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284 lines
10 KiB
C++
284 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) 2007-2022 Juan Linietsky, Ariel Manzur. */
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/* Copyright (c) 2014-2022 Godot Engine contributors (cf. AUTHORS.md). */
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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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const double to_fill_step = Math_TAU / (double)fft_size;
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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(to_fill_step * (double)temporal_fft_pos) + 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::instantiate() {
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Ref<AudioEffectSpectrumAnalyzerInstance> ins;
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ins.instantiate();
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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(FFTSize 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::FFTSize 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::FLOAT, "buffer_length", PROPERTY_HINT_RANGE, "0.1,4,0.1,suffix:s"), "set_buffer_length", "get_buffer_length");
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ADD_PROPERTY(PropertyInfo(Variant::FLOAT, "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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