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

webrtc/base/asyncpacketsocket.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 16 matching lines @@
 
   int rtp_sendtime_extension_id;    // extension header id present in packet.
   std::vector<char> srtp_auth_key;  // Authentication key.
   int srtp_auth_tag_len;            // Authentication tag length.
   int64_t srtp_packet_index;        // Required for Rtp Packet authentication.
 };
 
 // This structure holds meta information for the packet which is about to send
 // over network.
 struct PacketOptions {
-  PacketOptions() : dscp(DSCP_NO_CHANGE) {}
-  explicit PacketOptions(DiffServCodePoint dscp) : dscp(dscp) {}
+  PacketOptions() : dscp(DSCP_NO_CHANGE), packet_id(-1) {}
+  explicit PacketOptions(DiffServCodePoint dscp) : dscp(dscp), packet_id(-1) {}
 
   DiffServCodePoint dscp;
+  int packet_id;  // 16 bits, -1 represents "not set".
   PacketTimeUpdateParams packet_time_params;
 };
 
 // This structure will have the information about when packet is actually
 // received by socket.
 struct PacketTime {
   PacketTime() : timestamp(-1), not_before(-1) {}
   PacketTime(int64_t timestamp, int64_t not_before)
       : timestamp(timestamp), not_before(not_before) {}
 
@@ skipping 51 matching lines @@
   // TODO: Remove SetError().
   virtual int GetError() const = 0;
   virtual void SetError(int error) = 0;
 
   // Emitted each time a packet is read. Used only for UDP and
   // connected TCP sockets.
   sigslot::signal5<AsyncPacketSocket*, const char*, size_t,
                    const SocketAddress&,
                    const PacketTime&> SignalReadPacket;
 
+  // Emitted each time a packet is sent.
+  sigslot::signal2<AsyncPacketSocket*, const SentPacket&> SignalSentPacket;
+
   // Emitted when the socket is currently able to send.
   sigslot::signal1<AsyncPacketSocket*> SignalReadyToSend;
 
   // Emitted after address for the socket is allocated, i.e. binding
   // is finished. State of the socket is changed from BINDING to BOUND
   // (for UDP and server TCP sockets) or CONNECTING (for client TCP
   // sockets).
   sigslot::signal2<AsyncPacketSocket*, const SocketAddress&> SignalAddressReady;
 
   // Emitted for client TCP sockets when state is changed from
   // CONNECTING to CONNECTED.
   sigslot::signal1<AsyncPacketSocket*> SignalConnect;
 
   // Emitted for client TCP sockets when state is changed from
   // CONNECTED to CLOSED.
   sigslot::signal2<AsyncPacketSocket*, int> SignalClose;
 
   // Used only for listening TCP sockets.
   sigslot::signal2<AsyncPacketSocket*, AsyncPacketSocket*> SignalNewConnection;
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(AsyncPacketSocket);
 };
 
 }  // namespace rtc
 
 #endif  // WEBRTC_BASE_ASYNCPACKETSOCKET_H_