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sine-based, getting close to real though

This commit is contained in:
Gordon JC Pearce 2025-12-21 22:08:09 +00:00
parent 36db28a11f
commit 8c091c2a05
3 changed files with 29 additions and 38 deletions
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58
plugin/chorus.cpp
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@ -30,17 +30,16 @@ Chorus::Chorus() {
ram = new float[DELAYSIZE]; // probably needs to be calculated based on sample rate ram = new float[DELAYSIZE]; // probably needs to be calculated based on sample rate
fastPhase = 0; fastPhase = 0;
slowPhase = 0;
postFilter1l = new SVF(POSTCUTOFF, .546); // not quite Butterworth but you'd never hear the difference
postFilter2l = new SVF(POSTCUTOFF, 1.324); postFilter1l = new SVF(9688, .549);
postFilter1r = new SVF(POSTCUTOFF, .546); postFilter2l = new SVF(10377, 1.291);
postFilter2r = new SVF(POSTCUTOFF, 1.324); postFilter1r = new SVF(9688, .549);
postFilter2r = new SVF(10377, 1.291);
// lfo values taken from a rough simulation // lfo values taken from a rough simulation
fastOmega = 6.283 * 5.7 / sampleRate; // approximate, can be adjusted fastOmega = 6.283 * 0.7 / sampleRate; // approximate, can be adjusted
slowOmega = 6.283 * 0.7 / sampleRate; // again approximate
// zero out the delay buffer // zero out the delay buffer
memset(ram, 0, sizeof(float) * DELAYSIZE); memset(ram, 0, sizeof(float) * DELAYSIZE);
memset(lpfOut1, 0, sizeof(float) * bufferSize); memset(lpfOut1, 0, sizeof(float) * bufferSize);
@ -62,7 +61,7 @@ void Chorus::run(const float* input, float** outputs, uint32_t frames) {
// actual effects here // actual effects here
// now run the DSP // now run the DSP
float out0 = 0, out120 = 0, out240 = 0, s0 = 0, s1 = 0; float s0 = 0, s1 = 0;
float lfoMod, dly1, frac; float lfoMod, dly1, frac;
uint16_t tap, delay; uint16_t tap, delay;
@ -70,16 +69,14 @@ void Chorus::run(const float* input, float** outputs, uint32_t frames) {
// run a step of LFO // run a step of LFO
fastPhase += fastOmega; fastPhase += fastOmega;
if (fastPhase > 6.283) fastPhase -= 6.283; if (fastPhase > 6.283) fastPhase -= 6.283;
slowPhase += slowOmega;
if (slowPhase > 6.283) slowPhase -= 6.283;
ram[delayptr] = input[i]; ram[delayptr] = input[i];
#define BASE 0.05 #define BASE 0.05
#define AMT 0.00175 #define AMT 0.003175
// 0 degree delay line // 0 degree delay line
lfoMod = 0.203 * sin(fastPhase) + 0.835 * sin(slowPhase); lfoMod = 0.603 * sin(fastPhase);
dly1 = (BASE + (AMT * lfoMod)) * sampleRate; dly1 = (BASE + (AMT * lfoMod)) * sampleRate;
delay = (int)dly1; delay = (int)dly1;
frac = dly1 - delay; frac = dly1 - delay;
@ -87,39 +84,34 @@ void Chorus::run(const float* input, float** outputs, uint32_t frames) {
tap = delayptr - delay; tap = delayptr - delay;
s1 = ram[(tap - 1) & 0x3ff]; s1 = ram[(tap - 1) & 0x3ff];
s0 = ram[tap & 0x3ff]; s0 = ram[tap & 0x3ff];
out0 = ((s1 - s0) * frac) + s0; lpfOut1[i] = ((s1 - s0) * frac) + s0;
// 120 degree delay line dly1 = (BASE - (AMT * lfoMod)) * sampleRate;
lfoMod = 0.248 * sin(fastPhase + 2.09) + 0.745 * sin(slowPhase + 2.09);
dly1 = (BASE + (AMT * lfoMod)) * sampleRate;
delay = (int)dly1; delay = (int)dly1;
frac = dly1 - delay; frac = dly1 - delay;
tap = delayptr - delay; tap = delayptr - delay;
s1 = ram[(tap - 1) & 0x3ff]; s1 = ram[(tap - 1) & 0x3ff];
s0 = ram[tap & 0x3ff]; s0 = ram[tap & 0x3ff];
out120 = ((s1 - s0) * frac) + s0; lpfOut2[i] = ((s1 - s0) * frac) + s0;
// 240 degree delay line // lpfOut1[i] = input[i] + 1.2 * out0; //(out0 + (out120 * 0.66) + (out240 * 0.33));
lfoMod = 0.252 * sin(fastPhase + 4.18) + 0.809 * sin(slowPhase + 4.18); // lpfOut2[i] = input[i] + 1.2 * out120; //(out0 + (out120 * 0.33) + (out240 * 0.66));
dly1 = (BASE + (AMT * lfoMod)) * sampleRate;
delay = (int)dly1;
frac = dly1 - delay;
tap = delayptr - delay;
s1 = ram[(tap - 1) & 0x3ff];
s0 = ram[tap & 0x3ff];
out240 = ((s1 - s0) * frac) + s0;
lpfOut1[i] = (out0 + (out120 * 0.66) + (out240 * 0.33));
lpfOut2[i] = (out0 + (out120 * 0.33) + (out240 * 0.66));
delayptr++; delayptr++;
delayptr &= 0x3ff; delayptr &= 0x3ff;
} }
postFilter1l->runSVF(lpfOut1, lpfOut1, frames); postFilter1l->runSVF(lpfOut1, lpfOut1, frames);
postFilter2l->runSVF(lpfOut1, outputs[0], frames); postFilter2l->runSVF(lpfOut1, lpfOut1, frames);
postFilter1r->runSVF(lpfOut2, lpfOut2, frames); postFilter1r->runSVF(lpfOut2, lpfOut2, frames);
postFilter2r->runSVF(lpfOut2, outputs[1], frames); postFilter2r->runSVF(lpfOut2, lpfOut2, frames);
for (uint32_t i = 0; i < frames; i++) {
float y = input[i];
outputs[0][i] = y + (gain * lpfOut1[i]);
outputs[1][i] = y + (gain * lpfOut2[i]);
}
} }

7
plugin/chorus.hpp
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@ -36,11 +36,10 @@ class Chorus {
void run(const float* input, float** outputs, uint32_t frames); void run(const float* input, float** outputs, uint32_t frames);
private: private:
double fastPhase, fastOmega; double fastPhase = 0, fastOmega = 0;
double slowPhase, slowOmega; float gain = 1.2;
double fastLfo, slowLfo;
uint16_t delayptr; uint16_t delayptr = 0;
float* ram; float* ram;
float* lpfIn; float* lpfIn;

2
plugin/voice.cpp
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@ -52,7 +52,7 @@ void Voice::off() {
void Voice::run(Module* m, float* buffer, uint32_t samples) { void Voice::run(Module* m, float* buffer, uint32_t samples) {
// carry out per-voice calculations for each block of samples // carry out per-voice calculations for each block of samples
float out, t, fb, res; float out, t, fb;
//float cut = 0.00513 + 0.0000075*env; //float cut = 0.00513 + 0.0000075*env;