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adjustable filter envelope and vca gate

This commit is contained in:
Gordon JC Pearce 2025-12-19 15:06:02 +00:00
parent 05ece71acd
commit 6a00521656
3 changed files with 60 additions and 26 deletions
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42
plugin/module.cpp
View File

@ -18,6 +18,8 @@
#include "module.hpp"
#include <stdio.h>
#include "tables.hpp"
Module::Module() {
@ -26,32 +28,50 @@ Module::Module() {
void Module::run(Voice* voice) {
// run updates for module board
// FIXME break these out to the patch setter
a = attackTable[patchRam.env_a]; // attack time coeff looked up in table
d = decayTable[patchRam.env_d]; // decay time coeff looked up in table
r = decayTable[patchRam.env_r]; // release time coeff looked up in table
s = patchRam.env_s << 7; // scale 0x00-0x7f to 0x0000-0x3f80
// FIXME break these out to the patch setter
a = attackTable[patchRam.env_a]; // attack time coeff looked up in table
d = decayTable[patchRam.env_d]; // decay time coeff looked up in table
r = decayTable[patchRam.env_r]; // release time coeff looked up in table
s = patchRam.env_s << 7; // scale 0x00-0x7f to 0x0000-0x3f80
square = (patchRam.switch2 & 0x08) ? 0.63 : 0;
saw = (patchRam.switch2 & 0x10) ? 0.8 : 0;
sub = patchRam.sub / 127.0f;
// printf("%f %f %f %02x\n", square, saw, sub, patchRam.switch2);
// work out the "master" cutoff
vcfCutoff = patchRam.vcfFreq / 127.0f;
vcfCutoff += lfo * (patchRam.vcfLfo/127.0f);
// vcfCutoff = patchRam.vcfFreq / 127.0f;
// vcfCutoff += lfo * (patchRam.vcfLfo / 127.0f);
// also needs pitch bend amount for the base level
int16_t vcf = (patchRam.vcfEnv << 7) * ((patchRam.switch2 & 0x02) ? -1 : 1);
// int16_t vca = (patchRam.vcfEnv << 7) * (patchRam.switch2 & 0x01) ? -1 : 1;
for (uint32_t i = 0; i < NUM_VOICES; i++) {
switch (voice[i].envPhase) {
case 0: // release phase FIXME use an enum I guess
case 0: // release phase FIXME use an enum I guess
voice[i].env = (voice[i].env * d) >> 16; // "RC" decay to zero
break;
case 1: // attack phase
case 1: // attack phase
voice[i].env += a; // linear attack to 0x3fff
break;
case 2:
voice[i].env = (((voice[i].env - s) * d) >>16 ) + s;
voice[i].env = (((voice[i].env - s) * d) >> 16) + s;
break;
}
if (voice[i].env > 0x3fff) {
voice[i].env = 0x3fff;
voice[i].envPhase = 2; // flip to decay
}
// per voice we need to calculate the key follow amount and envelope amount
// per voice we need to calculate the key follow amount and envelope amount
voice[i].vcfCut = (patchRam.vcfFreq << 7) + ((vcf * voice[i].env) >> 16);
if (voice[i].vcfCut > 0x3fff) voice[i].vcfCut = 0x3fff;
if (voice[i].vcfCut < 0) voice[i].vcfCut = 0;
voice[i].vcaEnv = (patchRam.switch2 & 0x04) ? (voice[i].envPhase ? 0x3fff : 0) : voice[i].env;
}
}

23
plugin/module.hpp
View File

@ -35,31 +35,37 @@ class Module {
float lfo = 0, lfoTheta = 0;
// precomputed values for all voices
float pw, saw, square, sub;
float pw;//, saw, square, sub;
// "internal state" values for patch parameters
uint16_t a, d, s, r;
float saw = 0, square = 0, sub = 0, noise = 0;
struct {
uint8_t lfoRate = 0x30; // lookup value defaults to 0x0200
uint8_t lfoDelay = 0x00;
uint8_t vcoLfo = 0x0a;
uint8_t pwmLfo = 0x30;
uint8_t noise = 0x00;
uint8_t vcfFreq = 0x3c; // 0x3f80
uint8_t vcfFreq = 0x4c; // 0x3f80
uint8_t vcfReso = 0x00;
uint8_t vcfEnv = 0x2e;
uint8_t vcfEnv = 0x4e;
uint8_t vcfLfo = 0;
uint8_t vcfKey = 0x47;
uint8_t vca = 0x28;
uint8_t env_a = 0x1b;
uint8_t env_a = 0x00;
uint8_t env_d = 0x39;
uint8_t env_s = 0x39; // 0x3f80
uint8_t env_r = 0x30;
uint8_t sub = 0x00;
uint8_t sub = 0x7f;
uint8_t switch1 = 0x1a;
uint8_t switch2 = 0x18;
} patchRam;
private:
// controls
float subRC = 0, outRC = 0, pwmRC = 0, resRC = 0, noiseRC = 0;
};
class Voice {
@ -70,9 +76,14 @@ class Voice {
void run(Module* m, float* buffer, uint32_t samples);
uint8_t envPhase = 0;
int16_t env = 0; // output amplitude
int16_t vcfCut;
int16_t vcaEnv;
private:
float omega = 0, theta = 0; // phase increment and angle
// control
float vcaRC = 0, vcfRC = 0;
float omega = 0, theta = 0; // phase increment and angle FIXME better names
float delay = 0, lastpw = 0; // delay slots for antialiasing
uint8_t pulseStage = 1; // pulse wave phase
float subosc = 1; // sub oscillator flipflop output

21
plugin/voice.cpp
View File

@ -51,14 +51,17 @@ void Voice::run(Module* m, float* buffer, uint32_t samples) {
// carry out per-voice calculations for each block of samples
float out, t, fb, res;
float cut = 0.00513 + 0.0000075*env;
// printf("%f ", delay);
m->saw = 1;
m->square = 1;
m->sub = .5;
m->pw = 0.5;
//float cut = 0.00513 + 0.0000075*env;
float amp = env / 4096.0f;
// calculate cutoff frequency
float cut = 248.0f * (powf(2, (vcfCut - 0x1880) / 1143.0f));
cut = 0.25 * 6.2832 * cut / 48000.0f; // FIXME hardcoded values
cut = cut/(1+cut); // correct tuning warp
// printf("%f ", delay);
m->pw = 0.5;
float amp = vcaEnv / 4096.0f;
for (uint32_t i = 0; i < samples; i++) {
out = delay;
@ -101,14 +104,14 @@ void Voice::run(Module* m, float* buffer, uint32_t samples) {
if (fb > 1) fb = 1;
if (fb < -1) fb = -1;
fb = out - (fb * 2);
fb = out - (fb * 1.7);
b1 = ((fb - b1) * cut) + b1;
b2 = ((b1 - b2) * cut) + b2;
b3 = ((b2 - b3) * cut) + b3;
b4 = ((b3 - b4) * cut) + b4;
}
buffer[i] += 0.125 * amp * b4;
buffer[i] += 0.0625 * amp * b4;
lastpw = m->pw;
}
// buffer[0] += 1;