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 44894b52e8c67f1c9eaa28ed582fffb9fee00b34..b227de9a6bf5164c65842da2004af3d8b115a115 100644
--- a/content/renderer/media/webrtc_audio_renderer.h
+++ b/content/renderer/media/webrtc_audio_renderer.h
@@ -53,6 +53,10 @@ 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.
@@ -102,10 +106,6 @@ 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_;
 
@@ -126,9 +126,11 @@ class CONTENT_EXPORT WebRtcAudioRenderer
   // Delay due to the FIFO in milliseconds.
   int fifo_delay_milliseconds_;
 
-  // 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.
+  int sample_rate_;
+  int number_of_channels_;
+  int frames_per_buffer_;
 
   DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioRenderer);
 };