Make OverheadObserver::OnOverheadChanged count RTP headers only

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 54 matching lines @@
   ~RTPSender();
 
   void ProcessBitrate();
 
   uint16_t ActualSendBitrateKbit() const;
 
   uint32_t VideoBitrateSent() const;
   uint32_t FecOverheadRate() const;
   uint32_t NackOverheadRate() const;
 
-  // Includes size of RTP and FEC headers.
-  size_t MaxDataPayloadLength() const;
+  // Excluding size of RTP and FEC headers.
+  size_t MaxPayloadSize() const;
 
   int32_t RegisterPayload(const char* payload_name,
                           const int8_t payload_type,
                           const uint32_t frequency,
                           const size_t channels,
                           const uint32_t rate);
 
   int32_t DeRegisterSendPayload(const int8_t payload_type);
 
   void SetSendPayloadType(int8_t payload_type);
@@ skipping 12 matching lines @@
   void SetTimestampOffset(uint32_t timestamp);
 
   uint32_t GenerateNewSSRC();
   void SetSSRC(uint32_t ssrc);
 
   uint16_t SequenceNumber() const;
   void SetSequenceNumber(uint16_t seq);
 
   void SetCsrcs(const std::vector<uint32_t>& csrcs);
 
-  void SetMaxPayloadLength(size_t max_payload_length);
+  void SetMaxRtpPacketSize(size_t max_packet_size);
 
   bool 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_header,
                         uint32_t* transport_frame_id_out);
@@ skipping 37 matching lines @@
   // Create empty packet, fills ssrc, csrcs and reserve place for header
   // extensions RtpSender updates before sending.
   std::unique_ptr<RtpPacketToSend> AllocatePacket() const;
   // Allocate sequence number for provided packet.
   // Save packet's fields to generate padding that doesn't break media stream.
   // Return false if sending was turned off.
   bool AssignSequenceNumber(RtpPacketToSend* packet);
 
   size_t RtpHeaderLength() const;
   uint16_t AllocateSequenceNumber(uint16_t packets_to_send);
-  size_t MaxPayloadLength() const;
+  // Including RTP headers.
+  size_t MaxRtpPacketSize() const;
 
   uint32_t SSRC() const;
 
   rtc::Optional<uint32_t> FlexfecSsrc() const;
 
   bool SendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
                      StorageType storage,
                      RtpPacketSender::Priority priority);
 
   // Audio.
@@ skipping 23 matching lines @@
   void RegisterRtpStatisticsCallback(StreamDataCountersCallback* callback);
   StreamDataCountersCallback* GetRtpStatisticsCallback() const;
 
   uint32_t BitrateSent() const;
 
   void SetRtpState(const RtpState& rtp_state);
   RtpState GetRtpState() const;
   void SetRtxRtpState(const RtpState& rtp_state);
   RtpState GetRtxRtpState() const;
 
-  void SetTransportOverhead(int transport_overhead);
-
  protected:
   int32_t CheckPayloadType(int8_t payload_type, RtpVideoCodecTypes* video_type);
 
  private:
   // Maps capture time in milliseconds to send-side delay in milliseconds.
   // Send-side delay is the difference between transmission time and capture
   // time.
   typedef std::map<int64_t, int> SendDelayMap;
 
   size_t SendPadData(size_t bytes, int probe_cluster_id);
@@ skipping 42 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_;
+  size_t max_packet_size_;
 
   int8_t payload_type_ GUARDED_BY(send_critsect_);
   std::map<int8_t, RtpUtility::Payload*> payload_type_map_;
 
   RtpHeaderExtensionMap rtp_header_extension_map_ GUARDED_BY(send_critsect_);
 
   // Tracks the current request for playout delay limits from application
   // and decides whether the current RTP frame should include the playout
   // delay extension on header.
   PlayoutDelayOracle playout_delay_oracle_;
@@ skipping 30 matching lines @@
   uint32_t last_rtp_timestamp_ GUARDED_BY(send_critsect_);
   int64_t capture_time_ms_ GUARDED_BY(send_critsect_);
   int64_t last_timestamp_time_ms_ GUARDED_BY(send_critsect_);
   bool media_has_been_sent_ GUARDED_BY(send_critsect_);
   bool last_packet_marker_bit_ GUARDED_BY(send_critsect_);
   std::vector<uint32_t> csrcs_ GUARDED_BY(send_critsect_);
   int rtx_ GUARDED_BY(send_critsect_);
   uint32_t ssrc_rtx_ GUARDED_BY(send_critsect_);
   // Mapping rtx_payload_type_map_[associated] = rtx.
   std::map<int8_t, int8_t> rtx_payload_type_map_ GUARDED_BY(send_critsect_);
-  size_t transport_overhead_bytes_per_packet_ GUARDED_BY(send_critsect_);
   size_t rtp_overhead_bytes_per_packet_ GUARDED_BY(send_critsect_);
 
   RateLimiter* const retransmission_rate_limiter_;
   OverheadObserver* overhead_observer_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_