Remove hardcoded kValueSizeBytes values from variable-length header extensions.

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.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 399 matching lines @@
   return rtp_sender_->TimeToSendPacket(ssrc, sequence_number, capture_time_ms,
                                       retransmission, pacing_info);
 }
 
 size_t ModuleRtpRtcpImpl::TimeToSendPadding(
     size_t bytes,
     const PacedPacketInfo& pacing_info) {
   return rtp_sender_->TimeToSendPadding(bytes, pacing_info);
 }
 
-size_t ModuleRtpRtcpImpl::MaxPayloadSize() const {
-  return rtp_sender_->MaxPayloadSize();
-}
-
 size_t ModuleRtpRtcpImpl::MaxRtpPacketSize() const {
   return rtp_sender_->MaxRtpPacketSize();
 }
 
 void ModuleRtpRtcpImpl::SetMaxRtpPacketSize(size_t rtp_packet_size) {
   RTC_DCHECK_LE(rtp_packet_size, IP_PACKET_SIZE)
       << "rtp packet size too large: " << rtp_packet_size;
   RTC_DCHECK_GT(rtp_packet_size, packet_overhead_)
       << "rtp packet size too small: " << rtp_packet_size;
 
@@ skipping 454 matching lines @@
 StreamDataCountersCallback*
     ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
   return rtp_sender_->GetRtpStatisticsCallback();
 }
 
 void ModuleRtpRtcpImpl::SetVideoBitrateAllocation(
     const BitrateAllocation& bitrate) {
   rtcp_sender_.SetVideoBitrateAllocation(bitrate);
 }
 }  // namespace webrtc