godot/servers/audio/audio_filter_sw.h
Rémi Verschelde d95794ec8a
One Copyright Update to rule them all
As many open source projects have started doing it, we're removing the
current year from the copyright notice, so that we don't need to bump
it every year.

It seems like only the first year of publication is technically
relevant for copyright notices, and even that seems to be something
that many companies stopped listing altogether (in a version controlled
codebase, the commits are a much better source of date of publication
than a hardcoded copyright statement).

We also now list Godot Engine contributors first as we're collectively
the current maintainers of the project, and we clarify that the
"exclusive" copyright of the co-founders covers the timespan before
opensourcing (their further contributions are included as part of Godot
Engine contributors).

Also fixed "cf." Frenchism - it's meant as "refer to / see".
2023-01-05 13:25:55 +01:00

127 lines
4.3 KiB
C++

/**************************************************************************/
/* audio_filter_sw.h */
/**************************************************************************/
/* This file is part of: */
/* GODOT ENGINE */
/* https://godotengine.org */
/**************************************************************************/
/* Copyright (c) 2014-present Godot Engine contributors (see AUTHORS.md). */
/* Copyright (c) 2007-2014 Juan Linietsky, Ariel Manzur. */
/* */
/* 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 "core/math/math_funcs.h"
class AudioFilterSW {
public:
struct Coeffs {
float a1 = 0.0f;
float a2 = 0.0f;
float b0 = 0.0f;
float b1 = 0.0f;
float b2 = 0.0f;
};
enum Mode {
BANDPASS,
HIGHPASS,
LOWPASS,
NOTCH,
PEAK,
BANDLIMIT,
LOWSHELF,
HIGHSHELF
};
class Processor { // Simple filter processor.
AudioFilterSW *filter = nullptr;
Coeffs coeffs;
// History.
float ha1 = 0.0f;
float ha2 = 0.0f;
float hb1 = 0.0f;
float hb2 = 0.0f;
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 = 5000.0f;
float resonance = 0.5f;
float gain = 1.0f;
float sampling_rate = 44100.0f;
int stages = 1;
Mode mode = LOWPASS;
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