Implement timing frames.

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

 /*
  *  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 722 matching lines @@
       return false;
     packet_to_send = packet_rtx.get();
   }
 
   int64_t now_ms = clock_->TimeInMilliseconds();
   int64_t diff_ms = now_ms - capture_time_ms;
   packet_to_send->SetExtension<TransmissionOffset>(kTimestampTicksPerMs *
                                                    diff_ms);
   packet_to_send->SetExtension<AbsoluteSendTime>(now_ms);
 
+  if (packet->HasExtension<VideoTimingExtension>())

[åsapersson, 2017/06/12 14:33:42]

packet_to_send?

[ilnik, 2017/06/13 08:43:13]

Done. Thank's for catching that.

+    packet->set_pacer_exit_time_ms(clock_->TimeInMilliseconds());

[åsapersson, 2017/06/12 14:33:42]

maybe use now_ms

[ilnik, 2017/06/13 08:43:13]

Done.

+

[åsapersson, 2017/06/12 14:33:42]

Maybe add a unit test, e.g. in rtp_sender_unittest.cc.

[ilnik, 2017/06/13 08:43:13]

Done. Please check rtp_sender_unittest.cc

   PacketOptions options;
   if (UpdateTransportSequenceNumber(packet_to_send, &options.packet_id)) {
     AddPacketToTransportFeedback(options.packet_id, *packet_to_send,
                                  pacing_info);
   }
 
   if (!is_retransmit && !send_over_rtx) {
     UpdateDelayStatistics(packet->capture_time_ms(), now_ms);
     UpdateOnSendPacket(options.packet_id, packet->capture_time_ms(),
                        packet->Ssrc());
@@ skipping 514 matching lines @@
     if (rtp_overhead_bytes_per_packet_ == packet.headers_size()) {
       return;
     }
     rtp_overhead_bytes_per_packet_ = packet.headers_size();
     overhead_bytes_per_packet = rtp_overhead_bytes_per_packet_;
   }
   overhead_observer_->OnOverheadChanged(overhead_bytes_per_packet);
 }
 
 }  // namespace webrtc