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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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13
modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc
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@ -33,10 +33,10 @@ bool AudioDecoderIlbcImpl::HasDecodePlc() const {
}
int AudioDecoderIlbcImpl::DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
RTC_DCHECK_EQ(sample_rate_hz, 8000);
int16_t temp_type = 1; // Default is speech.
int ret = WebRtcIlbcfix_Decode(dec_state_, encoded, encoded_len, decoded,
@ -86,10 +86,9 @@ std::vector<AudioDecoder::ParseResult> AudioDecoderIlbcImpl::ParsePayload(
} else {
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 += bytes_per_frame,
timestamp_offset += timestamps_per_frame) {
timestamp_offset += timestamps_per_frame) {
std::unique_ptr<EncodedAudioFrame> frame(new LegacyEncodedAudioFrame(
this, rtc::Buffer(payload.data() + byte_offset, bytes_per_frame)));
results.emplace_back(timestamp + timestamp_offset, 0, std::move(frame));