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

webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc

Index: webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
index e78fc04452d4b4cc8b4741cfe119b2c8aa20bea6..7151ab01a9d5d9e84c2f0049a87aa01e45e50f26 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
@@ -16,7 +16,7 @@ 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_);
 }
@@ -30,7 +30,7 @@ int AudioDecoderOpus::DecodeInternal(const uint8_t* encoded,
                                      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);
@@ -51,7 +51,7 @@ int AudioDecoderOpus::DecodeRedundantInternal(const uint8_t* encoded,
                           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);