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

webrtc/p2p/base/port.h

 /*
  *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 449 matching lines @@
   // covers.
   virtual int Send(const void* data, size_t size,
                    const rtc::PacketOptions& options) = 0;
 
   // Error if Send() returns < 0
   virtual int GetError() = 0;
 
   sigslot::signal4<Connection*, const char*, size_t, const rtc::PacketTime&>
       SignalReadPacket;
 
+  sigslot::signal2<Connection*, const rtc::SentPacket&> SignalSentPacket;

[pthatcher1, 2015/09/25 23:24:58]

I don't see this actually wired up to anything.  Does that come later?

[stefan-webrtc, 2015/10/02 13:29:12]

I dropped the ball here. I think it should be properly wired up now.

+
   sigslot::signal1<Connection*> SignalReadyToSend;
 
   // Called when a packet is received on this connection.
   void OnReadPacket(const char* data, size_t size,
                     const rtc::PacketTime& packet_time);
 
   // Called when the socket is currently able to send.
   void OnReadyToSend();
 
   // Called when a connection is determined to be no longer useful to us.  We
@@ skipping 149 matching lines @@
                    const rtc::PacketOptions& options);
   virtual int GetError() { return error_; }
 
  private:
   int error_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_P2P_BASE_PORT_H_