| 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 cf65d2f487af5ddd89679f5cf8de23c6ded96a92..aae04ada6c8d7b00de747f971c65205ed04558e0 100644
|
| --- a/content/renderer/media/webaudio_capturer_source.cc
|
| +++ b/content/renderer/media/webaudio_capturer_source.cc
|
| @@ -49,9 +49,8 @@ void WebAudioCapturerSource::setFormat(
|
| // Set the format used by this WebAudioCapturerSource. We are using 10ms data
|
| // 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, sample_rate, 16,
|
| - sample_rate / 100);
|
| + params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout,
|
| + sample_rate, 16, sample_rate / 100);
|
| audio_format_changed_ = true;
|
|
|
| wrapper_bus_ = AudioBus::CreateWrapper(params_.channels());
|
|
|