Prepare to convert various types to size_t.

webrtc/modules/audio_coding/codecs/g722/test/testG722.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 65 matching lines @@
         printf("framelength  :    Framelength in samples.\n\n");
         printf("infile       :    Normal speech input file\n\n");
         printf("outbitfile   :    Bitstream output file\n\n");
         printf("outspeechfile:    Speech output file\n\n");
         exit(0);
 
     }
 
     /* Get frame length */
     framelength = atoi(argv[1]);
+    if (framelength < 0) {
+        printf("  G.722: Invalid framelength %d.\n", framelength);
+        exit(1);
+    }
 
     /* Get Input and Output files */
     sscanf(argv[2], "%s", inname);
     sscanf(argv[3], "%s", outbit);
     sscanf(argv[4], "%s", outname);
 
     if ((inp = fopen(inname,"rb")) == NULL) {
         printf("  G.722: Cannot read file %s.\n", inname);
         exit(1);
     }
@@ skipping 58 matching lines @@
     printf("\n\nLength of speech file: %.1f s\n", length_file);
     printf("Time to run G.722:      %.2f s (%.2f %% of realtime)\n\n", runtime, (100*runtime/length_file));
     printf("---------------------END----------------------\n");
 
     fclose(inp);
     fclose(outbitp);
     fclose(outp);
 
     return 0;
 }