Update audio code to use size_t more correctly,

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

 /*
  *  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.
  */
 
@@ skipping 57 matching lines @@
   0.00910872, 0.00903832, 0.00896134, 0.00887763, 0.00878706,
   0.00868949, 0.00858478, 0.00847280, 0.00835343, 0.00822653, 0.00809199,
   0.00794970, 0.00779956, 0.00764145, 0.00747530, 0.00730103,
   0.00711857, 0.00692787, 0.00672888, 0.00652158, 0.00630597,
   0.00608208, 0.00584994, 0.00560962, 0.00536124, 0.00510493, 0.00484089,
   0.00456935, 0.00429062, 0.00400505, 0.00371310, 0.00341532,
   0.00311238, 0.00280511, 0.00249452, 0.00218184, 0.00186864,
   0.00155690, 0.00124918, 0.00094895, 0.00066112, 0.00039320, 0.00015881
 };
 
-double WebRtcIsac_LevDurb(double *a, double *k, double *r, int order)
+double WebRtcIsac_LevDurb(double *a, double *k, double *r, size_t order)
 {
 
   double sum, alpha;
-  int m, m_h, i;
+  size_t m, m_h, i;
   alpha = 0; //warning -DH
   a[0] = 1.0;
   if (r[0] < LEVINSON_EPS) { /* if r[0] <= 0, set LPC coeff. to zero */
     for (i = 0; i < order; i++) {
       k[i] = 0;
       a[i+1] = 0;
     }
   } else {
     a[1] = k[0] = -r[1]/r[0];
     alpha = r[0] + r[1] * k[0];
@@ skipping 433 matching lines @@
       {
         res_nrg += aPolynom[j] * corrMat[subFrameCntr][n-j] *
             aPolynom[n];
       }
     }
 
     /* add hearing threshold and compute the gain */
     gain[subFrameCntr] = S_N_R / (sqrt(res_nrg) / *varscale + H_T_H);
   }
 }