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/thread_checker.h"
@@ skipping 35 matching lines @@
   // is "playing", Pause() won't be called when the state already is "paused"
   // 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();
 
   // 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.
   // 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 29 matching lines @@
   const int source_render_view_id_;
   const int source_render_frame_id_;
   const int session_id_;
 
   // The sink (destination) for rendered audio.
   scoped_refptr<media::AudioOutputDevice> sink_;
 
   // 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_;
 
-  // 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.

[tommi (sloooow) - chröme, 2014/01/31 13:58:32]

now that these are no longer const, can you document on which thread they are
allowed to be touched (and any locking that they might need).

[no longer working on chromium, 2014/02/02 16:50:16]

Added a comment to explain they are modified only in the Initialize() on the
main render thread.

+  int sample_rate_;
+  int number_of_channels_;
+  int frames_per_buffer_;
 
   DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioRenderer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_