Only update VCMTiming on every received frame instead of every received packet.

webrtc/video/rtp_stream_receiver.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 238 matching lines @@
 
 int32_t RtpStreamReceiver::OnReceivedPayloadData(
     const uint8_t* payload_data,
     size_t payload_size,
     const WebRtcRTPHeader* rtp_header) {
   WebRtcRTPHeader rtp_header_with_ntp = *rtp_header;
   rtp_header_with_ntp.ntp_time_ms =
       ntp_estimator_.Estimate(rtp_header->header.timestamp);
   if (jitter_buffer_experiment_) {
     VCMPacket packet(payload_data, payload_size, rtp_header_with_ntp);
-    timing_->IncomingTimestamp(packet.timestamp, clock_->TimeInMilliseconds());
     packet.timesNacked = nack_module_->OnReceivedPacket(packet);
 
     if (packet.codec == kVideoCodecH264) {
       // Only when we start to receive packets will we know what payload type
       // that will be used. When we know the payload type insert the correct
       // sps/pps into the tracker.
       if (packet.payloadType != last_payload_type_) {
         last_payload_type_ = packet.payloadType;
         InsertSpsPpsIntoTracker(packet.payloadType);
       }
@@ skipping 136 matching lines @@
   rtp_rtcp_->SendNack(sequence_numbers);
 }
 
 int32_t RtpStreamReceiver::ResendPackets(const uint16_t* sequence_numbers,
                                          uint16_t length) {
   return rtp_rtcp_->SendNACK(sequence_numbers, length);
 }
 
 void RtpStreamReceiver::OnReceivedFrame(
     std::unique_ptr<video_coding::RtpFrameObject> frame) {
+  if (!frame->delayed_by_retransmission())
+    timing_->IncomingTimestamp(frame->timestamp, clock_->TimeInMilliseconds());
   reference_finder_->ManageFrame(std::move(frame));
 }
 
 void RtpStreamReceiver::OnCompleteFrame(
     std::unique_ptr<video_coding::FrameObject> frame) {
   {
     rtc::CritScope lock(&last_seq_num_cs_);
     video_coding::RtpFrameObject* rtp_frame =
         static_cast<video_coding::RtpFrameObject*>(frame.get());
     last_seq_num_for_pic_id_[rtp_frame->picture_id] = rtp_frame->last_seq_num();
@@ skipping 250 matching lines @@
     return;
 
   if (!sprop_decoder.DecodeSprop(sprop_base64_it->second.c_str()))
     return;
 
   tracker_.InsertSpsPpsNalus(sprop_decoder.sps_nalu(),
                              sprop_decoder.pps_nalu());
 }
 
 }  // namespace webrtc