Maybe stupid question about Digital PCM Filter and oversampling

[Vamp898]

The linear phase default filter is transparent. Minimum phase filters cause phase distortion that might not be transparent to human ears. What people prefer is not the same thing as what is transparent. If you prefer phase distortion then you probably prefer minimum phase filter over linear phase filters.

The default filter is not linear phase, that is my point. At least not with the DACs i own/tested.

It should be, i 100% agree with that, but for whatever reason i do not understand, with modern DAC from ESS and Cirrus (not sure about AKM), linear phase filter are no longer the default. At least not with the DAC i have used so far. They all default to minimum (non linear) phase out of the box...

The worst thing is, the manual doesn't even tell you that. Some DAC give you some nonsense descriptions likes "Makes everything sound bigger" or "Focuses on vocals" or stuff like that. I had to download the technical specification from the homepage of the DAC Maker to lookup which filter does what.

In the DAWN PRO, the linear phase filter is called "Phase compensated" which confused me because i haven't seen that wording for linear phase filter before. The CS43131 documentation also doesn't mention that, Moondrop made it up for whatever reason. I understand what they mean with it but it was confusing.

So sadly, in this day and age, we can no longer trust that an DAC, out of the box, sounds correct.

 

[JamoBroGuy]

The default filter is not linear phase, that is my point. At least not with the DACs i own/tested.

It should be, i 100% agree with that, but for whatever reason i do not understand, with modern DAC from ESS and Cirrus (not sure about AKM), linear phase filter are no longer the default. At least not with the DAC i have used so far. They all default to minimum (non linear) phase out of the box...

The worst thing is, the manual doesn't even tell you that. Some DAC give you some nonsense descriptions likes "Makes everything sound bigger" or "Focuses on vocals" or stuff like that. I had to download the technical specification from the homepage of the DAC Maker to lookup which filter does what.

In the DAWN PRO, the linear phase filter is called "Phase compensated" which confused me because i haven't seen that wording for linear phase filter before. The CS43131 documentation also doesn't mention that, Moondrop made it up for whatever reason. I understand what they mean with it but it was confusing.

So sadly, in this day and age, we can no longer trust that an DAC, out of the box, sounds correct.

That’s really strange.
If I’m not wrong, iBasso dongles are set to linear phase by default, which all manufacturers should be doing. No idea why Moondrop made that BS up (can’t say I’m surprised though, coming from a company like them).

 

[theveterans]

So if these blind tests really have been done correctly, the default filter for most modern DACs is not transparent, so there is a reason to care

Yep nothing is 100% transparent. And lastly it's audible to some, but not to others (and 100% not audible to those who believe in absolute in ALL cases definition of audibility thresholds)

So sadly, in this day and age, we can no longer trust that an DAC, out of the box, sounds correct.

That's the crux of subjectivity. Whichever configuration sounds best to you. You can force yourself to believe linear phase is the only correct way to listen to music and everything else is not correct in absolutes, or you can simply just play around and have that dialed-in sound that you want with swapping DAC filters, using oversampling software, DSP, convolution, EQ etc.

"Phase compensated"

I interpreted it as having a linear group delay which means all frequencies arrive at the same time hence phase compensated aka linear phase

 

[Vamp898]

That’s really strange.
If I’m not wrong, iBasso dongles are set to linear phase by default, which all manufacturers should be doing. No idea why Moondrop made that BS up (can’t say I’m surprised though, coming from a company like them).

The iBasso usually are clean of that issue, except for the DC07 Pro, this one also has minimum phase as default, at least the model i had with Firmware 2.x (there is Firmware 3.x now, maybe that changed or did change from 1.x? Not sure, just guessing)

But i always check the Manual when i get a DAC and check which filter is Fast roll-off. On my D50III it was Filter 3..., its not even second choice...

But they are not to blame, that is how ESS made the DAC: www.esstech.com/wp-content/uploads/2024/04/ES9039MPRO_ES9039PRO_v0.3.2.pdf

Linear phase fast roll-off is the third filter... sad. They even call it legacy filter, like its some old, outdated stuff that is soon going to be removed or so (It probably does not, likely just bad wording)

 

[castleofargh]

There are tons of discussions about if people can hear the difference in DAC filters or not.

Unrelated to if these are able to be distinguished, just assume someone can for the sake of this question

Wouldn't oversampling to >=88.2kHz just irrelevant that? Why not just resample everything to >=88.2kHz and don't care?

Did i overlook something? I mean it doesn't really matter in the end anyway as most people will never hear the difference between the DAC filters but given they do, as soon you upsample, these effects should all be gone (not gone but way beyond 30kHz) or not?

Somehow i see no reason why upsampling should not solve this unless i overlooked something.

What @danadam said. If you oversample a file before sending it to the DAC, that operation also uses/needs a filter(or several) just above the initial sample rate/2. You don't solve the issue of the filter, you just apply the digital filter early.

 

[colonelkernel8]

Yep nothing is 100% transparent. And lastly it's audible to some, but not to others (and 100% not audible to those who believe in absolute in ALL cases definition of audibility thresholds)

Most decent (as in not from a BS laden audiophoolery peddler) enthusiast audio electronics today are 100% transparent. There are "absolute in ALL cases" audibility thresholds because there are physical thresholds of the human ear. Everyone has slight variations in their hearing, but there are hard limits on the structure of the ear that prevent hearing ultrasonics for example. You will never hear 30kHz; no one will, and no one wants to.

 

[bigshot]

Multiple filters is like a child’s busy box for audiophools. They listen and strain to hear and build an expectation bias about which one sounds better. But it isn’t about “better”. It’s about worse. The standard filter built into DACs and players that don’t have a selection of filters works perfect- clean and transparent. The added filters degrade the sound in various ways to various degrees.

You don’t need 8 filters. You need one- the standard one with the highest fidelity to human ears.

 

[theveterans]

Most decent (as in not from a BS laden audiophoolery peddler) enthusiast audio electronics today are 100% transparent. There are "absolute in ALL cases" audibility thresholds because there are physical thresholds of the human ear. Everyone has slight variations in their hearing, but there are hard limits on the structure of the ear that prevent hearing ultrasonics for example. You will never hear 30kHz; no one will, and no one wants to.

I don't care about the 30Khz material. What I hear is difference in the presentation which is obviously less than 20KHz between different built-in (not outside like software oversampling) linear phase filters (steep or gentle or even brickwall) (so apples to apples in group delay)

 

[bigshot]

IMHO, there's literally no such thing as "accurate" that's because it doesn't exist

There is accurate playback for human ears. You know a lot of trivia about the bleeding edges of digital audio, but you don’t know much at all about the thresholds of human hearing. Our ears have hard and finite limits, and looking at the numbers on digital audio without a freakin’ clue about what your ears can and can’t hear is leading you down contextless rabbit holes of foolishness.

More isn’t always better. You need enough to do the job. You can go further, but the “sound” you’re scratching and clawing after is as useless as teats on a bull hog.

Most likely and even subjectively the filters matter less after feeding it with hi-res.

Hires just brings is closer to using the standard transparent filter with 16/44.1.

nothing is 100% transparent.

Complete hogwash. That comment reveals the depth of your lack of understanding of even the most basic concepts of audio reproduction.

Ignorance is OK. It’s curable by information. Mark Twain said we’re all ignorant, just on different subjects.

Willful ignorance is different. That’s being presented with the facts and making a conscious choice to ignore them. That isn’t ignorance. That is stupidity.

 

[Slaphead]

Simplification: 10*1/3 = 3.33 --> 3.33/(1/3)--> 9.99

For human ears, 9.99 or 10 does not matter, it will be the same, but in math 9.99 and 10 are not the same. 9.99 = 10 is wrong. So he is right, its irrelevant in the real world but the statement is true, you need unlimited bandwidth

Your simplification isn't correct.

10 x 1/3 is equal to 3.33... or exactly 1/3 not 3.33

Therefore 3.33.../(1/3) = 1

The ... after the number means recurring, meaning the repetition will go on to infinity.

Here's a quick proof of how this works:-

x = 0.999... - Therefore:

10x = 9.999... - Therefore:

10x = 9 + 0.999... - As x is equal to 0.999... the following is true

10x = 9 + x - Subtract x from both sides

9x = 9 - Divide both sides by 9 and you get:

x = 1

Which means that 9.99... is very definitely equal to 10

I guess I get what you're trying to say in that in any digital system the recurring part will get truncated. Shall we say 3.33 as an example, but an analogue system will also get truncated, but it will be more like 3.3327594.

Really it doesn't matter for what we hear, and also for what dogs, cats and bats hear to be honest.

 

[theveterans]

Really it doesn't matter for what we hear, and also for what dogs, cats and bats hear to be honest.

If you're peering at the analog signal beyond 20KHz in the analog domain, yes we won't hear it, but in digital domain, the differences in accuracy between 9.99 (bandwidth limited) and 9.9999 (still bandwidth limited) since 10 (infinite bandwidth is impossible to achieve) is audible to subjectivists

 

[eq1849]

Your simplification isn't correct.

10 x 1/3 is equal to 3.33... or exactly 1/3 not 3.33

Therefore 3.33.../(1/3) = 1

The ... after the number means recurring, meaning the repetition will go on to infinity.

Here's a quick proof of how this works:-

x = 0.999... - Therefore:

10x = 9.999... - Therefore:

10x = 9 + 0.999... - As x is equal to 0.999... the following is true

10x = 9 + x - Subtract x from both sides

9x = 9 - Divide both sides by 9 and you get:

x = 1

Which means that 9.99... is very definitely equal to 10

I guess I get what you're trying to say in that in any digital system the recurring part will get truncated. Shall we say 3.33 as an example, but an analogue system will also get truncated, but it will be more like 3.3327594.

Really it doesn't matter for what we hear, and also for what dogs, cats and bats hear to be honest.

9.99 is not the same number as 10.00.

Your argument seems to be small signals do not matter.

 

[bigshot]

I don’t ask for infinity. I ask for an attempt to be made to remove expectation bias and perceptual error from the test. It isn’t too much to ask since it’s done in just about all scientific tests of perception.

If you consider DBT to be impossible, you’re in the wrong forum, bruthuh.

 

[theveterans]

9.99 is not the same number as 10.00.

Your argument seems to be small signals do not matter.

They really don't according to their definition of audibility thresholds which to me does not hold at all

 

[bigshot]

You don’t know what the audible thresholds are.