Storing raw audio sink for default audio track.

talk/app/webrtc/rtpsenderreceiver_unittest.cc

 /*
  * libjingle
  * Copyright 2012 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 53 matching lines @@
                void(uint32_t ssrc,
                     bool enable));
   MOCK_METHOD4(SetAudioSend,
                void(uint32_t ssrc,
                     bool enable,
                     const cricket::AudioOptions& options,
                     cricket::AudioRenderer* renderer));
   MOCK_METHOD2(SetAudioPlayoutVolume, void(uint32_t ssrc, double volume));
 
   void SetRawAudioSink(uint32_t,
-                       rtc::scoped_ptr<AudioSinkInterface> sink) override {
-    sink_ = std::move(sink);
+                       rtc::scoped_refptr<AudioSinkInterface> sink) override {
+    sink_ = sink;
   }
 
  private:
-  rtc::scoped_ptr<AudioSinkInterface> sink_;
+  rtc::scoped_refptr<AudioSinkInterface> sink_;
 };
 
 // Helper class to test RtpSender/RtpReceiver.
 class MockVideoProvider : public VideoProviderInterface {
  public:
   virtual ~MockVideoProvider() {}
   MOCK_METHOD2(SetCaptureDevice,
                bool(uint32_t ssrc, cricket::VideoCapturer* camera));
   MOCK_METHOD3(SetVideoPlayout,
                void(uint32_t ssrc,
@@ skipping 415 matching lines @@
                                video_track_->GetSource()->GetVideoCapturer()));
   EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc2, true, _));
   sender->SetSsrc(kVideoSsrc2);
 
   // Calls expected from destructor.
   EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc2, nullptr)).Times(1);
   EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc2, false, _)).Times(1);
 }
 
 }  // namespace webrtc