Prepare to convert various types to size_t.

webrtc/modules/audio_coding/codecs/g711/test/testG711.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 68 matching lines @@
 
   }
 
   /* Get version and print */
   WebRtcG711_Version(versionNumber, 40);
 
   printf("-----------------------------------\n");
   printf("G.711 version: %s\n\n", versionNumber);
   /* Get frame length */
   framelength = atoi(argv[1]);
+  if (framelength < 0) {
+    printf("  G.711: Invalid framelength %d.\n", framelength);
+    exit(1);
+  }
 
   /* Get compression law */
   strcpy(law, argv[2]);
 
   /* Get Input and Output files */
   sscanf(argv[3], "%s", inname);
   sscanf(argv[4], "%s", outname);
   if (argc == 6) {
     sscanf(argv[5], "%s", bitname);
     if ((bitp = fopen(bitname, "wb")) == NULL) {
@@ skipping 71 matching lines @@
   printf("Time to run G.711:      %.2f s (%.2f %% of realtime)\n\n",
          runtime,
          (100 * runtime / length_file));
   printf("---------------------END----------------------\n");
 
   fclose(inp);
   fclose(outp);
 
   return 0;
 }