Use RtpPacketToSend in RtpSenderVideo

webrtc/modules/rtp_rtcp/source/rtp_sender.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 1075 matching lines @@
 std::unique_ptr<RtpPacketToSend> RTPSender::AllocatePacket() const {
   rtc::CritScope lock(&send_critsect_);
   std::unique_ptr<RtpPacketToSend> packet(
       new RtpPacketToSend(&rtp_header_extension_map_, max_payload_length_));
   packet->SetSsrc(ssrc_);
   packet->SetCsrcs(csrcs_);
   // Reserve extensions, if registered, RtpSender set in SendToNetwork.
   packet->ReserveExtension<AbsoluteSendTime>();
   packet->ReserveExtension<TransmissionOffset>();
   packet->ReserveExtension<TransportSequenceNumber>();
+  if (playout_delay_oracle_.send_playout_delay()) {
+    packet->SetExtension<PlayoutDelayLimits>(
+        playout_delay_oracle_.playout_delay());
+  }
   return packet;
 }
 
 bool RTPSender::AssignSequenceNumber(RtpPacketToSend* packet) {
   rtc::CritScope lock(&send_critsect_);
   if (!sending_media_)
     return false;
   RTC_DCHECK_EQ(packet->Ssrc(), ssrc_);
   packet->SetSequenceNumber(sequence_number_++);
 
@@ skipping 657 matching lines @@
   rtc::CritScope lock(&send_critsect_);
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = timestamp_offset_;
 
   return state;
 }
 
 }  // namespace webrtc