Match existing type usage better.

webrtc/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.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 50 matching lines @@
                       int bottleneck,           /* excluding headers; bits/s */
                       BottleNeckModel* BN_data) {
   const int HeaderSize = 35;
   int HeaderRate;
 
   HeaderRate = HeaderSize * 8 * FS / current_framesamples; /* bits/s */
 
   /* everything in samples */
   BN_data->sample_count = BN_data->sample_count + current_framesamples;
 
-  BN_data->arrival_time +=
-      ((packet_size + HeaderSize) * 8 * FS) / (bottleneck + HeaderRate);
+  BN_data->arrival_time += (uint32_t)
+      (((packet_size + HeaderSize) * 8 * FS) / (bottleneck + HeaderRate));
   BN_data->send_time += current_framesamples;
 
   if (BN_data->arrival_time < BN_data->sample_count)
     BN_data->arrival_time = BN_data->sample_count;
 
   BN_data->rtp_number++;
 }
 
 int main(int argc, char* argv[]) {
   char inname[50], outname[50], bottleneck_file[50], bitfilename[60],
@@ skipping 527 matching lines @@
   fclose(outp);
 
   WebRtcIsac_Free(ISAC_main_inst);
   WebRtcIsacfix_Free(ISACFIX_main_inst);
 
   // fclose(histfile);
   // fclose(ratefile);
 
   return 0;
 }