-------------------------------
/* Audio Library for Teensy, Ladder Filter
* Copyright (c) 2021, Richard van Hoesel
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice, development funding notice, and this permission
* notice shall be included in all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
//-----------------------------------------------------------
// Huovilainen New Moog (HNM) model as per CMJ jun 2006
// Implemented as Teensy Audio Library compatible object
// Richard van Hoesel, Feb. 9 2021
// v.1.02 now includes both cutoff and resonance "CV" modulation inputs
// please retain this header if you use this code.
//-----------------------------------------------------------
// [URL]https://forum.pjrc.com/threads/60488?p=269755&viewfull=1#post269755[/URL]
// [URL]https://forum.pjrc.com/threads/60488?p=269609&viewfull=1#post269609[/URL]
#include <Arduino.h>
#include "filter_ladder.h"
#include <math.h>
#include <stdint.h>
#define MOOG_PI ((float)3.14159265358979323846264338327950288)
//#define MAX_RESONANCE ((float)1.07)
#define MAX_RESONANCE ((float)1.2)
#define MAX_FREQUENCY ((float)(AUDIO_SAMPLE_RATE_EXACT * 0.249f))
float AudioFilterLadder::LPF(float s, int i)
{
float ft = s * (1.0f/1.3f) + (0.3f/1.3f) * z0[i] - z1[i];
ft = ft * alpha + z1[i];
z1[i] = ft;
z0[i] = s;
return ft;
}
void AudioFilterLadder::resonance(float res)
{
// maps resonance = 0->1 to K = 0 -> 4
if (res > MAX_RESONANCE) {
res = MAX_RESONANCE;
} else if (res < 0.0f) {
res = 0.0f;
}
K = 4.0f * res;
}
void AudioFilterLadder::frequency(float c)
{
Fbase = c;
compute_coeffs(c);
}
void AudioFilterLadder:: octaveControl(float octaves)
{
if (octaves > 7.0f) {
octaves = 7.0f;
} else if (octaves < 0.0f) {
octaves = 0.0f;
}
octaveScale = octaves / 32768.0f;
}
void AudioFilterLadder::compute_coeffs(float c)
{
if (c > MAX_FREQUENCY) {
c = MAX_FREQUENCY;
} else if (c < 1.0f) {
c = 1.0f;
}
float wc = c * (float)(2.0f * MOOG_PI / AUDIO_SAMPLE_RATE_EXACT);
float wc2 = wc * wc;
alpha = 0.9892f * wc - 0.4324f * wc2 + 0.1381f * wc * wc2 - 0.0202f * wc2 * wc2;
}
bool AudioFilterLadder::resonating()
{
for (int i=0; i < 4; i++) {
if (fabsf(z0[i]) > 0.0001f) return true;
if (fabsf(z1[i]) > 0.0001f) return true;
}
return false;
}
static inline float fast_exp2f(float x)
{
float i;
float f = modff(x, &i);
f *= 0.693147f / 256.0f;
f += 1.0f;
f *= f;
f *= f;
f *= f;
f *= f;
f *= f;
f *= f;
f *= f;
f *= f;
f = ldexpf(f, i);
return f;
}
static inline float fast_tanh(float x)
{
float x2 = x * x;
return x * (27.0f + x2) / (27.0f + 9.0f * x2);
}
void AudioFilterLadder::update(void)
{
audio_block_t *blocka, *blockb, *blockc;
float Ktot;
bool FCmodActive = true;
bool QmodActive = true;
blocka = receiveWritable(0);
blockb = receiveReadOnly(1);
blockc = receiveReadOnly(2);
if (!blocka) {
if (resonating()) {
// When no data arrives but the filter is still
// resonating, we must continue computing the filter
// with zero input to sustain the resonance
blocka = allocate();
}
if (!blocka) {
if (blockb) release(blockb);
if (blockc) release(blockc);
return;
}
for (int i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
blocka->data[i] = 0;
}
}
if (!blockb) {
FCmodActive = false;
}
if (!blockc) {
QmodActive = false;
Ktot = K;
}
for (int i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
float input = blocka->data[i] * (1.0f/32768.0f);
if (FCmodActive) {
float FCmod = blockb->data[i] * octaveScale;
float ftot = Fbase * fast_exp2f(FCmod);
if (ftot > MAX_FREQUENCY) ftot = MAX_FREQUENCY;
if (FCmod != 0) compute_coeffs(ftot);
}
if (QmodActive) {
float Qmod = blockc->data[i] * (1.0f/32768.0f);
Ktot = K + (MAX_RESONANCE * 4.0f) * Qmod;
if (Ktot > MAX_RESONANCE * 4.0f) Ktot = MAX_RESONANCE * 4.0f;
if (Ktot < 0.0f) Ktot = 0.0f;
}
float u = input - (z1[3] - 0.5f * input) * Ktot;
u = fast_tanh(u);
float stage1 = LPF(u, 0);
float stage2 = LPF(stage1, 1);
float stage3 = LPF(stage2, 2);
float stage4 = LPF(stage3, 3);
blocka->data[i] = stage4 * 32767.0f;
}
transmit(blocka);
release(blocka);
if (blockb) release(blockb);
if (blockc) release(blockc);
}