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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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89
modules/audio_processing/audio_buffer.cc
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@ -49,20 +49,20 @@ AudioBuffer::AudioBuffer(size_t input_num_frames,
size_t process_num_frames,
size_t num_process_channels,
size_t output_num_frames)
: input_num_frames_(input_num_frames),
num_input_channels_(num_input_channels),
proc_num_frames_(process_num_frames),
num_proc_channels_(num_process_channels),
output_num_frames_(output_num_frames),
num_channels_(num_process_channels),
num_bands_(NumBandsFromSamplesPerChannel(proc_num_frames_)),
num_split_frames_(rtc::CheckedDivExact(proc_num_frames_, num_bands_)),
mixed_low_pass_valid_(false),
reference_copied_(false),
activity_(AudioFrame::kVadUnknown),
keyboard_data_(NULL),
data_(new IFChannelBuffer(proc_num_frames_, num_proc_channels_)),
output_buffer_(new IFChannelBuffer(output_num_frames_, num_channels_)) {
: input_num_frames_(input_num_frames),
num_input_channels_(num_input_channels),
proc_num_frames_(process_num_frames),
num_proc_channels_(num_process_channels),
output_num_frames_(output_num_frames),
num_channels_(num_process_channels),
num_bands_(NumBandsFromSamplesPerChannel(proc_num_frames_)),
num_split_frames_(rtc::CheckedDivExact(proc_num_frames_, num_bands_)),
mixed_low_pass_valid_(false),
reference_copied_(false),
activity_(AudioFrame::kVadUnknown),
keyboard_data_(NULL),
data_(new IFChannelBuffer(proc_num_frames_, num_proc_channels_)),
output_buffer_(new IFChannelBuffer(output_num_frames_, num_channels_)) {
RTC_DCHECK_GT(input_num_frames_, 0);
RTC_DCHECK_GT(proc_num_frames_, 0);
RTC_DCHECK_GT(output_num_frames_, 0);
@ -73,8 +73,8 @@ AudioBuffer::AudioBuffer(size_t input_num_frames,
if (input_num_frames_ != proc_num_frames_ ||
output_num_frames_ != proc_num_frames_) {
// Create an intermediate buffer for resampling.
process_buffer_.reset(new ChannelBuffer<float>(proc_num_frames_,
num_proc_channels_));
process_buffer_.reset(
new ChannelBuffer<float>(proc_num_frames_, num_proc_channels_));
if (input_num_frames_ != proc_num_frames_) {
for (size_t i = 0; i < num_proc_channels_; ++i) {
@ -92,12 +92,10 @@ AudioBuffer::AudioBuffer(size_t input_num_frames,
}
if (num_bands_ > 1) {
split_data_.reset(new IFChannelBuffer(proc_num_frames_,
num_proc_channels_,
num_bands_));
splitting_filter_.reset(new SplittingFilter(num_proc_channels_,
num_bands_,
proc_num_frames_));
split_data_.reset(
new IFChannelBuffer(proc_num_frames_, num_proc_channels_, num_bands_));
splitting_filter_.reset(
new SplittingFilter(num_proc_channels_, num_bands_, proc_num_frames_));
}
}
@ -132,8 +130,7 @@ void AudioBuffer::CopyFrom(const float* const* data,
// Resample.
if (input_num_frames_ != proc_num_frames_) {
for (size_t i = 0; i < num_proc_channels_; ++i) {
input_resamplers_[i]->Resample(data_ptr[i],
input_num_frames_,
input_resamplers_[i]->Resample(data_ptr[i], input_num_frames_,
process_buffer_->channels()[i],
proc_num_frames_);
}
@ -142,8 +139,7 @@ void AudioBuffer::CopyFrom(const float* const* data,
// Convert to the S16 range.
for (size_t i = 0; i < num_proc_channels_; ++i) {
FloatToFloatS16(data_ptr[i],
proc_num_frames_,
FloatToFloatS16(data_ptr[i], proc_num_frames_,
data_->fbuf()->channels()[i]);
}
}
@ -161,17 +157,14 @@ void AudioBuffer::CopyTo(const StreamConfig& stream_config,
data_ptr = process_buffer_->channels();
}
for (size_t i = 0; i < num_channels_; ++i) {
FloatS16ToFloat(data_->fbuf()->channels()[i],
proc_num_frames_,
FloatS16ToFloat(data_->fbuf()->channels()[i], proc_num_frames_,
data_ptr[i]);
}
// Resample.
if (output_num_frames_ != proc_num_frames_) {
for (size_t i = 0; i < num_channels_; ++i) {
output_resamplers_[i]->Resample(data_ptr[i],
proc_num_frames_,
data[i],
output_resamplers_[i]->Resample(data_ptr[i], proc_num_frames_, data[i],
output_num_frames_);
}
}
@ -204,16 +197,14 @@ int16_t* const* AudioBuffer::channels() {
}
const int16_t* const* AudioBuffer::split_bands_const(size_t channel) const {
return split_data_.get() ?
split_data_->ibuf_const()->bands(channel) :
data_->ibuf_const()->bands(channel);
return split_data_.get() ? split_data_->ibuf_const()->bands(channel)
: data_->ibuf_const()->bands(channel);
}
int16_t* const* AudioBuffer::split_bands(size_t channel) {
mixed_low_pass_valid_ = false;
return split_data_.get() ?
split_data_->ibuf()->bands(channel) :
data_->ibuf()->bands(channel);
return split_data_.get() ? split_data_->ibuf()->bands(channel)
: data_->ibuf()->bands(channel);
}
const int16_t* const* AudioBuffer::split_channels_const(Band band) const {
@ -261,16 +252,14 @@ float* const* AudioBuffer::channels_f() {
}
const float* const* AudioBuffer::split_bands_const_f(size_t channel) const {
return split_data_.get() ?
split_data_->fbuf_const()->bands(channel) :
data_->fbuf_const()->bands(channel);
return split_data_.get() ? split_data_->fbuf_const()->bands(channel)
: data_->fbuf_const()->bands(channel);
}
float* const* AudioBuffer::split_bands_f(size_t channel) {
mixed_low_pass_valid_ = false;
return split_data_.get() ?
split_data_->fbuf()->bands(channel) :
data_->fbuf()->bands(channel);
return split_data_.get() ? split_data_->fbuf()->bands(channel)
: data_->fbuf()->bands(channel);
}
const float* const* AudioBuffer::split_channels_const_f(Band band) const {
@ -401,19 +390,16 @@ void AudioBuffer::DeinterleaveFrom(AudioFrame* frame) {
num_input_channels_, deinterleaved[0]);
} else {
RTC_DCHECK_EQ(num_proc_channels_, num_input_channels_);
Deinterleave(frame->data(),
input_num_frames_,
num_proc_channels_,
Deinterleave(frame->data(), input_num_frames_, num_proc_channels_,
deinterleaved);
}
// Resample.
if (input_num_frames_ != proc_num_frames_) {
for (size_t i = 0; i < num_proc_channels_; ++i) {
input_resamplers_[i]->Resample(input_buffer_->fbuf_const()->channels()[i],
input_num_frames_,
data_->fbuf()->channels()[i],
proc_num_frames_);
input_resamplers_[i]->Resample(
input_buffer_->fbuf_const()->channels()[i], input_num_frames_,
data_->fbuf()->channels()[i], proc_num_frames_);
}
}
}
@ -453,8 +439,7 @@ void AudioBuffer::CopyLowPassToReference() {
if (!low_pass_reference_channels_.get() ||
low_pass_reference_channels_->num_channels() != num_channels_) {
low_pass_reference_channels_.reset(
new ChannelBuffer<int16_t>(num_split_frames_,
num_proc_channels_));
new ChannelBuffer<int16_t>(num_split_frames_, num_proc_channels_));
}
for (size_t i = 0; i < num_proc_channels_; i++) {
memcpy(low_pass_reference_channels_->channels()[i],