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a099a77e82
LibDSP can greatly benefit from this nice FFT implementation, so let's move it into the fitting library :^) Note that this now requires linking SoundPlayer against LibDSP. That's not an issue (LibDSP is rather small currently anyways), as we can probably make great use of it in the future anyways.
62 lines
1.8 KiB
C++
62 lines
1.8 KiB
C++
/*
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* Copyright (c) 2021, Cesar Torres <shortanemoia@protonmail.com>
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*
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* SPDX-License-Identifier: BSD-2-Clause
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*/
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#include "FFT.h"
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#include <AK/Complex.h>
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#include <AK/Math.h>
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namespace LibDSP {
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// This function uses the input vector as output too. therefore, if you wish to
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// leave it intact, pass a copy to this function
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//
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// The sampling frequency must be more than twice the frequency to resolve.
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// The sample window must be at least large enough to reflect the periodicity
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// of the smallest frequency to be resolved.
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//
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// For example, to resolve a 10 KHz and a 2 Hz sine waves we need at least
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// a samplerate of 20 KHz and a window of 0.5 seconds
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//
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// If invert is true, this function computes the inverse discrete fourier transform.
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//
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// The data vector must be a power of 2
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// Adapted from https://cp-algorithms.com/algebra/fft.html
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void fft(Vector<Complex<double>>& sample_data, bool invert)
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{
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int n = sample_data.size();
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auto data = sample_data.data();
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for (int i = 1, j = 0; i < n; i++) {
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int bit = n >> 1;
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for (; j & bit; bit >>= 1)
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j ^= bit;
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j ^= bit;
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if (i < j)
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swap(data[i], data[j]);
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}
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for (int len = 2; len <= n; len <<= 1) {
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double ang = 2 * AK::Pi<double> / len * (invert ? -1 : 1);
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Complex<double> wlen(AK::cos(ang), AK::sin(ang));
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for (int i = 0; i < n; i += len) {
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Complex<double> w = { 1., 0. };
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for (int j = 0; j < len / 2; j++) {
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Complex<double> u = data[i + j], v = data[i + j + len / 2] * w;
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data[i + j] = u + v;
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data[i + j + len / 2] = u - v;
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w *= wlen;
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}
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}
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}
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if (invert) {
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for (int i = 0; i < n; i++)
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data[i] /= n;
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}
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}
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}
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