I'm completely lost now, brain overload. I'm listening to my CDs of the Beethoven piano concertos recorded by Gilels and Szell and they are magnificent. At the risk of sounding like a rampant subjectivist that's what really matters to me.
OK, so now phase response doesn't matter either. In fact, nothing seems to matter.
I'm completely lost now, brain overload. I'm listening to my CDs of the Beethoven piano concertos recorded by Gilels and Szell and they are magnificent. At the risk of sounding like a rampant subjectivist that's what really matters to me.
I'm completely lost now, brain overload. I'm listening to my CDs of the Beethoven piano concertos recorded by Gilels and Szell and they are magnificent. At the risk of sounding like a rampant subjectivist that's what really matters to me.
I guess this is what happens when you call another engineer’s technicals «faulty» without taking the time to back up the accusation with evidence.
I have done so.
I said real music energy (not test tones) in the transition band isnt high. I demonstrated so. The level its at puts aliases in the noise floor.
Archimagos data which I can replicate shows ultrasonic noise with real music (not test tones) is not an issue.
So even if you look at the highest signal of -36dB, the alias are buried in the noise floor. Where is the problem?
Broadcast radio starts at 148kHz
I said real music energy (not test tones) in the transition band isnt high. I demonstrated so. The level its at puts in band aliases in the noise floor.
I didnt say that. I asked you to show some data regarding the audibility of phase error.
@Blumlein 88 data showed a reduction of -85dB before you hit the audible band (20kHz).
Agreed, Going back to the OP,At this point the rest of it is just throwing barbs back and forth.
I measure also higher, but there's usually nothing because the analog reconstruction filter cuts out. Typical analog filter is 2nd order with fc=100kHz so it has rolled off dirty output by 5 MHz already.
Taking one of the dirties devices I have (in that respect), Focusrite Forte. Almost all the the junk is in first 10 MHz band.
Without starting to write my own story here now, here's one example from other people:
https://www.diyaudio.com/forums/class-d/321632-hypex-ncore-nc400-input-anti-alias-filter.html
So any intermodulation products they may create in later stages (like power amp) are also fully correlated. So if you hear it sounds like artificial hardness.
One of the things you can simulate...
Why not multi-tone? Images are correlated multi-tone because they reflect music spectra around multiples of sampling rate.
Where does this -60 dB come from? It is not necessarily even that much down.
If it can be measured - it matters. If it cannot be measured, it still matters, but measurement resolution needs to be improved...
I don't like to make positive assumptions about amp behavior, rather negative worst-case scenarios. So I aim for clean DAC output that represents proper reconstruction instead of "half-assed" one.
Yes, my own listening tests. Not blind. When I have two algorithms at hand, I have no reason to prefer A over B. I have my favorite algorithms based on listening. Some other people prefer other ones, that's completely OK for me. I offer options to choose from.
No you may not.
I don't know why what I hear matters to anybody else except me in first place. I usually don't talk much about my listening experiences unless specifically asked. Sometimes if people ask what are my favorite settings, then I tell my current favorite.
Now what I don't understand is why this particular thing has caused this kind of shit storm towards me. I was just sharing my point of view about this detail based on spending 20+ years designing digital filters.
It an Aura. Which is pretty much the M3 in better build quality. And yes depending on the settings on the interface it is around .75 ohm at the upper frequencies. Here is a plot from long ago of the A1, but it is pretty much the same as mine in impedance.Arse!
I knew some clever dick would spoil things Im aam revising my post to 0.5 ohms.
Which model is it? Are you sure its 0.75 ohms?
Thats not actually a speaker, thats a link of wire. That will cause a world of pain for amps.
Class D isn't what most people use and is very obvious that it needs an input filter, like wise ADCs.
You can simulate the sound of artificial hardness and perception ?
Strange thesis. If something measured is so far below audibility why would it matter for audio ?
Don't forget .. there are other folks with more than 20 years of experience in audio here as well.
I think you very well realize that not all 'experts' have similar thoughts.
Sure, I'm a bit confused about the response though.
I first explained how I think particular choice of ESS digital filter in his DAC fixes certain common problem from source material used in the test tracks created by @Blumlein 88 . I'm kind of surprised of the response. Based on the response he could of course also go back to his chip vendor (ESS) and complain that they in first place even offer any filter options at all because none of that matter. Instead of complaining about that to me... I have my opinion about the matter and he is not going to change that anyway.
But this thread has gone massively off-topic at least. The only relevance to OT is that with DSD ADC you don't need digital decimation filters at all so this issue being discussed now doesn't exist.
Some upper limit of hearing figures for a very small sample set of 32 people:
https://asa.scitation.org/doi/10.1121/1.2761883
Now for fun you can factor in potential boost of about 20 dB from metal dome tweeter resonance somewhere between 20 kHz and 25 kHz.
But anyway, I perplexed to see the gap between subjectivist who say "you cannot measure what I hear" and objectivists saying "you cannot hear what I measure". Now I understand better what kind gap there really is!
And we are still talking about some of the more basic stuff. We haven't even touched modulator properties and performance yet. I guess most here have not heard ESS presentation about their modulator design and subjective listening tests about the sound.
Really, huh!
The first generation has already cut off those transition band frequencies. The test file in this respect sounds like listening to the March DAC. Which is using filtering more or less as it should be. You then say the DAC has fixed the original file, but it would do this simply by listening to that DAC.
With the assumption such things do make a difference in sound. I don't see how this is a complaint. You seem to have made an issue of what some DACs do because the one I used doesn't do that. I still feel like I'm misunderstanding you somehow.
Yes, that was my point.
No, for some people it seems to be an issue that I said the DAC in question cleaned that part due to choice of the particular ESS filter. I didn't complain anything, it was an observation from the file and DAC data. First I was wondering which side it was, DA or AD until you told the DAC which pointed to the roll-off curve which immediately described it was DA side. Then I compared that to plots in ESS datasheet to figure out which of the filters it likely was.
Some/many other DACs don't do that, so listening to the DA+AD loop file may actually sound relatively better in some respects through such DACs even though it has gone through the extra conversion loop and degraded in other ways.
Maybe people got upset because they know my software gives choice of oversampling filters that do similar cleanup. Or something, I just don't know.
All it takes for the manufacturers is to use a good input filter and there should not be much of a problem.
Maybe you could persuade Amir to measure the output of DACs well into the MHZ range as well as amplifiers.
Some things don't need fixing though and work fine as they are.
They may not measure that great though.