Used 10ms native buffer size for webrtc audio renderer on Linux and Mac

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 81 matching lines @@
   // 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(); }
+  int frames_per_buffer() const { return sink_params_.frames_per_buffer(); }
 
  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;
@@ skipping 116 matching lines @@
   // 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_