barrverb/plugin/barrverb.cpp

/*
   BarrVerb reverb plugin

   Copyright 2024 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 "barrverb.hpp"
#include "rom.h"

START_NAMESPACE_DISTRHO

BarrVerb::BarrVerb() : Plugin(kParameterCount, 64, 0) {  // two parameters, one program, no states
    lowpass = new float[getBufferSize()];
    ram = new int16_t[16384];

    /*
        // calculate SVF params
        // hardcoded values for now
        float fc = 5019;
        float F = fc / 48000;  // assume 48kHz
        float w = 2 * tan(3.14159 * F);
        float a = w / 0.7845;  // 1dB Chebyshev, 2-pole
        float b = w * w;

        // "corrected" SVF params, per Fons Adriaensen
        c1_1 = (a + b) / (1 + a / 2 + b / 4);
        c2_1 = b / (a + b);
        d0_1 = c1_1 * c2_1 / 4;

        fc = 9433;
        F = fc / 48000;  // assume 48kHz
        w = 2 * tan(3.14159 * F);
        a = w / 3.5594;  // 1dB Chebyshev, 2-pole
        b = w * w;

        c1_2 = (a + b) / (1 + a / 2 + b / 4);
        c2_2 = b / (a + b);
        d0_2 = c1_2 * c2_2 / 4;*/
    // calculate SVF params
    // hardcoded values for now

    float fc = 10000;
    float F = fc / 48000;  // assume 48kHz
    float w = 2 * tan(3.14159 * F);
    float a = w / 0.5412;  // Butterworth 4-pole first stage
    float b = w * w;

    // "corrected" SVF params, per Fons Adriaensen
    c1_1 = (a + b) / (1 + a / 2 + b / 4);
    c2_1 = b / (a + b);
    d0_1 = c1_1 * c2_1 / 4;

    fc = 10000;
    F = fc / 48000;  // assume 48kHz
    w = 2 * tan(3.14159 * F);
    a = w / 1.3065;  // Butterworth 4-pole second stage
    b = w * w;

    c1_2 = (a + b) / (1 + a / 2 + b / 4);
    c2_2 = b / (a + b);
    d0_2 = c1_2 * c2_2 / 4;
}

// Initialisation functions

void BarrVerb::initAudioPort(bool input, uint32_t index, AudioPort &port) {
    port.groupId = kPortGroupStereo;
    Plugin::initAudioPort(input, index, port);
}

void BarrVerb::initProgramName(uint32_t index, String &programName) {

    programName = "init program"; //&prog_name[index & 0x3f];
    programName = prog_name[index & 0x3f].c_str();

}

void BarrVerb::loadProgram(uint32_t index) {
    prog_offset = (index & 0x3f) << 7;
}

// Processing functions

void BarrVerb::activate() {
    // calculate filter coefficients
    printf("called activate()\n");
}

void BarrVerb::deactivate() {
    // zero out the outputs, maybe
    printf("called deactivate()\n");
}

void BarrVerb::run(const float **inputs, float **outputs, uint32_t frames) {
    // actual effects here

    float x;
    uint16_t opcode;

    for (uint32_t i = 0; i < frames; i++) {
        // smash to mono
        lowpass[i] = (inputs[0][i] + inputs[1][i]) / 2;

        // 10kHz lowpass filter, 2x oversampling
        x = lowpass[i] - in_z1 - in_z2;
        in_z2 += c2_1 * in_z1;
        in_z1 += c1_1 * x;

        x = (d0_1 * x + in_z2) - in_z12 - in_z22;
        in_z22 += c2_2 * in_z12;
        in_z12 += c1_2 * x;
        lowpass[i] = d0_2 * x + in_z22;
    }

    // now run the DSP
    for (uint32_t i=0; i < frames; i+=2) {
        // run the actual DSP engine for each sample
        for (uint8_t step = 0; step < 128; step++) {
            opcode = rom[prog_offset + step];
            switch (opcode & 0xc000) {
                case 0x0000:
                    ai = ram[ptr];
                    li = acc + (ai >> 1);
                    break;
                case 0x4000:
                    ai = ram[ptr];
                    li = (ai >> 1);
                    break;
                case 0x8000:
                    ai = acc;
                    ram[ptr] = ai;
                    li = acc + (ai >> 1);
                    break;
                case 0xc000:
                    ai = acc;
                    ram[ptr] = -ai;
                    li = -(ai >> 1);
                    break;
            }
            if (step == 0x00) {
                // load RAM from ADC
                ram[ptr] = (int)(lowpass[i] * 4096);
            } else if (step == 0x60) {
                // output right channel
                outputs[1][i] = (float)ai / 4096;
                outputs[1][i+1] = (float)ai / 4096;

            } else if (step == 0x70) {
                // output left channel
                outputs[0][i] = (float)ai / 4096;
                outputs[0][i+1] = (float)ai / 4096;
            } else {
                // everything else
                // ADC and DAC operations don't affect the accumulator
                // every other step ends with the accumulator latched from the Latch Input reg
                acc = li;
            }

            // 16kW of RAM
            ptr += opcode & 0x3fff;
            ptr &= 0x3fff;
        }
    }
}

// create the plugin
Plugin *createPlugin() { return new BarrVerb(); }

END_NAMESPACE_DISTRHO