Add generic interfaces for the sinks of the media stream audio track

content/renderer/media/webrtc_local_audio_renderer.h

Index: content/renderer/media/webrtc_local_audio_renderer.h
diff --git a/content/renderer/media/webrtc_local_audio_renderer.h b/content/renderer/media/webrtc_local_audio_renderer.h
index 8e8c96dad8a08add3a35009fb690cbdadf39ade5..e9871ece81b5c21a4439b5ae3a0b9540d67a1247 100644
--- a/content/renderer/media/webrtc_local_audio_renderer.h
+++ b/content/renderer/media/webrtc_local_audio_renderer.h
@@ -13,9 +13,11 @@
 #include "base/synchronization/lock.h"
 #include "base/threading/thread_checker.h"
 #include "content/common/content_export.h"
+#include "content/public/renderer/media_stream_audio_sink.h"
 #include "content/renderer/media/media_stream_audio_renderer.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/media/webrtc_local_audio_track.h"
+#include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
 
 namespace media {
 class AudioBus;
@@ -34,7 +36,7 @@ class WebRtcAudioCapturer;
 // It also implements media::AudioRendererSink::RenderCallback to render audio
 // data provided from a WebRtcLocalAudioTrack source.
 // When the audio layer in the browser process asks for data to render, this
-// class provides the data by implementing the WebRtcAudioCapturerSink
+// class provides the data by implementing the MediaStreamAudioSink
 // interface, i.e., we are a sink seen from the WebRtcAudioCapturer perspective.
 // TODO(henrika): improve by using similar principles as in RTCVideoRenderer
 // which register itself to the video track when the provider is started and
@@ -42,13 +44,13 @@ class WebRtcAudioCapturer;
 // Tracking this at http://crbug.com/164813.
 class CONTENT_EXPORT WebRtcLocalAudioRenderer
     : NON_EXPORTED_BASE(public MediaStreamAudioRenderer),
-      NON_EXPORTED_BASE(public media::AudioRendererSink::RenderCallback),
-      NON_EXPORTED_BASE(public WebRtcAudioCapturerSink) {
+      NON_EXPORTED_BASE(public MediaStreamAudioSink),
+      NON_EXPORTED_BASE(public media::AudioRendererSink::RenderCallback) {
  public:
   // Creates a local renderer and registers a capturing |source| object.
   // The |source| is owned by the WebRtcAudioDeviceImpl.
   // Called on the main thread.
-  WebRtcLocalAudioRenderer(WebRtcLocalAudioTrack* audio_track,
+  WebRtcLocalAudioRenderer(const blink::WebMediaStreamTrack& audio_track,
                            int source_render_view_id,
                            int session_id,
                            int frames_per_buffer);
@@ -71,21 +73,16 @@ class CONTENT_EXPORT WebRtcLocalAudioRenderer
   virtual ~WebRtcLocalAudioRenderer();
 
  private:
-  // WebRtcAudioCapturerSink implementation.
+  // MediaStreamAudioSink implementation.
 
   // Called on the AudioInputDevice worker thread.
-  virtual int CaptureData(const std::vector<int>& channels,
-                          const int16* audio_data,
-                          int sample_rate,
-                          int number_of_channels,
-                          int number_of_frames,
-                          int audio_delay_milliseconds,
-                          int current_volume,
-                          bool need_audio_processing,
-                          bool key_pressed) OVERRIDE;
-
-  // Can be called on different user thread.
-  virtual void SetCaptureFormat(const media::AudioParameters& params) OVERRIDE;
+  virtual void OnData(const int16* audio_data,
+                      int sample_rate,
+                      int number_of_channels,
+                      int number_of_frames) OVERRIDE;
+
+  // Called on the AudioInputDevice worker thread.
+  virtual void OnSetFormat(const media::AudioParameters& params) OVERRIDE;
 
   // media::AudioRendererSink::RenderCallback implementation.
   // Render() is called on the AudioOutputDevice thread and OnRenderError()
@@ -108,8 +105,10 @@ class CONTENT_EXPORT WebRtcLocalAudioRenderer
   // instance like a selected microphone and forwards the recorded data to its
   // sinks. The recorded data is stored in a FIFO and consumed
   // by this class when the sink asks for new data.
-  // The WebRtcAudioCapturer is today created by WebRtcAudioDeviceImpl.
-  scoped_refptr<WebRtcLocalAudioTrack> audio_track_;
+  // This class is calling MediaStreamAudioSink::AddToAudioTrack() and
+  // MediaStreamAudioSink::RemoveFromAudioTrack() to connect and disconnect
+  // with the audio track.
+  blink::WebMediaStreamTrack audio_track_;
 
   // The render view in which the audio is rendered into |sink_|.
   const int source_render_view_id_;