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

webrtc/modules/audio_coding/codecs/isac/main/util/utility.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.
  */
 
@@ skipping 81 matching lines @@
     unsigned int whenPackGeneratedMs;
     unsigned int whenPrevPackLeftMs;
     unsigned int sendTimeMs ;          /* milisecond */
     unsigned int arrival_time;         /* samples */
     unsigned int sample_count;         /* samples, also used as "send time stamp" */
     unsigned int rtp_number;
   } BottleNeckModel;
 
   void get_arrival_time(
       int              current_framesamples,   /* samples */
-      int              packet_size,            /* bytes */
+      size_t           packet_size,            /* bytes */
       int              bottleneck,             /* excluding headers; bits/s */
       BottleNeckModel* BN_data,
       short            senderSampFreqHz,
       short            receiverSampFreqHz);
 
   /* function for reading audio data from PCM file */
   int readframe(
       short* data,
       FILE*  inp,
       int    length);
@@ skipping 22 matching lines @@
       char* stringParam,
       int   maxSize);
 
 #if defined(__cplusplus)
 }
 #endif
 
 
 
 #endif