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

webrtc/modules/audio_processing/logging/aec_logging_file_handling.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 16 matching lines @@
   if (*wav_file) {
     if (rtc_WavSampleRate(*wav_file) == sample_rate)
       return;
     rtc_WavClose(*wav_file);
   }
   char filename[64];
   int written = rtc::sprintfn(filename, sizeof(filename), "%s%d-%d.wav", name,
                               instance_index, process_rate);
 
   // Ensure there was no buffer output error.
-  DCHECK_GE(written, 0);
+  RTC_DCHECK_GE(written, 0);
   // Ensure that the buffer size was sufficient.
-  DCHECK_LT(static_cast<size_t>(written), sizeof(filename));
+  RTC_DCHECK_LT(static_cast<size_t>(written), sizeof(filename));
 
   *wav_file = rtc_WavOpen(filename, sample_rate, 1);
 }
 
 void WebRtcAec_RawFileOpen(const char* name, int instance_index, FILE** file) {
   char filename[64];
   int written = rtc::sprintfn(filename, sizeof(filename), "%s_%d.dat", name,
                               instance_index);
 
   // Ensure there was no buffer output error.
-  DCHECK_GE(written, 0);
+  RTC_DCHECK_GE(written, 0);
   // Ensure that the buffer size was sufficient.
-  DCHECK_LT(static_cast<size_t>(written), sizeof(filename));
+  RTC_DCHECK_LT(static_cast<size_t>(written), sizeof(filename));
 
   *file = fopen(filename, "wb");
 }
 
 #endif  // WEBRTC_AEC_DEBUG_DUMP