Update audio code to use size_t more correctly, webrtc/modules/audio_processing/

webrtc/modules/audio_processing/ns/nsx_core_c.c

 /*
  *  Copyright (c) 2013 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 15 matching lines @@
 void WebRtcNsx_SpeechNoiseProb(NoiseSuppressionFixedC* inst,
                                uint16_t* nonSpeechProbFinal,
                                uint32_t* priorLocSnr,
                                uint32_t* postLocSnr) {
   uint32_t zeros, num, den, tmpU32no1, tmpU32no2, tmpU32no3;
   int32_t invLrtFX, indPriorFX, tmp32, tmp32no1, tmp32no2, besselTmpFX32;
   int32_t frac32, logTmp;
   int32_t logLrtTimeAvgKsumFX;
   int16_t indPriorFX16;
   int16_t tmp16, tmp16no1, tmp16no2, tmpIndFX, tableIndex, frac, intPart;
-  int i, normTmp, normTmp2, nShifts;
+  size_t i;
+  int normTmp, normTmp2, nShifts;
 
   // compute feature based on average LR factor
   // this is the average over all frequencies of the smooth log LRT
   logLrtTimeAvgKsumFX = 0;
   for (i = 0; i < inst->magnLen; i++) {
     besselTmpFX32 = (int32_t)postLocSnr[i]; // Q11
     normTmp = WebRtcSpl_NormU32(postLocSnr[i]);
     num = postLocSnr[i] << normTmp;  // Q(11+normTmp)
     if (normTmp > 10) {
       den = priorLocSnr[i] << (normTmp - 11);  // Q(normTmp)
@@ skipping 204 matching lines @@
           tmp32no1 = (int32_t)inst->priorNonSpeechProb << 8;  // Q22
 
           nonSpeechProbFinal[i] = tmp32no1 /
               (inst->priorNonSpeechProb + invLrtFX);  // Q8
         }
       }
     }
   }
 }