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LibAudio+LibDSP: Switch samples to 32-bit float instead of 64-bit float

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This has been overkill from the start, and it has been bugging me for a
long time. With this change, we're probably a bit slower on most
platforms but save huge amounts of space with all in-memory sample
datastructures.
This commit is contained in:
kleines Filmröllchen 2022-05-06 22:14:16 +02:00 • committed by Linus Groh
parent 39c0f31009
commit 19a4b820c4
16 changed files with 329 additions and 329 deletions
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12
Userland/Libraries/LibDSP/FFT.h
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@ -12,7 +12,7 @@
namespace LibDSP {
constexpr void fft(Span<Complex<double>> sample_data, bool invert = false)
constexpr void fft(Span<Complex<float>> sample_data, bool invert = false)
{
int n = sample_data.size();
@ -27,13 +27,13 @@ constexpr void fft(Span<Complex<double>> sample_data, bool invert = false)
}
for (int len = 2; len <= n; len <<= 1) {
double ang = 2 * AK::Pi<double> / len * (invert ? -1 : 1);
Complex<double> wlen = Complex<double>::from_polar(1., ang);
float ang = 2 * AK::Pi<float> / static_cast<float>(len * (invert ? -1 : 1));
Complex<float> wlen = Complex<float>::from_polar(1.f, ang);
for (int i = 0; i < n; i += len) {
Complex<double> w = { 1., 0. };
Complex<float> w = { 1., 0. };
for (int j = 0; j < len / 2; j++) {
Complex<double> u = sample_data[i + j];
Complex<double> v = sample_data[i + j + len / 2] * w;
Complex<float> u = sample_data[i + j];
Complex<float> v = sample_data[i + j + len / 2] * w;
sample_data[i + j] = u + v;
sample_data[i + j + len / 2] = u - v;
w *= wlen;