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LibDSP: Generalize & improve FFT

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Several related improvements to our Fast Fourier Transform
implementation:
- FFT now operates on spans, allowing it to use many more container
  types other than Vector. It's intended anyways that FFT transmutes the
  input data.
- FFT is now constexpr, moving the implementation to the header and
  removing the cpp file. This means that if we have static collections
  of samples, we can transform them at compile time.
- sample_data.data() weirdness is now gone.
This commit is contained in:
kleines Filmröllchen 2022-03-03 21:16:30 +01:00 • committed by Andreas Kling
parent 428d4ae337
commit 00dd8f8fbe
4 changed files with 36 additions and 66 deletions
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2
Userland/Applications/SoundPlayer/BarsVisualizationWidget.cpp
View file

@ -26,7 +26,7 @@ void BarsVisualizationWidget::paint_event(GUI::PaintEvent& event)
if (m_sample_buffer.is_empty())
return;
LibDSP::fft(m_sample_buffer, false);
LibDSP::fft(m_sample_buffer.span(), false);
double max = AK::sqrt(m_sample_count * 2.);
double freq_bin = m_samplerate / (double)m_sample_count;

1
Userland/Libraries/LibDSP/CMakeLists.txt
View file

@ -3,7 +3,6 @@ set(SOURCES
Track.cpp
Effects.cpp
Synthesizers.cpp
FFT.cpp
)
serenity_lib(LibDSP dsp)

62
Userland/Libraries/LibDSP/FFT.cpp
View file

@ -1,62 +0,0 @@
/*
* Copyright (c) 2021, Cesar Torres <shortanemoia@protonmail.com>
*
* SPDX-License-Identifier: BSD-2-Clause
*/
#include "FFT.h"
#include <AK/Complex.h>
#include <AK/Math.h>
namespace LibDSP {
// This function uses the input vector as output too. therefore, if you wish to
// leave it intact, pass a copy to this function
//
// The sampling frequency must be more than twice the frequency to resolve.
// The sample window must be at least large enough to reflect the periodicity
// of the smallest frequency to be resolved.
//
// For example, to resolve a 10 KHz and a 2 Hz sine waves we need at least
// a samplerate of 20 KHz and a window of 0.5 seconds
//
// If invert is true, this function computes the inverse discrete fourier transform.
//
// The data vector must be a power of 2
// Adapted from https://cp-algorithms.com/algebra/fft.html
void fft(Vector<Complex<double>>& sample_data, bool invert)
{
int n = sample_data.size();
auto data = sample_data.data();
for (int i = 1, j = 0; i < n; i++) {
int bit = n >> 1;
for (; j & bit; bit >>= 1)
j ^= bit;
j ^= bit;
if (i < j)
swap(data[i], data[j]);
}
for (int len = 2; len <= n; len <<= 1) {
double ang = 2 * AK::Pi<double> / len * (invert ? -1 : 1);
Complex<double> wlen(AK::cos(ang), AK::sin(ang));
for (int i = 0; i < n; i += len) {
Complex<double> w = { 1., 0. };
for (int j = 0; j < len / 2; j++) {
Complex<double> u = data[i + j], v = data[i + j + len / 2] * w;
data[i + j] = u + v;
data[i + j + len / 2] = u - v;
w *= wlen;
}
}
}
if (invert) {
for (int i = 0; i < n; i++)
data[i] /= n;
}
}
}

37
Userland/Libraries/LibDSP/FFT.h
View file

@ -7,10 +7,43 @@
#pragma once
#include <AK/Complex.h>
#include <AK/Vector.h>
#include <AK/Math.h>
#include <AK/Span.h>
namespace LibDSP {
void fft(Vector<Complex<double>>& sample_data, bool invert = false);
constexpr void fft(Span<Complex<double>> sample_data, bool invert = false)
{
int n = sample_data.size();
for (int i = 1, j = 0; i < n; i++) {
int bit = n >> 1;
for (; j & bit; bit >>= 1)
j ^= bit;
j ^= bit;
if (i < j)
swap(sample_data[i], sample_data[j]);
}
for (int len = 2; len <= n; len <<= 1) {
double ang = 2 * AK::Pi<double> / len * (invert ? -1 : 1);
Complex<double> wlen(AK::cos(ang), AK::sin(ang));
for (int i = 0; i < n; i += len) {
Complex<double> w = { 1., 0. };
for (int j = 0; j < len / 2; j++) {
Complex<double> u = sample_data[i + j], v = sample_data[i + j + len / 2] * w;
sample_data[i + j] = u + v;
sample_data[i + j + len / 2] = u - v;
w *= wlen;
}
}
}
if (invert) {
for (int i = 0; i < n; i++)
sample_data[i] /= n;
}
}
}