Revert of Storing raw audio sink for default audio track.

talk/app/webrtc/webrtcsession.h

 /*
  * 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 232 matching lines @@
   virtual bool GetLocalTrackIdBySsrc(uint32_t ssrc, std::string* track_id);
   virtual bool GetRemoteTrackIdBySsrc(uint32_t ssrc, std::string* track_id);
 
   // AudioMediaProviderInterface implementation.
   void SetAudioPlayout(uint32_t ssrc, bool enable) override;
   void SetAudioSend(uint32_t ssrc,
                     bool enable,
                     const cricket::AudioOptions& options,
                     cricket::AudioRenderer* renderer) override;
   void SetAudioPlayoutVolume(uint32_t ssrc, double volume) override;
-  void SetRawAudioSink(
-      uint32_t ssrc,
-      const rtc::scoped_refptr<AudioSinkInterface>& sink) override;
+  void SetRawAudioSink(uint32_t ssrc,
+                       rtc::scoped_ptr<AudioSinkInterface> sink) override;
 
   // Implements VideoMediaProviderInterface.
   bool SetCaptureDevice(uint32_t ssrc, cricket::VideoCapturer* camera) override;
   void SetVideoPlayout(uint32_t ssrc,
                        bool enable,
                        cricket::VideoRenderer* renderer) override;
   void SetVideoSend(uint32_t ssrc,
                     bool enable,
                     const cricket::VideoOptions* options) override;
 
@@ skipping 245 matching lines @@
   PeerConnectionInterface::BundlePolicy bundle_policy_;
 
   // Declares the RTCP mux policy for the WebRTCSession.
   PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
 };
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_WEBRTCSESSION_H_