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

webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.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 130 matching lines @@
  *                           jitter as estimated by other side
  *
  * Return value            : 0 if everything went fine,
  *                           -1 otherwise
  */
 int32_t WebRtcIsacfix_UpdateUplinkBwImpl(BwEstimatorstr *bweStr,
                                          const uint16_t rtpNumber,
                                          const int16_t  frameSize,
                                          const uint32_t sendTime,
                                          const uint32_t arrivalTime,
-                                         const int16_t  pksize,
+                                         const size_t   pksize,
                                          const uint16_t Index)
 {
   uint16_t  weight = 0;
   uint32_t  currBwInv = 0;
   uint16_t  recRtpRate;
   uint32_t  arrTimeProj;
   int32_t   arrTimeDiff;
   int32_t   arrTimeNoise;
   int32_t   arrTimeNoiseAbs;
   int32_t   sendTimeDiff;
@@ skipping 210 matching lines @@
           arrTimeDiff = frameSizeSampl + kSamplesIn25msec;
         }
 
         /* don't allow it to be less than frame rate - 10 ms */
         if (arrTimeDiff < frameSizeSampl - FRAMESAMPLES_10ms) {
           arrTimeDiff = frameSizeSampl - FRAMESAMPLES_10ms;
         }
 
         /* compute inverse receiving rate for last packet, in Q19 */
         numBytesInv = (uint16_t) WebRtcSpl_DivW32W16(
-            524288 + ((pksize + HEADER_SIZE) >> 1),
+            (int32_t)(524288 + ((pksize + HEADER_SIZE) >> 1)),
             (int16_t)(pksize + HEADER_SIZE));
 
         /* 8389 is  ~ 1/128000 in Q30 */
         byteSecondsPerBit = (uint32_t)(arrTimeDiff * 8389);
 
         /* get upper N bits */
         tempUpper = WEBRTC_SPL_RSHIFT_U32(byteSecondsPerBit, 15);
 
         /* get lower 15 bits */
         tempLower = byteSecondsPerBit & 0x00007FFF;
@@ skipping 636 matching lines @@
       /*s2nr = -1*(a_60 << 10) + ((b_60 * bottle_neck) >> 10);*/
       s2nr = -22500 + (int16_t)(500 * bottle_neck >> 10);
       break;
     default:
       s2nr = -1; /* Error */
   }
 
   return s2nr; //return in Q10
 
 }