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

webrtc/modules/audio_coding/neteq/neteq_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 822 matching lines @@
   *num_channels = static_cast<int>(sync_buffer_->Channels());
   LOG(LS_VERBOSE) << "Sync buffer (" << *num_channels << " channel(s)):" <<
       " insert " << algorithm_buffer_->Size() << " samples, extract " <<
       samples_from_sync << " samples";
   if (sync_buffer_->FutureLength() < expand_->overlap_length()) {
     // The sync buffer should always contain |overlap_length| samples, but now
     // too many samples have been extracted. Reinstall the |overlap_length|
     // lookahead by moving the index.
     const size_t missing_lookahead_samples =
         expand_->overlap_length() - sync_buffer_->FutureLength();
-    DCHECK_GE(sync_buffer_->next_index(), missing_lookahead_samples);
+    RTC_DCHECK_GE(sync_buffer_->next_index(), missing_lookahead_samples);
     sync_buffer_->set_next_index(sync_buffer_->next_index() -
                                  missing_lookahead_samples);
   }
   if (samples_from_sync != output_size_samples_) {
     LOG(LS_ERROR) << "samples_from_sync (" << samples_from_sync
                   << ") != output_size_samples_ (" << output_size_samples_
                   << ")";
     // TODO(minyue): treatment of under-run, filling zeros
     memset(output, 0, num_output_samples * sizeof(int16_t));
     *samples_per_channel = output_size_samples_;
     return kSampleUnderrun;
   }
   *samples_per_channel = output_size_samples_;
 
   // Should always have overlap samples left in the |sync_buffer_|.
-  DCHECK_GE(sync_buffer_->FutureLength(), expand_->overlap_length());
+  RTC_DCHECK_GE(sync_buffer_->FutureLength(), expand_->overlap_length());
 
   if (play_dtmf) {
     return_value = DtmfOverdub(dtmf_event, sync_buffer_->Channels(), output);
   }
 
   // Update the background noise parameters if last operation wrote data
   // straight from the decoder to the |sync_buffer_|. That is, none of the
   // operations that modify the signal can be followed by a parameter update.
   if ((last_mode_ == kModeNormal) ||
       (last_mode_ == kModeAccelerateFail) ||
@@ skipping 1123 matching lines @@
 
 void NetEqImpl::CreateDecisionLogic() {
   decision_logic_.reset(DecisionLogic::Create(fs_hz_, output_size_samples_,
                                               playout_mode_,
                                               decoder_database_.get(),
                                               *packet_buffer_.get(),
                                               delay_manager_.get(),
                                               buffer_level_filter_.get()));
 }
 }  // namespace webrtc