/* * Copyright (c) 2021, Cesar Torres * * SPDX-License-Identifier: BSD-2-Clause */ #include "AudioAlgorithms.h" #include #include // 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>& 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 / len * (invert ? -1 : 1); Complex wlen(AK::cos(ang), AK::sin(ang)); for (int i = 0; i < n; i += len) { Complex w = { 1., 0. }; for (int j = 0; j < len / 2; j++) { Complex 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; } }