some cleanup and simplification, register filtering tweaks
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b70ab64dc1
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7b138536ef
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@ -138,6 +138,10 @@ void Assigner::noteOn(uint8_t note) {
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// limit highest note to C7, one octave above the Solina's maximum range
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while(note>96) note -= 12;
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// limit lowest note to C2, too
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while(note<36) note += 12;
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// I'm not sure these are correct; these string ensembles don't have MIDI
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// so it's all a bit of a guess
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voices[v].startNote(note);
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@ -20,7 +20,6 @@
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#define _ASSIGNER_HPP
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#include "DistrhoPlugin.hpp"
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#include "generator.hpp"
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class Assigner {
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@ -31,12 +30,12 @@ class Assigner {
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private:
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void noteOn(uint8_t note); // incoming note on (or off, if velocity = 0)
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void noteOff(uint8_t note); // incoming note off
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void dumpTables();
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uint8_t voiceTbl[NUM_VOICES]; // voices in order of use
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uint8_t noteTbl[NUM_VOICES]; // note played by voice
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Voice *voices; // used to gain access to generator voices
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Voice *voices;
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void dumpTables();
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};
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#endif
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@ -36,12 +36,11 @@ Chorus::Chorus(uint32_t xbufferSize, double xsampleRate) { // no parameters, pr
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fastPhase = 0;
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slowPhase = 0;
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preFilter = new SVF();
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postFilter1 = new SVF();
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postFilter2 = new SVF();
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postFilter1 = new SVF(8000, 1.3);
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postFilter2 = new SVF(8000, 0.54);
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// lfo values taken from a rough simulation
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fastOmega = 6.283 * 6.8 / sampleRate; // approximate, can be adjusted
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fastOmega = 6.283 * 5.7 / sampleRate; // approximate, can be adjusted
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slowOmega = 6.283 * 0.7 / sampleRate; // again approximate
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// zero out the delay buffer
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@ -49,9 +48,6 @@ Chorus::Chorus(uint32_t xbufferSize, double xsampleRate) { // no parameters, pr
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memset(lpfIn, 0, sizeof(float) * bufferSize);
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memset(lpfOut1, 0, sizeof(float) * bufferSize);
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memset(lpfOut2, 0, sizeof(float) * bufferSize);
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preFilter->setCutoff(12600, 1.3, sampleRate);
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postFilter1->setCutoff(11653, 6.6, sampleRate);
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postFilter2->setCutoff(5883, 1.1, sampleRate);
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}
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Chorus::~Chorus() {
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@ -59,7 +55,6 @@ Chorus::~Chorus() {
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delete lpfOut1;
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delete lpfOut2;
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delete ram;
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delete preFilter;
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delete postFilter1;
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delete postFilter2;
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}
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@ -72,9 +67,6 @@ void Chorus::run(const float *input, float **outputs, uint32_t frames) {
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float lfoMod, dly1, frac;
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uint16_t tap, delay;
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// filter the input
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preFilter->runSVF(input, lpfIn, frames);
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for (uint32_t i = 0; i < frames; i++) {
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// run a step of LFO
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fastPhase += fastOmega;
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@ -82,7 +74,7 @@ void Chorus::run(const float *input, float **outputs, uint32_t frames) {
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slowPhase += slowOmega;
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if (slowPhase > 6.283) slowPhase -= 6.283;
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ram[delayptr] = lpfIn[i];
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ram[delayptr] = input[i];
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// lowpass filter
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@ -93,11 +85,11 @@ void Chorus::run(const float *input, float **outputs, uint32_t frames) {
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// 120deg 0.248 slow 0.745 fast
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// 240deg 0.252 slow 0.609 fast
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#define BASE 0.005
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#define AMT 0.0015
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#define BASE 0.05
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#define AMT 0.00175
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// 0 degree delay line
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lfoMod = 0.203 * sin(fastPhase) + 0.635 * sin(slowPhase);
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lfoMod = 0.203 * sin(fastPhase) + 0.835 * sin(slowPhase);
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dly1 = (BASE + (AMT * lfoMod)) * sampleRate;
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delay = (int)dly1;
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frac = dly1 - delay;
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@ -119,7 +111,7 @@ void Chorus::run(const float *input, float **outputs, uint32_t frames) {
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out120 = ((s1 - s0) * frac) + s0;
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// 240 degree delay line
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lfoMod = 0.252 * sin(fastPhase + 4.18) + 0.609 * sin(slowPhase + 4.18);
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lfoMod = 0.252 * sin(fastPhase + 4.18) + 0.809 * sin(slowPhase + 4.18);
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dly1 = (BASE + (AMT * lfoMod)) * sampleRate;
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delay = (int)dly1;
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frac = dly1 - delay;
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@ -23,24 +23,31 @@
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#include <cstdio>
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#include "DistrhoPluginInfo.h"
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#include "svf.hpp"
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// some unit-local globals
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extern double sampleRate;
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extern uint32_t bufferSize;
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float sampleRate = 0;
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// unit-local global
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static float envTc[2];
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float envTc[2];
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Generator::Generator(uint32_t bufferSize, double xSampleRate) {
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sampleRate = xSampleRate;
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// Generator::Generator(uint32_t bufferSize, double xSampleRate) {
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Generator::Generator() {
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// sampleRate = xSampleRate;
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output = new float[bufferSize];
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// create the phase increments for each semitone
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for (uint8_t i = 0; i < 12; i++) {
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phase[i] = 0;
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uint32_t f;
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f = (1 << 31) * (65.406 * powf(2, 0.083334 * (i + 12)) / sampleRate);
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f = (1 << 31) * (32.703 * powf(2, 0.083334 * i) / sampleRate);
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omega[i] = f;
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}
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// output filters
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// trumpet 4200 2.5
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// horn 1500 2.5
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tr2 = SVF(9500.0f, 0.707f);
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setEnvelope(0.5, 0.5);
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}
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Generator::~Generator() {
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@ -50,7 +57,7 @@ Generator::~Generator() {
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void Generator::runBlock(uint32_t frames) {
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Voice *v;
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uint32_t i;
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uint8_t k, p, key, n1, n2, d;
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uint8_t k, p, d;
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float n;
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memset(output, 0, frames * sizeof(float));
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@ -63,7 +70,7 @@ void Generator::runBlock(uint32_t frames) {
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for (k = 0; k < NUM_VOICES; k++) {
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v = &voices[k];
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d = (phase[v->semi] & (0x40000000 >> v->oct)) != 0;
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d = (phase[v->semi] & (0x20000000 >> v->oct)) != 0;
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n = d ? 0.25 : -0.25;
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v->vc34 = ((n - v->vc34) * v->c34) + v->vc34;
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n -= v->vc34;
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@ -71,7 +78,7 @@ void Generator::runBlock(uint32_t frames) {
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v->vc78 = ((n - v->vc78) * v->c78) + v->vc78;
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v->vc107 = ((v->vc78 - v->vc107) * v->c107) + v->vc107;
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d = (phase[v->semi] & (0x80000000 >> v->oct)) != 0;
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d = (phase[v->semi] & (0x40000000 >> v->oct)) != 0;
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n = d ? 0.25 : -0.25;
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v->vc33 = ((n - v->vc33) * v->c33) + v->vc33;
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n -= v->vc33;
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@ -87,6 +94,7 @@ void Generator::runBlock(uint32_t frames) {
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output[i] += 0.25 * (v4 + v8) * v->vca;
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}
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}
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tr2.runSVF(output, output, frames);
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}
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void Voice::startNote(uint8_t key) {
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@ -117,6 +125,6 @@ void Voice::stopNote() {
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}
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void Generator::setEnvelope(float attack, float sustain) {
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envTc[0] = ((96* powf(100, -attack))/sampleRate);
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envTc[1] = ((48* powf(100, -sustain))/sampleRate);
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envTc[0] = ((96 * powf(100, -attack)) / sampleRate);
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envTc[1] = ((48 * powf(100, -sustain)) / sampleRate);
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}
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@ -22,6 +22,7 @@
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#include <stdint.h>
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#include "DistrhoPluginInfo.h"
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#include "svf.hpp"
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class Generator;
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@ -46,7 +47,9 @@ class Voice {
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class Generator {
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public:
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Generator(uint32_t bufferSize, double xSampleRate);
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// Generator(uint32_t bufferSize, double xSampleRate);
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Generator();
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~Generator();
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void setEnvelope(float attack, float sustain);
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void runBlock(uint32_t frames);
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@ -56,6 +59,7 @@ class Generator {
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private:
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uint32_t phase[12];
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uint32_t omega[12];
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SVF tr2;
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};
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#endif
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@ -18,14 +18,20 @@
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#include "sonnenlicht.hpp"
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double sampleRate;
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uint32_t bufferSize;
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START_NAMESPACE_DISTRHO
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Sonnenlicht::Sonnenlicht() : Plugin(kParameterCount, 0, 0), fSampleRate(getSampleRate()) {
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genny = new Generator(getBufferSize(), fSampleRate);
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Sonnenlicht::Sonnenlicht() : Plugin(kParameterCount, 0, 0) {
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// genny = new Generator(getBufferSize(), fSampleRate);
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sampleRate = getSampleRate();
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bufferSize = getBufferSize();
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genny = new Generator();
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assigner = new Assigner(genny->voices);
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chorus = new Chorus(getBufferSize(), fSampleRate);
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chorus = new Chorus(getBufferSize(), sampleRate);
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}
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Sonnenlicht::~Sonnenlicht() {
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@ -57,8 +63,7 @@ void Sonnenlicht::run(const float**, float** outputs, uint32_t frames,
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genny->runBlock(frames);
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if (prog.enableChorus) {
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chorus->run(genny->output, outputs, frames);
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chorus->run(genny->output, outputs, frames);
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} else {
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memcpy(outputs[0], genny->output, frames * sizeof(float));
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memcpy(outputs[1], genny->output, frames * sizeof(float));
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@ -77,7 +77,7 @@ class Sonnenlicht : public Plugin {
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const MidiEvent *midiEvents, uint32_t midiEventCount) override;
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private:
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double fSampleRate;
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//double fSampleRate;
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Program prog;
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Assigner *assigner;
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@ -22,16 +22,14 @@
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#include <cstdio>
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SVF::SVF() {
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SVF::SVF(float cutoff, float Q) {
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// zero out all values
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z1 = 0;
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z2 = 0;
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c1 = 0;
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c2 = 0;
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d0 = 0;
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setCutoff(cutoff, Q);
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}
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void SVF::setCutoff(float cutoff, float Q, float sampleRate) {
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void SVF::setCutoff(float cutoff, float Q) {
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float F = cutoff / sampleRate;
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float w = 2 * tan(3.14159 * F);
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float a = w / Q;
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@ -45,6 +43,7 @@ void SVF::setCutoff(float cutoff, float Q, float sampleRate) {
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void SVF::runSVF(const float *input, float *output, uint32_t frames) {
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float x;
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for (uint32_t i = 0; i < frames; i++) {
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// lowpass filter
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x = input[i] - z1 - z2;
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@ -21,10 +21,13 @@
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#include <stdint.h>
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extern double sampleRate;
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class SVF {
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public:
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SVF();
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void setCutoff(float cutoff, float Q, float sampleRate);
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SVF(float cutoff=1000, float Q=0.707);
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void setCutoff(float cutoff, float Q);
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void runSVF(const float *input, float *output, uint32_t frames);
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protected:
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