Make rtcp sender use max transfer unit.

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 115 matching lines @@
       audio_configured_(audio),
       audio_(audio ? new RTPSenderAudio(clock, this) : nullptr),
       video_(audio ? nullptr : new RTPSenderVideo(clock, this)),
       paced_sender_(paced_sender),
       transport_sequence_number_allocator_(sequence_number_allocator),
       transport_feedback_observer_(transport_feedback_observer),
       last_capture_time_ms_sent_(0),
       transport_(transport),
       sending_media_(true),                      // Default to sending media.
       max_payload_length_(IP_PACKET_SIZE - 28),  // Default is IP-v4/UDP.
-      packet_over_head_(28),
       payload_type_(-1),
       payload_type_map_(),
       rtp_header_extension_map_(),
       transmission_time_offset_(0),
       absolute_send_time_(0),
       rotation_(kVideoRotation_0),
       cvo_mode_(kCVONone),
       transport_sequence_number_(0),
       // NACK.
       nack_byte_count_times_(),
@@ skipping 221 matching lines @@
 
 int8_t RTPSender::SendPayloadType() const {
   rtc::CritScope lock(&send_critsect_);
   return payload_type_;
 }
 
 int RTPSender::SendPayloadFrequency() const {
   return audio_ != NULL ? audio_->AudioFrequency() : kVideoPayloadTypeFrequency;
 }
 
-int32_t RTPSender::SetMaxPayloadLength(size_t max_payload_length,
-                                       uint16_t packet_over_head) {
+void RTPSender::SetMaxPayloadLength(size_t max_payload_length) {
   // Sanity check.
   RTC_DCHECK(max_payload_length >= 100 && max_payload_length <= IP_PACKET_SIZE)
       << "Invalid max payload length: " << max_payload_length;
   rtc::CritScope lock(&send_critsect_);
   max_payload_length_ = max_payload_length;
-  packet_over_head_ = packet_over_head;
-  return 0;
 }
 
 size_t RTPSender::MaxDataPayloadLength() const {
   int rtx;
   {
     rtc::CritScope lock(&send_critsect_);
     rtx = rtx_;
   }
   if (audio_configured_) {
     return max_payload_length_ - RTPHeaderLength();
   } else {
     return max_payload_length_ - RTPHeaderLength()  // RTP overhead.
            - video_->FECPacketOverhead()            // FEC/ULP/RED overhead.
            - ((rtx) ? 2 : 0);                       // RTX overhead.
   }
 }
 
 size_t RTPSender::MaxPayloadLength() const {
   return max_payload_length_;
 }
 
-uint16_t RTPSender::PacketOverHead() const { return packet_over_head_; }
-
 void RTPSender::SetRtxStatus(int mode) {
   rtc::CritScope lock(&send_critsect_);
   rtx_ = mode;
 }
 
 int RTPSender::RtxStatus() const {
   rtc::CritScope lock(&send_critsect_);
   return rtx_;
 }
 
@@ skipping 1490 matching lines @@
   rtc::CritScope lock(&send_critsect_);
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = start_timestamp_;
 
   return state;
 }
 
 }  // namespace webrtc