Make rtcp sender use max transfer unit.

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 100 matching lines @@
       nack_last_seq_number_sent_(0),
       key_frame_req_method_(kKeyFrameReqPliRtcp),
       remote_bitrate_(configuration.remote_bitrate_estimator),
       rtt_stats_(configuration.rtt_stats),
       critical_section_rtt_(CriticalSectionWrapper::CreateCriticalSection()),
       rtt_ms_(0) {
   // Make sure that RTCP objects are aware of our SSRC.
   uint32_t SSRC = rtp_sender_.SSRC();
   rtcp_sender_.SetSSRC(SSRC);
   SetRtcpReceiverSsrcs(SSRC);
+  SetMaxTransferUnit(IP_PACKET_SIZE);
 }
 
 // Returns the number of milliseconds until the module want a worker thread
 // to call Process.
 int64_t ModuleRtpRtcpImpl::TimeUntilNextProcess() {
   const int64_t now = clock_->TimeInMilliseconds();
   const int64_t kRtpRtcpMaxIdleTimeProcessMs = 5;
   return kRtpRtcpMaxIdleTimeProcessMs - (now - last_process_time_);
 }
 
@@ skipping 341 matching lines @@
   } else {
     // UDP.
     packet_overhead += 8;
   }
   packet_overhead += authentication_overhead;
 
   if (packet_overhead == packet_overhead_) {
     // Ok same as before.
     return 0;
   }
-  // Calc diff.
-  int16_t packet_over_head_diff = packet_overhead - packet_overhead_;
 
-  // Store new.
+  size_t mtu = rtp_sender_.MaxPayloadLength() + packet_overhead_;
+  size_t max_payload_length = mtu - packet_overhead;
   packet_overhead_ = packet_overhead;
-
-  uint16_t length =
-      rtp_sender_.MaxPayloadLength() - packet_over_head_diff;
-  return rtp_sender_.SetMaxPayloadLength(length, packet_overhead_);
+  rtcp_sender_.SetMaxPayloadLength(max_payload_length);
+  rtp_sender_.SetMaxPayloadLength(max_payload_length);
+  return 0;
 }
 
-int32_t ModuleRtpRtcpImpl::SetMaxTransferUnit(const uint16_t mtu) {
-  RTC_DCHECK_LE(mtu, IP_PACKET_SIZE) << "Invalid mtu: " << mtu;
-  return rtp_sender_.SetMaxPayloadLength(mtu - packet_overhead_,
-                                         packet_overhead_);
+int32_t ModuleRtpRtcpImpl::SetMaxTransferUnit(uint16_t mtu) {
+  RTC_DCHECK_LE(mtu, IP_PACKET_SIZE) << "MTU too large: " << mtu;
+  RTC_DCHECK_GT(mtu, packet_overhead_) << "MTU too small: " << mtu;
+  size_t max_payload_length = mtu - packet_overhead_;
+  rtcp_sender_.SetMaxPayloadLength(max_payload_length);
+  rtp_sender_.SetMaxPayloadLength(max_payload_length);
+  return 0;
 }
 
 RtcpMode ModuleRtpRtcpImpl::RTCP() const {
   return rtcp_sender_.Status();
 }
 
 // Configure RTCP status i.e on/off.
 void ModuleRtpRtcpImpl::SetRTCPStatus(const RtcpMode method) {
   rtcp_sender_.SetRTCPStatus(method);
 }
@@ skipping 485 matching lines @@
 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
     StreamDataCountersCallback* callback) {
   rtp_sender_.RegisterRtpStatisticsCallback(callback);
 }
 
 StreamDataCountersCallback*
     ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
   return rtp_sender_.GetRtpStatisticsCallback();
 }
 }  // namespace webrtc