Making WebRtcSession fire a destroyed signal.

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 309 matching lines @@
   }
 
   // Called when voice_channel_, video_channel_ and data_channel_ are created
   // and destroyed. As a result of, for example, setting a new description.
   sigslot::signal0<> SignalVoiceChannelCreated;
   sigslot::signal0<> SignalVoiceChannelDestroyed;
   sigslot::signal0<> SignalVideoChannelCreated;
   sigslot::signal0<> SignalVideoChannelDestroyed;
   sigslot::signal0<> SignalDataChannelCreated;
   sigslot::signal0<> SignalDataChannelDestroyed;
+  // Called when the whole session is destroyed.
+  sigslot::signal0<> SignalDestroyed;
 
   // Called when a valid data channel OPEN message is received.
   // std::string represents the data channel label.
   sigslot::signal2<const std::string&, const InternalDataChannelInit&>
       SignalDataChannelOpenMessage;
 
  private:
   // Indicates the type of SessionDescription in a call to SetLocalDescription
   // and SetRemoteDescription.
   enum Action {
@@ skipping 174 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_