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Core: Use Math namespace for constants

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Thaddeus Crews 2025-04-10 11:21:05 -05:00
parent 06c71fbf40
commit 94282d88f9
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GPG key ID: 8C6E5FEB5FC03CCC
181 changed files with 812 additions and 818 deletions
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4
servers/audio/effects/audio_effect_chorus.cpp
View file

@ -84,7 +84,7 @@ void AudioEffectChorusInstance::_process_chunk(const AudioFrame *p_src_frames, A
if (v.cutoff == 0) {
continue;
}
float auxlp = expf(-Math_TAU * v.cutoff / mix_rate);
float auxlp = expf(-Math::TAU * v.cutoff / mix_rate);
float c1 = 1.0 - auxlp;
float c2 = auxlp;
AudioFrame h = filter_h[vc];
@ -104,7 +104,7 @@ void AudioEffectChorusInstance::_process_chunk(const AudioFrame *p_src_frames, A
float phase = (float)(local_cycles & AudioEffectChorus::CYCLES_MASK) / (float)(1 << AudioEffectChorus::CYCLES_FRAC);
float wave_delay = sinf(phase * Math_TAU) * max_depth_frames;
float wave_delay = sinf(phase * Math::TAU) * max_depth_frames;
int wave_delay_frames = lrint(floor(wave_delay));
float wave_delay_frac = wave_delay - (float)wave_delay_frames;

2
servers/audio/effects/audio_effect_delay.cpp
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@ -73,7 +73,7 @@ void AudioEffectDelayInstance::_process_chunk(const AudioFrame *p_src_frames, Au
tap2_vol.right *= CLAMP(1.0 + base->tap_2_pan, 0, 1);
// feedback lowpass here
float lpf_c = expf(-Math_TAU * base->feedback_lowpass / mix_rate); // 0 .. 10khz
float lpf_c = expf(-Math::TAU * base->feedback_lowpass / mix_rate); // 0 .. 10khz
float lpf_ic = 1.0 - lpf_c;
const AudioFrame *src = p_src_frames;

4
servers/audio/effects/audio_effect_distortion.cpp
View file

@ -36,8 +36,8 @@ void AudioEffectDistortionInstance::process(const AudioFrame *p_src_frames, Audi
const float *src = (const float *)p_src_frames;
float *dst = (float *)p_dst_frames;
//float lpf_c=expf(-Math_TAU*keep_hf_hz.get()/(mix_rate*(float)OVERSAMPLE));
float lpf_c = expf(-Math_TAU * base->keep_hf_hz / (AudioServer::get_singleton()->get_mix_rate()));
//float lpf_c=expf(-Math::TAU*keep_hf_hz.get()/(mix_rate*(float)OVERSAMPLE));
float lpf_c = expf(-Math::TAU * base->keep_hf_hz / (AudioServer::get_singleton()->get_mix_rate()));
float lpf_ic = 1.0 - lpf_c;
float drive_f = base->drive;

6
servers/audio/effects/audio_effect_phaser.cpp
View file

@ -37,13 +37,13 @@ void AudioEffectPhaserInstance::process(const AudioFrame *p_src_frames, AudioFra
float dmin = base->range_min / (sampling_rate / 2.0);
float dmax = base->range_max / (sampling_rate / 2.0);
float increment = Math_TAU * (base->rate / sampling_rate);
float increment = Math::TAU * (base->rate / sampling_rate);
for (int i = 0; i < p_frame_count; i++) {
phase += increment;
while (phase >= Math_TAU) {
phase -= Math_TAU;
while (phase >= Math::TAU) {
phase -= Math::TAU;
}
float d = dmin + (dmax - dmin) * ((sin(phase) + 1.f) / 2.f);

16
servers/audio/effects/audio_effect_pitch_shift.cpp
View file

@ -92,7 +92,7 @@ void SMBPitchShift::PitchShift(float pitchShift, long numSampsToProcess, long ff
fftFrameSize2 = fftFrameSize/2;
stepSize = fftFrameSize/osamp;
freqPerBin = sampleRate/(double)fftFrameSize;
expct = 2.*Math_PI*(double)stepSize/(double)fftFrameSize;
expct = 2.*Math::PI*(double)stepSize/(double)fftFrameSize;
inFifoLatency = fftFrameSize-stepSize;
if (gRover == 0) { gRover = inFifoLatency;
}
@ -112,7 +112,7 @@ void SMBPitchShift::PitchShift(float pitchShift, long numSampsToProcess, long ff
/* do windowing and re,im interleave */
for (k = 0; k < fftFrameSize;k++) {
window = -.5*cos(2.*Math_PI*(double)k/(double)fftFrameSize)+.5;
window = -.5*cos(2.*Math::PI*(double)k/(double)fftFrameSize)+.5;
gFFTworksp[2*k] = gInFIFO[k] * window;
gFFTworksp[2*k+1] = 0.;
}
@ -140,14 +140,14 @@ void SMBPitchShift::PitchShift(float pitchShift, long numSampsToProcess, long ff
tmp -= (double)k*expct;
/* map delta phase into +/- Pi interval */
qpd = tmp/Math_PI;
qpd = tmp/Math::PI;
if (qpd >= 0) { qpd += qpd&1;
} else { qpd -= qpd&1;
}
tmp -= Math_PI*(double)qpd;
tmp -= Math::PI*(double)qpd;
/* get deviation from bin frequency from the +/- Pi interval */
tmp = osamp*tmp/(2.*Math_PI);
tmp = osamp*tmp/(2.*Math::PI);
/* compute the k-th partials' true frequency */
tmp = (double)k*freqPerBin + tmp*freqPerBin;
@ -184,7 +184,7 @@ void SMBPitchShift::PitchShift(float pitchShift, long numSampsToProcess, long ff
tmp /= freqPerBin;
/* take osamp into account */
tmp = 2.*Math_PI*tmp/osamp;
tmp = 2.*Math::PI*tmp/osamp;
/* add the overlap phase advance back in */
tmp += (double)k*expct;
@ -207,7 +207,7 @@ void SMBPitchShift::PitchShift(float pitchShift, long numSampsToProcess, long ff
/* do windowing and add to output accumulator */
for(k=0; k < fftFrameSize; k++) {
window = -.5*cos(2.*Math_PI*(double)k/(double)fftFrameSize)+.5;
window = -.5*cos(2.*Math::PI*(double)k/(double)fftFrameSize)+.5;
gOutputAccum[k] += 2.*window*gFFTworksp[2*k]/(fftFrameSize2*osamp);
}
for (k = 0; k < stepSize; k++) { gOutFIFO[k] = gOutputAccum[k];
@ -260,7 +260,7 @@ void SMBPitchShift::smbFft(float *fftBuffer, long fftFrameSize, long sign)
le2 = le>>1;
ur = 1.0;
ui = 0.0;
arg = Math_PI / (le2>>1);
arg = Math::PI / (le2>>1);
wr = cos(arg);
wi = sign*sin(arg);
for (j = 0; j < le2; j += 2) {

4
servers/audio/effects/audio_effect_spectrum_analyzer.cpp
View file

@ -71,7 +71,7 @@ static void smbFft(float *fftBuffer, long fftFrameSize, long sign)
le2 = le >> 1;
ur = 1.0;
ui = 0.0;
arg = Math_PI / (le2 >> 1);
arg = Math::PI / (le2 >> 1);
wr = cos(arg);
wi = sign * sin(arg);
for (j = 0; j < le2; j += 2) {
@ -110,7 +110,7 @@ void AudioEffectSpectrumAnalyzerInstance::process(const AudioFrame *p_src_frames
while (p_frame_count) {
int to_fill = fft_size * 2 - temporal_fft_pos;
to_fill = MIN(to_fill, p_frame_count);
const double to_fill_step = Math_TAU / (double)fft_size;
const double to_fill_step = Math::TAU / (double)fft_size;
float *fftw = temporal_fft.ptrw();
for (int i = 0; i < to_fill; i++) { //left and right buffers

6
servers/audio/effects/eq_filter.cpp
View file

@ -88,9 +88,9 @@ void EQ::recalculate_band_coefficients() {
double frq_l = round(frq / pow(2.0, octave_size / 2.0));
double side_gain2 = POW2(Math_SQRT12);
double th = Math_TAU * frq / mix_rate;
double th_l = Math_TAU * frq_l / mix_rate;
double side_gain2 = POW2(Math::SQRT12);
double th = Math::TAU * frq / mix_rate;
double th_l = Math::TAU * frq_l / mix_rate;
double c2a = side_gain2 * POW2(cos(th)) - 2.0 * side_gain2 * cos(th_l) * cos(th) + side_gain2 - POW2(sin(th_l));

4
servers/audio/effects/reverb_filter.cpp
View file

@ -90,7 +90,7 @@ void Reverb::process(float *p_src, float *p_dst, int p_frames) {
}
if (params.hpf > 0) {
float hpaux = expf(-Math_TAU * params.hpf * 6000 / params.mix_rate);
float hpaux = expf(-Math::TAU * params.hpf * 6000 / params.mix_rate);
float hp_a1 = (1.0 + hpaux) / 2.0;
float hp_a2 = -(1.0 + hpaux) / 2.0;
float hp_b1 = hpaux;
@ -292,7 +292,7 @@ void Reverb::update_parameters() {
float auxdmp = params.damp / 2.0 + 0.5; //only half the range (0.5 .. 1.0 is enough)
auxdmp *= auxdmp;
c.damp = expf(-Math_TAU * auxdmp * 10000 / params.mix_rate); // 0 .. 10khz
c.damp = expf(-Math::TAU * auxdmp * 10000 / params.mix_rate); // 0 .. 10khz
}
}