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https://github.com/godotengine/godot
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d95794ec8a
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".
261 lines
8.1 KiB
C++
261 lines
8.1 KiB
C++
/**************************************************************************/
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/* audio_filter_sw.cpp */
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/**************************************************************************/
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/* This file is part of: */
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/* GODOT ENGINE */
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/* https://godotengine.org */
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/**************************************************************************/
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/* Copyright (c) 2014-present Godot Engine contributors (see AUTHORS.md). */
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/* Copyright (c) 2007-2014 Juan Linietsky, Ariel Manzur. */
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/* */
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/* Permission is hereby granted, free of charge, to any person obtaining */
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/* a copy of this software and associated documentation files (the */
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/* "Software"), to deal in the Software without restriction, including */
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/* without limitation the rights to use, copy, modify, merge, publish, */
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/* distribute, sublicense, and/or sell copies of the Software, and to */
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/* permit persons to whom the Software is furnished to do so, subject to */
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/* the following conditions: */
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/* */
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/* The above copyright notice and this permission notice shall be */
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/* included in all copies or substantial portions of the Software. */
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/* */
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/* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, */
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/* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF */
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/* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. */
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/* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY */
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/* CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, */
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/* TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE */
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/* SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. */
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/**************************************************************************/
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#include "audio_filter_sw.h"
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void AudioFilterSW::set_mode(Mode p_mode) {
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mode = p_mode;
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}
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void AudioFilterSW::set_cutoff(float p_cutoff) {
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cutoff = p_cutoff;
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}
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void AudioFilterSW::set_resonance(float p_resonance) {
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resonance = p_resonance;
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}
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void AudioFilterSW::set_gain(float p_gain) {
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gain = p_gain;
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}
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void AudioFilterSW::set_sampling_rate(float p_srate) {
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sampling_rate = p_srate;
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}
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void AudioFilterSW::prepare_coefficients(Coeffs *p_coeffs) {
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int sr_limit = (sampling_rate / 2) + 512;
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double final_cutoff = (cutoff > sr_limit) ? sr_limit : cutoff;
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if (final_cutoff < 1) {
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final_cutoff = 1; //don't allow less than this
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}
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double omega = Math_TAU * final_cutoff / sampling_rate;
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double sin_v = Math::sin(omega);
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double cos_v = Math::cos(omega);
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double Q = resonance;
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if (Q <= 0.0) {
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Q = 0.0001;
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}
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if (mode == BANDPASS) {
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Q *= 2.0;
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} else if (mode == PEAK) {
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Q *= 3.0;
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}
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double tmpgain = gain;
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if (tmpgain < 0.001) {
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tmpgain = 0.001;
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}
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if (stages > 1) {
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Q = (Q > 1.0 ? Math::pow(Q, 1.0 / stages) : Q);
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tmpgain = Math::pow(tmpgain, 1.0 / (stages + 1));
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}
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double alpha = sin_v / (2 * Q);
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double a0 = 1.0 + alpha;
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switch (mode) {
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case LOWPASS: {
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p_coeffs->b0 = (1.0 - cos_v) / 2.0;
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p_coeffs->b1 = 1.0 - cos_v;
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p_coeffs->b2 = (1.0 - cos_v) / 2.0;
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p_coeffs->a1 = -2.0 * cos_v;
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p_coeffs->a2 = 1.0 - alpha;
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} break;
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case HIGHPASS: {
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p_coeffs->b0 = (1.0 + cos_v) / 2.0;
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p_coeffs->b1 = -(1.0 + cos_v);
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p_coeffs->b2 = (1.0 + cos_v) / 2.0;
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p_coeffs->a1 = -2.0 * cos_v;
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p_coeffs->a2 = 1.0 - alpha;
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} break;
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case BANDPASS: {
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p_coeffs->b0 = alpha * sqrt(Q + 1);
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p_coeffs->b1 = 0.0;
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p_coeffs->b2 = -alpha * sqrt(Q + 1);
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p_coeffs->a1 = -2.0 * cos_v;
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p_coeffs->a2 = 1.0 - alpha;
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} break;
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case NOTCH: {
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p_coeffs->b0 = 1.0;
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p_coeffs->b1 = -2.0 * cos_v;
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p_coeffs->b2 = 1.0;
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p_coeffs->a1 = -2.0 * cos_v;
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p_coeffs->a2 = 1.0 - alpha;
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} break;
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case PEAK: {
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p_coeffs->b0 = (1.0 + alpha * tmpgain);
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p_coeffs->b1 = (-2.0 * cos_v);
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p_coeffs->b2 = (1.0 - alpha * tmpgain);
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p_coeffs->a1 = -2 * cos_v;
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p_coeffs->a2 = (1 - alpha / tmpgain);
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} break;
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case BANDLIMIT: {
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//this one is extra tricky
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double hicutoff = resonance;
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double centercutoff = (cutoff + resonance) / 2.0;
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double bandwidth = (Math::log(centercutoff) - Math::log(hicutoff)) / Math::log((double)2);
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omega = Math_TAU * centercutoff / sampling_rate;
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alpha = Math::sin(omega) * Math::sinh(Math::log((double)2) / 2 * bandwidth * omega / Math::sin(omega));
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a0 = 1 + alpha;
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p_coeffs->b0 = alpha;
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p_coeffs->b1 = 0;
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p_coeffs->b2 = -alpha;
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p_coeffs->a1 = -2 * Math::cos(omega);
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p_coeffs->a2 = 1 - alpha;
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} break;
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case LOWSHELF: {
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double tmpq = Math::sqrt(Q);
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if (tmpq <= 0) {
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tmpq = 0.001;
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}
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double beta = Math::sqrt(tmpgain) / tmpq;
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a0 = (tmpgain + 1.0) + (tmpgain - 1.0) * cos_v + beta * sin_v;
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p_coeffs->b0 = tmpgain * ((tmpgain + 1.0) - (tmpgain - 1.0) * cos_v + beta * sin_v);
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p_coeffs->b1 = 2.0 * tmpgain * ((tmpgain - 1.0) - (tmpgain + 1.0) * cos_v);
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p_coeffs->b2 = tmpgain * ((tmpgain + 1.0) - (tmpgain - 1.0) * cos_v - beta * sin_v);
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p_coeffs->a1 = -2.0 * ((tmpgain - 1.0) + (tmpgain + 1.0) * cos_v);
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p_coeffs->a2 = ((tmpgain + 1.0) + (tmpgain - 1.0) * cos_v - beta * sin_v);
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} break;
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case HIGHSHELF: {
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double tmpq = Math::sqrt(Q);
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if (tmpq <= 0) {
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tmpq = 0.001;
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}
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double beta = Math::sqrt(tmpgain) / tmpq;
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a0 = (tmpgain + 1.0) - (tmpgain - 1.0) * cos_v + beta * sin_v;
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p_coeffs->b0 = tmpgain * ((tmpgain + 1.0) + (tmpgain - 1.0) * cos_v + beta * sin_v);
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p_coeffs->b1 = -2.0 * tmpgain * ((tmpgain - 1.0) + (tmpgain + 1.0) * cos_v);
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p_coeffs->b2 = tmpgain * ((tmpgain + 1.0) + (tmpgain - 1.0) * cos_v - beta * sin_v);
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p_coeffs->a1 = 2.0 * ((tmpgain - 1.0) - (tmpgain + 1.0) * cos_v);
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p_coeffs->a2 = ((tmpgain + 1.0) - (tmpgain - 1.0) * cos_v - beta * sin_v);
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} break;
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}
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p_coeffs->b0 /= a0;
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p_coeffs->b1 /= a0;
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p_coeffs->b2 /= a0;
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p_coeffs->a1 /= 0.0 - a0;
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p_coeffs->a2 /= 0.0 - a0;
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}
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void AudioFilterSW::set_stages(int p_stages) {
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stages = p_stages;
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}
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/* Fourier transform kernel to obtain response */
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float AudioFilterSW::get_response(float p_freq, Coeffs *p_coeffs) {
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float freq = p_freq / sampling_rate * Math_TAU;
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float cx = p_coeffs->b0, cy = 0.0;
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cx += cos(freq) * p_coeffs->b1;
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cy -= sin(freq) * p_coeffs->b1;
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cx += cos(2 * freq) * p_coeffs->b2;
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cy -= sin(2 * freq) * p_coeffs->b2;
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float H = cx * cx + cy * cy;
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cx = 1.0;
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cy = 0.0;
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cx -= cos(freq) * p_coeffs->a1;
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cy += sin(freq) * p_coeffs->a1;
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cx -= cos(2 * freq) * p_coeffs->a2;
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cy += sin(2 * freq) * p_coeffs->a2;
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H = H / (cx * cx + cy * cy);
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return H;
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}
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AudioFilterSW::Processor::Processor() {
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set_filter(nullptr);
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}
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void AudioFilterSW::Processor::set_filter(AudioFilterSW *p_filter, bool p_clear_history) {
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if (p_clear_history) {
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ha1 = ha2 = hb1 = hb2 = 0;
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}
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filter = p_filter;
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}
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void AudioFilterSW::Processor::update_coeffs(int p_interp_buffer_len) {
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if (!filter) {
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return;
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}
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if (p_interp_buffer_len) { //interpolate
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Coeffs old_coeffs = coeffs;
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filter->prepare_coefficients(&coeffs);
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incr_coeffs.a1 = (coeffs.a1 - old_coeffs.a1) / p_interp_buffer_len;
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incr_coeffs.a2 = (coeffs.a2 - old_coeffs.a2) / p_interp_buffer_len;
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incr_coeffs.b0 = (coeffs.b0 - old_coeffs.b0) / p_interp_buffer_len;
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incr_coeffs.b1 = (coeffs.b1 - old_coeffs.b1) / p_interp_buffer_len;
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incr_coeffs.b2 = (coeffs.b2 - old_coeffs.b2) / p_interp_buffer_len;
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coeffs = old_coeffs;
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} else {
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filter->prepare_coefficients(&coeffs);
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}
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}
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void AudioFilterSW::Processor::process(float *p_samples, int p_amount, int p_stride, bool p_interpolate) {
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if (!filter) {
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return;
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}
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if (p_interpolate) {
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for (int i = 0; i < p_amount; i++) {
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process_one_interp(*p_samples);
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p_samples += p_stride;
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}
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} else {
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for (int i = 0; i < p_amount; i++) {
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process_one(*p_samples);
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p_samples += p_stride;
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}
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}
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}
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