Update Packet::GetHeader() to copy playout_delay

webrtc/modules/rtp_rtcp/source/rtp_packet.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_packet.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_packet.cc b/webrtc/modules/rtp_rtcp/source/rtp_packet.cc
index 19e898422f5b6157d6aaa5bdc0c060bfd201df40..6b54be7b4c8abb424d74085ba7c35d78eb1a6087 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_packet.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_packet.cc
@@ -174,6 +174,7 @@ void Packet::GetHeader(RTPHeader* header) const {
           &header->extension.videoContentType);
   GetExtension<RtpStreamId>(&header->extension.stream_id);
   GetExtension<RepairedRtpStreamId>(&header->extension.repaired_stream_id);
+  GetExtension<PlayoutDelayLimits>(&header->extension.playout_delay);
 }
 
 size_t Packet::headers_size() const {