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The Multibit DSD debate


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As more and more DSD-capable DACs become available ( see this wonderful database that Jesus from Sonore started, and I help co-ordinate......https://docs.google.com/spreadsheet/ccc?key=0AgVhKcl_3lHfdFVyenBBNjNpQ2lieG81WGpqQTNfVUE#gid=0) the debate rages on about what is really DSD and what is simply converted PCM (behind the scenes). Or more appropriately, does a DAC's architecture and chipset (or lack thereof) allow it to process DSD and/or PCM more effectively than others...i.e better sound? Because, frankly, who cares how it gets to the speaker, as long as its beautiful musically, right?

 

Well, as we all try to educate ourselves on hirez music, and find that the suppliers of said music want a premium for this stuff, we are more and more interested in making sure our DAC purchases are going to play back our favorite music properly and conveniently. We DO care how it arrives at our speakers; we want to be future-proof to some extent, yet are confused by all the options. One-bit, mutlibit SDM, multbit PCM, chipless, R2R, Ring DAC, FPGA, etc.

 

Well...enter a very well regarded DAC player, Berkeley Digital. They have a new $14k DAC that, according to their news release (which rumor says was not written by the principals, let alone Dr Keith Johnson, but possibly by a dealer of theirs) clearly, refreshingly (and likley erroneosuly) states it, like 99% of all DSD-capable DACs out their, is multibit PCM and does not process DSD directly, but instead converts to PCM...so they want the user to convert DSD to PCM outside the DAC, prior to its entry, to save the noise and heavy lifting from happening near the DAC chip.

 

OK about the heavy lifting, but it's the first part of that (99% of DSD-capable DACs are multibit PCM) that I and others question. So much so that I wanted to start a thread about this debate rather than hijack the Berkeley annoucment thread. I'd like other DSD-capable DAC mfhgers or knowledgeable folks to "dummy down" the tech talk so that many of us can understand. :) We have several threads here about math; this one needs to be about differences among DSD-capable DACs that potential purchasers can understand. I started with Michal Jurewicz, from Mytek. His comments are below.

 

First though, Here is the text from the Berkeley release comments:

 

"Careful consideration was given to providing the highest possible reproduction of DSD files by the Alpha DAC Reference Series. 99% of modern DAC’s, including the Alpha Reference Series use multi-bit D/A converters because they provide better performance than 1-bit converters – even those who advertise “native” DSD compatibility. So, at some point, the 1-bit DSD stream must be converted to multi-bit for all of those DAC’s.

 

We could, like many other manufacturers, convert 1-bit DSD to multi-bit within the Alpha DAC Reference Series and show “DSD” in the front panel display. That would be the easiest approach from a marketing perspective. But that would also mean increasing the amount of processing in the DAC during playback which would degrade audio quality, and audio quality is the reason the Alpha Reference Series exists.

 

Fortunately, virtually all reproduction of DSD files using external DAC’s occurs with a computer based music server as the source. If the 1-bit DSD to multi-bit conversion is done first in the computer it can be performed with extremely high precision and superior filtering that preserves all of the content of the DSD file. Computer DSD to multi-bit conversion can be at least as good as that performed in a DAC and without adding processing noise near or in the D/A converter chip. Another advantage of computer based DSD to PCM conversion is that if higher performance DSD versions such as DSD 4x appear in the future they can easily be supported with a software upgrade.

 

For all of those reasons, DSD capability for the Alpha DAC Reference Series is provided by an included state of the art software application that provides either real time conversion of DSD 1x and DSD 2x to 176.4 kHz 24 bit PCM during playback or conversion to 176.4 kHz 24 bit AIFF or WAV files. The software application is included in the price of the Alpha DAC Reference Series and is compatible with either Windows OS or Mac OS based music servers."

 

So, Michal Jurewicz, Mytek founder and chief designer, responded:

 

That quote (above) cannot be considered accurate.

 

1) Multibit DSD is not PCM.

 

PCM is typically 24 bit of the whole sample value while multi bit DSD is typically 5-6 bits of DIFFERENCE between adjacent samples. In 1 bit DSD this difference is binary(0 or 1) while in multi bit DSD it's the same difference between samples but quantized with 5-6 bits (Sabre has a 6 bit DAC). Multibit DSD is BETTER than 1 bit DSD, so there is nothing wrong with going 1 bit> 6bit which is what all these DAC chips do.

 

2) The " purity of 1 bit " is a marketing spin resulting from how DSD was marketed by Sony, but it's not 1 bit that makes it sound good, but it's nature. What makes DSD sound good is the conversion method (differential btwn samples- not the actual sample) and no digital filters , it's simply cleaner. 1 bit DAC conversion has the appeal of simplicity (perfect linearity theoretically) in the early 2000s. We are now well past that with performance requirement. We need more bits to resolve detail better.

 

3) All PCM conversion today (99% of ADCs) is a derivative of multi bit DSD.

PCM is always a subset of that, so naturally it can only have less information, not more.

 

In our latest experiments PCM has to have at least 32 bit to compete with DSD low level resolution and 352kFS to compete with lightness of DSD.

 

In the experiments with our newest prototype ADC, when we reduce wordlength of ADC from 32 to 24 bit, signal deteriorates.

 

Hope this clarifies things somewhat.

 

 

As I thought, it seems we have at least four categories of managing DSD within the DSD-capable DACs. People, please correct me.

1) one-bit architecture like EMM, Meitner and Playback Designs. The downside is that they upsample (or convert) PCM to a DSD multiple before it hits the analog stage. Michal's comments about one-bit DACs are, of course, his. :)

2) multibit-DSD like Mytek and other ESS SABRE designs.

3) multibit PCM like Berkeley. They convert everything to PCM before the chip..

4) chipless analog-filtered-only designs like Lampizator. They are DSD-only (for this process). In this case maybe we do the converse of the Berkeley recommendation; we ask a player like JRIver to convert all PCM to some DSD multiple. I can't speak for the Lampi PCM side, but it must require a traffic cop or separate USB signal path?

 

I'm not sure where the esoteric "logic" designed ring DAC (dCS), R2R (MSB, TotalDAC) and FPGA (Chord) designs fit in here.

 

I'd love for this thread to help folks understand the clear design choice differences, and choose wisely among them. I am not asking that Berkeley be chastised; far from it. Someone in their network took the marketing definitions a little too broadly (or so it seems). However, if they can process a converted DSD-to-PCM piece of music and sound great, so be it!! Me, I've yet to hear that category make DSD sound as good as categories 1 and 2...but there's a boatload of incredible and mandatory PCM out there that we all own, so the choices are not black and white.

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I think this topic has been discussed to death on other threads.

 

But none of the above (removed) talked about how the conversion process actually works... Important thing to understand is that in SDM converter there are various stages with different number of "bits" and the meaning of "bit" changes on each stage. For that reason in SDM context we usually talk about number of levels instead of bits. Especially because in most cases it is convenient to have odd number of levels and that's not even close to any power of two. So a typical number of output levels could be 5 which is around 2.5-bits in terms of power-of-two binary number.

 

As I thought, it seems we have at least four categories of managing DSD within the DSD-capable DACs. People, please correct me.

1) one-bit architecture like EMM, Meitner and Playback Designs. The downside is that they upsample (or convert) PCM to a DSD multiple before it hits the analog stage. Michal's comments about one-bit DACs are, of course, his. :)

 

I think these are not "1-bit" in the sense of "let's filter the bitstream"... But more similar to the dCS.

 

2) multibit-DSD like Mytek and other ESS SABRE designs.

 

When these can take DSD input, it usually gives shorter path to the conversion stages than when same chip is given PCM input.

 

3) multibit PCM like Berkeley. They convert everything to PCM before the chip..

 

My question was if it really is, if it uses same AD1955 as the earlier models, then it is just normal delta-sigma converter and will convert the PCM input to "multi-bit DSD" inside the DAC chip. Just like "99%" of all current DACs.

 

4) chipless analog-filtered-only designs like Lampizator. They are DSD-only (for this process).

 

These can be also "multi-bit"...

One example of such analog-filtered-only are the ones using TI chips, like TEAC UD-501.

 

I'm not sure where the esoteric "logic" designed ring DAC (dCS), R2R (MSB, TotalDAC) and FPGA (Chord) designs fit in here.

 

R2R is the "only way" to have a true PCM DAC with something like more than 8 bits (in practice).

 

dCS RingDAC is 25-level delta-sigma. And I guess Chord's implementation is actually pretty much "1-bit" implementation. You cannot implement actual conversion stage inside an FPGA (since it's a digital component), so it is always more or less visible outside on the PCB.

 

I'd love for this thread to help folks understand the clear design choice differences, and choose wisely among them.

 

It is not easy to split things nicely into two silos, because as usual in life, there's so much grey area between the extremes. And especially SDM gives a lot of freedom for designer to come up with various different architectures for the converter. What I've been measuring wideband output from different DAC chips it is clearly visible that there are huge differences in implementation.

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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Miska, thanks. Some of those answers are very helpful. :) And YES, I will fully agree that the MATH has been calculated back and forth to death on other threads. Those of us without the developers hat need some help, though.

 

Especially because in most cases it is convenient to have odd number of levels and that's not even close to any power of two. So a typical number of output levels could be 5 which is around 2.5-bits in terms of power-of-two binary number.

Seriously, this just doesn't help me buy a DAC, I'm sorry. Maybe I'm dense, but otherwise I wouldn't have started this thread.

 

Two silos? Hardly. I have four or five so far..... ? Maybe we need more...

 

And when you say (about Berkeley, mutltbit-PCM category)

 

My question was if it really is, if it uses same AD1955 as the earlier models, then it is just normal delta-sigma converter and will convert the PCM input to "multi-bit DSD" inside the DAC chip. Just like "99%" of all current DACs.

 

then why would one want to convert DSD to PCM prior to this chip, as Berkeley asks, if it goes through another conversion to multibit-DSD anyway??

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Seriously, this just doesn't help me buy a DAC, I'm sorry.

 

It doesn't, I agree and there's no simple answer between the tech and trying to make educated decision without actually taking each individual design apart and discussing all the tech involved in great detail.

 

I could draw analogy to the new LED lamps with multiple small leds inside to make one lamp. Now think each bit controlling a switch to each individual LED inside a lamp. If you turn on all switches (all small LEDs) the lamp is bright, if you turn only half of those, it's half as bright and if you turn on only one, it's pretty dim. Each of those bits/switches is an independent "1-bit DAC". If you have 10 LEDs you have a 11-level DAC (one extra level is all-off). With such lamp, you can find number combinations of 5 leds that all give about the same apparent half-brightness.

 

ESS Sabre has 64 of such "LEDs" inside.

 

This is vastly different from how PCM works.

 

then why would one want to convert DSD to PCM prior to this chip, as Berkeley asks, if it goes through another conversion to multibit-DSD anyway??

 

That's a good question. By converting to PCM prior to the chip you practically replicate the process that is done inside the ADC chip when recording to PCM with any modern ADC...

 

That conversion process done inside computer has potential to be technically better than when done inside ADC chip. But in any case you have extra round of unnecessary back and forth conversions DSD was designed to avoid...

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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In our latest experiments PCM has to have at least 32 bit to compete with DSD low level resolution and 352kFS to compete with lightness of DSD.

Marketing. No ADC has real analog resolution better than 19-20 bit. To achieve full 20 bit resolution, THD+N figures shall be at -120dB.

The claim that they can 'hear' difference between 24 and 32 bit ADC conversion is just bogus meant to hype the sales of 'DSD capable' toys. They realized that people will keep the good-ol PCM-only DACs forever if they don't push for a different format that will make those PCM-only DACs 'morally obsolete'.

 

The switch from 1 bit DS to multibit DS was done exactly because 1 bit didn't work right, it had obvious problems, highlighted in several whitepaper publications (as opposed to marketing materials like the ones above). The claim that the conversion from 1 bit DSD to multibit DS requires a lot of processing power from the DAC chip is hogwash. There is no processing done in our conventional way of thinking, everything is hardwired.

 

Until a native format multibit DS will be available, all this DSD fad is just a marketing smart idea.

 

As much as I hate the ABX bunch of people, they might be on something when they say most of our perceptions are influenced by our wishful thinking.

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The switch from 1 bit DS to multibit DS was done exactly because 1 bit didn't work right, it had obvious problems, highlighted in several whitepaper publications (as opposed to marketing materials like the ones above).

 

...and DSD is 1-bit DS transport format, it has nothing to do with how many bits are used to convert it to analog... :)

 

PCM has the ugly limitation that you are much more bounded between transport and conversion. (you would have more freedom and less wasted bandwidth, by using delta-encoded PCM where you only transfer difference between two adjacent samples)

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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Miska,

Thank you for that. In rereading my response I have to apologize for what sounded like ungratefulness. It is not, believe me. I'm trying to get educated in the technical aspects, too, but you developer guys are wayyyyyy ahead of most of us who simply (I know, it was never implied that it was simple) want to make educated purchases. This hirez stuff often sounds great (often does not) but ALWAYS costs us more than our redbook or even vinyl options. To compound that by buying a DAC that we think is format agnostic but instead is format specific..well, that education is very valuable, especially if it's not the format we bought it for!

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OK, I am going to moderate this thread like I used to moderate my sales force's strategy meetings, by sitting in the back and saying "who cares" when a feature or fact is presented with no benefit! No offense Sonic67 but what is it about transporting the dither that I should care about?? We know you guys know LOTS technically; please educate.

 

(Note: if you don't want to educate the average CA DAC buyers then the "to DSD or not to DSD" thread is always available for those with slide rules. :) ) Joking..kind of.

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Ted, here is what MSB Technology is doing with DSD,

"From there the DSD 1bit 2.8 MHz (DSD64x) and DSD 1bit 5.6 MHz (DSD128x) is mapped to our 1.4 MHz 24bit Ladder DAC with no analog filtering. It is as pure a playback as you can get as we do not suffer from all the Delta Sigma artifacts."

 

You may want to email Playback Designs to see if they use a one-bit architecture or not.

 

Please consider my PureDSD DAC for the chipless category. It's not a production unit as of yet, but I just want to be included in the category:)

 

Also, for the DSD database I guess we could add column?

 

Any issues if I make some comments about the use of the term "native DSD"?

 

Jesus R

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Adding a column is a good idea; and it would help us get some confirmation from mfg'ers. Of course, the columns would need some explanation (i.e facts or features without spelling out the benefits :) )

 

The PureDSD DAC sounds very interesting. And I think "native" DSD is as political as "hirez" or "jitter-free". :) I try to only use it when speaking of a DSD recording that has been mastered without any PCM EQ (i.e I consider Jared Sacks doing nothing but edits in DXD but recording and finalizing in DSD to be native). But am open to scrutiny.

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I guess for the average person it's not important that the database denote this. I was going to comment about the other end of the playback chain. It's not "native DSD" playback IMO if the gear is converting to PCM, up sampling, down sampling, over sampling, and the like. I'm also not judging the designs either way...

 

Jesus R

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Ted-

 

I appreciate this thread, but why not talk specifically about the DSD DACs that most of us own: Mytek, Teac, Benchmark, Resonessence, exaSound etc?

 

So I own a Mytek. My understanding from Michal is that the DAC takes a DSD stream from my PC and "converts" it to multi-bit DSD - not PCM - before conversion to analogue. Is this correct?

 

Is this type of processing true/native DSD or not? My understanding is yes. I guess others will look at it differently.

 

I'd suggest we try to post simple summations like this one about what each "DSD" capable DAC does. Otherwise it won't be helpful to many people.

Main listening (small home office):

Main setup: Surge protector +_iFi  AC iPurifiers >Isol-8 Mini sub Axis Power Conditioning+Isolation>QuietPC Low Noise Server>Roon (Audiolense DRC)>Stack Audio Link II>Kii Control>Kii Three >GIK Room Treatments.

Secondary Path: Server with Audiolense RC>RPi4 or analog>Cayin iDAC6 MKII (tube mode) (XLR)>Kii Three .

Bedroom: SBTouch to Cambridge Soundworks Desktop Setup.
Living Room/Kitchen: Ropieee (RPi3b+ with touchscreen) + Schiit Modi3E to a pair of Morel Hogtalare. 

All absolute statements about audio are false :)

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"From there the DSD 1bit 2.8 MHz (DSD64x) and DSD 1bit 5.6 MHz (DSD128x) is mapped to our 1.4 MHz 24bit Ladder DAC with no analog filtering. It is as pure a playback as you can get as we do not suffer from all the Delta Sigma artifacts."

 

I really hope they have at least some digital filtering, otherwise it'll push really scary amounts of HF noise out... Without analog filtering it will anyway have really high level replicas of the output every multiple of the 1.4 MHz sampling rate...

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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I appreciate this thread, but why not talk specifically about the DSD DACs that most of us own: Mytek, Teac, Benchmark, Resonessence, exaSound etc?

 

All those except Teac use ESS Sabre so they behave the same way...

 

Sabre works similar way as DSD processor mode in CS4398 (non-DirectDSD). So there is digital processing for the DSD stream (filter, volume and remodulator). This diagram of CS4398 illustrates it quite well:

4398blkdiag_mag.gif

(note on CS4398 there's a configuration option to choose "DSD Processor" path or "Direct DSD" path, controlling those "MUX" switches)

Output of both ESS and Cirrus DSD processing engines is still at the native sampling rate (2.8/5.6 MHz etc) but multi-bit SDM.

 

Teac (with TI chip) has four different configuration options for it's conversion stage to form analog filters and AFAIK no other processing.

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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Until a native format multibit DS will be available, all this DSD fad is just a marketing smart idea.

.

 

I really think you could do with boning up on the latest 1-bit SDM research:

 

Here is an interesting doctoral thesis defended in 2010 at the University of Technology at Eindhoven by Erwin Janssen and now readable online. Janssen and his colleague Reefman has published extensively in the past on 1 bit audio research for Philips. The most interesting chapter is the last where he demonstrates 190dB SNR with 1 bit audio which is nearly equivalent to 32 bit PCM resolution. The theoretical maximum, evidently, given enough computing power and ideal conditions, is 425dB which is equivalent to 70bit PCM resolution.

 

"... Although with this approach a world-record SNR for a 1-bit noise-shaped signal has been achieved, it is still far away from the limits imposed by information theory. As such, in practice the SNR is only limited by the amount of available computational power that is required to stabilize the higher order filters"

 

Look-ahead sigma-delta modulation and its application to super audio CD (Erwin Janssen, Look-ahead sigma-delta modulation and its application to super audio CD, Ph.D. Thesis, Technische Universiteit Eindhoven, 2010)

 

http://alexandria.tue.nl/extra2/691188.pdf (Full doctoral thesis)

Music Interests: http://www.onebitaudio.com

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Ted-

 

I appreciate this thread, but why not talk specifically about the DSD DACs that most of us own: Mytek, Teac, Benchmark, Resonessence, exaSound etc?

 

So I own a Mytek. My understanding from Michal is that the DAC takes a DSD stream from my PC and "converts" it to multi-bit DSD - not PCM - before conversion to analogue. Is this correct?

 

Is this type of processing true/native DSD or not?

 

? Dog, that is what this thread is about....and MUCH more specifically I had Michal from Mytek respond to exactly what you are asking, in post number one. ? I'm confused by your take. What abut Michal's explanation did you not like?

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? Dog, that is what this thread is about....and MUCH more specifically I had Michal from Mytek respond to exactly what you are asking, in post number one. ? I'm confused by your take. What abut Michal's explanation did you not like?

 

Ted -

 

You're confused b/c you are giving me too much credit for intelligence. I apparently did understand Michal's answer, but I wasn't sure. People like me need it laid out in the simplest way possible. Miska helped by saying all ESS based DACs do it the same way, and by giving us the flow chart.

 

The only thing that's left that confuses me is why some people insist on saying a DAC that changes single bit to multiple bit DSD is somehow not a real DSD DAC and that the manufacturers are somehow ripping us off. It seems pretty clear that changing the signal to multibit isn't going to harm the sound we get in the end.

Main listening (small home office):

Main setup: Surge protector +_iFi  AC iPurifiers >Isol-8 Mini sub Axis Power Conditioning+Isolation>QuietPC Low Noise Server>Roon (Audiolense DRC)>Stack Audio Link II>Kii Control>Kii Three >GIK Room Treatments.

Secondary Path: Server with Audiolense RC>RPi4 or analog>Cayin iDAC6 MKII (tube mode) (XLR)>Kii Three .

Bedroom: SBTouch to Cambridge Soundworks Desktop Setup.
Living Room/Kitchen: Ropieee (RPi3b+ with touchscreen) + Schiit Modi3E to a pair of Morel Hogtalare. 

All absolute statements about audio are false :)

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All those except Teac use ESS Sabre so they behave the same way...

 

Sabre works similar way as DSD processor mode in CS4398 (non-DirectDSD). So there is digital processing for the DSD stream (filter, volume and remodulator). This diagram of CS4398 illustrates it quite well:

4398blkdiag_mag.gif

(note on CS4398 there's a configuration option to choose "DSD Processor" path or "Direct DSD" path, controlling those "MUX" switches)

Output of both ESS and Cirrus DSD processing engines is still at the native sampling rate (2.8/5.6 MHz etc) but multi-bit SDM.

 

Teac (with TI chip) has four different configuration options for it's conversion stage to form analog filters and AFAIK no other processing.

 

 

Thanks for the information Miska, so how does Chord do it without a DAC for DSD. I know Chord talks about a Xilinx Virtex field-programmable gate array (FPGA) to handle the digital to analog conversion process but is it that different than the others ?, I don't know I just trying to wrap my head around all of this. ps; great thread Ted

The Truth Is Out There

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This thread is so far focussed heavily on the DAC chips. We must also consider every other component and its implementation within the digital to analog converter. The actual D to A chip is but one part of the design.

Interesting comment ... it has been said by a couple of manufacturers (sorry I can't remember who) that yes they could enable DSD playback, but they feel it would cause the PCM playback to suffer as the analogue stage would have to be compromised.

 

Eloise

Eloise

---

...in my opinion / experience...

While I agree "Everything may matter" working out what actually affects the sound is a trickier thing.

And I agree "Trust your ears" but equally don't allow them to fool you - trust them with a bit of skepticism.

keep your mind open... But mind your brain doesn't fall out.

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