Index: content/renderer/media/webrtc_audio_capturer.cc |
=================================================================== |
--- content/renderer/media/webrtc_audio_capturer.cc (revision 181622) |
+++ content/renderer/media/webrtc_audio_capturer.cc (working copy) |
@@ -52,6 +52,9 @@ |
// TODO(henrika): It might be possible to reduce the input buffer |
// size and reduce the delay even more. |
buffer_size = 2 * sample_rate / 100; |
+#elif defined(OS_ANDROID) |
+ // TODO(leozwang): Tune and adjust buffer size on Android. |
+ buffer_size = 2 * sample_rate / 100; |
no longer working on chromium
2013/02/11 18:20:36
Have you tested that 20ms buffer size is a good st
leozwang1
2013/02/11 21:00:48
In my local loopback test with video, I can still
no longer working on chromium
2013/02/12 20:04:46
Then why don't we increase it to a reasonable valu
leozwang1
2013/02/12 20:14:17
Most of tests I have done are video+audio. The nex
|
#endif |
return buffer_size; |