| Index: webrtc/api/mediastreamprovider.h
|
| diff --git a/webrtc/api/mediastreamprovider.h b/webrtc/api/mediastreamprovider.h
|
| index f3bb3f49f39ef4ecf5b90ef1561980d992a5406c..eef92846cbd15c47ce3014f9da8c518c6130cf71 100644
|
| --- a/webrtc/api/mediastreamprovider.h
|
| +++ b/webrtc/api/mediastreamprovider.h
|
| @@ -11,6 +11,8 @@
|
| #ifndef WEBRTC_API_MEDIASTREAMPROVIDER_H_
|
| #define WEBRTC_API_MEDIASTREAMPROVIDER_H_
|
|
|
| +#include <memory>
|
| +
|
| #include "webrtc/api/rtpsenderinterface.h"
|
| #include "webrtc/base/basictypes.h"
|
| #include "webrtc/base/scoped_ptr.h"
|
| @@ -60,7 +62,7 @@ class AudioProviderInterface {
|
| // passed to the provider.
|
| virtual void SetRawAudioSink(
|
| uint32_t ssrc,
|
| - rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) = 0;
|
| + std::unique_ptr<webrtc::AudioSinkInterface> sink) = 0;
|
|
|
| virtual RtpParameters GetAudioRtpParameters(uint32_t ssrc) const = 0;
|
| virtual bool SetAudioRtpParameters(uint32_t ssrc,
|
|
|