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

webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c

 /*
  *  Copyright (c) 2012 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 124 matching lines @@
 /* arr_ts        - value given by timeGetTime() time of arrival in samples of packet from NetEq      */
 /* pksize        - size of packet in bytes, from NetEq                                               */
 /* Index         - integer (range 0...23) indicating bottle neck & jitter as estimated by other side */
 /* returns 0 if everything went fine, -1 otherwise                                                   */
 int16_t WebRtcIsac_UpdateBandwidthEstimator(
     BwEstimatorstr* bwest_str,
     const uint16_t rtp_number,
     const int32_t frame_length,
     const uint32_t send_ts,
     const uint32_t arr_ts,
-    const int32_t pksize
+    const size_t pksize
     /*,    const uint16_t Index*/)
 {
   float weight = 0.0f;
   float curr_bw_inv = 0.0f;
   float rec_rtp_rate;
   float t_diff_proj;
   float arr_ts_diff;
   float send_ts_diff;
   float arr_time_noise;
   float arr_time_noise_abs;
@@ skipping 866 matching lines @@
       s2nr = a_60 + b_60 * bottle_neck * 0.001 + c_60 * bottle_neck *
           bottle_neck * 0.000001;
       break;
     default:
       s2nr = 0;
   }
 
   return s2nr;
 
 }