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DAC Filters

MarkWinston

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When it comes to filters, which one do you choose for the most honest and faithful reproduction? Everyone will have their own prefences but that does not mean its the closest to the source, if its even audible in the first place. How do we know which one is the most accurate?
 
The guys from SMSL have a description of what each filters does. Does that descriptions make any sense? Thanks.


smsl_filters.jpg
 
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If you care up sample in software and not in the dac.

Mathematically "correct" is https://en.wikipedia.org/wiki/Sinc_filter
But it requires infinite delay and preringing to be "perfect"
So you might make some compromises to get good but not perfect results without infinite time ;)

Brick-wall filters that run in realtime are not physically realizable as they have infinite latency (i.e., its compact support in the frequency domain forces its time response not to have compact support meaning that it is ever-lasting) and infinite order (i.e., the response cannot be expressed as a linear differential equation with a finite sum), but approximate implementations are sometimes used and they are frequently called brick-wall filters.
 
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The guys from SMSL have a description of what each filters does. Does that descriptions make any sense? Thanks.


View attachment 174591
The descriptions are BS. The sharp Rolloff filter is the correct one as all others do not adhere to a correctly implemented reconstruction filter (frequencies above fs/2 need to be suppressed 90 dB or more).
 
The descriptions are BS. The sharp Rolloff filter is the correct one as all others do not adhere to a correctly implemented reconstruction filter (frequencies above fs/2 need to be suppressed 90 dB or more).
It was taken right out from AKM's marketing material of the AK4493/5/7 DAC chips. "Super Slow" is actually NOS btw.
 
It was taken right out from AKM's marketing material of the AK4493/5/7 DAC chips. "Super Slow" is actually NOS btw.
I had the intuition it wasn't SMSL's material. I guess there should be some element of truth, or at least some basis to state such things.
 
I’m not capable of doing the math , but I do believe the others ( as I read elsewhere ) when they say brick wall is the most correct .

They’re may be other reasons to choose a slightly lees step filter , bu t I don’t design DAC’s

My understanding is that :

If you can discern these differences some or all off them are wrongly implemented ,for example some of the slow filters I’ve seen graphed makes an audible impact down towards 15kHz . I think the whole filter craze is actually due to the unwanted audible contributions . These are then by audiophile logic transformed to something good.

The DAC designer ( not the chip manufacturer) should be able to make the correct choice for each sample rate and it should not be a user parameter to fudge with .

The chip makers such as AKM or ESS ofcourse wants it to be fudge factor for the end user :) and rightly so we all now recognise the specific chip in our DAC’s and AVR’s and for some reason care even if we should not ?
Audiophile mythological tales have it to be akin choosing a moving coil cart to suite your turntable, when it’s nothing like that . Implementation is everything ( ESS hump anyone) in the correct design it would not matter sonically. But there can probably be countless other engineering decisions involved in picking the proper chip for your design.
 
Sharp Roll Off it is then. In fact thats the default on my dac.
 
The discussion about filters goes something like this:

Me: I prefer the manufacturer had used the standard sharp roll off linear phase filter.
My Protagonist: I hate that brick wall effect. It ruins the sound stage, the music sounds dead.

Me: I suspect people claiming they can hear a difference are just stating a preference for the aliasing artifacts i.e. they like certain types of distortion.
MP: Don't worry about it, in all likelihood cannot hear above 10K and you definitely cannot hear above 20K.

Me: I am embarrassed to admit it but probably true. If we can't hear it why change the filter.
MP: I hate that brick wall effect. It ruins the sound stage, the music sounds dead.

Me: I suspect people claiming they can hear a difference are just stating a preference for the aliasing artifacts i.e. they like certain types of distortion.
MP: Don't worry about it, in all likelihood cannot hear above 10K and you definitely cannot hear above 20K.

and so on ...
 
The mathematically ideal reconstruction filter is known (sinc) but impossible -- or at least inconvenient -- to realize. Therefore, approximations are used in DACs, with a careful weighting of what's more relevant vs. what is less relevant in practice.

When investigated thoroughly it turns out that the frequency response is the dominant factor. NOS and "Slow" filters have significant roll-off of the treble (up to several dB's by the time they reach 20kHz) and that is clearly audible as long as your hearing is still good to, say 15kHz and above... with source material that has enough content there (white noise preferred here as test signal).

The attenuation at fs/2 is way less important, as is the general attenuation above fs/2. Time-domain aspect (post-ringing only vs. pre-and post-ringing, or in other words, the phase response) is also not that important. And the higher the sampling rate the more so:
  • Music seldom contains significant amounts of energy around, let alone above 20kHz.
  • Any imaged components (frequencies mirrored at fs/2) are above normal hearing range for most people. With 48kHz or greater everything is 100% out of the hearing range of anybody in the world.
  • Any downstream components that are not total junk will not produce any relevant additional intermodulation distortion products from the mixing of the images with the original content. And if they are junk, the normal IMD from signals below fs/2 is already dominating the picture, a bit more junk added doesn't have any real impact anymore.
  • Any normal music signals also do not contain single dirac samples which would actually excite the ringing. But even when it is excited, it is at fs/2, so above normal hearing range.
Go here for a demonstration what different filters would sound like if we could hear beyond 80kHz.

Bottom line. Don't worry too much about DAC filters, and use the "Sharp" types which have the flattest frequency response. But even when using NOS, you can apply a bit of EQ to make it flat up to 20kHz and then the ususally reported differences don't expose themselves anymore.

For ADC's (Analog to Digital Converters), the filters are way more important as insufficient filters acctually corrupt the signals unrecoverably, adding aliased signals (stuff above fs/2 folded down below fs/2).
 
OK, it's an old thread. But I've found it while looking for information on this topic. And to add on the "Sharp Roll-off is the only filter that makes sense", I'd like to add a resource that I find great for this:
 
OK, it's an old thread. But I've found it while looking for information on this topic. And to add on the "Sharp Roll-off is the only filter that makes sense", I'd like to add a resource that I find great for this:
Glad you like this piece. There really isn't too much around about exactly what filters do, apart from various subjective impressions. So I measured them all to find out.
 
In my listening experience, one of the clearly distinguishable differences was the presence or absence of pre-ringing in DAC filters.

Energy occurring before the actual transient can smear spatial cues and reduce image focus, even when conventional measurements such as frequency response and distortion remain unchanged.
 
In my listening experience, one of the clearly distinguishable differences was the presence or absence of pre-ringing in DAC filters.

Energy occurring before the actual transient can smear spatial cues and reduce image focus, even when conventional measurements such as frequency response and distortion remain unchanged.
There is no pre-ringing with normal music content.

 
In my listening experience, one of the clearly distinguishable differences was the presence or absence of pre-ringing in DAC filters.

Energy occurring before the actual transient can smear spatial cues and reduce image focus, even when conventional measurements such as frequency response and distortion remain unchanged.
Did you test that using blind tests or with sighted listening? I would be very surprised if you managed to detect that in an unbiased test.

The explanation you give sounds good, but does not hold when looking at real music. The "awful" pre-ringing plots shown in DAC datasheets and many websites present the impulse response of those filters. An impulse has unlimited bandwidth, though - real music does not. The ringing would also mainly occur at Nyquist frequency, if you managed to provoke it using an artificial signal like an impulse. That would be about 22 kHz for CD quality signals and much higher for high res stuff. So unless you are below roughly 25 years old and listen to CD quality stuff at best, you would not even be able to hear the ringing frequency.

You can read more about digital filters and ringing here. It's a good overview.
 
My point is not that “pre-ringing exists in music,” but that a playback system can introduce time-domain artifacts that interact with musical transients in a way that some listeners may perceive as unnatural, even if those artifacts are not present as standalone audible events.
If you would have taken the times to look at the links provided, you would know that this is not so.
 
Did you test that using blind tests or with sighted listening? I would be very surprised if you managed to detect that in an unbiased test.
Whether you believe my experience or not is fine. I’m not presenting this as a test or a blind evaluation.

For context, when I was less familiar with DAC filters, I used a FiiO K9 ESS for a long time with the Slow Roll-off (linear-phase) filter, which involves pre-ringing. Over time, the sound consistently felt smeared and unfocused. I didn’t connect it to the filter at first, since I had forgotten which one was selected. (So it’s probably not just pre-ringing. Slow Roll-off filters can sound blurred for several time-domain reasons)

After revisiting the settings, switching to a minimum-phase filter resolved the issue for me. I’m not claiming universal audibility. Just describing how the time-domain behavior of the Slow Roll-off filter affected my own listening.
 
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