string ensemble chorus crudely grafted on

This commit is contained in:
Gordon JC Pearce 2025-12-20 23:36:34 +00:00
parent b736c6534e
commit 8f604d10c2
7 changed files with 282 additions and 1 deletions

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@ -15,6 +15,8 @@ FILES_DSP = \
voice.cpp \
tables.cpp \
parameters.cpp \
svf.cpp \
chorus.cpp \
peacock.cpp
FILES_UI = \

129
plugin/chorus.cpp Normal file
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@ -0,0 +1,129 @@
/*
sonnenlicht poly ensemble
Copyright 2025 Gordon JC Pearce <gordonjcp@gjcp.net>
Permission to use, copy, modify, and/or distribute this software for any
purpose with or without fee is hereby granted, provided that the above
copyright notice and this permission notice appear in all copies.
THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY
SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN ACTION
OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF OR IN
CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
*/
#include "chorus.hpp"
#include <math.h>
#include <string.h>
#include <cstdio>
//extern double sampleRate;
//extern uint32_t bufferSize;
double sampleRate = 48000;
uint32_t bufferSize = 1024;
Chorus::Chorus() { // no parameters, programs, or states
lpfOut1 = new float[bufferSize];
lpfOut2 = new float[bufferSize];
ram = new float[DELAYSIZE]; // probably needs to be calculated based on sample rate
fastPhase = 0;
slowPhase = 0;
postFilter1l = new SVF(POSTCUTOFF, .546);
postFilter2l = new SVF(POSTCUTOFF, 1.324);
postFilter1r = new SVF(POSTCUTOFF, .546);
postFilter2r = new SVF(POSTCUTOFF, 1.324);
// lfo values taken from a rough simulation
fastOmega = 6.283 * 5.7 / sampleRate; // approximate, can be adjusted
slowOmega = 6.283 * 0.7 / sampleRate; // again approximate
// zero out the delay buffer
memset(ram, 0, sizeof(float) * DELAYSIZE);
memset(lpfOut1, 0, sizeof(float) * bufferSize);
memset(lpfOut2, 0, sizeof(float) * bufferSize);
}
Chorus::~Chorus() {
delete lpfOut1;
delete lpfOut2;
delete ram;
delete postFilter1l;
delete postFilter2l;
delete postFilter1r;
delete postFilter2r;
}
void Chorus::run(const float *input, float **outputs, uint32_t frames) {
// actual effects here
// now run the DSP
float out0 = 0, out120 = 0, out240 = 0, s0 = 0, s1 = 0;
float lfoMod, dly1, frac;
uint16_t tap, delay;
for (uint32_t i = 0; i < frames; i++) {
// run a step of LFO
fastPhase += fastOmega;
if (fastPhase > 6.283) fastPhase -= 6.283;
slowPhase += slowOmega;
if (slowPhase > 6.283) slowPhase -= 6.283;
ram[delayptr] = input[i];
#define BASE 0.05
#define AMT 0.00175
// 0 degree delay line
lfoMod = 0.203 * sin(fastPhase) + 0.835 * sin(slowPhase);
dly1 = (BASE + (AMT * lfoMod)) * sampleRate;
delay = (int)dly1;
frac = dly1 - delay;
tap = delayptr - delay;
s1 = ram[(tap - 1) & 0x3ff];
s0 = ram[tap & 0x3ff];
out0 = ((s1 - s0) * frac) + s0;
// 120 degree delay line
lfoMod = 0.248 * sin(fastPhase + 2.09) + 0.745 * sin(slowPhase + 2.09);
dly1 = (BASE + (AMT * lfoMod)) * sampleRate;
delay = (int)dly1;
frac = dly1 - delay;
tap = delayptr - delay;
s1 = ram[(tap - 1) & 0x3ff];
s0 = ram[tap & 0x3ff];
out120 = ((s1 - s0) * frac) + s0;
// 240 degree delay line
lfoMod = 0.252 * sin(fastPhase + 4.18) + 0.809 * sin(slowPhase + 4.18);
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 &= 0x3ff;
}
postFilter1l->runSVF(lpfOut1, lpfOut1, frames);
postFilter2l->runSVF(lpfOut1, outputs[0], frames);
postFilter1r->runSVF(lpfOut2, lpfOut2, frames);
postFilter2r->runSVF(lpfOut2, outputs[1], frames);
}

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plugin/chorus.hpp Normal file
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/*
sonnenlicht poly ensemble
Copyright 2025 Gordon JC Pearce <gordonjcp@gjcp.net>
Permission to use, copy, modify, and/or distribute this software for any
purpose with or without fee is hereby granted, provided that the above
copyright notice and this permission notice appear in all copies.
THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY
SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN ACTION
OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF OR IN
CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
*/
#ifndef Chorus_HPP
#define Chorus_HPP
#include "svf.hpp"
// total size of delay line buffer
#define DELAYSIZE 1028
#define POSTCUTOFF 10000
class Chorus {
public:
Chorus();
~Chorus();
void run(const float *input, float **outputs, uint32_t frames);
private:
double fastPhase, fastOmega;
double slowPhase, slowOmega;
double fastLfo, slowLfo;
uint16_t delayptr;
float *ram;
float *lpfIn;
float *lpfOut1, *lpfOut2;
SVF *postFilter1l, *postFilter2l, *postFilter1r, *postFilter2r;
};
#endif

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@ -26,11 +26,14 @@ Peacock::Peacock() : Plugin(parameterCount, 0, 0) {
m = new Module;
ic1 = new Assigner;
ic1->voice = voice;
chorus = new Chorus;
}
Peacock::~Peacock() {
free(m);
free(ic1);
free(chorus);
}
void Peacock::initAudioPort(bool input, uint32_t index, AudioPort& port) {
@ -93,7 +96,9 @@ void Peacock::run(const float**, float** outputs, uint32_t frames, const MidiEve
// now we've assembled a full chunk of audio
// we'd apply the highpass filter and chorus here
// for now just copy left to right
memcpy(outputs[1], outputs[0], sizeof(float) * frames);
//memcpy(outputs[1], outputs[0], sizeof(float) * frames);
chorus->run(outputs[0], outputs, frames);
//outputs[0][0]=1;
}

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@ -22,6 +22,8 @@
#include "DistrhoPlugin.hpp"
#include "assigner.hpp"
#include "module.hpp"
#include "chorus.hpp"
START_NAMESPACE_DISTRHO
class Peacock : public Plugin {
@ -55,6 +57,7 @@ class Peacock : public Plugin {
private:
Assigner* ic1;
Module* m;
Chorus* chorus;
uint32_t sampleRate;

55
plugin/svf.cpp Normal file
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@ -0,0 +1,55 @@
/*
State Variable Filter
Copyright 2025 Gordon JC Pearce <gordonjcp@gjcp.net>
Permission to use, copy, modify, and/or distribute this software for any
purpose with or without fee is hereby granted, provided that the above
copyright notice and this permission notice appear in all copies.
THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY
SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN ACTION
OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF OR IN
CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
*/
#include "svf.hpp"
#include <math.h>
#include <cstdio>
SVF::SVF(float cutoff, float Q) {
// zero out all values
z1 = 0;
z2 = 0;
setCutoff(cutoff, Q);
}
void SVF::setCutoff(float cutoff, float Q) {
float F = cutoff / sampleRate;
float w = 2 * tan(3.14159 * F);
float a = w / Q;
float b = w * w;
// "corrected" SVF params, per Fons Adriaensen
c1 = (a + b) / (1 + a / 2 + b / 4);
c2 = b / (a + b);
d0 = c1 * c2 / 4;
}
void SVF::runSVF(const float *input, float *output, uint32_t frames) {
float x;
for (uint32_t i = 0; i < frames; i++) {
// lowpass filter
x = input[i] - z1 - z2;
z2 += c2 * z1;
z1 += c1 * x;
output[i] = d0 * x + z2;
}
// printf("%f\n", x);
}

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plugin/svf.hpp Normal file
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/*
State Variable Filter
Copyright 2025 Gordon JC Pearce <gordonjcp@gjcp.net>
Permission to use, copy, modify, and/or distribute this software for any
purpose with or without fee is hereby granted, provided that the above
copyright notice and this permission notice appear in all copies.
THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY
SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN ACTION
OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF OR IN
CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
*/
#ifndef SVF_HPP
#define SVF_HPP
#include <stdint.h>
extern double sampleRate;
class SVF {
public:
SVF(float cutoff=1000, float Q=0.707);
void setCutoff(float cutoff, float Q);
void runSVF(const float *input, float *output, uint32_t frames);
protected:
private:
float c1, c2, d0;
float z1, z2;
};
#endif