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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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5
modules/audio_coding/codecs/legacy_encoded_audio_frame.cc
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@ -68,10 +68,9 @@ std::vector<AudioDecoder::ParseResult> LegacyEncodedAudioFrame::SplitBySamples(
split_size_bytes * timestamps_per_ms / bytes_per_ms);
size_t byte_offset;
uint32_t timestamp_offset;
for (byte_offset = 0, timestamp_offset = 0;
byte_offset < payload.size();
for (byte_offset = 0, timestamp_offset = 0; byte_offset < payload.size();
byte_offset += split_size_bytes,
timestamp_offset += timestamps_per_chunk) {
timestamp_offset += timestamps_per_chunk) {
split_size_bytes =
std::min(split_size_bytes, payload.size() - byte_offset);
rtc::Buffer new_payload(payload.data() + byte_offset, split_size_bytes);