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

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 105 matching lines @@
                       public AudioProviderInterface,
                       public DataChannelFactory,
                       public VideoProviderInterface,
                       public DtmfProviderInterface,
                       public DataChannelProviderInterface {
  public:
   WebRtcSession(cricket::ChannelManager* channel_manager,
                 rtc::Thread* signaling_thread,
                 rtc::Thread* worker_thread,
                 cricket::PortAllocator* port_allocator,
-                MediaStreamSignaling* mediastream_signaling);
+                MediaStreamSignaling* mediastream_signaling,
+                webrtc::CallFactory* call_factory);
   virtual ~WebRtcSession();
 
   bool Initialize(
       const PeerConnectionFactoryInterface::Options& options,
       const MediaConstraintsInterface* constraints,
       rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
       const PeerConnectionInterface::RTCConfiguration& rtc_configuration);
   // Deletes the voice, video and data channel and changes the session state
   // to STATE_RECEIVEDTERMINATE.
   void Terminate();
@@ skipping 239 matching lines @@
 
   // Invoked when TransportController connection completion is signaled.
   // Reports stats for all transports in use.
   void ReportTransportStats();
 
   // Gather the usage of IPv4/IPv6 as best connection.
   void ReportBestConnectionState(const cricket::TransportStats& stats);
 
   void ReportNegotiatedCiphers(const cricket::TransportStats& stats);
 
+  void OnSentPacket(cricket::TransportChannel* channel,
+                    const rtc::SentPacket& sent_packet);
+
   rtc::scoped_ptr<MediaControllerInterface> media_controller_;
   rtc::scoped_ptr<cricket::VoiceChannel> voice_channel_;
   rtc::scoped_ptr<cricket::VideoChannel> video_channel_;
   rtc::scoped_ptr<cricket::DataChannel> data_channel_;
   cricket::ChannelManager* channel_manager_;
   MediaStreamSignaling* mediastream_signaling_;
+  webrtc::CallFactory* call_factory_;
   IceObserver* ice_observer_;
   PeerConnectionInterface::IceConnectionState ice_connection_state_;
   bool ice_connection_receiving_;
   rtc::scoped_ptr<SessionDescriptionInterface> local_desc_;
   rtc::scoped_ptr<SessionDescriptionInterface> remote_desc_;
   // Candidates that arrived before the remote description was set.
   std::vector<IceCandidateInterface*> saved_candidates_;
   // If the remote peer is using a older version of implementation.
   bool older_version_remote_peer_;
   bool dtls_enabled_;
@@ skipping 23 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_