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author | henrikg <henrikg@webrtc.org> | 2015-09-17 00:24:34 -0700 |
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committer | Commit bot <commit-bot@chromium.org> | 2015-09-17 07:24:51 +0000 |
commit | 91d6edef35e7275879c30ce16ecb8b6dc73c6e4a (patch) | |
tree | 313b64a3d88422a53f5b091f770a2d33dce823ce /webrtc/modules/audio_coding/codecs/audio_encoder.cc | |
parent | c0ac6cad00f51b99fafaa7c19e2e63b3b542735b (diff) | |
download | webrtc-91d6edef35e7275879c30ce16ecb8b6dc73c6e4a.tar.gz |
Add RTC_ prefix to (D)CHECKs and related macros.
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}
Diffstat (limited to 'webrtc/modules/audio_coding/codecs/audio_encoder.cc')
-rw-r--r-- | webrtc/modules/audio_coding/codecs/audio_encoder.cc | 6 |
1 files changed, 3 insertions, 3 deletions
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.cc b/webrtc/modules/audio_coding/codecs/audio_encoder.cc index c0c20bec4d..6d763005ac 100644 --- a/webrtc/modules/audio_coding/codecs/audio_encoder.cc +++ b/webrtc/modules/audio_coding/codecs/audio_encoder.cc @@ -26,11 +26,11 @@ AudioEncoder::EncodedInfo AudioEncoder::Encode(uint32_t rtp_timestamp, size_t num_samples_per_channel, size_t max_encoded_bytes, uint8_t* encoded) { - CHECK_EQ(num_samples_per_channel, - static_cast<size_t>(SampleRateHz() / 100)); + RTC_CHECK_EQ(num_samples_per_channel, + static_cast<size_t>(SampleRateHz() / 100)); EncodedInfo info = EncodeInternal(rtp_timestamp, audio, max_encoded_bytes, encoded); - CHECK_LE(info.encoded_bytes, max_encoded_bytes); + RTC_CHECK_LE(info.encoded_bytes, max_encoded_bytes); return info; } |