Added VideoTrackInterface::GetSink method, for use by VideoRtpReceiver. Get rid of FrameInput.

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 241 matching lines @@
   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,
                        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 SetVideoPlayout(
+      uint32_t ssrc,
+      bool enable,
+      rtc::VideoSinkInterface<cricket::VideoFrame>* sink) override;
   void SetVideoSend(uint32_t ssrc,
                     bool enable,
                     const cricket::VideoOptions* options) override;
 
   // Implements DtmfProviderInterface.
   virtual bool CanInsertDtmf(const std::string& track_id);
   virtual bool InsertDtmf(const std::string& track_id,
                           int code, int duration);
   virtual sigslot::signal0<>* GetOnDestroyedSignal();
 
@@ skipping 241 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_