Add support for unmixed audio from remote WebRTC remote tracks.

content/renderer/media/webrtc/media_stream_remote_audio_track.h

Index: content/renderer/media/webrtc/media_stream_remote_audio_track.h
diff --git a/content/renderer/media/webrtc/media_stream_remote_audio_track.h b/content/renderer/media/webrtc/media_stream_remote_audio_track.h
index 639c2e9616bf96c3e14f3b007a37c615f5cbbab3..1f0a7b5a1e89623cb203bbd467ac83b9af7b9ae9 100644
--- a/content/renderer/media/webrtc/media_stream_remote_audio_track.h
+++ b/content/renderer/media/webrtc/media_stream_remote_audio_track.h
@@ -6,13 +6,16 @@
 #define CONTENT_RENDERER_MEDIA_WEBRTC_MEDIA_STREAM_REMOTE_AUDIO_TRACK_H_
 
 #include "base/memory/ref_counted.h"
-#include "content/renderer/media/media_stream_track.h"
+#include "content/renderer/media/media_stream_audio_track.h"
 
 namespace content {
 
 // MediaStreamRemoteAudioTrack is a WebRTC specific implementation of an
 // audio track received from a PeerConnection.
-class MediaStreamRemoteAudioTrack : public MediaStreamTrack {
+// TODO(tommi): Chrome shouldn't have to care about remote vs local so
+// we should have a single track implementation that delegates to the
+// sources that do different things depending on the type of source.
+class MediaStreamRemoteAudioTrack : public MediaStreamAudioTrack {
  public:
   explicit MediaStreamRemoteAudioTrack(
       const scoped_refptr<webrtc::AudioTrackInterface>& track);
@@ -21,9 +24,15 @@ class MediaStreamRemoteAudioTrack : public MediaStreamTrack {
   void SetEnabled(bool enabled) override;
   void Stop() override;
 
+  void AddSink(MediaStreamAudioSink* sink) override;
+  void RemoveSink(MediaStreamAudioSink* sink) override;
+  media::AudioParameters GetOutputFormat() const override;
+
   webrtc::AudioTrackInterface* GetAudioAdapter() override;
 
  private:
+  class AudioSink;
+  scoped_ptr<AudioSink> sink_;
   const scoped_refptr<webrtc::AudioTrackInterface> track_;
 };