| Index: content/renderer/media/webrtc_audio_capturer.cc
|
| diff --git a/content/renderer/media/webrtc_audio_capturer.cc b/content/renderer/media/webrtc_audio_capturer.cc
|
| index 994f77b150c6887b2ffb6718ec6efe7c9d134ee9..e770ec1b614ee94a8210408809b0be6ba103a379 100644
|
| --- a/content/renderer/media/webrtc_audio_capturer.cc
|
| +++ b/content/renderer/media/webrtc_audio_capturer.cc
|
| @@ -327,7 +327,7 @@
|
| // bits_per_sample is always 16 for now.
|
| int buffer_size = GetBufferSize(sample_rate);
|
| media::AudioParameters params(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
|
| - channel_layout, 0, sample_rate,
|
| + channel_layout, sample_rate,
|
| 16, buffer_size,
|
| device_info_.device.input.effects);
|
|
|
|
|