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Review and Measurements of PS Audio PerfectWave DirectStream DAC

Music1969

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Anyway, I don't see the point to talk about the guy himself and his degrees...
The DAC is bad. Period. Nothing has to turn personal.

Amir: "Ted Smith is the designer behind this DAC (a software guy who used to work at Microsoft the same time I was, turned hardware designer)"

We are reviewing Ted's DAC - it's not just a software guy designing a DAC. Ted also studied Electrical Engineering... that's the only reason I added this...
 

Sal1950

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Of course it is an older model PS DAC, but when I kept reading the industry reviews about how all of the multiple new firmware updates of the PS DirectStream DAC were improving its sound each time so dramatically and marvelously, I began to get a little suspicious about the whole thing.
What made this an "older model"? It was a brand new unit running the very latest firmware?
 

Cortes

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In those years, before the Bologna Plan (90's, EU), engineering in Spain lasted 5 years, and only 3 years the techniques. Architecture and Medicine 6 years. And the final year project involved about a year. Now are only four years the and pfc is a work of little demand. And in some Schools (Escuelas), to be able to present the PFC, you must to have the level 9/10.

The level of academic demand has fallen dramatically in Spain, even worse every year. I think the same in the rest of the EU.
It is the same in the whole world, not only Spain or EU. Only exception I know is Korea (South).
 

TomJ

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PS Audio recently dropped all its US dealers on short notice. Some dealers responded by unloading their DS inventory at $3,000 each (new, sealed box). Haha, no thanks. Likewise their S700 monoblock amps, where JA measured 877mV (?!) RF noise on the output jacks. Won't be buying those either.
 

Cortes

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I worked with an MIT CS graduate some years ago, and the guy wasn't anything exceptional. So if you're putting that out there as some sort of indication of excellence, it's not in my experience - If they can get passing marks on the courses, they get the degree. It's just that simple, nothing more and nothing less.
MIT's (and Hardvard) demand level for undergraduates is a joke. There are many places in India with students better formed.
 

AnalogDE

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Doesn't matter where the guy studied. If you spend your whole career doing software, then turn around to making DACs..... the result won't be good. LOL
 

ObjectiveSubjectivist

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Hmm would love to see how other Chord products perform. Hugo TT/TT2 and Dave. Fingers crossed :)
It seems that in all this audiophile talk and snake oil, Chord is the one of very few companies that produce gear that performs as it should.
 

fredoamigo

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MIT's (and Hardvard) demand level for undergraduates is a joke. There are many places in India with students better formed.


a brilliant, intelligent engineer, with many diplomas, who designs a defective device with catastrophic measures is much less forgivable it is much closer to an usurper because he knows what he is doing ... the incompetent them, I could forgive them...
 

maty

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MIT's (and Hardvard) demand level for undergraduates is a joke. There are many places in India with students better formed.

In India, as in China, there is a wild competition to enter at University and after to some Schools and Faculties, so the students who enter are the best trained and intelligent.

In Spain there is a generalized exam of access to the University, regardless of the Faculty or School. Approve more than 90% of those who finish high school and show up, with that being said. Since 2000 the industry has gone from 18% of GDP to 13% in Spain -> country of services, low salaries. And also, more and more mediocre politicians, as in the West. Mediocrity is a merit today.

In 1970 Spain had the technology and technical training to build an atomic bomb. Today I doubt much of the latter.

- End off topic -
 
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TomJ

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To be fair, Ted Smith is a smart talented guy and deserves a lot of credit for his initial design idea to convert PCM to DSD using algorithms microcoded in FPGA and then use a simple low pass filter to convert DSD to analog, for the purpose of eliminating the PCM reconstruction filter which is still the biggest problem with sigma delta. An example of Ted's good work is how neatly he solved the source jitter problem, as Amir confirmed. And Paul deserves credit too for giving Ted the opportunity to run with it. It's too bad how it turned out. They painted themselves into a corner with their early hardware decisions that left them with more noise than anyone would want in a DAC, and then more hardware problems in the DS Jr which runs so hot because it doesn't have enough heat sinking for its voltage step-downs. Along the way, cleaner solutions emerged for converting PCM to DSD without any hardware complications, like HQPlayer software. What does surprise me however that Paul went ahead with product releases that weren't yet fully baked (or too baked, in the case of the DS Jr), like the hardware noise in the DS and more RF noise in their recent ICEpower amps than you typically find in nCore amps.
 

Interference

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To be fair, Ted Smith is a smart talented guy and deserves a lot of credit for his initial design idea to convert PCM to DSD using algorithms microcoded in FPGA and then use a simple low pass filter to convert DSD to analog, for the purpose of eliminating the PCM reconstruction filter which is still the biggest problem with sigma delta.

Could you further elaborate on this?

What is the advantage of using a FPGA over a standard sigma-delta modulator? In my understanding, both are able to provide a DSD-like stream. Aren't DSD ADCs basically analog-driven sigma-delta modulators, after all?

I don't see different implications for what concerns the filtering stage. What prevents you from using a "simple low pass filter" after a SDM?
 

dropbear

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I just...feel bad for those who bought...this. The worst thing is, though, that I am not angry, I am merely disappointed, all the time I thought that guys buying products like this may be paying for 800% margins, but they at least get what they paid for, an over-built quality, but as it seems, that is not a case at all, so many extremely-priced products, which magical unicorns under the hood being debunked make me, again, sad for people who bought them for whatever reason...

Let me express my compassion with those blokes with a second of the pure silence they would never get from this product.... .... ..... ......done, let's get back to a work.
 

beefkabob

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Now you have to test their amp and preamp (stack?) to see if they're garbage too. I'm guessing they're not so hot, but even Schiit makes some decent products.
 

TomJ

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Could you further elaborate on this?

What is the advantage of using a FPGA over a standard sigma-delta modulator? In my understanding, both are able to provide a DSD-like stream. Aren't DSD ADCs basically analog-driven sigma-delta modulators, after all?

I don't see different implications for what concerns the filtering stage. What prevents you from using a "simple low pass filter" after a SDM?
To your first question "why FPGA", the advantage of an FPGA solution is that it has the potential to side-step the need for a reconstruction filter. The ever increasing number of PCM reconstruction filters is a good sign that an optimal filter design is still an unsolved problem. It's very interesting how many different paths have been taken in recent years to avoid the need for that filter. Examples include what Rob Watts is doing at Chord with his enormous filter tap lengths (eg the M-Scaler), Jim Kinne and team at Exogal (poor them, now in a corner with the obsolete FPGAs in the Comet), and of course all the different implementations of R-2R ladders. Of all these approaches, it seems that Chord's has the greatest potential. Rob is now saying that 44/16 audio files processed by his algorithm using one million plus coefficients (filter taps) sounds better than 192/24. That's fascinating, that he can now reconstruct complex analog waveforms in real time so accurately.

To your second question "using sigma delta to convert to DSD", that would require the use of a reconstruction filter for error correction of the sigma delta approximations, so what you're converting to DSD has already been compromised ("contaminated") by the reconstruction filter you've chosen.

To your third question, the answer is the same as the paragraph above.

T+A has an interesting dual solution in their DAC 8 DSD, where as you know they've implemented two independent circuits, one sigma delta decoder for PCM input, and a simple high pass filter for DSD input. It's too bad they chose to use a hi shelf resonant filter class for their PCM Bezier curve option, perhaps to differentiate themselves in the market but at the cost of that HF bump before the roll-off.

Re my comment about releasing half baked products, I didn't mean to be disrespectful of anyone but I do wonder sometimes whether Paul and his team understood how they were painting themselves deeper into corners with each of the three products I mentioned in my post, and whether they'll later regret that they compromised product quality and even their own reputations in doing that.

- cheers, Tom

PS - In retrospect, “side-step the need for a reconstruction filter” was a poor choice of words, eg “to allow for a less stringent or severe filter” would have been better. With delta-sig filters we’re still choosing between optimal freq response (brick wall) at the expense of temporal response (pre-ringing), or optimal temporal response at the expense of the HF attenuation slope. Maybe DSD is the best solution for now with its linear filter?
 
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Music1969

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Of all these approaches, it seems that Chord's has the greatest potential. Rob is now saying that 44/16 audio files processed by his algorithm using one million plus coefficients (filter taps) sounds better than 192/24. That's fascinating, that he can now reconstruct complex analog waveforms in real time so accurately.

Beyond the fancy maths/code, it would be most fascinating to see if/how Rob's M-Scaler makes any difference at all, to the analogue output of the already superb measuring Qutest DAC...

https://www.audiosciencereview.com/...ew-and-measurements-of-chord-qutest-dac.5981/
 

MZKM

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ahofer

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I’m curious: since this design successfully eliminates jitter and other nasties upstream from the DAC, do they still recommend incredibly expensive USB cables and sources? Or can we hook up a discman?
 

Music1969

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Or can we hook up a discman?

It's been verified by Amir to essentially be jitter immune, so if your discman has bitperfect optical output, it can't get better.

No jitter, no RFI, no ground loop potential.
 

maty

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Of all these approaches, it seems that Chord's has the greatest potential. Rob is now saying that 44/16 audio files processed by his algorithm using one million plus coefficients (filter taps) sounds better than 192/24. That's fascinating, that he can now reconstruct complex analog waveforms in real time so accurately...

Comment on these forums that one is able to differentiate between MP3 320 kbps vs FLAC 16/44 or 16/44 vs 24/96-192 is a sacrilege for a few.
 
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