1426cd3b3a
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". Backported from #70885.
281 lines
10 KiB
C++
281 lines
10 KiB
C++
/**************************************************************************/
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/* audio_effect_spectrum_analyzer.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_effect_spectrum_analyzer.h"
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#include "servers/audio_server.h"
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static void smbFft(float *fftBuffer, long fftFrameSize, long sign)
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/*
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FFT routine, (C)1996 S.M.Bernsee. Sign = -1 is FFT, 1 is iFFT (inverse)
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Fills fftBuffer[0...2*fftFrameSize-1] with the Fourier transform of the
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time domain data in fftBuffer[0...2*fftFrameSize-1]. The FFT array takes
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and returns the cosine and sine parts in an interleaved manner, ie.
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fftBuffer[0] = cosPart[0], fftBuffer[1] = sinPart[0], asf. fftFrameSize
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must be a power of 2. It expects a complex input signal (see footnote 2),
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ie. when working with 'common' audio signals our input signal has to be
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passed as {in[0],0.,in[1],0.,in[2],0.,...} asf. In that case, the transform
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of the frequencies of interest is in fftBuffer[0...fftFrameSize].
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*/
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{
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float wr, wi, arg, *p1, *p2, temp;
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float tr, ti, ur, ui, *p1r, *p1i, *p2r, *p2i;
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long i, bitm, j, le, le2, k;
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for (i = 2; i < 2 * fftFrameSize - 2; i += 2) {
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for (bitm = 2, j = 0; bitm < 2 * fftFrameSize; bitm <<= 1) {
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if (i & bitm) {
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j++;
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}
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j <<= 1;
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}
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if (i < j) {
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p1 = fftBuffer + i;
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p2 = fftBuffer + j;
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temp = *p1;
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*(p1++) = *p2;
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*(p2++) = temp;
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temp = *p1;
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*p1 = *p2;
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*p2 = temp;
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}
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}
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for (k = 0, le = 2; k < (long)(log((double)fftFrameSize) / log(2.) + .5); k++) {
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le <<= 1;
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le2 = le >> 1;
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ur = 1.0;
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ui = 0.0;
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arg = Math_PI / (le2 >> 1);
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wr = cos(arg);
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wi = sign * sin(arg);
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for (j = 0; j < le2; j += 2) {
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p1r = fftBuffer + j;
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p1i = p1r + 1;
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p2r = p1r + le2;
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p2i = p2r + 1;
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for (i = j; i < 2 * fftFrameSize; i += le) {
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tr = *p2r * ur - *p2i * ui;
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ti = *p2r * ui + *p2i * ur;
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*p2r = *p1r - tr;
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*p2i = *p1i - ti;
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*p1r += tr;
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*p1i += ti;
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p1r += le;
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p1i += le;
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p2r += le;
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p2i += le;
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}
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tr = ur * wr - ui * wi;
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ui = ur * wi + ui * wr;
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ur = tr;
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}
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}
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}
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void AudioEffectSpectrumAnalyzerInstance::process(const AudioFrame *p_src_frames, AudioFrame *p_dst_frames, int p_frame_count) {
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uint64_t time = OS::get_singleton()->get_ticks_usec();
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//copy everything over first, since this only really does capture
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for (int i = 0; i < p_frame_count; i++) {
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p_dst_frames[i] = p_src_frames[i];
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}
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//capture spectrum
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while (p_frame_count) {
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int to_fill = fft_size * 2 - temporal_fft_pos;
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to_fill = MIN(to_fill, p_frame_count);
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float *fftw = temporal_fft.ptrw();
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for (int i = 0; i < to_fill; i++) { //left and right buffers
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float window = -0.5 * Math::cos(2.0 * Math_PI * (double)temporal_fft_pos / (double)fft_size) + 0.5;
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fftw[temporal_fft_pos * 2] = window * p_src_frames->l;
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fftw[temporal_fft_pos * 2 + 1] = 0;
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fftw[(temporal_fft_pos + fft_size * 2) * 2] = window * p_src_frames->r;
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fftw[(temporal_fft_pos + fft_size * 2) * 2 + 1] = 0;
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++p_src_frames;
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++temporal_fft_pos;
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}
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p_frame_count -= to_fill;
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if (temporal_fft_pos == fft_size * 2) {
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//time to do a FFT
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smbFft(fftw, fft_size * 2, -1);
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smbFft(fftw + fft_size * 4, fft_size * 2, -1);
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int next = (fft_pos + 1) % fft_count;
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AudioFrame *hw = (AudioFrame *)fft_history[next].ptr(); //do not use write, avoid cow
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for (int i = 0; i < fft_size; i++) {
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//abs(vec)/fft_size normalizes each frequency
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hw[i].l = Vector2(fftw[i * 2], fftw[i * 2 + 1]).length() / float(fft_size);
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hw[i].r = Vector2(fftw[fft_size * 4 + i * 2], fftw[fft_size * 4 + i * 2 + 1]).length() / float(fft_size);
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}
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fft_pos = next; //swap
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temporal_fft_pos = 0;
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}
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}
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//determine time of capture
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double remainer_sec = (temporal_fft_pos / mix_rate); //subtract remainder from mix time
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last_fft_time = time - uint64_t(remainer_sec * 1000000.0);
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}
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void AudioEffectSpectrumAnalyzerInstance::_bind_methods() {
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ClassDB::bind_method(D_METHOD("get_magnitude_for_frequency_range", "from_hz", "to_hz", "mode"), &AudioEffectSpectrumAnalyzerInstance::get_magnitude_for_frequency_range, DEFVAL(MAGNITUDE_MAX));
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BIND_ENUM_CONSTANT(MAGNITUDE_AVERAGE);
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BIND_ENUM_CONSTANT(MAGNITUDE_MAX);
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}
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Vector2 AudioEffectSpectrumAnalyzerInstance::get_magnitude_for_frequency_range(float p_begin, float p_end, MagnitudeMode p_mode) const {
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if (last_fft_time == 0) {
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return Vector2();
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}
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uint64_t time = OS::get_singleton()->get_ticks_usec();
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float diff = double(time - last_fft_time) / 1000000.0 + base->get_tap_back_pos();
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diff -= AudioServer::get_singleton()->get_output_latency();
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float fft_time_size = float(fft_size) / mix_rate;
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int fft_index = fft_pos;
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while (diff > fft_time_size) {
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diff -= fft_time_size;
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fft_index -= 1;
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if (fft_index < 0) {
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fft_index = fft_count - 1;
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}
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}
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int begin_pos = p_begin * fft_size / (mix_rate * 0.5);
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int end_pos = p_end * fft_size / (mix_rate * 0.5);
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begin_pos = CLAMP(begin_pos, 0, fft_size - 1);
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end_pos = CLAMP(end_pos, 0, fft_size - 1);
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if (begin_pos > end_pos) {
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SWAP(begin_pos, end_pos);
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}
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const AudioFrame *r = fft_history[fft_index].ptr();
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if (p_mode == MAGNITUDE_AVERAGE) {
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Vector2 avg;
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for (int i = begin_pos; i <= end_pos; i++) {
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avg += Vector2(r[i]);
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}
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avg /= float(end_pos - begin_pos + 1);
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return avg;
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} else {
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Vector2 max;
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for (int i = begin_pos; i <= end_pos; i++) {
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max.x = MAX(max.x, r[i].l);
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max.y = MAX(max.y, r[i].r);
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}
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return max;
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}
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}
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Ref<AudioEffectInstance> AudioEffectSpectrumAnalyzer::instance() {
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Ref<AudioEffectSpectrumAnalyzerInstance> ins;
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ins.instance();
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ins->base = Ref<AudioEffectSpectrumAnalyzer>(this);
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static const int fft_sizes[FFT_SIZE_MAX] = { 256, 512, 1024, 2048, 4096 };
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ins->fft_size = fft_sizes[fft_size];
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ins->mix_rate = AudioServer::get_singleton()->get_mix_rate();
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ins->fft_count = (buffer_length / (float(ins->fft_size) / ins->mix_rate)) + 1;
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ins->fft_pos = 0;
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ins->last_fft_time = 0;
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ins->fft_history.resize(ins->fft_count);
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ins->temporal_fft.resize(ins->fft_size * 8); //x2 stereo, x2 amount of samples for freqs, x2 for input
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ins->temporal_fft_pos = 0;
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for (int i = 0; i < ins->fft_count; i++) {
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ins->fft_history.write[i].resize(ins->fft_size); //only magnitude matters
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for (int j = 0; j < ins->fft_size; j++) {
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ins->fft_history.write[i].write[j] = AudioFrame(0, 0);
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}
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}
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return ins;
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}
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void AudioEffectSpectrumAnalyzer::set_buffer_length(float p_seconds) {
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buffer_length = p_seconds;
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}
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float AudioEffectSpectrumAnalyzer::get_buffer_length() const {
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return buffer_length;
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}
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void AudioEffectSpectrumAnalyzer::set_tap_back_pos(float p_seconds) {
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tapback_pos = p_seconds;
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}
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float AudioEffectSpectrumAnalyzer::get_tap_back_pos() const {
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return tapback_pos;
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}
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void AudioEffectSpectrumAnalyzer::set_fft_size(FFT_Size p_fft_size) {
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ERR_FAIL_INDEX(p_fft_size, FFT_SIZE_MAX);
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fft_size = p_fft_size;
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}
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AudioEffectSpectrumAnalyzer::FFT_Size AudioEffectSpectrumAnalyzer::get_fft_size() const {
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return fft_size;
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}
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void AudioEffectSpectrumAnalyzer::_bind_methods() {
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ClassDB::bind_method(D_METHOD("set_buffer_length", "seconds"), &AudioEffectSpectrumAnalyzer::set_buffer_length);
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ClassDB::bind_method(D_METHOD("get_buffer_length"), &AudioEffectSpectrumAnalyzer::get_buffer_length);
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ClassDB::bind_method(D_METHOD("set_tap_back_pos", "seconds"), &AudioEffectSpectrumAnalyzer::set_tap_back_pos);
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ClassDB::bind_method(D_METHOD("get_tap_back_pos"), &AudioEffectSpectrumAnalyzer::get_tap_back_pos);
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ClassDB::bind_method(D_METHOD("set_fft_size", "size"), &AudioEffectSpectrumAnalyzer::set_fft_size);
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ClassDB::bind_method(D_METHOD("get_fft_size"), &AudioEffectSpectrumAnalyzer::get_fft_size);
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ADD_PROPERTY(PropertyInfo(Variant::REAL, "buffer_length", PROPERTY_HINT_RANGE, "0.1,4,0.1"), "set_buffer_length", "get_buffer_length");
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ADD_PROPERTY(PropertyInfo(Variant::REAL, "tap_back_pos", PROPERTY_HINT_RANGE, "0.1,4,0.1"), "set_tap_back_pos", "get_tap_back_pos");
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ADD_PROPERTY(PropertyInfo(Variant::INT, "fft_size", PROPERTY_HINT_ENUM, "256,512,1024,2048,4096"), "set_fft_size", "get_fft_size");
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BIND_ENUM_CONSTANT(FFT_SIZE_256);
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BIND_ENUM_CONSTANT(FFT_SIZE_512);
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BIND_ENUM_CONSTANT(FFT_SIZE_1024);
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BIND_ENUM_CONSTANT(FFT_SIZE_2048);
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BIND_ENUM_CONSTANT(FFT_SIZE_4096);
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BIND_ENUM_CONSTANT(FFT_SIZE_MAX);
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}
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AudioEffectSpectrumAnalyzer::AudioEffectSpectrumAnalyzer() {
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buffer_length = 2;
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tapback_pos = 0.01;
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fft_size = FFT_SIZE_1024;
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}
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