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//=============================================================================
// Zerberus
// Filters implementation
//
// Copyright (C) 2018 Werner Schweer
//
// This program is free software; you can redistribute it and/or modify
// it under the terms of the GNU General Public License version 2
// as published by the Free Software Foundation and appearing in
// the file LICENCE.GPL
//=============================================================================
#include "filter.h"
#include "zerberus.h"
#include "zone.h"
#include <math.h>
#include <functional>
static constexpr int INTERP_MAX = 256;
static float interpCoeff[INTERP_MAX][4];
//---------------------------------------------------------
// FilterBQ
//---------------------------------------------------------
ZFilter::ZFilter()
{
constexpr double ff = 1.0 / 32768.0;
for (int i = 0; i < INTERP_MAX; i++) {
double x = (double) i / (double) INTERP_MAX;
interpCoeff[i][0] = (x * (-0.5 + x * (1 - 0.5 * x))) * ff;
interpCoeff[i][1] = (1.0 + x * x * (1.5 * x - 2.5)) * ff;
interpCoeff[i][2] = (x * (0.5 + x * (2.0 - 1.5 * x))) * ff;
interpCoeff[i][3] = (0.5 * x * x * (x - 1.0)) * ff;
}
}
//---------------------------------------------------------
// initialize
//---------------------------------------------------------
void ZFilter::initialize(const Zerberus* _zerberus, const Zone* z, int velocity)
{
this->zerberus = _zerberus;
this->sampleZone = z;
resonanceF = _zerberus->ct2hz(13500.0);
if (z->isCutoffDefined) {
//calculate current cutoff value
float cutoffHz = z->cutoff;
//Formula for converting the interval frequency ratio f2 / f1 to cents (c or ¢).
//¢ or c = 1200 × log2 (f2 / f1)
cutoffHz *= pow(2.0, velocity / 127.0f * z->fil_veltrack / 1200.0);
resonanceF = cutoffHz;
}
last_resonanceF = -1.0;
float q_db = 0; //no resonance by default
q_lin = pow(10.0f, q_db / 20.0f);
gain = 1.0 / sqrt(q_lin);
firstRun = true;
}
//---------------------------------------------------------
// update
//---------------------------------------------------------
void ZFilter::update()
{
float adaptedFrequency = resonanceF;
const float sr = zerberus->sampleRate();
if (adaptedFrequency > 0.45f * sr)
adaptedFrequency = 0.45f * sr;
else if (adaptedFrequency < 5.f)
adaptedFrequency = 5.f;
const bool freqWasUpdated = fabs(adaptedFrequency - last_resonanceF) > 0.01f;
if (!freqWasUpdated)
return;
last_resonanceF = adaptedFrequency;
// The filter coefficients have to be recalculated (filter
// parameters have changed). Recalculation for various reasons is
// forced by setting last_fres to -1. The flag filter_startup
// indicates, that the DSP loop runs for the first time, in this
// case, the filter is set directly, instead of smoothly fading
// between old and new settings.
//
// Those equations from Robert Bristow-Johnson's `Cookbook
// formulae for audio EQ biquad filter coefficients', obtained
// from Harmony-central.com / Computer / Programming. They are
// the result of the bilinear transform on an analogue filter
// prototype. To quote, `BLT frequency warping has been taken
// into account for both significant frequency relocation and for
// bandwidth readjustment'.
const float omega = (float) 2.0f * M_PI * (adaptedFrequency / sr);
const float sin_coeff = sin(omega);
const float cos_coeff = cos(omega);
const float alpha_coeff = sin_coeff / (2.0f * q_lin);
const float a0_inv = 1.0f / (1.0f + alpha_coeff);
const float linCosCoeff = 2.f - cos_coeff;
float a1_temp = 0.f;
float a2_temp = 0.f;
float b1_temp = 0.f;
float b0_temp = 0.f;
float b2_temp = 0.f;
switch (sampleZone->fil_type) {
case FilterType::lpf_2p: {
a1_temp = 2.0f * cos_coeff * a0_inv;
a2_temp = (alpha_coeff - 1.f) * a0_inv;
b1_temp = (1.0f - cos_coeff) * a0_inv;
b0_temp = b2_temp = b1_temp * 0.5f;
break;
}
case FilterType::hpf_2p: {
a1_temp = 2.0f * cos_coeff * a0_inv;
a2_temp = (alpha_coeff - 1.f) * a0_inv;
b1_temp = -(1.0f + cos_coeff) * a0_inv;
b0_temp = b2_temp = -b1_temp * 0.5f;
break;
}
case FilterType::bpf_2p: {
a1_temp = 2.0f * cos_coeff * a0_inv;
a2_temp = (alpha_coeff - 1.f) * a0_inv;
b0_temp = alpha_coeff * a0_inv;
b1_temp = 0.f;
b2_temp = -b0_temp;
break;
}
case FilterType::brf_2p: {
a1_temp = 2.0f * cos_coeff * a0_inv;
a2_temp = (alpha_coeff - 1.f) * a0_inv;
b1_temp = a0_inv * (-2.f * cos_coeff);
b0_temp = b2_temp = a0_inv;
break;
}
case FilterType::lpf_1p: {
a1_temp = -(linCosCoeff - sqrt(linCosCoeff * linCosCoeff - 1));
b0_temp = 1 + a1_temp;
break;
}
case FilterType::hpf_1p: {
a1_temp = -(linCosCoeff - sqrt(linCosCoeff * linCosCoeff - 1));
b0_temp = -a1_temp;
b1_temp = a1_temp;
break;
}
default:
qWarning() << "fil_type is not implemented: " << (int)sampleZone->fil_type;
}
if (firstRun) {
/* The filter is calculated, because the voice was started up.
* In this case set the filter coefficients without delay.
*/
a1 = a1_temp;
a2 = a2_temp;
b0 = b0_temp;
b2 = b2_temp;
b1 = b1_temp;
filter_coeff_incr_count = 0;
firstRun = false;
}
else {
/* The filter frequency is changed. Calculate an increment
* factor, so that the new setting is reached after some time.
*/
static const int FILTER_TRANSITION_SAMPLES = 64;
a1_incr = (a1_temp - a1) / FILTER_TRANSITION_SAMPLES;
a2_incr = (a2_temp - a2) / FILTER_TRANSITION_SAMPLES;
b0_incr = (b0_temp - b0) / FILTER_TRANSITION_SAMPLES;
b1_incr = (b1_temp - b1) / FILTER_TRANSITION_SAMPLES;
b2_incr = (b2_temp - b2) / FILTER_TRANSITION_SAMPLES;
/* Have to add the increments filter_coeff_incr_count times. */
filter_coeff_incr_count = FILTER_TRANSITION_SAMPLES;
}
}
//---------------------------------------------------------
// apply
//---------------------------------------------------------
float ZFilter::apply(float inputValue, bool leftChannel)
{
float value = 0.f;
const auto applyBqEquation = [this, &inputValue](float& histX1, float& histX2, float& histY1, float& histY2) {
//apply filter
/*
float y = d.b0 * x + d.b1 * d.x1 + d.b2 * d.x2 +
d.a1 * d.y1 + d.a2 * d.y2;
d.x2 = d.x1;
d.x1 = x;
d.y2 = d.y1;
d.y1 = y;
return y;
*/
float value = b0 * inputValue + b1 * histX1 + b2 * histX2 + a1 * histY1 + a2 * histY2;
histX2 = histX1;
histX1 = inputValue;
histY2 = histY1;
histY1 = value;
return value;
};
const auto applyHPF1PEquation = [this, &inputValue](float& histX1, float& histY1) {
float value = b0 * inputValue + b1 * histX1 - a1 * histY1;
histX1 = inputValue;
histY1 = value;
return value;
};
const auto applyLPF1PEquation = [this, &inputValue](float& histY1) {
float value = b0 * inputValue - a1 * histY1;
histY1 = value;
return value;
};
if (leftChannel) {
switch (sampleZone->fil_type) {
case FilterType::hpf_2p:
case FilterType::lpf_2p:
case FilterType::bpf_2p:
case FilterType::brf_2p: {
value = applyBqEquation(monoL.histX1, monoL.histX2, monoL.histY1, monoL.histY2);
break;
}
case FilterType::hpf_1p: {
value = applyHPF1PEquation(monoL.histX1, monoL.histY1);
break;
}
case FilterType::lpf_1p: {
value = applyLPF1PEquation(monoL.histY1);
break;
}
default:
qWarning() << "this equation is not implemented" << (int)sampleZone->fil_type;
}
}
else { //right channel in stereo samples
switch (sampleZone->fil_type) {
case FilterType::hpf_2p:
case FilterType::lpf_2p:
case FilterType::bpf_2p:
case FilterType::brf_2p: {
value = applyBqEquation(monoR.histX1, monoR.histX2, monoR.histY1, monoR.histY2);
break;
}
case FilterType::hpf_1p: {
value = applyHPF1PEquation(monoR.histX1, monoR.histY1);
break;
}
case FilterType::lpf_1p: {
value = applyLPF1PEquation(monoR.histY1);
break;
}
default:
qWarning() << "this equation is not implemented" << (int)sampleZone->fil_type;
}
}
if (filter_coeff_incr_count) {
--filter_coeff_incr_count;
a1 += a1_incr;
a2 += a2_incr;
b0 += b0_incr;
b1 += b1_incr;
b2 += b2_incr;
}
return value;
}
//---------------------------------------------------------
// interpolate
//---------------------------------------------------------
float ZFilter::interpolate(unsigned phase, short prevVal, short currVal, short nextVal, short nextNextVal) const
{
const auto& interpValTable = interpCoeff[phase];
return (interpValTable[0] * prevVal
+ interpValTable[1] * currVal
+ interpValTable[2] * nextVal
+ interpValTable[3] * nextNextVal);
}
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