Reland "Upconvert various types to int.", isac portion.

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 27 matching lines @@
 using namespace std;
 
 int main(int argc, char* argv[]) {
   char inname[100], outname[100], bottleneck_file[100], vadfile[100];
   FILE* inp, *outp, * f_bn = NULL, * vadp = NULL, *bandwidthp;
   int framecnt, endfile;
 
   int i, errtype, VADusage = 0, packetLossPercent = 0;
   int16_t CodingMode;
   int32_t bottleneck = 0;
-  int16_t framesize = 30; /* ms */
+  int framesize = 30; /* ms */
   int cur_framesmpls, err;
 
   /* Runtime statistics */
   double starttime, runtime, length_file;
 
   int16_t stream_len = 0;
-  int16_t declen = 0, declenTC = 0;
+  int 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;
@@ skipping 867 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);
 }