ANOTHER Example of Why I HATE DSD and Why Customers Who Bought Sony's Boloney Are So Annoying

[barrows]

It does not matter how big a market share R2R DACs have, my point is that there are plenty of R2R DACs to choose from for the audiophile who wants one.

The real world measurements of actual DACs one can buy are there at stereophile.com for anyone to view. Not talking theory here, what happens with real world DACs which audiophiles purchase. The measurements show more HF noise at a lower frequency with DSD playback. Yes, this test is with DSD 64 content, as that is what 99.9% of all available DSD recordings (SACD) are. What is possible in theory is not what I was referring to, I was referring to DACs (and disc players) which are available to go and purchase at your audio dealer.

 

There is big difference to what happens in the real world, where most of the audiophiles live, and the products which they can purchase for playing back music, and what may, or not, happen in the world of Miska, where SD modulators are custom, DACs are custom, other hardware is custom, etc. What I am saying is that the results which you have are not available to the average audiophile, and as such are irrelevant (except theoretically).

[Miska]

The real world measurements of actual DACs one can buy are there at stereophile.com for anyone to view.

 

Where are all the wideband measurements up to 5 MHz that could actually show the out of band noise?

 

Not talking theory here, what happens with real world DACs which audiophiles purchase. The measurements show more HF noise at a lower frequency with DSD playback.

 

Yes, that's what I'm talking about too. But could you point to some particular measurements?

 

Yes, this test is with DSD 64 content, as that is what 99.9% of all available DSD recordings (SACD) are. What is possible in theory is not what I was referring to, I was referring to DACs (and disc players) which are available to go and purchase at your audio dealer.

 

We were talking about what happens inside DAC chip here, not about the content.

 

But I upsample all PCM content to DSD and many other do too. There's no reason to stick with DSD64 in those cases.

 

There is big difference to what happens in the real world, where most of the audiophiles live, and the products which they can purchase for playing back music, and what may, or not, happen in the world of Miska, where SD modulators are custom, DACs are custom, other hardware is custom, etc. What I am saying is that the results which you have are not available to the average audiophile, and as such are irrelevant (except theoretically).

 

What!? They are very much relevant. Measurement results are from real world DACs. Simulations are simulations, because otherwise you have to laser-cut the DAC chip open to hook into insides. And sometimes it is necessary to discuss theory and technology development too.

 

But mostly I talk about modulators in (1) DAC chips, (2) in HQPlayer. Both are available to everybody.

[Miska]

Some more real world measurements.

 

Here is RedBook frequency sweep through a DAC based on Burr-Brown DAC chip (their hybrid design):

(notice the delta-sigma noise bump at 500 kHz vs the images every 352.8 kHz caused by SAH from the PCM data, noise is non-correlated and the images are directly correlated)

 

Here is DXD frequency sweep through the same BB based DAC:

 

Here is RedBook frequency sweep through a DAC based on R2R Burr-Brown DAC chip and their (brickwall) DF chip:

[wappinghigh]

Wonder if this will affect DSD's future.

 

http://www.nytimes.com/2013/05/28/business/global/sonys-bread-and-butter-its-not-electronics.html?pagewanted=all&_r=0

 

I saw this story on the news last week.

 

LOL! Yes that has been their problem all along. Ignoring audiophiles and playing with their gaming stations instead.. Darn it, they used to make some great sounding hifi. And yes the SACD was a fantastic development. But they never properly supported it out in the recording industry. (See that other lengthy blog "why DSD failed"..) And now distribution has moved online, the rest is history. Unfortunately for us, the rot set in after they got greedy with the walkman. It's been downhill ever since...

[wappinghigh]

You know Sony *could* give us the combined PCM/DSD player we want.. A disc less stereo/multichannel streamer. No disrespect, but at the end of the day, 99.9% of folks aren't interested in the esseteric arguments going on in this thread. What they want is a simple way to play *all* available files. If they buy a DSD track, they want to play it in native DSD. If it's PCM, then let it stay PCM. They don't give too hoots about the underlying reasons why one format might sound better than the other. At the end of the day they just want to play the file.... without a glitch. So they can go on and listen to their music. I'm sure if Sony could they put their mind to it, they could produce such a player in a flash. The disc less equivalent of a Universal disc player. They just chose not to. Why this is the case, has been a decade long mystery to me...

[Audio_ELF]

The new Naim DAC-V1 is delta-sigma DAC like their CD and network players too.

Just to be accurate...

The Naim DAC (as opposed to the DAC-V1), the NDS streamer and CDS and 555 players (Naim's top end products) all use the TI 1704 r2r DAC. The other devices use various TI delta-sigma DACs.

 

Eloise

[wappinghigh]

Let's turn back to the OP'r original grievances. And yes, they are very legitimate. and I quote:

 

"Now it turns out that the customer is DEEPLY DISTRESSED that he can no longer have the "purity" of avoiding a preamp in his system, EVEN THOUGH J.RIVER INCLUDES A VERY SOPHISTICATED DIGITAL VOLUME CONTROL UTILIZING DOUBLE PRECISION FLOATING POINT CALCULATIONS (never mind how we are going to feed those fractions of a bit to the DAC chip....!

 

And the customer is so upset that he is threatening to return his DAC or his computer or his software, or his DoP (is that pronounced with a "long" "o", by any chance?) files or SOMETHING.

 

BECAUSE HE WANTS PURITY, BY GOD, AND NOW HE'S GOT THE MOST PURE DIGITAL MODULATION SCHEME EVER INVENTED BY MAN (at least according to Sony in the early 'oughties) and the MOST PURE DIGITAL PLAYBACK SOFTWARE (according to most people that ever have used it although for some reason I think that Miska will cast a vote of dissension) and the MOST PURE DAC EVER DEVELOPED (according to me, who admittedly should be hospitalized for delusions of grandeur), BUT HE CANNOT NOW CONTROL HIS VOLUME LEVEL WITHOUT (UGH!, hold your nose while you picture this) inserting an ANALOG VOLUME CONTROL INTO HIS SUNCTUM PURITUM, which will obviously soil his entire musical one-ness....

 

O what to do, what to do...

Do any of you expert readers have any suggestions that will save this poor soul from a life of hardship and MUSICAL POLLUTION MOST FOUL?"

 

Now reading between the lines, what is upsetting this poor tormented soul is that having spent a small fortune setting up what he/she thought was an ideal playback system, they have discovered a significant flaw. Digital volume control. Or even if it's not a flaw, they perceive it is. The frustration occurs because he/she is well read on forums such as this, and knows perfectly well what he/she is after is in fact achievable. Using the well known knowledge accumulated by the learned folks that visit C.A. Can I just say, I here ya Charles!!!!

 

Just why oh why the industry hasn't produced such a player, with all you geniuses around to help them.. and marketed it properly so simple folks like me can understand it, then just get on with it and simply use it..knowing that it is doing it's "thing".... Well that is beyond me. All we want is it to stream every file: DSD or PCM. Stereo and multichannel. Gapless via UPnP. In the purest way possible. So "if it's DSD let me here DSD. If it's PCM, then let it be PCM" (to quote TedB again).. With Internet radio plug in's available for easy casual listening as well...With a minimum of fuss...?? And no loss of bit's with volume control.

 

Is that to much to ask?

 

You geniuses can all argue until the cows come home, but the reality is customers just want to know it will work and do what they have paid for. Cheers

[Charles Hansen]

The fair test here for listening would be a DSD recording and PCM (24/192) made at the same time, with the DSD made through the ADC of ones' choice, and the PCM using Ayre's QA-9.

 

Hello Barrows,

 

That's close, but a MUCH better test would be to use the QA-9 for BOTH recordings. Then you would have the same:

 

1) Power supplies

2) Analgo circuitry

3) A/D Chip

4) Connectors

5) PCB material

6) Metering circuitry

7) Master clock with identical phase noise spectrrum

8) Digital driver circuitry

9) Chassis construction

10) Et detera

 

The ONLY difference is that in one position the FPGA is loaded with a program to provide low pass filtering to give a PCM output In the other position the FPGA is loaded with a program to perform 7-th order noise shaping (as mandated by the Scarlet Book) to give 1-bit, 64x DSD output. In both positions, the FPGA clock is running at the identical frequency (this affects the sound quality too -- EVERYTHING MATTERS!)

 

Then on the other end use a QB-9-DSD for the DA conversion where, once again, EVERYTHING IS AS IDENTICAL AS IS POSSIBLE TO MAKE IT.

 

This test would actually give an advantage to the DSD, because the ESS chip is essentially a D-S Modulator type of chip, so there would be one less conversion using the DSD.

 

But it is trivial to do and would settle the question for once and all.

 

Cheers,

[Charles Hansen]

The only thing "magical" about DSD sound, when one gets around the DF issue, is likely the presence of large amounts of HF noise, and how this interacts with the rest of the system. I suspect, that this noise is often (but not always) responsible for the "sweet" "soft" sound of DSD often present in many systems.

 

This was confirmed to me back in the early 2000's by one of the Philips engineers. He said that by playing with the frequency response of the noise shaper that you could get the system to sound almost any way you wanted it o sound -- soft and sweet, hard and edgy, smooth and creamy, et cetera. So the noise shaping curve they used was deliberate selected to give a specify "tonality" to the process.

 

This is why the high end (in particular) of DSD tends to have the same sound on all SACD releases. (At least the ones from early on that were made using the Sony-owned hardware that was designed and built by Ed Meitner and Andreas Koch. After other companies started making their own "DSD" converters, all bets are off as there is no guarantee that the used the same noise shaping filter. BUT all of the recordings made by one particular converter will have a definite sonic signature, just recordings made with an RCA BX-44 ribbon all have a similar coloration.)

[wappinghigh]

This was confirmed to me back in the early 2000's by one of the Philips engineers. He said that by playing with the frequency response of the noise shaper that you could get the system to sound almost any way you wanted it o sound -- soft and sweet, hard and edgy, smooth and creamy, et cetera. So the noise shaping curve they used was deliberate selected to give a specify "tonality" to the process.

 

 

Charles go have a play with a Resonessence Invicta. Or at least visit their website. This is exactly what they do with all their individual filters. Read up on their site. It explains digital volume attentuation beautifully. And why filtering is necessary. Why it is needed and why it's difficult to get right. You either do it in the physical player (in their case). Or in the software (like JRiver of HQplayer). It's a philosophical choice which path you go down. Cheers Wap

[wappinghigh]

Charles are you really the founder of Ayre Accoustics?

 

If so, then surely you will know what I mean...

[Adyc]

Well, I don't see that matching the QA-9 either. Maybe I'll get one for evaluation and post some measurement results. But I wasn't at all impressed on the Stereophile measurement results either.

 

Can I ask which commercially available hardware you are impressed with? How about dcs Vivaldi or MSB Diamond?

[Charles Hansen]

And no loss of bit's with volume control.

 

 

Hello Rapping Hood,

 

This is where the poor guy got misguided. The ONLY way to implement a volume control with no loss of bits is to use an analog one. So he' already been fed a bunch of nonsense that a digital volume control is the best way to go.

 

But I think what REALLY chapped his britches was that he had been convinced (by someone!) that DSD was infinitely superior to (ugh!) nasty old PCM. And the fact remains that it is IMPOSSIBLE to change the volume of a DSD data stream unless you first convert it to PCM, adjust the volume (losing resolution along the way) and the remodulating it BACK to DSD (adding more noise to the signall).

 

So just when he thought he had finally reached audio nirvana, he actually found out that somewhere along the line he had been lied to, and now he has to go back and reinvent a completely NEW paradigm so that he can reach audio nirvana....

 

My best guess is that he will end up with an all LP-system completely based on single-ended triodes with zero feedback and horn speakers. But I could be wrong! (I often am...)

 

Cheers,

[Charles Hansen]

In my previous post a said that only an analog volume control can avoid the loss of resolution. Here is a great talk by Martin Mallinson of ESS explaining WHY this is so, but also explaining why most people can get a digital volume control that GOOD ENOUGH to satisfy them.

 

 

Cheers,

[Superdad]

Off-topic question for Mr. Hansen:

Did ARDA ever produce and send you samples of the AT-1401 R2R DAC chip they announced before Google bought out the company?

If so, then the die must be residing somewhere in the world and maybe someone will eventually decide to produce it. I was pretty crest-fallen when I found out they closed (and the president/chief scientist never retuned my e-mail inquiries).

 

Secondly, did you have to make a lifetime buy of the AT-1201 ADC, or are there still a couple of distributors in the world holding stock?

Just curious.

 

Thanks,

ALEX

[wappinghigh]

Hello Rapping Hood,

 

This is where the poor guy got misguided. The ONLY way to implement a volume control with no loss of bits is to use an analog one. So he' already been fed a bunch of nonsense that a digital volume control is the best way to go.

 

 

Well F me. Look's like I've been lied to as well. Thanks for the legs up. LOL!

 

But Charles old son. Ask yourself if it *really* matters... If you are in fact the founder and designer of Ayre and you are an analog hardware engineer, why not give the punters (like me) what they want. Just forget about JRiver and computers and usb. That's yesterdays tech anyway man..Why not design an all in one Ayre with UPnP support of stereo and multichannel PCM/DSD and give it a superb analog control. Then out comes that beautiful sound via analog outs. Your an analog engineer man..stick to what you know and are good at!! LOL! Ol coot's like us all know analog is not half bad... hahahaha. Then explain in simple layman's term why you have done this and be done with. It would sell like hot cakes. Audiophiles are trusting soles as you know. Cheers Rap.

[Hiro]

That's close, but a MUCH better test would be to use the QA-9 for BOTH recordings.

 

That's exactly what I suggested earlier. But you would have to use a different DAC, something that is actually capable of playing back DSD-256 recordings made with the QA-9. Merging HORUS, and exaSound e20 come to mind...

[Miska]

But I think what REALLY chapped his britches was that he had been convinced (by someone!) that DSD was infinitely superior to (ugh!) nasty old PCM. And the fact remains that it is IMPOSSIBLE to change the volume of a DSD data stream unless you first convert it to PCM, adjust the volume (losing resolution along the way) and the remodulating it BACK to DSD (adding more noise to the signall).

 

This is wrong, you don't have to convert to PCM. You can keep it in SDM domain all the way. If you don't know how to do it, doesn't mean it doesn't exist.

 

If you double the SDM rate while doing it, the output can easily have 10 - 20 dB lower noise floor than the source material.

 

Same amount of noise is added in digital in both PCM and SDM cases, if you use same input and output resolutions.

[Miska]

This was confirmed to me back in the early 2000's by one of the Philips engineers. He said that by playing with the frequency response of the noise shaper that you could get the system to sound almost any way you wanted it o sound -- soft and sweet, hard and edgy, smooth and creamy, et cetera. So the noise shaping curve they used was deliberate selected to give a specify "tonality" to the process.

 

Sort of yes, large part of how a delta-sigma DAC sounds like originates from the modulator design properties. But DSD doesn't sound better because of the noise. PCM sounds worse than DSD because PCM has either 1) brickwall filter or 2) hefty amount of directly correlate digital images that cause intermodulation products.

 

This has been very clear for a long time already.

[Miska]

Can I ask which commercially available hardware you are impressed with? How about dcs Vivaldi or MSB Diamond?

 

I'm very hard to impress, I'm almost never impressed. But I try to pick up my highlights of things I currently don't have as example.

 

I was impressed by Devialet, because it had some original thinking and good performance. And also by NAD's direct digital amps. I'm impressed by Accuphase how they keep producing nicely performing good sounding amps with great build quality.

 

I like the technology of dCS, Playback Designs and Meitner/EMM. And maybe MSB too, but I don't know enough of it to really have an opinion. But I'm not impressed by the price tags, not available here and outside of my budget.

 

I am particularly impressed by Chord QuteHD especially if it gains DSD128 capability, because it combines original engineering, good performance and reasonable price tag.

 

And of course I'm happy with the equipment I have.

[wappinghigh]

Miska you are an incredibly knowledgeable guy... why don't you put your mind to it and embed your HQ player into a dedicated UPnP multichannel streamer? Just sayin' "Rapping Hood".

[wappinghigh]

P.S this is a totally awesome thread. At the cutting edge people

[Adyc]

This is wrong, you don't have to convert to PCM. You can keep it in SDM domain all the way. If you don't know how to do it, doesn't mean it doesn't exist

 

Do you think the volume control in dcs DAC is also done in SDM domain?

[Adyc]

But I think what REALLY chapped his britches was that he had been convinced (by someone!) that DSD was infinitely superior to (ugh!) nasty old PCM. And the fact remains that it is IMPOSSIBLE to change the volume of a DSD data stream unless you first convert it to PCM, adjust the volume (losing resolution along the way) and the remodulating it BACK to DSD (adding more noise to the signall).

 

Can you prove it mathematically it is impossible to change the volume in DSD domain? One side says it is possible and the other side says it is impossible. To settle it, it is the best to show it mathematically one way or the other way.

[tailspn]

Can you prove it mathematically it is impossible to change the volume in DSD domain? One side says it is possible and the other side says it is impossible.

 

DSD is not a domain; it's a marketing term for a specific format of Sigma-Delta Modulator coding (SDM), and specifically 1-bit two level. It is not possible to do a level change in a 1-bit, two level data stream directly. That's because there's no magnitude information in any one sample time, just an indicator of whether there was a level change from the previous sample time.

 

2's compliment coding PCM solves that issue by having each sample being an absolute stand along magnitude value, and therefore is easily processed digitally. The drawback to PCM is there's no practical way to create it from an analog signal directly, as there is with 1-bit two level, or multi-bit Delta-Sigma Modulators, and therefore must be derived from the SDM stream.

 

Data in the form of multi-bit SDM (the stuff that comes out of the front end of an A/D converter, and is the inter element code within a D/A converter) does have a quasi magnitude value, and can be processed digitally to perform a level change. This can be done at the original sampling rate, or higher, It's just not 2's compliment weighted, nor does it have the detrimental effects of decimation needed to convert to the presently slower sample rates of 2's compliment coded PCM.