Refactor NACK bitrate allocation

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 291 matching lines @@
                        uint32_t* NTPfrac,
                        uint32_t* remote_sr) const;
 
   bool LastReceivedXrReferenceTimeInfo(RtcpReceiveTimeInfo* info) const;
 
   int32_t BoundingSet(bool* tmmbr_owner, TMMBRSet* bounding_set_rec);
 
   void BitrateSent(uint32_t* total_rate,
                    uint32_t* video_rate,
                    uint32_t* fec_rate,
-                   uint32_t* nackRate) const override;
+                   uint32_t* nackRate) override;
 
   // Good state of RTP receiver inform sender.
   int32_t SendRTCPReferencePictureSelection(uint64_t picture_id) override;
 
   void RegisterSendChannelRtpStatisticsCallback(
       StreamDataCountersCallback* callback) override;
   StreamDataCountersCallback* GetSendChannelRtpStatisticsCallback()
       const override;
 
   void OnReceivedNACK(const std::list<uint16_t>& nack_sequence_numbers);
@@ skipping 44 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_