Don't Fear The Reaper

Halloween Music: It Was a Graveyard Smash!
Tue, 10/31/2017

Music in the Round #87: Oppo UDP-205 Recordings in the Round

Tue, 10/31/2017

COMMENTS
Anton's picture

It was educational, as well....and I own one!

As a two channel guy, I will add: This baby is even better than Kal said!

tonykaz's picture

Hmm, $1,300.

All things being equal I figured OPPO stuff would be in the $5k range by now.

It's better in all manner of ways, isn't it?

Egads, seems like Oppo is lowering their prices.

And,
this is the same outfit that MSB in Watsonville says is so very high level. Phew.

And it's the Player that Mr. JVS has sitting amongst those very pricey Wilsons.

I'm heading to 6moons to double check your appraisal.

So-far, I'm stammered by your reviewing discoveries.

Tony in Michigan

ps. I'm not a TV / Video guy so I don't follow this 4K stuff.

Anton's picture

Awesome.

"Didn't" find a way to bring up vinyl!

;-D

Kal Rubinson's picture

In a "Manufacturers' Comment," Jason Liao announces that the UDP-203/205 models are now Roon Ready devices and, therefore, can be controlled by Roon via Ethernet. This is, of course, good but please note that this support does not extend to DSD or to multichannel. IMHO, this is unfortunate.

tonykaz's picture

Roon is only music, so far, aren't they?

I thought that they were an off-shoot of MQA people who are an off-shoot of Meridian.

Egads, so many intrigues.

Overall, I contend that things are getting pretty darn good, especially compared to those old 100lb. TV sets sitting in everyone's basement, patiently waiting for a safe way to dispose of them.

I admire your work ( and Professor Waldrip's) in this area buuuuuuuut, as a one time retailer, I can't see regular civilians willingness to spend for quality music gear at the entry level, much less the 5 Channel versions. ( as of today )

Futuristically, I see all music delivery as surround type format, it's the method that can place listeners in the experience. Music in the Round is like being in the Holidome, I suspect it's a logical application of the emerging 4th Generation of music formats.

We have a wonderful music future ahead of us ( I think; in the next 10 years ).

Tony in Michigan

ps. we've come a very long way in the last 70 years, which is only the beginning.

Kal Rubinson's picture

Yes, Roon is just music, afaik, not movies/video. Also, there is a back-story about their relationship to Meridian but they are now quite independent.

tonykaz's picture

I just got an email say'n that our Bernie Sanders is Larry David's cousin, he looks strong enough to run in 2020 ( where I'll again be delighted to help )

This Internet is Shrinking the world around us.

I'm watching Roon

Tony in Michigan

ps. the RMAF17 Seminars are starting to be YouTube released, they're pretty darn good.

Richard D. George's picture

Great review, Kal.

I used to be a big Oppo fan, and I fully realize that the focus of the review is music. But....
If you don't actually spin music discs anymore (which I don't, anymore), the new Sony UBP-X1000 ES is a better purchase if you want to use the player for spinning 4K HDR video discs. I traded in an Oppo 105 and a 103D for two of the Sony units in one house, and will trade in two similar Oppo's in our mountain house next year when the other equipment is 4K HDR ready. The build quality of the Sony is very good (not like "consumer" Sony Blu-ray players) and takes 3-prong IEC power cords so you can use an upgraded power cord. It also has features that are useful to custom installers (Control 4 friendly). And... you can stream video through it, a feature that Oppo has dropped. The Sony unit is half the price of the Oppo.
I love listening to music, particularly high rez files and Tidal (and Spotify) and bit-perfect ripped CD's. I just don't play CD's or SACD's anymore.

Kal Rubinson's picture

Great review, Kal.
Thanks.

I used to be a big Oppo fan, and I fully realize that the focus of the review is music. But....If you don't actually spin music discs anymore (which I don't, anymore), the new Sony UBP-X1000 ES is a better purchase if you want to use the player for spinning 4K HDR video discs.
Thanks but I am not particularly interested in 4K video although I have a Samsung 4k player in my work room.

Richard D. George's picture

Fair enough

Richard D. George's picture

I suspect that there are more than a few Stereophile readers that:

- Already have a decent DAC, in one form or another
- Do not regularly play SACD or DVD-A discs

For the same money as the Oppo, these good folks could buy a Bluesound Vault 2 and connect it to their existing DAC (with a good coaxial digital cable) and:

1) Buy high rez files from HD Tracks and have them automatically download to the Vault 2 (2 TB capacity)
2) Take their existing CD's and rip them (bit perfect) directly into the Vault 2
3) Access existing music files on other NAS's using the Vault 2
4) Stream high rez Tidal using the Vault 2. There is also support for MQA.
5) Access all of the above in different locations in the house if additional Bluesound devices are added later.

I used to be a huge fan of Oppo and have purchased 5 of their players, including a few flagship 105's. In the brave new world, their appeal is now quite narrow in my opinion.

Kal Rubinson's picture

For the same money as the Oppo, these good folks could buy a Bluesound Vault 2 and connect it to their existing DAC (with a good coaxial digital cable)
AFAIK, no BlueSound device does multichannel. Thus, I prefer using my Baetis Prodigy-X server with an external DAC to do those things.

Richard D. George's picture

Fair enough.

My macro level point is that Oppo players used to have very broad appeal (to people like me) and I submit that going forward they will have a much narrower appeal.

Kal Rubinson's picture

Oh, I do agree and a lot of it is that the market it is in has become splintered at the same that Oppo has deleted apps that broadened its appeal.

David Harper's picture

good review but an even better deal is the OPPO 203 which is 99% as good for half the price. I think I read that the two are identical except for a couple of relatively minor circuitry differences which wouldn't mean anything to the vast majority.

Kal Rubinson's picture

Indeed. There is a different DAC but that affects only the analog outputs. Otherwise, a great alternative.

David Harper's picture

and the replies above which criticise the OPPO on the basis of the lack of importance of playing discs anymore are simply wrong. Millions of audiophiles still want an excellent digital disc player. I have absolutely no interest in streaming and files and Tidal and spotify and all that BS.
Nobody has ever been able to reliably and repeatedly distinguish between CD and hi-res in a properly controlled double blind test.

Pages

Music in the Round #87: Oppo UDP-205 Page 2

Tue, 10/31/2017

COMMENTS
Anton's picture

It was educational, as well....and I own one!

As a two channel guy, I will add: This baby is even better than Kal said!

tonykaz's picture

Hmm, $1,300.

All things being equal I figured OPPO stuff would be in the $5k range by now.

It's better in all manner of ways, isn't it?

Egads, seems like Oppo is lowering their prices.

And,
this is the same outfit that MSB in Watsonville says is so very high level. Phew.

And it's the Player that Mr. JVS has sitting amongst those very pricey Wilsons.

I'm heading to 6moons to double check your appraisal.

So-far, I'm stammered by your reviewing discoveries.

Tony in Michigan

ps. I'm not a TV / Video guy so I don't follow this 4K stuff.

Anton's picture

Awesome.

"Didn't" find a way to bring up vinyl!

;-D

Kal Rubinson's picture

In a "Manufacturers' Comment," Jason Liao announces that the UDP-203/205 models are now Roon Ready devices and, therefore, can be controlled by Roon via Ethernet. This is, of course, good but please note that this support does not extend to DSD or to multichannel. IMHO, this is unfortunate.

tonykaz's picture

Roon is only music, so far, aren't they?

I thought that they were an off-shoot of MQA people who are an off-shoot of Meridian.

Egads, so many intrigues.

Overall, I contend that things are getting pretty darn good, especially compared to those old 100lb. TV sets sitting in everyone's basement, patiently waiting for a safe way to dispose of them.

I admire your work ( and Professor Waldrip's) in this area buuuuuuuut, as a one time retailer, I can't see regular civilians willingness to spend for quality music gear at the entry level, much less the 5 Channel versions. ( as of today )

Futuristically, I see all music delivery as surround type format, it's the method that can place listeners in the experience. Music in the Round is like being in the Holidome, I suspect it's a logical application of the emerging 4th Generation of music formats.

We have a wonderful music future ahead of us ( I think; in the next 10 years ).

Tony in Michigan

ps. we've come a very long way in the last 70 years, which is only the beginning.

Kal Rubinson's picture

Yes, Roon is just music, afaik, not movies/video. Also, there is a back-story about their relationship to Meridian but they are now quite independent.

tonykaz's picture

I just got an email say'n that our Bernie Sanders is Larry David's cousin, he looks strong enough to run in 2020 ( where I'll again be delighted to help )

This Internet is Shrinking the world around us.

I'm watching Roon

Tony in Michigan

ps. the RMAF17 Seminars are starting to be YouTube released, they're pretty darn good.

Richard D. George's picture

Great review, Kal.

I used to be a big Oppo fan, and I fully realize that the focus of the review is music. But....
If you don't actually spin music discs anymore (which I don't, anymore), the new Sony UBP-X1000 ES is a better purchase if you want to use the player for spinning 4K HDR video discs. I traded in an Oppo 105 and a 103D for two of the Sony units in one house, and will trade in two similar Oppo's in our mountain house next year when the other equipment is 4K HDR ready. The build quality of the Sony is very good (not like "consumer" Sony Blu-ray players) and takes 3-prong IEC power cords so you can use an upgraded power cord. It also has features that are useful to custom installers (Control 4 friendly). And... you can stream video through it, a feature that Oppo has dropped. The Sony unit is half the price of the Oppo.
I love listening to music, particularly high rez files and Tidal (and Spotify) and bit-perfect ripped CD's. I just don't play CD's or SACD's anymore.

Kal Rubinson's picture

Great review, Kal.
Thanks.

I used to be a big Oppo fan, and I fully realize that the focus of the review is music. But....If you don't actually spin music discs anymore (which I don't, anymore), the new Sony UBP-X1000 ES is a better purchase if you want to use the player for spinning 4K HDR video discs.
Thanks but I am not particularly interested in 4K video although I have a Samsung 4k player in my work room.

Richard D. George's picture

Fair enough

Richard D. George's picture

I suspect that there are more than a few Stereophile readers that:

- Already have a decent DAC, in one form or another
- Do not regularly play SACD or DVD-A discs

For the same money as the Oppo, these good folks could buy a Bluesound Vault 2 and connect it to their existing DAC (with a good coaxial digital cable) and:

1) Buy high rez files from HD Tracks and have them automatically download to the Vault 2 (2 TB capacity)
2) Take their existing CD's and rip them (bit perfect) directly into the Vault 2
3) Access existing music files on other NAS's using the Vault 2
4) Stream high rez Tidal using the Vault 2. There is also support for MQA.
5) Access all of the above in different locations in the house if additional Bluesound devices are added later.

I used to be a huge fan of Oppo and have purchased 5 of their players, including a few flagship 105's. In the brave new world, their appeal is now quite narrow in my opinion.

Kal Rubinson's picture

For the same money as the Oppo, these good folks could buy a Bluesound Vault 2 and connect it to their existing DAC (with a good coaxial digital cable)
AFAIK, no BlueSound device does multichannel. Thus, I prefer using my Baetis Prodigy-X server with an external DAC to do those things.

Richard D. George's picture

Fair enough.

My macro level point is that Oppo players used to have very broad appeal (to people like me) and I submit that going forward they will have a much narrower appeal.

Kal Rubinson's picture

Oh, I do agree and a lot of it is that the market it is in has become splintered at the same that Oppo has deleted apps that broadened its appeal.

David Harper's picture

good review but an even better deal is the OPPO 203 which is 99% as good for half the price. I think I read that the two are identical except for a couple of relatively minor circuitry differences which wouldn't mean anything to the vast majority.

Kal Rubinson's picture

Indeed. There is a different DAC but that affects only the analog outputs. Otherwise, a great alternative.

David Harper's picture

and the replies above which criticise the OPPO on the basis of the lack of importance of playing discs anymore are simply wrong. Millions of audiophiles still want an excellent digital disc player. I have absolutely no interest in streaming and files and Tidal and spotify and all that BS.
Nobody has ever been able to reliably and repeatedly distinguish between CD and hi-res in a properly controlled double blind test.

Pages

Music in the Round #87: Oppo UDP-205

For some months now, I've lived mostly without music. To survive the dust and grit of the renovation of our Manhattan apartment, all electronics had to be covered with heavy plastic, the speakers encapsulated in large green lawn bags, and the listening room partitioned off with a temporary wall. We could listen to music with our little 3.1-channel TV system in the den (eh) or through headphones (not!), or we could decamp to our house in Connecticut, which we did as much as possible. I felt deprived. Now that it's all over, I'm grateful to have it back—and grateful for the improvements in the main system, some of them direct byproducts of the renovation.
Tue, 10/31/2017

Recording of March 1977: Direct from Cleveland

Direct From Cleveland
Orchestral works by De Falla, Bizet, Tchaikovsky, Berlioz
The Cleveland Orchestra, Lorin Maazel (cond.)
Telarc 5020 DD1 (LP). Robert Woods, prod.; Jack Renner, sound eng.; Glenn Glancy, Michael Bishop, disc-cutting engs.

Potentially the best news for perfectionists in years is the announcement of the first stereophonic direct-to-disc recording (in the US, at least) of a major symphony orchestra. Advent records of Cleveland, in collaboration with Discwasher, Inc. of Columbia, MO put four complete and usable runsthrough onto two sets of lacquers. The program was a collection of potboilers—what Sir Thomas Beecham used to call "lollypops"—much of it musically rather trivial, but all ideally suited for demonstrating what a no-holds barred recording can do in terms of sonics: works with bass drum, percussion, deep double-bass material, rich string sonorities" and so on.

Tue, 03/01/1977

Capital Audiofest Starts Friday

From a small regional show, Gary Gill's seven-year old Capital Audiofest has grown into the East Coast Show of 2017. Set for November 3–5 in the Hilton Hotel Twinbrook in Rockville, MD, CAF will offer 57 exhibit rooms spread over three floors plus the hotel Atrium. That amounts to 93 exhibitors and over 200 brands, including a CanMania with 20 headphone vendors. For a show that, just last year, maxed out at 40 rooms with 65 exhibitors and 85 brands, this represents major growth.
Mon, 10/30/2017

Tripping with Terry Riley

Did I really listen to and love the hi-rez (24/44.1k) file equivalent of four CDs chock-full of piano music written by and for the great Terry Riley (b. 1935)? Not only is the answer in the affirmative, but I can now honestly attests that pianist/pedagogue Sarah Cahill's Eighty Trips Around the Sun abounds in opportunities to take you on multiple mind-bending excursions through the mind of a true master.
Sun, 10/29/2017

Benchmark DAC3 HGC D/A preamplifier-headphone amplifier Measurements

Thu, 10/26/2017

COMMENTS
tonykaz's picture

NwAvGuy was describing the Benchmark and using it to evaluate his own DAC designs.

Mark Waldrip relies on Benchmark gear

I've heard of a good many Pro Audio folks tout Benchmark as being the Benchmark.

I've been trying DACs for some years now, I can't discover any advantage behind some of the super pricy DACs.

I've blamed my hearing, to the point of having my hearing evaluated by Audiologists at the University of Michigan. My hearing tapers off above 8k but I can still hear significant differences in 12AU7 preamp tubes. I still can't hear greatness in super expensive DACs.

Those that can hear significant improvements in Super Expensive DACs are living with the Audiophile Curse. ( the King's New Suit Curse )

As an Engineer, I've demonstrated Zip Lamp Cord vs. Monster Speaker Cable vs. Bruce Brisson's MH-750 to astonished Engineers.

Back in 2015 Tyll had a headphone gear Shootout where Stax, Sennheiser, and Audeze were the finalists. He also had the finest DACs including the highly touted Antelope. Nobody could detect any DAC performance advantage.

Then there's Chord and their Field Programable Gate Array devices which I suspect they've designed to sound better but probably not sound accurate ( like Tubes that make music nicer sounding ).

So, go for Accuracy with Benchmark or better sounding with Chord.

Nice reporting, sir, I think.

Tony in Michigan

barrows's picture

I would suggest that there is not much evidence to support your assertion that Chord DACs are not designed to sound accurate, take a look at their measurements here at stereophile.com...

On this review in general, despite the lack of difference heard vs the DS, it is interesting to note that Jim found an immediately apparent difference to the Benchmark DAC1, subjectively speaking...

Staxguy's picture

Accurate? Chord DAC's can't even handle (do) 24-bit. Even their best DAC, the Chord DAVE can't even do 24-bit. 21-bit, that's it!

Their 17-order noise shaper can do 350 dB DNR, but that's internal aah only.

It's taken forever for this review to come out. Back when we (I) was recommending the Benchmark DAC1 HGC in my magazine as a recommended component, The Absolute Sound was only recommending DACs with sub-par 16-bit performance and not even mentioning it, even though it had came out.

Well, it's great to see this review. I might buy the Rogue Audio RH-5 as a headphone amplifier, reviewed just before this one, which likely functions worse (no better) than the Aurorasound HEDA, because it looks good on the desk!

People like benchmark, and good for/on them. It's got a great summing DAC (multiple DAC) design.

I'd rather have the GTE Trinity DAC (the one with the actual triangles not the newer ahh one), but that's for myspace!

barrows's picture

Your reply is kind of in error. Here is a quote from JA's measurements on this Benchmark:

"...the increase in bit depth dropped the noise floor by more than 30dB (fig.5), indicating that the Benchmark's resolution is at least 21 bits. This is as good as a DAC can currently get!"

There is no DAC of which I am currently aware that achieves better than 21 bit resolution at its analog outputs. The resolution measured here for the Benchmark is at the same level as Chord DAVE (look it up on this site). When DAC makers say they are 24 bits, or 32 bits, that is what level the DAC operates at in the digital domain, but no DAC achieves that resolution at its analog outputs.

I am curious, do you have any references for measurements of the Trinity? I would expect them to be way worse than that of the Benchmark, not saying anything against the Trinity as far as sound quality is concerned though, just, it is likely not nearly as "accurate" being an R2R DAC...

Sal1950's picture

"There's a danger of being misled, of repeating the same mistakes again and again, of spending way too much money on things of little value. My point is that, as a hobby, industry, and avocation, we may have shifted too far toward the subjectivist side."

What a breath of fresh air in these pages! When I first read this review in my subscriber copy my jaw hit the floor. Thank you Jim, we can only hope as audiophiles dedicated to the accurate reporting, that blind listening sessions become much more widespread and prevalent in the pages of Stereophile. Sighted listening is much too fallible to be the sole basis for accurate evaluations.

mrkaic's picture

I hope this year, this month, and this review mark the beginning of the end of the tyranny of subjectivists.

ChrisS's picture

...listening/reviewing of this component was done blinded.

He did it the way we all do.

He listens.

He thinks about it.

He writes about what he hears.

That's what they all do at Stereophile.

ChrisS's picture

"Yes. Listening is what Stereophile reviewers do."

mrkaic's picture

There is only one question in audio and it is rhetorical -- why buy anything but Benchmark?

They are the best, nothing comes close.

Charles Hansen's picture

You claim to have a PhD and you make the most egregious beginner's mistakes imaginable. It is laughable that even Streophile would publish this garbage.

First you start of the article with a story about "hum and buzz" from "poor quality control" in the interconnects. If you knew anything at all about balanced circuitry and how it works, you would realize that a truly balanced source sending a signal to a truly balanced downstream device wouldn't make a bit of "hum and buzz" if the cables were shielded or not.

But since you clearly have no idea about anything, you start with the wrong assumption and leap to an incorrect conclusion. [Flame deleted by John Atkinson]

Then you try to "compare" the Benchmark against your PS Audio. And although you could "easily" hear differences between the Benchmarks 1 and 3, you could hear no difference whatsoever between the Benchmark 3 and the PS Audio. Why don't you look at your ridiculous methodology before jumping to more false conclusions?

First you said you connected "sent the output of the two DACs to different channels of my PS Audio BHK Signature preamp". WTF? So you sent the left channel of one DAC to one input and the right channel of the other DAC to the other input? Or do you not know the difference between a "channel" and an "input"? At this point you are looking beyond unprofessional, beyond amateurish, and all the way to "knowing just enough to be dangerous". Or does your preamp have only one input? Or does it have more than one but the inputs were full, and you were too lazy to disconnect them?

Anybody serious would know that the only FAIR way to do this type of test is to not only use the same input on the preamp (different inputs will exhibit different amounts of break-in, as the trace on the PCB needs a signal to dissipate the residual electrostatic charge after it is tested with many kilovolts for UL compliance), but to also use the exact same pair of interconnects as they need to be not only the same brand and model, but also have the same amount of break-in time.

And finally you lose all rationality. (We may have to send the men in white coats to save you from harming yourself.) You send the analog output of the PS Audio DAC into the analog input of the Benchmark DAC. At the very least this signal is now going through the analog circuitry and volume control of the Benchmark, which is clearly coloring the sound if it sounds "identical" to you - either that or your hearing ability sucks or your playback system sucks. And if (like the version 2) the version 3 also has a "hybrid digital/analog" volume control then the volume control wouldn't even be active unless the analog input were being digitized by whatever cheap A/D chip Benchmark is using with their cheap IC op-amp based analog circuitry. More cheap crap to color the sound of the PS Audio, and you can't figure ANY of this out...

Finally it is clear that you cannot even add 2 + 2 to equal the correct answer. First you claim that the Benchmark "moves interpolation off-chip" to eliminate problems with "intersample overs", but that the latest version sounds better because of "lower passband ripple, facilitated by the new chip's superior filter choices". So which is it, Mr. PhD? Does it sound better because the processing is done off-chip or does it sound better because the on-chip processing has lower passband ripple? You can't have it both ways.

[Flame deleted by John Atkinson]

You are the one that makes a joke out of the high end, not the people who make claims you can't understand - you can't even understand the simplest of claims, as this so-called "review" clearly demonstrates.

AJ's picture
Quote:

Anybody serious would know that the only FAIR way to do this type of test is to not only use the same input on the preamp (different inputs will exhibit different amounts of break-in, as the trace on the PCB needs a signal to dissipate the residual electrostatic charge after it is tested with many kilovolts for UL compliance), but to also use the exact same pair of interconnects as they need to be not only the same brand and model, but also have the same amount of break-in time.
And finally you lose all rationality.

Can't make this stuff up ;-).
Charles, your rational scientific evidence of this audible "break-in" outside the imaginations inside your head?
TIA

mrkaic's picture

It boggles the mind that someone who believes in cable break-in has the chutzpah to write about electronics. It really does.

But we can learn here, for sure. In the words of the previous occupant of the White House— it is a teachable moment—sans beer, in this particular case.

It is tempting to get angry at the omni-present neglect of science education and fume about the ignorance of individuals who believe in cable break-in, directional fuses etc. But it is more fun is to realize that those esteemed gentlemen, who believe, for example, in cable break-in, provide some free comic relief in these trying times.

So, don’t get mad, get entertained.

Anton's picture

I'd hate to see how worked up you get over real world issues.

barrows's picture

Jim. I too was disturbed by some of your methodology in your comparisons, and dismissed your results because of my concerns. While I may not go as far as Charlie, please note that I felt disturbed enough to dismiss the results.
I would suggest that if you want to make quick A/B comparisons of source components in future reviews, you should have a balanced switch box made, who's only function is to take two balanced inputs, and switch them to a single balanced output. I would also suggest that this be wired the AES way for consistency (XLR pin 1 to ground) and that the signal path internally be identical for both inputs, same length wiring, etc. A high quality switch box like this should be a reasonable expense for any any reviewer who is interested in making quick A/B comparisons of source components.
Also of import would be keeping everything else the same, same cabling, same cabling lengths, etc.
In the case of comparing these two DACs (DS and DAC 3) such a methodology could then go direct into the amps, removing any additional components (preamp) and maximizing the apparent sonic differences. You should also volume match the DACs by measuring the output voltage with a test signal (like that available on stereophile test CDs) without very accurate level matching any such comparisons are mostly meaningless.

supamark's picture

won't work as you think. You're going to add two add'l sets of connections (in/out of the box) which will color the sound. Not every component has balanced outs, and a "dirty" little industry secret is that a majority of XLR outs are derived from the unbalanced RCA outs since few components are actually fully balanced internally. Okay, not really a secret since the few mfg's who make components that are fully balanced from input to output will make sure you know about it.

You will also still need a volume control since not everything being reviewed will have a variable output (and reducing volume in the digital domain adds its own.... issues). Jim's method of using two line inputs of his high quality preamp (that I'mma assume Jim knows the sound of quite well) is a much better methodology. The only way I could think of to improve his methodology would be for him to have someone else connect the two devices and make sure he doesn't know which input is which device.

Let me tell ya another little secret that every recording engineer knows about critical listening (this has happened to EVERY recording engineer at least once): You can imagine differences in sound that are not actually there. Every engineer has at some point reached for the EQ, started adjusting it and heard the sound change, and I mean literally hearing the changes being made to the sound.... but there's a problem - you never actually engaged the EQ (there's an on/off relay switch on every mixing console to switch the EQ into the signal path). It was all in your mind. This, to me, makes two very different points:

1. blind A/B/X testing really is the only fully valid method, but unfortunately nobody's come up with a truly transparent comparator (and likely never will).

2. those expensive cables from like Nordost et.al. are mostly snake oil. A much better system would be 100% copper all the way - internal component wiring (already happens), line level plugs/jacks, speaker connections all 100% copper (or silver/gold/whatever but must all be the same metal and the same purity/alloy). When the electrical signal is traveling along and the medium (metal) changes the signal will change. Were I a high end mfg, I would make all my connectors copper and provide high quality all copper cables at no add'l charge (because, seriously, the mark-up on cables is crazy - 30 years ago I picked up 40' of Kimber Cable at dealer cost, $1 per foot).

barrows's picture

won't work as you think. You're going to add two add'l sets of connections (in/out of the box) which will color the sound. Not every component has balanced outs, and a "dirty" little industry secret is that a majority of XLR outs are derived from the unbalanced RCA outs since few components are actually fully balanced internally. Okay, not really a secret since the few mfg's who make components that are fully balanced from input to output will make sure you know about it.
Read more at https://www.stereophile.com/content/benchmark-dac3-hgc-da-preamplifier-h..."

Only one xtra pair of interconnects, and this is not a problem, as long as they are the same, the difference will still be the same. Same thing with the switch box itself, the entire point is to give the same signal path for each DUT, the result being that one will only hear the difference. This set up, given decent construction and good interconnects will be transparent enough to hear any actual difference.
Additionally, while your contention that components are generally pseudo balanced may have been the case long ago, it is not common now-for example, the DACs mentioned in Jim's review are true balanced, as are his pre amp and amp.

And if you think Nordost cabling is snake oil, you are either deaf, or have not listened to it.

supamark's picture

It's not the 1 extra pair of interconnects (the extra wire is essentially meaningless*), it's the extra set of connections - jacks and plugs (along with the circuitry inside the box, including gain matching circuits, even if they're just variable resistors). You going with a relay or mechanical switch? Both degrade the sound in different ways. Adding that crap in will make it harder to hear any differences because you just added noise and made the signal path more complex. That is much worse than using two inputs of the same high quality preamp.

Most home audio gear is not fully balanced inside (nor should it be). Its adoption rate is higher in high-end home audio, but honestly it's not really necessary in most home environments. Pro audio, where there's much longer cable runs and a lot more electrical noise, is where it's really useful.

*when I was recording Austin Symphony Orchestra in Bass Concert Hall at The University of Texas in the early 90's, the mic cable run - from the stereo pair of Neumann U87 mic's through underground conduits below a couple other buildings into our machine room patch bay and into our Harrison Series 10 console was at least a quarter mile, probably close to 2,000 feet. The sound was not dull or even particularly weak. The U87's large presence boost probably helped but as I said, a couple feet of line level cable ain't gonna mean squat.

btw, if you think Nordost or anyone's cable can magically transform the sound that is otherwise running through traces on a f'ing circuit board inside each piece of your analog gear as it moves from cap to resistor to etc, you have a few things to learn about science (esp. physics). I mean, holy sh!t, they'll sell you little teflon and wood teepees for your speaker cables (for a pretty penny, 'natch). It's really no different than the Tice Clock or coloring the edges of your CD's with a green marker. Maybe Nordost has proprietary connectors that do a truly excellent job of passing signal with minimal alteration - great, that's a real upgrade and can be audible... but that wire between those connectors for which they're charging you several grand more than JA's preamp? that stuff is snake oil.

Buy a high quality power conditioner instead of fancy cables, it'll make a real audible difference and cost a lot less money.

barrows's picture

Are you nuts? I am talking about a passive switch box here, no active circuitry. Just two sets of XLR input jacks, some wire, a high quality switch (lets say a shallco) and two output jacks. That is all. No resistors, no circuitry of any any kind. Will this change the sound, perhaps a tiny, tiny bit, but not enough to obscure differences between the two DUTs, as the change will be identical to both products. A passive box like this will have much less influence on sound than a fully active preamp (which we eliminate from the chain with this testing method) which includes the wiring and switch, but also adds a power supply (noise source), transistors, resistors, etc. We do not need any circuitry, both DACs are designed to run direct to power amps, and we match levels via the DAC VC.

On cables, I never suggested that a cable can "magically transform sound"! Please do not put words in my mouth, neither did I make any comment re audio cable pricing. I just said that if you think Nordost cabling is snake oil, you are either deaf or have not listened to it, and I stand by that. To be more detailed: compare directly a Nordost cable to Mogami, and you will hear a significant difference. But the cable thing is OT here, so lets leave it for another place.

supamark's picture

Those added connections will in fact degrade the sound. You are adding 2 additional connections to each channel (XLR's, which really aren't as firm as RCA connectors - they wiggle a bit), each will degrade the sound. Also, since not every DAC has balanced outs, you'll want another box with RCA's (or just go unbalanced in the first place - the main advantage with balanced is if you have RF issues in your home or really long cable runs).

If you believe so strongly that wire has a sound, just use the same brand/length/termination/model for both and they'll be the same... it's cable, not 9' concert grand pianos (which actually do all sound different because they're handmade and no two are identical but brands do have "house" sounds - Hamburg Steinways are my favorite btw, and Baldwins are popular in rock because they're a little more mid-heavy with less sparkle than NY or Hamburg Steinways and therefore easier to cut through a dense mix).

Not every DAC has a volume control, and even for those that do you'll need a master volume control if you want adjust volume while keeping the levels matched - unless you think no volume adjustments should be made? Besides, the fixed output *should* always be superior (digital volume controls necessarily change the bits, and not for the better).

Also, what if the output impedences of the DACs are not close? going direct into a tube amp could cause differences with freq. response.

So, yeah, going into a high quality preamp is both simpler and more likely to yield consistent results with a much wider variety of DACs.

AJ's picture
Quote:

You going with a relay or mechanical switch? Both degrade the sound in different ways. Adding that crap in will make it harder to hear any differences because you just added noise and made the signal path more complex. That is much worse than using two inputs of the same high quality preamp.

You just need to wave your hands around as you say this stuff, then finish with "believe me" ;-)

supamark's picture

I'm livin' in your tiny little head, you can't quit me. it's really sad, really really sad.

AJ's picture
Quote:

but unfortunately nobody's come up with a truly transparent comparator (and likely never will).

Your factual evidence please.

supamark's picture

that's just stupid. Why don't you go out and find/build one and prove me wrong?

AJ's picture
Quote:

Why don't you go out and find/build one and prove me wrong?

Because folks with a modicum of intelligence know where Burden of Proof lies ;-).
Ok, so you admit to having zero evidence for your specious claim.
The question was rhetorical.

supamark's picture

when someone is stupid enough to ask someone to prove a negative, it simply proves that they're just stupid.

Again, find me a transparent A/B/X comparator system... maybe get Nelson Pass (or his pre-amp guy), Dan D'Augustino, and the guys at Schiit on it, they're all excellent engineers.

AJ's picture
Quote:

when someone is stupid enough to ask someone to prove a negative

Except I didn't. Your specious BS claim:

Quote:

nobody's come up with a truly transparent comparator (and likely never will)

I asked for evidence to support your BS claim of (all known) comparators (never mind I can only think of 2, the current AVA and a discontinued QSC. The really silly "all future ones" is too funny) are non-transparent.
You admit you have none. IOW, your claim is total unsupported evidence free BS. "Stupid"? Well, you decide ;-)

AJ's picture
Quote:

Again, find me a transparent A/B/X comparator system... maybe get Nelson Pass (or his pre-amp guy), Dan D'Augustino, and the guys at Schiit on it, they're all excellent engineers.

Though you could never comprehend it, that sir, is your burden, not mine ;-)

Corvaldt's picture

Actually the burden of proof is on you here. Because it is possible to create a non-transparent comparator in an actually infinite number of ways, your claim that creating a transparent comparator IS possible is the less likely to be true.

supamark's picture

but where is it? Also, as I said before, you can't prove a negative. I made the claim (and I'm far from the only one), you can choose to accept/believe my claim, you can try (possibly even succeed) in proving my claim wrong, or you can just ignore it because life is short and this shit ain't important.

AJ's picture
Quote:

I made the claim

Bingo!
..and then the hands waved frantically...because the evidence was zero ;-)

supamark's picture

I have my very own stalker, how pathetic of you.

rschryer's picture

But, AJ, here is where your argument doesn't hold water: Supamark is saying that a truly transparent comparator DOESN'T exist. With that in mind, if I were to tell you that your inability to prove that wood fairies DON'T exist means that they do, would you agree with me? If not, then the burden of proving that a truly transparent comparator does exist rests on your shoulders, because you are insinuating, by way of your challenge to supamark, that one does. Do you know of a truly transparent comparator? If not, then supamark wins by default.

AJ's picture
Quote:

Supamark is saying that a truly transparent comparator DOESN'T exist.

Exactly!
Now both of you are burdened with evidence for your fabricated imaginary assertions.
Let's see it....or more frantic hand waving.

rschryer's picture

I may be burdened with fabricated imaginary assertions, but yours isn't one of them.

AJ's picture

Of course you would have no comprehension of Argument from Ignorance,
but your false assumption that his non-transparent comparator claim is true, thus the burden shifts to me to provide evidence to disprove it...is exactly that.
No worries, we're in the age of hand waving "believe me", it's expected ;-)

SeanS's picture

Analog inputs stay analog.

From the manufacturers manual:
HGC™ is Benchmark's unique Hybrid Gain
Control™ system. The DAC3 combines active
analog gain control, passive low-impedance
attenuators, a 32-bit digital gain control, and
a servo-driven volume control.
All inputs are controlled by the rotary volume
control. This volume control moves in
response to commands from the remote
control. Analog inputs are never converted to
digital, and digital inputs never pass through
an analog potentiometer. Digital inputs are
precisely controlled in the 32-bit DSP system.
The DSP system preserves precise L/R
balance, and precise stereo imaging, while
avoiding any source of noise and distortion.

Sigh's picture

I too was shocked by this way of comparing the PS-Audio to the Benchmark. Wether you hear a difference or not doesn't even matter because you (and we) don't even know what it is that makes the sound of one chain or the other, (The DAC sections? The Cable? The differences between the analog or digital volume control?). Read what Mytek has to say about the effect of digital and analog volume controls in their Brooklyn. I was eagerly waiting for this review since I am a big fan of all Benchmark DACs.
The lack of rigor and the seven pages of blabla are almost offensive to the companies that put years of hard work into their products and put them in your hands. Some of the comments could have been phrased more diplomatically but the nonsense one reads in Stereophile, 6moons, DAR... the contradictions, the approximations, the lack of methodology are infuriating because your opinions greatly influence the success of these products. The spectacular measurements are useful. But what exactly is there to learn from what you wrote?

pma's picture

Nice review Jim and also thanks to John Atkinson for a valuable set of measurements, as always. We can see that the review has initiated quite strong reactions of a prominent high-end designer as it obviously targets very good objective parameters that tend to be overlooked and replaced by rather mystic beliefs.

I would like to encourage Stereophile team in doing more controlled tests. Though there may be objections to AB switch box transparencies, the same applies to 'high-end' components and usually at much higher degree. Please continue the good job and do not let you make disappointed from the offensive comments.

Last but not least, the switch box transparency might be well evaluated by SYS-2722 redaction system.

arve's picture

@John Atkinson: Have you considered adding a test to see how gracefully DACs handle intersample overs? While it's possible to create waveforms with a True Peak value with an arbitrary value above 0 dBFS by approaching Nyquist, here is one that generates a peak that's +3dBFS - the below example is done in Audacity, but any sample editor with similar functionality should suffice.

  1. Generate a 0 dBFS sine at 1/2 Nyquist, so each cycle is represented by the four (floating-point) values 0,1,0,-1
  2. Use "Change speed" in the effects menu, and set it to 0.5.
  3. Zoom way in on the start of the waveform, so individual samples become visible. Now, chop off the first sample
  4. Again, use "Change speed" and set the speed to 2.0
  5. Add a short fade-in and fade-out to the beginning and end of the generated tone
  6. (Peak) Normalize the track to 0 dBFS

This has the overall effect of introducing a 45 degree phase shift in the period of the sample, and the peak of each sinusoid will now occur halfway between the two equal-valued samples, causing a true peak that's ~3dB above the sample values in the file. A DAC with less than graceful handling of intersample peaks should thus exhibit clipping of the sinusoid, while a DAC like the Benchmark should still show a proper sine wave.

barrows's picture

arve, love you're idea here. I think testing for this problem would be great and I would love to see it in Stereophile measurements.
When we at Sonore were developing the OSF used in our USB interface we encountered this problem, and adjusted our filter parameters appropriately. With a lot of current music releases featuring full scale 0 dB signals, or even clipping (what are these recording engineers doing) it is important for DAC designers to take this problem into account.
Do note, as I recall, Ted Smith (main designer of the PS Audio DAC referenced here) has noted that the DS DAC uses 6 dB of headroom in its DSP stage. So it is not true that Benchmark is the only company addressing this problem.

arve's picture

@barrows : As a general comment (I haven't looked into the PS Audio DAC), but merely a general observation: DSP headroom isn't necessarily the same as DAC headroom - when processing a signal, you use headroom to prevent downright clipping of the processed signal, but if that DSP outputs digitally to a DAC, but if the DSP itself is _capable_ of outputing a sample value with an amplitude of "max value" for the DAC, there still needs to be headroom, as the intersample peaks occur _between_ the samples.

As an example: In my second setup, I use a little homegrown box running on shairport-sync, with full room correction using BruteFIR. Prior to DSP, that system uses (for the specific filters I use) -9.2 dB of attenuation to guarantee against digital clipping, it provides no guarantee against inter-sample overs, so that job has to be left to the DAC, as the DSP can still output 0dBFS.

But as said: I haven't tried to delve into precisely what Ted Smith and PS Audio is doing with a DSP in their DAC - so they might very well have tackled the problem

JimAustin's picture

See footnote 4 in the review.

Best,
Jim

JimAustin's picture

I've been having trouble posting, or I would have replied sooner.

I think what makes sense is to do some preliminary tests, see how common this problem is. I'll fool around with this a little, test some DACs I have on hand.

I did eventually manage to get good data contrasting the DAC1 and DAC3 response to a test signal similar to the one you describe. It's quite dramatic. Red is the DAC1; orange is the DAC3.

InersampleOvers

Jim

arve's picture

@JimAustin: That's quite dramatic, and something I'd readily take to be audible. Now on to convince JA to include that 11025 measurement as one of his standard measurements.

JimAustin's picture

this is a test signal. Absolutely audible. It's less clear how often intersample overs affect the sound of music--although if Benchmark's John Siau is to be believed, it's very often. (In his Manufacturer's Response, he agreed that the DAC1 sounds brighter than the DAC3--and attributed the difference entirely to intersample overs.) If you haven't already, read the essays on Benchmark's site:

https://benchmarkmedia.com/blogs/application_notes/intersample-overs-in-...

https://benchmarkmedia.com/blogs/application_notes/13545433-audio-that-g...

https://benchmarkmedia.com/blogs/application_notes/13740017-why-audio-go...

TBD, IMO, is how common this problem is in recent/new DACs. JA and no one else will decide what to add to regular testing, but before that's even considered, I think it makes sense to do a little preliminary investigating.

Best,
Jim

supamark's picture

probably are not audible - when I was recording rock bands in the early 90's, I used an Aphex Dominator to prevent overs (it's an analog brick wall limiter). I'd mix a track, check the Sony DAT and verify no overs (I set the limiter to kick in at 0.5dB before 0 using a test tone) and would still occasionally see overs on playback but they were inaudible (Genelec 1031A monitoring). My aim was usually to light it up a few times during the mix to make sure I maxed out the s/n but never used it as a mix buss compressor like some did.

JimAustin's picture

But take a look at John Siau's Manufacturer's Comment. John's not one to claim audibility without evidence. He attributes the DAC1's relative brightness to overs. I intend to do some more listening myself. I did some for this review--but not after I confirmed I was actually getting them--i.e., that the server was sending bit-perfect output. As the review documents, I had a little problem there.

Best,
Jim

supamark's picture

and the D/A converters were nowhere near as good as they are now. I'm also talking about a few overs per song in rock music and generally on a drum strike. I can see it being a much bigger problem with today's mastering techniques, but then again with the aggressive use of hard limiting today how would you hear it separate from all the clipping?

You probably listen to a lot more classical/acoustic music than I do, and I bet it's a lot more audible in that context. I spent a couple years recording classical (always with just a stereo pair of Neumann mics), but 2 channel stereo just can't capture that surrounded by luscious reverb sound of being in the hall and always dissapoints me.

AJ's picture

And you believe yourself to be the arbiter of all what is and isn't audible?
Comedy gold ;-)

supamark's picture

No? Then STFU troll.

arve's picture

… let me show you a pathological case

https://imgur.com/ACqrucC

That's the _additional_ inter-sample clipping introduced into a track (Muse - Map of the Problematique). Intersample peaks below 0.1 dB are ignored. Every vertical red line is an inter-sample over.

While AB(/X) testing this takes some effort, because you need a controlled and calibrated testing setup with precisely matched levels, I would be _very_ surprised if that track didn't reveal differences between a DAC with inter-sample headroom and one without.

supamark's picture

that is some of the worst brick wall limiting I've ever seen - the mastering engineer (Howie Weinberg, who really should know better) should be taken out behind the woodshed and beaten.

arve's picture

Not being a reviewer with access to expensive gear: I tried this on various DACs I have lying around. All but of them will clip the signal,, showing a distortion spectrum similar to yours.

Edited: In re-testing - all of them exhibit clipping when digital volume is set to beyond -3 dBFS

Bubbamike's picture
Quote:

This is the kind of scientistic nonsense that's so common this world--a just-so story (ad-hoc fallacy) that attempts to explain subjective impressions via nice stories or casually observed phenomena while never subjecting those claims to serious tests.

Waiting for these kind of tests for MQA that you and the others at Stereophile seem to have swooned over without ever conducting any scientifically serious tests.

John Atkinson's picture
Bubbamike wrote:
Waiting for these kind of tests for MQA that you and the others at Stereophile seem to have swooned over without ever conducting any scientifically serious tests.

Starting with the January 2018 issue, we are publishing a series of articles examining MQA's claims one-by-one.

But "swoon" - I respectfully suggest your comprehension has been a wee bit colored by what you might have read from ill-informed hot heads on the InterWebs. :-).

John Atkinson
Editor, Stereophile

Bubbamike's picture

Sorry, but I think swoon is a fair description. Not the first time, even you admitted, just an issue or two ago that you and the magazine over empathized the effect of HDCD. I think this MQA mania is of the same enthusiasm. I have never seen a negative word applied to MQA from the staff of Stereophile. Perhaps I'll be surprised by your January issue. I hope so.

tonykaz's picture

There aren't any, are there ?

Sure, there are negatives. Everything has a negative of some sort even if it's quite minor.

MQA's prominent negative is that so many other things have been heavily promoted but not achieved universal acceptance that dubious consumers are remaining skeptical.

After all, isn't Vinyl 'still' the finest music Format for the group of Audiophiles that began during vinyl's Big Era? A good many of these folk remain Digital Deniers although even 'that' position is getting difficult since 2009 or so ( when HP of TAS was started blessing digital stuff ).

Another negative is Bob Stuart and Meridian. I've loved Meridian and Boothroyd Suart since the 1980s but I feel very much alone in this.

MQA being British is not a good thing for us Yanks, why couldn't one of our sharpies figure this out, someone like Edison or Einstein. Dam it, why another Brit thing?. That Linn guy and that LP12 was probably all the Brit we could take. We're the RedWhiteBlue Team and we deserve to win one, don't we?

Maybe the worst part of all this MQA business is that 'our' Warner was the first Record Label to get on-board. phew. Now it seems everyone is in a hurry to do MQA stuff. ( except some Neanderthal outfits that steadfastly refuse to advance into this 21st Century, I won't mention any names except Shit. There are a few more. )

The MQA Positive for those who remain negatives: Noboby has to buy it and nobody will hold it against you, MQA is just better RedBook. As far as I can tell.

Besides, if a person can't hear any MQA difference it only means that they have lesser gear or hearing ( like an old geezer ).

Ancient Tony in Michigan

Bubbamike's picture

You realize that MQA starts out with a high resolution file? It isn't meant to replace Red Book but to allow the streaming of Hi Res files over the internet with reduced bandwidth. Well among other issues, such as DRM and loss of data. But that you didn't know that is an indictment of Stereophile's coverage of MQA. If you look around you'll find explanations and critiques of the method, as well as Bruno Putzey's recent criticism of the lack of reliable tests of MQA.

tonykaz's picture

I accept, MQA is high resolution transmitted via RedBook.

Loss of Data, hmm, is that true?, maybe folks like Bruno Putzey will explain these things or someone else.

So far, MQA seems to be blessed but not universally adopted.

If I were making gear, I'd accept MQA.

As I stand today, I'm happy with RedBook and happy that I don't have to have a vast vinyl collection to enjoy music.

I feel like I'm winning and I'm not letting anyone take that away from me.

Tony in Michigan

arve's picture

I accept, MQA is high resolution transmitted via RedBook.

It's not. "Redbook" is entirely specific to data stored on an Audio CD, and governs all topics related to storing audio on that CD, including data structures and data encoding. The sample rate is only one such aspect. A file stored on a computer or audio streamed over a network can never be redbook.

But, the point you were trying to make was probably related to 16/44.1 audio being stored or streamed. Which is also entirely wrong for MQA. MQA uses a 24-bit container, rather than a 16-bit container.

Loss of Data, hmm, is that true?, maybe folks like Bruno Putzey will explain these things or someone else.

Yes, the data loss is true. John Siau has already done this dissemination of the format for you: https://benchmarkmedia.com/blogs/application_notes/163302855-is-mqa-doa

tonykaz's picture

Ok, I just had a brief look at that Benchmark Media report on "is mqa doa".

Firstly, thank you for pointing out that its a 24Bit container, didn't know ( or probably care ). Egads, can I accept or embrace 144db of dynamic range?, seems way too much ( even on a Battlefield Re-inactment )

Lossless? Sure, it looses the old file as it folds it up. I don't know what to think beyond that. Is it like flour stops being flour as the Pizza is made?

MQA is a just a Streaming System, isn't it? MQA is for the Record Company and the Streaming Company, it's not something built into the CD that we buy, as far as I can tell.

So, the Record Company & MQA devise a way to distribute their owned music to us 'Renters' of their convenient listening system.

We can create our own tiny SD memory cards and own our own Astel & Kern players and not bother with Streaming.

We can also collect vinyl and own vinyl playback gear.

We can have Tape Machines and buy Tape from Acoustic Sounds.

We seem to have a wide range of options.

MQA is just another option.

I don't see the reason for all the fuss.

Tony in Michigan

ps. as far as Streaming listeners are concerned, that Newspaper study showed how people had a hard time hearing 320 being different than hi-res.

Camilo's picture

Arve has proposed a very pertinent and concrete suggestion with regards to your test routine of D/A converters, and its merit and rationale very much deserves your consideration.

The suggested test would allow to recognize the innovation made by Benchmark with regards to intersample overs, and would allow to document and evidence the performance of other D/A converters in this respect.

It would shed more light into the differences between D/A converters, and thus provide a better resource of information for consumers, which is - I believe - the objective of carrying out measurements as well as the mission of a dedicated consumer product magazine.

Best,

Camilo

John Atkinson's picture
Camilo wrote:
Arve has proposed a very pertinent and concrete suggestion with regards to your test routine of D/A converters, and its merit and rationale very much deserves your consideration.

We used routinely to examine digital filter headroom, as you can see from reviews on this site that were originally published in the early 1990s. But I admit that was before the Loudness Wars, when CDs were mastered so that there were never consecutive samples at 0dBFS.

Modern digital audio workstations do calculate the waveform on the assumption that it would be processed by a typical digital filter and I have looked at some modern CDs. Inter-sample overs are rarer than many people believe, though I have found a few pathological examples.

But yes, I think I will start looking again at a DAC's digital filter overload characteristics.

John Atkinson
Editor, Stereophile

arve's picture

Inter-sample overs are rarer than many people believe, though I have found a few pathological examples.

This is a bit preliminary - as there are still non-music (and much quieter) data in the corpus (voice memos, audio books, podcasts and a couple of test tracks), but I have examined a corpus of 14411 tracks for intersample overs. Of those tracks, 6082 tracks reported true peaks below 0 dB, 1237 tracks has true peaks exactly at 0.0, and 7092 tracks have intersample overs. In other words, very roughly half of the tracks contain such overs.

I have not tried to do a more qualitative analysis of the overs, such as how many overs there are in a track, or counting whether some of them have overs that last more than one sample.

SeanS's picture

Hi John,

Eagerly anticipating the MQA article(s).

As an audiophile, I have to say I am not interested in the benefits of smaller file sizes for hi-res files--the origami, etc.

What I am really interested in is what, if anything, MQA can offer to better sound quality vs. a well implemented hi-res AD-DA conversion chain that doesn't use MQA. Like, for example, if you simultaneously created two 24/96 digital recordings of a jazz band using the same recording and playback equipment, with one recording having MQA enabled throughout the chain, and the other without, could you tell the difference? I would really appreciate your expert opinion coming from this perspective.

Thanks,
Sean

NeilS's picture

With all the expert technical scrutiny, even reverse-engineering that MQA has been subjected to on sites like computeraudiophile and Archimago this year, it's hard for me not to get the sense that MQA and its related claims are already technically very well understood by anyone who wants to know.

So when I read about Stereophile announcing at the end of October 2017 that it plans to run a series of articles beginning in 2018 on MQA to test its claims, it sure seems like a lot of effort, but it also seems to me like a lot of effort a lot too late.

Raving about/"swooning" over MQA before testing its claims may be a reminder that sometimes, like putting on socks and shoes, the order in which things are done matters.

tonykaz's picture

Paul McGowan of PS Audio is now doing a Question & Answer Series on YouTube.

Paul just answered the Question about different sounding DACs.

Egads, he's taking the time to answer all kinds of Audiophile questions.

This is well worth checking out.

He even tried to reason thru StillPoints.

Tony in Michigan

Camilo's picture

Paraphrasing the sentence that immediately links to your 5th footnote, I would say: "If you're REVIEWING 24-bit DACs and hi-rez downloads, you'd best get your noise level down to where you can hear at least some of that extra resolution, and that's harder than you might think".

What surprised - and outraged - some who read your review, is that you failed to hear a difference between the Benchmark DAC3 and the PS Audio Perfect Wave DAC, which leads to conclude that if there is a difference, it is not audible and doesn’t matter.

Although your 5th footnote hints a clue to why you - no matter how hard you tried - couldn't hear a difference between the Benchmark DAC3's 21 bit state of the art resolution, and the rather poor 17bit performance of the PS Audio Perfect Wave DAC, your review fails to take into account precisely that advice offered by your own footnote.

That advice obviously draws attention to the even poorer performance delivered by the PS monoblocks, acting as bottleneck when it comes to rendering the 21 bits of resolution of the Benchmark DAC and even the 17 bit performance of the PS Audio DAC, let alone their difference.

As Benchmark's John Siau wrote in a application note back in March 2014: "Anyone who thinks they can hear the difference between 16-bit and 24-bit digital audio through a "17-bit" power amplifier is fooling themselves."("What is high resolution audio? - part1" https://benchmarkmedia.com/blogs/application_notes/13174001-what-is-high...)

Nevertheless, the lenghty passage you dedicated to the efforts taken to hearing the difference between the Benchmark and PS Audio DACs, can suggest to some that this your flawed and misleading conclusion are not not merely innocuous omission or mistake. You appear to ignore your own advice, and completely disavow the fact that your monoblocks aren't capable of delivering even 16 bits to your speakers.

I think it is fair to criticize your review and to dismiss your conclusion regarding the audible difference between the Benchmark DAC and the PS Audio DAC. I also believe it is fair to demand that in the future, stereophile reviews take into account basic specifications - which I want to believe reviewers understand but apparently and consistently fail to apply.

I also believe it is impossible and unacceptable to excuse the obviously flawed attempt to establish a difference between the mentioned components and the consequently misleading conclusion, with a statement like: "Yes, listening is what Stereophile reviewers do." If people read reviews here and elsewhere, it is based on the expectation that they will be offered more than casual listening impressions that completely ignore basic science, let alone the specifications of the components reviewed.

It is in this particular case, quite obvious that you need an amplifier that matches and even exceeds the performance of a DAC to deliver the resolution the DAC has to offer to the speakers, yet that was blatantly ignored by the review. Worse, to reinforce the apparent ignorance - I will nevertheless refrain from concluding foul play or second intentions here - of the reviewer with regards to importance the specifications of the components used have for his review, he proceeds to write a lengthy passage showing what lengths he went to in order to hear a difference that the equipment he used is clearly unable to render. This is also not the only review to be found on Stereophile or Audiostream, with this exact same flaw.

This is poor work, and ultimately undermines the credibility of Stereophile, as well as the effort made by John Atkinson to measure equipment as a way to offer transparency and accountability with regard to manufacturer specifications and in many cases supplement the lack thereof.

I am in my mid fourties and, after thousands of hours behind a drumkit, exposure to loud environments and extended listening periods, cannot argue to hear extremely subtle differences. But even having a well trained ear as a musician, I would not be above relying on the due diligence of reading the specs and setting up valid review, audition or test conditions first, and before I make conclusions regarding the audible differences between two audio components or recordings.

I own a Benchmark DAC2 DX and have owned a DAC1. I had the chance to audition them both side by side with recordings that are known to have intersample overs, and I could very much hear the difference using my Sennheiser HD 800s.

I would hope for a clarifying response from you, with regards to taking into account the components used for a valid review of the performance of specific components. I would also like to see a response from John Atkinson with regards to the very pertinent suggestion made by arve - further up in the thread - with regards to introducing a test that takes into account intersample overs. Atkinson made no remarks to this excellent and concrete proposal, and instead threw in the infamous MQA topic, which distracted from arve’s suggestion.

Updating measurement routines as components introduce new improvements and features that need to be account ted for in order to do fair comparisons between components, would only benefit the objectivity of the measurements and reliability of what Stereophile publishes as such.

Benchmark has clearly separated itself from all other DAC manufacturers I have knowledge of, by effectively dealing with the audible downside of intersample overs, and has clearly introduced a substantial improvement to sound quality that sets new standards. This can only be acknowledged by introducing the corresponding measurement to account for this innovation and the – recording industry – problem of intersample overs, whose existence has been acknowledged and documented, as well as as put forward by this review.

Best,

C. Rodriguez

Robocop's picture

I have owned both the DAC1 and DAC2L over 15 years. This review actually tells me little I havn't already read from other reviewers and the Benchmark web site.

What I find most disconcerting is the DAC2 was the current reference DAC for Benchmark which I own.

Why was the DAC2 not compared directly to the new reference DAC3 in listening sessions?

Comparison with the DAC1 is obsolete, well surpassed by the DAC2.

"How much audible improvement do these changes add up to? "I'm quite certain that there should be no audible difference between a DAC2 and a DAC3 given a single pass through the converters," Siau wrote to me in an e-mail."

What does this mean from John Siau Benchmark? Is he saying no audible improvement over the DAC2?

If it sounds the same, well why bother!!!

I really want to know how much better is the DAC3 over the 2 to justify its increased purchase price.

There must be a sound improvement, the Sabre 9028 chip is alone a sonic upgrade over the 9018. This must be audible and not just measured.

It is at the end of the day all about the "SOUND" compared to live instruments in an acoustic space.

Robert

Sigh's picture

It seems that the 9028 is the new 9018 that might be discountinued soon, I can see why Benchmark made the change even if it offers only a minute improvement that isn't audible. Maybe they should have called it DAC2.2 (Although they did have a little update to the DAC2 called 2.2 already) or DAC2+ like Mytek with their Brooklyn+.

To compare the PS-Audio to the DAC3 in a set up that would have made some sort of sense one could run the analog-out of one DAC3 into the analog-in of another DAC3 :) Is the reviewer suggesting that this too would have sounded exactly like a single DAC3 on its digital volume control?

I've owned every Benchmark, DAC1, USB, PRE, HDR, DAC2 and changed more for the new features (USB, remote, second line in) than the negligible sound differences. The one thing I didn't like with the DAC1s was that they ran hot. With the DAC2 I feel like they reached a perfect product, great interface, hybrid gain control, runs cool, asynchronous USB.

It probably makes no sense to trade a DAC2 for a DAC3, yet at the same time it would make no sense for Benchmark to continue using a older chip.

I find that whatever my questions are, the best way to get them answered is to ask Benchmark themselves. They answer promptly and won't push a new product onto you.

Charles Hansen's picture

Once again, I am baffled by Benchmark, designer John Siau, and reviewer Jim Austin. The existence of intersample overs has been well known for over 20 years. I'm surprised that JA did not catch this much earlier.

For evidence of this fact, we only need to look at the datasheet for the once-popular Pacific Micronicss PMD-100 digital filter. In the datasheet is a an unambiguous statement:

The PMD-100 has a design attenuation of 1 dB to allow for filter overshoot on transients.

During the '90s when Robert Harley was technical editor of Stereophile, he would routinely show CD players with no internal headroom, and how they would clip the "ringing" (Gibbs phenomenon) on the tops of a 1kHz, 0dBFS square wave. In contrast were other players that used (for example) the PMD-100 digital filter (and many other designs) that provided headroom to prevent internal overload from intersample overs.

While this phenomenon is understood more clearly now, with a better understanding of the degenerate (worst possible) case, this is hardly some sort of "breakthrough" as Siau and Benchmark would have us believe. In fact it's more surprising to me that he was previously unaware of a well-known issue regarding digital audio playback.

Camilo's picture

Knowing the problem does not automatically guarantee there to be an immediate solution, and blaming those who finally solve a pervasive and known problem for not having solved it before and for taking credit for it, doesn't seem like the right place to aim your criticism at.

The solution for the audible artifacts of intersample overs that Benchmark came up with, is not something trivial, as it is an inherent flaw of D/A chips including the ES9018 and ES9028PRO used in the DAC2 and DAC3, respectively:

"Every D/A chip and SRC chip that we have tested here at Benchmark has an intersample clipping problem! To the best of our knowledge, no chip manufacturer has adequately addressed this problem. For this reason, virtually every audio device on the market has an intersample overload problem. This problem is most noticeable when playing 44.1 kHz sample rates." ("Intersample Overs in CD Recordings"-John Siau https://benchmarkmedia.com/blogs/application_notes/intersample-overs-in-...)

Benchmark had to develop and implemenmt a solution that is not an option provided by the D/A chip, but which addresses a common problem that originates in D/A chips:

"It is possible to build interpolators that will not clip or overload, but this is not being done by the D/A and SRC chip manufacturers. For this reason, Benchmark has moved some of the digital processing outside of the D/A chip. In the Benchmark DAC2 and DAC3 converters we have an external interpolator that has 3.5 dB of headroom above 0 dBFS. This means that the worst-case +3.01 dBFS intersample peaks can be processed without clipping. We also drive the ESS D/A converter chips at -3.5 dB so that no clipping will occur inside the ES9018 and ES9028PRO converter chips." ("Intersample Overs in CD Recordings"-John Siau https://benchmarkmedia.com/blogs/application_notes/intersample-overs-in-...)

It is not out of mere negligence that Benchmark, or other manufacturers who rely more on the D/A chips - and who in some cases even list the specs of D/A chips rather than the measured performance of their components, and thus not the result of their implementation of D/A chips - haven't offered an effective solution to the intersample overs problem before. Being Benchmark the ones who came up with it,
hardly makes them the target to blame for said negligence, let alone undeserving of the merit for effectively addressing the issue of intersample overs.

If anything, it is other manufacturers who have not yet implemented a similar solution who could deserve some criticism in this respect.

As to the developing and final implementation of a solution to intersample overs, it is debatable if - according to your appreciation - it took the industry too long. It is however not debatable that Benchmark's solution is very much a welcome breakthrough for the performance and sound quality of D/A converters, and one that clearly sets them apart from all other manufacturers.

SeanS's picture

Wouldn't Siau's solution have to sacrifice dynamic range in the digital domain? He says they then bump up the 3.5db in the analog domain to bring the level back up. Definitely not perfect. Sounds like your trading dynamic range on all input data in order to correct the consequences of "bad" input data. On the other hand, maybe the amount of dynamic range lost is less than the noise floor?
Sean

arve's picture

Due to thermal noise, no DAC can achieve the full 24 bits of dynamic range inherent to such program material - at _best_ the analog outputs of a DAC will offer a bit above 21 bits of signal, with the three lowest bit drowning in the thermal noise. By adding 3.5 dB of attenuation to the signal, Benchmark is throwing away approximately the lowest half bit of the signal - this is signal that was already lost because analog electronics won't let us do better.

The loss in level is made up for through analog means adding the 3.5 dB of gain back after conversion. The risk here is that adding this gain causes the noise floor to rise, and thus lose dynamic range. However, as seen from JA's measurements, this is not the case with the Benchmark: The analog outputs have class-leading state-of-the art performance with regards to the noise floor.

SeanS's picture

Arve, the content of your message seems to elaborate on the thoughts I was having when I posted.

Throwing away the last half bit of the signal = throwing away dynamic range in the digital domain.

However, the low level thrown away was already lost because of the physical limitations on analog electronics.... = it is below the noise floor.

I was hoping to get opinions on the solution.

Sean

nikolaos's picture

First of all I'm a huge fan of Ayre and had several components from you and always enjoy whenever I hear Ayre gear. I'm also a huge fan of Stereophile and consider the mag.. as very serious. I will also say I enjoyed reading the review of DAC 3.

Regarding your comments (sorry I'm not english so forgive me for errors in the writing) I had to reply regarding things that where said.

I had lot's of equipment and tested a lot and my conclusion is that whatever the inputs or cables has to say to what I'm hearing does not even come close to other factors like room, components and the ultimate headroom in the system itself.
I had some quite expensive cables and I can for sure not say some where better than others even if the price difference was huge. My concusion is that it is so minor differences in cables that you will gain much more elsewhere in the system. I have heard amazing systems with cables that where the cheapest and I heard the worst systems with the most expensive cables.
I had simelar experience as the reviewer regarding hum with unshield Nordost cables but as Charlie say as they are balanced it can not happen. I guess the system was not truly balanced.
I agree that the reviewer or the editor maybe should have picked that up.
Regarding Benchmark it is well know that they have made great products for some time and as J.A show in the meassurements of this DAC it is a well designed product. I don't understand why a DAC should cost as much as some charge. AYRE used to have their top of the line DAC in around the same price range as Benchmark. You say Mr. Hansen that the DAC 3 is made of cheap components and it seams like you consider Benchmark bellow average quality.

I don't buy to much into all that voodoo anymore about lot's of the claims in the high-end industry. Not to say that from time to time some come up with new and better solutions.
Audio is to me much more about measurements and real world facts. Even if the reviewer did something not considered right by connecting one DAC into the other I'm not so sure it would be so easy to hear coloring of the sound. It would for sure act as a filter but so does a preamp.

As said I'm not English and maybe I'm stupid but I did not get the problem you had with how the reviewer explained how he connected the output of the DACs to the preamp.

It is a little industry. Let's enjoy the music and be friends. Everytime I meet someone with the same hobby it is always a pleasure. In a forum things are very unpersonal and I know people would not act like this when met for real.

Pages

Benchmark DAC3 HGC D/A preamplifier-headphone amplifier Associated Equipment

Thu, 10/26/2017

COMMENTS
tonykaz's picture

NwAvGuy was describing the Benchmark and using it to evaluate his own DAC designs.

Mark Waldrip relies on Benchmark gear

I've heard of a good many Pro Audio folks tout Benchmark as being the Benchmark.

I've been trying DACs for some years now, I can't discover any advantage behind some of the super pricy DACs.

I've blamed my hearing, to the point of having my hearing evaluated by Audiologists at the University of Michigan. My hearing tapers off above 8k but I can still hear significant differences in 12AU7 preamp tubes. I still can't hear greatness in super expensive DACs.

Those that can hear significant improvements in Super Expensive DACs are living with the Audiophile Curse. ( the King's New Suit Curse )

As an Engineer, I've demonstrated Zip Lamp Cord vs. Monster Speaker Cable vs. Bruce Brisson's MH-750 to astonished Engineers.

Back in 2015 Tyll had a headphone gear Shootout where Stax, Sennheiser, and Audeze were the finalists. He also had the finest DACs including the highly touted Antelope. Nobody could detect any DAC performance advantage.

Then there's Chord and their Field Programable Gate Array devices which I suspect they've designed to sound better but probably not sound accurate ( like Tubes that make music nicer sounding ).

So, go for Accuracy with Benchmark or better sounding with Chord.

Nice reporting, sir, I think.

Tony in Michigan

barrows's picture

I would suggest that there is not much evidence to support your assertion that Chord DACs are not designed to sound accurate, take a look at their measurements here at stereophile.com...

On this review in general, despite the lack of difference heard vs the DS, it is interesting to note that Jim found an immediately apparent difference to the Benchmark DAC1, subjectively speaking...

Staxguy's picture

Accurate? Chord DAC's can't even handle (do) 24-bit. Even their best DAC, the Chord DAVE can't even do 24-bit. 21-bit, that's it!

Their 17-order noise shaper can do 350 dB DNR, but that's internal aah only.

It's taken forever for this review to come out. Back when we (I) was recommending the Benchmark DAC1 HGC in my magazine as a recommended component, The Absolute Sound was only recommending DACs with sub-par 16-bit performance and not even mentioning it, even though it had came out.

Well, it's great to see this review. I might buy the Rogue Audio RH-5 as a headphone amplifier, reviewed just before this one, which likely functions worse (no better) than the Aurorasound HEDA, because it looks good on the desk!

People like benchmark, and good for/on them. It's got a great summing DAC (multiple DAC) design.

I'd rather have the GTE Trinity DAC (the one with the actual triangles not the newer ahh one), but that's for myspace!

barrows's picture

Your reply is kind of in error. Here is a quote from JA's measurements on this Benchmark:

"...the increase in bit depth dropped the noise floor by more than 30dB (fig.5), indicating that the Benchmark's resolution is at least 21 bits. This is as good as a DAC can currently get!"

There is no DAC of which I am currently aware that achieves better than 21 bit resolution at its analog outputs. The resolution measured here for the Benchmark is at the same level as Chord DAVE (look it up on this site). When DAC makers say they are 24 bits, or 32 bits, that is what level the DAC operates at in the digital domain, but no DAC achieves that resolution at its analog outputs.

I am curious, do you have any references for measurements of the Trinity? I would expect them to be way worse than that of the Benchmark, not saying anything against the Trinity as far as sound quality is concerned though, just, it is likely not nearly as "accurate" being an R2R DAC...

Sal1950's picture

"There's a danger of being misled, of repeating the same mistakes again and again, of spending way too much money on things of little value. My point is that, as a hobby, industry, and avocation, we may have shifted too far toward the subjectivist side."

What a breath of fresh air in these pages! When I first read this review in my subscriber copy my jaw hit the floor. Thank you Jim, we can only hope as audiophiles dedicated to the accurate reporting, that blind listening sessions become much more widespread and prevalent in the pages of Stereophile. Sighted listening is much too fallible to be the sole basis for accurate evaluations.

mrkaic's picture

I hope this year, this month, and this review mark the beginning of the end of the tyranny of subjectivists.

ChrisS's picture

...listening/reviewing of this component was done blinded.

He did it the way we all do.

He listens.

He thinks about it.

He writes about what he hears.

That's what they all do at Stereophile.

ChrisS's picture

"Yes. Listening is what Stereophile reviewers do."

mrkaic's picture

There is only one question in audio and it is rhetorical -- why buy anything but Benchmark?

They are the best, nothing comes close.

Charles Hansen's picture

You claim to have a PhD and you make the most egregious beginner's mistakes imaginable. It is laughable that even Streophile would publish this garbage.

First you start of the article with a story about "hum and buzz" from "poor quality control" in the interconnects. If you knew anything at all about balanced circuitry and how it works, you would realize that a truly balanced source sending a signal to a truly balanced downstream device wouldn't make a bit of "hum and buzz" if the cables were shielded or not.

But since you clearly have no idea about anything, you start with the wrong assumption and leap to an incorrect conclusion. [Flame deleted by John Atkinson]

Then you try to "compare" the Benchmark against your PS Audio. And although you could "easily" hear differences between the Benchmarks 1 and 3, you could hear no difference whatsoever between the Benchmark 3 and the PS Audio. Why don't you look at your ridiculous methodology before jumping to more false conclusions?

First you said you connected "sent the output of the two DACs to different channels of my PS Audio BHK Signature preamp". WTF? So you sent the left channel of one DAC to one input and the right channel of the other DAC to the other input? Or do you not know the difference between a "channel" and an "input"? At this point you are looking beyond unprofessional, beyond amateurish, and all the way to "knowing just enough to be dangerous". Or does your preamp have only one input? Or does it have more than one but the inputs were full, and you were too lazy to disconnect them?

Anybody serious would know that the only FAIR way to do this type of test is to not only use the same input on the preamp (different inputs will exhibit different amounts of break-in, as the trace on the PCB needs a signal to dissipate the residual electrostatic charge after it is tested with many kilovolts for UL compliance), but to also use the exact same pair of interconnects as they need to be not only the same brand and model, but also have the same amount of break-in time.

And finally you lose all rationality. (We may have to send the men in white coats to save you from harming yourself.) You send the analog output of the PS Audio DAC into the analog input of the Benchmark DAC. At the very least this signal is now going through the analog circuitry and volume control of the Benchmark, which is clearly coloring the sound if it sounds "identical" to you - either that or your hearing ability sucks or your playback system sucks. And if (like the version 2) the version 3 also has a "hybrid digital/analog" volume control then the volume control wouldn't even be active unless the analog input were being digitized by whatever cheap A/D chip Benchmark is using with their cheap IC op-amp based analog circuitry. More cheap crap to color the sound of the PS Audio, and you can't figure ANY of this out...

Finally it is clear that you cannot even add 2 + 2 to equal the correct answer. First you claim that the Benchmark "moves interpolation off-chip" to eliminate problems with "intersample overs", but that the latest version sounds better because of "lower passband ripple, facilitated by the new chip's superior filter choices". So which is it, Mr. PhD? Does it sound better because the processing is done off-chip or does it sound better because the on-chip processing has lower passband ripple? You can't have it both ways.

[Flame deleted by John Atkinson]

You are the one that makes a joke out of the high end, not the people who make claims you can't understand - you can't even understand the simplest of claims, as this so-called "review" clearly demonstrates.

AJ's picture
Quote:

Anybody serious would know that the only FAIR way to do this type of test is to not only use the same input on the preamp (different inputs will exhibit different amounts of break-in, as the trace on the PCB needs a signal to dissipate the residual electrostatic charge after it is tested with many kilovolts for UL compliance), but to also use the exact same pair of interconnects as they need to be not only the same brand and model, but also have the same amount of break-in time.
And finally you lose all rationality.

Can't make this stuff up ;-).
Charles, your rational scientific evidence of this audible "break-in" outside the imaginations inside your head?
TIA

mrkaic's picture

It boggles the mind that someone who believes in cable break-in has the chutzpah to write about electronics. It really does.

But we can learn here, for sure. In the words of the previous occupant of the White House— it is a teachable moment—sans beer, in this particular case.

It is tempting to get angry at the omni-present neglect of science education and fume about the ignorance of individuals who believe in cable break-in, directional fuses etc. But it is more fun is to realize that those esteemed gentlemen, who believe, for example, in cable break-in, provide some free comic relief in these trying times.

So, don’t get mad, get entertained.

Anton's picture

I'd hate to see how worked up you get over real world issues.

barrows's picture

Jim. I too was disturbed by some of your methodology in your comparisons, and dismissed your results because of my concerns. While I may not go as far as Charlie, please note that I felt disturbed enough to dismiss the results.
I would suggest that if you want to make quick A/B comparisons of source components in future reviews, you should have a balanced switch box made, who's only function is to take two balanced inputs, and switch them to a single balanced output. I would also suggest that this be wired the AES way for consistency (XLR pin 1 to ground) and that the signal path internally be identical for both inputs, same length wiring, etc. A high quality switch box like this should be a reasonable expense for any any reviewer who is interested in making quick A/B comparisons of source components.
Also of import would be keeping everything else the same, same cabling, same cabling lengths, etc.
In the case of comparing these two DACs (DS and DAC 3) such a methodology could then go direct into the amps, removing any additional components (preamp) and maximizing the apparent sonic differences. You should also volume match the DACs by measuring the output voltage with a test signal (like that available on stereophile test CDs) without very accurate level matching any such comparisons are mostly meaningless.

supamark's picture

won't work as you think. You're going to add two add'l sets of connections (in/out of the box) which will color the sound. Not every component has balanced outs, and a "dirty" little industry secret is that a majority of XLR outs are derived from the unbalanced RCA outs since few components are actually fully balanced internally. Okay, not really a secret since the few mfg's who make components that are fully balanced from input to output will make sure you know about it.

You will also still need a volume control since not everything being reviewed will have a variable output (and reducing volume in the digital domain adds its own.... issues). Jim's method of using two line inputs of his high quality preamp (that I'mma assume Jim knows the sound of quite well) is a much better methodology. The only way I could think of to improve his methodology would be for him to have someone else connect the two devices and make sure he doesn't know which input is which device.

Let me tell ya another little secret that every recording engineer knows about critical listening (this has happened to EVERY recording engineer at least once): You can imagine differences in sound that are not actually there. Every engineer has at some point reached for the EQ, started adjusting it and heard the sound change, and I mean literally hearing the changes being made to the sound.... but there's a problem - you never actually engaged the EQ (there's an on/off relay switch on every mixing console to switch the EQ into the signal path). It was all in your mind. This, to me, makes two very different points:

1. blind A/B/X testing really is the only fully valid method, but unfortunately nobody's come up with a truly transparent comparator (and likely never will).

2. those expensive cables from like Nordost et.al. are mostly snake oil. A much better system would be 100% copper all the way - internal component wiring (already happens), line level plugs/jacks, speaker connections all 100% copper (or silver/gold/whatever but must all be the same metal and the same purity/alloy). When the electrical signal is traveling along and the medium (metal) changes the signal will change. Were I a high end mfg, I would make all my connectors copper and provide high quality all copper cables at no add'l charge (because, seriously, the mark-up on cables is crazy - 30 years ago I picked up 40' of Kimber Cable at dealer cost, $1 per foot).

barrows's picture

won't work as you think. You're going to add two add'l sets of connections (in/out of the box) which will color the sound. Not every component has balanced outs, and a "dirty" little industry secret is that a majority of XLR outs are derived from the unbalanced RCA outs since few components are actually fully balanced internally. Okay, not really a secret since the few mfg's who make components that are fully balanced from input to output will make sure you know about it.
Read more at https://www.stereophile.com/content/benchmark-dac3-hgc-da-preamplifier-h..."

Only one xtra pair of interconnects, and this is not a problem, as long as they are the same, the difference will still be the same. Same thing with the switch box itself, the entire point is to give the same signal path for each DUT, the result being that one will only hear the difference. This set up, given decent construction and good interconnects will be transparent enough to hear any actual difference.
Additionally, while your contention that components are generally pseudo balanced may have been the case long ago, it is not common now-for example, the DACs mentioned in Jim's review are true balanced, as are his pre amp and amp.

And if you think Nordost cabling is snake oil, you are either deaf, or have not listened to it.

supamark's picture

It's not the 1 extra pair of interconnects (the extra wire is essentially meaningless*), it's the extra set of connections - jacks and plugs (along with the circuitry inside the box, including gain matching circuits, even if they're just variable resistors). You going with a relay or mechanical switch? Both degrade the sound in different ways. Adding that crap in will make it harder to hear any differences because you just added noise and made the signal path more complex. That is much worse than using two inputs of the same high quality preamp.

Most home audio gear is not fully balanced inside (nor should it be). Its adoption rate is higher in high-end home audio, but honestly it's not really necessary in most home environments. Pro audio, where there's much longer cable runs and a lot more electrical noise, is where it's really useful.

*when I was recording Austin Symphony Orchestra in Bass Concert Hall at The University of Texas in the early 90's, the mic cable run - from the stereo pair of Neumann U87 mic's through underground conduits below a couple other buildings into our machine room patch bay and into our Harrison Series 10 console was at least a quarter mile, probably close to 2,000 feet. The sound was not dull or even particularly weak. The U87's large presence boost probably helped but as I said, a couple feet of line level cable ain't gonna mean squat.

btw, if you think Nordost or anyone's cable can magically transform the sound that is otherwise running through traces on a f'ing circuit board inside each piece of your analog gear as it moves from cap to resistor to etc, you have a few things to learn about science (esp. physics). I mean, holy sh!t, they'll sell you little teflon and wood teepees for your speaker cables (for a pretty penny, 'natch). It's really no different than the Tice Clock or coloring the edges of your CD's with a green marker. Maybe Nordost has proprietary connectors that do a truly excellent job of passing signal with minimal alteration - great, that's a real upgrade and can be audible... but that wire between those connectors for which they're charging you several grand more than JA's preamp? that stuff is snake oil.

Buy a high quality power conditioner instead of fancy cables, it'll make a real audible difference and cost a lot less money.

barrows's picture

Are you nuts? I am talking about a passive switch box here, no active circuitry. Just two sets of XLR input jacks, some wire, a high quality switch (lets say a shallco) and two output jacks. That is all. No resistors, no circuitry of any any kind. Will this change the sound, perhaps a tiny, tiny bit, but not enough to obscure differences between the two DUTs, as the change will be identical to both products. A passive box like this will have much less influence on sound than a fully active preamp (which we eliminate from the chain with this testing method) which includes the wiring and switch, but also adds a power supply (noise source), transistors, resistors, etc. We do not need any circuitry, both DACs are designed to run direct to power amps, and we match levels via the DAC VC.

On cables, I never suggested that a cable can "magically transform sound"! Please do not put words in my mouth, neither did I make any comment re audio cable pricing. I just said that if you think Nordost cabling is snake oil, you are either deaf or have not listened to it, and I stand by that. To be more detailed: compare directly a Nordost cable to Mogami, and you will hear a significant difference. But the cable thing is OT here, so lets leave it for another place.

supamark's picture

Those added connections will in fact degrade the sound. You are adding 2 additional connections to each channel (XLR's, which really aren't as firm as RCA connectors - they wiggle a bit), each will degrade the sound. Also, since not every DAC has balanced outs, you'll want another box with RCA's (or just go unbalanced in the first place - the main advantage with balanced is if you have RF issues in your home or really long cable runs).

If you believe so strongly that wire has a sound, just use the same brand/length/termination/model for both and they'll be the same... it's cable, not 9' concert grand pianos (which actually do all sound different because they're handmade and no two are identical but brands do have "house" sounds - Hamburg Steinways are my favorite btw, and Baldwins are popular in rock because they're a little more mid-heavy with less sparkle than NY or Hamburg Steinways and therefore easier to cut through a dense mix).

Not every DAC has a volume control, and even for those that do you'll need a master volume control if you want adjust volume while keeping the levels matched - unless you think no volume adjustments should be made? Besides, the fixed output *should* always be superior (digital volume controls necessarily change the bits, and not for the better).

Also, what if the output impedences of the DACs are not close? going direct into a tube amp could cause differences with freq. response.

So, yeah, going into a high quality preamp is both simpler and more likely to yield consistent results with a much wider variety of DACs.

AJ's picture
Quote:

You going with a relay or mechanical switch? Both degrade the sound in different ways. Adding that crap in will make it harder to hear any differences because you just added noise and made the signal path more complex. That is much worse than using two inputs of the same high quality preamp.

You just need to wave your hands around as you say this stuff, then finish with "believe me" ;-)

supamark's picture

I'm livin' in your tiny little head, you can't quit me. it's really sad, really really sad.

AJ's picture
Quote:

but unfortunately nobody's come up with a truly transparent comparator (and likely never will).

Your factual evidence please.

supamark's picture

that's just stupid. Why don't you go out and find/build one and prove me wrong?

AJ's picture
Quote:

Why don't you go out and find/build one and prove me wrong?

Because folks with a modicum of intelligence know where Burden of Proof lies ;-).
Ok, so you admit to having zero evidence for your specious claim.
The question was rhetorical.

supamark's picture

when someone is stupid enough to ask someone to prove a negative, it simply proves that they're just stupid.

Again, find me a transparent A/B/X comparator system... maybe get Nelson Pass (or his pre-amp guy), Dan D'Augustino, and the guys at Schiit on it, they're all excellent engineers.

AJ's picture
Quote:

when someone is stupid enough to ask someone to prove a negative

Except I didn't. Your specious BS claim:

Quote:

nobody's come up with a truly transparent comparator (and likely never will)

I asked for evidence to support your BS claim of (all known) comparators (never mind I can only think of 2, the current AVA and a discontinued QSC. The really silly "all future ones" is too funny) are non-transparent.
You admit you have none. IOW, your claim is total unsupported evidence free BS. "Stupid"? Well, you decide ;-)

AJ's picture
Quote:

Again, find me a transparent A/B/X comparator system... maybe get Nelson Pass (or his pre-amp guy), Dan D'Augustino, and the guys at Schiit on it, they're all excellent engineers.

Though you could never comprehend it, that sir, is your burden, not mine ;-)

Corvaldt's picture

Actually the burden of proof is on you here. Because it is possible to create a non-transparent comparator in an actually infinite number of ways, your claim that creating a transparent comparator IS possible is the less likely to be true.

supamark's picture

but where is it? Also, as I said before, you can't prove a negative. I made the claim (and I'm far from the only one), you can choose to accept/believe my claim, you can try (possibly even succeed) in proving my claim wrong, or you can just ignore it because life is short and this shit ain't important.

AJ's picture
Quote:

I made the claim

Bingo!
..and then the hands waved frantically...because the evidence was zero ;-)

supamark's picture

I have my very own stalker, how pathetic of you.

rschryer's picture

But, AJ, here is where your argument doesn't hold water: Supamark is saying that a truly transparent comparator DOESN'T exist. With that in mind, if I were to tell you that your inability to prove that wood fairies DON'T exist means that they do, would you agree with me? If not, then the burden of proving that a truly transparent comparator does exist rests on your shoulders, because you are insinuating, by way of your challenge to supamark, that one does. Do you know of a truly transparent comparator? If not, then supamark wins by default.

AJ's picture
Quote:

Supamark is saying that a truly transparent comparator DOESN'T exist.

Exactly!
Now both of you are burdened with evidence for your fabricated imaginary assertions.
Let's see it....or more frantic hand waving.

rschryer's picture

I may be burdened with fabricated imaginary assertions, but yours isn't one of them.

AJ's picture

Of course you would have no comprehension of Argument from Ignorance,
but your false assumption that his non-transparent comparator claim is true, thus the burden shifts to me to provide evidence to disprove it...is exactly that.
No worries, we're in the age of hand waving "believe me", it's expected ;-)

SeanS's picture

Analog inputs stay analog.

From the manufacturers manual:
HGC™ is Benchmark's unique Hybrid Gain
Control™ system. The DAC3 combines active
analog gain control, passive low-impedance
attenuators, a 32-bit digital gain control, and
a servo-driven volume control.
All inputs are controlled by the rotary volume
control. This volume control moves in
response to commands from the remote
control. Analog inputs are never converted to
digital, and digital inputs never pass through
an analog potentiometer. Digital inputs are
precisely controlled in the 32-bit DSP system.
The DSP system preserves precise L/R
balance, and precise stereo imaging, while
avoiding any source of noise and distortion.

Sigh's picture

I too was shocked by this way of comparing the PS-Audio to the Benchmark. Wether you hear a difference or not doesn't even matter because you (and we) don't even know what it is that makes the sound of one chain or the other, (The DAC sections? The Cable? The differences between the analog or digital volume control?). Read what Mytek has to say about the effect of digital and analog volume controls in their Brooklyn. I was eagerly waiting for this review since I am a big fan of all Benchmark DACs.
The lack of rigor and the seven pages of blabla are almost offensive to the companies that put years of hard work into their products and put them in your hands. Some of the comments could have been phrased more diplomatically but the nonsense one reads in Stereophile, 6moons, DAR... the contradictions, the approximations, the lack of methodology are infuriating because your opinions greatly influence the success of these products. The spectacular measurements are useful. But what exactly is there to learn from what you wrote?

pma's picture

Nice review Jim and also thanks to John Atkinson for a valuable set of measurements, as always. We can see that the review has initiated quite strong reactions of a prominent high-end designer as it obviously targets very good objective parameters that tend to be overlooked and replaced by rather mystic beliefs.

I would like to encourage Stereophile team in doing more controlled tests. Though there may be objections to AB switch box transparencies, the same applies to 'high-end' components and usually at much higher degree. Please continue the good job and do not let you make disappointed from the offensive comments.

Last but not least, the switch box transparency might be well evaluated by SYS-2722 redaction system.

arve's picture

@John Atkinson: Have you considered adding a test to see how gracefully DACs handle intersample overs? While it's possible to create waveforms with a True Peak value with an arbitrary value above 0 dBFS by approaching Nyquist, here is one that generates a peak that's +3dBFS - the below example is done in Audacity, but any sample editor with similar functionality should suffice.

  1. Generate a 0 dBFS sine at 1/2 Nyquist, so each cycle is represented by the four (floating-point) values 0,1,0,-1
  2. Use "Change speed" in the effects menu, and set it to 0.5.
  3. Zoom way in on the start of the waveform, so individual samples become visible. Now, chop off the first sample
  4. Again, use "Change speed" and set the speed to 2.0
  5. Add a short fade-in and fade-out to the beginning and end of the generated tone
  6. (Peak) Normalize the track to 0 dBFS

This has the overall effect of introducing a 45 degree phase shift in the period of the sample, and the peak of each sinusoid will now occur halfway between the two equal-valued samples, causing a true peak that's ~3dB above the sample values in the file. A DAC with less than graceful handling of intersample peaks should thus exhibit clipping of the sinusoid, while a DAC like the Benchmark should still show a proper sine wave.

barrows's picture

arve, love you're idea here. I think testing for this problem would be great and I would love to see it in Stereophile measurements.
When we at Sonore were developing the OSF used in our USB interface we encountered this problem, and adjusted our filter parameters appropriately. With a lot of current music releases featuring full scale 0 dB signals, or even clipping (what are these recording engineers doing) it is important for DAC designers to take this problem into account.
Do note, as I recall, Ted Smith (main designer of the PS Audio DAC referenced here) has noted that the DS DAC uses 6 dB of headroom in its DSP stage. So it is not true that Benchmark is the only company addressing this problem.

arve's picture

@barrows : As a general comment (I haven't looked into the PS Audio DAC), but merely a general observation: DSP headroom isn't necessarily the same as DAC headroom - when processing a signal, you use headroom to prevent downright clipping of the processed signal, but if that DSP outputs digitally to a DAC, but if the DSP itself is _capable_ of outputing a sample value with an amplitude of "max value" for the DAC, there still needs to be headroom, as the intersample peaks occur _between_ the samples.

As an example: In my second setup, I use a little homegrown box running on shairport-sync, with full room correction using BruteFIR. Prior to DSP, that system uses (for the specific filters I use) -9.2 dB of attenuation to guarantee against digital clipping, it provides no guarantee against inter-sample overs, so that job has to be left to the DAC, as the DSP can still output 0dBFS.

But as said: I haven't tried to delve into precisely what Ted Smith and PS Audio is doing with a DSP in their DAC - so they might very well have tackled the problem

JimAustin's picture

See footnote 4 in the review.

Best,
Jim

JimAustin's picture

I've been having trouble posting, or I would have replied sooner.

I think what makes sense is to do some preliminary tests, see how common this problem is. I'll fool around with this a little, test some DACs I have on hand.

I did eventually manage to get good data contrasting the DAC1 and DAC3 response to a test signal similar to the one you describe. It's quite dramatic. Red is the DAC1; orange is the DAC3.

InersampleOvers

Jim

arve's picture

@JimAustin: That's quite dramatic, and something I'd readily take to be audible. Now on to convince JA to include that 11025 measurement as one of his standard measurements.

JimAustin's picture

this is a test signal. Absolutely audible. It's less clear how often intersample overs affect the sound of music--although if Benchmark's John Siau is to be believed, it's very often. (In his Manufacturer's Response, he agreed that the DAC1 sounds brighter than the DAC3--and attributed the difference entirely to intersample overs.) If you haven't already, read the essays on Benchmark's site:

https://benchmarkmedia.com/blogs/application_notes/intersample-overs-in-...

https://benchmarkmedia.com/blogs/application_notes/13545433-audio-that-g...

https://benchmarkmedia.com/blogs/application_notes/13740017-why-audio-go...

TBD, IMO, is how common this problem is in recent/new DACs. JA and no one else will decide what to add to regular testing, but before that's even considered, I think it makes sense to do a little preliminary investigating.

Best,
Jim

supamark's picture

probably are not audible - when I was recording rock bands in the early 90's, I used an Aphex Dominator to prevent overs (it's an analog brick wall limiter). I'd mix a track, check the Sony DAT and verify no overs (I set the limiter to kick in at 0.5dB before 0 using a test tone) and would still occasionally see overs on playback but they were inaudible (Genelec 1031A monitoring). My aim was usually to light it up a few times during the mix to make sure I maxed out the s/n but never used it as a mix buss compressor like some did.

JimAustin's picture

But take a look at John Siau's Manufacturer's Comment. John's not one to claim audibility without evidence. He attributes the DAC1's relative brightness to overs. I intend to do some more listening myself. I did some for this review--but not after I confirmed I was actually getting them--i.e., that the server was sending bit-perfect output. As the review documents, I had a little problem there.

Best,
Jim

supamark's picture

and the D/A converters were nowhere near as good as they are now. I'm also talking about a few overs per song in rock music and generally on a drum strike. I can see it being a much bigger problem with today's mastering techniques, but then again with the aggressive use of hard limiting today how would you hear it separate from all the clipping?

You probably listen to a lot more classical/acoustic music than I do, and I bet it's a lot more audible in that context. I spent a couple years recording classical (always with just a stereo pair of Neumann mics), but 2 channel stereo just can't capture that surrounded by luscious reverb sound of being in the hall and always dissapoints me.

AJ's picture

And you believe yourself to be the arbiter of all what is and isn't audible?
Comedy gold ;-)

supamark's picture

No? Then STFU troll.

arve's picture

… let me show you a pathological case

https://imgur.com/ACqrucC

That's the _additional_ inter-sample clipping introduced into a track (Muse - Map of the Problematique). Intersample peaks below 0.1 dB are ignored. Every vertical red line is an inter-sample over.

While AB(/X) testing this takes some effort, because you need a controlled and calibrated testing setup with precisely matched levels, I would be _very_ surprised if that track didn't reveal differences between a DAC with inter-sample headroom and one without.

supamark's picture

that is some of the worst brick wall limiting I've ever seen - the mastering engineer (Howie Weinberg, who really should know better) should be taken out behind the woodshed and beaten.

arve's picture

Not being a reviewer with access to expensive gear: I tried this on various DACs I have lying around. All but of them will clip the signal,, showing a distortion spectrum similar to yours.

Edited: In re-testing - all of them exhibit clipping when digital volume is set to beyond -3 dBFS

Bubbamike's picture
Quote:

This is the kind of scientistic nonsense that's so common this world--a just-so story (ad-hoc fallacy) that attempts to explain subjective impressions via nice stories or casually observed phenomena while never subjecting those claims to serious tests.

Waiting for these kind of tests for MQA that you and the others at Stereophile seem to have swooned over without ever conducting any scientifically serious tests.

John Atkinson's picture
Bubbamike wrote:
Waiting for these kind of tests for MQA that you and the others at Stereophile seem to have swooned over without ever conducting any scientifically serious tests.

Starting with the January 2018 issue, we are publishing a series of articles examining MQA's claims one-by-one.

But "swoon" - I respectfully suggest your comprehension has been a wee bit colored by what you might have read from ill-informed hot heads on the InterWebs. :-).

John Atkinson
Editor, Stereophile

Bubbamike's picture

Sorry, but I think swoon is a fair description. Not the first time, even you admitted, just an issue or two ago that you and the magazine over empathized the effect of HDCD. I think this MQA mania is of the same enthusiasm. I have never seen a negative word applied to MQA from the staff of Stereophile. Perhaps I'll be surprised by your January issue. I hope so.

tonykaz's picture

There aren't any, are there ?

Sure, there are negatives. Everything has a negative of some sort even if it's quite minor.

MQA's prominent negative is that so many other things have been heavily promoted but not achieved universal acceptance that dubious consumers are remaining skeptical.

After all, isn't Vinyl 'still' the finest music Format for the group of Audiophiles that began during vinyl's Big Era? A good many of these folk remain Digital Deniers although even 'that' position is getting difficult since 2009 or so ( when HP of TAS was started blessing digital stuff ).

Another negative is Bob Stuart and Meridian. I've loved Meridian and Boothroyd Suart since the 1980s but I feel very much alone in this.

MQA being British is not a good thing for us Yanks, why couldn't one of our sharpies figure this out, someone like Edison or Einstein. Dam it, why another Brit thing?. That Linn guy and that LP12 was probably all the Brit we could take. We're the RedWhiteBlue Team and we deserve to win one, don't we?

Maybe the worst part of all this MQA business is that 'our' Warner was the first Record Label to get on-board. phew. Now it seems everyone is in a hurry to do MQA stuff. ( except some Neanderthal outfits that steadfastly refuse to advance into this 21st Century, I won't mention any names except Shit. There are a few more. )

The MQA Positive for those who remain negatives: Noboby has to buy it and nobody will hold it against you, MQA is just better RedBook. As far as I can tell.

Besides, if a person can't hear any MQA difference it only means that they have lesser gear or hearing ( like an old geezer ).

Ancient Tony in Michigan

Bubbamike's picture

You realize that MQA starts out with a high resolution file? It isn't meant to replace Red Book but to allow the streaming of Hi Res files over the internet with reduced bandwidth. Well among other issues, such as DRM and loss of data. But that you didn't know that is an indictment of Stereophile's coverage of MQA. If you look around you'll find explanations and critiques of the method, as well as Bruno Putzey's recent criticism of the lack of reliable tests of MQA.

tonykaz's picture

I accept, MQA is high resolution transmitted via RedBook.

Loss of Data, hmm, is that true?, maybe folks like Bruno Putzey will explain these things or someone else.

So far, MQA seems to be blessed but not universally adopted.

If I were making gear, I'd accept MQA.

As I stand today, I'm happy with RedBook and happy that I don't have to have a vast vinyl collection to enjoy music.

I feel like I'm winning and I'm not letting anyone take that away from me.

Tony in Michigan

arve's picture

I accept, MQA is high resolution transmitted via RedBook.

It's not. "Redbook" is entirely specific to data stored on an Audio CD, and governs all topics related to storing audio on that CD, including data structures and data encoding. The sample rate is only one such aspect. A file stored on a computer or audio streamed over a network can never be redbook.

But, the point you were trying to make was probably related to 16/44.1 audio being stored or streamed. Which is also entirely wrong for MQA. MQA uses a 24-bit container, rather than a 16-bit container.

Loss of Data, hmm, is that true?, maybe folks like Bruno Putzey will explain these things or someone else.

Yes, the data loss is true. John Siau has already done this dissemination of the format for you: https://benchmarkmedia.com/blogs/application_notes/163302855-is-mqa-doa

tonykaz's picture

Ok, I just had a brief look at that Benchmark Media report on "is mqa doa".

Firstly, thank you for pointing out that its a 24Bit container, didn't know ( or probably care ). Egads, can I accept or embrace 144db of dynamic range?, seems way too much ( even on a Battlefield Re-inactment )

Lossless? Sure, it looses the old file as it folds it up. I don't know what to think beyond that. Is it like flour stops being flour as the Pizza is made?

MQA is a just a Streaming System, isn't it? MQA is for the Record Company and the Streaming Company, it's not something built into the CD that we buy, as far as I can tell.

So, the Record Company & MQA devise a way to distribute their owned music to us 'Renters' of their convenient listening system.

We can create our own tiny SD memory cards and own our own Astel & Kern players and not bother with Streaming.

We can also collect vinyl and own vinyl playback gear.

We can have Tape Machines and buy Tape from Acoustic Sounds.

We seem to have a wide range of options.

MQA is just another option.

I don't see the reason for all the fuss.

Tony in Michigan

ps. as far as Streaming listeners are concerned, that Newspaper study showed how people had a hard time hearing 320 being different than hi-res.

Camilo's picture

Arve has proposed a very pertinent and concrete suggestion with regards to your test routine of D/A converters, and its merit and rationale very much deserves your consideration.

The suggested test would allow to recognize the innovation made by Benchmark with regards to intersample overs, and would allow to document and evidence the performance of other D/A converters in this respect.

It would shed more light into the differences between D/A converters, and thus provide a better resource of information for consumers, which is - I believe - the objective of carrying out measurements as well as the mission of a dedicated consumer product magazine.

Best,

Camilo

John Atkinson's picture
Camilo wrote:
Arve has proposed a very pertinent and concrete suggestion with regards to your test routine of D/A converters, and its merit and rationale very much deserves your consideration.

We used routinely to examine digital filter headroom, as you can see from reviews on this site that were originally published in the early 1990s. But I admit that was before the Loudness Wars, when CDs were mastered so that there were never consecutive samples at 0dBFS.

Modern digital audio workstations do calculate the waveform on the assumption that it would be processed by a typical digital filter and I have looked at some modern CDs. Inter-sample overs are rarer than many people believe, though I have found a few pathological examples.

But yes, I think I will start looking again at a DAC's digital filter overload characteristics.

John Atkinson
Editor, Stereophile

arve's picture

Inter-sample overs are rarer than many people believe, though I have found a few pathological examples.

This is a bit preliminary - as there are still non-music (and much quieter) data in the corpus (voice memos, audio books, podcasts and a couple of test tracks), but I have examined a corpus of 14411 tracks for intersample overs. Of those tracks, 6082 tracks reported true peaks below 0 dB, 1237 tracks has true peaks exactly at 0.0, and 7092 tracks have intersample overs. In other words, very roughly half of the tracks contain such overs.

I have not tried to do a more qualitative analysis of the overs, such as how many overs there are in a track, or counting whether some of them have overs that last more than one sample.

SeanS's picture

Hi John,

Eagerly anticipating the MQA article(s).

As an audiophile, I have to say I am not interested in the benefits of smaller file sizes for hi-res files--the origami, etc.

What I am really interested in is what, if anything, MQA can offer to better sound quality vs. a well implemented hi-res AD-DA conversion chain that doesn't use MQA. Like, for example, if you simultaneously created two 24/96 digital recordings of a jazz band using the same recording and playback equipment, with one recording having MQA enabled throughout the chain, and the other without, could you tell the difference? I would really appreciate your expert opinion coming from this perspective.

Thanks,
Sean

NeilS's picture

With all the expert technical scrutiny, even reverse-engineering that MQA has been subjected to on sites like computeraudiophile and Archimago this year, it's hard for me not to get the sense that MQA and its related claims are already technically very well understood by anyone who wants to know.

So when I read about Stereophile announcing at the end of October 2017 that it plans to run a series of articles beginning in 2018 on MQA to test its claims, it sure seems like a lot of effort, but it also seems to me like a lot of effort a lot too late.

Raving about/"swooning" over MQA before testing its claims may be a reminder that sometimes, like putting on socks and shoes, the order in which things are done matters.

tonykaz's picture

Paul McGowan of PS Audio is now doing a Question & Answer Series on YouTube.

Paul just answered the Question about different sounding DACs.

Egads, he's taking the time to answer all kinds of Audiophile questions.

This is well worth checking out.

He even tried to reason thru StillPoints.

Tony in Michigan

Camilo's picture

Paraphrasing the sentence that immediately links to your 5th footnote, I would say: "If you're REVIEWING 24-bit DACs and hi-rez downloads, you'd best get your noise level down to where you can hear at least some of that extra resolution, and that's harder than you might think".

What surprised - and outraged - some who read your review, is that you failed to hear a difference between the Benchmark DAC3 and the PS Audio Perfect Wave DAC, which leads to conclude that if there is a difference, it is not audible and doesn’t matter.

Although your 5th footnote hints a clue to why you - no matter how hard you tried - couldn't hear a difference between the Benchmark DAC3's 21 bit state of the art resolution, and the rather poor 17bit performance of the PS Audio Perfect Wave DAC, your review fails to take into account precisely that advice offered by your own footnote.

That advice obviously draws attention to the even poorer performance delivered by the PS monoblocks, acting as bottleneck when it comes to rendering the 21 bits of resolution of the Benchmark DAC and even the 17 bit performance of the PS Audio DAC, let alone their difference.

As Benchmark's John Siau wrote in a application note back in March 2014: "Anyone who thinks they can hear the difference between 16-bit and 24-bit digital audio through a "17-bit" power amplifier is fooling themselves."("What is high resolution audio? - part1" https://benchmarkmedia.com/blogs/application_notes/13174001-what-is-high...)

Nevertheless, the lenghty passage you dedicated to the efforts taken to hearing the difference between the Benchmark and PS Audio DACs, can suggest to some that this your flawed and misleading conclusion are not not merely innocuous omission or mistake. You appear to ignore your own advice, and completely disavow the fact that your monoblocks aren't capable of delivering even 16 bits to your speakers.

I think it is fair to criticize your review and to dismiss your conclusion regarding the audible difference between the Benchmark DAC and the PS Audio DAC. I also believe it is fair to demand that in the future, stereophile reviews take into account basic specifications - which I want to believe reviewers understand but apparently and consistently fail to apply.

I also believe it is impossible and unacceptable to excuse the obviously flawed attempt to establish a difference between the mentioned components and the consequently misleading conclusion, with a statement like: "Yes, listening is what Stereophile reviewers do." If people read reviews here and elsewhere, it is based on the expectation that they will be offered more than casual listening impressions that completely ignore basic science, let alone the specifications of the components reviewed.

It is in this particular case, quite obvious that you need an amplifier that matches and even exceeds the performance of a DAC to deliver the resolution the DAC has to offer to the speakers, yet that was blatantly ignored by the review. Worse, to reinforce the apparent ignorance - I will nevertheless refrain from concluding foul play or second intentions here - of the reviewer with regards to importance the specifications of the components used have for his review, he proceeds to write a lengthy passage showing what lengths he went to in order to hear a difference that the equipment he used is clearly unable to render. This is also not the only review to be found on Stereophile or Audiostream, with this exact same flaw.

This is poor work, and ultimately undermines the credibility of Stereophile, as well as the effort made by John Atkinson to measure equipment as a way to offer transparency and accountability with regard to manufacturer specifications and in many cases supplement the lack thereof.

I am in my mid fourties and, after thousands of hours behind a drumkit, exposure to loud environments and extended listening periods, cannot argue to hear extremely subtle differences. But even having a well trained ear as a musician, I would not be above relying on the due diligence of reading the specs and setting up valid review, audition or test conditions first, and before I make conclusions regarding the audible differences between two audio components or recordings.

I own a Benchmark DAC2 DX and have owned a DAC1. I had the chance to audition them both side by side with recordings that are known to have intersample overs, and I could very much hear the difference using my Sennheiser HD 800s.

I would hope for a clarifying response from you, with regards to taking into account the components used for a valid review of the performance of specific components. I would also like to see a response from John Atkinson with regards to the very pertinent suggestion made by arve - further up in the thread - with regards to introducing a test that takes into account intersample overs. Atkinson made no remarks to this excellent and concrete proposal, and instead threw in the infamous MQA topic, which distracted from arve’s suggestion.

Updating measurement routines as components introduce new improvements and features that need to be account ted for in order to do fair comparisons between components, would only benefit the objectivity of the measurements and reliability of what Stereophile publishes as such.

Benchmark has clearly separated itself from all other DAC manufacturers I have knowledge of, by effectively dealing with the audible downside of intersample overs, and has clearly introduced a substantial improvement to sound quality that sets new standards. This can only be acknowledged by introducing the corresponding measurement to account for this innovation and the – recording industry – problem of intersample overs, whose existence has been acknowledged and documented, as well as as put forward by this review.

Best,

C. Rodriguez

Robocop's picture

I have owned both the DAC1 and DAC2L over 15 years. This review actually tells me little I havn't already read from other reviewers and the Benchmark web site.

What I find most disconcerting is the DAC2 was the current reference DAC for Benchmark which I own.

Why was the DAC2 not compared directly to the new reference DAC3 in listening sessions?

Comparison with the DAC1 is obsolete, well surpassed by the DAC2.

"How much audible improvement do these changes add up to? "I'm quite certain that there should be no audible difference between a DAC2 and a DAC3 given a single pass through the converters," Siau wrote to me in an e-mail."

What does this mean from John Siau Benchmark? Is he saying no audible improvement over the DAC2?

If it sounds the same, well why bother!!!

I really want to know how much better is the DAC3 over the 2 to justify its increased purchase price.

There must be a sound improvement, the Sabre 9028 chip is alone a sonic upgrade over the 9018. This must be audible and not just measured.

It is at the end of the day all about the "SOUND" compared to live instruments in an acoustic space.

Robert

Sigh's picture

It seems that the 9028 is the new 9018 that might be discountinued soon, I can see why Benchmark made the change even if it offers only a minute improvement that isn't audible. Maybe they should have called it DAC2.2 (Although they did have a little update to the DAC2 called 2.2 already) or DAC2+ like Mytek with their Brooklyn+.

To compare the PS-Audio to the DAC3 in a set up that would have made some sort of sense one could run the analog-out of one DAC3 into the analog-in of another DAC3 :) Is the reviewer suggesting that this too would have sounded exactly like a single DAC3 on its digital volume control?

I've owned every Benchmark, DAC1, USB, PRE, HDR, DAC2 and changed more for the new features (USB, remote, second line in) than the negligible sound differences. The one thing I didn't like with the DAC1s was that they ran hot. With the DAC2 I feel like they reached a perfect product, great interface, hybrid gain control, runs cool, asynchronous USB.

It probably makes no sense to trade a DAC2 for a DAC3, yet at the same time it would make no sense for Benchmark to continue using a older chip.

I find that whatever my questions are, the best way to get them answered is to ask Benchmark themselves. They answer promptly and won't push a new product onto you.

Charles Hansen's picture

Once again, I am baffled by Benchmark, designer John Siau, and reviewer Jim Austin. The existence of intersample overs has been well known for over 20 years. I'm surprised that JA did not catch this much earlier.

For evidence of this fact, we only need to look at the datasheet for the once-popular Pacific Micronicss PMD-100 digital filter. In the datasheet is a an unambiguous statement:

The PMD-100 has a design attenuation of 1 dB to allow for filter overshoot on transients.

During the '90s when Robert Harley was technical editor of Stereophile, he would routinely show CD players with no internal headroom, and how they would clip the "ringing" (Gibbs phenomenon) on the tops of a 1kHz, 0dBFS square wave. In contrast were other players that used (for example) the PMD-100 digital filter (and many other designs) that provided headroom to prevent internal overload from intersample overs.

While this phenomenon is understood more clearly now, with a better understanding of the degenerate (worst possible) case, this is hardly some sort of "breakthrough" as Siau and Benchmark would have us believe. In fact it's more surprising to me that he was previously unaware of a well-known issue regarding digital audio playback.

Camilo's picture

Knowing the problem does not automatically guarantee there to be an immediate solution, and blaming those who finally solve a pervasive and known problem for not having solved it before and for taking credit for it, doesn't seem like the right place to aim your criticism at.

The solution for the audible artifacts of intersample overs that Benchmark came up with, is not something trivial, as it is an inherent flaw of D/A chips including the ES9018 and ES9028PRO used in the DAC2 and DAC3, respectively:

"Every D/A chip and SRC chip that we have tested here at Benchmark has an intersample clipping problem! To the best of our knowledge, no chip manufacturer has adequately addressed this problem. For this reason, virtually every audio device on the market has an intersample overload problem. This problem is most noticeable when playing 44.1 kHz sample rates." ("Intersample Overs in CD Recordings"-John Siau https://benchmarkmedia.com/blogs/application_notes/intersample-overs-in-...)

Benchmark had to develop and implemenmt a solution that is not an option provided by the D/A chip, but which addresses a common problem that originates in D/A chips:

"It is possible to build interpolators that will not clip or overload, but this is not being done by the D/A and SRC chip manufacturers. For this reason, Benchmark has moved some of the digital processing outside of the D/A chip. In the Benchmark DAC2 and DAC3 converters we have an external interpolator that has 3.5 dB of headroom above 0 dBFS. This means that the worst-case +3.01 dBFS intersample peaks can be processed without clipping. We also drive the ESS D/A converter chips at -3.5 dB so that no clipping will occur inside the ES9018 and ES9028PRO converter chips." ("Intersample Overs in CD Recordings"-John Siau https://benchmarkmedia.com/blogs/application_notes/intersample-overs-in-...)

It is not out of mere negligence that Benchmark, or other manufacturers who rely more on the D/A chips - and who in some cases even list the specs of D/A chips rather than the measured performance of their components, and thus not the result of their implementation of D/A chips - haven't offered an effective solution to the intersample overs problem before. Being Benchmark the ones who came up with it,
hardly makes them the target to blame for said negligence, let alone undeserving of the merit for effectively addressing the issue of intersample overs.

If anything, it is other manufacturers who have not yet implemented a similar solution who could deserve some criticism in this respect.

As to the developing and final implementation of a solution to intersample overs, it is debatable if - according to your appreciation - it took the industry too long. It is however not debatable that Benchmark's solution is very much a welcome breakthrough for the performance and sound quality of D/A converters, and one that clearly sets them apart from all other manufacturers.

SeanS's picture

Wouldn't Siau's solution have to sacrifice dynamic range in the digital domain? He says they then bump up the 3.5db in the analog domain to bring the level back up. Definitely not perfect. Sounds like your trading dynamic range on all input data in order to correct the consequences of "bad" input data. On the other hand, maybe the amount of dynamic range lost is less than the noise floor?
Sean

arve's picture

Due to thermal noise, no DAC can achieve the full 24 bits of dynamic range inherent to such program material - at _best_ the analog outputs of a DAC will offer a bit above 21 bits of signal, with the three lowest bit drowning in the thermal noise. By adding 3.5 dB of attenuation to the signal, Benchmark is throwing away approximately the lowest half bit of the signal - this is signal that was already lost because analog electronics won't let us do better.

The loss in level is made up for through analog means adding the 3.5 dB of gain back after conversion. The risk here is that adding this gain causes the noise floor to rise, and thus lose dynamic range. However, as seen from JA's measurements, this is not the case with the Benchmark: The analog outputs have class-leading state-of-the art performance with regards to the noise floor.

SeanS's picture

Arve, the content of your message seems to elaborate on the thoughts I was having when I posted.

Throwing away the last half bit of the signal = throwing away dynamic range in the digital domain.

However, the low level thrown away was already lost because of the physical limitations on analog electronics.... = it is below the noise floor.

I was hoping to get opinions on the solution.

Sean

nikolaos's picture

First of all I'm a huge fan of Ayre and had several components from you and always enjoy whenever I hear Ayre gear. I'm also a huge fan of Stereophile and consider the mag.. as very serious. I will also say I enjoyed reading the review of DAC 3.

Regarding your comments (sorry I'm not english so forgive me for errors in the writing) I had to reply regarding things that where said.

I had lot's of equipment and tested a lot and my conclusion is that whatever the inputs or cables has to say to what I'm hearing does not even come close to other factors like room, components and the ultimate headroom in the system itself.
I had some quite expensive cables and I can for sure not say some where better than others even if the price difference was huge. My concusion is that it is so minor differences in cables that you will gain much more elsewhere in the system. I have heard amazing systems with cables that where the cheapest and I heard the worst systems with the most expensive cables.
I had simelar experience as the reviewer regarding hum with unshield Nordost cables but as Charlie say as they are balanced it can not happen. I guess the system was not truly balanced.
I agree that the reviewer or the editor maybe should have picked that up.
Regarding Benchmark it is well know that they have made great products for some time and as J.A show in the meassurements of this DAC it is a well designed product. I don't understand why a DAC should cost as much as some charge. AYRE used to have their top of the line DAC in around the same price range as Benchmark. You say Mr. Hansen that the DAC 3 is made of cheap components and it seams like you consider Benchmark bellow average quality.

I don't buy to much into all that voodoo anymore about lot's of the claims in the high-end industry. Not to say that from time to time some come up with new and better solutions.
Audio is to me much more about measurements and real world facts. Even if the reviewer did something not considered right by connecting one DAC into the other I'm not so sure it would be so easy to hear coloring of the sound. It would for sure act as a filter but so does a preamp.

As said I'm not English and maybe I'm stupid but I did not get the problem you had with how the reviewer explained how he connected the output of the DACs to the preamp.

It is a little industry. Let's enjoy the music and be friends. Everytime I meet someone with the same hobby it is always a pleasure. In a forum things are very unpersonal and I know people would not act like this when met for real.

Pages

Benchmark DAC3 HGC D/A preamplifier-headphone amplifier Specifications

Thu, 10/26/2017

COMMENTS
tonykaz's picture

NwAvGuy was describing the Benchmark and using it to evaluate his own DAC designs.

Mark Waldrip relies on Benchmark gear

I've heard of a good many Pro Audio folks tout Benchmark as being the Benchmark.

I've been trying DACs for some years now, I can't discover any advantage behind some of the super pricy DACs.

I've blamed my hearing, to the point of having my hearing evaluated by Audiologists at the University of Michigan. My hearing tapers off above 8k but I can still hear significant differences in 12AU7 preamp tubes. I still can't hear greatness in super expensive DACs.

Those that can hear significant improvements in Super Expensive DACs are living with the Audiophile Curse. ( the King's New Suit Curse )

As an Engineer, I've demonstrated Zip Lamp Cord vs. Monster Speaker Cable vs. Bruce Brisson's MH-750 to astonished Engineers.

Back in 2015 Tyll had a headphone gear Shootout where Stax, Sennheiser, and Audeze were the finalists. He also had the finest DACs including the highly touted Antelope. Nobody could detect any DAC performance advantage.

Then there's Chord and their Field Programable Gate Array devices which I suspect they've designed to sound better but probably not sound accurate ( like Tubes that make music nicer sounding ).

So, go for Accuracy with Benchmark or better sounding with Chord.

Nice reporting, sir, I think.

Tony in Michigan

barrows's picture

I would suggest that there is not much evidence to support your assertion that Chord DACs are not designed to sound accurate, take a look at their measurements here at stereophile.com...

On this review in general, despite the lack of difference heard vs the DS, it is interesting to note that Jim found an immediately apparent difference to the Benchmark DAC1, subjectively speaking...

Staxguy's picture

Accurate? Chord DAC's can't even handle (do) 24-bit. Even their best DAC, the Chord DAVE can't even do 24-bit. 21-bit, that's it!

Their 17-order noise shaper can do 350 dB DNR, but that's internal aah only.

It's taken forever for this review to come out. Back when we (I) was recommending the Benchmark DAC1 HGC in my magazine as a recommended component, The Absolute Sound was only recommending DACs with sub-par 16-bit performance and not even mentioning it, even though it had came out.

Well, it's great to see this review. I might buy the Rogue Audio RH-5 as a headphone amplifier, reviewed just before this one, which likely functions worse (no better) than the Aurorasound HEDA, because it looks good on the desk!

People like benchmark, and good for/on them. It's got a great summing DAC (multiple DAC) design.

I'd rather have the GTE Trinity DAC (the one with the actual triangles not the newer ahh one), but that's for myspace!

barrows's picture

Your reply is kind of in error. Here is a quote from JA's measurements on this Benchmark:

"...the increase in bit depth dropped the noise floor by more than 30dB (fig.5), indicating that the Benchmark's resolution is at least 21 bits. This is as good as a DAC can currently get!"

There is no DAC of which I am currently aware that achieves better than 21 bit resolution at its analog outputs. The resolution measured here for the Benchmark is at the same level as Chord DAVE (look it up on this site). When DAC makers say they are 24 bits, or 32 bits, that is what level the DAC operates at in the digital domain, but no DAC achieves that resolution at its analog outputs.

I am curious, do you have any references for measurements of the Trinity? I would expect them to be way worse than that of the Benchmark, not saying anything against the Trinity as far as sound quality is concerned though, just, it is likely not nearly as "accurate" being an R2R DAC...

Sal1950's picture

"There's a danger of being misled, of repeating the same mistakes again and again, of spending way too much money on things of little value. My point is that, as a hobby, industry, and avocation, we may have shifted too far toward the subjectivist side."

What a breath of fresh air in these pages! When I first read this review in my subscriber copy my jaw hit the floor. Thank you Jim, we can only hope as audiophiles dedicated to the accurate reporting, that blind listening sessions become much more widespread and prevalent in the pages of Stereophile. Sighted listening is much too fallible to be the sole basis for accurate evaluations.

mrkaic's picture

I hope this year, this month, and this review mark the beginning of the end of the tyranny of subjectivists.

ChrisS's picture

...listening/reviewing of this component was done blinded.

He did it the way we all do.

He listens.

He thinks about it.

He writes about what he hears.

That's what they all do at Stereophile.

ChrisS's picture

"Yes. Listening is what Stereophile reviewers do."

mrkaic's picture

There is only one question in audio and it is rhetorical -- why buy anything but Benchmark?

They are the best, nothing comes close.

Charles Hansen's picture

You claim to have a PhD and you make the most egregious beginner's mistakes imaginable. It is laughable that even Streophile would publish this garbage.

First you start of the article with a story about "hum and buzz" from "poor quality control" in the interconnects. If you knew anything at all about balanced circuitry and how it works, you would realize that a truly balanced source sending a signal to a truly balanced downstream device wouldn't make a bit of "hum and buzz" if the cables were shielded or not.

But since you clearly have no idea about anything, you start with the wrong assumption and leap to an incorrect conclusion. [Flame deleted by John Atkinson]

Then you try to "compare" the Benchmark against your PS Audio. And although you could "easily" hear differences between the Benchmarks 1 and 3, you could hear no difference whatsoever between the Benchmark 3 and the PS Audio. Why don't you look at your ridiculous methodology before jumping to more false conclusions?

First you said you connected "sent the output of the two DACs to different channels of my PS Audio BHK Signature preamp". WTF? So you sent the left channel of one DAC to one input and the right channel of the other DAC to the other input? Or do you not know the difference between a "channel" and an "input"? At this point you are looking beyond unprofessional, beyond amateurish, and all the way to "knowing just enough to be dangerous". Or does your preamp have only one input? Or does it have more than one but the inputs were full, and you were too lazy to disconnect them?

Anybody serious would know that the only FAIR way to do this type of test is to not only use the same input on the preamp (different inputs will exhibit different amounts of break-in, as the trace on the PCB needs a signal to dissipate the residual electrostatic charge after it is tested with many kilovolts for UL compliance), but to also use the exact same pair of interconnects as they need to be not only the same brand and model, but also have the same amount of break-in time.

And finally you lose all rationality. (We may have to send the men in white coats to save you from harming yourself.) You send the analog output of the PS Audio DAC into the analog input of the Benchmark DAC. At the very least this signal is now going through the analog circuitry and volume control of the Benchmark, which is clearly coloring the sound if it sounds "identical" to you - either that or your hearing ability sucks or your playback system sucks. And if (like the version 2) the version 3 also has a "hybrid digital/analog" volume control then the volume control wouldn't even be active unless the analog input were being digitized by whatever cheap A/D chip Benchmark is using with their cheap IC op-amp based analog circuitry. More cheap crap to color the sound of the PS Audio, and you can't figure ANY of this out...

Finally it is clear that you cannot even add 2 + 2 to equal the correct answer. First you claim that the Benchmark "moves interpolation off-chip" to eliminate problems with "intersample overs", but that the latest version sounds better because of "lower passband ripple, facilitated by the new chip's superior filter choices". So which is it, Mr. PhD? Does it sound better because the processing is done off-chip or does it sound better because the on-chip processing has lower passband ripple? You can't have it both ways.

[Flame deleted by John Atkinson]

You are the one that makes a joke out of the high end, not the people who make claims you can't understand - you can't even understand the simplest of claims, as this so-called "review" clearly demonstrates.

AJ's picture
Quote:

Anybody serious would know that the only FAIR way to do this type of test is to not only use the same input on the preamp (different inputs will exhibit different amounts of break-in, as the trace on the PCB needs a signal to dissipate the residual electrostatic charge after it is tested with many kilovolts for UL compliance), but to also use the exact same pair of interconnects as they need to be not only the same brand and model, but also have the same amount of break-in time.
And finally you lose all rationality.

Can't make this stuff up ;-).
Charles, your rational scientific evidence of this audible "break-in" outside the imaginations inside your head?
TIA

mrkaic's picture

It boggles the mind that someone who believes in cable break-in has the chutzpah to write about electronics. It really does.

But we can learn here, for sure. In the words of the previous occupant of the White House— it is a teachable moment—sans beer, in this particular case.

It is tempting to get angry at the omni-present neglect of science education and fume about the ignorance of individuals who believe in cable break-in, directional fuses etc. But it is more fun is to realize that those esteemed gentlemen, who believe, for example, in cable break-in, provide some free comic relief in these trying times.

So, don’t get mad, get entertained.

Anton's picture

I'd hate to see how worked up you get over real world issues.

barrows's picture

Jim. I too was disturbed by some of your methodology in your comparisons, and dismissed your results because of my concerns. While I may not go as far as Charlie, please note that I felt disturbed enough to dismiss the results.
I would suggest that if you want to make quick A/B comparisons of source components in future reviews, you should have a balanced switch box made, who's only function is to take two balanced inputs, and switch them to a single balanced output. I would also suggest that this be wired the AES way for consistency (XLR pin 1 to ground) and that the signal path internally be identical for both inputs, same length wiring, etc. A high quality switch box like this should be a reasonable expense for any any reviewer who is interested in making quick A/B comparisons of source components.
Also of import would be keeping everything else the same, same cabling, same cabling lengths, etc.
In the case of comparing these two DACs (DS and DAC 3) such a methodology could then go direct into the amps, removing any additional components (preamp) and maximizing the apparent sonic differences. You should also volume match the DACs by measuring the output voltage with a test signal (like that available on stereophile test CDs) without very accurate level matching any such comparisons are mostly meaningless.

supamark's picture

won't work as you think. You're going to add two add'l sets of connections (in/out of the box) which will color the sound. Not every component has balanced outs, and a "dirty" little industry secret is that a majority of XLR outs are derived from the unbalanced RCA outs since few components are actually fully balanced internally. Okay, not really a secret since the few mfg's who make components that are fully balanced from input to output will make sure you know about it.

You will also still need a volume control since not everything being reviewed will have a variable output (and reducing volume in the digital domain adds its own.... issues). Jim's method of using two line inputs of his high quality preamp (that I'mma assume Jim knows the sound of quite well) is a much better methodology. The only way I could think of to improve his methodology would be for him to have someone else connect the two devices and make sure he doesn't know which input is which device.

Let me tell ya another little secret that every recording engineer knows about critical listening (this has happened to EVERY recording engineer at least once): You can imagine differences in sound that are not actually there. Every engineer has at some point reached for the EQ, started adjusting it and heard the sound change, and I mean literally hearing the changes being made to the sound.... but there's a problem - you never actually engaged the EQ (there's an on/off relay switch on every mixing console to switch the EQ into the signal path). It was all in your mind. This, to me, makes two very different points:

1. blind A/B/X testing really is the only fully valid method, but unfortunately nobody's come up with a truly transparent comparator (and likely never will).

2. those expensive cables from like Nordost et.al. are mostly snake oil. A much better system would be 100% copper all the way - internal component wiring (already happens), line level plugs/jacks, speaker connections all 100% copper (or silver/gold/whatever but must all be the same metal and the same purity/alloy). When the electrical signal is traveling along and the medium (metal) changes the signal will change. Were I a high end mfg, I would make all my connectors copper and provide high quality all copper cables at no add'l charge (because, seriously, the mark-up on cables is crazy - 30 years ago I picked up 40' of Kimber Cable at dealer cost, $1 per foot).

barrows's picture

won't work as you think. You're going to add two add'l sets of connections (in/out of the box) which will color the sound. Not every component has balanced outs, and a "dirty" little industry secret is that a majority of XLR outs are derived from the unbalanced RCA outs since few components are actually fully balanced internally. Okay, not really a secret since the few mfg's who make components that are fully balanced from input to output will make sure you know about it.
Read more at https://www.stereophile.com/content/benchmark-dac3-hgc-da-preamplifier-h..."

Only one xtra pair of interconnects, and this is not a problem, as long as they are the same, the difference will still be the same. Same thing with the switch box itself, the entire point is to give the same signal path for each DUT, the result being that one will only hear the difference. This set up, given decent construction and good interconnects will be transparent enough to hear any actual difference.
Additionally, while your contention that components are generally pseudo balanced may have been the case long ago, it is not common now-for example, the DACs mentioned in Jim's review are true balanced, as are his pre amp and amp.

And if you think Nordost cabling is snake oil, you are either deaf, or have not listened to it.

supamark's picture

It's not the 1 extra pair of interconnects (the extra wire is essentially meaningless*), it's the extra set of connections - jacks and plugs (along with the circuitry inside the box, including gain matching circuits, even if they're just variable resistors). You going with a relay or mechanical switch? Both degrade the sound in different ways. Adding that crap in will make it harder to hear any differences because you just added noise and made the signal path more complex. That is much worse than using two inputs of the same high quality preamp.

Most home audio gear is not fully balanced inside (nor should it be). Its adoption rate is higher in high-end home audio, but honestly it's not really necessary in most home environments. Pro audio, where there's much longer cable runs and a lot more electrical noise, is where it's really useful.

*when I was recording Austin Symphony Orchestra in Bass Concert Hall at The University of Texas in the early 90's, the mic cable run - from the stereo pair of Neumann U87 mic's through underground conduits below a couple other buildings into our machine room patch bay and into our Harrison Series 10 console was at least a quarter mile, probably close to 2,000 feet. The sound was not dull or even particularly weak. The U87's large presence boost probably helped but as I said, a couple feet of line level cable ain't gonna mean squat.

btw, if you think Nordost or anyone's cable can magically transform the sound that is otherwise running through traces on a f'ing circuit board inside each piece of your analog gear as it moves from cap to resistor to etc, you have a few things to learn about science (esp. physics). I mean, holy sh!t, they'll sell you little teflon and wood teepees for your speaker cables (for a pretty penny, 'natch). It's really no different than the Tice Clock or coloring the edges of your CD's with a green marker. Maybe Nordost has proprietary connectors that do a truly excellent job of passing signal with minimal alteration - great, that's a real upgrade and can be audible... but that wire between those connectors for which they're charging you several grand more than JA's preamp? that stuff is snake oil.

Buy a high quality power conditioner instead of fancy cables, it'll make a real audible difference and cost a lot less money.

barrows's picture

Are you nuts? I am talking about a passive switch box here, no active circuitry. Just two sets of XLR input jacks, some wire, a high quality switch (lets say a shallco) and two output jacks. That is all. No resistors, no circuitry of any any kind. Will this change the sound, perhaps a tiny, tiny bit, but not enough to obscure differences between the two DUTs, as the change will be identical to both products. A passive box like this will have much less influence on sound than a fully active preamp (which we eliminate from the chain with this testing method) which includes the wiring and switch, but also adds a power supply (noise source), transistors, resistors, etc. We do not need any circuitry, both DACs are designed to run direct to power amps, and we match levels via the DAC VC.

On cables, I never suggested that a cable can "magically transform sound"! Please do not put words in my mouth, neither did I make any comment re audio cable pricing. I just said that if you think Nordost cabling is snake oil, you are either deaf or have not listened to it, and I stand by that. To be more detailed: compare directly a Nordost cable to Mogami, and you will hear a significant difference. But the cable thing is OT here, so lets leave it for another place.

supamark's picture

Those added connections will in fact degrade the sound. You are adding 2 additional connections to each channel (XLR's, which really aren't as firm as RCA connectors - they wiggle a bit), each will degrade the sound. Also, since not every DAC has balanced outs, you'll want another box with RCA's (or just go unbalanced in the first place - the main advantage with balanced is if you have RF issues in your home or really long cable runs).

If you believe so strongly that wire has a sound, just use the same brand/length/termination/model for both and they'll be the same... it's cable, not 9' concert grand pianos (which actually do all sound different because they're handmade and no two are identical but brands do have "house" sounds - Hamburg Steinways are my favorite btw, and Baldwins are popular in rock because they're a little more mid-heavy with less sparkle than NY or Hamburg Steinways and therefore easier to cut through a dense mix).

Not every DAC has a volume control, and even for those that do you'll need a master volume control if you want adjust volume while keeping the levels matched - unless you think no volume adjustments should be made? Besides, the fixed output *should* always be superior (digital volume controls necessarily change the bits, and not for the better).

Also, what if the output impedences of the DACs are not close? going direct into a tube amp could cause differences with freq. response.

So, yeah, going into a high quality preamp is both simpler and more likely to yield consistent results with a much wider variety of DACs.

AJ's picture
Quote:

You going with a relay or mechanical switch? Both degrade the sound in different ways. Adding that crap in will make it harder to hear any differences because you just added noise and made the signal path more complex. That is much worse than using two inputs of the same high quality preamp.

You just need to wave your hands around as you say this stuff, then finish with "believe me" ;-)

supamark's picture

I'm livin' in your tiny little head, you can't quit me. it's really sad, really really sad.

AJ's picture
Quote:

but unfortunately nobody's come up with a truly transparent comparator (and likely never will).

Your factual evidence please.

supamark's picture

that's just stupid. Why don't you go out and find/build one and prove me wrong?

AJ's picture
Quote:

Why don't you go out and find/build one and prove me wrong?

Because folks with a modicum of intelligence know where Burden of Proof lies ;-).
Ok, so you admit to having zero evidence for your specious claim.
The question was rhetorical.

supamark's picture

when someone is stupid enough to ask someone to prove a negative, it simply proves that they're just stupid.

Again, find me a transparent A/B/X comparator system... maybe get Nelson Pass (or his pre-amp guy), Dan D'Augustino, and the guys at Schiit on it, they're all excellent engineers.

AJ's picture
Quote:

when someone is stupid enough to ask someone to prove a negative

Except I didn't. Your specious BS claim:

Quote:

nobody's come up with a truly transparent comparator (and likely never will)

I asked for evidence to support your BS claim of (all known) comparators (never mind I can only think of 2, the current AVA and a discontinued QSC. The really silly "all future ones" is too funny) are non-transparent.
You admit you have none. IOW, your claim is total unsupported evidence free BS. "Stupid"? Well, you decide ;-)

AJ's picture
Quote:

Again, find me a transparent A/B/X comparator system... maybe get Nelson Pass (or his pre-amp guy), Dan D'Augustino, and the guys at Schiit on it, they're all excellent engineers.

Though you could never comprehend it, that sir, is your burden, not mine ;-)

Corvaldt's picture

Actually the burden of proof is on you here. Because it is possible to create a non-transparent comparator in an actually infinite number of ways, your claim that creating a transparent comparator IS possible is the less likely to be true.

supamark's picture

but where is it? Also, as I said before, you can't prove a negative. I made the claim (and I'm far from the only one), you can choose to accept/believe my claim, you can try (possibly even succeed) in proving my claim wrong, or you can just ignore it because life is short and this shit ain't important.

AJ's picture
Quote:

I made the claim

Bingo!
..and then the hands waved frantically...because the evidence was zero ;-)

supamark's picture

I have my very own stalker, how pathetic of you.

rschryer's picture

But, AJ, here is where your argument doesn't hold water: Supamark is saying that a truly transparent comparator DOESN'T exist. With that in mind, if I were to tell you that your inability to prove that wood fairies DON'T exist means that they do, would you agree with me? If not, then the burden of proving that a truly transparent comparator does exist rests on your shoulders, because you are insinuating, by way of your challenge to supamark, that one does. Do you know of a truly transparent comparator? If not, then supamark wins by default.

AJ's picture
Quote:

Supamark is saying that a truly transparent comparator DOESN'T exist.

Exactly!
Now both of you are burdened with evidence for your fabricated imaginary assertions.
Let's see it....or more frantic hand waving.

rschryer's picture

I may be burdened with fabricated imaginary assertions, but yours isn't one of them.

AJ's picture

Of course you would have no comprehension of Argument from Ignorance,
but your false assumption that his non-transparent comparator claim is true, thus the burden shifts to me to provide evidence to disprove it...is exactly that.
No worries, we're in the age of hand waving "believe me", it's expected ;-)

SeanS's picture

Analog inputs stay analog.

From the manufacturers manual:
HGC™ is Benchmark's unique Hybrid Gain
Control™ system. The DAC3 combines active
analog gain control, passive low-impedance
attenuators, a 32-bit digital gain control, and
a servo-driven volume control.
All inputs are controlled by the rotary volume
control. This volume control moves in
response to commands from the remote
control. Analog inputs are never converted to
digital, and digital inputs never pass through
an analog potentiometer. Digital inputs are
precisely controlled in the 32-bit DSP system.
The DSP system preserves precise L/R
balance, and precise stereo imaging, while
avoiding any source of noise and distortion.

Sigh's picture

I too was shocked by this way of comparing the PS-Audio to the Benchmark. Wether you hear a difference or not doesn't even matter because you (and we) don't even know what it is that makes the sound of one chain or the other, (The DAC sections? The Cable? The differences between the analog or digital volume control?). Read what Mytek has to say about the effect of digital and analog volume controls in their Brooklyn. I was eagerly waiting for this review since I am a big fan of all Benchmark DACs.
The lack of rigor and the seven pages of blabla are almost offensive to the companies that put years of hard work into their products and put them in your hands. Some of the comments could have been phrased more diplomatically but the nonsense one reads in Stereophile, 6moons, DAR... the contradictions, the approximations, the lack of methodology are infuriating because your opinions greatly influence the success of these products. The spectacular measurements are useful. But what exactly is there to learn from what you wrote?

pma's picture

Nice review Jim and also thanks to John Atkinson for a valuable set of measurements, as always. We can see that the review has initiated quite strong reactions of a prominent high-end designer as it obviously targets very good objective parameters that tend to be overlooked and replaced by rather mystic beliefs.

I would like to encourage Stereophile team in doing more controlled tests. Though there may be objections to AB switch box transparencies, the same applies to 'high-end' components and usually at much higher degree. Please continue the good job and do not let you make disappointed from the offensive comments.

Last but not least, the switch box transparency might be well evaluated by SYS-2722 redaction system.

arve's picture

@John Atkinson: Have you considered adding a test to see how gracefully DACs handle intersample overs? While it's possible to create waveforms with a True Peak value with an arbitrary value above 0 dBFS by approaching Nyquist, here is one that generates a peak that's +3dBFS - the below example is done in Audacity, but any sample editor with similar functionality should suffice.

  1. Generate a 0 dBFS sine at 1/2 Nyquist, so each cycle is represented by the four (floating-point) values 0,1,0,-1
  2. Use "Change speed" in the effects menu, and set it to 0.5.
  3. Zoom way in on the start of the waveform, so individual samples become visible. Now, chop off the first sample
  4. Again, use "Change speed" and set the speed to 2.0
  5. Add a short fade-in and fade-out to the beginning and end of the generated tone
  6. (Peak) Normalize the track to 0 dBFS

This has the overall effect of introducing a 45 degree phase shift in the period of the sample, and the peak of each sinusoid will now occur halfway between the two equal-valued samples, causing a true peak that's ~3dB above the sample values in the file. A DAC with less than graceful handling of intersample peaks should thus exhibit clipping of the sinusoid, while a DAC like the Benchmark should still show a proper sine wave.

barrows's picture

arve, love you're idea here. I think testing for this problem would be great and I would love to see it in Stereophile measurements.
When we at Sonore were developing the OSF used in our USB interface we encountered this problem, and adjusted our filter parameters appropriately. With a lot of current music releases featuring full scale 0 dB signals, or even clipping (what are these recording engineers doing) it is important for DAC designers to take this problem into account.
Do note, as I recall, Ted Smith (main designer of the PS Audio DAC referenced here) has noted that the DS DAC uses 6 dB of headroom in its DSP stage. So it is not true that Benchmark is the only company addressing this problem.

arve's picture

@barrows : As a general comment (I haven't looked into the PS Audio DAC), but merely a general observation: DSP headroom isn't necessarily the same as DAC headroom - when processing a signal, you use headroom to prevent downright clipping of the processed signal, but if that DSP outputs digitally to a DAC, but if the DSP itself is _capable_ of outputing a sample value with an amplitude of "max value" for the DAC, there still needs to be headroom, as the intersample peaks occur _between_ the samples.

As an example: In my second setup, I use a little homegrown box running on shairport-sync, with full room correction using BruteFIR. Prior to DSP, that system uses (for the specific filters I use) -9.2 dB of attenuation to guarantee against digital clipping, it provides no guarantee against inter-sample overs, so that job has to be left to the DAC, as the DSP can still output 0dBFS.

But as said: I haven't tried to delve into precisely what Ted Smith and PS Audio is doing with a DSP in their DAC - so they might very well have tackled the problem

JimAustin's picture

See footnote 4 in the review.

Best,
Jim

JimAustin's picture

I've been having trouble posting, or I would have replied sooner.

I think what makes sense is to do some preliminary tests, see how common this problem is. I'll fool around with this a little, test some DACs I have on hand.

I did eventually manage to get good data contrasting the DAC1 and DAC3 response to a test signal similar to the one you describe. It's quite dramatic. Red is the DAC1; orange is the DAC3.

InersampleOvers

Jim

arve's picture

@JimAustin: That's quite dramatic, and something I'd readily take to be audible. Now on to convince JA to include that 11025 measurement as one of his standard measurements.

JimAustin's picture

this is a test signal. Absolutely audible. It's less clear how often intersample overs affect the sound of music--although if Benchmark's John Siau is to be believed, it's very often. (In his Manufacturer's Response, he agreed that the DAC1 sounds brighter than the DAC3--and attributed the difference entirely to intersample overs.) If you haven't already, read the essays on Benchmark's site:

https://benchmarkmedia.com/blogs/application_notes/intersample-overs-in-...

https://benchmarkmedia.com/blogs/application_notes/13545433-audio-that-g...

https://benchmarkmedia.com/blogs/application_notes/13740017-why-audio-go...

TBD, IMO, is how common this problem is in recent/new DACs. JA and no one else will decide what to add to regular testing, but before that's even considered, I think it makes sense to do a little preliminary investigating.

Best,
Jim

supamark's picture

probably are not audible - when I was recording rock bands in the early 90's, I used an Aphex Dominator to prevent overs (it's an analog brick wall limiter). I'd mix a track, check the Sony DAT and verify no overs (I set the limiter to kick in at 0.5dB before 0 using a test tone) and would still occasionally see overs on playback but they were inaudible (Genelec 1031A monitoring). My aim was usually to light it up a few times during the mix to make sure I maxed out the s/n but never used it as a mix buss compressor like some did.

JimAustin's picture

But take a look at John Siau's Manufacturer's Comment. John's not one to claim audibility without evidence. He attributes the DAC1's relative brightness to overs. I intend to do some more listening myself. I did some for this review--but not after I confirmed I was actually getting them--i.e., that the server was sending bit-perfect output. As the review documents, I had a little problem there.

Best,
Jim

supamark's picture

and the D/A converters were nowhere near as good as they are now. I'm also talking about a few overs per song in rock music and generally on a drum strike. I can see it being a much bigger problem with today's mastering techniques, but then again with the aggressive use of hard limiting today how would you hear it separate from all the clipping?

You probably listen to a lot more classical/acoustic music than I do, and I bet it's a lot more audible in that context. I spent a couple years recording classical (always with just a stereo pair of Neumann mics), but 2 channel stereo just can't capture that surrounded by luscious reverb sound of being in the hall and always dissapoints me.

AJ's picture

And you believe yourself to be the arbiter of all what is and isn't audible?
Comedy gold ;-)

supamark's picture

No? Then STFU troll.

arve's picture

… let me show you a pathological case

https://imgur.com/ACqrucC

That's the _additional_ inter-sample clipping introduced into a track (Muse - Map of the Problematique). Intersample peaks below 0.1 dB are ignored. Every vertical red line is an inter-sample over.

While AB(/X) testing this takes some effort, because you need a controlled and calibrated testing setup with precisely matched levels, I would be _very_ surprised if that track didn't reveal differences between a DAC with inter-sample headroom and one without.

supamark's picture

that is some of the worst brick wall limiting I've ever seen - the mastering engineer (Howie Weinberg, who really should know better) should be taken out behind the woodshed and beaten.

arve's picture

Not being a reviewer with access to expensive gear: I tried this on various DACs I have lying around. All but of them will clip the signal,, showing a distortion spectrum similar to yours.

Edited: In re-testing - all of them exhibit clipping when digital volume is set to beyond -3 dBFS

Bubbamike's picture
Quote:

This is the kind of scientistic nonsense that's so common this world--a just-so story (ad-hoc fallacy) that attempts to explain subjective impressions via nice stories or casually observed phenomena while never subjecting those claims to serious tests.

Waiting for these kind of tests for MQA that you and the others at Stereophile seem to have swooned over without ever conducting any scientifically serious tests.

John Atkinson's picture
Bubbamike wrote:
Waiting for these kind of tests for MQA that you and the others at Stereophile seem to have swooned over without ever conducting any scientifically serious tests.

Starting with the January 2018 issue, we are publishing a series of articles examining MQA's claims one-by-one.

But "swoon" - I respectfully suggest your comprehension has been a wee bit colored by what you might have read from ill-informed hot heads on the InterWebs. :-).

John Atkinson
Editor, Stereophile

Bubbamike's picture

Sorry, but I think swoon is a fair description. Not the first time, even you admitted, just an issue or two ago that you and the magazine over empathized the effect of HDCD. I think this MQA mania is of the same enthusiasm. I have never seen a negative word applied to MQA from the staff of Stereophile. Perhaps I'll be surprised by your January issue. I hope so.

tonykaz's picture

There aren't any, are there ?

Sure, there are negatives. Everything has a negative of some sort even if it's quite minor.

MQA's prominent negative is that so many other things have been heavily promoted but not achieved universal acceptance that dubious consumers are remaining skeptical.

After all, isn't Vinyl 'still' the finest music Format for the group of Audiophiles that began during vinyl's Big Era? A good many of these folk remain Digital Deniers although even 'that' position is getting difficult since 2009 or so ( when HP of TAS was started blessing digital stuff ).

Another negative is Bob Stuart and Meridian. I've loved Meridian and Boothroyd Suart since the 1980s but I feel very much alone in this.

MQA being British is not a good thing for us Yanks, why couldn't one of our sharpies figure this out, someone like Edison or Einstein. Dam it, why another Brit thing?. That Linn guy and that LP12 was probably all the Brit we could take. We're the RedWhiteBlue Team and we deserve to win one, don't we?

Maybe the worst part of all this MQA business is that 'our' Warner was the first Record Label to get on-board. phew. Now it seems everyone is in a hurry to do MQA stuff. ( except some Neanderthal outfits that steadfastly refuse to advance into this 21st Century, I won't mention any names except Shit. There are a few more. )

The MQA Positive for those who remain negatives: Noboby has to buy it and nobody will hold it against you, MQA is just better RedBook. As far as I can tell.

Besides, if a person can't hear any MQA difference it only means that they have lesser gear or hearing ( like an old geezer ).

Ancient Tony in Michigan

Bubbamike's picture

You realize that MQA starts out with a high resolution file? It isn't meant to replace Red Book but to allow the streaming of Hi Res files over the internet with reduced bandwidth. Well among other issues, such as DRM and loss of data. But that you didn't know that is an indictment of Stereophile's coverage of MQA. If you look around you'll find explanations and critiques of the method, as well as Bruno Putzey's recent criticism of the lack of reliable tests of MQA.

tonykaz's picture

I accept, MQA is high resolution transmitted via RedBook.

Loss of Data, hmm, is that true?, maybe folks like Bruno Putzey will explain these things or someone else.

So far, MQA seems to be blessed but not universally adopted.

If I were making gear, I'd accept MQA.

As I stand today, I'm happy with RedBook and happy that I don't have to have a vast vinyl collection to enjoy music.

I feel like I'm winning and I'm not letting anyone take that away from me.

Tony in Michigan

arve's picture

I accept, MQA is high resolution transmitted via RedBook.

It's not. "Redbook" is entirely specific to data stored on an Audio CD, and governs all topics related to storing audio on that CD, including data structures and data encoding. The sample rate is only one such aspect. A file stored on a computer or audio streamed over a network can never be redbook.

But, the point you were trying to make was probably related to 16/44.1 audio being stored or streamed. Which is also entirely wrong for MQA. MQA uses a 24-bit container, rather than a 16-bit container.

Loss of Data, hmm, is that true?, maybe folks like Bruno Putzey will explain these things or someone else.

Yes, the data loss is true. John Siau has already done this dissemination of the format for you: https://benchmarkmedia.com/blogs/application_notes/163302855-is-mqa-doa

tonykaz's picture

Ok, I just had a brief look at that Benchmark Media report on "is mqa doa".

Firstly, thank you for pointing out that its a 24Bit container, didn't know ( or probably care ). Egads, can I accept or embrace 144db of dynamic range?, seems way too much ( even on a Battlefield Re-inactment )

Lossless? Sure, it looses the old file as it folds it up. I don't know what to think beyond that. Is it like flour stops being flour as the Pizza is made?

MQA is a just a Streaming System, isn't it? MQA is for the Record Company and the Streaming Company, it's not something built into the CD that we buy, as far as I can tell.

So, the Record Company & MQA devise a way to distribute their owned music to us 'Renters' of their convenient listening system.

We can create our own tiny SD memory cards and own our own Astel & Kern players and not bother with Streaming.

We can also collect vinyl and own vinyl playback gear.

We can have Tape Machines and buy Tape from Acoustic Sounds.

We seem to have a wide range of options.

MQA is just another option.

I don't see the reason for all the fuss.

Tony in Michigan

ps. as far as Streaming listeners are concerned, that Newspaper study showed how people had a hard time hearing 320 being different than hi-res.

Camilo's picture

Arve has proposed a very pertinent and concrete suggestion with regards to your test routine of D/A converters, and its merit and rationale very much deserves your consideration.

The suggested test would allow to recognize the innovation made by Benchmark with regards to intersample overs, and would allow to document and evidence the performance of other D/A converters in this respect.

It would shed more light into the differences between D/A converters, and thus provide a better resource of information for consumers, which is - I believe - the objective of carrying out measurements as well as the mission of a dedicated consumer product magazine.

Best,

Camilo

John Atkinson's picture
Camilo wrote:
Arve has proposed a very pertinent and concrete suggestion with regards to your test routine of D/A converters, and its merit and rationale very much deserves your consideration.

We used routinely to examine digital filter headroom, as you can see from reviews on this site that were originally published in the early 1990s. But I admit that was before the Loudness Wars, when CDs were mastered so that there were never consecutive samples at 0dBFS.

Modern digital audio workstations do calculate the waveform on the assumption that it would be processed by a typical digital filter and I have looked at some modern CDs. Inter-sample overs are rarer than many people believe, though I have found a few pathological examples.

But yes, I think I will start looking again at a DAC's digital filter overload characteristics.

John Atkinson
Editor, Stereophile

arve's picture

Inter-sample overs are rarer than many people believe, though I have found a few pathological examples.

This is a bit preliminary - as there are still non-music (and much quieter) data in the corpus (voice memos, audio books, podcasts and a couple of test tracks), but I have examined a corpus of 14411 tracks for intersample overs. Of those tracks, 6082 tracks reported true peaks below 0 dB, 1237 tracks has true peaks exactly at 0.0, and 7092 tracks have intersample overs. In other words, very roughly half of the tracks contain such overs.

I have not tried to do a more qualitative analysis of the overs, such as how many overs there are in a track, or counting whether some of them have overs that last more than one sample.

SeanS's picture

Hi John,

Eagerly anticipating the MQA article(s).

As an audiophile, I have to say I am not interested in the benefits of smaller file sizes for hi-res files--the origami, etc.

What I am really interested in is what, if anything, MQA can offer to better sound quality vs. a well implemented hi-res AD-DA conversion chain that doesn't use MQA. Like, for example, if you simultaneously created two 24/96 digital recordings of a jazz band using the same recording and playback equipment, with one recording having MQA enabled throughout the chain, and the other without, could you tell the difference? I would really appreciate your expert opinion coming from this perspective.

Thanks,
Sean

NeilS's picture

With all the expert technical scrutiny, even reverse-engineering that MQA has been subjected to on sites like computeraudiophile and Archimago this year, it's hard for me not to get the sense that MQA and its related claims are already technically very well understood by anyone who wants to know.

So when I read about Stereophile announcing at the end of October 2017 that it plans to run a series of articles beginning in 2018 on MQA to test its claims, it sure seems like a lot of effort, but it also seems to me like a lot of effort a lot too late.

Raving about/"swooning" over MQA before testing its claims may be a reminder that sometimes, like putting on socks and shoes, the order in which things are done matters.

tonykaz's picture

Paul McGowan of PS Audio is now doing a Question & Answer Series on YouTube.

Paul just answered the Question about different sounding DACs.

Egads, he's taking the time to answer all kinds of Audiophile questions.

This is well worth checking out.

He even tried to reason thru StillPoints.

Tony in Michigan

Camilo's picture

Paraphrasing the sentence that immediately links to your 5th footnote, I would say: "If you're REVIEWING 24-bit DACs and hi-rez downloads, you'd best get your noise level down to where you can hear at least some of that extra resolution, and that's harder than you might think".

What surprised - and outraged - some who read your review, is that you failed to hear a difference between the Benchmark DAC3 and the PS Audio Perfect Wave DAC, which leads to conclude that if there is a difference, it is not audible and doesn’t matter.

Although your 5th footnote hints a clue to why you - no matter how hard you tried - couldn't hear a difference between the Benchmark DAC3's 21 bit state of the art resolution, and the rather poor 17bit performance of the PS Audio Perfect Wave DAC, your review fails to take into account precisely that advice offered by your own footnote.

That advice obviously draws attention to the even poorer performance delivered by the PS monoblocks, acting as bottleneck when it comes to rendering the 21 bits of resolution of the Benchmark DAC and even the 17 bit performance of the PS Audio DAC, let alone their difference.

As Benchmark's John Siau wrote in a application note back in March 2014: "Anyone who thinks they can hear the difference between 16-bit and 24-bit digital audio through a "17-bit" power amplifier is fooling themselves."("What is high resolution audio? - part1" https://benchmarkmedia.com/blogs/application_notes/13174001-what-is-high...)

Nevertheless, the lenghty passage you dedicated to the efforts taken to hearing the difference between the Benchmark and PS Audio DACs, can suggest to some that this your flawed and misleading conclusion are not not merely innocuous omission or mistake. You appear to ignore your own advice, and completely disavow the fact that your monoblocks aren't capable of delivering even 16 bits to your speakers.

I think it is fair to criticize your review and to dismiss your conclusion regarding the audible difference between the Benchmark DAC and the PS Audio DAC. I also believe it is fair to demand that in the future, stereophile reviews take into account basic specifications - which I want to believe reviewers understand but apparently and consistently fail to apply.

I also believe it is impossible and unacceptable to excuse the obviously flawed attempt to establish a difference between the mentioned components and the consequently misleading conclusion, with a statement like: "Yes, listening is what Stereophile reviewers do." If people read reviews here and elsewhere, it is based on the expectation that they will be offered more than casual listening impressions that completely ignore basic science, let alone the specifications of the components reviewed.

It is in this particular case, quite obvious that you need an amplifier that matches and even exceeds the performance of a DAC to deliver the resolution the DAC has to offer to the speakers, yet that was blatantly ignored by the review. Worse, to reinforce the apparent ignorance - I will nevertheless refrain from concluding foul play or second intentions here - of the reviewer with regards to importance the specifications of the components used have for his review, he proceeds to write a lengthy passage showing what lengths he went to in order to hear a difference that the equipment he used is clearly unable to render. This is also not the only review to be found on Stereophile or Audiostream, with this exact same flaw.

This is poor work, and ultimately undermines the credibility of Stereophile, as well as the effort made by John Atkinson to measure equipment as a way to offer transparency and accountability with regard to manufacturer specifications and in many cases supplement the lack thereof.

I am in my mid fourties and, after thousands of hours behind a drumkit, exposure to loud environments and extended listening periods, cannot argue to hear extremely subtle differences. But even having a well trained ear as a musician, I would not be above relying on the due diligence of reading the specs and setting up valid review, audition or test conditions first, and before I make conclusions regarding the audible differences between two audio components or recordings.

I own a Benchmark DAC2 DX and have owned a DAC1. I had the chance to audition them both side by side with recordings that are known to have intersample overs, and I could very much hear the difference using my Sennheiser HD 800s.

I would hope for a clarifying response from you, with regards to taking into account the components used for a valid review of the performance of specific components. I would also like to see a response from John Atkinson with regards to the very pertinent suggestion made by arve - further up in the thread - with regards to introducing a test that takes into account intersample overs. Atkinson made no remarks to this excellent and concrete proposal, and instead threw in the infamous MQA topic, which distracted from arve’s suggestion.

Updating measurement routines as components introduce new improvements and features that need to be account ted for in order to do fair comparisons between components, would only benefit the objectivity of the measurements and reliability of what Stereophile publishes as such.

Benchmark has clearly separated itself from all other DAC manufacturers I have knowledge of, by effectively dealing with the audible downside of intersample overs, and has clearly introduced a substantial improvement to sound quality that sets new standards. This can only be acknowledged by introducing the corresponding measurement to account for this innovation and the – recording industry – problem of intersample overs, whose existence has been acknowledged and documented, as well as as put forward by this review.

Best,

C. Rodriguez

Robocop's picture

I have owned both the DAC1 and DAC2L over 15 years. This review actually tells me little I havn't already read from other reviewers and the Benchmark web site.

What I find most disconcerting is the DAC2 was the current reference DAC for Benchmark which I own.

Why was the DAC2 not compared directly to the new reference DAC3 in listening sessions?

Comparison with the DAC1 is obsolete, well surpassed by the DAC2.

"How much audible improvement do these changes add up to? "I'm quite certain that there should be no audible difference between a DAC2 and a DAC3 given a single pass through the converters," Siau wrote to me in an e-mail."

What does this mean from John Siau Benchmark? Is he saying no audible improvement over the DAC2?

If it sounds the same, well why bother!!!

I really want to know how much better is the DAC3 over the 2 to justify its increased purchase price.

There must be a sound improvement, the Sabre 9028 chip is alone a sonic upgrade over the 9018. This must be audible and not just measured.

It is at the end of the day all about the "SOUND" compared to live instruments in an acoustic space.

Robert

Sigh's picture

It seems that the 9028 is the new 9018 that might be discountinued soon, I can see why Benchmark made the change even if it offers only a minute improvement that isn't audible. Maybe they should have called it DAC2.2 (Although they did have a little update to the DAC2 called 2.2 already) or DAC2+ like Mytek with their Brooklyn+.

To compare the PS-Audio to the DAC3 in a set up that would have made some sort of sense one could run the analog-out of one DAC3 into the analog-in of another DAC3 :) Is the reviewer suggesting that this too would have sounded exactly like a single DAC3 on its digital volume control?

I've owned every Benchmark, DAC1, USB, PRE, HDR, DAC2 and changed more for the new features (USB, remote, second line in) than the negligible sound differences. The one thing I didn't like with the DAC1s was that they ran hot. With the DAC2 I feel like they reached a perfect product, great interface, hybrid gain control, runs cool, asynchronous USB.

It probably makes no sense to trade a DAC2 for a DAC3, yet at the same time it would make no sense for Benchmark to continue using a older chip.

I find that whatever my questions are, the best way to get them answered is to ask Benchmark themselves. They answer promptly and won't push a new product onto you.

Charles Hansen's picture

Once again, I am baffled by Benchmark, designer John Siau, and reviewer Jim Austin. The existence of intersample overs has been well known for over 20 years. I'm surprised that JA did not catch this much earlier.

For evidence of this fact, we only need to look at the datasheet for the once-popular Pacific Micronicss PMD-100 digital filter. In the datasheet is a an unambiguous statement:

The PMD-100 has a design attenuation of 1 dB to allow for filter overshoot on transients.

During the '90s when Robert Harley was technical editor of Stereophile, he would routinely show CD players with no internal headroom, and how they would clip the "ringing" (Gibbs phenomenon) on the tops of a 1kHz, 0dBFS square wave. In contrast were other players that used (for example) the PMD-100 digital filter (and many other designs) that provided headroom to prevent internal overload from intersample overs.

While this phenomenon is understood more clearly now, with a better understanding of the degenerate (worst possible) case, this is hardly some sort of "breakthrough" as Siau and Benchmark would have us believe. In fact it's more surprising to me that he was previously unaware of a well-known issue regarding digital audio playback.

Camilo's picture

Knowing the problem does not automatically guarantee there to be an immediate solution, and blaming those who finally solve a pervasive and known problem for not having solved it before and for taking credit for it, doesn't seem like the right place to aim your criticism at.

The solution for the audible artifacts of intersample overs that Benchmark came up with, is not something trivial, as it is an inherent flaw of D/A chips including the ES9018 and ES9028PRO used in the DAC2 and DAC3, respectively:

"Every D/A chip and SRC chip that we have tested here at Benchmark has an intersample clipping problem! To the best of our knowledge, no chip manufacturer has adequately addressed this problem. For this reason, virtually every audio device on the market has an intersample overload problem. This problem is most noticeable when playing 44.1 kHz sample rates." ("Intersample Overs in CD Recordings"-John Siau https://benchmarkmedia.com/blogs/application_notes/intersample-overs-in-...)

Benchmark had to develop and implemenmt a solution that is not an option provided by the D/A chip, but which addresses a common problem that originates in D/A chips:

"It is possible to build interpolators that will not clip or overload, but this is not being done by the D/A and SRC chip manufacturers. For this reason, Benchmark has moved some of the digital processing outside of the D/A chip. In the Benchmark DAC2 and DAC3 converters we have an external interpolator that has 3.5 dB of headroom above 0 dBFS. This means that the worst-case +3.01 dBFS intersample peaks can be processed without clipping. We also drive the ESS D/A converter chips at -3.5 dB so that no clipping will occur inside the ES9018 and ES9028PRO converter chips." ("Intersample Overs in CD Recordings"-John Siau https://benchmarkmedia.com/blogs/application_notes/intersample-overs-in-...)

It is not out of mere negligence that Benchmark, or other manufacturers who rely more on the D/A chips - and who in some cases even list the specs of D/A chips rather than the measured performance of their components, and thus not the result of their implementation of D/A chips - haven't offered an effective solution to the intersample overs problem before. Being Benchmark the ones who came up with it,
hardly makes them the target to blame for said negligence, let alone undeserving of the merit for effectively addressing the issue of intersample overs.

If anything, it is other manufacturers who have not yet implemented a similar solution who could deserve some criticism in this respect.

As to the developing and final implementation of a solution to intersample overs, it is debatable if - according to your appreciation - it took the industry too long. It is however not debatable that Benchmark's solution is very much a welcome breakthrough for the performance and sound quality of D/A converters, and one that clearly sets them apart from all other manufacturers.

SeanS's picture

Wouldn't Siau's solution have to sacrifice dynamic range in the digital domain? He says they then bump up the 3.5db in the analog domain to bring the level back up. Definitely not perfect. Sounds like your trading dynamic range on all input data in order to correct the consequences of "bad" input data. On the other hand, maybe the amount of dynamic range lost is less than the noise floor?
Sean

arve's picture

Due to thermal noise, no DAC can achieve the full 24 bits of dynamic range inherent to such program material - at _best_ the analog outputs of a DAC will offer a bit above 21 bits of signal, with the three lowest bit drowning in the thermal noise. By adding 3.5 dB of attenuation to the signal, Benchmark is throwing away approximately the lowest half bit of the signal - this is signal that was already lost because analog electronics won't let us do better.

The loss in level is made up for through analog means adding the 3.5 dB of gain back after conversion. The risk here is that adding this gain causes the noise floor to rise, and thus lose dynamic range. However, as seen from JA's measurements, this is not the case with the Benchmark: The analog outputs have class-leading state-of-the art performance with regards to the noise floor.

SeanS's picture

Arve, the content of your message seems to elaborate on the thoughts I was having when I posted.

Throwing away the last half bit of the signal = throwing away dynamic range in the digital domain.

However, the low level thrown away was already lost because of the physical limitations on analog electronics.... = it is below the noise floor.

I was hoping to get opinions on the solution.

Sean

nikolaos's picture

First of all I'm a huge fan of Ayre and had several components from you and always enjoy whenever I hear Ayre gear. I'm also a huge fan of Stereophile and consider the mag.. as very serious. I will also say I enjoyed reading the review of DAC 3.

Regarding your comments (sorry I'm not english so forgive me for errors in the writing) I had to reply regarding things that where said.

I had lot's of equipment and tested a lot and my conclusion is that whatever the inputs or cables has to say to what I'm hearing does not even come close to other factors like room, components and the ultimate headroom in the system itself.
I had some quite expensive cables and I can for sure not say some where better than others even if the price difference was huge. My concusion is that it is so minor differences in cables that you will gain much more elsewhere in the system. I have heard amazing systems with cables that where the cheapest and I heard the worst systems with the most expensive cables.
I had simelar experience as the reviewer regarding hum with unshield Nordost cables but as Charlie say as they are balanced it can not happen. I guess the system was not truly balanced.
I agree that the reviewer or the editor maybe should have picked that up.
Regarding Benchmark it is well know that they have made great products for some time and as J.A show in the meassurements of this DAC it is a well designed product. I don't understand why a DAC should cost as much as some charge. AYRE used to have their top of the line DAC in around the same price range as Benchmark. You say Mr. Hansen that the DAC 3 is made of cheap components and it seams like you consider Benchmark bellow average quality.

I don't buy to much into all that voodoo anymore about lot's of the claims in the high-end industry. Not to say that from time to time some come up with new and better solutions.
Audio is to me much more about measurements and real world facts. Even if the reviewer did something not considered right by connecting one DAC into the other I'm not so sure it would be so easy to hear coloring of the sound. It would for sure act as a filter but so does a preamp.

As said I'm not English and maybe I'm stupid but I did not get the problem you had with how the reviewer explained how he connected the output of the DACs to the preamp.

It is a little industry. Let's enjoy the music and be friends. Everytime I meet someone with the same hobby it is always a pleasure. In a forum things are very unpersonal and I know people would not act like this when met for real.

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