| Index: content/renderer/media/webaudio_capturer_source.cc
|
| diff --git a/content/renderer/media/webaudio_capturer_source.cc b/content/renderer/media/webaudio_capturer_source.cc
|
| index 7076c24f329ce8ff592d270f44a49fbca5144212..2095ac68f8cd1d1658ce3568c0362eb22554ee4f 100644
|
| --- a/content/renderer/media/webaudio_capturer_source.cc
|
| +++ b/content/renderer/media/webaudio_capturer_source.cc
|
| @@ -48,7 +48,7 @@
|
| // as buffer size since that is the native buffer size of WebRtc packet
|
| // running on.
|
| params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
|
| - channel_layout, number_of_channels, 0, sample_rate, 16,
|
| + channel_layout, number_of_channels, sample_rate, 16,
|
| sample_rate / 100);
|
| audio_format_changed_ = true;
|
|
|
|
|