Implement timing frames.

webrtc/video/rtp_video_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.
  */
 
 #include "webrtc/video/rtp_video_stream_receiver.h"
 
+#include <utility>
 #include <vector>
-#include <utility>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/location.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/common_types.h"
 #include "webrtc/config.h"
 #include "webrtc/media/base/mediaconstants.h"
 #include "webrtc/modules/pacing/packet_router.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
@@ skipping 207 matching lines @@
 int32_t RtpVideoStreamReceiver::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);
   VCMPacket packet(payload_data, payload_size, rtp_header_with_ntp);
   packet.timesNacked =
       nack_module_ ? nack_module_->OnReceivedPacket(packet) : -1;
+  packet.receive_time_ms = clock_->TimeInMilliseconds();
 
   // In the case of a video stream without picture ids and no rtx the
   // RtpFrameReferenceFinder will need to know about padding to
   // correctly calculate frame references.
   if (packet.sizeBytes == 0) {
     reference_finder_->PaddingReceived(packet.seqNum);
     packet_buffer_->PaddingReceived(packet.seqNum);
     return 0;
   }
 
@@ skipping 261 matching lines @@
   }
   rtp_header.type.Video.codec = payload_specific.Video.videoCodecType;
   rtp_header.type.Video.rotation = kVideoRotation_0;
   if (header.extension.hasVideoRotation) {
     rtp_header.type.Video.rotation = header.extension.videoRotation;
   }
   rtp_header.type.Video.content_type = VideoContentType::UNSPECIFIED;
   if (header.extension.hasVideoContentType) {
     rtp_header.type.Video.content_type = header.extension.videoContentType;
   }
+  rtp_header.type.Video.video_timing = {0u, 0u, 0u, 0u, 0u, 0u, false};
+  if (header.extension.has_video_timing) {
+    rtp_header.type.Video.video_timing = header.extension.video_timing;
+    rtp_header.type.Video.video_timing.is_timing_frame = true;
+  }
   rtp_header.type.Video.playout_delay = header.extension.playout_delay;
 
   OnReceivedPayloadData(nullptr, 0, &rtp_header);
 }
 
 bool RtpVideoStreamReceiver::DeliverRtcp(const uint8_t* rtcp_packet,
                                          size_t rtcp_packet_length) {
   {
     rtc::CritScope lock(&receive_cs_);
     if (!receiving_) {
@@ skipping 141 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