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.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 117 matching lines @@
             }
             break;
         }
     }
     return paramLenght;
 }
 
 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)
 {
     unsigned int travelTimeMs;
         const int headerSizeByte = 35;
 
         int headerRate;
 
@@ skipping 20 matching lines @@
     }
 
     BN_data->arrival_time = (BN_data->whenPrevPackLeftMs *
         (receiverSampFreqHz / 1000));
 
 //      if (BN_data->arrival_time < BN_data->sample_count)
 //              BN_data->arrival_time = BN_data->sample_count;
 
         BN_data->rtp_number++;
 }