webrtc/modules/audio_processing: Use RTC_DCHECK() instead of assert()

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 <assert.h>
 #include <math.h>
 #include <stdlib.h>
 #include <string.h>
 
+#include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_processing/aec/aec_core.h"
 
 namespace webrtc {
 
 enum { kEstimateLengthFrames = 400 };
 
 typedef struct {
   float buffer[kResamplerBufferSize];
   float position;
 
@@ skipping 35 matching lines @@
                               size_t size,
                               float skew,
                               float* outspeech,
                               size_t* size_out) {
   AecResampler* obj = static_cast<AecResampler*>(resampInst);
 
   float* y;
   float be, tnew;
   size_t tn, mm;
 
-  assert(size <= 2 * FRAME_LEN);
-  assert(resampInst != NULL);
-  assert(inspeech != NULL);
-  assert(outspeech != NULL);
-  assert(size_out != NULL);
+  RTC_DCHECK_LE(size, 2u * FRAME_LEN);
+  RTC_DCHECK(resampInst);
+  RTC_DCHECK(inspeech);
+  RTC_DCHECK(outspeech);
+  RTC_DCHECK(size_out);
 
   // Add new frame data in lookahead
   memcpy(&obj->buffer[FRAME_LEN + kResamplingDelay], inspeech,
          size * sizeof(inspeech[0]));
 
   // Sample rate ratio
   be = 1 + skew;
 
   // Loop over input frame
   mm = 0;
@@ skipping 64 matching lines @@
   for (i = 0; i < size; i++) {
     if ((rawSkew[i] < absLimitOuter && rawSkew[i] > -absLimitOuter)) {
       n++;
       rawAvg += rawSkew[i];
     }
   }
 
   if (n == 0) {
     return -1;
   }
-  assert(n > 0);
+  RTC_DCHECK_GT(n, 0);
   rawAvg /= n;
 
   for (i = 0; i < size; i++) {
     if ((rawSkew[i] < absLimitOuter && rawSkew[i] > -absLimitOuter)) {
       err = rawSkew[i] - rawAvg;
       rawAbsDev += err >= 0 ? err : -err;
     }
   }
-  assert(n > 0);
+  RTC_DCHECK_GT(n, 0);
   rawAbsDev /= n;
   upperLimit = static_cast<int>(rawAvg + 5 * rawAbsDev + 1);  // +1 for ceiling.
   lowerLimit = static_cast<int>(rawAvg - 5 * rawAbsDev - 1);  // -1 for floor.
 
   n = 0;
   for (i = 0; i < size; i++) {
     if ((rawSkew[i] < absLimitInner && rawSkew[i] > -absLimitInner) ||
         (rawSkew[i] < upperLimit && rawSkew[i] > lowerLimit)) {
       n++;
       cumSum += rawSkew[i];
       x += n;
       x2 += n * n;
       y += cumSum;
       xy += n * cumSum;
     }
   }
 
   if (n == 0) {
     return -1;
   }
-  assert(n > 0);
+  RTC_DCHECK_GT(n, 0);
   xAvg = x / n;
   denom = x2 - xAvg * x;
 
   if (denom != 0) {
     skew = (xy - xAvg * y) / denom;
   }
 
   *skewEst = skew;
   return 0;
 }
 }  // namespace webrtc