Prepare to convert various types to size_t.

webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc

 /*
  *  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 167 matching lines @@
   printf("iSAC version %s \n\n", version_number);
 
   /* Loop over all command line arguments */
   CodingMode = 0;
   testNum = 0;
   useAssign = 0;
   // logFile = NULL;
   char transCodingFileName[500];
   int16_t totFileLoop = 0;
   int16_t numFileLoop = 0;
-  for (i = 1; i < argc - 2; i++) {
+  for (i = 1; i + 2 < argc; i++) {
     if (!strcmp("-LOOP", argv[i])) {
       i++;
       totFileLoop = (int16_t)atol(argv[i]);
       if (totFileLoop <= 0) {
         fprintf(stderr, "Invalid number of runs for the given input file, %d.",
                 totFileLoop);
         exit(0);
       }
     }
 
@@ skipping 734 matching lines @@
     fprintf(stderr, "  transcoding rate %.0f kbps",
             (double)numTransCodingBytes * 8.0 * (sampFreqKHz) / totalsmpls);
   }
 
   fclose(inp);
   fclose(outp);
   WebRtcIsac_Free(ISAC_main_inst);
 
   exit(0);
 }