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

webrtc/modules/audio_processing/voice_detection_impl.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.
  */
 
@@ skipping 85 matching lines @@
 int VoiceDetectionImpl::set_stream_has_voice(bool has_voice) {
   rtc::CritScope cs(crit_);
   using_external_vad_ = true;
   stream_has_voice_ = has_voice;
   return AudioProcessing::kNoError;
 }
 
 bool VoiceDetectionImpl::stream_has_voice() const {
   rtc::CritScope cs(crit_);
   // TODO(ajm): enable this assertion?
-  //assert(using_external_vad_ || is_component_enabled());
+  //RTC_DCHECK(using_external_vad_ || is_component_enabled());
   return stream_has_voice_;
 }
 
 int VoiceDetectionImpl::set_likelihood(VoiceDetection::Likelihood likelihood) {
   rtc::CritScope cs(crit_);
   likelihood_ = likelihood;
   if (enabled_) {
     int mode = 2;
     switch (likelihood) {
       case VoiceDetection::kVeryLowLikelihood:
@@ skipping 29 matching lines @@
   frame_size_ms_ = size;
   Initialize(sample_rate_hz_);
   return AudioProcessing::kNoError;
 }
 
 int VoiceDetectionImpl::frame_size_ms() const {
   rtc::CritScope cs(crit_);
   return frame_size_ms_;
 }
 }  // namespace webrtc