Use the optimal buffer size for the local audio renderer.

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 51 matching lines @@
     void set_volume(float volume) {
       DCHECK(CalledOnValidThread());
       volume_ = volume;
     }
 
    private:
     bool playing_;
     float volume_;
   };
 
+
+  // Returns platform specific optimal buffer size for rendering audio.
+  static int GetOptimalBufferSize(int sample_rate, int hardware_buffer_size);
+
   WebRtcAudioRenderer(
       const scoped_refptr<webrtc::MediaStreamInterface>& media_stream,
       int source_render_view_id,
       int source_render_frame_id,
       int session_id,
       int sample_rate,
       int frames_per_buffer);
 
   // Initialize function called by clients like WebRtcAudioDeviceImpl.
   // Stop() has to be called before |source| is deleted.
@@ skipping 150 matching lines @@
   // Used for triggering new UMA histogram. Counts number of render
   // callbacks modulo |kNumCallbacksBetweenRenderTimeHistograms|.
   int render_callback_count_;
 
   DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioRenderer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_