Update a ton of audio code to use size_t more correctly and in general reduce

webrtc/modules/audio_coding/codecs/isac/main/source/codec.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.
  */
 
 /*
  * codec.h
  *
  * This header file contains the calls to the internal encoder
  * and decoder functions.
  *
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_
 
 #include "structs.h"
 
 
 void WebRtcIsac_ResetBitstream(Bitstr* bit_stream);
 
 int WebRtcIsac_EstimateBandwidth(BwEstimatorstr* bwest_str, Bitstr* streamdata,
-                                 int32_t packet_size,
+                                 size_t packet_size,
                                  uint16_t rtp_seq_number,
                                  uint32_t send_ts, uint32_t arr_ts,
                                  enum IsacSamplingRate encoderSampRate,
                                  enum IsacSamplingRate decoderSampRate);
 
 int WebRtcIsac_DecodeLb(const TransformTables* transform_tables,
                         float* signal_out,
                         ISACLBDecStruct* ISACdec_obj,
                         int16_t* current_framesamples,
                         int16_t isRCUPayload);
@@ skipping 149 matching lines @@
 
 void WebRtcIsac_Spec2time(const TransformTables* tables,
                           double* inre,
                           double* inim,
                           double* outre1,
                           double* outre2,
                           FFTstr* fftstr_obj);
 
 /******************************* filter functions ****************************/
 
-void WebRtcIsac_AllPoleFilter(double* InOut, double* Coef, int lengthInOut,
+void WebRtcIsac_AllPoleFilter(double* InOut, double* Coef, size_t lengthInOut,
                               int orderCoef);
 
-void WebRtcIsac_AllZeroFilter(double* In, double* Coef, int lengthInOut,
+void WebRtcIsac_AllZeroFilter(double* In, double* Coef, size_t lengthInOut,
                               int orderCoef, double* Out);
 
 void WebRtcIsac_ZeroPoleFilter(double* In, double* ZeroCoef, double* PoleCoef,
-                               int lengthInOut, int orderCoef, double* Out);
+                               size_t lengthInOut, int orderCoef, double* Out);
 
 
 /***************************** filterbank functions **************************/
 
 void WebRtcIsac_SplitAndFilterFloat(float* in, float* LP, float* HP,
                                     double* LP_la, double* HP_la,
                                     PreFiltBankstr* prefiltdata);
 
 
 void WebRtcIsac_FilterAndCombineFloat(float* InLP, float* InHP, float* Out,
                                       PostFiltBankstr* postfiltdata);
 
 
 /************************* normalized lattice filters ************************/
 
 void WebRtcIsac_NormLatticeFilterMa(int orderCoef, float* stateF, float* stateG,
                                     float* lat_in, double* filtcoeflo,
                                     double* lat_out);
 
 void WebRtcIsac_NormLatticeFilterAr(int orderCoef, float* stateF, float* stateG,
                                     double* lat_in, double* lo_filt_coef,
                                     float* lat_out);
 
 void WebRtcIsac_Dir2Lat(double* a, int orderCoef, float* sth, float* cth);
 
-void WebRtcIsac_AutoCorr(double* r, const double* x, int N, int order);
+void WebRtcIsac_AutoCorr(double* r, const double* x, size_t N, size_t order);
 
 #endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_ */