
  private:
-  std::unique_ptr<webrtc::MdnsResponderInterface> mdns_responder_;
+  std::unique_ptr<MdnsResponderInterface> mdns_responder_;
 };
 
 // Base class for tests connecting a StunPort to a fake STUN server
-// (webrtc::StunServer).
-class StunPortTestBase : public ::testing::Test {
+// (StunServer).
+class StunPortTest : public ::testing::Test {
  public:
-  StunPortTestBase()
-      : StunPortTestBase(kPrivateIP.ipaddr(),
-                         {kStunServerAddr1, kStunServerAddr2},
-                         kNatAddr) {}
+  StunPortTest()
+      : StunPortTest(kPrivateIP.ipaddr(),
+                     {kStunServerAddr1, kStunServerAddr2},
+                     kNatAddr) {}
 
-  StunPortTestBase(const webrtc::IPAddress address,
-                   const std::set<webrtc::SocketAddress>& stun_server_addresses,
-                   const webrtc::SocketAddress& nat_server_address)
-      : env_(CreateTestEnvironment()),
-        ss_(new webrtc::VirtualSocketServer()),
-        thread_(ss_.get()),
+  StunPortTest(const IPAddress address,
+               const std::set<SocketAddress>& stun_server_addresses,
+               const SocketAddress& nat_server_address)
+      : ss_(std::make_unique<VirtualSocketServer>()),
+        time_controller_(Timestamp::Zero(), ss_.get()),
+        env_(CreateTestEnvironment({.time = &time_controller_})),
+        network_thread_(time_controller_.GetMainThread()),
         nat_factory_(ss_.get(), nat_server_address, nat_server_address),
         nat_socket_factory_(&nat_factory_),
         mdns_responder_provider_(new FakeMdnsResponderProvider()),
-        nat_server_(CreateNatServer(nat_server_address, webrtc::NAT_OPEN_CONE)),
         done_(false),
         error_(false),
         stun_keepalive_delay_(TimeDelta::Millis(1)) {
+    nat_server_ = CreateNatServer(nat_server_address, NAT_OPEN_CONE);
     network_ = MakeNetwork(address);
     RTC_CHECK(address.family() == nat_server_address.family());
     for (const auto& addr : stun_server_addresses) {
       RTC_CHECK(addr.family() == address.family());
       stun_servers_.push_back(
-          webrtc::TestStunServer::Create(env_, addr, *ss_, thread_));
+          TestStunServer::Create(env_, addr, *ss_, *network_thread_));
     }
   }
 
-  std::unique_ptr<webrtc::NATServer> CreateNatServer(const SocketAddress& addr,
-                                                     webrtc::NATType type) {
-    return std::make_unique<webrtc::NATServer>(
-        env_, type, thread_, ss_.get(), addr, addr, thread_, ss_.get(), addr);
+  std::unique_ptr<NATServer> CreateNatServer(const SocketAddress& addr,
+                                             NATType type) {
+    return std::make_unique<NATServer>(env_, type, *network_thread_, ss_.get(),
+                                       addr, addr, *network_thread_, ss_.get(),
+                                       addr);
   }
 
-  virtual webrtc::PacketSocketFactory* socket_factory() {
-    return &nat_socket_factory_;
-  }
+  virtual PacketSocketFactory* socket_factory() { return &nat_socket_factory_; }
 
-  webrtc::SocketServer* ss() const { return ss_.get(); }
-  webrtc::UDPPort* port() const { return stun_port_.get(); }
-  webrtc::AsyncPacketSocket* socket() const { return socket_.get(); }
+  SocketServer* ss() const { return ss_.get(); }
+  UDPPort* port() const { return stun_port_.get(); }
+  AsyncPacketSocket* socket() const { return socket_.get(); }
   bool done() const { return done_; }
   bool error() const { return error_; }
 
@@ -187,32 +180,32 @@ class StunPortTestBase : public ::testing::Test {
     return stun_port_->request_manager().HasRequestForTest(msg_type);
   }
 
-  void SetNetworkType(webrtc::AdapterType adapter_type) {
+  void SetNetworkType(AdapterType adapter_type) {
     network_->set_type(adapter_type);
   }
 
-  void CreateStunPort(const webrtc::SocketAddress& server_addr,
-                      const webrtc::FieldTrialsView* field_trials = nullptr) {
+  void CreateStunPort(const SocketAddress& server_addr,
+                      const FieldTrialsView* field_trials = nullptr) {
     ServerAddresses stun_servers;
     stun_servers.insert(server_addr);
     CreateStunPort(stun_servers, field_trials);
   }
 
   void CreateStunPort(const ServerAddresses& stun_servers,
-                      const webrtc::FieldTrialsView* field_trials = nullptr) {
+                      const FieldTrialsView* field_trials = nullptr) {
     // Overwrite field trials if provided.
     EnvironmentFactory env_factory(env_);
     env_factory.Set(field_trials);
     Environment env = env_factory.Create();
 
-    stun_port_ = webrtc::StunPort::Create(
-        {.env = env,
-         .network_thread = &thread_,
-         .socket_factory = socket_factory(),
-         .network = network_,
-         .ice_username_fragment = webrtc::CreateRandomString(16),
-         .ice_password = webrtc::CreateRandomString(22)},
-        0, 0, stun_servers, std::nullopt);
+    stun_port_ =
+        StunPort::Create({.env = env,
+                          .network_thread = network_thread_,
+                          .socket_factory = socket_factory(),
+                          .network = network_,
+                          .ice_username_fragment = CreateRandomString(16),
+                          .ice_password = CreateRandomString(22)},
+                         0, 0, stun_servers, std::nullopt);
     stun_port_->SetIceTiebreaker(kTiebreakerDefault);
     stun_port_->set_stun_keepalive_delay(stun_keepalive_delay_);
     // If `stun_keepalive_lifetime_` is not set, let the stun port choose its
@@ -245,20 +238,19 @@ class StunPortTestBase : public ::testing::Test {
     }
     ASSERT_TRUE(socket_ != nullptr);
     socket_->RegisterReceivedPacketCallback(
-        [&](webrtc::AsyncPacketSocket* socket,
-            const webrtc::ReceivedIpPacket& packet) {
+        [&](AsyncPacketSocket* socket, const ReceivedIpPacket& packet) {
           OnReadPacket(socket, packet);
         });
     ServerAddresses stun_servers;
     stun_servers.insert(server_addr);
-    stun_port_ = webrtc::UDPPort::Create(
-        {.env = env_,
-         .network_thread = &thread_,
-         .socket_factory = socket_factory(),
-         .network = network_,
-         .ice_username_fragment = webrtc::CreateRandomString(16),
-         .ice_password = webrtc::CreateRandomString(22)},
-        socket_.get(), false, std::nullopt);
+    stun_port_ =
+        UDPPort::Create({.env = env_,
+                         .network_thread = network_thread_,
+                         .socket_factory = socket_factory(),
+                         .network = network_,
+                         .ice_username_fragment = CreateRandomString(16),
+                         .ice_password = CreateRandomString(22)},
+                        socket_.get(), false, std::nullopt);
     stun_port_->set_server_addresses(stun_servers);
     ASSERT_TRUE(stun_port_ != nullptr);
     stun_port_->SetIceTiebreaker(kTiebreakerDefault);
@@ -268,18 +260,20 @@ class StunPortTestBase : public ::testing::Test {
                                    [this](Port* port) { OnPortError(port); });
   }
 
-  void PrepareAddress() { stun_port_->PrepareAddress(); }
+  void PrepareAddress() {
+    stun_port_->PrepareAddress();
+    time_controller_.AdvanceTime(TimeDelta::Zero());
+  }
 
-  void OnReadPacket(webrtc::AsyncPacketSocket* socket,
-                    const webrtc::ReceivedIpPacket& packet) {
+  void OnReadPacket(AsyncPacketSocket* socket, const ReceivedIpPacket& packet) {
     stun_port_->HandleIncomingPacket(socket, packet);
   }
 
   void SendData(const char* data, size_t len) {
     stun_port_->HandleIncomingPacket(
-        socket_.get(), webrtc::ReceivedIpPacket::CreateFromLegacy(
-                           data, len, /* packet_time_us */ -1,
-                           webrtc::SocketAddress("22.22.22.22", 0)));
+        socket_.get(),
+        ReceivedIpPacket::CreateFromLegacy(data, len, /* packet_time_us */ -1,
+                                           SocketAddress("22.22.22.22", 0)));
   }
 
   void EnableMdnsObfuscation() {
@@ -287,17 +281,16 @@ class StunPortTestBase : public ::testing::Test {
   }
 
  protected:
-  void OnPortComplete(webrtc::Port* /* port */) {
+  void OnPortComplete(Port* /* port */) {
     ASSERT_FALSE(done_);
     done_ = true;
     error_ = false;
   }
-  void OnPortError(webrtc::Port* /* port */) {
+  void OnPortError(Port* /* port */) {
     done_ = true;
     error_ = true;
   }
-  void OnCandidateError(webrtc::Port* /* port */,
-                        const webrtc::IceCandidateErrorEvent& event) {
+  void OnCandidateError(Port* /* port */, const IceCandidateErrorEvent& event) {
     error_event_ = event;
   }
   void SetKeepaliveDelay(TimeDelta delay) { stun_keepalive_delay_ = delay; }
@@ -306,53 +299,45 @@ class StunPortTestBase : public ::testing::Test {
     stun_keepalive_lifetime_ = lifetime;
   }
 
-  webrtc::Network* MakeNetwork(const webrtc::IPAddress& addr) {
+  Network* MakeNetwork(const IPAddress& addr) {
     networks_.emplace_back(
         std::make_unique<Network>("unittest", "unittest", addr, 32));
     networks_.back()->AddIP(addr);
     return networks_.back().get();
   }
 
-  webrtc::TestStunServer* stun_server_1() { return stun_servers_[0].get(); }
-  webrtc::TestStunServer* stun_server_2() { return stun_servers_[1].get(); }
+  TestStunServer* stun_server_1() { return stun_servers_[0].get(); }
+  TestStunServer* stun_server_2() { return stun_servers_[1].get(); }
 
-  webrtc::AutoSocketServerThread& thread() { return thread_; }
-  webrtc::SocketFactory* nat_factory() { return &nat_factory_; }
+  Thread& thread() { return *network_thread_; }
+  SocketFactory* nat_factory() { return &nat_factory_; }
 
- private:
+ protected:
+  std::unique_ptr<VirtualSocketServer> ss_;
+  GlobalSimulatedTimeController time_controller_;
   const Environment env_;
-  std::vector<std::unique_ptr<webrtc::Network>> networks_;
-  webrtc::Network* network_;
+  std::vector<std::unique_ptr<Network>> networks_;
+  Network* network_;
 
-  std::unique_ptr<webrtc::VirtualSocketServer> ss_;
-  webrtc::AutoSocketServerThread thread_;
-  webrtc::NATSocketFactory nat_factory_;
-  webrtc::BasicPacketSocketFactory nat_socket_factory_;
+  Thread* network_thread_;
+  NATSocketFactory nat_factory_;
+  BasicPacketSocketFactory nat_socket_factory_;
   // Note that stun_port_ can refer to socket_, so must be destroyed
   // before it.
-  std::unique_ptr<webrtc::AsyncPacketSocket> socket_;
-  std::unique_ptr<webrtc::UDPPort> stun_port_;
-  std::vector<webrtc::TestStunServer::StunServerPtr> stun_servers_;
-  std::unique_ptr<webrtc::MdnsResponderProvider> mdns_responder_provider_;
-  std::unique_ptr<webrtc::NATServer> nat_server_;
+  std::unique_ptr<AsyncPacketSocket> socket_;
+  std::unique_ptr<UDPPort> stun_port_;
+  std::vector<TestStunServer::StunServerPtr> stun_servers_;
+  std::unique_ptr<MdnsResponderProvider> mdns_responder_provider_;
+  std::unique_ptr<NATServer> nat_server_;
   bool done_;
   bool error_;
   TimeDelta stun_keepalive_delay_;
   std::optional<TimeDelta> stun_keepalive_lifetime_;
 
  protected:
-  webrtc::IceCandidateErrorEvent error_event_;
+  IceCandidateErrorEvent error_event_;
 };
 
-class StunPortTestWithRealClock : public StunPortTestBase {};
-
-class FakeClockBase {
- public:
-  webrtc::ScopedFakeClock fake_clock;
-};
-
-class StunPortTest : public FakeClockBase, public StunPortTestBase {};
-
 // Test that we can create a STUN port.
 TEST_F(StunPortTest, TestCreateStunPort) {
   CreateStunPort(kStunServerAddr1);
@@ -371,11 +356,8 @@ TEST_F(StunPortTest, TestCreateUdpPort) {
 TEST_F(StunPortTest, TestPrepareAddress) {
   CreateStunPort(kStunServerAddr1);
   PrepareAddress();
-  EXPECT_THAT(
-      webrtc::WaitUntil([&] { return done(); }, IsTrue(),
-                        {.timeout = webrtc::TimeDelta::Millis(kTimeoutMs),
-                         .clock = &fake_clock}),
-      webrtc::IsRtcOk());
+  EXPECT_TRUE(WaitUntil([this] { return done(); },
+                        {.timeout = kTimeout, .clock = &time_controller_}));
   ASSERT_EQ(1U, port()->Candidates().size());
   EXPECT_TRUE(kPublicIP.EqualIPs(port()->Candidates()[0].address()));
   std::string expected_server_url = "stun:" + kStunServerAddr1.ToString();
@@ -386,19 +368,11 @@ TEST_F(StunPortTest, TestPrepareAddress) {
 TEST_F(StunPortTest, TestPrepareAddressFail) {
   CreateStunPort(kBadAddr);
   PrepareAddress();
-  EXPECT_THAT(
-      webrtc::WaitUntil([&] { return done(); }, IsTrue(),
-                        {.timeout = webrtc::TimeDelta::Millis(kTimeoutMs),
-                         .clock = &fake_clock}),
-      webrtc::IsRtcOk());
+  EXPECT_TRUE(WaitUntil([this] { return done(); },
+                        {.timeout = kTimeout, .clock = &time_controller_}));
   EXPECT_TRUE(error());
   EXPECT_EQ(0U, port()->Candidates().size());
-  EXPECT_THAT(
-      webrtc::WaitUntil([&] { return error_event_.error_code; },
-                        Eq(webrtc::STUN_ERROR_SERVER_NOT_REACHABLE),
-                        {.timeout = webrtc::TimeDelta::Millis(kTimeoutMs),
-                         .clock = &fake_clock}),
-      webrtc::IsRtcOk());
+  EXPECT_EQ(error_event_.error_code, STUN_ERROR_SERVER_NOT_REACHABLE);
   EXPECT_NE(error_event_.error_text.find('.'), std::string::npos);
   EXPECT_NE(error_event_.address.find(kPrivateIP.HostAsSensitiveURIString()),
             std::string::npos);
@@ -411,11 +385,8 @@ TEST_F(StunPortTest, TestPrepareAddressFail) {
 TEST_F(StunPortTest, TestServerAddressFamilyMismatch) {
   CreateStunPort(kStunServerAddrIPv6Addr);
   PrepareAddress();
-  EXPECT_THAT(
-      webrtc::WaitUntil([&] { return done(); }, IsTrue(),
-                        {.timeout = webrtc::TimeDelta::Millis(kTimeoutMs),
-                         .clock = &fake_clock}),
-      webrtc::IsRtcOk());
+  EXPECT_TRUE(WaitUntil([this] { return done(); },
+                        {.timeout = kTimeout, .clock = &time_controller_}));
   EXPECT_TRUE(error());
   EXPECT_EQ(0U, port()->Candidates().size());
   EXPECT_EQ(0, error_event_.error_code);
@@ -426,42 +397,35 @@ class StunPortWithMockDnsResolverTest : public StunPortTest {
   StunPortWithMockDnsResolverTest()
       : StunPortTest(), socket_factory_(nat_factory()) {}
 
-  webrtc::PacketSocketFactory* socket_factory() override {
-    return &socket_factory_;
-  }
+  PacketSocketFactory* socket_factory() override { return &socket_factory_; }
 
   void SetDnsResolverExpectations(
-      webrtc::MockDnsResolvingPacketSocketFactory::Expectations expectations) {
+      MockDnsResolvingPacketSocketFactory::Expectations expectations) {
     socket_factory_.SetExpectations(expectations);
   }
 
  private:
-  webrtc::MockDnsResolvingPacketSocketFactory socket_factory_;
+  MockDnsResolvingPacketSocketFactory socket_factory_;
 };
 
 // Test that we can get an address from a STUN server specified by a hostname.
 TEST_F(StunPortWithMockDnsResolverTest, TestPrepareAddressHostname) {
-  SetDnsResolverExpectations(
-      [](webrtc::MockAsyncDnsResolver* resolver,
-         webrtc::MockAsyncDnsResolverResult* resolver_result) {
-        EXPECT_CALL(*resolver, Start(kValidHostnameAddr, /*family=*/AF_INET, _))
-            .WillOnce([](const webrtc::SocketAddress& /* addr */,
-                         int /* family */,
-                         absl::AnyInvocable<void()> callback) { callback(); });
+  SetDnsResolverExpectations([](MockAsyncDnsResolver* resolver,
+                                MockAsyncDnsResolverResult* resolver_result) {
+    EXPECT_CALL(*resolver, Start(kValidHostnameAddr, /*family=*/AF_INET, _))
+        .WillOnce([](const SocketAddress& /* addr */, int /* family */,
+                     absl::AnyInvocable<void()> callback) { callback(); });
 
-        EXPECT_CALL(*resolver, result)
-            .WillRepeatedly(ReturnPointee(resolver_result));
-        EXPECT_CALL(*resolver_result, GetError).WillOnce(Return(0));
-        EXPECT_CALL(*resolver_result, GetResolvedAddress(AF_INET, _))
-            .WillOnce(DoAll(SetArgPointee<1>(kStunServerAddr1), Return(true)));
-      });
+    EXPECT_CALL(*resolver, result)
+        .WillRepeatedly(ReturnPointee(resolver_result));
+    EXPECT_CALL(*resolver_result, GetError).WillOnce(Return(0));
+    EXPECT_CALL(*resolver_result, GetResolvedAddress(AF_INET, _))
+        .WillOnce(DoAll(SetArgPointee<1>(kStunServerAddr1), Return(true)));
+  });
   CreateStunPort(kValidHostnameAddr);
   PrepareAddress();
-  EXPECT_THAT(
-      webrtc::WaitUntil([&] { return done(); }, IsTrue(),
-                        {.timeout = webrtc::TimeDelta::Millis(kTimeoutMs),
-                         .clock = &fake_clock}),
-      webrtc::IsRtcOk());
+  EXPECT_TRUE(WaitUntil([this] { return done(); },
+                        {.timeout = kTimeout, .clock = &time_controller_}));
   ASSERT_EQ(1U, port()->Candidates().size());
   EXPECT_TRUE(kPublicIP.EqualIPs(port()->Candidates()[0].address()));
   EXPECT_EQ(kStunCandidatePriority, port()->Candidates()[0].priority());
@@ -471,47 +435,47 @@ TEST_F(StunPortWithMockDnsResolverTest,
        TestPrepareAddressHostnameWithPriorityAdjustment) {
   FieldTrials field_trials = CreateTestFieldTrials(
       "WebRTC-IncreaseIceCandidatePriorityHostSrflx/Enabled/");
-  SetDnsResolverExpectations(
-      [](webrtc::MockAsyncDnsResolver* resolver,
-         webrtc::MockAsyncDnsResolverResult* resolver_result) {
-        EXPECT_CALL(*resolver, Start(kValidHostnameAddr, /*family=*/AF_INET, _))
-            .WillOnce([](const webrtc::SocketAddress& /* addr */,
-                         int /* family */,
-                         absl::AnyInvocable<void()> callback) { callback(); });
-        EXPECT_CALL(*resolver, result)
-            .WillRepeatedly(ReturnPointee(resolver_result));
-        EXPECT_CALL(*resolver_result, GetError).WillOnce(Return(0));
-        EXPECT_CALL(*resolver_result, GetResolvedAddress(AF_INET, _))
-            .WillOnce(DoAll(SetArgPointee<1>(kStunServerAddr1), Return(true)));
-      });
+  SetDnsResolverExpectations([](MockAsyncDnsResolver* resolver,
+                                MockAsyncDnsResolverResult* resolver_result) {
+    EXPECT_CALL(*resolver, Start(kValidHostnameAddr, /*family=*/AF_INET, _))
+        .WillOnce([](const SocketAddress& /* addr */, int /* family */,
+                     absl::AnyInvocable<void()> callback) { callback(); });
+    EXPECT_CALL(*resolver, result)
+        .WillRepeatedly(ReturnPointee(resolver_result));
+    EXPECT_CALL(*resolver_result, GetError).WillOnce(Return(0));
+    EXPECT_CALL(*resolver_result, GetResolvedAddress(AF_INET, _))
+        .WillOnce(DoAll(SetArgPointee<1>(kStunServerAddr1), Return(true)));
+  });
   CreateStunPort(kValidHostnameAddr, &field_trials);
   PrepareAddress();
-  EXPECT_THAT(
-      webrtc::WaitUntil([&] { return done(); }, IsTrue(),
-                        {.timeout = webrtc::TimeDelta::Millis(kTimeoutMs),
-                         .clock = &fake_clock}),
-      webrtc::IsRtcOk());
+  EXPECT_TRUE(WaitUntil([this] { return done(); },
+                        {.timeout = kTimeout, .clock = &time_controller_}));
   ASSERT_EQ(1U, port()->Candidates().size());
   EXPECT_TRUE(kPublicIP.EqualIPs(port()->Candidates()[0].address()));
-  EXPECT_EQ(kStunCandidatePriority + (webrtc::kMaxTurnServers << 8),
+  EXPECT_EQ(kStunCandidatePriority + (kMaxTurnServers << 8),
             port()->Candidates()[0].priority());
 }
 
 // Test that we handle hostname lookup failures properly.
-TEST_F(StunPortTestWithRealClock, TestPrepareAddressHostnameFail) {
+// Test that we fail to get an address from a STUN server specified by an
+// invalid hostname.
+TEST_F(StunPortWithMockDnsResolverTest, TestPrepareAddressHostnameFail) {
+  SetDnsResolverExpectations([](MockAsyncDnsResolver* resolver,
+                                MockAsyncDnsResolverResult* resolver_result) {
+    EXPECT_CALL(*resolver, Start(kBadHostnameAddr, /*family=*/AF_INET, _))
+        .WillOnce([](const SocketAddress& /* addr */, int /* family */,
+                     absl::AnyInvocable<void()> callback) { callback(); });
+
+    EXPECT_CALL(*resolver, result)
+        .WillRepeatedly(ReturnPointee(resolver_result));
+    EXPECT_CALL(*resolver_result, GetError).WillOnce(Return(-2));
+  });
   CreateStunPort(kBadHostnameAddr);
   PrepareAddress();
-  EXPECT_THAT(
-      webrtc::WaitUntil([&] { return done(); }, IsTrue(),
-                        {.timeout = webrtc::TimeDelta::Millis(kTimeoutMs)}),
-      webrtc::IsRtcOk());
-  EXPECT_TRUE(error());
+  EXPECT_TRUE(WaitUntil([this] { return error(); },
+                        {.timeout = kTimeout, .clock = &time_controller_}));
   EXPECT_EQ(0U, port()->Candidates().size());
-  EXPECT_THAT(
-      webrtc::WaitUntil([&] { return error_event_.error_code; },
-                        Eq(webrtc::STUN_ERROR_SERVER_NOT_REACHABLE),
-                        {.timeout = webrtc::TimeDelta::Millis(kTimeoutMs)}),
-      webrtc::IsRtcOk());
+  EXPECT_EQ(error_event_.error_code, STUN_ERROR_SERVER_NOT_REACHABLE);
 }
 
 // This test verifies keepalive response messages don't result in
@@ -520,14 +484,11 @@ TEST_F(StunPortTest, TestKeepAliveResponse) {
   SetKeepaliveDelay(TimeDelta::Millis(500));
   CreateStunPort(kStunServerAddr1);
   PrepareAddress();
-  EXPECT_THAT(
-      webrtc::WaitUntil([&] { return done(); }, IsTrue(),
-                        {.timeout = webrtc::TimeDelta::Millis(kTimeoutMs),
-                         .clock = &fake_clock}),
-      webrtc::IsRtcOk());
+  EXPECT_TRUE(WaitUntil([this] { return done(); },
+                        {.timeout = kTimeout, .clock = &time_controller_}));
   ASSERT_EQ(1U, port()->Candidates().size());
   EXPECT_TRUE(kPublicIP.EqualIPs(port()->Candidates()[0].address()));
-  SIMULATED_WAIT(false, 1000, fake_clock);
+  time_controller_.AdvanceTime(TimeDelta::Millis(1000));
   EXPECT_EQ(1U, port()->Candidates().size());
 }
 
@@ -535,11 +496,8 @@ TEST_F(StunPortTest, TestKeepAliveResponse) {
 TEST_F(StunPortTest, TestSharedSocketPrepareAddress) {
   CreateSharedUdpPort(kStunServerAddr1, nullptr);
   PrepareAddress();
-  EXPECT_THAT(
-      webrtc::WaitUntil([&] { return done(); }, IsTrue(),
-                        {.timeout = webrtc::TimeDelta::Millis(kTimeoutMs),
-                         .clock = &fake_clock}),
-      webrtc::IsRtcOk());
+  EXPECT_TRUE(WaitUntil([this] { return done(); },
+                        {.timeout = kTimeout, .clock = &time_controller_}));
   ASSERT_EQ(2U, port()->Candidates().size());
   EXPECT_EQ(port()->Candidates()[0].type(), IceCandidateType::kHost);
   EXPECT_TRUE(kPrivateIP.EqualIPs(port()->Candidates()[0].address()));
@@ -550,20 +508,28 @@ TEST_F(StunPortTest, TestSharedSocketPrepareAddress) {
 // Test that we still get a local candidate with invalid stun server hostname.
 // Also verifing that UDPPort can receive packets when stun address can't be
 // resolved.
-TEST_F(StunPortTestWithRealClock,
+TEST_F(StunPortWithMockDnsResolverTest,
        TestSharedSocketPrepareAddressInvalidHostname) {
+  SetDnsResolverExpectations([](MockAsyncDnsResolver* resolver,
+                                MockAsyncDnsResolverResult* resolver_result) {
+    EXPECT_CALL(*resolver, Start(kBadHostnameAddr, /*family=*/AF_INET, _))
+        .WillOnce([](const SocketAddress& /* addr */, int /* family */,
+                     absl::AnyInvocable<void()> callback) { callback(); });
+
+    EXPECT_CALL(*resolver, result)
+        .WillRepeatedly(ReturnPointee(resolver_result));
+    EXPECT_CALL(*resolver_result, GetError).WillOnce(Return(-2));
+  });
   CreateSharedUdpPort(kBadHostnameAddr, nullptr);
   PrepareAddress();
-  EXPECT_THAT(
-      webrtc::WaitUntil([&] { return done(); }, IsTrue(),
-                        {.timeout = webrtc::TimeDelta::Millis(kTimeoutMs)}),
-      webrtc::IsRtcOk());
+  EXPECT_TRUE(WaitUntil([this] { return done(); },
+                        {.timeout = kTimeout, .clock = &time_controller_}));
   ASSERT_EQ(1U, port()->Candidates().size());
   EXPECT_TRUE(kPrivateIP.EqualIPs(port()->Candidates()[0].address()));
 
   // Send data to port after it's ready. This is to make sure, UDP port can
   // handle data with unresolved stun server address.
-  std::string data = "some random data, sending to webrtc::Port.";
+  std::string data = "some random data, sending to Port.";
   SendData(data.c_str(), data.length());
   // No crash is success.
 }
@@ -574,11 +540,8 @@ TEST_F(StunPortTest, TestStunCandidateGeneratedWithMdnsObfuscationEnabled) {
   EnableMdnsObfuscation();
   CreateSharedUdpPort(kStunServerAddr1, nullptr);
   PrepareAddress();
-  EXPECT_THAT(
-      webrtc::WaitUntil([&] { return done(); }, IsTrue(),
-                        {.timeout = webrtc::TimeDelta::Millis(kTimeoutMs),
-                         .clock = &fake_clock}),
-      webrtc::IsRtcOk());
+  EXPECT_TRUE(WaitUntil([this] { return done(); },
+                        {.timeout = kTimeout, .clock = &time_controller_}));
   ASSERT_EQ(2U, port()->Candidates().size());
 
   // One of the generated candidates is a local candidate and the other is a
@@ -607,11 +570,8 @@ TEST_F(StunPortTest, TestNoDuplicatedAddressWithTwoStunServers) {
   CreateStunPort(stun_servers);
   EXPECT_EQ(IceCandidateType::kSrflx, port()->Type());
   PrepareAddress();
-  EXPECT_THAT(
-      webrtc::WaitUntil([&] { return done(); }, IsTrue(),
-                        {.timeout = webrtc::TimeDelta::Millis(kTimeoutMs),
-                         .clock = &fake_clock}),
-      webrtc::IsRtcOk());
+  EXPECT_TRUE(WaitUntil([this] { return done(); },
+                        {.timeout = kTimeout, .clock = &time_controller_}));
   EXPECT_EQ(1U, port()->Candidates().size());
   EXPECT_EQ(port()->Candidates()[0].relay_protocol(), "");
 }
@@ -625,18 +585,11 @@ TEST_F(StunPortTest, TestMultipleStunServersWithBadServer) {
   CreateStunPort(stun_servers);
   EXPECT_EQ(IceCandidateType::kSrflx, port()->Type());
   PrepareAddress();
-  EXPECT_THAT(
-      webrtc::WaitUntil([&] { return done(); }, IsTrue(),
-                        {.timeout = webrtc::TimeDelta::Millis(kTimeoutMs),
-                         .clock = &fake_clock}),
-      webrtc::IsRtcOk());
+  EXPECT_TRUE(WaitUntil([this] { return done(); },
+                        {.timeout = kTimeout, .clock = &time_controller_}));
   EXPECT_EQ(1U, port()->Candidates().size());
   std::string server_url = "stun:" + kBadAddr.ToString();
-  ASSERT_THAT(
-      webrtc::WaitUntil([&] { return error_event_.url; }, Eq(server_url),
-                        {.timeout = webrtc::TimeDelta::Millis(kTimeoutMs),
-                         .clock = &fake_clock}),
-      webrtc::IsRtcOk());
+  EXPECT_EQ(error_event_.url, server_url);
 }
 
 // Test that two candidates are allocated if the two STUN servers return
@@ -653,11 +606,8 @@ TEST_F(StunPortTest, TestTwoCandidatesWithTwoStunServersAcrossNat) {
   CreateStunPort(stun_servers);
   EXPECT_EQ(IceCandidateType::kSrflx, port()->Type());
   PrepareAddress();
-  EXPECT_THAT(
-      webrtc::WaitUntil([&] { return done(); }, IsTrue(),
-                        {.timeout = webrtc::TimeDelta::Millis(kTimeoutMs),
-                         .clock = &fake_clock}),
-      webrtc::IsRtcOk());
+  EXPECT_TRUE(WaitUntil([this] { return done(); },
+                        {.timeout = kTimeout, .clock = &time_controller_}));
   EXPECT_EQ(2U, port()->Candidates().size());
   EXPECT_EQ(port()->Candidates()[0].relay_protocol(), "");
   EXPECT_EQ(port()->Candidates()[1].relay_protocol(), "");
@@ -671,11 +621,11 @@ TEST_F(StunPortTest, TestStunPortGetStunKeepaliveLifetime) {
   CreateStunPort(kStunServerAddr1);
   EXPECT_EQ(port()->stun_keepalive_lifetime(), TimeDelta::PlusInfinity());
   // Lifetime for the cellular network is `kHighCostPortKeepaliveLifetime`
-  SetNetworkType(webrtc::ADAPTER_TYPE_CELLULAR);
+  SetNetworkType(ADAPTER_TYPE_CELLULAR);
   EXPECT_EQ(port()->stun_keepalive_lifetime(), kHighCostPortKeepaliveLifetime);
 
   // Lifetime for the wifi network is infinite.
-  SetNetworkType(webrtc::ADAPTER_TYPE_WIFI);
+  SetNetworkType(ADAPTER_TYPE_WIFI);
   CreateStunPort(kStunServerAddr2);
   EXPECT_EQ(port()->stun_keepalive_lifetime(), TimeDelta::PlusInfinity());
 }
@@ -688,11 +638,11 @@ TEST_F(StunPortTest, TestUdpPortGetStunKeepaliveLifetime) {
   CreateSharedUdpPort(kStunServerAddr1, nullptr);
   EXPECT_EQ(port()->stun_keepalive_lifetime(), TimeDelta::PlusInfinity());
   // Lifetime for the cellular network is `kHighCostPortKeepaliveLifetime`.
-  SetNetworkType(webrtc::ADAPTER_TYPE_CELLULAR);
+  SetNetworkType(ADAPTER_TYPE_CELLULAR);
   EXPECT_EQ(port()->stun_keepalive_lifetime(), kHighCostPortKeepaliveLifetime);
 
   // Lifetime for the wifi network type is infinite.
-  SetNetworkType(webrtc::ADAPTER_TYPE_WIFI);
+  SetNetworkType(ADAPTER_TYPE_WIFI);
   CreateSharedUdpPort(kStunServerAddr2, nullptr);
   EXPECT_EQ(port()->stun_keepalive_lifetime(), TimeDelta::PlusInfinity());
 }
@@ -704,16 +654,11 @@ TEST_F(StunPortTest, TestStunBindingRequestShortLifetime) {
   SetKeepaliveLifetime(TimeDelta::Millis(100));
   CreateStunPort(kStunServerAddr1);
   PrepareAddress();
-  EXPECT_THAT(
-      webrtc::WaitUntil([&] { return done(); }, IsTrue(),
-                        {.timeout = webrtc::TimeDelta::Millis(kTimeoutMs),
-                         .clock = &fake_clock}),
-      webrtc::IsRtcOk());
-  EXPECT_THAT(
-      webrtc::WaitUntil(
-          [&] { return !HasPendingRequest(webrtc::STUN_BINDING_REQUEST); },
-          IsTrue(), {.clock = &fake_clock}),
-      webrtc::IsRtcOk());
+  EXPECT_TRUE(WaitUntil([this] { return done(); },
+                        {.timeout = kTimeout, .clock = &time_controller_}));
+  EXPECT_TRUE(
+      WaitUntil([&] { return !HasPendingRequest(STUN_BINDING_REQUEST); },
+                {.timeout = kTimeout, .clock = &time_controller_}));
 }
 
 // Test that by default, the STUN binding requests will last for a long time.
@@ -721,16 +666,10 @@ TEST_F(StunPortTest, TestStunBindingRequestLongLifetime) {
   SetKeepaliveDelay(TimeDelta::Millis(101));
   CreateStunPort(kStunServerAddr1);
   PrepareAddress();
-  EXPECT_THAT(
-      webrtc::WaitUntil([&] { return done(); }, IsTrue(),
-                        {.timeout = webrtc::TimeDelta::Millis(kTimeoutMs),
-                         .clock = &fake_clock}),
-      webrtc::IsRtcOk());
-  EXPECT_THAT(
-      webrtc::WaitUntil(
-          [&] { return HasPendingRequest(webrtc::STUN_BINDING_REQUEST); },
-          IsTrue(), {.clock = &fake_clock}),
-      webrtc::IsRtcOk());
+  EXPECT_TRUE(WaitUntil([this] { return done(); },
+                        {.timeout = kTimeout, .clock = &time_controller_}));
+  EXPECT_TRUE(WaitUntil([&] { return HasPendingRequest(STUN_BINDING_REQUEST); },
+                        {.timeout = kTimeout, .clock = &time_controller_}));
 }
 
 class StunPortIPAddressTypeMetricsTest
@@ -738,51 +677,46 @@ class StunPortIPAddressTypeMetricsTest
       public ::testing::WithParamInterface<IPAddressTypeTestConfig> {};
 
 TEST_P(StunPortIPAddressTypeMetricsTest, TestIPAddressTypeMetrics) {
-  SetDnsResolverExpectations(
-      [](webrtc::MockAsyncDnsResolver* resolver,
-         webrtc::MockAsyncDnsResolverResult* resolver_result) {
-        EXPECT_CALL(*resolver, Start(SocketAddress("localhost", 5000),
-                                     /*family=*/AF_INET, _))
-            .WillOnce([](const webrtc::SocketAddress& /* addr */,
-                         int /* family */,
-                         absl::AnyInvocable<void()> callback) { callback(); });
+  SetDnsResolverExpectations([](MockAsyncDnsResolver* resolver,
+                                MockAsyncDnsResolverResult* resolver_result) {
+    EXPECT_CALL(*resolver, Start(SocketAddress("localhost", 5000),
+                                 /*family=*/AF_INET, _))
+        .WillOnce([](const SocketAddress& /* addr */, int /* family */,
+                     absl::AnyInvocable<void()> callback) { callback(); });
 
-        EXPECT_CALL(*resolver, result)
-            .WillRepeatedly(ReturnPointee(resolver_result));
-        EXPECT_CALL(*resolver_result, GetError).WillOnce(Return(0));
-        EXPECT_CALL(*resolver_result, GetResolvedAddress(AF_INET, _))
-            .WillOnce(DoAll(SetArgPointee<1>(SocketAddress("127.0.0.1", 5000)),
-                            Return(true)));
-      });
+    EXPECT_CALL(*resolver, result)
+        .WillRepeatedly(ReturnPointee(resolver_result));
+    EXPECT_CALL(*resolver_result, GetError).WillOnce(Return(0));
+    EXPECT_CALL(*resolver_result, GetResolvedAddress(AF_INET, _))
+        .WillOnce(DoAll(SetArgPointee<1>(SocketAddress("127.0.0.1", 5000)),
+                        Return(true)));
+  });
 
-  webrtc::metrics::Reset();
+  metrics::Reset();
 
   CreateStunPort({GetParam().address, 5000});
   PrepareAddress();
-  EXPECT_THAT(
-      webrtc::WaitUntil([&] { return done(); }, IsTrue(),
-                        {.timeout = webrtc::TimeDelta::Millis(kTimeoutMs),
-                         .clock = &fake_clock}),
-      webrtc::IsRtcOk());
+  EXPECT_TRUE(WaitUntil([this] { return done(); },
+                        {.timeout = kTimeout, .clock = &time_controller_}));
 
   auto samples =
-      webrtc::metrics::Samples("WebRTC.PeerConnection.Stun.ServerAddressType");
+      metrics::Samples("WebRTC.PeerConnection.Stun.ServerAddressType");
   ASSERT_EQ(samples.size(), 1u);
   EXPECT_EQ(samples[static_cast<int>(GetParam().address_type)], 1);
 }
 
 const IPAddressTypeTestConfig kAllIPAddressTypeTestConfigs[] = {
-    {.address = "127.0.0.1", .address_type = webrtc::IPAddressType::kLoopback},
-    {.address = "localhost", .address_type = webrtc::IPAddressType::kLoopback},
-    {.address = "10.0.0.3", .address_type = webrtc::IPAddressType::kPrivate},
-    {.address = "1.1.1.1", .address_type = webrtc::IPAddressType::kPublic},
+    {.address = "127.0.0.1", .address_type = IPAddressType::kLoopback},
+    {.address = "localhost", .address_type = IPAddressType::kLoopback},
+    {.address = "10.0.0.3", .address_type = IPAddressType::kPrivate},
+    {.address = "1.1.1.1", .address_type = IPAddressType::kPublic},
 };
 
 INSTANTIATE_TEST_SUITE_P(All,
                          StunPortIPAddressTypeMetricsTest,
                          ::testing::ValuesIn(kAllIPAddressTypeTestConfigs));
 
-class MockAsyncPacketSocket : public webrtc::AsyncPacketSocket {
+class MockAsyncPacketSocket : public AsyncPacketSocket {
  public:
   ~MockAsyncPacketSocket() override = default;
 
@@ -792,7 +726,7 @@ class MockAsyncPacketSocket : public webrtc::AsyncPacketSocket {
               Send,
               (const void* pv,
                size_t cb,
-               const webrtc::AsyncSocketPacketOptions& options),
+               const AsyncSocketPacketOptions& options),
               (override));
 
   MOCK_METHOD(int,
@@ -800,18 +734,12 @@ class MockAsyncPacketSocket : public webrtc::AsyncPacketSocket {
               (const void* pv,
                size_t cb,
                const SocketAddress& addr,
-               const webrtc::AsyncSocketPacketOptions& options),
+               const AsyncSocketPacketOptions& options),
               (override));
   MOCK_METHOD(int, Close, (), (override));
   MOCK_METHOD(State, GetState, (), (const, override));
-  MOCK_METHOD(int,
-              GetOption,
-              (webrtc::Socket::Option opt, int* value),
-              (override));
-  MOCK_METHOD(int,
-              SetOption,
-              (webrtc::Socket::Option opt, int value),
-              (override));
+  MOCK_METHOD(int, GetOption, (Socket::Option opt, int* value), (override));
+  MOCK_METHOD(int, SetOption, (Socket::Option opt, int value), (override));
   MOCK_METHOD(int, GetError, (), (const, override));
   MOCK_METHOD(void, SetError, (int error), (override));
 };
@@ -833,6 +761,7 @@ TEST_F(StunPortTest, TestStunPacketsHaveDscpPacketOption) {
                                    Eq(DSCP_NO_CHANGE))))
       .WillOnce(Return(100));
   PrepareAddress();
+  port()->request_manager().Clear();
 
   // Once it is set transport wide, they should inherit that value.
   port()->SetOption(Socket::OPT_DSCP, DSCP_AF41);
@@ -841,32 +770,25 @@ TEST_F(StunPortTest, TestStunPacketsHaveDscpPacketOption) {
       SendTo(_, _, _, Field(&AsyncSocketPacketOptions::dscp, Eq(DSCP_AF41))))
       .WillRepeatedly(Return(100));
 
-  EXPECT_TRUE(WaitUntil(
-      [&] { return done(); },
-      {.timeout = TimeDelta::Millis(kTimeoutMs), .clock = &fake_clock}));
+  PrepareAddress();
+  EXPECT_TRUE(WaitUntil([this] { return done(); },
+                        {.timeout = kTimeout, .clock = &time_controller_}));
 }
 
-class StunIPv6PortTestBase : public StunPortTestBase {
+class StunIPv6PortTest : public StunPortTest {
  public:
-  StunIPv6PortTestBase()
-      : StunPortTestBase(kPrivateIPv6.ipaddr(),
-                         {kStunServerAddrIPv6Addr},
-                         kNatAddrIPv6) {}
+  StunIPv6PortTest()
+      : StunPortTest(kPrivateIPv6.ipaddr(),
+                     {kStunServerAddrIPv6Addr},
+                     kNatAddrIPv6) {}
 };
 
-class StunIPv6PortTestWithRealClock : public StunIPv6PortTestBase {};
-
-class StunIPv6PortTest : public FakeClockBase, public StunIPv6PortTestBase {};
-
 // Test that we can get an address from a STUN server.
 TEST_F(StunIPv6PortTest, TestPrepareAddress) {
   CreateStunPort(kStunServerAddrIPv6Addr);
   PrepareAddress();
-  EXPECT_THAT(
-      webrtc::WaitUntil([&] { return done(); }, IsTrue(),
-                        {.timeout = webrtc::TimeDelta::Millis(kTimeoutMs),
-                         .clock = &fake_clock}),
-      webrtc::IsRtcOk());
+  EXPECT_TRUE(WaitUntil([this] { return done(); },
+                        {.timeout = kTimeout, .clock = &time_controller_}));
   ASSERT_EQ(1U, port()->Candidates().size());
   EXPECT_TRUE(kPublicIPv6.EqualIPs(port()->Candidates()[0].address()));
   std::string expected_server_url = "stun:2003:4860:4860::8844:5000";
@@ -877,19 +799,11 @@ TEST_F(StunIPv6PortTest, TestPrepareAddress) {
 TEST_F(StunIPv6PortTest, TestPrepareAddressFail) {
   CreateStunPort(kIPv6BadAddr);
   PrepareAddress();
-  EXPECT_THAT(
-      webrtc::WaitUntil([&] { return done(); }, IsTrue(),
-                        {.timeout = webrtc::TimeDelta::Millis(kTimeoutMs),
-                         .clock = &fake_clock}),
-      webrtc::IsRtcOk());
+  EXPECT_TRUE(WaitUntil([this] { return done(); },
+                        {.timeout = kTimeout, .clock = &time_controller_}));
   EXPECT_TRUE(error());
   EXPECT_EQ(0U, port()->Candidates().size());
-  EXPECT_THAT(
-      webrtc::WaitUntil([&] { return error_event_.error_code; },
-                        Eq(webrtc::STUN_ERROR_SERVER_NOT_REACHABLE),
-                        {.timeout = webrtc::TimeDelta::Millis(kTimeoutMs),
-                         .clock = &fake_clock}),
-      webrtc::IsRtcOk());
+  EXPECT_EQ(error_event_.error_code, STUN_ERROR_SERVER_NOT_REACHABLE);
   EXPECT_NE(error_event_.error_text.find('.'), std::string::npos);
   EXPECT_NE(error_event_.address.find(kPrivateIPv6.HostAsSensitiveURIString()),
             std::string::npos);
@@ -902,75 +816,68 @@ TEST_F(StunIPv6PortTest, TestPrepareAddressFail) {
 TEST_F(StunIPv6PortTest, TestServerAddressFamilyMismatch) {
   CreateStunPort(kStunServerAddr1);
   PrepareAddress();
-  EXPECT_THAT(
-      webrtc::WaitUntil([&] { return done(); }, IsTrue(),
-                        {.timeout = webrtc::TimeDelta::Millis(kTimeoutMs),
-                         .clock = &fake_clock}),
-      webrtc::IsRtcOk());
-  EXPECT_TRUE(error());
+  EXPECT_TRUE(WaitUntil([this] { return error(); },
+                        {.timeout = kTimeout, .clock = &time_controller_}));
   EXPECT_EQ(0U, port()->Candidates().size());
   EXPECT_EQ(0, error_event_.error_code);
 }
 
-// Test that we handle hostname lookup failures properly with a real clock.
-TEST_F(StunIPv6PortTestWithRealClock, TestPrepareAddressHostnameFail) {
-  CreateStunPort(kBadHostnameAddr);
-  PrepareAddress();
-  EXPECT_THAT(
-      webrtc::WaitUntil([&] { return done(); }, IsTrue(),
-                        {.timeout = webrtc::TimeDelta::Millis(kTimeoutMs)}),
-      webrtc::IsRtcOk());
-  EXPECT_TRUE(error());
-  EXPECT_EQ(0U, port()->Candidates().size());
-  EXPECT_THAT(
-      webrtc::WaitUntil([&] { return error_event_.error_code; },
-                        Eq(webrtc::STUN_ERROR_SERVER_NOT_REACHABLE),
-                        {.timeout = webrtc::TimeDelta::Millis(kTimeoutMs)}),
-      webrtc::IsRtcOk());
-}
-
 class StunIPv6PortTestWithMockDnsResolver : public StunIPv6PortTest {
  public:
   StunIPv6PortTestWithMockDnsResolver()
       : StunIPv6PortTest(), socket_factory_(ss()) {}
 
-  webrtc::PacketSocketFactory* socket_factory() override {
-    return &socket_factory_;
-  }
+  PacketSocketFactory* socket_factory() override { return &socket_factory_; }
 
   void SetDnsResolverExpectations(
-      webrtc::MockDnsResolvingPacketSocketFactory::Expectations expectations) {
+      MockDnsResolvingPacketSocketFactory::Expectations expectations) {
     socket_factory_.SetExpectations(expectations);
   }
 
  private:
-  webrtc::MockDnsResolvingPacketSocketFactory socket_factory_;
+  MockDnsResolvingPacketSocketFactory socket_factory_;
 };
 
+// Test that we fail to get an address from a STUN server specified by an
+// invalid hostname.
+TEST_F(StunIPv6PortTestWithMockDnsResolver, TestPrepareAddressHostnameFail) {
+  SetDnsResolverExpectations([](MockAsyncDnsResolver* resolver,
+                                MockAsyncDnsResolverResult* resolver_result) {
+    EXPECT_CALL(*resolver, Start(kBadHostnameAddr, /*family=*/AF_INET6, _))
+        .WillOnce([](const SocketAddress& /* addr */, int /* family */,
+                     absl::AnyInvocable<void()> callback) { callback(); });
+
+    EXPECT_CALL(*resolver, result)
+        .WillRepeatedly(ReturnPointee(resolver_result));
+    EXPECT_CALL(*resolver_result, GetError).WillOnce(Return(-2));
+  });
+  CreateStunPort(kBadHostnameAddr);
+  PrepareAddress();
+  EXPECT_TRUE(WaitUntil([this] { return error(); },
+                        {.timeout = kTimeout, .clock = &time_controller_}));
+  EXPECT_EQ(0U, port()->Candidates().size());
+  EXPECT_EQ(error_event_.error_code, STUN_ERROR_SERVER_NOT_REACHABLE);
+}
+
 // Test that we can get an address from a STUN server specified by a hostname.
 TEST_F(StunIPv6PortTestWithMockDnsResolver, TestPrepareAddressHostname) {
-  SetDnsResolverExpectations(
-      [](webrtc::MockAsyncDnsResolver* resolver,
-         webrtc::MockAsyncDnsResolverResult* resolver_result) {
-        EXPECT_CALL(*resolver,
-                    Start(kValidHostnameAddr, /*family=*/AF_INET6, _))
-            .WillOnce([](const webrtc::SocketAddress& addr, int family,
-                         absl::AnyInvocable<void()> callback) { callback(); });
+  SetDnsResolverExpectations([](MockAsyncDnsResolver* resolver,
+                                MockAsyncDnsResolverResult* resolver_result) {
+    EXPECT_CALL(*resolver, Start(kValidHostnameAddr, /*family=*/AF_INET6, _))
+        .WillOnce([](const SocketAddress& addr, int family,
+                     absl::AnyInvocable<void()> callback) { callback(); });
 
-        EXPECT_CALL(*resolver, result)
-            .WillRepeatedly(ReturnPointee(resolver_result));
-        EXPECT_CALL(*resolver_result, GetError).WillOnce(Return(0));
-        EXPECT_CALL(*resolver_result, GetResolvedAddress(AF_INET6, _))
-            .WillOnce(
-                DoAll(SetArgPointee<1>(kStunServerAddrIPv6Addr), Return(true)));
-      });
+    EXPECT_CALL(*resolver, result)
+        .WillRepeatedly(ReturnPointee(resolver_result));
+    EXPECT_CALL(*resolver_result, GetError).WillOnce(Return(0));
+    EXPECT_CALL(*resolver_result, GetResolvedAddress(AF_INET6, _))
+        .WillOnce(
+            DoAll(SetArgPointee<1>(kStunServerAddrIPv6Addr), Return(true)));
+  });
   CreateStunPort(kValidHostnameAddr);
   PrepareAddress();
-  EXPECT_THAT(
-      webrtc::WaitUntil([&] { return done(); }, IsTrue(),
-                        {.timeout = webrtc::TimeDelta::Millis(kTimeoutMs),
-                         .clock = &fake_clock}),
-      webrtc::IsRtcOk());
+  EXPECT_TRUE(WaitUntil([this] { return done(); },
+                        {.timeout = kTimeout, .clock = &time_controller_}));
   ASSERT_EQ(1U, port()->Candidates().size());
   EXPECT_TRUE(kPrivateIPv6.EqualIPs(port()->Candidates()[0].address()));
   EXPECT_EQ(kIPv6StunCandidatePriority, port()->Candidates()[0].priority());
@@ -981,30 +888,25 @@ TEST_F(StunIPv6PortTestWithMockDnsResolver,
        TestPrepareAddressHostnameWithPriorityAdjustment) {
   FieldTrials field_trials = CreateTestFieldTrials(
       "WebRTC-IncreaseIceCandidatePriorityHostSrflx/Enabled/");
-  SetDnsResolverExpectations(
-      [](webrtc::MockAsyncDnsResolver* resolver,
-         webrtc::MockAsyncDnsResolverResult* resolver_result) {
-        EXPECT_CALL(*resolver,
-                    Start(kValidHostnameAddr, /*family=*/AF_INET6, _))
-            .WillOnce([](const webrtc::SocketAddress& addr, int family,
-                         absl::AnyInvocable<void()> callback) { callback(); });
-        EXPECT_CALL(*resolver, result)
-            .WillRepeatedly(ReturnPointee(resolver_result));
-        EXPECT_CALL(*resolver_result, GetError).WillOnce(Return(0));
-        EXPECT_CALL(*resolver_result, GetResolvedAddress(AF_INET6, _))
-            .WillOnce(
-                DoAll(SetArgPointee<1>(kStunServerAddrIPv6Addr), Return(true)));
-      });
+  SetDnsResolverExpectations([](MockAsyncDnsResolver* resolver,
+                                MockAsyncDnsResolverResult* resolver_result) {
+    EXPECT_CALL(*resolver, Start(kValidHostnameAddr, /*family=*/AF_INET6, _))
+        .WillOnce([](const SocketAddress& addr, int family,
+                     absl::AnyInvocable<void()> callback) { callback(); });
+    EXPECT_CALL(*resolver, result)
+        .WillRepeatedly(ReturnPointee(resolver_result));
+    EXPECT_CALL(*resolver_result, GetError).WillOnce(Return(0));
+    EXPECT_CALL(*resolver_result, GetResolvedAddress(AF_INET6, _))
+        .WillOnce(
+            DoAll(SetArgPointee<1>(kStunServerAddrIPv6Addr), Return(true)));
+  });
   CreateStunPort(kValidHostnameAddr, &field_trials);
   PrepareAddress();
-  EXPECT_THAT(
-      webrtc::WaitUntil([&] { return done(); }, IsTrue(),
-                        {.timeout = webrtc::TimeDelta::Millis(kTimeoutMs),
-                         .clock = &fake_clock}),
-      webrtc::IsRtcOk());
+  EXPECT_TRUE(WaitUntil([this] { return done(); },
+                        {.timeout = kTimeout, .clock = &time_controller_}));
   ASSERT_EQ(1U, port()->Candidates().size());
   EXPECT_TRUE(kPrivateIPv6.EqualIPs(port()->Candidates()[0].address()));
-  EXPECT_EQ(kIPv6StunCandidatePriority + (webrtc::kMaxTurnServers << 8),
+  EXPECT_EQ(kIPv6StunCandidatePriority + (kMaxTurnServers << 8),
             port()->Candidates()[0].priority());
 }
 
@@ -1013,28 +915,24 @@ class StunIPv6PortIPAddressTypeMetricsTest
       public ::testing::WithParamInterface<IPAddressTypeTestConfig> {};
 
 TEST_P(StunIPv6PortIPAddressTypeMetricsTest, TestIPAddressTypeMetrics) {
-  webrtc::metrics::Reset();
+  metrics::Reset();
 
   CreateStunPort({GetParam().address, 5000});
   PrepareAddress();
-  EXPECT_THAT(
-      webrtc::WaitUntil([&] { return done(); }, IsTrue(),
-                        {.timeout = webrtc::TimeDelta::Millis(kTimeoutMs),
-                         .clock = &fake_clock}),
-      webrtc::IsRtcOk());
+  EXPECT_TRUE(WaitUntil([this] { return done(); },
+                        {.timeout = kTimeout, .clock = &time_controller_}));
 
   auto samples =
-      webrtc::metrics::Samples("WebRTC.PeerConnection.Stun.ServerAddressType");
+      metrics::Samples("WebRTC.PeerConnection.Stun.ServerAddressType");
   ASSERT_EQ(samples.size(), 1u);
   EXPECT_EQ(samples[static_cast<int>(GetParam().address_type)], 1);
 }
 
 const IPAddressTypeTestConfig kAllIPv6AddressTypeTestConfigs[] = {
-    {.address = "::1", .address_type = webrtc::IPAddressType::kLoopback},
+    {.address = "::1", .address_type = IPAddressType::kLoopback},
     {.address = "fd00:4860:4860::8844",
-     .address_type = webrtc::IPAddressType::kPrivate},
-    {.address = "2001:4860:4860::8888",
-     .address_type = webrtc::IPAddressType::kPublic},
+     .address_type = IPAddressType::kPrivate},
+    {.address = "2001:4860:4860::8888", .address_type = IPAddressType::kPublic},
 };
 
 INSTANTIATE_TEST_SUITE_P(All,

From 8551776812a7ec4d992b226a9c22ddfaa8438b48 Mon Sep 17 00:00:00 2001
From: Evan Shrubsole <eshr@webrtc.org>
Date: Tue, 24 Mar 2026 12:08:28 +0000
Subject: [PATCH 406/472] Add gtest-parallel skill

Bug: webrtc:465491622
Change-Id: I94531c0098b71093ff439f68d82b36db6a6a6964
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/459720
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47259}
---
 .agents/.style.mdformat                |  0
 .agents/skills/gtest-parallel/SKILL.md | 60 ++++++++++++++++++++++++++
 api/DEPS                               |  1 +
 3 files changed, 61 insertions(+)
 create mode 100644 .agents/.style.mdformat
 create mode 100644 .agents/skills/gtest-parallel/SKILL.md

diff --git a/.agents/.style.mdformat b/.agents/.style.mdformat
new file mode 100644
index 0000000000..e69de29bb2
diff --git a/.agents/skills/gtest-parallel/SKILL.md b/.agents/skills/gtest-parallel/SKILL.md
new file mode 100644
index 0000000000..f1e8b32d99
--- /dev/null
+++ b/.agents/skills/gtest-parallel/SKILL.md
@@ -0,0 +1,60 @@
+---
+name: gtest-parallel
+description: Run Google Test binaries in parallel using the gtest-parallel script. Use when needing to speed up test execution, run flaky tests with repeat, or filter specific tests.
+---
+
+# gtest-parallel
+
+`gtest-parallel` is a script that executes Google Test binaries in parallel,
+providing speedup for single-threaded tests and tests that do not run at 100%
+CPU.
+
+## Location
+
+The script is located at `third_party/gtest-parallel/gtest-parallel`.
+
+## Core Flags
+
+### Filtering Tests
+
+Use `--gtest_filter` to run a select set of tests. It supports the same syntax
+as Google Test (including exclusion with `-`).
+
+```bash
+third_party/gtest-parallel/gtest-parallel path/to/binary --gtest_filter=Foo.*:Bar.*
+```
+
+### Timeouts
+
+- `--timeout=TIMEOUT`: Interrupt all remaining processes after the given time
+  (in seconds).
+- `--timeout_per_test=TIMEOUT_PER_TEST`: Interrupt single processes after the
+  given time (in seconds).
+
+### Output and Logging
+
+- `-d OUTPUT_DIR`, `--output_dir=OUTPUT_DIR`: Output directory for test logs.
+  Logs will be available under `gtest-parallel-logs/` inside the specified
+  directory.
+- `--dump_json_test_results=DUMP_JSON_TEST_RESULTS`: Saves the results of the
+  tests as a JSON machine-readable file.
+
+## Advanced Usage
+
+### Repeating Tests (Flakiness Testing)
+
+Use `--repeat=N` to run tests multiple times.
+
+```bash
+third_party/gtest-parallel/gtest-parallel path/to/binary --repeat=1000
+```
+
+### Workers
+
+Use `-w WORKERS` or `--workers=WORKERS` to specify the number of parallel
+workers (defaults to the number of cores).
+
+### Serializing Test Cases
+
+Use `--serialize_test_cases` to run tests within the same test case sequentially
+(useful if they share resources).
diff --git a/api/DEPS b/api/DEPS
index 01d7057f7d..90b81d2dee 100644
--- a/api/DEPS
+++ b/api/DEPS
@@ -1,6 +1,7 @@
 # This is supposed to be a complete list of top-level directories,
 # excepting only api/ itself.
 include_rules = [
+  "-.agents",
   "-agents",
   "-audio",
   "-base",

From 84c9fda5cba7f74b17403f58fe7c910e0ec6c69e Mon Sep 17 00:00:00 2001
From: Jakob Ivarsson <jakobi@webrtc.org>
Date: Wed, 25 Mar 2026 10:30:44 +0000
Subject: [PATCH 407/472] Initialize NetEqNetworkStatistics fields.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Bug: None
Change-Id: I40fd4039db67d92de41ea637832840b1623c76ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/460000
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47260}
---
 api/neteq/neteq.h | 41 +++++++++++++++++++++--------------------
 1 file changed, 21 insertions(+), 20 deletions(-)

diff --git a/api/neteq/neteq.h b/api/neteq/neteq.h
index 0afb04cc4b..b0aed233f0 100644
--- a/api/neteq/neteq.h
+++ b/api/neteq/neteq.h
@@ -31,28 +31,29 @@ namespace webrtc {
 class AudioFrame;
 
 struct NetEqNetworkStatistics {
-  uint16_t current_buffer_size_ms;    // Current jitter buffer size in ms.
-  uint16_t preferred_buffer_size_ms;  // Target buffer size in ms.
-  uint16_t jitter_peaks_found;        // 1 if adding extra delay due to peaky
-                                      // jitter; 0 otherwise.
-  uint16_t expand_rate;         // Fraction (of original stream) of synthesized
-                                // audio inserted through expansion (in Q14).
-  uint16_t speech_expand_rate;  // Fraction (of original stream) of synthesized
-                                // speech inserted through expansion (in Q14).
-  uint16_t preemptive_rate;     // Fraction of data inserted through pre-emptive
-                                // expansion (in Q14).
-  uint16_t accelerate_rate;     // Fraction of data removed through acceleration
-                                // (in Q14).
-  uint16_t secondary_decoded_rate;    // Fraction of data coming from FEC/RED
-                                      // decoding (in Q14).
-  uint16_t secondary_discarded_rate;  // Fraction of discarded FEC/RED data (in
-                                      // Q14).
+  uint16_t current_buffer_size_ms = 0;    // Current jitter buffer size in ms.
+  uint16_t preferred_buffer_size_ms = 0;  // Target buffer size in ms.
+  uint16_t jitter_peaks_found =
+      0;  // 1 if adding extra delay due to peaky jitter; 0 otherwise.
+  uint16_t expand_rate = 0;  // Fraction (of original stream) of synthesized
+                             // audio inserted through expansion (in Q14).
+  uint16_t speech_expand_rate =
+      0;  // Fraction (of original stream) of synthesized speech inserted
+          // through expansion (in Q14).
+  uint16_t preemptive_rate =
+      0;  // Fraction of data inserted through preemptive expansion (in Q14).
+  uint16_t accelerate_rate =
+      0;  // Fraction of data removed through acceleration (in Q14).
+  uint16_t secondary_decoded_rate =
+      0;  // Fraction of data coming from FEC/RED decoding (in Q14).
+  uint16_t secondary_discarded_rate =
+      0;  // Fraction of discarded FEC/RED data (in Q14).
   // Statistics for packet waiting times, i.e., the time between a packet
   // arrives until it is decoded.
-  int mean_waiting_time_ms;
-  int median_waiting_time_ms;
-  int min_waiting_time_ms;
-  int max_waiting_time_ms;
+  int mean_waiting_time_ms = 0;
+  int median_waiting_time_ms = 0;
+  int min_waiting_time_ms = 0;
+  int max_waiting_time_ms = 0;
 };
 
 // NetEq statistics that persist over the lifetime of the class.

From 80dd204abacdeb45850ce62a1581f054cef3cbc2 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Wed, 25 Mar 2026 07:15:30 -0700
Subject: [PATCH 408/472] Roll chromium_revision d3079a981f..5b78ada5e8
 (1604639:1604779)

Change log: https://chromium.googlesource.com/chromium/src/+log/d3079a981f..5b78ada5e8
Full diff: https://chromium.googlesource.com/chromium/src/+/d3079a981f..5b78ada5e8

Changed dependencies
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/087d2d6a3f..8a3d0d636d
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/7274f92de1..578948b7b7
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/7ad8972446..c77362d590
* src/third_party/androidx/cipd: TUGHGHq_2U0C6NzCTxxwEIYyf1o0qWVn7Ta_T8xndiAC..Ht6zouRsLwQucvrnaYcrAOwHvmqKCS2-hoi5sQBaMjgC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/aa0acca1ac..5a706f2d40
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/54c1a44a6e..4fcf84ea39
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/ff09f3d9ee..6e9d681b49
DEPS diff: https://chromium.googlesource.com/chromium/src/+/d3079a981f..5b78ada5e8/DEPS

BUG=None

Change-Id: I412d14be4bfe2a1890aa42664309cb38759ea673
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/460100
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47261}
---
 DEPS | 16 ++++++++--------
 1 file changed, 8 insertions(+), 8 deletions(-)

diff --git a/DEPS b/DEPS
index b7df5b3b06..5102922197 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': 'd3079a981fd07f619d0c1fc18d30b9208749b94b',
+  'chromium_revision': '5b78ada5e8c55ae9835f554516689406049e8ac1',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@087d2d6a3fcdf817624bc6a547becb6a50892a0e',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@8a3d0d636df83b5152f5d1a1440e0191bfb25d9f',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@7274f92de1882295073078afb648515ced0185ea',
+    'https://chromium.googlesource.com/chromium/src/testing@578948b7b70c93754c5686cc5bbd0bbde44ca2fb',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@7ad8972446f6cc626c193804fdb71d5413b0cea7',
+    'https://chromium.googlesource.com/chromium/src/third_party@c77362d5905e1f64c64b219797d020c08fabe0ed',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -488,7 +488,7 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@aa0acca1acf36b7fb16aa8a7b60b97d5cb01404d',
+    'https://boringssl.googlesource.com/boringssl.git@5a706f2d40fd0fbe287ad0067aac886d078c75f0',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@8be0e3114685fcc1589561067282edf75ea1259a',
   'src/third_party/catapult':
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '54c1a44a6e524789175c80da61ccae18987dc903',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '4fcf84ea39cb3595ccec5f358bb8b77071507543',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@ff09f3d9ee8b2f215be6ecd7592757c98010403d',
+    'https://chromium.googlesource.com/chromium/src/tools@6e9d681b49648576c49aab10ed9289b74b158d8d',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'TUGHGHq_2U0C6NzCTxxwEIYyf1o0qWVn7Ta_T8xndiAC',
+          'version': 'Ht6zouRsLwQucvrnaYcrAOwHvmqKCS2-hoi5sQBaMjgC',
       },
     ],
     'condition': 'checkout_android and non_git_source',

From a32d5d87ac77978794dbe0fbcca076c72b1617a1 Mon Sep 17 00:00:00 2001
From: chromium-webrtc-autoroll
 <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Date: Wed, 25 Mar 2026 11:07:38 -0700
Subject: [PATCH 409/472] Roll chromium_revision 5b78ada5e8..a3f5fcb392
 (1604779:1604961)

Change log: https://chromium.googlesource.com/chromium/src/+log/5b78ada5e8..a3f5fcb392
Full diff: https://chromium.googlesource.com/chromium/src/+/5b78ada5e8..a3f5fcb392

Changed dependencies
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/4f3f3e34dd..dd54dd5186
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/8a3d0d636d..b3b1914b7b
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/578948b7b7..629b7bb605
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/c77362d590..f54c8ff588
* src/third_party/androidx/cipd: Ht6zouRsLwQucvrnaYcrAOwHvmqKCS2-hoi5sQBaMjgC..xbJffE4gDOA4g5njwbpWZtgk4ouGMWOHiqq2ymCZ2ggC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/5a706f2d40..8dce4fd20a
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/d7bc9113b8..5a34891efa
* src/third_party/fuzztest/src: https://chromium.googlesource.com/external/github.com/google/fuzztest.git/+log/36b8640bfd..dc32713409
* src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/4fcf84ea39..40b1342aa7
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/6e9d681b49..f363a79871
DEPS diff: https://chromium.googlesource.com/chromium/src/+/5b78ada5e8..a3f5fcb392/DEPS

BUG=None

Change-Id: I115d372a1cfdc2dc55b9e4c355e24a70d3e4bc5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/460140
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47262}
---
 DEPS | 22 +++++++++++-----------
 1 file changed, 11 insertions(+), 11 deletions(-)

diff --git a/DEPS b/DEPS
index 5102922197..56b4be5b95 100644
--- a/DEPS
+++ b/DEPS
@@ -10,7 +10,7 @@ vars = {
   # chromium waterfalls. More info at: crbug.com/570091.
   'checkout_configuration': 'default',
   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': '5b78ada5e8c55ae9835f554516689406049e8ac1',
+  'chromium_revision': 'a3f5fcb392f2902650ca2b71820e7e418787e18b',
 
   # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
   # process the raw profiles produced by instrumented targets (built with
@@ -69,7 +69,7 @@ vars = {
 
 deps = {
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@4f3f3e34ddfc6f4df4fa398cbe6996fcddc9bf3d',
+    'https://chromium.googlesource.com/chromium/src/build@dd54dd5186566a13bda647123c22540666b12ace',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@95ed44cf5f06dbb5861030b91c9db9ccb4316762',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
@@ -78,13 +78,13 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@8a3d0d636df83b5152f5d1a1440e0191bfb25d9f',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@b3b1914b7bd50a64ec13fabaf5e42edcf22e99b8',
     'condition': 'checkout_ios',
   },
   'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@578948b7b70c93754c5686cc5bbd0bbde44ca2fb',
+    'https://chromium.googlesource.com/chromium/src/testing@629b7bb6055714e23d8125bf790cfc8d94a94159',
   'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@c77362d5905e1f64c64b219797d020c08fabe0ed',
+    'https://chromium.googlesource.com/chromium/src/third_party@f54c8ff58874b977b8ba20c146dd312d2102168e',
 
   'src/buildtools/linux64': {
     'packages': [
@@ -488,11 +488,11 @@ deps = {
   },
 
   'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@5a706f2d40fd0fbe287ad0067aac886d078c75f0',
+    'https://boringssl.googlesource.com/boringssl.git@8dce4fd20ab7e768c0a5103edc1d8cb7e54366ba',
   'src/third_party/breakpad/breakpad':
     'https://chromium.googlesource.com/breakpad/breakpad.git@8be0e3114685fcc1589561067282edf75ea1259a',
   'src/third_party/catapult':
-    'https://chromium.googlesource.com/catapult.git@d7bc9113b86bdc56239ff7a7d5d1f773d37ca567',
+    'https://chromium.googlesource.com/catapult.git@5a34891efa6e41c8aca8842386b8ee528963ffdf',
   'src/third_party/ced/src': {
     'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5',
   },
@@ -593,7 +593,7 @@ deps = {
   'src/third_party/libFuzzer/src':
     'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt/lib/fuzzer.git@bea408a6e01f0f7e6c82a43121fe3af4506c932e',
   'src/third_party/fuzztest/src':
-    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@36b8640bfd7cd2cf868cd3989cf1e8e30b304be4',
+    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@dc327134097700121e4ecd6e1d54d1d0a832a18d',
   'src/third_party/libjpeg_turbo':
     'https://chromium.googlesource.com/chromium/deps/libjpeg_turbo.git@d1f5f2393e0d51f840207342ae86e55a86443288',
   'src/third_party/libsrtp':
@@ -609,7 +609,7 @@ deps = {
       'condition': 'checkout_android',
   },
   'src/third_party/perfetto':
-    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '4fcf84ea39cb3595ccec5f358bb8b77071507543',
+    Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '40b1342aa7bd47d9c963c3617fd98ec1551528f9',
   'src/third_party/protobuf-javascript/src':
     Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e6d763860001ba1a76a63adcff5efb12b1c96024',
   'src/third_party/libvpx/source/libvpx':
@@ -668,7 +668,7 @@ deps = {
     'condition': 'checkout_android',
   },
   'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@6e9d681b49648576c49aab10ed9289b74b158d8d',
+    'https://chromium.googlesource.com/chromium/src/tools@f363a79871a91f36322e845e3134e2f04e1fc18a',
 
   'src/third_party/espresso': {
       'packages': [
@@ -707,7 +707,7 @@ deps = {
     'packages': [
       {
           'package': 'chromium/third_party/androidx',
-          'version': 'Ht6zouRsLwQucvrnaYcrAOwHvmqKCS2-hoi5sQBaMjgC',
+          'version': 'xbJffE4gDOA4g5njwbpWZtgk4ouGMWOHiqq2ymCZ2ggC',
       },
     ],
     'condition': 'checkout_android and non_git_source',

From 246b8dee2304df4a20513f595e315a0ca3df079c Mon Sep 17 00:00:00 2001
From: Philipp Hancke <philipp.hancke@googlemail.com>
Date: Wed, 25 Mar 2026 16:40:32 +0100
Subject: [PATCH 410/472] build: clean up modules audio build files

using gn_check_autofix with cherry-picking removals

Bug: webrtc:467294026
Change-Id: Ia8fe74995724c193d7e92db1ed3e26d77e34422a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/460060
Reviewed-by: Christoffer Dewerin <jansson@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#47263}
---
 modules/audio_coding/BUILD.gn                 | 61 -------------------
 .../audio_coding/codecs/opus/test/BUILD.gn    |  2 -
 modules/audio_device/BUILD.gn                 | 10 ---
 modules/audio_mixer/BUILD.gn                  |  9 ---
 modules/audio_processing/BUILD.gn             | 33 ----------
 modules/audio_processing/aec3/BUILD.gn        |  7 ---
 modules/audio_processing/aec_dump/BUILD.gn    |  7 ---
 modules/audio_processing/agc/BUILD.gn         | 10 ---
 modules/audio_processing/agc2/BUILD.gn        | 24 --------
 .../audio_processing/agc2/rnn_vad/BUILD.gn    |  4 --
 .../capture_levels_adjuster/BUILD.gn          |  2 -
 .../audio_processing/capture_mixer/BUILD.gn   |  3 -
 modules/audio_processing/ns/BUILD.gn          | 18 ------
 .../test/conversational_speech/BUILD.gn       |  1 -
 modules/audio_processing/utility/BUILD.gn     |  1 -
 modules/audio_processing/vad/BUILD.gn         |  2 -
 16 files changed, 194 deletions(-)

diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index d8e1e9788a..6c23fca430 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -36,27 +36,19 @@ rtc_library("audio_coding") {
 
   deps = [
     ":audio_coding_module_typedefs",
-    ":neteq",
-    "..:module_api",
     "..:module_api_public",
     "../../api:array_view",
     "../../api:function_view",
     "../../api/audio:audio_frame_api",
     "../../api/audio_codecs:audio_codecs_api",
-    "../../api/environment",
-    "../../api/neteq:default_neteq_factory",
-    "../../api/neteq:neteq_api",
-    "../../api/units:timestamp",
     "../../audio/utility:audio_frame_operations",
     "../../common_audio",
-    "../../common_audio:common_audio_c",
     "../../rtc_base:buffer",
     "../../rtc_base:checks",
     "../../rtc_base:logging",
     "../../rtc_base:macromagic",
     "../../rtc_base:safe_conversions",
     "../../rtc_base/synchronization:mutex",
-    "../../system_wrappers",
     "../../system_wrappers:metrics",
     "//third_party/abseil-cpp/absl/algorithm:container",
     "//third_party/abseil-cpp/absl/strings",
@@ -124,11 +116,9 @@ rtc_library("red") {
     "../../api:field_trials_view",
     "../../api/audio_codecs:audio_codecs_api",
     "../../api/units:time_delta",
-    "../../common_audio",
     "../../rtc_base:buffer",
     "../../rtc_base:byte_order",
     "../../rtc_base:checks",
-    "../../rtc_base:logging",
     "//third_party/abseil-cpp/absl/strings:string_view",
   ]
 }
@@ -181,7 +171,6 @@ rtc_library("g722") {
     "../../api/units:time_delta",
     "../../rtc_base:buffer",
     "../../rtc_base:checks",
-    "../../rtc_base:safe_conversions",
   ]
   public_deps += [ ":g722_c" ]  # no-presubmit-check TODO(webrtc:8603)
 }
@@ -213,7 +202,6 @@ rtc_library("isac_vad") {
   deps = [
     ":isac_bwinfo",
     "../../rtc_base:compile_assert_c",
-    "../../rtc_base/system:arch",
     "../../rtc_base/system:ignore_warnings",
     "../third_party/fft",
   ]
@@ -239,7 +227,6 @@ rtc_library("pcm16b") {
   deps = [
     ":g711",
     ":legacy_encoded_audio_frame",
-    "../../api:array_view",
     "../../api/audio_codecs:audio_codecs_api",
     "../../rtc_base:buffer",
     "../../rtc_base:checks",
@@ -265,7 +252,6 @@ rtc_library("audio_coding_opus_common") {
     "../../api:array_view",
     "../../api/audio_codecs:audio_codecs_api",
     "../../rtc_base:buffer",
-    "../../rtc_base:checks",
     "../../rtc_base:stringutils",
     "//third_party/abseil-cpp/absl/strings:string_view",
   ]
@@ -333,7 +319,6 @@ rtc_library("webrtc_multiopus") {
     "../../rtc_base:checks",
     "../../rtc_base:logging",
     "../../rtc_base:safe_conversions",
-    "../../rtc_base:safe_minmax",
     "../../rtc_base:stringutils",
     "//third_party/abseil-cpp/absl/memory",
     "//third_party/abseil-cpp/absl/strings",
@@ -517,7 +502,6 @@ rtc_library("neteq") {
   ]
 
   deps = [
-    ":audio_coding_module_typedefs",
     ":webrtc_cng",
     "..:module_api_public",
     "../../api:array_view",
@@ -534,21 +518,17 @@ rtc_library("neteq") {
     "../../api/neteq:tick_timer",
     "../../api/units:time_delta",
     "../../api/units:timestamp",
-    "../../common_audio",
     "../../common_audio:common_audio_c",
     "../../common_audio:common_audio_cc",
     "../../rtc_base:buffer",
     "../../rtc_base:checks",
     "../../rtc_base:event_tracer",
-    "../../rtc_base:gtest_prod",
     "../../rtc_base:logging",
-    "../../rtc_base:macromagic",
     "../../rtc_base:rtc_numerics",
     "../../rtc_base:safe_conversions",
     "../../rtc_base:safe_minmax",
     "../../rtc_base:sanitizer",
     "../../rtc_base/experiments:field_trial_parser",
-    "../../rtc_base/synchronization:mutex",
     "../../system_wrappers",
     "../../system_wrappers:metrics",
     "//third_party/abseil-cpp/absl/strings",
@@ -576,9 +556,7 @@ rtc_library("neteq_tools_minimal") {
   ]
 
   deps = [
-    ":neteq",
     "../../api:array_view",
-    "../../api:field_trials",
     "../../api:field_trials_view",
     "../../api:neteq_simulator_api",
     "../../api:rtp_headers",
@@ -587,14 +565,11 @@ rtc_library("neteq_tools_minimal") {
     "../../api/audio_codecs:audio_codecs_api",
     "../../api/environment",
     "../../api/environment:environment_factory",
-    "../../api/neteq:custom_neteq_factory",
-    "../../api/neteq:default_neteq_controller_factory",
     "../../api/neteq:default_neteq_factory",
     "../../api/neteq:neteq_api",
     "../../api/units:timestamp",
     "../../rtc_base:buffer",
     "../../rtc_base:checks",
-    "../../rtc_base:copy_on_write_buffer",
     "../../rtc_base:safe_conversions",
     "../../rtc_base:stringutils",
     "../../system_wrappers",
@@ -663,7 +638,6 @@ rtc_library("neteq_tools") {
   deps = [
     ":neteq_input_audio_tools",
     ":neteq_tools_minimal",
-    "..:module_api_public",
     "../../api:array_view",
     "../../api:rtp_headers",
     "../../api:rtp_packet_info",
@@ -678,7 +652,6 @@ rtc_library("neteq_tools") {
     "../../rtc_base:safe_conversions",
     "../../rtc_base:stringutils",
     "../../rtc_base:timeutils",
-    "../rtp_rtcp",
     "../rtp_rtcp:rtp_rtcp_format",
     "//third_party/abseil-cpp/absl/strings:string_view",
   ]
@@ -868,7 +841,6 @@ if (rtc_include_tests) {
       ":pcm16b_c",
       ":red",
       "../../api:array_view",
-      "../../api:field_trials",
       "../../api:rtp_headers",
       "../../api:scoped_refptr",
       "../../api/audio:audio_frame_api",
@@ -923,8 +895,6 @@ if (rtc_include_tests) {
       "../../api/test/metrics:global_metrics_logger_and_exporter",
       "../../api/test/metrics:metric",
       "../../rtc_base:buffer",
-      "../../rtc_base:timeutils",
-      "../../system_wrappers",
       "../../test:fileutils",
       "../../test:test_flags",
       "../../test:test_support",
@@ -943,9 +913,7 @@ if (rtc_include_tests) {
 
     deps = [
       ":audio_coding",
-      ":neteq_tools",
       ":neteq_tools_minimal",
-      "../../api:array_view",
       "../../api:scoped_refptr",
       "../../api/audio_codecs:audio_codecs_api",
       "../../api/audio_codecs:builtin_audio_decoder_factory",
@@ -973,18 +941,13 @@ if (rtc_include_tests) {
       ":audio_coding",
       ":audio_coding_module_typedefs",
       ":neteq_input_audio_tools",
-      ":neteq_tools",
       ":neteq_tools_minimal",
       "../../api/audio:audio_frame_api",
       "../../api/audio_codecs:audio_codecs_api",
-      "../../api/audio_codecs:builtin_audio_decoder_factory",
       "../../api/audio_codecs:builtin_audio_encoder_factory",
       "../../api/environment",
       "../../api/environment:environment_factory",
       "../../rtc_base:checks",
-      "../../rtc_base:copy_on_write_buffer",
-      "../../rtc_base:stringutils",
-      "../../test:test_support",
       "../rtp_rtcp:rtp_rtcp_format",
       "//testing/gtest",
       "//third_party/abseil-cpp/absl/strings",
@@ -1045,14 +1008,12 @@ if (rtc_include_tests) {
       visibility += webrtc_default_visibility
       defines = audio_codec_defines
       deps = [
-        ":neteq",
         ":neteq_input_audio_tools",
         ":neteq_test_tools",
         ":neteq_tools",
         ":neteq_tools_minimal",
         "../../api:field_trials",
         "../../api:make_ref_counted",
-        "../../api:rtp_headers",
         "../../api:scoped_refptr",
         "../../api/audio_codecs:audio_codecs_api",
         "../../api/audio_codecs:builtin_audio_decoder_factory",
@@ -1060,10 +1021,7 @@ if (rtc_include_tests) {
         "../../api/neteq:neteq_api",
         "../../logging:rtc_event_log_parser",
         "../../rtc_base:checks",
-        "../../rtc_base:refcount",
         "../../test:audio_test_common",
-        "../../test:fileutils",
-        "../../test:test_support",
         "../rtp_rtcp:rtp_rtcp_format",
         "//third_party/abseil-cpp/absl/strings:string_view",
       ]
@@ -1165,7 +1123,6 @@ if (rtc_include_tests) {
       "../../rtc_base:checks",
       "../../system_wrappers",
       "../../test:fileutils",
-      "../../test:test_support",
       "//testing/gtest",
     ]
   }
@@ -1179,7 +1136,6 @@ if (rtc_include_tests) {
       ]
 
       deps = [
-        ":neteq",
         ":neteq_input_audio_tools",
         ":neteq_test_tools",
         ":neteq_tools_minimal",
@@ -1195,7 +1151,6 @@ if (rtc_include_tests) {
         "../../rtc_base:buffer",
         "../../rtc_base:checks",
         "../../rtc_base:stringutils",
-        "../../system_wrappers",
         "../../test:fileutils",
         "../../test:test_support",
         "//testing/gtest",
@@ -1210,7 +1165,6 @@ if (rtc_include_tests) {
       deps = [
         ":audio_coding",
         ":audio_coding_module_typedefs",
-        ":audio_encoder_cng",
         ":neteq_input_audio_tools",
         "../../api/audio:audio_frame_api",
         "../../api/audio_codecs/L16:audio_encoder_L16",
@@ -1267,10 +1221,7 @@ if (rtc_include_tests) {
       sources = [ "neteq/tools/rtp_analyze.cc" ]
 
       deps = [
-        ":neteq",
         ":neteq_test_tools",
-        ":neteq_tools_minimal",
-        ":pcm16b",
         "../../api:rtp_headers",
         "../../rtc_base:checks",
         "../rtp_rtcp:rtp_rtcp_format",
@@ -1308,10 +1259,8 @@ if (rtc_include_tests) {
       sources = [ "neteq/test/neteq_speed_test.cc" ]
 
       deps = [
-        ":neteq",
         ":neteq_test_support",
         "../../rtc_base:checks",
-        "../../test:test_support",
         "//third_party/abseil-cpp/absl/flags:flag",
         "//third_party/abseil-cpp/absl/flags:parse",
       ]
@@ -1324,14 +1273,12 @@ if (rtc_include_tests) {
 
       deps = [
         ":g711",
-        ":neteq",
         ":neteq_quality_test_support",
         "../../api:array_view",
         "../../api/audio_codecs:audio_codecs_api",
         "../../rtc_base:buffer",
         "../../rtc_base:checks",
         "../../rtc_base:safe_conversions",
-        "../../test:fileutils",
         "../../test:test_support",
         "//testing/gtest",
         "//third_party/abseil-cpp/absl/flags:flag",
@@ -1344,7 +1291,6 @@ if (rtc_include_tests) {
       sources = [ "neteq/test/neteq_pcm16b_quality_test.cc" ]
 
       deps = [
-        ":neteq",
         ":neteq_quality_test_support",
         ":pcm16b",
         "../../api:array_view",
@@ -1352,7 +1298,6 @@ if (rtc_include_tests) {
         "../../rtc_base:buffer",
         "../../rtc_base:checks",
         "../../rtc_base:safe_conversions",
-        "../../test:fileutils",
         "../../test:test_support",
         "//testing/gtest",
         "//third_party/abseil-cpp/absl/flags:flag",
@@ -1528,18 +1473,12 @@ if (rtc_include_tests) {
         "../../logging:rtc_event_audio",
         "../../rtc_base:buffer",
         "../../rtc_base:checks",
-        "../../rtc_base:copy_on_write_buffer",
         "../../rtc_base:digest",
         "../../rtc_base:ip_address",
-        "../../rtc_base:macromagic",
-        "../../rtc_base:platform_thread",
         "../../rtc_base:random",
         "../../rtc_base:rtc_base_tests_utils",
-        "../../rtc_base:rtc_event",
         "../../rtc_base:safe_conversions",
         "../../rtc_base:stringutils",
-        "../../rtc_base:threading",
-        "../../rtc_base/synchronization:mutex",
         "../../rtc_base/system:arch",
         "../../system_wrappers",
         "../../test:audio_codec_mocks",
diff --git a/modules/audio_coding/codecs/opus/test/BUILD.gn b/modules/audio_coding/codecs/opus/test/BUILD.gn
index 0b2fc17a36..37cf935be7 100644
--- a/modules/audio_coding/codecs/opus/test/BUILD.gn
+++ b/modules/audio_coding/codecs/opus/test/BUILD.gn
@@ -46,9 +46,7 @@ if (rtc_include_tests) {
     deps = [
       ":test",
       "../../../../../common_audio",
-      "../../../../../common_audio:common_audio_c",
       "../../../../../rtc_base:checks",
-      "../../../../../rtc_base:macromagic",
       "../../../../../test:test_support",
       "//testing/gtest",
     ]
diff --git a/modules/audio_device/BUILD.gn b/modules/audio_device/BUILD.gn
index bad43a5c91..6cd3749f63 100644
--- a/modules/audio_device/BUILD.gn
+++ b/modules/audio_device/BUILD.gn
@@ -75,7 +75,6 @@ rtc_library("audio_device_buffer") {
     "../../rtc_base:timeutils",
     "../../rtc_base/synchronization:mutex",
     "../../rtc_base/system:no_unique_address",
-    "../../system_wrappers",
     "../../system_wrappers:metrics",
   ]
 }
@@ -88,7 +87,6 @@ rtc_library("audio_device_generic") {
   deps = [
     ":audio_device_buffer",
     "../../api/audio:audio_device",
-    "../../rtc_base:logging",
   ]
 }
 
@@ -196,11 +194,8 @@ if (!build_with_chromium) {
       "../../common_audio",
       "../../rtc_base:buffer",
       "../../rtc_base:checks",
-      "../../rtc_base:logging",
       "../../rtc_base:macromagic",
-      "../../rtc_base:platform_thread",
       "../../rtc_base:random",
-      "../../rtc_base:rtc_event",
       "../../rtc_base:safe_conversions",
       "../../rtc_base:timeutils",
       "../../rtc_base/synchronization:mutex",
@@ -247,7 +242,6 @@ if (!build_with_chromium) {
       "../../rtc_base:platform_thread",
       "../../rtc_base:stringutils",
       "../../rtc_base:threading",
-      "../../rtc_base:timeutils",
       "../../rtc_base/synchronization:mutex",
       "../../rtc_base/system:file_wrapper",
       "//third_party/abseil-cpp/absl/strings:string_view",
@@ -455,7 +449,6 @@ if (rtc_include_tests && !build_with_chromium) {
       "test_audio_device_impl_test.cc",
     ]
     deps = [
-      ":audio_device",
       ":audio_device_buffer",
       ":audio_device_generic",
       ":audio_device_impl",
@@ -468,8 +461,6 @@ if (rtc_include_tests && !build_with_chromium) {
       "../../api/audio:create_audio_device_module",
       "../../api/environment",
       "../../api/environment:environment_factory",
-      "../../api/task_queue",
-      "../../api/task_queue:default_task_queue_factory",
       "../../api/units:time_delta",
       "../../api/units:timestamp",
       "../../common_audio",
@@ -483,7 +474,6 @@ if (rtc_include_tests && !build_with_chromium) {
       "../../rtc_base:task_queue_for_test",
       "../../rtc_base:timeutils",
       "../../rtc_base/synchronization:mutex",
-      "../../system_wrappers",
       "../../test:create_test_environment",
       "../../test:fileutils",
       "../../test:test_support",
diff --git a/modules/audio_mixer/BUILD.gn b/modules/audio_mixer/BUILD.gn
index d2fa8dbaec..7dc868a242 100644
--- a/modules/audio_mixer/BUILD.gn
+++ b/modules/audio_mixer/BUILD.gn
@@ -46,21 +46,14 @@ rtc_library("audio_mixer_impl") {
     "../../api/audio:audio_frame_api",
     "../../api/audio:audio_mixer_api",
     "../../api/audio:audio_processing",
-    "../../api/audio:audio_processing",
-    "../../audio/utility:audio_frame_operations",
     "../../common_audio",
     "../../rtc_base:checks",
     "../../rtc_base:event_tracer",
     "../../rtc_base:logging",
     "../../rtc_base:macromagic",
-    "../../rtc_base:race_checker",
-    "../../rtc_base:refcount",
-    "../../rtc_base:safe_conversions",
     "../../rtc_base/synchronization:mutex",
-    "../../system_wrappers",
     "../../system_wrappers:metrics",
     "../audio_processing:apm_logging",
-    "../audio_processing:audio_frame_view",
     "../audio_processing/agc2:fixed_digital",
   ]
 }
@@ -95,8 +88,6 @@ if (rtc_include_tests) {
     ]
 
     deps = [
-      ":audio_frame_manipulator",
-      ":audio_mixer_impl",
       "../../api:array_view",
       "../../api/audio:audio_frame_api",
       "../../rtc_base:checks",
diff --git a/modules/audio_processing/BUILD.gn b/modules/audio_processing/BUILD.gn
index 2997c457a3..3116c2618c 100644
--- a/modules/audio_processing/BUILD.gn
+++ b/modules/audio_processing/BUILD.gn
@@ -32,7 +32,6 @@ rtc_library("audio_frame_proxies") {
     "include/audio_frame_proxies.h",
   ]
   deps = [
-    ":audio_frame_view",
     "../../api/audio:audio_frame_api",
     "../../api/audio:audio_processing",
     "../../api/audio:audio_processing_statistics",
@@ -138,7 +137,6 @@ rtc_library("gain_controller2") {
     "../../common_audio",
     "../../rtc_base:checks",
     "../../rtc_base:logging",
-    "../../rtc_base:stringutils",
     "agc2:adaptive_digital_gain_controller",
     "agc2:common",
     "agc2:cpu_features",
@@ -169,15 +167,12 @@ rtc_library("audio_processing") {
     ":apm_logging",
     ":audio_buffer",
     ":audio_frame_proxies",
-    ":audio_frame_view",
     ":gain_controller2",
     ":high_pass_filter",
     ":post_filter",
     ":rms_level",
     "../../api:array_view",
     "../../api:field_trials_view",
-    "../../api:function_view",
-    "../../api:make_ref_counted",
     "../../api:scoped_refptr",
     "../../api/audio:aec3_config",
     "../../api/audio:audio_frame_api",
@@ -187,25 +182,16 @@ rtc_library("audio_processing") {
     "../../api/audio:neural_residual_echo_estimator_api",
     "../../api/environment",
     "../../api/task_queue",
-    "../../audio/utility:audio_frame_operations",
     "../../common_audio",
-    "../../common_audio:common_audio_c",
-    "../../common_audio:fir_filter",
-    "../../common_audio:fir_filter_factory",
-    "../../common_audio/third_party/ooura:fft_size_256",
     "../../rtc_base:checks",
     "../../rtc_base:denormal_disabler",
     "../../rtc_base:event_tracer",
     "../../rtc_base:gtest_prod",
     "../../rtc_base:logging",
     "../../rtc_base:macromagic",
-    "../../rtc_base:safe_minmax",
-    "../../rtc_base:sanitizer",
     "../../rtc_base:swap_queue",
     "../../rtc_base:timeutils",
     "../../rtc_base/synchronization:mutex",
-    "../../rtc_base/system:rtc_export",
-    "../../system_wrappers",
     "../../system_wrappers:metrics",
     "aec3",
     "aec_dump",
@@ -309,8 +295,6 @@ if (rtc_include_tests) {
     sources = [ "include/mock_audio_processing.h" ]
     deps = [
       ":aec_dump_interface",
-      ":audio_buffer",
-      ":audio_processing",
       "../../api:array_view",
       "../../api:scoped_refptr",
       "../../api/audio:audio_processing",
@@ -368,7 +352,6 @@ if (rtc_include_tests) {
         "../../api:ref_count",
         "../../api:scoped_refptr",
         "../../api/audio:aec3_config",
-        "../../api/audio:aec3_factory",
         "../../api/audio:audio_frame_api",
         "../../api/audio:audio_processing",
         "../../api/audio:audio_processing_statistics",
@@ -378,11 +361,8 @@ if (rtc_include_tests) {
         "../../api/environment:environment_factory",
         "../../api/units:time_delta",
         "../../common_audio",
-        "../../common_audio:common_audio_c",
         "../../rtc_base:checks",
         "../../rtc_base:cpu_info",
-        "../../rtc_base:denormal_disabler",
-        "../../rtc_base:gtest_prod",
         "../../rtc_base:macromagic",
         "../../rtc_base:platform_thread",
         "../../rtc_base:protobuf_utils",
@@ -396,7 +376,6 @@ if (rtc_include_tests) {
         "../../rtc_base:task_queue_for_test",
         "../../rtc_base:threading",
         "../../rtc_base/synchronization:mutex",
-        "../../rtc_base/system:arch",
         "../../rtc_base/system:file_wrapper",
         "../../test:fileutils",
         "../../test:rtc_expect_death",
@@ -407,7 +386,6 @@ if (rtc_include_tests) {
         "agc:gain_control_interface",
         "agc2:adaptive_digital_gain_controller_unittest",
         "agc2:biquad_filter_unittests",
-        "agc2:cpu_features",
         "agc2:fixed_digital_unittests",
         "agc2:gain_applier_unittest",
         "agc2:input_volume_controller",
@@ -417,10 +395,8 @@ if (rtc_include_tests) {
         "agc2:saturation_protector_unittest",
         "agc2:speech_level_estimator_unittest",
         "agc2:test_utils",
-        "agc2:vad_wrapper",
         "agc2:vad_wrapper_unittests",
         "agc2/rnn_vad:unittests",
-        "capture_levels_adjuster",
         "capture_levels_adjuster:capture_levels_adjuster_unittests",
         "capture_mixer:capture_mixer_unittests",
         "test/conversational_speech:unittest",
@@ -496,10 +472,8 @@ if (rtc_include_tests) {
       "../../api/units:time_delta",
       "../../rtc_base:checks",
       "../../rtc_base:platform_thread",
-      "../../rtc_base:protobuf_utils",
       "../../rtc_base:random",
       "../../rtc_base:rtc_event",
-      "../../rtc_base:safe_conversions",
       "../../system_wrappers",
       "../../test:test_support",
       "//third_party/abseil-cpp/absl/strings:string_view",
@@ -513,7 +487,6 @@ if (rtc_include_tests) {
     ]
     deps = [
       "../../api:array_view",
-      "../../api/audio:audio_frame_api",
       "../../common_audio",
       "../../rtc_base:checks",
       "../../rtc_base:logging",
@@ -547,14 +520,12 @@ if (rtc_include_tests) {
         ":aec3_config_json",
         ":analog_mic_simulation",
         ":apm_logging",
-        ":audio_processing",
         ":audioproc_debug_proto",
         ":audioproc_protobuf_utils",
         ":audioproc_test_utils",
         "../../api:field_trials",
         "../../api:scoped_refptr",
         "../../api/audio:aec3_config",
-        "../../api/audio:aec3_factory",
         "../../api/audio:audio_processing",
         "../../api/audio:builtin_audio_processing_builder",
         "../../api/audio:echo_detector_creator",
@@ -595,7 +566,6 @@ if (rtc_include_tests) {
 
       deps = [
         ":audioproc_debug_proto",
-        "../../rtc_base:checks",
         "../../rtc_base:protobuf_utils",
         "../../rtc_base/system:arch",
       ]
@@ -638,14 +608,12 @@ rtc_library("audioproc_test_utils") {
 
   deps = [
     ":audio_buffer",
-    ":audio_processing",
     "../../api:array_view",
     "../../api/audio:audio_frame_api",
     "../../api/audio:audio_processing",
     "../../common_audio",
     "../../rtc_base:checks",
     "../../rtc_base:random",
-    "../../rtc_base/system:arch",
     "../../system_wrappers",
     "../../test:fileutils",
     "../../test:test_support",
@@ -668,7 +636,6 @@ rtc_library("aec3_config_json") {
     "../../rtc_base:logging",
     "../../rtc_base:rtc_json",
     "../../rtc_base:stringutils",
-    "../../rtc_base/system:rtc_export",
     "//third_party/abseil-cpp/absl/strings:string_view",
   ]
 }
diff --git a/modules/audio_processing/aec3/BUILD.gn b/modules/audio_processing/aec3/BUILD.gn
index 6b5279b56a..f52017d935 100644
--- a/modules/audio_processing/aec3/BUILD.gn
+++ b/modules/audio_processing/aec3/BUILD.gn
@@ -148,7 +148,6 @@ rtc_library("aec3") {
     "../../../api/audio:echo_control",
     "../../../api/audio:neural_residual_echo_estimator_api",
     "../../../api/environment",
-    "../../../common_audio:common_audio_c",
     "../../../rtc_base:checks",
     "../../../rtc_base:cpu_info",
     "../../../rtc_base:gtest_prod",
@@ -185,7 +184,6 @@ rtc_source_set("aec3_fft") {
     ":fft_data",
     "../../../common_audio/third_party/ooura:fft_size_128",
     "../../../rtc_base:checks",
-    "../../../rtc_base/system:arch",
   ]
 }
 
@@ -201,7 +199,6 @@ rtc_source_set("render_buffer") {
     ":aec3_common",
     ":fft_data",
     "../../../rtc_base:checks",
-    "../../../rtc_base/system:arch",
   ]
 }
 
@@ -230,7 +227,6 @@ rtc_source_set("matched_filter") {
   sources = [ "matched_filter.h" ]
   deps = [
     ":aec3_common",
-    "../../../rtc_base:gtest_prod",
     "../../../rtc_base/system:arch",
   ]
 }
@@ -315,7 +311,6 @@ if (rtc_include_tests) {
       ":vector_math",
       "..:apm_logging",
       "..:audio_buffer",
-      "..:audio_processing",
       "..:high_pass_filter",
       "../../../api:field_trials",
       "../../../api/audio:aec3_config",
@@ -325,14 +320,12 @@ if (rtc_include_tests) {
       "../../../api/environment:environment_factory",
       "../../../rtc_base:checks",
       "../../../rtc_base:cpu_info",
-      "../../../rtc_base:gunit_helpers",
       "../../../rtc_base:random",
       "../../../rtc_base:safe_minmax",
       "../../../rtc_base:stringutils",
       "../../../rtc_base/system:arch",
       "../../../system_wrappers:metrics",
       "../../../test:create_test_field_trials",
-      "../../../test:fileutils",
       "../../../test:test_support",
       "../utility:cascaded_biquad_filter",
     ]
diff --git a/modules/audio_processing/aec_dump/BUILD.gn b/modules/audio_processing/aec_dump/BUILD.gn
index 227d6dec6b..586a38ed79 100644
--- a/modules/audio_processing/aec_dump/BUILD.gn
+++ b/modules/audio_processing/aec_dump/BUILD.gn
@@ -33,8 +33,6 @@ if (rtc_include_tests) {
     deps = [
       "..:aec_dump_interface",
       "..:audio_frame_view",
-      "..:audioproc_test_utils",
-      "../",
       "../../../api/audio:audio_frame_api",
       "../../../api/audio:audio_processing",
       "../../../test:test_support",
@@ -48,8 +46,6 @@ if (rtc_include_tests) {
 
     deps = [
       ":mock_aec_dump",
-      "..:audioproc_test_utils",
-      "../",
       "../../../api:scoped_refptr",
       "../../../api/audio:audio_processing",
       "../../../api/audio:builtin_audio_processing_builder",
@@ -98,10 +94,7 @@ if (rtc_enable_protobuf) {
       defines = []
       deps = [
         ":aec_dump",
-        ":aec_dump_impl",
         "..:aec_dump_interface",
-        "..:audioproc_debug_proto",
-        "../",
         "../../../api/audio:audio_processing",
         "../../../rtc_base:task_queue_for_test",
         "../../../test:fileutils",
diff --git a/modules/audio_processing/agc/BUILD.gn b/modules/audio_processing/agc/BUILD.gn
index 090fbd4604..f5bad6f189 100644
--- a/modules/audio_processing/agc/BUILD.gn
+++ b/modules/audio_processing/agc/BUILD.gn
@@ -29,7 +29,6 @@ rtc_library("agc") {
     "../../../api/audio:audio_processing",
     "../../../api/environment",
     "../../../common_audio",
-    "../../../common_audio:common_audio_c",
     "../../../rtc_base:checks",
     "../../../rtc_base:gtest_prod",
     "../../../rtc_base:logging",
@@ -38,7 +37,6 @@ rtc_library("agc") {
     "../agc2:clipping_predictor",
     "../agc2:gain_map",
     "../agc2:input_volume_stats_reporter",
-    "../vad",
   ]
 }
 
@@ -75,12 +73,9 @@ rtc_library("legacy_agc") {
   ]
 
   deps = [
-    "../../../common_audio",
     "../../../common_audio:common_audio_c",
     "../../../common_audio:common_audio_cc",
-    "../../../common_audio/third_party/ooura:fft_size_256",
     "../../../rtc_base:checks",
-    "../../../system_wrappers",
   ]
 
   if (rtc_build_with_neon) {
@@ -107,18 +102,13 @@ if (rtc_include_tests) {
       ":gain_control_interface",
       ":level_estimation",
       "..:audio_buffer",
-      "..:mocks",
       "../../../api:array_view",
-      "../../../api:field_trials",
       "../../../api/audio:audio_processing",
       "../../../api/environment",
       "../../../api/environment:environment_factory",
       "../../../rtc_base:checks",
-      "../../../rtc_base:random",
-      "../../../rtc_base:safe_conversions",
       "../../../rtc_base:safe_minmax",
       "../../../rtc_base:stringutils",
-      "../../../system_wrappers:metrics",
       "../../../test:create_test_field_trials",
       "../../../test:fileutils",
       "../../../test:test_support",
diff --git a/modules/audio_processing/agc2/BUILD.gn b/modules/audio_processing/agc2/BUILD.gn
index 0de1c32bd7..4a6b6eaaa9 100644
--- a/modules/audio_processing/agc2/BUILD.gn
+++ b/modules/audio_processing/agc2/BUILD.gn
@@ -28,7 +28,6 @@ rtc_library("speech_level_estimator") {
   deps = [
     ":common",
     "..:apm_logging",
-    "../../../api:array_view",
     "../../../api:field_trials_view",
     "../../../api/audio:audio_processing",
     "../../../rtc_base:checks",
@@ -149,10 +148,8 @@ rtc_library("fixed_digital") {
   deps = [
     ":common",
     "..:apm_logging",
-    "..:audio_frame_view",
     "../../../api:array_view",
     "../../../api/audio:audio_frame_api",
-    "../../../common_audio",
     "../../../rtc_base:checks",
     "../../../rtc_base:gtest_prod",
     "../../../rtc_base:safe_conversions",
@@ -216,7 +213,6 @@ rtc_library("input_volume_controller") {
     ":input_volume_stats_reporter",
     "..:audio_buffer",
     "..:audio_frame_view",
-    "../../../api:array_view",
     "../../../api:field_trials_view",
     "../../../api/audio:audio_processing",
     "../../../rtc_base:checks",
@@ -233,11 +229,9 @@ rtc_library("noise_level_estimator") {
     "noise_level_estimator.h",
   ]
   deps = [
-    ":biquad_filter",
     "..:apm_logging",
     "../../../api/audio:audio_frame_api",
     "../../../rtc_base:checks",
-    "../../../system_wrappers",
   ]
 
   visibility = [
@@ -310,7 +304,6 @@ rtc_library("speech_level_estimator_unittest") {
     "..:apm_logging",
     "../../../api/audio:audio_processing",
     "../../../rtc_base:checks",
-    "../../../rtc_base:gunit_helpers",
     "../../../test:test_support",
   ]
 }
@@ -326,10 +319,8 @@ rtc_library("adaptive_digital_gain_controller_unittest") {
     ":common",
     ":test_utils",
     "..:apm_logging",
-    "..:audio_frame_view",
     "../../../api/audio:audio_processing",
     "../../../common_audio",
-    "../../../rtc_base:gunit_helpers",
     "../../../test:test_support",
   ]
 }
@@ -343,7 +334,6 @@ rtc_library("gain_applier_unittest") {
     ":gain_applier",
     ":test_utils",
     "../../../api/audio:audio_frame_api",
-    "../../../rtc_base:gunit_helpers",
     "../../../test:test_support",
   ]
 }
@@ -360,7 +350,6 @@ rtc_library("saturation_protector_unittest") {
     ":common",
     ":saturation_protector",
     "..:apm_logging",
-    "../../../rtc_base:gunit_helpers",
     "../../../test:test_support",
   ]
 }
@@ -371,7 +360,6 @@ rtc_library("biquad_filter_unittests") {
   deps = [
     ":biquad_filter",
     "../../../api:array_view",
-    "../../../rtc_base:gunit_helpers",
     "../../../test:test_support",
   ]
 }
@@ -396,13 +384,9 @@ rtc_library("fixed_digital_unittests") {
     ":fixed_digital",
     ":test_utils",
     "..:apm_logging",
-    "..:audio_frame_view",
-    "../../../api:array_view",
     "../../../api/audio:audio_frame_api",
     "../../../common_audio",
     "../../../rtc_base:checks",
-    "../../../rtc_base:gunit_helpers",
-    "../../../system_wrappers:metrics",
     "../../../test:test_support",
   ]
 }
@@ -420,18 +404,13 @@ rtc_library("input_volume_controller_unittests") {
 
   deps = [
     ":clipping_predictor",
-    ":gain_map",
     ":input_volume_controller",
     "..:audio_buffer",
     "..:audio_frame_view",
-    "../../../api:array_view",
     "../../../api/audio:audio_processing",
     "../../../api/environment:environment_factory",
     "../../../rtc_base:checks",
-    "../../../rtc_base:random",
-    "../../../rtc_base:safe_conversions",
     "../../../rtc_base:safe_minmax",
-    "../../../rtc_base:stringutils",
     "../../../system_wrappers:metrics",
     "../../../test:fileutils",
     "../../../test:test_support",
@@ -451,7 +430,6 @@ rtc_library("noise_estimator_unittests") {
     "../../../api:function_view",
     "../../../api/audio:audio_frame_api",
     "../../../rtc_base:checks",
-    "../../../rtc_base:gunit_helpers",
     "../../../test:test_support",
   ]
 }
@@ -465,7 +443,6 @@ rtc_library("vad_wrapper_unittests") {
     "../../../api:array_view",
     "../../../api/audio:audio_frame_api",
     "../../../rtc_base:checks",
-    "../../../rtc_base:gunit_helpers",
     "../../../rtc_base:safe_compare",
     "../../../test:test_support",
   ]
@@ -500,7 +477,6 @@ rtc_library("input_volume_stats_reporter") {
     "../../../rtc_base:checks",
     "../../../rtc_base:gtest_prod",
     "../../../rtc_base:logging",
-    "../../../rtc_base:safe_minmax",
     "../../../rtc_base:stringutils",
     "../../../system_wrappers:metrics",
     "//third_party/abseil-cpp/absl/strings:string_view",
diff --git a/modules/audio_processing/agc2/rnn_vad/BUILD.gn b/modules/audio_processing/agc2/rnn_vad/BUILD.gn
index 075e2a8110..07aa16ae75 100644
--- a/modules/audio_processing/agc2/rnn_vad/BUILD.gn
+++ b/modules/audio_processing/agc2/rnn_vad/BUILD.gn
@@ -34,8 +34,6 @@ rtc_library("rnn_vad") {
     "..:cpu_features",
     "../../../../api:array_view",
     "../../../../rtc_base:checks",
-    "../../../../rtc_base:safe_compare",
-    "../../../../rtc_base:safe_conversions",
     "//third_party/rnnoise:rnn_vad",
   ]
 }
@@ -319,12 +317,10 @@ if (rtc_include_tests) {
         ":rnn_vad",
         ":rnn_vad_common",
         "..:cpu_features",
-        "../../../../api:array_view",
         "../../../../common_audio",
         "../../../../rtc_base:checks",
         "../../../../rtc_base:logging",
         "../../../../rtc_base:safe_compare",
-        "../../../../test:test_support",
         "//third_party/abseil-cpp/absl/flags:flag",
         "//third_party/abseil-cpp/absl/flags:parse",
       ]
diff --git a/modules/audio_processing/capture_levels_adjuster/BUILD.gn b/modules/audio_processing/capture_levels_adjuster/BUILD.gn
index 20b2e395a8..ebd8bc3a68 100644
--- a/modules/audio_processing/capture_levels_adjuster/BUILD.gn
+++ b/modules/audio_processing/capture_levels_adjuster/BUILD.gn
@@ -39,8 +39,6 @@ rtc_library("capture_levels_adjuster_unittests") {
     ":capture_levels_adjuster",
     "..:audio_buffer",
     "..:audioproc_test_utils",
-    "../../../rtc_base:gunit_helpers",
-    "../../../rtc_base:stringutils",
     "../../../test:test_support",
   ]
 }
diff --git a/modules/audio_processing/capture_mixer/BUILD.gn b/modules/audio_processing/capture_mixer/BUILD.gn
index ec4653d358..2b9a8c4b73 100644
--- a/modules/audio_processing/capture_mixer/BUILD.gn
+++ b/modules/audio_processing/capture_mixer/BUILD.gn
@@ -50,10 +50,7 @@ rtc_library("capture_mixer_unittests") {
   ]
   deps = [
     ":capture_mixer",
-    "..:audioproc_test_utils",
     "../../../api:array_view",
-    "../../../rtc_base:gunit_helpers",
-    "../../../rtc_base:stringutils",
     "../../../test:test_support",
   ]
 }
diff --git a/modules/audio_processing/ns/BUILD.gn b/modules/audio_processing/ns/BUILD.gn
index e90ef0e06a..1b5ebafff1 100644
--- a/modules/audio_processing/ns/BUILD.gn
+++ b/modules/audio_processing/ns/BUILD.gn
@@ -49,19 +49,10 @@ rtc_static_library("ns") {
   }
 
   deps = [
-    "..:apm_logging",
     "..:audio_buffer",
-    "..:high_pass_filter",
     "../../../api:array_view",
-    "../../../common_audio:common_audio_c",
-    "../../../common_audio/third_party/ooura:fft_size_128",
     "../../../common_audio/third_party/ooura:fft_size_256",
     "../../../rtc_base:checks",
-    "../../../rtc_base:safe_minmax",
-    "../../../rtc_base/system:arch",
-    "../../../system_wrappers",
-    "../../../system_wrappers:metrics",
-    "../utility:cascaded_biquad_filter",
   ]
 }
 
@@ -74,18 +65,9 @@ if (rtc_include_tests) {
 
     deps = [
       ":ns",
-      "..:apm_logging",
       "..:audio_buffer",
-      "..:audio_processing",
-      "..:high_pass_filter",
-      "../../../api:array_view",
-      "../../../rtc_base:checks",
-      "../../../rtc_base:safe_minmax",
       "../../../rtc_base:stringutils",
-      "../../../rtc_base/system:arch",
-      "../../../system_wrappers",
       "../../../test:test_support",
-      "../utility:cascaded_biquad_filter",
     ]
 
     defines = []
diff --git a/modules/audio_processing/test/conversational_speech/BUILD.gn b/modules/audio_processing/test/conversational_speech/BUILD.gn
index 28f27daae4..10fd9dda33 100644
--- a/modules/audio_processing/test/conversational_speech/BUILD.gn
+++ b/modules/audio_processing/test/conversational_speech/BUILD.gn
@@ -21,7 +21,6 @@ if (!build_with_chromium) {
       ":lib",
       "../../../../rtc_base:checks",
       "../../../../test:fileutils",
-      "../../../../test:test_support",
       "//third_party/abseil-cpp/absl/flags:flag",
       "//third_party/abseil-cpp/absl/flags:parse",
     ]
diff --git a/modules/audio_processing/utility/BUILD.gn b/modules/audio_processing/utility/BUILD.gn
index 63ee6d066e..eaf22ec0c7 100644
--- a/modules/audio_processing/utility/BUILD.gn
+++ b/modules/audio_processing/utility/BUILD.gn
@@ -50,7 +50,6 @@ if (rtc_include_tests) {
     sources = [ "cascaded_biquad_filter_unittest.cc" ]
     deps = [
       ":cascaded_biquad_filter",
-      "../../../rtc_base:checks",
       "../../../test:test_support",
       "//api:array_view",
       "//rtc_base:checks",
diff --git a/modules/audio_processing/vad/BUILD.gn b/modules/audio_processing/vad/BUILD.gn
index 71e079d3a3..34b3003e6d 100644
--- a/modules/audio_processing/vad/BUILD.gn
+++ b/modules/audio_processing/vad/BUILD.gn
@@ -35,7 +35,6 @@ rtc_library("vad") {
     "voice_gmm_tables.h",
   ]
   deps = [
-    "../../../audio/utility:audio_frame_operations",
     "../../../common_audio",
     "../../../common_audio:common_audio_c",
     "../../../common_audio/third_party/ooura:fft_size_256",
@@ -59,7 +58,6 @@ if (rtc_include_tests) {
     ]
     deps = [
       ":vad",
-      "../../../common_audio",
       "../../../test:fileutils",
       "../../../test:test_support",
       "//testing/gmock",

From 1b9f86f801e9a5ee887adc2ba733474263e90ef7 Mon Sep 17 00:00:00 2001
From: Philipp Hancke <philipp.hancke@googlemail.com>
Date: Wed, 25 Mar 2026 08:35:58 +0100
Subject: [PATCH 411/472] build: remove unnecessary targets in video_coding

using
  tools_webrtc/gn_check_autofix -r modules/video_coding/BUILD.gn
followed by
  tools_webrtc/gn_check_autofix -C out/Default
with some platform-specific manual reverts and dependencies on test fixtures

Bug: webrtc:467294026
Change-Id: I17731b0bea95110fd05ee7aa4da36ce0c0489c08
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/459940
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Christoffer Dewerin <jansson@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#47264}
---
 modules/video_coding/BUILD.gn | 127 +---------------------------------
 1 file changed, 3 insertions(+), 124 deletions(-)

diff --git a/modules/video_coding/BUILD.gn b/modules/video_coding/BUILD.gn
index 34ff20e3f5..0ca6954084 100644
--- a/modules/video_coding/BUILD.gn
+++ b/modules/video_coding/BUILD.gn
@@ -23,12 +23,8 @@ rtc_library("encoded_frame") {
     "../../api/video:video_frame",
     "../../api/video:video_frame_type",
     "../../api/video:video_rtp_headers",
-    "../../modules:module_api_public",
-    "../../modules/rtp_rtcp:rtp_video_header",
-    "../../rtc_base:checks",
-    "../../rtc_base/experiments:alr_experiment",
     "../../rtc_base/system:rtc_export",
-    "../../system_wrappers",
+    "../rtp_rtcp:rtp_video_header",
   ]
 }
 
@@ -39,7 +35,6 @@ rtc_library("chain_diff_calculator") {
   ]
 
   deps = [
-    "../../rtc_base:checks",
     "../../rtc_base:logging",
     "//third_party/abseil-cpp/absl/container:inlined_vector",
   ]
@@ -82,7 +77,6 @@ rtc_library("nack_requester") {
     "../../rtc_base:macromagic",
     "../../rtc_base:mod_ops",
     "../../rtc_base:rtc_numerics",
-    "../../rtc_base/experiments:field_trial_parser",
     "../../rtc_base/system:no_unique_address",
     "../../rtc_base/task_utils:repeating_task",
     "../../system_wrappers",
@@ -96,10 +90,6 @@ rtc_library("packet_buffer") {
   ]
   deps = [
     ":codec_globals_headers",
-    "../../api:array_view",
-    "../../api:rtp_packet_info",
-    "../../api/units:timestamp",
-    "../../api/video:encoded_image",
     "../../api/video:video_frame",
     "../../api/video:video_frame_type",
     "../../common_video",
@@ -124,9 +114,6 @@ rtc_library("h26x_packet_buffer") {
     ":h264_sprop_parameter_sets",
     ":packet_buffer",
     "../../api:array_view",
-    "../../api:rtp_packet_info",
-    "../../api/units:timestamp",
-    "../../api/video:encoded_image",
     "../../api/video:video_frame",
     "../../api/video:video_frame_type",
     "../../common_video",
@@ -134,7 +121,6 @@ rtc_library("h26x_packet_buffer") {
     "../../rtc_base:copy_on_write_buffer",
     "../../rtc_base:logging",
     "../../rtc_base:rtc_numerics",
-    "../rtp_rtcp:rtp_rtcp_format",
     "../rtp_rtcp:rtp_video_header",
     "//third_party/abseil-cpp/absl/algorithm:container",
     "//third_party/abseil-cpp/absl/base:core_headers",
@@ -206,8 +192,6 @@ rtc_library("video_coding") {
   deps = [
     ":codec_globals_headers",
     ":encoded_frame",
-    ":frame_helpers",
-    ":h264_sprop_parameter_sets",
     ":video_codec_interface",
     ":video_coding_utility",
     ":webrtc_vp8_scalability",
@@ -218,36 +202,25 @@ rtc_library("video_coding") {
     "../../api:array_view",
     "../../api:fec_controller_api",
     "../../api:field_trials_view",
-    "../../api:rtp_headers",
     "../../api:rtp_packet_info",
     "../../api:scoped_refptr",
     "../../api:sequence_checker",
     "../../api/environment",
-    "../../api/task_queue",
     "../../api/units:data_rate",
-    "../../api/units:data_size",
-    "../../api/units:frequency",
     "../../api/units:time_delta",
     "../../api/units:timestamp",
-    "../../api/video:builtin_video_bitrate_allocator_factory",
     "../../api/video:encoded_frame",
     "../../api/video:encoded_image",
     "../../api/video:render_resolution",
-    "../../api/video:video_adaptation",
-    "../../api/video:video_bitrate_allocation",
-    "../../api/video:video_bitrate_allocator",
-    "../../api/video:video_bitrate_allocator_factory",
     "../../api/video:video_codec_constants",
     "../../api/video:video_frame",
     "../../api/video:video_frame_type",
     "../../api/video:video_rtp_headers",
     "../../api/video/corruption_detection:frame_instrumentation_data",
-    "../../api/video/corruption_detection:frame_instrumentation_evaluation",
     "../../api/video_codecs:scalability_mode",
     "../../api/video_codecs:video_codecs_api",
     "../../common_video",
     "../../common_video:corruption_score_calculator",
-    "../../rtc_base:base64",
     "../../rtc_base:buffer",
     "../../rtc_base:byte_buffer",
     "../../rtc_base:checks",
@@ -256,30 +229,19 @@ rtc_library("video_coding") {
     "../../rtc_base:logging",
     "../../rtc_base:macromagic",
     "../../rtc_base:mod_ops",
-    "../../rtc_base:rtc_event",
     "../../rtc_base:rtc_numerics",
     "../../rtc_base:safe_conversions",
-    "../../rtc_base:stringutils",
-    "../../rtc_base:threading",
-    "../../rtc_base:timeutils",
-    "../../rtc_base/experiments:alr_experiment",
-    "../../rtc_base/experiments:field_trial_parser",
     "../../rtc_base/experiments:min_video_bitrate_experiment",
     "../../rtc_base/experiments:rate_control_settings",
     "../../rtc_base/synchronization:mutex",
     "../../rtc_base/system:no_unique_address",
-    "../../rtc_base/task_utils:repeating_task",
     "../../system_wrappers",
     "../../system_wrappers:metrics",
     "../../video/config:encoder_config",
     "../rtp_rtcp",
-    "../rtp_rtcp:rtp_rtcp_format",
     "../rtp_rtcp:rtp_video_header",
     "codecs/av1:av1_svc_config",
     "svc:scalability_mode_util",
-    "timing:inter_frame_delay_variation_calculator",
-    "timing:jitter_estimator",
-    "timing:rtt_filter",
     "timing:timing_module",
     "//third_party/abseil-cpp/absl/algorithm:container",
     "//third_party/abseil-cpp/absl/container:inlined_vector",
@@ -307,9 +269,7 @@ rtc_library("video_codec_interface") {
     "../../api/video/corruption_detection:frame_instrumentation_data",
     "../../api/video_codecs:scalability_mode",
     "../../api/video_codecs:video_codecs_api",
-    "../../common_video",
     "../../common_video/generic_frame_descriptor",
-    "../../rtc_base:checks",
     "../../rtc_base/system:rtc_export",
   ]
 }
@@ -342,7 +302,6 @@ rtc_library("frame_sampler") {
     "..:module_api_public",
     "../../api/units:time_delta",
     "../../api/video:video_frame",
-    "../../rtc_base:timeutils",
   ]
 }
 
@@ -379,7 +338,6 @@ rtc_library("video_coding_utility") {
   ]
 
   deps = [
-    ":video_codec_interface",
     "../../api:array_view",
     "../../api:field_trials_view",
     "../../api:scoped_refptr",
@@ -388,39 +346,30 @@ rtc_library("video_coding_utility") {
     "../../api/task_queue",
     "../../api/units:data_rate",
     "../../api/units:time_delta",
-    "../../api/video:encoded_frame",
     "../../api/video:encoded_image",
-    "../../api/video:video_adaptation",
     "../../api/video:video_bitrate_allocation",
     "../../api/video:video_bitrate_allocator",
     "../../api/video:video_codec_constants",
     "../../api/video:video_frame",
     "../../api/video:video_frame_type",
-    "../../api/video/corruption_detection:corruption_detection_settings_generator",
-    "../../api/video/corruption_detection:filter_settings",
     "../../api/video_codecs:video_codecs_api",
     "../../common_video",
-    "../../modules/rtp_rtcp",
     "../../rtc_base:bitstream_reader",
     "../../rtc_base:checks",
     "../../rtc_base:logging",
     "../../rtc_base:macromagic",
     "../../rtc_base:rate_statistics",
-    "../../rtc_base:refcount",
     "../../rtc_base:rtc_numerics",
     "../../rtc_base:stringutils",
-    "../../rtc_base:timeutils",
     "../../rtc_base:weak_ptr",
     "../../rtc_base/experiments:encoder_info_settings",
     "../../rtc_base/experiments:quality_scaler_settings",
     "../../rtc_base/experiments:quality_scaling_experiment",
     "../../rtc_base/experiments:rate_control_settings",
     "../../rtc_base/synchronization:mutex",
-    "../../rtc_base/system:arch",
     "../../rtc_base/system:file_wrapper",
     "../../rtc_base/system:no_unique_address",
     "../../rtc_base/system:rtc_export",
-    "../../rtc_base/task_utils:repeating_task",
     "../../video/config:encoder_config",
     "../rtp_rtcp:rtp_rtcp_format",
     "svc:scalability_mode_util",
@@ -436,7 +385,6 @@ rtc_library("encoder_speed_controller_impl") {
     "utility/encoder_speed_controller_impl.h",
   ]
   deps = [
-    "../../api:array_view",
     "../../api/units:time_delta",
     "../../api/units:timestamp",
     "../../api/video_codecs:video_codecs_api",
@@ -466,7 +414,6 @@ rtc_library("webrtc_h264") {
     ":video_coding_utility",
     "../../api:scoped_refptr",
     "../../api/environment",
-    "../../api/transport/rtp:dependency_descriptor",
     "../../api/units:data_rate",
     "../../api/video:encoded_image",
     "../../api/video:render_resolution",
@@ -481,15 +428,12 @@ rtc_library("webrtc_h264") {
     "../../api/video_codecs:video_codecs_api",
     "../../common_video",
     "../../media:media_constants",
-    "../../media:rtc_media_base",
     "../../rtc_base:checks",
     "../../rtc_base:event_tracer",
     "../../rtc_base:logging",
-    "../../rtc_base:timeutils",
     "../../rtc_base/experiments:psnr_experiment",
     "../../rtc_base/system:rtc_export",
     "../../system_wrappers:metrics",
-    "../rtp_rtcp:rtp_rtcp_format",
     "svc:scalability_structures",
     "svc:scalable_video_controller",
     "//third_party/abseil-cpp/absl/base:nullability",
@@ -556,7 +500,6 @@ rtc_library("webrtc_vp8") {
     ":video_coding_utility",
     ":webrtc_libvpx_interface",
     ":webrtc_vp8_scalability",
-    ":webrtc_vp8_temporal_layers",
     "../../api:fec_controller_api",
     "../../api:field_trials_view",
     "../../api:scoped_refptr",
@@ -588,7 +531,6 @@ rtc_library("webrtc_vp8") {
     "../../rtc_base/experiments:rate_control_settings",
     "../../system_wrappers:metrics",
     "../rtp_rtcp:rtp_rtcp_format",
-    "svc:scalability_mode_util",
     "//third_party/abseil-cpp/absl/algorithm:container",
     "//third_party/abseil-cpp/absl/base:nullability",
     "//third_party/abseil-cpp/absl/container:inlined_vector",
@@ -622,8 +564,6 @@ rtc_library("webrtc_vp8_temporal_layers") {
   deps = [
     ":codec_globals_headers",
     ":video_codec_interface",
-    ":video_coding_utility",
-    "../../api:fec_controller_api",
     "../../api/environment",
     "../../api/transport/rtp:dependency_descriptor",
     "../../api/video:video_codec_constants",
@@ -648,15 +588,9 @@ rtc_library("webrtc_vp9_helpers") {
 
   deps = [
     ":codec_globals_headers",
-    ":video_codec_interface",
-    "../../api/video:video_bitrate_allocation",
-    "../../api/video:video_bitrate_allocator",
-    "../../api/video:video_codec_constants",
     "../../api/video:video_frame",
     "../../api/video_codecs:scalability_mode",
     "../../api/video_codecs:video_codecs_api",
-    "../../common_video",
-    "../../media:video_common",
     "../../rtc_base:checks",
     "../../rtc_base:logging",
     "svc:scalability_mode_util",
@@ -785,7 +719,6 @@ if (rtc_include_tests) {
       ":video_codec_interface",
       "../../api:create_frame_generator",
       "../../api:frame_generator_api",
-      "../../api/transport/rtp:dependency_descriptor",
       "../../api/units:timestamp",
       "../../api/video:encoded_image",
       "../../api/video:render_resolution",
@@ -805,7 +738,6 @@ if (rtc_include_tests) {
 
     deps = [
       ":video_codec_interface",
-      ":video_coding",
       ":video_coding_utility",
       "../../api:mock_video_decoder",
       "../../api:mock_video_encoder",
@@ -819,7 +751,6 @@ if (rtc_include_tests) {
       "../../api/video:video_frame_type",
       "../../api/video:video_rtp_headers",
       "../../api/video_codecs:video_codecs_api",
-      "../../common_video",
       "../../rtc_base:checks",
       "../../test:test_support",
     ]
@@ -836,12 +767,10 @@ if (rtc_include_tests) {
 
     deps = [
       ":codec_globals_headers",
+      ":simulcast_test_fixture_impl",
       ":video_codec_interface",
-      ":video_coding",
       ":video_coding_utility",
       ":videocodec_test_stats_impl",
-      ":webrtc_vp9_helpers",
-      "..:module_api",
       "../../api:create_frame_generator",
       "../../api:field_trials",
       "../../api:frame_generator_api",
@@ -851,24 +780,18 @@ if (rtc_include_tests) {
       "../../api:videocodec_test_stats_api",
       "../../api/environment",
       "../../api/environment:environment_factory",
-      "../../api/numerics",
       "../../api/task_queue",
-      "../../api/task_queue:default_task_queue_factory",
-      "../../api/test/metrics:global_metrics_logger_and_exporter",
       "../../api/units:time_delta",
       "../../api/video:builtin_video_bitrate_allocator_factory",
       "../../api/video:encoded_image",
       "../../api/video:resolution",
-      "../../api/video:video_bitrate_allocation",
       "../../api/video:video_bitrate_allocator",
       "../../api/video:video_bitrate_allocator_factory",
-      "../../api/video:video_codec_constants",
       "../../api/video:video_frame",
       "../../api/video:video_frame_type",
       "../../api/video:video_rtp_headers",
       "../../api/video_codecs:video_codecs_api",
       "../../common_video",
-      "../../rtc_base:buffer",
       "../../rtc_base:checks",
       "../../rtc_base:macromagic",
       "../../rtc_base:rtc_event",
@@ -927,7 +850,6 @@ if (rtc_include_tests) {
     ]
     deps = [
       ":codec_globals_headers",
-      ":video_codec_interface",
       ":video_codecs_test_framework",
       ":video_coding_utility",
       ":videocodec_test_stats_impl",
@@ -942,13 +864,10 @@ if (rtc_include_tests) {
       "../../api/environment:environment_factory",
       "../../api/test/metrics:global_metrics_logger_and_exporter",
       "../../api/test/metrics:metric",
-      "../../api/test/video:function_video_factory",
       "../../api/video:encoded_image",
       "../../api/video:resolution",
-      "../../api/video:video_bitrate_allocation",
       "../../api/video:video_codec_constants",
       "../../api/video:video_frame",
-      "../../api/video:video_frame_type",
       "../../api/video_codecs:video_codecs_api",
       "../../api/video_codecs:video_decoder_factory_template",
       "../../api/video_codecs:video_decoder_factory_template_dav1d_adapter",
@@ -962,7 +881,6 @@ if (rtc_include_tests) {
       "../../api/video_codecs:video_encoder_factory_template_open_h264_adapter",
       "../../common_video",
       "../../media:media_constants",
-      "../../media:rtc_audio_video",
       "../../rtc_base:checks",
       "../../rtc_base:cpu_info",
       "../../rtc_base:logging",
@@ -992,16 +910,11 @@ if (rtc_include_tests) {
     ]
     deps = [
       "../../api:videocodec_test_stats_api",
-      "../../api/numerics",
-      "../../api/test/metrics:global_metrics_logger_and_exporter",
-      "../../api/test/metrics:metric",
       "../../api/units:data_rate",
       "../../api/units:frequency",
       "../../api/video:video_frame_type",
       "../../rtc_base:checks",
       "../../rtc_base:rtc_numerics",
-      "../../rtc_base:stringutils",
-      "../../test:test_common",
       "../rtp_rtcp:rtp_rtcp_format",
     ]
   }
@@ -1012,8 +925,6 @@ if (rtc_include_tests) {
     sources = [ "codecs/test/video_codec_test.cc" ]
 
     deps = [
-      ":video_codec_interface",
-      "../../api:field_trials",
       "../../api/environment",
       "../../api/environment:environment_factory",
       "../../api/test/metrics:global_metrics_logger_and_exporter",
@@ -1024,7 +935,6 @@ if (rtc_include_tests) {
       "../../api/video_codecs:builtin_video_encoder_factory",
       "../../api/video_codecs:scalability_mode",
       "../../api/video_codecs:video_codecs_api",
-      "../../modules/video_coding/svc:scalability_mode_util",
       "../../rtc_base:checks",
       "../../rtc_base:logging",
       "../../rtc_base:stringutils",
@@ -1033,6 +943,7 @@ if (rtc_include_tests) {
       "../../test:test_flags",
       "../../test:test_support",
       "../../test:video_codec_tester",
+      "svc:scalability_mode_util",
       "//third_party/abseil-cpp/absl/flags:flag",
       "//third_party/abseil-cpp/absl/functional:any_invocable",
     ]
@@ -1088,7 +999,6 @@ if (rtc_include_tests) {
       "../../api:field_trials",
       "../../api:frame_generator_api",
       "../../api:make_ref_counted",
-      "../../api:mock_video_codec_factory",
       "../../api:mock_video_decoder",
       "../../api:mock_video_encoder",
       "../../api:scoped_refptr",
@@ -1096,7 +1006,6 @@ if (rtc_include_tests) {
       "../../api:videocodec_test_stats_api",
       "../../api/environment",
       "../../api/environment:environment_factory",
-      "../../api/test/metrics:global_metrics_logger_and_exporter",
       "../../api/test/video:function_video_factory",
       "../../api/units:data_rate",
       "../../api/units:time_delta",
@@ -1108,25 +1017,19 @@ if (rtc_include_tests) {
       "../../api/video:video_frame",
       "../../api/video:video_frame_type",
       "../../api/video:video_rtp_headers",
-      "../../api/video_codecs:rtc_software_fallback_wrappers",
       "../../api/video_codecs:scalability_mode",
       "../../api/video_codecs:video_codecs_api",
       "../../common_video",
-      "../../common_video/test:utilities",
-      "../../media:codec",
       "../../media:media_constants",
       "../../media:rtc_internal_video_codecs",
       "../../media:rtc_simulcast_encoder_adapter",
       "../../rtc_base:checks",
-      "../../rtc_base:refcount",
-      "../../rtc_base:stringutils",
       "../../rtc_base:timeutils",
       "../../test:create_test_field_trials",
       "../../test:fileutils",
       "../../test:test_support",
       "../../test:video_test_common",
       "../rtp_rtcp:rtp_rtcp_format",
-      "codecs/av1:dav1d_decoder",
       "svc:scalability_mode_util",
       "//third_party/abseil-cpp/absl/container:inlined_vector",
       "//third_party/abseil-cpp/absl/memory",
@@ -1210,16 +1113,12 @@ if (rtc_include_tests) {
     deps = [
       ":chain_diff_calculator",
       ":codec_globals_headers",
-      ":encoded_frame",
-      ":encoder_speed_controller_impl",
       ":frame_dependencies_calculator",
       ":frame_helpers",
-      ":frame_sampler",
       ":h264_sprop_parameter_sets",
       ":h26x_packet_buffer",
       ":nack_requester",
       ":packet_buffer",
-      ":simulcast_test_fixture_impl",
       ":video_codec_interface",
       ":video_codecs_test_framework",
       ":video_coding",
@@ -1229,7 +1128,6 @@ if (rtc_include_tests) {
       ":webrtc_h264",
       ":webrtc_vp8",
       ":webrtc_vp8_temporal_layers",
-      ":webrtc_vp9",
       ":webrtc_vp9_helpers",
       "..:module_api",
       "..:module_fec_api",
@@ -1244,7 +1142,6 @@ if (rtc_include_tests) {
       "../../api:mock_fec_controller_override",
       "../../api:mock_video_decoder",
       "../../api:mock_video_encoder",
-      "../../api:rtp_headers",
       "../../api:rtp_packet_info",
       "../../api:scoped_refptr",
       "../../api:simulcast_test_fixture_api",
@@ -1252,20 +1149,15 @@ if (rtc_include_tests) {
       "../../api/environment",
       "../../api/environment:environment_factory",
       "../../api/task_queue",
-      "../../api/task_queue:default_task_queue_factory",
       "../../api/test/video:function_video_factory",
       "../../api/transport/rtp:dependency_descriptor",
       "../../api/units:data_rate",
-      "../../api/units:data_size",
-      "../../api/units:frequency",
       "../../api/units:time_delta",
       "../../api/units:timestamp",
       "../../api/video:builtin_video_bitrate_allocator_factory",
       "../../api/video:encoded_frame",
       "../../api/video:encoded_image",
-      "../../api/video:frame_buffer",
       "../../api/video:render_resolution",
-      "../../api/video:video_adaptation",
       "../../api/video:video_bitrate_allocation",
       "../../api/video:video_bitrate_allocator",
       "../../api/video:video_bitrate_allocator_factory",
@@ -1274,7 +1166,6 @@ if (rtc_include_tests) {
       "../../api/video:video_frame_type",
       "../../api/video:video_rtp_headers",
       "../../api/video/corruption_detection:corruption_detection_unittests",
-      "../../api/video/corruption_detection:filter_settings",
       "../../api/video/corruption_detection:frame_instrumentation_data",
       "../../api/video_codecs:scalability_mode",
       "../../api/video_codecs:video_codecs_api",
@@ -1284,30 +1175,20 @@ if (rtc_include_tests) {
       "../../common_video/generic_frame_descriptor",
       "../../common_video/test:utilities",
       "../../media:media_constants",
-      "../../media:rtc_internal_video_codecs",
       "../../rtc_base:checks",
       "../../rtc_base:copy_on_write_buffer",
-      "../../rtc_base:gunit_helpers",
-      "../../rtc_base:histogram_percentile_counter",
-      "../../rtc_base:platform_thread",
       "../../rtc_base:random",
-      "../../rtc_base:refcount",
       "../../rtc_base:rtc_base_tests_utils",
       "../../rtc_base:rtc_event",
-      "../../rtc_base:rtc_numerics",
-      "../../rtc_base:stringutils",
       "../../rtc_base:task_queue_for_test",
       "../../rtc_base:threading",
-      "../../rtc_base:timeutils",
       "../../rtc_base/experiments:encoder_info_settings",
-      "../../rtc_base/synchronization:mutex",
       "../../rtc_base/system:file_wrapper",
       "../../rtc_base/system:unused",
       "../../system_wrappers",
       "../../system_wrappers:metrics",
       "../../test:create_test_environment",
       "../../test:create_test_field_trials",
-      "../../test:fake_encoded_frame",
       "../../test:fake_video_codecs",
       "../../test:fileutils",
       "../../test:run_loop",
@@ -1315,7 +1196,6 @@ if (rtc_include_tests) {
       "../../test:video_test_common",
       "../../test:video_test_support",
       "../../test/time_controller",
-      "../../third_party/libyuv",
       "../../video/config:encoder_config",
       "../rtp_rtcp",
       "../rtp_rtcp:rtp_rtcp_format",
@@ -1324,7 +1204,6 @@ if (rtc_include_tests) {
       "svc:scalability_structure_tests",
       "svc:simulcast_to_svc_converter_tests",
       "svc:svc_rate_allocator_tests",
-      "timing:jitter_estimator",
       "timing:timing_module",
       "//third_party/abseil-cpp/absl/container:inlined_vector",
     ]

From 18f6a998909d7eabe24c3f7f85938e171db1ef41 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?=C3=85sa=20Persson?= <asapersson@webrtc.org>
Date: Thu, 26 Mar 2026 09:02:08 +0000
Subject: [PATCH 412/472] Change how TimingFrameInfo is passed to
 ReceiveStatisticsProxy.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Instead of being stored in VCMTiming and pulled by VideoStreamBufferController, it is now passed directly after a frame is decoded through OnFrameToRender and OnDecodedFrame callbacks.

Bug: b/493549134
Change-Id: I3e2ce002805b3d3f30411848bfb0144b10338605
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/459780
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47265}
---
 modules/video_coding/generic_decoder.cc       |   4 +-
 .../include/video_coding_defines.h            |   2 +
 modules/video_coding/timing/timing.cc         |  10 -
 modules/video_coding/timing/timing.h          |   4 -
 video/receive_statistics_proxy.cc             |  18 +-
 video/receive_statistics_proxy.h              |   5 +-
 video/receive_statistics_proxy_unittest.cc    | 193 +++++++++++-------
 video/timing/simulator/rendering_tracker.h    |   2 -
 video/video_stream_buffer_controller.cc       |   8 -
 video/video_stream_buffer_controller.h        |   4 -
 ...video_stream_buffer_controller_unittest.cc |   4 -
 video/video_stream_decoder2.cc                |   3 +-
 12 files changed, 136 insertions(+), 121 deletions(-)

diff --git a/modules/video_coding/generic_decoder.cc b/modules/video_coding/generic_decoder.cc
index 2320dfdffc..dac95bfaa7 100644
--- a/modules/video_coding/generic_decoder.cc
+++ b/modules/video_coding/generic_decoder.cc
@@ -236,7 +236,6 @@ void VCMDecodedFrameCallback::Decoded(VideoFrame& decodedImage,
   RTC_HISTOGRAM_COUNTS_1000(
       "WebRTC.Video.GenericDecoder.DecodeDelay",
       timing_frame_info.decode_finish_ms - timing_frame_info.decode_start_ms);
-  timing_->SetTimingFrameInfo(timing_frame_info);
   decodedImage.set_timestamp_us(
       frame_info->render_time ? frame_info->render_time->us() : -1);
   decodedImage.set_content_type(frame_info->content_type);
@@ -244,7 +243,8 @@ void VCMDecodedFrameCallback::Decoded(VideoFrame& decodedImage,
                                       .qp = qp,
                                       .decode_time = decode_time,
                                       .content_type = frame_info->content_type,
-                                      .frame_type = frame_info->frame_type});
+                                      .frame_type = frame_info->frame_type,
+                                      .timing_frame_info = timing_frame_info});
 
   if (corruption_score_calculator_ &&
       frame_info->frame_instrumentation_data.has_value()) {
diff --git a/modules/video_coding/include/video_coding_defines.h b/modules/video_coding/include/video_coding_defines.h
index 88c16ca733..f2e852f181 100644
--- a/modules/video_coding/include/video_coding_defines.h
+++ b/modules/video_coding/include/video_coding_defines.h
@@ -20,6 +20,7 @@
 #include "api/video/video_content_type.h"
 #include "api/video/video_frame.h"
 #include "api/video/video_frame_type.h"
+#include "api/video/video_timing.h"
 #include "api/video_codecs/video_decoder.h"
 
 namespace webrtc {
@@ -58,6 +59,7 @@ class VCMReceiveCallback {
     TimeDelta decode_time;
     VideoContentType content_type;
     VideoFrameType frame_type;
+    TimingFrameInfo timing_frame_info;
   };
 
   virtual int32_t OnFrameToRender(const FrameToRender& arguments) = 0;
diff --git a/modules/video_coding/timing/timing.cc b/modules/video_coding/timing/timing.cc
index 63a7befaf0..79f71f39a8 100644
--- a/modules/video_coding/timing/timing.cc
+++ b/modules/video_coding/timing/timing.cc
@@ -259,16 +259,6 @@ VCMTiming::VideoDelayTimings VCMTiming::GetTimings() const {
       .current_delay = current_delay_};
 }
 
-void VCMTiming::SetTimingFrameInfo(const TimingFrameInfo& info) {
-  MutexLock lock(&mutex_);
-  timing_frame_info_.emplace(info);
-}
-
-std::optional<TimingFrameInfo> VCMTiming::GetTimingFrameInfo() {
-  MutexLock lock(&mutex_);
-  return timing_frame_info_;
-}
-
 void VCMTiming::SetMaxCompositionDelayInFrames(
     std::optional<int> max_composition_delay_in_frames) {
   MutexLock lock(&mutex_);
diff --git a/modules/video_coding/timing/timing.h b/modules/video_coding/timing/timing.h
index d0b863864e..9ee0d8add0 100644
--- a/modules/video_coding/timing/timing.h
+++ b/modules/video_coding/timing/timing.h
@@ -114,9 +114,6 @@ class VCMTiming {
   // Returns current timing information.
   VideoDelayTimings GetTimings() const;
 
-  void SetTimingFrameInfo(const TimingFrameInfo& info);
-  std::optional<TimingFrameInfo> GetTimingFrameInfo();
-
   void SetMaxCompositionDelayInFrames(
       std::optional<int> max_composition_delay_in_frames);
 
@@ -150,7 +147,6 @@ class VCMTiming {
   TimeDelta max_playout_delay_ RTC_GUARDED_BY(mutex_);
   TimeDelta jitter_delay_ RTC_GUARDED_BY(mutex_);
   TimeDelta current_delay_ RTC_GUARDED_BY(mutex_);
-  std::optional<TimingFrameInfo> timing_frame_info_ RTC_GUARDED_BY(mutex_);
   size_t num_decoded_frames_ RTC_GUARDED_BY(mutex_);
   std::optional<int> max_composition_delay_in_frames_ RTC_GUARDED_BY(mutex_);
   // Set by the field trial WebRTC-ZeroPlayoutDelay. The parameter min_pacing
diff --git a/video/receive_statistics_proxy.cc b/video/receive_statistics_proxy.cc
index 2d1a283a56..ca4b7dfb49 100644
--- a/video/receive_statistics_proxy.cc
+++ b/video/receive_statistics_proxy.cc
@@ -609,11 +609,13 @@ void ReceiveStatisticsProxy::OnCname(uint32_t ssrc, absl::string_view cname) {
   stats_.c_name = std::string(cname);
 }
 
-void ReceiveStatisticsProxy::OnDecodedFrame(const VideoFrame& frame,
-                                            std::optional<uint8_t> qp,
-                                            TimeDelta decode_time,
-                                            VideoContentType content_type,
-                                            VideoFrameType frame_type) {
+void ReceiveStatisticsProxy::OnDecodedFrame(
+    const VideoFrame& frame,
+    std::optional<uint8_t> qp,
+    TimeDelta decode_time,
+    VideoContentType content_type,
+    VideoFrameType frame_type,
+    const TimingFrameInfo& timing_frame_info) {
   TimeDelta processing_delay = TimeDelta::Zero();
   Timestamp current_time = clock_->CurrentTime();
   // TODO(bugs.webrtc.org/13984): some tests do not fill packet_infos().
@@ -637,10 +639,12 @@ void ReceiveStatisticsProxy::OnDecodedFrame(const VideoFrame& frame,
   // "com.apple.coremedia.decompressionsession.clientcallback"
   VideoFrameMetaData meta(frame, current_time);
   worker_thread_->PostTask(SafeTask(
-      task_safety_.flag(), [meta, qp, decode_time, processing_delay,
-                            assembly_time, content_type, frame_type, this]() {
+      task_safety_.flag(),
+      [meta, qp, decode_time, processing_delay, assembly_time, content_type,
+       frame_type, timing_frame_info, this]() {
         OnDecodedFrame(meta, qp, decode_time, processing_delay, assembly_time,
                        content_type, frame_type);
+        OnTimingFrameInfoUpdated(timing_frame_info);
       }));
 }
 
diff --git a/video/receive_statistics_proxy.h b/video/receive_statistics_proxy.h
index b72ed92160..16f42bd02d 100644
--- a/video/receive_statistics_proxy.h
+++ b/video/receive_statistics_proxy.h
@@ -70,7 +70,8 @@ class ReceiveStatisticsProxy : public VideoStreamBufferControllerStatsObserver,
                       std::optional<uint8_t> qp,
                       TimeDelta decode_time,
                       VideoContentType content_type,
-                      VideoFrameType frame_type);
+                      VideoFrameType frame_type,
+                      const TimingFrameInfo& timing_frame_info);
 
   // Called asyncronously on the worker thread as a result of a call to the
   // above OnDecodedFrame method, which is called back on the thread where
@@ -82,6 +83,7 @@ class ReceiveStatisticsProxy : public VideoStreamBufferControllerStatsObserver,
                       TimeDelta assembly_time,
                       VideoContentType content_type,
                       VideoFrameType frame_type);
+  void OnTimingFrameInfoUpdated(const TimingFrameInfo& info);
 
   void OnSyncOffsetUpdated(int64_t video_playout_ntp_ms,
                            int64_t sync_offset_ms,
@@ -111,7 +113,6 @@ class ReceiveStatisticsProxy : public VideoStreamBufferControllerStatsObserver,
                                    int jitter_delay_ms,
                                    int min_playout_delay_ms,
                                    int render_delay_ms) override;
-  void OnTimingFrameInfoUpdated(const TimingFrameInfo& info) override;
 
   // Implements RtcpCnameCallback.
   void OnCname(uint32_t ssrc, absl::string_view cname) override;
diff --git a/video/receive_statistics_proxy_unittest.cc b/video/receive_statistics_proxy_unittest.cc
index 40ba0338b4..b38b310c99 100644
--- a/video/receive_statistics_proxy_unittest.cc
+++ b/video/receive_statistics_proxy_unittest.cc
@@ -48,6 +48,8 @@ namespace webrtc {
 namespace internal {
 namespace {
 using ::testing::DoubleEq;
+using ::testing::Field;
+using ::testing::Optional;
 
 constexpr TimeDelta kFreqOffsetProcessInterval = TimeDelta::Seconds(40);
 constexpr uint32_t kRemoteSsrc = 456;
@@ -69,7 +71,7 @@ class ReceiveStatisticsProxyTest : public ::testing::Test {
   ~ReceiveStatisticsProxyTest() override { statistics_proxy_.reset(); }
 
  protected:
-  // Convenience method to avoid too many explict flushes.
+  // Convenience method to avoid too many explicit flushes.
   VideoReceiveStreamInterface::Stats FlushAndGetStats() {
     time_controller_.AdvanceTime(TimeDelta::Zero());
     return statistics_proxy_->GetStats();
@@ -120,6 +122,13 @@ class ReceiveStatisticsProxyTest : public ::testing::Test {
     return VideoFrameMetaData(frame, Now());
   }
 
+  // Creates a TimingFrameInfo with invalid flag.
+  TimingFrameInfo Info() {
+    TimingFrameInfo info;
+    info.flags = VideoSendTiming::kInvalid;
+    return info;
+  }
+
   GlobalSimulatedTimeController time_controller_;
   std::unique_ptr<ReceiveStatisticsProxy> statistics_proxy_;
 };
@@ -131,7 +140,7 @@ TEST_F(ReceiveStatisticsProxyTest, OnDecodedFrameIncreasesFramesDecoded) {
   for (uint32_t i = 1; i <= 3; ++i) {
     statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
                                       VideoContentType::UNSPECIFIED,
-                                      VideoFrameType::kVideoFrameKey);
+                                      VideoFrameType::kVideoFrameKey, Info());
     EXPECT_EQ(i, FlushAndGetStats().frames_decoded);
     EXPECT_EQ(0, FlushAndGetStats().frame_counts.key_frames);
   }
@@ -144,7 +153,7 @@ TEST_F(ReceiveStatisticsProxyTest, DecodedFpsIsReported) {
   for (int i = 0; i < kRequiredSamples; ++i) {
     statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
                                       VideoContentType::UNSPECIFIED,
-                                      VideoFrameType::kVideoFrameKey);
+                                      VideoFrameType::kVideoFrameKey, Info());
     time_controller_.AdvanceTime(1 / kFps);
   }
   FlushAndUpdateHistograms(std::nullopt, StreamDataCounters(), nullptr);
@@ -161,7 +170,7 @@ TEST_F(ReceiveStatisticsProxyTest, DecodedFpsIsNotReportedForTooFewSamples) {
   for (int i = 0; i < kRequiredSamples - 1; ++i) {
     statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
                                       VideoContentType::UNSPECIFIED,
-                                      VideoFrameType::kVideoFrameKey);
+                                      VideoFrameType::kVideoFrameKey, Info());
     time_controller_.AdvanceTime(1 / kFps);
   }
   FlushAndUpdateHistograms(std::nullopt, StreamDataCounters(), nullptr);
@@ -178,7 +187,7 @@ TEST_F(ReceiveStatisticsProxyTest,
   for (uint32_t i = 1; i <= 3; ++i) {
     statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Millis(1),
                                       VideoContentType::UNSPECIFIED,
-                                      VideoFrameType::kVideoFrameKey);
+                                      VideoFrameType::kVideoFrameKey, Info());
     expected_total_decode_time += TimeDelta::Millis(1);
     ++expected_frames_decoded;
     time_controller_.AdvanceTime(TimeDelta::Zero());
@@ -189,7 +198,7 @@ TEST_F(ReceiveStatisticsProxyTest,
   }
   statistics_proxy_->OnDecodedFrame(frame, 1u, TimeDelta::Millis(3),
                                     VideoContentType::UNSPECIFIED,
-                                    VideoFrameType::kVideoFrameKey);
+                                    VideoFrameType::kVideoFrameKey, Info());
   ++expected_frames_decoded;
   expected_total_decode_time += TimeDelta::Millis(3);
   time_controller_.AdvanceTime(TimeDelta::Zero());
@@ -215,7 +224,7 @@ TEST_F(ReceiveStatisticsProxyTest, OnDecodedFrameIncreasesProcessingDelay) {
     time_controller_.AdvanceTime(kProcessingDelay);
     statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Millis(1),
                                       VideoContentType::UNSPECIFIED,
-                                      VideoFrameType::kVideoFrameKey);
+                                      VideoFrameType::kVideoFrameKey, Info());
     expected_total_processing_delay += i * kProcessingDelay;
     ++expected_frames_decoded;
     time_controller_.AdvanceTime(TimeDelta::Zero());
@@ -227,7 +236,7 @@ TEST_F(ReceiveStatisticsProxyTest, OnDecodedFrameIncreasesProcessingDelay) {
   time_controller_.AdvanceTime(kProcessingDelay);
   statistics_proxy_->OnDecodedFrame(frame, 1u, TimeDelta::Millis(3),
                                     VideoContentType::UNSPECIFIED,
-                                    VideoFrameType::kVideoFrameKey);
+                                    VideoFrameType::kVideoFrameKey, Info());
   ++expected_frames_decoded;
   expected_total_processing_delay += 4 * kProcessingDelay;
   time_controller_.AdvanceTime(TimeDelta::Zero());
@@ -252,7 +261,7 @@ TEST_F(ReceiveStatisticsProxyTest, OnDecodedFrameIncreasesAssemblyTime) {
   frame.set_packet_infos(RtpPacketInfos(single_packet_frame));
   statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Millis(1),
                                     VideoContentType::UNSPECIFIED,
-                                    VideoFrameType::kVideoFrameKey);
+                                    VideoFrameType::kVideoFrameKey, Info());
   ++expected_frames_decoded;
   time_controller_.AdvanceTime(TimeDelta::Zero());
   EXPECT_EQ(expected_total_assembly_time,
@@ -273,7 +282,7 @@ TEST_F(ReceiveStatisticsProxyTest, OnDecodedFrameIncreasesAssemblyTime) {
   frame.set_packet_infos(RtpPacketInfos(ordered_frame));
   statistics_proxy_->OnDecodedFrame(frame, 1u, TimeDelta::Millis(3),
                                     VideoContentType::UNSPECIFIED,
-                                    VideoFrameType::kVideoFrameKey);
+                                    VideoFrameType::kVideoFrameKey, Info());
   ++expected_frames_decoded;
   ++expected_frames_assembled_from_multiple_packets;
   expected_total_assembly_time += 2 * kAssemblyTime;
@@ -298,7 +307,7 @@ TEST_F(ReceiveStatisticsProxyTest, OnDecodedFrameIncreasesAssemblyTime) {
   frame.set_packet_infos(RtpPacketInfos(unordered_frame));
   statistics_proxy_->OnDecodedFrame(frame, 1u, TimeDelta::Millis(3),
                                     VideoContentType::UNSPECIFIED,
-                                    VideoFrameType::kVideoFrameKey);
+                                    VideoFrameType::kVideoFrameKey, Info());
   ++expected_frames_decoded;
   ++expected_frames_assembled_from_multiple_packets;
   expected_total_assembly_time += 2 * kAssemblyTime;
@@ -317,11 +326,11 @@ TEST_F(ReceiveStatisticsProxyTest, OnDecodedFrameIncreasesQpSum) {
   VideoFrame frame = CreateFrame(kWidth, kHeight);
   statistics_proxy_->OnDecodedFrame(frame, 3u, TimeDelta::Zero(),
                                     VideoContentType::UNSPECIFIED,
-                                    VideoFrameType::kVideoFrameKey);
+                                    VideoFrameType::kVideoFrameKey, Info());
   EXPECT_EQ(3u, FlushAndGetStats().qp_sum);
   statistics_proxy_->OnDecodedFrame(frame, 127u, TimeDelta::Zero(),
                                     VideoContentType::UNSPECIFIED,
-                                    VideoFrameType::kVideoFrameDelta);
+                                    VideoFrameType::kVideoFrameDelta, Info());
   EXPECT_EQ(130u, FlushAndGetStats().qp_sum);
 }
 
@@ -330,11 +339,11 @@ TEST_F(ReceiveStatisticsProxyTest, OnDecodedFrameIncreasesTotalDecodeTime) {
   VideoFrame frame = CreateFrame(kWidth, kHeight);
   statistics_proxy_->OnDecodedFrame(frame, 3u, TimeDelta::Millis(4),
                                     VideoContentType::UNSPECIFIED,
-                                    VideoFrameType::kVideoFrameKey);
+                                    VideoFrameType::kVideoFrameKey, Info());
   EXPECT_EQ(4u, FlushAndGetStats().total_decode_time.ms());
   statistics_proxy_->OnDecodedFrame(frame, 127u, TimeDelta::Millis(7),
                                     VideoContentType::UNSPECIFIED,
-                                    VideoFrameType::kVideoFrameDelta);
+                                    VideoFrameType::kVideoFrameDelta, Info());
   EXPECT_EQ(11u, FlushAndGetStats().total_decode_time.ms());
 }
 
@@ -346,12 +355,12 @@ TEST_F(ReceiveStatisticsProxyTest, ReportsContentType) {
                                  statistics_proxy_->GetStats().content_type));
   statistics_proxy_->OnDecodedFrame(frame, 3u, TimeDelta::Zero(),
                                     VideoContentType::SCREENSHARE,
-                                    VideoFrameType::kVideoFrameKey);
+                                    VideoFrameType::kVideoFrameKey, Info());
   EXPECT_EQ(kScreenshareString,
             videocontenttypehelpers::ToString(FlushAndGetStats().content_type));
   statistics_proxy_->OnDecodedFrame(frame, 3u, TimeDelta::Zero(),
                                     VideoContentType::UNSPECIFIED,
-                                    VideoFrameType::kVideoFrameDelta);
+                                    VideoFrameType::kVideoFrameDelta, Info());
   EXPECT_EQ(kRealtimeString,
             videocontenttypehelpers::ToString(FlushAndGetStats().content_type));
 }
@@ -364,25 +373,25 @@ TEST_F(ReceiveStatisticsProxyTest, ReportsMaxInterframeDelay) {
   EXPECT_EQ(-1, statistics_proxy_->GetStats().interframe_delay_max_ms);
   statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
                                     VideoContentType::UNSPECIFIED,
-                                    VideoFrameType::kVideoFrameKey);
+                                    VideoFrameType::kVideoFrameKey, Info());
   EXPECT_EQ(-1, FlushAndGetStats().interframe_delay_max_ms);
 
   time_controller_.AdvanceTime(kInterframeDelay1);
   statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
                                     VideoContentType::UNSPECIFIED,
-                                    VideoFrameType::kVideoFrameDelta);
+                                    VideoFrameType::kVideoFrameDelta, Info());
   EXPECT_EQ(kInterframeDelay1.ms(), FlushAndGetStats().interframe_delay_max_ms);
 
   time_controller_.AdvanceTime(kInterframeDelay2);
   statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
                                     VideoContentType::UNSPECIFIED,
-                                    VideoFrameType::kVideoFrameDelta);
+                                    VideoFrameType::kVideoFrameDelta, Info());
   EXPECT_EQ(kInterframeDelay2.ms(), FlushAndGetStats().interframe_delay_max_ms);
 
   time_controller_.AdvanceTime(kInterframeDelay3);
   statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
                                     VideoContentType::UNSPECIFIED,
-                                    VideoFrameType::kVideoFrameDelta);
+                                    VideoFrameType::kVideoFrameDelta, Info());
   // kInterframeDelay3 is smaller than kInterframeDelay2.
   EXPECT_EQ(kInterframeDelay2.ms(), FlushAndGetStats().interframe_delay_max_ms);
 }
@@ -395,26 +404,26 @@ TEST_F(ReceiveStatisticsProxyTest, ReportInterframeDelayInWindow) {
   EXPECT_EQ(-1, statistics_proxy_->GetStats().interframe_delay_max_ms);
   statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
                                     VideoContentType::UNSPECIFIED,
-                                    VideoFrameType::kVideoFrameKey);
+                                    VideoFrameType::kVideoFrameKey, Info());
   EXPECT_EQ(-1, FlushAndGetStats().interframe_delay_max_ms);
 
   time_controller_.AdvanceTime(kInterframeDelay1);
   statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
                                     VideoContentType::UNSPECIFIED,
-                                    VideoFrameType::kVideoFrameDelta);
+                                    VideoFrameType::kVideoFrameDelta, Info());
   EXPECT_EQ(kInterframeDelay1.ms(), FlushAndGetStats().interframe_delay_max_ms);
 
   time_controller_.AdvanceTime(kInterframeDelay2);
   statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
                                     VideoContentType::UNSPECIFIED,
-                                    VideoFrameType::kVideoFrameDelta);
+                                    VideoFrameType::kVideoFrameDelta, Info());
   // Still first delay is the maximum
   EXPECT_EQ(kInterframeDelay1.ms(), FlushAndGetStats().interframe_delay_max_ms);
 
   time_controller_.AdvanceTime(kInterframeDelay3);
   statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
                                     VideoContentType::UNSPECIFIED,
-                                    VideoFrameType::kVideoFrameDelta);
+                                    VideoFrameType::kVideoFrameDelta, Info());
   // Now the first sample is out of the window, so the second is the maximum.
   EXPECT_EQ(kInterframeDelay2.ms(), FlushAndGetStats().interframe_delay_max_ms);
 }
@@ -524,7 +533,7 @@ TEST_F(ReceiveStatisticsProxyTest, OnDecodedFrameWithoutQpQpSumWontExist) {
   EXPECT_EQ(std::nullopt, statistics_proxy_->GetStats().qp_sum);
   statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
                                     VideoContentType::UNSPECIFIED,
-                                    VideoFrameType::kVideoFrameKey);
+                                    VideoFrameType::kVideoFrameKey, Info());
   EXPECT_EQ(std::nullopt, FlushAndGetStats().qp_sum);
 }
 
@@ -533,11 +542,11 @@ TEST_F(ReceiveStatisticsProxyTest, OnDecodedFrameWithoutQpResetsQpSum) {
   EXPECT_EQ(std::nullopt, statistics_proxy_->GetStats().qp_sum);
   statistics_proxy_->OnDecodedFrame(frame, 3u, TimeDelta::Zero(),
                                     VideoContentType::UNSPECIFIED,
-                                    VideoFrameType::kVideoFrameKey);
+                                    VideoFrameType::kVideoFrameKey, Info());
   EXPECT_EQ(3u, FlushAndGetStats().qp_sum);
   statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
                                     VideoContentType::UNSPECIFIED,
-                                    VideoFrameType::kVideoFrameDelta);
+                                    VideoFrameType::kVideoFrameDelta, Info());
   EXPECT_EQ(std::nullopt, FlushAndGetStats().qp_sum);
 }
 
@@ -618,8 +627,7 @@ TEST_F(ReceiveStatisticsProxyTest, GetStatsReportsOnDroppedFrame) {
     statistics_proxy_->OnDroppedFrames(i);
     dropped_frames += i;
   }
-  VideoReceiveStreamInterface::Stats stats = FlushAndGetStats();
-  EXPECT_EQ(dropped_frames, stats.frames_dropped);
+  EXPECT_EQ(dropped_frames, FlushAndGetStats().frames_dropped);
 }
 
 TEST_F(ReceiveStatisticsProxyTest, GetStatsReportsDecodeTimingStats) {
@@ -697,12 +705,12 @@ TEST_F(ReceiveStatisticsProxyTest, GetStatsReportsFrameCounts) {
   for (int i = 0; i < kKeyFrames; i++) {
     statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
                                       VideoContentType::UNSPECIFIED,
-                                      VideoFrameType::kVideoFrameKey);
+                                      VideoFrameType::kVideoFrameKey, Info());
   }
   for (int i = 0; i < kDeltaFrames; i++) {
     statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
                                       VideoContentType::UNSPECIFIED,
-                                      VideoFrameType::kVideoFrameDelta);
+                                      VideoFrameType::kVideoFrameDelta, Info());
   }
 
   VideoReceiveStreamInterface::Stats stats = statistics_proxy_->GetStats();
@@ -722,13 +730,43 @@ TEST_F(ReceiveStatisticsProxyTest, GetStatsReportsNoCNameForUnknownSsrc) {
   EXPECT_STREQ("", statistics_proxy_->GetStats().c_name.c_str());
 }
 
+TEST_F(ReceiveStatisticsProxyTest, ReportsTimingFrameInfo) {
+  TimingFrameInfo info;
+  info.rtp_timestamp = 123u;
+  statistics_proxy_->OnDecodedFrame(
+      CreateFrame(kWidth, kHeight), std::nullopt, TimeDelta::Zero(),
+      VideoContentType::UNSPECIFIED, VideoFrameType::kVideoFrameDelta, info);
+  EXPECT_THAT(FlushAndGetStats().timing_frame_info,
+              Optional(Field(&TimingFrameInfo::rtp_timestamp, 123u)));
+}
+
+TEST_F(ReceiveStatisticsProxyTest, ReportsNoTimingFrameInfoWhenInvalid) {
+  TimingFrameInfo info;
+  info.flags = VideoSendTiming::kInvalid;
+  statistics_proxy_->OnDecodedFrame(
+      CreateFrame(kWidth, kHeight), std::nullopt, TimeDelta::Zero(),
+      VideoContentType::UNSPECIFIED, VideoFrameType::kVideoFrameDelta, info);
+  EXPECT_EQ(FlushAndGetStats().timing_frame_info, std::nullopt);
+}
+
+TEST_F(ReceiveStatisticsProxyTest, UpdatesFirstFrameReceivedToDecodedTime) {
+  const TimeDelta kDecodeTime = TimeDelta::Millis(14);
+  TimingFrameInfo info;
+  info.receive_finish_ms = Now().ms();
+  time_controller_.AdvanceTime(kDecodeTime);
+  statistics_proxy_->OnDecodedFrame(
+      CreateFrame(kWidth, kHeight), std::nullopt, TimeDelta::Zero(),
+      VideoContentType::UNSPECIFIED, VideoFrameType::kVideoFrameDelta, info);
+  EXPECT_EQ(FlushAndGetStats().first_frame_received_to_decoded_ms,
+            kDecodeTime.ms());
+}
+
 TEST_F(ReceiveStatisticsProxyTest, ReportsLongestTimingFrameInfo) {
   const int64_t kShortEndToEndDelay = 10;
   const int64_t kMedEndToEndDelay = 20;
   const int64_t kLongEndToEndDelay = 100;
   const uint32_t kExpectedRtpTimestamp = 2;
   TimingFrameInfo info;
-  std::optional<TimingFrameInfo> result;
   info.rtp_timestamp = kExpectedRtpTimestamp - 1;
   info.capture_time_ms = 0;
   info.decode_finish_ms = kShortEndToEndDelay;
@@ -742,29 +780,27 @@ TEST_F(ReceiveStatisticsProxyTest, ReportsLongestTimingFrameInfo) {
   info.capture_time_ms = 0;
   info.decode_finish_ms = kMedEndToEndDelay;
   statistics_proxy_->OnTimingFrameInfoUpdated(info);
-  result = FlushAndGetStats().timing_frame_info;
-  EXPECT_TRUE(result);
-  EXPECT_EQ(kExpectedRtpTimestamp, result->rtp_timestamp);
+  EXPECT_THAT(
+      FlushAndGetStats().timing_frame_info,
+      Optional(Field(&TimingFrameInfo::rtp_timestamp, kExpectedRtpTimestamp)));
 }
 
 TEST_F(ReceiveStatisticsProxyTest, RespectsReportingIntervalForTimingFrames) {
-  TimingFrameInfo info;
   const int64_t kShortEndToEndDelay = 10;
   const uint32_t kExpectedRtpTimestamp = 2;
   const TimeDelta kShortDelay = TimeDelta::Seconds(1);
   const TimeDelta kLongDelay = TimeDelta::Seconds(10);
-  std::optional<TimingFrameInfo> result;
+  TimingFrameInfo info;
   info.rtp_timestamp = kExpectedRtpTimestamp;
   info.capture_time_ms = 0;
   info.decode_finish_ms = kShortEndToEndDelay;
   statistics_proxy_->OnTimingFrameInfoUpdated(info);
   time_controller_.AdvanceTime(kShortDelay);
-  result = FlushAndGetStats().timing_frame_info;
-  EXPECT_TRUE(result);
-  EXPECT_EQ(kExpectedRtpTimestamp, result->rtp_timestamp);
+  EXPECT_THAT(
+      FlushAndGetStats().timing_frame_info,
+      Optional(Field(&TimingFrameInfo::rtp_timestamp, kExpectedRtpTimestamp)));
   time_controller_.AdvanceTime(kLongDelay);
-  result = statistics_proxy_->GetStats().timing_frame_info;
-  EXPECT_FALSE(result);
+  EXPECT_EQ(statistics_proxy_->GetStats().timing_frame_info, std::nullopt);
 }
 
 TEST_F(ReceiveStatisticsProxyTest, LifetimeHistogramIsUpdated) {
@@ -776,7 +812,7 @@ TEST_F(ReceiveStatisticsProxyTest, LifetimeHistogramIsUpdated) {
   statistics_proxy_->OnCompleteFrame(true, 1000, VideoContentType::UNSPECIFIED);
   statistics_proxy_->OnDecodedFrame(
       frame, std::nullopt, TimeDelta::Millis(1000),
-      VideoContentType::UNSPECIFIED, VideoFrameType::kVideoFrameKey);
+      VideoContentType::UNSPECIFIED, VideoFrameType::kVideoFrameKey, Info());
   FlushAndGetStats();
 
   statistics_proxy_->UpdateHistograms(std::nullopt, StreamDataCounters(),
@@ -922,9 +958,10 @@ TEST_F(ReceiveStatisticsProxyTest,
   for (int i = 0; i < kMinRequiredSamples - 1; ++i) {
     statistics_proxy_->OnCompleteFrame(kIsKeyFrame, kFrameSizeBytes,
                                        VideoContentType::UNSPECIFIED);
-    statistics_proxy_->OnDecodedFrame(
-        frame, std::nullopt, TimeDelta::Millis(1000),
-        VideoContentType::UNSPECIFIED, VideoFrameType::kVideoFrameDelta);
+    statistics_proxy_->OnDecodedFrame(frame, std::nullopt,
+                                      TimeDelta::Millis(1000),
+                                      VideoContentType::UNSPECIFIED,
+                                      VideoFrameType::kVideoFrameDelta, Info());
   }
   FlushAndGetStats();
 
@@ -947,9 +984,10 @@ TEST_F(ReceiveStatisticsProxyTest,
   for (int i = 0; i < kMinRequiredSamples; ++i) {
     statistics_proxy_->OnCompleteFrame(kIsKeyFrame, kFrameSizeBytes,
                                        VideoContentType::UNSPECIFIED);
-    statistics_proxy_->OnDecodedFrame(
-        frame, std::nullopt, TimeDelta::Millis(1000),
-        VideoContentType::UNSPECIFIED, VideoFrameType::kVideoFrameDelta);
+    statistics_proxy_->OnDecodedFrame(frame, std::nullopt,
+                                      TimeDelta::Millis(1000),
+                                      VideoContentType::UNSPECIFIED,
+                                      VideoFrameType::kVideoFrameDelta, Info());
   }
   FlushAndGetStats();
 
@@ -974,14 +1012,15 @@ TEST_F(ReceiveStatisticsProxyTest, KeyFrameHistogramIsUpdated) {
                                        VideoContentType::UNSPECIFIED);
     statistics_proxy_->OnDecodedFrame(
         frame, std::nullopt, TimeDelta::Millis(1000),
-        VideoContentType::UNSPECIFIED, VideoFrameType::kVideoFrameKey);
+        VideoContentType::UNSPECIFIED, VideoFrameType::kVideoFrameKey, Info());
   }
   for (int i = 0; i < kMinRequiredSamples; ++i) {
     statistics_proxy_->OnCompleteFrame(false, kFrameSizeBytes,
                                        VideoContentType::UNSPECIFIED);
-    statistics_proxy_->OnDecodedFrame(
-        frame, std::nullopt, TimeDelta::Millis(1000),
-        VideoContentType::UNSPECIFIED, VideoFrameType::kVideoFrameDelta);
+    statistics_proxy_->OnDecodedFrame(frame, std::nullopt,
+                                      TimeDelta::Millis(1000),
+                                      VideoContentType::UNSPECIFIED,
+                                      VideoFrameType::kVideoFrameDelta, Info());
   }
   FlushAndGetStats();
 
@@ -1064,7 +1103,7 @@ TEST_F(ReceiveStatisticsProxyTest, DoesNotReportStaleFramerates) {
         time_controller_.GetClock()->CurrentNtpInMilliseconds());
     statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
                                       VideoContentType::UNSPECIFIED,
-                                      VideoFrameType::kVideoFrameKey);
+                                      VideoFrameType::kVideoFrameKey, Info());
     statistics_proxy_->OnRenderedFrame(MetaData(frame));
     time_controller_.AdvanceTime(1 / kDefaultFps);
   }
@@ -1138,7 +1177,7 @@ TEST_F(ReceiveStatisticsProxyTest, ZeroDelayReportedIfFrameNotDelayed) {
   VideoFrame frame = CreateFrame(kWidth, kHeight);
   statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
                                     VideoContentType::UNSPECIFIED,
-                                    VideoFrameType::kVideoFrameKey);
+                                    VideoFrameType::kVideoFrameKey, Info());
 
   // Frame not delayed, delayed frames to render: 0%.
   statistics_proxy_->OnRenderedFrame(
@@ -1160,7 +1199,7 @@ TEST_F(ReceiveStatisticsProxyTest,
   VideoFrame frame = CreateFrame(kWidth, kHeight);
   statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
                                     VideoContentType::UNSPECIFIED,
-                                    VideoFrameType::kVideoFrameKey);
+                                    VideoFrameType::kVideoFrameKey, Info());
 
   // Frame not delayed, delayed frames to render: 0%.
   statistics_proxy_->OnRenderedFrame(
@@ -1181,7 +1220,7 @@ TEST_F(ReceiveStatisticsProxyTest,
   VideoFrame frame = CreateFrame(kWidth, kHeight);
   statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
                                     VideoContentType::UNSPECIFIED,
-                                    VideoFrameType::kVideoFrameKey);
+                                    VideoFrameType::kVideoFrameKey, Info());
 
   // Min run time has passed. No rendered frames.
   time_controller_.AdvanceTime(metrics::kMinRunTime);
@@ -1197,7 +1236,7 @@ TEST_F(ReceiveStatisticsProxyTest, DelayReportedIfFrameIsDelayed) {
   VideoFrame frame = CreateFrame(kWidth, kHeight);
   statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
                                     VideoContentType::UNSPECIFIED,
-                                    VideoFrameType::kVideoFrameKey);
+                                    VideoFrameType::kVideoFrameKey, Info());
 
   // Frame delayed 1 ms, delayed frames to render: 100%.
   statistics_proxy_->OnRenderedFrame(
@@ -1221,7 +1260,7 @@ TEST_F(ReceiveStatisticsProxyTest, AverageDelayOfDelayedFramesIsReported) {
   VideoFrame frame = CreateFrame(kWidth, kHeight);
   statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
                                     VideoContentType::UNSPECIFIED,
-                                    VideoFrameType::kVideoFrameKey);
+                                    VideoFrameType::kVideoFrameKey, Info());
 
   // Two frames delayed (6 ms, 10 ms), delayed frames to render: 50%.
   const int64_t kNowMs = Now().ms();
@@ -1372,14 +1411,14 @@ TEST_P(ReceiveStatisticsProxyTestWithContent, InterFrameDelaysAreReported) {
   for (int i = 0; i < kMinRequiredSamples; ++i) {
     statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
                                       content_type_,
-                                      VideoFrameType::kVideoFrameKey);
+                                      VideoFrameType::kVideoFrameKey, Info());
     time_controller_.AdvanceTime(kInterFrameDelay);
   }
   // One extra with double the interval.
   time_controller_.AdvanceTime(kInterFrameDelay);
   statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
                                     content_type_,
-                                    VideoFrameType::kVideoFrameDelta);
+                                    VideoFrameType::kVideoFrameDelta, Info());
 
   FlushAndUpdateHistograms(std::nullopt, StreamDataCounters(), nullptr);
   const TimeDelta kExpectedInterFrame =
@@ -1410,20 +1449,20 @@ TEST_P(ReceiveStatisticsProxyTestWithContent,
     time_controller_.AdvanceTime(kInterFrameDelay);
     statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
                                       content_type_,
-                                      VideoFrameType::kVideoFrameKey);
+                                      VideoFrameType::kVideoFrameKey, Info());
   }
   // Last 5% of intervals are double in size.
   for (int i = 0; i < kLastFivePercentsSamples; ++i) {
     time_controller_.AdvanceTime(2 * kInterFrameDelay);
     statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
                                       content_type_,
-                                      VideoFrameType::kVideoFrameKey);
+                                      VideoFrameType::kVideoFrameKey, Info());
   }
   // Final sample is outlier and 10 times as big.
   time_controller_.AdvanceTime(10 * kInterFrameDelay);
   statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
                                     content_type_,
-                                    VideoFrameType::kVideoFrameKey);
+                                    VideoFrameType::kVideoFrameKey, Info());
 
   FlushAndUpdateHistograms(std::nullopt, StreamDataCounters(), nullptr);
   const TimeDelta kExpectedInterFrame = kInterFrameDelay * 2;
@@ -1447,7 +1486,7 @@ TEST_P(ReceiveStatisticsProxyTestWithContent,
   for (int i = 0; i < kMinRequiredSamples; ++i) {
     statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
                                       content_type_,
-                                      VideoFrameType::kVideoFrameKey);
+                                      VideoFrameType::kVideoFrameKey, Info());
     time_controller_.AdvanceTime(kInterFrameDelay);
   }
 
@@ -1471,7 +1510,7 @@ TEST_P(ReceiveStatisticsProxyTestWithContent, MaxInterFrameDelayOnlyWithPause) {
   for (int i = 0; i <= kMinRequiredSamples; ++i) {
     statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
                                       content_type_,
-                                      VideoFrameType::kVideoFrameKey);
+                                      VideoFrameType::kVideoFrameKey, Info());
     time_controller_.AdvanceTime(kInterFrameDelay);
   }
 
@@ -1483,11 +1522,11 @@ TEST_P(ReceiveStatisticsProxyTestWithContent, MaxInterFrameDelayOnlyWithPause) {
   // disregarded in the stats.
   statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
                                     content_type_,
-                                    VideoFrameType::kVideoFrameKey);
+                                    VideoFrameType::kVideoFrameKey, Info());
   time_controller_.AdvanceTime(kInterFrameDelay);
   statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
                                     content_type_,
-                                    VideoFrameType::kVideoFrameDelta);
+                                    VideoFrameType::kVideoFrameDelta, Info());
 
   FlushAndUpdateHistograms(std::nullopt, StreamDataCounters(), nullptr);
   if (videocontenttypehelpers::IsScreenshare(content_type_)) {
@@ -1609,7 +1648,7 @@ TEST_P(ReceiveStatisticsProxyTestWithContent, HarmonicFrameRateIsReported) {
     time_controller_.AdvanceTime(kFrameDuration);
     statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
                                       content_type_,
-                                      VideoFrameType::kVideoFrameKey);
+                                      VideoFrameType::kVideoFrameKey, Info());
     statistics_proxy_->OnRenderedFrame(MetaData(frame));
   }
 
@@ -1618,7 +1657,7 @@ TEST_P(ReceiveStatisticsProxyTestWithContent, HarmonicFrameRateIsReported) {
   time_controller_.AdvanceTime(kFreezeDuration);
   statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
                                     content_type_,
-                                    VideoFrameType::kVideoFrameDelta);
+                                    VideoFrameType::kVideoFrameDelta, Info());
   statistics_proxy_->OnRenderedFrame(MetaData(frame));
 
   // Add pause.
@@ -1626,7 +1665,7 @@ TEST_P(ReceiveStatisticsProxyTestWithContent, HarmonicFrameRateIsReported) {
   statistics_proxy_->OnStreamInactive();
   statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
                                     content_type_,
-                                    VideoFrameType::kVideoFrameDelta);
+                                    VideoFrameType::kVideoFrameDelta, Info());
   statistics_proxy_->OnRenderedFrame(MetaData(frame));
 
   FlushAndUpdateHistograms(std::nullopt, StreamDataCounters(), nullptr);
@@ -1702,13 +1741,13 @@ TEST_P(ReceiveStatisticsProxyTestWithContent, ManyPausesAtTheBeginning) {
   for (int i = 0; i <= kMinRequiredSamples; ++i) {
     statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
                                       content_type_,
-                                      VideoFrameType::kVideoFrameKey);
+                                      VideoFrameType::kVideoFrameKey, Info());
     time_controller_.AdvanceTime(kInterFrameDelay);
     statistics_proxy_->OnStreamInactive();
     time_controller_.AdvanceTime(kPauseDuration);
     statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
                                       content_type_,
-                                      VideoFrameType::kVideoFrameDelta);
+                                      VideoFrameType::kVideoFrameDelta, Info());
     time_controller_.AdvanceTime(kInterFrameDelay);
   }
 
@@ -1791,7 +1830,7 @@ TEST_P(ReceiveStatisticsProxyTestWithContent, TimeInBlockyVideoReported) {
   // Extra last frame.
   statistics_proxy_->OnDecodedFrame(frame, kHighQp, TimeDelta::Zero(),
                                     content_type_,
-                                    VideoFrameType::kVideoFrameKey);
+                                    VideoFrameType::kVideoFrameKey, Info());
   statistics_proxy_->OnRenderedFrame(MetaData(frame));
 
   FlushAndUpdateHistograms(std::nullopt, StreamDataCounters(), nullptr);
@@ -1819,7 +1858,7 @@ TEST_P(ReceiveStatisticsProxyTestWithContent, DownscalesReported) {
   // Call once to pass content type.
   statistics_proxy_->OnDecodedFrame(frame_hd, std::nullopt, TimeDelta::Zero(),
                                     content_type_,
-                                    VideoFrameType::kVideoFrameKey);
+                                    VideoFrameType::kVideoFrameKey, Info());
 
   time_controller_.AdvanceTime(TimeDelta::Zero());
   statistics_proxy_->OnRenderedFrame(MetaData(frame_hd));
@@ -1849,7 +1888,7 @@ TEST_P(ReceiveStatisticsProxyTestWithContent, DecodeTimeReported) {
 
   for (int i = 0; i < kMinRequiredSamples; ++i) {
     statistics_proxy_->OnDecodedFrame(frame, kLowQp, kDecodeTime, content_type_,
-                                      VideoFrameType::kVideoFrameKey);
+                                      VideoFrameType::kVideoFrameKey, Info());
     time_controller_.AdvanceTime(kInterFrameDelay);
   }
   FlushAndUpdateHistograms(std::nullopt, StreamDataCounters(), nullptr);
diff --git a/video/timing/simulator/rendering_tracker.h b/video/timing/simulator/rendering_tracker.h
index de33fae833..0e0dc75727 100644
--- a/video/timing/simulator/rendering_tracker.h
+++ b/video/timing/simulator/rendering_tracker.h
@@ -26,7 +26,6 @@
 #include "api/video/video_content_type.h"
 #include "api/video/video_frame.h"
 #include "api/video/video_sink_interface.h"
-#include "api/video/video_timing.h"
 #include "modules/video_coding/timing/timing.h"
 #include "rtc_base/thread_annotations.h"
 #include "video/timing/simulator/assembler.h"
@@ -99,7 +98,6 @@ class RenderingTracker : public AssembledFrameCallback,
                         TimeDelta jitter_buffer_target_delay,
                         TimeDelta jitter_buffer_minimum_delay) override;
   void OnFrameBufferTimingsUpdated(int, int, int, int, int, int) override {}
-  void OnTimingFrameInfoUpdated(const TimingFrameInfo&) override {}
 
   // Implements `VideoSinkInterface<VideoFrame>`.
   void OnFrame(const VideoFrame& decoded_frame) override;
diff --git a/video/video_stream_buffer_controller.cc b/video/video_stream_buffer_controller.cc
index ff778633c9..21e94242b9 100644
--- a/video/video_stream_buffer_controller.cc
+++ b/video/video_stream_buffer_controller.cc
@@ -28,7 +28,6 @@
 #include "api/video/encoded_frame.h"
 #include "api/video/frame_buffer.h"
 #include "api/video/video_content_type.h"
-#include "api/video/video_timing.h"
 #include "modules/video_coding/frame_helpers.h"
 #include "modules/video_coding/include/video_coding_defines.h"
 #include "modules/video_coding/timing/inter_frame_delay_variation_calculator.h"
@@ -265,7 +264,6 @@ void VideoStreamBufferController::OnFrameReady(
   // Update stats.
   UpdateDroppedFrames();
   UpdateFrameBufferTimings(min_receive_time, now);
-  UpdateTimingFrameInfo();
 
   std::unique_ptr<EncodedFrame> frame =
       CombineAndDeleteFrames(std::move(frames));
@@ -356,12 +354,6 @@ void VideoStreamBufferController::UpdateFrameBufferTimings(
                                  timings.minimum_delay);
 }
 
-void VideoStreamBufferController::UpdateTimingFrameInfo() {
-  std::optional<TimingFrameInfo> info = timing_->GetTimingFrameInfo();
-  if (info)
-    stats_proxy_->OnTimingFrameInfoUpdated(*info);
-}
-
 bool VideoStreamBufferController::IsTooManyFramesQueued() const
     RTC_RUN_ON(&worker_sequence_checker_) {
   return buffer_->CurrentSize() > zero_playout_delay_max_decode_queue_size_;
diff --git a/video/video_stream_buffer_controller.h b/video/video_stream_buffer_controller.h
index 17227871cc..93a98fc7e3 100644
--- a/video/video_stream_buffer_controller.h
+++ b/video/video_stream_buffer_controller.h
@@ -26,7 +26,6 @@
 #include "api/video/encoded_frame.h"
 #include "api/video/frame_buffer.h"
 #include "api/video/video_content_type.h"
-#include "api/video/video_timing.h"
 #include "modules/video_coding/include/video_coding_defines.h"
 #include "modules/video_coding/timing/inter_frame_delay_variation_calculator.h"
 #include "modules/video_coding/timing/jitter_estimator.h"
@@ -73,8 +72,6 @@ class VideoStreamBufferControllerStatsObserver {
                                            int jitter_delay_ms,
                                            int min_playout_delay_ms,
                                            int render_delay_ms) = 0;
-
-  virtual void OnTimingFrameInfoUpdated(const TimingFrameInfo& info) = 0;
 };
 
 class VideoStreamBufferController {
@@ -109,7 +106,6 @@ class VideoStreamBufferController {
   void FrameReadyForDecode(uint32_t rtp_timestamp, Timestamp render_time);
   void UpdateDroppedFrames() RTC_RUN_ON(&worker_sequence_checker_);
   void UpdateFrameBufferTimings(Timestamp min_receive_time, Timestamp now);
-  void UpdateTimingFrameInfo();
   bool IsTooManyFramesQueued() const RTC_RUN_ON(&worker_sequence_checker_);
   void ForceKeyFrameReleaseImmediately() RTC_RUN_ON(&worker_sequence_checker_);
   void MaybeScheduleFrameForRelease() RTC_RUN_ON(&worker_sequence_checker_);
diff --git a/video/video_stream_buffer_controller_unittest.cc b/video/video_stream_buffer_controller_unittest.cc
index e7e8dc1caa..26aacb7c7d 100644
--- a/video/video_stream_buffer_controller_unittest.cc
+++ b/video/video_stream_buffer_controller_unittest.cc
@@ -123,10 +123,6 @@ class VideoStreamBufferControllerStatsObserverMock
                int min_playout_delay_ms,
                int render_delay_ms),
               (override));
-  MOCK_METHOD(void,
-              OnTimingFrameInfoUpdated,
-              (const TimingFrameInfo& info),
-              (override));
 };
 
 }  // namespace
diff --git a/video/video_stream_decoder2.cc b/video/video_stream_decoder2.cc
index 45379a087a..f8b8a82f51 100644
--- a/video/video_stream_decoder2.cc
+++ b/video/video_stream_decoder2.cc
@@ -52,7 +52,8 @@ VideoStreamDecoder::~VideoStreamDecoder() {
 int32_t VideoStreamDecoder::OnFrameToRender(const FrameToRender& arguments) {
   receive_stats_callback_->OnDecodedFrame(
       arguments.video_frame, arguments.qp, arguments.decode_time,
-      arguments.content_type, arguments.frame_type);
+      arguments.content_type, arguments.frame_type,
+      arguments.timing_frame_info);
   incoming_video_stream_->OnFrame(arguments.video_frame);
   return 0;
 }

From 5560f130af58aaf10395207f601d2f2fbad4b975 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Wed, 25 Mar 2026 18:54:08 +0100
Subject: [PATCH 413/472] Truncate CSRC list in RtpPacket::SetCsrcs if too
 large

Truncates the CSRC list to a maximum of 15 entries as per the 4-bit RTP
header field limitation, and logs a warning when truncation occurs. Also
updates the buffer capacity check and allocation to happen prior to
writing data to avoid heap buffer overflow.

Additionally, ensure that the underlying buffer size is updated before
writing the CSRC data. By calling SetSize prior to WriteAt, we prevent
potential heap buffer overflows that could occur if the buffer was not
sufficiently allocated before the write operation.

Bug: chromium:486317116
Change-Id: I2821867f94c0a8d174fb25047ac1f4b94ee5b867
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/459943
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47266}
---
 modules/rtp_rtcp/source/rtp_packet.cc         | 12 +++++++---
 modules/rtp_rtcp/source/rtp_packet.h          |  6 +++++
 .../rtp_rtcp/source/rtp_packet_unittest.cc    | 23 +++++++++++++++++++
 3 files changed, 38 insertions(+), 3 deletions(-)

diff --git a/modules/rtp_rtcp/source/rtp_packet.cc b/modules/rtp_rtcp/source/rtp_packet.cc
index 97f6b54148..2b6cd006df 100644
--- a/modules/rtp_rtcp/source/rtp_packet.cc
+++ b/modules/rtp_rtcp/source/rtp_packet.cc
@@ -220,16 +220,22 @@ void RtpPacket::SetCsrcs(std::span<const uint32_t> csrcs) {
   RTC_DCHECK_EQ(extensions_size_, 0);
   RTC_DCHECK_EQ(payload_size_, 0);
   RTC_DCHECK_EQ(padding_size_, 0);
-  RTC_DCHECK_LE(csrcs.size(), 0x0fu);
-  RTC_DCHECK_LE(kFixedHeaderSize + 4 * csrcs.size(), capacity());
+
+  if (csrcs.size() > kMaxCsrcs) {
+    RTC_LOG(LS_ERROR) << "Truncating CSRC list, length exceeded " << kMaxCsrcs
+                      << ": " << csrcs.size();
+    csrcs = csrcs.first(kMaxCsrcs);
+  }
+
   payload_offset_ = kFixedHeaderSize + 4 * csrcs.size();
+  buffer_.SetSize(payload_offset_);  // SetSize before WriteAt.
+
   WriteAt(0, (data()[0] & 0xF0) | dchecked_cast<uint8_t>(csrcs.size()));
   size_t offset = kFixedHeaderSize;
   for (uint32_t csrc : csrcs) {
     ByteWriter<uint32_t>::WriteBigEndian(WriteAt(offset), csrc);
     offset += 4;
   }
-  buffer_.SetSize(payload_offset_);
 }
 
 std::span<uint8_t> RtpPacket::AllocateRawExtension(int id, size_t length) {
diff --git a/modules/rtp_rtcp/source/rtp_packet.h b/modules/rtp_rtcp/source/rtp_packet.h
index 1a9f91799b..78cccb347d 100644
--- a/modules/rtp_rtcp/source/rtp_packet.h
+++ b/modules/rtp_rtcp/source/rtp_packet.h
@@ -30,6 +30,12 @@ class RtpPacket {
   using ExtensionType = RTPExtensionType;
   using ExtensionManager = RtpHeaderExtensionMap;
 
+  // Maximum number of CSRCs in an RTP packet as specified in section
+  // "5.1 RTP Fixed Header Fields" of RFC 3550.
+  // Note: This is a different limit than the one that applies to RTCP packets
+  // (which is specified in section 6.1).
+  static constexpr size_t kMaxCsrcs = 15;
+
   // `extensions` required for SetExtension/ReserveExtension functions during
   // packet creating and used if available in Parse function.
   // Adding and getting extensions will fail until `extensions` is
diff --git a/modules/rtp_rtcp/source/rtp_packet_unittest.cc b/modules/rtp_rtcp/source/rtp_packet_unittest.cc
index 37b5f62d66..1343d0a7b4 100644
--- a/modules/rtp_rtcp/source/rtp_packet_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_packet_unittest.cc
@@ -17,6 +17,7 @@
 #include <string>
 #include <type_traits>
 #include <utility>
+#include <vector>
 
 #include "absl/strings/string_view.h"
 #include "api/rtp_headers.h"
@@ -1347,5 +1348,27 @@ TEST(RtpPacketTest, SetExtensionWithArray) {
               ElementsAreArray(extension_data));
 }
 
+TEST(RtpPacketTest, SetCsrcsTruncatesWhenExceedingMax) {
+  RtpPacketToSend packet(nullptr);
+  packet.SetPayloadType(kPayloadType);
+  packet.SetSequenceNumber(kSeqNum);
+  packet.SetTimestamp(kTimestamp);
+  packet.SetSsrc(kSsrc);
+
+  std::vector<uint32_t> many_csrcs;
+  for (uint32_t i = 0; i < 20; ++i) {
+    many_csrcs.push_back(kSsrc + i);
+  }
+
+  // SetCsrcs should truncate to maximum elements allowed.
+  packet.SetCsrcs(many_csrcs);
+
+  std::vector<uint32_t> csrcs = packet.Csrcs();
+  EXPECT_EQ(csrcs.size(), RtpPacket::kMaxCsrcs);
+  for (size_t i = 0; i < RtpPacket::kMaxCsrcs; ++i) {
+    EXPECT_EQ(csrcs[i], many_csrcs[i]);
+  }
+}
+
 }  // namespace
 }  // namespace webrtc

From 989cdaba9845f4946690da7e6a7efb6cd506d40b Mon Sep 17 00:00:00 2001
From: Sam Zackrisson <saza@webrtc.org>
Date: Mon, 16 Mar 2026 09:38:39 +0000
Subject: [PATCH 414/472] Delete the AEC3 json config fuzzer

The code is test-only nowadays.
Additionally, the JSON parser we use is not robust enough for fuzzing the config: It crashes on deep nesting.

Bug: chromium:440374794, chromium:441805811, chromium:441805815
Change-Id: Icd6daad3dfc500184c5a9dc3dce0bb15815cb516
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/457240
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47267}
---
 test/fuzzers/BUILD.gn                         | 13 --------
 test/fuzzers/aec3_config_json_fuzzer.cc       | 31 -------------------
 .../aec3-config-json-corpus/default.json      |  1 -
 3 files changed, 45 deletions(-)
 delete mode 100644 test/fuzzers/aec3_config_json_fuzzer.cc
 delete mode 100644 test/fuzzers/corpora/aec3-config-json-corpus/default.json

diff --git a/test/fuzzers/BUILD.gn b/test/fuzzers/BUILD.gn
index e690727fd8..34244b1a03 100644
--- a/test/fuzzers/BUILD.gn
+++ b/test/fuzzers/BUILD.gn
@@ -593,18 +593,6 @@ webrtc_fuzzer_test("agc_fuzzer") {
   seed_corpus = "corpora/agc-corpus"
 }
 
-webrtc_fuzzer_test("aec3_config_json_fuzzer") {
-  sources = [ "aec3_config_json_fuzzer.cc" ]
-  deps = [
-    ":fuzz_data_helper",
-    "../../api/audio:aec3_config",
-    "../../modules/audio_processing:aec3_config_json",
-    "//third_party/abseil-cpp/absl/strings:string_view",
-  ]
-  dict = "//testing/libfuzzer/fuzzers/dicts/json.dict"
-  seed_corpus = "corpora/aec3-config-json-corpus"
-}
-
 webrtc_fuzzer_test("aec3_fuzzer") {
   defines = []
   if (apm_debug_dump) {
@@ -818,7 +806,6 @@ webrtc_fuzzer_test("windowed_min_filter_fuzzer") {
 group("fuzzers") {
   testonly = true
   deps = [
-    ":aec3_config_json_fuzzer",
     ":aec3_fuzzer",
     ":agc_fuzzer",
     ":audio_decoder_fuzzer",
diff --git a/test/fuzzers/aec3_config_json_fuzzer.cc b/test/fuzzers/aec3_config_json_fuzzer.cc
deleted file mode 100644
index 0eb19e9c1c..0000000000
--- a/test/fuzzers/aec3_config_json_fuzzer.cc
+++ /dev/null
@@ -1,31 +0,0 @@
-/*
- *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <cstddef>
-
-#include "absl/strings/string_view.h"
-#include "api/audio/echo_canceller3_config.h"
-#include "modules/audio_processing/test/echo_canceller3_config_json.h"
-#include "test/fuzzers/fuzz_data_helper.h"
-
-namespace webrtc {
-void FuzzOneInput(FuzzDataHelper fuzz_data) {
-  if (fuzz_data.size() > 10'000) {
-    return;
-  }
-  absl::string_view config_json = fuzz_data.ReadString();
-
-  EchoCanceller3Config config;
-  bool success;
-  Aec3ConfigFromJsonString(config_json, &config, &success);
-  EchoCanceller3Config::Validate(&config);
-}
-
-}  // namespace webrtc
diff --git a/test/fuzzers/corpora/aec3-config-json-corpus/default.json b/test/fuzzers/corpora/aec3-config-json-corpus/default.json
deleted file mode 100644
index 4dcbcee53a..0000000000
--- a/test/fuzzers/corpora/aec3-config-json-corpus/default.json
+++ /dev/null
@@ -1 +0,0 @@
-{"aec3": {"buffering": {"excess_render_detection_interval_blocks": 250,"max_allowed_excess_render_blocks": 8},"delay": {"default_delay": 5,"down_sampling_factor": 4,"num_filters": 5,"delay_headroom_samples": 32,"hysteresis_limit_blocks": 1,"fixed_capture_delay_samples": 0,"delay_estimate_smoothing": 0.7,"delay_candidate_detection_threshold": 0.2,"delay_selection_thresholds": {"initial": 5,"converged": 20},"use_external_delay_estimator": false,"downmix_before_delay_estimation": true,"log_warning_on_delay_changes": false},"filter": {"main": [13,5e-05,0.05,0.001,2,2.00753e+07],"shadow": [13,0.7,2.00753e+07],"main_initial": [12,0.005,0.5,0.001,2,2.00753e+07],"shadow_initial": [12,0.9,2.00753e+07],"config_change_duration_blocks": 250,"initial_state_seconds": 2.5,"conservative_initial_phase": false,"enable_shadow_filter_output_usage": true},"erle": {"min": 1,"max_l": 4,"max_h": 1.5,"onset_detection": true,"num_sections": 1,"clamp_quality_estimate_to_zero": true,"clamp_quality_estimate_to_one": true},"ep_strength": {"default_gain": 1,"default_len": 0.83,"echo_can_saturate": true,"bounded_erl": false},"echo_audibility": {"low_render_limit": 256,"normal_render_limit": 64,"floor_power": 128,"audibility_threshold_lf": 10,"audibility_threshold_mf": 10,"audibility_threshold_hf": 10,"use_stationarity_properties": false,"use_stationarity_properties_at_init": false},"render_levels": {"active_render_limit": 100,"poor_excitation_render_limit": 150,"poor_excitation_render_limit_ds8": 20},"echo_removal_control": {"has_clock_drift": false,"linear_and_stable_echo_path": false},"echo_model": {"noise_floor_hold": 50,"min_noise_floor_power": 1.6384e+06,"stationary_gate_slope": 10,"noise_gate_power": 27509.4,"noise_gate_slope": 0.3,"render_pre_window_size": 1,"render_post_window_size": 1},"suppressor": {"nearend_average_blocks": 4,"normal_tuning": {"mask_lf": [0.3,0.4,0.3],"mask_hf": [0.07,0.1,0.3],"max_inc_factor": 2,"max_dec_factor_lf": 0.25},"nearend_tuning": {"mask_lf": [1.09,1.1,0.3],"mask_hf": [0.1,0.3,0.3],"max_inc_factor": 2,"max_dec_factor_lf": 0.25},"dominant_nearend_detection": {"enr_threshold": 0.25,"enr_exit_threshold": 10,"snr_threshold": 30,"hold_duration": 50,"trigger_threshold": 12,"use_during_initial_phase": 1},"high_bands_suppression": {"enr_threshold": 1,"max_gain_during_echo": 1},"floor_first_increase": 1e-05,"enforce_transparent": false,"enforce_empty_higher_bands": false}}}

From 05946612a4cd0e295edba02b74c8cd3b45063fa6 Mon Sep 17 00:00:00 2001
From: Mirko Bonadei <mbonadei@webrtc.org>
Date: Thu, 26 Mar 2026 08:27:42 -0700
Subject: [PATCH 415/472] Revert "build: clean up modules audio build files"

This reverts commit 246b8dee2304df4a20513f595e315a0ca3df079c.

Reason for revert: Breaks downstream project

Original change's description:
> build: clean up modules audio build files
>
> using gn_check_autofix with cherry-picking removals
>
> Bug: webrtc:467294026
> Change-Id: Ia8fe74995724c193d7e92db1ed3e26d77e34422a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/460060
> Reviewed-by: Christoffer Dewerin <jansson@webrtc.org>
> Reviewed-by: Jeremy Leconte <jleconte@google.com>
> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
> Cr-Commit-Position: refs/heads/main@{#47263}

Bug: webrtc:467294026
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: Ia13ae59ae025dbf9a6ca28922d061c2d6c1441c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/460460
Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47268}
---
 modules/audio_coding/BUILD.gn                 | 61 +++++++++++++++++++
 .../audio_coding/codecs/opus/test/BUILD.gn    |  2 +
 modules/audio_device/BUILD.gn                 | 10 +++
 modules/audio_mixer/BUILD.gn                  |  9 +++
 modules/audio_processing/BUILD.gn             | 33 ++++++++++
 modules/audio_processing/aec3/BUILD.gn        |  7 +++
 modules/audio_processing/aec_dump/BUILD.gn    |  7 +++
 modules/audio_processing/agc/BUILD.gn         | 10 +++
 modules/audio_processing/agc2/BUILD.gn        | 24 ++++++++
 .../audio_processing/agc2/rnn_vad/BUILD.gn    |  4 ++
 .../capture_levels_adjuster/BUILD.gn          |  2 +
 .../audio_processing/capture_mixer/BUILD.gn   |  3 +
 modules/audio_processing/ns/BUILD.gn          | 18 ++++++
 .../test/conversational_speech/BUILD.gn       |  1 +
 modules/audio_processing/utility/BUILD.gn     |  1 +
 modules/audio_processing/vad/BUILD.gn         |  2 +
 16 files changed, 194 insertions(+)

diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index 6c23fca430..d8e1e9788a 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -36,19 +36,27 @@ rtc_library("audio_coding") {
 
   deps = [
     ":audio_coding_module_typedefs",
+    ":neteq",
+    "..:module_api",
     "..:module_api_public",
     "../../api:array_view",
     "../../api:function_view",
     "../../api/audio:audio_frame_api",
     "../../api/audio_codecs:audio_codecs_api",
+    "../../api/environment",
+    "../../api/neteq:default_neteq_factory",
+    "../../api/neteq:neteq_api",
+    "../../api/units:timestamp",
     "../../audio/utility:audio_frame_operations",
     "../../common_audio",
+    "../../common_audio:common_audio_c",
     "../../rtc_base:buffer",
     "../../rtc_base:checks",
     "../../rtc_base:logging",
     "../../rtc_base:macromagic",
     "../../rtc_base:safe_conversions",
     "../../rtc_base/synchronization:mutex",
+    "../../system_wrappers",
     "../../system_wrappers:metrics",
     "//third_party/abseil-cpp/absl/algorithm:container",
     "//third_party/abseil-cpp/absl/strings",
@@ -116,9 +124,11 @@ rtc_library("red") {
     "../../api:field_trials_view",
     "../../api/audio_codecs:audio_codecs_api",
     "../../api/units:time_delta",
+    "../../common_audio",
     "../../rtc_base:buffer",
     "../../rtc_base:byte_order",
     "../../rtc_base:checks",
+    "../../rtc_base:logging",
     "//third_party/abseil-cpp/absl/strings:string_view",
   ]
 }
@@ -171,6 +181,7 @@ rtc_library("g722") {
     "../../api/units:time_delta",
     "../../rtc_base:buffer",
     "../../rtc_base:checks",
+    "../../rtc_base:safe_conversions",
   ]
   public_deps += [ ":g722_c" ]  # no-presubmit-check TODO(webrtc:8603)
 }
@@ -202,6 +213,7 @@ rtc_library("isac_vad") {
   deps = [
     ":isac_bwinfo",
     "../../rtc_base:compile_assert_c",
+    "../../rtc_base/system:arch",
     "../../rtc_base/system:ignore_warnings",
     "../third_party/fft",
   ]
@@ -227,6 +239,7 @@ rtc_library("pcm16b") {
   deps = [
     ":g711",
     ":legacy_encoded_audio_frame",
+    "../../api:array_view",
     "../../api/audio_codecs:audio_codecs_api",
     "../../rtc_base:buffer",
     "../../rtc_base:checks",
@@ -252,6 +265,7 @@ rtc_library("audio_coding_opus_common") {
     "../../api:array_view",
     "../../api/audio_codecs:audio_codecs_api",
     "../../rtc_base:buffer",
+    "../../rtc_base:checks",
     "../../rtc_base:stringutils",
     "//third_party/abseil-cpp/absl/strings:string_view",
   ]
@@ -319,6 +333,7 @@ rtc_library("webrtc_multiopus") {
     "../../rtc_base:checks",
     "../../rtc_base:logging",
     "../../rtc_base:safe_conversions",
+    "../../rtc_base:safe_minmax",
     "../../rtc_base:stringutils",
     "//third_party/abseil-cpp/absl/memory",
     "//third_party/abseil-cpp/absl/strings",
@@ -502,6 +517,7 @@ rtc_library("neteq") {
   ]
 
   deps = [
+    ":audio_coding_module_typedefs",
     ":webrtc_cng",
     "..:module_api_public",
     "../../api:array_view",
@@ -518,17 +534,21 @@ rtc_library("neteq") {
     "../../api/neteq:tick_timer",
     "../../api/units:time_delta",
     "../../api/units:timestamp",
+    "../../common_audio",
     "../../common_audio:common_audio_c",
     "../../common_audio:common_audio_cc",
     "../../rtc_base:buffer",
     "../../rtc_base:checks",
     "../../rtc_base:event_tracer",
+    "../../rtc_base:gtest_prod",
     "../../rtc_base:logging",
+    "../../rtc_base:macromagic",
     "../../rtc_base:rtc_numerics",
     "../../rtc_base:safe_conversions",
     "../../rtc_base:safe_minmax",
     "../../rtc_base:sanitizer",
     "../../rtc_base/experiments:field_trial_parser",
+    "../../rtc_base/synchronization:mutex",
     "../../system_wrappers",
     "../../system_wrappers:metrics",
     "//third_party/abseil-cpp/absl/strings",
@@ -556,7 +576,9 @@ rtc_library("neteq_tools_minimal") {
   ]
 
   deps = [
+    ":neteq",
     "../../api:array_view",
+    "../../api:field_trials",
     "../../api:field_trials_view",
     "../../api:neteq_simulator_api",
     "../../api:rtp_headers",
@@ -565,11 +587,14 @@ rtc_library("neteq_tools_minimal") {
     "../../api/audio_codecs:audio_codecs_api",
     "../../api/environment",
     "../../api/environment:environment_factory",
+    "../../api/neteq:custom_neteq_factory",
+    "../../api/neteq:default_neteq_controller_factory",
     "../../api/neteq:default_neteq_factory",
     "../../api/neteq:neteq_api",
     "../../api/units:timestamp",
     "../../rtc_base:buffer",
     "../../rtc_base:checks",
+    "../../rtc_base:copy_on_write_buffer",
     "../../rtc_base:safe_conversions",
     "../../rtc_base:stringutils",
     "../../system_wrappers",
@@ -638,6 +663,7 @@ rtc_library("neteq_tools") {
   deps = [
     ":neteq_input_audio_tools",
     ":neteq_tools_minimal",
+    "..:module_api_public",
     "../../api:array_view",
     "../../api:rtp_headers",
     "../../api:rtp_packet_info",
@@ -652,6 +678,7 @@ rtc_library("neteq_tools") {
     "../../rtc_base:safe_conversions",
     "../../rtc_base:stringutils",
     "../../rtc_base:timeutils",
+    "../rtp_rtcp",
     "../rtp_rtcp:rtp_rtcp_format",
     "//third_party/abseil-cpp/absl/strings:string_view",
   ]
@@ -841,6 +868,7 @@ if (rtc_include_tests) {
       ":pcm16b_c",
       ":red",
       "../../api:array_view",
+      "../../api:field_trials",
       "../../api:rtp_headers",
       "../../api:scoped_refptr",
       "../../api/audio:audio_frame_api",
@@ -895,6 +923,8 @@ if (rtc_include_tests) {
       "../../api/test/metrics:global_metrics_logger_and_exporter",
       "../../api/test/metrics:metric",
       "../../rtc_base:buffer",
+      "../../rtc_base:timeutils",
+      "../../system_wrappers",
       "../../test:fileutils",
       "../../test:test_flags",
       "../../test:test_support",
@@ -913,7 +943,9 @@ if (rtc_include_tests) {
 
     deps = [
       ":audio_coding",
+      ":neteq_tools",
       ":neteq_tools_minimal",
+      "../../api:array_view",
       "../../api:scoped_refptr",
       "../../api/audio_codecs:audio_codecs_api",
       "../../api/audio_codecs:builtin_audio_decoder_factory",
@@ -941,13 +973,18 @@ if (rtc_include_tests) {
       ":audio_coding",
       ":audio_coding_module_typedefs",
       ":neteq_input_audio_tools",
+      ":neteq_tools",
       ":neteq_tools_minimal",
       "../../api/audio:audio_frame_api",
       "../../api/audio_codecs:audio_codecs_api",
+      "../../api/audio_codecs:builtin_audio_decoder_factory",
       "../../api/audio_codecs:builtin_audio_encoder_factory",
       "../../api/environment",
       "../../api/environment:environment_factory",
       "../../rtc_base:checks",
+      "../../rtc_base:copy_on_write_buffer",
+      "../../rtc_base:stringutils",
+      "../../test:test_support",
       "../rtp_rtcp:rtp_rtcp_format",
       "//testing/gtest",
       "//third_party/abseil-cpp/absl/strings",
@@ -1008,12 +1045,14 @@ if (rtc_include_tests) {
       visibility += webrtc_default_visibility
       defines = audio_codec_defines
       deps = [
+        ":neteq",
         ":neteq_input_audio_tools",
         ":neteq_test_tools",
         ":neteq_tools",
         ":neteq_tools_minimal",
         "../../api:field_trials",
         "../../api:make_ref_counted",
+        "../../api:rtp_headers",
         "../../api:scoped_refptr",
         "../../api/audio_codecs:audio_codecs_api",
         "../../api/audio_codecs:builtin_audio_decoder_factory",
@@ -1021,7 +1060,10 @@ if (rtc_include_tests) {
         "../../api/neteq:neteq_api",
         "../../logging:rtc_event_log_parser",
         "../../rtc_base:checks",
+        "../../rtc_base:refcount",
         "../../test:audio_test_common",
+        "../../test:fileutils",
+        "../../test:test_support",
         "../rtp_rtcp:rtp_rtcp_format",
         "//third_party/abseil-cpp/absl/strings:string_view",
       ]
@@ -1123,6 +1165,7 @@ if (rtc_include_tests) {
       "../../rtc_base:checks",
       "../../system_wrappers",
       "../../test:fileutils",
+      "../../test:test_support",
       "//testing/gtest",
     ]
   }
@@ -1136,6 +1179,7 @@ if (rtc_include_tests) {
       ]
 
       deps = [
+        ":neteq",
         ":neteq_input_audio_tools",
         ":neteq_test_tools",
         ":neteq_tools_minimal",
@@ -1151,6 +1195,7 @@ if (rtc_include_tests) {
         "../../rtc_base:buffer",
         "../../rtc_base:checks",
         "../../rtc_base:stringutils",
+        "../../system_wrappers",
         "../../test:fileutils",
         "../../test:test_support",
         "//testing/gtest",
@@ -1165,6 +1210,7 @@ if (rtc_include_tests) {
       deps = [
         ":audio_coding",
         ":audio_coding_module_typedefs",
+        ":audio_encoder_cng",
         ":neteq_input_audio_tools",
         "../../api/audio:audio_frame_api",
         "../../api/audio_codecs/L16:audio_encoder_L16",
@@ -1221,7 +1267,10 @@ if (rtc_include_tests) {
       sources = [ "neteq/tools/rtp_analyze.cc" ]
 
       deps = [
+        ":neteq",
         ":neteq_test_tools",
+        ":neteq_tools_minimal",
+        ":pcm16b",
         "../../api:rtp_headers",
         "../../rtc_base:checks",
         "../rtp_rtcp:rtp_rtcp_format",
@@ -1259,8 +1308,10 @@ if (rtc_include_tests) {
       sources = [ "neteq/test/neteq_speed_test.cc" ]
 
       deps = [
+        ":neteq",
         ":neteq_test_support",
         "../../rtc_base:checks",
+        "../../test:test_support",
         "//third_party/abseil-cpp/absl/flags:flag",
         "//third_party/abseil-cpp/absl/flags:parse",
       ]
@@ -1273,12 +1324,14 @@ if (rtc_include_tests) {
 
       deps = [
         ":g711",
+        ":neteq",
         ":neteq_quality_test_support",
         "../../api:array_view",
         "../../api/audio_codecs:audio_codecs_api",
         "../../rtc_base:buffer",
         "../../rtc_base:checks",
         "../../rtc_base:safe_conversions",
+        "../../test:fileutils",
         "../../test:test_support",
         "//testing/gtest",
         "//third_party/abseil-cpp/absl/flags:flag",
@@ -1291,6 +1344,7 @@ if (rtc_include_tests) {
       sources = [ "neteq/test/neteq_pcm16b_quality_test.cc" ]
 
       deps = [
+        ":neteq",
         ":neteq_quality_test_support",
         ":pcm16b",
         "../../api:array_view",
@@ -1298,6 +1352,7 @@ if (rtc_include_tests) {
         "../../rtc_base:buffer",
         "../../rtc_base:checks",
         "../../rtc_base:safe_conversions",
+        "../../test:fileutils",
         "../../test:test_support",
         "//testing/gtest",
         "//third_party/abseil-cpp/absl/flags:flag",
@@ -1473,12 +1528,18 @@ if (rtc_include_tests) {
         "../../logging:rtc_event_audio",
         "../../rtc_base:buffer",
         "../../rtc_base:checks",
+        "../../rtc_base:copy_on_write_buffer",
         "../../rtc_base:digest",
         "../../rtc_base:ip_address",
+        "../../rtc_base:macromagic",
+        "../../rtc_base:platform_thread",
         "../../rtc_base:random",
         "../../rtc_base:rtc_base_tests_utils",
+        "../../rtc_base:rtc_event",
         "../../rtc_base:safe_conversions",
         "../../rtc_base:stringutils",
+        "../../rtc_base:threading",
+        "../../rtc_base/synchronization:mutex",
         "../../rtc_base/system:arch",
         "../../system_wrappers",
         "../../test:audio_codec_mocks",
diff --git a/modules/audio_coding/codecs/opus/test/BUILD.gn b/modules/audio_coding/codecs/opus/test/BUILD.gn
index 37cf935be7..0b2fc17a36 100644
--- a/modules/audio_coding/codecs/opus/test/BUILD.gn
+++ b/modules/audio_coding/codecs/opus/test/BUILD.gn
@@ -46,7 +46,9 @@ if (rtc_include_tests) {
     deps = [
       ":test",
       "../../../../../common_audio",
+      "../../../../../common_audio:common_audio_c",
       "../../../../../rtc_base:checks",
+      "../../../../../rtc_base:macromagic",
       "../../../../../test:test_support",
       "//testing/gtest",
     ]
diff --git a/modules/audio_device/BUILD.gn b/modules/audio_device/BUILD.gn
index 6cd3749f63..bad43a5c91 100644
--- a/modules/audio_device/BUILD.gn
+++ b/modules/audio_device/BUILD.gn
@@ -75,6 +75,7 @@ rtc_library("audio_device_buffer") {
     "../../rtc_base:timeutils",
     "../../rtc_base/synchronization:mutex",
     "../../rtc_base/system:no_unique_address",
+    "../../system_wrappers",
     "../../system_wrappers:metrics",
   ]
 }
@@ -87,6 +88,7 @@ rtc_library("audio_device_generic") {
   deps = [
     ":audio_device_buffer",
     "../../api/audio:audio_device",
+    "../../rtc_base:logging",
   ]
 }
 
@@ -194,8 +196,11 @@ if (!build_with_chromium) {
       "../../common_audio",
       "../../rtc_base:buffer",
       "../../rtc_base:checks",
+      "../../rtc_base:logging",
       "../../rtc_base:macromagic",
+      "../../rtc_base:platform_thread",
       "../../rtc_base:random",
+      "../../rtc_base:rtc_event",
       "../../rtc_base:safe_conversions",
       "../../rtc_base:timeutils",
       "../../rtc_base/synchronization:mutex",
@@ -242,6 +247,7 @@ if (!build_with_chromium) {
       "../../rtc_base:platform_thread",
       "../../rtc_base:stringutils",
       "../../rtc_base:threading",
+      "../../rtc_base:timeutils",
       "../../rtc_base/synchronization:mutex",
       "../../rtc_base/system:file_wrapper",
       "//third_party/abseil-cpp/absl/strings:string_view",
@@ -449,6 +455,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "test_audio_device_impl_test.cc",
     ]
     deps = [
+      ":audio_device",
       ":audio_device_buffer",
       ":audio_device_generic",
       ":audio_device_impl",
@@ -461,6 +468,8 @@ if (rtc_include_tests && !build_with_chromium) {
       "../../api/audio:create_audio_device_module",
       "../../api/environment",
       "../../api/environment:environment_factory",
+      "../../api/task_queue",
+      "../../api/task_queue:default_task_queue_factory",
       "../../api/units:time_delta",
       "../../api/units:timestamp",
       "../../common_audio",
@@ -474,6 +483,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "../../rtc_base:task_queue_for_test",
       "../../rtc_base:timeutils",
       "../../rtc_base/synchronization:mutex",
+      "../../system_wrappers",
       "../../test:create_test_environment",
       "../../test:fileutils",
       "../../test:test_support",
diff --git a/modules/audio_mixer/BUILD.gn b/modules/audio_mixer/BUILD.gn
index 7dc868a242..d2fa8dbaec 100644
--- a/modules/audio_mixer/BUILD.gn
+++ b/modules/audio_mixer/BUILD.gn
@@ -46,14 +46,21 @@ rtc_library("audio_mixer_impl") {
     "../../api/audio:audio_frame_api",
     "../../api/audio:audio_mixer_api",
     "../../api/audio:audio_processing",
+    "../../api/audio:audio_processing",
+    "../../audio/utility:audio_frame_operations",
     "../../common_audio",
     "../../rtc_base:checks",
     "../../rtc_base:event_tracer",
     "../../rtc_base:logging",
     "../../rtc_base:macromagic",
+    "../../rtc_base:race_checker",
+    "../../rtc_base:refcount",
+    "../../rtc_base:safe_conversions",
     "../../rtc_base/synchronization:mutex",
+    "../../system_wrappers",
     "../../system_wrappers:metrics",
     "../audio_processing:apm_logging",
+    "../audio_processing:audio_frame_view",
     "../audio_processing/agc2:fixed_digital",
   ]
 }
@@ -88,6 +95,8 @@ if (rtc_include_tests) {
     ]
 
     deps = [
+      ":audio_frame_manipulator",
+      ":audio_mixer_impl",
       "../../api:array_view",
       "../../api/audio:audio_frame_api",
       "../../rtc_base:checks",
diff --git a/modules/audio_processing/BUILD.gn b/modules/audio_processing/BUILD.gn
index 3116c2618c..2997c457a3 100644
--- a/modules/audio_processing/BUILD.gn
+++ b/modules/audio_processing/BUILD.gn
@@ -32,6 +32,7 @@ rtc_library("audio_frame_proxies") {
     "include/audio_frame_proxies.h",
   ]
   deps = [
+    ":audio_frame_view",
     "../../api/audio:audio_frame_api",
     "../../api/audio:audio_processing",
     "../../api/audio:audio_processing_statistics",
@@ -137,6 +138,7 @@ rtc_library("gain_controller2") {
     "../../common_audio",
     "../../rtc_base:checks",
     "../../rtc_base:logging",
+    "../../rtc_base:stringutils",
     "agc2:adaptive_digital_gain_controller",
     "agc2:common",
     "agc2:cpu_features",
@@ -167,12 +169,15 @@ rtc_library("audio_processing") {
     ":apm_logging",
     ":audio_buffer",
     ":audio_frame_proxies",
+    ":audio_frame_view",
     ":gain_controller2",
     ":high_pass_filter",
     ":post_filter",
     ":rms_level",
     "../../api:array_view",
     "../../api:field_trials_view",
+    "../../api:function_view",
+    "../../api:make_ref_counted",
     "../../api:scoped_refptr",
     "../../api/audio:aec3_config",
     "../../api/audio:audio_frame_api",
@@ -182,16 +187,25 @@ rtc_library("audio_processing") {
     "../../api/audio:neural_residual_echo_estimator_api",
     "../../api/environment",
     "../../api/task_queue",
+    "../../audio/utility:audio_frame_operations",
     "../../common_audio",
+    "../../common_audio:common_audio_c",
+    "../../common_audio:fir_filter",
+    "../../common_audio:fir_filter_factory",
+    "../../common_audio/third_party/ooura:fft_size_256",
     "../../rtc_base:checks",
     "../../rtc_base:denormal_disabler",
     "../../rtc_base:event_tracer",
     "../../rtc_base:gtest_prod",
     "../../rtc_base:logging",
     "../../rtc_base:macromagic",
+    "../../rtc_base:safe_minmax",
+    "../../rtc_base:sanitizer",
     "../../rtc_base:swap_queue",
     "../../rtc_base:timeutils",
     "../../rtc_base/synchronization:mutex",
+    "../../rtc_base/system:rtc_export",
+    "../../system_wrappers",
     "../../system_wrappers:metrics",
     "aec3",
     "aec_dump",
@@ -295,6 +309,8 @@ if (rtc_include_tests) {
     sources = [ "include/mock_audio_processing.h" ]
     deps = [
       ":aec_dump_interface",
+      ":audio_buffer",
+      ":audio_processing",
       "../../api:array_view",
       "../../api:scoped_refptr",
       "../../api/audio:audio_processing",
@@ -352,6 +368,7 @@ if (rtc_include_tests) {
         "../../api:ref_count",
         "../../api:scoped_refptr",
         "../../api/audio:aec3_config",
+        "../../api/audio:aec3_factory",
         "../../api/audio:audio_frame_api",
         "../../api/audio:audio_processing",
         "../../api/audio:audio_processing_statistics",
@@ -361,8 +378,11 @@ if (rtc_include_tests) {
         "../../api/environment:environment_factory",
         "../../api/units:time_delta",
         "../../common_audio",
+        "../../common_audio:common_audio_c",
         "../../rtc_base:checks",
         "../../rtc_base:cpu_info",
+        "../../rtc_base:denormal_disabler",
+        "../../rtc_base:gtest_prod",
         "../../rtc_base:macromagic",
         "../../rtc_base:platform_thread",
         "../../rtc_base:protobuf_utils",
@@ -376,6 +396,7 @@ if (rtc_include_tests) {
         "../../rtc_base:task_queue_for_test",
         "../../rtc_base:threading",
         "../../rtc_base/synchronization:mutex",
+        "../../rtc_base/system:arch",
         "../../rtc_base/system:file_wrapper",
         "../../test:fileutils",
         "../../test:rtc_expect_death",
@@ -386,6 +407,7 @@ if (rtc_include_tests) {
         "agc:gain_control_interface",
         "agc2:adaptive_digital_gain_controller_unittest",
         "agc2:biquad_filter_unittests",
+        "agc2:cpu_features",
         "agc2:fixed_digital_unittests",
         "agc2:gain_applier_unittest",
         "agc2:input_volume_controller",
@@ -395,8 +417,10 @@ if (rtc_include_tests) {
         "agc2:saturation_protector_unittest",
         "agc2:speech_level_estimator_unittest",
         "agc2:test_utils",
+        "agc2:vad_wrapper",
         "agc2:vad_wrapper_unittests",
         "agc2/rnn_vad:unittests",
+        "capture_levels_adjuster",
         "capture_levels_adjuster:capture_levels_adjuster_unittests",
         "capture_mixer:capture_mixer_unittests",
         "test/conversational_speech:unittest",
@@ -472,8 +496,10 @@ if (rtc_include_tests) {
       "../../api/units:time_delta",
       "../../rtc_base:checks",
       "../../rtc_base:platform_thread",
+      "../../rtc_base:protobuf_utils",
       "../../rtc_base:random",
       "../../rtc_base:rtc_event",
+      "../../rtc_base:safe_conversions",
       "../../system_wrappers",
       "../../test:test_support",
       "//third_party/abseil-cpp/absl/strings:string_view",
@@ -487,6 +513,7 @@ if (rtc_include_tests) {
     ]
     deps = [
       "../../api:array_view",
+      "../../api/audio:audio_frame_api",
       "../../common_audio",
       "../../rtc_base:checks",
       "../../rtc_base:logging",
@@ -520,12 +547,14 @@ if (rtc_include_tests) {
         ":aec3_config_json",
         ":analog_mic_simulation",
         ":apm_logging",
+        ":audio_processing",
         ":audioproc_debug_proto",
         ":audioproc_protobuf_utils",
         ":audioproc_test_utils",
         "../../api:field_trials",
         "../../api:scoped_refptr",
         "../../api/audio:aec3_config",
+        "../../api/audio:aec3_factory",
         "../../api/audio:audio_processing",
         "../../api/audio:builtin_audio_processing_builder",
         "../../api/audio:echo_detector_creator",
@@ -566,6 +595,7 @@ if (rtc_include_tests) {
 
       deps = [
         ":audioproc_debug_proto",
+        "../../rtc_base:checks",
         "../../rtc_base:protobuf_utils",
         "../../rtc_base/system:arch",
       ]
@@ -608,12 +638,14 @@ rtc_library("audioproc_test_utils") {
 
   deps = [
     ":audio_buffer",
+    ":audio_processing",
     "../../api:array_view",
     "../../api/audio:audio_frame_api",
     "../../api/audio:audio_processing",
     "../../common_audio",
     "../../rtc_base:checks",
     "../../rtc_base:random",
+    "../../rtc_base/system:arch",
     "../../system_wrappers",
     "../../test:fileutils",
     "../../test:test_support",
@@ -636,6 +668,7 @@ rtc_library("aec3_config_json") {
     "../../rtc_base:logging",
     "../../rtc_base:rtc_json",
     "../../rtc_base:stringutils",
+    "../../rtc_base/system:rtc_export",
     "//third_party/abseil-cpp/absl/strings:string_view",
   ]
 }
diff --git a/modules/audio_processing/aec3/BUILD.gn b/modules/audio_processing/aec3/BUILD.gn
index f52017d935..6b5279b56a 100644
--- a/modules/audio_processing/aec3/BUILD.gn
+++ b/modules/audio_processing/aec3/BUILD.gn
@@ -148,6 +148,7 @@ rtc_library("aec3") {
     "../../../api/audio:echo_control",
     "../../../api/audio:neural_residual_echo_estimator_api",
     "../../../api/environment",
+    "../../../common_audio:common_audio_c",
     "../../../rtc_base:checks",
     "../../../rtc_base:cpu_info",
     "../../../rtc_base:gtest_prod",
@@ -184,6 +185,7 @@ rtc_source_set("aec3_fft") {
     ":fft_data",
     "../../../common_audio/third_party/ooura:fft_size_128",
     "../../../rtc_base:checks",
+    "../../../rtc_base/system:arch",
   ]
 }
 
@@ -199,6 +201,7 @@ rtc_source_set("render_buffer") {
     ":aec3_common",
     ":fft_data",
     "../../../rtc_base:checks",
+    "../../../rtc_base/system:arch",
   ]
 }
 
@@ -227,6 +230,7 @@ rtc_source_set("matched_filter") {
   sources = [ "matched_filter.h" ]
   deps = [
     ":aec3_common",
+    "../../../rtc_base:gtest_prod",
     "../../../rtc_base/system:arch",
   ]
 }
@@ -311,6 +315,7 @@ if (rtc_include_tests) {
       ":vector_math",
       "..:apm_logging",
       "..:audio_buffer",
+      "..:audio_processing",
       "..:high_pass_filter",
       "../../../api:field_trials",
       "../../../api/audio:aec3_config",
@@ -320,12 +325,14 @@ if (rtc_include_tests) {
       "../../../api/environment:environment_factory",
       "../../../rtc_base:checks",
       "../../../rtc_base:cpu_info",
+      "../../../rtc_base:gunit_helpers",
       "../../../rtc_base:random",
       "../../../rtc_base:safe_minmax",
       "../../../rtc_base:stringutils",
       "../../../rtc_base/system:arch",
       "../../../system_wrappers:metrics",
       "../../../test:create_test_field_trials",
+      "../../../test:fileutils",
       "../../../test:test_support",
       "../utility:cascaded_biquad_filter",
     ]
diff --git a/modules/audio_processing/aec_dump/BUILD.gn b/modules/audio_processing/aec_dump/BUILD.gn
index 586a38ed79..227d6dec6b 100644
--- a/modules/audio_processing/aec_dump/BUILD.gn
+++ b/modules/audio_processing/aec_dump/BUILD.gn
@@ -33,6 +33,8 @@ if (rtc_include_tests) {
     deps = [
       "..:aec_dump_interface",
       "..:audio_frame_view",
+      "..:audioproc_test_utils",
+      "../",
       "../../../api/audio:audio_frame_api",
       "../../../api/audio:audio_processing",
       "../../../test:test_support",
@@ -46,6 +48,8 @@ if (rtc_include_tests) {
 
     deps = [
       ":mock_aec_dump",
+      "..:audioproc_test_utils",
+      "../",
       "../../../api:scoped_refptr",
       "../../../api/audio:audio_processing",
       "../../../api/audio:builtin_audio_processing_builder",
@@ -94,7 +98,10 @@ if (rtc_enable_protobuf) {
       defines = []
       deps = [
         ":aec_dump",
+        ":aec_dump_impl",
         "..:aec_dump_interface",
+        "..:audioproc_debug_proto",
+        "../",
         "../../../api/audio:audio_processing",
         "../../../rtc_base:task_queue_for_test",
         "../../../test:fileutils",
diff --git a/modules/audio_processing/agc/BUILD.gn b/modules/audio_processing/agc/BUILD.gn
index f5bad6f189..090fbd4604 100644
--- a/modules/audio_processing/agc/BUILD.gn
+++ b/modules/audio_processing/agc/BUILD.gn
@@ -29,6 +29,7 @@ rtc_library("agc") {
     "../../../api/audio:audio_processing",
     "../../../api/environment",
     "../../../common_audio",
+    "../../../common_audio:common_audio_c",
     "../../../rtc_base:checks",
     "../../../rtc_base:gtest_prod",
     "../../../rtc_base:logging",
@@ -37,6 +38,7 @@ rtc_library("agc") {
     "../agc2:clipping_predictor",
     "../agc2:gain_map",
     "../agc2:input_volume_stats_reporter",
+    "../vad",
   ]
 }
 
@@ -73,9 +75,12 @@ rtc_library("legacy_agc") {
   ]
 
   deps = [
+    "../../../common_audio",
     "../../../common_audio:common_audio_c",
     "../../../common_audio:common_audio_cc",
+    "../../../common_audio/third_party/ooura:fft_size_256",
     "../../../rtc_base:checks",
+    "../../../system_wrappers",
   ]
 
   if (rtc_build_with_neon) {
@@ -102,13 +107,18 @@ if (rtc_include_tests) {
       ":gain_control_interface",
       ":level_estimation",
       "..:audio_buffer",
+      "..:mocks",
       "../../../api:array_view",
+      "../../../api:field_trials",
       "../../../api/audio:audio_processing",
       "../../../api/environment",
       "../../../api/environment:environment_factory",
       "../../../rtc_base:checks",
+      "../../../rtc_base:random",
+      "../../../rtc_base:safe_conversions",
       "../../../rtc_base:safe_minmax",
       "../../../rtc_base:stringutils",
+      "../../../system_wrappers:metrics",
       "../../../test:create_test_field_trials",
       "../../../test:fileutils",
       "../../../test:test_support",
diff --git a/modules/audio_processing/agc2/BUILD.gn b/modules/audio_processing/agc2/BUILD.gn
index 4a6b6eaaa9..0de1c32bd7 100644
--- a/modules/audio_processing/agc2/BUILD.gn
+++ b/modules/audio_processing/agc2/BUILD.gn
@@ -28,6 +28,7 @@ rtc_library("speech_level_estimator") {
   deps = [
     ":common",
     "..:apm_logging",
+    "../../../api:array_view",
     "../../../api:field_trials_view",
     "../../../api/audio:audio_processing",
     "../../../rtc_base:checks",
@@ -148,8 +149,10 @@ rtc_library("fixed_digital") {
   deps = [
     ":common",
     "..:apm_logging",
+    "..:audio_frame_view",
     "../../../api:array_view",
     "../../../api/audio:audio_frame_api",
+    "../../../common_audio",
     "../../../rtc_base:checks",
     "../../../rtc_base:gtest_prod",
     "../../../rtc_base:safe_conversions",
@@ -213,6 +216,7 @@ rtc_library("input_volume_controller") {
     ":input_volume_stats_reporter",
     "..:audio_buffer",
     "..:audio_frame_view",
+    "../../../api:array_view",
     "../../../api:field_trials_view",
     "../../../api/audio:audio_processing",
     "../../../rtc_base:checks",
@@ -229,9 +233,11 @@ rtc_library("noise_level_estimator") {
     "noise_level_estimator.h",
   ]
   deps = [
+    ":biquad_filter",
     "..:apm_logging",
     "../../../api/audio:audio_frame_api",
     "../../../rtc_base:checks",
+    "../../../system_wrappers",
   ]
 
   visibility = [
@@ -304,6 +310,7 @@ rtc_library("speech_level_estimator_unittest") {
     "..:apm_logging",
     "../../../api/audio:audio_processing",
     "../../../rtc_base:checks",
+    "../../../rtc_base:gunit_helpers",
     "../../../test:test_support",
   ]
 }
@@ -319,8 +326,10 @@ rtc_library("adaptive_digital_gain_controller_unittest") {
     ":common",
     ":test_utils",
     "..:apm_logging",
+    "..:audio_frame_view",
     "../../../api/audio:audio_processing",
     "../../../common_audio",
+    "../../../rtc_base:gunit_helpers",
     "../../../test:test_support",
   ]
 }
@@ -334,6 +343,7 @@ rtc_library("gain_applier_unittest") {
     ":gain_applier",
     ":test_utils",
     "../../../api/audio:audio_frame_api",
+    "../../../rtc_base:gunit_helpers",
     "../../../test:test_support",
   ]
 }
@@ -350,6 +360,7 @@ rtc_library("saturation_protector_unittest") {
     ":common",
     ":saturation_protector",
     "..:apm_logging",
+    "../../../rtc_base:gunit_helpers",
     "../../../test:test_support",
   ]
 }
@@ -360,6 +371,7 @@ rtc_library("biquad_filter_unittests") {
   deps = [
     ":biquad_filter",
     "../../../api:array_view",
+    "../../../rtc_base:gunit_helpers",
     "../../../test:test_support",
   ]
 }
@@ -384,9 +396,13 @@ rtc_library("fixed_digital_unittests") {
     ":fixed_digital",
     ":test_utils",
     "..:apm_logging",
+    "..:audio_frame_view",
+    "../../../api:array_view",
     "../../../api/audio:audio_frame_api",
     "../../../common_audio",
     "../../../rtc_base:checks",
+    "../../../rtc_base:gunit_helpers",
+    "../../../system_wrappers:metrics",
     "../../../test:test_support",
   ]
 }
@@ -404,13 +420,18 @@ rtc_library("input_volume_controller_unittests") {
 
   deps = [
     ":clipping_predictor",
+    ":gain_map",
     ":input_volume_controller",
     "..:audio_buffer",
     "..:audio_frame_view",
+    "../../../api:array_view",
     "../../../api/audio:audio_processing",
     "../../../api/environment:environment_factory",
     "../../../rtc_base:checks",
+    "../../../rtc_base:random",
+    "../../../rtc_base:safe_conversions",
     "../../../rtc_base:safe_minmax",
+    "../../../rtc_base:stringutils",
     "../../../system_wrappers:metrics",
     "../../../test:fileutils",
     "../../../test:test_support",
@@ -430,6 +451,7 @@ rtc_library("noise_estimator_unittests") {
     "../../../api:function_view",
     "../../../api/audio:audio_frame_api",
     "../../../rtc_base:checks",
+    "../../../rtc_base:gunit_helpers",
     "../../../test:test_support",
   ]
 }
@@ -443,6 +465,7 @@ rtc_library("vad_wrapper_unittests") {
     "../../../api:array_view",
     "../../../api/audio:audio_frame_api",
     "../../../rtc_base:checks",
+    "../../../rtc_base:gunit_helpers",
     "../../../rtc_base:safe_compare",
     "../../../test:test_support",
   ]
@@ -477,6 +500,7 @@ rtc_library("input_volume_stats_reporter") {
     "../../../rtc_base:checks",
     "../../../rtc_base:gtest_prod",
     "../../../rtc_base:logging",
+    "../../../rtc_base:safe_minmax",
     "../../../rtc_base:stringutils",
     "../../../system_wrappers:metrics",
     "//third_party/abseil-cpp/absl/strings:string_view",
diff --git a/modules/audio_processing/agc2/rnn_vad/BUILD.gn b/modules/audio_processing/agc2/rnn_vad/BUILD.gn
index 07aa16ae75..075e2a8110 100644
--- a/modules/audio_processing/agc2/rnn_vad/BUILD.gn
+++ b/modules/audio_processing/agc2/rnn_vad/BUILD.gn
@@ -34,6 +34,8 @@ rtc_library("rnn_vad") {
     "..:cpu_features",
     "../../../../api:array_view",
     "../../../../rtc_base:checks",
+    "../../../../rtc_base:safe_compare",
+    "../../../../rtc_base:safe_conversions",
     "//third_party/rnnoise:rnn_vad",
   ]
 }
@@ -317,10 +319,12 @@ if (rtc_include_tests) {
         ":rnn_vad",
         ":rnn_vad_common",
         "..:cpu_features",
+        "../../../../api:array_view",
         "../../../../common_audio",
         "../../../../rtc_base:checks",
         "../../../../rtc_base:logging",
         "../../../../rtc_base:safe_compare",
+        "../../../../test:test_support",
         "//third_party/abseil-cpp/absl/flags:flag",
         "//third_party/abseil-cpp/absl/flags:parse",
       ]
diff --git a/modules/audio_processing/capture_levels_adjuster/BUILD.gn b/modules/audio_processing/capture_levels_adjuster/BUILD.gn
index ebd8bc3a68..20b2e395a8 100644
--- a/modules/audio_processing/capture_levels_adjuster/BUILD.gn
+++ b/modules/audio_processing/capture_levels_adjuster/BUILD.gn
@@ -39,6 +39,8 @@ rtc_library("capture_levels_adjuster_unittests") {
     ":capture_levels_adjuster",
     "..:audio_buffer",
     "..:audioproc_test_utils",
+    "../../../rtc_base:gunit_helpers",
+    "../../../rtc_base:stringutils",
     "../../../test:test_support",
   ]
 }
diff --git a/modules/audio_processing/capture_mixer/BUILD.gn b/modules/audio_processing/capture_mixer/BUILD.gn
index 2b9a8c4b73..ec4653d358 100644
--- a/modules/audio_processing/capture_mixer/BUILD.gn
+++ b/modules/audio_processing/capture_mixer/BUILD.gn
@@ -50,7 +50,10 @@ rtc_library("capture_mixer_unittests") {
   ]
   deps = [
     ":capture_mixer",
+    "..:audioproc_test_utils",
     "../../../api:array_view",
+    "../../../rtc_base:gunit_helpers",
+    "../../../rtc_base:stringutils",
     "../../../test:test_support",
   ]
 }
diff --git a/modules/audio_processing/ns/BUILD.gn b/modules/audio_processing/ns/BUILD.gn
index 1b5ebafff1..e90ef0e06a 100644
--- a/modules/audio_processing/ns/BUILD.gn
+++ b/modules/audio_processing/ns/BUILD.gn
@@ -49,10 +49,19 @@ rtc_static_library("ns") {
   }
 
   deps = [
+    "..:apm_logging",
     "..:audio_buffer",
+    "..:high_pass_filter",
     "../../../api:array_view",
+    "../../../common_audio:common_audio_c",
+    "../../../common_audio/third_party/ooura:fft_size_128",
     "../../../common_audio/third_party/ooura:fft_size_256",
     "../../../rtc_base:checks",
+    "../../../rtc_base:safe_minmax",
+    "../../../rtc_base/system:arch",
+    "../../../system_wrappers",
+    "../../../system_wrappers:metrics",
+    "../utility:cascaded_biquad_filter",
   ]
 }
 
@@ -65,9 +74,18 @@ if (rtc_include_tests) {
 
     deps = [
       ":ns",
+      "..:apm_logging",
       "..:audio_buffer",
+      "..:audio_processing",
+      "..:high_pass_filter",
+      "../../../api:array_view",
+      "../../../rtc_base:checks",
+      "../../../rtc_base:safe_minmax",
       "../../../rtc_base:stringutils",
+      "../../../rtc_base/system:arch",
+      "../../../system_wrappers",
       "../../../test:test_support",
+      "../utility:cascaded_biquad_filter",
     ]
 
     defines = []
diff --git a/modules/audio_processing/test/conversational_speech/BUILD.gn b/modules/audio_processing/test/conversational_speech/BUILD.gn
index 10fd9dda33..28f27daae4 100644
--- a/modules/audio_processing/test/conversational_speech/BUILD.gn
+++ b/modules/audio_processing/test/conversational_speech/BUILD.gn
@@ -21,6 +21,7 @@ if (!build_with_chromium) {
       ":lib",
       "../../../../rtc_base:checks",
       "../../../../test:fileutils",
+      "../../../../test:test_support",
       "//third_party/abseil-cpp/absl/flags:flag",
       "//third_party/abseil-cpp/absl/flags:parse",
     ]
diff --git a/modules/audio_processing/utility/BUILD.gn b/modules/audio_processing/utility/BUILD.gn
index eaf22ec0c7..63ee6d066e 100644
--- a/modules/audio_processing/utility/BUILD.gn
+++ b/modules/audio_processing/utility/BUILD.gn
@@ -50,6 +50,7 @@ if (rtc_include_tests) {
     sources = [ "cascaded_biquad_filter_unittest.cc" ]
     deps = [
       ":cascaded_biquad_filter",
+      "../../../rtc_base:checks",
       "../../../test:test_support",
       "//api:array_view",
       "//rtc_base:checks",
diff --git a/modules/audio_processing/vad/BUILD.gn b/modules/audio_processing/vad/BUILD.gn
index 34b3003e6d..71e079d3a3 100644
--- a/modules/audio_processing/vad/BUILD.gn
+++ b/modules/audio_processing/vad/BUILD.gn
@@ -35,6 +35,7 @@ rtc_library("vad") {
     "voice_gmm_tables.h",
   ]
   deps = [
+    "../../../audio/utility:audio_frame_operations",
     "../../../common_audio",
     "../../../common_audio:common_audio_c",
     "../../../common_audio/third_party/ooura:fft_size_256",
@@ -58,6 +59,7 @@ if (rtc_include_tests) {
     ]
     deps = [
       ":vad",
+      "../../../common_audio",
       "../../../test:fileutils",
       "../../../test:test_support",
       "//testing/gmock",

From bb96cbad593ae84c8bea85d8295d57756e5087e0 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Thu, 26 Mar 2026 15:28:47 +0100
Subject: [PATCH 416/472] Spring cleanup of WATCHLIST and OWNER files

Bug: none
Change-Id: I168e1f456905c96e57066721543ed9cd8925e4b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/460420
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47269}
---
 WATCHLISTS                                    | 41 ++++++-------------
 api/audio_codecs/OWNERS                       |  1 -
 audio/OWNERS                                  |  1 -
 modules/audio_mixer/OWNERS                    |  1 -
 modules/audio_processing/OWNERS               |  2 -
 .../test/conversational_speech/OWNERS         |  1 -
 rtc_base/OWNERS                               |  1 -
 rtc_tools/unpack_aecdump/OWNERS               |  1 -
 system_wrappers/OWNERS                        |  1 -
 9 files changed, 13 insertions(+), 37 deletions(-)

diff --git a/WATCHLISTS b/WATCHLISTS
index 3c35dd4be7..dbb0f3c02b 100644
--- a/WATCHLISTS
+++ b/WATCHLISTS
@@ -103,26 +103,20 @@
     'this_file': [],
     'all_webrtc': [],
     'root_files': ['peah@webrtc.org',
-                   'qiang.lu@intel.com',
-                   'yujie.mao@webrtc.org'],
+                   'qiang.lu@intel.com'],
     'build_files': ['mbonadei@webrtc.org'],
-    'common_audio': ['alessiob@webrtc.org',
-                     'audio-team@agora.io',
+    'common_audio': ['audio-team@agora.io',
                      'peah@webrtc.org',
                      'saza@webrtc.org'],
     'audio': ['peah@webrtc.org'],
     'api': ['hta@webrtc.org',
             'peah@webrtc.org'],
     'base': ['hta@webrtc.org'],
-    'call': ['mflodman@webrtc.org',
-             'stefan@webrtc.org'],
-    'video': ['mflodman@webrtc.org',
-              'stefan@webrtc.org',
+    'call': ['stefan@webrtc.org'],
+    'video': ['stefan@webrtc.org',
               'video-team@agora.io',
-              'yujie.mao@webrtc.org',
               'zhengzhonghou@agora.io'],
-    'video_capture': ['mflodman@webrtc.org',
-                      'perkj@webrtc.org',
+    'video_capture': ['perkj@webrtc.org',
                       'sdk-team@agora.io',
                       'zhengzhonghou@agora.io'],
     'audio_device': ['audio-team@agora.io',
@@ -130,45 +124,36 @@
                      'peah@webrtc.org',
                      'saza@webrtc.org',
                      'sdk-team@agora.io'],
-    'audio_coding': ['alessiob@webrtc.org',
-                     'audio-team@agora.io',
+    'audio_coding': ['audio-team@agora.io',
                      'henrik.lundin@webrtc.org',
                      'peah@webrtc.org',
                      'saza@webrtc.org'],
-    'neteq': ['alessiob@webrtc.org',
-              'audio-team@agora.io',
+    'neteq': ['audio-team@agora.io',
               'henrik.lundin@webrtc.org',
               'saza@webrtc.org'],
     'audio_mixer': ['aleloi@webrtc.org',
                     'henrik.lundin@webrtc.org',
                     'peah@webrtc.org',
                     'saza@webrtc.org'],
-    'audio_processing': ['alessiob@webrtc.org',
-                         'audio-team@agora.io',
+    'audio_processing': ['audio-team@agora.io',
                          'henrik.lundin@webrtc.org',
                          'peah@webrtc.org',
                          'saza@webrtc.org'],
-    'video_coding': ['mflodman@webrtc.org',
-                     'stefan@webrtc.org',
+    'video_coding': ['stefan@webrtc.org',
                      'video-team@agora.io',
                      'zhengzhonghou@agora.io'],
-    'bitrate_controller': ['mflodman@webrtc.org',
-                           'stefan@webrtc.org',
+    'bitrate_controller': ['stefan@webrtc.org',
                            'zhuangzesen@agora.io'],
     'congestion_controller': [],
-    'remote_bitrate_estimator': ['mflodman@webrtc.org',
-                                 'stefan@webrtc.org',
+    'remote_bitrate_estimator': ['stefan@webrtc.org',
                                  'zhuangzesen@agora.io'],
-    'pacing': ['mflodman@webrtc.org',
-               'stefan@webrtc.org',
+    'pacing': ['stefan@webrtc.org',
                'zhuangzesen@agora.io'],
-    'rtp_rtcp': ['mflodman@webrtc.org',
-                 'stefan@webrtc.org',
+    'rtp_rtcp': ['stefan@webrtc.org',
                  'danilchap@webrtc.org',
                  'zhuangzesen@agora.io'],
     'system_wrappers': ['fengyue@agora.io',
                         'henrika@webrtc.org',
-                        'mflodman@webrtc.org',
                         'peah@webrtc.org',
                         'zhengzhonghou@agora.io'],
     'pc': ['steveanton+watch@webrtc.org'],
diff --git a/api/audio_codecs/OWNERS b/api/audio_codecs/OWNERS
index 77b414abc3..f18110a4b6 100644
--- a/api/audio_codecs/OWNERS
+++ b/api/audio_codecs/OWNERS
@@ -1,3 +1,2 @@
-alessiob@webrtc.org
 henrik.lundin@webrtc.org
 jakobi@webrtc.org
diff --git a/audio/OWNERS b/audio/OWNERS
index e629bc1815..547018deb7 100644
--- a/audio/OWNERS
+++ b/audio/OWNERS
@@ -1,4 +1,3 @@
-alessiob@webrtc.org
 gustaf@webrtc.org
 henrik.lundin@webrtc.org
 jakobi@webrtc.org
diff --git a/modules/audio_mixer/OWNERS b/modules/audio_mixer/OWNERS
index 5edc304ab3..6e7454380e 100644
--- a/modules/audio_mixer/OWNERS
+++ b/modules/audio_mixer/OWNERS
@@ -1,2 +1 @@
-alessiob@webrtc.org
 henrik.lundin@webrtc.org
diff --git a/modules/audio_processing/OWNERS b/modules/audio_processing/OWNERS
index f5dc59ea35..cc5ef82097 100644
--- a/modules/audio_processing/OWNERS
+++ b/modules/audio_processing/OWNERS
@@ -1,8 +1,6 @@
-alessiob@webrtc.org
 devicentepena@webrtc.org
 gustaf@webrtc.org
 henrik.lundin@webrtc.org
-ivoc@webrtc.org
 lionelk@webrtc.org
 peah@webrtc.org
 saza@webrtc.org
diff --git a/modules/audio_processing/test/conversational_speech/OWNERS b/modules/audio_processing/test/conversational_speech/OWNERS
index 07cff405e6..18a48034b1 100644
--- a/modules/audio_processing/test/conversational_speech/OWNERS
+++ b/modules/audio_processing/test/conversational_speech/OWNERS
@@ -1,3 +1,2 @@
-alessiob@webrtc.org
 henrik.lundin@webrtc.org
 peah@webrtc.org
diff --git a/rtc_base/OWNERS b/rtc_base/OWNERS
index 680815b65f..251a43426b 100644
--- a/rtc_base/OWNERS
+++ b/rtc_base/OWNERS
@@ -1,5 +1,4 @@
 hta@webrtc.org
-mflodman@webrtc.org
 tommi@webrtc.org
 mbonadei@webrtc.org
 jonaso@webrtc.org
diff --git a/rtc_tools/unpack_aecdump/OWNERS b/rtc_tools/unpack_aecdump/OWNERS
index 3ef412054e..45953af31e 100644
--- a/rtc_tools/unpack_aecdump/OWNERS
+++ b/rtc_tools/unpack_aecdump/OWNERS
@@ -1,3 +1,2 @@
-alessiob@webrtc.org
 peah@webrtc.org
 saza@webrtc.org
diff --git a/system_wrappers/OWNERS b/system_wrappers/OWNERS
index f7bd06a0c3..95662c195c 100644
--- a/system_wrappers/OWNERS
+++ b/system_wrappers/OWNERS
@@ -1,2 +1 @@
 henrika@webrtc.org
-mflodman@webrtc.org

From 1f9fdd2bc21c397c0ed4a389b72f00e3f56e59a4 Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Thu, 26 Mar 2026 16:48:09 +0100
Subject: [PATCH 417/472] Replace ArrayView with std::span in
 modules/audio_processing

ArrayView is an alias to std::span. This change switch to use
std::span directly instead of through the alias.

Search&Replace MakeArrayView and ArrayView with std::span
Search&Replace include "api/array_view.h" with include <span>
Remove <span> include where std::span is not mentioned in the file
Remove build dependencies on array_view target

Replace ExpectArrayViewEquality helper with ElementsAreArray matcher
in pffft_wrapper_unittest.cc

Bug: webrtc:439801349
Change-Id: I9503bc9003ee6a7e349459989b03f59a7de37ea6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/460320
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47270}
---
 modules/audio_processing/BUILD.gn             | 16 +---
 modules/audio_processing/agc/BUILD.gn         |  3 -
 modules/audio_processing/agc/agc.cc           |  4 +-
 modules/audio_processing/agc/agc.h            |  4 +-
 .../agc/agc_manager_direct.cc                 |  4 +-
 .../audio_processing/agc/agc_manager_direct.h |  4 +-
 modules/audio_processing/agc/mock_agc.h       |  4 +-
 modules/audio_processing/agc2/BUILD.gn        | 12 +--
 .../audio_processing/agc2/biquad_filter.cc    |  4 +-
 modules/audio_processing/agc2/biquad_filter.h |  4 +-
 .../agc2/biquad_filter_unittest.cc            |  9 +-
 .../agc2/fixed_digital_level_estimator.cc     |  1 -
 modules/audio_processing/agc2/limiter.cc      |  4 +-
 .../audio_processing/agc2/rnn_vad/BUILD.gn    | 21 +----
 .../agc2/rnn_vad/auto_correlation.cc          |  6 +-
 .../agc2/rnn_vad/auto_correlation.h           |  6 +-
 .../agc2/rnn_vad/features_extraction.cc       |  6 +-
 .../agc2/rnn_vad/features_extraction.h        | 12 +--
 .../rnn_vad/features_extraction_unittest.cc   | 10 +-
 .../agc2/rnn_vad/lp_residual.cc               | 22 ++---
 .../agc2/rnn_vad/lp_residual.h                | 12 +--
 .../agc2/rnn_vad/pitch_search.cc              | 10 +-
 .../agc2/rnn_vad/pitch_search.h               |  4 +-
 .../agc2/rnn_vad/pitch_search_internal.cc     | 38 ++++----
 .../agc2/rnn_vad/pitch_search_internal.h      | 22 ++---
 .../rnn_vad/pitch_search_internal_unittest.cc |  8 +-
 .../agc2/rnn_vad/ring_buffer.h                |  8 +-
 .../agc2/rnn_vad/ring_buffer_unittest.cc      |  4 +-
 modules/audio_processing/agc2/rnn_vad/rnn.cc  |  5 +-
 modules/audio_processing/agc2/rnn_vad/rnn.h   |  4 +-
 .../audio_processing/agc2/rnn_vad/rnn_fc.cc   | 14 +--
 .../audio_processing/agc2/rnn_vad/rnn_fc.h    | 12 +--
 .../agc2/rnn_vad/rnn_fc_unittest.cc           |  1 -
 .../audio_processing/agc2/rnn_vad/rnn_gru.cc  | 46 +++++-----
 .../audio_processing/agc2/rnn_vad/rnn_gru.h   | 14 +--
 .../agc2/rnn_vad/rnn_gru_unittest.cc          |  8 +-
 .../agc2/rnn_vad/rnn_unittest.cc              |  1 -
 .../agc2/rnn_vad/rnn_vad_unittest.cc          |  4 +-
 .../agc2/rnn_vad/sequence_buffer.h            | 12 +--
 .../agc2/rnn_vad/spectral_features.cc         | 28 +++---
 .../agc2/rnn_vad/spectral_features.h          | 24 ++---
 .../rnn_vad/spectral_features_internal.cc     | 22 ++---
 .../agc2/rnn_vad/spectral_features_internal.h | 22 ++---
 .../spectral_features_internal_unittest.cc    |  4 +-
 .../rnn_vad/spectral_features_unittest.cc     | 30 +++---
 .../agc2/rnn_vad/symmetric_matrix_buffer.h    |  4 +-
 .../symmetric_matrix_buffer_unittest.cc       |  4 +-
 .../agc2/rnn_vad/test_utils.cc                | 12 +--
 .../agc2/rnn_vad/test_utils.h                 | 24 ++---
 .../agc2/rnn_vad/vector_math.h                |  8 +-
 .../agc2/rnn_vad/vector_math_avx2.cc          |  7 +-
 .../agc2/vad_wrapper_unittest.cc              |  6 +-
 .../audio_processing/audio_processing_impl.cc | 16 ++--
 .../audio_processing/audio_processing_impl.h  |  4 +-
 .../audio_processing_impl_locking_unittest.cc |  6 +-
 .../audio_processing_impl_unittest.cc         |  8 +-
 .../audio_processing_unittest.cc              | 26 +++---
 .../capture_levels_adjuster/BUILD.gn          |  1 -
 .../audio_samples_scaler.cc                   | 10 +-
 .../audio_processing/capture_mixer/BUILD.gn   |  6 +-
 .../capture_mixer/audio_content_analyzer.cc   |  9 +-
 .../capture_mixer/audio_content_analyzer.h    | 13 +--
 .../audio_content_analyzer_unittest.cc        |  1 -
 .../capture_mixer/capture_mixer.cc            | 12 +--
 .../capture_mixer/capture_mixer.h             |  7 +-
 .../capture_mixer/channel_content_remixer.cc  | 56 +++++------
 .../capture_mixer/channel_content_remixer.h   | 57 ++++++------
 .../channel_content_remixer_unittest.cc       |  6 +-
 .../capture_mixer/dc_levels_estimator.cc      |  9 +-
 .../capture_mixer/dc_levels_estimator.h       |  7 +-
 .../dc_levels_estimator_unittest.cc           |  8 +-
 .../capture_mixer/energy_estimator.cc         | 10 +-
 .../capture_mixer/energy_estimator.h          | 11 +--
 .../energy_estimator_unittest.cc              |  4 +-
 .../capture_mixer/remixing_logic.cc           | 30 +++---
 .../capture_mixer/remixing_logic.h            | 19 ++--
 .../capture_mixer/saturation_estimator.cc     | 11 +--
 .../capture_mixer/saturation_estimator.h      | 13 ++-
 .../saturation_estimator_unittest.cc          | 16 ++--
 modules/audio_processing/gain_control_impl.cc |  6 +-
 modules/audio_processing/gain_control_impl.h  |  4 +-
 .../audio_processing/gain_control_unittest.cc |  4 +-
 modules/audio_processing/high_pass_filter.cc  | 18 ++--
 .../high_pass_filter_unittest.cc              | 12 +--
 .../include/mock_audio_processing.h           |  8 +-
 .../logging/apm_data_dumper.h                 | 31 ++++---
 modules/audio_processing/ns/BUILD.gn          |  2 -
 modules/audio_processing/ns/fast_math.cc      |  8 +-
 modules/audio_processing/ns/fast_math.h       |  8 +-
 modules/audio_processing/ns/histograms.h      |  8 +-
 .../audio_processing/ns/noise_estimator.cc    |  8 +-
 modules/audio_processing/ns/noise_estimator.h | 16 ++--
 .../audio_processing/ns/noise_suppressor.cc   | 92 +++++++++----------
 .../audio_processing/ns/noise_suppressor.h    |  4 +-
 modules/audio_processing/ns/ns_fft.cc         | 14 +--
 modules/audio_processing/ns/ns_fft.h          | 14 +--
 .../ns/prior_signal_model_estimator.cc        |  6 +-
 .../ns/quantile_noise_estimator.cc            |  8 +-
 .../ns/quantile_noise_estimator.h             |  6 +-
 .../ns/signal_model_estimator.cc              | 22 ++---
 .../ns/signal_model_estimator.h               | 10 +-
 .../ns/speech_probability_estimator.cc        | 10 +-
 .../ns/speech_probability_estimator.h         | 12 +--
 modules/audio_processing/ns/wiener_filter.cc  | 10 +-
 modules/audio_processing/ns/wiener_filter.h   | 12 +--
 modules/audio_processing/post_filter.cc       |  8 +-
 modules/audio_processing/post_filter.h        |  4 +-
 .../audio_processing/post_filter_unittest.cc  |  4 +-
 .../residual_echo_detector.cc                 |  8 +-
 .../audio_processing/residual_echo_detector.h |  6 +-
 modules/audio_processing/rms_level.cc         |  6 +-
 modules/audio_processing/rms_level.h          |  7 +-
 .../audio_processing/rms_level_unittest.cc    |  6 +-
 modules/audio_processing/splitting_filter.cc  | 10 +-
 .../test/audio_buffer_tools.cc                |  4 +-
 .../test/audio_buffer_tools.h                 |  4 +-
 .../test/bitexactness_tools.cc                | 14 +--
 .../test/bitexactness_tools.h                 | 12 +--
 .../test/conversational_speech/BUILD.gn       |  2 -
 .../conversational_speech/mock_wavreader.h    |  9 +-
 .../conversational_speech/multiend_call.cc    |  4 +-
 .../conversational_speech/multiend_call.h     |  6 +-
 .../test/conversational_speech/simulator.cc   |  8 +-
 .../test/conversational_speech/timing.cc      |  4 +-
 .../test/conversational_speech/timing.h       |  4 +-
 .../wavreader_factory.cc                      |  6 +-
 .../wavreader_interface.h                     |  7 +-
 .../test/echo_canceller_test_tools.cc         |  8 +-
 .../test/echo_canceller_test_tools.h          |  8 +-
 .../echo_canceller_test_tools_unittest.cc     | 10 +-
 .../test/fake_recording_device.cc             | 12 +--
 .../test/fake_recording_device.h              |  4 +-
 .../three_band_filter_bank.cc                 | 30 +++---
 .../audio_processing/three_band_filter_bank.h | 11 +--
 modules/audio_processing/utility/BUILD.gn     |  8 +-
 .../utility/cascaded_biquad_filter.cc         | 14 +--
 .../utility/cascaded_biquad_filter.h          | 13 ++-
 .../cascaded_biquad_filter_unittest.cc        | 14 +--
 .../audio_processing/utility/pffft_wrapper.cc |  6 +-
 .../audio_processing/utility/pffft_wrapper.h  |  9 +-
 .../utility/pffft_wrapper_unittest.cc         | 22 ++---
 .../vad/voice_activity_detector.h             |  2 +-
 142 files changed, 772 insertions(+), 849 deletions(-)

diff --git a/modules/audio_processing/BUILD.gn b/modules/audio_processing/BUILD.gn
index 2997c457a3..e169aa76a5 100644
--- a/modules/audio_processing/BUILD.gn
+++ b/modules/audio_processing/BUILD.gn
@@ -57,7 +57,6 @@ rtc_library("audio_buffer") {
   defines = []
 
   deps = [
-    "../../api:array_view",
     "../../api/audio:audio_frame_api",
     "../../api/audio:audio_processing",
     "../../common_audio",
@@ -80,7 +79,6 @@ rtc_library("high_pass_filter") {
 
   deps = [
     ":audio_buffer",
-    "../../api:array_view",
     "../../rtc_base:checks",
     "utility:cascaded_biquad_filter",
   ]
@@ -98,7 +96,6 @@ rtc_library("post_filter") {
 
   deps = [
     ":audio_buffer",
-    "../../api:array_view",
     "../../rtc_base:checks",
     "utility:cascaded_biquad_filter",
   ]
@@ -174,7 +171,6 @@ rtc_library("audio_processing") {
     ":high_pass_filter",
     ":post_filter",
     ":rms_level",
-    "../../api:array_view",
     "../../api:field_trials_view",
     "../../api:function_view",
     "../../api:make_ref_counted",
@@ -245,7 +241,6 @@ rtc_library("residual_echo_detector") {
   ]
   deps = [
     ":apm_logging",
-    "../../api:array_view",
     "../../api/audio:audio_processing",
     "../../rtc_base:checks",
     "../../rtc_base:logging",
@@ -259,10 +254,7 @@ rtc_library("rms_level") {
     "rms_level.cc",
     "rms_level.h",
   ]
-  deps = [
-    "../../api:array_view",
-    "../../rtc_base:checks",
-  ]
+  deps = [ "../../rtc_base:checks" ]
 }
 
 rtc_library("audio_processing_statistics") {
@@ -294,7 +286,6 @@ rtc_library("apm_logging") {
     "logging/apm_data_dumper.h",
   ]
   deps = [
-    "../../api:array_view",
     "../../common_audio",
     "../../rtc_base:checks",
     "../../rtc_base:stringutils",
@@ -311,7 +302,6 @@ if (rtc_include_tests) {
       ":aec_dump_interface",
       ":audio_buffer",
       ":audio_processing",
-      "../../api:array_view",
       "../../api:scoped_refptr",
       "../../api/audio:audio_processing",
       "../../api/audio:audio_processing_statistics",
@@ -363,7 +353,6 @@ if (rtc_include_tests) {
         ":high_pass_filter",
         ":mocks",
         ":post_filter",
-        "../../api:array_view",
         "../../api:make_ref_counted",
         "../../api:ref_count",
         "../../api:scoped_refptr",
@@ -485,7 +474,6 @@ if (rtc_include_tests) {
     deps = [
       ":audio_processing",
       ":audioproc_test_utils",
-      "../../api:array_view",
       "../../api:scoped_refptr",
       "../../api/audio:audio_processing",
       "../../api/audio:builtin_audio_processing_builder",
@@ -512,7 +500,6 @@ if (rtc_include_tests) {
       "test/fake_recording_device.h",
     ]
     deps = [
-      "../../api:array_view",
       "../../api/audio:audio_frame_api",
       "../../common_audio",
       "../../rtc_base:checks",
@@ -639,7 +626,6 @@ rtc_library("audioproc_test_utils") {
   deps = [
     ":audio_buffer",
     ":audio_processing",
-    "../../api:array_view",
     "../../api/audio:audio_frame_api",
     "../../api/audio:audio_processing",
     "../../common_audio",
diff --git a/modules/audio_processing/agc/BUILD.gn b/modules/audio_processing/agc/BUILD.gn
index 090fbd4604..d2a4a5f05c 100644
--- a/modules/audio_processing/agc/BUILD.gn
+++ b/modules/audio_processing/agc/BUILD.gn
@@ -24,7 +24,6 @@ rtc_library("agc") {
     "..:apm_logging",
     "..:audio_buffer",
     "..:audio_frame_view",
-    "../../../api:array_view",
     "../../../api:field_trials_view",
     "../../../api/audio:audio_processing",
     "../../../api/environment",
@@ -52,7 +51,6 @@ rtc_library("level_estimation") {
     "utility.h",
   ]
   deps = [
-    "../../../api:array_view",
     "../../../rtc_base:checks",
     "../vad",
   ]
@@ -108,7 +106,6 @@ if (rtc_include_tests) {
       ":level_estimation",
       "..:audio_buffer",
       "..:mocks",
-      "../../../api:array_view",
       "../../../api:field_trials",
       "../../../api/audio:audio_processing",
       "../../../api/environment",
diff --git a/modules/audio_processing/agc/agc.cc b/modules/audio_processing/agc/agc.cc
index 29e84b4380..fb5a9ed1cb 100644
--- a/modules/audio_processing/agc/agc.cc
+++ b/modules/audio_processing/agc/agc.cc
@@ -13,9 +13,9 @@
 #include <cmath>
 #include <cstdint>
 #include <cstdlib>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/agc/loudness_histogram.h"
 #include "modules/audio_processing/agc/utility.h"
 #include "rtc_base/checks.h"
@@ -39,7 +39,7 @@ Agc::Agc()
 
 Agc::~Agc() = default;
 
-void Agc::Process(ArrayView<const int16_t> audio) {
+void Agc::Process(std::span<const int16_t> audio) {
   const int sample_rate_hz = audio.size() * kNum10msFramesInOneSecond;
   RTC_DCHECK_LE(sample_rate_hz, kMaxSampleRateHz);
   vad_.ProcessChunk(audio.data(), audio.size(), sample_rate_hz);
diff --git a/modules/audio_processing/agc/agc.h b/modules/audio_processing/agc/agc.h
index dfc769bce6..f5bd192587 100644
--- a/modules/audio_processing/agc/agc.h
+++ b/modules/audio_processing/agc/agc.h
@@ -13,8 +13,8 @@
 
 #include <cstdint>
 #include <memory>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/vad/voice_activity_detector.h"
 
 namespace webrtc {
@@ -28,7 +28,7 @@ class Agc {
 
   // `audio` must be mono; in a multi-channel stream, provide the first (usually
   // left) channel.
-  virtual void Process(ArrayView<const int16_t> audio);
+  virtual void Process(std::span<const int16_t> audio);
 
   // Retrieves the difference between the target RMS level and the current
   // signal RMS level in dB. Returns true if an update is available and false
diff --git a/modules/audio_processing/agc/agc_manager_direct.cc b/modules/audio_processing/agc/agc_manager_direct.cc
index a8d9ebd0b6..3d5f7fa3d6 100644
--- a/modules/audio_processing/agc/agc_manager_direct.cc
+++ b/modules/audio_processing/agc/agc_manager_direct.cc
@@ -19,8 +19,8 @@
 #include <cstdio>
 #include <memory>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/audio/audio_processing.h"
 #include "api/environment/environment.h"
 #include "api/field_trials_view.h"
@@ -200,7 +200,7 @@ void MonoAgc::Initialize() {
   is_first_frame_ = true;
 }
 
-void MonoAgc::Process(ArrayView<const int16_t> audio,
+void MonoAgc::Process(std::span<const int16_t> audio,
                       std::optional<int> rms_error_override) {
   new_compression_to_set_ = std::nullopt;
 
diff --git a/modules/audio_processing/agc/agc_manager_direct.h b/modules/audio_processing/agc/agc_manager_direct.h
index 5bbbf675cc..04b2e1b9a7 100644
--- a/modules/audio_processing/agc/agc_manager_direct.h
+++ b/modules/audio_processing/agc/agc_manager_direct.h
@@ -15,9 +15,9 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/audio_processing.h"
 #include "api/environment/environment.h"
 #include "modules/audio_processing/agc/agc.h"
@@ -217,7 +217,7 @@ class MonoAgc {
   // the (digital) compression gain to be applied by `agc_`. Must be called
   // after `HandleClipping()`. If `rms_error_override` has a value, RMS error
   // from AGC is overridden by it.
-  void Process(ArrayView<const int16_t> audio,
+  void Process(std::span<const int16_t> audio,
                std::optional<int> rms_error_override);
 
   // Returns the recommended input volume. Must be called after `Process()`.
diff --git a/modules/audio_processing/agc/mock_agc.h b/modules/audio_processing/agc/mock_agc.h
index f958d16e22..8086aa9f01 100644
--- a/modules/audio_processing/agc/mock_agc.h
+++ b/modules/audio_processing/agc/mock_agc.h
@@ -12,8 +12,8 @@
 #define MODULES_AUDIO_PROCESSING_AGC_MOCK_AGC_H_
 
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/agc/agc.h"
 #include "test/gmock.h"
 
@@ -22,7 +22,7 @@ namespace webrtc {
 class MockAgc : public Agc {
  public:
   ~MockAgc() override {}
-  MOCK_METHOD(void, Process, (ArrayView<const int16_t> audio), (override));
+  MOCK_METHOD(void, Process, (std::span<const int16_t> audio), (override));
   MOCK_METHOD(bool, GetRmsErrorDb, (int* error), (override));
   MOCK_METHOD(void, Reset, (), (override));
   MOCK_METHOD(int, set_target_level_dbfs, (int level), (override));
diff --git a/modules/audio_processing/agc2/BUILD.gn b/modules/audio_processing/agc2/BUILD.gn
index 0de1c32bd7..92276ee60c 100644
--- a/modules/audio_processing/agc2/BUILD.gn
+++ b/modules/audio_processing/agc2/BUILD.gn
@@ -28,7 +28,6 @@ rtc_library("speech_level_estimator") {
   deps = [
     ":common",
     "..:apm_logging",
-    "../../../api:array_view",
     "../../../api:field_trials_view",
     "../../../api/audio:audio_processing",
     "../../../rtc_base:checks",
@@ -94,10 +93,7 @@ rtc_library("biquad_filter") {
     "biquad_filter.cc",
     "biquad_filter.h",
   ]
-  deps = [
-    "../../../api:array_view",
-    "../../../rtc_base:checks",
-  ]
+  deps = [ "../../../rtc_base:checks" ]
 }
 
 rtc_library("clipping_predictor") {
@@ -150,7 +146,6 @@ rtc_library("fixed_digital") {
     ":common",
     "..:apm_logging",
     "..:audio_frame_view",
-    "../../../api:array_view",
     "../../../api/audio:audio_frame_api",
     "../../../common_audio",
     "../../../rtc_base:checks",
@@ -216,7 +211,6 @@ rtc_library("input_volume_controller") {
     ":input_volume_stats_reporter",
     "..:audio_buffer",
     "..:audio_frame_view",
-    "../../../api:array_view",
     "../../../api:field_trials_view",
     "../../../api/audio:audio_processing",
     "../../../rtc_base:checks",
@@ -370,7 +364,6 @@ rtc_library("biquad_filter_unittests") {
   sources = [ "biquad_filter_unittest.cc" ]
   deps = [
     ":biquad_filter",
-    "../../../api:array_view",
     "../../../rtc_base:gunit_helpers",
     "../../../test:test_support",
   ]
@@ -397,7 +390,6 @@ rtc_library("fixed_digital_unittests") {
     ":test_utils",
     "..:apm_logging",
     "..:audio_frame_view",
-    "../../../api:array_view",
     "../../../api/audio:audio_frame_api",
     "../../../common_audio",
     "../../../rtc_base:checks",
@@ -424,7 +416,6 @@ rtc_library("input_volume_controller_unittests") {
     ":input_volume_controller",
     "..:audio_buffer",
     "..:audio_frame_view",
-    "../../../api:array_view",
     "../../../api/audio:audio_processing",
     "../../../api/environment:environment_factory",
     "../../../rtc_base:checks",
@@ -462,7 +453,6 @@ rtc_library("vad_wrapper_unittests") {
   deps = [
     ":common",
     ":vad_wrapper",
-    "../../../api:array_view",
     "../../../api/audio:audio_frame_api",
     "../../../rtc_base:checks",
     "../../../rtc_base:gunit_helpers",
diff --git a/modules/audio_processing/agc2/biquad_filter.cc b/modules/audio_processing/agc2/biquad_filter.cc
index 9ff831a04e..e64521c0b0 100644
--- a/modules/audio_processing/agc2/biquad_filter.cc
+++ b/modules/audio_processing/agc2/biquad_filter.cc
@@ -11,8 +11,8 @@
 #include "modules/audio_processing/agc2/biquad_filter.h"
 
 #include <cstddef>
+#include <span>
 
-#include "api/array_view.h"
 #include "rtc_base/checks.h"
 
 namespace webrtc {
@@ -31,7 +31,7 @@ void BiQuadFilter::Reset() {
   state_ = {};
 }
 
-void BiQuadFilter::Process(ArrayView<const float> x, ArrayView<float> y) {
+void BiQuadFilter::Process(std::span<const float> x, std::span<float> y) {
   RTC_DCHECK_EQ(x.size(), y.size());
   const float config_a0 = config_.a[0];
   const float config_a1 = config_.a[1];
diff --git a/modules/audio_processing/agc2/biquad_filter.h b/modules/audio_processing/agc2/biquad_filter.h
index 766a750ea8..8613500440 100644
--- a/modules/audio_processing/agc2/biquad_filter.h
+++ b/modules/audio_processing/agc2/biquad_filter.h
@@ -11,7 +11,7 @@
 #ifndef MODULES_AUDIO_PROCESSING_AGC2_BIQUAD_FILTER_H_
 #define MODULES_AUDIO_PROCESSING_AGC2_BIQUAD_FILTER_H_
 
-#include "api/array_view.h"
+#include <span>
 
 namespace webrtc {
 
@@ -41,7 +41,7 @@ class BiQuadFilter {
 
   // Filters `x` and writes the output in `y`, which must have the same length
   // of `x`. In-place processing is supported.
-  void Process(ArrayView<const float> x, ArrayView<float> y);
+  void Process(std::span<const float> x, std::span<float> y);
 
  private:
   Config config_;
diff --git a/modules/audio_processing/agc2/biquad_filter_unittest.cc b/modules/audio_processing/agc2/biquad_filter_unittest.cc
index bdfdfba009..f0906c82b2 100644
--- a/modules/audio_processing/agc2/biquad_filter_unittest.cc
+++ b/modules/audio_processing/agc2/biquad_filter_unittest.cc
@@ -17,7 +17,8 @@
 
 // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
 // #include "test/fpe_observer.h"
-#include "api/array_view.h"
+#include <span>
+
 #include "test/gtest.h"
 
 namespace webrtc {
@@ -58,11 +59,11 @@ constexpr FloatArraySequence kBiQuadOutputSeq = {
      {{24.84286614f, -62.18094158f, 57.91488056f, -106.65685933f, 13.38760103f,
        -36.60367134f, -94.44880104f, -3.59920354f}}}};
 
-// Fails for every pair from two equally sized ArrayView<float> views
+// Fails for every pair from two equally sized std::span<float> views
 // such that their relative error is above a given threshold. If the expected
 // value of a pair is 0, `tolerance` is used to check the absolute error.
-void ExpectNearRelative(ArrayView<const float> expected,
-                        ArrayView<const float> computed,
+void ExpectNearRelative(std::span<const float> expected,
+                        std::span<const float> computed,
                         const float tolerance) {
   // The relative error is undefined when the expected value is 0.
   // When that happens, check the absolute error instead. `safe_den` is used
diff --git a/modules/audio_processing/agc2/fixed_digital_level_estimator.cc b/modules/audio_processing/agc2/fixed_digital_level_estimator.cc
index a84bb2c0b0..99a4534d10 100644
--- a/modules/audio_processing/agc2/fixed_digital_level_estimator.cc
+++ b/modules/audio_processing/agc2/fixed_digital_level_estimator.cc
@@ -15,7 +15,6 @@
 #include <cmath>
 #include <cstddef>
 
-#include "api/array_view.h"
 #include "api/audio/audio_frame.h"
 #include "api/audio/audio_view.h"
 #include "modules/audio_processing/agc2/agc2_common.h"
diff --git a/modules/audio_processing/agc2/limiter.cc b/modules/audio_processing/agc2/limiter.cc
index faa252c2ae..c564a83744 100644
--- a/modules/audio_processing/agc2/limiter.cc
+++ b/modules/audio_processing/agc2/limiter.cc
@@ -14,9 +14,9 @@
 #include <array>
 #include <cmath>
 #include <cstddef>
+#include <span>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio/audio_view.h"
 #include "modules/audio_processing/agc2/agc2_common.h"
 #include "modules/audio_processing/agc2/interpolated_gain_curve.h"
@@ -38,7 +38,7 @@ constexpr float kAttackFirstSubframeInterpolationPower = 8.0f;
 
 void InterpolateFirstSubframe(float last_factor,
                               float current_factor,
-                              ArrayView<float> subframe) {
+                              std::span<float> subframe) {
   const int n = dchecked_cast<int>(subframe.size());
   constexpr float p = kAttackFirstSubframeInterpolationPower;
   for (int i = 0; i < n; ++i) {
diff --git a/modules/audio_processing/agc2/rnn_vad/BUILD.gn b/modules/audio_processing/agc2/rnn_vad/BUILD.gn
index 075e2a8110..53c9af257b 100644
--- a/modules/audio_processing/agc2/rnn_vad/BUILD.gn
+++ b/modules/audio_processing/agc2/rnn_vad/BUILD.gn
@@ -32,7 +32,6 @@ rtc_library("rnn_vad") {
     ":rnn_vad_spectral_features",
     "..:biquad_filter",
     "..:cpu_features",
-    "../../../../api:array_view",
     "../../../../rtc_base:checks",
     "../../../../rtc_base:safe_compare",
     "../../../../rtc_base:safe_conversions",
@@ -47,7 +46,6 @@ rtc_library("rnn_vad_auto_correlation") {
   ]
   deps = [
     ":rnn_vad_common",
-    "../../../../api:array_view",
     "../../../../rtc_base:checks",
     "../../utility:pffft_wrapper",
   ]
@@ -71,7 +69,6 @@ rtc_library("rnn_vad_lp_residual") {
     "lp_residual.h",
   ]
   deps = [
-    "../../../../api:array_view",
     "../../../../rtc_base:checks",
     "../../../../rtc_base:safe_compare",
   ]
@@ -95,7 +92,6 @@ rtc_library("rnn_vad_layers") {
     ":rnn_vad_common",
     ":vector_math",
     "..:cpu_features",
-    "../../../../api:array_view",
     "../../../../api:function_view",
     "../../../../rtc_base:checks",
     "../../../../rtc_base:safe_conversions",
@@ -111,7 +107,6 @@ rtc_source_set("vector_math") {
   sources = [ "vector_math.h" ]
   deps = [
     "..:cpu_features",
-    "../../../../api:array_view",
     "../../../../rtc_base:checks",
     "../../../../rtc_base:safe_conversions",
     "../../../../rtc_base/system:arch",
@@ -131,7 +126,6 @@ if (current_cpu == "x86" || current_cpu == "x64") {
     }
     deps = [
       ":vector_math",
-      "../../../../api:array_view",
       "../../../../rtc_base:checks",
       "../../../../rtc_base:safe_conversions",
     ]
@@ -157,7 +151,6 @@ rtc_library("rnn_vad_pitch") {
     ":rnn_vad_common",
     ":vector_math",
     "..:cpu_features",
-    "../../../../api:array_view",
     "../../../../rtc_base:checks",
     "../../../../rtc_base:gtest_prod",
     "../../../../rtc_base:safe_compare",
@@ -171,18 +164,12 @@ rtc_library("rnn_vad_pitch") {
 
 rtc_source_set("rnn_vad_ring_buffer") {
   sources = [ "ring_buffer.h" ]
-  deps = [
-    "../../../../api:array_view",
-    "../../../../rtc_base:checks",
-  ]
+  deps = [ "../../../../rtc_base:checks" ]
 }
 
 rtc_source_set("rnn_vad_sequence_buffer") {
   sources = [ "sequence_buffer.h" ]
-  deps = [
-    "../../../../api:array_view",
-    "../../../../rtc_base:checks",
-  ]
+  deps = [ "../../../../rtc_base:checks" ]
 }
 
 rtc_library("rnn_vad_spectral_features") {
@@ -196,7 +183,6 @@ rtc_library("rnn_vad_spectral_features") {
     ":rnn_vad_common",
     ":rnn_vad_ring_buffer",
     ":rnn_vad_symmetric_matrix_buffer",
-    "../../../../api:array_view",
     "../../../../rtc_base:checks",
     "../../../../rtc_base:safe_compare",
     "../../utility:pffft_wrapper",
@@ -206,7 +192,6 @@ rtc_library("rnn_vad_spectral_features") {
 rtc_source_set("rnn_vad_symmetric_matrix_buffer") {
   sources = [ "symmetric_matrix_buffer.h" ]
   deps = [
-    "../../../../api:array_view",
     "../../../../rtc_base:checks",
     "../../../../rtc_base:safe_compare",
   ]
@@ -222,7 +207,6 @@ if (rtc_include_tests) {
     deps = [
       ":rnn_vad",
       ":rnn_vad_common",
-      "../../../../api:array_view",
       "../../../../api:scoped_refptr",
       "../../../../rtc_base:checks",
       "../../../../rtc_base:safe_compare",
@@ -290,7 +274,6 @@ if (rtc_include_tests) {
       ":vector_math",
       "..:cpu_features",
       "../..:audioproc_test_utils",
-      "../../../../api:array_view",
       "../../../../common_audio/",
       "../../../../rtc_base:checks",
       "../../../../rtc_base:logging",
diff --git a/modules/audio_processing/agc2/rnn_vad/auto_correlation.cc b/modules/audio_processing/agc2/rnn_vad/auto_correlation.cc
index b5f9f758a3..971e00d866 100644
--- a/modules/audio_processing/agc2/rnn_vad/auto_correlation.cc
+++ b/modules/audio_processing/agc2/rnn_vad/auto_correlation.cc
@@ -11,8 +11,8 @@
 #include "modules/audio_processing/agc2/rnn_vad/auto_correlation.h"
 
 #include <algorithm>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/agc2/rnn_vad/common.h"
 #include "modules/audio_processing/utility/pffft_wrapper.h"
 #include "rtc_base/checks.h"
@@ -47,8 +47,8 @@ AutoCorrelationCalculator::~AutoCorrelationCalculator() = default;
 // inverted lag equal to 0 that corresponds to a lag equal to the maximum
 // pitch period.
 void AutoCorrelationCalculator::ComputeOnPitchBuffer(
-    ArrayView<const float, kBufSize12kHz> pitch_buf,
-    ArrayView<float, kNumLags12kHz> auto_corr) {
+    std::span<const float, kBufSize12kHz> pitch_buf,
+    std::span<float, kNumLags12kHz> auto_corr) {
   RTC_DCHECK_LT(auto_corr.size(), kMaxPitch12kHz);
   RTC_DCHECK_GT(pitch_buf.size(), kMaxPitch12kHz);
   constexpr int kFftFrameSize = 1 << kAutoCorrelationFftOrder;
diff --git a/modules/audio_processing/agc2/rnn_vad/auto_correlation.h b/modules/audio_processing/agc2/rnn_vad/auto_correlation.h
index 127b259860..edda13f721 100644
--- a/modules/audio_processing/agc2/rnn_vad/auto_correlation.h
+++ b/modules/audio_processing/agc2/rnn_vad/auto_correlation.h
@@ -12,8 +12,8 @@
 #define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_AUTO_CORRELATION_H_
 
 #include <memory>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/agc2/rnn_vad/common.h"
 #include "modules/audio_processing/utility/pffft_wrapper.h"
 
@@ -32,8 +32,8 @@ class AutoCorrelationCalculator {
 
   // Computes the auto-correlation coefficients for a target pitch interval.
   // `auto_corr` indexes are inverted lags.
-  void ComputeOnPitchBuffer(ArrayView<const float, kBufSize12kHz> pitch_buf,
-                            ArrayView<float, kNumLags12kHz> auto_corr);
+  void ComputeOnPitchBuffer(std::span<const float, kBufSize12kHz> pitch_buf,
+                            std::span<float, kNumLags12kHz> auto_corr);
 
  private:
   Pffft fft_;
diff --git a/modules/audio_processing/agc2/rnn_vad/features_extraction.cc b/modules/audio_processing/agc2/rnn_vad/features_extraction.cc
index 1dbfa1a841..53362ea86e 100644
--- a/modules/audio_processing/agc2/rnn_vad/features_extraction.cc
+++ b/modules/audio_processing/agc2/rnn_vad/features_extraction.cc
@@ -12,8 +12,8 @@
 
 #include <array>
 #include <cstddef>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/agc2/biquad_filter.h"
 #include "modules/audio_processing/agc2/cpu_features.h"
 #include "modules/audio_processing/agc2/rnn_vad/common.h"
@@ -55,8 +55,8 @@ void FeaturesExtractor::Reset() {
 }
 
 bool FeaturesExtractor::CheckSilenceComputeFeatures(
-    ArrayView<const float, kFrameSize10ms24kHz> samples,
-    ArrayView<float, kFeatureVectorSize> feature_vector) {
+    std::span<const float, kFrameSize10ms24kHz> samples,
+    std::span<float, kFeatureVectorSize> feature_vector) {
   // Pre-processing.
   if (use_high_pass_filter_) {
     std::array<float, kFrameSize10ms24kHz> samples_filtered;
diff --git a/modules/audio_processing/agc2/rnn_vad/features_extraction.h b/modules/audio_processing/agc2/rnn_vad/features_extraction.h
index 2928a7fa00..f952045d6f 100644
--- a/modules/audio_processing/agc2/rnn_vad/features_extraction.h
+++ b/modules/audio_processing/agc2/rnn_vad/features_extraction.h
@@ -11,9 +11,9 @@
 #ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_FEATURES_EXTRACTION_H_
 #define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_FEATURES_EXTRACTION_H_
 
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/agc2/biquad_filter.h"
 #include "modules/audio_processing/agc2/cpu_features.h"
 #include "modules/audio_processing/agc2/rnn_vad/common.h"
@@ -37,8 +37,8 @@ class FeaturesExtractor {
   // `feature_vector` is partially written and therefore must not be used to
   // feed the VAD RNN.
   bool CheckSilenceComputeFeatures(
-      ArrayView<const float, kFrameSize10ms24kHz> samples,
-      ArrayView<float, kFeatureVectorSize> feature_vector);
+      std::span<const float, kFrameSize10ms24kHz> samples,
+      std::span<float, kFeatureVectorSize> feature_vector);
 
  private:
   const bool use_high_pass_filter_;
@@ -47,11 +47,11 @@ class FeaturesExtractor {
   BiQuadFilter hpf_;
   SequenceBuffer<float, kBufSize24kHz, kFrameSize10ms24kHz, kFrameSize20ms24kHz>
       pitch_buf_24kHz_;
-  ArrayView<const float, kBufSize24kHz> pitch_buf_24kHz_view_;
+  std::span<const float, kBufSize24kHz> pitch_buf_24kHz_view_;
   std::vector<float> lp_residual_;
-  ArrayView<float, kBufSize24kHz> lp_residual_view_;
+  std::span<float, kBufSize24kHz> lp_residual_view_;
   PitchEstimator pitch_estimator_;
-  ArrayView<const float, kFrameSize20ms24kHz> reference_frame_view_;
+  std::span<const float, kFrameSize20ms24kHz> reference_frame_view_;
   SpectralFeaturesExtractor spectral_features_extractor_;
   int pitch_period_48kHz_;
 };
diff --git a/modules/audio_processing/agc2/rnn_vad/features_extraction_unittest.cc b/modules/audio_processing/agc2/rnn_vad/features_extraction_unittest.cc
index 7c0b575217..b63198d448 100644
--- a/modules/audio_processing/agc2/rnn_vad/features_extraction_unittest.cc
+++ b/modules/audio_processing/agc2/rnn_vad/features_extraction_unittest.cc
@@ -11,9 +11,9 @@
 #include "modules/audio_processing/agc2/rnn_vad/features_extraction.h"
 
 #include <cmath>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/agc2/cpu_features.h"
 #include "modules/audio_processing/agc2/rnn_vad/common.h"
 #include "rtc_base/numerics/safe_compare.h"
@@ -43,7 +43,7 @@ bool PitchIsValid(float pitch_hz) {
          pitch_period <= kMaxPitch24kHz;
 }
 
-void CreatePureTone(float amplitude, float freq_hz, ArrayView<float> dst) {
+void CreatePureTone(float amplitude, float freq_hz, std::span<float> dst) {
   for (int i = 0; SafeLt(i, dst.size()); ++i) {
     dst[i] = amplitude * std::sin(2.f * kPi * freq_hz * i / kSampleRate24kHz);
   }
@@ -53,8 +53,8 @@ void CreatePureTone(float amplitude, float freq_hz, ArrayView<float> dst) {
 // For every frame, the output is written into `feature_vector`. Returns true
 // if silence is detected in the last frame.
 bool FeedTestData(FeaturesExtractor& features_extractor,
-                  ArrayView<const float> samples,
-                  ArrayView<float, kFeatureVectorSize> feature_vector) {
+                  std::span<const float> samples,
+                  std::span<float, kFeatureVectorSize> feature_vector) {
   // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
   // FloatingPointExceptionObserver fpe_observer;
   bool is_silence = true;
@@ -81,7 +81,7 @@ TEST(RnnVadTest, FeatureExtractionLowHighPitch) {
   std::vector<float> samples(kNumTestDataSize);
   std::vector<float> feature_vector(kFeatureVectorSize);
   ASSERT_EQ(kFeatureVectorSize, dchecked_cast<int>(feature_vector.size()));
-  ArrayView<float, kFeatureVectorSize> feature_vector_view(
+  std::span<float, kFeatureVectorSize> feature_vector_view(
       feature_vector.data(), kFeatureVectorSize);
 
   // Extract the normalized scalar feature that is proportional to the estimated
diff --git a/modules/audio_processing/agc2/rnn_vad/lp_residual.cc b/modules/audio_processing/agc2/rnn_vad/lp_residual.cc
index 46cb1eb299..5960552699 100644
--- a/modules/audio_processing/agc2/rnn_vad/lp_residual.cc
+++ b/modules/audio_processing/agc2/rnn_vad/lp_residual.cc
@@ -14,8 +14,8 @@
 #include <array>
 #include <cmath>
 #include <numeric>
+#include <span>
 
-#include "api/array_view.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/numerics/safe_compare.h"
 
@@ -26,8 +26,8 @@ namespace {
 // Computes auto-correlation coefficients for `x` and writes them in
 // `auto_corr`. The lag values are in {0, ..., max_lag - 1}, where max_lag
 // equals the size of `auto_corr`.
-void ComputeAutoCorrelation(ArrayView<const float> x,
-                            ArrayView<float, kNumLpcCoefficients> auto_corr) {
+void ComputeAutoCorrelation(std::span<const float> x,
+                            std::span<float, kNumLpcCoefficients> auto_corr) {
   constexpr int max_lag = auto_corr.size();
   RTC_DCHECK_LT(max_lag, x.size());
   for (int lag = 0; lag < max_lag; ++lag) {
@@ -37,7 +37,7 @@ void ComputeAutoCorrelation(ArrayView<const float> x,
 }
 
 // Applies denoising to the auto-correlation coefficients.
-void DenoiseAutoCorrelation(ArrayView<float, kNumLpcCoefficients> auto_corr) {
+void DenoiseAutoCorrelation(std::span<float, kNumLpcCoefficients> auto_corr) {
   // Assume -40 dB white noise floor.
   auto_corr[0] *= 1.0001f;
   // Hard-coded values obtained as
@@ -52,8 +52,8 @@ void DenoiseAutoCorrelation(ArrayView<float, kNumLpcCoefficients> auto_corr) {
 // Computes the initial inverse filter coefficients given the auto-correlation
 // coefficients of an input frame.
 void ComputeInitialInverseFilterCoefficients(
-    ArrayView<const float, kNumLpcCoefficients> auto_corr,
-    ArrayView<float, kNumLpcCoefficients - 1> lpc_coeffs) {
+    std::span<const float, kNumLpcCoefficients> auto_corr,
+    std::span<float, kNumLpcCoefficients - 1> lpc_coeffs) {
   float error = auto_corr[0];
   for (int i = 0; i < kNumLpcCoefficients - 1; ++i) {
     float reflection_coeff = 0.f;
@@ -87,8 +87,8 @@ void ComputeInitialInverseFilterCoefficients(
 }  // namespace
 
 void ComputeAndPostProcessLpcCoefficients(
-    ArrayView<const float> x,
-    ArrayView<float, kNumLpcCoefficients> lpc_coeffs) {
+    std::span<const float> x,
+    std::span<float, kNumLpcCoefficients> lpc_coeffs) {
   std::array<float, kNumLpcCoefficients> auto_corr;
   ComputeAutoCorrelation(x, auto_corr);
   if (auto_corr[0] == 0.f) {  // Empty frame.
@@ -113,9 +113,9 @@ void ComputeAndPostProcessLpcCoefficients(
   static_assert(kNumLpcCoefficients == 5, "Update `lpc_coeffs(_pre)`.");
 }
 
-void ComputeLpResidual(ArrayView<const float, kNumLpcCoefficients> lpc_coeffs,
-                       ArrayView<const float> x,
-                       ArrayView<float> y) {
+void ComputeLpResidual(std::span<const float, kNumLpcCoefficients> lpc_coeffs,
+                       std::span<const float> x,
+                       std::span<float> y) {
   RTC_DCHECK_GT(x.size(), kNumLpcCoefficients);
   RTC_DCHECK_EQ(x.size(), y.size());
   // The code below implements the following operation:
diff --git a/modules/audio_processing/agc2/rnn_vad/lp_residual.h b/modules/audio_processing/agc2/rnn_vad/lp_residual.h
index f29dfe03ad..115163709b 100644
--- a/modules/audio_processing/agc2/rnn_vad/lp_residual.h
+++ b/modules/audio_processing/agc2/rnn_vad/lp_residual.h
@@ -13,7 +13,7 @@
 
 #include <stddef.h>
 
-#include "api/array_view.h"
+#include <span>
 
 namespace webrtc {
 namespace rnn_vad {
@@ -24,15 +24,15 @@ constexpr int kNumLpcCoefficients = 5;
 // Given a frame `x`, computes a post-processed version of LPC coefficients
 // tailored for pitch estimation.
 void ComputeAndPostProcessLpcCoefficients(
-    ArrayView<const float> x,
-    ArrayView<float, kNumLpcCoefficients> lpc_coeffs);
+    std::span<const float> x,
+    std::span<float, kNumLpcCoefficients> lpc_coeffs);
 
 // Computes the LP residual for the input frame `x` and the LPC coefficients
 // `lpc_coeffs`. `y` and `x` can point to the same array for in-place
 // computation.
-void ComputeLpResidual(ArrayView<const float, kNumLpcCoefficients> lpc_coeffs,
-                       ArrayView<const float> x,
-                       ArrayView<float> y);
+void ComputeLpResidual(std::span<const float, kNumLpcCoefficients> lpc_coeffs,
+                       std::span<const float> x,
+                       std::span<float> y);
 
 }  // namespace rnn_vad
 }  // namespace webrtc
diff --git a/modules/audio_processing/agc2/rnn_vad/pitch_search.cc b/modules/audio_processing/agc2/rnn_vad/pitch_search.cc
index d6fe28be58..cd734d2d3b 100644
--- a/modules/audio_processing/agc2/rnn_vad/pitch_search.cc
+++ b/modules/audio_processing/agc2/rnn_vad/pitch_search.cc
@@ -11,8 +11,8 @@
 #include "modules/audio_processing/agc2/rnn_vad/pitch_search.h"
 
 #include <cstddef>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/agc2/cpu_features.h"
 #include "modules/audio_processing/agc2/rnn_vad/common.h"
 #include "modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h"
@@ -30,11 +30,11 @@ PitchEstimator::PitchEstimator(const AvailableCpuFeatures& cpu_features)
 PitchEstimator::~PitchEstimator() = default;
 
 int PitchEstimator::Estimate(
-    ArrayView<const float, kBufSize24kHz> pitch_buffer) {
-  ArrayView<float, kBufSize12kHz> pitch_buffer_12kHz_view(
+    std::span<const float, kBufSize24kHz> pitch_buffer) {
+  std::span<float, kBufSize12kHz> pitch_buffer_12kHz_view(
       pitch_buffer_12kHz_.data(), kBufSize12kHz);
   RTC_DCHECK_EQ(pitch_buffer_12kHz_.size(), pitch_buffer_12kHz_view.size());
-  ArrayView<float, kNumLags12kHz> auto_correlation_12kHz_view(
+  std::span<float, kNumLags12kHz> auto_correlation_12kHz_view(
       auto_correlation_12kHz_.data(), kNumLags12kHz);
   RTC_DCHECK_EQ(auto_correlation_12kHz_.size(),
                 auto_correlation_12kHz_view.size());
@@ -54,7 +54,7 @@ int PitchEstimator::Estimate(
 
   // Refine the initial pitch period estimation from 12 kHz to 48 kHz.
   // Pre-compute frame energies at 24 kHz.
-  ArrayView<float, kRefineNumLags24kHz> y_energy_24kHz_view(
+  std::span<float, kRefineNumLags24kHz> y_energy_24kHz_view(
       y_energy_24kHz_.data(), kRefineNumLags24kHz);
   RTC_DCHECK_EQ(y_energy_24kHz_.size(), y_energy_24kHz_view.size());
   ComputeSlidingFrameSquareEnergies24kHz(pitch_buffer, y_energy_24kHz_view,
diff --git a/modules/audio_processing/agc2/rnn_vad/pitch_search.h b/modules/audio_processing/agc2/rnn_vad/pitch_search.h
index 2804424893..9f228bcf8f 100644
--- a/modules/audio_processing/agc2/rnn_vad/pitch_search.h
+++ b/modules/audio_processing/agc2/rnn_vad/pitch_search.h
@@ -11,9 +11,9 @@
 #ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_PITCH_SEARCH_H_
 #define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_PITCH_SEARCH_H_
 
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/agc2/cpu_features.h"
 #include "modules/audio_processing/agc2/rnn_vad/auto_correlation.h"
 #include "modules/audio_processing/agc2/rnn_vad/common.h"
@@ -31,7 +31,7 @@ class PitchEstimator {
   PitchEstimator& operator=(const PitchEstimator&) = delete;
   ~PitchEstimator();
   // Returns the estimated pitch period at 48 kHz.
-  int Estimate(ArrayView<const float, kBufSize24kHz> pitch_buffer);
+  int Estimate(std::span<const float, kBufSize24kHz> pitch_buffer);
 
  private:
   FRIEND_TEST_ALL_PREFIXES(RnnVadTest, PitchSearchWithinTolerance);
diff --git a/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc b/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc
index 084eb21b8f..25f33fa71d 100644
--- a/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc
+++ b/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc
@@ -15,8 +15,8 @@
 #include <cmath>
 #include <cstddef>
 #include <cstdlib>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/agc2/cpu_features.h"
 #include "modules/audio_processing/agc2/rnn_vad/common.h"
 #include "modules/audio_processing/agc2/rnn_vad/vector_math.h"
@@ -27,7 +27,7 @@ namespace rnn_vad {
 namespace {
 
 float ComputeAutoCorrelation(int inverted_lag,
-                             ArrayView<const float, kBufSize24kHz> pitch_buffer,
+                             std::span<const float, kBufSize24kHz> pitch_buffer,
                              const VectorMath& vector_math) {
   RTC_DCHECK_LT(inverted_lag, kBufSize24kHz);
   RTC_DCHECK_LT(inverted_lag, kRefineNumLags24kHz);
@@ -65,7 +65,7 @@ int GetPitchPseudoInterpolationOffset(float prev_auto_correlation,
 // output sample rate is twice as that of `lag`.
 int PitchPseudoInterpolationLagPitchBuf(
     int lag,
-    ArrayView<const float, kBufSize24kHz> pitch_buffer,
+    std::span<const float, kBufSize24kHz> pitch_buffer,
     const VectorMath& vector_math) {
   int offset = 0;
   // Cannot apply pseudo-interpolation at the boundaries.
@@ -158,8 +158,8 @@ class InvertedLagsIndex {
 // the inverted lags for the computed auto correlation values.
 void ComputeAutoCorrelation(
     Range inverted_lags,
-    ArrayView<const float, kBufSize24kHz> pitch_buffer,
-    ArrayView<float, kInitialNumLags24kHz> auto_correlation,
+    std::span<const float, kBufSize24kHz> pitch_buffer,
+    std::span<float, kInitialNumLags24kHz> auto_correlation,
     InvertedLagsIndex& inverted_lags_index,
     const VectorMath& vector_math) {
   // Check valid range.
@@ -186,10 +186,10 @@ void ComputeAutoCorrelation(
 // Searches the strongest pitch period at 24 kHz and returns its inverted lag at
 // 48 kHz.
 int ComputePitchPeriod48kHz(
-    ArrayView<const float, kBufSize24kHz> /* pitch_buffer */,
-    ArrayView<const int> inverted_lags,
-    ArrayView<const float, kInitialNumLags24kHz> auto_correlation,
-    ArrayView<const float, kRefineNumLags24kHz> y_energy,
+    std::span<const float, kBufSize24kHz> /* pitch_buffer */,
+    std::span<const int> inverted_lags,
+    std::span<const float, kInitialNumLags24kHz> auto_correlation,
+    std::span<const float, kRefineNumLags24kHz> y_energy,
     const VectorMath& /* vector_math */) {
   static_assert(kMaxPitch24kHz > kInitialNumLags24kHz, "");
   static_assert(kMaxPitch24kHz < kBufSize24kHz, "");
@@ -287,8 +287,8 @@ bool IsAlternativePitchStrongerThanInitial(PitchInfo last,
 
 }  // namespace
 
-void Decimate2x(ArrayView<const float, kBufSize24kHz> src,
-                ArrayView<float, kBufSize12kHz> dst) {
+void Decimate2x(std::span<const float, kBufSize24kHz> src,
+                std::span<float, kBufSize12kHz> dst) {
   // TODO(bugs.webrtc.org/9076): Consider adding anti-aliasing filter.
   static_assert(2 * kBufSize12kHz == kBufSize24kHz, "");
   for (int i = 0; i < kBufSize12kHz; ++i) {
@@ -297,8 +297,8 @@ void Decimate2x(ArrayView<const float, kBufSize24kHz> src,
 }
 
 void ComputeSlidingFrameSquareEnergies24kHz(
-    ArrayView<const float, kBufSize24kHz> pitch_buffer,
-    ArrayView<float, kRefineNumLags24kHz> y_energy,
+    std::span<const float, kBufSize24kHz> pitch_buffer,
+    std::span<float, kRefineNumLags24kHz> y_energy,
     AvailableCpuFeatures cpu_features) {
   VectorMath vector_math(cpu_features);
   static_assert(kFrameSize20ms24kHz < kBufSize24kHz, "");
@@ -317,8 +317,8 @@ void ComputeSlidingFrameSquareEnergies24kHz(
 }
 
 CandidatePitchPeriods ComputePitchPeriod12kHz(
-    ArrayView<const float, kBufSize12kHz> pitch_buffer,
-    ArrayView<const float, kNumLags12kHz> auto_correlation,
+    std::span<const float, kBufSize12kHz> pitch_buffer,
+    std::span<const float, kNumLags12kHz> auto_correlation,
     AvailableCpuFeatures cpu_features) {
   static_assert(kMaxPitch12kHz > kNumLags12kHz, "");
   static_assert(kMaxPitch12kHz < kBufSize12kHz, "");
@@ -376,8 +376,8 @@ CandidatePitchPeriods ComputePitchPeriod12kHz(
 }
 
 int ComputePitchPeriod48kHz(
-    ArrayView<const float, kBufSize24kHz> pitch_buffer,
-    ArrayView<const float, kRefineNumLags24kHz> y_energy,
+    std::span<const float, kBufSize24kHz> pitch_buffer,
+    std::span<const float, kRefineNumLags24kHz> y_energy,
     CandidatePitchPeriods pitch_candidates,
     AvailableCpuFeatures cpu_features) {
   // Compute the auto-correlation terms only for neighbors of the two pitch
@@ -414,8 +414,8 @@ int ComputePitchPeriod48kHz(
 }
 
 PitchInfo ComputeExtendedPitchPeriod48kHz(
-    ArrayView<const float, kBufSize24kHz> pitch_buffer,
-    ArrayView<const float, kRefineNumLags24kHz> y_energy,
+    std::span<const float, kBufSize24kHz> pitch_buffer,
+    std::span<const float, kRefineNumLags24kHz> y_energy,
     int initial_pitch_period_48kHz,
     PitchInfo last_pitch_48kHz,
     AvailableCpuFeatures cpu_features) {
diff --git a/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h b/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h
index 5609d5a058..9ff1d27085 100644
--- a/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h
+++ b/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h
@@ -12,8 +12,8 @@
 #define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_PITCH_SEARCH_INTERNAL_H_
 
 #include <cstddef>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/agc2/cpu_features.h"
 #include "modules/audio_processing/agc2/rnn_vad/common.h"
 
@@ -21,8 +21,8 @@ namespace webrtc {
 namespace rnn_vad {
 
 // Performs 2x decimation without any anti-aliasing filter.
-void Decimate2x(ArrayView<const float, kBufSize24kHz> src,
-                ArrayView<float, kBufSize12kHz> dst);
+void Decimate2x(std::span<const float, kBufSize24kHz> src,
+                std::span<float, kBufSize12kHz> dst);
 
 // Key concepts and keywords used below in this file.
 //
@@ -62,8 +62,8 @@ void Decimate2x(ArrayView<const float, kBufSize24kHz> src,
 // Computes the sum of squared samples for every sliding frame `y` in the pitch
 // buffer. The indexes of `y_energy` are inverted lags.
 void ComputeSlidingFrameSquareEnergies24kHz(
-    ArrayView<const float, kBufSize24kHz> pitch_buffer,
-    ArrayView<float, kRefineNumLags24kHz> y_energy,
+    std::span<const float, kBufSize24kHz> pitch_buffer,
+    std::span<float, kRefineNumLags24kHz> y_energy,
     AvailableCpuFeatures cpu_features);
 
 // Top-2 pitch period candidates. Unit: number of samples - i.e., inverted lags.
@@ -76,16 +76,16 @@ struct CandidatePitchPeriods {
 // pitch buffer and the auto-correlation values (having inverted lags as
 // indexes).
 CandidatePitchPeriods ComputePitchPeriod12kHz(
-    ArrayView<const float, kBufSize12kHz> pitch_buffer,
-    ArrayView<const float, kNumLags12kHz> auto_correlation,
+    std::span<const float, kBufSize12kHz> pitch_buffer,
+    std::span<const float, kNumLags12kHz> auto_correlation,
     AvailableCpuFeatures cpu_features);
 
 // Computes the pitch period at 48 kHz given a view on the 24 kHz pitch buffer,
 // the energies for the sliding frames `y` at 24 kHz and the pitch period
 // candidates at 24 kHz (encoded as inverted lag).
 int ComputePitchPeriod48kHz(
-    ArrayView<const float, kBufSize24kHz> pitch_buffer,
-    ArrayView<const float, kRefineNumLags24kHz> y_energy,
+    std::span<const float, kBufSize24kHz> pitch_buffer,
+    std::span<const float, kRefineNumLags24kHz> y_energy,
     CandidatePitchPeriods pitch_candidates_24kHz,
     AvailableCpuFeatures cpu_features);
 
@@ -99,8 +99,8 @@ struct PitchInfo {
 // `y` at 24 kHz, the initial 48 kHz estimation (computed by
 // `ComputePitchPeriod48kHz()`) and the last estimated pitch.
 PitchInfo ComputeExtendedPitchPeriod48kHz(
-    ArrayView<const float, kBufSize24kHz> pitch_buffer,
-    ArrayView<const float, kRefineNumLags24kHz> y_energy,
+    std::span<const float, kBufSize24kHz> pitch_buffer,
+    std::span<const float, kRefineNumLags24kHz> y_energy,
     int initial_pitch_period_48kHz,
     PitchInfo last_pitch_48kHz,
     AvailableCpuFeatures cpu_features);
diff --git a/modules/audio_processing/agc2/rnn_vad/pitch_search_internal_unittest.cc b/modules/audio_processing/agc2/rnn_vad/pitch_search_internal_unittest.cc
index 8235ccac3f..3f39e4fadc 100644
--- a/modules/audio_processing/agc2/rnn_vad/pitch_search_internal_unittest.cc
+++ b/modules/audio_processing/agc2/rnn_vad/pitch_search_internal_unittest.cc
@@ -11,10 +11,10 @@
 #include "modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h"
 
 #include <array>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/agc2/cpu_features.h"
 #include "modules/audio_processing/agc2/rnn_vad/common.h"
 #include "modules/audio_processing/agc2/rnn_vad/test_utils.h"
@@ -93,7 +93,7 @@ TEST(RnnVadTest, ComputePitchPeriod48kHzBitExactness) {
 
   PitchTestData test_data;
   std::vector<float> y_energy(kRefineNumLags24kHz);
-  ArrayView<float, kRefineNumLags24kHz> y_energy_view(y_energy.data(),
+  std::span<float, kRefineNumLags24kHz> y_energy_view(y_energy.data(),
                                                       kRefineNumLags24kHz);
   ComputeSlidingFrameSquareEnergies24kHz(test_data.PitchBuffer24kHzView(),
                                          y_energy_view, cpu_features);
@@ -128,7 +128,7 @@ TEST_P(PitchCandidatesParametrization,
 
   PitchTestData test_data;
   std::vector<float> y_energy(kRefineNumLags24kHz);
-  ArrayView<float, kRefineNumLags24kHz> y_energy_view(y_energy.data(),
+  std::span<float, kRefineNumLags24kHz> y_energy_view(y_energy.data(),
                                                       kRefineNumLags24kHz);
   ComputeSlidingFrameSquareEnergies24kHz(test_data.PitchBuffer24kHzView(),
                                          y_energy_view, params.cpu_features);
@@ -179,7 +179,7 @@ TEST_P(ExtendedPitchPeriodSearchParametrizaion,
 
   PitchTestData test_data;
   std::vector<float> y_energy(kRefineNumLags24kHz);
-  ArrayView<float, kRefineNumLags24kHz> y_energy_view(y_energy.data(),
+  std::span<float, kRefineNumLags24kHz> y_energy_view(y_energy.data(),
                                                       kRefineNumLags24kHz);
   ComputeSlidingFrameSquareEnergies24kHz(test_data.PitchBuffer24kHzView(),
                                          y_energy_view, params.cpu_features);
diff --git a/modules/audio_processing/agc2/rnn_vad/ring_buffer.h b/modules/audio_processing/agc2/rnn_vad/ring_buffer.h
index 61ecc380e2..6c8e4db303 100644
--- a/modules/audio_processing/agc2/rnn_vad/ring_buffer.h
+++ b/modules/audio_processing/agc2/rnn_vad/ring_buffer.h
@@ -13,9 +13,9 @@
 
 #include <array>
 #include <cstring>
+#include <span>
 #include <type_traits>
 
-#include "api/array_view.h"
 #include "rtc_base/checks.h"
 
 namespace webrtc {
@@ -37,7 +37,7 @@ class RingBuffer {
   // Set the ring buffer values to zero.
   void Reset() { buffer_.fill(0); }
   // Replace the least recently pushed array in the buffer with `new_values`.
-  void Push(ArrayView<const T, S> new_values) {
+  void Push(std::span<const T, S> new_values) {
     std::memcpy(buffer_.data() + S * tail_, new_values.data(), S * sizeof(T));
     tail_ += 1;
     if (tail_ == N)
@@ -46,13 +46,13 @@ class RingBuffer {
   // Return an array view onto the array with a given delay. A view on the last
   // and least recently push array is returned when `delay` is 0 and N - 1
   // respectively.
-  ArrayView<const T, S> GetArrayView(int delay) const {
+  std::span<const T, S> GetArrayView(int delay) const {
     RTC_DCHECK_LE(0, delay);
     RTC_DCHECK_LT(delay, N);
     int offset = tail_ - 1 - delay;
     if (offset < 0)
       offset += N;
-    return ArrayView<const T, S>(buffer_.data() + S * offset, S);
+    return std::span<const T, S>(buffer_.data() + S * offset, S);
   }
 
  private:
diff --git a/modules/audio_processing/agc2/rnn_vad/ring_buffer_unittest.cc b/modules/audio_processing/agc2/rnn_vad/ring_buffer_unittest.cc
index 82bd01f591..14f51aba58 100644
--- a/modules/audio_processing/agc2/rnn_vad/ring_buffer_unittest.cc
+++ b/modules/audio_processing/agc2/rnn_vad/ring_buffer_unittest.cc
@@ -13,8 +13,8 @@
 #include <array>
 #include <cstddef>
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
 
@@ -31,7 +31,7 @@ void TestRingBuffer() {
   SCOPED_TRACE(S);
   std::array<T, S> prev_pushed_array;
   std::array<T, S> pushed_array;
-  ArrayView<const T, S> pushed_array_view(pushed_array.data(), S);
+  std::span<const T, S> pushed_array_view(pushed_array.data(), S);
 
   // Init.
   RingBuffer<T, S, N> ring_buf;
diff --git a/modules/audio_processing/agc2/rnn_vad/rnn.cc b/modules/audio_processing/agc2/rnn_vad/rnn.cc
index 775ea86343..62195cf85d 100644
--- a/modules/audio_processing/agc2/rnn_vad/rnn.cc
+++ b/modules/audio_processing/agc2/rnn_vad/rnn.cc
@@ -10,7 +10,8 @@
 
 #include "modules/audio_processing/agc2/rnn_vad/rnn.h"
 
-#include "api/array_view.h"
+#include <span>
+
 #include "modules/audio_processing/agc2/cpu_features.h"
 #include "modules/audio_processing/agc2/rnn_vad/common.h"
 #include "modules/audio_processing/agc2/rnn_vad/rnn_fc.h"
@@ -79,7 +80,7 @@ void RnnVad::Reset() {
 }
 
 float RnnVad::ComputeVadProbability(
-    ArrayView<const float, kFeatureVectorSize> feature_vector,
+    std::span<const float, kFeatureVectorSize> feature_vector,
     bool is_silence) {
   if (is_silence) {
     Reset();
diff --git a/modules/audio_processing/agc2/rnn_vad/rnn.h b/modules/audio_processing/agc2/rnn_vad/rnn.h
index 7b47d23e2d..caa2d360ed 100644
--- a/modules/audio_processing/agc2/rnn_vad/rnn.h
+++ b/modules/audio_processing/agc2/rnn_vad/rnn.h
@@ -13,8 +13,8 @@
 
 #include <stddef.h>
 
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/agc2/cpu_features.h"
 #include "modules/audio_processing/agc2/rnn_vad/common.h"
 #include "modules/audio_processing/agc2/rnn_vad/rnn_fc.h"
@@ -35,7 +35,7 @@ class RnnVad {
   // Observes `feature_vector` and `is_silence`, updates the RNN and returns the
   // current voice probability.
   float ComputeVadProbability(
-      ArrayView<const float, kFeatureVectorSize> feature_vector,
+      std::span<const float, kFeatureVectorSize> feature_vector,
       bool is_silence);
 
  private:
diff --git a/modules/audio_processing/agc2/rnn_vad/rnn_fc.cc b/modules/audio_processing/agc2/rnn_vad/rnn_fc.cc
index 0c4b7bbe4f..011c77372d 100644
--- a/modules/audio_processing/agc2/rnn_vad/rnn_fc.cc
+++ b/modules/audio_processing/agc2/rnn_vad/rnn_fc.cc
@@ -12,10 +12,10 @@
 
 #include <algorithm>
 #include <cstdint>
+#include <span>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/function_view.h"
 #include "modules/audio_processing/agc2/cpu_features.h"
 #include "rtc_base/checks.h"
@@ -27,7 +27,7 @@ namespace webrtc {
 namespace rnn_vad {
 namespace {
 
-std::vector<float> GetScaledParams(ArrayView<const int8_t> params) {
+std::vector<float> GetScaledParams(std::span<const int8_t> params) {
   std::vector<float> scaled_params(params.size());
   std::transform(params.begin(), params.end(), scaled_params.begin(),
                  [](int8_t x) -> float {
@@ -39,7 +39,7 @@ std::vector<float> GetScaledParams(ArrayView<const int8_t> params) {
 // TODO(bugs.chromium.org/10480): Hard-code optimized layout and remove this
 // function to improve setup time.
 // Casts and scales `weights` and re-arranges the layout.
-std::vector<float> PreprocessWeights(ArrayView<const int8_t> weights,
+std::vector<float> PreprocessWeights(std::span<const int8_t> weights,
                                      int output_size) {
   if (output_size == 1) {
     return GetScaledParams(weights);
@@ -72,8 +72,8 @@ FunctionView<float(float)> GetActivationFunction(
 FullyConnectedLayer::FullyConnectedLayer(
     const int input_size,
     const int output_size,
-    const ArrayView<const int8_t> bias,
-    const ArrayView<const int8_t> weights,
+    const std::span<const int8_t> bias,
+    const std::span<const int8_t> weights,
     ActivationFunction activation_function,
     const AvailableCpuFeatures& cpu_features,
     absl::string_view layer_name)
@@ -95,9 +95,9 @@ FullyConnectedLayer::FullyConnectedLayer(
 
 FullyConnectedLayer::~FullyConnectedLayer() = default;
 
-void FullyConnectedLayer::ComputeOutput(ArrayView<const float> input) {
+void FullyConnectedLayer::ComputeOutput(std::span<const float> input) {
   RTC_DCHECK_EQ(input.size(), input_size_);
-  ArrayView<const float> weights(weights_);
+  std::span<const float> weights(weights_);
   for (int o = 0; o < output_size_; ++o) {
     output_[o] = activation_function_(
         bias_[o] + vector_math_.DotProduct(
diff --git a/modules/audio_processing/agc2/rnn_vad/rnn_fc.h b/modules/audio_processing/agc2/rnn_vad/rnn_fc.h
index 12ed2f5453..bde2d67d8b 100644
--- a/modules/audio_processing/agc2/rnn_vad/rnn_fc.h
+++ b/modules/audio_processing/agc2/rnn_vad/rnn_fc.h
@@ -13,10 +13,10 @@
 
 #include <array>
 #include <cstdint>
+#include <span>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/function_view.h"
 #include "modules/audio_processing/agc2/cpu_features.h"
 #include "modules/audio_processing/agc2/rnn_vad/vector_math.h"
@@ -37,8 +37,8 @@ class FullyConnectedLayer {
   // Ctor. `output_size` cannot be greater than `kFullyConnectedLayerMaxUnits`.
   FullyConnectedLayer(int input_size,
                       int output_size,
-                      ArrayView<const int8_t> bias,
-                      ArrayView<const int8_t> weights,
+                      std::span<const int8_t> bias,
+                      std::span<const int8_t> weights,
                       ActivationFunction activation_function,
                       const AvailableCpuFeatures& cpu_features,
                       absl::string_view layer_name);
@@ -54,12 +54,12 @@ class FullyConnectedLayer {
   int size() const { return output_size_; }
 
   // Returns the output buffer.
-  ArrayView<const float> output() const {
-    return MakeArrayView(output_.data(), output_size_);
+  std::span<const float> output() const {
+    return std::span(output_.data(), output_size_);
   }
 
   // Computes the fully-connected layer output.
-  void ComputeOutput(ArrayView<const float> input);
+  void ComputeOutput(std::span<const float> input);
 
  private:
   const int input_size_;
diff --git a/modules/audio_processing/agc2/rnn_vad/rnn_fc_unittest.cc b/modules/audio_processing/agc2/rnn_vad/rnn_fc_unittest.cc
index 705f43aa2e..db85581c20 100644
--- a/modules/audio_processing/agc2/rnn_vad/rnn_fc_unittest.cc
+++ b/modules/audio_processing/agc2/rnn_vad/rnn_fc_unittest.cc
@@ -13,7 +13,6 @@
 #include <array>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/agc2/cpu_features.h"
 #include "modules/audio_processing/agc2/rnn_vad/test_utils.h"
 #include "modules/audio_processing/test/performance_timer.h"
diff --git a/modules/audio_processing/agc2/rnn_vad/rnn_gru.cc b/modules/audio_processing/agc2/rnn_vad/rnn_gru.cc
index 5bd7119332..a31b48acf1 100644
--- a/modules/audio_processing/agc2/rnn_vad/rnn_gru.cc
+++ b/modules/audio_processing/agc2/rnn_vad/rnn_gru.cc
@@ -14,10 +14,10 @@
 #include <array>
 #include <cstddef>
 #include <cstdint>
+#include <span>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "modules/audio_processing/agc2/cpu_features.h"
 #include "modules/audio_processing/agc2/rnn_vad/vector_math.h"
 #include "rtc_base/checks.h"
@@ -31,7 +31,7 @@ namespace {
 
 constexpr int kNumGruGates = 3;  // Update, reset, output.
 
-std::vector<float> PreprocessGruTensor(ArrayView<const int8_t> tensor_src,
+std::vector<float> PreprocessGruTensor(std::span<const int8_t> tensor_src,
                                        int output_size) {
   // Transpose, cast and scale.
   // `n` is the size of the first dimension of the 3-dim tensor `weights`.
@@ -64,12 +64,12 @@ std::vector<float> PreprocessGruTensor(ArrayView<const int8_t> tensor_src,
 void ComputeUpdateResetGate(int input_size,
                             int output_size,
                             const VectorMath& vector_math,
-                            ArrayView<const float> input,
-                            ArrayView<const float> state,
-                            ArrayView<const float> bias,
-                            ArrayView<const float> weights,
-                            ArrayView<const float> recurrent_weights,
-                            ArrayView<float> gate) {
+                            std::span<const float> input,
+                            std::span<const float> state,
+                            std::span<const float> bias,
+                            std::span<const float> weights,
+                            std::span<const float> recurrent_weights,
+                            std::span<float> gate) {
   RTC_DCHECK_EQ(input.size(), input_size);
   RTC_DCHECK_EQ(state.size(), output_size);
   RTC_DCHECK_EQ(bias.size(), output_size);
@@ -100,13 +100,13 @@ void ComputeUpdateResetGate(int input_size,
 void ComputeStateGate(int input_size,
                       int output_size,
                       const VectorMath& vector_math,
-                      ArrayView<const float> input,
-                      ArrayView<const float> update,
-                      ArrayView<const float> reset,
-                      ArrayView<const float> bias,
-                      ArrayView<const float> weights,
-                      ArrayView<const float> recurrent_weights,
-                      ArrayView<float> state) {
+                      std::span<const float> input,
+                      std::span<const float> update,
+                      std::span<const float> reset,
+                      std::span<const float> bias,
+                      std::span<const float> weights,
+                      std::span<const float> recurrent_weights,
+                      std::span<float> state) {
   RTC_DCHECK_EQ(input.size(), input_size);
   RTC_DCHECK_GE(update.size(), output_size);  // `update` is over-allocated.
   RTC_DCHECK_GE(reset.size(), output_size);   // `reset` is over-allocated.
@@ -134,9 +134,9 @@ void ComputeStateGate(int input_size,
 GatedRecurrentLayer::GatedRecurrentLayer(
     const int input_size,
     const int output_size,
-    const ArrayView<const int8_t> bias,
-    const ArrayView<const int8_t> weights,
-    const ArrayView<const int8_t> recurrent_weights,
+    const std::span<const int8_t> bias,
+    const std::span<const int8_t> weights,
+    const std::span<const int8_t> recurrent_weights,
     const AvailableCpuFeatures& cpu_features,
     absl::string_view layer_name)
     : input_size_(input_size),
@@ -167,19 +167,19 @@ void GatedRecurrentLayer::Reset() {
   state_.fill(0.f);
 }
 
-void GatedRecurrentLayer::ComputeOutput(ArrayView<const float> input) {
+void GatedRecurrentLayer::ComputeOutput(std::span<const float> input) {
   RTC_DCHECK_EQ(input.size(), input_size_);
 
   // The tensors below are organized as a sequence of flattened tensors for the
   // `update`, `reset` and `state` gates.
-  ArrayView<const float> bias(bias_);
-  ArrayView<const float> weights(weights_);
-  ArrayView<const float> recurrent_weights(recurrent_weights_);
+  std::span<const float> bias(bias_);
+  std::span<const float> weights(weights_);
+  std::span<const float> recurrent_weights(recurrent_weights_);
   // Strides to access to the flattened tensors for a specific gate.
   const int stride_weights = input_size_ * output_size_;
   const int stride_recurrent_weights = output_size_ * output_size_;
 
-  ArrayView<float> state(state_.data(), output_size_);
+  std::span<float> state(state_.data(), output_size_);
 
   // Update gate.
   std::array<float, kGruLayerMaxUnits> update;
diff --git a/modules/audio_processing/agc2/rnn_vad/rnn_gru.h b/modules/audio_processing/agc2/rnn_vad/rnn_gru.h
index 367f3cda6b..502802dbd4 100644
--- a/modules/audio_processing/agc2/rnn_vad/rnn_gru.h
+++ b/modules/audio_processing/agc2/rnn_vad/rnn_gru.h
@@ -13,10 +13,10 @@
 
 #include <array>
 #include <cstdint>
+#include <span>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "modules/audio_processing/agc2/cpu_features.h"
 #include "modules/audio_processing/agc2/rnn_vad/vector_math.h"
 
@@ -33,9 +33,9 @@ class GatedRecurrentLayer {
   // Ctor. `output_size` cannot be greater than `kGruLayerMaxUnits`.
   GatedRecurrentLayer(int input_size,
                       int output_size,
-                      ArrayView<const int8_t> bias,
-                      ArrayView<const int8_t> weights,
-                      ArrayView<const int8_t> recurrent_weights,
+                      std::span<const int8_t> bias,
+                      std::span<const int8_t> weights,
+                      std::span<const int8_t> recurrent_weights,
                       const AvailableCpuFeatures& cpu_features,
                       absl::string_view layer_name);
   GatedRecurrentLayer(const GatedRecurrentLayer&) = delete;
@@ -50,14 +50,14 @@ class GatedRecurrentLayer {
   int size() const { return output_size_; }
 
   // Returns the output buffer.
-  ArrayView<const float> output() const {
-    return MakeArrayView(state_.data(), output_size_);
+  std::span<const float> output() const {
+    return std::span(state_.data(), output_size_);
   }
 
   // Resets the GRU state.
   void Reset();
   // Computes the recurrent layer output and updates the status.
-  void ComputeOutput(ArrayView<const float> input);
+  void ComputeOutput(std::span<const float> input);
 
  private:
   const int input_size_;
diff --git a/modules/audio_processing/agc2/rnn_vad/rnn_gru_unittest.cc b/modules/audio_processing/agc2/rnn_vad/rnn_gru_unittest.cc
index 738c2219e0..044bb3b12b 100644
--- a/modules/audio_processing/agc2/rnn_vad/rnn_gru_unittest.cc
+++ b/modules/audio_processing/agc2/rnn_vad/rnn_gru_unittest.cc
@@ -14,9 +14,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/agc2/cpu_features.h"
 #include "modules/audio_processing/agc2/rnn_vad/test_utils.h"
 #include "modules/audio_processing/test/performance_timer.h"
@@ -31,8 +31,8 @@ namespace rnn_vad {
 namespace {
 
 void TestGatedRecurrentLayer(GatedRecurrentLayer& gru,
-                             ArrayView<const float> input_sequence,
-                             ArrayView<const float> expected_output_sequence) {
+                             std::span<const float> input_sequence,
+                             std::span<const float> expected_output_sequence) {
   const int input_sequence_length = CheckedDivExact(
       dchecked_cast<int>(input_sequence.size()), gru.input_size());
   const int output_sequence_length = CheckedDivExact(
@@ -138,7 +138,7 @@ TEST_P(RnnGruParametrization, DISABLED_BenchmarkGatedRecurrentLayer) {
                           /*cpu_features=*/GetParam(),
                           /*layer_name=*/"GRU");
 
-  ArrayView<const float> input_sequence(gru_input_sequence);
+  std::span<const float> input_sequence(gru_input_sequence);
   ASSERT_EQ(input_sequence.size() % kInputLayerOutputSize,
             static_cast<size_t>(0));
   const int input_sequence_length =
diff --git a/modules/audio_processing/agc2/rnn_vad/rnn_unittest.cc b/modules/audio_processing/agc2/rnn_vad/rnn_unittest.cc
index a944e2b933..b5a8a7969b 100644
--- a/modules/audio_processing/agc2/rnn_vad/rnn_unittest.cc
+++ b/modules/audio_processing/agc2/rnn_vad/rnn_unittest.cc
@@ -12,7 +12,6 @@
 
 #include <array>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/agc2/cpu_features.h"
 #include "modules/audio_processing/agc2/rnn_vad/common.h"
 #include "test/gtest.h"
diff --git a/modules/audio_processing/agc2/rnn_vad/rnn_vad_unittest.cc b/modules/audio_processing/agc2/rnn_vad/rnn_vad_unittest.cc
index 37e319743c..7ab9ce1b8a 100644
--- a/modules/audio_processing/agc2/rnn_vad/rnn_vad_unittest.cc
+++ b/modules/audio_processing/agc2/rnn_vad/rnn_vad_unittest.cc
@@ -11,9 +11,9 @@
 #include <array>
 #include <cstdlib>
 #include <memory>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "common_audio/resampler/push_sinc_resampler.h"
 #include "modules/audio_processing/agc2/cpu_features.h"
 #include "modules/audio_processing/agc2/rnn_vad/common.h"
@@ -141,7 +141,7 @@ TEST_P(RnnVadProbabilityParametrization, DISABLED_RnnVadPerformance) {
     perf_timer.StartTimer();
     for (int i = 0; i < num_frames; ++i) {
       bool is_silence = features_extractor.CheckSilenceComputeFeatures(
-          ArrayView<const float, kFrameSize10ms24kHz>(
+          std::span<const float, kFrameSize10ms24kHz>(
               &prefetched_decimated_samples[i * kFrameSize10ms24kHz],
               kFrameSize10ms24kHz),
           feature_vector);
diff --git a/modules/audio_processing/agc2/rnn_vad/sequence_buffer.h b/modules/audio_processing/agc2/rnn_vad/sequence_buffer.h
index 6870f951d1..9526d6e6bb 100644
--- a/modules/audio_processing/agc2/rnn_vad/sequence_buffer.h
+++ b/modules/audio_processing/agc2/rnn_vad/sequence_buffer.h
@@ -13,10 +13,10 @@
 
 #include <algorithm>
 #include <cstring>
+#include <span>
 #include <type_traits>
 #include <vector>
 
-#include "api/array_view.h"
 #include "rtc_base/checks.h"
 
 namespace webrtc {
@@ -50,18 +50,18 @@ class SequenceBuffer {
   // Sets the sequence buffer values to zero.
   void Reset() { std::fill(buffer_.begin(), buffer_.end(), 0); }
   // Returns a view on the whole buffer.
-  ArrayView<const T, S> GetBufferView() const {
-    return ArrayView<const T, S>(buffer_.data(), S);
+  std::span<const T, S> GetBufferView() const {
+    return std::span<const T, S>(buffer_.data(), S);
   }
   // Returns a view on the M most recent values of the buffer.
-  ArrayView<const T, M> GetMostRecentValuesView() const {
+  std::span<const T, M> GetMostRecentValuesView() const {
     static_assert(M <= S,
                   "The number of most recent values cannot be larger than the "
                   "sequence buffer size.");
-    return ArrayView<const T, M>(buffer_.data() + S - M, M);
+    return std::span<const T, M>(buffer_.data() + S - M, M);
   }
   // Shifts left the buffer by N items and add new N items at the end.
-  void Push(ArrayView<const T, N> new_values) {
+  void Push(std::span<const T, N> new_values) {
     // Make space for the new values.
     if (S > N)
       std::memmove(buffer_.data(), buffer_.data() + N, (S - N) * sizeof(T));
diff --git a/modules/audio_processing/agc2/rnn_vad/spectral_features.cc b/modules/audio_processing/agc2/rnn_vad/spectral_features.cc
index 1712c7d933..ea4dcaf9bb 100644
--- a/modules/audio_processing/agc2/rnn_vad/spectral_features.cc
+++ b/modules/audio_processing/agc2/rnn_vad/spectral_features.cc
@@ -15,8 +15,8 @@
 #include <cmath>
 #include <limits>
 #include <numeric>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/agc2/rnn_vad/common.h"
 #include "modules/audio_processing/agc2/rnn_vad/ring_buffer.h"
 #include "modules/audio_processing/agc2/rnn_vad/spectral_features_internal.h"
@@ -34,7 +34,7 @@ constexpr float kSilenceThreshold = 0.04f;
 // Computes the new cepstral difference stats and pushes them into the passed
 // symmetric matrix buffer.
 void UpdateCepstralDifferenceStats(
-    ArrayView<const float, kNumBands> new_cepstral_coeffs,
+    std::span<const float, kNumBands> new_cepstral_coeffs,
     const RingBuffer<float, kNumBands, kCepstralCoeffsHistorySize>& ring_buf,
     SymmetricMatrixBuffer<float, kCepstralCoeffsHistorySize>* sym_matrix_buf) {
   RTC_DCHECK(sym_matrix_buf);
@@ -71,7 +71,7 @@ std::array<float, kFrameSize20ms24kHz / 2> ComputeScaledHalfVorbisWindow(
 // applied. The Fourier coefficient corresponding to the Nyquist frequency is
 // set to zero (it is never used and this allows to simplify the code).
 void ComputeWindowedForwardFft(
-    ArrayView<const float, kFrameSize20ms24kHz> frame,
+    std::span<const float, kFrameSize20ms24kHz> frame,
     const std::array<float, kFrameSize20ms24kHz / 2>& half_window,
     Pffft::FloatBuffer* fft_input_buffer,
     Pffft::FloatBuffer* fft_output_buffer,
@@ -109,13 +109,13 @@ void SpectralFeaturesExtractor::Reset() {
 }
 
 bool SpectralFeaturesExtractor::CheckSilenceComputeFeatures(
-    ArrayView<const float, kFrameSize20ms24kHz> reference_frame,
-    ArrayView<const float, kFrameSize20ms24kHz> lagged_frame,
-    ArrayView<float, kNumBands - kNumLowerBands> higher_bands_cepstrum,
-    ArrayView<float, kNumLowerBands> average,
-    ArrayView<float, kNumLowerBands> first_derivative,
-    ArrayView<float, kNumLowerBands> second_derivative,
-    ArrayView<float, kNumLowerBands> bands_cross_corr,
+    std::span<const float, kFrameSize20ms24kHz> reference_frame,
+    std::span<const float, kFrameSize20ms24kHz> lagged_frame,
+    std::span<float, kNumBands - kNumLowerBands> higher_bands_cepstrum,
+    std::span<float, kNumLowerBands> average,
+    std::span<float, kNumLowerBands> first_derivative,
+    std::span<float, kNumLowerBands> second_derivative,
+    std::span<float, kNumLowerBands> bands_cross_corr,
     float* variability) {
   // Compute the Opus band energies for the reference frame.
   ComputeWindowedForwardFft(reference_frame, half_window_, fft_buffer_.get(),
@@ -161,9 +161,9 @@ bool SpectralFeaturesExtractor::CheckSilenceComputeFeatures(
 }
 
 void SpectralFeaturesExtractor::ComputeAvgAndDerivatives(
-    ArrayView<float, kNumLowerBands> average,
-    ArrayView<float, kNumLowerBands> first_derivative,
-    ArrayView<float, kNumLowerBands> second_derivative) const {
+    std::span<float, kNumLowerBands> average,
+    std::span<float, kNumLowerBands> first_derivative,
+    std::span<float, kNumLowerBands> second_derivative) const {
   auto curr = cepstral_coeffs_ring_buf_.GetArrayView(0);
   auto prev1 = cepstral_coeffs_ring_buf_.GetArrayView(1);
   auto prev2 = cepstral_coeffs_ring_buf_.GetArrayView(2);
@@ -181,7 +181,7 @@ void SpectralFeaturesExtractor::ComputeAvgAndDerivatives(
 }
 
 void SpectralFeaturesExtractor::ComputeNormalizedCepstralCorrelation(
-    ArrayView<float, kNumLowerBands> bands_cross_corr) {
+    std::span<float, kNumLowerBands> bands_cross_corr) {
   spectral_correlator_.ComputeCrossCorrelation(
       reference_frame_fft_->GetConstView(), lagged_frame_fft_->GetConstView(),
       bands_cross_corr_);
diff --git a/modules/audio_processing/agc2/rnn_vad/spectral_features.h b/modules/audio_processing/agc2/rnn_vad/spectral_features.h
index 6994e451d2..fe15f4b70a 100644
--- a/modules/audio_processing/agc2/rnn_vad/spectral_features.h
+++ b/modules/audio_processing/agc2/rnn_vad/spectral_features.h
@@ -14,8 +14,8 @@
 #include <array>
 #include <cstddef>
 #include <memory>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/agc2/rnn_vad/common.h"
 #include "modules/audio_processing/agc2/rnn_vad/ring_buffer.h"
 #include "modules/audio_processing/agc2/rnn_vad/spectral_features_internal.h"
@@ -39,22 +39,22 @@ class SpectralFeaturesExtractor {
   // detects silence and computes features. If silence is detected, the output
   // is neither computed nor written.
   bool CheckSilenceComputeFeatures(
-      ArrayView<const float, kFrameSize20ms24kHz> reference_frame,
-      ArrayView<const float, kFrameSize20ms24kHz> lagged_frame,
-      ArrayView<float, kNumBands - kNumLowerBands> higher_bands_cepstrum,
-      ArrayView<float, kNumLowerBands> average,
-      ArrayView<float, kNumLowerBands> first_derivative,
-      ArrayView<float, kNumLowerBands> second_derivative,
-      ArrayView<float, kNumLowerBands> bands_cross_corr,
+      std::span<const float, kFrameSize20ms24kHz> reference_frame,
+      std::span<const float, kFrameSize20ms24kHz> lagged_frame,
+      std::span<float, kNumBands - kNumLowerBands> higher_bands_cepstrum,
+      std::span<float, kNumLowerBands> average,
+      std::span<float, kNumLowerBands> first_derivative,
+      std::span<float, kNumLowerBands> second_derivative,
+      std::span<float, kNumLowerBands> bands_cross_corr,
       float* variability);
 
  private:
   void ComputeAvgAndDerivatives(
-      ArrayView<float, kNumLowerBands> average,
-      ArrayView<float, kNumLowerBands> first_derivative,
-      ArrayView<float, kNumLowerBands> second_derivative) const;
+      std::span<float, kNumLowerBands> average,
+      std::span<float, kNumLowerBands> first_derivative,
+      std::span<float, kNumLowerBands> second_derivative) const;
   void ComputeNormalizedCepstralCorrelation(
-      ArrayView<float, kNumLowerBands> bands_cross_corr);
+      std::span<float, kNumLowerBands> bands_cross_corr);
   float ComputeVariability() const;
 
   const std::array<float, kFrameSize20ms24kHz / 2> half_window_;
diff --git a/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.cc b/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.cc
index 7cbdcecb91..d370ac7586 100644
--- a/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.cc
+++ b/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.cc
@@ -14,8 +14,8 @@
 #include <array>
 #include <cmath>
 #include <cstddef>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/agc2/rnn_vad/common.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/numerics/safe_compare.h"
@@ -95,15 +95,15 @@ SpectralCorrelator::SpectralCorrelator()
 SpectralCorrelator::~SpectralCorrelator() = default;
 
 void SpectralCorrelator::ComputeAutoCorrelation(
-    ArrayView<const float> x,
-    ArrayView<float, kOpusBands24kHz> auto_corr) const {
+    std::span<const float> x,
+    std::span<float, kOpusBands24kHz> auto_corr) const {
   ComputeCrossCorrelation(x, x, auto_corr);
 }
 
 void SpectralCorrelator::ComputeCrossCorrelation(
-    ArrayView<const float> x,
-    ArrayView<const float> y,
-    ArrayView<float, kOpusBands24kHz> cross_corr) const {
+    std::span<const float> x,
+    std::span<const float> y,
+    std::span<float, kOpusBands24kHz> cross_corr) const {
   RTC_DCHECK_EQ(x.size(), kFrameSize20ms24kHz);
   RTC_DCHECK_EQ(x.size(), y.size());
   RTC_DCHECK_EQ(x[1], 0.f) << "The Nyquist coefficient must be zeroed.";
@@ -126,8 +126,8 @@ void SpectralCorrelator::ComputeCrossCorrelation(
 }
 
 void ComputeSmoothedLogMagnitudeSpectrum(
-    ArrayView<const float> bands_energy,
-    ArrayView<float, kNumBands> log_bands_energy) {
+    std::span<const float> bands_energy,
+    std::span<float, kNumBands> log_bands_energy) {
   RTC_DCHECK_LE(bands_energy.size(), kNumBands);
   constexpr float kOneByHundred = 1e-2f;
   constexpr float kLogOneByHundred = -2.f;
@@ -161,9 +161,9 @@ std::array<float, kNumBands * kNumBands> ComputeDctTable() {
   return dct_table;
 }
 
-void ComputeDct(ArrayView<const float> in,
-                ArrayView<const float, kNumBands * kNumBands> dct_table,
-                ArrayView<float> out) {
+void ComputeDct(std::span<const float> in,
+                std::span<const float, kNumBands * kNumBands> dct_table,
+                std::span<float> out) {
   // DCT scaling factor - i.e., sqrt(2 / kNumBands).
   constexpr float kDctScalingFactor = 0.301511345f;
   constexpr float kDctScalingFactorError =
diff --git a/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.h b/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.h
index 4d9fd52b97..4434291dff 100644
--- a/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.h
+++ b/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.h
@@ -14,9 +14,9 @@
 #include <stddef.h>
 
 #include <array>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/agc2/rnn_vad/common.h"
 
 namespace webrtc {
@@ -55,8 +55,8 @@ class SpectralCorrelator {
   //  - be encoded as vectors of interleaved real-complex FFT coefficients
   //    where x[1] = y[1] = 0 (the Nyquist frequency coefficient is omitted).
   void ComputeAutoCorrelation(
-      ArrayView<const float> x,
-      ArrayView<float, kOpusBands24kHz> auto_corr) const;
+      std::span<const float> x,
+      std::span<float, kOpusBands24kHz> auto_corr) const;
 
   // Computes the band-wise spectral cross-correlations.
   // `x` and `y` must:
@@ -64,9 +64,9 @@ class SpectralCorrelator {
   //  - be encoded as vectors of interleaved real-complex FFT coefficients where
   //    x[1] = y[1] = 0 (the Nyquist frequency coefficient is omitted).
   void ComputeCrossCorrelation(
-      ArrayView<const float> x,
-      ArrayView<const float> y,
-      ArrayView<float, kOpusBands24kHz> cross_corr) const;
+      std::span<const float> x,
+      std::span<const float> y,
+      std::span<float, kOpusBands24kHz> cross_corr) const;
 
  private:
   const std::vector<float> weights_;  // Weights for each Fourier coefficient.
@@ -77,8 +77,8 @@ class SpectralCorrelator {
 // computes the log magnitude spectrum applying smoothing both over time and
 // over frequency. Declared here for unit testing.
 void ComputeSmoothedLogMagnitudeSpectrum(
-    ArrayView<const float> bands_energy,
-    ArrayView<float, kNumBands> log_bands_energy);
+    std::span<const float> bands_energy,
+    std::span<float, kNumBands> log_bands_energy);
 
 // TODO(bugs.webrtc.org/10480): Move to anonymous namespace in
 // spectral_features.cc. Creates a DCT table for arrays having size equal to
@@ -90,9 +90,9 @@ std::array<float, kNumBands * kNumBands> ComputeDctTable();
 // In-place computation is not allowed and `out` can be smaller than `in` in
 // order to only compute the first DCT coefficients. Declared here for unit
 // testing.
-void ComputeDct(ArrayView<const float> in,
-                ArrayView<const float, kNumBands * kNumBands> dct_table,
-                ArrayView<float> out);
+void ComputeDct(std::span<const float> in,
+                std::span<const float, kNumBands * kNumBands> dct_table,
+                std::span<float> out);
 
 }  // namespace rnn_vad
 }  // namespace webrtc
diff --git a/modules/audio_processing/agc2/rnn_vad/spectral_features_internal_unittest.cc b/modules/audio_processing/agc2/rnn_vad/spectral_features_internal_unittest.cc
index 668a4eef2d..63bf3cf491 100644
--- a/modules/audio_processing/agc2/rnn_vad/spectral_features_internal_unittest.cc
+++ b/modules/audio_processing/agc2/rnn_vad/spectral_features_internal_unittest.cc
@@ -13,9 +13,9 @@
 #include <algorithm>
 #include <array>
 #include <numeric>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/agc2/rnn_vad/common.h"
 #include "modules/audio_processing/agc2/rnn_vad/test_utils.h"
 #include "modules/audio_processing/utility/pffft_wrapper.h"
@@ -73,7 +73,7 @@ TEST(RnnVadTest, DISABLED_TestOpusScaleWeights) {
   int i = 0;
   for (int band_size : GetOpusScaleNumBins24kHz20ms()) {
     SCOPED_TRACE(band_size);
-    ArrayView<float> band_weights(weights.data() + i, band_size);
+    std::span<float> band_weights(weights.data() + i, band_size);
     float prev = -1.f;
     for (float weight : band_weights) {
       EXPECT_LT(prev, weight);
diff --git a/modules/audio_processing/agc2/rnn_vad/spectral_features_unittest.cc b/modules/audio_processing/agc2/rnn_vad/spectral_features_unittest.cc
index 7c9403391d..6a01ab02d1 100644
--- a/modules/audio_processing/agc2/rnn_vad/spectral_features_unittest.cc
+++ b/modules/audio_processing/agc2/rnn_vad/spectral_features_unittest.cc
@@ -12,8 +12,8 @@
 
 #include <algorithm>
 #include <array>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/agc2/rnn_vad/common.h"
 #include "modules/audio_processing/agc2/rnn_vad/test_utils.h"
 #include "rtc_base/numerics/safe_compare.h"
@@ -28,40 +28,40 @@ namespace {
 constexpr int kTestFeatureVectorSize = kNumBands + 3 * kNumLowerBands + 1;
 
 // Writes non-zero sample values.
-void WriteTestData(ArrayView<float> samples) {
+void WriteTestData(std::span<float> samples) {
   for (int i = 0; SafeLt(i, samples.size()); ++i) {
     samples[i] = i % 100;
   }
 }
 
-ArrayView<float, kNumBands - kNumLowerBands> GetHigherBandsSpectrum(
-    ArrayView<float, kTestFeatureVectorSize> feature_vector) {
+std::span<float, kNumBands - kNumLowerBands> GetHigherBandsSpectrum(
+    std::span<float, kTestFeatureVectorSize> feature_vector) {
   return feature_vector.subspan<kNumLowerBands, kNumBands - kNumLowerBands>();
 }
 
-ArrayView<float, kNumLowerBands> GetAverage(
-    ArrayView<float, kTestFeatureVectorSize> feature_vector) {
+std::span<float, kNumLowerBands> GetAverage(
+    std::span<float, kTestFeatureVectorSize> feature_vector) {
   return feature_vector.first<kNumLowerBands>();
 }
 
-ArrayView<float, kNumLowerBands> GetFirstDerivative(
-    ArrayView<float, kTestFeatureVectorSize> feature_vector) {
+std::span<float, kNumLowerBands> GetFirstDerivative(
+    std::span<float, kTestFeatureVectorSize> feature_vector) {
   return feature_vector.subspan<kNumBands, kNumLowerBands>();
 }
 
-ArrayView<float, kNumLowerBands> GetSecondDerivative(
-    ArrayView<float, kTestFeatureVectorSize> feature_vector) {
+std::span<float, kNumLowerBands> GetSecondDerivative(
+    std::span<float, kTestFeatureVectorSize> feature_vector) {
   return feature_vector.subspan<kNumBands + kNumLowerBands, kNumLowerBands>();
 }
 
-ArrayView<float, kNumLowerBands> GetCepstralCrossCorrelation(
-    ArrayView<float, kTestFeatureVectorSize> feature_vector) {
+std::span<float, kNumLowerBands> GetCepstralCrossCorrelation(
+    std::span<float, kTestFeatureVectorSize> feature_vector) {
   return feature_vector
       .subspan<kNumBands + 2 * kNumLowerBands, kNumLowerBands>();
 }
 
 float* GetCepstralVariability(
-    ArrayView<float, kTestFeatureVectorSize> feature_vector) {
+    std::span<float, kTestFeatureVectorSize> feature_vector) {
   return &feature_vector[kNumBands + 3 * kNumLowerBands];
 }
 
@@ -73,7 +73,7 @@ TEST(RnnVadTest, SpectralFeaturesWithAndWithoutSilence) {
   // Initialize.
   SpectralFeaturesExtractor sfe;
   std::array<float, kFrameSize20ms24kHz> samples;
-  ArrayView<float, kFrameSize20ms24kHz> samples_view(samples);
+  std::span<float, kFrameSize20ms24kHz> samples_view(samples);
   bool is_silence;
   std::array<float, kTestFeatureVectorSize> feature_vector;
 
@@ -118,7 +118,7 @@ TEST(RnnVadTest, CepstralFeaturesConstantAverageZeroDerivative) {
   // Initialize.
   SpectralFeaturesExtractor sfe;
   std::array<float, kFrameSize20ms24kHz> samples;
-  ArrayView<float, kFrameSize20ms24kHz> samples_view(samples);
+  std::span<float, kFrameSize20ms24kHz> samples_view(samples);
   WriteTestData(samples);
 
   // Fill the spectral features with test data.
diff --git a/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer.h b/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer.h
index bd1ace8f5e..ad189db06d 100644
--- a/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer.h
+++ b/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer.h
@@ -14,10 +14,10 @@
 #include <algorithm>
 #include <array>
 #include <cstring>
+#include <span>
 #include <type_traits>
 #include <utility>
 
-#include "api/array_view.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/numerics/safe_compare.h"
 
@@ -52,7 +52,7 @@ class SymmetricMatrixBuffer {
   // most recent one in the ring buffer, whereas the last element in `values`
   // must correspond to the comparison between the most recent item and the
   // oldest one in the ring buffer.
-  void Push(ArrayView<T, S - 1> values) {
+  void Push(std::span<T, S - 1> values) {
     // Move the lower-right sub-matrix of size (S-2) x (S-2) one row up and one
     // column left.
     std::memmove(buf_.data(), buf_.data() + S, (buf_.size() - S) * sizeof(T));
diff --git a/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer_unittest.cc b/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer_unittest.cc
index 43ca270586..8c45d01ef7 100644
--- a/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer_unittest.cc
+++ b/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer_unittest.cc
@@ -12,9 +12,9 @@
 
 #include <algorithm>
 #include <array>
+#include <span>
 #include <utility>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/agc2/rnn_vad/ring_buffer.h"
 #include "test/gtest.h"
 
@@ -62,7 +62,7 @@ TEST(RnnVadTest, SymmetricMatrixBufferUseCase) {
   for (int t = 1; t <= 100; ++t) {  // Evolution steps.
     SCOPED_TRACE(t);
     const int t_removed = ring_buf.GetArrayView(kRingBufSize - 1)[0];
-    ring_buf.Push(ArrayView<int, 1>(&t, 1));
+    ring_buf.Push(std::span<int, 1>(&t, 1));
     // The head of the ring buffer is `t`.
     ASSERT_EQ(t, ring_buf.GetArrayView(0)[0]);
     // Create the comparisons between `t` and the older elements in the ring
diff --git a/modules/audio_processing/agc2/rnn_vad/test_utils.cc b/modules/audio_processing/agc2/rnn_vad/test_utils.cc
index de3045005d..251a425b81 100644
--- a/modules/audio_processing/agc2/rnn_vad/test_utils.cc
+++ b/modules/audio_processing/agc2/rnn_vad/test_utils.cc
@@ -15,13 +15,13 @@
 #include <fstream>
 #include <ios>
 #include <memory>
+#include <span>
 #include <string>
 #include <type_traits>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "modules/audio_processing/agc2/rnn_vad/common.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/numerics/safe_compare.h"
@@ -49,7 +49,7 @@ class FloatFileReader : public FileReader {
   ~FloatFileReader() override = default;
 
   int size() const override { return size_; }
-  bool ReadChunk(ArrayView<float> dst) override {
+  bool ReadChunk(std::span<float> dst) override {
     const std::streamsize bytes_to_read = dst.size() * sizeof(T);
     if (std::is_same<T, float>::value) {
       is_.read(reinterpret_cast<char*>(dst.data()), bytes_to_read);
@@ -77,8 +77,8 @@ class FloatFileReader : public FileReader {
 
 using test::ResourcePath;
 
-void ExpectEqualFloatArray(ArrayView<const float> expected,
-                           ArrayView<const float> computed) {
+void ExpectEqualFloatArray(std::span<const float> expected,
+                           std::span<const float> computed) {
   ASSERT_EQ(expected.size(), computed.size());
   for (int i = 0; SafeLt(i, expected.size()); ++i) {
     SCOPED_TRACE(i);
@@ -86,8 +86,8 @@ void ExpectEqualFloatArray(ArrayView<const float> expected,
   }
 }
 
-void ExpectNearAbsolute(ArrayView<const float> expected,
-                        ArrayView<const float> computed,
+void ExpectNearAbsolute(std::span<const float> expected,
+                        std::span<const float> computed,
                         float tolerance) {
   ASSERT_EQ(expected.size(), computed.size());
   for (int i = 0; SafeLt(i, expected.size()); ++i) {
diff --git a/modules/audio_processing/agc2/rnn_vad/test_utils.h b/modules/audio_processing/agc2/rnn_vad/test_utils.h
index 4345b04dd1..f8b4ff8c8c 100644
--- a/modules/audio_processing/agc2/rnn_vad/test_utils.h
+++ b/modules/audio_processing/agc2/rnn_vad/test_utils.h
@@ -16,10 +16,10 @@
 #include <ios>
 #include <limits>
 #include <memory>
+#include <span>
 #include <string>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "modules/audio_processing/agc2/rnn_vad/common.h"
 
 namespace webrtc {
@@ -27,15 +27,15 @@ namespace rnn_vad {
 
 constexpr float kFloatMin = std::numeric_limits<float>::min();
 
-// Fails for every pair from two equally sized webrtc::ArrayView<float> views
+// Fails for every pair from two equally sized std::span<float> views
 // such that the values in the pair do not match.
-void ExpectEqualFloatArray(ArrayView<const float> expected,
-                           ArrayView<const float> computed);
+void ExpectEqualFloatArray(std::span<const float> expected,
+                           std::span<const float> computed);
 
-// Fails for every pair from two equally sized webrtc::ArrayView<float> views
+// Fails for every pair from two equally sized std::span<float> views
 // such that their absolute error is above a given threshold.
-void ExpectNearAbsolute(ArrayView<const float> expected,
-                        ArrayView<const float> computed,
+void ExpectNearAbsolute(std::span<const float> expected,
+                        std::span<const float> computed,
                         float tolerance);
 
 // File reader interface.
@@ -49,7 +49,7 @@ class FileReader {
   // values are correctly read. If the number of remaining bytes in the file is
   // not sufficient to read `dst.size()` float values, `dst` is partially
   // modified and false is returned.
-  virtual bool ReadChunk(ArrayView<float> dst) = 0;
+  virtual bool ReadChunk(std::span<float> dst) = 0;
   // Reads a single float value, advances the internal file position according
   // to the number of read bytes and returns true if the value is correctly
   // read. If the number of remaining bytes in the file is not sufficient to
@@ -90,13 +90,13 @@ class PitchTestData {
  public:
   PitchTestData();
   ~PitchTestData();
-  ArrayView<const float, kBufSize24kHz> PitchBuffer24kHzView() const {
+  std::span<const float, kBufSize24kHz> PitchBuffer24kHzView() const {
     return pitch_buffer_24k_;
   }
-  ArrayView<const float, kRefineNumLags24kHz> SquareEnergies24kHzView() const {
+  std::span<const float, kRefineNumLags24kHz> SquareEnergies24kHzView() const {
     return square_energies_24k_;
   }
-  ArrayView<const float, kNumLags12kHz> AutoCorrelation12kHzView() const {
+  std::span<const float, kNumLags12kHz> AutoCorrelation12kHzView() const {
     return auto_correlation_12k_;
   }
 
@@ -114,7 +114,7 @@ class FileWriter {
   FileWriter(const FileWriter&) = delete;
   FileWriter& operator=(const FileWriter&) = delete;
   ~FileWriter() = default;
-  void WriteChunk(ArrayView<const float> value) {
+  void WriteChunk(std::span<const float> value) {
     const std::streamsize bytes_to_write = value.size() * sizeof(float);
     os_.write(reinterpret_cast<const char*>(value.data()), bytes_to_write);
   }
diff --git a/modules/audio_processing/agc2/rnn_vad/vector_math.h b/modules/audio_processing/agc2/rnn_vad/vector_math.h
index 0cf4e34efb..0bfe5af3f3 100644
--- a/modules/audio_processing/agc2/rnn_vad/vector_math.h
+++ b/modules/audio_processing/agc2/rnn_vad/vector_math.h
@@ -22,8 +22,8 @@
 #endif
 
 #include <numeric>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/agc2/cpu_features.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/numerics/safe_conversions.h"
@@ -40,7 +40,7 @@ class VectorMath {
       : cpu_features_(cpu_features) {}
 
   // Computes the dot product between two equally sized vectors.
-  float DotProduct(ArrayView<const float> x, ArrayView<const float> y) const {
+  float DotProduct(std::span<const float> x, std::span<const float> y) const {
     RTC_DCHECK_EQ(x.size(), y.size());
 #if defined(WEBRTC_ARCH_X86_FAMILY)
     if (cpu_features_.avx2) {
@@ -101,8 +101,8 @@ class VectorMath {
   }
 
  private:
-  float DotProductAvx2(ArrayView<const float> x,
-                       ArrayView<const float> y) const;
+  float DotProductAvx2(std::span<const float> x,
+                       std::span<const float> y) const;
 
   const AvailableCpuFeatures cpu_features_;
 };
diff --git a/modules/audio_processing/agc2/rnn_vad/vector_math_avx2.cc b/modules/audio_processing/agc2/rnn_vad/vector_math_avx2.cc
index 71466dab20..98c4fdcadf 100644
--- a/modules/audio_processing/agc2/rnn_vad/vector_math_avx2.cc
+++ b/modules/audio_processing/agc2/rnn_vad/vector_math_avx2.cc
@@ -10,7 +10,8 @@
 
 #include <immintrin.h>
 
-#include "api/array_view.h"
+#include <span>
+
 #include "modules/audio_processing/agc2/rnn_vad/vector_math.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/numerics/safe_conversions.h"
@@ -18,8 +19,8 @@
 namespace webrtc {
 namespace rnn_vad {
 
-float VectorMath::DotProductAvx2(ArrayView<const float> x,
-                                 ArrayView<const float> y) const {
+float VectorMath::DotProductAvx2(std::span<const float> x,
+                                 std::span<const float> y) const {
   RTC_DCHECK(cpu_features_.avx2);
   RTC_DCHECK_EQ(x.size(), y.size());
   __m256 accumulator = _mm256_setzero_ps();
diff --git a/modules/audio_processing/agc2/vad_wrapper_unittest.cc b/modules/audio_processing/agc2/vad_wrapper_unittest.cc
index 7d51591456..e159354925 100644
--- a/modules/audio_processing/agc2/vad_wrapper_unittest.cc
+++ b/modules/audio_processing/agc2/vad_wrapper_unittest.cc
@@ -12,11 +12,11 @@
 
 #include <limits>
 #include <memory>
+#include <span>
 #include <tuple>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/audio_view.h"
 #include "modules/audio_processing/agc2/agc2_common.h"
 #include "rtc_base/checks.h"
@@ -43,7 +43,7 @@ class MockVad : public VoiceActivityDetectorWrapper::MonoVad {
  public:
   MOCK_METHOD(int, SampleRateHz, (), (const, override));
   MOCK_METHOD(void, Reset, (), (override));
-  MOCK_METHOD(float, Analyze, (ArrayView<const float> frame), (override));
+  MOCK_METHOD(float, Analyze, (std::span<const float> frame), (override));
 };
 
 // Checks that the ctor and `Initialize()` read the sample rate of the wrapped
@@ -159,7 +159,7 @@ TEST_P(VadResamplingParametrization, CheckResampledFrameSize) {
       .Times(AnyNumber())
       .WillRepeatedly(Return(vad_sample_rate_hz()));
   EXPECT_CALL(*vad, Reset).Times(1);
-  EXPECT_CALL(*vad, Analyze(Truly([this](ArrayView<const float> frame) {
+  EXPECT_CALL(*vad, Analyze(Truly([this](std::span<const float> frame) {
     return SafeEq(frame.size(),
                   CheckedDivExact(vad_sample_rate_hz(), kNumFramesPerSecond));
   }))).Times(1);
diff --git a/modules/audio_processing/audio_processing_impl.cc b/modules/audio_processing/audio_processing_impl.cc
index e86cc3cf69..89d537f6a6 100644
--- a/modules/audio_processing/audio_processing_impl.cc
+++ b/modules/audio_processing/audio_processing_impl.cc
@@ -18,13 +18,13 @@
 #include <cstring>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/base/nullability.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio/audio_processing.h"
 #include "api/audio/audio_processing_statistics.h"
 #include "api/audio/audio_view.h"
@@ -1248,7 +1248,7 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
         *capture_buffer);
   }
 
-  capture_input_rms_.Analyze(ArrayView<const float>(
+  capture_input_rms_.Analyze(std::span<const float>(
       capture_buffer->channels_const()[0],
       capture_nonlocked_.capture_processing_format.num_frames()));
   const bool log_rms = ++capture_rms_interval_counter_ >= 1000;
@@ -1395,7 +1395,7 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
     }
 
     if (submodules_.echo_detector) {
-      submodules_.echo_detector->AnalyzeCaptureAudio(ArrayView<const float>(
+      submodules_.echo_detector->AnalyzeCaptureAudio(std::span<const float>(
           capture_buffer->channels()[0], capture_buffer->num_frames()));
     }
 
@@ -1419,7 +1419,7 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
       submodules_.capture_post_processor->Process(capture_buffer);
     }
 
-    capture_output_rms_.Analyze(ArrayView<const float>(
+    capture_output_rms_.Analyze(std::span<const float>(
         capture_buffer->channels_const()[0],
         capture_nonlocked_.capture_processing_format.num_frames()));
     if (log_rms) {
@@ -1479,7 +1479,7 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
   if (!capture_.capture_output_used_last_frame &&
       capture_.capture_output_used) {
     for (size_t ch = 0; ch < capture_buffer->num_channels(); ++ch) {
-      ArrayView<float> channel_view(capture_buffer->channels()[ch],
+      std::span<float> channel_view(capture_buffer->channels()[ch],
                                     capture_buffer->num_frames());
       std::fill(channel_view.begin(), channel_view.end(), 0.f);
     }
@@ -1642,7 +1642,7 @@ int AudioProcessingImpl::set_stream_delay_ms(int delay) {
 }
 
 bool AudioProcessingImpl::GetLinearAecOutput(
-    ArrayView<std::array<float, 160>> linear_output) const {
+    std::span<std::array<float, 160>> linear_output) const {
   MutexLock lock(&mutex_capture_);
   AudioBuffer* linear_aec_buffer = capture_.linear_aec_output.get();
 
@@ -1653,8 +1653,8 @@ bool AudioProcessingImpl::GetLinearAecOutput(
 
     for (size_t ch = 0; ch < linear_aec_buffer->num_channels(); ++ch) {
       RTC_DCHECK_EQ(linear_output[ch].size(), linear_aec_buffer->num_frames());
-      ArrayView<const float> channel_view =
-          ArrayView<const float>(linear_aec_buffer->channels_const()[ch],
+      std::span<const float> channel_view =
+          std::span<const float>(linear_aec_buffer->channels_const()[ch],
                                  linear_aec_buffer->num_frames());
       FloatS16ToFloat(channel_view.data(), channel_view.size(),
                       linear_output[ch].data());
diff --git a/modules/audio_processing/audio_processing_impl.h b/modules/audio_processing/audio_processing_impl.h
index c058961472..2d50d8d041 100644
--- a/modules/audio_processing/audio_processing_impl.h
+++ b/modules/audio_processing/audio_processing_impl.h
@@ -17,12 +17,12 @@
 #include <cstdio>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
 #include "absl/base/nullability.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio/audio_processing.h"
 #include "api/audio/audio_processing_statistics.h"
 #include "api/audio/echo_canceller3_config.h"
@@ -104,7 +104,7 @@ class AudioProcessingImpl : public AudioProcessing {
                     const StreamConfig& output_config,
                     float* const* dest) override;
   bool GetLinearAecOutput(
-      ArrayView<std::array<float, 160>> linear_output) const override;
+      std::span<std::array<float, 160>> linear_output) const override;
   void set_output_will_be_muted(bool muted) override;
   void HandleCaptureOutputUsedSetting(bool capture_output_used)
       RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
diff --git a/modules/audio_processing/audio_processing_impl_locking_unittest.cc b/modules/audio_processing/audio_processing_impl_locking_unittest.cc
index debd5b1b96..3bd808d210 100644
--- a/modules/audio_processing/audio_processing_impl_locking_unittest.cc
+++ b/modules/audio_processing/audio_processing_impl_locking_unittest.cc
@@ -11,9 +11,9 @@
 #include <algorithm>
 #include <cstddef>
 #include <cstdint>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/audio_processing.h"
 #include "api/audio/builtin_audio_processing_builder.h"
 #include "api/environment/environment_factory.h"
@@ -226,7 +226,7 @@ struct TestConfig {
 
       std::vector<TestConfig> out;
       for (auto test_config : in) {
-        auto available_rates = ArrayView<const int>(sample_rates);
+        auto available_rates = std::span<const int>(sample_rates);
 
         for (auto rate : available_rates) {
           test_config.initial_sample_rate_hz = rate;
@@ -460,7 +460,7 @@ void PopulateAudioFrame(float** frame,
 void PopulateAudioFrame(float amplitude,
                         size_t num_channels,
                         size_t samples_per_channel,
-                        ArrayView<int16_t> frame,
+                        std::span<int16_t> frame,
                         RandomGenerator* rand_gen) {
   ASSERT_GT(amplitude, 0);
   ASSERT_LE(amplitude, 32767);
diff --git a/modules/audio_processing/audio_processing_impl_unittest.cc b/modules/audio_processing/audio_processing_impl_unittest.cc
index 27a41344d4..2bbf10b717 100644
--- a/modules/audio_processing/audio_processing_impl_unittest.cc
+++ b/modules/audio_processing/audio_processing_impl_unittest.cc
@@ -15,12 +15,12 @@
 #include <cstddef>
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <string>
 #include <tuple>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/audio_processing.h"
 #include "api/audio/builtin_audio_processing_builder.h"
 #include "api/audio/echo_control.h"
@@ -90,12 +90,12 @@ class TestEchoDetector : public EchoDetector {
       : analyze_render_audio_called_(false),
         last_render_audio_first_sample_(0.f) {}
   ~TestEchoDetector() override = default;
-  void AnalyzeRenderAudio(ArrayView<const float> render_audio) override {
+  void AnalyzeRenderAudio(std::span<const float> render_audio) override {
     last_render_audio_first_sample_ = render_audio[0];
     analyze_render_audio_called_ = true;
   }
   void AnalyzeCaptureAudio(
-      ArrayView<const float> /* capture_audio */) override {}
+      std::span<const float> /* capture_audio */) override {}
   void Initialize(int /* capture_sample_rate_hz */,
                   int /* num_capture_channels */,
                   int /* render_sample_rate_hz */,
@@ -125,7 +125,7 @@ class TestRenderPreProcessor : public CustomProcessing {
   void Initialize(int /* sample_rate_hz */, int /* num_channels */) override {}
   void Process(AudioBuffer* audio) override {
     for (size_t k = 0; k < audio->num_channels(); ++k) {
-      ArrayView<float> channel_view(audio->channels()[k], audio->num_frames());
+      std::span<float> channel_view(audio->channels()[k], audio->num_frames());
       std::transform(channel_view.begin(), channel_view.end(),
                      channel_view.begin(), ProcessSample);
     }
diff --git a/modules/audio_processing/audio_processing_unittest.cc b/modules/audio_processing/audio_processing_unittest.cc
index 7078370c64..e77357dfa7 100644
--- a/modules/audio_processing/audio_processing_unittest.cc
+++ b/modules/audio_processing/audio_processing_unittest.cc
@@ -21,6 +21,7 @@
 #include <memory>
 #include <numeric>
 #include <queue>
+#include <span>
 #include <string>
 #include <tuple>
 #include <utility>
@@ -28,7 +29,6 @@
 
 #include "absl/flags/flag.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio/audio_processing_statistics.h"
 #include "api/audio/audio_view.h"
 #include "api/audio/builtin_audio_processing_builder.h"
@@ -316,8 +316,8 @@ void ExpectEventFieldsEq(const audioproc::Event& actual,
 // and contain the same data. If they differ and `kDumpWhenExpectMessageEqFails`
 // is true, checks the equality of a subset of `audioproc::Event` (nested)
 // fields.
-bool ExpectMessageEq(ArrayView<const uint8_t> actual,
-                     ArrayView<const uint8_t> expected) {
+bool ExpectMessageEq(std::span<const uint8_t> actual,
+                     std::span<const uint8_t> expected) {
   EXPECT_EQ(actual.size(), expected.size());
   if (actual.size() != expected.size()) {
     return false;
@@ -839,7 +839,7 @@ TEST_F(ApmTest, PreAmplifier) {
   tmp_frame.CopyFrom(frame_);
 
   auto compute_power = [](const Int16FrameData& frame) {
-    ArrayView<const int16_t> data = frame.view().data();
+    std::span<const int16_t> data = frame.view().data();
     return std::accumulate(data.begin(), data.end(), 0.0f,
                            [](float a, float b) { return a + b * b; }) /
            data.size() / 32768 / 32768;
@@ -942,7 +942,7 @@ TEST_F(ApmTest, CaptureLevelAdjustment) {
   tmp_frame.CopyFrom(frame_);
 
   auto compute_power = [](const Int16FrameData& frame) {
-    ArrayView<const int16_t> data = frame.view().data();
+    std::span<const int16_t> data = frame.view().data();
     return std::accumulate(data.begin(), data.end(), 0.0f,
                            [](float a, float b) { return a + b * b; }) /
            data.size() / 32768 / 32768;
@@ -2051,10 +2051,10 @@ std::string ProduceDebugText(int render_input_sample_rate_hz,
 
 // Validates that running the audio processing module using various combinations
 // of sample rates and number of channels works as intended.
-void RunApmRateAndChannelTest(ArrayView<const int> sample_rates_hz,
-                              ArrayView<const int> render_channel_counts,
-                              ArrayView<const int> capture_channel_counts,
-                              ArrayView<const bool> mono_capture_output) {
+void RunApmRateAndChannelTest(std::span<const int> sample_rates_hz,
+                              std::span<const int> render_channel_counts,
+                              std::span<const int> capture_channel_counts,
+                              std::span<const bool> mono_capture_output) {
   AudioProcessing::Config apm_config;
   apm_config.pipeline.multi_channel_render = true;
   apm_config.pipeline.multi_channel_capture = true;
@@ -2326,11 +2326,11 @@ TEST(ApmConfiguration, EchoDetectorInjection) {
 
   // The echo detector is included in processing when enabled.
   EXPECT_CALL(*mock_echo_detector, AnalyzeRenderAudio(_))
-      .WillOnce([](ArrayView<const float> render_audio) {
+      .WillOnce([](std::span<const float> render_audio) {
         EXPECT_EQ(render_audio.size(), 160u);
       });
   EXPECT_CALL(*mock_echo_detector, AnalyzeCaptureAudio(_))
-      .WillOnce([](ArrayView<const float> capture_audio) {
+      .WillOnce([](std::span<const float> capture_audio) {
         EXPECT_EQ(capture_audio.size(), 160u);
       });
   EXPECT_CALL(*mock_echo_detector, GetMetrics()).Times(1);
@@ -2356,12 +2356,12 @@ TEST(ApmConfiguration, EchoDetectorInjection) {
                          /*render_sample_rate_hz=*/48000, _))
       .Times(1);
   EXPECT_CALL(*mock_echo_detector, AnalyzeRenderAudio(_))
-      .WillOnce([](ArrayView<const float> render_audio) {
+      .WillOnce([](std::span<const float> render_audio) {
         EXPECT_EQ(render_audio.size(), 480u);
       });
   EXPECT_CALL(*mock_echo_detector, AnalyzeCaptureAudio(_))
       .Times(2)
-      .WillRepeatedly([](ArrayView<const float> capture_audio) {
+      .WillRepeatedly([](std::span<const float> capture_audio) {
         EXPECT_EQ(capture_audio.size(), 480u);
       });
   EXPECT_CALL(*mock_echo_detector, GetMetrics()).Times(2);
diff --git a/modules/audio_processing/capture_levels_adjuster/BUILD.gn b/modules/audio_processing/capture_levels_adjuster/BUILD.gn
index 20b2e395a8..b8df947692 100644
--- a/modules/audio_processing/capture_levels_adjuster/BUILD.gn
+++ b/modules/audio_processing/capture_levels_adjuster/BUILD.gn
@@ -22,7 +22,6 @@ rtc_library("capture_levels_adjuster") {
 
   deps = [
     "..:audio_buffer",
-    "../../../api:array_view",
     "../../../rtc_base:checks",
     "../../../rtc_base:safe_minmax",
   ]
diff --git a/modules/audio_processing/capture_levels_adjuster/audio_samples_scaler.cc b/modules/audio_processing/capture_levels_adjuster/audio_samples_scaler.cc
index a2430ab4f1..b32f8d2c00 100644
--- a/modules/audio_processing/capture_levels_adjuster/audio_samples_scaler.cc
+++ b/modules/audio_processing/capture_levels_adjuster/audio_samples_scaler.cc
@@ -11,8 +11,8 @@
 
 #include <algorithm>
 #include <cstddef>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/audio_buffer.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/numerics/safe_minmax.h"
@@ -40,7 +40,7 @@ void AudioSamplesScaler::Process(AudioBuffer& audio_buffer) {
   if (previous_gain_ == target_gain_) {
     // Apply a non-changing gain.
     for (size_t channel = 0; channel < audio_buffer.num_channels(); ++channel) {
-      ArrayView<float> channel_view(audio_buffer.channels()[channel],
+      std::span<float> channel_view(audio_buffer.channels()[channel],
                                     samples_per_channel_);
       for (float& sample : channel_view) {
         sample *= gain;
@@ -55,7 +55,7 @@ void AudioSamplesScaler::Process(AudioBuffer& audio_buffer) {
       for (size_t channel = 0; channel < audio_buffer.num_channels();
            ++channel) {
         gain = previous_gain_;
-        ArrayView<float> channel_view(audio_buffer.channels()[channel],
+        std::span<float> channel_view(audio_buffer.channels()[channel],
                                       samples_per_channel_);
         for (float& sample : channel_view) {
           gain = std::min(gain + increment, target_gain_);
@@ -67,7 +67,7 @@ void AudioSamplesScaler::Process(AudioBuffer& audio_buffer) {
       for (size_t channel = 0; channel < audio_buffer.num_channels();
            ++channel) {
         gain = previous_gain_;
-        ArrayView<float> channel_view(audio_buffer.channels()[channel],
+        std::span<float> channel_view(audio_buffer.channels()[channel],
                                       samples_per_channel_);
         for (float& sample : channel_view) {
           gain = std::max(gain + increment, target_gain_);
@@ -80,7 +80,7 @@ void AudioSamplesScaler::Process(AudioBuffer& audio_buffer) {
 
   // Saturate the samples to be in the S16 range.
   for (size_t channel = 0; channel < audio_buffer.num_channels(); ++channel) {
-    ArrayView<float> channel_view(audio_buffer.channels()[channel],
+    std::span<float> channel_view(audio_buffer.channels()[channel],
                                   samples_per_channel_);
     for (float& sample : channel_view) {
       constexpr float kMinFloatS16Value = -32768.f;
diff --git a/modules/audio_processing/capture_mixer/BUILD.gn b/modules/audio_processing/capture_mixer/BUILD.gn
index ec4653d358..16dfba79e6 100644
--- a/modules/audio_processing/capture_mixer/BUILD.gn
+++ b/modules/audio_processing/capture_mixer/BUILD.gn
@@ -30,10 +30,7 @@ rtc_library("capture_mixer") {
 
   defines = []
 
-  deps = [
-    "../../../api:array_view",
-    "../../../rtc_base:checks",
-  ]
+  deps = [ "../../../rtc_base:checks" ]
 }
 
 rtc_library("capture_mixer_unittests") {
@@ -51,7 +48,6 @@ rtc_library("capture_mixer_unittests") {
   deps = [
     ":capture_mixer",
     "..:audioproc_test_utils",
-    "../../../api:array_view",
     "../../../rtc_base:gunit_helpers",
     "../../../rtc_base:stringutils",
     "../../../test:test_support",
diff --git a/modules/audio_processing/capture_mixer/audio_content_analyzer.cc b/modules/audio_processing/capture_mixer/audio_content_analyzer.cc
index 4f079427ec..db47fb12a1 100644
--- a/modules/audio_processing/capture_mixer/audio_content_analyzer.cc
+++ b/modules/audio_processing/capture_mixer/audio_content_analyzer.cc
@@ -10,8 +10,7 @@
 #include "modules/audio_processing/capture_mixer/audio_content_analyzer.h"
 
 #include <cstddef>
-
-#include "api/array_view.h"
+#include <span>
 
 namespace webrtc {
 
@@ -19,8 +18,8 @@ AudioContentAnalyzer::AudioContentAnalyzer(size_t num_samples_per_channel)
     : dc_levels_estimator_(num_samples_per_channel),
       saturation_estimator_(num_samples_per_channel) {}
 
-bool AudioContentAnalyzer::Analyze(ArrayView<const float> channel0,
-                                   ArrayView<const float> channel1) {
+bool AudioContentAnalyzer::Analyze(std::span<const float> channel0,
+                                   std::span<const float> channel1) {
   ++num_frames_analyzed_;
 
   // Exclude the first frame from the analysis to avoid reacting on any
@@ -40,7 +39,7 @@ bool AudioContentAnalyzer::Analyze(ArrayView<const float> channel0,
     return false;
   }
 
-  ArrayView<const float, 2> dc_levels = dc_levels_estimator_.GetLevels();
+  std::span<const float, 2> dc_levels = dc_levels_estimator_.GetLevels();
   energy_estimator_.Update(channel0, channel1, dc_levels);
   saturation_estimator_.Update(channel0, channel1, dc_levels);
 
diff --git a/modules/audio_processing/capture_mixer/audio_content_analyzer.h b/modules/audio_processing/capture_mixer/audio_content_analyzer.h
index e4d8eb1fa7..436c15c76f 100644
--- a/modules/audio_processing/capture_mixer/audio_content_analyzer.h
+++ b/modules/audio_processing/capture_mixer/audio_content_analyzer.h
@@ -12,7 +12,8 @@
 
 #include <stddef.h>
 
-#include "api/array_view.h"
+#include <span>
+
 #include "modules/audio_processing/capture_mixer/dc_levels_estimator.h"
 #include "modules/audio_processing/capture_mixer/energy_estimator.h"
 #include "modules/audio_processing/capture_mixer/saturation_estimator.h"
@@ -33,17 +34,17 @@ class AudioContentAnalyzer {
   // Analyzes the provided audio samples for the two channels.
   // Updates the internal energy, and saturation estimators.
   // Returns true if the current frame is considered to contain activity.
-  bool Analyze(ArrayView<const float> channel0,
-               ArrayView<const float> channel1);
+  bool Analyze(std::span<const float> channel0,
+               std::span<const float> channel1);
 
   // Returns the current average energy estimates for the two channels.
-  ArrayView<const float, 2> GetChannelEnergies() const {
+  std::span<const float, 2> GetChannelEnergies() const {
     return energy_estimator_.GetChannelEnergies();
   }
 
   // Returns the number of frames since the last activity was detected in each
   // of the channels.
-  ArrayView<const int, 2> GetNumFramesSinceActivity() const {
+  std::span<const int, 2> GetNumFramesSinceActivity() const {
     return saturation_estimator_.GetNumFramesSinceActivity();
   }
 
@@ -51,7 +52,7 @@ class AudioContentAnalyzer {
   // saturation factor is a value between 0 and 1, where 1 means that the signal
   // has recently been fully saturated and 0 means that no saturation has been
   // observed in the resent past.
-  ArrayView<const float, 2> GetSaturationFactors() const {
+  std::span<const float, 2> GetSaturationFactors() const {
     return saturation_estimator_.GetSaturationFactors();
   }
 
diff --git a/modules/audio_processing/capture_mixer/audio_content_analyzer_unittest.cc b/modules/audio_processing/capture_mixer/audio_content_analyzer_unittest.cc
index 51abf94cfa..ded4c0e83b 100644
--- a/modules/audio_processing/capture_mixer/audio_content_analyzer_unittest.cc
+++ b/modules/audio_processing/capture_mixer/audio_content_analyzer_unittest.cc
@@ -13,7 +13,6 @@
 #include <tuple>
 #include <vector>
 
-#include "api/array_view.h"
 #include "test/gtest.h"
 
 namespace webrtc {
diff --git a/modules/audio_processing/capture_mixer/capture_mixer.cc b/modules/audio_processing/capture_mixer/capture_mixer.cc
index d1d011632f..cbbfb928a6 100644
--- a/modules/audio_processing/capture_mixer/capture_mixer.cc
+++ b/modules/audio_processing/capture_mixer/capture_mixer.cc
@@ -10,8 +10,8 @@
 #include "modules/audio_processing/capture_mixer/capture_mixer.h"
 
 #include <cstddef>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/capture_mixer/channel_content_remixer.h"
 #include "rtc_base/checks.h"
 
@@ -28,8 +28,8 @@ CaptureMixer::CaptureMixer(size_t num_samples_per_channel)
       remixing_logic_(num_samples_per_channel) {}
 
 void CaptureMixer::Mix(size_t num_output_channels,
-                       ArrayView<float> channel0,
-                       ArrayView<float> channel1) {
+                       std::span<float> channel0,
+                       std::span<float> channel1) {
   RTC_DCHECK_GE(num_output_channels, 1);
   RTC_DCHECK_LE(num_output_channels, 2);
 
@@ -45,11 +45,11 @@ void CaptureMixer::Mix(size_t num_output_channels,
     return;
   }
 
-  ArrayView<const float, 2> average_energies =
+  std::span<const float, 2> average_energies =
       audio_content_analyzer_.GetChannelEnergies();
-  ArrayView<const int, 2> num_frames_since_activity =
+  std::span<const int, 2> num_frames_since_activity =
       audio_content_analyzer_.GetNumFramesSinceActivity();
-  ArrayView<const float, 2> saturation_factors =
+  std::span<const float, 2> saturation_factors =
       audio_content_analyzer_.GetSaturationFactors();
 
   mixing_variant_ = remixing_logic_.SelectStereoChannelMixing(
diff --git a/modules/audio_processing/capture_mixer/capture_mixer.h b/modules/audio_processing/capture_mixer/capture_mixer.h
index 88c35f9ad3..2cc9605f36 100644
--- a/modules/audio_processing/capture_mixer/capture_mixer.h
+++ b/modules/audio_processing/capture_mixer/capture_mixer.h
@@ -12,7 +12,8 @@
 
 #include <stddef.h>
 
-#include "api/array_view.h"
+#include <span>
+
 #include "modules/audio_processing/capture_mixer/audio_content_analyzer.h"
 #include "modules/audio_processing/capture_mixer/channel_content_remixer.h"
 #include "modules/audio_processing/capture_mixer/remixing_logic.h"
@@ -26,8 +27,8 @@ class CaptureMixer {
   CaptureMixer& operator=(const CaptureMixer&) = delete;
 
   void Mix(size_t num_output_channels,
-           ArrayView<float> channel0,
-           ArrayView<float> channel1);
+           std::span<float> channel0,
+           std::span<float> channel1);
 
  private:
   AudioContentAnalyzer audio_content_analyzer_;
diff --git a/modules/audio_processing/capture_mixer/channel_content_remixer.cc b/modules/audio_processing/capture_mixer/channel_content_remixer.cc
index 073abdb8f8..8293f47909 100644
--- a/modules/audio_processing/capture_mixer/channel_content_remixer.cc
+++ b/modules/audio_processing/capture_mixer/channel_content_remixer.cc
@@ -11,8 +11,8 @@
 
 #include <algorithm>
 #include <cstddef>
+#include <span>
 
-#include "api/array_view.h"
 #include "rtc_base/checks.h"
 
 namespace webrtc {
@@ -28,8 +28,8 @@ ChannelContentRemixer::ChannelContentRemixer(size_t num_samples_per_channel,
 
 bool ChannelContentRemixer::Mix(size_t num_output_channels,
                                 StereoMixingVariant mixing_variant,
-                                ArrayView<float> channel0,
-                                ArrayView<float> channel1) {
+                                std::span<float> channel0,
+                                std::span<float> channel1) {
   RTC_DCHECK_EQ(channel0.size(), num_samples_per_channel_);
   RTC_DCHECK_EQ(channel1.size(), num_samples_per_channel_);
 
@@ -304,14 +304,14 @@ bool ChannelContentRemixer::IsCrossfadeCompleted() {
 }
 
 void ChannelContentRemixer::CopyChannelContent(
-    ArrayView<const float> source,
-    ArrayView<float> destination) const {
+    std::span<const float> source,
+    std::span<float> destination) const {
   std::copy(source.begin(), source.end(), destination.begin());
 }
 
 void ChannelContentRemixer::StoreChannelAverageIntoBothChannels(
-    ArrayView<float> channel0,
-    ArrayView<float> channel1) const {
+    std::span<float> channel0,
+    std::span<float> channel1) const {
   for (size_t k = 0; k < channel0.size(); ++k) {
     float average = (channel0[k] + channel1[k]) * 0.5f;
     channel0[k] = average;
@@ -320,9 +320,9 @@ void ChannelContentRemixer::StoreChannelAverageIntoBothChannels(
 }
 
 void ChannelContentRemixer::CrossFadeFromSingleChannelToSingleChannel(
-    ArrayView<const float> crossfade_from,
-    ArrayView<const float> crossfade_to,
-    ArrayView<float> destination,
+    std::span<const float> crossfade_from,
+    std::span<const float> crossfade_to,
+    std::span<float> destination,
     size_t& crossfade_sample_counter) const {
   for (size_t k = 0; k < destination.size(); ++k, ++crossfade_sample_counter) {
     const float scaling =
@@ -333,9 +333,9 @@ void ChannelContentRemixer::CrossFadeFromSingleChannelToSingleChannel(
 }
 
 void ChannelContentRemixer::CrossFadeFromSingleChannelContentToAverage(
-    ArrayView<const float> crossfade_from,
-    ArrayView<float> channel0,
-    ArrayView<float> channel1,
+    std::span<const float> crossfade_from,
+    std::span<float> channel0,
+    std::span<float> channel1,
     size_t& crossfade_sample_counter) const {
   for (size_t k = 0; k < channel0.size(); ++k, ++crossfade_sample_counter) {
     const float scaling =
@@ -350,9 +350,9 @@ void ChannelContentRemixer::CrossFadeFromSingleChannelContentToAverage(
 }
 
 void ChannelContentRemixer::CrossFadeFromAverageToSingleChannelContent(
-    ArrayView<const float> crossfade_to,
-    ArrayView<float> channel0,
-    ArrayView<float> channel1,
+    std::span<const float> crossfade_to,
+    std::span<float> channel0,
+    std::span<float> channel1,
     size_t& crossfade_sample_counter) const {
   for (size_t k = 0; k < channel0.size(); ++k, ++crossfade_sample_counter) {
     const float scaling =
@@ -366,8 +366,8 @@ void ChannelContentRemixer::CrossFadeFromAverageToSingleChannelContent(
 }
 
 void ChannelContentRemixer::CrossFadeFromAverageToBothChannels(
-    ArrayView<float> channel0,
-    ArrayView<float> channel1,
+    std::span<float> channel0,
+    std::span<float> channel1,
     size_t& crossfade_sample_counter) const {
   for (size_t k = 0; k < channel0.size(); ++k, ++crossfade_sample_counter) {
     const float scaling =
@@ -381,8 +381,8 @@ void ChannelContentRemixer::CrossFadeFromAverageToBothChannels(
 }
 
 void ChannelContentRemixer::CrossFadeFromBothChannelsToAverage(
-    ArrayView<float> channel0,
-    ArrayView<float> channel1,
+    std::span<float> channel0,
+    std::span<float> channel1,
     size_t& crossfade_sample_counter) const {
   for (size_t k = 0; k < channel0.size(); ++k, ++crossfade_sample_counter) {
     const float scaling =
@@ -395,9 +395,9 @@ void ChannelContentRemixer::CrossFadeFromBothChannelsToAverage(
 }
 
 void ChannelContentRemixer::CrossFadeFromAverageInToChannel0(
-    ArrayView<const float> crossfade_to,
-    ArrayView<float> channel0,
-    ArrayView<float> channel1,
+    std::span<const float> crossfade_to,
+    std::span<float> channel0,
+    std::span<float> channel1,
     size_t& crossfade_sample_counter) const {
   for (size_t k = 0; k < channel0.size(); ++k, ++crossfade_sample_counter) {
     const float scaling =
@@ -409,9 +409,9 @@ void ChannelContentRemixer::CrossFadeFromAverageInToChannel0(
 }
 
 void ChannelContentRemixer::CrossFadeChannel0ToAverage(
-    ArrayView<const float> crossfade_from,
-    ArrayView<float> channel0,
-    ArrayView<const float> channel1,
+    std::span<const float> crossfade_from,
+    std::span<float> channel0,
+    std::span<const float> channel1,
     size_t& crossfade_sample_counter) const {
   for (size_t k = 0; k < channel0.size(); ++k, ++crossfade_sample_counter) {
     const float scaling =
@@ -423,8 +423,8 @@ void ChannelContentRemixer::CrossFadeChannel0ToAverage(
 }
 
 void ChannelContentRemixer::StoreChannelAverageIntoChannel0(
-    ArrayView<float> channel0,
-    ArrayView<const float> channel1) const {
+    std::span<float> channel0,
+    std::span<const float> channel1) const {
   for (size_t k = 0; k < channel0.size(); ++k) {
     float average = (channel0[k] + channel1[k]) * 0.5f;
     channel0[k] = average;
diff --git a/modules/audio_processing/capture_mixer/channel_content_remixer.h b/modules/audio_processing/capture_mixer/channel_content_remixer.h
index 39d52bb7ed..3226dbe156 100644
--- a/modules/audio_processing/capture_mixer/channel_content_remixer.h
+++ b/modules/audio_processing/capture_mixer/channel_content_remixer.h
@@ -11,8 +11,7 @@
 #define MODULES_AUDIO_PROCESSING_CAPTURE_MIXER_CHANNEL_CONTENT_REMIXER_H_
 
 #include <cstddef>
-
-#include "api/array_view.h"
+#include <span>
 
 namespace webrtc {
 
@@ -50,8 +49,8 @@ class ChannelContentRemixer {
   // crossfades are completed.
   bool Mix(size_t num_output_channels,
            StereoMixingVariant mixing_variant,
-           ArrayView<float> channel0,
-           ArrayView<float> channel1);
+           std::span<float> channel0,
+           std::span<float> channel1);
 
  private:
   const size_t num_samples_per_channel_;
@@ -67,63 +66,63 @@ class ChannelContentRemixer {
   bool IsCrossfadeCompleted();
 
   // Copies content from source to destination.
-  void CopyChannelContent(ArrayView<const float> source,
-                          ArrayView<float> destination) const;
+  void CopyChannelContent(std::span<const float> source,
+                          std::span<float> destination) const;
 
   // Calculates the average of channel0 and channel1 and writes it to both.
-  void StoreChannelAverageIntoBothChannels(ArrayView<float> channel0,
-                                           ArrayView<float> channel1) const;
+  void StoreChannelAverageIntoBothChannels(std::span<float> channel0,
+                                           std::span<float> channel1) const;
 
   // Performs a linear cross-fade from `crossfade_from` to `crossfade_to` into
   // `destination`.
   void CrossFadeFromSingleChannelToSingleChannel(
-      ArrayView<const float> crossfade_from,
-      ArrayView<const float> crossfade_to,
-      ArrayView<float> destination,
+      std::span<const float> crossfade_from,
+      std::span<const float> crossfade_to,
+      std::span<float> destination,
       size_t& crossfade_sample_counter) const;
 
   // Cross-fades from a single channel to the average of both channels.
   void CrossFadeFromSingleChannelContentToAverage(
-      ArrayView<const float> crossfade_from,
-      ArrayView<float> channel0,
-      ArrayView<float> channel1,
+      std::span<const float> crossfade_from,
+      std::span<float> channel0,
+      std::span<float> channel1,
       size_t& crossfade_sample_counter) const;
 
   // Cross-fades from the average of both channels to a single channel.
   void CrossFadeFromAverageToSingleChannelContent(
-      ArrayView<const float> crossfade_to,
-      ArrayView<float> channel0,
-      ArrayView<float> channel1,
+      std::span<const float> crossfade_to,
+      std::span<float> channel0,
+      std::span<float> channel1,
       size_t& crossfade_sample_counter) const;
 
   // Cross-fades from the average of both channels to using both channels
   // independently.
   void CrossFadeFromAverageToBothChannels(
-      ArrayView<float> channel0,
-      ArrayView<float> channel1,
+      std::span<float> channel0,
+      std::span<float> channel1,
       size_t& crossfade_sample_counter) const;
 
   // Cross-fades from using both channels independently to their average.
   void CrossFadeFromBothChannelsToAverage(
-      ArrayView<float> channel0,
-      ArrayView<float> channel1,
+      std::span<float> channel0,
+      std::span<float> channel1,
       size_t& crossfade_sample_counter) const;
 
   // specific helper for Mix when num_output_channels == 1.
-  void CrossFadeFromAverageInToChannel0(ArrayView<const float> crossfade_to,
-                                        ArrayView<float> channel0,
-                                        ArrayView<float> channel1,
+  void CrossFadeFromAverageInToChannel0(std::span<const float> crossfade_to,
+                                        std::span<float> channel0,
+                                        std::span<float> channel1,
                                         size_t& crossfade_sample_counter) const;
 
   // specific helper for Mix when num_output_channels == 1.
-  void CrossFadeChannel0ToAverage(ArrayView<const float> crossfade_from,
-                                  ArrayView<float> channel0,
-                                  ArrayView<const float> channel1,
+  void CrossFadeChannel0ToAverage(std::span<const float> crossfade_from,
+                                  std::span<float> channel0,
+                                  std::span<const float> channel1,
                                   size_t& crossfade_sample_counter) const;
 
   // specific helper for Mix when num_output_channels == 1.
-  void StoreChannelAverageIntoChannel0(ArrayView<float> channel0,
-                                       ArrayView<const float> channel1) const;
+  void StoreChannelAverageIntoChannel0(std::span<float> channel0,
+                                       std::span<const float> channel1) const;
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_processing/capture_mixer/channel_content_remixer_unittest.cc b/modules/audio_processing/capture_mixer/channel_content_remixer_unittest.cc
index 48ca99c698..bfc2ef7380 100644
--- a/modules/audio_processing/capture_mixer/channel_content_remixer_unittest.cc
+++ b/modules/audio_processing/capture_mixer/channel_content_remixer_unittest.cc
@@ -12,10 +12,10 @@
 
 #include <algorithm>
 #include <cstddef>
+#include <span>
 #include <tuple>
 #include <vector>
 
-#include "api/array_view.h"
 #include "test/gtest.h"
 
 namespace webrtc {
@@ -33,7 +33,7 @@ void PopulateChannels(std::vector<float>& channel0,
 
 void VerifyCrossFade(float value_begin,
                      float value_end,
-                     ArrayView<const float> channel_data) {
+                     std::span<const float> channel_data) {
   const float one_by_num_samples_per_channel = 1.0f / channel_data.size();
   for (size_t k = 0; k < channel_data.size(); ++k) {
     const float expected_value =
@@ -44,7 +44,7 @@ void VerifyCrossFade(float value_begin,
 }
 
 void VerifyConstantValue(float expected_value,
-                         ArrayView<const float> channel_data) {
+                         std::span<const float> channel_data) {
   for (size_t k = 0; k < channel_data.size(); ++k) {
     EXPECT_NEAR(channel_data[k], expected_value, 1e-3);
   }
diff --git a/modules/audio_processing/capture_mixer/dc_levels_estimator.cc b/modules/audio_processing/capture_mixer/dc_levels_estimator.cc
index 5f8d1fdd45..1651d37379 100644
--- a/modules/audio_processing/capture_mixer/dc_levels_estimator.cc
+++ b/modules/audio_processing/capture_mixer/dc_levels_estimator.cc
@@ -11,15 +11,14 @@
 
 #include <cstddef>
 #include <numeric>
-
-#include "api/array_view.h"
+#include <span>
 
 namespace webrtc {
 
 namespace {
 
 void UpdateDcEstimate(float one_by_num_samples_per_channel,
-                      ArrayView<const float> audio,
+                      std::span<const float> audio,
                       float& dc_estimate) {
   constexpr float kForgettingFactor = 0.05f;
   float mean = std::accumulate(audio.begin(), audio.end(), 0.0f) *
@@ -34,8 +33,8 @@ DcLevelsEstimator::DcLevelsEstimator(size_t num_samples_per_channel)
   dc_levels_.fill(0.0f);
 }
 
-void DcLevelsEstimator::Update(ArrayView<const float> channel0,
-                               ArrayView<const float> channel1) {
+void DcLevelsEstimator::Update(std::span<const float> channel0,
+                               std::span<const float> channel1) {
   UpdateDcEstimate(one_by_num_samples_per_channel_, channel0, dc_levels_[0]);
   UpdateDcEstimate(one_by_num_samples_per_channel_, channel1, dc_levels_[1]);
 }
diff --git a/modules/audio_processing/capture_mixer/dc_levels_estimator.h b/modules/audio_processing/capture_mixer/dc_levels_estimator.h
index 8ab00ad62d..b7181feef5 100644
--- a/modules/audio_processing/capture_mixer/dc_levels_estimator.h
+++ b/modules/audio_processing/capture_mixer/dc_levels_estimator.h
@@ -13,8 +13,7 @@
 #include <stddef.h>
 
 #include <array>
-
-#include "api/array_view.h"
+#include <span>
 
 namespace webrtc {
 
@@ -30,10 +29,10 @@ class DcLevelsEstimator {
 
   // Updates the DC level estimates.
   // `channel0` and `channel1` contain the samples of the two channels.
-  void Update(ArrayView<const float> channel0, ArrayView<const float> channel1);
+  void Update(std::span<const float> channel0, std::span<const float> channel1);
 
   // Returns the current DC level estimates for the two channels.
-  ArrayView<const float, 2> GetLevels() const { return dc_levels_; }
+  std::span<const float, 2> GetLevels() const { return dc_levels_; }
 
  private:
   const float one_by_num_samples_per_channel_;
diff --git a/modules/audio_processing/capture_mixer/dc_levels_estimator_unittest.cc b/modules/audio_processing/capture_mixer/dc_levels_estimator_unittest.cc
index e6676f4199..ad5d8e4385 100644
--- a/modules/audio_processing/capture_mixer/dc_levels_estimator_unittest.cc
+++ b/modules/audio_processing/capture_mixer/dc_levels_estimator_unittest.cc
@@ -13,10 +13,10 @@
 
 #include <algorithm>
 #include <numbers>
+#include <span>
 #include <tuple>
 #include <vector>
 
-#include "api/array_view.h"
 #include "test/gtest.h"
 
 namespace webrtc {
@@ -25,8 +25,8 @@ namespace {
 void PopulateStereoChannelsWithSinusoid(int sample_rate_hz,
                                         float dc_level,
                                         int& generated_sample_counter,
-                                        ArrayView<float> channel0,
-                                        ArrayView<float> channel1) {
+                                        std::span<float> channel0,
+                                        std::span<float> channel1) {
   constexpr float kPi = std::numbers::pi;
   constexpr float kApmlitudeScaling = 1000.0f;
   constexpr float kBaseSinusoidFrequencyHz = 100.0f;
@@ -78,7 +78,7 @@ TEST_P(DcLevelsEstimatorParametrizedTest, VerifyEstimates) {
     estimator.Update(channel0, channel1);
   }
 
-  ArrayView<const float> levels = estimator.GetLevels();
+  std::span<const float> levels = estimator.GetLevels();
 
   for (const float level : levels) {
     EXPECT_NEAR(level, true_dc_level,
diff --git a/modules/audio_processing/capture_mixer/energy_estimator.cc b/modules/audio_processing/capture_mixer/energy_estimator.cc
index 8a43798d60..e471aa6129 100644
--- a/modules/audio_processing/capture_mixer/energy_estimator.cc
+++ b/modules/audio_processing/capture_mixer/energy_estimator.cc
@@ -9,13 +9,13 @@
  */
 #include "modules/audio_processing/capture_mixer/energy_estimator.h"
 
-#include "api/array_view.h"
+#include <span>
 
 namespace webrtc {
 
 namespace {
 
-void UpdateChannelEnergyEstimate(ArrayView<const float> audio,
+void UpdateChannelEnergyEstimate(std::span<const float> audio,
                                  float dc_level,
                                  float& channel_energy_estimate) {
   float energy = 0.0f;
@@ -35,9 +35,9 @@ AverageEnergyEstimator::AverageEnergyEstimator() {
   average_energy_in_channels_.fill(0.0f);
 }
 
-void AverageEnergyEstimator::Update(ArrayView<const float> channel0,
-                                    ArrayView<const float> channel1,
-                                    ArrayView<const float, 2> dc_levels) {
+void AverageEnergyEstimator::Update(std::span<const float> channel0,
+                                    std::span<const float> channel1,
+                                    std::span<const float, 2> dc_levels) {
   UpdateChannelEnergyEstimate(channel0, dc_levels[0],
                               average_energy_in_channels_[0]);
   UpdateChannelEnergyEstimate(channel1, dc_levels[1],
diff --git a/modules/audio_processing/capture_mixer/energy_estimator.h b/modules/audio_processing/capture_mixer/energy_estimator.h
index 3b3fa8f34d..cc32cb147a 100644
--- a/modules/audio_processing/capture_mixer/energy_estimator.h
+++ b/modules/audio_processing/capture_mixer/energy_estimator.h
@@ -13,8 +13,7 @@
 #include <stddef.h>
 
 #include <array>
-
-#include "api/array_view.h"
+#include <span>
 
 namespace webrtc {
 
@@ -31,12 +30,12 @@ class AverageEnergyEstimator {
   // `channel0` and `channel1` contain the samples of the two channels.
   // `dc_levels` contains the estimated DC offsets for the two channels, which
   // are subtracted from the samples before energy calculation.
-  void Update(ArrayView<const float> channel0,
-              ArrayView<const float> channel1,
-              ArrayView<const float, 2> dc_levels);
+  void Update(std::span<const float> channel0,
+              std::span<const float> channel1,
+              std::span<const float, 2> dc_levels);
 
   // Returns the current average energy estimates for the two channels.
-  ArrayView<const float, 2> GetChannelEnergies() const {
+  std::span<const float, 2> GetChannelEnergies() const {
     return average_energy_in_channels_;
   }
 
diff --git a/modules/audio_processing/capture_mixer/energy_estimator_unittest.cc b/modules/audio_processing/capture_mixer/energy_estimator_unittest.cc
index 715f700a5b..31b0c3de60 100644
--- a/modules/audio_processing/capture_mixer/energy_estimator_unittest.cc
+++ b/modules/audio_processing/capture_mixer/energy_estimator_unittest.cc
@@ -11,10 +11,10 @@
 
 #include <array>
 #include <cstddef>
+#include <span>
 #include <tuple>
 #include <vector>
 
-#include "api/array_view.h"
 #include "test/gtest.h"
 
 namespace webrtc {
@@ -63,7 +63,7 @@ TEST_P(AverageEnergyEstimatorParametrizedTest, VerifyEstimates) {
     estimator.Update(channel0, channel1, dc_levels);
   }
 
-  ArrayView<const float, 2> energies = estimator.GetChannelEnergies();
+  std::span<const float, 2> energies = estimator.GetChannelEnergies();
 
   constexpr float kToleranceError = 0.0001f;
   const float expected_energy_channel_0 =
diff --git a/modules/audio_processing/capture_mixer/remixing_logic.cc b/modules/audio_processing/capture_mixer/remixing_logic.cc
index d490fb5b83..d96bd63375 100644
--- a/modules/audio_processing/capture_mixer/remixing_logic.cc
+++ b/modules/audio_processing/capture_mixer/remixing_logic.cc
@@ -11,8 +11,8 @@
 
 #include <cstddef>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/capture_mixer/channel_content_remixer.h"
 #include "rtc_base/checks.h"
 
@@ -33,7 +33,7 @@ bool ChoiceOfChannelMatchesSingleChannelMixing(int channel,
 }
 
 bool EnoughContentForUpdatingMixing(
-    ArrayView<const int, 2> num_frames_since_activity) {
+    std::span<const int, 2> num_frames_since_activity) {
   const bool channel0_inactive =
       num_frames_since_activity[0] > kInactivityThresholdFrames;
   const bool channel1_inactive =
@@ -44,8 +44,8 @@ bool EnoughContentForUpdatingMixing(
 
 bool SingleSilentChannelDetected(
     size_t num_samples_per_channel,
-    ArrayView<const float, 2> average_energies,
-    ArrayView<const int, 2> num_frames_since_activity) {
+    std::span<const float, 2> average_energies,
+    std::span<const int, 2> num_frames_since_activity) {
   RTC_DCHECK(EnoughContentForUpdatingMixing(num_frames_since_activity));
 
   const bool channel0_inactive =
@@ -72,7 +72,7 @@ bool SingleSilentChannelDetected(
 }
 
 std::optional<int> IdentifyLargelyImbalancedChannel(
-    ArrayView<const float, 2> average_energies) {
+    std::span<const float, 2> average_energies) {
   constexpr float kEnergyRatioThreshold = 50.0f;
   const float& energy0 = average_energies[0];
   const float& energy1 = average_energies[1];
@@ -87,8 +87,8 @@ std::optional<int> IdentifyLargelyImbalancedChannel(
 }
 
 std::optional<int> IdentifyModerateImbalancedAndSaturatedChannel(
-    ArrayView<const float, 2> average_energies,
-    ArrayView<const float, 2> saturation_factors) {
+    std::span<const float, 2> average_energies,
+    std::span<const float, 2> saturation_factors) {
   constexpr float kEnergyRatioModerateThreshold = 4.0f;
   constexpr float kSignificantSaturationThreshold = 0.8f;
   constexpr float kNoSaturationThreshold = 0.1f;
@@ -127,9 +127,9 @@ RemixingLogic::RemixingLogic(size_t num_samples_per_channel,
     : settings_(settings), num_samples_per_channel_(num_samples_per_channel) {}
 
 StereoMixingVariant RemixingLogic::SelectStereoChannelMixing(
-    ArrayView<const float, 2> average_energies,
-    ArrayView<const int, 2> num_frames_since_activity,
-    ArrayView<const float, 2> saturation_factors) {
+    std::span<const float, 2> average_energies,
+    std::span<const int, 2> num_frames_since_activity,
+    std::span<const float, 2> saturation_factors) {
   // Only update the mixing when there is sufficient audio activity.
   if (!EnoughContentForUpdatingMixing(num_frames_since_activity)) {
     return mixing_;
@@ -173,8 +173,8 @@ StereoMixingVariant RemixingLogic::SelectStereoChannelMixing(
 }
 
 bool RemixingLogic::HandleAnySilentChannels(
-    ArrayView<const float, 2> average_energies,
-    ArrayView<const int, 2> num_frames_since_activity) {
+    std::span<const float, 2> average_energies,
+    std::span<const int, 2> num_frames_since_activity) {
   RTC_DCHECK(mode_ != Mode::kSilentChannel ||
              mixing_ == StereoMixingVariant::kUseAverage);
 
@@ -209,8 +209,8 @@ bool RemixingLogic::HandleAnySilentChannels(
 }
 
 bool RemixingLogic::HandleAnyImbalancedAndSaturatedChannels(
-    ArrayView<const float, 2> average_energies,
-    ArrayView<const float, 2> saturation_factors) {
+    std::span<const float, 2> average_energies,
+    std::span<const float, 2> saturation_factors) {
   RTC_DCHECK(mode_ != Mode::kSaturatedChannel ||
              (mixing_ == StereoMixingVariant::kUseChannel0 ||
               mixing_ == StereoMixingVariant::kUseChannel1));
@@ -257,7 +257,7 @@ bool RemixingLogic::HandleAnyImbalancedAndSaturatedChannels(
 }
 
 bool RemixingLogic::HandleAnyLargelyImbalancedChannels(
-    ArrayView<const float, 2> average_energies) {
+    std::span<const float, 2> average_energies) {
   RTC_DCHECK(mode_ != Mode::kImbalancedChannels ||
              (mixing_ == StereoMixingVariant::kUseChannel0 ||
               mixing_ == StereoMixingVariant::kUseChannel1));
diff --git a/modules/audio_processing/capture_mixer/remixing_logic.h b/modules/audio_processing/capture_mixer/remixing_logic.h
index 4104fc7464..d5653c9a11 100644
--- a/modules/audio_processing/capture_mixer/remixing_logic.h
+++ b/modules/audio_processing/capture_mixer/remixing_logic.h
@@ -12,7 +12,8 @@
 
 #include <stddef.h>
 
-#include "api/array_view.h"
+#include <span>
+
 #include "modules/audio_processing/capture_mixer/channel_content_remixer.h"
 
 namespace webrtc {
@@ -48,29 +49,29 @@ class RemixingLogic {
   // active. `saturation_factors`: Saturation measure for each channel. Returns
   // the chosen StereoMixingVariant.
   StereoMixingVariant SelectStereoChannelMixing(
-      ArrayView<const float, 2> average_energies,
-      ArrayView<const int, 2> num_frames_since_activity,
-      ArrayView<const float, 2> saturation_factors);
+      std::span<const float, 2> average_energies,
+      std::span<const int, 2> num_frames_since_activity,
+      std::span<const float, 2> saturation_factors);
 
  private:
   // Checks if any channel is silent and updates the mode and mixing variant
   // accordingly. Returns true if a mode change occurred.
   bool HandleAnySilentChannels(
-      ArrayView<const float, 2> average_energies,
-      ArrayView<const int, 2> num_frames_since_activity);
+      std::span<const float, 2> average_energies,
+      std::span<const int, 2> num_frames_since_activity);
 
   // Checks for channels that are moderately imbalanced and have differing
   // saturation levels, updating mode and mixing variant to favor the less
   // saturated channel. Returns true if a mode change occurred.
   bool HandleAnyImbalancedAndSaturatedChannels(
-      ArrayView<const float, 2> average_energies,
-      ArrayView<const float, 2> saturation_factors);
+      std::span<const float, 2> average_energies,
+      std::span<const float, 2> saturation_factors);
 
   // Checks for channels with a large energy imbalance and updates mode and
   // mixing variant to favor the louder channel. Returns true if a mode change
   // occurred.
   bool HandleAnyLargelyImbalancedChannels(
-      ArrayView<const float, 2> average_energies);
+      std::span<const float, 2> average_energies);
 
   // Represents the current state of the remixing logic.
   enum class Mode {
diff --git a/modules/audio_processing/capture_mixer/saturation_estimator.cc b/modules/audio_processing/capture_mixer/saturation_estimator.cc
index 4c6df83b72..fffcd122d7 100644
--- a/modules/audio_processing/capture_mixer/saturation_estimator.cc
+++ b/modules/audio_processing/capture_mixer/saturation_estimator.cc
@@ -12,15 +12,14 @@
 #include <math.h>
 
 #include <cstddef>
-
-#include "api/array_view.h"
+#include <span>
 
 namespace webrtc {
 
 namespace {
 
 void AnalyzeChannel(float one_by_num_samples_per_channel,
-                    ArrayView<const float> audio,
+                    std::span<const float> audio,
                     float dc_level,
                     int& num_frames_since_activity,
                     float& saturation_factors) {
@@ -59,9 +58,9 @@ SaturationEstimator::SaturationEstimator(size_t num_samples_per_channel)
   saturation_factors_.fill(0.0f);
 }
 
-void SaturationEstimator::Update(ArrayView<const float> channel0,
-                                 ArrayView<const float> channel1,
-                                 ArrayView<const float, 2> dc_levels) {
+void SaturationEstimator::Update(std::span<const float> channel0,
+                                 std::span<const float> channel1,
+                                 std::span<const float, 2> dc_levels) {
   AnalyzeChannel(one_by_num_samples_per_channel_, channel0, dc_levels[0],
                  num_frames_since_activity_[0], saturation_factors_[0]);
   AnalyzeChannel(one_by_num_samples_per_channel_, channel1, dc_levels[1],
diff --git a/modules/audio_processing/capture_mixer/saturation_estimator.h b/modules/audio_processing/capture_mixer/saturation_estimator.h
index d1cd3847bc..87d8cb9882 100644
--- a/modules/audio_processing/capture_mixer/saturation_estimator.h
+++ b/modules/audio_processing/capture_mixer/saturation_estimator.h
@@ -13,8 +13,7 @@
 #include <stddef.h>
 
 #include <array>
-
-#include "api/array_view.h"
+#include <span>
 
 namespace webrtc {
 
@@ -32,13 +31,13 @@ class SaturationEstimator {
   // `channel0` and `channel1` contain the samples of the two channels.
   // `dc_levels` contains the estimated DC offsets for the two channels, which
   // are subtracted from the samples before saturation calculation.
-  void Update(ArrayView<const float> channel0,
-              ArrayView<const float> channel1,
-              ArrayView<const float, 2> dc_levels);
+  void Update(std::span<const float> channel0,
+              std::span<const float> channel1,
+              std::span<const float, 2> dc_levels);
 
   // Returns the number of frames since the last activity was detected in each
   // of the channels.
-  ArrayView<const int, 2> GetNumFramesSinceActivity() const {
+  std::span<const int, 2> GetNumFramesSinceActivity() const {
     return num_frames_since_activity_;
   }
 
@@ -46,7 +45,7 @@ class SaturationEstimator {
   // saturation factor is a value between 0 and 1, where 1 means that the signal
   // has recently been fully saturated and 0 means that no saturation has been
   // observed in the resent past.
-  ArrayView<const float, 2> GetSaturationFactors() const {
+  std::span<const float, 2> GetSaturationFactors() const {
     return saturation_factors_;
   }
 
diff --git a/modules/audio_processing/capture_mixer/saturation_estimator_unittest.cc b/modules/audio_processing/capture_mixer/saturation_estimator_unittest.cc
index 24c5a386d9..682486ff88 100644
--- a/modules/audio_processing/capture_mixer/saturation_estimator_unittest.cc
+++ b/modules/audio_processing/capture_mixer/saturation_estimator_unittest.cc
@@ -11,10 +11,10 @@
 
 #include <algorithm>
 #include <array>
+#include <span>
 #include <tuple>
 #include <vector>
 
-#include "api/array_view.h"
 #include "test/gtest.h"
 
 namespace webrtc {
@@ -56,7 +56,7 @@ TEST_P(SaturationDetectorParametrizedTest, VerifyLowValueActivityDetection) {
     constexpr int kNumFramesToAnalyze = 10;
     for (int k = 0; k < kNumFramesToAnalyze; ++k) {
       estimator.Update(channel, channel, dc_levels);
-      ArrayView<const int, 2> num_frames_since_activity =
+      std::span<const int, 2> num_frames_since_activity =
           estimator.GetNumFramesSinceActivity();
       EXPECT_EQ(num_frames_since_activity[0], k + 1);
       EXPECT_EQ(num_frames_since_activity[1], k + 1);
@@ -81,7 +81,7 @@ TEST_P(SaturationDetectorParametrizedTest,
     constexpr int kNumFramesToAnalyze = 10;
     for (int k = 0; k < kNumFramesToAnalyze; ++k) {
       estimator.Update(channel, channel, dc_levels);
-      ArrayView<const int, 2> num_frames_since_activity =
+      std::span<const int, 2> num_frames_since_activity =
           estimator.GetNumFramesSinceActivity();
       EXPECT_EQ(num_frames_since_activity[0], 0);
       EXPECT_EQ(num_frames_since_activity[1], 0);
@@ -108,7 +108,7 @@ TEST_P(SaturationDetectorParametrizedTest,
       estimator.Update(channel, channel, dc_levels);
     }
 
-    ArrayView<const int, 2> num_frames_since_activity =
+    std::span<const int, 2> num_frames_since_activity =
         estimator.GetNumFramesSinceActivity();
     EXPECT_EQ(num_frames_since_activity[0], 1);
     EXPECT_EQ(num_frames_since_activity[1], 1);
@@ -153,7 +153,7 @@ TEST_P(SaturationDetectorParametrizedTest,
     constexpr int kNumFramesToAnalyze = 10;
     for (int k = 0; k < kNumFramesToAnalyze; ++k) {
       estimator.Update(channel, channel, dc_levels);
-      ArrayView<const float, 2> saturation_factors =
+      std::span<const float, 2> saturation_factors =
           estimator.GetSaturationFactors();
       EXPECT_EQ(saturation_factors[0], 0.0f);
       EXPECT_EQ(saturation_factors[1], 0.0f);
@@ -179,7 +179,7 @@ TEST_P(SaturationDetectorParametrizedTest,
     std::vector<float> previous_factors(2, 0.0f);
     for (int k = 0; k < kNumFramesToAnalyze; ++k) {
       estimator.Update(channel, channel, dc_levels);
-      ArrayView<const float, 2> saturation_factors =
+      std::span<const float, 2> saturation_factors =
           estimator.GetSaturationFactors();
       EXPECT_GT(saturation_factors[0], 0.0f);
       EXPECT_GT(saturation_factors[1], 0.0f);
@@ -209,7 +209,7 @@ TEST_P(SaturationDetectorParametrizedTest,
     for (int k = 0; k < kNumFramesToAnalyze; ++k) {
       estimator.Update(channel, channel, dc_levels);
     }
-    ArrayView<const float, 2> saturation_factors =
+    std::span<const float, 2> saturation_factors =
         estimator.GetSaturationFactors();
     EXPECT_GT(saturation_factors[0], 0.99f);
     EXPECT_GT(saturation_factors[1], 0.99f);
@@ -233,7 +233,7 @@ TEST_P(SaturationDetectorParametrizedTest, VerifyDecayingSaturationFactor) {
           num_samples_per_channel, dc_level + sign * kSampleValueSaturation);
       estimator.Update(channel, channel, dc_levels);
     }
-    ArrayView<const float, 2> saturation_factors =
+    std::span<const float, 2> saturation_factors =
         estimator.GetSaturationFactors();
     EXPECT_GT(saturation_factors[0], 0.0f);
     EXPECT_GT(saturation_factors[1], 0.0f);
diff --git a/modules/audio_processing/gain_control_impl.cc b/modules/audio_processing/gain_control_impl.cc
index 9b29e7ba6b..7e40e9fb25 100644
--- a/modules/audio_processing/gain_control_impl.cc
+++ b/modules/audio_processing/gain_control_impl.cc
@@ -15,9 +15,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/audio_processing.h"
 #include "common_audio/include/audio_util.h"
 #include "modules/audio_processing/agc/gain_control.h"
@@ -112,7 +112,7 @@ GainControlImpl::GainControlImpl()
 GainControlImpl::~GainControlImpl() = default;
 
 void GainControlImpl::ProcessRenderAudio(
-    ArrayView<const int16_t> packed_render_audio) {
+    std::span<const int16_t> packed_render_audio) {
   for (size_t ch = 0; ch < mono_agcs_.size(); ++ch) {
     WebRtcAgc_AddFarend(mono_agcs_[ch]->state, packed_render_audio.data(),
                         packed_render_audio.size());
@@ -125,7 +125,7 @@ void GainControlImpl::PackRenderAudioBuffer(
   RTC_DCHECK_GE(AudioBuffer::kMaxSplitFrameLength, audio.num_frames_per_band());
   std::array<int16_t, AudioBuffer::kMaxSplitFrameLength>
       mixed_16_kHz_render_data;
-  ArrayView<const int16_t> mixed_16_kHz_render(mixed_16_kHz_render_data.data(),
+  std::span<const int16_t> mixed_16_kHz_render(mixed_16_kHz_render_data.data(),
                                                audio.num_frames_per_band());
   if (audio.num_channels() == 1) {
     FloatS16ToS16(audio.split_bands_const(0)[kBand0To8kHz],
diff --git a/modules/audio_processing/gain_control_impl.h b/modules/audio_processing/gain_control_impl.h
index 8a3e94d1e5..cd8c912cfd 100644
--- a/modules/audio_processing/gain_control_impl.h
+++ b/modules/audio_processing/gain_control_impl.h
@@ -16,9 +16,9 @@
 
 #include <memory>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/agc/gain_control.h"
 
 namespace webrtc {
@@ -34,7 +34,7 @@ class GainControlImpl : public GainControl {
 
   ~GainControlImpl() override;
 
-  void ProcessRenderAudio(ArrayView<const int16_t> packed_render_audio);
+  void ProcessRenderAudio(std::span<const int16_t> packed_render_audio);
   int AnalyzeCaptureAudio(const AudioBuffer& audio);
   int ProcessCaptureAudio(AudioBuffer* audio, bool stream_has_echo);
 
diff --git a/modules/audio_processing/gain_control_unittest.cc b/modules/audio_processing/gain_control_unittest.cc
index 41123b98e0..1ecb623b89 100644
--- a/modules/audio_processing/gain_control_unittest.cc
+++ b/modules/audio_processing/gain_control_unittest.cc
@@ -11,9 +11,9 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/audio_processing.h"
 #include "modules/audio_coding/neteq/tools/input_audio_file.h"
 #include "modules/audio_processing/audio_buffer.h"
@@ -77,7 +77,7 @@ void RunBitExactnessTest(int sample_rate_hz,
                          int analog_level_min,
                          int analog_level_max,
                          int achieved_stream_analog_level_reference,
-                         ArrayView<const float> output_reference) {
+                         std::span<const float> output_reference) {
   GainControlImpl gain_controller;
   SetupComponent(sample_rate_hz, mode, target_level_dbfs, stream_analog_level,
                  compression_gain_db, enable_limiter, analog_level_min,
diff --git a/modules/audio_processing/high_pass_filter.cc b/modules/audio_processing/high_pass_filter.cc
index 85b64608e8..7c64d61322 100644
--- a/modules/audio_processing/high_pass_filter.cc
+++ b/modules/audio_processing/high_pass_filter.cc
@@ -12,9 +12,9 @@
 
 #include <array>
 #include <cstddef>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/audio_buffer.h"
 #include "modules/audio_processing/utility/cascaded_biquad_filter.h"
 #include "rtc_base/checks.h"
@@ -54,23 +54,23 @@ constexpr std::array<CascadedBiQuadFilter::BiQuadCoefficients, 3>
          .a = {-1.9983570340145236f, 0.9984928491805198f}},
     }};
 
-ArrayView<const CascadedBiQuadFilter::BiQuadCoefficients> ChooseCoefficients(
+std::span<const CascadedBiQuadFilter::BiQuadCoefficients> ChooseCoefficients(
     int sample_rate_hz) {
   switch (sample_rate_hz) {
     case 16000:
-      return ArrayView<const CascadedBiQuadFilter::BiQuadCoefficients>(
+      return std::span<const CascadedBiQuadFilter::BiQuadCoefficients>(
           kHighPassFilterCoefficients16kHz);
     case 32000:
-      return ArrayView<const CascadedBiQuadFilter::BiQuadCoefficients>(
+      return std::span<const CascadedBiQuadFilter::BiQuadCoefficients>(
           kHighPassFilterCoefficients32kHz);
     case 48000:
-      return ArrayView<const CascadedBiQuadFilter::BiQuadCoefficients>(
+      return std::span<const CascadedBiQuadFilter::BiQuadCoefficients>(
           kHighPassFilterCoefficients48kHz);
     default:
       RTC_DCHECK_NOTREACHED();
   }
   RTC_DCHECK_NOTREACHED();
-  return ArrayView<const CascadedBiQuadFilter::BiQuadCoefficients>(
+  return std::span<const CascadedBiQuadFilter::BiQuadCoefficients>(
       kHighPassFilterCoefficients16kHz);
 }
 
@@ -92,14 +92,14 @@ void HighPassFilter::Process(AudioBuffer* audio, bool use_split_band_data) {
   RTC_DCHECK_EQ(filters_.size(), audio->num_channels());
   if (use_split_band_data) {
     for (size_t k = 0; k < audio->num_channels(); ++k) {
-      ArrayView<float> channel_data = ArrayView<float>(
+      std::span<float> channel_data = std::span<float>(
           audio->split_bands(k)[0], audio->num_frames_per_band());
       filters_[k]->Process(channel_data);
     }
   } else {
     for (size_t k = 0; k < audio->num_channels(); ++k) {
-      ArrayView<float> channel_data =
-          ArrayView<float>(&audio->channels()[k][0], audio->num_frames());
+      std::span<float> channel_data =
+          std::span<float>(&audio->channels()[k][0], audio->num_frames());
       filters_[k]->Process(channel_data);
     }
   }
diff --git a/modules/audio_processing/high_pass_filter_unittest.cc b/modules/audio_processing/high_pass_filter_unittest.cc
index 93a56aef97..76c405189b 100644
--- a/modules/audio_processing/high_pass_filter_unittest.cc
+++ b/modules/audio_processing/high_pass_filter_unittest.cc
@@ -11,9 +11,9 @@
 
 #include <cmath>
 #include <cstddef>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/audio_processing.h"
 #include "modules/audio_processing/audio_buffer.h"
 #include "modules/audio_processing/test/audio_buffer_tools.h"
@@ -124,7 +124,7 @@ void RunBitexactnessTest(int num_channels,
 
 // Method for forming a vector out of an array.
 // TODO(peah): Remove once braced initialization is allowed.
-std::vector<float> CreateVector(const ArrayView<const float>& array_view) {
+std::vector<float> CreateVector(const std::span<const float>& array_view) {
   std::vector<float> v;
   for (auto value : array_view) {
     v.push_back(value);
@@ -230,8 +230,8 @@ TEST(HighPassFilterAccuracyTest, MonoInitial) {
 
   for (bool use_audio_buffer_interface : {true, false}) {
     RunBitexactnessTest(1, use_audio_buffer_interface,
-                        CreateVector(ArrayView<const float>(kReferenceInput)),
-                        CreateVector(ArrayView<const float>(kReference)));
+                        CreateVector(std::span<const float>(kReferenceInput)),
+                        CreateVector(std::span<const float>(kReference)));
   }
 }
 
@@ -324,8 +324,8 @@ TEST(HighPassFilterAccuracyTest, MonoConverged) {
 
   for (bool use_audio_buffer_interface : {true, false}) {
     RunBitexactnessTest(1, use_audio_buffer_interface,
-                        CreateVector(ArrayView<const float>(kReferenceInput)),
-                        CreateVector(ArrayView<const float>(kReference)));
+                        CreateVector(std::span<const float>(kReferenceInput)),
+                        CreateVector(std::span<const float>(kReference)));
   }
 }
 
diff --git a/modules/audio_processing/include/mock_audio_processing.h b/modules/audio_processing/include/mock_audio_processing.h
index 25ae9f8ff3..2b6f468677 100644
--- a/modules/audio_processing/include/mock_audio_processing.h
+++ b/modules/audio_processing/include/mock_audio_processing.h
@@ -16,11 +16,11 @@
 #include <cstdint>
 #include <cstdio>
 #include <memory>
+#include <span>
 #include <string>
 
 #include "absl/base/nullability.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio/audio_processing.h"
 #include "api/audio/audio_processing_statistics.h"
 #include "api/audio/echo_control.h"
@@ -91,11 +91,11 @@ class MockEchoDetector : public EchoDetector {
               (override));
   MOCK_METHOD(void,
               AnalyzeRenderAudio,
-              (webrtc::ArrayView<const float> render_audio),
+              (std::span<const float> render_audio),
               (override));
   MOCK_METHOD(void,
               AnalyzeCaptureAudio,
-              (webrtc::ArrayView<const float> capture_audio),
+              (std::span<const float> capture_audio),
               (override));
   MOCK_METHOD(Metrics, GetMetrics, (), (const, override));
 };
@@ -155,7 +155,7 @@ class MockAudioProcessing : public AudioProcessing {
               (override));
   MOCK_METHOD(bool,
               GetLinearAecOutput,
-              ((webrtc::ArrayView<std::array<float, 160>> linear_output)),
+              ((std::span<std::array<float, 160>> linear_output)),
               (const, override));
   MOCK_METHOD(int, set_stream_delay_ms, (int delay), (override));
   MOCK_METHOD(int, stream_delay_ms, (), (const, override));
diff --git a/modules/audio_processing/logging/apm_data_dumper.h b/modules/audio_processing/logging/apm_data_dumper.h
index e49845f95f..5b3afd133a 100644
--- a/modules/audio_processing/logging/apm_data_dumper.h
+++ b/modules/audio_processing/logging/apm_data_dumper.h
@@ -19,8 +19,9 @@
 #include <unordered_map>
 #endif
 
+#include <span>
+
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #if WEBRTC_APM_DEBUG_DUMP == 1
 #include "common_audio/wav_file.h"
 #include "rtc_base/checks.h"
@@ -123,12 +124,12 @@ class ApmDataDumper {
                [[maybe_unused]] const double* v,
                [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
-    DumpRaw(name, ArrayView<const double>(v, v_length), dump_set);
+    DumpRaw(name, std::span<const double>(v, v_length), dump_set);
 #endif
   }
 
   void DumpRaw([[maybe_unused]] absl::string_view name,
-               [[maybe_unused]] ArrayView<const double> v,
+               [[maybe_unused]] std::span<const double> v,
                [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
     if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
@@ -160,12 +161,12 @@ class ApmDataDumper {
                [[maybe_unused]] const float* v,
                [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
-    DumpRaw(name, ArrayView<const float>(v, v_length), dump_set);
+    DumpRaw(name, std::span<const float>(v, v_length), dump_set);
 #endif
   }
 
   void DumpRaw([[maybe_unused]] absl::string_view name,
-               [[maybe_unused]] ArrayView<const float> v,
+               [[maybe_unused]] std::span<const float> v,
                [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
     if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
@@ -196,12 +197,12 @@ class ApmDataDumper {
                [[maybe_unused]] const bool* v,
                [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
-    DumpRaw(name, ArrayView<const bool>(v, v_length), dump_set);
+    DumpRaw(name, std::span<const bool>(v, v_length), dump_set);
 #endif
   }
 
   void DumpRaw([[maybe_unused]] absl::string_view name,
-               [[maybe_unused]] ArrayView<const bool> v,
+               [[maybe_unused]] std::span<const bool> v,
                [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
     if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
@@ -236,12 +237,12 @@ class ApmDataDumper {
                [[maybe_unused]] const int16_t* v,
                [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
-    DumpRaw(name, ArrayView<const int16_t>(v, v_length), dump_set);
+    DumpRaw(name, std::span<const int16_t>(v, v_length), dump_set);
 #endif
   }
 
   void DumpRaw([[maybe_unused]] absl::string_view name,
-               [[maybe_unused]] ArrayView<const int16_t> v,
+               [[maybe_unused]] std::span<const int16_t> v,
                [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
     if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
@@ -273,12 +274,12 @@ class ApmDataDumper {
                [[maybe_unused]] const int32_t* v,
                [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
-    DumpRaw(name, ArrayView<const int32_t>(v, v_length), dump_set);
+    DumpRaw(name, std::span<const int32_t>(v, v_length), dump_set);
 #endif
   }
 
   void DumpRaw([[maybe_unused]] absl::string_view name,
-               [[maybe_unused]] ArrayView<const int32_t> v,
+               [[maybe_unused]] std::span<const int32_t> v,
                [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
     if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
@@ -310,12 +311,12 @@ class ApmDataDumper {
                [[maybe_unused]] const size_t* v,
                [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
-    DumpRaw(name, ArrayView<const size_t>(v, v_length), dump_set);
+    DumpRaw(name, std::span<const size_t>(v, v_length), dump_set);
 #endif
   }
 
   void DumpRaw([[maybe_unused]] absl::string_view name,
-               [[maybe_unused]] ArrayView<const size_t> v,
+               [[maybe_unused]] std::span<const size_t> v,
                [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
     if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
@@ -335,13 +336,13 @@ class ApmDataDumper {
                [[maybe_unused]] int num_channels,
                [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
-    DumpWav(name, ArrayView<const float>(v, v_length), sample_rate_hz,
+    DumpWav(name, std::span<const float>(v, v_length), sample_rate_hz,
             num_channels, dump_set);
 #endif
   }
 
   void DumpWav([[maybe_unused]] absl::string_view name,
-               [[maybe_unused]] ArrayView<const float> v,
+               [[maybe_unused]] std::span<const float> v,
                [[maybe_unused]] int sample_rate_hz,
                [[maybe_unused]] int num_channels,
                [[maybe_unused]] int dump_set = kDefaultDumpSet) {
diff --git a/modules/audio_processing/ns/BUILD.gn b/modules/audio_processing/ns/BUILD.gn
index e90ef0e06a..83cce6076e 100644
--- a/modules/audio_processing/ns/BUILD.gn
+++ b/modules/audio_processing/ns/BUILD.gn
@@ -52,7 +52,6 @@ rtc_static_library("ns") {
     "..:apm_logging",
     "..:audio_buffer",
     "..:high_pass_filter",
-    "../../../api:array_view",
     "../../../common_audio:common_audio_c",
     "../../../common_audio/third_party/ooura:fft_size_128",
     "../../../common_audio/third_party/ooura:fft_size_256",
@@ -78,7 +77,6 @@ if (rtc_include_tests) {
       "..:audio_buffer",
       "..:audio_processing",
       "..:high_pass_filter",
-      "../../../api:array_view",
       "../../../rtc_base:checks",
       "../../../rtc_base:safe_minmax",
       "../../../rtc_base:stringutils",
diff --git a/modules/audio_processing/ns/fast_math.cc b/modules/audio_processing/ns/fast_math.cc
index 7d52252201..9f5ed6cfb1 100644
--- a/modules/audio_processing/ns/fast_math.cc
+++ b/modules/audio_processing/ns/fast_math.cc
@@ -14,8 +14,8 @@
 #include <cstddef>
 #include <cstdint>
 #include <numbers>
+#include <span>
 
-#include "api/array_view.h"
 #include "rtc_base/checks.h"
 
 namespace webrtc {
@@ -60,7 +60,7 @@ float LogApproximation(float x) {
   return FastLog2f(x) * kLogOf2;
 }
 
-void LogApproximation(ArrayView<const float> x, ArrayView<float> y) {
+void LogApproximation(std::span<const float> x, std::span<float> y) {
   for (size_t k = 0; k < x.size(); ++k) {
     y[k] = LogApproximation(x[k]);
   }
@@ -71,13 +71,13 @@ float ExpApproximation(float x) {
   return PowApproximation(10.f, x * kLog10Ofe);
 }
 
-void ExpApproximation(ArrayView<const float> x, ArrayView<float> y) {
+void ExpApproximation(std::span<const float> x, std::span<float> y) {
   for (size_t k = 0; k < x.size(); ++k) {
     y[k] = ExpApproximation(x[k]);
   }
 }
 
-void ExpApproximationSignFlip(ArrayView<const float> x, ArrayView<float> y) {
+void ExpApproximationSignFlip(std::span<const float> x, std::span<float> y) {
   for (size_t k = 0; k < x.size(); ++k) {
     y[k] = ExpApproximation(-x[k]);
   }
diff --git a/modules/audio_processing/ns/fast_math.h b/modules/audio_processing/ns/fast_math.h
index 598b31cfd7..1064780e7f 100644
--- a/modules/audio_processing/ns/fast_math.h
+++ b/modules/audio_processing/ns/fast_math.h
@@ -11,7 +11,7 @@
 #ifndef MODULES_AUDIO_PROCESSING_NS_FAST_MATH_H_
 #define MODULES_AUDIO_PROCESSING_NS_FAST_MATH_H_
 
-#include "api/array_view.h"
+#include <span>
 
 namespace webrtc {
 
@@ -20,7 +20,7 @@ float SqrtFastApproximation(float f);
 
 // Log base conversion log(x) = log2(x)/log2(e).
 float LogApproximation(float x);
-void LogApproximation(ArrayView<const float> x, ArrayView<float> y);
+void LogApproximation(std::span<const float> x, std::span<float> y);
 
 // 2^x approximation.
 float Pow2Approximation(float p);
@@ -30,8 +30,8 @@ float PowApproximation(float x, float p);
 
 // e^x approximation.
 float ExpApproximation(float x);
-void ExpApproximation(ArrayView<const float> x, ArrayView<float> y);
-void ExpApproximationSignFlip(ArrayView<const float> x, ArrayView<float> y);
+void ExpApproximation(std::span<const float> x, std::span<float> y);
+void ExpApproximationSignFlip(std::span<const float> x, std::span<float> y);
 }  // namespace webrtc
 
 #endif  // MODULES_AUDIO_PROCESSING_NS_FAST_MATH_H_
diff --git a/modules/audio_processing/ns/histograms.h b/modules/audio_processing/ns/histograms.h
index 3f4445486b..3f65917245 100644
--- a/modules/audio_processing/ns/histograms.h
+++ b/modules/audio_processing/ns/histograms.h
@@ -12,8 +12,8 @@
 #define MODULES_AUDIO_PROCESSING_NS_HISTOGRAMS_H_
 
 #include <array>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/ns/signal_model.h"
 
 namespace webrtc {
@@ -35,11 +35,11 @@ class Histograms {
   void Update(const SignalModel& features_);
 
   // Methods for accessing the histograms.
-  ArrayView<const int, kHistogramSize> get_lrt() const { return lrt_; }
-  ArrayView<const int, kHistogramSize> get_spectral_flatness() const {
+  std::span<const int, kHistogramSize> get_lrt() const { return lrt_; }
+  std::span<const int, kHistogramSize> get_spectral_flatness() const {
     return spectral_flatness_;
   }
-  ArrayView<const int, kHistogramSize> get_spectral_diff() const {
+  std::span<const int, kHistogramSize> get_spectral_diff() const {
     return spectral_diff_;
   }
 
diff --git a/modules/audio_processing/ns/noise_estimator.cc b/modules/audio_processing/ns/noise_estimator.cc
index 00b647ca35..4661b6d32a 100644
--- a/modules/audio_processing/ns/noise_estimator.cc
+++ b/modules/audio_processing/ns/noise_estimator.cc
@@ -15,8 +15,8 @@
 #include <cstddef>
 #include <cstdint>
 #include <numbers>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/ns/fast_math.h"
 #include "modules/audio_processing/ns/ns_common.h"
 #include "modules/audio_processing/ns/suppression_params.h"
@@ -70,7 +70,7 @@ void NoiseEstimator::PrepareAnalysis() {
 
 void NoiseEstimator::PreUpdate(
     int32_t num_analyzed_frames,
-    ArrayView<const float, kFftSizeBy2Plus1> signal_spectrum,
+    std::span<const float, kFftSizeBy2Plus1> signal_spectrum,
     float signal_spectral_sum) {
   quantile_noise_estimator_.Estimate(signal_spectrum, noise_spectrum_);
 
@@ -159,8 +159,8 @@ void NoiseEstimator::PreUpdate(
 }
 
 void NoiseEstimator::PostUpdate(
-    ArrayView<const float> speech_probability,
-    ArrayView<const float, kFftSizeBy2Plus1> signal_spectrum) {
+    std::span<const float> speech_probability,
+    std::span<const float, kFftSizeBy2Plus1> signal_spectrum) {
   // Time-avg parameter for noise_spectrum update.
   constexpr float kNoiseUpdate = 0.9f;
 
diff --git a/modules/audio_processing/ns/noise_estimator.h b/modules/audio_processing/ns/noise_estimator.h
index 2ae89d5ed3..fd06b3b138 100644
--- a/modules/audio_processing/ns/noise_estimator.h
+++ b/modules/audio_processing/ns/noise_estimator.h
@@ -13,8 +13,8 @@
 
 #include <array>
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/ns/ns_common.h"
 #include "modules/audio_processing/ns/quantile_noise_estimator.h"
 #include "modules/audio_processing/ns/suppression_params.h"
@@ -32,29 +32,29 @@ class NoiseEstimator {
 
   // Performs the first step of the estimator update.
   void PreUpdate(int32_t num_analyzed_frames,
-                 ArrayView<const float, kFftSizeBy2Plus1> signal_spectrum,
+                 std::span<const float, kFftSizeBy2Plus1> signal_spectrum,
                  float signal_spectral_sum);
 
   // Performs the second step of the estimator update.
-  void PostUpdate(ArrayView<const float> speech_probability,
-                  ArrayView<const float, kFftSizeBy2Plus1> signal_spectrum);
+  void PostUpdate(std::span<const float> speech_probability,
+                  std::span<const float, kFftSizeBy2Plus1> signal_spectrum);
 
   // Returns the noise spectral estimate.
-  ArrayView<const float, kFftSizeBy2Plus1> get_noise_spectrum() const {
+  std::span<const float, kFftSizeBy2Plus1> get_noise_spectrum() const {
     return noise_spectrum_;
   }
 
   // Returns the noise from the previous frame.
-  ArrayView<const float, kFftSizeBy2Plus1> get_prev_noise_spectrum() const {
+  std::span<const float, kFftSizeBy2Plus1> get_prev_noise_spectrum() const {
     return prev_noise_spectrum_;
   }
 
   // Returns a noise spectral estimate based on white and pink noise parameters.
-  ArrayView<const float, kFftSizeBy2Plus1> get_parametric_noise_spectrum()
+  std::span<const float, kFftSizeBy2Plus1> get_parametric_noise_spectrum()
       const {
     return parametric_noise_spectrum_;
   }
-  ArrayView<const float, kFftSizeBy2Plus1> get_conservative_noise_spectrum()
+  std::span<const float, kFftSizeBy2Plus1> get_conservative_noise_spectrum()
       const {
     return conservative_noise_spectrum_;
   }
diff --git a/modules/audio_processing/ns/noise_suppressor.cc b/modules/audio_processing/ns/noise_suppressor.cc
index 47030c2acd..5f535e5124 100644
--- a/modules/audio_processing/ns/noise_suppressor.cc
+++ b/modules/audio_processing/ns/noise_suppressor.cc
@@ -16,8 +16,8 @@
 #include <cstdlib>
 #include <cstring>
 #include <memory>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/audio_buffer.h"
 #include "modules/audio_processing/ns/fast_math.h"
 #include "modules/audio_processing/ns/ns_common.h"
@@ -76,7 +76,7 @@ constexpr std::array<float, 96> kBlocks160w256FirstHalf = {
     0.99986614f};
 
 // Applies the filterbank window to a buffer.
-void ApplyFilterBankWindow(ArrayView<float, kFftSize> x) {
+void ApplyFilterBankWindow(std::span<float, kFftSize> x) {
   for (size_t i = 0; i < 96; ++i) {
     x[i] = kBlocks160w256FirstHalf[i] * x[i];
   }
@@ -88,9 +88,9 @@ void ApplyFilterBankWindow(ArrayView<float, kFftSize> x) {
 }
 
 // Extends a frame with previous data.
-void FormExtendedFrame(ArrayView<const float, kNsFrameSize> frame,
-                       ArrayView<float, kFftSize - kNsFrameSize> old_data,
-                       ArrayView<float, kFftSize> extended_frame) {
+void FormExtendedFrame(std::span<const float, kNsFrameSize> frame,
+                       std::span<float, kFftSize - kNsFrameSize> old_data,
+                       std::span<float, kFftSize> extended_frame) {
   std::copy(old_data.begin(), old_data.end(), extended_frame.begin());
   std::copy(frame.begin(), frame.end(),
             extended_frame.begin() + old_data.size());
@@ -99,9 +99,9 @@ void FormExtendedFrame(ArrayView<const float, kNsFrameSize> frame,
 }
 
 // Uses overlap-and-add to produce an output frame.
-void OverlapAndAdd(ArrayView<const float, kFftSize> extended_frame,
-                   ArrayView<float, kOverlapSize> overlap_memory,
-                   ArrayView<float, kNsFrameSize> output_frame) {
+void OverlapAndAdd(std::span<const float, kFftSize> extended_frame,
+                   std::span<float, kOverlapSize> overlap_memory,
+                   std::span<float, kNsFrameSize> output_frame) {
   for (size_t i = 0; i < kOverlapSize; ++i) {
     output_frame[i] = overlap_memory[i] + extended_frame[i];
   }
@@ -113,9 +113,9 @@ void OverlapAndAdd(ArrayView<const float, kFftSize> extended_frame,
 }
 
 // Produces a delayed frame.
-void DelaySignal(ArrayView<const float, kNsFrameSize> frame,
-                 ArrayView<float, kFftSize - kNsFrameSize> delay_buffer,
-                 ArrayView<float, kNsFrameSize> delayed_frame) {
+void DelaySignal(std::span<const float, kNsFrameSize> frame,
+                 std::span<float, kFftSize - kNsFrameSize> delay_buffer,
+                 std::span<float, kNsFrameSize> delayed_frame) {
   constexpr size_t kSamplesFromFrame = kNsFrameSize - (kFftSize - kNsFrameSize);
   std::copy(delay_buffer.begin(), delay_buffer.end(), delayed_frame.begin());
   std::copy(frame.begin(), frame.begin() + kSamplesFromFrame,
@@ -126,7 +126,7 @@ void DelaySignal(ArrayView<const float, kNsFrameSize> frame,
 }
 
 // Computes the energy of an extended frame.
-float ComputeEnergyOfExtendedFrame(ArrayView<const float, kFftSize> x) {
+float ComputeEnergyOfExtendedFrame(std::span<const float, kFftSize> x) {
   float energy = 0.f;
   for (float x_k : x) {
     energy += x_k * x_k;
@@ -137,8 +137,8 @@ float ComputeEnergyOfExtendedFrame(ArrayView<const float, kFftSize> x) {
 
 // Computes the energy of an extended frame based on its subcomponents.
 float ComputeEnergyOfExtendedFrame(
-    ArrayView<const float, kNsFrameSize> frame,
-    ArrayView<float, kFftSize - kNsFrameSize> old_data) {
+    std::span<const float, kNsFrameSize> frame,
+    std::span<float, kFftSize - kNsFrameSize> old_data) {
   float energy = 0.f;
   for (float v : old_data) {
     energy += v * v;
@@ -152,9 +152,9 @@ float ComputeEnergyOfExtendedFrame(
 
 // Computes the magnitude spectrum based on an FFT output.
 void ComputeMagnitudeSpectrum(
-    ArrayView<const float, kFftSize> real,
-    ArrayView<const float, kFftSize> imag,
-    ArrayView<float, kFftSizeBy2Plus1> signal_spectrum) {
+    std::span<const float, kFftSize> real,
+    std::span<const float, kFftSize> imag,
+    std::span<float, kFftSizeBy2Plus1> signal_spectrum) {
   signal_spectrum[0] = fabsf(real[0]) + 1.f;
   signal_spectrum[kFftSizeBy2Plus1 - 1] =
       fabsf(real[kFftSizeBy2Plus1 - 1]) + 1.f;
@@ -166,13 +166,13 @@ void ComputeMagnitudeSpectrum(
 }
 
 // Compute prior and post SNR.
-void ComputeSnr(ArrayView<const float, kFftSizeBy2Plus1> filter,
-                ArrayView<const float> prev_signal_spectrum,
-                ArrayView<const float> signal_spectrum,
-                ArrayView<const float> prev_noise_spectrum,
-                ArrayView<const float> noise_spectrum,
-                ArrayView<float> prior_snr,
-                ArrayView<float> post_snr) {
+void ComputeSnr(std::span<const float, kFftSizeBy2Plus1> filter,
+                std::span<const float> prev_signal_spectrum,
+                std::span<const float> signal_spectrum,
+                std::span<const float> prev_noise_spectrum,
+                std::span<const float> noise_spectrum,
+                std::span<float> prior_snr,
+                std::span<float> post_snr) {
   for (size_t i = 0; i < kFftSizeBy2Plus1; ++i) {
     // Previous post SNR.
     // Previous estimate: based on previous frame with gain filter.
@@ -193,10 +193,10 @@ void ComputeSnr(ArrayView<const float, kFftSizeBy2Plus1> filter,
 // Computes the attenuating gain for the noise suppression of the upper bands.
 float ComputeUpperBandsGain(
     float minimum_attenuating_gain,
-    ArrayView<const float, kFftSizeBy2Plus1> filter,
-    ArrayView<const float> speech_probability,
-    ArrayView<const float, kFftSizeBy2Plus1> prev_analysis_signal_spectrum,
-    ArrayView<const float, kFftSizeBy2Plus1> signal_spectrum) {
+    std::span<const float, kFftSizeBy2Plus1> filter,
+    std::span<const float> speech_probability,
+    std::span<const float, kFftSizeBy2Plus1> prev_analysis_signal_spectrum,
+    std::span<const float, kFftSizeBy2Plus1> signal_spectrum) {
   // Average speech prob and filter gain for the end of the lowest band.
   constexpr int kNumAvgBins = 32;
   constexpr float kOneByNumAvgBins = 1.f / kNumAvgBins;
@@ -277,13 +277,13 @@ NoiseSuppressor::NoiseSuppressor(const NsConfig& config,
 }
 
 void NoiseSuppressor::AggregateWienerFilters(
-    ArrayView<float, kFftSizeBy2Plus1> filter) const {
-  ArrayView<const float, kFftSizeBy2Plus1> filter0 =
+    std::span<float, kFftSizeBy2Plus1> filter) const {
+  std::span<const float, kFftSizeBy2Plus1> filter0 =
       channels_[0]->wiener_filter.get_filter();
   std::copy(filter0.begin(), filter0.end(), filter.begin());
 
   for (size_t ch = 1; ch < num_channels_; ++ch) {
-    ArrayView<const float, kFftSizeBy2Plus1> filter_ch =
+    std::span<const float, kFftSizeBy2Plus1> filter_ch =
         channels_[ch]->wiener_filter.get_filter();
 
     for (size_t k = 0; k < kFftSizeBy2Plus1; ++k) {
@@ -301,7 +301,7 @@ void NoiseSuppressor::Analyze(const AudioBuffer& audio) {
   // Check for zero frames.
   bool zero_frame = true;
   for (size_t ch = 0; ch < num_channels_; ++ch) {
-    ArrayView<const float, kNsFrameSize> y_band0(
+    std::span<const float, kNsFrameSize> y_band0(
         &audio.split_bands_const(ch)[0][0], kNsFrameSize);
     float energy = ComputeEnergyOfExtendedFrame(
         y_band0, channels_[ch]->analyze_analysis_memory);
@@ -331,7 +331,7 @@ void NoiseSuppressor::Analyze(const AudioBuffer& audio) {
   // Analyze all channels.
   for (size_t ch = 0; ch < num_channels_; ++ch) {
     std::unique_ptr<ChannelState>& ch_p = channels_[ch];
-    ArrayView<const float, kNsFrameSize> y_band0(
+    std::span<const float, kNsFrameSize> y_band0(
         &audio.split_bands_const(ch)[0][0], kNsFrameSize);
 
     // Form an extended frame and apply analysis filter bank windowing.
@@ -389,34 +389,34 @@ void NoiseSuppressor::Analyze(const AudioBuffer& audio) {
 void NoiseSuppressor::Process(AudioBuffer* audio) {
   // Select the space for storing data during the processing.
   std::array<FilterBankState, kMaxNumChannelsOnStack> filter_bank_states_stack;
-  ArrayView<FilterBankState> filter_bank_states(filter_bank_states_stack.data(),
+  std::span<FilterBankState> filter_bank_states(filter_bank_states_stack.data(),
                                                 num_channels_);
   std::array<float, kMaxNumChannelsOnStack> upper_band_gains_stack;
-  ArrayView<float> upper_band_gains(upper_band_gains_stack.data(),
+  std::span<float> upper_band_gains(upper_band_gains_stack.data(),
                                     num_channels_);
   std::array<float, kMaxNumChannelsOnStack> energies_before_filtering_stack;
-  ArrayView<float> energies_before_filtering(
+  std::span<float> energies_before_filtering(
       energies_before_filtering_stack.data(), num_channels_);
   std::array<float, kMaxNumChannelsOnStack> gain_adjustments_stack;
-  ArrayView<float> gain_adjustments(gain_adjustments_stack.data(),
+  std::span<float> gain_adjustments(gain_adjustments_stack.data(),
                                     num_channels_);
   if (NumChannelsOnHeap(num_channels_) > 0) {
     // If the stack-allocated space is too small, use the heap for storing the
     // data.
-    filter_bank_states = ArrayView<FilterBankState>(
+    filter_bank_states = std::span<FilterBankState>(
         filter_bank_states_heap_.data(), num_channels_);
     upper_band_gains =
-        ArrayView<float>(upper_band_gains_heap_.data(), num_channels_);
+        std::span<float>(upper_band_gains_heap_.data(), num_channels_);
     energies_before_filtering =
-        ArrayView<float>(energies_before_filtering_heap_.data(), num_channels_);
+        std::span<float>(energies_before_filtering_heap_.data(), num_channels_);
     gain_adjustments =
-        ArrayView<float>(gain_adjustments_heap_.data(), num_channels_);
+        std::span<float>(gain_adjustments_heap_.data(), num_channels_);
   }
 
   // Compute the suppression filters for all channels.
   for (size_t ch = 0; ch < num_channels_; ++ch) {
     // Form an extended frame and apply analysis filter bank windowing.
-    ArrayView<float, kNsFrameSize> y_band0(&audio->split_bands(ch)[0][0],
+    std::span<float, kNsFrameSize> y_band0(&audio->split_bands(ch)[0][0],
                                            kNsFrameSize);
 
     FormExtendedFrame(y_band0, channels_[ch]->process_analysis_memory,
@@ -463,7 +463,7 @@ void NoiseSuppressor::Process(AudioBuffer* audio) {
 
   // Aggregate the Wiener filters for all channels.
   std::array<float, kFftSizeBy2Plus1> filter_data;
-  ArrayView<const float, kFftSizeBy2Plus1> filter = filter_data;
+  std::span<const float, kFftSizeBy2Plus1> filter = filter_data;
   if (num_channels_ == 1) {
     filter = channels_[0]->wiener_filter.get_filter();
   } else {
@@ -515,7 +515,7 @@ void NoiseSuppressor::Process(AudioBuffer* audio) {
 
   // Use overlap-and-add to form the output frame of the lowest band.
   for (size_t ch = 0; ch < num_channels_; ++ch) {
-    ArrayView<float, kNsFrameSize> y_band0(&audio->split_bands(ch)[0][0],
+    std::span<float, kNsFrameSize> y_band0(&audio->split_bands(ch)[0][0],
                                            kNsFrameSize);
     OverlapAndAdd(filter_bank_states[ch].extended_frame,
                   channels_[ch]->process_synthesis_memory, y_band0);
@@ -533,7 +533,7 @@ void NoiseSuppressor::Process(AudioBuffer* audio) {
       for (size_t b = 1; b < num_bands_; ++b) {
         // Delay the upper bands to match the delay of the filterbank applied to
         // the lowest band.
-        ArrayView<float, kNsFrameSize> y_band(&audio->split_bands(ch)[b][0],
+        std::span<float, kNsFrameSize> y_band(&audio->split_bands(ch)[b][0],
                                               kNsFrameSize);
         std::array<float, kNsFrameSize> delayed_frame;
         DelaySignal(y_band, channels_[ch]->process_delay_memory[b - 1],
@@ -550,7 +550,7 @@ void NoiseSuppressor::Process(AudioBuffer* audio) {
   // Limit the output the allowed range.
   for (size_t ch = 0; ch < num_channels_; ++ch) {
     for (size_t b = 0; b < num_bands_; ++b) {
-      ArrayView<float, kNsFrameSize> y_band(&audio->split_bands(ch)[b][0],
+      std::span<float, kNsFrameSize> y_band(&audio->split_bands(ch)[b][0],
                                             kNsFrameSize);
       for (size_t j = 0; j < kNsFrameSize; j++) {
         y_band[j] = std::min(std::max(y_band[j], -32768.f), 32767.f);
diff --git a/modules/audio_processing/ns/noise_suppressor.h b/modules/audio_processing/ns/noise_suppressor.h
index 59312ef0fb..5e71decf59 100644
--- a/modules/audio_processing/ns/noise_suppressor.h
+++ b/modules/audio_processing/ns/noise_suppressor.h
@@ -15,9 +15,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/audio_buffer.h"
 #include "modules/audio_processing/ns/noise_estimator.h"
 #include "modules/audio_processing/ns/ns_common.h"
@@ -87,7 +87,7 @@ class NoiseSuppressor {
   std::vector<std::unique_ptr<ChannelState>> channels_;
 
   // Aggregates the Wiener filters into a single filter to use.
-  void AggregateWienerFilters(ArrayView<float, kFftSizeBy2Plus1> filter) const;
+  void AggregateWienerFilters(std::span<float, kFftSizeBy2Plus1> filter) const;
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_processing/ns/ns_fft.cc b/modules/audio_processing/ns/ns_fft.cc
index 07e63739aa..43112191db 100644
--- a/modules/audio_processing/ns/ns_fft.cc
+++ b/modules/audio_processing/ns/ns_fft.cc
@@ -12,8 +12,8 @@
 
 #include <array>
 #include <cstddef>
+#include <span>
 
-#include "api/array_view.h"
 #include "common_audio/third_party/ooura/fft_size_256/fft4g.h"
 #include "modules/audio_processing/ns/ns_common.h"
 
@@ -29,9 +29,9 @@ NrFft::NrFft() : bit_reversal_state_(kFftSize / 2), tables_(kFftSize / 2) {
               tables_.data());
 }
 
-void NrFft::Fft(ArrayView<float, kFftSize> time_data,
-                ArrayView<float, kFftSize> real,
-                ArrayView<float, kFftSize> imag) {
+void NrFft::Fft(std::span<float, kFftSize> time_data,
+                std::span<float, kFftSize> real,
+                std::span<float, kFftSize> imag) {
   WebRtc_rdft(kFftSize, 1, time_data.data(), bit_reversal_state_.data(),
               tables_.data());
 
@@ -47,9 +47,9 @@ void NrFft::Fft(ArrayView<float, kFftSize> time_data,
   }
 }
 
-void NrFft::Ifft(ArrayView<const float> real,
-                 ArrayView<const float> imag,
-                 ArrayView<float> time_data) {
+void NrFft::Ifft(std::span<const float> real,
+                 std::span<const float> imag,
+                 std::span<float> time_data) {
   time_data[0] = real[0];
   time_data[1] = real[kFftSizeBy2Plus1 - 1];
   for (size_t i = 1; i < kFftSizeBy2Plus1 - 1; ++i) {
diff --git a/modules/audio_processing/ns/ns_fft.h b/modules/audio_processing/ns/ns_fft.h
index 78872512da..358d23157b 100644
--- a/modules/audio_processing/ns/ns_fft.h
+++ b/modules/audio_processing/ns/ns_fft.h
@@ -12,9 +12,9 @@
 #define MODULES_AUDIO_PROCESSING_NS_NS_FFT_H_
 
 #include <cstddef>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/ns/ns_common.h"
 
 namespace webrtc {
@@ -27,14 +27,14 @@ class NrFft {
   NrFft& operator=(const NrFft&) = delete;
 
   // Transforms the signal from time to frequency domain.
-  void Fft(ArrayView<float, kFftSize> time_data,
-           ArrayView<float, kFftSize> real,
-           ArrayView<float, kFftSize> imag);
+  void Fft(std::span<float, kFftSize> time_data,
+           std::span<float, kFftSize> real,
+           std::span<float, kFftSize> imag);
 
   // Transforms the signal from frequency to time domain.
-  void Ifft(ArrayView<const float> real,
-            ArrayView<const float> imag,
-            ArrayView<float> time_data);
+  void Ifft(std::span<const float> real,
+            std::span<const float> imag,
+            std::span<float> time_data);
 
  private:
   std::vector<size_t> bit_reversal_state_;
diff --git a/modules/audio_processing/ns/prior_signal_model_estimator.cc b/modules/audio_processing/ns/prior_signal_model_estimator.cc
index f401764554..dcfe297f2e 100644
--- a/modules/audio_processing/ns/prior_signal_model_estimator.cc
+++ b/modules/audio_processing/ns/prior_signal_model_estimator.cc
@@ -12,8 +12,8 @@
 
 #include <algorithm>
 #include <cmath>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/ns/histograms.h"
 #include "modules/audio_processing/ns/ns_common.h"
 #include "rtc_base/checks.h"
@@ -25,7 +25,7 @@ namespace {
 // Identifies the first of the two largest peaks in the histogram.
 void FindFirstOfTwoLargestPeaks(
     float bin_size,
-    ArrayView<const int, kHistogramSize> spectral_flatness,
+    std::span<const int, kHistogramSize> spectral_flatness,
     float* peak_position,
     int* peak_weight) {
   RTC_DCHECK(peak_position);
@@ -66,7 +66,7 @@ void FindFirstOfTwoLargestPeaks(
   }
 }
 
-void UpdateLrt(ArrayView<const int, kHistogramSize> lrt_histogram,
+void UpdateLrt(std::span<const int, kHistogramSize> lrt_histogram,
                float* prior_model_lrt,
                bool* low_lrt_fluctuations) {
   RTC_DCHECK(prior_model_lrt);
diff --git a/modules/audio_processing/ns/quantile_noise_estimator.cc b/modules/audio_processing/ns/quantile_noise_estimator.cc
index fec852b531..d322f6294d 100644
--- a/modules/audio_processing/ns/quantile_noise_estimator.cc
+++ b/modules/audio_processing/ns/quantile_noise_estimator.cc
@@ -14,8 +14,8 @@
 #include <array>
 #include <cmath>
 #include <cstddef>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/ns/fast_math.h"
 #include "modules/audio_processing/ns/ns_common.h"
 
@@ -33,8 +33,8 @@ QuantileNoiseEstimator::QuantileNoiseEstimator() {
 }
 
 void QuantileNoiseEstimator::Estimate(
-    ArrayView<const float, kFftSizeBy2Plus1> signal_spectrum,
-    ArrayView<float, kFftSizeBy2Plus1> noise_spectrum) {
+    std::span<const float, kFftSizeBy2Plus1> signal_spectrum,
+    std::span<float, kFftSizeBy2Plus1> noise_spectrum) {
   std::array<float, kFftSizeBy2Plus1> log_spectrum;
   LogApproximation(signal_spectrum, log_spectrum);
 
@@ -82,7 +82,7 @@ void QuantileNoiseEstimator::Estimate(
 
   if (quantile_index_to_return >= 0) {
     ExpApproximation(
-        ArrayView<const float>(&log_quantile_[quantile_index_to_return],
+        std::span<const float>(&log_quantile_[quantile_index_to_return],
                                kFftSizeBy2Plus1),
         quantile_);
   }
diff --git a/modules/audio_processing/ns/quantile_noise_estimator.h b/modules/audio_processing/ns/quantile_noise_estimator.h
index 79a1b3e1cc..6eaf9d2e97 100644
--- a/modules/audio_processing/ns/quantile_noise_estimator.h
+++ b/modules/audio_processing/ns/quantile_noise_estimator.h
@@ -14,8 +14,8 @@
 #include <math.h>
 
 #include <array>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/ns/ns_common.h"
 
 namespace webrtc {
@@ -30,8 +30,8 @@ class QuantileNoiseEstimator {
   QuantileNoiseEstimator& operator=(const QuantileNoiseEstimator&) = delete;
 
   // Estimate noise.
-  void Estimate(ArrayView<const float, kFftSizeBy2Plus1> signal_spectrum,
-                ArrayView<float, kFftSizeBy2Plus1> noise_spectrum);
+  void Estimate(std::span<const float, kFftSizeBy2Plus1> signal_spectrum,
+                std::span<float, kFftSizeBy2Plus1> noise_spectrum);
 
  private:
   std::array<float, kSimult * kFftSizeBy2Plus1> density_;
diff --git a/modules/audio_processing/ns/signal_model_estimator.cc b/modules/audio_processing/ns/signal_model_estimator.cc
index 84f175f71f..0c45bd2d38 100644
--- a/modules/audio_processing/ns/signal_model_estimator.cc
+++ b/modules/audio_processing/ns/signal_model_estimator.cc
@@ -12,8 +12,8 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/ns/fast_math.h"
 #include "modules/audio_processing/ns/ns_common.h"
 #include "rtc_base/checks.h"
@@ -27,8 +27,8 @@ constexpr float kOneByFftSizeBy2Plus1 = 1.f / kFftSizeBy2Plus1;
 // Computes the difference measure between input spectrum and a template/learned
 // noise spectrum.
 float ComputeSpectralDiff(
-    ArrayView<const float, kFftSizeBy2Plus1> conservative_noise_spectrum,
-    ArrayView<const float, kFftSizeBy2Plus1> signal_spectrum,
+    std::span<const float, kFftSizeBy2Plus1> conservative_noise_spectrum,
+    std::span<const float, kFftSizeBy2Plus1> signal_spectrum,
     float signal_spectral_sum,
     float diff_normalization) {
   // spectral_diff = var(signal_spectrum) - cov(signal_spectrum, magnAvgPause)^2
@@ -67,7 +67,7 @@ float ComputeSpectralDiff(
 
 // Updates the spectral flatness based on the input spectrum.
 void UpdateSpectralFlatness(
-    ArrayView<const float, kFftSizeBy2Plus1> signal_spectrum,
+    std::span<const float, kFftSizeBy2Plus1> signal_spectrum,
     float signal_spectral_sum,
     float* spectral_flatness) {
   RTC_DCHECK(spectral_flatness);
@@ -100,9 +100,9 @@ void UpdateSpectralFlatness(
 }
 
 // Updates the log LRT measures.
-void UpdateSpectralLrt(ArrayView<const float, kFftSizeBy2Plus1> prior_snr,
-                       ArrayView<const float, kFftSizeBy2Plus1> post_snr,
-                       ArrayView<float, kFftSizeBy2Plus1> avg_log_lrt,
+void UpdateSpectralLrt(std::span<const float, kFftSizeBy2Plus1> prior_snr,
+                       std::span<const float, kFftSizeBy2Plus1> post_snr,
+                       std::span<float, kFftSizeBy2Plus1> avg_log_lrt,
                        float* lrt) {
   RTC_DCHECK(lrt);
 
@@ -135,10 +135,10 @@ void SignalModelEstimator::AdjustNormalization(int32_t num_analyzed_frames,
 
 // Update the noise features.
 void SignalModelEstimator::Update(
-    ArrayView<const float, kFftSizeBy2Plus1> prior_snr,
-    ArrayView<const float, kFftSizeBy2Plus1> post_snr,
-    ArrayView<const float, kFftSizeBy2Plus1> conservative_noise_spectrum,
-    ArrayView<const float, kFftSizeBy2Plus1> signal_spectrum,
+    std::span<const float, kFftSizeBy2Plus1> prior_snr,
+    std::span<const float, kFftSizeBy2Plus1> post_snr,
+    std::span<const float, kFftSizeBy2Plus1> conservative_noise_spectrum,
+    std::span<const float, kFftSizeBy2Plus1> signal_spectrum,
     float signal_spectral_sum,
     float signal_energy) {
   // Compute spectral flatness on input spectrum.
diff --git a/modules/audio_processing/ns/signal_model_estimator.h b/modules/audio_processing/ns/signal_model_estimator.h
index 540e478661..ef0e5e4401 100644
--- a/modules/audio_processing/ns/signal_model_estimator.h
+++ b/modules/audio_processing/ns/signal_model_estimator.h
@@ -12,8 +12,8 @@
 #define MODULES_AUDIO_PROCESSING_NS_SIGNAL_MODEL_ESTIMATOR_H_
 
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/ns/histograms.h"
 #include "modules/audio_processing/ns/ns_common.h"
 #include "modules/audio_processing/ns/prior_signal_model.h"
@@ -32,10 +32,10 @@ class SignalModelEstimator {
   void AdjustNormalization(int32_t num_analyzed_frames, float signal_energy);
 
   void Update(
-      ArrayView<const float, kFftSizeBy2Plus1> prior_snr,
-      ArrayView<const float, kFftSizeBy2Plus1> post_snr,
-      ArrayView<const float, kFftSizeBy2Plus1> conservative_noise_spectrum,
-      ArrayView<const float, kFftSizeBy2Plus1> signal_spectrum,
+      std::span<const float, kFftSizeBy2Plus1> prior_snr,
+      std::span<const float, kFftSizeBy2Plus1> post_snr,
+      std::span<const float, kFftSizeBy2Plus1> conservative_noise_spectrum,
+      std::span<const float, kFftSizeBy2Plus1> signal_spectrum,
       float signal_spectral_sum,
       float signal_energy);
 
diff --git a/modules/audio_processing/ns/speech_probability_estimator.cc b/modules/audio_processing/ns/speech_probability_estimator.cc
index 0f7369382f..44b90fe047 100644
--- a/modules/audio_processing/ns/speech_probability_estimator.cc
+++ b/modules/audio_processing/ns/speech_probability_estimator.cc
@@ -15,8 +15,8 @@
 #include <cmath>
 #include <cstddef>
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/ns/fast_math.h"
 #include "modules/audio_processing/ns/ns_common.h"
 #include "modules/audio_processing/ns/prior_signal_model.h"
@@ -30,10 +30,10 @@ SpeechProbabilityEstimator::SpeechProbabilityEstimator() {
 
 void SpeechProbabilityEstimator::Update(
     int32_t num_analyzed_frames,
-    ArrayView<const float, kFftSizeBy2Plus1> prior_snr,
-    ArrayView<const float, kFftSizeBy2Plus1> post_snr,
-    ArrayView<const float, kFftSizeBy2Plus1> conservative_noise_spectrum,
-    ArrayView<const float, kFftSizeBy2Plus1> signal_spectrum,
+    std::span<const float, kFftSizeBy2Plus1> prior_snr,
+    std::span<const float, kFftSizeBy2Plus1> post_snr,
+    std::span<const float, kFftSizeBy2Plus1> conservative_noise_spectrum,
+    std::span<const float, kFftSizeBy2Plus1> signal_spectrum,
     float signal_spectral_sum,
     float signal_energy) {
   // Update models.
diff --git a/modules/audio_processing/ns/speech_probability_estimator.h b/modules/audio_processing/ns/speech_probability_estimator.h
index 8651e8a883..0516a6d461 100644
--- a/modules/audio_processing/ns/speech_probability_estimator.h
+++ b/modules/audio_processing/ns/speech_probability_estimator.h
@@ -13,8 +13,8 @@
 
 #include <array>
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/ns/ns_common.h"
 #include "modules/audio_processing/ns/signal_model_estimator.h"
 
@@ -31,15 +31,15 @@ class SpeechProbabilityEstimator {
   // Compute speech probability.
   void Update(
       int32_t num_analyzed_frames,
-      ArrayView<const float, kFftSizeBy2Plus1> prior_snr,
-      ArrayView<const float, kFftSizeBy2Plus1> post_snr,
-      ArrayView<const float, kFftSizeBy2Plus1> conservative_noise_spectrum,
-      ArrayView<const float, kFftSizeBy2Plus1> signal_spectrum,
+      std::span<const float, kFftSizeBy2Plus1> prior_snr,
+      std::span<const float, kFftSizeBy2Plus1> post_snr,
+      std::span<const float, kFftSizeBy2Plus1> conservative_noise_spectrum,
+      std::span<const float, kFftSizeBy2Plus1> signal_spectrum,
       float signal_spectral_sum,
       float signal_energy);
 
   float get_prior_probability() const { return prior_speech_prob_; }
-  ArrayView<const float> get_probability() { return speech_probability_; }
+  std::span<const float> get_probability() { return speech_probability_; }
 
  private:
   SignalModelEstimator signal_model_estimator_;
diff --git a/modules/audio_processing/ns/wiener_filter.cc b/modules/audio_processing/ns/wiener_filter.cc
index ff8ec4d638..521fd71297 100644
--- a/modules/audio_processing/ns/wiener_filter.cc
+++ b/modules/audio_processing/ns/wiener_filter.cc
@@ -15,8 +15,8 @@
 #include <cstdio>
 #include <cstdlib>
 #include <cstring>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/ns/fast_math.h"
 #include "modules/audio_processing/ns/ns_common.h"
 #include "modules/audio_processing/ns/suppression_params.h"
@@ -32,10 +32,10 @@ WienerFilter::WienerFilter(const SuppressionParams& suppression_params)
 
 void WienerFilter::Update(
     int32_t num_analyzed_frames,
-    ArrayView<const float, kFftSizeBy2Plus1> noise_spectrum,
-    ArrayView<const float, kFftSizeBy2Plus1> prev_noise_spectrum,
-    ArrayView<const float, kFftSizeBy2Plus1> parametric_noise_spectrum,
-    ArrayView<const float, kFftSizeBy2Plus1> signal_spectrum) {
+    std::span<const float, kFftSizeBy2Plus1> noise_spectrum,
+    std::span<const float, kFftSizeBy2Plus1> prev_noise_spectrum,
+    std::span<const float, kFftSizeBy2Plus1> parametric_noise_spectrum,
+    std::span<const float, kFftSizeBy2Plus1> signal_spectrum) {
   for (size_t i = 0; i < kFftSizeBy2Plus1; ++i) {
     // Previous estimate based on previous frame with gain filter.
     float prev_tsa = spectrum_prev_process_[i] /
diff --git a/modules/audio_processing/ns/wiener_filter.h b/modules/audio_processing/ns/wiener_filter.h
index d34cd56e15..5f956b79dc 100644
--- a/modules/audio_processing/ns/wiener_filter.h
+++ b/modules/audio_processing/ns/wiener_filter.h
@@ -13,8 +13,8 @@
 
 #include <array>
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/ns/ns_common.h"
 #include "modules/audio_processing/ns/suppression_params.h"
 
@@ -30,10 +30,10 @@ class WienerFilter {
   // Updates the filter estimate.
   void Update(
       int32_t num_analyzed_frames,
-      ArrayView<const float, kFftSizeBy2Plus1> noise_spectrum,
-      ArrayView<const float, kFftSizeBy2Plus1> prev_noise_spectrum,
-      ArrayView<const float, kFftSizeBy2Plus1> parametric_noise_spectrum,
-      ArrayView<const float, kFftSizeBy2Plus1> signal_spectrum);
+      std::span<const float, kFftSizeBy2Plus1> noise_spectrum,
+      std::span<const float, kFftSizeBy2Plus1> prev_noise_spectrum,
+      std::span<const float, kFftSizeBy2Plus1> parametric_noise_spectrum,
+      std::span<const float, kFftSizeBy2Plus1> signal_spectrum);
 
   // Compute an overall gain scaling factor.
   float ComputeOverallScalingFactor(int32_t num_analyzed_frames,
@@ -42,7 +42,7 @@ class WienerFilter {
                                     float energy_after_filtering) const;
 
   // Returns the filter.
-  ArrayView<const float, kFftSizeBy2Plus1> get_filter() const {
+  std::span<const float, kFftSizeBy2Plus1> get_filter() const {
     return filter_;
   }
 
diff --git a/modules/audio_processing/post_filter.cc b/modules/audio_processing/post_filter.cc
index ab089b10ce..919d7cf5b6 100644
--- a/modules/audio_processing/post_filter.cc
+++ b/modules/audio_processing/post_filter.cc
@@ -13,9 +13,9 @@
 #include <array>
 #include <cstddef>
 #include <memory>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/audio_buffer.h"
 #include "modules/audio_processing/utility/cascaded_biquad_filter.h"
 #include "rtc_base/checks.h"
@@ -52,7 +52,7 @@ std::unique_ptr<PostFilter> PostFilter::CreateIfNeeded(int sample_rate_hz,
 }
 
 PostFilter::PostFilter(
-    ArrayView<const CascadedBiQuadFilter::BiQuadCoefficients> coefficients,
+    std::span<const CascadedBiQuadFilter::BiQuadCoefficients> coefficients,
     size_t num_channels) {
   RTC_DCHECK(!coefficients.empty());
 
@@ -65,8 +65,8 @@ PostFilter::PostFilter(
 void PostFilter::Process(AudioBuffer& audio) {
   RTC_DCHECK_EQ(filters_.size(), audio.num_channels());
   for (size_t k = 0; k < audio.num_channels(); ++k) {
-    ArrayView<float> channel_data =
-        ArrayView<float>(&audio.channels()[k][0], audio.num_frames());
+    std::span<float> channel_data =
+        std::span<float>(&audio.channels()[k][0], audio.num_frames());
     filters_[k]->Process(channel_data);
   }
 }
diff --git a/modules/audio_processing/post_filter.h b/modules/audio_processing/post_filter.h
index cce42608a6..2817c6ece9 100644
--- a/modules/audio_processing/post_filter.h
+++ b/modules/audio_processing/post_filter.h
@@ -13,9 +13,9 @@
 
 #include <cstddef>
 #include <memory>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/audio_buffer.h"
 #include "modules/audio_processing/utility/cascaded_biquad_filter.h"
 
@@ -39,7 +39,7 @@ class PostFilter {
 
  private:
   PostFilter(
-      ArrayView<const CascadedBiQuadFilter::BiQuadCoefficients> coefficients,
+      std::span<const CascadedBiQuadFilter::BiQuadCoefficients> coefficients,
       size_t num_channels);
 
   std::vector<std::unique_ptr<CascadedBiQuadFilter>> filters_;
diff --git a/modules/audio_processing/post_filter_unittest.cc b/modules/audio_processing/post_filter_unittest.cc
index cc62f1308d..facb613af7 100644
--- a/modules/audio_processing/post_filter_unittest.cc
+++ b/modules/audio_processing/post_filter_unittest.cc
@@ -13,9 +13,9 @@
 #include <cmath>
 #include <memory>
 #include <numbers>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/audio_processing.h"
 #include "modules/audio_processing/audio_buffer.h"
 #include "modules/audio_processing/test/audio_buffer_tools.h"
@@ -48,7 +48,7 @@ std::vector<float> ProcessOneFrameAsAudioBuffer(
   return frame_output;
 }
 
-float ComputePower(ArrayView<const float> audio) {
+float ComputePower(std::span<const float> audio) {
   double energy = 0.0;
   std::for_each(audio.begin(), audio.end(),
                 [&energy](float x) { energy += x * x; });
diff --git a/modules/audio_processing/residual_echo_detector.cc b/modules/audio_processing/residual_echo_detector.cc
index 67e2552768..4db2a1e15a 100644
--- a/modules/audio_processing/residual_echo_detector.cc
+++ b/modules/audio_processing/residual_echo_detector.cc
@@ -15,8 +15,8 @@
 #include <cstddef>
 #include <numeric>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/audio/audio_processing.h"
 #include "modules/audio_processing/logging/apm_data_dumper.h"
 #include "rtc_base/checks.h"
@@ -27,7 +27,7 @@ namespace webrtc {
 
 namespace {
 
-float Power(ArrayView<const float> input) {
+float Power(std::span<const float> input) {
   if (input.empty()) {
     return 0.f;
   }
@@ -58,7 +58,7 @@ ResidualEchoDetector::ResidualEchoDetector()
 ResidualEchoDetector::~ResidualEchoDetector() = default;
 
 void ResidualEchoDetector::AnalyzeRenderAudio(
-    ArrayView<const float> render_audio) {
+    std::span<const float> render_audio) {
   // Dump debug data assuming 48 kHz sample rate (if this assumption is not
   // valid the dumped audio will need to be converted offline accordingly).
   data_dumper_->DumpWav("ed_render", render_audio.size(), render_audio.data(),
@@ -79,7 +79,7 @@ void ResidualEchoDetector::AnalyzeRenderAudio(
 }
 
 void ResidualEchoDetector::AnalyzeCaptureAudio(
-    ArrayView<const float> capture_audio) {
+    std::span<const float> capture_audio) {
   // Dump debug data assuming 48 kHz sample rate (if this assumption is not
   // valid the dumped audio will need to be converted offline accordingly).
   data_dumper_->DumpWav("ed_capture", capture_audio.size(),
diff --git a/modules/audio_processing/residual_echo_detector.h b/modules/audio_processing/residual_echo_detector.h
index 292b6ce658..4c5823ebc9 100644
--- a/modules/audio_processing/residual_echo_detector.h
+++ b/modules/audio_processing/residual_echo_detector.h
@@ -14,9 +14,9 @@
 #include <atomic>
 #include <cstddef>
 #include <memory>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/audio_processing.h"
 #include "modules/audio_processing/echo_detector/circular_buffer.h"
 #include "modules/audio_processing/echo_detector/mean_variance_estimator.h"
@@ -34,10 +34,10 @@ class ResidualEchoDetector : public EchoDetector {
   ~ResidualEchoDetector() override;
 
   // This function should be called while holding the render lock.
-  void AnalyzeRenderAudio(ArrayView<const float> render_audio) override;
+  void AnalyzeRenderAudio(std::span<const float> render_audio) override;
 
   // This function should be called while holding the capture lock.
-  void AnalyzeCaptureAudio(ArrayView<const float> capture_audio) override;
+  void AnalyzeCaptureAudio(std::span<const float> capture_audio) override;
 
   // This function should be called while holding the capture lock.
   void Initialize(int capture_sample_rate_hz,
diff --git a/modules/audio_processing/rms_level.cc b/modules/audio_processing/rms_level.cc
index 8ae21fcb29..8cfab7f3f6 100644
--- a/modules/audio_processing/rms_level.cc
+++ b/modules/audio_processing/rms_level.cc
@@ -16,8 +16,8 @@
 #include <cstdint>
 #include <numeric>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "rtc_base/checks.h"
 
 namespace webrtc {
@@ -61,7 +61,7 @@ void RmsLevel::Reset() {
   block_size_ = std::nullopt;
 }
 
-void RmsLevel::Analyze(ArrayView<const int16_t> data) {
+void RmsLevel::Analyze(std::span<const int16_t> data) {
   if (data.empty()) {
     return;
   }
@@ -78,7 +78,7 @@ void RmsLevel::Analyze(ArrayView<const int16_t> data) {
   max_sum_square_ = std::max(max_sum_square_, sum_square);
 }
 
-void RmsLevel::Analyze(ArrayView<const float> data) {
+void RmsLevel::Analyze(std::span<const float> data) {
   if (data.empty()) {
     return;
   }
diff --git a/modules/audio_processing/rms_level.h b/modules/audio_processing/rms_level.h
index 879fdff224..0d804d0b36 100644
--- a/modules/audio_processing/rms_level.h
+++ b/modules/audio_processing/rms_level.h
@@ -15,8 +15,7 @@
 #include <stdint.h>
 
 #include <optional>
-
-#include "api/array_view.h"
+#include <span>
 
 namespace webrtc {
 
@@ -45,8 +44,8 @@ class RmsLevel {
   void Reset();
 
   // Pass each chunk of audio to Analyze() to accumulate the level.
-  void Analyze(ArrayView<const int16_t> data);
-  void Analyze(ArrayView<const float> data);
+  void Analyze(std::span<const int16_t> data);
+  void Analyze(std::span<const float> data);
 
   // If all samples with the given `length` have a magnitude of zero, this is
   // a shortcut to avoid some computation.
diff --git a/modules/audio_processing/rms_level_unittest.cc b/modules/audio_processing/rms_level_unittest.cc
index ed9372dbb5..0fc3c26276 100644
--- a/modules/audio_processing/rms_level_unittest.cc
+++ b/modules/audio_processing/rms_level_unittest.cc
@@ -14,9 +14,9 @@
 #include <cstdint>
 #include <memory>
 #include <numbers>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/numerics/safe_conversions.h"
 #include "test/gtest.h"
@@ -26,7 +26,7 @@ namespace {
 constexpr int kSampleRateHz = 48000;
 constexpr size_t kBlockSizeSamples = kSampleRateHz / 100;
 
-std::unique_ptr<RmsLevel> RunTest(ArrayView<const int16_t> input) {
+std::unique_ptr<RmsLevel> RunTest(std::span<const int16_t> input) {
   std::unique_ptr<RmsLevel> level(new RmsLevel);
   for (size_t n = 0; n + kBlockSizeSamples <= input.size();
        n += kBlockSizeSamples) {
@@ -35,7 +35,7 @@ std::unique_ptr<RmsLevel> RunTest(ArrayView<const int16_t> input) {
   return level;
 }
 
-std::unique_ptr<RmsLevel> RunTest(ArrayView<const float> input) {
+std::unique_ptr<RmsLevel> RunTest(std::span<const float> input) {
   std::unique_ptr<RmsLevel> level(new RmsLevel);
   for (size_t n = 0; n + kBlockSizeSamples <= input.size();
        n += kBlockSizeSamples) {
diff --git a/modules/audio_processing/splitting_filter.cc b/modules/audio_processing/splitting_filter.cc
index 6513bf80f1..76d77abec3 100644
--- a/modules/audio_processing/splitting_filter.cc
+++ b/modules/audio_processing/splitting_filter.cc
@@ -13,8 +13,8 @@
 #include <array>
 #include <cstddef>
 #include <cstring>
+#include <span>
 
-#include "api/array_view.h"
 #include "common_audio/channel_buffer.h"
 #include "common_audio/signal_processing/include/signal_processing_library.h"
 #include "modules/audio_processing/three_band_filter_bank.h"
@@ -113,10 +113,10 @@ void SplittingFilter::ThreeBandsAnalysis(const ChannelBuffer<float>* data,
 
   for (size_t i = 0; i < three_band_filter_banks_.size(); ++i) {
     three_band_filter_banks_[i].Analysis(
-        ArrayView<const float, ThreeBandFilterBank::kFullBandSize>(
+        std::span<const float, ThreeBandFilterBank::kFullBandSize>(
             data->channels_view()[i].data(),
             ThreeBandFilterBank::kFullBandSize),
-        ArrayView<const ArrayView<float>, ThreeBandFilterBank::kNumBands>(
+        std::span<const std::span<float>, ThreeBandFilterBank::kNumBands>(
             bands->bands_view(i).data(), ThreeBandFilterBank::kNumBands));
   }
 }
@@ -134,9 +134,9 @@ void SplittingFilter::ThreeBandsSynthesis(const ChannelBuffer<float>* bands,
 
   for (size_t i = 0; i < data->num_channels(); ++i) {
     three_band_filter_banks_[i].Synthesis(
-        ArrayView<const ArrayView<float>, ThreeBandFilterBank::kNumBands>(
+        std::span<const std::span<float>, ThreeBandFilterBank::kNumBands>(
             bands->bands_view(i).data(), ThreeBandFilterBank::kNumBands),
-        ArrayView<float, ThreeBandFilterBank::kFullBandSize>(
+        std::span<float, ThreeBandFilterBank::kFullBandSize>(
             data->channels_view()[i].data(),
             ThreeBandFilterBank::kFullBandSize));
   }
diff --git a/modules/audio_processing/test/audio_buffer_tools.cc b/modules/audio_processing/test/audio_buffer_tools.cc
index 8c9f58ed83..1fa7d86bf0 100644
--- a/modules/audio_processing/test/audio_buffer_tools.cc
+++ b/modules/audio_processing/test/audio_buffer_tools.cc
@@ -11,9 +11,9 @@
 #include "modules/audio_processing/test/audio_buffer_tools.h"
 
 #include <cstring>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/audio_processing.h"
 #include "modules/audio_processing/audio_buffer.h"
 #include "rtc_base/checks.h"
@@ -33,7 +33,7 @@ void SetupFrame(const StreamConfig& stream_config,
 }
 
 void CopyVectorToAudioBuffer(const StreamConfig& stream_config,
-                             ArrayView<const float> source,
+                             std::span<const float> source,
                              AudioBuffer* destination) {
   std::vector<float*> input;
   std::vector<float> input_samples;
diff --git a/modules/audio_processing/test/audio_buffer_tools.h b/modules/audio_processing/test/audio_buffer_tools.h
index a324bb23ca..8bdebe041e 100644
--- a/modules/audio_processing/test/audio_buffer_tools.h
+++ b/modules/audio_processing/test/audio_buffer_tools.h
@@ -11,9 +11,9 @@
 #ifndef MODULES_AUDIO_PROCESSING_TEST_AUDIO_BUFFER_TOOLS_H_
 #define MODULES_AUDIO_PROCESSING_TEST_AUDIO_BUFFER_TOOLS_H_
 
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/audio_processing.h"
 #include "modules/audio_processing/audio_buffer.h"
 
@@ -22,7 +22,7 @@ namespace test {
 
 // Copies a vector into an audiobuffer.
 void CopyVectorToAudioBuffer(const StreamConfig& stream_config,
-                             ArrayView<const float> source,
+                             std::span<const float> source,
                              AudioBuffer* destination);
 
 // Extracts a vector from an audiobuffer.
diff --git a/modules/audio_processing/test/bitexactness_tools.cc b/modules/audio_processing/test/bitexactness_tools.cc
index 64538b3fea..01ba02cfd6 100644
--- a/modules/audio_processing/test/bitexactness_tools.cc
+++ b/modules/audio_processing/test/bitexactness_tools.cc
@@ -15,10 +15,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <ostream>  // no-presubmit-check TODO(webrtc:8982)
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_coding/neteq/tools/input_audio_file.h"
 #include "rtc_base/checks.h"
 #include "test/gtest.h"
@@ -62,7 +62,7 @@ std::string GetApmCaptureTestVectorFileName(int sample_rate_hz) {
 void ReadFloatSamplesFromStereoFile(size_t samples_per_channel,
                                     size_t num_channels,
                                     InputAudioFile* stereo_pcm_file,
-                                    ArrayView<float> data) {
+                                    std::span<float> data) {
   RTC_DCHECK_LE(num_channels, 2);
   RTC_DCHECK_EQ(data.size(), samples_per_channel * num_channels);
   std::vector<int16_t> read_samples(samples_per_channel * 2);
@@ -80,8 +80,8 @@ void ReadFloatSamplesFromStereoFile(size_t samples_per_channel,
 ::testing::AssertionResult VerifyDeinterleavedArray(
     size_t samples_per_channel,
     size_t num_channels,
-    ArrayView<const float> reference,
-    ArrayView<const float> output,
+    std::span<const float> reference,
+    std::span<const float> output,
     float element_error_bound) {
   // Form vectors to compare the reference to. Only the first values of the
   // outputs are compared in order not having to specify all preceeding frames
@@ -100,8 +100,8 @@ void ReadFloatSamplesFromStereoFile(size_t samples_per_channel,
   return VerifyArray(reference, output_to_verify, element_error_bound);
 }
 
-::testing::AssertionResult VerifyArray(ArrayView<const float> reference,
-                                       ArrayView<const float> output,
+::testing::AssertionResult VerifyArray(std::span<const float> reference,
+                                       std::span<const float> output,
                                        float element_error_bound) {
   // The vectors are deemed to be bitexact only if
   // a) output have a size at least as long as the reference.
@@ -127,7 +127,7 @@ void ReadFloatSamplesFromStereoFile(size_t samples_per_channel,
 
   // Lambda function that produces a formatted string with the data in the
   // vector.
-  auto print_vector_in_c_format = [](ArrayView<const float> v,
+  auto print_vector_in_c_format = [](std::span<const float> v,
                                      size_t num_values_to_print) {
     std::string s = "{ ";
     for (size_t k = 0; k < std::min(num_values_to_print, v.size()); ++k) {
diff --git a/modules/audio_processing/test/bitexactness_tools.h b/modules/audio_processing/test/bitexactness_tools.h
index f87658e3c6..6f8a1f47c5 100644
--- a/modules/audio_processing/test/bitexactness_tools.h
+++ b/modules/audio_processing/test/bitexactness_tools.h
@@ -13,9 +13,9 @@
 #define MODULES_AUDIO_PROCESSING_TEST_BITEXACTNESS_TOOLS_H_
 
 #include <cstddef>
+#include <span>
 #include <string>
 
-#include "api/array_view.h"
 #include "modules/audio_coding/neteq/tools/input_audio_file.h"
 #include "test/gtest.h"
 
@@ -34,21 +34,21 @@ std::string GetApmCaptureTestVectorFileName(int sample_rate_hz);
 void ReadFloatSamplesFromStereoFile(size_t samples_per_channel,
                                     size_t num_channels,
                                     InputAudioFile* stereo_pcm_file,
-                                    ArrayView<float> data);
+                                    std::span<float> data);
 
 // Verifies a frame against a reference and returns the results as an
 // AssertionResult.
 ::testing::AssertionResult VerifyDeinterleavedArray(
     size_t samples_per_channel,
     size_t num_channels,
-    ArrayView<const float> reference,
-    ArrayView<const float> output,
+    std::span<const float> reference,
+    std::span<const float> output,
     float element_error_bound);
 
 // Verifies a vector against a reference and returns the results as an
 // AssertionResult.
-::testing::AssertionResult VerifyArray(ArrayView<const float> reference,
-                                       ArrayView<const float> output,
+::testing::AssertionResult VerifyArray(std::span<const float> reference,
+                                       std::span<const float> output,
                                        float element_error_bound);
 
 }  // namespace test
diff --git a/modules/audio_processing/test/conversational_speech/BUILD.gn b/modules/audio_processing/test/conversational_speech/BUILD.gn
index 28f27daae4..88e2613c48 100644
--- a/modules/audio_processing/test/conversational_speech/BUILD.gn
+++ b/modules/audio_processing/test/conversational_speech/BUILD.gn
@@ -45,7 +45,6 @@ rtc_library("lib") {
     "wavreader_interface.h",
   ]
   deps = [
-    "../../../../api:array_view",
     "../../../../common_audio",
     "../../../../rtc_base:checks",
     "../../../../rtc_base:logging",
@@ -68,7 +67,6 @@ rtc_library("unittest") {
   ]
   deps = [
     ":lib",
-    "../../../../api:array_view",
     "../../../../common_audio",
     "../../../../rtc_base:logging",
     "../../../../rtc_base:safe_conversions",
diff --git a/modules/audio_processing/test/conversational_speech/mock_wavreader.h b/modules/audio_processing/test/conversational_speech/mock_wavreader.h
index 32acec6003..a1881586b3 100644
--- a/modules/audio_processing/test/conversational_speech/mock_wavreader.h
+++ b/modules/audio_processing/test/conversational_speech/mock_wavreader.h
@@ -13,8 +13,8 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/test/conversational_speech/wavreader_interface.h"
 #include "test/gmock.h"
 
@@ -27,11 +27,8 @@ class MockWavReader : public WavReaderInterface {
   MockWavReader(int sample_rate, size_t num_channels, size_t num_samples);
   ~MockWavReader() override;
 
-  MOCK_METHOD(size_t, ReadFloatSamples, (webrtc::ArrayView<float>), (override));
-  MOCK_METHOD(size_t,
-              ReadInt16Samples,
-              (webrtc::ArrayView<int16_t>),
-              (override));
+  MOCK_METHOD(size_t, ReadFloatSamples, (std::span<float>), (override));
+  MOCK_METHOD(size_t, ReadInt16Samples, (std::span<int16_t>), (override));
 
   MOCK_METHOD(int, SampleRate, (), (const, override));
   MOCK_METHOD(size_t, NumChannels, (), (const, override));
diff --git a/modules/audio_processing/test/conversational_speech/multiend_call.cc b/modules/audio_processing/test/conversational_speech/multiend_call.cc
index d0b5d7cf11..bf3f19c9b6 100644
--- a/modules/audio_processing/test/conversational_speech/multiend_call.cc
+++ b/modules/audio_processing/test/conversational_speech/multiend_call.cc
@@ -15,13 +15,13 @@
 #include <iterator>
 #include <map>
 #include <memory>
+#include <span>
 #include <string>
 #include <tuple>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "modules/audio_processing/test/conversational_speech/timing.h"
 #include "modules/audio_processing/test/conversational_speech/wavreader_abstract_factory.h"
 #include "modules/audio_processing/test/conversational_speech/wavreader_interface.h"
@@ -34,7 +34,7 @@ namespace test {
 namespace conversational_speech {
 
 MultiEndCall::MultiEndCall(
-    ArrayView<const Turn> timing,
+    std::span<const Turn> timing,
     absl::string_view audiotracks_path,
     std::unique_ptr<WavReaderAbstractFactory> wavreader_abstract_factory)
     : timing_(timing),
diff --git a/modules/audio_processing/test/conversational_speech/multiend_call.h b/modules/audio_processing/test/conversational_speech/multiend_call.h
index 4ea53c9ff5..fbf926bdf0 100644
--- a/modules/audio_processing/test/conversational_speech/multiend_call.h
+++ b/modules/audio_processing/test/conversational_speech/multiend_call.h
@@ -16,11 +16,11 @@
 #include <map>
 #include <memory>
 #include <set>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "modules/audio_processing/test/conversational_speech/timing.h"
 #include "modules/audio_processing/test/conversational_speech/wavreader_abstract_factory.h"
 #include "modules/audio_processing/test/conversational_speech/wavreader_interface.h"
@@ -51,7 +51,7 @@ class MultiEndCall {
   };
 
   MultiEndCall(
-      ArrayView<const Turn> timing,
+      std::span<const Turn> timing,
       absl::string_view audiotracks_path,
       std::unique_ptr<WavReaderAbstractFactory> wavreader_abstract_factory);
   ~MultiEndCall();
@@ -84,7 +84,7 @@ class MultiEndCall {
   // only up to 2 speakers. Rejects unordered turns and self cross-talk.
   bool CheckTiming();
 
-  ArrayView<const Turn> timing_;
+  std::span<const Turn> timing_;
   std::string audiotracks_path_;
   std::unique_ptr<WavReaderAbstractFactory> wavreader_abstract_factory_;
   std::set<std::string> speaker_names_;
diff --git a/modules/audio_processing/test/conversational_speech/simulator.cc b/modules/audio_processing/test/conversational_speech/simulator.cc
index f95b508cc3..2dcfbdbd2a 100644
--- a/modules/audio_processing/test/conversational_speech/simulator.cc
+++ b/modules/audio_processing/test/conversational_speech/simulator.cc
@@ -16,13 +16,13 @@
 #include <map>
 #include <memory>
 #include <set>
+#include <span>
 #include <string>
 #include <tuple>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "common_audio/include/audio_util.h"
 #include "common_audio/wav_file.h"
 #include "modules/audio_processing/test/conversational_speech/multiend_call.h"
@@ -134,7 +134,7 @@ std::unique_ptr<std::map<std::string, std::vector<int16_t>>> PreloadAudioTracks(
 // previously written samples in `wav_writer` is less than `interval_begin`, it
 // adds zeros as left padding. The padding corresponds to intervals during which
 // a speaker is not active.
-void PadLeftWriteChunk(ArrayView<const int16_t> source_samples,
+void PadLeftWriteChunk(std::span<const int16_t> source_samples,
                        size_t interval_begin,
                        WavWriter* wav_writer) {
   // Add left padding.
@@ -163,9 +163,9 @@ void PadRightWrite(WavWriter* wav_writer, size_t pad_samples) {
   }
 }
 
-void ScaleSignal(ArrayView<const int16_t> source_samples,
+void ScaleSignal(std::span<const int16_t> source_samples,
                  int gain,
-                 ArrayView<int16_t> output_samples) {
+                 std::span<int16_t> output_samples) {
   const float gain_linear = DbToRatio(gain);
   RTC_DCHECK_EQ(source_samples.size(), output_samples.size());
   std::transform(source_samples.begin(), source_samples.end(),
diff --git a/modules/audio_processing/test/conversational_speech/timing.cc b/modules/audio_processing/test/conversational_speech/timing.cc
index 6f21113a09..bd1998ed40 100644
--- a/modules/audio_processing/test/conversational_speech/timing.cc
+++ b/modules/audio_processing/test/conversational_speech/timing.cc
@@ -12,11 +12,11 @@
 
 #include <fstream>
 #include <iostream>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/string_encode.h"
 #include "rtc_base/string_to_number.h"
@@ -62,7 +62,7 @@ std::vector<Turn> LoadTiming(absl::string_view timing_filepath) {
 }
 
 void SaveTiming(absl::string_view timing_filepath,
-                ArrayView<const Turn> timing) {
+                std::span<const Turn> timing) {
   std::ofstream outfile(std::string{timing_filepath});
   RTC_CHECK(outfile.is_open());
   for (const Turn& turn : timing) {
diff --git a/modules/audio_processing/test/conversational_speech/timing.h b/modules/audio_processing/test/conversational_speech/timing.h
index 56e19c6e8e..edf8cd6bce 100644
--- a/modules/audio_processing/test/conversational_speech/timing.h
+++ b/modules/audio_processing/test/conversational_speech/timing.h
@@ -11,11 +11,11 @@
 #ifndef MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_TIMING_H_
 #define MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_TIMING_H_
 
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 
 namespace webrtc {
 namespace test {
@@ -42,7 +42,7 @@ std::vector<Turn> LoadTiming(absl::string_view timing_filepath);
 
 // Writes a list of turns into a file.
 void SaveTiming(absl::string_view timing_filepath,
-                ArrayView<const Turn> timing);
+                std::span<const Turn> timing);
 
 }  // namespace conversational_speech
 }  // namespace test
diff --git a/modules/audio_processing/test/conversational_speech/wavreader_factory.cc b/modules/audio_processing/test/conversational_speech/wavreader_factory.cc
index 4cd7d096e2..63f4d68f3e 100644
--- a/modules/audio_processing/test/conversational_speech/wavreader_factory.cc
+++ b/modules/audio_processing/test/conversational_speech/wavreader_factory.cc
@@ -13,9 +13,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <memory>
+#include <span>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "common_audio/wav_file.h"
 #include "modules/audio_processing/test/conversational_speech/wavreader_interface.h"
 
@@ -31,11 +31,11 @@ class WavReaderAdaptor final : public WavReaderInterface {
       : wav_reader_(filepath) {}
   ~WavReaderAdaptor() override = default;
 
-  size_t ReadFloatSamples(ArrayView<float> samples) override {
+  size_t ReadFloatSamples(std::span<float> samples) override {
     return wav_reader_.ReadSamples(samples.size(), samples.data());
   }
 
-  size_t ReadInt16Samples(ArrayView<int16_t> samples) override {
+  size_t ReadInt16Samples(std::span<int16_t> samples) override {
     return wav_reader_.ReadSamples(samples.size(), samples.data());
   }
 
diff --git a/modules/audio_processing/test/conversational_speech/wavreader_interface.h b/modules/audio_processing/test/conversational_speech/wavreader_interface.h
index c589c579d4..012c70497f 100644
--- a/modules/audio_processing/test/conversational_speech/wavreader_interface.h
+++ b/modules/audio_processing/test/conversational_speech/wavreader_interface.h
@@ -13,8 +13,7 @@
 
 #include <cstddef>
 #include <cstdint>
-
-#include "api/array_view.h"
+#include <span>
 
 namespace webrtc {
 namespace test {
@@ -25,8 +24,8 @@ class WavReaderInterface {
   virtual ~WavReaderInterface() = default;
 
   // Returns the number of samples read.
-  virtual size_t ReadFloatSamples(ArrayView<float> samples) = 0;
-  virtual size_t ReadInt16Samples(ArrayView<int16_t> samples) = 0;
+  virtual size_t ReadFloatSamples(std::span<float> samples) = 0;
+  virtual size_t ReadInt16Samples(std::span<int16_t> samples) = 0;
 
   // Getters.
   virtual int SampleRate() const = 0;
diff --git a/modules/audio_processing/test/echo_canceller_test_tools.cc b/modules/audio_processing/test/echo_canceller_test_tools.cc
index 0d0e4077d6..1e0c84956b 100644
--- a/modules/audio_processing/test/echo_canceller_test_tools.cc
+++ b/modules/audio_processing/test/echo_canceller_test_tools.cc
@@ -12,20 +12,20 @@
 
 #include <algorithm>
 #include <cstddef>
+#include <span>
 
-#include "api/array_view.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/random.h"
 
 namespace webrtc {
 
-void RandomizeSampleVector(Random* random_generator, ArrayView<float> v) {
+void RandomizeSampleVector(Random* random_generator, std::span<float> v) {
   RandomizeSampleVector(random_generator, v,
                         /*amplitude=*/32767.f);
 }
 
 void RandomizeSampleVector(Random* random_generator,
-                           ArrayView<float> v,
+                           std::span<float> v,
                            float amplitude) {
   for (auto& v_k : v) {
     v_k = 2 * amplitude * random_generator->Rand<float>() - amplitude;
@@ -33,7 +33,7 @@ void RandomizeSampleVector(Random* random_generator,
 }
 
 template <typename T>
-void DelayBuffer<T>::Delay(ArrayView<const T> x, ArrayView<T> x_delayed) {
+void DelayBuffer<T>::Delay(std::span<const T> x, std::span<T> x_delayed) {
   RTC_DCHECK_EQ(x.size(), x_delayed.size());
   if (buffer_.empty()) {
     std::copy(x.begin(), x.end(), x_delayed.begin());
diff --git a/modules/audio_processing/test/echo_canceller_test_tools.h b/modules/audio_processing/test/echo_canceller_test_tools.h
index f8a805a726..cb5ef8514f 100644
--- a/modules/audio_processing/test/echo_canceller_test_tools.h
+++ b/modules/audio_processing/test/echo_canceller_test_tools.h
@@ -12,19 +12,19 @@
 #define MODULES_AUDIO_PROCESSING_TEST_ECHO_CANCELLER_TEST_TOOLS_H_
 
 #include <cstddef>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "rtc_base/random.h"
 
 namespace webrtc {
 
 // Randomizes the elements in a vector with values -32767.f:32767.f.
-void RandomizeSampleVector(Random* random_generator, ArrayView<float> v);
+void RandomizeSampleVector(Random* random_generator, std::span<float> v);
 
 // Randomizes the elements in a vector with values -amplitude:amplitude.
 void RandomizeSampleVector(Random* random_generator,
-                           ArrayView<float> v,
+                           std::span<float> v,
                            float amplitude);
 
 // Class for delaying a signal a fixed number of samples.
@@ -35,7 +35,7 @@ class DelayBuffer {
   ~DelayBuffer() = default;
 
   // Produces a delayed signal copy of x.
-  void Delay(ArrayView<const T> x, ArrayView<T> x_delayed);
+  void Delay(std::span<const T> x, std::span<T> x_delayed);
 
  private:
   std::vector<T> buffer_;
diff --git a/modules/audio_processing/test/echo_canceller_test_tools_unittest.cc b/modules/audio_processing/test/echo_canceller_test_tools_unittest.cc
index b22bd452a0..c1340c8bcd 100644
--- a/modules/audio_processing/test/echo_canceller_test_tools_unittest.cc
+++ b/modules/audio_processing/test/echo_canceller_test_tools_unittest.cc
@@ -12,9 +12,9 @@
 
 #include <algorithm>
 #include <cstddef>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/random.h"
 #include "test/gtest.h"
@@ -32,8 +32,8 @@ TEST(EchoCancellerTestTools, FloatDelayBuffer) {
   constexpr size_t kBlockSize = 50;
   for (size_t k = 0; k < CheckedDivExact(v.size(), kBlockSize); ++k) {
     delay_buffer.Delay(
-        ArrayView<const float>(&v[k * kBlockSize], kBlockSize),
-        ArrayView<float>(&v_delayed[k * kBlockSize], kBlockSize));
+        std::span<const float>(&v[k * kBlockSize], kBlockSize),
+        std::span<float>(&v_delayed[k * kBlockSize], kBlockSize));
   }
   for (size_t k = kDelay; k < v.size(); ++k) {
     EXPECT_EQ(v[k - kDelay], v_delayed[k]);
@@ -50,8 +50,8 @@ TEST(EchoCancellerTestTools, IntDelayBuffer) {
   std::vector<int> v_delayed = v;
   const size_t kBlockSize = 50;
   for (size_t k = 0; k < CheckedDivExact(v.size(), kBlockSize); ++k) {
-    delay_buffer.Delay(ArrayView<const int>(&v[k * kBlockSize], kBlockSize),
-                       ArrayView<int>(&v_delayed[k * kBlockSize], kBlockSize));
+    delay_buffer.Delay(std::span<const int>(&v[k * kBlockSize], kBlockSize),
+                       std::span<int>(&v_delayed[k * kBlockSize], kBlockSize));
   }
   for (size_t k = kDelay; k < v.size(); ++k) {
     EXPECT_EQ(v[k - kDelay], v_delayed[k]);
diff --git a/modules/audio_processing/test/fake_recording_device.cc b/modules/audio_processing/test/fake_recording_device.cc
index 6706fe56ac..a3f90ff85a 100644
--- a/modules/audio_processing/test/fake_recording_device.cc
+++ b/modules/audio_processing/test/fake_recording_device.cc
@@ -14,8 +14,8 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "common_audio/channel_buffer.h"
 #include "common_audio/include/audio_util.h"
 #include "modules/audio_processing/agc2/gain_map_internal.h"
@@ -43,7 +43,7 @@ class FakeRecordingDeviceWorker {
   void set_mic_level(const int level) { mic_level_ = level; }
   void set_undo_mic_level(const int level) { undo_mic_level_ = level; }
   virtual ~FakeRecordingDeviceWorker() = default;
-  virtual void ModifyBufferInt16(ArrayView<int16_t> buffer) = 0;
+  virtual void ModifyBufferInt16(std::span<int16_t> buffer) = 0;
   virtual void ModifyBufferFloat(ChannelBuffer<float>* buffer) = 0;
 
  protected:
@@ -62,7 +62,7 @@ class FakeRecordingDeviceIdentity final : public FakeRecordingDeviceWorker {
   explicit FakeRecordingDeviceIdentity(const int initial_mic_level)
       : FakeRecordingDeviceWorker(initial_mic_level) {}
   ~FakeRecordingDeviceIdentity() override = default;
-  void ModifyBufferInt16(ArrayView<int16_t> /* buffer */) override {}
+  void ModifyBufferInt16(std::span<int16_t> /* buffer */) override {}
   void ModifyBufferFloat(ChannelBuffer<float>* /* buffer */) override {}
 };
 
@@ -73,7 +73,7 @@ class FakeRecordingDeviceLinear final : public FakeRecordingDeviceWorker {
   explicit FakeRecordingDeviceLinear(const int initial_mic_level)
       : FakeRecordingDeviceWorker(initial_mic_level) {}
   ~FakeRecordingDeviceLinear() override = default;
-  void ModifyBufferInt16(ArrayView<int16_t> buffer) override {
+  void ModifyBufferInt16(std::span<int16_t> buffer) override {
     const size_t number_of_samples = buffer.size();
     int16_t* data = buffer.data();
     // If an undo level is specified, virtually restore the unmodified
@@ -115,7 +115,7 @@ class FakeRecordingDeviceAgc final : public FakeRecordingDeviceWorker {
   explicit FakeRecordingDeviceAgc(const int initial_mic_level)
       : FakeRecordingDeviceWorker(initial_mic_level) {}
   ~FakeRecordingDeviceAgc() override = default;
-  void ModifyBufferInt16(ArrayView<int16_t> buffer) override {
+  void ModifyBufferInt16(std::span<int16_t> buffer) override {
     const float scaling_factor =
         ComputeAgcLinearFactor(undo_mic_level_, mic_level_);
     const size_t number_of_samples = buffer.size();
@@ -178,7 +178,7 @@ void FakeRecordingDevice::SetUndoMicLevel(const int level) {
   worker_->set_undo_mic_level(level);
 }
 
-void FakeRecordingDevice::SimulateAnalogGain(ArrayView<int16_t> buffer) {
+void FakeRecordingDevice::SimulateAnalogGain(std::span<int16_t> buffer) {
   RTC_DCHECK(worker_);
   worker_->ModifyBufferInt16(buffer);
 }
diff --git a/modules/audio_processing/test/fake_recording_device.h b/modules/audio_processing/test/fake_recording_device.h
index 0fcb4d18ee..84c34ff74a 100644
--- a/modules/audio_processing/test/fake_recording_device.h
+++ b/modules/audio_processing/test/fake_recording_device.h
@@ -13,8 +13,8 @@
 
 #include <cstdint>
 #include <memory>
+#include <span>
 
-#include "api/array_view.h"
 #include "common_audio/channel_buffer.h"
 
 namespace webrtc {
@@ -53,7 +53,7 @@ class FakeRecordingDevice final {
   // If `real_device_level` is a valid level, the unmodified mic signal is
   // virtually restored. To skip the latter step set `real_device_level` to
   // an empty value.
-  void SimulateAnalogGain(ArrayView<int16_t> buffer);
+  void SimulateAnalogGain(std::span<int16_t> buffer);
 
   // Simulates the analog gain.
   // If `real_device_level` is a valid level, the unmodified mic signal is
diff --git a/modules/audio_processing/three_band_filter_bank.cc b/modules/audio_processing/three_band_filter_bank.cc
index a04852a4c7..0b12b938b8 100644
--- a/modules/audio_processing/three_band_filter_bank.cc
+++ b/modules/audio_processing/three_band_filter_bank.cc
@@ -35,8 +35,8 @@
 #include <algorithm>
 #include <array>
 #include <numbers>
+#include <span>
 
-#include "api/array_view.h"
 #include "rtc_base/checks.h"
 
 namespace webrtc {
@@ -109,11 +109,11 @@ const float kDctModulation[ThreeBandFilterBank::kNumNonZeroFilters][kDctSize] =
 
 // Filters the input signal `in` with the filter `filter` using a shift by
 // `in_shift`, taking into account the previous state.
-void FilterCore(ArrayView<const float, kFilterSize> filter,
-                ArrayView<const float, ThreeBandFilterBank::kSplitBandSize> in,
+void FilterCore(std::span<const float, kFilterSize> filter,
+                std::span<const float, ThreeBandFilterBank::kSplitBandSize> in,
                 const int in_shift,
-                ArrayView<float, ThreeBandFilterBank::kSplitBandSize> out,
-                ArrayView<float, kMemorySize> state) {
+                std::span<float, ThreeBandFilterBank::kSplitBandSize> out,
+                std::span<float, kMemorySize> state) {
   constexpr int kMaxInShift = (kStride - 1);
   RTC_DCHECK_GE(in_shift, 0);
   RTC_DCHECK_LE(in_shift, kMaxInShift);
@@ -176,8 +176,8 @@ ThreeBandFilterBank::~ThreeBandFilterBank() = default;
 //      of `kSparsity`.
 //   3. Modulating with cosines and accumulating to get the desired band.
 void ThreeBandFilterBank::Analysis(
-    ArrayView<const float, kFullBandSize> in,
-    ArrayView<const ArrayView<float>, ThreeBandFilterBank::kNumBands> out) {
+    std::span<const float, kFullBandSize> in,
+    std::span<const std::span<float>, ThreeBandFilterBank::kNumBands> out) {
   // Initialize the output to zero.
   for (int band = 0; band < ThreeBandFilterBank::kNumBands; ++band) {
     RTC_DCHECK_EQ(out[band].size(), kSplitBandSize);
@@ -204,10 +204,10 @@ void ThreeBandFilterBank::Analysis(
               ? index
               : (index < kZeroFilterIndex2 ? index - 1 : index - 2);
 
-      ArrayView<const float, kFilterSize> filter(kFilterCoeffs[filter_index]);
-      ArrayView<const float, kDctSize> dct_modulation(
+      std::span<const float, kFilterSize> filter(kFilterCoeffs[filter_index]);
+      std::span<const float, kDctSize> dct_modulation(
           kDctModulation[filter_index]);
-      ArrayView<float, kMemorySize> state(state_analysis_[filter_index]);
+      std::span<float, kMemorySize> state(state_analysis_[filter_index]);
 
       // Filter.
       std::array<float, kSplitBandSize> out_subsampled;
@@ -231,8 +231,8 @@ void ThreeBandFilterBank::Analysis(
 //      `kSparsity` signals with different delays.
 //   3. Parallel to serial upsampling by a factor of `kNumBands`.
 void ThreeBandFilterBank::Synthesis(
-    ArrayView<const ArrayView<float>, ThreeBandFilterBank::kNumBands> in,
-    ArrayView<float, kFullBandSize> out) {
+    std::span<const std::span<float>, ThreeBandFilterBank::kNumBands> in,
+    std::span<float, kFullBandSize> out) {
   std::fill(out.begin(), out.end(), 0);
   for (int upsampling_index = 0; upsampling_index < kSubSampling;
        ++upsampling_index) {
@@ -247,10 +247,10 @@ void ThreeBandFilterBank::Synthesis(
               ? index
               : (index < kZeroFilterIndex2 ? index - 1 : index - 2);
 
-      ArrayView<const float, kFilterSize> filter(kFilterCoeffs[filter_index]);
-      ArrayView<const float, kDctSize> dct_modulation(
+      std::span<const float, kFilterSize> filter(kFilterCoeffs[filter_index]);
+      std::span<const float, kDctSize> dct_modulation(
           kDctModulation[filter_index]);
-      ArrayView<float, kMemorySize> state(state_synthesis_[filter_index]);
+      std::span<float, kMemorySize> state(state_synthesis_[filter_index]);
 
       // Prepare filter input by modulating the banded input.
       std::array<float, kSplitBandSize> in_subsampled;
diff --git a/modules/audio_processing/three_band_filter_bank.h b/modules/audio_processing/three_band_filter_bank.h
index 84feb39d07..2ee95f8502 100644
--- a/modules/audio_processing/three_band_filter_bank.h
+++ b/modules/audio_processing/three_band_filter_bank.h
@@ -13,8 +13,7 @@
 
 #include <array>
 #include <cstring>
-
-#include "api/array_view.h"
+#include <span>
 
 namespace webrtc {
 
@@ -55,13 +54,13 @@ class ThreeBandFilterBank final {
 
   // Splits `in` of size kFullBandSize into 3 downsampled frequency bands in
   // `out`, each of size 160.
-  void Analysis(ArrayView<const float, kFullBandSize> in,
-                ArrayView<const ArrayView<float>, kNumBands> out);
+  void Analysis(std::span<const float, kFullBandSize> in,
+                std::span<const std::span<float>, kNumBands> out);
 
   // Merges the 3 downsampled frequency bands in `in`, each of size 160, into
   // `out`, which is of size kFullBandSize.
-  void Synthesis(ArrayView<const ArrayView<float>, kNumBands> in,
-                 ArrayView<float, kFullBandSize> out);
+  void Synthesis(std::span<const std::span<float>, kNumBands> in,
+                 std::span<float, kFullBandSize> out);
 
  private:
   std::array<std::array<float, kMemorySize>, kNumNonZeroFilters>
diff --git a/modules/audio_processing/utility/BUILD.gn b/modules/audio_processing/utility/BUILD.gn
index 63ee6d066e..413aba48d6 100644
--- a/modules/audio_processing/utility/BUILD.gn
+++ b/modules/audio_processing/utility/BUILD.gn
@@ -13,10 +13,7 @@ rtc_library("cascaded_biquad_filter") {
     "cascaded_biquad_filter.cc",
     "cascaded_biquad_filter.h",
   ]
-  deps = [
-    "../../../api:array_view",
-    "../../../rtc_base:checks",
-  ]
+  deps = [ "../../../rtc_base:checks" ]
 }
 
 rtc_library("legacy_delay_estimator") {
@@ -37,7 +34,6 @@ rtc_library("pffft_wrapper") {
     "pffft_wrapper.h",
   ]
   deps = [
-    "../../../api:array_view",
     "../../../rtc_base:checks",
     "//third_party/pffft",
   ]
@@ -52,7 +48,6 @@ if (rtc_include_tests) {
       ":cascaded_biquad_filter",
       "../../../rtc_base:checks",
       "../../../test:test_support",
-      "//api:array_view",
       "//rtc_base:checks",
       "//testing/gtest",
     ]
@@ -74,7 +69,6 @@ if (rtc_include_tests) {
     sources = [ "pffft_wrapper_unittest.cc" ]
     deps = [
       ":pffft_wrapper",
-      "../../../api:array_view",
       "../../../test:test_support",
       "//testing/gtest",
       "//third_party/pffft",
diff --git a/modules/audio_processing/utility/cascaded_biquad_filter.cc b/modules/audio_processing/utility/cascaded_biquad_filter.cc
index 53c9d15506..2abbda0d9c 100644
--- a/modules/audio_processing/utility/cascaded_biquad_filter.cc
+++ b/modules/audio_processing/utility/cascaded_biquad_filter.cc
@@ -11,9 +11,9 @@
 
 #include <algorithm>
 #include <cstddef>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "rtc_base/checks.h"
 
 namespace webrtc {
@@ -23,7 +23,7 @@ void CascadedBiQuadFilter::BiQuad::BiQuad::Reset() {
 }
 
 CascadedBiQuadFilter::CascadedBiQuadFilter(
-    ArrayView<const CascadedBiQuadFilter::BiQuadCoefficients> coefficients) {
+    std::span<const CascadedBiQuadFilter::BiQuadCoefficients> coefficients) {
   for (const auto& single_biquad_coefficients : coefficients) {
     biquads_.push_back(BiQuad(single_biquad_coefficients));
   }
@@ -31,8 +31,8 @@ CascadedBiQuadFilter::CascadedBiQuadFilter(
 
 CascadedBiQuadFilter::~CascadedBiQuadFilter() = default;
 
-void CascadedBiQuadFilter::Process(ArrayView<const float> x,
-                                   ArrayView<float> y) {
+void CascadedBiQuadFilter::Process(std::span<const float> x,
+                                   std::span<float> y) {
   if (!biquads_.empty()) {
     ApplyBiQuad(x, y, &biquads_[0]);
     for (size_t k = 1; k < biquads_.size(); ++k) {
@@ -43,7 +43,7 @@ void CascadedBiQuadFilter::Process(ArrayView<const float> x,
   }
 }
 
-void CascadedBiQuadFilter::Process(ArrayView<float> y) {
+void CascadedBiQuadFilter::Process(std::span<float> y) {
   for (auto& biquad : biquads_) {
     ApplyBiQuad(y, y, &biquad);
   }
@@ -55,8 +55,8 @@ void CascadedBiQuadFilter::Reset() {
   }
 }
 
-void CascadedBiQuadFilter::ApplyBiQuad(ArrayView<const float> x,
-                                       ArrayView<float> y,
+void CascadedBiQuadFilter::ApplyBiQuad(std::span<const float> x,
+                                       std::span<float> y,
                                        CascadedBiQuadFilter::BiQuad* biquad) {
   RTC_DCHECK_EQ(x.size(), y.size());
   const float c_a_0 = biquad->coefficients.a[0];
diff --git a/modules/audio_processing/utility/cascaded_biquad_filter.h b/modules/audio_processing/utility/cascaded_biquad_filter.h
index 25d8d504ec..3f1ab71e73 100644
--- a/modules/audio_processing/utility/cascaded_biquad_filter.h
+++ b/modules/audio_processing/utility/cascaded_biquad_filter.h
@@ -13,10 +13,9 @@
 
 #include <stddef.h>
 
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
-
 namespace webrtc {
 
 // Applies a number of biquads in a cascaded manner. The filter implementation
@@ -38,21 +37,21 @@ class CascadedBiQuadFilter {
   };
 
   CascadedBiQuadFilter(
-      ArrayView<const CascadedBiQuadFilter::BiQuadCoefficients> coefficients);
+      std::span<const CascadedBiQuadFilter::BiQuadCoefficients> coefficients);
   ~CascadedBiQuadFilter();
   CascadedBiQuadFilter(const CascadedBiQuadFilter&) = delete;
   CascadedBiQuadFilter& operator=(const CascadedBiQuadFilter&) = delete;
 
   // Applies the biquads on the values in x in order to form the output in y.
-  void Process(ArrayView<const float> x, ArrayView<float> y);
+  void Process(std::span<const float> x, std::span<float> y);
   // Applies the biquads on the values in y in an in-place manner.
-  void Process(ArrayView<float> y);
+  void Process(std::span<float> y);
   // Resets the filter to its initial state.
   void Reset();
 
  private:
-  void ApplyBiQuad(ArrayView<const float> x,
-                   ArrayView<float> y,
+  void ApplyBiQuad(std::span<const float> x,
+                   std::span<float> y,
                    CascadedBiQuadFilter::BiQuad* biquad);
 
   std::vector<BiQuad> biquads_;
diff --git a/modules/audio_processing/utility/cascaded_biquad_filter_unittest.cc b/modules/audio_processing/utility/cascaded_biquad_filter_unittest.cc
index 23e1e87820..c6e14df023 100644
--- a/modules/audio_processing/utility/cascaded_biquad_filter_unittest.cc
+++ b/modules/audio_processing/utility/cascaded_biquad_filter_unittest.cc
@@ -12,9 +12,9 @@
 
 #include <array>
 #include <cstddef>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "rtc_base/checks.h"
 #include "test/gtest.h"
 
@@ -61,7 +61,7 @@ TEST(CascadedBiquadFilter, BlockingConfiguration) {
   std::vector<float> values = CreateInputWithIncreasingValues(1000);
 
   CascadedBiQuadFilter filter(
-      (ArrayView<const CascadedBiQuadFilter::BiQuadCoefficients>(
+      (std::span<const CascadedBiQuadFilter::BiQuadCoefficients>(
           kBlockingCoefficients)));
   filter.Process(values);
 
@@ -78,7 +78,7 @@ TEST(CascadedBiquadFilter, HighPassConfiguration) {
   }
 
   CascadedBiQuadFilter filter(
-      (ArrayView<const CascadedBiQuadFilter::BiQuadCoefficients>(
+      (std::span<const CascadedBiQuadFilter::BiQuadCoefficients>(
           kHighPassFilterCoefficients)));
   filter.Process(values);
 
@@ -90,7 +90,7 @@ TEST(CascadedBiquadFilter, HighPassConfiguration) {
 // Verifies that the reset functionality works as intended.
 TEST(CascadedBiquadFilter, HighPassConfigurationResetFunctionality) {
   CascadedBiQuadFilter filter(
-      (ArrayView<const CascadedBiQuadFilter::BiQuadCoefficients>(
+      (std::span<const CascadedBiQuadFilter::BiQuadCoefficients>(
           kHighPassFilterCoefficients)));
 
   std::vector<float> values1(100, 1.f);
@@ -114,7 +114,7 @@ TEST(CascadedBiquadFilter, TransparentConfiguration) {
   std::vector<float> output(input.size());
 
   CascadedBiQuadFilter filter(
-      (ArrayView<const CascadedBiQuadFilter::BiQuadCoefficients>(
+      (std::span<const CascadedBiQuadFilter::BiQuadCoefficients>(
           kTransparentCoefficients)));
   filter.Process(input, output);
 
@@ -127,7 +127,7 @@ TEST(CascadedBiquadFilter, CascadedConfiguration) {
   std::vector<float> output(input.size());
 
   CascadedBiQuadFilter filter(
-      (ArrayView<const CascadedBiQuadFilter::BiQuadCoefficients>(
+      (std::span<const CascadedBiQuadFilter::BiQuadCoefficients>(
           kCascadedCoefficients)));
   filter.Process(input, output);
 
@@ -145,7 +145,7 @@ TEST(CascadedBiquadFilterDeathTest, InputSizeCheckVerification) {
   std::vector<float> output(input.size() - 1);
 
   CascadedBiQuadFilter filter(
-      (ArrayView<const CascadedBiQuadFilter::BiQuadCoefficients>(
+      (std::span<const CascadedBiQuadFilter::BiQuadCoefficients>(
           kTransparentCoefficients)));
   EXPECT_DEATH(filter.Process(input, output), "");
 }
diff --git a/modules/audio_processing/utility/pffft_wrapper.cc b/modules/audio_processing/utility/pffft_wrapper.cc
index 59b590005d..87f5e9504b 100644
--- a/modules/audio_processing/utility/pffft_wrapper.cc
+++ b/modules/audio_processing/utility/pffft_wrapper.cc
@@ -12,8 +12,8 @@
 
 #include <cstddef>
 #include <memory>
+#include <span>
 
-#include "api/array_view.h"
 #include "rtc_base/checks.h"
 #include "third_party/pffft/src/pffft.h"
 
@@ -38,11 +38,11 @@ Pffft::FloatBuffer::~FloatBuffer() {
   pffft_aligned_free(data_);
 }
 
-ArrayView<const float> Pffft::FloatBuffer::GetConstView() const {
+std::span<const float> Pffft::FloatBuffer::GetConstView() const {
   return {data_, size_};
 }
 
-ArrayView<float> Pffft::FloatBuffer::GetView() {
+std::span<float> Pffft::FloatBuffer::GetView() {
   return {data_, size_};
 }
 
diff --git a/modules/audio_processing/utility/pffft_wrapper.h b/modules/audio_processing/utility/pffft_wrapper.h
index 590310e5dc..0a0a70890f 100644
--- a/modules/audio_processing/utility/pffft_wrapper.h
+++ b/modules/audio_processing/utility/pffft_wrapper.h
@@ -13,8 +13,7 @@
 
 #include <cstddef>
 #include <memory>
-
-#include "api/array_view.h"
+#include <span>
 
 // Forward declaration.
 struct PFFFT_Setup;
@@ -35,8 +34,8 @@ class Pffft {
     FloatBuffer& operator=(const FloatBuffer&) = delete;
     ~FloatBuffer();
 
-    ArrayView<const float> GetConstView() const;
-    ArrayView<float> GetView();
+    std::span<const float> GetConstView() const;
+    std::span<float> GetView();
 
    private:
     friend class Pffft;
@@ -68,7 +67,7 @@ class Pffft {
   // Creates a buffer of the right size.
   std::unique_ptr<FloatBuffer> CreateBuffer() const;
 
-  // TODO(https://crbug.com/webrtc/9577): Overload with ArrayView args.
+  // TODO(https://crbug.com/webrtc/9577): Overload with std::span args.
   // Computes the forward fast Fourier transform.
   void ForwardTransform(const FloatBuffer& in, FloatBuffer* out, bool ordered);
   // Computes the backward fast Fourier transform.
diff --git a/modules/audio_processing/utility/pffft_wrapper_unittest.cc b/modules/audio_processing/utility/pffft_wrapper_unittest.cc
index f1ce5209d2..c7704e715d 100644
--- a/modules/audio_processing/utility/pffft_wrapper_unittest.cc
+++ b/modules/audio_processing/utility/pffft_wrapper_unittest.cc
@@ -13,8 +13,9 @@
 #include <algorithm>
 #include <cstdlib>
 #include <memory>
+#include <span>
 
-#include "api/array_view.h"
+#include "test/gmock.h"
 #include "test/gtest.h"
 #include "third_party/pffft/src/pffft.h"
 
@@ -22,6 +23,8 @@ namespace webrtc {
 namespace test {
 namespace {
 
+using ::testing::ElementsAreArray;
+
 constexpr size_t kMaxValidSizeCheck = 1024;
 
 constexpr int kFftSizes[] = {16,   32,   64,     96,   128,  160,     192,  256,
@@ -41,15 +44,6 @@ double frand() {
   return std::rand() / static_cast<double>(RAND_MAX);
 }
 
-void ExpectArrayViewsEquality(ArrayView<const float> a,
-                              ArrayView<const float> b) {
-  ASSERT_EQ(a.size(), b.size());
-  for (size_t i = 0; i < a.size(); ++i) {
-    SCOPED_TRACE(i);
-    EXPECT_EQ(a[i], b[i]);
-  }
-}
-
 // Compares the output of the PFFFT C++ wrapper to that of the C PFFFT.
 // Bit-exactness is expected.
 void PffftValidateWrapper(size_t fft_size, bool complex_fft) {
@@ -75,8 +69,8 @@ void PffftValidateWrapper(size_t fft_size, bool complex_fft) {
   auto out_wrapper = pffft_wrapper.CreateBuffer();
 
   // Input and output buffers views.
-  ArrayView<float> in_view(in, num_floats);
-  ArrayView<float> out_view(out, num_floats);
+  std::span<float> in_view(in, num_floats);
+  std::span<float> out_view(out, num_floats);
   auto in_wrapper_view = in_wrapper->GetView();
   EXPECT_EQ(in_wrapper_view.size(), num_floats);
   auto out_wrapper_view = out_wrapper->GetConstView();
@@ -91,7 +85,7 @@ void PffftValidateWrapper(size_t fft_size, bool complex_fft) {
   pffft_transform(pffft_status, in, out, scratch, PFFFT_FORWARD);
   pffft_wrapper.ForwardTransform(*in_wrapper, out_wrapper.get(),
                                  /*ordered=*/false);
-  ExpectArrayViewsEquality(out_view, out_wrapper_view);
+  EXPECT_THAT(out_wrapper_view, ElementsAreArray(out_view));
 
   // Copy the FFT results into the input buffers to compute the backward FFT.
   std::copy(out_view.begin(), out_view.end(), in_view.begin());
@@ -102,7 +96,7 @@ void PffftValidateWrapper(size_t fft_size, bool complex_fft) {
   pffft_transform(pffft_status, in, out, scratch, PFFFT_BACKWARD);
   pffft_wrapper.BackwardTransform(*in_wrapper, out_wrapper.get(),
                                   /*ordered=*/false);
-  ExpectArrayViewsEquality(out_view, out_wrapper_view);
+  EXPECT_THAT(out_wrapper_view, ElementsAreArray(out_view));
 
   pffft_destroy_setup(pffft_status);
   pffft_aligned_free(in);
diff --git a/modules/audio_processing/vad/voice_activity_detector.h b/modules/audio_processing/vad/voice_activity_detector.h
index 5754e89523..64c42a478b 100644
--- a/modules/audio_processing/vad/voice_activity_detector.h
+++ b/modules/audio_processing/vad/voice_activity_detector.h
@@ -33,7 +33,7 @@ class VoiceActivityDetector {
   ~VoiceActivityDetector();
 
   // Processes each audio chunk and estimates the voice probability.
-  // TODO(bugs.webrtc.org/7494): Switch to ArrayView and remove
+  // TODO(bugs.webrtc.org/7494): Switch to std::span and remove
   // `sample_rate_hz`.
   void ProcessChunk(const int16_t* audio, size_t length, int sample_rate_hz);
 

From 4070f1298af0dedb669526aa829d353820bd93f0 Mon Sep 17 00:00:00 2001
From: Evan Shrubsole <eshr@webrtc.org>
Date: Thu, 19 Mar 2026 08:14:22 +0000
Subject: [PATCH 418/472] Use GlobalSimulatedTimeController in PortTest

Bug: webrtc:381524905,webrtc:42223992,webrtc:469327588
Change-Id: If4bb795c0ab03e889fbe71b968c4be786a6a6964
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/458301
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47271}
---
 p2p/BUILD.gn              |   1 +
 p2p/base/port_unittest.cc | 949 +++++++++++++++++++++-----------------
 2 files changed, 523 insertions(+), 427 deletions(-)

diff --git a/p2p/BUILD.gn b/p2p/BUILD.gn
index a18edbb3aa..bc12178f5e 100644
--- a/p2p/BUILD.gn
+++ b/p2p/BUILD.gn
@@ -1220,6 +1220,7 @@ if (rtc_include_tests) {
       "../rtc_base:timeutils",
       "../rtc_base/network:received_packet",
       "../rtc_base/network:sent_packet",
+      "../rtc_base/system:unused",
       "../system_wrappers:metrics",
       "../test:create_test_environment",
       "../test:create_test_field_trials",
diff --git a/p2p/base/port_unittest.cc b/p2p/base/port_unittest.cc
index 82b005ccf2..45d1d90397 100644
--- a/p2p/base/port_unittest.cc
+++ b/p2p/base/port_unittest.cc
@@ -25,7 +25,6 @@
 #include "api/async_dns_resolver.h"
 #include "api/candidate.h"
 #include "api/environment/environment.h"
-#include "api/environment/environment_factory.h"
 #include "api/field_trials.h"
 #include "api/packet_socket_factory.h"
 #include "api/test/mock_packet_socket_factory.h"
@@ -56,8 +55,6 @@
 #include "rtc_base/checks.h"
 #include "rtc_base/crypto_random.h"
 #include "rtc_base/dscp.h"
-#include "rtc_base/fake_clock.h"
-#include "rtc_base/gunit.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/net_helper.h"
 #include "rtc_base/network.h"
@@ -65,12 +62,14 @@
 #include "rtc_base/network_constants.h"
 #include "rtc_base/socket.h"
 #include "rtc_base/socket_address.h"
+#include "rtc_base/system/unused.h"
 #include "rtc_base/thread.h"
 #include "rtc_base/virtual_socket_server.h"
 #include "test/create_test_environment.h"
 #include "test/create_test_field_trials.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/time_controller/simulated_time_controller.h"
 #include "test/wait_until.h"
 
 namespace webrtc {
@@ -82,8 +81,8 @@ using ::testing::IsTrue;
 using ::testing::NotNull;
 using ::testing::Return;
 
-constexpr int kDefaultTimeout = 3000;
-constexpr int kShortTimeout = 1000;
+constexpr TimeDelta kDefaultTimeout = TimeDelta::Millis(3000);
+constexpr TimeDelta kShortTimeout = TimeDelta::Millis(1000);
 constexpr TimeDelta kMaxExpectedSimulatedRtt = TimeDelta::Millis(200);
 constexpr TimeDelta kEpsilon = TimeDelta::Millis(1);
 const SocketAddress kLocalAddr1("192.168.1.2", 0);
@@ -132,7 +131,6 @@ bool WriteStunMessage(const StunMessage& msg, ByteBufferWriter* buf) {
   return msg.Write(buf);
 }
 
-
 bool GetStunMessageFromBufferWriter(TestPort* port,
                                     ByteBufferWriter* buf,
                                     const SocketAddress& addr,
@@ -146,20 +144,22 @@ void SendPingAndReceiveResponse(Connection* lconn,
                                 TestPort* lport,
                                 Connection* rconn,
                                 TestPort* rport,
-                                ScopedFakeClock* clock,
+                                GlobalSimulatedTimeController& time_controller,
                                 int64_t ms) {
   lconn->Ping();
-  ASSERT_THAT(WaitUntil([&] { return lport->last_stun_msg(); }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return lport->last_stun_msg(); }, IsTrue(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller}),
+      IsRtcOk());
   ASSERT_GT(lport->last_stun_buf().size(), 0u);
   rconn->OnReadPacket(
       ReceivedIpPacket(lport->last_stun_buf(), SocketAddress(), std::nullopt));
 
-  clock->AdvanceTime(TimeDelta::Millis(ms));
-  ASSERT_THAT(WaitUntil([&] { return rport->last_stun_msg(); }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  time_controller.AdvanceTime(TimeDelta::Millis(ms));
+  ASSERT_THAT(
+      WaitUntil([&] { return rport->last_stun_msg(); }, IsTrue(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller}),
+      IsRtcOk());
   ASSERT_GT(rport->last_stun_buf().size(), 0u);
   lconn->OnReadPacket(
       ReceivedIpPacket(rport->last_stun_buf(), SocketAddress(), std::nullopt));
@@ -168,7 +168,9 @@ void SendPingAndReceiveResponse(Connection* lconn,
 class TestChannel {
  public:
   // Takes ownership of `p1` (but not `p2`).
-  explicit TestChannel(std::unique_ptr<Port> p1) : port_(std::move(p1)) {
+  TestChannel(std::unique_ptr<Port> p1,
+              GlobalSimulatedTimeController& time_controller)
+      : time_controller_(time_controller), port_(std::move(p1)) {
     port_->SubscribePortComplete(this,
                                  [this](Port* port) { OnPortComplete(port); });
     port_->SubscribeUnknownAddress(
@@ -188,7 +190,10 @@ class TestChannel {
   const SocketAddress& remote_address() { return remote_address_; }
   std::string remote_fragment() { return remote_frag_; }
 
-  void Start() { port_->PrepareAddress(); }
+  void Start() {
+    port_->PrepareAddress();
+    time_controller_.AdvanceTime(TimeDelta::Zero());
+  }
   void CreateConnection(const Candidate& remote_candidate) {
     RTC_DCHECK(!conn_);
     conn_ = port_->CreateConnection(remote_candidate, Port::ORIGIN_MESSAGE);
@@ -298,6 +303,7 @@ class TestChannel {
     connection_ready_to_send_ = true;
   }
 
+  GlobalSimulatedTimeController& time_controller_;
   IceMode ice_mode_ = ICEMODE_FULL;
   std::unique_ptr<Port> port_;
 
@@ -314,14 +320,16 @@ class PortTest : public ::testing::Test {
  public:
   PortTest()
       : ss_(new VirtualSocketServer()),
-        main_(ss_.get()),
+        time_controller_(Timestamp::Millis(1000), ss_.get()),
+        main_(time_controller_.GetMainThread()),
+        env_(CreateTestEnvironment({.time = &time_controller_})),
         socket_factory_(ss_.get()),
         nat_factory1_(ss_.get(), kNatAddr1, SocketAddress()),
         nat_factory2_(ss_.get(), kNatAddr2, SocketAddress()),
         nat_socket_factory1_(&nat_factory1_),
         nat_socket_factory2_(&nat_factory2_),
-        stun_server_(TestStunServer::Create(env_, kStunAddr, *ss_, main_)),
-        turn_server_(env_, &main_, ss_.get(), kTurnUdpIntAddr, kTurnUdpExtAddr),
+        stun_server_(TestStunServer::Create(env_, kStunAddr, *ss_, *main_)),
+        turn_server_(env_, main_, ss_.get(), kTurnUdpIntAddr, kTurnUdpExtAddr),
         username_(CreateRandomString(ICE_UFRAG_LENGTH)),
         password_(CreateRandomString(ICE_PWD_LENGTH)),
         role_conflict_(false),
@@ -331,10 +339,13 @@ class PortTest : public ::testing::Test {
     // Workaround for tests that trigger async destruction of objects that we
     // need to give an opportunity here to run, before proceeding with other
     // teardown.
-    Thread::Current()->ProcessMessages(0);
+    time_controller_.AdvanceTime(TimeDelta::Zero());
   }
 
  protected:
+  VirtualSocketServer* vss() { return ss_.get(); }
+  const Environment& env() const { return env_; }
+
   std::string password() { return password_; }
 
   void TestLocalToLocal() {
@@ -442,7 +453,7 @@ class PortTest : public ::testing::Test {
   std::unique_ptr<UDPPort> CreateUdpPort(const SocketAddress& addr,
                                          PacketSocketFactory* socket_factory) {
     auto port = UDPPort::Create({.env = env_,
-                                 .network_thread = &main_,
+                                 .network_thread = main_,
                                  .socket_factory = socket_factory,
                                  .network = MakeNetwork(addr),
                                  .ice_username_fragment = username_,
@@ -458,7 +469,7 @@ class PortTest : public ::testing::Test {
       PacketSocketFactory* socket_factory) {
     auto port = UDPPort::Create(
         {.env = env_,
-         .network_thread = &main_,
+         .network_thread = main_,
          .socket_factory = socket_factory,
          .network = MakeNetworkMultipleAddrs(global_addr, link_local_addr),
          .ice_username_fragment = username_,
@@ -473,7 +484,7 @@ class PortTest : public ::testing::Test {
   std::unique_ptr<TCPPort> CreateTcpPort(const SocketAddress& addr,
                                          PacketSocketFactory* socket_factory) {
     auto port = TCPPort::Create({.env = env_,
-                                 .network_thread = &main_,
+                                 .network_thread = main_,
                                  .socket_factory = socket_factory,
                                  .network = MakeNetwork(addr),
                                  .ice_username_fragment = username_,
@@ -488,7 +499,7 @@ class PortTest : public ::testing::Test {
     ServerAddresses stun_servers;
     stun_servers.insert(kStunAddr);
     auto port = StunPort::Create({.env = env_,
-                                  .network_thread = &main_,
+                                  .network_thread = main_,
                                   .socket_factory = socket_factory,
                                   .network = MakeNetwork(addr),
                                   .ice_username_fragment = username_,
@@ -520,7 +531,7 @@ class PortTest : public ::testing::Test {
     config.credentials = kRelayCredentials;
     ProtocolAddress server_address(server_addr, int_proto);
     CreateRelayPortArgs args = {.env = env_};
-    args.network_thread = &main_;
+    args.network_thread = main_;
     args.socket_factory = socket_factory;
     args.network = MakeNetwork(addr);
     args.username = username_;
@@ -535,8 +546,8 @@ class PortTest : public ::testing::Test {
 
   std::unique_ptr<NATServer> CreateNatServer(const SocketAddress& addr,
                                              NATType type) {
-    return std::make_unique<NATServer>(env_, type, main_, ss_.get(), addr, addr,
-                                       main_, ss_.get(), addr);
+    return std::make_unique<NATServer>(env_, type, *main_, ss_.get(), addr,
+                                       addr, *main_, ss_.get(), addr);
   }
   static const char* StunName(NATType type) {
     switch (type) {
@@ -586,20 +597,23 @@ class PortTest : public ::testing::Test {
   // TCP reconnecting mechanism before entering this function.
   void ConnectStartedChannels(TestChannel* ch1, TestChannel* ch2) {
     ASSERT_TRUE(ch1->conn());
-    EXPECT_THAT(WaitUntil([&] { return ch1->conn()->connected(); }, IsTrue(),
-                          {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-                IsRtcOk());  // for TCP connect
+    EXPECT_THAT(
+        WaitUntil([&] { return ch1->conn()->connected(); }, IsTrue(),
+                  {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+        IsRtcOk());  // for TCP connect
     ch1->Ping();
-    EXPECT_TRUE(WaitUntil([&] { return !ch2->remote_address().IsNil(); },
-                          {.timeout = TimeDelta::Millis(kShortTimeout)}));
+    EXPECT_TRUE(
+        WaitUntil([&] { return !ch2->remote_address().IsNil(); },
+                  {.timeout = kShortTimeout, .clock = &time_controller_}));
 
     // Send a ping from dst to src.
     ch2->AcceptConnection(GetCandidate(ch1->port()));
     ch2->Ping();
-    EXPECT_THAT(WaitUntil([&] { return ch2->conn()->write_state(); },
-                          Eq(Connection::STATE_WRITABLE),
-                          {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-                IsRtcOk());
+    EXPECT_THAT(
+        WaitUntil([&] { return ch2->conn()->write_state(); },
+                  Eq(Connection::STATE_WRITABLE),
+                  {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+        IsRtcOk());
   }
 
   // This connects and disconnects the provided channels in the same sequence as
@@ -628,12 +642,14 @@ class PortTest : public ::testing::Test {
                                  tcp_conn2->socket()->GetLocalAddress()));
 
     // Wait for both OnClose are delivered.
-    EXPECT_THAT(WaitUntil([&] { return !ch1->conn()->connected(); }, IsTrue(),
-                          {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-                IsRtcOk());
-    EXPECT_THAT(WaitUntil([&] { return !ch2->conn()->connected(); }, IsTrue(),
-                          {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-                IsRtcOk());
+    EXPECT_THAT(
+        WaitUntil([&] { return !ch1->conn()->connected(); }, IsTrue(),
+                  {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+        IsRtcOk());
+    EXPECT_THAT(
+        WaitUntil([&] { return !ch2->conn()->connected(); }, IsTrue(),
+                  {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+        IsRtcOk());
 
     // Ensure redundant SignalClose events on TcpConnection won't break tcp
     // reconnection. Chromium will fire SignalClose for all outstanding IPC
@@ -644,9 +660,10 @@ class PortTest : public ::testing::Test {
     // Speed up destroying ch2's connection such that the test is ready to
     // accept a new connection from ch1 before ch1's connection destroys itself.
     ch2->Stop();
-    EXPECT_THAT(WaitUntil([&] { return ch2->conn(); }, IsNull(),
-                          {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-                IsRtcOk());
+    EXPECT_THAT(
+        WaitUntil([&] { return ch2->conn(); }, IsNull(),
+                  {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+        IsRtcOk());
   }
 
   void TestTcpReconnect(bool ping_after_disconnected,
@@ -660,30 +677,32 @@ class PortTest : public ::testing::Test {
     port2->set_component(ICE_CANDIDATE_COMPONENT_DEFAULT);
 
     // Set up channels and ensure both ports will be deleted.
-    TestChannel ch1(std::move(port1));
-    TestChannel ch2(std::move(port2));
+    TestChannel ch1(std::move(port1), time_controller_);
+    TestChannel ch2(std::move(port2), time_controller_);
     EXPECT_EQ(0, ch1.complete_count());
     EXPECT_EQ(0, ch2.complete_count());
 
     ch1.Start();
     ch2.Start();
-    ASSERT_THAT(WaitUntil([&] { return ch1.complete_count(); }, Eq(1),
-                          {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-                IsRtcOk());
-    ASSERT_THAT(WaitUntil([&] { return ch2.complete_count(); }, Eq(1),
-                          {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-                IsRtcOk());
+    ASSERT_THAT(
+        WaitUntil([&] { return ch1.complete_count(); }, Eq(1),
+                  {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+        IsRtcOk());
+    ASSERT_THAT(
+        WaitUntil([&] { return ch2.complete_count(); }, Eq(1),
+                  {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+        IsRtcOk());
 
     // Initial connecting the channel, create connection on channel1.
     ch1.CreateConnection(GetCandidate(ch2.port()));
     ConnectStartedChannels(&ch1, &ch2);
 
     // Shorten the timeout period.
-    const int kTcpReconnectTimeout = kDefaultTimeout;
+    const TimeDelta kTcpReconnectTimeout = kDefaultTimeout;
     static_cast<TCPConnection*>(ch1.conn())
-        ->set_reconnection_timeout(kTcpReconnectTimeout);
+        ->set_reconnection_timeout(kTcpReconnectTimeout.ms());
     static_cast<TCPConnection*>(ch2.conn())
-        ->set_reconnection_timeout(kTcpReconnectTimeout);
+        ->set_reconnection_timeout(kTcpReconnectTimeout.ms());
 
     EXPECT_FALSE(ch1.connection_ready_to_send());
     EXPECT_FALSE(ch2.connection_ready_to_send());
@@ -705,15 +724,17 @@ class PortTest : public ::testing::Test {
       }
 
       // Wait for channel's outgoing TCPConnection connected.
-      EXPECT_THAT(WaitUntil([&] { return ch1.conn()->connected(); }, IsTrue(),
-                            {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-                  IsRtcOk());
+      EXPECT_THAT(
+          WaitUntil([&] { return ch1.conn()->connected(); }, IsTrue(),
+                    {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+          IsRtcOk());
 
       // Verify that we could still connect channels.
       ConnectStartedChannels(&ch1, &ch2);
       EXPECT_THAT(
-          WaitUntil([&] { return ch1.connection_ready_to_send(); }, IsTrue(),
-                    {.timeout = TimeDelta::Millis(kTcpReconnectTimeout)}),
+          WaitUntil(
+              [&] { return ch1.connection_ready_to_send(); }, IsTrue(),
+              {.timeout = kTcpReconnectTimeout, .clock = &time_controller_}),
           IsRtcOk());
       // Channel2 is the passive one so a new connection is created during
       // reconnect. This new connection should never have issued ENOTCONN
@@ -724,8 +745,8 @@ class PortTest : public ::testing::Test {
       // Since the reconnection never happens, the connections should have been
       // destroyed after the timeout.
       EXPECT_THAT(WaitUntil([&] { return !ch1.conn(); }, IsTrue(),
-                            {.timeout = TimeDelta::Millis(kTcpReconnectTimeout +
-                                                          kDefaultTimeout)}),
+                            {.timeout = kTcpReconnectTimeout + kDefaultTimeout,
+                             .clock = &time_controller_}),
                   IsRtcOk());
       EXPECT_TRUE(!ch2.conn());
     }
@@ -733,12 +754,14 @@ class PortTest : public ::testing::Test {
     // Tear down and ensure that goes smoothly.
     ch1.Stop();
     ch2.Stop();
-    EXPECT_THAT(WaitUntil([&] { return ch1.conn(); }, IsNull(),
-                          {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-                IsRtcOk());
-    EXPECT_THAT(WaitUntil([&] { return ch2.conn(); }, IsNull(),
-                          {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-                IsRtcOk());
+    EXPECT_THAT(
+        WaitUntil([&] { return ch1.conn(); }, IsNull(),
+                  {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+        IsRtcOk());
+    EXPECT_THAT(
+        WaitUntil([&] { return ch2.conn(); }, IsNull(),
+                  {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+        IsRtcOk());
   }
 
   std::unique_ptr<IceMessage> CreateStunMessage(StunMessageType type) {
@@ -758,12 +781,14 @@ class PortTest : public ::testing::Test {
       absl::string_view username,
       absl::string_view password,
       const FieldTrialsView* field_trials = nullptr) {
-    Port::PortParametersRef args = {.env = CreateEnvironment(field_trials),
-                                    .network_thread = &main_,
-                                    .socket_factory = &socket_factory_,
-                                    .network = MakeNetwork(addr),
-                                    .ice_username_fragment = username,
-                                    .ice_password = password};
+    Port::PortParametersRef args = {
+        .env = CreateTestEnvironment(
+            {.field_trials = field_trials, .time = &time_controller_}),
+        .network_thread = main_,
+        .socket_factory = &socket_factory_,
+        .network = MakeNetwork(addr),
+        .ice_username_fragment = username,
+        .ice_password = password};
     auto port = std::make_unique<TestPort>(args, 0, 0);
     port->SubscribeRoleConflict([this]() { OnRoleConflict(); });
     return port;
@@ -783,7 +808,7 @@ class PortTest : public ::testing::Test {
                                            absl::string_view username,
                                            absl::string_view password) {
     Port::PortParametersRef args = {.env = env_,
-                                    .network_thread = &main_,
+                                    .network_thread = main_,
                                     .socket_factory = &socket_factory_,
                                     .network = network,
                                     .ice_username_fragment = username,
@@ -796,7 +821,7 @@ class PortTest : public ::testing::Test {
   std::unique_ptr<TestPort> CreateRawTestPort() {
     Port::PortParametersRef args = {
         .env = env_,
-        .network_thread = &main_,
+        .network_thread = main_,
         .socket_factory = &socket_factory_,
         .network = MakeNetwork(kLocalAddr1),
     };
@@ -818,18 +843,16 @@ class PortTest : public ::testing::Test {
     return &nat_socket_factory1_;
   }
 
-  VirtualSocketServer* vss() { return ss_.get(); }
+ protected:
+  std::unique_ptr<VirtualSocketServer> ss_;
+  GlobalSimulatedTimeController time_controller_;
+  Thread* main_;
+  const Environment env_;
 
-  const Environment& env() const { return env_; }
-
- private:
-  const Environment env_ = CreateTestEnvironment();
   // When a "create port" helper method is called with an IP, we create a
   // Network with that IP and add it to this list. Using a list instead of a
   // vector so that when it grows, pointers aren't invalidated.
   std::list<Network> networks_;
-  std::unique_ptr<VirtualSocketServer> ss_;
-  AutoSocketServerThread main_;
   BasicPacketSocketFactory socket_factory_;
   std::unique_ptr<NATServer> nat_server1_;
   std::unique_ptr<NATServer> nat_server2_;
@@ -853,40 +876,43 @@ void PortTest::TestConnectivity(absl::string_view name1,
                                 bool same_addr1,
                                 bool same_addr2,
                                 bool possible) {
-  ScopedFakeClock clock;
   RTC_LOG(LS_INFO) << "Test: " << name1 << " to " << name2 << ": ";
   port1->set_component(ICE_CANDIDATE_COMPONENT_DEFAULT);
   port2->set_component(ICE_CANDIDATE_COMPONENT_DEFAULT);
 
   // Set up channels and ensure both ports will be deleted.
-  TestChannel ch1(std::move(port1));
-  TestChannel ch2(std::move(port2));
+  TestChannel ch1(std::move(port1), time_controller_);
+  TestChannel ch2(std::move(port2), time_controller_);
   EXPECT_EQ(0, ch1.complete_count());
   EXPECT_EQ(0, ch2.complete_count());
 
   // Acquire addresses.
   ch1.Start();
   ch2.Start();
-  ASSERT_THAT(WaitUntil([&] { return ch1.complete_count(); }, Eq(1),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout),
-                         .clock = &clock}),
-              IsRtcOk());
-  ASSERT_THAT(WaitUntil([&] { return ch2.complete_count(); }, Eq(1),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout),
-                         .clock = &clock}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return ch1.complete_count(); }, Eq(1),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return ch2.complete_count(); }, Eq(1),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
 
   // Send a ping from src to dst. This may or may not make it.
   ch1.CreateConnection(GetCandidate(ch2.port()));
   ASSERT_TRUE(ch1.conn() != nullptr);
-  EXPECT_THAT(WaitUntil([&] { return ch1.conn()->connected(); }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout),
-                         .clock = &clock}),
-              IsRtcOk());  // for TCP connect
-  ch1.Ping();
-  SIMULATED_WAIT(!ch2.remote_address().IsNil(), kShortTimeout, clock);
 
+  EXPECT_THAT(
+      WaitUntil([&] { return ch1.conn()->connected(); }, IsTrue(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());  // for TCP connect
+  ch1.Ping();
   if (accept) {
+    EXPECT_THAT(
+        WaitUntil([&] { return !ch2.remote_address().IsNil(); }, IsTrue(),
+                  {.timeout = kShortTimeout, .clock = &time_controller_}),
+        IsRtcOk());
+
     // We are able to send a ping from src to dst. This is the case when
     // sending to UDP ports and cone NATs.
     EXPECT_TRUE(ch1.remote_address().IsNil());
@@ -901,12 +927,13 @@ void PortTest::TestConnectivity(absl::string_view name1,
     // Send a ping from dst to src.
     ch2.AcceptConnection(GetCandidate(ch1.port()));
     ASSERT_TRUE(ch2.conn() != nullptr);
+
     ch2.Ping();
-    EXPECT_THAT(WaitUntil([&] { return ch2.conn()->write_state(); },
-                          Eq(Connection::STATE_WRITABLE),
-                          {.timeout = TimeDelta::Millis(kDefaultTimeout),
-                           .clock = &clock}),
-                IsRtcOk());
+    EXPECT_THAT(
+        WaitUntil([&] { return ch2.conn()->write_state(); },
+                  Eq(Connection::STATE_WRITABLE),
+                  {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+        IsRtcOk());
   } else {
     // We can't send a ping from src to dst, so flip it around. This will happen
     // when the destination NAT is addr/port restricted or symmetric.
@@ -916,9 +943,12 @@ void PortTest::TestConnectivity(absl::string_view name1,
     // Send a ping from dst to src. Again, this may or may not make it.
     ch2.CreateConnection(GetCandidate(ch1.port()));
     ASSERT_TRUE(ch2.conn() != nullptr);
+
     ch2.Ping();
-    SIMULATED_WAIT(ch2.conn()->write_state() == Connection::STATE_WRITABLE,
-                   kShortTimeout, clock);
+    RTC_UNUSED(
+        WaitUntil([&] { return ch2.conn()->write_state(); },
+                  Eq(Connection::STATE_WRITABLE),
+                  {.timeout = kShortTimeout, .clock = &time_controller_}));
 
     if (same_addr1 && same_addr2) {
       // The new ping got back to the source.
@@ -929,25 +959,24 @@ void PortTest::TestConnectivity(absl::string_view name1,
       // through.  So we will have to do another.
       if (ch1.conn()->write_state() == Connection::STATE_WRITE_INIT) {
         ch1.Ping();
-        EXPECT_THAT(WaitUntil([&] { return ch1.conn()->write_state(); },
-                              Eq(Connection::STATE_WRITABLE),
-                              {.timeout = TimeDelta::Millis(kDefaultTimeout),
-                               .clock = &clock}),
-                    IsRtcOk());
+        EXPECT_THAT(
+            WaitUntil([&] { return ch1.conn()->write_state(); },
+                      Eq(Connection::STATE_WRITABLE),
+                      {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+            IsRtcOk());
       }
     } else if (!same_addr1 && possible) {
       // The new ping went to the candidate address, but that address was bad.
       // This will happen when the source NAT is symmetric.
-      EXPECT_TRUE(ch1.remote_address().IsNil());
-      EXPECT_TRUE(ch2.remote_address().IsNil());
+      // With simulated time and some NAT types (like Turn/AR-Nat), the packet
+      // can be delivered early.
 
       // However, since we have now sent a ping to the source IP, we should be
       // able to get a ping from it. This gives us the real source address.
       ch1.Ping();
       EXPECT_THAT(
-          WaitUntil(
-              [&] { return !ch2.remote_address().IsNil(); }, IsTrue(),
-              {.timeout = TimeDelta::Millis(kDefaultTimeout), .clock = &clock}),
+          WaitUntil([&] { return !ch2.remote_address().IsNil(); }, IsTrue(),
+                    {.timeout = kDefaultTimeout, .clock = &time_controller_}),
           IsRtcOk());
       EXPECT_FALSE(ch2.conn()->receiving());
       EXPECT_TRUE(ch1.remote_address().IsNil());
@@ -955,12 +984,13 @@ void PortTest::TestConnectivity(absl::string_view name1,
       // Pick up the actual address and establish the connection.
       ch2.AcceptConnection(GetCandidate(ch1.port()));
       ASSERT_TRUE(ch2.conn() != nullptr);
+
       ch2.Ping();
-      EXPECT_THAT(WaitUntil([&] { return ch2.conn()->write_state(); },
-                            Eq(Connection::STATE_WRITABLE),
-                            {.timeout = TimeDelta::Millis(kDefaultTimeout),
-                             .clock = &clock}),
-                  IsRtcOk());
+      EXPECT_THAT(
+          WaitUntil([&] { return ch2.conn()->write_state(); },
+                    Eq(Connection::STATE_WRITABLE),
+                    {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+          IsRtcOk());
     } else if (!same_addr2 && possible) {
       // The new ping came in, but from an unexpected address. This will happen
       // when the destination NAT is symmetric.
@@ -969,18 +999,21 @@ void PortTest::TestConnectivity(absl::string_view name1,
 
       // Update our address and complete the connection.
       ch1.AcceptConnection(GetCandidate(ch2.port()));
+
       ch1.Ping();
-      EXPECT_THAT(WaitUntil([&] { return ch1.conn()->write_state(); },
-                            Eq(Connection::STATE_WRITABLE),
-                            {.timeout = TimeDelta::Millis(kDefaultTimeout),
-                             .clock = &clock}),
-                  IsRtcOk());
+      EXPECT_THAT(
+          WaitUntil([&] { return ch1.conn()->write_state(); },
+                    Eq(Connection::STATE_WRITABLE),
+                    {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+          IsRtcOk());
     } else {  // (!possible)
       // There should be s no way for the pings to reach each other. Check it.
       EXPECT_TRUE(ch1.remote_address().IsNil());
       EXPECT_TRUE(ch2.remote_address().IsNil());
       ch1.Ping();
-      SIMULATED_WAIT(!ch2.remote_address().IsNil(), kShortTimeout, clock);
+      RTC_UNUSED(
+          WaitUntil([&] { return !ch2.remote_address().IsNil(); }, IsTrue(),
+                    {.timeout = kShortTimeout, .clock = &time_controller_}));
       EXPECT_TRUE(ch1.remote_address().IsNil());
       EXPECT_TRUE(ch2.remote_address().IsNil());
     }
@@ -990,9 +1023,13 @@ void PortTest::TestConnectivity(absl::string_view name1,
   ASSERT_TRUE(ch1.conn() != nullptr);
   ASSERT_TRUE(ch2.conn() != nullptr);
   if (possible) {
-    EXPECT_TRUE(ch1.conn()->receiving());
+    EXPECT_THAT(WaitUntil([&] { return ch1.conn()->receiving(); }, IsTrue(),
+                          {.clock = &time_controller_}),
+                IsRtcOk());
     EXPECT_EQ(Connection::STATE_WRITABLE, ch1.conn()->write_state());
-    EXPECT_TRUE(ch2.conn()->receiving());
+    EXPECT_THAT(WaitUntil([&] { return ch2.conn()->receiving(); }, IsTrue(),
+                          {.clock = &time_controller_}),
+                IsRtcOk());
     EXPECT_EQ(Connection::STATE_WRITABLE, ch2.conn()->write_state());
   } else {
     EXPECT_FALSE(ch1.conn()->receiving());
@@ -1004,14 +1041,14 @@ void PortTest::TestConnectivity(absl::string_view name1,
   // Tear down and ensure that goes smoothly.
   ch1.Stop();
   ch2.Stop();
-  EXPECT_THAT(WaitUntil([&] { return ch1.conn(); }, IsNull(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout),
-                         .clock = &clock}),
-              IsRtcOk());
-  EXPECT_THAT(WaitUntil([&] { return ch2.conn(); }, IsNull(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout),
-                         .clock = &clock}),
-              IsRtcOk());
+  EXPECT_THAT(
+      WaitUntil([&] { return ch1.conn(); }, IsNull(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
+  EXPECT_THAT(
+      WaitUntil([&] { return ch2.conn(); }, IsNull(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
 }
 
 class FakePacketSocketFactory : public PacketSocketFactory {
@@ -1301,13 +1338,14 @@ TEST_F(PortTest, TestTcpNeverConnect) {
   port1->set_component(ICE_CANDIDATE_COMPONENT_DEFAULT);
 
   // Set up a channel and ensure the port will be deleted.
-  TestChannel ch1(std::move(port1));
+  TestChannel ch1(std::move(port1), time_controller_);
   EXPECT_EQ(0, ch1.complete_count());
 
   ch1.Start();
-  ASSERT_THAT(WaitUntil([&] { return ch1.complete_count(); }, Eq(1),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return ch1.complete_count(); }, Eq(1),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
 
   std::unique_ptr<Socket> server(
       vss()->CreateSocket(kLocalAddr2.family(), SOCK_STREAM));
@@ -1319,9 +1357,10 @@ TEST_F(PortTest, TestTcpNeverConnect) {
 
   ch1.CreateConnection(c);
   EXPECT_TRUE(ch1.conn());
-  EXPECT_THAT(WaitUntil([&] { return !ch1.conn(); }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());  // for TCP connect
+  EXPECT_THAT(
+      WaitUntil([&] { return !ch1.conn(); }, IsTrue(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());  // for TCP connect
 }
 
 /* TODO(?): Enable these once testrelayserver can accept external TCP.
@@ -1351,17 +1390,19 @@ TEST_F(PortTest, TestSslTcpToSslTcpRelay) {
 // ii) it has not received anything for kDeadConnectionReceiveTimeout since
 //     last receiving.
 TEST_F(PortTest, TestConnectionDead) {
-  TestChannel ch1(CreateUdpPort(kLocalAddr1));
-  TestChannel ch2(CreateUdpPort(kLocalAddr2));
+  TestChannel ch1(CreateUdpPort(kLocalAddr1), time_controller_);
+  TestChannel ch2(CreateUdpPort(kLocalAddr2), time_controller_);
   // Acquire address.
   ch1.Start();
   ch2.Start();
-  ASSERT_THAT(WaitUntil([&] { return ch1.complete_count(); }, Eq(1),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
-  ASSERT_THAT(WaitUntil([&] { return ch2.complete_count(); }, Eq(1),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return ch1.complete_count(); }, Eq(1),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return ch2.complete_count(); }, Eq(1),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
 
   // Test case that the connection has never received anything.
   Timestamp before_created = env().clock().CurrentTime();
@@ -1371,18 +1412,19 @@ TEST_F(PortTest, TestConnectionDead) {
   ASSERT_NE(conn, nullptr);
   // It is not dead if it is after kMinConnectionLifetime but not pruned.
   conn->UpdateState(after_created + kMinConnectionLifetime + kEpsilon);
-  Thread::Current()->ProcessMessages(0);
+  time_controller_.AdvanceTime(TimeDelta::Zero());
   EXPECT_TRUE(ch1.conn() != nullptr);
   // It is not dead if it is before kMinConnectionLifetime and pruned.
   conn->UpdateState(before_created + kMinConnectionLifetime - kEpsilon);
   conn->Prune();
-  Thread::Current()->ProcessMessages(0);
+  time_controller_.AdvanceTime(TimeDelta::Zero());
   EXPECT_TRUE(ch1.conn() != nullptr);
   // It will be dead after kMinConnectionLifetime and pruned.
   conn->UpdateState(after_created + kMinConnectionLifetime + kEpsilon);
-  EXPECT_THAT(WaitUntil([&] { return ch1.conn(); }, Eq(nullptr),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  EXPECT_THAT(
+      WaitUntil([&] { return ch1.conn(); }, Eq(nullptr),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
 
   // Test case that the connection has received something.
   // Create a connection again and receive a ping.
@@ -1395,27 +1437,30 @@ TEST_F(PortTest, TestConnectionDead) {
   // The connection will be dead after kDeadConnectionReceiveTimeout
   conn->UpdateState(before_last_receiving + kDeadConnectionReceiveTimeout -
                     kEpsilon);
-  Thread::Current()->ProcessMessages(100);
+  time_controller_.AdvanceTime(TimeDelta::Zero());
   EXPECT_TRUE(ch1.conn() != nullptr);
   conn->UpdateState(after_last_receiving + kDeadConnectionReceiveTimeout +
                     kEpsilon);
-  EXPECT_THAT(WaitUntil([&] { return ch1.conn(); }, Eq(nullptr),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  EXPECT_THAT(
+      WaitUntil([&] { return ch1.conn(); }, Eq(nullptr),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
 }
 
 TEST_F(PortTest, TestConnectionDeadWithDeadConnectionTimeout) {
-  TestChannel ch1(CreateUdpPort(kLocalAddr1));
-  TestChannel ch2(CreateUdpPort(kLocalAddr2));
+  TestChannel ch1(CreateUdpPort(kLocalAddr1), time_controller_);
+  TestChannel ch2(CreateUdpPort(kLocalAddr2), time_controller_);
   // Acquire address.
   ch1.Start();
   ch2.Start();
-  ASSERT_THAT(WaitUntil([&] { return ch1.complete_count(); }, Eq(1),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
-  ASSERT_THAT(WaitUntil([&] { return ch2.complete_count(); }, Eq(1),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return ch1.complete_count(); }, Eq(1),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return ch2.complete_count(); }, Eq(1),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
 
   // Note: set field trials manually since they are parsed by
   // P2PTransportChannel but P2PTransportChannel is not used in this test.
@@ -1433,12 +1478,13 @@ TEST_F(PortTest, TestConnectionDeadWithDeadConnectionTimeout) {
   Timestamp after_last_receiving = env().clock().CurrentTime();
   // The connection will be dead after 90s
   conn->UpdateState(before_last_receiving + TimeDelta::Seconds(90) - kEpsilon);
-  Thread::Current()->ProcessMessages(100);
+  time_controller_.AdvanceTime(TimeDelta::Zero());
   EXPECT_TRUE(ch1.conn() != nullptr);
   conn->UpdateState(after_last_receiving + TimeDelta::Seconds(90) + kEpsilon);
-  EXPECT_THAT(WaitUntil([&] { return ch1.conn(); }, Eq(nullptr),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  EXPECT_THAT(
+      WaitUntil([&] { return ch1.conn(); }, Eq(nullptr),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
 }
 
 TEST_F(PortTest, TestConnectionDeadOutstandingPing) {
@@ -1449,17 +1495,19 @@ TEST_F(PortTest, TestConnectionDeadOutstandingPing) {
   port2->SetIceRole(ICEROLE_CONTROLLED);
   port2->SetIceTiebreaker(kTiebreaker2);
 
-  TestChannel ch1(std::move(port1));
-  TestChannel ch2(std::move(port2));
+  TestChannel ch1(std::move(port1), time_controller_);
+  TestChannel ch2(std::move(port2), time_controller_);
   // Acquire address.
   ch1.Start();
   ch2.Start();
-  ASSERT_THAT(WaitUntil([&] { return ch1.complete_count(); }, Eq(1),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
-  ASSERT_THAT(WaitUntil([&] { return ch2.complete_count(); }, Eq(1),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return ch1.complete_count(); }, Eq(1),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return ch2.complete_count(); }, Eq(1),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
 
   // Note: set field trials manually since they are parsed by
   // P2PTransportChannel but P2PTransportChannel is not used in this test.
@@ -1480,13 +1528,14 @@ TEST_F(PortTest, TestConnectionDeadOutstandingPing) {
   // The connection will be dead 30s after the ping was sent.
   conn->UpdateState(send_ping_timestamp + kDeadConnectionReceiveTimeout -
                     kEpsilon);
-  Thread::Current()->ProcessMessages(100);
+  time_controller_.AdvanceTime(TimeDelta::Zero());
   EXPECT_TRUE(ch1.conn() != nullptr);
   conn->UpdateState(send_ping_timestamp + kDeadConnectionReceiveTimeout +
                     kEpsilon);
-  EXPECT_THAT(WaitUntil([&] { return ch1.conn(); }, Eq(nullptr),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  EXPECT_THAT(
+      WaitUntil([&] { return ch1.conn(); }, Eq(nullptr),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
 }
 
 // This test case verifies standard ICE features in STUN messages. Currently it
@@ -1518,16 +1567,18 @@ TEST_F(PortTest, TestLoopbackCall) {
       lport->CreateConnection(lport->Candidates()[0], Port::ORIGIN_MESSAGE);
   conn->Ping();
 
-  ASSERT_THAT(WaitUntil([&] { return lport->last_stun_msg(); }, NotNull(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return lport->last_stun_msg(); }, NotNull(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   IceMessage* msg = lport->last_stun_msg();
   EXPECT_EQ(STUN_BINDING_REQUEST, msg->type());
   conn->OnReadPacket(
       ReceivedIpPacket(lport->last_stun_buf(), SocketAddress(), std::nullopt));
-  ASSERT_THAT(WaitUntil([&] { return lport->last_stun_msg(); }, NotNull(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return lport->last_stun_msg(); }, NotNull(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   msg = lport->last_stun_msg();
   EXPECT_EQ(STUN_BINDING_RESPONSE, msg->type());
 
@@ -1540,9 +1591,10 @@ TEST_F(PortTest, TestLoopbackCall) {
       lport->CreateConnection(lport->Candidates()[1], Port::ORIGIN_MESSAGE);
   conn1->Ping();
 
-  ASSERT_THAT(WaitUntil([&] { return lport->last_stun_msg(); }, NotNull(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return lport->last_stun_msg(); }, NotNull(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   msg = lport->last_stun_msg();
   EXPECT_EQ(STUN_BINDING_REQUEST, msg->type());
   std::unique_ptr<IceMessage> modified_req(
@@ -1564,9 +1616,10 @@ TEST_F(PortTest, TestLoopbackCall) {
   conn1->OnReadPacket(ReceivedIpPacket::CreateFromLegacy(
       reinterpret_cast<const char*>(buf->Data()), buf->Length(),
       /*packet_time_us=*/-1));
-  ASSERT_THAT(WaitUntil([&] { return lport->last_stun_msg(); }, NotNull(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return lport->last_stun_msg(); }, NotNull(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   msg = lport->last_stun_msg();
   EXPECT_EQ(STUN_BINDING_ERROR_RESPONSE, msg->type());
 }
@@ -1594,24 +1647,25 @@ TEST_F(PortTest, TestIceRoleConflict) {
       rport->CreateConnection(lport->Candidates()[0], Port::ORIGIN_MESSAGE);
   rconn->Ping();
 
-  ASSERT_THAT(WaitUntil([&] { return rport->last_stun_msg(); }, NotNull(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return rport->last_stun_msg(); }, NotNull(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   IceMessage* msg = rport->last_stun_msg();
   EXPECT_EQ(STUN_BINDING_REQUEST, msg->type());
   // Send rport binding request to lport.
   lconn->OnReadPacket(
       ReceivedIpPacket(rport->last_stun_buf(), SocketAddress(), std::nullopt));
 
-  ASSERT_THAT(WaitUntil([&] { return lport->last_stun_msg(); }, NotNull(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return lport->last_stun_msg(); }, NotNull(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   EXPECT_EQ(STUN_BINDING_RESPONSE, lport->last_stun_msg()->type());
   EXPECT_TRUE(role_conflict());
 }
 
 TEST_F(PortTest, TestTcpNoDelay) {
-  ScopedFakeClock clock;
   auto port1 = CreateTcpPort(kLocalAddr1);
   port1->SetIceRole(ICEROLE_CONTROLLING);
   int option_value = -1;
@@ -1624,28 +1678,30 @@ TEST_F(PortTest, TestTcpNoDelay) {
 
   // Set up a connection, and verify that option is set on connected sockets at
   // both ends.
-  TestChannel ch1(std::move(port1));
-  TestChannel ch2(std::move(port2));
+  TestChannel ch1(std::move(port1), time_controller_);
+  TestChannel ch2(std::move(port2), time_controller_);
   // Acquire addresses.
   ch1.Start();
   ch2.Start();
-  ASSERT_THAT(WaitUntil([&] { return ch1.complete_count(); }, Eq(1),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout),
-                         .clock = &clock}),
-              IsRtcOk());
-  ASSERT_THAT(WaitUntil([&] { return ch2.complete_count(); }, Eq(1),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout),
-                         .clock = &clock}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return ch1.complete_count(); }, Eq(1),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return ch2.complete_count(); }, Eq(1),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   // Connect and send a ping from src to dst.
   ch1.CreateConnection(GetCandidate(ch2.port()));
   ASSERT_TRUE(ch1.conn() != nullptr);
-  EXPECT_THAT(WaitUntil([&] { return ch1.conn()->connected(); }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout),
-                         .clock = &clock}),
-              IsRtcOk());  // for TCP connect
+  EXPECT_THAT(
+      WaitUntil([&] { return ch1.conn()->connected(); }, IsTrue(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());  // for TCP connect
   ch1.Ping();
-  SIMULATED_WAIT(!ch2.remote_address().IsNil(), kShortTimeout, clock);
+  EXPECT_THAT(WaitUntil([&] { return !ch2.remote_address().IsNil(); }, IsTrue(),
+                        {.clock = &time_controller_}),
+              IsRtcOk());
 
   // Accept the connection.
   ch2.AcceptConnection(GetCandidate(ch1.port()));
@@ -1917,9 +1973,10 @@ TEST_F(PortTest, TestSendStunMessage) {
   lconn->Ping();
 
   // Check that it's a proper BINDING-REQUEST.
-  ASSERT_THAT(WaitUntil([&] { return lport->last_stun_msg(); }, NotNull(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return lport->last_stun_msg(); }, NotNull(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   IceMessage* msg = lport->last_stun_msg();
   EXPECT_EQ(STUN_BINDING_REQUEST, msg->type());
   EXPECT_FALSE(msg->IsLegacy());
@@ -2021,9 +2078,10 @@ TEST_F(PortTest, TestSendStunMessage) {
   rconn->Ping();
   rconn->Ping();
   rconn->Ping();
-  ASSERT_THAT(WaitUntil([&] { return rport->last_stun_msg(); }, NotNull(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return rport->last_stun_msg(); }, NotNull(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   msg = rport->last_stun_msg();
   EXPECT_EQ(STUN_BINDING_REQUEST, msg->type());
   const StunUInt64Attribute* ice_controlled_attr =
@@ -2094,9 +2152,10 @@ TEST_F(PortTest, TestNomination) {
   // Send ping (including the nomination value) from `lconn` to `rconn`. This
   // should set the remote nomination of `rconn`.
   lconn->Ping();
-  ASSERT_THAT(WaitUntil([&] { return lport->last_stun_msg(); }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return lport->last_stun_msg(); }, IsTrue(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   ASSERT_GT(lport->last_stun_buf().size(), 0u);
   rconn->OnReadPacket(
       ReceivedIpPacket(lport->last_stun_buf(), SocketAddress(), std::nullopt));
@@ -2109,9 +2168,10 @@ TEST_F(PortTest, TestNomination) {
 
   // This should result in an acknowledgment sent back from `rconn` to `lconn`,
   // updating the acknowledged nomination of `lconn`.
-  ASSERT_THAT(WaitUntil([&] { return rport->last_stun_msg(); }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return rport->last_stun_msg(); }, IsTrue(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   ASSERT_GT(rport->last_stun_buf().size(), 0u);
   lconn->OnReadPacket(
       ReceivedIpPacket(rport->last_stun_buf(), SocketAddress(), std::nullopt));
@@ -2124,8 +2184,6 @@ TEST_F(PortTest, TestNomination) {
 }
 
 TEST_F(PortTest, TestRoundTripTime) {
-  ScopedFakeClock clock;
-
   auto lport = CreateTestPort(kLocalAddr1, "lfrag", "lpass");
   auto rport = CreateTestPort(kLocalAddr2, "rfrag", "rpass");
   lport->SetIceRole(ICEROLE_CONTROLLING);
@@ -2145,20 +2203,20 @@ TEST_F(PortTest, TestRoundTripTime) {
   EXPECT_EQ(0u, lconn->stats().total_round_trip_time_ms);
   EXPECT_FALSE(lconn->stats().current_round_trip_time_ms);
 
-  SendPingAndReceiveResponse(lconn, lport.get(), rconn, rport.get(), &clock,
-                             10);
+  SendPingAndReceiveResponse(lconn, lport.get(), rconn, rport.get(),
+                             time_controller_, 10);
   EXPECT_EQ(10u, lconn->stats().total_round_trip_time_ms);
   ASSERT_TRUE(lconn->stats().current_round_trip_time_ms);
   EXPECT_EQ(10u, *lconn->stats().current_round_trip_time_ms);
 
-  SendPingAndReceiveResponse(lconn, lport.get(), rconn, rport.get(), &clock,
-                             20);
+  SendPingAndReceiveResponse(lconn, lport.get(), rconn, rport.get(),
+                             time_controller_, 20);
   EXPECT_EQ(30u, lconn->stats().total_round_trip_time_ms);
   ASSERT_TRUE(lconn->stats().current_round_trip_time_ms);
   EXPECT_EQ(20u, *lconn->stats().current_round_trip_time_ms);
 
-  SendPingAndReceiveResponse(lconn, lport.get(), rconn, rport.get(), &clock,
-                             30);
+  SendPingAndReceiveResponse(lconn, lport.get(), rconn, rport.get(),
+                             time_controller_, 30);
   EXPECT_EQ(60u, lconn->stats().total_round_trip_time_ms);
   ASSERT_TRUE(lconn->stats().current_round_trip_time_ms);
   EXPECT_EQ(30u, *lconn->stats().current_round_trip_time_ms);
@@ -2179,9 +2237,10 @@ TEST_F(PortTest, TestUseCandidateAttribute) {
   Connection* lconn =
       lport->CreateConnection(rport->Candidates()[0], Port::ORIGIN_MESSAGE);
   lconn->Ping();
-  ASSERT_THAT(WaitUntil([&] { return lport->last_stun_msg(); }, NotNull(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return lport->last_stun_msg(); }, NotNull(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   IceMessage* msg = lport->last_stun_msg();
   const StunUInt64Attribute* ice_controlling_attr =
       msg->GetUInt64(STUN_ATTR_ICE_CONTROLLING);
@@ -2239,9 +2298,10 @@ TEST_F(PortTest, TestNetworkCostChange) {
   // message is handled in rconn, The rconn's remote candidate will have cost
   // kNetworkCostHigh;
   EXPECT_EQ(kNetworkCostLow, rconn->remote_candidate().network_cost());
-  ASSERT_THAT(WaitUntil([&] { return lport->last_stun_msg(); }, NotNull(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return lport->last_stun_msg(); }, NotNull(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   IceMessage* msg = lport->last_stun_msg();
   EXPECT_EQ(STUN_BINDING_REQUEST, msg->type());
   // Pass the binding request to rport.
@@ -2249,9 +2309,10 @@ TEST_F(PortTest, TestNetworkCostChange) {
       ReceivedIpPacket(lport->last_stun_buf(), SocketAddress(), std::nullopt));
 
   // Wait until rport sends the response and then check the remote network cost.
-  ASSERT_THAT(WaitUntil([&] { return rport->last_stun_msg(); }, NotNull(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return rport->last_stun_msg(); }, NotNull(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   EXPECT_EQ(kNetworkCostHigh, rconn->remote_candidate().network_cost());
 }
 
@@ -2272,9 +2333,10 @@ TEST_F(PortTest, TestNetworkInfoAttribute) {
   Connection* lconn =
       lport->CreateConnection(rport->Candidates()[0], Port::ORIGIN_MESSAGE);
   lconn->Ping();
-  ASSERT_THAT(WaitUntil([&] { return lport->last_stun_msg(); }, NotNull(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return lport->last_stun_msg(); }, NotNull(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   IceMessage* msg = lport->last_stun_msg();
   const StunUInt32Attribute* network_info_attr =
       msg->GetUInt32(STUN_ATTR_GOOG_NETWORK_INFO);
@@ -2292,9 +2354,10 @@ TEST_F(PortTest, TestNetworkInfoAttribute) {
   Connection* rconn =
       rport->CreateConnection(lport->Candidates()[0], Port::ORIGIN_MESSAGE);
   rconn->Ping();
-  ASSERT_THAT(WaitUntil([&] { return rport->last_stun_msg(); }, NotNull(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return rport->last_stun_msg(); }, NotNull(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   msg = rport->last_stun_msg();
   network_info_attr = msg->GetUInt32(STUN_ATTR_GOOG_NETWORK_INFO);
   ASSERT_TRUE(network_info_attr != nullptr);
@@ -2581,16 +2644,18 @@ TEST_F(PortTest,
 
   // Send request.
   lconn->Ping();
-  ASSERT_THAT(WaitUntil([&] { return lport->last_stun_msg(); }, NotNull(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return lport->last_stun_msg(); }, NotNull(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   rconn->OnReadPacket(
       ReceivedIpPacket(lport->last_stun_buf(), SocketAddress(), std::nullopt));
 
   // Intercept request and add comprehension required attribute.
-  ASSERT_THAT(WaitUntil([&] { return rport->last_stun_msg(); }, NotNull(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return rport->last_stun_msg(); }, NotNull(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   auto modified_response = rport->last_stun_msg()->Clone();
   modified_response->AddAttribute(StunAttribute::CreateUInt32(0x7777));
   modified_response->RemoveAttribute(STUN_ATTR_FINGERPRINT);
@@ -2670,23 +2735,25 @@ TEST_F(PortTest, TestHandleStunBindingIndication) {
       rport->CreateConnection(lport->Candidates()[0], Port::ORIGIN_MESSAGE);
   rconn->Ping();
 
-  ASSERT_THAT(WaitUntil([&] { return rport->last_stun_msg(); }, NotNull(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return rport->last_stun_msg(); }, NotNull(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   IceMessage* msg = rport->last_stun_msg();
   EXPECT_EQ(STUN_BINDING_REQUEST, msg->type());
   // Send rport binding request to lport.
   lconn->OnReadPacket(
       ReceivedIpPacket(rport->last_stun_buf(), SocketAddress(), std::nullopt));
 
-  ASSERT_THAT(WaitUntil([&] { return lport->last_stun_msg(); }, NotNull(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return lport->last_stun_msg(); }, NotNull(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   EXPECT_EQ(STUN_BINDING_RESPONSE, lport->last_stun_msg()->type());
   Timestamp last_ping_received1 = lconn->LastPingReceived();
 
   // Adding a delay of 100ms.
-  Thread::Current()->ProcessMessages(100);
+  time_controller_.AdvanceTime(TimeDelta::Millis(100));
   // Pinging lconn using stun indication message.
   lconn->OnReadPacket(ReceivedIpPacket::CreateFromLegacy(
       buf->Data(), buf->Length(), /*packet_time_us=*/-1));
@@ -2801,9 +2868,10 @@ TEST_F(PortTest, TestCandidateFoundation) {
             tcpport2->Candidates()[0].foundation());
   auto stunport = CreateStunPort(kLocalAddr1, nat_socket_factory1());
   stunport->PrepareAddress();
-  ASSERT_THAT(WaitUntil([&] { return stunport->Candidates().size(); }, Eq(1U),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return stunport->Candidates().size(); }, Eq(1U),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   EXPECT_NE(tcpport1->Candidates()[0].foundation(),
             stunport->Candidates()[0].foundation());
   EXPECT_NE(tcpport2->Candidates()[0].foundation(),
@@ -2816,9 +2884,10 @@ TEST_F(PortTest, TestCandidateFoundation) {
   auto turnport1 =
       CreateTurnPort(kLocalAddr1, nat_socket_factory1(), PROTO_UDP, PROTO_UDP);
   turnport1->PrepareAddress();
-  ASSERT_THAT(WaitUntil([&] { return turnport1->Candidates().size(); }, Eq(1U),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return turnport1->Candidates().size(); }, Eq(1U),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   EXPECT_NE(udpport1->Candidates()[0].foundation(),
             turnport1->Candidates()[0].foundation());
   EXPECT_NE(udpport2->Candidates()[0].foundation(),
@@ -2828,9 +2897,10 @@ TEST_F(PortTest, TestCandidateFoundation) {
   auto turnport2 =
       CreateTurnPort(kLocalAddr1, nat_socket_factory1(), PROTO_UDP, PROTO_UDP);
   turnport2->PrepareAddress();
-  ASSERT_THAT(WaitUntil([&] { return turnport2->Candidates().size(); }, Eq(1U),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return turnport2->Candidates().size(); }, Eq(1U),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   EXPECT_EQ(turnport1->Candidates()[0].foundation(),
             turnport2->Candidates()[0].foundation());
 
@@ -2842,9 +2912,10 @@ TEST_F(PortTest, TestCandidateFoundation) {
   auto turnport3 = CreateTurnPort(kLocalAddr1, nat_socket_factory1(), PROTO_UDP,
                                   PROTO_UDP, kTurnUdpIntAddr2);
   turnport3->PrepareAddress();
-  ASSERT_THAT(WaitUntil([&] { return turnport3->Candidates().size(); }, Eq(1U),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return turnport3->Candidates().size(); }, Eq(1U),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   EXPECT_NE(turnport3->Candidates()[0].foundation(),
             turnport2->Candidates()[0].foundation());
 
@@ -2855,9 +2926,10 @@ TEST_F(PortTest, TestCandidateFoundation) {
   auto turnport4 =
       CreateTurnPort(kLocalAddr1, nat_socket_factory1(), PROTO_TCP, PROTO_UDP);
   turnport4->PrepareAddress();
-  ASSERT_THAT(WaitUntil([&] { return turnport4->Candidates().size(); }, Eq(1U),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return turnport4->Candidates().size(); }, Eq(1U),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   EXPECT_NE(turnport2->Candidates()[0].foundation(),
             turnport4->Candidates()[0].foundation());
 }
@@ -2875,9 +2947,10 @@ TEST_F(PortTest, TestCandidateRelatedAddress) {
   // socket address.
   auto stunport = CreateStunPort(kLocalAddr1, nat_socket_factory1());
   stunport->PrepareAddress();
-  ASSERT_THAT(WaitUntil([&] { return stunport->Candidates().size(); }, Eq(1U),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return stunport->Candidates().size(); }, Eq(1U),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   // Check STUN candidate address.
   EXPECT_EQ(stunport->Candidates()[0].address().ipaddr(), kNatAddr1.ipaddr());
   // Check STUN candidate related address.
@@ -2888,9 +2961,10 @@ TEST_F(PortTest, TestCandidateRelatedAddress) {
   auto turnport =
       CreateTurnPort(kLocalAddr1, nat_socket_factory1(), PROTO_UDP, PROTO_UDP);
   turnport->PrepareAddress();
-  ASSERT_THAT(WaitUntil([&] { return turnport->Candidates().size(); }, Eq(1U),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return turnport->Candidates().size(); }, Eq(1U),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   EXPECT_EQ(kTurnUdpExtAddr.ipaddr(),
             turnport->Candidates()[0].address().ipaddr());
   EXPECT_EQ(kNatAddr1.ipaddr(),
@@ -3004,39 +3078,40 @@ TEST_F(PortTest, TestConnectionPriorityWithPriorityAdjustment) {
 // the default setup where the RTT is deterministically one, which generates an
 // estimate given by `MINIMUM_RTT` = 100.
 TEST_F(PortTest, TestWritableState) {
-  ScopedFakeClock clock;
   auto port1 = CreateUdpPort(kLocalAddr1);
   port1->SetIceRole(ICEROLE_CONTROLLING);
   auto port2 = CreateUdpPort(kLocalAddr2);
   port2->SetIceRole(ICEROLE_CONTROLLED);
 
   // Set up channels.
-  TestChannel ch1(std::move(port1));
-  TestChannel ch2(std::move(port2));
+  TestChannel ch1(std::move(port1), time_controller_);
+  TestChannel ch2(std::move(port2), time_controller_);
 
   // Acquire addresses.
   ch1.Start();
   ch2.Start();
-  ASSERT_THAT(WaitUntil([&] { return ch1.complete_count(); }, Eq(1),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout),
-                         .clock = &clock}),
-              IsRtcOk());
-  ASSERT_THAT(WaitUntil([&] { return ch2.complete_count(); }, Eq(1),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout),
-                         .clock = &clock}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return ch1.complete_count(); }, Eq(1),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return ch2.complete_count(); }, Eq(1),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
 
   // Send a ping from src to dst.
   ch1.CreateConnection(GetCandidate(ch2.port()));
   ASSERT_TRUE(ch1.conn() != nullptr);
   EXPECT_EQ(Connection::STATE_WRITE_INIT, ch1.conn()->write_state());
   // for TCP connect
-  EXPECT_THAT(WaitUntil([&] { return ch1.conn()->connected(); }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout),
-                         .clock = &clock}),
-              IsRtcOk());
+  EXPECT_THAT(
+      WaitUntil([&] { return ch1.conn()->connected(); }, IsTrue(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   ch1.Ping();
-  SIMULATED_WAIT(!ch2.remote_address().IsNil(), kShortTimeout, clock);
+  EXPECT_THAT(WaitUntil([&] { return !ch2.remote_address().IsNil(); }, IsTrue(),
+                        {.clock = &time_controller_}),
+              IsRtcOk());
 
   // Data should be sendable before the connection is accepted.
   char data[] = "abcd";
@@ -3047,11 +3122,11 @@ TEST_F(PortTest, TestWritableState) {
   // Accept the connection to return the binding response, transition to
   // writable, and allow data to be sent.
   ch2.AcceptConnection(GetCandidate(ch1.port()));
-  EXPECT_THAT(WaitUntil([&] { return ch1.conn()->write_state(); },
-                        Eq(Connection::STATE_WRITABLE),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout),
-                         .clock = &clock}),
-              IsRtcOk());
+  EXPECT_THAT(
+      WaitUntil([&] { return ch1.conn()->write_state(); },
+                Eq(Connection::STATE_WRITABLE),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   EXPECT_EQ(data_size, ch1.conn()->Send(data, data_size, options));
 
   // Ask the connection to update state as if enough time has passed to lose
@@ -3071,11 +3146,11 @@ TEST_F(PortTest, TestWritableState) {
 
   // And now allow the other side to process the pings and send binding
   // responses.
-  EXPECT_THAT(WaitUntil([&] { return ch1.conn()->write_state(); },
-                        Eq(Connection::STATE_WRITABLE),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout),
-                         .clock = &clock}),
-              IsRtcOk());
+  EXPECT_THAT(
+      WaitUntil([&] { return ch1.conn()->write_state(); },
+                Eq(Connection::STATE_WRITABLE),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   // Wait long enough for a full timeout (past however long we've already
   // waited).
   for (uint32_t i = 1; i <= kConnectionWriteConnectFailures; ++i) {
@@ -3097,42 +3172,43 @@ TEST_F(PortTest, TestWritableState) {
 // the default value given by `kConnectionWriteConnectTimeout` and
 // `kConnectionWriteConnectFailures`.
 TEST_F(PortTest, TestWritableStateWithConfiguredThreshold) {
-  ScopedFakeClock clock;
   auto port1 = CreateUdpPort(kLocalAddr1);
   port1->SetIceRole(ICEROLE_CONTROLLING);
   auto port2 = CreateUdpPort(kLocalAddr2);
   port2->SetIceRole(ICEROLE_CONTROLLED);
 
   // Set up channels.
-  TestChannel ch1(std::move(port1));
-  TestChannel ch2(std::move(port2));
+  TestChannel ch1(std::move(port1), time_controller_);
+  TestChannel ch2(std::move(port2), time_controller_);
 
   // Acquire addresses.
   ch1.Start();
   ch2.Start();
-  ASSERT_THAT(WaitUntil([&] { return ch1.complete_count(); }, Eq(1),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout),
-                         .clock = &clock}),
-              IsRtcOk());
-  ASSERT_THAT(WaitUntil([&] { return ch2.complete_count(); }, Eq(1),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout),
-                         .clock = &clock}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return ch1.complete_count(); }, Eq(1),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return ch2.complete_count(); }, Eq(1),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
 
   // Send a ping from src to dst.
   ch1.CreateConnection(GetCandidate(ch2.port()));
   ASSERT_TRUE(ch1.conn() != nullptr);
   ch1.Ping();
-  SIMULATED_WAIT(!ch2.remote_address().IsNil(), kShortTimeout, clock);
+  EXPECT_THAT(WaitUntil([&] { return !ch2.remote_address().IsNil(); }, IsTrue(),
+                        {.clock = &time_controller_}),
+              IsRtcOk());
 
   // Accept the connection to return the binding response, transition to
   // writable, and allow data to be sent.
   ch2.AcceptConnection(GetCandidate(ch1.port()));
-  EXPECT_THAT(WaitUntil([&] { return ch1.conn()->write_state(); },
-                        Eq(Connection::STATE_WRITABLE),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout),
-                         .clock = &clock}),
-              IsRtcOk());
+  EXPECT_THAT(
+      WaitUntil([&] { return ch1.conn()->write_state(); },
+                Eq(Connection::STATE_WRITABLE),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
 
   ch1.conn()->SetUnwritableTimeout(TimeDelta::Seconds(1));
   ch1.conn()->set_unwritable_min_checks(3);
@@ -3167,8 +3243,8 @@ TEST_F(PortTest, TestTimeoutForNeverWritable) {
   port2->SetIceRole(ICEROLE_CONTROLLED);
 
   // Set up channels.
-  TestChannel ch1(std::move(port1));
-  TestChannel ch2(std::move(port2));
+  TestChannel ch1(std::move(port1), time_controller_);
+  TestChannel ch2(std::move(port2), time_controller_);
 
   // Acquire addresses.
   ch1.Start();
@@ -3180,11 +3256,14 @@ TEST_F(PortTest, TestTimeoutForNeverWritable) {
 
   // Attempt to go directly to write timeout.
   for (uint32_t i = 1; i <= kConnectionWriteConnectFailures; ++i) {
-    ch1.Ping(Timestamp::Millis(i));
+    time_controller_.AdvanceTime(TimeDelta::Millis(1));
+    ch1.Ping();
   }
-  ch1.conn()->UpdateState(Timestamp::Zero() + kConnectionWriteTimeout +
-                          kMaxExpectedSimulatedRtt);
-  EXPECT_EQ(Connection::STATE_WRITE_TIMEOUT, ch1.conn()->write_state());
+  time_controller_.AdvanceTime(kConnectionWriteTimeout);
+  ch1.conn()->UpdateState(env_.clock().CurrentTime());
+  // The connection should be destroyed because no data received before the
+  // write timeout.
+  EXPECT_THAT(ch1.conn(), IsNull());
 }
 
 // This test verifies the connection setup between ICEMODE_FULL
@@ -3199,16 +3278,17 @@ TEST_F(PortTest, TestIceLiteConnectivity) {
   auto ice_lite_port = CreateTestPort(kLocalAddr2, "rfrag", "rpass",
                                       ICEROLE_CONTROLLED, kTiebreaker2);
   // Setup TestChannel. This behaves like FULL mode client.
-  TestChannel ch1(std::move(ice_full_port));
+  TestChannel ch1(std::move(ice_full_port), time_controller_);
   ch1.SetIceMode(ICEMODE_FULL);
 
   // Start gathering candidates.
   ch1.Start();
   ice_lite_port->PrepareAddress();
 
-  ASSERT_THAT(WaitUntil([&] { return ch1.complete_count(); }, Eq(1),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return ch1.complete_count(); }, Eq(1),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   ASSERT_FALSE(ice_lite_port->Candidates().empty());
 
   ch1.CreateConnection(GetCandidate(ice_lite_port.get()));
@@ -3223,7 +3303,7 @@ TEST_F(PortTest, TestIceLiteConnectivity) {
   // from port.
   ASSERT_THAT(
       WaitUntil([&] { return ice_full_port_ptr->last_stun_msg(); }, NotNull(),
-                {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
       IsRtcOk());
   IceMessage* msg = ice_full_port_ptr->last_stun_msg();
   EXPECT_TRUE(msg->GetByteString(STUN_ATTR_USE_CANDIDATE) == nullptr);
@@ -3242,13 +3322,15 @@ TEST_F(PortTest, TestIceLiteConnectivity) {
                                             SocketAddress(), std::nullopt));
 
   // Verifying full mode connection becomes writable from the response.
-  EXPECT_THAT(WaitUntil([&] { return ch1.conn()->write_state(); },
-                        Eq(Connection::STATE_WRITABLE),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
-  EXPECT_THAT(WaitUntil([&] { return ch1.nominated(); }, IsTrue(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  EXPECT_THAT(
+      WaitUntil([&] { return ch1.conn()->write_state(); },
+                Eq(Connection::STATE_WRITABLE),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
+  EXPECT_THAT(
+      WaitUntil([&] { return ch1.nominated(); }, IsTrue(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
 
   // Clear existing stun messsages. Otherwise we will process old stun
   // message right after we send ping.
@@ -3257,7 +3339,7 @@ TEST_F(PortTest, TestIceLiteConnectivity) {
   ch1.Ping();
   ASSERT_THAT(
       WaitUntil([&] { return ice_full_port_ptr->last_stun_msg(); }, NotNull(),
-                {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
       IsRtcOk());
   msg = ice_full_port_ptr->last_stun_msg();
   EXPECT_TRUE(msg->GetByteString(STUN_ATTR_USE_CANDIDATE) != nullptr);
@@ -3321,7 +3403,7 @@ TEST_P(GoogPingTest, TestGoogPingAnnounceEnable) {
   auto port2 = CreateTestPort(kLocalAddr2, "rfrag", "rpass", ICEROLE_CONTROLLED,
                               kTiebreaker2);
 
-  TestChannel ch1(std::move(port1_unique));
+  TestChannel ch1(std::move(port1_unique), time_controller_);
   // Block usage of STUN_ATTR_USE_CANDIDATE so that
   // ch1.conn() will sent GOOG_PING_REQUEST directly.
   // This only makes test a bit shorter...
@@ -3330,9 +3412,10 @@ TEST_P(GoogPingTest, TestGoogPingAnnounceEnable) {
   ch1.Start();
   port2->PrepareAddress();
 
-  ASSERT_THAT(WaitUntil([&] { return ch1.complete_count(); }, Eq(1),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return ch1.complete_count(); }, Eq(1),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   ASSERT_FALSE(port2->Candidates().empty());
 
   ch1.CreateConnection(GetCandidate(port2.get()));
@@ -3343,9 +3426,10 @@ TEST_P(GoogPingTest, TestGoogPingAnnounceEnable) {
   // Send ping.
   ch1.Ping();
 
-  ASSERT_THAT(WaitUntil([&] { return port1->last_stun_msg(); }, NotNull(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return port1->last_stun_msg(); }, NotNull(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   const IceMessage* request1 = port1->last_stun_msg();
 
   ASSERT_EQ(trials.enable_goog_ping,
@@ -3373,9 +3457,10 @@ TEST_P(GoogPingTest, TestGoogPingAnnounceEnable) {
   port2->Reset();
 
   ch1.Ping();
-  ASSERT_THAT(WaitUntil([&] { return port1->last_stun_msg(); }, NotNull(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return port1->last_stun_msg(); }, NotNull(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   const IceMessage* request2 = port1->last_stun_msg();
 
   // It should be a GOOG_PING if both of these are TRUE
@@ -3417,7 +3502,7 @@ TEST_F(PortTest, TestGoogPingUnsupportedVersionInStunBinding) {
   auto port2 = CreateTestPort(kLocalAddr2, "rfrag", "rpass", ICEROLE_CONTROLLED,
                               kTiebreaker2);
 
-  TestChannel ch1(std::move(port1_unique));
+  TestChannel ch1(std::move(port1_unique), time_controller_);
   // Block usage of STUN_ATTR_USE_CANDIDATE so that
   // ch1.conn() will sent GOOG_PING_REQUEST directly.
   // This only makes test a bit shorter...
@@ -3426,9 +3511,10 @@ TEST_F(PortTest, TestGoogPingUnsupportedVersionInStunBinding) {
   ch1.Start();
   port2->PrepareAddress();
 
-  ASSERT_THAT(WaitUntil([&] { return ch1.complete_count(); }, Eq(1),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return ch1.complete_count(); }, Eq(1),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   ASSERT_FALSE(port2->Candidates().empty());
 
   ch1.CreateConnection(GetCandidate(port2.get()));
@@ -3439,9 +3525,10 @@ TEST_F(PortTest, TestGoogPingUnsupportedVersionInStunBinding) {
   // Send ping.
   ch1.Ping();
 
-  ASSERT_THAT(WaitUntil([&] { return port1->last_stun_msg(); }, NotNull(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return port1->last_stun_msg(); }, NotNull(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   const IceMessage* request1 = port1->last_stun_msg();
 
   ASSERT_TRUE(GetSupportedGoogPingVersion(request1) &&
@@ -3490,7 +3577,7 @@ TEST_F(PortTest, TestGoogPingUnsupportedVersionInStunBindingResponse) {
   auto port2 = CreateTestPort(kLocalAddr2, "rfrag", "rpass", ICEROLE_CONTROLLED,
                               kTiebreaker2);
 
-  TestChannel ch1(std::move(port1_unique));
+  TestChannel ch1(std::move(port1_unique), time_controller_);
   // Block usage of STUN_ATTR_USE_CANDIDATE so that
   // ch1.conn() will sent GOOG_PING_REQUEST directly.
   // This only makes test a bit shorter...
@@ -3499,9 +3586,10 @@ TEST_F(PortTest, TestGoogPingUnsupportedVersionInStunBindingResponse) {
   ch1.Start();
   port2->PrepareAddress();
 
-  ASSERT_THAT(WaitUntil([&] { return ch1.complete_count(); }, Eq(1),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return ch1.complete_count(); }, Eq(1),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   ASSERT_FALSE(port2->Candidates().empty());
 
   ch1.CreateConnection(GetCandidate(port2.get()));
@@ -3512,9 +3600,10 @@ TEST_F(PortTest, TestGoogPingUnsupportedVersionInStunBindingResponse) {
   // Send ping.
   ch1.Ping();
 
-  ASSERT_THAT(WaitUntil([&] { return port1->last_stun_msg(); }, NotNull(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return port1->last_stun_msg(); }, NotNull(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   const IceMessage* request1 = port1->last_stun_msg();
 
   ASSERT_TRUE(GetSupportedGoogPingVersion(request1) &&
@@ -3563,9 +3652,10 @@ TEST_F(PortTest, TestGoogPingUnsupportedVersionInStunBindingResponse) {
   port2->Reset();
 
   ch1.Ping();
-  ASSERT_THAT(WaitUntil([&] { return port1->last_stun_msg(); }, NotNull(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return port1->last_stun_msg(); }, NotNull(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
 
   // This should now be a STUN_BINDING...without a kGoogPingVersion
   const IceMessage* request2 = port1->last_stun_msg();
@@ -3595,7 +3685,7 @@ TEST_F(PortTest, TestChangeInAttributeMakesGoogPingFallsbackToStunBinding) {
   auto port2 = CreateTestPort(kLocalAddr2, "rfrag", "rpass", ICEROLE_CONTROLLED,
                               kTiebreaker2);
 
-  TestChannel ch1(std::move(port1_unique));
+  TestChannel ch1(std::move(port1_unique), time_controller_);
   // Block usage of STUN_ATTR_USE_CANDIDATE so that
   // ch1.conn() will sent GOOG_PING_REQUEST directly.
   // This only makes test a bit shorter...
@@ -3604,9 +3694,10 @@ TEST_F(PortTest, TestChangeInAttributeMakesGoogPingFallsbackToStunBinding) {
   ch1.Start();
   port2->PrepareAddress();
 
-  ASSERT_THAT(WaitUntil([&] { return ch1.complete_count(); }, Eq(1),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return ch1.complete_count(); }, Eq(1),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   ASSERT_FALSE(port2->Candidates().empty());
 
   ch1.CreateConnection(GetCandidate(port2.get()));
@@ -3617,9 +3708,10 @@ TEST_F(PortTest, TestChangeInAttributeMakesGoogPingFallsbackToStunBinding) {
   // Send ping.
   ch1.Ping();
 
-  ASSERT_THAT(WaitUntil([&] { return port1->last_stun_msg(); }, NotNull(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return port1->last_stun_msg(); }, NotNull(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   const IceMessage* msg = port1->last_stun_msg();
   auto* con =
       port2->CreateConnection(port1->Candidates()[0], Port::ORIGIN_MESSAGE);
@@ -3641,9 +3733,10 @@ TEST_F(PortTest, TestChangeInAttributeMakesGoogPingFallsbackToStunBinding) {
   port2->Reset();
 
   ch1.Ping();
-  ASSERT_THAT(WaitUntil([&] { return port1->last_stun_msg(); }, NotNull(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return port1->last_stun_msg(); }, NotNull(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   const IceMessage* msg2 = port1->last_stun_msg();
 
   // It should be a GOOG_PING if both of these are TRUE
@@ -3664,9 +3757,10 @@ TEST_F(PortTest, TestChangeInAttributeMakesGoogPingFallsbackToStunBinding) {
   ch1.conn()->set_use_candidate_attr(!ch1.conn()->use_candidate_attr());
 
   ch1.Ping();
-  ASSERT_THAT(WaitUntil([&] { return port1->last_stun_msg(); }, NotNull(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return port1->last_stun_msg(); }, NotNull(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   const IceMessage* msg3 = port1->last_stun_msg();
 
   // It should be a STUN_BINDING_REQUEST
@@ -3687,7 +3781,7 @@ TEST_F(PortTest, TestErrorResponseMakesGoogPingFallBackToStunBinding) {
   auto port2 = CreateTestPort(kLocalAddr2, "rfrag", "rpass", ICEROLE_CONTROLLED,
                               kTiebreaker2);
 
-  TestChannel ch1(std::move(port1_unique));
+  TestChannel ch1(std::move(port1_unique), time_controller_);
   // Block usage of STUN_ATTR_USE_CANDIDATE so that
   // ch1.conn() will sent GOOG_PING_REQUEST directly.
   // This only makes test a bit shorter...
@@ -3696,9 +3790,10 @@ TEST_F(PortTest, TestErrorResponseMakesGoogPingFallBackToStunBinding) {
   ch1.Start();
   port2->PrepareAddress();
 
-  ASSERT_THAT(WaitUntil([&] { return ch1.complete_count(); }, Eq(1),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return ch1.complete_count(); }, Eq(1),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   ASSERT_FALSE(port2->Candidates().empty());
 
   ch1.CreateConnection(GetCandidate(port2.get()));
@@ -3709,9 +3804,10 @@ TEST_F(PortTest, TestErrorResponseMakesGoogPingFallBackToStunBinding) {
   // Send ping.
   ch1.Ping();
 
-  ASSERT_THAT(WaitUntil([&] { return port1->last_stun_msg(); }, NotNull(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return port1->last_stun_msg(); }, NotNull(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   const IceMessage* msg = port1->last_stun_msg();
   auto* con =
       port2->CreateConnection(port1->Candidates()[0], Port::ORIGIN_MESSAGE);
@@ -3733,9 +3829,10 @@ TEST_F(PortTest, TestErrorResponseMakesGoogPingFallBackToStunBinding) {
   port2->Reset();
 
   ch1.Ping();
-  ASSERT_THAT(WaitUntil([&] { return port1->last_stun_msg(); }, NotNull(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return port1->last_stun_msg(); }, NotNull(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   const IceMessage* msg2 = port1->last_stun_msg();
 
   // It should be a GOOG_PING.
@@ -3763,9 +3860,10 @@ TEST_F(PortTest, TestErrorResponseMakesGoogPingFallBackToStunBinding) {
   port2->Reset();
 
   ch1.Ping();
-  ASSERT_THAT(WaitUntil([&] { return port1->last_stun_msg(); }, NotNull(),
-                        {.timeout = TimeDelta::Millis(kDefaultTimeout)}),
-              IsRtcOk());
+  ASSERT_THAT(
+      WaitUntil([&] { return port1->last_stun_msg(); }, NotNull(),
+                {.timeout = kDefaultTimeout, .clock = &time_controller_}),
+      IsRtcOk());
   const IceMessage* msg3 = port1->last_stun_msg();
 
   // It should be a STUN_BINDING_REQUEST
@@ -3778,7 +3876,6 @@ TEST_F(PortTest, TestErrorResponseMakesGoogPingFallBackToStunBinding) {
 // port will time out after connectivity is lost, if they are not marked as
 // "keep alive until pruned."
 TEST_F(PortTest, TestPortTimeoutIfNotKeptAlive) {
-  ScopedFakeClock clock;
   int timeout_delay = 100;
   auto port1 = CreateUdpPort(kLocalAddr1);
   ConnectToSignalDestroyed(port1.get());
@@ -3793,23 +3890,22 @@ TEST_F(PortTest, TestPortTimeoutIfNotKeptAlive) {
   port2->SetIceTiebreaker(kTiebreaker2);
 
   // Set up channels and ensure both ports will be deleted.
-  TestChannel ch1(std::move(port1));
-  TestChannel ch2(std::move(port2));
+  TestChannel ch1(std::move(port1), time_controller_);
+  TestChannel ch2(std::move(port2), time_controller_);
 
   // Simulate a connection that succeeds, and then is destroyed.
   StartConnectAndStopChannels(&ch1, &ch2);
   // After the connection is destroyed, the port will be destroyed because
   // none of them is marked as "keep alive until pruned.
-  EXPECT_THAT(
-      WaitUntil([&] { return ports_destroyed(); }, Eq(2), {.clock = &clock}),
-      IsRtcOk());
+  EXPECT_THAT(WaitUntil([&] { return ports_destroyed(); }, Eq(2),
+                        {.clock = &time_controller_}),
+              IsRtcOk());
 }
 
 // Test that if after all connection are destroyed, new connections are created
 // and destroyed again, ports won't be destroyed until a timeout period passes
 // after the last set of connections are all destroyed.
 TEST_F(PortTest, TestPortTimeoutAfterNewConnectionCreatedAndDestroyed) {
-  ScopedFakeClock clock;
   int timeout_delay = 100;
   auto port1 = CreateUdpPort(kLocalAddr1);
   ConnectToSignalDestroyed(port1.get());
@@ -3825,12 +3921,12 @@ TEST_F(PortTest, TestPortTimeoutAfterNewConnectionCreatedAndDestroyed) {
   port2->SetIceTiebreaker(kTiebreaker2);
 
   // Set up channels and ensure both ports will be deleted.
-  TestChannel ch1(std::move(port1));
-  TestChannel ch2(std::move(port2));
+  TestChannel ch1(std::move(port1), time_controller_);
+  TestChannel ch2(std::move(port2), time_controller_);
 
   // Simulate a connection that succeeds, and then is destroyed.
   StartConnectAndStopChannels(&ch1, &ch2);
-  SIMULATED_WAIT(ports_destroyed() > 0, 80, clock);
+  time_controller_.AdvanceTime(TimeDelta::Millis(80));
   EXPECT_EQ(0, ports_destroyed());
 
   // Start the second set of connection and destroy them.
@@ -3839,20 +3935,19 @@ TEST_F(PortTest, TestPortTimeoutAfterNewConnectionCreatedAndDestroyed) {
   ch1.Stop();
   ch2.Stop();
 
-  SIMULATED_WAIT(ports_destroyed() > 0, 80, clock);
+  time_controller_.AdvanceTime(TimeDelta::Millis(80));
   EXPECT_EQ(0, ports_destroyed());
 
   // The ports on both sides should be destroyed after timeout.
-  EXPECT_THAT(
-      WaitUntil([&] { return ports_destroyed(); }, Eq(2), {.clock = &clock}),
-      IsRtcOk());
+  EXPECT_THAT(WaitUntil([&] { return ports_destroyed(); }, Eq(2),
+                        {.clock = &time_controller_}),
+              IsRtcOk());
 }
 
 // This test case verifies that neither the controlling port nor the controlled
 // port will time out after connectivity is lost if they are marked as "keep
 // alive until pruned". They will time out after they are pruned.
 TEST_F(PortTest, TestPortNotTimeoutUntilPruned) {
-  ScopedFakeClock clock;
   int timeout_delay = 100;
   auto port1 = CreateUdpPort(kLocalAddr1);
   ConnectToSignalDestroyed(port1.get());
@@ -3872,23 +3967,23 @@ TEST_F(PortTest, TestPortNotTimeoutUntilPruned) {
   port2->set_component(ICE_CANDIDATE_COMPONENT_DEFAULT);
 
   // Set up channels and keep the port alive.
-  TestChannel ch1(std::move(port1));
-  TestChannel ch2(std::move(port2));
+  TestChannel ch1(std::move(port1), time_controller_);
+  TestChannel ch2(std::move(port2), time_controller_);
   // Simulate a connection that succeeds, and then is destroyed. But ports
   // are kept alive. Ports won't be destroyed.
   StartConnectAndStopChannels(&ch1, &ch2);
   ch1.port()->KeepAliveUntilPruned();
   ch2.port()->KeepAliveUntilPruned();
-  SIMULATED_WAIT(ports_destroyed() > 0, 150, clock);
+  time_controller_.AdvanceTime(TimeDelta::Millis(150));
   EXPECT_EQ(0, ports_destroyed());
 
   // If they are pruned now, they will be destroyed right away.
   ch1.port()->Prune();
   ch2.port()->Prune();
   // The ports on both sides should be destroyed after timeout.
-  EXPECT_THAT(
-      WaitUntil([&] { return ports_destroyed(); }, Eq(2), {.clock = &clock}),
-      IsRtcOk());
+  EXPECT_THAT(WaitUntil([&] { return ports_destroyed(); }, Eq(2),
+                        {.clock = &time_controller_}),
+              IsRtcOk());
 }
 
 TEST_F(PortTest, TestSupportsProtocol) {
@@ -3951,7 +4046,7 @@ TEST_F(PortTest, TestAddConnectionWithSameAddress) {
   EXPECT_EQ(2u, conn_in_use->remote_candidate().generation());
 
   // Make sure the new connection was not deleted.
-  Thread::Current()->ProcessMessages(300);
+  time_controller_.AdvanceTime(TimeDelta::Millis(300));
   EXPECT_TRUE(port->GetConnection(address) != nullptr);
 }
 

From b188c9f2d9fbc385418d08100a44d69891317684 Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Thu, 26 Mar 2026 21:10:03 -0700
Subject: [PATCH 419/472] Update WebRTC code version (2026-03-27T04:10:01).

Bug: None
Change-Id: I0c4ef4bb7336674109e9c6475ccf3e8b3aea5b87
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/460642
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47272}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index 8f1b8fd42b..0a885eda5b 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-25T04:04:33";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-27T04:10:01";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From b466437c7e4ebf0d57d6525a711c5426e7d09780 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Thu, 26 Mar 2026 23:33:14 +0100
Subject: [PATCH 420/472] Use first_ssrc() instead of ssrcs[0] in channel.cc

This is just a btw change that didn't fit with some other CLs, but a
thing will occasionally distract while not exactly fitting in with other
changes. :)

Bug: none
Change-Id: Ib15aee691478da8d4b7669009fce963165b7925e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/460580
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47273}
---
 pc/channel.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/pc/channel.cc b/pc/channel.cc
index 8925882e17..5761f2aafe 100644
--- a/pc/channel.cc
+++ b/pc/channel.cc
@@ -827,7 +827,7 @@ RTCError BaseChannel::UpdateLocalStreams_w(
     }
 
     if (media_send_channel()->AddSendStream(new_stream)) {
-      RTC_LOG(LS_INFO) << "Add send stream ssrc: " << new_stream.ssrcs[0]
+      RTC_LOG(LS_INFO) << "Add send stream ssrc: " << new_stream.first_ssrc()
                        << " into " << ToString();
     } else {
       return RTCError::InvalidParameter()

From 47601ff208f695fb04ba146fb1d854895b8f0058 Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Thu, 26 Mar 2026 16:20:25 +0100
Subject: [PATCH 421/472] Replace ArrayView with std::span in common_video/

ArrayView is an alias to std::span. This change switch to use
std::span directly instead of through the alias.

Search&Replace MakeArrayView and ArrayView with std::span
Search&Replace include "api/array_view.h" with include <span>
Remove <span> include where std::span is not mentioned in the file
Remove build dependencies on array_view target

Bug: webrtc:439801349
Change-Id: Iabcb3e93425e1c0189e866ac3b249078fed833a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/460440
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47274}
---
 common_video/BUILD.gn                              |  2 --
 common_video/h264/h264_bitstream_parser.cc         |  8 ++++----
 common_video/h264/h264_bitstream_parser.h          |  8 ++++----
 common_video/h264/h264_common.cc                   |  8 ++++----
 common_video/h264/h264_common.h                    | 12 ++++++------
 common_video/h264/pps_parser.cc                    | 10 +++++-----
 common_video/h264/pps_parser.h                     | 13 ++++++-------
 common_video/h264/pps_parser_unittest.cc           |  6 +++---
 common_video/h264/sps_parser.cc                    |  4 ++--
 common_video/h264/sps_parser.h                     |  4 ++--
 common_video/h264/sps_parser_unittest.cc           |  4 ++--
 common_video/h264/sps_vui_rewriter.cc              | 12 ++++++------
 common_video/h264/sps_vui_rewriter.h               |  8 ++++----
 common_video/h264/sps_vui_rewriter_unittest.cc     |  4 ++--
 common_video/h265/h265_bitstream_parser.cc         | 12 ++++++------
 common_video/h265/h265_bitstream_parser.h          | 12 ++++++------
 .../h265/h265_bitstream_parser_unittest.cc         |  1 -
 common_video/h265/h265_common.cc                   |  8 ++++----
 common_video/h265/h265_common.h                    | 14 +++++++-------
 common_video/h265/h265_pps_parser.cc               |  8 ++++----
 common_video/h265/h265_pps_parser.h                | 12 ++++++------
 common_video/h265/h265_pps_parser_unittest.cc      |  4 ++--
 common_video/h265/h265_sps_parser.cc               |  6 +++---
 common_video/h265/h265_sps_parser.h                |  8 ++++----
 common_video/h265/h265_sps_parser_unittest.cc      |  4 ++--
 common_video/h265/h265_vps_parser.cc               |  6 +++---
 common_video/h265/h265_vps_parser.h                |  8 ++++----
 27 files changed, 101 insertions(+), 105 deletions(-)

diff --git a/common_video/BUILD.gn b/common_video/BUILD.gn
index 527c1ee4ae..52753212af 100644
--- a/common_video/BUILD.gn
+++ b/common_video/BUILD.gn
@@ -64,7 +64,6 @@ rtc_library("common_video") {
   }
 
   deps = [
-    "../api:array_view",
     "../api:make_ref_counted",
     "../api:scoped_refptr",
     "../api/units:time_delta",
@@ -144,7 +143,6 @@ if (rtc_include_tests && !build_with_chromium) {
 
     deps = [
       ":common_video",
-      "../api:array_view",
       "../api:scoped_refptr",
       "../api/units:time_delta",
       "../api/units:timestamp",
diff --git a/common_video/h264/h264_bitstream_parser.cc b/common_video/h264/h264_bitstream_parser.cc
index 1a66e5a88a..145e4d2d8d 100644
--- a/common_video/h264/h264_bitstream_parser.cc
+++ b/common_video/h264/h264_bitstream_parser.cc
@@ -12,9 +12,9 @@
 #include <cstdint>
 #include <cstdlib>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "common_video/h264/h264_common.h"
 #include "common_video/h264/pps_parser.h"
 #include "common_video/h264/sps_parser.h"
@@ -34,7 +34,7 @@ H264BitstreamParser::H264BitstreamParser() = default;
 H264BitstreamParser::~H264BitstreamParser() = default;
 
 H264BitstreamParser::Result H264BitstreamParser::ParseNonParameterSetNalu(
-    ArrayView<const uint8_t> source,
+    std::span<const uint8_t> source,
     uint8_t nalu_type) {
   if (!sps_ || !pps_)
     return kInvalidStream;
@@ -311,7 +311,7 @@ H264BitstreamParser::Result H264BitstreamParser::ParseNonParameterSetNalu(
   return kOk;
 }
 
-void H264BitstreamParser::ParseSlice(ArrayView<const uint8_t> slice) {
+void H264BitstreamParser::ParseSlice(std::span<const uint8_t> slice) {
   if (slice.empty()) {
     return;
   }
@@ -343,7 +343,7 @@ void H264BitstreamParser::ParseSlice(ArrayView<const uint8_t> slice) {
   }
 }
 
-void H264BitstreamParser::ParseBitstream(ArrayView<const uint8_t> bitstream) {
+void H264BitstreamParser::ParseBitstream(std::span<const uint8_t> bitstream) {
   std::vector<H264::NaluIndex> nalu_indices = H264::FindNaluIndices(bitstream);
   for (const H264::NaluIndex& index : nalu_indices)
     ParseSlice(
diff --git a/common_video/h264/h264_bitstream_parser.h b/common_video/h264/h264_bitstream_parser.h
index 2dafa86185..810b91cbdd 100644
--- a/common_video/h264/h264_bitstream_parser.h
+++ b/common_video/h264/h264_bitstream_parser.h
@@ -14,8 +14,8 @@
 #include <stdint.h>
 
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/video_codecs/bitstream_parser.h"
 #include "common_video/h264/pps_parser.h"
 #include "common_video/h264/sps_parser.h"
@@ -33,7 +33,7 @@ class H264BitstreamParser : public BitstreamParser {
   H264BitstreamParser();
   ~H264BitstreamParser() override;
 
-  void ParseBitstream(ArrayView<const uint8_t> bitstream) override;
+  void ParseBitstream(std::span<const uint8_t> bitstream) override;
   std::optional<int> GetLastSliceQp() const override;
 
  protected:
@@ -42,8 +42,8 @@ class H264BitstreamParser : public BitstreamParser {
     kInvalidStream,
     kUnsupportedStream,
   };
-  void ParseSlice(ArrayView<const uint8_t> slice);
-  Result ParseNonParameterSetNalu(ArrayView<const uint8_t> source,
+  void ParseSlice(std::span<const uint8_t> slice);
+  Result ParseNonParameterSetNalu(std::span<const uint8_t> source,
                                   uint8_t nalu_type);
 
   // SPS/PPS state, updated when parsing new SPS/PPS, used to parse slices.
diff --git a/common_video/h264/h264_common.cc b/common_video/h264/h264_common.cc
index 4ab704c4eb..387d360d4a 100644
--- a/common_video/h264/h264_common.cc
+++ b/common_video/h264/h264_common.cc
@@ -12,9 +12,9 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "rtc_base/buffer.h"
 
 namespace webrtc {
@@ -22,7 +22,7 @@ namespace H264 {
 
 const uint8_t kNaluTypeMask = 0x1F;
 
-std::vector<NaluIndex> FindNaluIndices(ArrayView<const uint8_t> buffer) {
+std::vector<NaluIndex> FindNaluIndices(std::span<const uint8_t> buffer) {
   // This is sorta like Boyer-Moore, but with only the first optimization step:
   // given a 3-byte sequence we're looking at, if the 3rd byte isn't 1 or 0,
   // skip ahead to the next 3-byte sequence. 0s and 1s are relatively rare, so
@@ -72,7 +72,7 @@ NaluType ParseNaluType(uint8_t data) {
   return static_cast<NaluType>(data & kNaluTypeMask);
 }
 
-std::vector<uint8_t> ParseRbsp(ArrayView<const uint8_t> data) {
+std::vector<uint8_t> ParseRbsp(std::span<const uint8_t> data) {
   std::vector<uint8_t> out;
   out.reserve(data.size());
 
@@ -95,7 +95,7 @@ std::vector<uint8_t> ParseRbsp(ArrayView<const uint8_t> data) {
   return out;
 }
 
-void WriteRbsp(ArrayView<const uint8_t> bytes, Buffer* destination) {
+void WriteRbsp(std::span<const uint8_t> bytes, Buffer* destination) {
   static const uint8_t kZerosInStartSequence = 2;
   static const uint8_t kEmulationByte = 0x03u;
   size_t num_consecutive_zeros = 0;
diff --git a/common_video/h264/h264_common.h b/common_video/h264/h264_common.h
index 68c9504107..932fb54604 100644
--- a/common_video/h264/h264_common.h
+++ b/common_video/h264/h264_common.h
@@ -14,9 +14,9 @@
 #include <stddef.h>
 #include <stdint.h>
 
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/system/rtc_export.h"
 
@@ -65,7 +65,7 @@ struct NaluIndex {
 
 // Returns a vector of the NALU indices in the given buffer.
 RTC_EXPORT std::vector<NaluIndex> FindNaluIndices(
-    ArrayView<const uint8_t> buffer);
+    std::span<const uint8_t> buffer);
 
 // Get the NAL type from the header byte immediately following start sequence.
 RTC_EXPORT NaluType ParseNaluType(uint8_t data);
@@ -84,23 +84,23 @@ RTC_EXPORT NaluType ParseNaluType(uint8_t data);
 // the 03 emulation byte.
 
 // Parse the given data and remove any emulation byte escaping.
-std::vector<uint8_t> ParseRbsp(ArrayView<const uint8_t> data);
+std::vector<uint8_t> ParseRbsp(std::span<const uint8_t> data);
 
 // TODO: bugs.webrtc.org/42225170 - Deprecate.
 inline std::vector<uint8_t> ParseRbsp(const uint8_t* data, size_t length) {
-  return ParseRbsp(MakeArrayView(data, length));
+  return ParseRbsp(std::span(data, length));
 }
 
 // Write the given data to the destination buffer, inserting and emulation
 // bytes in order to escape any data the could be interpreted as a start
 // sequence.
-void WriteRbsp(ArrayView<const uint8_t> bytes, Buffer* destination);
+void WriteRbsp(std::span<const uint8_t> bytes, Buffer* destination);
 
 // TODO: bugs.webrtc.org/42225170 - Deprecate.
 inline void WriteRbsp(const uint8_t* bytes,
                       size_t length,
                       Buffer* destination) {
-  WriteRbsp(MakeArrayView(bytes, length), destination);
+  WriteRbsp(std::span(bytes, length), destination);
 }
 }  // namespace H264
 }  // namespace webrtc
diff --git a/common_video/h264/pps_parser.cc b/common_video/h264/pps_parser.cc
index dc36ba1d10..2ecfaa55a1 100644
--- a/common_video/h264/pps_parser.cc
+++ b/common_video/h264/pps_parser.cc
@@ -13,10 +13,10 @@
 #include <cstdint>
 #include <limits>
 #include <optional>
+#include <span>
 #include <vector>
 
 #include "absl/numeric/bits.h"
-#include "api/array_view.h"
 #include "common_video/h264/h264_common.h"
 #include "rtc_base/bitstream_reader.h"
 #include "rtc_base/checks.h"
@@ -32,14 +32,14 @@ constexpr int kMinPicInitQpDeltaValue = -26;
 // http://www.itu.int/rec/T-REC-H.264
 
 std::optional<PpsParser::PpsState> PpsParser::ParsePps(
-    ArrayView<const uint8_t> data) {
+    std::span<const uint8_t> data) {
   // First, parse out rbsp, which is basically the source buffer minus emulation
   // bytes (the last byte of a 0x00 0x00 0x03 sequence). RBSP is defined in
   // section 7.3.1 of the H.264 standard.
   return ParseInternal(H264::ParseRbsp(data));
 }
 
-bool PpsParser::ParsePpsIds(ArrayView<const uint8_t> data,
+bool PpsParser::ParsePpsIds(std::span<const uint8_t> data,
                             uint32_t* pps_id,
                             uint32_t* sps_id) {
   RTC_DCHECK(pps_id);
@@ -55,7 +55,7 @@ bool PpsParser::ParsePpsIds(ArrayView<const uint8_t> data,
 }
 
 std::optional<PpsParser::SliceHeader> PpsParser::ParseSliceHeader(
-    ArrayView<const uint8_t> data) {
+    std::span<const uint8_t> data) {
   std::vector<uint8_t> unpacked_buffer = H264::ParseRbsp(data);
   BitstreamReader slice_reader(unpacked_buffer);
   PpsParser::SliceHeader slice_header;
@@ -76,7 +76,7 @@ std::optional<PpsParser::SliceHeader> PpsParser::ParseSliceHeader(
 }
 
 std::optional<PpsParser::PpsState> PpsParser::ParseInternal(
-    ArrayView<const uint8_t> buffer) {
+    std::span<const uint8_t> buffer) {
   BitstreamReader reader(buffer);
   PpsState pps;
   pps.id = reader.ReadExponentialGolomb();
diff --git a/common_video/h264/pps_parser.h b/common_video/h264/pps_parser.h
index cdf3fb79a0..ef3ba21859 100644
--- a/common_video/h264/pps_parser.h
+++ b/common_video/h264/pps_parser.h
@@ -15,8 +15,7 @@
 #include <stdint.h>
 
 #include <optional>
-
-#include "api/array_view.h"
+#include <span>
 
 namespace webrtc {
 
@@ -48,24 +47,24 @@ class PpsParser {
   };
 
   // Unpack RBSP and parse PPS state from the supplied buffer.
-  static std::optional<PpsState> ParsePps(ArrayView<const uint8_t> data);
+  static std::optional<PpsState> ParsePps(std::span<const uint8_t> data);
   // TODO: bugs.webrtc.org/42225170 - Deprecate.
   static inline std::optional<PpsState> ParsePps(const uint8_t* data,
                                                  size_t length) {
-    return ParsePps(MakeArrayView(data, length));
+    return ParsePps(std::span(data, length));
   }
 
-  static bool ParsePpsIds(ArrayView<const uint8_t> data,
+  static bool ParsePpsIds(std::span<const uint8_t> data,
                           uint32_t* pps_id,
                           uint32_t* sps_id);
 
   static std::optional<SliceHeader> ParseSliceHeader(
-      ArrayView<const uint8_t> data);
+      std::span<const uint8_t> data);
 
  protected:
   // Parse the PPS state, for a buffer where RBSP decoding has already been
   // performed.
-  static std::optional<PpsState> ParseInternal(ArrayView<const uint8_t> buffer);
+  static std::optional<PpsState> ParseInternal(std::span<const uint8_t> buffer);
 };
 
 }  // namespace webrtc
diff --git a/common_video/h264/pps_parser_unittest.cc b/common_video/h264/pps_parser_unittest.cc
index eb2891c8cb..8bf1117bf5 100644
--- a/common_video/h264/pps_parser_unittest.cc
+++ b/common_video/h264/pps_parser_unittest.cc
@@ -13,9 +13,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "common_video/h264/h264_common.h"
 #include "rtc_base/bit_buffer.h"
 #include "rtc_base/buffer.h"
@@ -138,7 +138,7 @@ void WritePps(const PpsParser::PpsState& pps,
     bit_buffer.GetCurrentOffset(&byte_offset, &bit_offset);
   }
 
-  H264::WriteRbsp(MakeArrayView(data, byte_offset), out_buffer);
+  H264::WriteRbsp(std::span(data, byte_offset), out_buffer);
 }
 
 class PpsParserTest : public ::testing::Test {
@@ -223,7 +223,7 @@ TEST_F(PpsParserTest, MaxPps) {
 }
 
 TEST_F(PpsParserTest, ParseSliceHeader) {
-  ArrayView<const uint8_t> chunk(kH264BitstreamChunk);
+  std::span<const uint8_t> chunk(kH264BitstreamChunk);
   std::vector<H264::NaluIndex> nalu_indices = H264::FindNaluIndices(chunk);
   EXPECT_EQ(nalu_indices.size(), 3ull);
   for (const auto& index : nalu_indices) {
diff --git a/common_video/h264/sps_parser.cc b/common_video/h264/sps_parser.cc
index 2ee431c7e5..77945fe308 100644
--- a/common_video/h264/sps_parser.cc
+++ b/common_video/h264/sps_parser.cc
@@ -13,9 +13,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "common_video/h264/h264_common.h"
 #include "rtc_base/bitstream_reader.h"
 
@@ -36,7 +36,7 @@ SpsParser::SpsState::~SpsState() = default;
 
 // Unpack RBSP and parse SPS state from the supplied buffer.
 std::optional<SpsParser::SpsState> SpsParser::ParseSps(
-    ArrayView<const uint8_t> data) {
+    std::span<const uint8_t> data) {
   std::vector<uint8_t> unpacked_buffer = H264::ParseRbsp(data);
   BitstreamReader reader(unpacked_buffer);
   return ParseSpsUpToVui(reader);
diff --git a/common_video/h264/sps_parser.h b/common_video/h264/sps_parser.h
index b24194c6e9..3091703a21 100644
--- a/common_video/h264/sps_parser.h
+++ b/common_video/h264/sps_parser.h
@@ -13,8 +13,8 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "rtc_base/bitstream_reader.h"
 #include "rtc_base/system/rtc_export.h"
 
@@ -45,7 +45,7 @@ class RTC_EXPORT SpsParser {
   };
 
   // Unpack RBSP and parse SPS state from the supplied buffer.
-  static std::optional<SpsState> ParseSps(ArrayView<const uint8_t> data);
+  static std::optional<SpsState> ParseSps(std::span<const uint8_t> data);
 
  protected:
   // Parse the SPS state, up till the VUI part, for a buffer where RBSP
diff --git a/common_video/h264/sps_parser_unittest.cc b/common_video/h264/sps_parser_unittest.cc
index d2396d016a..6ab2152725 100644
--- a/common_video/h264/sps_parser_unittest.cc
+++ b/common_video/h264/sps_parser_unittest.cc
@@ -13,8 +13,8 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "common_video/h264/h264_common.h"
 #include "rtc_base/bit_buffer.h"
 #include "rtc_base/buffer.h"
@@ -111,7 +111,7 @@ void GenerateFakeSps(uint16_t width,
   }
 
   out_buffer->Clear();
-  H264::WriteRbsp(MakeArrayView(rbsp, byte_count), out_buffer);
+  H264::WriteRbsp(std::span(rbsp, byte_count), out_buffer);
 }
 
 TEST(H264SpsParserTest, TestSampleSPSHdLandscape) {
diff --git a/common_video/h264/sps_vui_rewriter.cc b/common_video/h264/sps_vui_rewriter.cc
index 4481afdd37..be0cbf123d 100644
--- a/common_video/h264/sps_vui_rewriter.cc
+++ b/common_video/h264/sps_vui_rewriter.cc
@@ -15,9 +15,9 @@
 #include <cstdint>
 #include <cstring>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/video/color_space.h"
 #include "common_video/h264/h264_common.h"
 #include "common_video/h264/sps_parser.h"
@@ -134,7 +134,7 @@ void SpsVuiRewriter::UpdateStats(ParseResult result, Direction direction) {
 }
 
 SpsVuiRewriter::ParseResult SpsVuiRewriter::ParseAndRewriteSps(
-    ArrayView<const uint8_t> buffer,
+    std::span<const uint8_t> buffer,
     std::optional<SpsParser::SpsState>* sps,
     const ColorSpace* color_space,
     Buffer* destination) {
@@ -211,7 +211,7 @@ SpsVuiRewriter::ParseResult SpsVuiRewriter::ParseAndRewriteSps(
 }
 
 SpsVuiRewriter::ParseResult SpsVuiRewriter::ParseAndRewriteSps(
-    ArrayView<const uint8_t> buffer,
+    std::span<const uint8_t> buffer,
     std::optional<SpsParser::SpsState>* sps,
     const ColorSpace* color_space,
     Buffer* destination,
@@ -223,7 +223,7 @@ SpsVuiRewriter::ParseResult SpsVuiRewriter::ParseAndRewriteSps(
 }
 
 Buffer SpsVuiRewriter::ParseOutgoingBitstreamAndRewrite(
-    ArrayView<const uint8_t> buffer,
+    std::span<const uint8_t> buffer,
     const ColorSpace* color_space) {
   std::vector<H264::NaluIndex> nalus = H264::FindNaluIndices(buffer);
 
@@ -233,10 +233,10 @@ Buffer SpsVuiRewriter::ParseOutgoingBitstreamAndRewrite(
 
   for (const H264::NaluIndex& nalu_index : nalus) {
     // Copy NAL unit start code.
-    ArrayView<const uint8_t> start_code = buffer.subspan(
+    std::span<const uint8_t> start_code = buffer.subspan(
         nalu_index.start_offset,
         nalu_index.payload_start_offset - nalu_index.start_offset);
-    ArrayView<const uint8_t> nalu = buffer.subspan(
+    std::span<const uint8_t> nalu = buffer.subspan(
         nalu_index.payload_start_offset, nalu_index.payload_size);
     if (nalu.empty()) {
       continue;
diff --git a/common_video/h264/sps_vui_rewriter.h b/common_video/h264/sps_vui_rewriter.h
index 665142f3a0..d4e85cc9f4 100644
--- a/common_video/h264/sps_vui_rewriter.h
+++ b/common_video/h264/sps_vui_rewriter.h
@@ -16,8 +16,8 @@
 #include <stdint.h>
 
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/video/color_space.h"
 #include "common_video/h264/sps_parser.h"
 #include "rtc_base/buffer.h"
@@ -44,7 +44,7 @@ class SpsVuiRewriter : private SpsParser {
   // SPS state. This function assumes that any previous headers
   // (NALU start, type, Stap-A, etc) have already been parsed and that RBSP
   // decoding has been performed.
-  static ParseResult ParseAndRewriteSps(ArrayView<const uint8_t> buffer,
+  static ParseResult ParseAndRewriteSps(std::span<const uint8_t> buffer,
                                         std::optional<SpsParser::SpsState>* sps,
                                         const ColorSpace* color_space,
                                         Buffer* destination,
@@ -53,11 +53,11 @@ class SpsVuiRewriter : private SpsParser {
   // Parses NAL units from `buffer`, strips AUD blocks and rewrites VUI in SPS
   // blocks if necessary.
   static Buffer ParseOutgoingBitstreamAndRewrite(
-      ArrayView<const uint8_t> buffer,
+      std::span<const uint8_t> buffer,
       const ColorSpace* color_space);
 
  private:
-  static ParseResult ParseAndRewriteSps(ArrayView<const uint8_t> buffer,
+  static ParseResult ParseAndRewriteSps(std::span<const uint8_t> buffer,
                                         std::optional<SpsParser::SpsState>* sps,
                                         const ColorSpace* color_space,
                                         Buffer* destination);
diff --git a/common_video/h264/sps_vui_rewriter_unittest.cc b/common_video/h264/sps_vui_rewriter_unittest.cc
index c1ea8038a3..ea31466a97 100644
--- a/common_video/h264/sps_vui_rewriter_unittest.cc
+++ b/common_video/h264/sps_vui_rewriter_unittest.cc
@@ -13,9 +13,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <tuple>
 
-#include "api/array_view.h"
 #include "api/video/color_space.h"
 #include "common_video/h264/h264_common.h"
 #include "common_video/h264/sps_parser.h"
@@ -250,7 +250,7 @@ void GenerateFakeSps(const VuiHeader& vui, Buffer* out_buffer) {
     byte_count++;
   }
 
-  H264::WriteRbsp(MakeArrayView(rbsp, byte_count), out_buffer);
+  H264::WriteRbsp(std::span(rbsp, byte_count), out_buffer);
 }
 
 void TestSps(const VuiHeader& vui,
diff --git a/common_video/h265/h265_bitstream_parser.cc b/common_video/h265/h265_bitstream_parser.cc
index b00080dae6..960f88c9c8 100644
--- a/common_video/h265/h265_bitstream_parser.cc
+++ b/common_video/h265/h265_bitstream_parser.cc
@@ -14,9 +14,9 @@
 #include <cstdlib>
 #include <limits>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "common_video/h265/h265_common.h"
 #include "common_video/h265/h265_pps_parser.h"
 #include "common_video/h265/h265_sps_parser.h"
@@ -84,7 +84,7 @@ H265BitstreamParser::~H265BitstreamParser() = default;
 // section 7.3.6.1. You can find it on this page:
 // http://www.itu.int/rec/T-REC-H.265
 H265BitstreamParser::Result H265BitstreamParser::ParseNonParameterSetNalu(
-    ArrayView<const uint8_t> source,
+    std::span<const uint8_t> source,
     uint8_t nalu_type) {
   last_slice_qp_delta_ = std::nullopt;
   last_slice_pps_id_ = std::nullopt;
@@ -520,7 +520,7 @@ const H265SpsParser::SpsState* H265BitstreamParser::GetSPS(uint32_t id) const {
   return &it->second;
 }
 
-void H265BitstreamParser::ParseSlice(ArrayView<const uint8_t> slice) {
+void H265BitstreamParser::ParseSlice(std::span<const uint8_t> slice) {
   if (slice.empty()) {
     RTC_LOG(LS_WARNING) << "Empty slice in H265 bitstream.";
     return;
@@ -594,7 +594,7 @@ void H265BitstreamParser::ParseSlice(ArrayView<const uint8_t> slice) {
 
 std::optional<uint32_t>
 H265BitstreamParser::ParsePpsIdFromSliceSegmentLayerRbsp(
-    ArrayView<const uint8_t> data,
+    std::span<const uint8_t> data,
     uint8_t nalu_type) {
   std::vector<uint8_t> unpacked_buffer = H265::ParseRbsp(data);
   BitstreamReader slice_reader(unpacked_buffer);
@@ -622,7 +622,7 @@ H265BitstreamParser::ParsePpsIdFromSliceSegmentLayerRbsp(
 }
 
 std::optional<bool> H265BitstreamParser::IsFirstSliceSegmentInPic(
-    ArrayView<const uint8_t> data) {
+    std::span<const uint8_t> data) {
   std::vector<uint8_t> unpacked_buffer = H265::ParseRbsp(data);
   BitstreamReader slice_reader(unpacked_buffer);
 
@@ -635,7 +635,7 @@ std::optional<bool> H265BitstreamParser::IsFirstSliceSegmentInPic(
   return first_slice_segment_in_pic_flag;
 }
 
-void H265BitstreamParser::ParseBitstream(ArrayView<const uint8_t> bitstream) {
+void H265BitstreamParser::ParseBitstream(std::span<const uint8_t> bitstream) {
   std::vector<H265::NaluIndex> nalu_indices = H265::FindNaluIndices(bitstream);
   for (const H265::NaluIndex& index : nalu_indices)
     ParseSlice(
diff --git a/common_video/h265/h265_bitstream_parser.h b/common_video/h265/h265_bitstream_parser.h
index 20eb8dfb74..f7b19297fe 100644
--- a/common_video/h265/h265_bitstream_parser.h
+++ b/common_video/h265/h265_bitstream_parser.h
@@ -15,8 +15,8 @@
 #include <stdint.h>
 
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/video_codecs/bitstream_parser.h"
 #include "common_video/h265/h265_pps_parser.h"
 #include "common_video/h265/h265_sps_parser.h"
@@ -34,19 +34,19 @@ class RTC_EXPORT H265BitstreamParser : public BitstreamParser {
   ~H265BitstreamParser() override;
 
   // New interface.
-  void ParseBitstream(ArrayView<const uint8_t> bitstream) override;
+  void ParseBitstream(std::span<const uint8_t> bitstream) override;
   std::optional<int> GetLastSliceQp() const override;
 
   std::optional<uint32_t> GetLastSlicePpsId() const;
 
   static std::optional<uint32_t> ParsePpsIdFromSliceSegmentLayerRbsp(
-      ArrayView<const uint8_t> data,
+      std::span<const uint8_t> data,
       uint8_t nalu_type);
 
   // Returns true if the slice segment is the first in the picture; otherwise
   // return false. If parse failed, return nullopt.
   static std::optional<bool> IsFirstSliceSegmentInPic(
-      ArrayView<const uint8_t> data);
+      std::span<const uint8_t> data);
 
  protected:
   enum Result {
@@ -54,8 +54,8 @@ class RTC_EXPORT H265BitstreamParser : public BitstreamParser {
     kInvalidStream,
     kUnsupportedStream,
   };
-  void ParseSlice(ArrayView<const uint8_t> slice);
-  Result ParseNonParameterSetNalu(ArrayView<const uint8_t> source,
+  void ParseSlice(std::span<const uint8_t> slice);
+  Result ParseNonParameterSetNalu(std::span<const uint8_t> source,
                                   uint8_t nalu_type);
 
   const H265PpsParser::PpsState* GetPPS(uint32_t id) const;
diff --git a/common_video/h265/h265_bitstream_parser_unittest.cc b/common_video/h265/h265_bitstream_parser_unittest.cc
index 1045a47ebd..e37b686f01 100644
--- a/common_video/h265/h265_bitstream_parser_unittest.cc
+++ b/common_video/h265/h265_bitstream_parser_unittest.cc
@@ -13,7 +13,6 @@
 #include <cstdint>
 #include <optional>
 
-#include "api/array_view.h"
 #include "common_video/h265/h265_common.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
diff --git a/common_video/h265/h265_common.cc b/common_video/h265/h265_common.cc
index 332bab7acc..528145b055 100644
--- a/common_video/h265/h265_common.cc
+++ b/common_video/h265/h265_common.cc
@@ -11,9 +11,9 @@
 #include "common_video/h265/h265_common.h"
 
 #include <cstdint>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "common_video/h264/h264_common.h"
 #include "common_video/h265/h265_inline.h"
 #include "rtc_base/buffer.h"
@@ -23,7 +23,7 @@ namespace H265 {
 
 constexpr uint8_t kNaluTypeMask = 0x7E;
 
-std::vector<NaluIndex> FindNaluIndices(ArrayView<const uint8_t> buffer) {
+std::vector<NaluIndex> FindNaluIndices(std::span<const uint8_t> buffer) {
   std::vector<H264::NaluIndex> indices = H264::FindNaluIndices(buffer);
   std::vector<NaluIndex> results;
   results.reserve(indices.size());
@@ -39,11 +39,11 @@ NaluType ParseNaluType(uint8_t data) {
   return static_cast<NaluType>((data & kNaluTypeMask) >> 1);
 }
 
-std::vector<uint8_t> ParseRbsp(ArrayView<const uint8_t> data) {
+std::vector<uint8_t> ParseRbsp(std::span<const uint8_t> data) {
   return H264::ParseRbsp(data);
 }
 
-void WriteRbsp(ArrayView<const uint8_t> bytes, Buffer* destination) {
+void WriteRbsp(std::span<const uint8_t> bytes, Buffer* destination) {
   H264::WriteRbsp(bytes, destination);
 }
 
diff --git a/common_video/h265/h265_common.h b/common_video/h265/h265_common.h
index 9ce9fbee58..2cc83ea5ec 100644
--- a/common_video/h265/h265_common.h
+++ b/common_video/h265/h265_common.h
@@ -13,9 +13,9 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/system/rtc_export.h"
 
@@ -80,12 +80,12 @@ struct NaluIndex {
 
 // Returns a vector of the NALU indices in the given buffer.
 RTC_EXPORT std::vector<NaluIndex> FindNaluIndices(
-    ArrayView<const uint8_t> buffer);
+    std::span<const uint8_t> buffer);
 
 // TODO: bugs.webrtc.org/42225170 - Deprecate.
 inline std::vector<NaluIndex> FindNaluIndices(const uint8_t* buffer,
                                               size_t buffer_size) {
-  return FindNaluIndices(MakeArrayView(buffer, buffer_size));
+  return FindNaluIndices(std::span(buffer, buffer_size));
 }
 
 // Get the NAL type from the header byte immediately following start sequence.
@@ -105,23 +105,23 @@ RTC_EXPORT NaluType ParseNaluType(uint8_t data);
 // the 03 emulation byte.
 
 // Parse the given data and remove any emulation byte escaping.
-std::vector<uint8_t> ParseRbsp(ArrayView<const uint8_t> data);
+std::vector<uint8_t> ParseRbsp(std::span<const uint8_t> data);
 
 // TODO: bugs.webrtc.org/42225170 - Deprecate.
 inline std::vector<uint8_t> ParseRbsp(const uint8_t* data, size_t length) {
-  return ParseRbsp(MakeArrayView(data, length));
+  return ParseRbsp(std::span(data, length));
 }
 
 // Write the given data to the destination buffer, inserting and emulation
 // bytes in order to escape any data the could be interpreted as a start
 // sequence.
-void WriteRbsp(ArrayView<const uint8_t> bytes, Buffer* destination);
+void WriteRbsp(std::span<const uint8_t> bytes, Buffer* destination);
 
 // TODO: bugs.webrtc.org/42225170 -  Deprecate.
 inline void WriteRbsp(const uint8_t* bytes,
                       size_t length,
                       Buffer* destination) {
-  WriteRbsp(MakeArrayView(bytes, length), destination);
+  WriteRbsp(std::span(bytes, length), destination);
 }
 
 uint32_t Log2Ceiling(uint32_t value);
diff --git a/common_video/h265/h265_pps_parser.cc b/common_video/h265/h265_pps_parser.cc
index a5f7adb7f9..460e936a50 100644
--- a/common_video/h265/h265_pps_parser.cc
+++ b/common_video/h265/h265_pps_parser.cc
@@ -12,9 +12,9 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "common_video/h265/h265_common.h"
 #include "common_video/h265/h265_sps_parser.h"
 #include "rtc_base/bitstream_reader.h"
@@ -65,7 +65,7 @@ constexpr int kMaxRefIdxActive = 15;
 // http://www.itu.int/rec/T-REC-H.265
 
 std::optional<H265PpsParser::PpsState> H265PpsParser::ParsePps(
-    ArrayView<const uint8_t> data,
+    std::span<const uint8_t> data,
     const H265SpsParser::SpsState* sps) {
   // First, parse out rbsp, which is basically the source buffer minus emulation
   // bytes (the last byte of a 0x00 0x00 0x03 sequence). RBSP is defined in
@@ -73,7 +73,7 @@ std::optional<H265PpsParser::PpsState> H265PpsParser::ParsePps(
   return ParseInternal(H265::ParseRbsp(data), sps);
 }
 
-bool H265PpsParser::ParsePpsIds(ArrayView<const uint8_t> data,
+bool H265PpsParser::ParsePpsIds(std::span<const uint8_t> data,
                                 uint32_t* pps_id,
                                 uint32_t* sps_id) {
   RTC_DCHECK(pps_id);
@@ -91,7 +91,7 @@ bool H265PpsParser::ParsePpsIds(ArrayView<const uint8_t> data,
 }
 
 std::optional<H265PpsParser::PpsState> H265PpsParser::ParseInternal(
-    ArrayView<const uint8_t> buffer,
+    std::span<const uint8_t> buffer,
     const H265SpsParser::SpsState* sps) {
   BitstreamReader reader(buffer);
   PpsState pps;
diff --git a/common_video/h265/h265_pps_parser.h b/common_video/h265/h265_pps_parser.h
index 2cb3f67dab..c73be46bce 100644
--- a/common_video/h265/h265_pps_parser.h
+++ b/common_video/h265/h265_pps_parser.h
@@ -14,8 +14,8 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "common_video/h265/h265_sps_parser.h"
 #include "rtc_base/bitstream_reader.h"
 #include "rtc_base/system/rtc_export.h"
@@ -46,17 +46,17 @@ class RTC_EXPORT H265PpsParser {
   };
 
   // Unpack RBSP and parse PPS state from the supplied buffer.
-  static std::optional<PpsState> ParsePps(ArrayView<const uint8_t> data,
+  static std::optional<PpsState> ParsePps(std::span<const uint8_t> data,
                                           const H265SpsParser::SpsState* sps);
   // TODO: bugs.webrtc.org/42225170 - Deprecate.
   static inline std::optional<PpsState> ParsePps(
       const uint8_t* data,
       size_t length,
       const H265SpsParser::SpsState* sps) {
-    return ParsePps(MakeArrayView(data, length), sps);
+    return ParsePps(std::span(data, length), sps);
   }
 
-  static bool ParsePpsIds(ArrayView<const uint8_t> data,
+  static bool ParsePpsIds(std::span<const uint8_t> data,
                           uint32_t* pps_id,
                           uint32_t* sps_id);
   // TODO: bugs.webrtc.org/42225170 - Deprecate.
@@ -64,14 +64,14 @@ class RTC_EXPORT H265PpsParser {
                                  size_t length,
                                  uint32_t* pps_id,
                                  uint32_t* sps_id) {
-    return ParsePpsIds(MakeArrayView(data, length), pps_id, sps_id);
+    return ParsePpsIds(std::span(data, length), pps_id, sps_id);
   }
 
  protected:
   // Parse the PPS state, for a bit buffer where RBSP decoding has already been
   // performed.
   static std::optional<PpsState> ParseInternal(
-      ArrayView<const uint8_t> buffer,
+      std::span<const uint8_t> buffer,
       const H265SpsParser::SpsState* sps);
   static bool ParsePpsIdsInternal(BitstreamReader& reader,
                                   uint32_t& pps_id,
diff --git a/common_video/h265/h265_pps_parser_unittest.cc b/common_video/h265/h265_pps_parser_unittest.cc
index 688e9e98d7..668b89ace4 100644
--- a/common_video/h265/h265_pps_parser_unittest.cc
+++ b/common_video/h265/h265_pps_parser_unittest.cc
@@ -14,8 +14,8 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "common_video/h265/h265_common.h"
 #include "common_video/h265/h265_sps_parser.h"
 #include "rtc_base/bit_buffer.h"
@@ -164,7 +164,7 @@ void WritePps(const H265PpsParser::PpsState& pps,
     bit_buffer.GetCurrentOffset(&byte_offset, &bit_offset);
   }
 
-  H265::WriteRbsp(MakeArrayView(data, byte_offset), out_buffer);
+  H265::WriteRbsp(std::span(data, byte_offset), out_buffer);
 }
 
 class H265PpsParserTest : public ::testing::Test {
diff --git a/common_video/h265/h265_sps_parser.cc b/common_video/h265/h265_sps_parser.cc
index 0916de5ad3..da6f913f3c 100644
--- a/common_video/h265/h265_sps_parser.cc
+++ b/common_video/h265/h265_sps_parser.cc
@@ -15,9 +15,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "common_video/h265/h265_common.h"
 #include "rtc_base/bitstream_reader.h"
 #include "rtc_base/logging.h"
@@ -107,7 +107,7 @@ size_t H265SpsParser::GetDpbMaxPicBuf(int general_profile_idc) {
 
 // Unpack RBSP and parse SPS state from the supplied buffer.
 std::optional<H265SpsParser::SpsState> H265SpsParser::ParseSps(
-    ArrayView<const uint8_t> data) {
+    std::span<const uint8_t> data) {
   return ParseSpsInternal(H265::ParseRbsp(data));
 }
 
@@ -388,7 +388,7 @@ H265SpsParser::ParseProfileTierLevel(bool profile_present,
 }
 
 std::optional<H265SpsParser::SpsState> H265SpsParser::ParseSpsInternal(
-    ArrayView<const uint8_t> buffer) {
+    std::span<const uint8_t> buffer) {
   BitstreamReader reader(buffer);
 
   // Now, we need to use a bit buffer to parse through the actual H265 SPS
diff --git a/common_video/h265/h265_sps_parser.h b/common_video/h265/h265_sps_parser.h
index 1bccf6adcc..03e588d267 100644
--- a/common_video/h265/h265_sps_parser.h
+++ b/common_video/h265/h265_sps_parser.h
@@ -14,9 +14,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "rtc_base/bitstream_reader.h"
 #include "rtc_base/system/rtc_export.h"
 
@@ -106,11 +106,11 @@ class RTC_EXPORT H265SpsParser {
   };
 
   // Unpack RBSP and parse SPS state from the supplied buffer.
-  static std::optional<SpsState> ParseSps(ArrayView<const uint8_t> data);
+  static std::optional<SpsState> ParseSps(std::span<const uint8_t> data);
   // TODO: bugs.webrtc.org/42225170 - Deprecate.
   static inline std::optional<SpsState> ParseSps(const uint8_t* data,
                                                  size_t length) {
-    return ParseSps(MakeArrayView(data, length));
+    return ParseSps(std::span(data, length));
   }
 
   static bool ParseScalingListData(BitstreamReader& reader);
@@ -131,7 +131,7 @@ class RTC_EXPORT H265SpsParser {
   // Parse the SPS state, for a bit buffer where RBSP decoding has already been
   // performed.
   static std::optional<SpsState> ParseSpsInternal(
-      ArrayView<const uint8_t> buffer);
+      std::span<const uint8_t> buffer);
 
   // From Table A.8 - General tier and level limits.
   static int GetMaxLumaPs(int general_level_idc);
diff --git a/common_video/h265/h265_sps_parser_unittest.cc b/common_video/h265/h265_sps_parser_unittest.cc
index 1d3e1b2859..5a8e051554 100644
--- a/common_video/h265/h265_sps_parser_unittest.cc
+++ b/common_video/h265/h265_sps_parser_unittest.cc
@@ -13,9 +13,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "common_video/h265/h265_common.h"
 #include "rtc_base/bit_buffer.h"
 #include "rtc_base/buffer.h"
@@ -370,7 +370,7 @@ void WriteSps(uint16_t width,
   }
 
   out_buffer->Clear();
-  H265::WriteRbsp(MakeArrayView(rbsp, byte_count), out_buffer);
+  H265::WriteRbsp(std::span(rbsp, byte_count), out_buffer);
 }
 
 class H265SpsParserTest : public ::testing::Test {
diff --git a/common_video/h265/h265_vps_parser.cc b/common_video/h265/h265_vps_parser.cc
index 8ebe5f96a1..83821c5ad3 100644
--- a/common_video/h265/h265_vps_parser.cc
+++ b/common_video/h265/h265_vps_parser.cc
@@ -12,8 +12,8 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "common_video/h265/h265_common.h"
 #include "rtc_base/bitstream_reader.h"
 
@@ -27,12 +27,12 @@ H265VpsParser::VpsState::VpsState() = default;
 
 // Unpack RBSP and parse VPS state from the supplied buffer.
 std::optional<H265VpsParser::VpsState> H265VpsParser::ParseVps(
-    ArrayView<const uint8_t> data) {
+    std::span<const uint8_t> data) {
   return ParseInternal(H265::ParseRbsp(data));
 }
 
 std::optional<H265VpsParser::VpsState> H265VpsParser::ParseInternal(
-    ArrayView<const uint8_t> buffer) {
+    std::span<const uint8_t> buffer) {
   BitstreamReader reader(buffer);
 
   // Now, we need to use a bit buffer to parse through the actual H265 VPS
diff --git a/common_video/h265/h265_vps_parser.h b/common_video/h265/h265_vps_parser.h
index e890e4e64f..ad423d22f7 100644
--- a/common_video/h265/h265_vps_parser.h
+++ b/common_video/h265/h265_vps_parser.h
@@ -14,8 +14,8 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
@@ -32,17 +32,17 @@ class RTC_EXPORT H265VpsParser {
   };
 
   // Unpack RBSP and parse VPS state from the supplied buffer.
-  static std::optional<VpsState> ParseVps(ArrayView<const uint8_t> data);
+  static std::optional<VpsState> ParseVps(std::span<const uint8_t> data);
   // TODO: bugs.webrtc.org/42225170 - Deprecate.
   static inline std::optional<VpsState> ParseVps(const uint8_t* data,
                                                  size_t length) {
-    return ParseVps(MakeArrayView(data, length));
+    return ParseVps(std::span(data, length));
   }
 
  protected:
   // Parse the VPS state, for a bit buffer where RBSP decoding has already been
   // performed.
-  static std::optional<VpsState> ParseInternal(ArrayView<const uint8_t> buffer);
+  static std::optional<VpsState> ParseInternal(std::span<const uint8_t> buffer);
 };
 
 }  // namespace webrtc

From fef1bf4b3fa35f5a9fb13131b957ba0905a94bb0 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Fri, 27 Mar 2026 09:18:05 +0100
Subject: [PATCH 422/472] Refactor RTP Header Extension Management into
 RtpTransport

Moves the caching, validation, and history tracking of RTP header
extensions from BaseChannel to RtpTransport. This aligns with the BUNDLE
model and simplifies the concurrency model by handling this on the
network thread.

Bug: webrtc:42222117
Change-Id: Iae4c65c938ee085d242cc3a27b5abaff5d7471de
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/455460
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47275}
---
 pc/BUILD.gn                  |   6 ++
 pc/channel.cc                | 139 +++++++++++------------------------
 pc/channel.h                 |   7 --
 pc/channel_unittest.cc       |  12 ++-
 pc/rtp_transport.cc          |  93 ++++++++++++++++++++++-
 pc/rtp_transport.h           |  11 ++-
 pc/rtp_transport_internal.h  |   8 +-
 pc/rtp_transport_unittest.cc |  26 +++++++
 8 files changed, 183 insertions(+), 119 deletions(-)

diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index ca8dcd86b9..1f19fdc878 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -103,6 +103,7 @@ rtc_library("channel") {
     "../rtc_base/containers:flat_set",
     "../rtc_base/network:sent_packet",
     "//third_party/abseil-cpp/absl/algorithm:container",
+    "//third_party/abseil-cpp/absl/cleanup",
     "//third_party/abseil-cpp/absl/functional:any_invocable",
     "//third_party/abseil-cpp/absl/strings:string_view",
   ]
@@ -588,6 +589,8 @@ rtc_library("rtp_transport") {
     ":rtp_transport_internal",
     ":session_description",
     "../api:field_trials_view",
+    "../api:rtc_error",
+    "../api:rtp_parameters",
     "../api:sequence_checker",
     "../api/task_queue",
     "../api/task_queue:pending_task_safety_flag",
@@ -606,11 +609,13 @@ rtc_library("rtp_transport") {
     "../rtc_base:macromagic",
     "../rtc_base:network_route",
     "../rtc_base:socket",
+    "../rtc_base:stringutils",
     "../rtc_base/containers:flat_map",
     "../rtc_base/containers:flat_set",
     "../rtc_base/network:received_packet",
     "../rtc_base/network:sent_packet",
     "../rtc_base/system:no_unique_address",
+    "//third_party/abseil-cpp/absl/algorithm:container",
     "//third_party/abseil-cpp/absl/strings:string_view",
   ]
 }
@@ -623,6 +628,7 @@ rtc_source_set("rtp_transport_internal") {
   sources = [ "rtp_transport_internal.h" ]
   deps = [
     ":session_description",
+    "../api:rtc_error",
     "../api/task_queue:pending_task_safety_flag",
     "../api/transport:ecn_marking",
     "../api/units:timestamp",
diff --git a/pc/channel.cc b/pc/channel.cc
index 5761f2aafe..1774778eec 100644
--- a/pc/channel.cc
+++ b/pc/channel.cc
@@ -11,7 +11,6 @@
 #include "pc/channel.h"
 
 #include <algorithm>
-#include <bitset>
 #include <cstdint>
 #include <memory>
 #include <optional>
@@ -20,6 +19,7 @@
 #include <vector>
 
 #include "absl/algorithm/container.h"
+#include "absl/cleanup/cleanup.h"
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
 #include "api/crypto/crypto_options.h"
@@ -267,23 +267,6 @@ bool BaseChannel::SetRtpTransport(RtpTransportInternal* rtp_transport) {
 
   if (rtp_transport_) {
     DisconnectFromRtpTransport_n();
-    // Clear the cached header extensions on the worker.
-    // If the network and worker thread pointers are configured to map to the
-    // same thread object, we'll do this synchronously. To start with, we're on
-    // the correct thread anyway, but an important second reason is that other
-    // parts of the code (SetLocalContent_w, SetRemoteContent_w) may execute a
-    // BlockingCall that touches `rtp_header_extensions` which, for the case
-    // where the threads are the same, will be executed before the lambda in the
-    // PostTask and not after, which may lead to unexpected behavior.
-    if (worker_thread_ == network_thread_) {
-      RTC_DCHECK_RUN_ON(worker_thread());
-      rtp_header_extensions_.clear();
-    } else {
-      worker_thread_->PostTask(SafeTask(alive_, [this] {
-        RTC_DCHECK_RUN_ON(worker_thread());
-        rtp_header_extensions_.clear();
-      }));
-    }
   }
 
   RTC_DCHECK(!rtp_transport_);
@@ -578,50 +561,27 @@ RTCError BaseChannel::MaybeUpdateDemuxerAndRtpExtensions_w(
     std::optional<flat_set<uint8_t>> payload_types,
     const RtpHeaderExtensions& extensions,
     std::optional<flat_set<uint32_t>> ssrcs) {
-  bool update_extensions = true;
-  if (rtp_header_extensions_ == extensions) {
-    update_extensions = false;  // No need to update header extensions.
-  } else {
-    RTCError error = CheckRtpExtensionValidity(extensions);
-    if (!error.ok()) {
-      return error;
-    }
-    rtp_header_extensions_ = extensions;
-
-    for (const auto& extension : extensions) {
-      if (extension.id == 0)
-        continue;
-      if (absl::c_find_if(historical_rtp_header_extensions_,
-                          [&](const RtpExtension& ext) {
-                            return ext.id == extension.id;
-                          }) == historical_rtp_header_extensions_.end()) {
-        historical_rtp_header_extensions_.push_back(extension);
-      }
-    }
-  }
-
-  if (!update_demuxer && !update_extensions && !payload_types.has_value() &&
-      !ssrcs.has_value()) {
-    return RTCError::OK();  // No update needed.
-  }
-
-  // TODO(bugs.webrtc.org/13536): See if we can do this asynchronously.
-
-  if (update_demuxer || payload_types.has_value() || ssrcs.has_value()) {
+  const bool pending_update =
+      update_demuxer || payload_types.has_value() || ssrcs.has_value();
+  if (pending_update) {
     media_receive_channel()->OnDemuxerCriteriaUpdatePending();
   }
+  absl::Cleanup cleanup = [this, pending_update] {
+    if (pending_update) {
+      media_receive_channel()->OnDemuxerCriteriaUpdateComplete();
+    }
+  };
 
+  // TODO(bugs.webrtc.org/13536): See if we can do this asynchronously.
   RTCError error = network_thread()->BlockingCall([&]() -> RTCError {
     RTC_DCHECK_RUN_ON(network_thread());
-    if (!rtp_transport_) {
-      // To repro this situation, run the
-      // `ApplyDescriptionWithSameSsrcsBundledFails` test.
-      return LOG_ERROR(RTCError::InvalidState()
-                       << "No transport assigned for mid=" << mid());
-    }
-
-    if (update_extensions) {
-      rtp_transport_->RegisterRtpHeaderExtensionMap(mid(), extensions);
+    RTCError error =
+        rtp_transport_
+            ? rtp_transport_->RegisterRtpHeaderExtensionMap(mid(), extensions)
+            : (RTCError::InvalidState() << "No transport assigned.");
+    if (!error.ok()) {
+      error.string_builder() << " (mid=" << mid() << ")";
+      return LOG_ERROR(error);
     }
 
     if (payload_types) {
@@ -650,9 +610,6 @@ RTCError BaseChannel::MaybeUpdateDemuxerAndRtpExtensions_w(
     return RTCError::OK();
   });
 
-  if (update_demuxer || payload_types.has_value() || ssrcs.has_value())
-    media_receive_channel()->OnDemuxerCriteriaUpdateComplete();
-
   return error;
 }
 
@@ -660,6 +617,9 @@ bool BaseChannel::RegisterRtpDemuxerSink_w(
     bool clear_payload_types,
     std::optional<flat_set<uint32_t>> ssrcs) {
   media_receive_channel()->OnDemuxerCriteriaUpdatePending();
+  absl::Cleanup cleanup = [this] {
+    media_receive_channel()->OnDemuxerCriteriaUpdateComplete();
+  };
   bool ret = network_thread_->BlockingCall([&] {
     RTC_DCHECK_RUN_ON(network_thread());
     if (!rtp_transport_) {
@@ -694,8 +654,6 @@ bool BaseChannel::RegisterRtpDemuxerSink_w(
     return rtp_transport_->RegisterRtpDemuxerSink(criteria, this);
   });
 
-  media_receive_channel()->OnDemuxerCriteriaUpdateComplete();
-
   return ret;
 }
 
@@ -928,41 +886,24 @@ RtpHeaderExtensions BaseChannel::GetDeduplicatedRtpHeaderExtensions(
                                                    extensions_filter_);
 }
 
+// TODO: webrtc:42222117 - Move header extension logic in the channel classes
+// to the network thread. At the moment, this function does a BlockingCall
+// to the network thread in order to delegate the check to the transport.
+// The worker and network threads are commonly configured to map to the same
+// actual thread, so a blocking call in those cases isn't expensive, although
+// not ideal.
 RTCError BaseChannel::CheckRtpExtensionValidity(
     const RtpHeaderExtensions& extensions) const {
-  std::bitset<1 + RtpExtension::kMaxId> id_used;
-  for (const auto& extension : extensions) {
-    if (extension.id == 0)
-      continue;
-    if (extension.id < RtpExtension::kMinId ||
-        extension.id > RtpExtension::kMaxId) {
-      return RTCError::InvalidParameter()
-             << "Bad RTP extension ID: " << extension.ToString();
+  return network_thread()->BlockingCall([&]() -> RTCError {
+    RTC_DCHECK_RUN_ON(network_thread());
+    RTCError error =
+        rtp_transport_ ? rtp_transport_->VerifyRtpHeaderExtensionMap(extensions)
+                       : (RTCError::InvalidState() << "No transport assigned.");
+    if (!error.ok()) {
+      error.string_builder() << " (mid=" << mid() << ")";
     }
-    if (id_used[extension.id]) {
-      return RTCError::InvalidParameter()
-             << "Duplicate RTP extension ID: " << extension.ToString();
-    }
-    id_used[extension.id] = true;
-  }
-
-  for (const auto& new_extension : extensions) {
-    if (new_extension.id == 0)
-      continue;
-    auto it = absl::c_find_if(
-        historical_rtp_header_extensions_,
-        [&](const RtpExtension& ext) { return ext.id == new_extension.id; });
-    if (it != historical_rtp_header_extensions_.end() &&
-        it->uri != new_extension.uri) {
-      return RTCError::InvalidParameter()
-             << "Failed to update RTP header extensions for m-section with "
-             << "mid='" << mid() << "'. RTP extension ID reassignment from "
-             << it->uri << " to " << new_extension.uri << " for ID "
-             << new_extension.id << ".";
-    }
-  }
-
-  return RTCError::OK();
+    return error;
+  });
 }
 
 void BaseChannel::SignalSentPacket_n(const SentPacketInfo& sent_packet) {
@@ -1186,8 +1127,6 @@ RTCError VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
                                          SdpType type) {
   TRACE_EVENT0("webrtc", "VideoChannel::SetLocalContent_w");
 
-  RTC_LOG_THREAD_BLOCK_COUNT();
-
   RtpHeaderExtensions header_extensions =
       GetDeduplicatedRtpHeaderExtensions(content->rtp_header_extensions());
 
@@ -1196,6 +1135,8 @@ RTCError VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
     return error;
   }
 
+  RTC_LOG_THREAD_BLOCK_COUNT();
+
   // TODO: issues.webrtc.org/383078466 - remove if pushdown on answer is enough.
   media_send_channel()->SetExtmapAllowMixed(content->extmap_allow_mixed());
 
@@ -1256,9 +1197,13 @@ RTCError VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
 
   RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(0);
 
-  return MaybeUpdateDemuxerAndRtpExtensions_w(
+  error = MaybeUpdateDemuxerAndRtpExtensions_w(
       /*update_demuxer=*/false, std::move(payload_types),
       recv_params.extensions, /*ssrcs=*/std::nullopt);
+
+  RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(1);
+
+  return error;
 }
 
 RTCError VideoChannel::SetRemoteContent_w(
diff --git a/pc/channel.h b/pc/channel.h
index 6b4438ab19..fcb01262a9 100644
--- a/pc/channel.h
+++ b/pc/channel.h
@@ -354,13 +354,6 @@ class BaseChannel : public ChannelInterface,
 
   // Cached list of payload types, used if payload type demuxing is re-enabled.
   flat_set<uint8_t> payload_types_ RTC_GUARDED_BY(network_thread());
-  // A stored copy of the rtp header extensions as applied to the transport.
-  RtpHeaderExtensions rtp_header_extensions_ RTC_GUARDED_BY(worker_thread());
-
-  // Set of all historic RTP header extensions mapped, keyed by URI,
-  // to ensure no ID-URI reassignment occurs per RFC 8285.
-  RtpHeaderExtensions historical_rtp_header_extensions_
-      RTC_GUARDED_BY(worker_thread());
 
   const std::string mid_;
   flat_set<uint32_t> ssrcs_ RTC_GUARDED_BY(network_thread());
diff --git a/pc/channel_unittest.cc b/pc/channel_unittest.cc
index aae219ffd5..1705685fe6 100644
--- a/pc/channel_unittest.cc
+++ b/pc/channel_unittest.cc
@@ -69,6 +69,7 @@ namespace {
 using ::testing::AllOf;
 using ::testing::ElementsAre;
 using ::testing::Field;
+using ::testing::HasSubstr;
 using ::webrtc::CreateTestFieldTrials;
 using ::webrtc::DtlsTransportInternal;
 using ::webrtc::FakeVoiceMediaReceiveChannel;
@@ -708,8 +709,7 @@ class ChannelTest : public ::testing::Test {
     CreateChannels(0, 0);
     RTCError error = channel1_->SetLocalContent(&local, SdpType::kOffer);
     EXPECT_FALSE(error.ok());
-    EXPECT_THAT(error.message(),
-                ::testing::HasSubstr("Duplicate RTP extension ID"));
+    EXPECT_THAT(error.message(), HasSubstr("Duplicate extension ID"));
   }
 
   void TestInvalidRtpHeaderExtensionIds() {
@@ -722,7 +722,7 @@ class ChannelTest : public ::testing::Test {
     CreateChannels(0, 0);
     RTCError error = channel1_->SetLocalContent(&local, SdpType::kOffer);
     EXPECT_FALSE(error.ok());
-    EXPECT_THAT(error.message(), ::testing::HasSubstr("Bad RTP extension ID"));
+    EXPECT_THAT(error.message(), HasSubstr("Bad extension ID"));
   }
 
   void TestRtpHeaderExtensionIdReassignment() {
@@ -743,8 +743,7 @@ class ChannelTest : public ::testing::Test {
     RTCError error =
         channel1_->SetLocalContent(&local_updated, SdpType::kOffer);
     EXPECT_FALSE(error.ok());
-    EXPECT_THAT(error.message(),
-                ::testing::HasSubstr("RTP extension ID reassignment"));
+    EXPECT_THAT(error.message(), HasSubstr("RTP extension ID reassignment"));
   }
 
   void TestRtpHeaderExtensionIdHistoryReassignment() {
@@ -770,8 +769,7 @@ class ChannelTest : public ::testing::Test {
     RTCError error =
         channel1_->SetLocalContent(&local_updated, SdpType::kOffer);
     EXPECT_FALSE(error.ok());
-    EXPECT_THAT(error.message(),
-                ::testing::HasSubstr("RTP extension ID reassignment"));
+    EXPECT_THAT(error.message(), HasSubstr("RTP extension ID reassignment"));
   }
 
   // Test that SetLocalContent and SetRemoteContent properly configure
diff --git a/pc/rtp_transport.cc b/pc/rtp_transport.cc
index c23046f51c..20db5d3a03 100644
--- a/pc/rtp_transport.cc
+++ b/pc/rtp_transport.cc
@@ -11,6 +11,8 @@
 #include "pc/rtp_transport.h"
 
 #include <algorithm>
+#include <bitset>
+#include <cstddef>
 #include <cstdint>
 #include <memory>
 #include <optional>
@@ -18,7 +20,10 @@
 #include <utility>
 #include <vector>
 
+#include "absl/algorithm/container.h"
 #include "absl/strings/string_view.h"
+#include "api/rtc_error.h"
+#include "api/rtp_parameters.h"
 #include "api/sequence_checker.h"
 #include "api/task_queue/pending_task_safety_flag.h"
 #include "api/task_queue/task_queue_base.h"
@@ -55,6 +60,28 @@ void RemoveExtensionMapForMid(
   }
 }
 
+RTCError VerifyExtensionIds(const RtpHeaderExtensions& extensions) {
+  using ExtensionsUsed = std::bitset<1 + RtpExtension::kMaxId>;
+  ExtensionsUsed id_used;
+  for (const auto& extension : extensions) {
+    if (extension.id == 0) {
+      continue;
+    }
+    if (extension.id < RtpExtension::kMinId ||
+        extension.id > RtpExtension::kMaxId) {
+      return RTCError::InvalidParameter()
+             << "Bad extension ID: " << extension.ToString();
+    }
+    ExtensionsUsed::reference entry = id_used[extension.id];
+    if (entry) {
+      return RTCError::InvalidParameter()
+             << "Duplicate extension ID: " << extension.ToString();
+    }
+    entry = true;
+  }
+  return RTCError::OK();
+}
+
 }  // namespace
 
 void RtpTransport::SetRtcpMuxEnabled(bool enable) {
@@ -202,14 +229,72 @@ bool RtpTransport::SendPacket(bool rtcp,
   return true;
 }
 
-void RtpTransport::RegisterRtpHeaderExtensionMap(
-    absl::string_view mid,
-    const RtpHeaderExtensions& header_extensions) {
+RTCError RtpTransport::VerifyRtpHeaderExtensionMap(
+    const RtpHeaderExtensions& extensions) const {
   RTC_DCHECK_RUN_ON(&network_thread_checker_);
+
+  RTCError error = VerifyExtensionIds(extensions);
+  if (!error.ok()) {
+    return error;
+  }
+
+  for (const auto& new_extension : extensions) {
+    if (new_extension.id == 0) {
+      continue;
+    }
+    auto it = absl::c_find_if(
+        historical_rtp_header_extensions_,
+        [&](const RtpExtension& ext) { return ext.id == new_extension.id; });
+    if (it != historical_rtp_header_extensions_.end() &&
+        it->uri != new_extension.uri) {
+      return RTCError::InvalidParameter()
+             << "RTP extension ID reassignment not supported (id="
+             << new_extension.id << ", old_uri=\"" << it->uri
+             << "\", new_uri=\"" << new_extension.uri << "\").";
+    }
+  }
+
+  return RTCError::OK();
+}
+
+RTCError RtpTransport::RegisterRtpHeaderExtensionMap(
+    absl::string_view mid,
+    const RtpHeaderExtensions& extensions) {
+  RTC_DCHECK_RUN_ON(&network_thread_checker_);
+
+  RTCError error = VerifyRtpHeaderExtensionMap(extensions);
+  if (!error.ok()) {
+    return error;
+  }
+
+  auto existing_extensions =
+      absl::c_find_if(header_extensions_by_mid_,
+                      [mid](const auto& kv) { return kv.first == mid; });
+  if (existing_extensions != header_extensions_by_mid_.end() &&
+      existing_extensions->second == extensions) {
+    return RTCError::OK();
+  }
+
+  for (const RtpExtension& extension : extensions) {
+    if (extension.id == 0) {
+      continue;
+    }
+    auto it = absl::c_find_if(historical_rtp_header_extensions_,
+                              [&extension](const RtpExtension& ext) {
+                                return ext.id == extension.id;
+                              });
+    if (it == historical_rtp_header_extensions_.end()) {
+      historical_rtp_header_extensions_.push_back(extension);
+    } else {
+      RTC_DCHECK_EQ(it->uri, extension.uri);
+    }
+  }
+
   RemoveExtensionMapForMid(mid, header_extensions_by_mid_);
-  header_extensions_by_mid_.emplace_back(std::string(mid), header_extensions);
+  header_extensions_by_mid_.emplace_back(std::string(mid), extensions);
 
   RebuildMergedMap();
+  return RTCError::OK();
 }
 
 void RtpTransport::UnregisterRtpHeaderExtensionMap(absl::string_view mid) {
diff --git a/pc/rtp_transport.h b/pc/rtp_transport.h
index 4e1fd93ec3..1b0d65b96a 100644
--- a/pc/rtp_transport.h
+++ b/pc/rtp_transport.h
@@ -22,6 +22,7 @@
 
 #include "absl/strings/string_view.h"
 #include "api/field_trials_view.h"
+#include "api/rtc_error.h"
 #include "api/sequence_checker.h"
 #include "api/task_queue/pending_task_safety_flag.h"
 #include "api/transport/ecn_marking.h"
@@ -92,9 +93,12 @@ class RtpTransport : public RtpTransportInternal {
 
   bool IsSrtpActive() const override { return false; }
 
-  void RegisterRtpHeaderExtensionMap(
+  RTCError VerifyRtpHeaderExtensionMap(
+      const RtpHeaderExtensions& extensions) const override;
+
+  RTCError RegisterRtpHeaderExtensionMap(
       absl::string_view mid,
-      const RtpHeaderExtensions& header_extensions) override;
+      const RtpHeaderExtensions& extensions) override;
 
   // Currently only used for testing. In production, unregistration isn't needed
   // because leaving the registered extensions in `RtpTransport` is harmless
@@ -175,6 +179,9 @@ class RtpTransport : public RtpTransportInternal {
   std::vector<std::pair<std::string, RtpHeaderExtensions>>
       header_extensions_by_mid_ RTC_GUARDED_BY(network_thread_checker_);
 
+  RtpHeaderExtensions historical_rtp_header_extensions_
+      RTC_GUARDED_BY(network_thread_checker_);
+
   // Guard against recursive "ready to send" signals
   bool processing_ready_to_send_ = false;
   RTC_NO_UNIQUE_ADDRESS SequenceChecker network_thread_checker_;
diff --git a/pc/rtp_transport_internal.h b/pc/rtp_transport_internal.h
index a2c9967545..96aaa3ea3c 100644
--- a/pc/rtp_transport_internal.h
+++ b/pc/rtp_transport_internal.h
@@ -17,6 +17,7 @@
 
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
+#include "api/rtc_error.h"
 #include "api/task_queue/pending_task_safety_flag.h"
 #include "api/transport/ecn_marking.h"
 #include "api/units/timestamp.h"
@@ -134,9 +135,12 @@ class RtpTransportInternal {
   //   UpdateSendEncryptedHeaderExtensionIds,
   //   UpdateRecvEncryptedHeaderExtensionIds,
   //   CacheRtpAbsSendTimeHeaderExtension,
-  virtual void RegisterRtpHeaderExtensionMap(
+  virtual RTCError RegisterRtpHeaderExtensionMap(
       absl::string_view mid,
-      const RtpHeaderExtensions& header_extensions) = 0;
+      const RtpHeaderExtensions& extensions) = 0;
+
+  virtual RTCError VerifyRtpHeaderExtensionMap(
+      const RtpHeaderExtensions& extensions) const = 0;
 
   virtual void UnregisterRtpHeaderExtensionMap(absl::string_view mid) = 0;
 
diff --git a/pc/rtp_transport_unittest.cc b/pc/rtp_transport_unittest.cc
index 4c2aca61af..adbe70822a 100644
--- a/pc/rtp_transport_unittest.cc
+++ b/pc/rtp_transport_unittest.cc
@@ -14,6 +14,7 @@
 #include <cstdint>
 #include <optional>
 
+#include "api/rtc_error.h"
 #include "api/rtp_parameters.h"
 #include "api/test/rtc_error_matchers.h"
 #include "api/transport/ecn_marking.h"
@@ -44,6 +45,7 @@ namespace {
 
 using ::testing::Eq;
 using ::testing::Ge;
+using ::testing::HasSubstr;
 
 constexpr bool kMuxDisabled = false;
 constexpr bool kMuxEnabled = true;
@@ -365,6 +367,30 @@ TEST(RtpTransportTest, RegisterAndUnregisterRtpHeaderExtensionMap) {
   transport.UnregisterRtpDemuxerSink(&observer);
 }
 
+TEST(RtpTransportTest, VerifyRtpHeaderExtensionMapRejectsIdReassignment) {
+  test::RunLoop loop;
+  RtpTransport transport(kMuxDisabled, CreateTestFieldTrials());
+  RtpHeaderExtensions extensions1 = {
+      RtpExtension("urn:ietf:params:rtp-hdrext:ssrc-audio-level", 1)};
+  RtpHeaderExtensions extensions2 = {RtpExtension(
+      "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time", 1)};
+
+  // Registering the first map should succeed.
+  EXPECT_TRUE(
+      transport.RegisterRtpHeaderExtensionMap("audio", extensions1).ok());
+
+  // Verifying a map that tries to reassign ID 1 to a different URI should fail.
+  RTCError error = transport.VerifyRtpHeaderExtensionMap(extensions2);
+  EXPECT_FALSE(error.ok());
+  EXPECT_EQ(error.type(), RTCErrorType::INVALID_PARAMETER);
+  EXPECT_THAT(error.message(), HasSubstr("RTP extension ID reassignment"));
+
+  // Registering the second map should also fail.
+  error = transport.RegisterRtpHeaderExtensionMap("video", extensions2);
+  EXPECT_FALSE(error.ok());
+  EXPECT_EQ(error.type(), RTCErrorType::INVALID_PARAMETER);
+}
+
 // Test that SignalPacketReceived fires with rtcp=true when a RTCP packet is
 // received.
 TEST(RtpTransportTest, SignalDemuxedRtcp) {

From c95a4a938ac2a5fa3c93a956b6d03651f5fee7c8 Mon Sep 17 00:00:00 2001
From: Evan Shrubsole <eshr@webrtc.org>
Date: Tue, 17 Feb 2026 10:24:04 +0000
Subject: [PATCH 423/472] Add .gemini to .gitignore

This allows one to add workspace settings for Gemini in WebRTC.

Bug: webrtc:465491622
Change-Id: Ib120fcdad6b12aa542f984a169a9c0486a6a6964
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/460360
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47276}
---
 .gitignore | 1 +
 1 file changed, 1 insertion(+)

diff --git a/.gitignore b/.gitignore
index 42b16e532e..05d65d0076 100644
--- a/.gitignore
+++ b/.gitignore
@@ -34,6 +34,7 @@
 .cproject
 .gdb_history
 .gdbinit
+.gemini
 .landmines
 .metadata
 .project

From b83ca3e728dd0b55db5268276a011b978c004f2a Mon Sep 17 00:00:00 2001
From: Evan Shrubsole <eshr@webrtc.org>
Date: Wed, 25 Mar 2026 10:51:18 +0000
Subject: [PATCH 424/472] Create include-cleaner skill

Bug: webrtc:465491622
Change-Id: I277172143a77a6ea88102ae1a9668f476a6a6964
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/460020
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47277}
---
 .agents/skills/include-cleaner/SKILL.md | 67 +++++++++++++++++++++++++
 1 file changed, 67 insertions(+)
 create mode 100644 .agents/skills/include-cleaner/SKILL.md

diff --git a/.agents/skills/include-cleaner/SKILL.md b/.agents/skills/include-cleaner/SKILL.md
new file mode 100644
index 0000000000..d664a81e64
--- /dev/null
+++ b/.agents/skills/include-cleaner/SKILL.md
@@ -0,0 +1,67 @@
+---
+name: webrtc-include-cleaner
+description: Runs the WebRTC include-cleaner tool (IWYU replacement) to fix headers in C++ files. Use when preparing a CL for upload, after modifying .cc or .h files, or when instructed to fix include regressions.
+---
+
+# WebRTC Include Cleaner
+
+This skill provides instructions for using
+`tools_webrtc/iwyu/apply-include-cleaner`, a tool that automatically manages C++
+`#include` directives in the WebRTC codebase. It ensures that every header used
+is explicitly included and that unused headers are removed.
+
+## When to Use
+
+- **Pre-upload**: Run this tool before uploading a CL to ensure clean includes.
+- **After refactoring**: When moving code or changing dependencies, use this to
+  update `#include` blocks.
+- **Fixing regressions**: Use this if a presubmit or bot identifies
+  include-related issues.
+
+## Basic Usage
+
+To run the include cleaner on specific files:
+
+```bash
+tools_webrtc/iwyu/apply-include-cleaner path/to/file.cc path/to/file.h
+```
+
+To run it on all modified files relative to the upstream branch (ideal for CL
+preparation):
+
+```bash
+tools_webrtc/iwyu/apply-include-cleaner
+```
+
+Note: This is as expensive as a build for each file, so use it sparingly.
+
+## Options
+
+- `-p`, `--print`: Don't modify the files, just print the proposed changes.
+- `-w WORK_DIR`, `--work-dir WORK_DIR`: Specify the GN work directory (default:
+  `out/Default`).
+
+## Post-Execution Steps
+
+After running the include cleaner, it is recommended to perform the following
+steps to ensure build and style consistency:
+
+1. **Check for build errors**: The tool might occasionally make mistakes. Run a
+   build to verify.
+1. **Fix GN dependencies**: Use `tools_webrtc/gn_check_autofix.py` to fix any
+   `deps` issues caused by include changes.
+   ```bash
+   tools_webrtc/gn_check_autofix.py -C out/Default
+   ```
+1. **Format code**: Run `git cl format` to fix any formatting issues in the
+   `#include` blocks.
+   ```bash
+   git cl format
+   ```
+
+## Prerequisites
+
+- The tool automatically generates `compile_commands.json` in the output
+  directory if `out/Default` exists.
+- `clangd` must be checked out in your `.gclient` file
+  (`"checkout_clangd": True`).

From 0085c1c6c74d928284283d189589062267600b9e Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Fri, 27 Mar 2026 10:25:09 +0100
Subject: [PATCH 425/472] Rename methods in ScopedOperationsBatcher

* Rename `push_back` to `Add`
* Rename the previously overloaded push_back method to
  `AddWithFinalizer`.

This disambiguates the two functions to avoid running into conflicts, as
well as clarify the intent.

Bug: webrtc:42222804
Change-Id: Ic8471da8d62229d73ac40fbfae813da31d9d97dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/460520
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47278}
---
 pc/scoped_operations_batcher.cc          |  4 ++--
 pc/scoped_operations_batcher.h           |  5 +++--
 pc/scoped_operations_batcher_unittest.cc | 12 ++++++------
 pc/sdp_offer_answer.cc                   | 12 ++++++------
 4 files changed, 17 insertions(+), 16 deletions(-)

diff --git a/pc/scoped_operations_batcher.cc b/pc/scoped_operations_batcher.cc
index b5b277fea9..eea4157110 100644
--- a/pc/scoped_operations_batcher.cc
+++ b/pc/scoped_operations_batcher.cc
@@ -70,14 +70,14 @@ void ScopedOperationsBatcher::Run() {
   }
 }
 
-void ScopedOperationsBatcher::push_back(absl::AnyInvocable<void() &&> task) {
+void ScopedOperationsBatcher::Add(absl::AnyInvocable<void() &&> task) {
   RTC_DCHECK_RUN_ON(&sequence_checker_);
   if (task) {
     tasks_.emplace_back(std::move(task));
   }
 }
 
-void ScopedOperationsBatcher::push_back(
+void ScopedOperationsBatcher::AddWithFinalizer(
     absl::AnyInvocable<absl::AnyInvocable<void() &&>() &&> task) {
   RTC_DCHECK_RUN_ON(&sequence_checker_);
   if (task) {
diff --git a/pc/scoped_operations_batcher.h b/pc/scoped_operations_batcher.h
index f5c32c1a74..6a5f905701 100644
--- a/pc/scoped_operations_batcher.h
+++ b/pc/scoped_operations_batcher.h
@@ -44,8 +44,9 @@ class ScopedOperationsBatcher {
 
   // Queues non-nullptr tasks to be executed on the target thread when the
   // ScopedOperationsBatcher goes out of scope.
-  void push_back(absl::AnyInvocable<void() &&> task);
-  void push_back(absl::AnyInvocable<absl::AnyInvocable<void() &&>() &&> task);
+  void Add(absl::AnyInvocable<void() &&> task);
+  void AddWithFinalizer(
+      absl::AnyInvocable<absl::AnyInvocable<void() &&>() &&> task);
 
  private:
   using BatchedTask =
diff --git a/pc/scoped_operations_batcher_unittest.cc b/pc/scoped_operations_batcher_unittest.cc
index 03aa42d89d..d8699967a5 100644
--- a/pc/scoped_operations_batcher_unittest.cc
+++ b/pc/scoped_operations_batcher_unittest.cc
@@ -30,7 +30,7 @@ TEST(ScopedOperationsBatcherTest, ExecutesTasksOnTargetThread) {
 
   {
     ScopedOperationsBatcher batcher(target_thread.get());
-    batcher.push_back([&] {
+    batcher.Add([&] {
       task_executed = true;
       target_checked = target_thread->IsCurrent();
     });
@@ -66,7 +66,7 @@ TEST(ScopedOperationsBatcherTest, ExecutesReturnedTasksOnCallingThread) {
         return_task_thread = Thread::Current();
       };
     };
-    batcher.push_back(std::move(task));
+    batcher.AddWithFinalizer(std::move(task));
   }
 
   EXPECT_TRUE(task_executed);
@@ -83,15 +83,15 @@ TEST(ScopedOperationsBatcherTest, YieldsToOtherTasks) {
 
   {
     ScopedOperationsBatcher batcher(target_thread.get());
-    batcher.push_back([&] { execution_order.push_back(1); });
-    batcher.push_back([&] {
+    batcher.Add([&] { execution_order.push_back(1); });
+    batcher.Add([&] {
       execution_order.push_back(2);
       // Post a task that should interrupt the batch since we now yield to any
       // pending task.
       target_thread->PostTask([&] { execution_order.push_back(3); });
     });
-    batcher.push_back([&] { execution_order.push_back(4); });
-    batcher.push_back([&] { execution_order.push_back(5); });
+    batcher.Add([&] { execution_order.push_back(4); });
+    batcher.Add([&] { execution_order.push_back(5); });
   }
 
   // Expect the task (3) to execute immediately after the task
diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc
index e4e996fa48..5a3ce99ff3 100644
--- a/pc/sdp_offer_answer.cc
+++ b/pc/sdp_offer_answer.cc
@@ -2414,17 +2414,17 @@ void SdpOfferAnswerHandler::ApplyRemoteDescriptionUpdateTransceiverState(
     if (content->rejected && !transceiver->stopped()) {
       RTC_LOG(LS_INFO) << "Stopping transceiver for MID=" << content->mid()
                        << " since the media section was rejected.";
-      worker_tasks.push_back(transceiver->GetStopTransceiverProcedure());
+      worker_tasks.Add(transceiver->GetStopTransceiverProcedure());
     }
     if (!content->rejected && RtpTransceiverDirectionHasRecv(local_direction)) {
       if (!media_desc->streams().empty() &&
           media_desc->streams()[0].has_ssrcs()) {
         uint32_t ssrc = media_desc->streams()[0].first_ssrc();
-        worker_tasks.push_back(
+        worker_tasks.Add(
             transceiver->receiver_internal()->GetSetupForMediaChannel(ssrc));
       } else {
-        worker_tasks.push_back(transceiver->receiver_internal()
-                                   ->GetSetupForUnsignaledMediaChannel());
+        worker_tasks.Add(transceiver->receiver_internal()
+                             ->GetSetupForUnsignaledMediaChannel());
       }
     }
   }
@@ -3484,7 +3484,7 @@ RTCError SdpOfferAnswerHandler::Rollback(SdpType desc_type) {
       if (transceiver->internal()->reused_for_addtrack()) {
         transceiver->internal()->set_created_by_addtrack(true);
       } else {
-        worker_tasks.push_back(
+        worker_tasks.Add(
             transceiver->internal()->GetStopTransceiverProcedure());
         transceivers()->Remove(transceiver);
       }
@@ -4094,7 +4094,7 @@ RTCError SdpOfferAnswerHandler::UpdateTransceiversAndDataChannels(
           // is not already stopped, SDP munging has happened and we need to
           // ensure the transceiver is stopped.
           if (!transceiver->internal()->stopped()) {
-            worker_tasks.push_back(
+            worker_tasks.Add(
                 transceiver->internal()->GetStopTransceiverProcedure());
           }
           RTC_DCHECK(transceiver->internal()->stopped());

From c6ef49b61eff05fab303fdb451e6006838371ee2 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Fri, 27 Mar 2026 09:50:35 +0100
Subject: [PATCH 426/472] More spring cleaning

Remove entries from the watchlist that aren't associated with activity
for over 10 years as well as some expired addresses.

Most of these were added here: https://codereview.webrtc.org/1470833002/

No-try: true
Bug: none
Change-Id: Ib122604e762587d0c1ce55fdbe1df7ca2c13e1e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/460740
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47279}
---
 WATCHLISTS | 58 ++++++++++++++++++------------------------------------
 1 file changed, 19 insertions(+), 39 deletions(-)

diff --git a/WATCHLISTS b/WATCHLISTS
index dbb0f3c02b..7b38c2084f 100644
--- a/WATCHLISTS
+++ b/WATCHLISTS
@@ -102,61 +102,41 @@
   'WATCHLISTS': {
     'this_file': [],
     'all_webrtc': [],
-    'root_files': ['peah@webrtc.org',
-                   'qiang.lu@intel.com'],
+    'root_files': ['peah@webrtc.org'],
     'build_files': ['mbonadei@webrtc.org'],
-    'common_audio': ['audio-team@agora.io',
-                     'peah@webrtc.org',
+    'common_audio': ['peah@webrtc.org',
                      'saza@webrtc.org'],
     'audio': ['peah@webrtc.org'],
     'api': ['hta@webrtc.org',
             'peah@webrtc.org'],
     'base': ['hta@webrtc.org'],
     'call': ['stefan@webrtc.org'],
-    'video': ['stefan@webrtc.org',
-              'video-team@agora.io',
-              'zhengzhonghou@agora.io'],
-    'video_capture': ['perkj@webrtc.org',
-                      'sdk-team@agora.io',
-                      'zhengzhonghou@agora.io'],
-    'audio_device': ['audio-team@agora.io',
-                     'henrika@webrtc.org',
-                     'peah@webrtc.org',
-                     'saza@webrtc.org',
-                     'sdk-team@agora.io'],
-    'audio_coding': ['audio-team@agora.io',
-                     'henrik.lundin@webrtc.org',
+    'video': ['stefan@webrtc.org'],
+    'video_capture': ['perkj@webrtc.org'],
+    'audio_device': ['henrika@webrtc.org',
                      'peah@webrtc.org',
                      'saza@webrtc.org'],
-    'neteq': ['audio-team@agora.io',
-              'henrik.lundin@webrtc.org',
+    'audio_coding': ['henrik.lundin@webrtc.org',
+                     'peah@webrtc.org',
+                     'saza@webrtc.org'],
+    'neteq': ['henrik.lundin@webrtc.org',
               'saza@webrtc.org'],
-    'audio_mixer': ['aleloi@webrtc.org',
-                    'henrik.lundin@webrtc.org',
+    'audio_mixer': ['henrik.lundin@webrtc.org',
                     'peah@webrtc.org',
                     'saza@webrtc.org'],
-    'audio_processing': ['audio-team@agora.io',
-                         'henrik.lundin@webrtc.org',
+    'audio_processing': ['henrik.lundin@webrtc.org',
                          'peah@webrtc.org',
                          'saza@webrtc.org'],
-    'video_coding': ['stefan@webrtc.org',
-                     'video-team@agora.io',
-                     'zhengzhonghou@agora.io'],
-    'bitrate_controller': ['stefan@webrtc.org',
-                           'zhuangzesen@agora.io'],
+    'video_coding': ['stefan@webrtc.org'],
+    'bitrate_controller': ['stefan@webrtc.org'],
     'congestion_controller': [],
-    'remote_bitrate_estimator': ['stefan@webrtc.org',
-                                 'zhuangzesen@agora.io'],
-    'pacing': ['stefan@webrtc.org',
-               'zhuangzesen@agora.io'],
+    'remote_bitrate_estimator': ['stefan@webrtc.org'],
+    'pacing': ['stefan@webrtc.org'],
     'rtp_rtcp': ['stefan@webrtc.org',
-                 'danilchap@webrtc.org',
-                 'zhuangzesen@agora.io'],
-    'system_wrappers': ['fengyue@agora.io',
-                        'henrika@webrtc.org',
-                        'peah@webrtc.org',
-                        'zhengzhonghou@agora.io'],
-    'pc': ['steveanton+watch@webrtc.org'],
+                 'danilchap@webrtc.org'],
+    'system_wrappers': ['henrika@webrtc.org',
+                        'peah@webrtc.org'],
+    'pc': [],
     'logging': ['terelius@webrtc.org'],
   },
 }

From 2395c1f954c04f718c805d910f20d5b283fe3dc6 Mon Sep 17 00:00:00 2001
From: Evan Shrubsole <eshr@webrtc.org>
Date: Wed, 25 Mar 2026 10:51:18 +0000
Subject: [PATCH 427/472] Remove SIMULATED_WAIT

Bug: webrtc:381524905
Change-Id: I24c455d6656412a9ce2e16250564d9986a6a6964
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/460080
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47280}
---
 media/BUILD.gn                                |  1 -
 modules/audio_processing/aec3/BUILD.gn        |  1 -
 modules/audio_processing/agc2/BUILD.gn        |  8 ------
 .../capture_levels_adjuster/BUILD.gn          |  1 -
 .../audio_processing/capture_mixer/BUILD.gn   |  1 -
 modules/video_capture/BUILD.gn                |  1 -
 p2p/BUILD.gn                                  |  1 -
 rtc_base/BUILD.gn                             | 26 ------------------
 rtc_base/experiments/BUILD.gn                 |  1 -
 rtc_base/gunit.h                              | 27 -------------------
 rtc_base/win32_unittest.cc                    |  1 -
 rtc_tools/network_tester/BUILD.gn             |  1 -
 sdk/BUILD.gn                                  |  2 +-
 sdk/objc/unittests/RTCAudioSessionTest.mm     |  9 ++++---
 sdk/objc/unittests/RTCCertificateTest.mm      |  4 +--
 sdk/objc/unittests/RTCConfigurationTest.mm    |  4 +--
 .../RTCDataChannelConfigurationTest.mm        |  2 +-
 .../unittests/RTCFileVideoCapturer_xctest.mm  |  3 ++-
 sdk/objc/unittests/RTCIceCandidateTest.mm     |  4 +--
 sdk/objc/unittests/RTCIceServerTest.mm        |  4 +--
 sdk/objc/unittests/RTCMediaConstraintsTest.mm |  4 +--
 .../RTCPeerConnectionFactoryBuilderTest.mm    |  3 ++-
 sdk/objc/unittests/RTCPeerConnectionTest.mm   |  4 +--
 .../unittests/RTCSessionDescriptionTest.mm    |  3 ++-
 sdk/objc/unittests/RTCTracingTest.mm          |  4 +--
 sdk/objc/unittests/avformatmappertests.mm     |  2 +-
 sdk/objc/unittests/nalu_rewriter_xctest.mm    |  8 ++----
 .../objc_video_decoder_factory_tests.mm       |  3 ++-
 .../objc_video_encoder_factory_tests.mm       |  3 ++-
 stats/BUILD.gn                                |  1 -
 test/pc/e2e/BUILD.gn                          |  1 -
 31 files changed, 35 insertions(+), 103 deletions(-)
 delete mode 100644 rtc_base/gunit.h

diff --git a/media/BUILD.gn b/media/BUILD.gn
index d02dea6768..54d8b8b160 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -871,7 +871,6 @@ if (rtc_include_tests) {
       "../rtc_base:checks",
       "../rtc_base:copy_on_write_buffer",
       "../rtc_base:dscp",
-      "../rtc_base:gunit_helpers",
       "../rtc_base:logging",
       "../rtc_base:macromagic",
       "../rtc_base:network_route",
diff --git a/modules/audio_processing/aec3/BUILD.gn b/modules/audio_processing/aec3/BUILD.gn
index 6b5279b56a..d0e1c0d7a5 100644
--- a/modules/audio_processing/aec3/BUILD.gn
+++ b/modules/audio_processing/aec3/BUILD.gn
@@ -325,7 +325,6 @@ if (rtc_include_tests) {
       "../../../api/environment:environment_factory",
       "../../../rtc_base:checks",
       "../../../rtc_base:cpu_info",
-      "../../../rtc_base:gunit_helpers",
       "../../../rtc_base:random",
       "../../../rtc_base:safe_minmax",
       "../../../rtc_base:stringutils",
diff --git a/modules/audio_processing/agc2/BUILD.gn b/modules/audio_processing/agc2/BUILD.gn
index 92276ee60c..55e28a79ba 100644
--- a/modules/audio_processing/agc2/BUILD.gn
+++ b/modules/audio_processing/agc2/BUILD.gn
@@ -304,7 +304,6 @@ rtc_library("speech_level_estimator_unittest") {
     "..:apm_logging",
     "../../../api/audio:audio_processing",
     "../../../rtc_base:checks",
-    "../../../rtc_base:gunit_helpers",
     "../../../test:test_support",
   ]
 }
@@ -323,7 +322,6 @@ rtc_library("adaptive_digital_gain_controller_unittest") {
     "..:audio_frame_view",
     "../../../api/audio:audio_processing",
     "../../../common_audio",
-    "../../../rtc_base:gunit_helpers",
     "../../../test:test_support",
   ]
 }
@@ -337,7 +335,6 @@ rtc_library("gain_applier_unittest") {
     ":gain_applier",
     ":test_utils",
     "../../../api/audio:audio_frame_api",
-    "../../../rtc_base:gunit_helpers",
     "../../../test:test_support",
   ]
 }
@@ -354,7 +351,6 @@ rtc_library("saturation_protector_unittest") {
     ":common",
     ":saturation_protector",
     "..:apm_logging",
-    "../../../rtc_base:gunit_helpers",
     "../../../test:test_support",
   ]
 }
@@ -364,7 +360,6 @@ rtc_library("biquad_filter_unittests") {
   sources = [ "biquad_filter_unittest.cc" ]
   deps = [
     ":biquad_filter",
-    "../../../rtc_base:gunit_helpers",
     "../../../test:test_support",
   ]
 }
@@ -393,7 +388,6 @@ rtc_library("fixed_digital_unittests") {
     "../../../api/audio:audio_frame_api",
     "../../../common_audio",
     "../../../rtc_base:checks",
-    "../../../rtc_base:gunit_helpers",
     "../../../system_wrappers:metrics",
     "../../../test:test_support",
   ]
@@ -442,7 +436,6 @@ rtc_library("noise_estimator_unittests") {
     "../../../api:function_view",
     "../../../api/audio:audio_frame_api",
     "../../../rtc_base:checks",
-    "../../../rtc_base:gunit_helpers",
     "../../../test:test_support",
   ]
 }
@@ -455,7 +448,6 @@ rtc_library("vad_wrapper_unittests") {
     ":vad_wrapper",
     "../../../api/audio:audio_frame_api",
     "../../../rtc_base:checks",
-    "../../../rtc_base:gunit_helpers",
     "../../../rtc_base:safe_compare",
     "../../../test:test_support",
   ]
diff --git a/modules/audio_processing/capture_levels_adjuster/BUILD.gn b/modules/audio_processing/capture_levels_adjuster/BUILD.gn
index b8df947692..61292c7d6a 100644
--- a/modules/audio_processing/capture_levels_adjuster/BUILD.gn
+++ b/modules/audio_processing/capture_levels_adjuster/BUILD.gn
@@ -38,7 +38,6 @@ rtc_library("capture_levels_adjuster_unittests") {
     ":capture_levels_adjuster",
     "..:audio_buffer",
     "..:audioproc_test_utils",
-    "../../../rtc_base:gunit_helpers",
     "../../../rtc_base:stringutils",
     "../../../test:test_support",
   ]
diff --git a/modules/audio_processing/capture_mixer/BUILD.gn b/modules/audio_processing/capture_mixer/BUILD.gn
index 16dfba79e6..c7b2b4c6f4 100644
--- a/modules/audio_processing/capture_mixer/BUILD.gn
+++ b/modules/audio_processing/capture_mixer/BUILD.gn
@@ -48,7 +48,6 @@ rtc_library("capture_mixer_unittests") {
   deps = [
     ":capture_mixer",
     "..:audioproc_test_utils",
-    "../../../rtc_base:gunit_helpers",
     "../../../rtc_base:stringutils",
     "../../../test:test_support",
   ]
diff --git a/modules/video_capture/BUILD.gn b/modules/video_capture/BUILD.gn
index 741380d68e..ef4c4ff848 100644
--- a/modules/video_capture/BUILD.gn
+++ b/modules/video_capture/BUILD.gn
@@ -193,7 +193,6 @@ if (!build_with_chromium || is_linux || is_chromeos) {
         "../../api/video:video_rtp_headers",
         "../../common_video",
         "../../rtc_base:checks",
-        "../../rtc_base:gunit_helpers",
         "../../rtc_base:macromagic",
         "../../rtc_base:timeutils",
         "../../rtc_base/synchronization:mutex",
diff --git a/p2p/BUILD.gn b/p2p/BUILD.gn
index bc12178f5e..8ff23691b9 100644
--- a/p2p/BUILD.gn
+++ b/p2p/BUILD.gn
@@ -1195,7 +1195,6 @@ if (rtc_include_tests) {
       "../rtc_base:copy_on_write_buffer",
       "../rtc_base:crypto_random",
       "../rtc_base:dscp",
-      "../rtc_base:gunit_helpers",
       "../rtc_base:ip_address",
       "../rtc_base:logging",
       "../rtc_base:mdns_responder_interface",
diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
index 29888f043b..7f9af04c8c 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -751,7 +751,6 @@ if (rtc_include_tests) {
 
     sources = [ "task_queue_stdlib_unittest.cc" ]
     deps = [
-      ":gunit_helpers",
       ":logging",
       ":rtc_event",
       ":rtc_task_queue_stdlib",
@@ -1415,7 +1414,6 @@ if (rtc_include_tests) {
     sources = [ "async_packet_socket_unittest.cc" ]
     deps = [
       ":async_packet_socket",
-      ":gunit_helpers",
       ":socket",
       ":socket_address",
       "../test:test_support",
@@ -1429,7 +1427,6 @@ if (rtc_include_tests) {
     deps = [
       ":async_packet_socket",
       ":async_udp_socket",
-      ":gunit_helpers",
       ":mock_socket",
       ":rtc_base_tests_utils",
       ":socket",
@@ -1450,7 +1447,6 @@ if (rtc_include_tests) {
     deps = [
       ":async_packet_socket",
       ":async_tcp_socket",
-      ":gunit_helpers",
       ":net_helpers",
       ":rtc_base_tests_utils",
       ":socket",
@@ -1739,19 +1735,6 @@ rtc_source_set("gtest_prod") {
   sources = [ "gtest_prod_util.h" ]
 }
 
-rtc_library("gunit_helpers") {
-  testonly = true
-  sources = [ "gunit.h" ]
-  deps = [
-    ":logging",
-    ":rtc_base_tests_utils",
-    ":stringutils",
-    ":threading",
-    "../test:test_support",
-    "//third_party/abseil-cpp/absl/strings:string_view",
-  ]
-}
-
 rtc_library("testclient") {
   testonly = true
   sources = [
@@ -1762,7 +1745,6 @@ rtc_library("testclient") {
     ":async_packet_socket",
     ":async_udp_socket",
     ":buffer",
-    ":gunit_helpers",
     ":rtc_base_tests_utils",
     ":socket",
     ":socket_address",
@@ -1783,7 +1765,6 @@ rtc_library("callback_list_unittests") {
   sources = [ "callback_list_unittest.cc" ]
   deps = [
     ":callback_list",
-    ":gunit_helpers",
     ":untyped_function",
     "../api:function_view",
     "../test:test_support",
@@ -1995,7 +1976,6 @@ if (rtc_include_tests) {
     sources = [ "operations_chain_unittest.cc" ]
     deps = [
       ":checks",
-      ":gunit_helpers",
       ":rtc_event",
       ":rtc_operations_chain",
       ":threading",
@@ -2026,7 +2006,6 @@ if (rtc_include_tests) {
         ":buffer",
         ":checks",
         ":file_rotating_stream",
-        ":gunit_helpers",
         ":ip_address",
         ":logging",
         ":net_helpers",
@@ -2127,7 +2106,6 @@ if (rtc_include_tests) {
         ":divide_round",
         ":event_tracer",
         ":frequency_tracker",
-        ":gunit_helpers",
         ":histogram_percentile_counter",
         ":ip_address",
         ":logging",
@@ -2202,7 +2180,6 @@ if (rtc_include_tests) {
 
       sources = [ "task_queue_unittest.cc" ]
       deps = [
-        ":gunit_helpers",
         ":rtc_base_tests_utils",
         ":rtc_event",
         ":task_queue_for_test",
@@ -2219,7 +2196,6 @@ if (rtc_include_tests) {
 
       sources = [ "weak_ptr_unittest.cc" ]
       deps = [
-        ":gunit_helpers",
         ":rtc_base_tests_utils",
         ":rtc_event",
         ":task_queue_for_test",
@@ -2259,7 +2235,6 @@ if (rtc_include_tests) {
 
       sources = [ "strings/json_unittest.cc" ]
       deps = [
-        ":gunit_helpers",
         ":rtc_base_tests_utils",
         ":rtc_json",
         "../test:test_support",
@@ -2303,7 +2278,6 @@ if (rtc_include_tests) {
         ":data_rate_limiter",
         ":denormal_disabler",
         ":digest",
-        ":gunit_helpers",
         ":ifaddrs_converter",
         ":ip_address",
         ":logging",
diff --git a/rtc_base/experiments/BUILD.gn b/rtc_base/experiments/BUILD.gn
index 13601c302c..98d3025a04 100644
--- a/rtc_base/experiments/BUILD.gn
+++ b/rtc_base/experiments/BUILD.gn
@@ -235,7 +235,6 @@ if (rtc_include_tests && !build_with_chromium) {
       ":quality_scaler_settings",
       ":quality_scaling_experiment",
       ":rate_control_settings",
-      "..:gunit_helpers",
       "../../api:field_trials",
       "../../api:field_trials_view",
       "../../api/units:data_rate",
diff --git a/rtc_base/gunit.h b/rtc_base/gunit.h
deleted file mode 100644
index cf61fabf1a..0000000000
--- a/rtc_base/gunit.h
+++ /dev/null
@@ -1,27 +0,0 @@
-/*
- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef RTC_BASE_GUNIT_H_
-#define RTC_BASE_GUNIT_H_
-
-// TODO(bugs.webrtc.org/42226242): remove transitive includes
-#include "rtc_base/thread.h"  // IWYU pragma: keep
-#include "test/gtest.h"       // IWYU pragma: keep
-
-// Wait until "ex" is true, or "timeout" expires, using fake clock where
-// messages are processed every millisecond.
-// TODO(pthatcher): Allow tests to control how many milliseconds to advance.
-#define SIMULATED_WAIT(ex, timeout, clock)                          \
-  for (int64_t wait_start = ::webrtc::TimeMillis();                 \
-       !(ex) && ::webrtc::TimeMillis() < wait_start + (timeout);) { \
-    (clock).AdvanceTime(webrtc::TimeDelta::Millis(1));              \
-  }
-
-#endif  // RTC_BASE_GUNIT_H_
diff --git a/rtc_base/win32_unittest.cc b/rtc_base/win32_unittest.cc
index 902f91532a..863376f766 100644
--- a/rtc_base/win32_unittest.cc
+++ b/rtc_base/win32_unittest.cc
@@ -12,7 +12,6 @@
 
 #include <string>
 
-#include "rtc_base/gunit.h"
 #include "rtc_base/ip_address.h"
 #include "rtc_base/net_helpers.h"
 #include "test/gtest.h"
diff --git a/rtc_tools/network_tester/BUILD.gn b/rtc_tools/network_tester/BUILD.gn
index fd98be799b..f39654771e 100644
--- a/rtc_tools/network_tester/BUILD.gn
+++ b/rtc_tools/network_tester/BUILD.gn
@@ -90,7 +90,6 @@ if (rtc_enable_protobuf) {
       "../../api:rtc_error_matchers",
       "../../api/environment",
       "../../api/environment:environment_factory",
-      "../../rtc_base:gunit_helpers",
       "../../rtc_base:random",
       "../../rtc_base:threading",
       "../../rtc_base:timeutils",
diff --git a/sdk/BUILD.gn b/sdk/BUILD.gn
index 065df50da8..5252eb9193 100644
--- a/sdk/BUILD.gn
+++ b/sdk/BUILD.gn
@@ -1236,9 +1236,9 @@ if (is_ios || is_mac) {
             "../media:rtc_media_base",
             "../media:rtc_media_tests_utils",
             "../modules/video_coding:video_codec_interface",
-            "../rtc_base:gunit_helpers",
             "../rtc_base:refcount",
             "../rtc_base:rtc_event",
+            "../rtc_base:threading",
             "../rtc_base/system:unused",
             "../system_wrappers",
             "../test:create_test_environment",
diff --git a/sdk/objc/unittests/RTCAudioSessionTest.mm b/sdk/objc/unittests/RTCAudioSessionTest.mm
index 03f7b6942b..a8c0679f8e 100644
--- a/sdk/objc/unittests/RTCAudioSessionTest.mm
+++ b/sdk/objc/unittests/RTCAudioSessionTest.mm
@@ -12,10 +12,12 @@
 #import <OCMock/OCMock.h>
 #import <XCTest/XCTest.h>
 
+#include "rtc_base/thread.h"
+#include "test/gtest.h"
+
 #include <vector>
 
 #include "rtc_base/event.h"
-#include "rtc_base/gunit.h"
 
 #import "components/audio/RTCAudioSession+Private.h"
 
@@ -25,8 +27,9 @@
 @interface RTC_OBJC_TYPE (RTCAudioSession)
 (UnitTesting)
 
-    @property(nonatomic, readonly) std::vector<
-        __weak id<RTC_OBJC_TYPE(RTCAudioSessionDelegate)> > delegates;
+    @property(nonatomic, readonly)
+        std::vector<__weak id<RTC_OBJC_TYPE(RTCAudioSessionDelegate)> >
+            delegates;
 
 - (instancetype)initWithAudioSession:(id)audioSession;
 
diff --git a/sdk/objc/unittests/RTCCertificateTest.mm b/sdk/objc/unittests/RTCCertificateTest.mm
index 64c68e0e45..3af2f6ab18 100644
--- a/sdk/objc/unittests/RTCCertificateTest.mm
+++ b/sdk/objc/unittests/RTCCertificateTest.mm
@@ -11,9 +11,9 @@
 #import <Foundation/Foundation.h>
 #import <XCTest/XCTest.h>
 
-#include <vector>
+#include "test/gtest.h"
 
-#include "rtc_base/gunit.h"
+#include <vector>
 
 #import "api/peerconnection/RTCConfiguration+Private.h"
 #import "api/peerconnection/RTCConfiguration.h"
diff --git a/sdk/objc/unittests/RTCConfigurationTest.mm b/sdk/objc/unittests/RTCConfigurationTest.mm
index 5c8fca0383..2716b1ed91 100644
--- a/sdk/objc/unittests/RTCConfigurationTest.mm
+++ b/sdk/objc/unittests/RTCConfigurationTest.mm
@@ -11,9 +11,9 @@
 #import <Foundation/Foundation.h>
 #import <XCTest/XCTest.h>
 
-#include <vector>
+#include "test/gtest.h"
 
-#include "rtc_base/gunit.h"
+#include <vector>
 
 #import "api/peerconnection/RTCConfiguration+Private.h"
 #import "api/peerconnection/RTCConfiguration.h"
diff --git a/sdk/objc/unittests/RTCDataChannelConfigurationTest.mm b/sdk/objc/unittests/RTCDataChannelConfigurationTest.mm
index ccebd74198..e2a0ab0e7c 100644
--- a/sdk/objc/unittests/RTCDataChannelConfigurationTest.mm
+++ b/sdk/objc/unittests/RTCDataChannelConfigurationTest.mm
@@ -11,7 +11,7 @@
 #import <Foundation/Foundation.h>
 #import <XCTest/XCTest.h>
 
-#include "rtc_base/gunit.h"
+#include "test/gtest.h"
 
 #import "api/peerconnection/RTCDataChannelConfiguration+Private.h"
 #import "api/peerconnection/RTCDataChannelConfiguration.h"
diff --git a/sdk/objc/unittests/RTCFileVideoCapturer_xctest.mm b/sdk/objc/unittests/RTCFileVideoCapturer_xctest.mm
index 9e5f227913..e4a6847236 100644
--- a/sdk/objc/unittests/RTCFileVideoCapturer_xctest.mm
+++ b/sdk/objc/unittests/RTCFileVideoCapturer_xctest.mm
@@ -12,8 +12,9 @@
 
 #import <XCTest/XCTest.h>
 
+#include "test/gtest.h"
+
 #include "api/test/rtc_error_matchers.h"
-#include "rtc_base/gunit.h"
 #include "test/gmock.h"
 #include "test/wait_until.h"
 
diff --git a/sdk/objc/unittests/RTCIceCandidateTest.mm b/sdk/objc/unittests/RTCIceCandidateTest.mm
index 268d330815..b89a55de6d 100644
--- a/sdk/objc/unittests/RTCIceCandidateTest.mm
+++ b/sdk/objc/unittests/RTCIceCandidateTest.mm
@@ -11,9 +11,9 @@
 #import <Foundation/Foundation.h>
 #import <XCTest/XCTest.h>
 
-#include <memory>
+#include "test/gtest.h"
 
-#include "rtc_base/gunit.h"
+#include <memory>
 
 #import "api/peerconnection/RTCIceCandidate+Private.h"
 #import "api/peerconnection/RTCIceCandidate.h"
diff --git a/sdk/objc/unittests/RTCIceServerTest.mm b/sdk/objc/unittests/RTCIceServerTest.mm
index 29163961da..0d1919d8bd 100644
--- a/sdk/objc/unittests/RTCIceServerTest.mm
+++ b/sdk/objc/unittests/RTCIceServerTest.mm
@@ -11,9 +11,9 @@
 #import <Foundation/Foundation.h>
 #import <XCTest/XCTest.h>
 
-#include <vector>
+#include "test/gtest.h"
 
-#include "rtc_base/gunit.h"
+#include <vector>
 
 #import "api/peerconnection/RTCIceServer+Private.h"
 #import "api/peerconnection/RTCIceServer.h"
diff --git a/sdk/objc/unittests/RTCMediaConstraintsTest.mm b/sdk/objc/unittests/RTCMediaConstraintsTest.mm
index 60d68f3e31..7b67147d39 100644
--- a/sdk/objc/unittests/RTCMediaConstraintsTest.mm
+++ b/sdk/objc/unittests/RTCMediaConstraintsTest.mm
@@ -11,9 +11,9 @@
 #import <Foundation/Foundation.h>
 #import <XCTest/XCTest.h>
 
-#include <memory>
+#include "test/gtest.h"
 
-#include "rtc_base/gunit.h"
+#include <memory>
 
 #import "api/peerconnection/RTCMediaConstraints+Private.h"
 #import "api/peerconnection/RTCMediaConstraints.h"
diff --git a/sdk/objc/unittests/RTCPeerConnectionFactoryBuilderTest.mm b/sdk/objc/unittests/RTCPeerConnectionFactoryBuilderTest.mm
index 115b5e200e..15eb6ca746 100644
--- a/sdk/objc/unittests/RTCPeerConnectionFactoryBuilderTest.mm
+++ b/sdk/objc/unittests/RTCPeerConnectionFactoryBuilderTest.mm
@@ -10,6 +10,8 @@
 
 #import <Foundation/Foundation.h>
 #import <XCTest/XCTest.h>
+
+#include "test/gtest.h"
 #ifdef __cplusplus
 extern "C" {
 #endif
@@ -28,7 +30,6 @@ extern "C" {
 #include "api/video_codecs/video_decoder_factory.h"
 #include "api/video_codecs/video_encoder_factory.h"
 
-#include "rtc_base/gunit.h"
 #include "rtc_base/system/unused.h"
 
 @interface RTCPeerConnectionFactoryBuilderTests : XCTestCase
diff --git a/sdk/objc/unittests/RTCPeerConnectionTest.mm b/sdk/objc/unittests/RTCPeerConnectionTest.mm
index e2c05f90f7..57a089418f 100644
--- a/sdk/objc/unittests/RTCPeerConnectionTest.mm
+++ b/sdk/objc/unittests/RTCPeerConnectionTest.mm
@@ -11,11 +11,11 @@
 #import <Foundation/Foundation.h>
 #import <XCTest/XCTest.h>
 
+#include "test/gtest.h"
+
 #include <memory>
 #include <vector>
 
-#include "rtc_base/gunit.h"
-
 #import "api/peerconnection/RTCConfiguration+Private.h"
 #import "api/peerconnection/RTCConfiguration.h"
 #import "api/peerconnection/RTCCryptoOptions.h"
diff --git a/sdk/objc/unittests/RTCSessionDescriptionTest.mm b/sdk/objc/unittests/RTCSessionDescriptionTest.mm
index d64abce614..790fcd7bda 100644
--- a/sdk/objc/unittests/RTCSessionDescriptionTest.mm
+++ b/sdk/objc/unittests/RTCSessionDescriptionTest.mm
@@ -11,8 +11,9 @@
 #import <Foundation/Foundation.h>
 #import <XCTest/XCTest.h>
 
+#include "test/gtest.h"
+
 #include "api/jsep.h"
-#include "rtc_base/gunit.h"
 
 #import "api/peerconnection/RTCSessionDescription+Private.h"
 #import "api/peerconnection/RTCSessionDescription.h"
diff --git a/sdk/objc/unittests/RTCTracingTest.mm b/sdk/objc/unittests/RTCTracingTest.mm
index 63008a79d1..1d613ea962 100644
--- a/sdk/objc/unittests/RTCTracingTest.mm
+++ b/sdk/objc/unittests/RTCTracingTest.mm
@@ -11,9 +11,9 @@
 #import <Foundation/Foundation.h>
 #import <XCTest/XCTest.h>
 
-#include <vector>
+#include "test/gtest.h"
 
-#include "rtc_base/gunit.h"
+#include <vector>
 
 #import "api/peerconnection/RTCTracing.h"
 #import "helpers/NSString+StdString.h"
diff --git a/sdk/objc/unittests/avformatmappertests.mm b/sdk/objc/unittests/avformatmappertests.mm
index 06aefb4351..2d8d2d3640 100644
--- a/sdk/objc/unittests/avformatmappertests.mm
+++ b/sdk/objc/unittests/avformatmappertests.mm
@@ -11,7 +11,7 @@
 #import <Foundation/Foundation.h>
 #import <OCMock/OCMock.h>
 
-#include "rtc_base/gunit.h"
+#include "test/gtest.h"
 
 // Width and height don't play any role so lets use predefined values throughout
 // the tests.
diff --git a/sdk/objc/unittests/nalu_rewriter_xctest.mm b/sdk/objc/unittests/nalu_rewriter_xctest.mm
index 82260a51e1..ac98010a91 100644
--- a/sdk/objc/unittests/nalu_rewriter_xctest.mm
+++ b/sdk/objc/unittests/nalu_rewriter_xctest.mm
@@ -12,7 +12,6 @@
 
 #include "common_video/h264/h264_common.h"
 #include "components/video_codec/nalu_rewriter.h"
-#include "rtc_base/gunit.h"
 
 #import <XCTest/XCTest.h>
 
@@ -212,11 +211,8 @@ static const uint8_t SPS_PPS_BUFFER[] = {
   CMSampleBufferRef out_sample_buffer = nil;
   CMVideoFormatDescriptionRef description = [self createDescription];
 
-  Boolean result =
-      webrtc::H264AnnexBBufferToCMSampleBuffer(annex_b_test_data,
-                                               description,
-                                               &out_sample_buffer,
-                                               memory_pool);
+  Boolean result = webrtc::H264AnnexBBufferToCMSampleBuffer(
+      annex_b_test_data, description, &out_sample_buffer, memory_pool);
 
   XCTAssertTrue(result);
 
diff --git a/sdk/objc/unittests/objc_video_decoder_factory_tests.mm b/sdk/objc/unittests/objc_video_decoder_factory_tests.mm
index 3a8748e102..e3008a1139 100644
--- a/sdk/objc/unittests/objc_video_decoder_factory_tests.mm
+++ b/sdk/objc/unittests/objc_video_decoder_factory_tests.mm
@@ -12,6 +12,8 @@
 #import <OCMock/OCMock.h>
 #import <XCTest/XCTest.h>
 
+#include "test/gtest.h"
+
 #include "sdk/objc/native/src/objc_video_decoder_factory.h"
 
 #include "api/environment/environment_factory.h"
@@ -20,7 +22,6 @@
 #include "media/base/codec.h"
 #include "modules/video_coding/include/video_codec_interface.h"
 #include "modules/video_coding/include/video_error_codes.h"
-#include "rtc_base/gunit.h"
 #import "sdk/objc/base/RTCMacros.h"
 
 id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)> CreateDecoderFactoryReturning(
diff --git a/sdk/objc/unittests/objc_video_encoder_factory_tests.mm b/sdk/objc/unittests/objc_video_encoder_factory_tests.mm
index 12b9c99cb8..4c0c435ac3 100644
--- a/sdk/objc/unittests/objc_video_encoder_factory_tests.mm
+++ b/sdk/objc/unittests/objc_video_encoder_factory_tests.mm
@@ -12,6 +12,8 @@
 #import <OCMock/OCMock.h>
 #import <XCTest/XCTest.h>
 
+#include "test/gtest.h"
+
 #include "sdk/objc/native/src/objc_video_encoder_factory.h"
 
 #include "api/environment/environment_factory.h"
@@ -23,7 +25,6 @@
 #import "components/video_frame_buffer/RTCCVPixelBuffer.h"
 #include "modules/video_coding/include/video_codec_interface.h"
 #include "modules/video_coding/include/video_error_codes.h"
-#include "rtc_base/gunit.h"
 #include "sdk/objc/native/src/objc_frame_buffer.h"
 
 id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)> CreateEncoderFactoryReturning(
diff --git a/stats/BUILD.gn b/stats/BUILD.gn
index a523cb8e35..5626cc5aa6 100644
--- a/stats/BUILD.gn
+++ b/stats/BUILD.gn
@@ -63,7 +63,6 @@ if (rtc_include_tests && !build_with_chromium) {
       "../api:scoped_refptr",
       "../api/units:timestamp",
       "../rtc_base:checks",
-      "../rtc_base:gunit_helpers",
       "../rtc_base:rtc_json",
       "../test:test_support",
     ]
diff --git a/test/pc/e2e/BUILD.gn b/test/pc/e2e/BUILD.gn
index 058a6941db..20b8524eeb 100644
--- a/test/pc/e2e/BUILD.gn
+++ b/test/pc/e2e/BUILD.gn
@@ -348,7 +348,6 @@ if (!build_with_chromium) {
         "../../../modules/audio_device:audio_device_impl",
         "../../../pc:pc_test_utils",
         "../../../pc:peerconnection_wrapper",
-        "../../../rtc_base:gunit_helpers",
         "../../../rtc_base:logging",
         "../../../rtc_base:rtc_event",
         "../../../test:fileutils",

From 53987851c66c473620be403e788f105cdc9d51cb Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Fri, 27 Mar 2026 10:33:08 +0000
Subject: [PATCH 428/472] Convert kMax constants in video_codec_constants to
 size_t

There was a pattern of having to cast these to size_t when comparing
them to sizes of things. Since these are mostly used for sizing
arrays and lists, it seems logical to make them size_t.

Bug: None
Change-Id: I77e0642dfc0937e6367034af10b27f1ba95b495b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/459400
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47281}
---
 api/video/video_bitrate_allocation.cc                |  2 +-
 api/video/video_codec_constants.h                    | 12 +++++++-----
 call/rtp_payload_params_unittest.cc                  |  4 +++-
 media/BUILD.gn                                       |  3 +++
 media/base/media_engine.cc                           |  5 +++--
 media/engine/simulcast_encoder_adapter.cc            |  3 ++-
 modules/rtp_rtcp/source/rtcp_sender.cc               |  4 ++--
 modules/rtp_rtcp/source/rtp_sender_video_unittest.cc |  4 ++--
 modules/video_coding/BUILD.gn                        |  3 +++
 .../codecs/vp8/default_temporal_layers_unittest.cc   |  2 +-
 modules/video_coding/rtp_generic_ref_finder.cc       |  3 ++-
 sdk/android/src/jni/video_encoder_wrapper.cc         |  5 +++--
 video/send_statistics_proxy.cc                       |  6 +++---
 video/video_stream_encoder.cc                        |  6 +++---
 video/video_stream_encoder_unittest.cc               |  2 +-
 15 files changed, 39 insertions(+), 25 deletions(-)

diff --git a/api/video/video_bitrate_allocation.cc b/api/video/video_bitrate_allocation.cc
index 6e47777621..98735cb096 100644
--- a/api/video/video_bitrate_allocation.cc
+++ b/api/video/video_bitrate_allocation.cc
@@ -119,7 +119,7 @@ VideoBitrateAllocation::GetSimulcastAllocations() const {
     std::optional<VideoBitrateAllocation> layer_bitrate;
     if (IsSpatialLayerUsed(si)) {
       layer_bitrate = VideoBitrateAllocation();
-      for (int tl = 0; tl < kMaxTemporalStreams; ++tl) {
+      for (size_t tl = 0; tl < kMaxTemporalStreams; ++tl) {
         if (HasBitrate(si, tl))
           layer_bitrate->SetBitrate(0, tl, GetBitrate(si, tl));
       }
diff --git a/api/video/video_codec_constants.h b/api/video/video_codec_constants.h
index 5859f9b4cf..0f84968879 100644
--- a/api/video/video_codec_constants.h
+++ b/api/video/video_codec_constants.h
@@ -11,13 +11,15 @@
 #ifndef API_VIDEO_VIDEO_CODEC_CONSTANTS_H_
 #define API_VIDEO_VIDEO_CODEC_CONSTANTS_H_
 
+#include <cstddef>
+
 namespace webrtc {
 
-enum : int { kMaxEncoderBuffers = 8 };
-enum : int { kMaxSimulcastStreams = 3 };
-enum : int { kMaxSpatialLayers = 5 };
-enum : int { kMaxTemporalStreams = 4 };
-enum : int { kMaxPreferredPixelFormats = 5 };
+inline constexpr size_t kMaxEncoderBuffers = 8;
+inline constexpr size_t kMaxSimulcastStreams = 3;
+inline constexpr size_t kMaxSpatialLayers = 5;
+inline constexpr size_t kMaxTemporalStreams = 4;
+inline constexpr size_t kMaxPreferredPixelFormats = 5;
 
 }  // namespace webrtc
 
diff --git a/call/rtp_payload_params_unittest.cc b/call/rtp_payload_params_unittest.cc
index 304acbb530..18a3cdfa86 100644
--- a/call/rtp_payload_params_unittest.cc
+++ b/call/rtp_payload_params_unittest.cc
@@ -10,6 +10,7 @@
 
 #include "call/rtp_payload_params.h"
 
+#include <cstddef>
 #include <cstdint>
 #include <map>
 #include <optional>
@@ -1012,7 +1013,8 @@ TEST(RtpPayloadParamsVp9ToGenericTest, SpatialScalabilityKSvc) {
   headers[3] = params.GetRtpVideoHeader(image, &info, /*shared_frame_id=*/7);
 
   ASSERT_TRUE(headers[0].generic);
-  int num_decode_targets = headers[0].generic->decode_target_indications.size();
+  size_t num_decode_targets =
+      headers[0].generic->decode_target_indications.size();
   // Rely on implementation detail there are always kMaxTemporalStreams temporal
   // layers assumed, in particular assume Decode Target#0 matches layer S0T0,
   // and Decode Target#kMaxTemporalStreams matches layer S1T0.
diff --git a/media/BUILD.gn b/media/BUILD.gn
index 54d8b8b160..e292762430 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -254,6 +254,8 @@ rtc_library("media_engine") {
     "../api/video_codecs:scalability_mode_helper",
     "../call:call_interfaces",
     "../rtc_base:checks",
+    "../rtc_base:safe_compare",
+    "../rtc_base:safe_conversions",
     "../rtc_base:stringutils",
     "../rtc_base/system:file_wrapper",
     "//third_party/abseil-cpp/absl/algorithm:container",
@@ -504,6 +506,7 @@ rtc_library("rtc_simulcast_encoder_adapter") {
     "../rtc_base:checks",
     "../rtc_base:logging",
     "../rtc_base:macromagic",
+    "../rtc_base:safe_conversions",
     "../rtc_base:stringutils",
     "../rtc_base/experiments:encoder_info_settings",
     "../rtc_base/experiments:rate_control_settings",
diff --git a/media/base/media_engine.cc b/media/base/media_engine.cc
index 73a87101c0..27fbdbc55f 100644
--- a/media/base/media_engine.cc
+++ b/media/base/media_engine.cc
@@ -33,6 +33,7 @@
 #include "media/base/rid_description.h"
 #include "media/base/stream_params.h"
 #include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_compare.h"
 
 namespace webrtc {
 namespace {
@@ -194,8 +195,8 @@ RTCError CheckRtpParametersValues(const RtpParameters& rtp_parameters,
     }
     if (rtp_parameters.encodings[i].num_temporal_layers) {
       if (*rtp_parameters.encodings[i].num_temporal_layers < 1 ||
-          *rtp_parameters.encodings[i].num_temporal_layers >
-              kMaxTemporalStreams) {
+          SafeGt(*rtp_parameters.encodings[i].num_temporal_layers,
+                 kMaxTemporalStreams)) {
         return LOG_ERROR(RTCError(RTCErrorType::INVALID_RANGE)
                          << "Attempted to set RtpParameters "
                             "num_temporal_layers to an invalid number.");
diff --git a/media/engine/simulcast_encoder_adapter.cc b/media/engine/simulcast_encoder_adapter.cc
index fbed6ef915..06aebdc685 100644
--- a/media/engine/simulcast_encoder_adapter.cc
+++ b/media/engine/simulcast_encoder_adapter.cc
@@ -59,6 +59,7 @@
 #include "rtc_base/checks.h"
 #include "rtc_base/experiments/rate_control_settings.h"
 #include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
 #include "rtc_base/strings/str_join.h"
 #include "rtc_base/strings/string_builder.h"
 #include "rtc_base/synchronization/mutex.h"
@@ -769,7 +770,7 @@ void SimulcastEncoderAdapter::SetRates(
     // the encoder handling the current simulcast stream.
     RateControlParameters stream_parameters = parameters;
     stream_parameters.bitrate = VideoBitrateAllocation();
-    for (int i = 0; i < kMaxTemporalStreams; ++i) {
+    for (int i = 0; checked_cast<size_t>(i) < kMaxTemporalStreams; ++i) {
       if (parameters.bitrate.HasBitrate(stream_idx, i)) {
         stream_parameters.bitrate.SetBitrate(
             0, i, parameters.bitrate.GetBitrate(stream_idx, i));
diff --git a/modules/rtp_rtcp/source/rtcp_sender.cc b/modules/rtp_rtcp/source/rtcp_sender.cc
index c26d3659a7..9b9bd22262 100644
--- a/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -552,8 +552,8 @@ void RTCPSender::BuildExtendedReports(const RtcpContext& ctx,
   if (send_video_bitrate_allocation_) {
     rtcp::TargetBitrate target_bitrate;
 
-    for (int sl = 0; sl < kMaxSpatialLayers; ++sl) {
-      for (int tl = 0; tl < kMaxTemporalStreams; ++tl) {
+    for (size_t sl = 0; sl < kMaxSpatialLayers; ++sl) {
+      for (size_t tl = 0; tl < kMaxTemporalStreams; ++tl) {
         if (video_bitrate_allocation_.HasBitrate(sl, tl)) {
           target_bitrate.AddTargetBitrate(
               sl, tl, video_bitrate_allocation_.GetBitrate(sl, tl) / 1000);
diff --git a/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
index fb8c98faf4..443b608836 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
@@ -467,7 +467,7 @@ TEST_F(RtpSenderVideoTest, RetransmissionTypesVP8HigherLayers) {
   header.codec = kVideoCodecVP8;
 
   auto& vp8_header = header.video_type_header.emplace<RTPVideoHeaderVP8>();
-  for (int tid = 1; tid <= kMaxTemporalStreams; ++tid) {
+  for (size_t tid = 1; tid <= kMaxTemporalStreams; ++tid) {
     vp8_header.temporalIdx = tid;
 
     EXPECT_FALSE(rtp_sender_video_->AllowRetransmission(
@@ -487,7 +487,7 @@ TEST_F(RtpSenderVideoTest, RetransmissionTypesVP9) {
   header.codec = kVideoCodecVP9;
 
   auto& vp9_header = header.video_type_header.emplace<RTPVideoHeaderVP9>();
-  for (int tid = 1; tid <= kMaxTemporalStreams; ++tid) {
+  for (size_t tid = 1; tid <= kMaxTemporalStreams; ++tid) {
     vp9_header.temporal_idx = tid;
 
     EXPECT_FALSE(rtp_sender_video_->AllowRetransmission(
diff --git a/modules/video_coding/BUILD.gn b/modules/video_coding/BUILD.gn
index 0ca6954084..029ddb6066 100644
--- a/modules/video_coding/BUILD.gn
+++ b/modules/video_coding/BUILD.gn
@@ -1180,6 +1180,9 @@ if (rtc_include_tests) {
       "../../rtc_base:random",
       "../../rtc_base:rtc_base_tests_utils",
       "../../rtc_base:rtc_event",
+      "../../rtc_base:rtc_numerics",
+      "../../rtc_base:safe_conversions",
+      "../../rtc_base:stringutils",
       "../../rtc_base:task_queue_for_test",
       "../../rtc_base:threading",
       "../../rtc_base/experiments:encoder_info_settings",
diff --git a/modules/video_coding/codecs/vp8/default_temporal_layers_unittest.cc b/modules/video_coding/codecs/vp8/default_temporal_layers_unittest.cc
index b50db65f96..b65be28140 100644
--- a/modules/video_coding/codecs/vp8/default_temporal_layers_unittest.cc
+++ b/modules/video_coding/codecs/vp8/default_temporal_layers_unittest.cc
@@ -441,7 +441,7 @@ class TemporalLayersReferenceTest : public TemporalLayersTest,
 
 INSTANTIATE_TEST_SUITE_P(DefaultTemporalLayersTest,
                          TemporalLayersReferenceTest,
-                         ::testing::Range(1, kMaxTemporalStreams + 1));
+                         ::testing::Range(1, int{kMaxTemporalStreams} + 1));
 
 TEST_P(TemporalLayersReferenceTest, ValidFrameConfigs) {
   const int num_layers = GetParam();
diff --git a/modules/video_coding/rtp_generic_ref_finder.cc b/modules/video_coding/rtp_generic_ref_finder.cc
index 85dcaf0832..42268ee4b8 100644
--- a/modules/video_coding/rtp_generic_ref_finder.cc
+++ b/modules/video_coding/rtp_generic_ref_finder.cc
@@ -21,6 +21,7 @@
 #include "modules/video_coding/codecs/interface/common_constants.h"
 #include "modules/video_coding/rtp_frame_reference_finder.h"
 #include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
 
 namespace webrtc {
 
@@ -28,7 +29,7 @@ RtpFrameReferenceFinder::ReturnVector RtpGenericFrameRefFinder::ManageFrame(
     std::unique_ptr<RtpFrameObject> frame,
     const RTPVideoHeader::GenericDescriptorInfo& descriptor) {
   RtpFrameReferenceFinder::ReturnVector res;
-  if (descriptor.spatial_index >= kMaxSpatialLayers) {
+  if (checked_cast<size_t>(descriptor.spatial_index) >= kMaxSpatialLayers) {
     RTC_LOG(LS_WARNING) << "Spatial index " << descriptor.spatial_index
                         << " is unsupported.";
     return res;
diff --git a/sdk/android/src/jni/video_encoder_wrapper.cc b/sdk/android/src/jni/video_encoder_wrapper.cc
index 93570b0304..023921bb2b 100644
--- a/sdk/android/src/jni/video_encoder_wrapper.cc
+++ b/sdk/android/src/jni/video_encoder_wrapper.cc
@@ -458,9 +458,10 @@ ScopedJavaLocalRef<jobject> VideoEncoderWrapper::ToJavaBitrateAllocation(
       ScopedJavaLocalRef<jobjectArray>::Adopt(
           jni, jni->NewObjectArray(kMaxSpatialLayers, int_array_class_.obj(),
                                    nullptr /* initial */));
-  for (int spatial_i = 0; spatial_i < kMaxSpatialLayers; ++spatial_i) {
+  for (size_t spatial_i = 0; spatial_i < kMaxSpatialLayers; ++spatial_i) {
     std::array<int32_t, kMaxTemporalStreams> spatial_layer;
-    for (int temporal_i = 0; temporal_i < kMaxTemporalStreams; ++temporal_i) {
+    for (size_t temporal_i = 0; temporal_i < kMaxTemporalStreams;
+         ++temporal_i) {
       spatial_layer[temporal_i] = allocation.GetBitrate(spatial_i, temporal_i);
     }
 
diff --git a/video/send_statistics_proxy.cc b/video/send_statistics_proxy.cc
index 4fc364bcc1..abb746da2d 100644
--- a/video/send_statistics_proxy.cc
+++ b/video/send_statistics_proxy.cc
@@ -1378,20 +1378,20 @@ void SendStatisticsProxy::OnBitrateAllocationUpdated(
     const VideoCodec& codec,
     const VideoBitrateAllocation& allocation) {
   int num_spatial_layers = 0;
-  for (int i = 0; i < kMaxSpatialLayers; i++) {
+  for (size_t i = 0; i < kMaxSpatialLayers; i++) {
     if (codec.spatialLayers[i].active) {
       num_spatial_layers++;
     }
   }
   int num_simulcast_streams = 0;
-  for (int i = 0; i < kMaxSimulcastStreams; i++) {
+  for (size_t i = 0; i < kMaxSimulcastStreams; i++) {
     if (codec.simulcastStream[i].active) {
       num_simulcast_streams++;
     }
   }
 
   std::array<bool, kMaxSpatialLayers> spatial_layers;
-  for (int i = 0; i < kMaxSpatialLayers; i++) {
+  for (size_t i = 0; i < kMaxSpatialLayers; i++) {
     spatial_layers[i] = (allocation.GetSpatialLayerSum(i) > 0);
   }
 
diff --git a/video/video_stream_encoder.cc b/video/video_stream_encoder.cc
index 177ae421a6..4faf80cec7 100644
--- a/video/video_stream_encoder.cc
+++ b/video/video_stream_encoder.cc
@@ -264,14 +264,14 @@ VideoBitrateAllocation UpdateAllocationFromEncoderInfo(
     return allocation;
   }
   VideoBitrateAllocation new_allocation;
-  for (int si = 0; si < kMaxSpatialLayers; ++si) {
+  for (size_t si = 0; si < kMaxSpatialLayers; ++si) {
     if (encoder_info.fps_allocation[si].size() == 1 &&
         allocation.IsSpatialLayerUsed(si)) {
       // One TL is signalled to be used by the encoder. Do not distribute
       // bitrate allocation across TLs (use sum at ti:0).
       new_allocation.SetBitrate(si, 0, allocation.GetSpatialLayerSum(si));
     } else {
-      for (int ti = 0; ti < kMaxTemporalStreams; ++ti) {
+      for (size_t ti = 0; ti < kMaxTemporalStreams; ++ti) {
         if (allocation.HasBitrate(si, ti))
           new_allocation.SetBitrate(si, ti, allocation.GetBitrate(si, ti));
       }
@@ -351,7 +351,7 @@ VideoLayersAllocation CreateVideoLayersAllocation(
 
     std::vector<DataRate> aggregated_spatial_bitrate(kMaxTemporalStreams,
                                                      DataRate::Zero());
-    for (int si = 0; si < kMaxSpatialLayers; ++si) {
+    for (size_t si = 0; si < kMaxSpatialLayers; ++si) {
       layers_allocation.resolution_and_frame_rate_is_valid = true;
       if (!target_bitrate.IsSpatialLayerUsed(si) ||
           target_bitrate.GetSpatialLayerSum(si) == 0) {
diff --git a/video/video_stream_encoder_unittest.cc b/video/video_stream_encoder_unittest.cc
index 41c91a79ff..64468c3448 100644
--- a/video/video_stream_encoder_unittest.cc
+++ b/video/video_stream_encoder_unittest.cc
@@ -1139,7 +1139,7 @@ class VideoStreamEncoderTest : public ::testing::Test {
               kQpLow, kQpHigh, kMinPixelsPerFrame);
         }
         info.is_hardware_accelerated = is_hardware_accelerated_;
-        for (int i = 0; i < kMaxSpatialLayers; ++i) {
+        for (size_t i = 0; i < kMaxSpatialLayers; ++i) {
           if (temporal_layers_supported_[i]) {
             info.fps_allocation[i].clear();
             int num_layers = temporal_layers_supported_[i].value() ? 2 : 1;

From 190a7806e991036b008ee88ea63955cd49fa54c9 Mon Sep 17 00:00:00 2001
From: Philipp Hancke <philipp.hancke@googlemail.com>
Date: Thu, 26 Mar 2026 07:42:11 +0100
Subject: [PATCH 429/472] p2ptc: clean up unused legacy API

after the remoting usage was cleaned up in
  https://chromium-review.googlesource.com/c/chromium/src/+/7698373
  https://chromium-review.googlesource.com/c/chromium/src/+/7702494

Bug: None
Change-Id: I7337615cce120487f5d684b4cec29b755d5d4fe6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/460240
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#47282}
---
 p2p/base/p2p_transport_channel.h | 7 -------
 1 file changed, 7 deletions(-)

diff --git a/p2p/base/p2p_transport_channel.h b/p2p/base/p2p_transport_channel.h
index 7339920c41..92f8dfe2bd 100644
--- a/p2p/base/p2p_transport_channel.h
+++ b/p2p/base/p2p_transport_channel.h
@@ -168,13 +168,6 @@ class RTC_EXPORT P2PTransportChannel : public IceTransportInternal,
   bool PruneConnections(
       std::span<const Connection* const> connections) override;
 
-  // TODO(honghaiz): Remove this method once the reference of it in
-  // Chromoting is removed.
-  const Connection* best_connection() const {
-    RTC_DCHECK_RUN_ON(&network_thread_);
-    return selected_connection_;
-  }
-
   void set_incoming_only(bool value) {
     RTC_DCHECK_RUN_ON(&network_thread_);
     incoming_only_ = value;

From fd2e7ea6206c28d5a567b7bf7e95d77b515851c8 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Fri, 27 Mar 2026 09:43:10 +0100
Subject: [PATCH 430/472] Fix potential dcSCTP NackBetweenAckBlocks OOB access

Add strict GapAckBlock boundary validation in
RetransmissionQueue::IsSackValid to drop malformed SACK chunks that
acknowledge TSNs beyond the currently outstanding sequence numbers.
Additionally, adds bounds checking within NackBetweenAckBlocks to
prevent Out-Of-Bounds iterating.

Bug: webrtc:495700476
Change-Id: I6ca4011cd074ef3777ff972488dad79227e6d45b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/460720
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47283}
---
 net/dcsctp/tx/outstanding_data.cc      |  3 ++-
 net/dcsctp/tx/outstanding_data_test.cc | 15 +++++++++++++++
 net/dcsctp/tx/retransmission_queue.cc  |  8 ++++++++
 3 files changed, 25 insertions(+), 1 deletion(-)

diff --git a/net/dcsctp/tx/outstanding_data.cc b/net/dcsctp/tx/outstanding_data.cc
index 89f1ea2a77..b996dbdbd4 100644
--- a/net/dcsctp/tx/outstanding_data.cc
+++ b/net/dcsctp/tx/outstanding_data.cc
@@ -256,7 +256,8 @@ void OutstandingData::NackBetweenAckBlocks(
     UnwrappedTSN cur_block_first_acked =
         UnwrappedTSN::AddTo(cumulative_tsn_ack, block.start);
     for (UnwrappedTSN tsn = prev_block_last_acked.next_value();
-         tsn < cur_block_first_acked && tsn <= max_tsn_to_nack;
+         tsn < cur_block_first_acked && tsn <= max_tsn_to_nack &&
+         tsn < next_tsn();
          tsn = tsn.next_value()) {
       ack_info.has_packet_loss |=
           NackItem(tsn, /*retransmit_now=*/false,
diff --git a/net/dcsctp/tx/outstanding_data_test.cc b/net/dcsctp/tx/outstanding_data_test.cc
index f56f1d5499..7818d9b08d 100644
--- a/net/dcsctp/tx/outstanding_data_test.cc
+++ b/net/dcsctp/tx/outstanding_data_test.cc
@@ -732,5 +732,20 @@ TEST_F(OutstandingDataTest,
                           Pair(TSN(16), State::kAcked)));
 }
 
+TEST_F(OutstandingDataTest, NackBetweenAckBlocksDoesNotAccessOutOfBounds) {
+  for (int i = 0; i < 5; ++i) {
+    buf_.Insert(kMessageId, gen_.Ordered({1}, ""), kNow);
+  }
+
+  // Inject a malformed SACK where the GapAckBlock exceeds the number of
+  // outstanding items, potentially triggering an OOB read/write.
+  std::vector<SackChunk::GapAckBlock> malformed_blocks = {
+      SackChunk::GapAckBlock(1, 40000)};
+
+  // This should not crash or trigger ASAN errors.
+  buf_.HandleSack(unwrapper_.Unwrap(TSN(10)), malformed_blocks,
+                  /*is_in_fast_recovery=*/false);
+}
+
 }  // namespace
 }  // namespace dcsctp
diff --git a/net/dcsctp/tx/retransmission_queue.cc b/net/dcsctp/tx/retransmission_queue.cc
index 3dc2bc7134..2cfc6ef087 100644
--- a/net/dcsctp/tx/retransmission_queue.cc
+++ b/net/dcsctp/tx/retransmission_queue.cc
@@ -250,6 +250,14 @@ bool RetransmissionQueue::IsSackValid(const SackChunk& sack) const {
   } else if (cumulative_tsn_ack > outstanding_data_.highest_outstanding_tsn()) {
     return false;
   }
+
+  for (const auto& block : sack.gap_ack_blocks()) {
+    if (UnwrappedTSN::AddTo(cumulative_tsn_ack, block.end) >
+        outstanding_data_.highest_outstanding_tsn()) {
+      return false;
+    }
+  }
+
   return true;
 }
 

From fe2da40be567c59ad25bf5517e64352e9ef5e7ea Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Fri, 27 Mar 2026 11:17:02 +0100
Subject: [PATCH 431/472] Audio: Reduce max channels to 16 to prevent buffer
 overflow
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Lowers kMaxNumberOfAudioChannels from 24 to 16 to fit within the statically allocated AudioFrame buffer (7680 max samples) when resampling at 48kHz (7680 = 16 channels * 480 samples). Also implies safe channel capping in resampler to prevent buffer overflows.

Bug: webrtc:495018167
Change-Id: I60b30832b34508deff036a3a63fcabac1a35c889
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/460680
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47284}
---
 api/audio/audio_view.h                        |  7 +++---
 modules/audio_coding/acm2/acm_resampler.cc    |  4 +++
 .../acm2/acm_resampler_unittest.cc            | 25 +++++++++++--------
 .../builtin_audio_decoder_factory_unittest.cc |  4 +--
 .../audio_mixer/frame_combiner_unittest.cc    | 13 +++++-----
 5 files changed, 29 insertions(+), 24 deletions(-)

diff --git a/api/audio/audio_view.h b/api/audio/audio_view.h
index b35c95e5a5..f3fcf8d918 100644
--- a/api/audio/audio_view.h
+++ b/api/audio/audio_view.h
@@ -49,10 +49,9 @@ using MonoView = ArrayView<T>;
 
 // The maximum number of audio channels supported by WebRTC encoders, decoders
 // and the AudioFrame class.
-// TODO(peah, tommi): Should kMaxNumberOfAudioChannels be 16 rather than 24?
-// The reason is that AudioFrame's max number of samples is 7680, which can
-// hold 16 10ms 16bit channels at 48 kHz (and not 24 channels).
-static constexpr size_t kMaxNumberOfAudioChannels = 24;
+// AudioFrame's max number of samples is 7680, which can hold 16 10ms 16bit
+// channels at 48 kHz.
+static constexpr size_t kMaxNumberOfAudioChannels = 16;
 
 // InterleavedView<> is a view over an interleaved audio buffer (e.g. from
 // AudioFrame).
diff --git a/modules/audio_coding/acm2/acm_resampler.cc b/modules/audio_coding/acm2/acm_resampler.cc
index e9c2861101..ddefdbe8f1 100644
--- a/modules/audio_coding/acm2/acm_resampler.cc
+++ b/modules/audio_coding/acm2/acm_resampler.cc
@@ -17,6 +17,7 @@
 #include "absl/algorithm/container.h"
 #include "api/audio/audio_frame.h"
 #include "api/audio/audio_view.h"
+#include "api/audio/channel_layout.h"
 #include "audio/utility/audio_frame_operations.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
@@ -49,6 +50,9 @@ bool ResamplerHelper::MaybeResample(int desired_sample_rate_hz,
       RTC_LOG(LS_ERROR) << "AudioFrame cannot hold resampled data.";
       AudioFrameOperations::Mute(audio_frame);
       audio_frame->SetSampleRateAndChannelSize(desired_sample_rate_hz);
+      audio_frame->SetLayoutAndNumChannels(
+          CHANNEL_LAYOUT_UNSUPPORTED,
+          AudioFrame::kMaxDataSizeSamples / audio_frame->samples_per_channel());
       return false;
     }
   }
diff --git a/modules/audio_coding/acm2/acm_resampler_unittest.cc b/modules/audio_coding/acm2/acm_resampler_unittest.cc
index f59715150b..49713a97d4 100644
--- a/modules/audio_coding/acm2/acm_resampler_unittest.cc
+++ b/modules/audio_coding/acm2/acm_resampler_unittest.cc
@@ -24,30 +24,33 @@ TEST(ResamplerHelperTest, MaybeResampleCheckForMaxSize) {
   ResamplerHelper resampler;
   AudioFrame audio_frame;
 
-  // Create an audio frame that requires resampling from 32kHz to 48kHz
-  // with a very high number of channels (24).
-  const int kCurrentSampleRateHz = 32000;
-  const int kDesiredSampleRateHz = 48000;
-  const size_t kChannels = 24;
+  // Create an audio frame that requires resampling from 48kHz to 96kHz
+  // with a high number of channels (16) to exceed the buffer size.
+  const int kCurrentSampleRateHz = 48000;
+  const int kDesiredSampleRateHz = 96000;
+  const size_t kChannels = 16;
 
-  // 10 ms of data at 32kHz = 320 samples per channel.
-  std::vector<int16_t> dummy_data(320 * 24, 0);
-  audio_frame.UpdateFrame(0, dummy_data.data(), 320, kCurrentSampleRateHz,
+  // 10 ms of data at 48kHz = 480 samples per channel.
+  std::vector<int16_t> dummy_data(480 * 16, 0);
+  audio_frame.UpdateFrame(0, dummy_data.data(), 480, kCurrentSampleRateHz,
                           AudioFrame::kNormalSpeech, AudioFrame::kVadActive,
                           kChannels);
 
   // The resampler prime path will attempt to allocate a buffer that is
-  // kChannels * (kDesiredSampleRateHz / 100) = 24 * 480 = 11520 samples,
+  // kChannels * (kDesiredSampleRateHz / 100) = 16 * 960 = 15360 samples,
   // which exceeds AudioFrame::kMaxDataSizeSamples (7680).
   const bool resample_success =
       resampler.MaybeResample(kDesiredSampleRateHz, &audio_frame);
 
   // Verify that MaybeResample correctly detects the buffer size condition and
-  // safely aborts the operation by returning false and muting the frame.
+  // safely aborts the operation by returning false, muting the frame, and
+  // capping the channel count to avoid a buffer overflow in the muted data
+  // array.
   EXPECT_FALSE(resample_success);
   EXPECT_TRUE(audio_frame.muted());
   EXPECT_EQ(audio_frame.sample_rate_hz_, kDesiredSampleRateHz);
-  EXPECT_EQ(audio_frame.num_channels_, kChannels);
+  EXPECT_EQ(audio_frame.num_channels_, AudioFrame::kMaxDataSizeSamples /
+                                           audio_frame.samples_per_channel());
 }
 
 TEST(ResamplerHelperTest, MaybeResampleValidMaxSize) {
diff --git a/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc b/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc
index fe3adbd823..0d678de98b 100644
--- a/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc
+++ b/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc
@@ -64,9 +64,9 @@ TEST(AudioDecoderFactoryTest, CreateL16) {
   const Environment env = CreateEnvironment();
   scoped_refptr<AudioDecoderFactory> adf = CreateBuiltinAudioDecoderFactory();
   ASSERT_TRUE(adf);
-  // L16 supports any clock rate and any number of channels up to 24.
+  // L16 supports any clock rate and any number of channels up to 16.
   const int clockrates[] = {8000, 16000, 32000, 48000};
-  const int num_channels[] = {1, 2, 3, 24};
+  const int num_channels[] = {1, 2, 3, 16};
   for (int clockrate : clockrates) {
     EXPECT_FALSE(
         adf->Create(env, SdpAudioFormat("l16", clockrate, 0), std::nullopt));
diff --git a/modules/audio_mixer/frame_combiner_unittest.cc b/modules/audio_mixer/frame_combiner_unittest.cc
index df01237e07..ae440462b4 100644
--- a/modules/audio_mixer/frame_combiner_unittest.cc
+++ b/modules/audio_mixer/frame_combiner_unittest.cc
@@ -19,7 +19,6 @@
 
 #include "api/array_view.h"
 #include "api/audio/audio_frame.h"
-#include "api/audio/audio_view.h"
 #include "api/audio/channel_layout.h"
 #include "api/rtp_packet_info.h"
 #include "api/rtp_packet_infos.h"
@@ -146,15 +145,12 @@ TEST(FrameCombiner, ContainsAllRtpPacketInfos) {
 TEST(FrameCombinerDeathTest, BuildCrashesWithManyChannels) {
   FrameCombiner combiner(true);
   for (const int rate : {8000, 18000, 34000, 48000}) {
-    for (const int number_of_channels : {10, 20, 21}) {
-      RTC_DCHECK_LE(number_of_channels, kMaxNumberOfAudioChannels);
+    for (const int number_of_channels : {10, 15, 17}) {
       if (static_cast<size_t>(rate / 100 * number_of_channels) >
           AudioFrame::kMaxDataSizeSamples) {
         continue;
       }
       const std::vector<AudioFrame*> all_frames = {&frame1, &frame2};
-      SetUpFrames(rate, number_of_channels);
-
       const int number_of_frames = 2;
       SCOPED_TRACE(
           ProduceDebugText(rate, number_of_channels, number_of_frames));
@@ -162,8 +158,11 @@ TEST(FrameCombinerDeathTest, BuildCrashesWithManyChannels) {
           all_frames.begin(), all_frames.begin() + number_of_frames);
       AudioFrame audio_frame_for_mixing;
       EXPECT_DEATH(
-          combiner.Combine(frames_to_combine, number_of_channels, rate,
-                           frames_to_combine.size(), &audio_frame_for_mixing),
+          {
+            SetUpFrames(rate, number_of_channels);
+            combiner.Combine(frames_to_combine, number_of_channels, rate,
+                             frames_to_combine.size(), &audio_frame_for_mixing);
+          },
           "");
     }
   }

From 39fcc21366cf7e523429f8919d5657164c19b36d Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Fri, 27 Mar 2026 10:59:37 +0100
Subject: [PATCH 432/472] Replace ArrayView with std::span in modules/

ArrayView is an alias to std::span. This change switch to use
std::span directly instead of through the alias.

Search&Replace MakeArrayView and ArrayView with std::span
Search&Replace include "api/array_view.h" with include <span>
Remove <span> include where std::span is not mentioned in the file
Remove build dependencies on array_view target

Bug: webrtc:439801349
Change-Id: I55a4978c265a0b8b6e873db93bbbf2241bd3e066
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/460800
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47285}
---
 modules/audio_coding/BUILD.gn                 | 27 ---------------
 modules/audio_coding/acm2/acm_remixing.cc     |  4 +--
 modules/audio_coding/acm2/acm_remixing.h      |  4 +--
 .../audio_coding/acm2/audio_coding_module.cc  |  4 +--
 .../acm2/audio_coding_module_unittest.cc      |  8 ++---
 .../controller_manager.cc                     |  8 ++---
 .../controller_manager.h                      |  4 +--
 .../frame_length_controller_v2.cc             |  5 +--
 .../frame_length_controller_v2.h              |  4 +--
 .../builtin_audio_encoder_factory_unittest.cc |  4 +--
 .../codecs/cng/audio_encoder_cng.cc           | 16 ++++-----
 .../codecs/cng/audio_encoder_cng_unittest.cc  |  4 +--
 .../audio_coding/codecs/cng/cng_unittest.cc   | 34 +++++++++----------
 modules/audio_coding/codecs/cng/webrtc_cng.cc | 10 +++---
 modules/audio_coding/codecs/cng/webrtc_cng.h  |  8 ++---
 .../codecs/g711/audio_encoder_pcm.cc          |  6 ++--
 .../codecs/g711/audio_encoder_pcm.h           |  4 +--
 .../codecs/g722/audio_encoder_g722.cc         |  6 ++--
 .../codecs/g722/audio_encoder_g722.h          |  4 +--
 .../codecs/legacy_encoded_audio_frame.cc      |  4 +--
 .../codecs/legacy_encoded_audio_frame.h       |  4 +--
 .../legacy_encoded_audio_frame_unittest.cc    |  4 +--
 .../codecs/opus/audio_coder_opus_common.h     |  4 +--
 .../codecs/opus/audio_decoder_opus.cc         |  4 +--
 .../opus/audio_decoder_opus_unittest.cc       | 16 ++++-----
 .../audio_encoder_multi_channel_opus_impl.cc  |  6 ++--
 .../audio_encoder_multi_channel_opus_impl.h   |  4 +--
 .../codecs/opus/audio_encoder_opus.cc         |  6 ++--
 .../codecs/opus/audio_encoder_opus.h          |  6 ++--
 .../opus/audio_encoder_opus_unittest.cc       |  8 ++---
 .../audio_coding/codecs/opus/opus_unittest.cc |  6 ++--
 .../codecs/red/audio_encoder_copy_red.cc      | 10 +++---
 .../codecs/red/audio_encoder_copy_red.h       |  6 ++--
 .../red/audio_encoder_copy_red_unittest.cc    |  4 +--
 modules/audio_coding/neteq/accelerate.cc      | 12 +++----
 .../neteq/audio_decoder_unittest.cc           |  6 ++--
 .../audio_coding/neteq/audio_multi_vector.cc  |  4 +--
 .../audio_coding/neteq/audio_multi_vector.h   |  4 +--
 .../audio_coding/neteq/background_noise.cc    |  6 ++--
 modules/audio_coding/neteq/background_noise.h |  7 ++--
 modules/audio_coding/neteq/comfort_noise.cc   |  4 +--
 modules/audio_coding/neteq/merge.cc           |  4 +--
 .../neteq/neteq_decoder_plc_unittest.cc       |  6 ++--
 modules/audio_coding/neteq/neteq_impl.cc      |  6 ++--
 modules/audio_coding/neteq/neteq_impl.h       |  6 ++--
 .../neteq/neteq_network_stats_unittest.cc     |  4 +--
 .../neteq/neteq_stereo_unittest.cc            |  6 ++--
 modules/audio_coding/neteq/neteq_unittest.cc  | 12 +++----
 modules/audio_coding/neteq/normal.cc          |  4 +--
 .../neteq/packet_buffer_unittest.cc           |  4 +--
 .../audio_coding/neteq/preemptive_expand.cc   | 12 +++----
 .../neteq/test/neteq_decoding_test.cc         |  6 ++--
 .../neteq/test/neteq_opus_quality_test.cc     |  4 +--
 .../neteq/test/neteq_pcm16b_quality_test.cc   |  4 +--
 .../neteq/test/neteq_pcmu_quality_test.cc     |  4 +--
 .../audio_coding/neteq/tools/audio_checksum.h |  4 +--
 .../audio_coding/neteq/tools/audio_loop.cc    |  8 ++---
 modules/audio_coding/neteq/tools/audio_loop.h |  4 +--
 .../neteq/tools/encode_neteq_input.h          |  4 +--
 .../neteq/tools/fake_decode_from_file.cc      |  6 ++--
 .../neteq/tools/fake_decode_from_file.h       |  4 +--
 .../neteq/tools/neteq_event_log_input.cc      |  4 +--
 .../neteq/tools/neteq_quality_test.cc         |  4 +--
 .../audio_coding/neteq/tools/neteq_test.cc    |  1 -
 .../audio_coding/neteq/tools/rtp_jitter.cc    |  6 ++--
 modules/audio_coding/test/Channel.cc          |  4 +--
 modules/audio_coding/test/EncodeDecodeTest.cc |  4 +--
 modules/audio_coding/test/PacketLossTest.cc   |  4 +--
 modules/audio_coding/test/TestAllCodecs.cc    |  4 +--
 modules/audio_coding/test/TestStereo.cc       |  4 +--
 .../test/target_delay_unittest.cc             |  4 +--
 modules/audio_device/BUILD.gn                 |  5 ---
 modules/audio_device/audio_device_unittest.cc | 23 ++++++-------
 modules/audio_device/fine_audio_buffer.cc     |  8 ++---
 modules/audio_device/fine_audio_buffer.h      |  8 ++---
 .../fine_audio_buffer_unittest.cc             |  6 ++--
 .../audio_device/include/test_audio_device.cc | 20 +++++------
 .../audio_device/include/test_audio_device.h  |  4 +--
 .../include/test_audio_device_unittest.cc     | 10 +++---
 .../audio_device/test_audio_device_impl.cc    |  4 +--
 .../audio_device/win/core_audio_input_win.cc  |  6 ++--
 .../audio_device/win/core_audio_output_win.cc |  6 ++--
 modules/audio_mixer/BUILD.gn                  |  4 ---
 modules/audio_mixer/audio_mixer_impl.cc       |  8 ++---
 modules/audio_mixer/audio_mixer_impl.h        |  4 +--
 .../audio_mixer/audio_mixer_impl_unittest.cc  |  6 ++--
 .../default_output_rate_calculator.cc         |  4 +--
 .../default_output_rate_calculator.h          |  5 +--
 modules/audio_mixer/frame_combiner.cc         | 10 +++---
 modules/audio_mixer/frame_combiner.h          |  4 +--
 .../audio_mixer/frame_combiner_unittest.cc    |  6 ++--
 modules/audio_mixer/gain_change_calculator.cc | 14 ++++----
 modules/audio_mixer/gain_change_calculator.h  | 14 ++++----
 modules/audio_mixer/output_rate_calculator.h  |  4 +--
 .../audio_processing/agc2/rnn_vad/BUILD.gn    |  1 -
 .../congestion_controller/goog_cc/BUILD.gn    |  1 -
 .../goog_cc/loss_based_bwe_v2.cc              |  6 ++--
 .../goog_cc/loss_based_bwe_v2.h               |  6 ++--
 modules/congestion_controller/rtp/BUILD.gn    |  1 -
 .../transport_feedback_adapter_unittest.cc    | 14 ++++----
 modules/desktop_capture/BUILD.gn              |  2 --
 .../desktop_capture/desktop_frame_unittest.cc |  4 +--
 .../screen_capturer_integration_test.cc       |  4 +--
 modules/pacing/BUILD.gn                       |  2 --
 modules/pacing/pacing_controller.cc           |  4 +--
 modules/pacing/pacing_controller.h            |  6 ++--
 modules/pacing/pacing_controller_unittest.cc  | 10 +++---
 modules/pacing/packet_router.cc               |  4 +--
 modules/pacing/packet_router.h                |  4 +--
 modules/rtp_rtcp/source/rtp_packet.h          |  2 +-
 modules/video_coding/BUILD.gn                 |  8 -----
 modules/video_coding/codecs/av1/BUILD.gn      |  1 -
 .../codecs/av1/dav1d_decoder_unittest.cc      |  4 +--
 .../test/videocodec_test_fixture_impl.cc      |  1 -
 .../codecs/vp9/libvpx_vp9_decoder.cc          |  4 +--
 .../codecs/vp9/libvpx_vp9_encoder.cc          |  4 +--
 .../codecs/vp9/test/vp9_impl_unittest.cc      |  4 +--
 .../frame_dependencies_calculator.cc          |  4 +--
 .../frame_dependencies_calculator.h           |  4 +--
 .../video_coding/generic_decoder_unittest.cc  |  4 +--
 modules/video_coding/h264_sps_pps_tracker.cc  |  8 ++---
 modules/video_coding/h264_sps_pps_tracker.h   |  4 +--
 .../h264_sps_pps_tracker_unittest.cc          |  4 +--
 modules/video_coding/h26x_packet_buffer.cc    |  8 ++---
 .../h26x_packet_buffer_unittest.cc            |  4 +--
 .../loss_notification_controller.cc           |  6 ++--
 .../loss_notification_controller.h            |  8 ++---
 .../video_coding/packet_buffer_unittest.cc    | 10 +++---
 .../rtp_vp8_ref_finder_unittest.cc            |  4 +--
 .../rtp_vp9_ref_finder_unittest.cc            |  4 +--
 modules/video_coding/svc/BUILD.gn             |  1 -
 .../scalability_structure_key_svc_unittest.cc |  4 +--
 ...ility_structure_l2t2_key_shift_unittest.cc |  5 ++-
 .../svc/scalability_structure_test_helpers.cc |  4 +--
 .../svc/scalability_structure_test_helpers.h  |  4 +--
 .../svc/scalability_structure_unittest.cc     | 16 ++++-----
 modules/video_coding/utility/qp_parser.cc     |  6 ++--
 .../utility/vp9_uncompressed_header_parser.cc |  6 ++--
 .../utility/vp9_uncompressed_header_parser.h  |  4 +--
 139 files changed, 397 insertions(+), 455 deletions(-)

diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index d8e1e9788a..0aeff4e301 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -39,7 +39,6 @@ rtc_library("audio_coding") {
     ":neteq",
     "..:module_api",
     "..:module_api_public",
-    "../../api:array_view",
     "../../api:function_view",
     "../../api/audio:audio_frame_api",
     "../../api/audio_codecs:audio_codecs_api",
@@ -70,7 +69,6 @@ rtc_library("legacy_encoded_audio_frame") {
     "codecs/legacy_encoded_audio_frame.h",
   ]
   deps = [
-    "../../api:array_view",
     "../../api/audio_codecs:audio_codecs_api",
     "../../rtc_base:buffer",
     "../../rtc_base:checks",
@@ -85,7 +83,6 @@ rtc_library("webrtc_cng") {
   ]
 
   deps = [
-    "../../api:array_view",
     "../../common_audio:common_audio_c",
     "../../rtc_base:buffer",
     "../../rtc_base:checks",
@@ -102,7 +99,6 @@ rtc_library("audio_encoder_cng") {
 
   deps = [
     ":webrtc_cng",
-    "../../api:array_view",
     "../../api/audio_codecs:audio_codecs_api",
     "../../api/units:time_delta",
     "../../common_audio",
@@ -119,7 +115,6 @@ rtc_library("red") {
   ]
 
   deps = [
-    "../../api:array_view",
     "../../api:bitrate_allocation",
     "../../api:field_trials_view",
     "../../api/audio_codecs:audio_codecs_api",
@@ -145,7 +140,6 @@ rtc_library("g711") {
 
   deps = [
     ":legacy_encoded_audio_frame",
-    "../../api:array_view",
     "../../api/audio_codecs:audio_codecs_api",
     "../../api/units:time_delta",
     "../../rtc_base:buffer",
@@ -175,7 +169,6 @@ rtc_library("g722") {
 
   deps = [
     ":legacy_encoded_audio_frame",
-    "../../api:array_view",
     "../../api/audio_codecs:audio_codecs_api",
     "../../api/audio_codecs/g722:audio_encoder_g722_config",
     "../../api/units:time_delta",
@@ -239,7 +232,6 @@ rtc_library("pcm16b") {
   deps = [
     ":g711",
     ":legacy_encoded_audio_frame",
-    "../../api:array_view",
     "../../api/audio_codecs:audio_codecs_api",
     "../../rtc_base:buffer",
     "../../rtc_base:checks",
@@ -262,7 +254,6 @@ rtc_library("audio_coding_opus_common") {
   ]
 
   deps = [
-    "../../api:array_view",
     "../../api/audio_codecs:audio_codecs_api",
     "../../rtc_base:buffer",
     "../../rtc_base:checks",
@@ -284,7 +275,6 @@ rtc_library("webrtc_opus") {
   deps = [
     ":audio_coding_opus_common",
     ":audio_network_adaptor",
-    "../../api:array_view",
     "../../api:bitrate_allocation",
     "../../api:field_trials_view",
     "../../api/audio_codecs:audio_codecs_api",
@@ -324,7 +314,6 @@ rtc_library("webrtc_multiopus") {
 
   deps = [
     ":audio_coding_opus_common",
-    "../../api:array_view",
     "../../api/audio_codecs:audio_codecs_api",
     "../../api/audio_codecs/opus:audio_decoder_opus_config",
     "../../api/audio_codecs/opus:audio_encoder_opus_config",
@@ -355,7 +344,6 @@ rtc_library("webrtc_opus_wrapper") {
   defines = audio_coding_defines
 
   deps = [
-    "../../api:array_view",
     "../../rtc_base:checks",
     "../../rtc_base:ignore_wundef",
   ]
@@ -423,7 +411,6 @@ rtc_library("audio_network_adaptor") {
       [ ":audio_network_adaptor_config" ]
 
   deps = [
-    "../../api:array_view",
     "../../api/audio_codecs:audio_codecs_api",
     "../../api/environment",
     "../../api/rtc_event_log",
@@ -520,7 +507,6 @@ rtc_library("neteq") {
     ":audio_coding_module_typedefs",
     ":webrtc_cng",
     "..:module_api_public",
-    "../../api:array_view",
     "../../api:field_trials_view",
     "../../api:rtp_headers",
     "../../api:rtp_packet_info",
@@ -577,7 +563,6 @@ rtc_library("neteq_tools_minimal") {
 
   deps = [
     ":neteq",
-    "../../api:array_view",
     "../../api:field_trials",
     "../../api:field_trials_view",
     "../../api:neteq_simulator_api",
@@ -629,7 +614,6 @@ rtc_library("neteq_test_tools") {
     ":neteq_tools",
     ":neteq_tools_minimal",
     ":pcm16b",
-    "../../api:array_view",
     "../../api:rtp_headers",
     "../../api/units:timestamp",
     "../../common_audio",
@@ -664,7 +648,6 @@ rtc_library("neteq_tools") {
     ":neteq_input_audio_tools",
     ":neteq_tools_minimal",
     "..:module_api_public",
-    "../../api:array_view",
     "../../api:rtp_headers",
     "../../api:rtp_packet_info",
     "../../api/audio:audio_frame_api",
@@ -747,7 +730,6 @@ rtc_library("audio_coding_modules_tests_shared") {
     ":neteq_tools_minimal",
     ":webrtc_opus_wrapper",
     "..:module_api",
-    "../../api:array_view",
     "../../api:rtp_headers",
     "../../api:rtp_parameters",
     "../../api/audio:audio_frame_api",
@@ -867,7 +849,6 @@ if (rtc_include_tests) {
       ":audio_encoder_cng",
       ":pcm16b_c",
       ":red",
-      "../../api:array_view",
       "../../api:field_trials",
       "../../api:rtp_headers",
       "../../api:scoped_refptr",
@@ -945,7 +926,6 @@ if (rtc_include_tests) {
       ":audio_coding",
       ":neteq_tools",
       ":neteq_tools_minimal",
-      "../../api:array_view",
       "../../api:scoped_refptr",
       "../../api/audio_codecs:audio_codecs_api",
       "../../api/audio_codecs:builtin_audio_decoder_factory",
@@ -1013,7 +993,6 @@ if (rtc_include_tests) {
 
       deps = [
                ":neteq_input_audio_tools",
-               "../../api:array_view",
                "../../api/audio_codecs:audio_codecs_api",
                "../../api/audio_codecs/g722:audio_encoder_g722_config",
                "../../api/audio_codecs/opus:audio_encoder_opus",
@@ -1183,7 +1162,6 @@ if (rtc_include_tests) {
         ":neteq_input_audio_tools",
         ":neteq_test_tools",
         ":neteq_tools_minimal",
-        "../../api:array_view",
         "../../api:rtp_headers",
         "../../api:scoped_refptr",
         "../../api/audio_codecs:audio_codecs_api",
@@ -1236,7 +1214,6 @@ if (rtc_include_tests) {
       testonly = true
 
       deps = [
-        "../../api:array_view",
         "../../rtc_base:buffer",
         "../../rtc_base:checks",
         "../rtp_rtcp:rtp_rtcp_format",
@@ -1290,7 +1267,6 @@ if (rtc_include_tests) {
         ":neteq_quality_test_support",
         ":neteq_tools",
         ":webrtc_opus",
-        "../../api:array_view",
         "../../api:rtp_parameters",
         "../../api/audio_codecs:audio_codecs_api",
         "../../rtc_base:buffer",
@@ -1326,7 +1302,6 @@ if (rtc_include_tests) {
         ":g711",
         ":neteq",
         ":neteq_quality_test_support",
-        "../../api:array_view",
         "../../api/audio_codecs:audio_codecs_api",
         "../../rtc_base:buffer",
         "../../rtc_base:checks",
@@ -1347,7 +1322,6 @@ if (rtc_include_tests) {
         ":neteq",
         ":neteq_quality_test_support",
         ":pcm16b",
-        "../../api:array_view",
         "../../api/audio_codecs:audio_codecs_api",
         "../../rtc_base:buffer",
         "../../rtc_base:checks",
@@ -1492,7 +1466,6 @@ if (rtc_include_tests) {
         ":webrtc_opus",
         ":webrtc_opus_wrapper",
         "..:module_api_public",
-        "../../api:array_view",
         "../../api:bitrate_allocation",
         "../../api:field_trials",
         "../../api:field_trials_view",
diff --git a/modules/audio_coding/acm2/acm_remixing.cc b/modules/audio_coding/acm2/acm_remixing.cc
index 3ad995542b..812929f06c 100644
--- a/modules/audio_coding/acm2/acm_remixing.cc
+++ b/modules/audio_coding/acm2/acm_remixing.cc
@@ -13,16 +13,16 @@
 #include <algorithm>
 #include <cstddef>
 #include <cstdint>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/audio_frame.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/numerics/safe_conversions.h"
 
 namespace webrtc {
 
-void DownMixFrame(const AudioFrame& input, ArrayView<int16_t> output) {
+void DownMixFrame(const AudioFrame& input, std::span<int16_t> output) {
   RTC_DCHECK_EQ(input.num_channels_, 2);
   RTC_DCHECK_EQ(output.size(), input.samples_per_channel_);
 
diff --git a/modules/audio_coding/acm2/acm_remixing.h b/modules/audio_coding/acm2/acm_remixing.h
index 0ccb7c0bf4..fc2d082405 100644
--- a/modules/audio_coding/acm2/acm_remixing.h
+++ b/modules/audio_coding/acm2/acm_remixing.h
@@ -13,16 +13,16 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/audio_frame.h"
 
 namespace webrtc {
 
 // Stereo-to-mono downmixing. The length of the output must equal to the number
 // of samples per channel in the input.
-void DownMixFrame(const AudioFrame& input, ArrayView<int16_t> output);
+void DownMixFrame(const AudioFrame& input, std::span<int16_t> output);
 
 // Remixes the interleaved input frame to an interleaved output data vector. The
 // remixed data replaces the data in the output vector which is resized if
diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index dfbfe18a2d..784ceb011f 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -15,11 +15,11 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio/audio_view.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/function_view.h"
@@ -262,7 +262,7 @@ int32_t AudioCodingModuleImpl::Encode(
   encode_buffer_.Clear();
   encoded_info = encoder_stack_->Encode(
       rtp_timestamp,
-      ArrayView<const int16_t>(
+      std::span<const int16_t>(
           input_data.audio,
           input_data.audio_channel * input_data.length_per_channel),
       &encode_buffer_);
diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index f8285f83c6..cf9a385cc2 100644
--- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -18,12 +18,12 @@
 #include <cstring>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_decoder_factory.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/audio_encoder_factory.h"
@@ -208,7 +208,7 @@ class AudioCodingModuleTestOldApi : public ::testing::Test {
     const uint8_t kPayload[kPayloadSizeBytes] = {0};
     ASSERT_EQ(0, neteq_->InsertPacket(
                      rtp_header_,
-                     ArrayView<const uint8_t>(kPayload, kPayloadSizeBytes),
+                     std::span<const uint8_t>(kPayload, kPayloadSizeBytes),
                      /*receive_time=*/Timestamp::MinusInfinity()));
     rtp_utility_->Forward(&rtp_header_);
   }
@@ -577,7 +577,7 @@ class AcmSenderBitExactnessOldApi : public ::testing::Test,
     Buffer checksum_result =
         Buffer::CreateWithCapacity(payload_checksum_->Size());
     checksum_result.AppendData(
-        payload_checksum_->Size(), [&](ArrayView<uint8_t> checksum_view) {
+        payload_checksum_->Size(), [&](std::span<uint8_t> checksum_view) {
           payload_checksum_->Finish(checksum_view.data(), checksum_view.size());
           return checksum_view.size();
         });
@@ -1117,7 +1117,7 @@ TEST_F(AcmSenderBitExactnessOldApi, External_Pcmu_20ms) {
       .Times(AtLeast(1))
       .WillRepeatedly(Invoke(
           &encoder, static_cast<AudioEncoder::EncodedInfo (AudioEncoder::*)(
-                        uint32_t, ArrayView<const int16_t>, Buffer*)>(
+                        uint32_t, std::span<const int16_t>, Buffer*)>(
                         &AudioEncoderPcmU::Encode)));
   ASSERT_TRUE(SetUpSender(kTestFileMono32kHz, 32000));
   ASSERT_NO_FATAL_FAILURE(
diff --git a/modules/audio_coding/audio_network_adaptor/controller_manager.cc b/modules/audio_coding/audio_network_adaptor/controller_manager.cc
index 87f771b911..f5cae7d822 100644
--- a/modules/audio_coding/audio_network_adaptor/controller_manager.cc
+++ b/modules/audio_coding/audio_network_adaptor/controller_manager.cc
@@ -18,12 +18,12 @@
 #include <memory>
 #include <optional>
 #include <set>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "modules/audio_coding/audio_network_adaptor/bitrate_controller.h"
 #include "modules/audio_coding/audio_network_adaptor/channel_controller.h"
@@ -88,7 +88,7 @@ std::unique_ptr<FecControllerPlrBased> CreateFecControllerPlrBased(
 
 std::unique_ptr<FrameLengthController> CreateFrameLengthController(
     const audio_network_adaptor::config::FrameLengthController& config,
-    ArrayView<const int> encoder_frame_lengths_ms,
+    std::span<const int> encoder_frame_lengths_ms,
     int initial_frame_length_ms,
     int min_encoder_bitrate_bps) {
   RTC_CHECK(config.has_fl_increasing_packet_loss_fraction());
@@ -212,7 +212,7 @@ std::unique_ptr<BitrateController> CreateBitrateController(
 
 std::unique_ptr<FrameLengthControllerV2> CreateFrameLengthControllerV2(
     const audio_network_adaptor::config::FrameLengthControllerV2& config,
-    ArrayView<const int> encoder_frame_lengths_ms) {
+    std::span<const int> encoder_frame_lengths_ms) {
   return std::make_unique<FrameLengthControllerV2>(
       encoder_frame_lengths_ms, config.min_payload_bitrate_bps(),
       config.use_slow_adaptation());
@@ -232,7 +232,7 @@ std::unique_ptr<ControllerManager> ControllerManagerImpl::Create(
     const Environment& env,
     absl::string_view config_string,
     size_t num_encoder_channels,
-    ArrayView<const int> encoder_frame_lengths_ms,
+    std::span<const int> encoder_frame_lengths_ms,
     int min_encoder_bitrate_bps,
     size_t intial_channels_to_encode,
     int initial_frame_length_ms,
diff --git a/modules/audio_coding/audio_network_adaptor/controller_manager.h b/modules/audio_coding/audio_network_adaptor/controller_manager.h
index 318db0c72b..f85a583919 100644
--- a/modules/audio_coding/audio_network_adaptor/controller_manager.h
+++ b/modules/audio_coding/audio_network_adaptor/controller_manager.h
@@ -16,11 +16,11 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "modules/audio_coding/audio_network_adaptor/controller.h"
 
@@ -55,7 +55,7 @@ class ControllerManagerImpl final : public ControllerManager {
       const Environment& env,
       absl::string_view config_string,
       size_t num_encoder_channels,
-      ArrayView<const int> encoder_frame_lengths_ms,
+      std::span<const int> encoder_frame_lengths_ms,
       int min_encoder_bitrate_bps,
       size_t intial_channels_to_encode,
       int initial_frame_length_ms,
diff --git a/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.cc b/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.cc
index bbb0a2de29..bb033b228e 100644
--- a/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.cc
+++ b/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.cc
@@ -10,8 +10,9 @@
 
 #include "modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.h"
 
+#include <span>
+
 #include "absl/algorithm/container.h"
-#include "api/array_view.h"
 #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
 #include "rtc_base/checks.h"
 
@@ -25,7 +26,7 @@ int OverheadBps(int overhead_bytes_per_packet, int frame_length_ms) {
 }  // namespace
 
 FrameLengthControllerV2::FrameLengthControllerV2(
-    ArrayView<const int> encoder_frame_lengths_ms,
+    std::span<const int> encoder_frame_lengths_ms,
     int min_payload_bitrate_bps,
     bool use_slow_adaptation)
     : encoder_frame_lengths_ms_(encoder_frame_lengths_ms.begin(),
diff --git a/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.h b/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.h
index e8ff5adfde..4ecfb0f4b6 100644
--- a/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.h
+++ b/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.h
@@ -12,9 +12,9 @@
 #define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FRAME_LENGTH_CONTROLLER_V2_H_
 
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_coding/audio_network_adaptor/controller.h"
 #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
 
@@ -22,7 +22,7 @@ namespace webrtc {
 
 class FrameLengthControllerV2 final : public Controller {
  public:
-  FrameLengthControllerV2(ArrayView<const int> encoder_frame_lengths_ms,
+  FrameLengthControllerV2(std::span<const int> encoder_frame_lengths_ms,
                           int min_payload_bitrate_bps,
                           bool use_slow_adaptation);
 
diff --git a/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc b/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc
index f1250a29f4..d651d7888f 100644
--- a/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc
+++ b/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc
@@ -14,10 +14,10 @@
 #include <cstdint>
 #include <limits>
 #include <memory>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/audio_encoder_factory.h"
 #include "api/audio_codecs/audio_format.h"
@@ -83,7 +83,7 @@ TEST_P(AudioEncoderFactoryTest, CanRunAllSupportedEncoders) {
                                               encoder->NumChannels() / 100);
     Buffer out;
     BufferT<int16_t> audio;
-    audio.SetData(num_samples, [](ArrayView<int16_t> audio) {
+    audio.SetData(num_samples, [](std::span<int16_t> audio) {
       for (size_t i = 0; i != audio.size(); ++i) {
         // Just put some numbers in there, ensure they're within range.
         audio[i] =
diff --git a/modules/audio_coding/codecs/cng/audio_encoder_cng.cc b/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
index fa7006fa75..384038c057 100644
--- a/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
+++ b/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
@@ -14,10 +14,10 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/units/time_delta.h"
 #include "common_audio/vad/include/vad.h"
@@ -49,14 +49,14 @@ class AudioEncoderCng final : public AudioEncoder {
   size_t Max10MsFramesInAPacket() const override;
   int GetTargetBitrate() const override;
   EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
-                         ArrayView<const int16_t> audio,
+                         std::span<const int16_t> audio,
                          Buffer* encoded) override;
   void Reset() override;
   bool SetFec(bool enable) override;
   bool SetDtx(bool enable) override;
   bool SetApplication(Application application) override;
   void SetMaxPlaybackRate(int frequency_hz) override;
-  ArrayView<std::unique_ptr<AudioEncoder>> ReclaimContainedEncoders() override;
+  std::span<std::unique_ptr<AudioEncoder>> ReclaimContainedEncoders() override;
   void OnReceivedUplinkPacketLossFraction(
       float uplink_packet_loss_fraction) override;
   void OnReceivedUplinkBandwidth(int target_audio_bitrate_bps,
@@ -125,7 +125,7 @@ int AudioEncoderCng::GetTargetBitrate() const {
 
 AudioEncoder::EncodedInfo AudioEncoderCng::EncodeImpl(
     uint32_t rtp_timestamp,
-    ArrayView<const int16_t> audio,
+    std::span<const int16_t> audio,
     Buffer* encoded) {
   const size_t samples_per_10ms_frame = SamplesPer10msFrame();
   RTC_CHECK_EQ(speech_buffer_.size(),
@@ -216,9 +216,9 @@ void AudioEncoderCng::SetMaxPlaybackRate(int frequency_hz) {
   speech_encoder_->SetMaxPlaybackRate(frequency_hz);
 }
 
-ArrayView<std::unique_ptr<AudioEncoder>>
+std::span<std::unique_ptr<AudioEncoder>>
 AudioEncoderCng::ReclaimContainedEncoders() {
-  return ArrayView<std::unique_ptr<AudioEncoder>>(&speech_encoder_, 1);
+  return std::span<std::unique_ptr<AudioEncoder>>(&speech_encoder_, 1);
 }
 
 void AudioEncoderCng::OnReceivedUplinkPacketLossFraction(
@@ -253,7 +253,7 @@ AudioEncoder::EncodedInfo AudioEncoderCng::EncodePassive(
     // that value, in which case we don't want to overwrite any value from
     // an earlier iteration.
     size_t encoded_bytes_tmp = cng_encoder_->Encode(
-        ArrayView<const int16_t>(&speech_buffer_[i * samples_per_10ms_frame],
+        std::span<const int16_t>(&speech_buffer_[i * samples_per_10ms_frame],
                                  samples_per_10ms_frame),
         force_sid, encoded);
 
@@ -279,7 +279,7 @@ AudioEncoder::EncodedInfo AudioEncoderCng::EncodeActive(size_t frames_to_encode,
   for (size_t i = 0; i < frames_to_encode; ++i) {
     info = speech_encoder_->Encode(
         rtp_timestamps_.front(),
-        ArrayView<const int16_t>(&speech_buffer_[i * samples_per_10ms_frame],
+        std::span<const int16_t>(&speech_buffer_[i * samples_per_10ms_frame],
                                  samples_per_10ms_frame),
         encoded);
     if (i + 1 == frames_to_encode) {
diff --git a/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc b/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
index 720e236693..81642e24c4 100644
--- a/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
+++ b/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
@@ -15,9 +15,9 @@
 #include <cstring>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/units/time_delta.h"
 #include "common_audio/vad/include/vad.h"
@@ -98,7 +98,7 @@ class AudioEncoderCngTest : public ::testing::Test {
   void Encode() {
     ASSERT_TRUE(cng_) << "Must call CreateCng() first.";
     encoded_info_ = cng_->Encode(
-        timestamp_, ArrayView<const int16_t>(audio_, num_audio_samples_10ms_),
+        timestamp_, std::span<const int16_t>(audio_, num_audio_samples_10ms_),
         &encoded_);
     timestamp_ += static_cast<uint32_t>(num_audio_samples_10ms_);
   }
diff --git a/modules/audio_coding/codecs/cng/cng_unittest.cc b/modules/audio_coding/codecs/cng/cng_unittest.cc
index 309e2b6880..d6a1305675 100644
--- a/modules/audio_coding/codecs/cng/cng_unittest.cc
+++ b/modules/audio_coding/codecs/cng/cng_unittest.cc
@@ -10,9 +10,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <cstdio>
+#include <span>
 #include <string>
 
-#include "api/array_view.h"
 #include "modules/audio_coding/codecs/cng/webrtc_cng.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/checks.h"
@@ -66,11 +66,11 @@ void CngTest::TestCngEncode(int sample_rate_hz, int quality) {
   ComfortNoiseEncoder cng_encoder(sample_rate_hz, kSidNormalIntervalUpdate,
                                   quality);
   EXPECT_EQ(0U, cng_encoder.Encode(
-                    ArrayView<const int16_t>(speech_data_, num_samples_10ms),
+                    std::span<const int16_t>(speech_data_, num_samples_10ms),
                     kNoSid, &sid_data));
   EXPECT_EQ(static_cast<size_t>(quality + 1),
             cng_encoder.Encode(
-                ArrayView<const int16_t>(speech_data_, num_samples_10ms),
+                std::span<const int16_t>(speech_data_, num_samples_10ms),
                 kForceSid, &sid_data));
 }
 
@@ -100,7 +100,7 @@ TEST_F(CngDeathTest, CngEncodeTooLong) {
   ComfortNoiseEncoder cng_encoder(8000, kSidNormalIntervalUpdate,
                                   kCNGNumParamsNormal);
   // Run encoder with too much data.
-  EXPECT_DEATH(cng_encoder.Encode(ArrayView<const int16_t>(speech_data_, 641),
+  EXPECT_DEATH(cng_encoder.Encode(std::span<const int16_t>(speech_data_, 641),
                                   kNoSid, &sid_data),
                "");
 }
@@ -137,7 +137,7 @@ TEST_F(CngTest, CngUpdateSid) {
 
   // Run normal Encode and UpdateSid.
   EXPECT_EQ(kCNGNumParamsNormal + 1,
-            cng_encoder.Encode(ArrayView<const int16_t>(speech_data_, 160),
+            cng_encoder.Encode(std::span<const int16_t>(speech_data_, 160),
                                kForceSid, &sid_data));
   cng_decoder.UpdateSid(sid_data);
 
@@ -146,14 +146,14 @@ TEST_F(CngTest, CngUpdateSid) {
   cng_decoder.Reset();
 
   // Expect 0 because of unstable parameters after switching length.
-  EXPECT_EQ(0U, cng_encoder.Encode(ArrayView<const int16_t>(speech_data_, 160),
+  EXPECT_EQ(0U, cng_encoder.Encode(std::span<const int16_t>(speech_data_, 160),
                                    kForceSid, &sid_data));
   EXPECT_EQ(
       kCNGNumParamsHigh + 1,
-      cng_encoder.Encode(ArrayView<const int16_t>(speech_data_ + 160, 160),
+      cng_encoder.Encode(std::span<const int16_t>(speech_data_ + 160, 160),
                          kForceSid, &sid_data));
   cng_decoder.UpdateSid(
-      ArrayView<const uint8_t>(sid_data.data(), kCNGNumParamsNormal + 1));
+      std::span<const uint8_t>(sid_data.data(), kCNGNumParamsNormal + 1));
 }
 
 // Update SID parameters, with wrong parameters or without calling decode.
@@ -165,7 +165,7 @@ TEST_F(CngTest, CngUpdateSidErroneous) {
                                   kCNGNumParamsNormal);
   ComfortNoiseDecoder cng_decoder;
   EXPECT_EQ(kCNGNumParamsNormal + 1,
-            cng_encoder.Encode(ArrayView<const int16_t>(speech_data_, 160),
+            cng_encoder.Encode(std::span<const int16_t>(speech_data_, 160),
                                kForceSid, &sid_data));
 
   // First run with valid parameters, then with too many CNG parameters.
@@ -193,18 +193,18 @@ TEST_F(CngTest, CngGenerate) {
 
   // Normal Encode.
   EXPECT_EQ(kCNGNumParamsNormal + 1,
-            cng_encoder.Encode(ArrayView<const int16_t>(speech_data_, 160),
+            cng_encoder.Encode(std::span<const int16_t>(speech_data_, 160),
                                kForceSid, &sid_data));
 
   // Normal UpdateSid.
   cng_decoder.UpdateSid(sid_data);
 
   // Two normal Generate, one with new_period.
-  EXPECT_TRUE(cng_decoder.Generate(ArrayView<int16_t>(out_data, 640), 1));
-  EXPECT_TRUE(cng_decoder.Generate(ArrayView<int16_t>(out_data, 640), 0));
+  EXPECT_TRUE(cng_decoder.Generate(std::span<int16_t>(out_data, 640), 1));
+  EXPECT_TRUE(cng_decoder.Generate(std::span<int16_t>(out_data, 640), 0));
 
   // Call Genereate with too much data.
-  EXPECT_FALSE(cng_decoder.Generate(ArrayView<int16_t>(out_data, 641), 0));
+  EXPECT_FALSE(cng_decoder.Generate(std::span<int16_t>(out_data, 641), 0));
 }
 
 // Test automatic SID.
@@ -219,13 +219,13 @@ TEST_F(CngTest, CngAutoSid) {
   // Normal Encode, 100 msec, where no SID data should be generated.
   for (int i = 0; i < 10; i++) {
     EXPECT_EQ(0U,
-              cng_encoder.Encode(ArrayView<const int16_t>(speech_data_, 160),
+              cng_encoder.Encode(std::span<const int16_t>(speech_data_, 160),
                                  kNoSid, &sid_data));
   }
 
   // We have reached 100 msec, and SID data should be generated.
   EXPECT_EQ(kCNGNumParamsNormal + 1,
-            cng_encoder.Encode(ArrayView<const int16_t>(speech_data_, 160),
+            cng_encoder.Encode(std::span<const int16_t>(speech_data_, 160),
                                kNoSid, &sid_data));
 }
 
@@ -239,13 +239,13 @@ TEST_F(CngTest, CngAutoSidShort) {
   ComfortNoiseDecoder cng_decoder;
 
   // First call will never generate SID, unless forced to.
-  EXPECT_EQ(0U, cng_encoder.Encode(ArrayView<const int16_t>(speech_data_, 160),
+  EXPECT_EQ(0U, cng_encoder.Encode(std::span<const int16_t>(speech_data_, 160),
                                    kNoSid, &sid_data));
 
   // Normal Encode, 100 msec, SID data should be generated all the time.
   for (int i = 0; i < 10; i++) {
     EXPECT_EQ(kCNGNumParamsNormal + 1,
-              cng_encoder.Encode(ArrayView<const int16_t>(speech_data_, 160),
+              cng_encoder.Encode(std::span<const int16_t>(speech_data_, 160),
                                  kNoSid, &sid_data));
   }
 }
diff --git a/modules/audio_coding/codecs/cng/webrtc_cng.cc b/modules/audio_coding/codecs/cng/webrtc_cng.cc
index 5961cb0e9c..f729bae42b 100644
--- a/modules/audio_coding/codecs/cng/webrtc_cng.cc
+++ b/modules/audio_coding/codecs/cng/webrtc_cng.cc
@@ -13,8 +13,8 @@
 #include <algorithm>
 #include <cstddef>
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "common_audio/signal_processing/include/signal_processing_library.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/checks.h"
@@ -75,7 +75,7 @@ void ComfortNoiseDecoder::Reset() {
   dec_used_scale_factor_ = 0;
 }
 
-void ComfortNoiseDecoder::UpdateSid(ArrayView<const uint8_t> sid) {
+void ComfortNoiseDecoder::UpdateSid(std::span<const uint8_t> sid) {
   int16_t refCs[WEBRTC_CNG_MAX_LPC_ORDER];
   int32_t targetEnergy;
   size_t length = sid.size();
@@ -112,7 +112,7 @@ void ComfortNoiseDecoder::UpdateSid(ArrayView<const uint8_t> sid) {
   }
 }
 
-bool ComfortNoiseDecoder::Generate(ArrayView<int16_t> out_data,
+bool ComfortNoiseDecoder::Generate(std::span<int16_t> out_data,
                                    bool new_period) {
   int16_t excitation[kCngMaxOutsizeOrder];
   int16_t low[kCngMaxOutsizeOrder];
@@ -236,7 +236,7 @@ void ComfortNoiseEncoder::Reset(int fs, int interval, int quality) {
   enc_seed_ = 7777; /* For debugging only. */
 }
 
-size_t ComfortNoiseEncoder::Encode(ArrayView<const int16_t> speech,
+size_t ComfortNoiseEncoder::Encode(std::span<const int16_t> speech,
                                    bool force_sid,
                                    Buffer* output) {
   int16_t arCoefs[WEBRTC_CNG_MAX_LPC_ORDER + 1];
@@ -367,7 +367,7 @@ size_t ComfortNoiseEncoder::Encode(ArrayView<const int16_t> speech,
       index = 94;
 
     const size_t output_coefs = enc_nrOfCoefs_ + 1;
-    output->AppendData(output_coefs, [&](ArrayView<uint8_t> output) {
+    output->AppendData(output_coefs, [&](std::span<uint8_t> output) {
       output[0] = (uint8_t)index;
 
       /* Quantize coefficients with tweak for WebRtc implementation of
diff --git a/modules/audio_coding/codecs/cng/webrtc_cng.h b/modules/audio_coding/codecs/cng/webrtc_cng.h
index 738f60aa1f..4a12612b29 100644
--- a/modules/audio_coding/codecs/cng/webrtc_cng.h
+++ b/modules/audio_coding/codecs/cng/webrtc_cng.h
@@ -14,8 +14,8 @@
 #include <stdint.h>
 
 #include <cstddef>
+#include <span>
 
-#include "api/array_view.h"
 #include "rtc_base/buffer.h"
 
 #define WEBRTC_CNG_MAX_LPC_ORDER 12
@@ -34,7 +34,7 @@ class ComfortNoiseDecoder {
 
   // Updates the CN state when a new SID packet arrives.
   // `sid` is a view of the SID packet without the headers.
-  void UpdateSid(ArrayView<const uint8_t> sid);
+  void UpdateSid(std::span<const uint8_t> sid);
 
   // Generates comfort noise.
   // `out_data` will be filled with samples - its size determines the number of
@@ -43,7 +43,7 @@ class ComfortNoiseDecoder {
   // currently 640 bytes (equalling 10ms at 64kHz).
   // TODO(ossu): Specify better limits for the size of out_data. Either let it
   //             be unbounded or limit to 10ms in the current sample rate.
-  bool Generate(ArrayView<int16_t> out_data, bool new_period);
+  bool Generate(std::span<int16_t> out_data, bool new_period);
 
  private:
   uint32_t dec_seed_;
@@ -79,7 +79,7 @@ class ComfortNoiseEncoder {
   // true, a SID frame is forced and the internal sid interval counter is reset.
   // Will fail if the input size is too large (> 640 samples, see
   // ComfortNoiseDecoder::Generate).
-  size_t Encode(ArrayView<const int16_t> speech,
+  size_t Encode(std::span<const int16_t> speech,
                 bool force_sid,
                 Buffer* output);
 
diff --git a/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc b/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
index 281be597cf..eaaa9b6321 100644
--- a/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
+++ b/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
@@ -13,9 +13,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <utility>
 
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/units/time_delta.h"
 #include "modules/audio_coding/codecs/g711/g711_interface.h"
@@ -69,7 +69,7 @@ int AudioEncoderPcm::GetTargetBitrate() const {
 
 AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeImpl(
     uint32_t rtp_timestamp,
-    ArrayView<const int16_t> audio,
+    std::span<const int16_t> audio,
     Buffer* encoded) {
   if (speech_buffer_.empty()) {
     first_timestamp_in_buffer_ = rtp_timestamp;
@@ -83,7 +83,7 @@ AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeImpl(
   info.encoded_timestamp = first_timestamp_in_buffer_;
   info.payload_type = payload_type_;
   info.encoded_bytes = encoded->AppendData(
-      full_frame_samples_ * BytesPerSample(), [&](ArrayView<uint8_t> encoded) {
+      full_frame_samples_ * BytesPerSample(), [&](std::span<uint8_t> encoded) {
         return EncodeCall(&speech_buffer_[0], full_frame_samples_,
                           encoded.data());
       });
diff --git a/modules/audio_coding/codecs/g711/audio_encoder_pcm.h b/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
index 0d56e58d02..1ba2e1f746 100644
--- a/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
+++ b/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
@@ -14,10 +14,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/units/time_delta.h"
 #include "rtc_base/buffer.h"
@@ -54,7 +54,7 @@ class AudioEncoderPcm : public AudioEncoder {
   AudioEncoderPcm(const Config& config, int sample_rate_hz);
 
   EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
-                         ArrayView<const int16_t> audio,
+                         std::span<const int16_t> audio,
                          Buffer* encoded) override;
 
   virtual size_t EncodeCall(const int16_t* audio,
diff --git a/modules/audio_coding/codecs/g722/audio_encoder_g722.cc b/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
index 45ffeff413..36c8440b72 100644
--- a/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
+++ b/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
@@ -13,9 +13,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <utility>
 
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/g722/audio_encoder_g722_config.h"
 #include "api/units/time_delta.h"
@@ -94,7 +94,7 @@ AudioEncoderG722Impl::GetFrameLengthRange() const {
 
 AudioEncoder::EncodedInfo AudioEncoderG722Impl::EncodeImpl(
     uint32_t rtp_timestamp,
-    ArrayView<const int16_t> audio,
+    std::span<const int16_t> audio,
     Buffer* encoded) {
   if (num_10ms_frames_buffered_ == 0)
     first_timestamp_in_buffer_ = rtp_timestamp;
@@ -124,7 +124,7 @@ AudioEncoder::EncodedInfo AudioEncoderG722Impl::EncodeImpl(
   const size_t bytes_to_encode = samples_per_channel / 2 * num_channels_;
   EncodedInfo info;
   info.encoded_bytes =
-      encoded->AppendData(bytes_to_encode, [&](ArrayView<uint8_t> encoded) {
+      encoded->AppendData(bytes_to_encode, [&](std::span<uint8_t> encoded) {
         // Interleave the encoded bytes of the different channels. Each separate
         // channel and the interleaved stream encodes two samples per byte, most
         // significant half first.
diff --git a/modules/audio_coding/codecs/g722/audio_encoder_g722.h b/modules/audio_coding/codecs/g722/audio_encoder_g722.h
index c7942029eb..97e5cfa271 100644
--- a/modules/audio_coding/codecs/g722/audio_encoder_g722.h
+++ b/modules/audio_coding/codecs/g722/audio_encoder_g722.h
@@ -15,10 +15,10 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/g722/audio_encoder_g722_config.h"
 #include "api/units/time_delta.h"
@@ -47,7 +47,7 @@ class AudioEncoderG722Impl final : public AudioEncoder {
 
  protected:
   EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
-                         ArrayView<const int16_t> audio,
+                         std::span<const int16_t> audio,
                          Buffer* encoded) override;
 
  private:
diff --git a/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc b/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc
index 8c914ca28f..cd26ef9ba3 100644
--- a/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc
+++ b/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc
@@ -15,10 +15,10 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_decoder.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/checks.h"
@@ -37,7 +37,7 @@ size_t LegacyEncodedAudioFrame::Duration() const {
 }
 
 std::optional<AudioDecoder::EncodedAudioFrame::DecodeResult>
-LegacyEncodedAudioFrame::Decode(ArrayView<int16_t> decoded) const {
+LegacyEncodedAudioFrame::Decode(std::span<int16_t> decoded) const {
   AudioDecoder::SpeechType speech_type = AudioDecoder::kSpeech;
   const int ret = decoder_->Decode(
       payload_.data(), payload_.size(), decoder_->SampleRateHz(),
diff --git a/modules/audio_coding/codecs/legacy_encoded_audio_frame.h b/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
index 50349e0aed..e8353bad34 100644
--- a/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
+++ b/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
@@ -15,9 +15,9 @@
 #include <stdint.h>
 
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_decoder.h"
 #include "rtc_base/buffer.h"
 
@@ -37,7 +37,7 @@ class LegacyEncodedAudioFrame final : public AudioDecoder::EncodedAudioFrame {
 
   size_t Duration() const override;
 
-  std::optional<DecodeResult> Decode(ArrayView<int16_t> decoded) const override;
+  std::optional<DecodeResult> Decode(std::span<int16_t> decoded) const override;
 
   // For testing:
   const Buffer& payload() const { return payload_; }
diff --git a/modules/audio_coding/codecs/legacy_encoded_audio_frame_unittest.cc b/modules/audio_coding/codecs/legacy_encoded_audio_frame_unittest.cc
index e6bd7cf20f..a9be6de2c6 100644
--- a/modules/audio_coding/codecs/legacy_encoded_audio_frame_unittest.cc
+++ b/modules/audio_coding/codecs/legacy_encoded_audio_frame_unittest.cc
@@ -12,8 +12,8 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/numerics/safe_conversions.h"
@@ -133,7 +133,7 @@ TEST_P(SplitBySamplesTest, PayloadSizes) {
     // sample was missed or repeated.
     const auto generate_payload = [](size_t num_bytes) {
       Buffer payload = Buffer::CreateWithCapacity(num_bytes);
-      payload.AppendData(num_bytes, [](ArrayView<uint8_t> payload_view) {
+      payload.AppendData(num_bytes, [](std::span<uint8_t> payload_view) {
         uint8_t value = 0;
         // Allow wrap-around of value in counter below.
         for (size_t i = 0; i != payload_view.size(); ++i, ++value) {
diff --git a/modules/audio_coding/codecs/opus/audio_coder_opus_common.h b/modules/audio_coding/codecs/opus/audio_coder_opus_common.h
index 011abfb25f..4d98cae858 100644
--- a/modules/audio_coding/codecs/opus/audio_coder_opus_common.h
+++ b/modules/audio_coding/codecs/opus/audio_coder_opus_common.h
@@ -14,12 +14,12 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_decoder.h"
 #include "api/audio_codecs/audio_format.h"
 #include "rtc_base/buffer.h"
@@ -61,7 +61,7 @@ class OpusFrame : public AudioDecoder::EncodedAudioFrame {
   bool IsDtxPacket() const override { return payload_.size() <= 2; }
 
   std::optional<DecodeResult> Decode(
-      ArrayView<int16_t> decoded) const override {
+      std::span<int16_t> decoded) const override {
     AudioDecoder::SpeechType speech_type = AudioDecoder::kSpeech;
     int ret;
     if (is_primary_payload_) {
diff --git a/modules/audio_coding/codecs/opus/audio_decoder_opus.cc b/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
index c4fb057718..203969f967 100644
--- a/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
+++ b/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
@@ -13,10 +13,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_decoder.h"
 #include "api/field_trials_view.h"
 #include "modules/audio_coding/codecs/opus/audio_coder_opus_common.h"
@@ -141,7 +141,7 @@ void AudioDecoderOpusImpl::GeneratePlc(
     return;
   }
   int plc_size = WebRtcOpus_PlcDuration(dec_state_) * channels_;
-  concealment_audio->AppendData(plc_size, [&](ArrayView<int16_t> decoded) {
+  concealment_audio->AppendData(plc_size, [&](std::span<int16_t> decoded) {
     int16_t temp_type = 1;
     int ret =
         WebRtcOpus_Decode(dec_state_, nullptr, 0, decoded.data(), &temp_type);
diff --git a/modules/audio_coding/codecs/opus/audio_decoder_opus_unittest.cc b/modules/audio_coding/codecs/opus/audio_decoder_opus_unittest.cc
index fb13f8ab35..b4d5df526a 100644
--- a/modules/audio_coding/codecs/opus/audio_decoder_opus_unittest.cc
+++ b/modules/audio_coding/codecs/opus/audio_decoder_opus_unittest.cc
@@ -16,10 +16,10 @@
 #include <cstdint>
 #include <limits>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/audio_frame.h"
 #include "api/audio_codecs/audio_decoder.h"
 #include "api/audio_codecs/audio_encoder.h"
@@ -82,7 +82,7 @@ class WhiteNoiseGenerator {
                                     std::numeric_limits<int16_t>::max())),
         random_generator_(42) {}
 
-  void GenerateNextFrame(ArrayView<int16_t> frame) {
+  void GenerateNextFrame(std::span<int16_t> frame) {
     for (size_t i = 0; i < frame.size(); ++i) {
       frame[i] = saturated_cast<int16_t>(
           random_generator_.Rand(-amplitude_, amplitude_));
@@ -94,7 +94,7 @@ class WhiteNoiseGenerator {
   Random random_generator_;
 };
 
-bool IsZeroedFrame(ArrayView<const int16_t> audio) {
+bool IsZeroedFrame(std::span<const int16_t> audio) {
   for (const int16_t& v : audio) {
     if (v != 0)
       return false;
@@ -102,7 +102,7 @@ bool IsZeroedFrame(ArrayView<const int16_t> audio) {
   return true;
 }
 
-bool IsTrivialStereo(ArrayView<const int16_t> audio) {
+bool IsTrivialStereo(std::span<const int16_t> audio) {
   const int num_samples = CheckedDivExact(audio.size(), static_cast<size_t>(2));
   for (int i = 0, j = 0; i < num_samples; ++i, j += 2) {
     if (audio[j] != audio[j + 1]) {
@@ -313,7 +313,7 @@ TEST(AudioDecoderOpusTest,
   ASSERT_EQ(speech_type, AudioDecoder::SpeechType::kComfortNoise);
   RTC_CHECK_GT(num_decoded_samples, 0);
   RTC_CHECK_LE(num_decoded_samples, decoded_frame.size());
-  ArrayView<const int16_t> decoded_view(decoded_frame.data(),
+  std::span<const int16_t> decoded_view(decoded_frame.data(),
                                         num_decoded_samples);
   // Make sure that comfort noise is not a muted frame.
   ASSERT_FALSE(IsZeroedFrame(decoded_view));
@@ -352,7 +352,7 @@ TEST(AudioDecoderOpusTest, MonoEncoderStereoDecoderOutputsTrivialStereoPlc) {
   decoder.GeneratePlc(/*requested_samples_per_channel=*/kIgnored,
                       &concealment_audio);
   RTC_CHECK_GT(concealment_audio.size(), 0);
-  ArrayView<const int16_t> decoded_view(concealment_audio.data(),
+  std::span<const int16_t> decoded_view(concealment_audio.data(),
                                         concealment_audio.size());
   // Make sure that packet loss concealment is not a muted frame.
   ASSERT_FALSE(IsZeroedFrame(decoded_view));
@@ -450,7 +450,7 @@ TEST(AudioDecoderOpusTest,
   ASSERT_EQ(speech_type, AudioDecoder::SpeechType::kComfortNoise);
   RTC_CHECK_GT(num_decoded_samples, 0);
   RTC_CHECK_LE(num_decoded_samples, decoded_frame.size());
-  ArrayView<const int16_t> decoded_view(decoded_frame.data(),
+  std::span<const int16_t> decoded_view(decoded_frame.data(),
                                         num_decoded_samples);
   // Make sure that comfort noise is not a muted frame.
   ASSERT_FALSE(IsZeroedFrame(decoded_view));
@@ -484,7 +484,7 @@ TEST(AudioDecoderOpusTest,
   decoder.GeneratePlc(/*requested_samples_per_channel=*/kIgnored,
                       &concealment_audio);
   RTC_CHECK_GT(concealment_audio.size(), 0);
-  ArrayView<const int16_t> decoded_view(concealment_audio.data(),
+  std::span<const int16_t> decoded_view(concealment_audio.data(),
                                         concealment_audio.size());
   // Make sure that packet loss concealment is not a muted frame.
   ASSERT_FALSE(IsZeroedFrame(decoded_view));
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc b/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc
index be55406ecb..646cd9f570 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc
@@ -25,12 +25,12 @@
 #include <iterator>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/match.h"
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.h"
@@ -339,7 +339,7 @@ int AudioEncoderMultiChannelOpusImpl::GetTargetBitrate() const {
 
 AudioEncoder::EncodedInfo AudioEncoderMultiChannelOpusImpl::EncodeImpl(
     uint32_t rtp_timestamp,
-    ArrayView<const int16_t> audio,
+    std::span<const int16_t> audio,
     Buffer* encoded) {
   if (input_buffer_.empty())
     first_timestamp_in_buffer_ = rtp_timestamp;
@@ -355,7 +355,7 @@ AudioEncoder::EncodedInfo AudioEncoderMultiChannelOpusImpl::EncodeImpl(
   const size_t max_encoded_bytes = SufficientOutputBufferSize();
   EncodedInfo info;
   info.encoded_bytes =
-      encoded->AppendData(max_encoded_bytes, [&](ArrayView<uint8_t> encoded) {
+      encoded->AppendData(max_encoded_bytes, [&](std::span<uint8_t> encoded) {
         int status = WebRtcOpus_Encode(
             inst_, &input_buffer_[0],
             CheckedDivExact(input_buffer_.size(), config_.num_channels),
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h b/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h
index ccf05a8429..b86fedcf64 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h
+++ b/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h
@@ -16,10 +16,10 @@
 
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.h"
@@ -59,7 +59,7 @@ class AudioEncoderMultiChannelOpusImpl final : public AudioEncoder {
 
  protected:
   EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
-                         ArrayView<const int16_t> audio,
+                         std::span<const int16_t> audio,
                          Buffer* encoded) override;
 
  private:
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index a8e1bff186..451fe8b9e2 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -17,6 +17,7 @@
 #include <iterator>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
@@ -24,7 +25,6 @@
 #include "absl/memory/memory.h"
 #include "absl/strings/match.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/audio_codecs/opus/audio_encoder_opus_config.h"
@@ -583,7 +583,7 @@ void AudioEncoderOpusImpl::SetReceiverFrameLengthRange(
 
 AudioEncoder::EncodedInfo AudioEncoderOpusImpl::EncodeImpl(
     uint32_t rtp_timestamp,
-    ArrayView<const int16_t> audio,
+    std::span<const int16_t> audio,
     Buffer* encoded) {
   MaybeUpdateUplinkBandwidth();
 
@@ -601,7 +601,7 @@ AudioEncoder::EncodedInfo AudioEncoderOpusImpl::EncodeImpl(
   const size_t max_encoded_bytes = SufficientOutputBufferSize();
   EncodedInfo info;
   info.encoded_bytes =
-      encoded->AppendData(max_encoded_bytes, [&](ArrayView<uint8_t> encoded) {
+      encoded->AppendData(max_encoded_bytes, [&](std::span<uint8_t> encoded) {
         int status = WebRtcOpus_Encode(
             inst_, &input_buffer_[0],
             CheckedDivExact(input_buffer_.size(), config_.num_channels),
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/modules/audio_coding/codecs/opus/audio_encoder_opus.h
index 5b873d9186..6900e0e61e 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus.h
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus.h
@@ -16,11 +16,11 @@
 #include <functional>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/audio_codecs/opus/audio_encoder_opus_config.h"
@@ -102,7 +102,7 @@ class AudioEncoderOpusImpl final : public AudioEncoder {
   ANAStats GetANAStats() const override;
   std::optional<std::pair<TimeDelta, TimeDelta> > GetFrameLengthRange()
       const override;
-  ArrayView<const int> supported_frame_lengths_ms() const {
+  std::span<const int> supported_frame_lengths_ms() const {
     return config_.supported_frame_lengths_ms;
   }
 
@@ -117,7 +117,7 @@ class AudioEncoderOpusImpl final : public AudioEncoder {
 
  protected:
   EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
-                         ArrayView<const int16_t> audio,
+                         std::span<const int16_t> audio,
                          Buffer* encoded) override;
 
  private:
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
index 1fcf7a70fc..f9ead462eb 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
@@ -14,12 +14,12 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/audio_codecs/opus/audio_encoder_opus_config.h"
@@ -512,7 +512,7 @@ TEST_P(AudioEncoderOpusTest, UpdateUplinkBandwidthInAudioNetworkAdaptor) {
       .WillOnce(Return(50000));
   EXPECT_CALL(*states->mock_audio_network_adaptor, SetUplinkBandwidth(50000));
   states->encoder->Encode(
-      0, ArrayView<const int16_t>(audio.data(), audio.size()), &encoded);
+      0, std::span<const int16_t>(audio.data(), audio.size()), &encoded);
 
   // Repeat update uplink bandwidth tests.
   for (int i = 0; i < 5; i++) {
@@ -520,7 +520,7 @@ TEST_P(AudioEncoderOpusTest, UpdateUplinkBandwidthInAudioNetworkAdaptor) {
     states->fake_clock->AdvanceTime(
         TimeDelta::Millis(states->uplink_bandwidth_update_interval_ms - 1));
     states->encoder->Encode(
-        0, ArrayView<const int16_t>(audio.data(), audio.size()), &encoded);
+        0, std::span<const int16_t>(audio.data(), audio.size()), &encoded);
 
     // Update when it is time to update.
     EXPECT_CALL(*states->mock_bitrate_smoother, GetAverage)
@@ -528,7 +528,7 @@ TEST_P(AudioEncoderOpusTest, UpdateUplinkBandwidthInAudioNetworkAdaptor) {
     EXPECT_CALL(*states->mock_audio_network_adaptor, SetUplinkBandwidth(40000));
     states->fake_clock->AdvanceTime(TimeDelta::Millis(1));
     states->encoder->Encode(
-        0, ArrayView<const int16_t>(audio.data(), audio.size()), &encoded);
+        0, std::span<const int16_t>(audio.data(), audio.size()), &encoded);
   }
 }
 
diff --git a/modules/audio_coding/codecs/opus/opus_unittest.cc b/modules/audio_coding/codecs/opus/opus_unittest.cc
index 3d86918339..298049aa0d 100644
--- a/modules/audio_coding/codecs/opus/opus_unittest.cc
+++ b/modules/audio_coding/codecs/opus/opus_unittest.cc
@@ -16,11 +16,11 @@
 #include <cstdlib>
 #include <map>
 #include <memory>
+#include <span>
 #include <string>
 #include <tuple>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/audio_coding/codecs/opus/opus_interface.h"
 #include "modules/audio_coding/neteq/tools/audio_loop.h"
 #include "rtc_base/checks.h"
@@ -131,7 +131,7 @@ class OpusTest
   void PrepareSpeechData(int block_length_ms, int loop_length_ms);
 
   int EncodeDecode(WebRtcOpusEncInst* encoder,
-                   ArrayView<const int16_t> input_audio,
+                   std::span<const int16_t> input_audio,
                    WebRtcOpusDecInst* decoder,
                    int16_t* output_audio,
                    int16_t* audio_type);
@@ -217,7 +217,7 @@ void OpusTest::CheckAudioBounded(const int16_t* audio,
 }
 
 int OpusTest::EncodeDecode(WebRtcOpusEncInst* encoder,
-                           ArrayView<const int16_t> input_audio,
+                           std::span<const int16_t> input_audio,
                            WebRtcOpusDecInst* decoder,
                            int16_t* output_audio,
                            int16_t* audio_type) {
diff --git a/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc b/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
index b77f891c42..b62d5f2b21 100644
--- a/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
+++ b/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
@@ -16,12 +16,12 @@
 #include <iterator>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/call/bitrate_allocation.h"
 #include "api/field_trials_view.h"
@@ -106,7 +106,7 @@ int AudioEncoderCopyRed::GetTargetBitrate() const {
 
 AudioEncoder::EncodedInfo AudioEncoderCopyRed::EncodeImpl(
     uint32_t rtp_timestamp,
-    ArrayView<const int16_t> audio,
+    std::span<const int16_t> audio,
     Buffer* encoded) {
   primary_encoded_.Clear();
   EncodedInfo info =
@@ -160,7 +160,7 @@ AudioEncoder::EncodedInfo AudioEncoderCopyRed::EncodeImpl(
     const uint32_t timestamp_delta =
         info.encoded_timestamp - it->first.encoded_timestamp;
     encoded->data()[header_offset] = it->first.payload_type | 0x80;
-    SetBE16(ArrayView<uint8_t>(*encoded).subspan(header_offset + 1, 2),
+    SetBE16(std::span<uint8_t>(*encoded).subspan(header_offset + 1, 2),
             (timestamp_delta << 2) | (it->first.encoded_bytes >> 8));
     encoded->data()[header_offset + 3] = it->first.encoded_bytes & 0xff;
     header_offset += kRedHeaderLength;
@@ -282,9 +282,9 @@ ANAStats AudioEncoderCopyRed::GetANAStats() const {
   return speech_encoder_->GetANAStats();
 }
 
-ArrayView<std::unique_ptr<AudioEncoder>>
+std::span<std::unique_ptr<AudioEncoder>>
 AudioEncoderCopyRed::ReclaimContainedEncoders() {
-  return ArrayView<std::unique_ptr<AudioEncoder>>(&speech_encoder_, 1);
+  return std::span<std::unique_ptr<AudioEncoder>>(&speech_encoder_, 1);
 }
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/codecs/red/audio_encoder_copy_red.h b/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
index 7d147802ff..070a242800 100644
--- a/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
+++ b/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
@@ -17,10 +17,10 @@
 #include <list>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/call/bitrate_allocation.h"
 #include "api/field_trials_view.h"
@@ -81,11 +81,11 @@ class AudioEncoderCopyRed final : public AudioEncoder {
   ANAStats GetANAStats() const override;
   std::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
       const override;
-  ArrayView<std::unique_ptr<AudioEncoder>> ReclaimContainedEncoders() override;
+  std::span<std::unique_ptr<AudioEncoder>> ReclaimContainedEncoders() override;
 
  protected:
   EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
-                         ArrayView<const int16_t> audio,
+                         std::span<const int16_t> audio,
                          Buffer* encoded) override;
 
  private:
diff --git a/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc b/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
index b1805f4459..c7ac10f0bf 100644
--- a/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
+++ b/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
@@ -15,11 +15,11 @@
 #include <cstring>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/field_trials.h"
 #include "api/units/time_delta.h"
@@ -71,7 +71,7 @@ class AudioEncoderCopyRedTest : public ::testing::Test {
     ASSERT_TRUE(red_.get() != nullptr);
     encoded_.Clear();
     encoded_info_ = red_->Encode(
-        timestamp_, ArrayView<const int16_t>(audio_, num_audio_samples_10ms),
+        timestamp_, std::span<const int16_t>(audio_, num_audio_samples_10ms),
         &encoded_);
     timestamp_ += checked_cast<uint32_t>(num_audio_samples_10ms);
   }
diff --git a/modules/audio_coding/neteq/accelerate.cc b/modules/audio_coding/neteq/accelerate.cc
index eda4370541..38bbdd695d 100644
--- a/modules/audio_coding/neteq/accelerate.cc
+++ b/modules/audio_coding/neteq/accelerate.cc
@@ -12,8 +12,8 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_coding/neteq/audio_multi_vector.h"
 #include "modules/audio_coding/neteq/time_stretch.h"
 #include "rtc_base/checks.h"
@@ -31,7 +31,7 @@ Accelerate::ReturnCodes Accelerate::Process(const int16_t* input,
       input_length / num_channels_ < (2 * k15ms - 1) * fs_mult_) {
     // Length of input data too short to do accelerate. Simply move all data
     // from input to output.
-    output->PushBackInterleaved(ArrayView<const int16_t>(input, input_length));
+    output->PushBackInterleaved(std::span<const int16_t>(input, input_length));
     return kError;
   }
   return TimeStretch::Process(input, input_length, fast_accelerate, output,
@@ -74,15 +74,15 @@ Accelerate::ReturnCodes Accelerate::CheckCriteriaAndStretch(
     RTC_DCHECK_GE(fs_mult_120, peak_index);  // Should be handled in Process().
     // Copy first part; 0 to 15 ms.
     output->PushBackInterleaved(
-        ArrayView<const int16_t>(input, fs_mult_120 * num_channels_));
+        std::span<const int16_t>(input, fs_mult_120 * num_channels_));
     // Copy the `peak_index` starting at 15 ms to `temp_vector`.
     AudioMultiVector temp_vector(num_channels_);
-    temp_vector.PushBackInterleaved(ArrayView<const int16_t>(
+    temp_vector.PushBackInterleaved(std::span<const int16_t>(
         &input[fs_mult_120 * num_channels_], peak_index * num_channels_));
     // Cross-fade `temp_vector` onto the end of `output`.
     output->CrossFade(temp_vector, peak_index);
     // Copy the last unmodified part, 15 ms + pitch period until the end.
-    output->PushBackInterleaved(ArrayView<const int16_t>(
+    output->PushBackInterleaved(std::span<const int16_t>(
         &input[(fs_mult_120 + peak_index) * num_channels_],
         input_length - (fs_mult_120 + peak_index) * num_channels_));
 
@@ -93,7 +93,7 @@ Accelerate::ReturnCodes Accelerate::CheckCriteriaAndStretch(
     }
   } else {
     // Accelerate not allowed. Simply move all data from decoded to outData.
-    output->PushBackInterleaved(ArrayView<const int16_t>(input, input_length));
+    output->PushBackInterleaved(std::span<const int16_t>(input, input_length));
     return kNoStretch;
   }
 }
diff --git a/modules/audio_coding/neteq/audio_decoder_unittest.cc b/modules/audio_coding/neteq/audio_decoder_unittest.cc
index 34dd59276d..0503dda2a6 100644
--- a/modules/audio_coding/neteq/audio_decoder_unittest.cc
+++ b/modules/audio_coding/neteq/audio_decoder_unittest.cc
@@ -15,11 +15,11 @@
 #include <cstdlib>
 #include <memory>
 #include <optional>
+#include <span>
 #include <tuple>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/g722/audio_encoder_g722_config.h"
 #include "api/audio_codecs/opus/audio_encoder_opus.h"
@@ -147,7 +147,7 @@ class AudioDecoderTest : public ::testing::Test {
 
       encoded_info = audio_encoder_->Encode(
           0,
-          ArrayView<const int16_t>(interleaved_input.get(),
+          std::span<const int16_t>(interleaved_input.get(),
                                    audio_encoder_->NumChannels() *
                                        audio_encoder_->SampleRateHz() / 100),
           output);
@@ -191,7 +191,7 @@ class AudioDecoderTest : public ::testing::Test {
           decoder_->ParsePayload(std::move(encoded), /*timestamp=*/0);
       RTC_CHECK_EQ(parse_result.size(), size_t{1});
       auto decode_result = parse_result[0].frame->Decode(
-          ArrayView<int16_t>(&decoded[processed_samples * channels_],
+          std::span<int16_t>(&decoded[processed_samples * channels_],
                              frame_size_ * channels_ * sizeof(int16_t)));
       RTC_CHECK(decode_result.has_value());
       EXPECT_EQ(frame_size_ * channels_, decode_result->num_decoded_samples);
diff --git a/modules/audio_coding/neteq/audio_multi_vector.cc b/modules/audio_coding/neteq/audio_multi_vector.cc
index 6ba4bc37f6..d0d1ddc1fa 100644
--- a/modules/audio_coding/neteq/audio_multi_vector.cc
+++ b/modules/audio_coding/neteq/audio_multi_vector.cc
@@ -14,9 +14,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/audio_view.h"
 #include "modules/audio_coding/neteq/audio_vector.h"
 #include "rtc_base/checks.h"
@@ -67,7 +67,7 @@ void AudioMultiVector::CopyTo(AudioMultiVector* copy_to) const {
 }
 
 void AudioMultiVector::PushBackInterleaved(
-    ArrayView<const int16_t> append_this) {
+    std::span<const int16_t> append_this) {
   RTC_DCHECK_EQ(append_this.size() % Channels(), 0);
   if (append_this.empty()) {
     return;
diff --git a/modules/audio_coding/neteq/audio_multi_vector.h b/modules/audio_coding/neteq/audio_multi_vector.h
index 486c13ab54..12b8f94799 100644
--- a/modules/audio_coding/neteq/audio_multi_vector.h
+++ b/modules/audio_coding/neteq/audio_multi_vector.h
@@ -15,9 +15,9 @@
 #include <string.h>
 
 #include <memory>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/audio_view.h"
 #include "modules/audio_coding/neteq/audio_vector.h"
 
@@ -55,7 +55,7 @@ class AudioMultiVector {
   // is assumed to be channel-interleaved. The length must be an even multiple
   // of this object's number of channels. The length of this object is increased
   // with the length of the array divided by the number of channels.
-  void PushBackInterleaved(ArrayView<const int16_t> append_this);
+  void PushBackInterleaved(std::span<const int16_t> append_this);
 
   // Appends the contents of AudioMultiVector `append_this` to this object. The
   // length of this object is increased with the length of `append_this`.
diff --git a/modules/audio_coding/neteq/background_noise.cc b/modules/audio_coding/neteq/background_noise.cc
index 8c5102c77f..ce3d343055 100644
--- a/modules/audio_coding/neteq/background_noise.cc
+++ b/modules/audio_coding/neteq/background_noise.cc
@@ -13,8 +13,8 @@
 #include <algorithm>  // min, max
 #include <cstdint>
 #include <cstring>  // memcpy
+#include <span>
 
-#include "api/array_view.h"
 #include "common_audio/signal_processing/dot_product_with_scale.h"
 #include "common_audio/signal_processing/include/signal_processing_library.h"
 #include "common_audio/signal_processing/include/spl_inl.h"
@@ -119,7 +119,7 @@ bool BackgroundNoise::Update(const AudioMultiVector& sync_buffer) {
 }
 
 void BackgroundNoise::GenerateBackgroundNoise(
-    ArrayView<const int16_t> random_vector,
+    std::span<const int16_t> random_vector,
     size_t channel,
     int /* mute_slope */,
     bool /* too_many_expands */,
@@ -194,7 +194,7 @@ const int16_t* BackgroundNoise::FilterState(size_t channel) const {
 }
 
 void BackgroundNoise::SetFilterState(size_t channel,
-                                     ArrayView<const int16_t> input) {
+                                     std::span<const int16_t> input) {
   RTC_DCHECK_LT(channel, num_channels_);
   size_t length = std::min(input.size(), kMaxLpcOrder);
   memcpy(channel_parameters_[channel].filter_state, input.data(),
diff --git a/modules/audio_coding/neteq/background_noise.h b/modules/audio_coding/neteq/background_noise.h
index 1a506a195b..00b7091101 100644
--- a/modules/audio_coding/neteq/background_noise.h
+++ b/modules/audio_coding/neteq/background_noise.h
@@ -15,8 +15,7 @@
 
 #include <cstdint>
 #include <memory>
-
-#include "api/array_view.h"
+#include <span>
 
 namespace webrtc {
 
@@ -46,7 +45,7 @@ class BackgroundNoise {
 
   // Generates background noise given a random vector and writes the output to
   // `buffer`.
-  void GenerateBackgroundNoise(ArrayView<const int16_t> random_vector,
+  void GenerateBackgroundNoise(std::span<const int16_t> random_vector,
                                size_t channel,
                                int mute_slope,
                                bool too_many_expands,
@@ -70,7 +69,7 @@ class BackgroundNoise {
 
   // Copies `input` to the filter state. Will not copy more than `kMaxLpcOrder`
   // elements.
-  void SetFilterState(size_t channel, ArrayView<const int16_t> input);
+  void SetFilterState(size_t channel, std::span<const int16_t> input);
 
   // Returns `scale_` for `channel`.
   int16_t Scale(size_t channel) const;
diff --git a/modules/audio_coding/neteq/comfort_noise.cc b/modules/audio_coding/neteq/comfort_noise.cc
index da94881585..1b57145b6c 100644
--- a/modules/audio_coding/neteq/comfort_noise.cc
+++ b/modules/audio_coding/neteq/comfort_noise.cc
@@ -13,8 +13,8 @@
 #include <cstddef>
 #include <cstdint>
 #include <memory>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_coding/codecs/cng/webrtc_cng.h"
 #include "modules/audio_coding/neteq/audio_multi_vector.h"
 #include "modules/audio_coding/neteq/audio_vector.h"
@@ -67,7 +67,7 @@ int ComfortNoise::Generate(size_t requested_length, AudioMultiVector* output) {
   }
 
   std::unique_ptr<int16_t[]> temp(new int16_t[number_of_samples]);
-  if (!cng_decoder->Generate(ArrayView<int16_t>(temp.get(), number_of_samples),
+  if (!cng_decoder->Generate(std::span<int16_t>(temp.get(), number_of_samples),
                              new_period)) {
     // Error returned.
     output->Zeros(requested_length);
diff --git a/modules/audio_coding/neteq/merge.cc b/modules/audio_coding/neteq/merge.cc
index 8abaff7dbb..ac1f51d0b7 100644
--- a/modules/audio_coding/neteq/merge.cc
+++ b/modules/audio_coding/neteq/merge.cc
@@ -15,8 +15,8 @@
 #include <cstring>  // memmove, memcpy, memset, size_t
 #include <limits>
 #include <memory>
+#include <span>
 
-#include "api/array_view.h"
 #include "common_audio/signal_processing/dot_product_with_scale.h"
 #include "common_audio/signal_processing/include/signal_processing_library.h"
 #include "common_audio/signal_processing/include/spl_inl.h"
@@ -66,7 +66,7 @@ size_t Merge::Process(int16_t* input,
   // Transfer input signal to an AudioMultiVector.
   AudioMultiVector input_vector(num_channels_);
   input_vector.PushBackInterleaved(
-      ArrayView<const int16_t>(input, input_length));
+      std::span<const int16_t>(input, input_length));
   size_t input_length_per_channel = input_vector.Size();
   RTC_DCHECK_EQ(input_length_per_channel, input_length / num_channels_);
 
diff --git a/modules/audio_coding/neteq/neteq_decoder_plc_unittest.cc b/modules/audio_coding/neteq/neteq_decoder_plc_unittest.cc
index 632d6921fb..116892a79d 100644
--- a/modules/audio_coding/neteq/neteq_decoder_plc_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_decoder_plc_unittest.cc
@@ -14,11 +14,11 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_decoder.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/make_ref_counted.h"
@@ -107,9 +107,9 @@ class AudioDecoderPlc : public AudioDecoder {
 // An input sample generator which generates only zero-samples.
 class ZeroSampleGenerator : public EncodeNetEqInput::Generator {
  public:
-  ArrayView<const int16_t> Generate(size_t num_samples) override {
+  std::span<const int16_t> Generate(size_t num_samples) override {
     vec.resize(num_samples, 0);
-    ArrayView<const int16_t> view(vec);
+    std::span<const int16_t> view(vec);
     RTC_DCHECK_EQ(view.size(), num_samples);
     return view;
   }
diff --git a/modules/audio_coding/neteq/neteq_impl.cc b/modules/audio_coding/neteq/neteq_impl.cc
index f0e96564f7..81420261c9 100644
--- a/modules/audio_coding/neteq/neteq_impl.cc
+++ b/modules/audio_coding/neteq/neteq_impl.cc
@@ -17,11 +17,11 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/str_cat.h"
-#include "api/array_view.h"
 #include "api/audio/audio_frame.h"
 #include "api/audio/audio_view.h"
 #include "api/audio_codecs/audio_decoder.h"
@@ -180,7 +180,7 @@ NetEqImpl::NetEqImpl(const NetEq::Config& config,
 NetEqImpl::~NetEqImpl() = default;
 
 int NetEqImpl::InsertPacket(const RTPHeader& rtp_header,
-                            ArrayView<const uint8_t> payload,
+                            std::span<const uint8_t> payload,
                             const RtpPacketInfo& packet_info) {
   MsanCheckInitialized(payload);
   TRACE_EVENT0("webrtc", "NetEqImpl::InsertPacket");
@@ -1340,7 +1340,7 @@ int NetEqImpl::DecodeLoop(PacketList* packet_list,
                operation == Operation::kPreemptiveExpand);
 
     auto opt_result = packet_list->front().frame->Decode(
-        ArrayView<int16_t>(&decoded_buffer_[*decoded_length],
+        std::span<int16_t>(&decoded_buffer_[*decoded_length],
                            decoded_buffer_length_ - *decoded_length));
     if (packet_list->front().packet_info) {
       last_decoded_packet_infos_.push_back(*packet_list->front().packet_info);
diff --git a/modules/audio_coding/neteq/neteq_impl.h b/modules/audio_coding/neteq/neteq_impl.h
index c80ad4fa41..c02282a1be 100644
--- a/modules/audio_coding/neteq/neteq_impl.h
+++ b/modules/audio_coding/neteq/neteq_impl.h
@@ -16,9 +16,9 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/audio_frame.h"
 #include "api/audio_codecs/audio_decoder.h"
 #include "api/audio_codecs/audio_decoder_factory.h"
@@ -130,7 +130,7 @@ class NetEqImpl : public NetEq {
   NetEqImpl& operator=(const NetEqImpl&) = delete;
 
   int InsertPacket(const RTPHeader& rtp_header,
-                   ArrayView<const uint8_t> payload) override {
+                   std::span<const uint8_t> payload) override {
     return InsertPacket(
         rtp_header, payload,
         RtpPacketInfo(rtp_header, /*receive_time=*/Timestamp::MinusInfinity()));
@@ -138,7 +138,7 @@ class NetEqImpl : public NetEq {
 
   // Inserts a new packet into NetEq. Returns 0 on success, -1 on failure.
   int InsertPacket(const RTPHeader& rtp_header,
-                   ArrayView<const uint8_t> payload,
+                   std::span<const uint8_t> payload,
                    const RtpPacketInfo& packet_info) override;
 
   int GetAudio(
diff --git a/modules/audio_coding/neteq/neteq_network_stats_unittest.cc b/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
index fb0d8e5d4b..79e8df0659 100644
--- a/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
@@ -13,10 +13,10 @@
 #include <cstring>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/audio_frame.h"
 #include "api/audio_codecs/audio_decoder.h"
 #include "api/audio_codecs/audio_decoder_factory.h"
@@ -76,7 +76,7 @@ class MockAudioDecoder final : public AudioDecoder {
     size_t Duration() const override { return kPacketDuration; }
 
     std::optional<DecodeResult> Decode(
-        ArrayView<int16_t> decoded) const override {
+        std::span<int16_t> decoded) const override {
       const size_t output_size =
           sizeof(int16_t) * kPacketDuration * num_channels_;
       if (decoded.size() >= output_size) {
diff --git a/modules/audio_coding/neteq/neteq_stereo_unittest.cc b/modules/audio_coding/neteq/neteq_stereo_unittest.cc
index bdb0cfea23..54c81b9869 100644
--- a/modules/audio_coding/neteq/neteq_stereo_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_stereo_unittest.cc
@@ -17,9 +17,9 @@
 #include <list>
 #include <memory>
 #include <ostream>
+#include <span>
 #include <string>
 
-#include "api/array_view.h"
 #include "api/audio/audio_frame.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
@@ -178,13 +178,13 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
         ASSERT_EQ(NetEq::kOK,
                   neteq_mono_->InsertPacket(
                       rtp_header_mono_,
-                      ArrayView<const uint8_t>(encoded_, payload_size_bytes_),
+                      std::span<const uint8_t>(encoded_, payload_size_bytes_),
                       Timestamp::Millis(time_now_ms)));
         // Insert packet in multi-channel instance.
         ASSERT_EQ(NetEq::kOK,
                   neteq_->InsertPacket(
                       rtp_header_,
-                      ArrayView<const uint8_t>(encoded_multi_channel_,
+                      std::span<const uint8_t>(encoded_multi_channel_,
                                                multi_payload_size_bytes_),
                       Timestamp::Millis(time_now_ms)));
         // Get next input packets (mono and multi-channel).
diff --git a/modules/audio_coding/neteq/neteq_unittest.cc b/modules/audio_coding/neteq/neteq_unittest.cc
index 897532fc51..4a25d56876 100644
--- a/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_unittest.cc
@@ -18,11 +18,11 @@
 #include <memory>
 #include <optional>
 #include <set>
+#include <span>
 #include <string>
 #include <utility>
 
 #include "absl/flags/flag.h"
-#include "api/array_view.h"
 #include "api/audio/audio_frame.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "api/rtp_headers.h"
@@ -341,7 +341,7 @@ class NetEqBgnTest : public NetEqDecodingTest {
 
       ASSERT_EQ(0,
                 neteq_->InsertPacket(
-                    rtp_info, ArrayView<const uint8_t>(payload, enc_len_bytes),
+                    rtp_info, std::span<const uint8_t>(payload, enc_len_bytes),
                     clock_.CurrentTime()));
       output.Reset();
       ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
@@ -464,7 +464,7 @@ TEST_F(NetEqDecodingTest, DiscardDuplicateCng) {
   PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
   // This is the first time this CNG packet is inserted.
   ASSERT_EQ(0, neteq_->InsertPacket(
-                   rtp_info, ArrayView<const uint8_t>(payload, payload_len),
+                   rtp_info, std::span<const uint8_t>(payload, payload_len),
                    clock_.CurrentTime()));
 
   // Pull audio once and make sure CNG is played.
@@ -479,7 +479,7 @@ TEST_F(NetEqDecodingTest, DiscardDuplicateCng) {
   // Insert the same CNG packet again. Note that at this point it is old, since
   // we have already decoded the first copy of it.
   ASSERT_EQ(0, neteq_->InsertPacket(
-                   rtp_info, ArrayView<const uint8_t>(payload, payload_len),
+                   rtp_info, std::span<const uint8_t>(payload, payload_len),
                    clock_.CurrentTime()));
 
   // Pull audio until we have played `kCngPeriodMs` of CNG. Start at 10 ms since
@@ -532,7 +532,7 @@ TEST_F(NetEqDecodingTest, CngFirst) {
   PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
   ASSERT_EQ(NetEq::kOK,
             neteq_->InsertPacket(rtp_info,
-                                 ArrayView<const uint8_t>(payload, payload_len),
+                                 std::span<const uint8_t>(payload, payload_len),
                                  clock_.CurrentTime()));
   ++seq_no;
   timestamp += kCngPeriodSamples;
@@ -585,7 +585,7 @@ class NetEqDecodingTestWithMutedState : public NetEqDecodingTest {
     PopulateCng(0, rtp_timestamp, &rtp_info, payload, &payload_len);
     EXPECT_EQ(NetEq::kOK,
               neteq_->InsertPacket(
-                  rtp_info, ArrayView<const uint8_t>(payload, payload_len),
+                  rtp_info, std::span<const uint8_t>(payload, payload_len),
                   clock_.CurrentTime()));
   }
 
diff --git a/modules/audio_coding/neteq/normal.cc b/modules/audio_coding/neteq/normal.cc
index 909e955684..117c480f8c 100644
--- a/modules/audio_coding/neteq/normal.cc
+++ b/modules/audio_coding/neteq/normal.cc
@@ -14,8 +14,8 @@
 #include <cstdint>
 #include <cstring>  // memset, memcpy
 #include <memory>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/neteq/neteq.h"
 #include "common_audio/signal_processing/dot_product_with_scale.h"
 #include "common_audio/signal_processing/include/signal_processing_library.h"
@@ -46,7 +46,7 @@ int Normal::Process(const int16_t* input,
     output->Clear();
     return 0;
   }
-  output->PushBackInterleaved(ArrayView<const int16_t>(input, length));
+  output->PushBackInterleaved(std::span<const int16_t>(input, length));
 
   const int fs_mult = fs_hz_ / 8000;
   RTC_DCHECK_GT(fs_mult, 0);
diff --git a/modules/audio_coding/neteq/packet_buffer_unittest.cc b/modules/audio_coding/neteq/packet_buffer_unittest.cc
index 3e34602e74..5851e4a052 100644
--- a/modules/audio_coding/neteq/packet_buffer_unittest.cc
+++ b/modules/audio_coding/neteq/packet_buffer_unittest.cc
@@ -17,10 +17,10 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_decoder.h"
 #include "api/neteq/tick_timer.h"
 #include "modules/audio_coding/neteq/mock/mock_decoder_database.h"
@@ -47,7 +47,7 @@ class MockEncodedAudioFrame : public AudioDecoder::EncodedAudioFrame {
 
   MOCK_METHOD(std::optional<DecodeResult>,
               Decode,
-              (ArrayView<int16_t> decoded),
+              (std::span<int16_t> decoded),
               (const, override));
 };
 
diff --git a/modules/audio_coding/neteq/preemptive_expand.cc b/modules/audio_coding/neteq/preemptive_expand.cc
index a183c3dd96..1e3976ebf4 100644
--- a/modules/audio_coding/neteq/preemptive_expand.cc
+++ b/modules/audio_coding/neteq/preemptive_expand.cc
@@ -13,8 +13,8 @@
 #include <algorithm>
 #include <cstddef>
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_coding/neteq/audio_multi_vector.h"
 #include "modules/audio_coding/neteq/time_stretch.h"
 
@@ -35,7 +35,7 @@ PreemptiveExpand::ReturnCodes PreemptiveExpand::Process(
       old_data_length >= input_length / num_channels_ - overlap_samples_) {
     // Length of input data too short to do preemptive expand. Simply move all
     // data from input to output.
-    output->PushBackInterleaved(ArrayView<const int16_t>(input, input_length));
+    output->PushBackInterleaved(std::span<const int16_t>(input, input_length));
     return kError;
   }
   const bool kFastMode = false;  // Fast mode is not available for PE Expand.
@@ -79,17 +79,17 @@ PreemptiveExpand::ReturnCodes PreemptiveExpand::CheckCriteriaAndStretch(
     size_t unmodified_length =
         std::max(old_data_length_per_channel_, fs_mult_120);
     // Copy first part, including cross-fade region.
-    output->PushBackInterleaved(ArrayView<const int16_t>(
+    output->PushBackInterleaved(std::span<const int16_t>(
         input, (unmodified_length + peak_index) * num_channels_));
     // Copy the last `peak_index` samples up to 15 ms to `temp_vector`.
     AudioMultiVector temp_vector(num_channels_);
-    temp_vector.PushBackInterleaved(ArrayView<const int16_t>(
+    temp_vector.PushBackInterleaved(std::span<const int16_t>(
         &input[(unmodified_length - peak_index) * num_channels_],
         peak_index * num_channels_));
     // Cross-fade `temp_vector` onto the end of `output`.
     output->CrossFade(temp_vector, peak_index);
     // Copy the last unmodified part, 15 ms + pitch period until the end.
-    output->PushBackInterleaved(ArrayView<const int16_t>(
+    output->PushBackInterleaved(std::span<const int16_t>(
         &input[unmodified_length * num_channels_],
         input_length - unmodified_length * num_channels_));
 
@@ -100,7 +100,7 @@ PreemptiveExpand::ReturnCodes PreemptiveExpand::CheckCriteriaAndStretch(
     }
   } else {
     // Accelerate not allowed. Simply move all data from decoded to outData.
-    output->PushBackInterleaved(ArrayView<const int16_t>(input, input_length));
+    output->PushBackInterleaved(std::span<const int16_t>(input, input_length));
     return kNoStretch;
   }
 }
diff --git a/modules/audio_coding/neteq/test/neteq_decoding_test.cc b/modules/audio_coding/neteq/test/neteq_decoding_test.cc
index f266081479..45c3d6ffc9 100644
--- a/modules/audio_coding/neteq/test/neteq_decoding_test.cc
+++ b/modules/audio_coding/neteq/test/neteq_decoding_test.cc
@@ -14,10 +14,10 @@
 #include <cstdint>
 #include <optional>
 #include <set>
+#include <span>
 #include <string>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio/audio_frame.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
@@ -320,7 +320,7 @@ void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
       RTPHeader rtp_info;
       PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
       ASSERT_EQ(0, neteq_->InsertPacket(
-                       rtp_info, ArrayView<const uint8_t>(payload, payload_len),
+                       rtp_info, std::span<const uint8_t>(payload, payload_len),
                        Timestamp::Millis(t_ms)));
       ++seq_no;
       timestamp += kCngPeriodSamples;
@@ -363,7 +363,7 @@ void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
       RTPHeader rtp_info;
       PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
       ASSERT_EQ(0, neteq_->InsertPacket(
-                       rtp_info, ArrayView<const uint8_t>(payload, payload_len),
+                       rtp_info, std::span<const uint8_t>(payload, payload_len),
                        Timestamp::Millis(t_ms)));
       ++seq_no;
       timestamp += kCngPeriodSamples;
diff --git a/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc b/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc
index ecf1a21937..e1c7e419dc 100644
--- a/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc
+++ b/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc
@@ -11,9 +11,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <ostream>
+#include <span>
 
 #include "absl/flags/flag.h"
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/rtp_parameters.h"
 #include "modules/audio_coding/codecs/opus/opus_inst.h"
@@ -161,7 +161,7 @@ int NetEqOpusQualityTest::EncodeBlock(int16_t* in_data,
   int value;
   opus_repacketizer_init(repacketizer_);
   for (int idx = 0; idx < sub_packets_; idx++) {
-    payload->AppendData(max_bytes, [&](ArrayView<uint8_t> payload) {
+    payload->AppendData(max_bytes, [&](std::span<uint8_t> payload) {
       value = WebRtcOpus_Encode(opus_encoder_, pointer, sub_block_size_samples_,
                                 max_bytes, payload.data());
 
diff --git a/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc b/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc
index d1f6cf7907..b709139713 100644
--- a/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc
+++ b/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc
@@ -11,9 +11,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <memory>
+#include <span>
 
 #include "absl/flags/flag.h"
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/audio_format.h"
 #include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
@@ -67,7 +67,7 @@ class NetEqPcm16bQualityTest : public NetEqQualityTest {
     AudioEncoder::EncodedInfo info;
     do {
       info = encoder_->Encode(dummy_timestamp,
-                              ArrayView<const int16_t>(
+                              std::span<const int16_t>(
                                   in_data + encoded_samples, kFrameSizeSamples),
                               payload);
       encoded_samples += kFrameSizeSamples;
diff --git a/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc b/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
index a4a624d3da..b060a92381 100644
--- a/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
+++ b/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
@@ -11,9 +11,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <memory>
+#include <span>
 
 #include "absl/flags/flag.h"
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/audio_format.h"
 #include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
@@ -66,7 +66,7 @@ class NetEqPcmuQualityTest : public NetEqQualityTest {
     AudioEncoder::EncodedInfo info;
     do {
       info = encoder_->Encode(dummy_timestamp,
-                              ArrayView<const int16_t>(
+                              std::span<const int16_t>(
                                   in_data + encoded_samples, kFrameSizeSamples),
                               payload);
       encoded_samples += kFrameSizeSamples;
diff --git a/modules/audio_coding/neteq/tools/audio_checksum.h b/modules/audio_coding/neteq/tools/audio_checksum.h
index b7ce1493df..de1c439522 100644
--- a/modules/audio_coding/neteq/tools/audio_checksum.h
+++ b/modules/audio_coding/neteq/tools/audio_checksum.h
@@ -14,9 +14,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <string>
 
-#include "api/array_view.h"
 #include "modules/audio_coding/neteq/tools/audio_sink.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/message_digest.h"
@@ -52,7 +52,7 @@ class AudioChecksum : public AudioSink {
     if (!finished_) {
       finished_ = true;
       checksum_result_.AppendData(checksum_->Size(),
-                                  [&](ArrayView<uint8_t> view) {
+                                  [&](std::span<uint8_t> view) {
                                     checksum_->Finish(view.data(), view.size());
                                     return view.size();
                                   });
diff --git a/modules/audio_coding/neteq/tools/audio_loop.cc b/modules/audio_coding/neteq/tools/audio_loop.cc
index ba67a66709..a7353b3100 100644
--- a/modules/audio_coding/neteq/tools/audio_loop.cc
+++ b/modules/audio_coding/neteq/tools/audio_loop.cc
@@ -13,10 +13,10 @@
 #include <cstdint>
 #include <cstdio>
 #include <cstring>
+#include <span>
 #include <string>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 
 namespace webrtc {
 namespace test {
@@ -50,14 +50,14 @@ bool AudioLoop::Init(absl::string_view file_name,
   return true;
 }
 
-ArrayView<const int16_t> AudioLoop::GetNextBlock() {
+std::span<const int16_t> AudioLoop::GetNextBlock() {
   // Check that the AudioLoop is initialized.
   if (block_length_samples_ == 0)
-    return ArrayView<const int16_t>();
+    return std::span<const int16_t>();
 
   const int16_t* output_ptr = &audio_array_[next_index_];
   next_index_ = (next_index_ + block_length_samples_) % loop_length_samples_;
-  return ArrayView<const int16_t>(output_ptr, block_length_samples_);
+  return std::span<const int16_t>(output_ptr, block_length_samples_);
 }
 
 }  // namespace test
diff --git a/modules/audio_coding/neteq/tools/audio_loop.h b/modules/audio_coding/neteq/tools/audio_loop.h
index d5c561d623..b00d50a503 100644
--- a/modules/audio_coding/neteq/tools/audio_loop.h
+++ b/modules/audio_coding/neteq/tools/audio_loop.h
@@ -14,9 +14,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <memory>
+#include <span>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 
 namespace webrtc {
 namespace test {
@@ -44,7 +44,7 @@ class AudioLoop {
 
   // Returns a (pointer,size) pair for the next block of audio. The size is
   // equal to the `block_length_samples` Init() argument.
-  ArrayView<const int16_t> GetNextBlock();
+  std::span<const int16_t> GetNextBlock();
 
  private:
   size_t next_index_;
diff --git a/modules/audio_coding/neteq/tools/encode_neteq_input.h b/modules/audio_coding/neteq/tools/encode_neteq_input.h
index 66927d2f15..d9b1cd3733 100644
--- a/modules/audio_coding/neteq/tools/encode_neteq_input.h
+++ b/modules/audio_coding/neteq/tools/encode_neteq_input.h
@@ -15,8 +15,8 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "modules/audio_coding/neteq/tools/neteq_input.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
@@ -33,7 +33,7 @@ class EncodeNetEqInput : public NetEqInput {
    public:
     virtual ~Generator() = default;
     // Returns the next num_samples values from the signal generator.
-    virtual ArrayView<const int16_t> Generate(size_t num_samples) = 0;
+    virtual std::span<const int16_t> Generate(size_t num_samples) = 0;
   };
 
   // The source will end after the given input duration.
diff --git a/modules/audio_coding/neteq/tools/fake_decode_from_file.cc b/modules/audio_coding/neteq/tools/fake_decode_from_file.cc
index c2a9c7752f..e94c4596f5 100644
--- a/modules/audio_coding/neteq/tools/fake_decode_from_file.cc
+++ b/modules/audio_coding/neteq/tools/fake_decode_from_file.cc
@@ -15,9 +15,9 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_decoder.h"
 #include "modules/audio_coding/neteq/tools/input_audio_file.h"
 #include "modules/rtp_rtcp/source/byte_io.h"
@@ -44,7 +44,7 @@ class FakeEncodedFrame : public AudioDecoder::EncodedAudioFrame {
   size_t Duration() const override { return duration_; }
 
   std::optional<DecodeResult> Decode(
-      ArrayView<int16_t> decoded) const override {
+      std::span<int16_t> decoded) const override {
     if (is_dtx_) {
       std::fill_n(decoded.data(), duration_, 0);
       return DecodeResult{.num_decoded_samples = duration_,
@@ -105,7 +105,7 @@ int FakeDecodeFromFile::DecodeInternal(const uint8_t* encoded,
 void FakeDecodeFromFile::PrepareEncoded(uint32_t timestamp,
                                         size_t samples,
                                         size_t original_payload_size_bytes,
-                                        ArrayView<uint8_t> encoded) {
+                                        std::span<uint8_t> encoded) {
   RTC_CHECK_GE(encoded.size(), 12);
   ByteWriter<uint32_t>::WriteLittleEndian(&encoded[0], timestamp);
   ByteWriter<uint32_t>::WriteLittleEndian(&encoded[4],
diff --git a/modules/audio_coding/neteq/tools/fake_decode_from_file.h b/modules/audio_coding/neteq/tools/fake_decode_from_file.h
index 39e4a5eb74..01c73c45aa 100644
--- a/modules/audio_coding/neteq/tools/fake_decode_from_file.h
+++ b/modules/audio_coding/neteq/tools/fake_decode_from_file.h
@@ -15,10 +15,10 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_decoder.h"
 #include "modules/audio_coding/neteq/tools/input_audio_file.h"
 #include "rtc_base/buffer.h"
@@ -68,7 +68,7 @@ class FakeDecodeFromFile : public AudioDecoder {
   static void PrepareEncoded(uint32_t timestamp,
                              size_t samples,
                              size_t original_payload_size_bytes,
-                             ArrayView<uint8_t> encoded);
+                             std::span<uint8_t> encoded);
 
  private:
   std::unique_ptr<InputAudioFile> input_;
diff --git a/modules/audio_coding/neteq/tools/neteq_event_log_input.cc b/modules/audio_coding/neteq/tools/neteq_event_log_input.cc
index 66494d5741..f7834c6d67 100644
--- a/modules/audio_coding/neteq/tools/neteq_event_log_input.cc
+++ b/modules/audio_coding/neteq/tools/neteq_event_log_input.cc
@@ -15,10 +15,10 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/rtp_headers.h"
 #include "logging/rtc_event_log/events/logged_rtp_rtcp.h"
 #include "logging/rtc_event_log/events/rtc_event_audio_playout.h"
@@ -130,7 +130,7 @@ class NetEqEventLogInput : public NetEqInput {
     packet_data->SetTimestamp(logged.header.timestamp);
     packet_data->SetSsrc(logged.header.ssrc);
     packet_data->SetCsrcs(
-        MakeArrayView(logged.header.arrOfCSRCs, logged.header.numCSRCs));
+        std::span(logged.header.arrOfCSRCs, logged.header.numCSRCs));
     packet_data->set_arrival_time(logged.log_time());
 
     // This is a header-only "dummy" packet. Set the payload to all zeros, with
diff --git a/modules/audio_coding/neteq/tools/neteq_quality_test.cc b/modules/audio_coding/neteq/tools/neteq_quality_test.cc
index 68e4fac1dc..666d2e993e 100644
--- a/modules/audio_coding/neteq/tools/neteq_quality_test.cc
+++ b/modules/audio_coding/neteq/tools/neteq_quality_test.cc
@@ -21,6 +21,7 @@
 #include <memory>
 #include <ostream>
 #include <set>
+#include <span>
 #include <sstream>
 #include <string>
 #include <utility>
@@ -28,7 +29,6 @@
 
 #include "absl/flags/flag.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_decoder_factory.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/environment/environment_factory.h"
@@ -434,7 +434,7 @@ int NetEqQualityTest::Transmit() {
     if (!PacketLost()) {
       int ret = neteq_->InsertPacket(
           rtp_header_,
-          ArrayView<const uint8_t>(payload_.data(), payload_size_bytes_),
+          std::span<const uint8_t>(payload_.data(), payload_size_bytes_),
           Timestamp::Millis(packet_input_time_ms));
       if (ret != NetEq::kOK)
         return -1;
diff --git a/modules/audio_coding/neteq/tools/neteq_test.cc b/modules/audio_coding/neteq/tools/neteq_test.cc
index bab757fae5..adf882025a 100644
--- a/modules/audio_coding/neteq/tools/neteq_test.cc
+++ b/modules/audio_coding/neteq/tools/neteq_test.cc
@@ -19,7 +19,6 @@
 #include <optional>
 #include <utility>
 
-#include "api/array_view.h"
 #include "api/audio/audio_frame.h"
 #include "api/audio_codecs/audio_decoder_factory.h"
 #include "api/audio_codecs/audio_format.h"
diff --git a/modules/audio_coding/neteq/tools/rtp_jitter.cc b/modules/audio_coding/neteq/tools/rtp_jitter.cc
index acf27445ba..03b346c645 100644
--- a/modules/audio_coding/neteq/tools/rtp_jitter.cc
+++ b/modules/audio_coding/neteq/tools/rtp_jitter.cc
@@ -13,11 +13,11 @@
 #include <cstdio>
 #include <fstream>
 #include <iostream>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/rtp_rtcp/source/byte_io.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/checks.h"
@@ -32,7 +32,7 @@ constexpr size_t kRtpDumpHeaderLength = 8;
 Buffer ReadNextPacket(FILE* file) {
   // Read the rtpdump header for the next packet.
   Buffer buffer;
-  buffer.SetData(kRtpDumpHeaderLength, [&](ArrayView<uint8_t> x) {
+  buffer.SetData(kRtpDumpHeaderLength, [&](std::span<uint8_t> x) {
     return fread(x.data(), 1, x.size(), file);
   });
   if (buffer.size() != kRtpDumpHeaderLength) {
@@ -45,7 +45,7 @@ Buffer ReadNextPacket(FILE* file) {
   RTC_CHECK_GE(len, kRtpDumpHeaderLength);
 
   // Read remaining data from file directly into buffer.
-  buffer.AppendData(len - kRtpDumpHeaderLength, [&](ArrayView<uint8_t> x) {
+  buffer.AppendData(len - kRtpDumpHeaderLength, [&](std::span<uint8_t> x) {
     return fread(x.data(), 1, x.size(), file);
   });
   if (buffer.size() != len) {
diff --git a/modules/audio_coding/test/Channel.cc b/modules/audio_coding/test/Channel.cc
index 5d3c2192d3..e2ebb90074 100644
--- a/modules/audio_coding/test/Channel.cc
+++ b/modules/audio_coding/test/Channel.cc
@@ -14,8 +14,8 @@
 #include <cstdint>
 #include <cstdio>
 #include <cstring>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/neteq/neteq.h"
 #include "api/rtp_headers.h"
 #include "api/units/timestamp.h"
@@ -92,7 +92,7 @@ int32_t Channel::SendData(AudioFrameType frameType,
   }
 
   status = _neteq->InsertPacket(
-      rtp_header, ArrayView<const uint8_t>(_payloadData, payloadDataSize),
+      rtp_header, std::span<const uint8_t>(_payloadData, payloadDataSize),
       /*receive_time=*/Timestamp::MinusInfinity());
 
   return status;
diff --git a/modules/audio_coding/test/EncodeDecodeTest.cc b/modules/audio_coding/test/EncodeDecodeTest.cc
index 848e833f42..e61d584eec 100644
--- a/modules/audio_coding/test/EncodeDecodeTest.cc
+++ b/modules/audio_coding/test/EncodeDecodeTest.cc
@@ -15,10 +15,10 @@
 #include <cstdlib>
 #include <map>
 #include <memory>
+#include <span>
 #include <string>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "api/audio_codecs/builtin_audio_encoder_factory.h"
@@ -182,7 +182,7 @@ bool Receiver::IncomingPacket() {
 
     EXPECT_GE(
         0, _neteq->InsertPacket(_rtpHeader,
-                                ArrayView<const uint8_t>(_incomingPayload,
+                                std::span<const uint8_t>(_incomingPayload,
                                                          _realPayloadSizeBytes),
                                 /*receive_time=*/Timestamp::Millis(_nextTime)));
     _realPayloadSizeBytes = _rtpStream->Read(&_rtpHeader, _incomingPayload,
diff --git a/modules/audio_coding/test/PacketLossTest.cc b/modules/audio_coding/test/PacketLossTest.cc
index 59b5309c80..129eebec40 100644
--- a/modules/audio_coding/test/PacketLossTest.cc
+++ b/modules/audio_coding/test/PacketLossTest.cc
@@ -12,10 +12,10 @@
 
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <string>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
@@ -73,7 +73,7 @@ bool ReceiverWithPacketLoss::IncomingPacket() {
     if (!PacketLost()) {
       _neteq->InsertPacket(
           _rtpHeader,
-          ArrayView<const uint8_t>(_incomingPayload, _realPayloadSizeBytes),
+          std::span<const uint8_t>(_incomingPayload, _realPayloadSizeBytes),
           Timestamp::Millis(_nextTime));
     }
     packet_counter_++;
diff --git a/modules/audio_coding/test/TestAllCodecs.cc b/modules/audio_coding/test/TestAllCodecs.cc
index e45f1fddb9..689c25868d 100644
--- a/modules/audio_coding/test/TestAllCodecs.cc
+++ b/modules/audio_coding/test/TestAllCodecs.cc
@@ -14,11 +14,11 @@
 #include <cstdio>
 #include <cstring>
 #include <limits>
+#include <span>
 #include <string>
 
 #include "absl/strings/match.h"
 #include "absl/strings/str_cat.h"
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "api/audio_codecs/builtin_audio_encoder_factory.h"
@@ -93,7 +93,7 @@ int32_t TestPack::SendData(AudioFrameType frame_type,
   memcpy(payload_data_, payload_data, payload_size);
 
   status = neteq_->InsertPacket(
-      rtp_header, ArrayView<const uint8_t>(payload_data_, payload_size),
+      rtp_header, std::span<const uint8_t>(payload_data_, payload_size),
       /*receive_time=*/Timestamp::MinusInfinity());
 
   payload_size_ = payload_size;
diff --git a/modules/audio_coding/test/TestStereo.cc b/modules/audio_coding/test/TestStereo.cc
index 78b0d8ca74..f0913f72ec 100644
--- a/modules/audio_coding/test/TestStereo.cc
+++ b/modules/audio_coding/test/TestStereo.cc
@@ -14,12 +14,12 @@
 #include <cstddef>
 #include <cstdint>
 #include <cstring>
+#include <span>
 #include <string>
 #include <utility>
 
 #include "absl/strings/match.h"
 #include "absl/strings/str_cat.h"
-#include "api/array_view.h"
 #include "api/audio/audio_frame.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
@@ -78,7 +78,7 @@ int32_t TestPackStereo::SendData(const AudioFrameType frame_type,
 
   if (lost_packet_ == false) {
     status = neteq_->InsertPacket(
-        rtp_header, ArrayView<const uint8_t>(payload_data, payload_size),
+        rtp_header, std::span<const uint8_t>(payload_data, payload_size),
         /*receive_time=*/Timestamp::MinusInfinity());
 
     if (frame_type != AudioFrameType::kAudioFrameCN) {
diff --git a/modules/audio_coding/test/target_delay_unittest.cc b/modules/audio_coding/test/target_delay_unittest.cc
index 96130b49d6..bf72bcd8a6 100644
--- a/modules/audio_coding/test/target_delay_unittest.cc
+++ b/modules/audio_coding/test/target_delay_unittest.cc
@@ -13,8 +13,8 @@
 #include <cstdlib>
 #include <map>
 #include <memory>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/audio/audio_frame.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
@@ -95,7 +95,7 @@ class TargetDelayTest : public ::testing::Test {
     rtp_header_.sequenceNumber++;
     ASSERT_EQ(0, neteq_->InsertPacket(
                      rtp_header_,
-                     ArrayView<const uint8_t>(payload_, kFrameSizeSamples * 2),
+                     std::span<const uint8_t>(payload_, kFrameSizeSamples * 2),
                      Timestamp::MinusInfinity()));
   }
 
diff --git a/modules/audio_device/BUILD.gn b/modules/audio_device/BUILD.gn
index bad43a5c91..a5105a8257 100644
--- a/modules/audio_device/BUILD.gn
+++ b/modules/audio_device/BUILD.gn
@@ -58,7 +58,6 @@ rtc_library("audio_device_buffer") {
     "fine_audio_buffer.h",
   ]
   deps = [
-    "../../api:array_view",
     "../../api:sequence_checker",
     "../../api/audio:audio_device",
     "../../api/environment",
@@ -149,7 +148,6 @@ rtc_library("audio_device_module_from_input_and_output") {
       ":audio_device_buffer",
       ":audio_device_name",
       ":windows_core_audio_utility",
-      "../../api:array_view",
       "../../api:make_ref_counted",
       "../../api:scoped_refptr",
       "../../api:sequence_checker",
@@ -186,7 +184,6 @@ if (!build_with_chromium) {
       ":audio_device_default",
       ":audio_device_generic",
       ":audio_device_impl",
-      "../../api:array_view",
       "../../api:make_ref_counted",
       "../../api:scoped_refptr",
       "../../api/audio:audio_device",
@@ -264,7 +261,6 @@ rtc_library("audio_device_impl") {
     ":audio_device_default",
     ":audio_device_dummy",
     ":audio_device_generic",
-    "../../api:array_view",
     "../../api:make_ref_counted",
     "../../api:ref_count",
     "../../api:refcountedbase",
@@ -461,7 +457,6 @@ if (rtc_include_tests && !build_with_chromium) {
       ":audio_device_impl",
       ":mock_audio_device",
       ":test_audio_device_module",
-      "../../api:array_view",
       "../../api:scoped_refptr",
       "../../api:sequence_checker",
       "../../api/audio:audio_device",
diff --git a/modules/audio_device/audio_device_unittest.cc b/modules/audio_device/audio_device_unittest.cc
index ef93f7bc4c..9dc10ea648 100644
--- a/modules/audio_device/audio_device_unittest.cc
+++ b/modules/audio_device/audio_device_unittest.cc
@@ -18,9 +18,9 @@
 #include <list>
 #include <numeric>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/audio_device_defines.h"
 #include "api/audio/create_audio_device_module.h"
 #include "api/environment/environment.h"
@@ -113,8 +113,8 @@ enum class TransportType {
 // measurements.
 class AudioStream {
  public:
-  virtual void Write(ArrayView<const int16_t> source) = 0;
-  virtual void Read(ArrayView<int16_t> destination) = 0;
+  virtual void Write(std::span<const int16_t> source) = 0;
+  virtual void Read(std::span<int16_t> destination) = 0;
 
   virtual ~AudioStream() = default;
 };
@@ -141,7 +141,7 @@ int IndexToMilliseconds(size_t index, size_t frames_per_10ms_buffer) {
 // change over time and that both sides will in most cases use the same size.
 class FifoAudioStream : public AudioStream {
  public:
-  void Write(ArrayView<const int16_t> source) override {
+  void Write(std::span<const int16_t> source) override {
     RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
     const size_t size = [&] {
       MutexLock lock(&lock_);
@@ -158,7 +158,7 @@ class FifoAudioStream : public AudioStream {
     written_elements_ += size;
   }
 
-  void Read(ArrayView<int16_t> destination) override {
+  void Read(std::span<int16_t> destination) override {
     MutexLock lock(&lock_);
     if (fifo_.empty()) {
       std::fill(destination.begin(), destination.end(), 0);
@@ -227,7 +227,7 @@ class LatencyAudioStream : public AudioStream {
   }
 
   // Insert periodic impulses in first two samples of `destination`.
-  void Read(ArrayView<int16_t> destination) override {
+  void Read(std::span<int16_t> destination) override {
     RTC_DCHECK_RUN_ON(&read_thread_checker_);
     if (read_count_ == 0) {
       PRINT("[");
@@ -249,7 +249,7 @@ class LatencyAudioStream : public AudioStream {
 
   // Detect received impulses in `source`, derive time between transmission and
   // detection and add the calculated delay to list of latencies.
-  void Write(ArrayView<const int16_t> source) override {
+  void Write(std::span<const int16_t> source) override {
     RTC_DCHECK_RUN_ON(&write_thread_checker_);
     RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
     MutexLock lock(&lock_);
@@ -402,9 +402,8 @@ class MockAudioTransport : public test::MockAudioTransport {
     }
     // Write audio data to audio stream object if one has been injected.
     if (audio_stream_) {
-      audio_stream_->Write(
-          MakeArrayView(static_cast<const int16_t*>(audio_buffer),
-                        samples_per_channel * channels));
+      audio_stream_->Write(std::span(static_cast<const int16_t*>(audio_buffer),
+                                     samples_per_channel * channels));
     }
     // Signal the event after given amount of callbacks.
     if (event_ && ReceivedEnoughCallbacks()) {
@@ -443,8 +442,8 @@ class MockAudioTransport : public test::MockAudioTransport {
     samples_out = samples_per_channel * channels;
     // Read audio data from audio stream object if one has been injected.
     if (audio_stream_) {
-      audio_stream_->Read(MakeArrayView(static_cast<int16_t*>(audio_buffer),
-                                        samples_per_channel * channels));
+      audio_stream_->Read(std::span(static_cast<int16_t*>(audio_buffer),
+                                    samples_per_channel * channels));
     } else {
       // Fill the audio buffer with zeros to avoid disturbing audio.
       const size_t num_bytes = samples_per_channel * bytes_per_frame;
diff --git a/modules/audio_device/fine_audio_buffer.cc b/modules/audio_device/fine_audio_buffer.cc
index 0ab7023448..04cf6713a6 100644
--- a/modules/audio_device/fine_audio_buffer.cc
+++ b/modules/audio_device/fine_audio_buffer.cc
@@ -13,8 +13,8 @@
 #include <cstdint>
 #include <cstring>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_device/audio_device_buffer.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
@@ -64,7 +64,7 @@ bool FineAudioBuffer::IsReadyForRecord() const {
   return record_samples_per_channel_10ms_ > 0 && record_channels_ > 0;
 }
 
-void FineAudioBuffer::GetPlayoutData(ArrayView<int16_t> audio_buffer,
+void FineAudioBuffer::GetPlayoutData(std::span<int16_t> audio_buffer,
                                      int playout_delay_ms) {
   RTC_DCHECK(IsReadyForPlayout());
   // Ask WebRTC for new data in chunks of 10ms until we have enough to
@@ -81,7 +81,7 @@ void FineAudioBuffer::GetPlayoutData(ArrayView<int16_t> audio_buffer,
       const size_t num_elements_10ms =
           playout_channels_ * playout_samples_per_channel_10ms_;
       const size_t written_elements = playout_buffer_.AppendData(
-          num_elements_10ms, [&](ArrayView<int16_t> buf) {
+          num_elements_10ms, [&](std::span<int16_t> buf) {
             const size_t samples_per_channel_10ms =
                 audio_device_buffer_->GetPlayoutData(buf.data());
             return playout_channels_ * samples_per_channel_10ms;
@@ -108,7 +108,7 @@ void FineAudioBuffer::GetPlayoutData(ArrayView<int16_t> audio_buffer,
 }
 
 void FineAudioBuffer::DeliverRecordedData(
-    ArrayView<const int16_t> audio_buffer,
+    std::span<const int16_t> audio_buffer,
     int record_delay_ms,
     std::optional<int64_t> capture_time_ns) {
   RTC_DCHECK(IsReadyForRecord());
diff --git a/modules/audio_device/fine_audio_buffer.h b/modules/audio_device/fine_audio_buffer.h
index dd4d456a68..a2cb3c2a32 100644
--- a/modules/audio_device/fine_audio_buffer.h
+++ b/modules/audio_device/fine_audio_buffer.h
@@ -14,8 +14,8 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "rtc_base/buffer.h"
 
 namespace webrtc {
@@ -52,7 +52,7 @@ class FineAudioBuffer {
   // silence instead. The provided delay estimate in `playout_delay_ms` should
   // contain an estimate of the latency between when an audio frame is read from
   // WebRTC and when it is played out on the speaker.
-  void GetPlayoutData(ArrayView<int16_t> audio_buffer, int playout_delay_ms);
+  void GetPlayoutData(std::span<int16_t> audio_buffer, int playout_delay_ms);
 
   // Consumes the audio data in `audio_buffer` and sends it to the WebRTC layer
   // in chunks of 10ms. The sum of the provided delay estimate in
@@ -63,11 +63,11 @@ class FineAudioBuffer {
   // Example: buffer size is 5ms => call #1 stores 5ms of data, call #2 stores
   // 5ms of data and sends a total of 10ms to WebRTC and clears the internal
   // cache. Call #3 restarts the scheme above.
-  void DeliverRecordedData(ArrayView<const int16_t> audio_buffer,
+  void DeliverRecordedData(std::span<const int16_t> audio_buffer,
                            int record_delay_ms) {
     DeliverRecordedData(audio_buffer, record_delay_ms, std::nullopt);
   }
-  void DeliverRecordedData(ArrayView<const int16_t> audio_buffer,
+  void DeliverRecordedData(std::span<const int16_t> audio_buffer,
                            int record_delay_ms,
                            std::optional<int64_t> capture_time_ns);
 
diff --git a/modules/audio_device/fine_audio_buffer_unittest.cc b/modules/audio_device/fine_audio_buffer_unittest.cc
index 2ec0a53796..2516b5f0c3 100644
--- a/modules/audio_device/fine_audio_buffer_unittest.cc
+++ b/modules/audio_device/fine_audio_buffer_unittest.cc
@@ -13,8 +13,8 @@
 #include <climits>
 #include <cstdint>
 #include <memory>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/audio_device/mock_audio_device_buffer.h"
 #include "test/create_test_environment.h"
 #include "test/gmock.h"
@@ -131,12 +131,12 @@ void RunFineBufferTest(int frame_size_in_samples) {
 
   for (int i = 0; i < kNumberOfFrames; ++i) {
     fine_buffer.GetPlayoutData(
-        ArrayView<int16_t>(out_buffer.get(), kChannels * kFrameSizeSamples), 0);
+        std::span<int16_t>(out_buffer.get(), kChannels * kFrameSizeSamples), 0);
     EXPECT_TRUE(
         VerifyBuffer(out_buffer.get(), i, kChannels * kFrameSizeSamples));
     UpdateInputBuffer(in_buffer.get(), i, kChannels * kFrameSizeSamples);
     fine_buffer.DeliverRecordedData(
-        ArrayView<const int16_t>(in_buffer.get(),
+        std::span<const int16_t>(in_buffer.get(),
                                  kChannels * kFrameSizeSamples),
         0);
   }
diff --git a/modules/audio_device/include/test_audio_device.cc b/modules/audio_device/include/test_audio_device.cc
index 132128e00a..01360b294d 100644
--- a/modules/audio_device/include/test_audio_device.cc
+++ b/modules/audio_device/include/test_audio_device.cc
@@ -14,12 +14,12 @@
 #include <cstdlib>
 #include <cstring>
 #include <memory>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio/audio_device.h"
 #include "api/environment/environment.h"
 #include "api/make_ref_counted.h"
@@ -74,7 +74,7 @@ class PulsedNoiseCapturerImpl final
     buffer->SetData(
         TestAudioDeviceModule::SamplesPerFrame(sampling_frequency_in_hz_) *
             num_channels_,
-        [&](ArrayView<int16_t> data) {
+        [&](std::span<int16_t> data) {
           if (fill_with_zero_) {
             std::fill(data.begin(), data.end(), 0);
           } else {
@@ -120,7 +120,7 @@ class WavFileReader final : public TestAudioDeviceModule::Capturer {
     buffer->SetData(
         TestAudioDeviceModule::SamplesPerFrame(sampling_frequency_in_hz_) *
             num_channels_,
-        [&](ArrayView<int16_t> data) {
+        [&](std::span<int16_t> data) {
           size_t read = wav_reader_->ReadSamples(data.size(), data.data());
           if (read < data.size() && repeat_) {
             do {
@@ -170,7 +170,7 @@ class WavFileWriter final : public TestAudioDeviceModule::Renderer {
 
   int NumChannels() const override { return num_channels_; }
 
-  bool Render(ArrayView<const int16_t> data) override {
+  bool Render(std::span<const int16_t> data) override {
     wav_writer_->WriteSamples(data.data(), data.size());
     return true;
   }
@@ -207,7 +207,7 @@ class BoundedWavFileWriter : public TestAudioDeviceModule::Renderer {
 
   int NumChannels() const override { return num_channels_; }
 
-  bool Render(ArrayView<const int16_t> data) override {
+  bool Render(std::span<const int16_t> data) override {
     const int16_t kAmplitudeThreshold = 5;
 
     const int16_t* begin = data.data();
@@ -266,7 +266,7 @@ class DiscardRenderer final : public TestAudioDeviceModule::Renderer {
 
   int NumChannels() const override { return num_channels_; }
 
-  bool Render(ArrayView<const int16_t> /* data */) override { return true; }
+  bool Render(std::span<const int16_t> /* data */) override { return true; }
 
  private:
   int sampling_frequency_in_hz_;
@@ -302,8 +302,8 @@ class RawFileReader final : public TestAudioDeviceModule::Capturer {
     buffer->SetData(
         TestAudioDeviceModule::SamplesPerFrame(SamplingFrequency()) *
             NumChannels(),
-        [&](ArrayView<int16_t> data) {
-          ArrayView<int8_t> read_buffer_view = ReadBufferView();
+        [&](std::span<int16_t> data) {
+          std::span<int8_t> read_buffer_view = ReadBufferView();
           size_t size = data.size() * 2;
           size_t read = input_file_.Read(read_buffer_view.data(), size);
           if (read < size && repeat_) {
@@ -322,7 +322,7 @@ class RawFileReader final : public TestAudioDeviceModule::Capturer {
   }
 
  private:
-  ArrayView<int8_t> ReadBufferView() { return read_buffer_; }
+  std::span<int8_t> ReadBufferView() { return read_buffer_; }
 
   const std::string input_file_name_;
   const int sampling_frequency_in_hz_;
@@ -360,7 +360,7 @@ class RawFileWriter : public TestAudioDeviceModule::Renderer {
 
   int NumChannels() const override { return num_channels_; }
 
-  bool Render(ArrayView<const int16_t> data) override {
+  bool Render(std::span<const int16_t> data) override {
     const int16_t kAmplitudeThreshold = 5;
 
     const int16_t* begin = data.data();
diff --git a/modules/audio_device/include/test_audio_device.h b/modules/audio_device/include/test_audio_device.h
index 79694846a6..e460d25d15 100644
--- a/modules/audio_device/include/test_audio_device.h
+++ b/modules/audio_device/include/test_audio_device.h
@@ -13,9 +13,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <memory>
+#include <span>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio/audio_device.h"
 #include "api/environment/environment.h"
 #include "api/scoped_refptr.h"
@@ -59,7 +59,7 @@ class TestAudioDeviceModule {
     virtual int NumChannels() const = 0;
     // Renders the passed audio data and returns true if the renderer wants
     // to keep receiving data, or false otherwise.
-    virtual bool Render(ArrayView<const int16_t> data) = 0;
+    virtual bool Render(std::span<const int16_t> data) = 0;
   };
 
   // A fake capturer that generates pulses with random samples between
diff --git a/modules/audio_device/include/test_audio_device_unittest.cc b/modules/audio_device/include/test_audio_device_unittest.cc
index 95eb74eb62..1ad4b9dbd8 100644
--- a/modules/audio_device/include/test_audio_device_unittest.cc
+++ b/modules/audio_device/include/test_audio_device_unittest.cc
@@ -20,11 +20,11 @@
 #include <cstring>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/audio_device.h"
 #include "api/audio/audio_device_defines.h"
 #include "api/environment/environment.h"
@@ -65,7 +65,7 @@ void RunWavTest(const std::vector<int16_t>& input_samples,
         TestAudioDeviceModule::CreateBoundedWavFileWriter(output_filename, 800);
 
     for (size_t i = 0; i < input_samples.size(); i += kSamplesPerFrame) {
-      EXPECT_TRUE(writer->Render(ArrayView<const int16_t>(
+      EXPECT_TRUE(writer->Render(std::span<const int16_t>(
           &input_samples[i],
           std::min(kSamplesPerFrame, input_samples.size() - i))));
     }
@@ -162,7 +162,7 @@ TEST(WavFileReaderTest, RepeatedTrueWithSingleFrameFileReadTwice) {
         TestAudioDeviceModule::CreateWavFileWriter(output_filename, 800);
 
     for (size_t i = 0; i < kInputSamples.size(); i += kSamplesPerFrame) {
-      EXPECT_TRUE(writer->Render(ArrayView<const int16_t>(
+      EXPECT_TRUE(writer->Render(std::span<const int16_t>(
           &kInputSamples[i],
           std::min(kSamplesPerFrame, kInputSamples.size() - i))));
     }
@@ -203,7 +203,7 @@ void RunRawTestNoRepeat(const std::vector<int16_t>& input_samples,
             output_filename, /*sampling_frequency_in_hz=*/800);
 
     for (size_t i = 0; i < input_samples.size(); i += kSamplesPerFrame) {
-      EXPECT_TRUE(writer->Render(ArrayView<const int16_t>(
+      EXPECT_TRUE(writer->Render(std::span<const int16_t>(
           &input_samples[i],
           std::min(kSamplesPerFrame, input_samples.size() - i))));
     }
@@ -311,7 +311,7 @@ TEST(RawFileWriterTest, Repeat) {
             output_filename, /*sampling_frequency_in_hz=*/800);
 
     for (size_t i = 0; i < kInputSamples.size(); i += kSamplesPerFrame) {
-      EXPECT_TRUE(writer->Render(ArrayView<const int16_t>(
+      EXPECT_TRUE(writer->Render(std::span<const int16_t>(
           &kInputSamples[i],
           std::min(kSamplesPerFrame, kInputSamples.size() - i))));
     }
diff --git a/modules/audio_device/test_audio_device_impl.cc b/modules/audio_device/test_audio_device_impl.cc
index 5cfb65bfdc..33f59a8539 100644
--- a/modules/audio_device/test_audio_device_impl.cc
+++ b/modules/audio_device/test_audio_device_impl.cc
@@ -13,9 +13,9 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/task_queue/task_queue_factory.h"
 #include "api/units/time_delta.h"
@@ -189,7 +189,7 @@ void TestAudioDevice::ProcessAudio() {
     size_t samples_out = samples_per_channel * renderer_->NumChannels();
     RTC_CHECK_LE(samples_out, playout_buffer_.size());
     const bool keep_rendering = renderer_->Render(
-        ArrayView<const int16_t>(playout_buffer_.data(), samples_out));
+        std::span<const int16_t>(playout_buffer_.data(), samples_out));
     if (!keep_rendering) {
       rendering_ = false;
     }
diff --git a/modules/audio_device/win/core_audio_input_win.cc b/modules/audio_device/win/core_audio_input_win.cc
index f6139aefb3..3b92e586f4 100644
--- a/modules/audio_device/win/core_audio_input_win.cc
+++ b/modules/audio_device/win/core_audio_input_win.cc
@@ -13,9 +13,9 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 
-#include "api/array_view.h"
 #include "api/audio/audio_device.h"
 #include "api/environment/environment.h"
 #include "api/sequence_checker.h"
@@ -368,8 +368,8 @@ bool CoreAudioInput::OnDataCallback(uint64_t device_frequency) {
       // Copy recorded audio in `audio_data` to the WebRTC sink using the
       // FineAudioBuffer object.
       fine_audio_buffer_->DeliverRecordedData(
-          webrtc::MakeArrayView(reinterpret_cast<const int16_t*>(audio_data),
-                                format_.Format.nChannels * num_frames_to_read),
+          std::span(reinterpret_cast<const int16_t*>(audio_data),
+                    format_.Format.nChannels * num_frames_to_read),
 
           latency_ms_);
     }
diff --git a/modules/audio_device/win/core_audio_output_win.cc b/modules/audio_device/win/core_audio_output_win.cc
index decc0ee653..f373c40bd1 100644
--- a/modules/audio_device/win/core_audio_output_win.cc
+++ b/modules/audio_device/win/core_audio_output_win.cc
@@ -12,9 +12,9 @@
 
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <string>
 
-#include "api/array_view.h"
 #include "api/audio/audio_device.h"
 #include "api/environment/environment.h"
 #include "api/sequence_checker.h"
@@ -346,8 +346,8 @@ bool CoreAudioOutput::OnDataCallback(uint64_t device_frequency) {
   // Get audio data from WebRTC and write it to the allocated buffer in
   // `audio_data`. The playout latency is not updated for each callback.
   fine_audio_buffer_->GetPlayoutData(
-      webrtc::MakeArrayView(reinterpret_cast<int16_t*>(audio_data),
-                            num_requested_frames * format_.Format.nChannels),
+      std::span(reinterpret_cast<int16_t*>(audio_data),
+                num_requested_frames * format_.Format.nChannels),
       latency_ms_);
 
   // Release the buffer space acquired in IAudioRenderClient::GetBuffer.
diff --git a/modules/audio_mixer/BUILD.gn b/modules/audio_mixer/BUILD.gn
index d2fa8dbaec..362abf5dc9 100644
--- a/modules/audio_mixer/BUILD.gn
+++ b/modules/audio_mixer/BUILD.gn
@@ -39,14 +39,12 @@ rtc_library("audio_mixer_impl") {
 
   deps = [
     ":audio_frame_manipulator",
-    "../../api:array_view",
     "../../api:make_ref_counted",
     "../../api:rtp_packet_info",
     "../../api:scoped_refptr",
     "../../api/audio:audio_frame_api",
     "../../api/audio:audio_mixer_api",
     "../../api/audio:audio_processing",
-    "../../api/audio:audio_processing",
     "../../audio/utility:audio_frame_operations",
     "../../common_audio",
     "../../rtc_base:checks",
@@ -97,7 +95,6 @@ if (rtc_include_tests) {
     deps = [
       ":audio_frame_manipulator",
       ":audio_mixer_impl",
-      "../../api:array_view",
       "../../api/audio:audio_frame_api",
       "../../rtc_base:checks",
       "../../rtc_base:safe_conversions",
@@ -116,7 +113,6 @@ if (rtc_include_tests) {
       ":audio_frame_manipulator",
       ":audio_mixer_impl",
       ":audio_mixer_test_utils",
-      "../../api:array_view",
       "../../api:rtp_packet_info",
       "../../api:scoped_refptr",
       "../../api/audio:audio_frame_api",
diff --git a/modules/audio_mixer/audio_mixer_impl.cc b/modules/audio_mixer/audio_mixer_impl.cc
index edf68284e4..e961279093 100644
--- a/modules/audio_mixer/audio_mixer_impl.cc
+++ b/modules/audio_mixer/audio_mixer_impl.cc
@@ -13,10 +13,10 @@
 #include <algorithm>
 #include <cstddef>
 #include <memory>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/audio_frame.h"
 #include "api/make_ref_counted.h"
 #include "api/scoped_refptr.h"
@@ -101,7 +101,7 @@ void AudioMixerImpl::Mix(size_t number_of_channels,
                  });
 
   int output_frequency = output_rate_calculator_->CalculateOutputRateFromRange(
-      ArrayView<const int>(helper_containers_->preferred_rates.data(),
+      std::span<const int>(helper_containers_->preferred_rates.data(),
                            number_of_streams));
 
   frame_combiner_.Combine(GetAudioFromSources(output_frequency),
@@ -129,7 +129,7 @@ void AudioMixerImpl::RemoveSource(Source* audio_source) {
   audio_source_list_.erase(iter);
 }
 
-ArrayView<AudioFrame* const> AudioMixerImpl::GetAudioFromSources(
+std::span<AudioFrame* const> AudioMixerImpl::GetAudioFromSources(
     int output_frequency) {
   int audio_to_mix_count = 0;
   for (auto& source_and_status : audio_source_list_) {
@@ -148,7 +148,7 @@ ArrayView<AudioFrame* const> AudioMixerImpl::GetAudioFromSources(
             &source_and_status->audio_frame;
     }
   }
-  return ArrayView<AudioFrame* const>(helper_containers_->audio_to_mix.data(),
+  return std::span<AudioFrame* const>(helper_containers_->audio_to_mix.data(),
                                       audio_to_mix_count);
 }
 
diff --git a/modules/audio_mixer/audio_mixer_impl.h b/modules/audio_mixer/audio_mixer_impl.h
index 104f1304b7..c70d6c9ca8 100644
--- a/modules/audio_mixer/audio_mixer_impl.h
+++ b/modules/audio_mixer/audio_mixer_impl.h
@@ -14,9 +14,9 @@
 #include <stddef.h>
 
 #include <memory>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/audio_frame.h"
 #include "api/audio/audio_mixer.h"
 #include "api/scoped_refptr.h"
@@ -63,7 +63,7 @@ class AudioMixerImpl : public AudioMixer {
   void UpdateSourceCountStats() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
 
   // Fetches audio frames to mix from sources.
-  ArrayView<AudioFrame* const> GetAudioFromSources(int output_frequency)
+  std::span<AudioFrame* const> GetAudioFromSources(int output_frequency)
       RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
 
   // The critical section lock guards audio source insertion and
diff --git a/modules/audio_mixer/audio_mixer_impl_unittest.cc b/modules/audio_mixer/audio_mixer_impl_unittest.cc
index 9cd6b424be..1e240cf09f 100644
--- a/modules/audio_mixer/audio_mixer_impl_unittest.cc
+++ b/modules/audio_mixer/audio_mixer_impl_unittest.cc
@@ -15,10 +15,10 @@
 #include <cstring>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/audio_frame.h"
 #include "api/audio/audio_mixer.h"
 #include "api/rtp_packet_info.h"
@@ -122,7 +122,7 @@ class CustomRateCalculator : public OutputRateCalculator {
  public:
   explicit CustomRateCalculator(int rate) : rate_(rate) {}
   int CalculateOutputRateFromRange(
-      ArrayView<const int> /* preferred_rates */) override {
+      std::span<const int> /* preferred_rates */) override {
     return rate_;
   }
 
@@ -484,7 +484,7 @@ class HighOutputRateCalculator : public OutputRateCalculator {
  public:
   static const int kDefaultFrequency = 76000;
   int CalculateOutputRateFromRange(
-      ArrayView<const int> /* preferred_sample_rates */) override {
+      std::span<const int> /* preferred_sample_rates */) override {
     return kDefaultFrequency;
   }
   ~HighOutputRateCalculator() override {}
diff --git a/modules/audio_mixer/default_output_rate_calculator.cc b/modules/audio_mixer/default_output_rate_calculator.cc
index be9819c027..8eb6a05f72 100644
--- a/modules/audio_mixer/default_output_rate_calculator.cc
+++ b/modules/audio_mixer/default_output_rate_calculator.cc
@@ -12,15 +12,15 @@
 
 #include <algorithm>
 #include <iterator>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/audio/audio_processing.h"
 #include "rtc_base/checks.h"
 
 namespace webrtc {
 
 int DefaultOutputRateCalculator::CalculateOutputRateFromRange(
-    ArrayView<const int> preferred_sample_rates) {
+    std::span<const int> preferred_sample_rates) {
   if (preferred_sample_rates.empty()) {
     return DefaultOutputRateCalculator::kDefaultFrequency;
   }
diff --git a/modules/audio_mixer/default_output_rate_calculator.h b/modules/audio_mixer/default_output_rate_calculator.h
index acae77fe0a..e91a2d884d 100644
--- a/modules/audio_mixer/default_output_rate_calculator.h
+++ b/modules/audio_mixer/default_output_rate_calculator.h
@@ -11,7 +11,8 @@
 #ifndef MODULES_AUDIO_MIXER_DEFAULT_OUTPUT_RATE_CALCULATOR_H_
 #define MODULES_AUDIO_MIXER_DEFAULT_OUTPUT_RATE_CALCULATOR_H_
 
-#include "api/array_view.h"
+#include <span>
+
 #include "modules/audio_mixer/output_rate_calculator.h"
 
 namespace webrtc {
@@ -25,7 +26,7 @@ class DefaultOutputRateCalculator : public OutputRateCalculator {
   // AudioProcessing::NativeRate. If `preferred_sample_rates` is
   // empty, returns `kDefaultFrequency`.
   int CalculateOutputRateFromRange(
-      ArrayView<const int> preferred_sample_rates) override;
+      std::span<const int> preferred_sample_rates) override;
   ~DefaultOutputRateCalculator() override {}
 };
 
diff --git a/modules/audio_mixer/frame_combiner.cc b/modules/audio_mixer/frame_combiner.cc
index 6f5e1b785a..7fd5caf377 100644
--- a/modules/audio_mixer/frame_combiner.cc
+++ b/modules/audio_mixer/frame_combiner.cc
@@ -15,10 +15,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/audio_frame.h"
 #include "api/audio/audio_view.h"
 #include "api/rtp_packet_info.h"
@@ -32,7 +32,7 @@
 namespace webrtc {
 namespace {
 
-void SetAudioFrameFields(ArrayView<const AudioFrame* const> mix_list,
+void SetAudioFrameFields(std::span<const AudioFrame* const> mix_list,
                          size_t number_of_channels,
                          int sample_rate,
                          size_t /* number_of_streams */,
@@ -70,7 +70,7 @@ void SetAudioFrameFields(ArrayView<const AudioFrame* const> mix_list,
   }
 }
 
-void MixFewFramesWithNoLimiter(ArrayView<const AudioFrame* const> mix_list,
+void MixFewFramesWithNoLimiter(std::span<const AudioFrame* const> mix_list,
                                AudioFrame* audio_frame_for_mixing) {
   if (mix_list.empty()) {
     audio_frame_for_mixing->Mute();
@@ -82,7 +82,7 @@ void MixFewFramesWithNoLimiter(ArrayView<const AudioFrame* const> mix_list,
   CopySamples(dst, mix_list[0]->data_view());
 }
 
-void MixToFloatFrame(ArrayView<const AudioFrame* const> mix_list,
+void MixToFloatFrame(std::span<const AudioFrame* const> mix_list,
                      DeinterleavedView<float>& mixing_buffer) {
   const size_t number_of_channels = NumChannels(mixing_buffer);
   // Clear the mixing buffer.
@@ -133,7 +133,7 @@ FrameCombiner::FrameCombiner(bool use_limiter)
 
 FrameCombiner::~FrameCombiner() = default;
 
-void FrameCombiner::Combine(ArrayView<AudioFrame* const> mix_list,
+void FrameCombiner::Combine(std::span<AudioFrame* const> mix_list,
                             size_t number_of_channels,
                             int sample_rate,
                             size_t number_of_streams,
diff --git a/modules/audio_mixer/frame_combiner.h b/modules/audio_mixer/frame_combiner.h
index 74c4547cc8..c6ee55c8ad 100644
--- a/modules/audio_mixer/frame_combiner.h
+++ b/modules/audio_mixer/frame_combiner.h
@@ -14,8 +14,8 @@
 #include <array>
 #include <cstddef>
 #include <memory>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/audio/audio_frame.h"
 #include "modules/audio_processing/agc2/limiter.h"
 
@@ -33,7 +33,7 @@ class FrameCombiner {
   // because 'mix_list' can be empty. The parameter
   // 'number_of_streams' is used for determining whether to pass the
   // data through a limiter.
-  void Combine(ArrayView<AudioFrame* const> mix_list,
+  void Combine(std::span<AudioFrame* const> mix_list,
                size_t number_of_channels,
                int sample_rate,
                size_t number_of_streams,
diff --git a/modules/audio_mixer/frame_combiner_unittest.cc b/modules/audio_mixer/frame_combiner_unittest.cc
index ae440462b4..e2afc47443 100644
--- a/modules/audio_mixer/frame_combiner_unittest.cc
+++ b/modules/audio_mixer/frame_combiner_unittest.cc
@@ -14,10 +14,10 @@
 #include <cstdint>
 #include <initializer_list>
 #include <numeric>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/audio_frame.h"
 #include "api/audio/channel_layout.h"
 #include "api/rtp_packet_info.h"
@@ -340,8 +340,8 @@ TEST(FrameCombiner, GainCurveIsSmoothForAlternatingNumberOfStreams) {
                        config.sample_rate_hz, number_of_streams,
                        &audio_frame_for_mixing);
       cumulative_change += change_calculator.CalculateGainChange(
-          ArrayView<const int16_t>(frame1.data(), number_of_samples),
-          ArrayView<const int16_t>(audio_frame_for_mixing.data(),
+          std::span<const int16_t>(frame1.data(), number_of_samples),
+          std::span<const int16_t>(audio_frame_for_mixing.data(),
                                    number_of_samples));
     }
 
diff --git a/modules/audio_mixer/gain_change_calculator.cc b/modules/audio_mixer/gain_change_calculator.cc
index ff14baaa52..6420468f40 100644
--- a/modules/audio_mixer/gain_change_calculator.cc
+++ b/modules/audio_mixer/gain_change_calculator.cc
@@ -13,9 +13,9 @@
 #include <cmath>
 #include <cstdint>
 #include <cstdlib>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "rtc_base/checks.h"
 
 namespace webrtc {
@@ -24,8 +24,8 @@ namespace {
 constexpr int16_t kReliabilityThreshold = 100;
 }  // namespace
 
-float GainChangeCalculator::CalculateGainChange(ArrayView<const int16_t> in,
-                                                ArrayView<const int16_t> out) {
+float GainChangeCalculator::CalculateGainChange(std::span<const int16_t> in,
+                                                std::span<const int16_t> out) {
   RTC_DCHECK_EQ(in.size(), out.size());
 
   std::vector<float> gain(in.size());
@@ -37,9 +37,9 @@ float GainChangeCalculator::LatestGain() const {
   return last_reliable_gain_;
 }
 
-void GainChangeCalculator::CalculateGain(ArrayView<const int16_t> in,
-                                         ArrayView<const int16_t> out,
-                                         ArrayView<float> gain) {
+void GainChangeCalculator::CalculateGain(std::span<const int16_t> in,
+                                         std::span<const int16_t> out,
+                                         std::span<float> gain) {
   RTC_DCHECK_EQ(in.size(), out.size());
   RTC_DCHECK_EQ(in.size(), gain.size());
 
@@ -52,7 +52,7 @@ void GainChangeCalculator::CalculateGain(ArrayView<const int16_t> in,
 }
 
 float GainChangeCalculator::CalculateDifferences(
-    ArrayView<const float> values) {
+    std::span<const float> values) {
   float res = 0;
   for (float f : values) {
     res += fabs(f - last_value_);
diff --git a/modules/audio_mixer/gain_change_calculator.h b/modules/audio_mixer/gain_change_calculator.h
index b17db3b197..2b5013a78c 100644
--- a/modules/audio_mixer/gain_change_calculator.h
+++ b/modules/audio_mixer/gain_change_calculator.h
@@ -13,7 +13,7 @@
 
 #include <stdint.h>
 
-#include "api/array_view.h"
+#include <span>
 
 namespace webrtc {
 
@@ -22,17 +22,17 @@ class GainChangeCalculator {
   // The 'out' signal is assumed to be produced from 'in' by applying
   // a smoothly varying gain. This method computes variations of the
   // gain and handles special cases when the samples are small.
-  float CalculateGainChange(ArrayView<const int16_t> in,
-                            ArrayView<const int16_t> out);
+  float CalculateGainChange(std::span<const int16_t> in,
+                            std::span<const int16_t> out);
 
   float LatestGain() const;
 
  private:
-  void CalculateGain(ArrayView<const int16_t> in,
-                     ArrayView<const int16_t> out,
-                     ArrayView<float> gain);
+  void CalculateGain(std::span<const int16_t> in,
+                     std::span<const int16_t> out,
+                     std::span<float> gain);
 
-  float CalculateDifferences(ArrayView<const float> values);
+  float CalculateDifferences(std::span<const float> values);
   float last_value_ = 0.f;
   float last_reliable_gain_ = 1.0f;
 };
diff --git a/modules/audio_mixer/output_rate_calculator.h b/modules/audio_mixer/output_rate_calculator.h
index 755da3285f..e9938a08f3 100644
--- a/modules/audio_mixer/output_rate_calculator.h
+++ b/modules/audio_mixer/output_rate_calculator.h
@@ -11,7 +11,7 @@
 #ifndef MODULES_AUDIO_MIXER_OUTPUT_RATE_CALCULATOR_H_
 #define MODULES_AUDIO_MIXER_OUTPUT_RATE_CALCULATOR_H_
 
-#include "api/array_view.h"
+#include <span>
 
 namespace webrtc {
 
@@ -20,7 +20,7 @@ namespace webrtc {
 class OutputRateCalculator {
  public:
   virtual int CalculateOutputRateFromRange(
-      ArrayView<const int> preferred_sample_rates) = 0;
+      std::span<const int> preferred_sample_rates) = 0;
 
   virtual ~OutputRateCalculator() {}
 };
diff --git a/modules/audio_processing/agc2/rnn_vad/BUILD.gn b/modules/audio_processing/agc2/rnn_vad/BUILD.gn
index 53c9af257b..2e9c473d5f 100644
--- a/modules/audio_processing/agc2/rnn_vad/BUILD.gn
+++ b/modules/audio_processing/agc2/rnn_vad/BUILD.gn
@@ -302,7 +302,6 @@ if (rtc_include_tests) {
         ":rnn_vad",
         ":rnn_vad_common",
         "..:cpu_features",
-        "../../../../api:array_view",
         "../../../../common_audio",
         "../../../../rtc_base:checks",
         "../../../../rtc_base:logging",
diff --git a/modules/congestion_controller/goog_cc/BUILD.gn b/modules/congestion_controller/goog_cc/BUILD.gn
index 837b1085f4..a8c9da49db 100644
--- a/modules/congestion_controller/goog_cc/BUILD.gn
+++ b/modules/congestion_controller/goog_cc/BUILD.gn
@@ -125,7 +125,6 @@ rtc_library("loss_based_bwe_v2") {
     "loss_based_bwe_v2.h",
   ]
   deps = [
-    "../../../api:array_view",
     "../../../api:field_trials_view",
     "../../../api/transport:network_control",
     "../../../api/units:data_rate",
diff --git a/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc b/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc
index 55c78735bc..9649332bbc 100644
--- a/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc
+++ b/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc
@@ -16,10 +16,10 @@
 #include <cstdlib>
 #include <limits>
 #include <optional>
+#include <span>
 #include <vector>
 
 #include "absl/algorithm/container.h"
-#include "api/array_view.h"
 #include "api/field_trials_view.h"
 #include "api/transport/network_types.h"
 #include "api/units/data_rate.h"
@@ -187,7 +187,7 @@ void LossBasedBweV2::SetMinMaxBitrate(DataRate min_bitrate,
 }
 
 void LossBasedBweV2::UpdateBandwidthEstimate(
-    ArrayView<const PacketResult> packet_results,
+    std::span<const PacketResult> packet_results,
     DataRate delay_based_estimate,
     bool in_alr) {
   delay_based_estimate_ = delay_based_estimate;
@@ -1147,7 +1147,7 @@ void LossBasedBweV2::NewtonsMethodUpdate(
 }
 
 bool LossBasedBweV2::PushBackObservation(
-    ArrayView<const PacketResult> packet_results) {
+    std::span<const PacketResult> packet_results) {
   if (packet_results.empty()) {
     return false;
   }
diff --git a/modules/congestion_controller/goog_cc/loss_based_bwe_v2.h b/modules/congestion_controller/goog_cc/loss_based_bwe_v2.h
index b088420f19..ce3219e0ee 100644
--- a/modules/congestion_controller/goog_cc/loss_based_bwe_v2.h
+++ b/modules/congestion_controller/goog_cc/loss_based_bwe_v2.h
@@ -13,10 +13,10 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <unordered_map>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/field_trials_view.h"
 #include "api/transport/network_types.h"
 #include "api/units/data_rate.h"
@@ -72,7 +72,7 @@ class LossBasedBweV2 {
 
   void SetAcknowledgedBitrate(DataRate acknowledged_bitrate);
   void SetMinMaxBitrate(DataRate min_bitrate, DataRate max_bitrate);
-  void UpdateBandwidthEstimate(ArrayView<const PacketResult> packet_results,
+  void UpdateBandwidthEstimate(std::span<const PacketResult> packet_results,
                                DataRate delay_based_estimate,
                                bool in_alr);
 
@@ -193,7 +193,7 @@ class LossBasedBweV2 {
   void NewtonsMethodUpdate(ChannelParameters& channel_parameters) const;
 
   // Returns false if no observation was created.
-  bool PushBackObservation(ArrayView<const PacketResult> packet_results);
+  bool PushBackObservation(std::span<const PacketResult> packet_results);
   bool IsEstimateIncreasingWhenLossLimited(DataRate old_estimate,
                                            DataRate new_estimate);
   bool IsInLossLimitedState() const;
diff --git a/modules/congestion_controller/rtp/BUILD.gn b/modules/congestion_controller/rtp/BUILD.gn
index 27c108a4f0..23e6a060a1 100644
--- a/modules/congestion_controller/rtp/BUILD.gn
+++ b/modules/congestion_controller/rtp/BUILD.gn
@@ -69,7 +69,6 @@ if (rtc_include_tests) {
     ]
     deps = [
       ":transport_feedback",
-      "../../../api:array_view",
       "../../../api/transport:ecn_marking",
       "../../../api/transport:network_control",
       "../../../api/units:data_size",
diff --git a/modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc b/modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc
index bdb63b7aad..cf36947a1c 100644
--- a/modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc
+++ b/modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc
@@ -15,10 +15,10 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/transport/ecn_marking.h"
 #include "api/transport/network_types.h"
 #include "api/units/data_size.h"
@@ -145,7 +145,7 @@ RtpPacketToSend CreatePacketToSend(PacketTemplate packet) {
 }
 
 rtcp::TransportFeedback BuildRtcpTransportFeedbackPacket(
-    ArrayView<const PacketTemplate> packets) {
+    std::span<const PacketTemplate> packets) {
   rtcp::TransportFeedback feedback;
   feedback.SetBase(packets[0].transport_sequence_number,
                    packets[0].receive_timestamp);
@@ -160,7 +160,7 @@ rtcp::TransportFeedback BuildRtcpTransportFeedbackPacket(
 }
 
 rtcp::CongestionControlFeedback BuildRtcpCongestionControlFeedbackPacket(
-    ArrayView<const PacketTemplate> packets) {
+    std::span<const PacketTemplate> packets) {
   // Assume the feedback was sent when the last packet was received.
   Timestamp feedback_sent_time = Timestamp::MinusInfinity();
   for (auto it = packets.rbegin(); it != packets.rend(); ++it) {
@@ -215,7 +215,7 @@ class TransportFeedbackAdapterTest : public ::testing::TestWithParam<bool> {
   bool UseRfc8888CongestionControlFeedback() const { return GetParam(); }
 
   std::optional<TransportPacketsFeedback> CreateAndProcessFeedback(
-      ArrayView<const PacketTemplate> packets,
+      std::span<const PacketTemplate> packets,
       TransportFeedbackAdapter& adapter) {
     if (UseRfc8888CongestionControlFeedback()) {
       rtcp::CongestionControlFeedback rtcp_feedback =
@@ -599,9 +599,9 @@ TEST_P(TransportFeedbackAdapterTest, TransportPacketFeedbackHasDataInFlight) {
   }
 
   std::optional<TransportPacketsFeedback> adapted_feedback_1 =
-      CreateAndProcessFeedback(MakeArrayView(&packets[0], 1), adapter);
+      CreateAndProcessFeedback(std::span(&packets[0], 1), adapter);
   std::optional<TransportPacketsFeedback> adapted_feedback_2 =
-      CreateAndProcessFeedback(MakeArrayView(&packets[1], 1), adapter);
+      CreateAndProcessFeedback(std::span(&packets[1], 1), adapter);
   EXPECT_EQ(adapted_feedback_1->data_in_flight, packets[1].packet_size);
   EXPECT_EQ(adapted_feedback_2->data_in_flight, DataSize::Zero());
 }
@@ -812,7 +812,7 @@ TEST(TransportFeedbackAdapterCongestionFeedbackTest,
   const TimeDelta kExpectedRtt = TimeDelta::Millis(20);
   for (int i = 0; i < 4; i = i + 2) {
     rtcp::CongestionControlFeedback rtcp_feedback =
-        BuildRtcpCongestionControlFeedbackPacket(MakeArrayView(&packets[i], 2));
+        BuildRtcpCongestionControlFeedbackPacket(std::span(&packets[i], 2));
     std::optional<TransportPacketsFeedback> adapted_feedback =
         adapter.ProcessCongestionControlFeedback(
             rtcp_feedback,
diff --git a/modules/desktop_capture/BUILD.gn b/modules/desktop_capture/BUILD.gn
index 3da5304447..6b15ad7b80 100644
--- a/modules/desktop_capture/BUILD.gn
+++ b/modules/desktop_capture/BUILD.gn
@@ -69,7 +69,6 @@ if (rtc_include_tests) {
         ":desktop_capture_mock",
         ":primitives",
         ":screen_drawer",
-        "../../api:array_view",
         "../../rtc_base:base64",
         "../../rtc_base:threading",
         "../../test:test_support",
@@ -163,7 +162,6 @@ if (rtc_include_tests) {
       ":desktop_capture",
       ":desktop_capture_mock",
       ":primitives",
-      "../../api:array_view",
       "../../api/units:time_delta",
       "../../api/units:timestamp",
       "../../rtc_base:checks",
diff --git a/modules/desktop_capture/desktop_frame_unittest.cc b/modules/desktop_capture/desktop_frame_unittest.cc
index ea3b2bb14a..a29cc8eed3 100644
--- a/modules/desktop_capture/desktop_frame_unittest.cc
+++ b/modules/desktop_capture/desktop_frame_unittest.cc
@@ -14,8 +14,8 @@
 #include <cstring>
 #include <memory>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "modules/desktop_capture/desktop_geometry.h"
 #include "test/gtest.h"
 
@@ -77,7 +77,7 @@ void RunTest(const TestData& test) {
   }
 }
 
-void RunTests(ArrayView<const TestData> tests) {
+void RunTests(std::span<const TestData> tests) {
   for (const TestData& test : tests) {
     SCOPED_TRACE(test.description);
 
diff --git a/modules/desktop_capture/screen_capturer_integration_test.cc b/modules/desktop_capture/screen_capturer_integration_test.cc
index a1f61e343a..34996fca10 100644
--- a/modules/desktop_capture/screen_capturer_integration_test.cc
+++ b/modules/desktop_capture/screen_capturer_integration_test.cc
@@ -14,11 +14,11 @@
 #include <initializer_list>
 #include <iostream>  // TODO(zijiehe): Remove once flaky has been resolved.
 #include <memory>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/desktop_capture/desktop_capture_options.h"
 #include "modules/desktop_capture/desktop_capturer.h"
 #include "modules/desktop_capture/desktop_frame.h"
@@ -216,7 +216,7 @@ class ScreenCapturerIntegrationTest : public ::testing::Test {
           // The else if statement is for debugging purpose only,
           // which should be removed after flakiness of
           // ScreenCapturerIntegrationTest has been resolved.
-          ArrayView<const uint8_t> frame_data(
+          std::span<const uint8_t> frame_data(
               frame->data(), frame->size().height() * frame->stride());
           std::string result = Base64Encode(frame_data);
           std::cout << frame->size().width() << " x " << frame->size().height()
diff --git a/modules/pacing/BUILD.gn b/modules/pacing/BUILD.gn
index 11c8d02cc1..083b6ecf2a 100644
--- a/modules/pacing/BUILD.gn
+++ b/modules/pacing/BUILD.gn
@@ -29,7 +29,6 @@ rtc_library("pacing") {
   ]
 
   deps = [
-    "../../api:array_view",
     "../../api:field_trials_view",
     "../../api:rtp_headers",
     "../../api:rtp_packet_sender",
@@ -87,7 +86,6 @@ if (rtc_include_tests) {
     deps = [
       ":interval_budget",
       ":pacing",
-      "../../api:array_view",
       "../../api:field_trials",
       "../../api:rtp_headers",
       "../../api:sequence_checker",
diff --git a/modules/pacing/pacing_controller.cc b/modules/pacing/pacing_controller.cc
index 96fb3cbd0d..8c2dda696e 100644
--- a/modules/pacing/pacing_controller.cc
+++ b/modules/pacing/pacing_controller.cc
@@ -16,12 +16,12 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
 #include "absl/cleanup/cleanup.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/field_trials_view.h"
 #include "api/transport/network_types.h"
 #include "api/units/data_rate.h"
@@ -102,7 +102,7 @@ PacingController::PacingController(Clock* clock,
 PacingController::~PacingController() = default;
 
 void PacingController::CreateProbeClusters(
-    ArrayView<const ProbeClusterConfig> probe_cluster_configs) {
+    std::span<const ProbeClusterConfig> probe_cluster_configs) {
   for (const ProbeClusterConfig probe_cluster_config : probe_cluster_configs) {
     prober_.CreateProbeCluster(probe_cluster_config);
   }
diff --git a/modules/pacing/pacing_controller.h b/modules/pacing/pacing_controller.h
index 0311423915..6e244014cd 100644
--- a/modules/pacing/pacing_controller.h
+++ b/modules/pacing/pacing_controller.h
@@ -17,10 +17,10 @@
 #include <array>
 #include <memory>
 #include <optional>
+#include <span>
 #include <vector>
 
 #include "absl/base/attributes.h"
-#include "api/array_view.h"
 #include "api/field_trials_view.h"
 #include "api/rtp_packet_sender.h"
 #include "api/transport/network_types.h"
@@ -59,7 +59,7 @@ class PacingController {
     // have been updated.
     virtual void OnAbortedRetransmissions(
         uint32_t /* ssrc */,
-        ArrayView<const uint16_t> /* sequence_numbers */) {}
+        std::span<const uint16_t> /* sequence_numbers */) {}
     virtual std::optional<uint32_t> GetRtxSsrcForMedia(
         uint32_t /* ssrc */) const {
       return std::nullopt;
@@ -133,7 +133,7 @@ class PacingController {
   void EnqueuePacket(std::unique_ptr<RtpPacketToSend> packet);
 
   void CreateProbeClusters(
-      ArrayView<const ProbeClusterConfig> probe_cluster_configs);
+      std::span<const ProbeClusterConfig> probe_cluster_configs);
 
   void Pause();   // Temporarily pause all sending.
   void Resume();  // Resume sending packets.
diff --git a/modules/pacing/pacing_controller_unittest.cc b/modules/pacing/pacing_controller_unittest.cc
index 84fa97c5c7..0370248aca 100644
--- a/modules/pacing/pacing_controller_unittest.cc
+++ b/modules/pacing/pacing_controller_unittest.cc
@@ -17,10 +17,10 @@
 #include <cstdlib>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/field_trials.h"
 #include "api/transport/network_types.h"
 #include "api/units/data_rate.h"
@@ -139,7 +139,7 @@ class MockPacingControllerCallback : public PacingController::PacketSender {
   MOCK_METHOD(size_t, SendPadding, (size_t target_size));
   MOCK_METHOD(void,
               OnAbortedRetransmissions,
-              (uint32_t, ArrayView<const uint16_t>),
+              (uint32_t, std::span<const uint16_t>),
               (override));
   MOCK_METHOD(std::optional<uint32_t>,
               GetRtxSsrcForMedia,
@@ -167,7 +167,7 @@ class MockPacketSender : public PacingController::PacketSender {
               (override));
   MOCK_METHOD(void,
               OnAbortedRetransmissions,
-              (uint32_t, ArrayView<const uint16_t>),
+              (uint32_t, std::span<const uint16_t>),
               (override));
   MOCK_METHOD(std::optional<uint32_t>,
               GetRtxSsrcForMedia,
@@ -205,7 +205,7 @@ class PacingControllerPadding : public PacingController::PacketSender {
     return packets;
   }
 
-  void OnAbortedRetransmissions(uint32_t, ArrayView<const uint16_t>) override {}
+  void OnAbortedRetransmissions(uint32_t, std::span<const uint16_t>) override {}
   std::optional<uint32_t> GetRtxSsrcForMedia(uint32_t) const override {
     return std::nullopt;
   }
@@ -265,7 +265,7 @@ class PacingControllerProbing : public PacingController::PacketSender {
     return packets;
   }
 
-  void OnAbortedRetransmissions(uint32_t, ArrayView<const uint16_t>) override {}
+  void OnAbortedRetransmissions(uint32_t, std::span<const uint16_t>) override {}
   std::optional<uint32_t> GetRtxSsrcForMedia(uint32_t) const override {
     return std::nullopt;
   }
diff --git a/modules/pacing/packet_router.cc b/modules/pacing/packet_router.cc
index 67c3a63fad..a8b4aea186 100644
--- a/modules/pacing/packet_router.cc
+++ b/modules/pacing/packet_router.cc
@@ -14,11 +14,11 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
 #include "absl/functional/any_invocable.h"
-#include "api/array_view.h"
 #include "api/rtp_headers.h"
 #include "api/sequence_checker.h"
 #include "api/transport/network_types.h"
@@ -291,7 +291,7 @@ std::vector<std::unique_ptr<RtpPacketToSend>> PacketRouter::GeneratePadding(
 
 void PacketRouter::OnAbortedRetransmissions(
     uint32_t ssrc,
-    ArrayView<const uint16_t> sequence_numbers) {
+    std::span<const uint16_t> sequence_numbers) {
   RTC_DCHECK_RUN_ON(&thread_checker_);
   auto it = send_modules_map_.find(ssrc);
   if (it != send_modules_map_.end()) {
diff --git a/modules/pacing/packet_router.h b/modules/pacing/packet_router.h
index 2a7336ab91..7b0ab39465 100644
--- a/modules/pacing/packet_router.h
+++ b/modules/pacing/packet_router.h
@@ -19,11 +19,11 @@
 #include <memory>
 #include <optional>
 #include <set>
+#include <span>
 #include <unordered_map>
 #include <vector>
 
 #include "absl/functional/any_invocable.h"
-#include "api/array_view.h"
 #include "api/sequence_checker.h"
 #include "api/transport/network_types.h"
 #include "api/units/data_size.h"
@@ -79,7 +79,7 @@ class PacketRouter : public PacingController::PacketSender {
       DataSize size) override;
   void OnAbortedRetransmissions(
       uint32_t ssrc,
-      ArrayView<const uint16_t> sequence_numbers) override;
+      std::span<const uint16_t> sequence_numbers) override;
   std::optional<uint32_t> GetRtxSsrcForMedia(uint32_t ssrc) const override;
   void OnBatchComplete() override;
 
diff --git a/modules/rtp_rtcp/source/rtp_packet.h b/modules/rtp_rtcp/source/rtp_packet.h
index 78cccb347d..027969aeff 100644
--- a/modules/rtp_rtcp/source/rtp_packet.h
+++ b/modules/rtp_rtcp/source/rtp_packet.h
@@ -203,7 +203,7 @@ class RtpPacket {
   ExtensionInfo& FindOrCreateExtensionInfo(int id);
 
   // Allocates and returns place to store rtp header extension.
-  // Returns empty arrayview on failure.
+  // Returns empty std::span on failure.
   std::span<uint8_t> AllocateRawExtension(int id, size_t length);
 
   // Promotes existing one-byte header extensions to two-byte header extensions
diff --git a/modules/video_coding/BUILD.gn b/modules/video_coding/BUILD.gn
index 029ddb6066..74c0115bb6 100644
--- a/modules/video_coding/BUILD.gn
+++ b/modules/video_coding/BUILD.gn
@@ -47,7 +47,6 @@ rtc_library("frame_dependencies_calculator") {
   ]
 
   deps = [
-    "../../api:array_view",
     "../../common_video/generic_frame_descriptor",
     "../../rtc_base:checks",
     "../../rtc_base:logging",
@@ -113,7 +112,6 @@ rtc_library("h26x_packet_buffer") {
     ":codec_globals_headers",
     ":h264_sprop_parameter_sets",
     ":packet_buffer",
-    "../../api:array_view",
     "../../api/video:video_frame",
     "../../api/video:video_frame_type",
     "../../common_video",
@@ -199,7 +197,6 @@ rtc_library("video_coding") {
     "..:module_api",
     "..:module_api_public",
     "..:module_fec_api",
-    "../../api:array_view",
     "../../api:fec_controller_api",
     "../../api:field_trials_view",
     "../../api:rtp_packet_info",
@@ -338,7 +335,6 @@ rtc_library("video_coding_utility") {
   ]
 
   deps = [
-    "../../api:array_view",
     "../../api:field_trials_view",
     "../../api:scoped_refptr",
     "../../api:sequence_checker",
@@ -619,7 +615,6 @@ rtc_library("webrtc_vp9") {
     ":video_codec_interface",
     ":video_coding_utility",
     ":webrtc_libvpx_interface",
-    "../../api:array_view",
     "../../api:fec_controller_api",
     "../../api:field_trials_view",
     "../../api:refcountedbase",
@@ -854,7 +849,6 @@ if (rtc_include_tests) {
       ":video_coding_utility",
       ":videocodec_test_stats_impl",
       ":webrtc_vp9_helpers",
-      "../../api:array_view",
       "../../api:field_trials_view",
       "../../api:make_ref_counted",
       "../../api:rtp_parameters",
@@ -993,7 +987,6 @@ if (rtc_include_tests) {
       ":webrtc_vp8",
       ":webrtc_vp9",
       ":webrtc_vp9_helpers",
-      "../../api:array_view",
       "../../api:create_frame_generator",
       "../../api:create_videocodec_test_fixture_api",
       "../../api:field_trials",
@@ -1131,7 +1124,6 @@ if (rtc_include_tests) {
       ":webrtc_vp9_helpers",
       "..:module_api",
       "..:module_fec_api",
-      "../../api:array_view",
       "../../api:create_frame_generator",
       "../../api:create_simulcast_test_fixture_api",
       "../../api:fec_controller_api",
diff --git a/modules/video_coding/codecs/av1/BUILD.gn b/modules/video_coding/codecs/av1/BUILD.gn
index 5d6156f3a7..3490a9fcf8 100644
--- a/modules/video_coding/codecs/av1/BUILD.gn
+++ b/modules/video_coding/codecs/av1/BUILD.gn
@@ -121,7 +121,6 @@ if (rtc_include_tests) {
       ":av1_svc_config",
       ":dav1d_decoder",
       "../..:video_codec_interface",
-      "../../../../api:array_view",
       "../../../../api:field_trials",
       "../../../../api:make_ref_counted",
       "../../../../api:scoped_refptr",
diff --git a/modules/video_coding/codecs/av1/dav1d_decoder_unittest.cc b/modules/video_coding/codecs/av1/dav1d_decoder_unittest.cc
index 217edc1048..edb969e99d 100644
--- a/modules/video_coding/codecs/av1/dav1d_decoder_unittest.cc
+++ b/modules/video_coding/codecs/av1/dav1d_decoder_unittest.cc
@@ -13,9 +13,9 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/environment/environment_factory.h"
 #include "api/video/color_space.h"
@@ -45,7 +45,7 @@ constexpr uint8_t kAv1FrameWithBT709FullRangeColorSpace[] = {
     0x22, 0x02, 0x02, 0x03, 0x08, 0x32, 0x0e, 0x10, 0x00, 0xac, 0x02, 0x05,
     0x14, 0x20, 0x81, 0x00, 0x02, 0x00, 0x95, 0xe1, 0xe0};
 
-EncodedImage CreateEncodedImage(ArrayView<const uint8_t> data) {
+EncodedImage CreateEncodedImage(std::span<const uint8_t> data) {
   EncodedImage image;
   image.SetEncodedData(EncodedImageBuffer::Create(data.data(), data.size()));
   return image;
diff --git a/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc b/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc
index 4260669d5b..5c60bdbb46 100644
--- a/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc
+++ b/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc
@@ -24,7 +24,6 @@
 #include "absl/strings/match.h"
 #include "absl/strings/str_replace.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/environment/environment_factory.h"
 #include "api/field_trials_view.h"
 #include "api/make_ref_counted.h"
diff --git a/modules/video_coding/codecs/vp9/libvpx_vp9_decoder.cc b/modules/video_coding/codecs/vp9/libvpx_vp9_decoder.cc
index 77740ef590..0dafd3e201 100644
--- a/modules/video_coding/codecs/vp9/libvpx_vp9_decoder.cc
+++ b/modules/video_coding/codecs/vp9/libvpx_vp9_decoder.cc
@@ -17,8 +17,8 @@
 #include <cstdint>
 #include <cstring>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/scoped_refptr.h"
 #include "api/video/color_space.h"
 #include "api/video/encoded_image.h"
@@ -209,7 +209,7 @@ int LibvpxVp9Decoder::Decode(const EncodedImage& input_image,
   if (input_image.IsKey()) {
     std::optional<Vp9UncompressedHeader> frame_info =
         ParseUncompressedVp9Header(
-            MakeArrayView(input_image.data(), input_image.size()));
+            std::span(input_image.data(), input_image.size()));
     if (frame_info) {
       RenderResolution frame_resolution(frame_info->frame_width,
                                         frame_info->frame_height);
diff --git a/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc b/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc
index 105a6e8215..d757f1aa11 100644
--- a/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc
+++ b/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc
@@ -21,12 +21,12 @@
 #include <memory>
 #include <numeric>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
 #include "absl/algorithm/container.h"
 #include "absl/container/inlined_vector.h"
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/fec_controller_override.h"
 #include "api/field_trials_view.h"
@@ -203,7 +203,7 @@ std::unique_ptr<ScalableVideoController> CreateVp9ScalabilityStructure(
 }
 
 vpx_svc_ref_frame_config_t Vp9References(
-    ArrayView<const ScalableVideoController::LayerFrameConfig> layers) {
+    std::span<const ScalableVideoController::LayerFrameConfig> layers) {
   vpx_svc_ref_frame_config_t ref_config = {};
   for (const ScalableVideoController::LayerFrameConfig& layer_frame : layers) {
     const auto& buffers = layer_frame.Buffers();
diff --git a/modules/video_coding/codecs/vp9/test/vp9_impl_unittest.cc b/modules/video_coding/codecs/vp9/test/vp9_impl_unittest.cc
index c6ffdb941e..30884f44e4 100644
--- a/modules/video_coding/codecs/vp9/test/vp9_impl_unittest.cc
+++ b/modules/video_coding/codecs/vp9/test/vp9_impl_unittest.cc
@@ -14,13 +14,13 @@
 #include <functional>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <tuple>
 #include <vector>
 
 #include "absl/container/inlined_vector.h"
 #include "absl/memory/memory.h"
-#include "api/array_view.h"
 #include "api/environment/environment_factory.h"
 #include "api/field_trials.h"
 #include "api/make_ref_counted.h"
@@ -2116,7 +2116,7 @@ TEST_P(Vp9ImplWithLayeringTest, FlexibleMode) {
     if (picture_idx == 0) {
       EXPECT_EQ(vp9.num_ref_pics, 0) << "Frame " << i;
     } else {
-      EXPECT_THAT(MakeArrayView(vp9.p_diff, vp9.num_ref_pics),
+      EXPECT_THAT(std::span(vp9.p_diff, vp9.num_ref_pics),
                   UnorderedElementsAreArray(gof.pid_diff[gof_idx],
                                             gof.num_ref_pics[gof_idx]))
           << "Frame " << i;
diff --git a/modules/video_coding/frame_dependencies_calculator.cc b/modules/video_coding/frame_dependencies_calculator.cc
index ca046f24bc..2d28a415a4 100644
--- a/modules/video_coding/frame_dependencies_calculator.cc
+++ b/modules/video_coding/frame_dependencies_calculator.cc
@@ -14,10 +14,10 @@
 #include <iterator>
 #include <optional>
 #include <set>
+#include <span>
 
 #include "absl/algorithm/container.h"
 #include "absl/container/inlined_vector.h"
-#include "api/array_view.h"
 #include "common_video/generic_frame_descriptor/generic_frame_info.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
@@ -26,7 +26,7 @@ namespace webrtc {
 
 absl::InlinedVector<int64_t, 5> FrameDependenciesCalculator::FromBuffersUsage(
     int64_t frame_id,
-    ArrayView<const CodecBufferUsage> buffers_usage) {
+    std::span<const CodecBufferUsage> buffers_usage) {
   absl::InlinedVector<int64_t, 5> dependencies;
   RTC_DCHECK_GT(buffers_usage.size(), 0);
   for (const CodecBufferUsage& buffer_usage : buffers_usage) {
diff --git a/modules/video_coding/frame_dependencies_calculator.h b/modules/video_coding/frame_dependencies_calculator.h
index 3a354c563a..d9da436892 100644
--- a/modules/video_coding/frame_dependencies_calculator.h
+++ b/modules/video_coding/frame_dependencies_calculator.h
@@ -14,9 +14,9 @@
 #include <stdint.h>
 
 #include <optional>
+#include <span>
 
 #include "absl/container/inlined_vector.h"
-#include "api/array_view.h"
 #include "common_video/generic_frame_descriptor/generic_frame_info.h"
 
 namespace webrtc {
@@ -32,7 +32,7 @@ class FrameDependenciesCalculator {
   // Calculates frame dependencies based on previous encoder buffer usage.
   absl::InlinedVector<int64_t, 5> FromBuffersUsage(
       int64_t frame_id,
-      ArrayView<const CodecBufferUsage> buffers_usage);
+      std::span<const CodecBufferUsage> buffers_usage);
 
  private:
   struct BufferUsage {
diff --git a/modules/video_coding/generic_decoder_unittest.cc b/modules/video_coding/generic_decoder_unittest.cc
index 7c083459c7..23ecf2664c 100644
--- a/modules/video_coding/generic_decoder_unittest.cc
+++ b/modules/video_coding/generic_decoder_unittest.cc
@@ -12,10 +12,10 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/field_trials.h"
 #include "api/rtp_packet_infos.h"
 #include "api/scoped_refptr.h"
@@ -75,7 +75,7 @@ class ReceiveCallback : public VCMReceiveCallback {
     return ret;
   }
 
-  ArrayView<const VideoFrame> GetAllFrames() const { return frames_; }
+  std::span<const VideoFrame> GetAllFrames() const { return frames_; }
 
   void OnDroppedFrames(uint32_t frames_dropped) override {
     frames_dropped_ += frames_dropped;
diff --git a/modules/video_coding/h264_sps_pps_tracker.cc b/modules/video_coding/h264_sps_pps_tracker.cc
index 3f76077e3e..ebd8035725 100644
--- a/modules/video_coding/h264_sps_pps_tracker.cc
+++ b/modules/video_coding/h264_sps_pps_tracker.cc
@@ -13,10 +13,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/video/video_codec_type.h"
 #include "common_video/h264/h264_common.h"
 #include "common_video/h264/pps_parser.h"
@@ -35,7 +35,7 @@ const uint8_t start_code_h264[] = {0, 0, 0, 1};
 }  // namespace
 
 H264SpsPpsTracker::FixedBitstream H264SpsPpsTracker::CopyAndFixBitstream(
-    ArrayView<const uint8_t> bitstream,
+    std::span<const uint8_t> bitstream,
     RTPVideoHeader* video_header) {
   RTC_DCHECK(video_header);
   RTC_DCHECK(video_header->codec == kVideoCodecH264);
@@ -206,9 +206,9 @@ void H264SpsPpsTracker::InsertSpsPpsNalus(const std::vector<uint8_t>& sps,
     return;
   }
   std::optional<SpsParser::SpsState> parsed_sps = SpsParser::ParseSps(
-      ArrayView<const uint8_t>(sps).subspan(kNaluHeaderOffset));
+      std::span<const uint8_t>(sps).subspan(kNaluHeaderOffset));
   std::optional<PpsParser::PpsState> parsed_pps = PpsParser::ParsePps(
-      ArrayView<const uint8_t>(pps).subspan(kNaluHeaderOffset));
+      std::span<const uint8_t>(pps).subspan(kNaluHeaderOffset));
 
   if (!parsed_sps) {
     RTC_LOG(LS_WARNING) << "Failed to parse SPS.";
diff --git a/modules/video_coding/h264_sps_pps_tracker.h b/modules/video_coding/h264_sps_pps_tracker.h
index 132aaa0568..b8c04c09ab 100644
--- a/modules/video_coding/h264_sps_pps_tracker.h
+++ b/modules/video_coding/h264_sps_pps_tracker.h
@@ -14,9 +14,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <map>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "modules/rtp_rtcp/source/rtp_video_header.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/copy_on_write_buffer.h"
@@ -38,7 +38,7 @@ class H264SpsPpsTracker {
   ~H264SpsPpsTracker() = default;
 
   // Returns fixed bitstream and modifies `video_header`.
-  FixedBitstream CopyAndFixBitstream(ArrayView<const uint8_t> bitstream,
+  FixedBitstream CopyAndFixBitstream(std::span<const uint8_t> bitstream,
                                      RTPVideoHeader* video_header);
 
   void InsertSpsPpsNalus(const std::vector<uint8_t>& sps,
diff --git a/modules/video_coding/h264_sps_pps_tracker_unittest.cc b/modules/video_coding/h264_sps_pps_tracker_unittest.cc
index 079049d59e..adb2307526 100644
--- a/modules/video_coding/h264_sps_pps_tracker_unittest.cc
+++ b/modules/video_coding/h264_sps_pps_tracker_unittest.cc
@@ -11,9 +11,9 @@
 #include "modules/video_coding/h264_sps_pps_tracker.h"
 
 #include <cstdint>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/video/video_codec_type.h"
 #include "common_video/h264/h264_common.h"
 #include "modules/rtp_rtcp/source/rtp_video_header.h"
@@ -30,7 +30,7 @@ using ::testing::SizeIs;
 
 const uint8_t start_code[] = {0, 0, 0, 1};
 
-ArrayView<const uint8_t> Bitstream(
+std::span<const uint8_t> Bitstream(
     const H264SpsPpsTracker::FixedBitstream& fixed) {
   return fixed.bitstream;
 }
diff --git a/modules/video_coding/h26x_packet_buffer.cc b/modules/video_coding/h26x_packet_buffer.cc
index 10c22f3dc9..cee203a757 100644
--- a/modules/video_coding/h26x_packet_buffer.cc
+++ b/modules/video_coding/h26x_packet_buffer.cc
@@ -17,12 +17,12 @@
 #include <limits>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/algorithm/container.h"
-#include "api/array_view.h"
 #include "api/video/video_codec_type.h"
 #include "api/video/video_frame_type.h"
 #include "common_video/h264/h264_common.h"
@@ -77,7 +77,7 @@ bool HasSps(const H26xPacketBuffer::Packet& packet) {
   });
 }
 
-int64_t* GetContinuousSequence(ArrayView<int64_t> last_continuous,
+int64_t* GetContinuousSequence(std::span<int64_t> last_continuous,
                                int64_t unwrapped_seq_num) {
   for (int64_t& last : last_continuous) {
     if (unwrapped_seq_num - 1 == last) {
@@ -361,9 +361,9 @@ void H26xPacketBuffer::InsertSpsPpsNalus(const std::vector<uint8_t>& sps,
     return;
   }
   std::optional<SpsParser::SpsState> parsed_sps = SpsParser::ParseSps(
-      ArrayView<const uint8_t>(sps).subspan(kNaluHeaderOffset));
+      std::span<const uint8_t>(sps).subspan(kNaluHeaderOffset));
   std::optional<PpsParser::PpsState> parsed_pps = PpsParser::ParsePps(
-      ArrayView<const uint8_t>(pps).subspan(kNaluHeaderOffset));
+      std::span<const uint8_t>(pps).subspan(kNaluHeaderOffset));
 
   if (!parsed_sps) {
     RTC_LOG(LS_WARNING) << "Failed to parse SPS.";
diff --git a/modules/video_coding/h26x_packet_buffer_unittest.cc b/modules/video_coding/h26x_packet_buffer_unittest.cc
index f786776d41..8a50b786d2 100644
--- a/modules/video_coding/h26x_packet_buffer_unittest.cc
+++ b/modules/video_coding/h26x_packet_buffer_unittest.cc
@@ -12,11 +12,11 @@
 #include <cstdint>
 #include <cstring>
 #include <memory>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/video/render_resolution.h"
 #include "api/video/video_codec_type.h"
 #include "api/video/video_frame_type.h"
@@ -378,7 +378,7 @@ H265Packet& H265Packet::SeqNum(int64_t rtp_seq_num) {
 }
 #endif
 
-ArrayView<const uint8_t> PacketPayload(
+std::span<const uint8_t> PacketPayload(
     const std::unique_ptr<H26xPacketBuffer::Packet>& packet) {
   return packet->video_payload;
 }
diff --git a/modules/video_coding/loss_notification_controller.cc b/modules/video_coding/loss_notification_controller.cc
index 97f86dc57a..4f4a0087ff 100644
--- a/modules/video_coding/loss_notification_controller.cc
+++ b/modules/video_coding/loss_notification_controller.cc
@@ -12,8 +12,8 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/sequence_checker.h"
 #include "modules/include/module_common_types.h"
 #include "rtc_base/checks.h"
@@ -113,7 +113,7 @@ void LossNotificationController::OnAssembledFrame(
     uint16_t first_seq_num,
     int64_t frame_id,
     bool discardable,
-    ArrayView<const int64_t> frame_dependencies) {
+    std::span<const int64_t> frame_dependencies) {
   RTC_DCHECK_RUN_ON(&sequence_checker_);
 
   DiscardOldInformation();  // Prevent memory overconsumption.
@@ -139,7 +139,7 @@ void LossNotificationController::DiscardOldInformation() {
 }
 
 bool LossNotificationController::AllDependenciesDecodable(
-    ArrayView<const int64_t> frame_dependencies) const {
+    std::span<const int64_t> frame_dependencies) const {
   RTC_DCHECK_RUN_ON(&sequence_checker_);
 
   // Due to packet reordering, frame buffering and asynchronous decoders, it is
diff --git a/modules/video_coding/loss_notification_controller.h b/modules/video_coding/loss_notification_controller.h
index f6b99925e2..07265842fa 100644
--- a/modules/video_coding/loss_notification_controller.h
+++ b/modules/video_coding/loss_notification_controller.h
@@ -15,8 +15,8 @@
 
 #include <optional>
 #include <set>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/sequence_checker.h"
 #include "modules/include/module_common_types.h"
 #include "rtc_base/system/no_unique_address.h"
@@ -29,7 +29,7 @@ class LossNotificationController {
   struct FrameDetails {
     bool is_keyframe;
     int64_t frame_id;
-    ArrayView<const int64_t> frame_dependencies;
+    std::span<const int64_t> frame_dependencies;
   };
 
   LossNotificationController(KeyFrameRequestSender* key_frame_request_sender,
@@ -45,13 +45,13 @@ class LossNotificationController {
   void OnAssembledFrame(uint16_t first_seq_num,
                         int64_t frame_id,
                         bool discardable,
-                        ArrayView<const int64_t> frame_dependencies);
+                        std::span<const int64_t> frame_dependencies);
 
  private:
   void DiscardOldInformation();
 
   bool AllDependenciesDecodable(
-      ArrayView<const int64_t> frame_dependencies) const;
+      std::span<const int64_t> frame_dependencies) const;
 
   // When the loss of a packet or the non-decodability of a frame is detected,
   // produces a key frame request or a loss notification.
diff --git a/modules/video_coding/packet_buffer_unittest.cc b/modules/video_coding/packet_buffer_unittest.cc
index af994c1aed..29f13e4f3e 100644
--- a/modules/video_coding/packet_buffer_unittest.cc
+++ b/modules/video_coding/packet_buffer_unittest.cc
@@ -14,10 +14,10 @@
 #include <limits>
 #include <memory>
 #include <ostream>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/video/video_codec_type.h"
 #include "api/video/video_frame_type.h"
 #include "common_video/h264/h264_common.h"
@@ -48,7 +48,7 @@ void IgnoreResult(PacketBuffer::InsertResult /*result*/) {}
 // Validates frame boundaries are valid and returns first sequence_number for
 // each frame.
 std::vector<uint16_t> StartSeqNums(
-    ArrayView<const std::unique_ptr<PacketBuffer::Packet>> packets) {
+    std::span<const std::unique_ptr<PacketBuffer::Packet>> packets) {
   std::vector<uint16_t> result;
   bool frame_boundary = true;
   for (const auto& packet : packets) {
@@ -117,7 +117,7 @@ class PacketBufferTest : public ::testing::Test {
                                   IsKeyFrame keyframe,  // is keyframe
                                   IsFirst first,  // is first packet of frame
                                   IsLast last,    // is last packet of frame
-                                  ArrayView<const uint8_t> data = {},
+                                  std::span<const uint8_t> data = {},
                                   uint32_t timestamp = 123u) {  // rtp timestamp
     auto packet = std::make_unique<PacketBuffer::Packet>();
     packet->video_header.codec = kVideoCodecGeneric;
@@ -421,7 +421,7 @@ class PacketBufferH264Test : public PacketBufferTest {
       IsFirst first,        // is first packet of frame
       IsLast last,          // is last packet of frame
       uint32_t timestamp,   // rtp timestamp
-      ArrayView<const uint8_t> data = {},
+      std::span<const uint8_t> data = {},
       uint32_t width = 0,      // width of frame (SPS/IDR)
       uint32_t height = 0,     // height of frame (SPS/IDR)
       bool generic = false) {  // has generic descriptor
@@ -459,7 +459,7 @@ class PacketBufferH264Test : public PacketBufferTest {
       IsFirst first,        // is first packet of frame
       IsLast last,          // is last packet of frame
       uint32_t timestamp,   // rtp timestamp
-      ArrayView<const uint8_t> data = {},
+      std::span<const uint8_t> data = {},
       uint32_t width = 0,     // width of frame (SPS/IDR)
       uint32_t height = 0) {  // height of frame (SPS/IDR)
     auto packet = std::make_unique<PacketBuffer::Packet>();
diff --git a/modules/video_coding/rtp_vp8_ref_finder_unittest.cc b/modules/video_coding/rtp_vp8_ref_finder_unittest.cc
index 0ed70b2e39..b52d1fb811 100644
--- a/modules/video_coding/rtp_vp8_ref_finder_unittest.cc
+++ b/modules/video_coding/rtp_vp8_ref_finder_unittest.cc
@@ -13,10 +13,10 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/rtp_packet_infos.h"
 #include "api/video/encoded_frame.h"
 #include "api/video/encoded_image.h"
@@ -44,7 +44,7 @@ namespace {
 MATCHER_P2(HasIdAndRefs, id, refs, "") {
   return Matches(Eq(id))(arg->Id()) &&
          Matches(UnorderedElementsAreArray(refs))(
-             ArrayView<int64_t>(arg->references, arg->num_references));
+             std::span<int64_t>(arg->references, arg->num_references));
 }
 
 Matcher<const std::vector<std::unique_ptr<EncodedFrame>>&>
diff --git a/modules/video_coding/rtp_vp9_ref_finder_unittest.cc b/modules/video_coding/rtp_vp9_ref_finder_unittest.cc
index ab3c67bf04..08633251a9 100644
--- a/modules/video_coding/rtp_vp9_ref_finder_unittest.cc
+++ b/modules/video_coding/rtp_vp9_ref_finder_unittest.cc
@@ -16,10 +16,10 @@
 #include <memory>
 #include <optional>
 #include <ostream>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/rtp_packet_infos.h"
 #include "api/video/encoded_frame.h"
 #include "api/video/encoded_image.h"
@@ -192,7 +192,7 @@ class HasFrameMatcher : public MatcherInterface<const FrameVector&> {
       return false;
     }
 
-    ArrayView<int64_t> actual_refs((*it)->references, (*it)->num_references);
+    std::span<int64_t> actual_refs((*it)->references, (*it)->num_references);
     if (!Matches(UnorderedElementsAreArray(expected_refs_))(actual_refs)) {
       if (result_listener->IsInterested()) {
         *result_listener << "Frame with frame_id:" << frame_id_ << " and "
diff --git a/modules/video_coding/svc/BUILD.gn b/modules/video_coding/svc/BUILD.gn
index 7a7864b645..dc08441516 100644
--- a/modules/video_coding/svc/BUILD.gn
+++ b/modules/video_coding/svc/BUILD.gn
@@ -127,7 +127,6 @@ if (rtc_include_tests) {
       ":scalable_video_controller",
       "..:chain_diff_calculator",
       "..:frame_dependencies_calculator",
-      "../../../api:array_view",
       "../../../api/transport/rtp:dependency_descriptor",
       "../../../api/video:video_bitrate_allocation",
       "../../../api/video:video_frame",
diff --git a/modules/video_coding/svc/scalability_structure_key_svc_unittest.cc b/modules/video_coding/svc/scalability_structure_key_svc_unittest.cc
index 145a278234..71d64a3f47 100644
--- a/modules/video_coding/svc/scalability_structure_key_svc_unittest.cc
+++ b/modules/video_coding/svc/scalability_structure_key_svc_unittest.cc
@@ -9,9 +9,9 @@
  */
 #include "modules/video_coding/svc/scalability_structure_key_svc.h"
 
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "common_video/generic_frame_descriptor/generic_frame_info.h"
 #include "modules/video_coding/svc/scalability_structure_test_helpers.h"
 #include "test/gmock.h"
@@ -234,7 +234,7 @@ TEST(ScalabilityStructureL3T3KeyTest, ReenablingSpatialLayerTriggersKeyFrame) {
   EXPECT_EQ(frames[13].temporal_id, 0);
   EXPECT_EQ(frames[14].temporal_id, 0);
   EXPECT_EQ(frames[15].temporal_id, 0);
-  auto all_frames = MakeArrayView(frames.data(), frames.size());
+  auto all_frames = std::span(frames.data(), frames.size());
   EXPECT_TRUE(wrapper.FrameReferencesAreValid(all_frames.subspan(0, 13)));
   // Frames starting from the frame#13 should not reference any earlier frames.
   EXPECT_TRUE(wrapper.FrameReferencesAreValid(all_frames.subspan(13)));
diff --git a/modules/video_coding/svc/scalability_structure_l2t2_key_shift_unittest.cc b/modules/video_coding/svc/scalability_structure_l2t2_key_shift_unittest.cc
index 153294014d..ebf3ce42a1 100644
--- a/modules/video_coding/svc/scalability_structure_l2t2_key_shift_unittest.cc
+++ b/modules/video_coding/svc/scalability_structure_l2t2_key_shift_unittest.cc
@@ -9,9 +9,9 @@
  */
 #include "modules/video_coding/svc/scalability_structure_l2t2_key_shift.h"
 
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "common_video/generic_frame_descriptor/generic_frame_info.h"
 #include "modules/video_coding/svc/scalability_structure_test_helpers.h"
 #include "test/gmock.h"
@@ -236,8 +236,7 @@ TEST(ScalabilityStructureL2T2KeyShiftTest, ReenableS1TriggersKeyFrame) {
   EXPECT_THAT(frames[4].temporal_id, 1);
 
   // Expect frame[5] to be a key frame.
-  EXPECT_TRUE(
-      wrapper.FrameReferencesAreValid(MakeArrayView(frames.data() + 5, 4)));
+  EXPECT_TRUE(wrapper.FrameReferencesAreValid(std::span(frames.data() + 5, 4)));
 
   EXPECT_THAT(frames[5].spatial_id, 0);
   EXPECT_THAT(frames[6].spatial_id, 1);
diff --git a/modules/video_coding/svc/scalability_structure_test_helpers.cc b/modules/video_coding/svc/scalability_structure_test_helpers.cc
index 004b7d245e..7b4f1e4764 100644
--- a/modules/video_coding/svc/scalability_structure_test_helpers.cc
+++ b/modules/video_coding/svc/scalability_structure_test_helpers.cc
@@ -12,10 +12,10 @@
 #include <bitset>
 #include <cstddef>
 #include <cstdint>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/video/video_bitrate_allocation.h"
 #include "common_video/generic_frame_descriptor/generic_frame_info.h"
 #include "modules/video_coding/chain_diff_calculator.h"
@@ -66,7 +66,7 @@ void ScalabilityStructureWrapper::GenerateFrames(
 }
 
 bool ScalabilityStructureWrapper::FrameReferencesAreValid(
-    ArrayView<const GenericFrameInfo> frames) const {
+    std::span<const GenericFrameInfo> frames) const {
   bool valid = true;
   // VP9 and AV1 supports up to 8 buffers. Expect no more buffers are not used.
   std::bitset<8> buffer_contains_frame;
diff --git a/modules/video_coding/svc/scalability_structure_test_helpers.h b/modules/video_coding/svc/scalability_structure_test_helpers.h
index 42257b59a1..ddc492a8a2 100644
--- a/modules/video_coding/svc/scalability_structure_test_helpers.h
+++ b/modules/video_coding/svc/scalability_structure_test_helpers.h
@@ -12,9 +12,9 @@
 
 #include <stdint.h>
 
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/video/video_bitrate_allocation.h"
 #include "common_video/generic_frame_descriptor/generic_frame_info.h"
 #include "modules/video_coding/chain_diff_calculator.h"
@@ -43,7 +43,7 @@ class ScalabilityStructureWrapper {
   // Returns false and ADD_FAILUREs for frames with invalid references.
   // In particular validates no frame frame reference to frame before frames[0].
   // In error messages frames are indexed starting with 0.
-  bool FrameReferencesAreValid(ArrayView<const GenericFrameInfo> frames) const;
+  bool FrameReferencesAreValid(std::span<const GenericFrameInfo> frames) const;
 
  private:
   ScalableVideoController& structure_controller_;
diff --git a/modules/video_coding/svc/scalability_structure_unittest.cc b/modules/video_coding/svc/scalability_structure_unittest.cc
index 849f08f147..1d95c6fffd 100644
--- a/modules/video_coding/svc/scalability_structure_unittest.cc
+++ b/modules/video_coding/svc/scalability_structure_unittest.cc
@@ -14,11 +14,11 @@
 #include <optional>
 #include <ostream>
 #include <set>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/transport/rtp/dependency_descriptor.h"
 #include "api/video/video_bitrate_allocation.h"
 #include "api/video_codecs/scalability_mode.h"
@@ -115,14 +115,12 @@ TEST_P(ScalabilityStructureTest,
       controller->StreamConfig();
   EXPECT_EQ(config.num_spatial_layers, static_config->num_spatial_layers);
   EXPECT_EQ(config.num_temporal_layers, static_config->num_temporal_layers);
-  EXPECT_THAT(
-      MakeArrayView(config.scaling_factor_num, config.num_spatial_layers),
-      ElementsAreArray(static_config->scaling_factor_num,
-                       static_config->num_spatial_layers));
-  EXPECT_THAT(
-      MakeArrayView(config.scaling_factor_den, config.num_spatial_layers),
-      ElementsAreArray(static_config->scaling_factor_den,
-                       static_config->num_spatial_layers));
+  EXPECT_THAT(std::span(config.scaling_factor_num, config.num_spatial_layers),
+              ElementsAreArray(static_config->scaling_factor_num,
+                               static_config->num_spatial_layers));
+  EXPECT_THAT(std::span(config.scaling_factor_den, config.num_spatial_layers),
+              ElementsAreArray(static_config->scaling_factor_den,
+                               static_config->num_spatial_layers));
 }
 
 TEST_P(ScalabilityStructureTest,
diff --git a/modules/video_coding/utility/qp_parser.cc b/modules/video_coding/utility/qp_parser.cc
index 491bae8e56..95cf4fdb7c 100644
--- a/modules/video_coding/utility/qp_parser.cc
+++ b/modules/video_coding/utility/qp_parser.cc
@@ -13,8 +13,8 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/video/video_codec_constants.h"
 #include "api/video/video_codec_type.h"
 #include "modules/video_coding/utility/vp8_header_parser.h"
@@ -58,7 +58,7 @@ std::optional<uint32_t> QpParser::H264QpParser::Parse(const uint8_t* frame_data,
                                                       size_t frame_size) {
   MutexLock lock(&mutex_);
   bitstream_parser_.ParseBitstream(
-      ArrayView<const uint8_t>(frame_data, frame_size));
+      std::span<const uint8_t>(frame_data, frame_size));
   return bitstream_parser_.GetLastSliceQp();
 }
 
@@ -67,7 +67,7 @@ std::optional<uint32_t> QpParser::H265QpParser::Parse(const uint8_t* frame_data,
                                                       size_t frame_size) {
   MutexLock lock(&mutex_);
   bitstream_parser_.ParseBitstream(
-      ArrayView<const uint8_t>(frame_data, frame_size));
+      std::span<const uint8_t>(frame_data, frame_size));
   return bitstream_parser_.GetLastSliceQp();
 }
 #endif
diff --git a/modules/video_coding/utility/vp9_uncompressed_header_parser.cc b/modules/video_coding/utility/vp9_uncompressed_header_parser.cc
index ca3c8595c4..4219ecd4de 100644
--- a/modules/video_coding/utility/vp9_uncompressed_header_parser.cc
+++ b/modules/video_coding/utility/vp9_uncompressed_header_parser.cc
@@ -12,10 +12,10 @@
 #include <cstddef>
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "modules/video_coding/utility/vp9_constants.h"
 #include "rtc_base/bitstream_reader.h"
 #include "rtc_base/logging.h"
@@ -513,7 +513,7 @@ void Parse(BitstreamReader& br,
 }
 
 std::optional<Vp9UncompressedHeader> ParseUncompressedVp9Header(
-    ArrayView<const uint8_t> buf) {
+    std::span<const uint8_t> buf) {
   BitstreamReader reader(buf);
   Vp9UncompressedHeader frame_info;
   Parse(reader, &frame_info, /*qp_only=*/false);
@@ -526,7 +526,7 @@ std::optional<Vp9UncompressedHeader> ParseUncompressedVp9Header(
 namespace vp9 {
 
 bool GetQp(const uint8_t* buf, size_t length, int* qp) {
-  BitstreamReader reader(MakeArrayView(buf, length));
+  BitstreamReader reader(std::span(buf, length));
   Vp9UncompressedHeader frame_info;
   Parse(reader, &frame_info, /*qp_only=*/true);
   if (!reader.Ok()) {
diff --git a/modules/video_coding/utility/vp9_uncompressed_header_parser.h b/modules/video_coding/utility/vp9_uncompressed_header_parser.h
index 0153a3b579..400559b17d 100644
--- a/modules/video_coding/utility/vp9_uncompressed_header_parser.h
+++ b/modules/video_coding/utility/vp9_uncompressed_header_parser.h
@@ -17,9 +17,9 @@
 #include <array>
 #include <bitset>
 #include <optional>
+#include <span>
 #include <string>
 
-#include "api/array_view.h"
 #include "modules/video_coding/utility/vp9_constants.h"
 
 namespace webrtc {
@@ -151,7 +151,7 @@ struct Vp9UncompressedHeader {
 // Parses the uncompressed header and populates (most) values in a
 // UncompressedHeader struct. Returns nullopt on failure.
 std::optional<Vp9UncompressedHeader> ParseUncompressedVp9Header(
-    ArrayView<const uint8_t> buf);
+    std::span<const uint8_t> buf);
 
 }  // namespace webrtc
 

From 830e031e2c6671b0ece4d89703ba828e1bff0d14 Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Fri, 27 Mar 2026 13:00:36 +0000
Subject: [PATCH 433/472] Use the PayloadType class rather than uint8_t in
 Codec class
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Also replace "int" with "PayloadType" in a number of interfaces, and
make appropriate adjustments.

This is one step towards controlling the issuing and consumption
of PayloadType values.

Bug: webrtc:360058654
Change-Id: I146bc4e3e9966b98864277eee6f52b75ba10d89a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/460021
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47286}
---
 api/BUILD.gn                                 |  1 +
 api/payload_type.h                           | 49 ++++++++++++++++----
 api/webrtc_sdp.cc                            | 39 +++++++++-------
 call/BUILD.gn                                |  1 +
 call/payload_type_picker_unittest.cc         |  2 +-
 media/BUILD.gn                               |  4 ++
 media/base/codec.cc                          | 38 +++++++++------
 media/base/codec.h                           | 41 +++++++++-------
 media/base/codec_comparators.cc              | 31 ++++++++-----
 media/base/codec_list.cc                     |  9 ++--
 media/base/media_channel.h                   |  4 +-
 media/engine/webrtc_video_engine.cc          |  9 ++--
 media/engine/webrtc_video_engine_unittest.cc |  3 +-
 media/engine/webrtc_voice_engine.cc          | 16 ++++---
 media/engine/webrtc_voice_engine.h           |  3 +-
 media/engine/webrtc_voice_engine_unittest.cc |  3 +-
 pc/BUILD.gn                                  |  6 +++
 pc/channel.cc                                |  5 +-
 pc/codec_vendor.cc                           |  4 +-
 pc/codec_vendor_unittest.cc                  |  3 +-
 pc/rtp_parameters_conversion.cc              |  2 +-
 pc/sdp_payload_type_suggester.cc             |  7 +++
 pc/sdp_payload_type_suggester_unittest.cc    | 14 ++++--
 pc/simulcast_sdp_serializer.cc               |  3 +-
 rtc_base/strong_alias.h                      |  7 +++
 25 files changed, 203 insertions(+), 101 deletions(-)

diff --git a/api/BUILD.gn b/api/BUILD.gn
index 5e0e0f4fe0..3555ec0202 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -340,6 +340,7 @@ rtc_library("jsep") {
   ]
   deps = [
     ":candidate",
+    ":payload_type",
     ":ref_count",
     ":rtc_error",
     ":rtp_parameters",
diff --git a/api/payload_type.h b/api/payload_type.h
index 9d4a4434cf..dc40b1f696 100644
--- a/api/payload_type.h
+++ b/api/payload_type.h
@@ -11,31 +11,64 @@
 #ifndef API_PAYLOAD_TYPE_H_
 #define API_PAYLOAD_TYPE_H_
 
-#include <cstdint>
+#include <optional>
 
 #include "absl/strings/str_format.h"
 #include "rtc_base/strong_alias.h"
 
 namespace webrtc {
 
-class PayloadType : public StrongAlias<class PayloadTypeTag, uint8_t> {
+class PayloadType : public StrongAlias<class PayloadTypeTag, int> {
  public:
+  // The default constructor makes a NotSet.
+  PayloadType() : StrongAlias(-1) {}
   // Non-explicit conversions from and to ints are to be deprecated and
   // removed once calling code is upgraded.
-  PayloadType(uint8_t pt) { value_ = pt; }                // NOLINT: explicit
-  constexpr operator uint8_t() const& { return value_; }  // NOLINT: Explicit
-  static bool IsValid(PayloadType id, bool rtcp_mux) {
+  constexpr PayloadType(int pt) : StrongAlias(pt) {  // NOLINT: explicit
+    // The number of tests that use invalid values is high enough that
+    // this DCHECK can't be deployed yet.
+    // Also, allow -1 as argument as a temporary measure. Those calls should
+    // eventually be replaced with PayloadType::NotSet() values.
+    // Intended check:
+    // RTC_DCHECK(pt >= -1 && pt <= 127) << "Payload type " << pt << " is
+    // invalid";
+  }
+
+  constexpr operator int() const& { return value(); }  // NOLINT: explicit
+
+  // Factory function to create a value if you need to check for
+  // values in the valid range.
+  static std::optional<PayloadType> Create(int pt) {
+    if (pt < 0 || pt > 127) {
+      return std::nullopt;
+    }
+    return PayloadType(pt);
+  }
+  // Factory function for the NotSet value. This should be the only way
+  // to create a value outside the valid range.
+  static constexpr PayloadType NotSet() { return PayloadType(Internal{}, -1); }
+  bool Valid(bool rtcp_mux = false) {
     // A payload type is a 7-bit value in the RTP header, so max = 127.
     // If RTCP multiplexing is used, the numbers from 64 to 95 are reserved
     // for RTCP packets.
-    if (rtcp_mux && (id > 63 && id < 96)) {
+    if (rtcp_mux && (value() > 63 && value() < 96)) {
       return false;
     }
-    return id >= 0 && id <= 127;
+    return value() >= 0 && value() <= 127;
   }
+  // Older interface to validity check.
+  static bool IsValid(PayloadType id, bool rtcp_mux) {
+    return id.Valid(rtcp_mux);
+  }
+  bool IsSet() { return value() >= 0; }
+
+ private:
+  class Internal {};
+  // Allow -1 for "NotSet"
+  explicit constexpr PayloadType(Internal tag, int pt) : StrongAlias(pt) {}
   template <typename Sink>
   friend void AbslStringify(Sink& sink, const PayloadType pt) {
-    absl::Format(&sink, "%d", pt.value_);
+    absl::Format(&sink, "%d", pt.value());
   }
 };
 
diff --git a/api/webrtc_sdp.cc b/api/webrtc_sdp.cc
index aab9251257..67eee80801 100644
--- a/api/webrtc_sdp.cc
+++ b/api/webrtc_sdp.cc
@@ -33,6 +33,7 @@
 #include "api/candidate.h"
 #include "api/jsep.h"
 #include "api/media_types.h"
+#include "api/payload_type.h"
 #include "api/rtc_error.h"
 #include "api/rtp_parameters.h"
 #include "api/rtp_transceiver_direction.h"
@@ -192,7 +193,7 @@ const char kDefaultSctpmapProtocol[] = "webrtc-datachannel";
 
 // RTP payload type is in the 0-127 range. Use -1 to indicate "all" payload
 // types.
-const int kWildcardPayloadType = -1;
+constexpr PayloadType kWildcardPayloadType = PayloadType::NotSet();
 
 // Check if passed character is a "token-char" from RFC 4566.
 // https://datatracker.ietf.org/doc/html/rfc4566#section-9
@@ -1013,7 +1014,7 @@ bool GetParameter(const std::string& name,
   return true;
 }
 
-void WriteFmtpHeader(int payload_type, StringBuilder* os) {
+void WriteFmtpHeader(PayloadType payload_type, StringBuilder* os) {
   // fmtp header: a=fmtp:`payload_type` <parameters>
   // Add a=fmtp
   InitAttrLine(kAttributeFmtp, os);
@@ -1021,7 +1022,7 @@ void WriteFmtpHeader(int payload_type, StringBuilder* os) {
   *os << kSdpDelimiterColon << payload_type;
 }
 
-void WritePacketizationHeader(int payload_type, StringBuilder* os) {
+void WritePacketizationHeader(PayloadType payload_type, StringBuilder* os) {
   // packetization header: a=packetization:`payload_type` <packetization_format>
   // Add a=packetization
   InitAttrLine(kAttributePacketization, os);
@@ -1029,7 +1030,7 @@ void WritePacketizationHeader(int payload_type, StringBuilder* os) {
   *os << kSdpDelimiterColon << payload_type;
 }
 
-void WriteRtcpFbHeader(int payload_type, StringBuilder* os) {
+void WriteRtcpFbHeader(PayloadType payload_type, StringBuilder* os) {
   // rtcp-fb header: a=rtcp-fb:`payload_type`
   // <parameters>/<ccm <ccm_parameters>>
   // Add a=rtcp-fb
@@ -2035,7 +2036,7 @@ void BackfillCodecParameters(std::vector<Codec>& codecs) {
 // with that payload type.
 Codec GetCodecWithPayloadType(MediaType type,
                               const std::vector<Codec>& codecs,
-                              int payload_type) {
+                              PayloadType payload_type) {
   const Codec* codec = FindCodecById(codecs, payload_type);
   if (codec)
     return *codec;
@@ -2050,7 +2051,7 @@ Codec GetCodecWithPayloadType(MediaType type,
 // Adds or updates existing codec corresponding to `payload_type` according
 // to `parameters`.
 void UpdateCodec(MediaContentDescription* content_desc,
-                 int payload_type,
+                 PayloadType payload_type,
                  const CodecParameterMap& parameters) {
   // Codec might already have been populated (from rtpmap).
   Codec new_codec = GetCodecWithPayloadType(
@@ -2062,7 +2063,7 @@ void UpdateCodec(MediaContentDescription* content_desc,
 // Adds or updates existing codec corresponding to `payload_type` according
 // to `feedback_param`.
 void UpdateCodec(MediaContentDescription* content_desc,
-                 int payload_type,
+                 PayloadType payload_type,
                  const FeedbackParam& feedback_param) {
   // Codec might already have been populated (from rtpmap).
   Codec new_codec = GetCodecWithPayloadType(
@@ -2098,11 +2099,12 @@ bool ParseFmtpAttributes(absl::string_view line,
     return false;
   }
 
-  int payload_type = 0;
-  if (!GetPayloadTypeFromString(line_payload, payload_type_str, &payload_type,
+  int pt_int = 0;
+  if (!GetPayloadTypeFromString(line_payload, payload_type_str, &pt_int,
                                 error)) {
     return false;
   }
+  PayloadType payload_type = PayloadType(pt_int);
 
   // Parse out format specific parameters.
   CodecParameterMap codec_params;
@@ -2224,7 +2226,7 @@ bool ParseSsrcAttribute(absl::string_view line,
 
 // Updates or creates a new codec entry in the audio description with according
 // to `name`, `clockrate`, `bitrate`, and `channels`.
-void UpdateCodec(int payload_type,
+void UpdateCodec(PayloadType payload_type,
                  absl::string_view name,
                  int clockrate,
                  int bitrate,
@@ -2243,7 +2245,7 @@ void UpdateCodec(int payload_type,
 
 // Updates or creates a new codec entry in the video description according to
 // `name`, `width`, `height`, and `framerate`.
-void UpdateCodec(int payload_type,
+void UpdateCodec(PayloadType payload_type,
                  absl::string_view name,
                  MediaContentDescription* desc) {
   // Codec may already be populated with (only) optional parameters
@@ -2358,7 +2360,7 @@ bool ParseRtpmapAttribute(absl::string_view line,
 // Adds or updates existing video codec corresponding to `payload_type`
 // according to `packetization`.
 void UpdateVideoCodecPacketization(MediaContentDescription* desc,
-                                   int payload_type,
+                                   PayloadType payload_type,
                                    absl::string_view packetization) {
   if (packetization != kPacketizationParamRaw) {
     // Ignore unsupported packetization attribute.
@@ -2389,11 +2391,11 @@ bool ParsePacketizationAttribute(absl::string_view line,
                 &payload_type_string, error)) {
     return false;
   }
-  int payload_type;
-  if (!GetPayloadTypeFromString(line, payload_type_string, &payload_type,
-                                error)) {
+  int pt_int;
+  if (!GetPayloadTypeFromString(line, payload_type_string, &pt_int, error)) {
     return false;
   }
+  PayloadType payload_type = PayloadType(pt_int);
   absl::string_view packetization = packetization_fields[1];
   UpdateVideoCodecPacketization(media_desc, payload_type, packetization);
   return true;
@@ -2416,12 +2418,13 @@ bool ParseRtcpFbAttribute(absl::string_view line,
                 error)) {
     return false;
   }
-  int payload_type = kWildcardPayloadType;
+  PayloadType payload_type = kWildcardPayloadType;
   if (payload_type_string != "*") {
-    if (!GetPayloadTypeFromString(line, payload_type_string, &payload_type,
-                                  error)) {
+    int pt_int;
+    if (!GetPayloadTypeFromString(line, payload_type_string, &pt_int, error)) {
       return false;
     }
+    payload_type = PayloadType(pt_int);
   }
   absl::string_view id = rtcp_fb_fields[1];
   std::string param = "";
diff --git a/call/BUILD.gn b/call/BUILD.gn
index 232c66aef8..b11616353c 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -530,6 +530,7 @@ if (rtc_include_tests) {
         "../api:mock_audio_mixer",
         "../api:mock_frame_transformer",
         "../api:mock_video_codec_factory",
+        "../api:payload_type",
         "../api:rtp_headers",
         "../api:rtp_parameters",
         "../api:scoped_refptr",
diff --git a/call/payload_type_picker_unittest.cc b/call/payload_type_picker_unittest.cc
index 46e0de7ef1..562bb2cb5e 100644
--- a/call/payload_type_picker_unittest.cc
+++ b/call/payload_type_picker_unittest.cc
@@ -13,6 +13,7 @@
 #include <string>
 
 #include "absl/strings/str_cat.h"
+#include "api/payload_type.h"
 #include "api/video_codecs/sdp_video_format.h"
 #include "call/payload_type.h"
 #include "media/base/codec.h"
@@ -96,7 +97,6 @@ TEST(PayloadTypePicker, RollbackAndCommit) {
   PayloadTypeRecorder recorder(picker);
   const PayloadType a_payload_type(123);
   const PayloadType b_payload_type(124);
-  const PayloadType not_a_payload_type(44);
 
   Codec a_codec = CreateVideoCodec(0, "vp8");
 
diff --git a/media/BUILD.gn b/media/BUILD.gn
index e292762430..15c2cf92dc 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -356,6 +356,7 @@ rtc_source_set("media_channel") {
     "../rtc_base:stringutils",
     "../rtc_base/network:sent_packet",
     "../video/config:encoder_config",
+    "//third_party/abseil-cpp/absl/container:flat_hash_map",
     "//third_party/abseil-cpp/absl/functional:any_invocable",
     "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/strings:string_view",
@@ -374,6 +375,7 @@ rtc_library("codec") {
   ]
   deps = [
     ":media_constants",
+    "../api:payload_type",
     "../api:rtp_parameters",
     "../api/audio_codecs:audio_codecs_api",
     "../api/video_codecs:scalability_mode",
@@ -400,6 +402,7 @@ rtc_library("codec_list") {
   deps = [
     ":codec",
     ":media_constants",
+    "../api:payload_type",
     "../api:rtc_error",
     "../rtc_base:checks",
     "../rtc_base:logging",
@@ -970,6 +973,7 @@ if (rtc_include_tests) {
         "../api:mock_video_codec_factory",
         "../api:mock_video_decoder",
         "../api:mock_video_encoder",
+        "../api:payload_type",
         "../api:priority",
         "../api:ref_count",
         "../api:rtc_error",
diff --git a/media/base/codec.cc b/media/base/codec.cc
index 8379f9996b..535fcd80a9 100644
--- a/media/base/codec.cc
+++ b/media/base/codec.cc
@@ -22,6 +22,7 @@
 #include "absl/strings/str_cat.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/media_types.h"
+#include "api/payload_type.h"
 #include "api/rtp_parameters.h"
 #include "api/video_codecs/h264_profile_level_id.h"
 #include "api/video_codecs/sdp_video_format.h"
@@ -97,10 +98,10 @@ bool FeedbackParams::HasDuplicateEntries() const {
   return false;
 }
 
-Codec::Codec(Type type, int id, const std::string& name, int clockrate)
+Codec::Codec(Type type, PayloadType id, const std::string& name, int clockrate)
     : Codec(type, id, name, clockrate, 0) {}
 Codec::Codec(Type type,
-             int id,
+             PayloadType id,
              const std::string& name,
              int clockrate,
              size_t channels)
@@ -115,18 +116,22 @@ Codec::Codec(Type type,
 
 Codec::Codec(Type type)
     : Codec(type,
-            kIdNotSet,
+            PayloadType::NotSet(),
             "",
             type == Type::kVideo ? kDefaultVideoClockRateHz
                                  : kDefaultAudioClockRateHz) {}
 
 Codec::Codec(const SdpAudioFormat& c)
-    : Codec(Type::kAudio, kIdNotSet, c.name, c.clockrate_hz, c.num_channels) {
+    : Codec(Type::kAudio,
+            PayloadType::NotSet(),
+            c.name,
+            c.clockrate_hz,
+            c.num_channels) {
   params = c.parameters;
 }
 
 Codec::Codec(const SdpVideoFormat& c)
-    : Codec(Type::kVideo, kIdNotSet, c.name, kVideoCodecClockrate) {
+    : Codec(Type::kVideo, PayloadType::NotSet(), c.name, kVideoCodecClockrate) {
   params = c.parameters;
   scalability_modes = c.scalability_modes;
 }
@@ -290,20 +295,25 @@ std::string Codec::ToString() const {
   return sb.str();
 }
 
-Codec CreateAudioRtxCodec(int rtx_payload_type, int associated_payload_type) {
+Codec CreateAudioRtxCodec(PayloadType rtx_payload_type,
+                          PayloadType associated_payload_type) {
   Codec rtx_codec = CreateAudioCodec(rtx_payload_type, kRtxCodecName,
                                      kDefaultAudioClockRateHz, 1);
-  rtx_codec.SetParam(kCodecParamAssociatedPayloadType, associated_payload_type);
+  rtx_codec.SetParam(kCodecParamAssociatedPayloadType,
+                     associated_payload_type.value());
   return rtx_codec;
 }
 
-Codec CreateVideoRtxCodec(int rtx_payload_type, int associated_payload_type) {
+Codec CreateVideoRtxCodec(PayloadType rtx_payload_type,
+                          PayloadType associated_payload_type) {
   Codec rtx_codec = CreateVideoCodec(rtx_payload_type, kRtxCodecName);
-  rtx_codec.SetParam(kCodecParamAssociatedPayloadType, associated_payload_type);
+  rtx_codec.SetParam(kCodecParamAssociatedPayloadType,
+                     associated_payload_type.value());
   return rtx_codec;
 }
 
-const Codec* FindCodecById(const std::vector<Codec>& codecs, int payload_type) {
+const Codec* FindCodecById(const std::vector<Codec>& codecs,
+                           PayloadType payload_type) {
   for (const auto& codec : codecs) {
     if (codec.id == payload_type)
       return &codec;
@@ -395,7 +405,7 @@ void AddH264ConstrainedBaselineProfileToSupportedFormats(
   }
 }
 
-Codec CreateAudioCodec(int id,
+Codec CreateAudioCodec(PayloadType id,
                        const std::string& name,
                        int clockrate,
                        size_t channels) {
@@ -407,10 +417,10 @@ Codec CreateAudioCodec(const SdpAudioFormat& c) {
 }
 
 Codec CreateVideoCodec(const std::string& name) {
-  return CreateVideoCodec(Codec::kIdNotSet, name);
+  return CreateVideoCodec(PayloadType::NotSet(), name);
 }
 
-Codec CreateVideoCodec(int id, const std::string& name) {
+Codec CreateVideoCodec(PayloadType id, const std::string& name) {
   Codec c(Codec::Type::kVideo, id, name, kVideoCodecClockrate);
   if (absl::EqualsIgnoreCase(kH264CodecName, name)) {
     // This default is set for all H.264 codecs created because
@@ -426,7 +436,7 @@ Codec CreateVideoCodec(const SdpVideoFormat& c) {
   return Codec(c);
 }
 
-Codec CreateVideoCodec(int id, const SdpVideoFormat& sdp) {
+Codec CreateVideoCodec(PayloadType id, const SdpVideoFormat& sdp) {
   Codec c = CreateVideoCodec(sdp);
   c.id = id;
   return c;
diff --git a/media/base/codec.h b/media/base/codec.h
index bc970c5a6b..5f197441a9 100644
--- a/media/base/codec.h
+++ b/media/base/codec.h
@@ -20,6 +20,7 @@
 #include "absl/strings/str_format.h"
 #include "absl/strings/string_view.h"
 #include "api/audio_codecs/audio_format.h"
+#include "api/payload_type.h"
 #include "api/rtp_parameters.h"
 #include "api/video_codecs/scalability_mode.h"
 #include "api/video_codecs/sdp_video_format.h"
@@ -87,7 +88,7 @@ struct RTC_EXPORT Codec {
   static const int kIdNotSet = -1;
 
   Type type;
-  int id;
+  PayloadType id;
   std::string name;
   int clockrate;
 
@@ -173,7 +174,11 @@ struct RTC_EXPORT Codec {
   std::string ToString() const;
 
   // Default constructor, for initialization.
-  Codec() : Codec(Type::kAudio, kIdNotSet, "", kDefaultAudioClockRateHz) {}
+  Codec()
+      : Codec(Type::kAudio,
+              PayloadType::NotSet(),
+              "",
+              kDefaultAudioClockRateHz) {}
   Codec& operator=(const Codec& c);
   Codec& operator=(Codec&& c);
 
@@ -183,7 +188,7 @@ struct RTC_EXPORT Codec {
 
   template <typename Sink>
   friend void AbslStringify(Sink& sink, const Codec& c) {
-    absl::Format(&sink, "[%d:", c.id);
+    absl::Format(&sink, "[%v:", c.id);
     switch (c.type) {
       case Codec::Type::kAudio:
         sink.Append("audio/");
@@ -208,9 +213,9 @@ struct RTC_EXPORT Codec {
   // Creates an empty codec.
   explicit Codec(Type type);
   // Creates a codec with the given parameters.
-  Codec(Type type, int id, const std::string& name, int clockrate);
+  Codec(Type type, PayloadType id, const std::string& name, int clockrate);
   Codec(Type type,
-        int id,
+        PayloadType id,
         const std::string& name,
         int clockrate,
         size_t channels);
@@ -218,36 +223,38 @@ struct RTC_EXPORT Codec {
   explicit Codec(const SdpAudioFormat& c);
   explicit Codec(const SdpVideoFormat& c);
 
-  friend Codec CreateAudioCodec(int id,
+  friend Codec CreateAudioCodec(PayloadType id,
                                 const std::string& name,
                                 int clockrate,
                                 size_t channels);
   friend Codec CreateAudioCodec(const SdpAudioFormat& c);
-  friend Codec CreateAudioRtxCodec(int rtx_payload_type,
-                                   int associated_payload_type);
-  friend Codec CreateVideoCodec(int id, const std::string& name);
+  friend Codec CreateAudioRtxCodec(PayloadType rtx_payload_type,
+                                   PayloadType associated_payload_type);
+  friend Codec CreateVideoCodec(PayloadType id, const std::string& name);
   friend Codec CreateVideoCodec(const SdpVideoFormat& c);
-  friend Codec CreateVideoRtxCodec(int rtx_payload_type,
-                                   int associated_payload_type);
+  friend Codec CreateVideoCodec(PayloadType id, const SdpVideoFormat& sdp);
 };
 
 using Codecs = std::vector<Codec>;
 
-Codec CreateAudioCodec(int id,
+Codec CreateAudioCodec(PayloadType id,
                        const std::string& name,
                        int clockrate,
                        size_t channels);
 Codec CreateAudioCodec(const SdpAudioFormat& c);
-Codec CreateAudioRtxCodec(int rtx_payload_type, int associated_payload_type);
+Codec CreateAudioRtxCodec(PayloadType rtx_payload_type,
+                          PayloadType associated_payload_type);
 Codec CreateVideoCodec(const std::string& name);
-Codec CreateVideoCodec(int id, const std::string& name);
+Codec CreateVideoCodec(PayloadType id, const std::string& name);
 Codec CreateVideoCodec(const SdpVideoFormat& c);
-Codec CreateVideoCodec(int id, const SdpVideoFormat& sdp);
-Codec CreateVideoRtxCodec(int rtx_payload_type, int associated_payload_type);
+Codec CreateVideoCodec(PayloadType id, const SdpVideoFormat& sdp);
+Codec CreateVideoRtxCodec(PayloadType rtx_payload_type,
+                          PayloadType associated_payload_type);
 
 // Get the codec setting associated with `payload_type`. If there
 // is no codec associated with that payload type it returns nullptr.
-const Codec* FindCodecById(const std::vector<Codec>& codecs, int payload_type);
+const Codec* FindCodecById(const std::vector<Codec>& codecs,
+                           PayloadType payload_type);
 
 bool HasLntf(const Codec& codec);
 bool HasNack(const Codec& codec);
diff --git a/media/base/codec_comparators.cc b/media/base/codec_comparators.cc
index 942e3a2973..3b4b198ad0 100644
--- a/media/base/codec_comparators.cc
+++ b/media/base/codec_comparators.cc
@@ -11,6 +11,7 @@
 
 #include <algorithm>
 #include <cstddef>
+#include <cstdint>
 #include <optional>
 #include <string>
 #include <vector>
@@ -20,6 +21,7 @@
 #include "absl/strings/match.h"
 #include "absl/strings/string_view.h"
 #include "api/media_types.h"
+#include "api/payload_type.h"
 #include "api/rtp_parameters.h"
 #include "api/video_codecs/av1_profile.h"
 #include "api/video_codecs/h264_profile_level_id.h"
@@ -119,9 +121,9 @@ bool IsSameCodecSpecific(const std::string& name1,
 }
 
 bool ReferencedCodecsMatch(const std::vector<Codec>& codecs1,
-                           const int codec1_id,
+                           const PayloadType codec1_id,
                            const std::vector<Codec>& codecs2,
-                           const int codec2_id) {
+                           const PayloadType codec2_id) {
   const Codec* codec1 = FindCodecById(codecs1, codec1_id);
   const Codec* codec2 = FindCodecById(codecs2, codec2_id);
   return codec1 != nullptr && codec2 != nullptr && codec1->Matches(*codec2);
@@ -130,21 +132,25 @@ bool ReferencedCodecsMatch(const std::vector<Codec>& codecs1,
 bool MatchesWithReferenceAttributesAndComparator(
     const Codec& codec_to_match,
     const Codec& potential_match,
-    absl::AnyInvocable<bool(int, int)> reference_comparator) {
+    absl::AnyInvocable<bool(PayloadType, PayloadType)> reference_comparator) {
   if (!MatchesWithCodecRules(codec_to_match, potential_match)) {
     return false;
   }
   Codec::ResiliencyType resiliency_type = codec_to_match.GetResiliencyType();
   if (resiliency_type == Codec::ResiliencyType::kRtx) {
-    int apt_value_1 = 0;
-    int apt_value_2 = 0;
+    int apt_value_1_int = 0;
+    int apt_value_2_int = 0;
     if (!codec_to_match.GetParam(kCodecParamAssociatedPayloadType,
-                                 &apt_value_1) ||
+                                 &apt_value_1_int) ||
         !potential_match.GetParam(kCodecParamAssociatedPayloadType,
-                                  &apt_value_2)) {
+                                  &apt_value_2_int)) {
       RTC_LOG(LS_WARNING) << "RTX missing associated payload type.";
       return false;
     }
+    PayloadType apt_value_1 =
+        PayloadType(static_cast<uint8_t>(apt_value_1_int));
+    PayloadType apt_value_2 =
+        PayloadType(static_cast<uint8_t>(apt_value_2_int));
     if (reference_comparator(apt_value_1, apt_value_2)) {
       return true;
     }
@@ -196,8 +202,8 @@ bool MatchesWithReferenceAttributesAndComparator(
     // codecs has an unassigned payload type or they have the same ID.
     if (codec_to_match.type == Codec::Type::kAudio &&
         codec_to_match.name == kRedCodecName &&
-        (codec_to_match.id == Codec::kIdNotSet ||
-         potential_match.id == Codec::kIdNotSet ||
+        (codec_to_match.id == PayloadType::NotSet() ||
+         potential_match.id == PayloadType::NotSet() ||
          codec_to_match.id == potential_match.id)) {
       return true;
     }
@@ -297,7 +303,8 @@ bool MatchesWithCodecRules(const Codec& left_codec, const Codec& right_codec) {
 
   bool matches_id;
   if ((is_id_in_dynamic_range && is_codec_id_in_dynamic_range) ||
-      left_codec.id == Codec::kIdNotSet || right_codec.id == Codec::kIdNotSet) {
+      left_codec.id == PayloadType::NotSet() ||
+      right_codec.id == PayloadType::NotSet()) {
     matches_id = absl::EqualsIgnoreCase(left_codec.name, right_codec.name);
   } else {
     matches_id = (left_codec.id == right_codec.id);
@@ -330,7 +337,7 @@ bool MatchesWithCodecRules(const Codec& left_codec, const Codec& right_codec) {
 
 bool MatchesWithReferenceAttributes(const Codec& codec1, const Codec& codec2) {
   return MatchesWithReferenceAttributesAndComparator(
-      codec1, codec2, [](int a, int b) { return a == b; });
+      codec1, codec2, [](PayloadType a, PayloadType b) { return a == b; });
 }
 
 // Finds a codec in `codecs2` that matches `codec_to_match`, which is
@@ -348,7 +355,7 @@ std::optional<Codec> FindMatchingCodec(const std::vector<Codec>& codecs1,
   for (const Codec& potential_match : codecs2) {
     if (MatchesWithReferenceAttributesAndComparator(
             codec_to_match, potential_match,
-            [&codecs1, &codecs2](int a, int b) {
+            [&codecs1, &codecs2](PayloadType a, PayloadType b) {
               return ReferencedCodecsMatch(codecs1, a, codecs2, b);
             })) {
       return potential_match;
diff --git a/media/base/codec_list.cc b/media/base/codec_list.cc
index 4be60c4776..a293244b11 100644
--- a/media/base/codec_list.cc
+++ b/media/base/codec_list.cc
@@ -14,6 +14,7 @@
 #include <map>
 #include <vector>
 
+#include "api/payload_type.h"
 #include "api/rtc_error.h"
 #include "media/base/codec.h"
 #include "media/base/media_constants.h"
@@ -32,8 +33,8 @@ RTCError CheckInputConsistency(const std::vector<Codec>& codecs) {
   // that there are no duplicates.
   for (size_t i = 0; i < codecs.size(); i++) {
     const Codec& codec = codecs[i];
-    if (codec.id != Codec::kIdNotSet) {
-      auto [it, success] = pt_to_index.insert({codec.id, i});
+    if (codec.id != PayloadType::NotSet()) {
+      auto [it, success] = pt_to_index.insert({codec.id, static_cast<int>(i)});
       if (!success) {
         RTC_LOG(LS_ERROR) << "Duplicate payload type in codec list, " << codec
                           << " and " << codecs[it->second]
@@ -56,7 +57,7 @@ RTCError CheckInputConsistency(const std::vector<Codec>& codecs) {
         // TODO: https://issues.webrtc.org/384756622 - reject codec earlier and
         // enable check. RTC_DCHECK(apt_it != codec.params.end()); Until that is
         // fixed:
-        if (codec.id == Codec::kIdNotSet) {
+        if (codec.id == PayloadType::NotSet()) {
           // Should not have an apt parameter.
           if (apt_it != codec.params.end()) {
             RTC_LOG(LS_WARNING) << "Surprising condition: RTX codec without "
@@ -77,7 +78,7 @@ RTCError CheckInputConsistency(const std::vector<Codec>& codecs) {
           return LOG_ERROR(RTCError(RTCErrorType::INVALID_PARAMETER)
                            << "Non-numeric argument to rtx apt parameter");
         }
-        if (codec.id != Codec::kIdNotSet &&
+        if (codec.id != PayloadType::NotSet() &&
             pt_to_index.count(associated_pt) != 1) {
           RTC_LOG(LS_WARNING)
               << "Surprising condition: RTX codec APT not found: " << codec
diff --git a/media/base/media_channel.h b/media/base/media_channel.h
index cdc19280cc..70756943ff 100644
--- a/media/base/media_channel.h
+++ b/media/base/media_channel.h
@@ -22,6 +22,7 @@
 #include <utility>
 #include <vector>
 
+#include "absl/container/flat_hash_map.h"
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/str_cat.h"
 #include "absl/strings/string_view.h"
@@ -705,7 +706,8 @@ struct BandwidthEstimationInfo {
 };
 
 // Maps from payload type to `RtpCodecParameters`.
-typedef std::map<int, RtpCodecParameters> RtpCodecParametersMap;
+typedef absl::flat_hash_map<decltype(Codec::id), RtpCodecParameters>
+    RtpCodecParametersMap;
 
 // Stats returned from VoiceMediaSendChannel.GetStats()
 struct VoiceMediaSendInfo {
diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc
index 91f2c9b333..6b3e0b506b 100644
--- a/media/engine/webrtc_video_engine.cc
+++ b/media/engine/webrtc_video_engine.cc
@@ -1715,13 +1715,13 @@ void WebRtcVideoSendChannel::FillSendCodecStats(
   // we can omit the codec information for those here and only insert the
   // primary codec that is being used to send here.
   video_media_info->send_codecs.insert(std::make_pair(
-      send_codec()->codec.id, send_codec()->codec.ToCodecParameters()));
+      send_codec()->codec.id.value(), send_codec()->codec.ToCodecParameters()));
 
   for (const auto& it : send_codecs_) {
     auto codec_param_it = video_media_info->send_codecs.find(it.codec.id);
     if (codec_param_it == video_media_info->send_codecs.end()) {
       video_media_info->send_codecs.insert(
-          std::make_pair(it.codec.id, it.codec.ToCodecParameters()));
+          std::make_pair(it.codec.id.value(), it.codec.ToCodecParameters()));
     }
   }
 }
@@ -2472,7 +2472,8 @@ WebRtcVideoSendChannel::WebRtcVideoSendStream::GetPerLayerVideoSenderInfos(
   VideoSenderInfo common_info;
   if (parameters_.codec_settings) {
     common_info.codec_name = parameters_.codec_settings->codec.name;
-    common_info.codec_payload_type = parameters_.codec_settings->codec.id;
+    common_info.codec_payload_type =
+        parameters_.codec_settings->codec.id.value();
   }
   // If SVC is used, one stream is configured but multiple encodings exist. This
   // is not spec-compliant, but it is how we've implemented SVC so this affects
@@ -3223,7 +3224,7 @@ void WebRtcVideoReceiveChannel::FillReceiveCodecStats(
                  *receiver.codec_payload_type == c.id;
         });
     if (codec != recv_params_.codecs.end()) {
-      video_media_info->receive_codecs.insert(
+      video_media_info->receive_codecs.emplace(
           std::make_pair(codec->id, codec->ToCodecParameters()));
     }
   }
diff --git a/media/engine/webrtc_video_engine_unittest.cc b/media/engine/webrtc_video_engine_unittest.cc
index 368554440c..c9de2e55a8 100644
--- a/media/engine/webrtc_video_engine_unittest.cc
+++ b/media/engine/webrtc_video_engine_unittest.cc
@@ -36,6 +36,7 @@
 #include "api/field_trials.h"
 #include "api/make_ref_counted.h"
 #include "api/media_types.h"
+#include "api/payload_type.h"
 #include "api/priority.h"
 #include "api/rtc_error.h"
 #include "api/rtp_headers.h"
@@ -5626,7 +5627,7 @@ TEST_F(WebRtcVideoChannelTest,
   VideoReceiverParameters parameters;
   parameters.codecs.push_back(GetEngineCodec("VP8"));
   parameters.codecs.push_back(GetEngineCodec("VP8"));
-  parameters.codecs[1].id += 1;
+  parameters.codecs[1].id = PayloadType(parameters.codecs[1].id + 1);
   EXPECT_TRUE(receive_channel_->SetReceiverParameters(parameters));
 }
 
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index c8bda9884f..a3c8c98089 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -50,6 +50,7 @@
 #include "api/frame_transformer_interface.h"
 #include "api/make_ref_counted.h"
 #include "api/media_types.h"
+#include "api/payload_type.h"
 #include "api/priority.h"
 #include "api/rtc_error.h"
 #include "api/rtp_headers.h"
@@ -1431,7 +1432,7 @@ bool WebRtcVoiceSendChannel::SetSendCodecs(
     const std::vector<Codec>& codecs,
     std::optional<Codec> preferred_codec) {
   RTC_DCHECK_RUN_ON(worker_thread_);
-  dtmf_payload_type_ = std::nullopt;
+  dtmf_payload_type_ = PayloadType::NotSet();
   dtmf_payload_freq_ = -1;
 
   // Validate supplied codecs list.
@@ -1453,7 +1454,8 @@ bool WebRtcVoiceSendChannel::SetSendCodecs(
   for (const Codec& codec : codecs) {
     if (IsCodec(codec, kDtmfCodecName)) {
       dtmf_codecs.push_back(codec);
-      if (!dtmf_payload_type_ || codec.clockrate < dtmf_payload_freq_) {
+      if ((dtmf_payload_type_ == PayloadType::NotSet()) ||
+          codec.clockrate < dtmf_payload_freq_) {
         dtmf_payload_type_ = codec.id;
         dtmf_payload_freq_ = codec.clockrate;
       }
@@ -1513,7 +1515,7 @@ bool WebRtcVoiceSendChannel::SetSendCodecs(
           RTC_LOG(LS_WARNING)
               << "CN frequency " << cn_codec.clockrate << " not supported.";
         } else {
-          send_codec_spec->cng_payload_type = cn_codec.id;
+          send_codec_spec->cng_payload_type = cn_codec.id.value();
         }
         break;
       }
@@ -1537,7 +1539,7 @@ bool WebRtcVoiceSendChannel::SetSendCodecs(
     const Codec& red_codec = codecs[i];
     if (IsCodec(red_codec, kRedCodecName) &&
         CheckRedParameters(red_codec, *send_codec_spec)) {
-      send_codec_spec->red_payload_type = red_codec.id;
+      send_codec_spec->red_payload_type = red_codec.id.value();
       break;
     }
   }
@@ -1698,7 +1700,7 @@ bool WebRtcVoiceSendChannel::SetLocalSource(uint32_t ssrc,
 
 bool WebRtcVoiceSendChannel::CanInsertDtmf() {
   RTC_DCHECK_RUN_ON(worker_thread_);
-  return dtmf_payload_type_.has_value() && send_;
+  return (dtmf_payload_type_ != PayloadType::NotSet()) && send_;
 }
 
 void WebRtcVoiceSendChannel::SetFrameEncryptor(
@@ -1884,7 +1886,7 @@ void WebRtcVoiceSendChannel::FillSendCodecStats(
     });
     if (codec != send_codecs_.end()) {
       voice_media_info->send_codecs.insert(
-          std::make_pair(codec->id, codec->ToCodecParameters()));
+          std::make_pair(codec->id.value(), codec->ToCodecParameters()));
     }
   }
 }
@@ -2772,7 +2774,7 @@ void WebRtcVoiceReceiveChannel::FillReceiveCodecStats(
     });
     if (codec != recv_codecs_.end()) {
       voice_media_info->receive_codecs.insert(
-          std::make_pair(codec->id, codec->ToCodecParameters()));
+          std::make_pair(codec->id.value(), codec->ToCodecParameters()));
     }
   }
 }
diff --git a/media/engine/webrtc_voice_engine.h b/media/engine/webrtc_voice_engine.h
index 97907fd4f9..c38a47f6aa 100644
--- a/media/engine/webrtc_voice_engine.h
+++ b/media/engine/webrtc_voice_engine.h
@@ -39,6 +39,7 @@
 #include "api/field_trials_view.h"
 #include "api/frame_transformer_interface.h"
 #include "api/media_types.h"
+#include "api/payload_type.h"
 #include "api/rtc_error.h"
 #include "api/rtp_headers.h"
 #include "api/rtp_parameters.h"
@@ -280,7 +281,7 @@ class WebRtcVoiceSendChannel final : public MediaChannelUtil,
 
   int max_send_bitrate_bps_ RTC_GUARDED_BY(worker_thread_) = 0;
   AudioOptions options_ RTC_GUARDED_BY(worker_thread_);
-  std::optional<int> dtmf_payload_type_ RTC_GUARDED_BY(worker_thread_);
+  PayloadType dtmf_payload_type_ RTC_GUARDED_BY(worker_thread_);
   int dtmf_payload_freq_ RTC_GUARDED_BY(worker_thread_) = -1;
   bool enable_non_sender_rtt_ RTC_GUARDED_BY(worker_thread_) = false;
   bool send_ RTC_GUARDED_BY(worker_thread_) = false;
diff --git a/media/engine/webrtc_voice_engine_unittest.cc b/media/engine/webrtc_voice_engine_unittest.cc
index 8195403501..526a6cbf25 100644
--- a/media/engine/webrtc_voice_engine_unittest.cc
+++ b/media/engine/webrtc_voice_engine_unittest.cc
@@ -38,6 +38,7 @@
 #include "api/field_trials.h"
 #include "api/make_ref_counted.h"
 #include "api/media_types.h"
+#include "api/payload_type.h"
 #include "api/priority.h"
 #include "api/ref_count.h"
 #include "api/rtc_error.h"
@@ -1110,7 +1111,7 @@ TEST_P(WebRtcVoiceEngineTestFake, ChangeRecvCodecPayloadType) {
   parameters.codecs.push_back(kOpusCodec);
   EXPECT_TRUE(receive_channel_->SetReceiverParameters(parameters));
 
-  ++parameters.codecs[0].id;
+  parameters.codecs[0].id = PayloadType(parameters.codecs[0].id + 1);
   EXPECT_TRUE(receive_channel_->SetReceiverParameters(parameters));
 }
 
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 1f19fdc878..a903e359d8 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -105,6 +105,7 @@ rtc_library("channel") {
     "//third_party/abseil-cpp/absl/algorithm:container",
     "//third_party/abseil-cpp/absl/cleanup",
     "//third_party/abseil-cpp/absl/functional:any_invocable",
+    "//third_party/abseil-cpp/absl/strings:str_format",
     "//third_party/abseil-cpp/absl/strings:string_view",
   ]
 }
@@ -1228,6 +1229,7 @@ rtc_library("sdp_payload_type_suggester") {
     ":jsep_transport_collection",
     ":session_description",
     "../api:jsep",
+    "../api:payload_type",
     "../api:peer_connection_interface",
     "../api:rtc_error",
     "../call:payload_type",
@@ -1235,6 +1237,7 @@ rtc_library("sdp_payload_type_suggester") {
     "../media:codec",
     "../rtc_base:checks",
     "../rtc_base:event_tracer",
+    "../rtc_base:logging",
     "../rtc_base:threading",
     "//third_party/abseil-cpp/absl/strings:string_view",
   ]
@@ -1390,6 +1393,7 @@ rtc_library("simulcast_sdp_serializer") {
   deps = [
     ":session_description",
     ":simulcast_description",
+    "../api:payload_type",
     "../api:rtc_error",
     "../api:rtp_parameters",
     "../media:codec",
@@ -2268,6 +2272,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "../api:make_ref_counted",
       "../api:media_stream_interface",
       "../api:mock_datagram_connection_observer",
+      "../api:payload_type",
       "../api:peer_connection_interface",
       "../api:priority",
       "../api:rtc_error",
@@ -2625,6 +2630,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "../api:mock_encoder_selector",
       "../api:mock_packet_socket_factory",
       "../api:mock_video_track",
+      "../api:payload_type",
       "../api:peer_connection_interface",
       "../api:priority",
       "../api:ref_count",
diff --git a/pc/channel.cc b/pc/channel.cc
index 1774778eec..610f507b2d 100644
--- a/pc/channel.cc
+++ b/pc/channel.cc
@@ -21,6 +21,7 @@
 #include "absl/algorithm/container.h"
 #include "absl/cleanup/cleanup.h"
 #include "absl/functional/any_invocable.h"
+#include "absl/strings/str_format.h"
 #include "absl/strings/string_view.h"
 #include "api/crypto/crypto_options.h"
 #include "api/jsep.h"
@@ -156,9 +157,9 @@ RTCError MaybeIgnorePacketization(const MediaChannelParameters& new_params,
       // Note: this writes into old_params
       codec.packetization = std::nullopt;
     } else if (!has_matching_packetization) {
-      std::string error_desc = StringFormat(
+      std::string error_desc = absl::StrFormat(
           "Failed to set local answer due to incompatible codec "
-          "packetization for pt='%d' specified.",
+          "packetization for pt='%v' specified.",
           codec.id);
       return RTCError(RTCErrorType::INTERNAL_ERROR, error_desc);
     }
diff --git a/pc/codec_vendor.cc b/pc/codec_vendor.cc
index 2adc3a62e0..cb9f65cecd 100644
--- a/pc/codec_vendor.cc
+++ b/pc/codec_vendor.cc
@@ -563,7 +563,7 @@ RTCError AssignCodecIdsAndLinkRed(PayloadTypeSuggester* pt_suggester,
   RTC_DCHECK_DISALLOW_THREAD_BLOCKING_CALLS();
   int codec_payload_type = Codec::kIdNotSet;
   for (Codec& codec : codecs) {
-    if (codec.id == Codec::kIdNotSet) {
+    if (codec.id == PayloadType::NotSet()) {
       // Add payload types to codecs, if needed
       // This should only happen if WebRTC-PayloadTypesInTransport field trial
       // is enabled.
@@ -577,7 +577,7 @@ RTCError AssignCodecIdsAndLinkRed(PayloadTypeSuggester* pt_suggester,
     // record first Opus codec id
     if (absl::EqualsIgnoreCase(codec.name, kOpusCodecName) &&
         codec_payload_type == Codec::kIdNotSet) {
-      codec_payload_type = codec.id;
+      codec_payload_type = codec.id.value();
     }
   }
   if (codec_payload_type != Codec::kIdNotSet) {
diff --git a/pc/codec_vendor_unittest.cc b/pc/codec_vendor_unittest.cc
index d8f88f6f58..76b4cbe676 100644
--- a/pc/codec_vendor_unittest.cc
+++ b/pc/codec_vendor_unittest.cc
@@ -18,6 +18,7 @@
 #include "api/environment/environment_factory.h"
 #include "api/field_trials.h"
 #include "api/media_types.h"
+#include "api/payload_type.h"
 #include "api/rtc_error.h"
 #include "api/rtp_transceiver_direction.h"
 #include "api/test/rtc_error_matchers.h"
@@ -194,7 +195,7 @@ TEST(CodecVendorTest, PreferencesAffectCodecChoice) {
   MediaDescriptionOptions options(MediaType::VIDEO, "mid",
                                   RtpTransceiverDirection::kSendOnly, false);
   options.codec_preferences = {
-      ToRtpCodecCapability(CreateVideoCodec(-1, "vp9")),
+      ToRtpCodecCapability(CreateVideoCodec(PayloadType::NotSet(), "vp9")),
   };
   FakePayloadTypeSuggester pt_suggester;
 
diff --git a/pc/rtp_parameters_conversion.cc b/pc/rtp_parameters_conversion.cc
index 63d50d23cc..ca19b217b8 100644
--- a/pc/rtp_parameters_conversion.cc
+++ b/pc/rtp_parameters_conversion.cc
@@ -81,7 +81,7 @@ RtpCodecCapability ToRtpCodecCapability(const Codec& cricket_codec) {
   codec.kind = cricket_codec.type == Codec::Type::kAudio ? MediaType::AUDIO
                                                          : MediaType::VIDEO;
   codec.clock_rate.emplace(cricket_codec.clockrate);
-  codec.preferred_payload_type.emplace(cricket_codec.id);
+  codec.preferred_payload_type.emplace(cricket_codec.id.value());
   for (const FeedbackParam& cricket_feedback :
        cricket_codec.feedback_params.params()) {
     std::optional<RtcpFeedback> feedback = ToRtcpFeedback(cricket_feedback);
diff --git a/pc/sdp_payload_type_suggester.cc b/pc/sdp_payload_type_suggester.cc
index acaca73b83..23868649d1 100644
--- a/pc/sdp_payload_type_suggester.cc
+++ b/pc/sdp_payload_type_suggester.cc
@@ -16,12 +16,14 @@
 
 #include "absl/strings/string_view.h"
 #include "api/jsep.h"
+#include "api/payload_type.h"
 #include "api/rtc_error.h"
 #include "call/payload_type.h"
 #include "call/payload_type_picker.h"
 #include "media/base/codec.h"
 #include "pc/session_description.h"
 #include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
 #include "rtc_base/thread.h"
 #include "rtc_base/trace_event.h"
 
@@ -45,6 +47,8 @@ RTCErrorOr<PayloadType> SdpPayloadTypeSuggester::SuggestPayloadType(
   RTCErrorOr<PayloadType> remote_result =
       remote_recorder.LookupPayloadType(codec);
   if (remote_result.ok()) {
+    RTC_LOG(LS_INFO) << "SuggestPayloadType: Found remote mapping for " << codec
+                     << " to PT " << remote_result.value();
     RTCErrorOr<Codec> local_codec =
         local_recorder.LookupCodec(remote_result.value());
     if (!local_codec.ok()) {
@@ -55,6 +59,9 @@ RTCErrorOr<PayloadType> SdpPayloadTypeSuggester::SuggestPayloadType(
     }
     // If we get here, PT is already in use, possibly for something else.
     // Fall through to SuggestMapping.
+  } else {
+    RTC_LOG(LS_INFO) << "SuggestPayloadType: FAILED to find remote mapping for "
+                     << codec;
   }
   return payload_type_picker_.SuggestMapping(codec, &local_recorder);
 }
diff --git a/pc/sdp_payload_type_suggester_unittest.cc b/pc/sdp_payload_type_suggester_unittest.cc
index d543d1b1fb..1610822615 100644
--- a/pc/sdp_payload_type_suggester_unittest.cc
+++ b/pc/sdp_payload_type_suggester_unittest.cc
@@ -16,9 +16,9 @@
 
 #include "absl/strings/string_view.h"
 #include "api/jsep.h"
+#include "api/payload_type.h"
 #include "api/peer_connection_interface.h"
 #include "api/rtc_error.h"
-#include "call/payload_type.h"
 #include "media/base/codec.h"
 #include "media/base/media_constants.h"
 #include "pc/session_description.h"
@@ -48,7 +48,8 @@ class SdpPayloadTypeSuggesterTest : public testing::Test {
 };
 
 TEST_F(SdpPayloadTypeSuggesterTest, SuggestPayloadTypeBasic) {
-  Codec pcmu_codec = CreateAudioCodec(-1, kPcmuCodecName, 8000, 1);
+  Codec pcmu_codec =
+      CreateAudioCodec(PayloadType::NotSet(), kPcmuCodecName, 8000, 1);
   RTCErrorOr<PayloadType> pcmu_pt =
       suggester_.SuggestPayloadType("mid", pcmu_codec);
   ASSERT_TRUE(pcmu_pt.ok());
@@ -63,7 +64,8 @@ TEST_F(SdpPayloadTypeSuggesterTest, SuggestPayloadTypeReusesRemotePayloadType) {
   offer->contents()[0].media_description()->set_codecs({remote_lyra_codec});
   EXPECT_TRUE(
       suggester_.Update(offer.get(), /* local= */ false, SdpType::kOffer).ok());
-  Codec local_lyra_codec = CreateAudioCodec(-1, "lyra", 8000, 1);
+  Codec local_lyra_codec =
+      CreateAudioCodec(PayloadType::NotSet(), "lyra", 8000, 1);
   RTCErrorOr<PayloadType> lyra_pt =
       suggester_.SuggestPayloadType(kAudioMid1, local_lyra_codec);
   ASSERT_TRUE(lyra_pt.ok());
@@ -81,12 +83,14 @@ TEST_F(SdpPayloadTypeSuggesterTest,
   EXPECT_TRUE(
       suggester_.Update(offer.get(), /* local= */ false, SdpType::kOffer).ok());
   // Check that we get the Opus codec back with the remote PT
-  Codec local_opus_codec = CreateAudioCodec(-1, "opus", 48000, 2);
+  Codec local_opus_codec =
+      CreateAudioCodec(PayloadType::NotSet(), "opus", 48000, 2);
   RTCErrorOr<PayloadType> local_opus_pt =
       suggester_.SuggestPayloadType(kAudioMid1, local_opus_codec);
   EXPECT_EQ(local_opus_pt.value(), remote_opus_pt);
   // Check that we don't get 110 allocated for DTMF, since it's in use for opus
-  Codec local_other_codec = CreateAudioCodec(-1, kDtmfCodecName, 48000, 1);
+  Codec local_other_codec =
+      CreateAudioCodec(PayloadType::NotSet(), kDtmfCodecName, 48000, 1);
   RTCErrorOr<PayloadType> other_pt =
       suggester_.SuggestPayloadType(kAudioMid1, local_other_codec);
   ASSERT_TRUE(other_pt.ok());
diff --git a/pc/simulcast_sdp_serializer.cc b/pc/simulcast_sdp_serializer.cc
index 9f3539923d..68aea17af2 100644
--- a/pc/simulcast_sdp_serializer.cc
+++ b/pc/simulcast_sdp_serializer.cc
@@ -19,6 +19,7 @@
 
 #include "absl/algorithm/container.h"
 #include "absl/strings/string_view.h"
+#include "api/payload_type.h"
 #include "api/rtc_error.h"
 #include "api/rtp_parameters.h"
 #include "media/base/codec.h"
@@ -302,7 +303,7 @@ std::string SimulcastSdpSerializer::SerializeRidDescription(
     if (it == media_desc.codecs().end()) {
       break;
     }
-    if (it->id == Codec::kIdNotSet) {
+    if (it->id == PayloadType::NotSet()) {
       RTC_DCHECK_NOTREACHED();
       break;
     }
diff --git a/rtc_base/strong_alias.h b/rtc_base/strong_alias.h
index 548253de6c..8071a7e443 100644
--- a/rtc_base/strong_alias.h
+++ b/rtc_base/strong_alias.h
@@ -68,6 +68,13 @@ class StrongAlias {
 
  protected:
   UnderlyingType value_;
+
+ private:
+  // Helper function for using abseil hash types.
+  template <typename H>
+  friend H AbslHashValue(H h, const StrongAlias& st) {
+    return H::combine(std::move(h), st.value_);
+  }
 };
 
 }  // namespace webrtc

From 978f78b10334ee3d93808a50140e6af98a43ad6e Mon Sep 17 00:00:00 2001
From: Evan Shrubsole <eshr@webrtc.org>
Date: Fri, 27 Mar 2026 14:26:16 +0000
Subject: [PATCH 434/472] Support custom socket server in test::RunLoop

Puts this to use in stun_server_unittest.

Bug: webrtc:469327588
Change-Id: I66e904cac27c9fafe67c2061650f85996a6a6964
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/460880
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47287}
---
 p2p/test/stun_server_unittest.cc |  4 ++--
 test/BUILD.gn                    |  1 +
 test/run_loop.cc                 | 38 ++++++++++++++++++++++++++++----
 test/run_loop.h                  | 11 ++++++---
 4 files changed, 45 insertions(+), 9 deletions(-)

diff --git a/p2p/test/stun_server_unittest.cc b/p2p/test/stun_server_unittest.cc
index 95f5298b4d..8a63a149d9 100644
--- a/p2p/test/stun_server_unittest.cc
+++ b/p2p/test/stun_server_unittest.cc
@@ -20,10 +20,10 @@
 #include "rtc_base/byte_buffer.h"
 #include "rtc_base/socket_address.h"
 #include "rtc_base/test_client.h"
-#include "rtc_base/thread.h"
 #include "rtc_base/virtual_socket_server.h"
 #include "test/create_test_environment.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 
 namespace webrtc {
 
@@ -57,7 +57,7 @@ class StunServerTest : public ::testing::Test {
  private:
   const Environment env_ = CreateTestEnvironment();
   VirtualSocketServer ss_;
-  AutoSocketServerThread main_thread_{&ss_};
+  test::RunLoop run_loop_{&ss_};
   StunServer server_{AsyncUDPSocket::Create(env_, server_addr, ss_)};
   TestClient client_{AsyncUDPSocket::Create(env_, client_addr, ss_)};
 };
diff --git a/test/BUILD.gn b/test/BUILD.gn
index dbe5920024..5b1b5ef9ba 100644
--- a/test/BUILD.gn
+++ b/test/BUILD.gn
@@ -1098,6 +1098,7 @@ rtc_library("run_loop") {
     "../rtc_base:threading",
     "../rtc_base:timeutils",
     "../rtc_base:weak_ptr",
+    "//third_party/abseil-cpp/absl/base:nullability",
     "//third_party/abseil-cpp/absl/functional:any_invocable",
   ]
 }
diff --git a/test/run_loop.cc b/test/run_loop.cc
index 82d46bb08a..a412147e5d 100644
--- a/test/run_loop.cc
+++ b/test/run_loop.cc
@@ -9,6 +9,7 @@
  */
 #include "test/run_loop.h"
 
+#include "absl/base/nullability.h"
 #include "absl/functional/any_invocable.h"
 #include "api/task_queue/pending_task_safety_flag.h"
 #include "api/task_queue/task_queue_base.h"
@@ -21,7 +22,14 @@
 namespace webrtc {
 namespace test {
 
-RunLoop::RunLoop() : weak_factory_(this) {
+RunLoop::RunLoop() : worker_thread_(&socket_server_), weak_factory_(this) {
+  worker_thread_.WrapCurrent();
+}
+
+RunLoop::RunLoop(SocketServer* absl_nullable custom_ss)
+    : socket_server_(custom_ss),
+      worker_thread_(&socket_server_),
+      weak_factory_(this) {
   worker_thread_.WrapCurrent();
 }
 
@@ -69,28 +77,50 @@ void RunLoop::Flush() {
 }
 
 RunLoop::FakeSocketServer::FakeSocketServer() = default;
+
+RunLoop::FakeSocketServer::FakeSocketServer(
+    SocketServer* absl_nullable custom_ss)
+    : custom_ss_(custom_ss) {}
+
 RunLoop::FakeSocketServer::~FakeSocketServer() = default;
 
 void RunLoop::FakeSocketServer::FailNextWait() {
   fail_next_wait_ = true;
 }
 
+void RunLoop::FakeSocketServer::SetMessageQueue(Thread* absl_nullable queue) {
+  if (custom_ss_) {
+    custom_ss_->SetMessageQueue(queue);
+  }
+}
+
 bool RunLoop::FakeSocketServer::Wait(TimeDelta max_wait_duration,
                                      bool process_io) {
   if (fail_next_wait_) {
     fail_next_wait_ = false;
     return false;
   }
+  if (custom_ss_) {
+    return custom_ss_->Wait(max_wait_duration, process_io);
+  }
   return true;
 }
 
-void RunLoop::FakeSocketServer::WakeUp() {}
+void RunLoop::FakeSocketServer::WakeUp() {
+  if (custom_ss_) {
+    custom_ss_->WakeUp();
+  }
+}
 
-Socket* RunLoop::FakeSocketServer::CreateSocket(int family, int type) {
+Socket* absl_nullable RunLoop::FakeSocketServer::CreateSocket(int family,
+                                                              int type) {
+  if (custom_ss_) {
+    return custom_ss_->CreateSocket(family, type);
+  }
   return nullptr;
 }
 
-RunLoop::WorkerThread::WorkerThread(SocketServer* ss)
+RunLoop::WorkerThread::WorkerThread(SocketServer* absl_nullable ss)
     : Thread(ss), tq_setter_(this) {}
 
 }  // namespace test
diff --git a/test/run_loop.h b/test/run_loop.h
index 3050ce6f67..5ae3394b36 100644
--- a/test/run_loop.h
+++ b/test/run_loop.h
@@ -12,6 +12,7 @@
 
 #include <utility>
 
+#include "absl/base/nullability.h"
 #include "absl/functional/any_invocable.h"
 #include "api/task_queue/task_queue_base.h"
 #include "api/units/time_delta.h"
@@ -30,6 +31,7 @@ namespace test {
 class RunLoop {
  public:
   RunLoop();
+  explicit RunLoop(SocketServer* absl_nullable custom_ss);
   ~RunLoop();
 
   // Returns a pointer to the task queue implementation managed by this RunLoop.
@@ -62,30 +64,33 @@ class RunLoop {
   class FakeSocketServer : public SocketServer {
    public:
     FakeSocketServer();
+    explicit FakeSocketServer(SocketServer* absl_nullable custom_ss);
     ~FakeSocketServer() override;
 
     void FailNextWait();
 
    private:
+    void SetMessageQueue(Thread* absl_nullable queue) override;
     bool Wait(webrtc::TimeDelta max_wait_duration, bool process_io) override;
     void WakeUp() override;
 
-    Socket* CreateSocket(int family, int type) override;
+    Socket* absl_nullable CreateSocket(int family, int type) override;
 
    private:
     bool fail_next_wait_ = false;
+    SocketServer* const absl_nullable custom_ss_ = nullptr;
   };
 
   class WorkerThread : public Thread {
    public:
-    explicit WorkerThread(SocketServer* ss);
+    explicit WorkerThread(SocketServer* absl_nullable ss);
 
    private:
     CurrentTaskQueueSetter tq_setter_;
   };
 
   FakeSocketServer socket_server_;
-  WorkerThread worker_thread_{&socket_server_};
+  WorkerThread worker_thread_;
   WeakPtrFactory<RunLoop> weak_factory_;
 };
 

From c25f3da902e15331f2c295960646e8b2148b4da8 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Fri, 27 Mar 2026 16:11:11 +0100
Subject: [PATCH 435/472] Update ScopedOperationsBatcher tasks with finalizers
 to propagate errors

The Run() method now returns RTCError instead of void. Tasks with a
finalizer, now return RTCError. Upon encountering an error while running
batched tasks on the target thread, the batcher will abort and return.
This is to align with how these operations are today applied in a non
batched manner. Already queued up finalizers will be invoked.

Adding new type declarations for the callbacks for improved readability:
SimpleBatchTask, FinalizerTask and BatchTaskWithFinalizer.

Bug: webrtc:42222804
Change-Id: I03150abcdcc4fed41b18de5033fdb256f5802822
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/460540
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47288}
---
 pc/BUILD.gn                              |  2 +
 pc/scoped_operations_batcher.cc          | 42 +++++++++++------
 pc/scoped_operations_batcher.h           | 20 ++++----
 pc/scoped_operations_batcher_unittest.cc | 58 ++++++++++++++++++++++--
 4 files changed, 96 insertions(+), 26 deletions(-)

diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index a903e359d8..5db70e761f 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -1088,8 +1088,10 @@ rtc_library("scoped_operations_batcher") {
     "scoped_operations_batcher.h",
   ]
   deps = [
+    "../api:rtc_error",
     "../api:sequence_checker",
     "../rtc_base:checks",
+    "../rtc_base:logging",
     "../rtc_base:threading",
     "../rtc_base/system:no_unique_address",
     "//third_party/abseil-cpp/absl/functional:any_invocable",
diff --git a/pc/scoped_operations_batcher.cc b/pc/scoped_operations_batcher.cc
index eea4157110..ad6ccc5b55 100644
--- a/pc/scoped_operations_batcher.cc
+++ b/pc/scoped_operations_batcher.cc
@@ -16,8 +16,10 @@
 #include <vector>
 
 #include "absl/functional/any_invocable.h"
+#include "api/rtc_error.h"
 #include "api/sequence_checker.h"
 #include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
 #include "rtc_base/thread.h"
 
 namespace webrtc {
@@ -28,57 +30,69 @@ ScopedOperationsBatcher::ScopedOperationsBatcher(Thread* target_thread)
 }
 
 ScopedOperationsBatcher::~ScopedOperationsBatcher() {
-  Run();
+  RTCError error = Run();
+  if (!error.ok()) {
+    RTC_LOG(LS_ERROR) << "Batcher failed: " << error.message();
+  }
 }
 
-void ScopedOperationsBatcher::Run() {
+RTCError ScopedOperationsBatcher::Run() {
   RTC_DCHECK_RUN_ON(&sequence_checker_);
-  std::vector<absl::AnyInvocable<void() &&>> return_tasks;
+  std::vector<FinalizerTask> return_tasks;
 
   size_t task_idx = 0;
   bool target_thread_is_current = target_thread_->IsCurrent();
 
+  RTCError error = RTCError::OK();
   while (task_idx < tasks_.size()) {
     target_thread_->BlockingCall([&] {
       while (task_idx < tasks_.size()) {
-        if (auto* void_task =
-                std::get_if<absl::AnyInvocable<void() &&>>(&tasks_[task_idx])) {
+        if (auto* void_task = std::get_if<SimpleBatchTask>(&tasks_[task_idx])) {
           std::move (*void_task)();
         } else {
-          auto* returning_task = std::get_if<
-              absl::AnyInvocable<absl::AnyInvocable<void() &&>() &&>>(
-              &tasks_[task_idx]);
+          auto* returning_task =
+              std::get_if<BatchTaskWithFinalizer>(&tasks_[task_idx]);
           RTC_DCHECK(returning_task);
           auto ret = std::move(*returning_task)();
-          if (ret) {
-            return_tasks.push_back(std::move(ret));
+          if (ret.ok()) {
+            if (ret.value()) {
+              return_tasks.push_back(std::move(ret.value()));
+            }
+          } else {
+            error = ret.MoveError();
           }
         }
         ++task_idx;
+        if (!error.ok()) {
+          return;
+        }
         if (!target_thread_is_current && target_thread_->HasPendingTasks()) {
           return;
         }
       }
     });
+    if (!error.ok()) {
+      break;
+    }
   }
 
-  RTC_DCHECK_EQ(task_idx, tasks_.size());
   tasks_.clear();
 
   for (auto& task : return_tasks) {
     std::move(task)();
   }
+
+  return error;
 }
 
-void ScopedOperationsBatcher::Add(absl::AnyInvocable<void() &&> task) {
+void ScopedOperationsBatcher::Add(SimpleBatchTask task) {
   RTC_DCHECK_RUN_ON(&sequence_checker_);
   if (task) {
     tasks_.emplace_back(std::move(task));
   }
 }
 
-void ScopedOperationsBatcher::AddWithFinalizer(
-    absl::AnyInvocable<absl::AnyInvocable<void() &&>() &&> task) {
+void ScopedOperationsBatcher::AddWithFinalizer(BatchTaskWithFinalizer task) {
   RTC_DCHECK_RUN_ON(&sequence_checker_);
   if (task) {
     tasks_.emplace_back(std::move(task));
diff --git a/pc/scoped_operations_batcher.h b/pc/scoped_operations_batcher.h
index 6a5f905701..2c481d4407 100644
--- a/pc/scoped_operations_batcher.h
+++ b/pc/scoped_operations_batcher.h
@@ -15,6 +15,7 @@
 #include <vector>
 
 #include "absl/functional/any_invocable.h"
+#include "api/rtc_error.h"
 #include "api/sequence_checker.h"
 #include "rtc_base/system/no_unique_address.h"
 #include "rtc_base/thread.h"
@@ -31,27 +32,30 @@ namespace webrtc {
 // batcher will cooperatively yield the thread and resume execution in a
 // subsequent `BlockingCall`.
 //
-// Tasks can either have a `void` return type, or return a new task.
+// Tasks can either have a `void` return type, or return a new task or return an
+// error.
 // Any tasks returned by the executed worker thread tasks are collected
 // and subsequently executed on the calling thread (typically the signaling
 // thread) after the worker thread operations have completed.
 class ScopedOperationsBatcher {
  public:
+  using SimpleBatchTask = absl::AnyInvocable<void() &&>;
+  using FinalizerTask = absl::AnyInvocable<void() &&>;
+  using BatchTaskWithFinalizer =
+      absl::AnyInvocable<RTCErrorOr<FinalizerTask>() &&>;
+
   explicit ScopedOperationsBatcher(Thread* target_thread);
   ~ScopedOperationsBatcher();
 
-  void Run();
+  RTCError Run();
 
   // Queues non-nullptr tasks to be executed on the target thread when the
   // ScopedOperationsBatcher goes out of scope.
-  void Add(absl::AnyInvocable<void() &&> task);
-  void AddWithFinalizer(
-      absl::AnyInvocable<absl::AnyInvocable<void() &&>() &&> task);
+  void Add(SimpleBatchTask task);
+  void AddWithFinalizer(BatchTaskWithFinalizer task);
 
  private:
-  using BatchedTask =
-      std::variant<absl::AnyInvocable<void() &&>,
-                   absl::AnyInvocable<absl::AnyInvocable<void() &&>() &&>>;
+  using BatchedTask = std::variant<SimpleBatchTask, BatchTaskWithFinalizer>;
 
   RTC_NO_UNIQUE_ADDRESS SequenceChecker sequence_checker_;
   Thread* const target_thread_;
diff --git a/pc/scoped_operations_batcher_unittest.cc b/pc/scoped_operations_batcher_unittest.cc
index d8699967a5..afe36b7084 100644
--- a/pc/scoped_operations_batcher_unittest.cc
+++ b/pc/scoped_operations_batcher_unittest.cc
@@ -15,6 +15,7 @@
 #include <vector>
 
 #include "absl/functional/any_invocable.h"
+#include "api/rtc_error.h"
 #include "rtc_base/thread.h"
 #include "test/gtest.h"
 
@@ -57,14 +58,14 @@ TEST(ScopedOperationsBatcherTest, ExecutesReturnedTasksOnCallingThread) {
 
   {
     ScopedOperationsBatcher batcher(target_thread.get());
-    absl::AnyInvocable<absl::AnyInvocable<void() &&>() &&> task =
-        [&]() -> absl::AnyInvocable<void() &&> {
+    ScopedOperationsBatcher::BatchTaskWithFinalizer task =
+        [&]() -> RTCErrorOr<ScopedOperationsBatcher::FinalizerTask> {
       task_executed = true;
       task_thread = Thread::Current();
-      return [&]() {
+      return ScopedOperationsBatcher::FinalizerTask([&]() {
         return_task_executed = true;
         return_task_thread = Thread::Current();
-      };
+      });
     };
     batcher.AddWithFinalizer(std::move(task));
   }
@@ -100,5 +101,54 @@ TEST(ScopedOperationsBatcherTest, YieldsToOtherTasks) {
   EXPECT_EQ(execution_order, std::vector<int>({1, 2, 3, 4, 5}));
 }
 
+TEST(ScopedOperationsBatcherTest, StopsExecutionOnError) {
+  auto target_thread = Thread::Create();
+  target_thread->Start();
+
+  bool task1_executed = false;
+  bool task2_executed = false;
+  bool task3_executed = false;
+  // There's no finalizer for task2.
+  bool finalizer1_executed = false;
+  bool finalizer3_executed = false;
+
+  RTCError error = RTCError::OK();
+  {
+    ScopedOperationsBatcher batcher(target_thread.get());
+    // Task 1.
+    batcher.AddWithFinalizer(
+        [&]() -> RTCErrorOr<ScopedOperationsBatcher::FinalizerTask> {
+          task1_executed = true;
+          return ScopedOperationsBatcher::FinalizerTask(
+              [&] { finalizer1_executed = true; });
+        });
+    // Task 2.
+    batcher.AddWithFinalizer(
+        [&]() -> RTCErrorOr<ScopedOperationsBatcher::FinalizerTask> {
+          task2_executed = true;
+          return RTCError(RTCErrorType::INVALID_STATE, "Failed");
+        });
+    // Task 3.
+    batcher.AddWithFinalizer(
+        [&]() -> RTCErrorOr<ScopedOperationsBatcher::FinalizerTask> {
+          task3_executed = true;
+          return ScopedOperationsBatcher::FinalizerTask(
+              [&] { finalizer3_executed = true; });
+        });
+
+    error = batcher.Run();
+  }
+
+  EXPECT_FALSE(error.ok());
+  EXPECT_EQ(error.type(), RTCErrorType::INVALID_STATE);
+  EXPECT_STREQ(error.message(), "Failed");
+
+  EXPECT_TRUE(task1_executed);
+  EXPECT_TRUE(finalizer1_executed);
+  EXPECT_TRUE(task2_executed);
+  EXPECT_FALSE(task3_executed);
+  EXPECT_FALSE(finalizer3_executed);
+}
+
 }  // namespace
 }  // namespace webrtc

From 876820e1307aecbede8285fa56f0696d101e4ec4 Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Fri, 27 Mar 2026 21:05:14 -0700
Subject: [PATCH 436/472] Update WebRTC code version (2026-03-28T04:04:44).

Bug: None
Change-Id: Ic4ded4662ef9feefad2b221cbd10907a29b7791a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/460940
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47289}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index 0a885eda5b..ae06ea787d 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-27T04:10:01";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-28T04:04:44";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From 4487631837501e9ff44195f4a718f0b743c8e5d9 Mon Sep 17 00:00:00 2001
From: Jan Grulich <grulja@gmail.com>
Date: Wed, 25 Mar 2026 15:51:22 +0100
Subject: [PATCH 437/472] PipeWire capture: fix camera capability enumeration
 race and duplicates

Don't enumerate capabilities without actual format changes. Push the new
discovered capabilities into a pending list and swap pending list into
actual capabilities only after all params from the enumeration have been
received to ensure capabilities are never empty while enumerating.

Bug: webrtc:478386505
Change-Id: Iad8d076de067d4c663b8b281cb65158a38fd1309
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/457941
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Andreas Pehrson <apehrson@mozilla.com>
Cr-Commit-Position: refs/heads/main@{#47290}
---
 modules/portal/pipewire.sigs                  |  6 +++
 .../video_capture/linux/pipewire_session.cc   | 41 ++++++++++++++++---
 .../video_capture/linux/pipewire_session.h    |  5 +++
 3 files changed, 46 insertions(+), 6 deletions(-)

diff --git a/modules/portal/pipewire.sigs b/modules/portal/pipewire.sigs
index cab90280ef..f922048548 100644
--- a/modules/portal/pipewire.sigs
+++ b/modules/portal/pipewire.sigs
@@ -54,5 +54,11 @@ pw_context *pw_context_new(pw_loop *main_loop, pw_properties *props, size_t user
 pw_core * pw_context_connect(pw_context *context, pw_properties *properties, size_t user_data_size);
 pw_core * pw_context_connect_fd(pw_context *context, int fd, pw_properties *properties, size_t user_data_size);
 
+// node.h
+struct pw_node_info * pw_node_info_update(struct pw_node_info *info, const struct pw_node_info *update);
+void pw_node_info_free(struct pw_node_info *info);
+
 // proxy.h
 void pw_proxy_destroy(struct pw_proxy *proxy);
+void pw_proxy_add_listener(struct pw_proxy *proxy, struct spa_hook *listener, const struct pw_proxy_events *events, void *data);
+int pw_proxy_sync(struct pw_proxy *proxy, int seq);
diff --git a/modules/video_capture/linux/pipewire_session.cc b/modules/video_capture/linux/pipewire_session.cc
index a6a87fde3e..ba841a7f19 100644
--- a/modules/video_capture/linux/pipewire_session.cc
+++ b/modules/video_capture/linux/pipewire_session.cc
@@ -29,6 +29,7 @@
 #include <cstring>
 #include <memory>
 #include <optional>
+#include <utility>
 
 #include "absl/strings/string_view.h"
 #include "common_video/libyuv/include/webrtc_libyuv.h"
@@ -75,7 +76,9 @@ VideoType PipeWireRawFormatToVideoType(uint32_t id) {
 void PipeWireNode::PipeWireNodeDeleter::operator()(
     PipeWireNode* node) const noexcept {
   spa_hook_remove(&node->node_listener_);
+  spa_hook_remove(&node->proxy_listener_);
   pw_proxy_destroy(node->proxy_);
+  pw_node_info_free(node->info_);
 }
 
 // static
@@ -105,6 +108,13 @@ PipeWireNode::PipeWireNode(PipeWireSession* session,
   };
 
   pw_node_add_listener(reinterpret_cast<pw_node*>(proxy_), &node_listener_, &node_events, this);
+
+  static const pw_proxy_events proxy_events{
+      .version = PW_VERSION_PROXY_EVENTS,
+      .done = OnProxyDone,
+  };
+
+  pw_proxy_add_listener(proxy_, &proxy_listener_, &proxy_events, this);
 }
 
 // static
@@ -112,6 +122,12 @@ RTC_NO_SANITIZE("cfi-icall")
 void PipeWireNode::OnNodeInfo(void* data, const pw_node_info* info) {
   PipeWireNode* that = static_cast<PipeWireNode*>(data);
 
+  that->info_ = pw_node_info_update(that->info_, info);
+  if (!that->info_)
+    return;
+
+  info = that->info_;
+
   if (info->change_mask & PW_NODE_CHANGE_MASK_PROPS) {
     const char* vid_str;
     const char* pid_str;
@@ -133,18 +149,20 @@ void PipeWireNode::OnNodeInfo(void* data, const pw_node_info* info) {
 
   if (info->change_mask & PW_NODE_CHANGE_MASK_PARAMS) {
     for (uint32_t i = 0; i < info->n_params; i++) {
+      if (info->params[i].user == 0)
+        continue;
+      info->params[i].user = 0;
+
       uint32_t id = info->params[i].id;
       if (id == SPA_PARAM_EnumFormat &&
           info->params[i].flags & SPA_PARAM_INFO_READ) {
-        RTC_LOG(LS_INFO)
-            << "Clearing existing capabilities before re-enumeration for "
-            << that->display_name_;
-        that->capabilities_.clear();
+        that->pending_capabilities_.clear();
         pw_node_enum_params(reinterpret_cast<pw_node*>(that->proxy_), 0, id, 0, UINT32_MAX, nullptr);
+        that->sync_seq_ = pw_proxy_sync(that->proxy_, that->sync_seq_);
+        that->session_->PipeWireSync();
         break;
       }
     }
-    that->session_->PipeWireSync();
   }
 }
 
@@ -212,7 +230,18 @@ void PipeWireNode::OnNodeParam(void* data,
                       << cap.width << "x" << cap.height << "@" << cap.maxFPS
                       << ")";
 
-  that->capabilities_.push_back(cap);
+  that->pending_capabilities_.push_back(cap);
+}
+
+// static
+void PipeWireNode::OnProxyDone(void* data, int seq) {
+  PipeWireNode* that = static_cast<PipeWireNode*>(data);
+
+  if (seq != that->sync_seq_)
+    return;
+
+  that->capabilities_ = std::move(that->pending_capabilities_);
+  that->pending_capabilities_.clear();
 }
 
 // static
diff --git a/modules/video_capture/linux/pipewire_session.h b/modules/video_capture/linux/pipewire_session.h
index d192c09444..f03ada7528 100644
--- a/modules/video_capture/linux/pipewire_session.h
+++ b/modules/video_capture/linux/pipewire_session.h
@@ -72,15 +72,20 @@ class PipeWireNode {
                           uint32_t next,
                           const spa_pod* param);
   static bool ParseFormat(const spa_pod* param, VideoCaptureCapability* cap);
+  static void OnProxyDone(void* data, int seq);
 
   struct pw_proxy* proxy_;
   struct spa_hook node_listener_;
+  struct spa_hook proxy_listener_;
   PipeWireSession* session_;
   uint32_t id_;
   std::string display_name_;
   std::string unique_id_;
   std::string model_id_;
+  struct pw_node_info* info_ = nullptr;
+  int sync_seq_ = 0;
   std::vector<VideoCaptureCapability> capabilities_;
+  std::vector<VideoCaptureCapability> pending_capabilities_;
 };
 
 class CameraPortalNotifier : public CameraPortal::PortalNotifier {

From a73939ff93c0f82ad226d2fd8bc9d2b060909495 Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Sat, 28 Mar 2026 21:05:35 -0700
Subject: [PATCH 438/472] Update WebRTC code version (2026-03-29T04:05:05).

Bug: None
Change-Id: Ie8d6685c269746f919cdb060ad25cc723744cc92
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/460980
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47291}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index ae06ea787d..41aa24aa38 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-28T04:04:44";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-29T04:05:04";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From 65dd518777f0e0bb084367a9b5eb6af869cb5365 Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Sun, 29 Mar 2026 21:12:50 -0700
Subject: [PATCH 439/472] Update WebRTC code version (2026-03-30T04:12:15).

Bug: None
Change-Id: If3a89563f6f4cf61bc48a7eaef24d1574eb8425d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/461020
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47292}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index 41aa24aa38..8232446b74 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-29T04:05:04";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-30T04:12:15";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From 50f35f0996a9e63d3a6618c4dae807e119ba23bb Mon Sep 17 00:00:00 2001
From: Philipp Hancke <philipp.hancke@googlemail.com>
Date: Fri, 27 Mar 2026 16:31:20 +0100
Subject: [PATCH 440/472] Reapply "build: clean up modules audio build files"

This reverts commit 05946612a4cd0e295edba02b74c8cd3b45063fa6.
after restoring the problematic dependency and resolving
merge conflicts.

Bug: webrtc:467294026
Change-Id: I0105ce5918b4f3299483b253f98eb7c38a58e243
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/460760
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#47293}
---
 modules/audio_coding/BUILD.gn                 | 62 +------------------
 .../audio_coding/codecs/opus/test/BUILD.gn    |  2 -
 modules/audio_device/BUILD.gn                 | 10 ---
 modules/audio_mixer/BUILD.gn                  |  8 ---
 modules/audio_processing/BUILD.gn             | 33 ----------
 modules/audio_processing/aec3/BUILD.gn        |  6 --
 modules/audio_processing/aec_dump/BUILD.gn    |  7 ---
 modules/audio_processing/agc/BUILD.gn         | 10 ---
 modules/audio_processing/agc2/BUILD.gn        | 12 ----
 .../audio_processing/agc2/rnn_vad/BUILD.gn    |  3 -
 .../capture_levels_adjuster/BUILD.gn          |  1 -
 .../audio_processing/capture_mixer/BUILD.gn   |  2 -
 modules/audio_processing/ns/BUILD.gn          | 17 -----
 .../test/conversational_speech/BUILD.gn       |  1 -
 modules/audio_processing/utility/BUILD.gn     |  1 -
 modules/audio_processing/vad/BUILD.gn         |  2 -
 16 files changed, 3 insertions(+), 174 deletions(-)

diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index 0aeff4e301..25748a0a16 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -36,26 +36,18 @@ rtc_library("audio_coding") {
 
   deps = [
     ":audio_coding_module_typedefs",
-    ":neteq",
-    "..:module_api",
     "..:module_api_public",
     "../../api:function_view",
     "../../api/audio:audio_frame_api",
     "../../api/audio_codecs:audio_codecs_api",
-    "../../api/environment",
-    "../../api/neteq:default_neteq_factory",
-    "../../api/neteq:neteq_api",
-    "../../api/units:timestamp",
     "../../audio/utility:audio_frame_operations",
     "../../common_audio",
-    "../../common_audio:common_audio_c",
     "../../rtc_base:buffer",
     "../../rtc_base:checks",
     "../../rtc_base:logging",
     "../../rtc_base:macromagic",
     "../../rtc_base:safe_conversions",
     "../../rtc_base/synchronization:mutex",
-    "../../system_wrappers",
     "../../system_wrappers:metrics",
     "//third_party/abseil-cpp/absl/algorithm:container",
     "//third_party/abseil-cpp/absl/strings",
@@ -119,11 +111,9 @@ rtc_library("red") {
     "../../api:field_trials_view",
     "../../api/audio_codecs:audio_codecs_api",
     "../../api/units:time_delta",
-    "../../common_audio",
     "../../rtc_base:buffer",
     "../../rtc_base:byte_order",
     "../../rtc_base:checks",
-    "../../rtc_base:logging",
     "//third_party/abseil-cpp/absl/strings:string_view",
   ]
 }
@@ -174,7 +164,6 @@ rtc_library("g722") {
     "../../api/units:time_delta",
     "../../rtc_base:buffer",
     "../../rtc_base:checks",
-    "../../rtc_base:safe_conversions",
   ]
   public_deps += [ ":g722_c" ]  # no-presubmit-check TODO(webrtc:8603)
 }
@@ -206,7 +195,6 @@ rtc_library("isac_vad") {
   deps = [
     ":isac_bwinfo",
     "../../rtc_base:compile_assert_c",
-    "../../rtc_base/system:arch",
     "../../rtc_base/system:ignore_warnings",
     "../third_party/fft",
   ]
@@ -256,7 +244,6 @@ rtc_library("audio_coding_opus_common") {
   deps = [
     "../../api/audio_codecs:audio_codecs_api",
     "../../rtc_base:buffer",
-    "../../rtc_base:checks",
     "../../rtc_base:stringutils",
     "//third_party/abseil-cpp/absl/strings:string_view",
   ]
@@ -322,7 +309,6 @@ rtc_library("webrtc_multiopus") {
     "../../rtc_base:checks",
     "../../rtc_base:logging",
     "../../rtc_base:safe_conversions",
-    "../../rtc_base:safe_minmax",
     "../../rtc_base:stringutils",
     "//third_party/abseil-cpp/absl/memory",
     "//third_party/abseil-cpp/absl/strings",
@@ -504,7 +490,6 @@ rtc_library("neteq") {
   ]
 
   deps = [
-    ":audio_coding_module_typedefs",
     ":webrtc_cng",
     "..:module_api_public",
     "../../api:field_trials_view",
@@ -520,21 +505,20 @@ rtc_library("neteq") {
     "../../api/neteq:tick_timer",
     "../../api/units:time_delta",
     "../../api/units:timestamp",
-    "../../common_audio",
+
+    # TODO: bugs.webrtc.org/496700738 - fix dependency.
+    "../../common_audio",  # keep
     "../../common_audio:common_audio_c",
     "../../common_audio:common_audio_cc",
     "../../rtc_base:buffer",
     "../../rtc_base:checks",
     "../../rtc_base:event_tracer",
-    "../../rtc_base:gtest_prod",
     "../../rtc_base:logging",
-    "../../rtc_base:macromagic",
     "../../rtc_base:rtc_numerics",
     "../../rtc_base:safe_conversions",
     "../../rtc_base:safe_minmax",
     "../../rtc_base:sanitizer",
     "../../rtc_base/experiments:field_trial_parser",
-    "../../rtc_base/synchronization:mutex",
     "../../system_wrappers",
     "../../system_wrappers:metrics",
     "//third_party/abseil-cpp/absl/strings",
@@ -562,8 +546,6 @@ rtc_library("neteq_tools_minimal") {
   ]
 
   deps = [
-    ":neteq",
-    "../../api:field_trials",
     "../../api:field_trials_view",
     "../../api:neteq_simulator_api",
     "../../api:rtp_headers",
@@ -572,14 +554,11 @@ rtc_library("neteq_tools_minimal") {
     "../../api/audio_codecs:audio_codecs_api",
     "../../api/environment",
     "../../api/environment:environment_factory",
-    "../../api/neteq:custom_neteq_factory",
-    "../../api/neteq:default_neteq_controller_factory",
     "../../api/neteq:default_neteq_factory",
     "../../api/neteq:neteq_api",
     "../../api/units:timestamp",
     "../../rtc_base:buffer",
     "../../rtc_base:checks",
-    "../../rtc_base:copy_on_write_buffer",
     "../../rtc_base:safe_conversions",
     "../../rtc_base:stringutils",
     "../../system_wrappers",
@@ -647,7 +626,6 @@ rtc_library("neteq_tools") {
   deps = [
     ":neteq_input_audio_tools",
     ":neteq_tools_minimal",
-    "..:module_api_public",
     "../../api:rtp_headers",
     "../../api:rtp_packet_info",
     "../../api/audio:audio_frame_api",
@@ -661,7 +639,6 @@ rtc_library("neteq_tools") {
     "../../rtc_base:safe_conversions",
     "../../rtc_base:stringutils",
     "../../rtc_base:timeutils",
-    "../rtp_rtcp",
     "../rtp_rtcp:rtp_rtcp_format",
     "//third_party/abseil-cpp/absl/strings:string_view",
   ]
@@ -849,7 +826,6 @@ if (rtc_include_tests) {
       ":audio_encoder_cng",
       ":pcm16b_c",
       ":red",
-      "../../api:field_trials",
       "../../api:rtp_headers",
       "../../api:scoped_refptr",
       "../../api/audio:audio_frame_api",
@@ -904,8 +880,6 @@ if (rtc_include_tests) {
       "../../api/test/metrics:global_metrics_logger_and_exporter",
       "../../api/test/metrics:metric",
       "../../rtc_base:buffer",
-      "../../rtc_base:timeutils",
-      "../../system_wrappers",
       "../../test:fileutils",
       "../../test:test_flags",
       "../../test:test_support",
@@ -924,7 +898,6 @@ if (rtc_include_tests) {
 
     deps = [
       ":audio_coding",
-      ":neteq_tools",
       ":neteq_tools_minimal",
       "../../api:scoped_refptr",
       "../../api/audio_codecs:audio_codecs_api",
@@ -953,18 +926,13 @@ if (rtc_include_tests) {
       ":audio_coding",
       ":audio_coding_module_typedefs",
       ":neteq_input_audio_tools",
-      ":neteq_tools",
       ":neteq_tools_minimal",
       "../../api/audio:audio_frame_api",
       "../../api/audio_codecs:audio_codecs_api",
-      "../../api/audio_codecs:builtin_audio_decoder_factory",
       "../../api/audio_codecs:builtin_audio_encoder_factory",
       "../../api/environment",
       "../../api/environment:environment_factory",
       "../../rtc_base:checks",
-      "../../rtc_base:copy_on_write_buffer",
-      "../../rtc_base:stringutils",
-      "../../test:test_support",
       "../rtp_rtcp:rtp_rtcp_format",
       "//testing/gtest",
       "//third_party/abseil-cpp/absl/strings",
@@ -1024,14 +992,12 @@ if (rtc_include_tests) {
       visibility += webrtc_default_visibility
       defines = audio_codec_defines
       deps = [
-        ":neteq",
         ":neteq_input_audio_tools",
         ":neteq_test_tools",
         ":neteq_tools",
         ":neteq_tools_minimal",
         "../../api:field_trials",
         "../../api:make_ref_counted",
-        "../../api:rtp_headers",
         "../../api:scoped_refptr",
         "../../api/audio_codecs:audio_codecs_api",
         "../../api/audio_codecs:builtin_audio_decoder_factory",
@@ -1039,10 +1005,7 @@ if (rtc_include_tests) {
         "../../api/neteq:neteq_api",
         "../../logging:rtc_event_log_parser",
         "../../rtc_base:checks",
-        "../../rtc_base:refcount",
         "../../test:audio_test_common",
-        "../../test:fileutils",
-        "../../test:test_support",
         "../rtp_rtcp:rtp_rtcp_format",
         "//third_party/abseil-cpp/absl/strings:string_view",
       ]
@@ -1144,7 +1107,6 @@ if (rtc_include_tests) {
       "../../rtc_base:checks",
       "../../system_wrappers",
       "../../test:fileutils",
-      "../../test:test_support",
       "//testing/gtest",
     ]
   }
@@ -1158,7 +1120,6 @@ if (rtc_include_tests) {
       ]
 
       deps = [
-        ":neteq",
         ":neteq_input_audio_tools",
         ":neteq_test_tools",
         ":neteq_tools_minimal",
@@ -1173,7 +1134,6 @@ if (rtc_include_tests) {
         "../../rtc_base:buffer",
         "../../rtc_base:checks",
         "../../rtc_base:stringutils",
-        "../../system_wrappers",
         "../../test:fileutils",
         "../../test:test_support",
         "//testing/gtest",
@@ -1188,7 +1148,6 @@ if (rtc_include_tests) {
       deps = [
         ":audio_coding",
         ":audio_coding_module_typedefs",
-        ":audio_encoder_cng",
         ":neteq_input_audio_tools",
         "../../api/audio:audio_frame_api",
         "../../api/audio_codecs/L16:audio_encoder_L16",
@@ -1244,10 +1203,7 @@ if (rtc_include_tests) {
       sources = [ "neteq/tools/rtp_analyze.cc" ]
 
       deps = [
-        ":neteq",
         ":neteq_test_tools",
-        ":neteq_tools_minimal",
-        ":pcm16b",
         "../../api:rtp_headers",
         "../../rtc_base:checks",
         "../rtp_rtcp:rtp_rtcp_format",
@@ -1284,10 +1240,8 @@ if (rtc_include_tests) {
       sources = [ "neteq/test/neteq_speed_test.cc" ]
 
       deps = [
-        ":neteq",
         ":neteq_test_support",
         "../../rtc_base:checks",
-        "../../test:test_support",
         "//third_party/abseil-cpp/absl/flags:flag",
         "//third_party/abseil-cpp/absl/flags:parse",
       ]
@@ -1300,13 +1254,11 @@ if (rtc_include_tests) {
 
       deps = [
         ":g711",
-        ":neteq",
         ":neteq_quality_test_support",
         "../../api/audio_codecs:audio_codecs_api",
         "../../rtc_base:buffer",
         "../../rtc_base:checks",
         "../../rtc_base:safe_conversions",
-        "../../test:fileutils",
         "../../test:test_support",
         "//testing/gtest",
         "//third_party/abseil-cpp/absl/flags:flag",
@@ -1319,14 +1271,12 @@ if (rtc_include_tests) {
       sources = [ "neteq/test/neteq_pcm16b_quality_test.cc" ]
 
       deps = [
-        ":neteq",
         ":neteq_quality_test_support",
         ":pcm16b",
         "../../api/audio_codecs:audio_codecs_api",
         "../../rtc_base:buffer",
         "../../rtc_base:checks",
         "../../rtc_base:safe_conversions",
-        "../../test:fileutils",
         "../../test:test_support",
         "//testing/gtest",
         "//third_party/abseil-cpp/absl/flags:flag",
@@ -1501,18 +1451,12 @@ if (rtc_include_tests) {
         "../../logging:rtc_event_audio",
         "../../rtc_base:buffer",
         "../../rtc_base:checks",
-        "../../rtc_base:copy_on_write_buffer",
         "../../rtc_base:digest",
         "../../rtc_base:ip_address",
-        "../../rtc_base:macromagic",
-        "../../rtc_base:platform_thread",
         "../../rtc_base:random",
         "../../rtc_base:rtc_base_tests_utils",
-        "../../rtc_base:rtc_event",
         "../../rtc_base:safe_conversions",
         "../../rtc_base:stringutils",
-        "../../rtc_base:threading",
-        "../../rtc_base/synchronization:mutex",
         "../../rtc_base/system:arch",
         "../../system_wrappers",
         "../../test:audio_codec_mocks",
diff --git a/modules/audio_coding/codecs/opus/test/BUILD.gn b/modules/audio_coding/codecs/opus/test/BUILD.gn
index 0b2fc17a36..37cf935be7 100644
--- a/modules/audio_coding/codecs/opus/test/BUILD.gn
+++ b/modules/audio_coding/codecs/opus/test/BUILD.gn
@@ -46,9 +46,7 @@ if (rtc_include_tests) {
     deps = [
       ":test",
       "../../../../../common_audio",
-      "../../../../../common_audio:common_audio_c",
       "../../../../../rtc_base:checks",
-      "../../../../../rtc_base:macromagic",
       "../../../../../test:test_support",
       "//testing/gtest",
     ]
diff --git a/modules/audio_device/BUILD.gn b/modules/audio_device/BUILD.gn
index a5105a8257..e62ce13dd8 100644
--- a/modules/audio_device/BUILD.gn
+++ b/modules/audio_device/BUILD.gn
@@ -74,7 +74,6 @@ rtc_library("audio_device_buffer") {
     "../../rtc_base:timeutils",
     "../../rtc_base/synchronization:mutex",
     "../../rtc_base/system:no_unique_address",
-    "../../system_wrappers",
     "../../system_wrappers:metrics",
   ]
 }
@@ -87,7 +86,6 @@ rtc_library("audio_device_generic") {
   deps = [
     ":audio_device_buffer",
     "../../api/audio:audio_device",
-    "../../rtc_base:logging",
   ]
 }
 
@@ -193,11 +191,8 @@ if (!build_with_chromium) {
       "../../common_audio",
       "../../rtc_base:buffer",
       "../../rtc_base:checks",
-      "../../rtc_base:logging",
       "../../rtc_base:macromagic",
-      "../../rtc_base:platform_thread",
       "../../rtc_base:random",
-      "../../rtc_base:rtc_event",
       "../../rtc_base:safe_conversions",
       "../../rtc_base:timeutils",
       "../../rtc_base/synchronization:mutex",
@@ -244,7 +239,6 @@ if (!build_with_chromium) {
       "../../rtc_base:platform_thread",
       "../../rtc_base:stringutils",
       "../../rtc_base:threading",
-      "../../rtc_base:timeutils",
       "../../rtc_base/synchronization:mutex",
       "../../rtc_base/system:file_wrapper",
       "//third_party/abseil-cpp/absl/strings:string_view",
@@ -451,7 +445,6 @@ if (rtc_include_tests && !build_with_chromium) {
       "test_audio_device_impl_test.cc",
     ]
     deps = [
-      ":audio_device",
       ":audio_device_buffer",
       ":audio_device_generic",
       ":audio_device_impl",
@@ -463,8 +456,6 @@ if (rtc_include_tests && !build_with_chromium) {
       "../../api/audio:create_audio_device_module",
       "../../api/environment",
       "../../api/environment:environment_factory",
-      "../../api/task_queue",
-      "../../api/task_queue:default_task_queue_factory",
       "../../api/units:time_delta",
       "../../api/units:timestamp",
       "../../common_audio",
@@ -478,7 +469,6 @@ if (rtc_include_tests && !build_with_chromium) {
       "../../rtc_base:task_queue_for_test",
       "../../rtc_base:timeutils",
       "../../rtc_base/synchronization:mutex",
-      "../../system_wrappers",
       "../../test:create_test_environment",
       "../../test:fileutils",
       "../../test:test_support",
diff --git a/modules/audio_mixer/BUILD.gn b/modules/audio_mixer/BUILD.gn
index 362abf5dc9..49daca411a 100644
--- a/modules/audio_mixer/BUILD.gn
+++ b/modules/audio_mixer/BUILD.gn
@@ -45,20 +45,14 @@ rtc_library("audio_mixer_impl") {
     "../../api/audio:audio_frame_api",
     "../../api/audio:audio_mixer_api",
     "../../api/audio:audio_processing",
-    "../../audio/utility:audio_frame_operations",
     "../../common_audio",
     "../../rtc_base:checks",
     "../../rtc_base:event_tracer",
     "../../rtc_base:logging",
     "../../rtc_base:macromagic",
-    "../../rtc_base:race_checker",
-    "../../rtc_base:refcount",
-    "../../rtc_base:safe_conversions",
     "../../rtc_base/synchronization:mutex",
-    "../../system_wrappers",
     "../../system_wrappers:metrics",
     "../audio_processing:apm_logging",
-    "../audio_processing:audio_frame_view",
     "../audio_processing/agc2:fixed_digital",
   ]
 }
@@ -93,8 +87,6 @@ if (rtc_include_tests) {
     ]
 
     deps = [
-      ":audio_frame_manipulator",
-      ":audio_mixer_impl",
       "../../api/audio:audio_frame_api",
       "../../rtc_base:checks",
       "../../rtc_base:safe_conversions",
diff --git a/modules/audio_processing/BUILD.gn b/modules/audio_processing/BUILD.gn
index e169aa76a5..f1330d634d 100644
--- a/modules/audio_processing/BUILD.gn
+++ b/modules/audio_processing/BUILD.gn
@@ -32,7 +32,6 @@ rtc_library("audio_frame_proxies") {
     "include/audio_frame_proxies.h",
   ]
   deps = [
-    ":audio_frame_view",
     "../../api/audio:audio_frame_api",
     "../../api/audio:audio_processing",
     "../../api/audio:audio_processing_statistics",
@@ -135,7 +134,6 @@ rtc_library("gain_controller2") {
     "../../common_audio",
     "../../rtc_base:checks",
     "../../rtc_base:logging",
-    "../../rtc_base:stringutils",
     "agc2:adaptive_digital_gain_controller",
     "agc2:common",
     "agc2:cpu_features",
@@ -166,14 +164,11 @@ rtc_library("audio_processing") {
     ":apm_logging",
     ":audio_buffer",
     ":audio_frame_proxies",
-    ":audio_frame_view",
     ":gain_controller2",
     ":high_pass_filter",
     ":post_filter",
     ":rms_level",
     "../../api:field_trials_view",
-    "../../api:function_view",
-    "../../api:make_ref_counted",
     "../../api:scoped_refptr",
     "../../api/audio:aec3_config",
     "../../api/audio:audio_frame_api",
@@ -183,25 +178,16 @@ rtc_library("audio_processing") {
     "../../api/audio:neural_residual_echo_estimator_api",
     "../../api/environment",
     "../../api/task_queue",
-    "../../audio/utility:audio_frame_operations",
     "../../common_audio",
-    "../../common_audio:common_audio_c",
-    "../../common_audio:fir_filter",
-    "../../common_audio:fir_filter_factory",
-    "../../common_audio/third_party/ooura:fft_size_256",
     "../../rtc_base:checks",
     "../../rtc_base:denormal_disabler",
     "../../rtc_base:event_tracer",
     "../../rtc_base:gtest_prod",
     "../../rtc_base:logging",
     "../../rtc_base:macromagic",
-    "../../rtc_base:safe_minmax",
-    "../../rtc_base:sanitizer",
     "../../rtc_base:swap_queue",
     "../../rtc_base:timeutils",
     "../../rtc_base/synchronization:mutex",
-    "../../rtc_base/system:rtc_export",
-    "../../system_wrappers",
     "../../system_wrappers:metrics",
     "aec3",
     "aec_dump",
@@ -300,8 +286,6 @@ if (rtc_include_tests) {
     sources = [ "include/mock_audio_processing.h" ]
     deps = [
       ":aec_dump_interface",
-      ":audio_buffer",
-      ":audio_processing",
       "../../api:scoped_refptr",
       "../../api/audio:audio_processing",
       "../../api/audio:audio_processing_statistics",
@@ -357,7 +341,6 @@ if (rtc_include_tests) {
         "../../api:ref_count",
         "../../api:scoped_refptr",
         "../../api/audio:aec3_config",
-        "../../api/audio:aec3_factory",
         "../../api/audio:audio_frame_api",
         "../../api/audio:audio_processing",
         "../../api/audio:audio_processing_statistics",
@@ -367,11 +350,8 @@ if (rtc_include_tests) {
         "../../api/environment:environment_factory",
         "../../api/units:time_delta",
         "../../common_audio",
-        "../../common_audio:common_audio_c",
         "../../rtc_base:checks",
         "../../rtc_base:cpu_info",
-        "../../rtc_base:denormal_disabler",
-        "../../rtc_base:gtest_prod",
         "../../rtc_base:macromagic",
         "../../rtc_base:platform_thread",
         "../../rtc_base:protobuf_utils",
@@ -385,7 +365,6 @@ if (rtc_include_tests) {
         "../../rtc_base:task_queue_for_test",
         "../../rtc_base:threading",
         "../../rtc_base/synchronization:mutex",
-        "../../rtc_base/system:arch",
         "../../rtc_base/system:file_wrapper",
         "../../test:fileutils",
         "../../test:rtc_expect_death",
@@ -396,7 +375,6 @@ if (rtc_include_tests) {
         "agc:gain_control_interface",
         "agc2:adaptive_digital_gain_controller_unittest",
         "agc2:biquad_filter_unittests",
-        "agc2:cpu_features",
         "agc2:fixed_digital_unittests",
         "agc2:gain_applier_unittest",
         "agc2:input_volume_controller",
@@ -406,10 +384,8 @@ if (rtc_include_tests) {
         "agc2:saturation_protector_unittest",
         "agc2:speech_level_estimator_unittest",
         "agc2:test_utils",
-        "agc2:vad_wrapper",
         "agc2:vad_wrapper_unittests",
         "agc2/rnn_vad:unittests",
-        "capture_levels_adjuster",
         "capture_levels_adjuster:capture_levels_adjuster_unittests",
         "capture_mixer:capture_mixer_unittests",
         "test/conversational_speech:unittest",
@@ -484,10 +460,8 @@ if (rtc_include_tests) {
       "../../api/units:time_delta",
       "../../rtc_base:checks",
       "../../rtc_base:platform_thread",
-      "../../rtc_base:protobuf_utils",
       "../../rtc_base:random",
       "../../rtc_base:rtc_event",
-      "../../rtc_base:safe_conversions",
       "../../system_wrappers",
       "../../test:test_support",
       "//third_party/abseil-cpp/absl/strings:string_view",
@@ -500,7 +474,6 @@ if (rtc_include_tests) {
       "test/fake_recording_device.h",
     ]
     deps = [
-      "../../api/audio:audio_frame_api",
       "../../common_audio",
       "../../rtc_base:checks",
       "../../rtc_base:logging",
@@ -534,14 +507,12 @@ if (rtc_include_tests) {
         ":aec3_config_json",
         ":analog_mic_simulation",
         ":apm_logging",
-        ":audio_processing",
         ":audioproc_debug_proto",
         ":audioproc_protobuf_utils",
         ":audioproc_test_utils",
         "../../api:field_trials",
         "../../api:scoped_refptr",
         "../../api/audio:aec3_config",
-        "../../api/audio:aec3_factory",
         "../../api/audio:audio_processing",
         "../../api/audio:builtin_audio_processing_builder",
         "../../api/audio:echo_detector_creator",
@@ -582,7 +553,6 @@ if (rtc_include_tests) {
 
       deps = [
         ":audioproc_debug_proto",
-        "../../rtc_base:checks",
         "../../rtc_base:protobuf_utils",
         "../../rtc_base/system:arch",
       ]
@@ -625,13 +595,11 @@ rtc_library("audioproc_test_utils") {
 
   deps = [
     ":audio_buffer",
-    ":audio_processing",
     "../../api/audio:audio_frame_api",
     "../../api/audio:audio_processing",
     "../../common_audio",
     "../../rtc_base:checks",
     "../../rtc_base:random",
-    "../../rtc_base/system:arch",
     "../../system_wrappers",
     "../../test:fileutils",
     "../../test:test_support",
@@ -654,7 +622,6 @@ rtc_library("aec3_config_json") {
     "../../rtc_base:logging",
     "../../rtc_base:rtc_json",
     "../../rtc_base:stringutils",
-    "../../rtc_base/system:rtc_export",
     "//third_party/abseil-cpp/absl/strings:string_view",
   ]
 }
diff --git a/modules/audio_processing/aec3/BUILD.gn b/modules/audio_processing/aec3/BUILD.gn
index d0e1c0d7a5..f52017d935 100644
--- a/modules/audio_processing/aec3/BUILD.gn
+++ b/modules/audio_processing/aec3/BUILD.gn
@@ -148,7 +148,6 @@ rtc_library("aec3") {
     "../../../api/audio:echo_control",
     "../../../api/audio:neural_residual_echo_estimator_api",
     "../../../api/environment",
-    "../../../common_audio:common_audio_c",
     "../../../rtc_base:checks",
     "../../../rtc_base:cpu_info",
     "../../../rtc_base:gtest_prod",
@@ -185,7 +184,6 @@ rtc_source_set("aec3_fft") {
     ":fft_data",
     "../../../common_audio/third_party/ooura:fft_size_128",
     "../../../rtc_base:checks",
-    "../../../rtc_base/system:arch",
   ]
 }
 
@@ -201,7 +199,6 @@ rtc_source_set("render_buffer") {
     ":aec3_common",
     ":fft_data",
     "../../../rtc_base:checks",
-    "../../../rtc_base/system:arch",
   ]
 }
 
@@ -230,7 +227,6 @@ rtc_source_set("matched_filter") {
   sources = [ "matched_filter.h" ]
   deps = [
     ":aec3_common",
-    "../../../rtc_base:gtest_prod",
     "../../../rtc_base/system:arch",
   ]
 }
@@ -315,7 +311,6 @@ if (rtc_include_tests) {
       ":vector_math",
       "..:apm_logging",
       "..:audio_buffer",
-      "..:audio_processing",
       "..:high_pass_filter",
       "../../../api:field_trials",
       "../../../api/audio:aec3_config",
@@ -331,7 +326,6 @@ if (rtc_include_tests) {
       "../../../rtc_base/system:arch",
       "../../../system_wrappers:metrics",
       "../../../test:create_test_field_trials",
-      "../../../test:fileutils",
       "../../../test:test_support",
       "../utility:cascaded_biquad_filter",
     ]
diff --git a/modules/audio_processing/aec_dump/BUILD.gn b/modules/audio_processing/aec_dump/BUILD.gn
index 227d6dec6b..586a38ed79 100644
--- a/modules/audio_processing/aec_dump/BUILD.gn
+++ b/modules/audio_processing/aec_dump/BUILD.gn
@@ -33,8 +33,6 @@ if (rtc_include_tests) {
     deps = [
       "..:aec_dump_interface",
       "..:audio_frame_view",
-      "..:audioproc_test_utils",
-      "../",
       "../../../api/audio:audio_frame_api",
       "../../../api/audio:audio_processing",
       "../../../test:test_support",
@@ -48,8 +46,6 @@ if (rtc_include_tests) {
 
     deps = [
       ":mock_aec_dump",
-      "..:audioproc_test_utils",
-      "../",
       "../../../api:scoped_refptr",
       "../../../api/audio:audio_processing",
       "../../../api/audio:builtin_audio_processing_builder",
@@ -98,10 +94,7 @@ if (rtc_enable_protobuf) {
       defines = []
       deps = [
         ":aec_dump",
-        ":aec_dump_impl",
         "..:aec_dump_interface",
-        "..:audioproc_debug_proto",
-        "../",
         "../../../api/audio:audio_processing",
         "../../../rtc_base:task_queue_for_test",
         "../../../test:fileutils",
diff --git a/modules/audio_processing/agc/BUILD.gn b/modules/audio_processing/agc/BUILD.gn
index d2a4a5f05c..498971fbe6 100644
--- a/modules/audio_processing/agc/BUILD.gn
+++ b/modules/audio_processing/agc/BUILD.gn
@@ -28,7 +28,6 @@ rtc_library("agc") {
     "../../../api/audio:audio_processing",
     "../../../api/environment",
     "../../../common_audio",
-    "../../../common_audio:common_audio_c",
     "../../../rtc_base:checks",
     "../../../rtc_base:gtest_prod",
     "../../../rtc_base:logging",
@@ -37,7 +36,6 @@ rtc_library("agc") {
     "../agc2:clipping_predictor",
     "../agc2:gain_map",
     "../agc2:input_volume_stats_reporter",
-    "../vad",
   ]
 }
 
@@ -73,12 +71,9 @@ rtc_library("legacy_agc") {
   ]
 
   deps = [
-    "../../../common_audio",
     "../../../common_audio:common_audio_c",
     "../../../common_audio:common_audio_cc",
-    "../../../common_audio/third_party/ooura:fft_size_256",
     "../../../rtc_base:checks",
-    "../../../system_wrappers",
   ]
 
   if (rtc_build_with_neon) {
@@ -105,17 +100,12 @@ if (rtc_include_tests) {
       ":gain_control_interface",
       ":level_estimation",
       "..:audio_buffer",
-      "..:mocks",
-      "../../../api:field_trials",
       "../../../api/audio:audio_processing",
       "../../../api/environment",
       "../../../api/environment:environment_factory",
       "../../../rtc_base:checks",
-      "../../../rtc_base:random",
-      "../../../rtc_base:safe_conversions",
       "../../../rtc_base:safe_minmax",
       "../../../rtc_base:stringutils",
-      "../../../system_wrappers:metrics",
       "../../../test:create_test_field_trials",
       "../../../test:fileutils",
       "../../../test:test_support",
diff --git a/modules/audio_processing/agc2/BUILD.gn b/modules/audio_processing/agc2/BUILD.gn
index 55e28a79ba..0e3ee94a5a 100644
--- a/modules/audio_processing/agc2/BUILD.gn
+++ b/modules/audio_processing/agc2/BUILD.gn
@@ -145,9 +145,7 @@ rtc_library("fixed_digital") {
   deps = [
     ":common",
     "..:apm_logging",
-    "..:audio_frame_view",
     "../../../api/audio:audio_frame_api",
-    "../../../common_audio",
     "../../../rtc_base:checks",
     "../../../rtc_base:gtest_prod",
     "../../../rtc_base:safe_conversions",
@@ -227,11 +225,9 @@ rtc_library("noise_level_estimator") {
     "noise_level_estimator.h",
   ]
   deps = [
-    ":biquad_filter",
     "..:apm_logging",
     "../../../api/audio:audio_frame_api",
     "../../../rtc_base:checks",
-    "../../../system_wrappers",
   ]
 
   visibility = [
@@ -319,7 +315,6 @@ rtc_library("adaptive_digital_gain_controller_unittest") {
     ":common",
     ":test_utils",
     "..:apm_logging",
-    "..:audio_frame_view",
     "../../../api/audio:audio_processing",
     "../../../common_audio",
     "../../../test:test_support",
@@ -384,11 +379,9 @@ rtc_library("fixed_digital_unittests") {
     ":fixed_digital",
     ":test_utils",
     "..:apm_logging",
-    "..:audio_frame_view",
     "../../../api/audio:audio_frame_api",
     "../../../common_audio",
     "../../../rtc_base:checks",
-    "../../../system_wrappers:metrics",
     "../../../test:test_support",
   ]
 }
@@ -406,17 +399,13 @@ rtc_library("input_volume_controller_unittests") {
 
   deps = [
     ":clipping_predictor",
-    ":gain_map",
     ":input_volume_controller",
     "..:audio_buffer",
     "..:audio_frame_view",
     "../../../api/audio:audio_processing",
     "../../../api/environment:environment_factory",
     "../../../rtc_base:checks",
-    "../../../rtc_base:random",
-    "../../../rtc_base:safe_conversions",
     "../../../rtc_base:safe_minmax",
-    "../../../rtc_base:stringutils",
     "../../../system_wrappers:metrics",
     "../../../test:fileutils",
     "../../../test:test_support",
@@ -482,7 +471,6 @@ rtc_library("input_volume_stats_reporter") {
     "../../../rtc_base:checks",
     "../../../rtc_base:gtest_prod",
     "../../../rtc_base:logging",
-    "../../../rtc_base:safe_minmax",
     "../../../rtc_base:stringutils",
     "../../../system_wrappers:metrics",
     "//third_party/abseil-cpp/absl/strings:string_view",
diff --git a/modules/audio_processing/agc2/rnn_vad/BUILD.gn b/modules/audio_processing/agc2/rnn_vad/BUILD.gn
index 2e9c473d5f..0f581c6922 100644
--- a/modules/audio_processing/agc2/rnn_vad/BUILD.gn
+++ b/modules/audio_processing/agc2/rnn_vad/BUILD.gn
@@ -33,8 +33,6 @@ rtc_library("rnn_vad") {
     "..:biquad_filter",
     "..:cpu_features",
     "../../../../rtc_base:checks",
-    "../../../../rtc_base:safe_compare",
-    "../../../../rtc_base:safe_conversions",
     "//third_party/rnnoise:rnn_vad",
   ]
 }
@@ -306,7 +304,6 @@ if (rtc_include_tests) {
         "../../../../rtc_base:checks",
         "../../../../rtc_base:logging",
         "../../../../rtc_base:safe_compare",
-        "../../../../test:test_support",
         "//third_party/abseil-cpp/absl/flags:flag",
         "//third_party/abseil-cpp/absl/flags:parse",
       ]
diff --git a/modules/audio_processing/capture_levels_adjuster/BUILD.gn b/modules/audio_processing/capture_levels_adjuster/BUILD.gn
index 61292c7d6a..5bd81487d4 100644
--- a/modules/audio_processing/capture_levels_adjuster/BUILD.gn
+++ b/modules/audio_processing/capture_levels_adjuster/BUILD.gn
@@ -38,7 +38,6 @@ rtc_library("capture_levels_adjuster_unittests") {
     ":capture_levels_adjuster",
     "..:audio_buffer",
     "..:audioproc_test_utils",
-    "../../../rtc_base:stringutils",
     "../../../test:test_support",
   ]
 }
diff --git a/modules/audio_processing/capture_mixer/BUILD.gn b/modules/audio_processing/capture_mixer/BUILD.gn
index c7b2b4c6f4..8d7b0eb444 100644
--- a/modules/audio_processing/capture_mixer/BUILD.gn
+++ b/modules/audio_processing/capture_mixer/BUILD.gn
@@ -47,8 +47,6 @@ rtc_library("capture_mixer_unittests") {
   ]
   deps = [
     ":capture_mixer",
-    "..:audioproc_test_utils",
-    "../../../rtc_base:stringutils",
     "../../../test:test_support",
   ]
 }
diff --git a/modules/audio_processing/ns/BUILD.gn b/modules/audio_processing/ns/BUILD.gn
index 83cce6076e..6324ed2bb7 100644
--- a/modules/audio_processing/ns/BUILD.gn
+++ b/modules/audio_processing/ns/BUILD.gn
@@ -49,18 +49,9 @@ rtc_static_library("ns") {
   }
 
   deps = [
-    "..:apm_logging",
     "..:audio_buffer",
-    "..:high_pass_filter",
-    "../../../common_audio:common_audio_c",
-    "../../../common_audio/third_party/ooura:fft_size_128",
     "../../../common_audio/third_party/ooura:fft_size_256",
     "../../../rtc_base:checks",
-    "../../../rtc_base:safe_minmax",
-    "../../../rtc_base/system:arch",
-    "../../../system_wrappers",
-    "../../../system_wrappers:metrics",
-    "../utility:cascaded_biquad_filter",
   ]
 }
 
@@ -73,17 +64,9 @@ if (rtc_include_tests) {
 
     deps = [
       ":ns",
-      "..:apm_logging",
       "..:audio_buffer",
-      "..:audio_processing",
-      "..:high_pass_filter",
-      "../../../rtc_base:checks",
-      "../../../rtc_base:safe_minmax",
       "../../../rtc_base:stringutils",
-      "../../../rtc_base/system:arch",
-      "../../../system_wrappers",
       "../../../test:test_support",
-      "../utility:cascaded_biquad_filter",
     ]
 
     defines = []
diff --git a/modules/audio_processing/test/conversational_speech/BUILD.gn b/modules/audio_processing/test/conversational_speech/BUILD.gn
index 88e2613c48..19b8beebda 100644
--- a/modules/audio_processing/test/conversational_speech/BUILD.gn
+++ b/modules/audio_processing/test/conversational_speech/BUILD.gn
@@ -21,7 +21,6 @@ if (!build_with_chromium) {
       ":lib",
       "../../../../rtc_base:checks",
       "../../../../test:fileutils",
-      "../../../../test:test_support",
       "//third_party/abseil-cpp/absl/flags:flag",
       "//third_party/abseil-cpp/absl/flags:parse",
     ]
diff --git a/modules/audio_processing/utility/BUILD.gn b/modules/audio_processing/utility/BUILD.gn
index 413aba48d6..7a23781ce5 100644
--- a/modules/audio_processing/utility/BUILD.gn
+++ b/modules/audio_processing/utility/BUILD.gn
@@ -46,7 +46,6 @@ if (rtc_include_tests) {
     sources = [ "cascaded_biquad_filter_unittest.cc" ]
     deps = [
       ":cascaded_biquad_filter",
-      "../../../rtc_base:checks",
       "../../../test:test_support",
       "//rtc_base:checks",
       "//testing/gtest",
diff --git a/modules/audio_processing/vad/BUILD.gn b/modules/audio_processing/vad/BUILD.gn
index 71e079d3a3..34b3003e6d 100644
--- a/modules/audio_processing/vad/BUILD.gn
+++ b/modules/audio_processing/vad/BUILD.gn
@@ -35,7 +35,6 @@ rtc_library("vad") {
     "voice_gmm_tables.h",
   ]
   deps = [
-    "../../../audio/utility:audio_frame_operations",
     "../../../common_audio",
     "../../../common_audio:common_audio_c",
     "../../../common_audio/third_party/ooura:fft_size_256",
@@ -59,7 +58,6 @@ if (rtc_include_tests) {
     ]
     deps = [
       ":vad",
-      "../../../common_audio",
       "../../../test:fileutils",
       "../../../test:test_support",
       "//testing/gmock",

From c9752917bd482d964d6a8a1de9bbe3b300fdc1a5 Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Fri, 27 Mar 2026 15:27:59 +0100
Subject: [PATCH 441/472] Replace ArrayView with std::span everywhere except
 api

ArrayView is an alias to std::span. This change switch to use
std::span directly instead of through the alias.

Search&Replace MakeArrayView and ArrayView with std::span
Search&Replace include "api/array_view.h" with include <span>
Remove <span> include where std::span is not mentioned in the file
Remove build dependencies on array_view target

Bug: webrtc:439801349
Change-Id: I64bbdcf1300126286158ccbeda62c49e0c3ec83f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/460501
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47294}
---
 audio/BUILD.gn                                |  3 --
 audio/audio_receive_stream.cc                 |  4 +-
 audio/audio_receive_stream.h                  |  4 +-
 audio/audio_send_stream.cc                    |  4 +-
 audio/audio_send_stream.h                     |  4 +-
 audio/audio_send_stream_tests.cc              | 12 ++---
 audio/channel_receive.cc                      | 14 ++---
 ...nnel_receive_frame_transformer_delegate.cc | 14 ++---
 ...annel_receive_frame_transformer_delegate.h |  6 +--
 ...ive_frame_transformer_delegate_unittest.cc | 20 +++----
 audio/channel_receive_unittest.cc             |  8 +--
 audio/channel_send.cc                         | 22 ++++----
 audio/channel_send.h                          |  4 +-
 ...channel_send_frame_transformer_delegate.cc |  8 +--
 .../channel_send_frame_transformer_delegate.h |  6 +--
 ...end_frame_transformer_delegate_unittest.cc | 12 ++---
 audio/channel_send_unittest.cc                | 16 +++---
 audio/mock_voe_channel_proxy.h                |  4 +-
 audio/voip/BUILD.gn                           |  4 --
 audio/voip/audio_channel.h                    |  6 +--
 audio/voip/audio_egress.cc                    |  4 +-
 audio/voip/audio_ingress.cc                   |  8 +--
 audio/voip/audio_ingress.h                    |  6 +--
 audio/voip/test/BUILD.gn                      |  3 --
 audio/voip/test/audio_channel_unittest.cc     | 16 +++---
 audio/voip/test/audio_egress_unittest.cc      | 12 ++---
 audio/voip/test/audio_ingress_unittest.cc     | 10 ++--
 audio/voip/voip_core.cc                       |  6 +--
 audio/voip/voip_core.h                        |  6 +--
 call/BUILD.gn                                 |  8 ---
 call/call.cc                                  |  4 +-
 call/call_perf_tests.cc                       |  6 +--
 call/fake_network_pipe.cc                     | 10 ++--
 call/fake_network_pipe.h                      |  6 +--
 call/flexfec_receive_stream_impl.cc           |  1 -
 call/flexfec_receive_stream_unittest.cc       |  4 +-
 call/rtp_config.cc                            |  1 -
 call/rtp_transport_controller_send.cc         |  4 +-
 call/rtp_transport_controller_send.h          |  4 +-
 call/rtp_video_sender.cc                      | 10 ++--
 call/rtp_video_sender.h                       |  8 +--
 call/rtp_video_sender_interface.h             |  8 +--
 call/rtp_video_sender_unittest.cc             | 54 +++++++++----------
 call/rtx_receive_stream.cc                    |  4 +-
 call/rtx_receive_stream_unittest.cc           | 28 +++++-----
 call/video_send_stream.h                      |  4 +-
 common_audio/BUILD.gn                         |  1 -
 common_audio/channel_buffer.h                 | 24 ++++-----
 common_audio/include/audio_util.h             |  2 +-
 examples/BUILD.gn                             |  2 -
 examples/androidvoip/BUILD.gn                 |  1 -
 .../androidvoip/jni/android_voip_client.cc    | 10 ++--
 .../androidvoip/jni/android_voip_client.h     |  6 +--
 .../peerconnection/client/linux/main_wnd.h    |  4 +-
 examples/turnserver/read_auth_file.cc         |  4 +-
 system_wrappers/BUILD.gn                      |  1 -
 56 files changed, 220 insertions(+), 245 deletions(-)

diff --git a/audio/BUILD.gn b/audio/BUILD.gn
index c39c42dd70..c710be09b9 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -40,7 +40,6 @@ rtc_library("audio") {
   ]
 
   deps = [
-    "../api:array_view",
     "../api:bitrate_allocation",
     "../api:call_api",
     "../api:field_trials_view",
@@ -165,7 +164,6 @@ if (rtc_include_tests) {
       ":audio",
       ":audio_end_to_end_test",
       ":channel_receive_unittest",
-      "../api:array_view",
       "../api:bitrate_allocation",
       "../api:call_api",
       "../api:field_trials",
@@ -252,7 +250,6 @@ if (rtc_include_tests) {
     sources = [ "channel_receive_unittest.cc" ]
     deps = [
       ":audio",
-      "../api:array_view",
       "../api:make_ref_counted",
       "../api:mock_frame_transformer",
       "../api:scoped_refptr",
diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
index 4f31b489ea..4f052c6a21 100644
--- a/audio/audio_receive_stream.cc
+++ b/audio/audio_receive_stream.cc
@@ -15,12 +15,12 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio/audio_frame.h"
 #include "api/audio/audio_mixer.h"
 #include "api/audio_codecs/audio_format.h"
@@ -446,7 +446,7 @@ bool AudioReceiveStreamImpl::SetMinimumPlayoutDelay(TimeDelta delay) {
   return channel_receive_->SetMinimumPlayoutDelay(delay);
 }
 
-void AudioReceiveStreamImpl::DeliverRtcp(ArrayView<const uint8_t> packet) {
+void AudioReceiveStreamImpl::DeliverRtcp(std::span<const uint8_t> packet) {
   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   channel_receive_->ReceivedRTCPPacket(packet.data(), packet.size());
 }
diff --git a/audio/audio_receive_stream.h b/audio/audio_receive_stream.h
index 111cd898dc..41a470f8b8 100644
--- a/audio/audio_receive_stream.h
+++ b/audio/audio_receive_stream.h
@@ -16,11 +16,11 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio/audio_frame.h"
 #include "api/audio/audio_mixer.h"
 #include "api/audio_codecs/audio_format.h"
@@ -128,7 +128,7 @@ class AudioReceiveStreamImpl final : public webrtc::AudioReceiveStreamInterface,
                                        Timestamp time) override;
   bool SetMinimumPlayoutDelay(TimeDelta delay) override;
 
-  void DeliverRtcp(ArrayView<const uint8_t> packet);
+  void DeliverRtcp(std::span<const uint8_t> packet);
 
   void SetSyncGroup(absl::string_view sync_group);
 
diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index 4c53757e74..25efcb8f92 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -15,13 +15,13 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/str_cat.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio/audio_frame.h"
 #include "api/audio/audio_processing.h"
 #include "api/audio_codecs/audio_encoder.h"
@@ -495,7 +495,7 @@ webrtc::AudioSendStream::Stats AudioSendStream::GetStats(
   return stats;
 }
 
-void AudioSendStream::DeliverRtcp(ArrayView<const uint8_t> packet) {
+void AudioSendStream::DeliverRtcp(std::span<const uint8_t> packet) {
   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   channel_send_->ReceivedRTCPPacket(packet.data(), packet.size());
   // Poll if overhead has changed, which it can do if ack triggers us to stop
diff --git a/audio/audio_send_stream.h b/audio/audio_send_stream.h
index 9bac56005a..7cc204c7bd 100644
--- a/audio/audio_send_stream.h
+++ b/audio/audio_send_stream.h
@@ -15,10 +15,10 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/call/bitrate_allocation.h"
 #include "api/environment/environment.h"
 #include "api/field_trials_view.h"
@@ -103,7 +103,7 @@ class AudioSendStream final : public webrtc::AudioSendStream,
   webrtc::AudioSendStream::Stats GetStats(
       bool has_remote_tracks) const override;
 
-  void DeliverRtcp(ArrayView<const uint8_t> packet);
+  void DeliverRtcp(std::span<const uint8_t> packet);
 
   // Implements BitrateAllocatorObserver.
   uint32_t OnBitrateUpdated(BitrateAllocationUpdate update) override;
diff --git a/audio/audio_send_stream_tests.cc b/audio/audio_send_stream_tests.cc
index 6c754a1878..5379026c1d 100644
--- a/audio/audio_send_stream_tests.cc
+++ b/audio/audio_send_stream_tests.cc
@@ -10,10 +10,10 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/rtp_headers.h"
 #include "api/rtp_parameters.h"
 #include "call/audio_receive_stream.h"
@@ -56,7 +56,7 @@ TEST_F(AudioSendStreamCallTest, SupportsCName) {
     CNameObserver() = default;
 
    private:
-    Action OnSendRtcp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtcp(std::span<const uint8_t> packet) override {
       RtcpPacketParser parser;
       EXPECT_TRUE(parser.Parse(packet));
       if (parser.sdes()->num_packets() > 0) {
@@ -89,7 +89,7 @@ TEST_F(AudioSendStreamCallTest, NoExtensionsByDefault) {
     NoExtensionsObserver() = default;
 
    private:
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       RtpPacket rtp_packet;
       EXPECT_TRUE(rtp_packet.Parse(packet));  // rtp packet is valid.
       EXPECT_EQ(packet[0] & 0b0001'0000, 0);  // extension bit not set.
@@ -119,7 +119,7 @@ TEST_F(AudioSendStreamCallTest, SupportsAudioLevel) {
       extensions_.Register<AudioLevelExtension>(kAudioLevelExtensionId);
     }
 
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       RtpPacket rtp_packet(&extensions_);
       EXPECT_TRUE(rtp_packet.Parse(packet));
 
@@ -164,7 +164,7 @@ class TransportWideSequenceNumberObserver : public AudioSendTest {
   }
 
  private:
-  Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+  Action OnSendRtp(std::span<const uint8_t> packet) override {
     RtpPacket rtp_packet(&extensions_);
     EXPECT_TRUE(rtp_packet.Parse(packet));
 
@@ -210,7 +210,7 @@ TEST_F(AudioSendStreamCallTest, SendDtmf) {
     DtmfObserver() = default;
 
    private:
-    Action OnSendRtp(ArrayView<const uint8_t> packet) override {
+    Action OnSendRtp(std::span<const uint8_t> packet) override {
       RtpPacket rtp_packet;
       EXPECT_TRUE(rtp_packet.Parse(packet));
 
diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index 2eb4a6202a..dfd4cd4091 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -17,11 +17,11 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/audio_device.h"
 #include "api/audio/audio_mixer.h"
 #include "api/audio_codecs/audio_decoder_factory.h"
@@ -220,7 +220,7 @@ class ChannelReceive : public ChannelReceiveInterface,
 
   int GetRtpTimestampRateHz() const;
 
-  void OnReceivedPayloadData(ArrayView<const uint8_t> payload,
+  void OnReceivedPayloadData(std::span<const uint8_t> payload,
                              const RTPHeader& header,
                              Timestamp receive_time)
       RTC_RUN_ON(worker_thread_checker_);
@@ -334,7 +334,7 @@ class ChannelReceive : public ChannelReceiveInterface,
       RTC_GUARDED_BY(worker_thread_checker_);
 };
 
-void ChannelReceive::OnReceivedPayloadData(ArrayView<const uint8_t> payload,
+void ChannelReceive::OnReceivedPayloadData(std::span<const uint8_t> payload,
                                            const RTPHeader& header,
                                            Timestamp receive_time) {
   if (!playing_) {
@@ -379,7 +379,7 @@ void ChannelReceive::InitFrameTransformerDelegate(
   // Pass a callback to ChannelReceive::OnReceivedPayloadData, to be called by
   // the delegate to receive transformed audio.
   ChannelReceiveFrameTransformerDelegate::ReceiveFrameCallback
-      receive_audio_callback = [this](ArrayView<const uint8_t> packet,
+      receive_audio_callback = [this](std::span<const uint8_t> packet,
                                       const RTPHeader& header,
                                       Timestamp receive_time) {
         RTC_DCHECK_RUN_ON(&worker_thread_checker_);
@@ -737,7 +737,7 @@ void ChannelReceive::ReceivePacket(const uint8_t* packet,
         frame_decryptor_->Decrypt(
             MediaType::AUDIO, csrcs,
             /*additional_data=*/{},
-            ArrayView<const uint8_t>(payload, payload_data_length),
+            std::span<const uint8_t>(payload, payload_data_length),
             decrypted_audio_payload);
 
     if (decrypt_result.IsOk()) {
@@ -755,7 +755,7 @@ void ChannelReceive::ReceivePacket(const uint8_t* packet,
     payload_data_length = 0;
   }
 
-  ArrayView<const uint8_t> payload_data(payload, payload_data_length);
+  std::span<const uint8_t> payload_data(payload, payload_data_length);
   if (frame_transformer_delegate_) {
     // Asynchronously transform the received payload. After the payload is
     // transformed, the delegate will call OnReceivedPayloadData to handle it.
@@ -778,7 +778,7 @@ void ChannelReceive::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
   UpdatePlayoutTimestamp(true, env_.clock().CurrentTime());
 
   // Deliver RTCP packet to RTP/RTCP module for parsing
-  rtp_rtcp_->IncomingRtcpPacket(MakeArrayView(data, length));
+  rtp_rtcp_->IncomingRtcpPacket(std::span(data, length));
 
   std::optional<TimeDelta> rtt = rtp_rtcp_->LastRtt();
   if (!rtt.has_value()) {
diff --git a/audio/channel_receive_frame_transformer_delegate.cc b/audio/channel_receive_frame_transformer_delegate.cc
index 887edaa2ee..b2422d0aab 100644
--- a/audio/channel_receive_frame_transformer_delegate.cc
+++ b/audio/channel_receive_frame_transformer_delegate.cc
@@ -14,10 +14,10 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 
-#include "api/array_view.h"
 #include "api/frame_transformer_interface.h"
 #include "api/rtp_headers.h"
 #include "api/scoped_refptr.h"
@@ -34,7 +34,7 @@ namespace webrtc {
 class TransformableIncomingAudioFrame
     : public TransformableAudioFrameInterface {
  public:
-  TransformableIncomingAudioFrame(ArrayView<const uint8_t> payload,
+  TransformableIncomingAudioFrame(std::span<const uint8_t> payload,
                                   const RTPHeader& header,
                                   uint32_t ssrc,
                                   const std::string& codec_mime_type,
@@ -46,9 +46,9 @@ class TransformableIncomingAudioFrame
         codec_mime_type_(codec_mime_type),
         receive_time_(receive_time) {}
   ~TransformableIncomingAudioFrame() override = default;
-  ArrayView<const uint8_t> GetData() const override { return payload_; }
+  std::span<const uint8_t> GetData() const override { return payload_; }
 
-  void SetData(ArrayView<const uint8_t> data) override {
+  void SetData(std::span<const uint8_t> data) override {
     payload_.SetData(data.data(), data.size());
   }
 
@@ -59,8 +59,8 @@ class TransformableIncomingAudioFrame
   uint8_t GetPayloadType() const override { return header_.payloadType; }
   uint32_t GetSsrc() const override { return ssrc_; }
   uint32_t GetTimestamp() const override { return header_.timestamp; }
-  ArrayView<const uint32_t> GetContributingSources() const override {
-    return ArrayView<const uint32_t>(header_.arrOfCSRCs, header_.numCSRCs);
+  std::span<const uint32_t> GetContributingSources() const override {
+    return std::span<const uint32_t>(header_.arrOfCSRCs, header_.numCSRCs);
   }
   Direction GetDirection() const override { return Direction::kReceiver; }
 
@@ -159,7 +159,7 @@ void ChannelReceiveFrameTransformerDelegate::Reset() {
 }
 
 void ChannelReceiveFrameTransformerDelegate::Transform(
-    ArrayView<const uint8_t> packet,
+    std::span<const uint8_t> packet,
     const RTPHeader& header,
     uint32_t ssrc,
     const std::string& codec_mime_type,
diff --git a/audio/channel_receive_frame_transformer_delegate.h b/audio/channel_receive_frame_transformer_delegate.h
index 1b5f04eeee..a0ba8fd0f2 100644
--- a/audio/channel_receive_frame_transformer_delegate.h
+++ b/audio/channel_receive_frame_transformer_delegate.h
@@ -14,9 +14,9 @@
 #include <cstdint>
 #include <functional>
 #include <memory>
+#include <span>
 #include <string>
 
-#include "api/array_view.h"
 #include "api/frame_transformer_interface.h"
 #include "api/rtp_headers.h"
 #include "api/scoped_refptr.h"
@@ -34,7 +34,7 @@ namespace webrtc {
 class ChannelReceiveFrameTransformerDelegate : public TransformedFrameCallback {
  public:
   using ReceiveFrameCallback =
-      std::function<void(webrtc::ArrayView<const uint8_t> packet,
+      std::function<void(std::span<const uint8_t> packet,
                          const RTPHeader& header,
                          Timestamp receive_time)>;
   ChannelReceiveFrameTransformerDelegate(
@@ -54,7 +54,7 @@ class ChannelReceiveFrameTransformerDelegate : public TransformedFrameCallback {
 
   // Delegates the call to FrameTransformerInterface::Transform, to transform
   // the frame asynchronously.
-  void Transform(ArrayView<const uint8_t> packet,
+  void Transform(std::span<const uint8_t> packet,
                  const RTPHeader& header,
                  uint32_t ssrc,
                  const std::string& codec_mime_type,
diff --git a/audio/channel_receive_frame_transformer_delegate_unittest.cc b/audio/channel_receive_frame_transformer_delegate_unittest.cc
index ae9a01cd69..01dbaee486 100644
--- a/audio/channel_receive_frame_transformer_delegate_unittest.cc
+++ b/audio/channel_receive_frame_transformer_delegate_unittest.cc
@@ -12,9 +12,9 @@
 
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <utility>
 
-#include "api/array_view.h"
 #include "api/frame_transformer_factory.h"
 #include "api/frame_transformer_interface.h"
 #include "api/make_ref_counted.h"
@@ -44,12 +44,12 @@ class MockChannelReceive {
  public:
   MOCK_METHOD(void,
               ReceiveFrame,
-              (ArrayView<const uint8_t> packet,
+              (std::span<const uint8_t> packet,
                const RTPHeader& header,
                Timestamp receive_time));
 
   ChannelReceiveFrameTransformerDelegate::ReceiveFrameCallback callback() {
-    return [this](ArrayView<const uint8_t> packet, const RTPHeader& header,
+    return [this](std::span<const uint8_t> packet, const RTPHeader& header,
                   Timestamp receive_time) {
       ReceiveFrame(packet, header, receive_time);
     };
@@ -101,7 +101,7 @@ TEST(ChannelReceiveFrameTransformerDelegateTest,
   ASSERT_TRUE(callback);
 
   const uint8_t data[] = {1, 2, 3, 4};
-  ArrayView<const uint8_t> packet(data, sizeof(data));
+  std::span<const uint8_t> packet(data, sizeof(data));
   RTPHeader header;
   EXPECT_CALL(mock_channel, ReceiveFrame);
   ON_CALL(*mock_frame_transformer, Transform)
@@ -132,7 +132,7 @@ TEST(ChannelReceiveFrameTransformerDelegateTest,
   ASSERT_TRUE(callback);
 
   const uint8_t data[] = {1, 2, 3, 4};
-  ArrayView<const uint8_t> packet(data, sizeof(data));
+  std::span<const uint8_t> packet(data, sizeof(data));
   RTPHeader header;
   EXPECT_CALL(mock_channel,
               ReceiveFrame(ElementsAre(1, 2, 3, 4), _, kFakeReceiveTimestamp));
@@ -177,7 +177,7 @@ TEST(ChannelReceiveFrameTransformerDelegateTest,
       make_ref_counted<ChannelReceiveFrameTransformerDelegate>(
           mock_channel.callback(), mock_frame_transformer, Thread::Current());
   const uint8_t data[] = {1, 2, 3, 4};
-  ArrayView<const uint8_t> packet(data, sizeof(data));
+  std::span<const uint8_t> packet(data, sizeof(data));
   RTPHeader header;
 
   delegate->StartShortCircuiting();
@@ -207,7 +207,7 @@ TEST(ChannelReceiveFrameTransformerDelegateTest,
   ASSERT_TRUE(callback);
 
   const uint8_t data[] = {1, 2, 3, 4};
-  ArrayView<const uint8_t> packet(data, sizeof(data));
+  std::span<const uint8_t> packet(data, sizeof(data));
   RTPHeader header;
   std::unique_ptr<TransformableFrameInterface> frame;
   ON_CALL(*mock_frame_transformer, Transform)
@@ -244,7 +244,7 @@ TEST(ChannelReceiveFrameTransformerDelegateTest,
   ASSERT_TRUE(callback);
 
   const uint8_t data[] = {1, 2, 3, 4};
-  ArrayView<const uint8_t> packet(data, sizeof(data));
+  std::span<const uint8_t> packet(data, sizeof(data));
   RTPHeader header;
   uint8_t audio_level_dbov = 67;
   AudioLevel audio_level(/*voice_activity=*/true, audio_level_dbov);
@@ -282,7 +282,7 @@ TEST(ChannelReceiveFrameTransformerDelegateTest,
   ASSERT_TRUE(callback);
 
   const uint8_t data[] = {1, 2, 3, 4};
-  ArrayView<const uint8_t> packet(data, sizeof(data));
+  std::span<const uint8_t> packet(data, sizeof(data));
   Timestamp capture_time = Timestamp::Millis(1234);
   TimeDelta sender_capture_time_offsets[] = {TimeDelta::Millis(56),
                                              TimeDelta::Millis(-79)};
@@ -320,7 +320,7 @@ TEST(ChannelReceiveFrameTransformerDelegateTest, SetAudioLevel) {
           Thread::Current());
   delegate->Init();
   const uint8_t data[] = {1, 2, 3, 4};
-  ArrayView<const uint8_t> packet(data, sizeof(data));
+  std::span<const uint8_t> packet(data, sizeof(data));
   std::unique_ptr<TransformableFrameInterface> frame;
   ON_CALL(*mock_frame_transformer, Transform)
       .WillByDefault(
diff --git a/audio/channel_receive_unittest.cc b/audio/channel_receive_unittest.cc
index 72e45c3958..83cc6d56e0 100644
--- a/audio/channel_receive_unittest.cc
+++ b/audio/channel_receive_unittest.cc
@@ -14,10 +14,10 @@
 #include <cstdint>
 #include <cstring>
 #include <memory>
+#include <span>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio/audio_frame.h"
 #include "api/audio_codecs/audio_decoder_factory.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
@@ -142,7 +142,7 @@ class ChannelReceiveTest : public Test {
   }
 
   void HandleGeneratedRtcp(ChannelReceiveInterface& /* channel */,
-                           ArrayView<const uint8_t> packet) {
+                           std::span<const uint8_t> packet) {
     if (packet[1] == rtcp::ReceiverReport::kPacketType) {
       // Ignore RR, it requires no response
     } else {
@@ -185,7 +185,7 @@ TEST_F(ChannelReceiveTest, ReceiveReportGeneratedOnTime) {
 
   bool receiver_report_sent = false;
   EXPECT_CALL(transport_, SendRtcp)
-      .WillRepeatedly([&](ArrayView<const uint8_t> packet,
+      .WillRepeatedly([&](std::span<const uint8_t> packet,
                           const PacketOptions& options) {
         if (packet.size() >= 2 &&
             packet[1] == rtcp::ReceiverReport::kPacketType) {
@@ -205,7 +205,7 @@ TEST_F(ChannelReceiveTest, CaptureStartTimeBecomesValid) {
 
   EXPECT_CALL(transport_, SendRtcp)
       .WillRepeatedly(
-          [&](ArrayView<const uint8_t> packet, const PacketOptions& options) {
+          [&](std::span<const uint8_t> packet, const PacketOptions& options) {
             HandleGeneratedRtcp(*channel, packet);
             return true;
           });
diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index 68fd2f62d9..528bd26851 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -16,13 +16,13 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/call/bitrate_allocation.h"
@@ -165,7 +165,7 @@ class ChannelSend : public ChannelSendInterface,
   void SetInputMute(bool enable) override;
 
   // CSRCs.
-  void SetCsrcs(ArrayView<const uint32_t> csrcs) override;
+  void SetCsrcs(std::span<const uint32_t> csrcs) override;
 
   // Stats.
   ANAStats GetANAStatistics() const override;
@@ -243,9 +243,9 @@ class ChannelSend : public ChannelSendInterface,
   int32_t SendRtpAudio(AudioFrameType frameType,
                        uint8_t payloadType,
                        uint32_t rtp_timestamp_without_offset,
-                       ArrayView<const uint8_t> payload,
+                       std::span<const uint8_t> payload,
                        int64_t absolute_capture_timestamp_ms,
-                       ArrayView<const uint32_t> csrcs,
+                       std::span<const uint32_t> csrcs,
                        std::optional<uint8_t> audio_level_dbov)
       RTC_RUN_ON(worker_thread_);
 
@@ -418,9 +418,9 @@ int32_t ChannelSend::SendData(AudioFrameType frameType,
 int32_t ChannelSend::SendRtpAudio(AudioFrameType frameType,
                                   uint8_t payloadType,
                                   uint32_t rtp_timestamp_without_offset,
-                                  ArrayView<const uint8_t> payload,
+                                  std::span<const uint8_t> payload,
                                   int64_t absolute_capture_timestamp_ms,
-                                  ArrayView<const uint32_t> csrcs,
+                                  std::span<const uint32_t> csrcs,
                                   std::optional<uint8_t> audio_level_dbov) {
   // E2EE Custom Audio Frame Encryption (This is optional).
   // Keep this buffer around for the lifetime of the send call.
@@ -695,7 +695,7 @@ void ChannelSend::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
   RTC_DCHECK_RUN_ON(worker_thread_);
 
   // Deliver RTCP packet to RTP/RTCP module for parsing
-  rtp_rtcp_->IncomingRtcpPacket(MakeArrayView(data, length));
+  rtp_rtcp_->IncomingRtcpPacket(std::span(data, length));
 
   std::optional<TimeDelta> rtt = rtp_rtcp_->LastRtt();
   if (!rtt.has_value()) {
@@ -721,7 +721,7 @@ bool ChannelSend::InputMute() const {
   return input_mute_;
 }
 
-void ChannelSend::SetCsrcs(ArrayView<const uint32_t> csrcs) {
+void ChannelSend::SetCsrcs(std::span<const uint32_t> csrcs) {
   RTC_DCHECK_RUN_ON(worker_thread_);
   std::vector<uint32_t> csrcs_copy(
       csrcs.begin(),
@@ -919,7 +919,7 @@ void ChannelSend::ProcessAndEncodeAudio(
             rms_level_.AnalyzeMuted(length);
           } else {
             rms_level_.Analyze(
-                ArrayView<const int16_t>(audio_frame->data(), length));
+                std::span<const int16_t>(audio_frame->data(), length));
           }
         }
         previous_frame_muted_ = is_muted;
@@ -984,9 +984,9 @@ void ChannelSend::InitFrameTransformerDelegate(
   ChannelSendFrameTransformerDelegate::SendFrameCallback send_audio_callback =
       [this](AudioFrameType frameType, uint8_t payloadType,
              uint32_t rtp_timestamp_with_offset,
-             ArrayView<const uint8_t> payload,
+             std::span<const uint8_t> payload,
              int64_t absolute_capture_timestamp_ms,
-             ArrayView<const uint32_t> csrcs,
+             std::span<const uint32_t> csrcs,
              std::optional<uint8_t> audio_level_dbov) {
         RTC_DCHECK_RUN_ON(worker_thread_);
         return SendRtpAudio(
diff --git a/audio/channel_send.h b/audio/channel_send.h
index 185f3e9975..ab97c3b64e 100644
--- a/audio/channel_send.h
+++ b/audio/channel_send.h
@@ -15,10 +15,10 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio/audio_frame.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/audio_format.h"
@@ -96,7 +96,7 @@ class ChannelSendInterface {
   // Sets the list of CSRCs to be included in the RTP header. If more than
   // kRtpCsrcSize CSRCs are provided, only the first kRtpCsrcSize elements are
   // kept.
-  virtual void SetCsrcs(ArrayView<const uint32_t> csrcs) = 0;
+  virtual void SetCsrcs(std::span<const uint32_t> csrcs) = 0;
 
   virtual void ProcessAndEncodeAudio(
       std::unique_ptr<AudioFrame> audio_frame) = 0;
diff --git a/audio/channel_send_frame_transformer_delegate.cc b/audio/channel_send_frame_transformer_delegate.cc
index 044f1c3d31..57079c3605 100644
--- a/audio/channel_send_frame_transformer_delegate.cc
+++ b/audio/channel_send_frame_transformer_delegate.cc
@@ -14,11 +14,11 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/frame_transformer_interface.h"
 #include "api/scoped_refptr.h"
 #include "api/sequence_checker.h"
@@ -91,8 +91,8 @@ class TransformableOutgoingAudioFrame
         sequence_number_(sequence_number),
         audio_level_dbov_(audio_level_dbov) {}
   ~TransformableOutgoingAudioFrame() override = default;
-  ArrayView<const uint8_t> GetData() const override { return payload_; }
-  void SetData(ArrayView<const uint8_t> data) override {
+  std::span<const uint8_t> GetData() const override { return payload_; }
+  void SetData(std::span<const uint8_t> data) override {
     payload_.SetData(data.data(), data.size());
   }
   uint32_t GetTimestamp() const override { return rtp_timestamp_with_offset_; }
@@ -110,7 +110,7 @@ class TransformableOutgoingAudioFrame
   Direction GetDirection() const override { return Direction::kSender; }
   std::string GetMimeType() const override { return codec_mime_type_; }
 
-  ArrayView<const uint32_t> GetContributingSources() const override {
+  std::span<const uint32_t> GetContributingSources() const override {
     return csrcs_;
   }
 
diff --git a/audio/channel_send_frame_transformer_delegate.h b/audio/channel_send_frame_transformer_delegate.h
index c495af6a69..678f51fcb3 100644
--- a/audio/channel_send_frame_transformer_delegate.h
+++ b/audio/channel_send_frame_transformer_delegate.h
@@ -16,10 +16,10 @@
 #include <functional>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/frame_transformer_interface.h"
 #include "api/scoped_refptr.h"
 #include "api/task_queue/task_queue_base.h"
@@ -40,9 +40,9 @@ class ChannelSendFrameTransformerDelegate : public TransformedFrameCallback {
       std::function<int32_t(AudioFrameType frameType,
                             uint8_t payloadType,
                             uint32_t rtp_timestamp_with_offset,
-                            webrtc::ArrayView<const uint8_t> payload,
+                            std::span<const uint8_t> payload,
                             int64_t absolute_capture_timestamp_ms,
-                            webrtc::ArrayView<const uint32_t> csrcs,
+                            std::span<const uint32_t> csrcs,
                             std::optional<uint8_t> audio_level_dbov)>;
   ChannelSendFrameTransformerDelegate(
       SendFrameCallback send_frame_callback,
diff --git a/audio/channel_send_frame_transformer_delegate_unittest.cc b/audio/channel_send_frame_transformer_delegate_unittest.cc
index 20acb49c6d..d5e19c1827 100644
--- a/audio/channel_send_frame_transformer_delegate_unittest.cc
+++ b/audio/channel_send_frame_transformer_delegate_unittest.cc
@@ -13,11 +13,11 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
 #include "absl/memory/memory.h"
-#include "api/array_view.h"
 #include "api/frame_transformer_interface.h"
 #include "api/make_ref_counted.h"
 #include "api/scoped_refptr.h"
@@ -51,16 +51,16 @@ class MockChannelSend {
               (AudioFrameType frameType,
                uint8_t payloadType,
                uint32_t rtp_timestamp,
-               ArrayView<const uint8_t> payload,
+               std::span<const uint8_t> payload,
                int64_t absolute_capture_timestamp_ms,
-               ArrayView<const uint32_t> csrcs,
+               std::span<const uint32_t> csrcs,
                std::optional<uint8_t> audio_level_dbov));
 
   ChannelSendFrameTransformerDelegate::SendFrameCallback callback() {
     return [this](AudioFrameType frameType, uint8_t payloadType,
-                  uint32_t rtp_timestamp, ArrayView<const uint8_t> payload,
+                  uint32_t rtp_timestamp, std::span<const uint8_t> payload,
                   int64_t absolute_capture_timestamp_ms,
-                  ArrayView<const uint32_t> csrcs,
+                  std::span<const uint32_t> csrcs,
                   std::optional<uint8_t> audio_level_dbov) {
       return SendFrame(frameType, payloadType, rtp_timestamp, payload,
                        absolute_capture_timestamp_ms, csrcs, audio_level_dbov);
@@ -73,7 +73,7 @@ std::unique_ptr<TransformableAudioFrameInterface> CreateMockReceiverFrame(
     std::optional<uint8_t> audio_level_dbov) {
   std::unique_ptr<MockTransformableAudioFrame> mock_frame =
       std::make_unique<NiceMock<MockTransformableAudioFrame>>();
-  ArrayView<const uint8_t> payload(mock_data);
+  std::span<const uint8_t> payload(mock_data);
   ON_CALL(*mock_frame, GetData).WillByDefault(Return(payload));
   ON_CALL(*mock_frame, GetPayloadType).WillByDefault(Return(0));
   ON_CALL(*mock_frame, GetDirection)
diff --git a/audio/channel_send_unittest.cc b/audio/channel_send_unittest.cc
index 4835056070..e777cf6bef 100644
--- a/audio/channel_send_unittest.cc
+++ b/audio/channel_send_unittest.cc
@@ -14,10 +14,10 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/audio_frame.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/audio_encoder_factory.h"
@@ -155,7 +155,7 @@ TEST_F(ChannelSendTest, IncreaseRtpTimestampByPauseDuration) {
   channel_->StartSend();
   uint32_t timestamp;
   int sent_packets = 0;
-  auto send_rtp = [&](ArrayView<const uint8_t> data,
+  auto send_rtp = [&](std::span<const uint8_t> data,
                       const PacketOptions& /* options */) {
     ++sent_packets;
     RtpPacketReceived packet;
@@ -190,7 +190,7 @@ TEST_F(ChannelSendTest, FrameTransformerGetsCorrectTimestamp) {
   EXPECT_CALL(*mock_frame_transformer, UnregisterTransformedFrameCallback);
 
   std::optional<uint32_t> sent_timestamp;
-  auto send_rtp = [&](ArrayView<const uint8_t> data,
+  auto send_rtp = [&](std::span<const uint8_t> data,
                       const PacketOptions& /* options */) {
     RtpPacketReceived packet;
     packet.Parse(data);
@@ -239,7 +239,7 @@ TEST_F(ChannelSendTest, AudioLevelsAttachedToCorrectTransformedFrame) {
   EXPECT_CALL(*mock_frame_transformer, UnregisterTransformedFrameCallback);
 
   std::vector<uint8_t> sent_audio_levels;
-  auto send_rtp = [&](ArrayView<const uint8_t> data,
+  auto send_rtp = [&](std::span<const uint8_t> data,
                       const PacketOptions& /* options */) {
     RtpPacketReceived packet(&extension_manager);
     packet.Parse(data);
@@ -303,7 +303,7 @@ TEST_F(ChannelSendTest, AudioLevelsAttachedToInsertedTransformedFrame) {
   EXPECT_CALL(*mock_frame_transformer, UnregisterTransformedFrameCallback);
 
   std::optional<uint8_t> sent_audio_level;
-  auto send_rtp = [&](ArrayView<const uint8_t> data,
+  auto send_rtp = [&](std::span<const uint8_t> data,
                       const PacketOptions& /* options */) {
     RtpPacketReceived packet(&extension_manager);
     packet.Parse(data);
@@ -323,7 +323,7 @@ TEST_F(ChannelSendTest, AudioLevelsAttachedToInsertedTransformedFrame) {
   ON_CALL(*mock_frame, AudioLevel()).WillByDefault(Return(audio_level));
   uint8_t payload[10];
   ON_CALL(*mock_frame, GetData())
-      .WillByDefault(Return(ArrayView<uint8_t>(&payload[0], 10)));
+      .WillByDefault(Return(std::span<uint8_t>(&payload[0], 10)));
   EXPECT_THAT(WaitUntil([&] { return callback; }, IsTrue()), IsRtcOk());
   callback->OnTransformedFrame(std::move(mock_frame));
 
@@ -415,7 +415,7 @@ TEST_F(ChannelSendTest, ConfiguredCsrcsAreIncludedInRtpPackets) {
   channel_->SetCsrcs(expected_csrcs);
 
   std::vector<uint32_t> csrcs;
-  auto send_rtp = [&](ArrayView<const uint8_t> data,
+  auto send_rtp = [&](std::span<const uint8_t> data,
                       const PacketOptions& /* options */) {
     RtpPacketReceived packet;
     packet.Parse(data);
@@ -500,7 +500,7 @@ TEST_F(ChannelSendTest, FrameTransformerTakesPrecedenceOverSetCsrcs) {
   channel_->StartSend();
 
   std::vector<uint32_t> sent_csrcs;
-  auto send_rtp = [&](ArrayView<const uint8_t> data,
+  auto send_rtp = [&](std::span<const uint8_t> data,
                       const PacketOptions& /* options */) {
     RtpPacketReceived packet;
     packet.Parse(data);
diff --git a/audio/mock_voe_channel_proxy.h b/audio/mock_voe_channel_proxy.h
index db4d73adaf..c460bb119b 100644
--- a/audio/mock_voe_channel_proxy.h
+++ b/audio/mock_voe_channel_proxy.h
@@ -16,11 +16,11 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio/audio_frame.h"
 #include "api/audio/audio_mixer.h"
 #include "api/audio_codecs/audio_encoder.h"
@@ -208,7 +208,7 @@ class MockChannelSend : public voe::ChannelSendInterface {
               RegisterPacketOverhead,
               (int packet_byte_overhead),
               (override));
-  MOCK_METHOD(void, SetCsrcs, (ArrayView<const uint32_t> csrcs), (override));
+  MOCK_METHOD(void, SetCsrcs, (std::span<const uint32_t> csrcs), (override));
 };
 }  // namespace test
 }  // namespace webrtc
diff --git a/audio/voip/BUILD.gn b/audio/voip/BUILD.gn
index 5780fd2c92..dd5e127e13 100644
--- a/audio/voip/BUILD.gn
+++ b/audio/voip/BUILD.gn
@@ -16,7 +16,6 @@ rtc_library("voip_core") {
   deps = [
     ":audio_channel",
     "..:audio",
-    "../../api:array_view",
     "../../api:make_ref_counted",
     "../../api:scoped_refptr",
     "../../api:transport_api",
@@ -43,7 +42,6 @@ rtc_library("audio_channel") {
   deps = [
     ":audio_egress",
     ":audio_ingress",
-    "../../api:array_view",
     "../../api:ref_count",
     "../../api:rtp_headers",
     "../../api:scoped_refptr",
@@ -67,7 +65,6 @@ rtc_library("audio_ingress") {
   ]
   deps = [
     "..:audio",
-    "../../api:array_view",
     "../../api:scoped_refptr",
     "../../api/audio:audio_mixer_api",
     "../../api/audio_codecs:audio_codecs_api",
@@ -97,7 +94,6 @@ rtc_library("audio_egress") {
   ]
   deps = [
     "..:audio",
-    "../../api:array_view",
     "../../api:sequence_checker",
     "../../api/audio:audio_frame_api",
     "../../api/audio_codecs:audio_codecs_api",
diff --git a/audio/voip/audio_channel.h b/audio/voip/audio_channel.h
index b670b310f3..268e15eb76 100644
--- a/audio/voip/audio_channel.h
+++ b/audio/voip/audio_channel.h
@@ -15,9 +15,9 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 
-#include "api/array_view.h"
 #include "api/audio/audio_mixer.h"
 #include "api/audio_codecs/audio_decoder_factory.h"
 #include "api/audio_codecs/audio_encoder.h"
@@ -83,10 +83,10 @@ class AudioChannel : public RefCountInterface {
 
   // APIs relayed to AudioIngress.
   bool IsPlaying() const { return ingress_->IsPlaying(); }
-  void ReceivedRTPPacket(ArrayView<const uint8_t> rtp_packet) {
+  void ReceivedRTPPacket(std::span<const uint8_t> rtp_packet) {
     ingress_->ReceivedRTPPacket(rtp_packet);
   }
-  void ReceivedRTCPPacket(ArrayView<const uint8_t> rtcp_packet) {
+  void ReceivedRTCPPacket(std::span<const uint8_t> rtcp_packet) {
     ingress_->ReceivedRTCPPacket(rtcp_packet);
   }
   void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs) {
diff --git a/audio/voip/audio_egress.cc b/audio/voip/audio_egress.cc
index fc69d2b6d7..36ffe21d44 100644
--- a/audio/voip/audio_egress.cc
+++ b/audio/voip/audio_egress.cc
@@ -13,9 +13,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <utility>
 
-#include "api/array_view.h"
 #include "api/audio/audio_frame.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/audio_format.h"
@@ -132,7 +132,7 @@ int32_t AudioEgress::SendData(AudioFrameType frame_type,
                               size_t payload_size) {
   RTC_DCHECK_RUN_ON(&encoder_queue_checker_);
 
-  ArrayView<const uint8_t> payload(payload_data, payload_size);
+  std::span<const uint8_t> payload(payload_data, payload_size);
 
   // Currently we don't get a capture time from downstream modules (ADM,
   // AudioTransportImpl).
diff --git a/audio/voip/audio_ingress.cc b/audio/voip/audio_ingress.cc
index b4525a18cd..1e2fa34838 100644
--- a/audio/voip/audio_ingress.cc
+++ b/audio/voip/audio_ingress.cc
@@ -14,10 +14,10 @@
 #include <ctime>
 #include <map>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/audio_mixer.h"
 #include "api/audio_codecs/audio_decoder_factory.h"
 #include "api/audio_codecs/audio_format.h"
@@ -160,7 +160,7 @@ void AudioIngress::SetReceiveCodecs(
   neteq_->SetCodecs(codecs);
 }
 
-void AudioIngress::ReceivedRTPPacket(ArrayView<const uint8_t> rtp_packet) {
+void AudioIngress::ReceivedRTPPacket(std::span<const uint8_t> rtp_packet) {
   RtpPacketReceived rtp_packet_received;
   rtp_packet_received.Parse(rtp_packet.data(), rtp_packet.size());
 
@@ -204,7 +204,7 @@ void AudioIngress::ReceivedRTPPacket(ArrayView<const uint8_t> rtp_packet) {
   const uint8_t* payload = rtp_packet_received.data() + header.headerLength;
   size_t payload_length = packet_length - header.headerLength;
   size_t payload_data_length = payload_length - header.paddingLength;
-  auto data_view = ArrayView<const uint8_t>(payload, payload_data_length);
+  auto data_view = std::span<const uint8_t>(payload, payload_data_length);
 
   // Push the incoming payload (parsed and ready for decoding) into NetEq.
   if (!data_view.empty()) {
@@ -217,7 +217,7 @@ void AudioIngress::ReceivedRTPPacket(ArrayView<const uint8_t> rtp_packet) {
   }
 }
 
-void AudioIngress::ReceivedRTCPPacket(ArrayView<const uint8_t> rtcp_packet) {
+void AudioIngress::ReceivedRTCPPacket(std::span<const uint8_t> rtcp_packet) {
   rtcp::CommonHeader rtcp_header;
   if (rtcp_header.Parse(rtcp_packet.data(), rtcp_packet.size()) &&
       (rtcp_header.type() == rtcp::SenderReport::kPacketType ||
diff --git a/audio/voip/audio_ingress.h b/audio/voip/audio_ingress.h
index e6d9836fd4..a8ec34f02d 100644
--- a/audio/voip/audio_ingress.h
+++ b/audio/voip/audio_ingress.h
@@ -17,8 +17,8 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/audio/audio_mixer.h"
 #include "api/audio_codecs/audio_decoder_factory.h"
 #include "api/audio_codecs/audio_format.h"
@@ -72,8 +72,8 @@ class AudioIngress : public AudioMixer::Source {
   void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs);
 
   // APIs to handle received RTP/RTCP packets from caller.
-  void ReceivedRTPPacket(ArrayView<const uint8_t> rtp_packet);
-  void ReceivedRTCPPacket(ArrayView<const uint8_t> rtcp_packet);
+  void ReceivedRTPPacket(std::span<const uint8_t> rtp_packet);
+  void ReceivedRTCPPacket(std::span<const uint8_t> rtcp_packet);
 
   // See comments on LevelFullRange, TotalEnergy, TotalDuration from
   // audio/audio_level.h.
diff --git a/audio/voip/test/BUILD.gn b/audio/voip/test/BUILD.gn
index c0efeb6c8a..0c1acf0d68 100644
--- a/audio/voip/test/BUILD.gn
+++ b/audio/voip/test/BUILD.gn
@@ -49,7 +49,6 @@ if (rtc_include_tests) {
     deps = [
       ":mock_task_queue",
       "..:audio_channel",
-      "../../../api:array_view",
       "../../../api:make_ref_counted",
       "../../../api:scoped_refptr",
       "../../../api/audio:audio_frame_api",
@@ -76,7 +75,6 @@ if (rtc_include_tests) {
     deps = [
       "..:audio_egress",
       "..:audio_ingress",
-      "../../../api:array_view",
       "../../../api:rtp_headers",
       "../../../api:scoped_refptr",
       "../../../api/audio:audio_frame_api",
@@ -102,7 +100,6 @@ if (rtc_include_tests) {
     sources = [ "audio_egress_unittest.cc" ]
     deps = [
       "..:audio_egress",
-      "../../../api:array_view",
       "../../../api:rtp_headers",
       "../../../api:scoped_refptr",
       "../../../api:transport_api",
diff --git a/audio/voip/test/audio_channel_unittest.cc b/audio/voip/test/audio_channel_unittest.cc
index 8b4c32f273..a707e2d5ad 100644
--- a/audio/voip/test/audio_channel_unittest.cc
+++ b/audio/voip/test/audio_channel_unittest.cc
@@ -14,10 +14,10 @@
 #include <cstring>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 
 #include "absl/functional/any_invocable.h"
-#include "api/array_view.h"
 #include "api/audio/audio_frame.h"
 #include "api/audio/audio_mixer.h"
 #include "api/audio_codecs/audio_decoder_factory.h"
@@ -120,7 +120,7 @@ class AudioChannelTest : public ::testing::Test {
 // Resulted RTP packet is looped back into AudioChannel and gets decoded into
 // audio frame to see if it has some signal to indicate its validity.
 TEST_F(AudioChannelTest, PlayRtpByLocalLoop) {
-  auto loop_rtp = [&](ArrayView<const uint8_t> packet, Unused) {
+  auto loop_rtp = [&](std::span<const uint8_t> packet, Unused) {
     audio_channel_->ReceivedRTPPacket(packet);
     return true;
   };
@@ -145,7 +145,7 @@ TEST_F(AudioChannelTest, PlayRtpByLocalLoop) {
 // Validate assigned local SSRC is resulted in RTP packet.
 TEST_F(AudioChannelTest, VerifyLocalSsrcAsAssigned) {
   RtpPacketReceived rtp;
-  auto loop_rtp = [&](ArrayView<const uint8_t> packet, Unused) {
+  auto loop_rtp = [&](std::span<const uint8_t> packet, Unused) {
     rtp.Parse(packet);
     return true;
   };
@@ -160,7 +160,7 @@ TEST_F(AudioChannelTest, VerifyLocalSsrcAsAssigned) {
 
 // Check metrics after processing an RTP packet.
 TEST_F(AudioChannelTest, TestIngressStatistics) {
-  auto loop_rtp = [&](ArrayView<const uint8_t> packet, Unused) {
+  auto loop_rtp = [&](std::span<const uint8_t> packet, Unused) {
     audio_channel_->ReceivedRTPPacket(packet);
     return true;
   };
@@ -237,11 +237,11 @@ TEST_F(AudioChannelTest, TestIngressStatistics) {
 
 // Check ChannelStatistics metric after processing RTP and RTCP packets.
 TEST_F(AudioChannelTest, TestChannelStatistics) {
-  auto loop_rtp = [&](ArrayView<const uint8_t> packet, Unused) {
+  auto loop_rtp = [&](std::span<const uint8_t> packet, Unused) {
     audio_channel_->ReceivedRTPPacket(packet);
     return true;
   };
-  auto loop_rtcp = [&](ArrayView<const uint8_t> packet, Unused) {
+  auto loop_rtcp = [&](std::span<const uint8_t> packet, Unused) {
     audio_channel_->ReceivedRTCPPacket(packet);
     return true;
   };
@@ -306,7 +306,7 @@ TEST_F(AudioChannelTest, RttIsAvailableAfterChangeOfRemoteSsrc) {
   auto send_recv_rtp = [&](scoped_refptr<AudioChannel> rtp_sender,
                            scoped_refptr<AudioChannel> rtp_receiver) {
     // Setup routing logic via transport_.
-    auto route_rtp = [&](ArrayView<const uint8_t> packet, Unused) {
+    auto route_rtp = [&](std::span<const uint8_t> packet, Unused) {
       rtp_receiver->ReceivedRTPPacket(packet);
       return true;
     };
@@ -328,7 +328,7 @@ TEST_F(AudioChannelTest, RttIsAvailableAfterChangeOfRemoteSsrc) {
   auto send_recv_rtcp = [&](scoped_refptr<AudioChannel> rtcp_sender,
                             scoped_refptr<AudioChannel> rtcp_receiver) {
     // Setup routing logic via transport_.
-    auto route_rtcp = [&](ArrayView<const uint8_t> packet, Unused) {
+    auto route_rtcp = [&](std::span<const uint8_t> packet, Unused) {
       rtcp_receiver->ReceivedRTCPPacket(packet);
       return true;
     };
diff --git a/audio/voip/test/audio_egress_unittest.cc b/audio/voip/test/audio_egress_unittest.cc
index a086ab9e04..5cf9a2aa43 100644
--- a/audio/voip/test/audio_egress_unittest.cc
+++ b/audio/voip/test/audio_egress_unittest.cc
@@ -14,8 +14,8 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/audio/audio_frame.h"
 #include "api/audio_codecs/audio_encoder_factory.h"
 #include "api/audio_codecs/audio_format.h"
@@ -131,7 +131,7 @@ TEST_F(AudioEgressTest, ProcessAudioWithMute) {
   Event event;
   int rtp_count = 0;
   RtpPacketReceived rtp;
-  auto rtp_sent = [&](ArrayView<const uint8_t> packet, Unused) {
+  auto rtp_sent = [&](std::span<const uint8_t> packet, Unused) {
     rtp.Parse(packet);
     if (++rtp_count == kExpected) {
       event.Set();
@@ -169,7 +169,7 @@ TEST_F(AudioEgressTest, ProcessAudioWithSineWave) {
   Event event;
   int rtp_count = 0;
   RtpPacketReceived rtp;
-  auto rtp_sent = [&](ArrayView<const uint8_t> packet, Unused) {
+  auto rtp_sent = [&](std::span<const uint8_t> packet, Unused) {
     rtp.Parse(packet);
     if (++rtp_count == kExpected) {
       event.Set();
@@ -204,7 +204,7 @@ TEST_F(AudioEgressTest, SkipAudioEncodingAfterStopSend) {
   constexpr int kExpected = 10;
   Event event;
   int rtp_count = 0;
-  auto rtp_sent = [&](ArrayView<const uint8_t> /* packet */, Unused) {
+  auto rtp_sent = [&](std::span<const uint8_t> /* packet */, Unused) {
     if (++rtp_count == kExpected) {
       event.Set();
     }
@@ -278,7 +278,7 @@ TEST_F(AudioEgressTest, SendDTMF) {
   // It's possible that we may have actual audio RTP packets along with
   // DTMF packtets.  We are only interested in the exact number of DTMF
   // packets rtp stack is emitting.
-  auto rtp_sent = [&](ArrayView<const uint8_t> packet, Unused) {
+  auto rtp_sent = [&](std::span<const uint8_t> packet, Unused) {
     RtpPacketReceived rtp;
     rtp.Parse(packet);
     if (is_dtmf(rtp) && ++dtmf_count == kExpected) {
@@ -305,7 +305,7 @@ TEST_F(AudioEgressTest, TestAudioInputLevelAndEnergyDuration) {
   constexpr int kExpected = 6;
   Event event;
   int rtp_count = 0;
-  auto rtp_sent = [&](ArrayView<const uint8_t> /* packet */, Unused) {
+  auto rtp_sent = [&](std::span<const uint8_t> /* packet */, Unused) {
     if (++rtp_count == kExpected) {
       event.Set();
     }
diff --git a/audio/voip/test/audio_ingress_unittest.cc b/audio/voip/test/audio_ingress_unittest.cc
index 477e653e59..23e39aefe5 100644
--- a/audio/voip/test/audio_ingress_unittest.cc
+++ b/audio/voip/test/audio_ingress_unittest.cc
@@ -13,8 +13,8 @@
 #include <cstddef>
 #include <cstdint>
 #include <memory>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/audio/audio_frame.h"
 #include "api/audio/audio_mixer.h"
 #include "api/audio_codecs/audio_decoder_factory.h"
@@ -125,7 +125,7 @@ TEST_F(AudioIngressTest, PlayingAfterStartAndStop) {
 
 TEST_F(AudioIngressTest, GetAudioFrameAfterRtpReceived) {
   Event event;
-  auto handle_rtp = [&](ArrayView<const uint8_t> packet, Unused) {
+  auto handle_rtp = [&](std::span<const uint8_t> packet, Unused) {
     ingress_->ReceivedRTPPacket(packet);
     event.Set();
     return true;
@@ -155,7 +155,7 @@ TEST_F(AudioIngressTest, TestSpeechOutputLevelAndEnergyDuration) {
   constexpr int kNumRtp = 6;
   int rtp_count = 0;
   Event event;
-  auto handle_rtp = [&](ArrayView<const uint8_t> packet, Unused) {
+  auto handle_rtp = [&](std::span<const uint8_t> packet, Unused) {
     ingress_->ReceivedRTPPacket(packet);
     if (++rtp_count == kNumRtp) {
       event.Set();
@@ -186,7 +186,7 @@ TEST_F(AudioIngressTest, TestSpeechOutputLevelAndEnergyDuration) {
 
 TEST_F(AudioIngressTest, PreferredSampleRate) {
   Event event;
-  auto handle_rtp = [&](ArrayView<const uint8_t> packet, Unused) {
+  auto handle_rtp = [&](std::span<const uint8_t> packet, Unused) {
     ingress_->ReceivedRTPPacket(packet);
     event.Set();
     return true;
@@ -215,7 +215,7 @@ TEST_F(AudioIngressTest, GetMutedAudioFrameAfterRtpReceivedAndStopPlay) {
   constexpr int kNumRtp = 6;
   int rtp_count = 0;
   Event event;
-  auto handle_rtp = [&](ArrayView<const uint8_t> packet, Unused) {
+  auto handle_rtp = [&](std::span<const uint8_t> packet, Unused) {
     ingress_->ReceivedRTPPacket(packet);
     if (++rtp_count == kNumRtp) {
       event.Set();
diff --git a/audio/voip/voip_core.cc b/audio/voip/voip_core.cc
index d3ce340ac9..a5d5fc3d9f 100644
--- a/audio/voip/voip_core.cc
+++ b/audio/voip/voip_core.cc
@@ -16,10 +16,10 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/audio_device.h"
 #include "api/audio/audio_processing.h"
 #include "api/audio_codecs/audio_decoder_factory.h"
@@ -367,7 +367,7 @@ VoipResult VoipCore::StopPlayout(ChannelId channel_id) {
 }
 
 VoipResult VoipCore::ReceivedRTPPacket(ChannelId channel_id,
-                                       ArrayView<const uint8_t> rtp_packet) {
+                                       std::span<const uint8_t> rtp_packet) {
   scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
 
   if (!channel) {
@@ -380,7 +380,7 @@ VoipResult VoipCore::ReceivedRTPPacket(ChannelId channel_id,
 }
 
 VoipResult VoipCore::ReceivedRTCPPacket(ChannelId channel_id,
-                                        ArrayView<const uint8_t> rtcp_packet) {
+                                        std::span<const uint8_t> rtcp_packet) {
   scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
 
   if (!channel) {
diff --git a/audio/voip/voip_core.h b/audio/voip/voip_core.h
index dbd9150be9..45eca080c8 100644
--- a/audio/voip/voip_core.h
+++ b/audio/voip/voip_core.h
@@ -15,9 +15,9 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <unordered_map>
 
-#include "api/array_view.h"
 #include "api/audio/audio_device.h"
 #include "api/audio/audio_mixer.h"
 #include "api/audio/audio_processing.h"
@@ -81,9 +81,9 @@ class VoipCore : public VoipEngine,
 
   // Implements VoipNetwork interfaces.
   VoipResult ReceivedRTPPacket(ChannelId channel_id,
-                               ArrayView<const uint8_t> rtp_packet) override;
+                               std::span<const uint8_t> rtp_packet) override;
   VoipResult ReceivedRTCPPacket(ChannelId channel_id,
-                                ArrayView<const uint8_t> rtcp_packet) override;
+                                std::span<const uint8_t> rtcp_packet) override;
 
   // Implements VoipCodec interfaces.
   VoipResult SetSendCodec(ChannelId channel_id,
diff --git a/call/BUILD.gn b/call/BUILD.gn
index b11616353c..a8ad708290 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -109,7 +109,6 @@ rtc_library("rtp_interfaces") {
     "rtp_transport_controller_send_interface.h",
   ]
   deps = [
-    "../api:array_view",
     "../api:fec_controller_api",
     "../api:frame_transformer_interface",
     "../api:network_state_predictor_api",
@@ -148,7 +147,6 @@ rtc_library("rtp_receiver") {
   ]
   deps = [
     ":rtp_interfaces",
-    "../api:array_view",
     "../api:sequence_checker",
     "../api/environment",
     "../logging:rtc_event_rtp_rtcp",
@@ -178,7 +176,6 @@ rtc_library("rtp_sender") {
   deps = [
     ":bitrate_configurator",
     ":rtp_interfaces",
-    "../api:array_view",
     "../api:bitrate_allocation",
     "../api:fec_controller_api",
     "../api:field_trials_view",
@@ -306,7 +303,6 @@ rtc_library("call") {
     ":version",
     ":video_receive_stream_api",
     ":video_send_stream_api",
-    "../api:array_view",
     "../api:fec_controller_api",
     "../api:field_trials_view",
     "../api:rtp_headers",
@@ -407,7 +403,6 @@ rtc_library("video_send_stream_api") {
   ]
   deps = [
     ":rtp_interfaces",
-    "../api:array_view",
     "../api:frame_transformer_interface",
     "../api:rtp_parameters",
     "../api:rtp_sender_interface",
@@ -474,7 +469,6 @@ rtc_library("fake_network") {
   ]
   deps = [
     ":simulated_packet_receiver",
-    "../api:array_view",
     "../api:rtp_parameters",
     "../api:simulated_network_api",
     "../api:transport_api",
@@ -521,7 +515,6 @@ if (rtc_include_tests) {
         ":rtp_sender",
         ":video_receive_stream_api",
         ":video_send_stream_api",
-        "../api:array_view",
         "../api:bitrate_allocation",
         "../api:create_frame_generator",
         "../api:field_trials",
@@ -625,7 +618,6 @@ if (rtc_include_tests) {
         ":fake_network",
         ":video_receive_stream_api",
         ":video_send_stream_api",
-        "../api:array_view",
         "../api:field_trials_view",
         "../api:make_ref_counted",
         "../api:rtc_event_log_output_file",
diff --git a/call/call.cc b/call/call.cc
index a7fd3e3142..df8786c886 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -18,6 +18,7 @@
 #include <memory>
 #include <optional>
 #include <set>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
@@ -25,7 +26,6 @@
 #include "absl/functional/bind_front.h"
 #include "absl/strings/string_view.h"
 #include "api/adaptation/resource.h"
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/fec_controller.h"
 #include "api/media_types.h"
@@ -1343,7 +1343,7 @@ void Call::DeliverRtcpPacket(CopyOnWriteBuffer packet) {
 
   receive_stats_.AddReceivedRtcpBytes(static_cast<int>(packet.size()));
   bool rtcp_delivered = false;
-  ArrayView<const uint8_t> packet_view(packet.cdata(), packet.size());
+  std::span<const uint8_t> packet_view(packet.cdata(), packet.size());
   for (VideoReceiveStream2* stream : video_receive_streams_) {
     if (stream->DeliverRtcp(packet_view))
       rtcp_delivered = true;
diff --git a/call/call_perf_tests.cc b/call/call_perf_tests.cc
index 9cd29d8d1c..2eae4f4290 100644
--- a/call/call_perf_tests.cc
+++ b/call/call_perf_tests.cc
@@ -16,13 +16,13 @@
 #include <limits>
 #include <map>
 #include <memory>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/flags/flag.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio/audio_device.h"
 #include "api/audio/builtin_audio_processing_builder.h"
 #include "api/audio_codecs/builtin_audio_encoder_factory.h"
@@ -566,7 +566,7 @@ void CallPerfTest::TestMinTransmitBitrate(bool pad_to_min_bitrate) {
 
    private:
     // TODO(holmer): Run this with a timer instead of once per packet.
-    Action OnSendRtp(ArrayView<const uint8_t> /* packet */) override {
+    Action OnSendRtp(std::span<const uint8_t> /* packet */) override {
       task_queue_->PostTask(SafeTask(task_safety_flag_, [this]() {
         VideoSendStream::Stats stats = send_stream_->GetStats();
 
@@ -1021,7 +1021,7 @@ void CallPerfTest::TestEncodeFramerate(VideoEncoderFactory* encoder_factory,
       }
     }
 
-    Action OnSendRtp(ArrayView<const uint8_t> /* packet */) override {
+    Action OnSendRtp(std::span<const uint8_t> /* packet */) override {
       const Timestamp now = clock_->CurrentTime();
       if (now - last_getstats_time_ > kMinGetStatsInterval) {
         last_getstats_time_ = now;
diff --git a/call/fake_network_pipe.cc b/call/fake_network_pipe.cc
index 4504b6c704..a816593728 100644
--- a/call/fake_network_pipe.cc
+++ b/call/fake_network_pipe.cc
@@ -16,10 +16,10 @@
 #include <memory>
 #include <optional>
 #include <queue>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/call/transport.h"
 #include "api/media_types.h"
 #include "api/test/simulated_network.h"
@@ -142,7 +142,7 @@ void FakeNetworkPipe::RemoveActiveTransport(Transport* transport) {
   }
 }
 
-bool FakeNetworkPipe::SendRtp(ArrayView<const uint8_t> packet,
+bool FakeNetworkPipe::SendRtp(std::span<const uint8_t> packet,
                               const PacketOptions& options,
                               Transport* transport) {
   RTC_DCHECK(transport);
@@ -150,7 +150,7 @@ bool FakeNetworkPipe::SendRtp(ArrayView<const uint8_t> packet,
   return true;
 }
 
-bool FakeNetworkPipe::SendRtcp(ArrayView<const uint8_t> packet,
+bool FakeNetworkPipe::SendRtcp(std::span<const uint8_t> packet,
                                Transport* transport) {
   RTC_DCHECK(transport);
   EnqueuePacket(CopyOnWriteBuffer(packet), std::nullopt, true, transport);
@@ -322,10 +322,10 @@ void FakeNetworkPipe::DeliverNetworkPacket(NetworkPacket* packet) {
       return;
     }
     if (packet->is_rtcp()) {
-      transport->SendRtcp(MakeArrayView(packet->data(), packet->data_length()),
+      transport->SendRtcp(std::span(packet->data(), packet->data_length()),
                           packet->packet_options());
     } else {
-      transport->SendRtp(MakeArrayView(packet->data(), packet->data_length()),
+      transport->SendRtp(std::span(packet->data(), packet->data_length()),
                          packet->packet_options());
     }
   } else if (receiver_) {
diff --git a/call/fake_network_pipe.h b/call/fake_network_pipe.h
index a30c50eecb..168f8afffa 100644
--- a/call/fake_network_pipe.h
+++ b/call/fake_network_pipe.h
@@ -17,8 +17,8 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/call/transport.h"
 #include "api/test/simulated_network.h"
 #include "call/simulated_packet_receiver.h"
@@ -133,10 +133,10 @@ class FakeNetworkPipe : public SimulatedPacketReceiverInterface {
   // Methods for use with Transport interface. When/if packets are delivered,
   // they will be passed to the instance specified by the `transport` parameter.
   // Note that that instance must be in the map of active transports.
-  bool SendRtp(ArrayView<const uint8_t> packet,
+  bool SendRtp(std::span<const uint8_t> packet,
                const PacketOptions& options,
                Transport* transport);
-  bool SendRtcp(ArrayView<const uint8_t> packet, Transport* transport);
+  bool SendRtcp(std::span<const uint8_t> packet, Transport* transport);
 
   // Implements the PacketReceiver interface. When/if packets are delivered,
   // they will be passed directly to the receiver instance given in
diff --git a/call/flexfec_receive_stream_impl.cc b/call/flexfec_receive_stream_impl.cc
index 8b16d4cbc8..d694d06f14 100644
--- a/call/flexfec_receive_stream_impl.cc
+++ b/call/flexfec_receive_stream_impl.cc
@@ -14,7 +14,6 @@
 #include <memory>
 #include <string>
 
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/sequence_checker.h"
 #include "call/flexfec_receive_stream.h"
diff --git a/call/flexfec_receive_stream_unittest.cc b/call/flexfec_receive_stream_unittest.cc
index be769db5d1..27c8d2c285 100644
--- a/call/flexfec_receive_stream_unittest.cc
+++ b/call/flexfec_receive_stream_unittest.cc
@@ -12,9 +12,9 @@
 
 #include <cstdint>
 #include <memory>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/call/transport.h"
 #include "api/environment/environment_factory.h"
 #include "api/rtp_headers.h"
@@ -52,7 +52,7 @@ FlexfecReceiveStream::Config CreateDefaultConfig(
   return config;
 }
 
-RtpPacketReceived ParsePacket(ArrayView<const uint8_t> packet) {
+RtpPacketReceived ParsePacket(std::span<const uint8_t> packet) {
   RtpPacketReceived parsed_packet(nullptr);
   EXPECT_TRUE(parsed_packet.Parse(packet));
   return parsed_packet;
diff --git a/call/rtp_config.cc b/call/rtp_config.cc
index 115912cd46..f48ae30462 100644
--- a/call/rtp_config.cc
+++ b/call/rtp_config.cc
@@ -19,7 +19,6 @@
 #include <vector>
 
 #include "absl/algorithm/container.h"
-#include "api/array_view.h"
 #include "api/rtp_headers.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/strings/string_builder.h"
diff --git a/call/rtp_transport_controller_send.cc b/call/rtp_transport_controller_send.cc
index cd5fe0e9fc..edc91904d8 100644
--- a/call/rtp_transport_controller_send.cc
+++ b/call/rtp_transport_controller_send.cc
@@ -14,12 +14,12 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/call/transport.h"
 #include "api/fec_controller.h"
 #include "api/frame_transformer_interface.h"
@@ -898,7 +898,7 @@ void RtpTransportControllerSend::PostUpdates(NetworkControlUpdate update) {
 
 void RtpTransportControllerSend::OnReport(
     Timestamp receive_time,
-    ArrayView<const ReportBlockData> report_blocks) {
+    std::span<const ReportBlockData> report_blocks) {
   RTC_DCHECK_RUN_ON(&sequence_checker_);
   if (report_blocks.empty())
     return;
diff --git a/call/rtp_transport_controller_send.h b/call/rtp_transport_controller_send.h
index 26becb0f91..f5d0cdef5f 100644
--- a/call/rtp_transport_controller_send.h
+++ b/call/rtp_transport_controller_send.h
@@ -16,11 +16,11 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/fec_controller.h"
 #include "api/frame_transformer_interface.h"
@@ -129,7 +129,7 @@ class RtpTransportControllerSend final
   void OnReceiverEstimatedMaxBitrate(Timestamp receive_time,
                                      DataRate bitrate) override;
   void OnReport(Timestamp receive_time,
-                ArrayView<const ReportBlockData> report_blocks) override;
+                std::span<const ReportBlockData> report_blocks) override;
   void OnRttUpdate(Timestamp receive_time, TimeDelta rtt) override;
   void OnTransportFeedback(Timestamp receive_time,
                            const rtcp::TransportFeedback& feedback) override;
diff --git a/call/rtp_video_sender.cc b/call/rtp_video_sender.cc
index c25fe194bf..73e8898784 100644
--- a/call/rtp_video_sender.cc
+++ b/call/rtp_video_sender.cc
@@ -17,6 +17,7 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
@@ -24,7 +25,6 @@
 #include "absl/algorithm/container.h"
 #include "absl/base/nullability.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/call/bitrate_allocation.h"
 #include "api/crypto/crypto_options.h"
 #include "api/environment/environment.h"
@@ -719,7 +719,7 @@ DataRate RtpVideoSender::GetPostEncodeOverhead() const {
   return post_encode_overhead;
 }
 
-void RtpVideoSender::DeliverRtcp(ArrayView<const uint8_t> packet) {
+void RtpVideoSender::DeliverRtcp(std::span<const uint8_t> packet) {
   // Runs on a network thread.
   for (const RtpStreamSender& stream : rtp_streams_)
     stream.rtp_rtcp->IncomingRtcpPacket(packet);
@@ -920,7 +920,7 @@ uint32_t RtpVideoSender::GetProtectionBitrateBps() const {
 
 std::vector<RtpSequenceNumberMap::Info> RtpVideoSender::GetSentRtpPacketInfos(
     uint32_t ssrc,
-    ArrayView<const uint16_t> sequence_numbers) const {
+    std::span<const uint16_t> sequence_numbers) const {
   for (const auto& rtp_stream : rtp_streams_) {
     if (ssrc == rtp_stream.rtp_rtcp->SSRC()) {
       return rtp_stream.rtp_rtcp->GetSentRtpPacketInfos(sequence_numbers);
@@ -1015,7 +1015,7 @@ void RtpVideoSender::OnPacketFeedbackVector(
       // clean up anyway.
       continue;
     }
-    ArrayView<const uint16_t> rtp_sequence_numbers(kv.second);
+    std::span<const uint16_t> rtp_sequence_numbers(kv.second);
     it->second->OnPacketsAcknowledged(rtp_sequence_numbers);
   }
 }
@@ -1027,7 +1027,7 @@ void RtpVideoSender::SetEncodingData(size_t width,
                                    rtp_config_.max_packet_size);
 }
 
-void RtpVideoSender::SetCsrcs(ArrayView<const uint32_t> csrcs) {
+void RtpVideoSender::SetCsrcs(std::span<const uint32_t> csrcs) {
   MutexLock lock(&mutex_);
   csrcs_.assign(csrcs.begin(),
                 csrcs.begin() + std::min<size_t>(csrcs.size(), kRtpCsrcSize));
diff --git a/call/rtp_video_sender.h b/call/rtp_video_sender.h
index ca7d0989d1..5d35e5d492 100644
--- a/call/rtp_video_sender.h
+++ b/call/rtp_video_sender.h
@@ -15,10 +15,10 @@
 #include <cstdint>
 #include <map>
 #include <memory>
+#include <span>
 #include <vector>
 
 #include "absl/base/nullability.h"
-#include "api/array_view.h"
 #include "api/call/bitrate_allocation.h"
 #include "api/call/transport.h"
 #include "api/crypto/crypto_options.h"
@@ -113,7 +113,7 @@ class RtpVideoSender : public RtpVideoSenderInterface,
   std::map<uint32_t, RtpPayloadState> GetRtpPayloadStates() const
       RTC_LOCKS_EXCLUDED(mutex_) override;
 
-  void DeliverRtcp(ArrayView<const uint8_t> packet)
+  void DeliverRtcp(std::span<const uint8_t> packet)
       RTC_LOCKS_EXCLUDED(mutex_) override;
   // Implements webrtc::VCMProtectionCallback.
   int ProtectionRequest(const FecProtectionParams* delta_params,
@@ -158,12 +158,12 @@ class RtpVideoSender : public RtpVideoSenderInterface,
   // Sets the list of CSRCs to be included in every packet. If more than
   // kRtpCsrcSize CSRCs are provided, only the first kRtpCsrcSize elements are
   // kept.
-  void SetCsrcs(ArrayView<const uint32_t> csrcs)
+  void SetCsrcs(std::span<const uint32_t> csrcs)
       RTC_LOCKS_EXCLUDED(mutex_) override;
 
   std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos(
       uint32_t ssrc,
-      ArrayView<const uint16_t> sequence_numbers) const
+      std::span<const uint16_t> sequence_numbers) const
       RTC_LOCKS_EXCLUDED(mutex_) override;
 
   // From StreamFeedbackObserver.
diff --git a/call/rtp_video_sender_interface.h b/call/rtp_video_sender_interface.h
index 754fe03cd4..035044cd82 100644
--- a/call/rtp_video_sender_interface.h
+++ b/call/rtp_video_sender_interface.h
@@ -14,9 +14,9 @@
 #include <cstddef>
 #include <cstdint>
 #include <map>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/call/bitrate_allocation.h"
 #include "api/fec_controller_override.h"
 #include "api/video/video_layers_allocation.h"
@@ -40,7 +40,7 @@ class RtpVideoSenderInterface : public EncodedImageCallback,
   virtual std::map<uint32_t, RtpState> GetRtpStates() const = 0;
   virtual std::map<uint32_t, RtpPayloadState> GetRtpPayloadStates() const = 0;
 
-  virtual void DeliverRtcp(ArrayView<const uint8_t> packet) = 0;
+  virtual void DeliverRtcp(std::span<const uint8_t> packet) = 0;
 
   virtual void OnBitrateAllocationUpdated(
       const VideoBitrateAllocation& bitrate) = 0;
@@ -55,10 +55,10 @@ class RtpVideoSenderInterface : public EncodedImageCallback,
   virtual void SetEncodingData(size_t width,
                                size_t height,
                                size_t num_temporal_layers) = 0;
-  virtual void SetCsrcs(ArrayView<const uint32_t> csrcs) = 0;
+  virtual void SetCsrcs(std::span<const uint32_t> csrcs) = 0;
   virtual std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos(
       uint32_t ssrc,
-      ArrayView<const uint16_t> sequence_numbers) const = 0;
+      std::span<const uint16_t> sequence_numbers) const = 0;
 
   // Implements FecControllerOverride.
   void SetFecAllowed(bool fec_allowed) override = 0;
diff --git a/call/rtp_video_sender_unittest.cc b/call/rtp_video_sender_unittest.cc
index 5014cf5dad..a979e05db7 100644
--- a/call/rtp_video_sender_unittest.cc
+++ b/call/rtp_video_sender_unittest.cc
@@ -15,10 +15,10 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/call/bitrate_allocation.h"
 #include "api/call/transport.h"
 #include "api/crypto/crypto_options.h"
@@ -137,7 +137,7 @@ VideoSendStream::Config CreateVideoSendStreamConfig(
     const std::vector<uint32_t>& ssrcs,
     const std::vector<uint32_t>& rtx_ssrcs,
     int payload_type,
-    ArrayView<const int> payload_types) {
+    std::span<const int> payload_types) {
   VideoSendStream::Config config(transport);
   config.rtp.ssrcs = ssrcs;
   config.rtp.rtx.ssrcs = rtx_ssrcs;
@@ -511,7 +511,7 @@ TEST(RtpVideoSenderTest, DoesNotRetrasmitAckedPackets) {
       .Times(2)
       .WillRepeatedly(
           [&rtp_sequence_numbers, &transport_sequence_numbers](
-              ArrayView<const uint8_t> packet, const PacketOptions& options) {
+              std::span<const uint8_t> packet, const PacketOptions& options) {
             RtpPacket rtp_packet;
             EXPECT_TRUE(rtp_packet.Parse(packet));
             rtp_sequence_numbers.push_back(rtp_packet.SequenceNumber());
@@ -537,13 +537,13 @@ TEST(RtpVideoSenderTest, DoesNotRetrasmitAckedPackets) {
   EXPECT_CALL(test.transport(), SendRtp)
       .Times(2)
       .WillRepeatedly([&retransmitted_rtp_sequence_numbers](
-                          ArrayView<const uint8_t> packet,
+                          std::span<const uint8_t> packet,
                           const PacketOptions& options) {
         RtpPacket rtp_packet;
         EXPECT_TRUE(rtp_packet.Parse(packet));
         EXPECT_EQ(rtp_packet.Ssrc(), kRtxSsrc1);
         // Capture the retransmitted sequence number from the RTX header.
-        ArrayView<const uint8_t> payload = rtp_packet.payload();
+        std::span<const uint8_t> payload = rtp_packet.payload();
         retransmitted_rtp_sequence_numbers.push_back(
             ByteReader<uint16_t>::ReadBigEndian(payload.data()));
         return true;
@@ -578,13 +578,13 @@ TEST(RtpVideoSenderTest, DoesNotRetrasmitAckedPackets) {
   // still be retransmitted.
   test.AdvanceTime(TimeDelta::Millis(33));
   EXPECT_CALL(test.transport(), SendRtp)
-      .WillOnce([&lost_packet_feedback](ArrayView<const uint8_t> packet,
+      .WillOnce([&lost_packet_feedback](std::span<const uint8_t> packet,
                                         const PacketOptions& options) {
         RtpPacket rtp_packet;
         EXPECT_TRUE(rtp_packet.Parse(packet));
         EXPECT_EQ(rtp_packet.Ssrc(), kRtxSsrc1);
         // Capture the retransmitted sequence number from the RTX header.
-        ArrayView<const uint8_t> payload = rtp_packet.payload();
+        std::span<const uint8_t> payload = rtp_packet.payload();
         EXPECT_EQ(lost_packet_feedback.rtp_sequence_number,
                   ByteReader<uint16_t>::ReadBigEndian(payload.data()));
         return true;
@@ -681,7 +681,7 @@ TEST(RtpVideoSenderTest, EarlyRetransmits) {
   EXPECT_CALL(test.transport(), SendRtp)
       .WillOnce(
           [&frame1_rtp_sequence_number, &frame1_transport_sequence_number](
-              ArrayView<const uint8_t> packet, const PacketOptions& options) {
+              std::span<const uint8_t> packet, const PacketOptions& options) {
             RtpPacket rtp_packet;
             EXPECT_TRUE(rtp_packet.Parse(packet));
             frame1_rtp_sequence_number = rtp_packet.SequenceNumber();
@@ -700,7 +700,7 @@ TEST(RtpVideoSenderTest, EarlyRetransmits) {
   EXPECT_CALL(test.transport(), SendRtp)
       .WillOnce(
           [&frame2_rtp_sequence_number, &frame2_transport_sequence_number](
-              ArrayView<const uint8_t> packet, const PacketOptions& options) {
+              std::span<const uint8_t> packet, const PacketOptions& options) {
             RtpPacket rtp_packet;
             EXPECT_TRUE(rtp_packet.Parse(packet));
             frame2_rtp_sequence_number = rtp_packet.SequenceNumber();
@@ -717,7 +717,7 @@ TEST(RtpVideoSenderTest, EarlyRetransmits) {
   // Inject a transport feedback where the packet for the first frame is lost,
   // expect a retransmission for it.
   EXPECT_CALL(test.transport(), SendRtp)
-      .WillOnce([&frame1_rtp_sequence_number](ArrayView<const uint8_t> packet,
+      .WillOnce([&frame1_rtp_sequence_number](std::span<const uint8_t> packet,
                                               const PacketOptions& options) {
         RtpPacket rtp_packet;
         EXPECT_TRUE(rtp_packet.Parse(packet));
@@ -725,7 +725,7 @@ TEST(RtpVideoSenderTest, EarlyRetransmits) {
 
         // Retransmitted sequence number from the RTX header should match
         // the lost packet.
-        ArrayView<const uint8_t> payload = rtp_packet.payload();
+        std::span<const uint8_t> payload = rtp_packet.payload();
         EXPECT_EQ(ByteReader<uint16_t>::ReadBigEndian(payload.data()),
                   frame1_rtp_sequence_number);
         return true;
@@ -760,7 +760,7 @@ TEST(RtpVideoSenderTest, SupportsDependencyDescriptor) {
   std::vector<RtpPacket> sent_packets;
   ON_CALL(test.transport(), SendRtp)
       .WillByDefault(
-          [&](ArrayView<const uint8_t> packet, const PacketOptions& options) {
+          [&](std::span<const uint8_t> packet, const PacketOptions& options) {
             sent_packets.emplace_back(&extensions);
             EXPECT_TRUE(sent_packets.back().Parse(packet));
             return true;
@@ -824,7 +824,7 @@ TEST(RtpVideoSenderTest, SimulcastIndependentFrameIds) {
   std::vector<RtpPacket> sent_packets;
   ON_CALL(test.transport(), SendRtp)
       .WillByDefault(
-          [&](ArrayView<const uint8_t> packet, const PacketOptions& options) {
+          [&](std::span<const uint8_t> packet, const PacketOptions& options) {
             sent_packets.emplace_back(&extensions);
             EXPECT_TRUE(sent_packets.back().Parse(packet));
             return true;
@@ -883,7 +883,7 @@ TEST(RtpVideoSenderTest,
   std::vector<RtpPacket> sent_packets;
   ON_CALL(test.transport(), SendRtp)
       .WillByDefault(
-          [&](ArrayView<const uint8_t> packet, const PacketOptions& options) {
+          [&](std::span<const uint8_t> packet, const PacketOptions& options) {
             sent_packets.emplace_back(&extensions);
             EXPECT_TRUE(sent_packets.back().Parse(packet));
             return true;
@@ -938,7 +938,7 @@ TEST(RtpVideoSenderTest, MixedCodecSimulcastPayloadType) {
   std::vector<RtpPacket> sent_packets;
   EXPECT_CALL(test.transport(), SendRtp)
       .Times(3)
-      .WillRepeatedly([&](ArrayView<const uint8_t> packet,
+      .WillRepeatedly([&](std::span<const uint8_t> packet,
                           const PacketOptions& options) -> bool {
         RtpPacket& rtp_packet = sent_packets.emplace_back();
         EXPECT_TRUE(rtp_packet.Parse(packet));
@@ -983,7 +983,7 @@ TEST(RtpVideoSenderTest, MixedCodecSimulcastPayloadType) {
   EXPECT_CALL(test.transport(), SendRtp)
       .Times(3)
       .WillRepeatedly(
-          [&](ArrayView<const uint8_t> packet, const PacketOptions& options) {
+          [&](std::span<const uint8_t> packet, const PacketOptions& options) {
             RtpPacket& rtp_packet = sent_rtx_packets.emplace_back();
             EXPECT_TRUE(rtp_packet.Parse(packet));
             return true;
@@ -1009,7 +1009,7 @@ TEST(RtpVideoSenderTest,
   std::vector<RtpPacket> sent_packets;
   ON_CALL(test.transport(), SendRtp)
       .WillByDefault(
-          [&](ArrayView<const uint8_t> packet, const PacketOptions&) {
+          [&](std::span<const uint8_t> packet, const PacketOptions&) {
             EXPECT_TRUE(sent_packets.emplace_back(&extensions).Parse(packet));
             return true;
           });
@@ -1055,7 +1055,7 @@ TEST(RtpVideoSenderTest, SupportsDependencyDescriptorForVp9) {
   std::vector<RtpPacket> sent_packets;
   ON_CALL(test.transport(), SendRtp)
       .WillByDefault(
-          [&](ArrayView<const uint8_t> packet, const PacketOptions& options) {
+          [&](std::span<const uint8_t> packet, const PacketOptions& options) {
             sent_packets.emplace_back(&extensions);
             EXPECT_TRUE(sent_packets.back().Parse(packet));
             return true;
@@ -1111,7 +1111,7 @@ TEST(RtpVideoSenderTest,
   std::vector<RtpPacket> sent_packets;
   ON_CALL(test.transport(), SendRtp)
       .WillByDefault(
-          [&](ArrayView<const uint8_t> packet, const PacketOptions& options) {
+          [&](std::span<const uint8_t> packet, const PacketOptions& options) {
             sent_packets.emplace_back(&extensions);
             EXPECT_TRUE(sent_packets.back().Parse(packet));
             return true;
@@ -1165,7 +1165,7 @@ TEST(RtpVideoSenderTest,
   std::vector<RtpPacket> sent_packets;
   EXPECT_CALL(test.transport(), SendRtp(_, _))
       .Times(2)
-      .WillRepeatedly([&](ArrayView<const uint8_t> packet,
+      .WillRepeatedly([&](std::span<const uint8_t> packet,
                           const PacketOptions& options) -> bool {
         sent_packets.emplace_back(&extensions);
         EXPECT_TRUE(sent_packets.back().Parse(packet));
@@ -1223,7 +1223,7 @@ TEST(RtpVideoSenderTest, GenerateDependecyDescriptorForGenericCodecs) {
   std::vector<RtpPacket> sent_packets;
   ON_CALL(test.transport(), SendRtp)
       .WillByDefault(
-          [&](ArrayView<const uint8_t> packet, const PacketOptions& options) {
+          [&](std::span<const uint8_t> packet, const PacketOptions& options) {
             sent_packets.emplace_back(&extensions);
             EXPECT_TRUE(sent_packets.back().Parse(packet));
             return true;
@@ -1269,7 +1269,7 @@ TEST(RtpVideoSenderTest, SupportsStoppingUsingDependencyDescriptor) {
   std::vector<RtpPacket> sent_packets;
   ON_CALL(test.transport(), SendRtp)
       .WillByDefault(
-          [&](ArrayView<const uint8_t> packet, const PacketOptions& options) {
+          [&](std::span<const uint8_t> packet, const PacketOptions& options) {
             sent_packets.emplace_back(&extensions);
             EXPECT_TRUE(sent_packets.back().Parse(packet));
             return true;
@@ -1381,7 +1381,7 @@ TEST(RtpVideoSenderTest, ClearsPendingPacketsOnInactivation) {
   std::vector<RtpPacket> sent_packets;
   ON_CALL(test.transport(), SendRtp)
       .WillByDefault(
-          [&](ArrayView<const uint8_t> packet, const PacketOptions& options) {
+          [&](std::span<const uint8_t> packet, const PacketOptions& options) {
             sent_packets.emplace_back(&extensions);
             EXPECT_TRUE(sent_packets.back().Parse(packet));
             return true;
@@ -1459,7 +1459,7 @@ TEST(RtpVideoSenderTest,
   std::vector<RtpPacket> sent_packets;
   ON_CALL(test.transport(), SendRtp)
       .WillByDefault(
-          [&](ArrayView<const uint8_t> packet, const PacketOptions& options) {
+          [&](std::span<const uint8_t> packet, const PacketOptions& options) {
             sent_packets.emplace_back(&extensions);
             EXPECT_TRUE(sent_packets.back().Parse(packet));
             return true;
@@ -1552,7 +1552,7 @@ TEST(RtpVideoSenderTest, RetransmitsBaseLayerOnly) {
       .Times(2)
       .WillRepeatedly(
           [&rtp_sequence_numbers, &transport_sequence_numbers](
-              ArrayView<const uint8_t> packet, const PacketOptions& options) {
+              std::span<const uint8_t> packet, const PacketOptions& options) {
             RtpPacket rtp_packet;
             EXPECT_TRUE(rtp_packet.Parse(packet));
             rtp_sequence_numbers.push_back(rtp_packet.SequenceNumber());
@@ -1587,13 +1587,13 @@ TEST(RtpVideoSenderTest, RetransmitsBaseLayerOnly) {
   EXPECT_CALL(test.transport(), SendRtp)
       .Times(1)
       .WillRepeatedly([&retransmitted_rtp_sequence_numbers](
-                          ArrayView<const uint8_t> packet,
+                          std::span<const uint8_t> packet,
                           const PacketOptions& options) {
         RtpPacket rtp_packet;
         EXPECT_TRUE(rtp_packet.Parse(packet));
         EXPECT_EQ(rtp_packet.Ssrc(), kRtxSsrc1);
         // Capture the retransmitted sequence number from the RTX header.
-        ArrayView<const uint8_t> payload = rtp_packet.payload();
+        std::span<const uint8_t> payload = rtp_packet.payload();
         retransmitted_rtp_sequence_numbers.push_back(
             ByteReader<uint16_t>::ReadBigEndian(payload.data()));
         return true;
diff --git a/call/rtx_receive_stream.cc b/call/rtx_receive_stream.cc
index d725d285da..2f773802de 100644
--- a/call/rtx_receive_stream.cc
+++ b/call/rtx_receive_stream.cc
@@ -14,9 +14,9 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/sequence_checker.h"
 #include "call/rtp_packet_sink_interface.h"
@@ -78,7 +78,7 @@ void RtxReceiveStream::OnRtpPacket(const RtpPacketReceived& rtx_packet) {
   if (rtp_receive_statistics_) {
     rtp_receive_statistics_->OnRtpPacket(rtx_packet);
   }
-  ArrayView<const uint8_t> payload = rtx_packet.payload();
+  std::span<const uint8_t> payload = rtx_packet.payload();
 
   if (payload.size() < kRtxHeaderSize) {
     LogRtpPacketToEventLog(rtx_packet, /*osn=*/std::nullopt);
diff --git a/call/rtx_receive_stream_unittest.cc b/call/rtx_receive_stream_unittest.cc
index 80fd8c96ca..9a30cc437e 100644
--- a/call/rtx_receive_stream_unittest.cc
+++ b/call/rtx_receive_stream_unittest.cc
@@ -14,8 +14,8 @@
 #include <cstdint>
 #include <cstring>
 #include <map>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/rtc_event_log/rtc_event.h"
 #include "api/units/timestamp.h"
@@ -123,7 +123,7 @@ std::map<int, int> PayloadTypeMapping() {
 }
 
 template <typename T>
-ArrayView<T> Truncate(ArrayView<T> a, size_t drop) {
+std::span<T> Truncate(std::span<T> a, size_t drop) {
   return a.subspan(0, a.size() - drop);
 }
 
@@ -134,7 +134,7 @@ TEST(RtxReceiveStreamTest, RestoresPacketPayload) {
   Environment env = CreateTestEnvironment();
   RtxReceiveStream rtx_sink(env, &media_sink, PayloadTypeMapping(), kMediaSSRC);
   RtpPacketReceived rtx_packet;
-  EXPECT_TRUE(rtx_packet.Parse(ArrayView<const uint8_t>(kRtxPacket)));
+  EXPECT_TRUE(rtx_packet.Parse(std::span<const uint8_t>(kRtxPacket)));
 
   EXPECT_CALL(media_sink, OnRtpPacket)
       .WillOnce([](const RtpPacketReceived& packet) {
@@ -152,7 +152,7 @@ TEST(RtxReceiveStreamTest, SetsRecoveredFlag) {
   Environment env = CreateTestEnvironment();
   RtxReceiveStream rtx_sink(env, &media_sink, PayloadTypeMapping(), kMediaSSRC);
   RtpPacketReceived rtx_packet;
-  EXPECT_TRUE(rtx_packet.Parse(ArrayView<const uint8_t>(kRtxPacket)));
+  EXPECT_TRUE(rtx_packet.Parse(std::span<const uint8_t>(kRtxPacket)));
   EXPECT_FALSE(rtx_packet.recovered());
   EXPECT_CALL(media_sink, OnRtpPacket)
       .WillOnce([](const RtpPacketReceived& packet) {
@@ -171,7 +171,7 @@ TEST(RtxReceiveStreamTest, IgnoresUnknownPayloadType) {
   Environment env = CreateTestEnvironment({.event_log = &log});
   RtxReceiveStream rtx_sink(env, &media_sink, payload_type_mapping, kMediaSSRC);
   RtpPacketReceived rtx_packet;
-  EXPECT_TRUE(rtx_packet.Parse(ArrayView<const uint8_t>(kRtxPacket)));
+  EXPECT_TRUE(rtx_packet.Parse(std::span<const uint8_t>(kRtxPacket)));
   EXPECT_CALL(log, LogProxy(_));
   rtx_sink.OnRtpPacket(rtx_packet);
 }
@@ -183,7 +183,7 @@ TEST(RtxReceiveStreamTest, IgnoresTruncatedPacket) {
   RtxReceiveStream rtx_sink(env, &media_sink, PayloadTypeMapping(), kMediaSSRC);
   RtpPacketReceived rtx_packet;
   EXPECT_TRUE(
-      rtx_packet.Parse(Truncate(ArrayView<const uint8_t>(kRtxPacket), 2)));
+      rtx_packet.Parse(Truncate(std::span<const uint8_t>(kRtxPacket), 2)));
   EXPECT_CALL(log, LogProxy(_));
   rtx_sink.OnRtpPacket(rtx_packet);
 }
@@ -195,7 +195,7 @@ TEST(RtxReceiveStreamTest, CopiesRtpHeaderExtensions) {
   RtpHeaderExtensionMap extension_map;
   extension_map.RegisterByType(3, kRtpExtensionVideoRotation);
   RtpPacketReceived rtx_packet(&extension_map);
-  EXPECT_TRUE(rtx_packet.Parse(ArrayView<const uint8_t>(kRtxPacketWithCVO)));
+  EXPECT_TRUE(rtx_packet.Parse(std::span<const uint8_t>(kRtxPacketWithCVO)));
 
   VideoRotation rotation = kVideoRotation_0;
   EXPECT_TRUE(rtx_packet.GetExtension<VideoOrientation>(&rotation));
@@ -220,7 +220,7 @@ TEST(RtxReceiveStreamTest, PropagatesArrivalTime) {
   Environment env = CreateTestEnvironment();
   RtxReceiveStream rtx_sink(env, &media_sink, PayloadTypeMapping(), kMediaSSRC);
   RtpPacketReceived rtx_packet(nullptr);
-  EXPECT_TRUE(rtx_packet.Parse(ArrayView<const uint8_t>(kRtxPacket)));
+  EXPECT_TRUE(rtx_packet.Parse(std::span<const uint8_t>(kRtxPacket)));
   rtx_packet.set_arrival_time(Timestamp::Millis(123));
   EXPECT_CALL(media_sink, OnRtpPacket(Property(&RtpPacketReceived::arrival_time,
                                                Timestamp::Millis(123))));
@@ -236,14 +236,14 @@ TEST(RtxReceiveStreamTest, SupportsLargePacket) {
   constexpr int kRtxPayloadOffset = 14;
   uint8_t large_rtx_packet[kRtxPacketSize];
   memcpy(large_rtx_packet, kRtxPacket, sizeof(kRtxPacket));
-  ArrayView<uint8_t> payload(large_rtx_packet + kRtxPayloadOffset,
+  std::span<uint8_t> payload(large_rtx_packet + kRtxPayloadOffset,
                              kRtxPacketSize - kRtxPayloadOffset);
 
   // Fill payload.
   for (size_t i = 0; i < payload.size(); i++) {
     payload[i] = i;
   }
-  EXPECT_TRUE(rtx_packet.Parse(ArrayView<const uint8_t>(large_rtx_packet)));
+  EXPECT_TRUE(rtx_packet.Parse(std::span<const uint8_t>(large_rtx_packet)));
 
   EXPECT_CALL(media_sink, OnRtpPacket)
       .WillOnce([&](const RtpPacketReceived& packet) {
@@ -267,10 +267,10 @@ TEST(RtxReceiveStreamTest, SupportsLargePacketWithPadding) {
   uint8_t large_rtx_packet[kRtxPacketSize];
   memcpy(large_rtx_packet, kRtxPacketWithPadding,
          sizeof(kRtxPacketWithPadding));
-  ArrayView<uint8_t> payload(
+  std::span<uint8_t> payload(
       large_rtx_packet + kRtxPayloadOffset,
       kRtxPacketSize - kRtxPayloadOffset - kRtxPaddingSize);
-  ArrayView<uint8_t> padding(
+  std::span<uint8_t> padding(
       large_rtx_packet + kRtxPacketSize - kRtxPaddingSize, kRtxPaddingSize);
 
   // Fill payload.
@@ -282,7 +282,7 @@ TEST(RtxReceiveStreamTest, SupportsLargePacketWithPadding) {
     padding[i] = kRtxPaddingSize;
   }
 
-  EXPECT_TRUE(rtx_packet.Parse(ArrayView<const uint8_t>(large_rtx_packet)));
+  EXPECT_TRUE(rtx_packet.Parse(std::span<const uint8_t>(large_rtx_packet)));
 
   EXPECT_CALL(media_sink, OnRtpPacket)
       .WillOnce([&](const RtpPacketReceived& packet) {
@@ -314,7 +314,7 @@ TEST(RtxReceiveStreamTest, LogsRtpPacketIncoming) {
   Environment env = CreateTestEnvironment({.event_log = &log});
   RtxReceiveStream rtx_sink(env, &media_sink, PayloadTypeMapping(), kMediaSSRC);
   RtpPacketReceived rtx_packet;
-  EXPECT_TRUE(rtx_packet.Parse(ArrayView<const uint8_t>(kRtxPacket)));
+  EXPECT_TRUE(rtx_packet.Parse(std::span<const uint8_t>(kRtxPacket)));
 
   EXPECT_CALL(log, LogProxy(IsRtcEventRtpPacketIncomingPtrWithSsrcAndOsn(
                        kRtxSSRC, kMediaSeqno)));
diff --git a/call/video_send_stream.h b/call/video_send_stream.h
index c5a24d2c48..683be5f469 100644
--- a/call/video_send_stream.h
+++ b/call/video_send_stream.h
@@ -15,11 +15,11 @@
 
 #include <map>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "api/adaptation/resource.h"
-#include "api/array_view.h"
 #include "api/call/transport.h"
 #include "api/crypto/crypto_options.h"
 #include "api/frame_transformer_interface.h"
@@ -264,7 +264,7 @@ class VideoSendStream {
   virtual void SetStats(const Stats& stats) { RTC_CHECK_NOTREACHED(); }
 
   // Sets the list of CSRCs to be included in every packet.
-  virtual void SetCsrcs(ArrayView<const uint32_t> csrcs) = 0;
+  virtual void SetCsrcs(std::span<const uint32_t> csrcs) = 0;
 
   virtual void GenerateKeyFrame(const std::vector<std::string>& rids) = 0;
 
diff --git a/common_audio/BUILD.gn b/common_audio/BUILD.gn
index 83e9ef98a0..c54c797a84 100644
--- a/common_audio/BUILD.gn
+++ b/common_audio/BUILD.gn
@@ -45,7 +45,6 @@ rtc_library("common_audio") {
   deps = [
     ":common_audio_c",
     ":sinc_resampler",
-    "../api:array_view",
     "../api/audio:audio_frame_api",
     "../api/units:timestamp",
     "../rtc_base:checks",
diff --git a/common_audio/channel_buffer.h b/common_audio/channel_buffer.h
index 7d040e5d5f..5a79f00450 100644
--- a/common_audio/channel_buffer.h
+++ b/common_audio/channel_buffer.h
@@ -14,9 +14,9 @@
 #include <cstdint>
 #include <cstring>
 #include <memory>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/audio_view.h"
 #include "rtc_base/checks.h"
 
@@ -89,19 +89,19 @@ class ChannelBuffer {
         num_channels_(num_channels),
         num_bands_(num_bands),
         bands_view_(num_allocated_channels_,
-                    std::vector<ArrayView<T>>(num_bands_)),
+                    std::vector<std::span<T>>(num_bands_)),
         channels_view_(num_bands_,
-                       std::vector<ArrayView<T>>(num_allocated_channels_)) {
+                       std::vector<std::span<T>>(num_allocated_channels_)) {
     // Temporarily cast away const_ness to allow populating the array views.
     auto* bands_view =
-        const_cast<std::vector<std::vector<ArrayView<T>>>*>(&bands_view_);
+        const_cast<std::vector<std::vector<std::span<T>>>*>(&bands_view_);
     auto* channels_view =
-        const_cast<std::vector<std::vector<ArrayView<T>>>*>(&channels_view_);
+        const_cast<std::vector<std::vector<std::span<T>>>*>(&channels_view_);
 
     for (size_t ch = 0; ch < num_allocated_channels_; ++ch) {
       for (size_t band = 0; band < num_bands_; ++band) {
         (*channels_view)[band][ch] =
-            ArrayView<T>(&data_[ch * num_frames_ + band * num_frames_per_band_],
+            std::span<T>(&data_[ch * num_frames_ + band * num_frames_per_band_],
                          num_frames_per_band_);
         (*bands_view)[ch][band] = channels_view_[band][ch];
         channels_[band * num_allocated_channels_ + ch] =
@@ -133,10 +133,10 @@ class ChannelBuffer {
     const ChannelBuffer<T>* t = this;
     return const_cast<T* const*>(t->channels(band));
   }
-  ArrayView<const ArrayView<T>> channels_view(size_t band = 0) {
+  std::span<const std::span<T>> channels_view(size_t band = 0) {
     return channels_view_[band];
   }
-  ArrayView<const ArrayView<T>> channels_view(size_t band = 0) const {
+  std::span<const std::span<T>> channels_view(size_t band = 0) const {
     return channels_view_[band];
   }
 
@@ -157,10 +157,10 @@ class ChannelBuffer {
     return const_cast<T* const*>(t->bands(channel));
   }
 
-  ArrayView<const ArrayView<T>> bands_view(size_t channel) {
+  std::span<const std::span<T>> bands_view(size_t channel) {
     return bands_view_[channel];
   }
-  ArrayView<const ArrayView<T>> bands_view(size_t channel) const {
+  std::span<const std::span<T>> bands_view(size_t channel) const {
     return bands_view_[channel];
   }
 
@@ -204,8 +204,8 @@ class ChannelBuffer {
   // Number of channels the user sees.
   size_t num_channels_;
   const size_t num_bands_;
-  const std::vector<std::vector<ArrayView<T>>> bands_view_;
-  const std::vector<std::vector<ArrayView<T>>> channels_view_;
+  const std::vector<std::vector<std::span<T>>> bands_view_;
+  const std::vector<std::vector<std::span<T>>> channels_view_;
 };
 
 // One int16_t and one float ChannelBuffer that are kept in sync. The sync is
diff --git a/common_audio/include/audio_util.h b/common_audio/include/audio_util.h
index 672ebd1aa0..0b04f812bb 100644
--- a/common_audio/include/audio_util.h
+++ b/common_audio/include/audio_util.h
@@ -106,7 +106,7 @@ inline float FloatS16ToDbfs(float v) {
 // Copy audio from `src` channels to `dest` channels unless `src` and `dest`
 // point to the same address. `src` and `dest` must have the same number of
 // channels, and there must be sufficient space allocated in `dest`.
-// TODO: b/335805780 - Accept ArrayView.
+// TODO: b/335805780 - Accept std::span.
 template <typename T>
 void CopyAudioIfNeeded(const T* const* src,
                        int num_frames,
diff --git a/examples/BUILD.gn b/examples/BUILD.gn
index 4dd7670cd3..dae37ec578 100644
--- a/examples/BUILD.gn
+++ b/examples/BUILD.gn
@@ -65,7 +65,6 @@ rtc_library("read_auth_file") {
     "turnserver/read_auth_file.h",
   ]
   deps = [
-    "../api:array_view",
     "../rtc_base:stringutils",
     "//third_party/abseil-cpp/absl/strings:string_view",
   ]
@@ -677,7 +676,6 @@ if (is_linux || is_chromeos || is_win) {
     ]
 
     deps = [
-      "../api:array_view",
       "../api:async_dns_resolver",
       "../api:audio_options_api",
       "../api:create_frame_generator",
diff --git a/examples/androidvoip/BUILD.gn b/examples/androidvoip/BUILD.gn
index 1954875050..c26d2b2b50 100644
--- a/examples/androidvoip/BUILD.gn
+++ b/examples/androidvoip/BUILD.gn
@@ -55,7 +55,6 @@ if (is_android) {
 
     deps = [
       ":generated_jni",
-      "../../api:array_view",
       "../../api:sequence_checker",
       "../../api/audio:builtin_audio_processing_builder",
       "../../api/environment",
diff --git a/examples/androidvoip/jni/android_voip_client.cc b/examples/androidvoip/jni/android_voip_client.cc
index 2c1fd43d02..53489da5e2 100644
--- a/examples/androidvoip/jni/android_voip_client.cc
+++ b/examples/androidvoip/jni/android_voip_client.cc
@@ -20,12 +20,12 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/memory/memory.h"
-#include "api/array_view.h"
 #include "api/audio/builtin_audio_processing_builder.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
@@ -481,7 +481,7 @@ void AndroidVoipClient::SendRtpPacket(const std::vector<uint8_t>& packet_copy) {
   }
 }
 
-bool AndroidVoipClient::SendRtp(webrtc::ArrayView<const uint8_t> packet,
+bool AndroidVoipClient::SendRtp(std::span<const uint8_t> packet,
                                 const webrtc::PacketOptions& options) {
   std::vector<uint8_t> packet_copy(packet.begin(), packet.end());
   voip_thread_->PostTask([this, packet_copy = std::move(packet_copy)] {
@@ -501,7 +501,7 @@ void AndroidVoipClient::SendRtcpPacket(
   }
 }
 
-bool AndroidVoipClient::SendRtcp(webrtc::ArrayView<const uint8_t> packet,
+bool AndroidVoipClient::SendRtcp(std::span<const uint8_t> packet,
                                  const webrtc::PacketOptions& options) {
   std::vector<uint8_t> packet_copy(packet.begin(), packet.end());
   voip_thread_->PostTask([this, packet_copy = std::move(packet_copy)] {
@@ -519,7 +519,7 @@ void AndroidVoipClient::ReadRTPPacket(const std::vector<uint8_t>& packet_copy) {
   }
   webrtc::VoipResult result = voip_engine_->Network().ReceivedRTPPacket(
       *channel_,
-      webrtc::ArrayView<const uint8_t>(packet_copy.data(), packet_copy.size()));
+      std::span<const uint8_t>(packet_copy.data(), packet_copy.size()));
   RTC_CHECK(result == webrtc::VoipResult::kOk);
 }
 
@@ -543,7 +543,7 @@ void AndroidVoipClient::ReadRTCPPacket(
   }
   webrtc::VoipResult result = voip_engine_->Network().ReceivedRTCPPacket(
       *channel_,
-      webrtc::ArrayView<const uint8_t>(packet_copy.data(), packet_copy.size()));
+      std::span<const uint8_t>(packet_copy.data(), packet_copy.size()));
   RTC_CHECK(result == webrtc::VoipResult::kOk);
 }
 
diff --git a/examples/androidvoip/jni/android_voip_client.h b/examples/androidvoip/jni/android_voip_client.h
index 7d9e499ae3..885921fd50 100644
--- a/examples/androidvoip/jni/android_voip_client.h
+++ b/examples/androidvoip/jni/android_voip_client.h
@@ -16,10 +16,10 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/call/transport.h"
 #include "api/environment/environment.h"
@@ -121,9 +121,9 @@ class AndroidVoipClient : public webrtc::Transport {
   void Delete(JNIEnv* env);
 
   // Implementation for Transport.
-  bool SendRtp(webrtc::ArrayView<const uint8_t> packet,
+  bool SendRtp(std::span<const uint8_t> packet,
                const webrtc::PacketOptions& options) override;
-  bool SendRtcp(webrtc::ArrayView<const uint8_t> packet,
+  bool SendRtcp(std::span<const uint8_t> packet,
                 const webrtc::PacketOptions& options) override;
 
   void OnSignalReadRTPPacket(webrtc::AsyncPacketSocket* socket,
diff --git a/examples/peerconnection/client/linux/main_wnd.h b/examples/peerconnection/client/linux/main_wnd.h
index cee1d3ab64..db730e662e 100644
--- a/examples/peerconnection/client/linux/main_wnd.h
+++ b/examples/peerconnection/client/linux/main_wnd.h
@@ -14,9 +14,9 @@
 #include <stdint.h>
 
 #include <memory>
+#include <span>
 #include <string>
 
-#include "api/array_view.h"
 #include "api/media_stream_interface.h"
 #include "api/scoped_refptr.h"
 #include "api/video/video_frame.h"
@@ -94,7 +94,7 @@ class GtkMainWnd : public MainWindow {
     // VideoSinkInterface implementation
     void OnFrame(const webrtc::VideoFrame& frame) override;
 
-    webrtc::ArrayView<const uint8_t> image() const { return image_; }
+    std::span<const uint8_t> image() const { return image_; }
 
     int width() const { return width_; }
 
diff --git a/examples/turnserver/read_auth_file.cc b/examples/turnserver/read_auth_file.cc
index c1b6eff2d6..65d1e41971 100644
--- a/examples/turnserver/read_auth_file.cc
+++ b/examples/turnserver/read_auth_file.cc
@@ -13,10 +13,10 @@
 #include <cstddef>
 #include <istream>
 #include <map>
+#include <span>
 #include <string>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "rtc_base/string_encode.h"
 
 namespace webrtc_examples {
@@ -28,7 +28,7 @@ std::map<std::string, std::string> ReadAuthFile(std::istream* s) {
     if (sep == std::string::npos)
       continue;
     char buf[32];
-    size_t len = webrtc::hex_decode(webrtc::ArrayView<char>(buf),
+    size_t len = webrtc::hex_decode(std::span<char>(buf),
                                     absl::string_view(line).substr(sep + 1));
     if (len > 0) {
       name_to_key.emplace(line.substr(0, sep), std::string(buf, len));
diff --git a/system_wrappers/BUILD.gn b/system_wrappers/BUILD.gn
index 12a5a828f8..a3e9a49204 100644
--- a/system_wrappers/BUILD.gn
+++ b/system_wrappers/BUILD.gn
@@ -23,7 +23,6 @@ rtc_library("system_wrappers") {
   defines = []
   libs = []
   deps = [
-    "../api:array_view",
     "../api/units:time_delta",
     "../api/units:timestamp",
     "../rtc_base:checks",

From 443e7ada92c3247dd294c6d8d1d95217b6dec715 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Sun, 29 Mar 2026 18:33:30 +0200
Subject: [PATCH 442/472] Synchronize final stats collection in E2E quality
 tests
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

The PeerConnectionE2EQualityTest framework previously lost final
metrics, such as payload_bytes_received, because it did not wait for
asynchronous RTCStatsCollector::GetStats() calls to complete during test
teardown.

This change ensures all stats are captured by:
- Tracking pending asynchronous requests in InternalStatsObserver
  using a counter and a new IsPolling() method.
- Implementing a synchronization point in the test runner that uses
  the time controller to wait until StatsPoller reports no active
  polling.
- Adding an optional polling delay to StatsPoller to help simulate
  and verify fixes for race conditions in asynchronous stat
  delivery.

Bug: webrtc:481443652
Change-Id: Ie689bdab4843780c765e8ffd2668d63863328026
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/461000
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47295}
---
 test/pc/e2e/BUILD.gn                          |  5 ++
 test/pc/e2e/peer_connection_quality_test.cc   |  8 ++-
 test/pc/e2e/peer_connection_quality_test.h    |  3 +
 .../e2e/peer_connection_quality_test_test.cc  | 67 ++++++++++++++++++-
 test/pc/e2e/stats_poller.cc                   | 45 +++++++++++--
 test/pc/e2e/stats_poller.h                    | 32 ++++++---
 6 files changed, 144 insertions(+), 16 deletions(-)

diff --git a/test/pc/e2e/BUILD.gn b/test/pc/e2e/BUILD.gn
index 20b8524eeb..d52cae5c5b 100644
--- a/test/pc/e2e/BUILD.gn
+++ b/test/pc/e2e/BUILD.gn
@@ -431,6 +431,9 @@ if (!build_with_chromium) {
         "../../../api:create_network_emulation_manager",
         "../../../api:network_emulation_manager_api",
         "../../../api:peer_connection_quality_test_fixture_api",
+        "../../../api:rtc_stats_api",
+        "../../../api:scoped_refptr",
+        "../../../api:track_id_stream_info_map",
         "../../../api/test/metrics:global_metrics_logger_and_exporter",
         "../../../api/test/network_emulation",
         "../../../api/test/pclf:media_configuration",
@@ -461,6 +464,8 @@ if (!build_with_chromium) {
         "../../../api:rtc_stats_api",
         "../../../api:scoped_refptr",
         "../../../api:stats_observer_interface",
+        "../../../api/task_queue",
+        "../../../api/units:time_delta",
         "../../../rtc_base:logging",
         "../../../rtc_base:macromagic",
         "../../../rtc_base/synchronization:mutex",
diff --git a/test/pc/e2e/peer_connection_quality_test.cc b/test/pc/e2e/peer_connection_quality_test.cc
index 05251f616a..ae2ea68723 100644
--- a/test/pc/e2e/peer_connection_quality_test.cc
+++ b/test/pc/e2e/peer_connection_quality_test.cc
@@ -371,7 +371,8 @@ void PeerConnectionE2EQualityTest::Run(RunParams run_params) {
   StatsPoller stats_poller(observers,
                            std::map<std::string, StatsProvider*>{
                                {*alice_->params().name, alice_.get()},
-                               {*bob_->params().name, bob_.get()}});
+                               {*bob_->params().name, bob_.get()}},
+                           stats_polling_delay_);
   executor_->ScheduleActivity(TimeDelta::Zero(), kStatsUpdateInterval,
                               [&stats_poller](TimeDelta) {
                                 stats_poller.PollStatsAndNotifyObservers();
@@ -415,6 +416,11 @@ void PeerConnectionE2EQualityTest::Run(RunParams run_params) {
     // Get final end-of-call stats.
     stats_poller.PollStatsAndNotifyObservers();
   });
+
+  // Wait until the final stats collection is fully resolved across all peers.
+  ASSERT_TRUE(time_controller_.Wait(
+      [&stats_poller]() { return !stats_poller.IsPolling(); },
+      kDefaultTimeout));
   // We need to detach AEC dumping from peers, because dump uses `task_queue_`
   // inside.
   alice_->DetachAecDump();
diff --git a/test/pc/e2e/peer_connection_quality_test.h b/test/pc/e2e/peer_connection_quality_test.h
index 77bf73507b..1888f66098 100644
--- a/test/pc/e2e/peer_connection_quality_test.h
+++ b/test/pc/e2e/peer_connection_quality_test.h
@@ -86,6 +86,8 @@ class PeerConnectionE2EQualityTest
     return real_test_duration_;
   }
 
+  void SetStatsPollingDelay(TimeDelta delay) { stats_polling_delay_ = delay; }
+
  private:
   class PeerHandleImpl : public PeerHandle {
    public:
@@ -114,6 +116,7 @@ class PeerConnectionE2EQualityTest
   Timestamp Now() const;
 
   TimeController& time_controller_;
+  TimeDelta stats_polling_delay_ = TimeDelta::Zero();
   const std::unique_ptr<TaskQueueFactory> task_queue_factory_;
   std::string test_case_name_;
   std::unique_ptr<VideoQualityAnalyzerInjectionHelper>
diff --git a/test/pc/e2e/peer_connection_quality_test_test.cc b/test/pc/e2e/peer_connection_quality_test_test.cc
index 94e2d58c31..8168b53a6c 100644
--- a/test/pc/e2e/peer_connection_quality_test_test.cc
+++ b/test/pc/e2e/peer_connection_quality_test_test.cc
@@ -18,6 +18,8 @@
 #include <vector>
 
 #include "absl/strings/string_view.h"
+#include "api/scoped_refptr.h"
+#include "api/stats/rtc_stats_report.h"
 #include "api/test/create_network_emulation_manager.h"
 #include "api/test/metrics/global_metrics_logger_and_exporter.h"
 #include "api/test/network_emulation/network_emulation_interfaces.h"
@@ -25,6 +27,8 @@
 #include "api/test/pclf/media_configuration.h"
 #include "api/test/pclf/media_quality_test_params.h"
 #include "api/test/pclf/peer_configurer.h"
+#include "api/test/peerconnection_quality_test_fixture.h"
+#include "api/test/track_id_stream_info_map.h"
 #include "api/units/time_delta.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
@@ -45,7 +49,8 @@ void CleanDir(absl::string_view dir, size_t expected_output_files_count) {
   std::optional<std::vector<std::string>> dir_content =
       test::ReadDirectory(dir);
   if (expected_output_files_count == 0) {
-    ASSERT_FALSE(dir_content.has_value()) << "Empty directory is expected";
+    ASSERT_TRUE(!dir_content.has_value() || dir_content->empty())
+        << "Empty directory is expected";
   } else {
     ASSERT_TRUE(dir_content.has_value()) << "Test directory is empty!";
     EXPECT_EQ(dir_content->size(), expected_output_files_count);
@@ -136,6 +141,66 @@ TEST_F(PeerConnectionE2EQualityTestTest, OutputVideoIsDumpedWhenRequested) {
   ExpectOutputFilesCount(1);
 }
 
+class FinalStatsWaitReporter
+    : public PeerConnectionE2EQualityTestFixture::QualityMetricsReporter {
+ public:
+  void Start(absl::string_view test_case_name,
+             const TrackIdStreamInfoMap* reporter_helper) override {}
+
+  void OnStatsReports(
+      absl::string_view pc_label,
+      const scoped_refptr<const RTCStatsReport>& report) override {
+    ++calls_count_;
+  }
+
+  void StopAndReportResults() override { EXPECT_EQ(calls_count_, 4u); }
+
+ private:
+  size_t calls_count_ = 0;
+};
+
+TEST_F(PeerConnectionE2EQualityTestTest, StatsAreCollectedAtTheEndOfTheTest) {
+  std::unique_ptr<NetworkEmulationManager> network_emulation =
+      CreateNetworkEmulationManager({.time_mode = TimeMode::kSimulated});
+  PeerConnectionE2EQualityTest fixture(
+      "test_case", *network_emulation->time_controller(),
+      /*audio_quality_analyzer=*/nullptr, /*video_quality_analyzer=*/nullptr,
+      test::GetGlobalMetricsLogger());
+  // Inject a delay that simulates stats gathering taking some time.
+  fixture.SetStatsPollingDelay(TimeDelta::Millis(100));
+
+  EmulatedEndpoint* alice_endpoint =
+      network_emulation->CreateEndpoint(EmulatedEndpointConfig());
+  EmulatedEndpoint* bob_endpoint =
+      network_emulation->CreateEndpoint(EmulatedEndpointConfig());
+
+  network_emulation->CreateRoute(
+      alice_endpoint, {network_emulation->CreateUnconstrainedEmulatedNode()},
+      bob_endpoint);
+  network_emulation->CreateRoute(
+      bob_endpoint, {network_emulation->CreateUnconstrainedEmulatedNode()},
+      alice_endpoint);
+
+  EmulatedNetworkManagerInterface* alice_network =
+      network_emulation->CreateEmulatedNetworkManagerInterface(
+          {alice_endpoint});
+  EmulatedNetworkManagerInterface* bob_network =
+      network_emulation->CreateEmulatedNetworkManagerInterface({bob_endpoint});
+
+  PeerConfigurer alice(*alice_network);
+  alice.SetName("alice");
+  alice.AddVideoConfig(VideoConfig("alice_video", 320, 180, 30));
+  fixture.AddPeer(std::make_unique<PeerConfigurer>(std::move(alice)));
+
+  PeerConfigurer bob(*bob_network);
+  bob.SetName("bob");
+  fixture.AddPeer(std::make_unique<PeerConfigurer>(std::move(bob)));
+
+  fixture.AddQualityMetricsReporter(std::make_unique<FinalStatsWaitReporter>());
+
+  fixture.Run(RunParams(TimeDelta::Millis(1)));
+}
+
 }  // namespace
 }  // namespace webrtc_pc_e2e
 }  // namespace webrtc
diff --git a/test/pc/e2e/stats_poller.cc b/test/pc/e2e/stats_poller.cc
index 29533ac637..d2ca30e008 100644
--- a/test/pc/e2e/stats_poller.cc
+++ b/test/pc/e2e/stats_poller.cc
@@ -19,7 +19,9 @@
 #include "api/make_ref_counted.h"
 #include "api/scoped_refptr.h"
 #include "api/stats/rtc_stats_report.h"
+#include "api/task_queue/task_queue_base.h"
 #include "api/test/stats_observer_interface.h"
+#include "api/units/time_delta.h"
 #include "rtc_base/synchronization/mutex.h"
 #include "test/pc/e2e/stats_provider.h"
 #include "test/pc/e2e/test_peer.h"
@@ -28,7 +30,19 @@ namespace webrtc {
 namespace webrtc_pc_e2e {
 
 void InternalStatsObserver::PollStats() {
-  peer_->GetStats(this);
+  {
+    MutexLock lock(&mutex_);
+    ++pending_requests_;
+  }
+  if (stats_delay_.IsZero()) {
+    peer_->GetStats(this);
+  } else {
+    // Artificial delay for testing race conditions.
+    TaskQueueBase::Current()->PostDelayedTask(
+        [this, captured_observer = scoped_refptr<InternalStatsObserver>(
+                   this)]() { peer_->GetStats(captured_observer.get()); },
+        stats_delay_);
+  }
 }
 
 void InternalStatsObserver::OnStatsDelivered(
@@ -36,25 +50,36 @@ void InternalStatsObserver::OnStatsDelivered(
   for (auto* observer : observers_) {
     observer->OnStatsReports(pc_label_, report);
   }
+  {
+    MutexLock lock(&mutex_);
+    --pending_requests_;
+  }
+}
+
+bool InternalStatsObserver::IsPolling() const {
+  MutexLock lock(&mutex_);
+  return pending_requests_ > 0;
 }
 
 StatsPoller::StatsPoller(std::vector<StatsObserverInterface*> observers,
-                         std::map<std::string, StatsProvider*> peers)
+                         std::map<std::string, StatsProvider*> peers,
+                         TimeDelta stats_delay)
     : observers_(std::move(observers)) {
   MutexLock lock(&mutex_);
   for (auto& peer : peers) {
     pollers_.push_back(make_ref_counted<InternalStatsObserver>(
-        peer.first, peer.second, observers_));
+        peer.first, peer.second, observers_, stats_delay));
   }
 }
 
 StatsPoller::StatsPoller(std::vector<StatsObserverInterface*> observers,
-                         std::map<std::string, TestPeer*> peers)
+                         std::map<std::string, TestPeer*> peers,
+                         TimeDelta stats_delay)
     : observers_(std::move(observers)) {
   MutexLock lock(&mutex_);
   for (auto& peer : peers) {
     pollers_.push_back(make_ref_counted<InternalStatsObserver>(
-        peer.first, peer.second, observers_));
+        peer.first, peer.second, observers_, stats_delay));
   }
 }
 
@@ -83,5 +108,15 @@ bool StatsPoller::UnregisterParticipantInCall(absl::string_view peer_name) {
   return false;
 }
 
+bool StatsPoller::IsPolling() const {
+  MutexLock lock(&mutex_);
+  for (const auto& poller : pollers_) {
+    if (poller->IsPolling()) {
+      return true;
+    }
+  }
+  return false;
+}
+
 }  // namespace webrtc_pc_e2e
 }  // namespace webrtc
diff --git a/test/pc/e2e/stats_poller.h b/test/pc/e2e/stats_poller.h
index f9b8890652..541e943ada 100644
--- a/test/pc/e2e/stats_poller.h
+++ b/test/pc/e2e/stats_poller.h
@@ -21,6 +21,7 @@
 #include "api/stats/rtc_stats_collector_callback.h"
 #include "api/stats/rtc_stats_report.h"
 #include "api/test/stats_observer_interface.h"
+#include "api/units/time_delta.h"
 #include "rtc_base/synchronization/mutex.h"
 #include "rtc_base/thread_annotations.h"
 #include "test/pc/e2e/stats_provider.h"
@@ -29,14 +30,18 @@
 namespace webrtc {
 namespace webrtc_pc_e2e {
 
-// Helper class that will notify all the webrtc::test::StatsObserverInterface
+// Helper class that will notify all the test::StatsObserverInterface
 // objects subscribed.
 class InternalStatsObserver : public RTCStatsCollectorCallback {
  public:
   InternalStatsObserver(absl::string_view pc_label,
                         StatsProvider* peer,
-                        std::vector<StatsObserverInterface*> observers)
-      : pc_label_(pc_label), peer_(peer), observers_(std::move(observers)) {}
+                        std::vector<StatsObserverInterface*> observers,
+                        TimeDelta stats_delay = TimeDelta::Zero())
+      : pc_label_(pc_label),
+        peer_(peer),
+        observers_(std::move(observers)),
+        stats_delay_(stats_delay) {}
 
   std::string pc_label() const { return pc_label_; }
 
@@ -45,24 +50,33 @@ class InternalStatsObserver : public RTCStatsCollectorCallback {
   void OnStatsDelivered(
       const scoped_refptr<const RTCStatsReport>& report) override;
 
+  bool IsPolling() const;
+
  private:
   std::string pc_label_;
-  StatsProvider* peer_;
+  StatsProvider* const peer_;
   std::vector<StatsObserverInterface*> observers_;
+  TimeDelta stats_delay_;
+  mutable Mutex mutex_;
+  int pending_requests_ RTC_GUARDED_BY(mutex_) = 0;
 };
 
 // Helper class to invoke GetStats on a PeerConnection by passing a
-// webrtc::StatsObserver that will notify all the
-// webrtc::test::StatsObserverInterface subscribed.
+// StatsObserver that will notify all the test::StatsObserverInterface
+// subscribed.
 class StatsPoller {
  public:
   StatsPoller(std::vector<StatsObserverInterface*> observers,
-              std::map<std::string, StatsProvider*> peers_to_observe);
+              std::map<std::string, StatsProvider*> peers_to_observe,
+              TimeDelta stats_delay = TimeDelta::Zero());
   StatsPoller(std::vector<StatsObserverInterface*> observers,
-              std::map<std::string, TestPeer*> peers_to_observe);
+              std::map<std::string, TestPeer*> peers_to_observe,
+              TimeDelta stats_delay = TimeDelta::Zero());
 
   void PollStatsAndNotifyObservers();
 
+  bool IsPolling() const;
+
   void RegisterParticipantInCall(absl::string_view peer_name,
                                  StatsProvider* peer);
   // Unregister participant from stats poller. Returns true if participant was
@@ -71,7 +85,7 @@ class StatsPoller {
 
  private:
   const std::vector<StatsObserverInterface*> observers_;
-  webrtc::Mutex mutex_;
+  mutable Mutex mutex_;
   std::vector<scoped_refptr<InternalStatsObserver>> pollers_
       RTC_GUARDED_BY(mutex_);
 };

From 36def04dd851b0ea850bb1c9f48603f4f2af971e Mon Sep 17 00:00:00 2001
From: Per K <perkj@webrtc.org>
Date: Mon, 30 Mar 2026 12:34:55 +0000
Subject: [PATCH 443/472] Scream, ensure feedback contains at least one
 received packet

Bug: webrtc:447037083
Change-Id: I62b414c2b4331d81b243584c9a7d367d771e550e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/461080
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47296}
---
 .../scream_network_controller_unittest.cc     | 25 +++++++++++++++++++
 .../congestion_controller/scream/scream_v2.cc |  4 +++
 2 files changed, 29 insertions(+)

diff --git a/modules/congestion_controller/scream/scream_network_controller_unittest.cc b/modules/congestion_controller/scream/scream_network_controller_unittest.cc
index 6331b8dfc0..c5046b3f08 100644
--- a/modules/congestion_controller/scream/scream_network_controller_unittest.cc
+++ b/modules/congestion_controller/scream/scream_network_controller_unittest.cc
@@ -14,6 +14,7 @@
 #include "api/transport/network_control.h"
 #include "api/transport/network_types.h"
 #include "api/units/data_rate.h"
+#include "api/units/data_size.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
 #include "modules/congestion_controller/scream/test/cc_feedback_generator.h"
@@ -573,5 +574,29 @@ TEST(ScreamControllerTest, CanSetStartBitrate) {
   EXPECT_LT(update.target_rate->target_rate, DataRate::KilobitsPerSec(3300));
 }
 
+TEST(ScreamControllerTest, IgnoreFeedbackWithoutReceivedPackets) {
+  SimulatedClock clock(Timestamp::Seconds(1'234));
+  Environment env = CreateTestEnvironment({.time = &clock});
+  NetworkControllerConfig config(env);
+  config.constraints.starting_rate = DataRate::KilobitsPerSec(300);
+  ScreamNetworkController scream_controller(config);
+
+  TransportPacketsFeedback msg;
+  msg.feedback_time = clock.CurrentTime();
+  PacketResult result;
+  result.sent_packet.send_time = clock.CurrentTime();
+  result.sent_packet.sequence_number = 1;
+  result.sent_packet.size = DataSize::Bytes(1000);
+  ASSERT_FALSE(result.IsReceived());
+  msg.packet_feedbacks.push_back(result);
+
+  NetworkControlUpdate update =
+      scream_controller.OnTransportPacketsFeedback(msg);
+  // Scream should not change the target rate if there are no received packets.
+  // Since this is the first feedback, the target rate should be the starting
+  // rate.
+  EXPECT_EQ(update.target_rate->target_rate, DataRate::KilobitsPerSec(300));
+}
+
 }  // namespace
 }  // namespace webrtc
diff --git a/modules/congestion_controller/scream/scream_v2.cc b/modules/congestion_controller/scream/scream_v2.cc
index ea458762cf..786ee7e756 100644
--- a/modules/congestion_controller/scream/scream_v2.cc
+++ b/modules/congestion_controller/scream/scream_v2.cc
@@ -87,6 +87,10 @@ void ScreamV2::OnPacketSent(DataSize data_in_flight) {
 }
 
 void ScreamV2::OnTransportPacketsFeedback(const TransportPacketsFeedback& msg) {
+  if (msg.ReceivedWithSendInfo().empty()) {
+    RTC_LOG(LS_INFO) << "No received packets in feedback, ignoring.";
+    return;
+  }
   max_data_in_flight_this_rtt_ =
       std::max(max_data_in_flight_this_rtt_, msg.data_in_flight);
 

From f1c55505df567d20868076c3c3284cfd63da3fb4 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Erik=20Spr=C3=A5ng?= <sprang@webrtc.org>
Date: Mon, 30 Mar 2026 15:50:02 +0200
Subject: [PATCH 444/472] Move the LibaomAv1Encoder V2 encoder implementation
 to separate file.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This makes it easier to work with than when it's a prive class in the
encoder factory implementation.

Bug: webrtc:496266459
Change-Id: Ib7f167494f8f19a63ac0a490f5c34212e0765cdc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/460380
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47297}
---
 api/video_codecs/BUILD.gn                     |  15 +-
 .../libaom_av1_encoder_factory.cc             | 817 +-----------------
 modules/video_coding/codecs/av1/BUILD.gn      |  30 +
 .../codecs/av1/libaom_av1_encoder_v2.cc       | 807 +++++++++++++++++
 .../codecs/av1/libaom_av1_encoder_v2.h        |  62 ++
 5 files changed, 924 insertions(+), 807 deletions(-)
 create mode 100644 modules/video_coding/codecs/av1/libaom_av1_encoder_v2.cc
 create mode 100644 modules/video_coding/codecs/av1/libaom_av1_encoder_v2.h

diff --git a/api/video_codecs/BUILD.gn b/api/video_codecs/BUILD.gn
index 7106dc197d..2b16682954 100644
--- a/api/video_codecs/BUILD.gn
+++ b/api/video_codecs/BUILD.gn
@@ -385,23 +385,12 @@ rtc_library("libaom_av1_encoder_factory") {
   ]
 
   deps = [
-    ":video_codecs_api",
     ":video_encoder_factory_interface",
     ":video_encoder_interface",
     ":video_encoding_general",
-    "..:array_view",
-    "..:scoped_refptr",
-    "../../api/units:time_delta",
-    "../../rtc_base:checks",
-    "../../rtc_base:logging",
+    "../../api/video:video_frame",
+    "../../modules/video_coding/codecs/av1:libaom_av1_encoder_v2",
     "../../rtc_base:rtc_numerics",
-    "../../rtc_base:stringutils",
-    "../units:data_rate",
-    "../units:data_size",
-    "../video:resolution",
-    "../video:video_frame",
-    "//third_party/abseil-cpp/absl/algorithm:container",
-    "//third_party/abseil-cpp/absl/cleanup",
     "//third_party/libaom",
   ]
 }
diff --git a/api/video_codecs/libaom_av1_encoder_factory.cc b/api/video_codecs/libaom_av1_encoder_factory.cc
index 207b0b142c..2fa1e27861 100644
--- a/api/video_codecs/libaom_av1_encoder_factory.cc
+++ b/api/video_codecs/libaom_av1_encoder_factory.cc
@@ -11,810 +11,19 @@
 #include "api/video_codecs/libaom_av1_encoder_factory.h"
 
 #include <array>
-#include <cstddef>
-#include <cstdint>
-#include <cstring>
 #include <map>
 #include <memory>
-#include <optional>
 #include <string>
-#include <type_traits>
-#include <variant>
-#include <vector>
 
-#include "absl/algorithm/container.h"
-#include "absl/cleanup/cleanup.h"
-#include "api/array_view.h"
-#include "api/scoped_refptr.h"
-#include "api/units/data_rate.h"
-#include "api/units/data_size.h"
-#include "api/units/time_delta.h"
-#include "api/video/resolution.h"
 #include "api/video/video_frame_buffer.h"
-#include "api/video_codecs/video_codec.h"
 #include "api/video_codecs/video_encoder_factory_interface.h"
 #include "api/video_codecs/video_encoder_interface.h"
 #include "api/video_codecs/video_encoding_general.h"
-#include "rtc_base/checks.h"
-#include "rtc_base/logging.h"
+#include "modules/video_coding/codecs/av1/libaom_av1_encoder_v2.h"
 #include "rtc_base/numerics/rational.h"
-#include "rtc_base/strings/string_builder.h"
-#include "third_party/libaom/source/libaom/aom/aom_codec.h"
-#include "third_party/libaom/source/libaom/aom/aom_encoder.h"
-#include "third_party/libaom/source/libaom/aom/aom_image.h"
-#include "third_party/libaom/source/libaom/aom/aomcx.h"
-
-#define SET_OR_RETURN(param_id, param_value)                          \
-  do {                                                                \
-    if (!SetEncoderControlParameters(&ctx_, param_id, param_value)) { \
-      return;                                                         \
-    }                                                                 \
-  } while (0)
-
-#define SET_OR_RETURN_FALSE(param_id, param_value)                    \
-  do {                                                                \
-    if (!SetEncoderControlParameters(&ctx_, param_id, param_value)) { \
-      return false;                                                   \
-    }                                                                 \
-  } while (0)
 
 namespace webrtc {
 
-using FrameEncodeSettings = VideoEncoderInterface::FrameEncodeSettings;
-using Cbr = FrameEncodeSettings::Cbr;
-using Cqp = FrameEncodeSettings::Cqp;
-using aom_img_ptr = std::unique_ptr<aom_image_t, decltype(&aom_img_free)>;
-
-namespace {
-// MaxQp defined here:
-// http://google3/third_party/libaom/git_root/av1/av1_cx_iface.c;l=3510;rcl=527067478
-constexpr int kMaxQp = 63;
-constexpr int kNumBuffers = 8;
-constexpr int kMaxReferences = 3;
-constexpr int kMinEffortLevel = -2;
-constexpr int kMaxEffortLevel = 2;
-constexpr int kMaxSpatialLayersWtf = 4;
-constexpr int kMaxTemporalLayers = 4;
-constexpr int kRtpTicksPerSecond = 90000;
-constexpr std::array<VideoFrameBuffer::Type, 2> kSupportedInputFormats = {
-    VideoFrameBuffer::Type::kI420, VideoFrameBuffer::Type::kNV12};
-
-constexpr std::array<Rational, 7> kSupportedScalingFactors = {
-    {{.numerator = 8, .denominator = 1},
-     {.numerator = 4, .denominator = 1},
-     {.numerator = 2, .denominator = 1},
-     {.numerator = 1, .denominator = 1},
-     {.numerator = 1, .denominator = 2},
-     {.numerator = 1, .denominator = 4},
-     {.numerator = 1, .denominator = 8}}};
-
-std::optional<Rational> GetScalingFactor(const Resolution& from,
-                                         const Resolution& to) {
-  auto it = absl::c_find_if(kSupportedScalingFactors, [&](const Rational& r) {
-    return (from.width * r.numerator / r.denominator) == to.width &&
-           (from.height * r.numerator / r.denominator) == to.height;
-  });
-
-  if (it != kSupportedScalingFactors.end()) {
-    return *it;
-  }
-
-  return {};
-}
-
-class LibaomAv1Encoder : public VideoEncoderInterface {
- public:
-  LibaomAv1Encoder() = default;
-  ~LibaomAv1Encoder() override;
-
-  bool InitEncode(
-      const VideoEncoderFactoryInterface::StaticEncoderSettings& settings,
-      const std::map<std::string, std::string>& encoder_specific_settings);
-
-  void Encode(scoped_refptr<VideoFrameBuffer> frame_buffer,
-              const TemporalUnitSettings& tu_settings,
-              std::vector<FrameEncodeSettings> frame_settings) override;
-
- private:
-  aom_img_ptr image_to_encode_ = aom_img_ptr(nullptr, aom_img_free);
-  aom_codec_ctx_t ctx_;
-  aom_codec_enc_cfg_t cfg_;
-
-  std::optional<VideoCodecMode> current_content_type_;
-  std::array<std::optional<int>, kMaxSpatialLayersWtf> current_effort_level_;
-  int max_number_of_threads_;
-  std::array<std::optional<Resolution>, 8> last_resolution_in_buffer_;
-};
-
-template <typename T>
-bool SetEncoderControlParameters(aom_codec_ctx_t* ctx, int id, T value) {
-  aom_codec_err_t error_code = aom_codec_control(ctx, id, value);
-  if (error_code != AOM_CODEC_OK) {
-    RTC_LOG(LS_WARNING) << "aom_codec_control returned " << error_code
-                        << " with id:  " << id << ".";
-  }
-  return error_code == AOM_CODEC_OK;
-}
-
-LibaomAv1Encoder::~LibaomAv1Encoder() {
-  aom_codec_destroy(&ctx_);
-}
-
-bool LibaomAv1Encoder::InitEncode(
-    const VideoEncoderFactoryInterface::StaticEncoderSettings& settings,
-    const std::map<std::string, std::string>& encoder_specific_settings) {
-  if (!encoder_specific_settings.empty()) {
-    RTC_LOG(LS_ERROR)
-        << "libaom av1 encoder accepts no encoder specific settings";
-    return false;
-  }
-
-  if (aom_codec_err_t ret = aom_codec_enc_config_default(
-          aom_codec_av1_cx(), &cfg_, AOM_USAGE_REALTIME);
-      ret != AOM_CODEC_OK) {
-    RTC_LOG(LS_ERROR) << "aom_codec_enc_config_default returned " << ret;
-    return false;
-  }
-
-  max_number_of_threads_ = settings.max_number_of_threads;
-
-  // The encode resolution is set dynamically for each call to `Encode`, but for
-  // `aom_codec_enc_init` to not fail we set it here as well.
-  cfg_.g_w = settings.max_encode_dimensions.width;
-  cfg_.g_h = settings.max_encode_dimensions.height;
-  cfg_.g_timebase.num = 1;
-  // TD: does 90khz timebase make sense, use microseconds instead maybe?
-  cfg_.g_timebase.den = kRtpTicksPerSecond;
-  cfg_.g_input_bit_depth = settings.encoding_format.bit_depth;
-  cfg_.kf_mode = AOM_KF_DISABLED;
-  // TD: rc_undershoot_pct and rc_overshoot_pct should probably be removed.
-  cfg_.rc_undershoot_pct = 50;
-  cfg_.rc_overshoot_pct = 50;
-  auto* cbr =
-      std::get_if<VideoEncoderFactoryInterface::StaticEncoderSettings::Cbr>(
-          &settings.rc_mode);
-  cfg_.rc_buf_initial_sz = cbr ? cbr->target_buffer_size.ms() : 600;
-  cfg_.rc_buf_optimal_sz = cbr ? cbr->target_buffer_size.ms() : 600;
-  cfg_.rc_buf_sz = cbr ? cbr->max_buffer_size.ms() : 1000;
-  cfg_.g_usage = AOM_USAGE_REALTIME;
-  cfg_.g_pass = AOM_RC_ONE_PASS;
-  cfg_.g_lag_in_frames = 0;
-  cfg_.g_error_resilient = 0;
-  cfg_.rc_end_usage = cbr ? AOM_CBR : AOM_Q;
-
-  if (aom_codec_err_t ret =
-          aom_codec_enc_init(&ctx_, aom_codec_av1_cx(), &cfg_, /*flags=*/0);
-      ret != AOM_CODEC_OK) {
-    RTC_LOG(LS_ERROR) << "aom_codec_enc_init returned " << ret;
-    return false;
-  }
-
-  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_CDEF, 1);
-  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_TPL_MODEL, 0);
-  SET_OR_RETURN_FALSE(AV1E_SET_DELTAQ_MODE, 0);
-  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_ORDER_HINT, 0);
-  SET_OR_RETURN_FALSE(AV1E_SET_AQ_MODE, 3);
-  SET_OR_RETURN_FALSE(AOME_SET_MAX_INTRA_BITRATE_PCT, 300);
-  SET_OR_RETURN_FALSE(AV1E_SET_COEFF_COST_UPD_FREQ, 3);
-  SET_OR_RETURN_FALSE(AV1E_SET_MODE_COST_UPD_FREQ, 3);
-  SET_OR_RETURN_FALSE(AV1E_SET_MV_COST_UPD_FREQ, 3);
-  SET_OR_RETURN_FALSE(AV1E_SET_ROW_MT, 1);
-  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_OBMC, 0);
-  SET_OR_RETURN_FALSE(AV1E_SET_NOISE_SENSITIVITY, 0);
-  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_WARPED_MOTION, 0);
-  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_GLOBAL_MOTION, 0);
-  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_REF_FRAME_MVS, 0);
-  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_CFL_INTRA, 0);
-  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_SMOOTH_INTRA, 0);
-  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_ANGLE_DELTA, 0);
-  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_FILTER_INTRA, 0);
-  SET_OR_RETURN_FALSE(AV1E_SET_INTRA_DEFAULT_TX_ONLY, 1);
-  SET_OR_RETURN_FALSE(AV1E_SET_DISABLE_TRELLIS_QUANT, 1);
-  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_DIST_WTD_COMP, 0);
-  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_DIFF_WTD_COMP, 0);
-  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_DUAL_FILTER, 0);
-  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_INTERINTRA_COMP, 0);
-  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_INTERINTRA_WEDGE, 0);
-  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_INTRA_EDGE_FILTER, 0);
-  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_INTRABC, 0);
-  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_MASKED_COMP, 0);
-  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_PAETH_INTRA, 0);
-  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_QM, 0);
-  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_RECT_PARTITIONS, 0);
-  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_RESTORATION, 0);
-  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_SMOOTH_INTERINTRA, 0);
-  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_TX64, 0);
-  SET_OR_RETURN_FALSE(AV1E_SET_MAX_REFERENCE_FRAMES, 3);
-
-  return true;
-}
-
-struct ThreadTilesAndSuperblockSizeInfo {
-  int num_threads;
-  int exp_tile_rows;
-  int exp_tile_colums;
-  aom_superblock_size_t superblock_size;
-};
-
-ThreadTilesAndSuperblockSizeInfo GetThreadingTilesAndSuperblockSize(
-    int width,
-    int height,
-    int max_number_of_threads) {
-  ThreadTilesAndSuperblockSizeInfo res;
-  const int num_pixels = width * height;
-  if (num_pixels >= 1920 * 1080 && max_number_of_threads > 8) {
-    res.num_threads = 8;
-    res.exp_tile_rows = 2;
-    res.exp_tile_colums = 1;
-  } else if (num_pixels >= 640 * 360 && max_number_of_threads > 4) {
-    res.num_threads = 4;
-    res.exp_tile_rows = 1;
-    res.exp_tile_colums = 1;
-  } else if (num_pixels >= 320 * 180 && max_number_of_threads > 2) {
-    res.num_threads = 2;
-    res.exp_tile_rows = 1;
-    res.exp_tile_colums = 0;
-  } else {
-    res.num_threads = 1;
-    res.exp_tile_rows = 0;
-    res.exp_tile_colums = 0;
-  }
-
-  if (res.num_threads > 4 && num_pixels >= 960 * 540) {
-    res.superblock_size = AOM_SUPERBLOCK_SIZE_64X64;
-  } else {
-    res.superblock_size = AOM_SUPERBLOCK_SIZE_DYNAMIC;
-  }
-
-  RTC_LOG(LS_WARNING) << __FUNCTION__ << " res.num_threads=" << res.num_threads
-                      << " res.exp_tile_rows=" << res.exp_tile_rows
-                      << " res.exp_tile_colums=" << res.exp_tile_colums
-                      << " res.superblock_size=" << res.superblock_size;
-
-  return res;
-}
-
-bool ValidateEncodeParams(
-    const VideoFrameBuffer& /* frame_buffer */,
-    const VideoEncoderInterface::TemporalUnitSettings& /* tu_settings */,
-    const std::vector<VideoEncoderInterface::FrameEncodeSettings>&
-        frame_settings,
-    const std::array<std::optional<Resolution>, 8>& last_resolution_in_buffer,
-    aom_rc_mode rc_mode) {
-  if (frame_settings.empty()) {
-    RTC_LOG(LS_ERROR) << "No frame settings provided.";
-    return false;
-  }
-
-  auto in_range = [](int low, int high, int val) {
-    return low <= val && val < high;
-  };
-
-  for (size_t i = 0; i < frame_settings.size(); ++i) {
-    const VideoEncoderInterface::FrameEncodeSettings& settings =
-        frame_settings[i];
-
-    if (!settings.frame_output) {
-      RTC_LOG(LS_ERROR) << "No frame output provided.";
-      return false;
-    }
-
-    if (!in_range(kMinEffortLevel, kMaxEffortLevel + 1,
-                  settings.effort_level)) {
-      RTC_LOG(LS_ERROR) << "Unsupported effort level " << settings.effort_level;
-      return false;
-    }
-
-    if (!in_range(0, kMaxSpatialLayersWtf, settings.spatial_id)) {
-      RTC_LOG(LS_ERROR) << "invalid spatial id " << settings.spatial_id;
-      return false;
-    }
-
-    if (!in_range(0, kMaxTemporalLayers, settings.temporal_id)) {
-      RTC_LOG(LS_ERROR) << "invalid temporal id " << settings.temporal_id;
-      return false;
-    }
-
-    if ((settings.frame_type == FrameType::kKeyframe ||
-         settings.frame_type == FrameType::kStartFrame) &&
-        !settings.reference_buffers.empty()) {
-      RTC_LOG(LS_ERROR) << "Reference buffers can not be used for keyframes.";
-      return false;
-    }
-
-    if ((settings.frame_type == FrameType::kKeyframe ||
-         settings.frame_type == FrameType::kStartFrame) &&
-        !settings.update_buffer) {
-      RTC_LOG(LS_ERROR)
-          << "Buffer to update must be specified for keyframe/startframe";
-      return false;
-    }
-
-    if (settings.update_buffer &&
-        !in_range(0, kNumBuffers, *settings.update_buffer)) {
-      RTC_LOG(LS_ERROR) << "Invalid update buffer id.";
-      return false;
-    }
-
-    if (settings.reference_buffers.size() > kMaxReferences) {
-      RTC_LOG(LS_ERROR) << "Too many referenced buffers.";
-      return false;
-    }
-
-    for (size_t j = 0; j < settings.reference_buffers.size(); ++j) {
-      if (!in_range(0, kNumBuffers, settings.reference_buffers[j])) {
-        RTC_LOG(LS_ERROR) << "Invalid reference buffer id.";
-        return false;
-      }
-
-      // Figure out which frame resolution a certain buffer will hold when the
-      // frame described by `settings` is encoded.
-      std::optional<Resolution> referenced_resolution;
-      bool keyframe_on_previous_layer = false;
-
-      // Will some other frame in this temporal unit update the buffer?
-      for (size_t k = 0; k < i; ++k) {
-        if (frame_settings[k].frame_type == FrameType::kKeyframe) {
-          keyframe_on_previous_layer = true;
-          referenced_resolution.reset();
-        }
-        if (frame_settings[k].update_buffer == settings.reference_buffers[j]) {
-          referenced_resolution = frame_settings[k].resolution;
-        }
-      }
-
-      // Not updated by another frame in the temporal unit, what is the
-      // resolution of the last frame stored into that buffer?
-      if (!referenced_resolution && !keyframe_on_previous_layer) {
-        referenced_resolution =
-            last_resolution_in_buffer[settings.reference_buffers[j]];
-      }
-
-      if (!referenced_resolution) {
-        RTC_LOG(LS_ERROR) << "Referenced buffer holds no frame.";
-        return false;
-      }
-
-      if (!GetScalingFactor(*referenced_resolution, settings.resolution)) {
-        RTC_LOG(LS_ERROR)
-            << "Required resolution scaling factor not supported.";
-        return false;
-      }
-
-      for (size_t l = i + 1; l < settings.reference_buffers.size(); ++l) {
-        if (settings.reference_buffers[i] == settings.reference_buffers[l]) {
-          RTC_LOG(LS_ERROR) << "Duplicate reference buffer specified.";
-          return false;
-        }
-      }
-    }
-
-    if ((rc_mode == AOM_CBR &&
-         std::holds_alternative<Cqp>(settings.rate_options)) ||
-        (rc_mode == AOM_Q &&
-         std::holds_alternative<Cbr>(settings.rate_options))) {
-      RTC_LOG(LS_ERROR) << "Invalid rate options, encoder configured with "
-                        << (rc_mode == AOM_CBR ? "AOM_CBR" : "AOM_Q");
-      return false;
-    }
-
-    for (size_t j = i + 1; j < frame_settings.size(); ++j) {
-      if (settings.spatial_id >= frame_settings[j].spatial_id) {
-        RTC_LOG(LS_ERROR) << "Frame spatial id specified out of order.";
-        return false;
-      }
-    }
-  }
-
-  return true;
-}
-
-void PrepareInputImage(const VideoFrameBuffer& input_buffer,
-                       aom_img_ptr& out_aom_image) {
-  aom_img_fmt_t input_format;
-  switch (input_buffer.type()) {
-    case VideoFrameBuffer::Type::kI420:
-      input_format = AOM_IMG_FMT_I420;
-      break;
-    case VideoFrameBuffer::Type::kNV12:
-      input_format = AOM_IMG_FMT_NV12;
-      break;
-    default:
-      RTC_CHECK_NOTREACHED();
-      return;
-  }
-
-  if (!out_aom_image || out_aom_image->fmt != input_format ||
-      static_cast<int>(out_aom_image->w) != input_buffer.width() ||
-      static_cast<int>(out_aom_image->h) != input_buffer.height()) {
-    out_aom_image.reset(
-        aom_img_wrap(/*img=*/nullptr, input_format, input_buffer.width(),
-                     input_buffer.height(), /*align=*/1, /*img_data=*/nullptr));
-
-    RTC_LOG(LS_WARNING) << __FUNCTION__ << " input_format=" << input_format
-                        << " input_buffer.width()=" << input_buffer.width()
-                        << " input_buffer.height()=" << input_buffer.height()
-                        << " w=" << out_aom_image->w
-                        << " h=" << out_aom_image->h
-                        << " d_w=" << out_aom_image->d_w
-                        << " d_h=" << out_aom_image->d_h
-                        << " r_w=" << out_aom_image->r_w
-                        << " r_h=" << out_aom_image->r_h;
-  }
-
-  if (input_format == AOM_IMG_FMT_I420) {
-    const I420BufferInterface* i420_buffer = input_buffer.GetI420();
-    RTC_DCHECK(i420_buffer);
-    out_aom_image->planes[AOM_PLANE_Y] =
-        const_cast<unsigned char*>(i420_buffer->DataY());
-    out_aom_image->planes[AOM_PLANE_U] =
-        const_cast<unsigned char*>(i420_buffer->DataU());
-    out_aom_image->planes[AOM_PLANE_V] =
-        const_cast<unsigned char*>(i420_buffer->DataV());
-    out_aom_image->stride[AOM_PLANE_Y] = i420_buffer->StrideY();
-    out_aom_image->stride[AOM_PLANE_U] = i420_buffer->StrideU();
-    out_aom_image->stride[AOM_PLANE_V] = i420_buffer->StrideV();
-  } else {
-    const NV12BufferInterface* nv12_buffer = input_buffer.GetNV12();
-    RTC_DCHECK(nv12_buffer);
-    out_aom_image->planes[AOM_PLANE_Y] =
-        const_cast<unsigned char*>(nv12_buffer->DataY());
-    out_aom_image->planes[AOM_PLANE_U] =
-        const_cast<unsigned char*>(nv12_buffer->DataUV());
-    out_aom_image->planes[AOM_PLANE_V] = nullptr;
-    out_aom_image->stride[AOM_PLANE_Y] = nv12_buffer->StrideY();
-    out_aom_image->stride[AOM_PLANE_U] = nv12_buffer->StrideUV();
-    out_aom_image->stride[AOM_PLANE_V] = 0;
-  }
-}
-
-aom_svc_ref_frame_config_t GetSvcRefFrameConfig(
-    const VideoEncoderInterface::FrameEncodeSettings& settings) {
-  // Buffer alias to use for each position. In particular when there are two
-  // buffers being used, prefer to alias them as LAST and GOLDEN, since the AV1
-  // bitstream format has dedicated fields for them. See last_frame_idx and
-  // golden_frame_idx in the av1 spec
-  // https://aomediacodec.github.io/av1-spec/av1-spec.pdf.
-
-  // Libaom is also compiled for RTC, which limits the number of references to
-  // at most three, and they must be aliased as LAST, GOLDEN and ALTREF. Also
-  // note that libaom favors LAST the most, and GOLDEN second most, so buffers
-  // should be specified in order of how useful they are for prediction. Libaom
-  // could be updated to make LAST, GOLDEN and ALTREF equivalent, but that is
-  // not a priority for now. All aliases can be used to update buffers.
-  // TD: Automatically select LAST, GOLDEN and ALTREF depending on previous
-  //       buffer usage.
-  static constexpr int kPreferedAlias[] = {0,  // LAST
-                                           3,  // GOLDEN
-                                           6,  // ALTREF
-                                           1, 2, 4, 5};
-
-  aom_svc_ref_frame_config_t ref_frame_config = {};
-
-  int alias_index = 0;
-  if (!settings.reference_buffers.empty()) {
-    for (size_t i = 0; i < settings.reference_buffers.size(); ++i) {
-      ref_frame_config.ref_idx[kPreferedAlias[alias_index]] =
-          settings.reference_buffers[i];
-      ref_frame_config.reference[kPreferedAlias[alias_index]] = 1;
-      alias_index++;
-    }
-
-    // Delta frames must not alias unused buffers, and since start frames only
-    // update some buffers it is not safe to leave unused aliases to simply
-    // point to buffer 0.
-    for (size_t i = settings.reference_buffers.size();
-         i < std::size(ref_frame_config.ref_idx); ++i) {
-      ref_frame_config.ref_idx[kPreferedAlias[i]] =
-          settings.reference_buffers.back();
-    }
-  }
-
-  if (settings.update_buffer) {
-    if (!absl::c_linear_search(settings.reference_buffers,
-                               *settings.update_buffer)) {
-      ref_frame_config.ref_idx[kPreferedAlias[alias_index]] =
-          *settings.update_buffer;
-      alias_index++;
-    }
-    ref_frame_config.refresh[*settings.update_buffer] = 1;
-  }
-
-  char buf[256];
-  SimpleStringBuilder sb(buf);
-  sb << " spatial_id=" << settings.spatial_id;
-  sb << "  ref_idx=[ ";
-  for (auto r : ref_frame_config.ref_idx) {
-    sb << r << " ";
-  }
-  sb << "]  reference=[ ";
-  for (auto r : ref_frame_config.reference) {
-    sb << r << " ";
-  }
-  sb << "]  refresh=[ ";
-  for (auto r : ref_frame_config.refresh) {
-    sb << r << " ";
-  }
-  sb << "]";
-
-  RTC_LOG(LS_WARNING) << __FUNCTION__ << sb.str();
-
-  return ref_frame_config;
-}
-
-aom_svc_params_t GetSvcParams(
-    const VideoFrameBuffer& frame_buffer,
-    const std::vector<VideoEncoderInterface::FrameEncodeSettings>&
-        frame_settings) {
-  aom_svc_params_t svc_params = {};
-  svc_params.number_spatial_layers = frame_settings.back().spatial_id + 1;
-  svc_params.number_temporal_layers = kMaxTemporalLayers;
-
-  // TD: What about svc_params.framerate_factor?
-  // If `framerate_factors` are left at 0 then configured bitrate values will
-  // not be picked up by libaom.
-  for (int tid = 0; tid < svc_params.number_temporal_layers; ++tid) {
-    svc_params.framerate_factor[tid] = 1;
-  }
-
-  // If the scaling factor is left at zero for unused layers a division by zero
-  // will happen inside libaom, default all layers to one.
-  for (int sid = 0; sid < svc_params.number_spatial_layers; ++sid) {
-    svc_params.scaling_factor_num[sid] = 1;
-    svc_params.scaling_factor_den[sid] = 1;
-  }
-
-  for (const VideoEncoderInterface::FrameEncodeSettings& settings :
-       frame_settings) {
-    std::optional<Rational> scaling_factor = GetScalingFactor(
-        {.width = frame_buffer.width(), .height = frame_buffer.height()},
-        settings.resolution);
-    RTC_CHECK(scaling_factor);
-    svc_params.scaling_factor_num[settings.spatial_id] =
-        scaling_factor->numerator;
-    svc_params.scaling_factor_den[settings.spatial_id] =
-        scaling_factor->denominator;
-
-    const int flat_layer_id =
-        settings.spatial_id * svc_params.number_temporal_layers +
-        settings.temporal_id;
-
-    RTC_LOG(LS_WARNING) << __FUNCTION__ << " flat_layer_id=" << flat_layer_id
-                        << " num="
-                        << svc_params.scaling_factor_num[settings.spatial_id]
-                        << " den="
-                        << svc_params.scaling_factor_den[settings.spatial_id];
-
-    std::visit(
-        [&](auto&& arg) {
-          using T = std::decay_t<decltype(arg)>;
-          if constexpr (std::is_same_v<T, Cbr>) {
-            // Libaom calculates the total bitrate across all spatial layers by
-            // summing the bitrate of the last temporal layer in each spatial
-            // layer. This means the bitrate for the top temporal layer always
-            // has to be set even if that temporal layer is not being encoded.
-            const int last_temporal_layer_in_spatial_layer_id =
-                settings.spatial_id * svc_params.number_temporal_layers +
-                (kMaxTemporalLayers - 1);
-            svc_params
-                .layer_target_bitrate[last_temporal_layer_in_spatial_layer_id] =
-                arg.target_bitrate.kbps();
-
-            svc_params.layer_target_bitrate[flat_layer_id] =
-                arg.target_bitrate.kbps();
-            // When libaom is configured with `AOM_CBR` it will still limit QP
-            // to stay between `min_quantizers` and `max_quantizers'. Set
-            // `max_quantizers` to max QP to avoid the encoder overshooting.
-            svc_params.max_quantizers[flat_layer_id] = kMaxQp;
-            svc_params.min_quantizers[flat_layer_id] = 0;
-          } else if constexpr (std::is_same_v<T, Cqp>) {
-            // When libaom is configured with `AOM_Q` it will still look at the
-            // `layer_target_bitrate` to determine whether the layer is disabled
-            // or not. Set `layer_target_bitrate` to 1 so that libaom knows the
-            // layer is active.
-            svc_params.layer_target_bitrate[flat_layer_id] = 1;
-            svc_params.max_quantizers[flat_layer_id] = arg.target_qp;
-            svc_params.min_quantizers[flat_layer_id] = arg.target_qp;
-            RTC_LOG(LS_WARNING) << __FUNCTION__ << " svc_params.qp["
-                                << flat_layer_id << "]=" << arg.target_qp;
-            // TD: Does libaom look at both max and min? Shouldn't it just be
-            //       one of them
-          }
-        },
-        settings.rate_options);
-  }
-
-  char buf[512];
-  SimpleStringBuilder sb(buf);
-  sb << "GetSvcParams" << " layer bitrates kbps";
-  for (int s = 0; s < svc_params.number_spatial_layers; ++s) {
-    sb << " S" << s << "=[ ";
-    for (int t = 0; t < svc_params.number_temporal_layers; ++t) {
-      int id = s * svc_params.number_temporal_layers + t;
-      sb << "T" << t << "=" << svc_params.layer_target_bitrate[id] << " ";
-    }
-    sb << "]";
-  }
-
-  RTC_LOG(LS_WARNING) << sb.str();
-
-  return svc_params;
-}
-
-void LibaomAv1Encoder::Encode(scoped_refptr<VideoFrameBuffer> frame_buffer,
-                              const TemporalUnitSettings& tu_settings,
-                              std::vector<FrameEncodeSettings> frame_settings) {
-  absl::Cleanup on_return = [&] {
-    // On return call `EncodeComplete` with EncodingError result unless they
-    // were already called with an EncodedData result.
-    for (FrameEncodeSettings& settings : frame_settings) {
-      if (settings.frame_output) {
-        settings.frame_output->EncodeComplete(EncodingError());
-      }
-    }
-  };
-
-  if (!ValidateEncodeParams(*frame_buffer, tu_settings, frame_settings,
-                            last_resolution_in_buffer_, cfg_.rc_end_usage)) {
-    return;
-  }
-
-  if (current_content_type_ != tu_settings.content_hint) {
-    if (tu_settings.content_hint == VideoCodecMode::kScreensharing) {
-      // TD: Set speed 11?
-      SET_OR_RETURN(AV1E_SET_TUNE_CONTENT, AOM_CONTENT_SCREEN);
-      SET_OR_RETURN(AV1E_SET_ENABLE_PALETTE, 1);
-    } else {
-      SET_OR_RETURN(AV1E_SET_TUNE_CONTENT, AOM_CONTENT_DEFAULT);
-      SET_OR_RETURN(AV1E_SET_ENABLE_PALETTE, 0);
-    }
-    current_content_type_ = tu_settings.content_hint;
-  }
-
-  if (cfg_.rc_end_usage == AOM_CBR) {
-    DataRate accum_rate = DataRate::Zero();
-    for (const FrameEncodeSettings& settings : frame_settings) {
-      accum_rate += std::get<Cbr>(settings.rate_options).target_bitrate;
-    }
-    cfg_.rc_target_bitrate = accum_rate.kbps();
-    RTC_LOG(LS_WARNING) << __FUNCTION__
-                        << " cfg_.rc_target_bitrate=" << cfg_.rc_target_bitrate;
-  }
-
-  if (static_cast<int>(cfg_.g_w) != frame_buffer->width() ||
-      static_cast<int>(cfg_.g_h) != frame_buffer->height()) {
-    RTC_LOG(LS_WARNING) << __FUNCTION__ << " resolution changed from "
-                        << cfg_.g_w << "x" << cfg_.g_h << " to "
-                        << frame_buffer->width() << "x"
-                        << frame_buffer->height();
-    ThreadTilesAndSuperblockSizeInfo ttsbi = GetThreadingTilesAndSuperblockSize(
-        frame_buffer->width(), frame_buffer->height(), max_number_of_threads_);
-    SET_OR_RETURN(AV1E_SET_SUPERBLOCK_SIZE, ttsbi.superblock_size);
-    SET_OR_RETURN(AV1E_SET_TILE_ROWS, ttsbi.exp_tile_rows);
-    SET_OR_RETURN(AV1E_SET_TILE_COLUMNS, ttsbi.exp_tile_colums);
-    cfg_.g_threads = ttsbi.num_threads;
-    cfg_.g_w = frame_buffer->width();
-    cfg_.g_h = frame_buffer->height();
-  }
-
-  PrepareInputImage(*frame_buffer, image_to_encode_);
-
-  // The bitrates caluclated internally in libaom when `AV1E_SET_SVC_PARAMS` is
-  // called depends on the currently configured `cfg_.rc_target_bitrate`. If the
-  // total target bitrate is not updated first a division by zero could happen.
-  if (aom_codec_err_t ret = aom_codec_enc_config_set(&ctx_, &cfg_);
-      ret != AOM_CODEC_OK) {
-    RTC_LOG(LS_ERROR) << "aom_codec_enc_config_set returned " << ret;
-    return;
-  }
-  aom_svc_params_t svc_params = GetSvcParams(*frame_buffer, frame_settings);
-  SET_OR_RETURN(AV1E_SET_SVC_PARAMS, &svc_params);
-
-  // The libaom AV1 encoder requires that `aom_codec_encode` is called for
-  // every spatial layer, even if no frame should be encoded for that layer.
-  std::array<FrameEncodeSettings*, kMaxSpatialLayersWtf>
-      settings_for_spatial_id;
-  settings_for_spatial_id.fill(nullptr);
-  FrameEncodeSettings settings_for_unused_layer;
-  for (FrameEncodeSettings& settings : frame_settings) {
-    settings_for_spatial_id[settings.spatial_id] = &settings;
-  }
-
-  for (int sid = frame_settings[0].spatial_id;
-       sid < svc_params.number_spatial_layers; ++sid) {
-    const bool layer_enabled = settings_for_spatial_id[sid] != nullptr;
-    FrameEncodeSettings& settings = layer_enabled
-                                        ? *settings_for_spatial_id[sid]
-                                        : settings_for_unused_layer;
-
-    aom_svc_layer_id_t layer_id = {
-        .spatial_layer_id = sid,
-        .temporal_layer_id = settings.temporal_id,
-    };
-    SET_OR_RETURN(AV1E_SET_SVC_LAYER_ID, &layer_id);
-    aom_svc_ref_frame_config_t ref_config = GetSvcRefFrameConfig(settings);
-    SET_OR_RETURN(AV1E_SET_SVC_REF_FRAME_CONFIG, &ref_config);
-
-    // TD: Duration can't be zero, what does it matter when the layer is
-    // not being encoded?
-    TimeDelta duration = TimeDelta::Millis(1);
-    if (layer_enabled) {
-      if (const Cbr* cbr = std::get_if<Cbr>(&settings.rate_options)) {
-        duration = cbr->duration;
-      } else {
-        // TD: What should duration be when Cqp is used?
-        duration = TimeDelta::Millis(1);
-      }
-
-      if (settings.effort_level != current_effort_level_[settings.spatial_id]) {
-        // For RTC we use speed level 6 to 10, with 8 being the default. Note
-        // that low effort means higher speed.
-        SET_OR_RETURN(AOME_SET_CPUUSED, 8 - settings.effort_level);
-        current_effort_level_[settings.spatial_id] = settings.effort_level;
-      }
-    }
-
-    RTC_LOG(LS_WARNING)
-        << __FUNCTION__ << " timestamp="
-        << (tu_settings.presentation_timestamp.ms() * kRtpTicksPerSecond / 1000)
-        << "  duration=" << (duration.ms() * kRtpTicksPerSecond / 1000)
-        << "  type="
-        << (settings.frame_type == FrameType::kKeyframe ? "key" : "delta");
-    aom_codec_err_t ret = aom_codec_encode(
-        &ctx_, &*image_to_encode_, tu_settings.presentation_timestamp.ms() * 90,
-        duration.ms() * 90,
-        settings.frame_type == FrameType::kKeyframe ? AOM_EFLAG_FORCE_KF : 0);
-    if (ret != AOM_CODEC_OK) {
-      RTC_LOG(LS_WARNING) << "aom_codec_encode returned " << ret;
-      return;
-    }
-
-    if (!layer_enabled) {
-      continue;
-    }
-
-    if (settings.frame_type == FrameType::kKeyframe) {
-      last_resolution_in_buffer_ = {};
-    }
-
-    if (settings.update_buffer) {
-      last_resolution_in_buffer_[*settings.update_buffer] = settings.resolution;
-    }
-
-    EncodedData result;
-    aom_codec_iter_t iter = nullptr;
-    bool bitstream_produced = false;
-    while (const aom_codec_cx_pkt_t* pkt =
-               aom_codec_get_cx_data(&ctx_, &iter)) {
-      if (pkt->kind == AOM_CODEC_CX_FRAME_PKT && pkt->data.frame.sz > 0) {
-        SET_OR_RETURN(AOME_GET_LAST_QUANTIZER_64, &result.encoded_qp);
-        result.frame_type = pkt->data.frame.flags & AOM_FRAME_IS_KEY
-                                ? FrameType::kKeyframe
-                                : FrameType::kDeltaFrame;
-        ArrayView<uint8_t> output_buffer =
-            settings.frame_output->GetBitstreamOutputBuffer(
-                DataSize::Bytes(pkt->data.frame.sz));
-        if (output_buffer.size() != pkt->data.frame.sz) {
-          return;
-        }
-        memcpy(output_buffer.data(), pkt->data.frame.buf, pkt->data.frame.sz);
-        bitstream_produced = true;
-        break;
-      }
-    }
-
-    if (!bitstream_produced) {
-      return;
-    } else {
-      RTC_CHECK(settings.frame_output);
-      settings.frame_output->EncodeComplete(result);
-      // To avoid invoking any callback more than once.
-      settings.frame_output = nullptr;
-    }
-  }
-}
-}  // namespace
-
 std::string LibaomAv1EncoderFactory::CodecName() const {
   return "AV1";
 }
@@ -832,6 +41,26 @@ std::map<std::string, std::string> LibaomAv1EncoderFactory::CodecSpecifics()
 // The formater and cpplint have conflicting ideas.
 VideoEncoderFactoryInterface::Capabilities
 LibaomAv1EncoderFactory::GetEncoderCapabilities() const {
+  constexpr int kMaxQp = 63;
+  constexpr int kNumBuffers = 8;
+  constexpr int kMaxReferences = 3;
+  constexpr int kMinEffortLevel = -2;
+  constexpr int kMaxEffortLevel = 2;
+  constexpr int kMaxSpatialLayersLimit = 4;
+  constexpr int kMaxTemporalLayers = 4;
+
+  constexpr std::array<VideoFrameBuffer::Type, 2> kSupportedInputFormats = {
+      VideoFrameBuffer::Type::kI420, VideoFrameBuffer::Type::kNV12};
+
+  constexpr std::array<Rational, 7> kSupportedScalingFactors = {
+      {{.numerator = 8, .denominator = 1},
+       {.numerator = 4, .denominator = 1},
+       {.numerator = 2, .denominator = 1},
+       {.numerator = 1, .denominator = 1},
+       {.numerator = 1, .denominator = 2},
+       {.numerator = 1, .denominator = 4},
+       {.numerator = 1, .denominator = 8}}};
+
   return {
       .prediction_constraints = {
            .num_buffers = kNumBuffers,
@@ -839,7 +68,7 @@ LibaomAv1EncoderFactory::GetEncoderCapabilities() const {
            .max_temporal_layers = kMaxTemporalLayers,
            .buffer_space_type = VideoEncoderFactoryInterface::Capabilities::
                PredictionConstraints::BufferSpaceType::kSingleKeyframe,
-           .max_spatial_layers = kMaxSpatialLayersWtf,
+           .max_spatial_layers = kMaxSpatialLayersLimit,
            .scaling_factors = {kSupportedScalingFactors.begin(),
                                kSupportedScalingFactors.end()},
            .supported_frame_types = {FrameType::kKeyframe,
@@ -868,7 +97,7 @@ LibaomAv1EncoderFactory::GetEncoderCapabilities() const {
 std::unique_ptr<VideoEncoderInterface> LibaomAv1EncoderFactory::CreateEncoder(
     const StaticEncoderSettings& settings,
     const std::map<std::string, std::string>& encoder_specific_settings) {
-  auto encoder = std::make_unique<LibaomAv1Encoder>();
+  auto encoder = std::make_unique<LibaomAv1EncoderV2>();
   if (!encoder->InitEncode(settings, encoder_specific_settings)) {
     return nullptr;
   }
diff --git a/modules/video_coding/codecs/av1/BUILD.gn b/modules/video_coding/codecs/av1/BUILD.gn
index 3490a9fcf8..c903ff4367 100644
--- a/modules/video_coding/codecs/av1/BUILD.gn
+++ b/modules/video_coding/codecs/av1/BUILD.gn
@@ -95,6 +95,36 @@ rtc_library("libaom_av1_encoder") {
   ]
 }
 
+rtc_library("libaom_av1_encoder_v2") {
+  visibility = [ "../../../../api/video_codecs:libaom_av1_encoder_factory" ]
+  poisonous = [ "software_video_codecs" ]
+  sources = [
+    "libaom_av1_encoder_v2.cc",
+    "libaom_av1_encoder_v2.h",
+  ]
+  deps = [
+    "../../../../api:array_view",
+    "../../../../api:scoped_refptr",
+    "../../../../api/units:data_rate",
+    "../../../../api/units:data_size",
+    "../../../../api/units:time_delta",
+    "../../../../api/video:resolution",
+    "../../../../api/video:video_codec_constants",
+    "../../../../api/video:video_frame",
+    "../../../../api/video_codecs:video_codecs_api",
+    "../../../../api/video_codecs:video_encoder_factory_interface",
+    "../../../../api/video_codecs:video_encoder_interface",
+    "../../../../api/video_codecs:video_encoding_general",
+    "../../../../rtc_base:checks",
+    "../../../../rtc_base:logging",
+    "../../../../rtc_base:rtc_numerics",
+    "../../../../rtc_base:stringutils",
+    "//third_party/abseil-cpp/absl/algorithm:container",
+    "//third_party/abseil-cpp/absl/cleanup",
+    "//third_party/libaom",
+  ]
+}
+
 rtc_library("libaom_speed_config_factory") {
   sources = [
     "libaom_speed_config_factory.cc",
diff --git a/modules/video_coding/codecs/av1/libaom_av1_encoder_v2.cc b/modules/video_coding/codecs/av1/libaom_av1_encoder_v2.cc
new file mode 100644
index 0000000000..d1009e7e97
--- /dev/null
+++ b/modules/video_coding/codecs/av1/libaom_av1_encoder_v2.cc
@@ -0,0 +1,807 @@
+/*
+ *  Copyright (c) 2026 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/av1/libaom_av1_encoder_v2.h"
+
+#include <array>
+#include <cstddef>
+#include <cstdint>
+#include <cstring>
+#include <map>
+#include <memory>
+#include <optional>
+#include <string>
+#include <type_traits>
+#include <variant>
+#include <vector>
+
+#include "absl/algorithm/container.h"
+#include "absl/cleanup/cleanup.h"
+#include "api/array_view.h"
+#include "api/scoped_refptr.h"
+#include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/video/resolution.h"
+#include "api/video/video_frame_buffer.h"
+#include "api/video_codecs/video_codec.h"
+#include "api/video_codecs/video_encoder_factory_interface.h"
+#include "api/video_codecs/video_encoder_interface.h"
+#include "api/video_codecs/video_encoding_general.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/rational.h"
+#include "rtc_base/strings/string_builder.h"
+#include "third_party/libaom/source/libaom/aom/aom_codec.h"
+#include "third_party/libaom/source/libaom/aom/aom_encoder.h"
+#include "third_party/libaom/source/libaom/aom/aom_image.h"
+#include "third_party/libaom/source/libaom/aom/aomcx.h"
+
+#define SET_OR_RETURN(param_id, param_value)                          \
+  do {                                                                \
+    if (!SetEncoderControlParameters(&ctx_, param_id, param_value)) { \
+      return;                                                         \
+    }                                                                 \
+  } while (0)
+
+#define SET_OR_RETURN_FALSE(param_id, param_value)                    \
+  do {                                                                \
+    if (!SetEncoderControlParameters(&ctx_, param_id, param_value)) { \
+      return false;                                                   \
+    }                                                                 \
+  } while (0)
+
+namespace webrtc {
+
+using FrameEncodeSettings = VideoEncoderInterface::FrameEncodeSettings;
+using Cbr = FrameEncodeSettings::Cbr;
+using Cqp = FrameEncodeSettings::Cqp;
+using aom_img_ptr = std::unique_ptr<aom_image_t, decltype(&aom_img_free)>;
+
+namespace {
+
+constexpr int kMaxQp = 63;
+constexpr int kNumBuffers = 8;
+constexpr int kMaxReferences = 3;
+constexpr int kMinEffortLevel = -2;
+constexpr int kMaxEffortLevel = 2;
+constexpr int kMaxSpatialLayersLimit = 4;
+constexpr int kMaxTemporalLayers = 4;
+constexpr int kRtpTicksPerSecond = 90000;
+
+static_assert(kMaxSpatialLayersLimit <= AOM_MAX_SS_LAYERS);
+static_assert(kMaxTemporalLayers <= AOM_MAX_TS_LAYERS);
+
+constexpr std::array<Rational, 7> kSupportedScalingFactors = {
+    {{.numerator = 8, .denominator = 1},
+     {.numerator = 4, .denominator = 1},
+     {.numerator = 2, .denominator = 1},
+     {.numerator = 1, .denominator = 1},
+     {.numerator = 1, .denominator = 2},
+     {.numerator = 1, .denominator = 4},
+     {.numerator = 1, .denominator = 8}}};
+
+std::optional<Rational> GetScalingFactor(const Resolution& from,
+                                         const Resolution& to) {
+  auto it = absl::c_find_if(kSupportedScalingFactors, [&](const Rational& r) {
+    return (from.width * r.numerator / r.denominator) == to.width &&
+           (from.height * r.numerator / r.denominator) == to.height;
+  });
+
+  if (it != kSupportedScalingFactors.end()) {
+    return *it;
+  }
+
+  return {};
+}
+
+template <typename T>
+bool SetEncoderControlParameters(aom_codec_ctx_t* ctx, int id, T value) {
+  aom_codec_err_t error_code = aom_codec_control(ctx, id, value);
+  if (error_code != AOM_CODEC_OK) {
+    RTC_LOG(LS_WARNING) << "aom_codec_control returned " << error_code
+                        << " with id:  " << id << ".";
+  }
+  return error_code == AOM_CODEC_OK;
+}
+
+struct ThreadTilesAndSuperblockSizeInfo {
+  int num_threads;
+  int exp_tile_rows;
+  int exp_tile_colums;
+  aom_superblock_size_t superblock_size;
+};
+
+ThreadTilesAndSuperblockSizeInfo GetThreadingTilesAndSuperblockSize(
+    int width,
+    int height,
+    int max_number_of_threads) {
+  ThreadTilesAndSuperblockSizeInfo res;
+  const int num_pixels = width * height;
+  if (num_pixels >= 1920 * 1080 && max_number_of_threads > 8) {
+    res.num_threads = 8;
+    res.exp_tile_rows = 2;
+    res.exp_tile_colums = 1;
+  } else if (num_pixels >= 640 * 360 && max_number_of_threads > 4) {
+    res.num_threads = 4;
+    res.exp_tile_rows = 1;
+    res.exp_tile_colums = 1;
+  } else if (num_pixels >= 320 * 180 && max_number_of_threads > 2) {
+    res.num_threads = 2;
+    res.exp_tile_rows = 1;
+    res.exp_tile_colums = 0;
+  } else {
+    res.num_threads = 1;
+    res.exp_tile_rows = 0;
+    res.exp_tile_colums = 0;
+  }
+
+  if (res.num_threads > 4 && num_pixels >= 960 * 540) {
+    res.superblock_size = AOM_SUPERBLOCK_SIZE_64X64;
+  } else {
+    res.superblock_size = AOM_SUPERBLOCK_SIZE_DYNAMIC;
+  }
+
+  RTC_LOG(LS_WARNING) << __FUNCTION__ << " res.num_threads=" << res.num_threads
+                      << " res.exp_tile_rows=" << res.exp_tile_rows
+                      << " res.exp_tile_colums=" << res.exp_tile_colums
+                      << " res.superblock_size=" << res.superblock_size;
+
+  return res;
+}
+
+bool ValidateEncodeParams(
+    const VideoFrameBuffer& /* frame_buffer */,
+    const VideoEncoderInterface::TemporalUnitSettings& /* tu_settings */,
+    const std::vector<FrameEncodeSettings>& frame_settings,
+    const std::array<std::optional<Resolution>, 8>& last_resolution_in_buffer,
+    aom_rc_mode rc_mode) {
+  if (frame_settings.empty()) {
+    RTC_LOG(LS_ERROR) << "No frame settings provided.";
+    return false;
+  }
+
+  auto in_range = [](int low, int high, int val) {
+    return low <= val && val < high;
+  };
+
+  for (size_t i = 0; i < frame_settings.size(); ++i) {
+    const FrameEncodeSettings& settings = frame_settings[i];
+
+    if (!settings.frame_output) {
+      RTC_LOG(LS_ERROR) << "No frame output provided.";
+      return false;
+    }
+
+    if (!in_range(kMinEffortLevel, kMaxEffortLevel + 1,
+                  settings.effort_level)) {
+      RTC_LOG(LS_ERROR) << "Unsupported effort level " << settings.effort_level;
+      return false;
+    }
+
+    if (!in_range(0, kMaxSpatialLayersLimit, settings.spatial_id)) {
+      RTC_LOG(LS_ERROR) << "invalid spatial id " << settings.spatial_id;
+      return false;
+    }
+
+    if (!in_range(0, kMaxTemporalLayers, settings.temporal_id)) {
+      RTC_LOG(LS_ERROR) << "invalid temporal id " << settings.temporal_id;
+      return false;
+    }
+
+    if ((settings.frame_type == VideoEncoderInterface::FrameType::kKeyframe ||
+         settings.frame_type ==
+             VideoEncoderInterface::FrameType::kStartFrame) &&
+        !settings.reference_buffers.empty()) {
+      RTC_LOG(LS_ERROR) << "Reference buffers can not be used for keyframes.";
+      return false;
+    }
+
+    if ((settings.frame_type == VideoEncoderInterface::FrameType::kKeyframe ||
+         settings.frame_type ==
+             VideoEncoderInterface::FrameType::kStartFrame) &&
+        !settings.update_buffer) {
+      RTC_LOG(LS_ERROR)
+          << "Buffer to update must be specified for keyframe/startframe";
+      return false;
+    }
+
+    if (settings.update_buffer &&
+        !in_range(0, kNumBuffers, *settings.update_buffer)) {
+      RTC_LOG(LS_ERROR) << "Invalid update buffer id.";
+      return false;
+    }
+
+    if (settings.reference_buffers.size() > kMaxReferences) {
+      RTC_LOG(LS_ERROR) << "Too many referenced buffers.";
+      return false;
+    }
+
+    for (size_t j = 0; j < settings.reference_buffers.size(); ++j) {
+      if (!in_range(0, kNumBuffers, settings.reference_buffers[j])) {
+        RTC_LOG(LS_ERROR) << "Invalid reference buffer id.";
+        return false;
+      }
+
+      // Figure out which frame resolution a certain buffer will hold when the
+      // frame described by `settings` is encoded.
+      std::optional<Resolution> referenced_resolution;
+      bool keyframe_on_previous_layer = false;
+
+      // Will some other frame in this temporal unit update the buffer?
+      for (size_t k = 0; k < i; ++k) {
+        if (frame_settings[k].frame_type ==
+            VideoEncoderInterface::FrameType::kKeyframe) {
+          keyframe_on_previous_layer = true;
+          referenced_resolution.reset();
+        }
+        if (frame_settings[k].update_buffer == settings.reference_buffers[j]) {
+          referenced_resolution = frame_settings[k].resolution;
+        }
+      }
+
+      // Not updated by another frame in the temporal unit, what is the
+      // resolution of the last frame stored into that buffer?
+      if (!referenced_resolution && !keyframe_on_previous_layer) {
+        referenced_resolution =
+            last_resolution_in_buffer[settings.reference_buffers[j]];
+      }
+
+      if (!referenced_resolution) {
+        RTC_LOG(LS_ERROR) << "Referenced buffer holds no frame.";
+        return false;
+      }
+
+      if (!GetScalingFactor(*referenced_resolution, settings.resolution)) {
+        RTC_LOG(LS_ERROR)
+            << "Required resolution scaling factor not supported.";
+        return false;
+      }
+
+      for (size_t l = i + 1; l < settings.reference_buffers.size(); ++l) {
+        if (settings.reference_buffers[i] == settings.reference_buffers[l]) {
+          RTC_LOG(LS_ERROR) << "Duplicate reference buffer specified.";
+          return false;
+        }
+      }
+    }
+
+    if ((rc_mode == AOM_CBR &&
+         std::holds_alternative<Cqp>(settings.rate_options)) ||
+        (rc_mode == AOM_Q &&
+         std::holds_alternative<Cbr>(settings.rate_options))) {
+      RTC_LOG(LS_ERROR) << "Invalid rate options, encoder configured with "
+                        << (rc_mode == AOM_CBR ? "AOM_CBR" : "AOM_Q");
+      return false;
+    }
+
+    for (size_t j = i + 1; j < frame_settings.size(); ++j) {
+      if (settings.spatial_id >= frame_settings[j].spatial_id) {
+        RTC_LOG(LS_ERROR) << "Frame spatial id specified out of order.";
+        return false;
+      }
+    }
+  }
+
+  return true;
+}
+
+void PrepareInputImage(const VideoFrameBuffer& input_buffer,
+                       aom_img_ptr& out_aom_image) {
+  aom_img_fmt_t input_format;
+  switch (input_buffer.type()) {
+    case VideoFrameBuffer::Type::kI420:
+      input_format = AOM_IMG_FMT_I420;
+      break;
+    case VideoFrameBuffer::Type::kNV12:
+      input_format = AOM_IMG_FMT_NV12;
+      break;
+    default:
+      RTC_CHECK_NOTREACHED();
+      return;
+  }
+
+  if (!out_aom_image || out_aom_image->fmt != input_format ||
+      static_cast<int>(out_aom_image->w) != input_buffer.width() ||
+      static_cast<int>(out_aom_image->h) != input_buffer.height()) {
+    out_aom_image.reset(
+        aom_img_wrap(/*img=*/nullptr, input_format, input_buffer.width(),
+                     input_buffer.height(), /*align=*/1, /*img_data=*/nullptr));
+
+    RTC_LOG(LS_WARNING) << __FUNCTION__ << " input_format=" << input_format
+                        << " input_buffer.width()=" << input_buffer.width()
+                        << " input_buffer.height()=" << input_buffer.height()
+                        << " w=" << out_aom_image->w
+                        << " h=" << out_aom_image->h
+                        << " d_w=" << out_aom_image->d_w
+                        << " d_h=" << out_aom_image->d_h
+                        << " r_w=" << out_aom_image->r_w
+                        << " r_h=" << out_aom_image->r_h;
+  }
+
+  if (input_format == AOM_IMG_FMT_I420) {
+    const I420BufferInterface* i420_buffer = input_buffer.GetI420();
+    RTC_DCHECK(i420_buffer);
+    out_aom_image->planes[AOM_PLANE_Y] =
+        const_cast<unsigned char*>(i420_buffer->DataY());
+    out_aom_image->planes[AOM_PLANE_U] =
+        const_cast<unsigned char*>(i420_buffer->DataU());
+    out_aom_image->planes[AOM_PLANE_V] =
+        const_cast<unsigned char*>(i420_buffer->DataV());
+    out_aom_image->stride[AOM_PLANE_Y] = i420_buffer->StrideY();
+    out_aom_image->stride[AOM_PLANE_U] = i420_buffer->StrideU();
+    out_aom_image->stride[AOM_PLANE_V] = i420_buffer->StrideV();
+  } else {
+    const NV12BufferInterface* nv12_buffer = input_buffer.GetNV12();
+    RTC_DCHECK(nv12_buffer);
+    out_aom_image->planes[AOM_PLANE_Y] =
+        const_cast<unsigned char*>(nv12_buffer->DataY());
+    out_aom_image->planes[AOM_PLANE_U] =
+        const_cast<unsigned char*>(nv12_buffer->DataUV());
+    out_aom_image->planes[AOM_PLANE_V] = nullptr;
+    out_aom_image->stride[AOM_PLANE_Y] = nv12_buffer->StrideY();
+    out_aom_image->stride[AOM_PLANE_U] = nv12_buffer->StrideUV();
+    out_aom_image->stride[AOM_PLANE_V] = 0;
+  }
+}
+
+aom_svc_ref_frame_config_t GetSvcRefFrameConfig(
+    const FrameEncodeSettings& settings) {
+  // Buffer alias to use for each position. In particular when there are two
+  // buffers being used, prefer to alias them as LAST and GOLDEN, since the AV1
+  // bitstream format has dedicated fields for them. See last_frame_idx and
+  // golden_frame_idx in the av1 spec
+  // https://aomediacodec.github.io/av1-spec/av1-spec.pdf.
+
+  // Libaom is also compiled for RTC, which limits the number of references to
+  // at most three, and they must be aliased as LAST, GOLDEN and ALTREF. Also
+  // note that libaom favors LAST the most, and GOLDEN second most, so buffers
+  // should be specified in order of how useful they are for prediction. Libaom
+  // could be updated to make LAST, GOLDEN and ALTREF equivalent, but that is
+  // not a priority for now. All aliases can be used to update buffers.
+  // TD: Automatically select LAST, GOLDEN and ALTREF depending on previous
+  //       buffer usage.
+  static constexpr std::array<int, 7> kPreferedAlias = {0,  // LAST
+                                                        3,  // GOLDEN
+                                                        6,  // ALTREF
+                                                        1, 2, 4, 5};
+
+  aom_svc_ref_frame_config_t ref_frame_config = {};
+  ArrayView<int> ref_idx_view(ref_frame_config.ref_idx, 7);
+  ArrayView<int> reference_view(ref_frame_config.reference, 7);
+  ArrayView<int> refresh_view(ref_frame_config.refresh, 8);
+
+  int alias_index = 0;
+  if (!settings.reference_buffers.empty()) {
+    for (size_t i = 0; i < settings.reference_buffers.size(); ++i) {
+      ref_idx_view[kPreferedAlias[alias_index]] = settings.reference_buffers[i];
+      reference_view[kPreferedAlias[alias_index]] = 1;
+      alias_index++;
+    }
+
+    // Delta frames must not alias unused buffers, and since start frames only
+    // update some buffers it is not safe to leave unused aliases to simply
+    // point to buffer 0.
+    for (size_t i = settings.reference_buffers.size(); i < ref_idx_view.size();
+         ++i) {
+      ref_idx_view[kPreferedAlias[i]] = settings.reference_buffers.back();
+    }
+  }
+
+  if (settings.update_buffer) {
+    if (!absl::c_linear_search(settings.reference_buffers,
+                               *settings.update_buffer)) {
+      ref_idx_view[kPreferedAlias[alias_index]] = *settings.update_buffer;
+      alias_index++;
+    }
+    refresh_view[*settings.update_buffer] = 1;
+  }
+
+  char buf[256];
+  SimpleStringBuilder sb(buf);
+  sb << " spatial_id=" << settings.spatial_id;
+  sb << "  ref_idx=[ ";
+  for (auto r : ref_frame_config.ref_idx) {
+    sb << r << " ";
+  }
+  sb << "]  reference=[ ";
+  for (auto r : ref_frame_config.reference) {
+    sb << r << " ";
+  }
+  sb << "]  refresh=[ ";
+  for (auto r : ref_frame_config.refresh) {
+    sb << r << " ";
+  }
+  sb << "]";
+
+  RTC_LOG(LS_WARNING) << __FUNCTION__ << sb.str();
+
+  return ref_frame_config;
+}
+
+aom_svc_params_t GetSvcParams(
+    const VideoFrameBuffer& frame_buffer,
+    const std::vector<FrameEncodeSettings>& frame_settings) {
+  aom_svc_params_t svc_params = {};
+  svc_params.number_spatial_layers = frame_settings.back().spatial_id + 1;
+  svc_params.number_temporal_layers = kMaxTemporalLayers;
+
+  ArrayView<int> framerate_factor_view(svc_params.framerate_factor,
+                                       AOM_MAX_TS_LAYERS);
+  ArrayView<int> scaling_factor_num_view(svc_params.scaling_factor_num,
+                                         AOM_MAX_SS_LAYERS);
+  ArrayView<int> scaling_factor_den_view(svc_params.scaling_factor_den,
+                                         AOM_MAX_SS_LAYERS);
+  ArrayView<int> layer_target_bitrate_view(svc_params.layer_target_bitrate,
+                                           AOM_MAX_LAYERS);
+  ArrayView<int> max_quantizers_view(svc_params.max_quantizers, AOM_MAX_LAYERS);
+  ArrayView<int> min_quantizers_view(svc_params.min_quantizers, AOM_MAX_LAYERS);
+
+  // TD: What about svc_params.framerate_factor?
+  // If `framerate_factors` are left at 0 then configured bitrate values will
+  // not be picked up by libaom.
+  for (int tid = 0; tid < svc_params.number_temporal_layers; ++tid) {
+    framerate_factor_view[tid] = 1;
+  }
+
+  // If the scaling factor is left at zero for unused layers a division by zero
+  // will happen inside libaom, default all layers to one.
+  for (int sid = 0; sid < svc_params.number_spatial_layers; ++sid) {
+    scaling_factor_num_view[sid] = 1;
+    scaling_factor_den_view[sid] = 1;
+  }
+
+  for (const FrameEncodeSettings& settings : frame_settings) {
+    std::optional<Rational> scaling_factor = GetScalingFactor(
+        {.width = frame_buffer.width(), .height = frame_buffer.height()},
+        settings.resolution);
+    RTC_CHECK(scaling_factor);
+    scaling_factor_num_view[settings.spatial_id] = scaling_factor->numerator;
+    scaling_factor_den_view[settings.spatial_id] = scaling_factor->denominator;
+
+    const int flat_layer_id =
+        settings.spatial_id * svc_params.number_temporal_layers +
+        settings.temporal_id;
+
+    RTC_LOG(LS_WARNING) << __FUNCTION__ << " flat_layer_id=" << flat_layer_id
+                        << " num="
+                        << scaling_factor_num_view[settings.spatial_id]
+                        << " den="
+                        << scaling_factor_den_view[settings.spatial_id];
+
+    std::visit(
+        [&](auto&& arg) {
+          using T = std::decay_t<decltype(arg)>;
+          if constexpr (std::is_same_v<T, Cbr>) {
+            // Libaom calculates the total bitrate across all spatial layers by
+            // summing the bitrate of the last temporal layer in each spatial
+            // layer. This means the bitrate for the top temporal layer always
+            // has to be set even if that temporal layer is not being encoded.
+            const int last_temporal_layer_in_spatial_layer_id =
+                settings.spatial_id * svc_params.number_temporal_layers +
+                (kMaxTemporalLayers - 1);
+            layer_target_bitrate_view[last_temporal_layer_in_spatial_layer_id] =
+                arg.target_bitrate.kbps();
+
+            layer_target_bitrate_view[flat_layer_id] =
+                arg.target_bitrate.kbps();
+            // When libaom is configured with `AOM_CBR` it will still limit QP
+            // to stay between `min_quantizers` and `max_quantizers'. Set
+            // `max_quantizers` to max QP to avoid the encoder overshooting.
+            max_quantizers_view[flat_layer_id] = kMaxQp;
+            min_quantizers_view[flat_layer_id] = 0;
+          } else if constexpr (std::is_same_v<T, Cqp>) {
+            // When libaom is configured with `AOM_Q` it will still look at the
+            // `layer_target_bitrate` to determine whether the layer is disabled
+            // or not. Set `layer_target_bitrate` to 1 so that libaom knows the
+            // layer is active.
+            layer_target_bitrate_view[flat_layer_id] = 1;
+            max_quantizers_view[flat_layer_id] = arg.target_qp;
+            min_quantizers_view[flat_layer_id] = arg.target_qp;
+            // TD: Does libaom look at both max and min? Shouldn't it just be
+            // one of them?
+            RTC_LOG(LS_WARNING) << __FUNCTION__ << " svc_params.qp["
+                                << flat_layer_id << "]=" << arg.target_qp;
+          }
+        },
+        settings.rate_options);
+  }
+
+  char buf[512];
+  SimpleStringBuilder sb(buf);
+  sb << "GetSvcParams" << " layer bitrates kbps";
+  for (int s = 0; s < svc_params.number_spatial_layers; ++s) {
+    sb << " S" << s << "=[ ";
+    for (int t = 0; t < svc_params.number_temporal_layers; ++t) {
+      int id = s * svc_params.number_temporal_layers + t;
+      sb << "T" << t << "=" << layer_target_bitrate_view[id] << " ";
+    }
+    sb << "]";
+  }
+
+  RTC_LOG(LS_WARNING) << sb.str();
+
+  return svc_params;
+}
+
+}  // namespace
+
+LibaomAv1EncoderV2::~LibaomAv1EncoderV2() {
+  aom_codec_destroy(&ctx_);
+}
+
+bool LibaomAv1EncoderV2::InitEncode(
+    const VideoEncoderFactoryInterface::StaticEncoderSettings& settings,
+    const std::map<std::string, std::string>& encoder_specific_settings) {
+  if (!encoder_specific_settings.empty()) {
+    RTC_LOG(LS_ERROR)
+        << "libaom av1 encoder accepts no encoder specific settings";
+    return false;
+  }
+
+  if (aom_codec_err_t ret = aom_codec_enc_config_default(
+          aom_codec_av1_cx(), &cfg_, AOM_USAGE_REALTIME);
+      ret != AOM_CODEC_OK) {
+    RTC_LOG(LS_ERROR) << "aom_codec_enc_config_default returned " << ret;
+    return false;
+  }
+
+  max_number_of_threads_ = settings.max_number_of_threads;
+
+  // The encode resolution is set dynamically for each call to `Encode`, but for
+  // `aom_codec_enc_init` to not fail we set it here as well.
+  cfg_.g_w = settings.max_encode_dimensions.width;
+  cfg_.g_h = settings.max_encode_dimensions.height;
+  cfg_.g_timebase.num = 1;
+  // TD: does 90khz timebase make sense, use microseconds instead maybe?
+  cfg_.g_timebase.den = kRtpTicksPerSecond;
+  cfg_.g_input_bit_depth = settings.encoding_format.bit_depth;
+  cfg_.kf_mode = AOM_KF_DISABLED;
+  // TD: rc_undershoot_pct and rc_overshoot_pct should probably be removed.
+  cfg_.rc_undershoot_pct = 50;
+  cfg_.rc_overshoot_pct = 50;
+  auto* cbr =
+      std::get_if<VideoEncoderFactoryInterface::StaticEncoderSettings::Cbr>(
+          &settings.rc_mode);
+  cfg_.rc_buf_initial_sz = cbr ? cbr->target_buffer_size.ms() : 600;
+  cfg_.rc_buf_optimal_sz = cbr ? cbr->target_buffer_size.ms() : 600;
+  cfg_.rc_buf_sz = cbr ? cbr->max_buffer_size.ms() : 1000;
+  cfg_.g_usage = AOM_USAGE_REALTIME;
+  cfg_.g_pass = AOM_RC_ONE_PASS;
+  cfg_.g_lag_in_frames = 0;
+  cfg_.g_error_resilient = 0;
+  cfg_.rc_end_usage = cbr ? AOM_CBR : AOM_Q;
+
+  if (aom_codec_err_t ret =
+          aom_codec_enc_init(&ctx_, aom_codec_av1_cx(), &cfg_, /*flags=*/0);
+      ret != AOM_CODEC_OK) {
+    RTC_LOG(LS_ERROR) << "aom_codec_enc_init returned " << ret;
+    return false;
+  }
+
+  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_CDEF, 1);
+  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_TPL_MODEL, 0);
+  SET_OR_RETURN_FALSE(AV1E_SET_DELTAQ_MODE, 0);
+  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_ORDER_HINT, 0);
+  SET_OR_RETURN_FALSE(AV1E_SET_AQ_MODE, 3);
+  SET_OR_RETURN_FALSE(AOME_SET_MAX_INTRA_BITRATE_PCT, 300);
+  SET_OR_RETURN_FALSE(AV1E_SET_COEFF_COST_UPD_FREQ, 3);
+  SET_OR_RETURN_FALSE(AV1E_SET_MODE_COST_UPD_FREQ, 3);
+  SET_OR_RETURN_FALSE(AV1E_SET_MV_COST_UPD_FREQ, 3);
+  SET_OR_RETURN_FALSE(AV1E_SET_ROW_MT, 1);
+  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_OBMC, 0);
+  SET_OR_RETURN_FALSE(AV1E_SET_NOISE_SENSITIVITY, 0);
+  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_WARPED_MOTION, 0);
+  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_GLOBAL_MOTION, 0);
+  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_REF_FRAME_MVS, 0);
+  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_CFL_INTRA, 0);
+  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_SMOOTH_INTRA, 0);
+  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_ANGLE_DELTA, 0);
+  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_FILTER_INTRA, 0);
+  SET_OR_RETURN_FALSE(AV1E_SET_INTRA_DEFAULT_TX_ONLY, 1);
+  SET_OR_RETURN_FALSE(AV1E_SET_DISABLE_TRELLIS_QUANT, 1);
+  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_DIST_WTD_COMP, 0);
+  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_DIFF_WTD_COMP, 0);
+  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_DUAL_FILTER, 0);
+  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_INTERINTRA_COMP, 0);
+  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_INTERINTRA_WEDGE, 0);
+  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_INTRA_EDGE_FILTER, 0);
+  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_INTRABC, 0);
+  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_MASKED_COMP, 0);
+  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_PAETH_INTRA, 0);
+  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_QM, 0);
+  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_RECT_PARTITIONS, 0);
+  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_RESTORATION, 0);
+  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_SMOOTH_INTERINTRA, 0);
+  SET_OR_RETURN_FALSE(AV1E_SET_ENABLE_TX64, 0);
+  SET_OR_RETURN_FALSE(AV1E_SET_MAX_REFERENCE_FRAMES, 3);
+
+  return true;
+}
+
+void LibaomAv1EncoderV2::Encode(
+    scoped_refptr<VideoFrameBuffer> frame_buffer,
+    const TemporalUnitSettings& tu_settings,
+    std::vector<FrameEncodeSettings> frame_settings) {
+  // On return call `EncodeComplete` with EncodingError result unless they
+  // were already called with an EncodedData result.
+  absl::Cleanup on_return = [&] {
+    for (FrameEncodeSettings& settings : frame_settings) {
+      if (settings.frame_output) {
+        settings.frame_output->EncodeComplete(
+            VideoEncoderInterface::EncodingError());
+      }
+    }
+  };
+
+  if (!ValidateEncodeParams(*frame_buffer, tu_settings, frame_settings,
+                            last_resolution_in_buffer_, cfg_.rc_end_usage)) {
+    return;
+  }
+
+  if (current_content_type_ != tu_settings.content_hint) {
+    if (tu_settings.content_hint == VideoCodecMode::kScreensharing) {
+      // TD: Set speed 11?
+      SET_OR_RETURN(AV1E_SET_TUNE_CONTENT, AOM_CONTENT_SCREEN);
+      SET_OR_RETURN(AV1E_SET_ENABLE_PALETTE, 1);
+    } else {
+      SET_OR_RETURN(AV1E_SET_TUNE_CONTENT, AOM_CONTENT_DEFAULT);
+      SET_OR_RETURN(AV1E_SET_ENABLE_PALETTE, 0);
+    }
+    current_content_type_ = tu_settings.content_hint;
+  }
+
+  if (cfg_.rc_end_usage == AOM_CBR) {
+    DataRate accum_rate = DataRate::Zero();
+    for (const FrameEncodeSettings& settings : frame_settings) {
+      accum_rate += std::get<Cbr>(settings.rate_options).target_bitrate;
+    }
+    cfg_.rc_target_bitrate = accum_rate.kbps();
+    RTC_LOG(LS_WARNING) << __FUNCTION__
+                        << " cfg_.rc_target_bitrate=" << cfg_.rc_target_bitrate;
+  }
+
+  if (static_cast<int>(cfg_.g_w) != frame_buffer->width() ||
+      static_cast<int>(cfg_.g_h) != frame_buffer->height()) {
+    RTC_LOG(LS_WARNING) << __FUNCTION__ << " resolution changed from "
+                        << cfg_.g_w << "x" << cfg_.g_h << " to "
+                        << frame_buffer->width() << "x"
+                        << frame_buffer->height();
+    ThreadTilesAndSuperblockSizeInfo ttsbi = GetThreadingTilesAndSuperblockSize(
+        frame_buffer->width(), frame_buffer->height(), max_number_of_threads_);
+    SET_OR_RETURN(AV1E_SET_SUPERBLOCK_SIZE, ttsbi.superblock_size);
+    SET_OR_RETURN(AV1E_SET_TILE_ROWS, ttsbi.exp_tile_rows);
+    SET_OR_RETURN(AV1E_SET_TILE_COLUMNS, ttsbi.exp_tile_colums);
+    cfg_.g_threads = ttsbi.num_threads;
+    cfg_.g_w = frame_buffer->width();
+    cfg_.g_h = frame_buffer->height();
+  }
+
+  PrepareInputImage(*frame_buffer, image_to_encode_);
+
+  // The bitrates calculated internally in libaom when `AV1E_SET_SVC_PARAMS` is
+  // called depends on the currently configured `cfg_.rc_target_bitrate`. If the
+  // total target bitrate is not updated first a division by zero could happen.
+  if (aom_codec_err_t ret = aom_codec_enc_config_set(&ctx_, &cfg_);
+      ret != AOM_CODEC_OK) {
+    RTC_LOG(LS_ERROR) << "aom_codec_enc_config_set returned " << ret;
+    return;
+  }
+  aom_svc_params_t svc_params = GetSvcParams(*frame_buffer, frame_settings);
+  SET_OR_RETURN(AV1E_SET_SVC_PARAMS, &svc_params);
+
+  // The libaom AV1 encoder requires that `aom_codec_encode` is called for
+  // every spatial layer, even if no frame should be encoded for that layer.
+  std::array<FrameEncodeSettings*, kMaxSpatialLayersLimit>
+      settings_for_spatial_id;
+  settings_for_spatial_id.fill(nullptr);
+  FrameEncodeSettings settings_for_unused_layer;
+  for (FrameEncodeSettings& settings : frame_settings) {
+    settings_for_spatial_id[settings.spatial_id] = &settings;
+  }
+
+  for (int sid = frame_settings[0].spatial_id;
+       sid < svc_params.number_spatial_layers; ++sid) {
+    const bool layer_enabled = settings_for_spatial_id[sid] != nullptr;
+    FrameEncodeSettings& settings = layer_enabled
+                                        ? *settings_for_spatial_id[sid]
+                                        : settings_for_unused_layer;
+
+    aom_svc_layer_id_t layer_id = {
+        .spatial_layer_id = sid,
+        .temporal_layer_id = settings.temporal_id,
+    };
+    SET_OR_RETURN(AV1E_SET_SVC_LAYER_ID, &layer_id);
+    aom_svc_ref_frame_config_t ref_config = GetSvcRefFrameConfig(settings);
+    SET_OR_RETURN(AV1E_SET_SVC_REF_FRAME_CONFIG, &ref_config);
+
+    // TD: Duration can't be zero, what does it matter when the layer is
+    // not being encoded?
+    TimeDelta duration = TimeDelta::Millis(1);
+    if (layer_enabled) {
+      if (const Cbr* cbr = std::get_if<Cbr>(&settings.rate_options)) {
+        duration = cbr->duration;
+      } else {
+        // TD: What should duration be when Cqp is used?
+        duration = TimeDelta::Millis(1);
+      }
+
+      if (settings.effort_level != current_effort_level_[settings.spatial_id]) {
+        // For RTC we use speed level 6 to 10, with 8 being the default. Note
+        // that low effort means higher speed.
+        SET_OR_RETURN(AOME_SET_CPUUSED, 8 - settings.effort_level);
+        current_effort_level_[settings.spatial_id] = settings.effort_level;
+      }
+    }
+
+    RTC_LOG(LS_WARNING)
+        << __FUNCTION__ << " timestamp="
+        << (tu_settings.presentation_timestamp.ms() * kRtpTicksPerSecond / 1000)
+        << "  duration=" << (duration.ms() * kRtpTicksPerSecond / 1000)
+        << "  type="
+        << (settings.frame_type == VideoEncoderInterface::FrameType::kKeyframe
+                ? "key"
+                : "delta");
+    aom_codec_err_t ret = aom_codec_encode(
+        &ctx_, &*image_to_encode_, tu_settings.presentation_timestamp.ms() * 90,
+        duration.ms() * 90,
+        settings.frame_type == VideoEncoderInterface::FrameType::kKeyframe
+            ? AOM_EFLAG_FORCE_KF
+            : 0);
+    if (ret != AOM_CODEC_OK) {
+      RTC_LOG(LS_WARNING) << "aom_codec_encode returned " << ret;
+      return;
+    }
+
+    if (!layer_enabled) {
+      continue;
+    }
+
+    if (settings.frame_type == VideoEncoderInterface::FrameType::kKeyframe) {
+      last_resolution_in_buffer_ = {};
+    }
+
+    if (settings.update_buffer) {
+      last_resolution_in_buffer_[*settings.update_buffer] = settings.resolution;
+    }
+
+    VideoEncoderInterface::EncodedData result;
+    aom_codec_iter_t iter = nullptr;
+    bool bitstream_produced = false;
+    while (const aom_codec_cx_pkt_t* pkt =
+               aom_codec_get_cx_data(&ctx_, &iter)) {
+      if (pkt->kind == AOM_CODEC_CX_FRAME_PKT && pkt->data.frame.sz > 0) {
+        SET_OR_RETURN(AOME_GET_LAST_QUANTIZER_64, &result.encoded_qp);
+        result.frame_type = pkt->data.frame.flags & AOM_FRAME_IS_KEY
+                                ? VideoEncoderInterface::FrameType::kKeyframe
+                                : VideoEncoderInterface::FrameType::kDeltaFrame;
+        ArrayView<uint8_t> output_buffer =
+            settings.frame_output->GetBitstreamOutputBuffer(
+                DataSize::Bytes(pkt->data.frame.sz));
+        if (output_buffer.size() != pkt->data.frame.sz) {
+          return;
+        }
+        memcpy(output_buffer.data(), pkt->data.frame.buf, pkt->data.frame.sz);
+        bitstream_produced = true;
+        break;
+      }
+    }
+
+    if (!bitstream_produced) {
+      return;
+    } else {
+      RTC_CHECK(settings.frame_output);
+      settings.frame_output->EncodeComplete(result);
+      settings.frame_output = nullptr;
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/modules/video_coding/codecs/av1/libaom_av1_encoder_v2.h b/modules/video_coding/codecs/av1/libaom_av1_encoder_v2.h
new file mode 100644
index 0000000000..75f75bcbe3
--- /dev/null
+++ b/modules/video_coding/codecs/av1/libaom_av1_encoder_v2.h
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2026 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_CODECS_AV1_LIBAOM_AV1_ENCODER_V2_H_
+#define MODULES_VIDEO_CODING_CODECS_AV1_LIBAOM_AV1_ENCODER_V2_H_
+
+#include <array>
+#include <map>
+#include <memory>
+#include <optional>
+#include <string>
+#include <vector>
+
+#include "api/scoped_refptr.h"
+#include "api/video/resolution.h"
+#include "api/video/video_codec_constants.h"
+#include "api/video/video_frame_buffer.h"
+#include "api/video_codecs/video_codec.h"
+#include "api/video_codecs/video_encoder_factory_interface.h"
+#include "api/video_codecs/video_encoder_interface.h"
+#include "third_party/libaom/source/libaom/aom/aom_codec.h"
+#include "third_party/libaom/source/libaom/aom/aom_encoder.h"
+#include "third_party/libaom/source/libaom/aom/aom_image.h"
+
+namespace webrtc {
+
+class LibaomAv1EncoderV2 : public VideoEncoderInterface {
+ public:
+  LibaomAv1EncoderV2() = default;
+  ~LibaomAv1EncoderV2() override;
+
+  bool InitEncode(
+      const VideoEncoderFactoryInterface::StaticEncoderSettings& settings,
+      const std::map<std::string, std::string>& encoder_specific_settings);
+
+  void Encode(scoped_refptr<VideoFrameBuffer> frame_buffer,
+              const TemporalUnitSettings& tu_settings,
+              std::vector<FrameEncodeSettings> frame_settings) override;
+
+ private:
+  using aom_img_ptr = std::unique_ptr<aom_image_t, decltype(&aom_img_free)>;
+
+  aom_img_ptr image_to_encode_ = aom_img_ptr(nullptr, aom_img_free);
+  aom_codec_ctx_t ctx_;
+  aom_codec_enc_cfg_t cfg_;
+
+  std::optional<VideoCodecMode> current_content_type_;
+  std::array<std::optional<int>, kMaxSpatialLayers> current_effort_level_;
+  int max_number_of_threads_;
+  std::array<std::optional<Resolution>, 8> last_resolution_in_buffer_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_CODECS_AV1_LIBAOM_AV1_ENCODER_V2_H_

From f79b11597724fe7c30a9d38705e78456dda36672 Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Mon, 30 Mar 2026 17:25:52 +0200
Subject: [PATCH 445/472] Replace ArrayView with std::span in
 modules/video_coding/av1
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Bug: webrtc:439801349
Change-Id: Ia0a8a5e91c868a844c8d63f9f7de04bda3407149
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/461140
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47298}
---
 modules/video_coding/codecs/av1/BUILD.gn      |  1 -
 .../codecs/av1/libaom_av1_encoder_v2.cc       | 22 +++++++++----------
 2 files changed, 11 insertions(+), 12 deletions(-)

diff --git a/modules/video_coding/codecs/av1/BUILD.gn b/modules/video_coding/codecs/av1/BUILD.gn
index c903ff4367..a5bb945da1 100644
--- a/modules/video_coding/codecs/av1/BUILD.gn
+++ b/modules/video_coding/codecs/av1/BUILD.gn
@@ -103,7 +103,6 @@ rtc_library("libaom_av1_encoder_v2") {
     "libaom_av1_encoder_v2.h",
   ]
   deps = [
-    "../../../../api:array_view",
     "../../../../api:scoped_refptr",
     "../../../../api/units:data_rate",
     "../../../../api/units:data_size",
diff --git a/modules/video_coding/codecs/av1/libaom_av1_encoder_v2.cc b/modules/video_coding/codecs/av1/libaom_av1_encoder_v2.cc
index d1009e7e97..255e8d8807 100644
--- a/modules/video_coding/codecs/av1/libaom_av1_encoder_v2.cc
+++ b/modules/video_coding/codecs/av1/libaom_av1_encoder_v2.cc
@@ -17,6 +17,7 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <type_traits>
 #include <variant>
@@ -24,7 +25,6 @@
 
 #include "absl/algorithm/container.h"
 #include "absl/cleanup/cleanup.h"
-#include "api/array_view.h"
 #include "api/scoped_refptr.h"
 #include "api/units/data_rate.h"
 #include "api/units/data_size.h"
@@ -374,9 +374,9 @@ aom_svc_ref_frame_config_t GetSvcRefFrameConfig(
                                                         1, 2, 4, 5};
 
   aom_svc_ref_frame_config_t ref_frame_config = {};
-  ArrayView<int> ref_idx_view(ref_frame_config.ref_idx, 7);
-  ArrayView<int> reference_view(ref_frame_config.reference, 7);
-  ArrayView<int> refresh_view(ref_frame_config.refresh, 8);
+  std::span<int> ref_idx_view(ref_frame_config.ref_idx, 7);
+  std::span<int> reference_view(ref_frame_config.reference, 7);
+  std::span<int> refresh_view(ref_frame_config.refresh, 8);
 
   int alias_index = 0;
   if (!settings.reference_buffers.empty()) {
@@ -433,16 +433,16 @@ aom_svc_params_t GetSvcParams(
   svc_params.number_spatial_layers = frame_settings.back().spatial_id + 1;
   svc_params.number_temporal_layers = kMaxTemporalLayers;
 
-  ArrayView<int> framerate_factor_view(svc_params.framerate_factor,
+  std::span<int> framerate_factor_view(svc_params.framerate_factor,
                                        AOM_MAX_TS_LAYERS);
-  ArrayView<int> scaling_factor_num_view(svc_params.scaling_factor_num,
+  std::span<int> scaling_factor_num_view(svc_params.scaling_factor_num,
                                          AOM_MAX_SS_LAYERS);
-  ArrayView<int> scaling_factor_den_view(svc_params.scaling_factor_den,
+  std::span<int> scaling_factor_den_view(svc_params.scaling_factor_den,
                                          AOM_MAX_SS_LAYERS);
-  ArrayView<int> layer_target_bitrate_view(svc_params.layer_target_bitrate,
+  std::span<int> layer_target_bitrate_view(svc_params.layer_target_bitrate,
                                            AOM_MAX_LAYERS);
-  ArrayView<int> max_quantizers_view(svc_params.max_quantizers, AOM_MAX_LAYERS);
-  ArrayView<int> min_quantizers_view(svc_params.min_quantizers, AOM_MAX_LAYERS);
+  std::span<int> max_quantizers_view(svc_params.max_quantizers, AOM_MAX_LAYERS);
+  std::span<int> min_quantizers_view(svc_params.min_quantizers, AOM_MAX_LAYERS);
 
   // TD: What about svc_params.framerate_factor?
   // If `framerate_factors` are left at 0 then configured bitrate values will
@@ -782,7 +782,7 @@ void LibaomAv1EncoderV2::Encode(
         result.frame_type = pkt->data.frame.flags & AOM_FRAME_IS_KEY
                                 ? VideoEncoderInterface::FrameType::kKeyframe
                                 : VideoEncoderInterface::FrameType::kDeltaFrame;
-        ArrayView<uint8_t> output_buffer =
+        std::span<uint8_t> output_buffer =
             settings.frame_output->GetBitstreamOutputBuffer(
                 DataSize::Bytes(pkt->data.frame.sz));
         if (output_buffer.size() != pkt->data.frame.sz) {

From 3f11f145666a9d4fb20405518a8f71adda7c052f Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Mon, 30 Mar 2026 21:11:15 -0700
Subject: [PATCH 446/472] Update WebRTC code version (2026-03-31T04:10:44).

Bug: None
Change-Id: I780017500e3d0cb9f6088fce64fd9770bf61f1d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/461200
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47299}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index 8232446b74..a4b2de53d2 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-30T04:12:15";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-31T04:10:44";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From 6075b7e5d8ffa585ad2b2f63ee91c762ecc7240a Mon Sep 17 00:00:00 2001
From: Evan Shrubsole <eshr@webrtc.org>
Date: Tue, 31 Mar 2026 10:48:07 +0000
Subject: [PATCH 447/472] Add gn-check-autofix skill

Bug: webrtc:465491622
Change-Id: If333dd7b35663795864270af3d8292526a6a6964
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/461240
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47300}
---
 .agents/skills/gn-check-autofix/SKILL.md | 72 ++++++++++++++++++++++++
 1 file changed, 72 insertions(+)
 create mode 100644 .agents/skills/gn-check-autofix/SKILL.md

diff --git a/.agents/skills/gn-check-autofix/SKILL.md b/.agents/skills/gn-check-autofix/SKILL.md
new file mode 100644
index 0000000000..67943ae188
--- /dev/null
+++ b/.agents/skills/gn-check-autofix/SKILL.md
@@ -0,0 +1,72 @@
+---
+name: gn-check-autofix
+description: Automatically fix GN check errors in WebRTC BUILD.gn files. Use when encountering "Include not allowed", "rtc_source_set shall not contain cc files", or when needing to clean up non-absolute dependencies.
+---
+
+# GN Check Autofix
+
+This skill provides instructions for using `tools_webrtc/gn_check_autofix.py` to
+automatically resolve common GN build configuration errors in WebRTC.
+
+## Core Workflows
+
+### Fix Missing Dependencies
+
+When you see "Include not allowed" errors from `gn gen --check`, use this
+workflow:
+
+1. Run the autofix tool on your output directory:
+   ```bash
+   tools_webrtc/gn_check_autofix.py -C <dir>
+   ```
+2. The tool will:
+   - Identify targets missing dependencies.
+   - Add the missing `deps` to the appropriate `BUILD.gn` files.
+   - Automatically run `gn format` on modified files.
+
+### Fix rtc_source_set Violations
+
+If you see "rtc_source_set shall not contain cc files", the tool will
+automatically convert them to `rtc_library`:
+
+1. Run the tool:
+   ```bash
+   tools_webrtc/gn_check_autofix.py -C <dir>
+   ```
+
+### Clean Up Dependencies
+
+To remove all non-absolute dependencies (those not starting with `//`) from
+specific `BUILD.gn` files:
+
+```bash
+tools_webrtc/gn_check_autofix.py -r path/to/BUILD.gn
+```
+
+*Note: This preserves absolute dependencies (starting with `//`) and targets
+ending in `_test`, `_tests`, `_unittest`, or `_unittests`.*
+
+## Integration with Include Cleaner
+
+This tool is the recommended second step after running `webrtc-include-cleaner`.
+While the include cleaner updates your C++ source files, `gn-check-autofix`
+synchronizes your `BUILD.gn` files to match.
+
+## Parameters
+
+- `-C <dir>`: Path to a local build directory (e.g., `out/Default`). The tool
+  internally runs `gn gen --check --error-limit=20000`.
+- `-r <files>`: Remove all non-absolute dependencies from the specified files.
+- `--error-limit`: Can be used to override the default error cap.
+
+## Post-Fix Steps
+
+After the tool runs, always verify the changes:
+
+1. **Deduplicate**: The tool may occasionally add a dependency that is already
+   present. Check the `deps` list for duplicates.
+2. **Regenerate GN**: Run `gn gen <dir>` to confirm errors are resolved.
+3. **Format**: The tool runs `gn format`, but a final `git cl format` is
+   recommended for consistency.
+4. **Review**: Check the diff to ensure dependencies were added to the correct
+   targets.

From 0ead3cbf042136747b703ed493acf08418db2741 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Sat, 28 Mar 2026 17:00:13 +0100
Subject: [PATCH 448/472] Use ScopedOperationsBatcher for media description
 pushdown

Modify RtpTransceiver and SdpOfferAnswer to batch media description
updates using ScopedOperationsBatcher. This replaces the previous
approach where each transceiver update triggered a synchronous blocking
call to the worker thread.

Sequentially blocking the signaling thread for each transceiver during
SDP renegotiation can cause performance bottlenecks and audio
impairments, so the responsibility of "backing off" is now with the
ScopedOperationsBatcher class instead of doing individual blocking
calls. By batching these operations, the work on the worker thread is
consolidated in a best-effort manner, and state updates to the signaling
thread are handled via finalizer tasks once the batch completes.

Bug: webrtc:42222804, webrtc:42223005
Change-Id: If826fdd30c1340afb0d426d2a7d1dac864976055
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/458360
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47301}
---
 pc/BUILD.gn            |  1 +
 pc/rtp_transceiver.cc  | 89 +++++++++++++++++++++++-------------------
 pc/rtp_transceiver.h   | 52 +++++++++++++++++++++---
 pc/sdp_offer_answer.cc | 29 ++++++--------
 4 files changed, 110 insertions(+), 61 deletions(-)

diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 5db70e761f..9b04a2f1f7 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -1702,6 +1702,7 @@ rtc_library("rtp_transceiver") {
     ":rtp_sender",
     ":rtp_sender_proxy",
     ":rtp_transport_internal",
+    ":scoped_operations_batcher",
     ":session_description",
     ":video_rtp_receiver",
     "../api:audio_options_api",
diff --git a/pc/rtp_transceiver.cc b/pc/rtp_transceiver.cc
index 010574895d..946601a19c 100644
--- a/pc/rtp_transceiver.cc
+++ b/pc/rtp_transceiver.cc
@@ -62,6 +62,7 @@
 #include "pc/rtp_sender.h"
 #include "pc/rtp_sender_proxy.h"
 #include "pc/rtp_transport_internal.h"
+#include "pc/scoped_operations_batcher.h"
 #include "pc/session_description.h"
 #include "pc/video_rtp_receiver.h"
 #include "rtc_base/checks.h"
@@ -1299,32 +1300,34 @@ bool RtpTransceiver::SetChannelRtpTransport(
   return channel_->SetRtpTransport(rtp_transport);
 }
 
-RTCError RtpTransceiver::SetChannelLocalContent(
+void RtpTransceiver::SetChannelLocalContent(
     const MediaContentDescription* content,
-    SdpType type) {
+    SdpType type,
+    ScopedOperationsBatcher& batcher) {
   RTC_DCHECK_RUN_ON(context()->signaling_thread());
-  return SetChannelContent([&]() {
-    RTC_DCHECK_RUN_ON(context()->worker_thread());
-    return channel_->SetLocalContent(content, type);
-  });
+  SetChannelContent(
+      [this, content, type]() {
+        return channel_->SetLocalContent(content, type);
+      },
+      batcher);
 }
 
-RTCError RtpTransceiver::SetChannelRemoteContent(
+void RtpTransceiver::SetChannelRemoteContent(
     const MediaContentDescription* content,
-    SdpType type) {
+    SdpType type,
+    ScopedOperationsBatcher& batcher) {
   RTC_DCHECK_RUN_ON(context()->signaling_thread());
-  return SetChannelContent([&]() {
-    RTC_DCHECK_RUN_ON(context()->worker_thread());
-    return channel_->SetRemoteContent(content, type);
-  });
+  SetChannelContent(
+      [this, content, type]() {
+        return channel_->SetRemoteContent(content, type);
+      },
+      batcher);
 }
 
-RTCError RtpTransceiver::SetChannelContent(
-    absl::AnyInvocable<RTCError() &&> set_content) {
+void RtpTransceiver::SetChannelContent(
+    absl::AnyInvocable<RTCError() &&> set_content,
+    ScopedOperationsBatcher& batcher) {
   RTC_DCHECK_RUN_ON(context()->signaling_thread());
-  if (!channel_) {
-    return RTCError::InvalidState() << "No channel";
-  }
 
   struct SenderParameters {
     const uint32_t ssrc;
@@ -1339,29 +1342,35 @@ RTCError RtpTransceiver::SetChannelContent(
         {.ssrc = sender->ssrc(), .sender = sender->internal()});
   }
 
-  // Calls the callback on the worker thread, fetches and returns the
-  // RtpParameters for the senders.
-  RTCError result = context()->worker_thread()->BlockingCall([&]() {
-    RTCError error = std::move(set_content)();
-    if (!error.ok()) {
-      return error;
-    }
-    for (auto& entry : sender_parameters) {
-      if (entry.ssrc != 0) {
-        entry.parameters =
-            channel_->media_send_channel()->GetRtpSendParameters(entry.ssrc);
-      }
-    }
-    return RTCError::OK();
-  });
-
-  for (auto& entry : sender_parameters) {
-    if (entry.parameters) {
-      entry.sender->SetCachedParameters(std::move(*entry.parameters));
-    }
-  }
-
-  return result;
+  batcher.AddWithFinalizer(
+      [this, set_content = std::move(set_content),
+       sender_parameters = std::move(sender_parameters)]() mutable
+          -> RTCErrorOr<ScopedOperationsBatcher::FinalizerTask> {
+        RTC_DCHECK_RUN_ON(context()->worker_thread());
+        if (!channel_) {
+          return RTCError::InvalidState() << "No channel";
+        }
+        RTCError result = std::move(set_content)();
+        if (!result.ok()) {
+          return result;
+        }
+        for (auto& entry : sender_parameters) {
+          if (entry.ssrc != 0) {
+            entry.parameters =
+                channel_->media_send_channel()->GetRtpSendParameters(
+                    entry.ssrc);
+          }
+        }
+        return ScopedOperationsBatcher::FinalizerTask(
+            [sender_parameters = std::move(sender_parameters)]() mutable {
+              for (auto& entry : sender_parameters) {
+                if (entry.parameters) {
+                  entry.sender->SetCachedParameters(
+                      std::move(*entry.parameters));
+                }
+              }
+            });
+      });
 }
 
 bool RtpTransceiver::SetChannelPayloadTypeDemuxingEnabled(bool enabled) {
diff --git a/pc/rtp_transceiver.h b/pc/rtp_transceiver.h
index 730978b9d2..a69d17be67 100644
--- a/pc/rtp_transceiver.h
+++ b/pc/rtp_transceiver.h
@@ -65,6 +65,7 @@ namespace webrtc {
 class DtlsTransport;
 
 class PeerConnectionSdpMethods;
+class ScopedOperationsBatcher;
 
 // Implementation of the public RtpTransceiverInterface.
 //
@@ -377,10 +378,20 @@ class RtpTransceiver : public RtpTransceiverInterface {
   }
 
   bool SetChannelRtpTransport(RtpTransportInternal* rtp_transport);
-  RTCError SetChannelLocalContent(const MediaContentDescription* content,
-                                  SdpType type);
-  RTCError SetChannelRemoteContent(const MediaContentDescription* content,
-                                   SdpType type);
+  // Configures the channel with local content description.
+  // Pushes a multi-stage execution task into the provided
+  // `ScopedOperationsBatcher`. See `SetChannelContent` for details on the
+  // execution model.
+  void SetChannelLocalContent(const MediaContentDescription* content,
+                              SdpType type,
+                              ScopedOperationsBatcher& batcher);
+  // Configures the channel with remote content description.
+  // Pushes a multi-stage execution task into the provided
+  // `ScopedOperationsBatcher`. See `SetChannelContent` for details on the
+  // execution model.
+  void SetChannelRemoteContent(const MediaContentDescription* content,
+                               SdpType type,
+                               ScopedOperationsBatcher& batcher);
   bool SetChannelPayloadTypeDemuxingEnabled(bool enabled);
   void EnableChannel(bool enable);
   const std::vector<StreamParams>& channel_local_streams() const;
@@ -431,7 +442,38 @@ class RtpTransceiver : public RtpTransceiverInterface {
   GetOfferedAndImplementedHeaderExtensions(
       const MediaContentDescription* content) const;
 
-  RTCError SetChannelContent(absl::AnyInvocable<RTCError() &&> set_content);
+  // Configures the channel with the provided content description.
+  // Pushes a multi-stage execution task into the provided
+  // `ScopedOperationsBatcher`.
+  //
+  //  Signaling Thread                   Worker Thread
+  //  +-------------------+              +-------------------+
+  //  | SetChannelContent |              |                   |
+  //  | - Capture SSRC    |              |                   |
+  //  | - Push Outer -----|------------->| [Outer Lambda]    |
+  //  |                   |              | - Run set_content |
+  //  |                   |              | - GetRtpSendParams|
+  //  | [Inner Lambda] <--|--------------| - Return Inner    |
+  //  | - SetCachedParams |              |                   |
+  //  +-------------------+              +-------------------+
+  //
+  // Execution Stages:
+  // 1. **Preparation (Signaling Thread)**: Called when this method is invoked.
+  //    Captures current sender SSRC and cached parameters.
+  //
+  // 2. **Execution (Worker Thread)**: The pushed outer lambda is executed
+  //    by the ScopedOperationsBatcher on the worker thread. Runs
+  //    `set_content()`. If successful, fetches updated `RtpSendParameters`
+  //    from the media channel and returns an inner completion lambda.
+  //
+  // 3. **Completion (Signaling Thread)**: The inner lambda returned by the
+  //    worker thread task is collected by `ScopedOperationsBatcher::Run()`.
+  //    After all batched operations on the worker thread are complete,
+  //    `Run()` executes these inner lambdas on the thread that called
+  //    `Run()` (in this case, the Signaling Thread). This stage updates
+  //    the cached parameters on the senders.
+  void SetChannelContent(absl::AnyInvocable<RTCError() &&> set_content,
+                         ScopedOperationsBatcher& batcher);
 
   const Environment env_;
   // Enforce that this object is created, used and destroyed on one thread.
diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc
index 5a3ce99ff3..9ef845c176 100644
--- a/pc/sdp_offer_answer.cc
+++ b/pc/sdp_offer_answer.cc
@@ -5220,25 +5220,22 @@ RTCError SdpOfferAnswerHandler::PushdownMediaDescription(
                            "types, ignoring all.";
     }
 
-    // This for-loop of invokes helps audio impairment during re-negotiations.
-    // One of the causes is that downstairs decoder creation is synchronous at
-    // the moment, and that a decoder is created for each codec listed in the
-    // SDP.
-    //
-    // TODO(bugs.webrtc.org/12840): consider merging the invokes again after
-    // these projects have shipped:
-    // - bugs.webrtc.org/12462
-    // - crbug.com/1157227
-    // - crbug.com/1187289
+    // Batch up the calls to set the local/remote channel content.
+    ScopedOperationsBatcher batcher(context_->worker_thread());
+
     for (const auto& [transceiver, content] : channels) {
-      RTCError error =
-          (source == CS_LOCAL)
-              ? transceiver->SetChannelLocalContent(content, type)
-              : transceiver->SetChannelRemoteContent(content, type);
-      if (!error.ok()) {
-        return error;
+      if (source == CS_LOCAL) {
+        transceiver->SetChannelLocalContent(content, type, batcher);
+      } else {
+        transceiver->SetChannelRemoteContent(content, type, batcher);
       }
     }
+
+    RTCError error = batcher.Run();
+    if (!error.ok()) {
+      return error;
+    }
+
     // If local and remote are both set, we assume that it's safe to trigger
     // CCFB.
     if (pc_->trials().IsEnabled("WebRTC-RFC8888CongestionControlFeedback")) {

From 5a90c2dc89a490af39f8b945d8bc786fa90df65f Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Tue, 31 Mar 2026 16:10:48 +0200
Subject: [PATCH 449/472] Delete deprecated reinterpret_array_view

Bug: webrtc:439801349
Change-Id: I61cdbef2b6f5e30b561766356258a463087db9de
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/461280
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47302}
---
 api/array_view.h           | 19 -------------------
 api/array_view_unittest.cc | 23 -----------------------
 2 files changed, 42 deletions(-)

diff --git a/api/array_view.h b/api/array_view.h
index c670314852..8414ac0471 100644
--- a/api/array_view.h
+++ b/api/array_view.h
@@ -13,7 +13,6 @@
 
 #include <cstddef>
 #include <span>
-#include <type_traits>
 
 namespace webrtc {
 
@@ -26,24 +25,6 @@ inline ArrayView<T> MakeArrayView(T* data, size_t size) {
   return ArrayView<T>(data, size);
 }
 
-// Only for primitive types that have the same size and aligment.
-// Allow reinterpret cast of the array view to another primitive type of the
-// same size.
-// Template arguments order is (U, T, Size) to allow deduction of the template
-// arguments in client calls: reinterpret_array_view<target_type>(array_view).
-template <typename U, typename T, size_t Size>
-[[deprecated]] ArrayView<U, Size> reinterpret_array_view(
-    ArrayView<T, Size> view) {
-  static_assert(sizeof(U) == sizeof(T) && alignof(U) == alignof(T),
-                "ArrayView reinterpret_cast is only supported for casting "
-                "between views that represent the same chunk of memory.");
-  static_assert(
-      std::is_fundamental<T>::value && std::is_fundamental<U>::value,
-      "ArrayView reinterpret_cast is only supported for casting between "
-      "fundamental types.");
-  return ArrayView<U, Size>(reinterpret_cast<U*>(view.data()), view.size());
-}
-
 }  //  namespace webrtc
 
 
diff --git a/api/array_view_unittest.cc b/api/array_view_unittest.cc
index bd7e491984..ae0f830a82 100644
--- a/api/array_view_unittest.cc
+++ b/api/array_view_unittest.cc
@@ -611,27 +611,4 @@ TEST(ArrayViewTest, LastFixed) {
   EXPECT_EQ((av.last<2>()).extent, 2u);
 }
 
-#pragma clang diagnostic push
-#pragma clang diagnostic ignored "-Wdeprecated-declarations"
-TEST(ArrayViewTest, TestReinterpretCastFixedSize) {
-  uint8_t bytes[] = {1, 2, 3};
-  ArrayView<uint8_t, 3> uint8_av(bytes);
-  ArrayView<int8_t, 3> int8_av = reinterpret_array_view<int8_t>(uint8_av);
-  EXPECT_EQ(int8_av.size(), uint8_av.size());
-  EXPECT_EQ(int8_av[0], 1);
-  EXPECT_EQ(int8_av[1], 2);
-  EXPECT_EQ(int8_av[2], 3);
-}
-
-TEST(ArrayViewTest, TestReinterpretCastVariableSize) {
-  std::vector<int8_t> v = {1, 2, 3};
-  ArrayView<int8_t> int8_av(v);
-  ArrayView<uint8_t> uint8_av = reinterpret_array_view<uint8_t>(int8_av);
-  EXPECT_EQ(int8_av.size(), uint8_av.size());
-  EXPECT_EQ(uint8_av[0], 1);
-  EXPECT_EQ(uint8_av[1], 2);
-  EXPECT_EQ(uint8_av[2], 3);
-}
-#pragma clang diagnostic pop
-
 }  // namespace webrtc

From e10cd19640a9bf6f53cd071d1dd58051e1c0b0fc Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Jes=C3=BAs=20de=20Vicente=20Pe=C3=B1a?=
 <devicentepena@webrtc.org>
Date: Tue, 31 Mar 2026 18:54:36 +0200
Subject: [PATCH 450/472] Audio: Enable dynamic on-the-fly AEC3 configuration
 updates for the suppressor gain computation
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This CL refactors suppression gain to allow dynamic configuration changes without requiring reinitialization. This allows switching to a specific ML-REE configuration once the linear mode is used.

These changes are bitexact when not running the neural residual echo estimator. The refinement in MovingAverageSpectrum does not introduce  any change when the residual residual echo estimator is disabled. When running neural residual echo estimator, there is a reduction of initial echo blips for recordings that manifest such issue.

Bug: webrtc:442444736
No-Iwyu: Not changing the tflite includes.
Change-Id: If7c7bdb55ab5a95bb846ea75d45fe5b6387a2fea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456140
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47303}
---
 api/audio/neural_residual_echo_estimator.h    |   5 +
 .../aec3/dominant_nearend_detector.cc         |  29 +--
 .../aec3/dominant_nearend_detector.h          |  10 +-
 .../aec3/echo_canceller3_unittest.cc          |   5 +
 modules/audio_processing/aec3/echo_remover.cc |  25 ++-
 .../aec3/moving_average_spectrum.cc           |  22 ++-
 .../aec3/moving_average_spectrum.h            |  12 +-
 .../aec3/moving_average_spectrum_unittest.cc  |  36 +++-
 .../audio_processing/aec3/nearend_detector.h  |   4 +
 .../neural_residual_echo_estimator_impl.cc    |  27 ++-
 .../neural_residual_echo_estimator_impl.h     |   2 +
 .../aec3/residual_echo_estimator.cc           |  80 ++++----
 .../aec3/residual_echo_estimator.h            |   4 +
 .../aec3/subband_nearend_detector.cc          |  13 ++
 .../aec3/subband_nearend_detector.h           |   9 +-
 .../audio_processing/aec3/suppression_gain.cc | 179 +++++++++++-------
 .../audio_processing/aec3/suppression_gain.h  |  35 ++--
 .../aec3/suppression_gain_unittest.cc         |  58 ++++--
 .../test/audioproc_float_impl.cc              |   6 +-
 19 files changed, 370 insertions(+), 191 deletions(-)

diff --git a/api/audio/neural_residual_echo_estimator.h b/api/audio/neural_residual_echo_estimator.h
index 93ea2fae89..87e814906e 100644
--- a/api/audio/neural_residual_echo_estimator.h
+++ b/api/audio/neural_residual_echo_estimator.h
@@ -57,6 +57,11 @@ class NeuralResidualEchoEstimator {
   // Returns a recommended AEC3 configuration for this estimator.
   virtual EchoCanceller3Config GetConfiguration(bool multi_channel) const = 0;
 
+  // Adjusts the provided AEC3 suppressor configuration based on the estimator's
+  // requirements.
+  virtual EchoCanceller3Config::Suppressor AdjustConfig(
+      const EchoCanceller3Config::Suppressor& config) const = 0;
+
   // Resets the internal state of the estimator.
   virtual void Reset() = 0;
 };
diff --git a/modules/audio_processing/aec3/dominant_nearend_detector.cc b/modules/audio_processing/aec3/dominant_nearend_detector.cc
index 35aff9341b..d7e01fbaeb 100644
--- a/modules/audio_processing/aec3/dominant_nearend_detector.cc
+++ b/modules/audio_processing/aec3/dominant_nearend_detector.cc
@@ -24,12 +24,7 @@ namespace webrtc {
 DominantNearendDetector::DominantNearendDetector(
     const EchoCanceller3Config::Suppressor::DominantNearendDetection& config,
     size_t num_capture_channels)
-    : enr_threshold_(config.enr_threshold),
-      enr_exit_threshold_(config.enr_exit_threshold),
-      snr_threshold_(config.snr_threshold),
-      hold_duration_(config.hold_duration),
-      trigger_threshold_(config.trigger_threshold),
-      use_during_initial_phase_(config.use_during_initial_phase),
+    : config_(config),
       num_capture_channels_(num_capture_channels),
       trigger_counters_(num_capture_channels_),
       hold_counters_(num_capture_channels_) {}
@@ -55,13 +50,13 @@ void DominantNearendDetector::Update(
 
     // Detect strong active nearend if the nearend is sufficiently stronger than
     // the echo and the nearend noise.
-    if ((!initial_state || use_during_initial_phase_) &&
-        echo_sum < enr_threshold_ * ne_sum &&
-        ne_sum > snr_threshold_ * noise_sum) {
-      if (++trigger_counters_[ch] >= trigger_threshold_) {
+    if ((!initial_state || config_.use_during_initial_phase) &&
+        echo_sum < config_.enr_threshold * ne_sum &&
+        ne_sum > config_.snr_threshold * noise_sum) {
+      if (++trigger_counters_[ch] >= config_.trigger_threshold) {
         // After a period of strong active nearend activity, flag nearend mode.
-        hold_counters_[ch] = hold_duration_;
-        trigger_counters_[ch] = trigger_threshold_;
+        hold_counters_[ch] = config_.hold_duration;
+        trigger_counters_[ch] = config_.trigger_threshold;
       }
     } else {
       // Forget previously detected strong active nearend activity.
@@ -69,8 +64,8 @@ void DominantNearendDetector::Update(
     }
 
     // Exit nearend-state early at strong echo.
-    if (echo_sum > enr_exit_threshold_ * ne_sum &&
-        echo_sum > snr_threshold_ * noise_sum) {
+    if (echo_sum > config_.enr_exit_threshold * ne_sum &&
+        echo_sum > config_.snr_threshold * noise_sum) {
       hold_counters_[ch] = 0;
     }
 
@@ -79,4 +74,10 @@ void DominantNearendDetector::Update(
     nearend_state_ = nearend_state_ || hold_counters_[ch] > 0;
   }
 }
+
+void DominantNearendDetector::SetConfig(
+    const EchoCanceller3Config::Suppressor& config) {
+  config_ = config.dominant_nearend_detection;
+}
+
 }  // namespace webrtc
diff --git a/modules/audio_processing/aec3/dominant_nearend_detector.h b/modules/audio_processing/aec3/dominant_nearend_detector.h
index cbab36e3a4..877697c2e4 100644
--- a/modules/audio_processing/aec3/dominant_nearend_detector.h
+++ b/modules/audio_processing/aec3/dominant_nearend_detector.h
@@ -40,13 +40,11 @@ class DominantNearendDetector : public NearendDetector {
           comfort_noise_spectrum,
       bool initial_state) override;
 
+  // Sets the configuration.
+  void SetConfig(const EchoCanceller3Config::Suppressor& config) override;
+
  private:
-  const float enr_threshold_;
-  const float enr_exit_threshold_;
-  const float snr_threshold_;
-  const int hold_duration_;
-  const int trigger_threshold_;
-  const bool use_during_initial_phase_;
+  EchoCanceller3Config::Suppressor::DominantNearendDetection config_;
   const size_t num_capture_channels_;
 
   bool nearend_state_ = false;
diff --git a/modules/audio_processing/aec3/echo_canceller3_unittest.cc b/modules/audio_processing/aec3/echo_canceller3_unittest.cc
index 69d71ad810..a27b3e899a 100644
--- a/modules/audio_processing/aec3/echo_canceller3_unittest.cc
+++ b/modules/audio_processing/aec3/echo_canceller3_unittest.cc
@@ -1195,6 +1195,11 @@ TEST(EchoCanceller3, InjectedNeuralResidualEchoEstimatorIsUsed) {
       return EchoCanceller3Config();
     }
 
+    EchoCanceller3Config::Suppressor AdjustConfig(
+        const EchoCanceller3Config::Suppressor& config) const override {
+      return config;
+    }
+
     MOCK_METHOD(void, Reset, (), (override));
 
    private:
diff --git a/modules/audio_processing/aec3/echo_remover.cc b/modules/audio_processing/aec3/echo_remover.cc
index 0d3e1d1c7a..9919b6fe4f 100644
--- a/modules/audio_processing/aec3/echo_remover.cc
+++ b/modules/audio_processing/aec3/echo_remover.cc
@@ -184,6 +184,8 @@ class EchoRemoverImpl final : public EchoRemover {
  private:
   static std::atomic<int> instance_count_;
   const EchoCanceller3Config config_;
+  const std::optional<EchoCanceller3Config::Suppressor>
+      ml_ree_suppressor_config_;
   const Aec3Fft fft_;
   std::unique_ptr<ApmDataDumper> data_dumper_;
   const Aec3Optimization optimization_;
@@ -206,6 +208,7 @@ class EchoRemoverImpl final : public EchoRemover {
   size_t block_counter_ = 0;
   int gain_change_hangover_ = 0;
   bool refined_filter_output_last_selected_ = true;
+  bool ml_ree_was_active_ = false;
 
   std::vector<std::array<float, kFftLengthBy2>> e_heap_;
   std::vector<std::array<float, kFftLengthBy2Plus1>> Y2_heap_;
@@ -230,6 +233,11 @@ EchoRemoverImpl::EchoRemoverImpl(
     size_t num_capture_channels,
     NeuralResidualEchoEstimator* neural_residual_echo_estimator)
     : config_(config),
+      ml_ree_suppressor_config_(
+          neural_residual_echo_estimator
+              ? std::make_optional(neural_residual_echo_estimator->AdjustConfig(
+                    config.suppressor))
+              : std::nullopt),
       fft_(),
       data_dumper_(new ApmDataDumper(instance_count_.fetch_add(1) + 1)),
       optimization_(DetectOptimization()),
@@ -416,9 +424,11 @@ void EchoRemoverImpl::ProcessCapture(
   subtractor_.Process(*render_buffer, *y, render_signal_analyzer_, aec_state_,
                       subtractor_output);
 
+  const bool ml_ree_is_active = residual_echo_estimator_.IsMlReeActive();
+
   // Compute spectra.
   bool use_refined_output;
-  if (use_coarse_filter_output_) {
+  if (use_coarse_filter_output_ && !ml_ree_is_active) {
     use_refined_output = false;
     for (size_t ch = 0; ch < num_capture_channels_; ++ch) {
       if (UseRefinedOutput(subtractor_output[ch])) {
@@ -499,9 +509,20 @@ void EchoRemoverImpl::ProcessCapture(
     const bool clock_drift = config_.echo_removal_control.has_clock_drift ||
                              echo_path_variability.clock_drift;
 
+    // Select the active suppressor configuration.
+    const EchoCanceller3Config::Suppressor& active_suppressor_config =
+        (ml_ree_is_active && ml_ree_suppressor_config_.has_value())
+            ? *ml_ree_suppressor_config_
+            : config_.suppressor;
+
+    // Determine if the configuration affecting the suppressor has changed.
+    const bool config_changed = (ml_ree_is_active != ml_ree_was_active_);
+    ml_ree_was_active_ = ml_ree_is_active;
+
     // Compute preferred gains.
     float high_bands_gain;
-    suppression_gain_.GetGain(nearend_spectrum, echo_spectrum, R2, R2_unbounded,
+    suppression_gain_.GetGain(active_suppressor_config, config_changed,
+                              nearend_spectrum, echo_spectrum, R2, R2_unbounded,
                               cng_.NoiseSpectrum(), render_signal_analyzer_,
                               aec_state_, x, clock_drift, &high_bands_gain, &G);
 
diff --git a/modules/audio_processing/aec3/moving_average_spectrum.cc b/modules/audio_processing/aec3/moving_average_spectrum.cc
index 3884c4a57a..fb469182e1 100644
--- a/modules/audio_processing/aec3/moving_average_spectrum.cc
+++ b/modules/audio_processing/aec3/moving_average_spectrum.cc
@@ -23,9 +23,9 @@ namespace webrtc {
 MovingAverageSpectrum::MovingAverageSpectrum(size_t num_elem, size_t mem_len)
     : num_elem_(num_elem),
       mem_len_(mem_len - 1),
-      scaling_(1.0f / static_cast<float>(mem_len)),
       memory_(num_elem * mem_len_, 0.f),
-      mem_index_(0) {
+      mem_index_(0),
+      number_updates_(0) {
   RTC_DCHECK(num_elem_ > 0);
   RTC_DCHECK(mem_len > 0);
 }
@@ -44,9 +44,10 @@ void MovingAverageSpectrum::Average(std::span<const float> input,
                    std::plus<float>());
   }
 
-  // Divide by mem_len_.
+  // Divide by the number of points used to compute the average.
+  const float scaling = 1.0f / static_cast<float>(number_updates_ + 1);
   for (float& o : output) {
-    o *= scaling_;
+    o *= scaling;
   }
 
   // Update memory.
@@ -55,6 +56,19 @@ void MovingAverageSpectrum::Average(std::span<const float> input,
               memory_.begin() + mem_index_ * num_elem_);
     mem_index_ = (mem_index_ + 1) % mem_len_;
   }
+  number_updates_ = std::min(static_cast<int>(mem_len_), number_updates_ + 1);
+}
+
+void MovingAverageSpectrum::UpdateMemoryLength(size_t mem_len) {
+  if (mem_len_ + 1 == mem_len) {
+    return;
+  }
+  RTC_DCHECK(mem_len > 0);
+  mem_len_ = mem_len - 1;
+  memory_.resize(num_elem_ * mem_len_);
+  std::fill(memory_.begin(), memory_.end(), 0.0f);
+  mem_index_ = 0;
+  number_updates_ = 0;
 }
 
 }  // namespace webrtc
diff --git a/modules/audio_processing/aec3/moving_average_spectrum.h b/modules/audio_processing/aec3/moving_average_spectrum.h
index b4a3763311..e8a2909ee3 100644
--- a/modules/audio_processing/aec3/moving_average_spectrum.h
+++ b/modules/audio_processing/aec3/moving_average_spectrum.h
@@ -25,16 +25,20 @@ class MovingAverageSpectrum {
   MovingAverageSpectrum(size_t num_elem, size_t mem_len);
   ~MovingAverageSpectrum();
 
-  // Computes the average of input and mem_len-1 previous inputs and stores the
-  // result in output.
+  // Computes the average of the current input and up to mem_len-1 previous
+  // inputs and stores the result in output.
   void Average(std::span<const float> input, std::span<float> output);
 
+  // If a new memory length is provided, resets the state and clears the
+  // average memory to use the new window size.
+  void UpdateMemoryLength(size_t mem_len);
+
  private:
   const size_t num_elem_;
-  const size_t mem_len_;
-  const float scaling_;
+  size_t mem_len_;
   std::vector<float> memory_;
   size_t mem_index_;
+  int number_updates_;
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_processing/aec3/moving_average_spectrum_unittest.cc b/modules/audio_processing/aec3/moving_average_spectrum_unittest.cc
index 718069b611..5dbfca2d40 100644
--- a/modules/audio_processing/aec3/moving_average_spectrum_unittest.cc
+++ b/modules/audio_processing/aec3/moving_average_spectrum_unittest.cc
@@ -29,16 +29,16 @@ TEST(MovingAverageSpectrum, Average) {
   std::array<float, num_elem> output;
 
   ma.Average(data1, output);
-  EXPECT_NEAR(output[0], data1[0] / 3.0f, e);
-  EXPECT_NEAR(output[1], data1[1] / 3.0f, e);
-  EXPECT_NEAR(output[2], data1[2] / 3.0f, e);
-  EXPECT_NEAR(output[3], data1[3] / 3.0f, e);
+  EXPECT_NEAR(output[0], data1[0] / 1.0f, e);
+  EXPECT_NEAR(output[1], data1[1] / 1.0f, e);
+  EXPECT_NEAR(output[2], data1[2] / 1.0f, e);
+  EXPECT_NEAR(output[3], data1[3] / 1.0f, e);
 
   ma.Average(data2, output);
-  EXPECT_NEAR(output[0], (data1[0] + data2[0]) / 3.0f, e);
-  EXPECT_NEAR(output[1], (data1[1] + data2[1]) / 3.0f, e);
-  EXPECT_NEAR(output[2], (data1[2] + data2[2]) / 3.0f, e);
-  EXPECT_NEAR(output[3], (data1[3] + data2[3]) / 3.0f, e);
+  EXPECT_NEAR(output[0], (data1[0] + data2[0]) / 2.0f, e);
+  EXPECT_NEAR(output[1], (data1[1] + data2[1]) / 2.0f, e);
+  EXPECT_NEAR(output[2], (data1[2] + data2[2]) / 2.0f, e);
+  EXPECT_NEAR(output[3], (data1[3] + data2[3]) / 2.0f, e);
 
   ma.Average(data3, output);
   EXPECT_NEAR(output[0], (data1[0] + data2[0] + data3[0]) / 3.0f, e);
@@ -53,6 +53,26 @@ TEST(MovingAverageSpectrum, Average) {
   EXPECT_NEAR(output[3], (data2[3] + data3[3] + data4[3]) / 3.0f, e);
 }
 
+TEST(MovingAverageSpectrum, UpdateMemoryLength) {
+  constexpr size_t num_elem = 4;
+  constexpr float e = 1e-6f;
+  MovingAverageSpectrum ma(num_elem, 3);
+  std::array<float, num_elem> data1 = {1, 2, 3, 4};
+  std::array<float, num_elem> data2 = {5, 1, 9, 7};
+  std::array<float, num_elem> output;
+
+  ma.Average(data1, output);
+  EXPECT_NEAR(output[0], data1[0], e);
+
+  ma.UpdateMemoryLength(1);
+  ma.Average(data2, output);
+  // After update, it should behave as if it was just created with mem_len = 1.
+  EXPECT_NEAR(output[0], data2[0], e);
+  EXPECT_NEAR(output[1], data2[1], e);
+  EXPECT_NEAR(output[2], data2[2], e);
+  EXPECT_NEAR(output[3], data2[3], e);
+}
+
 TEST(MovingAverageSpectrum, PassThrough) {
   constexpr size_t num_elem = 4;
   constexpr size_t mem_len = 1;
diff --git a/modules/audio_processing/aec3/nearend_detector.h b/modules/audio_processing/aec3/nearend_detector.h
index c95c19edc1..ff30f423f9 100644
--- a/modules/audio_processing/aec3/nearend_detector.h
+++ b/modules/audio_processing/aec3/nearend_detector.h
@@ -14,6 +14,7 @@
 #include <array>
 #include <span>
 
+#include "api/audio/echo_canceller3_config.h"
 #include "modules/audio_processing/aec3/aec3_common.h"
 
 namespace webrtc {
@@ -33,6 +34,9 @@ class NearendDetector {
       std::span<const std::array<float, kFftLengthBy2Plus1>>
           comfort_noise_spectrum,
       bool initial_state) = 0;
+
+  // Sets the configuration.
+  virtual void SetConfig(const EchoCanceller3Config::Suppressor& config) = 0;
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_processing/aec3/neural_residual_echo_estimator/neural_residual_echo_estimator_impl.cc b/modules/audio_processing/aec3/neural_residual_echo_estimator/neural_residual_echo_estimator_impl.cc
index 0c80ef18ce..a18f1e511f 100644
--- a/modules/audio_processing/aec3/neural_residual_echo_estimator/neural_residual_echo_estimator_impl.cc
+++ b/modules/audio_processing/aec3/neural_residual_echo_estimator/neural_residual_echo_estimator_impl.cc
@@ -569,6 +569,15 @@ void NeuralResidualEchoEstimatorImpl::Estimate(
 EchoCanceller3Config NeuralResidualEchoEstimatorImpl::GetConfiguration(
     bool multi_channel) const {
   EchoCanceller3Config config;
+  config.suppressor = AdjustConfig(config.suppressor);
+  config.filter.enable_coarse_filter_output_usage = false;
+  return config;
+}
+
+EchoCanceller3Config::Suppressor NeuralResidualEchoEstimatorImpl::AdjustConfig(
+    const EchoCanceller3Config::Suppressor& suppressor_config) const {
+  EchoCanceller3Config::Suppressor adjusted_suppressor_config =
+      suppressor_config;
   EchoCanceller3Config::Suppressor::MaskingThresholds tuning_masking_thresholds(
       /*enr_transparent=*/0.0f, /*enr_suppress=*/1.0f,
       /*emr_transparent=*/0.3f);
@@ -576,15 +585,15 @@ EchoCanceller3Config NeuralResidualEchoEstimatorImpl::GetConfiguration(
       /*mask_lf=*/tuning_masking_thresholds,
       /*mask_hf=*/tuning_masking_thresholds, /*max_inc_factor=*/100.0f,
       /*max_dec_factor_lf=*/0.0f);
-  config.filter.enable_coarse_filter_output_usage = false;
-  config.suppressor.nearend_average_blocks = 1;
-  config.suppressor.normal_tuning = tuning;
-  config.suppressor.nearend_tuning = tuning;
-  config.suppressor.dominant_nearend_detection.enr_threshold = 0.5f;
-  config.suppressor.dominant_nearend_detection.trigger_threshold = 2;
-  config.suppressor.high_frequency_suppression.limiting_gain_band = 24;
-  config.suppressor.high_frequency_suppression.bands_in_limiting_gain = 3;
-  return config;
+  adjusted_suppressor_config.nearend_average_blocks = 1;
+  adjusted_suppressor_config.normal_tuning = tuning;
+  adjusted_suppressor_config.nearend_tuning = tuning;
+  adjusted_suppressor_config.dominant_nearend_detection.enr_threshold = 0.5f;
+  adjusted_suppressor_config.dominant_nearend_detection.trigger_threshold = 2;
+  adjusted_suppressor_config.high_frequency_suppression.limiting_gain_band = 24;
+  adjusted_suppressor_config.high_frequency_suppression.bands_in_limiting_gain =
+      3;
+  return adjusted_suppressor_config;
 }
 
 void NeuralResidualEchoEstimatorImpl::DumpInputs(
diff --git a/modules/audio_processing/aec3/neural_residual_echo_estimator/neural_residual_echo_estimator_impl.h b/modules/audio_processing/aec3/neural_residual_echo_estimator/neural_residual_echo_estimator_impl.h
index 2c2f091089..f92358eef5 100644
--- a/modules/audio_processing/aec3/neural_residual_echo_estimator/neural_residual_echo_estimator_impl.h
+++ b/modules/audio_processing/aec3/neural_residual_echo_estimator/neural_residual_echo_estimator_impl.h
@@ -81,6 +81,8 @@ class NeuralResidualEchoEstimatorImpl : public NeuralResidualEchoEstimator {
       std::span<std::array<float, kFftLengthBy2Plus1>> R2_unbounded) override;
 
   EchoCanceller3Config GetConfiguration(bool multi_channel) const override;
+  EchoCanceller3Config::Suppressor AdjustConfig(
+      const EchoCanceller3Config::Suppressor& config) const override;
 
   void Reset() override;
 
diff --git a/modules/audio_processing/aec3/residual_echo_estimator.cc b/modules/audio_processing/aec3/residual_echo_estimator.cc
index 65793d46c1..fd821cc3b1 100644
--- a/modules/audio_processing/aec3/residual_echo_estimator.cc
+++ b/modules/audio_processing/aec3/residual_echo_estimator.cc
@@ -206,6 +206,8 @@ void ResidualEchoEstimator::Estimate(
 
   const size_t num_capture_channels = R2.size();
 
+  is_ml_ree_active_ = neural_residual_echo_estimator_ != nullptr &&
+                      aec_state.UsableLinearEstimate();
   // Estimate the power of the stationary noise in the render signal.
   UpdateRenderNoisePower(render_buffer);
 
@@ -235,9 +237,9 @@ void ResidualEchoEstimator::Estimate(
         R2, R2_unbounded);
   }
 
-  // Estimate the residual echo power.
-  if (aec_state.UsableLinearEstimate()) {
-    if (neural_residual_echo_estimator_ == nullptr) {
+  // Estimate the residual echo power, used when ml_ree is not active.
+  if (!is_ml_ree_active_) {
+    if (aec_state.UsableLinearEstimate()) {
       // When there is saturated echo, assume the same spectral content as is
       // present in the microphone signal.
       if (aec_state.SaturatedEcho()) {
@@ -256,48 +258,48 @@ void ResidualEchoEstimator::Estimate(
                    dominant_nearend);
       AddReverb(R2);
       AddReverb(R2_unbounded);
-    }
-  } else {
-    const float echo_path_gain =
-        GetEchoPathGain(aec_state, /*gain_for_early_reflections=*/true);
-
-    // When there is saturated echo, assume the same spectral content as is
-    // present in the microphone signal.
-    if (aec_state.SaturatedEcho()) {
-      for (size_t ch = 0; ch < num_capture_channels; ++ch) {
-        std::copy(Y2[ch].begin(), Y2[ch].end(), R2[ch].begin());
-        std::copy(Y2[ch].begin(), Y2[ch].end(), R2_unbounded[ch].begin());
-      }
     } else {
-      // Estimate the echo generating signal power.
-      std::array<float, kFftLengthBy2Plus1> X2;
-      EchoGeneratingPower(num_render_channels_,
-                          render_buffer.GetSpectrumBuffer(), config_.echo_model,
-                          aec_state.MinDirectPathFilterDelay(), X2);
-      if (!aec_state.UseStationarityProperties()) {
-        ApplyNoiseGate(config_.echo_model, X2);
+      const float echo_path_gain =
+          GetEchoPathGain(aec_state, /*gain_for_early_reflections=*/true);
+
+      // When there is saturated echo, assume the same spectral content as is
+      // present in the microphone signal.
+      if (aec_state.SaturatedEcho()) {
+        for (size_t ch = 0; ch < num_capture_channels; ++ch) {
+          std::copy(Y2[ch].begin(), Y2[ch].end(), R2[ch].begin());
+          std::copy(Y2[ch].begin(), Y2[ch].end(), R2_unbounded[ch].begin());
+        }
+      } else {
+        // Estimate the echo generating signal power.
+        std::array<float, kFftLengthBy2Plus1> X2;
+        EchoGeneratingPower(
+            num_render_channels_, render_buffer.GetSpectrumBuffer(),
+            config_.echo_model, aec_state.MinDirectPathFilterDelay(), X2);
+        if (!aec_state.UseStationarityProperties()) {
+          ApplyNoiseGate(config_.echo_model, X2);
+        }
+
+        // Subtract the stationary noise power to avoid stationary noise causing
+        // excessive echo suppression.
+        for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) {
+          X2[k] -=
+              config_.echo_model.stationary_gate_slope * X2_noise_floor_[k];
+          X2[k] = std::max(0.f, X2[k]);
+        }
+
+        NonLinearEstimate(echo_path_gain, X2, R2);
+        NonLinearEstimate(echo_path_gain, X2, R2_unbounded);
       }
 
-      // Subtract the stationary noise power to avoid stationary noise causing
-      // excessive echo suppression.
-      for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) {
-        X2[k] -= config_.echo_model.stationary_gate_slope * X2_noise_floor_[k];
-        X2[k] = std::max(0.f, X2[k]);
+      if (config_.echo_model.model_reverb_in_nonlinear_mode &&
+          !aec_state.TransparentModeActive()) {
+        UpdateReverb(ReverbType::kNonLinear, aec_state, render_buffer,
+                     dominant_nearend);
+        AddReverb(R2);
+        AddReverb(R2_unbounded);
       }
-
-      NonLinearEstimate(echo_path_gain, X2, R2);
-      NonLinearEstimate(echo_path_gain, X2, R2_unbounded);
-    }
-
-    if (config_.echo_model.model_reverb_in_nonlinear_mode &&
-        !aec_state.TransparentModeActive()) {
-      UpdateReverb(ReverbType::kNonLinear, aec_state, render_buffer,
-                   dominant_nearend);
-      AddReverb(R2);
-      AddReverb(R2_unbounded);
     }
   }
-
   if (aec_state.UseStationarityProperties()) {
     // Scale the echo according to echo audibility.
     std::array<float, kFftLengthBy2Plus1> residual_scaling;
diff --git a/modules/audio_processing/aec3/residual_echo_estimator.h b/modules/audio_processing/aec3/residual_echo_estimator.h
index fe1a4f868d..0a53e0d203 100644
--- a/modules/audio_processing/aec3/residual_echo_estimator.h
+++ b/modules/audio_processing/aec3/residual_echo_estimator.h
@@ -49,6 +49,9 @@ class ResidualEchoEstimator {
       std::span<std::array<float, kFftLengthBy2Plus1>> R2,
       std::span<std::array<float, kFftLengthBy2Plus1>> R2_unbounded);
 
+  // Returns true if ML-REE is active.
+  bool IsMlReeActive() const { return is_ml_ree_active_; }
+
  private:
   enum class ReverbType { kLinear, kNonLinear };
 
@@ -84,6 +87,7 @@ class ResidualEchoEstimator {
   std::array<int, kFftLengthBy2Plus1> X2_noise_floor_counter_;
   ReverbModel echo_reverb_;
   NeuralResidualEchoEstimator* neural_residual_echo_estimator_;
+  bool is_ml_ree_active_ = false;
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_processing/aec3/subband_nearend_detector.cc b/modules/audio_processing/aec3/subband_nearend_detector.cc
index 39908acf59..8cd85e4ca0 100644
--- a/modules/audio_processing/aec3/subband_nearend_detector.cc
+++ b/modules/audio_processing/aec3/subband_nearend_detector.cc
@@ -73,4 +73,17 @@ void SubbandNearendDetector::Update(
          (nearend_power_subband1 > config_.snr_threshold * noise_power));
   }
 }
+
+void SubbandNearendDetector::SetConfig(
+    const EchoCanceller3Config::Suppressor& config) {
+  config_ = config.subband_nearend_detection;
+  one_over_subband_length1_ =
+      1.f / (config_.subband1.high - config_.subband1.low + 1);
+  one_over_subband_length2_ =
+      1.f / (config_.subband2.high - config_.subband2.low + 1);
+  for (auto& smoother : nearend_smoothers_) {
+    smoother.UpdateMemoryLength(config_.nearend_average_blocks);
+  }
+}
+
 }  // namespace webrtc
diff --git a/modules/audio_processing/aec3/subband_nearend_detector.h b/modules/audio_processing/aec3/subband_nearend_detector.h
index 5b4333199d..e28391f61d 100644
--- a/modules/audio_processing/aec3/subband_nearend_detector.h
+++ b/modules/audio_processing/aec3/subband_nearend_detector.h
@@ -41,12 +41,15 @@ class SubbandNearendDetector : public NearendDetector {
           comfort_noise_spectrum,
       bool initial_state) override;
 
+  // Sets the configuration.
+  void SetConfig(const EchoCanceller3Config::Suppressor& config) override;
+
  private:
-  const EchoCanceller3Config::Suppressor::SubbandNearendDetection config_;
+  EchoCanceller3Config::Suppressor::SubbandNearendDetection config_;
   const size_t num_capture_channels_;
   std::vector<MovingAverageSpectrum> nearend_smoothers_;
-  const float one_over_subband_length1_;
-  const float one_over_subband_length2_;
+  float one_over_subband_length1_;
+  float one_over_subband_length2_;
   bool nearend_state_ = false;
 };
 
diff --git a/modules/audio_processing/aec3/suppression_gain.cc b/modules/audio_processing/aec3/suppression_gain.cc
index aab8a325a3..c5ca8ff233 100644
--- a/modules/audio_processing/aec3/suppression_gain.cc
+++ b/modules/audio_processing/aec3/suppression_gain.cc
@@ -41,14 +41,16 @@ void LimitLowFrequencyGains(std::array<float, kFftLengthBy2Plus1>* gain) {
   (*gain)[0] = (*gain)[1] = std::min((*gain)[1], (*gain)[2]);
 }
 
-void LimitHighFrequencyGains(const EchoCanceller3Config::Suppressor& config,
-                             std::array<float, kFftLengthBy2Plus1>* gain) {
+void LimitHighFrequencyGains(
+    const EchoCanceller3Config::Suppressor::HighFrequencySuppression&
+        high_frequency_suppression,
+    bool conservative_hf_suppression,
+    std::array<float, kFftLengthBy2Plus1>* gain) {
   // Limit the high frequency gains to avoid echo leakage due to an imperfect
   // filter.
-  const int limiting_gain_band =
-      config.high_frequency_suppression.limiting_gain_band;
+  const int limiting_gain_band = high_frequency_suppression.limiting_gain_band;
   const int bands_in_limiting_gain =
-      config.high_frequency_suppression.bands_in_limiting_gain;
+      high_frequency_suppression.bands_in_limiting_gain;
   if (bands_in_limiting_gain > 0) {
     RTC_DCHECK_GE(limiting_gain_band, 0);
     RTC_DCHECK_LE(limiting_gain_band + bands_in_limiting_gain, gain->size());
@@ -63,7 +65,7 @@ void LimitHighFrequencyGains(const EchoCanceller3Config::Suppressor& config,
   }
   (*gain)[kFftLengthBy2] = (*gain)[kFftLengthBy2Minus1];
 
-  if (config.conservative_hf_suppression) {
+  if (conservative_hf_suppression) {
     // Limits the gain in the frequencies for which the adaptive filter has not
     // converged.
     // TODO(peah): Make adaptive to take the actual filter error into account.
@@ -83,9 +85,10 @@ void LimitHighFrequencyGains(const EchoCanceller3Config::Suppressor& config,
 }
 
 // Scales the echo according to assessed audibility at the other end.
-void WeightEchoForAudibility(const EchoCanceller3Config& config,
-                             std::span<const float> echo,
-                             std::span<float> weighted_echo) {
+void WeightEchoForAudibility(
+    const EchoCanceller3Config::EchoAudibility& echo_audibility,
+    std::span<const float> echo,
+    std::span<float> weighted_echo) {
   RTC_DCHECK_EQ(kFftLengthBy2Plus1, echo.size());
   RTC_DCHECK_EQ(kFftLengthBy2Plus1, weighted_echo.size());
 
@@ -101,19 +104,19 @@ void WeightEchoForAudibility(const EchoCanceller3Config& config,
     }
   };
 
-  float threshold = config.echo_audibility.floor_power *
-                    config.echo_audibility.audibility_threshold_lf;
-  float normalizer = 1.f / (threshold - config.echo_audibility.floor_power);
+  float threshold =
+      echo_audibility.floor_power * echo_audibility.audibility_threshold_lf;
+  float normalizer = 1.f / (threshold - echo_audibility.floor_power);
   weigh(threshold, normalizer, 0, 3, echo, weighted_echo);
 
-  threshold = config.echo_audibility.floor_power *
-              config.echo_audibility.audibility_threshold_mf;
-  normalizer = 1.f / (threshold - config.echo_audibility.floor_power);
+  threshold =
+      echo_audibility.floor_power * echo_audibility.audibility_threshold_mf;
+  normalizer = 1.f / (threshold - echo_audibility.floor_power);
   weigh(threshold, normalizer, 3, 7, echo, weighted_echo);
 
-  threshold = config.echo_audibility.floor_power *
-              config.echo_audibility.audibility_threshold_hf;
-  normalizer = 1.f / (threshold - config.echo_audibility.floor_power);
+  threshold =
+      echo_audibility.floor_power * echo_audibility.audibility_threshold_hf;
+  normalizer = 1.f / (threshold - echo_audibility.floor_power);
   weigh(threshold, normalizer, 7, kFftLengthBy2Plus1, echo, weighted_echo);
 }
 
@@ -122,6 +125,7 @@ void WeightEchoForAudibility(const EchoCanceller3Config& config,
 std::atomic<int> SuppressionGain::instance_count_(0);
 
 float SuppressionGain::UpperBandsGain(
+    const EchoCanceller3Config::Suppressor& suppressor_config,
     std::span<const std::array<float, kFftLengthBy2Plus1>> echo_spectrum,
     std::span<const std::array<float, kFftLengthBy2Plus1>>
         comfort_noise_spectrum,
@@ -172,7 +176,7 @@ float SuppressionGain::UpperBandsGain(
   // upper bands.
   float anti_howling_gain;
   const float activation_threshold =
-      kBlockSize * config_.suppressor.high_bands_suppression
+      kBlockSize * suppressor_config.high_bands_suppression
                        .anti_howling_activation_threshold;
   if (high_band_energy < std::max(low_band_energy, activation_threshold)) {
     anti_howling_gain = 1.f;
@@ -181,14 +185,14 @@ float SuppressionGain::UpperBandsGain(
     RTC_DCHECK_LE(low_band_energy, high_band_energy);
     RTC_DCHECK_NE(0.f, high_band_energy);
     anti_howling_gain =
-        config_.suppressor.high_bands_suppression.anti_howling_gain *
+        suppressor_config.high_bands_suppression.anti_howling_gain *
         sqrtf(low_band_energy / high_band_energy);
   }
 
   float gain_bound = 1.f;
   if (!dominant_nearend_detector_->IsNearendState()) {
     // Bound the upper gain during significant echo activity.
-    const auto& cfg = config_.suppressor.high_bands_suppression;
+    const auto& cfg = suppressor_config.high_bands_suppression;
     auto low_frequency_energy = [](std::span<const float> spectrum) {
       RTC_DCHECK_LE(16, spectrum.size());
       return std::accumulate(spectrum.begin() + 1, spectrum.begin() + 16, 0.f);
@@ -230,16 +234,18 @@ void SuppressionGain::GainToNoAudibleEcho(
 
 // Compute the minimum gain as the attenuating gain to put the signal just
 // above the zero sample values.
-void SuppressionGain::GetMinGain(std::span<const float> weighted_residual_echo,
-                                 std::span<const float> last_nearend,
-                                 std::span<const float> last_echo,
-                                 bool low_noise_render,
-                                 bool saturated_echo,
-                                 std::span<float> min_gain) const {
+void SuppressionGain::GetMinGain(
+    const EchoCanceller3Config::Suppressor& suppressor_config,
+    std::span<const float> weighted_residual_echo,
+    std::span<const float> last_nearend,
+    std::span<const float> last_echo,
+    bool low_noise_render,
+    bool saturated_echo,
+    std::span<float> min_gain) const {
   if (!saturated_echo) {
     const float min_echo_power =
-        low_noise_render ? config_.echo_audibility.low_render_limit
-                         : config_.echo_audibility.normal_render_limit;
+        low_noise_render ? echo_audibility_config_.low_render_limit
+                         : echo_audibility_config_.normal_render_limit;
 
     for (size_t k = 0; k < min_gain.size(); ++k) {
       min_gain[k] = weighted_residual_echo[k] > 0.f
@@ -249,16 +255,16 @@ void SuppressionGain::GetMinGain(std::span<const float> weighted_residual_echo,
     }
 
     if (!initial_state_ ||
-        config_.suppressor.lf_smoothing_during_initial_phase) {
+        suppressor_config.lf_smoothing_during_initial_phase) {
       const float& dec = dominant_nearend_detector_->IsNearendState()
                              ? nearend_params_.max_dec_factor_lf
                              : normal_params_.max_dec_factor_lf;
 
-      for (int k = 0; k <= config_.suppressor.last_lf_smoothing_band; ++k) {
+      for (int k = 0; k <= suppressor_config.last_lf_smoothing_band; ++k) {
         // Make sure the gains of the low frequencies do not decrease too
         // quickly after strong nearend.
         if (last_nearend[k] > last_echo[k] ||
-            k <= config_.suppressor.last_permanent_lf_smoothing_band) {
+            k <= suppressor_config.last_permanent_lf_smoothing_band) {
           min_gain[k] = std::max(min_gain[k], last_gain_[k] * dec);
           min_gain[k] = std::min(min_gain[k], 1.f);
         }
@@ -271,17 +277,19 @@ void SuppressionGain::GetMinGain(std::span<const float> weighted_residual_echo,
 
 // Compute the maximum gain by limiting the gain increase from the previous
 // gain.
-void SuppressionGain::GetMaxGain(std::span<float> max_gain) const {
+void SuppressionGain::GetMaxGain(float floor_first_increase,
+                                 std::span<float> max_gain) const {
   const auto& inc = dominant_nearend_detector_->IsNearendState()
                         ? nearend_params_.max_inc_factor
                         : normal_params_.max_inc_factor;
-  const auto& floor = config_.suppressor.floor_first_increase;
   for (size_t k = 0; k < max_gain.size(); ++k) {
-    max_gain[k] = std::min(std::max(last_gain_[k] * inc, floor), 1.f);
+    max_gain[k] =
+        std::min(std::max(last_gain_[k] * inc, floor_first_increase), 1.f);
   }
 }
 
 void SuppressionGain::LowerBandGain(
+    const EchoCanceller3Config::Suppressor& suppressor_config,
     bool low_noise_render,
     const AecState& aec_state,
     std::span<const std::array<float, kFftLengthBy2Plus1>> suppressor_input,
@@ -292,7 +300,7 @@ void SuppressionGain::LowerBandGain(
   gain->fill(1.f);
   const bool saturated_echo = aec_state.SaturatedEcho();
   std::array<float, kFftLengthBy2Plus1> max_gain;
-  GetMaxGain(max_gain);
+  GetMaxGain(suppressor_config.floor_first_increase, max_gain);
 
   for (size_t ch = 0; ch < num_capture_channels_; ++ch) {
     std::array<float, kFftLengthBy2Plus1> G;
@@ -301,11 +309,12 @@ void SuppressionGain::LowerBandGain(
 
     // Weight echo power in terms of audibility.
     std::array<float, kFftLengthBy2Plus1> weighted_residual_echo;
-    WeightEchoForAudibility(config_, residual_echo[ch], weighted_residual_echo);
+    WeightEchoForAudibility(echo_audibility_config_, residual_echo[ch],
+                            weighted_residual_echo);
 
     std::array<float, kFftLengthBy2Plus1> min_gain;
-    GetMinGain(weighted_residual_echo, last_nearend_[ch], last_echo_[ch],
-               low_noise_render, saturated_echo, min_gain);
+    GetMinGain(suppressor_config, weighted_residual_echo, last_nearend_[ch],
+               last_echo_[ch], low_noise_render, saturated_echo, min_gain);
 
     GainToNoAudibleEcho(nearend, weighted_residual_echo, comfort_noise[0], &G);
 
@@ -325,8 +334,10 @@ void SuppressionGain::LowerBandGain(
   // Use conservative high-frequency gains during clock-drift or when not in
   // dominant nearend.
   if (!dominant_nearend_detector_->IsNearendState() || clock_drift ||
-      config_.suppressor.conservative_hf_suppression) {
-    LimitHighFrequencyGains(config_.suppressor, gain);
+      suppressor_config.conservative_hf_suppression) {
+    LimitHighFrequencyGains(suppressor_config.high_frequency_suppression,
+                            suppressor_config.conservative_hf_suppression,
+                            gain);
   }
 
   // Store computed gains.
@@ -342,32 +353,29 @@ SuppressionGain::SuppressionGain(const EchoCanceller3Config& config,
                                  size_t num_capture_channels)
     : data_dumper_(new ApmDataDumper(instance_count_.fetch_add(1) + 1)),
       optimization_(optimization),
-      config_(config),
       num_capture_channels_(num_capture_channels),
-      state_change_duration_blocks_(
-          static_cast<int>(config_.filter.config_change_duration_blocks)),
+      echo_audibility_config_(config.echo_audibility),
+      use_subband_nearend_detection_(
+          config.suppressor.use_subband_nearend_detection),
       last_nearend_(num_capture_channels_, {0}),
       last_echo_(num_capture_channels_, {0}),
       nearend_smoothers_(
           num_capture_channels_,
           MovingAverageSpectrum(kFftLengthBy2Plus1,
                                 config.suppressor.nearend_average_blocks)),
-      nearend_params_(config_.suppressor.last_lf_band,
-                      config_.suppressor.first_hf_band,
-                      config_.suppressor.nearend_tuning),
-      normal_params_(config_.suppressor.last_lf_band,
-                     config_.suppressor.first_hf_band,
-                     config_.suppressor.normal_tuning),
-      use_unbounded_echo_spectrum_(config.suppressor.dominant_nearend_detection
-                                       .use_unbounded_echo_spectrum) {
-  RTC_DCHECK_LT(0, state_change_duration_blocks_);
+      nearend_params_(config.suppressor.last_lf_band,
+                      config.suppressor.first_hf_band,
+                      config.suppressor.nearend_tuning),
+      normal_params_(config.suppressor.last_lf_band,
+                     config.suppressor.first_hf_band,
+                     config.suppressor.normal_tuning) {
   last_gain_.fill(1.f);
-  if (config_.suppressor.use_subband_nearend_detection) {
+  if (config.suppressor.use_subband_nearend_detection) {
     dominant_nearend_detector_ = std::make_unique<SubbandNearendDetector>(
-        config_.suppressor.subband_nearend_detection, num_capture_channels_);
+        config.suppressor.subband_nearend_detection, num_capture_channels_);
   } else {
     dominant_nearend_detector_ = std::make_unique<DominantNearendDetector>(
-        config_.suppressor.dominant_nearend_detection, num_capture_channels_);
+        config.suppressor.dominant_nearend_detection, num_capture_channels_);
   }
   RTC_DCHECK(dominant_nearend_detector_);
 }
@@ -375,6 +383,8 @@ SuppressionGain::SuppressionGain(const EchoCanceller3Config& config,
 SuppressionGain::~SuppressionGain() = default;
 
 void SuppressionGain::GetGain(
+    const EchoCanceller3Config::Suppressor& suppressor_config,
+    bool config_changed,
     std::span<const std::array<float, kFftLengthBy2Plus1>> nearend_spectrum,
     std::span<const std::array<float, kFftLengthBy2Plus1>> echo_spectrum,
     std::span<const std::array<float, kFftLengthBy2Plus1>>
@@ -392,10 +402,15 @@ void SuppressionGain::GetGain(
   RTC_DCHECK(high_bands_gain);
   RTC_DCHECK(low_band_gain);
 
+  if (config_changed) {
+    UpdateStateDependingOnConfig(suppressor_config);
+  }
+
   // Choose residual echo spectrum for dominant nearend detection.
-  const auto echo = use_unbounded_echo_spectrum_
-                        ? residual_echo_spectrum_unbounded
-                        : residual_echo_spectrum;
+  const auto echo =
+      suppressor_config.dominant_nearend_detection.use_unbounded_echo_spectrum
+          ? residual_echo_spectrum_unbounded
+          : residual_echo_spectrum;
 
   // Update the nearend state selection.
   dominant_nearend_detector_->Update(nearend_spectrum, echo,
@@ -403,17 +418,17 @@ void SuppressionGain::GetGain(
 
   // Compute gain for the lower band.
   bool low_noise_render = low_render_detector_.Detect(render);
-  LowerBandGain(low_noise_render, aec_state, nearend_spectrum,
-                residual_echo_spectrum, comfort_noise_spectrum, clock_drift,
-                low_band_gain);
+  LowerBandGain(suppressor_config, low_noise_render, aec_state,
+                nearend_spectrum, residual_echo_spectrum,
+                comfort_noise_spectrum, clock_drift, low_band_gain);
 
   // Compute the gain for the upper bands.
   const std::optional<int> narrow_peak_band =
       render_signal_analyzer.NarrowPeakBand();
 
-  *high_bands_gain =
-      UpperBandsGain(echo_spectrum, comfort_noise_spectrum, narrow_peak_band,
-                     aec_state.SaturatedEcho(), render, *low_band_gain);
+  *high_bands_gain = UpperBandsGain(
+      suppressor_config, echo_spectrum, comfort_noise_spectrum,
+      narrow_peak_band, aec_state.SaturatedEcho(), render, *low_band_gain);
 
   data_dumper_->DumpRaw("aec3_dominant_nearend",
                         dominant_nearend_detector_->IsNearendState());
@@ -421,11 +436,24 @@ void SuppressionGain::GetGain(
 
 void SuppressionGain::SetInitialState(bool state) {
   initial_state_ = state;
-  if (state) {
-    initial_state_change_counter_ = state_change_duration_blocks_;
-  } else {
-    initial_state_change_counter_ = 0;
+}
+
+void SuppressionGain::UpdateStateDependingOnConfig(
+    const EchoCanceller3Config::Suppressor& suppressor_config) {
+  RTC_DCHECK_EQ(suppressor_config.use_subband_nearend_detection,
+                use_subband_nearend_detection_);
+  // Update nearend average blocks.
+  for (auto& smoother : nearend_smoothers_) {
+    smoother.UpdateMemoryLength(suppressor_config.nearend_average_blocks);
   }
+  dominant_nearend_detector_->SetConfig(suppressor_config);
+  nearend_params_.SetConfig(suppressor_config.last_lf_band,
+                            suppressor_config.first_hf_band,
+                            suppressor_config.nearend_tuning);
+
+  normal_params_.SetConfig(suppressor_config.last_lf_band,
+                           suppressor_config.first_hf_band,
+                           suppressor_config.normal_tuning);
 }
 
 // Detects when the render signal can be considered to have low power and
@@ -452,9 +480,16 @@ bool SuppressionGain::LowNoiseRenderDetector::Detect(const Block& render) {
 SuppressionGain::GainParameters::GainParameters(
     int last_lf_band,
     int first_hf_band,
-    const EchoCanceller3Config::Suppressor::Tuning& tuning)
-    : max_inc_factor(tuning.max_inc_factor),
-      max_dec_factor_lf(tuning.max_dec_factor_lf) {
+    const EchoCanceller3Config::Suppressor::Tuning& tuning) {
+  SetConfig(last_lf_band, first_hf_band, tuning);
+}
+
+void SuppressionGain::GainParameters::SetConfig(
+    int last_lf_band,
+    int first_hf_band,
+    const EchoCanceller3Config::Suppressor::Tuning& tuning) {
+  max_inc_factor = tuning.max_inc_factor;
+  max_dec_factor_lf = tuning.max_dec_factor_lf;
   // Compute per-band masking thresholds.
   RTC_DCHECK_LT(last_lf_band, first_hf_band);
   auto& lf = tuning.mask_lf;
diff --git a/modules/audio_processing/aec3/suppression_gain.h b/modules/audio_processing/aec3/suppression_gain.h
index 4bcd0d23c3..6d0920774c 100644
--- a/modules/audio_processing/aec3/suppression_gain.h
+++ b/modules/audio_processing/aec3/suppression_gain.h
@@ -27,6 +27,7 @@
 #include "modules/audio_processing/aec3/nearend_detector.h"
 #include "modules/audio_processing/aec3/render_signal_analyzer.h"
 #include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/gtest_prod_util.h"
 
 namespace webrtc {
 
@@ -42,6 +43,8 @@ class SuppressionGain {
   SuppressionGain& operator=(const SuppressionGain&) = delete;
 
   void GetGain(
+      const EchoCanceller3Config::Suppressor& suppressor_config,
+      bool config_changed,
       std::span<const std::array<float, kFftLengthBy2Plus1>> nearend_spectrum,
       std::span<const std::array<float, kFftLengthBy2Plus1>> echo_spectrum,
       std::span<const std::array<float, kFftLengthBy2Plus1>>
@@ -65,8 +68,15 @@ class SuppressionGain {
   void SetInitialState(bool state);
 
  private:
+  FRIEND_TEST_ALL_PREFIXES(SuppressionGainTest, UpdateStateDependingOnConfig);
+
+  // Updates the internal state, e.g. sizes and parameters, if the config
+  // changes.
+  void UpdateStateDependingOnConfig(
+      const EchoCanceller3Config::Suppressor& suppressor_config);
   // Computes the gain to apply for the bands beyond the first band.
   float UpperBandsGain(
+      const EchoCanceller3Config::Suppressor& suppressor_config,
       std::span<const std::array<float, kFftLengthBy2Plus1>> echo_spectrum,
       std::span<const std::array<float, kFftLengthBy2Plus1>>
           comfort_noise_spectrum,
@@ -81,6 +91,7 @@ class SuppressionGain {
                            std::array<float, kFftLengthBy2Plus1>* gain) const;
 
   void LowerBandGain(
+      const EchoCanceller3Config::Suppressor& suppressor_config,
       bool stationary_with_low_power,
       const AecState& aec_state,
       std::span<const std::array<float, kFftLengthBy2Plus1>> suppressor_input,
@@ -89,14 +100,15 @@ class SuppressionGain {
       bool clock_drift,
       std::array<float, kFftLengthBy2Plus1>* gain);
 
-  void GetMinGain(std::span<const float> weighted_residual_echo,
+  void GetMinGain(const EchoCanceller3Config::Suppressor& suppressor_config,
+                  std::span<const float> weighted_residual_echo,
                   std::span<const float> last_nearend,
                   std::span<const float> last_echo,
                   bool low_noise_render,
                   bool saturated_echo,
                   std::span<float> min_gain) const;
 
-  void GetMaxGain(std::span<float> max_gain) const;
+  void GetMaxGain(float floor_first_increase, std::span<float> max_gain) const;
 
   class LowNoiseRenderDetector {
    public:
@@ -111,8 +123,11 @@ class SuppressionGain {
         int last_lf_band,
         int first_hf_band,
         const EchoCanceller3Config::Suppressor::Tuning& tuning);
-    const float max_inc_factor;
-    const float max_dec_factor_lf;
+    void SetConfig(int last_lf_band,
+                   int first_hf_band,
+                   const EchoCanceller3Config::Suppressor::Tuning& tuning);
+    float max_inc_factor;
+    float max_dec_factor_lf;
     std::array<float, kFftLengthBy2Plus1> enr_transparent_;
     std::array<float, kFftLengthBy2Plus1> enr_suppress_;
     std::array<float, kFftLengthBy2Plus1> emr_transparent_;
@@ -121,21 +136,17 @@ class SuppressionGain {
   static std::atomic<int> instance_count_;
   std::unique_ptr<ApmDataDumper> data_dumper_;
   const Aec3Optimization optimization_;
-  const EchoCanceller3Config config_;
   const size_t num_capture_channels_;
-  const int state_change_duration_blocks_;
+  const EchoCanceller3Config::EchoAudibility echo_audibility_config_;
+  const bool use_subband_nearend_detection_;
   std::array<float, kFftLengthBy2Plus1> last_gain_;
   std::vector<std::array<float, kFftLengthBy2Plus1>> last_nearend_;
   std::vector<std::array<float, kFftLengthBy2Plus1>> last_echo_;
   LowNoiseRenderDetector low_render_detector_;
   bool initial_state_ = true;
-  int initial_state_change_counter_ = 0;
   std::vector<MovingAverageSpectrum> nearend_smoothers_;
-  const GainParameters nearend_params_;
-  const GainParameters normal_params_;
-  // Determines if the dominant nearend detector uses the unbounded residual
-  // echo spectrum.
-  const bool use_unbounded_echo_spectrum_;
+  GainParameters nearend_params_;
+  GainParameters normal_params_;
   std::unique_ptr<NearendDetector> dominant_nearend_detector_;
 };
 
diff --git a/modules/audio_processing/aec3/suppression_gain_unittest.cc b/modules/audio_processing/aec3/suppression_gain_unittest.cc
index 60d290d78b..0bd36a13ae 100644
--- a/modules/audio_processing/aec3/suppression_gain_unittest.cc
+++ b/modules/audio_processing/aec3/suppression_gain_unittest.cc
@@ -58,23 +58,26 @@ TEST(SuppressionGainDeathTest, NullOutputGains) {
   AecState aec_state(CreateEnvironment(), EchoCanceller3Config{}, 1);
   EXPECT_DEATH(
       SuppressionGain(EchoCanceller3Config{}, DetectOptimization(), 16000, 1)
-          .GetGain(E2, S2, R2, R2_unbounded, N2,
+          .GetGain(EchoCanceller3Config{}.suppressor, /*config_changed=*/false,
+                   E2, S2, R2, R2_unbounded, N2,
                    RenderSignalAnalyzer((EchoCanceller3Config{})), aec_state,
-                   Block(3, 1), false, &high_bands_gain, nullptr),
+                   Block(3, 1), /*clock_drift=*/false, &high_bands_gain,
+                   nullptr),
       "");
 }
 
 #endif
 
 // Does a sanity check that the gains are correctly computed.
-TEST(SuppressionGain, BasicGainComputation) {
+TEST(SuppressionGainTest, BasicGainComputation) {
   constexpr size_t kNumRenderChannels = 1;
   constexpr size_t kNumCaptureChannels = 2;
   constexpr int kSampleRateHz = 16000;
   constexpr size_t kNumBands = NumBandsForRate(kSampleRateHz);
-  SuppressionGain suppression_gain(EchoCanceller3Config(), DetectOptimization(),
-                                   kSampleRateHz, kNumCaptureChannels);
-  RenderSignalAnalyzer analyzer(EchoCanceller3Config{});
+  EchoCanceller3Config config;
+  SuppressionGain suppression_gain(config, DetectOptimization(), kSampleRateHz,
+                                   kNumCaptureChannels);
+  RenderSignalAnalyzer analyzer(config);
   float high_bands_gain;
   std::vector<std::array<float, kFftLengthBy2Plus1>> E2(kNumCaptureChannels);
   std::vector<std::array<float, kFftLengthBy2Plus1>> S2(kNumCaptureChannels,
@@ -88,7 +91,6 @@ TEST(SuppressionGain, BasicGainComputation) {
   std::vector<SubtractorOutput> output(kNumCaptureChannels);
   Block x(kNumBands, kNumRenderChannels);
   const Environment env = CreateEnvironment();
-  EchoCanceller3Config config;
   AecState aec_state(env, config, kNumCaptureChannels);
   ApmDataDumper data_dumper(42);
   Subtractor subtractor(env, config, kNumRenderChannels, kNumCaptureChannels,
@@ -120,8 +122,9 @@ TEST(SuppressionGain, BasicGainComputation) {
     aec_state.Update(delay_estimate, subtractor.FilterFrequencyResponses(),
                      subtractor.FilterImpulseResponses(),
                      *render_delay_buffer->GetRenderBuffer(), E2, Y2, output);
-    suppression_gain.GetGain(E2, S2, R2, R2_unbounded, N2, analyzer, aec_state,
-                             x, false, &high_bands_gain, &g);
+    suppression_gain.GetGain(config.suppressor, /*config_changed=*/false, E2,
+                             S2, R2, R2_unbounded, N2, analyzer, aec_state, x,
+                             /*clock_drift=*/false, &high_bands_gain, &g);
   }
   std::for_each(g.begin(), g.end(),
                 [](float a) { EXPECT_NEAR(1.0f, a, 0.001f); });
@@ -140,8 +143,9 @@ TEST(SuppressionGain, BasicGainComputation) {
     aec_state.Update(delay_estimate, subtractor.FilterFrequencyResponses(),
                      subtractor.FilterImpulseResponses(),
                      *render_delay_buffer->GetRenderBuffer(), E2, Y2, output);
-    suppression_gain.GetGain(E2, S2, R2, R2_unbounded, N2, analyzer, aec_state,
-                             x, false, &high_bands_gain, &g);
+    suppression_gain.GetGain(config.suppressor, /*config_changed=*/false, E2,
+                             S2, R2, R2_unbounded, N2, analyzer, aec_state, x,
+                             /*clock_drift=*/false, &high_bands_gain, &g);
   }
   std::for_each(g.begin(), g.end(),
                 [](float a) { EXPECT_NEAR(1.0f, a, 0.001f); });
@@ -152,11 +156,39 @@ TEST(SuppressionGain, BasicGainComputation) {
   R2_unbounded[1].fill(10000000000000.0f);
 
   for (int k = 0; k < 10; ++k) {
-    suppression_gain.GetGain(E2, S2, R2, R2_unbounded, N2, analyzer, aec_state,
-                             x, false, &high_bands_gain, &g);
+    suppression_gain.GetGain(config.suppressor, /*config_changed=*/false, E2,
+                             S2, R2, R2_unbounded, N2, analyzer, aec_state, x,
+                             /*clock_drift=*/false, &high_bands_gain, &g);
   }
   std::for_each(g.begin(), g.end(),
                 [](float a) { EXPECT_NEAR(0.0f, a, 0.001f); });
 }
 
+TEST(SuppressionGainTest, UpdateStateDependingOnConfig) {
+  constexpr size_t kNumCaptureChannels = 1;
+  constexpr int kSampleRateHz = 16000;
+  EchoCanceller3Config config;
+  config.suppressor.nearend_tuning.max_inc_factor = 2.0f;
+  config.suppressor.normal_tuning.max_dec_factor_lf = 0.2f;
+
+  SuppressionGain suppression_gain(config, DetectOptimization(), kSampleRateHz,
+                                   kNumCaptureChannels);
+
+  // Initial call to set up the state.
+  suppression_gain.UpdateStateDependingOnConfig(config.suppressor);
+
+  EXPECT_EQ(suppression_gain.nearend_params_.max_inc_factor, 2.0f);
+  EXPECT_EQ(suppression_gain.normal_params_.max_dec_factor_lf, 0.2f);
+
+  // Change config and verify state is updated.
+  EchoCanceller3Config new_config = config;
+  new_config.suppressor.nearend_tuning.max_inc_factor = 3.0f;
+  new_config.suppressor.normal_tuning.max_dec_factor_lf = 0.3f;
+
+  suppression_gain.UpdateStateDependingOnConfig(new_config.suppressor);
+
+  EXPECT_EQ(suppression_gain.nearend_params_.max_inc_factor, 3.0f);
+  EXPECT_EQ(suppression_gain.normal_params_.max_dec_factor_lf, 0.3f);
+}
+
 }  // namespace webrtc
diff --git a/modules/audio_processing/test/audioproc_float_impl.cc b/modules/audio_processing/test/audioproc_float_impl.cc
index 589fa1ad97..39ee1766e9 100644
--- a/modules/audio_processing/test/audioproc_float_impl.cc
+++ b/modules/audio_processing/test/audioproc_float_impl.cc
@@ -780,7 +780,6 @@ void SetDependencies(const SimulationSettings& settings,
                      BuiltinAudioProcessingBuilder& builder,
                      AudioProcessingBuilderState& builder_state) {
   EchoCanceller3Config aec3_config;
-  std::optional<EchoCanceller3Config> aec3_multichannel_config;
   if (settings.neural_echo_residual_estimator_model) {
     tflite::ops::builtin::BuiltinOpResolver op_resolver;
     builder_state.model = tflite::FlatBufferModel::BuildFromFile(
@@ -789,9 +788,6 @@ void SetDependencies(const SimulationSettings& settings,
     std::unique_ptr<NeuralResidualEchoEstimator> estimator =
         CreateNeuralResidualEchoEstimator(builder_state.model.get(),
                                           &op_resolver);
-    aec3_config = estimator->GetConfiguration(/*multi_channel=*/false);
-    aec3_multichannel_config =
-        estimator->GetConfiguration(/*multi_channel=*/true);
     RTC_CHECK(estimator);
     builder.SetNeuralResidualEchoEstimator(std::move(estimator));
   }
@@ -813,7 +809,7 @@ void SetDependencies(const SimulationSettings& settings,
     }
     std::cout << Aec3ConfigToJsonString(aec3_config) << std::endl;
   }
-  builder.SetEchoCancellerConfig(aec3_config, aec3_multichannel_config);
+  builder.SetEchoCancellerConfig(aec3_config, std::nullopt);
 
   if (settings.use_ed && *settings.use_ed) {
     builder.SetEchoDetector(CreateEchoDetector());

From ae72a714e07b85cf1bd1889e1f9b3ea0387da798 Mon Sep 17 00:00:00 2001
From: Evan Shrubsole <eshr@webrtc.org>
Date: Mon, 30 Mar 2026 14:15:15 +0000
Subject: [PATCH 451/472] Use RunLoop in pc/ tests

Bug: webrtc:469327588
Change-Id: I0648024795ba1e490e7b0a583aeec1996a6a6964
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/461120
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47304}
---
 pc/BUILD.gn                                   |  5 +++
 pc/channel_unittest.cc                        |  7 ++--
 pc/dtls_srtp_transport_integrationtest.cc     | 22 ++++++------
 pc/dtls_srtp_transport_unittest.cc            |  4 +--
 pc/dtls_transport_unittest.cc                 |  4 +--
 pc/ice_transport_unittest.cc                  |  4 +--
 pc/jsep_transport_controller_unittest.cc      |  3 +-
 pc/jsep_transport_unittest.cc                 |  4 +--
 pc/legacy_stats_collector_unittest.cc         |  3 +-
 pc/media_stream_unittest.cc                   |  3 +-
 pc/peer_connection_bundle_unittest.cc         |  3 +-
 pc/peer_connection_crypto_unittest.cc         |  3 +-
 pc/peer_connection_data_channel_unittest.cc   |  3 +-
 pc/peer_connection_end_to_end_unittest.cc     | 36 +++++++++----------
 pc/peer_connection_factory_unittest.cc        |  3 +-
 pc/peer_connection_field_trial_tests.cc       |  3 +-
 ...er_connection_header_extension_unittest.cc |  3 +-
 pc/peer_connection_histogram_unittest.cc      |  3 +-
 pc/peer_connection_interface_unittest.cc      |  4 ++-
 pc/peer_connection_media_unittest.cc          | 21 ++++++-----
 pc/peer_connection_rampup_tests.cc            |  2 +-
 pc/peer_connection_signaling_unittest.cc      |  3 +-
 pc/rtp_transport_unittest.cc                  | 17 +++++----
 pc/sctp_transport_unittest.cc                 |  6 ++--
 pc/sdp_munging_detector_unittest.cc           |  3 +-
 pc/sdp_offer_answer_unittest.cc               |  3 +-
 pc/srtp_transport_unittest.cc                 |  4 +--
 pc/test/fake_audio_capture_module_unittest.cc |  4 +--
 pc/track_media_info_map_unittest.cc           |  3 +-
 pc/video_track_unittest.cc                    |  3 +-
 30 files changed, 104 insertions(+), 85 deletions(-)

diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 9b04a2f1f7..91cc293538 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -2425,6 +2425,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "../p2p:transport_description",
       "../rtc_base:checks",
       "../rtc_base:crypto_random",
+      "../rtc_base:net_helper",
       "../rtc_base:rtc_base_tests_utils",
       "../rtc_base:socket_address",
       "../rtc_base:socket_factory",
@@ -2433,6 +2434,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "../system_wrappers",
       "../test:create_test_environment",
       "../test:create_test_field_trials",
+      "../test:run_loop",
       "../test:test_support",
       "../test:wait_until",
       "//third_party/abseil-cpp/absl/strings",
@@ -2492,6 +2494,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "../rtc_base:logging",
       "../rtc_base:rtc_base_tests_utils",
       "../rtc_base:socket_address",
+      "../test:run_loop",
       "../test:test_support",
       "../test:wait_until",
       "//third_party/abseil-cpp/absl/algorithm:container",
@@ -3103,6 +3106,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "../rtc_base/task_utils:repeating_task",
       "../system_wrappers",
       "../test:frame_generator_capturer",
+      "../test:run_loop",
       "../test:test_support",
       "../test:wait_until",
       "//testing/gmock",
@@ -3158,6 +3162,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "../rtc_base:logging",
       "../rtc_base/containers:flat_map",
       "../system_wrappers",
+      "../test:run_loop",
       "../test:test_support",
       "../test/pc/e2e/analyzer/video:default_video_quality_analyzer",
       "../test/pc/e2e/analyzer/video:default_video_quality_analyzer_shared",
diff --git a/pc/channel_unittest.cc b/pc/channel_unittest.cc
index 1705685fe6..8a91c4750a 100644
--- a/pc/channel_unittest.cc
+++ b/pc/channel_unittest.cc
@@ -63,6 +63,7 @@
 #include "test/create_test_field_trials.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 
 namespace {
 
@@ -1616,9 +1617,7 @@ class ChannelTest : public ::testing::Test {
       thread->ProcessMessages(0);
     }
   }
-  static void FlushCurrentThread() {
-    webrtc::Thread::Current()->ProcessMessages(0);
-  }
+  void FlushCurrentThread() { main_thread_.Flush(); }
   void WaitForThreads(std::span<webrtc::Thread*> threads) {
     // `threads` and current thread post packets to network thread.
     for (webrtc::Thread* thread : threads) {
@@ -1679,7 +1678,7 @@ class ChannelTest : public ::testing::Test {
         channel2_->media_receive_channel());
   }
 
-  webrtc::AutoThread main_thread_;
+  webrtc::test::RunLoop main_thread_;
   // TODO(pbos): Remove playout from all media channels and let renderers mute
   // themselves.
   const bool verify_playout_;
diff --git a/pc/dtls_srtp_transport_integrationtest.cc b/pc/dtls_srtp_transport_integrationtest.cc
index 7a88f7ea77..4c3144a9c3 100644
--- a/pc/dtls_srtp_transport_integrationtest.cc
+++ b/pc/dtls_srtp_transport_integrationtest.cc
@@ -40,10 +40,10 @@
 #include "rtc_base/ssl_fingerprint.h"
 #include "rtc_base/ssl_identity.h"
 #include "rtc_base/ssl_stream_adapter.h"
-#include "rtc_base/thread.h"
 #include "test/create_test_environment.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 #include "test/wait_until.h"
 
 namespace webrtc {
@@ -130,15 +130,15 @@ class DtlsSrtpTransportIntegrationTest : public ::testing::Test {
     client_ice_transport_->SetDestination(server_ice_transport_.get());
 
     // Wait for the DTLS connection to be up.
-    EXPECT_THAT(WaitUntil(
-                    [&] {
-                      return client_dtls_transport_->writable() &&
-                             server_dtls_transport_->writable();
-                    },
-                    IsTrue(),
-                    {.timeout = TimeDelta::Millis(kTimeout),
-                     .clock = &fake_clock_}),
-                IsRtcOk());
+    EXPECT_THAT(
+        WaitUntil(
+            [&] {
+              return client_dtls_transport_->writable() &&
+                     server_dtls_transport_->writable();
+            },
+            IsTrue(),
+            {.timeout = TimeDelta::Millis(kTimeout), .clock = &fake_clock_}),
+        IsRtcOk());
     EXPECT_EQ(client_dtls_transport_->dtls_state(),
               DtlsTransportState::kConnected);
     EXPECT_EQ(server_dtls_transport_->dtls_state(),
@@ -220,7 +220,7 @@ class DtlsSrtpTransportIntegrationTest : public ::testing::Test {
   }
 
  private:
-  AutoThread main_thread_;
+  test::RunLoop main_thread_;
   ScopedFakeClock fake_clock_;
   const Environment env_ = CreateTestEnvironment();
 
diff --git a/pc/dtls_srtp_transport_unittest.cc b/pc/dtls_srtp_transport_unittest.cc
index e42d9b75f4..ab58a8c6a1 100644
--- a/pc/dtls_srtp_transport_unittest.cc
+++ b/pc/dtls_srtp_transport_unittest.cc
@@ -38,9 +38,9 @@
 #include "rtc_base/copy_on_write_buffer.h"
 #include "rtc_base/rtc_certificate.h"
 #include "rtc_base/ssl_identity.h"
-#include "rtc_base/thread.h"
 #include "test/create_test_field_trials.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 
 namespace webrtc {
 namespace {
@@ -267,7 +267,7 @@ class DtlsSrtpTransportTest : public ::testing::Test {
     SendRecvRtcpPackets();
   }
 
-  AutoThread main_thread_;
+  test::RunLoop main_thread_;
   std::unique_ptr<DtlsSrtpTransport> dtls_srtp_transport1_;
   std::unique_ptr<DtlsSrtpTransport> dtls_srtp_transport2_;
   TransportObserver transport_observer1_;
diff --git a/pc/dtls_transport_unittest.cc b/pc/dtls_transport_unittest.cc
index 480357b71c..44e17d46b9 100644
--- a/pc/dtls_transport_unittest.cc
+++ b/pc/dtls_transport_unittest.cc
@@ -25,9 +25,9 @@
 #include "rtc_base/fake_ssl_identity.h"
 #include "rtc_base/rtc_certificate.h"
 #include "rtc_base/ssl_identity.h"
-#include "rtc_base/thread.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 #include "test/wait_until.h"
 
 namespace webrtc {
@@ -99,7 +99,7 @@ class DtlsTransportTest : public ::testing::Test {
     fake_dtls1->SetDestination(fake_dtls2.get());
   }
 
-  AutoThread main_thread_;
+  test::RunLoop main_thread_;
   scoped_refptr<DtlsTransport> transport_;
   std::unique_ptr<FakeDtlsTransport> internal_transport_;
   TestDtlsTransportObserver observer_;
diff --git a/pc/ice_transport_unittest.cc b/pc/ice_transport_unittest.cc
index 91d04a06c4..9b91e86e7e 100644
--- a/pc/ice_transport_unittest.cc
+++ b/pc/ice_transport_unittest.cc
@@ -22,9 +22,9 @@
 #include "p2p/test/fake_port_allocator.h"
 #include "rtc_base/internal/default_socket_server.h"
 #include "rtc_base/socket_server.h"
-#include "rtc_base/thread.h"
 #include "test/create_test_environment.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 
 namespace webrtc {
 
@@ -38,7 +38,7 @@ class IceTransportTest : public ::testing::Test {
 
  private:
   std::unique_ptr<SocketServer> socket_server_;
-  AutoSocketServerThread main_thread_;
+  test::RunLoop main_thread_;
 };
 
 TEST_F(IceTransportTest, CreateNonSelfDeletingTransport) {
diff --git a/pc/jsep_transport_controller_unittest.cc b/pc/jsep_transport_controller_unittest.cc
index e271753e24..136d2e4d72 100644
--- a/pc/jsep_transport_controller_unittest.cc
+++ b/pc/jsep_transport_controller_unittest.cc
@@ -67,6 +67,7 @@
 #include "test/create_test_field_trials.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 #include "test/wait_until.h"
 
 namespace webrtc {
@@ -357,7 +358,7 @@ class JsepTransportControllerTest : public JsepTransportController::Observer,
 
   FieldTrials field_trials_ = CreateTestFieldTrials();
   Environment env_;
-  AutoThread main_thread_;
+  test::RunLoop main_thread_;
   // Information received from signals from transport controller.
   IceConnectionState connection_state_ = kIceConnectionConnecting;
   PeerConnectionInterface::IceConnectionState ice_connection_state_ =
diff --git a/pc/jsep_transport_unittest.cc b/pc/jsep_transport_unittest.cc
index 3f49e7a755..b72267d808 100644
--- a/pc/jsep_transport_unittest.cc
+++ b/pc/jsep_transport_unittest.cc
@@ -53,9 +53,9 @@
 #include "rtc_base/ssl_fingerprint.h"
 #include "rtc_base/ssl_identity.h"
 #include "rtc_base/ssl_stream_adapter.h"
-#include "rtc_base/thread.h"
 #include "test/create_test_field_trials.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 
 namespace webrtc {
 namespace {
@@ -186,7 +186,7 @@ class JsepTransport2Test : public ::testing::Test {
 
   void OnRtcpMuxActive() { signal_rtcp_mux_active_received_ = true; }
 
-  AutoThread main_thread_;
+  test::RunLoop main_thread_;
   std::unique_ptr<JsepTransport> jsep_transport_;
   bool signal_rtcp_mux_active_received_ = false;
   FieldTrials field_trials_ = CreateTestFieldTrials();
diff --git a/pc/legacy_stats_collector_unittest.cc b/pc/legacy_stats_collector_unittest.cc
index f03911589b..94f7810a10 100644
--- a/pc/legacy_stats_collector_unittest.cc
+++ b/pc/legacy_stats_collector_unittest.cc
@@ -67,6 +67,7 @@
 #include "test/create_test_environment.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 
 namespace webrtc {
 
@@ -695,7 +696,7 @@ class LegacyStatsCollectorTest : public ::testing::Test {
       TimeDelta::Seconds(int64_t{365} * 50 * 86400)};  // 50 years offset.
 
  private:
-  AutoThread main_thread_;
+  test::RunLoop main_thread_;
 };
 
 static scoped_refptr<MockRtpSenderInternal> CreateMockSender(
diff --git a/pc/media_stream_unittest.cc b/pc/media_stream_unittest.cc
index de82b201e7..8611ee465e 100644
--- a/pc/media_stream_unittest.cc
+++ b/pc/media_stream_unittest.cc
@@ -20,6 +20,7 @@
 #include "rtc_base/thread.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 
 static const char kStreamId1[] = "local_stream_1";
 static const char kVideoTrackId[] = "dummy_video_cam_1";
@@ -82,7 +83,7 @@ class MediaStreamTest : public ::testing::Test {
     EXPECT_FALSE(track->enabled());
   }
 
-  AutoThread main_thread_;
+  test::RunLoop main_thread_;
   scoped_refptr<MediaStreamInterface> stream_;
   scoped_refptr<AudioTrackInterface> audio_track_;
   scoped_refptr<VideoTrackInterface> video_track_;
diff --git a/pc/peer_connection_bundle_unittest.cc b/pc/peer_connection_bundle_unittest.cc
index 192de541e4..6d4dc983cd 100644
--- a/pc/peer_connection_bundle_unittest.cc
+++ b/pc/peer_connection_bundle_unittest.cc
@@ -63,6 +63,7 @@
 #include "rtc_base/virtual_socket_server.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 #include "test/wait_until.h"
 
 #ifdef WEBRTC_ANDROID
@@ -264,7 +265,7 @@ class PeerConnectionBundleBaseTest : public ::testing::Test {
   }
 
   VirtualSocketServer vss_;
-  AutoSocketServerThread main_;
+  test::RunLoop main_;
   const SdpSemantics sdp_semantics_;
 };
 
diff --git a/pc/peer_connection_crypto_unittest.cc b/pc/peer_connection_crypto_unittest.cc
index 20d8553990..f7fee87ed5 100644
--- a/pc/peer_connection_crypto_unittest.cc
+++ b/pc/peer_connection_crypto_unittest.cc
@@ -52,6 +52,7 @@
 #include "rtc_base/thread.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 #include "test/wait_until.h"
 #ifdef WEBRTC_ANDROID
 #include "pc/test/android_test_initializer.h"
@@ -156,7 +157,7 @@ class PeerConnectionCryptoBaseTest : public ::testing::Test {
   }
 
   std::unique_ptr<VirtualSocketServer> vss_;
-  AutoSocketServerThread main_;
+  test::RunLoop main_;
   scoped_refptr<PeerConnectionFactoryInterface> pc_factory_;
   const SdpSemantics sdp_semantics_;
 };
diff --git a/pc/peer_connection_data_channel_unittest.cc b/pc/peer_connection_data_channel_unittest.cc
index 694f20702a..e086ce01a5 100644
--- a/pc/peer_connection_data_channel_unittest.cc
+++ b/pc/peer_connection_data_channel_unittest.cc
@@ -34,6 +34,7 @@
 #include "test/gmock.h"
 #include "test/gtest.h"
 #include "test/pc/sctp/fake_sctp_transport.h"
+#include "test/run_loop.h"
 
 #ifdef WEBRTC_ANDROID
 #include "pc/test/android_test_initializer.h"
@@ -158,7 +159,7 @@ class PeerConnectionDataChannelBaseTest : public ::testing::Test {
   }
 
   std::unique_ptr<VirtualSocketServer> vss_;
-  AutoSocketServerThread main_;
+  test::RunLoop main_;
   const SdpSemantics sdp_semantics_;
 };
 
diff --git a/pc/peer_connection_end_to_end_unittest.cc b/pc/peer_connection_end_to_end_unittest.cc
index 53c8daf9cd..ee6799cb3f 100644
--- a/pc/peer_connection_end_to_end_unittest.cc
+++ b/pc/peer_connection_end_to_end_unittest.cc
@@ -48,6 +48,7 @@
 #include "test/create_test_environment.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 #include "test/wait_until.h"
 
 #ifdef WEBRTC_ANDROID
@@ -246,7 +247,7 @@ class PeerConnectionEndToEndBaseTest : public ::testing::Test {
   }
 
  protected:
-  AutoThread main_thread_;
+  test::RunLoop main_thread_;
   PhysicalSocketServer pss_;
   Environment env_;
   std::unique_ptr<Thread> network_thread_;
@@ -287,13 +288,13 @@ std::unique_ptr<AudioDecoder> CreateForwardingMockDecoder(
       .WillRepeatedly([dec] { return dec->Channels(); });
   EXPECT_CALL(*mock_decoder, DecodeInternal(_, _, _, _, _))
       .Times(AtLeast(1))
-      .WillRepeatedly(
-          [dec](const uint8_t* encoded, size_t encoded_len, int sample_rate_hz,
-                int16_t* decoded, AudioDecoder::SpeechType* speech_type) {
-            return dec->Decode(encoded, encoded_len, sample_rate_hz,
-                               std::numeric_limits<size_t>::max(), decoded,
-                               speech_type);
-          });
+      .WillRepeatedly([dec](const uint8_t* encoded, size_t encoded_len,
+                            int sample_rate_hz, int16_t* decoded,
+                            AudioDecoder::SpeechType* speech_type) {
+        return dec->Decode(encoded, encoded_len, sample_rate_hz,
+                           std::numeric_limits<size_t>::max(), decoded,
+                           speech_type);
+      });
   EXPECT_CALL(*mock_decoder, Die());
   EXPECT_CALL(*mock_decoder, HasDecodePlc()).WillRepeatedly([dec] {
     return dec->HasDecodePlc();
@@ -326,15 +327,14 @@ scoped_refptr<AudioDecoderFactory> CreateForwardingMockDecoderFactory(
       });
   EXPECT_CALL(*mock_decoder_factory, Create)
       .Times(AtLeast(2))
-      .WillRepeatedly(
-          [real_decoder_factory](const Environment& env,
-                                 const SdpAudioFormat& format,
-                                 std::optional<AudioCodecPairId> /* pair */) {
-            auto real_decoder = real_decoder_factory->Create(env, format);
-            return real_decoder
-                       ? CreateForwardingMockDecoder(std::move(real_decoder))
-                       : nullptr;
-          });
+      .WillRepeatedly([real_decoder_factory](
+                          const Environment& env, const SdpAudioFormat& format,
+                          std::optional<AudioCodecPairId> /* pair */) {
+        auto real_decoder = real_decoder_factory->Create(env, format);
+        return real_decoder
+                   ? CreateForwardingMockDecoder(std::move(real_decoder))
+                   : nullptr;
+      });
   return mock_decoder_factory;
 }
 
@@ -690,7 +690,7 @@ TEST_P(PeerConnectionEndToEndTest, CloseDataChannelRemotelyWhileNotReferenced) {
 
   // Wait for a bit longer so the remote data channel will receive the
   // close message and be destroyed.
-  Thread::Current()->ProcessMessages(100);
+  main_thread_.RunFor(webrtc::TimeDelta::Millis(100));
 }
 
 // Test behavior of creating too many datachannels.
diff --git a/pc/peer_connection_factory_unittest.cc b/pc/peer_connection_factory_unittest.cc
index ed93e64c75..346125c287 100644
--- a/pc/peer_connection_factory_unittest.cc
+++ b/pc/peer_connection_factory_unittest.cc
@@ -68,6 +68,7 @@
 #include "rtc_base/thread.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 
 #ifdef WEBRTC_ANDROID
 #include "pc/test/android_test_initializer.h"
@@ -262,7 +263,7 @@ class PeerConnectionFactoryTest : public ::testing::Test {
   }
 
   std::unique_ptr<SocketServer> socket_server_;
-  AutoSocketServerThread main_thread_;
+  test::RunLoop main_thread_;
   scoped_refptr<PeerConnectionFactoryInterface> factory_;
   NullPeerConnectionObserver observer_;
   std::unique_ptr<FakePortAllocator> port_allocator_;
diff --git a/pc/peer_connection_field_trial_tests.cc b/pc/peer_connection_field_trial_tests.cc
index 4977bdd07f..fa323677b0 100644
--- a/pc/peer_connection_field_trial_tests.cc
+++ b/pc/peer_connection_field_trial_tests.cc
@@ -36,6 +36,7 @@
 #include "system_wrappers/include/clock.h"
 #include "test/create_test_field_trials.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 
 #ifdef WEBRTC_ANDROID
 #include "pc/test/android_test_initializer.h"
@@ -93,7 +94,7 @@ class PeerConnectionFieldTrialTest : public ::testing::Test {
 
   Clock* const clock_;
   std::unique_ptr<SocketServer> socket_server_;
-  AutoSocketServerThread main_thread_;
+  test::RunLoop main_thread_;
   scoped_refptr<PeerConnectionFactoryInterface> pc_factory_ = nullptr;
   PeerConnectionInterface::RTCConfiguration config_;
 };
diff --git a/pc/peer_connection_header_extension_unittest.cc b/pc/peer_connection_header_extension_unittest.cc
index efd84f94f5..a81cd9b2c7 100644
--- a/pc/peer_connection_header_extension_unittest.cc
+++ b/pc/peer_connection_header_extension_unittest.cc
@@ -40,6 +40,7 @@
 #include "test/create_test_field_trials.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 
 namespace webrtc {
 
@@ -116,7 +117,7 @@ class PeerConnectionHeaderExtensionTest
   }
 
   std::unique_ptr<SocketServer> socket_server_;
-  AutoSocketServerThread main_thread_;
+  test::RunLoop main_thread_;
   std::vector<RtpHeaderExtensionCapability> extensions_;
 };
 
diff --git a/pc/peer_connection_histogram_unittest.cc b/pc/peer_connection_histogram_unittest.cc
index 53b034c1ff..251d46df52 100644
--- a/pc/peer_connection_histogram_unittest.cc
+++ b/pc/peer_connection_histogram_unittest.cc
@@ -44,6 +44,7 @@
 #include "system_wrappers/include/metrics.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 #include "test/wait_until.h"
 
 namespace webrtc {
@@ -344,7 +345,7 @@ class PeerConnectionUsageHistogramTest : public ::testing::Test {
 
   int next_local_address_ = 0;
   VirtualSocketServer vss_;
-  AutoSocketServerThread main_;
+  test::RunLoop main_;
 };
 
 TEST_F(PeerConnectionUsageHistogramTest, UsageFingerprintHistogramFromTimeout) {
diff --git a/pc/peer_connection_interface_unittest.cc b/pc/peer_connection_interface_unittest.cc
index 15f4d319a5..7135acb106 100644
--- a/pc/peer_connection_interface_unittest.cc
+++ b/pc/peer_connection_interface_unittest.cc
@@ -92,6 +92,7 @@
 #include "rtc_base/weak_ptr.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 #include "test/wait_until.h"
 
 #ifdef WEBRTC_ANDROID
@@ -1290,7 +1291,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test {
   SocketServer* socket_server() const { return vss_.get(); }
 
   std::unique_ptr<VirtualSocketServer> vss_;
-  AutoSocketServerThread main_;
+  test::RunLoop main_;
   std::unique_ptr<Thread> network_thread_;
   std::unique_ptr<Thread> worker_thread_;
   scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
@@ -3978,6 +3979,7 @@ class PeerConnectionMediaConfigTest : public ::testing::Test {
 
   std::unique_ptr<Thread> network_thread_;
   std::unique_ptr<Thread> worker_thread_;
+  test::RunLoop signaling_thread_;
   scoped_refptr<PeerConnectionFactoryForTest> pcf_;
   MockPeerConnectionObserver observer_;
 };
diff --git a/pc/peer_connection_media_unittest.cc b/pc/peer_connection_media_unittest.cc
index a4671c9906..ea2a077d41 100644
--- a/pc/peer_connection_media_unittest.cc
+++ b/pc/peer_connection_media_unittest.cc
@@ -55,6 +55,7 @@
 #include "rtc_base/ref_counted_object.h"
 #include "rtc_base/thread.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 #ifdef WEBRTC_ANDROID
 #include "pc/test/android_test_initializer.h"
 #endif
@@ -245,7 +246,7 @@ class PeerConnectionMediaBaseTest : public ::testing::Test {
   }
 
   std::unique_ptr<VirtualSocketServer> vss_;
-  AutoSocketServerThread main_;
+  test::RunLoop main_;
   const SdpSemantics sdp_semantics_;
 };
 
@@ -1576,11 +1577,10 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan,
   auto codecs =
       caller->pc_factory()->GetRtpSenderCapabilities(MediaType::AUDIO).codecs;
   auto codecs_only_rtx_red_fec = codecs;
-  std::erase_if(
-      codecs_only_rtx_red_fec, [](const RtpCodecCapability& codec) {
-        return !(codec.name == kRtxCodecName || codec.name == kRedCodecName ||
-                 codec.name == kUlpfecCodecName);
-      });
+  std::erase_if(codecs_only_rtx_red_fec, [](const RtpCodecCapability& codec) {
+    return !(codec.name == kRtxCodecName || codec.name == kRedCodecName ||
+             codec.name == kUlpfecCodecName);
+  });
   ASSERT_THAT(codecs_only_rtx_red_fec.size(), Gt(0));
   auto result = transceiver->SetCodecPreferences(codecs_only_rtx_red_fec);
   EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, result.type());
@@ -1652,11 +1652,10 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan,
   auto codecs =
       caller->pc_factory()->GetRtpSenderCapabilities(MediaType::VIDEO).codecs;
   auto codecs_only_rtx_red_fec = codecs;
-  std::erase_if(
-      codecs_only_rtx_red_fec, [](const RtpCodecCapability& codec) {
-        return !(codec.name == kRtxCodecName || codec.name == kRedCodecName ||
-                 codec.name == kUlpfecCodecName);
-      });
+  std::erase_if(codecs_only_rtx_red_fec, [](const RtpCodecCapability& codec) {
+    return !(codec.name == kRtxCodecName || codec.name == kRedCodecName ||
+             codec.name == kUlpfecCodecName);
+  });
 
   auto result = transceiver->SetCodecPreferences(codecs_only_rtx_red_fec);
   EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, result.type());
diff --git a/pc/peer_connection_rampup_tests.cc b/pc/peer_connection_rampup_tests.cc
index e967536357..b0c76c2576 100644
--- a/pc/peer_connection_rampup_tests.cc
+++ b/pc/peer_connection_rampup_tests.cc
@@ -40,7 +40,6 @@
 #include "api/video_codecs/video_encoder_factory_template_libvpx_vp8_adapter.h"
 #include "api/video_codecs/video_encoder_factory_template_libvpx_vp9_adapter.h"
 #include "api/video_codecs/video_encoder_factory_template_open_h264_adapter.h"
-#include "p2p/base/port_interface.h"
 #include "p2p/test/test_turn_server.h"
 #include "pc/peer_connection.h"
 #include "pc/peer_connection_wrapper.h"
@@ -51,6 +50,7 @@
 #include "rtc_base/crypto_random.h"
 #include "rtc_base/fake_network.h"
 #include "rtc_base/firewall_socket_server.h"
+#include "rtc_base/net_helper.h"
 #include "rtc_base/socket_address.h"
 #include "rtc_base/socket_factory.h"
 #include "rtc_base/task_queue_for_test.h"
diff --git a/pc/peer_connection_signaling_unittest.cc b/pc/peer_connection_signaling_unittest.cc
index 368ef4a440..6bc9c294f5 100644
--- a/pc/peer_connection_signaling_unittest.cc
+++ b/pc/peer_connection_signaling_unittest.cc
@@ -65,6 +65,7 @@
 #include "rtc_base/virtual_socket_server.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 #include "test/wait_until.h"
 
 #ifdef WEBRTC_ANDROID
@@ -198,7 +199,7 @@ class PeerConnectionSignalingBaseTest : public ::testing::Test {
   }
 
   std::unique_ptr<VirtualSocketServer> vss_;
-  AutoSocketServerThread main_;
+  test::RunLoop main_;
   scoped_refptr<PeerConnectionFactoryInterface> pc_factory_;
   const SdpSemantics sdp_semantics_;
 };
diff --git a/pc/rtp_transport_unittest.cc b/pc/rtp_transport_unittest.cc
index adbe70822a..65762faec8 100644
--- a/pc/rtp_transport_unittest.cc
+++ b/pc/rtp_transport_unittest.cc
@@ -32,7 +32,6 @@
 #include "rtc_base/logging.h"
 #include "rtc_base/network/sent_packet.h"
 #include "rtc_base/network_route.h"
-#include "rtc_base/thread.h"
 #include "test/create_test_field_trials.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
@@ -240,7 +239,7 @@ TEST(RtpTransportTest, SetRtcpTransportWithNetworkRouteChanged) {
 TEST(RtpTransportTest, RtcpPacketSentOverCorrectTransport) {
   // If the RTCP-mux is not enabled, RTCP packets are expected to be sent over
   // the RtcpPacketTransport.
-  AutoThread thread;
+  test::RunLoop thread;
   RtpTransport transport(kMuxDisabled, CreateTestFieldTrials());
   FakePacketTransport fake_rtcp("fake_rtcp");
   FakePacketTransport fake_rtp("fake_rtp");
@@ -291,7 +290,7 @@ TEST(RtpTransportTest, ChangingReadyToSendStateOnlySignalsWhenChanged) {
 }
 
 TEST(RtpTransportTest, RegisterAndUnregisterRtpHeaderExtensionMap) {
-  AutoThread thread;
+  test::RunLoop thread;
   RtpTransport transport(kMuxDisabled, CreateTestFieldTrials());
   RtpHeaderExtensions extensions1 = {
       RtpExtension("urn:ietf:params:rtp-hdrext:ssrc-audio-level", 1)};
@@ -394,7 +393,7 @@ TEST(RtpTransportTest, VerifyRtpHeaderExtensionMapRejectsIdReassignment) {
 // Test that SignalPacketReceived fires with rtcp=true when a RTCP packet is
 // received.
 TEST(RtpTransportTest, SignalDemuxedRtcp) {
-  AutoThread thread;
+  test::RunLoop thread;
   RtpTransport transport(kMuxDisabled, CreateTestFieldTrials());
   FakePacketTransport fake_rtp("fake_rtp");
   fake_rtp.SetDestination(&fake_rtp, true);
@@ -418,7 +417,7 @@ const int kRtpLen = 12;
 // Test that SignalPacketReceived fires with rtcp=false when a RTP packet with a
 // handled payload type is received.
 TEST(RtpTransportTest, SignalHandledRtpPayloadType) {
-  AutoThread thread;
+  test::RunLoop thread;
   RtpTransport transport(kMuxDisabled, CreateTestFieldTrials());
   FakePacketTransport fake_rtp("fake_rtp");
   fake_rtp.SetDestination(&fake_rtp, true);
@@ -443,7 +442,7 @@ TEST(RtpTransportTest, SignalHandledRtpPayloadType) {
 }
 
 TEST(RtpTransportTest, ReceivedPacketEcnMarkingPropagatedToDemuxedPacket) {
-  AutoThread thread;
+  test::RunLoop thread;
   RtpTransport transport(kMuxDisabled, CreateTestFieldTrials());
   // Setup FakePacketTransport to send packets to itself.
   FakePacketTransport fake_rtp("fake_rtp");
@@ -468,7 +467,7 @@ TEST(RtpTransportTest, ReceivedPacketEcnMarkingPropagatedToDemuxedPacket) {
 }
 
 TEST(RtpTransportTest, RtcpSentAsEct1IfReceivedRtpPacketAsEct1) {
-  AutoThread thread;
+  test::RunLoop thread;
   RtpTransport transport(kMuxDisabled, CreateTestFieldTrials());
   // Setup FakePacketTransport to send packets to itself.
   FakePacketTransport fake_rtp("fake_rtp");
@@ -504,7 +503,7 @@ TEST(RtpTransportTest, RtcpSentAsEct1IfReceivedRtpPacketAsEct1) {
 // Test that SignalPacketReceived does not fire when a RTP packet with an
 // unhandled payload type is received.
 TEST(RtpTransportTest, DontSignalUnhandledRtpPayloadType) {
-  AutoThread thread;
+  test::RunLoop thread;
   RtpTransport transport(kMuxDisabled, CreateTestFieldTrials());
   FakePacketTransport fake_rtp("fake_rtp");
   fake_rtp.SetDestination(&fake_rtp, true);
@@ -530,7 +529,7 @@ TEST(RtpTransportTest, DontSignalUnhandledRtpPayloadType) {
 
 TEST(RtpTransportTest, DontChangeReadyToSendStateOnSendFailure) {
   // ReadyToSendState should only care about if transport is writable.
-  AutoThread thread;
+  test::RunLoop thread;
   RtpTransport transport(kMuxEnabled, CreateTestFieldTrials());
   TransportObserver observer(&transport);
 
diff --git a/pc/sctp_transport_unittest.cc b/pc/sctp_transport_unittest.cc
index c7a8a4bd97..d3b796e02f 100644
--- a/pc/sctp_transport_unittest.cc
+++ b/pc/sctp_transport_unittest.cc
@@ -32,9 +32,9 @@
 #include "p2p/dtls/fake_dtls_transport.h"
 #include "pc/dtls_transport.h"
 #include "rtc_base/copy_on_write_buffer.h"
-#include "rtc_base/thread.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 #include "test/wait_until.h"
 
 namespace webrtc {
@@ -161,7 +161,7 @@ class SctpTransportTest : public ::testing::Test {
     return static_cast<FakeSctpTransportInternal*>(transport_->internal());
   }
 
-  AutoThread main_thread_;
+  test::RunLoop main_thread_;
   scoped_refptr<SctpTransport> transport_;
   scoped_refptr<DtlsTransport> dtls_transport_;
   std::unique_ptr<FakeDtlsTransport> internal_transport_;
@@ -169,7 +169,7 @@ class SctpTransportTest : public ::testing::Test {
 };
 
 TEST(SctpTransportSimpleTest, CreateClearDelete) {
-  AutoThread main_thread;
+  test::RunLoop main_thread;
   std::unique_ptr<DtlsTransportInternal> internal_transport =
       std::make_unique<FakeDtlsTransport>("audio", ICE_CANDIDATE_COMPONENT_RTP);
   scoped_refptr<DtlsTransport> dtls_transport =
diff --git a/pc/sdp_munging_detector_unittest.cc b/pc/sdp_munging_detector_unittest.cc
index 2c8430c209..369837d060 100644
--- a/pc/sdp_munging_detector_unittest.cc
+++ b/pc/sdp_munging_detector_unittest.cc
@@ -66,6 +66,7 @@
 #include "test/create_test_field_trials.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 #include "test/wait_until.h"
 
 // This file contains unit tests that relate to the behavior of the
@@ -149,7 +150,7 @@ class SdpMungingTest : public ::testing::Test {
   scoped_refptr<PeerConnectionFactoryInterface> pc_factory_;
 
  private:
-  AutoThread main_thread_;
+  test::RunLoop main_thread_;
 };
 
 TEST_F(SdpMungingTest, DISABLED_ReportUMAMetricsWithNoMunging) {
diff --git a/pc/sdp_offer_answer_unittest.cc b/pc/sdp_offer_answer_unittest.cc
index a4a9c62843..0749978dcd 100644
--- a/pc/sdp_offer_answer_unittest.cc
+++ b/pc/sdp_offer_answer_unittest.cc
@@ -57,6 +57,7 @@
 #include "test/create_test_field_trials.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 
 // This file contains unit tests that relate to the behavior of the
 // SdpOfferAnswer module.
@@ -153,7 +154,7 @@ class SdpOfferAnswerTest : public ::testing::Test {
   scoped_refptr<PeerConnectionFactoryInterface> pc_factory_;
 
  private:
-  AutoThread main_thread_;
+  test::RunLoop main_thread_;
 };
 
 TEST_F(SdpOfferAnswerTest, OnTrackReturnsProxiedObject) {
diff --git a/pc/srtp_transport_unittest.cc b/pc/srtp_transport_unittest.cc
index 20a7c458fb..163c5095db 100644
--- a/pc/srtp_transport_unittest.cc
+++ b/pc/srtp_transport_unittest.cc
@@ -33,9 +33,9 @@
 #include "rtc_base/containers/flat_set.h"
 #include "rtc_base/copy_on_write_buffer.h"
 #include "rtc_base/ssl_stream_adapter.h"
-#include "rtc_base/thread.h"
 #include "test/create_test_field_trials.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 
 using ::webrtc::kSrtpAeadAes128Gcm;
 using ::webrtc::kTestKey1;
@@ -280,7 +280,7 @@ class SrtpTransportTest : public ::testing::Test {
     TestSendRecvPacketWithEncryptedHeaderExtension(crypto_suite,
                                                    encrypted_headers);
   }
-  AutoThread main_thread;
+  test::RunLoop main_thread;
 
   std::unique_ptr<SrtpTransport> srtp_transport1_;
   std::unique_ptr<SrtpTransport> srtp_transport2_;
diff --git a/pc/test/fake_audio_capture_module_unittest.cc b/pc/test/fake_audio_capture_module_unittest.cc
index 42d1a7202e..63d3580432 100644
--- a/pc/test/fake_audio_capture_module_unittest.cc
+++ b/pc/test/fake_audio_capture_module_unittest.cc
@@ -19,9 +19,9 @@
 #include "api/test/rtc_error_matchers.h"
 #include "api/units/time_delta.h"
 #include "rtc_base/synchronization/mutex.h"
-#include "rtc_base/thread.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 #include "test/wait_until.h"
 
 class FakeAdmTest : public ::testing::Test, public webrtc::AudioTransport {
@@ -127,7 +127,7 @@ class FakeAdmTest : public ::testing::Test, public webrtc::AudioTransport {
     return min_buffer_size;
   }
 
-  webrtc::AutoThread main_thread_;
+  webrtc::test::RunLoop main_thread_;
 
   mutable webrtc::Mutex mutex_;
 
diff --git a/pc/track_media_info_map_unittest.cc b/pc/track_media_info_map_unittest.cc
index fd46ed5759..2b50e4078f 100644
--- a/pc/track_media_info_map_unittest.cc
+++ b/pc/track_media_info_map_unittest.cc
@@ -33,6 +33,7 @@
 #include "rtc_base/thread.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 
 using ::testing::ElementsAre;
 
@@ -210,7 +211,7 @@ class TrackMediaInfoMapTest : public ::testing::Test {
   }
 
  private:
-  AutoThread main_thread_;
+  test::RunLoop main_thread_;
   VoiceMediaInfo voice_media_info_;
   VideoMediaInfo video_media_info_;
 
diff --git a/pc/video_track_unittest.cc b/pc/video_track_unittest.cc
index a0a078c84b..259d415226 100644
--- a/pc/video_track_unittest.cc
+++ b/pc/video_track_unittest.cc
@@ -22,6 +22,7 @@
 #include "pc/test/fake_video_track_source.h"
 #include "rtc_base/thread.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 
 namespace webrtc {
 namespace {
@@ -38,7 +39,7 @@ class VideoTrackTest : public ::testing::Test {
   }
 
  protected:
-  AutoThread main_thread_;
+  test::RunLoop main_thread_;
   scoped_refptr<FakeVideoTrackSource> video_track_source_;
   scoped_refptr<VideoTrack> video_track_;
   FakeFrameSource frame_source_;

From 9f086f0600c608bae2966bb5b005e92d1ac371c6 Mon Sep 17 00:00:00 2001
From: Evan Shrubsole <eshr@webrtc.org>
Date: Tue, 31 Mar 2026 08:40:33 +0000
Subject: [PATCH 452/472] Use RunLoop in test/ tests

Bug: webrtc:469327588
Change-Id: I5306d9aaa98e12522d829f85cd405ce76a6a6964
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/461341
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47305}
---
 test/wait_until.cc          |  2 +-
 test/wait_until_unittest.cc | 12 ++++++------
 2 files changed, 7 insertions(+), 7 deletions(-)

diff --git a/test/wait_until.cc b/test/wait_until.cc
index 56ce22c12a..36ca1bac71 100644
--- a/test/wait_until.cc
+++ b/test/wait_until.cc
@@ -28,7 +28,7 @@ namespace webrtc {
                              WaitUntilSettings settings) {
   if (std::holds_alternative<std::monostate>(settings.clock)) {
     RTC_CHECK(Thread::Current()) << "A current thread is required. An "
-                                    "webrtc::AutoThread can work for tests.";
+                                    "webrtc::test::RunLoop can work for tests.";
   }
 
   auto now = [&] {
diff --git a/test/wait_until_unittest.cc b/test/wait_until_unittest.cc
index a78a377010..b0391ccd66 100644
--- a/test/wait_until_unittest.cc
+++ b/test/wait_until_unittest.cc
@@ -21,10 +21,10 @@
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
 #include "rtc_base/fake_clock.h"
-#include "rtc_base/thread.h"
 #include "system_wrappers/include/clock.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 
 namespace webrtc {
 namespace {
@@ -39,7 +39,7 @@ using ::testing::MatchesRegex;
 using ::testing::Property;
 
 TEST(WaitUntilTest, ReturnsTrueWhenConditionIsMet) {
-  AutoThread thread;
+  test::RunLoop thread;
 
   int counter = 0;
   EXPECT_TRUE(WaitUntil([&] { return ++counter == 3; }));
@@ -49,7 +49,7 @@ TEST(WaitUntilTest, ReturnsTrueWhenConditionIsMet) {
 }
 
 TEST(WaitUntilTest, ReturnsWhenConditionIsMet) {
-  AutoThread thread;
+  test::RunLoop thread;
 
   int counter = 0;
   RTCErrorOr<int> result = WaitUntil([&] { return ++counter; }, Eq(3));
@@ -57,7 +57,7 @@ TEST(WaitUntilTest, ReturnsWhenConditionIsMet) {
 }
 
 TEST(WaitUntilTest, ReturnsErrorWhenTimeoutIsReached) {
-  AutoThread thread;
+  test::RunLoop thread;
   int counter = 0;
   RTCErrorOr<int> result =
       WaitUntil([&] { return --counter; }, Eq(1),
@@ -73,7 +73,7 @@ TEST(WaitUntilTest, ReturnsErrorWhenTimeoutIsReached) {
 }
 
 TEST(WaitUntilTest, ErrorContainsMatcherExplanation) {
-  AutoThread thread;
+  test::RunLoop thread;
   int counter = 0;
   auto matcher = AllOf(Gt(0), Lt(10));
   RTCErrorOr<int> result =
@@ -153,7 +153,7 @@ class CustomType {
 };
 
 TEST(WaitUntilTest, RequiresOnlyMoveCopyConstructionForReturnedType) {
-  AutoThread thread;
+  test::RunLoop thread;
 
   int counter = 0;
   RTCErrorOr<CustomType> result =

From 6331862bcd72668587a59fa1aae395bc153104ef Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?=C3=85sa=20Persson?= <asapersson@webrtc.org>
Date: Tue, 31 Mar 2026 12:36:10 +0000
Subject: [PATCH 453/472] Group members in VCMTiming into a VideoDelayTimings
 struct.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Bug: b/493549134
Change-Id: I8af7d30ecb49bc0180c6d631166277052cfb1d8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/461220
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47306}
---
 modules/video_coding/timing/timing.cc         | 107 +++++++++---------
 modules/video_coding/timing/timing.h          |  39 ++++---
 .../video_coding/timing/timing_unittest.cc    |  29 +++--
 3 files changed, 90 insertions(+), 85 deletions(-)

diff --git a/modules/video_coding/timing/timing.cc b/modules/video_coding/timing/timing.cc
index 79f71f39a8..7bea914ac3 100644
--- a/modules/video_coding/timing/timing.cc
+++ b/modules/video_coding/timing/timing.cc
@@ -31,7 +31,6 @@
 namespace webrtc {
 namespace {
 
-constexpr TimeDelta kDefaultRenderDelay = TimeDelta::Millis(10);
 // Default pacing that is used for the low-latency renderer path.
 constexpr TimeDelta kZeroPlayoutDelayDefaultMinPacing = TimeDelta::Millis(8);
 constexpr TimeDelta kLowLatencyStreamMaxPlayoutDelayThreshold =
@@ -54,18 +53,26 @@ void CheckDelaysValid(TimeDelta min_delay, TimeDelta max_delay) {
 
 }  // namespace
 
+void VCMTiming::VideoDelayTimings::Reset() {
+  minimum_delay = TimeDelta::Zero();
+  estimated_max_decode_time = TimeDelta::Zero();
+  render_delay = kDefaultRenderDelay;
+  min_playout_delay = TimeDelta::Zero();
+  target_delay = TimeDelta::Zero();
+  current_delay = TimeDelta::Zero();
+}
+
+bool VCMTiming::VideoDelayTimings::UseLowLatencyRendering() const {
+  return min_playout_delay.IsZero() &&
+         max_playout_delay <= kLowLatencyStreamMaxPlayoutDelayThreshold;
+}
+
 VCMTiming::VCMTiming(Clock* clock, const FieldTrialsView& field_trials)
     : clock_(clock),
       ts_extrapolator_(
           std::make_unique<TimestampExtrapolator>(clock_->CurrentTime(),
                                                   field_trials)),
       decode_time_filter_(std::make_unique<DecodeTimePercentileFilter>()),
-      render_delay_(kDefaultRenderDelay),
-      min_playout_delay_(TimeDelta::Zero()),
-      max_playout_delay_(TimeDelta::Seconds(10)),
-      jitter_delay_(TimeDelta::Zero()),
-      current_delay_(TimeDelta::Zero()),
-      num_decoded_frames_(0),
       zero_playout_delay_min_pacing_("min_pacing",
                                      kZeroPlayoutDelayDefaultMinPacing),
       last_decode_scheduled_(Timestamp::Zero()) {
@@ -77,27 +84,24 @@ void VCMTiming::Reset() {
   MutexLock lock(&mutex_);
   ts_extrapolator_->Reset(clock_->CurrentTime());
   decode_time_filter_ = std::make_unique<DecodeTimePercentileFilter>();
-  render_delay_ = kDefaultRenderDelay;
-  min_playout_delay_ = TimeDelta::Zero();
-  jitter_delay_ = TimeDelta::Zero();
-  current_delay_ = TimeDelta::Zero();
+  timings_.Reset();
 }
 
 void VCMTiming::set_render_delay(TimeDelta render_delay) {
   MutexLock lock(&mutex_);
-  render_delay_ = render_delay;
+  timings_.render_delay = render_delay;
 }
 
 TimeDelta VCMTiming::min_playout_delay() const {
   MutexLock lock(&mutex_);
-  return min_playout_delay_;
+  return timings_.min_playout_delay;
 }
 
 void VCMTiming::set_min_playout_delay(TimeDelta min_playout_delay) {
   MutexLock lock(&mutex_);
-  if (min_playout_delay_ != min_playout_delay) {
-    CheckDelaysValid(min_playout_delay, max_playout_delay_);
-    min_playout_delay_ = min_playout_delay;
+  if (timings_.min_playout_delay != min_playout_delay) {
+    CheckDelaysValid(min_playout_delay, timings_.max_playout_delay);
+    timings_.min_playout_delay = min_playout_delay;
   }
 }
 
@@ -105,17 +109,17 @@ void VCMTiming::set_playout_delay(const VideoPlayoutDelay& playout_delay) {
   MutexLock lock(&mutex_);
   // No need to call `CheckDelaysValid` as the same invariant (min <= max)
   // is guaranteed by the `VideoPlayoutDelay` type.
-  min_playout_delay_ = playout_delay.min();
-  max_playout_delay_ = playout_delay.max();
+  timings_.min_playout_delay = playout_delay.min();
+  timings_.max_playout_delay = playout_delay.max();
 }
 
-void VCMTiming::SetJitterDelay(TimeDelta jitter_delay) {
+void VCMTiming::SetJitterDelay(TimeDelta minimum_delay) {
   MutexLock lock(&mutex_);
-  if (jitter_delay != jitter_delay_) {
-    jitter_delay_ = jitter_delay;
+  if (minimum_delay != timings_.minimum_delay) {
+    timings_.minimum_delay = minimum_delay;
     // When in initial state, set current delay to minimum delay.
-    if (current_delay_.IsZero()) {
-      current_delay_ = jitter_delay_;
+    if (timings_.current_delay.IsZero()) {
+      timings_.current_delay = timings_.minimum_delay;
     }
   }
 }
@@ -125,16 +129,16 @@ void VCMTiming::UpdateCurrentDelay(Timestamp render_time,
   MutexLock lock(&mutex_);
   TimeDelta target_delay = TargetDelayInternal();
   TimeDelta delayed = (actual_decode_time - render_time) +
-                      EstimatedMaxDecodeTime() + render_delay_;
+                      EstimatedMaxDecodeTime() + timings_.render_delay;
 
   // Only consider `delayed` as negative by more than a few microseconds.
   if (delayed.ms() < 0) {
     return;
   }
-  if (current_delay_ + delayed <= target_delay) {
-    current_delay_ += delayed;
+  if (timings_.current_delay + delayed <= target_delay) {
+    timings_.current_delay += delayed;
   } else {
-    current_delay_ = target_delay;
+    timings_.current_delay = target_delay;
   }
 }
 
@@ -142,7 +146,7 @@ void VCMTiming::StopDecodeTimer(TimeDelta decode_time, Timestamp now) {
   MutexLock lock(&mutex_);
   decode_time_filter_->AddTiming(decode_time.ms(), now.ms());
   RTC_DCHECK_GE(decode_time, TimeDelta::Zero());
-  ++num_decoded_frames_;
+  ++timings_.num_decoded_frames;
 }
 
 void VCMTiming::IncomingTimestamp(uint32_t rtp_timestamp, Timestamp now) {
@@ -163,7 +167,7 @@ void VCMTiming::SetLastDecodeScheduledTimestamp(
 
 Timestamp VCMTiming::RenderTimeInternal(uint32_t frame_timestamp,
                                         Timestamp now) const {
-  if (UseLowLatencyRendering()) {
+  if (timings_.UseLowLatencyRendering()) {
     // Render as soon as possible or with low-latency renderer algorithm.
     return Timestamp::Zero();
   }
@@ -176,10 +180,11 @@ Timestamp VCMTiming::RenderTimeInternal(uint32_t frame_timestamp,
   }
   Timestamp estimated_complete_time = *local_time;
 
-  // Make sure the actual delay stays in the range of `min_playout_delay_`
-  // and `max_playout_delay_`.
+  // Make sure the actual delay stays in the range of `min_playout_delay`
+  // and `max_playout_delay`.
   TimeDelta actual_delay =
-      std::clamp(current_delay_, min_playout_delay_, max_playout_delay_);
+      std::clamp(timings_.current_delay, timings_.min_playout_delay,
+                 timings_.max_playout_delay);
   return estimated_complete_time + actual_delay;
 }
 
@@ -195,7 +200,8 @@ TimeDelta VCMTiming::MaxWaitingTime(Timestamp render_time,
   MutexLock lock(&mutex_);
 
   if (render_time.IsZero() && zero_playout_delay_min_pacing_->us() > 0 &&
-      min_playout_delay_.IsZero() && max_playout_delay_ > TimeDelta::Zero()) {
+      timings_.min_playout_delay.IsZero() &&
+      timings_.max_playout_delay > TimeDelta::Zero()) {
     // `render_time` == 0 indicates that the frame should be decoded and
     // rendered as soon as possible. However, the decoder can be choked if too
     // many frames are sent at once. Therefore, limit the interframe delay to
@@ -211,7 +217,7 @@ TimeDelta VCMTiming::MaxWaitingTime(Timestamp render_time,
                                   : earliest_next_decode_start_time - now;
     return max_wait_time;
   }
-  return render_time - now - EstimatedMaxDecodeTime() - render_delay_;
+  return render_time - now - EstimatedMaxDecodeTime() - timings_.render_delay;
 }
 
 TimeDelta VCMTiming::TargetVideoDelay() const {
@@ -220,43 +226,32 @@ TimeDelta VCMTiming::TargetVideoDelay() const {
 }
 
 TimeDelta VCMTiming::TargetDelayInternal() const {
-  return std::max(min_playout_delay_,
-                  jitter_delay_ + EstimatedMaxDecodeTime() + render_delay_);
+  return std::max(timings_.min_playout_delay, timings_.minimum_delay +
+                                                  EstimatedMaxDecodeTime() +
+                                                  timings_.render_delay);
 }
 
 // TODO(crbug.com/webrtc/15197): Centralize delay arithmetic.
 TimeDelta VCMTiming::StatsTargetDelayInternal() const {
   TimeDelta stats_target_delay =
-      TargetDelayInternal() - (EstimatedMaxDecodeTime() + render_delay_);
+      TargetDelayInternal() -
+      (EstimatedMaxDecodeTime() + timings_.render_delay);
   return std::max(TimeDelta::Zero(), stats_target_delay);
 }
 
 VideoFrame::RenderParameters VCMTiming::RenderParameters() const {
   MutexLock lock(&mutex_);
-  return {.use_low_latency_rendering = UseLowLatencyRendering(),
+  return {.use_low_latency_rendering = timings_.UseLowLatencyRendering(),
           .max_composition_delay_in_frames = max_composition_delay_in_frames_};
 }
 
-bool VCMTiming::UseLowLatencyRendering() const {
-  // min_playout_delay_==0,
-  // max_playout_delay_<=kLowLatencyStreamMaxPlayoutDelayThreshold indicates
-  // that the low-latency path should be used, which means that frames should be
-  // decoded and rendered as soon as possible.
-  return min_playout_delay_.IsZero() &&
-         max_playout_delay_ <= kLowLatencyStreamMaxPlayoutDelayThreshold;
-}
-
 VCMTiming::VideoDelayTimings VCMTiming::GetTimings() const {
   MutexLock lock(&mutex_);
-  return VideoDelayTimings{
-      .num_decoded_frames = num_decoded_frames_,
-      .minimum_delay = jitter_delay_,
-      .estimated_max_decode_time = EstimatedMaxDecodeTime(),
-      .render_delay = render_delay_,
-      .min_playout_delay = min_playout_delay_,
-      .max_playout_delay = max_playout_delay_,
-      .target_delay = StatsTargetDelayInternal(),
-      .current_delay = current_delay_};
+  VideoDelayTimings timings = timings_;
+  // TODO(b/493549134): Update in StopDecodeTimer.
+  timings.estimated_max_decode_time = EstimatedMaxDecodeTime();
+  timings.target_delay = StatsTargetDelayInternal();
+  return timings;
 }
 
 void VCMTiming::SetMaxCompositionDelayInFrames(
diff --git a/modules/video_coding/timing/timing.h b/modules/video_coding/timing/timing.h
index 9ee0d8add0..5ae772a05a 100644
--- a/modules/video_coding/timing/timing.h
+++ b/modules/video_coding/timing/timing.h
@@ -33,27 +33,34 @@ namespace webrtc {
 class VCMTiming {
  public:
   struct VideoDelayTimings {
-    size_t num_decoded_frames;
+    static constexpr TimeDelta kDefaultRenderDelay = TimeDelta::Millis(10);
+
+    void Reset();
+    // Returns whether the low-latency path should be used, i.e., frames should
+    // be decoded and rendered as soon as possible.
+    bool UseLowLatencyRendering() const;
+
+    size_t num_decoded_frames = 0;
     // Pre-decode delay added to smooth out frame delay variation ("jitter")
     // caused by the network. The target delay will be no smaller than this
     // delay, thus it is called `minimum_delay`.
-    TimeDelta minimum_delay;
+    TimeDelta minimum_delay = TimeDelta::Zero();
     // Estimated time needed to decode a video frame. Obtained as the 95th
     // percentile decode time over a recent time window.
-    TimeDelta estimated_max_decode_time;
+    TimeDelta estimated_max_decode_time = TimeDelta::Zero();
     // Post-decode delay added to smooth out frame delay variation caused by
     // decoding and rendering. Set to a constant.
-    TimeDelta render_delay;
+    TimeDelta render_delay = kDefaultRenderDelay;
     // Minimum total delay used when determining render time for a frame.
     // Obtained from API, `playout-delay` RTP header extension, or A/V sync.
-    TimeDelta min_playout_delay;
+    TimeDelta min_playout_delay = TimeDelta::Zero();
     // Maximum total delay used when determining render time for a frame.
     // Obtained from `playout-delay` RTP header extension.
-    TimeDelta max_playout_delay;
+    TimeDelta max_playout_delay = TimeDelta::Seconds(10);
     // Target total delay. Obtained from all the elements above.
-    TimeDelta target_delay;
+    TimeDelta target_delay = TimeDelta::Zero();
     // Current total delay. Obtained by smoothening the `target_delay`.
-    TimeDelta current_delay;
+    TimeDelta current_delay = TimeDelta::Zero();
   };
 
   VCMTiming(Clock* clock, const FieldTrialsView& field_trials);
@@ -129,7 +136,6 @@ class VCMTiming {
   TimeDelta TargetDelayInternal() const RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
   TimeDelta StatsTargetDelayInternal() const
       RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
-  bool UseLowLatencyRendering() const RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
 
   mutable Mutex mutex_;
   Clock* const clock_;
@@ -137,17 +143,10 @@ class VCMTiming {
       RTC_PT_GUARDED_BY(mutex_);
   std::unique_ptr<DecodeTimePercentileFilter> decode_time_filter_
       RTC_GUARDED_BY(mutex_) RTC_PT_GUARDED_BY(mutex_);
-  TimeDelta render_delay_ RTC_GUARDED_BY(mutex_);
-  // Best-effort playout delay range for frames from capture to render.
-  // The receiver tries to keep the delay between `min_playout_delay_ms_`
-  // and `max_playout_delay_ms_` taking the network jitter into account.
-  // A special case is where min_playout_delay_ms_ = max_playout_delay_ms_ = 0,
-  // in which case the receiver tries to play the frames as they arrive.
-  TimeDelta min_playout_delay_ RTC_GUARDED_BY(mutex_);
-  TimeDelta max_playout_delay_ RTC_GUARDED_BY(mutex_);
-  TimeDelta jitter_delay_ RTC_GUARDED_BY(mutex_);
-  TimeDelta current_delay_ RTC_GUARDED_BY(mutex_);
-  size_t num_decoded_frames_ RTC_GUARDED_BY(mutex_);
+
+  // Holds the current video delay timings.
+  VideoDelayTimings timings_ RTC_GUARDED_BY(mutex_);
+
   std::optional<int> max_composition_delay_in_frames_ RTC_GUARDED_BY(mutex_);
   // Set by the field trial WebRTC-ZeroPlayoutDelay. The parameter min_pacing
   // determines the minimum delay between frames scheduled for decoding that is
diff --git a/modules/video_coding/timing/timing_unittest.cc b/modules/video_coding/timing/timing_unittest.cc
index afe11d0217..b2ea1ad7ef 100644
--- a/modules/video_coding/timing/timing_unittest.cc
+++ b/modules/video_coding/timing/timing_unittest.cc
@@ -119,7 +119,6 @@ TEST(VCMTimingTest, UseLowLatencyRenderer) {
   FieldTrials field_trials = CreateTestFieldTrials();
   SimulatedClock clock(0);
   VCMTiming timing(&clock, field_trials);
-  timing.Reset();
   // Default is false.
   EXPECT_FALSE(timing.RenderParameters().use_low_latency_rendering);
   // False if min playout delay > 0.
@@ -150,7 +149,6 @@ TEST(VCMTimingTest, MaxWaitingTimeIsZeroForZeroRenderTime) {
   SimulatedClock clock(kStartTimeUs);
   FieldTrials field_trials = CreateTestFieldTrials();
   VCMTiming timing(&clock, field_trials);
-  timing.Reset();
   timing.set_playout_delay({TimeDelta::Zero(), TimeDelta::Zero()});
   for (int i = 0; i < 10; ++i) {
     clock.AdvanceTime(kTimeDelta);
@@ -190,7 +188,6 @@ TEST(VCMTimingTest, MaxWaitingTimeZeroDelayPacingExperiment) {
   constexpr auto kZeroRenderTime = Timestamp::Zero();
   SimulatedClock clock(kStartTimeUs);
   VCMTiming timing(&clock, field_trials);
-  timing.Reset();
   // MaxWaitingTime() returns zero for evenly spaced video frames.
   for (int i = 0; i < 10; ++i) {
     clock.AdvanceTime(kTimeDelta);
@@ -240,7 +237,6 @@ TEST(VCMTimingTest, DefaultMaxWaitingTimeUnaffectedByPacingExperiment) {
   const TimeDelta kTimeDelta = TimeDelta::Millis(1000.0 / 60.0);
   SimulatedClock clock(kStartTimeUs);
   VCMTiming timing(&clock, field_trials);
-  timing.Reset();
   clock.AdvanceTime(kTimeDelta);
   auto now = clock.CurrentTime();
   Timestamp render_time = now + TimeDelta::Millis(30);
@@ -274,7 +270,6 @@ TEST(VCMTimingTest, MaxWaitingTimeReturnsZeroIfTooManyFramesQueuedIsTrue) {
   constexpr auto kZeroRenderTime = Timestamp::Zero();
   SimulatedClock clock(kStartTimeUs);
   VCMTiming timing(&clock, field_trials);
-  timing.Reset();
   // MaxWaitingTime() returns zero for evenly spaced video frames.
   for (int i = 0; i < 10; ++i) {
     clock.AdvanceTime(kTimeDelta);
@@ -334,7 +329,6 @@ TEST(VCMTimingTest, UpdateCurrentDelayCapsWhenOffByMicroseconds) {
   FieldTrials field_trials = CreateTestFieldTrials();
   SimulatedClock clock(0);
   VCMTiming timing(&clock, field_trials);
-  timing.Reset();
 
   // Set larger initial current delay.
   timing.set_min_playout_delay(TimeDelta::Millis(200));
@@ -353,6 +347,23 @@ TEST(VCMTimingTest, UpdateCurrentDelayCapsWhenOffByMicroseconds) {
   // EXPECT_THAT(timing.GetTimings(), HasConsistentVideoDelayTimings());
 }
 
+TEST(VCMTimingTest, InitialVideoDelayTimings) {
+  FieldTrials field_trials = CreateTestFieldTrials();
+  SimulatedClock clock(0);
+  VCMTiming timing(&clock, field_trials);
+
+  VCMTiming::VideoDelayTimings timings = timing.GetTimings();
+  EXPECT_EQ(timings.num_decoded_frames, 0u);
+  EXPECT_EQ(timings.minimum_delay, TimeDelta::Zero());
+  EXPECT_EQ(timings.estimated_max_decode_time, TimeDelta::Zero());
+  EXPECT_EQ(timings.render_delay,
+            VCMTiming::VideoDelayTimings::kDefaultRenderDelay);
+  EXPECT_EQ(timings.min_playout_delay, TimeDelta::Zero());
+  EXPECT_EQ(timings.target_delay, TimeDelta::Zero());
+  EXPECT_EQ(timings.current_delay, TimeDelta::Zero());
+  EXPECT_THAT(timings, HasConsistentVideoDelayTimings());
+}
+
 TEST(VCMTimingTest, GetTimings) {
   FieldTrials field_trials = CreateTestFieldTrials();
   SimulatedClock clock(33);
@@ -397,8 +408,7 @@ TEST(VCMTimingTest, Reset) {
   SimulatedClock clock(Timestamp::Millis(33));
   VCMTiming timing(&clock, field_trials);
 
-  const TimeDelta default_render_delay = timing.GetTimings().render_delay;
-  timing.set_render_delay(default_render_delay + TimeDelta::Millis(1));
+  timing.set_render_delay(TimeDelta::Millis(11));
   TimeDelta min_playout_delay = TimeDelta::Millis(50);
   TimeDelta max_playout_delay = TimeDelta::Millis(500);
   timing.set_playout_delay({min_playout_delay, max_playout_delay});
@@ -420,7 +430,8 @@ TEST(VCMTimingTest, Reset) {
   EXPECT_GT(timings.num_decoded_frames, 0u);
   EXPECT_EQ(timings.minimum_delay, TimeDelta::Zero());
   EXPECT_EQ(timings.estimated_max_decode_time, TimeDelta::Zero());
-  EXPECT_EQ(timings.render_delay, default_render_delay);
+  EXPECT_EQ(timings.render_delay,
+            VCMTiming::VideoDelayTimings::kDefaultRenderDelay);
   EXPECT_EQ(timings.min_playout_delay, TimeDelta::Zero());
   EXPECT_EQ(timings.max_playout_delay, max_playout_delay);
   EXPECT_EQ(timings.target_delay, TimeDelta::Zero());

From 6815c67b00bf8b744172b20e4334d54dd2795942 Mon Sep 17 00:00:00 2001
From: Evan Shrubsole <eshr@webrtc.org>
Date: Tue, 31 Mar 2026 08:10:02 +0000
Subject: [PATCH 454/472] Use RunLoop in rtc_base/ tests

Bug: webrtc:469327588
Change-Id: Ib419c8d1d26c5db2107048ba4780d16d6a6a6964
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/461342
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47307}
---
 rtc_base/BUILD.gn                              | 12 ++++++++++++
 rtc_base/memory/BUILD.gn                       |  1 +
 rtc_base/memory/fifo_buffer_unittest.cc        |  6 +++---
 rtc_base/network_unittest.cc                   | 10 +++++-----
 rtc_base/openssl_adapter_unittest.cc           | 12 ++++++------
 rtc_base/operations_chain_unittest.cc          |  3 ++-
 rtc_base/physical_socket_server_unittest.cc    |  4 ++--
 rtc_base/rtc_certificate_generator_unittest.cc |  3 ++-
 rtc_base/ssl_adapter_unittest.cc               |  3 ++-
 9 files changed, 35 insertions(+), 19 deletions(-)

diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
index 7f9af04c8c..380090a9f7 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -759,6 +759,7 @@ if (rtc_include_tests) {
       "../api/task_queue",
       "../api/task_queue:task_queue_test",
       "../api/units:time_delta",
+      "../test:run_loop",
       "../test:test_support",
       "//third_party/abseil-cpp/absl/strings:string_view",
     ]
@@ -1128,6 +1129,7 @@ if (rtc_include_tests) {
     deps = [
       ":socket",
       ":socket_address",
+      "../test:run_loop",
       "../test:test_support",
     ]
   }
@@ -1416,6 +1418,7 @@ if (rtc_include_tests) {
       ":async_packet_socket",
       ":socket",
       ":socket_address",
+      "../test:run_loop",
       "../test:test_support",
       "network:received_packet",
     ]
@@ -1436,6 +1439,7 @@ if (rtc_include_tests) {
       "../api/units:timestamp",
       "../system_wrappers",
       "../test:create_test_environment",
+      "../test:run_loop",
       "../test:test_support",
       "network:received_packet",
       "//third_party/abseil-cpp/absl/memory",
@@ -1451,6 +1455,7 @@ if (rtc_include_tests) {
       ":rtc_base_tests_utils",
       ":socket",
       "../test:create_test_environment",
+      "../test:run_loop",
       "../test:test_support",
     ]
   }
@@ -1966,6 +1971,7 @@ if (rtc_include_tests) {
     sources = [ "untyped_function_unittest.cc" ]
     deps = [
       ":untyped_function",
+      "../test:run_loop",
       "../test:test_support",
     ]
   }
@@ -1982,6 +1988,7 @@ if (rtc_include_tests) {
       "../api:rtc_error_matchers",
       "../api:scoped_refptr",
       "../api/units:time_delta",
+      "../test:run_loop",
       "../test:test_support",
       "../test:wait_until",
     ]
@@ -2154,6 +2161,7 @@ if (rtc_include_tests) {
         "../system_wrappers",
         "../test:create_test_environment",
         "../test:fileutils",
+        "../test:run_loop",
         "../test:test_support",
         "../test/time_controller",
         "containers:flat_map",
@@ -2186,6 +2194,7 @@ if (rtc_include_tests) {
         ":timeutils",
         "../api/task_queue",
         "../api/units:time_delta",
+        "../test:run_loop",
         "../test:test_support",
         "//third_party/abseil-cpp/absl/memory",
       ]
@@ -2200,6 +2209,7 @@ if (rtc_include_tests) {
         ":rtc_event",
         ":task_queue_for_test",
         ":weak_ptr",
+        "../test:run_loop",
         "../test:test_support",
       ]
     }
@@ -2224,6 +2234,7 @@ if (rtc_include_tests) {
         ":timeutils",
         ":windowed_min_filter",
         "../api/units:time_delta",
+        "../test:run_loop",
         "../test:test_support",
         "//testing/gtest",
         "//third_party/abseil-cpp/absl/algorithm:container",
@@ -2237,6 +2248,7 @@ if (rtc_include_tests) {
       deps = [
         ":rtc_base_tests_utils",
         ":rtc_json",
+        "../test:run_loop",
         "../test:test_support",
       ]
     }
diff --git a/rtc_base/memory/BUILD.gn b/rtc_base/memory/BUILD.gn
index 60da01950a..2a0ec551a2 100644
--- a/rtc_base/memory/BUILD.gn
+++ b/rtc_base/memory/BUILD.gn
@@ -59,6 +59,7 @@ rtc_library("unittests") {
     ":fifo_buffer",
     "..:stream",
     "..:threading",
+    "../../test:run_loop",
     "../../test:test_support",
   ]
 }
diff --git a/rtc_base/memory/fifo_buffer_unittest.cc b/rtc_base/memory/fifo_buffer_unittest.cc
index e5f2070449..a5ebd880ba 100644
--- a/rtc_base/memory/fifo_buffer_unittest.cc
+++ b/rtc_base/memory/fifo_buffer_unittest.cc
@@ -15,13 +15,13 @@
 #include <span>
 
 #include "rtc_base/stream.h"
-#include "rtc_base/thread.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 
 namespace webrtc {
 
 TEST(FifoBufferTest, TestAll) {
-  AutoThread main_thread;
+  test::RunLoop main_thread;
   const size_t kSize = 16;
   const uint8_t in[kSize * 2 + 1] = "0123456789ABCDEFGHIJKLMNOPQRSTUV";
   uint8_t out[kSize * 2];
@@ -219,7 +219,7 @@ TEST(FifoBufferTest, TestAll) {
 }
 
 TEST(FifoBufferTest, FullBufferCheck) {
-  AutoThread main_thread;
+  test::RunLoop main_thread;
   FifoBuffer buff(10);
   buff.ConsumeWriteBuffer(10);
 
diff --git a/rtc_base/network_unittest.cc b/rtc_base/network_unittest.cc
index 380580f2b2..c93de941e8 100644
--- a/rtc_base/network_unittest.cc
+++ b/rtc_base/network_unittest.cc
@@ -38,10 +38,10 @@
 #include "rtc_base/network_monitor_factory.h"
 #include "rtc_base/physical_socket_server.h"
 #include "rtc_base/socket_address.h"
-#include "rtc_base/thread.h"
 #include "test/create_test_field_trials.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 #include "test/wait_until.h"
 
 // IWYU pragma: begin_keep
@@ -332,7 +332,7 @@ class NetworkTest : public ::testing::Test {
  protected:
   const FieldTrials field_trials_ = CreateTestFieldTrials();
   const Environment env_ = CreateEnvironment(&field_trials_);
-  AutoThread main_thread_;
+  test::RunLoop main_thread_;
   bool callback_called_ = false;
 };
 
@@ -430,14 +430,14 @@ TEST_F(NetworkTest, TestUpdateNetworks) {
   EXPECT_EQ(NetworkManager::ENUMERATION_ALLOWED,
             manager.enumeration_permission());
   manager.StartUpdating();
-  Thread::Current()->ProcessMessages(0);
+  main_thread_.Flush();
   EXPECT_TRUE(callback_called_);
   callback_called_ = false;
   // Callback should be triggered immediately when StartUpdating
   // is called, after network update signal is already sent.
   manager.StartUpdating();
   EXPECT_TRUE(manager.started());
-  Thread::Current()->ProcessMessages(0);
+  main_thread_.Flush();
   EXPECT_TRUE(callback_called_);
   manager.StopUpdating();
   EXPECT_TRUE(manager.started());
@@ -451,7 +451,7 @@ TEST_F(NetworkTest, TestUpdateNetworks) {
   // Callback should be triggered immediately after StartUpdating is called
   // when start_count_ is reset to 0.
   manager.StartUpdating();
-  Thread::Current()->ProcessMessages(0);
+  main_thread_.Flush();
   EXPECT_TRUE(callback_called_);
 }
 
diff --git a/rtc_base/openssl_adapter_unittest.cc b/rtc_base/openssl_adapter_unittest.cc
index 7f64cbfaa2..8d40fd8c6c 100644
--- a/rtc_base/openssl_adapter_unittest.cc
+++ b/rtc_base/openssl_adapter_unittest.cc
@@ -26,10 +26,10 @@
 #include "rtc_base/socket.h"
 #include "rtc_base/socket_address.h"
 #include "rtc_base/ssl_certificate.h"
-#include "rtc_base/ssl_stream_adapter.h"           // IWYU pragma: keep
-#include "rtc_base/thread.h"
+#include "rtc_base/ssl_stream_adapter.h"  // IWYU pragma: keep
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 #include "test/wait_until.h"  // IWYU pragma: keep
 
 namespace webrtc {
@@ -145,7 +145,7 @@ TEST(OpenSSLAdapterTest, TestTransformAlpnProtocols) {
 // Verifies that SSLStart works when OpenSSLAdapter is started in standalone
 // mode.
 TEST(OpenSSLAdapterTest, TestBeginSSLBeforeConnection) {
-  AutoThread main_thread;
+  test::RunLoop main_thread;
   Socket* async_socket = new MockAsyncSocket();
   OpenSSLAdapter adapter(async_socket);
   EXPECT_EQ(adapter.StartSSL("webrtc.org"), 0);
@@ -158,7 +158,7 @@ TEST(OpenSSLAdapterTest, TestBeginSSLBeforeConnection) {
 // build and run this test.
 TEST(OpenSSLAdaptorTest, TestRealSSLConnection) {
   PhysicalSocketServer socket_server;
-  AutoSocketServerThread main_thread(&socket_server);
+  test::RunLoop main_thread(&socket_server);
 
   constexpr absl::string_view kHostname = "webrtc.org";
   constexpr int kPort = 443;
@@ -213,7 +213,7 @@ TEST(OpenSSLAdaptorTest, TestRealSSLConnection) {
 
 // Verifies that the adapter factory can create new adapters.
 TEST(OpenSSLAdapterFactoryTest, CreateSingleOpenSSLAdapter) {
-  AutoThread main_thread;
+  test::RunLoop main_thread;
   OpenSSLAdapterFactory adapter_factory;
   Socket* async_socket = new MockAsyncSocket();
   auto simple_adapter = std::unique_ptr<OpenSSLAdapter>(
@@ -224,7 +224,7 @@ TEST(OpenSSLAdapterFactoryTest, CreateSingleOpenSSLAdapter) {
 // Verifies that setting a custom verifier still allows for adapters to be
 // created.
 TEST(OpenSSLAdapterFactoryTest, CreateWorksWithCustomVerifier) {
-  AutoThread main_thread;
+  test::RunLoop main_thread;
   MockCertVerifier* mock_verifier = new MockCertVerifier();
   EXPECT_CALL(*mock_verifier, Verify(_)).WillRepeatedly(Return(true));
   auto cert_verifier = std::unique_ptr<SSLCertificateVerifier>(mock_verifier);
diff --git a/rtc_base/operations_chain_unittest.cc b/rtc_base/operations_chain_unittest.cc
index 9f7ce22758..667b189069 100644
--- a/rtc_base/operations_chain_unittest.cc
+++ b/rtc_base/operations_chain_unittest.cc
@@ -25,6 +25,7 @@
 #include "rtc_base/thread.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 #include "test/wait_until.h"
 
 namespace webrtc {
@@ -397,7 +398,7 @@ TEST(OperationsChainTest, IsEmpty) {
 }
 
 TEST(OperationsChainTest, OnChainEmptyCallback) {
-  AutoThread main_thread;
+  test::RunLoop main_thread;
   OperationTrackerProxy operation_tracker_proxy;
   operation_tracker_proxy.Initialize()->Wait(Event::kForever);
 
diff --git a/rtc_base/physical_socket_server_unittest.cc b/rtc_base/physical_socket_server_unittest.cc
index 56ec60903a..909c80e63a 100644
--- a/rtc_base/physical_socket_server_unittest.cc
+++ b/rtc_base/physical_socket_server_unittest.cc
@@ -24,9 +24,9 @@
 #include "rtc_base/socket_address.h"
 #include "rtc_base/socket_unittest.h"
 #include "rtc_base/test_utils.h"
-#include "rtc_base/thread.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 #include "test/wait_until.h"
 
 #define MAYBE_SKIP_IPV4                        \
@@ -130,7 +130,7 @@ class PhysicalSocketTest : public SocketTest {
   void WritableAfterPartialWrite(const IPAddress& loopback);
 
   FakePhysicalSocketServer server_;
-  AutoSocketServerThread thread_;
+  test::RunLoop thread_;
   bool fail_accept_;
   int max_send_size_;
 };
diff --git a/rtc_base/rtc_certificate_generator_unittest.cc b/rtc_base/rtc_certificate_generator_unittest.cc
index 06bb5812e4..10b9b4b511 100644
--- a/rtc_base/rtc_certificate_generator_unittest.cc
+++ b/rtc_base/rtc_certificate_generator_unittest.cc
@@ -24,6 +24,7 @@
 #include "rtc_base/thread.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 #include "test/wait_until.h"
 
 namespace webrtc {
@@ -74,7 +75,7 @@ class RTCCertificateGeneratorTest : public ::testing::Test {
  protected:
   static constexpr TimeDelta kGenerationTimeoutMs = TimeDelta::Millis(10000);
 
-  AutoThread main_thread_;
+  test::RunLoop main_thread_;
   RTCCertificateGeneratorFixture fixture_;
 };
 
diff --git a/rtc_base/ssl_adapter_unittest.cc b/rtc_base/ssl_adapter_unittest.cc
index 79c987a89e..a4a069b829 100644
--- a/rtc_base/ssl_adapter_unittest.cc
+++ b/rtc_base/ssl_adapter_unittest.cc
@@ -30,6 +30,7 @@
 #include "rtc_base/virtual_socket_server.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 #include "test/wait_until.h"
 
 namespace webrtc {
@@ -291,7 +292,7 @@ class SSLAdapterTestBase : public ::testing::Test {
 
  protected:
   std::unique_ptr<VirtualSocketServer> vss_;
-  AutoSocketServerThread thread_;
+  test::RunLoop thread_;
   std::unique_ptr<SSLAdapterTestDummyServer> server_;
   std::unique_ptr<SSLAdapterTestDummyClient> client_;
   std::unique_ptr<SSLCertificateVerifier> cert_verifier_;

From 507cc3d8b900ef2c5bf70e6f8ad04f5b7ee55461 Mon Sep 17 00:00:00 2001
From: Evan Shrubsole <eshr@webrtc.org>
Date: Tue, 31 Mar 2026 08:40:34 +0000
Subject: [PATCH 455/472] Use RunLoop in video/ tests
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Bug: webrtc:469327588
Change-Id: I23eb639d3c0add2a40c1295418f41f2d6a6a6964
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/461340
Commit-Queue: Erik Språng <sprang@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47308}
---
 video/adaptation/BUILD.gn                            | 1 +
 video/adaptation/quality_scaler_resource_unittest.cc | 4 ++--
 2 files changed, 3 insertions(+), 2 deletions(-)

diff --git a/video/adaptation/BUILD.gn b/video/adaptation/BUILD.gn
index 04a00da3e0..9b5cfd753f 100644
--- a/video/adaptation/BUILD.gn
+++ b/video/adaptation/BUILD.gn
@@ -107,6 +107,7 @@ if (rtc_include_tests) {
       "../../system_wrappers",
       "../../test:create_test_environment",
       "../../test:create_test_field_trials",
+      "../../test:run_loop",
       "../../test:test_support",
       "../../test/time_controller",
       "../config:encoder_config",
diff --git a/video/adaptation/quality_scaler_resource_unittest.cc b/video/adaptation/quality_scaler_resource_unittest.cc
index 32dfa3bab3..a33ae1c95a 100644
--- a/video/adaptation/quality_scaler_resource_unittest.cc
+++ b/video/adaptation/quality_scaler_resource_unittest.cc
@@ -16,9 +16,9 @@
 #include "api/task_queue/task_queue_base.h"
 #include "call/adaptation/degradation_preference_provider.h"
 #include "call/adaptation/test/mock_resource_listener.h"
-#include "rtc_base/thread.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 
 namespace webrtc {
 
@@ -53,7 +53,7 @@ class QualityScalerResourceTest : public ::testing::Test {
   }
 
  protected:
-  AutoThread main_thread_;
+  test::RunLoop main_thread_;
   StrictMock<MockResourceListener> fake_resource_listener_;
   FakeDegradationPreferenceProvider degradation_preference_provider_;
   scoped_refptr<QualityScalerResource> quality_scaler_resource_;

From 6b3d4efe0dead10382c1f56b85c96f9c4e0c5b96 Mon Sep 17 00:00:00 2001
From: Evan Shrubsole <eshr@webrtc.org>
Date: Tue, 31 Mar 2026 08:11:47 +0000
Subject: [PATCH 456/472] Use RunLoop in media/ tests
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Bug: webrtc:469327588
Change-Id: Ife8aeee90aeabfb2a123a85f125358746a6a6964
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/461343
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47309}
---
 media/BUILD.gn                          |  1 +
 media/sctp/dcsctp_transport_unittest.cc | 19 ++++++++++---------
 2 files changed, 11 insertions(+), 9 deletions(-)

diff --git a/media/BUILD.gn b/media/BUILD.gn
index 15c2cf92dc..dcdb269995 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -1128,6 +1128,7 @@ if (rtc_include_tests) {
           "../net/dcsctp/public:factory",
           "../net/dcsctp/public:mocks",
           "../net/dcsctp/public:socket",
+          "../test:run_loop",
         ]
       }
     }
diff --git a/media/sctp/dcsctp_transport_unittest.cc b/media/sctp/dcsctp_transport_unittest.cc
index dffbc7a011..bfe797cb91 100644
--- a/media/sctp/dcsctp_transport_unittest.cc
+++ b/media/sctp/dcsctp_transport_unittest.cc
@@ -31,6 +31,7 @@
 #include "system_wrappers/include/clock.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 
 using ::testing::_;
 using ::testing::ByMove;
@@ -100,7 +101,7 @@ class Peer {
 }  // namespace
 
 TEST(DcSctpTransportTest, OpenSequence) {
-  AutoThread main_thread;
+  test::RunLoop main_thread;
   Peer peer_a;
   peer_a.fake_dtls_transport_.SetWritable(true);
 
@@ -118,7 +119,7 @@ TEST(DcSctpTransportTest, OpenSequence) {
 // Tests that the close sequence invoked from one end results in the stream to
 // be reset from both ends and all the proper signals are sent.
 TEST(DcSctpTransportTest, CloseSequence) {
-  AutoThread main_thread;
+  test::RunLoop main_thread;
   Peer peer_a;
   Peer peer_b;
   peer_a.fake_dtls_transport_.SetDestination(&peer_b.fake_dtls_transport_,
@@ -168,7 +169,7 @@ TEST(DcSctpTransportTest, CloseSequence) {
 // terminates properly. Both peers will think they initiated it, so no
 // OnClosingProcedureStartedRemotely should be called.
 TEST(DcSctpTransportTest, CloseSequenceSimultaneous) {
-  AutoThread main_thread;
+  test::RunLoop main_thread;
   Peer peer_a;
   Peer peer_b;
   peer_a.fake_dtls_transport_.SetDestination(&peer_b.fake_dtls_transport_,
@@ -212,7 +213,7 @@ TEST(DcSctpTransportTest, CloseSequenceSimultaneous) {
 }
 
 TEST(DcSctpTransportTest, SetStreamPriority) {
-  AutoThread main_thread;
+  test::RunLoop main_thread;
   Peer peer_a;
 
   {
@@ -236,7 +237,7 @@ TEST(DcSctpTransportTest, SetStreamPriority) {
 }
 
 TEST(DcSctpTransportTest, DiscardMessageClosedChannel) {
-  AutoThread main_thread;
+  test::RunLoop main_thread;
   Peer peer_a;
 
   EXPECT_CALL(*peer_a.socket_, Send(_, _)).Times(0);
@@ -252,7 +253,7 @@ TEST(DcSctpTransportTest, DiscardMessageClosedChannel) {
 }
 
 TEST(DcSctpTransportTest, DiscardMessageClosingChannel) {
-  AutoThread main_thread;
+  test::RunLoop main_thread;
   Peer peer_a;
 
   EXPECT_CALL(*peer_a.socket_, Send(_, _)).Times(0);
@@ -270,7 +271,7 @@ TEST(DcSctpTransportTest, DiscardMessageClosingChannel) {
 }
 
 TEST(DcSctpTransportTest, SendDataOpenChannel) {
-  AutoThread main_thread;
+  test::RunLoop main_thread;
   Peer peer_a;
   dcsctp::DcSctpOptions options;
 
@@ -288,7 +289,7 @@ TEST(DcSctpTransportTest, SendDataOpenChannel) {
 }
 
 TEST(DcSctpTransportTest, DeliversMessage) {
-  AutoThread main_thread;
+  test::RunLoop main_thread;
   Peer peer_a;
 
   EXPECT_CALL(peer_a.sink_, OnDataReceived(1, DataMessageType::kBinary, _))
@@ -305,7 +306,7 @@ TEST(DcSctpTransportTest, DeliversMessage) {
 }
 
 TEST(DcSctpTransportTest, DropMessageWithUnknownPpid) {
-  AutoThread main_thread;
+  test::RunLoop main_thread;
   Peer peer_a;
 
   EXPECT_CALL(peer_a.sink_, OnDataReceived(_, _, _)).Times(0);

From 4f399386d9ff6a5c14801a39162869c470a801f7 Mon Sep 17 00:00:00 2001
From: Evan Shrubsole <eshr@webrtc.org>
Date: Tue, 31 Mar 2026 12:20:34 +0000
Subject: [PATCH 457/472] Use real Thread in examples/ instead of AutoThread

Bug: webrtc:469327588
Change-Id: I99fa0af31513400270490ee6ee7a4d0a6a6a6964
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/461347
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47310}
---
 examples/BUILD.gn                            |  1 +
 examples/peerconnection/client/linux/main.cc |  7 +++++--
 examples/peerconnection/client/main.cc       |  5 ++++-
 examples/turnserver/turnserver_main.cc       | 11 +++++++----
 4 files changed, 17 insertions(+), 7 deletions(-)

diff --git a/examples/BUILD.gn b/examples/BUILD.gn
index dae37ec578..507b575764 100644
--- a/examples/BUILD.gn
+++ b/examples/BUILD.gn
@@ -820,6 +820,7 @@ if (is_linux || is_chromeos || is_win) {
       "../pc:rtc_pc",
       "../rtc_base:async_udp_socket",
       "../rtc_base:ip_address",
+      "../rtc_base:net_helper",
       "../rtc_base:socket_address",
       "../rtc_base:socket_server",
       "../rtc_base:threading",
diff --git a/examples/peerconnection/client/linux/main.cc b/examples/peerconnection/client/linux/main.cc
index 60b7b53a34..358d28ffe1 100644
--- a/examples/peerconnection/client/linux/main.cc
+++ b/examples/peerconnection/client/linux/main.cc
@@ -92,7 +92,9 @@ int main(int argc, char* argv[]) {
   wnd.Create();
 
   CustomSocketServer socket_server(&wnd);
-  webrtc::AutoSocketServerThread thread(&socket_server);
+  std::unique_ptr<webrtc::Thread> thread =
+      std::make_unique<webrtc::Thread>(&socket_server);
+  webrtc::ThreadManager::Instance()->SetCurrentThread(thread.get());
 
   webrtc::InitializeSSL();
   // Must be constructed after we set the socketserver.
@@ -101,7 +103,8 @@ int main(int argc, char* argv[]) {
   socket_server.set_client(&client);
   socket_server.set_conductor(conductor.get());
 
-  thread.Run();
+  thread->Run();
+  webrtc::ThreadManager::Instance()->SetCurrentThread(nullptr);
 
   // gtk_main();
   wnd.Destroy();
diff --git a/examples/peerconnection/client/main.cc b/examples/peerconnection/client/main.cc
index 9320fd9909..83876e4f39 100644
--- a/examples/peerconnection/client/main.cc
+++ b/examples/peerconnection/client/main.cc
@@ -84,7 +84,9 @@ int PASCAL wWinMain(HINSTANCE instance,
                     int cmd_show) {
   webrtc::WinsockInitializer winsock_init;
   webrtc::PhysicalSocketServer ss;
-  webrtc::AutoSocketServerThread main_thread(&ss);
+  std::unique_ptr<webrtc::Thread> main_thread =
+      std::make_unique<webrtc::Thread>(&ss);
+  webrtc::ThreadManager::Instance()->SetCurrentThread(main_thread.get());
 
   WindowsCommandLineArguments win_args;
   int argc = win_args.argc();
@@ -140,5 +142,6 @@ int PASCAL wWinMain(HINSTANCE instance,
   }
 
   webrtc::CleanupSSL();
+  webrtc::ThreadManager::Instance()->SetCurrentThread(nullptr);
   return 0;
 }
diff --git a/examples/turnserver/turnserver_main.cc b/examples/turnserver/turnserver_main.cc
index 3bbf406f4d..41d767254d 100644
--- a/examples/turnserver/turnserver_main.cc
+++ b/examples/turnserver/turnserver_main.cc
@@ -20,10 +20,10 @@
 #include "api/environment/environment_factory.h"
 #include "examples/turnserver/read_auth_file.h"
 #include "p2p/base/basic_packet_socket_factory.h"
-#include "p2p/base/port_interface.h"
 #include "p2p/test/turn_server.h"
 #include "rtc_base/async_udp_socket.h"
 #include "rtc_base/ip_address.h"
+#include "rtc_base/net_helper.h"
 #include "rtc_base/physical_socket_server.h"
 #include "rtc_base/socket_address.h"
 #include "rtc_base/thread.h"
@@ -75,7 +75,9 @@ int main(int argc, char* argv[]) {
 
   const webrtc::Environment env = webrtc::CreateEnvironment();
   webrtc::PhysicalSocketServer socket_server;
-  webrtc::AutoSocketServerThread main(&socket_server);
+  std::unique_ptr<webrtc::Thread> main =
+      std::make_unique<webrtc::Thread>(&socket_server);
+  webrtc::ThreadManager::Instance()->SetCurrentThread(main.get());
   std::unique_ptr<webrtc::AsyncUDPSocket> int_socket =
       webrtc::AsyncUDPSocket::Create(env, int_addr, socket_server);
   if (!int_socket) {
@@ -84,7 +86,7 @@ int main(int argc, char* argv[]) {
     return 1;
   }
 
-  webrtc::TurnServer server(env, &main);
+  webrtc::TurnServer server(env, main.get());
   std::fstream auth_file(argv[4], std::fstream::in);
 
   TurnFileAuth auth(auth_file.is_open()
@@ -100,6 +102,7 @@ int main(int argc, char* argv[]) {
 
   std::cout << "Listening internally at " << int_addr.ToString() << std::endl;
 
-  main.Run();
+  main->Run();
+  webrtc::ThreadManager::Instance()->SetCurrentThread(nullptr);
   return 0;
 }

From 06d113a66cef88e1d21673309ac0ad5ecea97f38 Mon Sep 17 00:00:00 2001
From: Evan Shrubsole <eshr@webrtc.org>
Date: Tue, 31 Mar 2026 08:09:45 +0000
Subject: [PATCH 458/472] Use RunLoop in p2p/ tests

Bug: webrtc:469327588
Change-Id: I4765d009202bba7bee45a0e944f797766a6a6964
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/461380
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47311}
---
 p2p/BUILD.gn                                       |  2 ++
 p2p/base/async_stun_tcp_socket_unittest.cc         |  4 ++--
 p2p/base/p2p_transport_channel_unittest.cc         |  7 ++++---
 p2p/base/port_allocator_unittest.cc                |  4 ++--
 p2p/base/tcp_port_unittest.cc                      | 10 +++++-----
 .../wrapping_active_ice_controller_unittest.cc     | 14 +++++++-------
 p2p/dtls/dtls_transport_unittest.cc                |  4 ++--
 p2p/test/nat_unittest.cc                           |  7 ++++---
 p2p/test/turn_server_unittest.cc                   |  6 +++---
 9 files changed, 31 insertions(+), 27 deletions(-)

diff --git a/p2p/BUILD.gn b/p2p/BUILD.gn
index 8ff23691b9..344d12756c 100644
--- a/p2p/BUILD.gn
+++ b/p2p/BUILD.gn
@@ -1075,6 +1075,7 @@ if (rtc_include_tests) {
       "../rtc_base/network:received_packet",
       "../rtc_base/network:sent_packet",
       "../rtc_base/synchronization:mutex",
+      "../test:run_loop",
       "../test:test_support",
       "//third_party/abseil-cpp/absl/base:core_headers",
       "//third_party/abseil-cpp/absl/memory",
@@ -1285,6 +1286,7 @@ if (rtc_include_tests) {
       "../rtc_base:ssl_adapter",
       "../rtc_base:threading",
       "../test:create_test_field_trials",
+      "../test:run_loop",
       "../test:test_support",
       "../test:wait_until",
       "//third_party/abseil-cpp/absl/algorithm:container",
diff --git a/p2p/base/async_stun_tcp_socket_unittest.cc b/p2p/base/async_stun_tcp_socket_unittest.cc
index 574561c062..7857cc37f9 100644
--- a/p2p/base/async_stun_tcp_socket_unittest.cc
+++ b/p2p/base/async_stun_tcp_socket_unittest.cc
@@ -26,11 +26,11 @@
 #include "rtc_base/network/sent_packet.h"
 #include "rtc_base/socket.h"
 #include "rtc_base/socket_address.h"
-#include "rtc_base/thread.h"
 #include "rtc_base/virtual_socket_server.h"
 #include "test/create_test_environment.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 
 namespace webrtc {
 
@@ -151,7 +151,7 @@ class AsyncStunTCPSocketTest : public ::testing::Test {
   }
 
   std::unique_ptr<VirtualSocketServer> vss_;
-  AutoSocketServerThread thread_;
+  test::RunLoop thread_;
   std::unique_ptr<AsyncStunTCPSocket> send_socket_;
   std::unique_ptr<AsyncListenSocket> listen_socket_;
   std::unique_ptr<AsyncPacketSocket> recv_socket_;
diff --git a/p2p/base/p2p_transport_channel_unittest.cc b/p2p/base/p2p_transport_channel_unittest.cc
index 97a9feb949..8b596e0b43 100644
--- a/p2p/base/p2p_transport_channel_unittest.cc
+++ b/p2p/base/p2p_transport_channel_unittest.cc
@@ -96,6 +96,7 @@
 #include "test/create_test_field_trials.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 #include "test/time_controller/simulated_time_controller.h"
 #include "test/wait_until.h"
 
@@ -5404,7 +5405,7 @@ TEST(P2PTransportChannelResolverTest, HostnameCandidateIsResolved) {
   const Environment env = CreateTestEnvironment();
   ResolverFactoryFixture resolver_fixture;
   std::unique_ptr<SocketServer> socket_server = CreateDefaultSocketServer();
-  AutoSocketServerThread main_thread(socket_server.get());
+  test::RunLoop main_thread(socket_server.get());
   FakePortAllocator allocator(env, socket_server.get());
   IceTransportInit init(env);
   init.set_port_allocator(&allocator);
@@ -6378,7 +6379,7 @@ TEST_F(P2PTransportChannelPingTest, TestInitialSelectDampeningBoth) {
 TEST(P2PTransportChannelIceControllerTest, InjectIceController) {
   const Environment env = CreateTestEnvironment();
   std::unique_ptr<SocketServer> socket_server = CreateDefaultSocketServer();
-  AutoSocketServerThread main_thread(socket_server.get());
+  test::RunLoop main_thread(socket_server.get());
   MockIceControllerFactory factory;
   FakePortAllocator pa(env, socket_server.get());
   EXPECT_CALL(factory, RecordIceControllerCreated()).Times(1);
@@ -6393,7 +6394,7 @@ TEST(P2PTransportChannelIceControllerTest, InjectIceController) {
 TEST(P2PTransportChannel, InjectActiveIceController) {
   const Environment env = CreateTestEnvironment();
   std::unique_ptr<SocketServer> socket_server = CreateDefaultSocketServer();
-  AutoSocketServerThread main_thread(socket_server.get());
+  test::RunLoop main_thread(socket_server.get());
   MockActiveIceControllerFactory factory;
   FakePortAllocator pa(env, socket_server.get());
   EXPECT_CALL(factory, RecordActiveIceControllerCreated()).Times(1);
diff --git a/p2p/base/port_allocator_unittest.cc b/p2p/base/port_allocator_unittest.cc
index 9e68874f52..10f9d2a478 100644
--- a/p2p/base/port_allocator_unittest.cc
+++ b/p2p/base/port_allocator_unittest.cc
@@ -22,9 +22,9 @@
 #include "rtc_base/ip_address.h"
 #include "rtc_base/net_helper.h"
 #include "rtc_base/socket_address.h"
-#include "rtc_base/thread.h"
 #include "rtc_base/virtual_socket_server.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 
 using ::webrtc::CreateEnvironment;
 using ::webrtc::IceCandidateType;
@@ -98,7 +98,7 @@ class PortAllocatorTest : public ::testing::Test {
   }
 
   std::unique_ptr<webrtc::VirtualSocketServer> vss_;
-  webrtc::AutoSocketServerThread main_;
+  webrtc::test::RunLoop main_;
   std::unique_ptr<webrtc::FakePortAllocator> allocator_;
   webrtc::SocketAddress stun_server_1{"11.11.11.11", 3478};
   webrtc::SocketAddress stun_server_2{"22.22.22.22", 3478};
diff --git a/p2p/base/tcp_port_unittest.cc b/p2p/base/tcp_port_unittest.cc
index 6d50e41ff3..97de11c021 100644
--- a/p2p/base/tcp_port_unittest.cc
+++ b/p2p/base/tcp_port_unittest.cc
@@ -34,10 +34,10 @@
 #include "rtc_base/network/sent_packet.h"
 #include "rtc_base/socket.h"
 #include "rtc_base/socket_address.h"
-#include "rtc_base/thread.h"
 #include "rtc_base/virtual_socket_server.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 #include "test/wait_until.h"
 
 using ::testing::Eq;
@@ -105,7 +105,7 @@ class TCPPortTest : public ::testing::Test {
                                          int port_number = 0) {
     auto port = std::unique_ptr<TCPPort>(
         TCPPort::Create({.env = env_,
-                         .network_thread = &main_,
+                         .network_thread = main_.task_queue(),
                          .socket_factory = &socket_factory_,
                          .network = MakeNetwork(addr),
                          .ice_username_fragment = username_,
@@ -118,7 +118,7 @@ class TCPPortTest : public ::testing::Test {
   std::unique_ptr<TCPPort> CreateTCPPort(const webrtc::Network* network) {
     auto port = std::unique_ptr<TCPPort>(
         TCPPort::Create({.env = env_,
-                         .network_thread = &main_,
+                         .network_thread = main_.task_queue(),
                          .socket_factory = &socket_factory_,
                          .network = network,
                          .ice_username_fragment = username_,
@@ -135,7 +135,7 @@ class TCPPortTest : public ::testing::Test {
   // vector so that when it grows, pointers aren't invalidated.
   std::list<webrtc::Network> networks_;
   std::unique_ptr<webrtc::VirtualSocketServer> ss_;
-  webrtc::AutoSocketServerThread main_;
+  webrtc::test::RunLoop main_;
   webrtc::BasicPacketSocketFactory socket_factory_;
   std::string username_;
   std::string password_;
@@ -431,7 +431,7 @@ TEST_F(TCPPortTest, SignalSentPacketAfterReconnect) {
                         {.timeout = webrtc::TimeDelta::Millis(kTimeout)}),
       webrtc::IsRtcOk());
   // Wait a bit for the Stun response to be received.
-  webrtc::Thread::Current()->ProcessMessages(100);
+  main_.RunFor(webrtc::TimeDelta::Millis(100));
 
   // After the Stun Ping response has been received, packets can be sent again
   // and SignalSentPacket should be invoked.
diff --git a/p2p/base/wrapping_active_ice_controller_unittest.cc b/p2p/base/wrapping_active_ice_controller_unittest.cc
index 459af04f4f..41c605631b 100644
--- a/p2p/base/wrapping_active_ice_controller_unittest.cc
+++ b/p2p/base/wrapping_active_ice_controller_unittest.cc
@@ -26,10 +26,10 @@
 #include "p2p/test/mock_ice_controller.h"
 #include "rtc_base/event.h"
 #include "rtc_base/fake_clock.h"
-#include "rtc_base/thread.h"
 #include "test/create_test_environment.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 
 namespace webrtc {
 namespace {
@@ -55,7 +55,7 @@ const std::vector<const Connection*> kEmptyConnsList =
 constexpr TimeDelta kTick = TimeDelta::Millis(1);
 
 TEST(WrappingActiveIceControllerTest, CreateLegacyIceControllerFromFactory) {
-  AutoThread main;
+  test::RunLoop main;
   MockIceAgent agent;
   IceControllerFactoryArgs args = {.env = CreateTestEnvironment()};
   MockIceControllerFactory legacy_controller_factory;
@@ -65,7 +65,7 @@ TEST(WrappingActiveIceControllerTest, CreateLegacyIceControllerFromFactory) {
 }
 
 TEST(WrappingActiveIceControllerTest, PassthroughIceControllerInterface) {
-  AutoThread main;
+  test::RunLoop main;
   MockIceAgent agent;
   auto will_move = std::make_unique<MockIceController>();
   MockIceController* wrapped = will_move.get();
@@ -100,7 +100,7 @@ TEST(WrappingActiveIceControllerTest, PassthroughIceControllerInterface) {
 }
 
 TEST(WrappingActiveIceControllerTest, HandlesImmediateSwitchRequest) {
-  AutoThread main;
+  test::RunLoop main;
   ScopedFakeClock clock;
   NiceMock<MockIceAgent> agent;
   auto will_move = std::make_unique<NiceMockIceController>();
@@ -144,7 +144,7 @@ TEST(WrappingActiveIceControllerTest, HandlesImmediateSwitchRequest) {
 }
 
 TEST(WrappingActiveIceControllerTest, HandlesImmediateSortAndSwitchRequest) {
-  AutoThread main;
+  test::RunLoop main;
   ScopedFakeClock clock;
   NiceMock<MockIceAgent> agent;
   auto will_move = std::make_unique<NiceMockIceController>();
@@ -195,7 +195,7 @@ TEST(WrappingActiveIceControllerTest, HandlesImmediateSortAndSwitchRequest) {
 }
 
 TEST(WrappingActiveIceControllerTest, HandlesSortAndSwitchRequest) {
-  AutoThread main;
+  test::RunLoop main;
   ScopedFakeClock clock;
 
   // Block the main task queue until ready.
@@ -239,7 +239,7 @@ TEST(WrappingActiveIceControllerTest, HandlesSortAndSwitchRequest) {
 }
 
 TEST(WrappingActiveIceControllerTest, StartPingingAfterSortAndSwitch) {
-  AutoThread main;
+  test::RunLoop main;
   ScopedFakeClock clock;
 
   // Block the main task queue until ready.
diff --git a/p2p/dtls/dtls_transport_unittest.cc b/p2p/dtls/dtls_transport_unittest.cc
index 98b0ba5eee..3d3dd6a9ab 100644
--- a/p2p/dtls/dtls_transport_unittest.cc
+++ b/p2p/dtls/dtls_transport_unittest.cc
@@ -57,11 +57,11 @@
 #include "rtc_base/ssl_identity.h"
 #include "rtc_base/ssl_stream_adapter.h"
 #include "rtc_base/stream.h"
-#include "rtc_base/thread.h"
 #include "system_wrappers/include/metrics.h"
 #include "test/create_test_environment.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 #include "test/wait_until.h"
 
 namespace webrtc {
@@ -728,7 +728,7 @@ class DtlsTransportInternalImplTestBase {
   }
 
  protected:
-  AutoThread main_thread_;
+  test::RunLoop main_thread_;
   ScopedFakeClock fake_clock_;
   DtlsTestClient client1_;
   DtlsTestClient client2_;
diff --git a/p2p/test/nat_unittest.cc b/p2p/test/nat_unittest.cc
index f50ea7328c..502e0f2653 100644
--- a/p2p/test/nat_unittest.cc
+++ b/p2p/test/nat_unittest.cc
@@ -36,6 +36,7 @@
 #include "rtc_base/virtual_socket_server.h"
 #include "test/create_test_environment.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 #include "test/wait_until.h"
 
 namespace webrtc {
@@ -231,13 +232,13 @@ bool TestConnectivity(const SocketAddress& src, const IPAddress& dst) {
 }
 
 void TestPhysicalInternal(const SocketAddress& int_addr) {
-  AutoThread main_thread;
+  test::RunLoop main_thread;
   PhysicalSocketServer socket_server;
   const Environment env = CreateTestEnvironment();
   BasicNetworkManager network_manager(env, &socket_server);
   network_manager.StartUpdating();
   // Process pending messages so the network list is updated.
-  Thread::Current()->ProcessMessages(0);
+  main_thread.Flush();
 
   std::vector<const Network*> networks = network_manager.GetNetworks();
   std::erase_if(networks, [](const Network* network) {
@@ -301,7 +302,7 @@ class TestVirtualSocketServer : public VirtualSocketServer {
 }  // namespace
 
 void TestVirtualInternal(int family) {
-  AutoThread main_thread;
+  test::RunLoop main_thread;
   const Environment env = CreateTestEnvironment();
   std::unique_ptr<TestVirtualSocketServer> int_vss(
       new TestVirtualSocketServer());
diff --git a/p2p/test/turn_server_unittest.cc b/p2p/test/turn_server_unittest.cc
index 2b63b13933..455e1da4d8 100644
--- a/p2p/test/turn_server_unittest.cc
+++ b/p2p/test/turn_server_unittest.cc
@@ -14,13 +14,13 @@
 
 #include "api/environment/environment.h"
 #include "p2p/base/basic_packet_socket_factory.h"
-#include "p2p/base/port_interface.h"
 #include "rtc_base/async_packet_socket.h"
+#include "rtc_base/net_helper.h"
 #include "rtc_base/socket_address.h"
-#include "rtc_base/thread.h"
 #include "rtc_base/virtual_socket_server.h"
 #include "test/create_test_environment.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 
 // NOTE: This is a work in progress. Currently this file only has tests for
 // TurnServerConnection, a primitive class used by TurnServer.
@@ -48,7 +48,7 @@ class TurnServerConnectionTest : public ::testing::Test {
 
  protected:
   VirtualSocketServer vss_;
-  AutoSocketServerThread thread_;
+  test::RunLoop thread_;
   BasicPacketSocketFactory socket_factory_;
 };
 

From 2dab14a4fce46b9f3fdcdeba5ffd71311180d005 Mon Sep 17 00:00:00 2001
From: Evan Shrubsole <eshr@webrtc.org>
Date: Tue, 31 Mar 2026 08:16:39 +0000
Subject: [PATCH 459/472] Use RunLoop in call/ tests

Bug: webrtc:469327588
Change-Id: Ia88b913a96175f8926208f7fd6bb4d1d6a6a6964
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/461345
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47312}
---
 call/BUILD.gn                                        |  5 +++++
 call/adaptation/BUILD.gn                             |  1 +
 .../resource_adaptation_processor_unittest.cc        |  8 +++-----
 call/flexfec_receive_stream_unittest.cc              |  4 ++--
 call/rtp_transport_controller_send_unittest.cc       | 12 ++++++------
 5 files changed, 17 insertions(+), 13 deletions(-)

diff --git a/call/BUILD.gn b/call/BUILD.gn
index a8ad708290..815d28a81b 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -660,6 +660,7 @@ if (rtc_include_tests) {
         "../test:encoder_settings",
         "../test:fake_video_codecs",
         "../test:frame_generator_capturer",
+        "../test:run_loop",
         "../test:test_common",
         "../test:test_flags",
         "../test:test_support",
@@ -699,6 +700,7 @@ if (rtc_include_tests) {
       "../rtc_base:network_route",
       "../rtc_base/containers:flat_map",
       "../rtc_base/network:sent_packet",
+      "../test:run_loop",
       "../test:test_support",
       "//third_party/abseil-cpp/absl/strings:string_view",
     ]
@@ -709,6 +711,7 @@ if (rtc_include_tests) {
     sources = [ "test/mock_bitrate_allocator.h" ]
     deps = [
       ":bitrate_allocator",
+      "../test:run_loop",
       "../test:test_support",
     ]
   }
@@ -730,6 +733,7 @@ if (rtc_include_tests) {
       "../api/audio_codecs:audio_codecs_api",
       "../api/crypto:frame_decryptor_interface",
       "../api/transport/rtp:rtp_source",
+      "../test:run_loop",
       "../test:test_support",
     ]
   }
@@ -760,6 +764,7 @@ if (rtc_include_tests) {
       "../rtc_base:checks",
       "../rtc_base:copy_on_write_buffer",
       "../system_wrappers",
+      "../test:run_loop",
       "../test:test_support",
       "../test/network:simulated_network",
     ]
diff --git a/call/adaptation/BUILD.gn b/call/adaptation/BUILD.gn
index 3149c136b3..0b0da3477d 100644
--- a/call/adaptation/BUILD.gn
+++ b/call/adaptation/BUILD.gn
@@ -89,6 +89,7 @@ if (rtc_include_tests) {
       "../../rtc_base:task_queue_for_test",
       "../../rtc_base:threading",
       "../../test:create_test_field_trials",
+      "../../test:run_loop",
       "../../test:test_support",
       "../../test:wait_until",
       "../../video/config:encoder_config",
diff --git a/call/adaptation/resource_adaptation_processor_unittest.cc b/call/adaptation/resource_adaptation_processor_unittest.cc
index 4f6396082b..413f6fe016 100644
--- a/call/adaptation/resource_adaptation_processor_unittest.cc
+++ b/call/adaptation/resource_adaptation_processor_unittest.cc
@@ -27,11 +27,11 @@
 #include "call/adaptation/video_stream_input_state_provider.h"
 #include "rtc_base/event.h"
 #include "rtc_base/task_queue_for_test.h"
-#include "rtc_base/thread.h"
 #include "rtc_base/thread_annotations.h"
 #include "test/create_test_field_trials.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 #include "test/wait_until.h"
 
 namespace webrtc {
@@ -141,12 +141,10 @@ class ResourceAdaptationProcessorTest : public ::testing::Test {
     processor_.reset();
   }
 
-  static void WaitUntilTaskQueueIdle() {
-    ASSERT_TRUE(Thread::Current()->ProcessMessages(0));
-  }
+  void WaitUntilTaskQueueIdle() { main_thread_.Flush(); }
 
  protected:
-  AutoThread main_thread_;
+  test::RunLoop main_thread_;
   FakeFrameRateProvider frame_rate_provider_;
   VideoStreamInputStateProvider input_state_provider_;
   scoped_refptr<FakeResource> resource_;
diff --git a/call/flexfec_receive_stream_unittest.cc b/call/flexfec_receive_stream_unittest.cc
index 27c8d2c285..c4df175cdf 100644
--- a/call/flexfec_receive_stream_unittest.cc
+++ b/call/flexfec_receive_stream_unittest.cc
@@ -25,10 +25,10 @@
 #include "modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h"
 #include "modules/rtp_rtcp/source/byte_io.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
-#include "rtc_base/thread.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
 #include "test/mock_transport.h"
+#include "test/run_loop.h"
 
 namespace webrtc {
 
@@ -94,7 +94,7 @@ class FlexfecReceiveStreamTest : public ::testing::Test {
     receive_stream_->UnregisterFromTransport();
   }
 
-  AutoThread main_thread_;
+  test::RunLoop main_thread_;
   MockTransport rtcp_send_transport_;
   MockRtcEventLog log_;
   FlexfecReceiveStream::Config config_;
diff --git a/call/rtp_transport_controller_send_unittest.cc b/call/rtp_transport_controller_send_unittest.cc
index 3e07614c36..5bf29ea7d6 100644
--- a/call/rtp_transport_controller_send_unittest.cc
+++ b/call/rtp_transport_controller_send_unittest.cc
@@ -23,10 +23,10 @@
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/rtcp_packet/congestion_control_feedback.h"
 #include "rtc_base/containers/flat_map.h"
-#include "rtc_base/thread.h"
 #include "test/create_test_environment.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 
 namespace webrtc {
 namespace {
@@ -102,7 +102,7 @@ rtcp::CongestionControlFeedback GenerateFeedback(
 
 TEST(RtpTransportControllerSendTest,
      IgnoresFeedbackForReportedReceivedPacketThatWereNotSent) {
-  AutoThread main_thread;
+  test::RunLoop main_thread;
   RtpTransportControllerSend transport({.env = CreateTestEnvironment()});
   transport.SetPreferredRtcpCcAckType(RtcpFeedbackType::CCFB);
   PacketSender sender(transport);
@@ -131,7 +131,7 @@ TEST(RtpTransportControllerSendTest,
      AccumulatesNumberOfReportedReceivedPacketsPerSsrcPerEcnMarkingType) {
   constexpr uint32_t kSsrc1 = 1'000;
   constexpr uint32_t kSsrc2 = 2'000;
-  AutoThread main_thread;
+  test::RunLoop main_thread;
   RtpTransportControllerSend transport({.env = CreateTestEnvironment()});
   transport.SetPreferredRtcpCcAckType(RtcpFeedbackType::CCFB);
 
@@ -176,7 +176,7 @@ TEST(RtpTransportControllerSendTest,
 }
 
 TEST(RtpTransportControllerSendTest, CalculatesNumberOfBleachedPackets) {
-  AutoThread main_thread;
+  test::RunLoop main_thread;
   RtpTransportControllerSend transport({.env = CreateTestEnvironment()});
   transport.SetPreferredRtcpCcAckType(RtcpFeedbackType::CCFB);
   PacketSender sender(transport);
@@ -211,7 +211,7 @@ TEST(RtpTransportControllerSendTest, CalculatesNumberOfBleachedPackets) {
 
 TEST(RtpTransportControllerSendTest,
      AccumulatesNumberOfReportedLostAndRecoveredPackets) {
-  AutoThread main_thread;
+  test::RunLoop main_thread;
   RtpTransportControllerSend transport({.env = CreateTestEnvironment()});
   transport.SetPreferredRtcpCcAckType(RtcpFeedbackType::CCFB);
 
@@ -258,7 +258,7 @@ TEST(RtpTransportControllerSendTest,
 
 TEST(RtpTransportControllerSendTest,
      DoesNotCountGapsInSequenceNumberBetweenReportsAsLoss) {
-  AutoThread main_thread;
+  test::RunLoop main_thread;
   RtpTransportControllerSend transport({.env = CreateTestEnvironment()});
   transport.SetPreferredRtcpCcAckType(RtcpFeedbackType::CCFB);
 

From f2b4180c16602c4ee23c0df1e921c5624b400222 Mon Sep 17 00:00:00 2001
From: Per K <perkj@webrtc.org>
Date: Wed, 1 Apr 2026 14:07:24 +0000
Subject: [PATCH 460/472] Allow EventLogAnalyzer to be created without a valid
 parsed log

The purpose is to be able to list all possible plots without opening an eventlog.

Bug: None
Change-Id: I947114d59b5ec393357748c0a0f78d95edd63663
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/461421
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47313}
---
 .../rtc_event_log_visualizer/analyzer.cc      | 26 ++++++++++++-------
 1 file changed, 16 insertions(+), 10 deletions(-)

diff --git a/rtc_tools/rtc_event_log_visualizer/analyzer.cc b/rtc_tools/rtc_event_log_visualizer/analyzer.cc
index a44d6832c7..5821ad7bd4 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyzer.cc
+++ b/rtc_tools/rtc_event_log_visualizer/analyzer.cc
@@ -47,22 +47,28 @@ EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& parsed_log,
   }
   neteq_simulator_ = std::make_unique<LazyNetEqSimulator>(parsed_log_, config_);
 
-  RTC_LOG(LS_INFO) << "Log is "
-                   << (parsed_log_.last_timestamp().ms() -
-                       parsed_log_.first_timestamp().ms()) /
-                          1000
-                   << " seconds long.";
+  if (parsed_log_.first_timestamp().IsFinite() &&
+      parsed_log_.last_timestamp().IsFinite()) {
+    RTC_LOG(LS_INFO) << "Log is "
+                     << (parsed_log_.last_timestamp().ms() -
+                         parsed_log_.first_timestamp().ms()) /
+                            1000
+                     << " seconds long.";
+  }
 }
 
 EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& parsed_log,
                                    const AnalyzerConfig& config)
     : parsed_log_(parsed_log), config_(config) {
   neteq_simulator_ = std::make_unique<LazyNetEqSimulator>(parsed_log_, config_);
-  RTC_LOG(LS_INFO) << "Log is "
-                   << (parsed_log_.last_timestamp().ms() -
-                       parsed_log_.first_timestamp().ms()) /
-                          1000
-                   << " seconds long.";
+  if (parsed_log_.first_timestamp().IsFinite() &&
+      parsed_log_.last_timestamp().IsFinite()) {
+    RTC_LOG(LS_INFO) << "Log is "
+                     << (parsed_log_.last_timestamp().ms() -
+                         parsed_log_.first_timestamp().ms()) /
+                            1000
+                     << " seconds long.";
+  }
 }
 
 EventLogAnalyzer::~EventLogAnalyzer() = default;

From c6b486dd18e59e51a207d51ca8b4f8a11921a146 Mon Sep 17 00:00:00 2001
From: webrtc-version-updater
 <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Date: Wed, 1 Apr 2026 21:08:12 -0700
Subject: [PATCH 461/472] Update WebRTC code version (2026-04-02T04:08:10).

Bug: None
Change-Id: I9e5b318d6e65d0ff27d044b7ff1de24925cfb965
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/461460
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#47314}
---
 call/version.cc | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/call/version.cc b/call/version.cc
index a4b2de53d2..1d4ee7360f 100644
--- a/call/version.cc
+++ b/call/version.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2026-03-31T04:10:44";
+const char* const kSourceTimestamp = "WebRTC source stamp 2026-04-02T04:08:10";
 
 void LoadWebRTCVersionInRegister() {
   // Using volatile to instruct the compiler to not optimize `p` away even

From 52e8522cb7fdc00ab72b337dd778bb79a37b25f0 Mon Sep 17 00:00:00 2001
From: Danil Chapovalov <danilchap@webrtc.org>
Date: Mon, 30 Mar 2026 17:17:59 +0200
Subject: [PATCH 462/472] Replace ArrayView with std::span in api/

ArrayView is an alias to std::span. This change switch to use
std::span directly instead of through the alias.

Search&Replace MakeArrayView and ArrayView with std::span
Search&Replace include "api/array_view.h" with include <span>
Remove <span> include where std::span is not mentioned in the file
Remove build dependencies on array_view target

Updated comment in audio_frame.cc to be more accurate matching one of
the behavior changes between ArrayView and std::span

Bug: webrtc:439801349
Change-Id: Iec42e1bf3ff5539211e8a4c85773ba08cf1412dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/461060
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47315}
---
 api/BUILD.gn                                  | 25 +-----------
 api/audio/BUILD.gn                            |  7 +---
 api/audio/audio_frame.cc                      | 16 ++++----
 api/audio/audio_frame.h                       |  4 +-
 api/audio/audio_processing.h                  |  8 ++--
 api/audio/audio_view.h                        | 14 +++----
 api/audio/neural_residual_echo_estimator.h    | 16 ++++----
 api/audio/test/BUILD.gn                       |  1 -
 api/audio/test/audio_frame_unittest.cc        |  2 +-
 api/audio/test/audio_view_unittest.cc         | 10 ++---
 api/audio_codecs/BUILD.gn                     |  1 -
 api/audio_codecs/audio_decoder.cc             |  8 ++--
 api/audio_codecs/audio_decoder.h              |  4 +-
 api/audio_codecs/audio_encoder.cc             |  6 +--
 api/audio_codecs/audio_encoder.h              |  8 ++--
 api/audio_options.cc                          |  1 -
 api/call/transport.h                          |  6 +--
 api/crypto/BUILD.gn                           |  2 -
 api/crypto/frame_decryptor_interface.h        |  8 ++--
 api/crypto/frame_encryptor_interface.h        |  8 ++--
 api/data_channel_event_observer_interface.h   |  4 +-
 api/datagram_connection.h                     |  8 ++--
 api/frame_transformer_interface.h             |  8 ++--
 api/neteq/BUILD.gn                            |  1 -
 api/neteq/neteq.h                             |  8 ++--
 api/numerics/BUILD.gn                         |  1 -
 api/numerics/samples_stats_counter.h          |  4 +-
 api/rtp_transceiver_interface.h               |  6 +--
 api/test/fake_frame_decryptor.cc              |  8 ++--
 api/test/fake_frame_decryptor.h               |  8 ++--
 api/test/fake_frame_encryptor.cc              |  8 ++--
 api/test/fake_frame_encryptor.h               |  9 +++--
 api/test/metrics/BUILD.gn                     | 10 +----
 .../chrome_perf_dashboard_metrics_exporter.cc |  4 +-
 .../chrome_perf_dashboard_metrics_exporter.h  |  4 +-
 ...global_metrics_logger_and_exporter_test.cc |  4 +-
 api/test/metrics/metrics_exporter.h           |  5 ++-
 .../metrics_set_proto_file_exporter.cc        |  4 +-
 .../metrics/metrics_set_proto_file_exporter.h |  4 +-
 .../print_result_proxy_metrics_exporter.cc    |  4 +-
 .../print_result_proxy_metrics_exporter.h     |  5 ++-
 api/test/metrics/stdout_metrics_exporter.cc   |  4 +-
 api/test/metrics/stdout_metrics_exporter.h    |  4 +-
 api/test/mock_datagram_connection.h           |  8 ++--
 api/test/mock_datagram_connection_observer.h  |  4 +-
 api/test/mock_frame_decryptor.h               |  8 ++--
 api/test/mock_frame_encryptor.h               |  8 ++--
 api/test/mock_rtp_transceiver.h               |  6 +--
 api/test/mock_transformable_audio_frame.h     |  8 ++--
 api/test/mock_transformable_frame.h           |  6 +--
 api/test/mock_transformable_video_frame.h     |  6 +--
 api/test/network_emulation_manager.h          |  6 +--
 api/test/pclf/BUILD.gn                        |  1 -
 api/test/pclf/media_configuration.cc          |  6 +--
 api/test/pclf/media_configuration.h           |  6 +--
 api/test/video_quality_analyzer_interface.h   |  4 +-
 api/transport/BUILD.gn                        |  3 --
 api/transport/rtp/BUILD.gn                    |  1 -
 .../rtp/corruption_detection_message.cc       |  4 +-
 .../rtp/corruption_detection_message.h        |  8 ++--
 api/transport/stun.cc                         | 38 +++++++++----------
 api/transport/stun.h                          |  8 ++--
 api/transport/stun_unittest.cc                | 12 +++---
 api/units/BUILD.gn                            |  5 ---
 api/units/data_rate.cc                        |  1 -
 api/units/data_size.cc                        |  1 -
 api/units/time_delta.cc                       |  1 -
 api/units/timestamp.cc                        |  1 -
 api/video/BUILD.gn                            |  8 ----
 .../frame_instrumentation_data.cc             |  5 +--
 .../frame_instrumentation_data.h              |  7 ++--
 .../frame_instrumentation_data_reader.cc      |  4 +-
 .../frame_instrumentation_evaluation.cc       |  6 +--
 api/video/frame_buffer.cc                     |  4 +-
 api/video/rtp_video_frame_assembler.cc        |  4 +-
 .../rtp_video_frame_assembler_unittests.cc    | 10 ++---
 ...tp_video_frame_h265_assembler_unittests.cc | 10 ++---
 api/video/video_frame_buffer.cc               |  4 +-
 api/video/video_frame_buffer.h                |  4 +-
 api/video/video_frame_metadata.cc             | 10 ++---
 api/video/video_frame_metadata.h              | 10 ++---
 api/video/video_timing.cc                     |  1 -
 api/video_codecs/BUILD.gn                     |  8 +---
 api/video_codecs/bitstream_parser.h           |  5 +--
 .../libaom_av1_encoder_factory_test.cc        | 10 ++---
 api/video_codecs/sdp_video_format.cc          |  6 +--
 api/video_codecs/sdp_video_format.h           |  6 +--
 api/video_codecs/simple_encoder_wrapper.cc    |  4 +-
 .../video_decoder_factory_template.h          |  4 +-
 .../video_encoder_factory_template.h          |  4 +-
 api/video_codecs/video_encoder_interface.h    |  4 +-
 api/voip/BUILD.gn                             |  2 -
 api/voip/test/mock_voip_engine.h              |  6 +--
 api/voip/voip_network.h                       |  6 +--
 api/webrtc_sdp_unittest.cc                    |  1 -
 95 files changed, 258 insertions(+), 335 deletions(-)

diff --git a/api/BUILD.gn b/api/BUILD.gn
index 3555ec0202..df24e793e4 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -523,7 +523,6 @@ rtc_library("rtp_transceiver_interface") {
     "rtp_transceiver_interface.h",
   ]
   deps = [
-    ":array_view",
     ":ref_count",
     ":rtc_error",
     ":rtp_parameters",
@@ -594,7 +593,6 @@ rtc_library("libjingle_peerconnection_api") {
     "video_track_source_proxy_factory.h",  # Used downstream
   ]
   deps = [
-    ":array_view",
     ":async_dns_resolver",
     ":audio_options_api",
     ":candidate",
@@ -684,7 +682,6 @@ rtc_library("frame_transformer_interface") {
     "frame_transformer_interface.h",
   ]
   deps = [
-    ":array_view",
     ":ref_count",
     ":scoped_refptr",
     "../rtc_base:checks",
@@ -784,7 +781,6 @@ rtc_source_set("video_quality_analyzer_api") {
   sources = [ "test/video_quality_analyzer_interface.h" ]
 
   deps = [
-    ":array_view",
     ":rtc_stats_api",
     ":scoped_refptr",
     ":stats_observer_interface",
@@ -838,7 +834,6 @@ rtc_library("rtp_parameters") {
     "rtp_parameters.h",
   ]
   deps = [
-    ":array_view",
     ":priority",
     ":rtc_error",
     ":rtp_transceiver_direction",
@@ -1040,10 +1035,7 @@ rtc_library("create_peer_connection_quality_test_frame_generator") {
 rtc_source_set("data_channel_event_observer_interface") {
   visibility = [ "*" ]
   sources = [ "data_channel_event_observer_interface.h" ]
-  deps = [
-    ":array_view",
-    "//third_party/abseil-cpp/absl/strings:string_view",
-  ]
+  deps = [ "//third_party/abseil-cpp/absl/strings:string_view" ]
 }
 
 rtc_source_set("libjingle_logging_api") {
@@ -1098,7 +1090,6 @@ rtc_library("audio_options_api") {
   ]
 
   deps = [
-    ":array_view",
     "../rtc_base:stringutils",
     "../rtc_base/system:rtc_export",
   ]
@@ -1110,7 +1101,7 @@ rtc_library("transport_api") {
     "call/transport.cc",
     "call/transport.h",
   ]
-  deps = [ ":array_view" ]
+  deps = []
 }
 
 rtc_source_set("bitrate_allocation") {
@@ -1143,7 +1134,6 @@ rtc_library("network_emulation_manager_api") {
     "test/network_emulation_manager.h",
   ]
   deps = [
-    ":array_view",
     ":field_trials_view",
     ":peer_network_dependencies",
     ":simulated_network_api",
@@ -1262,7 +1252,6 @@ rtc_library("datagram_connection") {
   visibility = [ "*" ]
   sources = [ "datagram_connection.h" ]
   deps = [
-    ":array_view",
     ":candidate",
     ":ref_count",
     "../p2p:transport_description",
@@ -1459,7 +1448,6 @@ if (rtc_include_tests) {
     testonly = true
     sources = [ "test/mock_frame_encryptor.h" ]
     deps = [
-      ":array_view",
       ":rtp_parameters",
       "../test:test_support",
       "crypto:frame_encryptor_interface",
@@ -1471,7 +1459,6 @@ if (rtc_include_tests) {
     testonly = true
     sources = [ "test/mock_frame_decryptor.h" ]
     deps = [
-      ":array_view",
       ":rtp_parameters",
       "../test:test_support",
       "crypto:frame_decryptor_interface",
@@ -1509,7 +1496,6 @@ if (rtc_include_tests) {
       "test/fake_frame_encryptor.h",
     ]
     deps = [
-      ":array_view",
       ":rtp_parameters",
       "../rtc_base:checks",
       "../rtc_base:refcount",
@@ -1525,7 +1511,6 @@ if (rtc_include_tests) {
       "test/fake_frame_decryptor.h",
     ]
     deps = [
-      ":array_view",
       ":rtp_parameters",
       "../rtc_base:checks",
       "crypto:frame_decryptor_interface",
@@ -1616,7 +1601,6 @@ if (rtc_include_tests) {
     testonly = true
     sources = [ "test/mock_transformable_frame.h" ]
     deps = [
-      ":array_view",
       ":frame_transformer_interface",
       "../test:test_support",
       "units:time_delta",
@@ -1662,7 +1646,6 @@ if (rtc_include_tests) {
     ]
 
     deps = [
-      ":array_view",
       ":dtls_transport_interface",
       ":dtmf_sender_interface",
       ":frame_transformer_interface",
@@ -1690,7 +1673,6 @@ if (rtc_include_tests) {
     sources = [ "test/mock_transformable_audio_frame.h" ]
 
     deps = [
-      ":array_view",
       ":frame_transformer_interface",
       "../test:test_support",
       "units:time_delta",
@@ -1704,7 +1686,6 @@ if (rtc_include_tests) {
     sources = [ "test/mock_transformable_video_frame.h" ]
 
     deps = [
-      ":array_view",
       ":frame_transformer_interface",
       "../test:test_support",
       "units:time_delta",
@@ -1801,7 +1782,6 @@ if (rtc_include_tests) {
     testonly = true
     sources = [ "test/mock_datagram_connection.h" ]
     deps = [
-      ":array_view",
       ":candidate",
       ":datagram_connection",
       "../p2p:transport_description",
@@ -1816,7 +1796,6 @@ if (rtc_include_tests) {
     testonly = true
     sources = [ "test/mock_datagram_connection_observer.h" ]
     deps = [
-      ":array_view",
       ":candidate",
       ":datagram_connection",
       "//test:test_support",
diff --git a/api/audio/BUILD.gn b/api/audio/BUILD.gn
index db356e6b56..578930e2eb 100644
--- a/api/audio/BUILD.gn
+++ b/api/audio/BUILD.gn
@@ -48,7 +48,6 @@ rtc_library("audio_frame_api") {
   ]
 
   deps = [
-    "..:array_view",
     "..:rtp_packet_info",
     "../../rtc_base:checks",
     "../../rtc_base:logging",
@@ -84,7 +83,6 @@ rtc_library("audio_processing") {
     ":aec3_config",
     ":audio_processing_statistics",
     ":echo_control",
-    "..:array_view",
     "..:ref_count",
     "..:scoped_refptr",
     "../../rtc_base:checks",
@@ -176,10 +174,7 @@ rtc_source_set("echo_control") {
 rtc_source_set("neural_residual_echo_estimator_api") {
   visibility = [ "*" ]
   sources = [ "neural_residual_echo_estimator.h" ]
-  deps = [
-    ":aec3_config",
-    "..:array_view",
-  ]
+  deps = [ ":aec3_config" ]
 }
 
 if (rtc_enable_protobuf) {
diff --git a/api/audio/audio_frame.cc b/api/audio/audio_frame.cc
index b894673ebf..14afaded89 100644
--- a/api/audio/audio_frame.cc
+++ b/api/audio/audio_frame.cc
@@ -13,9 +13,9 @@
 #include <cstdint>
 #include <cstring>
 #include <optional>
+#include <span>
 
 #include "absl/algorithm/container.h"
-#include "api/array_view.h"
 #include "api/audio/audio_view.h"
 #include "api/audio/channel_layout.h"
 #include "api/rtp_packet_infos.h"
@@ -133,12 +133,10 @@ const int16_t* AudioFrame::data() const {
 InterleavedView<const int16_t> AudioFrame::data_view() const {
   // If you get a nullptr from `data_view()`, it's likely because the
   // samples_per_channel_ and/or num_channels_ members haven't been properly
-  // set. Since `data_view()` returns an InterleavedView<> (which internally
-  // uses ArrayView<>), we inherit the behavior in InterleavedView when
-  // the view size is 0 that ArrayView<>::data() returns nullptr. So, even when
-  // an AudioFrame is muted and we want to return `zeroed_data()`, if
-  // samples_per_channel_ or  num_channels_ is 0, the view will point to
-  // nullptr.
+  // set. `data_view()` returns an InterleavedView<> which internally
+  // uses std::span<>. So, even when an AudioFrame is muted and we want to
+  // return `zeroed_data()`, if samples_per_channel_ or  num_channels_ is 0,
+  // the view might point to nullptr.
   return InterleavedView<const int16_t>(muted_ ? &zeroed_data()[0] : &data_[0],
                                         samples_per_channel_, num_channels_);
 }
@@ -213,9 +211,9 @@ void AudioFrame::SetSampleRateAndChannelSize(int sample_rate) {
 }
 
 // static
-ArrayView<const int16_t> AudioFrame::zeroed_data() {
+std::span<const int16_t> AudioFrame::zeroed_data() {
   static int16_t* null_data = new int16_t[kMaxDataSizeSamples]();
-  return ArrayView<const int16_t>(null_data, kMaxDataSizeSamples);
+  return std::span<const int16_t>(null_data, kMaxDataSizeSamples);
 }
 
 }  // namespace webrtc
diff --git a/api/audio/audio_frame.h b/api/audio/audio_frame.h
index c4c93031e6..095db192a4 100644
--- a/api/audio/audio_frame.h
+++ b/api/audio/audio_frame.h
@@ -16,8 +16,8 @@
 
 #include <array>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/audio/audio_view.h"
 #include "api/audio/channel_layout.h"
 #include "api/rtp_packet_infos.h"
@@ -199,7 +199,7 @@ class AudioFrame {
   // A permanently zeroed out buffer to represent muted frames. This is a
   // header-only class, so the only way to avoid creating a separate zeroed
   // buffer per translation unit is to wrap a static in an inline function.
-  static ArrayView<const int16_t> zeroed_data();
+  static std::span<const int16_t> zeroed_data();
 
   std::array<int16_t, kMaxDataSizeSamples> data_;
   bool muted_ = true;
diff --git a/api/audio/audio_processing.h b/api/audio/audio_processing.h
index 1e0ea0beee..a4f3f52fe1 100644
--- a/api/audio/audio_processing.h
+++ b/api/audio/audio_processing.h
@@ -18,11 +18,11 @@
 #include <cstring>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 
 #include "absl/base/nullability.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio/audio_processing_statistics.h"
 #include "api/audio/echo_control.h"
 #include "api/environment/environment.h"
@@ -582,7 +582,7 @@ class RTC_EXPORT AudioProcessing : public RefCountInterface {
   // representation of the input is returned. Returns true/false to indicate
   // whether an output returned.
   virtual bool GetLinearAecOutput(
-      ArrayView<std::array<float, 160>> linear_output) const = 0;
+      std::span<std::array<float, 160>> linear_output) const = 0;
 
   // This must be called prior to ProcessStream() if and only if adaptive analog
   // gain control is enabled, to pass the current analog level from the audio
@@ -864,10 +864,10 @@ class EchoDetector : public RefCountInterface {
                           int num_render_channels) = 0;
 
   // Analysis (not changing) of the first channel of the render signal.
-  virtual void AnalyzeRenderAudio(ArrayView<const float> render_audio) = 0;
+  virtual void AnalyzeRenderAudio(std::span<const float> render_audio) = 0;
 
   // Analysis (not changing) of the capture signal.
-  virtual void AnalyzeCaptureAudio(ArrayView<const float> capture_audio) = 0;
+  virtual void AnalyzeCaptureAudio(std::span<const float> capture_audio) = 0;
 
   struct Metrics {
     std::optional<double> echo_likelihood;
diff --git a/api/audio/audio_view.h b/api/audio/audio_view.h
index f3fcf8d918..9f7291b9d8 100644
--- a/api/audio/audio_view.h
+++ b/api/audio/audio_view.h
@@ -13,11 +13,11 @@
 
 #include <cstddef>
 #include <iterator>
+#include <span>
 #include <variant>
 #include <vector>
 
 #include "absl/algorithm/container.h"
-#include "api/array_view.h"
 #include "rtc_base/checks.h"
 
 namespace webrtc {
@@ -34,7 +34,7 @@ namespace webrtc {
 //   buffer. Channels can be enumerated and accessing the individual channel
 //   data is done via MonoView<>.
 //
-// The views are comparable to and built on ArrayView<> but add
+// The views are comparable to and built on std::span<> but add
 // audio specific properties for the dimensions of the buffer and the above
 // specialized [de]interleaved support.
 //
@@ -45,7 +45,7 @@ namespace webrtc {
 // can be either an single channel (mono) interleaved buffer (e.g. AudioFrame),
 // or a de-interleaved channel (e.g. from AudioBuffer).
 template <typename T>
-using MonoView = ArrayView<T>;
+using MonoView = std::span<T>;
 
 // The maximum number of audio channels supported by WebRTC encoders, decoders
 // and the AudioFrame class.
@@ -59,8 +59,8 @@ template <typename T>
 class InterleavedView {
  public:
   using value_type = T;
-  using iterator = typename ArrayView<T>::iterator;
-  using const_iterator = typename ArrayView<const T>::iterator;
+  using iterator = typename std::span<T>::iterator;
+  using const_iterator = typename std::span<const T>::iterator;
 
   InterleavedView() = default;
 
@@ -90,7 +90,7 @@ class InterleavedView {
 
   size_t num_channels() const { return num_channels_; }
   size_t samples_per_channel() const { return samples_per_channel_; }
-  ArrayView<T> data() const { return data_; }
+  std::span<T> data() const { return data_; }
   bool empty() const { return data_.empty(); }
   size_t size() const { return data_.size(); }
 
@@ -131,7 +131,7 @@ class InterleavedView {
   // construction.
   size_t num_channels_ = 0u;
   size_t samples_per_channel_ = 0u;
-  ArrayView<T> data_;
+  std::span<T> data_;
 };
 
 template <typename T>
diff --git a/api/audio/neural_residual_echo_estimator.h b/api/audio/neural_residual_echo_estimator.h
index 87e814906e..974bc67263 100644
--- a/api/audio/neural_residual_echo_estimator.h
+++ b/api/audio/neural_residual_echo_estimator.h
@@ -12,8 +12,8 @@
 #define API_AUDIO_NEURAL_RESIDUAL_ECHO_ESTIMATOR_H_
 
 #include <array>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/audio/echo_canceller3_config.h"
 
 namespace webrtc {
@@ -45,14 +45,14 @@ class NeuralResidualEchoEstimator {
   // Other inputs:
   //   * dominant_nearend: True if dominant nearend is active
   virtual void Estimate(const Block& render,
-                        ArrayView<const std::array<float, 64>> y,
-                        ArrayView<const std::array<float, 64>> e,
-                        ArrayView<const std::array<float, 65>> S2,
-                        ArrayView<const std::array<float, 65>> Y2,
-                        ArrayView<const std::array<float, 65>> E2,
+                        std::span<const std::array<float, 64>> y,
+                        std::span<const std::array<float, 64>> e,
+                        std::span<const std::array<float, 65>> S2,
+                        std::span<const std::array<float, 65>> Y2,
+                        std::span<const std::array<float, 65>> E2,
                         bool dominant_nearend,
-                        ArrayView<std::array<float, 65>> R2,
-                        ArrayView<std::array<float, 65>> R2_unbounded) = 0;
+                        std::span<std::array<float, 65>> R2,
+                        std::span<std::array<float, 65>> R2_unbounded) = 0;
 
   // Returns a recommended AEC3 configuration for this estimator.
   virtual EchoCanceller3Config GetConfiguration(bool multi_channel) const = 0;
diff --git a/api/audio/test/BUILD.gn b/api/audio/test/BUILD.gn
index b1060faae3..edd3dcd701 100644
--- a/api/audio/test/BUILD.gn
+++ b/api/audio/test/BUILD.gn
@@ -23,7 +23,6 @@ if (rtc_include_tests) {
     deps = [
       "..:aec3_config",
       "..:audio_frame_api",
-      "../..:array_view",
       "../../../modules/audio_processing:aec3_config_json",
       "../../../rtc_base:checks",
       "../../../test:test_support",
diff --git a/api/audio/test/audio_frame_unittest.cc b/api/audio/test/audio_frame_unittest.cc
index a7c0166330..e9a1e6fd81 100644
--- a/api/audio/test/audio_frame_unittest.cc
+++ b/api/audio/test/audio_frame_unittest.cc
@@ -66,7 +66,7 @@ TEST(AudioFrameTest, FrameStartsZeroedAndMuted) {
   EXPECT_TRUE(AllSamplesAre(0, frame));
 }
 
-// TODO: b/335805780 - Delete test when `mutable_data()` returns ArrayView.
+// TODO: b/335805780 - Delete test when `mutable_data()` returns std::span.
 TEST(AudioFrameTest, UnmutedFrameIsInitiallyZeroedLegacy) {
   AudioFrame frame(kSampleRateHz, kNumChannelsMono, CHANNEL_LAYOUT_NONE);
   frame.mutable_data();
diff --git a/api/audio/test/audio_view_unittest.cc b/api/audio/test/audio_view_unittest.cc
index 4ac914f1bc..833f936fe7 100644
--- a/api/audio/test/audio_view_unittest.cc
+++ b/api/audio/test/audio_view_unittest.cc
@@ -13,9 +13,9 @@
 #include <array>
 #include <cstddef>
 #include <cstdint>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "test/gtest.h"
 
 namespace webrtc {
@@ -37,7 +37,7 @@ void Increment(int16_t& t) {
 
 // Fills a given buffer with monotonically increasing values.
 template <typename T>
-void FillBuffer(ArrayView<T> buffer) {
+void FillBuffer(std::span<T> buffer) {
   T value = {};
   for (T& t : buffer) {
     Increment<T>(value);
@@ -50,7 +50,7 @@ void FillBuffer(ArrayView<T> buffer) {
 TEST(AudioViewTest, MonoView) {
   const size_t kArraySize = 100u;
   int16_t arr[kArraySize];
-  FillBuffer(ArrayView<int16_t>(arr));
+  FillBuffer(std::span<int16_t>(arr));
 
   MonoView<int16_t> mono(arr);
   MonoView<const int16_t> const_mono(arr);
@@ -69,7 +69,7 @@ TEST(AudioViewTest, MonoView) {
 TEST(AudioViewTest, InterleavedView) {
   const size_t kArraySize = 100u;
   int16_t arr[kArraySize];
-  FillBuffer(ArrayView<int16_t>(arr));
+  FillBuffer(std::span<int16_t>(arr));
 
   InterleavedView<int16_t> interleaved(arr, kArraySize, 1);
   EXPECT_EQ(NumChannels(interleaved), 1u);
@@ -144,7 +144,7 @@ TEST(AudioViewTest, CopySamples) {
   const size_t kArraySize = 100u;
   int16_t source_arr[kArraySize] = {};
   int16_t dest_arr[kArraySize] = {};
-  FillBuffer(ArrayView<int16_t>(source_arr));
+  FillBuffer(std::span<int16_t>(source_arr));
 
   InterleavedView<const int16_t> source(source_arr, 2);
   InterleavedView<int16_t> destination(dest_arr, 2);
diff --git a/api/audio_codecs/BUILD.gn b/api/audio_codecs/BUILD.gn
index 25ea12557b..0d5b919e8a 100644
--- a/api/audio_codecs/BUILD.gn
+++ b/api/audio_codecs/BUILD.gn
@@ -29,7 +29,6 @@ rtc_library("audio_codecs_api") {
     "audio_format.h",
   ]
   deps = [
-    "..:array_view",
     "..:bitrate_allocation",
     "..:make_ref_counted",
     "..:ref_count",
diff --git a/api/audio_codecs/audio_decoder.cc b/api/audio_codecs/audio_decoder.cc
index ecc1decfa5..dd2920e3fb 100644
--- a/api/audio_codecs/audio_decoder.cc
+++ b/api/audio_codecs/audio_decoder.cc
@@ -14,10 +14,10 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/sanitizer.h"
@@ -41,7 +41,7 @@ class OldStyleEncodedFrame final : public AudioDecoder::EncodedAudioFrame {
   }
 
   std::optional<DecodeResult> Decode(
-      ArrayView<int16_t> decoded) const override {
+      std::span<int16_t> decoded) const override {
     auto speech_type = AudioDecoder::kSpeech;
     const int ret = decoder_->Decode(
         payload_.data(), payload_.size(), decoder_->SampleRateHz(),
@@ -94,7 +94,7 @@ int AudioDecoder::Decode(const uint8_t* encoded,
                          int16_t* decoded,
                          SpeechType* speech_type) {
   TRACE_EVENT0("webrtc", "AudioDecoder::Decode");
-  MsanCheckInitialized(MakeArrayView(encoded, encoded_len));
+  MsanCheckInitialized(std::span(encoded, encoded_len));
   int duration = PacketDuration(encoded, encoded_len);
   if (duration >= 0 &&
       duration * Channels() * sizeof(int16_t) > max_decoded_bytes) {
@@ -111,7 +111,7 @@ int AudioDecoder::DecodeRedundant(const uint8_t* encoded,
                                   int16_t* decoded,
                                   SpeechType* speech_type) {
   TRACE_EVENT0("webrtc", "AudioDecoder::DecodeRedundant");
-  MsanCheckInitialized(MakeArrayView(encoded, encoded_len));
+  MsanCheckInitialized(std::span(encoded, encoded_len));
   int duration = PacketDurationRedundant(encoded, encoded_len);
   if (duration >= 0 &&
       duration * Channels() * sizeof(int16_t) > max_decoded_bytes) {
diff --git a/api/audio_codecs/audio_decoder.h b/api/audio_codecs/audio_decoder.h
index a085d3b059..e15ffc90c3 100644
--- a/api/audio_codecs/audio_decoder.h
+++ b/api/audio_codecs/audio_decoder.h
@@ -16,9 +16,9 @@
 
 #include <memory>
 #include <optional>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio/audio_view.h"
 #include "rtc_base/buffer.h"
 
@@ -63,7 +63,7 @@ class AudioDecoder {
     // decoder produced comfort noise or speech. On failure, returns an empty
     // std::optional. Decode may be called at most once per frame object.
     virtual std::optional<DecodeResult> Decode(
-        ArrayView<int16_t> decoded) const = 0;
+        std::span<int16_t> decoded) const = 0;
   };
 
   struct ParseResult {
diff --git a/api/audio_codecs/audio_encoder.cc b/api/audio_codecs/audio_encoder.cc
index cb86adcb35..3fef371f2b 100644
--- a/api/audio_codecs/audio_encoder.cc
+++ b/api/audio_codecs/audio_encoder.cc
@@ -14,9 +14,9 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/call/bitrate_allocation.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/checks.h"
@@ -45,7 +45,7 @@ int AudioEncoder::RtpTimestampRateHz() const {
 }
 
 AudioEncoder::EncodedInfo AudioEncoder::Encode(uint32_t rtp_timestamp,
-                                               ArrayView<const int16_t> audio,
+                                               std::span<const int16_t> audio,
                                                Buffer* encoded) {
   TRACE_EVENT0("webrtc", "AudioEncoder::Encode");
   RTC_CHECK_EQ(audio.size(),
@@ -77,7 +77,7 @@ void AudioEncoder::SetMaxPlaybackRate(int /* frequency_hz */) {}
 
 void AudioEncoder::SetTargetBitrate(int /* target_bps */) {}
 
-ArrayView<std::unique_ptr<AudioEncoder>>
+std::span<std::unique_ptr<AudioEncoder>>
 AudioEncoder::ReclaimContainedEncoders() {
   return {};
 }
diff --git a/api/audio_codecs/audio_encoder.h b/api/audio_codecs/audio_encoder.h
index 8c4fe33b11..d82a074c29 100644
--- a/api/audio_codecs/audio_encoder.h
+++ b/api/audio_codecs/audio_encoder.h
@@ -16,12 +16,12 @@
 
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
 #include "absl/base/attributes.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio/audio_view.h"
 #include "api/call/bitrate_allocation.h"
 #include "api/units/data_rate.h"
@@ -150,7 +150,7 @@ class AudioEncoder {
   // EncodeImpl() which does the actual work, and then checks some
   // postconditions.
   EncodedInfo Encode(uint32_t rtp_timestamp,
-                     ArrayView<const int16_t> audio,
+                     std::span<const int16_t> audio,
                      Buffer* encoded);
 
   // Resets the encoder to its starting state, discarding any input that has
@@ -196,7 +196,7 @@ class AudioEncoder {
   // not call any methods on this encoder afterwards, except for the
   // destructor. The default implementation just returns an empty array.
   // NOTE: This method is subject to change. Do not call or override it.
-  virtual ArrayView<std::unique_ptr<AudioEncoder>> ReclaimContainedEncoders();
+  virtual std::span<std::unique_ptr<AudioEncoder>> ReclaimContainedEncoders();
 
   // Enables audio network adaptor. Returns true if successful.
   virtual bool EnableAudioNetworkAdaptor(absl::string_view config);
@@ -260,7 +260,7 @@ class AudioEncoder {
   // Subclasses implement this to perform the actual encoding. Called by
   // Encode().
   virtual EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
-                                 ArrayView<const int16_t> audio,
+                                 std::span<const int16_t> audio,
                                  Buffer* encoded) = 0;
 };
 }  // namespace webrtc
diff --git a/api/audio_options.cc b/api/audio_options.cc
index 46913b8a4c..83429206d8 100644
--- a/api/audio_options.cc
+++ b/api/audio_options.cc
@@ -13,7 +13,6 @@
 #include <optional>
 #include <string>
 
-#include "api/array_view.h"
 #include "rtc_base/strings/string_builder.h"
 
 namespace webrtc {
diff --git a/api/call/transport.h b/api/call/transport.h
index 74d7addf6c..aade6a5db5 100644
--- a/api/call/transport.h
+++ b/api/call/transport.h
@@ -13,7 +13,7 @@
 
 #include <stdint.h>
 
-#include "api/array_view.h"
+#include <span>
 
 namespace webrtc {
 
@@ -41,9 +41,9 @@ struct PacketOptions {
 
 class Transport {
  public:
-  virtual bool SendRtp(ArrayView<const uint8_t> packet,
+  virtual bool SendRtp(std::span<const uint8_t> packet,
                        const PacketOptions& options) = 0;
-  virtual bool SendRtcp(ArrayView<const uint8_t> packet,
+  virtual bool SendRtcp(std::span<const uint8_t> packet,
                         const PacketOptions& options) = 0;
 
  protected:
diff --git a/api/crypto/BUILD.gn b/api/crypto/BUILD.gn
index 00a9debba1..e89016a653 100644
--- a/api/crypto/BUILD.gn
+++ b/api/crypto/BUILD.gn
@@ -35,7 +35,6 @@ rtc_source_set("frame_decryptor_interface") {
   visibility = [ "*" ]
   sources = [ "frame_decryptor_interface.h" ]
   deps = [
-    "..:array_view",
     "..:ref_count",
     "..:rtp_parameters",
     "../../rtc_base:refcount",
@@ -46,7 +45,6 @@ rtc_source_set("frame_encryptor_interface") {
   visibility = [ "*" ]
   sources = [ "frame_encryptor_interface.h" ]
   deps = [
-    "..:array_view",
     "..:ref_count",
     "..:rtp_parameters",
     "../../rtc_base:refcount",
diff --git a/api/crypto/frame_decryptor_interface.h b/api/crypto/frame_decryptor_interface.h
index 659f16654c..c0b8801473 100644
--- a/api/crypto/frame_decryptor_interface.h
+++ b/api/crypto/frame_decryptor_interface.h
@@ -13,9 +13,9 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/media_types.h"
 #include "api/ref_count.h"
 
@@ -62,9 +62,9 @@ class FrameDecryptorInterface : public RefCountInterface {
   // cases.
   virtual Result Decrypt(MediaType media_type,
                          const std::vector<uint32_t>& csrcs,
-                         ArrayView<const uint8_t> additional_data,
-                         ArrayView<const uint8_t> encrypted_frame,
-                         ArrayView<uint8_t> frame) = 0;
+                         std::span<const uint8_t> additional_data,
+                         std::span<const uint8_t> encrypted_frame,
+                         std::span<uint8_t> frame) = 0;
 
   // Returns the total required length in bytes for the output of the
   // decryption. This can be larger than the actual number of bytes you need but
diff --git a/api/crypto/frame_encryptor_interface.h b/api/crypto/frame_encryptor_interface.h
index 6a1797249d..4360cd5bbc 100644
--- a/api/crypto/frame_encryptor_interface.h
+++ b/api/crypto/frame_encryptor_interface.h
@@ -13,8 +13,8 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/media_types.h"
 #include "api/ref_count.h"
 
@@ -40,9 +40,9 @@ class FrameEncryptorInterface : public RefCountInterface {
   // selected by the implementer to represent error codes.
   virtual int Encrypt(MediaType media_type,
                       uint32_t ssrc,
-                      ArrayView<const uint8_t> additional_data,
-                      ArrayView<const uint8_t> frame,
-                      ArrayView<uint8_t> encrypted_frame,
+                      std::span<const uint8_t> additional_data,
+                      std::span<const uint8_t> frame,
+                      std::span<uint8_t> encrypted_frame,
                       size_t* bytes_written) = 0;
 
   // Returns the total required length in bytes for the output of the
diff --git a/api/data_channel_event_observer_interface.h b/api/data_channel_event_observer_interface.h
index f9b59020f4..135db62ea6 100644
--- a/api/data_channel_event_observer_interface.h
+++ b/api/data_channel_event_observer_interface.h
@@ -12,11 +12,11 @@
 #define API_DATA_CHANNEL_EVENT_OBSERVER_INTERFACE_H_
 
 #include <cstdint>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 
 namespace webrtc {
 
@@ -55,7 +55,7 @@ class DataChannelEventObserverInterface {
     void set_data_type(DataType type) { data_type_ = type; }
 
     const std::vector<uint8_t>& data() const { return data_; }
-    void set_data(ArrayView<const uint8_t> d) {
+    void set_data(std::span<const uint8_t> d) {
       data_.assign(d.begin(), d.end());
     }
 
diff --git a/api/datagram_connection.h b/api/datagram_connection.h
index 9b52e97ec7..e18588b52c 100644
--- a/api/datagram_connection.h
+++ b/api/datagram_connection.h
@@ -12,11 +12,11 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 #include <string_view>
 
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/candidate.h"
 #include "api/ref_count.h"
 #include "api/units/timestamp.h"
@@ -48,7 +48,7 @@ class RTC_EXPORT DatagramConnection : public RefCountInterface {
     struct PacketMetadata {
       Timestamp receive_time;
     };
-    virtual void OnPacketReceived(ArrayView<const uint8_t> data,
+    virtual void OnPacketReceived(std::span<const uint8_t> data,
                                   PacketMetadata metadata) = 0;
 
     // Notification of outcome of an earlier call to SendPacket.
@@ -96,13 +96,13 @@ class RTC_EXPORT DatagramConnection : public RefCountInterface {
     // performed, the caller is responsible for ensuring uniqueness and handing
     // rollovers.
     PacketId id = 0;
-    ArrayView<const uint8_t> payload;
+    std::span<const uint8_t> payload;
   };
 
   // Send a batch of packets on this connection. Listen to
   // Observer::OnSendOutcome for notification of whether each was sent
   // successfully.
-  virtual void SendPackets(ArrayView<PacketSendParameters> packets) = 0;
+  virtual void SendPackets(std::span<PacketSendParameters> packets) = 0;
 
   // Initiate closing connection and releasing resources. Must be called before
   // destruction.
diff --git a/api/frame_transformer_interface.h b/api/frame_transformer_interface.h
index 42ac2c1266..9f4ed927f2 100644
--- a/api/frame_transformer_interface.h
+++ b/api/frame_transformer_interface.h
@@ -14,9 +14,9 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 
-#include "api/array_view.h"
 #include "api/ref_count.h"
 #include "api/scoped_refptr.h"
 #include "api/units/time_delta.h"
@@ -46,10 +46,10 @@ class TransformableFrameInterface {
 
   // Returns the frame payload data. The data is valid until the next non-const
   // method call.
-  virtual ArrayView<const uint8_t> GetData() const = 0;
+  virtual std::span<const uint8_t> GetData() const = 0;
 
   // Copies `data` into the owned frame payload data.
-  virtual void SetData(ArrayView<const uint8_t> data) = 0;
+  virtual void SetData(std::span<const uint8_t> data) = 0;
 
   virtual uint8_t GetPayloadType() const = 0;
   virtual bool CanSetPayloadType() const { return false; }
@@ -125,7 +125,7 @@ class TransformableAudioFrameInterface : public TransformableFrameInterface {
   RTC_EXPORT explicit TransformableAudioFrameInterface(Passkey passkey);
   ~TransformableAudioFrameInterface() override = default;
 
-  virtual ArrayView<const uint32_t> GetContributingSources() const = 0;
+  virtual std::span<const uint32_t> GetContributingSources() const = 0;
 
   virtual const std::optional<uint16_t> SequenceNumber() const = 0;
 
diff --git a/api/neteq/BUILD.gn b/api/neteq/BUILD.gn
index 826e426e86..b68fbe8b57 100644
--- a/api/neteq/BUILD.gn
+++ b/api/neteq/BUILD.gn
@@ -17,7 +17,6 @@ rtc_library("neteq_api") {
   ]
 
   deps = [
-    "..:array_view",
     "..:rtp_headers",
     "..:rtp_packet_info",
     "..:scoped_refptr",
diff --git a/api/neteq/neteq.h b/api/neteq/neteq.h
index b0aed233f0..da667b5991 100644
--- a/api/neteq/neteq.h
+++ b/api/neteq/neteq.h
@@ -16,10 +16,10 @@
 
 #include <map>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/rtp_headers.h"
 #include "api/rtp_packet_info.h"
@@ -186,14 +186,14 @@ class NetEq {
   virtual ~NetEq() {}
 
   virtual int InsertPacket(const RTPHeader& rtp_header,
-                           ArrayView<const uint8_t> payload) {
+                           std::span<const uint8_t> payload) {
     return InsertPacket(rtp_header, payload,
                         /*receive_time=*/Timestamp::MinusInfinity());
   }
 
   // TODO: webrtc:343501093 - removed unused method.
   virtual int InsertPacket(const RTPHeader& rtp_header,
-                           ArrayView<const uint8_t> payload,
+                           std::span<const uint8_t> payload,
                            Timestamp receive_time) {
     return InsertPacket(rtp_header, payload,
                         RtpPacketInfo(rtp_header, receive_time));
@@ -203,7 +203,7 @@ class NetEq {
   // Returns 0 on success, -1 on failure.
   // TODO: webrtc:343501093 - Make this method pure virtual.
   virtual int InsertPacket(const RTPHeader& rtp_header,
-                           ArrayView<const uint8_t> payload,
+                           std::span<const uint8_t> payload,
                            const RtpPacketInfo& /* rtp_packet_info */) {
     return InsertPacket(rtp_header, payload);
   }
diff --git a/api/numerics/BUILD.gn b/api/numerics/BUILD.gn
index 3ec8c37391..2d9f6a2a85 100644
--- a/api/numerics/BUILD.gn
+++ b/api/numerics/BUILD.gn
@@ -16,7 +16,6 @@ rtc_library("numerics") {
     "samples_stats_counter.h",
   ]
   deps = [
-    "..:array_view",
     "../../rtc_base:checks",
     "../../rtc_base:rtc_numerics",
     "../units:timestamp",
diff --git a/api/numerics/samples_stats_counter.h b/api/numerics/samples_stats_counter.h
index 38d0cbe043..89bedb2a7c 100644
--- a/api/numerics/samples_stats_counter.h
+++ b/api/numerics/samples_stats_counter.h
@@ -15,10 +15,10 @@
 #include <stdint.h>
 
 #include <map>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/units/timestamp.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/numerics/running_statistics.h"
@@ -102,7 +102,7 @@ class SamplesStatsCounter {
   // guarantees of order, so samples can be in different order comparing to in
   // which they were added into counter. Also return value will be invalidate
   // after call to any non const method.
-  ArrayView<const StatsSample> GetTimedSamples() const { return samples_; }
+  std::span<const StatsSample> GetTimedSamples() const { return samples_; }
   std::vector<double> GetSamples() const {
     std::vector<double> out;
     out.reserve(samples_.size());
diff --git a/api/rtp_transceiver_interface.h b/api/rtp_transceiver_interface.h
index a4ff39ab74..021b215d7b 100644
--- a/api/rtp_transceiver_interface.h
+++ b/api/rtp_transceiver_interface.h
@@ -12,11 +12,11 @@
 #define API_RTP_TRANSCEIVER_INTERFACE_H_
 
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/base/attributes.h"
-#include "api/array_view.h"
 #include "api/media_types.h"
 #include "api/ref_count.h"
 #include "api/rtc_error.h"
@@ -152,7 +152,7 @@ class RTC_EXPORT RtpTransceiverInterface : public RefCountInterface {
   // by WebRTC for this transceiver.
   // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-setcodecpreferences
   virtual RTCError SetCodecPreferences(
-      ArrayView<RtpCodecCapability> codecs) = 0;
+      std::span<RtpCodecCapability> codecs) = 0;
   virtual std::vector<RtpCodecCapability> codec_preferences() const = 0;
 
   // Returns the set of header extensions that was set
@@ -172,7 +172,7 @@ class RTC_EXPORT RtpTransceiverInterface : public RefCountInterface {
   // so that it negotiates use of header extensions which are not kStopped.
   // https://w3c.github.io/webrtc-extensions/#rtcrtptransceiver-interface
   virtual RTCError SetHeaderExtensionsToNegotiate(
-      ArrayView<const RtpHeaderExtensionCapability> header_extensions) = 0;
+      std::span<const RtpHeaderExtensionCapability> header_extensions) = 0;
 
  protected:
   ~RtpTransceiverInterface() override = default;
diff --git a/api/test/fake_frame_decryptor.cc b/api/test/fake_frame_decryptor.cc
index 0e1da2c3f2..a57c086ab1 100644
--- a/api/test/fake_frame_decryptor.cc
+++ b/api/test/fake_frame_decryptor.cc
@@ -12,9 +12,9 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/media_types.h"
 #include "rtc_base/checks.h"
 
@@ -27,9 +27,9 @@ FakeFrameDecryptor::FakeFrameDecryptor(uint8_t fake_key,
 FakeFrameDecryptor::Result FakeFrameDecryptor::Decrypt(
     MediaType /* media_type */,
     const std::vector<uint32_t>& /* csrcs */,
-    ArrayView<const uint8_t> /* additional_data */,
-    ArrayView<const uint8_t> encrypted_frame,
-    ArrayView<uint8_t> frame) {
+    std::span<const uint8_t> /* additional_data */,
+    std::span<const uint8_t> encrypted_frame,
+    std::span<uint8_t> frame) {
   if (fail_decryption_) {
     return Result(Status::kFailedToDecrypt, 0);
   }
diff --git a/api/test/fake_frame_decryptor.h b/api/test/fake_frame_decryptor.h
index b3a79970d2..018a3e5b81 100644
--- a/api/test/fake_frame_decryptor.h
+++ b/api/test/fake_frame_decryptor.h
@@ -14,9 +14,9 @@
 #include <stddef.h>
 #include <stdint.h>
 
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/crypto/frame_decryptor_interface.h"
 #include "api/media_types.h"
 
@@ -36,9 +36,9 @@ class FakeFrameDecryptor : public FrameDecryptorInterface {
   // the postfix byte. This will always fail if fail_decryption_ is set to true.
   Result Decrypt(MediaType media_type,
                  const std::vector<uint32_t>& csrcs,
-                 ArrayView<const uint8_t> additional_data,
-                 ArrayView<const uint8_t> encrypted_frame,
-                 ArrayView<uint8_t> frame) override;
+                 std::span<const uint8_t> additional_data,
+                 std::span<const uint8_t> encrypted_frame,
+                 std::span<uint8_t> frame) override;
   // Always returns 1 less than the size of the encrypted frame.
   size_t GetMaxPlaintextByteSize(MediaType media_type,
                                  size_t encrypted_frame_size) override;
diff --git a/api/test/fake_frame_encryptor.cc b/api/test/fake_frame_encryptor.cc
index d35756e065..da996c478e 100644
--- a/api/test/fake_frame_encryptor.cc
+++ b/api/test/fake_frame_encryptor.cc
@@ -12,8 +12,8 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/media_types.h"
 #include "rtc_base/checks.h"
 
@@ -24,9 +24,9 @@ FakeFrameEncryptor::FakeFrameEncryptor(uint8_t fake_key, uint8_t postfix_byte)
 // FrameEncryptorInterface implementation
 int FakeFrameEncryptor::Encrypt(MediaType /* media_type */,
                                 uint32_t /* ssrc */,
-                                ArrayView<const uint8_t> /* additional_data */,
-                                ArrayView<const uint8_t> frame,
-                                ArrayView<uint8_t> encrypted_frame,
+                                std::span<const uint8_t> /* additional_data */,
+                                std::span<const uint8_t> frame,
+                                std::span<uint8_t> encrypted_frame,
                                 size_t* bytes_written) {
   if (fail_encryption_) {
     return static_cast<int>(FakeEncryptionStatus::FORCED_FAILURE);
diff --git a/api/test/fake_frame_encryptor.h b/api/test/fake_frame_encryptor.h
index eae0e41b3d..4f8d6c2957 100644
--- a/api/test/fake_frame_encryptor.h
+++ b/api/test/fake_frame_encryptor.h
@@ -14,7 +14,8 @@
 #include <stddef.h>
 #include <stdint.h>
 
-#include "api/array_view.h"
+#include <span>
+
 #include "api/crypto/frame_encryptor_interface.h"
 #include "api/media_types.h"
 #include "rtc_base/ref_counted_object.h"
@@ -34,9 +35,9 @@ class FakeFrameEncryptor : public RefCountedObject<FrameEncryptorInterface> {
   // bit to the end. This will always fail if fail_encryption_ is set to true.
   int Encrypt(MediaType media_type,
               uint32_t ssrc,
-              ArrayView<const uint8_t> additional_data,
-              ArrayView<const uint8_t> frame,
-              ArrayView<uint8_t> encrypted_frame,
+              std::span<const uint8_t> additional_data,
+              std::span<const uint8_t> frame,
+              std::span<uint8_t> encrypted_frame,
               size_t* bytes_written) override;
   // Always returns 1 more than the size of the frame.
   size_t GetMaxCiphertextByteSize(MediaType media_type,
diff --git a/api/test/metrics/BUILD.gn b/api/test/metrics/BUILD.gn
index abb6453bbe..7834e21c6d 100644
--- a/api/test/metrics/BUILD.gn
+++ b/api/test/metrics/BUILD.gn
@@ -95,10 +95,7 @@ rtc_library("metrics_accumulator") {
 rtc_library("metrics_exporter") {
   visibility = [ "*" ]
   sources = [ "metrics_exporter.h" ]
-  deps = [
-    ":metric",
-    "../..:array_view",
-  ]
+  deps = [ ":metric" ]
 }
 
 if (!build_with_chromium) {
@@ -114,7 +111,6 @@ if (!build_with_chromium) {
     deps = [
       ":metric",
       ":metrics_exporter",
-      "../..:array_view",
       "../../../rtc_base:stringutils",
       "../../../test:test_flags",
       "//third_party/abseil-cpp/absl/flags:flag",
@@ -133,7 +129,6 @@ rtc_library("chrome_perf_dashboard_metrics_exporter") {
   deps = [
     ":metric",
     ":metrics_exporter",
-    "../..:array_view",
     "../../../test:fileutils",
     "../../../test:perf_test",
     "//third_party/abseil-cpp/absl/memory",
@@ -160,7 +155,6 @@ rtc_library("metrics_set_proto_file_exporter") {
   deps = [
     ":metric",
     ":metrics_exporter",
-    "../..:array_view",
     "../../../rtc_base:logging",
     "../../../test:fileutils",
     "//third_party/abseil-cpp/absl/strings:string_view",
@@ -181,7 +175,6 @@ rtc_library("print_result_proxy_metrics_exporter") {
   deps = [
     ":metric",
     ":metrics_exporter",
-    "../..:array_view",
     "../../../test:perf_test",
     "../../numerics",
   ]
@@ -259,7 +252,6 @@ if (rtc_include_tests) {
       ":metric",
       ":metrics_exporter",
       ":metrics_logger",
-      "../..:array_view",
       "../../../system_wrappers",
       "../../../test:test_support",
     ]
diff --git a/api/test/metrics/chrome_perf_dashboard_metrics_exporter.cc b/api/test/metrics/chrome_perf_dashboard_metrics_exporter.cc
index 79d50d6c64..3ba87cb96d 100644
--- a/api/test/metrics/chrome_perf_dashboard_metrics_exporter.cc
+++ b/api/test/metrics/chrome_perf_dashboard_metrics_exporter.cc
@@ -11,12 +11,12 @@
 
 #include <cstdio>
 #include <memory>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/memory/memory.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/test/metrics/metric.h"
 #include "test/testsupport/file_utils.h"
 #include "test/testsupport/perf_test.h"
@@ -101,7 +101,7 @@ ChromePerfDashboardMetricsExporter::ChromePerfDashboardMetricsExporter(
     : export_file_path_(export_file_path) {}
 
 bool ChromePerfDashboardMetricsExporter::Export(
-    ArrayView<const Metric> metrics) {
+    std::span<const Metric> metrics) {
   std::unique_ptr<PerfTestResultWriter> writer =
       absl::WrapUnique<PerfTestResultWriter>(CreateHistogramWriter());
   for (const Metric& metric : metrics) {
diff --git a/api/test/metrics/chrome_perf_dashboard_metrics_exporter.h b/api/test/metrics/chrome_perf_dashboard_metrics_exporter.h
index 10447510a8..03959930cb 100644
--- a/api/test/metrics/chrome_perf_dashboard_metrics_exporter.h
+++ b/api/test/metrics/chrome_perf_dashboard_metrics_exporter.h
@@ -11,10 +11,10 @@
 #ifndef API_TEST_METRICS_CHROME_PERF_DASHBOARD_METRICS_EXPORTER_H_
 #define API_TEST_METRICS_CHROME_PERF_DASHBOARD_METRICS_EXPORTER_H_
 
+#include <span>
 #include <string>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/test/metrics/metric.h"
 #include "api/test/metrics/metrics_exporter.h"
 
@@ -29,7 +29,7 @@ class ChromePerfDashboardMetricsExporter : public MetricsExporter {
       absl::string_view export_file_path);
   ~ChromePerfDashboardMetricsExporter() override = default;
 
-  bool Export(ArrayView<const Metric> metrics) override;
+  bool Export(std::span<const Metric> metrics) override;
 
  private:
   const std::string export_file_path_;
diff --git a/api/test/metrics/global_metrics_logger_and_exporter_test.cc b/api/test/metrics/global_metrics_logger_and_exporter_test.cc
index bf462a6f3f..4e50a938c5 100644
--- a/api/test/metrics/global_metrics_logger_and_exporter_test.cc
+++ b/api/test/metrics/global_metrics_logger_and_exporter_test.cc
@@ -12,11 +12,11 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/test/metrics/metric.h"
 #include "api/test/metrics/metrics_exporter.h"
 #include "api/test/metrics/metrics_logger.h"
@@ -54,7 +54,7 @@ struct TestMetricsExporterFactory {
         : factory_(factory), export_result_(export_result) {}
     ~TestMetricsExporter() override = default;
 
-    bool Export(ArrayView<const Metric> metrics) override {
+    bool Export(std::span<const Metric> metrics) override {
       factory_->exported_metrics =
           std::vector<Metric>(metrics.begin(), metrics.end());
       return export_result_;
diff --git a/api/test/metrics/metrics_exporter.h b/api/test/metrics/metrics_exporter.h
index b71d4fbd93..2640fc8125 100644
--- a/api/test/metrics/metrics_exporter.h
+++ b/api/test/metrics/metrics_exporter.h
@@ -11,7 +11,8 @@
 #ifndef API_TEST_METRICS_METRICS_EXPORTER_H_
 #define API_TEST_METRICS_METRICS_EXPORTER_H_
 
-#include "api/array_view.h"
+#include <span>
+
 #include "api/test/metrics/metric.h"
 
 namespace webrtc {
@@ -24,7 +25,7 @@ class MetricsExporter {
 
   // Exports specified metrics in a format that depends on the implementation.
   // Returns true if export succeeded, false otherwise.
-  virtual bool Export(ArrayView<const Metric> metrics) = 0;
+  virtual bool Export(std::span<const Metric> metrics) = 0;
 };
 
 }  // namespace test
diff --git a/api/test/metrics/metrics_set_proto_file_exporter.cc b/api/test/metrics/metrics_set_proto_file_exporter.cc
index c3a7554585..88588e890d 100644
--- a/api/test/metrics/metrics_set_proto_file_exporter.cc
+++ b/api/test/metrics/metrics_set_proto_file_exporter.cc
@@ -11,11 +11,11 @@
 
 #include <cstdio>
 #include <map>
+#include <span>
 #include <string>
 #include <utility>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/test/metrics/metric.h"
 #include "rtc_base/logging.h"
 #include "test/testsupport/file_utils.h"
@@ -131,7 +131,7 @@ MetricsSetProtoFileExporter::Options::Options(
     std::map<std::string, std::string> metadata)
     : export_file_path(export_file_path), metadata(std::move(metadata)) {}
 
-bool MetricsSetProtoFileExporter::Export(ArrayView<const Metric> metrics) {
+bool MetricsSetProtoFileExporter::Export(std::span<const Metric> metrics) {
 #if WEBRTC_ENABLE_PROTOBUF
   test_metrics::MetricsSet metrics_set;
   for (const auto& [key, value] : options_.metadata) {
diff --git a/api/test/metrics/metrics_set_proto_file_exporter.h b/api/test/metrics/metrics_set_proto_file_exporter.h
index 6e38eb9981..6338527c1b 100644
--- a/api/test/metrics/metrics_set_proto_file_exporter.h
+++ b/api/test/metrics/metrics_set_proto_file_exporter.h
@@ -12,10 +12,10 @@
 #define API_TEST_METRICS_METRICS_SET_PROTO_FILE_EXPORTER_H_
 
 #include <map>
+#include <span>
 #include <string>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/test/metrics/metric.h"
 #include "api/test/metrics/metrics_exporter.h"
 
@@ -48,7 +48,7 @@ class MetricsSetProtoFileExporter : public MetricsExporter {
   MetricsSetProtoFileExporter& operator=(const MetricsSetProtoFileExporter&) =
       delete;
 
-  bool Export(ArrayView<const Metric> metrics) override;
+  bool Export(std::span<const Metric> metrics) override;
 
  private:
   const Options options_;
diff --git a/api/test/metrics/print_result_proxy_metrics_exporter.cc b/api/test/metrics/print_result_proxy_metrics_exporter.cc
index cb163facde..2a168585ef 100644
--- a/api/test/metrics/print_result_proxy_metrics_exporter.cc
+++ b/api/test/metrics/print_result_proxy_metrics_exporter.cc
@@ -10,10 +10,10 @@
 #include "api/test/metrics/print_result_proxy_metrics_exporter.h"
 
 #include <cstddef>
+#include <span>
 #include <string>
 #include <unordered_set>
 
-#include "api/array_view.h"
 #include "api/numerics/samples_stats_counter.h"
 #include "api/test/metrics/metric.h"
 #include "test/testsupport/perf_test.h"
@@ -78,7 +78,7 @@ bool NameEndsWithConnected(const std::string& name) {
 
 }  // namespace
 
-bool PrintResultProxyMetricsExporter::Export(ArrayView<const Metric> metrics) {
+bool PrintResultProxyMetricsExporter::Export(std::span<const Metric> metrics) {
   static const std::unordered_set<std::string> per_call_metrics{
       "actual_encode_bitrate",
       "encode_frame_rate",
diff --git a/api/test/metrics/print_result_proxy_metrics_exporter.h b/api/test/metrics/print_result_proxy_metrics_exporter.h
index cbc72ab212..a84304bda5 100644
--- a/api/test/metrics/print_result_proxy_metrics_exporter.h
+++ b/api/test/metrics/print_result_proxy_metrics_exporter.h
@@ -11,7 +11,8 @@
 #ifndef API_TEST_METRICS_PRINT_RESULT_PROXY_METRICS_EXPORTER_H_
 #define API_TEST_METRICS_PRINT_RESULT_PROXY_METRICS_EXPORTER_H_
 
-#include "api/array_view.h"
+#include <span>
+
 #include "api/test/metrics/metric.h"
 #include "api/test/metrics/metrics_exporter.h"
 
@@ -23,7 +24,7 @@ class PrintResultProxyMetricsExporter : public MetricsExporter {
  public:
   ~PrintResultProxyMetricsExporter() override = default;
 
-  bool Export(ArrayView<const Metric> metrics) override;
+  bool Export(std::span<const Metric> metrics) override;
 };
 
 }  // namespace test
diff --git a/api/test/metrics/stdout_metrics_exporter.cc b/api/test/metrics/stdout_metrics_exporter.cc
index 336817f8da..f87fe026b4 100644
--- a/api/test/metrics/stdout_metrics_exporter.cc
+++ b/api/test/metrics/stdout_metrics_exporter.cc
@@ -13,11 +13,11 @@
 #include <cstdint>
 #include <cstdio>
 #include <optional>
+#include <span>
 #include <string>
 
 #include "absl/flags/flag.h"
 #include "absl/strings/str_cat.h"
-#include "api/array_view.h"
 #include "api/test/metrics/metric.h"
 #include "rtc_base/strings/string_builder.h"
 #include "test/test_flags.h"
@@ -89,7 +89,7 @@ std::string TestCaseAndMetadata(const Metric& metric) {
 
 StdoutMetricsExporter::StdoutMetricsExporter() : output_(stdout) {}
 
-bool StdoutMetricsExporter::Export(ArrayView<const Metric> metrics) {
+bool StdoutMetricsExporter::Export(std::span<const Metric> metrics) {
   for (const Metric& metric : metrics) {
     PrintMetric(metric);
   }
diff --git a/api/test/metrics/stdout_metrics_exporter.h b/api/test/metrics/stdout_metrics_exporter.h
index 60354b58fd..bdf473bf6e 100644
--- a/api/test/metrics/stdout_metrics_exporter.h
+++ b/api/test/metrics/stdout_metrics_exporter.h
@@ -12,8 +12,8 @@
 #define API_TEST_METRICS_STDOUT_METRICS_EXPORTER_H_
 
 #include <cstdio>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/test/metrics/metric.h"
 #include "api/test/metrics/metrics_exporter.h"
 
@@ -29,7 +29,7 @@ class StdoutMetricsExporter : public MetricsExporter {
   StdoutMetricsExporter(const StdoutMetricsExporter&) = delete;
   StdoutMetricsExporter& operator=(const StdoutMetricsExporter&) = delete;
 
-  bool Export(ArrayView<const Metric> metrics) override;
+  bool Export(std::span<const Metric> metrics) override;
 
  private:
   void PrintMetric(const Metric& metric);
diff --git a/api/test/mock_datagram_connection.h b/api/test/mock_datagram_connection.h
index 128add909d..e31e4fe7b0 100644
--- a/api/test/mock_datagram_connection.h
+++ b/api/test/mock_datagram_connection.h
@@ -12,11 +12,11 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 #include <type_traits>
 
 #include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/candidate.h"
 #include "api/datagram_connection.h"
 #include "p2p/base/transport_description.h"
@@ -45,14 +45,14 @@ class MockDatagramConnection : public DatagramConnection {
               (override));
   MOCK_METHOD(void,
               SendPackets,
-              (ArrayView<PacketSendParameters> packets),
+              (std::span<PacketSendParameters> packets),
               (override));
   MOCK_METHOD(void,
               Terminate,
               (absl::AnyInvocable<void()> terminate_complete_callback),
               (override));
-  MOCK_METHOD(std::string_view, IceUsernameFragment, (), (override));
-  MOCK_METHOD(std::string_view, IcePassword, (), (override));
+  MOCK_METHOD(absl::string_view, IceUsernameFragment, (), (override));
+  MOCK_METHOD(absl::string_view, IcePassword, (), (override));
 };
 
 static_assert(!std::is_abstract_v<RefCountedObject<MockDatagramConnection>>,
diff --git a/api/test/mock_datagram_connection_observer.h b/api/test/mock_datagram_connection_observer.h
index 985087210d..c997865890 100644
--- a/api/test/mock_datagram_connection_observer.h
+++ b/api/test/mock_datagram_connection_observer.h
@@ -11,8 +11,8 @@
 #define API_TEST_MOCK_DATAGRAM_CONNECTION_OBSERVER_H_
 
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/candidate.h"
 #include "api/datagram_connection.h"
 #include "test/gmock.h"
@@ -27,7 +27,7 @@ class MockDatagramConnectionObserver : public DatagramConnection::Observer {
               (override));
   MOCK_METHOD(void,
               OnPacketReceived,
-              (ArrayView<const uint8_t> data, PacketMetadata metadata),
+              (std::span<const uint8_t> data, PacketMetadata metadata),
               (override));
   MOCK_METHOD(void, OnSendOutcome, (SendOutcome send_outcome), (override));
   MOCK_METHOD(void, OnConnectionError, (), (override));
diff --git a/api/test/mock_frame_decryptor.h b/api/test/mock_frame_decryptor.h
index 191a7dd83a..93ed397b6f 100644
--- a/api/test/mock_frame_decryptor.h
+++ b/api/test/mock_frame_decryptor.h
@@ -13,9 +13,9 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/crypto/frame_decryptor_interface.h"
 #include "api/media_types.h"
 #include "test/gmock.h"
@@ -28,9 +28,9 @@ class MockFrameDecryptor : public FrameDecryptorInterface {
               Decrypt,
               (MediaType,
                const std::vector<uint32_t>&,
-               ArrayView<const uint8_t>,
-               ArrayView<const uint8_t>,
-               ArrayView<uint8_t>),
+               std::span<const uint8_t>,
+               std::span<const uint8_t>,
+               std::span<uint8_t>),
               (override));
 
   MOCK_METHOD(size_t,
diff --git a/api/test/mock_frame_encryptor.h b/api/test/mock_frame_encryptor.h
index a449a29aee..85e3507777 100644
--- a/api/test/mock_frame_encryptor.h
+++ b/api/test/mock_frame_encryptor.h
@@ -13,8 +13,8 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/crypto/frame_encryptor_interface.h"
 #include "api/media_types.h"
 #include "test/gmock.h"
@@ -27,9 +27,9 @@ class MockFrameEncryptor : public FrameEncryptorInterface {
               Encrypt,
               (MediaType,
                uint32_t,
-               ArrayView<const uint8_t>,
-               ArrayView<const uint8_t>,
-               ArrayView<uint8_t>,
+               std::span<const uint8_t>,
+               std::span<const uint8_t>,
+               std::span<uint8_t>,
                size_t*),
               (override));
 
diff --git a/api/test/mock_rtp_transceiver.h b/api/test/mock_rtp_transceiver.h
index 1a55efa1e1..25a5c89e99 100644
--- a/api/test/mock_rtp_transceiver.h
+++ b/api/test/mock_rtp_transceiver.h
@@ -12,10 +12,10 @@
 #define API_TEST_MOCK_RTP_TRANSCEIVER_H_
 
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/make_ref_counted.h"
 #include "api/media_types.h"
 #include "api/rtc_error.h"
@@ -68,7 +68,7 @@ class MockRtpTransceiver : public RtpTransceiverInterface {
   MOCK_METHOD(void, Stop, (), (override));
   MOCK_METHOD(RTCError,
               SetCodecPreferences,
-              (ArrayView<RtpCodecCapability> codecs),
+              (std::span<RtpCodecCapability> codecs),
               (override));
   MOCK_METHOD(std::vector<RtpCodecCapability>,
               codec_preferences,
@@ -84,7 +84,7 @@ class MockRtpTransceiver : public RtpTransceiverInterface {
               (const, override));
   MOCK_METHOD(RTCError,
               SetHeaderExtensionsToNegotiate,
-              (ArrayView<const RtpHeaderExtensionCapability> header_extensions),
+              (std::span<const RtpHeaderExtensionCapability> header_extensions),
               (override));
 };
 
diff --git a/api/test/mock_transformable_audio_frame.h b/api/test/mock_transformable_audio_frame.h
index 9de3c3fa29..6abb24a547 100644
--- a/api/test/mock_transformable_audio_frame.h
+++ b/api/test/mock_transformable_audio_frame.h
@@ -13,9 +13,9 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 
-#include "api/array_view.h"
 #include "api/frame_transformer_interface.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
@@ -27,8 +27,8 @@ class MockTransformableAudioFrame : public TransformableAudioFrameInterface {
  public:
   MockTransformableAudioFrame() : TransformableAudioFrameInterface(Passkey()) {}
 
-  MOCK_METHOD(ArrayView<const uint8_t>, GetData, (), (const, override));
-  MOCK_METHOD(void, SetData, (ArrayView<const uint8_t>), (override));
+  MOCK_METHOD(std::span<const uint8_t>, GetData, (), (const, override));
+  MOCK_METHOD(void, SetData, (std::span<const uint8_t>), (override));
   MOCK_METHOD(void, SetRTPTimestamp, (uint32_t), (override));
   MOCK_METHOD(uint8_t, GetPayloadType, (), (const, override));
   MOCK_METHOD(bool, CanSetPayloadType, (), (const, override));
@@ -36,7 +36,7 @@ class MockTransformableAudioFrame : public TransformableAudioFrameInterface {
   MOCK_METHOD(uint32_t, GetSsrc, (), (const, override));
   MOCK_METHOD(uint32_t, GetTimestamp, (), (const, override));
   MOCK_METHOD(std::string, GetMimeType, (), (const, override));
-  MOCK_METHOD(ArrayView<const uint32_t>,
+  MOCK_METHOD(std::span<const uint32_t>,
               GetContributingSources,
               (),
               (const, override));
diff --git a/api/test/mock_transformable_frame.h b/api/test/mock_transformable_frame.h
index 252622abc4..804901d41b 100644
--- a/api/test/mock_transformable_frame.h
+++ b/api/test/mock_transformable_frame.h
@@ -14,10 +14,10 @@
 #include <stdint.h>
 
 #include <optional>
+#include <span>
 #include <string>
 #include <type_traits>
 
-#include "api/array_view.h"
 #include "api/frame_transformer_interface.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
@@ -29,8 +29,8 @@ class MockTransformableFrame : public TransformableFrameInterface {
  public:
   MockTransformableFrame() : TransformableFrameInterface(Passkey()) {}
 
-  MOCK_METHOD(ArrayView<const uint8_t>, GetData, (), (const, override));
-  MOCK_METHOD(void, SetData, (ArrayView<const uint8_t>), (override));
+  MOCK_METHOD(std::span<const uint8_t>, GetData, (), (const, override));
+  MOCK_METHOD(void, SetData, (std::span<const uint8_t>), (override));
   MOCK_METHOD(uint8_t, GetPayloadType, (), (const, override));
   MOCK_METHOD(bool, CanSetPayloadType, (), (const, override));
   MOCK_METHOD(void, SetPayloadType, (uint8_t), (override));
diff --git a/api/test/mock_transformable_video_frame.h b/api/test/mock_transformable_video_frame.h
index 2529dcca66..8f418a32b9 100644
--- a/api/test/mock_transformable_video_frame.h
+++ b/api/test/mock_transformable_video_frame.h
@@ -13,10 +13,10 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 #include <type_traits>
 
-#include "api/array_view.h"
 #include "api/frame_transformer_interface.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
@@ -28,8 +28,8 @@ namespace webrtc {
 class MockTransformableVideoFrame : public TransformableVideoFrameInterface {
  public:
   MockTransformableVideoFrame() : TransformableVideoFrameInterface(Passkey()) {}
-  MOCK_METHOD(ArrayView<const uint8_t>, GetData, (), (const, override));
-  MOCK_METHOD(void, SetData, (ArrayView<const uint8_t> data), (override));
+  MOCK_METHOD(std::span<const uint8_t>, GetData, (), (const, override));
+  MOCK_METHOD(void, SetData, (std::span<const uint8_t> data), (override));
   MOCK_METHOD(uint32_t, GetTimestamp, (), (const, override));
   MOCK_METHOD(void, SetRTPTimestamp, (uint32_t), (override));
   MOCK_METHOD(uint32_t, GetSsrc, (), (const, override));
diff --git a/api/test/network_emulation_manager.h b/api/test/network_emulation_manager.h
index 611a4b349f..de5afe8027 100644
--- a/api/test/network_emulation_manager.h
+++ b/api/test/network_emulation_manager.h
@@ -15,13 +15,13 @@
 #include <functional>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/base/nullability.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/field_trials_view.h"
 #include "api/test/network_emulation/cross_traffic.h"
 #include "api/test/network_emulation/network_emulation_interfaces.h"
@@ -346,14 +346,14 @@ class NetworkEmulationManager {
   // `stats_callback`. Callback will be executed on network emulation
   // internal task queue.
   virtual void GetStats(
-      ArrayView<EmulatedEndpoint* const> endpoints,
+      std::span<EmulatedEndpoint* const> endpoints,
       std::function<void(EmulatedNetworkStats)> stats_callback) = 0;
 
   // Passes combined network stats for all specified `nodes` into specified
   // `stats_callback`. Callback will be executed on network emulation
   // internal task queue.
   virtual void GetStats(
-      ArrayView<EmulatedNetworkNode* const> nodes,
+      std::span<EmulatedNetworkNode* const> nodes,
       std::function<void(EmulatedNetworkNodeStats)> stats_callback) = 0;
 
   // Create a EmulatedTURNServer.
diff --git a/api/test/pclf/BUILD.gn b/api/test/pclf/BUILD.gn
index 363beddc70..858b09ba0e 100644
--- a/api/test/pclf/BUILD.gn
+++ b/api/test/pclf/BUILD.gn
@@ -17,7 +17,6 @@ rtc_library("media_configuration") {
   ]
 
   deps = [
-    "../..:array_view",
     "../..:audio_options_api",
     "../..:media_stream_interface",
     "../..:rtp_parameters",
diff --git a/api/test/pclf/media_configuration.cc b/api/test/pclf/media_configuration.cc
index 679486ded4..1ce3f2a5e2 100644
--- a/api/test/pclf/media_configuration.cc
+++ b/api/test/pclf/media_configuration.cc
@@ -16,12 +16,12 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/test/video/video_frame_writer.h"
 #include "api/units/time_delta.h"
 #include "rtc_base/checks.h"
@@ -229,7 +229,7 @@ VideoCodecConfig::VideoCodecConfig(
     : name(name), required_params(std::move(required_params)) {}
 
 std::optional<VideoResolution> VideoSubscription::GetMaxResolution(
-    ArrayView<const VideoConfig> video_configs) {
+    std::span<const VideoConfig> video_configs) {
   std::vector<VideoResolution> resolutions;
   for (const auto& video_config : video_configs) {
     resolutions.push_back(video_config.GetResolution());
@@ -238,7 +238,7 @@ std::optional<VideoResolution> VideoSubscription::GetMaxResolution(
 }
 
 std::optional<VideoResolution> VideoSubscription::GetMaxResolution(
-    ArrayView<const VideoResolution> resolutions) {
+    std::span<const VideoResolution> resolutions) {
   if (resolutions.empty()) {
     return std::nullopt;
   }
diff --git a/api/test/pclf/media_configuration.h b/api/test/pclf/media_configuration.h
index 5f8e45cb76..c39978aa88 100644
--- a/api/test/pclf/media_configuration.h
+++ b/api/test/pclf/media_configuration.h
@@ -17,11 +17,11 @@
 #include <map>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/audio_options.h"
 #include "api/media_stream_interface.h"
 #include "api/rtp_parameters.h"
@@ -394,9 +394,9 @@ class VideoSubscription {
   // Returns the resolution constructed as maximum from all resolution
   // dimensions: width, height and fps.
   static std::optional<VideoResolution> GetMaxResolution(
-      ArrayView<const VideoConfig> video_configs);
+      std::span<const VideoConfig> video_configs);
   static std::optional<VideoResolution> GetMaxResolution(
-      ArrayView<const VideoResolution> resolutions);
+      std::span<const VideoResolution> resolutions);
 
   bool operator==(const VideoSubscription& other) const;
   bool operator!=(const VideoSubscription& other) const;
diff --git a/api/test/video_quality_analyzer_interface.h b/api/test/video_quality_analyzer_interface.h
index 43223364ec..a06342c3a8 100644
--- a/api/test/video_quality_analyzer_interface.h
+++ b/api/test/video_quality_analyzer_interface.h
@@ -13,10 +13,10 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <string>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/scoped_refptr.h"
 #include "api/stats/rtc_stats_report.h"
 #include "api/test/stats_observer_interface.h"
@@ -88,7 +88,7 @@ class VideoQualityAnalyzerInterface
   // thread in each method, but should remember, that it is the same thread,
   // that is used in video pipeline.
   virtual void Start(std::string /* test_case_name */,
-                     ArrayView<const std::string> /* peer_names */,
+                     std::span<const std::string> /* peer_names */,
                      int /* max_threads_count */) {}
 
   // Will be called when frame was generated from the input stream.
diff --git a/api/transport/BUILD.gn b/api/transport/BUILD.gn
index 136a8f937a..26737a2717 100644
--- a/api/transport/BUILD.gn
+++ b/api/transport/BUILD.gn
@@ -65,7 +65,6 @@ rtc_source_set("datagram_transport_interface") {
   visibility = [ "*" ]
   sources = [ "data_channel_transport_interface.h" ]
   deps = [
-    "..:array_view",
     "..:priority",
     "..:rtc_error",
     "../../rtc_base:copy_on_write_buffer",
@@ -110,7 +109,6 @@ rtc_library("stun_types") {
   ]
 
   deps = [
-    "../../api:array_view",
     "../../rtc_base:byte_buffer",
     "../../rtc_base:byte_order",
     "../../rtc_base:checks",
@@ -176,7 +174,6 @@ if (rtc_include_tests) {
     sources = [ "stun_unittest.cc" ]
     deps = [
       ":stun_types",
-      "..:array_view",
       "../../rtc_base:byte_buffer",
       "../../rtc_base:byte_order",
       "../../rtc_base:ip_address",
diff --git a/api/transport/rtp/BUILD.gn b/api/transport/rtp/BUILD.gn
index fa0dc97379..dc627f8a14 100644
--- a/api/transport/rtp/BUILD.gn
+++ b/api/transport/rtp/BUILD.gn
@@ -41,7 +41,6 @@ rtc_library("corruption_detection_message") {
     "corruption_detection_message.h",
   ]
   deps = [
-    "../../:array_view",
     "../../../rtc_base:checks",
     "../../video/corruption_detection:frame_instrumentation_data",
     "//third_party/abseil-cpp/absl/container:inlined_vector",
diff --git a/api/transport/rtp/corruption_detection_message.cc b/api/transport/rtp/corruption_detection_message.cc
index 68e1ca66ec..903b0ca16b 100644
--- a/api/transport/rtp/corruption_detection_message.cc
+++ b/api/transport/rtp/corruption_detection_message.cc
@@ -11,9 +11,9 @@
 #include "api/transport/rtp/corruption_detection_message.h"
 
 #include <optional>
+#include <span>
 #include <utility>
 
-#include "api/array_view.h"
 #include "api/video/corruption_detection/frame_instrumentation_data.h"
 #include "rtc_base/checks.h"
 
@@ -116,7 +116,7 @@ CorruptionDetectionMessage::Builder::WithChromaErrorThreshold(
 
 CorruptionDetectionMessage::Builder&
 CorruptionDetectionMessage::Builder::WithSampleValues(
-    const ArrayView<const double>& sample_values) {
+    const std::span<const double>& sample_values) {
   message_.sample_values_.assign(sample_values.begin(), sample_values.end());
   return *this;
 }
diff --git a/api/transport/rtp/corruption_detection_message.h b/api/transport/rtp/corruption_detection_message.h
index 81ea224c7c..2765b5145e 100644
--- a/api/transport/rtp/corruption_detection_message.h
+++ b/api/transport/rtp/corruption_detection_message.h
@@ -13,9 +13,9 @@
 
 #include <cstddef>
 #include <optional>
+#include <span>
 
 #include "absl/container/inlined_vector.h"
-#include "api/array_view.h"
 #include "api/video/corruption_detection/frame_instrumentation_data.h"
 
 namespace webrtc {
@@ -39,8 +39,8 @@ class CorruptionDetectionMessage {
   double std_dev() const { return std_dev_; }
   int luma_error_threshold() const { return luma_error_threshold_; }
   int chroma_error_threshold() const { return chroma_error_threshold_; }
-  ArrayView<const double> sample_values() const {
-    return MakeArrayView(sample_values_.data(), sample_values_.size());
+  std::span<const double> sample_values() const {
+    return std::span(sample_values_.data(), sample_values_.size());
   }
 
   static CorruptionDetectionMessage FromFrameInstrumentationData(
@@ -95,7 +95,7 @@ class CorruptionDetectionMessage::Builder {
   Builder& WithStdDev(double std_dev);
   Builder& WithLumaErrorThreshold(int luma_error_threshold);
   Builder& WithChromaErrorThreshold(int chroma_error_threshold);
-  Builder& WithSampleValues(const ArrayView<const double>& sample_values);
+  Builder& WithSampleValues(const std::span<const double>& sample_values);
 
  private:
   CorruptionDetectionMessage message_;
diff --git a/api/transport/stun.cc b/api/transport/stun.cc
index 6b9ca04d11..2849a5aa0c 100644
--- a/api/transport/stun.cc
+++ b/api/transport/stun.cc
@@ -17,12 +17,12 @@
 #include <iterator>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "rtc_base/byte_buffer.h"
 #include "rtc_base/byte_order.h"
 #include "rtc_base/checks.h"
@@ -51,8 +51,8 @@ uint32_t ReduceTransactionId(absl::string_view transaction_id) {
              transaction_id.length() == kStunLegacyTransactionIdLength)
       << transaction_id.length();
   ByteBufferReader reader(
-      MakeArrayView(reinterpret_cast<const uint8_t*>(transaction_id.data()),
-                    transaction_id.size()));
+      std::span(reinterpret_cast<const uint8_t*>(transaction_id.data()),
+                transaction_id.size()));
   uint32_t result = 0;
   uint32_t next;
   while (reader.ReadUInt32(&next)) {
@@ -379,7 +379,7 @@ bool StunMessage::ValidateMessageIntegrityOfType(int mi_attr_type,
     return false;
   }
 
-  ArrayView<const uint8_t> data_view(reinterpret_cast<const uint8_t*>(data),
+  std::span<const uint8_t> data_view(reinterpret_cast<const uint8_t*>(data),
                                      size);
 
   // Getting the message length from the STUN header.
@@ -438,7 +438,7 @@ bool StunMessage::ValidateMessageIntegrityOfType(int mi_attr_type,
     //     |0 0|     STUN Message Type     |         Message Length        |
     //     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     SetBE16(
-        ArrayView<uint8_t>(reinterpret_cast<uint8_t*>(temp_data.get() + 2), 2),
+        std::span<uint8_t>(reinterpret_cast<uint8_t*>(temp_data.get() + 2), 2),
         static_cast<uint16_t>(new_adjusted_len));
   }
 
@@ -510,7 +510,7 @@ bool StunMessage::ValidateFingerprint(const char* data, size_t size) {
   if (size % 4 != 0 || size < kStunHeaderSize + fingerprint_attr_size)
     return false;
 
-  ArrayView<const uint8_t> data_view(reinterpret_cast<const uint8_t*>(data),
+  std::span<const uint8_t> data_view(reinterpret_cast<const uint8_t*>(data),
                                      size);
 
   // Skip the rest if the magic cookie isn't present.
@@ -539,14 +539,14 @@ std::string StunMessage::GenerateTransactionId() {
   return CreateRandomString(kStunTransactionIdLength);
 }
 
-bool StunMessage::IsStunMethod(ArrayView<int> methods,
+bool StunMessage::IsStunMethod(std::span<int> methods,
                                const char* data,
                                size_t size) {
   // Check the message length.
   if (size % 4 != 0 || size < kStunHeaderSize)
     return false;
 
-  ArrayView<const uint8_t> data_view(reinterpret_cast<const uint8_t*>(data),
+  std::span<const uint8_t> data_view(reinterpret_cast<const uint8_t*>(data),
                                      size);
 
   // Skip the rest if the magic cookie isn't present.
@@ -816,7 +816,7 @@ void StunAttribute::WritePadding(ByteBufferWriter* buf) const {
   int remainder = length_ % 4;
   if (remainder > 0) {
     uint8_t zeroes[4] = {0};
-    buf->Write(ArrayView<const uint8_t>(zeroes, 4 - remainder));
+    buf->Write(std::span<const uint8_t>(zeroes, 4 - remainder));
   }
 }
 
@@ -915,8 +915,8 @@ bool StunAddressAttribute::Read(ByteBufferReader* buf) {
     if (length() != SIZE_IP4) {
       return false;
     }
-    if (!buf->ReadBytes(MakeArrayView(reinterpret_cast<uint8_t*>(&v4addr),
-                                      sizeof(v4addr)))) {
+    if (!buf->ReadBytes(
+            std::span(reinterpret_cast<uint8_t*>(&v4addr), sizeof(v4addr)))) {
       return false;
     }
     IPAddress ipaddr(v4addr);
@@ -926,8 +926,8 @@ bool StunAddressAttribute::Read(ByteBufferReader* buf) {
     if (length() != SIZE_IP6) {
       return false;
     }
-    if (!buf->ReadBytes(MakeArrayView(reinterpret_cast<uint8_t*>(&v6addr),
-                                      sizeof(v6addr)))) {
+    if (!buf->ReadBytes(
+            std::span(reinterpret_cast<uint8_t*>(&v6addr), sizeof(v6addr)))) {
       return false;
     }
     IPAddress ipaddr(v6addr);
@@ -950,13 +950,13 @@ bool StunAddressAttribute::Write(ByteBufferWriter* buf) const {
   switch (address_.family()) {
     case AF_INET: {
       in_addr v4addr = address_.ipaddr().ipv4_address();
-      buf->Write(ArrayView<const uint8_t>(reinterpret_cast<uint8_t*>(&v4addr),
+      buf->Write(std::span<const uint8_t>(reinterpret_cast<uint8_t*>(&v4addr),
                                           sizeof(v4addr)));
       break;
     }
     case AF_INET6: {
       in6_addr v6addr = address_.ipaddr().ipv6_address();
-      buf->Write(ArrayView<const uint8_t>(reinterpret_cast<uint8_t*>(&v6addr),
+      buf->Write(std::span<const uint8_t>(reinterpret_cast<uint8_t*>(&v6addr),
                                           sizeof(v6addr)));
       break;
     }
@@ -1040,13 +1040,13 @@ bool StunXorAddressAttribute::Write(ByteBufferWriter* buf) const {
   switch (xored_ip.family()) {
     case AF_INET: {
       in_addr v4addr = xored_ip.ipv4_address();
-      buf->Write(ArrayView<const uint8_t>(
+      buf->Write(std::span<const uint8_t>(
           reinterpret_cast<const uint8_t*>(&v4addr), sizeof(v4addr)));
       break;
     }
     case AF_INET6: {
       in6_addr v6addr = xored_ip.ipv6_address();
-      buf->Write(ArrayView<const uint8_t>(
+      buf->Write(std::span<const uint8_t>(
           reinterpret_cast<const uint8_t*>(&v6addr), sizeof(v6addr)));
       break;
     }
@@ -1185,7 +1185,7 @@ void StunByteStringAttribute::SetByte(size_t index, uint8_t value) {
 
 bool StunByteStringAttribute::Read(ByteBufferReader* buf) {
   bytes_ = new uint8_t[length()];
-  if (!buf->ReadBytes(ArrayView<uint8_t>(bytes_, length()))) {
+  if (!buf->ReadBytes(std::span<uint8_t>(bytes_, length()))) {
     return false;
   }
 
@@ -1198,7 +1198,7 @@ bool StunByteStringAttribute::Write(ByteBufferWriter* buf) const {
   if (!LengthValid(type(), length())) {
     return false;
   }
-  buf->Write(ArrayView<const uint8_t>(bytes_, length()));
+  buf->Write(std::span<const uint8_t>(bytes_, length()));
   WritePadding(buf);
   return true;
 }
diff --git a/api/transport/stun.h b/api/transport/stun.h
index 9066a5c654..f9e0a8f104 100644
--- a/api/transport/stun.h
+++ b/api/transport/stun.h
@@ -19,11 +19,11 @@
 #include <functional>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "rtc_base/byte_buffer.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/ip_address.h"
@@ -255,7 +255,7 @@ class StunMessage {
 
   // Verify that a buffer has stun magic cookie and one of the specified
   // methods. Note that it does not check for the existance of FINGERPRINT.
-  static bool IsStunMethod(ArrayView<int> methods,
+  static bool IsStunMethod(std::span<int> methods,
                            const char* data,
                            size_t size);
 
@@ -523,9 +523,7 @@ class StunByteStringAttribute : public StunAttribute {
   }
   // Returns the attribute value as an uint8_t view.
   // Use this function for values that are not text.
-  ArrayView<uint8_t> array_view() const {
-    return MakeArrayView(bytes_, length());
-  }
+  std::span<uint8_t> array_view() const { return std::span(bytes_, length()); }
 
   [[deprecated]] std::string GetString() const {
     return std::string(reinterpret_cast<const char*>(bytes_), length());
diff --git a/api/transport/stun_unittest.cc b/api/transport/stun_unittest.cc
index 1d70f5fd8a..21d26ee7d6 100644
--- a/api/transport/stun_unittest.cc
+++ b/api/transport/stun_unittest.cc
@@ -14,11 +14,11 @@
 #include <cstring>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "rtc_base/byte_buffer.h"
 #include "rtc_base/byte_order.h"
 #include "rtc_base/ip_address.h"
@@ -212,9 +212,9 @@ constexpr uint8_t kRtcpPacket[] = {
   0x00, 0x03, 0x73, 0x50,
 };
 
-const ArrayView<const uint8_t> kZeroLenView(kStunMessageWithZeroLength);
-const ArrayView<const uint8_t> kExcessLenView(kStunMessageWithExcessLength);
-const ArrayView<const uint8_t> kSmallLenView(kStunMessageWithSmallLength);
+const std::span<const uint8_t> kZeroLenView(kStunMessageWithZeroLength);
+const std::span<const uint8_t> kExcessLenView(kStunMessageWithExcessLength);
+const std::span<const uint8_t> kSmallLenView(kStunMessageWithSmallLength);
 
 
 // RFC5769 Test Vectors
@@ -546,7 +546,7 @@ class StunTest : public ::testing::Test {
   size_t ReadStunMessageTestCase(StunMessage* msg,
                                  const uint8_t* testcase,
                                  size_t size) {
-    ByteBufferReader buf(MakeArrayView(testcase, size));
+    ByteBufferReader buf(std::span(testcase, size));
     if (msg->Read(&buf)) {
       // Returns the size the stun message should report itself as being
       return (size - 20);
@@ -1133,7 +1133,7 @@ TEST_F(StunTest, WriteMessageWithAUInt16ListAttribute) {
 // Test that we fail to read messages with invalid lengths.
 void CheckFailureToRead(const uint8_t* testcase, size_t length) {
   StunMessage msg;
-  ByteBufferReader buf(MakeArrayView(testcase, length));
+  ByteBufferReader buf(std::span(testcase, length));
   ASSERT_FALSE(msg.Read(&buf));
 }
 
diff --git a/api/units/BUILD.gn b/api/units/BUILD.gn
index 379d5b5f7b..0c54b3e14e 100644
--- a/api/units/BUILD.gn
+++ b/api/units/BUILD.gn
@@ -19,7 +19,6 @@ rtc_library("data_rate") {
     ":data_size",
     ":frequency",
     ":time_delta",
-    "..:array_view",
     "../../rtc_base:checks",
     "../../rtc_base:stringutils",
     "../../rtc_base/system:rtc_export",
@@ -35,7 +34,6 @@ rtc_library("data_size") {
   ]
 
   deps = [
-    "..:array_view",
     "../../rtc_base:checks",
     "../../rtc_base:stringutils",
     "../../rtc_base/system:rtc_export",
@@ -51,7 +49,6 @@ rtc_library("time_delta") {
   ]
 
   deps = [
-    "..:array_view",
     "../../rtc_base:checks",
     "../../rtc_base:stringutils",
     "../../rtc_base/system:rtc_export",
@@ -68,7 +65,6 @@ rtc_library("frequency") {
 
   deps = [
     ":time_delta",
-    "..:array_view",
     "../../rtc_base:checks",
     "../../rtc_base:stringutils",
     "../../rtc_base/system:rtc_export",
@@ -85,7 +81,6 @@ rtc_library("timestamp") {
 
   deps = [
     ":time_delta",
-    "..:array_view",
     "../../rtc_base:checks",
     "../../rtc_base:stringutils",
     "../../rtc_base/system:rtc_export",
diff --git a/api/units/data_rate.cc b/api/units/data_rate.cc
index 7f99a172ed..5b3838f80f 100644
--- a/api/units/data_rate.cc
+++ b/api/units/data_rate.cc
@@ -12,7 +12,6 @@
 
 #include <string>
 
-#include "api/array_view.h"
 #include "rtc_base/strings/string_builder.h"
 
 namespace webrtc {
diff --git a/api/units/data_size.cc b/api/units/data_size.cc
index abe4906c92..24bb83c866 100644
--- a/api/units/data_size.cc
+++ b/api/units/data_size.cc
@@ -12,7 +12,6 @@
 
 #include <string>
 
-#include "api/array_view.h"
 #include "rtc_base/strings/string_builder.h"
 
 namespace webrtc {
diff --git a/api/units/time_delta.cc b/api/units/time_delta.cc
index efd5a2a518..9e1521bb96 100644
--- a/api/units/time_delta.cc
+++ b/api/units/time_delta.cc
@@ -12,7 +12,6 @@
 
 #include <string>
 
-#include "api/array_view.h"
 #include "rtc_base/strings/string_builder.h"
 
 namespace webrtc {
diff --git a/api/units/timestamp.cc b/api/units/timestamp.cc
index 38b0d1157d..6dd64ab061 100644
--- a/api/units/timestamp.cc
+++ b/api/units/timestamp.cc
@@ -12,7 +12,6 @@
 
 #include <string>
 
-#include "api/array_view.h"
 #include "rtc_base/strings/string_builder.h"
 
 namespace webrtc {
diff --git a/api/video/BUILD.gn b/api/video/BUILD.gn
index d6186040dc..0566d8b205 100644
--- a/api/video/BUILD.gn
+++ b/api/video/BUILD.gn
@@ -27,7 +27,6 @@ rtc_library("video_rtp_headers") {
   ]
 
   deps = [
-    "..:array_view",
     "../../rtc_base:checks",
     "../../rtc_base:logging",
     "../../rtc_base:safe_conversions",
@@ -61,7 +60,6 @@ rtc_library("video_frame") {
 
   deps = [
     ":video_rtp_headers",
-    "..:array_view",
     "..:make_ref_counted",
     "..:ref_count",
     "..:rtp_packet_info",
@@ -117,7 +115,6 @@ rtc_source_set("recordable_encoded_frame") {
     ":encoded_image",
     ":video_frame",
     ":video_rtp_headers",
-    "..:array_view",
     "..:make_ref_counted",
     "..:scoped_refptr",
     "../units:timestamp",
@@ -206,7 +203,6 @@ rtc_library("rtp_video_frame_assembler") {
     ":encoded_image",
     ":video_frame_type",
     ":video_rtp_headers",
-    "..:array_view",
     "..:rtp_packet_info",
     "..:scoped_refptr",
     "../../modules/rtp_rtcp",
@@ -231,7 +227,6 @@ rtc_library("rtp_video_frame_assembler_unittests") {
     ":rtp_video_frame_assembler",
     ":video_frame",
     ":video_frame_type",
-    "..:array_view",
     "../../modules/rtp_rtcp",
     "../../modules/rtp_rtcp:rtp_packetizer_av1_test_helper",
     "../../modules/rtp_rtcp:rtp_rtcp_format",
@@ -253,7 +248,6 @@ if (rtc_use_h265) {
       ":rtp_video_frame_assembler",
       ":video_frame",
       ":video_frame_type",
-      "..:array_view",
       "../../modules/rtp_rtcp",
       "../../modules/rtp_rtcp:rtp_rtcp_format",
       "../../modules/rtp_rtcp:rtp_video_header",
@@ -363,7 +357,6 @@ rtc_library("video_frame_metadata") {
     ":video_frame",
     ":video_frame_type",
     ":video_rtp_headers",
-    "..:array_view",
     "../../modules/video_coding:codec_globals_headers",
     "../../rtc_base/system:rtc_export",
     "../transport/rtp:dependency_descriptor",
@@ -400,7 +393,6 @@ rtc_library("frame_buffer") {
     "frame_buffer.h",
   ]
   deps = [
-    "..:array_view",
     "../../api:field_trials_view",
     "../../api/units:timestamp",
     "../../api/video:encoded_frame",
diff --git a/api/video/corruption_detection/frame_instrumentation_data.cc b/api/video/corruption_detection/frame_instrumentation_data.cc
index ecc076ad4c..d54aa4b7c1 100644
--- a/api/video/corruption_detection/frame_instrumentation_data.cc
+++ b/api/video/corruption_detection/frame_instrumentation_data.cc
@@ -10,11 +10,10 @@
 
 #include "api/video/corruption_detection/frame_instrumentation_data.h"
 
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
-
 namespace webrtc {
 
 constexpr int kMaxSequenceIndex = (1 << 14) - 1;
@@ -61,7 +60,7 @@ bool FrameInstrumentationData::SetChromaErrorThreshold(int threshold) {
 }
 
 bool FrameInstrumentationData::SetSampleValues(
-    webrtc::ArrayView<const double> samples) {
+    std::span<const double> samples) {
   for (double sample_value : samples) {
     if (sample_value < 0.0 || sample_value > 255.0) {
       return false;
diff --git a/api/video/corruption_detection/frame_instrumentation_data.h b/api/video/corruption_detection/frame_instrumentation_data.h
index 8d4dc41f9c..923a981539 100644
--- a/api/video/corruption_detection/frame_instrumentation_data.h
+++ b/api/video/corruption_detection/frame_instrumentation_data.h
@@ -11,10 +11,9 @@
 #ifndef API_VIDEO_CORRUPTION_DETECTION_FRAME_INSTRUMENTATION_DATA_H_
 #define API_VIDEO_CORRUPTION_DETECTION_FRAME_INSTRUMENTATION_DATA_H_
 
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
-
 namespace webrtc {
 
 class FrameInstrumentationData {
@@ -26,14 +25,14 @@ class FrameInstrumentationData {
   double std_dev() const { return std_dev_; }
   int luma_error_threshold() const { return luma_error_threshold_; }
   int chroma_error_threshold() const { return chroma_error_threshold_; }
-  ArrayView<const double> sample_values() const { return sample_values_; }
+  std::span<const double> sample_values() const { return sample_values_; }
 
   bool SetSequenceIndex(int index);
   void set_droppable(bool droppable) { droppable_ = droppable; }
   bool SetStdDev(double std_dev);
   bool SetLumaErrorThreshold(int threshold);
   bool SetChromaErrorThreshold(int threshold);
-  bool SetSampleValues(ArrayView<const double> samples);
+  bool SetSampleValues(std::span<const double> samples);
   bool SetSampleValues(std::vector<double>&& samples);
 
   // Convenience methods..
diff --git a/api/video/corruption_detection/frame_instrumentation_data_reader.cc b/api/video/corruption_detection/frame_instrumentation_data_reader.cc
index 760feadbbd..b025d25f30 100644
--- a/api/video/corruption_detection/frame_instrumentation_data_reader.cc
+++ b/api/video/corruption_detection/frame_instrumentation_data_reader.cc
@@ -11,8 +11,8 @@
 #include "api/video/corruption_detection/frame_instrumentation_data_reader.h"
 
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/transport/rtp/corruption_detection_message.h"
 #include "api/video/corruption_detection/frame_instrumentation_data.h"
 #include "rtc_base/logging.h"
@@ -45,7 +45,7 @@ FrameInstrumentationDataReader::ParseMessage(
   // The sequence index field of the message refers to the halton sequence index
   // for the first sample in the message. In order to figure out the next
   // expected sequence index we must increment it by the number of samples.
-  ArrayView<const double> sample_values = message.sample_values();
+  std::span<const double> sample_values = message.sample_values();
   last_seen_sequence_index_ = data.sequence_index() + sample_values.size();
 
   if (!sample_values.empty()) {
diff --git a/api/video/corruption_detection/frame_instrumentation_evaluation.cc b/api/video/corruption_detection/frame_instrumentation_evaluation.cc
index a3e271d455..d1beb7cac5 100644
--- a/api/video/corruption_detection/frame_instrumentation_evaluation.cc
+++ b/api/video/corruption_detection/frame_instrumentation_evaluation.cc
@@ -12,9 +12,9 @@
 
 #include <cstddef>
 #include <memory>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/video/corruption_detection/frame_instrumentation_data.h"
 #include "api/video/video_content_type.h"
 #include "api/video/video_frame.h"
@@ -28,8 +28,8 @@ namespace webrtc {
 namespace {
 
 std::vector<FilteredSample> ConvertSampleValuesToFilteredSamples(
-    ArrayView<const double> values,
-    ArrayView<const FilteredSample> samples) {
+    std::span<const double> values,
+    std::span<const FilteredSample> samples) {
   RTC_CHECK_EQ(values.size(), samples.size())
       << "values and samples must have the same size";
   std::vector<FilteredSample> filtered_samples;
diff --git a/api/video/frame_buffer.cc b/api/video/frame_buffer.cc
index 78426db4f6..fcff81f41c 100644
--- a/api/video/frame_buffer.cc
+++ b/api/video/frame_buffer.cc
@@ -16,11 +16,11 @@
 #include <iterator>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 
 #include "absl/algorithm/container.h"
 #include "absl/container/inlined_vector.h"
-#include "api/array_view.h"
 #include "api/field_trials_view.h"
 #include "api/video/encoded_frame.h"
 #include "rtc_base/logging.h"
@@ -46,7 +46,7 @@ bool ValidReferences(const EncodedFrame& frame) {
 // Since FrameBuffer::FrameInfo is private it can't be used in the function
 // signature, hence the FrameIteratorT type.
 template <typename FrameIteratorT>
-ArrayView<const int64_t> GetReferences(const FrameIteratorT& it) {
+std::span<const int64_t> GetReferences(const FrameIteratorT& it) {
   return {it->second.encoded_frame->references,
           std::min<size_t>(it->second.encoded_frame->num_references,
                            EncodedFrame::kMaxFrameReferences)};
diff --git a/api/video/rtp_video_frame_assembler.cc b/api/video/rtp_video_frame_assembler.cc
index e7bc37aee5..821e529268 100644
--- a/api/video/rtp_video_frame_assembler.cc
+++ b/api/video/rtp_video_frame_assembler.cc
@@ -13,11 +13,11 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
 #include "absl/container/inlined_vector.h"
-#include "api/array_view.h"
 #include "api/rtp_packet_infos.h"
 #include "api/scoped_refptr.h"
 #include "api/transport/rtp/dependency_descriptor.h"
@@ -161,7 +161,7 @@ RtpVideoFrameAssembler::Impl::RtpFrameVector
 RtpVideoFrameAssembler::Impl::AssembleFrames(
     video_coding::PacketBuffer::InsertResult insert_result) {
   video_coding::PacketBuffer::Packet* first_packet = nullptr;
-  std::vector<ArrayView<const uint8_t>> payloads;
+  std::vector<std::span<const uint8_t>> payloads;
   RtpFrameVector result;
 
   for (auto& packet : insert_result.packets) {
diff --git a/api/video/rtp_video_frame_assembler_unittests.cc b/api/video/rtp_video_frame_assembler_unittests.cc
index 826ae8ae3b..e5c0dba2ea 100644
--- a/api/video/rtp_video_frame_assembler_unittests.cc
+++ b/api/video/rtp_video_frame_assembler_unittests.cc
@@ -11,9 +11,9 @@
 #include <cstdint>
 #include <iterator>
 #include <memory>
+#include <span>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/transport/rtp/dependency_descriptor.h"
 #include "api/video/encoded_frame.h"
 #include "api/video/rtp_video_frame_assembler.h"
@@ -54,7 +54,7 @@ class PacketBuilder {
     return *this;
   }
 
-  PacketBuilder& WithPayload(ArrayView<const uint8_t> payload) {
+  PacketBuilder& WithPayload(std::span<const uint8_t> payload) {
     payload_.assign(payload.begin(), payload.end());
     return *this;
   }
@@ -131,11 +131,11 @@ void AppendFrames(RtpVideoFrameAssembler::FrameVector from,
             std::make_move_iterator(from.end()));
 }
 
-ArrayView<int64_t> References(const std::unique_ptr<EncodedFrame>& frame) {
-  return MakeArrayView(frame->references, frame->num_references);
+std::span<int64_t> References(const std::unique_ptr<EncodedFrame>& frame) {
+  return std::span(frame->references, frame->num_references);
 }
 
-ArrayView<const uint8_t> Payload(const std::unique_ptr<EncodedFrame>& frame) {
+std::span<const uint8_t> Payload(const std::unique_ptr<EncodedFrame>& frame) {
   return *frame->GetEncodedData();
 }
 
diff --git a/api/video/rtp_video_frame_h265_assembler_unittests.cc b/api/video/rtp_video_frame_h265_assembler_unittests.cc
index aff5c8a74d..db823076d4 100644
--- a/api/video/rtp_video_frame_h265_assembler_unittests.cc
+++ b/api/video/rtp_video_frame_h265_assembler_unittests.cc
@@ -11,10 +11,10 @@
 #include <cstdint>
 #include <iterator>
 #include <memory>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/video/encoded_frame.h"
 #include "api/video/rtp_video_frame_assembler.h"
 #include "api/video/video_frame_type.h"
@@ -44,7 +44,7 @@ class PacketBuilder {
     return *this;
   }
 
-  PacketBuilder& WithPayload(ArrayView<const uint8_t> payload) {
+  PacketBuilder& WithPayload(std::span<const uint8_t> payload) {
     payload_.assign(payload.begin(), payload.end());
     return *this;
   }
@@ -87,11 +87,11 @@ void AppendFrames(RtpVideoFrameAssembler::FrameVector&& from,
             std::make_move_iterator(from.end()));
 }
 
-ArrayView<int64_t> References(const std::unique_ptr<EncodedFrame>& frame) {
-  return MakeArrayView(frame->references, frame->num_references);
+std::span<int64_t> References(const std::unique_ptr<EncodedFrame>& frame) {
+  return std::span(frame->references, frame->num_references);
 }
 
-ArrayView<const uint8_t> Payload(const std::unique_ptr<EncodedFrame>& frame) {
+std::span<const uint8_t> Payload(const std::unique_ptr<EncodedFrame>& frame) {
   return *frame->GetEncodedData();
 }
 
diff --git a/api/video/video_frame_buffer.cc b/api/video/video_frame_buffer.cc
index 7b45658791..131aa3e647 100644
--- a/api/video/video_frame_buffer.cc
+++ b/api/video/video_frame_buffer.cc
@@ -11,9 +11,9 @@
 #include "api/video/video_frame_buffer.h"
 
 #include <cstddef>
+#include <span>
 #include <string>
 
-#include "api/array_view.h"
 #include "api/scoped_refptr.h"
 #include "api/video/i420_buffer.h"
 #include "api/video/i422_buffer.h"
@@ -89,7 +89,7 @@ const NV12BufferInterface* VideoFrameBuffer::GetNV12() const {
 }
 
 scoped_refptr<VideoFrameBuffer> VideoFrameBuffer::GetMappedFrameBuffer(
-    ArrayView<Type> /* types */) {
+    std::span<Type> /* types */) {
   RTC_CHECK(type() == Type::kNative);
   return nullptr;
 }
diff --git a/api/video/video_frame_buffer.h b/api/video/video_frame_buffer.h
index b160ab8bb7..5545239458 100644
--- a/api/video/video_frame_buffer.h
+++ b/api/video/video_frame_buffer.h
@@ -13,9 +13,9 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <span>
 #include <string>
 
-#include "api/array_view.h"
 #include "api/ref_count.h"
 #include "api/scoped_refptr.h"
 #include "rtc_base/system/rtc_export.h"
@@ -129,7 +129,7 @@ class RTC_EXPORT VideoFrameBuffer : public RefCountInterface {
   // frame type is not supported, mapping is not possible, or if the kNative
   // frame has not implemented this method. Only callable if type() is kNative.
   virtual scoped_refptr<VideoFrameBuffer> GetMappedFrameBuffer(
-      ArrayView<Type> types);
+      std::span<Type> types);
 
   // For logging: returns a textual representation of the storage.
   virtual std::string storage_representation() const;
diff --git a/api/video/video_frame_metadata.cc b/api/video/video_frame_metadata.cc
index f265fc335a..4441b98fac 100644
--- a/api/video/video_frame_metadata.cc
+++ b/api/video/video_frame_metadata.cc
@@ -12,10 +12,10 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/transport/rtp/dependency_descriptor.h"
 #include "api/video/video_codec_type.h"
 #include "api/video/video_content_type.h"
@@ -90,23 +90,23 @@ void VideoFrameMetadata::SetTemporalIndex(int temporal_index) {
   temporal_index_ = temporal_index;
 }
 
-ArrayView<const int64_t> VideoFrameMetadata::GetFrameDependencies() const {
+std::span<const int64_t> VideoFrameMetadata::GetFrameDependencies() const {
   return frame_dependencies_;
 }
 
 void VideoFrameMetadata::SetFrameDependencies(
-    ArrayView<const int64_t> frame_dependencies) {
+    std::span<const int64_t> frame_dependencies) {
   frame_dependencies_.assign(frame_dependencies.begin(),
                              frame_dependencies.end());
 }
 
-ArrayView<const DecodeTargetIndication>
+std::span<const DecodeTargetIndication>
 VideoFrameMetadata::GetDecodeTargetIndications() const {
   return decode_target_indications_;
 }
 
 void VideoFrameMetadata::SetDecodeTargetIndications(
-    ArrayView<const DecodeTargetIndication> decode_target_indications) {
+    std::span<const DecodeTargetIndication> decode_target_indications) {
   decode_target_indications_.assign(decode_target_indications.begin(),
                                     decode_target_indications.end());
 }
diff --git a/api/video/video_frame_metadata.h b/api/video/video_frame_metadata.h
index 342d69b4f9..1dfd1bb62d 100644
--- a/api/video/video_frame_metadata.h
+++ b/api/video/video_frame_metadata.h
@@ -13,11 +13,11 @@
 
 #include <cstdint>
 #include <optional>
+#include <span>
 #include <variant>
 #include <vector>
 
 #include "absl/container/inlined_vector.h"
-#include "api/array_view.h"
 #include "api/transport/rtp/dependency_descriptor.h"
 #include "api/video/video_codec_type.h"
 #include "api/video/video_content_type.h"
@@ -67,12 +67,12 @@ class RTC_EXPORT VideoFrameMetadata {
   int GetTemporalIndex() const;
   void SetTemporalIndex(int temporal_index);
 
-  ArrayView<const int64_t> GetFrameDependencies() const;
-  void SetFrameDependencies(ArrayView<const int64_t> frame_dependencies);
+  std::span<const int64_t> GetFrameDependencies() const;
+  void SetFrameDependencies(std::span<const int64_t> frame_dependencies);
 
-  ArrayView<const DecodeTargetIndication> GetDecodeTargetIndications() const;
+  std::span<const DecodeTargetIndication> GetDecodeTargetIndications() const;
   void SetDecodeTargetIndications(
-      ArrayView<const DecodeTargetIndication> decode_target_indications);
+      std::span<const DecodeTargetIndication> decode_target_indications);
 
   bool GetIsLastFrameInPicture() const;
   void SetIsLastFrameInPicture(bool is_last_frame_in_picture);
diff --git a/api/video/video_timing.cc b/api/video/video_timing.cc
index a5b7f78dc9..bc9ee808ad 100644
--- a/api/video/video_timing.cc
+++ b/api/video/video_timing.cc
@@ -14,7 +14,6 @@
 #include <cstdint>
 #include <string>
 
-#include "api/array_view.h"
 #include "api/units/time_delta.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/numerics/safe_conversions.h"
diff --git a/api/video_codecs/BUILD.gn b/api/video_codecs/BUILD.gn
index 2b16682954..1ec3dff92c 100644
--- a/api/video_codecs/BUILD.gn
+++ b/api/video_codecs/BUILD.gn
@@ -81,7 +81,6 @@ rtc_library("video_codecs_api") {
     ":scalability_mode",
     "..:fec_controller_api",
     "..:scoped_refptr",
-    "../../api:array_view",
     "../../api:rtp_parameters",
     "../../media:media_constants",
     "../../modules/video_coding:codec_globals_headers",
@@ -122,7 +121,7 @@ rtc_library("encoder_speed_controller_factory") {
 rtc_source_set("bitstream_parser_api") {
   visibility = [ "*" ]
   sources = [ "bitstream_parser.h" ]
-  deps = [ "..:array_view" ]
+  deps = []
 }
 
 rtc_library("builtin_video_decoder_factory") {
@@ -169,7 +168,6 @@ rtc_source_set("video_encoder_factory_template") {
   deps = [
     ":scalability_mode",
     ":video_codecs_api",
-    "..:array_view",
     "../../modules/video_coding/svc:scalability_mode_util",
     "../environment",
     "//third_party/abseil-cpp/absl/algorithm:container",
@@ -240,7 +238,6 @@ rtc_source_set("video_decoder_factory_template") {
 
   deps = [
     ":video_codecs_api",
-    "..:array_view",
     "../environment",
     "//third_party/abseil-cpp/absl/algorithm:container",
   ]
@@ -301,7 +298,6 @@ rtc_source_set("video_encoder_interface") {
 
   deps = [
     ":video_encoding_general",
-    "..:array_view",
     "..:scoped_refptr",
     "../../api/units:data_rate",
     "../../api/units:data_size",
@@ -338,7 +334,6 @@ rtc_library("simple_encoder_wrapper") {
   deps = [
     ":video_encoder_factory_interface",
     ":video_encoder_interface",
-    "..:array_view",
     "..:scoped_refptr",
     "../../api/units:data_rate",
     "../../api/video_codecs:scalability_mode",
@@ -405,7 +400,6 @@ rtc_library("libaom_av1_encoder_factory_test") {
     ":video_encoder_factory_interface",
     ":video_encoder_interface",
     ":video_encoding_general",
-    "..:array_view",
     "..:scoped_refptr",
     "../../api/video:video_frame",
     "../../api/video_codecs:video_codecs_api",
diff --git a/api/video_codecs/bitstream_parser.h b/api/video_codecs/bitstream_parser.h
index a7dd998f5e..85587c61dd 100644
--- a/api/video_codecs/bitstream_parser.h
+++ b/api/video_codecs/bitstream_parser.h
@@ -15,8 +15,7 @@
 #include <stdint.h>
 
 #include <optional>
-
-#include "api/array_view.h"
+#include <span>
 
 namespace webrtc {
 
@@ -26,7 +25,7 @@ class BitstreamParser {
   virtual ~BitstreamParser() = default;
 
   // Parse an additional chunk of the bitstream.
-  virtual void ParseBitstream(ArrayView<const uint8_t> bitstream) = 0;
+  virtual void ParseBitstream(std::span<const uint8_t> bitstream) = 0;
 
   // Get the last extracted QP value from the parsed bitstream. If no QP
   // value could be parsed, returns std::nullopt.
diff --git a/api/video_codecs/libaom_av1_encoder_factory_test.cc b/api/video_codecs/libaom_av1_encoder_factory_test.cc
index 27189fc9e7..8d58b5184d 100644
--- a/api/video_codecs/libaom_av1_encoder_factory_test.cc
+++ b/api/video_codecs/libaom_av1_encoder_factory_test.cc
@@ -15,12 +15,12 @@
 #include <memory>
 #include <optional>
 #include <ostream>
+#include <span>
 #include <string>
 #include <utility>
 #include <variant>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/scoped_refptr.h"
 #include "api/units/data_rate.h"
 #include "api/units/data_size.h"
@@ -98,7 +98,7 @@ class Av1Decoder : public DecodedImageCallback {
     return 0;
   }
 
-  VideoFrame Decode(ArrayView<uint8_t> bitstream_data) {
+  VideoFrame Decode(std::span<uint8_t> bitstream_data) {
     EncodedImage img;
     img.SetEncodedData(EncodedImageBuffer::Create(bitstream_data.data(),
                                                   bitstream_data.size()));
@@ -127,7 +127,7 @@ class FrameEncoderSettingsBuilder {
   FrameEncoderSettingsBuilder() {
     class IgnoredOutput : public VideoEncoderInterface::FrameOutput {
      public:
-      ArrayView<uint8_t> GetBitstreamOutputBuffer(DataSize size) override {
+      std::span<uint8_t> GetBitstreamOutputBuffer(DataSize size) override {
         unread_.resize(size.bytes());
         return unread_;
       }
@@ -203,9 +203,9 @@ class FrameEncoderSettingsBuilder {
  private:
   struct FrameOut : public VideoEncoderInterface::FrameOutput {
     explicit FrameOut(EncOut& e) : eo(e) {}
-    ArrayView<uint8_t> GetBitstreamOutputBuffer(DataSize size) override {
+    std::span<uint8_t> GetBitstreamOutputBuffer(DataSize size) override {
       eo.bitstream.resize(size.bytes());
-      return ArrayView<uint8_t>(eo.bitstream);
+      return std::span<uint8_t>(eo.bitstream);
     }
     void EncodeComplete(const EncodeResult& encode_result) override {
       eo.res = encode_result;
diff --git a/api/video_codecs/sdp_video_format.cc b/api/video_codecs/sdp_video_format.cc
index dcc966e6af..57cdbec1d1 100644
--- a/api/video_codecs/sdp_video_format.cc
+++ b/api/video_codecs/sdp_video_format.cc
@@ -11,11 +11,11 @@
 #include "api/video_codecs/sdp_video_format.h"
 
 #include <optional>
+#include <span>
 #include <string>
 
 #include "absl/container/inlined_vector.h"
 #include "absl/strings/match.h"
-#include "api/array_view.h"
 #include "api/rtp_parameters.h"
 #include "api/video/video_codec_type.h"
 #include "api/video_codecs/av1_profile.h"
@@ -172,7 +172,7 @@ bool SdpVideoFormat::IsSameCodec(const SdpVideoFormat& other) const {
 }
 
 bool SdpVideoFormat::IsCodecInList(
-    ArrayView<const SdpVideoFormat> formats) const {
+    std::span<const SdpVideoFormat> formats) const {
   for (const auto& format : formats) {
     if (IsSameCodec(format)) {
       return true;
@@ -245,7 +245,7 @@ const SdpVideoFormat SdpVideoFormat::AV1Profile1() {
 }
 
 std::optional<SdpVideoFormat> FuzzyMatchSdpVideoFormat(
-    ArrayView<const SdpVideoFormat> supported_formats,
+    std::span<const SdpVideoFormat> supported_formats,
     const SdpVideoFormat& format) {
   std::optional<SdpVideoFormat> res;
   int best_parameter_match = 0;
diff --git a/api/video_codecs/sdp_video_format.h b/api/video_codecs/sdp_video_format.h
index 494d1df856..37f03aad5c 100644
--- a/api/video_codecs/sdp_video_format.h
+++ b/api/video_codecs/sdp_video_format.h
@@ -13,10 +13,10 @@
 
 #include <map>
 #include <optional>
+#include <span>
 #include <string>
 
 #include "absl/container/inlined_vector.h"
-#include "api/array_view.h"
 #include "api/rtp_parameters.h"
 #include "api/video_codecs/scalability_mode.h"
 #include "rtc_base/system/rtc_export.h"
@@ -55,7 +55,7 @@ struct RTC_EXPORT SdpVideoFormat {
   // specific parameters. Please note that two SdpVideoFormats can represent the
   // same codec even though not all parameters are the same.
   bool IsSameCodec(const SdpVideoFormat& other) const;
-  bool IsCodecInList(ArrayView<const SdpVideoFormat> formats) const;
+  bool IsCodecInList(std::span<const SdpVideoFormat> formats) const;
 
   std::string ToString() const;
 
@@ -92,7 +92,7 @@ struct RTC_EXPORT SdpVideoFormat {
 // anymore. Until we stop misusing SdpVideoFormats provide this convenience
 // function to perform fuzzy matching.
 std::optional<SdpVideoFormat> FuzzyMatchSdpVideoFormat(
-    ArrayView<const SdpVideoFormat> supported_formats,
+    std::span<const SdpVideoFormat> supported_formats,
     const SdpVideoFormat& format);
 
 }  // namespace webrtc
diff --git a/api/video_codecs/simple_encoder_wrapper.cc b/api/video_codecs/simple_encoder_wrapper.cc
index 90fb4ce740..cb5ddced44 100644
--- a/api/video_codecs/simple_encoder_wrapper.cc
+++ b/api/video_codecs/simple_encoder_wrapper.cc
@@ -15,6 +15,7 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <utility>
 #include <variant>
@@ -22,7 +23,6 @@
 
 #include "absl/algorithm/container.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/scoped_refptr.h"
 #include "api/units/data_size.h"
 #include "api/units/frequency.h"
@@ -192,7 +192,7 @@ void SimpleEncoderWrapper::Encode(scoped_refptr<VideoFrameBuffer> frame_buffer,
     }
 
     struct FrameOut : public VideoEncoderInterface::FrameOutput {
-      ArrayView<uint8_t> GetBitstreamOutputBuffer(DataSize size) override {
+      std::span<uint8_t> GetBitstreamOutputBuffer(DataSize size) override {
         bitstream.resize(size.bytes());
         return bitstream;
       }
diff --git a/api/video_codecs/video_decoder_factory_template.h b/api/video_codecs/video_decoder_factory_template.h
index fba0cec496..45eeb0f8e2 100644
--- a/api/video_codecs/video_decoder_factory_template.h
+++ b/api/video_codecs/video_decoder_factory_template.h
@@ -12,11 +12,11 @@
 #define API_VIDEO_CODECS_VIDEO_DECODER_FACTORY_TEMPLATE_H_
 
 #include <memory>
+#include <span>
 #include <type_traits>
 #include <vector>
 
 #include "absl/algorithm/container.h"
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/video_codecs/sdp_video_format.h"
 #include "api/video_codecs/video_decoder.h"
@@ -53,7 +53,7 @@ class VideoDecoderFactoryTemplate : public VideoDecoderFactory {
 
  private:
   bool IsFormatInList(const SdpVideoFormat& format,
-                      ArrayView<const SdpVideoFormat> supported_formats) const {
+                      std::span<const SdpVideoFormat> supported_formats) const {
     return absl::c_any_of(
         supported_formats, [&](const SdpVideoFormat& supported_format) {
           return supported_format.name == format.name &&
diff --git a/api/video_codecs/video_encoder_factory_template.h b/api/video_codecs/video_encoder_factory_template.h
index f58a36ed87..938a73b844 100644
--- a/api/video_codecs/video_encoder_factory_template.h
+++ b/api/video_codecs/video_encoder_factory_template.h
@@ -13,11 +13,11 @@
 
 #include <memory>
 #include <optional>
+#include <span>
 #include <string>
 #include <vector>
 
 #include "absl/algorithm/container.h"
-#include "api/array_view.h"
 #include "api/environment/environment.h"
 #include "api/video_codecs/scalability_mode.h"
 #include "api/video_codecs/sdp_video_format.h"
@@ -76,7 +76,7 @@ class VideoEncoderFactoryTemplate : public VideoEncoderFactory {
 
  private:
   bool IsFormatInList(const SdpVideoFormat& format,
-                      ArrayView<const SdpVideoFormat> supported_formats) const {
+                      std::span<const SdpVideoFormat> supported_formats) const {
     return absl::c_any_of(
         supported_formats, [&](const SdpVideoFormat& supported_format) {
           return supported_format.name == format.name &&
diff --git a/api/video_codecs/video_encoder_interface.h b/api/video_codecs/video_encoder_interface.h
index 2e99d50770..5a3cb77217 100644
--- a/api/video_codecs/video_encoder_interface.h
+++ b/api/video_codecs/video_encoder_interface.h
@@ -14,10 +14,10 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
+#include <span>
 #include <variant>
 #include <vector>
 
-#include "api/array_view.h"
 #include "api/scoped_refptr.h"
 #include "api/units/data_rate.h"
 #include "api/units/data_size.h"
@@ -43,7 +43,7 @@ class VideoEncoderInterface {
 
   struct FrameOutput {
     virtual ~FrameOutput() = default;
-    virtual ArrayView<uint8_t> GetBitstreamOutputBuffer(DataSize size) = 0;
+    virtual std::span<uint8_t> GetBitstreamOutputBuffer(DataSize size) = 0;
     virtual void EncodeComplete(const EncodeResult& encode_result) = 0;
   };
 
diff --git a/api/voip/BUILD.gn b/api/voip/BUILD.gn
index b0ae871f41..0d3e7b7da8 100644
--- a/api/voip/BUILD.gn
+++ b/api/voip/BUILD.gn
@@ -20,7 +20,6 @@ rtc_source_set("voip_api") {
     "voip_volume_control.h",
   ]
   deps = [
-    "..:array_view",
     "../audio_codecs:audio_codecs_api",
     "../neteq:neteq_api",
     "//third_party/abseil-cpp/absl/base:core_headers",
@@ -56,7 +55,6 @@ if (rtc_include_tests) {
     sources = [ "test/mock_voip_engine.h" ]
     deps = [
       ":voip_api",
-      "..:array_view",
       "../../test:test_support",
       "../audio_codecs:audio_codecs_api",
     ]
diff --git a/api/voip/test/mock_voip_engine.h b/api/voip/test/mock_voip_engine.h
index 97006a5928..779824e28a 100644
--- a/api/voip/test/mock_voip_engine.h
+++ b/api/voip/test/mock_voip_engine.h
@@ -14,8 +14,8 @@
 #include <cstdint>
 #include <map>
 #include <optional>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/voip/voip_base.h"
 #include "api/voip/voip_codec.h"
@@ -69,11 +69,11 @@ class MockVoipNetwork : public VoipNetwork {
  public:
   MOCK_METHOD(VoipResult,
               ReceivedRTPPacket,
-              (ChannelId channel_id, ArrayView<const uint8_t> rtp_packet),
+              (ChannelId channel_id, std::span<const uint8_t> rtp_packet),
               (override));
   MOCK_METHOD(VoipResult,
               ReceivedRTCPPacket,
-              (ChannelId channel_id, ArrayView<const uint8_t> rtcp_packet),
+              (ChannelId channel_id, std::span<const uint8_t> rtcp_packet),
               (override));
 };
 
diff --git a/api/voip/voip_network.h b/api/voip/voip_network.h
index b239c7eaf8..7760891d78 100644
--- a/api/voip/voip_network.h
+++ b/api/voip/voip_network.h
@@ -12,8 +12,8 @@
 #define API_VOIP_VOIP_NETWORK_H_
 
 #include <cstdint>
+#include <span>
 
-#include "api/array_view.h"
 #include "api/voip/voip_base.h"
 
 namespace webrtc {
@@ -28,7 +28,7 @@ class VoipNetwork {
   //  kOk - received RTP packet is processed.
   //  kInvalidArgument - `channel_id` is invalid.
   virtual VoipResult ReceivedRTPPacket(ChannelId channel_id,
-                                       ArrayView<const uint8_t> rtp_packet) = 0;
+                                       std::span<const uint8_t> rtp_packet) = 0;
 
   // The data received from the network including RTCP header is passed here.
   // Returns following VoipResult;
@@ -36,7 +36,7 @@ class VoipNetwork {
   //  kInvalidArgument - `channel_id` is invalid.
   virtual VoipResult ReceivedRTCPPacket(
       ChannelId channel_id,
-      ArrayView<const uint8_t> rtcp_packet) = 0;
+      std::span<const uint8_t> rtcp_packet) = 0;
 
  protected:
   virtual ~VoipNetwork() = default;
diff --git a/api/webrtc_sdp_unittest.cc b/api/webrtc_sdp_unittest.cc
index cc58f50bce..d121d8d36f 100644
--- a/api/webrtc_sdp_unittest.cc
+++ b/api/webrtc_sdp_unittest.cc
@@ -25,7 +25,6 @@
 #include "absl/strings/str_cat.h"
 #include "absl/strings/str_replace.h"
 #include "absl/strings/string_view.h"
-#include "api/array_view.h"
 #include "api/candidate.h"
 #include "api/jsep.h"
 #include "api/media_types.h"

From 2b7dcedf3360a81d9edb061c2e6e3a8b458c0f00 Mon Sep 17 00:00:00 2001
From: Evan Shrubsole <eshr@webrtc.org>
Date: Tue, 31 Mar 2026 08:40:31 +0000
Subject: [PATCH 463/472] Use RunLoop in audio/ tests
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Bug: webrtc:469327588
Change-Id: I6f7a6f3d88184aa38f5a8597fc29acc56a6a6964
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/461346
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47316}
---
 ...ive_frame_transformer_delegate_unittest.cc | 19 ++++++++++---------
 1 file changed, 10 insertions(+), 9 deletions(-)

diff --git a/audio/channel_receive_frame_transformer_delegate_unittest.cc b/audio/channel_receive_frame_transformer_delegate_unittest.cc
index 01dbaee486..986cddf5ea 100644
--- a/audio/channel_receive_frame_transformer_delegate_unittest.cc
+++ b/audio/channel_receive_frame_transformer_delegate_unittest.cc
@@ -28,6 +28,7 @@
 #include "system_wrappers/include/ntp_time.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/run_loop.h"
 
 namespace webrtc {
 namespace {
@@ -87,7 +88,7 @@ TEST(ChannelReceiveFrameTransformerDelegateTest,
 // transformer, it passes it to the channel using the ReceiveFrameCallback.
 TEST(ChannelReceiveFrameTransformerDelegateTest,
      TransformRunsChannelReceiveCallback) {
-  AutoThread main_thread;
+  test::RunLoop main_thread;
   scoped_refptr<MockFrameTransformer> mock_frame_transformer =
       make_ref_counted<NiceMock<MockFrameTransformer>>();
   MockChannelReceive mock_channel;
@@ -118,7 +119,7 @@ TEST(ChannelReceiveFrameTransformerDelegateTest,
 // transformer, it passes it to the channel using the ReceiveFrameCallback.
 TEST(ChannelReceiveFrameTransformerDelegateTest,
      TransformRunsChannelReceiveCallbackForSenderFrame) {
-  AutoThread main_thread;
+  test::RunLoop main_thread;
   scoped_refptr<MockFrameTransformer> mock_frame_transformer =
       make_ref_counted<NiceMock<MockFrameTransformer>>();
   MockChannelReceive mock_channel;
@@ -153,7 +154,7 @@ TEST(ChannelReceiveFrameTransformerDelegateTest,
 // after resetting the delegate.
 TEST(ChannelReceiveFrameTransformerDelegateTest,
      OnTransformedDoesNotRunChannelReceiveCallbackAfterReset) {
-  AutoThread main_thread;
+  test::RunLoop main_thread;
   scoped_refptr<MockFrameTransformer> mock_frame_transformer =
       make_ref_counted<testing::NiceMock<MockFrameTransformer>>();
   MockChannelReceive mock_channel;
@@ -169,7 +170,7 @@ TEST(ChannelReceiveFrameTransformerDelegateTest,
 
 TEST(ChannelReceiveFrameTransformerDelegateTest,
      ShortCircuitingSkipsTransform) {
-  AutoThread main_thread;
+  test::RunLoop main_thread;
   scoped_refptr<MockFrameTransformer> mock_frame_transformer =
       make_ref_counted<testing::NiceMock<MockFrameTransformer>>();
   MockChannelReceive mock_channel;
@@ -193,7 +194,7 @@ TEST(ChannelReceiveFrameTransformerDelegateTest,
 
 TEST(ChannelReceiveFrameTransformerDelegateTest,
      AudioLevelAndCaptureTimeAbsentWithoutExtension) {
-  AutoThread main_thread;
+  test::RunLoop main_thread;
   scoped_refptr<MockFrameTransformer> mock_frame_transformer =
       make_ref_counted<NiceMock<MockFrameTransformer>>();
   scoped_refptr<ChannelReceiveFrameTransformerDelegate> delegate =
@@ -230,7 +231,7 @@ TEST(ChannelReceiveFrameTransformerDelegateTest,
 
 TEST(ChannelReceiveFrameTransformerDelegateTest,
      AudioLevelPresentWithExtension) {
-  AutoThread main_thread;
+  test::RunLoop main_thread;
   scoped_refptr<MockFrameTransformer> mock_frame_transformer =
       make_ref_counted<NiceMock<MockFrameTransformer>>();
   scoped_refptr<ChannelReceiveFrameTransformerDelegate> delegate =
@@ -268,7 +269,7 @@ TEST(ChannelReceiveFrameTransformerDelegateTest,
 
 TEST(ChannelReceiveFrameTransformerDelegateTest,
      CaptureTimePresentWithExtension) {
-  AutoThread main_thread;
+  test::RunLoop main_thread;
   scoped_refptr<MockFrameTransformer> mock_frame_transformer =
       make_ref_counted<NiceMock<MockFrameTransformer>>();
   scoped_refptr<ChannelReceiveFrameTransformerDelegate> delegate =
@@ -311,7 +312,7 @@ TEST(ChannelReceiveFrameTransformerDelegateTest,
 }
 
 TEST(ChannelReceiveFrameTransformerDelegateTest, SetAudioLevel) {
-  AutoThread main_thread;
+  test::RunLoop main_thread;
   scoped_refptr<MockFrameTransformer> mock_frame_transformer =
       make_ref_counted<NiceMock<MockFrameTransformer>>();
   scoped_refptr<ChannelReceiveFrameTransformerDelegate> delegate =
@@ -346,7 +347,7 @@ TEST(ChannelReceiveFrameTransformerDelegateTest, SetAudioLevel) {
 
 TEST(ChannelReceiveFrameTransformerDelegateTest,
      ReceivingSenderFrameWithAudioValueSetsAudioLevelInHeader) {
-  AutoThread main_thread;
+  test::RunLoop main_thread;
   scoped_refptr<MockFrameTransformer> mock_frame_transformer =
       make_ref_counted<NiceMock<MockFrameTransformer>>();
   MockChannelReceive mock_channel;

From 2432b699726449382e1870409f3e689d6d6b2c42 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Thu, 2 Apr 2026 09:59:48 +0200
Subject: [PATCH 464/472] Capture channel pointers on signaling thread to fix
 data race

The stats collector previously accessed channel getters directly from
transceivers during the collection process. This created a data race if
a transceiver's channel was cleared on the signaling thread while the
collection was still ongoing/queued on the worker thread.

This change introduces a callback-based mechanism for retrieving media
channel statistics and RTP parameters. By capturing these as
absl::AnyInvocable objects on the signaling thread at the start of the
stats collection process, the collector can safely execute them later on
the worker thread.

Bug: webrtc:481443652
Change-Id: Ia8ee334df710bc8655e2617e079bf200ab04173e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/461260
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47317}
---
 media/base/fake_media_engine.cc               |  41 +++++++
 media/base/fake_media_engine.h                |  13 +++
 media/base/media_channel.h                    |  40 +++++++
 media/engine/webrtc_video_engine.cc           |  40 +++++++
 media/engine/webrtc_video_engine.h            |   7 ++
 media/engine/webrtc_voice_engine.cc           |  41 +++++++
 media/engine/webrtc_voice_engine.h            |   6 +
 pc/rtc_stats_collector.cc                     | 103 ++++++------------
 pc/rtc_stats_collector.h                      |   8 ++
 pc/rtc_stats_collector_unittest.cc            |  65 +++++++++++
 pc/test/fake_peer_connection_for_stats.h      |  66 +++++++++++
 ...ck_voice_media_receive_channel_interface.h |   5 +
 12 files changed, 367 insertions(+), 68 deletions(-)

diff --git a/media/base/fake_media_engine.cc b/media/base/fake_media_engine.cc
index cad3c587ed..2603c7b4bc 100644
--- a/media/base/fake_media_engine.cc
+++ b/media/base/fake_media_engine.cc
@@ -20,6 +20,7 @@
 #include <utility>
 #include <vector>
 
+#include "absl/functional/any_invocable.h"
 #include "absl/strings/match.h"
 #include "api/audio/audio_device.h"
 #include "api/audio_options.h"
@@ -173,6 +174,16 @@ bool FakeVoiceMediaReceiveChannel::GetStats(
     bool /* get_and_clear_legacy_stats */) {
   return false;
 }
+absl::AnyInvocable<std::optional<VoiceMediaReceiveInfo>()>
+FakeVoiceMediaReceiveChannel::GetStatsCallback(bool reset_legacy) {
+  return [this, reset_legacy]() -> std::optional<VoiceMediaReceiveInfo> {
+    VoiceMediaReceiveInfo info;
+    if (GetStats(&info, reset_legacy)) {
+      return info;
+    }
+    return std::nullopt;
+  };
+}
 void FakeVoiceMediaReceiveChannel::SetRawAudioSink(
     uint32_t /* ssrc */,
     std::unique_ptr<AudioSinkInterface> sink) {
@@ -318,6 +329,16 @@ bool FakeVoiceMediaSendChannel::GetOutputVolume(uint32_t ssrc, double* volume) {
 bool FakeVoiceMediaSendChannel::GetStats(VoiceMediaSendInfo* /* info */) {
   return false;
 }
+absl::AnyInvocable<std::optional<VoiceMediaSendInfo>()>
+FakeVoiceMediaSendChannel::GetStatsCallback() {
+  return [this]() -> std::optional<VoiceMediaSendInfo> {
+    VoiceMediaSendInfo info;
+    if (GetStats(&info)) {
+      return info;
+    }
+    return std::nullopt;
+  };
+}
 bool FakeVoiceMediaSendChannel::SetSendCodecs(
     const std::vector<Codec>& codecs) {
   if (fail_set_send_codecs()) {
@@ -419,6 +440,16 @@ void FakeVideoMediaSendChannel::FillBitrateInfo(
 bool FakeVideoMediaSendChannel::GetStats(VideoMediaSendInfo* /* info */) {
   return false;
 }
+absl::AnyInvocable<std::optional<VideoMediaSendInfo>()>
+FakeVideoMediaSendChannel::GetStatsCallback() {
+  return [this]() -> std::optional<VideoMediaSendInfo> {
+    VideoMediaSendInfo info;
+    if (GetStats(&info)) {
+      return info;
+    }
+    return std::nullopt;
+  };
+}
 bool FakeVideoMediaSendChannel::SetSendCodecs(
     const std::vector<Codec>& codecs) {
   if (fail_set_send_codecs()) {
@@ -559,6 +590,16 @@ void FakeVideoMediaReceiveChannel::RequestRecvKeyFrame(uint32_t /* ssrc */) {}
 bool FakeVideoMediaReceiveChannel::GetStats(VideoMediaReceiveInfo* /* info */) {
   return false;
 }
+absl::AnyInvocable<std::optional<VideoMediaReceiveInfo>()>
+FakeVideoMediaReceiveChannel::GetStatsCallback() {
+  return [this]() -> std::optional<VideoMediaReceiveInfo> {
+    VideoMediaReceiveInfo info;
+    if (GetStats(&info)) {
+      return info;
+    }
+    return std::nullopt;
+  };
+}
 
 FakeVoiceEngine::FakeVoiceEngine()
     : encoder_factory_(make_ref_counted<FakeVoiceEncoderFactory>(this)),
diff --git a/media/base/fake_media_engine.h b/media/base/fake_media_engine.h
index 4e5e43a5e3..acff856d74 100644
--- a/media/base/fake_media_engine.h
+++ b/media/base/fake_media_engine.h
@@ -337,6 +337,11 @@ class RtpSendChannelHelper : public Base, public MediaChannelUtil {
     }
     return RtpParameters();
   }
+
+  absl::AnyInvocable<RtpParameters(uint32_t)> GetRtpSendParametersCallback()
+      const override {
+    return [this](uint32_t ssrc) { return GetRtpSendParameters(ssrc); };
+  }
   RTCError SetRtpSendParameters(uint32_t ssrc,
                                 const RtpParameters& parameters,
                                 SetParametersCallback callback) override {
@@ -519,6 +524,8 @@ class FakeVoiceMediaReceiveChannel
 
   bool GetStats(VoiceMediaReceiveInfo* info,
                 bool get_and_clear_legacy_stats) override;
+  absl::AnyInvocable<std::optional<VoiceMediaReceiveInfo>()> GetStatsCallback(
+      bool reset_legacy) override;
 
   void SetRawAudioSink(uint32_t ssrc,
                        std::unique_ptr<AudioSinkInterface> sink) override;
@@ -611,6 +618,8 @@ class FakeVoiceMediaSendChannel
   std::optional<Codec> GetSendCodec() const override;
 
   bool GetStats(VoiceMediaSendInfo* stats) override;
+  absl::AnyInvocable<std::optional<VoiceMediaSendInfo>()> GetStatsCallback()
+      override;
 
  private:
   class VoiceChannelAudioSink : public AudioSource::Sink {
@@ -695,6 +704,8 @@ class FakeVideoMediaReceiveChannel
   void ClearRecordableEncodedFrameCallback(uint32_t ssrc) override;
   void RequestRecvKeyFrame(uint32_t ssrc) override;
   bool GetStats(VideoMediaReceiveInfo* info) override;
+  absl::AnyInvocable<std::optional<VideoMediaReceiveInfo>()> GetStatsCallback()
+      override;
 
   bool AddDefaultRecvStreamForTesting(const StreamParams& /* sp */) override {
     RTC_CHECK_NOTREACHED();
@@ -755,6 +766,8 @@ class FakeVideoMediaSendChannel
 
   bool SendCodecHasNack() const override { return false; }
   bool GetStats(VideoMediaSendInfo* info) override;
+  absl::AnyInvocable<std::optional<VideoMediaSendInfo>()> GetStatsCallback()
+      override;
   bool SetOptions(const VideoOptions& options) override;
 
  private:
diff --git a/media/base/media_channel.h b/media/base/media_channel.h
index 70756943ff..95d1e7e7ec 100644
--- a/media/base/media_channel.h
+++ b/media/base/media_channel.h
@@ -238,6 +238,18 @@ class MediaSendChannelInterface {
       uint32_t /* ssrc */,
       VideoEncoderFactory::EncoderSelectorInterface* /* encoder_selector */) {}
   virtual RtpParameters GetRtpSendParameters(uint32_t ssrc) const = 0;
+  // Returns a callback that is to be used to retrieve RTP send parameters
+  // in the same way as `GetRtpSendParameters` does.
+  // The difference between GetRtpSendParametersCallback and
+  // `GetRtpSendParameters` is that GetRtpSendParametersCallback is callable
+  // from the signaling thread and allows binding the `alive` state of the
+  // channel with the returned callback object. The callback object itself
+  // however must be called on the worker thread, but does not require the
+  // caller to have a pointer to the channel. This avoids running into a race
+  // conditions during asynchronous teardown where signaling thread and worker
+  // thread state may be torn down asynchronously.
+  virtual absl::AnyInvocable<RtpParameters(uint32_t)>
+  GetRtpSendParametersCallback() const = 0;
   virtual bool SendCodecHasNack() const = 0;
   // Called whenever the list of sending SSRCs changes.
   virtual void SetSsrcListChangedCallback(
@@ -911,6 +923,13 @@ class VoiceMediaSendChannelInterface : public MediaSendChannelInterface {
   // DTMF event 0-9, *, #, A-D.
   virtual bool InsertDtmf(uint32_t ssrc, int event, int duration) = 0;
   virtual bool GetStats(VoiceMediaSendInfo* stats) = 0;
+  // Returns a callback that can be used to retrieve stats.
+  // The purpose is to allow binding state as it exists on the signaling thread
+  // to a callback that is run on the worker thread. This avoids race conditions
+  // during asynchronous teardown where signaling thread and worker thread state
+  // may be torn down asynchronously.
+  virtual absl::AnyInvocable<std::optional<VoiceMediaSendInfo>()>
+  GetStatsCallback() = 0;
   virtual bool SenderNackEnabled() const = 0;
   virtual bool SenderNonSenderRttEnabled() const = 0;
   virtual bool SetOptions(const AudioOptions& options) = 0;
@@ -936,6 +955,13 @@ class VoiceMediaReceiveChannelInterface : public MediaReceiveChannelInterface {
   virtual void SetDefaultRawAudioSink(
       std::unique_ptr<AudioSinkInterface> sink) = 0;
   virtual bool GetStats(VoiceMediaReceiveInfo* stats, bool reset_legacy) = 0;
+  // Returns a callback that can be used to retrieve stats.
+  // The purpose is to allow binding state as it exists on the signaling thread
+  // to a callback that is run on the worker thread. This avoids race conditions
+  // during asynchronous teardown where signaling thread and worker thread state
+  // may be torn down asynchronously.
+  virtual absl::AnyInvocable<std::optional<VoiceMediaReceiveInfo>()>
+  GetStatsCallback(bool reset_legacy) = 0;
   virtual enum RtcpMode RtcpMode() const = 0;
   virtual void SetRtcpMode(enum RtcpMode mode) = 0;
   virtual void SetReceiveNackEnabled(bool enabled) = 0;
@@ -979,6 +1005,13 @@ class VideoMediaSendChannelInterface : public MediaSendChannelInterface {
   virtual void GenerateSendKeyFrame(uint32_t ssrc,
                                     const std::vector<std::string>& rids) = 0;
   virtual bool GetStats(VideoMediaSendInfo* stats) = 0;
+  // Returns a callback that can be used to retrieve stats.
+  // The purpose is to allow binding state as it exists on the signaling thread
+  // to a callback that is run on the worker thread. This avoids race conditions
+  // during asynchronous teardown where signaling thread and worker thread state
+  // may be torn down asynchronously.
+  virtual absl::AnyInvocable<std::optional<VideoMediaSendInfo>()>
+  GetStatsCallback() = 0;
   // This fills the "bitrate parts" (rtx, video bitrate) of the
   // BandwidthEstimationInfo, since that part that isn't possible to get
   // through Call::GetStats, as they are statistics of the send
@@ -1016,6 +1049,13 @@ class VideoMediaReceiveChannelInterface : public MediaReceiveChannelInterface {
   // Clear recordable encoded frame callback for `ssrc`
   virtual void ClearRecordableEncodedFrameCallback(uint32_t ssrc) = 0;
   virtual bool GetStats(VideoMediaReceiveInfo* stats) = 0;
+  // Returns a callback that can be used to retrieve stats.
+  // The purpose is to allow binding state as it exists on the signaling thread
+  // to a callback that is run on the worker thread. This avoids race conditions
+  // during asynchronous teardown where signaling thread and worker thread state
+  // may be torn down asynchronously.
+  virtual absl::AnyInvocable<std::optional<VideoMediaReceiveInfo>()>
+  GetStatsCallback() = 0;
   virtual bool AddDefaultRecvStreamForTesting(const StreamParams& sp) = 0;
 };
 
diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc
index 6b3e0b506b..172c7a12b0 100644
--- a/media/engine/webrtc_video_engine.cc
+++ b/media/engine/webrtc_video_engine.cc
@@ -1408,6 +1408,16 @@ RtpParameters WebRtcVideoSendChannel::GetRtpSendParameters(
   return rtp_params;
 }
 
+absl::AnyInvocable<RtpParameters(uint32_t)>
+WebRtcVideoSendChannel::GetRtpSendParametersCallback() const {
+  return [this, safety = task_safety_.flag()](
+             uint32_t ssrc) mutable -> RtpParameters {
+    if (!safety->alive())
+      return RtpParameters();
+    return GetRtpSendParameters(ssrc);
+  };
+}
+
 bool WebRtcVideoSendChannel::SetOptions(const VideoOptions& options) {
   RTC_DCHECK_RUN_ON(&thread_checker_);
   default_send_options_ = options;
@@ -1726,6 +1736,21 @@ void WebRtcVideoSendChannel::FillSendCodecStats(
   }
 }
 
+absl::AnyInvocable<std::optional<VideoMediaSendInfo>()>
+WebRtcVideoSendChannel::GetStatsCallback() {
+  return [this, safety = task_safety_.flag()]() mutable
+             -> std::optional<VideoMediaSendInfo> {
+    if (!safety->alive()) {
+      return std::nullopt;
+    }
+    VideoMediaSendInfo info;
+    if (GetStats(&info)) {
+      return info;
+    }
+    return std::nullopt;
+  };
+}
+
 void WebRtcVideoSendChannel::OnPacketSent(const SentPacketInfo& sent_packet) {
   RTC_DCHECK_RUN_ON(&network_thread_checker_);
   // TODO(tommi): We shouldn't need to go through call_ to deliver this
@@ -3230,6 +3255,21 @@ void WebRtcVideoReceiveChannel::FillReceiveCodecStats(
   }
 }
 
+absl::AnyInvocable<std::optional<VideoMediaReceiveInfo>()>
+WebRtcVideoReceiveChannel::GetStatsCallback() {
+  return [this, safety = task_safety_.flag()]() mutable
+             -> std::optional<VideoMediaReceiveInfo> {
+    if (!safety->alive()) {
+      return std::nullopt;
+    }
+    VideoMediaReceiveInfo info;
+    if (GetStats(&info)) {
+      return info;
+    }
+    return std::nullopt;
+  };
+}
+
 void WebRtcVideoReceiveChannel::OnPacketReceived(RtpPacketReceived packet) {
   // Note: the network_thread_checker may refer to the worker thread if the two
   // threads are combined, but this is either always true or always false
diff --git a/media/engine/webrtc_video_engine.h b/media/engine/webrtc_video_engine.h
index b28211a41f..d497e42b8e 100644
--- a/media/engine/webrtc_video_engine.h
+++ b/media/engine/webrtc_video_engine.h
@@ -200,6 +200,8 @@ class WebRtcVideoSendChannel : public MediaChannelUtil,
                                 const RtpParameters& parameters,
                                 SetParametersCallback callback) override;
   RtpParameters GetRtpSendParameters(uint32_t ssrc) const override;
+  absl::AnyInvocable<RtpParameters(uint32_t)> GetRtpSendParametersCallback()
+      const override;
   std::optional<Codec> GetSendCodec() const override;
   bool SetSend(bool send) override;
   bool SetVideoSend(uint32_t ssrc,
@@ -209,6 +211,8 @@ class WebRtcVideoSendChannel : public MediaChannelUtil,
   bool RemoveSendStream(uint32_t ssrc) override;
   void FillBitrateInfo(BandwidthEstimationInfo* bwe_info) override;
   bool GetStats(VideoMediaSendInfo* info) override;
+  absl::AnyInvocable<std::optional<VideoMediaSendInfo>()> GetStatsCallback()
+      override;
 
   void OnPacketSent(const SentPacketInfo& sent_packet) override;
   void OnReadyToSend(bool ready) override;
@@ -531,6 +535,8 @@ class WebRtcVideoReceiveChannel : public MediaChannelUtil,
   bool SetSink(uint32_t ssrc, VideoSinkInterface<VideoFrame>* sink) override;
   void SetDefaultSink(VideoSinkInterface<VideoFrame>* sink) override;
   bool GetStats(VideoMediaReceiveInfo* info) override;
+  absl::AnyInvocable<std::optional<VideoMediaReceiveInfo>()> GetStatsCallback()
+      override;
   void OnPacketReceived(RtpPacketReceived packet) override;
   bool SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, int delay_ms) override;
 
@@ -764,6 +770,7 @@ class WebRtcVideoReceiveChannel : public MediaChannelUtil,
       RTC_GUARDED_BY(thread_checker_);
 
   const int receive_buffer_size_;
+  ScopedTaskSafety task_safety_;
 };
 
 // Keeping the old name "WebRtcVideoChannel" around because some external
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index a3c8c98089..0448da09de 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -1861,6 +1861,21 @@ bool WebRtcVoiceSendChannel::GetStats(VoiceMediaSendInfo* info) {
   return true;
 }
 
+absl::AnyInvocable<std::optional<VoiceMediaSendInfo>()>
+WebRtcVoiceSendChannel::GetStatsCallback() {
+  return [this, safety = task_safety_.flag()]() mutable
+             -> std::optional<VoiceMediaSendInfo> {
+    if (!safety->alive()) {
+      return std::nullopt;
+    }
+    VoiceMediaSendInfo info;
+    if (GetStats(&info)) {
+      return info;
+    }
+    return std::nullopt;
+  };
+}
+
 bool WebRtcVoiceSendChannel::SenderNackEnabled() const {
   RTC_DCHECK_RUN_ON(worker_thread_);
   if (!send_codec_spec_) {
@@ -1925,6 +1940,16 @@ RtpParameters WebRtcVoiceSendChannel::GetRtpSendParameters(
   return rtp_params;
 }
 
+absl::AnyInvocable<RtpParameters(uint32_t)>
+WebRtcVoiceSendChannel::GetRtpSendParametersCallback() const {
+  return [this, safety = task_safety_.flag()](
+             uint32_t ssrc) mutable -> RtpParameters {
+    if (!safety->alive())
+      return RtpParameters();
+    return GetRtpSendParameters(ssrc);
+  };
+}
+
 RTCError WebRtcVoiceSendChannel::SetRtpSendParameters(
     uint32_t ssrc,
     const RtpParameters& parameters,
@@ -2779,6 +2804,22 @@ void WebRtcVoiceReceiveChannel::FillReceiveCodecStats(
   }
 }
 
+absl::AnyInvocable<std::optional<VoiceMediaReceiveInfo>()>
+WebRtcVoiceReceiveChannel::GetStatsCallback(bool get_and_clear_legacy_stats) {
+  return
+      [this, get_and_clear_legacy_stats, safety = task_safety_.flag()]() mutable
+          -> std::optional<VoiceMediaReceiveInfo> {
+        if (!safety->alive()) {
+          return std::nullopt;
+        }
+        VoiceMediaReceiveInfo info;
+        if (GetStats(&info, get_and_clear_legacy_stats)) {
+          return info;
+        }
+        return std::nullopt;
+      };
+}
+
 void WebRtcVoiceReceiveChannel::SetRawAudioSink(
     uint32_t ssrc,
     std::unique_ptr<AudioSinkInterface> sink) {
diff --git a/media/engine/webrtc_voice_engine.h b/media/engine/webrtc_voice_engine.h
index c38a47f6aa..f033c2179e 100644
--- a/media/engine/webrtc_voice_engine.h
+++ b/media/engine/webrtc_voice_engine.h
@@ -218,6 +218,8 @@ class WebRtcVoiceSendChannel final : public MediaChannelUtil,
 
   bool SetSenderParameters(const AudioSenderParameter& params) override;
   RtpParameters GetRtpSendParameters(uint32_t ssrc) const override;
+  absl::AnyInvocable<RtpParameters(uint32_t)> GetRtpSendParametersCallback()
+      const override;
   RTCError SetRtpSendParameters(uint32_t ssrc,
                                 const RtpParameters& parameters,
                                 SetParametersCallback callback) override;
@@ -248,6 +250,8 @@ class WebRtcVoiceSendChannel final : public MediaChannelUtil,
                              const NetworkRoute& network_route) override;
   void OnReadyToSend(bool ready) override;
   bool GetStats(VoiceMediaSendInfo* info) override;
+  absl::AnyInvocable<std::optional<VoiceMediaSendInfo>()> GetStatsCallback()
+      override;
   bool SetOptions(const AudioOptions& options) override;
 
   // Sets a frame transformer between encoder and packetizer, to transform
@@ -377,6 +381,8 @@ class WebRtcVoiceReceiveChannel final
   void OnPacketReceived(RtpPacketReceived packet) override;
   bool GetStats(VoiceMediaReceiveInfo* info,
                 bool get_and_clear_legacy_stats) override;
+  absl::AnyInvocable<std::optional<VoiceMediaReceiveInfo>()> GetStatsCallback(
+      bool get_and_clear_legacy_stats) override;
 
   // Set the audio sink for an existing stream.
   void SetRawAudioSink(uint32_t ssrc,
diff --git a/pc/rtc_stats_collector.cc b/pc/rtc_stats_collector.cc
index 8030dd4cae..ce8c2b3b9e 100644
--- a/pc/rtc_stats_collector.cc
+++ b/pc/rtc_stats_collector.cc
@@ -2274,16 +2274,6 @@ RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w() {
 
   std::vector<RtpTransceiverStatsInfo> transceiver_stats_infos;
   std::vector<TransceiverReferences> transceiver_references;
-  // These are used to invoke GetStats for all the media channels together in
-  // one worker thread hop.
-  std::map<VoiceMediaSendChannelInterface*, VoiceMediaSendInfo>
-      voice_send_stats;
-  std::map<VideoMediaSendChannelInterface*, VideoMediaSendInfo>
-      video_send_stats;
-  std::map<VoiceMediaReceiveChannelInterface*, VoiceMediaReceiveInfo>
-      voice_receive_stats;
-  std::map<VideoMediaReceiveChannelInterface*, VideoMediaReceiveInfo>
-      video_receive_stats;
 
   auto transceivers = pc_->GetTransceiversInternal();
 
@@ -2319,18 +2309,28 @@ RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w() {
     stats.has_receivers = !refs.receivers.empty();
 
     if (stats.has_channel) {
+      auto* send_channel = transceiver->media_send_channel();
+      auto* receive_channel = transceiver->media_receive_channel();
       if (stats.media_type == MediaType::AUDIO) {
-        AddChannelStats(voice_send_stats,
-                        transceiver->voice_media_send_channel());
-        AddChannelStats(voice_receive_stats,
-                        transceiver->voice_media_receive_channel());
+        VoiceMediaSendChannelInterface* voice_send =
+            send_channel->AsVoiceSendChannel();
+        RTC_CHECK(voice_send);
+        VoiceMediaReceiveChannelInterface* voice_receive =
+            receive_channel->AsVoiceReceiveChannel();
+        RTC_CHECK(voice_receive);
+        refs.get_send_stats_voice = voice_send->GetStatsCallback();
+        refs.get_receive_stats_voice = voice_receive->GetStatsCallback(false);
+        refs.get_send_parameters = voice_send->GetRtpSendParametersCallback();
       } else if (stats.media_type == MediaType::VIDEO) {
-        AddChannelStats(video_send_stats,
-                        transceiver->video_media_send_channel());
-        AddChannelStats(video_receive_stats,
-                        transceiver->video_media_receive_channel());
-      } else {
-        RTC_DCHECK_NOTREACHED();
+        VideoMediaSendChannelInterface* video_send =
+            send_channel->AsVideoSendChannel();
+        RTC_CHECK(video_send);
+        VideoMediaReceiveChannelInterface* video_receive =
+            receive_channel->AsVideoReceiveChannel();
+        RTC_CHECK(video_receive);
+        refs.get_send_stats_video = video_send->GetStatsCallback();
+        refs.get_receive_stats_video = video_receive->GetStatsCallback();
+        refs.get_send_parameters = video_send->GetRtpSendParametersCallback();
       }
     }
 
@@ -2343,11 +2343,8 @@ RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w() {
   // as GetCallStats(). At the same time we construct the TrackMediaInfoMaps,
   // which also needs info from the worker thread.
   return [this, transceiver_stats_infos = std::move(transceiver_stats_infos),
-          transceiver_references = std::move(transceiver_references),
-          voice_send_stats = std::move(voice_send_stats),
-          voice_receive_stats = std::move(voice_receive_stats),
-          video_send_stats = std::move(video_send_stats),
-          video_receive_stats = std::move(video_receive_stats)]() mutable {
+          transceiver_references =
+              std::move(transceiver_references)]() mutable {
     Thread::ScopedDisallowBlockingCalls no_blocking_calls;
 
     WorkerThreadResult worker_result;
@@ -2355,28 +2352,6 @@ RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w() {
         std::move(transceiver_stats_infos);
     worker_result.transceiver_references = std::move(transceiver_references);
 
-    for (auto& pair : voice_send_stats) {
-      if (!pair.first->GetStats(&pair.second)) {
-        RTC_LOG(LS_WARNING) << "Failed to get voice send stats.";
-      }
-    }
-    for (auto& pair : voice_receive_stats) {
-      if (!pair.first->GetStats(&pair.second,
-                                /*get_and_clear_legacy_stats=*/false)) {
-        RTC_LOG(LS_WARNING) << "Failed to get voice receive stats.";
-      }
-    }
-    for (auto& pair : video_send_stats) {
-      if (!pair.first->GetStats(&pair.second)) {
-        RTC_LOG(LS_WARNING) << "Failed to get video send stats.";
-      }
-    }
-    for (auto& pair : video_receive_stats) {
-      if (!pair.first->GetStats(&pair.second)) {
-        RTC_LOG(LS_WARNING) << "Failed to get video receive stats.";
-      }
-    }
-
     // Create the TrackMediaInfoMap for each transceiver stats object
     // and keep track of whether we have at least one audio receiver.
     bool has_audio_receiver = false;
@@ -2396,16 +2371,7 @@ RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w() {
       for (const auto& sender : stats.sender_infos) {
         RtpParameters params;
         if (sender.ssrc != 0 && stats.has_channel) {
-          auto& transceiver = refs.transceiver;
-          MediaSendChannelInterface* media_channel =
-              (stats.media_type == MediaType::AUDIO)
-                  ? static_cast<MediaSendChannelInterface*>(
-                        transceiver->voice_media_send_channel())
-                  : static_cast<MediaSendChannelInterface*>(
-                        transceiver->video_media_send_channel());
-          if (media_channel) {
-            params = media_channel->GetRtpSendParameters(sender.ssrc);
-          }
+          params = refs.get_send_parameters(sender.ssrc);
         }
         // `sender_parameters` must be the same size as `stats.sender_infos` to
         // allow 1:1 mapping when TrackMediaInfoMap is constructed. Empty params
@@ -2421,21 +2387,22 @@ RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w() {
       std::optional<VoiceMediaInfo> voice_media_info;
       std::optional<VideoMediaInfo> video_media_info;
       if (stats.has_channel) {
-        auto& transceiver = refs.transceiver;
         if (stats.media_type == MediaType::AUDIO) {
-          auto voice_send_channel = transceiver->voice_media_send_channel();
-          auto voice_receive_channel =
-              transceiver->voice_media_receive_channel();
+          std::optional<VoiceMediaSendInfo> voice_send =
+              refs.get_send_stats_voice();
+          std::optional<VoiceMediaReceiveInfo> voice_receive =
+              refs.get_receive_stats_voice();
           voice_media_info = VoiceMediaInfo(
-              std::move(voice_send_stats[voice_send_channel]),
-              std::move(voice_receive_stats[voice_receive_channel]));
+              std::move(voice_send).value_or(VoiceMediaSendInfo{}),
+              std::move(voice_receive).value_or(VoiceMediaReceiveInfo{}));
         } else if (stats.media_type == MediaType::VIDEO) {
-          auto video_send_channel = transceiver->video_media_send_channel();
-          auto video_receive_channel =
-              transceiver->video_media_receive_channel();
+          std::optional<VideoMediaSendInfo> video_send =
+              refs.get_send_stats_video();
+          std::optional<VideoMediaReceiveInfo> video_receive =
+              refs.get_receive_stats_video();
           video_media_info = VideoMediaInfo(
-              std::move(video_send_stats[video_send_channel]),
-              std::move(video_receive_stats[video_receive_channel]));
+              std::move(video_send).value_or(VideoMediaSendInfo{}),
+              std::move(video_receive).value_or(VideoMediaReceiveInfo{}));
         }
       }
 
diff --git a/pc/rtc_stats_collector.h b/pc/rtc_stats_collector.h
index c873006066..f20a67d0e3 100644
--- a/pc/rtc_stats_collector.h
+++ b/pc/rtc_stats_collector.h
@@ -35,6 +35,7 @@
 #include "api/task_queue/task_queue_base.h"
 #include "api/units/timestamp.h"
 #include "call/call.h"
+#include "media/base/media_channel.h"
 #include "pc/data_channel_utils.h"
 #include "pc/peer_connection_internal.h"
 #include "pc/rtp_receiver.h"
@@ -78,6 +79,13 @@ struct RtpTransceiverStatsInfo {
 struct TransceiverReferences {
   scoped_refptr<RtpTransceiver> transceiver;
   std::vector<scoped_refptr<RtpReceiverInternal>> receivers;
+  absl::AnyInvocable<std::optional<VoiceMediaSendInfo>()> get_send_stats_voice;
+  absl::AnyInvocable<std::optional<VideoMediaSendInfo>()> get_send_stats_video;
+  absl::AnyInvocable<std::optional<VoiceMediaReceiveInfo>()>
+      get_receive_stats_voice;
+  absl::AnyInvocable<std::optional<VideoMediaReceiveInfo>()>
+      get_receive_stats_video;
+  absl::AnyInvocable<RtpParameters(uint32_t ssrc)> get_send_parameters;
 };
 
 // All public methods of the collector are to be called on the signaling thread.
diff --git a/pc/rtc_stats_collector_unittest.cc b/pc/rtc_stats_collector_unittest.cc
index 35905ed3f1..18b4df2997 100644
--- a/pc/rtc_stats_collector_unittest.cc
+++ b/pc/rtc_stats_collector_unittest.cc
@@ -371,6 +371,7 @@ class RTCStatsCollectorWrapper {
     });
     EXPECT_CALL(*sender, AttachmentId()).WillRepeatedly(Return(attachment_id));
     EXPECT_CALL(*sender, Stop()).WillRepeatedly(Return());
+    EXPECT_CALL(*sender, SetSendCodecs(_)).WillRepeatedly(Return());
     RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
     pc_->AddSender(sender);
     RTC_ALLOW_PLAN_B_DEPRECATION_END();
@@ -887,6 +888,70 @@ TEST_P(RTCStatsCollectorTest, MultipleCallbacksWithInvalidatedCacheInBetween) {
   EXPECT_NE(c.get(), b.get());
 }
 
+TEST_P(RTCStatsCollectorTest, StatsPreservedWhenChannelClearedConcurrently) {
+  VoiceMediaInfo voice_media_info;
+  voice_media_info.receivers.emplace_back();
+  voice_media_info.receivers[0].add_ssrc(1);
+  voice_media_info.receivers[0].packets_received = 123;
+
+  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
+  pc_->AddVoiceChannel("AudioMid", "TransportName", voice_media_info);
+  RTC_ALLOW_PLAN_B_DEPRECATION_END();
+  auto remote_receiver = stats_->SetupRemoteTrackAndReceiver(
+      MediaType::AUDIO, "RemoteAudioTrackID", "RemoteStreamId", 1);
+  auto local_sender = stats_->SetupLocalTrackAndSender(
+      MediaType::AUDIO, "LocalAudioTrackID", 2,
+      /*add_stream=*/true, /*attachment_id=*/2);
+
+  EXPECT_CALL(*local_sender, SetSendCodecs(_)).WillRepeatedly(Return());
+  EXPECT_CALL(*local_sender, SetMediaChannel(_)).WillRepeatedly(Return());
+  EXPECT_CALL(*local_sender, DetachTrackAndGetStopTask()).WillRepeatedly([] {
+    return nullptr;
+  });
+  EXPECT_CALL(*remote_receiver, SetMediaChannel(_)).WillRepeatedly(Return());
+
+  scoped_refptr<const RTCStatsReport> report;
+  bool stats_obtained = false;
+  stats_->stats_collector().GetStatsReport(
+      RTCStatsObtainer::Create(&report, [&] {
+        stats_obtained = true;
+        main_thread_.Quit();
+      }));
+
+  auto transceivers = pc_->GetTransceiversInternal();
+  ASSERT_FALSE(transceivers.empty());
+  auto* transceiver = transceivers[0]->internal();
+  EXPECT_TRUE(transceiver->HasChannel());
+
+  auto network_task = transceiver->GetClearChannelNetworkTask();
+  pc_->network_thread()->BlockingCall([&]() { std::move(network_task)(); });
+
+  auto delete_worker_task =
+      transceiver->GetDeleteChannelWorkerTask(/*stop_senders=*/false);
+  // Now the state of the transceiver is that on the signaling thread,
+  // there is no channel. However, the work of actually deleting the channels
+  // belongs to the worker thread task, which hasn't run yet.
+  // Let's post the execution of that task so that it ends up running
+  // *after* the task that the `stats_collector` has queued up for
+  // gathering the stats from the channel. The stats collector should
+  // be able to get the stats safely even though technically the
+  // transceiver by now, doesn't have a channel.
+  EXPECT_FALSE(transceiver->HasChannel());
+  Event worker_task_done;
+  pc_->worker_thread()->PostTask([&]() {
+    std::move(delete_worker_task)();
+    worker_task_done.Set();
+  });
+
+  main_thread_.Run();
+
+  EXPECT_TRUE(report);
+  auto inbound_rtps = report->GetStatsOfType<RTCInboundRtpStreamStats>();
+  EXPECT_EQ(inbound_rtps.size(), 1u);
+  // Wait for the cleanup task to complete.
+  worker_task_done.Wait(Event::kForever);
+}
+
 TEST_P(RTCStatsCollectorTest, ToJsonProducesParseableJson) {
   ExampleStatsGraph graph = SetupExampleStatsGraphForSelectorTests();
   scoped_refptr<const RTCStatsReport> report =
diff --git a/pc/test/fake_peer_connection_for_stats.h b/pc/test/fake_peer_connection_for_stats.h
index 1e380820d6..c2f0f07dda 100644
--- a/pc/test/fake_peer_connection_for_stats.h
+++ b/pc/test/fake_peer_connection_for_stats.h
@@ -20,6 +20,7 @@
 #include <utility>
 #include <vector>
 
+#include "absl/functional/any_invocable.h"
 #include "absl/strings/string_view.h"
 #include "api/audio/audio_device.h"
 #include "api/audio_options.h"
@@ -35,6 +36,7 @@
 #include "api/rtp_transceiver_direction.h"
 #include "api/scoped_refptr.h"
 #include "api/sequence_checker.h"
+#include "api/task_queue/pending_task_safety_flag.h"
 #include "api/task_queue/task_queue_base.h"
 #include "api/transport/data_channel_transport_interface.h"
 #include "call/call.h"
@@ -111,7 +113,23 @@ class FakeVoiceMediaSendChannelForStats : public FakeVoiceMediaSendChannel {
     return false;
   }
 
+  absl::AnyInvocable<std::optional<VoiceMediaSendInfo>()> GetStatsCallback()
+      override {
+    return [this, safety = task_safety_.flag()]() mutable
+               -> std::optional<VoiceMediaSendInfo> {
+      if (!safety->alive()) {
+        return std::nullopt;
+      }
+      VoiceMediaSendInfo info;
+      if (GetStats(&info)) {
+        return info;
+      }
+      return std::nullopt;
+    };
+  }
+
  private:
+  ScopedTaskSafety task_safety_;
   std::optional<VoiceMediaSendInfo> send_stats_;
 };
 
@@ -138,7 +156,23 @@ class FakeVoiceMediaReceiveChannelForStats
     return false;
   }
 
+  absl::AnyInvocable<std::optional<VoiceMediaReceiveInfo>()> GetStatsCallback(
+      bool get_and_clear_legacy_stats) override {
+    return [this, safety = task_safety_.flag()]() mutable
+               -> std::optional<VoiceMediaReceiveInfo> {
+      if (!safety->alive()) {
+        return std::nullopt;
+      }
+      VoiceMediaReceiveInfo info;
+      if (GetStats(&info, /*get_and_clear_legacy_stats=*/false)) {
+        return info;
+      }
+      return std::nullopt;
+    };
+  }
+
  private:
+  ScopedTaskSafety task_safety_;
   std::optional<VoiceMediaReceiveInfo> receive_stats_;
 };
 
@@ -164,7 +198,23 @@ class FakeVideoMediaSendChannelForStats : public FakeVideoMediaSendChannel {
     return false;
   }
 
+  absl::AnyInvocable<std::optional<VideoMediaSendInfo>()> GetStatsCallback()
+      override {
+    return [this, safety = task_safety_.flag()]() mutable
+               -> std::optional<VideoMediaSendInfo> {
+      if (!safety->alive()) {
+        return std::nullopt;
+      }
+      VideoMediaSendInfo info;
+      if (GetStats(&info)) {
+        return info;
+      }
+      return std::nullopt;
+    };
+  }
+
  private:
+  ScopedTaskSafety task_safety_;
   std::optional<VideoMediaSendInfo> send_stats_;
 };
 
@@ -189,7 +239,23 @@ class FakeVideoMediaReceiveChannelForStats
     return false;
   }
 
+  absl::AnyInvocable<std::optional<VideoMediaReceiveInfo>()> GetStatsCallback()
+      override {
+    return [this, safety = task_safety_.flag()]() mutable
+               -> std::optional<VideoMediaReceiveInfo> {
+      if (!safety->alive()) {
+        return std::nullopt;
+      }
+      VideoMediaReceiveInfo info;
+      if (GetStats(&info)) {
+        return info;
+      }
+      return std::nullopt;
+    };
+  }
+
  private:
+  ScopedTaskSafety task_safety_;
   std::optional<VideoMediaReceiveInfo> receive_stats_;
 };
 
diff --git a/pc/test/mock_voice_media_receive_channel_interface.h b/pc/test/mock_voice_media_receive_channel_interface.h
index 1c1f5e3211..feac0eeab0 100644
--- a/pc/test/mock_voice_media_receive_channel_interface.h
+++ b/pc/test/mock_voice_media_receive_channel_interface.h
@@ -16,6 +16,7 @@
 #include <type_traits>
 #include <vector>
 
+#include "absl/functional/any_invocable.h"
 #include "api/audio_options.h"
 #include "api/call/audio_sink.h"
 #include "api/crypto/frame_decryptor_interface.h"
@@ -74,6 +75,10 @@ class MockVoiceMediaReceiveChannelInterface
               GetStats,
               (webrtc::VoiceMediaReceiveInfo * stats, bool reset_legacy),
               (override));
+  MOCK_METHOD(absl::AnyInvocable<std::optional<VoiceMediaReceiveInfo>()>,
+              GetStatsCallback,
+              (bool reset_legacy),
+              (override));
   MOCK_METHOD(::webrtc::RtcpMode, RtcpMode, (), (const, override));
   MOCK_METHOD(void, SetRtcpMode, (::webrtc::RtcpMode mode), (override));
   MOCK_METHOD(void, SetReceiveNackEnabled, (bool enabled), (override));

From a2bd172110638168e74c393e910b5f5f5bf375b6 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Wed, 1 Apr 2026 22:51:25 +0200
Subject: [PATCH 465/472] Move payload type demuxing management to RtpTransport

Shift the responsibility for enabling and disabling payload type
demuxing from the Channel and Transceiver layers to the RtpTransport and
RtpDemuxer layer. This centralizes demuxing logic within the transport
layer.

Moving this logic to RtpTransport allows the SdpOfferAnswerHandler to
apply state changes directly to the transport and not need to go through
the Channel classes.

Key modifications:
- Remove SetPayloadTypeDemuxingEnabled from BaseChannel,
  ChannelInterface, and RtpTransceiver.
- Implement SetActivePayloadTypeDemuxing in RtpTransport to manage
  demuxer criteria filtering and coordinate with RtpDemuxer.
- Update SdpOfferAnswerHandler to apply demuxing state changes to
  RtpTransport instances directly.
- Ensure ResetUnsignaledRecvStream executes on the worker thread
  after applying transport state, to clear legacy streams when
  demuxing is disabled.

Bug: webrtc:42222117
Change-Id: If02194417dc8e4c3dcf5fda5e969aa81116004e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456720
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47318}
---
 call/rtp_demuxer.cc                 |  9 +++-
 call/rtp_demuxer.h                  |  5 ++
 media/engine/webrtc_video_engine.cc |  5 ++
 media/engine/webrtc_video_engine.h  |  1 +
 media/engine/webrtc_voice_engine.cc |  5 ++
 pc/BUILD.gn                         |  2 +
 pc/channel.cc                       | 40 +---------------
 pc/channel.h                        | 11 -----
 pc/channel_interface.h              |  1 -
 pc/rtp_transceiver.cc               | 14 +++---
 pc/rtp_transceiver.h                |  3 +-
 pc/rtp_transport.cc                 |  5 ++
 pc/rtp_transport.h                  |  2 +
 pc/rtp_transport_internal.h         |  2 +
 pc/sdp_offer_answer.cc              | 74 +++++++++++++++++++----------
 pc/sdp_offer_answer.h               |  2 +-
 pc/test/mock_channel_interface.h    |  1 -
 17 files changed, 93 insertions(+), 89 deletions(-)

diff --git a/call/rtp_demuxer.cc b/call/rtp_demuxer.cc
index c49d4fc4f8..598764b71c 100644
--- a/call/rtp_demuxer.cc
+++ b/call/rtp_demuxer.cc
@@ -400,8 +400,13 @@ RtpPacketSinkInterface* RtpDemuxer::ResolveSink(
     return ssrc_sink_it->second;
   }
 
-  // Legacy senders will only signal payload type, support that as last resort.
-  return ResolveSinkByPayloadType(packet.PayloadType(), ssrc);
+  if (use_payload_type_demuxing_) {
+    // Legacy senders will only signal payload type.
+    // Support that as a last resort.
+    return ResolveSinkByPayloadType(packet.PayloadType(), ssrc);
+  }
+
+  return nullptr;
 }
 
 RtpPacketSinkInterface* RtpDemuxer::ResolveSinkByMid(absl::string_view mid,
diff --git a/call/rtp_demuxer.h b/call/rtp_demuxer.h
index acc985f486..cc6a84c878 100644
--- a/call/rtp_demuxer.h
+++ b/call/rtp_demuxer.h
@@ -136,6 +136,10 @@ class RtpDemuxer {
   RtpDemuxer(const RtpDemuxer&) = delete;
   void operator=(const RtpDemuxer&) = delete;
 
+  void set_use_payload_type_demuxing(bool enable) {
+    use_payload_type_demuxing_ = enable;
+  }
+
   // Registers a sink that will be notified when RTP packets match its given
   // criteria according to the algorithm described in the class description.
   // Returns true if the sink was successfully added.
@@ -230,6 +234,7 @@ class RtpDemuxer {
   void AddSsrcSinkBinding(uint32_t ssrc, RtpPacketSinkInterface* sink);
 
   const bool use_mid_;
+  bool use_payload_type_demuxing_ = true;
 };
 
 }  // namespace webrtc
diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc
index 172c7a12b0..0e6ce01ac5 100644
--- a/media/engine/webrtc_video_engine.cc
+++ b/media/engine/webrtc_video_engine.cc
@@ -3143,6 +3143,11 @@ bool WebRtcVideoReceiveChannel::RemoveRecvStream(uint32_t ssrc) {
 }
 
 void WebRtcVideoReceiveChannel::ResetUnsignaledRecvStream() {
+  if (!worker_thread_->IsCurrent()) {
+    worker_thread_->PostTask(SafeTask(
+        task_safety_.flag(), [this]() { ResetUnsignaledRecvStream(); }));
+    return;
+  }
   RTC_DCHECK_RUN_ON(&thread_checker_);
   RTC_LOG(LS_INFO) << "ResetUnsignaledRecvStream.";
   unsignaled_stream_params_ = StreamParams();
diff --git a/media/engine/webrtc_video_engine.h b/media/engine/webrtc_video_engine.h
index d497e42b8e..334a871168 100644
--- a/media/engine/webrtc_video_engine.h
+++ b/media/engine/webrtc_video_engine.h
@@ -711,6 +711,7 @@ class WebRtcVideoReceiveChannel : public MediaChannelUtil,
   // Variables.
   const Environment env_;
   TaskQueueBase* const worker_thread_;
+  ScopedTaskSafety task_safety_;
   scoped_refptr<PendingTaskSafetyFlag> network_thread_safety_;
   RTC_NO_UNIQUE_ADDRESS SequenceChecker network_thread_checker_{
       SequenceChecker::kDetached};
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index 0448da09de..4000fb1980 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -2495,6 +2495,11 @@ bool WebRtcVoiceReceiveChannel::RemoveRecvStream(uint32_t ssrc) {
 }
 
 void WebRtcVoiceReceiveChannel::ResetUnsignaledRecvStream() {
+  if (!worker_thread_->IsCurrent()) {
+    worker_thread_->PostTask(SafeTask(
+        task_safety_.flag(), [this]() { ResetUnsignaledRecvStream(); }));
+    return;
+  }
   RTC_DCHECK_RUN_ON(worker_thread_);
   RTC_LOG(LS_INFO) << "ResetUnsignaledRecvStream.";
   unsignaled_stream_params_ = StreamParams();
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 91cc293538..322488a20c 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -617,6 +617,7 @@ rtc_library("rtp_transport") {
     "../rtc_base/network:sent_packet",
     "../rtc_base/system:no_unique_address",
     "//third_party/abseil-cpp/absl/algorithm:container",
+    "//third_party/abseil-cpp/absl/container:flat_hash_map",
     "//third_party/abseil-cpp/absl/strings:string_view",
   ]
 }
@@ -1150,6 +1151,7 @@ rtc_library("sdp_offer_answer") {
     ":rtp_sender_proxy",
     ":rtp_transceiver",
     ":rtp_transmission_manager",
+    ":rtp_transport_internal",
     ":scoped_operations_batcher",
     ":sdp_munging_detector",
     ":sdp_payload_type_suggester",
diff --git a/pc/channel.cc b/pc/channel.cc
index 610f507b2d..59921fa4ff 100644
--- a/pc/channel.cc
+++ b/pc/channel.cc
@@ -338,42 +338,6 @@ RTCError BaseChannel::SetRemoteContent(const MediaContentDescription* content,
   return SetRemoteContent_w(content, type);
 }
 
-bool BaseChannel::SetPayloadTypeDemuxingEnabled(bool enabled) {
-  // TODO(bugs.webrtc.org/11993): The demuxer state needs to be managed on the
-  // network thread. At the moment there's a workaround for inconsistent state
-  // between the worker and network thread because of this (see
-  // OnDemuxerCriteriaUpdatePending elsewhere in this file) and
-  // UpdateRemoteStreams_w has a BlockingCall over to the network
-  // thread to apply state updates.
-  RTC_DCHECK_RUN_ON(network_thread());
-  TRACE_EVENT0("webrtc", "BaseChannel::SetPayloadTypeDemuxingEnabled");
-
-  if (enabled == payload_type_demuxing_enabled_) {
-    return true;
-  }
-
-  payload_type_demuxing_enabled_ = enabled;
-
-  if (!enabled) {
-    worker_thread_->PostTask(SafeTask(alive_, [this] {
-      // TODO(crbug.com/11477): This will remove *all* unsignaled streams (those
-      // without an explicitly signaled SSRC), which may include streams that
-      // were matched to this channel by MID or RID. Ideally we'd remove only
-      // the streams that were matched based on payload type alone, but
-      // currently there is no straightforward way to identify those streams.
-      media_receive_channel()->ResetUnsignaledRecvStream();
-    }));
-  }
-
-  if (!payload_types_.empty()) {
-    if (!rtp_transport_) {
-      return false;
-    }
-    return rtp_transport_->RegisterRtpDemuxerSink(demuxer_criteria(), this);
-  }
-
-  return true;
-}
 
 bool BaseChannel::IsReadyToSendMedia_w() const {
   // Send outgoing data if we are enabled, have local and remote content,
@@ -661,9 +625,7 @@ bool BaseChannel::RegisterRtpDemuxerSink_w(
 RtpDemuxerCriteria BaseChannel::demuxer_criteria() const {
   RTC_DCHECK_RUN_ON(network_thread());
   RtpDemuxerCriteria criteria(mid_);
-  if (payload_type_demuxing_enabled_) {
-    criteria.payload_types() = payload_types_;
-  }
+  criteria.payload_types() = payload_types_;
   criteria.ssrcs() = ssrcs_;
   return criteria;
 }
diff --git a/pc/channel.h b/pc/channel.h
index fcb01262a9..a19950fbc5 100644
--- a/pc/channel.h
+++ b/pc/channel.h
@@ -127,16 +127,6 @@ class BaseChannel : public ChannelInterface,
                            SdpType type) override;
   RTCError SetRemoteContent(const MediaContentDescription* content,
                             SdpType type) override;
-  // Controls whether this channel will receive packets on the basis of
-  // matching payload type alone. This is needed for legacy endpoints that
-  // don't signal SSRCs or use MID/RID, but doesn't make sense if there is
-  // more than channel of specific media type, As that creates an ambiguity.
-  //
-  // This method will also remove any existing streams that were bound to this
-  // channel on the basis of payload type, since one of these streams might
-  // actually belong to a new channel. See: crbug.com/webrtc/11477
-  bool SetPayloadTypeDemuxingEnabled(bool enabled) override
-      RTC_RUN_ON(network_thread());
 
   void Enable(bool enable) override;
 
@@ -344,7 +334,6 @@ class BaseChannel : public ChannelInterface,
   // call to the worker thread, so it should be safe.
   bool enabled_ RTC_GUARDED_BY(worker_thread()) = false;
   bool enabled_s_ RTC_GUARDED_BY(signaling_thread()) = false;
-  bool payload_type_demuxing_enabled_ RTC_GUARDED_BY(network_thread()) = true;
   std::vector<StreamParams> local_streams_ RTC_GUARDED_BY(worker_thread());
   std::vector<StreamParams> remote_streams_ RTC_GUARDED_BY(worker_thread());
   RtpTransceiverDirection local_content_direction_
diff --git a/pc/channel_interface.h b/pc/channel_interface.h
index 97fed01e8f..dfbe7496cc 100644
--- a/pc/channel_interface.h
+++ b/pc/channel_interface.h
@@ -91,7 +91,6 @@ class ChannelInterface {
                                    SdpType type) = 0;
   virtual RTCError SetRemoteContent(const MediaContentDescription* content,
                                     SdpType type) = 0;
-  virtual bool SetPayloadTypeDemuxingEnabled(bool enabled) = 0;
 
   // Access to the local and remote streams that were set on the channel.
   virtual const std::vector<StreamParams>& local_streams() const = 0;
diff --git a/pc/rtp_transceiver.cc b/pc/rtp_transceiver.cc
index 946601a19c..58da29fe79 100644
--- a/pc/rtp_transceiver.cc
+++ b/pc/rtp_transceiver.cc
@@ -1372,19 +1372,19 @@ void RtpTransceiver::SetChannelContent(
             });
       });
 }
-
-bool RtpTransceiver::SetChannelPayloadTypeDemuxingEnabled(bool enabled) {
-  RTC_DCHECK_RUN_ON(context()->network_thread());
-  RTC_DCHECK(channel_);
-  return channel_->SetPayloadTypeDemuxingEnabled(enabled);
-}
-
 void RtpTransceiver::EnableChannel(bool enable) {
   RTC_DCHECK_RUN_ON(thread_);
   RTC_DCHECK(channel_);
   channel_->Enable(enable);
 }
 
+void RtpTransceiver::ResetUnsignaledRecvStream() {
+  RTC_DCHECK_RUN_ON(thread_);
+  if (MediaReceiveChannelInterface* receive_channel = media_receive_channel()) {
+    receive_channel->ResetUnsignaledRecvStream();
+  }
+}
+
 const std::vector<StreamParams>& RtpTransceiver::channel_local_streams() const {
   RTC_DCHECK_RUN_ON(thread_);
   RTC_DCHECK(channel_);
diff --git a/pc/rtp_transceiver.h b/pc/rtp_transceiver.h
index a69d17be67..86e76f824f 100644
--- a/pc/rtp_transceiver.h
+++ b/pc/rtp_transceiver.h
@@ -392,8 +392,9 @@ class RtpTransceiver : public RtpTransceiverInterface {
   void SetChannelRemoteContent(const MediaContentDescription* content,
                                SdpType type,
                                ScopedOperationsBatcher& batcher);
-  bool SetChannelPayloadTypeDemuxingEnabled(bool enabled);
   void EnableChannel(bool enable);
+  void ResetUnsignaledRecvStream();
+
   const std::vector<StreamParams>& channel_local_streams() const;
   const std::vector<StreamParams>& channel_remote_streams() const;
   absl::string_view channel_transport_name() const;
diff --git a/pc/rtp_transport.cc b/pc/rtp_transport.cc
index 20db5d3a03..05ef651486 100644
--- a/pc/rtp_transport.cc
+++ b/pc/rtp_transport.cc
@@ -341,9 +341,14 @@ void RtpTransport::RebuildMergedMap() {
   header_extension_map_ = std::move(merged_map);
 }
 
+void RtpTransport::SetActivePayloadTypeDemuxing(bool enabled) {
+  rtp_demuxer_.set_use_payload_type_demuxing(enabled);
+}
+
 bool RtpTransport::RegisterRtpDemuxerSink(const RtpDemuxerCriteria& criteria,
                                           RtpPacketSinkInterface* sink) {
   rtp_demuxer_.RemoveSink(sink);
+
   if (!rtp_demuxer_.AddSink(criteria, sink)) {
     RTC_LOG(LS_ERROR) << "Failed to register the sink for RTP demuxer.";
     return false;
diff --git a/pc/rtp_transport.h b/pc/rtp_transport.h
index 1b0d65b96a..e51a1247fa 100644
--- a/pc/rtp_transport.h
+++ b/pc/rtp_transport.h
@@ -112,6 +112,8 @@ class RtpTransport : public RtpTransportInternal {
 
   bool UnregisterRtpDemuxerSink(RtpPacketSinkInterface* sink) override;
 
+  void SetActivePayloadTypeDemuxing(bool enabled) override;
+
  protected:
   // These methods will be used in the subclasses.
   void DemuxPacket(CopyOnWriteBuffer packet,
diff --git a/pc/rtp_transport_internal.h b/pc/rtp_transport_internal.h
index 96aaa3ea3c..bb4b66961b 100644
--- a/pc/rtp_transport_internal.h
+++ b/pc/rtp_transport_internal.h
@@ -151,6 +151,8 @@ class RtpTransportInternal {
 
   virtual bool UnregisterRtpDemuxerSink(RtpPacketSinkInterface* sink) = 0;
 
+  virtual void SetActivePayloadTypeDemuxing(bool enabled) = 0;
+
  protected:
   void SendReadyToSend(bool arg) { callback_list_ready_to_send_.Send(arg); }
   void SendRtcpPacketReceived(CopyOnWriteBuffer packet,
diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc
index 9ef845c176..7a9348071f 100644
--- a/pc/sdp_offer_answer.cc
+++ b/pc/sdp_offer_answer.cc
@@ -86,6 +86,7 @@
 #include "pc/rtp_sender_proxy.h"
 #include "pc/rtp_transceiver.h"
 #include "pc/rtp_transmission_manager.h"
+#include "pc/rtp_transport_internal.h"
 #include "pc/scoped_operations_batcher.h"
 #include "pc/sdp_munging_detector.h"
 #include "pc/session_description.h"
@@ -5165,15 +5166,9 @@ RTCError SdpOfferAnswerHandler::PushdownMediaDescription(
   RTC_DCHECK(sdesc);
 
   if (ConfiguredForMedia()) {
-    // Note: This will perform a BlockingCall over to the worker thread, which
+    // Note: This may perform a BlockingCall over to the network thread, which
     // we'll also do in a loop below.
-    if (!UpdatePayloadTypeDemuxingState(source, bundle_groups_by_mid)) {
-      // Note that this is never expected to fail, since RtpDemuxer doesn't
-      // return an error when changing payload type demux criteria, which is all
-      // this does.
-      return LOG_ERROR(RTCError(RTCErrorType::INTERNAL_ERROR)
-                       << "Failed to update payload type demuxing state.");
-    }
+    UpdatePayloadTypeDemuxingState(source, bundle_groups_by_mid);
 
     // Push down the new SDP media section for each audio/video transceiver.
     auto rtp_transceivers = transceivers()->ListInternal();
@@ -5778,7 +5773,7 @@ SdpOfferAnswerHandler::GetMediaDescriptionOptionsForRejectedData(
   return options;
 }
 
-bool SdpOfferAnswerHandler::UpdatePayloadTypeDemuxingState(
+void SdpOfferAnswerHandler::UpdatePayloadTypeDemuxingState(
     ContentSource source,
     const flat_map<std::string, const ContentGroup*>& bundle_groups_by_mid) {
   TRACE_EVENT0("webrtc",
@@ -5879,13 +5874,13 @@ bool SdpOfferAnswerHandler::UpdatePayloadTypeDemuxingState(
                                         mid_header_extension_missing_video ||
                                         pt_demuxing_has_been_used_video_;
 
-  // Gather all updates ahead of time so that all channels can be updated in a
+  // Gather all updates ahead of time so that all transports can be updated in a
   // single BlockingCall; necessary due to thread guards.
-  std::vector<std::pair<bool, RtpTransceiver*>> channels_to_update;
-  for (const auto& transceiver : transceivers()->ListInternal()) {
+  flat_map<std::string, bool> transports_to_update;
+  for (RtpTransceiver* transceiver : transceivers()->ListInternal()) {
     const ContentInfo* content =
         FindMediaSectionForTransceiver(transceiver, sdesc);
-    if (!transceiver->HasChannel() || !content) {
+    if (!content) {
       continue;
     }
 
@@ -5923,26 +5918,53 @@ bool SdpOfferAnswerHandler::UpdatePayloadTypeDemuxingState(
       }
     }
 
-    channels_to_update.emplace_back(pt_demux_enabled, transceiver);
+    // Accumulate the pt_demux_enabled state per transport (represented by mid).
+    // If any transceiver associated with the transport requires PT demuxing,
+    // the transport will have it enabled.
+    transports_to_update[content->mid()] |= pt_demux_enabled;
   }
 
-  if (channels_to_update.empty()) {
-    return true;
+  if (transports_to_update.empty()) {
+    return;
   }
 
-  // TODO(bugs.webrtc.org/11993): The demuxer state is now fully managed on
-  // the network thread. So we do a single blocking call here to the network
-  // thread. Ideally we could also do this without blocking.
-  return context_->network_thread()->BlockingCall([&channels_to_update]() {
-    for (const auto& it : channels_to_update) {
-      if (!it.second->SetChannelPayloadTypeDemuxingEnabled(it.first)) {
-        // Note that the state has already been irrevocably changed at this
-        // point. Is it useful to stop the loop?
-        return false;
+  context_->network_thread()->BlockingCall([&]() {
+    JsepTransportController* transport_controller =
+        pc_->transport_controller_n();
+    for (const auto& [mid, pt_demux_enabled] : transports_to_update) {
+      RtpTransportInternal* rtp_transport =
+          transport_controller->GetRtpTransport(mid);
+      if (rtp_transport != nullptr) {
+        rtp_transport->SetActivePayloadTypeDemuxing(pt_demux_enabled);
       }
     }
-    return true;
   });
+
+  // If payload type demuxing is disabled for a transport, clear out any
+  // unsignaled streams that might have been created previously based on
+  // payload type matches. This prevents SSRC collisions if those SSRCs
+  // are later reused.
+  // TODO(tommi): Could/should we rather issue the calls to the transceivers
+  // on the network thread? This will post tasks per transceiver to the worker
+  // thread. A common configuration will have joined worker and network threads,
+  // so bundling with the network thread may be more efficient.
+  for (RtpTransceiver* transceiver : transceivers()->ListInternal()) {
+    const ContentInfo* content =
+        FindMediaSectionForTransceiver(transceiver, sdesc);
+    if (!content) {
+      continue;
+    }
+    auto it = transports_to_update.find(content->mid());
+    if (it != transports_to_update.end() && !it->second) {
+      // TODO: bugs.webrtc.org/42221580 - This will remove *all* unsignaled
+      // streams (those without an explicitly signaled SSRC), which may include
+      // streams that were matched to this channel by MID or RID. Ideally we'd
+      // remove only the streams that were matched based on payload type alone,
+      // but currently there is no straightforward way to identify those
+      // streams.
+      transceiver->ResetUnsignaledRecvStream();
+    }
+  }
 }
 
 bool SdpOfferAnswerHandler::ConfiguredForMedia() const {
diff --git a/pc/sdp_offer_answer.h b/pc/sdp_offer_answer.h
index 7be0f6626a..802f518fcf 100644
--- a/pc/sdp_offer_answer.h
+++ b/pc/sdp_offer_answer.h
@@ -574,7 +574,7 @@ class SdpOfferAnswerHandler : public SdpStateProvider {
 
   // Based on number of transceivers per media type, enabled or disable
   // payload type based demuxing in the affected channels.
-  bool UpdatePayloadTypeDemuxingState(
+  void UpdatePayloadTypeDemuxingState(
       ContentSource source,
       const flat_map<std::string, const ContentGroup*>& bundle_groups_by_mid);
 
diff --git a/pc/test/mock_channel_interface.h b/pc/test/mock_channel_interface.h
index b87ab9dc6a..2848905e89 100644
--- a/pc/test/mock_channel_interface.h
+++ b/pc/test/mock_channel_interface.h
@@ -81,7 +81,6 @@ class MockChannelInterface : public ChannelInterface {
               SetRemoteContent,
               (const webrtc::MediaContentDescription*, SdpType),
               (override));
-  MOCK_METHOD(bool, SetPayloadTypeDemuxingEnabled, (bool), (override));
   MOCK_METHOD(const std::vector<StreamParams>&,
               local_streams,
               (),

From ca896b7ffef011bbf6957c99d413c5aac602c99f Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Thu, 2 Apr 2026 13:06:38 +0200
Subject: [PATCH 466/472] Fix rebase issue of two colliding CLs

Remove the task_safety_ member variable from the video engine header to
resolve a regression introduced during a rebase of two colliding
changes.

Using no-try after verifying that compilation issues have been addressed
so that we can open the tree again.

No-try: true
Bug: none
Change-Id: I5ec3299806c4df8424dfded3b592f3709ba58685
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/461541
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47319}
---
 media/engine/webrtc_video_engine.h | 1 -
 1 file changed, 1 deletion(-)

diff --git a/media/engine/webrtc_video_engine.h b/media/engine/webrtc_video_engine.h
index 334a871168..8c5442b046 100644
--- a/media/engine/webrtc_video_engine.h
+++ b/media/engine/webrtc_video_engine.h
@@ -771,7 +771,6 @@ class WebRtcVideoReceiveChannel : public MediaChannelUtil,
       RTC_GUARDED_BY(thread_checker_);
 
   const int receive_buffer_size_;
-  ScopedTaskSafety task_safety_;
 };
 
 // Keeping the old name "WebRtcVideoChannel" around because some external

From 1822467bec1c91b8d782e9c7f24342d1d65f01f2 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Henrik=20Bostr=C3=B6m?= <hbos@google.com>
Date: Thu, 9 Apr 2026 12:19:32 +0200
Subject: [PATCH 467/472] [Merge-148] Cherry pick "Move the NullVideoDecoder
 into a separate file and target."
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Original change's description:
> Move the NullVideoDecoder into a separate file and target.
>
> This is necessary because the Chromium video decoder factory needs to
> be able to instantiate it for fallback purposes (e.g. Decode error
> causing the HW to no longer be available).
>
> Bug: chromium:500960863
> Change-Id: I724a5f28aa0a0a615972dd49eebde701c3b7ec90
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/462243
> Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#47366}

Bug: chromium:500960863
Change-Id: I724a5f28aa0a0a615972dd49eebde701c3b7ec90
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/464521
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/branch-heads/7778@{#1}
Cr-Branched-From: ca896b7ffef011bbf6957c99d413c5aac602c99f-refs/heads/main@{#47319}
---
 video/BUILD.gn                 | 18 +++++++++++++
 video/null_video_decoder.cc    | 48 ++++++++++++++++++++++++++++++++++
 video/null_video_decoder.h     | 39 +++++++++++++++++++++++++++
 video/video_receive_stream2.cc | 34 +++---------------------
 4 files changed, 108 insertions(+), 31 deletions(-)
 create mode 100644 video/null_video_decoder.cc
 create mode 100644 video/null_video_decoder.h

diff --git a/video/BUILD.gn b/video/BUILD.gn
index f7f773e62b..766f6b2fb0 100644
--- a/video/BUILD.gn
+++ b/video/BUILD.gn
@@ -73,6 +73,7 @@ rtc_library("video") {
     ":frame_cadence_adapter",
     ":frame_decode_scheduler",
     ":frame_dumping_decoder",
+    ":null_video_decoder",
     ":task_queue_frame_decode_scheduler",
     ":unique_timestamp_counter",
     ":video_stream_buffer_controller",
@@ -191,6 +192,23 @@ rtc_library("video") {
   }
 }
 
+rtc_library("null_video_decoder") {
+  visibility = [ "*" ]
+
+  sources = [
+    "null_video_decoder.cc",
+    "null_video_decoder.h",
+  ]
+
+  deps = [
+    "../api/video:encoded_image",
+    "../api/video_codecs:video_codecs_api",
+    "../modules/video_coding:video_codec_interface",
+    "../rtc_base:logging",
+    "../rtc_base/system:rtc_export",
+  ]
+}
+
 rtc_library("frame_dumping_decoder") {
   visibility = [ "*" ]
 
diff --git a/video/null_video_decoder.cc b/video/null_video_decoder.cc
new file mode 100644
index 0000000000..3ce581204a
--- /dev/null
+++ b/video/null_video_decoder.cc
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2026 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video/null_video_decoder.h"
+
+#include <cstdint>
+
+#include "api/video/encoded_image.h"
+#include "api/video_codecs/video_decoder.h"
+#include "modules/video_coding/include/video_error_codes.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+bool NullVideoDecoder::Configure(const Settings& settings) {
+  RTC_LOG(LS_ERROR) << "Can't initialize NullVideoDecoder.";
+  return true;
+}
+
+int32_t NullVideoDecoder::Decode(const EncodedImage& input_image,
+                                 int64_t render_time_ms) {
+  RTC_LOG(LS_ERROR) << "The NullVideoDecoder doesn't support decoding.";
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t NullVideoDecoder::RegisterDecodeCompleteCallback(
+    DecodedImageCallback* callback) {
+  RTC_LOG(LS_ERROR)
+      << "Can't register decode complete callback on NullVideoDecoder.";
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t NullVideoDecoder::Release() {
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+const char* NullVideoDecoder::ImplementationName() const {
+  return "NullVideoDecoder";
+}
+
+}  // namespace webrtc
diff --git a/video/null_video_decoder.h b/video/null_video_decoder.h
new file mode 100644
index 0000000000..3ed049c839
--- /dev/null
+++ b/video/null_video_decoder.h
@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2026 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef VIDEO_NULL_VIDEO_DECODER_H_
+#define VIDEO_NULL_VIDEO_DECODER_H_
+
+#include <cstdint>
+
+#include "api/video/encoded_image.h"
+#include "api/video_codecs/video_decoder.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// The decoder used when there is no real decoder implementation available.
+class RTC_EXPORT NullVideoDecoder : public VideoDecoder {
+ public:
+  // These are NO-OPs except for RTC_LOG lines.
+  bool Configure(const Settings& settings) override;
+  int32_t Decode(const EncodedImage& input_image,
+                 int64_t render_time_ms) override;
+  int32_t RegisterDecodeCompleteCallback(
+      DecodedImageCallback* callback) override;
+  int32_t Release() override;
+
+  // This is exposed in getStats() as "decoderImplementation".
+  const char* ImplementationName() const override;
+};
+
+}  // namespace webrtc
+
+#endif  // VIDEO_NULL_VIDEO_DECODER_H_
diff --git a/video/video_receive_stream2.cc b/video/video_receive_stream2.cc
index 8e3f6a8132..79624b6f51 100644
--- a/video/video_receive_stream2.cc
+++ b/video/video_receive_stream2.cc
@@ -88,6 +88,7 @@
 #include "video/decode_synchronizer.h"
 #include "video/frame_decode_scheduler.h"
 #include "video/frame_dumping_decoder.h"
+#include "video/null_video_decoder.h"
 #include "video/receive_statistics_proxy.h"
 #include "video/render/incoming_video_stream.h"
 #include "video/task_queue_frame_decode_scheduler.h"
@@ -170,33 +171,6 @@ RenderResolution InitialDecoderResolution(const FieldTrialsView& field_trials) {
   return RenderResolution(320, 180);
 }
 
-// Video decoder class to be used for unknown codecs. Doesn't support decoding
-// but logs messages to LS_ERROR.
-class NullVideoDecoder : public VideoDecoder {
- public:
-  bool Configure(const Settings& settings) override {
-    RTC_LOG(LS_ERROR) << "Can't initialize NullVideoDecoder.";
-    return true;
-  }
-
-  int32_t Decode(const EncodedImage& input_image,
-                 int64_t render_time_ms) override {
-    RTC_LOG(LS_ERROR) << "The NullVideoDecoder doesn't support decoding.";
-    return WEBRTC_VIDEO_CODEC_OK;
-  }
-
-  int32_t RegisterDecodeCompleteCallback(
-      DecodedImageCallback* callback) override {
-    RTC_LOG(LS_ERROR)
-        << "Can't register decode complete callback on NullVideoDecoder.";
-    return WEBRTC_VIDEO_CODEC_OK;
-  }
-
-  int32_t Release() override { return WEBRTC_VIDEO_CODEC_OK; }
-
-  const char* ImplementationName() const override { return "NullVideoDecoder"; }
-};
-
 bool IsKeyFrameAndUnspecifiedResolution(const EncodedFrame& frame) {
   return frame.IsKey() && frame.EncodedImage()._encodedWidth == 0 &&
          frame.EncodedImage()._encodedHeight == 0;
@@ -555,10 +529,8 @@ void VideoReceiveStream2::CreateAndRegisterExternalDecoder(
                "VideoReceiveStream2::CreateAndRegisterExternalDecoder");
   std::unique_ptr<VideoDecoder> video_decoder =
       config_.decoder_factory->Create(env_, decoder.video_format);
-  // If we still have no valid decoder, we have to create a "Null" decoder
-  // that ignores all calls. The reason we can get into this state is that the
-  // old decoder factory interface doesn't have a way to query supported
-  // codecs.
+  // The factory can end up in this state either if the format is not supported
+  // or because a creation step failed, e.g. HW is unavailable.
   if (!video_decoder) {
     video_decoder = std::make_unique<NullVideoDecoder>();
   }

From 2c3c984b00a5b6e15f661acfae23a512c81d5030 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Thu, 16 Apr 2026 14:48:05 +0200
Subject: [PATCH 468/472] [M148] Allow signaled SSRCs to override learned
 bindings in RtpDemuxer
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Update RtpDemuxer to permit signaled SSRCs to override existing learned
SSRC bindings. This prevents registration failures during session
renegotiation when an SSRC, previously identified through payload type
demuxing, is subsequently explicitly signaled.

The demuxer now maintains a set of signaled SSRCs to differentiate
explicit configurations from learned state. Conflict detection is
modified to allow these overrides while maintaining strict checks
between multiple signaled SSRCs. This ensures that explicit signaling
always takes precedence over internally learned bindings, resolving
routing issues during media stream updates.

(cherry picked from commit 186d42a3010981882a8679a145ce36aa84581c78)

Fixed: chromium:503666505
Bug: webrtc:502130956
Change-Id: I551c2b7ee5af18fc9792ee52be08c5349c20b641
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/464220
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Original-Commit-Position: refs/heads/main@{#47446}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/464764
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/branch-heads/7778@{#2}
Cr-Branched-From: ca896b7ffef011bbf6957c99d413c5aac602c99f-refs/heads/main@{#47319}
---
 call/rtp_demuxer.cc          | 22 +++++++---
 call/rtp_demuxer.h           |  1 +
 call/rtp_demuxer_unittest.cc | 83 ++++++++++++++++++++++++++++++++++++
 3 files changed, 101 insertions(+), 5 deletions(-)

diff --git a/call/rtp_demuxer.cc b/call/rtp_demuxer.cc
index 598764b71c..e078347544 100644
--- a/call/rtp_demuxer.cc
+++ b/call/rtp_demuxer.cc
@@ -161,7 +161,13 @@ bool RtpDemuxer::AddSink(const RtpDemuxerCriteria& criteria,
   }
 
   for (uint32_t ssrc : criteria.ssrcs()) {
-    sink_by_ssrc_.emplace(ssrc, sink);
+    auto result = sink_by_ssrc_.emplace(ssrc, sink);
+    if (!result.second) {
+      if (signaled_ssrcs_.find(ssrc) == signaled_ssrcs_.end()) {
+        result.first->second = sink;
+      }
+    }
+    signaled_ssrcs_.insert(ssrc);
   }
 
   for (uint8_t payload_type : criteria.payload_types()) {
@@ -231,10 +237,12 @@ bool RtpDemuxer::CriteriaWouldConflict(
   for (uint32_t ssrc : criteria.ssrcs()) {
     const auto sink_by_ssrc = sink_by_ssrc_.find(ssrc);
     if (sink_by_ssrc != sink_by_ssrc_.end()) {
-      RTC_LOG(LS_INFO) << criteria.ToString()
-                       << " would conflict with existing sink = "
-                       << sink_by_ssrc->second << " binding by SSRC=" << ssrc;
-      return true;
+      if (signaled_ssrcs_.find(ssrc) != signaled_ssrcs_.end()) {
+        RTC_LOG(LS_INFO) << criteria.ToString()
+                         << " would conflict with existing sink = "
+                         << sink_by_ssrc->second << " binding by SSRC=" << ssrc;
+        return true;
+      }
     }
   }
 
@@ -270,6 +278,10 @@ void RtpDemuxer::AddSink(absl::string_view rsid, RtpPacketSinkInterface* sink) {
 
 bool RtpDemuxer::RemoveSink(const RtpPacketSinkInterface* sink) {
   RTC_DCHECK(sink);
+  flat_set<uint32_t> ssrcs = GetSsrcsForSink(sink);
+  for (uint32_t ssrc : ssrcs) {
+    signaled_ssrcs_.erase(ssrc);
+  }
   size_t num_removed = RemoveFromMapByValue(&sink_by_mid_, sink) +
                        RemoveFromMapByValue(&sink_by_ssrc_, sink) +
                        RemoveFromMultimapByValue(&sinks_by_pt_, sink) +
diff --git a/call/rtp_demuxer.h b/call/rtp_demuxer.h
index cc6a84c878..f2f76cb562 100644
--- a/call/rtp_demuxer.h
+++ b/call/rtp_demuxer.h
@@ -217,6 +217,7 @@ class RtpDemuxer {
   flat_map<std::pair<std::string, std::string>, RtpPacketSinkInterface*>
       sink_by_mid_and_rsid_;
   flat_map<std::string, RtpPacketSinkInterface*> sink_by_rsid_;
+  flat_set<uint32_t> signaled_ssrcs_;
 
   // Tracks all the MIDs that have been identified in added criteria. Used to
   // determine if a packet should be dropped right away because the MID is
diff --git a/call/rtp_demuxer_unittest.cc b/call/rtp_demuxer_unittest.cc
index 363d58aa13..b878d2bbae 100644
--- a/call/rtp_demuxer_unittest.cc
+++ b/call/rtp_demuxer_unittest.cc
@@ -369,6 +369,89 @@ TEST_F(RtpDemuxerTest, OnRtpPacketCalledOnCorrectSinkByPayloadType) {
   EXPECT_TRUE(demuxer_.OnRtpPacket(*packet));
 }
 
+TEST_F(RtpDemuxerTest, DontSignalRtpPayloadTypeWhenPtDemuxingDisabled) {
+  constexpr uint32_t ssrc1 = 10;
+  constexpr uint32_t ssrc2 = 11;
+  constexpr uint8_t pt1 = 30;
+  constexpr uint8_t pt2 = 31;
+
+  // Sink 1 registered when PT demuxing is enabled (default).
+  MockRtpPacketSink sink1;
+  RtpDemuxerCriteria criteria1;
+  criteria1.payload_types() = {pt1};
+  EXPECT_TRUE(AddSink(criteria1, &sink1));
+
+  // Disable PT demuxing.
+  demuxer_.set_use_payload_type_demuxing(false);
+
+  // Sink 2 registered when PT demuxing is disabled.
+  MockRtpPacketSink sink2;
+  RtpDemuxerCriteria criteria2;
+  criteria2.payload_types() = {pt2};
+  EXPECT_TRUE(AddSink(criteria2, &sink2));
+
+  // Packet with pt1 should not go to sink1 because fallback is disabled.
+  auto packet1 = CreatePacketWithSsrc(ssrc1);
+  packet1->SetPayloadType(pt1);
+  EXPECT_CALL(sink1, OnRtpPacket(_)).Times(0);
+  EXPECT_FALSE(demuxer_.OnRtpPacket(*packet1));
+
+  // Packet with pt2 should not go to sink2 because fallback is disabled and it
+  // was not registered by PT.
+  auto packet2 = CreatePacketWithSsrc(ssrc2);
+  packet2->SetPayloadType(pt2);
+  EXPECT_CALL(sink2, OnRtpPacket(_)).Times(0);
+  EXPECT_FALSE(demuxer_.OnRtpPacket(*packet2));
+}
+
+TEST_F(RtpDemuxerTest, DynamicPayloadTypeDemuxing) {
+  constexpr uint32_t ssrc = 10;
+  constexpr uint8_t payload_type = 30;
+
+  MockRtpPacketSink sink;
+  RtpDemuxerCriteria criteria;
+  criteria.payload_types() = {payload_type};
+
+  demuxer_.set_use_payload_type_demuxing(false);
+  EXPECT_TRUE(AddSink(criteria, &sink));
+
+  auto packet = CreatePacketWithSsrc(ssrc);
+  packet->SetPayloadType(payload_type);
+
+  EXPECT_CALL(sink, OnRtpPacket(_)).Times(0);
+  EXPECT_FALSE(demuxer_.OnRtpPacket(*packet));
+
+  demuxer_.set_use_payload_type_demuxing(true);
+
+  EXPECT_CALL(sink, OnRtpPacket(SamePacketAs(*packet))).Times(1);
+  EXPECT_TRUE(demuxer_.OnRtpPacket(*packet));
+}
+
+TEST_F(RtpDemuxerTest, SignaledSsrcOverridesLearnedBinding) {
+  constexpr uint32_t ssrc = 10;
+  constexpr uint8_t payload_type = 30;
+
+  MockRtpPacketSink sink1;
+  RtpDemuxerCriteria criteria1;
+  criteria1.payload_types() = {payload_type};
+  EXPECT_TRUE(AddSink(criteria1, &sink1));
+
+  auto packet1 = CreatePacketWithSsrc(ssrc);
+  packet1->SetPayloadType(payload_type);
+  EXPECT_CALL(sink1, OnRtpPacket(SamePacketAs(*packet1))).Times(1);
+  EXPECT_TRUE(demuxer_.OnRtpPacket(*packet1));
+
+  MockRtpPacketSink sink2;
+  RtpDemuxerCriteria criteria2;
+  criteria2.ssrcs().insert(ssrc);
+  EXPECT_TRUE(AddSink(criteria2, &sink2));
+
+  auto packet2 = CreatePacketWithSsrc(ssrc);
+  EXPECT_CALL(sink1, OnRtpPacket(_)).Times(0);
+  EXPECT_CALL(sink2, OnRtpPacket(SamePacketAs(*packet2))).Times(1);
+  EXPECT_TRUE(demuxer_.OnRtpPacket(*packet2));
+}
+
 TEST_F(RtpDemuxerTest, PacketsDeliveredInRightOrder) {
   constexpr uint32_t ssrc = 101;
   MockRtpPacketSink sink;

From 9c2497e795477d16dac808e12d7f53f12e90ebc6 Mon Sep 17 00:00:00 2001
From: Jakob Ivarsson <jakobi@webrtc.org>
Date: Thu, 16 Apr 2026 10:35:57 +0000
Subject: [PATCH 469/472] [M148] Check vector sizes when crossfading from
 CNG/expand to normal.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Original change's description:
> Check vector sizes when crossfading from CNG/expand to normal.
>
> This fixes a potential out-of-bounds write.
>
> Bug: chromium:502661101
> Change-Id: I6f03b522643d7a55040d7f5403f342b32d47f0c7
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/464500
> Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#47447}

(cherry picked from commit ff1be2e8524fe63069fc2406d5e3f3ddccafee1a)

Bug: chromium:504500274,chromium:502661101
Change-Id: I6f03b522643d7a55040d7f5403f342b32d47f0c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/465220
Commit-Queue: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Auto-Submit: Chrome Cherry Picker <chrome-cherry-picker@chops-service-accounts.iam.gserviceaccount.com>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/branch-heads/7778@{#3}
Cr-Branched-From: ca896b7ffef011bbf6957c99d413c5aac602c99f-refs/heads/main@{#47319}
---
 modules/audio_coding/neteq/normal.cc          | 57 ++++++++-----------
 modules/audio_coding/neteq/normal.h           |  4 --
 modules/audio_coding/neteq/normal_unittest.cc | 15 +++++
 3 files changed, 39 insertions(+), 37 deletions(-)

diff --git a/modules/audio_coding/neteq/normal.cc b/modules/audio_coding/neteq/normal.cc
index 117c480f8c..be727fb799 100644
--- a/modules/audio_coding/neteq/normal.cc
+++ b/modules/audio_coding/neteq/normal.cc
@@ -22,12 +22,32 @@
 #include "common_audio/signal_processing/include/spl_inl.h"
 #include "modules/audio_coding/codecs/cng/webrtc_cng.h"
 #include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "modules/audio_coding/neteq/audio_vector.h"
 #include "modules/audio_coding/neteq/background_noise.h"
 #include "modules/audio_coding/neteq/decoder_database.h"
 #include "modules/audio_coding/neteq/expand.h"
 #include "rtc_base/checks.h"
 
 namespace webrtc {
+namespace {
+
+void Crossfade(const AudioVector& from, AudioVector& to, size_t win_length) {
+  const size_t win_length_clamped =
+      std::min({win_length, from.Size(), to.Size()});
+  if (win_length_clamped == 0) {
+    return;
+  }
+  int16_t win_slope_Q14 = (1 << 14) / static_cast<int16_t>(win_length_clamped);
+  int16_t win_up_Q14 = 0;
+  for (size_t i = 0; i < win_length_clamped; i++) {
+    win_up_Q14 += win_slope_Q14;
+    to[i] =
+        (win_up_Q14 * to[i] + ((1 << 14) - win_up_Q14) * from[i] + (1 << 13)) >>
+        14;
+  }
+}
+
+}  // namespace
 
 int Normal::Process(const int16_t* input,
                     size_t length,
@@ -138,23 +158,7 @@ int Normal::Process(const int16_t* input,
 
       // Interpolate the expanded data into the new vector.
       // (NB/WB/SWB32/SWB48 8/16/32/48 samples.)
-      size_t win_length = samples_per_ms_;
-      int16_t win_slope_Q14 = default_win_slope_Q14_;
-      RTC_DCHECK_LT(channel_ix, output->Channels());
-      if (win_length > output->Size()) {
-        win_length = output->Size();
-        win_slope_Q14 = (1 << 14) / static_cast<int16_t>(win_length);
-      }
-      int16_t win_up_Q14 = 0;
-      for (size_t i = 0; i < win_length; i++) {
-        win_up_Q14 += win_slope_Q14;
-        (*output)[channel_ix][i] =
-            (win_up_Q14 * (*output)[channel_ix][i] +
-             ((1 << 14) - win_up_Q14) * expanded[channel_ix][i] + (1 << 13)) >>
-            14;
-      }
-      RTC_DCHECK_GT(win_up_Q14,
-                    (1 << 14) - 32);  // Worst case rouding is a length of 34
+      Crossfade(expanded[channel_ix], (*output)[channel_ix], samples_per_ms_);
     }
   } else if (last_mode == NetEq::Mode::kRfc3389Cng) {
     RTC_DCHECK_EQ(output->Channels(), 1);  // Not adapted for multi-channel yet.
@@ -176,22 +180,9 @@ int Normal::Process(const int16_t* input,
     }
     // Interpolate the CNG into the new vector.
     // (NB/WB/SWB32/SWB48 8/16/32/48 samples.)
-    size_t win_length = samples_per_ms_;
-    int16_t win_slope_Q14 = default_win_slope_Q14_;
-    if (win_length > kCngLength) {
-      win_length = kCngLength;
-      win_slope_Q14 = (1 << 14) / static_cast<int16_t>(win_length);
-    }
-    int16_t win_up_Q14 = 0;
-    for (size_t i = 0; i < win_length; i++) {
-      win_up_Q14 += win_slope_Q14;
-      (*output)[0][i] =
-          (win_up_Q14 * (*output)[0][i] +
-           ((1 << 14) - win_up_Q14) * cng_output[i] + (1 << 13)) >>
-          14;
-    }
-    RTC_DCHECK_GT(win_up_Q14,
-                  (1 << 14) - 32);  // Worst case rouding is a length of 34
+    AudioVector temp_vector(kCngLength);
+    temp_vector.OverwriteAt(cng_output, kCngLength, 0);
+    Crossfade(temp_vector, (*output)[0], samples_per_ms_);
   }
 
   return static_cast<int>(length);
diff --git a/modules/audio_coding/neteq/normal.h b/modules/audio_coding/neteq/normal.h
index e6c918764f..d0dfc2f6cb 100644
--- a/modules/audio_coding/neteq/normal.h
+++ b/modules/audio_coding/neteq/normal.h
@@ -17,7 +17,6 @@
 #include "api/neteq/neteq.h"
 #include "modules/audio_coding/neteq/statistics_calculator.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/numerics/safe_conversions.h"
 
 namespace webrtc {
 
@@ -42,8 +41,6 @@ class Normal {
         background_noise_(background_noise),
         expand_(expand),
         samples_per_ms_(CheckedDivExact(fs_hz_, 1000)),
-        default_win_slope_Q14_(
-            dchecked_cast<uint16_t>((1 << 14) / samples_per_ms_)),
         statistics_(statistics) {}
 
   virtual ~Normal() {}
@@ -68,7 +65,6 @@ class Normal {
   const BackgroundNoise& background_noise_;
   Expand* expand_;
   const size_t samples_per_ms_;
-  const int16_t default_win_slope_Q14_;
   StatisticsCalculator* const statistics_;
 };
 
diff --git a/modules/audio_coding/neteq/normal_unittest.cc b/modules/audio_coding/neteq/normal_unittest.cc
index 272869d701..c29a2c3314 100644
--- a/modules/audio_coding/neteq/normal_unittest.cc
+++ b/modules/audio_coding/neteq/normal_unittest.cc
@@ -14,6 +14,7 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <vector>
 
 #include "api/neteq/neteq.h"
 #include "api/neteq/tick_timer.h"
@@ -147,6 +148,20 @@ TEST(Normal, LastModeExpand120msPacket) {
   EXPECT_CALL(expand, Die());  // Called when `expand` goes out of scope.
 }
 
+TEST(Normal, LastModeRfc3389CngSmallInput) {
+  constexpr size_t kChannels = 1;
+  constexpr size_t kInputFrames = 10;
+  MockDecoderDatabase db;
+  Normal normal(/*fs_hz=*/48000, /*decoder_database=*/&db,
+                /*background_noise=*/BackgroundNoise(kChannels),
+                /*expand=*/nullptr, /*statistics=*/nullptr);
+  AudioMultiVector output(kChannels);
+  std::vector<int16_t> input(kChannels * kInputFrames, 0);
+  EXPECT_EQ(normal.Process(input.data(), input.size(), NetEq::Mode::kRfc3389Cng,
+                           &output),
+            static_cast<int>(input.size()));
+}
+
 // TODO(hlundin): Write more tests.
 
 }  // namespace webrtc

From 3fda3224b32e72ebaab8c67adc61aad0814a4a58 Mon Sep 17 00:00:00 2001
From: Tommi <tommi@webrtc.org>
Date: Mon, 20 Apr 2026 10:25:57 +0200
Subject: [PATCH 470/472] [M148] Relax RTP header extension ID reuse checks in
 RtpTransport

Removing the historical_rtp_header_extensions_ list and instead only
check header_extensions_by_mid_ for conflicts with active extensions.

Also borrowing a test for this provided by philipp.hancke.

(cherry picked from commit 8915e3564c0593501ba1a0ffa1ac08b8cb7cd00f)

Fixed: chromium:504980502
Bug: webrtc:503013383
Change-Id: I08874edb1d46d638641c74255fe5379622a43c0e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/464501
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Original-Commit-Position: refs/heads/main@{#47472}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/465760
Cr-Commit-Position: refs/branch-heads/7778@{#4}
Cr-Branched-From: ca896b7ffef011bbf6957c99d413c5aac602c99f-refs/heads/main@{#47319}
---
 pc/channel_unittest.cc                   |  6 ++-
 pc/peer_connection_simulcast_unittest.cc | 55 ++++++++++++++++++++++++
 pc/rtp_transport.cc                      | 40 +++++++----------
 pc/rtp_transport.h                       |  3 --
 pc/rtp_transport_unittest.cc             | 22 ++++++++++
 5 files changed, 97 insertions(+), 29 deletions(-)

diff --git a/pc/channel_unittest.cc b/pc/channel_unittest.cc
index 8a91c4750a..62ea4d8f72 100644
--- a/pc/channel_unittest.cc
+++ b/pc/channel_unittest.cc
@@ -769,8 +769,10 @@ class ChannelTest : public ::testing::Test {
     });
     RTCError error =
         channel1_->SetLocalContent(&local_updated, SdpType::kOffer);
-    EXPECT_FALSE(error.ok());
-    EXPECT_THAT(error.message(), HasSubstr("RTP extension ID reassignment"));
+    // Expected to succeed because the mapping for ID 1 was cleared by the
+    // previous SetLocalContent call (which set extensions to empty).
+    // RtpTransport allows reuse when not in use by any active MID.
+    EXPECT_TRUE(error.ok());
   }
 
   // Test that SetLocalContent and SetRemoteContent properly configure
diff --git a/pc/peer_connection_simulcast_unittest.cc b/pc/peer_connection_simulcast_unittest.cc
index 2cd0df043b..e8cd73f95d 100644
--- a/pc/peer_connection_simulcast_unittest.cc
+++ b/pc/peer_connection_simulcast_unittest.cc
@@ -556,4 +556,59 @@ TEST_F(PeerConnectionSimulcastTests, SimulcastSldModificationRejected) {
   EXPECT_TRUE(modified_offer);
   EXPECT_TRUE(local->SetLocalDescription(std::move(modified_offer)));
 }
+
+// Reproduces the bug reported by @ibc where RTP extension IDs are reassigned
+// to different URIs in subsequent offers, causing SetLocalDescription to fail.
+TEST_F(PeerConnectionSimulcastTests,
+       NoRtpExtensionIdReassignmentWhenAddingTransceiver) {
+  auto local = CreatePeerConnectionWrapper();
+  auto layers = CreateLayers({"f", "h", "q"}, true);
+
+  // Add video transceiver with simulcast.
+  AddTransceiver(local.get(), layers);
+  ASSERT_TRUE(local->CreateOfferAndSetAsLocal());
+
+  // Set remote answer without header extensions.
+  std::string remote_answer_sdp =
+      "v=0\r\n"
+      "o=- 8403615332048243445 2 IN IP4 127.0.0.1\r\n"
+      "s=-\r\n"
+      "t=0 0\r\n"
+      "a=group:BUNDLE 0\r\n"
+      "m=video 9 UDP/TLS/RTP/SAVPF 96\r\n"
+      "c=IN IP4 0.0.0.0\r\n"
+      "a=mid:0\r\n"
+      "a=ice-ufrag:IZeV\r\n"
+      "a=ice-pwd:uaZhQD4rYM/Tta2qWBT1Bbt4\r\n"
+      "a=fingerprint:sha-256 "
+      "D8:6C:3D:FA:23:E2:2C:63:11:2D:D0:86:BE:C4:D0:65:F9:42:F7:1C:06:04:27:E6:"
+      "1C:2C:74:01:8D:50:67:23\r\n"
+      "a=setup:active\r\n"
+      "a=rtcp-mux\r\n"
+      "a=extmap:9 urn:ietf:params:rtp-hdrext:sdes:mid\r\n"
+      "a=extmap:10 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id\r\n"
+      "a=extmap:11 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id\r\n"
+      "a=extmap:2 "
+      "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\r\n"
+      "a=extmap:4 "
+      "http://www.ietf.org/id/"
+      "draft-holmer-rmcat-transport-wide-cc-extensions-01\r\n"
+      "a=extmap:3 urn:3gpp:video-orientation\r\n"
+      "a=extmap:1 urn:ietf:params:rtp-hdrext:toffset\r\n"
+      "a=extmap:5 "
+      "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay\r\n"
+      "a=rtpmap:96 VP8/90000\r\n";
+  // Answer recvonly.
+  ASSERT_TRUE(local->SetRemoteDescription(CreateSessionDescription(
+      SdpType::kAnswer, remote_answer_sdp + "a=recvonly\r\n")));
+
+  ASSERT_TRUE(local->CreateOfferAndSetAsLocal());
+  // Answer inactive.
+  ASSERT_TRUE(local->SetRemoteDescription(CreateSessionDescription(
+      SdpType::kAnswer, remote_answer_sdp + "a=inactive\r\n")));
+
+  // Add an audio transceiver.
+  local->AddAudioTrack("audio");
+  EXPECT_TRUE(local->CreateOfferAndSetAsLocal());
+}
 }  // namespace webrtc
diff --git a/pc/rtp_transport.cc b/pc/rtp_transport.cc
index 05ef651486..086ea72869 100644
--- a/pc/rtp_transport.cc
+++ b/pc/rtp_transport.cc
@@ -242,15 +242,22 @@ RTCError RtpTransport::VerifyRtpHeaderExtensionMap(
     if (new_extension.id == 0) {
       continue;
     }
-    auto it = absl::c_find_if(
-        historical_rtp_header_extensions_,
-        [&](const RtpExtension& ext) { return ext.id == new_extension.id; });
-    if (it != historical_rtp_header_extensions_.end() &&
-        it->uri != new_extension.uri) {
-      return RTCError::InvalidParameter()
-             << "RTP extension ID reassignment not supported (id="
-             << new_extension.id << ", old_uri=\"" << it->uri
-             << "\", new_uri=\"" << new_extension.uri << "\").";
+    // TODO: bugs.webrtc.org/503013383 - Introduce checking against IDs that are
+    // currently not present in the SDP, but have been used in previous
+    // negotiation rounds. Reusing extensions with a different ID is a protocol
+    // violation, but we cannot check this until we check against the same
+    // protocol violation on the sender side.
+    for (const auto& [mid, active_extensions] : header_extensions_by_mid_) {
+      auto it = absl::c_find_if(
+          active_extensions,
+          [&](const RtpExtension& ext) { return ext.id == new_extension.id; });
+      if (it != active_extensions.end() && it->uri != new_extension.uri) {
+        return RTCError::InvalidParameter()
+               << "RTP extension ID reassignment not supported (collision on "
+                  "active MID "
+               << mid << ", id=" << new_extension.id << ", old_uri=\""
+               << it->uri << "\", new_uri=\"" << new_extension.uri << "\").";
+      }
     }
   }
 
@@ -275,21 +282,6 @@ RTCError RtpTransport::RegisterRtpHeaderExtensionMap(
     return RTCError::OK();
   }
 
-  for (const RtpExtension& extension : extensions) {
-    if (extension.id == 0) {
-      continue;
-    }
-    auto it = absl::c_find_if(historical_rtp_header_extensions_,
-                              [&extension](const RtpExtension& ext) {
-                                return ext.id == extension.id;
-                              });
-    if (it == historical_rtp_header_extensions_.end()) {
-      historical_rtp_header_extensions_.push_back(extension);
-    } else {
-      RTC_DCHECK_EQ(it->uri, extension.uri);
-    }
-  }
-
   RemoveExtensionMapForMid(mid, header_extensions_by_mid_);
   header_extensions_by_mid_.emplace_back(std::string(mid), extensions);
 
diff --git a/pc/rtp_transport.h b/pc/rtp_transport.h
index e51a1247fa..185201586c 100644
--- a/pc/rtp_transport.h
+++ b/pc/rtp_transport.h
@@ -181,9 +181,6 @@ class RtpTransport : public RtpTransportInternal {
   std::vector<std::pair<std::string, RtpHeaderExtensions>>
       header_extensions_by_mid_ RTC_GUARDED_BY(network_thread_checker_);
 
-  RtpHeaderExtensions historical_rtp_header_extensions_
-      RTC_GUARDED_BY(network_thread_checker_);
-
   // Guard against recursive "ready to send" signals
   bool processing_ready_to_send_ = false;
   RTC_NO_UNIQUE_ADDRESS SequenceChecker network_thread_checker_;
diff --git a/pc/rtp_transport_unittest.cc b/pc/rtp_transport_unittest.cc
index 65762faec8..07430e6427 100644
--- a/pc/rtp_transport_unittest.cc
+++ b/pc/rtp_transport_unittest.cc
@@ -390,6 +390,28 @@ TEST(RtpTransportTest, VerifyRtpHeaderExtensionMapRejectsIdReassignment) {
   EXPECT_EQ(error.type(), RTCErrorType::INVALID_PARAMETER);
 }
 
+TEST(RtpTransportTest,
+     VerifyRtpHeaderExtensionMapAllowsIdReuseAfterUnregister) {
+  test::RunLoop loop;
+  RtpTransport transport(kMuxDisabled, CreateTestFieldTrials());
+  RtpHeaderExtensions extensions1 = {
+      RtpExtension("urn:ietf:params:rtp-hdrext:ssrc-audio-level", 1)};
+  RtpHeaderExtensions extensions2 = {RtpExtension(
+      "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time", 1)};
+
+  // Registering the first map should succeed.
+  EXPECT_TRUE(
+      transport.RegisterRtpHeaderExtensionMap("audio", extensions1).ok());
+
+  // Unregister the first map.
+  transport.UnregisterRtpHeaderExtensionMap("audio");
+
+  // Registering the second map with same ID but different URI should now
+  // succeed!
+  EXPECT_TRUE(
+      transport.RegisterRtpHeaderExtensionMap("video", extensions2).ok());
+}
+
 // Test that SignalPacketReceived fires with rtcp=true when a RTCP packet is
 // received.
 TEST(RtpTransportTest, SignalDemuxedRtcp) {

From 9600e77d854090669817d22aa2fc941ee92aaacd Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Fri, 10 Apr 2026 11:52:19 +0000
Subject: [PATCH 471/472] [M148] Fix TOCTOU race in
 RtpVideoSender::OnVideoLayersAllocationUpdated
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

RtpVideoSender::OnVideoLayersAllocationUpdated runs on the encoder thread,
checks if it's active, and then posts a task to the transport queue to
update module status.

If SetSending(false) is called on the transport queue before the posted
task runs, the task would previously re-register modules in PacketRouter
even though active_ was false. This lead to dangling pointers in
PacketRouter when the RtpVideoSender was destroyed.

This CL adds an IsActive() check inside the posted task to ensure we don't
re-register modules if the sender has been deactivated.

(cherry picked from commit 77d265670fa7d7f358c86c54704a6e97c27a6005)

Bug: chromium:500767595
Test: RtpVideoSenderTest.PostTaskRaceDoesNotLeadToDanglingPointer
Change-Id: I18827c34bece7f5ebc41d159d9e19e356afbc404
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/462880
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Original-Commit-Position: refs/heads/main@{#47384}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/465961
Cr-Commit-Position: refs/branch-heads/7778@{#5}
Cr-Branched-From: ca896b7ffef011bbf6957c99d413c5aac602c99f-refs/heads/main@{#47319}
---
 call/BUILD.gn                     |  1 +
 call/rtp_video_sender.cc          |  5 +++
 call/rtp_video_sender_unittest.cc | 70 +++++++++++++++++++++++++++++++
 3 files changed, 76 insertions(+)

diff --git a/call/BUILD.gn b/call/BUILD.gn
index 815d28a81b..a78433233a 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -550,6 +550,7 @@ if (rtc_include_tests) {
         "../api/video:video_codec_constants",
         "../api/video:video_frame",
         "../api/video:video_frame_type",
+        "../api/video:video_layers_allocation",
         "../api/video:video_rtp_headers",
         "../api/video_codecs:video_codecs_api",
         "../audio",
diff --git a/call/rtp_video_sender.cc b/call/rtp_video_sender.cc
index 73e8898784..7267add42b 100644
--- a/call/rtp_video_sender.cc
+++ b/call/rtp_video_sender.cc
@@ -695,6 +695,11 @@ void RtpVideoSender::OnVideoLayersAllocationUpdated(
     transport_queue_.PostTask(
         SafeTask(safety_.flag(), [this, sending = std::move(sending)] {
           RTC_DCHECK_RUN_ON(&transport_checker_);
+          // It's possible for another task to be scheduled on the transport
+          // checker ahead of this call that makes the sender not active.
+          if (!IsActive()) {
+            return;
+          }
           RTC_CHECK_EQ(sending.size(), rtp_streams_.size());
           for (size_t i = 0; i < sending.size(); ++i) {
             SetModuleIsActive(sending[i], *rtp_streams_[i].rtp_rtcp);
diff --git a/call/rtp_video_sender_unittest.cc b/call/rtp_video_sender_unittest.cc
index a979e05db7..d7d30dfb29 100644
--- a/call/rtp_video_sender_unittest.cc
+++ b/call/rtp_video_sender_unittest.cc
@@ -40,6 +40,7 @@
 #include "api/video/encoded_image.h"
 #include "api/video/video_codec_type.h"
 #include "api/video/video_frame_type.h"
+#include "api/video/video_layers_allocation.h"
 #include "api/video_codecs/video_encoder.h"
 #include "call/rtp_config.h"
 #include "call/rtp_transport_config.h"
@@ -1607,4 +1608,73 @@ TEST(RtpVideoSenderTest, RetransmitsBaseLayerOnly) {
   EXPECT_EQ(retransmitted_rtp_sequence_numbers, base_rtp_sequence_numbers);
 }
 
+TEST(RtpVideoSenderTest, PostTaskRaceDoesNotLeadToDanglingPointer) {
+  NiceMock<MockTransport> transport;
+  NiceMock<MockRtcpIntraFrameObserver> encoder_feedback;
+
+  GlobalSimulatedTimeController time_controller(Timestamp::Millis(1000000));
+  Environment env = CreateEnvironment(time_controller.GetClock(),
+                                      time_controller.CreateTaskQueueFactory());
+
+  VideoSendStream::Config config(&transport);
+  config.rtp.ssrcs = {kSsrc1};
+
+  SendStatisticsProxy stats_proxy(
+      time_controller.GetClock(), config,
+      VideoEncoderConfig::ContentType::kRealtimeVideo, env.field_trials());
+
+  BitrateConstraints bitrate_config = GetBitrateConfig();
+  RtpTransportConfig transport_config{.env = env,
+                                      .bitrate_config = bitrate_config};
+  RtpTransportControllerSend transport_controller(transport_config);
+  transport_controller.EnsureStarted();
+
+  RateLimiter retransmission_rate_limiter(time_controller.GetClock(),
+                                          kRetransmitWindowSizeMs);
+
+  std::map<uint32_t, RtpState> suspended_ssrcs;
+  std::map<uint32_t, RtpPayloadState> suspended_payload_states;
+
+  auto router = std::make_unique<RtpVideoSender>(
+      env, time_controller.GetMainThread(), suspended_ssrcs,
+      suspended_payload_states, config.rtp, config.rtcp_report_interval_ms,
+      &transport,
+      CreateObservers(&encoder_feedback, &stats_proxy, &stats_proxy,
+                      &stats_proxy, nullptr, &stats_proxy),
+      &transport_controller, &retransmission_rate_limiter,
+      std::make_unique<FecControllerDefault>(env), nullptr, CryptoOptions{},
+      nullptr);
+
+  router->SetSending(true);
+  // Verify it's registered
+  EXPECT_TRUE(
+      transport_controller.packet_router()->SsrcOfFirstSender().has_value());
+
+  // Trigger race:
+  // 1. OnVideoLayersAllocationUpdated posts a task to re-register modules
+  // because it sees active_ is true.
+  VideoLayersAllocation allocation;
+  allocation.active_spatial_layers.push_back({.rtp_stream_index = 0});
+  router->OnVideoLayersAllocationUpdated(allocation);
+
+  // 2. Immediately set active_ to false. This happens BEFORE the posted task
+  // runs.
+  router->SetSending(false);
+
+  // 3. Let the posted task run. With the fix, it should return early and not
+  // re-register the module because active_ is now false.
+  time_controller.AdvanceTime(TimeDelta::Zero());
+
+  // 4. Verify that PacketRouter does NOT have a sender.
+  EXPECT_FALSE(
+      transport_controller.packet_router()->SsrcOfFirstSender().has_value());
+
+  // 5. Destroy the router.
+  router.reset();
+
+  // 6. Verify that PacketRouter still does not have a sender.
+  EXPECT_FALSE(
+      transport_controller.packet_router()->SsrcOfFirstSender().has_value());
+}
+
 }  // namespace webrtc

From 6316b66d7794a66de1959c55c0f841ceb52aac83 Mon Sep 17 00:00:00 2001
From: "mark a. foltz" <mfoltz@chromium.org>
Date: Mon, 20 Apr 2026 15:16:47 -0700
Subject: [PATCH 472/472] [M148] [Pipewire] Fix mouse cursor data race.

Original change's description:
> [Pipewire] Fix mouse cursor data race.
>
> This addresses a potential data race when the mouse cursor is updated
> on the Pipewire thread and read by the capture thread.
>
> Bug: chromium:504551032
> Change-Id: I1afb9febe8bb41ce62c63872e2cb5514e908dd38
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/465440
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Auto-Submit: Mark Foltz <mfoltz@chromium.org>
> Cr-Commit-Position: refs/heads/main@{#47502}

(cherry picked from commit 90b22181ec922900e26de71463b8e0ed47a5c2dd)

Bug: chromium:505447260,chromium:504551032
Change-Id: I1afb9febe8bb41ce62c63872e2cb5514e908dd38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/468140
Commit-Queue: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Auto-Submit: Chrome Cherry Picker <chrome-cherry-picker@chops-service-accounts.iam.gserviceaccount.com>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/branch-heads/7778@{#6}
Cr-Branched-From: ca896b7ffef011bbf6957c99d413c5aac602c99f-refs/heads/main@{#47319}
---
 .../linux/wayland/shared_screencast_stream.cc | 21 ++++++++++++++-----
 1 file changed, 16 insertions(+), 5 deletions(-)

diff --git a/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc b/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc
index d611ca0858..1aa8be5505 100644
--- a/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc
+++ b/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc
@@ -128,8 +128,9 @@ class SharedScreenCastStreamPrivate {
   Mutex latest_frame_lock_ RTC_ACQUIRED_AFTER(queue_lock_);
   SharedDesktopFrame* latest_available_frame_
       RTC_GUARDED_BY(&latest_frame_lock_) = nullptr;
-  std::unique_ptr<MouseCursor> mouse_cursor_;
-  DesktopVector mouse_cursor_position_ = DesktopVector(-1, -1);
+  std::unique_ptr<MouseCursor> mouse_cursor_ RTC_GUARDED_BY(&latest_frame_lock_);
+  DesktopVector mouse_cursor_position_ RTC_GUARDED_BY(&latest_frame_lock_) =
+      DesktopVector(-1, -1);
 
   int64_t modifier_;
   std::unique_ptr<EglDmaBuf> egl_dmabuf_;
@@ -695,6 +696,7 @@ SharedScreenCastStreamPrivate::CaptureFrame() {
 }
 
 std::unique_ptr<MouseCursor> SharedScreenCastStreamPrivate::CaptureCursor() {
+  MutexLock latest_frame_lock(&latest_frame_lock_);
   if (!mouse_cursor_) {
     return nullptr;
   }
@@ -703,6 +705,7 @@ std::unique_ptr<MouseCursor> SharedScreenCastStreamPrivate::CaptureCursor() {
 }
 
 DesktopVector SharedScreenCastStreamPrivate::CaptureCursorPosition() {
+  MutexLock latest_frame_lock(&latest_frame_lock_);
   return mouse_cursor_position_;
 }
 
@@ -774,20 +777,28 @@ void SharedScreenCastStreamPrivate::ProcessBuffer(pw_buffer* buffer) {
           mouse_frame->CopyPixelsFrom(
               bitmap_data, bitmap->stride,
               DesktopRect::MakeWH(bitmap->size.width, bitmap->size.height));
-          mouse_cursor_ = std::make_unique<MouseCursor>(
-              mouse_frame, DesktopVector(cursor->hotspot.x, cursor->hotspot.y));
+          {
+            MutexLock latest_frame_lock(&latest_frame_lock_);
+            mouse_cursor_ = std::make_unique<MouseCursor>(
+                mouse_frame,
+                DesktopVector(cursor->hotspot.x, cursor->hotspot.y));
+          }
 
           if (observer_) {
             observer_->OnCursorShapeChanged();
           }
         }
-        mouse_cursor_position_.set(cursor->position.x, cursor->position.y);
+        {
+          MutexLock latest_frame_lock(&latest_frame_lock_);
+          mouse_cursor_position_.set(cursor->position.x, cursor->position.y);
+        }
 
         if (observer_) {
           observer_->OnCursorPositionChanged();
         }
       } else {
         // Indicate an invalid cursor
+        MutexLock latest_frame_lock(&latest_frame_lock_);
         mouse_cursor_position_.set(-1, -1);
       }
     }