Make OverheadObserver::OnOverheadChanged count RTP headers only

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.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 189 matching lines @@
   void SetREMBData(uint32_t bitrate,
                    const std::vector<uint32_t>& ssrcs) override;
 
   // (TMMBR) Temporary Max Media Bit Rate.
   bool TMMBR() const override;
 
   void SetTMMBRStatus(bool enable) override;
 
   void SetTmmbn(std::vector<rtcp::TmmbItem> bounding_set) override;
 
-  uint16_t MaxPayloadLength() const override;
+  size_t MaxPayloadSize() const override;
 
-  uint16_t MaxDataPayloadLength() const override;
+  size_t MaxRtpPacketSize() const override;
 
-  int32_t SetMaxTransferUnit(uint16_t size) override;
-
-  // TODO(michaelt): deprecate the function.
-  int32_t SetTransportOverhead(bool tcp,
-                               bool ipv6,
-                               uint8_t authentication_overhead = 0) override;
-
-  void SetTransportOverhead(int transport_overhead_per_packet) override;
+  void SetMaxRtpPacketSize(size_t size) override;

[stefan-webrtc, 2016/12/20 13:05:01]

max_packet_size to align with rtp_rtcp.h

[nisse-webrtc, 2016/12/20 13:38:13]

Done.

 
   // (NACK) Negative acknowledgment part.
 
   int SelectiveRetransmissions() const override;
 
   int SetSelectiveRetransmissions(uint8_t settings) override;
 
   // Send a Negative acknowledgment packet.
   // TODO(philipel): Deprecate SendNACK and use SendNack instead.
   int32_t SendNACK(const uint16_t* nack_list, uint16_t size) override;
@@ skipping 126 matching lines @@
   PacketLossStats receive_loss_stats_;
 
   // The processed RTT from RtcpRttStats.
   rtc::CriticalSection critical_section_rtt_;
   int64_t rtt_ms_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_