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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
Download Patch File Download Diff File Expand all files Collapse all files

21
modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
View File

@ -42,8 +42,8 @@ AudioEncoderPcm::AudioEncoderPcm(const Config& config, int sample_rate_hz)
payload_type_(config.payload_type),
num_10ms_frames_per_packet_(
static_cast<size_t>(config.frame_size_ms / 10)),
full_frame_samples_(
config.num_channels * config.frame_size_ms * sample_rate_hz / 1000),
full_frame_samples_(config.num_channels * config.frame_size_ms *
sample_rate_hz / 1000),
first_timestamp_in_buffer_(0) {
RTC_CHECK_GT(sample_rate_hz, 0) << "Sample rate must be larger than 0 Hz";
RTC_CHECK_EQ(config.frame_size_ms % 10, 0)
@ -70,8 +70,8 @@ size_t AudioEncoderPcm::Max10MsFramesInAPacket() const {
}
int AudioEncoderPcm::GetTargetBitrate() const {
return static_cast<int>(
8 * BytesPerSample() * SampleRateHz() * NumChannels());
return static_cast<int>(8 * BytesPerSample() * SampleRateHz() *
NumChannels());
}
AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeImpl(
@ -89,13 +89,12 @@ AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeImpl(
EncodedInfo info;
info.encoded_timestamp = first_timestamp_in_buffer_;
info.payload_type = payload_type_;
info.encoded_bytes =
encoded->AppendData(full_frame_samples_ * BytesPerSample(),
[&] (rtc::ArrayView<uint8_t> encoded) {
return EncodeCall(&speech_buffer_[0],
full_frame_samples_,
encoded.data());
});
info.encoded_bytes = encoded->AppendData(
full_frame_samples_ * BytesPerSample(),
[&](rtc::ArrayView<uint8_t> encoded) {
return EncodeCall(&speech_buffer_[0], full_frame_samples_,
encoded.data());
});
speech_buffer_.clear();
info.encoder_type = GetCodecType();
return info;