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

content/renderer/media/webrtc_audio_device_impl.h

Index: content/renderer/media/webrtc_audio_device_impl.h
diff --git a/content/renderer/media/webrtc_audio_device_impl.h b/content/renderer/media/webrtc_audio_device_impl.h
index f9279f5f6ff5c4986b1afd62e5223228426ae690..4543a8a788319d5b04278a9eda4a6e679241b55f 100644
--- a/content/renderer/media/webrtc_audio_device_impl.h
+++ b/content/renderer/media/webrtc_audio_device_impl.h
@@ -186,15 +186,10 @@ class WebRtcAudioRenderer;
 class WebRtcAudioRendererSource {
  public:
   // Callback to get the rendered interleaved data.
-  // TODO(xians): Change uint8* to int16*.
-  virtual void RenderData(uint8* audio_data,
-                          int number_of_channels,
-                          int number_of_frames,
+  virtual void RenderData(media::AudioBus* audio_bus,
+                          int sample_rate,
                           int audio_delay_milliseconds) = 0;
 
-  // Set the format for the capture audio parameters.
-  virtual void SetRenderFormat(const media::AudioParameters& params) = 0;
-
   // Callback to notify the client that the renderer is going away.
   virtual void RemoveAudioRenderer(WebRtcAudioRenderer* renderer) = 0;
 
@@ -305,6 +300,12 @@ class CONTENT_EXPORT WebRtcAudioDeviceImpl
   void AddAudioCapturer(const scoped_refptr<WebRtcAudioCapturer>& capturer);
   void RemoveAudioCapturer(const scoped_refptr<WebRtcAudioCapturer>& capturer);
 
+  // Adds/Removes the observer of WebRtcAudioRendererSource to the ADM.
+  // These methods are used by the MediaStreamAudioProcesssor to get the
+  // render data for AEC.
+  void AddRenderDataObserver(WebRtcAudioRendererSource* observer);
+  void RemoveRenderDataObserver(WebRtcAudioRendererSource* observer);
+
   // Gets paired device information of the capture device for the audio
   // renderer. This is used to pass on a session id, sample rate and buffer
   // size to a webrtc audio renderer (either local or remote), so that audio
@@ -318,18 +319,11 @@ class CONTENT_EXPORT WebRtcAudioDeviceImpl
   const scoped_refptr<WebRtcAudioRenderer>& renderer() const {
     return renderer_;
   }
-  int output_buffer_size() const {
-    return output_audio_parameters_.frames_per_buffer();
-  }
-  int output_channels() const {
-    return output_audio_parameters_.channels();
-  }
-  int output_sample_rate() const {
-    return output_audio_parameters_.sample_rate();
-  }
 
  private:
   typedef std::list<scoped_refptr<WebRtcAudioCapturer> > CapturerList;
+  typedef std::list<WebRtcAudioRendererSource* > RenderDataObservers;
+  class RenderBuffer;
 
   // Make destructor private to ensure that we can only be deleted by Release().
   virtual ~WebRtcAudioDeviceImpl();
@@ -353,13 +347,11 @@ class CONTENT_EXPORT WebRtcAudioDeviceImpl
   // WebRtcAudioRendererSource implementation.
 
   // Called on the AudioInputDevice worker thread.
-  virtual void RenderData(uint8* audio_data,
-                          int number_of_channels,
-                          int number_of_frames,
+  virtual void RenderData(media::AudioBus* audio_bus,
+                          int sample_rate,
                           int audio_delay_milliseconds) OVERRIDE;
 
   // Called on the main render thread.
-  virtual void SetRenderFormat(const media::AudioParameters& params) OVERRIDE;
   virtual void RemoveAudioRenderer(WebRtcAudioRenderer* renderer) OVERRIDE;
 
   // Helper to get the default capturer, which is the last capturer in
@@ -378,14 +370,14 @@ class CONTENT_EXPORT WebRtcAudioDeviceImpl
   // Provides access to the audio renderer in the browser process.
   scoped_refptr<WebRtcAudioRenderer> renderer_;
 
+  // List of observers which requires access to the render data.
+  RenderDataObservers render_data_observers_;
+
   // Weak reference to the audio callback.
   // The webrtc client defines |audio_transport_callback_| by calling
   // RegisterAudioCallback().
   webrtc::AudioTransport* audio_transport_callback_;
 
-  // Cached values of used output audio parameters. Platform dependent.
-  media::AudioParameters output_audio_parameters_;
-
   // Cached value of the current audio delay on the input/capture side.
   int input_delay_ms_;
 
@@ -412,6 +404,11 @@ class CONTENT_EXPORT WebRtcAudioDeviceImpl
   // Range is [0, 255].
   uint32_t microphone_volume_;
 
+  // Buffer used for temporary storage during render callback.
+  // It is only accessed by the audio render thread.
+  scoped_ptr<int16[]> render_buffer_;
+  int render_buffer_size_;
+
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl);
 };