Revert of With these changes, WebRtc x86 will build Visual Studio 2015.

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 15 matching lines @@
 /* Defines */
 #define SEED_FILE "randseed.txt" /* Used when running decoder on garbage data */
 #define MAX_FRAMESAMPLES 960     /* max number of samples per frame
                                     (= 60 ms frame & 16 kHz) or
                                     (= 30 ms frame & 32 kHz) */
 #define FRAMESAMPLES_10ms 160 /* number of samples per 10ms frame */
 #define SWBFRAMESAMPLES_10ms 320
 //#define FS 16000 /* sampling frequency (Hz) */
 
 #ifdef WIN32
-#ifndef CLOCKS_PER_SEC
 #define CLOCKS_PER_SEC 1000 /* Runtime statistics */
 #endif
-#endif
 
 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;
 
   size_t i;
   int errtype, VADusage = 0, packetLossPercent = 0;
@@ skipping 890 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);
 }