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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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21
modules/audio_coding/codecs/opus/audio_encoder_opus.cc
View File

@ -613,20 +613,17 @@ AudioEncoder::EncodedInfo AudioEncoderOpusImpl::EncodeImpl(
const size_t max_encoded_bytes = SufficientOutputBufferSize();
EncodedInfo info;
info.encoded_bytes =
encoded->AppendData(
max_encoded_bytes, [&] (rtc::ArrayView<uint8_t> encoded) {
int status = WebRtcOpus_Encode(
inst_, &input_buffer_[0],
rtc::CheckedDivExact(input_buffer_.size(),
config_.num_channels),
rtc::saturated_cast<int16_t>(max_encoded_bytes),
encoded.data());
info.encoded_bytes = encoded->AppendData(
max_encoded_bytes, [&](rtc::ArrayView<uint8_t> encoded) {
int status = WebRtcOpus_Encode(
inst_, &input_buffer_[0],
rtc::CheckedDivExact(input_buffer_.size(), config_.num_channels),
rtc::saturated_cast<int16_t>(max_encoded_bytes), encoded.data());
RTC_CHECK_GE(status, 0); // Fails only if fed invalid data.
RTC_CHECK_GE(status, 0); // Fails only if fed invalid data.
return static_cast<size_t>(status);
});
return static_cast<size_t>(status);
});
input_buffer_.clear();
bool dtx_frame = (info.encoded_bytes <= 2);