Replace scoped_ptr with unique_ptr in webrtc/api/

webrtc/api/mediastreamprovider.h

 /*
  *  Copyright 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #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"
 #include "webrtc/media/base/videosinkinterface.h"
 #include "webrtc/media/base/videosourceinterface.h"
 
 namespace cricket {
 
 class AudioSource;
 class VideoFrame;
@@ skipping 29 matching lines @@
 
   // Sets the audio playout volume of a remote audio track with |ssrc|.
   // |volume| is in the range of [0, 10].
   virtual void SetAudioPlayoutVolume(uint32_t ssrc, double volume) = 0;
 
   // Allows for setting a direct audio sink for an incoming audio source.
   // Only one audio sink is supported per ssrc and ownership of the sink is
   // 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,
                                      const RtpParameters& parameters) = 0;
 
  protected:
   virtual ~AudioProviderInterface() {}
 };
 
 // This interface is called by VideoRtpSender/Receivers to change the settings
@@ skipping 17 matching lines @@
   virtual bool SetVideoRtpParameters(uint32_t ssrc,
                                      const RtpParameters& parameters) = 0;
 
  protected:
   virtual ~VideoProviderInterface() {}
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_MEDIASTREAMPROVIDER_H_