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Reformat the WebRTC code base

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Running clang-format with chromium's style guide.

The goal is n-fold:
 * providing consistency and readability (that's what code guidelines are for)
 * preventing noise with presubmit checks and git cl format
 * building on the previous point: making it easier to automatically fix format issues
 * you name it

Please consider using git-hyper-blame to ignore this commit.

Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
This commit is contained in:
Yves Gerey
2018-06-19 15:03:05 +02:00
parent b602123a5a
commit 665174fdbb
1569 changed files with 30495 additions and 30309 deletions
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6
modules/audio_processing/audio_processing_impl_locking_unittest.cc
View File

@ -146,8 +146,7 @@ struct TestConfig {
// Create test config for the first processing API function set.
test_configs.push_back(test_config);
test_config.render_api_function =
RenderApiImpl::AnalyzeReverseStreamImpl;
test_config.render_api_function = RenderApiImpl::AnalyzeReverseStreamImpl;
test_config.capture_api_function = CaptureApiImpl::ProcessStreamImpl3;
test_configs.push_back(test_config);
}
@ -482,8 +481,7 @@ void PopulateAudioFrame(AudioFrame* frame,
for (size_t k = 0; k < frame->samples_per_channel_; k++) {
// Store random 16 bit number between -(amplitude+1) and
// amplitude.
frame_data[k * ch] =
rand_gen->RandInt(2 * amplitude + 1) - amplitude - 1;
frame_data[k * ch] = rand_gen->RandInt(2 * amplitude + 1) - amplitude - 1;
}
}
}