Use the optimal buffer size for the local audio renderer.

content/renderer/media/webrtc_audio_renderer.cc

Index: content/renderer/media/webrtc_audio_renderer.cc
diff --git a/content/renderer/media/webrtc_audio_renderer.cc b/content/renderer/media/webrtc_audio_renderer.cc
index 41e7cb9604bdcfe7fca6666e325d216caf69a417..34e0f48f61b93f59161ff9ee45017c1036f15a34 100644
--- a/content/renderer/media/webrtc_audio_renderer.cc
+++ b/content/renderer/media/webrtc_audio_renderer.cc
@@ -140,8 +140,10 @@ int GetCurrentDuckingFlag(int render_frame_id) {
   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
@@ -173,8 +175,6 @@ int GetOptimalBufferSize(int sample_rate, int hardware_buffer_size) {
   return frames_per_buffer;
 }
 
-}  // namespace
-
 WebRtcAudioRenderer::WebRtcAudioRenderer(
     const scoped_refptr<webrtc::MediaStreamInterface>& media_stream,
     int source_render_view_id,