5f2c9278f3
Done using
tools/clang/scripts/build_clang_tools_extra.py \
--fetch out/Default clang-tidy clang-apply-replacements
ninja -C out/Default
gn gen out/Default --export-compile-commands
cd out/Default
tools/clang/third_party/llvm/clang-tools-extra/clang-tidy/tool/run-clang-tidy.py -p . \
-clang-tidy-binary out/Default/tools/clang/third_party/llvm/build/bin/clang-tidy \
-clang-apply-replacements-binary \
out/Default/tools/clang/third_party/llvm/build/bin/clang-apply-replacements \
-checks='-*,modernize-deprecated-headers' \
-fix
followed by
git cl format
and a round of IWYU followed by more git cl format.
Bug: webrtc:424706384
Change-Id: Icd004fa4c834002416f2196261b6ef215ebe47d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/401145
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#45196}
118 lines
3.6 KiB
C++
118 lines
3.6 KiB
C++
/*
|
|
* Copyright (c) 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
|
|
* 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 "call/audio_send_stream.h"
|
|
|
|
#include <cstddef>
|
|
#include <string>
|
|
|
|
#include "absl/strings/str_cat.h"
|
|
#include "api/audio_codecs/audio_format.h"
|
|
#include "api/call/transport.h"
|
|
#include "rtc_base/strings/string_builder.h"
|
|
|
|
namespace webrtc {
|
|
|
|
AudioSendStream::Stats::Stats() = default;
|
|
AudioSendStream::Stats::~Stats() = default;
|
|
|
|
AudioSendStream::Config::Config(Transport* send_transport)
|
|
: send_transport(send_transport) {}
|
|
|
|
AudioSendStream::Config::~Config() = default;
|
|
|
|
std::string AudioSendStream::Config::ToString() const {
|
|
StringBuilder ss;
|
|
ss << "{rtp: " << rtp.ToString();
|
|
ss << ", rtcp_report_interval_ms: " << rtcp_report_interval_ms;
|
|
ss << ", send_transport: " << (send_transport ? "(Transport)" : "null");
|
|
ss << ", min_bitrate_bps: " << min_bitrate_bps;
|
|
ss << ", max_bitrate_bps: " << max_bitrate_bps;
|
|
ss << ", has audio_network_adaptor_config: "
|
|
<< (audio_network_adaptor_config ? "true" : "false");
|
|
ss << ", has_dscp: " << (has_dscp ? "true" : "false");
|
|
ss << ", send_codec_spec: "
|
|
<< (send_codec_spec ? send_codec_spec->ToString() : "<unset>");
|
|
ss << "}";
|
|
return ss.Release();
|
|
}
|
|
|
|
AudioSendStream::Config::Rtp::Rtp() = default;
|
|
|
|
AudioSendStream::Config::Rtp::~Rtp() = default;
|
|
|
|
std::string AudioSendStream::Config::Rtp::ToString() const {
|
|
char buf[1024];
|
|
SimpleStringBuilder ss(buf);
|
|
ss << "{ssrc: " << ssrc;
|
|
if (!rid.empty()) {
|
|
ss << ", rid: " << rid;
|
|
}
|
|
if (!mid.empty()) {
|
|
ss << ", mid: " << mid;
|
|
}
|
|
ss << ", csrcs: [";
|
|
for (size_t i = 0; i < csrcs.size(); ++i) {
|
|
ss << csrcs[i];
|
|
if (i != csrcs.size() - 1) {
|
|
ss << ", ";
|
|
}
|
|
}
|
|
ss << ']';
|
|
ss << ", extmap-allow-mixed: " << (extmap_allow_mixed ? "true" : "false");
|
|
ss << ", extensions: [";
|
|
for (size_t i = 0; i < extensions.size(); ++i) {
|
|
ss << extensions[i].ToString();
|
|
if (i != extensions.size() - 1) {
|
|
ss << ", ";
|
|
}
|
|
}
|
|
ss << ']';
|
|
ss << ", c_name: " << c_name;
|
|
ss << '}';
|
|
return ss.str();
|
|
}
|
|
|
|
AudioSendStream::Config::SendCodecSpec::SendCodecSpec(
|
|
int payload_type,
|
|
const SdpAudioFormat& format)
|
|
: payload_type(payload_type), format(format) {}
|
|
AudioSendStream::Config::SendCodecSpec::~SendCodecSpec() = default;
|
|
|
|
std::string AudioSendStream::Config::SendCodecSpec::ToString() const {
|
|
char buf[1024];
|
|
SimpleStringBuilder ss(buf);
|
|
ss << "{nack_enabled: " << (nack_enabled ? "true" : "false");
|
|
ss << ", enable_non_sender_rtt: "
|
|
<< (enable_non_sender_rtt ? "true" : "false");
|
|
ss << ", cng_payload_type: "
|
|
<< (cng_payload_type ? absl::StrCat(*cng_payload_type) : "<unset>");
|
|
ss << ", red_payload_type: "
|
|
<< (red_payload_type ? absl::StrCat(*red_payload_type) : "<unset>");
|
|
ss << ", payload_type: " << payload_type;
|
|
ss << ", format: " << absl::StrCat(format);
|
|
ss << '}';
|
|
return ss.str();
|
|
}
|
|
|
|
bool AudioSendStream::Config::SendCodecSpec::operator==(
|
|
const AudioSendStream::Config::SendCodecSpec& rhs) const {
|
|
if (nack_enabled == rhs.nack_enabled &&
|
|
enable_non_sender_rtt == rhs.enable_non_sender_rtt &&
|
|
cng_payload_type == rhs.cng_payload_type &&
|
|
red_payload_type == rhs.red_payload_type &&
|
|
payload_type == rhs.payload_type && format == rhs.format &&
|
|
target_bitrate_bps == rhs.target_bitrate_bps) {
|
|
return true;
|
|
}
|
|
return false;
|
|
}
|
|
} // namespace webrtc
|