Replace scoped_ptr with unique_ptr in webrtc/api/

webrtc/api/rtpsenderreceiver_unittest.cc

 /*
  *  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.
  */
 
+#include <memory>
 #include <string>
 #include <utility>
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/api/audiotrack.h"
 #include "webrtc/api/mediastream.h"
 #include "webrtc/api/remoteaudiosource.h"
 #include "webrtc/api/rtpreceiver.h"
 #include "webrtc/api/rtpsender.h"
@@ skipping 30 matching lines @@
                void(uint32_t ssrc,
                     bool enable,
                     const cricket::AudioOptions& options,
                     cricket::AudioSource* source));
   MOCK_METHOD2(SetAudioPlayoutVolume, void(uint32_t ssrc, double volume));
   MOCK_CONST_METHOD1(GetAudioRtpParameters, RtpParameters(uint32_t ssrc));
   MOCK_METHOD2(SetAudioRtpParameters,
                bool(uint32_t ssrc, const RtpParameters&));
 
   void SetRawAudioSink(uint32_t,
-                       rtc::scoped_ptr<AudioSinkInterface> sink) override {
+                       std::unique_ptr<AudioSinkInterface> sink) override {
     sink_ = std::move(sink);
   }
 
  private:
-  rtc::scoped_ptr<AudioSinkInterface> sink_;
+  std::unique_ptr<AudioSinkInterface> sink_;
 };
 
 // Helper class to test RtpSender/RtpReceiver.
 class MockVideoProvider : public VideoProviderInterface {
  public:
   virtual ~MockVideoProvider() {}
   MOCK_METHOD2(SetSource,
                bool(uint32_t ssrc,
                     rtc::VideoSourceInterface<cricket::VideoFrame>* source));
   MOCK_METHOD3(SetVideoPlayout,
@@ skipping 420 matching lines @@
       .WillOnce(Return(RtpParameters()));
   EXPECT_CALL(video_provider_, SetVideoRtpParameters(kVideoSsrc, _))
       .WillOnce(Return(true));
   RtpParameters params = video_rtp_sender_->GetParameters();
   EXPECT_TRUE(video_rtp_sender_->SetParameters(params));
 
   DestroyVideoRtpSender();
 }
 
 }  // namespace webrtc