peacock/plugin/chorus.cpp

/*
   sonnenlicht poly ensemble

   Copyright 2025 Gordon JC Pearce <gordonjcp@gjcp.net>

   Permission to use, copy, modify, and/or distribute this software for any
   purpose with or without fee is hereby granted, provided that the above
   copyright notice and this permission notice appear in all copies.

   THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
   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
   CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
*/

#include "chorus.hpp"

#include <math.h>
#include <string.h>

#include <cstdio>

Chorus::Chorus() {
    lpfOut1 = new float[bufferSize];
    lpfOut2 = new float[bufferSize];
    ram = new float[DELAYSIZE];  // probably needs to be calculated based on sample rate

    fastPhase = 0;

    // not quite Butterworth but you'd never hear the difference
    postFilter1l = new SVF(9688, .549);
    postFilter2l = new SVF(10377, 1.291);
    postFilter1r = new SVF(9688, .549);
    postFilter2r = new SVF(10377, 1.291);

    // lfo values taken from a rough simulation
    fastOmega = 6.283 * 0.7 / sampleRate;  // approximate, can be adjusted

    // zero out the delay buffer
    memset(ram, 0, sizeof(float) * DELAYSIZE);
    memset(lpfOut1, 0, sizeof(float) * bufferSize);
    memset(lpfOut2, 0, sizeof(float) * bufferSize);
}

Chorus::~Chorus() {
    delete lpfOut1;
    delete lpfOut2;
    delete ram;
    delete postFilter1l;
    delete postFilter2l;
    delete postFilter1r;
    delete postFilter2r;
}

void Chorus::run(float* input, float** outputs, uint32_t frames) {
    // actual effects here

    // FIXME add highpass filter

    // now run the DSP
    float s0 = 0, s1 = 0;
    float lfoMod, dly1, frac, flt;
    uint16_t tap, delay;

    for (uint32_t i = 0; i < frames; i++) {
        // run a step of LFO
        fastPhase += fastOmega;
        if (fastPhase > 6.283) fastPhase -= 6.283;

        flt = ((hpdelay - input[i]) * hpcut) + input[i];
        hpdelay = flt;
        input[i] += (flt * hpgain);

        ram[delayptr] = input[i];

#define BASE 0.005
#define AMT 0.00175

        // FIXME make this triangle not sine, and move arithmetic out of mainloop
        lfoMod = sin(fastPhase);
        dly1 = (BASE + (AMT * lfoMod)) * sampleRate;
        delay = (int)dly1;
        frac = dly1 - delay;

        tap = delayptr - delay;
        s1 = ram[(tap - 1) & 0x3ff];
        s0 = ram[tap & 0x3ff];
        lpfOut1[i] = ((s1 - s0) * frac) + s0;

        dly1 = (BASE - (AMT * lfoMod)) * sampleRate;
        delay = (int)dly1;
        frac = dly1 - delay;

        tap = delayptr - delay;
        s1 = ram[(tap - 1) & 0x3ff];
        s0 = ram[tap & 0x3ff];
        lpfOut2[i] = ((s1 - s0) * frac) + s0;

        // lpfOut1[i] = input[i] + 1.2 * out0; //(out0 + (out120 * 0.66) + (out240 * 0.33));
        //  lpfOut2[i] = input[i] + 1.2 * out120; //(out0 + (out120 * 0.33) + (out240 * 0.66));

        delayptr++;
        delayptr &= 0x3ff;
    }

    postFilter1l->runSVF(lpfOut1, lpfOut1, frames);
    postFilter2l->runSVF(lpfOut1, lpfOut1, frames);
    postFilter1r->runSVF(lpfOut2, lpfOut2, frames);
    postFilter2r->runSVF(lpfOut2, lpfOut2, frames);

    for (uint32_t i = 0; i < frames; i++) {
        float y = input[i];
        outputs[0][i] = y + (gain * lpfOut1[i]);
        outputs[1][i] = y + (gain * lpfOut2[i]);
    }
}

void Chorus::setHpf(uint8_t mode) {
    switch (mode) {
        case 0x00:
            hpcut = 0.91245;
            hpgain = -1;
            break;
        case 0x08:
            hpcut = 0.97097;
            hpgain = -1;
            break;
        case 0x10:
            hpcut = 1;
            hpgain = 0;
            break;
        case 0x18:
            hpcut = 0.99088;
            hpgain = 1;
            break;
    }
}

void Chorus::setChorus(uint8_t mode) {
    // switch chorus mode
    switch (mode) {
        case 0x60:
            gain = 0;
            break;
        case 0x40:
            gain = 1.2;
            fastOmega = 6.283 * 0.5 / sampleRate;
            break;  // approximate, can be adjusted
        case 0x00:
            gain = 1.2;
            fastOmega = 6.283 * 0.9 / sampleRate;
            break;  // approximate, can be adjusted
    }
}