Add support for unmixed audio from remote WebRTC remote tracks.

content/renderer/media/webrtc/media_stream_remote_audio_track.cc

 // 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 <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() {
   DCHECK(main_render_thread_checker_.CalledOnValidThread());
 }
 
 void MediaStreamRemoteAudioTrack::SetEnabled(bool enabled) {
   DCHECK(main_render_thread_checker_.CalledOnValidThread());
   track_->set_enabled(enabled);
 }
 
 void MediaStreamRemoteAudioTrack::Stop() {
   DCHECK(main_render_thread_checker_.CalledOnValidThread());
   // 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());
+
+  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();
 }
 
 }  // namespace content