Set actual transport overhead in rtp_rtcp

webrtc/modules/rtp_rtcp/include/rtp_rtcp.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 91 matching lines @@
 
   // **************************************************************************
   // Sender
   // **************************************************************************
 
   // Sets MTU.
   // |size| - Max transfer unit in bytes, default is 1500.
   // Returns -1 on failure else 0.
   virtual int32_t SetMaxTransferUnit(uint16_t size) = 0;
 
-  // Sets transtport overhead. Default is IPv4 and UDP with no encryption.
-  // |tcp| - true for TCP false UDP.
-  // |ipv6| - true for IP version 6 false for version 4.
-  // |authentication_overhead| - number of bytes to leave for an authentication
-  // header.
+  // Sets transtport overhead per packet.
   // Returns -1 on failure else 0
-  virtual int32_t SetTransportOverhead(bool tcp,
-                                       bool ipv6,
-                                       uint8_t authentication_overhead = 0) = 0;
+  virtual int32_t SetTransportOverhead(

[minyue-webrtc, 2016/10/20 09:45:22]

Removing/Modifying API may require to send some notice to webrtc-discussion 2
weeks ahead of time. 

Easy way is to 

1) keep the old one, let new one direct to the old one
2) prepare a separate cleaning up CL and send to webrtc-discussion, then you can
land this CL first
3) after two weeks time without objection, land the cleaning up CL.

[the sun, 2016/10/20 10:40:52]

Sorry, I didn't think about the rtprtcp module being "public"... :/ Perhaps best
to just change back to returning the useless int32.

+      int transport_overhead_per_packet_byte) = 0;
 
   // Returns max payload length, which is a combination of the configuration
   // MaxTransferUnit and TransportOverhead.
   // Does not account for RTP headers and FEC/ULP/RED overhead (when FEC is
   // enabled).
   virtual uint16_t MaxPayloadLength() const = 0;
 
   // Returns max data payload length, which is a combination of the
   // configuration MaxTransferUnit, headers and TransportOverhead.
   // Takes into account RTP headers and FEC/ULP/RED overhead (when FEC is
@@ skipping 340 matching lines @@
   virtual int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) = 0;
 
   // Sends a request for a keyframe.
   // Returns -1 on failure else 0.
   virtual int32_t RequestKeyFrame() = 0;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_