Merging BaseSession code into WebRtcSession.

talk/app/webrtc/peerconnection.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 191 matching lines @@
                           VideoTrackInterface* video_track);
 
   // Implements IceObserver
   void OnIceConnectionChange(IceConnectionState new_state) override;
   void OnIceGatheringChange(IceGatheringState new_state) override;
   void OnIceCandidate(const IceCandidateInterface* candidate) override;
   void OnIceComplete() override;
   void OnIceConnectionReceivingChange(bool receiving) override;
 
   // Signals from WebRtcSession.
-  void OnSessionStateChange(cricket::BaseSession* session,
-                            cricket::BaseSession::State state);
+  void OnSessionStateChange(WebRtcSession* session, WebRtcSession::State state);
   void ChangeSignalingState(SignalingState signaling_state);
 
   rtc::Thread* signaling_thread() const {
     return factory_->signaling_thread();
   }
 
   void PostSetSessionDescriptionFailure(SetSessionDescriptionObserver* observer,
                                         const std::string& error);
   void PostCreateSessionDescriptionFailure(
       CreateSessionDescriptionObserver* observer,
@@ skipping 165 matching lines @@
   // because its destruction fires signals (such as VoiceChannelDestroyed)
   // which will trigger some final actions in PeerConnection...
   rtc::scoped_ptr<WebRtcSession> session_;
   // ... But stats_ depends on session_ so it should be destroyed even earlier.
   rtc::scoped_ptr<StatsCollector> stats_;
 };
 
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_PEERCONNECTION_H_