Implement timing frames.

webrtc/video/payload_router.cc

 /*
  *  Copyright (c) 2015 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 112 matching lines @@
   RTC_DCHECK(!rtp_modules_.empty());
   if (!active_)
     return Result(Result::ERROR_SEND_FAILED);
 
   RTPVideoHeader rtp_video_header;
   memset(&rtp_video_header, 0, sizeof(RTPVideoHeader));
   if (codec_specific_info)
     CopyCodecSpecific(codec_specific_info, &rtp_video_header);
   rtp_video_header.rotation = encoded_image.rotation_;
   rtp_video_header.content_type = encoded_image.content_type_;
+  if (encoded_image.timing_.is_timing_frame) {
+    rtp_video_header.video_timing.encode_start_ms_delta =
+        VideoTiming::GetDeltaCappedMs(encoded_image.capture_time_ms_,
+                                      encoded_image.timing_.encode_start_ms);

[holmer, 2017/06/19 07:48:26]

Is there a reason why the delta is computed here and not when the
encode_start_ms timestamp is taken? AFAICT you might just as well do it there
already?

[ilnik, 2017/06/19 08:41:51]

Reason is: we have the same data-structure to store timestamps on sending time
and on receiving time in EncodedImage. Doing otherwise would require to separate
data-structures in EncodedImage for send and receive sides. Thus, we can't store
deltas in EncodedImage, and they are computed then rtp video header is formed.

[holmer, 2017/06/19 12:17:47]

Acknowledged.

+    rtp_video_header.video_timing.encode_finish_ms_delta =
+        VideoTiming::GetDeltaCappedMs(encoded_image.capture_time_ms_,
+                                      encoded_image.timing_.encode_finish_ms);
+    rtp_video_header.video_timing.packetization_finish_ms_delta = 0;
+    rtp_video_header.video_timing.pacer_exit_ms_delta = 0;
+    rtp_video_header.video_timing.network_timstamp_ms_delta = 0;
+    rtp_video_header.video_timing.network2_timstamp_ms_delta = 0;

[holmer, 2017/06/19 07:48:26]

Maybe add a default ctor which initializes everything to zero instead?

[ilnik, 2017/06/19 08:41:51]

Tried that, it causes some weird problems with RTPVideoHeader class:
error: [chromium-style] Complex class/struct needs an explicit out-of-line
constructor.

Adding there constructor seems like a too big of a change, it also creates a
cascade change to WebRtcRTPHeader and so on.

[holmer, 2017/06/19 12:17:47]

Acknowledged.

+    rtp_video_header.video_timing.is_timing_frame = true;
+  } else {
+    rtp_video_header.video_timing.is_timing_frame = false;
+  }
   rtp_video_header.playout_delay = encoded_image.playout_delay_;
 
   int stream_index = rtp_video_header.simulcastIdx;
   RTC_DCHECK_LT(stream_index, rtp_modules_.size());
   uint32_t frame_id;
   bool send_result = rtp_modules_[stream_index]->SendOutgoingData(
       encoded_image._frameType, payload_type_, encoded_image._timeStamp,
       encoded_image.capture_time_ms_, encoded_image._buffer,
       encoded_image._length, fragmentation, &rtp_video_header, &frame_id);
   if (!send_result)
@@ skipping 16 matching lines @@
         BitrateAllocation layer_bitrate;
         for (int tl = 0; tl < kMaxTemporalStreams; ++tl)
           layer_bitrate.SetBitrate(0, tl, bitrate.GetBitrate(si, tl));
         rtp_modules_[si]->SetVideoBitrateAllocation(layer_bitrate);
       }
     }
   }
 }
 
 }  // namespace webrtc