Use the optimal buffer size for the local audio renderer.

content/renderer/media/webrtc_audio_renderer.cc

 // 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.
 
 #include "content/renderer/media/webrtc_audio_renderer.h"
 
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/strings/string_util.h"
 #include "base/strings/stringprintf.h"
@@ skipping 122 matching lines @@
       RenderFrameImpl::FromRoutingID(render_frame_id);
   MediaStreamDispatcher* const dispatcher = frame ?
       frame->GetMediaStreamDispatcher() : NULL;
   if (dispatcher && dispatcher->IsAudioDuckingActive()) {
     return media::AudioParameters::DUCKING;
   }
 
   return media::AudioParameters::NO_EFFECTS;
 }
 
-// Helper method to get platform specific optimal buffer size.
-int GetOptimalBufferSize(int sample_rate, int hardware_buffer_size) {
+}  // namespace
+
+int WebRtcAudioRenderer::GetOptimalBufferSize(int sample_rate,
+                                              int hardware_buffer_size) {
   // Use native hardware buffer size as default. On Windows, we strive to open
   // up using this native hardware buffer size to achieve best
   // possible performance and to ensure that no FIFO is needed on the browser
   // side to match the client request. That is why there is no #if case for
   // Windows below.
   int frames_per_buffer = hardware_buffer_size;
 
 #if defined(OS_LINUX) || defined(OS_MACOSX)
   // On Linux and MacOS, the low level IO implementations on the browser side
   // supports all buffer size the clients want. We use the native peer
@@ skipping 11 matching lines @@
     //  Galaxy Nexus: 144              => 2*441 = 882
     frames_per_buffer = 2 * frames_per_10ms;
     DVLOG(1) << "Low-latency output detected on Android";
   }
 #endif
 
   DVLOG(1) << "Using sink output buffer size: " << frames_per_buffer;
   return frames_per_buffer;
 }
 
-}  // namespace
-
 WebRtcAudioRenderer::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)
     : state_(UNINITIALIZED),
       source_render_view_id_(source_render_view_id),
       source_render_frame_id_(source_render_frame_id),
@@ skipping 364 matching lines @@
       if (RemovePlayingState(source, state))
         EnterPauseState();
     } else if (AddPlayingState(source, state)) {
       EnterPlayState();
     }
     UpdateSourceVolume(source);
   }
 }
 
 }  // namespace content