Remove definition of non-existent method.

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 281 matching lines @@
   virtual void OnTransportRequestSignaling(cricket::Transport* transport);
   virtual void OnTransportConnecting(cricket::Transport* transport);
   virtual void OnTransportWritable(cricket::Transport* transport);
   virtual void OnTransportCompleted(cricket::Transport* transport);
   virtual void OnTransportFailed(cricket::Transport* transport);
   virtual void OnTransportProxyCandidatesReady(
       cricket::TransportProxy* proxy,
       const cricket::Candidates& candidates);
   virtual void OnCandidatesAllocationDone();
 
-  // Creates local session description with audio and video contents.
-  bool CreateDefaultLocalDescription();
   // Enables media channels to allow sending of media.
   void EnableChannels();
   // Creates a JsepIceCandidate and adds it to the local session description
   // and notify observers. Called when a new local candidate have been found.
   void ProcessNewLocalCandidate(const std::string& content_name,
                                 const cricket::Candidates& candidates);
   // Returns the media index for a local ice candidate given the content name.
   // Returns false if the local session description does not have a media
   // content called  |content_name|.
   bool GetLocalCandidateMediaIndex(const std::string& content_name,
@@ skipping 100 matching lines @@
   PeerConnectionInterface::BundlePolicy bundle_policy_;
 
   // Declares the RTCP mux policy for the WebRTCSession.
   PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
 };
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_WEBRTCSESSION_H_