DSD comparison to PCM.

The ADC was an 18 bit unit purchased from Analogic...

Rene, thanks so much for this. It's the sort of information I've been seeking for ages and ages.

I'm assuming that 24 bit ladder ADCs have simply never existed. Any thoughts?

After the Company was sold, HDCD going to Microsoft, the last M2s were built by Euphonics. I don't know if they had the skill or inclination to make mods to the ADC. The dither ensured at least that the resolution extended to 20 bits, if not the linearity.

Well my particular PM2 is an early unit that was used by PM for in-house demoing purposes, according to Dave Peck. Hopefully it was tweaked as well as possible... maybe by KOJ himself??? Nice thought.

Rene, along with Bud and Lynn, you obviously have a wealth of knowledge about the subject at hand. And it's just great that you're happy to share this with the rest of us. But I'm not getting a sense of your preference for DSD or PCM. Would you care to share?

Mani.
 
Rene, thanks so much for this. It's the sort of information I've been seeking for ages and ages.

I'm assuming that 24 bit ladder ADCs have simply never existed. Any thoughts?

As with DACs so with ADCs; getting even to 20 bits is a heroic effort. At this level of precision one has to craft current sources or resistor ladders with better than 1ppm precision - and this over the temperature range at which the part has to operate. That is why most successful designs were done using either multi-stage or gain-ranging approaches.

Well my particular PM2 is an early unit that was used by PM for in-house demoing purposes, according to Dave Peck. Hopefully it was tweaked as well as possible... maybe by KOJ himself??? Nice thought.

Yes, probably by Keith.

Rene, along with Bud and Lynn, you obviously have a wealth of knowledge about the subject at hand. And it's just great that you're happy to share this with the rest of us. But I'm not getting a sense of your preference for DSD or PCM. Would you care to share?

Mani.

As an engineer who had been involved with delta mod and pcm convertors for almost 20 years, my complaint with DSD from the beginning was: WHY?
As mentioned earlier, I was involved in an early shootout between 30 ips half-track analog, HDCD and DSD at Sony studios. Notables were there, including a famous producer who later went over to the Dark Side (Digital.) All present at that session were pretty much tied between analog and HDCD. Sony wasn't running. When I saw that Sony was persistent, I would venture in to demos at various hifi shows, hoping to hear - what? The same modulation noise I'd heard from the beginning? Maybe that Sony was marketing it for personal players or just because it allowed cheap DACs. Through the best of playback equipment, DSD sounds to me like good analog tape. Perhaps the lack of direct to DSD recordings done at least at 128FS, where it can have technical specs slightly better than RBCD.

I have heard PCM hires done right and I am happy.
 
As you go down in small increments, it's surprising how much crap pops up on the FFT at specific input levels.

Oh yes oh yes oh yes. There is *so* much crud right at the noise level. Funny little harmonics that come and go, you can really see if the analog circuits are getting in trouble or not. The ESS presentation at the 2011 RMAF really left an impression on me: all kinds of awful, low-level things going on with noise-shaping algorithms, like a Class H amplifier gone mad.

I was surprised just how unstable noise-shaping really is, and I kind of wondered if ESS really had chased out all the problems. They said they did, but, well ... it was their presentation, after all, and what are they going to say: "Well, folks, we have it 90% licked, trust us." Nobody is going to ever say that. So, who knows which noise-shaping technique is the best ... and DSD relies on noise-shaping even more than delta-sigma converters.

Along with so many others, I had cheerfully assumed the latest-n-greatest delta-sigma converters were as good as it gets: I certainly didn't expect funky old ladder converters would be better. Then I auditioned some of the highly-reviewed DACs with big price tags, and wondered "what's going on here?", my old Burr-Brown DAC sounds better! That made no sense at all ... until I remembered the ESS talk. Uh-oh, down the rabbit hole again!
 
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Along with so many others, I had cheerfully assumed the latest-n-greatest delta-sigma converters were as good as it gets: I certainly didn't expect funky old ladder converters would be better. Then I auditioned some of the highly-reviewed DACs with big price tags, and wondered "what's going on here?", my old Burr-Brown DAC sounds better! That made no sense at all ... until I remembered the ESS talk. Uh-oh, down the rabbit hole again!

Have you measured or listened to any proprietary FPGA converters?
 
Having tried about all I/V means known to the planet and refusing to ever try OpAmps apart from discrete ones which also did not work out, I ended up with something carrying a slew rate of 670V/us. I know, not 1K, but better for my environment (which is about noise figures, voltages used and some more configurative stuff like differential setup etc.) I could not find. And you know what ? it sounded better. Way better.

My reasoning back at the time was that the high frequencies were eating the current, especially when seeing that not the bass improved but the highs. This is sort of related to the images behind the mirror which are there in my NOS/Arc Prediction Filtered case and although they are sufficiently down, they eat energy and by a higher frequency than normal audio band frequencies, obviously.

When you dig deeper in my explanation, you can see that it is all over unofficial "reasoning out" what is for the better and what won't work. I mean, let pass higher frequencies beyond the audio band less distorted (see slew rate) on purpose ? Well, if we don't do that then the in-band higher frequencies become muffled. A nice natural sort of filter ? no way. No way since those "un-muffled" frequencies come out the most undistorted. And so the fun : instead of seeing a roll off in the higher frequencies dedicated to NOS/Filterless, we perceive a high frequency output (for level) which is unsurpassed and of which at least I dare claim that it is unsurpassed indeed.

And let's say (and hope) that by now I should have sufficient experience in knowing the difference between "freshness" coming from unfiltered NOS hence a mountain of piled up harmonic distortion on one side, and super silk and undistorted highs with a perceived maybe over 20dB more output than any normally filtered DAC on the other. Ever heard sheer SPL from brushes on a snare ? 5 years back the whole d*mn brushes weren't even auadible in my system. And yes, brushes may be the toughest "instruments" to render well.

Peter

Thanks for sharing your experience, Peter. I've been very curious about the Phasure DAC for some time, and I appreciate your experience and insight. That quality of "freshness" is exactly what was missing from the expensive delta-sigma DACs I was auditioning ... and is also absent from many DSD converters, as well. It's hard to describe, but when the "freshness" is missing, the music has a sort of monotone quality, and the performers sound bored.

I'm guessing the noise-shaping algorithms are doing strange things to the dynamics ... it certainly sounds like the dynamics are altered. With one visiting DAC, the dynamics were so different that I could never get it subjectively level-matched with my Burr-Brown DAC. I'd match 'em up on a quiet passage, then they would sound totally different when it got loud.

Regarding slewing ... well, that's another down-the-rabbit-hole topic all by itself. Slewing is a function of current distortion, not voltage distortion. When an internal amplifier drives a capacitive load, and starts to lose linearity, it doesn't happen all at once. There can be a large region between the onset of slewing (gentle distortion) and hard slewing. Depending on the topology and the devices used, this region can be fairly small (1:1.5) or pretty big, like (1:10). It can be the difference between hard, gritty, 2-dimensional sound when the HF content gets loud, or a soft, syrupy sound that's dull and velvety sounding. Slewing can sound like either, depending on how the device reacts to driving current into a capacitance.

You might have noticed that's what many people describe as the difference between PCM and DSD. Uh, yeah. Which is why I don't think PCM vs DSD comparisons have much validity until we know more about how the analog circuits are responding to 1~20 MHz content. Typical audio circuits (with feedback) that have dominant poles anywhere from 10 Hz to 1 kHz get into real trouble at RF frequencies; feedback is mostly gone, the output stage is not happy driving the load of the feedback circuit (as well as external loads), and opamps typically have Class AB output stages as well as problems with thermal tails. (Transient gain shifts due to temperature changes on the die which occur in the 10 mSec range.)

We have two problem areas: the hard-to-measure things noise-shaping does to the signal, and the way the analog circuits respond to significant amounts of RF content. DSD converters have very substantial amounts of noise-shaping: by my rough calculations, 60 dB or more in the 1~5 kHz region.

But the 1~5 MHz decorrelated noise spectrum of DSD may be kinder to hard-slewing analog circuits than the spiky images of PCM. So right there we have two major sources of sonic difference between the two systems - in fact, we have two flavors of PCM: delta-sigma with noise-shaping, and ladder converters without noise-shaping.
 
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I was surprised just how unstable noise-shaping really is, and I kind of wondered if ESS really had chased out all the problems. They said they did, but, well ... it was their presentation, after all, and what are they going to say: "Well, folks, we have it 90% licked, trust us." Nobody is going to ever say that. So, who knows which noise-shaping technique is the best ... and DSD relies on noise-shaping even more than delta-sigma converters.

The ESS presentation was interesting, not least for what it left out. They talked only about noise shaping in the modulator, they omitted any talk (at least from my recollection) about noise shaping in the resistor array itself. That's another rabbit hole to fall into, how the errors in resistor values (which are at best around 0.1%) get turned into 'benign noise' via stuff like 'data weighted averaging'. Philips solved this in their designs with DEM - 'dynamic element matching' which actually did adjust the bit weights dynamically. As that's an analog adjustment its been discarded in S-D DACs in favour of handling it digitally with noise-shaping of the errors. There's no feedback involved though as far as I'm aware so perhaps these techniques really are benign?
 
Isn't the problem though Lynn is that both have trade offs, going NOS ends up with more complex and noticable pattern from the perspective of jitter spectra analysis and noise-distortion related to signal dBFS, compounded by bit resolution from those I have seen measured so far is around 16bit to 17bit at best.
I was keen on the idea of NOS with native high rez releases (so 88.2khz and above) as in theory I thought it would be ideal, however it still seems to suffer from much of the above comments when measured.
But against this is the traits associated with delta-sigma oversampling/filters-algorithms; lack of accuracy in both time and frequency domain, ringing-ripples consideration.
Did you ever listen to Chord Electronics DAC64, or thoughts on dCS (that to my mind is even more of a hybrid than ESS) and their sound?
Regarding the highly reviewed DAcs with big prices, anything else same about them such as DAC chipset-manufacturer used?
Appreciate you probably do not want to mention names of products and so refraining from asking that.
Cheers
Orb
 
Have you measured or listened to any proprietary FPGA converters?

No, I haven't. I'm curious about the technology: I assume deglitching is non-trivial, and requires a well-designed sample-and-hold following the converter.
 
The ESS presentation was interesting, not least for what it left out. They talked only about noise shaping in the modulator, they omitted any talk (at least from my recollection) about noise shaping in the resistor array itself. That's another rabbit hole to fall into, how the errors in resistor values (which are at best around 0.1%) get turned into 'benign noise' via stuff like 'data weighted averaging'. Philips solved this in their designs with DEM - 'dynamic element matching' which actually did adjust the bit weights dynamically. As that's an analog adjustment its been discarded in S-D DACs in favour of handling it digitally with noise-shaping of the errors. There's no feedback involved though as far as I'm aware so perhaps these techniques really are benign?

Well, the devil's in the details. I wouldn't assume any of these averaging or signal-processing techniques are sonically invisible. For one thing, a lot of vendors cheat on the spectrum analyzer display, using really long averaging times (1 to 30 sec) to make things look good. But the ear doesn't average over 1 second; a more realistic interval might be 10 to 100 mSec, and the averaging doesn't have as much time to work its miracles on the signal.

In fact, this is a deeper problem with dithering and its close cousin, noise-shaping. How heavily do they rely on averaging to make good the S/N and distortion specs? Perhaps more importantly, in a single Red Book sample of 22.7 uSec, what is the real resolution?

This is where flash converters look very different than delta-sigma or noise-shaped DSD. The flash converter does its thing, bang, and there's the output. Delta-sigma and DSD do a lot of signal processing to a low-bit converter to synthesize, through successive approximation, the intended analog value.
 
Have you measured or listened to any proprietary FPGA converters?

Added to this is what I was hinting about architecture, this can also include the oversampling being handled separately by a more powerful DSP and software-algorithms, which can be used with FPGA based converters or more standard delta-sigma DAC.
Appreciate some FPGA have enough capability that some manufacturers would include all functionality within said FPGA processor such as Chord Electronics (used to anyway I think).

Cheers
Orb
 
For one thing, a lot of vendors cheat on the spectrum analyzer display, using really long averaging times (1 to 30 sec) to make things look good.

To show this, here's a "same" picture of that noise line from my post yesterday (#416)

noiseline01.png


Hey, now that looks better eh ? This is averaged.
But compare with that other picture and see that this shows the noise at -160dB. How come ? deeper FFT. But at least I always mention it when it is unrealistic for comparison otherwise. Well, like I did in that post (256K) and like I should say in this post that this is averaged (forgot how much, but think dozens of seconds). Btw, this is an 8K FFT and "normal".
 
That quality of "freshness" is exactly what was missing from the expensive delta-sigma DACs I was auditioning ... and is also absent from many DSD converters, as well. It's hard to describe, but when the "freshness" is missing, the music has a sort of monotone quality, and the performers sound bored.

It's funny, this description of SDM/DSD sound is exactly how I hear things too. Was it Opus111 who described the SDM/DSD sound as "elevator music"? Another great description. Pleasant enough, but hardly emotionally stirring.

I have heard PCM hires done right and I am happy.

+1

Mani.
 
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Hey, now that looks better eh ? This is averaged.

As with DACs so with ADCs; getting even to 20 bits is a heroic effort. At this level of precision one has to craft current sources or resistor ladders with better than 1ppm precision - and this over the temperature range at which the part has to operate. That is why most successful designs were done using either multi-stage or gain-ranging approaches.

Hey Peter, your graphs look substantially better than 20 bits resolution to me... not from just 20Hz-20KHz, but from 20Hz-90KHz! Could we call the NOS1 a true 24 bit DAC? And this is achieved with no noise-shaping whatsoever??? Is this really possible?

Mani.
 
Hi Mani - No. All what this noise line shows is that the ~-142dB you see here should be capable of resolving close to 24 bits (which needs a noise lower than 144.48dB).

Assuming that you refer to PCM with the "noise shaping" phrase, it doesn't really need that because no noise is to be shaped away anyway. But then of course this is about the filtering applied which works on the "brickwall" and without any "shaping" the stepping is now small enough to have residuals far out of the audio band anyway. This is about 705.6KHz applied to 16/44.1 material and now ...

Well, now I actually don't understand much of DSD to begin with;
For me and at this moment, all is too much unclear for starters, but try to see yourself in my situation where no further filter is applied, this implies stepping of, say, 768KHz which is a quarter of DSD (2.8MHz) and where my stepping is LARGER than that of DSD (think horizontally for the time domain and vertically for the frequency domain where the stepping happens just the same (level changes) ... and where thus my LARGER stepping which is about squares do not need a thing ?
So, how would DSD need something ? In my so far theoretical view it can only need less.
But then my theories are always different because the base is different (all over). Look here for another example, but with the notice I should supply a picture with it, which I don't have at hand right now :

When I play (through the NOS1 that is) an e.g. 1KHz close to full scale signal and which is, mind you, 16 bits (44.1) and attenuate this with 141dBFS, this nicely sticks out of the field. Of course this is Arc Prediction upsampled (to 705.6KHz) but in the base it is and remains 16 bits. And you know what ? no dither used. Nothing.
I won't go into the details of dither and what it can bring (well, more bits of resolution that is), but as you may know I always stayed away from dither because a first thing it will do is take out bits (at least one). But as you also know I'm a sort of 100% noise guy and when we can't get rid of it we should utilize it. Now, since no real random dither exist while analog noise shoud be rather random (that too never is, but say it is more random than what we can create by means of digital randomizers), this is ... a perfect dither. At least in my thinking it is, but it also shows easily on the analyser.

I don't say the NOS1 is resolving 24 bits perfectly and with the notice of doing this linearly is quite another matter. But let's at least keep in mind that the 1Khz can be resolved up to 23/24 bits while it is not even a 24 bit signal. This should tell something about "possibilities".
Btw, stupidly enough I never even tested this with a 24 bit signal, most probably because I am not into Hires at all. Say I don't care. But from extrapolation should follow that when first that FFT looks like in that earlier post (noise 20dB down because the deeper FFT) any 24 bit signal will easily show when e.g. 155dBFS attenuated. One problem : XXHighEnd doesn't allow more than 144dBFS of attenuation (so it would be a lab application).

Then, it is doubtful to what degree we would ever be able to perceive such a deep resolution, because "our dynamic range" won't be more than some 70dB anyway. And as I have said it before, apply a signal which is 96dB attenuated (normal RBCD dynamic range), turn up the volume of your amp so you can just hear the faint music, and next undo the digital attenuation. I don't know what will break, but if nothing like windows it will be your ears. And this is not about 96dBSPL which will be there suddenly, because the "bottom" of inherent noise (molecule etc.) is to be added first (say 30dB, I don't know the exact figure). So, something like 130dBSPL will be the result.
And this is only Redbook.

Still still still, it will do something to my firm believe, because all is about modulation. If not in mid air already, then at least in the D/A domain where any (!) higher frequency modulates on any (!) lower one. And so many frequencies around ...
And here it will be that DSD is different. Well, in my view; I don't think we can speak of any D/A where the higher frequency modulates upon the lower frequency and this is merely because no D/A as such is in order. However, now there is noise modulated on to the actual signal just because some lunatic invented computers and designed them around 1's and 0's. If only it would have been 1, 0 an -1. Then no "modulator" noise would be there as an unavoidable con.

Ok, I'll stop here because I see I run into too much explanation needed. See it as a teaser. But for fun, see next post.
Peter
 
Peter sorry if I am missing something but what does the chart show with 1khz signal at 0dBFS and -10dBFS, possibly even -30dBFS to see trend?
Also interesting to do the same at 10khz and 17khz (this is the one that really lowers the figure, not as much as intermodulation test though).

Cheers
Orb
 
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Fun, or ?

Since indeed I am trying to make something from DSD with the most ignorant base and which is about avoiding any noise shaping - at least for as long as I think it can be possible and with tricks in mind which are above my own comprehension ... I may have been digging more into the why's and how's of DSD (SACD) than anyone else. This is partly based on me not understanding most of if which by itself relates to lacking some nowlegde. Lacking or lacked - not sure yet. Anyway :

I found papers from Philips (not officially available on the Internet) where the whole SACD thing was layed out in a way that I would not have dared to do in primary school where I learned about 0-'s and 1's. The whole document talks only and only about +1 and -1 and from there tells what DSD/SACD is. Mind you, this was 2002 and SACD was on the market already.

A next nice observation (but is subjective because through my eyes) is that that fameous Vanderkooy and Lipshitz (telling that DSD can't be any good) out of all where the authoring guys of that +1/-1 paper I just talked about. To me this looks like that they first sort of invented DSD to next see it coud not work.

Then there is this 2004 paper (also not available) and this explains in more decent fashion how DSD works, but what does it say in there ?
"At this moment it is not quite understood why the modulator runs out of control, can overload, saturate or start self-oscillating".
Mind you, words similar to this (but no phenomenon made up) as I do this by heart now.
So, 2004 and I don't know how many SACDs around at that time, but even Philips herself doesn't know how the damn thing works. No wonder maybe when all starts out with -1 and +1 to explain how the modulator works. And to this regard, notice that +1 plus -1 equals zero which would be 100% convenient to someone like me, knowing that any "silent" stream consists of +1' and 0's which digitally added is not 0 at all. However, electrically it will be with a small DC offset to the plus side.

Then there are all the pictures all over the world which show how the pulses of 1's and 0's result in a sine. Fine. But no single one seems to tell the truth to me because I can't see how it could ever work *without* corrections in computer math which anyone can do to his liking. This includes the staying away from overloading and sadly ALL is mushed with noise shaping which will take out the errors coming from the modulator's result (and which is the DSD stream input to e.g. my software).
... and where I like to start out without noise shaping which seems to bring the conclusion that no DSD stream is correct.

Done ? Almost.
Where I find myself having difficulties with examining the real merits of such streams and which merely always lead to DC offset but with the notice that I really don't know how to cough up a "DC Zero" result because possibly subjective to unknown math (at least to me), I next run into DAW users complaining about their sort of consumer Korg DSD recorders that they exhibit DC Offset. Oh. These users complain and here and there get their Korgs replaced but the problem remains. Never anywhere I see this sorted out except for the advices to remove DC which is a most common task to a DAW engineer, that is, were it about PCM (and electrical environmental behavior which implies DC easily). Notice that any common DSP software tool will contain DC removal.
But with DSD the main key little thingy is that when all 0's and 1's (for electrical result !) are added, the sum has to be zero. This is not only logical but also a necessity because otherwise we do have DC Offset in our gear, and that's not a good thing (up to burned speaker driver coils).

Parallel to the above there's the -also from Korg- PCM to DSD free conversion program, Audiogate. Nice for testing a few things. Here too I couldn't find my way because that too lead to strange results but in different fashion (with always in my mind that I didn't know the heck what I was doing). Until ... until I found reports about that conversion program lead to DC offset. Here too all is diminished by the good guys telling about a checkbox in there, which when ticked will remove DC offset. Aha, that helps.
Yea well, but still there should be no DC offset anywhere to begin with.

And because all people are satisfied with the backdoor solutions of which they think it is a front door, nothing ever happened about it that I could find (and this goes back to 2007).

In the mean time, all I could do to have a decent stream with to my eyes valid data, is averaging the stream(-data) like a noise shaper tends to do. Then I just don't see the offset error because it can't happen. Now of course it still is so that I don't know how to do it anyway, but combined with my little story above I feel more safe by thinking that nobody understood one bit of it, while still music can come from it.


Maybe Bruce can shed a light on DC offset, found in DSD in particular ? In different A/D converters ?
 

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