Feed the render data to MediaStreamAudioProcessor and used AudioBus in render callback

content/renderer/media/webrtc_audio_renderer.h

Index: content/renderer/media/webrtc_audio_renderer.h
diff --git a/content/renderer/media/webrtc_audio_renderer.h b/content/renderer/media/webrtc_audio_renderer.h
index 983929a120059ff8b35267dbfd7c338e21e8f859..92c85cc55c2c3d2b5eef0848e5e3a71aea6edee5 100644
--- a/content/renderer/media/webrtc_audio_renderer.h
+++ b/content/renderer/media/webrtc_audio_renderer.h
@@ -96,6 +96,10 @@ class CONTENT_EXPORT WebRtcAudioRenderer
   // Used to DCHECK on the expected state.
   bool IsStarted() const;
 
+  // Accessors to the sink audio parameters.
+  int channels() const { return number_of_channels_; }
+  int sample_rate() const { return sample_rate_; }
+
  private:
   // MediaStreamAudioRenderer implementation.  This is private since we want
   // callers to use proxy objects.
@@ -190,10 +194,6 @@ class CONTENT_EXPORT WebRtcAudioRenderer
   // Audio data source from the browser process.
   WebRtcAudioRendererSource* source_;
 
-  // Buffers used for temporary storage during render callbacks.
-  // Allocated during initialization.
-  scoped_ptr<int16[]> buffer_;
-
   // Protects access to |state_|, |source_| and |sink_|.
   base::Lock lock_;
 
@@ -217,9 +217,12 @@ class CONTENT_EXPORT WebRtcAudioRenderer
   // Saved volume and playing state of the root renderer.
   PlayingState playing_state_;
 
-  // The preferred sample rate and buffer sizes provided via the ctor.
-  const int sample_rate_;
-  const int frames_per_buffer_;
+  // The sample rate, number of channels and buffer sizes used by the sink of
+  // the renderer.
+  // Only modified in the Initialize() on the main render thread.
+  int sample_rate_;
+  int number_of_channels_;
+  int frames_per_buffer_;
 
   // Maps audio sources to a list of active audio renderers.
   // Pointers to PlayingState objects are only kept in this map while the