Refactor NACK bitrate allocation

webrtc/modules/rtp_rtcp/mocks/mock_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 95 matching lines @@
       void(uint32_t));
   MOCK_METHOD2(SetRtxSendPayloadType, void(int, int));
   MOCK_CONST_METHOD0(RtxSendPayloadType, std::pair<int, int>());
   MOCK_METHOD1(SetSendingStatus,
       int32_t(const bool sending));
   MOCK_CONST_METHOD0(Sending,
       bool());
   MOCK_METHOD1(SetSendingMediaStatus, void(const bool sending));
   MOCK_CONST_METHOD0(SendingMedia,
       bool());
-  MOCK_CONST_METHOD4(BitrateSent,
-                     void(uint32_t* totalRate,
-                          uint32_t* videoRate,
-                          uint32_t* fecRate,
-                          uint32_t* nackRate));
+  MOCK_METHOD4(BitrateSent,
+               void(uint32_t* totalRate,
+                    uint32_t* videoRate,
+                    uint32_t* fecRate,
+                    uint32_t* nackRate));
   MOCK_METHOD1(RegisterVideoBitrateObserver, void(BitrateStatisticsObserver*));
   MOCK_CONST_METHOD0(GetVideoBitrateObserver, BitrateStatisticsObserver*(void));
   MOCK_CONST_METHOD1(EstimatedReceiveBandwidth,
       int(uint32_t* available_bandwidth));
   MOCK_METHOD8(SendOutgoingData,
       int32_t(const FrameType frameType,
               const int8_t payloadType,
               const uint32_t timeStamp,
               int64_t capture_time_ms,
               const uint8_t* payloadData,
@@ skipping 125 matching lines @@
       void(StreamDataCountersCallback*));
   MOCK_CONST_METHOD0(GetSendChannelRtpStatisticsCallback,
       StreamDataCountersCallback*(void));
   // Members.
   unsigned int remote_ssrc_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_