Use the optimal buffer size for the local audio renderer.

content/renderer/media/webrtc_local_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_local_audio_renderer.h"
 
 #include "base/debug/trace_event.h"
 #include "base/logging.h"
 #include "base/message_loop/message_loop_proxy.h"
 #include "base/metrics/histogram.h"
 #include "base/synchronization/lock.h"
 #include "content/renderer/media/audio_device_factory.h"
 #include "content/renderer/media/media_stream_dispatcher.h"
 #include "content/renderer/media/webrtc_audio_capturer.h"
+#include "content/renderer/media/webrtc_audio_renderer.h"
 #include "content/renderer/render_frame_impl.h"
 #include "media/audio/audio_output_device.h"
 #include "media/base/audio_block_fifo.h"
 #include "media/base/audio_bus.h"
 
 namespace content {
 
 namespace {
 
 enum LocalRendererSinkStates {
@@ skipping 252 matching lines @@
     return;
 
   // Reset the |source_params_|, |sink_params_| and |loopback_fifo_| to match
   // the new format.
 
   source_params_ = params;
 
   sink_params_ = media::AudioParameters(source_params_.format(),
       source_params_.channel_layout(), source_params_.sample_rate(),
       source_params_.bits_per_sample(),
-#if defined(OS_ANDROID)
-      // On Android, input and output use the same sample rate. In order to
-      // use the low latency mode, we need to use the buffer size suggested by
-      // the AudioManager for the sink.  It will later be used to decide
-      // the buffer size of the shared memory buffer.
-      frames_per_buffer_,
-#else
-      2 * source_params_.frames_per_buffer(),
-#endif
+      WebRtcAudioRenderer::GetOptimalBufferSize(source_params_.sample_rate(),
+                                                frames_per_buffer_),
       // If DUCKING is enabled on the source, it needs to be enabled on the
       // sink as well.
       source_params_.effects() | implicit_ducking_effect);
 
   {
     // TODO(henrika): we could add a more dynamic solution here but I prefer
     // a fixed size combined with bad audio at overflow. The alternative is
     // that we start to build up latency and that can be more difficult to
     // detect. Tests have shown that the FIFO never contains more than 2 or 3
     // audio frames but I have selected a max size of ten buffers just
@@ skipping 18 matching lines @@
     sink_->Stop();
     sink_started_ = false;
   }
 
   sink_ = AudioDeviceFactory::NewOutputDevice(source_render_view_id_,
                                               source_render_frame_id_);
   MaybeStartSink();
 }
 
 }  // namespace content