Reformat the WebRTC code base
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}
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@ -50,8 +50,7 @@ void PreemptiveExpand::SetParametersForPassiveSpeech(size_t len,
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// but we must ensure that best_correlation is not larger than the length of
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// the new data.
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// but we must ensure that best_correlation is not larger than the new data.
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*peak_index = std::min(*peak_index,
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len - old_data_length_per_channel_);
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*peak_index = std::min(*peak_index, len - old_data_length_per_channel_);
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}
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PreemptiveExpand::ReturnCodes PreemptiveExpand::CheckCriteriaAndStretch(
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@ -68,13 +67,13 @@ PreemptiveExpand::ReturnCodes PreemptiveExpand::CheckCriteriaAndStretch(
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// Check for strong correlation (>0.9 in Q14) and at least 15 ms new data,
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// or passive speech.
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if (((best_correlation > kCorrelationThreshold) &&
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(old_data_length_per_channel_ <= fs_mult_120)) ||
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(old_data_length_per_channel_ <= fs_mult_120)) ||
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!active_speech) {
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// Do accelerate operation by overlap add.
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// Set length of the first part, not to be modified.
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size_t unmodified_length = std::max(old_data_length_per_channel_,
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fs_mult_120);
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size_t unmodified_length =
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std::max(old_data_length_per_channel_, fs_mult_120);
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// Copy first part, including cross-fade region.
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output->PushBackInterleaved(
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input, (unmodified_length + peak_index) * num_channels_);
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@ -107,8 +106,8 @@ PreemptiveExpand* PreemptiveExpandFactory::Create(
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size_t num_channels,
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const BackgroundNoise& background_noise,
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size_t overlap_samples) const {
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return new PreemptiveExpand(
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sample_rate_hz, num_channels, background_noise, overlap_samples);
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return new PreemptiveExpand(sample_rate_hz, num_channels, background_noise,
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overlap_samples);
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}
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} // namespace webrtc
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