This one? https://www.signalyst.com/consumer.html
That's the one. Now again, I'm not hearing much of a difference in actual sound from DSD/PCM - but this will upsample all your stuff on the fly in the software as opposed to the DAC doing the heavy lifting. Which arguably has its benefits.
The impressive DSD jitter results are mostly by Jussi (Miska) the author of HQPlayer.
He’s also the one showing charts up into the MHz range. His claim is that noise there might become audible with some equipment due to intermodulation into the audible band. I’m skeptical that this is the case, but don’t have the equipment to measure it.
If there is IM in the normal audio band we can measure it. Without commenting on the likely hood of this, even though its technically feasible, do we have any measurements that correlate and support the proposition?
Another view is don't use an amp that is wide open to RF frequencies.
Do you have any preferred settings for 16/44.1, 16/48, 24/88.2, 24/96, 24/192, DSD64 and DSD128?By measuring both audio band and out of band behavior. Traditional measurements on audio band (20 kHz and 100 kHz). And checking that there are no correlated components or discrete spurious tones above audio band, such as images for example.
Do you have any preferred settings for 16/44.1, 16/48, 24/88.2, 24/96, 24/192, DSD64 and DSD128?
If there is IM in the normal audio band we can measure it. Without commenting on the likely hood of this, even though its technically feasible, do we have any measurements that correlate and support the proposition?
Another view is don't use an amp that is wide open to RF frequencies.
Digital audio systems are band limited by definition. They will be so at the front end whether it is due to the microphone, it's amp, the mixer etc or the ADC it's bandwidth setting and anti alias filter.
To reproduce a square wave perfectly you require infinite bandwidth as it has harmonics that stretch to infinity. The distorted part of a square wave are harmonics outside of the bandwidth of the system. In audio system that should be outside of the normal audio band.
The iFi Nano post-DAC filter is two RC stages. I've never seen a Micro, so I can't say if it differs.Yeah, I think both iFi and Marantz in question above have 2nd order analog filters.
Accidentally building such a thing was a useful lesson for my 12-year-old self.Basically I don't want an amp that picks up my local AM radio station
Aahh, didn't see the bit about bw limited square wave.
So we are talking about the effects of any phase shift.
Do you have any research links that show the subjective audibility?
Secondly what about the band limiting that I mentioned already exists in any recording?
Accidentally building such a thing was a useful lesson for my 12-year-old self.
I don't start with research about audibility, but about absolute perfectness. Mathematically we are supposed to be able to reconstruct the signal perfectly, so that's what I'm looking for. I don't take stance on how many out of billions of people are able to hear what.
I'm talking about reconstruction in it's all aspects, for which one part is phase response. Others are clean step response, etc. These are all related but different views of the same thing.
P.S. Reason why I chose 7 kHz square is that is not in sync with 44.1k sampling rate and it still has fundamental and first harmonic within the RedBook Nyquist band. But the test is really about hires (PCM or DSD, doesn't matter), at or above 352.8k sampling rate the analog filter usually begins to dominate anyway. 3 kHz would probably work fine too.
I applaud your quest but searching for perfection IMO is a fruitless search. You will never get there.
We have a test on this forum where member @Blumlein 88 has performed an 8x loopback recording of some tracks and so far everybody is really struggling to correctly identify the 8th generation against the original.
It would be interesting to see how you perform, its just good fun dont take it seriously, but I think it does put things into perspective.
I know, and I don't have a problem with that. Keeps me busy. I have long ago accepted that we mere humans will never reach perfectness on anything.
Are they? Elsewhere I explained why in my opinion the DAC in question (yours!) is actually partially fixing faults of the original. Opening the files in audio editor the difference is immediately obvious.
But that loop is heavily bandlimited, so one needs to understand what it can represent and what it cannot. It is more bandlimited than for example output of your DAC. (further band-limiting is done by the ADC's anti-alias and decimation filters that also have other effects)
I don't generally use myself as a yardstick. I have my interest areas and I work on those and I hear differences on those areas. For hard numbers I put my measurement rigs at work. My quest is "what can I do to with DSP to make this overall system perform better". Since I'm limited by time and resources I just hope hardware manufacturers come up with better DACs and don't settle in sentiment of "good enough". Imagine if people would have done that when CD was first rolled out! If I could put 72 hours in a day I could do more hardware stuff. Now I just applaud every time I see new DACs coming out from someone!
I have to admit I'm sometimes tired of listening the same test material 100th time during a day!