Index: content/renderer/media/webrtc/media_stream_remote_audio_track.cc |
diff --git a/content/renderer/media/webrtc/media_stream_remote_audio_track.cc b/content/renderer/media/webrtc/media_stream_remote_audio_track.cc |
index e345b883d6446441f1dff2f9488c4f0d54839809..5ca07a9dba415acbd0d552ec18e5e0fb4500132f 100644 |
--- a/content/renderer/media/webrtc/media_stream_remote_audio_track.cc |
+++ b/content/renderer/media/webrtc/media_stream_remote_audio_track.cc |
@@ -4,14 +4,91 @@ |
#include "content/renderer/media/webrtc/media_stream_remote_audio_track.h" |
+#include <list> |
+ |
#include "base/logging.h" |
+#include "content/public/renderer/media_stream_audio_sink.h" |
#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" |
namespace content { |
+class MediaStreamRemoteAudioTrack::AudioSink |
+ : public webrtc::AudioTrackSinkInterface { |
+ public: |
+ AudioSink() { |
+ } |
+ ~AudioSink() override { |
+ DCHECK(sinks_.empty()); |
+ } |
+ |
+ void Add(MediaStreamAudioSink* sink) { |
+ DCHECK(thread_checker_.CalledOnValidThread()); |
+ base::AutoLock lock(lock_); |
+ sinks_.push_back(sink); |
+ } |
+ |
+ void Remove(MediaStreamAudioSink* sink) { |
+ DCHECK(thread_checker_.CalledOnValidThread()); |
+ base::AutoLock lock(lock_); |
+ sinks_.remove(sink); |
+ } |
+ |
+ bool IsNeeded() const { |
+ DCHECK(thread_checker_.CalledOnValidThread()); |
+ return !sinks_.empty(); |
+ } |
+ |
+ const media::AudioParameters GetOutputFormat() const { |
+ base::AutoLock lock(lock_); |
+ return params_; |
+ } |
+ |
+ private: |
+ void OnData(const void* audio_data, int bits_per_sample, int sample_rate, |
+ int number_of_channels, size_t number_of_frames) override { |
+ if (!audio_bus_ || audio_bus_->channels() != number_of_channels || |
+ audio_bus_->frames() != static_cast<int>(number_of_frames)) { |
+ audio_bus_ = media::AudioBus::Create(number_of_channels, |
+ number_of_frames); |
+ } |
+ |
+ audio_bus_->FromInterleaved(audio_data, number_of_frames, |
+ bits_per_sample / 8); |
+ |
+ bool format_changed = false; |
+ if (params_.format() != media::AudioParameters::AUDIO_PCM_LOW_LATENCY || |
+ params_.channels() != number_of_channels || |
+ params_.sample_rate() != sample_rate || |
+ params_.frames_per_buffer() != static_cast<int>(number_of_frames)) { |
+ base::AutoLock lock(lock_); // Only lock on this thread when writing. |
+ params_ = media::AudioParameters( |
+ media::AudioParameters::AUDIO_PCM_LOW_LATENCY, |
+ media::GuessChannelLayout(number_of_channels), |
+ sample_rate, 16, number_of_frames); |
+ format_changed = true; |
+ } |
+ |
+ // TODO(tommi): We should get the timestamp from WebRTC. |
+ base::TimeTicks estimated_capture_time(base::TimeTicks::Now()); |
+ |
+ base::AutoLock lock(lock_); |
+ for (auto* sink : sinks_) { |
+ if (format_changed) |
+ sink->OnSetFormat(params_); |
+ sink->OnData(*audio_bus_.get(), estimated_capture_time); |
+ } |
+ } |
+ |
+ mutable base::Lock lock_; |
+ std::list<MediaStreamAudioSink*> sinks_; |
+ base::ThreadChecker thread_checker_; |
+ media::AudioParameters params_; |
+ scoped_ptr<media::AudioBus> audio_bus_; // Only used on the callback thread. |
+}; |
+ |
MediaStreamRemoteAudioTrack::MediaStreamRemoteAudioTrack( |
const scoped_refptr<webrtc::AudioTrackInterface>& track) |
- : MediaStreamTrack(false), track_(track) { |
+ : MediaStreamAudioTrack(false), track_(track) { |
} |
MediaStreamRemoteAudioTrack::~MediaStreamRemoteAudioTrack() { |
@@ -32,6 +109,35 @@ void MediaStreamRemoteAudioTrack::Stop() { |
SetEnabled(false); |
} |
+void MediaStreamRemoteAudioTrack::AddSink(MediaStreamAudioSink* sink) { |
+ DCHECK(main_render_thread_checker_.CalledOnValidThread()); |
+ |
+ if (!sink_) { |
+ sink_.reset(new AudioSink()); |
+ track_->AddSink(sink_.get()); |
+ } |
+ |
+ sink_->Add(sink); |
+} |
+ |
+void MediaStreamRemoteAudioTrack::RemoveSink(MediaStreamAudioSink* sink) { |
+ DCHECK(main_render_thread_checker_.CalledOnValidThread()); |
+ |
+ DCHECK(sink_); |
+ |
+ sink_->Remove(sink); |
+ |
+ if (!sink_->IsNeeded()) { |
+ track_->RemoveSink(sink_.get()); |
+ sink_.reset(); |
+ } |
+} |
+ |
+media::AudioParameters MediaStreamRemoteAudioTrack::GetOutputFormat() const { |
+ DCHECK(main_render_thread_checker_.CalledOnValidThread()); |
+ return sink_ ? sink_->GetOutputFormat() : media::AudioParameters(); |
+} |
+ |
webrtc::AudioTrackInterface* MediaStreamRemoteAudioTrack::GetAudioAdapter() { |
DCHECK(main_render_thread_checker_.CalledOnValidThread()); |
return track_.get(); |