Phasure NOS1 vs. Pacific Microsonics Model Two

[manisandher]

For a while now, I’ve been saying that there is a DAC on the way that will be a 'game-changer'. Well, a year or so later than anticipated, it’s finally here. The Phasure NOS1 is the brainchild of Phasure's Peter Stordiau (PeterSt on this forum), who spent over two years on its development. And as the name implies, it’s a non-oversampling DAC – actually, it’s a totally filterless design with no SDM or analog filtering stage.

 

The NOS1 currently accepts data rates of 32/384 (though my understanding is that it may be able to accept 32/768 rates in the future). So, how does it achieve this? Well actually, there is only one way of getting data into the NOS1… and that is by the use of a computer. But not by the means of a USB, firewire or spdif protocol. Rather, Phasure apply the clever use a PCI extender to allow them to place an OEM ESI soundcard in the NOS1’s chassis. The soundcard uses ultra-stable clocks with a jitter of only 0.5ps and is simply used to produce a digital I2S stream which is fed directly to 8x PCM1704U-K D/A converter chips, sitting in close proximity to, but screened from, the soundcard.

 

And what about the Pacific Microsonics Model Two? Well, I’d happily spend hours describing this gem of audio design, but I once promised that I wouldn’t write anything more about it on CA. So, if you want to learn a little more about it, just look here: http://www.computeraudiophile.com/content/Pacific-Microsonics-Model-Two. I’ll just say though that I’m using a Weiss AFI1 to pipe data from the computer to the Model Two.

 

So, how do the two DACs compare sonically?

 

The NOS1 is like taking an x-ray of the music. It’s so crystal clear and pure. And yet it’s so easy to listen to. It’s exciting and yet relaxing at the same time. I’ve never been too bothered about imaging, but I’ll just mention that imaging with the NOS1 is super, super sharp. I think people for whom imaging is a big deal would love this aspect of the NOS1. I have no hesitation in saying that the NOS1 is the most accurate DAC I have ever heard.

 

In contrast, the Model Two is more full-bodied. It gives the impression of richer harmonic content. It retains some of the NOS1’s detail but loses its super clarity – it’s simply no match for the NOS1 in this respect. The bottom end is more pronounced and weightier – but then perhaps a little less tuneful for it. It is an incredibly musically engaging DAC though.

 

Over the weekend, we had a friend staying over. Although not a professional musician himself, Paul plays many instruments (very well) and has also spent some time in a professional recording studio. We played a number of tracks that he’s very familiar with. He really, really liked the NOS1. He said that he’d heard things in these tracks that he’d never ever heard before. The NOS1 gave him an insight into the recording/mixing/mastering process in a way that he’d never experienced. After the session, I asked Paul a hypothetical question: He’s just walked into a hifi store which has a NOS1 and a Model Two for sale at exactly the same price. Which one would he buy? He said that he’d want to take the NOS1 home because he knows that his whole collection would sound totally different through it – he would finally be hearing what’s really on his recordings.

 

Well, so much for hypothetical questions. Right now, there are two Model Twos for sale on a popular used gear site. They are $17.5K and $20K respectively. For around 1/4 of the price, you can have a brand new NOS1!

 

However, there is a very important caveat to bear in mind when considering the NOS1. Although the DAC itself is indeed totally filterless, it expects the computer software player feeding it to either interpolate the data or to ‘upscale’ it to a point where any aliasing distortion is well beyond the audio band. The philosophy is then to do no further filtering of any sorts with the data. I’ve used it very successfully with Signalyst’s HQPlayer, using one of HQPlayer’s advanced minimum phase filters set to 32/176.4 output. But not surprisingly, I get the best results by using Phasure’s own XXHighEnd player which is capable of upscaling any data rate (including 16/44.1, 24/88.2, 24/96, 24/176.4 and 24/192) up to 32/384 (and beyond) with zero pre- and post-ringing. At these rates, any aliasing distortion is above 192KHz and well out of harm’s way.

 

I think it's fair to say that the Model Two set the bar for all others to follow when it was released some 10 years ago. And now comes along a DAC that matches it in some areas and handily beats it in all others. That's close enough to 'game-changer' for me.

 

Mani.

 

NOTE 1: I am not affiliated to Phasure in any way, shape or form. I paid the full retail price for my NOS1 (and for my 3 copies of XXHighEnd). I am considering buying a second NOS1 for my office system and again will pay the full retail price if I do so.

NOTE 2: I have no interest in becoming a dealer of any sorts, but if anyone reading this thread is seriously considering buying a NOS1, I would happily offer a demonstration of the DAC in my home. However, it may be better to accept Phasure’s 3 month return policy and listen to it in your own system.

NOTE 3: I have purposely not passed what I’ve written here by Phasure before posting it. If there are any technical inaccuracies, they are purely of my own doing.

 

 

[The Computer Audiophile]

Hi Mani - Thanks for the post. I'm trying to imagine how the interface looks but can't quite picture it in my head. Can you post any photos?

 

[ted_b]

upsample everything to 32/384 (whether via player on-the-fly or via Saracon/Philips/Korg/etc)? What would the computer interface actually be then (i.e does it require a pc with some PCI extender?)

 

I've been searching Peter's web forum and have read your other comments there (nice and thorough, thanks), but there doesn't seem to be an introduction or basic description of the DAC...only that it sounds great and is unique. The external pic (Sonicweld-like photo)looks quite interesting, but again, no real understanding of where the music files reside, etc

 

 

[ziggyzack]

Soundcard in the middle, separate left and right DAC banks?

 

Or, power on one side, DAC on the other, soundcard... still in the middle?

 

I have to say that when I read '8x PCM1704U-K D/A converter chips' I first thought to myself '8x oversampling...? B-b-b-b-but there's supposed to be none!' Then it dawned on me.

 

What is it??? Specs please!

 

Nice one Peter.

 

[manisandher]

Hi Chris, I bet Peter's got some nice photos of the NOS1 that would do it justice. Peter? If not, I'll post some when I have a chance.

 

But you can probably just make out some of the cutouts in the chassis on the photo that ted_b posted. If you look towards the bottom, you'll see a rectangular cutout - this is for the PCI extender connection.

 

Mani.

 

[manisandher]

"I've been searching Peter's web forum and have read your other comments there (nice and thorough, thanks), but there doesn't seem to be an introduction or basic description of the DAC...only that it sounds great and is unique."

 

OK, the way that it's configured in my system is as follows:

- PCI extender output card placed in PCIe slot of computer

- PCI extender cable links computer to DAC

- OEM ESI soundcard placed in PCI extender input card which sits in DAC itself

 

It's a bit like using a Magma PCI extender and placing the whole DAC in the external Magma box, with its own power supply etc.

 

Neat idea. It sounds simple, but I'm sure was a PITA to get working correctly. I know that Peter spent quite a while on all this, and on keeping the noise down to (I think) an unmeasureable level at the output of the DAC. You'll probably understand slightly better now why the NOS1 looks the way it does - there's a lot of isolation and regulation going on in there.

 

Mani.

 

[manisandher]

Here's what it says in the Manual:

 

- 100% Filterless Non Oversampling, using 8 x PCM1704U-K Burr Brown (Texas Instruments) multibit 24 bit D/A converter chips

- Filtering to be applied in software

- Output Impedance is 33 Ohm

- The Clock section consists of replaceable oscillators, separately for the 44100 Hz and 48000 Hz base. The provided oscillators carry an overall jitter of better than 0.5ps and phase noise of better than 120dBc/Hz at 100Hz and better than 108dBc/Hz at 50Hz. Oscillators sit 12cm from DAC chips.

- S/N ratio in-(audio)band is at –132dBFS at full scale (-0dBFS) differential (XLR) output.

Out band (from off 31000Hz and up to 96000Hz which is the measureable limit) this is better than –144dB, and at the limit of the analyser operating at 24 bit and 192KHz.

- Mains related products measure 17uVRMS at –121dBFS (measured at 50Hz mains)

- SE (RCA) output measures in(-audio)band Unweighed 0.0018% THD+N at 16 or 24 bit input and a 1KHz test signal with a sample rate of 44100Hz and Arc Prediction filtered into 176400Hz. The THD+N from Differential (XLR) is beter than 0.0015% which is under the typical specs of the DAC chips.

 

Perhaps Peter has some nice graphs he could show us?

 

Mani.

 

[Sasaki]

This is quite interesting. Thank you for sharing info.

How do you install driver for this?

I guess we can use ESI driver.

 

[ted_b]

Windows pc's or can the newly vaunted Alix and other MDP Linux boxes produce a PCI extender capability....I assume a MAC can't. Also, I realize that the players mentioned, especially Peter's, are Windows-based, but if one were to take the upconvert seriously and do it with Audiogate or Saracon, etc (i.e offline) then we could use any of the chosen OS's players capable of 24/384k right? The downside with offline upconversion.....gonna need to invest in storage.

 

[PeterSt]

Hi Mani - That's some nice sounding post ! Of course I knew half of it already, but that was only half. Well, thank you - and I feel honoured.

 

Maybe it's nice to say outloud that at the time you first heard the NOS1 (well over a year back now), you didn't own a PM Model Two yet; you bought that later and from that moment on it has been quite exciting to wait for what would happen when you'd be able to compare the both. I guess you must have been more confident than I myself.

 

I will make some pictures allright and add some graphs too.

 

Sasaki - Yes, standard ESI driver. And sorry I didn't respond to your email yet (same apologies to others).

 

Thanks,

Peter

 

[sr1329]

Wow. This sounds like a great concept. Why mess with hardware filtering and/or FPGA when you can just use the 95% CPU that goes unused when playing music and use filtering that is upgradeable in plain PC software?

 

I suppose trying this require me to get a PC desktop machine or can one use a mini PCI-E port in a laptop and draw a wire out?

 

Maybe this is the computer to get:

 

http://xi3.org/product_gallery.php

 

I've been dying to find an excuse to acquire one.

 

[PeterSt]

I suppose trying this require me to get a PC desktop machine or can one use a mini PCI-E port in a laptop

 

No, I'm afraid that won't go.

But, PC-Express (card) slot : yes. But, only "officially" for now and it has never been tested yet.

 

On another note : laptops really don't cut it. They are really too slow (never mind the type and processor) and won't give you a means of convenient playback. Ok, the playback itself will be allright, but not the preprocessing needed. Please notice that I am talking XXHighEnd here, and think about a FLAC needing to be pre-converted which really can go well within half of a second for a 10 minute track. A laptop ? could be 30 seconds.

But it's easy to try out; it's not DAC related.

 

Regards,

Peter

 

[ted_b]

No excessive on-the-fly processor load, just a need for larger files.

 

[PeterSt]

Ted, Hi,

 

Of course you are correct, but it is not really a friendly means for playback. It's a bit similar to copying files to special media and play them from there (like SSD, RAMDisk) because it sounds better; who really likes doing that ?

 

So, a player like what's suggested best here, does that all for you and it goes "on the fly". This suggests real time and it will come so to you indeed once the PC is fast enough. I mean, something like half of a second is sufficiently that (IMO). Just play like always, but in the mean time it plays fully from memory and as in the Windows case, just PCM (WAV).

It doesn't matter what (sequence of) conversions were needed, and it's just done for you, including ending up in that media you asked for (set).

 

Of course, all this can be done in advance by yourself too, and the cpu etc. burden for it is not there. But you will be busy copying etc. those files more than listening to music.

 

About things like "up conversion" in general ... this too is not to be underestimated. I mean, my cons from above can be eliminated largely to convert your whole collection in advance, but :

 

a. It will be a kind of fixed (while still many variations exist you may want to try out tomorrow);

b. It still needs the original to be kept on disk;

c. Look at the size of a 32/352.8 album; 2-3GB !

 

I hope this enlightens a bit, but of course not to forget : is takes for starting point these things can go all automatically. If they don't anyway (or let's say when a player plays FLAC and converts real time during playback itself) - yes, your general option will be "do it yourself in advance". But suppose we're only talking about the ever hot subject of FLAC ... who really does this ?

 

All 'n all we could say that maybe the way I approach this all is "taking distance" from things we shouldn't be bothered with. And now this includes the filtering (which is the underlaying subject really) which is not fixed in-DAC, but which is flexible outside it. It really shouldn't imply you suddenly must start copying things. So, it's a concept and I hope to have made it consistent. But now it does need a PC which is sufficiently fast ...

 

Thanks Ted,

Peter

 

[manisandher]

I've written before (http://www.computeraudiophile.com/content/How-can-influence-SQ) that the sound from the Model Two seems very dependent on the AES interface and cables that feed it.

 

I'm using a Weiss AFI1 interface connected to the Model Two with short AES cables with built-in group loop isolators. The Model Two remains the clock master, with the AFI1 slaved to it via a short BNC cable. I believe this setup is at least as good as using a Lynx or RME PCI card. However, I'm not sure how it would compare to a Mykerinos PCI card. It could well be that I have never heard the Model Two working at its best.

 

Mani.

 

[manisandher]

I was sent a PM with a few questions, and thought that others might be interested in my answers.

 

”I'm very used to the Mac environment and I wondered how you see the learning curve on the XXXhighend part on a PC?”

 

I usually find any player that I’m using for the first time really frustrating. And XXHighEnd is certainly one of the more sophisticated players out there. Look, I would be lying if I said the learning curve is easy. HOWEVER… the Phasure forum will see you through - that, I can almost promise you. It’s a very friendly and helpful forum. Peter himself is very active and always helpful, although I’ll warn you now in case you don’t know, he’s a ‘typical’ Dutchman – I love the Dutch (some of my best friends are Dutch - I don't hold it against them) but to some other cultures, they can come across as a little... 'direct', let's say. And of course, you have to learn Double Dutch to understand a word they say - anyone who's read any of Peter's posts on CA will know exactly what I mean...

 

”Peter's posts seemed to emphasize that a laptop didn't have enough processing power to be ideal. Do you have any comment on that from your experience?”

 

The issue here is that XXHighEnd is a true memory player. It conducts any pre-processing that needs to be done before it starts playing. As part of this pre-processing, it will convert all non-WAV formats into WAV before playing. Now, if you’re feeding it a WAV file and not applying any on-board upsampling then there likely won’t be much load. Here, a laptop might be OK. Indeed, I can even use XXHighEnd on my CAPS machine like this... at a very long push. However, with a NOS1, you have to apply upsampling (32/352.8 or 32/384 recommended), unless you like having headaches. A laptop will still work, but you may get frustrated at having to wait a while before anything plays.

 

Many of us have spent a lot of time comparing various setups, and the pretty unanimous verdict is that XXHighEnd works best with a pretty powerful machine with lots of fast RAM. If you look at the forum, many of us use at least i7 machines with 8GB RAM – certainly not gaming machines, but better than most (all?) laptops. (Actually, being a memory player, the speed of the RAM may be even more important the CPU.)

 

”My interest is in going right from the DAC to amp, so the output section matters a lot too.”

 

I’ll let Peter chime in here, as I don’t actually know much about the output section. But I know that Peter has specifically designed the NOS1 to drive power amps directly. Many of us do this and enjoy improved SQ as a result.

 

”I wondered if after a little more listening time, your impressions stay so positive?”

 

I’ll write them up in a follow-on…

 

HTH.

 

Mani.

 

 

[barrows]

I just wanted to thank you guys for all the information you are providing on Peter's DAC and XXhighend. Although I am not currently in the market for setting up an entirely new front end, I do find Peter's approach to computer playback both unique and fascinating, and I enjoy learning about this unusual approach to digital file playback.

 

[SteveChicago]

... how does it sound in Amsterdam?

 

-- Q

 

[1audio]

I'm curious which ESI card and driver supports this sample rate? I checked and the only current pro cards that goe above 96 seem to be the Juli@ and the Maxio. The control panel for those stop at 192.

 

I think the idea of packaging the card and the dacs in a separate box is a very good one. I'm not enthusiastic about PCI express because its a serial interface with a serdes at each end and a not too synchronous communications unlike PCI, but its much faster so that may not affect things as much. Also PCI may well be obsoleted in the not too distant future so we may all be migrating.

 

[Miska]

I'm curious which ESI card and driver supports this sample rate?

 

My guess is none, but at least one common trick is to use a dedicated I2S per DAC, thus one can get 384/32 for example. With a bit of FPGA it is possible to take it to 768/32 by combining four 192 kHz I2S channels into one.

 

I'm not enthusiastic about PCI express because its a serial interface with a serdes at each end and a not too synchronous communications unlike PCI

 

I don't see that as much of an issue, since PCI is a shared bus operating in transaction based burst mode. Both will need some FIFO on the board anyway.

 

Good side on serial systems is better tolerance for length of the bus at high clock speeds. Due to parallel nature, physical length of the PCI bus is limited by the timing requirements. If you look at size of a 64-bit PCI-X (not to be confused with PCI-Express) slot and maximum length for the max 533 MHz clock speed, it is not very practical...

 

 

[PeterSt]

”My interest is in going right from the DAC to amp, so the output section matters a lot too.”

 

and

 

I’ll let Peter chime in here, as I don’t actually know much about the output section. But I know that Peter has specifically designed the NOS1 to drive power amps directly. Many of us do this and enjoy improved SQ as a result.

 

Since I'm asked I guess I have to say something here, but what can be added (apart from the output impedance being 33 Ohms, which Mani told in an earlier post) ?

But I guess I can tell a little story maybe ...

 

At knowing how much SQ will improve at skipping any means of "pre-amp" (but replace "preamp" by "any means in the signal path, up to a single $$ resistor"), it has been my explicit objective right from the start to let it work (regardless of amps and interlink lengths (~~)) "on its own" so to say;

If we skip some years of thinking about everything, and now knowing that the whole physical traject took some two years, I can tell you that it took around two years to do just this very subject well. It is related to nearly everything.

 

People who followed the thread over at Phasure about the first physical steps and how all emerged to a ready product at the end, will have seen that during the time of waiting for new PCBs, parts, sorting out other things, there has been one large big red thread all over : the "gain stage" as I'd like to call it for a general means of getting enough juice into the power amps. And as a side subject (though major) there has always been the "nothing in the signal path" subject, which is a sheer impossible thing. And for those experimenting with the PCM1704 chips ... they are a pain to this regard.

 

While wanting the direct connection (and which really is the explicit advice right now), there's always been de desire for having in some means of volume control. The theoretical possibility for this depends largely on the means of "gain" itself, and each means could you make sit back for a week and think about the now possibilities for that. It never worked, and without exception always degraded sound. Not that new customers would notice that, but that would only be because of not having the reference.

Even when the first DACs were in production, I was still trying out some newly found means with some sort of transformers (but luckily that did not work either, haha).

 

Btw, the fact that right from the start there was a very well working digital volume per means of XXHighEnd, never scared me much for the possible impossibility of not succeeding on the volume matter. This does, however, not mean that any potential customer will like it that way (the breaking windows syndrome), but in the end I choose that "commercial risk" over degration of SQ.

 

In a nutshell this little story -very indirectly- tells about the direct connection to power amps, but what I actually wanted to say is that it was this which took 2 years. Or let's say it really needed that throughput time.

Right now there's a fairly nice 1.5VRMS output for SE-RCA and 2.5VRMS for balanced-XLR which is on the limits of the 2.5V rails (5V plus/minus), allowing for a nice headroom for overshooting as well. No switches anywhere, and the only degrading element in the signal path (serial) is one (high quality) resistor.

 

Of course the implied question (quoted above) assumes a better SQ from a direct connection to begin with. So yes, this is good thinking. But you don't want to know how difficult is to do that right, and to not cause more harm than good. This morning I just wrote a little story on my own forum about me by now having more bass output than an amplified live band can do well with dedicated amps and speakers for the bass player. And the most interesting is : it really needs the software to complete that little part of it. Bits = still bits, but you don't want to know ...

(meaning something like : without the drive of a pre-amp or explicit "buffer" this will be the first thing going wrong; in the very end though, it will be the wobbling bass frequencies which may produce low perceived frequencies allright, but nothing like a wall of bass; low jitter combined with software which "controls", really does that job. Well, as it turned out.)

 

Peter

 

[TimDH]

What seems great here is that Peter is actually exploiting the computer part of computer audio. Can I try to summarize what I think is going on and check with Peter and Mani?

 

1. The XXHigh End massively upsamples. This lets you get an almost perfect impulse response from the DAC- would pretty much eliminate pre and post-ringing. Some high end DACS like dCs and Weiss (medea at 352.8 khz) do this too, but with hardware.

2. XXHigh end uses some proprietary interpolation: "Arc prediction" for a more analogue processing

3. the NOS design avoids filtering done by DAC chips, which Peter dislikes. Again, some high end DAC makers do their own DSP which may result in avoiding the "bad" filtering of DAC chips.

4. By pre-processing the data in XXHighEnd, the computer isn't working hard at the time of playback.

4. The Phasure Dac has extremely low jitter.

 

So I wonder if Peter hasn't (entirely to his credit) recreated much of the hard work done by certain DAC manufacturers, but through software? Is that fair to say? What is hard to know is whether other respected DAC makers such as Metric Halo have incorporated some of these tricks into their hardware designs or not.

 

Also, Mani, Peter, could you comment more on the specs and cost of the PC that you found you need to do the job?

 

Thanks so much,

TDH

 

 

 

[barrows]

I would like to know a little more about the output stage if you are willing to share some details. In general, what type of circuit is used, how do you handle DC offsets, and what gain devices are used (JFETS, MOSFETS, Bipolars, IC opamps)?

 

I still would love to see you develop a DAC which incorporates your oversampling with Predictive Arc, or to license this oversampling code to third parties to use in their designs. With the power available in contemporary DSP/FPGA chips I suspect that this would be possible. This approach would allow people to use any computer, and send the native sample rate to the DAC. I do not doubt that your approach works very well, but it is a little inflexible when it comes to computer choice and interface to the DAC. A DAC with the oversampling built in would be a much more flexible product, and could still allow for programability for different settings.

 

[PeterSt]

I'm curious which ESI card and driver supports this sample rate? I checked and the only current pro cards that goe above 96 seem to be the Juli@ and the Maxio.

 

It is the Juli@ allright.

We can use Miska's guess on how to achieve the sample rate as a good one; It needs programming on the DAC side (that FPGA = Field Programmable Gate Array) to get things together into a 32/384 stream, the software at the other end doing similar - there taking it apart. A switch on the DAC lets it operate this way, that switch being Off making it all behave normally (32/192 for physical input).

 

I'm not enthusiastic about PCI express because its a serial interface with a serdes at each end and a not too synchronous communications unlike PCI, but its much faster so that may not affect things as much.

 

Apart from what Miska again explained very well, the length of the interface runs OK on 10 meters and with a modest cable. Nothing can go wrong either (it's all normally error-checked); Jitter can't play a role here. That's further down stream, the oscillators directly clocking the DAC chips via i2s.

 

Peter

 

[PeterSt]

1. The XXHigh End massively upsamples. This lets you get an almost perfect impulse response from the DAC- would pretty much eliminate pre and post-ringing. Some high end DACS like dCs and Weiss (medea at 352.8 khz) do this too, but with hardware.

 

I can imagine how vague or confusing this all is, but no;

 

There is no "massive upsampling" as such at all. It is only 8 times (for redbook) which is just enough to sufficiently eliminate harmonic distortion from the too large steps for the too low sample rate (44.1 redbook) - while on the other hand transients are not killed. I always try to illustrate this by asking to grab a piece of paper, draw a squared wave on it with only a little rounded corners (like analogue will behave) and now start to evenly cut off those corners by means of a diagonal line from one side (vertical oine of the square) to the other (horizontal line). Do this 256 times (which I'd call massive upsampling) and you will end up with a nice sine. Here is your flute which ever was a violin ...

 

A perfect impulse response is not related to this at all (thinking this is about an impulse which does not "ring" (pre- or post echoes). The means of filtering *is* related to this though, and Arc Prediction Filtering from XXHighEnd (not coincidentally) does not ring one sample.

Let the "not coincidentally" be related to "NOS" which should be just like that, where part of that (from origine) is related to "no filtering". So, the "no filtering" *is* there, because it's really not filtering as such what happens. Instead it is interpolated upsampling (upscaling) which nicely takes care of getting the stepping distortion out of the way. So it filters as should, but it's no filter ...

 

For additional confusement : the DAC really "physically" is NOS, but including the software it needs (previous alinea) it's not. Of course it can be used without it (to your decision), but it would be wrong. Not more wrong than current NOS lovers use NOS/Filterless, but wrong.

 

No other DACs (high-end or not) do this, guaranteed.

The most close will be the Pasific Microsonics Model Two (emphasis on "Two") because of its 24/192 design which allows for a similar "NOS" principle, but with the difference that it contains filtering - we all hoping to have them eliminated by the process, because the process shouldn't be doing really much when fed with "already filtered data".

I'll stop here before it's confusing only.

 

2. XXHigh end uses some proprietary interpolation: "Arc prediction" for a more analogue processing

 

Well, true. Maybe not so much "for a more analogue processing", but for a more 1:1 analogue result;

In the other thread I talked about "head nor tail" when the analogue result would be held against the original file; this (A.P.)) just can. And you don't want to see how much off it all still is ... (this was clearly no commercial statement :-)

 

3. the NOS design avoids filtering done by DAC chips, which Peter dislikes. Again, some high end DAC makers do their own DSP which may result in avoiding the "bad" filtering of DAC chips.

 

I don't think that something like "bad filtering in DAC chips" exists, although quite many can, and for almost as many it can be shut off. But the point is, next comes that DSP chip (or separate filtering chip), and there you again have what I indeed don't like. Mind you, just for theories.

 

4. By pre-processing the data in XXHighEnd, the computer isn't working hard at the time of playback.

 

Assuming you refer to the upsampling process ... it depends on how you look at this;

Mani's outlay on laptops not being fast enough is true; however, it is true only for when music starts, or for when the digital volume changes; you'd have to wait for it "to come through". More to the real merits this description isn't literally correct because of parallel processing and settings dictating that. So, without what you say being untrue, during playback this process happens just the same, because when a next track comes along you don't want silence. So, in a parallel process this happens - during playback itself. You won't notice it. Still for the first notes to play, you'd have to wait for it.

 

4. The Phasure Dac has extremely low jitter.

 

I guess it has. Partly this is because of the best low jitter oscillators available (mind you, for audio and which is all about phase noise), but the largest deal here is the direct connection; as I said in the earlier post, there's just nothing in between those oscillators (each separately shunt regulated) and the chips that can add jitter. So the figures should be net. I must honestly add that so far I had no good means to really measure "figures" as such, and all I can tell that there isn't any way I can find a single spur of jitter by means of graphs. Only when something is wrong (poor oscillator) I can show it, and what shows, shows 120dB down. Says nothing, and let's stick to lacking equipment for it.

 

So I wonder if Peter hasn't (entirely to his credit) recreated much of the hard work done by certain DAC manufacturers, but through software? Is that fair to say?

 

Please allow me to speak for myself, but I really don't think so. Hopefully this shows from my responses above ?

It will be true that The Phasure NOS1 does things through sofware which others do in hardware, but what it does through software is different from anything else. Besides that, it allows you to choose yourself what that "through software" exactly is. You are given a hint by the software provided via XX, but nothing witholds you from improving on it if you think it can. Or more easily thought : just use and compare anything you can find (online (real time) or offline (pre-processed into a new file) and pick your choice. One thing you will have a guarantee about : nothing in the DAC will change whatever it is you do.

 

Let me thank you for putting up questions which really matter; You didn't get the answers you maybe expected, but just for that reasons the questions (or statements) were good.

 

Regards,

Peter

 

PS: I bought a new PC recently; it costed 1000 euro exactly including one 90GB SSD. Not 100% silent, but silent enough. But notice that any more modern PC will do. If there's only no energy saving stuff in there (laptop processors etc.).