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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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26
modules/audio_coding/acm2/audio_coding_module.cc
View File

@ -323,9 +323,10 @@ int DownMix(const AudioFrame& frame,
if (!frame.muted()) {
const int16_t* frame_data = frame.data();
for (size_t n = 0; n < frame.samples_per_channel_; ++n) {
out_buff[n] = static_cast<int16_t>(
(static_cast<int32_t>(frame_data[2 * n]) +
static_cast<int32_t>(frame_data[2 * n + 1])) >> 1);
out_buff[n] =
static_cast<int16_t>((static_cast<int32_t>(frame_data[2 * n]) +
static_cast<int32_t>(frame_data[2 * n + 1])) >>
1);
}
} else {
std::fill(out_buff, out_buff + frame.samples_per_channel_, 0);
@ -472,7 +473,7 @@ int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
if (!HaveValidEncoder("Process"))
return -1;
if(!first_frame_) {
if (!first_frame_) {
RTC_DCHECK(IsNewerTimestamp(input_data.input_timestamp, last_timestamp_))
<< "Time should not move backwards";
}
@ -493,9 +494,10 @@ int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
// Clear the buffer before reuse - encoded data will get appended.
encode_buffer_.Clear();
encoded_info = encoder_stack_->Encode(
rtp_timestamp, rtc::ArrayView<const int16_t>(
input_data.audio, input_data.audio_channel *
input_data.length_per_channel),
rtp_timestamp,
rtc::ArrayView<const int16_t>(
input_data.audio,
input_data.audio_channel * input_data.length_per_channel),
&encode_buffer_);
bitrate_logger_.MaybeLog(encoder_stack_->GetTargetBitrate() / 1000);
@ -767,7 +769,6 @@ int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
expected_in_ts_ = in_frame.timestamp_;
}
if (!down_mix && !resample) {
// No pre-processing is required.
if (expected_in_ts_ == expected_codec_ts_) {
@ -793,8 +794,8 @@ int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
if (down_mix) {
// If a resampling is required the output of a down-mix is written into a
// local buffer, otherwise, it will be written to the output frame.
int16_t* dest_ptr_audio = resample ?
audio : preprocess_frame_.mutable_data();
int16_t* dest_ptr_audio =
resample ? audio : preprocess_frame_.mutable_data();
if (DownMix(in_frame, WEBRTC_10MS_PCM_AUDIO, dest_ptr_audio) < 0)
return -1;
preprocess_frame_.num_channels_ = 1;
@ -912,7 +913,8 @@ int AudioCodingModuleImpl::SetVAD(bool enable_dtx,
}
// Get VAD/DTX settings.
int AudioCodingModuleImpl::VAD(bool* dtx_enabled, bool* vad_enabled,
int AudioCodingModuleImpl::VAD(bool* dtx_enabled,
bool* vad_enabled,
ACMVADMode* mode) const {
rtc::CritScope lock(&acm_crit_sect_);
const auto* sp = encoder_factory_->codec_manager.GetStackParams();
@ -1229,7 +1231,7 @@ int AudioCodingModuleImpl::LeastRequiredDelayMs() const {
}
void AudioCodingModuleImpl::GetDecodingCallStatistics(
AudioDecodingCallStats* call_stats) const {
AudioDecodingCallStats* call_stats) const {
receiver_.GetDecodingCallStatistics(call_stats);
}