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

webrtc/modules/audio_processing/aec/aec_resampler.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.
  */
 
 /* Resamples a signal to an arbitrary rate. Used by the AEC to compensate for
  * clock skew by resampling the farend signal.
  */
 
 #include "webrtc/modules/audio_processing/aec/aec_resampler.h"
 
 #include <math.h>
 #include <stdlib.h>
 #include <string.h>
 
-#include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_processing/aec/aec_core.h"
+#include "webrtc/rtc_base/checks.h"
 
 namespace webrtc {
 
 enum { kEstimateLengthFrames = 400 };
 
 typedef struct {
   float buffer[kResamplerBufferSize];
   float position;
 
   int deviceSampleRateHz;
@@ skipping 165 matching lines @@
   denom = x2 - xAvg * x;
 
   if (denom != 0) {
     skew = (xy - xAvg * y) / denom;
   }
 
   *skewEst = skew;
   return 0;
 }
 }  // namespace webrtc