Reformat the WebRTC code base
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}diff --git a/modules/audio_processing/audio_buffer.cc b/modules/audio_processing/audio_buffer.cc
index 16f1174..f163f5a 100644
--- a/modules/audio_processing/audio_buffer.cc
+++ b/modules/audio_processing/audio_buffer.cc
@@ -49,20 +49,20 @@
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 @@
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 @@
}
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 @@
// 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 @@
// 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 @@
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 @@
}
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 @@
}
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 @@
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 @@
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],