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13 changed files with 283 additions and 301 deletions

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@ -1,43 +0,0 @@
name: build
on: [push, pull_request]
jobs:
linux:
strategy:
matrix:
target: [linux-arm64, linux-armhf, linux-i686, linux-riscv64, linux-x86_64]
runs-on: ubuntu-22.04
steps:
- uses: actions/checkout@v4
with:
submodules: recursive
- uses: distrho/dpf-makefile-action@v1
with:
target: ${{ matrix.target }}
macos:
strategy:
matrix:
target: [macos-intel, macos-universal]
runs-on: macos-15
steps:
- uses: actions/checkout@v4
with:
submodules: recursive
- uses: distrho/dpf-makefile-action@v1
with:
target: ${{ matrix.target }}
windows:
strategy:
matrix:
target: [win32, win64]
runs-on: ubuntu-22.04
steps:
- uses: actions/checkout@v4
with:
submodules: recursive
- uses: distrho/dpf-makefile-action@v1
with:
target: ${{ matrix.target }}

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@ -23,15 +23,6 @@
#define DISTRHO_PLUGIN_NAME "peacock-8"
#define DISTRHO_PLUGIN_URI "https://gjcp.net/plugins/peacock"
#define DISTRHO_PLUGIN_CLAP_ID "net.gjcp.peacock"
#define DISTRHO_PLUGIN_CLAP_FEATURES "instrument","synthesizer","stereo"
#define DISTRHO_PLUGIN_BRAND_ID GJCP
#define DISTRHO_PLUGIN_UNIQUE_ID Pfau
#define DISTRHO_PLUGIN_LV2_CATEGORY "lv2:InstrumentPlugin"
#define DISTRHO_PLUGIN_VST_CATEGORY "Fx|Instrument"
#define DISTRHO_PLUGIN_NUM_INPUTS 0
#define DISTRHO_PLUGIN_NUM_OUTPUTS 2
#define DISTRHO_PLUGIN_IS_SYNTH 1
@ -74,10 +65,6 @@
pChorusMode,
pVcoBend,
pVcfBend,
pModDepth,
parameterCount
};

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@ -34,8 +34,7 @@ include ../dpf/Makefile.plugins.mk
SKIP_NATIVE_AUDIO_FALLBACK = true
# omitting LV2 for the moment until I figure out cross-compiling
TARGETS += jack vst2 vst3 clap
TARGETS += jack lv2_sep
all: $(TARGETS)

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@ -43,12 +43,7 @@ Assigner::Assigner() {
}
void Assigner::handleMidi(MidiEvent* ev) {
uint8_t status = ev->data[0];
//printf("called with event %04x (%02x): %02x %02x %02x\n", ev->frame, ev->size, ev->data[0], ev->data[1], ev->data[2]);
if (ev->size > 3) return; // sysex bug
switch (status & 0xf0) {
case 0x80:
noteOff(ev->data[1]);
@ -61,11 +56,9 @@ void Assigner::handleMidi(MidiEvent* ev) {
break;
case 0xb0:
switch (ev->data[1]) {
case 0x01: // modwheel
printf("mod wheel %02x\n", ev->data[2]);
m->modWheel = ev->data[2];
// handle the following
// CC 64 - sustain // FIXME sustain not implemented
// CC 1 - modwheel
// CC 64 - sustain
// possibly JU-06 CC values
default:
break;
@ -73,8 +66,7 @@ void Assigner::handleMidi(MidiEvent* ev) {
break; // nothing to do here except in special cases where we don't expect the host to pass on controls
case 0xc0: // program change
break;
case 0xe0: // pitch bend
m->bend = ((ev->data[1] + (ev->data[2]<<7))>>5)-0x100;
case 0xe0: // pitch bend;
break;
case 0xf0: // sysex
break;

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@ -19,8 +19,9 @@
#include "chorus.hpp"
#include <math.h>
#include <stdio.h>
#include <string.h>
#include <stdio.h>
Chorus::Chorus() {
lpfOut1 = new float[bufferSize];
lpfOut2 = new float[bufferSize];
@ -29,8 +30,7 @@ Chorus::Chorus() {
lfoPhase = 1;
lfoSpeed = 6.283 * 10.7 / sampleRate; // plainly silly value to show if it hasn't been set
gainTC = 1 - exp(-M_PI * 10 / sampleRate); // 1/10th of a second declick
bbdTC = 1 - exp(-M_PI * 60 / sampleRate); // hpf into BBD
gainTC = 1 - exp(-6.283 * 10 / sampleRate);
// not quite Butterworth but you'd never hear the difference
// these are calculated from the real-world component values
@ -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 dly1, frac, flt;
float lfoMod, dly1, frac, flt;
uint16_t tap, delay;
for (uint32_t i = 0; i < frames; i++) {
@ -76,14 +76,11 @@ void Chorus::run(float* input, float** outputs, uint32_t frames) {
hpDelay = flt;
input[i] += (flt * hpGain);
flt = ((input[i] - bbdRC) * bbdTC) + bbdRC;
bbdRC = flt;
ram[delayptr] = input[i] - flt;
ram[delayptr] = input[i];
// delays in milliseconds
#define BASE 0.0035
#define AMT 0.002
#define BASE 0.005
#define AMT 0.00175
dly1 = (BASE + (AMT * lfoPhase)) * sampleRate;
delay = (int)dly1;
@ -106,6 +103,8 @@ void Chorus::run(float* input, float** outputs, uint32_t frames) {
delayptr++;
delayptr &= 0x3ff;
}
//printf("dly1 = %f\n", dly1);
postFilter1l->runSVF(lpfOut1, lpfOut1, frames);
postFilter2l->runSVF(lpfOut1, lpfOut1, frames);
postFilter1r->runSVF(lpfOut2, lpfOut2, frames);
@ -114,6 +113,7 @@ void Chorus::run(float* input, float** outputs, uint32_t frames) {
for (uint32_t i = 0; i < frames; i++) {
float y = input[i];
gainRC = (gain - gainRC) * gainTC + gainRC;
outputs[0][i] = y + (gainRC * lpfOut1[i]);
outputs[1][i] = y + (gainRC * lpfOut2[i]);
}
@ -127,11 +127,11 @@ void Chorus::setHpf(uint8_t mode) {
// k = 1-exp(-2pi * Fc * sampleRate)
switch (mode) {
case 0x00:
hpCut = 1 - exp(-M_PI * 720 / sampleRate);
hpCut = 1 - exp(-6.283 * 720 / sampleRate);
hpGain = -1;
break;
case 0x08:
hpCut = 1 - exp(-M_PI * 225 / sampleRate);
hpCut = 1 - exp(-6.283 * 225 / sampleRate);
hpGain = -1;
break;
case 0x10:
@ -140,7 +140,7 @@ void Chorus::setHpf(uint8_t mode) {
break;
case 0x18:
hpCut = 1 - exp(-6.283 * 85 / sampleRate);
hpGain = 1.0;
hpGain = 1.707;
break;
}
}
@ -154,11 +154,11 @@ void Chorus::setChorus(uint8_t mode) {
break;
case 0x40:
gain = 1.2;
lfoSpeed = M_PI * 0.525 / sampleRate;
lfoSpeed = 6.283 * 0.3 / sampleRate;
break;
case 0x00:
gain = 1.2;
lfoSpeed = M_PI * 0.85 / sampleRate;
lfoSpeed = 6.283 * 0.5 / sampleRate;
break;
}
}

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@ -46,8 +46,6 @@ class Chorus {
float gainRC = 0;
float gainTC = 0;
float bbdRC=0, bbdTC=0;
uint16_t delayptr = 0;

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@ -25,14 +25,13 @@
Module::Module() {
// cutoff frequencies for various RC networks
vcaTC = 1 - exp(-M_PI * 159 / sampleRate); // VCA and VCF 10k/0.1u time constant
subTC = 1 - exp(-M_PI * 15 / sampleRate); // Main VCA and Sub Level 1k + 10u time constant
pwmTC = 1 - exp(-M_PI * 40 / sampleRate); // integrator with 100k/0.047u time constant
vcaTC = 1 - exp(-6.283 * 159 / sampleRate); // VCA and VCF 10k/0.1u time constant
subTC = 1 - exp(-6.283 * 15 / sampleRate); // Main VCA and Sub Level 1k + 10u time constant
pwmTC = 1 - exp(-6.283 * 40 / sampleRate); // integrator with 100k/0.047u time constant
vcaBuf = new float[bufferSize];
subBuf = new float[bufferSize];
pwmBuf = new float[bufferSize];
noiseBuf = new float[bufferSize];
}
Module::~Module() {
@ -42,56 +41,27 @@ Module::~Module() {
delete pwmBuf;
}
void Module::genNoise() {
for (uint32_t i = 0; i < bufferSize; i++) {
noiseRNG *= 0x8088405;
noiseRNG++;
noiseBuf[i] = 1 - (noiseRNG & 0xffff) / 32768.0f;
}
}
void Module::lfoRampOn() {
lfoDelayState = 1;
lfoDelayTimer = 0;
lfoDelay = 0;
}
void Module::runLFO() {
if (lfoDelayState == 1) {
lfoDelayTimer += attackTable[patchRam.lfoDelay];
if (lfoDelayTimer > 0x3fff) lfoDelayState = 2;
}
if ((lfoDelayState == 2)) {
lfoDelay += lfoDelayTable[patchRam.lfoDelay >> 4];
}
if (lfoDelay > 0xff) {
lfoDelayState = 0;
lfoDelay = 0xff;
}
lfoRate = lfoRateTable[patchRam.lfoRate]; // FIXME move to parameters
lfoPhase += (lfoState & 0x01) ? -lfoRate : lfoRate;
if (lfoPhase > 0x1fff) {
lfoPhase = 0x1fff;
lfoState++;
}
if (lfoPhase < 0x0000) {
lfoPhase = 0x0000;
lfoState++;
}
lfo = (lfoState & 0x02) ? -lfoPhase : lfoPhase;
pw = (lfoState & 0x02) ? lfoPhase + 0x2000 : 0x2000 - lfoPhase; // PW LFO is unipolar
pw = (patchRam.switch2 & 0x01) ? 0x3fff : pw; // either LFO or "all on"
pw = 0x3fff - ((pw * (int)(patchRam.pwmLfo*0.9125)) >> 7); // FIXME tidy up this bit
}
void Module::run(Voice* voices, uint32_t blockSize) {
// run updates for module board
int16_t lfoToVco = 0, lfoToVcf = 0;
if (lfoDelayState == 1) {
lfoDelayTimer += lfoDelayTable[patchRam.lfoDelay >> 4];
if (lfoDelayTimer & 0xc000) lfoDelayState = 2;
}
if ((lfoDelayState == 2)) {
lfoDelay += attackTable[patchRam.lfoDelay];
}
if (lfoDelay & 0xc000) {
lfoDelayState = 0;
lfoDelay = 0x3fff;
}
// FIXME break these out to the patch setter
a = attackTable[patchRam.env_a]; // attack time coeff looked up in table
@ -99,26 +69,35 @@ void Module::run(Voice* voices, uint32_t blockSize) {
r = decayTable[patchRam.env_r]; // release time coeff looked up in table
s = patchRam.env_s << 7; // scale 0x00-0x7f to 0x0000-0x3f80
master = powf(2, (patchRam.vca / 31.75 - 4.0f)) * 0.1;
square = (patchRam.switch1 & 0x08) ? 1 : 0;
saw = (patchRam.switch1 & 0x10) ? 1 : 0;
sub = (patchRam.sub / 127.0f) * 1.4;
square = (patchRam.switch1 & 0x08) ? 0.63 : 0;
saw = (patchRam.switch1 & 0x10) ? 0.8 : 0;
sub = patchRam.sub / 127.0f;
lfoPhase += lfoRateTable[patchRam.lfoRate];
res = patchRam.vcfReso / 127.0;
noise = (patchRam.noise / 127.0);
noise = patchRam.noise / 127.0;
// FIXME the exp in these is expensive, don't call it all the time
chorus->setChorus(patchRam.switch1 & 0x60);
chorus->setHpf(patchRam.switch2 & 0x18);
runLFO();
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;
// calculate "smoothed" parameters
// these are single outputs with heavy RC smoothing
for (uint32_t i = 0; i < blockSize; i++) {
vcaRC = (master - vcaRC) * subTC + vcaRC;
pwmRC = ((pw / 32768.0f) - pwmRC) * pwmTC + pwmRC;
pwmRC = (pw - pwmRC) * pwmTC + pwmRC;
subRC = (sub - subRC) * vcaTC + subRC;
vcaBuf[i] = vcaRC;
@ -128,28 +107,13 @@ void Module::run(Voice* voices, uint32_t blockSize) {
if (bufPtr < bufferSize) bufPtr++;
}
lfoToVco = (lfoDepthTable[patchRam.vcoLfo] * lfoDelay) >> 8; // lookup table is 0-255
lfoToVco += ((int)(modWheel * modDepth));
int16_t vcf = (patchRam.vcfEnv << 7) * ((patchRam.switch2 & 0x02) ? -1 : 1);
if (lfoToVco > 0xff) lfoToVco = 0xff;
lfoToVco = (lfo * lfoToVco) >> 11; // 8 for normalisation plus three additional DSLR EA
int16_t pitchBase = 0x1818;
pitchBase += (lfo * lfoDepthTable[patchRam.vcoLfo]) >> 11;
lfoToVcf = (patchRam.vcfLfo * lfoDelay) >> 7; // value is 0-127
lfoToVcf = (lfo * lfoToVcf) >> 9; // 8 for normalisation plus one additional DSLR EA
int16_t pitchBase = 0x1818, vcfBase = 0;
pitchBase += lfoToVco;
pitchBase += vcoBendDepth * bend;
vcfBase = (patchRam.vcfFreq << 7);
vcfBase += lfoToVcf;
vcfBase += vcfBendDepth * bend;
if (vcfBase > 0x3fff) vcfBase = 0x3fff;
if (vcfBase < 0x0000) vcfBase = 0x0000;
// per-voice calculations
for (uint32_t i = 0; i < NUM_VOICES; i++) {
// run one step of the envelope
// maybe move all this into voice.cpp FIXME
Voice* v = &voices[i];
switch (v->envPhase) {
case 0: // release phase FIXME use an enum I guess
@ -168,30 +132,23 @@ void Module::run(Voice* voices, uint32_t blockSize) {
}
// pitch
uint16_t pitch = pitchBase + (v->note << 8);
int8_t semi = pitch >> 8;
semi -= 36;
// FIXME clean this all up a bit
int16_t pitch = pitchBase + (v->note << 8);
int16_t semi = pitch >> 8;
float frac = (pitch & 0xff) / 256.0;
if (semi < 0) {
semi = 0;
frac = 0;
}
if (semi >= 103) {
semi = 103;
frac = 0;
};
float p1 = pitchTable[semi], p2 = pitchTable[semi + 1];
int16_t px = ((p2 - p1) * frac + p1); // interpolated pitch from table
// octave divider
float p1 = pitchTable[semi], p2 = pitchTable[semi + 1];
int16_t px = ((p2 - p1) * frac + p1);
px *= (patchRam.switch1 & 0x07);
v->omega = px / sampleRate; // FIXME recalculate table using proper scaler
v->omega = px / (sampleRate * 8.0f); // fixme use proper scaler
// per voice we need to calculate the key follow amount and envelope amount
v->vcfCut = vcfBase + (((v->env * patchRam.vcfEnv) >> 7) * ((patchRam.switch2 & 0x02) ? -1 : 1));
v->vcfCut = (patchRam.vcfFreq << 7) + ((vcf * v->env) >> 14);
v->vcfCut += (lfo * patchRam.vcfLfo) >> 9;
v->vcfCut += (int)((v->note - 60) * (patchRam.vcfKey << 1) * 0.375);
v->vcfCut += (int)((v->note - 36) * (patchRam.vcfKey << 1) * 0.375);
if (v->vcfCut > 0x3fff) v->vcfCut = 0x3fff;
if (v->vcfCut < 0) v->vcfCut = 0;

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@ -33,37 +33,87 @@ 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;
/*
#if 0
struct {
uint8_t lfoRate = 0x58;
uint8_t lfoDelay = 0x00;
uint8_t vcoLfo = 0x00;
uint8_t pwmLfo = 0x3b;
uint8_t noise = 0x00;
uint8_t vcfFreq = 0x25; // 1c; // 0x3f80
uint8_t vcfReso = 0x6a;
uint8_t vcfEnv = 0x25; // 4e;
uint8_t vcfLfo = 0x00;
uint8_t vcfKey = 0x00; // 47;
uint8_t vca = 0x35;
uint8_t env_a = 0x00;
uint8_t env_d = 0x3c;
uint8_t env_s = 0x00; // 0x3f80
uint8_t env_r = 0x3c;
uint8_t sub = 0x7f;
uint8_t switch1 = 0x4a;
uint8_t switch2 = 0x18;
} patchRam;
#else
struct {
uint8_t lfoRate = 0x40;
uint8_t lfoDelay = 0x00;
uint8_t vcoLfo = 0x00;
uint8_t pwmLfo = 0x00;
uint8_t noise = 0x01;
uint8_t vcfFreq = 0x31;
uint8_t vcfReso = 0x7f;
uint8_t vcfEnv = 0x00;
uint8_t vcfLfo = 0x00;
uint8_t vcfKey = 0x7f;
uint8_t vca = 0x40;
uint8_t env_a = 0x00;
uint8_t env_d = 0x00;
uint8_t env_s = 0x00; // 0x3f80
uint8_t env_r = 0x00;
uint8_t sub = 0x00;
uint8_t switch1 = 0x22;
uint8_t switch2 = 0x1d;
} patchRam;
#endif
*/
float saw = 0, square = 0, sub = 0, noise = 0, master = 0;
int16_t bend = 0, modWheel=0;
float vcoBendDepth = 4, vcfBendDepth=1.5, modDepth=.5;
struct {
uint8_t lfoRate = 0x1f;
uint8_t lfoRate = 0x58;
uint8_t lfoDelay = 0x00;
uint8_t vcoLfo = 0x00;
uint8_t pwmLfo = 0x3c;
uint8_t pwmLfo = 0x00;
uint8_t noise = 0x00;
uint8_t vcfFreq = 0x25; // 1c; // 0x3f80
uint8_t vcfReso = 0x1d;
uint8_t vcfEnv = 0x1c; // 4e;
uint8_t vcfFreq = 0x00; // 1c; // 0x3f80
uint8_t vcfReso = 0x7f;
uint8_t vcfEnv = 0x7f; // 4e;
uint8_t vcfLfo = 0x00;
uint8_t vcfKey = 0x2b; // 47;
uint8_t vca = 0x5c;
uint8_t vcfKey = 0x00; // 47;
uint8_t vca = 0x20;
uint8_t env_a = 0x00;
uint8_t env_d = 0x2a;
uint8_t env_s = 0x23; // 0x3f80
uint8_t env_r = 0x00;
uint8_t sub = 0x40;
uint8_t switch1 = 0x19;
uint8_t switch2 = 0x18;
uint8_t env_d = 0x5c;
uint8_t env_s = 0x00; // 0x3f80
uint8_t env_r = 0x3c;
uint8_t sub = 0x7f;
uint8_t switch1 = 0x3a;
uint8_t switch2 = 0x19;
} patchRam;
Chorus* chorus;
@ -74,22 +124,12 @@ 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;
int16_t lfoPhase;
uint8_t lfoState = 0;
uint16_t lfoRate;
uint32_t noiseRNG = 1;
uint16_t lfoDelay = 0;
uint8_t lfoDelayState = 0;
uint16_t lfoDelayTimer = 0;
@ -102,7 +142,7 @@ class Voice {
Voice();
void on(uint8_t midiNote);
void off();
void run(Module* m, float* buffer, uint32_t framePos, uint32_t samples);
void run(Module* m, float* buffer, uint32_t samples);
private:
float omega = 0, theta = 0; // phase increment and angle FIXME better names
@ -110,16 +150,18 @@ class Voice {
uint8_t pulseStage = 1; // pulse wave phase
float subosc = 1; // sub oscillator flipflop output
uint8_t envPhase = 0; // current running state of envelope
int16_t env = 0; // calculated envelope amount
int16_t vcfCut; // calculated cutoff to filter
int16_t vcaEnv; // calculated level to VCA (env/gate)
float vcaRC = 0, vcfRC = 0; // RC circuit state values
uint8_t envPhase = 0;
int16_t env = 0; // output amplitude
uint16_t vcfCut;
int16_t vcaEnv;
float vcaRC = 0, vcfRC = 0;
uint8_t note = 0;
// filter
float y0 = 0, y1 = 0, y2 = 0, y3 = 0;
double s[4] = {0, 0, 0, 0};
float zi = 0;
};
#endif

View File

@ -309,35 +309,19 @@ void Peacock::initParameter(uint32_t index, Parameter& parameter) {
enumValues[2].label = "Fast";
parameter.enumValues.values = enumValues;
}
break;
case pVcoBend:
parameter.hints = kParameterIsAutomatable;
parameter.name = "VCO Bend";
parameter.symbol = "pfau_vcobend";
parameter.ranges.min = 0.0f;
parameter.ranges.max = 12.0f;
parameter.ranges.def = 4.0f;
break;
case pVcfBend:
parameter.hints = kParameterIsAutomatable;
parameter.name = "VCF Bend";
parameter.symbol = "pfau_vcfbend";
parameter.ranges.min = 0.0f;
parameter.ranges.max = 42.0f;
parameter.ranges.def = 4.0f;
break;
case pModDepth:
parameter.hints = kParameterIsAutomatable;
parameter.name = "Mod Depth";
parameter.symbol = "pfau_moddepth";
parameter.ranges.min = 0.0f;
parameter.ranges.max = 4.0f;
parameter.ranges.def = 0.5f;
break;
/*
case pModWheel:
parameter.hints = kParameterIsAutomatable | kParameterIsHidden;
parameter.name = "Mod wheel";
parameter.symbol = "pfau_modwheel";
parameter.ranges.min = 0.0f;
parameter.ranges.max = 127.0f;
parameter.ranges.def = 0.0f;
parameter.midiCC = 1;
break;
*/
}
// chorus, porta, bend range, key mode still to do
}
void Peacock::setParameterValue(uint32_t index, float value) {
@ -357,7 +341,7 @@ void Peacock::setParameterValue(uint32_t index, float value) {
m->patchRam.vcoLfo = value;
break;
case pPWMDepth:
m->patchRam.pwmLfo = value;
m->patchRam.pwmLfo = value / 1.27;
break;
case pSubLevel:
m->patchRam.sub = value;
@ -450,15 +434,10 @@ void Peacock::setParameterValue(uint32_t index, float value) {
m->patchRam.switch2 &= 0xe7;
m->patchRam.switch2 |= (3 - (int)value) << 3;
break;
case pVcoBend:
m->vcoBendDepth = value;
break;
case pVcfBend:
m->vcfBendDepth = value / 2.66f;
break;
case pModDepth:
m->modDepth = value;
break;
/*
case pModWheel:
//s.ff64 = (int)value << 1;
break;*/
}
}
@ -487,7 +466,7 @@ float Peacock::getParameterValue(uint32_t index) const {
return m->patchRam.vcoLfo;
break;
case pPWMDepth:
return m->patchRam.pwmLfo;
return m->patchRam.pwmLfo * 1.27f;
break;
case pPWMMode:
@ -557,16 +536,6 @@ float Peacock::getParameterValue(uint32_t index) const {
default:
break;
}
break;
case pVcoBend:
return m->vcoBendDepth;
break;
case pVcfBend:
return m->vcfBendDepth * 2.66f;
break;
case pModDepth:
return m->modDepth;
break;
}
return 0;
}

View File

@ -28,7 +28,6 @@ Peacock::Peacock() : Plugin(parameterCount, 0, 0) {
sampleRate = getSampleRate();
bufferSize = getBufferSize();
chorus = new Chorus();
m = new Module();
ic1 = new Assigner;
@ -71,27 +70,27 @@ 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; // reset the output buffer pointer
m->bufPtr = 0;
runMidi(midiEvents, midiEventCount, blockLeft);
while (framePos < frames) {
if (blockLeft == 0) {
// no more samples to calculate in this update period
blockLeft = sampleRate / 233.5; // update rate in Hz, measured
blockLeft = sampleRate / 238; // update rate in Hz
runMidi(midiEvents, midiEventCount, framePos + blockLeft);
m->run(voice, blockLeft);
}
// how many frames to do? Are we about to run off an update block
sizeThisTime = (framesLeft < blockLeft) ? framesLeft : blockLeft;
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;
@ -100,7 +99,7 @@ void Peacock::run(const float**, float** outputs, uint32_t frames, const MidiEve
}
// now we've assembled a full chunk of audio
// memcpy(outputs[0], m->vcaBuf, sizeof(float)* frames);
//memcpy(outputs[0], m->vcaBuf, sizeof(float)* frames);
chorus->run(outputs[0], outputs, frames);
}

View File

@ -89,19 +89,19 @@ uint16_t lfoDelayTable[8] = {
0xffff, 0x0419, 0x020c, 0x015e, 0x0100, 0x0100, 0x0100, 0x0100};
float pitchTable[104] = {
8.123, 8.607, 9.122, 9.664, 10.240, 10.850, 11.497, 12.183,
12.908, 13.678, 14.492, 15.355, 16.268, 17.236, 18.265, 19.351,
20.504, 21.723, 23.019, 24.389, 25.841, 27.382, 29.011, 30.733,
32.561, 34.497, 36.555, 38.730, 41.034, 43.474, 46.066, 48.804,
51.712, 54.795, 58.052, 61.493, 65.147, 69.023, 73.142, 77.495,
82.102, 86.987, 92.166, 97.637, 103.455, 109.625, 116.144, 123.001,
130.310, 138.083, 146.327, 155.039, 164.258, 174.034, 184.366, 195.312,
206.954, 219.298, 232.342, 246.063, 260.688, 276.243, 292.740, 310.174,
328.515, 348.189, 368.732, 390.625, 413.907, 438.596, 464.684, 492.126,
521.376, 552.486, 585.480, 620.347, 657.030, 696.379, 737.463, 781.250,
827.815, 877.193, 929.368, 984.252, 1043.841, 1106.195, 1170.960, 1240.695,
1315.789, 1392.758, 1474.926, 1562.500, 1655.629, 1754.386, 1858.736, 1968.504,
2083.333, 2212.389, 2347.418, 2487.562, 2631.579, 2793.296, 2958.580, 3125.000,
32.494, 34.430, 36.486, 38.658, 40.962, 43.399, 45.990, 48.731,
51.633, 54.711, 57.969, 61.419, 65.072, 68.944, 73.059, 77.405,
82.014, 86.892, 92.077, 97.556, 103.365, 109.529, 116.043, 122.933,
130.242, 137.988, 146.220, 154.919, 164.136, 173.898, 184.264, 195.217,
206.847, 219.178, 232.207, 245.972, 260.586, 276.091, 292.569, 309.981,
328.407, 347.947, 368.664, 390.549, 413.822, 438.500, 464.576, 492.005,
521.241, 552.334, 585.309, 620.155, 657.030, 696.136, 737.463, 781.250,
827.815, 877.193, 929.368, 984.252, 1042.753, 1104.972, 1170.960, 1240.695,
1314.060, 1392.758, 1474.926, 1562.500, 1655.629, 1754.386, 1858.736, 1968.504,
2085.506, 2209.945, 2341.920, 2481.390, 2628.121, 2785.515, 2949.853, 3125.000,
3311.258, 3508.772, 3717.472, 3937.008, 4175.365, 4424.779, 4683.841, 4962.779,
5263.158, 5571.031, 5899.705, 6250.000, 6622.517, 7017.544, 7434.944, 7874.016,
8333.333, 8849.558, 9389.671, 9950.249, 10526.316, 11173.184, 11834.320, 12500.000
};
#endif

View File

@ -258,17 +258,15 @@ void DistrhoUIPeacock::parameterChanged(uint32_t index, float value) {
sw1 &= 0xf8; // mask
if (value > 2) value = 2;
sw1 |= (1 << (int)value);
repaint();
xBtn16ft->repaint(); // will repaint all the panel
break;
case pSqr:
sw1 &= 0xf7;
sw1 |= ((value >= 0.5)) << 3;
repaint();
break;
case pSaw:
sw1 &= 0xef;
sw1 |= (value > 0.5) << 4;
repaint();
break;
case pChorusMode:
@ -285,8 +283,6 @@ void DistrhoUIPeacock::parameterChanged(uint32_t index, float value) {
sw1 |= 0x00;
break;
}
repaint();
break;
}
}
@ -344,7 +340,6 @@ void DistrhoUIPeacock::imageButtonClicked(ImageButton* imgBtn, int) {
default:
break;
}
repaint();
}
void DistrhoUIPeacock::onDisplay() {

View File

@ -40,9 +40,9 @@ Voice::Voice() {
}
void Voice::on(uint8_t midiNote) {
while (midiNote < 24) midiNote += 12; // limit lowest note to C1
while (midiNote > 108) midiNote -= 12; // limit highest note to C8
note = midiNote;
while (midiNote < 24) midiNote += 12;
while (midiNote > 108) midiNote -= 12;
note = midiNote - 24;
envPhase = 1;
}
@ -50,19 +50,28 @@ void Voice::off() {
envPhase = 0;
}
void Voice::run(Module* m, float* buffer, uint32_t framePos, uint32_t samples) {
// tanh(x)/x approximation, flatline at very high inputs
// 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 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) {
// 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)); // FIXME explain magic numbers
float cut = 261.0f * (powf(2, (vcfCut - 0x1880) / 1143.0f));
cut = M_PI * cut / sampleRate;
cut = cut / (1 + cut); // correct tuning warp
if (cut > 0.7) cut = 0.7;
// if (cut > 0.7) cut = 0.7;
double r = 5 * m->res;
float r = 6 * m->res;
float amp = vcaEnv / 32768.0f;
float amp = vcaEnv / 4096.0f;
for (uint32_t i = 0; i < samples; i++) {
out = delay;
@ -94,31 +103,109 @@ void Voice::run(Module* m, float* buffer, uint32_t framePos, 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 * m->noiseBuf[i + framePos];
out += m->noise * (0.8 - 1.6 * (rand() & 0xffff) / 65536.0);
// out *= 0.1;
// 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 > 6) fb = 6;
if (fb < -6) fb = -6;
if (fb > 4) fb = 4;
if (fb < -4) fb = -4;
// fb = 1.5 * fb - 0.5 * fb * fb * fb;
//
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[framePos + i] += m->vcaBuf[i] * vcaRC * y3;
buffer[i] += m->vcaBuf[i] * vcaRC * out;
lastpw = m->pwmBuf[i];
}
// buffer[0] += 1; // buzzing noise to test
// buffer[0] += 1;
}