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/dsp_helper.cc b/modules/audio_coding/neteq/dsp_helper.cc
index 2a1d81b..05b0f70 100644
--- a/modules/audio_coding/neteq/dsp_helper.cc
+++ b/modules/audio_coding/neteq/dsp_helper.cc
@@ -21,41 +21,29 @@
// Table of constants used in method DspHelper::ParabolicFit().
const int16_t DspHelper::kParabolaCoefficients[17][3] = {
- { 120, 32, 64 },
- { 140, 44, 75 },
- { 150, 50, 80 },
- { 160, 57, 85 },
- { 180, 72, 96 },
- { 200, 89, 107 },
- { 210, 98, 112 },
- { 220, 108, 117 },
- { 240, 128, 128 },
- { 260, 150, 139 },
- { 270, 162, 144 },
- { 280, 174, 149 },
- { 300, 200, 160 },
- { 320, 228, 171 },
- { 330, 242, 176 },
- { 340, 257, 181 },
- { 360, 288, 192 } };
+ {120, 32, 64}, {140, 44, 75}, {150, 50, 80}, {160, 57, 85},
+ {180, 72, 96}, {200, 89, 107}, {210, 98, 112}, {220, 108, 117},
+ {240, 128, 128}, {260, 150, 139}, {270, 162, 144}, {280, 174, 149},
+ {300, 200, 160}, {320, 228, 171}, {330, 242, 176}, {340, 257, 181},
+ {360, 288, 192}};
// Filter coefficients used when downsampling from the indicated sample rates
// (8, 16, 32, 48 kHz) to 4 kHz. Coefficients are in Q12. The corresponding Q0
// values are provided in the comments before each array.
// Q0 values: {0.3, 0.4, 0.3}.
-const int16_t DspHelper::kDownsample8kHzTbl[3] = { 1229, 1638, 1229 };
+const int16_t DspHelper::kDownsample8kHzTbl[3] = {1229, 1638, 1229};
// Q0 values: {0.15, 0.2, 0.3, 0.2, 0.15}.
-const int16_t DspHelper::kDownsample16kHzTbl[5] = { 614, 819, 1229, 819, 614 };
+const int16_t DspHelper::kDownsample16kHzTbl[5] = {614, 819, 1229, 819, 614};
// Q0 values: {0.1425, 0.1251, 0.1525, 0.1628, 0.1525, 0.1251, 0.1425}.
-const int16_t DspHelper::kDownsample32kHzTbl[7] = {
- 584, 512, 625, 667, 625, 512, 584 };
+const int16_t DspHelper::kDownsample32kHzTbl[7] = {584, 512, 625, 667,
+ 625, 512, 584};
// Q0 values: {0.2487, 0.0952, 0.1042, 0.1074, 0.1042, 0.0952, 0.2487}.
-const int16_t DspHelper::kDownsample48kHzTbl[7] = {
- 1019, 390, 427, 440, 427, 390, 1019 };
+const int16_t DspHelper::kDownsample48kHzTbl[7] = {1019, 390, 427, 440,
+ 427, 390, 1019};
int DspHelper::RampSignal(const int16_t* input,
size_t length,
@@ -115,9 +103,12 @@
return end_factor;
}
-void DspHelper::PeakDetection(int16_t* data, size_t data_length,
- size_t num_peaks, int fs_mult,
- size_t* peak_index, int16_t* peak_value) {
+void DspHelper::PeakDetection(int16_t* data,
+ size_t data_length,
+ size_t num_peaks,
+ int fs_mult,
+ size_t* peak_index,
+ int16_t* peak_value) {
size_t min_index = 0;
size_t max_index = 0;
@@ -163,8 +154,10 @@
}
}
-void DspHelper::ParabolicFit(int16_t* signal_points, int fs_mult,
- size_t* peak_index, int16_t* peak_value) {
+void DspHelper::ParabolicFit(int16_t* signal_points,
+ int fs_mult,
+ size_t* peak_index,
+ int16_t* peak_value) {
uint16_t fit_index[13];
if (fs_mult == 1) {
fit_index[0] = 0;
@@ -204,23 +197,26 @@
// num = -3 * signal_points[0] + 4 * signal_points[1] - signal_points[2];
// den = signal_points[0] - 2 * signal_points[1] + signal_points[2];
- int32_t num = (signal_points[0] * -3) + (signal_points[1] * 4)
- - signal_points[2];
+ int32_t num =
+ (signal_points[0] * -3) + (signal_points[1] * 4) - signal_points[2];
int32_t den = signal_points[0] + (signal_points[1] * -2) + signal_points[2];
int32_t temp = num * 120;
int flag = 1;
- int16_t stp = kParabolaCoefficients[fit_index[fs_mult]][0]
- - kParabolaCoefficients[fit_index[fs_mult - 1]][0];
- int16_t strt = (kParabolaCoefficients[fit_index[fs_mult]][0]
- + kParabolaCoefficients[fit_index[fs_mult - 1]][0]) / 2;
+ int16_t stp = kParabolaCoefficients[fit_index[fs_mult]][0] -
+ kParabolaCoefficients[fit_index[fs_mult - 1]][0];
+ int16_t strt = (kParabolaCoefficients[fit_index[fs_mult]][0] +
+ kParabolaCoefficients[fit_index[fs_mult - 1]][0]) /
+ 2;
int16_t lmt;
if (temp < -den * strt) {
lmt = strt - stp;
while (flag) {
if ((flag == fs_mult) || (temp > -den * lmt)) {
- *peak_value = (den * kParabolaCoefficients[fit_index[fs_mult - flag]][1]
- + num * kParabolaCoefficients[fit_index[fs_mult - flag]][2]
- + signal_points[0] * 256) / 256;
+ *peak_value =
+ (den * kParabolaCoefficients[fit_index[fs_mult - flag]][1] +
+ num * kParabolaCoefficients[fit_index[fs_mult - flag]][2] +
+ signal_points[0] * 256) /
+ 256;
*peak_index = *peak_index * 2 * fs_mult - flag;
flag = 0;
} else {
@@ -233,9 +229,9 @@
while (flag) {
if ((flag == fs_mult) || (temp < -den * lmt)) {
int32_t temp_term_1 =
- den * kParabolaCoefficients[fit_index[fs_mult+flag]][1];
+ den * kParabolaCoefficients[fit_index[fs_mult + flag]][1];
int32_t temp_term_2 =
- num * kParabolaCoefficients[fit_index[fs_mult+flag]][2];
+ num * kParabolaCoefficients[fit_index[fs_mult + flag]][2];
int32_t temp_term_3 = signal_points[0] * 256;
*peak_value = (temp_term_1 + temp_term_2 + temp_term_3) / 256;
*peak_index = *peak_index * 2 * fs_mult + flag;
@@ -251,8 +247,10 @@
}
}
-size_t DspHelper::MinDistortion(const int16_t* signal, size_t min_lag,
- size_t max_lag, size_t length,
+size_t DspHelper::MinDistortion(const int16_t* signal,
+ size_t min_lag,
+ size_t max_lag,
+ size_t length,
int32_t* distortion_value) {
size_t best_index = 0;
int32_t min_distortion = WEBRTC_SPL_WORD32_MAX;
@@ -273,9 +271,12 @@
return best_index;
}
-void DspHelper::CrossFade(const int16_t* input1, const int16_t* input2,
- size_t length, int16_t* mix_factor,
- int16_t factor_decrement, int16_t* output) {
+void DspHelper::CrossFade(const int16_t* input1,
+ const int16_t* input2,
+ size_t length,
+ int16_t* mix_factor,
+ int16_t factor_decrement,
+ int16_t* output) {
int16_t factor = *mix_factor;
int16_t complement_factor = 16384 - factor;
for (size_t i = 0; i < length; i++) {
@@ -287,8 +288,10 @@
*mix_factor = factor;
}
-void DspHelper::UnmuteSignal(const int16_t* input, size_t length,
- int16_t* factor, int increment,
+void DspHelper::UnmuteSignal(const int16_t* input,
+ size_t length,
+ int16_t* factor,
+ int increment,
int16_t* output) {
uint16_t factor_16b = *factor;
int32_t factor_32b = (static_cast<int32_t>(factor_16b) << 6) + 32;
@@ -308,17 +311,20 @@
}
}
-int DspHelper::DownsampleTo4kHz(const int16_t* input, size_t input_length,
- size_t output_length, int input_rate_hz,
- bool compensate_delay, int16_t* output) {
+int DspHelper::DownsampleTo4kHz(const int16_t* input,
+ size_t input_length,
+ size_t output_length,
+ int input_rate_hz,
+ bool compensate_delay,
+ int16_t* output) {
// Set filter parameters depending on input frequency.
// NOTE: The phase delay values are wrong compared to the true phase delay
// of the filters. However, the error is preserved (through the +1 term) for
// consistency.
const int16_t* filter_coefficients; // Filter coefficients.
- size_t filter_length; // Number of coefficients.
- size_t filter_delay; // Phase delay in samples.
- int16_t factor; // Conversion rate (inFsHz / 8000).
+ size_t filter_length; // Number of coefficients.
+ size_t filter_delay; // Phase delay in samples.
+ int16_t factor; // Conversion rate (inFsHz / 8000).
switch (input_rate_hz) {
case 8000: {
filter_length = 3;