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Audio stack traces

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Bug: webrtc:0
Change-Id: I90ea6301f02c2ebe72711ddbeda0bf000a6873aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276940
Auto-Submit: Olga Sharonova <olka@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38223}
This commit is contained in:
Olga Sharonova 2022-09-27 15:22:34 +02:00 committed by WebRTC LUCI CQ
parent c42eb8346f
commit 2d0ba28e25
14 changed files with 35 additions and 1 deletions
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1
audio/BUILD.gn
View File

@ -87,6 +87,7 @@ rtc_library("audio") {
"../rtc_base:audio_format_to_string",
"../rtc_base:buffer",
"../rtc_base:checks",
"../rtc_base:event_tracer",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:race_checker",

2
audio/audio_send_stream.cc
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@ -39,6 +39,7 @@
#include "rtc_base/logging.h"
#include "rtc_base/strings/audio_format_to_string.h"
#include "rtc_base/task_queue.h"
#include "rtc_base/trace_event.h"
namespace webrtc {
namespace {
@ -392,6 +393,7 @@ void AudioSendStream::Stop() {
void AudioSendStream::SendAudioData(std::unique_ptr<AudioFrame> audio_frame) {
RTC_CHECK_RUNS_SERIALIZED(&audio_capture_race_checker_);
RTC_DCHECK_GT(audio_frame->sample_rate_hz_, 0);
TRACE_EVENT0("webrtc", "AudioSendStream::SendAudioData");
double duration = static_cast<double>(audio_frame->samples_per_channel_) /
audio_frame->sample_rate_hz_;
{

5
audio/audio_transport_impl.cc
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@ -20,6 +20,7 @@
#include "modules/async_audio_processing/async_audio_processing.h"
#include "modules/audio_processing/include/audio_frame_proxies.h"
#include "rtc_base/checks.h"
#include "rtc_base/trace_event.h"
namespace webrtc {
@ -177,6 +178,7 @@ int32_t AudioTransportImpl::RecordedDataIsAvailable(
void AudioTransportImpl::SendProcessedData(
std::unique_ptr<AudioFrame> audio_frame) {
TRACE_EVENT0("webrtc", "AudioTransportImpl::SendProcessedData");
RTC_DCHECK_GT(audio_frame->samples_per_channel_, 0);
MutexLock lock(&capture_lock_);
if (audio_senders_.empty())
@ -202,6 +204,7 @@ int32_t AudioTransportImpl::NeedMorePlayData(const size_t nSamples,
size_t& nSamplesOut,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) {
TRACE_EVENT0("webrtc", "AudioTransportImpl::SendProcessedData");
RTC_DCHECK_EQ(sizeof(int16_t) * nChannels, nBytesPerSample);
RTC_DCHECK_GE(nChannels, 1);
RTC_DCHECK_LE(nChannels, 2);
@ -239,6 +242,8 @@ void AudioTransportImpl::PullRenderData(int bits_per_sample,
void* audio_data,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) {
TRACE_EVENT2("webrtc", "AudioTransportImpl::PullRenderData", "sample_rate",
sample_rate, "number_of_frames", number_of_frames);
RTC_DCHECK_EQ(bits_per_sample, 16);
RTC_DCHECK_GE(number_of_channels, 1);
RTC_DCHECK_GE(sample_rate, AudioProcessing::NativeRate::kSampleRate8kHz);

4
audio/channel_receive.cc
View File

@ -48,6 +48,7 @@
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/system/no_unique_address.h"
#include "rtc_base/time_utils.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/metrics.h"
#include "system_wrappers/include/ntp_time.h"
@ -375,6 +376,8 @@ void ChannelReceive::InitFrameTransformerDelegate(
AudioMixer::Source::AudioFrameInfo ChannelReceive::GetAudioFrameWithInfo(
int sample_rate_hz,
AudioFrame* audio_frame) {
TRACE_EVENT1("webrtc", "ChannelReceive::GetAudioFrameWithInfo",
"sample_rate_hz", sample_rate_hz);
RTC_DCHECK_RUNS_SERIALIZED(&audio_thread_race_checker_);
audio_frame->sample_rate_hz_ = sample_rate_hz;
@ -440,7 +443,6 @@ AudioMixer::Source::AudioFrameInfo ChannelReceive::GetAudioFrameWithInfo(
if (capture_start_rtp_time_stamp_ >= 0) {
// audio_frame.timestamp_ should be valid from now on.
// Compute elapsed time.
int64_t unwrap_timestamp =
rtp_ts_wraparound_handler_->Unwrap(audio_frame->timestamp_);

3
audio/channel_send.cc
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@ -40,6 +40,7 @@
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/task_queue.h"
#include "rtc_base/time_utils.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/clock.h"
#include "system_wrappers/include/metrics.h"
@ -801,6 +802,8 @@ void ChannelSend::RtcpPacketTypesCounterUpdated(
void ChannelSend::ProcessAndEncodeAudio(
std::unique_ptr<AudioFrame> audio_frame) {
TRACE_EVENT0("webrtc", "ChannelSend::ProcessAndEncodeAudio");
RTC_DCHECK_RUNS_SERIALIZED(&audio_thread_race_checker_);
RTC_DCHECK_GT(audio_frame->samples_per_channel_, 0);
RTC_DCHECK_LE(audio_frame->num_channels_, 8);

4
media/engine/webrtc_voice_engine.cc
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@ -923,6 +923,8 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
size_t number_of_channels,
size_t number_of_frames,
absl::optional<int64_t> absolute_capture_timestamp_ms) override {
TRACE_EVENT_BEGIN2("webrtc", "WebRtcAudioSendStream::OnData", "sample_rate",
sample_rate, "number_of_frames", number_of_frames);
RTC_DCHECK_EQ(16, bits_per_sample);
RTC_CHECK_RUNS_SERIALIZED(&audio_capture_race_checker_);
RTC_DCHECK(stream_);
@ -938,6 +940,8 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
*absolute_capture_timestamp_ms);
}
stream_->SendAudioData(std::move(audio_frame));
TRACE_EVENT_END1("webrtc", "WebRtcAudioSendStream::OnData",
"number_of_channels", number_of_channels);
}
// Callback from the `source_` when it is going away. In case Start() has

1
modules/audio_device/BUILD.gn
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@ -72,6 +72,7 @@ rtc_library("audio_device_buffer") {
"../../common_audio:common_audio_c",
"../../rtc_base:buffer",
"../../rtc_base:checks",
"../../rtc_base:event_tracer",
"../../rtc_base:logging",
"../../rtc_base:macromagic",
"../../rtc_base:rtc_task_queue",

4
modules/audio_device/audio_device_buffer.cc
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@ -20,6 +20,7 @@
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/time_utils.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/metrics.h"
namespace webrtc {
@ -297,6 +298,9 @@ int32_t AudioDeviceBuffer::DeliverRecordedData() {
}
int32_t AudioDeviceBuffer::RequestPlayoutData(size_t samples_per_channel) {
TRACE_EVENT1("webrtc", "AudioDeviceBuffer::RequestPlayoutData",
"samples_per_channel", samples_per_channel);
// The consumer can change the requested size on the fly and we therefore
// resize the buffer accordingly. Also takes place at the first call to this
// method.

1
modules/audio_mixer/BUILD.gn
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@ -46,6 +46,7 @@ rtc_library("audio_mixer_impl") {
"../../audio/utility:audio_frame_operations",
"../../common_audio",
"../../rtc_base:checks",
"../../rtc_base:event_tracer",
"../../rtc_base:logging",
"../../rtc_base:macromagic",
"../../rtc_base:race_checker",

2
modules/audio_mixer/audio_mixer_impl.cc
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@ -21,6 +21,7 @@
#include "modules/audio_mixer/default_output_rate_calculator.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/trace_event.h"
namespace webrtc {
@ -156,6 +157,7 @@ rtc::scoped_refptr<AudioMixerImpl> AudioMixerImpl::Create(
void AudioMixerImpl::Mix(size_t number_of_channels,
AudioFrame* audio_frame_for_mixing) {
TRACE_EVENT0("webrtc", "AudioMixerImpl::Mix");
RTC_DCHECK(number_of_channels >= 1);
MutexLock lock(&mutex_);

4
modules/rtp_rtcp/source/source_tracker.cc
View File

@ -13,6 +13,8 @@
#include <algorithm>
#include <utility>
#include "rtc_base/trace_event.h"
namespace webrtc {
constexpr int64_t SourceTracker::kTimeoutMs;
@ -24,6 +26,8 @@ void SourceTracker::OnFrameDelivered(const RtpPacketInfos& packet_infos) {
return;
}
TRACE_EVENT0("webrtc", "SourceTracker::OnFrameDelivered");
int64_t now_ms = clock_->TimeInMilliseconds();
MutexLock lock_scope(&lock_);

1
pc/BUILD.gn
View File

@ -1828,6 +1828,7 @@ rtc_library("remote_audio_source") {
"../media:rtc_media_base",
"../rtc_base",
"../rtc_base:checks",
"../rtc_base:event_tracer",
"../rtc_base:logging",
"../rtc_base:safe_conversions",
"../rtc_base:stringutils",

2
pc/remote_audio_source.cc
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@ -23,6 +23,7 @@
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/strings/string_format.h"
#include "rtc_base/trace_event.h"
namespace webrtc {
@ -149,6 +150,7 @@ void RemoteAudioSource::RemoveSink(AudioTrackSinkInterface* sink) {
void RemoteAudioSource::OnData(const AudioSinkInterface::Data& audio) {
// Called on the externally-owned audio callback thread, via/from webrtc.
TRACE_EVENT0("webrtc", "RemoteAudioSource::OnData");
MutexLock lock(&sink_lock_);
for (auto* sink : sinks_) {
// When peerconnection acts as an audio source, it should not provide

2
pc/rtp_sender.cc
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@ -474,6 +474,8 @@ void LocalAudioSinkAdapter::OnData(
size_t number_of_channels,
size_t number_of_frames,
absl::optional<int64_t> absolute_capture_timestamp_ms) {
TRACE_EVENT2("webrtc", "LocalAudioSinkAdapter::OnData", "sample_rate",
sample_rate, "number_of_frames", number_of_frames);
MutexLock lock(&lock_);
if (sink_) {
sink_->OnData(audio_data, bits_per_sample, sample_rate, number_of_channels,