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

content/renderer/media/webrtc_audio_renderer.h

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_
 
 #include "base/memory/ref_counted.h"
 #include "base/synchronization/lock.h"
 #include "base/threading/non_thread_safe.h"
@@ skipping 78 matching lines @@
   // etc and similarly maintains the same state for Stop().
   // When Stop() is called or when the proxy goes out of scope, the proxy
   // will ensure that Pause() is called followed by a call to Stop(), which
   // is the usage pattern that WebRtcAudioRenderer requires.
   scoped_refptr<MediaStreamAudioRenderer> CreateSharedAudioRendererProxy(
       const scoped_refptr<webrtc::MediaStreamInterface>& media_stream);
 
   // Used to DCHECK on the expected state.
   bool IsStarted() const;
 
+  // Accessors to the sink audio parameters.
+  int channels() const { return sink_params_.channels(); }
+  int sample_rate() const { return sink_params_.sample_rate(); }
+
  private:
   // MediaStreamAudioRenderer implementation.  This is private since we want
   // callers to use proxy objects.
   // TODO(tommi): Make the MediaStreamAudioRenderer implementation a pimpl?
   virtual void Start() OVERRIDE;
   virtual void Play() OVERRIDE;
   virtual void Pause() OVERRIDE;
   virtual void Stop() OVERRIDE;
   virtual void SetVolume(float volume) OVERRIDE;
   virtual base::TimeDelta GetCurrentRenderTime() const OVERRIDE;
@@ skipping 74 matching lines @@
 
   // The sink (destination) for rendered audio.
   scoped_refptr<media::AudioOutputDevice> sink_;
 
   // The media stream that holds the audio tracks that this renderer renders.
   const scoped_refptr<webrtc::MediaStreamInterface> media_stream_;
 
   // 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_;
 
   // Ref count for the MediaPlayers which are playing audio.
   int play_ref_count_;
 
   // Ref count for the MediaPlayers which have called Start() but not Stop().
   int start_ref_count_;
 
   // Used to buffer data between the client and the output device in cases where
   // the client buffer size is not the same as the output device buffer size.
   scoped_ptr<media::AudioPullFifo> audio_fifo_;
 
   // Contains the accumulated delay estimate which is provided to the WebRTC
   // AEC.
   int audio_delay_milliseconds_;
 
   // Delay due to the FIFO in milliseconds.
   int fifo_delay_milliseconds_;
 
   // 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_;
+  // Audio params used by the sink of the renderer.
+  media::AudioParameters sink_params_;
 
   // Maps audio sources to a list of active audio renderers.
   // Pointers to PlayingState objects are only kept in this map while the
   // associated renderer is actually playing the stream.  Ownership of the
   // state objects lies with the renderers and they must leave the playing state
   // before being destructed (PlayingState object goes out of scope).
   SourcePlayingStates source_playing_states_;
 
   DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioRenderer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_