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PCM filters - useful, simple understanding! (Topping DX3 Pro+)

Fraxo

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First I'd like to thank any of you who comment, I appreciate your time reaching out to help!

So I'm not highly familiar with different filters in hardware and what they cause. If it's similar to LP filters in the digital audio engineering domain - then I'd assume some pre ringing could be an issue if it's linear, phasing could be prevalent in others, high frequency rolloff etc...
I care a lot about transient accuracy for subtle mix work, but in hardware the terminology is not so clear to me and I assume there's more to it...
Could you please simplify this for me for the different choices in hardware?

1) Which do you recommend for DX3 Pro+ (photos below) for transient accuracy (percussions\plucks) and why.
2) What main artifacts are to be expected with music.
3) What's the most common choice in the Hi-Fi world? (curiosity mainly).


WhatsApp Image 2022-03-28 at 07.30.01.jpeg
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WhatsApp Image 2022-03-28 at 07.30.01 (1).jpeg


Thanks again people!
 
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dc655321

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Those are low pass filters.
Choose a linear phase filter with a 20kHz passband. That will avoid any potential phase issues. Pre ringing is not a thing for properly bandwidth limited input.

No idea what the most common choice would be in the hifi community. Probably not the one that the makes the most sense from an electrical/signal theory perspective… :rolleyes:
 
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Fraxo

Fraxo

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@Ricardus @dc655321
Thank you for the comments.
Could you touch on how transients\impulse response would be affected by different options?

I was hoping to get a deeper understanding of this subject, since people give different opinions - I'd love to understand what the recommendations are based on.
Thanks :)
 
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dc655321

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If the filter passband is flat and covers the audio bandwidth, 20Hz-20kHz, then input signals in that passband will be output unscathed.

For deeper understanding I recommend studying signal theory at the university level.

Question for you: define transient.
Are you thinking of a percussion strike? Something like that?
 
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Fraxo

Fraxo

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If the filter passband is flat and covers the audio bandwidth, 20Hz-20kHz, then input signals in that passband will be output unscathed.

For deeper understanding I recommend studying signal theory at the university level.

Question for you: define transient.
Are you thinking of a percussion strike? Something like that?
I will do my best to dive into that, and yes, percussive strikes indeed. I'm obsessed with it for accuracy.

I was thinking that even without the deep science atm - the bottom line of how it affects sound is known to knowledgeable people like yourself - so at least with a simplified explanation I'll pick correctly until the extremely detailed info follows...
 
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Fraxo

Fraxo

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The default, F-3. This chart from AKM may assist;

index.php



JSmith
Thanks! The filter titles don't directly correspond to the Pro+ version but this gives some idea.
Still very confused as to what I'm seeing here. Ik I want no enhancements so I'd prioritize the original sound, but I see some things I don't understand:

1) On some - the impulse response is shifted to the right and on some it starts right away, does that mean anything?
2) "Super Slow Roll Off" looks very different from the rest, not sure what that means.
3) How does all of it translate to recreating the sound of percussive elements such as drums and plucks?

Ugh I looked through threads and couldn't find anything... I tried reading this source but was still left confused as it didn't really explain what I practically needed. It's very hard to find sources about this, I only ask here as a last resort guys, I hope that it's clear I'm willing to do the research myself when possible.
 

JSmith

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I want no enhancements
To be honest mate most people can't pick or hear any difference between these filters.

This blog may assist you with some more detailed info, testing etc.; (edit: oh sorry you've found this already!)
But if you want what is clearly the most technically accurate – that is, high fidelity – output from your DAC, choose fast roll-off or sharp-roll-off, and linear phase (or no named phase) rather than minimal phase.
This blurb by AKM is rather subjective, but may assist in understanding the intention of allowing user access to these filters;


JSmith
 
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Fraxo

Fraxo

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To be honest mate most people can't pick or hear any difference between these filters.

This blog may assist you with some more detailed info, testing etc.; (edit: oh sorry you've found this already!)

This blurb by AKM is rather subjective, but may assist in understanding the intention of allowing user access to these filters;


JSmith
Haha yea u see..? :) I've actually tried to do my due diligence and found some sources but non address this... My ears are extremely trained so even if it's subtle I'd really like to choose what's best for accuracy (in dynamics) for my needs.

Could you tell me though - in the photos I've sent of the Pro+'s filters - which one is "short delay"?
I'd assume that linear isn't short delay but not quite sure...

I'll check out the second link you've sent.
 

Jimbob54

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This gives an interesting look at DA filters http://archimago.blogspot.com/2013/06/measurements-digital-filters-and.html

Regarding superslow- have a look at p52 of the RME Dac manual for some interesting commentary. https://www.rme-audio.de/downloads/adi2dac_e.pdf

I think F3 is SD sharp and F4 is SD slow (see also the RME link above- same page)

PS if your ears are that good, you might be the ideal candidate to do some blind testing to determine which sound best to you. Easiest blind test ever- just need a blindfold and helper to switch the filters on the DAC during playback- record if there is any pattern to whether you can discern a difference with a looped music sample and the filter switched (or not) each time- record if you have a preference.

Also- regarding the position of the impulse in the window- thats time. the further to the left, the lower the delay in ms. Of course, if you are just listening to stereo audio with no accompanying visuals, a delay isnt an issue.

My very limited understanding is that there is always a trade off between the technically most perfect impluse response and most perfect frequency response. So linear slow filters give the best impulse response but are terribly leaky for out of band frequencies )the yellow on your first pic). Sharp filters (like f1) are messy in the time domain but offer the best attenuation of out of band frequencies.

I would imagine in unsighted listening tests for people of a decent age with natural roll off of the higher frequencies you would be lucky to tell which is which in normal playback. Of course, if you select the filter looking at all the narrative and images, the linear slow filters will give you the most perfect transients ;-)
 
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AnalogSteph

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The traditional default is F7, the sharp rolloff linear phase half-band filter. It has the best in-band characteristics in terms of frequency response flatness and ripple, at the expense of a minimal amount of images past fs/2 (but those generally being of signals past 20 kHz, and there tends to be little signal energy up there to begin with, so you are very unlikely to bother any following amplifiers). I still consider this a very worthwhile tradeoff.

If you need lowest latency, its minimal phase cousin F5 is not a bad choice. Being minimal phase, phase response invariably goes south near fs/2. (I think it's an elliptic filter.)

Once you start insisting on a response that really stops at fs/2 but cannot afford increased filter length (which in turn affects latency), you are paying the price in reduced bandwidth (F2) or higher in-band periodic ripple (F1, F3). F3 still is a decent filter, but I don't think it's the best tradeoff in a practical audio application.

I honestly struggle to think of any circumstances where one of the slow rolloff filters (F4, F6) would be worth it.

If you want super awesome digital filtering, you can always output at a much higher sample rate (say, 192 kHz or even 384 kHz) and use the SoX resampler for upsampling in software. Its characteristics put basically any of the on-chip filters to shame. Yes, you'll add a few ms of latency, but who cares in a playback-only application. (This is only critical for live monitoring through an A/D-D/A chain, where even a few ms may throw off a vocalist.)
 
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Fraxo

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Yess I was hoping for a response like this. Please allow me pick your brain a bit :)
...It has the best in-band characteristics in terms of frequency response flatness and ripple, at the expense of a minimal amount of images past fs/2 (but those generally being of signals past 20 kHz
If you need lowest latency, its minimal phase cousin F5 is not a bad choice. Being minimal phase, phase response invariably goes south near fs/2. (I think it's an elliptic filter.)
Once you start insisting on a response that really stops at fs/2 but cannot afford increased filter length (which in turn affects latency), you are paying the price in reduced bandwidth (F2) or higher in-band periodic ripple (F1, F3). F3 still is a decent filter, but I don't think it's the best tradeoff in a practical audio application.
Could you simplify those terms (or link a reading source for it) in the context of audio gear?
- Ripple
- Images past fs/2
- Latency
- I'm very familiar with filters and eq in the digital mixing world where linear phase would cause more latency but won't affect FR as much etc, but how does it relate to hardware and how would you quantify it? The latency specs didn't change on the DX3 Pro+ win10 panel when I changed filters, I suspect there's a different meaning that I wish someone could explain or link a source to read because I'm not sure how it affects the sound of a device.
- Filter length - I assume it's referring to the slope of the cutoff point. What I would usually call the Q Factor. Once again, ik what it does in mixing (creates more phasing\ringing), not sure how it relates to hardware that goes beyond 20K Hz and how it works exactly.
- Tradeoffs - Perhaps a summarized version of what is attempted to be achieved by applying filters, and what are the downsides\audible side affects of each method.

I can not tell you how much this could simplify things for me... I'd read for days if I had the sources for it and I'm not new to audio, I am new to the scientific audio gear science so could use help with it. Thanks!
 
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