Refactor RTPPacketHistory to use a packet struct.

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 1008 matching lines @@
   if (capture_time_ms > 0) {
     UpdateTransmissionTimeOffset(buffer, payload_length + rtp_header_length,
                                  rtp_header, now_ms - capture_time_ms);
   }
 
   UpdateAbsoluteSendTime(buffer, payload_length + rtp_header_length,
                          rtp_header, now_ms);
 
   // 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;
   }
 
   if (paced_sender_ && storage != kDontStore) {
     // Correct offset between implementations of millisecond time stamps in
     // TickTime and Clock.
     int64_t corrected_time_ms = capture_time_ms + clock_delta_ms_;
     if (!paced_sender_->SendPacket(priority, rtp_header.ssrc,
                                    rtp_header.sequenceNumber, corrected_time_ms,
                                    payload_length, false)) {
@@ skipping 857 matching lines @@
   CriticalSectionScoped lock(send_critsect_.get());
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = start_timestamp_;
 
   return state;
 }
 
 }  // namespace webrtc