Replace calls to assert() with RTC_DCHECK_*() in .c code

webrtc/common_audio/signal_processing/filter_ar_fast_q12.c

 /*
  *  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.
  */
-#include <assert.h>
 
+#include "webrtc/base/checks.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 
 // TODO(bjornv): Change the return type to report errors.
 
 void WebRtcSpl_FilterARFastQ12(const int16_t* data_in,
                                int16_t* data_out,
                                const int16_t* __restrict coefficients,
                                size_t coefficients_length,
                                size_t data_length) {
   size_t i = 0;
   size_t j = 0;
 
-  assert(data_length > 0);
-  assert(coefficients_length > 1);
+  RTC_DCHECK_GT(data_length, 0);
+  RTC_DCHECK_GT(coefficients_length, 1);
 
   for (i = 0; i < data_length; i++) {
     int32_t output = 0;
     int32_t sum = 0;
 
     for (j = coefficients_length - 1; j > 0; j--) {
       sum += coefficients[j] * data_out[i - j];
     }
 
     output = coefficients[0] * data_in[i];
     output -= sum;
 
     // Saturate and store the output.
     output = WEBRTC_SPL_SAT(134215679, output, -134217728);
     data_out[i] = (int16_t)((output + 2048) >> 12);
   }
 }