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Review and Measurements of PS Audio PerfectWave DirectStream DAC

Interference

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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.

How an FPGA allows you to side-step the need for a reconstruction filter? This was the meaning of my original question. I would be brought to think that no matter how complicate you get the machinery, you will always come out with a digital signal that needs a low pass.

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.

This is some heavy convolution after which you will still have a digital stream in need for a DAC and reconstruction filter, so it does not really answer my question. Even R2R DACs require a reconstruction filter.

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.

From the block diagrams I saw for SDM DACs, the feedback loop of the modulator closes before the output filter, so this is quite unclear as well. Could you elaborate further or provide me a reference?
 

solderdude

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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.

How do you see that happening ?
 

AudioSceptic

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I guess the real problem with Chord is the pricing bracket they target and their choice of cozying up with subjectivist mysticism.

But as previous Chord devices measured here have shown, they all seem to measure and perform quite well at least for their generation of technical implementation. The current generation product, the Qutest actually still ranks quite high up in the range of DAC's measured here.

How good are they "audibly" is subjective too since we all hear them differently. Not forgetting that dynamic loads and static measurement loads can yield different results, be it slight or dramatic. It definitely bears out for my area of work with regards to mechanical implementations, so I'm confident they are analogous with electronics too.
There was a graph from @amirm for the THX789 that somehow showed the THD+N to swing once connected to a headphone or something.
Its something that's quite hard to consistently duplicate or measure, so perhaps its one aspect of the testing that ASR could put everyone's expertise together to devise a suitable test rig for such dynamic loads.

I have heard mumblings from subjectivists with regards to the THX 789's sound signature being pretty 'dull / dry / thin' but yet gush profusely over the JDS Atom's. Both being excellent, if not exceptional measuring headphone amps here on ASR. Which is quite puzzling if you ask me and the only postulation I can attribute it to is their varied response to dynamic versus static loads.

I do like the Atom which I have but have yet to do a A/B with the THX 789.

I have tried both the Qutest, Dave+Blu MkII and the Hugo2. They sound really nice to my ears. Only issue I have with them is the price they are asking wrt to the various other well measuring devices that have been uncovered here on ASR.
That's pretty much what I'm saying. Not all pricey stuff is rubbish, but it's still overpriced for the performance. In the case (!) of Chord and a few others, you can see that the enclosure is expensive and that is something that the buyer must factor in (audio jewellery), but how much of the money goes into what you hear, compared with what you see?
 

AudioSceptic

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We'll see. Hope it is since Chord touts their measurements so much in their marketing. Chances are it might not even beat the current king Matrix X-Sabre MQA DAC since Chord is custom FPGA implemented. Based on empirical data, those who know how to truly maximize off-the-shelf DAC chips are the ones who measure extremely well. Any further complications/implementation just deviate from the absolute transparency. That's also why I'm very curious with the Linn Klimax DSM which probably costs about 1.5x DAVE but it uses an off-the-shelf AK4497EQ DAC chip
The Linn Akurate has been measured here and came out fairly well, albeit not commensurate with its price. I'd expect the Klimax to do a little better but not break any records.

Edit: I would pay extra for the Klimax's gorgeous styling and build, but not £10k+ extra!
 
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maxxevv

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Yes agreed.

Even more so for me as I can make a nice case for budget device such as the JDS Atom should I choose to. So definitely don't see the rationale for paying over the top prices just for the aesthetics.
 

w1000i

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We'll see. Hope it is since Chord touts their measurements so much in their marketing. Chances are it might not even beat the current king Matrix X-Sabre MQA DAC since Chord is custom FPGA implemented. Based on empirical data, those who know how to truly maximize off-the-shelf DAC chips are the ones who measure extremely well. Any further complications/implementation just deviate from the absolute transparency. That's also why I'm very curious with the Linn Klimax DSM which probably costs about 1.5x DAVE but it uses an off-the-shelf AK4497EQ DAC chip
The transparency you mean is what ADC recorded. And Chord they don't believe that ADC was done right so they apply there algorithm before playing it.
 

Harry1973

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I have tried both the Qutest, Dave+Blu MkII and the Hugo2. They sound really nice to my ears. Only issue I have with them is the price they are asking wrt to the various other well measuring devices that have been uncovered here on ASR.

Mojo is relatively affordable and it sounds really nice. -I have been sweating to find a filter on Sajab D5 (top player on the SINAD list) that sounds as good. Mojo with an optical cable in a main set is a nice way to find out what Chord is about. (IMO).
 

Rja4000

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A transformer is fully able to provide a balanced output.
That's even one of the goals to include them to begin with.

Which makes me come back to my question above:
Is the transformer also in the path for the Unbalanced out ?
Is that something Amir measured ?
 

pozz

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The level of academic demand has fallen dramatically in Spain, even worse every year. I think the same in the rest of the EU.
I wouldn't put it that pessimistically, but the Bologna "streamlining" process did make the higher education system more factory-like (goal was to increase the number who finish the entire program from Bachelor's to PhD within 8-10 years (many spend that much time just writing their dissertation), and to apply curriculum standardization applied across the EU and make it stricter internationally to ensure transferrability of course credits).

Although this is way off-topic.
 

ahofer

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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.
Not a sacrilege, so much as a proposition with little controlled evidence to support it. I’d be happy if I could hear the difference every time but I can’t.
 
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vert

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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.
That comes as no suprise to me. Some Spanish friends and colleagues have told me the same thing. I finished the equivalent of high school in France in the late 80ies. There were a few extremely smart math & science kids in my school (I wasn't one of them); all of them were told to avoid engineering and go into finance after high school, which most of them did. The French so-called authorities had already decided by then France would no longer have an industry, a process that's nearing completion. I can imagine the same decisions were made in Spain.
 

maxxevv

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Mojo is relatively affordable and it sounds really nice. -I have been sweating to find a filter on Sajab D5 (top player on the SINAD list) that sounds as good. Mojo with an optical cable in a main set is a nice way to find out what Chord is about. (IMO).

Compared to its equivalent ? Say a NX4 DSD in price and performance ?
 

solderdude

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So the high end equipment, I mean TotalDac or this thing, are in par in term of performance.

Perhaps on par in certain aspects, in other aspects they differ.
Both don't measure well but have a fan-base that are quite happy with the audio-result.
At least now we know how it actually performs and that his bla-bla-bla is just that (but most of us knew that already)
 

TomJ

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How an FPGA allows you to side-step the need for a reconstruction filter? This was the meaning of my original question. I would be brought to think that no matter how complicate you get the machinery, you will always come out with a digital signal that needs a low pass.

This is some heavy convolution after which you will still have a digital stream in need for a DAC and reconstruction filter, so it does not really answer my question. Even R2R DACs require a reconstruction filter.

From the block diagrams I saw for SDM DACs, the feedback loop of the modulator closes before the output filter, so this is quite unclear as well. Could you elaborate further or provide me a reference?

How an FPGA allows you to side-step the need for a reconstruction filter? This was the meaning of my original question. I would be brought to think that no matter how complicate you get the machinery, you will always come out with a digital signal that needs a low pass.

This is some heavy convolution after which you will still have a digital stream in need for a DAC and reconstruction filter, so it does not really answer my question. Even R2R DACs require a reconstruction filter.

From the block diagrams I saw for SDM DACs, the feedback loop of the modulator closes before the output filter, so this is quite unclear as well. Could you elaborate further or provide me a reference?

Right on… generalities are seldom helpful in comparison to case study specifics.

For more details on the internals of sigma delta modulation, the Matlab DS toolbox is a great resource
https://pdfs.semanticscholar.org/c45b/0a23369e43e68ccfdcb8b71f894952b8184d.pdf
or the Python port of same which is freeware, no Matlab license required
http://www.python-deltasigma.io/.

For Rob Watts’ approach at Chord, here are two good starting points:
http://tagnz.co.nz/wp-content/uploads/2015/01/Rob-Watts-DAC-technology-for-Asia-2014.pdf
https://www.moon-audio.com/mwdownloads/download/link/id/159/

For the PS Audio Direct Stream design, here’s a brief overview and you’ll find some lengthy interviews with Ted that provide more details:
https://www.psaudio.com/wp-content/uploads/2014/02/DirectStream-DAC-white-paper.pdf

For the Exogal Comet, this is a good introduction:
https://www.exogal.com/images/PDFs/EXOGAL_DAC_and_PowerDAC_White_Paper.pdf

For R-2R, here’s a recent overview and you’ll find lots of software simulations online which allow you to explore different design strategies and step through code with test data
https://pdfs.semanticscholar.org/c314/7d9cbc59f187dcdd99763262f4b3205c3d3b.pdf

FWIW both my DACs are sigma delta so not taking sides, just impressed with all the hard work that these companies and others are doing to raise the bar on playback quality.

- cheers
 

solderdude

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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.

There is nothing in there stating they skip the reconstruction filter.
In fact theirs is really steep (as it should be).
 

Music1969

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Right on… generalities are seldom helpful in comparison to case study specifics.

For more details on the internals of sigma delta modulation, the Matlab DS toolbox is a great resource
https://pdfs.semanticscholar.org/c45b/0a23369e43e68ccfdcb8b71f894952b8184d.pdf
or the Python port of same which is freeware, no Matlab license required
http://www.python-deltasigma.io/.

For Rob Watts’ approach at Chord, here are two good starting points:
http://tagnz.co.nz/wp-content/uploads/2015/01/Rob-Watts-DAC-technology-for-Asia-2014.pdf
https://www.moon-audio.com/mwdownloads/download/link/id/159/

For the PS Audio Direct Stream design, here’s a brief overview and you’ll find some lengthy interviews with Ted that provide more details:
https://www.psaudio.com/wp-content/uploads/2014/02/DirectStream-DAC-white-paper.pdf

For the Exogal Comet, this is a good introduction:
https://www.exogal.com/images/PDFs/EXOGAL_DAC_and_PowerDAC_White_Paper.pdf

For R-2R, here’s a recent overview and you’ll find lots of software simulations online which allow you to explore different design strategies and step through code with test data
https://pdfs.semanticscholar.org/c314/7d9cbc59f187dcdd99763262f4b3205c3d3b.pdf

FWIW both my DACs are sigma delta so not taking sides, just impressed with all the hard work that these companies and others are doing to raise the bar on playback quality.

- cheers

By the way, Rob Watts' DACs are SDM also... 5 bit SDM running at ~ 104 MHz...

So it's not really FPGA vs SDM... I'm not suggesting you are suggesting this but just in case others think it is FPGA vs SDM.

1569245673083.png
 

tensor9

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Keep in mind that with the DS DAC, the FPGA is NOT doing the DA conversion. It only does digital processing. It upsamples to 20x DSD, does all the digit filtering stuff, then it outputs a 2xDSD stream to voltage rails that is low pass filtered by the transformers. They do the “conversion” in a sense.
 

maty

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Not a sacrilege, so much as a proposition with little controlled evidence to support it. I’d be happy if I could hear the difference ever

Some hours ago I made tests with SoX implementation of AP2 - Album Player soft player.
https://www.audiosciencereview.com/forum/index.php?threads/foobar-or-jriver.8364/post-233239

I could appreciate the differences between the three types of filters implemented. Many say they are incapable. Of course, the interface was that Russian program but the engine was JRiver MC.

AP2-Album-Player-config-SoX.png


And also when marking the Ultra High Quality and Steep filter options.

http://albumplayer.ru/english.html
 
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