R2R/multibit vs Delta-Sigma - Is There A Measurable Scientific Difference That's Audible
Dec 15, 2016 at 1:43 PM Post #901 of 1,344
Could anyone summarise this topic? Do two different DACs with flat fr differ any bit?

 
1. Implementation is just as important as the specific DAC chip technology.  Don't pick something just because it has your favorite species of chip.
 
2. Remember that almost all modern recordings are made using Sigma-Delta AD converters before you get too crazy purist about the superiority of R2R (unless you're only going to listen to digital recordings from the last century, and no modern re-masterings).
 
3. R2R vs Delta-Sigma is most importantly about what a given designer is most comfortable with.  They're both just tools, capable of glory or disaster in the right or wrong hands.  If a designer does her best work using R2R, then you may have a solid reason for preferring R2R from that vendor (e.g. Schiit). On the other hand, if a given designer pushes the envelope of DS design, that's probably the better product from that vendor (e.g. NAD Master Series).
 
3. For R2R DACs that use DSP filters, the filter is really the secret sauce that has the biggest effect on the sound (aside from the analog output stage).  There are real, measurable differences between IIR and FIR filters in the realm of phase and impulse response.  How much this is audible is a different question.  But it is real.
 
4. Except for the cheapest gear, old gear, or disastrously crappy connections, jitter is probably not worth worrying much about now.  The difference between 500 picoseconds and 100 picoseconds of jitter in the audio band is more about epeen and bragging rights than anything audible.
 
5. Pay more attention to the analog stage than the chipset (unless you need/want DSD).  A beefy analog stage with good power supply and isolation, coupled to a ho-hum DAC chip, will beat a razzle-dazzle expensive DAC chip coupled to a ho-hum analog stage.
 
6. DACs are not turntables or tape decks or tuners or other highly flawed, distorted, electromechanical analog sources, where the difference between the best and the worst is both huge and costly.  $99 DACs can be perfectly listenable, regardless of chipset, with only minor flaws.  The same wasn't true of budget tape decks or budget turntables.  So keep things in perspective.
 
Dec 15, 2016 at 3:15 PM Post #902 of 1,344
One part of the problem is that the measurable and visible differences constitute differences in ringing waveforms that are themselves ultrasonic in nature. For example, how does a short sharp spike sound as compared to the same spike with several cycles of ultrasonic ringing before and after it? Some would argue that, since the ringing is at inaudible frequencies, you can't possibly hear it, so it doesn't matter. Others will say that, because the ringing takes power away from the primary spike and spreads it in time, the spike itself is stretched, and its energy delivered over a longer period of time. The result is that nothing extra is heard, but the "character" of what is heard seems to alter slightly. There's also the possibility that, even if the ringing itself is inaudible, it could still cause interactions or distortions which are audible.
 
The other problem is that the differences we're talking about are difficult to define. To me, it's the difference between whether a wire brush hitting a cymbal sounds more like, well, little wires hitting a hunk of metal, or more like a steam valve going "tssss  tsssss tssss". I'm not quite sure how I'd describe the differences between those two sounds, but they seem different to me. (I can say that it sounds to me more like, with the latter, the individual little noises of wires hitting metal seem blurred together, but that image in my head could be due to what I BELIEVE to be the cause.)
 
I can also say that I've tried a sort of double blind test on several DACs I currently own and been able to reliably detect differences between filters.
We're talking about a test where I had a friend switch between the filters, and I could identify which filter was being used, as could he when our roles were reversed.
 
Note:
1) we're talking about filters which were both arbitrarily very flat in terms of frequency response (so the differences there should be quite inaudible)
2) the difference was only apparent on certain few recordings - and not on many others
3) the difference was only apparent with certain models of speakers and headphones
4) on one of the DACs, a Wyred4Sound DAC2, of the six filters it offers, only three of them sounded distinctly different (to me)
5) we are talking about very tiny differences of the sort that are only noticeable when you switch back and forth between the choices multiple times
 
Please note that I'm not suggesting which is better, or even which is more technically correct, or even whether the differences matter - simply that there are in fact audible differences... which suggests to me that we don't know everything yet. For example, while there have been plenty of studies done on the range of frequencies audible to humans, they've all been done with continuous steady state sine waves. (Most folks agree that most humans with excellent hearing can hear frequencies between 20 Hz and 20 kHz - although one study extends the bottom of the range to 10 Hz.) However, less direct concepts, like how much phase shift between the left and right it takes to produce an apparent shift in the left/right position of something in the sound stage, are more difficult to measure, And measurements that involve transient and varying waveforms even more so. (Virtually all of the math that describes things like the Nyquist frequency, and the ability to accurately represent waveforms with limited numbers of samples, assumes a continuous sine wave signal.... which music is NOT. Likewise, even the math that describes how a DAC works assumes that the DAC uses something called a "SinC function" when, in reality, the sampling process used by DACs is really only an approximation of it.)  
  Quote:
   
That was a wonderful synopsis and as fair as I've read on the topic under discussion.  Though, it should be noted exactly how you go about noticing these subtle differences in filters.  Is there a high probability that what you seem to be noticing is actually nothing more than sighted bias?  Also, for those filter changes, could some of these actually be altering the sound significantly enough to show a measurable difference that many would consider to be audible?

 
Dec 15, 2016 at 3:31 PM Post #903 of 1,344
Thanks for the additional details, if only for my sake.
 
Have you heard inexpensive (<$200) DACs that could reproduce wire brushes accurately?  In other words, did you find a correlation to price and accuracy, or is it just a specific filtering technique that you found preferable?
 
Dec 15, 2016 at 4:07 PM Post #904 of 1,344
To be quite honest, I've heard some DACs in the sub-$200 range that sound very good to me, and I've also heard some very expensive DACs that seemed to me to NOT do a very good job..... so I wouldn't say that I've found much correlation between cost and performance at all. To me, the biggest distinction seems to be between DACs that do their best to be neutral (which I prefer), and DACs which were clearly designed with the intent of "sounding a certain way" - which is a nice way of saying that they have some sort of deliberate coloration.
 
I would also have to say that I haven't found a specific correlation between accurate sound and a particular type of filter. (While I notice slight differences between the filter choices on a given DAC, and also find that some DAcs seem to sound more natural than others to me, there's not one specific filter TYPE that I prefer across multiple DACs. Of the two DACs I often use which have multiple filters, I do NOT prefer the same filter on both, and I have other DACs which only offer a single filter which I find to be equally good.)
 
(While having multiple filters on a single DAC is a convenient way of demonstrating that the filters make a difference, while holding everything else equal, I do not find having multiple filters to be an important feature. I'm quite satisfied with a DAC that only has one filter, as long as it is one which sounds neutral to me, and seems to handle transients well.)
  Quote:
  Thanks for the additional details, if only for my sake.
 
Have you heard inexpensive (<$200) DACs that could reproduce wire brushes accurately?  In other words, did you find a correlation to price and accuracy, or is it just a specific filtering technique that you found preferable?

 
Dec 15, 2016 at 4:29 PM Post #905 of 1,344
  Thanks for the additional details, if only for my sake.
 
Have you heard inexpensive (<$200) DACs that could reproduce wire brushes accurately?  In other words, did you find a correlation to price and accuracy, or is it just a specific filtering technique that you found preferable?

 
I find wire brushes to be more dependent on tweeters and crossovers than on DACs.
 
Dec 15, 2016 at 6:17 PM Post #906 of 1,344
I agree entirely.......
 
And, as I said, we're talking about subtle differences here.
I have a pair of powered monitors (Emotiva Stealth 8's) with folded ribbon (AMT) tweeters - and I noticed the differences with those.
And I found them somewhat obvious on my Koss electrostatic headphones.
However, I didn't hear any difference on my AKG headphones or another pair of speakers I had at the time with soft dome tweeters.
And, out of a dozen or so high-res albums I had at the time, I only noticed a difference with one or two of them.
 
So, as I've said before, I'm not suggesting that we're talking about significant or important differences.
Or even that most people would notice or care......
(However, if even a few people, and only under certain circumstances, hear them, then they are in fact audible.....)
 
Personally, I've kind of reached the conclusion that, if I have to pull out one or two select "'test albums" in order to be able to hear something, then it's probably not terribly important. However, if you do happen to notice the difference between various DACs, then it does make sense to buy the one that sounds better to you.
 
 
Quote:
   
I find wire brushes to be more dependent on tweeters and crossovers than on DACs.

 
Dec 17, 2016 at 6:25 AM Post #907 of 1,344
...simply that there are in fact audible differences... which suggests to me that we don't know everything yet. [1] For example, while there have been plenty of studies done on the range of frequencies audible to humans, they've all been done with continuous steady state sine waves. ([2] Most folks agree that most humans with excellent hearing can hear frequencies between 20 Hz and 20 kHz - although one study extends the bottom of the range to 10 Hz.) [3] However, less direct concepts, like how much phase shift between the left and right it takes to produce an apparent shift in the left/right position of something in the sound stage, are more difficult to measure, And measurements that involve transient and varying waveforms even more so. ([4] Virtually all of the math that describes things like the Nyquist frequency, and the ability to accurately represent waveforms with limited numbers of samples, assumes a continuous sine wave signal.... which music is NOT. [4] Likewise, even the math that describes how a DAC works assumes that the DAC uses something called a "SinC function" when, in reality, the sampling process used by DACs is really only an approximation of it.)

 
Hopefully without causing too much ill feeling, I have to say that I find some of your posts a little troubling. It's not that they're overtly incorrect, it's that they sometimes appear (IMHO) to expand the grey areas, to suggest implications which are, or very easily could be, misleading. Although it maybe entirely a coincidence, some of your statements appear virtually idetical to those exploited by marketing departments in order to deliberately mislead consumers and then quoted (typically inappropriately) by audiophiles. How many times have we seen these types of responses from audiophiles when challenged over some ludicrous claim: "Science doesn't know everything yet", "we can't measure everything", "music isn't just a sine wave", "digital audio is just an approximation", etc?
 
1. While strictly true, this statement omits some pertinent facts and the implication is: Maybe if we used music instead of "continuous steady state sine waves" we might discover that the range of audible frequencies extends higher than currently accepted. However: A. With music, other test signals (such as noise) and short duration signals (like transients) our ability to discern high frequencies diminishes. We use steady state sine waves in this case because it presents the best case scenario for success, which allows us to say with significant confidence that if (for example) your limit is a 19kHz isolated, continuous, steady state sine wave, frequencies higher than 19kHz (of equal amplitude) will be inaudible to you in any other circumstances. B. There have been many more tests/studies which effectively test the human audibility of high and ultrasonic frequencies and which used music rather than sine waves as the test material (HiRez vs CD tests for example). However, as they were not tests aimed solely/specifically at testing the frequency limits of audibility, your statement is strictly true though not, IMHO, generally true.
 
2. I can't speak for "most folks" but again "generally" I would disagree. When I was a lecturer in audio engineering, my colleagues and I informally blind tested first year students, the vast majority of whom were 18-20 year olds, in small groups, as part of the "Listening Skills" module. We tested about 300 a year and I was there for 6 years. The vast majority struggled beyond 17kHz and were out by 18kHz, a dozen or so made it to 19kHz, not one of the roughly 1800 tested could detect 20kHz. In the analogue days, the BBC restricted broadcast TV to 15kHz but for several years broadcast an accompanying 19kHz pilot tone. None of the tens of millions of viewers appeared to notice when it started, when it stopped or during the period. In fact, it wasn't until many years later (the early 90's I believe), when some BBC broadcast engineer mentioned it, that anyone was aware it had ever occurred.
 
3. It's not difficult to measure phase shift, although it is difficult to measure the human perception of positioning from it, not least because that perception varies with frequency (and amplitude). Therefore, we don't know THE answer because there is no one answer. What we do know is that phase shift which affects positional perception is typically in the millisecond range and with best case test signals can extend down to the microsecond range. If we take any perception at all (not necessary just perception of positioning) of timing/phase then we're down into the hundreds of nano second range but still hundreds of times above what any moderately competent DAC should achieve.
 
4. This is the most obviously incorrect statement, although I'm not a mathematician and so might be mistaken. As I understand it, Nyquist's theory does not mention or assume sine waves, in fact it does not consider continuous signals of any type. The actual Sampling Theorem (which describes and proves the Nyquist frequency), also does not assume a sine wave, it assumes any mathematical function which has a Fourier transform, EG. All acoustic sound waves (regardless of complexity), which obviously does include all music! In fact, in his seminal 1948 paper "A Mathematical Theory of Communication" (the paper which proved the sampling theorem and why it is commonly called the Nyquist/Shannon Sampling Theorem), Shannon specifically states: "A communication system is designed not for a particular speech function and still less for a sine wave ...". From what I understand (which is admittedly limited) of the mathematical contributions to the sampling theorem, rather than say "virtually all the math ... assumes a continuous sine wave", I would say the opposite; none of the math assumes a sine wave! I'm not however disputing that many of the explanations/demonstrations of the sampling theorem use a single continuous sine wave as an example.
 
5. Although Shannon/Whittaker's perfect interpolation formula is a sinc function, the maths which explain how a modern DAC works does not, as far as I'm aware, assume this perfect sinc function, which is impractical to implement. Therefore other math (which doesn't assume a perfect sinc function) has been developed to get around the engineering practicalities. While I agree that a DAC is effectively approximating, we have to be very careful as the word "approximate" is both relative and frequently abused. Every work of engineering has a practical limit of accuracy, so the question isn't whether a DAC is approximating, it's by how much. With modern technology, even cheap DACs can/should be astonishingly accurate, with any approximations (interpolation errors) being well below audibility.
 
Again, this is NOT intended as an attack, I'm not even disputing your observations, just some of your wording/rationale suggesting an explanation for those observations. For example, with the first two sentences I've quoted; I'm not disputing your observation and I obviously can't dispute what those observations may suggest to you personally. However, what it suggests to me is not that "we" (science) doesn't know everything yet but that "we" (personally) don't know how the manufacturer/s are actually implementing their filters. It seems to me far more likely that some DAC manufacturers, especially those offering filter choices, maybe implementing filters deliberately designed to be audibly distinguishable. In all fairness, you do imply this possibility in a subsequent post. I think we need to be very careful, particularly here on head-fi, not to appear to invoke, support, rationalise or add fuel to the fire of audiophile myths. Just sayin'
 
G
 
Dec 22, 2016 at 11:21 PM Post #908 of 1,344
4. This is the most obviously incorrect statement, although I'm not a mathematician and so might be mistaken. As I understand it, Nyquist's theory does not mention or assume sine waves, in fact it does not consider continuous signals of any type. The actual Sampling Theorem (which describes and proves the Nyquist frequency), also does not assume a sine wave, it assumes any mathematical function which has a Fourier transform, EG. All acoustic sound waves (regardless of complexity), which obviously does include all music! In fact, in his seminal 1948 paper "A Mathematical Theory of Communication" (the paper which proved the sampling theorem and why it is commonly called the Nyquist/Shannon Sampling Theorem), Shannon specifically states: "A communication system is designed not for a particular speech function and still less for a sine wave ...". From what I understand (which is admittedly limited) of the mathematical contributions to the sampling theorem, rather than say "virtually all the math ... assumes a continuous sine wave", I would say the opposite; none of the math assumes a sine wave! I'm not however disputing that many of the explanations/demonstrations of the sampling theorem use a single continuous sine wave as an example.  
5. Although Shannon/Whittaker's perfect interpolation formula is a sinc function, the maths which explain how a modern DAC works does not, as far as I'm aware, assume this perfect sinc function, which is impractical to implement. Therefore other math (which doesn't assume a perfect sinc function) has been developed to get around the engineering practicalities. While I agree that a DAC is effectively approximating, we have to be very careful as the word "approximate" is both relative and frequently abused. Every work of engineering has a practical limit of accuracy, so the question isn't whether a DAC is approximating, it's by how much. With modern technology, even cheap DACs can/should be astonishingly accurate, with any approximations (interpolation errors) being well below audibility.

To elaborate on a few concepts of sampling, informally:
 
Just as you said, the sampling theorem does not assume sine waves, only a frequency representation(the existence of a Fourier transform). Sampling theorem invokes what is now called the Discrete-time Fourier Transform (DTFT) using discrete samples, impulses, to get a continuous spectrum. This DTFT representation does allow for the representation of sampled sinusoids as discrete spikes, but is not restricted to it.
 
The sampling theorem and sinc interpolation are too closely related to really be separate. Sampling has the pesky property of representing the spectrum from DC to Nyquist and the spectrum flipped and folded over integer multiples of Fs/2. To reject those aliases, we would like to have filter that preserves the amplitude and phase everything within +/-(Fs/2) and totally reject everything outside that range. The impulse response of this is the sinc function, hence sinc interpolation is the ideal.
The sampled signal in time, an audio signal for example, can be thought as a sequences of impulses that scales the amplitude of the sincs. The ideal DAC has the impulse response of a sinc, with a smooth sinc interpolation between the sample points in time. However, the sinc has the unfortunate properties of not following causality and an infinite rise and settle time. We have to make our peace with causality, and we generally limit ourselves to finite time intervals; the combination of these leaves a somewhat different impulse response. The interpolation is then given by the sum of the impulse responses scaled by the amplitudes of the signal points.
 
Actual DACs tend not to work like an impulse train, they operate closer to a zero-order hold that "holds" the voltage constant in between samples, the much maligned stairstep representation. For the normal DACs that have analog reconstruction filters, that output is immediately filtered into a bandlimited signal that interpolates a smooth time curve. For the sake of analysis, we can still consider the zero-order hold or filtered version as a sequence of pulses. The DAC still has its own impulse and frequency response to consider, the ability of the DAC to accurately interpolate signals can still be evaluated.
 
Dec 23, 2016 at 2:45 AM Post #909 of 1,344
... Actual DACs tend not to work like an impulse train, they operate closer to a zero-order hold that "holds" the voltage constant in between samples, the much maligned stairstep representation. For the normal DACs that have analog reconstruction filters, that output is immediately filtered into a bandlimited signal that interpolates a smooth time curve. For the sake of analysis, we can still consider the zero-order hold or filtered version as a sequence of pulses.

I'm just nit picking here.  but in the topic of R2R vs delta sigma, calling R2R behavior normal DACs, and delta sigma's model becoming the shadow of "for the sake of analysis", doesn't seem representative of the actual DAC world IMO. isn't pulse modulation vastly dominating the DAC market in quantity?
 
 
 
 
 
geek ON:  page 61 was a great page! you guys rock.
 
Dec 23, 2016 at 11:21 AM Post #910 of 1,344
 
Without getting into minutae, as you've noted, real DACs are not following a true SinC function, and do each have their own impulse, phase and frequency response. They are also delivering 'good approximations" in other areas as well. In simplest terms, if you put a complex signal into several DACs, what comes out will not be identical. Which puts us right back to the question of whether those differences will be audible or not...and, if so, which ones.
 
In general, we have adequate studies to agree that, in terms of steady state sine waves, our human frequency response extends from about 20 Hz to about 20 kHz (some studies have shown that humans can hear as low as 10 Hz under certain conditions). We also have lots of studies attempting to determine the minimum difference in amplitude that we can hear - but they tend to vary somewhat in their results. (Most people agree that you can hear a 3 dB jump in loudness; and most agree that you cannot hear a 0.01 dB step; but some set the "cutoff point" at 0.25 dB, while other claims a 1 dB step isn't usually audible.)
 
I personally suspect that even these assumptions have been oversimplified. For example, you probably wouldn't notice a difference in frequency response between two speakers that are both "flat to within 0.5 dB" when listening to music. However, if one has a gradual rise of 1 dB between 200 Hz and 500 Hz, while the other zig-zags up and down a dozen times between those two frequencies, and you play a 100 Hz to 1 kHz sweep through both, I'll bet you won't hear the slow rise but you'll hear a "warble" in the one with the zig-zag response. It would be interesting to do a study to determine exactly how small a modulation amplitude is audible at 3 kHz.... my bet is that it's a tiny fraction of a dB.
 
In the case of DACs, just to pick an example, we all know that differences in phase between the left and right channels are audible as shifts in the apparent position of the sound source; this, along with amplitude differences is what gives us "sound stage". Some studies have suggested that VERY tiny shifts in relative phase produce audible shifts in apparent position. (One study claimed that a frequency response of at least 50 kHz would be required to resolve the phase shifts beyond the point where they're audible.) Perhaps, in the context of DACs, a few degrees of relative phase shift are in fact audible, or, perhaps, a fixed shift isn't audible, but one that varies rapidly can be heard as "a blurring or apparent position".
 
Note that I'm NOT especially claiming that they're right in that specific claim. All I can claim with certainty is that I've listened to many DACs whose frequency response is flat enough, and whose THD is low enough, that, according to "the standard assumptions" I SHOULDN'T be able to hear any difference between them. However, in some cases, there are in fact audible differences.  This suggests to me that either the standard assumptions aren't entirely correct, or that there are other differences that haven't been adequately studied yet.
 
There's a big difference between claiming that "there are no measurable differences - so you must be imagining them" and "well, yes, there are lots of measurable differences, but we're sure that none of them can possibly be audible". And I'm not aware of anyone who has actually done a full-scale study to determine if things like variations in transient response are in fact audible or not. (Wolfson offers a total of 21 different filter choices on their top of the line DAC chip; and they offer oscilloscope images to show the specifics; but, as far as I know, they haven't done a controlled study to prove which of them are audibly different.)
 
Quote:
  To elaborate on a few concepts of sampling, informally:
 
Just as you said, the sampling theorem does not assume sine waves, only a frequency representation(the existence of a Fourier transform). Sampling theorem invokes what is now called the Discrete-time Fourier Transform (DTFT) using discrete samples, impulses, to get a continuous spectrum. This DTFT representation does allow for the representation of sampled sinusoids as discrete spikes, but is not restricted to it.
 
The sampling theorem and sinc interpolation are too closely related to really be separate. Sampling has the pesky property of representing the spectrum from DC to Nyquist and the spectrum flipped and folded over integer multiples of Fs/2. To reject those aliases, we would like to have filter that preserves the amplitude and phase everything within +/-(Fs/2) and totally reject everything outside that range. The impulse response of this is the sinc function, hence sinc interpolation is the ideal.
The sampled signal in time, an audio signal for example, can be thought as a sequences of impulses that scales the amplitude of the sincs. The ideal DAC has the impulse response of a sinc, with a smooth sinc interpolation between the sample points in time. However, the sinc has the unfortunate properties of not following causality and an infinite rise and settle time. We have to make our peace with causality, and we generally limit ourselves to finite time intervals; the combination of these leaves a somewhat different impulse response. The interpolation is then given by the sum of the impulse responses scaled by the amplitudes of the signal points.
 
Actual DACs tend not to work like an impulse train, they operate closer to a zero-order hold that "holds" the voltage constant in between samples, the much maligned stairstep representation. For the normal DACs that have analog reconstruction filters, that output is immediately filtered into a bandlimited signal that interpolates a smooth time curve. For the sake of analysis, we can still consider the zero-order hold or filtered version as a sequence of pulses. The DAC still has its own impulse and frequency response to consider, the ability of the DAC to accurately interpolate signals can still be evaluated.

 
Dec 23, 2016 at 1:26 PM Post #911 of 1,344
Quote:
  I'm just nit picking here.  but in the topic of R2R vs delta sigma, calling R2R behavior normal DACs, and delta sigma's model becoming the shadow of "for the sake of analysis", doesn't seem representative of the actual DAC world IMO.

I phrased that quite poorly. A Delta-Sigma DAC isn't necessarily restricted to a single bit and the output still operates by ZOH, albeit at a much higher rate than signal bandwidth.
 
Sep 13, 2018 at 2:32 AM Post #912 of 1,344
I am considering use of signal purifiing product with my R2R-11 dac (f.e. iFi iPurifier3). Is it a good idea to connect something between Windows PC and USB-B input on R2R dac to alter that 32bit PCM signal flowing via audiocable (Schiit Pyst)?
 
Sep 13, 2018 at 1:20 PM Post #913 of 1,344
It's better to address the specific problem with your source if you have one, rather than try to cobble together some sort of fix for it downstream. What exactly is wrong with your Windows PC's output?
 
Sep 13, 2018 at 3:02 PM Post #914 of 1,344
It's better to address the specific problem with your source if you have one, rather than try to cobble together some sort of fix for it downstream. What exactly is wrong with your Windows PC's output?
Nothing. It sounds superb. I just would like to know, if there is something that should NOT be done specifically to R2R dacs in contrast with DS dacs in terms of signal treatment with sound purifiers.
 
Sep 13, 2018 at 4:32 PM Post #915 of 1,344
First off, assuming that the bits are correct to begin with, you should NOT be altering the data - at all.
If it starts out being correct, and you alter it, then the result must no longer be correct.

In general, computers have relatively noisy power supplies and grounds, and the data they deliver may be transmit on a less than ideal waveform (the square waves may be a bit rounded, or a bit noisy, or not timed perfectly).
And, in general, a well-designed DAC should be designed to avoid being adversely influenced by all these minor flaws.
(In fact, since USB data is delivered in little chunks, the DAC has to re-create all the timing as it receives and decodes the signal anyway).
Therefore, the short answer is that, assuming that the DAC does its job well, then there's no purpose in doing anything to the signal along the way.
Therefore, the purported purpose of the various "signal purifying products" is to compensate for various shortcomings in the DAC.
(And, since nothing is perfect, some DACs do in fact have shortcomings.)

However, in the context of your question.....
If the performance of your DAC is being compromised by imperfections in the signal it's receiving from your computer, then adding a device to filter out some of those imperfections may enable it to work better.
If it's working perfectly to begin with, then there's nothing to improve, and you risk making things worse instead of better.

So the answer really depends on how good the input circuitry on your DAC is, and how bad the output circuitry on your computer is, and there is a LOT or variation on both of those between different models.

I am considering use of signal purifiing product with my R2R-11 dac (f.e. iFi iPurifier3). Is it a good idea to connect something between Windows PC and USB-B input on R2R dac to alter that 32bit PCM signal flowing via audiocable (Schiit Pyst)?
 

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