Add support for transport wide sequence numbers

webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h

 /*
  *  Copyright (c) 2012 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 275 matching lines @@
   virtual void OnReceivedEstimatedBitrate(uint32_t bitrate) = 0;
 
   virtual void OnReceivedRtcpReceiverReport(
       const ReportBlockList& report_blocks,
       int64_t rtt,
       int64_t now_ms) = 0;
 
   virtual ~RtcpBandwidthObserver() {}
 };
 
+class SendTimeObserver {
+ public:
+  SendTimeObserver() {}
+  virtual ~SendTimeObserver() {}
+
+  // Transport-wide sequence number and timestamp (system time in milliseconds),
+  // of when the packet was put on the wire.
+  virtual void OnPacketSent(uint16_t sequence_number, int64_t send_time) = 0;

[stefan-webrtc, 2015/07/29 09:04:11]

Maybe transport_sequence_number?

[sprang_webrtc, 2015/07/29 10:03:25]

Done.

+};
+
 class RtcpRttStats {
  public:
   virtual void OnRttUpdate(int64_t rtt) = 0;
 
   virtual int64_t LastProcessedRtt() const = 0;
 
   virtual ~RtcpRttStats() {};
 };
 
 // Null object version of RtpFeedback.
@@ skipping 52 matching lines @@
   uint64_t single_packet_loss_count;
   // The number of events in which more than one adjacent packet was lost.
   uint64_t multiple_packet_loss_event_count;
   // The number of packets lost in events where more than one adjacent packet
   // was lost.
   uint64_t multiple_packet_loss_packet_count;
 };
 
 }  // namespace webrtc
 #endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_