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()); |