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pandemonium_engine/servers/audio/effects/audio_effect_spectrum_analyzer.cpp

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/*************************************************************************/
/* audio_effect_spectrum_analyzer.cpp */
/*************************************************************************/
/* This file is part of: */
/* GODOT ENGINE */
/* https://godotengine.org */
/*************************************************************************/
/* Copyright (c) 2007-2022 Juan Linietsky, Ariel Manzur. */
/* Copyright (c) 2014-2022 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. */
/*************************************************************************/
#include "audio_effect_spectrum_analyzer.h"
#include "servers/audio_server.h"
static void smbFft(float *fftBuffer, long fftFrameSize, long sign)
/*
FFT routine, (C)1996 S.M.Bernsee. Sign = -1 is FFT, 1 is iFFT (inverse)
Fills fftBuffer[0...2*fftFrameSize-1] with the Fourier transform of the
time domain data in fftBuffer[0...2*fftFrameSize-1]. The FFT array takes
and returns the cosine and sine parts in an interleaved manner, ie.
fftBuffer[0] = cosPart[0], fftBuffer[1] = sinPart[0], asf. fftFrameSize
must be a power of 2. It expects a complex input signal (see footnote 2),
ie. when working with 'common' audio signals our input signal has to be
passed as {in[0],0.,in[1],0.,in[2],0.,...} asf. In that case, the transform
of the frequencies of interest is in fftBuffer[0...fftFrameSize].
*/
{
float wr, wi, arg, *p1, *p2, temp;
float tr, ti, ur, ui, *p1r, *p1i, *p2r, *p2i;
long i, bitm, j, le, le2, k;
for (i = 2; i < 2 * fftFrameSize - 2; i += 2) {
for (bitm = 2, j = 0; bitm < 2 * fftFrameSize; bitm <<= 1) {
if (i & bitm) {
j++;
}
j <<= 1;
}
if (i < j) {
p1 = fftBuffer + i;
p2 = fftBuffer + j;
temp = *p1;
*(p1++) = *p2;
*(p2++) = temp;
temp = *p1;
*p1 = *p2;
*p2 = temp;
}
}
for (k = 0, le = 2; k < (long)(log((double)fftFrameSize) / log(2.) + .5); k++) {
le <<= 1;
le2 = le >> 1;
ur = 1.0;
ui = 0.0;
arg = Math_PI / (le2 >> 1);
wr = cos(arg);
wi = sign * sin(arg);
for (j = 0; j < le2; j += 2) {
p1r = fftBuffer + j;
p1i = p1r + 1;
p2r = p1r + le2;
p2i = p2r + 1;
for (i = j; i < 2 * fftFrameSize; i += le) {
tr = *p2r * ur - *p2i * ui;
ti = *p2r * ui + *p2i * ur;
*p2r = *p1r - tr;
*p2i = *p1i - ti;
*p1r += tr;
*p1i += ti;
p1r += le;
p1i += le;
p2r += le;
p2i += le;
}
tr = ur * wr - ui * wi;
ui = ur * wi + ui * wr;
ur = tr;
}
}
}
void AudioEffectSpectrumAnalyzerInstance::process(const AudioFrame *p_src_frames, AudioFrame *p_dst_frames, int p_frame_count) {
uint64_t time = OS::get_singleton()->get_ticks_usec();
//copy everything over first, since this only really does capture
for (int i = 0; i < p_frame_count; i++) {
p_dst_frames[i] = p_src_frames[i];
}
//capture spectrum
while (p_frame_count) {
int to_fill = fft_size * 2 - temporal_fft_pos;
to_fill = MIN(to_fill, p_frame_count);
float *fftw = temporal_fft.ptrw();
for (int i = 0; i < to_fill; i++) { //left and right buffers
float window = -0.5 * Math::cos(2.0 * Math_PI * (double)temporal_fft_pos / (double)fft_size) + 0.5;
fftw[temporal_fft_pos * 2] = window * p_src_frames->l;
fftw[temporal_fft_pos * 2 + 1] = 0;
fftw[(temporal_fft_pos + fft_size * 2) * 2] = window * p_src_frames->r;
fftw[(temporal_fft_pos + fft_size * 2) * 2 + 1] = 0;
++p_src_frames;
++temporal_fft_pos;
}
p_frame_count -= to_fill;
if (temporal_fft_pos == fft_size * 2) {
//time to do a FFT
smbFft(fftw, fft_size * 2, -1);
smbFft(fftw + fft_size * 4, fft_size * 2, -1);
int next = (fft_pos + 1) % fft_count;
AudioFrame *hw = (AudioFrame *)fft_history[next].ptr(); //do not use write, avoid cow
for (int i = 0; i < fft_size; i++) {
//abs(vec)/fft_size normalizes each frequency
hw[i].l = Vector2(fftw[i * 2], fftw[i * 2 + 1]).length() / float(fft_size);
hw[i].r = Vector2(fftw[fft_size * 4 + i * 2], fftw[fft_size * 4 + i * 2 + 1]).length() / float(fft_size);
}
fft_pos = next; //swap
temporal_fft_pos = 0;
}
}
//determine time of capture
double remainer_sec = (temporal_fft_pos / mix_rate); //subtract remainder from mix time
last_fft_time = time - uint64_t(remainer_sec * 1000000.0);
}
void AudioEffectSpectrumAnalyzerInstance::_bind_methods() {
ClassDB::bind_method(D_METHOD("get_magnitude_for_frequency_range", "from_hz", "to_hz", "mode"), &AudioEffectSpectrumAnalyzerInstance::get_magnitude_for_frequency_range, DEFVAL(MAGNITUDE_MAX));
BIND_ENUM_CONSTANT(MAGNITUDE_AVERAGE);
BIND_ENUM_CONSTANT(MAGNITUDE_MAX);
}
Vector2 AudioEffectSpectrumAnalyzerInstance::get_magnitude_for_frequency_range(float p_begin, float p_end, MagnitudeMode p_mode) const {
if (last_fft_time == 0) {
return Vector2();
}
uint64_t time = OS::get_singleton()->get_ticks_usec();
float diff = double(time - last_fft_time) / 1000000.0 + base->get_tap_back_pos();
diff -= AudioServer::get_singleton()->get_output_latency();
float fft_time_size = float(fft_size) / mix_rate;
int fft_index = fft_pos;
while (diff > fft_time_size) {
diff -= fft_time_size;
fft_index -= 1;
if (fft_index < 0) {
fft_index = fft_count - 1;
}
}
int begin_pos = p_begin * fft_size / (mix_rate * 0.5);
int end_pos = p_end * fft_size / (mix_rate * 0.5);
begin_pos = CLAMP(begin_pos, 0, fft_size - 1);
end_pos = CLAMP(end_pos, 0, fft_size - 1);
if (begin_pos > end_pos) {
SWAP(begin_pos, end_pos);
}
const AudioFrame *r = fft_history[fft_index].ptr();
if (p_mode == MAGNITUDE_AVERAGE) {
Vector2 avg;
for (int i = begin_pos; i <= end_pos; i++) {
avg += Vector2(r[i]);
}
avg /= float(end_pos - begin_pos + 1);
return avg;
} else {
Vector2 max;
for (int i = begin_pos; i <= end_pos; i++) {
max.x = MAX(max.x, r[i].l);
max.y = MAX(max.y, r[i].r);
}
return max;
}
}
Ref<AudioEffectInstance> AudioEffectSpectrumAnalyzer::instance() {
Ref<AudioEffectSpectrumAnalyzerInstance> ins;
ins.instance();
ins->base = Ref<AudioEffectSpectrumAnalyzer>(this);
static const int fft_sizes[FFT_SIZE_MAX] = { 256, 512, 1024, 2048, 4096 };
ins->fft_size = fft_sizes[fft_size];
ins->mix_rate = AudioServer::get_singleton()->get_mix_rate();
ins->fft_count = (buffer_length / (float(ins->fft_size) / ins->mix_rate)) + 1;
ins->fft_pos = 0;
ins->last_fft_time = 0;
ins->fft_history.resize(ins->fft_count);
ins->temporal_fft.resize(ins->fft_size * 8); //x2 stereo, x2 amount of samples for freqs, x2 for input
ins->temporal_fft_pos = 0;
for (int i = 0; i < ins->fft_count; i++) {
ins->fft_history.write[i].resize(ins->fft_size); //only magnitude matters
for (int j = 0; j < ins->fft_size; j++) {
ins->fft_history.write[i].write[j] = AudioFrame(0, 0);
}
}
return ins;
}
void AudioEffectSpectrumAnalyzer::set_buffer_length(float p_seconds) {
buffer_length = p_seconds;
}
float AudioEffectSpectrumAnalyzer::get_buffer_length() const {
return buffer_length;
}
void AudioEffectSpectrumAnalyzer::set_tap_back_pos(float p_seconds) {
tapback_pos = p_seconds;
}
float AudioEffectSpectrumAnalyzer::get_tap_back_pos() const {
return tapback_pos;
}
void AudioEffectSpectrumAnalyzer::set_fft_size(FFT_Size p_fft_size) {
ERR_FAIL_INDEX(p_fft_size, FFT_SIZE_MAX);
fft_size = p_fft_size;
}
AudioEffectSpectrumAnalyzer::FFT_Size AudioEffectSpectrumAnalyzer::get_fft_size() const {
return fft_size;
}
void AudioEffectSpectrumAnalyzer::_bind_methods() {
ClassDB::bind_method(D_METHOD("set_buffer_length", "seconds"), &AudioEffectSpectrumAnalyzer::set_buffer_length);
ClassDB::bind_method(D_METHOD("get_buffer_length"), &AudioEffectSpectrumAnalyzer::get_buffer_length);
ClassDB::bind_method(D_METHOD("set_tap_back_pos", "seconds"), &AudioEffectSpectrumAnalyzer::set_tap_back_pos);
ClassDB::bind_method(D_METHOD("get_tap_back_pos"), &AudioEffectSpectrumAnalyzer::get_tap_back_pos);
ClassDB::bind_method(D_METHOD("set_fft_size", "size"), &AudioEffectSpectrumAnalyzer::set_fft_size);
ClassDB::bind_method(D_METHOD("get_fft_size"), &AudioEffectSpectrumAnalyzer::get_fft_size);
ADD_PROPERTY(PropertyInfo(Variant::REAL, "buffer_length", PROPERTY_HINT_RANGE, "0.1,4,0.1"), "set_buffer_length", "get_buffer_length");
ADD_PROPERTY(PropertyInfo(Variant::REAL, "tap_back_pos", PROPERTY_HINT_RANGE, "0.1,4,0.1"), "set_tap_back_pos", "get_tap_back_pos");
ADD_PROPERTY(PropertyInfo(Variant::INT, "fft_size", PROPERTY_HINT_ENUM, "256,512,1024,2048,4096"), "set_fft_size", "get_fft_size");
BIND_ENUM_CONSTANT(FFT_SIZE_256);
BIND_ENUM_CONSTANT(FFT_SIZE_512);
BIND_ENUM_CONSTANT(FFT_SIZE_1024);
BIND_ENUM_CONSTANT(FFT_SIZE_2048);
BIND_ENUM_CONSTANT(FFT_SIZE_4096);
BIND_ENUM_CONSTANT(FFT_SIZE_MAX);
}
AudioEffectSpectrumAnalyzer::AudioEffectSpectrumAnalyzer() {
buffer_length = 2;
tapback_pos = 0.01;
fft_size = FFT_SIZE_1024;
}
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