Match existing type usage better.

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 356 matching lines @@
         }
 
         /* 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),
-            pksize + HEADER_SIZE);
+            (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 629 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
 
 }