Update includes for webrtc/{base => rtc_base} rename (1/3)

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.
  */
 
 // ReleaseTest-API.cpp : Defines the entry point for the console application.
 //
 
 #include <stdio.h>
 #include <stdlib.h>
 #include <string.h>
 #include <time.h>
 #include <ctype.h>
 #include <iostream>
 
 /* include API */
 #include "isac.h"
 #include "utility.h"
-#include "webrtc/base/format_macros.h"
+#include "webrtc/rtc_base/format_macros.h"
 
 /* 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) */
 
@@ skipping 905 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);
 }