Use the optimal buffer size for the local audio renderer.

content/renderer/media/webrtc_local_audio_renderer.cc

Index: content/renderer/media/webrtc_local_audio_renderer.cc
diff --git a/content/renderer/media/webrtc_local_audio_renderer.cc b/content/renderer/media/webrtc_local_audio_renderer.cc
index 61aca702ca4abc64a7d95d22f558b3e593a19c43..dd657eb7c3e89eb67682ee93151fae178c11cd0a 100644
--- a/content/renderer/media/webrtc_local_audio_renderer.cc
+++ b/content/renderer/media/webrtc_local_audio_renderer.cc
@@ -12,6 +12,7 @@
 #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"
@@ -284,15 +285,8 @@ void WebRtcLocalAudioRenderer::ReconfigureSink(
   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);