| 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 f396813387a821176df21a827af07ed2e550ab58..6cf1195a46a8a8d3f24e52e4ca3168b287f4ee40 100644
|
| --- a/content/renderer/media/webrtc_audio_capturer.cc
|
| +++ b/content/renderer/media/webrtc_audio_capturer.cc
|
| @@ -304,7 +304,8 @@ void WebRtcAudioCapturer::SetCapturerSource(
|
| 16, buffer_size, effects);
|
| scoped_refptr<MediaStreamAudioProcessor> new_audio_processor(
|
| new talk_base::RefCountedObject<MediaStreamAudioProcessor>(
|
| - params, constraints, effects, audio_device_));
|
| + params, constraints, effects, device_info_.device.type,
|
| + audio_device_));
|
| {
|
| base::AutoLock auto_lock(lock_);
|
| audio_processor_ = new_audio_processor;
|
|
|