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

webrtc/modules/audio_coding/codecs/isac/main/source/lpc_analysis.h

 /*
  *  Copyright (c) 2011 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.
  */
 
 /*
  * lpc_analysis.h
  *
  * LPC functions
  *
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_ANALYSIS_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_ANALYSIS_H_
 
 #include "settings.h"
 #include "structs.h"
 
-double WebRtcIsac_LevDurb(double *a, double *k, double *r, int order);
+double WebRtcIsac_LevDurb(double *a, double *k, double *r, size_t order);
 
 void WebRtcIsac_GetVars(const double *input, const int16_t *pitchGains_Q12,
                        double *oldEnergy, double *varscale);
 
 void WebRtcIsac_GetLpcCoefLb(double *inLo, double *inHi, MaskFiltstr *maskdata,
                              double signal_noise_ratio, const int16_t *pitchGains_Q12,
                              double *lo_coeff, double *hi_coeff);
 
 
 void WebRtcIsac_GetLpcGain(
     double         signal_noise_ratio,
     const double*  filtCoeffVecs,
     int            numVecs,
     double*        gain,
     double         corrLo[][UB_LPC_ORDER + 1],
     const double*  varscale);
 
 void WebRtcIsac_GetLpcCoefUb(
     double*      inSignal,
     MaskFiltstr* maskdata,
     double*      lpCoeff,
     double       corr[][UB_LPC_ORDER + 1],
     double*      varscale,
     int16_t  bandwidth);
 
 #endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_ANALYIS_H_ */