Refactor NACK bitrate allocation

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.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 874 matching lines @@
     }
     // Change local SSRC and inform all objects about the new SSRC.
     rtcp_sender_.SetSSRC(new_ssrc);
     SetRtcpReceiverSsrcs(new_ssrc);
   }
 }
 
 void ModuleRtpRtcpImpl::BitrateSent(uint32_t* total_rate,
                                     uint32_t* video_rate,
                                     uint32_t* fec_rate,
-                                    uint32_t* nack_rate) const {
+                                    uint32_t* nack_rate) {
   *total_rate = rtp_sender_.BitrateSent();
   *video_rate = rtp_sender_.VideoBitrateSent();
   *fec_rate = rtp_sender_.FecOverheadRate();
   *nack_rate = rtp_sender_.NackOverheadRate();
 }
 
 void ModuleRtpRtcpImpl::OnRequestSendReport() {
   SendRTCP(kRtcpSr);
 }
 
@@ skipping 90 matching lines @@
 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
     StreamDataCountersCallback* callback) {
   rtp_sender_.RegisterRtpStatisticsCallback(callback);
 }
 
 StreamDataCountersCallback*
     ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
   return rtp_sender_.GetRtpStatisticsCallback();
 }
 }  // namespace webrtc