XXHighEnd

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Edit, May 2011 :

I just changed the title for its "24/192" part into "24/384", because this is of course what the Phasure NOS1 DAC turned out to be.
I also removed the question mark about 24/192 being the first NOS filterless DAC, because by now we really know it didn't exist. As it still doesn't if we count out the NOS1 (which of course is filterless at all sample rates).

Also, the topic has been moved from the Chatter board to the Phasure NOS1 DAC board.

For those who are new, well, you can try to read it all through, because it really shows the history of how someone like me started to produce D/A converters, which after over two years ended up in the very best DAC existing at this time. Not because I say so, but because by now it replaced all the great names people owned, them people usually gaining a fair net amount of money after putting their old DACs to eBay.
You may ask for your DAC to be potentially replaced whether it has (you could do it in this topic), and or I'll wait a few days for the now NOS1 owner to respond himself but will answer myself if nothing pops up, or I will respond right away if I know it's a "new" one.

Anway, of course all ended up in a DAC which uses completely uncommon technology (interface, less is more approach with hardly anything in the signal path, no switches, NON-oversampling which measures as good as OS but showing the *real* figures, in-software filtering (e.g. by means of XXHighEnd), really zero in-DAC filtering, special differential setup of the 8x PCM1704U-K chips with better figures than even the chips themselves, super low noise and jitter levels) ... all from an external DAC.

But maybe it is more efficient to start reading somewhere at the back of this topic. :)

Lastly for here in this edited first post, maybe it's nice so see some new posts in this topic, because it has been a great time throughout creating it with many of us, still having the nice memories about it. Or maybe I will add some newbies to the topic myself later ...

Regards,
Peter

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Guys,

I must be honest with you. I have been working on other things than software lately ... :)
See below, which as far as I know is the first 24 bit 192KHz DAC explicitly assembled for non oversampling and filterless mode. :prankster:
36 hours ago the champagne was opened because it produced sound for the first time.

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Some technical insight for those interested :

- Balanced passive I/V conversion by means of two balanced DACs (PCM1704) per channel as the base;
- Balanced or RCA out;
- Output 1.5 VRMS (2 VRMS possible);
- Possibility for oversampling to 211KHz with 1.2ps RMS total jitter;
- Possibility for filtering (2 pole Bessel 100KHz);
- Drives long interlinks sufficiently to avoid a pre-amp or other means of buffer;
- S/PDIF connected, and due : directly Firewire connected without soundcard into I2S;
- Terminals for an external clock;
- The most stable shunt regulated power supply imagineable, both channels completely separated throuhgout;
- Everything operating in pure Class A;
- Housing : WxDxH : 43 x 26 x 8.5 cm = 17.2 x 10.4 x 3.4 " worth of 15Kgs weight;
- The various elements not designed by me, but chosen as *the* combination for the best DAC ever (so 1 % credit for me really :innocent:);
- Name I dedicated to this combination : NOS1.

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You may ask yourself : Can I build this too ?
If you are like me, forget it. I drove various people to complete madness and received some burnouts myself. :swoon:
Anyway, it took me over a year thinking about it and "designing" various concepts, 7 weeks throughput of ordering modules and parts, three weeks of understanding, building and getting more parts, and occupuying the dinner table. The (programming of the) Firewire connection still to do.

Background

I didn't start this project well over a year ago to perceive better sound, and I was very happy with my current NOS 18 bit 96KHz DAC (the TwinDAC+). However, since the hires material is on the horizon closer and closer, including the *fact* that a pre-amp should be avoided encouraging for digital volume control - that by itself taking out bits, I got obessed of wanting more. Technically more.
Being a convict of non oversampling filterless DACs, I ran into a problem : 24 bit 192KHz DACs do not exist.
Many months of thinking and designing brought me the concept of a 32 bit 384KHz DAC completely software realtime controlled (yep, the firmware part), of which the hardware arrived May 2008. But after I squeezed out the first sines from it, I coulnd't find enough time for it. :swoon:

While I started this off thinking 32 bits would be needed for a proper digital volume control, underway I created the digital volume for XXHighEnd which very sufficiently operates at 18 bits or even 16 when the output voltage of the DAC isn't too high and the gain of the amplifier isn't too much. This brought me to again searching the internet and it came to me that the PCM1704 should theoretically be able to do the job, but nobody just made something out of it. Or ?
For over a month I had bookmarked a company which expressed specs that looked promising but kind of vague in the mean time. I kept on coming back to this because there wasn't anything else and in the end I started to send an email to that company ...

Ok Peter, why don't you mention the name of that company ?

This is a strange story, and possibly I can't even justify it 100%, but I have the feeling that right now this company shouldn't receive dozens of emails from you guys asking away because they just can't bear it. Maybe later, but not right now. This is all my fault and prices would double instantly ... :blush1:
People who know me, will know that the occasions I asked someone for help in my professional life can be counted on one hand. Until one month ago that was. Right now, some email boxes have been overflown, just because I started some DIY project with modules provided as kits and a withgoing FAQ that says all is for skilled people only. Hmm ...
Add to that this isn't about a "DAC kit" as such, but a couple of modules which makes it a DAC. Add to *that* again that I asked for modifications which were politely met, but that these functional modifications ended up in technical stuff beyond my knowledge, and no existing manual could explain to me how to deal with things. Just being honest here ... :whistle:

So even if you know what company this is about, please don't go there right now, and if you indeed know, you also know this is related to respect. I'm sure you will understand in a later stage ...

First listening impressions (with 0.9w-3 and which is theoratically not the best (thing to do))

Since rather many combinations of settings exist, all with their own merits, right now it is rather hard to determine what is the best setting. But no matter what setting I tried ... it is one big step ahead in audio playback. To mention a few things I definitely recognize (just 16/44K1) :

• Enourmeously increased resolution.
  
  This is unrelated to the higher samplerate (because I'm just not using that right now) and also this is unrelated to the nos/filterless principle;
  It is related to the enormeous stability of the PSU, which, mind you, must deal with such small voltages (.0001 etc. mV). And, knowing that creating 1V of output here easily degrades the .0001mV output there, you will know what this is actually about, hence what I was after.
  Also, of course I know how accurately things were trimmed by me, but more importantly : how accurately they *can* be trimmed by design.
  
  Cymbals now receive an additional dimension, which I generally express as : now you can see the size of the cymbals much better. Also, compared to reality (live cymbals) it comes creapily close now.
  
  
• A bass response of which I didn't think it was possible.
  
  For the first time in my life I "heard" a powerful bass response as how we all perceive the large heavy amplifier being able to produce good bass response. We all know that the latter is non-sense, since the class D and other 100g chipamp stuff can do it just the same. What I did not know is how a very good PSU for a DAC incurs for the same but in exponential form. But it can well be that it is the other way around from what I sad right above : the small voltage spikes needed for the higher frequencies won't tear down the more current eating bass waves (never mind this all occurs at micro levels). That the I/V (current to voltage) conversion occurs in a high current domain is another thing that for sure contributes.
  
  As everyone I use a bunch of test tracks and throughout the subwoofers speak where they did not before. This is the most occurring, because - apart from XXHighEnd anomalies as some may recall - usually this comes together with a more profound "just low bass" region (say 40-200 Hz) while right now there is no spur of this. Oh, the bass is more profound allright, but it doesn't incur for any colouration or the tempt to change the cross over. It is just all good good good.
  
  
• Compared to hires it is now an apples with apples comparison.
  
  Since this DAC allows for NOS/Filterless at the higher resolutions, I now can compare 16/44K1 with 24/96 and 24/192 without having to hop over to a oversampling sigma delta DAC.
  I sure did not recognize any different nature at the higher resolutions (tried 192 as well), but I also did not perceive it as better. This latter is not meant to be related to the DAC but merely is a confirmation of what I already though before but couldn't prove (apples and oranges thing) : redbook (hence 16/44K1) seems to suffice and any higher samplerate plus bitdepth doesn't bring more to our brains. Something for another discussion !
  
  
• There's an unary "experience" listening to my speakers now, and I don't know how to put it into words yet.
  
  It is not "black" as we often hear or experience, but in that area. An "emptyness" maybe. A being swallowed by the music, you being in that black hole. It's not a negative, but a strange one. Again things are happening in mid air (I talk about this more often), but now they more happen in 3D while not being able to point at it.
  It sprung to my mind that bass is more directive as ever (I use subwoofers in stereo setup) and besides that the veils from drums suck on you. The latter is again not a negative, but at trying to express what I feel, this has to be in the equation somehow. Hmm ... it could well be that absolute phase is wrong (just change it in XXHighEnd of course), which btw *is* very well possible because of the various taps which can be used, and each "stage" inverts the absolute phase.
  Anyway, it looks that the "boost" of music is more profound now, where boost becomes inverted boost (suck) when indeed the absolute phase is wrong.
  
  
• Very good micro detail at the larger instruments.
  
  This may sound strange and also looks similar to the beforementioned higher resolution, but I am fairly sure this is about the inheritent speed of the DAC (about slew rates and such, possibly even feedback :yes:).
  Under "increased resolution" I talked about the other dimension in cymbals, and here I talk about yet another dimension : the attack of cymbals. Ok, maybe you must have the experience of holding a drum stick a couple of times, but when a drum stick hits a cymbal this doesn't need to be one surface touch. I mean : when a ride cymbal is "ridden" this *is* one touch only, but when a smash cymbal is hit, before the stick goes away the cymbal hits the stick again. It is this which just is audible now.
  
  Brushes, even more delicate and far worse to express properly as I found lately, can't avoid the "feedback" of a cymbal and snare (or tom). That too is audible now, though not exactly in the leage of the "larger instruments". I mean this opposite to the more commonly known detail like breath in a flute or sax. So it is not about this.
  Similar IMO is the wood of a bass becoming audible. I hear you say "man, I heard this long time". But no, I don't think you did. I know my abouts, and this is in another leage. This is about jitter (in good combination with the other virtues of this DAC);
  
  I don't know whether it is a world record, but the overall jitter of 1.2ps seems to incur for what I just described. Why do I think that ?
  Because this low jitter only can be achieved in oversampling mode of the DAC and it is there where this is happening. This is *not* because of the oversampling itself hence impeeding higher "resolution", because at least I claim to know that oversampling only degrades. And for sure it won't "create" the beauties I hear here. If that would be so, XXHighEnd could do it just the same, and it can't.
  
  All 'n all this subject is not really honest, because it uses oversampling mode which net I don't like. It colours the music, and all starts to sound similar because of it. Nothing different from what I was used to.
  But for the more or less solution, the direct Firewire connection comes into play, because that will incur for less jitter because of the I2S connection beind it. And to get the real merits of this : at non oversampling mode the DAC depends on the clock of the incoming data. At oversampling mode this is just unrelated ... :secret:
  
  
  

Might you want to know : none of the modules I used, emerged from more or less known DIY projects as known from e.g. DIYAudio. I can tell you that each single part was designed from highly qualified engineers and that the designs themselves testify high quality engineering. Not that "DIY engineering" wouldn't be any good or the engineers being out there are not trustworthy for their profession, but what I have here contains all the consistent knowledge of the most qualified people imagineable, for whom I make a deep bow. :thankyou:

This is not a matter of simple is less, and looking at the picture below I can tell you that the DAC (ok, SMD) comprises 1/25th of the total PCB surface. 24/25 comprises of PSU stuff and getting the output current from the DAC into the output terminals.
Oh, I didn't count all the switches needed for the various input and output settings.

I think in the commercial version, the cardboard should be black :)

Very interesting!

My previous DAC was a NOS filterless construction. (DDDAC 16/44 only). It had the most dynamic sound I have ever heard from CD. Very "immediate" sounding.

BUT: With no filter, there will be a lot of HF noise at the sampling frequency, and above. This noise can be quite stressful for wideband amplifiers. Also it will create IM distortion at audible frequencies. Do you have the possibilty to measure this noise?

You need to send this to me for, ah hmm, evaluation.
(ya, that's the ticket)

Yes, but it won't be different from what you are used to (at 44.1). But still it is so that with the filter it sounds less good/natural.
Just take a look at what manishandher produced by means of (RME's) Digicheck which is just about that. As commonly known, nos DACs should not be "measured".
Not an excuse of any kind, but just the net result of filtering which debets to the sound. As I say it : not filtering makes all "snappier" (or more dynamically if you want), but also more natural.

Btw, don't confuse the "HF noise" you mention with HF noise as the left over of downsampling from a higher sample rate. This sure is *not* anything you'd like at all, and this happened for XX at downsampling from 352800 to 176400 at first, which really needs a filter (the AA filter as is there since). This is important, because nothing in there (the DAC) is downsampling again. Not at oversampling (to 210 KHz) and not at nos.

Not necessary. You'll accept it blindly. :)
But honestly, yeah, I'd dare to invest the time of building, send it yo you, pay for the shipping charges, and be sure it won't come back.
A kind of Internet means btw, because how else to judge products from far away countries without a local distributor and dealers ?
What goes with it, would be a "maybe some other user tried this very product before you" and each scratch would allow for some discount.

You may know I am a provocer of "no such thing as a real bad recording exists until the real proof of it is there".
Well, one of those recordings prone to that is Musicology from Prince, expecially the 2nd track "Illusion, Coma, Pimp & Circumstance";
I recall being at someone else's place, pointing out the extraordinary excursion of the woofer being 2cm or so in one direction, which at normal circumstances should not be more than a few mm's only.

Today I thought about "control" of the DAC in that area, and if one track would prove that, it would be this one.

As you already guessed, it worked out beautifully (to the sense of how "beautiful" can workout for Prince :)).

This is a track with rather low synth fundamentals (a bit Madonna like) which in this case before worked out the most rough as possible. It just made vomating (a bit of a dutch expression) the woofer before. Just completely out of control as how it came to me, although without really knowing, and virtually blaming it on the "bad recording" (like completely overstreered). But today ?
Ha ! not so today. Today it appeared to be an indeed low synth bass line, but with a bunch of higher frequency harmonics, just "good" disco like.

Might you have the album, try it. But uhhm, output should be at an average of 90dB @ 5m distance. Run this at a lower output, and it's just some fumbling around.

Good control of a DAC ... who would have guessed that ...

Peter,

This NOSDAC project is interesting - but only if we could eventually get the opportunity to buy one or construct one ourselves. What is your thinking about how that might happen?

Cheers
Frank

Since you haven't built the FW input yet, you must be still using the FireFace?  so to listen to hirez you still had to increase the the sample rate right?  Is that something you'll have to do always? or when you feed it directly with Firewire... how low can you go? and will it have to increase like with the FireFace?

I was off 4 days and only got to listen once this whole time :/  holiday stuff and building a computer for my daughter at her mom's place... and, out of all the computers I've put together, this is the first time something wacky happened and it didn't fire up right away... had to take it apart piece by piece to see what was up ... 2 days wasted to find out I think I either didn't seat the cpu correctly or some "arctic silver" paste spilled over when trying to place the heatsink fan ... :/ bleh  anyway, just got it going tonight yea, but wasted time... Casual listening I've been lazy and haven't gone back to W3, but you're still using it over V7... I must bring myself to give it another try with different settings.

You are completely right of course, and I didn't mention that on kind of purpose;
Firstly, it is relatively less important than having to use a complete other DAC as how it was before (with me).
Secondly, I didn't want to confuse things unnecessary, because I first have to see whether the larger buffer still makes a difference. And in other words too : whether XX can still make a difference by means of the various settings.

The latter is very complicated because not is all known (not by me anyway), although this project / DAC might be just the test pilot to find out. I mean, whether all is PSU impeeded (which whould be unable to influence in this case) or the jitter (which is completely detached from the incoming stream in this case *if* oversampling mode is used (which I won't hehe)) ... it shouldn't make a difference anymore. Shouldn't, but it will take some ages to test everything, while I'm not even through the base of it all.

I didn't use 0.9w-3 because I think it is better or whatever; instead it was just lazyness with the combination of nothing being shielded yet, using unshielded cables all over ... well ... look at the picture. It is just impossible to be the best right now, hence it is useless to try to squeeze out the best of it at this moment. Nevertheless it is the best I ever heard.
Also, it can hardly have been broken in right now.

A complete other matter is that indeed I'm trying to get to the buffer of this firewire connection, and at this moment I can't judge yet what can be done because I don't have the drivers running yet (the hardware should all be set already). If I want I can program the complete firmware and driver (I mean, right from the beginning) so I could make it the best for "our" purpose. But it is not to underestimate, because that would need just another development board (besides the one I have now), and the additional $1500 it takes is not funny, not even knowing what can be achieved with it.
I have to keep up that I am going to produce XXX DACs already :secret:, otherwise you don't get access to all the stuff anyway.

:clapping: Wow! Are you expanding your enterprise into hardware products? That's cool.

When it comes to designing and building a DAC there are several critical points. PSU is of course one of them. Personally I would like my future "ultimate" DAC to feature a BNC clock input. I belive an external clock with huge PSU can be good. I am not an engineer, so I wonder if this option is easy to implement in your new DAC prototype?

Another thing is that most DAC's are working from x48kHz samplingfrequency. (48, 96, 192, etc). But the CD is 44,1kHz. So inside the DAC, there has to be a samplerate convertion (SRC) chip in front of or integrated inside the DAC chip. I believe this SRC converters are "not good". My ideal DAC, then shoud be runninig on 88,2 then. Eventually the SRC process could be done in a "better" way. Maybe the SRC can be done in a software, implemented as a plug-in in the digital loop available in your FF800 software? (I know there are software solutions in the marked).

What do you think?

Hahaha, wait a minute pedal ...

What I expressed for features and all were not much commercial expressions, but merely what I had in mind for myself. I little show-off if you like ...


As the original features show, a terminal is there for an external clock. What they don't show, is that a BNC input for a word clock is there too. However, for me, myself and I it would not be connected. But it can be ...
The configuration as I have it in mind  (and which is just there, although not everything is connected) is as follows (besides the beforementioned in the first post) :

- As said, BNC Word clock input;
- Direct TOSLink S/PDIF input;
- Firewire input 2x (useless by itself, by see below);
- AES input (over Firewire);
- S/PDIF input (over Firewire);
- ADAT input (over Firewire);
- TDIF input (over Firewire);
- Direct I2S receiver;
- I2S input 3x (over Firewire);
- Direct S/PDIF receiver 2x;
- I2C interface;
- Each of the above also available as output (not Firewire);
- Some more Pro stuff;
- Last but not least : DSD input.

Huh ? :yes:

The last mentioned "feauture" needs some additional explanation :

During the process the ESS Sabre DAC came available (I think March this year); Besides it has the best specs ever, its internal working is very much similar to my own design about the 32/384 DAC for jitter specs and more. My "DAC" will contain the ESS Sabre as well, and although it is a (heavy) oversampling DAC, as said, the specs are special. Otoh, the jitter specs are not better than the DAC I have running right now ... in oversampling mode.
:scratching:
Here all are apples and oranges again, currently knowing that the oversampling mode of my "NOS1" just doesn't touch it. So far I tried each night, and it really doesn't last for one track ...

Besides that, when I really implement the ESS Sabre (due here for a couple of weeks) it won't fit in the same cabinet. But then I anticipated on that with having two cabinets with sharing connections where needed (like the inputs, outputs, some PSU parts and routing switches).

So do I overdo it ? probably yes, and out of all available options and combinations only one will be used. However, there is a significant difference with how I had it before :
Before, in my case, always the Fireface was the intermediate. It was the "routing" device, but it always routed over S/PDIF. That now can be avoided, because the DAC can be its own router. For example : any CDPlayer, DVD(A)player, DATDevice, SACDPlayer, SATReceiver and the like (??) can use the DAC as a DAC, and it is not PC-dedicated. In the mean time, there's nothing that makes it dependent on the PC (the direct Firewire connection causing that, and - upside down - a soundcard is not involved hence does not disturb).

Did I mention USB input ? No. But optionally it is there too, but now completely without reason *and* currently it doesn't support 24/192 (and not DSD of course ... afaik).

What can I say ? not any element was designed or created by me. But with some soldering and driver programming it just allows for all.
This "all" is just about backups. Right now I again enjoyed for several hours the SUPERB bass coming from this single - far from optimal - connection. All still unshielded as the picture in the first post shows.

Peter

I agree with you, that the new Sabre DAC seems to be the new champ. Today I did some research and learned that actually I have friend waiting for delivery of same (or similar) DAC-kit as you. So, perhaps I can lay my ears to it within the end of the year. Can't wait!


Q: Regarding the I/V converter following the DAC-chip: Will you use a SS (opamp), transformer or tube circuit?

Pedal: LMC ?

Neither. But it is passive anyway. :grazy:

If it interests you : any transformer just kills transients, "sharpness" and sprankling following that, never mind it's passive.
OpAmps ... well ... YMMV but if they're not noisy they colour the sound (ah, I am very much generalizing here).
Tubes ? hmm ...

Tubes can colour, but IMO this is unwanted. I think they can just as well be neutral if picked properly for the job.
People say tubes are slow, which I personally don't believe. What I do believe is that tubes don't last forever and I can't stand that I'd never know when they are worn out. But that's personal.

All'n all passive is the more sure way to go for. If it can be achieved of course. Anyway it was one of my sure objectives and requirements, friendly met after politely asking.
But who knows ... when the I/V was setup in an active design (as it originally was so), what would have brought *that* ?

If I may say so ... my TwinDAC+ also takes this explicitly into account ... but it just hasn't got the drive for longer interlinks. But hey, weren't we suppose to use pre-amps ?

:veryhappy:

God that sounds like a dream to me. :drool:
I have to build this one someday!

Hi Peter,

How do you think to connect a SACD transport to the NOS1? maybe via HDMI?. I have 3 SACD players, OPPO DVD player, PS3 and Cary SACD 306 Professional, I think that only the Oppo and PS3 can output the SACD digital signal, and only via HDMI, and I don't know SACD drives for computer player for use firewire connection. 

Good question and an honest answer : currently I don't know yet.
I think anyway the Oppo is known to be able to capture the DSD output and the ESS Sabre for sure is able to digest DSD.
But as with all features from my second list above, they have to be created explicitly. So, that is a theoretical list and I can tell you ... to implement them all may take many weeks. Besides that, some connections just go over Firewire, and others require dedicated inputs + input switches. So please remember : what I listed is indeed theory only and allowing for it all at the same time would require a "switchboard" which may look virtually imposant, but which may be undoable at the same time. This is exactly why this "routing stuff" goes by software these days, which ... requires the programming of that. It may look all nice for you, but is a huge task for me at the same time. I think ...

Peter

Does even the PS3 output sacd through even hdmi?   I thought there was no way to get that SACD digital signal... If I play my SACDs through my ps3 they get downsampled through the digital optical output, I know that ... in any case my SCD-1 and SACDs lie dormant as I've played with computer audio for low these many years.... wait, guess it hasn't been that long.  Man, so much has changed though, been very exciting this last year.

Officially (at least that's what I know of it) the raw DSD stream can't be output from an SACD player (it's not allowed, whatever). But, from some players it can be picked up, and the Oppo (don't know the type) is known for it. It you Google around a bit, you will find it.
You'd have to take up the solder iron though ... :yes:

Btw, I don't own an SACD player, and there was nothing in my mind that planned to use the DSD input. But about theoretical options ...
Peter

Hi, only the two first generations of PS3 can play SACD, and unfortunately the PS3 can't output a pure DSD signal through HDMI, it convert DSD to PCM 24bit/176khz.

HDMI from versions 1.2 and up can transmit DSD securely with HDCP (High-bandwidth Digital Content Protection).

I just asked to Oppo if its players can output via HDMI pure dsd signal, I am waiting its response. I have read that the DV 980HD can do it, but the DV 983HD converts DSD to PCM, I don't know about the DV 981HD model that I own.

Also there is some Pioneer, Sony and Denon player that has a SACD digital interpace call iLink, that it is firewire, but I have read that Pioneer convert DSD to PCM.

:scratching: ... Ok, apparantly I must dive into this. My, say, personal problem is that I derive much from the movie world, where getting audio over HDMI seems to be a tough job, and no one rule seems to exist (mind you, this is related to multi channel vs. 2 channel (where things *do* work) but in relation to AC3 and DTS *and* that 2 channel is not an option there obviously.

I could also say : do you know (definetely !) about a soundcard with HDMI input, which ... well, just works. From what output ? ahh, I don't know, because where is the CDPlayer with HDMI output ? and would there even be a reason for it ?

Right now two external DACs spring to my mind which officially support DSD meant to be read from files (computers), and both live in the recording world only (no, I didn't say Pro world !). So, an example is DXD (which is 352800) which nowadays often is used for mastering DSD. Also it goes the other way around : DXD can be used to edit DSD (DSD can't be edited, or it is too cumbersome to do it).
From this, I am fairly sure (but not 100%) that where DSD is converted to PCM without any losses, it would be 352800 PCM. Now, you might have a DAC that can do it (but you won't), but now where is the soundcard to pass it through ?

If I keep on typing, I just as well might find the solution automatically, but I only wanted to indicate this is all not so easy (at all).

Note that many DAC chips (!) are able to receive DSD, just because they're delta sigma, and because of the high oversampling rates needed for that principle, it's relatively easy to support DSD as well. This does not mean that "we" can use such chips for SACD material, which for 100% sure won't be playable from within a PC anyway (although mr. Putzeys created one in private). Thus, those chips are meant for being in SACD players ...

If you are looking for an external DAC being able to play the contents of e.g. your Oppo, you must catch the DSD stream (I think several external DACs exist with DSD input, just like I could provide it with the ESS Sabre).
If you can only catch the PCM stream it should not be downsampled, and if I'm well informed (see above) it's a dead end, unless you have a 384000 (352800) DAC.

All 'n all, right now, I don't see where HDMI comes into play. But hey, I sure can't know everything, so if anyone knows more, please tell it and consider this as (well meant) BS. :)

Peter

Reply from Oppo:

The only player which we have which supports native DSD transportation is the DV-980H. All other products convert DSD to PCM.

An external clock shouldnt be needed. If done well (really well) the DAC has to be the master clock.


A NOS design like this (Peter correct me if I'm wrong) does not change the sampling rate, whichever you put in, that will be out. It is filterless. The beauty of peter creation is that it can accept 48, 88.2, 96,176,4 and up to 192k. Just to cite all available sampling rates :)


No, they are cr*p, they ruin the signal and add jitter, to avoid like all digital filters.

Just some background on "non oversampling" (NOS) for those interested :

For me, unlike the first explicit provocer of it (Peter Qvortrup working for Audio Note) it is all about the squariness of waves. In music "square waves" exist all over, and if it isn't a synthesizer which can exhibit them as exact as can be (when no analogue devices where in the mix at recording), it is the coincidence of matters. An example of the latter in nature is thunder, which can be visualised by looking at the front of a surf water wave. And, while thunder is an example of the large level (think bass like) of squarish waves, instruments like a trumpet exhibit them at a more detailed level (higher frequencies). A tick on something like the edge of a snare or tom drum would be another (kind of) example, where the "square" exposes as a very fast (read : steep) transient.
At least in 44K1 sampled WAV data, transients exist ranging easily over 2/3 of the total voltage range, meaning that the voltage (for a 2V RMS output DAC) will change from e.g. -1.3V to +1.3V in one go ! So, in analogue form this cannot exist, because it would imply an inifitly fast rise time, but with 44K100 samples per second there's just no more "resolution" to catch the steps which may be there in (analogue) reality.
Note that with e.g. 176K400 samples per second there's 4 times more room for in between voltage steps, and a transient which is captured at 44K100 exposing one go, could show 4 steps at 176K400. Or still one, because the sampling rate is way way too low to mimic analogue.

Above I mixed two principles :
1. When a steep transient (or square sound) is really there, it should be expressed like that (first part of the above);
2. When a not so steep transient (or less square sound) is there in reality, it should not be expressed more square than reality was (second part of the above).

The second part is theoretically solved by a higher sample rate;
The first part is solved by just not rounding the squares.

The story becomes confusing, knowing that the first part will be emulated (!!) by oversampling. Note that oversampling as such, is nothing else than adding interpolated samples which are not real, but do round squares which were not perfectly square in the first place. In order to understand this principe, do this :

Draw a square 2d wave with the corners not being 90 degrees exactly. This mimics what happens in sound. Those corners must be drawn in digital steps and not in an analogue round fashion. The steps you draw are the samples take.
Now adjust thee drawing, by interpolating the steps with a factor of 2. Thus, each step becomes 2 steps. This must be done at the outside bondaries of where the rounding of the corners started. Thus, looking upwards, the rounding goes downwards. And, looking sideways (the top of the wave) the rounding goes sideways.
After this one step of interpolation, you can already see that at doing it again, the square wave becomes more and more rounded. Do this an infinit number of times, and a pure sine will be left !

The latter is what this is all about : oversampling DACs make sines out of original squares, and the sound is destroyed. Oh, it will be less harsh because of less squares, but "harsh" is relative, and when a trumpet shows the "harshness" of just that instrument, it should stay like that.

It is obvious that this all can be related to transients, and that steep transients will become less steep, and dynamics will be destroyed.
Whether we talk about squares as such or about dynamics, both are the exact same subject.

As a side note, keep in mind that when an exact square is fed to the DAC, oversampling doesn't matter a thing, because there will be no steps to interpolate (draw an exact square wave now, and try !).

When you understand the above, you will see why some people like oversampling DACs and others like non oversampling DACs;
Both contribute to something which could be worked out for the better, but when the oversamplig DAC works out for the better, this can only be because of too few samples in the data. With this I refer to the exmple of a transient being recorded too steeply, because there was no room to have more steps, those steps always being there in (analogue) reality. Thus, the higher sample rate allows for those steps which are better for theory on one side, but since the steps are no reality at all (the real steps were different), it works out worse at the same time.

Now here is the important part :
An oversampling DAC creates less reality because a too steeply captured transient will be flattened in an unrealistic way;
An oversampling DAC creates better reality because a too steeply captured transient, which is unrealistic by itself, will be flattened.

Got this ?
There will be no scientific answer to which of each "features" works out for the better, and both have an unreal result.
BUT :
There's also the matter of the real transients and squares which are reality from the start, and the oversampling DAC will get those out of the way just the same ! And thus :
The "feature" of the non oversampling DAC to at least pertain those transients and squares comes out of the equation as a positive.
And thus all 'n all the non oversampling DAC wins with 2-1.

There is quite some more to it, like oversampling shifting the nyquist frequency and therefore requiering less filtering, and for example of 352K800 it is said that just no filtering is needed at all. 44K100 just officially needs the filtering, which will operate in the audible domain.
The NOS filterless DAC just doesn't do that, and where filtering by itself will again destroy sound, the artifacts (if audible at all) from "aliasing" are taken for granted.
Again, "if audible at all", while at the same time synthesizer music is completely destroyed by oversampling. And this is audible for everyone.


Having said this all, the phenomenon "oversampling" needs some additional clarification;

An oversampling DAC, in 95% of cases is so, because it can't operate without it. The sigma-delta DACs are the example, with DSD (= SACD) in thee same line of working.
These 1 bit principles can operate only by heavy oversampling, meaning 256 times or more, up to MHz's.

The other 5% of cases, are about multi bit DACs which just do not need the oversampling in order to operate, but, it is done anyway for the reasons of shifting the nyquist frequency as briefly referenced to above, and besides that will need filtering which is always needed just *because* of the oversampling. In this case we talk about oversampling to e.g. 176K400 which is 4 times only. Note though that 176K400 actually is a strange number, and 192K is more normal, but since 192K can't be divided by 44K100, first a common denominator has to be found, and from there on heavy oversampling (underway) emerges again.

Although the definitions do not exist explicitly, one could say the "oversampling" is the heavy kind, here posed as a pure negative, while "upsampling" is going from e.g. 44K100 to 176K400 directly.


Lastly, and to get a bit from the abouts, at 44K100 sampling rate, a 22050Hz pure sine is represented as a pure square. There's just no more (sample) room to make it less than pure square. Similarly, a 11025Hz pure sine is represented as a square built from 2 steps. This is just audible, knowing that squares produce a ton of harmonics.
Now, upsample this 1 time, and the anomaly (for what it's worth) at the very audible 11025 is shifted to 22050 which is not audible (for most).
This is why heavy "oversampling" is beneficial by itself, because it shifts the anomaly into the inaudible area. But keep in mind : in a fake fashion.

All together you may get the grasp of the importance of a 192K non oversapling DAC, meaning :
When 192K native material is played, the anomalies have been shifted "over 4 times better" into the inaudible area anyway (but now with real samples !), which makes non oversampling "over 4 times more legit".

Peter

Ok, after a week of playing with the DAC (I still don't think is is burned in sufficiently), I want to express some feelings about it :

I think I said it before : For a longer time I regard my own system as a whole to be the best I ever heard. Of course this doesn't say much, and a chance exists that many of you just have it better. Only statistics of me being around at other places tell that the chance is little.
Small conclusion : whether or not I have this "best system" or not, it should be better than average, and this is my reference.

The above was with my old 18/96 nos DAC.

Now, with the 24/192 in NOS non filterless mode, and if I had to apply a mathematical figure to it, I'd say it is 10 times better. Whoops, that's a lot.

I don't think that I, right now, can find commonly known wordings for it. I mean, there's nothing like spatiousness, wider stage, instrument positioning. That is, I don't recognize this or can't as off yet. You could also say : this was good already before, and improvements must be in other areas (at talking about "10 times better" :)). But I'll try a few expressions as far as I can recognize them;

First of all, I think I can now at last hear the benefits of higher sample rate recordings. Remember, I could not do this before because of lacking bits 18 only) at 96Khz, and not being able to play 192Khz at all. Of course I could use my oversampling sigma delta (Fireface), but this is apples and oranges to begin with, and btw nothing good at all. So :
There appears to be a phenomenon to my judgement, which is related to being able to differentiate in voices singing in parallel. Remember, I'm talking about the higher samplerates, and how to hear they are better.
It doesn't matter where I'm in the room, as soon as background voices (male or female) start to sing, you can kind of count with how many they are. I don't talk about where they are, but with how many. So, a matter of being able to hear person A, B and C separately.
Maybe this is nothing new (for you), but so far I couldn't manage to find the absolute difference because of the apples and oranges thing.
In other areas I don't perceive a difference. Or not yet.

Back to normal 44100 playback;

The longer I play with the DAC, and the better album examples I can find for it, the more amazing the bass becomes. I have no words for it, but the sheer difference of hearing beautiful bass before (which already was a tough job for me, and is much related to XXHighEnd versions), with complete life-like basses from now ... I can't express it differently. I think earlier I already talked about the ability to hear the wood of a bass, but right now I'd say that someone with knowledge should easily be able to tell the "manufacturer" of the bass instrument. I could also say : before I was very happy when I could hear the strings of low bass vibrate (good resolution in that area), but now each (double) bass and cello just sound different. Give me a few months and I will be able to tell the artist by listening to the bass only ! (and I don't mean the means of how it's played, just the sound of it).

There are also strange things going on;
My last tweak being the SSD with the OS on it, already brought the clear vibration of strings being pulled too loudly. So, the metal itself makes the sound in this case, and while this is a very profound sound (dzziinnggg) I never heard it before from a speaker. I checked it with others to be sure, but it really is so. Why ? I'm puzzled, just because the sound is so profound, and by the way so very naturally. But, I never missed it before, so how the hell could I tell those strings were pulled so hard on the specific recordings ?
Now, with this DAC, it appears that it will be very hard for those musicians *not* to pull those strings too hard. They just do, and they do it always. Man man man, I get tears in my eyes at writing this, because it is really a complete new dimension in audio playback.
I don't care anymore whether there's a stupid clarinet showing off as often with jazz albums, it's the bass man (and often woman) making the sound !

A few days ago I bumped into James Cotton and his album "Deep in The Blues", and besides I seem to be able to clearly hear he is playing an acoustic bass guitar throughout the album, he is playing that guitar "toggling" (5 individual fingers) like a spanish guitar would be played without chords. But now imagine the deep sound of a bass guitar, and that those fast individual bass notes just work out as intended (instead of a smeared bassy sound). Man man man.

I think I said earlier that everything seems to be supported by bass. This is literal;
This works so much throughout, that it merely looks like an anomaly. I kept on paying attention to it though, and the only conclusion I can draw is that - like the wood from the bass - it is the recording room/space I am hearing. Think of a room and bass measurement, and the knowlegde that the room adds bass to your sub woofer, just like the cabinet the bass driver is in does. Now I hear the same throughout, and this wasn't there before *at all*. It is strange though, because you are not used to it. Even voices can express some sub low which sure can't come from the voice itself, but with some hall and reverberation it just sounds natural.
Besides this general phenomenon, I experienced quite some times the whole house started rambling because of sub low output, and this is on recordings I never heard it before.
Added to this that I have a couple of tracks to trim the bass vs. sub woofer output where the subwoofer output should stay normal in all cases, these recordings do not exhibit more sub woofer output. To me this is the most strange, but proves all is still right, and it is not just "more bass output". It merely looks like slow waves with not too much amplitude just being able to express now, where they were killed before.

Where I was keen before on having just directional bass output - knowing that much of that is caused by higher frequencies around that bass fooling you - this is now just "completely directional". And this is the most interesting, because if you now can hear where the man playing the bass really is, or what about two of them, this is just again another dimension. But keep in mind what I said above : when the metal of the strings becomes audible, this is the base of the "guideance" of the directional bass !

About my stories about standing waves disappearing when things are all right : this again vastly improved. But, know in a very understandable way;
As a strange example I want to mention Madonna with the ever accompanying synth bass. I used to know this as "bass" of which was audible it is a synth. Strangely enough now this isn't so much of "bass" as such anymore. The synth now expresses it's short pulsed output, and where things got smeared before, it's now just the pulses you are hearing, and the deep sunding coming from it disappears.

Read the latter again; it is contradictionary to all the other phenomena around the bass. Everywhere we have more bass, and here we have less ?
Well, in fact I just explained it. The waves are less smeared, and the individual vibes are expressed all over, and now the "individual vibes" and the standing waves come into play : the more the vibes can be expressed individually, the less they interact in space, the less anomalies come from that.
That in the mean time bass output itself is "higher" (read : better) is just another phenomenon I think, caused by the dedicated PSU. So, the better bass output seems PSU related, and the better accuracy is DAC related (though the PSU will enable to follow the accuracy).

Talking about accuracy, well, this expresses all over. So, now we're up to the higher frequency regions, and may it be "speed" (slew rate of the DAC) or the better translation of it because of the PSU, there is detail which ... well ... can't be expressed properly either.
This too is strange, because normally I would be able to give examples, talk about hi-hats and cymbals etc., but somehow here too other definitions are needed. I don't think hi-hats and cymbals etc. are better. I do think brushes are far better and it might be the only thing I could reason out why. The other things express in things which can't be told. Not by me, not yet;
Yello is always my example whether the NOS principle acts as intended (because synthesizers in an interesting fashion only). One can't talk about synthesizers and how they should behave and all. One *can* say though, that many more interesting sounds were heard, and the better music reproduction becomes, the more value those two guys put into their music. Might it be some sub low intended (!) reverberation, or a dazzling sound sweeping from left to right, neither is audible when the playback system isn't up to it.
My point is merely that these gagdets (when audible to start with) have less or more fragility. With synthesizers I think you can say that - besides things being audible or not - it is the fragility with which the sounds are expressed, that matter. Just think about some triangle and saw tooth waves interacting with eachother, and what might come of it. You can bet though that the Mini Moog guys from the early days, as well as the modern synth people from today, try to create interesting sounds, as long as they don't try to emulate violins. If you are not into synthesizers I am sure that this is because you never obtained the so much interesting intended sounds coming from them. Try Momita (uhhm, emulating volins) and combine classical with synthesizers, but in the mean time try to imagine how in the world the guy was able to create all those orchestra sounds from his synthesizers, all to be programmed right from the base.
But don't try your PC speakers ...

Lastly for now something commonly known, but happening here just the same :
Look at the picture below. For me this doesn't invite much to sit down and listen to it. Of course I tried, but she sings exactly as she looks like there. Yesterday - at testing various hires albums - I had to try her too (this is 24/88K2). Well ... since I am a man and thus can't cry I, didn't. But it was very hard to keep the tears inside. Whether it is the superbly played bass or just she herself, I imagined a woman with a history all laid in her songs on this album. Before it came to me as some shattering around, but now it just worked (as intended I suppose).
As said, a commonly known phenomenon, but for me in this case a radical change.
Of course, when the bass is not there (as it wasn't before) the "technical" fun has gone already. So, wanting to listen to the next track for the bass already, makes you dive into this album more than before. But still ...


Enough said for now. As you can see I don't have much to talk about, except for that bass. I guess it is too overwhelming to be ready for other merits at this moment. Later ...

Peter

hello P
talking about system:
what amps and speakers do you use?
would be useful for my own reference to evaluate the "value" of your opinions
not that I don´t trust them,but useful to know in what context U talk about  "best bass" etc
if you e.g. had minimonitors :),that would be a useless statement to me
best
Leif "curious" Christensen(LMC)
Norway

and Yes Pedal´s friend waiting for Buffalo Dac (ESS SABRE) is probably me.
it left US y-day and me going to Florida the 19th , I won´t build it until early jan.

Of course not, that would be extraordinaire ! :whistle:

Let me first try to explain that good deep bass hardly is related to the surface of the bass drivers' diaphragm. Oh, it might, but it can hardly make a difference in the living room. The only thing what matters IMO is less distortion with more drivers, just because their excursion doesn't need to be so high per driver.
Btw I tell this, because I have in mind that you have those double BD15 red bass cabinets ... but maybe I'm wrong. And I only want to express : I just cannot for te life of me imagine ANY "bass setup" be better than mine ... with which I only want to say : a tad smaller (than a double BD 15) is way enough ...
But theoretically, of course, more will be better, but IMHO merely in the area of less distortion at high output.
My room is 290m3 and my listening level is always around 100dB @ 4 meters. Careful here, because I use the BD-15 Ultra with Orphean, hence rather directional, and @1m this is only 110 or so.
The port of the BD-15 Ultra is closed.

Btw, maybe hard to imagine, but with the better control of the bass the excursion gets less and the output more. Think of a straight 30Hz tone, making expand and distract the diaphragm 30 times per second, displacing the amount of air the surface of the diaphragm incurs for. Now make this 10 Hz because of improper control, but keep in mind it should be 30. Theoretically the excursion must be 3 times more now, and of course it will be fumbling bass.

Ok, this lot is driven by 4 33W GainClones, and mentioned 100dB SPL is at -30dB and 1.5V DAC output (no preamp).

Each channel bears a SVS PB12+ 550W active subwoofer capable of 12Hz driven by the LS output of the bass amp, and customly crossed at 40Hz (the BD15 goes straight to 27Hz).
Since I closed all the ports, output will drop off under approx. 16Hz (16 still works perfectly, which I can tell because it is the resonance frequency of the doors surrounding the living room -> they ramble).

I can tell you, the low B of a 5 string electric bass, when played somewhat more profoundly, keeps on vibrating on your stomache, giving the unary feeling I talked about earlier, just because that 60Hz (or whatever it is) is so powerful (at each vibration !).


I'm afraid I will be earlier then (expect it any day now). Of course I will again express my honest judgement about it.

Peter

You only miss analogue then! :)
interesting; I seriously considered changing my Trio´s for a similar setup with custom hornloaded double ultra with Orpheans in D´Appolito
less floorspace!
my listening area is of similar volume and I play at about same levels at same distance
I backed out because of little secondhand market for trios and some serious considerations about the modified BMS driver.
I ordered the Buffalo with the I/V stage incl.x-formers and psu s but obviously one could use Borbely´s discreet buffer
however this way I get up testing quicker and can alway develop it further.
by the way, it´s only the usb drivers that prevent the usb interphase from sending hi-rez.couldn´t someone rewrite one for audio use.or else we´ll have to wait for the Buffalo firewire I2S bus or USB 3.0
best
Leif

I don't know where you got that from (hence you imply that this is coming, and I never read about it :)) but ...

I was working on that ?!  E.g. :


Ah, but maybe you just were talking about "my" solution ? (from over at Twisted Pear)
Well, *that* specific solution won't get to work. But there will be another one ...


Hmm ... we might have more in common than you can see now ...
:) :)

I did not think of your "solution"
only picked up some noise through "the grapewine" about the possibility of someone launching a firewire to I2S module :)
would suit me fine
in the meantime usb 3.0 is on it´s way
I´ll se when I get back from holiday in jan.
will be a busy winther before sailing season kicks off again!:
I´ll change my power amp driver tubes to we437a as well  :clapping:
so long buddy
best
Leif "forward" Christensen
Norway

It's always good to have a plan B :)))

ps: signature updated

hello P
got the Buffalo today and have built the first psu
looks really good
hope to do more tom and sunday
L8R
have a nice evening
best
Leif

Hehe Leif, looked at the USPS tracks this morning, but mine seems not even has sent off yet !
So keep going, you have 6 days left at least !

soon finished! :)
just need to read the IVY manual and hook up x-formers etc
got a fiberboard to screw it to for the time being :veryhappy:
maybe sound tomorrow? tonight is dinner with guest and redwine :whistle:
L8R
best
Leif
Norway

Mine arrived today afterall ! But I'll be working on XXHighEnd first. :heat:
And next I'll try "some" Firewire connection hehe.

Careful with the wine now !

good 4 U!
I am sitting with a Carlsberg and listen to 9w-1 with q-4 unattended and it sounds good 2 me
have a nice evening
best
Leif

Any update on the Buffalo?

Hello
I´m struggling with time since I´m leaving 4 x-mas holiday on friday morning.
I´m finished soldering the Buffalo + IVY + the 2 psu´s.What´s missing is making the regulator for the DD USB pcb and hooking up the wires between pcb´s and the output rca connectors.All is put in a previously used Hammond chassis just 4 testing purposes.
Maybe I´ll test thursday evening.Tomorrow I´m attending a heart starter medical training course since I´m a volunteer in a Norwegian naval rescue-organisation.They´re pushing ahead to make me head of one of the watch-crews.
Time is indeed a scarce commodity!
L8R
best
Leif

Hi Leif,

I'm not sure this is what you are looking for, but just in case ...

(mine is still in its packing, but careful, because I only need to solder the dac board really, and then I can hook it up ... but I'll wait for you first :) ... and of course I have NO TIME ... :wacko:).

Although the NOS1 DAC is still not finished, by now I had a fair amount of listening hours, and I think it has sufficiently broken in.

First funny thing is that I still can't find the proper wording to describe how it sounds. I think this is because "we" can't describe in cold stupid phenomena anymore because they are from an other leage and behind me for a longer time. So the only *real* thing I am definite about is : it is better better better.
Yeah, "better".

Again, I think this is about the earlier idea to have achieved reality already, allthough you can always find some things which can improve. A kind of main point here is that earlier I listened mainly in the cymbals area for further improvement, while right now I am totally overwhelmed by the improvement in complete other areas, where I wasn't looking for improvement. So, this kind of put my both legs to the ground again, and I have to re-learn listening.

This isn't anymore about "staging" (although if anything I'd say this is more laid back than I'm used to), and this isn't anymore about placement of instruments. Oh, it might be, but somehow you don't come to that anymore. It is now about the instruments themselves, how many of them (same type) are there, and listening to your own room has shifted to listening to the recording room/hall etc.
Btw, I might repeat myself, but these are the things which spring to my mind at wanting to describe what happened.

Am I estatic ? strangely enough not even that. Being estatic from overwhelming goodness suddenly has changed into being very critical only. And the ever again wondering how things happened. Like the so profound metal sound from bass strings of which I can't imagine where it's coming from (better the other way around : how is it possible that this wasn't there before while it is so profound).
Or other kind of "technical" angles which jump into my mind : the better the reproduction is, the better oscillating amplifiers can be noticed. I already heard that coming up better and better (when things got better), but now this is near to overvoice the music itself sometimes. Not only with guys like Julian Sas, but also with respected jazz players like our Toots Thielemans. Btw this happens when an amplified instrument is played way softly and the artist wants to let that hear loud, so the amp is cranked up.

I hear the double/triple strings from pianos/wing now. They interact with eachother, and buzz into eachother. It is that which forms the character of that one piano/wing and which differentiates the one (manufacturer) from the other. But what must I say ? I hear more detail ? nope, because detail as such is a phenomenon which can exist without buzzing strings.

Higher frequency squarish synth sounds seem to fly through the air now. I think I said it before : in a much more fragile way than before, *if* audible at all before.

I have listened to several complete jazz albums, where I couldn't find *any* anomaly in the cymbals and hi-hats used, or IOW just for real, and can't be improved anymore. Ok, that is what I think today.
Before I would have to save the rare track with the same judgement in a "nice stuff" etc. Gallery because it sounded so cool. Today ... today I'd need to save complete albums where the cymbals do not sound right, in order not to run that album anymore ...

And there you have it. Right now there seems to be a huge difference between recordings. O yes, there always was, but I always managed to let poor recordings sound rather right just the same. It was my hobby.
It still is, but I wonder how to close the gap between those poor(er) recordings, and those who just sound 100% right. And now I have another problem :

Once you can obtain the "100% right" (to your idea of it anyway), there is no going back to less. For example, now I know that all basses can express their individual vibes, playing an album which does not do that looses my interest immediately. And they exist. Most probably because the room where the artist was had bad cancellation or the mike was too far away.
For me things have become a huge challenge, and right now I am kind of obsessed to queeze out more again. The Firewire->I2S interface should be one of them, and a special setup of balanced interlinks to the main amps another (I hope both to apply this week). Then there is XXHighEnd which more than ever has my attention, and of which I already have a version with improved SQ (well, that's what I think).

Here's an example of a stupid challenge :

It was not long ago (what about 20 days or so :)) that I was kind of hunting for the lowest basses to jump on you with the real power they have. Well, I have that now. Now start laughing for the challenge ... :dancing: ...

These low basses will be something like 60Hz and they work (with the power needed that is). But from there on, I realized that there are "boundaries" upwards. Similar (I think !) to hearing the wood of the bass, there are these individual vibes UPWARDS. Just think of it : an e.g. 40 Hz tone is easily audible for its individual vibes (you must have the experience of it of course). However, before, with not enough drive, those vibes would smear and no good vibes would be there. Now this is arranged for, and IMO all the way down to e.g. 16Hz, it is MORE DIFFICULT to have the same effect more upwards, in the, say, 120Hz area. Why ? well, because it may need almost as much power NET because of the doubled energy put in. Of course, the higher frequency by itself requires less energy, but since the amplitude might be nearly the same, the total energy required is higher.

What the h*ll am I blabbering about ?
Well, in fact something I hear coming (read : being improved next week etc.). I hear that a cello may be able to express the same vibes in its higher frequency region. This is the interaction with the wood which is already audible, and which springs from individual vibes as well. All is amplitude related, and if one thing is so so so much profound already, it is the dynamics with which a (double) bass is played. The string can be pulled hard and soft and anything in between. It can be pulled with full resonance to the cabinet, and with less (making metal sound only). If one of the guys (of three) on the Superbass vol-1 album pulls a string 5 cm sideways, you now can hear he is doing it. This is confirmed by the audience laughing loud because of it. Before ? before I thought the guy made a handstand or whatever. I couldn't understand.
The same things happen on a cello but all in a more fragile way (higher frequency etc.).

Similarly I can clearly hear the veil of a floor tom vibrating, while knowing from my owm drum stuff that a higher pitched tom clearly can do that too audibly.
So it is these areas where the next improvements will be I think; obtaining the higher frequency individual vibes.

Now you all could tell me that you experience these things for many years already ...

So far for now.
Peter

hello P
I know how to connect but I´m not sure I´ll make it in time.
Don´t you lack a common star ground in your drawing? to chassis even?
only difference in my place is that I will use Doede Douma´s USB pcb to start with and it needs a separate psu that´s almost completed.
Status:buff and ivy are connected together
psu´s  to BUFF and Ivy are not
x-formers to psu are connected and delivering
usb to Buff is not
ivy to output is not.
need a couple of hours more to complete.
PUHHH
 :) :( :)
best
Leif

by the way
why are we obsessed with firewire to i2s ?
the bottleneck of the usb 2.0 is obviously not the cable but the chip and the driver.(if I understood right)
once usb 3.0 is released we will probaly have a usb 3.0 i2s bus soon.mentioned this for DD and he was not unwilling to make a new adapted usb pcb for the usb 3.0  std once it was released.
the beauty of the usb std is the "autorecognition" feature once you hook up a new device.has firewire this as well?
OR?
best
Leif

Leif,


I don't think it is needed, and may even be unwanted (isn't all setup in "double" ?). And if you don't have a chassis, what to connect to then ? hehe
I think you'd connect ground from the wall to the chassis (and then sure not signal ground or whatever either !).
Note that overhere we're not all that difficult about proper grounding (for safety I mean now), which is very different to e.g. the US ...


This depends a bit on the design, and the whole chain. In my case (NOS1) the DAC relies on the jitter of SPDIF which is higher than from I2S.
Besides that, again in my case, I'm passing through SPDIF from "a" soundcard (the Fireface) which obviously isn't the best when it can go directly. Also in this case, I would not have a soundcard in the chain, similar to USB. And of course, this allows for 24/192 right at this moment. Added to that that Firewire isn't depending on proper (timed) audio data as USB is, so all together this is IMO *the* option to go for.

Important : The ESS is *not* influenced by SPDIF vs. I2S for jitter, as the NOS1 is not in oversampling mode (which I don't want to use :nea:).


No ... And this is exactly my major problem. I could write all the driver stuff myself, and I *still* don't succeed. There is just no market for this right now, unless someone orders a million of these "interfaces". But usually I get where I want. In this occasion it will take much time.


Just be careful to measure the output V of the PSUs before connecting (+/-15V vs. 6.6 vs. 6.0).
Also, (I think !) do not test the IVY without the DAC hence load connected. If you want to test it without the DAC, ask Russ et al for the load hence what 5W etc. resistors to simulate the load with.

I hope you get it going !

I'm reading with interest, if not real understanding, through this thread.  But, I thought I'd chime in on the usb.  Perhaps usb standards will change and whatnot... but, the thing that threw me off usb (usb out of computer into my Stello-usb dac) is that everything I tried seem to make a difference under usb.  On the usb outs from the computer, (both laptop and the "super" computer I built heh) I could always find one that sounded better.  Having any other usb device on with the dac-usb made a difference.  Shutting off all but the one usb port feeding the stello made a difference.  Every cheap usb cable I tried sounded differently (I still hold onto my favorite one in case there's a reason to go from the FW I use now back to USB ... it is a no name silver braid type that is much better than all the other decent cables I tried) ... and don't even try cables with ferrite beads, all three I tried sounded worse than my silver braid.  I even tried optical usb with a battery supply; it did not sound very good compared to my cheap silver braid 2m one...
point is
so much variation with usb makes me feel it is fragile  ... firewire has seemed more robust.

IT PLAYS! :) :heat: :clapping:
excellent heights on my sennheiser hd650 in the basement workshop
have made an extremely loooong playlist in xx to burn it in
will test on big system on thursday
now: REDWINE and wife! :wacko:
best
Leif

GOOOoooood !

Might you have any special tips for me (including which wine), I'll be happy to receive them.
Very glad to hear that it works !!

Now, stress is over, so prepare for Florida. And for your wife of course. :)
Peter

no tips needed; very straightforward
if you stack Buffalo on top of Ivy to have access to the Ivy dip-switches it´s smart to connect inputs from B.,outputs and  psu connections on Ivy BEFORE you mount the Buffalo on top.otherwise you´ll have difficulties.
also take into account signal cable lenght between the cards so you can "flip" them apart(as if they´re hinged in one end), even though they are connected with wires,for possible future access needs.
ref .support it´s no problem firing up Ivy w/o anything connected to outputs as "load"(e.g.for testing purposes)
I went for scheme 1 in Ivy manual but R 17-20 are replaced by jumpers(0 ohm resistors) and you´re supplied with C 13-16:THE MANUAL IS NOT UPDATED ON THIS.
also with usb in all dipswitches are in "-" position
DD´s usb card worked just fine.
good luck
brb
best
Leif

Thank you Leif. I will do mine most probably the upcoming weekend. :heat:
(still working on XX "problems" currently)

good luck
the funny thing about this dac is that I noticed details in the hights,that I never knew existed! :)
this despite that the hd650 is only driven by a vintage Pioneer integrated from the early eighties
extreme resolution all over the spectrum
will test on trios tom.
best
Leif

#¤#"/&&%&%/  it sounds way too lightweight! :( no impact!??
hey but wait ,my wifie has stumbled in the powercables to the right basstower :innocent:
he-he .....now we´re talking business! the bass resolution and impact is tremendous!and everything is ahh,soooooo relaxed, despite the extreme resolution.3D is out of this world!in full width :o
whow this is hot :heat:
anyone want a ticket and packet holiday with complete family to Florida? :teasing:
I don´t think I have the time :)
This is f#¤%)/¤ close to a clean world top analogue rig w/o the sparkling fire clicks! :clapping:
and this is with a relatively lightweight, if well implemented psu
the whole shabong is only thrown into an old hammond steel chassis
who cares,it sings anyway
all 4 now! the packing was done in 5 minutes and I´m into my Cancannon Petit Sirah redwine glass.
holy sh*t,I´ve never heard those miniscule details be4.
Peter get on 2 it right away! it wipes the floor with the 5 tower dddac and the 4xad1865 AD labs from Romania
merry x-mas and a happy new year to all
best
Leif "soon off to Florida" Christensen

Me !!! Do you have a ticked for the whole family ?
We were there last May (and in a couple of other states), and we're still talking about it every day.

Anyway, this is a nice base ...
I will build the DAC next Saturday, and I will be using that powerful PSU ... (but I have the IVY as well).
So, may you be able to find an internet connection any 8 hours after Saturday ... I will post the results here of course.
But I'm not sure yet how it can be better than what I have right now hahaha.

Have as much fun as we had and say hello to many nice people !
Peter

he-he good luck
I´m off in 8 hours
best
Leif

Quote:

"I will build the DAC next Saturday, and I will be using that powerful PSU ... (but I have the IVY as well).
So, may you be able to find an internet connection any 8 hours after Saturday ... I will post the results here of course.
But I'm not sure yet how it can be better than what I have right now hahaha."



Hi Peter,

You ventilated your thoughts about completing the Buffalo last Saturday
I am just curious about the progress with your Buffalo and Ivy. Looking forward to your esteemed thoughts referring to the two items, maybe with its standard powersupply. Hope you share with us what your planning to mail to Leifchristensen about your findings?
Thanks for the beautifull player btw!

Ed

Hi Ed,

Things worked out differently from what I planned.

1.
It appeared that it is not just a matter of attaching the Buffalo to the "powerplant" :) because the output of the Buffalo is + and - volt while the powerplant was setup for balanced DACs with + only, hence + and - connected (I hope you can follow what I mean, but this is about 2 DACs per channel, not in parallel, but balanced).
Of course I can change this, but it means work which implies work again when it needs to be reversed (and remember, this is only testing around a bit). So ...

2.
So I deciced to build the IVY and all, which was completed Sunday morning.

3.
From then on I'm struggeling with the Buffalo DAC ...
RCA out gave hum all over (and I mean MUCH hum and a little bit of music), and I decided to setup a balanced means of output. Officially my amps are not balanced, but since I planned to make 'm like so, it was a good reason to do it now.

4.
This means of "balanced", gives some (but too much) hum by itself, thus also to the NOS1 DAC.

One very well could say that right now I am in a big knot of ground loops and 10ths of meters of shunting wires are all over, and the one setup doesn't suit the other change (and creating one acceptable setup really takes hours).

And additional problem I have, is that I play without preamp, but when such a hum is exposed as the Buffalo does so far, this is not allowed (breaking windows). So, the pre-amp has to come in again, and that by itself destroys the ground loop setup.
So I'm in circles ... :wacko::wacko:

But as soon I have some result, I will post it !
Peter

PS:

Hahaha, I gave a smart answer there. Now I better ask you Leif !

Colony Hotel Miami Beach calling 8)
I use the original psu´s that deliver +/- 15 to the Ivy and the 6,0 and 6,6 to the dac.
I did not get fifnished with grounding scheme but it was dead quiet but had sonic dropouts due to gnd loops.
Acc to TP guys I will connect all gnd between the psu outs (the minus on the dac supply and the gnd on the Ivy supply)together and the DD usb pcb that only gets 11vdc will also be connected to the other psu´s by the gnd terminal.NO gnd between dac and usb card I2S out.
this way all should be well
if problems I´ll check this page L8R 2 day
best
Leif

Hey, thanks Leif. I will probably working on it tomorrow or the day after (in between eating stuff :)).

Ok, up and running here. Can't think of why it didn't work before, other than the rather unofficial way I stacked it (see the locomotive below).
Before I disassembled that several times at trying, so it wasn't about a loose connection I think. Now both halves are loose and everything looks right at first glance.

Must test it for its merits later; today there's visitors. And Chritsmas music only of course. :)

ok looks fine but I would not use screw terminals for signal connections
just my 2 cents
best
leif
and now I´m off to radio shack key west with my boys and wifie,since it´s raining anyway :)

Ok, this is merely for fun;

Look at the description below. THD is less than 0.0001%. That is goooood for a DAC. But what does it actually say ?

The two pictures show the result (measured at the output of the DAC) of feeding the ESS Sabre DAC with a pure square wave. The first picture shows what's left of it at 3000Hz and the second shows a nice pure sine at 10000Hz. I talked about that earlier in this topic.
To me the second picture shows a 100% distortion. But hey, what I do here won't be a measurement of THD (http://en.wikipedia.org/wiki/Total_harmonic_distortion). Right, but what to do with THD if this is the real life result ?

hmmmm
have you actually LISTENED to it?
or like Gary Dews of Borderpatrol said:
if it sounds good , it probably is 8)
best
Leif

But he copied Duke Ellington I think. :)

But wait, I have a verdict by now. Just writing it ...

... which is about the ESS Sabre DAC-chip in Buffalo application, with a sauce of "my" NOS1.

Well, first things first :

If there is any DAC that mimics vinyl, it is the Buffalo !!
I have been listening to the Buffalo for two (rather full) days, and stuff like from Henry Mancini just pictured my parents' living room with Thorens, Leak tube amp and Leak speakers (my father was a musician in an orhestra, teacher at the conservatorium and had a recording studio during his last years).

Until some time ago I always tried to copy that sound, and I never succeeded. And suddenly there it was ...
There is this massive wall of violins, and a great ease of letting everything sound right. Good positioning of the orchestra's instruments.
A great ease ...

Yeah, violins roll over you indeed like a wall of sound. That orchestra really is massive.
A bit messy too ...

And then it started to disturbe me. Why ? how ?
It seemed I had a brain damage;
Something inside me wanted to hear which instruments were playing, but it could not happen. I heard copper, but what copper ?
I heard reeds, but was in disagreement with my wife whether it was a hobo or a clarinet. No, a sax would be out of the question. Or maybe it was ?

I heard massive violins from my early days, but I could not hear *the* violins. It could be 4 or 12 or 35.
I heard bassy sound, but from what ?
I heard smashing cymbals, but they were grey ...

They were grey like cymbals from vinyl are grey. Yeah, everything is grey. Greyish.

Throughout S's are too harsh. When 20 background girls are singing the volume must go down.
Still, the same sound as from my young days. That was hi-fi ...

Today at 10 am I started to play my regular bunch of test records, and if one thing was not lacking it was detail.
Hmm ...
Also played the "square wave" tests (mainly Yello) and it ocurred to me I wanted to live with it. But (like with vinly) after an hours or so it occurred to me that all cymbals sounded the same. And again grey. Good by itself, but grey and uninteresting.
Bass seemed to work ok (better than my exptectations), but not as how I got used to lately. But still ...

Then I found that actually nothing got me into the music. I heard basses, I heard saxes, I heard voices, but they were all without feeling.
Was I focusing too much on technicalities maybe ?

Then it occuurred to me that actually dynamics were missing. The thrill of the metal at (double) basses wasn't there. There wasn't any smashing, no hitting on cymbals, and, well, trumpets were too smooth.

A great deal of the day I had in minf that the ESS people possibly had overcome the rounding of squares because of (massive) oversampling, just because all the detail which could be heard. Yello sounded different, but again I was ready to accept that.
But why were the musicians not playing for *me* ?
That's when I grabbed the scope (see earlier thread).
So I really tried and was completely open to it. In fact I at last heard what I had been seeking for so long. The sound of the living room I grew up ...

---

I must tell you that there another small background to really want to try to get the Buffalo "work";
The NOS1 exposed noise which made me mad. Some high frequency stuff I couldn't bear. HF stuff that goes with you to bed, and with which you get up in the morning. It got me crazy.
Most of last week I have been trying to get it away, after I first could measure (by microphone) it really was there. A 17,5Khz tone which was 20dB louder than the music ...
This story by itself is many pages longer than the few lines I write about it here, but what it comes down to in short is that while the Buffalo would not allow this tone (which really is a steady tone) to be "amplified" hence changed at attenuating via the TVC I have, the NOS1 would. But this works exactly the other way around as well : the TVC can attenuate it with the NOS1, and in combination of things, it could get me the tone just above the noise level, as it could in the first place with the Buffalo.
More importantly, early this morning I found what causes this tone : my 380V pump of the central heating ... Also, other tones (at 15KHz and 19KHz) are there from the freezer in the basement. And although I have completely separated audio power groups with own earth, nothing can get it away. Not even a sine re-generating battery power plant !
But in the end I could get it down into the noise level, so all is fine for the moment. One thing : I have to use the TVC for it, which I really want not. New freezer and a new pump I guess ...

---

So at 4pm I changed back to the NOS1, and ran Mancini again. Right, so now you hear *the* violins. The background women can be counted again, and there's an enourmeous separation in everything. Background is black, and a brush sounds different from a hit on a chinese cymbal.
Pwew.

No, Mancini does not sound like in the young days, but hey, I don't want that !!
It is spirit I want. So yes, now the musicians are performing ... performing expecially for me.
No harsh S's anymore.
And that bass ... ha ! back again.


All together the difference between the two DACs is huge;
I truly believe the Buffalo may be the best oversampling DAC around, but it doesn't work *at all*. You're faked by its presented detail (btw the biggest trap anyway, like with T-amps and such).
Right now Jazz at the Lincoln is performing. I can tell you ... *that's* talking trumpets !

But what remains is the vinyl addicted. If I was one, I really would go for the Buffalo. I even don't recognize the PSU is underpowered. What made it sing, though, was connecting the ground of the SPDIF to both of the interlink grounds (at the output of the DAC). Possibly this is my situation only.
The plus opposed to vinly is that this goes lower (anyone heard sub-low from vinly ?).
So, if you imagine that squares can't be in vinyl grooved anyway (but what about reel-to-reel), technically the ESS Sabre / Buffalo might come just close - if not equal. Including the lacking dynamics ...

Now, don't shoot the piano player please. I really tried. :evil:
Peter


PS: The first picture below is the NOS1 filtered, and the second one filterless. Both at 22050 Hz this time. Not that this shows anything of the filtering itself (which would be about the frequency spectrum to pass through or not, although I tried 30000 Hz which shows the same; didn't try further), but there clearly is a difference electrically. I use filterless ... (which might change at decent A-B-ing later)
Dunno why there isn't any ringing with the filtered version, but as you can see the slopes of the squares aren't as steep.

Have you ever dreamed of running the audio completely off the electrical grid? I have, heh.  Hoping for solar power, wind power... discovering some fast running underground stream to tap into...
I've shut off everything in the house at the circuit breaker level except the subpanel that runs the music system.... still get noise from the power lines... and I have a psaudio power plant running the front end and the computer on opposite leg of the audio equipment in the sub panel.  hehe oh well, I don't think I have a 17 khz tone, but a bit of a hum that can grow inexplicably louder once in a great while... knock on wood haven't heard it for a long while.

Be careful. Why would I think I have such a thing ? It is the pure coincidence that the NOS1 amplifies it, and after two weeks of not attenuating that (no preamp) I got crazy of something. I heard it was more noisy than it sould, but that is just white noise. It's not what got me crazy.
And then I got the glorious idea of measuring with a microphone and prove whether I maybe heard ghosts.

With the amplification of the NOS1, this goes to -60dB ...

It seems things all turned out well for the NOS1.
Now, I'm only waiting to listen to it myself ;)

Me too Telstar... :soundsgood:

Gerner

Hi Ed,

Although I gave some implied answer by describing the Buffalo experience, here is a response from some other angles :

I think we are all seeking for 100% natural playback through speakers. Possibly 99,99% of people may think this will never be possible because of various indirect reasons like "the room", "the speaker", "our ears", and any other IMHO *not* legitimate reason not to try.
I do, and I do that since I got myself the combination of an NOS DAC and (very) high resolution speakers, a few years ago. I took the hurdle of leaving "getting rid of disturbing things" and plunged in the pond of "how to get it more real". It is my assumption that 99,99% of people (not necessarily the same as the beforementioned) are still struggling with the "getting rid of the disturbances".

What I tried to express about the Buffalo (but did not spend too many words on in the end) is that it is so easy to see how the Buffalo will fit into "a" system. BUT this is related to how difficult that is for the NOS DAC.
With the Buffalo nothing seems critical, and the music flows to you as -indeed- vinyl would back in my old (ehh, young) days. I have things in mind like
- when the amp is not 100% it is not problem; the Buffalo will make it sing anyway;
- when the resolution of the speaker (or tweeter) isn't all that much, no problem; the Buffalo will get to you anyway;
- when you are not a critical listener, the Buffalo will be way better opposed to whatever it was you had ...

... when you had an oversampling DAC ...

To put the above into the perspective I felt all the time at listening to the Buffalo, each and every second springs to my mind how much more difficult it is for the non-oversampling DAC to perform;
You may recall the earlier days of the NOS DAC, where known people (a "Garmt" springs to my mind for our area) tend to be so sure that NOS was nice, as long as the whole orchestra didn't start to play together. Things get wild and rough and "difficult".
It is this what I got rid of a few years ago, and it is exactly this what I heard back through the Buffalo as being no problem at all (which says something about my gotting rid of it ... uhm), and ... it is more of a problem with the NOS1. :unsure:

Since I haven't been used to an OS DAC for a few years, the step "back" to that is bigger as how I recall the stepping from it back then;
Strangely enough the Buffalo even emphasizes the OS character, and I mean far more than my Fireface which I now and then used for testing 24/96/192. Thus, the Fireface is poor on all fronts, but I never perceived the OS principle form it so much, apart from synths being uninteresting.

If I would try to put things in a negative perspective for the Buffalo, I would say : this makes a mess of it all the best way possible.
But my context is not negative, and I really want to say something like : it is able to cohere all the different voices and things so that all is smooth and more smooth. As a bonus there's loads of detail as well.
And my objective expression would be : but smooth = smeared. Smeared at all the levels. The lack of dynamics (remember, compared to NOS) contributes to that. No, it is not the other way around, because the process itsels (heavy oversampling) just implies smearing. Or smoothing if you like. And *then* the dynamics have gone.

Dynamics is a technical matter, and it don't think it is a fact than you need them. It is a dimension yes, but by itself a dangerous one if your system is not up to it. E.g. when the speaker can't follow you end up with distortion because of it. So it depends ...
But :
Without the speed of transients, it is sheer impossible to mimic cymbals in the most natural way. Cymbals as *the* example for me to achieve in the most natural way the pas few years, and where I reached a limit at some stage. Or better : XxHighEnd seemed to have reached the limit, and I imagine an earlier version to be better at it than the current one. But has it ?

In fact the Buffalo -as a good OS DAC- shows what is happening really :
Where XXHighEnd can hardly deal with more than jitter influence at the source side (and a little more), the OS DAC brings you back into the world of disturbances. The cymbals are just gray again. Not really distortion like, but just gray. Nothing much different from vinyl I'd say.
In *my* case another type of "disturbing" entered my life : my brain fighting with the instruments not being recognized anymore. Try to imagine this : without the knowledge of this being possible, no problem (no matter a live performance tells you different), but with the knowledge it disturbs all over.
I am fairly sure that anyone being used to an OS DAC in the first place, would not be disturbed by this. Like gray cymbals are common, and the best to achieve as long as no distortion is heard.

Now back to the NOS1 ...

Besides the bass I raved about throughout in this topic, I now rave the same about the cymbals;
I have loads of recordings from which I would swear the whole of the performance can't be bettered. Including the cymbals and all the variants of it.
I told it earlier, now it comes to be that brushes (on a snare) are more difficult than enything else. Before (with my old NOS DAC) they already got out of the noise (two years ago I was listening with my ears in the speaker to find out what noise I heard), but only being a so loud(er) noise that it couldn't be noise itself, and you could regocnize that brushes were going on. So, that was one "disturbance" less. Today ? hmm ... like every bassis is suddenly plunking the strings by 5cm, all those Jazz idiots seem to prefer brushes over sticks. Yes, they are up front now ... that loud.
But they are not good enough. Too less colour opposed to reality. Too much "noise".

The latter might be an example of what may happen with the NOS DAC. It puts every instrument in itself, with the consequence it must be good. If it is not good enough, but the instrument is profoundly there anyway, you have a problem. A relative problem of course, and maybe I am the only one having it (possibly stribing too much for perfection).

When I say I have loads of recordings which just mimic reality, I must add to that that I also now have loads of recordings that shout;
Most people (if not all but me) would say that this is a bad recording. Maybe so, but my experience tells me that when an album doesn't sound good, it is the equipment (the software player being "equipment" just the same) doing it to you. It happened so many times, and I just know I am right in this.

It very well can be so that the NOS1 runs me into limits of my system. And it can be;
A long time ago I measured my old DAC for squares with the scope, and I don't recall anything so beautifully remaining a square as I showed a few posts back. This was merely like the 3000Hz picture of the Buffalo, but then at 20KHz. So, the NOS1 should be much "sharper";
Besides the more fragile detail (like hearing the feedback of a cymbal onto the stick) this creates a "force" in the mid area which currently comes to me a standing waves in that area, 95% certainly incurred by my horn speakers and the internal resonance. With XXHighEnd tweaks this happened before, and a trumpet started to be able to resonate the horn, and perceived the sound of the horn. This too was just more square sound, and the voice of Mark Knopfler could do it too.

The latter again is an example of something which would never be incurred by the Buffalo (hence ESS Sabre chip). Never.

So ... all 'n all you see that NOS (and certainly the NOS1) is way way way more live like and natural when the rest of your system coorporates. But if your system does not, you run into problems much quicker with NOS.
For me, right now, this means that I must try tweaking with XXHighEnd first. I never did that so far, and just used the old settings. Wasn't of any use so far either, because just yesterday I closed the case the NOS1 is in, and all should sound like ... well, intended.

Peter

Good, you closed the case!
And did you put a wire from the case to the real earth?
I experimented with this lately and found this removes all the electrosmog and makes the high frequencies really shine.
I think the best way is to make a central (city!) heating (in my case) or a waterpipe as a star earth to all the metal cases.
But then the cases have to float and not be connected through a balanced wire or electrical earth connection through te mains.
In my cas the vinyl got new live after I connected it this way at the most sensitive part (the arm) to the central city heating.
Big surprise, XXHE sounds better too! (Just my 1 cent)
But only do experiments with this if you know what your doing!!!

No, not yet. First I want to see how things work out the usual way (for me, which is exactly like you are doing).
In this case things might me different, and since it would be the first device connected to mains ground anyway ...

Don't make your starearth like this:

Ok, a small update.

I think I have everything under control for connections and the best sound, and I must say that besides all the virtues I mentioned earlier, the DAC now also sounds "sweet".

Sweet is in contrast to the crispyness and sharpness (sharp in the good sense !) an NOS DAC usually shows, and which can be more refined. Thus, that crispyness and all, also shows a kind of roughness, and where "roughness" is a negative, I managed to sweeten that.
Btw in due time I will start measuring the DAC, and show some figures.

What I didn't tell so far, is that nearly all preparations have been taken in order to produce this DAC and sell it. Yep, and in its own dedicated "Phasure NOS1" cabinet. Before production really can start though, one thing has to be worked out and this is a tough one. This is btw also the reason of telling about the production, because I know quite some people are waiting for answers on this, but in the mean time I now know this is going to take a while. So Yes, but Wait.

So what are we waiting for ?

With my ever planned 32/384 DAC I planned a special connection to the PC which has never been done before. That is, not that I know of, and there isn't any reason for it also with other manufacturers;
Then, in this topic I expressed about wanting a general I2S connection. This I2S connection should work for PC's just the same. And as you may know, it is (AFAIK) the most jitter free connection;
I tried this with Firewire as the first carrier, but this is a dead end. I just can't find any Firewire boards with manufacturers that want to sell something, or otherwise the drivers s*ck and access to the SDKs etc. is (too) difficult.
So now I combined my old plan with this new DAC ...

The good news is : I have it working. Thus, no SPDIF, no USB, no Firewire/ADAT or something, but native I2S with a "secret" connection to the PC.
The bad news is : the stupid DAC board only accepts I2S when in Oversampling mode ...

So, after all the hard work it now appears that I2S cannot be used in NOS mode. Yuk.
And "sadly" the nature of the I2S connection indeed sounds better than SPDIF, and it just gives the sweetness I talked about in the above. I know, comparing is a bit of apples and oranges, but since SPDIF can be used in OS mode just the same, this difference in the nature of the sound is very clear.

Again (like with the ESS) 1 million people most certainly will love the sound coming from OS mode. But this will last until they hear NOS mode. Again OS just doesn't cut it, and compared with NOS mode, OS to me sounds like a dead bird. Too sweet, and nothing seems to jump out, while I am sure each musician does his/her stinking best to let his/her instrument jump out. With NOS they do ...

All 'n all right now I am into deep trouble. Well, kind of;
The DAC board wasn't explicitly designed for NOS mode, and although everything looked good so far, this is where the tweaking stops.
Someone has to go back to the drawing board ... :cry:

Peter

pity
but you´ll come up with something
in the meantime:
the buffalo sounds excellent despite os.
BUT the sound varied very much with gnd scheme and after I replaced the onboard LM317 regulators on the double psu for the buffalo with LT1085 the sound took a turn towards sweeter and more nos like
will do same on ivy where 317 is replaced by lt 1085 and 337 by lt1033.
that´s it for now
best
Leif

Wow. Breaking news! Do you have any estimated price? (At least, can you give an indication?).

Yes ...

For those who already knew (offline), what I calculated so far was 3450 euro ex (if applicable) vat and shipping. However, this anticipated on an onboard Firewire connection which now is to be replaced with the "secret" never done before other connection. This is more expensive (in fact just additional), and although I know what the kind of maximum is, theories tell me that there is a low boundary of 350-450 euro. But :

At this moment I can't tell whether it really will make THE difference because it cannot be tested in NOS mode (read : I won't construct a first version of it, not knowing yet whether it can be used or not).

Peter

 :o :wacko: :scare:

Hahaha..... I think i am going to work some extra hours from now on.....

And offcourse a special candlelight diner with my wife!!! :rofl: :rofl:

Hahaha, not quite ...
Three rather important things "happened" :

1.
Yesterday I thought "well, if the thing wants I2S only at high sample rate mode (which is what it comes down to), I'll give it that".
And thus, from within XXHighEnd I could use the Quattro mode (for 44.1 files).

2.
While initially the above had been in my mind the whole day yesterday, and I was wondering whether upsampling from the software would workout better than the upsampling by the SRC chip, at attemping this means of playback it sprung to my mind that I ever developed "just Quattro" mode, meaning a higher sample rate only, and no upsampling. Hahaha, that would do the trick !
Note : this means that all the squares from the earlier scope picture just would stay like that, and each sample is just fed 4 times without adjustment (this was back then made for better jitter rejection).

3.
Playing for 1 minute with immediate good result (at 176400) a next thing sprung to my mind, not thinking it would work, but hoping for it ...
The means of source :secret: I use for the I2S connection is quite different from usual (no Fireface in between), and I notched down the buffer of that means. Note : for 176400 via the Fireface a sample buffer of 128 had to be used (96 or less would not work), which btw is 128 + 64 in real life (the 64 is always there, and is called a Safety Buffer). Now ... I notched down and down and ended up at 48.
Hahaha, so with I2S I now can use the 48 sample buffer for all resolutions ! This is very important because the size of that buffer changes SQ (for SPDIF it does anyway), so the apples and organges thing is out of my life forever ! Only now lowres files can really be compared to hires files !

How does it sound ? (said the reviewer :))

Well, first of all the bird is fully alive now !! But wow, what a different sound once again !
I must again emphasize on "sweet"; Sweetness is what can be heard all over, but the combination with other phenomena is strange :

Now the crispyness is still there. Ok, hard to explain, but as it appears now the sweetness I heard from I2S at OS mode indeed is a property by itself (as I expected). Thus, while OS mode removes the cryspiness and the lively sound, this is back now, but the sweet touch of everything smelling like Sses is still there. Cymbals did not loose there metal (which added "sweetness" would incur for), and all squariness from the NOS DAC is nicely there.

Before I continue I want to make clear somehow how huuuge the difference is with SPDIF. In fact this is just two complete different DACs now. The both connections cannot be compared. Just not. Additionally I must say that I never read about such an enormeous difference between SPDIF and I2S, although everybody does hear "the" difference. This is just not normal, and I am not sure what to think of this. However, theoretically there is this difference :

In OS mode the jitter specs are a few ps only. I am using that now.
In NOS mode jitter is subject to the amount of jitter the DAC is fed with. Remember, this is why I wanted the I2S connection, because I wanted NOS mode. The jitter from SPDIF can well be 200ps or much more. From I2S I don't know it, but I already have my thoughts, looking at the scope and the way steady picture of the perceived clock data (I will measure everything officially in due time).
So, I think I can fairly state that the sweetness is coming from less jitter. Or the other way around, the roughness as how I know it, comes from high amounts of jitter.

Keep in mind, I am doing something here which other people didn't do before me, and I mean not in the consistent setup of (actually) NOS, a most high grade PSU and tweaking an I2S connection from a PC. And some fine speakers to hear the difference within a second and blahblahblah.

Now, proceeding with the sound quality it seems that I have to remap my mind/memory/expectations from music through loudspeakers;
I know the sound of NOS so well (what about 30 months in a row listening for differences at developing XXHighEnd) that it (yeah what) has become a reference for me. A kind of absolute reference on how e.g. a synthesizer must sound. The same counts for trumpets and everything with very high transients. Of course because it all sounded natural it was allowed to have that reference. But the reference was wrong ...

Earlier in this topic I think I said (or maybe it was offline somewhere) that the "sharpness" of this DAC (just think squares) made my horn speakers resonate at certain frequencies. Also, those frequencies would create standing waves in the room, and e.g. women voices may become unevenly harsh spread over the frequency spectrum concerned. What I did not tell is that I got rid of that by means of some "softener", and the softener just was the (TVC) preamp again. It rounds all squariness so much that all just is fine again.

But guys, the only thing this DAC does is presenting everything very very good as how it receives it. I showed you the square wave at 22050 Hz, and if you can find one picture on the Net that shows a better one ... be my guest. And that was SPDIF with its jitter ...
Now, what the I2S connection shows is that the additional "roughness" the DAC expresses is just jitter from SPDIF; Currently there is no such thing anymore of horn resonance or standing waves at higher frequencies.

The strange thing (for me) at listening to synthesizers, and of course compared to before, is that there now is again "sweet" detail in between the squary tones. Possibly this is best explained when I refer to the feedback of the cymbal to the stick which became audible with this DAC, which has the sweetness I refer to. This is now everywhere, and even a trumpet is not *as* squareish as I thought before.
And please keep in mind how difficult it all is, because like one can just never know how a synth should sound, one also can't know how square a trumpet is blown, which in the end is the same as one can't know how hard the strings are pulled from a bass so metal sound comes from the strings. So, all these things could trick at least me, and it now appears that beyond "natural sound" another dimension exists : real sound. So let's say that natural sound doesn't sound wrong and mimics instruments in general, but real sound makes an instrument sound as how it was played. Hmm ...

The latter gives quite some opportunities. I mean, having all things sounding natural is what I have been working on for a long time, and once you think it can't be improved, the fun is over. But when we now discover that there's also this other dimension, we can gladly be in search of improvements forever, because we just won't/can't know how instruments were played.
I say it once again, the metal from the basses is the most profound thinkable, and there was really nothing in me that could expect it was in the data. Still it is and it could be unveiled.

Right. So the DAC is ready now ? Hell no.
With the just learned data I think some other things can be tweaked. The I2S connection is not the final one, but I first want to measure whether this is ever going to make a difference.
Whether I faked NOS or not, with the current setup all is going through the SRC. Not the best for theories.

To be continued. :yes:
Peter

This is is fantastic reading, Peter!

I had a NOS DAC myself for more than a year, and I agree 100% with the "sharpness of transients" which you refer to. This kind of realisme I have not heard from any other DAC. In the end I sold it though, and bought a more "conventional" OS DAC - my present Stello Eximus. It doesn't have quite the same "sharpness" or slam, but is muuuuch more detailed and relaxed. All in all, a better SQ. (But I still think about my old NOS when I am lonely in the night. Like an old lover). The point is that my NOS DAC (DDDAC) was made with obsolent mid-fi Philips DAC-chipsets, which probably put a negative stamp on the final result. I am very interested in re-trying a NOS DAC which is based on a modern DAC-chip like the Buffalo.

Something else, Reference Recordings  is now selling true mastertape copies with original quality 24/176.4. 176,4 is quad of 44,1. Can you make XX work on this 176,4 samplingfrequency?

Haha, I don't know whether you meant to say that the Buffalo is NOS ? but if you did ... it is far from it !


Well old lover, this should be in there since 09u-12 (April 4 2008).
Did you miss it all this long time, or don't I understand what you want here (like playing a tape maby ? :blob8:)

I missed it! These hi-rez files from RR has just entered the marked. I havn't bought any, but it is in the pipeline. Good to know XX are allready there.

Sorry for mixing up the words here. What I was unclear about, was that my previous DAC also was FILTERLESS. No anti-aliasing, no roll-of. My guess was/is that you are using the Buffalo in filterless mode too?

Ah, the old lover is a bit confused !

It looks like you haven't read the whole topic here;
Although the Buffalo was mentioned here and there, and although I indeed built one (as Leif did) this topic is about my own assmbled DAC, worldwide commonly known as "Phasure NOS1". :swoon: Okay, in a year that is, I hope.

The Buffalo (AKA ESS Sabre) can't work filterless, as this is a heavily oversampling 1 bit Delta Sigma whatever DAC.
I would not call that sweet, but very good instead. Very good if you don't like NOS or just didn't experience NOS. Although I really cannot tell, with a decent PSU it is most probably better than your Eximus. However, if I had to listen to it as "my DAC", I'd go to bed early. Like an old lover acting young. Well, it's all been said in an earlier post in this topic.

The NOS1 indeed is assembled as NOS and filterless. But it can run in several combined modes.
To avoid confusion : what I was talking about in the earlier large post, was OS mode, tweaked to NOS behaviour running 16/44.1 at 176400 and without filter, connected via native I2S.

For your interest on "modern DAC chips", this runs on 4 PCM1704UK chips (2 x 2 balanced mode).
Peter

OK. That is also a top of the shelf DAC chip. Looking forward to follow your development!

What material are you going to mount this beauty in?  I'd prefer your cardboard setup to metal.  Think of a nice thick piece of not very dense type of wood  (experiment with varnishes) with a nice thick acrylic top...

and a
12V car- or motorcylce battery

acrylic gets static and has no schreening properties and is thus a STUPID choice
imo
best
Leif

I don't think it will be wood, or it must be additional decoration. Wood doesn't act much as a shield, and to me it doesn't look exactly better either. Personal of course.
And as Leif said acrylic ... the same.

I am still fighting with the I2S solution and the space it will need, which is the main reason I can't proceed on this hence show anything. But I will as soon as I'm done with it.

On battery power ...

So far I don't see (by measuring) how that would impact the performance of the DAC. Not in this case, not with this PSU.
This is all in the context of the distortion (http://www.phasure.com/index.php?topic=693.0) I have on my mains, which is rather - no the most bad anyway. Note though that a distortion like this - and as how I expressed it can be noticed (just through speakers) is unrelated to the provision of power. IOW, there is no way this tone is coming into the DAC, and that it amplifies it or wants to D/A it or whatever.
The DAC is about the most steady PSU for voltage and current imaginable, and it does just that. For example and your imagination, if I'd connect a 100W lightbulp to the supply the DAC is using, and assuming all can bear the amperage coming from that, nothing will ripple or change. This is the self defending mechanism of the PSU, which comprises 95% of the DAC circuitry. This is also why the DAC has such a low forceful output, because nothing can steal current from it, and with a DAC this would be mainly the DAC fighting itself.

Besides this, like many I have my doubts about battery power;
Dave, is you have a PS Audio and you have it for quite a while, move it out of the way for a few minutes and listen. If you say "all is worse" then you're ok. But I don't think you will say that.

Of course, battery power for main amps etc. is very different from battery power for DACs, and main amps surge more. But there is more to it (the theories) and together with me not liking the result those theories might just as well be true.

Then there is also the given fact that this is a device which fully operates in Class A. You can read the newspaper by the light emitting diodes used (which are used as speciasl diodes) and the thing gets warmer than my Audio Note tuby DAC. And there goes the battery solution.
Of course, on this matter you could use a car battery which will last 24 hours for sure, but it will impact the design of the PSU because all is not exactly running on 12 Volts (and not less but more).

During my measuring experiments (starting this weekend) I will try to find proof of a battery making a difference by measurement. And if any mains is prone to improvement, it is mine.

Peter

why don´t you move into one of these!:

My experience with them is mixed:
Because I wanted less noise I made a psu for my tubepreamp which converts 12V from a caravan-battery to 220V.
A lot of capacitance behind it so worked OK. The switching frequency is below the noise.
When I put a AC/DC supply of 12V instead of the battery it sounded really bad.
Back to my old AC/DC 220V supply was really bad too.(and it was supposed to sound very good)
So I settled for the battery, but still some noise, maybe the tubes?
Then an extra AudynCap capacitor close to the battery and a much lower noisefloor!
Now I don't like this because it means I can hear the quality of this capacitor and of the battery...
Besides the overnight charging which I sometimes forget.
So for A DAC it's a nice experiment but not garanteed to work.
If you can make it to work let us know how it sounds!

Maybe its me, but I LIKE the sweetness you mention. Get that from the absolute best situations here, with exceptionally clean power during holidays etc. when much of the people living around me is away.

Sweetness=Nirvana.

But that's only me... :)

As Leif mentiones, he has very good results WITH OS. Don't think thats a dead cow.

You seem to refer to something that someone (probably I) mentioned as a negative ? or ?
I most certainly did not intend to ...

Found this one 500 meters from my house (this is really true). I could move to there of course, but first I'll investigate getting a cable from here to there.
:)

This one's 150th birthday was celebrated some three months ago. Must have real ancient power.

I just thought it´s a great way to solve your polluted power system,by generating your own in-house clean power
best
Leif

ah-ha!

I knew that you sometime would have made my same reasoning and trying to get i2s directly from the computer.
That is the best single cable solution. The other best is dual ST, bu too cumbersome to be done with a computer source.

Your last email confused me ;)

Sometime ? hehe, I think it is in the first post of this topic, and planned long before that !

But just 10 minutes ago the theoretical solution for I2S in NOS mode was found. Practical takes a few days more.
Will keep you informed. :)

:scratching: Ah ... I now think you meant without Firewire ...

Peter.  With direct I2S coming off the computer... is there a worry about electrical grunge riding along into the dac?  We know there's more to the transport/dac link influencing sound than just jitter... any way to isolate the two better?  any small optical thingamajig right before/after the I2S... obviously I don't know techno, but just throwing it out there.

Yeah some photocoupler like AVM used in one of they're CD players, they lid the whole thing up in a red shine, nice.
Good idea.

I know Doede Douma is very peculiar about keeping I2S cabling from his spdif and usb pcb to dac pcb´s as short as possible,probably due to risk of interference
best
Leif

Yep :)

and i'm waiting for that email :)))

Maybe this will sound "commercial", maybe it is just good to know for people in search for a DAC and bump into this ... but here is another kind of "review".

Spread over two days, a session of 15 hours was spent on a comparison between a DAC based on the ESS Sabre, and my own Phasure NOS1.

The ESS Sabre based DAC this time makes use of the original Evaluation Board, though modded to a high(est ?) degree. The bass output therefore (?) is better compared to the Buffalo DAC. Sound character otherwise was the same, to my ears.

The NOS1 was used with SPDIF connection only for more honest comparison, and it was used in NOS mode as well as in OS mode, and either with and without filter.

Software was XXHighEnd, and only 16/44.1 files were used in a normal fashion (not upsampled or anything).

A kind of interesting twist was that various persons came with different presumptions :

a. The owner of the ESS Sabre DAC being there right from the start of the Sabre, working on it in a fashion of creating the best DAC existing. This is visitor A.
b. Visitor B looking for the best DAC in the world (maybe not regardless price :)) in a most serious fashion I would dedicate to myself only.
c. Me, indeed looking for the best as well, and with a search behind me similar to visitor B, and means similar to vistor A (DIY).
d. Some person D without real interests, apart from the ability to bear the noises day in day out, my wife.

Important to know is that we did not use any music for testing that would put the NOS DAC in the advantage explicitly. In other words, only candle light music was used, much piano, and some toucher of classical here and there.
For those not knowing "my" merits on this one : nothing "square like" music, hence nothing the NOS DAC would do better because of that (remaining squares as they are).

The reason for this kind of music actually was visitor B; He already knew the DAC from visitor A, recognized the virtues, but without being able to describe it, felt "something was missing". You could say that it was the stupid idea to possibly find the missing parts in my own DAC, which is a highly pretentious idea to start with, nobody knowing in advance what would come out. On this matter, I did not know the visitors besides some dozens of emails, and about the seriousness ... they came by airplane.


To get used to my system and how it sounds, we started with the Sabre for some 90 minutes. Where I was behaving the most objective, visitor A got used to his own DAC again, while visitor B again came to the conclusion that technically everything sounded well, but something was missing.
At that stage I knew what that was, but said nothing.

In those cases where objectivity was not needed, I certainly spoke up;
On a side note, I did not know any of the music we listened to.

At one stage, I asked visitor A what instrument was playing. If you followed this topic, I of course internally referred to my notification of the Sabre (in Buffalo appearance) did not do that, and here it was the same. The innocent answer of visitor A was "trumpet. ehh, sax. ehh no ... ?".
Of course this was a kind of mean from me, but was nevertheless related to possibly "something missing", where I experienced that myself and did not recognize it immediately. For the visitors of course this was more difficult because of a lacking reference.

Already at this stage of listening to the Sabre only, I recognized the most strange violins, which were indeed so strange to my ears, that I later (at trying the same track on the NOS1) promised a recognizeable synthesizer at hopping over to the NOS1.

It was really a pure coincidence that my "promise" turned out the most untrue; Instead of a synthesizer a normal violin came out.
Well, with this as the very first thing happening, nobody was really objective anymore, because a violing sounding as an unrecognizeable thing can't be standed. Not once you have the reference. I was ok with it before when the instrument sounded like a combination of a violin with a handsaw and a bag pipe. Of course, the latter was placeboed by the Keltish music (which it was), but still. So, if you hear this back as a normal violin, your brains won't allow to like the non-violin DAC better, no matter for what other reason.
Btw, of course this is similar to what I told a few posts back about Little Feat and the strange noise which just appeared to be a guitar.

In an earlier stage we where kind of always talking about the realism of the music. Somehow it never sounded right, although I must admit we were nitpicks to a high degree. I am not sure whether visitor A would really come up which such messages, but he sure agreed once things were pointed out.
Person D could point out that all was too fuzzy, which was independant of my own stating that things "buzzed" too much. So, I myself recognized the uneven notes on a piano, but since this appeared to be throughout the spectrum, it kind of annoyed throughout. Or at least with my that was so, and apparantly with person D the same. But remember, we (me and D) had a reference.

At a certain stage we at last found a beautiful piano piece of which we could all agree that the Sabre was performing very well on it. So, to be as honest and open as possible, it was my idea to switch the cables for the first time. And to make it a bit pretentious from my side, I had a prediction as detailed as possible on how that piano would change. Remember, that piano (a large wing) that was sounding 100% good to begin with.
I predicted a zillion additional harmonics, a more fresh piano (it sounded late late late night dull), hammers to be heard (they did not at all), and interhamonic "reverbaration", which I later explained by pionting out the 3 strings forming 1 tone on the higher piano notes.

As you have guessed, of course this came out exactly. What I could not have guessed though, is that those harmonics are so much of importance, that the highest notes appeared an octave higher. I never knew that (but never made such extensive comparisons in this area of good music), but at the highest notes the "base tone" seems to be created from harmonics only. They are louder than the actual base tone. In the end I know because I have a wing myself, but it never occurred to me really.

Right. So the nice thing happening (nice for comparing in general) was that while nobody lacked anything on that piano piece, everyone agreed within one second that it was *totally* different. And better. And real.

We were switching a lot, and at some stage a request of visitor B came for a certain track we listened to before over the Sabre, and first we listened to the Sabre again. So, this was the second time;
Try to imagine, me hopping around with cables, your wish is my command at the volumes and everything (all perceivement had to be of a volume as the gentlemen were used to), and as a nice dog I listened again to this track.
Now the track had to be run on the NOS1, and after forfilling my job again, of course the waiting came to listen out that stupid track once more. Life is hard sometimes.
But wait a minute, this time I got something from it, and it sucked me in. With the danger of person D seeing it, I had to wipe my eyes. I saw person B doing the same, and person A had dropped is head down on the bar he was behind. No idea what that meant, but he didn't do it before.

Well, anyway visitor B was done with it. He now knew what he had been missing on the Sabre DAC. Emotion.
I too was confronted with that again (a kind of a hard way to do it), and in the end this too is just the same as what I said before : the ESS Sabre sings like a dead bird. It just is so. No matter all the detail, too many things are - or go wrong at heavy oversampling.


Since everybody now was able to watch out for differences and where they could be, we found many more tracks, particular in the classical regions, where things just did not work at all. I can tell (we all could) that the recordings were the best existing, but on the Sabre violins easily became a flute. Yes, I'm serious. A flute.

Visitor A came to the conclusion that something must be wrong with his DAC. He did not recognize these "errors" from his home. But what visitor A probably forgets is that
a. I asked him before about what instrument he heard through is DAC, with the just innocent answer "don't know", but with the flair of "who actually cares, as long as I like it !"
b. He did not have the reference before.
Besides that I just recognized the Sabre sound.

A little less relevant for the whole exhercition, was that I let the two visitors decide in what mode the NOS1 sounded best to their ears. Or actually is was visitor B wanting to find out, with the notion that visitor B is not NOS minded at all. He knew how I think about that, but never found it relevant for his choice of a DAC. Also visitor B has good theories (confirmed by, say, the books) that oversampling to a certain extend is good.
The unanimous result was NOS/Filterless. No matter I told filterless is wrong and measures wrong, no matter I told I played with filter for the last couple of weeks, filterless came out of it. And indeed, if your measure "the right" things, that may come out of measuring just the same.

As an anecdote, I come near the end with this :

Visitor A, who told that he had some nice hours of good sleep (:)) woke up with the solution;
His DAC received a 50 Ohm internal SPDIF cable at some last mod, and this really should be 75 Ohms. So, let's change that cable !
We did, and it did not make a difference. Then of course what visitor A did not know, is that I myself have an internal 50 Ohms cable. :biglol:
Indeed wrong of course, but to my ears that 25 or so cm did not make an audible difference with a 75 Ohms coax, and at this moment I don't own a 75 Ohm very flexible cable which really needs to be flexible at pulling off the cover and the PCB where the SPDIF is connected to is mounted on the cover.
Anyway, I told visitor A to forget about the reference he now has, and most probably back at home things will be allright again. His fine tube amp may mask the anomalies my amps reveil, and I really hope this is just so (for him).

On a last note, and a kind of relevant to not only myself, but also to you listening to my brabbling about "my fine system with the best sound in the world and blahblahblah", I really worship the ears and the paying of attention to everything of vistor B. Not that vistor A is less on this, but visitor B is the critical complaining and "nagging" user. Only the best will do for him;
He told me that in my system he heard one of the best systems he ever heard, but not "the" best. Well, that is enough compliment for me, and I must say that I wasn't all that confident in advance. Remember, I know visitor B as searching for a DAC and all for many years, which may not exactly be avarage. If you are so much into things, you know you merits, and for me this was a kind of scary.

Dear visitors, thank you for a great experience on my side and lending your ears. I'm sure at some stage you will be reading this, and maybe you want to add something, or disagree with things I just said. Don't hesitate to jump in in that case, or do not if you just don't like to  write something here. A small hint for visitor B might be to create another nickname just because of this. In the case you don't want to show who you really are on the internet. Do as you please.

Thanks you both,
Peter

As someone may have guessed, I am visitor B.

First of all, I want to thank again Peter for the hospitality and for giving me the chance to do an important A/B comparison of DACs.

If "cost no object", I would have already shed a few dozens of thousands euros to get a system alike my reference. Instead my goal is to get a reference-grade sound at a fraction of that cost. No bullsh*t. I know that it can be done. But I had to enter the diy world.

Everybody has its own parameters, so I will never say that what I like has to be agreed by everyone, though it makes me happy when is the case. My friend, visitor "A", happens to agree with my ears most of the times. I was glad to hear things similarly from Peter, whose ears -I guessed and I know now- can be trusted.
I'm not posting my comments online anywhere but here, although I will express all my thoughts privately to people worthy of my esteem.
I dont see the reason to bash the ESS Sabre, especially in the DIY community, and I have no commercial interest whatsoever, unlike Peter, (whom i believe to be motivated by the reach of good sound, rather than money) :) .

My approach in the evaluation stage is to listen just anything, without knowing what's inside and which is the pricetag. This helps avoiding the placebo effect (to which I think to be quite resistant). So, when I auditioned the system which then became my reference, I didnt know exactly what parts were in the audio gear.

I think that more or less, the evaluation stage is finished for me, at least regarding the source. I believe to have made the right choice about the speakers and the amplification, but it is too soon to tell.
I will also not comment in detail about my impression of Peter system, I can confirm what he already wrote here, that is, his system is "one of the best that I have ever heard, but not the best". It has one special merit, it is utterly transparent, and i expected that considering the speakers alone, and I wasnt disappointed. This is very helpful in doing a comparison of a single component.
I haven't auditioned hundreds of systems, only a few dozens, but I think to have acquired a decent knowledge.

During the past auditions, I have found to like, more or less, sources based on the pcm1704 converters. I wrongly assumed that my reference cd player employed the older 16-bit tda1541a DACs, and that's what I told Peter. But I was wrong. I did a quick search yesterday and found out that it does instead use 4x 1704, in NOS mode.
The family sound of these DACs is characterized by realistic timbre of instruments, PRAT, and lack of high-frequency distortions*.
Being multibit mono chips, they require an extreme care in the design and implementation of ALL other circuits around. This is not often found in commercial products. A single opamp in the output stage would compromise irreparably the purity of the sound, while an undersized psu would make the bass muddy.

The NOS1**  built by Peter comes quite close to my reference. His system as a whole, though, failed to give me complete emotional involvement. I felt like i was in front of a concert hall, looking through a window, instead of being inside. Still, it was pleasant to listen and I wanted to enjoy the music of my favourite discs for longer.

---
*   for lack of a better word. In worse systems this is the mark of a digital sound, metallic, unpleasant and that cannot be stand for long. In better systems, like it was in the case of the Sabre in Peter chain, it is just some annoyance with high tones and desire to tune the volume down. This is the effect that my ears receive typically when there is a sigma-delta DAC in the chain. I dont know why, but it is so.

** configured NOS and filterless, with spdif input provided by the fireface

Poor guy. He flew in with his DAC just to learn that it transformed violins into flutes.


LMC has a ESS Sabre based DAC, and he has reported that he is very happy with the sound. I doubt he would say so if it made violins into flutes! (Eventually, if so, I will tease him for the rest of my life. He-he). Are you sure that the DAC of "Visitor A" is not faulty? After all, it's a DIY kit. Probably many places where you can go wrong when putting the parts together.

I was about to order a ESS Sabre kit myself. But know I'll put it on hold. Do you have any release date for your Phasure NOS1 DAC?

------------

BTW: Previously I had a DIY DDDAC with 5 towers (60 pcs) of stacked Philips TDA1543 dac chips, without oversampling (NOS) and no filtering (Filterless). It was build without compromise, including the most expensive CERAFINE and BG capacitors, etc. It was truly an extremly dynamic and "live" sounding DAC. It made most other DACs sound like dead byrds. BUT it also turned violins into synthezisers. It was great on trio 60s-jazz and rock, but not resolved enough for massed strings or choral music. It homogenized crashing cymbals too. I don't know if these artifacts are due to the poor performance of the Philips 1543 (it's a cheap economical DAC chip from 1991...) (http://www.acec13.fr/tvc/Philips/documentation/TDA1543.pdf), or because of the heavy figures of folded-down IM distortion because of Filterless operation.

After reading your listening evaluation I am even more confused, because you say that your NOS/Filterless DAC is the real deal, while the ESS Sabre makes violins sound like flutes. -This is contrary to my experience.


PS: Fortunately, in the mean time, I am happy with the Eximus CD/DAC-10. The DAC inside is really a great product. So I have very good SQ while waiting for "the next big thing".

I dont know the ddac very well, but I dont think that it has nothing to do with OS/NOS.
It is more a matter of resolution and dac typology.

I can tell you that the NOS1 did not turn the Violins into synthesizers.

Hi pedal,


Not that you should take my expression on this with some grains of salt, but ...

But of course it is not so that any violin suddenly sounds like a flute. This is under certain conditions only, and you can well say that these conditions are the more difficult ones. I cannot explain these conditions, but can only refer to the ways a violin can be played, and hope you know what I mean. Just (try to) compare more or less resin on the stick (the horse hair), and the more resin is there, the more harmonics will be created. Why ? the stick (micro) stutters on the string. This might be a typicle example the violin may start to sound like a synth. When few resin is there, few stutters are there, and the viloin will sound more like a flute.

Neither is true in practice of course, but since a DAC may be profound in either way, the DAC may make it sound like that. The NOS DAC will impeed for the synth more, and the heavy OS DAC will impeed more for the flute.

All is exlained by my earlier pictures. The heacy OS DAC making a pure sine of pure squares, while the NOS DAC pertains the squares.

The latter is always the reality BUT, it now depends for 100% on how the DAC further deals with that;
I think I said it before, but the NOS DAC will repeat a square infinitly into the spectrum, meaning right over the Nyquist frequency, that causing aliasing NQ being the mirror, which is 22050 for 44100 sample frequency. So the squares bump back into the audible spectrum when nothing is done about it.

A lot more can be said about this (and some I talked about earlier in this topic), but the main point is and remains :
Harmonics are the results of square(ish) sounds, and they are as square as their origin, though lower in amplitude. When the original sound is turned into sines, the harmonics disappear, because a sine just does not have harmonics. Thus, a sound which is that sound because of harmonics, becomes a flute. A flute, because it just has no harmonics.


In between the above lines much more is to say and explain, but I think you get the grasp of it : a heavy oversampling DAC is just NOT ABLE to produce harmonics, if you only believe my earlier pictures. And they are not faked.
A violin is a "crackling" instrument, similar to a trumpet on that matter. That crackling is squares. Remove them, and both the trumpet and the violin start to sound like flutes.

I did not want my own ESS Sabre (Buffalo) to sound bad or anything, and the measuring I did afterwards. But it is just a fact that
a. inrtuments cannot be discerned for their origin, especially not in a larger orchestra;
b. how can they, if the harmonics creating them have been removed;
c. you need the reference to recognize it.

As I said earlier, maybe more in between the lines : nobody was complaining in advance of listening to the NOS1. I was there, they did not. That something was lacking (as described) is another matter (well, I think). But as soon as the reference was there, the ESS became, well, unlistenable. Not because I said so. Also, each and every time I noticed the nodding within two seconds after the start of a track, I mean it.

Lastly, to be serious on your remark an NOS DAC making a synth of a violin .. it should. It should just because of theories I briefly explained, and which IMO are not too difficult to understand. However, this is only about the part of aliasing (because of filterless) and that aliasing for 100% sure (because I know by meassuring) is only at very few places at an SPL that equals the listening level. So, this is 5 or 6 (by heard) sheer "points" in the audible band. Although this looks very wrong, in fact this is only about a few spikes like a speaker may (no, will !) let emerge here and there.

I put it my task to
a. measure properly;
c. interpret that towards how bad it is for actually hearing.

So far for me this is difficult, because I am not experienced on this, but on the other hand, actually nobody is, because official AES17 measurement measures the wrong things. Says who ? me. This is about THD+N of 0.0001% while a square became a sine.
I like to turn that into more truthful and meaningful figures. Just because it is as much audible as my theories tell.

Lastly, and I hope I am allowed to emphasize it this time, this time only music was compared which should just matter not all that much about OS/NOS. We did not compare trumpets, no cymbals, no sibilance reed instruments, no synthesizers, no high timbre male voices.
And *there* NOS really makes the difference.
Besides this, and this really matters too, we compared SPDIF only just to be honest in comparisons. I2S is just from another leage really ...
(but the ESS Sabre can theoretically do that too)

So far,
Peter

Ok...just some thoughts on an sunday afternoon here. Read what Telstar wrote, and although not heard your DACs, I kinda symphatize with some remarks: Several releases of XX have had too much energy in the treble and upper mid. MB this is because Peter has tweaked it that way to suit his own PB chain and speakers, I don't know.

But these are characteristics which also has come forward in my system, which is very different fra Peters.

Also some releases are very 2D and you get the feeling of looking thorugh a window as Telstar mentioned. But then again, some versions are VERY 3D.

The problem is that one versions good SQ is often ruined in the next version due to implementing more formats, changes etc. that has nothing to do with SQ in themselves.

Looking forward to hearing some of these DACs before long.

Pwew, difficult ...
Ok, some brief (objective I hope) remarks :

1. This should hardly be about my system, but then personal likings, if any. My high output is 16dB uplevelled already opposed to how it officially should be.
2. I don't recall to (wanting to) head into those directions by means of the software. By accident it does though, sometimes.
3. It may be useless to refer to my playback chain from what we heard those 15 hours, because that was SPDIF while the virtues of the NOS1 are with I2S.

The differences between SPDIF and I2S are rather large, but (or *thus*) SPDIF was used for both DACS to compare apples with apples better.

Important or not, I just copied the remarks of the visitors without any colouring of myself, and might the statement have been "the worst system" or "the best system" either wouldn't have said much because "the system" is so much created by the software already, let alone the DAC. But of course the rest of the chain does have its influence, and in the case of the visitors it may be good to say that both were explicit dipole lovers, which is quite some opposite from horns. :yes:

In the very end it is dangerous - or not a good thing anyway - to explain (XX) things through a system, no matter how truthful and logical that would be. We never did because we never felt the need to, and that is very very useful because we all could discuss about the same thing. However, I think that a. we all have become much more experienced on listening (I sure have in the past 2 years) and b. we may be reaching limits where the rest of the PB chain just can't be avoided anymore. Therefore, let me add to this "pwew" response my latest experience :

Referring to the post from today from SeVeReD (Re: Q experiment anyone ? (http://www.phasure.com/index.php?topic=723.msg5387#msg5387)) he talks (including earlier posts in that topic) about "a wall of sound". Apparently he can encourage for this by means of Invert On;
Then, this week I visited the designer of these horn speakers, and by means of a small tweak in the filters (about eliminating a dip in the 120 Hz region) he created a massive wall of sound, which I "stated" to be way too much a wall of sound. Voiced and instruments became to big of it (imagine a voice to be over a meter in diameter).
Lastly, by pure coincidence (that it happened right tonight) I was able to play the NOS1 how it should be and how I want(ed) it : completely NOS withpout filter over I2S. Together with Invert now *that* creates a wall of sound in my room and PB system which I could not with this DAC so far (and the software versions since ??).

Now if you watch above alinea closely you see that "wall of sound" (which may be very deterministic for "a system") was created 3 times in 3 very different ways, the 2nd one actually changing "the system" really. The first is just a software setting, the third a change of the DAC including a software setting.
I can't speak for SeVeRed really of course, but if I didn't know otherwise, the other two means of changes would make me swear I was listening to complete different systems. And as you can imagine, more laid back opposed to a wall of sound just *is* completely different.

Btw, the most characteristic of the NOS1 operating really NOS (and filterless) I2S connected, is the unrivaled dynamics going together with the sweetness of I2S. SPDIF connected the dynamics should be as high, but as how it comes to me, is "smeared" again because of the roughness (opposed to sweetness) of the SPDIF connection. So, these dynamics come to you as very very clean. The fun is, this would lead to sterile without further changes, but including the wall of sound andthe warmth added because of that, just makes it a (3D 3D 3D) mixture I LOVE.
With normal sized voices. Haha.


I hear you very, very well;
As far as I can tell, those reasons for changes are behind us, because it just all has been done. At this moment I can only hope that the current version is good for SQ for everybody, and if not, the Q2-Q5 can do something to it. I am sure (!). The problem of course is how.

Right, having said this all, I think I only now see how to interpret this one :


... which makes all I just said unnecessary.:fool:  After proper interpretation (I think) I said the same as you wanted to express. I never like to scratch carefully typed words (can I ? hehe), but what it comes down to is that earlier versions sounded better, right ?
Hmm ...

Earlier versions was not uniformly better than todays, no, IMO. The right description would be more like a rollercoaster ride with incredible ups and downs...

There has been gold nuggets all around. But there's been a lot of cr*p too... :)

(Beeing truthfully here, hope you don't mind, P!)

What I really would like to see in XX is support for either a simple EQ with some 16-20 bands, or possibilities for linking to e.g. Audiolense etc.

Then each of our (very different) choice of TASTE in SQ reproduction will be taken care of in a good way. I for one would be very happy with an standard EQ.

Sry about the distraction somewhat from the disc. about the NOS and other DACs but the issue kind of cropped up earlier in this thread...

But don't typical EQs mess with the phase in a way that just destroys... you'll be chasing your tail forever? Not saying "don't do it Peter", but ... maybe there's a reason he hasn't done it.  EQs and xovers settings are harder to implement (and maybe impossible somewhat) than we all may think.  I agree it's been an up and down ride, but I think we've been heading in the right direction...(says me with being very happy right now and too scared to move to X version hehe).  What worries me is when Peter ups several changes quickly and I dl them without evaluating them against a good old player... I think that happened with later V versions and early W versions and I got lost ... There was some good things and some bad things that happened.... made it confusing.

The "simple EQ" cant do a good job in most cases (including mine).
What would be really good, is to interface (i.e. use a filter) with DRC:
http://drc-fir.sourceforge.net/

It's open source and on total different level than paremetric eq, or even more sophisticate piece of hardware such as Lyngdorf and DEQX.

I know that Peter is working on some filters, so this may be the right moment to look over room correction.

If you want EQ and/or roomcorrection there is one amazing program to buy: AUDIOLENSE XO (http://www.juicehifi.com/index.html). It even includes digital XO functions if you want to, for as many channels you like at whatever samplingfrequenzy. At €360.- it's a bargin.

I use it as a EQ/DRC in 2-chanel mode with final Group Delay Correction to minimize any phase errors occuring after the FIR filters. After measurements the program creats a correction file, which I enter into the playback chain, through Convolver plug-in running in a digital loop inside the RME FFMixer software. Life will never be the same.

SOMETHING ELSE:
Peter, -does your new DAC prototype have a capacitor on the output, or is it DC-coupled?

Why should I buy Audiolense (which is good btw), when DRC is free? :)
BTW can audiolense create filters for each channel separately? ;)

PS: I dont have the Fireface and if things go as they should, I'll never have it.

what´s wrong with flutes? :)
in my system which pedal know quite well the , the sabre dac fed through I2S bus is the best so far
it equals the 5 tower dddac in all over AND bass slam
it surpasses it in resolution(not difficult)
with resolution and low noise-floor comes 3d and the rest by itself imo
so there´s no problem following complex passages of massed instruments and voices e.g. G.F.Haendel Messiah( my desert island favourit)
and this is with the stock IVY buffer version and the std psu´s
(will at a later stage try to build a tube based buffer ala  lampizator, but my tube psu´s tend to be big.......my riaa psu is 48kg for 4 x E810F!))

when it comes to XX I agree that close to perfect sounding versions are sometimes ruined in the persuit of  tweaking of features
agree that som prior versions sounds as if they were tuned to a specific need/purpose(soundwise)
( maybe the treshold for releasing new (improved) versions should go through a beta panel before "final" release)
so e.g let 0.9w-9b stay as the official latest working version till the next level is quality-proven.

maybe remove versions older than ????      there is something called too many options   (also on the different settings)
maybe there should be a working reference with fixed settings(at least some)
e.g 9b with 4/17/17/0/0   
easier to move on fw. when the new (intermediate) reference is set.......no more turning back except for P to locate weird faults
then the beta users can communicate on a select and"closed" part of the forum in the pursuit of REAL PROGRESS
this would also save a lot of ranting in the forum time for Peter

ALL this is of course IMO
ANYWAY THXS 4 the effort Peter,but make it easier on yourself
best
Leif

Why did you pay for XXHighEnd when Foobar is free? 8) 
-The point is that digital EQ with lots of FIR filters generate phase errors. They sing like a dead bird, to quote Mr P. So you need a program which can perform a Group Delay Correction in the final stage, before D/A conversion, in order to get (most of) the transient sharpness and dynamics back into the music. But of course, if you wanna live a dull life for free, it's your choice. He-he. (No offense!)

Yes. For as many channels you like.

Some players, like Foobar and WMP, accept plug-ins. Then you don't need aditional soundcard software with digital loop facilities. Hopefully XXhighend will have a possibility of plug-ins too in the near future?

Because XXHE sounds better?
AFAIK XXHE is the only WASAPI player available. WASAPI is a much more solid api than ASIO, it's less dependent on the drivers produced by the manifacturers.


First of all, did you try DRC? AFAIK it creates only one filter (per channel), and the filters can be 24/192.
I assume that you must create at least ONE filter to apply any kind of digital correction.

AFAIK, the features of DRC are superior to any other software, but I could be wrong, so if I am, correct me.
It's not like I dont want to spend the 300€, but I believe(d) that besides graphic representation and ease of use, the functionalities of DRC surpass the ones of Audiolense.

I hope so, but at first I hope the possibility to use digital filters.

No capacitors on any of the output possibilities, and no capacitors anywhere in the signal path.
Who'd want those ? hehe

There will be filters in XX. The first will be for applying proper filtering for NOS DACs that work without - or can switch off filtering.
The mechanism of applying the filter (of any kind) will be general and applied onto the music file. Thus, not real time, compared to FLAC decoding.
Any filter available for such an application can be applied then. In a later stage more output channels will be supported, so XOvers will work as well.

The first one is the most important, because it will give NOS DAC users the opportunity to use a filter afterall, that filter not compromising anything to music (output) data ... assuming that I know what I am doing, can mesasure all I want/need and ... it will be dynamic. It will be dynamic/adaptive  to the various input rates, knowing that 44.1 needs a different filter than 96 etc.
When this works out as I intend, it will be a most good reason why XXHighEnd will sound better in NOS circumstances. But of course this is a kind of DSPing, although one of a special kind, and with the best purpose imagineable *and* it will just replace something a DAC normaly does - or should do.
Of course my NOS1 is the "perfect" example to test with, because it just has the options in hardware. If you'd only see that
a. filterless sounds best
b. what a complete mess measuring shows in that case
you'd know how much better it can be theoretically. Of course, unless this mess is what creates the sound kind of explicitly ... then the mess should just stay.

Please note the difference with something like equalizing. I am not in favour of that, and will "allow" it by the grace of the means to apply it (because : any filter can be applied, as explained before). So the only thing which I like about it is that it can be applied without loosing on SQ just because of applying it (mind you, like decoding FLAC is audible when done in real time).

Peter

Hey Leif, thanks for your input. A few remarks which I think deserve some attention;


Yes, this is just equal to my first "review" of my Buffalo, and in the end is about my remark how very much more difficult it is to let sound an NOS good in such circumstances. With the Buffalo one can immediately feel with how much ease such passages are interpreted.


In itself this is true of course. The latter though is not true. Each and every explicit SQ change emerged from
a. some theory of mine *not* knowing the result in advance
or
b. undoing a degration and hoping that some counteracting means would help.


Haha, but maybe you forgot that XX just *is* in beta ... all the time so far.
Ranting (if any ... I don't think so) is for me about remarks steering me in the right direction again. If remarks/complaints were not there, it can only be me myself discovering anomalies, and who knows I'd never hear them.
Please keep in mind that I hardly have time for listening myself, and that relative to that already one of you may have 10 times more listening opportunities. IOW, don't underestimate what a huge amount of time it takes to find something like "No track given", while the whole stupid thing emerges at, say, 16 minutes of testing. It could take 100 times of testing these stupid 16 minutes, which I rather do without sound after the 25th time. :heat:
In the end you could say that I abuse you all, but the more there are, the earlier someone finds this anomaly (which as we know often is dedicated to a few only). In the end it is for the good cause of course. That it takes a few years more than I expected in the beginning, is only because at some stage I thought of the Unattened avoiding the GUI interference, with the example of right now : remote control. It is a kind of stupid of course to remotely control a player which is not there. But okay, with some additional months it will work (the best is yest to come).


And Leif, might you be sailing the North Sea some time ... you're welcome. :yes:
Peter

where do you live? I have driven to Nunspeet to Bert a few years ago. after the Kiel ferry it isn´t that far.
reg. all the beta versions:
I´ve just learned slowly to resist trying all the new versions all the time;
out of experience I´ve learned the hard way that I often get into all sorts of hopeless "unsolveable" trouble when installing a more up to date version
I simply don´t have time to test them all
and now I´m very happy with 9b
best
Leif

20 Km closer from the Kiel direction ...

ok then that´s just a nice days drive in the countryside ;)
and since you have a speaker close to what I planned building it could be fun
hmmmmm
best
Leif

Equalization or "phase delay" as DRC do are needed in some rooms like mine. I do understand why you do NOT need equalization in your room, but in most other cases it will help. Those 16db (or 13?) you added to the HF could have been done with a digital filter. Think of that if you DIDN'T have a crossover. :)

I'm not sure how you are planning to apply the filters, are we still talking of not touching the (thousands of) music files, right? :)
What do you mean by not being real-time? Do you mean some preload in memory or a way to hardcode in the program?

Anyway, our goal is the best SQ, so I'm pretty sure that you will find the proper way.

You are definitely right Telstar. But somehow I don't feel it like that. This is part of the speaker and an explicit part of the xover in there. This is mechanical stuff and about resonances and all. It belongs to the speaker.
Of course a digital crossover could do that job (I don't say it is easy, and I don't talk about which is better), but it still should belong to the speaker.

Now we have a problem, because the PC or softwareplayer if you like, does not belong to the speaker, and e.g. the output of my SAT box would not use the xover.
The way pedal does it, is coincidentally a good one. The filter (whavetever) is software wise added as a kind of plugin to the soundcard, and e.g. my SAT box could go through the soundcard just the same, picking up the filter underway.

Add to this all, that when one would apply a filter in a box outside the PC it really should be one with digital input and digital output. And as when done in software or "in" the soundcard, this needs an amplifier per speaker driver. I only want to say, having a box with analogue in and outs but which is digital, is just another DAC (including ADC) and this is the very very last one (being serious about DACS) would want.

Of course I am bringing up all kind of subjects here, but they are all related, and it is not so easy to do it right ir conveniently right.


No, this would be DSP (I said that I think). But DSP for another cause and replacing what a DAC otherwise is doing when talking about the filter only (and not about xovers and all).
But regading the last part of your question here, maybe you got confused by a. it not happening in real time, and b. you not needing to convert all of your files;

With real time I mean per window part which would be the smallest part needed when filters are applied (they need to read ahead to see what will happen in order to apply the filter, and that can be named a window).
With "not real time" I mean the stage of preconverting the file which XXHighEnd is just doing before handing the file to XXEngine3 (or to itself for Engine#1 and #2).

So your precious physical files remain untouched, but a copy of them will contain the filter. Keep in mind, this would be the "everything is possible" part, because it just needs an existing converter (SOX could be one) that allows for batch processing;
The "not everything is possible part" would work in interal memory, which is always available as an option because the file is present there too (the memory player is in luck this time :)).

Silence Before The Storm,

Must be a Hell of a Post coming up!
 :blob8: :blob8:

Does it work for you ? Peter

I have been typing for quite an hour tonight allright. But I could not finish it.
By now typing starts to be too difficult. :party:
:)

Later ...

and???

Anxiously awaiting news from Mr P. :bored:
There must be some which is hiding...

Ava, LOLOLOLOLOL ... I don't know whether you got that smiley application from somewhere else or whether you made it up, but I so far never saw it anywhere (but maybe I'm not hanging around all that much elsewhere).

So simple ...
Don't look strange if we are going to see that more in here !

Anyway, I got up this morning just to write some piece at last ...
So I will be doing that right now. It will be one of my longer posts I'm afraid ... :swoon:

Hope it's positief....  :) :xx: :soundsgood:

Please don't chivvy yourself.
We all know that that kind of work takes time, time we all want you to invest.
We're just anxiously waiting for things you might come up with.
And if I may say that, there are very few things that are not worth waiting for that you allways come up with!

So DON'T chivvy yourself!

Oh and news about something like the PNOS1 is allways interesting to follow!

Greetz Ava

PS: this running from that "secret" smiley is an official "add". You can see it when you're writing something. There's a moving "M" with an arrow. You just have to mark the things you want to let move, and click it. There you go^^

Well, it's been a long time since anything really was heard from me, but that actually meant much much time has been spent on it, and from one improvement over the other, any story would have been obsolete before it got posted.
This one too will be obsolete within two weeks, but I didn't want to keep you wait longer, and the phase things are in now is a separate phase anyway (not that the phase itself can be described ... it's juat a "quality" phase, independent from the other stuff).

First, roughly what has been going on lately ...

As some will know I bought the analyser equipment (5 months ago), which I bought for software measuring purposes. However, at that time I felt some better justification for buying at designing and building this DAC, but since the DAC was a kind of coincidence (although planned for some years), I didn't actually use the analyser at that time. That is, not for *creating* the DAC. It was already there in its stage you have seen from the earlier posts in this topic.

In the mean time (and along the way of listening to the DAC, finding its best settings, etc.) the analyser did not bring me what I was hoping for, and urged by the wanting to know what actually happens when software makes things sound different (you know, all being in the "bit perfect domain"), around the time of the last XXHighEnd upgrade I started to write my own analysis software. And although that by itself was planned right from the start of XXHighEnd, I was encouraged by Jeffc who came up with blu ray ripped CDs vs. normal DVDRom etc. drives, and the first sounding better, all files containing the same bits and bytes.
Although I could make something of that rather quickly, it took an inmense amount of time to get it workable for you (because I like to have it within XX), and right at this moment it is not finished. It is not finished because of the results I obtained from it, told me how bad a DAC actually performs ...

From one thing came the other, and where my own software told me "a" behavior of the DAC, I started to use the analyser to look into things better, and right now I couldn't have made a better investment than buying that apparatus.
So, I did the most obvious when an engineer (which I am actually not) wants to know what is going on and ... started to improve.

In between the lines other developments around the DAC were and are on-going, mostly related to seeing better my own wishes after it was playing, and in close co-working with the engineers who help me with this in the first place. And not to forget : I started this DAC project for my own self but got enthusiast by the so many who wanted to buy it, some in DIY mode, for price reasons or other.

Although it may be hard for you to follow, but at the start of my own analysis software, suddenly all came together. I'll try to sum up a few :

a. The engineer, now having designed a four later board, really found that filtering -if to be there at all- should be in software;
b. My measurement of the "old" DAC showed that things could be better in some theoretical areas;
c. The whole design of the "old" DAC was too large. It looks impressive allright, but it would impeed for a steep price;
d. The general idea growing was : speed. Speed speed speed, which means throw out components (also see a);
e. From the original design I learned that jitter is not *the* aspect it is all about, never mind we like to think so;
f. I found that so many options should be in the DAC (not expressed so far), that the price would go up and up;
g. While the latter may be no problem for many, I really want just everyone to enjoy this DAC.

And thus ... while the new DAC board finished yesterday including some heavy programming by the engineer concerned (oh yes, a DAC runs on software too), I finished my own project on getting the whole into a minimalistic design with enough headroom (price) for all I want to have additionally in there, but which turns out to measure as ... well, an oversampling DAC. Not that this went by itself, because it took hundreds of hours of investigating, trying, testing and listening.
In the mean time I can tell you that theoretically the DAC can be full of switches for different sound (all legitimate, because about all I would have been writing screaming stories if I would have had the time), but right now I'll stick to "one sound", which is the one I finished yesterday.

In between the lines, and for understanding : what I have been doing myself is outside the DAC board, and can be considered the analogue stage. So, what I will be writing about below is about that, and the "old" DAC board. Why not the new one ? well, because that will again need all the trimming, and knowing myself it will take another few weeks before I will be able to report about that. But the fun is : it will be better again (not that I can imagine that at this moment).

Right. So now you understand the environmental conditions the next is about, I must emphasize on the importance of what actually happened, which is why the next step (the new DAC board) will be as important for better sound. In the end I mean : this is not theory only, but it is about measurements proving that what you see for better measurements, sounds better indeed. In fact it is *the most* important, because it was never understood why an NOS DAC could sound better while it measured so much worse.
Of course, I have my own explanations to that, which is about the squares vs. the sines and the destroyment of both (the first by OS the second by NOS). However, the net results could be resoned only, and no science could ever show the absolute right. Now, this will all be obsolete when the NOS DAC measures nearly the same as an OS DAC, right ?

Because I want to be longwinded today, I can add that harmonic distortion theoretically is all over in the NOS DAC, knowing that squares (or squareish sounds) create harmonics, *assuming* that the squares we talk about were sines originally. However, nobody knows, looking at the CD data, when a square which is there, was original or not. No matter when it was a sine, it can be a square in digital only, so whether (e.g.) a synthesizer put out a pure square at 22050Hz, or whether it was a pure sine at 22050Hz, in digital it looks exactly the same : a square. Now, when an analyser puts in a sine, and it detects harmonics (2nd, 3rd, 4th, etc.), it tells you : no good ! And the higher those harmonics are, the more no good it is. Now, while this is something we have to live with, during the process I found quite some other parameters influencing these "harmonics" ...

Allright, because I really feel longwinded today, and the chance is fair that you start to fall alseep by now, let's look at a picture from another topic in here, which was about me being bothered by the 17.7KHz frequency a modulating pump throws on my mains and which I could capture my microphone through the loudspeakers :

(http://www.stordiau.nl/XXHighEnd\Pump02.png)

Already earlier I found that adding noise to the electrical system, would make a peak (and its audible sound !!) like this go higher. Hmm ...
Mind you, this is not an analyser fault, and the picture you see is just from a microphone who captured this sound in front of the speaker, visualized by as program that can show it in real time.

With the pure coincidence of this knowledge (noise let rise such anomalies), at working at the analogue stage of the DAC I could see this stupid thing grow higher or go down, the harmonic distortion of the DAC going along with it. From that point on I knew this was caused by noise, or noisy connections if you want.

In the mean time, also coincidence, I received FFT pictures from the engineer from the measurement of the new DAC board, and THD+N (Total Harmonic Distortion + Noise) which showed such low distortion from the DAC itself, that I got urged to get there too, but now for the whole chain.
Yeah, if you don't know any better, why not be foolish and try it !

Allright, at first I had 60 screenschots to show everything at the various frequencies, but I thought that is really too much, so I'll stick to six for now. All will be better at the definitive version anyway, so more pictures then.

The first of the three pictures below show (all left channel only)
- my old NOS DAC which is considered to be a good one; no oversampling, no filter (orange).
- Phasure NOS1; no oversampling, no filter (red);
- Phasure NOS1; upsampled 4x (which is different from oversampling !), no filter (also red).


10000Hz with low pass filter :

(http://www.stordiau.nl/XXHighEnd\Good NOS FFT NOS 10000Hz.png)
(http://www.stordiau.nl/XXHighEnd\Phasure NOS1 FFT NOS 10000Hz.png)
(http://www.stordiau.nl/XXHighEnd\Phasure NOS1 FFT 10000Hz.png)

The numbers show the Total Harmonic Distortion relative to the amplitude of the base tone (which is a 10000Hz sine here);
Note that the amplitude of the base tone for official test measurement should be at -60dBFS, but that squeezes the picture vertically and things will be harder to see. Also the amplitudes can't start at -0dBFS because the output voltage of both DACs is not normalized (which would be 2VRMS), hence it is less. For both DACs this does not matter for the figures.

You can clearly see that the "Good" NOS DAC is much more wild at the bottom side. Note that the THD+N(oise) figures don't show an avarage, but in fact the worst peak anywhere occurring relative to the frequency band measured. Because the figures shown only take into account the normal audio band, this will be those first two peaks to the right of the 10000Hz original tone, topping -120dB, the original tone being at -40dB. So, the distortion is at -80dB relative to the original tone.
Note that these are aliasing residues (the mirror being at 22050 Hz), and not harmonics.

If you compare the first picture with the second, you can notice some 10dB of difference between the general noise floors in both pictures.
Of course it is very clear that the first picture looks much worse, and never mind the distortion figure (remember, which is about the peaks), much more harmonic distortion is present in the first picture.

On to the last picture; this shows what happens when the DAC upsamples the original data 4 times (so, from 44.1 to 176.4 in this case), and that this prevents the aliasing, or in other words the distortion at the left side of the Nyquist mirror. However, because aliasing also takes place at the right side of the mirror, there too all the sh*t has vanished.


To get the grasp of another kind of leage, below are the same pictures again (but it were separate takes), but now the distortion figure reflects more than the audio band (in this case, up to 96KHz). Aha, those figures really show different now :


10000Hz no low pass filter :

(http://www.stordiau.nl/XXHighEnd\Good NOS FFT NOS NoLowPass 10000Hz.png)
(http://www.stordiau.nl/XXHighEnd\Phasure NOS1 FFT NOS NoLowPass 10000Hz.png)
(http://www.stordiau.nl/XXHighEnd\Phasure NOS1 FFT NoLowPass 10000Hz.png)


Now, while keeping in mind that the "Good DAC" really is a good DAC, and not any worst example to, say, proove the NOS1 can do better, you can see that the difference emerges by the upsampling. Ehh, but wait a minute, wasn't the NOS about "non oversampling" ? True, true ...

The whole point is, that oversampling, like any sigma-delta DAC would do (and then I mean times 64 to times 256) is killing the squares, while 4 times upsampling shifts away the Nyquist mirror, removing the distortion in the audible band *and* outside it, the latter being bad for your amplifier, which might not be able to cope with it. And now it comes : where the OS DAC can't do without the heavy oversampling because otherwise it can't operate, the NOS DAC can upsample to any rate we think is necessary if it can do that in the first place. And this is where 24/192 comes in as an important phenomenon, because the Good DAC just can't do that. So, now you also know why I did not show any picture of the Good DAC without all that distortion, because it just can't do the upsampling ... (mind you, some DAC chips can, but they are not 24 bits).

In the end, while this is achieved as a basis feature, you can see (at understanding all a little) that no filter is needed in order to get those nasty fake harmonics down, because they are way shifted out of the audible band, and even up to a region where your amplifier may not be bothered by it anymore (which is above 176.4KHz in this case).

Going further again, we are thus comparing a "nice and easy" DAC (which can do 18/96 in this case) with a crazily more complicated 24/192 DAC for which it is not so easy at all to achieve the same good figures. That is, the DAC chip itself can do it allright, but the analogue stage behind it must be able to do too, just like your amplifier must be able to cope with higher frequencies in order to perceive them right. And it is this where the tweaks were applied ... wrong ... where they *had* to be applied, in order to achieve a same sound quality a good NOS DAC with lower sample rate and bit depth, well, can't to begin with. So, now we are confused ...

The confusement starts by looking at the OS DAC chip, which can reach so called very good THD+N levels, which
a. are true when sines are assumed to get in as the only kind of wave
b. are not true for reality because the analogue stage is needed just the same, but since everything which is squareish in the first place is destroyed within the DAC itself, we will not be able to measure the net results at the end of the analogue stage of the DAC.

Also to keep in mind : any squareish wave requires more speed from the analogue stage because a square is "transient" by nature, while a sine is, well, a nice soft and easy thing.

Now, on to the real merits, what you saw in brief in the pictures above, for both DACs it includes the analogue stage including 2 meters of interlink cable, while for the NOS1 also an analogue volume was included in those measurements (:teasing:).
For the "Good DAC" this is not sufficient to drive a main amp at that cable length, while for the NOS1 it is. This means that when it were for measurements and figures only, this all looks good (this can't be shown by these few pictures only, but trust me for now that the other 60 or so pictures proove that, and keep in mind that it will get better only with the new DAC board).

Last on this subject, and actually the most important, is that where the tweaks by itself were necessary to obtain good measurement results (which I think are important), the reason for the tweaks has been the speed of the analogue section and the means I got there. These means can well be called "illegal" for an engineering point of view, only meaning that no engineer would ever have tried what I did to begin with, and while the job has been tough, together with measurements it got me exactly where I wanted. So, those figures mean much more than only good figures : they tell that a not understood means of arranging an anologue stage with enough drive to directly drive a main amp, just works.

What the pictures can't show is how it sounds.

For me, this is is relative to how it sounded before the tweaking, and well, when you hear this for the first time, you won't believe what you are hearing.

The first thing is, that this is completely fluid. Second is that this is completely quiet (call that blackness). Third is that it is in a fashion of balance and easy only an OS DAC can show. Fourth is that the level of detail is unheard. Let the fifth be that this sucks and sucks and sucks you into the music.

The fourth, the level of detail, is the most interesting, and in the end the most important I think. The point here is, it is not difficult at all to "create" detail, but usually this occurs in the higher frequencies, and most often it appears to be fake (harmonics !) afterwards. Mind you, not always, because when I wrote the story like this the first time (never posted), it was as good IMO, but lacked the overall balance, and in the end the all over smashing cymbals did not allow for background music. The most interesting though, and since I think it was as good, it might well become a switch in the DAC (very easy to setup different "sound" arrangements). However ... the version of which I think is final (the one I am listening to right now) has another feature : unveil details at the lower frequencies just because the smashing higher frequencies are *not* there. Things like background voices never heard before, singing saws, wowing guitars while no guitar ever was there before, and in the end you could say that the quitness allows for hearing this all. Keep in mind though : creating quietness also is easy : just get out all the detail and you are *there* too. But hey ...

You must be warned a bit; for me it is rather easy to hear that once you stuff in your favorite preamp, all detail will be lost. Yes, so super-fine all is now. I mean, there is no way anymore that a preamp could make your sound less harsh. It just isn't harsh to begin with, so everything flattening will destroy the detail. But then the DAC can drive your main amp, and it contains a volume control ... (of a means I'm sure it's not applied so far :)). And no, this isn't supposed anything like a preamp as you might expect, so don't ask for analogue inputs.

One last thing for now;
There won't be anything left for DIY I think. All is SMD stuff (this is about resistors and all of 2mm wide) but also there won't be much price reason left to want this. Although at this moment I can't tell what prices will be, it will be quite affordable I'm sure. Besides, because everything will be about the most special trimming, grounding, wireing and all, destroying sound instead of bettering is the most likely. Different options for more or less valid sound quality will be in there as far as they don't destroy general SQ to begin with, and filtering options will be by means of (player) software, and uploadable to the DAC in a later stage. This latter is a subject by itself, and will be an option to apply to your choice. Later more about that.

Within the next two weeks I hope to tell more about "the DAC" as a whole, and about how the NOS1 from then on will set a reference for music playback through loudspeakers for once and for all. At least that is where my intentions are.
Right now I am the most satisfied as it is. However, I anticipate on you wanting more.

Peter (who didn't check for typos nor inconsistencies because of changing this little story a couple of times, and which really took him a whole day to write this, and of which he hopes it is a little informative afterall)

Ok first reply this night, more tomorrow.

"...little informative..." ... you're kidding me right!?

 :goodjob: still have to :15a: this over :sleeping: .

Still impresses me, which endless stories you can write, which strangely won't make me stop reading, although(and thank God) they're greatly detailed!

Greetz Ava

Thanks Peter. This is facinating stuff!


So, my understanding is that XXHE and the hardware will work together seamlessly. But will the DAC be able to work with other software players and/or digital inputs?

Mani.

This sounds a bit like the Q1=-4 setting at the moment!



Peter, I would have loved to have seen the following included in your analysis:

1)  a delt-sigma "24"/192 DAC
2) a true 24/192 Multibit DAC... with oversampling switched in but without upsampling

My (limited) understanding is that there should be no difference between 2) and an upsampled NOS DAC.(If you'd like to borrow my D70, I'm sure we could arrange that. Is this a true 24 bit DAC?)

Mani.

Well if it's all SMD and it will be quite affordable, then  :dancing: less work, more time to listen^^.

I'll be happily waiting for more news.

Ava

The NOS1 is a 24/192* multibit dac.

He has a sabre buffalo so if he got the time he can do the measurements for it.

[*actually accepting higher than 192k]

OK Telstar, that's cool.

But I'm still just a little confused. Peter, are you simply proving that if you upsample, you don't need to oversample... and if you oversample, you don't need to upsample???

This fits with my (limited) understanding, because oversampling and upsampling are achieving the same result, but at different points, no? FWIW, I've never liked upsampling, whether done in hardware or software... BUT, I've only ever used an OS DAC, where IMHO upsampling simply isn't necessary.

But your thinking on getting the output stage right is facinating...

Mani.

There's never been a clear definition of oversampling and upsampling, somebody refers to the bitrate (which does nothing and its trascurable), others to integer vs floating point resampling, the latter being sort of universally accepted.
ex. 4x upsampling as Peter said = 44,1k x4 = 176,4k
ex 4,something oversampling: 44,1k => 192k

Integer upsampling is the best way to do it, because it creates the interpolated points at fixed points in space (sorry for my poor explanation -i'm abroad and i dont have my reference links).

Long time ago (like 6 months or more) i had a long discussion with Peter about this subject. the conclusion was that if OS has to be done, be it integer of at least 4x (8x would be ideal according to Peter, but there are other issues that are not solvable at present times, so he is settling for 4x -in software).

If you scavenge on diyaudio and audio asylum you would find some of my posts about possible ways to use the computer as transport, about doing the resampling IN SOFTWARE and why it is better than any SRC available to use in electronic devices, including state of the art algorithms such as the ones used by Lavry, dCS, Meitner, chord and so on.
If we consider the FLEXIBILITY of the computer medium for any calculation done in the digital domain, the choice becomes obvious.

I'm very excited by Peter (and the digital engineer) progress of making all this theory a reality, and for a reasonable price. Yes, i'm one that cannot afford 6 figures and that i would be skeptical spending even if i could.
I understand his happiness regarding the output stage, because it is THE most important part in defining the sound of a DAC. Now, I'm hoping that MY amplifiers will be fast enought to follow it... (the tube folks wont be happy)

The new digital board that Peter is waiting will mostly improve on jitter only and will be more crucial for high-res material, so i believe that the sound that he is hearing now is 99% of what he'll experience with the new board with 16/44 tracks.

I'm not sure which amount of information i can post publically at this stage, so i'll stop here. I'm really looking at the final production unit of this dac and i have no doubts that it will be the best that can be made to be connected to a computer transport, short of programming a custom multibit converter.

When I have some time, I'm going to dust off my old 'Sampling Theory' texts from university. I'm absolutely convinced that 4x oversampling (with no upsampling) should theoretically achieve the same result as 4x upsampling (with no oversampling). (Let's just stick to integers, because anything else just complicates things further.]

The fact that we've had oversampling for over 20 years and upsampling for only 10 or so is because, in IMHO, the marketing people hadn't realised that they could scam consumers with upsampling, knowing full well that their DACs/CD players were oversampling anyway and just didn't need upsampling. Actually, to my ears, upsampling just messes things up with an OS DAC.

Now, if you have a NOS DAC, I can totally understand why you would want to upsample. But you'll just get the same result as an oversampled DAC, surely?

I'd love to be shown wrong  :smile:

Mani.

Well, before getting to some answers, it may be a good thing if I try to explain from my views how I see both Upsampling and Oversampling; Indeed there is no strict definition for this (not that I read anywhere), but for sure two very different phenomena are going on, and it is a good thing if we all know what we're talking about, when.

Oversampling

This is the thing a sigma-delta can't do without. Now watch the part "over", which may tell us something like "overdone". Too much. More than we actually want. However, the sigma-delta (1 bit for any sigma-delta DAC and which may be a part of the whole functional bit structure) NEEDS to oversample, and afaik the least will be 64 times, and the common-most is 256 times (I think 512 times will meet the SSD standard, so any DAC (in a cabinet) which tells to support SSD, will be oversampling a 512 times (relative to 44.1 of course).
Never to forget : the more oversampling takes place, the more squares will become sines. At 64 times oversampling, for the higher frequencies this will be a fact; the larger the square wave (low frequency) the more oversampling it takes to round the square into a sine.

Upsampling

See this is a nice and easy "upsample" from 44.1 to e.g. 88.2, or maybe one or two steps further. Now understand the difference with "oversampling" as how I describe it : there's nothing "over". It goes to where we need or want, and it is not more destructive than what we require from it.
So yes, a 22050 Hz original full square, will become a 2 stepped square when 44.1 is upsampled to 88.2 (but it will be far from a sine yet).

Thus, it is the whole point that any not true 24 bit DAC, needs oversampling to operate, and while this process shifts the Nyquist frequency nicely as I described earier (like 256 times out of the audio band) it destroys the squares and thus harmonics. Besides that the heavy oversampling creates HF noise which must be filtered out again, but that is another matter. Otoh, this never allows the OS DAC to be "filterless", and those filters by itself destory sound again (roll off in the highs).

So Mani, if we'd "oversample" 4 times and compare that to "upsample" 4 times, of coure it is the same. But an Oversampling DAC is called like that because its physcis require the (heavy) oversampling as I described it, and it won't work with 4 times only.

Because an oversampling DAC just measures good because it *can't* have the false aliasing, one may tend to believe it is/sounds better. And of course, because of the inherently not present aliasing, it really is. But in the mean time all transients were turned into soft sines.

Now, what happens if we - preceeding thee DAC - upsample in software ? the only thing -besides that the software may use another "upsampling" (I say this on purpose here) algorithm than the OS DAC will use- is that we feed the DAC with e.g. 176.4, and now that will be oversampled a 256 times once more.
(note that there may be a difference here with SRC's involved, who for an NOS DAC may go from 176.4 to 192 instead of going from 44.1 to 192, once fed with 176.4 -> this is also related to the maximum sample rate the DAC can handle. Thus upsampling yourself in software may incur for different results in the SRC opposed to doing nothing in software (already because of the different algorithms used)).


Once an NOS DAC is used, we must realize that various "tricks" can be in order, like playing a 176.4 (or 192) file which may not need any treatment at all, because no aliasing will occur in the audible band. So, this time we don't need to upsample at all, and because it's NOS it wasn't oversampled also. This is IMO why it is so important that the DAC can inherently do 24/192 in NOS, assuming that more and more hirez albums will emerge, and NOS playing at that same rate without any conversions, will be theoretically the best one can think of, so better than 44.1 for that reason (no aliasing) alone. That you are listening to a higher resolution which may come to you as more refined ... well ... personally I have my doubts that *that* is audible. But the lacking distortion just is.
To this I can add that the virtues I heard before from hirez files (and which I heard since the NOS1 only), now seem to have disappeared again, because 44.1 now sounds as refined as the hirez files. Not because 4 x upsampling makes the sample steps more fine, but because the distortion is not there.

It is all rather complicated and nothing is real science as long as science seems to tell that OS is good. Example : we have 44.1 which is oversampled 256 times and we have a hirez file of 96 which is 256 times oversampled. We do realize that the resulting resolution is a matter ogf difference of 256 vs 257, right ? and that one of that 256/257 was a real higher not-fake step. Do we think we can hear that ?
Or another example : we have 44.1 vs. 96 and this time we use an NOS DAC. All the analogue stuff is so much slower that the "huge" steps of 44.1 can't be followed anyway, and the approx. 2 times smaller steps can .... 't be folowed equally (inthe time domain the steps are as large). So can we ever perceive a difference ? no, not with slow analogue. But then again, the 44.1 contains all the aliasing sh*t, while 96 does not. The chance is fairly high that we perceive *that* for a difference, right ?

But still no science ... just some reasoning.

Back to business : NOS 24/192 is nice for playing hirez material, but it is mandatory for Redbook !
At least that is what I say. :)

That depends ...

There will be options for more DACs (this can be in several arrangements, but I will tell about that later in a later stadium (weeks)), and this is meant for multichannel XOver. :secret:
Now, this will be done by an on board processor, but, the convolver files can be uploaded to the DAC. So, you make them how you want them, and the DAC will internally use them. And this is not what I was telling about yesterday of course ...

Yesterday I talked about the filter protecting the amplifiers ... if needed at all. Now, it is the idea that these filters will be applied in software (XXHighEnd), while the XOver "filters" are run by the DAC as I just explained. In either case, the both filter types will not be dealt with by the same instance.
Now, when the XOver is just not there, the on board processor can deal with the normal DAC filter, and the software does nothing in that case.

In either case all has a dynamic setup, because it can always be changed when you like.

Assuming that the software to upload the filter to the DAC is outside of XXHighEnd (and why not), the DAC will be running independently. However, if both filters are used at the same time, the software must take one part, and then the (XX) software is additionally needed to run the DAC properly.

I must be careful what I'm promising, because there is more I didn't talk about yet, and that "more" most probably will need software too, and *that* software can only be XXHighEnd or a redundant derival of it. I don't know about this yet, but I hope in a couple of weeks I do.

Peter

I hope this is sufficiently clear now, since my earlier post from a few hours back ?
But what you will see from the delta-sigma DAC (fully delta-sigma or not) is that it measures even better for the FFT graphs I showed. However, other graphs will show the roll off near 20KHz, which will not be there with the NOS DAC when measured properly (this is not even easy).
N.b.: My Buffalo is a kind of out of order right now, but then I have the Fireface. Note that this is not about how it sounds (like the FF doesn't "sound" at all) but how it behaves. I must see when I have some time (better : want to spend it), because in between the jobs I now rather want to finish the "analysis" stuff in XXHighEnd so I can at last upgrade again, in the mean time solving a few bugs which came around lately (and Mani, your 32 bit file support won't be in there yet :sorry:).

Peter

Haha, thanks. At this stage there are no secrets really, but not everything might come true. So, indeed it is good to stop "there".

Apart from better jitter specs (of which I don't think they are important at all at these already OK levels), the noise level will go down by 3dB at least, but possibly around 6dB. This is partly because of the now 4 layer board, and partly because of something else, yet to tell. But in the mean time something very different is going on, which is of great importance for the analogue stage. This actually is the reason that posting all those 60 screenshots was without real sense, because they will be overruled largely. Well, that is the expectation, when digital and analogue meet.

So many things are going on once that big hurdle has been taken. Now, the hurdle has been taken allright, but behind it may be the water pool. The reccie (good old rally term) showed all was dry though. But as you know it can always start raining unexpectedly ...

Peter, thanks for taking the time to expound your thoughts - very helpful.

I'm not sure if I'm knowledgeable enough to agree or disagree, but one thing is for sure; I totally agree that a NOS 24/192 would be nice for playing 24/176.4 and 24/192 material - I don't see any need to manipulate the data in any way, shape or form at these resolutions.

As for 16/44.1, I've always felt that the 'old' multibit DACs/CD players with 8x oversampling were absolutely fine (with no upsampling required). But I can see that 4x upsampling with a NOS 24/192 would work nicely...

Mani.

That "8 x oversampling" would be the same as what I today call upsampling. So, with a multibit that can be done (at only 8 times) while a not multi-bit can only over do it.
What applied in the old days, still applies, weren't it that the old days chips (like 1541) just don't exist anymore, which for those 16 bit chips is even a good reason (at screaming for 24 bit these days).

So ... since the 24 bit chips also were killed, and aparantly the world thinks it needs to go single bit ... let's see what really is better (oh, I know hehe).

>I'm hoping that MY amplifiers will be fast enought to follow it... (the tube folks wont be happy)

Are you saying that 'tube folks' should not even try this DAC?

Hi Josef,

I'm not saying anything. In fact I'd say that a tube is super fast. This is different from possible "sluggyness" in the bass area, which IMO is unrelated to the speed I'm talking about in the DAC case. But maybe I don't know everything ... :)

I'm afraid that some SET amplifiers could not be fast enough, yeah. But maybe i'm wrong.
My amp is a SET of jfet (the lovely firstwatt f3).

is it possible to have "an executive summary" of the present status: I don´t have time (or patience) to read all these replies
thxs in advance
best
Leif

Hi Leif - This summarizes it a bit I think : http://www.phasure.com/index.php?topic=642.msg6497#msg6497

Now, a few days later, I can add that all your Tangerine Dreams, Beach Boys, Black Sabbaths and even Stealers Wheel sound as from today. But then without the compression of course. It is really weird ...

In one - two weeks I will present more definite results and all the measurements.

Peter

In between the lines ... the new setup is playing right now. And if anything, it's better again !
Another (:secret:) part is to arrive later this week.

And then measuring and fine-tuning.

:soundsgood: (Hans Theessink seemed a good one to start with)

Allright. No long stories this time, just graphs ... :)

Below applies to all graphs :

• Phasure NOS1 2 channel 8 x mono DAC, 4 layer board;
• Setup Single Ended (SE). Note that with Differential (Balanced) all figures will be 3dB better and around half the distortion percentage. So, worst case scenario is shown;
• The DAC running at 176400Hz, apart from the last two pictures, where the DAC runs at 192000Hz;
• All measurements include the normal (2m / 6') interlinks and connectors used. As before there's a volume pot in the chain;
• Each graph shows the worse harmonic distortion or aliasing peaks. IOW if there were worse outside the bands of the picture, I had said it;
• What you look at, is besides Total Harmonic Distortion + Noise (THD+N), the best sound, meaning : THD+N figures can be better, but sound will degrade.
• No filter is applied and frequency response is flat over the whole audio band within 0.1dB;
• All is unweighed, and all figures incorporate a bandpass of 22Hz - 22Khz (but see last few pictures as well);
• Impedance of the load is 100KOhm.

FFT 5 Hz

(http://www.stordiau.nl/XXHighEnd\Phasure NOS1-8 FFT 5Hz.png)

Is 5Hz important ? maybe not. But if it's in the "data" it shouldn't destroy other frequencies, and NOS DACs can do that !


FFT 16 Hz

(http://www.stordiau.nl/XXHighEnd\Phasure NOS1-8 FFT 16Hz.png)

16Hz for sure occurs in music. So it just should be right.


FFT 50 Hz

(http://www.stordiau.nl/XXHighEnd\Phasure NOS1-8 FFT 50Hz.png)

50Hz, and still looking good.

For your reference : this is how some Zanden looks like at 50Hz (courtesy Stereophile) :

(http://www.stordiau.nl/XXHighEnd\Zanden01.jpg)

Keep in mind though, this last picture has an output of -0dBFS, while my pictures have -40dBFS. So, it is about the difference in levels.
So, suppose you can see a -145dB in the NOS1 50Hz picture, this is actually -145 + 40 = 105dB down.
For the Zanden this is -58 + 0 = 58dB down. :scratching:


FFT 200 Hz

(http://www.stordiau.nl/XXHighEnd\Phasure NOS1-8 FFT 200Hz.png)

Everything still over 98dB down.


FFT 1000 Hz

(http://www.stordiau.nl/XXHighEnd\Phasure NOS1-8 FFT 1000Hz.png)

Here we have the worst case of all frequencies; 1000Hz is only just over 90dB down. :whistle:


FFT 2000 Hz

(http://www.stordiau.nl/XXHighEnd\Phasure NOS1-8 FFT 2000Hz.png)

Ah, 2000Hz and already looking better again. Note that to the right (oudside of the picture) nothing is "worse" than you see here. This counts for all, or otherwise I'd say it.


FFT 5000 Hz

(http://www.stordiau.nl/XXHighEnd\Phasure NOS1-8 FFT 5000Hz.png)

Not much to say about this.


FFT 8000 Hz

(http://www.stordiau.nl/XXHighEnd\Phasure NOS1-8 FFT 8000Hz.png)

Note that with a Balanced connection this will be just over 100dB down (as everything is).


FFT 10000 Hz

(http://www.stordiau.nl/XXHighEnd\Phasure NOS1-8 FFT 10000Hz.png)

Keep in mind that the peaks you see at equal distances of the base tone (20000Hz vs. 10000Hz here) are harmonics. IOW, no spurs of aliasing (which are tones reflecting on the Nyquist mirror which would be 22050Hz for 44100 sample rate, but which mirror is here 88200 for the 176400 sample rate.


FFT 18000 Hz

(http://www.stordiau.nl/XXHighEnd\Phasure NOS1-8 FFT 18000Hz.png)

This one is just for the excitement. So, *if* we'd be able to hear 18000Hz, then we wouldn't want distortion products in the more audible band (because of aliasing) coming along with it, right ?


FFT 24000 Hz

(http://www.stordiau.nl/XXHighEnd\Phasure NOS1-8 FFT 24000Hz.png)

Aha, now it gets interesting (or not). This is 24000Hz and more than the normal Nyquist mirror. Well, let's not forget that if we run the DAC at 176400, we formally allow frequencies above 22050 Hz (in fact up to 88200Hz) ... may they be fake or not. Thus, technically this again shouldn't cause distortions of any kind.

FFT 44000 Hz

(http://www.stordiau.nl/XXHighEnd\Phasure NOS1-8 FFT 44000Hz.png)

Whoops ... what happened ?!
Well, I told you, when things would get worse outside the picture, I'll show you and get it in;
What you see here is something which is close to "illegal"; I officially started playing 44000Hz which should be okay for the Nyquist mirror being at 96000Hz here (the DAC operates at 192000Hz now !), and what you see is that a mirror tone comes mightly close to the original (whatever is "mighty close" at 60000Hz vs. 44000Hz). It is 60dB down, way out of our hearing capabilities, but our amplifiers might be bothered by it !

But wait a minute ... would a 44000Hz tone be in the data - hence music ?
Yes, theoretically it can, because I am playing a 24/96 (or 24/192) here, and 96000/2 = 48000 which is always more than 44000, so it is legal; Is it in the data, then it will be processed by the DAC.

While this may be less good in the situation an amplifier gets disturbed by it, nothing is the matter when not, because no mirror products jump into the audible range. And as you can see, they don't even go pass the original tone (or otherwise everything is 98dB or more down).


FFT 94000 Hz

(http://www.stordiau.nl/XXHighEnd\Phasure NOS1-8 FFT 94000Hz.png)

This one is merely for fun, and it proves that when the 48000 border is crossed while running at 192000Hz, mirror products occur in the audible range. It is still 60dB down, but mind you, this is audible !

Ok, not anything to worry about, because no "official" sound creating a 94000Hz tone at this level will exist. But just for understandings : when running 24/192 material, in theory (technically) it can be in there.


FFT 1000 Hz whole spectrumm

(http://www.stordiau.nl/XXHighEnd\Phasure NOS1-8 FFT 1000Hz WholeSpectrum.png)

This is the 1000Hz from before, and as being the worse (see text there), I thought to show the whole spectrum up to 96KHz.
As promised, you can see that nowhere harmonic distortion is worst than you already saw from the zoomed picture.
The worst you see is at the right, being a mirror product from the 96000Hz mirror.


Well, concluded for now, I don't think you'll find an NOS measuring better than this one (and not to forget, all figures will be better again when Balanced connected). Not a 16/44.1, not a 18/96, not a 16/192 (they exist), and for sure not a 24/192. But I'll be happy to make a deep bow for any device showing better. And remember please : this is about sound in the first place. I think it is the most obvious that when NOS (ok, running at 176400 which is still NOS to the terms, but which is upsampling on the other side) can show these figures, OS (oversampling 64 times at least !) is nowhere.
This is what I wanted, and this now has been done. :heat:

However ...

When you got the grasp a bit from what is happening to which kind of distortions, *and* you incorporate that nature will expose frequencies to well above the normal Nyquist mirror (being 22050Hz for red book CD) -but - they can technically exist on 96Khz (or 88.2) material only, you should see that we're really out of trouble when the Nyquist mirror would be at 192000Hz.

Oh, I forgot to tell ... this DAC is a 384KHz DAC now ...
:swoon:

Maybe I am trying to be funny, but I think this is really needed to make the best job of it.
And to keep in mind : the DAC is not upsampling (no SRC in there anymore), and it is up to the software in front of it to make the best of this.

And yeah, I hear you first question : how to connect that 384Khz to the PC ?
Working on that ... working on that ...

Peter

Man you really should get a reward for this!!  :o :wacko: :veryhappy:

:pleasantry:

:clapping: :clapping: :clapping:

Peter I'm totally speechles, :o

I can't even comprehend with the idea how that must sound.

QUOTE
And yeah, I hear you first question : how to connect that 384Khz to the PC ?
Working on that ... working on that ...

After all a phasure soundcard ??
"The Missing Link"

 :clapping: :dancing: :good: :yahoo:  :wub:

Q about volume control:

No digital volume control possible during playback?

just found out is doesn't. so

The DAC has a volume control done the right way.

It will have a I2S input, for CDP or iPod?
Greetz Ava

Sure ...

Hi Peter,

Wow - these graphs show impressive low THD+N even when measured at -20dB. (0dB should be even better measurewise - but I am not an engineer so my understanding is pure layman on this). Have you measured liniarity at low levels - for instance at -80dB which is how some Hi-Fi magazines often test CDs and DACs.

Thanks again for sharing your work and enthusiasm. Now we all anxious await to hear the news

"PHASURE DAC-1 is ready to ship - introductory price - $499.99" ;)

Take care.

Best Regards,

Per Borgen

PS: Hmmm - I still wonder how you get that I2S signal out of the PC... ;-)

With a modified soundcard, easier is one with via envy24 chipset. Easiest on esi juli@.

Hi Per !

Well, I must confess something ... and found that out just this morning ...

I never trusted that very low S/N data (which seems to be some -157dB at the pictures), and so never talked about that either - while I should have, because they are so unbelieveably low. But they are just too low to be real. And they aren't ...

I just found some stupid mistake in the analyser software which had an "Auto" checkbox ticked for auto-sensing the input level (which is not 2VRMS (the standard) in my case), but which only auto sensed the input level when UNticked. So, now the noise is shown at the proper level which elsewhere officially measures -130dB in this case (this case : the status of things, that changing in a daily basis for testing).

As you can see below this actually looks better, because no harmonics (HD) can be seen anymore (compare with the earlier shown 1000Hz picture), but this is only necause they are well down into the noise.
Nothing changed about the real noise of course, but we now know this is at -130dB and not at an impossible -157.

On a side note, something I didn't mention before, but which might be interesting by itself to some :
I may expect that a few people with high sensivity horn speakers float around in here (like mine are 115dB), and as you know it is always a problem to have it all enough hum and noise free, not to hear (mainly) the noise from a distance. This can be done allright, but I never encountered a situation (at others as well) that you could be with your head in the horn, and hear NOTHING. Well, with the now hopefully real figures shown below (but just the same with the unreal figures from before of course :)) you can hear NOTHING. Just nothing. And to keep in mind : in my case this is with the main amps at full gain because I don't use a pre-amp.

About so good THD+N figures, I'm afraid they went worse because of the before wrong analyser setting. THD+N is measured against the noise (the +N), and when the analyaser thinks the noise is at -157dBFS (FS = digital input level), it measures the relative amplitude of -40dBFS (as in the picture below) against that perceived noise level. And if that perceived noise level is wrong, THD+N is also wrong ...
All means that (as it came out) where the noise was 28dB down too much, the THD+N figures were just over 5 times too good opposed to reality. This too doesn't mean that suddenly things are super bad, but for sure it isn't as shown before (below shown THD+N is an indication only, properly not correct -> see below too).


At this moment (read : with the proper settings as how they are now) this is true. Before it was not, because of mis-interpretation by the analyser. The stupid thing is, as how it was before it should have been right; now it is (seems) not. This is related to the linearity for THD of the DAC at the specific input levels (digital output level to it), so I guess - at this moment - I am not sure how to interpret this correctly, or how to set the analyser's setting so that I see what I expect. For example, below picture shows an input to the DAC of -12dBFS, and at 0dBFS THD+N drops to 0.02%. This really shouldn't be so, unless at the attenuation the headroom of the 24 bits is not used *OR* it is used in the first place, while it really should not. So, relation to your quoted question above, for 16 bit source material, digital attenuation up to 48dB really won't affect sound quality (hence THD), while the analyser now shows it does (note that it is not XX playing here, it's the analyser's generator). So in the end something is not right now, and I can't see quickly how to fix it.

Peter

I think I know what is going on ...

I'm not sure how others do this, but how to measure THD when it is under the noise floor. Ok, it can be done (just look at the earlier pictures) but that is wrong, because obviously THD+Noise should measure the noise along with it. Well, this is exactly what the last picture does, and what to do about the fact that the harmonic distortion is just under that ...

It is not as simple as "allright, so THD+N is equal to N only";
If you imagine an amplifier behind this, and it gains by e.g. 24 dB, it will gain the fundamental, it will gain the harmonic distortion (now invisible) and it will also gain the noise. Because ot the latter, HD will remain inaudible, and noise comes first.
You could say that it can become quite noisy then, and theoretically spoken this is true, and nothing different from usual. This, however, depends on the initial output of the DAC (say, that fundamental) and if that is quite low to begin with, the output will be more noisier because of the additional gain needed to compensate for the lower output.

Although I don't know at this moment how to properly express the relation between output amplitude, HD which is under the noise floor and the noise itself, one thing I already know, and that is the noise being totally inaudible (with my current gain of 24dB creating an SPL of well over 90dB) and no music running. This by itself tells that with far of sufficient gain HD will remain inaudible just the same, but this isn't enough for me.

I'll be back on this later.
Peter

Hi all,

Before nobody is going to believe in this project anymore, it is alive as can be. And, as I am known to never give up on something which once came to my head as possible, this time too it seems to work out. At last ...

Many of you will know (mostly PM) that I'm hunting a Firewire connection. Actually this is right from the start, although at that time I though it could be done relatively easy. Well, as it turned out, not so (at all). And I really found all the reasons why no audiophile Firewire DACs exist. One reall really must me a complete fool to hunt this ... but since I am ...

Besides each and every potential option I investigated, one of them I started to work on from off December (2008), and only today I can anounce that one is going to work.
Think of phone calls, many many emails, attempts to let others do all the work, all seemed to end up in nothing, and the throughput time this takes is inmense (well, just 8 months at this moment !). And during that time there never has been a moment of not outstanding emails or answers, but one thing I know : If I had to start a next life, I sure wouldn't be an electrical engineer. What a world that is. To me it looks like the least efficient world existing on this globe. Emails without answers, one-per-day email conversations after months ending in nothing which could have ended in nothing on day 1, etc. etc. Nothing for me actually.
But I want this ...

So, after now two weeks of trying to communicate with the software department of a large chip manufacturer, and feeling that this would never end at making clear specs, and probably ending up with wrongly working stuff afterall, I now decided for that particular Firewire product to write the "settings" interface and driver myself. And yes, the nice thing of this product is that it comes with an SDK (System Development Kit) and DDK (Driver Development Kit), actually created to do the development, which in practice nobody does because it is a hell of a job. The good thing is, I already know that the things I want are possible, based on "them" saying they could do it (thus it can be done) and furthermore by buying ready products which are similar, virtually combine the features they have, and thus now have a good idea what the possibilities are to be combined into this one unique product : NOS Phasure1.

So, let's sum up some features this subject will bring, if all succeeds - which it of course will : :fishy:

• Low incoming jitter
  
  For those keen on few jitter, the data connection (read : Firewire which is normal data, opposed to SPDIF which is audio) is right next to the DAC (think in terms of 1cm), and from there on it is I2S. Can't be better for incoming jitter, which (by now) is known to siple through right to the output, never mind what is done underway to prevent that.
  
  
• Accepts 24/192 over Firewire ... and USB. :yes::yes:
  
  
• Accepts 24/384 over Firewire. Yeah yeah, cannot be. Well, watch it !
  
  Small remark : This will only be possible with XXHighEnd as playback sofware.
  
  
• Choice of clocking by the PC or by the DAC (Firewire part).
  
  
• Parallel to the above (thus actually unrelated), WordClock In, WordClock out.
  
  
• Galvanic separation with Firewire/USB from the PC.
  
  
• Automatic Sample Rate changes to-from all Sample Rates.
  
  
• Asynchronous connection for Firewire.
  
  Here a disclaimer is in order;
  These days rumours go that Firewire would be an asynchronous connection by itself. I don't think this is so, or anyway not of the "real asynchronous" some USB DACs these days can do. This is related to the earlier mentioned subject "Firewire can clock the data" (opposed to the PC), and on that matter it seems just to be already there. But, first of all this is not a common Firewire feature. For example, if my Fireface800 can do it, please let me know how - apart from some Master setting which IMHO does exactly nothing in this case (being the master over the PC). Anyway ...
  When this "Firewire can clock the data" is already just it (fully asynchronous) well, then it's there inherently. And, I can only learn this with the SDK at hand, which I don't have yet. But if it's not "it", I can create it myself, a bit depending on the possibilities of accessing the on board memory, which is needed for this. That too can only be seen when the SDK is at hand.
  When all is accessible as I now think it is, all is based on the good clock the DAC uses, and which deviates a handful PPM only. I won't go too technical, but with a very low deviation of the internal clock, the internal memory can serve as a buffer that may last an album or more, and no such thing as incoming jitter can exist. A PLL (jitter creating by itself) will not be there (is not needed) and the inherent jitter will be that few PPM I just mentioned.
  Theory for now, and I strive for making it practice.
  
  
• The best sound !
  
  This needs a small explanation of course;
  The explanation will come through as I intent, only to those knowing/using XXHighEnd, and the means I use to let it sound good;
  As many know/assume or expect, the way XXHighEnd does this is all the most indirect, and while the software outputs bitperfectly, it is just waiting what the rest of the chain (up to the D/A) will do with it. This time, the chain is mine, so to say. In other words, I will be able to apply the exact same as we are used from XXHighEnd, which undoubtedly will boost SQ further. You will not be able to recognize it or compare, unless improvements at this driver level come along, and you upgrade the driver or the firmware which again is a level closer to the DAC (that would be the Firewire chip).
  So yes, here is a crazy fool who will offer firmware upgrades for a DAC to improve its sound.
  
  And oh, before someone asks, since all is programmable in hardware to my (or our) creativity, it all may and up with filtering (like it now happens at the software side in XXHighEnd) right in there. I talked about this before I think, but then this was about theories and virtual processing hardware devices (like "some" DSP). Here, the DSP is right at my hands and the theories are now narrowed down to "program it". But, for later.
  
  

Although there will be a complete separate SPDIF input (and AES/EBU) this really will be all. Thus, no Toslink, because it doesn't fit the architecture (although it could be made) and will sound lousy anyway. At least that is what I think at this time.
I2S out as well as SPDIF out can be there, but I didn't decide about that yet.
Analogue out was and remains RCA + Balanced.

I have a disappointment too, and the word has to come out at some time anyway :oops: :
As you may have imagined, I only go for the very best, and if there is only one slight little degradation of sound opposed to the possibilities (which are sometimes theories to start with), I won't do it. And so it is :

Throughout the time I have been examining all the dozens of means the analogue stage can be comprised of, plus one.
That one is a finding of myself, and it is not official. No, it is officially wrong by the grace of no engineer understanding how it can work.
The number of times I went back and forth on one of the dozens of options mentioned, and "my own" is countless by now, but each and every time it sounds the best by far, and measures the best at the same time.
I found this one relatively long ago (must have been April), so you'd say "why bother further". Well, apart from that it is not official which for me is a good reason by itself, the DAC will have a very low output. Summarized I make use of the DAC's best operational level (read : where are its best distortion figures), combined with a fully passive approach without any downsides (no components holding back or squeezing anything).
Well, opposed to what is normal, in this case the output is 18dB less which may be too much for you. If you currently play at 12 o'clock or something, really nothing is the matter. But if you recognize to reach a quarter past often, you will find yourself without juice too soon.

That's it for now. Ok, maybe I should add that again the DAC won't be ready tomorrow. I can tell more after the first looks at the SDK and all, which also will be the time I can predict the price. And I assure you, it will be worth it !
Don't even think "tomorrow" to hear me back on this, because this world is as it is, and I don't expect it to change. I mean, it was day before yesterday I took the decision on buying the SDK/DDK (which is only another 10K, yahoo !) and although I asked in the morning how to proceed, I heard nothing yesterday, sent a reminder today (reminders are always necessary somehow), received an answer at 2pm that I had to provide a purchase order and a FedEx account number - which I don't have, asked about other means and whether that was necessary and ... heard nothing. Just as expected. So tomorrow morning a reminder is needed, and I'm sure another question will come from that, as I am sure it will take weeks before I receive something here. I don't know what's wrong with ALL of these guys. And you can't get mad at them.

So far,
Peter

AMAZING,

Really like that clocking of memory thought, really smart Peter (You Crazy Fool :grazy:).
So just loading music data on the memory and start clocking from there?
And writing your own drivers in SDK/DDK (its that or waiting for aliens to land, I guess)
Btw 24/384 needs your own drivers anyway

 :ok: :clapping:

Please take your time, you are already lightyears ahead of all out there.

Roy

Well, the decisions taken and all, the now definitely to use interface I had laying around for two months already, today was soldered next to the DAC. The I2S lines are still 18cm (1cm to be) and the below looks pretty good to me.

Firewire is clocking, buffers set to 1ms (can be less), I'm listening to Taj Mahal - The Real Thing, and I have the shivers for more than 50 minutes now (who needs Viagra :grazy:).
A tuba already was an instrument I recognized as a very good representative for sounding honest and real - or just not, but man, this really beats everything and all. Phew, the album just finished minutes ago, and the shivers won't go away.
And then imagine it was the last album I played yesterday, sounding awfully good already (you just can't sit still on that last 19 minute track), but shivers ? no, not that I was aware of.

Or maybe it's because I at last got where I wanted to be for so long (and theories start to work out) ?

So, the firewire is working, hey? I know this has been a real 'crusade' for you, so well done!

Now, what are the most obvious sonic benefits, as compared to the 18cm I2S lines that you've been using to date?

Are you using a similar firewire implementation to Weiss?

Mani.

(sorry for typos, didn't read it back)

Hey Mani,

No, no dice with DICE, haha. Something is just wrong around that chip (the organization), and I can't find my ways there. Nowhere actually, at seeing that it takes 8 months to get somewhere with one of those products, which more often are virtual then real. Anyway, I've signed an NDA, didn't read all the small print, so I better keep quiet a bit (for now).
The implementation itself though is proprietary, and while Daniel never offered me his "implementation" (no name one) I guess I will never do that either. And then we'll see how long this DAC lasts as the only Firewire connected audiophile DAC ... :whistle:

On the sonic qualities I first must try to explain something else;
During my quest on the analogue stages (man, did I try many) you just won't believe how TOTALLY different every single solution sounds. It is always a difference of night and day, and the worst of all is : most of them sound good. Of course this can't be, and we all would say "but one will be the best". Yes, but it doesn't work like that with night and day differences and both are "good". I'll try to give one example, this time in the area of just the connection;

For many months the connection has been I2S by means of a tweaked soundcard in the PC. This was by means of a Cat5 twisted pair, carefully grounded, unterminated. On the scope it looks slightly worse than the pictures from my last post, but, not as steady which tells me "jitter". 80cm is or ir not too long, but if anything ... too long (ref. Gordon R.). It sounded just GREAT, and everything became very sweet and nice of it. Fragile if you want. On a side note, this is a very dangerous judgement when you're working on the analogue stage at the same time, knowing that a very first thing happening there is ... filtering. Filtering of resolution, of bass, of treble, add colour, add flair and anything you may be able to imagine, also thinking there's no further preamp to filter further, and thus you hear everything. However, compared to SPDIF from the Fireface, the I2S connection was a for me most clear very much more clean sound. Call SPDIF plainly "rough" in comparison. Rough vs. sweet.

In a before post I was talking about buying other products (external soundcards) in order to try out native possibilties for the chip I chose, see whether the I2S lines could be taken from it etc., and since those products never have a native I2S connection, I obviously started off with connecting such a device with SPDIF. Not different from how the Fireface would be connected. This is what I have been using the last 3 or 4 weeks, and I just let it be because this by itself again brought a COMPLETE different sound, with the major property of things being fresh fresh and more fresh. Funny thing is (if you followed the Relase Notes on XXHighEnd) I started using this device right at the moment I eliminated the Mem and LDN checkbox from XXHighEnd and immediately was sorry because here again I had the super dynamics I experienced before, and of which I thought it wasn't good and why the LDN checkbox in XXHighEnd emerged. This is hard to imagine and hard to explain, but it comes down to each and every album you start, starts with some smash which makes you dive away, so frightening.
This is not fatigueing (at all) and merely is the most interesting throughout. It brings detail at a level from another dimension (the attacks), and when you're not all that much analytical you would like it forever. But, in the end I don't think it is reality BUT I cannot be sure because why would an attack be more loud than reality. It can though, and this again is related to the analogue stage (think of overshooting), but how this is related to the digital connection ... hmm ... (not much different from XXHighEnd could incur for it in one of its early (never public) versions).

At listening to it for these three weeks - and if something lasts for three weeks it must be good - in the end I thought "the hell with it, it is going to stay like this, it is just too good". Remember, this was SPDIF. I must tell though (and I'll come back on this in a next post) that this was a Firewire clocked connection, so it could be very different from what I was used to with the Fireface/SPDIF, because I never saw something in there "clocking the stream" (again, see next post).
Now, only because I had to build some PSUs and do some analysis and soldering on another similar device I already had laying around for a couple of months, that ever meant to be a final solution to go into the DAC's cabinet, plus it would be out of sense because it would need (driver) programming which didn't come together, it was only yesterday that I finally stuffed in that interface, this time I2S connected.
Long story short this again outbettered everything, just because everything sounds "normal". So, no smashes and crashes, no superfresh, but also no supersweet from the earlier I2S connection and better colours in the highs (cymbals). And much more important : very much (more) involving, which for me always is a last resort when other judgements fail, or day and night has too much contrast while both seem to sound good.

If you are still with me, in the end it is somewhat (no much) more complex, because as you know I'm also on my filtering project. This is a tough one, because it is so so hard to judge for its merits. So, a little story about this :
When I first started using the filtering (which is in softare, XXHighEnd) it was the most obvious that everything became dead black and quiet of it. This was with the earlier I2S connection, and I imagined things got overdone, and I was listening to an oversampling DAC. Well, of course it was (4x). Now, at moving to the SPDIF connectoin I just talked about, and at hearing those relatively (to I2S) emphasized freshness, I thought to switch off the filtering because if anything this would show the effect of it. I was playing Abdullah Ibrahim - Senzo, and I never never heard such a good recording in my life. Sadly I just received the album, so I didn't know it from before. It is taken a kind of loud, and at all that beautifully singing and interacting piano notes I heard harmonics so much emhpasized, I thought it would be alisaing or whatever. But it did not sound wrong. So, I swicthed ON the filtering and ... really heard no difference on that matter. Thus, really no perceived "anomalies" disappeared, and the only thing I knew is that the ambiance of the superb recording disappeared. I didn't see him working at the wing anymore (this is one long 60 minute recording without breaks) and I didn't hear of feel him breading anymore. It stopped working.

Out of all this was this superfresh SPDIF situation, and while the already so sweet I2S got even more quiet from the filtering which I perceived as good, I expected harmonic anomalies to disappear from the filtering in this SPDIF situation, but it didn't change a thing to that (if I perceived anomalies in the first place,and you really have to listen to that album to understand what I mean ... btw, get that album, it is the best I own (just piano, nothing else)).
And so it happened that I never switched the filtering on anymore, with in my mind "if I ever hear aliasing anomalies or stuff my amps can't cope with, I will hear it with this superfresh and dynamic sound". Well, that too stayed for 3 weeks, and again, anything lasting that long is there to stay.

The now implemented I2S connection (which btw is the one of 18cm long and will be 1cm or less in the end situation) is theoretically the best, and as said, sounds the most normal. If it is still there in a week's time, it is there to stay. I don't know that yet because as always it takes many music styles to judge it to its merits. It sure brings the emotion, which doesn't mean hiphop will sound good as well (somehow hiphop is one of the toughest styles to let sound good via PC playback ... ah ... who cares :)).


Well, in the mean time some may wonder how it can get better and better over and over, and the possibility exists I am perceived as that reviewer who always and always finds the again better sounding product each and every subsequent week. Not so. If I say something sounds better from before I mean that, and if I have said it 10 times by now, it really got better over and over, and the only thing it should tell is that by now you can't have the slightest clue on what I'm actually listening to. Well, for those perceiving the sonic improvements on XXHighEnd (of which I hope that's all of you out there), it happened there over time (again and again), and this is the same, but multiply the difference with a before version by 3 or so to get the grasp of the more huge differences which can be achieved at this (hardware) level.
But then I only report the improvements, and not the failures. Maybe I should post a photograph of quite some KGs of failed analogue stages. One of them even exploded (well, some caps).

Mani, allow me to ask you a question, since I my self am not certain about it;

I know you are experienced on the Fireface and using the Wordclock or possibly internal clock stuff to create a better (clocked) connection with the DAC (or PC for that matter). I never understood that quite per the lack of devices allowing such conections. With the experience you have on this, can you tell me :

Whether - with a normal Firewire connection used as SPDIF, the "Master" setting or anything else can influence the way the stream from the PC is sent or clocked.
Although I can tick checkboxes in the Firewire driver screen, and supposedly they should do something, I never saw any difference, and combined with the fact that the Fireface does not "lock" to the PC stream - or anyway not visibly by the lock lights present - I always thought it just doesn't do anything.
Is this correct ?

I ask this because the Internet seems to have started rumours that Firewire would be asynchrounous for the connection which I find hard to believe as such, but seems to be in the area of this "Master" setting anyway. Besides, when so much fuzz is made about USB/Async, why would Firewire suddenly be the exact same.
However ...
With what I use now I can explicitly choose for the PC being the master or Firewire being the master, and in that case it really does something (both settings do not allow the same small buffersize).

Do you have opinions on this regarding your own experience ? (I'm not asking for babble from other forums, just your opinion ;)). 

Thank you !
Peter

Peter,

A very short response for now - I'm working in Houston all next week and need to get a lot prepared before my flight first thing tomorrow morning. Sorry...

I remember when I first started using the Fireface (slave) with the Esoteric D70 (master), I wanted to experiment with getting better quality sound out of DVDs. Using Media Player to play the DVD, I found that I could indeed control the 'speed' of the video by adjusting the clock frequency on the D70. More recently, I recall playing a programme on the BBC iPlayer and again being able to adjust the speed of the video with the D70.

Ergo, the D70 in master mode was definitely affecting the way the stream from the PC was being sent...

Not sure if this answers your question. I'll post again from the States when I have a minute.

Mani.

Hi Mani - thanks so far !
Indeed I knew that, and for the exact same reasons (say, better sync with video hence display frequency). But now I never saw that *not* working. And uhmm ... I did that with the Fireface itself (DDS tab) ...
In the end I guess I never understood it well.

I'll be happy to learn more from your experiences on this one, but don't make any fuzz about it. When time permits, and even then, no obligations.
Peter

Fantastic achievement, Peter.  :goodjob:

How does your dac sound when fed a 16 / 44.1 signal from a ordinary CD transport through either SPDIF og AES? Do you have ANY idea if the price of your dac will end above or beyond the 2000 Euro mark and when it will be ready for sale? Are we talking 1 or 2 years from now cause I might be tempted to live with my humble TEC 24 / 96 spdif and toslink dac and rig a PC with XXHighend and souncard with SPDIF out until then. And start saving up....

Keep up the good work - and take care.

Best Regards,

Per

Shivers.
This is what I wanted.
I didn't get them when I heard the very early prototype of the NOS1. It sounded better than most digital sources that I have ever listened to but it dint get into the emotional level. I think (hope) you are there now. I just got back from short vacations and I think I have to fly again soon... :)

I'm pretty certain the RME and Weiss firewire implementations have always been 'asynchronous' (a term which I still find really confusing in the context in which it's being used - but maybe I'm just too stupid...).

Thanks for describing the sonic differences. I have to admit that I lost you a little though - too many different permutations going on.

I'm not sure if I have any other insights to share from my own experiences. All I can say is that 'master' mode has always sounded significantly better to my ears than any other word clock mode. To me, this makes perfect sense because the clock sits right next to the DAC chips... as it should.

The exact mechanism by which this clock then controls the stream from the PC, I have no idea about. Sorry.

Mani.

Thank you Mani.


Well, who is asking for differences, not knowing the base (reference) in the first place ? :) :)

If I had to describe the sound as how it is now, I would say

- Aggressive;
- Very good separation of individual elements like a plunk on a nylon string;
- Something like the latter being very forward and straight (no vibrato no flanger) indicating few jitter;
- The sense of wanting to jump along. It is "performing".

The "aggressive" is a dangerous one, because it is quite the opposite of "dead bird" and comes along with "not laid back". You could almost say "less suitable for background music". Dynamics are fairly high (but not too much this time).


One other means of, say, objective judging :
Yesterday I sat down to compare a few means of filtering. Never mind why, but this testing needed an SPDIF connection, so I used the same setup but with Firewire used as SPDIF passthrough (still clocked by firewire). Didn't listen to anything else earlier on. Tried many things during 90 minutes or so and couldn't decide for anything. After testing time was up and normal listening hours should start, I connected everything back, and immediately there it was, WAM.

This connection is of huge importance, as in fact everything is. What "you" will never be able to grab without listening, is how in the world it is possible to create such ENORMEOUS differences by these kind of means. I believe it because I hear it. And then to think that people of good reputation exist telling that all DACs sound the same ?! Well, this is one only, and I guess there will be a day soon that I will write out what switches could be there without pain (SQ degradation) so you'll have 5 completely different DACs instead of 1.

Peter

Hi Per,

How does the DAC sound from an ordinary CD transport ... good question, and also the reason I respond a bit late because I wanted to try. But, I still didn't get around to that. So, before I do (and you might remind me when it takes too long), some general blahblah :

In the last post, and earlier as well, I in fact described the sound coming from SPDIF (well, indirectly). To me it looks like the second best sound coming from "a" connection". It is special somehow, and the character is super fresh without being able to recognize much what's wrong. But, between not-wrong and "working" quite some world exists, see my before post.

We must be very careful here, because what I by now reject, could have been a 100% desire one year ago. By now I am spoiled like hell, and nothing is good enough except for the new and better than before. Stupid example : If you read Telstar's post again (a few posts back) ... we spend a day (that includes a night :)) at comparing a perceived good OS DAC with mine of that time. We were honest, and compared with SPDIF only, because the OS DAC could connect to SPDIF only. But, I and "we" knew there was also I2S, and the only thing what happened with that is that I told it was tremendously better, but nobody ever asked to listen to that, and so we never did. Isn't that strange ? (btw, through the tears in my eyes I've seen them in Telstars' but never mind that haha).
I only want to say : after perceiving the sound of the good OS DAC and listening to mine, the OS DAC suddenly sounded as bad as completely broken, which really was the sudden thinking of the owner of it (which btw was not Telstar).
Now, hopping over to I2S would have encouraged for the same idea : "my" SPDIF is broken.

To emphasize further about "being spoiled", since then a few things changed. And, each topic I mention below is really such another step of "the before must have been broken" :

1. The I/V and output stage has been fully replaced with another one. Maybe this has brought the biggest change.
2. The DAC itself has been replaced by a complete new design (before it was differential 4 chips, now it's differential + parallelled 8 chips).
3. In fact including in #2 but of major importance : the means of clocking has been changed completely from the ground up.
4. The direct Firewire connection, that interface being right next to the DAC with internal I2S connection. This too brings "a biggest change".

So what to say, other than that you most probably would have gone for above not mentioned step -1, being the old situation and SPDIF connected, step 0 being the I2S connection Telstar already never heard. And if I say things got better, well, I hope some people trust this.

All 'n all it is my guess that your CD player will improve by many miles. But, to to it really well your CDPlayer should be tweaked for I2S-out;
The DAC board will be as close to the cabinet terminals as possible, and I estimate that maybe 2-3cm is needed internally; If you put your CDPlayer on top of that, there may be, say, 20cm needed to get out of the CDP's cabinet plus 15cm to go to the DAC. All together some 37cm.
In this context I should note that my earlier I2S connection was about 80cm and that worked perfectly. As a matter of fact, I don't see where it fails, BUT the sound is 100% different. In the before post I described the current sound as "aggressive", which is not only the opposite of laid back, but also the opposite of "sweet". That other I2S connection (from a tweaked PCI interface) is pure sweet, also "works" if it is about emotion, but totally different sounding. On that matter it could be such a switch I mentioned in my before post, and with an additional EUR 150-170 you'd have that too. Ehh, for a PC connection.


No, this won't take two years anymore. At this moment it is hard to predict, and I recall telling a few people last spring it would be "this summer". Well, this summer is almost over, and I won't be able to make that. The biggest hurdle has been the Firewire though, and that seems to be behind me. However, as I told in an earlier post, the communication and all with a few companies and people I just need on ths subject, is slower than the slowest snale. Last thing what happened on that is that I was "allowed" to put a purchase order for the SDK/DDK I need, and never heard anything since. And to keep in mind : this Firewire interface was ordered last December, while I received it in June. I guess this is how things go when you're really into something not existing, and you are depended on others. But, I will persist. Maybe "you" can't wait and this is your right.
If you ask me now, I'd say it will be strange when "this year" is not in order. But keep in mind : I said similar about "this summer". And what's REALLY to be kept in mind is that it is all for the good - no, best cause.


The price is still out in the open at this moment, in fact awaitening two things :

1. I must look at the SDK/DDK (system and driver development kits) to see whether what is promised really can be done - and can be done by me. If it can't be done be me (or my company) it will take $$$$ to have it done, and this will influence the price. As said, I never heard anything back on my P.O. so far.

2. The current setup looks like finished for SQ (although programming on the driver may change that more or less), and the current PSU used for the digital section is a good one, but is a kind of large. So, the base SQ known now, I want to "play" with PSUs here, and possibly use a super shunt regulated in the mean time. Because this is for the digital section it shouldn't matter, and *thus* nobody does it. Aha, well, in *that* case I should try it. :prankster:

But 2000 will not be in reach I'm afraid. My earlier calculations showed 3400; I just summed it up again, and although every single part has changed opposed to the before situation, I now come up with 3310 excl. VAT and shipping. This includes a dedicated manufactered housing (so not a stupid DIY box).
I think it is a bargain since no hours are in there for the design and all, and if it takes, say, 500 hours to get the driver and all where I want, there is no 50K+ calculated in there, which even for a 100 units should add 500 to the price.
Besides, there is hardly anything to find not being in there, and I'd say that everything everybody always wanted comes together in this box.

Although you must go for NOS/Filterless to really like this, do not forget it can behave like a nice oversampling DAC at the same time with filtering of choice to apply in software. The filtering by itself is upgradeable and possibly ... maybe leading to an even better SQ than NOS/Filterless, meaning not ending up in a dead bird. This is what I'm still working on, but unimportant for the NOS die hards. But wait ... if it can get better afterall "we" might give in a little ... :)

Peter

I thought I listened to i2s at the very beginning  :( then switching to SPDIF for a fair comparison with the other DAC.

If you read my post of 6 months ago, after the audition, maybe between the lines, cuz i dont remember, I said that the NOS1 sounded very very good, but it lacked that extra, the "emotion" that I'm looking for and that so far I got only from the Naim cd555.

By all means, Peter, go over the top with the power supplies (in particular for the clock*). I'm pretty sure some of the magic of digital (not sounding such, in the good sense) comes from there.

*if you haven't, take a look at Terra Firma XO writeup on their site (forgot who are they, but they are $wi$$).

No no ... we never listened to my DAC in the beginning. We just started off with the OS. Maybe you recall I wanted that for an hour at least. In the end I started predicting what the differences would be ...

So, I could be wrong of course, but I think you just never heard the I2S connection. But then remember, it wasn't a show off or anything, and all was under the subject of your friend's OS. Or at least that should have been his subject for the trip. From my part I saw no reason to emphasize "that" (outcome) by outbettering it even more and start laughing or whatever. Not my style.
Strange situation of course.

I'll dive into the PSU matters for sure, and let's not forget, have some (new Firewire) clock connections to test too.

Tadaa, something for this topic I guess :

I can confirm that 24/384 input to the DAC with the PC as source will be possible !

As known I wanted this right from the start, and today I managed to let it work.
This gives us acces to native 24/352.8 DXD (so, 2L go ahead with it !), and of course Octad Arc Prediction Upsampling, AKA OAP (haha).

I'm always glad when something works out I set my mind to. :yes: :)

There is one small downside :
The DAC needs a new PCB which design by itself is already there, but which has to be prototyped and after that, produced (yes, again :heat:). This may imply that January 2010 (as told in another topic) is not feaseable, and it may get to February.
But I guess it is worth it !!

Yahoo ! Peter



PS: There is a difference between the practice I just got running, and the real practice with the new DAC board (think in areas of unexpected higher noise levels). So this is the small disclaimer for now (throughput time). The expected upside of it all, is that the new design will once again be about less = more, like leaving out all the SPDIF sh*t because it just isn't needed (SPDIF - for those using a CDP - comes in by other means now). So, it has become a pure i2s connected DAC with no incoming jitter (did I say No ? yes I did). :whistle:
:secret:

 :goodjob: Man you are a piece off ......

 :)

terra firma is from allan wright of vacuumstate.com
best
Leif

:scratching:

Everything allright Leif ?

Peter,

Any update on the Phasure NOS1?   :biglol:

Sorry for being most curious!  :)

You better come over one of these days. Haha.
A picture says more than a thousand words doesn't apply.

Dear all,

After a rather nervous afternoon of soldering, press Play and first looking whether the VU meters were still jumping up and down, next switching on the main amps - fingers crossed ... imagining the latest Kernel Streaming improvements couldn't be surpassed for better sound ... oh man ...

Those who visited me, shorter or longer ago, expressing something like "never heard such a thing before" ... let's call that history.

Allright, with all the things I did so far, a while ago it appeared to me I had the beautiful opportunity of connecting the "clock source" so much directly to the chips that, well, nothing would be in between it. So, I took it up as a final unplanned little project and thus I got myself a couple of low jitter clocks (3ps max over the whole frequency range), fed them with separate shunt regulated power supplies, and now the net result should be 3ps (max) of jitter. Not that I know how to measure it, but my ears tell me enough ! And I don't even know whether clocks have to burn in or something. Or run in maybe, haha.

There's loads of more tight bass (AGAIN), woman voices sound like beautiful bells, and the Jens Gad (ex Enigma, with the known "grayish" sound in the highs) I'm currently playing, now just show very high resolution sprankling highs.
:smirk::smirk:

The means of connection remains a secret, but I promise you it is still an external DAC.

Well, for all the elements which may make good or bad sound, the *very best* solution has been applied now. There is just nothing left, or it sould be the clock itself with again lower jitter, and which can be replaced in 5 seconds really ...

Before the question comes ... I have over 200 DAC chips here now (alone costing a small fortune already), and I'm in the middle of the purchasing process by now, so please don't think I'm not serious. It took a lot of time though, and while I can only say "sorry" for that, it sure has been for the good cause.
The earlier promised webshop is already there somewhere in the back ground; the ERP system behind it not yet, but this is a matter of spending some time on it. Now I definitely know all the hardware elements to go in, a nice slick cabinet can be made, and this shouldn't take all that long (but undoubtedly longer than I want).

So far for now, and thanks for waiting,
Peter

I hope my name is down on your list of first buyers.

I can't wait to put the NOS1 up against a real DAC ;)

Mani.

Hello Peter
Do I understand that right, there will be a new website for the hardware?
Thanks
Jack

Oh yes. But if you look at the current (Domain) names, you know what will be happening ...
:)

Peter

I can't remember. Is there going to be a faciltiy for driving headphones on your DAC?

Thanks
Frank

Peter,

That is fantastic news!!!  :goodjob: You are having a big party i guess..... :blob8:

Hope you will sell thousends :biglol:  :NY01: :xx:

Frank, sadly, no. Different from usual, in my case this needs an extra facility which we just as well might call an headphone amp. So, since this is needed, I guess it is a kind of useless to build it in; it will cost the extra money (though it would be an option to have it in), and maybe it is better if people have the headphone amp to their likings. Notice though that I sure would be able to provide one of the (considered) very best, (Fet based) and that there will be space in the cabinet for it.

Please notice that I never use headphones myself, and thus don't know really what makes/brings "a best" headphone amp (you might tell me !). What I do know though, is that it will be in there without a real cable connection other than lead the wires to the jack terminal, which will be 1-2cm. Also, the special design of the cabinet will do some additionally good, as I expect (wait till you see that, haha).

But all together, knowing that it will need an extra amp anyway, I'm not sure people will prefer that.
Peter

Hi Peter,

Happy Easter to you - and all the forum members / guests. Any updates on the NOS1 DAC project of yours?

A short briefing will do, too  ;)

Per

Ok, since you ask ... From off last Friday the 24/384 is a fact. So, all I wanted works now.

Now I am "struggeling" with a display, but because nothing really good is available I may give up on that soon (but a display would be nice !).
It really can't take months anymore now ...

Peter

Consider VFD or full color lcd, the former being cheaper and still good looking (see nuforce cd, and many others).

How did it sound?  :soundsgood:  :)

But it was an april's fool! wasn't it?  :prankster:

Haha Telstar, sure not !

Gerard, yesterday I finished what I named "Octo ArcPrediction" (8 x) and immediately I noticed the separation and profoundness of everything calling itself a cymbal. Things were creapy already and this keeps on continuing. So, this 8x step seems to bring something special - something unusual. I don't have words for it yet and it needs a get used to.

One little :secret: though ... Today I created DoubleOcto Arc Prediction.

:whistle:

Yeah, why ?

I would also like to reserve a place in the line for a DAC.

Thanks.

For those of us who are not engineers (but only informed enthusiasts), would someone care to explain in relatively simple language why filterless DACs are better? I have been spending a few days trawling through the web and have come across some articles (written by Dan Lavry, Kusoniki etc.) but I can't understand them for the life of me.  :(

I also have a Weiss Dac 2 which is 24/192 but not filterless, which my wife thinks would be hard to beat, but I am keen on understanding the NOS1 better...

Thanks for the help.

Ok ... It will be obvious that I try to answer this, which does not mean others aren't allowed to go against it, or doubt my thinkings. Allright ?

In simple wording, I hope :

---

First of all this is about Non Oversampling (NOS). This should be read as : The topology (design) of the DAC chips used is so, that it doesn't *require* oversampling in order to operate. Thus, the 1 bit chips (with sigma-delta design) *do* require oversampling, and this is always to be read as "heavy" oversampling (like 256 times etc.). This kind of oversampling by itself requires filtering or otherwise you will be left with noise only. Now :

A Non-Oversampling DAC does not require this filtering, and this is because it doesn't make use of the oversampling technique. It just doesn't need it. This can only count for multi bit chips, with as many bits as the output actually tells. Thus, a 24 bit NOS chip, makes use of 24 physically available bits. A sigma-delta chip makes use of 1 physical bit only, or possibly a mixture of the multi-bit principle (like 5 bits) and further based on the 1 bit principle.

So, an NOS chip does not require filtering in order to operate, hence produces sound without noise without any filter. However :

This doesn't mean NOS does not require filtering at all, which is related to the harmonic distortion coming from an actually too low sample rate. For example, a tone of 22050 Hz captured with a 44100 sample rate, will be an exact square wave. There are just not enough samples to even come close to the original sine (assumed the 22050 Hz tone is to be a sine). Next, it is clear I think that a square wave doesn't sound like a sine wave. The square carries harmonics, and what you will perceive from it are tones at lower frequencies than the original 22050Hz. In other words, the original 22050Hz sine -inaudible to most- now will be very much audible.
The way this works is not easy to explain, but what it comes down to is that not only the 22050Hz is a square when captured at 44100 Hz, but when the original 22050Hz is captured (!) it won't be captured at its exact "square" points. Thus, if coincidentally the capturing machine is a few micro seconds ahead on the time axis, nothing like a square will be captured, but merely a "mess". So, the original sine won't be a sine, it won't be a square, it will be a mess. And from this the lower frequency tones emerge.

The higher the sample rate at capturing, the better the original sine will be represented. And, at an infinit sample rate, the sine will be captured / registered as a sine again.

Since redbook (CD format) is about a registered sample rate of 44100 (times per second), it is so that everything above just under 5KHz will be "messy". Thus, looking at a scope, original sines of 5KHz and up don't show as nice sines anymore. This means distortion.

If during playback the missing samples (for a good representative of sines at the higher sample rates) are filled in, the problem has gone, as will be the distortion. However, in general there is no means known (but see later) which is able to fill in the missing samples;

Clever mathematical techniques were able to create a "filtering" means, which is able to restore the original wave form. The sine from our example. This is what -amongst others- Dan Lavry explains about. If you read that, it seems waterproof. But is it ?

The (digital, mathematical) filtering techniques as decribed by DL are based upon the "knowledge" of upcoming (or past) samples. Very simplified, if you'd capture that 22050 Hz sine many times, you will know how the waveform really should be, which is by itself comes from the frequency not being an "even" amount of the sample rate. Thus, now thinking of a frequency of 22049Hz but a sample rate of 44100, if you sample this wave many many times, in the end you will have captured every point of the wave as how it really was. Just, say, average all, and there you'll have it.
The downside of this is just that necessity of many other samples, which means your actual to be output sample is subject to others. This vaguens. And as how we speak of it, this "rings" (echos). Now, the major problem is that at measuring a single frequency this will just look (and be) how it really should. There is nothing to ring, and if it rings it rings as should. But this is no real life audio !

So ...
The only way to overcome this, is creating a "filtering" means that injects the samples right on the spot, and without the knowledge of other samples. And remember, it doesn't (didn't) exist.

Each DAC which is not filterless comes with a filter of the wrong type. This is because indeed the right type doesn't exist. Of course one with the right formula could choose to hard-code that formula in the digital filtering of the DAC, but this is a not so friendly means. I mean, the development area of a DAC (which is to be seen as hardware) is not the most flexible. And now the crux : so this is why I developed a filterless DAC (which must be NOS for it), in order to provide my own means of filtering from the playback software ...

And so it happened that I first created the (prototype of) the 24 bit NOS filterless DAC, and next created the Arc Prediction Upsampling, which just is that means of "filtering" that injects the samples at the proper places.
With a filtering DAC I'd be running behind the facts.

To complete this story : In order to get rid of the harmonic distortion because of too few samples, the DAC must be 192 (176.4) KHz at least, and it should be 24 bits because upsampling can't go without using additional bits (which is another story).
384 ( 352.8 ) KHz would be more optimal because it will shift away harmonic distortion from above 96KHz to above 176.4KHz, for those who are sure that even way in the inaudible range we still perceive things. So, indeed for over a week now I am playing with "Octo Arc Prediction".

Lastly, while the filtering hence upsampling (or heavy oversampling) usually is just part of the filtering means (read : no matter what, getting rid of the harmonic distortion because of a too low sample rate in the source material always needs upsampling), it occurred to me already long ago that Arc Prediction Upsampling not only does that (the filtering), but also brings sheer higher resolution. And, at going from 176.4 to 352.8 it even occurred to me that for this reason "things" in the hardware (DAC) couldn't cope anymore, hence the sound got worse from it. So I changed those "things", and now it is better than ever before.

Anticipating on the resolution increase not being at its end yet, I already created Double Octo Arc Prediction in XXHighEnd.
Well, you can guess what that actually leads to ...

Peter

PS: I hope I didn't bring in too much of my own "stuff", but since I'm the only one approaching everything like this, I hope it is allowed without sounding like a real commercial. :)

Nice explanation Peter.


I think people will be shocked when they hear the NOS1 DAC with Quad (and greater) Arc Prediction. The benefits of this means of filtering is not subtle. It's totally obvious. BUT...

... it really requires a 24bit NOS filterless DAC to appreciate it. I have an oversampling DAC in my office system and prefer not using Arc Prediction at all with it. In my main listening room, I have a 24bit ADC/DAC that I believe is non-oversampling at >176.4 KHz, and I much prefer using QAP to anything else.

But this raises an interesting issue (interesting to me, at least):

Most recordings nowadays are made with delta-sigma ADC chips. What affect, if any, will this have? I mean, is the recording 'doomed' from the outset?

Any thoughts?

Mani.

Peter - Thanks for the explanation. It is all much clearer to me now (I think). If I understand correctly, the "filtering" function now occurs at the software playback level (as opposed to the DAC level).

I have some further questions, if I may. Can you please expand on why you say:


I am not sure I understand how multiple samples can mess up output.

Also if I don't hear much of an improvement between Arc Prediction / Quad mode and non Arc Prediction / non Quad mode, could it be because my DAC is filtering the output of XXhighend and making them both sound the same?

And finally I thought about Manisandher's question as well and it seems to be that the arc prediction + NOS solution is designed to replicate whatever is recorded on the CD faithfully so to the extent that the recording is not as good (whether because of delta-sigma or anything else), there will be a limitation, but an Arc Predicted / NOS DAC outputted sound should still be better than a filtered DAC output in any situation.

Admiring your quest for better and better sound.

Yes, indeed I prefer straight 44.1k without any upsampling on my OS AKM DAC.

Hi there Mani,

I must honestly say that not long ago I wondered this myself but couldn't come to proper conclusions. And this still is so : I just don't know - or can't reason it out. However, maybe it starts to occur to me that it is not just record companies (like HR) that produce better recordings because of their recording capabilities, but that it is "just" the Microsonics being used. Now it becomes difficult, because the one record company would be using the same "recorder" all the time.

Funnily enough I am a step further again on sound quality, and it (as it seems to me) unveils a character of sound from ... recordings with the Microsonics. It is no coincidence that I now can recognize this (again, as it seems) because I am concentrating on HDCD recordings the past few days. Just from random recording companies. Of course this *should* unveil a character of sound, because or I'm listening to undecoded HDCD, or I'm listening to decoded HDCD (hahaha) and both are not normal redbook. Ok, turning this some other way around, at listening to random (non-HDCD) HRx recordings, they are not a bit better, they are HUGELY better. Actually there's no comparison, and I don't think this comes from special mikes or whatever.
FYI: Yes, I am using *and* HDCD decoding *and* Arc Prediction (8x).

I guess it need some digging out (by me) on how A/D works in detail, and *if* it even is able to work with multi-bit only (yes, thus far I doubt that). Also, if the Microsonics would have worked with pure multibit A/D's I really wonder which that ever would have been for 192KHz (up to 2002). Did 24/192 multibit chips ever exist ? and what would have been the famous parts out of production so they had to quit the Microsonics production ?
(if true at all, because it also can be related to MS (eh. I mean M$ here) obtaining the HDCD license just before that time).

You know, this alll *is* interesting, because when I know how things are molested during the recording stage (if at all), I may be able to restore that ...
I know, this is highly opportunistic.

Peter

Ha Peter,

Good to see you followed up, scary huh
Was about to ask about your findings on this, but with HDCD.exe available i could only imagine it will be in the next version.
anyway, would be insanely great to find and restore studio/recording flaws, maybe impossible, who knows.
Something is going on, maybe these Pacific Microsonics are just right.

How to obtain such info about recordings ?
How many other recording (software, mastering) systems are being used elsewhere ?

after that it would be the case "How to recognize such recordings by software",


You are right about unveiling certain recording characters, because, well........ this is just the case.

maybe this is next step in the playback evolution of digital music !?
Who knows.........

With the common filtering many other samples (following and/or preceeding) "create" the current sample to be output. So, the other samples weigh in. Something like (very simplistically) the current sample weighs in for 50%, the next for 20%, the next for 18% the next for 16%, the last for 2 % (totalling to 100%) and there is the output sample. This is far from the original sample (which isn't good either), and the main objective one (like me) could have is that none of the original sample points are in the orignal sample anymore. This, however, should be looked at as a full wave of a frequency : none of the originally captured samples comprising that wave, are at their original time-axis location anymore. This can also mean the wave has shifted in time, and creates a phase error (which happens with IIR filtering and is called "minimum phase" as in "almost nothing is good anymore, haha). So, with proper interpolation (Arc Prediction is about interpolation), all the original sample points remain, and only new ones are injected.
Hoping you are not confused even more now, this is why Arc Prediction can only do "even" (like from 44.1 to 88.2) upsampling rates because uneven (like from 44.1 to 96) requires the original sample points to be lost.


This would be very true. There's a small mystery going on though : for most people with OS DACs or NOS/Filtering DACs, Arc Prediction still improves;
Although I can't reason this out for 100%, this has to be related to the smartness of the filtering algorithms, which can be highly based upon "is there anything to correct, and if not, never mind". At least for Arc Prediction it works like this, and any file already subject to Arc Prediction (might it exist) won't change a thing to the wave form anymore. Thus, IMO it depends on the filter used, and I don't know them ...


Mwah, a little different : The point is, what is recorded on the CD is wrong always, but it is the trick to bring that back to what was recorded (the nice sine etc.).
On a side note though (and this might be what you mean), if I record the output of NOS (with or without Arc Prediction, but filterless) and digitize that again (capture by means of an A/D converter), I can very well compare the CD data with the output analogue data coming from that. There's still a huge error, but it is comparable. If I do the same with an OS/Filtering DAC, no head nor tail can be made from it. It has become COMPLETELY different data. It is even a miracle that we perceive music from it. And do note please that I created the comparising software for it (actually in XXHighEnd but disabled -> the Analysis tab), and what I do in there has not been done before (and is thus unknown to "the world"). If you'd look at the graphs coming from this, well, then you know at last and for sure that not only a zillion things are wrong with audio, but that as many things can still be improved just because of that ! Isn't that a nice future ? (in case I'd be running empty of work, haha).
In the end, although nobody using good equipment with a nice OS DAC will believe it, this is exactly what you hear/perceive from OS : actually nothing much. It may be nice sound, it may be not fatigueing, the background may be nicely black, but what are the instruments you are hearing ? It just doesn't work. And if you'd see those graphs, well, you just can see why. There is no comparison with reality, that reality being the CD data, though (and I can't emphasize this enough) also wrong. For fun, do this :

Draw one cycle of a sine on a paper. The line starts at 0V, the first peak is at 2V, next it crosses 0V and dips into -2V and it ends at 0V again. So, this is one cycle of a frequency. Now assume a samplerate that catches this sine with 4 samples, and divide those samples nicely on the time line (horizontally). Make a sample where the sine starts (0V) make one at the first peak (2V), make one where it crosses 0, make one at -2V, and lastly make one where it ends, which is the same as the start of the next one. Imagine the volts (which produce the sound) jumping up and down, which is nowhere near the sine you drew. But it it not about that ... What is, is this :

You assumed the sampler starting at the starting point of the sine. But why would it ? So, assuming your drawing is 10cm or 4 inches wide, create your first sample point 1 cm to the right of the original one. Shift the other samples 1 cm to the right similarly. Now, do you see what happens ? you have the same base frequency, but the voltages (on the Y-axis) have become totally different. And you bet the tone coming from that is totally different as well. And the only thing what actually happened, is after the A/D recorder started recording, the pianoplayer started 1us later playing opposed to your first "recording" (sample drawing). Now, if the piano player would be able to play very consistently (hence the time difference with the sampler wouldn't change all over the track), the sound will be TOTALLY different. Same recording engineer, same piano player, same recording device, same everything but on thing not under control : time shift compared to the start of the sampler.

In either case it is the intention of it all that your nicely drawn sine is reconstructed. You know, that sine which already isn't in the CD data anymore ...

Never minding Arc Prediction, now think again about the bunch of next samples contributing to the current one to be output like normal filtering does. Now think of not the pianoplayer starting to play at a fraction of a ms later, not the sampler starting later etc., but your EAC rip not starting at the offset as "intended". Let's say the offset is one sample later only. Look at your sine again. When it starts at the right place the first value to calculate with is 0. But one sample later it is 37767 (as the digital representation of 2V redbook). Ouch. What will that do to the calculation of the next output samples, THROUGHOUT the track ?? it will all sound different !

Things may be not as black and white as I suggest, but they all influence ...

Peter

FYI,Did you know there is software to restore those drive offsets in flac files ? (to AccurateRip compliant)
Maybe do some testing
It can also recognize HDCD's and many more, love it !

Its called Cuetools 2.0.6

(http://img684.imageshack.us/img684/8392/deskq.png)

Hi Peter,

Here's what Michael Ritter (of Pacific Microsonics) wrote in 'Mix' magazine in 1999, before the Model Two was released (the highlights in bold are mine):

"The actual A/D converter in the Model One runs at 24 bits and 176.4 kHz currently; the Model Two will also convert at 192 kHz. We improve the linearity of our conversion with a high-amplitude broadband dither signal that we mix in with the program in the analog domain. The dither appears to be random, but the system knows at any given instant precisely what the amplitude of that dither signal is. And because we use our own custom, discrete, full-ladder converter with excellent amplitude and phase accuracy, we are able to apply an 'anti-dither' signal, exactly out-of-phase and matched in time, in the digital domain after conversion. That nulls the dither noise out of the signal.

If it's going to be a 176.4 or 192kHz DVD-Audio release, then we will not decimate that signal; we use a proprietary filter [non-oversampled] optimized to that sample rate. If it's going to be 88.2/96 kHz, we use 2:1 decimation, and once again we use a filter optimized to that frequency. But in both high-resolution settings, the Nyquist frequency is high enough that we don't use the 'dynamic decimation' process that becomes necessary when we go down to 44.1 or 48 kHz."

Note that there may be some confusion with regard to the editing by Mix magazine. Ritter mentions a 'proprietary filter' and the editor has included '[non-oversampled]'. I'm not sure how accurate this is...

HTH.

Mani.

Yes, my experience too. But I think you need a 24/176.4 NOS DAC to appreciate this...

This weekend, I'll try to find some time to make some recordings from vinyl for you - 16/44.1, 16/44.1 with peak extension, 24/176.4 and 24/192. You won't like the music much (it'll be classical or jazz - that's all I've got on vinyl), but hopefully it might help you understand what's going on a bit better - or at least eliminate Keith Johnson's recording techniques from the equation.

Cheers,
Mani.

But key is the "full ladder" thing (which I missed thus far in my readings). So, I will believe that, but which chip was (at that time) able to do it. So, this goes back to before 1999, although possibly 176.4 only for that chip. The latter makes it kind of suspectible, because (at that time !) nobody was dealing with 88.2 or 176.4. Of course, the useable derival of it, 44.1, was.

Hmm ... Did you actually ever try to open the box and look for chips ? (I am not saying you should do this now on behalf of this little discussion !). Or, if it's easy to open the box you may take a photo of it and send it to me (maybe by normal email because of who knows what's an actual secret in there).
But again, you may have better things to do !

Thanks,
Peter


PS: About your recordings ... mwah, that might lead to something. But I don't know yet. You may leave 192 just as well out.

PPS: It intrigues me. I just won't believe Keith stopped all this because of parts running obsolete. Just as that I won't believe it will take a million to restart production with the currently available parts. Well, let's say that I didn't consume a million yet. Haha.

I haven't, but Romy has! Well his Model One at least:

http://goodsoundclub.com/Forums/ShowPost.aspx?postID=6566#6566

Not sure if you can make the chips out though...

Mani.

I'm not sure if this is relevant, but Ritter talked about the 'upcoming' Model Two back in 1999. At that point, only the Model One was actually available. However, my particular Model Two was one of the first to be built... and that was in 2003!

Perhaps they were waiting for commercial 24/192 ADC chips to become available for the Model Two? However, Ritter talks about 'custom, discrete, full-ladder converters'...

Unfortunately, I'm not willing to take my unit apart to have a peek inside those PMADC-1 modules and see what chips they're using. Sorry!

Mani.

Peter,

Won't a lot of the things you describe as problems with OS dacs and advantages of NOS dacs be masked by the following electronics / speakers anyway?

Per

Hi Per,

Yes, of course. But I guess this is about those zillion things being wrong. I mean, the data coming from a DAC (and also the software as, we know now) can be so totally different (over setups/designs) that it easily overrules. I also could say : you have no idea what can be squeezed out of your loudspeakers. So, also the lousy ones.

If you take for the example the relatively very tiny influence software can have (remember, bit perfect), but eveybody hears it through no matter what speakers and further system, well, then you know the truth of it.
Fact remains that that too (speakers and everything) matter a LOT.

I don't want to tease you all, but I think I have entered the stage which might be a kind of end stage. Btw, important to know is that everything I actually (or really) did was about speed speed speed. Nothing else. Thus for example, not a high quality well known whatever amplifier, but a very fast one. Now :

I don't know if someone has the OST version of The Wall, but yesterday I played it for the first time, and (thus) I have no relative judgement of it. Man, this was totally creapy. This album is so much ful of "real life" sounds that in the end I got tired from it. This starts somewhere (IIRC) at the second track with someone whisteling. I looked in the room, looked again, looked at the speakers, and I just refused to believe this was not someone in the room whisteling.
Well, we all have this experience sometimes (the crying baby is a good example), but now it never stopped. Really, it tires (but with a happy face).

Later throughout the album my wife walked into the room, and I asked her "am I crazy, or does this all sound so realistic ?". She misinterpreted the question and thought I was talking about the performance of the system instead of the album. Her response : "it sure does. You know, lately, when I'm sitting outside, I'm constantly thinking someone inside is talking to you or anything" (outside is with the doors closed btw).

At a certain stage I thought "this must be the best recording I ever heard", which is total nonsense of course (unless plainly true, which theoretically can be so), but then the conversation I had some 2 years back with Peter from 2L (recording engineer) slipped through my mind; he told me something like "you don't want to know, but at 352.8 things suddenly work". So yes, I was using 352.8 indeed, though upsampled ...

All 'n all I can't be sure what will be happening to your systems, but if I look at all the improvements I could create for myself (with a. XXHighEnd and b. the Phasure NOS1 DAC) it should be so that you will be totally shocked by now. Maybe a slower speaker or a slower amp will debet on it, and maybe it *requires* the fast "back end". We'll have to see that. But I tell you : this just starts to be unbelieveable (indeed says me myself and I :)).

Peter

Thanks a lot for your reply, Peter. VERY exciting indeed...

I must say that judging from what you have done with the software / computer side of digital playback with XXHighEnd I think we all might be in for some real (pleasant it seems to) surprises with your dac - IF your ideas and electronic artistry works out in real life with different setups judged by different users, that is.

Can't wait to hear the feedback from the first customers of the PHASURE NOS1 DAC. Might turn everything upside down in regards to building a highend setup. Pondering about some things / questions but that is for the ideal amp / speaker setup thread http://www.phasure.com/index.php?topic=1182.0

All the best,

Per

Peter

In relation to measuring the noise in your DAC project you stated that noise would be (at least 3 dB) better if measured with balanced lines.
I take that to mean same signal from the DAC's digital section, but processed through the balanced section of the analog stage in your DAC, rather than processed through the unbalanced section and measuerd at the unbalanced analog output (as in your posted measurement pictures).

In other words, you are not speaking about the difference in the DAC's performance when fed a balanced source (digital IN) signal versus an unbalanced source (digital IN) signal. 

Would be interersting to hear your thoughts on feeding unbalanced versus balanced digital source signals to an external DAC - in general as well as related to your DAC project (listening and mesurement findings).

Best regards

Bjorn

Hi Bjorn,

Ok, after taking some deep breaths on how to answer this, here is my well meant attempt :

First of all (don't get scared) : There is no such thing as balanced or unbalanced digital-In in the Phasure NOS1. It is direct i2s (you will learn how, later) and no such thing as balanced/single ended exists there.

But in general the answer could be : it may matter in jitter because a balanced connection will eliminate noise, if there in the first place. Notice though that any normal means of us consumers connecting DACs, such noise won't be there. Merely think about pro-use, and hundreds of meters digital connection.

The real merits of the base of your question is different though, although you will not have noticed that yourself; :)

When I told about this noise, and better with balanced etc., this is was not about the elimination of (possible !) noise in the outgoing interlinks (or otherwise I don't know what text you are referring to), but merely about this :

The DAC chips themselves (and there are 8 of them) for the half of them are lined up in balanced (differential) operation. This means that what's fed as positive voltage to the one chip, is fed as negative to its counterpart (just like balanced line). This eliminates a type of distortion which may occur at one side of the 0 voltage line (think like negative voltage creating another amount of (inherent) distortion than positive voltage). It "balances out". Thus, whatever the text is you refer to, this will eliminate distortion to begin with.

While this is one half of the line up of the DACs, the other half is setup parallelled. Notice this was created later, and wasn't so at the beginning of this topic. This causes the output to be higher in dB while the "PCB-inherent" noise (not distortion this time !) keeps the same. Thus, relatively the noise drops.

Now, the first setup of the DAC had a choice for Balanced and Single Ended (unbalanced), and it was a 4 chip version. As per my explanation above, both imply something for distortion and noise respectively. It was *or* balanced *or* single ended;

The current version is somewhat smarter :
It still has the choice for balanced and single ended BUT with the balanced setup the parallel configuration is also at work (this is because only 2 chips (per channel) are useful at creating the differential (balanced) setup; 4 don't add anything more). So, now there's less distortion *and* less noise at the same time.
When you choose for Single Ended, all chips are parallelled and there's eve less noise (but not as few distortion as possible).

Well, at trying to be as vague as possible (as it seems to me), you must also think about how you use the configuration options. I mean, while there are actually two options to choose from (balanced/SE), in either case you can use half of the output lines vs "added". Notice that "added" is wrongly put for Balanced (you'd just be using a negative and a positive line), but looked at it the other way around : you can use half of the balanced output for a SE connection, and you can also use half of the SE output for a SE connection. Although this just works perfectly, both these options are actually stupid, because they produce half of the possible output, either with unnessecary distortion or unnessecary noise (if you could ever follow my above story haha). BUT :

It produces less net output towards the main amps just the same, so, if indeed working without preamp, it may just give you the attenuation needed, not causing to loose one bit of resolution (6dB) otherwise. Otoh ... I don't think there will be a switch for this option (other than internally) because that switch would be in the signal path, and I don't like anything in the signal path which is not really necessary.
So, the story about it was merely to support the way it works and to hopefully make it more understandable, than that it will be used in practice.


Lastly, because this is as important I think :
Notice (from above story) that the balanced output is nothing which is created in the analogue domain (hence no transformers or anything). Instead, it just starts in the DAC chips and from there goes (via the analogue stage) direct out to the XLR pins.
:secret:

Peter

Peter

Thanks for your explanation - provides a good outline of technical side of the operation of (earlier and latest version) NOS1.

Let me just state my understanding of these operating principals and how you intend to implement those in the NOS1, and you can correct me if I have misunderstood.

1. Switching btw Balanced and SE operation.
The switching of internal operating mode of NOS1 btw Balanced and SE. In other words, not a switching btw Balanced vs SE digital input.
In effect, the DAC does not even need a Balanced input as as a balanced +/- digital signal can be generated in the DAC from a digital SE(+) IN signal.

2. Balanced vs SE output
Balanced vs SE output is linked to the selection of operating mode; i.e. Balanced (XLR) output only active when Balanced operating mode selected and SE output only active when SE operating mode selected.

Bjorn

Bjorn,


It is hard to explain (further), because the phenomena just are not applicable ! But, the operation of the DAC makes you confused, I'm sure.

Think about a pre-amp. Does it need balanced inputs of some kind in order to have balanced outputs ? no, because it is totally unrelated. But :
*If* a preamp would have balanced inputs, I'd put that through all over (doubling really everything for parts etc.) and you could even have separate power supplies for the plus and min part (and notice that you'd need plus and min power rails anyway, but they usually come from the same PSU).

With more DAC chips you could do anything, and you could even let one chip deal with the plus voltage (from the music data this time !) and the other chip with the minus voltage, while at the end they come together again as a normal signal (with +/- to drive your speakers). If this were necessary it would have been done, but the PCM1704 does that internally already ! (eliminating cross-over distortion).
So, the useage of more chips just allows the trick for balanced operation to be applied, and it is totally unrelated to the input. Also notice that if the input were balanced, this is just an analogue thing (no matter it is carrying functionally digital data). Within the DAC it is no analogue thing at all, BUT since there are two outputs (separated in plus and min) you just as well can let that continue in the I/V conversion to the outputs (where it has become pure analogue). Thus, a trick which avoids the transformers normally needed to make a SE analogue signal balanced.
Less = More --> Less = Speed.


True. But by the grace of me not applying something strange, like using half of the balanced output only, making that SE again.


I'm afraid this didn't help much, but keep in mind it is all about technical decrease of distortion and noise. With technical I mean : not subject to coincidental picking up of noise which would be eliminated by balanced cabling. Instead you can just calculate on it with this as the best example :
When the inherent noise of the DAC board (think PCB) is -120dB while the output harmonic distortion of the DAC chips (and all) is -130dB THD+N(oise) will be -120dB. When I'm able to increase the output of the DAC chips (and all) with 3dB because more chips contribute to the output (which is current at first), the THD+N will be -123dB. Why ? because the signal got higher but the (PCB) noise remained the same.
Nothing much different from the good old analogue stuff, and louder recording implied less noise (S/N ratio !). -> The recorded signal is higher but the equipment and tape noise remain the same.

The above was about the parallelled part of it, which is there anyway (4 chips). So as said, when all is parallelled (SE output) S/N ratio gets better again, but THD gets worse.

To finish this off, when the noise floor is above the HD floor, HD doesn't matter anymore, because it disappears in the noise. But when you improve the noise (lower the noise floor) HD get apparent. All 'n all, one of the tricks is to have both at the same level, which is "efficient" so to speak (hence, why improve further on HD while you can't hear it anyway because of the noise).

Oh well :)
Peter

a-a! This is a false belief.
Our ears are able to listen 10 to 20db below the noise floor.
To measure HD under the noise floor the notch filter trick can (and should) be used.

So, in first instance the noise floor should be very low (and <120dB IS very good for a low-level signal), then the harmonics should be another 10db under it.

Haha, yes I agree. But it is difficult enough already to explain it all (from the angle we talk about here).

But it is also about practice. I think I told this before :

Take a 1 bit recording (RME has (or had) such a file for download). This is -96dB. Now try to set your equipment so loud you can hear the music play. The very first you'll hear (in a random system) will be a lot of noise, but you will be able to hear the music playing as well (unless you really have too much noise). This proves it.

Sorry for the wrong suggestion !
Peter

FWIW, it looks like the DAC section uses UltraAnalog chips, bearing a striking resemblance as they do to the older 20bit chips used in the Spectral SDR-2000 (compare the attachment to pics on Romy's site; http://goodsoundclub.com/Forums/ShowPost.aspx?postID=6566#6566). This isn't surprising seeing as Keith Johnson (one of the designers of the Model Two) happens to be the Director of Engineering at Spectral. I'm not sure about the ADC chips though...

Looking at the specs for the SDR-2000, it seems that the people at Spectral share your obsession speed...

Mani.

I must admit that those boxes look veeery similar (while it should be a cover for the real chips only) although they seem to be not equal in size.

UltraAnalog ... hmm ...

Hi Peter and All

Finished reading the tread.  Wow what a lot of info.  Will need to read it again and maybe again again to fully glean all the stuff it stands.  But right now for me I still don't understand how you get the 384/32 into the DAC?

Thanks
Bill

Funny Bill ... but you REALLY are the first one to ask. I guess it is known that I don't tell everything which is regarded a secret (propriatary) in the first place ? Or everybody thinks this is no problem ?

Your number is on the low side though. :swoon:
Peter


PS: But 24 bits !

Cant wait to hear what you can divulge - it really has piqued my curiosity.

Thanks
Bill

I think I know. But I wont tell :grazy:

Fair enough.  But that does still leaves another issue.  XX has an option to select 384/32 as your dac.  My understanding is that the windows drivers doesn't support that high a transfer rate.  How can you use that kind of dac?

Thanks
Bill

Of course, many people are controlled by their computers. But I learned to be in control over them machines hehe

You can't guess even half of what all happened ... :)

Peter

The thread has earlier touched on the possibility of digital filtering (in software) as well as the possibility for the DAC to perform the ADC of the analog signal from a turntable.

From that perspective, I am interested in your thoughts on the pros and cons of applying the RIAA filtering in the analog vs digital domain.
From a practical perspective, I assume that for implementation of digital RIAA filtering  in XX software, analog signal (from turntable w-out analog phono stage) would first need to be fed to the DAC for ADC conversion, then digital signal fed back to computer for RIAA filtering in XX software and finally digital signal fed back to the DAC for conversion to analog before signal OUT fed to the amp.

In general, if analog signal from the turntable (w/out analog phono pre-amp) is too low for processing in your DAC and some form of phono pre-amp required between the turntable and the DAC, analog RIAA filtering may just as well be performed in the phono pre-amp (and forget about RIAA filtering in the digital domain).

Best regards

Bjorn

Hi Bjorn,

A not so easy question (for me :)). Why ? well, because you are (I think) using the digital RIAA correction at real time playing vinyl, and the (obvious, to me) reason to do it is to improve on the phono stage. I don't know ...

I don't know because it needs the experience, and I guess I won't ever have that, lacking a good turn table. But what I can reason out (who can't) is this tradeoff :

a. Play vinyl as analoguely as it is, with the best phono stage to your knowlegde;
b. Record vinyl to digital, and play it back from there with an undoubtedly better RIAA correction (I'm rather sure).

So the point is : what harm does the transfer to digital ?

This question alone is a can of worms by itself, for which one has to be thinking quite controversely. Ok, I can do that and I will, but in the end it doesn't make sense I think. Look :


Whether it is a digital transfer of direct cut vinyl (the guys doing that CLEARLY don't apply stupid "corrections" as common todays digital mastering does), or whether it is an uploaded "vinyl rip" from someone (those seeming to have good turntables in the first place) ... the files coming from that to me CLEARLY sound analogue. Thus, what I'm actually saying is : they sound better than random good recordings from a CD.
Of course, those vinyl rips will have been done at 24/96 or so, but the direct cut etc. stuff I talked about is not (just 16/44.1) and the character of both is very much the same CLEARLY, to me.
Notice that this tells at least me that digital mastering is flawed (generally).

But what to do with this ?
Will it mean that when I play back my vinyl through my modest turntable and through digital RIAA correction - through XXHighEnd, it will sound better than directly via the phono stage and without transfer to digital ?
or
Will it mean that when you play back your vinyl through your super turntable and through digital RIAA correction - through XXHighEnd, it will sound better than directly via the phono stange and without transfer to digital ?
That would be odd if so.

But sadly this is about a heavy case of apples and oranges because of the two variables in both equations : transfer your nice analogue to digital on one hand, and the digital RIAA correction on the other.

So ... ONLY when the digital RIAA correction is very much profoundly better than analogue to digital transfer destroys, there is a (very) good reason to play back vinyl this way. And I don't say this won't be the case ...

I can add to this, that IMO (no, plain experience) there really is nothing wrong with digital 16/44.1. In other words, assuming the A/D conversion is performed allright, it just doesn't harm at all. I also derive this from rather random "vinyl rips" I own, and not one of them sounds worse than normal digital to me. And worse ... they sound more analogue ... it should even happen with my modest turntable ...

When you read back the above (and assumed my own observations are correct) there would be one logical conclusion only : it is the RIAA correction in the analogue domain destroying the stuff. Thus, as long as nobody tells me where my story flaws, we should indeed make the provision. It should even be able to "bear" the randomly modest A/D the DAC theoretically has (practice when I provide the connections at the back of the cabinet), because remember, it is a DAC; nothing has been done to let the A/D excel. One little iny whiny exception to this : the A/D will be able to record in in 24/384 ...
:swoon:

Yes, this latter may make you laugh, but I never thought of the combination until I wrote the previous alinea. So I'm at least laughing myself ! Rather outloud !
Hmm ...

It would be quite a joke actually;
No filtering whatsoever would be needed at the recording stage, and no filtering (like Arc Prediction Upsampling) would be needed at the playback stage either. No dither anywhere, nothing. All will be as HD free as the equipment (turntable, ADC, DAC, amp) permits, and you could well say it should be the best "vinyl rip" we've ever heard (unless people are used to listen to 24/384 takes from vinyl right now, which at least will go through OS DACs, so not the best at all (remember, my view)).

I still can't type of laughing ...

Well, at least this has made me decide to activate the recording capabilities in the DAC, which so far seemed an unnessecary thing to do (because it is just there, but not explicitly made on par with the D/A capabilities (for SQ)). But if *this* comes from it for the better, I just should do it.
As I (I think) told earlier somewhere, the recording capabilities are in XXHighEnd, and they are real time. Thus, exactly what we need here. The only thing it needs is the RIAA correction in between ...

but

This will not be conform the way XXHighEnd operates normally, because the RIAA correction now needs to be performed in realtime too. And no such process exists at the moment. So, SQ itself will degrade because of that, but YMMV to what degree and the net result (compared with direct analogue playback from the turntable).


Keep those ideas coming ! It only makes the DAC ready later again. :oops:
Peter


PS: ... which really seems to be a problem for me; although this improvement of the "DAC" is in a complete different area, I (or we :)) keep on having ideas to improve the DAC, and it is REALLY worth while !

PPS: I didn't tell it yet, but the DAC is in production for two weeks now, and the ideas I keep on having are all at the back end, so to speak. So, they don't hold up current production, but until everything is assembled into the cabinet, at that stage things can still change. Allowing input terminals is an example of that, but there is more.

Dear all,

A few -I think- important milestones have been achieved, so I thought to inform you about it.
Hopefully it also shows better why things seem to take a long time, while it's really worthwhile the long waiting. And, I'd hate to pronounce improvements later, while some of you may have a "less" version in the mean time. So it better be right the first time !

Allright. Starting with what I mentioned in my before post, production really has started a few weeks back. A 100 units will be produced as a first batch (I may sell 10 only haha), so hopefully enough people can try the DAC (money back guarantee) without being in a long back log while others already have it. Thus, only when the 100 are ready, the first will ship.

In the mean time, this will give me the "opportunity" to wait for some parts with a long lead time. This just is so and can't be helped, but for some parts only the best can be used, and they are produced on (my) order. Currently the longest pending lead time is 8 weeks. :sorry:
But, these are parts which can be assembled quickly, and the assembly won't hold up production really (the waiting for the parts does).

Although you didn't know it, only yesterday another technical highstand has been achieved, and I must say this is about one of the most nice (and difficult !) coorperations not in the same office I have ever met in my life. Think of hundreds of emails about a certain design at the bit level (this is literal), that in the end needing independend programming at two sides, one side being XXHighEnd outputting data, the other side being CPLD programming, which is actually in-DAC and at the engineering level. And, after 6 months of careful preparation, yesterday it worked in one go (neither side being able to test without the other's efforts).
Some know (via offline) what this is about, and actually it is about creating some kind of "double buffer" to be 100% sure it can't be done better. It is about the 24/384 input of which we can reason that better (higher sample rate) isn't necessarym but because the technology to go further was there in (my) theory in the first place, it just was done : 24/768. Mind you, this is for input, and no master files exist for it.
However, since Arc Prediction Upsampling turned out to be a quality phenomenon by itself, it *will* be useful for just that : sound quality.
On this matter, please notice that the step from 24/176.4 to 24/352.8 (with a source of plain 16/44.1) already brought a significant improvement, so stepping further to 24/705.6 most probably will again (I couldn't test this myself, because I don't have the hardware for it yet).

By pure coincidence, also yesterday I was able AT LAST to apply a gain "mechanism" with THD+N specs better than the DAC chips itself, or in other words, which does not degrade sound. Maybe, just maybe I have to come back on this later (because I don't like the sound of it afterall), but chances are very small because of the enormous improvement ot brought (see below story).
This must have been the most time consuming of it all, and in the end is about a decent I/V stage which most probably is the most discussed subject in the DIY community on the Internet. This is also about my earlier expression on "you have to live with 24dBFS under 2VRMS" (which by itself is very doable with enough gain in the main amps and/or enough efficiency in the loudspeakers), which now is just 2VRMS (which is the standard). Also, this will allow for headphone output, although in theory only (read : this has not been tested yet).

To my own surprise, the at last working of this, brought a new dimension in music reproduction through loudspeakers for me. If I had to rate the change in SQ, I would say "10 times better". Again ? yes, again. Look :

People who are here from the beginning (or those who ran into the post(s) concerned by accident) know that I have always had one major negative remark about music reproduction through loudspeakers : the volume level of cymbals and the like. And since we are talking about it anyway, you may come to the same conclusion while I put your attention to it :
Go to a live perfromance, and have the bass player, the singer, the piano player and the drummer (etc.) in front of you. Now put your attention to the balance of all the sounds *but* the cymbals on one hand, and those cymbals at the other. What actually sounds the most loud ? It is the cymbals !
Now go back to your listening room, and try to perceive the same balance. No way it is going to be met. Ha ! you are even lucky if you perceive (real) cymbals in the first place !! ... so bad it is ...
On this matter, maybe three years ago by now, I tweaked my crossovers so I would have 16dB more output at 20KHz, with a nice slope starting at 5KHz and up (to that max at 20KHz). That really helped a lot, and was actually the first time hi-hats became nicely profound while before I never heard them as an explicit instrument playing (while it is the most important instrument for a drummer).
I never removed that tweak up till today (and I still won't). So, although this may sound wrongish to you, it is my opinion this is just needed in order to receive a more or less representative playback for cymbal like instruments. Also, this can't be blamed to our ears, because I really don't have a problem with the drum kit in the house here. The live cymbals perform perfectly right and they really don't need cranking up.
So what is going on really ? And to remember, a self respecting loudspeaker just measures flat.

I think I now found the answer in a (to me) new phenomenon - or dimension if you like : The SPL of high frequencies.
Oh yes ...

I am quite sure (not 100%) that most of you look at SPL (Sound Pressure Level) as a phenomenon which occurs in the lower regions (I mean under 5KHz or so). SPL is equallish to sound pressure you can feel, and it comes from base drums, general low frequency sounds and maybe ar loud singing voice. But not from something like a tambourine. No ?

Ha, come here and be startled. And this is exactly why I rather call this a new dimension than just a phenomenon. It makes your whatever music completely new. There's a whole other band playing besides the one you were used to. It is the drummer with his gags.
Btw, I must admit, the mid level increased with it, and those sensitive to that will be quite happy just because of that alone. But that's another subject (as important btw, but not as a new dimension as such).

Quite a few virtues spring from this, like really fresh sound is one of them. So, people who tend to talk in terms of removed blankets ... here you go ! it is really amazing.
But what actually happened ?

To be honest, at this moment I don't know yet, although some of it could be predicted from earlier attempts in a similar setup. I think the other day I talked about cymbals sounding for 20 seconds instead of the "usual" 5 seconds, which I remembered from this earlier setup, though not good sounding. So, it seems logic that (relative) additional amplification of the high frequencies make jump out a cymbal for 20 seconds long, while before it could only jump out for 5 seconds. It was there for 20 seconds allright, but the sound of the piano etc. made it disappear. Not so anymore. And, obviously along with it goes the as louder attack. So, just louder (and not a bit, but severely more).
If I had to explain it from a technical point of view, I would say this :
The setup from before was about speed. You have read about it, just explicit speed. But since this is a.o. about leaving out parts, it is also about leaving out "drive" (notice : in my case, and of what I can reason only after the happening). Now, we tend to derive drive from low frequency sounds. In other words, I think it is recognizeable that when we don't have enough drive, the sound gets too thinny and the bass disappears. And, since this was not the case in my setup, I had enough drive. However ...

A low(er) frequency sound moves slowly for its up and down (plus and minus voltage) wave. It is not difficult for such a wave to go down again after it reached the (volume) top, because it is slooooow. It needs power (yes, drive) to go up and down, and it even needs additional power when the frequency is low because it has to move a lot of air (simply said, and expressed in the wrong domain perhaps), but what about the high frequencies which have to be pulled back in time ? So, a high frequency may not need much power to move the air (which is related to the directionality of it !) but it still needs power to retract from its top in the wave back down to its low side, etc. And *this* goes 1000 times faster for 20KHz opposed to 20Hz. It may even need more drive to control it, and in the end is similar to reracting a loudspeaker diaphragm from its excursed state back to the other side. If this would be left to the mechanical properties of the diaphragm it would be way too slow ...
Further, try to imagine that everything which is too slow to cause that reraction, will cause peaks not to be reached (because the power to create the peak is slow, the creation of the peak is behind, and before it's at it's top it has to go down again ... and peak = SPL (volume)).

The above summarized :
While I had created a most speedy setup with lowest distortion figures possible, apparantly (!) this setup didn't contain the drive to perform in the high frequencies. Notice also that without a speedy setup, the higher frequencies don't perform for the same reason (all is too slow again, read above alinea). Now though, I was able to create a situation with the same speed, but with the power needed. Ehm, apparently, and as long as my own reasoning will stand.

Still here ? well, get intrigued on the next then :

I have always been shouting that my cymbals improved and improved, and whereever I go they are not there at all. But, whenever I said that, I also said (a bit depending on the stage I was at) that they could be improved still. One time they are too plastic, the other time they hiss too much, and the next time the color ain't exactly right. But please imagine, this always was in the "environment" of the cymbals being too much in the background. And as I said in the beginning, many of you will be glad to perceive them explicitly in the first place, although you always will be hearing that they are there. Also, even with my 16dB increase of the higher frequencies, it was allright for me, because they were not too profound (as said, still in the background). But hey, not so anymore !
And so you can imagine that I created myself a huge problem, because at any smashing cymbal in your face, they better sound real !!

:oops:

Thus, in the very end I may be much excited what has been achieved on this, and it may sound the most interesting to me at the moment, but the next thing now is to get it completely right. It needs much more albums to play to perceive the real merits of what I just said, but I expect some problems here. One thing : I already know that the cymbals sound way better than before for their colour, so nothing turns out to be worse opposed to before. This means that the route itself is the right one which is important for the need to (not) undo that. It is to be awaitened though whether I can "hear" where to tweak further, but hey, I have some 8 more weeks for that (according to the beginning of this post), because this too is back end stuff.


Last thing for now :

Currently I am working on the rather tough decision to lower the jitter which officially is under the audible level already (4ps RMS max);
The reason to do this is an attempt to remove the influence from software (players). This anticipates on software influencing the jitter level itself, which by that means reaches the audible level (like 100ns or whatever). Notice that this possibly happening is only a thought and can't be proven (measurement will show just 4ps, but measurement implies a fluent load on current and is no music (with varying load)). The idea is that implementing a much lower jitter clock (if possible at all) will let rise the jitter again during music playback, but not above audible levels.

So, *or* I may succeed in finding a lower jitter clock solution and if I'm right we can throw out Q1 with unpredicted result (will SQ now always be at its best ??) *or* I let it be deliberately as it is now, so we can keep on using XXHighEnd as we are used to (and have fun with !).

The latter seems stupid to do when the first would be feasable. This is why it is a tough decision (and this too is a back end application).
Notice that with the current (4ps for net measureable result) XXHighEnd influences as much as ever before.

So far for now !
Peter

Good morning Peter,

Don't forget my almost grey haired position on the waiting list for the DAC   8)

Great news that the production has started,

Leo

Only one thing.


So, there will be a volume control? I dont remember if you canceled this option or not.

The headphone out would be VERY VERY handy. Lots of people like to listen with those (not me unless i'm working), but also allows for an easy comparison with the normal system (amps and speakers), which would be VERY handy for me :)

If I stay with this option, yes, there will be a volume control. But ... only when it doesn't degrade. If it does, it just isn't a good idea. It shouldn't degrade, but the setup is a little strange. So I'll have to measure it. When I have done that, I will let you know.

With remote? :)

<-- lazy.


OK, thanks. You know how eager i am to listen to the "really finished" DAC :)

Peter

As you invited comments, let me expand on the issue of playing analog (vinyl LPs) and RIAA conversion through XX software and NOS1 DAC.

First, my thoughts to tranfer my vinyl collection (vinyl rip to 96/24 and save files to disc) was more driven by the convenience factor; with vinyl in digital domain benefits would be
a. stored in central digital music library and easily accesible (togteher with CD rips and digital dowloads) via computer network to play at any location in my home
b. ability to play vinyl tracks in any order (from one or several LPs)
c. possibility to play through XX and take advantage of its superior abilities

Thus, the above driving the decision to look into option to transfer vinyl into the digital domain, and the discussion of better to apply RIAA in the analog or digital domain stems from that.
In other words, transfer vinyl to digital has benefits over and above the advantage of possibly applying RIAA filtering in the digital domain.

I think useful to outline three different categories of LP listening/ digital conversion using XX and NOS DAC

1. Playing LPs (in real time) with ADC conversion in NOS1 and applying (real time) filtering in XX.
  The reason for vinyl analog to digital conversion is presumably to take advantage of XX digital filtering capabilities; if not for RIAA filtering, then room adjustment and/or x-over filtering. In fact, I see no real separation between digital room adjustment filtering and RIAA filtering. In other words, would it not be feasible for a system set up to use a digital room adjustment filter in XX to have two separate filters – one room adjustment filter for regular digital (CD) files and the second the same room adjustment filter with the RIAA curve imbedded (rather than two separate filters) for vinyl playback.
Also would be great to have NOS1 capability to pipe out the signal both in digital and analog simultaneously; i.e. while playing vinyl and listening to the NOS1 analog OUT signal, the NOS1 digital OUT signal could simultaneously be routed back to the computer and saved. This way, vinyl ripping would not be a separate chore, but a by-product of listening to LPs (optimal if option provided to save the digital file either with digital RIAA filtering imbedded or w/out RIAA adjustment, see below).     

2. Playing a digital file converted from vinyl, but stored on disc as a RAW file without imbedded RIAA filtering.
The file would be subject to digital RIAA filtering in XX when played. However, as the RIAA is not embedded in the file before storing on disc, playback of the file would benefit from continued improvements in XX digital RIAA filter capabilities. (Similar idea to RAW file format for photos).  Playback would require XX (for RIAA filtering), but not necessarily require use of NOS1 DAC.

3.  Playing a digital file converted from vinyl, and stored on disc as a file with RIAA adjustments imbedded (as a result of either analog or digital RIAA filtering).
As for any other digital file neither use of XX or NOS1 DAC strictly required.
However, would be great to have XX capability to upsample a stored 96/24 file to 192/24 and 384/24, the latter presumably requiring NOS1 DAC.

Bjorn

 

Oh, I have another request Peter, if it's not destroyed, leave a spdif digital input - thinking to connect my tv.

I live
to
 :drool:


This would be nice ... if it doesn't hurt.

Peter,
It is still not clear to me how I should connect the NOS dac to my computer.
Do I need to prepare my computer in some way (hardware, connections?).
I understand that I have still some time to do that  ;)
Cheers, Eric. 

I want to know that as well.  I asked a few questions trying to get to the bottom of it but never really got a reply - which is quite understandable of course since it probably touches on proprietary information.  But as a potential customer I think we at least should know what we need to do at our end.  With the availability of powerful notebooks that incorporate I5's I have decided against a dedicated PC in favor of a notebook with a long USB cable connecting where I sit (or lie in my case because of my bad arthritis - but that is another story) to the USB dac.  From a convenience point of view this is a real plus and I would need one hell of a good reason to move away from it.

Thanks
Bill

Bjorn, thanks. So, I was quite on the wrong track with my before post. Of course this is now only your opinion (and mine was just another), but assumed your opinion is shared by everyone (and I won't be playing vinyl anyway - not real time from the turn table at least), I rather want to make it myself more comfortable plus better (SQ) playback means;

In that case I'd indeed store the file raw, and apply the RIAA correction afterwards, like with the real time turn table playback. One big difference (and this is my conveniency and the bettr SQ), it then can be done as a preprocess and fits the general approach again.

At this moment I am not sure what it takes to store a raw take from vinyl, but notice it doesn't require XXHighEnd when other software exists for it. Key will be the "no RIAA correction" and the (digital) levels the file should be recorded in. This may be about necessary headroom (for the correction) and may not be an easy thing at all. I just don't know.
The downside will be the storage space it needs, and assumed 384 will be better than 192 (or 96) the files coming from it will be huge. Also, the preprocessing may take ages (you could try current 352.8 DXD (free) downloads from 2L to see what I mean (they can be played via XXHighEnd when you have a 176.4 capable DAC)).

Anyway, what is to do is :
1. Find the good way to capture non-RIAA corrected vinyl data;
2. Find RIAA correction software.

The latter may be convenient because I maybe can just use it. Its importance for now is though, that you can proove for yourself it works. When that has been proven I can build it myself via the (hopefully) known algorithms.
Both 1 and 2 above I won't be doing now, so others may. When something has been found we (I) will proceed on it. And yes, you will need a recording device to ever be able to try. So, if necessary I can try it, after someone found potentially good software (hey, I could do all in the end. Yes, AT the END haha).

If I'm totally off again, let me know ! (and I'm aware that I don't propose real time playback anymore)

Peter

Haha, nobody has to be afraid of not being able to connect to the DAC (says Scotty, kind of.). Therefore you don't need to know. Or at least not now. But in the end you will be knowing what you buy of course !


No matter the i5, you don't know what you are missing. A bit difficult in your case, because you don't use XXHighEnd at all at the moment. But here are a few reasons :

• A notebook is underpowered. It always was, and it ever will, relative to desktops and the like.
  
  
• You *will* need all the power there is, no matter today it is two times more compared to one year back.
  
  
• Notice that this power is felt at nearly everything you do because of the preprocessing needed. The shorter that takes, the better it is. Always. And it will get more important when other preprocessing stuff comes around (xovers, room correction, be creative).
  
  
• There is no single means (at the moment) that can replace the speed of SATAII(I). Today you may be happy with an internal 500GB (by itself fast enough) disk, but in 3 years time (at the most) you will be in the same situation as many, and won't be knowing where and how to connect the disks. And if you do, a copy of one of them takes more than one day. This, while it can take 2 hours.
  
  
• When a notebook fails something (think in the area of glitches and the like) there is most often not more to do than throw it away. Better not be ignorant about this, because it just is so. Not so with a desktop etc. system, because virtually everything can be replaced.
  
  
• Similarly, this is all about SQ, SQ and more SQ, and a great deal of it comes from the components used. Now try to replace such a component for the better, in a notebook.
  
  
• Again similarly are your convenience options. Take the example from wimck. He bought (a new) system, but that system actually doesn't alllow him to use the HiFace. Allright, it is the fault of the HiFace, but still holds for useage. The solution was another USB controller. Try that with your notebook. And this is only one example, while dozens exist. Not today, then tomorrow (like the HiFace didn't exist "yesterday").
  
  
• Only very few things can be done with a notebook when it comes down to other OS versions. You'll need the version from the notebook manufacturer, or otherwise nothing will work. I know, these things can be done allright, but can you ? can everybody in general ? No, it is the other way around : virtually nobody can. With a notebook you are stuck in its shell.
  
  
• Apart from the underpowered thing, notebooks are slow. I wouldn't be able to work with them, because I really have better things to do than waiting for the notebook to come up with whatever it is it is doing in order to get whatever it is to be done, done. You may not recognize it, because you don't know better (just used to it), but take it from me a todays notebook is still way more slow than a normal desktop from 10 years back.
  
  

So where does this leave you for conveniency ?

You best option (I think) is to get yourself a nice flatscreen, a couple of lengthy PS2 and VGA (or DVI) cables. Drop your screen, mouse and keyboard at the place you want (which is not on you lap in my ideas) and let your music PC be as large as needed, and even close to the stereo (so you can use an USB cable (etc. !) of normal length instead of something which implies bad SQ (in theory)).

A second best option is a laptop connected by Wifi to ANOTHER PC in the same network, that PC ethernet connected to the music PC (never stuff Wifi in the music PC !). The laptop controls the music PC my means of RDC. Notice this already will degrade SQ because of the RDC activities in the music PC.

A third best option is a laptop which also controls the main machine by means of RDC but is connected by ethernet directly. This is third because less convenient, and will degrade SQ as much (whatever much means) as the before option.

So, IMO there are a hell of a lot of reasons enough to change things a bit. You do want the best sound, right ?
Peter

Peter

Converting vinyl to digital and digital RIAA filter

1. Digital RIAA filter plug-in
see, free download (VST)
http://www.kvraudio.com/get/3550.html

2. Saving files
My thought was to record and save vinyl as 96/24 files.
As for upsampling in XX, I suppose upsampling of 96/24 files will require a 32 bit DAC, such as NOS1 (similar to upsampling of 16 bit files requires a 24 bit DAC).

Bjorn

Hi Bjorn, hi Peter,

there is not only one RIAA curve. As implemented by behold company there are "RIAA, RIAA (IEC), TELDEC, AES, Columbia LP M33, NAB,
NARTB and London M33+M45". This is the only product I know which should be near your wishes Bjorn --  http://www.behold.eu/page.php?en32118 --.

Georg

Hi Bjorn, thank you.

On the Saving Files :

First off, personally I wouldn't capture at 24/96 and think it's best to upsample to 192 next. Then I would capture in 192 ...

Next, you are right that upsampling from 24/96 to 192 requires more bits officially (actually, 1 bit more in this case), but I wouldn't bother about that at all. There is nothing to perceive from even 24 bits hence from the dynamic range that brings. It is just way too large already (144dB).

If you wonder how or why, I have written more often about it. Maybe you can find it (I'd search for "dynamic range", "24 and "32" at the same time). And otherwise maybe open a new topic for it ?

Notice that the NOS1 is 24 bits, but that the output to it is 32 bits (even at 768). Here too, 32 bits would bring nothing useful !

Regards,
Peter

Georg - But isn't it so that there are more of those curves allright, but only one is RIAA ? ... and *that* is the one used the most if you're only not playing vinyl from before the 1950's or so ?

Also, I think I have read that what was used before a certain year is unknown anyway, so we wouldn't know how to "recurve" it. As your phono stage can't ...

But I don't know much about it !
Peter

Hi Peter,

that's right - starting from that time (sometimes with a delay of months or years) there was one type of the RIAA curve standard. But a lot of the good records are pressed on vinyl before  ;) But there is a cutting machine at the end of the process, mostly Neumann cutting machines. They invented another cutting time constant (50 KHz roll-off) which is part of most newer LP's. It is implemented in the aquvov phono preamp -- http://www.aqvox.com/phono.html --

Georg   

Peter,
the schematic is private or public ?

Have you used a design like this one?
http://tech.juaneda.com/en/projects/digitaldecoder.html


Andrea Ciuffoli
www.audiodesignguide.com

Andrea,

I looked, but no. I don't see any resemblence. The 1704 maybe, but there it stops.

Btw, that is not allowed to be used commercially, but I guess you knew that.

The design is proprietary (and owned by me).

I hope this answers your questions !
Peter

Do you seel the dac pcb board ?

Any simplified schematic is available ?

Andrea Ciuffoli
www.audiodesignguide.com

Andrea,

You wouldn't be able to do anything with the DAC board, so the answer is No for already that reason. :)
Estimating correctly what you want to do with it (I searched my life out of me at the time), it won't work. The whole thing comprises of elements all not done before, so it is a consistent whole, no part being able to work without the other.
You might just as well start from scratch yourself ... I did that in the end (by kind of accident) ...

Thanks,
Peter

Peter,
Are this design able to play at 192KHz 24bit ?

There is No bit shift, no digital filter and No oversampling

The PCM63 are changed with PCM1704

Hi again Andrea,

So ... in my earlier post I said I estimated what you were up to. Now please allow me to ask you ... :)

?

Peter

I think he wants to build a cheap 2ch NOS dac able to play highres material.

Yes, I would like to create a project to play high resolution materials like the 24bit Studio 192 format (www.linnrecords.co.uk and www.2l.no).

Offcourse this should play also low res. material like cd ripped.

I am thinking to use a 192KHz 24bit usb to spdif interface like these and get I2S signals:

http://cgi.ebay.it/HLLY-MUSILAND-Monitor-01-US-HI-FI-Mini-USB-sound-card-/260573890807?cmd=ViewItem&pt=PCA_Speakers&hash=item3cab6a08f7

http://www.m2tech.biz/products.html

Allright Andrea,

I think you may be confused by the way I write about the Phasure NOS1 (this topic), plus that you might have read the first couple of posts and not the middle. So yes, this started out as a DIY project, but ended up in a commercial DAC (IOW, you can buy it if you want, but there's not much to DIY on ... :sorry:).

And notice the studio format (at least from 2L) is not 24/192 but 24/352.8 (which is DXD). The NOS1 does just that (and more), might you be interested.

Regards,
Peter

In any case I have ordered a pair of PCM1704 to start some tests.

Probably I will use a CS8416 followed by a AD1896 to resampler all the I2S input to 192KHz 24bit and after a direct connection to the PCM1704 (inv port for left ch).

I have lot of experence with AD1896:
http://www.audiodesignguide.com/DAC_final/DacFinal.html

Passive I/V with a pair of 50ohm MK132 Caddock and vaccum tube stage D3a SE with CCS.

Sorry Peter for the long time lost with me.

I have started a new forum about my project.

http://www.diyaudio.com/forums/digital-source/166290-192khz-24bit-dac-no-oversampling-no-digital-filter.html

bye

Peter, any idea of release date yet? I mean, are we talking Q3, Q4 or into 2011?

Cheers,
Mani.

Hi all, time again for a small update.

Let's start with my ever hunting me subject : the gain ...


I never told it publically, but I already told a few people off line that I again threw it out. This time it lasted two weeks though.
You could say that I thought it was "commercially" too dangerous. For many albums just great, but for most albums unlistenable again. At least through my system, and horn speakers which resonate at certain frequencies when those frequencies are too loud compared to others (and compared to as should). So, maybe, maybe I would be able to keep on listening to that myself, but then I thought and recognized that frequencies were just jumping out and it had to be wrong. Again.

Because it was much closer to what I think is good already, plus I could recognize frequencies jumping out, I now had some stick to hold on to, and search for the cause of this anomaly. And, with a special type of measurement I could find something not optimal in the circuitry. Something normally not important, but with the stupid PCM1704 as important as can be, once you learned how these chips behave, and react to an incorrect load.
So, with the help of measurement I could tune this optimally, and two weeks back again the module was stuffed in. No, I never give up.

This time I was sure this was commercially allowed, and this time it sounded BETTER compared to without the gain (a gain I don't need myself).
The only "difficulties" left were the more massive productions with a lot of cymbal hitting. So, if we'd take Led Zeppelin or the old Deep Purple as the example of this, I would say this wasn't quite right, but my wife keeps on telling me "this kind of music can have that" ... which almost sounds as "should have that" ...
Since anyway me hardly listens to this stuff anymore (but for testing), I thought it would be okay. So, at last I had found the combination of a sufficient amount of gain, together with outrageous detail, and highs as never before, while those highs we sufficiently acceptable throughout all music.

Now, maybe you can a "flaw" admosphere in the above. If you do, this is fine, because this is how I feel at writing it. Thus, it should be huray all over, but it is not. It is "just good", and nothing to write home about.
Oh ? so wasn't this what I have been working on for over 18 months now, and didn't I just tell that I at las achieved it ?

Yes, but I guess this atmosphere is created automatically when in the mean time something else has happened ...


Oh boy. :coocoo:

Allright. Let's first put things in the proper perspective again, or otherwise everybody may be lost on this. Especially those who auditioned my system in the past, and at whichever stage that was, it was already good, always ready for further improvement, or the best ever possible. The last stage "the best ever possible" is already from 6 months or so back, and this was before improvements which created crazy more detail in mainly the mid area.
Maybe a small example of this for those owning the album and because it is easy to check, if you listen to the first Dire Straits album, and the track Six Blade Knife, after 20-30 seconds there is this Trrrrr percussion instrument coming from the right speaker. That is, it must have been 6 months or so back that I first heard it, faintly. Now, with a few steps of improvement throughout these 6 months, now this instrument can not be missed ever anymore, and it almost sounds as loud as the other sounds from that track. I myself just can't believe that at first it wasn't there at all, while now it's nearly blastingly there. Btw, XXHighEnd contributes to this too.

If you try to imagine the above, or better, if you check it yourself for its audibility, you will perceive it to be there to some degree. Now, especially if you have difficulties in recognizing it, now hear me say that music, generally, is completely full of these kind of sounds. If these sounds are not there, okay, you might be having music allright, but if these sounds suddenly *are* there, well, you won't be able to recognize the album you even played yesterday. Nothing new by itself, but in this case so drastic, well, what to say.

The above was the stage from before last weekend, and where the gaining stage at last brought much much more than without it, so I was in heaven for that. Not only for the good sound for myself, but because I now knew this could go to you without any danger. Your jaws would drop from all what is there in your stupid redbook CDs. Together with it you'd have a very (very) fresh sound which would not fatigue, as long as you stayed out of Deep Purple ...
You'd be having something of which I would think "Well, this is it, at last I have done it. I can't get it better".

As said, before last weekend. :huh:

Then, on Saturday one of the more or less (haha) crucial parts arrived at last : the lastest PCBs for the DAC itself. The one which does 768KHz for input.
Sunday afternoon I soldered it together with the (new) special caps I had in mind for it, and after that no sound came from it. Yeah, that happens when you overlook for more than an hour that the cabling was still setup for Saturday's movie, in the end coming down to "forgot to put in the plug".:fool:
Then, by the time it was almost too late to listen and dinner almost was ready, it became clear within 0.2 seconds that something had drastically changed. Djeezz, how the most normal this sounded ! Yes, NORMAL. But what the heck is "normal" anyway ?

Yesterday evening, and time for listening again;
Those caps, needing a break-in of 3 days to my experience, just had one night behind them because yesterday all being powered off because of other (DAC) works. So, I started off with an album I run weekly (or anyway when something new is to test), Cornershop - When I was born for the 7th time. Not much familar to you I guess, but rather good quality, much varying, and enough strange sounds to appreciate it as Indian (hey Mani !).
I must have played this album some 50 times by now.
Well, to be brief : I did not recognize anything of it anymore. I tried to focus on before heard stuff, but I couldn't. Ok, melodies maybe, but further ? no, all the attention went to the new stuff now there. Hmm ...

After this I thought not to loose time to right away put Deep Purple to the test. Here we go again : Made in Japan (normal version). I can't count how many times I played this one ... can be 500.
Now imagine ... this is just "stupid" hardrock, and really nothing to get tears in your eyes from. Especially not if you heard it a 500 times before. And no, not even when I played it for the first time by means of CD, while the LP was played gray at the time. It is just those most familar tracks and tunes.
This one too I must be playing each week for some tracks, mainly to let squeeze out those cymbals, being fairly gray to begin with (well, as usual).
I was alone in the room, and at the swelling of Highway Star I immediately knew it ...

When I get tears in my eyes, it is always doubtful whether this is because of the music, or whether it is from a :yahoo: yahoo, I have done it !, but since the "I have done it" was already behind me (with genuine satisfaction), this must have been about the music. Music ? well, some say. Haha.
I was shivering from top to toe, and couldn't think otherwise than that I just was there. No, no new sounds in this music, but just COMPLETELY live.
Imagining the live concert I really tried to see where improvement could be, but I couldn't. Including the cymbals and all ... man ... this is just IT.

I tried to analyse what actually happened that this all works, but I don't have real answers. One of it could be that just nothing disturbs. Another could be the sensation of being in open air because the sounds themselves were so much full of air. Also don't underestimate the rough deep low synth sounds coming from John Lord's "organ", and btw the lowest bass on Smoke On the Water (jumping in after a minute or so) is just that organ (and not a bass guitar). It must all be related to something like "nothing being wrong". Actually a mathematical thing. At least I couldn't hear anything being wrong, and this is the very first time in my life ...


We tend to relate good sound to the volume we can use. I too think this is related, but maybe now I found some reasoning to really justify it. Look :

The DAC now outputs 1.5VRMS (this is around 2V peak-peak, see later). The before best sounding version had an output of 324mVRMS. Notice that times 2 (6dB) this is 648 and again times 2 (another 6dB) this is 1300;
With the before 324VRMS I played this Made in Japan album at -12dB(FS) or so. In fact the maximum level without expecting complaints from the family (and which level is already shouting in eachother's ear when something must be said). Yesterday I played the album at -10.5dB, which means some 14dB louder as usual. 14dB is a LOT ...

We tend to think that we can play louder because things don't shout or get harsh etc. This sure is true. "Shout" will happen with frequencies put out unevenly (for level), and hars will happen when things are wrong (or not the best) all over. But as to how I think now, there is another reason ...

To emphasize how crazy it all is, and indirectly how sound has improved, let me also tell you that my level of playing always has been around -10dBFS. This was also the case when the DAC didn't output more than 164mVRMS (so that's another 6dB less). Now, always when I recognize I output higher than before without "pain" I measure the SPL. Important to know is that in my case the output level I use is never squeezed because of harsh sound (or shouting etc.) otherwise, because in such a case things are not good to begin with;
Whenever I measure the SPL, it is always around 90dB. So, the "technical" output is more, but the net output is not ?

Yesterday I broke another record for technical output level. I also could sense it because of the energy around me. The airyness which itself seemed to create sound pressure ...
I measured the SPL, and no way it would reach 90dB this time. It was way way lower, and merely 70dB.
Huh ?

The only thing I can think of is the official math laying in squares versus sines. So, if I output, say, 1Vp-p in squares, this will be 1 VRMS as well. Notice that "VRMS" is the average output voltage, and p-p (peak-peak) is the output voltage you would measure with a multi meter. Now, a square will always be at the peaks (horizontal line), whereas the sine only reaches the peak for one moment in the wave cycle and then drops back. The relation between a sine and its average voltage output is 1.414. In other words, 1Vp-p output = 1/1.414 = 0.7V.
So my gut feeling tells me that something like "before all was far too squary, and now it's nice sines" is going on. But it sure is audible, whatever it is.


Are we on schedule ? In theory yes, but the parts with 8 weeks lead time still have to arrive (5 weeks being past now), and it is hard to tell whether they will be there in three weeks or not. So, actually nothing much changed to the plan, but for two things maybe :
- The gain stage will definitely be in there, and has to be produced (PCB stuff);
- I have found a nice display, but it needs programming, and so far I have only 10 out of a total stock of 100 and I can't get more from that stock. So, that too needs production.

:heat:
Peter

thanks for the update  :)
I think i can start ordering parts for the amplifiers now  ;)

I think so ... :)
So keep in mind : output is 1.5VRMS now.

Additionally for you Telstar, this version of the DAC supports the 8 channel business, but it is not programmed yet. Not in the CPLD, not in XXHighEnd, and not for the real purpose : the XOver stuff.

But the parts for your amps should stay away for 8 weeks, to keep things fair.
:) :)

There is time... The XO stuff i think at the beginning i can handle it like now (which means no highe than 4x AP)


Probably it'll take longer, who knows :) In the meanwhile I seem to have found the best VAS (which worked even with 300mV out and will have easier job now).

And i have yet those tickets to buy...

Peter, congratulations, it seems that everything is on a proper way. There is only one thing that concerns me, is it immune to PC's noise? Or shall I put it this way - does it sound the same with every PC if XXHE set properly?

Best regards,
Marcin

Wow. All this reports are wetting my appetite. Peter, you have become a real…… dacteaser! :whistle:

Hehe Pedal ... my dear Buffalo friend ... :whistle::whistle:

Marcin, the only honext answer is this, and you actually gave it yourself :

The DAC is galvanically isolated from the PC, and on that matter I would dare state that the PC can not make a difference.

Right ?

But XXHighEnd still *does* make a difference (actually I am happy about that by now).
How can that be ?
Or else : If XXHighEnd can make "the" difference (luckily in an expected way for everybody as it seems), why would something else in the PC (could be another piece of software) not do that too ?

If I don't count "per channel", there are 4 separate shunt regulated power supplies in operation (including for the low jitter clock of which there is only ONE (per frequency type) hence no master-slave relation), and XXHighEnd keeps on whisteling while doing its ever designed task to influence.

So what would be your answer ?

But I think you must try to approach this somewhat differently; Already last week the sound was so drastically different from what you ever heard before (promise !), that you don't bother about such a thing as possibly influencing noise (which technically for what we know can't even be there in the first place with galvanical isolation). Now, today it is again so drastically better compared to what I was used to two days ago, that now *I* don't care at all. Let's say the "so much better" just overrules everything and all.

Let me try to give another example, which hopefully can be recognized;
Somewhat longer ago, after aquiering my equipment from then, the only thing left to do for me was making special ground connections (and I don't care about cabling, at all). This always helpt, and the more time proceeded, the more grounding cables I had everywhere, and the better the sound became.
If I would now, with the same equipment but without the grounding, let XXHighEnd loose on that, there would be no no no way that grounding stuff would have helped better. XX would win at all times, and the most easy. And so, in my current setup I don't care about grounding anymore. There is nothing. But ...

If you look at the first post in this topic ... there it was another matter. That design was even depending on your grounding in Poland, as it seemed. You can better say : nothing helped to make that good, and it even allowed for getting out deep down earth mysterious sounds (which later appeared to be a pump in my house) - nobody wants. So, with a grounding scheme you have to start right, and the scheme in this DAC is, well, very special (I'm serious). Let's say it is all floating ground ...

As you know by know, I have high efficiency horn speakers (115dB), and everybody with such speakers will know the foremost problem : audible noise from the buggers when the music is silent. Everybody has that. What not everybody has (but I do) is not using a preamp, even in that situation. This means full gain from the amps (and everything in front of it), and still with your ears in the horn, you hear just nothing. If someone else has achieved this, let me know. :) So, this is the base, and the real base is the DAC, because that is doing - or killing it. I have no ferrite stuff or anything to prevent digital stuff to get back into the mains. It just is not necessary (while officially it is, as it was with all of my former setups).
But measure your own inherent noise. It will probably be at -100/-110dB. Here that is over -130dB.
But remove your DAC (I mean *remove*, and not only shut off because that doesn't help !) and look again ...

But ok, you still can be right that some PC setup can incur for noise. But only theoretically because XXHighEnd still influences ...
And then : no good, money back. :yes:

Peter

I cried too, on Sunday. Well at least my eyes got wet. I was playing the ELP Brain Salad Surgery DVD-A, bonus track "Luck Man" at +100dB with the latest XX version and 1024 latency. Oh my god. The realisme could make an old prog rock fan cry anytime. (I am really into hi-rez now. Bought all the titles I could find on eBay and Amazone + downloads from HDTracks. When you get used to music mastered in 24/96 or 24/192, then you easily forget about compressed CD).

WISHLIST: upsampling and ARC with hi-rez formats, not only 16/44.


PS: Maybe you should make a new section in the forum called "The last time I cried", as the ultimate listening experience? (You can rename the "Measurements" section. It has never been used). It can also be used for hardware accidents, like "yesterday I cries beacuse my amps blew up", or "my harddisk crashed without backup".
 :smile:

LOL Pedal. Nice idea. Btw, what about COL eh ?
Crying OutLoud. That is what I did yesterday. :cry:
With all your pussys around me now I dare say it ...


Coming up in a couple of hours. :grazy:

Compliments to the Chef,


So, WiFi, seperate videocard, overclocking are also allowed in the dedicated audio PC.
So no more power tweaking and such (CPU, RAM)

Wow, dedicated audio PC, is no longer a necessity, just get a fast PC.

Also a good thing for our American audio friends, is the "good" dollar-euro value. (0.83)
By the time your sales start the price will probably be even lower.

 :drinks:

Hi Roy. Well, don't make the quotes out of context. Or at least I tried by best to state the opposite; As long as XXHighEnd keeps on influencing, it is my guess everything will. But, I also said the other stuff will be in the minority (hence overwhelmed by the DAC itself).

Now, this too is not completely true; once you have heard what it can do, a different XX setting immediately makes it less again. So, that doesn't differ from before. That this "less" still is 5 times better than my day before is another matter. So, in the end it is all about highering the bar, and what you don't know, well, you don't know (only your bar will be lower than could be, but it still will be 5 meters about the before world record).

If I only could stop XX influencing ...
(what a world)

Peter

I hardly believe it. My M-Audio Transit (TAS1020B) also gives galvanic isolation and I couldn't stand the sound until I made all the tweaks and undervolting/underclocking. But the Transit is fed by USB power... I haven't got any firewire device so far, but I know that for USB, the PC's power/noise will affect SQ even when you spend money for a fancy transport like ART Legato with separate power. The only way to virtually eliminate computer's negative impact on DAC is to cut the power leg from USB, but sometimes it's required for a receiver to operate. My question for you Peter is: does NOS1 feed 30V from firewire on PC's side?

Why should it, if no Firewire is in there. Hahahaha

But to answer your question hopefully more satisfactory : it is completely self contained for the power supply (supplies). And no, not by means of a switched brick whatever. Lowest ripple possible.

Btw, I agree with your last post, if that wasn't clear in the first place.
Peter

Sorry for the confusion of my last post,

It was ment as a question not as a statement. (how would i know, anyway ?)

But i totally believe you on:

Compared to different cables, capacitors and so on.
But still minor changes in software can have drastic effect on SQ, for eg. 48 vs 1024 samples.

The whole project of xx was like this, its not a little better or worse with every version, its totally different each time a new version is
launched.
This can be hard sometimes, as in what is the "Reference" (my reference got lost a long time ago)
I am still amazed how this can sound so different each time.
Try to explain this to the simple minded people, who say its just ones and zero's (I remember a discussion with someone, a couple of years ago)

But I really enjoy the whole learning curve into achieving better sound.

Houdoe

No, or better ni.
1) Wi-Fi contaminates the AIR and all electronics are subject to the dangerous RFI and EMI. We cannot do a sh*t, but shield every piece of equipment very well.
Why? Because unless you live in an isolated, very isolated villa, you'll have already a few wi-fi networks hanging on you. Try with any modem or smartphone. My iphone picks up an average of 2-4 from my neighbors.
So the recommendation of Peter to avoid it only apply if you dont have it already. Otherwise, I dont see other solution than shield every piece of electronic gear in a thick mu-metal cage.

2) Non dedicated PC - I think Peter answered you on this. Various processes can still influence xxhe, that is the sound, so best to avoid. But the computer can be fast, as long as fanless and EMI/RFI controlled (note that Peter has his computer FAR from the DAC and his system, AFAI Remember).

3) PC PSU. Here instead special care to the computer power source ***should*** NOT be needed anymore, when using the NOS1, with its special interface.
But, what is needed is a filter to block the noise FROM the PC psu and any other switching equipment that is connected to the same power line (I mean line not plug) to protect the OTHER electronics in picking it up, namely the amplifiers (the NOS1 seem to have an excellent PSRR by itself)
I'm going to build such filter for about 300W power, single rail and 6 plugs. It'll be open source/diy.
At this regard if someone can source the fabric used by Furutech to absorb EMI/RFI, it would be a nice addition.

Ahem... my engineers made me notice that vRMS is half of the p-p value (but i dont exactly understand why).
So my question is: did you measure the peak to peak tension or directly RMS (my multimeter can do that) and calculated the other?

Uhoh, better be careful what your engineer is going to make for you ! :) :)

Draw a sine (or let a program do it, so it will be a real sine). Then imagine the average voltage coming from it. Where it crosses zero, it is zero. Where it is at the peak, it is "peak" (what the multi meter measures). Would you say the average (above zero) is half of the peak ? I don't think so ...

Anyway, the RMS comes from the analyser (the 10K multimeter :)).

Btw, for clarification, I saw that I wrote this :

"1Vp-p output = 1/1.414 = 0.7V"

which should have been

"1Vp-p output = 1/1.414 = 0.7VRMS"


http://en.wikipedia.org/wiki/Root_mean_square

Look below, this gives 1.41421356237

I think the misunderstanding lies in the difference between peak voltage and peak to peak voltage, which are not the same thing (although i still dont understand why). So I believe that you measured or calculated the peak voltage, in which case to use the squareroot of 2 divider is correct.

What matters for me to match the sensitivity is the vRMS (taken with sinewaves) :)

See here:
http://en.wikipedia.org/wiki/Amplitude

Exactly, in your attachment there's the formula for peak to RMS.

Someone must make a mistake here. It is allright if that is me. :)

I only now understand your message "VRMS is half of p-p" which at first to me looked like "when V = 2, VRMS = 1". So never mind my previous post.

Fact is that 1.5VRMS = ~4Vp-p (and I said 2 in my earlier post -> WRONG).

So yes, I calculated the p-p (wrongly) but measured the RMS -> 1.5.

Sorry for accusing your engineer. I stand corrected !
Peter

This is what matters in the end :)

Here is another status update ...

First of all, the parts I was waiting for, I am still waiting for. They are over due already, and day before yesterday I heard that they will be shipped July 16 now. Assumed that will be true, with another week of shipping time the first DACs should be ready for shipping near the end of July. If I'm only ready for the administrative side of things (never got around that anymore, so far).
So, I guess we are really almost there now !

In the mean time, again the additional throughput time has been the most fruitful;

A few weeks back the latest version of the DAC board was received, and only last week I got around to integrating the CPLD logic (made at a distance of which I told about I think) with the XXHighEnd logic, both providing the special 384/768 mode. And, after 2 days of work on the XXHighEnd side, I can say that it all works above expectations. Above ? how ?

On a side note : The whole technical and electrical idea behind this was created November-December last year (2009), and while this implied a complete new design of the DAC (board), while that was produced I got some other ideas that could nicely combine. And, now really having that DAC board, forced by some technical problems in the 768 area I again got some other idea. All the (three) ideas could combine into one solution, and that is what I have been listening to for the last two days ...

Let me start at the end, and an actual by now rather common story : yesterday I played one of the albums we run almost weekly for explicit testing, and I asked my wife "do you know what this is ?". Allright, you know the answer ... she did not.
What the heck happened this time ?

This time it is all about jitter. 3 folded.

1.
By means of a plain dirty trick, I was able to decrease the phase noise to -128dBc @ 100Hz. Very very good is -110dBc, and much more common is -60dBc. Mind you, this is for 100Hz, a common measurement point for phase noise. Also, jitter is the most audible in the lower frequencies, or better : it is the most applicable to the lower frequencies (something like : the longer the wave, the more time there is to notice fluctuations).
The trick comprises of a technical application, and the figure is derived from the base specs from the oscillators I use.

2.
By means of a "logical thinking" trick I reduced the jitter by another factor 2 (but think in dB, which would be 6). This makes use of available headroom when the 768 input is NOT used. Hahaha.
Like I said in the above, forced by technical problems in the 768KHz area (which is ~25MHz at the bit level and which operates outside of chips in my case) I thought to trade the (thus far) unuseable headroom for better jitter specs. And at this moment I think I don't even want the 768 input to operate anymore, because what is gained by (whatever it is) I applied is much much more powerful then shifting away high ferquencies further (and remember, 384 (352.8 ) was enough for that already). But, if 768 gets to work afterall, it is just a choice in XXHighEnd.

3.
Last but not least at all, something completely new, and let me be the inventor of it. I am not sure yet how it works or what "influence" is eliminated, but I tend to call it interchannel jitter;
We all know about the good merits of separating analogue devices like mono blocks instead of one stereo amplifier, separate PSUs for the separate channels or even separate DACs for the channels (inherently present at the chip level for the Phasure NOS1), but I found that there is also an "opportunity" at the plain digital level, where nothing in that path is routed to one channel anymore. The effect of this should be equal jitter for both channels. Or at least the way I set it up, should have that effect.
Huh ? :huh:

Sound wise ...

Let us start with the latter ... interchannel jitter.
It is the most drastical sound change I ever met in any of the steps I ever applied, might it be in my younger days, or just the "XXHighEnd years".
It implies a channel "separation" in a dimension we didn't know it was there. So, there is channel separation, and when measured it can be 80 or 90dB or whatever figure we know it is inaudible. Or, when you listen to a really hard panned guitar from the left speaker and just can not hear it from the right speaker, well, that is good channel separation, and generally we are not worried we are bothered by it. It is too inaudible.
Now, in the above I put "separation" between quotes, because it is just about the opposite : channel integration. :huh::huh:
Let's keep in mind that I dedicated this phenomenon to what I perceive myself, and maybe there's a more official/known phenomenon for it. But now let's try to explain :

Let's first assume and aknowledge the audibility of jitter. I, or I think I can well say "we" don't care about low jitter specs of a device, and as long as XXHighEnd can influence *and* assumed that the carrier of this can be jitter only, we hear jitter. Oh yeah, we hear it very very well (the differences).
Now suppose that the differences we can create with XXHighEnd would be there between the left and right channel. That would be a quite messy sound, a bit depending on the extremes you found in XXHighEnd settings. Now go one step further, and let's learn that those differences just *are* there;
Also good to recognize is that we aren't used otherwise. It is all normal for us, because no tool was available to tweak that difference. The only thing fair to say (without real scientific knowledge) is that when the general jitter level is very low, the differences between left and right are also very low (and the other way around).
Now, what I did, was 100% elminate that difference between left and right, meaning : The jitter is still there (that assumed), but however it exhibits, it does this the same in the left and the right channel, and this is at the sample level. The audible effect ?

A so super full warm sound that you really can not imagine it is coming from the same speakers and amplifiers.
Yesterday I was explicitly listening for what to write in this description, but I can't describe it in a technical fashion. Oh, I could say that you will have the perception of a wider sound stage, but it is not that. Physically (for dimensions) is it not that. But there is a stage full of sound, instead 5 microphones at 5 places. There is totally no sterility left. It feels like how it was real, at the performance. It is also not like a mouth has become too big or something, but merely a far more exact projection of a small mouth which can't sound small at the distance you listen.
Let's take one of my albums I always try after a bigger change. Get Yer Ya-Ya's Out! from The Stones. Do you have it ? well, put it on now;
I think I talked before about the improvements achieved on this album, and that it is actually completely flat far away sounding. It will have vastly improved, but still, if you compare it with other good recordings, no, this is not it.
But now ? man, you would bet 10 million that I had a unique very well done remaster. There is completely *nothing* left of the flatness, and I even imagined Charly Watts to look happy, and applying variations I never saw him do in clips.

Btw, yes, I watched carefully whether the focus on voices or instruments was still there as should (instead of DSP-not).

Allright, on to the next two jitter subjects, far more normal.
Ehh ...
This is what makes your albums (again) unrecognizeable. In the end it is (at least to me) a very strange thing. Although I must admit that along the line the mids again came somewhat more forward - which has its own merits -, this time all kind of melody lines come forward, never heard before. It can be bass, it can be by background bells, it can be a far away guitar, but it is always about a melody line. Well, you can imagine that if you were used to listen to a track with a certain melody (whatever that was) and suddenly the melody changes because another line comes forward, well, your track sounds different up to unrecognizeable.
Part of it is created because of more energy which comes forward most in bass instruments. But it is also about "sub harmonics" (which don't exist) forming an attack layer for the particular instrument and even voice. To make this clear I better refer to drum veils and their attack. This was already very good, but got better again. There's more warmth under it now (the so called sub haronics).

Here's another example : Faith from George Michael. I mention this one, because it occurred to me that it wouldn't work as I wanted before. You could also say that I didn't get the feelings with it, I recalled from when being a tad younger, women involved. It was too sterile and had no slam. It didn't have the power the tough guy on the cover expresses, no matter how gay he is.
Well, you saw it coming ... now it works. But wait, what does that mean ?
Throughout the album a bass is used which really does the job on all of the tracks. It should be the most profound of all, and it should incorporate the synth created vibes in the mean time. So, very very powerful, but at keeping those vibes audible. That it goes along with the super duper warmth from the former subject is another matter, but together with those most hard drum rythm slams it draws tears (yes, it did). So, it is the combination at work here. It *is* a super tough setup album, but it *is* sung by a gay guy with gay warmth.
Without the slam the combination isn't there, and it's only one of those dozens of two gays behind a keyboard guys with women voices.

Wow, you really can't imagine what can happen when all has come together. I tried some of the Shulman albums, which is ambient and only 10 minute or longer tracks. I know so well how boring it actually is, because it is not only 10 minutes of the same, but a whole album of the same. Still I like it for a kind of background louder music (hehe ambient eh). But today ? today I can't stop from being attrackted to each of those seconds, the album throughout. It is one big pile of intelligent detail harmonically formed into a story you are sucked into.

Ok, unless 768 brings something additional, I'm done;
The Phasure NOS1 now works for all types of music, and I wouldn't know what to do further. It should be the best on the globe.

Peter

Man what a fantastic story this is!!!  :toomuch:  :)

Looking forward to "official" photos of ready-to-ship product.

In short George Michael is not Boy George  :whistle:

Lately, my main music source are hi-rez files from various disks and downloads. Sampling frequencies are mainly 96k, but also 88k2, 176k4 and 192kHz. It strikes me that a filterless DAC is perfect for hi-rez files, because they have a natural roll-of with very little(!) amplitude left above, let's say, 40kHz. So no *need* for low pass filtering, because the aliasing distortion will be very low anyway.

The official release pictures will be out shortly, but here's a sneak peak of Peter working on an early version. It clearly shows he's been working on the experimental NOS in a compact form for some time now.























 :innocent:

Hey hey ... ho ho ... :)

Who informed you that a filterless DAC is "perfect" for hires ?
(but let me softly add that a while back we talked offline about these things IIRC, and maybe there I said or implied something like that ?)

So no, this is not what I want to express, and I hope I never did somewhere. And as a matter of fact, the whole discussion since (0.9z) hires upsampling (filtering !) is that it should NOT happen. IOW, no filter ! But as an exception Arc Prediction is allowed, because it doesn't imply a roll off; it only increases (fake) resolution.

But since you brought up the subject anyway, let's be confusing. :)
I think it would be true that applying an Oversampling DAC onto hires material is extra-wrong. This is not so much about the "filter", but merely about the oversampling which just is not needed for hires. So here you go again, because now it is not only the wrong means of filtering which *will* be applied for OS, but also the destroying oversampling.

On this very last, try to see this : As long as no "upsamplers" can be found which definitely make the sound better because of better (but fake) resolution, any means of upsampling is wrong. So, for hires this for sure counts, because you will be manipulating the data while it is not necessary at all. So yes, this is where NOS comes into play, because it doesn't touch anything. That is, if you don't apply any means of filtering, which includes (no) upsampling. But :


with Arc Prediction as the exception, unless suddenly everybody disagrees with this.
But then still ... Arc Prediction is made for letting it loose on a DAC which doesn't again does something to the data, and here we end up again at the necessity of NOS 24/192 (or better). 16 bits can't do it and 96 can't do it.

I hope this is somewhat more clear than confusing ...
Peter

Thanks for your interesting reply.

-Actually it came to my mind while using my new software Adobe Audition 3, doing a spectrum analyze of several hi-rez files. Then I realised there is not much(!) HF content above, lets say, 25kHz. Close to the half of the samplingfrequency (48, 88 or 96) there is practically speaking only noisefloor.
[Even if the digital recording technology can capture signals >50kHz, the mics does not. If we are speaking about music recorded in the haydays of analog, then most multitrack mastertape machines rolled of above 20kHz, too].
 
Conclusion: *Nothing* left to filter at sf/2 when the music file has 88k2Hz or higher sampling frequency. The less lowpass filtering, the better. I look forward to hear you DAC!

Hi all, I am new to this forum, I found here just when searching in google for anything about NOS DAC, and this project I really have great interesting.

I have not yet read all posts, it is a bit difficult for me since I am from Hong Kong, English not my mother language.

I would like to ask is this project fionally will open sell as kit or built?

May I ask the designer for a favour, mind to keep me update via email? I am eager to buy one, thanks  ;)

Hi there - Thank you kindly for your interest !

Although I started this myself as a DIY project, there will be a built version only. Let's say this is because it is meant to be the best (for sound quality) ever, and I rather guarantee that by our own building and measurements here, than leave it up to you with the possibility of making mistakes ending up in unsolveable hum or noise, or worse : no hum or noise but just poor sonic performance, while you don't know it is more poor than intended (you don't have the reference). And not to forget : nothing is normal in this DAC, and if it doesn't sound good I'd rather have that you blame me, than that I blame you for an unknown mistake somewhere while it still could be our mistake here.

I don't maintain something like an email (news) list, but I think it suffices to keep a watch on this topic; When something significantly changes you will read it in here, although I must admit that sometimes things pop up in other topics "by accident".


A general remark for everyone :
Last Monday I was notified by the company which builds the cabinet that the promised proto which should be ready around today, is postponed until the week of July 19. Next they foresee that because of vacations the final cabinet won't be ready before 5 weeks after that (notice that this is related to a couple of "external" rounds like painting and printing).
In the end this is my own fault I think, because I postponed the assignment for making the cabinet "forever", but which by itself was related to the unknown of what had to be in there in detail (and this is final for only a few weeks now). And so I dropped myself right in the middle of the vacation period. Well, at least this gives some space to the other (3) suppliers who are all late now, while the assembly of the parts concerned will take around one week anyway (for the first 100 DACs).

Peter

Thank you Peter, a built version will be fine but if a full, or semi kit, or PCB available will be more fun, since I can have my choice of components such as caps, and maybe saving cost by using some my stock parts, as a qualified electronic engineer, I have confidence to face any problem you worry about, although I usually stuck with analog circuit but I do design amplifiers, and nothing low end.

Anyway, I will stuck with the post and attention for any update.

As told I am from Hong Kong, do you having any preparation for international payment such as via PayPal?

Peter:

Today I use PC > Fireface400 > spdif coax cable > DAC. [which is superior SQ compared to USB adaptive mode, BTW].

But you are using PC > DIY modification > I2S cable > DAC. (Is that correct?)

I don't know if you have mentioned it in the past, but do you consider your DIY 12S solution superior to SPDIF?
Eventually, what have you done to get the I2S output from your PC?

(Probably you DAC will accept several digital inputs, but the whole point about your DAC is to get the best sound quality. Right?)

All the best.

I2S is always superiour to spdif,

I2S is a so called chip-to-chip connection, the output ic from the souncard gets directly connected with the receiver chip in the dac.
This means that data and bit- and wordclock signals stays separated were spdif merges these together and get separated again at the dac side.
I2S has a 3 wire configuration while spdif just uses 1. (this is strange because a cd mechanism outputs I2S)
This skips the need for PLL at the DAC side, wow, less jitter better sound, but this is just half the story, add a good master word-clock and the fun just begins.
Although I think the length of the I2S connection should remain as short as possible.

A simple mod to for e.g. an ESI Juli@ soundcard, will give you I2S output from your pc. (if you have a dac that can input I2S ofcourse)

Cheers,

Roy

Yes, that is correct.


Generate it somewhere outside of the PC. :swoon:
But did that in a fashion that no clock (software or hardware) in the PC plays any role.


SPDIF sounds completely raw in comparison. "Raw" indeed is the best description, although you will perceive it like that once you heard "the other way".
Notice that we can't speek of "an i2s connection" in general, because something like that can't exist. This by itself is because it isn't a standard for external connection, and thus it is your "implementation" which makes or breaks it. What Roy suggested works too but it can only be far from optimal. Still better than SPDIF though.
i2s is just the description of 3 wires, while two of them already aren't related to the logical connection in the sense "must come from one source". This leaves one other of which you methematically can't say "must come from one source".
:smirk:

Would you like to share the recipe with us?  :yes:

Yesterday I didn't want to suggest to replace caps and everything, but I felt you were heading that way. So here it is :

The main PSU board (the one creating the sound) explicitly consists of a configuration where you can mount 10 smaller caps per channel, 2 big ones, or any combination (but the big ones take space for the smaller ones, so 10 small + 2 big is not possible). It comes with 1x 40000uF audio grade low ESR per channel.
There are 33 other (electrolytical) caps to replace, devided over 4 functional areas. None is in any signal path, and all matter for sound. They should be all the best already and take 3-4 days to break in (so no black gates), although I'm fairly sure that after a month you will still see improvement which you will find significant at going back to new ones after 3-4 days breaking in (I just did that for fun a couple of days back, and while the one month old made me say "this is just live music", I can't say that (yet) from the new ones).
Resistors are Vishay 0,1% non-magnetic throughout (for the non-SMD stuff).

PayPal is no problem !
Peter

:no:

Later. :)

This thread has become a DICK DAC teaser...    :tongue2:

Haha, yes, but it's not really on purpose.

While I'm here anyway, maybe a small update again :

First of all, the most important lacking parts arrived yesterday after nearly 5 months of waiting. :heat:

Next, this afternoon I am going to collect the proto of the cabinet. Honestly, this has been the most "disturbing" part of the whole project. Now I am really excited how it will impress even myself. :yes:
When all is in there, I will try to make some vague first photos ... but notice it will be unpainted at this time.


Then there is positive news with a little downside;

I mentioned it in between the lines in some other topic that the capacitors need a longer break-in than I thought at first. This is a downside, because it disallows "fast" judgement for a new DAC. I guess this will take the money-back guarantee to perhaps even 90 days. I can't even tell yet how long the break-in takes !
This is how it goes :

Day one (for me this is a comparison with random other caps) :
Ah, totally different sound. Wow, now you can see the size of cymbals.

Day two :
Sound is a bit worse from day one. More grayish.

Day three :
Wow, yes, this is it !

Day four :
A little improvement over day three.

Day five :
Clearly broken in.

Day 30 or so :
Doesn't it sound better than yesterday ? what did I change ??

Day 31 :
What the heck. Where does this improvement come from ?

Day 32 :
Djeezz, was I deaf lately ?

Currently I am on day 45 or so, and the changes compared to "yesterday" still are on-going with steps that don't seem to decrease. It is a very consistent improving without any fallbacks (Black Gates have fall backs during a crazy period of over 6 months).
If you only know how super-enthusiast I was at day three ... you can imagine I am getting a kind of crazy today.

Peter

Peter

Can you please confirm again your latest configuration for the Phasure NOS Dac:

Maximum 2 channel or 8 channel output?

If 8 channel output can it be configured to have 2 channel input with 8 channel output (for active crossovers etc.)?

Headphone amp or not?

Frank

Hi Frank,

There's the 2 channel input and 2 channel output.

There's a theoreretical 2 channel input and 4 channel output with the normal hardware (but extra output connectors) (via XXHighEnd, the software to be designed yet, and no real use case at hand (unless 2 channel xover))

There's 2 channel input and 8 (or 6) channel output (via XXHighend, the software to be programmed yet).

There will be no headphone output at first, for the stupid reason it needs a 24V PSU (or more) to do it well, and it isn't in there. So, it needs attention which will hold up again, and I don't want that. But Frank, it is very very good that you mention this again, because I forgot to let create a (dummy) headphone output in the prototype of the cabinet and it really should be there for later. So, a sincere thanks for just mentioning it !

Peter

Peter


Clarification please. I assume this means:
(1) It will need only 1 interconnect per channel between PC and Phasure DAC for this multi-output DAC option?
(2) Owners will not be able to use the 8 channel output option in Phasure DAC unless they use the XXHighEnd player?


Does this mean a phasure DAC would have to be returned to you for an headphone out upgrade when this option was available? (Not really feasible from NZ)


Cheers
Frank

I'm sorry if I missed this... will the DAC connect via firewire? It would be great if I didn't have to buy a converter or soundcard and just run it straight from the PC (in my limited experience firewire is better than asynchronous USB)

No, you didn't miss anything. I just never told it. The connection is proprietary, and you won't need anything. Full blown specs will be available before you order. :)

Frank,

(1)
Between PC and DAC is one connection which carries all. But I guess it is your own earlier question which confuses a bit. Let me try to explain better (but it is all very obvious !) :

The DAC is connected to the PC. For now and some visualisation, let's say it is proprietary i2s. This one cable can carry 8 cannels of high speed data. One channel could be a normal left speaker, another channel the right.

The more than 2 channels for normal stereo, is explicitly meant for crossovers in loudspeakers.

(2)
Although it may look strange, the DAC is an extension of XXHighEnd, but xx times more powerful in better sound than XXHighEnd ever can cause (without equalizing and stuff of course). So, it is not the other way around.
In order to understand this better, think of the DAC being filterless for a reason : now the software can control it. And that is XXHighEnd with Arc Prediction. Arc Prediction was there earlier than the DAC,  and most use it. But it excels with the DAC because it is made for it.

In normal 2 channel mode the DAC can be used with any software (Foobar etc.).
In "more than 2 channel mode" (set by switches) no other software can control it because it is all proprietary how it's done. Not that it is so much of a secret, but things are being done which are not much normal, and it requires coorporation with the controlling software.

We must not forget the reason of XXHighEnd being there : better sound. Now, the (max) 8 channel hires operation of the DAC is made for xovers, while the xover math always will be done in the PC (nothing strange here). But, like with normal XXHighEnd operation, the processing and inherently degrading (SQ) effect of such a thing - will be covered for by XXHighEnd, and playback will be as good as always. But, now spread over (max) 8 channels, each with their own bandlimit and roll offs.

All 'n all, because it is the explicit idea that XXHighEnd provides the xovering, there also must be a "superb" DAC dealing with it. So, both belong to eachother, although XXHighEnd will be able to send the 8 channels to any DAC able to receive 8 channels (but you will be ending up in Pro ugly sounding stuff (or 4 stereo DACs etc.).

While the normal 2 channel version comes with both RCA and XLR, the 8 channel version must be ordered with either, because there's no space in the cabinet for *and* 8x RCA output *and* 8x XLR output. So you have to choose this in advance (4 channel can still be equipped with both).

To be clear : The 8 channel version can still be used in normal 2 channel mode (switch).


Lastly, and more or less off topic, out of everyone I now know what "less is more" means, and what any out(PCB)board connection does to sound quality. If you'd see my loudspeaker filter/xovers ... I'm not sure what will happen, but SQ must be subject to a vast improvement again, just by leaving out all these resistors, caps and coils + soldering joints. So, I am the most eager to start with this myself !
(and next I'll have a pair of Duelund caps for sale for 500 euro or so. :swoon:)

Peter

No, it will be done differently;

I don't know of any people asking for the heaphone output but you. I think you will have seen it yourself in this topic, because not much response (not a single in fact) was there to this. Also not offline, where people tend to post when things get "tough" (like you did that). For me this means I can't hold up the further process just because of one person. But, obviously it means that you will have to wait, but it will be for your good reason.
Don't be afraid I won't do it, if you only imagine that the 8 channel version has been developed for one person only also, and it really took a very large deal of the development time (still not finished, and he will have to wait some longer too). But, I seriously think there will be quite some people wanting it, if you only know what it can bring for sound quality (see previous post ... I'm fairly sure).
When the headphone output is there, more people will take it, I'm sure. But don't underestimate this feature, because doing it really well is quite costly, and I won't go for less than "really well". Could be in between 350 - 500 euro.

But ...

A better and honest advice from my side would be not to want it at all, which is just because you won't be able to imagine what will be happening in your normal listening room and the sound through loudspeakers. This is such a happening and sheer happyness all over ... you won't be able to perceive this through headphones. I won't say it will be a waste of money to get the headphone output, but it will be a waste of pleasure and time when you listen through headphones. You'll see ... PROMISE !

Peter

Peter,

What if whe choose for a 2 channel and like to have more later. Can we send it back for an upgrade?? What are your thoughts about that?

 :)

 ;)


I've tried it, not with a SOTA dac, and the improvement is huge especially in the transparency department.
There's no way back to passive XOs for me.

Must be you then ! haha

Yes. I once tried to play the mid-high without any filters (which can happen in my case, but then all is unprotected), and the result was air air air. Warm air.

But this was two years ago, and by now this warm air has been superceeded 10 times or so by other means. So I really don't know what can happen to it now. Maybe musicians jump out of the speakers. :)

Very good question, and actually I wanted to tell about just that in the earlier post from today. But I forgot.

Yes, no problem with that, but of course there's the additional shipping costs. Notice that on my side this will be in euros :

Netherlands : 40
Europe : 80
Outside Europe : 110

But this includes some overhead for packing and stuff. On "your side" it won't be much less though, so it's good to mutiply this by 2 for, say, wasted costs.


What will happen is that the PSU for the DAC will be modified, and a second DAC board will go into the cabinet. 6 output stages will be added, meaning that the standard 2 will be replaced by 8 new. One half of the output connectors will be replaced with the type you want (RCA/XLR) and 6 more of the chosen type will be added. The input wireing needs to be changed, as well as the output wireing. Maybe - but this is not sure yet - the receiver board needs to be changed.
Cost ? unknown at this moment (as the cost for the 8 channel is not known at this moment). It seems reasonable though that the additional work compared to do it right away, may set you back for 200-300 euro. Could be less.

Peter

one m ore question - will we finally be able to run convolver with the XXhighend - DAC solution? not that I believe that room correction will produce an even better sound, but because sometimes adjustments still need to be made for DIY multi-way speakers or 2.1 setups (correct x-overs!). Will it cost as much time to include the feature?

Yes of course, but it has to be made within XXHighEnd.
But is this related to the DAC ?

Maybe you are referring to more channels input ? On that matter ... At the DAC side nothing changes of course (you'd have 8 channels available), but for XXHighEnd it would be something new. Not that it would be a big problem to get one situation running (like 16/48 6.1) but to incorporate it into every possible situation will be quite a lot of work ...

Peter

:yes: :yes: Thats great!

:soundsgood: Does this mean you configure with crossover processing done before ARC Prediction? (In other 8 channels DAC the PC/MAC is used to programme the crossover software built into the DAC)
The crossover function is important to me because I want to design and build my own DIY active 2.2 speakers. I know others will want to do things like converting their existing passive speakers into active speakers.

:clapping: Peter, Didn't expect you to have all these things up front, but did want to know that when I purchase the 8 channel version that the headphone option will eventually come.
If I look at where my "ideal" requirements have come from, my original interest was excited by the Benchmark HDR Pre DAC with headphones, then the Metro Halo Lio 8. I will wait for the Phasure 8 channel.
350 - 500 Euro is not out of the way for a good headphone stage.

Frank

Hi Frank,



Then you talk about stuff like DEQX which is not what I was referring to. So I'm afraid I confused you a bit.

It will be done after Arc Precition, because only then the exact frequencies are known. Arc Prediction is a preprocess, and the xover part will come behind it (like a convolver would).


As in two channel two-way ?

Peter

Then it will use a LOT of computing power, unless you can let the DAC do some of the work.

Peter hi,

I have been dipping into the Phasure NOS Dac thread for a while and reading the updates and progress with mounting excitement . Your XX approach carried all the way through to the DAC output has got to be a good thing for listening. I am really interested in a 2 channel model when they become available. I was not sure if your collecting indications of interest / orders, but if you have a list please, please place me on it !

Nick.

Peter
Yes, Two channel two-way + (.2) two subwoofers, or it might end up two-channel three-way after suitable experimentation.
Frank

In anticipation of buying Peter’s DAC, I’ve been trying to get used to using only the volume control in XXHE.  Why?  Because Peter’s DAC won’t have a remote volume control and I’m too used to not having to get off the couch to change the volume.  My current DAC, a Berkeley Alpha, has remote volume and I use a wireless remote connection from my macbook or ipad to control xxhe on my music server.  My listening room doubles as our living room so I don’t want to run wires and have a monitor and keyboard in the listening position.
 
I’m completely frustrated for many reasons:

1) In many listening situations like mine, you fiddle with the volume more frequently than anything else, and interacting with XXHE to do this is cumbersome at best.

2) XXHE is very slow to respond to volume-changes.  My own experience says this is even worse in unattended mode. You’ve missed the phone call before you’ve turned the volume down!

3) Remote Desktop Connection does not work in Windows 7 if you set a core appointment scheme in XXHE.

4) I’ve yet to find a VPN solution, at least for a mac client, that works reliably either.

5) The ipad in many ways is the ideal remote control.  It’s small and light; it runs cool; it has great battery-life.  However, working XXHE’s volume slider and knob takes too much dexterity for a touch-screen interface and the soft keyboard adds too many steps and effort and delay to making what should be a simple volume adjustment.

So why not include a remote control volume with the DAC?  I can’t be the only person who wants one.  Peter says he hasn’t found one that doesn’t damage the sound.  But what about making it defeat-able? That way, purists can use an optimal setting on the DAC and make further volume adjustments on the music server.  For example, on my Berkeley, a setting of 54 is supposedly optimal when you’re using a pre-amp to control volume.  On my 20-year-old EAD, there’s a combination of analog and digital volume: every 6 clicks of the volume switches in a different Vishay resistor and between the 6 clicks adjustments are made in the digital domain.  Obviously, remote volume has been implemented before without a huge sonic penalty.

I don’t think I can justify buying his DAC without remote volume, and from a business perspective, I think his market is significantly reduced without it.  I hope Peter sees the light and figures out a way to provide this very common functionality.  It’s pretty much a show-stopper for me.

How do other people feel? 

Hi there Danny,

Not to be ahead of others (so, speakup everybody !), and with the notice your remarks are appreciated at least by me, some remarks to your (in itself valid) topics :

1)
I do too, but far far less with the Normalized Volume active. Do you use it ? you really should ...

2)
I use Unattended Mode exclusively, and the volume changes between 1-2 seconds. Of course you could call that "very slow" (because it is), but is it TOO slow ? not IMO, if you only can trust it "happens". And it can be trusted. As it can at changing the volume 3 times in a row (say 3 presses within 0.5 secs, and the total result will be there within 4 secs). Now it is *this* which can use some improvement, and that improvement is planned : just use the 1,2,3,4 etc. on the Remote to order for 1,2,3,4 steps at a time. Of course at ordering for 4 steps it will be dealt with as 1 (leap of 4).
Will that help you ?

3)
If not solved otherwise, it should be solved "anyway". But notice I don't use any means of remote desktop etc. A normal (but selected MCE) remote will do fine.
Of course, your solution is more versatile, but it will trade that for SQ. Well, theoretically, and I don't even try. :)
But let's keep it to the "should be solved" for now.

4)
Ditto.

5)
This I can dedicate only to wrong useage.
a. you imply Attended Playback, which is not meant to be in the first place;
b. you should use "remote commands" just the same, never mind you have the option of a desktop in front of you.


Maybe I don't get exactly what you want here, but if, say, one resistor would be ok, there would be a fixed resistor volume pot. But even one resistor is not ok.
Of course I realize how difficult it is to grasp, but I gues you must have heard something from the DAC before you are "able" to say something like this.


Haha, how can you tell ? and btw, what is "huge" ?
I go for the best only, and you should too. Suppose I'd give you the volume control, with the prerequisite you are not allowed to listen to women voices anymore. All other is fine, but not that. This is not such a huge minor, is it ?
:swoon:
No, when we talk about minor or huge degradation, we must first have the reference. I have that of course, so again, it will not be easy at all for you to grasp. Not yet. :)


I saw the light allright. 600 euro for a fine and long lasting LDR volume control, including remote and display. This is for SE output only. Balanced will set you back for more than double.
I have it allright, but can just as well sell it to you (the SE version only). I never even tried it, because the price is ridiculous.

Now, if people could respond to the latter as a nice alternative, I can get a few more so it can go along with the DAC (as an option).
But personally I thought this would be adding way too much to the price of the DAC, and it ain't word sh*t when the digital volume is all so much better.
*If* that operates in a workable fashion ... TRUE !!

?

Peter

One thing to add (so you see, it is ALWAYS good to "complain" a little (or more :)) :

I must admit that I haven't investigated one situation, while just now suddenly I thought that *additionally* I can bypass it the most easy :

I expect a new, say, construction of the output stage in my mailbox tomorrow, and this is the very final one. But :

The way it is constructed allows me to parallal a 24 fixed (hence step) resistor (Vishay/Dale 0,1%) volume pot, with remote control. The kind of downside is that it "steps" and you will hear that at changing the volume (at low volume). By no way I can predict the result on SQ, because it is not in the signal path, but influences the gain of the output stage anyway. And, because it is not in the signal path, its "being there" can be cut by means of a switch, that switch not being in the signal path either. :scratching:
Chances even exist that it won't influence SQ at all.

If it works, I think I can apply it for somewhere between 150-200 euro. I must sort that out though.

Peter

For me there is no need to have a volume control capability in the DAC. Any money I need to spend more on the DAcCto have this would be a waste ....for me.
Cheers
Eric

Peter,

Thanks for the thoughtful responses. 

I do use normalized volume and, yes, it helps a lot.

Can you explain what you mean by using 1,2,3,4 on the Remote? Are you referring to an MCE remote?  Can you recommend one if this is a workable solution?  (I have to add that my experience with the reliability of XXHE’s volume control is not as good as yours.  Less than four seconds? Sometimes it takes a good ten.  Is it affected by Player Priority maybe?)

I’d be interested in the stepped Vishay attenuator you mentioned if it works well.  I’ve used this kind of volume on two conrad-johnson pre-amps I’ve owned. 

Dan

24 steps are way too few for me.
But if it works good for SQ, maybe Peter can offer a more fine attenuator.

Telstar,

I don't think more than 24 is possible with this concept, but also, it is not necessary. This is a bit in the area Dan talked about :

1. Set a "reference" digital volume (in XXHE);
2. Use the volume within that range;
but also
3. It will reach from maximum output up to some level I have to find out.

Ad 3.
Think something like it can attenuate 24dB, and the 24 steps are available for that range. So, at full attenuation from the pot you still only went 24dB down in effect. #1 above defines the maximum output, which is good for when there's too much gain, altough the pot itself can do that too, when next the digital volume is used (but that is not the subject here, I think).

Don't get too happy, because maybe it doesn't work.

Ok, I have worked this out a bit, and there is positive news as well as somewhat negative news.

The negative is that I now realize that this all operates in the differential (balanced) domain, so it takes 4 "channels" of this attenuation, meaning a price of near 400 euro. And no, "differential" means that I can't do only half when you'd want SE output only.

The positive news is that this is relais controlled (no clickclicklcick), and uses 64 steps, for normal operation meant to be 1dB steps. In our application here this means the dB steps will be far less (maybe 1/3dB, but I have to find out what the useable range is).

I just ordered one, but probably everybody will tell me that 400 euro really is too much for a stupid volume control.
Oh well ...
:)
Peter

When I said 24 steps are too few, i was thinking about full output and each step of 1db each which is too coarse for me.
But this way, 0,5db or less, and setting a reference volume in xxhe (maybe or maybe not for highres files), it would work for me, very well indeed.

I have a question, though: will it work with 8 channels? I can have all balanced outputs for what i care of that ^^
The most important thing is that all steps have the same impedance output seen by the amp (which IS different than without it).
400 euro is fine by me - any decent pre costs more. The savings on cables alone is very much worth it.

I'm really happy of this news.

I agree.  400 euros is a lot.  I'm just a 2-channel, balanced listener.  But if it works best...

First I'd like to try the MCE remote approach.  I’ve spent the last few hours looking for one, and all I’ve seen are not programmable but learn-able where you need another working remote to teach them the codes.  I need to be able to program them with Alt-N, Alt-U, Alt-D, etc.  Also, the ones I’ve found don’t come with a USB IR, which I need to plug into my PC. 

Does anyone have a good remote/IR receiver to work with XXHE?

Dan, look here for a Remote :  A Working Remote (http://www.phasure.com/index.php?topic=1340.0).

It will work with 8 channels allright, if you only multiply the costs with 4. :swoon:
A kind of crazy, right ?

The impedance isn't influenced a sinlge bit though ...

Ahiahiahi, now i have to think about it a little bit. It'll depend on the base cost of the 8ch version which is not known yet, right?

But tell me one thing, if there would be 4 pots, can you factory set to have a different default (i.e 0, -4db, 0, -8db) and then change all at the same time with the remote? This would cover the different gain of the amps in a totally transparent way (probably better than doing it in the digital domain like i do now).

So when will the 2 channel version of the NOS1 start shipping and what will the cost be?  Thanks.

Well, actually it can, and I thought about telling you this, but then I couldn't see a good solution in it. But, I only thought of the different levels for the XOver channels, and didn't see a good reason to change the individual level of them *after* the XOver filters have been defined. And so I never mentioned it. Your reason though is a most legit one. But factory set ? hmm ... how to do that ? of course, if you'd tell me what the difference in gain is, it could. But it would be the most dangerous, and fully dependend on your amps (and changes to them ... others later maybe ...). So, what I would propose in that case is to have an additional small trimpot at the output of one such a "stage" to control the individual attenuation, which would be as legit as the whole thing not being in the signal path, *if* it works in the first place.

But for the 8 channel analogue stages, which as you know are not ready yet, it would be better to make that single ended from the ground up, which avoids the differential attenuation setup, so the 1600 drops to 800 again. This still allows for others to choose for balanced, if they only know it has its price.
And to remember : having both SE and Balanced in one cabinet won't go anyway, because there's no space for both connectors. This, unless the relatively expensive jack types are applied, which take both SE and Balanced in one terminal.

Hmm ...

Changing the gain of a power amp is more difficult and often cause much worse sound. Doing it in the source would be just perfect.
I would rather consider a jumper (internal) than another pot, since i wouldnt change in years. Bu yes of course it wont be in the signal path since itìs set as default by another mean. For factory set I meant set by you according to my needs (that i dont know yet haha).


For me it doesnt matter if balanced or single ended, as long as SE sounds as good in the DAC part as balanced.

Peter

Sorry if this has been covered before. I did a quick search but couldn't find the answer.

Your XXHE player/8 Channel DAC will have adjustments in each channel for: a) level b) phase c) variable crossover points with independently variable slopes d) delay?

I seem to remember you saying there would not be any facility for minimal EQ adjustment (usually to slightly pull down any response peaks in the bass) - correct?

Thanks
Frank

Hi Frank,

a) No (I think :), b) Yes, c) Yes, d) No.

Ad a
See before posts addressed to Telstar. I judt didn't have in my mind to control that. But maybe it can't be avoided ... I mean, if I want my tweeter to nicely fit the mid, first the tweeter has to go down. Doing that by means of an analogue pot again would be a kind of wrongish, don't you (we) think ? But the point is, this makes all subject to "wrong" digital attennuation, and I could just as well make the current volume steps 0.1dB ...
Of course, it may turn out unavoidable that the whole band is influenced because of the crossover slope, and when that is so, better have the (most) fine digital attenuation. Once that is there, Telstar doesn't need his adjustments per channel anymore.

Ad d
Although in passive analogue some things can be done about this, it is not the way to do it ... I think. Better solve this in the physical driver alignment. But if you're talking about a subwoofer ... it should be in the subwoofer (overall phase).


About the EQ ... that is correct. But in the speaker filter (design) area it can't be avoided. But these should be parametric.


Hey, I guess you are somewhat ahead of me. :yes:
All very valid points Frank.

Well, there are cases in which physical alignment can cause aesthetic issues. I dont need it i think, but it would be easy (i think) to implement and very useful for many, probably lossless (you tell me).
About subwoofers, think about who DIY them and dont use very adjustable amps... i dont see many 2kw subwoofer amps and if there are, too expensive to buy rather than build (ok this is only my case which is rather an exception).


It's early to talk about it, but some drivers need some correction/EQ, yes parametric and with adjustable slope (Q). Not everyone can afford very expensive drivers with quite linear response, which are the MINORITY, anyway.
Mechanical equalization like is done with the driver construction or the speaker construction (horn size/shape etc), baffle size/shape etc. is THE best, second is digital correction IMO.

There was a very interesting opinion on a white paper from Harman international about speakers design and measurements and they differentiate the kind of adjustments needed because of the drivers/cabinet and because of the room. I know that you disagree on the second point, but the first... I do a notch filter in the digital domain, it requires higher attenuation than if i used a cap and a res... but i just didnt want to use them and the result has been defined "flat" (albeit is not in the HF, but nobody but me seems to hear them  :grazy:).

Peter

You are busy just getting the two channel DAC out, and that should be your priority.

Can I suggest that you continue to reconsider/rethink your configuration of the 8 channel DAC before finalising.

As I see it the most prevalent use of this 8-channel DAC will be by audiophiles wanting to run an active system. All of the features I asked about are what I would consider "basic" reqirements for such a system (Certainly in the Pro world, where it has been done successfully for ages - but not of course with the same resolution and fidelity as you are achieving).

Basic software requirements, usually available with an 8-Channel DAC would include:
1) Level adjustment per channel  
2) Phase adjustment  
3) Variable crossover points with independently variable slopes (i.e different slope high pass & low pass filters available at any crossover point)  
4) Simple delay for each channel [easy to achieve digitally and essential for good crossover design - physical alignment of drivers is not always possible or a design goal]    
5) Some EQ options.

Would building any of these software options into the DAC rather than XXHE make it easier?

Peter, please take some extra time to investigate before pushing yourself to get your 8-channel DAC to market.

Cheers
Frank

No answer the first time....so may I ask again when will the 2 channel version of the NOS1 start shipping and what will the cost be?  Thanks.

Yeah, I know. But I was waiting until the day was over, and I'd had some "results" on a stupid power inlet not doing what it promises (switching voltage between 115 and 230).

So, last Monday the cabinet should have gone into production (meaning the cabinet itself), but at assembling everything in the proto, this power inlet laughed at me within 2 minutes.
After fighting with and over this the whole week, a few hours back I finally received the message from the manufacturer that indeed it can't work.
So now I need another 100 of which I don't know where to get them at this moment, then they must be shipped from whereever it will be, and then at last I can try whether that works (but hey, why wouldn't it do what is promised eh ?), and finally I can let produce that cabinet.

Eerflappin, you are experienced in asking when the shipping date will be. I hope you are better than me in estimating it.
Btw, all other parts are here now, except for that volume trial where all apparently depends on one person which is only there next week.
Final price is 2900 excluding volume control *if* that works, which I can't try at the moment.

One thing I know, I am not ever going to design a DAC again, unless it's the same. It is all one big pile of hopelessly being dependend on suppliers which never do what they promise. I've been waiting for nearly 5 months for stupid capacitors. Come on now.

I have to say that I overall agree with you.
Some things are more important,some less but all can be useful and it's not difficult to do at software/drivers level.

All,

First a few general, boring things.

It has become clear that this "channel integration" thing really is THE thing to hunt for. When I wrote about this the first time I was only used to it two days I think, so that was extasy all over. But actually -at running more and more albums- it is an amazing thing you just haven't heard *anywhere*. Yes, I'm pretty sure about that.
This too excels with certain types of music, and since this is all about separation - think about King's Singers. I listened too Good Vibrations of them the other day, and really thought it was the most beautiful I EVER heard.

Further (but this is really the last one) improvement has been made on the sound in general, which partly came from the lot being in its cabinet now (I perceived somewhat more body from that). I must add to this that this time I used the analyser to squeeze down every bit of noise which is in there, and what comes from the DAC can not be measured anymore (at the end of the intterlink) from an FFT. IOW, this is beyond -144dB, as far as the analyser goes, which thus shows -144dB. And I must say, I never heard completely *no noise at all* from my horn speakers (which as you may know by now are 115dB sensitive, and with the amps at full gain of 27dB).
The larger deal tough came from the DAC and gain stage now being "integrated" by means of two "click connect" PCBs, while the proto of the gain stage was connected with soldered wires and two connectors. And yes, that all matters. This time it brought the snare drum on Six Blade Knife (Dire Straits) to more reality, while it was sounding like firm cardboard before (what one can hunt for :swoon:).

And this brings me to the volume control ...

Let me (again nag) first that this just took over 2 weeks of wasted time. Wasted time to the sense of that it has to go (hence fit) in the cabinet (of which I'm producing a 100 at first) or not. I mean, I can't produce the definite cabinet when I don't know what will be in there.
So as I told the other time, I immediately ordered one of which I thought it would be good (I recall that was on a Thursday) and heard nothing. The next Sunday though I received an email it would be shipped the next Tuesday. And so I received it last Wednesday. A week later than could have been.
The last two days I have been working on it, to finally come to the conclusion that it can't work not being in the signal path. This is about adjusting a resistance somewhere which should be on a tracelength of 3mm, while here I can't do it otherwise than over a length of 20cm. So, distortion all over, and it didn't even attenuate.

At wrapping all up yesterday, I thought to at least check what would happen if I would use it as intended. Thus, in the signal path as any normal attenuator;
I have been measuring the whole afternoon, which merely was the result from not seeing bad things at all. Oh, I saw a decrease of almost a factor two on harmonic distortion, but you must know that isn't all *that* bad, while we're at -105dB, and this factor two implies 6 dB (thus now -99).
The only thing I could find as strange was the slowly increasing noise level beyond the audio band, but since this is inaudible for both the frequency (25KHz and up) and the level (-130dB), I actually could not "see" where it would be wrong. Not in hours of time.

And so, at listening time the volume control was in the chain, and I ran a couple of tracks that I'm familiar with.
Now, with the explicit notice that all is relative, and that I'm used to what I'm used to (which is, say, a far too high standard), I could hear immediately that it didn't "work". The width of the soundstage seems compressed (yea, must find some new phenomena sometimes haha), bass doesn't go deep at all, but worse, is coloured, I sensed some standing wave behaviour in the low areas and for sure perceived standing (buzzing) waves in the higher regions. It was mushy. It didn't fit anymore. Piano notes shouted. Te music seemed too loud to be comfortable.

I imagined "you" listening to it, and I imagined that you still could say "wow", not knowing better. But, it would be critical. It would be critical to "the best DAC ever" and I would not dare to bet on it.

At this moment I can't decide;
*If* it would be in there, there would always be a direct path to the (different set of) outputs as well. But heck, why to do it in the first place ?

1. To comfortably attenuate the sound ?
Maybe, but in the very end non-sense. When talking about other players it is not much of a problem to begin with, and when talking about XXHighEnd a lot can be done to make it more comfortable. It can be made more fast (responding), it can be made more direct (like choosing the number of steps to in/decrease). But no, there won't be a physical knob. So what.

2. To attenuate with smaller steps
Yes, but this is very relative. This one works with 64 1dB steps, and using the knob isn't even comfortable, knowing that you need 10 full cycles or so of the knob to go from 64 to 0. I didn't try it with the remote, but imagine it takes ages to find "your" volume, because the steps can't be addressed directly, and it takes a "wait time" when the up/down key is continuesly pressed to start moving, and after that it will go too fast to stop where you want, or will go too slow to be comfortable.
Allright, where this is 1dB, XX works with 1.5dB. Well, I tell you that the annoying thing at the moment is those small steps when you want larger. It really really never occurred to me that I couldn't find "my" volume in those 1.5dB steps. So ... if you really think you need smaller steps (like Telstar) better think twice if this may be nagging, while in reality you are after the best sound quality. So, if I *tell* you this is nagging, maybe it helps ! haha

3. To protect from static
Is this only stupid theory ? I mean, when we play loud with our well matched amps, we play near full volume, and the static will be as bad as without the volume.
Of course this is different when we use 700W amps on speakers who need 50 only, so we only talk about that (and these people sure will exist, so why not take it as a legit reason).
For me it is clear that we just need an inherent solution for this. This is not a panic-mute button (you won't be able to press it from shock) but something which detects it and just cuts the signal. In theory this can be done in-DAC, but I think it will be too complicated to have a program running in there which can "see" this, like the Crack Detection in XXHighEnd can. The Crack Detection in XXHighEnd by itself works allright, weren't it that it is hooked in the chain too soon. I mean, after the Crack Detection process there's more which can mangle the data, so theoretically it can be done later. However, while it can be there at the latest at filling the buffers towards the OS and driver, there's always the driver at latest which can do things wrong. Not that we need to expect it will do things wrong (I never encountered that except for one report of a user with an explanation which goes in that direction), but we can imply it ourselves so much. Thus, when we squeeze down the latency so much that we aren't able to pass on the bytes in the proper sequence - meaning just skip one - we'd have it. On this matter, notice that an audio sample comprises of more bytes, and if one is left out you'd have static. This too I never encountered, which may imply that the OS (or driver ?) is very much responsible for keeping these things together, so if we really could rely on *that*, then I could guarantee it in the software.
But then this is only *my* software, while the DAC won't prevent you from using Foobar etc. ...

This brings me to the last subject for today : the inputs.
As it looks now, it won't be possible to use the available digital input (SPDIF) for measuring means. This never has been a subject much, but for me it is a kind of important. This is in the area of people "testing" the DAC (could be a review), while without such an input there is not much to test. The input is physically there allright, but it implies such a bad "stream" that this itself masks all the good stuff from the DAC. Most probably this is because the SPDIF input will route via the OS, or better : I can't get the SPDIF input passing through to i2s (which I use), unless I switch on the "Listen to this" in the device's properties. So, while the both are available on the same board (think like a sound card), the OS apparently is used for the routing, and nothing than harmonic distortion comes from that (like over 30%).
I didn't get around yet to testing the analogue inputs. But I need to add something to this :

I guess I made a thinking error on combining two "connection" situations where either at this moment won't allow analogue inputs to work (but the combination does while it can't exist as combination). The one situation needs a space in the cabinet which isn't there because the PCB with the clocks occupies it (and which I won't move farther away because it should be as close to the DAC as possible), while the other situation will be in the PC, but it will have a too jittery clock connection because the clocks are in the DAC cabinet. And please keep in mind : this "inputs" thing was about "your measuring" (so not really recording as such) which by itself needs the same clock for playback and the recording (of that playback).
Hmm ... something to sort out. But maybe I'll leave it be; It ever was a nice "withgoing" feauture, but if it's not "withgoing", then bye.
Anyone there who needs 400 Neutrik connectors ? you can also use them for outputs ! hahaha :cry:
Ok, I'll see what I can make of it.

First the Crack Detection I'd say.
Peter

Hi Peter,

in my opinion you should deliver the first 100pcs DAC as planned without volume control. GO for the best SQ as intended.
This was your intention until the "volume control" came up and you should not waggle.
As I see from your story, there is a lot of research involved,  and we would like to have the "World's first NOS 24/192 filterless DAC" 

NOW

without volume control and the best SQ available.

Enthusiastics may have a look to the" Weiss ATT202 Passive Attenuator".

Joachim

------------------------------------
Windows VISTA SP2> XXHE 09z-2 - Adaptive mode -  Q1/2/3/4/5 = 1/0/0/0/0 / No Invert / Playerprio = Low / ThreadPrio = RealTime / Scheme = 3 @ UnAttended / Arc Prediction * > Acer , Intel i750>|- Weiss Minerva -|- Dartzeel NHB-108B > speakers: Anat Referenz II Studio>

 :yahoo: :yahoo: :yahoo:

I wish I was living within driving distance from Peter too!

I know those Beatles tracks well. I have them in "all" versions, including 24bit/44k. They allways sounded grainy. I look forward to hear it the way you did! (I think TAS author Robert Harley was floored too, when he heard Get Back in the million dollar reference system of the owner of Magico in USA. Read about it in TAS a couple of years ago).
-What about the bass pipe in the end of Yellow Submarine. Did it really "shake the house"?

We only played it for two minutes or so, as the follow up on the remark of Gerard and Leon about those "separate" voices on Get Back. I had Yellow Submarine in mind as an even better example of that, so we played it briefly. I'll try it for myself later today (it's a bit eary now to shake the house (5:20 am) :)).

Btw Pedal ...


I let my wife listen to this one the morning before Geard and Leon arrived, just because I picked it as a funny first track to listen to and wanted to show it to her too. Her only remark (full amazement) : Huh ? I recall this is that grainy song ??

It is just the original from 196x and there is nothing grainy about it, as it is not with all the Beatles stuff. I think their repertoire is *the* example of how bad sounding old stuff isn't bad at all, but somehow our equipment can't deal with it very well. Actually this is with all material which seems over-emphasized in the highs, once you are able to squeeze out those highs. Later again, those highs appear to bring the refinement while ealier it looked the other way around. And deep deep down (I guess when the highs won't come out "yet") it's grainy ...

Peter

Gerard, Leon, what a "report" !

Let me first thank you both for your honesty, and the excessive time it must have taken you to detail this all out. Especially the emphasis on the "bad stuff" is good; Nothing less than I would like when it's about the software, and of course a necessity to improve (upon).

So to start with this "bad stuff" ... yes, I am the first to "state" that no such thing like a DAC which is "good in" certain types of music should exist, but that day -to my own surprise- it happened that it was not the case. And indeed, only a matter of the volume control being in there (though not active at all) is enough to "destroy". That this can (audibly) happen to one type of music (that we found) is a most peculiar thing, but just the case (as I know and knew already). Anyway, although done off line already, apologizes that it happened like that, which of course wasn't inentional and I didn't know it myself.

More interesting for you both maybe, is the again "learning" what one can listen for, without even knowing the music itself. In this case -and as you briefly implied yourself in your report- I could hear in general that something was wrong, just by not knowing what I was listening to. Thus, in this case this was about the impossibility to hear whether there were violins or synths in the background; two things happen here : 1. it should be distinguishable in the first place (ok, with this DAC it would) and 2. there shouldn't even be attention to it. The latter like I said in my quoted response (similar) "I can't even listen anymore to violins or synths because now it's about the voice". So, no matter how impossible it may sound to everybody, this is just a measure, once you are over some hurdles. :yes:


That "strange" design ... Wel, for others, here it is :

(http://www.stordiau.nl/images/img_4224a.jpg)

It could have been "square" (as large) just the same, that middle compartment being somewhere in there. But actually this is about 5 different compartments, them being there to avoid radiating influences. The DAC itself is in the middle.

At this moment there is a connection at the left back (top) side which just is a mistake (it now takes unnecessary "sticking out" space at the back, as there are connections at the right outside side of the upper left leg. I must honesty say that I thought these wouldn't harm, but since Gerard mentioned that it could be an ugly "out sticking" thing in a nice stereo tower, it should go to the inner side somewhere. Not sure that can be done though.

(http://www.stordiau.nl/images/img_4232a.jpg)

Btw, quite unbelieveable, but I have this cabinet for over 6 weeks now, and it still changes. :sorry: !
Oh, of course this in an unpainted proto.


It is not important at all, but I recall my own joy with the Diana Krall track, everybody being astouned about the power of her voice and seeing her literally singing with her head above the wing (piano), me not in the last place because I don't like her much. So why my joy ? Well, while everybody was happy with the result (I think this followed the bummer on the other two ladies) I *said* it was all nothing much, and nobody could understand. And, with a "wait, I'll let you hear some Ray Brown" and the following question "now, what was wrong with Krall there ?" the answer was unanimous : the wing. This time, however, it was about the poor quality (and for me a real disturbance) how indeed badly recorded the wing was, if compared with a better one. The essence though was (for all of us I think), that this is so clearly audible, and a virtue with that.
But as said, not important at all, and I just had a good time with it, so much saying in advance I don't like Krall, the next evening immediately playing full records of her; I said it before, I said it when you were there, and I keep on saying it : once things really work out, there isn't much music not to like. May it be Krall, Beatles and even house (what about the details Infected Mushroom spreads out. Oops).


Hmm ... This didn't quite come forward during the "session", but I feel this must be related to your remark afterwards "we should have been prepared better". Am I right ? Right or not, you both said that, so how would you prepare a next time ? Just curious ...


Thank you guys. I can't be sure whether you mean "we at our homes with a new DAC" or "the future of music";
I won't force you to say the latter, but it sure is what *I* am after, knowing it is just possible to outrageously outperform anything what exists at the moment. But it is so fragile ... I mean, no matter a cymbal at last sounds like a cymbal, there's also this strange dimension of "it has to work" which almost seems a contradiction. So, let's admit it : without any cymbal, and even with grainy stuff, music can still work, and may still draw tears. Also, I think we'll all agree that the latter may be even more important than the more technical thing like a cymbal sounding like a cymbal (no matter how very hard that is by itself). Both have to work, and the "working" thing is hard to control. Also hard to notice. That's why I always say that it takes 5 days to judge a change; it gives the opportunity to run into anomalies, may it be a cymbal not working with hiphop somehow, or an Emma Shapplin suddenly eating herself. So ok ... it's taking me long enough to make it 100% (to my own ideas), and while I know what that is, I still have to learn to change exactly *nothing* when an audition is coming up, no matter I think it will be unrelated. So that won't ever happen again.

One more thing to add, just because I feel it's importance :
Telstar too experienced the same "doesn't work" phenomenon. Of course this was 16 months ago, and with a version of the DAC which was totally "nothing" compared to what it is now. So, then too, this was about the "interest" of the technical thing (not heard in other DACs I know of) against the "does it daw tears" which is just in another leage, and btw also unrelated to foot tapping;
I guess it will be THE challenge for me, having Telstar over again in due time, with the advice of being prepared at saving some tears for this visit. In other words, at this moment I will leave it to him whether that now works out in full, knowing that not everybody has as many tears spare for a thing like this. So, Lisa Gerard and Emma Shapplin just were out of key because some technical failure, which is really in a different leage again from something which sounds normal to me, but would let another man cry ... or not because it just doesn't happen.
Telstar, you're welcome. And needed I guess. :)

Peter

(http://www.stordiau.nl/images/img_4230a.jpg)

(http://www.stordiau.nl/images/img_4233a.jpg)

I'm sorting out some issues at work here, but I'm coming soon for my final audition (you have a pm).
Then i'll quiet myself all the time needed for the 8ch version to be ready.

Hey Gerard, thanks for the write up of your visit with Leon to Peter's place. I really appreciate the time you've taken to do this.

Mani.

First Gerard and Leon, nice post you guys.

Peter,

Little bit strange casing, but the more you think about it, its a good design.
Shielding the various sections, proper room for interlinks (kinda like that).

Made in Holland

Roy

Yeah, nothing's perfect... right?

Mani.

 :thankyou: It's been a pleasure for us. And than to think of  it's is only a 45 min drive to the place where it all happends!  :yahoo:

Lucky us!  :)

Practically ALL multitrack rock recordings from the 60s sounds somewhat grainy. That is, compared with "modern recordings" at least. The recording equipment wasnt much good. I've checked the technical data on those recorders and mixers, and it wasn't even "Hi-Fi" (DIN 45.500)... Also, 50 years old magnetic master tapes are starting to wear out.
Interestingly, jazz and blues (acoustic) recordings sounds better, even those from the late 50's. It's the minimalistic aproach of course. Think Rudy Van Gelder etc.

Lately I have purchaed a lot of hi-rez albums in 24/96 and 24/192. Some of them are transfers of old music. It's very fascinating listening, because I easily get lost into the music, enjoying the very good SQ. It's only AFTER, when switching to a modern recording that you realize all the wow and flutter, distortion and limited bandwidth. Funnily, going back to those old recordings after a few days, my hearing is reset and I enjoy them just as musch as I did the first time. Aural perception is certainly a mystery!

Sorry for the OT. To get back on track I would only add that when those old Beatles recording are decoded by the XX software/hardware brobably 2-3 coats of haze is removed and the squarewave impulses are restored. So they take on another level. As I said; I cant wait to hear it for myself!

BTW: All the Beatles records were done with the same old studio equipment (tube mixer and old tape recorders), except Abbey Road which had new "modern" solid state mixers and recorders. White album SQ stands out above the others, maybe because it is more acoustic, less edited. But Abbey Road is clearly the very best soundwise. It is also the only one which was mixed in stereo originally. On Abbey Road I feel that the grainyness is not present. It sounds quite transparent and clear, with deep bass and very good definition all over. The Beatles was "first" with doing just about everything in rock. The Abbey Road album might be characterized as the first audiophile rock recording, because they were the very first to use this brand new equipment inside Abbey Road Studios!

PS: After spending some many years developing your DAC I think you are entitled to shape it as you like. The H shape might be a good marketing gimic, too. Altough I would personally prefer the letter "P". Like in Pet(t)er...
;-)

EDIT: It was Come Together, the opening track on Abbey Road. Here is a link to the TAS article (http://www.magico.net/support/MAGICO_Ultimate.pdf). Worth a read.
Come Together is also featured on the recent LOVE soundtrack, available as DVD-A remastered in 24/96. I'll try it out tonight!

Peter

It looks like I will have take delivery of the "bare-bones" Phasure NOS1 DAC, rather than waiting for the 8-channel version (with extras), if I want the purest sound. Can't wait to hear what your local 'first-users' report back, when they take delivery (soon I hope).

Can you please remind me what physical PC output connection you are using to go from PC to 12s latency (as per your signature)?

Thanks

Frank

I'm guessing RJ45...

Of course this is okay. But what I didn't tell yet (because still not 100% ready with it) is that I am now theoretically able to have a volume control in, which doesn't degrade sound for a bit. So :
I have had that working, while not any measureably degradation was there, as well as that it wasn't audible. However, this was with two mono-pots and a 0,0001ct resistor somewhere. So, I found exactly *one* stereo pot on the globe which could do the job (apart from two others, one mil spec and one med spec of over 300 euro), and together with it the cheap resistor was replaced by a 11 euro one (per channel). And now it doesn't work anymore. :aggressive:
Thus, theoretically solveable, but practically still not. :cry:

This VC is, by now, of more importance than I ever thought of myself. So watch this, as a small explanation of a rather huge thinking error :

Long ago, I created the "lossless" digital volume in XXHighEnd; It anticipates on the headroom of 8 additional bits at playing 16/44.1 onto the 24/xx DAC. That works. Good.
Later, I developed Arc Prediction, and that too uses that available headroom. So, imagine that 2-step wave at 20KHz (that being a theoretical square (practical sawtooth) while a sine is intended), which is turned into an 8 times better "shape" which nicely looks like a sine again. This *needs* those bits ...

Do you see it coming ? yes, you do;
At using the digital attenuation the next thing which happens is that I "throw out" those bloodily created additional bits at Arc Prediction, with the result of AP not doing what it intended. That is, when I have to attenuate digitally 24dB (which is very normal for my modest 27dB gaining amps), I've thrown out 4 bits again ...

:swoon:

So, both the digital volume (as implemented within XXHighEnd) as AP are as good as they are, when each of them is used individually. But not both at the same time.

While this is as logic as can be, at least I never thought of this, but came to it automatically at observing that the VC which measures as good as without it, sounded BETTER. There's just more resolution, if only the digital volume isn't used (as much).

The stupid thing is, at measuring one wouldn't notice this, because this always happens at -0dBFS (no digital attenuation) because otherwise the noise floor comes closer and the figures get worse because only that (the same counts for measuring with a fair amount of analogue attenuation).

And so, in the far end we don't want to use a preamp because it degrades, we want Arc Prediction on 16/44.1 material because it improves so much, and we *have* to attenuate somehow ...
And this is already because our amps have different gain.

I won't say that the whole thing backfired on me (or us if you want), but it could be better for SQ again ... :yes:
Now it *needs* the VC, just as a means of good calibration and to "keep in" those 24 bits. That this implies that digital attenuation is no good after all is another matter, but no problem if the analogue attenuation is just there anyway.

Well, you could say that just because of giving myself another additional month (if not two), it again improved;
Now I have to turn the theoretically working situation into one which really works (forget about remote control please), and all should be done. It's all quite a lengthy job, because there's a nest of wireing needed here, and each other attempt needs to resolder it all. When I'm done I'll let you know. :)


Frank, that shouldn't read "12 seconds of latancy" or whatever you made of it, but i2s (i-square-s) which is the protocol used. You'll need nothing, because what's needed will be in the box. I'm trying to keep this in the blue as long as I can. :sorry:

Peter