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 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) {}
+  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 timestamp, int64 not_before)
       : timestamp(timestamp), not_before(not_before) {
   }
 
   int64 timestamp;  // Receive time after socket delivers the data.
   int64 not_before; // Earliest possible time the data could have arrived,
                     // indicating the potential error in the |timestamp| value,
                     // in case the system, is busy. For example, the time of
                     // the last select() call.
                     // If unknown, this value will be set to zero.
 };
 
 inline PacketTime CreatePacketTime(int64 not_before) {
   return PacketTime(TimeMicros(), not_before);
 }
 
+struct SentPacket {
+  SentPacket() : packet_id(-1), send_time_ms(-1) {}
+  SentPacket(int32 packet_id, int64 send_time_ms)
+      : packet_id(packet_id), send_time_ms(send_time_ms) {}
+
+  int packet_id;
+  int64 send_time_ms;
+};
+
 // Provides the ability to receive packets asynchronously. Sends are not
 // buffered since it is acceptable to drop packets under high load.
 class AsyncPacketSocket : public sigslot::has_slots<> {
  public:
   enum State {
     STATE_CLOSED,
     STATE_BINDING,
     STATE_BOUND,
     STATE_CONNECTING,
     STATE_CONNECTED
@@ skipping 28 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::signal3<AsyncPacketSocket*, const SocketAddress&, const SentPacket&>
+      SignalPacketSent;

[stefan-webrtc, 2015/10/05 13:43:19]

I'm not sure if I actually need SocketAddress here, I suspect not? It doesn't
matter to me if the address changes, the packet ids are still the same. Or can
multiple peer connections share the same socket?

[pthatcher1, 2015/10/05 18:30:14]

You shouldn't care what socket it is, as long as the packet IDs don't overlap
between different transports (different audio and video streams if they are not
bundled).  If the IDs don't overlap, then you're fine.  However, the typical
style is to include the object as the first parameter.  So even though you don't
need it, it's consistent to include it.

[stefan-webrtc, 2015/10/07 16:55:25]

Acknowledged.

+
   // 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_