Add sender controlled playout delay limits

webrtc/modules/rtp_rtcp/source/rtp_sender_video.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.
  */
 
@@ skipping 230 matching lines @@
     fec_enabled = fec_enabled_;
   }
 
   // Register CVO rtp header extension at the first time when we receive a frame
   // with pending rotation.
   RTPSenderInterface::CVOMode cvo_mode = RTPSenderInterface::kCVONone;
   if (video_header && video_header->rotation != kVideoRotation_0) {
     cvo_mode = _rtpSender.ActivateCVORtpHeaderExtension();
   }
 
-  uint16_t rtp_header_length = _rtpSender.RTPHeaderLength();
   size_t payload_bytes_to_send = payloadSize;
   const uint8_t* data = payloadData;
 
   // TODO(changbin): we currently don't support to configure the codec to
   // output multiple partitions for VP8. Should remove below check after the
   // issue is fixed.
   const RTPFragmentationHeader* frag =
       (videoType == kRtpVideoVp8) ? NULL : fragmentation;
 
   packetizer->SetPayloadData(data, payload_bytes_to_send, frag);
 
   bool first = true;
   bool last = false;
   while (!last) {
     uint8_t dataBuffer[IP_PACKET_SIZE] = {0};
     size_t payload_bytes_in_packet = 0;
+
+    // Write RTP header.
+    // Note that RTP header size is dynamically computed since there may be
+    // an optional RTP header extension.
+    uint16_t rtp_header_length = _rtpSender.BuildRTPheader(
+        dataBuffer, payloadType, false, captureTimeStamp, capture_time_ms);
+
     if (!packetizer->NextPacket(&dataBuffer[rtp_header_length],
                                 &payload_bytes_in_packet, &last)) {
       return -1;
     }
 
-    // Write RTP header.
     // Set marker bit true if this is the last packet in frame.
-    _rtpSender.BuildRTPheader(
-        dataBuffer, payloadType, last, captureTimeStamp, capture_time_ms);
+    if (last)
+      dataBuffer[1] |= kRtpMarkerBitMask;
+
     // According to
     // 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)). The MTSI client may also add the payload bytes onto the last RTP
     // packet in each group of packets which make up another type of frame
     // (e.g. a P-Frame) only if the current value is different from the previous
     // value sent.
@@ skipping 60 matching lines @@
   rtc::CritScope cs(&crit_);
   return _retransmissionSettings;
 }
 
 void RTPSenderVideo::SetSelectiveRetransmissions(uint8_t settings) {
   rtc::CritScope cs(&crit_);
   _retransmissionSettings = settings;
 }
 
 }  // namespace webrtc