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 1064 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 75 matching lines @@
       case kRtpExtensionAbsoluteSendTime:
         block_length = BuildAbsoluteSendTimeExtension(extension_data);
         break;
       case kRtpExtensionVideoRotation:
         block_length = BuildVideoRotationExtension(extension_data);
         break;
       case kRtpExtensionTransportSequenceNumber:
         block_length = BuildTransportSequenceNumberExtension(
             extension_data, transport_sequence_number_);
         break;
-      case kRtpExtensionPlayoutDelay:
+      case kRtpExtensionPlayoutDelay: {
+        PlayoutDelay playout_delay = playout_delay_oracle_.playout_delay();
         block_length = BuildPlayoutDelayExtension(
-            extension_data, playout_delay_oracle_.min_playout_delay_ms(),
-            playout_delay_oracle_.max_playout_delay_ms());
+            extension_data, playout_delay.min_ms, playout_delay.max_ms);
         break;
+      }
       default:
         assert(false);
     }
     total_block_length += block_length;
     type = rtp_header_extension_map_.Next(type);
   }
   if (total_block_length == 0) {
     // No extension added.
     return 0;
   }
@@ skipping 556 matching lines @@
   rtc::CritScope lock(&send_critsect_);
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = timestamp_offset_;
 
   return state;
 }
 
 }  // namespace webrtc