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

webrtc/modules/audio_coding/codecs/isac/main/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.
  */
 
 #include "structs.h"
 #include "bandwidth_estimator.h"
 #include "entropy_coding.h"
 #include "codec.h"
 
 
 int
 WebRtcIsac_EstimateBandwidth(
     BwEstimatorstr*           bwest_str,
     Bitstr*                   streamdata,
-    int32_t               packet_size,
+    size_t                packet_size,
     uint16_t              rtp_seq_number,
     uint32_t              send_ts,
     uint32_t              arr_ts,
     enum IsacSamplingRate encoderSampRate,
     enum IsacSamplingRate decoderSampRate)
 {
   int16_t  index;
   int16_t  frame_samples;
   uint32_t sendTimestampIn16kHz;
   uint32_t arrivalTimestampIn16kHz;
@@ skipping 48 matching lines @@
                                             (frame_samples * 1000) / FS, sendTimestampIn16kHz,
                                             arrivalTimestampIn16kHz, packet_size);
   // error check
   if(err < 0)
   {
     return err;
   }
 
   return 0;
 }