| Index: content/renderer/media/webrtc_local_audio_track.h
|
| diff --git a/content/renderer/media/webrtc_local_audio_track.h b/content/renderer/media/webrtc_local_audio_track.h
|
| index 0f4e712de257f00c80a7574062d7cef088bfdce4..a3b818e1a37db8a2f724c9c1dd1049be962d8cbb 100644
|
| --- a/content/renderer/media/webrtc_local_audio_track.h
|
| +++ b/content/renderer/media/webrtc_local_audio_track.h
|
| @@ -11,6 +11,7 @@
|
| #include "base/synchronization/lock.h"
|
| #include "base/threading/thread_checker.h"
|
| #include "content/renderer/media/webrtc_audio_device_impl.h"
|
| +#include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h"
|
| #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
|
| #include "third_party/libjingle/source/talk/app/webrtc/mediastreamtrack.h"
|
| #include "third_party/libjingle/source/talk/media/base/audiorenderer.h"
|
| @@ -37,7 +38,8 @@ class CONTENT_EXPORT WebRtcLocalAudioTrack
|
| static scoped_refptr<WebRtcLocalAudioTrack> Create(
|
| const std::string& id,
|
| const scoped_refptr<WebRtcAudioCapturer>& capturer,
|
| - webrtc::AudioSourceInterface* stream_source);
|
| + webrtc::AudioSourceInterface* stream_source,
|
| + const webrtc::MediaConstraintsInterface* constraints);
|
|
|
| // Add a sink to the track. This function will trigger a SetCaptureFormat()
|
| // call on the |sink|.
|
| @@ -72,7 +74,8 @@ class CONTENT_EXPORT WebRtcLocalAudioTrack
|
| protected:
|
| WebRtcLocalAudioTrack(const std::string& label,
|
| const scoped_refptr<WebRtcAudioCapturer>& capturer,
|
| - webrtc::AudioSourceInterface* track_source);
|
| + webrtc::AudioSourceInterface* track_source,
|
| + const webrtc::MediaConstraintsInterface* constraints);
|
| virtual ~WebRtcLocalAudioTrack();
|
|
|
| private:
|
|
|