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/*************************************************************************/
/* audio_filter_sw.h */
/*************************************************************************/
/* This file is part of: */
/* GODOT ENGINE */
/* https://godotengine.org */
/*************************************************************************/
/* Copyright (c) 2007-2018 Juan Linietsky, Ariel Manzur. */
/* Copyright (c) 2014-2018 Godot Engine contributors (cf. AUTHORS.md) */
/* */
/* 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 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. */
/*************************************************************************/
#ifndef AUDIO_FILTER_SW_H
#define AUDIO_FILTER_SW_H
#include "math_funcs.h"
class AudioFilterSW {
public:
struct Coeffs {
float a1, a2;
float b0, b1, b2;
//bool operator==(const Coeffs &p_rv) { return (FLOATS_EQ(a1,p_rv.a1) && FLOATS_EQ(a2,p_rv.a2) && FLOATS_EQ(b1,p_rv.b1) && FLOATS_EQ(b2,p_rv.b2) && FLOATS_EQ(b0,p_rv.b0) ); }
Coeffs() { a1 = a2 = b0 = b1 = b2 = 0.0; }
};
enum Mode {
BANDPASS,
HIGHPASS,
LOWPASS,
NOTCH,
PEAK,
BANDLIMIT,
LOWSHELF,
HIGHSHELF
};
class Processor { // simple filter processor
AudioFilterSW *filter;
Coeffs coeffs;
float ha1, ha2, hb1, hb2; //history
Coeffs incr_coeffs;
public:
void set_filter(AudioFilterSW *p_filter, bool p_clear_history = true);
void process(float *p_samples, int p_amount, int p_stride = 1, bool p_interpolate = false);
void update_coeffs(int p_interp_buffer_len = 0);
_ALWAYS_INLINE_ void process_one(float &p_sample);
_ALWAYS_INLINE_ void process_one_interp(float &p_sample);
Processor();
};
private:
float cutoff;
float resonance;
float gain;
float sampling_rate;
int stages;
Mode mode;
public:
float get_response(float p_freq, Coeffs *p_coeffs);
void set_mode(Mode p_mode);
void set_cutoff(float p_cutoff);
void set_resonance(float p_resonance);
void set_gain(float p_gain);
void set_sampling_rate(float p_srate);
void set_stages(int p_stages); //adjust for multiple stages
void prepare_coefficients(Coeffs *p_coeffs);
AudioFilterSW();
};
/* inline methods */
void AudioFilterSW::Processor::process_one(float &p_sample) {
float pre = p_sample;
p_sample = (p_sample * coeffs.b0 + hb1 * coeffs.b1 + hb2 * coeffs.b2 + ha1 * coeffs.a1 + ha2 * coeffs.a2);
ha2 = ha1;
hb2 = hb1;
hb1 = pre;
ha1 = p_sample;
}
void AudioFilterSW::Processor::process_one_interp(float &p_sample) {
float pre = p_sample;
p_sample = (p_sample * coeffs.b0 + hb1 * coeffs.b1 + hb2 * coeffs.b2 + ha1 * coeffs.a1 + ha2 * coeffs.a2);
ha2 = ha1;
hb2 = hb1;
hb1 = pre;
ha1 = p_sample;
coeffs.b0 += incr_coeffs.b0;
coeffs.b1 += incr_coeffs.b1;
coeffs.b2 += incr_coeffs.b2;
coeffs.a1 += incr_coeffs.a1;
coeffs.a2 += incr_coeffs.a2;
}
#endif // AUDIO_FILTER_SW_H
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