Use RtpPacketToSend in RtpSenderVideo

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_sender.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index fc677d77601c0785fc3a55aa25d3900c464463f7..ffb6a81fcb15bd877e15d31055e2495d89cc3259 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -1082,6 +1082,10 @@ std::unique_ptr<RtpPacketToSend> RTPSender::AllocatePacket() const {
   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;
 }
 
@@ -1177,11 +1181,12 @@ uint16_t RTPSender::BuildRtpHeaderExtension(uint8_t* data_buffer,
         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);
     }