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better noise, sticking with original filter, lfo cleanup

This commit is contained in:
Gordon JC Pearce 2026-01-01 01:06:51 +00:00
parent 2ebf7fac1c
commit a8464e6d12
5 changed files with 63 additions and 126 deletions
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2
plugin/chorus.cpp
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@ -59,7 +59,7 @@ void Chorus::run(float* input, float** outputs, uint32_t frames) {
// run highpass / bass boost and stereo chorus effect for one full block
float s0 = 0, s1 = 0;
float lfoMod, dly1, frac, flt;
float dly1, frac, flt;
uint16_t tap, delay;
for (uint32_t i = 0; i < frames; i++) {

55
plugin/module.cpp
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@ -32,6 +32,7 @@ Module::Module() {
vcaBuf = new float[bufferSize];
subBuf = new float[bufferSize];
pwmBuf = new float[bufferSize];
noiseBuf = new float[bufferSize];
}
Module::~Module() {
@ -41,15 +42,21 @@ Module::~Module() {
delete pwmBuf;
}
void Module::genNoise() {
for (uint32_t i = 0; i < bufferSize; i++) {
noiseRNG *= 0x8088405;
noiseRNG++;
noiseBuf[i] = 2 - (noiseRNG & 0xffff) / 16384.0f;
}
}
void Module::lfoRampOn() {
lfoDelayState = 1;
lfoDelayTimer = 0;
lfoDelay = 0;
}
void Module::run(Voice* voices, uint32_t blockSize) {
// run updates for module board
void Module::runLFO() {
if (lfoDelayState == 1) {
lfoDelayTimer += lfoDelayTable[patchRam.lfoDelay >> 4];
if (lfoDelayTimer & 0xc000) lfoDelayState = 2;
@ -63,41 +70,45 @@ void Module::run(Voice* voices, uint32_t blockSize) {
lfoDelay = 0x3fff;
}
lfoPhase += lfoRateTable[patchRam.lfoRate];
if (lfoPhase & 0x4000)
lfo = 0x1fff - (lfoPhase & 0x3fff);
else
lfo = (lfoPhase & 0x3fff) - 0x1fff;
pw = 0x3fff-(((0x2000 + lfo) * patchRam.pwmLfo) >> 7);
pw = (patchRam.switch2 & 0x01) ? 0x3fff - (patchRam.pwmLfo << 7 ) : pw;
lfo = (lfo * lfoDelay) >> 14;
}
void Module::run(Voice* voices, uint32_t blockSize) {
// 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
square = (patchRam.switch1 & 0x08) ? 0.63 : 0;
saw = (patchRam.switch1 & 0x10) ? 0.8 : 0;
sub = patchRam.sub / 127.0f;
lfoPhase += lfoRateTable[patchRam.lfoRate];
master = powf(2, (patchRam.vca / 31.75 - 4.0f)) * 0.1;
square = (patchRam.switch1 & 0x08) ? 0.28 : 0;
saw = (patchRam.switch1 & 0x10) ? .36 : 0;
sub = (patchRam.sub / 127.0f) * 0.4;
res = patchRam.vcfReso / 127.0;
noise = patchRam.noise / 127.0;
noise = (patchRam.noise / 127.0) * 0.4;
// FIXME the exp in these is expensive, don't call it all the time
chorus->setChorus(patchRam.switch1 & 0x60);
chorus->setHpf(patchRam.switch2 & 0x18);
if (lfoPhase & 0x4000)
lfo = 0x1fff - (lfoPhase & 0x3fff);
else
lfo = (lfoPhase & 0x3fff) - 0x1fff;
// FIXME represent PW as int until we calculate the block?
pw = 0.5 - ((0x2000 + lfo) * patchRam.pwmLfo) / (32768.0f * 128);
pw = (patchRam.switch2 & 0x01) ? 0.5 - (patchRam.pwmLfo / 256.0f) : pw;
lfo = (lfo * lfoDelay) >> 14;
float master = powf(2, (patchRam.vca / 31.75 - 4.0f));
float sub = patchRam.sub / 127.0f;
runLFO();
float pwf = pw / 32768.0f;
for (uint32_t i = 0; i < blockSize; i++) {
vcaRC = (master - vcaRC) * subTC + vcaRC;
pwmRC = (pw - pwmRC) * pwmTC + pwmRC;
pwmRC = (pwf - pwmRC) * pwmTC + pwmRC;
subRC = (sub - subRC) * vcaTC + subRC;
vcaBuf[i] = vcaRC;

19
plugin/module.hpp
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@ -33,20 +33,16 @@ class Module {
Module();
~Module();
void genNoise();
void lfoRampOn();
void run(Voice* voices, uint32_t blockLeft);
float res = 0;
// precomputed values for all voices
float pw; //, saw, square, sub;
// "internal state" values for patch parameters
uint16_t a, d, s, r;
int16_t lfo;
uint32_t lfoPhase;
float saw = 0, square = 0, sub = 0, noise = 0;
float saw = 0, square = 0, sub = 0, noise = 0, master = 0;
/*
#if 0
@ -124,12 +120,19 @@ class Module {
float* vcaBuf;
float* subBuf;
float* pwmBuf;
float* noiseBuf;
private:
void runLFO();
// precalculated coefficients for RC networks
float pwmTC = 0, subTC = 0, mVcaTC = 0;
float pwmRC = 0, subRC = 0, vcaRC = 0;
int16_t lfo, pw;
uint32_t lfoPhase;
uint32_t noiseRNG = 1;
uint16_t lfoDelay = 0;
uint8_t lfoDelayState = 0;
uint16_t lfoDelayTimer = 0;
@ -142,7 +145,7 @@ class Voice {
Voice();
void on(uint8_t midiNote);
void off();
void run(Module* m, float* buffer, uint32_t samples);
void run(Module* m, float* buffer, uint32_t framePos, uint32_t samples);
private:
float omega = 0, theta = 0; // phase increment and angle FIXME better names
@ -160,8 +163,6 @@ class Voice {
// filter
float y0 = 0, y1 = 0, y2 = 0, y3 = 0;
double s[4] = {0, 0, 0, 0};
float zi = 0;
};
#endif

10
plugin/peacock.cpp
View File

@ -70,9 +70,11 @@ void Peacock::run(const float**, float** outputs, uint32_t frames, const MidiEve
memset(outputs[0], 0, frames * sizeof(float));
memset(outputs[1], 0, frames * sizeof(float));
m->genNoise();
// if there were any events that happen between now and the end of this block, process them
lastEvent = 0;
m->bufPtr = 0;
m->bufPtr = 0; // reset the output buffer pointer
runMidi(midiEvents, midiEventCount, blockLeft);
while (framePos < frames) {
@ -80,17 +82,17 @@ void Peacock::run(const float**, float** outputs, uint32_t frames, const MidiEve
// no more samples to calculate in this update period
blockLeft = sampleRate / 238; // update rate in Hz
runMidi(midiEvents, midiEventCount, framePos + blockLeft);
}
// how many frames to do? Are we about to run off an update block
sizeThisTime = (framesLeft < blockLeft) ? framesLeft : blockLeft;
m->run(voice, sizeThisTime);
// update the module board for this block
m->run(voice, sizeThisTime);
// now run all the voices for this chunk of samples
for (uint32_t i = 0; i < NUM_VOICES; i++) {
voice[i].run(m, outputs[0] + framePos, sizeThisTime);
voice[i].run(m, outputs[0], framePos, sizeThisTime);
}
framePos += sizeThisTime;

101
plugin/voice.cpp
View File

@ -54,22 +54,23 @@ void Voice::off() {
// so might not be safe for very large feedback gains
// [limit is 1/15 so very large means ~15 or +23dB]
double tanhXdX(double x) {
float tanhXdX(float x) {
return 1 - 0.05 * abs(x);
float s = 0.0333, d = 30.0;
return 1.0f - s * (d + 1.0f) * x * x / (d + x * x);
}
void Voice::run(Module* m, float* buffer, uint32_t samples) {
void Voice::run(Module* m, float* buffer, uint32_t framePos, uint32_t samples) {
// carry out per-voice calculations for each block of samples
float out, t, fb;
// calculate cutoff frequency
float cut = 261.0f * (powf(2, (vcfCut - 0x1880) / 1143.0f));
float cut = 261.0f * (powf(2, (vcfCut - 0x1880) / 1143.0f)); // FIXME explain magic numbers
cut = M_PI * cut / sampleRate;
cut = cut / (1 + cut); // correct tuning warp
// if (cut > 0.7) cut = 0.7;
double r = 5 * m->res;
if (cut > 0.7) cut = 0.7;
float r = 5 * m->res;
float amp = vcaEnv / 4096.0f;
@ -103,109 +104,31 @@ void Voice::run(Module* m, float* buffer, uint32_t samples) {
}
}
// FIXME DC offset removal
delay += m->saw * (1 - (2 * theta));
delay += m->square * ((pulseStage ? -1.f : 1.f) - m->pwmBuf[i] + 0.5);
delay += m->subBuf[i] * subosc;
out += m->noise * (0.8 - 1.6 * (rand() & 0xffff) / 65536.0);
// out *= 0.1;
out += m->noise * m->noiseBuf[i + framePos];
// same time constant for both VCF and VCF RC circuits
vcfRC = (cut - vcfRC) * m->vcaTC + vcfRC;
#if 1
//// LICENSE TERMS: Copyright 2012 Teemu Voipio
//
// You can use this however you like for pretty much any purpose,
// as long as you don't claim you wrote it. There is no warranty.
//
// Distribution of substantial portions of this code in source form
// must include this copyright notice and list of conditions.
//
// input delay and state for member variables
// cutoff as normalized frequency (eg 0.5 = Nyquist)
// resonance from 0 to 1, self-oscillates at settings over 0.9
// void transistorLadder(
// double cutoff, double resonance,
// double * in, double * out, unsigned nsamples)
//{
// tuning and feedback
//------------------------------------------------------------------------------ sample loop
// for(unsigned n = 0; n < nsamples; ++n)
//{
out *= 0.025;
// input with half delay, for non-linearities
double ih = 0.5 * (out + zi);
zi = out;
// double ih = out;
// evaluate the non-linear gains
double t0 = tanhXdX((ih * (r + 1)) - r * s[3]);
double t1 = tanhXdX(s[0]);
double t2 = tanhXdX(s[1]);
double t3 = tanhXdX(s[2]);
double t4 = tanhXdX(s[3]);
double f = vcfRC;
// g# the denominators for solutions of individual stages
double g0 = 1 / (1 + f * t1), g1 = 1 / (1 + f * t2);
double g2 = 1 / (1 + f * t3), g3 = 1 / (1 + f * t4);
// f# are just factored out of the feedback solution
double f3 = f * t3 * g3, f2 = f * t2 * g2 * f3, f1 = f * t1 * g1 * f2, f0 = f * t0 * g0 * f1;
// solve feedback
double y3 = (g3 * s[3] + f3 * g2 * s[2] + f2 * g1 * s[1] + f1 * g0 * s[0] + f0 * out) / (1 + r * f0);
// then solve the remaining outputs (with the non-linear gains here)
double xx = t0 * ((out * (r + 1)) - r * y3);
double y0 = t1 * g0 * (s[0] + f * xx);
double y1 = t2 * g1 * (s[1] + f * y0);
double y2 = t3 * g2 * (s[2] + f * y1);
// update state
s[0] += 2 * f * (xx - y0);
s[1] += 2 * f * (y0 - y1);
s[2] += 2 * f * (y1 - y2);
s[3] += 2 * f * (y2 - t4 * y3);
// out[n] = y3;
// }
// out *= 0.1;
out = y3;
#else
out *= 0.5;
for (uint8_t ovs = 0; ovs < 2; ovs++) {
fb = y3;
// hard clip
fb = ((out * 0.5) - fb) * r;
if (fb > 4) fb = 4;
if (fb < -4) fb = -4;
// fb = 1.5 * fb - 0.5 * fb * fb * fb;
//
if (fb > 2) fb = 2;
if (fb < -2) fb = -2;
y0 = ((out + fb - y0) * vcfRC) + y0;
y1 = ((y0 - y1) * vcfRC) + y1;
y2 = ((y1 - y2) * vcfRC) + y2;
y3 = ((y2 - y3) * vcfRC) + y3;
}
#endif
vcaRC = (amp - vcaRC) * m->vcaTC + vcaRC;
buffer[i] += m->vcaBuf[i] * vcaRC * out;
buffer[framePos + i] += m->vcaBuf[i] * vcaRC * y3;
lastpw = m->pwmBuf[i];
}
// buffer[0] += 1;
// buffer[0] += 1; // buzzing noise to test
}