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}
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}
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}
This is a reland of commit e844bd819c
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}
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}
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}
This reverts commit e844bd819c.
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}
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}
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}
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}
This reverts commit db6adf2e9f.
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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
This is a reland of commit a7c29a690e
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}
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}
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}
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}
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}
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}
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}
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}
This is a reland of commit ed6507b57e
Reland safe: Use comment to deprecate rather than annotation
Original change's description:
> Use Clock in internal EventTracer
>
> This requires changes to X11 capturers due to a poorly defined macro in
> X11.
>
> Bug: webrtc:42223992
> Change-Id: If9f8102386dc93142f9facfba21deb1f6a6a6964
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/445903
> Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#46819}
Bug: webrtc:42223992
Change-Id: Ied9d07950cd35f132c17c3532581b5197a0fbfb1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/446780
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46832}
