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Sanskrit 10th MKII does anybody understand the 'filters'

Hotwetrat

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FL1 FL2 etc the manual just doesn't explain, I can't find anything useful on google or the review on here about these and what exactly they do...

Does anybody know ?

'Sharp Roll-Off Filter'
'Slow Roll-Off Filter'
ETC.....

WHAT !?

I mean I think I get the basics that it rolls off trying to reproduce frequencies so high we can't hear them anyway?? (is that right?) but .... Man I don't know exactly what does which and how and OML never been confused by a DAC before!
 
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companyja

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The filters are low-pass filters - they attenuate everything past the nyquist limit. For example, if you have a 44.1khz sampling rate, that means are frenquencies under 22.05khz are perfectly sampled. Anything above 22.05khz however will be garbage data unrelated to the original signal. Everything past 22.05khz should technically be completely eliminated, however you can't just cut audio with a knife so filters will attenuate to a certain degree. Modern AKM chips have the cutoff at 24khz even if you're sampling at 44.1khz, and offer a few different variants:

index.php


As you can see, if the filter is not sharp enough (if it's SLOW), it might start cutting off frequencies before nyquist, and in the audible range (20khz and under).

There's also several types of filters - in digital audio filtering isn't "free" and you're going to have to deal with one of two "artefacts" - if you use linear phase filters, you will add pre-ringing to your sound. This means that you're adding some frequencies before the actual sampled signal. This is important in music production as if you're filtering a lot of low-end for example with linear phase filters, you will start to hear pre-ringing as a whooshing type of sound. If you're however filtering things at 22-24khz, the pre-ringing frequencies will be around 22-24khz which is inaudible. So it's no problem to use a linear phase filter for low-pass DAC filtering.

If you use minimal phase filtering, you won't add pre-ringing which is good, but you will alter the phase of the signal. If you're altering phase of an instrument that was recorded by multiple microphones in the mix, it can cause cancellation of frequencies and make slight EQ sound extreme. If you're however filtering out frequencies over 24khz on a single audio stream using minimal phase, you're changing the phase of frequencies around 22-24khz which is again inaudible. There's a lot of debate about which one is more "perfect" but for all intents and purposes, it is inaudible.

Chip companies such as AKM offer a few of these filters with their chips - you can have your regular linear phase with pre-ringing, short delay which to my knowledge removes pre-ringing but increases post-ringing, and all these kinds of things. As you can see on the chart above, the impact to the audible range is minimal, and if you choose a sharp filter that doesn't start rolling off before 20khz, it will be inaudible

Here's I think an AKM marketing thing describing their filter names

soundcolor.jpg


Feel free to ignore the "remarks" and "positioning", there is nothing magical about these filters that will do any such thing. You can see the linear phase at the bottom with pre-ringing included
 
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Hotwetrat

Hotwetrat

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Really excellent explanation, my goodness that's a lot to be processing! I appreciate you taking the time.

I will use FL1 then. As it strikes me that's the 'inaudible' one.... Wait or FL3 Short Delay Sharp Roll-Off filter

Just for the record I did look at that graph before but had no idea what 'LEFT 1 LEFT 2' etc was....
 
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