Wire up transport sequence number / send time callbacks to webrtc via libjingle.

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 343 matching lines @@
   rtc::scoped_refptr<PeerConnectionFactory> factory_;
   PeerConnectionObserver* observer_;
   UMAObserver* uma_observer_;
   SignalingState signaling_state_;
   // TODO(bemasc): Remove ice_state_.
   IceState ice_state_;
   IceConnectionState ice_connection_state_;
   IceGatheringState ice_gathering_state_;
 
   rtc::scoped_ptr<cricket::PortAllocator> port_allocator_;
+  rtc::scoped_ptr<MediaControllerInterface> media_controller_;
 
   // Streams added via AddStream.
   rtc::scoped_refptr<StreamCollection> local_streams_;
   // Streams created as a result of SetRemoteDescription.
   rtc::scoped_refptr<StreamCollection> remote_streams_;
 
   // These lists store track info seen in local/remote descriptions.
   TrackInfos remote_audio_tracks_;
   TrackInfos remote_video_tracks_;
   TrackInfos local_audio_tracks_;
@@ skipping 14 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_