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; |