Galileo365

Two things: 1. You’re quoting me out context. I say in the article itself that I’m referring to a “pure” ladder DAC or binary-weighted resistor DAC, not an R/2R DAC, which I say in the article is an inherently compromised design. 2. Despit that, your picture is of a sine way. It just has slightly longer than usual intervals between the points on the curve. A square looks like an exact crossection of a staircase. it doesn’t matter to me, but you seem to be trying so hard to prove that I am wrong on almost everyhthing I write—for no practical purpose—that you must be really frightened of me. I suggest that you stop reading my blog because it’s making you so nervous and get some professional help.

Ecactly what slack did I get when? Are you saying that just because my first response was not, “Thank you proving to me that MQA sucks”, means that I get kicked out of the club? This is supposed to be Computer Audiophile, not We Hate MQA, or so I thought.

I assume you are referring to yourself here?

Wow, you got that all from a few words in marketing speak, so your basis that the Mytek ADC as a way to test MQA in practice is invalid comes without any formal analysis. Are you really so scared that you might be wrong that you need to poison the idea of a controlled experiment before it starts?

I am quite certain that I am not.

Your explanation was not brief and when I countered it logically you dismissively blew me off with yet another in a long series of personal attacks rather than new or more information and an explanation, a sign of a weak, feeble, and insecure mind for sure, and you only withdrew your attacks and dropped the subject when I reported your deragatory posts to the admins, and are now pretnding that I was the one being unreasonable because I did not, by default, regard you with god-like status. Sorry to bruise your ego, but that’s life in the big city. If you can’t take it, don’t dish it out.

I can spot a BS artist from a mile away, regardless of his or her background. mansr has already undermined his own credibility beyond repair from the perspective of any truly sensible and open-minded person.

I’m new to the forum and I don’t know those people at all, only know mansr by virtue of his abusive behavior. Maybe he forgets to take his Thorazine when they let him out of the nuthouse for the weekend. I said MQA is “supposed to do X”, something and that you can only prove it does or doesn’t do by looking both an MQA DAC and an MQA ADC, something that no one here has appparently thought to do. If you had read my very next post, you would have seen a controlled experiment designed to asses that, but maybe the person who reads for you given your apparent illiteracy had to leave for the day.

Myek is coming out with a consumer-level MQA ADC and, of course, they already make an MQA DAC. Ayre used to make a QA-9 and a QB-9 ADC and DAC, both of which you can find used. Take an analog recording, preferably from two-track R2R tape. Run that through an oscilloscope. Then connect the MQA ADC directly to the MQA DAC and the QA-9 to the QB-9. Run the signal from the tape through both parallel chains and analyze the output of both DAC’s with the oscilloscope. Then see if the trace on the MQA chain differs at all from the non-MQA chain and, if so, which trace more closely matches the pre-sampled trace. That will tell you more than reversing engineering an MQA DAC out of context. There may be no apparent difference at all or there might be but that’s hard empirical evidence. If all three traes look the same, perform some additional analysis using other measurement techniques to see if any difference ever shows up.

Without going into a lot of detail right now because I’m tired of getting flamed, reversing engineering MQA DAC’s is fine if you know how an MQA ADC works, because it is not supposed to be based on Shannon signal theory. Assuming that only the Shannon sampling model is valid and assessing MQA DAC’s from that perspective is like only evaluating quantum mechanics from a Newtonian perspective.

Quite sincerely and honestly, none whstsoever. I’m just trying to figure what it’s supposed to do, how it might work, and how to confirm that it does so. I would guess that most of those 21 posts involve defending myself for “asking stupid questions”.

It you read my post carefully, you will find the explanation. I don’t know if you have a background in analog electronics or just signal theory,, but certain terms overlap between the two areas with very different meanings, That's the last word I have to say on the matter.

I studied logic, math, computer science, philosophy, literature, and a field called semiotics at Stanford. The dictionary definition was meant to disambiguate the use of the term filter in the mainstream from how it applies to signal processing such as interpolation, although I did not make that exact point very clear. Fourier analysis is a family of mathematical techniques, all based on decomposing signals into sinusoids. The discrete Fourier transform (DFT) is usually used with digitized signals. The shift theorem says that a delay in the time domain corresponds to linear phase, a property of a filter. The result is that all frequency components of the input signal are shifted in time by the same amount, referred to as the group delay. There is no phase distortion due to the time delay of frequencies relative to one another. The predominant type of digital filter used in audio is called an FIR (Finite Impulse Response) filter. High-quality sampling ensures that no aliasing occurs by lowpass filtering the signal. A example digital filter would bring a 24 kHz signal down to about -100db. That's a really stiff slope and, in analog electronics, the filter would have a phase shift or non-uniform delay from frequency to frequency. However, digital “filters” are different from analog “filters” because digital “filters” interpolate, an active process in the digital domain, not “filter” passively in the analog domain. So FIR digital “filters” have no group delay. That brings me around to my original point, that the word “filter” in the digital domain is a misnomer given the common meaning of “filter”. So insisting, out of context, using self-accolades and derogatory comments aboit me rather than elucidation (a rhetorical strategy as opposed to a logically deductive proof) that interpolation and filtering are the same thing in general will only confuse the layperson. The “time smearing” or phase shift I refer to occurs as the result of passive filtering in the analog domain, often but not always employed in DAC’s for the consumer market. So when you say that MQA uses “filters” as opposed to interpolation without explication, you are making an inherently ambiguous and self-contradictory statement. Trying to dazzle anyone with long lists of marhemstivsl terms to appear to have superior knowledge without actually disambiguating terms and saying precisely what you mean will not work on me. It’s the equivalent of robbing a bank and offering, as your defense, not if you did or did not rob the bank, but saying simply, “I’m not the kind of person who would rob a bank”. So either say what you mean or stop commenting in dismissive and derogatory ways, because you’re making an ass of yourself as opposed to actually informing anyone.

I suggest that you stop assuming I haven’t already done so and consider the possibility that you are in error.

Yes, I know; although, the mathematics behind that presume a “perfect” brick-wall filter, as one example, which is something that at least currently cannot be built. Plus, Fourier transforms work well at expressing attributes of the frequency domain but not so much the time domain, convolution notwithstanding. So using this particular point to say that interpolation and filtering equivocate particularly in reference to “temporal smearing” has no mathematical support and little basis in reality.