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

talk/media/webrtc/webrtcvoiceengine.h

 /*
  * libjingle
  * Copyright 2004 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 222 matching lines @@
                                     int reporting_threshold,
                                     int penalty_decay,
                                     int type_event_delay) override;
   bool SetOutputScaling(uint32 ssrc, double left, double right) override;
 
   bool CanInsertDtmf() override;
   bool InsertDtmf(uint32 ssrc, int event, int duration, int flags) override;
 
   void OnPacketReceived(rtc::Buffer* packet,
                         const rtc::PacketTime& packet_time) override;
+  void OnPacketSent(const rtc::SentPacket& packet_sent) override;
   void OnRtcpReceived(rtc::Buffer* packet,
                       const rtc::PacketTime& packet_time) override;
   void OnReadyToSend(bool ready) override {}
   bool GetStats(VoiceMediaInfo* info) override;
   // Gets last reported error from WebRtc voice engine.  This should be only
   // called in response a failure.
   void GetLastMediaError(uint32* ssrc,
                          VoiceMediaChannel::Error* error) override;
 
   // implements Transport interface
@@ skipping 119 matching lines @@
   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
 
   // Do not lock this on the VoE media processor thread; potential for deadlock
   // exists.
   mutable rtc::CriticalSection receive_channels_cs_;
 };
 
 }  // namespace cricket
 
 #endif  // TALK_MEDIA_WEBRTCVOICEENGINE_H_