Storing raw audio sink for default audio track.

talk/app/webrtc/remoteaudiosource.cc

 /*
  * libjingle
  * Copyright 2014 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 78 matching lines @@
   RTC_DCHECK(audio_observers_.empty());
   RTC_DCHECK(sinks_.empty());
 }
 
 void RemoteAudioSource::Initialize(uint32_t ssrc,
                                    AudioProviderInterface* provider) {
   RTC_DCHECK(main_thread_->IsCurrent());
   // To make sure we always get notified when the provider goes out of scope,
   // we register for callbacks here and not on demand in AddSink.
   if (provider) {  // May be null in tests.
-    provider->SetRawAudioSink(
-        ssrc, rtc::scoped_ptr<AudioSinkInterface>(new Sink(this)));
+    provider->SetRawAudioSink(ssrc, new rtc::RefCountedObject<Sink>(this));
   }
 }
 
 MediaSourceInterface::SourceState RemoteAudioSource::state() const {
   RTC_DCHECK(main_thread_->IsCurrent());
   return state_;
 }
 
 bool RemoteAudioSource::remote() const {
   RTC_DCHECK(main_thread_->IsCurrent());
@@ skipping 56 matching lines @@
 }
 
 void RemoteAudioSource::OnMessage(rtc::Message* msg) {
   RTC_DCHECK(main_thread_->IsCurrent());
   sinks_.clear();
   state_ = MediaSourceInterface::kEnded;
   FireOnChanged();
 }
 
 }  // namespace webrtc