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

webrtc/modules/audio_processing/rms_level.cc

 /*
  *  Copyright (c) 2014 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 "webrtc/modules/audio_processing/rms_level.h"
 
-#include <assert.h>
 #include <math.h>
 
+#include "webrtc/base/checks.h"
+
 namespace webrtc {
 
 static const float kMaxSquaredLevel = 32768 * 32768;
 
 RMSLevel::RMSLevel()
     : sum_square_(0),
       sample_count_(0) {}
 
 RMSLevel::~RMSLevel() {}
 
@@ skipping 16 matching lines @@
 int RMSLevel::RMS() {
   if (sample_count_ == 0 || sum_square_ == 0) {
     Reset();
     return kMinLevel;
   }
 
   // Normalize by the max level.
   float rms = sum_square_ / (sample_count_ * kMaxSquaredLevel);
   // 20log_10(x^0.5) = 10log_10(x)
   rms = 10 * log10(rms);
-  assert(rms <= 0);
+  RTC_DCHECK_LE(rms, 0);
   if (rms < -kMinLevel)
     rms = -kMinLevel;
 
   rms = -rms;
   Reset();
   return static_cast<int>(rms + 0.5);
 }
 
 }  // namespace webrtc