Implement timing frames.

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 93 matching lines @@
   RTC_CHECK(last_packet && last_packet->markerBit);
   // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/
   // ts_126114v120700p.pdf Section 7.4.5.
   // The MTSI client shall add the payload bytes as defined in this clause
   // onto the last RTP packet in each group of packets which make up a key
   // frame (I-frame or IDR frame in H.264 (AVC), or an IRAP picture in H.265
   // (HEVC)).
   rotation_ = last_packet->video_header.rotation;
   _rotation_set = true;
   content_type_ = last_packet->video_header.content_type;
+  if (last_packet->video_header.video_timing.is_timing_frame) {
+    // ntp_time_ms_ may be -1 if not estimated yet. This is not a problem,
+    // as this will be dealt with at the time of reporting.
+    timing_.is_timing_frame = true;
+    timing_.encode_start_ms =
+        ntp_time_ms_ +
+        last_packet->video_header.video_timing.encode_start_ms_delta;
+    timing_.encode_finish_ms =
+        ntp_time_ms_ +
+        last_packet->video_header.video_timing.encode_finish_ms_delta;
+    timing_.packetization_finish_ms =
+        ntp_time_ms_ +
+        last_packet->video_header.video_timing.packetization_finish_ms_delta;
+    timing_.pacer_exit_ms =
+        ntp_time_ms_ +
+        last_packet->video_header.video_timing.pacer_exit_ms_delta;
+    timing_.network_timestamp_ms =
+        ntp_time_ms_ +
+        last_packet->video_header.video_timing.network_timstamp_ms_delta;
+    timing_.receive_start_ms = first_packet->receive_time_ms;
+    timing_.receive_finish_ms = last_packet->receive_time_ms;
+  } else {
+    timing_.is_timing_frame = false;
+  }
 }
 
 RtpFrameObject::~RtpFrameObject() {
   packet_buffer_->ReturnFrame(this);
 }
 
 uint16_t RtpFrameObject::first_seq_num() const {
   return first_seq_num_;
 }
 
@@ skipping 36 matching lines @@
 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