| 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 df46265d8fb519f93c5ca69bd1de88851467fbe2..adf49026b540c86e95eacfb7246262443e1d5aa6 100644 | 
| --- a/content/renderer/media/webrtc_audio_capturer.cc | 
| +++ b/content/renderer/media/webrtc_audio_capturer.cc | 
| @@ -244,7 +244,7 @@ WebRtcAudioCapturer::WebRtcAudioCapturer( | 
| : constraints_(constraints), | 
| audio_processor_(new rtc::RefCountedObject<MediaStreamAudioProcessor>( | 
| constraints, | 
| -          device_info.device.input.effects, | 
| +          device_info.device.input, | 
| audio_device)), | 
| running_(false), | 
| render_frame_id_(render_frame_id), | 
| @@ -343,11 +343,8 @@ void WebRtcAudioCapturer::SetCapturerSourceInternal( | 
| // which would normally be used by default. | 
| // bits_per_sample is always 16 for now. | 
| media::AudioParameters params(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, | 
| -                                channel_layout, | 
| -                                sample_rate, | 
| -                                16, | 
| -                                buffer_size, | 
| -                                device_info_.device.input.effects); | 
| +                                channel_layout, sample_rate, 16, buffer_size, | 
| +                                "", device_info_.device.input.effects); | 
|  | 
| { | 
| base::AutoLock auto_lock(lock_); | 
|  |