Make rtcp sender use max transfer unit.

webrtc/modules/rtp_rtcp/source/rtcp_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 81 matching lines @@
   void OnPacketReady(uint8_t* data, size_t length) override {
     if (transport_->SendRtcp(data, length)) {
       bytes_sent_ += length;
       if (event_log_) {
         event_log_->LogRtcpPacket(kOutgoingPacket, MediaType::ANY, data,
                                   length);
       }
     }
   }
 
-  size_t SendPackets() {
-    rtcp::CompoundPacket::Build(this);
+  size_t SendPackets(size_t max_payload_length) {
+    RTC_DCHECK_LE(max_payload_length, static_cast<size_t>(IP_PACKET_SIZE));
+    uint8_t buffer[IP_PACKET_SIZE];
+    BuildExternalBuffer(buffer, max_payload_length, this);
     return bytes_sent_;
   }
 
  private:
   Transport* transport_;
   RtcEventLog* const event_log_;
   size_t bytes_sent_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(PacketContainer);
 };
@@ skipping 48 matching lines @@
       remote_ssrc_(0),
       receive_statistics_(receive_statistics),
 
       sequence_number_fir_(0),
 
       remb_bitrate_(0),
 
       tmmbr_help_(),
       tmmbr_send_(0),
       packet_oh_send_(0),
+      max_payload_length_(IP_PACKET_SIZE - 28),  // IPv4 + UDP by default.
 
       app_sub_type_(0),
       app_name_(0),
       app_data_(nullptr),
       app_length_(0),
 
       xr_send_receiver_reference_time_enabled_(false),
       packet_type_counter_observer_(packet_type_counter_observer) {
   RTC_DCHECK(transport_ != nullptr);
 
@@ skipping 88 matching lines @@
 
 void RTCPSender::SetTMMBRStatus(bool enable) {
   rtc::CritScope lock(&critical_section_rtcp_sender_);
   if (enable) {
     SetFlag(RTCPPacketType::kRtcpTmmbr, false);
   } else {
     ConsumeFlag(RTCPPacketType::kRtcpTmmbr, true);
   }
 }
 
+void RTCPSender::SetMaxPayloadLength(size_t max_payload_length) {
+  max_payload_length_ = max_payload_length;
+}
+
 void RTCPSender::SetStartTimestamp(uint32_t start_timestamp) {
   rtc::CritScope lock(&critical_section_rtcp_sender_);
   start_timestamp_ = start_timestamp;
 }
 
 void RTCPSender::SetLastRtpTime(uint32_t rtp_timestamp,
                                 int64_t capture_time_ms) {
   rtc::CritScope lock(&critical_section_rtcp_sender_);
   last_rtp_timestamp_ = rtp_timestamp;
   if (capture_time_ms < 0) {
@@ skipping 512 matching lines @@
     }
 
     if (packet_type_counter_observer_ != nullptr) {
       packet_type_counter_observer_->RtcpPacketTypesCounterUpdated(
           remote_ssrc_, packet_type_counter_);
     }
 
     RTC_DCHECK(AllVolatileFlagsConsumed());
   }
 
-  size_t bytes_sent = container.SendPackets();
+  size_t bytes_sent = container.SendPackets(max_payload_length_);
   return bytes_sent == 0 ? -1 : 0;
 }
 
 void RTCPSender::PrepareReport(const std::set<RTCPPacketType>& packetTypes,
                                const FeedbackState& feedback_state) {
   // Add all flags as volatile. Non volatile entries will not be overwritten
   // and all new volatile flags added will be consumed by the end of this call.
   SetFlags(packetTypes, true);
 
   if (packet_type_counter_.first_packet_time_ms == -1)
@@ skipping 200 matching lines @@
     }
 
     Transport* const transport_;
     RtcEventLog* const event_log_;
     bool send_failure_;
     // TODO(terelius): We would like to
     // RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Sender);
     // but we can't because of an incorrect warning (C4822) in MVS 2013.
   } sender(transport_, event_log_);
 
+  RTC_DCHECK_LE(max_payload_length_, static_cast<size_t>(IP_PACKET_SIZE));
   uint8_t buffer[IP_PACKET_SIZE];
-  return packet.BuildExternalBuffer(buffer, IP_PACKET_SIZE, &sender) &&
+  return packet.BuildExternalBuffer(buffer, max_payload_length_, &sender) &&
          !sender.send_failure_;
 }
 
 }  // namespace webrtc