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_coding/neteq/expand.cc b/modules/audio_coding/neteq/expand.cc
index 73e8d07..5f671ad 100644
--- a/modules/audio_coding/neteq/expand.cc
+++ b/modules/audio_coding/neteq/expand.cc
@@ -14,7 +14,7 @@
#include <string.h> // memset
#include <algorithm> // min, max
-#include <limits> // numeric_limits<T>
+#include <limits> // numeric_limits<T>
#include "common_audio/signal_processing/include/signal_processing_library.h"
#include "modules/audio_coding/neteq/background_noise.h"
@@ -94,7 +94,6 @@
GenerateRandomVector(2, rand_length, random_vector);
}
-
// Generate signal.
UpdateLagIndex();
@@ -103,8 +102,8 @@
size_t expansion_vector_length = max_lag_ + overlap_length_;
size_t current_lag = expand_lags_[current_lag_index_];
// Copy lag+overlap data.
- size_t expansion_vector_position = expansion_vector_length - current_lag -
- overlap_length_;
+ size_t expansion_vector_position =
+ expansion_vector_length - current_lag - overlap_length_;
size_t temp_length = current_lag + overlap_length_;
for (size_t channel_ix = 0; channel_ix < num_channels_; ++channel_ix) {
ChannelParameters& parameters = channel_parameters_[channel_ix];
@@ -175,8 +174,10 @@
// Do overlap add between new vector and overlap.
(*sync_buffer_)[channel_ix][start_ix + i] =
(((*sync_buffer_)[channel_ix][start_ix + i] * muting_window) +
- (((parameters.mute_factor * voiced_vector_storage[i]) >> 14) *
- unmuting_window) + 16384) >> 15;
+ (((parameters.mute_factor * voiced_vector_storage[i]) >> 14) *
+ unmuting_window) +
+ 16384) >>
+ 15;
muting_window += muting_window_increment;
unmuting_window += unmuting_window_increment;
}
@@ -188,10 +189,10 @@
// parameters.expand_vector0 and parameters.expand_vector1 no longer
// match with expand_lags_, causing invalid reads and writes. Is it a good
// idea to enable this again, and solve the vector size problem?
-// max_lag_ = fs_mult * 120;
-// expand_lags_[0] = fs_mult * 120;
-// expand_lags_[1] = fs_mult * 120;
-// expand_lags_[2] = fs_mult * 120;
+ // max_lag_ = fs_mult * 120;
+ // expand_lags_[0] = fs_mult * 120;
+ // expand_lags_[1] = fs_mult * 120;
+ // expand_lags_[2] = fs_mult * 120;
}
// Unvoiced part.
@@ -204,8 +205,7 @@
}
WebRtcSpl_AffineTransformVector(scaled_random_vector, random_vector,
parameters.ar_gain, add_constant,
- parameters.ar_gain_scale,
- current_lag);
+ parameters.ar_gain_scale, current_lag);
WebRtcSpl_FilterARFastQ12(scaled_random_vector, unvoiced_vector,
parameters.ar_filter, kUnvoicedLpcOrder + 1,
current_lag);
@@ -230,8 +230,9 @@
// Create combined signal by shifting in more and more of unvoiced part.
temp_shift = 8 - temp_shift; // = getbits(mix_factor_increment).
- size_t temp_length = (parameters.current_voice_mix_factor -
- parameters.voice_mix_factor) >> temp_shift;
+ size_t temp_length =
+ (parameters.current_voice_mix_factor - parameters.voice_mix_factor) >>
+ temp_shift;
temp_length = std::min(temp_length, current_lag);
DspHelper::CrossFade(voiced_vector, unvoiced_vector, temp_length,
¶meters.current_voice_mix_factor,
@@ -266,9 +267,8 @@
// Mute segment according to slope value.
if ((consecutive_expands_ != 0) || !parameters.onset) {
// Mute to the previous level, then continue with the muting.
- WebRtcSpl_AffineTransformVector(temp_data, temp_data,
- parameters.mute_factor, 8192,
- 14, current_lag);
+ WebRtcSpl_AffineTransformVector(
+ temp_data, temp_data, parameters.mute_factor, 8192, 14, current_lag);
if (!stop_muting_) {
DspHelper::MuteSignal(temp_data, parameters.mute_slope, current_lag);
@@ -276,8 +276,8 @@
// Shift by 6 to go from Q20 to Q14.
// TODO(hlundin): Adding 8192 before shifting 6 steps seems wrong.
// Legacy.
- int16_t gain = static_cast<int16_t>(16384 -
- (((current_lag * parameters.mute_slope) + 8192) >> 6));
+ int16_t gain = static_cast<int16_t>(
+ 16384 - (((current_lag * parameters.mute_slope) + 8192) >> 6));
gain = ((gain * parameters.mute_factor) + 8192) >> 14;
// Guard against getting stuck with very small (but sometimes audible)
@@ -291,12 +291,9 @@
}
// Background noise part.
- GenerateBackgroundNoise(random_vector,
- channel_ix,
- channel_parameters_[channel_ix].mute_slope,
- TooManyExpands(),
- current_lag,
- unvoiced_array_memory);
+ GenerateBackgroundNoise(
+ random_vector, channel_ix, channel_parameters_[channel_ix].mute_slope,
+ TooManyExpands(), current_lag, unvoiced_array_memory);
// Add background noise to the combined voiced-unvoiced signal.
for (size_t i = 0; i < current_lag; i++) {
@@ -311,8 +308,9 @@
}
// Increase call number and cap it.
- consecutive_expands_ = consecutive_expands_ >= kMaxConsecutiveExpands ?
- kMaxConsecutiveExpands : consecutive_expands_ + 1;
+ consecutive_expands_ = consecutive_expands_ >= kMaxConsecutiveExpands
+ ? kMaxConsecutiveExpands
+ : consecutive_expands_ + 1;
expand_duration_samples_ += output->Size();
// Clamp the duration counter at 2 seconds.
expand_duration_samples_ = std::min(expand_duration_samples_,
@@ -329,7 +327,7 @@
}
void Expand::SetParametersForMergeAfterExpand() {
- current_lag_index_ = -1; /* out of the 3 possible ones */
+ current_lag_index_ = -1; /* out of the 3 possible ones */
lag_index_direction_ = 1; /* make sure we get the "optimal" lag */
stop_muting_ = true;
}
@@ -357,7 +355,7 @@
consecutive_expands_ = 0;
for (size_t ix = 0; ix < num_channels_; ++ix) {
channel_parameters_[ix].current_voice_mix_factor = 16384; // 1.0 in Q14.
- channel_parameters_[ix].mute_factor = 16384; // 1.0 in Q14.
+ channel_parameters_[ix].mute_factor = 16384; // 1.0 in Q14.
// Start with 0 gain for background noise.
background_noise_->SetMuteFactor(ix, 0);
}
@@ -420,10 +418,10 @@
// Calculate distortion around the |kNumCorrelationCandidates| best lags.
int distortion_scale = 0;
for (size_t i = 0; i < kNumCorrelationCandidates; i++) {
- size_t min_index = std::max(fs_mult_20,
- best_correlation_index[i] - fs_mult_4);
- size_t max_index = std::min(fs_mult_120 - 1,
- best_correlation_index[i] + fs_mult_4);
+ size_t min_index =
+ std::max(fs_mult_20, best_correlation_index[i] - fs_mult_4);
+ size_t max_index =
+ std::min(fs_mult_120 - 1, best_correlation_index[i] + fs_mult_4);
best_distortion_index[i] = DspHelper::MinDistortion(
&(audio_history[signal_length - fs_mult_dist_len]), min_index,
max_index, fs_mult_dist_len, &best_distortion_w32[i]);
@@ -459,23 +457,23 @@
// Calculate the exact best correlation in the range between
// |correlation_lag| and |distortion_lag|.
- correlation_length =
- std::max(std::min(distortion_lag + 10, fs_mult_120),
- static_cast<size_t>(60 * fs_mult));
+ correlation_length = std::max(std::min(distortion_lag + 10, fs_mult_120),
+ static_cast<size_t>(60 * fs_mult));
size_t start_index = std::min(distortion_lag, correlation_lag);
size_t correlation_lags = static_cast<size_t>(
- WEBRTC_SPL_ABS_W16((distortion_lag-correlation_lag)) + 1);
+ WEBRTC_SPL_ABS_W16((distortion_lag - correlation_lag)) + 1);
assert(correlation_lags <= static_cast<size_t>(99 * fs_mult + 1));
for (size_t channel_ix = 0; channel_ix < num_channels_; ++channel_ix) {
ChannelParameters& parameters = channel_parameters_[channel_ix];
// Calculate suitable scaling.
int16_t signal_max = WebRtcSpl_MaxAbsValueW16(
- &audio_history[signal_length - correlation_length - start_index
- - correlation_lags],
- correlation_length + start_index + correlation_lags - 1);
- int correlation_scale = (31 - WebRtcSpl_NormW32(signal_max * signal_max)) +
+ &audio_history[signal_length - correlation_length - start_index -
+ correlation_lags],
+ correlation_length + start_index + correlation_lags - 1);
+ int correlation_scale =
+ (31 - WebRtcSpl_NormW32(signal_max * signal_max)) +
(31 - WebRtcSpl_NormW32(static_cast<int32_t>(correlation_length))) - 31;
correlation_scale = std::max(0, correlation_scale);
@@ -520,8 +518,8 @@
// Calculate max_correlation / sqrt(energy1 * energy2) in Q14.
int cc_shift = 14 - (energy1_scale + energy2_scale) / 2;
max_correlation = WEBRTC_SPL_SHIFT_W32(max_correlation, cc_shift);
- corr_coefficient = WebRtcSpl_DivW32W16(max_correlation,
- sqrt_energy_product);
+ corr_coefficient =
+ WebRtcSpl_DivW32W16(max_correlation, sqrt_energy_product);
// Cap at 1.0 in Q14.
corr_coefficient = std::min(16384, corr_coefficient);
} else {
@@ -547,9 +545,9 @@
int32_t scaled_energy2 = std::max(16 - WebRtcSpl_NormW32(energy2), 0);
int32_t scaled_energy1 = scaled_energy2 - 13;
// Calculate scaled_energy1 / scaled_energy2 in Q13.
- int32_t energy_ratio = WebRtcSpl_DivW32W16(
- WEBRTC_SPL_SHIFT_W32(energy1, -scaled_energy1),
- static_cast<int16_t>(energy2 >> scaled_energy2));
+ int32_t energy_ratio =
+ WebRtcSpl_DivW32W16(WEBRTC_SPL_SHIFT_W32(energy1, -scaled_energy1),
+ static_cast<int16_t>(energy2 >> scaled_energy2));
// Calculate sqrt ratio in Q13 (sqrt of en1/en2 in Q26).
amplitude_ratio =
static_cast<int16_t>(WebRtcSpl_SqrtFloor(energy_ratio << 13));
@@ -558,16 +556,13 @@
parameters.expand_vector0.PushBack(vector1, expansion_length);
parameters.expand_vector1.Clear();
if (parameters.expand_vector1.Size() < expansion_length) {
- parameters.expand_vector1.Extend(
- expansion_length - parameters.expand_vector1.Size());
+ parameters.expand_vector1.Extend(expansion_length -
+ parameters.expand_vector1.Size());
}
std::unique_ptr<int16_t[]> temp_1(new int16_t[expansion_length]);
- WebRtcSpl_AffineTransformVector(temp_1.get(),
- const_cast<int16_t*>(vector2),
- amplitude_ratio,
- 4096,
- 13,
- expansion_length);
+ WebRtcSpl_AffineTransformVector(
+ temp_1.get(), const_cast<int16_t*>(vector2), amplitude_ratio, 4096,
+ 13, expansion_length);
parameters.expand_vector1.OverwriteAt(temp_1.get(), expansion_length, 0);
} else {
// Energy change constraint not fulfilled. Only use last vector.
@@ -606,11 +601,11 @@
// Calculate the LPC and the gain of the filters.
// Calculate kUnvoicedLpcOrder + 1 lags of the auto-correlation function.
- size_t temp_index = signal_length - fs_mult_lpc_analysis_len -
- kUnvoicedLpcOrder;
+ size_t temp_index =
+ signal_length - fs_mult_lpc_analysis_len - kUnvoicedLpcOrder;
// Copy signal to temporary vector to be able to pad with leading zeros.
- int16_t* temp_signal = new int16_t[fs_mult_lpc_analysis_len
- + kUnvoicedLpcOrder];
+ int16_t* temp_signal =
+ new int16_t[fs_mult_lpc_analysis_len + kUnvoicedLpcOrder];
memset(temp_signal, 0,
sizeof(int16_t) * (fs_mult_lpc_analysis_len + kUnvoicedLpcOrder));
memcpy(&temp_signal[kUnvoicedLpcOrder],
@@ -619,16 +614,15 @@
CrossCorrelationWithAutoShift(
&temp_signal[kUnvoicedLpcOrder], &temp_signal[kUnvoicedLpcOrder],
fs_mult_lpc_analysis_len, kUnvoicedLpcOrder + 1, -1, auto_correlation);
- delete [] temp_signal;
+ delete[] temp_signal;
// Verify that variance is positive.
if (auto_correlation[0] > 0) {
// Estimate AR filter parameters using Levinson-Durbin algorithm;
// kUnvoicedLpcOrder + 1 filter coefficients.
- int16_t stability = WebRtcSpl_LevinsonDurbin(auto_correlation,
- parameters.ar_filter,
- reflection_coeff,
- kUnvoicedLpcOrder);
+ int16_t stability =
+ WebRtcSpl_LevinsonDurbin(auto_correlation, parameters.ar_filter,
+ reflection_coeff, kUnvoicedLpcOrder);
// Keep filter parameters only if filter is stable.
if (stability != 1) {
@@ -671,10 +665,8 @@
&(audio_history[signal_length - 128 - kUnvoicedLpcOrder]),
sizeof(int16_t) * kUnvoicedLpcOrder);
WebRtcSpl_FilterMAFastQ12(&audio_history[signal_length - 128],
- unvoiced_vector,
- parameters.ar_filter,
- kUnvoicedLpcOrder + 1,
- 128);
+ unvoiced_vector, parameters.ar_filter,
+ kUnvoicedLpcOrder + 1, 128);
const int unvoiced_max_abs = [&] {
const int16_t max_abs = WebRtcSpl_MaxAbsValueW16(unvoiced_vector, 128);
// Since WebRtcSpl_MaxAbsValueW16 returns 2^15 - 1 when the input contains
@@ -689,10 +681,8 @@
int unvoiced_prescale =
std::max(0, 2 * WebRtcSpl_GetSizeInBits(unvoiced_max_abs) - 24);
- int32_t unvoiced_energy = WebRtcSpl_DotProductWithScale(unvoiced_vector,
- unvoiced_vector,
- 128,
- unvoiced_prescale);
+ int32_t unvoiced_energy = WebRtcSpl_DotProductWithScale(
+ unvoiced_vector, unvoiced_vector, 128, unvoiced_prescale);
// Normalize |unvoiced_energy| to 28 or 29 bits to preserve sqrt() accuracy.
int16_t unvoiced_scale = WebRtcSpl_NormW32(unvoiced_energy) - 3;
@@ -703,8 +693,8 @@
unvoiced_energy = WEBRTC_SPL_SHIFT_W32(unvoiced_energy, unvoiced_scale);
int16_t unvoiced_gain =
static_cast<int16_t>(WebRtcSpl_SqrtFloor(unvoiced_energy));
- parameters.ar_gain_scale = 13
- + (unvoiced_scale + 7 - unvoiced_prescale) / 2;
+ parameters.ar_gain_scale =
+ 13 + (unvoiced_scale + 7 - unvoiced_prescale) / 2;
parameters.ar_gain = unvoiced_gain;
// Calculate voice_mix_factor from corr_coefficient.
@@ -717,17 +707,17 @@
int16_t x1, x2, x3;
// |corr_coefficient| is in Q14.
x1 = static_cast<int16_t>(corr_coefficient);
- x2 = (x1 * x1) >> 14; // Shift 14 to keep result in Q14.
+ x2 = (x1 * x1) >> 14; // Shift 14 to keep result in Q14.
x3 = (x1 * x2) >> 14;
- static const int kCoefficients[4] = { -5179, 19931, -16422, 5776 };
+ static const int kCoefficients[4] = {-5179, 19931, -16422, 5776};
int32_t temp_sum = kCoefficients[0] * 16384;
temp_sum += kCoefficients[1] * x1;
temp_sum += kCoefficients[2] * x2;
temp_sum += kCoefficients[3] * x3;
parameters.voice_mix_factor =
static_cast<int16_t>(std::min(temp_sum / 4096, 16384));
- parameters.voice_mix_factor = std::max(parameters.voice_mix_factor,
- static_cast<int16_t>(0));
+ parameters.voice_mix_factor =
+ std::max(parameters.voice_mix_factor, static_cast<int16_t>(0));
} else {
parameters.voice_mix_factor = 0;
}
@@ -816,8 +806,8 @@
static const size_t kNumCorrelationLags = 54;
static const size_t kCorrelationLength = 60;
// Downsample to 4 kHz sample rate.
- static const size_t kDownsampledLength = kCorrelationStartLag
- + kNumCorrelationLags + kCorrelationLength;
+ static const size_t kDownsampledLength =
+ kCorrelationStartLag + kNumCorrelationLags + kCorrelationLength;
int16_t downsampled_input[kDownsampledLength];
static const size_t kFilterDelay = 0;
WebRtcSpl_DownsampleFast(
@@ -827,8 +817,8 @@
downsampling_factor, kFilterDelay);
// Normalize |downsampled_input| to using all 16 bits.
- int16_t max_value = WebRtcSpl_MaxAbsValueW16(downsampled_input,
- kDownsampledLength);
+ int16_t max_value =
+ WebRtcSpl_MaxAbsValueW16(downsampled_input, kDownsampledLength);
int16_t norm_shift = 16 - WebRtcSpl_NormW32(max_value);
WebRtcSpl_VectorBitShiftW16(downsampled_input, kDownsampledLength,
downsampled_input, norm_shift);
@@ -836,13 +826,13 @@
int32_t correlation[kNumCorrelationLags];
CrossCorrelationWithAutoShift(
&downsampled_input[kDownsampledLength - kCorrelationLength],
- &downsampled_input[kDownsampledLength - kCorrelationLength
- - kCorrelationStartLag],
+ &downsampled_input[kDownsampledLength - kCorrelationLength -
+ kCorrelationStartLag],
kCorrelationLength, kNumCorrelationLags, -1, correlation);
// Normalize and move data from 32-bit to 16-bit vector.
- int32_t max_correlation = WebRtcSpl_MaxAbsValueW32(correlation,
- kNumCorrelationLags);
+ int32_t max_correlation =
+ WebRtcSpl_MaxAbsValueW32(correlation, kNumCorrelationLags);
int16_t norm_shift2 = static_cast<int16_t>(
std::max(18 - WebRtcSpl_NormW32(max_correlation), 0));
WebRtcSpl_VectorBitShiftW32ToW16(output, kNumCorrelationLags, correlation,
@@ -894,19 +884,15 @@
// Scale random vector to correct energy level.
WebRtcSpl_AffineTransformVector(
- scaled_random_vector, random_vector,
- background_noise_->Scale(channel), dc_offset,
- background_noise_->ScaleShift(channel),
- num_noise_samples);
+ scaled_random_vector, random_vector, background_noise_->Scale(channel),
+ dc_offset, background_noise_->ScaleShift(channel), num_noise_samples);
WebRtcSpl_FilterARFastQ12(scaled_random_vector, noise_samples,
background_noise_->Filter(channel),
- kNoiseLpcOrder + 1,
- num_noise_samples);
+ kNoiseLpcOrder + 1, num_noise_samples);
background_noise_->SetFilterState(
- channel,
- &(noise_samples[num_noise_samples - kNoiseLpcOrder]),
+ channel, &(noise_samples[num_noise_samples - kNoiseLpcOrder]),
kNoiseLpcOrder);
// Unmute the background noise.