Refactor RTPPacketHistory to use a packet struct.

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 cec7e17e4b621fbe7984281633db41d0793e19df..bc312125533d13e2d38ad40ccd72debac1ac64f2 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -1026,8 +1026,7 @@ int32_t RTPSender::SendToNetwork(
 
   // Used for NACK and to spread out the transmission of packets.
   if (packet_history_.PutRTPPacket(buffer, rtp_header_length + payload_length,
-                                   max_payload_length_, capture_time_ms,
-                                   storage) != 0) {
+                                   capture_time_ms, storage) != 0) {
     return -1;
   }