Used 10ms native buffer size for webrtc audio renderer on Linux and Mac

content/renderer/media/webrtc_audio_renderer.h

Index: content/renderer/media/webrtc_audio_renderer.h
diff --git a/content/renderer/media/webrtc_audio_renderer.h b/content/renderer/media/webrtc_audio_renderer.h
index 61b0b24d141ba3c2ff2b6414df226ae8578614e7..47b0e8a713624b38b2eb487c89fa68edcdbc51e1 100644
--- a/content/renderer/media/webrtc_audio_renderer.h
+++ b/content/renderer/media/webrtc_audio_renderer.h
@@ -99,6 +99,7 @@ class CONTENT_EXPORT WebRtcAudioRenderer
   // Accessors to the sink audio parameters.
   int channels() const { return sink_params_.channels(); }
   int sample_rate() const { return sink_params_.sample_rate(); }
+  int buffer_size() const { return sink_params_.frames_per_buffer(); }

[tommi (sloooow) - chröme, 2014/09/03 13:22:11]

rename to frames_per_buffer for consistency.  buffer_size() isn't really correct
since this doesn't reflect number of bytes.

[no longer working on chromium, 2014/09/03 14:34:50]

Done.

 
  private:
   // MediaStreamAudioRenderer implementation.  This is private since we want