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

webrtc/modules/audio_coding/neteq/audio_decoder_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 30 matching lines @@
 }
 size_t AudioDecoderPcmU::Channels() const {
   return 1;
 }
 
 int AudioDecoderPcmU::DecodeInternal(const uint8_t* encoded,
                                      size_t encoded_len,
                                      int sample_rate_hz,
                                      int16_t* decoded,
                                      SpeechType* speech_type) {
-  DCHECK_EQ(sample_rate_hz, 8000);
+  RTC_DCHECK_EQ(sample_rate_hz, 8000);
   int16_t temp_type = 1;  // Default is speech.
   size_t ret = WebRtcG711_DecodeU(encoded, encoded_len, decoded, &temp_type);
   *speech_type = ConvertSpeechType(temp_type);
   return static_cast<int>(ret);
 }
 
 int AudioDecoderPcmU::PacketDuration(const uint8_t* encoded,
                                      size_t encoded_len) const {
   // One encoded byte per sample per channel.
   return static_cast<int>(encoded_len / Channels());
 }
 
 size_t AudioDecoderPcmUMultiCh::Channels() const {
   return channels_;
 }
 
 // PCMa
 
 void AudioDecoderPcmA::Reset() {
 }
 size_t AudioDecoderPcmA::Channels() const {
   return 1;
 }
 
 int AudioDecoderPcmA::DecodeInternal(const uint8_t* encoded,
                                      size_t encoded_len,
                                      int sample_rate_hz,
                                      int16_t* decoded,
                                      SpeechType* speech_type) {
-  DCHECK_EQ(sample_rate_hz, 8000);
+  RTC_DCHECK_EQ(sample_rate_hz, 8000);
   int16_t temp_type = 1;  // Default is speech.
   size_t ret = WebRtcG711_DecodeA(encoded, encoded_len, decoded, &temp_type);
   *speech_type = ConvertSpeechType(temp_type);
   return static_cast<int>(ret);
 }
 
 int AudioDecoderPcmA::PacketDuration(const uint8_t* encoded,
                                      size_t encoded_len) const {
   // One encoded byte per sample per channel.
   return static_cast<int>(encoded_len / Channels());
@@ skipping 16 matching lines @@
 
 bool AudioDecoderG722::HasDecodePlc() const {
   return false;
 }
 
 int AudioDecoderG722::DecodeInternal(const uint8_t* encoded,
                                      size_t encoded_len,
                                      int sample_rate_hz,
                                      int16_t* decoded,
                                      SpeechType* speech_type) {
-  DCHECK_EQ(sample_rate_hz, 16000);
+  RTC_DCHECK_EQ(sample_rate_hz, 16000);
   int16_t temp_type = 1;  // Default is speech.
   size_t ret =
       WebRtcG722_Decode(dec_state_, encoded, encoded_len, decoded, &temp_type);
   *speech_type = ConvertSpeechType(temp_type);
   return static_cast<int>(ret);
 }
 
 void AudioDecoderG722::Reset() {
   WebRtcG722_DecoderInit(dec_state_);
 }
@@ skipping 18 matching lines @@
 AudioDecoderG722Stereo::~AudioDecoderG722Stereo() {
   WebRtcG722_FreeDecoder(dec_state_left_);
   WebRtcG722_FreeDecoder(dec_state_right_);
 }
 
 int AudioDecoderG722Stereo::DecodeInternal(const uint8_t* encoded,
                                            size_t encoded_len,
                                            int sample_rate_hz,
                                            int16_t* decoded,
                                            SpeechType* speech_type) {
-  DCHECK_EQ(sample_rate_hz, 16000);
+  RTC_DCHECK_EQ(sample_rate_hz, 16000);
   int16_t temp_type = 1;  // Default is speech.
   // De-interleave the bit-stream into two separate payloads.
   uint8_t* encoded_deinterleaved = new uint8_t[encoded_len];
   SplitStereoPacket(encoded, encoded_len, encoded_deinterleaved);
   // Decode left and right.
   size_t decoded_len = WebRtcG722_Decode(dec_state_left_, encoded_deinterleaved,
                                          encoded_len / 2, decoded, &temp_type);
   size_t ret = WebRtcG722_Decode(
       dec_state_right_, &encoded_deinterleaved[encoded_len / 2],
       encoded_len / 2, &decoded[decoded_len], &temp_type);
@@ skipping 43 matching lines @@
   for (size_t i = 0; i < encoded_len / 2; i++) {
     uint8_t right_byte = encoded_deinterleaved[i + 1];
     memmove(&encoded_deinterleaved[i + 1], &encoded_deinterleaved[i + 2],
             encoded_len - i - 2);
     encoded_deinterleaved[encoded_len - 1] = right_byte;
   }
 }
 #endif
 
 AudioDecoderCng::AudioDecoderCng() {
-  CHECK_EQ(0, WebRtcCng_CreateDec(&dec_state_));
+  RTC_CHECK_EQ(0, WebRtcCng_CreateDec(&dec_state_));
   WebRtcCng_InitDec(dec_state_);
 }
 
 AudioDecoderCng::~AudioDecoderCng() {
   WebRtcCng_FreeDec(dec_state_);
 }
 
 void AudioDecoderCng::Reset() {
   WebRtcCng_InitDec(dec_state_);
 }
@@ skipping 184 matching lines @@
     case kDecoderRED:
     case kDecoderAVT:
     case kDecoderArbitrary:
     default: {
       return NULL;
     }
   }
 }
 
 }  // namespace webrtc