Add RTC_ prefix to (D)CHECKs and related macros.

webrtc/modules/audio_processing/vad/voice_activity_detector.cc

 /*
  *  Copyright (c) 2015 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 19 matching lines @@
     : last_voice_probability_(kDefaultVoiceValue),
       standalone_vad_(StandaloneVad::Create()) {
 }
 
 // Because ISAC has a different chunk length, it updates
 // |chunkwise_voice_probabilities_| and |chunkwise_rms_| when there is new data.
 // Otherwise it clears them.
 void VoiceActivityDetector::ProcessChunk(const int16_t* audio,
                                          size_t length,
                                          int sample_rate_hz) {
-  DCHECK_EQ(static_cast<int>(length), sample_rate_hz / 100);
-  DCHECK_LE(length, kMaxLength);
+  RTC_DCHECK_EQ(static_cast<int>(length), sample_rate_hz / 100);
+  RTC_DCHECK_LE(length, kMaxLength);
   // Resample to the required rate.
   const int16_t* resampled_ptr = audio;
   if (sample_rate_hz != kSampleRateHz) {
-    CHECK_EQ(
+    RTC_CHECK_EQ(
         resampler_.ResetIfNeeded(sample_rate_hz, kSampleRateHz, kNumChannels),
         0);
     resampler_.Push(audio, length, resampled_, kLength10Ms, length);
     resampled_ptr = resampled_;
   }
-  DCHECK_EQ(length, kLength10Ms);
+  RTC_DCHECK_EQ(length, kLength10Ms);
 
   // Each chunk needs to be passed into |standalone_vad_|, because internally it
   // buffers the audio and processes it all at once when GetActivity() is
   // called.
-  CHECK_EQ(standalone_vad_->AddAudio(resampled_ptr, length), 0);
+  RTC_CHECK_EQ(standalone_vad_->AddAudio(resampled_ptr, length), 0);
 
   audio_processing_.ExtractFeatures(resampled_ptr, length, &features_);
 
   chunkwise_voice_probabilities_.resize(features_.num_frames);
   chunkwise_rms_.resize(features_.num_frames);
   std::copy(features_.rms, features_.rms + chunkwise_rms_.size(),
             chunkwise_rms_.begin());
   if (features_.num_frames > 0) {
     if (features_.silence) {
       // The other features are invalid, so set the voice probabilities to an
       // arbitrary low value.
       std::fill(chunkwise_voice_probabilities_.begin(),
                 chunkwise_voice_probabilities_.end(), kLowProbability);
     } else {
       std::fill(chunkwise_voice_probabilities_.begin(),
                 chunkwise_voice_probabilities_.end(), kNeutralProbability);
-      CHECK_GE(
+      RTC_CHECK_GE(
           standalone_vad_->GetActivity(&chunkwise_voice_probabilities_[0],
                                        chunkwise_voice_probabilities_.size()),
           0);
-      CHECK_GE(pitch_based_vad_.VoicingProbability(
-                   features_, &chunkwise_voice_probabilities_[0]),
-               0);
+      RTC_CHECK_GE(pitch_based_vad_.VoicingProbability(
+                       features_, &chunkwise_voice_probabilities_[0]),
+                   0);
     }
     last_voice_probability_ = chunkwise_voice_probabilities_.back();
   }
 }
 
 }  // namespace webrtc