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

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We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.

Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.

BUG=chromium:468375
NOTRY=true

Review URL: https://codereview.webrtc.org/1335923002

Cr-Commit-Position: refs/heads/master@{#9964}
This commit is contained in:
henrikg
2015-09-17 00:24:34 -07:00
committed by Commit bot
parent c0ac6cad00
commit 91d6edef35
232 changed files with 1665 additions and 1646 deletions
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10
webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
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@ -48,7 +48,7 @@ int AudioDecoderPcmU::DecodeInternal(const uint8_t* encoded,
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);
@ -78,7 +78,7 @@ int AudioDecoderPcmA::DecodeInternal(const uint8_t* encoded,
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);
@ -115,7 +115,7 @@ int AudioDecoderG722::DecodeInternal(const uint8_t* encoded,
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);
@ -154,7 +154,7 @@ int AudioDecoderG722Stereo::DecodeInternal(const uint8_t* encoded,
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];
@ -218,7 +218,7 @@ void AudioDecoderG722Stereo::SplitStereoPacket(const uint8_t* encoded,
#endif
AudioDecoderCng::AudioDecoderCng() {
CHECK_EQ(0, WebRtcCng_CreateDec(&dec_state_));
RTC_CHECK_EQ(0, WebRtcCng_CreateDec(&dec_state_));
WebRtcCng_InitDec(dec_state_);
}