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

webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.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.
  */
 
 #include "webrtc/modules/audio_coding/codecs/opus/interface/audio_decoder_opus.h"
 
 #include "webrtc/base/checks.h"
 
 namespace webrtc {
 
 AudioDecoderOpus::AudioDecoderOpus(size_t num_channels)
     : channels_(num_channels) {
-  DCHECK(num_channels == 1 || num_channels == 2);
+  RTC_DCHECK(num_channels == 1 || num_channels == 2);
   WebRtcOpus_DecoderCreate(&dec_state_, static_cast<int>(channels_));
   WebRtcOpus_DecoderInit(dec_state_);
 }
 
 AudioDecoderOpus::~AudioDecoderOpus() {
   WebRtcOpus_DecoderFree(dec_state_);
 }
 
 int AudioDecoderOpus::DecodeInternal(const uint8_t* encoded,
                                      size_t encoded_len,
                                      int sample_rate_hz,
                                      int16_t* decoded,
                                      SpeechType* speech_type) {
-  DCHECK_EQ(sample_rate_hz, 48000);
+  RTC_DCHECK_EQ(sample_rate_hz, 48000);
   int16_t temp_type = 1;  // Default is speech.
   int ret =
       WebRtcOpus_Decode(dec_state_, encoded, encoded_len, decoded, &temp_type);
   if (ret > 0)
     ret *= static_cast<int>(channels_);  // Return total number of samples.
   *speech_type = ConvertSpeechType(temp_type);
   return ret;
 }
 
 int AudioDecoderOpus::DecodeRedundantInternal(const uint8_t* encoded,
                                               size_t encoded_len,
                                               int sample_rate_hz,
                                               int16_t* decoded,
                                               SpeechType* speech_type) {
   if (!PacketHasFec(encoded, encoded_len)) {
     // This packet is a RED packet.
     return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
                           speech_type);
   }
 
-  DCHECK_EQ(sample_rate_hz, 48000);
+  RTC_DCHECK_EQ(sample_rate_hz, 48000);
   int16_t temp_type = 1;  // Default is speech.
   int ret = WebRtcOpus_DecodeFec(dec_state_, encoded, encoded_len, decoded,
                                  &temp_type);
   if (ret > 0)
     ret *= static_cast<int>(channels_);  // Return total number of samples.
   *speech_type = ConvertSpeechType(temp_type);
   return ret;
 }
 
 void AudioDecoderOpus::Reset() {
@@ skipping 20 matching lines @@
   int fec;
   fec = WebRtcOpus_PacketHasFec(encoded, encoded_len);
   return (fec == 1);
 }
 
 size_t AudioDecoderOpus::Channels() const {
   return channels_;
 }
 
 }  // namespace webrtc