Make rtcp sender use max transfer unit.

webrtc/modules/rtp_rtcp/source/rtp_sender.h

 /*
  *  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 49 matching lines @@
                                  bool inc_sequence_number = true) = 0;
 
   virtual size_t RTPHeaderLength() const = 0;
   // Returns the next sequence number to use for a packet and allocates
   // 'packets_to_send' number of sequence numbers. It's important all allocated
   // sequence numbers are used in sequence to avoid perceived packet loss.
   virtual uint16_t AllocateSequenceNumber(uint16_t packets_to_send) = 0;
   virtual uint16_t SequenceNumber() const = 0;
   virtual size_t MaxPayloadLength() const = 0;
   virtual size_t MaxDataPayloadLength() const = 0;
-  virtual uint16_t PacketOverHead() const = 0;
   virtual uint16_t ActualSendBitrateKbit() const = 0;
 
   virtual int32_t SendToNetwork(uint8_t* data_buffer,
                                 size_t payload_length,
                                 size_t rtp_header_length,
                                 int64_t capture_time_ms,
                                 StorageType storage,
                                 RtpPacketSender::Priority priority) = 0;
 
   virtual bool UpdateVideoRotation(uint8_t* rtp_packet,
@@ skipping 58 matching lines @@
   void SetStartTimestamp(uint32_t timestamp, bool force);
 
   uint32_t GenerateNewSSRC();
   void SetSSRC(uint32_t ssrc);
 
   uint16_t SequenceNumber() const override;
   void SetSequenceNumber(uint16_t seq);
 
   void SetCsrcs(const std::vector<uint32_t>& csrcs);
 
-  int32_t SetMaxPayloadLength(size_t length, uint16_t packet_over_head);
+  void SetMaxPayloadLength(size_t max_payload_length);
 
   int32_t SendOutgoingData(FrameType frame_type,
                            int8_t payload_type,
                            uint32_t timestamp,
                            int64_t capture_time_ms,
                            const uint8_t* payload_data,
                            size_t payload_size,
                            const RTPFragmentationHeader* fragmentation,
                            const RTPVideoHeader* rtp_hdr = NULL);
 
@@ skipping 80 matching lines @@
                          int8_t payload_type,
                          bool marker_bit,
                          uint32_t capture_timestamp,
                          int64_t capture_time_ms,
                          const bool timestamp_provided = true,
                          const bool inc_sequence_number = true) override;
 
   size_t RTPHeaderLength() const override;
   uint16_t AllocateSequenceNumber(uint16_t packets_to_send) override;
   size_t MaxPayloadLength() const override;
-  uint16_t PacketOverHead() const override;
 
   // Current timestamp.
   uint32_t Timestamp() const override;
   uint32_t SSRC() const override;
 
   int32_t SendToNetwork(uint8_t* data_buffer,
                         size_t payload_length,
                         size_t rtp_header_length,
                         int64_t capture_time_ms,
                         StorageType storage,
@@ skipping 169 matching lines @@
   RtpPacketSender* const paced_sender_;
   TransportSequenceNumberAllocator* const transport_sequence_number_allocator_;
   TransportFeedbackObserver* const transport_feedback_observer_;
   int64_t last_capture_time_ms_sent_;
   rtc::CriticalSection send_critsect_;
 
   Transport *transport_;
   bool sending_media_ GUARDED_BY(send_critsect_);
 
   size_t max_payload_length_;
-  uint16_t packet_over_head_;
 
   int8_t payload_type_ GUARDED_BY(send_critsect_);
   std::map<int8_t, RtpUtility::Payload*> payload_type_map_;
 
   RtpHeaderExtensionMap rtp_header_extension_map_;
   int32_t transmission_time_offset_;
   uint32_t absolute_send_time_;
   VideoRotation rotation_;
   CVOMode cvo_mode_;
   uint16_t transport_sequence_number_;
@@ skipping 43 matching lines @@
   // that the target bitrate is still valid.
   rtc::scoped_ptr<CriticalSectionWrapper> target_bitrate_critsect_;
   uint32_t target_bitrate_ GUARDED_BY(target_bitrate_critsect_);
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_