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 |
new file mode 100644 |
index 0000000000000000000000000000000000000000..e3940ab72b3bce7149897913d4af49745b673da0 |
--- /dev/null |
+++ b/content/renderer/media/webrtc/media_stream_remote_audio_track.cc |
@@ -0,0 +1,237 @@ |
+// Copyright 2015 The Chromium Authors. All rights reserved. |
+// Use of this source code is governed by a BSD-style license that can be |
+// found in the LICENSE file. |
+ |
+#include "content/renderer/media/webrtc/media_stream_remote_audio_track.h" |
+ |
+#include <stddef.h> |
+ |
+#include <list> |
+ |
+#include "base/logging.h" |
+#include "content/public/renderer/media_stream_audio_sink.h" |
+#include "third_party/webrtc/api/mediastreaminterface.h" |
+ |
+namespace content { |
+ |
+class MediaStreamRemoteAudioSource::AudioSink |
+ : public webrtc::AudioTrackSinkInterface { |
+ public: |
+ AudioSink() { |
+ } |
+ ~AudioSink() override { |
+ DCHECK(sinks_.empty()); |
+ } |
+ |
+ void Add(MediaStreamAudioSink* sink, MediaStreamAudioTrack* track, |
+ bool enabled) { |
+ DCHECK(thread_checker_.CalledOnValidThread()); |
+ SinkInfo info(sink, track, enabled); |
+ base::AutoLock lock(lock_); |
+ sinks_.push_back(info); |
+ } |
+ |
+ void Remove(MediaStreamAudioSink* sink, MediaStreamAudioTrack* track) { |
+ DCHECK(thread_checker_.CalledOnValidThread()); |
+ base::AutoLock lock(lock_); |
+ sinks_.remove_if([&sink, &track](const SinkInfo& info) { |
+ return info.sink == sink && info.track == track; |
+ }); |
+ } |
+ |
+ void SetEnabled(MediaStreamAudioTrack* track, bool enabled) { |
+ DCHECK(thread_checker_.CalledOnValidThread()); |
+ base::AutoLock lock(lock_); |
+ for (SinkInfo& info : sinks_) { |
+ if (info.track == track) |
+ info.enabled = enabled; |
+ } |
+ } |
+ |
+ void RemoveAll(MediaStreamAudioTrack* track) { |
+ base::AutoLock lock(lock_); |
+ sinks_.remove_if([&track](const SinkInfo& info) { |
+ return info.track == track; |
+ }); |
+ } |
+ |
+ bool IsNeeded() const { |
+ DCHECK(thread_checker_.CalledOnValidThread()); |
+ return !sinks_.empty(); |
+ } |
+ |
+ private: |
+ void OnData(const void* audio_data, int bits_per_sample, int sample_rate, |
+ size_t number_of_channels, size_t number_of_frames) override { |
+ if (!audio_bus_ || |
+ static_cast<size_t>(audio_bus_->channels()) != number_of_channels || |
+ static_cast<size_t>(audio_bus_->frames()) != 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 || |
+ static_cast<size_t>(params_.channels()) != number_of_channels || |
+ params_.sample_rate() != sample_rate || |
+ static_cast<size_t>(params_.frames_per_buffer()) != number_of_frames) { |
+ 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 (const SinkInfo& info : sinks_) { |
+ if (info.enabled) { |
+ if (format_changed) |
+ info.sink->OnSetFormat(params_); |
+ info.sink->OnData(*audio_bus_.get(), estimated_capture_time); |
+ } |
+ } |
+ } |
+ |
+ mutable base::Lock lock_; |
+ struct SinkInfo { |
+ SinkInfo(MediaStreamAudioSink* sink, MediaStreamAudioTrack* track, |
+ bool enabled) : sink(sink), track(track), enabled(enabled) {} |
+ MediaStreamAudioSink* sink; |
+ MediaStreamAudioTrack* track; |
+ bool enabled; |
+ }; |
+ std::list<SinkInfo> sinks_; |
+ base::ThreadChecker thread_checker_; |
+ media::AudioParameters params_; // Only used on the callback thread. |
+ std::unique_ptr<media::AudioBus> |
+ audio_bus_; // Only used on the callback thread. |
+}; |
+ |
+MediaStreamRemoteAudioTrack::MediaStreamRemoteAudioTrack( |
+ const blink::WebMediaStreamSource& source, bool enabled) |
+ : MediaStreamAudioTrack(false), source_(source), enabled_(enabled) { |
+ DCHECK(source.getExtraData()); // Make sure the source has a native source. |
+} |
+ |
+MediaStreamRemoteAudioTrack::~MediaStreamRemoteAudioTrack() { |
+ DCHECK(main_render_thread_checker_.CalledOnValidThread()); |
+ // Ensure the track is stopped. |
+ MediaStreamAudioTrack::Stop(); |
+} |
+ |
+void MediaStreamRemoteAudioTrack::SetEnabled(bool enabled) { |
+ DCHECK(main_render_thread_checker_.CalledOnValidThread()); |
+ |
+ // 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. |
+ // All tracks from the same source will share this state and thus can step |
+ // on each other's toes. |
+ // This is also why we can't check the |enabled_| state for equality with |
+ // |enabled| before setting the mixing enabled state. |enabled_| and the |
+ // shared state might not be the same. |
+ source()->SetEnabledForMixing(enabled); |
+ |
+ enabled_ = enabled; |
+ source()->SetSinksEnabled(this, enabled); |
+} |
+ |
+void MediaStreamRemoteAudioTrack::OnStop() { |
+ DCHECK(main_render_thread_checker_.CalledOnValidThread()); |
+ DVLOG(1) << "MediaStreamRemoteAudioTrack::OnStop()"; |
+ |
+ source()->RemoveAll(this); |
+ |
+ // Stop means that a track should be stopped permanently. But |
+ // since there is no proper way of doing that on a remote track, we can |
+ // at least disable the track. Blink will not call down to the content layer |
+ // after a track has been stopped. |
+ SetEnabled(false); |
+} |
+ |
+void MediaStreamRemoteAudioTrack::AddSink(MediaStreamAudioSink* sink) { |
+ DCHECK(main_render_thread_checker_.CalledOnValidThread()); |
+ return source()->AddSink(sink, this, enabled_); |
+} |
+ |
+void MediaStreamRemoteAudioTrack::RemoveSink(MediaStreamAudioSink* sink) { |
+ DCHECK(main_render_thread_checker_.CalledOnValidThread()); |
+ return source()->RemoveSink(sink, this); |
+} |
+ |
+media::AudioParameters MediaStreamRemoteAudioTrack::GetOutputFormat() const { |
+ DCHECK(main_render_thread_checker_.CalledOnValidThread()); |
+ // This method is not implemented on purpose and should be removed. |
+ // TODO(tommi): See comment for GetOutputFormat in MediaStreamAudioTrack. |
+ NOTIMPLEMENTED(); |
+ return media::AudioParameters(); |
+} |
+ |
+webrtc::AudioTrackInterface* MediaStreamRemoteAudioTrack::GetAudioAdapter() { |
+ DCHECK(main_render_thread_checker_.CalledOnValidThread()); |
+ return source()->GetAudioAdapter(); |
+} |
+ |
+MediaStreamRemoteAudioSource* MediaStreamRemoteAudioTrack::source() const { |
+ return static_cast<MediaStreamRemoteAudioSource*>(source_.getExtraData()); |
+} |
+ |
+MediaStreamRemoteAudioSource::MediaStreamRemoteAudioSource( |
+ const scoped_refptr<webrtc::AudioTrackInterface>& track) : track_(track) {} |
+ |
+MediaStreamRemoteAudioSource::~MediaStreamRemoteAudioSource() { |
+ DCHECK(thread_checker_.CalledOnValidThread()); |
+} |
+ |
+void MediaStreamRemoteAudioSource::SetEnabledForMixing(bool enabled) { |
+ DCHECK(thread_checker_.CalledOnValidThread()); |
+ track_->set_enabled(enabled); |
+} |
+ |
+void MediaStreamRemoteAudioSource::AddSink(MediaStreamAudioSink* sink, |
+ MediaStreamAudioTrack* track, |
+ bool enabled) { |
+ DCHECK(thread_checker_.CalledOnValidThread()); |
+ if (!sink_) { |
+ sink_.reset(new AudioSink()); |
+ track_->AddSink(sink_.get()); |
+ } |
+ |
+ sink_->Add(sink, track, enabled); |
+} |
+ |
+void MediaStreamRemoteAudioSource::RemoveSink(MediaStreamAudioSink* sink, |
+ MediaStreamAudioTrack* track) { |
+ DCHECK(thread_checker_.CalledOnValidThread()); |
+ DCHECK(sink_); |
+ |
+ sink_->Remove(sink, track); |
+ |
+ if (!sink_->IsNeeded()) { |
+ track_->RemoveSink(sink_.get()); |
+ sink_.reset(); |
+ } |
+} |
+ |
+void MediaStreamRemoteAudioSource::SetSinksEnabled(MediaStreamAudioTrack* track, |
+ bool enabled) { |
+ if (sink_) |
+ sink_->SetEnabled(track, enabled); |
+} |
+ |
+void MediaStreamRemoteAudioSource::RemoveAll(MediaStreamAudioTrack* track) { |
+ if (sink_) |
+ sink_->RemoveAll(track); |
+} |
+ |
+webrtc::AudioTrackInterface* MediaStreamRemoteAudioSource::GetAudioAdapter() { |
+ DCHECK(thread_checker_.CalledOnValidThread()); |
+ return track_.get(); |
+} |
+ |
+} // namespace content |