Re: [LAU] jack/oversampling

From: Jörn Nettingsmeier <nettings@email-addr-hidden>
Date: Sun Mar 16 2014 - 22:54:02 EET

On 03/16/2014 09:49 PM, Fons Adriaensen wrote:
> On Sun, Mar 16, 2014 at 07:50:57PM +0100, Jörn Nettingsmeier wrote:
>
>> i don't understand this. can you elaborate? what is "blt"?
>
> Bi-Linear Transform, a mathematical trick used to transform
> the transfer function of an analog filter (in the s-domain)
> into that of a digital one (in the z-domain).

ah, thanks. i had guessed that it probably wasn't "bacon/lettuce/tomato"...

> It introduces a 'warping' of the frequency axis. If A(f)
> is the frequency response of the analog filter, and D(f)
> that of the digital one, they will be different but there
> is a function F(f) such that D(F(f)) = A(f). And it's
> always possible to arrange things such that the 'defining'
> point of the FR (the -3 dB point, or the center frequency)
> is correct - by applying the inverse of F(f).
>
> For low frequencies F(f) = f, so the two filter are the
> same. But as f->inf, F(f)->Fs/2. So filters in the upper
> part of the frequency range will have a different shape
> of the FR. For example a standard 2nd order parametric
> will have a symmetric shape when plotted on a log frequency
> scale, but the equivalent digital one won't if the center
> frequency is high. But there is no reason to say that one
> is 'better' than the other, there is no reason why any
> digital filter used in audio processing should be an exact
> copy of an analog one.

thanks! time to review some things... (/me grabs old hardcopy of
dspguide.com)

-- 
Jörn Nettingsmeier
Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487
Meister für Veranstaltungstechnik (Bühne/Studio)
Tonmeister VDT
http://stackingdwarves.net
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Received on Mon Mar 17 00:15:07 2014

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