godot/servers/audio/audio_filter_sw.h
Rémi Verschelde 0be6d925dc Style: clang-format: Disable KeepEmptyLinesAtTheStartOfBlocks
Which means that reduz' beloved style which we all became used to
will now be changed automatically to remove the first empty line.

This makes us lean closer to 1TBS (the one true brace style) instead
of hybridating it with some Allman-inspired spacing.

There's still the case of braces around single-statement blocks that
needs to be addressed (but clang-format can't help with that, but
clang-tidy may if we agree about it).

Part of #33027.
2020-05-14 16:54:55 +02:00

125 lines
4.3 KiB
C++

/*************************************************************************/
/* audio_filter_sw.h */
/*************************************************************************/
/* This file is part of: */
/* GODOT ENGINE */
/* https://godotengine.org */
/*************************************************************************/
/* Copyright (c) 2007-2020 Juan Linietsky, Ariel Manzur. */
/* Copyright (c) 2014-2020 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 "core/math/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