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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/sync_buffer.cc
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@ -27,7 +27,7 @@ void SyncBuffer::PushBack(const AudioMultiVector& append_this) {
next_index_ -= samples_added;
} else {
// This means that we are pushing out future data that was never used.
// assert(false);
// assert(false);
// TODO(hlundin): This assert must be disabled to support 60 ms frames.
// This should not happen even for 60 ms frames, but it does. Investigate
// why.
@ -75,9 +75,8 @@ void SyncBuffer::GetNextAudioInterleaved(size_t requested_len,
RTC_DCHECK(output);
const size_t samples_to_read = std::min(FutureLength(), requested_len);
output->ResetWithoutMuting();
const size_t tot_samples_read =
ReadInterleavedFromIndex(next_index_, samples_to_read,
output->mutable_data());
const size_t tot_samples_read = ReadInterleavedFromIndex(
next_index_, samples_to_read, output->mutable_data());
const size_t samples_read_per_channel = tot_samples_read / Channels();
next_index_ += samples_read_per_channel;
output->num_channels_ = Channels();