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

webrtc/modules/audio_coding/codecs/isac/fix/source/decode_bwe.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.
  */
 
 /*
  * decode_bwe.c
  *
  * This C file contains the internal decode bandwidth estimate function.
  *
  */
 
 
 #include "bandwidth_estimator.h"
 #include "codec.h"
 #include "entropy_coding.h"
 #include "structs.h"
 
 
 
 
 int WebRtcIsacfix_EstimateBandwidth(BwEstimatorstr *bwest_str,
                                     Bitstr_dec  *streamdata,
-                                    int32_t  packet_size,
+                                    size_t packet_size,
                                     uint16_t rtp_seq_number,
                                     uint32_t send_ts,
                                     uint32_t arr_ts)
 {
   int16_t index;
-  int16_t frame_samples;
+  size_t frame_samples;
   int err;
 
   /* decode framelength */
   err = WebRtcIsacfix_DecodeFrameLen(streamdata, &frame_samples);
   /* error check */
   if (err<0) {
     return err;
   }
 
   /* decode BW estimation */
   err = WebRtcIsacfix_DecodeSendBandwidth(streamdata, &index);
   /* error check */
   if (err<0) {
     return err;
   }
 
   /* Update BWE with received data */
   err = WebRtcIsacfix_UpdateUplinkBwImpl(
       bwest_str,
       rtp_seq_number,
-      frame_samples * 1000 / FS,
+      (int16_t)(frame_samples * 1000 / FS),
       send_ts,
       arr_ts,
-      (int16_t) packet_size,  /* in bytes */
+      packet_size,  /* in bytes */
       index);
 
   /* error check */
   if (err<0) {
     return err;
   }
 
   /* Succesful */
   return 0;
 }