Match existing type usage better.

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 34 matching lines @@
   int i, errtype, VADusage = 0, packetLossPercent = 0;
   int16_t CodingMode;
   int32_t bottleneck = 0;
   int16_t framesize = 30; /* ms */
   int cur_framesmpls, err;
 
   /* Runtime statistics */
   double starttime, runtime, length_file;
 
   int16_t stream_len = 0;
-  int16_t declen = 0, lostFrame = 0, declenTC = 0;
+  int16_t declen = 0, declenTC = 0;
+  bool lostFrame = false;
 
   int16_t shortdata[SWBFRAMESAMPLES_10ms];
   int16_t vaddata[SWBFRAMESAMPLES_10ms * 3];
   int16_t decoded[MAX_FRAMESAMPLES << 1];
   int16_t decodedTC[MAX_FRAMESAMPLES << 1];
   uint16_t streamdata[500];
   int16_t speechType[1];
   int16_t rateBPS = 0;
   int16_t fixedFL = 0;
   int16_t payloadSize = 0;
@@ skipping 623 matching lines @@
       maxStreamLen30 =
           (stream_len > maxStreamLen30) ? stream_len : maxStreamLen30;
     } else {
       maxStreamLen60 =
           (stream_len > maxStreamLen60) ? stream_len : maxStreamLen60;
     }
 
     if (!lostFrame) {
       lostFrame = ((rand() % 100) < packetLossPercent);
     } else {
-      lostFrame = 0;
+      lostFrame = false;
     }
 
     // RED.
     if (lostFrame) {
       stream_len = WebRtcIsac_GetRedPayload(
           ISAC_main_inst, reinterpret_cast<uint8_t*>(streamdata));
 
       if (doTransCoding) {
         streamLenTransCoding = WebRtcIsac_GetNewBitStream(
             ISAC_main_inst, bnIdxTC, jitterInfoTC, rateTransCoding,
@@ skipping 222 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);
 }