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- /* 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.
- //-----------------------------------------------------------
-
- // https://forum.pjrc.com/threads/60488?p=269755&viewfull=1#post269755
- // https://forum.pjrc.com/threads/60488?p=269609&viewfull=1#post269609
-
- #include <Arduino.h>
- #include "filter_ladder.h"
- #include <math.h>
- #include <stdint.h>
- #define MOOG_PI ((float)3.14159265358979323846264338327950288)
-
- #define MAX_RESONANCE ((float)1.1)
- #define MAX_FREQUENCY ((float)(AUDIO_SAMPLE_RATE_EXACT * 0.49f))
-
- 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 < 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);
- }
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