| Index: content/renderer/media/media_stream_audio_processor.cc
|
| diff --git a/content/renderer/media/media_stream_audio_processor.cc b/content/renderer/media/media_stream_audio_processor.cc
|
| index 4c1a2c90ab299c27846cebd1c56e730cc9f0f22a..11e889259c0045346b9e5714ade5649ef5aa6fdf 100644
|
| --- a/content/renderer/media/media_stream_audio_processor.cc
|
| +++ b/content/renderer/media/media_stream_audio_processor.cc
|
| @@ -376,7 +376,8 @@ void MediaStreamAudioProcessor::InitializeCaptureConverter(
|
| const int sink_sample_rate = audio_processing_ ?
|
| kAudioProcessingSampleRate : source_params.sample_rate();
|
| const media::ChannelLayout sink_channel_layout = audio_processing_ ?
|
| - media::CHANNEL_LAYOUT_MONO : source_params.channel_layout();
|
| + media::GuessChannelLayout(kAudioProcessingNumberOfChannel) :
|
| + source_params.channel_layout();
|
|
|
| // WebRtc AudioProcessing requires 10ms as its packet size. We use this
|
| // native size when processing is enabled. While processing is disabled, and
|
|
|