Storing raw audio sink for default audio track.

talk/media/webrtc/fakewebrtccall.cc

 /*
  * libjingle
  * Copyright 2015 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 75 matching lines @@
 
 void FakeAudioReceiveStream::IncrementReceivedPackets() {
   received_packets_++;
 }
 
 webrtc::AudioReceiveStream::Stats FakeAudioReceiveStream::GetStats() const {
   return stats_;
 }
 
 void FakeAudioReceiveStream::SetSink(
-    rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) {
-  sink_ = std::move(sink);
+    const rtc::scoped_refptr<webrtc::AudioSinkInterface>& sink) {
+  sink_ = sink;
 }
 
 FakeVideoSendStream::FakeVideoSendStream(
     const webrtc::VideoSendStream::Config& config,
     const webrtc::VideoEncoderConfig& encoder_config)
     : sending_(false),
       config_(config),
       codec_settings_set_(false),
       num_swapped_frames_(0) {
   RTC_DCHECK(config.encoder_settings.encoder != NULL);
@@ skipping 326 matching lines @@
 }
 
 void FakeCall::SignalNetworkState(webrtc::NetworkState state) {
   network_state_ = state;
 }
 
 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) {
   last_sent_packet_ = sent_packet;
 }
 }  // namespace cricket