| Index: content/renderer/media/webrtc_audio_capturer.cc
|
| ===================================================================
|
| --- content/renderer/media/webrtc_audio_capturer.cc (revision 181072)
|
| +++ content/renderer/media/webrtc_audio_capturer.cc (working copy)
|
| @@ -101,7 +101,7 @@
|
| int buffer_size = GetBufferSizeForSampleRate(sample_rate);
|
|
|
| // Configure audio parameters for the default source.
|
| - params_.Reset(format, channel_layout, sample_rate, 16, buffer_size);
|
| + params_.Reset(format, channel_layout, 0, sample_rate, 16, buffer_size);
|
|
|
| // Tell all sinks which format we use.
|
| for (SinkList::const_iterator it = sinks_.begin();
|
| @@ -194,6 +194,7 @@
|
|
|
| params_.Reset(params_.format(),
|
| channel_layout,
|
| + 0,
|
| sample_rate,
|
| 16, // ignored since the audio stack uses float32.
|
| buffer_size);
|
|
|