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

webrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.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.
  */
 
 /******************************************************************
 
  iLBC Speech Coder ANSI-C Source Code
 
  WebRtcIlbcfix_InterpolateSamples.c
 
 ******************************************************************/
 
 #include "defines.h"
 #include "constants.h"
 
 void WebRtcIlbcfix_InterpolateSamples(
     int16_t *interpSamples, /* (o) The interpolated samples */
     int16_t *CBmem,   /* (i) The CB memory */
-    int16_t lMem    /* (i) Length of the CB memory */
+    size_t lMem    /* (i) Length of the CB memory */
                                       ) {
   int16_t *ppi, *ppo, i, j, temp1, temp2;
   int16_t *tmpPtr;
 
   /* Calculate the 20 vectors of interpolated samples (4 samples each)
      that are used in the codebooks for lag 20 to 39 */
   tmpPtr = interpSamples;
   for (j=0; j<20; j++) {
     temp1 = 0;
     temp2 = 3;
     ppo = CBmem+lMem-4;
     ppi = CBmem+lMem-j-24;
     for (i=0; i<4; i++) {
 
       *tmpPtr++ = (int16_t)((WebRtcIlbcfix_kAlpha[temp2] * *ppo) >> 15) +
           (int16_t)((WebRtcIlbcfix_kAlpha[temp1] * *ppi) >> 15);
 
       ppo++;
       ppi++;
       temp1++;
       temp2--;
     }
   }
 
   return;
 }