built-in triple chorus
This commit is contained in:
Gordon JC Pearce
parent
24fd4718f5
commit
6bde8378ed
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@ -9,7 +9,7 @@
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NAME = sonnenlicht
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FILES_DSP = assigner.cpp generator.cpp sonnenlicht.cpp
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FILES_DSP = assigner.cpp generator.cpp chorus.cpp svf.cpp sonnenlicht.cpp
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include ../dpf/Makefile.plugins.mk
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TARGETS += vst2 vst3 jack lv2_dsp
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@ -0,0 +1,152 @@
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/*
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sonnenlicht poly ensemble
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Copyright 2025 Gordon JC Pearce <gordonjcp@gjcp.net>
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Permission to use, copy, modify, and/or distribute this software for any
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purpose with or without fee is hereby granted, provided that the above
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copyright notice and this permission notice appear in all copies.
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THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
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WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
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MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY
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SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
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WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN ACTION
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OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF OR IN
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CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
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*/
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#include "chorus.hpp"
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#include <math.h>
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#include <string.h>
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#include <cstdio>
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Chorus::Chorus(uint32_t xbufferSize, double xsampleRate) { // no parameters, programs, or states
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lpfIn = new float[bufferSize];
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lpfOut1 = new float[bufferSize];
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lpfOut2 = new float[bufferSize];
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ram = new float[DELAYSIZE]; // probably needs to be calculated based on sample rate
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sampleRate = xsampleRate;
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// lfo values taken from a rough simulation
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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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fastOmega = 6.283 * 6.8 / 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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memset(ram, 0, sizeof(float) * DELAYSIZE);
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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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// calculate SVF params
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// hardcoded values for now
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// this is the pre-chorus filter based around TR2
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// It's actually a Sallen-Key filter which is easy to realise in hardware
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// however a State Variable Filter is far easier to realise in software
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// simple is good, and using a little maths we can work out that for
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// R55 = R56 = 22k, C54 = 1.5nF, C76 = 220pF then the filter is at
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// 12.6kHz and a Q of about 1.3
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//
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// Here is the best writeup ever on SVFs
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// https://kokkinizita.linuxaudio.org/papers/digsvfilt.pdf
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}
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Chorus::~Chorus() {
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delete lpfIn;
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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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void Chorus::run(const float *input, float **outputs, uint32_t frames) {
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// actual effects here
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// now run the DSP
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float out0 = 0, out120 = 0, out240 = 0, s0 = 0, s1 = 0;
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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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//memcpy(lpfIn, input, sizeof(float) * 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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if (fastPhase > 6.283) fastPhase -= 6.283;
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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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// lowpass filter
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// now we need to calculate the delay
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// I don't know how long the Solina's delay lines are so I'm guessing 2-4ms for now
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// normalised mod depths, from a quick simulation of the LFO block:
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// 0deg 0.203 slow 0.635 fast
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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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// 0 degree delay line
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lfoMod = 0.203 * sin(fastPhase) + 0.635 * 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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tap = delayptr - delay;
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s1 = ram[(tap - 1) & 0x3ff];
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s0 = ram[tap & 0x3ff];
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out0 = ((s1 - s0) * frac) + s0;
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// 120 degree delay line
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lfoMod = 0.248 * sin(fastPhase + 2.09) + 0.745 * sin(slowPhase + 2.09);
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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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tap = delayptr - delay;
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s1 = ram[(tap - 1) & 0x3ff];
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s0 = ram[tap & 0x3ff];
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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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dly1 = (BASE + (AMT * lfoMod)) * sampleRate;
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delay = (int)dly1;
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frac = dly1 - delay;
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tap = delayptr - delay;
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s1 = ram[(tap - 1) & 0x3ff];
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s0 = ram[tap & 0x3ff];
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out240 = ((s1 - s0) * frac) + s0;
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lpfOut1[i] = (out0 + out120 + out240) / 3;
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delayptr++;
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delayptr &= 0x3ff;
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}
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postFilter1->runSVF(lpfOut1, lpfOut2, frames);
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postFilter2->runSVF(lpfOut2, outputs[0], frames);
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}
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@ -0,0 +1,48 @@
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/*
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sonnenlicht poly ensemble
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Copyright 2025 Gordon JC Pearce <gordonjcp@gjcp.net>
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Permission to use, copy, modify, and/or distribute this software for any
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purpose with or without fee is hereby granted, provided that the above
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copyright notice and this permission notice appear in all copies.
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THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
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WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
|
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MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY
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SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
|
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WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN ACTION
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OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF OR IN
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CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
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*/
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#ifndef Chorus_HPP
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#define Chorus_HPP
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#include "svf.hpp"
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// total size of delay line buffer
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#define DELAYSIZE 1028
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class Chorus {
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public:
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Chorus(uint32_t bufferSize, double sampleRate);
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~Chorus();
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void run(const float *input, float **outputs, uint32_t frames);
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private:
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uint32_t bufferSize;
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double sampleRate;
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double fastPhase, fastOmega;
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double slowPhase, slowOmega;
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double fastLfo, slowLfo;
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uint16_t delayptr;
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float *ram;
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float *lpfIn;
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float *lpfOut1, *lpfOut2;
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SVF *preFilter, *postFilter1, *postFilter2;
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};
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#endif
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@ -18,21 +18,21 @@
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#include "sonnenlicht.hpp"
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#include "assigner.hpp"
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START_NAMESPACE_DISTRHO
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Sonnenlicht::Sonnenlicht() : Plugin(kParameterCount, 0, 0), fSampleRate(getSampleRate()) {
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printf("initialiser called\n");
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genny = new Generator(getBufferSize());
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genny->setupGenerator(fSampleRate);
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assigner = new Assigner(genny->voices);
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chorus = new Chorus(getBufferSize(), fSampleRate);
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}
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Sonnenlicht::~Sonnenlicht() {
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delete assigner;
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delete genny;
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delete chorus;
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}
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void Sonnenlicht::setParameterValue(uint32_t index, float value) {
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@ -66,8 +66,8 @@ void Sonnenlicht::run(const float**, float** outputs, uint32_t frames,
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}
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genny->runBlock(assigner->noteTbl, frames);
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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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chorus->run(genny->output, outputs, frames);
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}
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Plugin* createPlugin() { return new Sonnenlicht(); }
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@ -22,6 +22,7 @@
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#include "DistrhoPlugin.hpp"
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#include "assigner.hpp"
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#include "generator.hpp"
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#include "chorus.hpp"
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START_NAMESPACE_DISTRHO
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@ -61,6 +62,7 @@ class Sonnenlicht : public Plugin {
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double fSampleRate;
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Assigner *assigner;
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Generator *genny;
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Chorus *chorus;
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DISTRHO_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR(Sonnenlicht);
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};
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@ -0,0 +1,56 @@
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/*
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State Variable Filter
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Copyright 2025 Gordon JC Pearce <gordonjcp@gjcp.net>
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Permission to use, copy, modify, and/or distribute this software for any
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purpose with or without fee is hereby granted, provided that the above
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copyright notice and this permission notice appear in all copies.
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||||
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THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
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WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
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MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY
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SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
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WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN ACTION
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OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF OR IN
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CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
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*/
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#include "svf.hpp"
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#include <math.h>
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#include <cstdio>
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SVF::SVF() {
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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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}
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void SVF::setCutoff(float cutoff, float Q, float sampleRate) {
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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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float b = w * w;
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// "corrected" SVF params, per Fons Adriaensen
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c1 = (a + b) / (1 + a / 2 + b / 4);
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c2 = b / (a + b);
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d0 = c1 * c2 / 4;
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}
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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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z2 += c2 * z1;
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z1 += c1 * x;
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output[i] = d0 * x + z2;
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}
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// printf("%f\n", x);
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}
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@ -0,0 +1,36 @@
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/*
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State Variable Filter
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Copyright 2025 Gordon JC Pearce <gordonjcp@gjcp.net>
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Permission to use, copy, modify, and/or distribute this software for any
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||||
purpose with or without fee is hereby granted, provided that the above
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copyright notice and this permission notice appear in all copies.
|
||||
|
||||
THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
|
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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
|
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CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
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*/
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#ifndef SVF_HPP
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#define SVF_HPP
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#include <stdint.h>
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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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void runSVF(const float *input, float *output, uint32_t frames);
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protected:
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private:
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float c1, c2, d0;
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float z1, z2;
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};
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#endif
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