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

webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.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 1852 matching lines @@
   0, 21845, 43690, 65535};
 
 /* pointer to cdf array for frame length indicator */
 const uint16_t *kFrameLenCdfPtr[1] = {kFrameLenCdf};
 
 /* initial cdf index for decoder of frame length indicator */
 const uint16_t kFrameLenInitIndex[1] = {1};
 
 
 int WebRtcIsacfix_DecodeFrameLen(Bitstr_dec *streamdata,
-                                 int16_t *framesamples)
+                                 size_t *framesamples)
 {
 
   int err;
   int16_t frame_mode;
 
   err = 0;
   /* entropy decoding of frame length [1:30ms,2:60ms] */
   err = WebRtcIsacfix_DecHistOneStepMulti(&frame_mode, streamdata, kFrameLenCdfPtr, kFrameLenInitIndex, 1);
   if (err<0)  // error check
     return -ISAC_RANGE_ERROR_DECODE_FRAME_LENGTH;
@@ skipping 141 matching lines @@
 
     index_gQQ[k] = pos2QQ + WebRtcIsacfix_kQuantMinGain[k]; //ATTN: ok?
     if (index_gQQ[k] < 0) {
       index_gQQ[k] = 0;
     }
     else if (index_gQQ[k] > WebRtcIsacfix_kMaxIndGain[k]) {
       index_gQQ[k] = WebRtcIsacfix_kMaxIndGain[k];
     }
   }
 }