Converting sampling frequency: best method?

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Converting sampling frequency: best method?

Unread post by EXer » Fri Oct 06, 2017 8:49 pm

I have extracted samples from a Roland S-550 sampler file and I have saved them to .wav files.

Now I want to use them in my EX5, but their sampling frequency is 30 kHz so I have to convert them to 44,1 kHz (I have tried them 'as is' but they sound horribly flat Image ).

My DAW asks me which interpolation method I want to use: linear, spline, 2nd degree polynomial, 3rd degree polynomial, 4th degree polynomial, narrow band, or none.

► Which method should I use to get the best possible result, given that those samples are single notes of a solo harpsichord?

It also asks me if I want to use anti-aliasing. I guess that for upsampling anti-aliasing is not needed?

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Re: Converting sampling frequency: best method?

Unread post by SeaGtGruff » Sat Oct 07, 2017 8:28 am

Hmm, on a guess, I'd think that a polynomial would be more accurate, and the higher the degree the better, so I'd try 4th degree polynomial and see how that works. If you don't like the results, try one of the other options. I'd say that "none" should give you the closest thing to the original 30 kHz in a 44.1 kHz form, since the sampled values would just be repeated as needed rather than having the computer try to guess (interpolate) what the missing values might be.
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