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

webrtc/modules/audio_processing/agc/agc.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 36 matching lines @@
       ++num_clipped;
   }
   return 1.0f * num_clipped / length;
 }
 
 int Agc::Process(const int16_t* audio, size_t length, int sample_rate_hz) {
   vad_.ProcessChunk(audio, length, sample_rate_hz);
   const std::vector<double>& rms = vad_.chunkwise_rms();
   const std::vector<double>& probabilities =
       vad_.chunkwise_voice_probabilities();
-  DCHECK_EQ(rms.size(), probabilities.size());
+  RTC_DCHECK_EQ(rms.size(), probabilities.size());
   for (size_t i = 0; i < rms.size(); ++i) {
     histogram_->Update(rms[i], probabilities[i]);
   }
   return 0;
 }
 
 bool Agc::GetRmsErrorDb(int* error) {
   if (!error) {
     assert(false);
     return false;
@@ skipping 24 matching lines @@
   // limits. The upper limit should be chosen such that the risk of clipping is
   // low. The lower limit should not result in a too quiet signal.
   if (level >= 0 || level <= -100)
     return -1;
   target_level_dbfs_ = level;
   target_level_loudness_ = Dbfs2Loudness(level);
   return 0;
 }
 
 }  // namespace webrtc