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, |