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Commit 751ef0ee authored by henrika's avatar henrika Committed by Commit Bot
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Refining native logs in UserMediaProcessor and related classes 

Extends existing logs related to media streams and media stream tracks
and aligns the style with recent work in this area.

Bug: 1017219
Change-Id: I88ed50cd4c0d9dd5e2f6104e8176a8ef99d6d46d
Reviewed-on: https://chromium-review.googlesource.com/c/chromium/src/+/1960278Reviewed-by: default avatarGuido Urdaneta <guidou@chromium.org>
Commit-Queue: Henrik Andreasson <henrika@chromium.org>
Cr-Commit-Position: refs/heads/master@{#726392}
parent 0890817f
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Showing with 464 additions and 155 deletions
third_party/blink/public/platform/modules/mediastream/media_stream_audio_track.h
View file @ 751ef0ee
...@@ -111,6 +111,10 @@ class BLINK_PLATFORM_EXPORT MediaStreamAudioTrack ...@@ -111,6 +111,10 @@ class BLINK_PLATFORM_EXPORT MediaStreamAudioTrack
// Buffer used to deliver silent audio data while this track is disabled. // Buffer used to deliver silent audio data while this track is disabled.
std::unique_ptr<media::AudioBus> silent_bus_; std::unique_ptr<media::AudioBus> silent_bus_;
// Set to true once at first audio callback after calling Start().
// Only used for logging purposes.
bool received_audio_callback_ = false;
// Provides weak pointers that are valid until Stop() is called. // Provides weak pointers that are valid until Stop() is called.
base::WeakPtrFactory<MediaStreamAudioTrack> weak_factory_{this}; base::WeakPtrFactory<MediaStreamAudioTrack> weak_factory_{this};
......
third_party/blink/renderer/modules/mediastream/processed_local_audio_source.cc
View file @ 751ef0ee
...@@ -36,11 +36,26 @@ using EchoCancellationType = ...@@ -36,11 +36,26 @@ using EchoCancellationType =
blink::AudioProcessingProperties::EchoCancellationType; blink::AudioProcessingProperties::EchoCancellationType;
namespace { namespace {
void SendLogMessage(const std::string& message) {
blink::WebRtcLogMessage("PLAS::" + message);
}
// Used as an identifier for ProcessedLocalAudioSource::From(). // Used as an identifier for ProcessedLocalAudioSource::From().
void* const kProcessedLocalAudioSourceIdentifier = void* const kProcessedLocalAudioSourceIdentifier =
const_cast<void**>(&kProcessedLocalAudioSourceIdentifier); const_cast<void**>(&kProcessedLocalAudioSourceIdentifier);
void LogAudioProcesingProperties( std::string GetEnsureSourceIsStartedLogString(
const blink::MediaStreamDevice& device) {
return base::StringPrintf(
"EnsureSourceIsStarted({session_id=%s}, {channel_layout=%d}, "
"{sample_rate=%d}, {buffer_size=%d}, {effects=%d})",
device.session_id().ToString().c_str(), device.input.channel_layout(),
device.input.sample_rate(), device.input.frames_per_buffer(),
device.input.effects());
}
std::string GetAudioProcesingPropertiesLogString(
const blink::AudioProcessingProperties& properties) { const blink::AudioProcessingProperties& properties) {
auto aec_to_string = auto aec_to_string =
[](blink::AudioProcessingProperties::EchoCancellationType type) { [](blink::AudioProcessingProperties::EchoCancellationType type) {
...@@ -56,18 +71,17 @@ void LogAudioProcesingProperties( ...@@ -56,18 +71,17 @@ void LogAudioProcesingProperties(
}; };
auto bool_to_string = [](bool value) { return value ? "true" : "false"; }; auto bool_to_string = [](bool value) { return value ? "true" : "false"; };
auto str = base::StringPrintf( auto str = base::StringPrintf(
"AudioProcessingProperties: " "aec: %s, "
"aec=%s, " "disable_hw_ns: %s, "
"disable_hw_ns=%s, " "goog_audio_mirroring: %s, "
"goog_audio_mirroring=%s, " "goog_auto_gain_control: %s, "
"goog_auto_gain_control=%s, " "goog_experimental_echo_cancellation: %s, "
"goog_experimental_echo_cancellation=%s, " "goog_typing_noise_detection: %s, "
"goog_typing_noise_detection=%s, " "goog_noise_suppression: %s, "
"goog_noise_suppression=%s, " "goog_experimental_noise_suppression: %s, "
"goog_experimental_noise_suppression=%s, " "goog_highpass_filter: %s, "
"goog_highpass_filter=%s, " "goog_experimental_agc: %s, "
"goog_experimental_agc=%s, " "hybrid_agc: %s",
"hybrid_agc=%s",
aec_to_string(properties.echo_cancellation_type), aec_to_string(properties.echo_cancellation_type),
bool_to_string(properties.disable_hw_noise_suppression), bool_to_string(properties.disable_hw_noise_suppression),
bool_to_string(properties.goog_audio_mirroring), bool_to_string(properties.goog_audio_mirroring),
...@@ -79,8 +93,7 @@ void LogAudioProcesingProperties( ...@@ -79,8 +93,7 @@ void LogAudioProcesingProperties(
bool_to_string(properties.goog_highpass_filter), bool_to_string(properties.goog_highpass_filter),
bool_to_string(properties.goog_experimental_auto_gain_control), bool_to_string(properties.goog_experimental_auto_gain_control),
bool_to_string(base::FeatureList::IsEnabled(features::kWebRtcHybridAgc))); bool_to_string(base::FeatureList::IsEnabled(features::kWebRtcHybridAgc)));
return str;
blink::WebRtcLogMessage(str);
} }
} // namespace } // namespace
...@@ -100,12 +113,14 @@ ProcessedLocalAudioSource::ProcessedLocalAudioSource( ...@@ -100,12 +113,14 @@ ProcessedLocalAudioSource::ProcessedLocalAudioSource(
started_callback_(std::move(started_callback)), started_callback_(std::move(started_callback)),
volume_(0), volume_(0),
allow_invalid_render_frame_id_for_testing_(false) { allow_invalid_render_frame_id_for_testing_(false) {
DVLOG(1) << "ProcessedLocalAudioSource::ProcessedLocalAudioSource()";
SetDevice(device); SetDevice(device);
SendLogMessage(
base::StringPrintf("ProcessedLocalAudioSource({session_id=%s})",
device.session_id().ToString().c_str()));
} }
ProcessedLocalAudioSource::~ProcessedLocalAudioSource() { ProcessedLocalAudioSource::~ProcessedLocalAudioSource() {
DVLOG(1) << "ProcessedLocalAudioSource::~ProcessedLocalAudioSource()"; DVLOG(1) << "PLAS::~ProcessedLocalAudioSource()";
EnsureSourceIsStopped(); EnsureSourceIsStopped();
} }
...@@ -118,6 +133,12 @@ ProcessedLocalAudioSource* ProcessedLocalAudioSource::From( ...@@ -118,6 +133,12 @@ ProcessedLocalAudioSource* ProcessedLocalAudioSource::From(
return nullptr; return nullptr;
} }
void ProcessedLocalAudioSource::SendLogMessageWithSessionId(
const std::string& message) const {
SendLogMessage(message + " [session_id=" + device().session_id().ToString() +
"]");
}
base::Optional<blink::AudioProcessingProperties> base::Optional<blink::AudioProcessingProperties>
ProcessedLocalAudioSource::GetAudioProcessingProperties() const { ProcessedLocalAudioSource::GetAudioProcessingProperties() const {
return audio_processing_properties_; return audio_processing_properties_;
...@@ -137,23 +158,17 @@ bool ProcessedLocalAudioSource::EnsureSourceIsStarted() { ...@@ -137,23 +158,17 @@ bool ProcessedLocalAudioSource::EnsureSourceIsStarted() {
// to initialize the audio source. // to initialize the audio source.
if (!allow_invalid_render_frame_id_for_testing_ && if (!allow_invalid_render_frame_id_for_testing_ &&
!internal_consumer_frame_->frame()) { !internal_consumer_frame_->frame()) {
blink::WebRtcLogMessage( SendLogMessageWithSessionId(
"ProcessedLocalAudioSource::EnsureSourceIsStarted() fails " "EnsureSourceIsStarted() => (ERROR: "
" because the render frame does not exist."); " render frame does not exist)");
return false; return false;
} }
std::string str = base::StringPrintf( SendLogMessage(GetEnsureSourceIsStartedLogString(device()));
"ProcessedLocalAudioSource::EnsureSourceIsStarted." SendLogMessageWithSessionId(base::StringPrintf(
"channel_layout=%d, sample_rate=%d, buffer_size=%d, session_id=%s" "EnsureSourceIsStarted() => (audio_processing_properties=[%s])",
", effects=%d. ", GetAudioProcesingPropertiesLogString(audio_processing_properties_)
device().input.channel_layout(), device().input.sample_rate(), .c_str()));
device().input.frames_per_buffer(),
device().session_id().ToString().c_str(), device().input.effects());
blink::WebRtcLogMessage(str);
DVLOG(1) << str;
LogAudioProcesingProperties(audio_processing_properties_);
blink::MediaStreamDevice modified_device(device()); blink::MediaStreamDevice modified_device(device());
bool device_is_modified = false; bool device_is_modified = false;
...@@ -197,9 +212,8 @@ bool ProcessedLocalAudioSource::EnsureSourceIsStarted() { ...@@ -197,9 +212,8 @@ bool ProcessedLocalAudioSource::EnsureSourceIsStarted() {
WebRtcAudioDeviceImpl* const rtc_audio_device = WebRtcAudioDeviceImpl* const rtc_audio_device =
PeerConnectionDependencyFactory::GetInstance()->GetWebRtcAudioDevice(); PeerConnectionDependencyFactory::GetInstance()->GetWebRtcAudioDevice();
if (!rtc_audio_device) { if (!rtc_audio_device) {
blink::WebRtcLogMessage( SendLogMessageWithSessionId(
"ProcessedLocalAudioSource::EnsureSourceIsStarted() fails" "EnsureSourceIsStarted() => (ERROR: no WebRTC ADM instance)");
" because there is no WebRtcAudioDeviceImpl instance.");
return false; return false;
} }
...@@ -227,10 +241,10 @@ bool ProcessedLocalAudioSource::EnsureSourceIsStarted() { ...@@ -227,10 +241,10 @@ bool ProcessedLocalAudioSource::EnsureSourceIsStarted() {
channel_layout != media::CHANNEL_LAYOUT_STEREO && channel_layout != media::CHANNEL_LAYOUT_STEREO &&
channel_layout != media::CHANNEL_LAYOUT_STEREO_AND_KEYBOARD_MIC && channel_layout != media::CHANNEL_LAYOUT_STEREO_AND_KEYBOARD_MIC &&
channel_layout != media::CHANNEL_LAYOUT_DISCRETE) { channel_layout != media::CHANNEL_LAYOUT_DISCRETE) {
blink::WebRtcLogMessage(base::StringPrintf( SendLogMessage(
"ProcessedLocalAudioSource::EnsureSourceIsStarted() fails " base::StringPrintf("EnsureSourceIsStarted() => (ERROR: "
" because the input channel layout (%d) is not supported.", "input channel layout (%d) is not supported.",
static_cast<int>(channel_layout))); static_cast<int>(channel_layout)));
return false; return false;
} }
...@@ -258,6 +272,7 @@ bool ProcessedLocalAudioSource::EnsureSourceIsStarted() { ...@@ -258,6 +272,7 @@ bool ProcessedLocalAudioSource::EnsureSourceIsStarted() {
} }
DVLOG(1) << params.AsHumanReadableString(); DVLOG(1) << params.AsHumanReadableString();
DCHECK(params.IsValid()); DCHECK(params.IsValid());
media::AudioSourceParameters source_params(device().session_id()); media::AudioSourceParameters source_params(device().session_id());
const bool use_remote_apm = const bool use_remote_apm =
media::IsWebRtcApmInAudioServiceEnabled() && media::IsWebRtcApmInAudioServiceEnabled() &&
...@@ -277,8 +292,9 @@ bool ProcessedLocalAudioSource::EnsureSourceIsStarted() { ...@@ -277,8 +292,9 @@ bool ProcessedLocalAudioSource::EnsureSourceIsStarted() {
source_params.processing->settings.automatic_gain_control = source_params.processing->settings.automatic_gain_control =
media::AutomaticGainControlType::kHybridExperimental; media::AutomaticGainControlType::kHybridExperimental;
} }
blink::WebRtcLogMessage(base::StringPrintf( SendLogMessageWithSessionId(base::StringPrintf(
"Using APM in audio process; settings: %s", "EnsureSourceIsStarted() => (using APM in audio process: "
"settings=[%s])",
source_params.processing->settings.ToString().c_str())); source_params.processing->settings.ToString().c_str()));
} else { } else {
...@@ -291,10 +307,11 @@ bool ProcessedLocalAudioSource::EnsureSourceIsStarted() { ...@@ -291,10 +307,11 @@ bool ProcessedLocalAudioSource::EnsureSourceIsStarted() {
} }
// Start the source. // Start the source.
DVLOG(1) << "Starting WebRTC audio source for consumption " SendLogMessageWithSessionId(base::StringPrintf(
<< "with input parameters={" << params.AsHumanReadableString() "EnsureSourceIsStarted() => (WebRTC audio source starts: "
<< "} and output parameters={" "input_parameters=[%s], output_parameters=[%s])",
<< GetAudioParameters().AsHumanReadableString() << '}'; params.AsHumanReadableString().c_str(),
GetAudioParameters().AsHumanReadableString().c_str()));
scoped_refptr<media::AudioCapturerSource> new_source = scoped_refptr<media::AudioCapturerSource> new_source =
Platform::Current()->NewAudioCapturerSource( Platform::Current()->NewAudioCapturerSource(
internal_consumer_frame_->web_frame(), source_params); internal_consumer_frame_->web_frame(), source_params);
...@@ -372,6 +389,7 @@ int ProcessedLocalAudioSource::MaxVolume() const { ...@@ -372,6 +389,7 @@ int ProcessedLocalAudioSource::MaxVolume() const {
} }
void ProcessedLocalAudioSource::OnCaptureStarted() { void ProcessedLocalAudioSource::OnCaptureStarted() {
SendLogMessageWithSessionId(base::StringPrintf("OnCaptureStarted()"));
std::move(started_callback_) std::move(started_callback_)
.Run(this, blink::mojom::MediaStreamRequestResult::OK, ""); .Run(this, blink::mojom::MediaStreamRequestResult::OK, "");
} }
...@@ -392,24 +410,29 @@ void ProcessedLocalAudioSource::Capture(const media::AudioBus* audio_bus, ...@@ -392,24 +410,29 @@ void ProcessedLocalAudioSource::Capture(const media::AudioBus* audio_bus,
} }
void ProcessedLocalAudioSource::OnCaptureError(const std::string& message) { void ProcessedLocalAudioSource::OnCaptureError(const std::string& message) {
blink::WebRtcLogMessage("ProcessedLocalAudioSource::OnCaptureError: " + SendLogMessageWithSessionId(
message); base::StringPrintf("OnCaptureError({message=%s})", message.c_str()));
StopSourceOnError(message); StopSourceOnError(message);
} }
void ProcessedLocalAudioSource::OnCaptureMuted(bool is_muted) { void ProcessedLocalAudioSource::OnCaptureMuted(bool is_muted) {
SendLogMessageWithSessionId(base::StringPrintf(
"OnCaptureMuted({is_muted=%s})", is_muted ? "true" : "false"));
SetMutedState(is_muted); SetMutedState(is_muted);
} }
void ProcessedLocalAudioSource::OnCaptureProcessorCreated( void ProcessedLocalAudioSource::OnCaptureProcessorCreated(
media::AudioProcessorControls* controls) { media::AudioProcessorControls* controls) {
SendLogMessageWithSessionId(
base::StringPrintf("OnCaptureProcessorCreated()"));
DCHECK(audio_processor_proxy_); DCHECK(audio_processor_proxy_);
audio_processor_proxy_->SetControls(controls); audio_processor_proxy_->SetControls(controls);
} }
void ProcessedLocalAudioSource::SetOutputDeviceForAec( void ProcessedLocalAudioSource::SetOutputDeviceForAec(
const std::string& output_device_id) { const std::string& output_device_id) {
DVLOG(1) << "ProcessedLocalAudioSource::SetOutputDeviceForAec()"; SendLogMessageWithSessionId(base::StringPrintf(
"SetOutputDeviceForAec({device_id=%s})", output_device_id.c_str()));
if (source_) if (source_)
source_->SetOutputDeviceForAec(output_device_id); source_->SetOutputDeviceForAec(output_device_id);
} }
......
third_party/blink/renderer/modules/mediastream/processed_local_audio_source.h
View file @ 751ef0ee
...@@ -119,6 +119,11 @@ class MODULES_EXPORT ProcessedLocalAudioSource final ...@@ -119,6 +119,11 @@ class MODULES_EXPORT ProcessedLocalAudioSource final
// processing will take place. // processing will take place.
int GetBufferSize(int sample_rate) const; int GetBufferSize(int sample_rate) const;
// Helper method which sends the log |message| to a native WebRTC log and
// adds the current session ID (from the associated media stream device) to
// make the log unique.
void SendLogMessageWithSessionId(const std::string& message) const;
// The LocalFrame that will consume the audio data. Used when creating // The LocalFrame that will consume the audio data. Used when creating
// AudioCapturerSources. // AudioCapturerSources.
std::unique_ptr<MediaStreamInternalFrameWrapper> internal_consumer_frame_; std::unique_ptr<MediaStreamInternalFrameWrapper> internal_consumer_frame_;
......
third_party/blink/renderer/modules/mediastream/user_media_processor.h
View file @ 751ef0ee
...@@ -161,9 +161,11 @@ class MODULES_EXPORT UserMediaProcessor ...@@ -161,9 +161,11 @@ class MODULES_EXPORT UserMediaProcessor
bool IsCurrentRequestInfo( bool IsCurrentRequestInfo(
const blink::WebUserMediaRequest& web_request) const; const blink::WebUserMediaRequest& web_request) const;
void DelayedGetUserMediaRequestSucceeded( void DelayedGetUserMediaRequestSucceeded(
int request_id,
const blink::WebMediaStream& stream, const blink::WebMediaStream& stream,
blink::WebUserMediaRequest web_request); blink::WebUserMediaRequest web_request);
void DelayedGetUserMediaRequestFailed( void DelayedGetUserMediaRequestFailed(
int request_id,
blink::WebUserMediaRequest web_request, blink::WebUserMediaRequest web_request,
blink::mojom::blink::MediaStreamRequestResult result, blink::mojom::blink::MediaStreamRequestResult result,
const String& constraint_name); const String& constraint_name);
......
third_party/blink/renderer/modules/peerconnection/webrtc_media_stream_track_adapter.cc
View file @ 751ef0ee
...@@ -5,11 +5,21 @@ ...@@ -5,11 +5,21 @@
#include "third_party/blink/renderer/modules/peerconnection/webrtc_media_stream_track_adapter.h" #include "third_party/blink/renderer/modules/peerconnection/webrtc_media_stream_track_adapter.h"
#include "base/bind.h" #include "base/bind.h"
#include "base/strings/stringprintf.h"
#include "third_party/blink/public/platform/modules/mediastream/media_stream_audio_track.h" #include "third_party/blink/public/platform/modules/mediastream/media_stream_audio_track.h"
#include "third_party/blink/public/platform/modules/webrtc/webrtc_logging.h"
#include "third_party/blink/renderer/modules/mediastream/processed_local_audio_source.h" #include "third_party/blink/renderer/modules/mediastream/processed_local_audio_source.h"
#include "third_party/blink/renderer/modules/peerconnection/media_stream_video_webrtc_sink.h" #include "third_party/blink/renderer/modules/peerconnection/media_stream_video_webrtc_sink.h"
#include "third_party/blink/renderer/modules/peerconnection/peer_connection_dependency_factory.h" #include "third_party/blink/renderer/modules/peerconnection/peer_connection_dependency_factory.h"
namespace {
void SendLogMessage(const std::string& message) {
blink::WebRtcLogMessage("WRMSTA::" + message);
}
} // namespace
namespace blink { namespace blink {
// static // static
...@@ -155,6 +165,8 @@ void WebRtcMediaStreamTrackAdapter::InitializeLocalAudioTrack( ...@@ -155,6 +165,8 @@ void WebRtcMediaStreamTrackAdapter::InitializeLocalAudioTrack(
DCHECK(!web_track.IsNull()); DCHECK(!web_track.IsNull());
DCHECK_EQ(web_track.Source().GetType(), DCHECK_EQ(web_track.Source().GetType(),
blink::WebMediaStreamSource::kTypeAudio); blink::WebMediaStreamSource::kTypeAudio);
SendLogMessage(base::StringPrintf("InitializeLocalAudioTrack({id=%s})",
web_track.Id().Utf8().c_str()));
web_track_ = web_track; web_track_ = web_track;
blink::MediaStreamAudioTrack* native_track = blink::MediaStreamAudioTrack* native_track =
blink::MediaStreamAudioTrack::From(web_track_); blink::MediaStreamAudioTrack::From(web_track_);
...@@ -208,6 +220,8 @@ void WebRtcMediaStreamTrackAdapter::InitializeRemoteAudioTrack( ...@@ -208,6 +220,8 @@ void WebRtcMediaStreamTrackAdapter::InitializeRemoteAudioTrack(
DCHECK(webrtc_audio_track); DCHECK(webrtc_audio_track);
DCHECK_EQ(webrtc_audio_track->kind(), DCHECK_EQ(webrtc_audio_track->kind(),
webrtc::MediaStreamTrackInterface::kAudioKind); webrtc::MediaStreamTrackInterface::kAudioKind);
SendLogMessage(
base::StringPrintf("InitializeRemoteAudioTrack([this=%p])", this));
remote_audio_track_adapter_ = remote_audio_track_adapter_ =
base::MakeRefCounted<blink::RemoteAudioTrackAdapter>( base::MakeRefCounted<blink::RemoteAudioTrackAdapter>(
main_thread_, webrtc_audio_track.get()); main_thread_, webrtc_audio_track.get());
......
third_party/blink/renderer/modules/webrtc/webrtc_audio_device_impl.cc
View file @ 751ef0ee
...@@ -7,10 +7,12 @@ ...@@ -7,10 +7,12 @@
#include "base/logging.h" #include "base/logging.h"
#include "base/metrics/histogram_macros.h" #include "base/metrics/histogram_macros.h"
#include "base/stl_util.h" #include "base/stl_util.h"
#include "base/strings/stringprintf.h"
#include "base/trace_event/trace_event.h" #include "base/trace_event/trace_event.h"
#include "media/base/audio_bus.h" #include "media/base/audio_bus.h"
#include "media/base/audio_parameters.h" #include "media/base/audio_parameters.h"
#include "media/base/sample_rates.h" #include "media/base/sample_rates.h"
#include "third_party/blink/public/platform/modules/webrtc/webrtc_logging.h"
#include "third_party/blink/renderer/modules/mediastream/processed_local_audio_source.h" #include "third_party/blink/renderer/modules/mediastream/processed_local_audio_source.h"
#include "third_party/blink/renderer/modules/webrtc/webrtc_audio_renderer.h" #include "third_party/blink/renderer/modules/webrtc/webrtc_audio_renderer.h"
...@@ -19,6 +21,14 @@ using media::ChannelLayout; ...@@ -19,6 +21,14 @@ using media::ChannelLayout;
namespace blink { namespace blink {
namespace {
void SendLogMessage(const std::string& message) {
blink::WebRtcLogMessage("WRADI::" + message);
}
} // namespace
WebRtcAudioDeviceImpl::WebRtcAudioDeviceImpl() WebRtcAudioDeviceImpl::WebRtcAudioDeviceImpl()
: audio_processing_id_(base::UnguessableToken::Create()), : audio_processing_id_(base::UnguessableToken::Create()),
audio_transport_callback_(nullptr), audio_transport_callback_(nullptr),
...@@ -26,7 +36,8 @@ WebRtcAudioDeviceImpl::WebRtcAudioDeviceImpl() ...@@ -26,7 +36,8 @@ WebRtcAudioDeviceImpl::WebRtcAudioDeviceImpl()
initialized_(false), initialized_(false),
playing_(false), playing_(false),
recording_(false) { recording_(false) {
DVLOG(1) << "WebRtcAudioDeviceImpl::WebRtcAudioDeviceImpl()"; SendLogMessage(base::StringPrintf("WebRtcAudioDeviceImpl({id=%s})",
GetAudioProcessingId().ToString().c_str()));
// This object can be constructed on either the signaling thread or the main // This object can be constructed on either the signaling thread or the main
// thread, so we need to detach these thread checkers here and have them // thread, so we need to detach these thread checkers here and have them
// initialize automatically when the first methods are called. // initialize automatically when the first methods are called.
...@@ -38,7 +49,8 @@ WebRtcAudioDeviceImpl::WebRtcAudioDeviceImpl() ...@@ -38,7 +49,8 @@ WebRtcAudioDeviceImpl::WebRtcAudioDeviceImpl()
} }
WebRtcAudioDeviceImpl::~WebRtcAudioDeviceImpl() { WebRtcAudioDeviceImpl::~WebRtcAudioDeviceImpl() {
DVLOG(1) << "WebRtcAudioDeviceImpl::~WebRtcAudioDeviceImpl()"; SendLogMessage(base::StringPrintf("~WebRtcAudioDeviceImpl([id=%s])",
GetAudioProcessingId().ToString().c_str()));
DCHECK_CALLED_ON_VALID_THREAD(main_thread_checker_); DCHECK_CALLED_ON_VALID_THREAD(main_thread_checker_);
DCHECK(!initialized_) << "Terminate must have been called."; DCHECK(!initialized_) << "Terminate must have been called.";
} }
...@@ -143,8 +155,9 @@ base::UnguessableToken WebRtcAudioDeviceImpl::GetAudioProcessingId() const { ...@@ -143,8 +155,9 @@ base::UnguessableToken WebRtcAudioDeviceImpl::GetAudioProcessingId() const {
int32_t WebRtcAudioDeviceImpl::RegisterAudioCallback( int32_t WebRtcAudioDeviceImpl::RegisterAudioCallback(
webrtc::AudioTransport* audio_callback) { webrtc::AudioTransport* audio_callback) {
DVLOG(1) << "WebRtcAudioDeviceImpl::RegisterAudioCallback()";
DCHECK_CALLED_ON_VALID_THREAD(signaling_thread_checker_); DCHECK_CALLED_ON_VALID_THREAD(signaling_thread_checker_);
SendLogMessage(base::StringPrintf("RegisterAudioCallback([id=%s])",
GetAudioProcessingId().ToString().c_str()));
base::AutoLock lock(lock_); base::AutoLock lock(lock_);
DCHECK_EQ(!audio_transport_callback_, !!audio_callback); DCHECK_EQ(!audio_transport_callback_, !!audio_callback);
audio_transport_callback_ = audio_callback; audio_transport_callback_ = audio_callback;
...@@ -254,9 +267,10 @@ bool WebRtcAudioDeviceImpl::Playing() const { ...@@ -254,9 +267,10 @@ bool WebRtcAudioDeviceImpl::Playing() const {
} }
int32_t WebRtcAudioDeviceImpl::StartRecording() { int32_t WebRtcAudioDeviceImpl::StartRecording() {
DVLOG(1) << "WebRtcAudioDeviceImpl::StartRecording()";
DCHECK_CALLED_ON_VALID_THREAD(worker_thread_checker_); DCHECK_CALLED_ON_VALID_THREAD(worker_thread_checker_);
DCHECK(initialized_); DCHECK(initialized_);
SendLogMessage(base::StringPrintf("StartRecording([id=%s])",
GetAudioProcessingId().ToString().c_str()));
base::AutoLock auto_lock(lock_); base::AutoLock auto_lock(lock_);
if (!audio_transport_callback_) { if (!audio_transport_callback_) {
LOG(ERROR) << "Audio transport is missing"; LOG(ERROR) << "Audio transport is missing";
...@@ -269,7 +283,6 @@ int32_t WebRtcAudioDeviceImpl::StartRecording() { ...@@ -269,7 +283,6 @@ int32_t WebRtcAudioDeviceImpl::StartRecording() {
} }
int32_t WebRtcAudioDeviceImpl::StopRecording() { int32_t WebRtcAudioDeviceImpl::StopRecording() {
DVLOG(1) << "WebRtcAudioDeviceImpl::StopRecording()";
DCHECK(initialized_); DCHECK(initialized_);
// Can be called both from the worker thread (e.g. when called from webrtc) // Can be called both from the worker thread (e.g. when called from webrtc)
// or the signaling thread (e.g. when we call it ourselves internally). // or the signaling thread (e.g. when we call it ourselves internally).
...@@ -279,6 +292,8 @@ int32_t WebRtcAudioDeviceImpl::StopRecording() { ...@@ -279,6 +292,8 @@ int32_t WebRtcAudioDeviceImpl::StopRecording() {
DCHECK(signaling_thread_checker_.CalledOnValidThread() || DCHECK(signaling_thread_checker_.CalledOnValidThread() ||
worker_thread_checker_.CalledOnValidThread()); worker_thread_checker_.CalledOnValidThread());
#endif #endif
SendLogMessage(base::StringPrintf("StopRecording([id=%s])",
GetAudioProcessingId().ToString().c_str()));
base::AutoLock auto_lock(lock_); base::AutoLock auto_lock(lock_);
recording_ = false; recording_ = false;
return 0; return 0;
...@@ -379,7 +394,8 @@ bool WebRtcAudioDeviceImpl::SetAudioRenderer( ...@@ -379,7 +394,8 @@ bool WebRtcAudioDeviceImpl::SetAudioRenderer(
void WebRtcAudioDeviceImpl::AddAudioCapturer( void WebRtcAudioDeviceImpl::AddAudioCapturer(
ProcessedLocalAudioSource* capturer) { ProcessedLocalAudioSource* capturer) {
DCHECK_CALLED_ON_VALID_THREAD(main_thread_checker_); DCHECK_CALLED_ON_VALID_THREAD(main_thread_checker_);
DVLOG(1) << "WebRtcAudioDeviceImpl::AddAudioCapturer()"; SendLogMessage(base::StringPrintf("AddAudioCapturer([id=%s])",
GetAudioProcessingId().ToString().c_str()));
DCHECK(capturer); DCHECK(capturer);
DCHECK(!capturer->device().id.empty()); DCHECK(!capturer->device().id.empty());
...@@ -392,7 +408,8 @@ void WebRtcAudioDeviceImpl::AddAudioCapturer( ...@@ -392,7 +408,8 @@ void WebRtcAudioDeviceImpl::AddAudioCapturer(
void WebRtcAudioDeviceImpl::RemoveAudioCapturer( void WebRtcAudioDeviceImpl::RemoveAudioCapturer(
ProcessedLocalAudioSource* capturer) { ProcessedLocalAudioSource* capturer) {
DCHECK_CALLED_ON_VALID_THREAD(main_thread_checker_); DCHECK_CALLED_ON_VALID_THREAD(main_thread_checker_);
DVLOG(1) << "WebRtcAudioDeviceImpl::RemoveAudioCapturer()"; SendLogMessage(base::StringPrintf("RemoveAudioCapturer([id=%s])",
GetAudioProcessingId().ToString().c_str()));
DCHECK(capturer); DCHECK(capturer);
base::AutoLock auto_lock(lock_); base::AutoLock auto_lock(lock_);
capturers_.remove(capturer); capturers_.remove(capturer);
......
third_party/blink/renderer/platform/exported/mediastream/media_stream_audio_track.cc
View file @ 751ef0ee
...@@ -8,21 +8,32 @@ ...@@ -8,21 +8,32 @@
#include <vector> #include <vector>
#include "base/logging.h" #include "base/logging.h"
#include "base/strings/stringprintf.h"
#include "media/base/audio_bus.h" #include "media/base/audio_bus.h"
#include "third_party/blink/public/platform/modules/mediastream/web_media_stream_audio_sink.h" #include "third_party/blink/public/platform/modules/mediastream/web_media_stream_audio_sink.h"
#include "third_party/blink/public/platform/modules/webrtc/webrtc_logging.h"
#include "third_party/blink/public/platform/web_media_stream_source.h" #include "third_party/blink/public/platform/web_media_stream_source.h"
namespace blink { namespace blink {
namespace {
void SendLogMessage(const std::string& message) {
blink::WebRtcLogMessage("MSAT::" + message);
}
} // namespace
MediaStreamAudioTrack::MediaStreamAudioTrack(bool is_local_track) MediaStreamAudioTrack::MediaStreamAudioTrack(bool is_local_track)
: WebPlatformMediaStreamTrack(is_local_track), is_enabled_(1) { : WebPlatformMediaStreamTrack(is_local_track), is_enabled_(1) {
DVLOG(1) << "MediaStreamAudioTrack@" << this << "::MediaStreamAudioTrack(" SendLogMessage(
<< (is_local_track ? "local" : "remote") << " track)"; base::StringPrintf("MediaStreamAudioTrack([this=%p] {is_local_track=%s})",
this, (is_local_track ? "local" : "remote")));
} }
MediaStreamAudioTrack::~MediaStreamAudioTrack() { MediaStreamAudioTrack::~MediaStreamAudioTrack() {
DCHECK_CALLED_ON_VALID_THREAD(thread_checker_); DCHECK_CALLED_ON_VALID_THREAD(thread_checker_);
DVLOG(1) << "MediaStreamAudioTrack@" << this << " is being destroyed."; SendLogMessage(base::StringPrintf("~MediaStreamAudioTrack([this=%p])", this));
Stop(); Stop();
} }
...@@ -38,9 +49,7 @@ MediaStreamAudioTrack* MediaStreamAudioTrack::From( ...@@ -38,9 +49,7 @@ MediaStreamAudioTrack* MediaStreamAudioTrack::From(
void MediaStreamAudioTrack::AddSink(WebMediaStreamAudioSink* sink) { void MediaStreamAudioTrack::AddSink(WebMediaStreamAudioSink* sink) {
DCHECK_CALLED_ON_VALID_THREAD(thread_checker_); DCHECK_CALLED_ON_VALID_THREAD(thread_checker_);
SendLogMessage(base::StringPrintf("AddSink([this=%p])", this));
DVLOG(1) << "Adding WebMediaStreamAudioSink@" << sink
<< " to MediaStreamAudioTrack@" << this << '.';
// If the track has already stopped, just notify the sink of this fact without // If the track has already stopped, just notify the sink of this fact without
// adding it. // adding it.
...@@ -55,9 +64,8 @@ void MediaStreamAudioTrack::AddSink(WebMediaStreamAudioSink* sink) { ...@@ -55,9 +64,8 @@ void MediaStreamAudioTrack::AddSink(WebMediaStreamAudioSink* sink) {
void MediaStreamAudioTrack::RemoveSink(WebMediaStreamAudioSink* sink) { void MediaStreamAudioTrack::RemoveSink(WebMediaStreamAudioSink* sink) {
DCHECK_CALLED_ON_VALID_THREAD(thread_checker_); DCHECK_CALLED_ON_VALID_THREAD(thread_checker_);
SendLogMessage(base::StringPrintf("RemoveSink([this=%p])", this));
deliverer_.RemoveConsumer(sink); deliverer_.RemoveConsumer(sink);
DVLOG(1) << "Removed WebMediaStreamAudioSink@" << sink
<< " from MediaStreamAudioTrack@" << this << '.';
} }
media::AudioParameters MediaStreamAudioTrack::GetOutputFormat() const { media::AudioParameters MediaStreamAudioTrack::GetOutputFormat() const {
...@@ -66,8 +74,8 @@ media::AudioParameters MediaStreamAudioTrack::GetOutputFormat() const { ...@@ -66,8 +74,8 @@ media::AudioParameters MediaStreamAudioTrack::GetOutputFormat() const {
void MediaStreamAudioTrack::SetEnabled(bool enabled) { void MediaStreamAudioTrack::SetEnabled(bool enabled) {
DCHECK_CALLED_ON_VALID_THREAD(thread_checker_); DCHECK_CALLED_ON_VALID_THREAD(thread_checker_);
DVLOG(1) << "MediaStreamAudioTrack@" << this << "::SetEnabled(" SendLogMessage(base::StringPrintf("SetEnabled([this=%p] {enabled=%s})", this,
<< (enabled ? 'Y' : 'N') << ')'; (enabled ? "true" : "false")));
const bool previously_enabled = const bool previously_enabled =
!!base::subtle::NoBarrier_AtomicExchange(&is_enabled_, enabled ? 1 : 0); !!base::subtle::NoBarrier_AtomicExchange(&is_enabled_, enabled ? 1 : 0);
...@@ -98,13 +106,13 @@ void MediaStreamAudioTrack::Start(base::OnceClosure stop_callback) { ...@@ -98,13 +106,13 @@ void MediaStreamAudioTrack::Start(base::OnceClosure stop_callback) {
DCHECK_CALLED_ON_VALID_THREAD(thread_checker_); DCHECK_CALLED_ON_VALID_THREAD(thread_checker_);
DCHECK(!stop_callback.is_null()); DCHECK(!stop_callback.is_null());
DCHECK(stop_callback_.is_null()); DCHECK(stop_callback_.is_null());
DVLOG(1) << "Starting MediaStreamAudioTrack@" << this << '.'; SendLogMessage(base::StringPrintf("Start([this=%p])", this));
stop_callback_ = std::move(stop_callback); stop_callback_ = std::move(stop_callback);
} }
void MediaStreamAudioTrack::StopAndNotify(base::OnceClosure callback) { void MediaStreamAudioTrack::StopAndNotify(base::OnceClosure callback) {
DCHECK_CALLED_ON_VALID_THREAD(thread_checker_); DCHECK_CALLED_ON_VALID_THREAD(thread_checker_);
DVLOG(1) << "Stopping MediaStreamAudioTrack@" << this << '.'; SendLogMessage(base::StringPrintf("StopAndNotify([this=%p])", this));
if (!stop_callback_.is_null()) if (!stop_callback_.is_null())
std::move(stop_callback_).Run(); std::move(stop_callback_).Run();
...@@ -122,11 +130,22 @@ void MediaStreamAudioTrack::StopAndNotify(base::OnceClosure callback) { ...@@ -122,11 +130,22 @@ void MediaStreamAudioTrack::StopAndNotify(base::OnceClosure callback) {
} }
void MediaStreamAudioTrack::OnSetFormat(const media::AudioParameters& params) { void MediaStreamAudioTrack::OnSetFormat(const media::AudioParameters& params) {
SendLogMessage(base::StringPrintf("OnSetFormat([this=%p] {params: [%s]})",
this,
params.AsHumanReadableString().c_str()));
deliverer_.OnSetFormat(params); deliverer_.OnSetFormat(params);
} }
void MediaStreamAudioTrack::OnData(const media::AudioBus& audio_bus, void MediaStreamAudioTrack::OnData(const media::AudioBus& audio_bus,
base::TimeTicks reference_time) { base::TimeTicks reference_time) {
if (!received_audio_callback_) {
// Add log message with unique this pointer id to mark the audio track as
// alive at the first data callback.
SendLogMessage(base::StringPrintf(
"OnData([this=%p] => (audio track is alive))", this));
received_audio_callback_ = true;
}
// Note: Using NoBarrier_Load because the timing of when the audio thread sees // Note: Using NoBarrier_Load because the timing of when the audio thread sees
// a changed |is_enabled_| value can be relaxed. // a changed |is_enabled_| value can be relaxed.
const bool deliver_data = !!base::subtle::NoBarrier_Load(&is_enabled_); const bool deliver_data = !!base::subtle::NoBarrier_Load(&is_enabled_);
......
third_party/blink/renderer/platform/mediastream/media_stream_audio_source.cc
View file @ 751ef0ee
...@@ -6,9 +6,11 @@ ...@@ -6,9 +6,11 @@
#include "base/bind.h" #include "base/bind.h"
#include "base/single_thread_task_runner.h" #include "base/single_thread_task_runner.h"
#include "base/strings/stringprintf.h"
#include "base/threading/thread_task_runner_handle.h" #include "base/threading/thread_task_runner_handle.h"
#include "build/build_config.h" #include "build/build_config.h"
#include "third_party/blink/public/platform/modules/mediastream/media_stream_audio_track.h" #include "third_party/blink/public/platform/modules/mediastream/media_stream_audio_track.h"
#include "third_party/blink/public/platform/modules/webrtc/webrtc_logging.h"
#include "third_party/blink/public/platform/web_media_stream_source.h" #include "third_party/blink/public/platform/web_media_stream_source.h"
#include "third_party/blink/public/platform/web_string.h" #include "third_party/blink/public/platform/web_string.h"
#include "third_party/blink/renderer/platform/scheduler/public/post_cross_thread_task.h" #include "third_party/blink/renderer/platform/scheduler/public/post_cross_thread_task.h"
...@@ -17,6 +19,14 @@ ...@@ -17,6 +19,14 @@
namespace blink { namespace blink {
namespace {
void SendLogMessage(const std::string& message) {
blink::WebRtcLogMessage("MSAS::" + message);
}
} // namespace
const int kMaxAudioLatencyMs = 5000; const int kMaxAudioLatencyMs = 5000;
static_assert(std::numeric_limits<int>::max() / media::limits::kMaxSampleRate > static_assert(std::numeric_limits<int>::max() / media::limits::kMaxSampleRate >
kMaxAudioLatencyMs, kMaxAudioLatencyMs,
...@@ -45,8 +55,9 @@ MediaStreamAudioSource::MediaStreamAudioSource( ...@@ -45,8 +55,9 @@ MediaStreamAudioSource::MediaStreamAudioSource(
disable_local_echo_(disable_local_echo), disable_local_echo_(disable_local_echo),
is_stopped_(false), is_stopped_(false),
task_runner_(std::move(task_runner)) { task_runner_(std::move(task_runner)) {
DVLOG(1) << "MediaStreamAudioSource@" << this << "::MediaStreamAudioSource(" SendLogMessage(base::StringPrintf(
<< (is_local_source_ ? "local" : "remote") << " source)"; "MediaStreamAudioSource([this=%p] {is_local_source=%s})", this,
(is_local_source ? "local" : "remote")));
} }
MediaStreamAudioSource::MediaStreamAudioSource( MediaStreamAudioSource::MediaStreamAudioSource(
...@@ -58,7 +69,8 @@ MediaStreamAudioSource::MediaStreamAudioSource( ...@@ -58,7 +69,8 @@ MediaStreamAudioSource::MediaStreamAudioSource(
MediaStreamAudioSource::~MediaStreamAudioSource() { MediaStreamAudioSource::~MediaStreamAudioSource() {
DCHECK(task_runner_->BelongsToCurrentThread()); DCHECK(task_runner_->BelongsToCurrentThread());
DVLOG(1) << "MediaStreamAudioSource@" << this << " is being destroyed."; SendLogMessage(
base::StringPrintf("~MediaStreamAudioSource([this=%p])", this));
} }
// static // static
...@@ -74,6 +86,8 @@ bool MediaStreamAudioSource::ConnectToTrack( ...@@ -74,6 +86,8 @@ bool MediaStreamAudioSource::ConnectToTrack(
const WebMediaStreamTrack& blink_track) { const WebMediaStreamTrack& blink_track) {
DCHECK(task_runner_->BelongsToCurrentThread()); DCHECK(task_runner_->BelongsToCurrentThread());
DCHECK(!blink_track.IsNull()); DCHECK(!blink_track.IsNull());
SendLogMessage(base::StringPrintf("ConnectToTrack({track_id=%s})",
blink_track.Id().Utf8().c_str()));
// Sanity-check that there is not already a MediaStreamAudioTrack instance // Sanity-check that there is not already a MediaStreamAudioTrack instance
// associated with |blink_track|. // associated with |blink_track|.
...@@ -167,6 +181,8 @@ void MediaStreamAudioSource::DoChangeSource( ...@@ -167,6 +181,8 @@ void MediaStreamAudioSource::DoChangeSource(
std::unique_ptr<MediaStreamAudioTrack> std::unique_ptr<MediaStreamAudioTrack>
MediaStreamAudioSource::CreateMediaStreamAudioTrack(const std::string& id) { MediaStreamAudioSource::CreateMediaStreamAudioTrack(const std::string& id) {
DCHECK(task_runner_->BelongsToCurrentThread()); DCHECK(task_runner_->BelongsToCurrentThread());
SendLogMessage(
base::StringPrintf("CreateMediaStreamAudioTrack({id=%s})", id.c_str()));
return std::unique_ptr<MediaStreamAudioTrack>( return std::unique_ptr<MediaStreamAudioTrack>(
new MediaStreamAudioTrack(is_local_source())); new MediaStreamAudioTrack(is_local_source()));
} }
...@@ -190,9 +206,10 @@ void MediaStreamAudioSource::ChangeSourceImpl( ...@@ -190,9 +206,10 @@ void MediaStreamAudioSource::ChangeSourceImpl(
} }
void MediaStreamAudioSource::SetFormat(const media::AudioParameters& params) { void MediaStreamAudioSource::SetFormat(const media::AudioParameters& params) {
DVLOG(1) << "MediaStreamAudioSource@" << this << "::SetFormat(" SendLogMessage(base::StringPrintf(
<< params.AsHumanReadableString() << "), was previously set to {" "SetFormat([this=%p] {params=[%s]}, {old_params=[%s]})", this,
<< deliverer_.GetAudioParameters().AsHumanReadableString() << "}."; params.AsHumanReadableString().c_str(),
deliverer_.GetAudioParameters().AsHumanReadableString().c_str()));
deliverer_.OnSetFormat(params); deliverer_.OnSetFormat(params);
} }
...@@ -210,6 +227,7 @@ void MediaStreamAudioSource::DoStopSource() { ...@@ -210,6 +227,7 @@ void MediaStreamAudioSource::DoStopSource() {
void MediaStreamAudioSource::StopAudioDeliveryTo(MediaStreamAudioTrack* track) { void MediaStreamAudioSource::StopAudioDeliveryTo(MediaStreamAudioTrack* track) {
DCHECK(task_runner_->BelongsToCurrentThread()); DCHECK(task_runner_->BelongsToCurrentThread());
SendLogMessage(base::StringPrintf("StopAudioDeliveryTo([this=%p])", this));
const bool did_remove_last_track = deliverer_.RemoveConsumer(track); const bool did_remove_last_track = deliverer_.RemoveConsumer(track);
DVLOG(1) << "Removed MediaStreamAudioTrack@" << track DVLOG(1) << "Removed MediaStreamAudioTrack@" << track
...@@ -222,8 +240,8 @@ void MediaStreamAudioSource::StopAudioDeliveryTo(MediaStreamAudioTrack* track) { ...@@ -222,8 +240,8 @@ void MediaStreamAudioSource::StopAudioDeliveryTo(MediaStreamAudioTrack* track) {
} }
void MediaStreamAudioSource::StopSourceOnError(const std::string& why) { void MediaStreamAudioSource::StopSourceOnError(const std::string& why) {
VLOG(1) << why; SendLogMessage(base::StringPrintf("StopSourceOnError([this=%p] {why=%s})",
this, why.c_str()));
// Stop source when error occurs. // Stop source when error occurs.
PostCrossThreadTask( PostCrossThreadTask(
*task_runner_, FROM_HERE, *task_runner_, FROM_HERE,
...@@ -232,7 +250,8 @@ void MediaStreamAudioSource::StopSourceOnError(const std::string& why) { ...@@ -232,7 +250,8 @@ void MediaStreamAudioSource::StopSourceOnError(const std::string& why) {
} }
void MediaStreamAudioSource::SetMutedState(bool muted_state) { void MediaStreamAudioSource::SetMutedState(bool muted_state) {
DVLOG(3) << "MediaStreamAudioSource::SetMutedState state=" << muted_state; SendLogMessage(base::StringPrintf("SetMutedState([this=%p] {muted_state=%s})",
this, (muted_state ? "true" : "false")));
PostCrossThreadTask( PostCrossThreadTask(
*task_runner_, FROM_HERE, *task_runner_, FROM_HERE,
WTF::CrossThreadBindOnce(&WebPlatformMediaStreamSource::SetSourceMuted, WTF::CrossThreadBindOnce(&WebPlatformMediaStreamSource::SetSourceMuted,
......
third_party/blink/renderer/platform/mediastream/media_stream_source.cc
View file @ 751ef0ee
...@@ -30,6 +30,7 @@ ...@@ -30,6 +30,7 @@
#include "third_party/blink/renderer/platform/mediastream/media_stream_source.h" #include "third_party/blink/renderer/platform/mediastream/media_stream_source.h"
#include "third_party/blink/public/platform/modules/webrtc/webrtc_logging.h"
#include "third_party/blink/renderer/platform/mediastream/media_stream_audio_source.h" #include "third_party/blink/renderer/platform/mediastream/media_stream_audio_source.h"
#include "third_party/blink/renderer/platform/wtf/assertions.h" #include "third_party/blink/renderer/platform/wtf/assertions.h"
...@@ -37,6 +38,36 @@ namespace blink { ...@@ -37,6 +38,36 @@ namespace blink {
namespace { namespace {
void SendLogMessage(const std::string& message) {
blink::WebRtcLogMessage("MSS::" + message);
}
const char* StreamTypeToString(MediaStreamSource::StreamType type) {
switch (type) {
case MediaStreamSource::kTypeAudio:
return "Audio";
case MediaStreamSource::kTypeVideo:
return "Video";
default:
NOTREACHED();
}
return "Invalid";
}
const char* ReadyStateToString(MediaStreamSource::ReadyState state) {
switch (state) {
case MediaStreamSource::kReadyStateLive:
return "Live";
case MediaStreamSource::kReadyStateMuted:
return "Muted";
case MediaStreamSource::kReadyStateEnded:
return "Ended";
default:
NOTREACHED();
}
return "Invalid";
}
void GetSourceSettings(const blink::WebMediaStreamSource& web_source, void GetSourceSettings(const blink::WebMediaStreamSource& web_source,
blink::WebMediaStreamTrack::Settings& settings) { blink::WebMediaStreamTrack::Settings& settings) {
blink::MediaStreamAudioSource* const source = blink::MediaStreamAudioSource* const source =
...@@ -68,13 +99,28 @@ MediaStreamSource::MediaStreamSource(const String& id, ...@@ -68,13 +99,28 @@ MediaStreamSource::MediaStreamSource(const String& id,
name_(name), name_(name),
remote_(remote), remote_(remote),
ready_state_(ready_state), ready_state_(ready_state),
requires_consumer_(requires_consumer) {} requires_consumer_(requires_consumer) {
SendLogMessage(
String::Format(
"MediaStreamSource({id=%s}, {type=%s}, {name=%s}, {remote=%d}, "
"{ready_state=%s}",
id.Utf8().c_str(), StreamTypeToString(type), name.Utf8().c_str(),
remote, ReadyStateToString(ready_state))
.Utf8());
}
void MediaStreamSource::SetGroupId(const String& group_id) { void MediaStreamSource::SetGroupId(const String& group_id) {
SendLogMessage(
String::Format("SetGroupId({group_id=%s})", group_id.Utf8().c_str())
.Utf8());
group_id_ = group_id; group_id_ = group_id;
} }
void MediaStreamSource::SetReadyState(ReadyState ready_state) { void MediaStreamSource::SetReadyState(ReadyState ready_state) {
SendLogMessage(String::Format("SetReadyState({id=%s}, {ready_state=%s})",
Id().Utf8().c_str(),
ReadyStateToString(ready_state))
.Utf8());
if (ready_state_ != kReadyStateEnded && ready_state_ != ready_state) { if (ready_state_ != kReadyStateEnded && ready_state_ != ready_state) {
ready_state_ = ready_state; ready_state_ = ready_state;
...@@ -171,6 +217,12 @@ void MediaStreamSource::GetSettings(WebMediaStreamTrack::Settings& settings) { ...@@ -171,6 +217,12 @@ void MediaStreamSource::GetSettings(WebMediaStreamTrack::Settings& settings) {
void MediaStreamSource::SetAudioFormat(size_t number_of_channels, void MediaStreamSource::SetAudioFormat(size_t number_of_channels,
float sample_rate) { float sample_rate) {
SendLogMessage(
String::Format(
"SetAudioFormat({id=%s}, {number_of_channels=%d}, {sample_rate=%f})",
Id().Utf8().c_str(), static_cast<int>(number_of_channels),
sample_rate)
.Utf8());
DCHECK(requires_consumer_); DCHECK(requires_consumer_);
MutexLocker locker(audio_consumers_lock_); MutexLocker locker(audio_consumers_lock_);
for (AudioDestinationConsumer* consumer : audio_consumers_) for (AudioDestinationConsumer* consumer : audio_consumers_)
......
third_party/blink/renderer/platform/peerconnection/webrtc_audio_sink.cc
View file @ 751ef0ee
...@@ -11,10 +11,20 @@ ...@@ -11,10 +11,20 @@
#include "base/location.h" #include "base/location.h"
#include "base/logging.h" #include "base/logging.h"
#include "base/stl_util.h" #include "base/stl_util.h"
#include "base/strings/stringprintf.h"
#include "third_party/blink/public/platform/modules/webrtc/webrtc_logging.h"
#include "third_party/blink/renderer/platform/scheduler/public/post_cross_thread_task.h" #include "third_party/blink/renderer/platform/scheduler/public/post_cross_thread_task.h"
#include "third_party/blink/renderer/platform/wtf/cross_thread_functional.h" #include "third_party/blink/renderer/platform/wtf/cross_thread_functional.h"
#include "third_party/webrtc/rtc_base/ref_counted_object.h" #include "third_party/webrtc/rtc_base/ref_counted_object.h"
namespace {
void SendLogMessage(const std::string& message) {
blink::WebRtcLogMessage("WRAS::" + message);
}
} // namespace
namespace WTF { namespace WTF {
template <> template <>
...@@ -48,12 +58,14 @@ WebRtcAudioSink::WebRtcAudioSink( ...@@ -48,12 +58,14 @@ WebRtcAudioSink::WebRtcAudioSink(
fifo_(ConvertToBaseRepeatingCallback( fifo_(ConvertToBaseRepeatingCallback(
CrossThreadBindRepeating(&WebRtcAudioSink::DeliverRebufferedAudio, CrossThreadBindRepeating(&WebRtcAudioSink::DeliverRebufferedAudio,
CrossThreadUnretained(this)))) { CrossThreadUnretained(this)))) {
DVLOG(1) << "WebRtcAudioSink::WebRtcAudioSink()"; SendLogMessage(base::StringPrintf("WebRtcAudioSink({label=%s})",
adapter_->label().c_str()));
} }
WebRtcAudioSink::~WebRtcAudioSink() { WebRtcAudioSink::~WebRtcAudioSink() {
DCHECK_CALLED_ON_VALID_THREAD(thread_checker_); DCHECK_CALLED_ON_VALID_THREAD(thread_checker_);
DVLOG(1) << "WebRtcAudioSink::~WebRtcAudioSink()"; SendLogMessage(base::StringPrintf("~WebRtcAudioSink([label=%s])",
adapter_->label().c_str()));
} }
void WebRtcAudioSink::SetAudioProcessor( void WebRtcAudioSink::SetAudioProcessor(
...@@ -72,6 +84,9 @@ void WebRtcAudioSink::SetLevel( ...@@ -72,6 +84,9 @@ void WebRtcAudioSink::SetLevel(
void WebRtcAudioSink::OnEnabledChanged(bool enabled) { void WebRtcAudioSink::OnEnabledChanged(bool enabled) {
DCHECK_CALLED_ON_VALID_THREAD(thread_checker_); DCHECK_CALLED_ON_VALID_THREAD(thread_checker_);
SendLogMessage(base::StringPrintf("OnEnabledChanged([label=%s] {enabled=%s})",
adapter_->label().c_str(),
(enabled ? "true" : "false")));
PostCrossThreadTask( PostCrossThreadTask(
*adapter_->signaling_task_runner(), FROM_HERE, *adapter_->signaling_task_runner(), FROM_HERE,
CrossThreadBindOnce( CrossThreadBindOnce(
...@@ -90,6 +105,9 @@ void WebRtcAudioSink::OnData(const media::AudioBus& audio_bus, ...@@ -90,6 +105,9 @@ void WebRtcAudioSink::OnData(const media::AudioBus& audio_bus,
void WebRtcAudioSink::OnSetFormat(const media::AudioParameters& params) { void WebRtcAudioSink::OnSetFormat(const media::AudioParameters& params) {
DCHECK(params.IsValid()); DCHECK(params.IsValid());
SendLogMessage(base::StringPrintf("OnSetFormat([label=%s] {params=[%s]})",
adapter_->label().c_str(),
params.AsHumanReadableString().c_str()));
params_ = params; params_ = params;
// Make sure that our params always reflect a buffer size of 10ms. // Make sure that our params always reflect a buffer size of 10ms.
params_.set_frames_per_buffer(params_.sample_rate() / 100); params_.set_frames_per_buffer(params_.sample_rate() / 100);
...@@ -129,14 +147,19 @@ WebRtcAudioSink::Adapter::Adapter( ...@@ -129,14 +147,19 @@ WebRtcAudioSink::Adapter::Adapter(
scoped_refptr<base::SingleThreadTaskRunner> signaling_task_runner, scoped_refptr<base::SingleThreadTaskRunner> signaling_task_runner,
scoped_refptr<base::SingleThreadTaskRunner> main_task_runner) scoped_refptr<base::SingleThreadTaskRunner> main_task_runner)
: webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>(label), : webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>(label),
label_(label),
source_(std::move(source)), source_(std::move(source)),
signaling_task_runner_(std::move(signaling_task_runner)), signaling_task_runner_(std::move(signaling_task_runner)),
main_task_runner_(std::move(main_task_runner)) { main_task_runner_(std::move(main_task_runner)) {
DCHECK(signaling_task_runner_); DCHECK(signaling_task_runner_);
DCHECK(main_task_runner_); DCHECK(main_task_runner_);
SendLogMessage(
base::StringPrintf("Adapter::Adapter({label=%s})", label_.c_str()));
} }
WebRtcAudioSink::Adapter::~Adapter() { WebRtcAudioSink::Adapter::~Adapter() {
SendLogMessage(
base::StringPrintf("Adapter::~Adapter([label=%s])", label_.c_str()));
if (audio_processor_) { if (audio_processor_) {
PostCrossThreadTask(*main_task_runner_.get(), FROM_HERE, PostCrossThreadTask(*main_task_runner_.get(), FROM_HERE,
CrossThreadBindOnce(&DereferenceOnMainThread, CrossThreadBindOnce(&DereferenceOnMainThread,
...@@ -163,6 +186,9 @@ std::string WebRtcAudioSink::Adapter::kind() const { ...@@ -163,6 +186,9 @@ std::string WebRtcAudioSink::Adapter::kind() const {
bool WebRtcAudioSink::Adapter::set_enabled(bool enable) { bool WebRtcAudioSink::Adapter::set_enabled(bool enable) {
DCHECK(!signaling_task_runner_ || DCHECK(!signaling_task_runner_ ||
signaling_task_runner_->RunsTasksInCurrentSequence()); signaling_task_runner_->RunsTasksInCurrentSequence());
SendLogMessage(
base::StringPrintf("Adapter::set_enabled([label=%s] {enable=%s})",
label_.c_str(), (enable ? "true" : "false")));
return webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>::set_enabled( return webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>::set_enabled(
enable); enable);
} }
...@@ -171,6 +197,8 @@ void WebRtcAudioSink::Adapter::AddSink(webrtc::AudioTrackSinkInterface* sink) { ...@@ -171,6 +197,8 @@ void WebRtcAudioSink::Adapter::AddSink(webrtc::AudioTrackSinkInterface* sink) {
DCHECK(!signaling_task_runner_ || DCHECK(!signaling_task_runner_ ||
signaling_task_runner_->RunsTasksInCurrentSequence()); signaling_task_runner_->RunsTasksInCurrentSequence());
DCHECK(sink); DCHECK(sink);
SendLogMessage(
base::StringPrintf("Adapter::AddSink({label=%s})", label_.c_str()));
base::AutoLock auto_lock(lock_); base::AutoLock auto_lock(lock_);
DCHECK(!base::Contains(sinks_, sink)); DCHECK(!base::Contains(sinks_, sink));
sinks_.push_back(sink); sinks_.push_back(sink);
...@@ -180,6 +208,8 @@ void WebRtcAudioSink::Adapter::RemoveSink( ...@@ -180,6 +208,8 @@ void WebRtcAudioSink::Adapter::RemoveSink(
webrtc::AudioTrackSinkInterface* sink) { webrtc::AudioTrackSinkInterface* sink) {
DCHECK(!signaling_task_runner_ || DCHECK(!signaling_task_runner_ ||
signaling_task_runner_->RunsTasksInCurrentSequence()); signaling_task_runner_->RunsTasksInCurrentSequence());
SendLogMessage(
base::StringPrintf("Adapter::RemoveSink([label=%s])", label_.c_str()));
base::AutoLock auto_lock(lock_); base::AutoLock auto_lock(lock_);
const auto it = std::find(sinks_.begin(), sinks_.end(), sink); const auto it = std::find(sinks_.begin(), sinks_.end(), sink);
if (it != sinks_.end()) if (it != sinks_.end())
...@@ -200,6 +230,9 @@ bool WebRtcAudioSink::Adapter::GetSignalLevel(int* level) { ...@@ -200,6 +230,9 @@ bool WebRtcAudioSink::Adapter::GetSignalLevel(int* level) {
// Convert from float in range [0.0,1.0] to an int in range [0,32767]. // Convert from float in range [0.0,1.0] to an int in range [0,32767].
*level = static_cast<int>(signal_level * std::numeric_limits<int16_t>::max() + *level = static_cast<int>(signal_level * std::numeric_limits<int16_t>::max() +
0.5f /* rounding to nearest int */); 0.5f /* rounding to nearest int */);
SendLogMessage(
base::StringPrintf("Adapter::GetSignalLevel([label=%s]) => (level=%d)",
label_.c_str(), *level));
return true; return true;
} }
......
third_party/blink/renderer/platform/peerconnection/webrtc_audio_sink.h
View file @ 751ef0ee
...@@ -102,6 +102,8 @@ class PLATFORM_EXPORT WebRtcAudioSink : public WebMediaStreamAudioSink { ...@@ -102,6 +102,8 @@ class PLATFORM_EXPORT WebRtcAudioSink : public WebMediaStreamAudioSink {
size_t number_of_channels, size_t number_of_channels,
size_t number_of_frames); size_t number_of_frames);
std::string label() const { return label_; }
// webrtc::MediaStreamTrack implementation. // webrtc::MediaStreamTrack implementation.
std::string kind() const override; std::string kind() const override;
bool set_enabled(bool enable) override; bool set_enabled(bool enable) override;
...@@ -118,6 +120,8 @@ class PLATFORM_EXPORT WebRtcAudioSink : public WebMediaStreamAudioSink { ...@@ -118,6 +120,8 @@ class PLATFORM_EXPORT WebRtcAudioSink : public WebMediaStreamAudioSink {
~Adapter() override; ~Adapter() override;
private: private:
const std::string label_;
const scoped_refptr<webrtc::AudioSourceInterface> source_; const scoped_refptr<webrtc::AudioSourceInterface> source_;
// Task runner for operations that must be done on libjingle's signaling // Task runner for operations that must be done on libjingle's signaling
......
third_party/blink/renderer/platform/webrtc/peer_connection_remote_audio_source.cc
View file @ 751ef0ee
...@@ -5,8 +5,10 @@ ...@@ -5,8 +5,10 @@
#include "third_party/blink/renderer/platform/webrtc/peer_connection_remote_audio_source.h" #include "third_party/blink/renderer/platform/webrtc/peer_connection_remote_audio_source.h"
#include "base/logging.h" #include "base/logging.h"
#include "base/strings/stringprintf.h"
#include "base/time/time.h" #include "base/time/time.h"
#include "media/base/audio_bus.h" #include "media/base/audio_bus.h"
#include "third_party/blink/public/platform/modules/webrtc/webrtc_logging.h"
namespace blink { namespace blink {
...@@ -14,19 +16,26 @@ namespace { ...@@ -14,19 +16,26 @@ namespace {
// Used as an identifier for the down-casters. // Used as an identifier for the down-casters.
void* const kPeerConnectionRemoteTrackIdentifier = void* const kPeerConnectionRemoteTrackIdentifier =
const_cast<void**>(&kPeerConnectionRemoteTrackIdentifier); const_cast<void**>(&kPeerConnectionRemoteTrackIdentifier);
void SendLogMessage(const std::string& message) {
blink::WebRtcLogMessage("PCRAS::" + message);
}
} // namespace } // namespace
PeerConnectionRemoteAudioTrack::PeerConnectionRemoteAudioTrack( PeerConnectionRemoteAudioTrack::PeerConnectionRemoteAudioTrack(
scoped_refptr<webrtc::AudioTrackInterface> track_interface) scoped_refptr<webrtc::AudioTrackInterface> track_interface)
: MediaStreamAudioTrack(false /* is_local_track */), : MediaStreamAudioTrack(false /* is_local_track */),
track_interface_(std::move(track_interface)) { track_interface_(std::move(track_interface)) {
DVLOG(1) blink::WebRtcLogMessage(
<< "PeerConnectionRemoteAudioTrack::PeerConnectionRemoteAudioTrack()"; base::StringPrintf("PCRAT::PeerConnectionRemoteAudioTrack({id=%s})",
track_interface_->id().c_str()));
} }
PeerConnectionRemoteAudioTrack::~PeerConnectionRemoteAudioTrack() { PeerConnectionRemoteAudioTrack::~PeerConnectionRemoteAudioTrack() {
DVLOG(1) blink::WebRtcLogMessage(
<< "PeerConnectionRemoteAudioTrack::~PeerConnectionRemoteAudioTrack()"; base::StringPrintf("PCRAT::~PeerConnectionRemoteAudioTrack([id=%s])",
track_interface_->id().c_str()));
// Ensure the track is stopped. // Ensure the track is stopped.
MediaStreamAudioTrack::Stop(); MediaStreamAudioTrack::Stop();
} }
...@@ -42,6 +51,9 @@ PeerConnectionRemoteAudioTrack* PeerConnectionRemoteAudioTrack::From( ...@@ -42,6 +51,9 @@ PeerConnectionRemoteAudioTrack* PeerConnectionRemoteAudioTrack::From(
void PeerConnectionRemoteAudioTrack::SetEnabled(bool enabled) { void PeerConnectionRemoteAudioTrack::SetEnabled(bool enabled) {
DCHECK_CALLED_ON_VALID_THREAD(thread_checker_); DCHECK_CALLED_ON_VALID_THREAD(thread_checker_);
blink::WebRtcLogMessage(base::StringPrintf(
"PCRAT::SetEnabled([id=%s] {enabled=%s})", track_interface_->id().c_str(),
(enabled ? "true" : "false")));
// This affects the shared state of the source for whether or not it's a part // This affects the shared state of the source for whether or not it's a part
// of the mixed audio that's rendered for remote tracks from WebRTC. // of the mixed audio that's rendered for remote tracks from WebRTC.
...@@ -67,13 +79,13 @@ PeerConnectionRemoteAudioSource::PeerConnectionRemoteAudioSource( ...@@ -67,13 +79,13 @@ PeerConnectionRemoteAudioSource::PeerConnectionRemoteAudioSource(
track_interface_(std::move(track_interface)), track_interface_(std::move(track_interface)),
is_sink_of_peer_connection_(false) { is_sink_of_peer_connection_(false) {
DCHECK(track_interface_); DCHECK(track_interface_);
DVLOG(1) SendLogMessage(base::StringPrintf("PeerConnectionRemoteAudioSource([id=%s])",
<< "PeerConnectionRemoteAudioSource::PeerConnectionRemoteAudioSource()"; track_interface_->id().c_str()));
} }
PeerConnectionRemoteAudioSource::~PeerConnectionRemoteAudioSource() { PeerConnectionRemoteAudioSource::~PeerConnectionRemoteAudioSource() {
DVLOG(1) SendLogMessage(base::StringPrintf("~PeerConnectionRemoteAudioSource([id=%s])",
<< "PeerConnectionRemoteAudioSource::~PeerConnectionRemoteAudioSource()"; track_interface_->id().c_str()));
EnsureSourceIsStopped(); EnsureSourceIsStopped();
} }
...@@ -88,8 +100,8 @@ bool PeerConnectionRemoteAudioSource::EnsureSourceIsStarted() { ...@@ -88,8 +100,8 @@ bool PeerConnectionRemoteAudioSource::EnsureSourceIsStarted() {
DCHECK_CALLED_ON_VALID_THREAD(thread_checker_); DCHECK_CALLED_ON_VALID_THREAD(thread_checker_);
if (is_sink_of_peer_connection_) if (is_sink_of_peer_connection_)
return true; return true;
VLOG(1) << "Starting PeerConnection remote audio source with id=" SendLogMessage(base::StringPrintf("EnsureSourceIsStarted([id=%s])",
<< track_interface_->id(); track_interface_->id().c_str()));
track_interface_->AddSink(this); track_interface_->AddSink(this);
is_sink_of_peer_connection_ = true; is_sink_of_peer_connection_ = true;
return true; return true;
...@@ -98,10 +110,10 @@ bool PeerConnectionRemoteAudioSource::EnsureSourceIsStarted() { ...@@ -98,10 +110,10 @@ bool PeerConnectionRemoteAudioSource::EnsureSourceIsStarted() {
void PeerConnectionRemoteAudioSource::EnsureSourceIsStopped() { void PeerConnectionRemoteAudioSource::EnsureSourceIsStopped() {
DCHECK_CALLED_ON_VALID_THREAD(thread_checker_); DCHECK_CALLED_ON_VALID_THREAD(thread_checker_);
if (is_sink_of_peer_connection_) { if (is_sink_of_peer_connection_) {
SendLogMessage(base::StringPrintf("EnsureSourceIsStopped([id=%s])",
track_interface_->id().c_str()));
track_interface_->RemoveSink(this); track_interface_->RemoveSink(this);
is_sink_of_peer_connection_ = false; is_sink_of_peer_connection_ = false;
VLOG(1) << "Stopped PeerConnection remote audio source with id="
<< track_interface_->id();
} }
} }
......
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