Re: [linux-audio-dev] Resampling

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Subject: Re: [linux-audio-dev] Resampling
From: Juhana Sadeharju (kouhia_AT_nic.funet.fi)
Date: Thu Aug 09 2001 - 20:08:59 EEST


>From: Alexander Ehlert <alexander.ehlert_AT_uni-tuebingen.de>
>
>Why 4096 FFT array?? I just use any blocksize. FFTW supports blocksizes
>other than exponents of 2. The problem is, that the so called

You would make a good patent lawyer because you would now be able to
grand a patent for any other FFT array length than 4096. ;-)
(It was just an example. :-)

>wether I should torture the user with real high quality converting
>rather than do it quick and dirty. Yesterday I imported a 4 Minutes
>Stereo mp3 file with 44,1Khz and imported it to 48khz and it took
>10 Minutes on my Dual PIII 600. The ifft used a blocksize of 2228
>with 8times oversampling. And for this blocksize FFTW performs really bad.
>Maybe 8 times oversampling is overkill, too? Benno?

I guess it doesn't matter if the cutoff frequency is not that exact.
Because FFT uses equal-width bands and because ear fits to logarithmic
scale, a loss of an extra frequencies (an FFT pin too much) on the top
end means quite less. You could reduce the oversampling if it only changes
the accuracy of the cutoff point, not the SNR.

I suggest to test the thing. I would be happy if frequencies up to
19000 Hz are preserved within -90 dB (or -120 dB). Note that common
A/D converters are good only for 80% - 90% of the total 22050 Hz
frequency range.

Regards,

Juhana

7140ab14c74516d269317a5a65373b1d


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