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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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7
modules/audio_coding/neteq/tools/input_audio_file.cc
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@ -20,7 +20,9 @@ InputAudioFile::InputAudioFile(const std::string file_name, bool loop_at_end)
fp_ = fopen(file_name.c_str(), "rb");
}
InputAudioFile::~InputAudioFile() { fclose(fp_); }
InputAudioFile::~InputAudioFile() {
fclose(fp_);
}
bool InputAudioFile::Read(size_t samples, int16_t* destination) {
if (!fp_) {
@ -73,7 +75,8 @@ bool InputAudioFile::Seek(int samples) {
return true;
}
void InputAudioFile::DuplicateInterleaved(const int16_t* source, size_t samples,
void InputAudioFile::DuplicateInterleaved(const int16_t* source,
size_t samples,
size_t channels,
int16_t* destination) {
// Start from the end of |source| and |destination|, and work towards the