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 |