Implement timing frames.

webrtc/modules/rtp_rtcp/source/rtp_packet.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 154 matching lines @@
   header->extension.hasTransportSequenceNumber =
       GetExtension<TransportSequenceNumber>(
           &header->extension.transportSequenceNumber);
   header->extension.hasAudioLevel = GetExtension<AudioLevel>(
       &header->extension.voiceActivity, &header->extension.audioLevel);
   header->extension.hasVideoRotation =
       GetExtension<VideoOrientation>(&header->extension.videoRotation);
   header->extension.hasVideoContentType =
       GetExtension<VideoContentTypeExtension>(
           &header->extension.videoContentType);
+  header->extension.hasVideoTiming =
+      GetExtension<VideoTimingExtension>(&header->extension.videoTiming);
   GetExtension<RtpStreamId>(&header->extension.stream_id);
   GetExtension<RepairedRtpStreamId>(&header->extension.repaired_stream_id);
   GetExtension<PlayoutDelayLimits>(&header->extension.playout_delay);
 }
 
 size_t Packet::headers_size() const {
   return payload_offset_;
 }
 
 size_t Packet::payload_size() const {
@@ skipping 388 matching lines @@
 uint8_t* Packet::WriteAt(size_t offset) {
   return buffer_.data() + offset;
 }
 
 void Packet::WriteAt(size_t offset, uint8_t byte) {
   buffer_.data()[offset] = byte;
 }
 
 }  // namespace rtp
 }  // namespace webrtc