Storing raw audio sink for default audio track.

talk/app/webrtc/rtpsenderreceiver_unittest.cc

Index: talk/app/webrtc/rtpsenderreceiver_unittest.cc
diff --git a/talk/app/webrtc/rtpsenderreceiver_unittest.cc b/talk/app/webrtc/rtpsenderreceiver_unittest.cc
index a590e1d01f8b07df94c5b9304b04fa4a85e1b158..375c41409b935dd8418cfe4d84ff0cd933e54a47 100644
--- a/talk/app/webrtc/rtpsenderreceiver_unittest.cc
+++ b/talk/app/webrtc/rtpsenderreceiver_unittest.cc
@@ -71,12 +71,12 @@ class MockAudioProvider : public AudioProviderInterface {
   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.