On Sat, May 20, 2023 at 05:15:03PM +0200, Florian Paul Schmidt wrote: > 1. Are there any glaring oversights with this approach?
The minimum phase step is not really necessary, you could just use the linear phase version, the output of the IFFT rotated by half its size. If you do the minphase operation, the best way to do this using an FFT that is at least 4 times as long as your IR. In other words, take the linear phase version in the time domain, add zeros, then do the minphase calculation. > 2. What is a sensible approach to regularize this approach. If spectrum > 1 has any components near zero magnitude then the division is ill > suited. It works fine for my experiments with distorted guitars, but I > wonder whether I should e.g. just clamp the values of spectrum 1 to > something like -70 dB? This probably should not depend on the absolute values, just on the ratio, i.e. the calculated gain. If r = s2 / s1 you could impose a smooth maximum: g = r / sqrt (1 + M * r * r) with M in the range 1e-3 to 1e-2 Or you could even reduce the gain if r is really very high (which probably means there is nothing of value in s1 at that frequency: g = r / (1 + M * r * r * r) with M in range 1e-5 to 1e-4 Try this in gnuplot: plot [0:100] x / sqrt (1 + 1e-2 * x * x), x / sqrt (1 + 1e-3 * x * x) plot [0:100] x / (1 + 1e-4 * x * x * x), x / (1 + 1e-5 * x * x * x) to get an idea of what these will do. Average levels in the two input files should be more or less matched, before applying any of these. > 3. If the user wants to smooth the spectrum before calculating the > responses what would be sensible approaches? I thought about smoothing s > with a filter that has a varying bandwidth when expressed in FFT bins or > Hz, but constant, when expressed in e.g. octaves or decades. I'm not > sure how to do that though. Any thoughts? That is a good idea. Above say 500 Hz anything smaller than say 1/3 octave probably doesn't matter much. There are many ways to do this. One would be to use the exact per-bin levels at low frequencies, and a moving average that increases in length as frequency goes up. Use an odd number of bins to sum (to avoid shifting peaks), and sum powers, not amplitudes. The window used in the analysis will also provide some smoothing, with a raised cosine anything at the center of a frequency bin will also show up 6 dB lower in the two adjacent bins. That limits the resolution as low frequencies, but that probably not a bad thing. Ciao, -- FA _______________________________________________ Linux-audio-dev mailing list -- linux-audio-dev@lists.linuxaudio.org To unsubscribe send an email to linux-audio-dev-le...@lists.linuxaudio.org
