Don't update the jitter estimate with frames containing retransmitted packets.

webrtc/modules/video_coding/frame_object.cc

 /*
  *  Copyright (c) 2016 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 103 matching lines @@
 }
 
 int64_t RtpFrameObject::ReceivedTime() const {
   return received_time_;
 }
 
 int64_t RtpFrameObject::RenderTime() const {
   return _renderTimeMs;
 }
 
+bool RtpFrameObject::delayed_by_retransmission() const {
+  return times_nacked() > 0;
+}
+
 rtc::Optional<RTPVideoTypeHeader> RtpFrameObject::GetCodecHeader() const {
   rtc::CritScope lock(&packet_buffer_->crit_);
   VCMPacket* packet = packet_buffer_->GetPacket(first_seq_num_);
   if (!packet)
     return rtc::Optional<RTPVideoTypeHeader>();
   return rtc::Optional<RTPVideoTypeHeader>(packet->video_header.codecHeader);
 }
 
 }  // namespace video_coding
 }  // namespace webrtc