World's first NOS 24/384 filterless DAC

[PeterSt]

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


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.
36 hours ago the champagne was opened because it produced sound for the first time.

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 );
- Name I dedicated to this combination : NOS1.

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.
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.

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 ...
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 ...

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 ).
  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 ...
  
  
  

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.

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.

[glynnw]

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

[pedal]

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?

[SeVeReD]

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

[PeterSt]

Very interesting!
[...]
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?

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.

[PeterSt]

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

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.

[PeterSt]

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 ...

[xp9433]

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

[SeVeReD]

[/li][li]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 !

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.

[PeterSt]

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 , otherwise you don't get access to all the stuff anyway.

[pedal]

I have to keep up that I am going to produce XXX DACs already , otherwise you don't get access to all the stuff anyway.

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?

[PeterSt]

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 ...

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?

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 ?

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.

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

[pedal]

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?

[LydMekk]

Pedal: LMC ?

[PeterSt]

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

Neither. But it is passive anyway.

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 ?

[Ava12]

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

[Bull-X]

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. 

[PeterSt]

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

[SeVeReD]

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. 

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.

[PeterSt]

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 ...

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

[Bull-X]

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. 

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.

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.

[Bull-X]

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 ...

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

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.

[PeterSt]

... 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

[Bull-X]

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.

[Telstar]

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?

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

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.

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

I believe this SRC converters are "not good".

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

[PeterSt]

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

[PeterSt]

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

[leifchristensen]

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.

[PeterSt]

if you e.g. had minimonitors ,that would be a useless statement to me

Of course not, that would be extraordinaire !

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 !).

and me going to Florida the 19th , I won´t build it until early jan.

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

[leifchristensen]

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

[PeterSt]

or else we´ll have to wait for the Buffalo firewire I2S bus

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. :

- S/PDIF connected, and due : directly Firewire connected without soundcard into I2S;

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 ...

[leifchristensen]

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 
so long buddy
best
Leif "forward" Christensen
Norway

[Telstar]

directly Firewire connected without soundcard into I2S;

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 ...

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

ps: signature updated

[leifchristensen]

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

[PeterSt]

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 !

[leifchristensen]

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
maybe sound tomorrow? tonight is dinner with guest and redwine
L8R
best
Leif
Norway

[PeterSt]

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

Careful with the wine now !

[leifchristensen]

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

[Telstar]

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?

[leifchristensen]

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

[PeterSt]

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 ... ).

[PeterSt]

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 ... ...

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

[leifchristensen]

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

[leifchristensen]

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

[PeterSt]

Leif,

Don´t you lack a common star ground in your drawing? to chassis even?

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 ...

why are we obsessed with firewire to i2s ?

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 ).

the beauty of the usb std is the "autorecognition" feature once you hook up a new device.has firewire this as well?

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.

[...] psu´s  to BUFF and Ivy are not [...]

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 !

[SeVeReD]

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

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.

[leifchristensen]

IT PLAYS!
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!
best
Leif

[PeterSt]

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

[leifchristensen]

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

[PeterSt]

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

[leifchristensen]

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

[leifchristensen]

#¤#"/&&%&%/  it sounds way too lightweight! no impact!??
hey but wait ,my wifie has stumbled in the powercables to the right basstower
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
whow this is hot
anyone want a ticket and packet holiday with complete family to Florida?
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!
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

[PeterSt]

anyone want a ticket and packet holiday with complete family to Florida?

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

[leifchristensen]

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

[ed linssen]

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

[PeterSt]

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 ...

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

PS:

Don´t you lack a common star ground in your drawing? to chassis even?

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

[ed linssen]

Hi Ed,

Things worked out differently from what I planned.


Morning, Peter,

I will keep my attention on your progressings.
I must say, I am really satisfied with my own DAC (so far!). I have built, about two years ago, the Doede Douma Dac DDDac version two. The same one  Leifcristensen has. Mine is USB and battery-powered.
The reason I am looking around is, I can only play-back 44/16 files.
Since I am a llifetime-liverecording enthausiast and and do my recordings (orchestral and choir) ,for the last eight years or so on hard-disk 96/32, I need a really good way of playing back those files as well without the need rebuilding them to 44/16, as you will understand.
The things I am using now for that matter are the prodigy HD2 soundcard(intern), or the M-audio USB(extern).
I am looking for a better solution!
But......since it's all DIY and hobbying, for the sake of good musical audio reproduction, low cost investments are always a preference!

Enjoy your music,

Ed

[leifchristensen]

Colony Hotel Miami Beach calling
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

[PeterSt]

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

[PeterSt]

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.

[leifchristensen]

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

[PeterSt]

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 ?

[leifchristensen]

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

[PeterSt]

if it sounds good , it probably is

But he copied Duke Ellington I think.

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

[PeterSt]

... 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.
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.

[SeVeReD]

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.

[PeterSt]

I don't think I have a 17 khz tone

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 ...

[Telstar]

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

[Gerner]

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

Me too Telstar...

Gerner

[PeterSt]

I will keep my attention on your progressings.
I must say, I am really satisfied with my own DAC (so far!). I have built, about two years ago, the Doede Douma Dac DDDac version two. The same one  Leifcristensen has. Mine is USB and battery-powered.
The reason I am looking around is, I can only play-back 44/16 files.
Since I am a llifetime-liverecording enthausiast and and do my recordings (orchestral and choir) ,for the last eight years or so on hard-disk 96/32, I need a really good way of playing back those files as well without the need rebuilding them to 44/16, as you will understand.
The things I am using now for that matter are the prodigy HD2 soundcard(intern), or the M-audio USB(extern).
I am looking for a better solution!
But......since it's all DIY and hobbying, for the sake of good musical audio reproduction, low cost investments are always a preference!

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.

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

[GerardA]

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!!!

[PeterSt]

And did you put a wire from the case to the real earth?

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 ...

[GerardA]

it would be the first device connected to mains ground anyway ...

Don't make your starearth like this:

[PeterSt]

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 ...

Peter

[leifchristensen]

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

[pedal]

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.

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

[PeterSt]

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

[Gerard]

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

 

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

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

[PeterSt]

The DAC board wasn't explicitly designed for NOS mode, and although everything looked good so far, this is where the tweaking stops.

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 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.
Peter

[pedal]

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?

[PeterSt]

I am very interested in re-trying a NOS DAC which is based on a modern DAC-chip like the Buffalo

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

Can you make XX work on this 176,4 samplingfrequency?

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 ? )

[pedal]

I am very interested in re-trying a NOS DAC which is based on a modern DAC-chip like the Buffalo

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

Can you make XX work on this 176,4 samplingfrequency?

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 ? )

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?

[PeterSt]

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". 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

[pedal]

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

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

[SeVeReD]

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

[leifchristensen]

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

[PeterSt]

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 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

[leifchristensen]

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

[GerardA]

Besides this, like many I have my doubts about battery power;

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!

[LydMekk]

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.

[PeterSt]

Maybe its me, but I LIKE the sweetness you mention.

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

[PeterSt]

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

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.

[leifchristensen]

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

[Telstar]

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.

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

[PeterSt]

I knew that you sometime

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.

[PeterSt]

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

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

[SeVeReD]

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.

[Ava12]

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.

[leifchristensen]

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

[Telstar]

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

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

Yep

and i'm waiting for that email ))

[PeterSt]

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.
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

[Telstar]

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

[pedal]

...I did not know the visitors besides some dozens of emails, and about the seriousness ... they came by airplane.

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

...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...the recordings were the best existing, but on the Sabre violins easily became a flute. Yes, I'm serious. A flute.

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...), 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".

[Telstar]

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.

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.

[PeterSt]

Hi pedal,

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!

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

[LydMekk]

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.

[PeterSt]

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.

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.

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 ?) 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.

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.

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 :

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

... which makes all I just said unnecessary.  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 ...

[LydMekk]

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...

[SeVeReD]

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.

[Telstar]

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.

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.

[pedal]

If you want EQ and/or roomcorrection there is one amazing program to buy: AUDIOLENSE XO. 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?

[Telstar]

If you want EQ and/or roomcorrection there is one amazing program to buy: AUDIOLENSE XO. 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.

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.

[leifchristensen]

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

[pedal]

Why should I buy Audiolense (which is good btw), when DRC is free?

Why did you pay for XXHighEnd when Foobar is free?  
-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!)

BTW can audiolense create filters for each channel separately?

Yes. For as many channels you like.

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

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?

[Telstar]

Why should I buy Audiolense (which is good btw), when DRC is free?

Why did you pay for XXHighEnd when Foobar is free?  

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.

-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!)

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.

Hopefully XXhighend will have a possibility of plug-ins too in the near future?

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

[PeterSt]

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

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

[PeterSt]

Hopefully XXhighend will have a possibility of plug-ins too in the near future?

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

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

[PeterSt]

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

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

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.

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)

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.

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

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.
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.
Peter

[leifchristensen]

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

[PeterSt]

20 Km closer from the Kiel direction ...

[leifchristensen]

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

[Telstar]

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

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.

[PeterSt]

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

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.

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?

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 ).

[AUDIODIDAKT]

Silence Before The Storm,

Must be a Hell of a Post coming up!
 

Does it work for you ? Peter

[PeterSt]

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


Later ...

[Telstar]

I have been typing for quite an hour tonight allright. But I could not finish it.

and???

[Ava12]

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

[PeterSt]

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 ...

[Gerard]

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 ...

Hope it's positief.... 

[Ava12]

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^^

[PeterSt]

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 :

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 :

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 :

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 ().
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)

[Ava12]

Ok first reply this night, more tomorrow.

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

  still have to this over .

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

[manisandher]

Thanks Peter. This is facinating stuff!

... filtering options will be by means of (player) software, and uploadable to the DAC in a later stage.

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.

[manisandher]

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.

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

... where the [delta-sigma] 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).

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.

[Ava12]

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

I'll be happily waiting for more news.

Ava

[Telstar]

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

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]

[manisandher]

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.

[Telstar]

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.

[manisandher]

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 

Mani.

[PeterSt]

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.

[PeterSt]

... filtering options will be by means of (player) software, and uploadable to the DAC in a later stage.

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?

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.
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

[PeterSt]

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.

... where the [delta-sigma] 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).

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?)

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 ).

Peter

[PeterSt]

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.

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 ...

[manisandher]

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.

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.

[PeterSt]

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).

[Josef]

>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?

[PeterSt]

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 ...

[Telstar]

>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?

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).

[leifchristensen]

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

[PeterSt]

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

[PeterSt]

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

And then measuring and fine-tuning.

(Hans Theessink seemed a good one to start with)

[PeterSt]

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

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

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


FFT 50 Hz

50Hz, and still looking good.

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

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.


FFT 200 Hz

Everything still over 98dB down.


FFT 1000 Hz

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


FFT 2000 Hz

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

Not much to say about this.


FFT 8000 Hz

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


FFT 10000 Hz

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

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

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

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

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

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.

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 ...


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

[Gerard]

Man you really should get a reward for this!! 

[Telstar]

[AUDIODIDAKT]

Peter I'm totally speechles,

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"

   

[AUDIODIDAKT]

Q about volume control:

No digital volume control possible during playback?

just found out is doesn't. so

[Telstar]

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.

[Ava12]

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

[PeterSt]

Sure ...

[Per]

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... ;-)

[Telstar]

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@.

[PeterSt]

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).

(0dB should be even better measurewise - but I am not an engineer so my understanding is pure layman on this)

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

[PeterSt]

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

[PeterSt]

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 :

• 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.
  
  
• 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 :
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,