Storing raw audio sink for default audio track.

talk/app/webrtc/webrtcsession.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 1302 matching lines @@
   if (!voice_channel_) {
     LOG(LS_ERROR) << "SetAudioPlayoutVolume: No audio channel exists.";
     return;
   }
 
   if (!voice_channel_->SetOutputVolume(ssrc, volume)) {
     ASSERT(false);
   }
 }
 
-void WebRtcSession::SetRawAudioSink(uint32_t ssrc,
-                                    rtc::scoped_ptr<AudioSinkInterface> sink) {
+void WebRtcSession::SetRawAudioSink(
+    uint32_t ssrc,
+    rtc::scoped_refptr<AudioSinkInterface> sink) {
   ASSERT(signaling_thread()->IsCurrent());
   if (!voice_channel_)
     return;
 
-  voice_channel_->SetRawAudioSink(ssrc, std::move(sink));
+  voice_channel_->SetRawAudioSink(ssrc, sink);
 }
 
 bool WebRtcSession::SetCaptureDevice(uint32_t ssrc,
                                      cricket::VideoCapturer* camera) {
   ASSERT(signaling_thread()->IsCurrent());
 
   if (!video_channel_) {
     // |video_channel_| doesnt't exist. Probably because the remote end doesnt't
     // support video.
     LOG(LS_WARNING) << "Video not used in this call.";
@@ skipping 856 matching lines @@
   }
 }
 
 void WebRtcSession::OnSentPacket_w(cricket::TransportChannel* channel,
                                    const rtc::SentPacket& sent_packet) {
   RTC_DCHECK(worker_thread()->IsCurrent());
   media_controller_->call_w()->OnSentPacket(sent_packet);
 }
 
 }  // namespace webrtc