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: |