Revert of Storing raw audio sink for default audio track.

talk/media/webrtc/fakewebrtccall.h

 /*
  * 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 71 matching lines @@
 
 class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream {
  public:
   explicit FakeAudioReceiveStream(
       const webrtc::AudioReceiveStream::Config& config);
 
   const webrtc::AudioReceiveStream::Config& GetConfig() const;
   void SetStats(const webrtc::AudioReceiveStream::Stats& stats);
   int received_packets() const { return received_packets_; }
   void IncrementReceivedPackets();
-  const rtc::scoped_refptr<webrtc::AudioSinkInterface>& sink() const {
-    return sink_;
-  }
 
  private:
   // webrtc::ReceiveStream implementation.
   void Start() override {}
   void Stop() override {}
   void SignalNetworkState(webrtc::NetworkState state) override {}
   bool DeliverRtcp(const uint8_t* packet, size_t length) override {
     return true;
   }
   bool DeliverRtp(const uint8_t* packet,
                   size_t length,
                   const webrtc::PacketTime& packet_time) override {
     return true;
   }
 
   // webrtc::AudioReceiveStream implementation.
   webrtc::AudioReceiveStream::Stats GetStats() const override;
-  void SetSink(
-      const rtc::scoped_refptr<webrtc::AudioSinkInterface>& sink) override;
+  void SetSink(rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) override;
 
   webrtc::AudioReceiveStream::Config config_;
   webrtc::AudioReceiveStream::Stats stats_;
   int received_packets_;
-  rtc::scoped_refptr<webrtc::AudioSinkInterface> sink_;
+  rtc::scoped_ptr<webrtc::AudioSinkInterface> sink_;
 };
 
 class FakeVideoSendStream final : public webrtc::VideoSendStream,
                                   public webrtc::VideoCaptureInput {
  public:
   FakeVideoSendStream(const webrtc::VideoSendStream::Config& config,
                       const webrtc::VideoEncoderConfig& encoder_config);
   webrtc::VideoSendStream::Config GetConfig() const;
   webrtc::VideoEncoderConfig GetEncoderConfig() const;
   std::vector<webrtc::VideoStream> GetVideoStreams();
@@ skipping 134 matching lines @@
   std::vector<FakeAudioSendStream*> audio_send_streams_;
   std::vector<FakeVideoReceiveStream*> video_receive_streams_;
   std::vector<FakeAudioReceiveStream*> audio_receive_streams_;
 
   int num_created_send_streams_;
   int num_created_receive_streams_;
 };
 
 }  // namespace cricket
 #endif  // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_