Remove BaseSession::SignalNewDescription. It was only used by GTP and now just clutters the code.

webrtc/p2p/base/session.h

 /*
  *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 394 matching lines @@
   // Handles messages posted to us.
   virtual void OnMessage(rtc::Message *pmsg);
 
  protected:
   bool IsCandidateAllocationDone() const;
 
   State state_;
   Error error_;
   std::string error_desc_;
 
-  // Fires the new description signal according to the current state.
-  virtual void SignalNewDescription();
   // This method will delete the Transport and TransportChannelImpls
   // and replace those with the Transport object of the first
   // MediaContent in bundle_group.
   bool BundleContentGroup(const ContentGroup* bundle_group);
 
  private:
   // Helper methods to push local and remote transport descriptions.
   bool PushdownLocalTransportDescription(
       const SessionDescription* sdesc, ContentAction action,
       std::string* error_desc);
   bool PushdownRemoteTransportDescription(
       const SessionDescription* sdesc, ContentAction action,
       std::string* error_desc);
 
   void MaybeCandidateAllocationDone();
 
   // Log session state.
   void LogState(State old_state, State new_state);
 
   // Returns true and the TransportInfo of the given |content_name|
   // from |description|. Returns false if it's not available.
   static bool GetTransportDescription(const SessionDescription* description,
                                       const std::string& content_name,
                                       TransportDescription* info);
 
-  // Gets the ContentAction and ContentSource according to the session state.
-  bool GetContentAction(ContentAction* action, ContentSource* source);
-
   rtc::Thread* const signaling_thread_;
   rtc::Thread* const worker_thread_;
   PortAllocator* const port_allocator_;
   const std::string sid_;
   const std::string content_type_;
   const std::string transport_type_;
   bool initiator_;
   rtc::SSLIdentity* identity_;
   rtc::SSLProtocolVersion ssl_max_version_;
   rtc::scoped_ptr<const SessionDescription> local_description_;
   rtc::scoped_ptr<SessionDescription> remote_description_;
   uint64 ice_tiebreaker_;
   // This flag will be set to true after the first role switch. This flag
   // will enable us to stop any role switch during the call.
   bool role_switch_;
   TransportMap transports_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_P2P_BASE_SESSION_H_