/**************************************************************************/ /* 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